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[alsa-devel] [PATCH 1/2] ASoC: tegra: improve goto error label
by Marcel Ziswiler 26 Jul '18

26 Jul '18

From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> While the two error labels "err" and "err_clk_put" goto the same place it is rather confusing that the earlier one is certainly used later again. Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> --- sound/soc/tegra/tegra20_ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index affad46bf188..682ef33afb5f 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -377,7 +377,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ret = clk_prepare_enable(ac97->clk_ac97); if (ret) { dev_err(&pdev->dev, "clk_enable failed: %d\n", ret); - goto err; + goto err_clk_put; } ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops); -- 2.14.4

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[alsa-devel] [PATCH] Added support for Encore mDSD USB DAC
by Jeff Crukley 26 Jul '18

26 Jul '18

From: JCruk <jcrukley(a)gmail.com> This patch adds native DSD playback support for the Encore mDSD USB DAC by specifying the vendor and product ID's Signed-off-by: Jeff Crukley <jcrukley(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc0..6fbafce 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1373,6 +1373,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ -- 2.7.4

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[alsa-devel] [PATCH] ASoC: AMD: Fix build warning
by Akshu Agrawal 26 Jul '18

26 Jul '18

Fixes sound/soc/amd/acp-da7219-max98357a.c: In function 'cz_probe': sound/soc/amd/acp-da7219-max98357a.c:367:3: warning: 'ret' may be used uninitialized in this function [-Wmaybe-uninitialized] dev_err(&pdev->dev, "Failed to register regulator: %d\n", ret); Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index cd3cf6e..48c1d94 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -365,7 +365,7 @@ static int cz_probe(struct platform_device *pdev) &acp_da7219_cfg); if (IS_ERR(rdev)) { dev_err(&pdev->dev, "Failed to register regulator: %d\n", - ret); + (int)PTR_ERR(rdev); return -EINVAL; } -- 1.9.1

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[alsa-devel] [PATCH 00/14] ALSA: Proper endian notations
by Takashi Iwai 25 Jul '18

25 Jul '18

Hi, this is a patchset for correcting the integer types to the endian-wise ones, as caught by sparse. Just cosmetic fixes for catching future bugs, no functional change by itself. Takashi === Takashi Iwai (14): ALSA: ymfpci: Proper endian notations ALSA: trident: Proper endian notations ALSA: hda: Proper endian notations for BDL pointers ALSA: riptide: Properly endian notations ALSA: mixart: Proper endian notations ALSA: lola: Proper endian notations ALSA: intel8x0: Proper endian notations ALSA: intel8x0m: Proper endian notations ALSA: maestro3: Proper endian notations ALSA: echoaudio: Proper endian notations ALSA: bt87x: Proper endian notations ALSA: atiixp: Proper endian notations ALSA: atiixp_modem: Proper endian notations ALSA: sb: Proper endian notations sound/hda/hdac_stream.c | 4 +- sound/isa/sb/sb16_csp.c | 30 +++++------ sound/pci/atiixp.c | 4 +- sound/pci/atiixp_modem.c | 4 +- sound/pci/bt87x.c | 4 +- sound/pci/echoaudio/echoaudio.h | 2 +- sound/pci/echoaudio/echoaudio_3g.c | 14 +++--- sound/pci/echoaudio/echoaudio_dsp.c | 6 +-- sound/pci/echoaudio/echoaudio_dsp.h | 50 +++++++++--------- sound/pci/echoaudio/echoaudio_gml.c | 8 +-- sound/pci/intel8x0.c | 6 +-- sound/pci/intel8x0m.c | 6 +-- sound/pci/lola/lola.c | 4 +- sound/pci/lola/lola.h | 4 +- sound/pci/lola/lola_pcm.c | 8 +-- sound/pci/maestro3.c | 6 +-- sound/pci/mixart/mixart_core.c | 4 +- sound/pci/mixart/mixart_hwdep.c | 42 ++++++++-------- sound/pci/mixart/mixart_hwdep.h | 8 +-- sound/pci/riptide/riptide.c | 8 +-- sound/pci/trident/trident.h | 2 +- sound/pci/trident/trident_main.c | 2 +- sound/pci/ymfpci/ymfpci.h | 78 ++++++++++++++--------------- sound/pci/ymfpci/ymfpci_main.c | 6 +-- 24 files changed, 157 insertions(+), 153 deletions(-) -- 2.18.0

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[alsa-devel] [PATCH 00/10] ALSA: yet more PCM format type conversion fixes
by Takashi Iwai 25 Jul '18

25 Jul '18

Hi, this is a series of similar fixes as the previous patchset for ASoC, covering other ALSA drivers to deal with PCM format type properly. Takashi === Takashi Iwai (10): ALSA: hda: Fix implicit PCM format type conversion ALSA: riptide: Fix PCM format type conversion ALSA: pcm: Fix sparse warning wrt PCM format type ALSA: ad1816a: Fix sparse warning wrt PCM format type ALSA: au88x0: Fix sparse warning wrt PCM format type ALSA: asihpi: Fix PCM format notations ALSA: wss: Fix sparse warning wrt PCM format type ALSA: sb: Fix PCM format bit calculation ALSA: sb: Fix sparse warning wrt PCM format type ALSA: xen: Use standard pcm_format_to_bits() for ALSA format bits include/sound/hdaudio.h | 3 ++- include/sound/sb16_csp.h | 2 +- sound/core/oss/pcm_oss.c | 2 +- sound/hda/hdac_device.c | 2 +- sound/isa/ad1816a/ad1816a_lib.c | 3 ++- sound/isa/sb/sb16_csp.c | 6 +++--- sound/isa/sb/sb16_main.c | 9 ++++++--- sound/isa/wss/wss_lib.c | 2 +- sound/pci/asihpi/asihpi.c | 24 +++++++++++++----------- sound/pci/au88x0/au88x0.h | 2 +- sound/pci/au88x0/au88x0_core.c | 2 +- sound/pci/riptide/riptide.c | 2 +- sound/xen/xen_snd_front_alsa.c | 4 ++-- 13 files changed, 35 insertions(+), 28 deletions(-) -- 2.18.0

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[alsa-devel] [PATCH 00/12] Fixes for sparse warnings and other misc bugs
by Takashi Iwai 25 Jul '18

25 Jul '18

Hi, this is a series of patches to correct bugs spotted by sparse and manual lookups, applied in a wide range of drivers. Takashi === Takashi Iwai (12): ALSA: usb-audio: Fix multiple definitions in AU0828_DEVICE() macro ALSA: msnd: Add missing __iomem annotations ALSA: msnd: Fix the default sample sizes ALSA: hda/ca0132 - Use NULL instead of 0 ALSA: msnd: Use NULL instead of 0 ALSA: hda - Fix a sparse warning about snd_ctl_elem_iface_t ALSA: opl3: Declare common variables properly ALSA: usb-audio: Declare the common variable in header file ALSA: pcm: Add __force to cast in snd_pcm_lib_read/write() ALSA: pcm: Use standard lower_32_bits() and upper_32_bits() ALSA: korg1212: Add __force annotation to cast in user-copy callbacks ALSA: emu10k1: Fix missing __force annotation for user/kernel pointer cast include/sound/pcm.h | 4 ++-- sound/core/oss/pcm_plugin.c | 8 ++++---- sound/drivers/opl3/opl3_drums.c | 2 -- sound/drivers/opl3/opl3_lib.c | 3 +-- sound/drivers/opl3/opl3_midi.c | 4 ---- sound/drivers/opl3/opl3_oss.c | 2 -- sound/drivers/opl3/opl3_synth.c | 1 + sound/drivers/opl3/opl3_voice.h | 4 ++++ sound/isa/msnd/msnd.c | 18 +++++++++--------- sound/isa/msnd/msnd.h | 2 +- sound/isa/msnd/msnd_midi.c | 2 +- sound/isa/msnd/msnd_pinnacle.c | 8 ++++---- sound/pci/emu10k1/emufx.c | 24 ++++++++++++------------ sound/pci/hda/hda_codec.c | 4 ++-- sound/pci/hda/patch_ca0132.c | 3 +-- sound/pci/korg1212/korg1212.c | 4 ++-- sound/usb/mixer.h | 2 ++ sound/usb/mixer_quirks.c | 2 -- sound/usb/quirks-table.h | 3 ++- 19 files changed, 48 insertions(+), 52 deletions(-) -- 2.18.0

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[alsa-devel] [PATCH 1/3] ALSA: vx222: Fix invalid endian conversions
by Takashi Iwai 25 Jul '18

25 Jul '18

The endian conversions used in vx2_dma_read() and vx2_dma_write() are superfluous and even wrong on big-endian machines, as inl() and outl() already do conversions. Kill them. Spotted by sparse, a warning like: sound/pci/vx222/vx222_ops.c:278:30: warning: incorrect type in argument 1 (different base types) Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/vx222/vx222_ops.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index d4298af6d3ee..c0d0bf44f365 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -275,7 +275,7 @@ static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, length >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ for (; length > 0; length--) { - outl(cpu_to_le32(*addr), port); + outl(*addr, port); addr++; } addr = (u32 *)runtime->dma_area; @@ -285,7 +285,7 @@ static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, count >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ for (; count > 0; count--) { - outl(cpu_to_le32(*addr), port); + outl(*addr, port); addr++; } @@ -313,7 +313,7 @@ static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, length >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ for (; length > 0; length--) - *addr++ = le32_to_cpu(inl(port)); + *addr++ = inl(port); addr = (u32 *)runtime->dma_area; pipe->hw_ptr = 0; } @@ -321,7 +321,7 @@ static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, count >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ for (; count > 0; count--) - *addr++ = le32_to_cpu(inl(port)); + *addr++ = inl(port); vx2_release_pseudo_dma(chip); } -- 2.18.0

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[alsa-devel] [PATCH] ALSA: seq: Fix poll() error return
by Takashi Iwai 25 Jul '18

25 Jul '18

The sanity checks in ALSA sequencer and OSS sequencer emulation codes return falsely -ENXIO from poll callback. They should be EPOLLERR instead. This was caught thanks to the recent change to the return value. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/oss/seq_oss.c | 2 +- sound/core/seq/seq_clientmgr.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index 5f64d0d88320..e1f44fc86885 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -203,7 +203,7 @@ odev_poll(struct file *file, poll_table * wait) struct seq_oss_devinfo *dp; dp = file->private_data; if (snd_BUG_ON(!dp)) - return -ENXIO; + return EPOLLERR; return snd_seq_oss_poll(dp, file, wait); } diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 56ca78423040..6fd4b074b206 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1101,7 +1101,7 @@ static __poll_t snd_seq_poll(struct file *file, poll_table * wait) /* check client structures are in place */ if (snd_BUG_ON(!client)) - return -ENXIO; + return EPOLLERR; if ((snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_INPUT) && client->data.user.fifo) { -- 2.18.0

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[alsa-devel] Need some help building alsa-utils
by Jamey Kirby 25 Jul '18

25 Jul '18

I have been working with alsa for a while. We use it to archive content from our radio station. I am using arecord. Build problem: 1) I cloned alsa-utils from github 2) Spent two hours on Google trying to figure out how to build the darn thing. 3) I am running Ubuntu Server 18.0 with build-essentials installed. 4) I get errors that I do not know how to fix when I try to setup ./configure. 5) I just need to make a build. What do I need to do? Feature: I am adding a feature. In the loop for capture(), when a new file is opened, I want to close the old file (when doing continuous recording). I added a new CLI parameter -K. When this is set, the capture() loop will close the old file before opening the new file. This is an issue for me for two reasons: 1) When will i run out of file handles? I capture one hour files continuously. After 10 hours, I see that all the capture files are still open? 2) I need to be notified when the file is closed so that I can do post-processing of rht capture. I've tried inotifywait in a script, and in another script I tried lsof. lsof works great, but the files are open,. See, the main issue is that I need to know when each file completes so that I can post process. Here is my post process script outline: #!/bin/bash for f in $HOME/vboxshare/jkirby/newarc/*.raw do filename=$(basename $f) open=$(lsof -t $f | wc -w) if [ $open -lt 1 ] then echo "$filename is closed" echo "Do post processing here" else echo "$filename is open by $open processe(s)" fi done If I could close the file before or after the previous file is closed, I can do my work. Otherwise, I have no way to know if the file is open and being processed or not. So, I've made the changes, I just need to build the darn thing. Thanks for any advice or help. -- Jamey Kirby Disrupting the establishment since 1964 This is a personal email account and as such, emails are not subject to archiving. Nothing else really matters.

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[alsa-devel] [asoc:topic/amd 1/1] sound/soc/amd/acp-da7219-max98357a.c:325:3: warning: 'ret' may be used uninitialized in this function
by kbuild test robot 25 Jul '18

25 Jul '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/amd head: 7b5317aa809f833d017196ad4a34e3b3b7dae101 commit: 7b5317aa809f833d017196ad4a34e3b3b7dae101 [1/1] ASoC: AMD: Add a fix voltage regulator for DA7219 and ADAU7002 config: x86_64-randconfig-x002-201829 (attached as .config) compiler: gcc-7 (Debian 7.3.0-16) 7.3.0 reproduce: git checkout 7b5317aa809f833d017196ad4a34e3b3b7dae101 # save the attached .config to linux build tree make ARCH=x86_64 Note: it may well be a FALSE warning. FWIW you are at least aware of it now. http://gcc.gnu.org/wiki/Better_Uninitialized_Warnings All warnings (new ones prefixed by >>): sound/soc/amd/acp-da7219-max98357a.c: In function 'cz_probe': >> sound/soc/amd/acp-da7219-max98357a.c:325:3: warning: 'ret' may be used uninitialized in this function [-Wmaybe-uninitialized] dev_err(&pdev->dev, "Failed to register regulator: %d\n", ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ret); ~~~~ vim +/ret +325 sound/soc/amd/acp-da7219-max98357a.c 313 314 static int cz_probe(struct platform_device *pdev) 315 { 316 int ret; 317 struct snd_soc_card *card; 318 struct acp_platform_info *machine; 319 struct regulator_dev *rdev; 320 321 acp_da7219_cfg.dev = &pdev->dev; 322 rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, 323 &acp_da7219_cfg); 324 if (IS_ERR(rdev)) { > 325 dev_err(&pdev->dev, "Failed to register regulator: %d\n", 326 ret); 327 return -EINVAL; 328 } 329 330 machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), 331 GFP_KERNEL); 332 if (!machine) 333 return -ENOMEM; 334 card = &cz_card; 335 cz_card.dev = &pdev->dev; 336 platform_set_drvdata(pdev, card); 337 snd_soc_card_set_drvdata(card, machine); 338 ret = devm_snd_soc_register_card(&pdev->dev, &cz_card); 339 if (ret) { 340 dev_err(&pdev->dev, 341 "devm_snd_soc_register_card(%s) failed: %d\n", 342 cz_card.name, ret); 343 return ret; 344 } 345 bt_uart_enable = !device_property_read_bool(&pdev->dev, 346 "bt-pad-enable"); 347 return 0; 348 } 349 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH v2] ASoC: AMD: Add a fix voltage regulator for DA7219 and ADAU7002
by Akshu Agrawal 25 Jul '18

25 Jul '18

DA7219 for our platform need to be configured for 1.8V. Hence, we add a volatge regulator with supplies of 1.8V in the machine driver. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp-da7219-max98357a.c | 43 ++++++++++++++++++++++++++++++++++++ 2 files changed, 44 insertions(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 6cbf9cf..58c1dcb 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -8,6 +8,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_ADAU7002 + select REGULATOR depends on SND_SOC_AMD_ACP && I2C help This option enables machine driver for DA7219 and MAX9835. diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index f42606e..cd3cf6e 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -32,6 +32,8 @@ #include <linux/clk.h> #include <linux/gpio.h> #include <linux/module.h> +#include <linux/regulator/machine.h> +#include <linux/regulator/driver.h> #include <linux/i2c.h> #include <linux/input.h> #include <linux/acpi.h> @@ -320,11 +322,52 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static struct regulator_consumer_supply acp_da7219_supplies[] = { + REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDIO", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("IOVDD", "ADAU7002:00"), +}; + +static struct regulator_init_data acp_da7219_data = { + .constraints = { + .always_on = 1, + }, + .num_consumer_supplies = ARRAY_SIZE(acp_da7219_supplies), + .consumer_supplies = acp_da7219_supplies, +}; + +static struct regulator_config acp_da7219_cfg = { + .init_data = &acp_da7219_data, +}; + +static struct regulator_ops acp_da7219_ops = { +}; + +static struct regulator_desc acp_da7219_desc = { + .name = "reg-fixed-1.8V", + .type = REGULATOR_VOLTAGE, + .owner = THIS_MODULE, + .ops = &acp_da7219_ops, + .fixed_uV = 1800000, /* 1.8V */ + .n_voltages = 1, +}; + static int cz_probe(struct platform_device *pdev) { int ret; struct snd_soc_card *card; struct acp_platform_info *machine; + struct regulator_dev *rdev; + + acp_da7219_cfg.dev = &pdev->dev; + rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, + &acp_da7219_cfg); + if (IS_ERR(rdev)) { + dev_err(&pdev->dev, "Failed to register regulator: %d\n", + ret); + return -EINVAL; + } machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); -- 1.9.1

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[alsa-devel] [PATCH 1/3] ASoC: rt5651: add Volume to the name of volume control
by Bard Liao 25 Jul '18

25 Jul '18

add Volume to the name of volume control Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5651.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 985852f..fe970ed 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -332,11 +332,11 @@ static const struct snd_kcontrol_new rt5651_snd_controls[] = { RT5651_L_VOL_SFT, RT5651_R_VOL_SFT, 175, 0, dac_vol_tlv), /* IN1/IN2/IN3 Control */ - SOC_SINGLE_TLV("IN1 Boost", RT5651_IN1_IN2, + SOC_SINGLE_TLV("IN1 Boost Volume", RT5651_IN1_IN2, RT5651_BST_SFT1, 8, 0, bst_tlv), - SOC_SINGLE_TLV("IN2 Boost", RT5651_IN1_IN2, + SOC_SINGLE_TLV("IN2 Boost Volume", RT5651_IN1_IN2, RT5651_BST_SFT2, 8, 0, bst_tlv), - SOC_SINGLE_TLV("IN3 Boost", RT5651_IN3, + SOC_SINGLE_TLV("IN3 Boost Volume", RT5651_IN3, RT5651_BST_SFT1, 8, 0, bst_tlv), /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5651_INL1_INR1_VOL, -- 2.7.4

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[alsa-devel] [PATCH 0/6] Fix Sparse warnings on sound/soc/intel
by Pierre-Louis Bossart 25 Jul '18

25 Jul '18

Since we found quite a few issues when checking the SOF code with Sparse, I figured I'd run a quick check on the rest of the Intel audio drivers. Some of the issues are trivial but in the end there are a couple of improvements identified by reducing the volume of warnings, so it's worth it overall. If there are concerns about the code readability, I would still recommend merging the last 3-4 patches that are really the interesting ones exposed by Sparse checks. Pierre-Louis Bossart (6): ASoC: Intel: boards: cleanup snd_mask_set usage ASoC: Intel: Haswell: fix endianness handling ASoC: Intel: Skylake: BDL definitions should be __le32 ASoC: Intel: common: make sst_dma functions static ASoC: Intel: Atom: simplify iomem address and casts ASoC: Intel: Atom: fix inversion between __iowrite32 and __ioread32 sound/soc/intel/atom/sst/sst_drv_interface.c | 29 ++++++------ sound/soc/intel/atom/sst/sst_loader.c | 6 +-- sound/soc/intel/boards/bdw-rt5677.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 6 ++- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 6 ++- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/common/sst-firmware.c | 6 +-- sound/soc/intel/haswell/sst-haswell-dsp.c | 53 ++++++++++++---------- sound/soc/intel/skylake/skl-sst-cldma.c | 8 ++-- sound/soc/intel/skylake/skl-sst-cldma.h | 2 +- 16 files changed, 70 insertions(+), 62 deletions(-) -- 2.14.1

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[alsa-devel] Applied "ASoC: Intel: Skylake: fix widget handling" to the asoc tree
by Mark Brown 25 Jul '18

25 Jul '18

The patch ASoC: Intel: Skylake: fix widget handling has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fe65324e3f5205072a2d55ac9c63ec77155fa528 Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> Date: Tue, 24 Jul 2018 19:50:48 -0500 Subject: [PATCH] ASoC: Intel: Skylake: fix widget handling include DAPM Mux and output widgets into the list. Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-topology.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 647e52aecdc3..9845d853644c 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -108,6 +108,9 @@ static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w, case snd_soc_dapm_aif_out: case snd_soc_dapm_dai_out: case snd_soc_dapm_switch: + case snd_soc_dapm_output: + case snd_soc_dapm_mux: + return false; default: return true; -- 2.18.0

1 0

[alsa-devel] [PATCH] Added support for Encore mDSD and Topping DX7s XMOS USB DACs
by JCruk 25 Jul '18

25 Jul '18

Signed-off-by: JCruk <jcrukley(a)gmail.com> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc0..7683d33 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1373,6 +1373,8 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + case USB_ID(0x152a, 0x8750): /* Topping DX7s */ + case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ -- 2.7.4

1 0

Re: [alsa-devel] [PATCH 1/3] ASoC: rt5651: add Volume to the name of volume control
by Mark Brown 25 Jul '18

25 Jul '18

On Wed, Jul 25, 2018 at 02:01:21AM +0000, Bard Liao wrote: > > The RT5631 doesn't appear to have any users except Arndale, certainly no > > x86 users? > As far as I know, there is no x86 users using RT5631. OK, I think it's probably safe to apply that one then as I'm not seeing any in tree users except that Arndale expansion.

1 0

[alsa-devel] [PATCH] ASoC: AMD: Add a fix voltage regulator for DA7219 and ADAU7002
by Akshu Agrawal 25 Jul '18

25 Jul '18

DA7219 for our platform need to be configured for 1.8V. Hence, we add a fixed volatge regulator with supplies of 1.8V in the machine driver. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/Kconfig | 2 ++ sound/soc/amd/acp-da7219-max98357a.c | 45 ++++++++++++++++++++++++++++++++++++ 2 files changed, 47 insertions(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 6cbf9cf..c447a51 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -8,6 +8,8 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_ADAU7002 + select REGULATOR + select REGULATOR_FIXED_VOLTAGE depends on SND_SOC_AMD_ACP && I2C help This option enables machine driver for DA7219 and MAX9835. diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index f42606e..fdf8972 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -31,7 +31,10 @@ #include <sound/jack.h> #include <linux/clk.h> #include <linux/gpio.h> +#include <linux/platform_device.h> #include <linux/module.h> +#include <linux/regulator/fixed.h> +#include <linux/regulator/machine.h> #include <linux/i2c.h> #include <linux/input.h> #include <linux/acpi.h> @@ -320,11 +323,53 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static struct regulator_consumer_supply acp_da7219_supplies[] = { + REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDIO", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("IOVDD", "ADAU7002:00"), +}; + +static struct regulator_init_data acp_da7219_data = { + .constraints = { + .always_on = 1, + }, + .num_consumer_supplies = ARRAY_SIZE(acp_da7219_supplies), + .consumer_supplies = acp_da7219_supplies, +}; + +static struct fixed_voltage_config acp_da7219 = { + .supply_name = "reg-fixed-1.8V", + .microvolts = 1800000, /* 1.8V */ + .gpio = -EINVAL, + .enabled_at_boot = 1, + .init_data = &acp_da7219_data, +}; + +static struct platform_device acp_da7219_regulator = { + .name = "reg-fixed-voltage", + .id = PLATFORM_DEVID_AUTO, + .dev = { + .platform_data = &acp_da7219, + }, +}; + static int cz_probe(struct platform_device *pdev) { int ret; struct snd_soc_card *card; struct acp_platform_info *machine; + static bool regulators_registered; + + if (!regulators_registered) { + ret = platform_device_register(&acp_da7219_regulator); + if (ret) { + dev_err(&pdev->dev, "Failed to register regulator: %d\n", + ret); + return ret; + } + regulators_registered = true; + } machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); -- 1.9.1

3 4

Re: [alsa-devel] 答复: [PATCH 3/4] drm/radeon: Add audio component support
by Takashi Iwai 25 Jul '18

25 Jul '18

On Wed, 25 Jul 2018 05:32:52 +0200, Qu, Jim wrote: > @@ -269,6 +271,10 @@ static void radeon_audio_enable(struct radeon_device *rdev, > > if (rdev->audio.funcs->enable) > rdev->audio.funcs->enable(rdev, pin, enable_mask); > + > + if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) > + acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, > + pin->id, -1); > > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> > Jim: radeon_audio_enable() can enable audios and also disable them, so eld noitfy callback should be called when enable_mask is true. It's intentional. The notifier needs to be called at disablement time, too. The audio driver has to follow the HDMI audio disablement, and notifier receives both on and off case. The actual state is inquired via get_eld call by HD-audio followed after the notification. thanks, Takashi

2 1

[alsa-devel] Building alsa-utils, adding changes
by Jamey Kirby 25 Jul '18

25 Jul '18

I have been working with alsa for a while. We use it to archive content from our radio station. I am using arecord. Build problem: 1) I cloned alsa-utils from github 2) Spent two hours on Google trying to figure out how to build the darn thing. 3) I am running Ubuntu Server 18.0 with build-essentials installed. 4) I get errors that I do not know how to fix when I try to setup ./configure. 5) I just need to make a build. What do I need to do? Feature: I am adding a feature. In the loop for capture(), when a new file is opened, I want to close the old file (when doing continuous recording). I added a new CLI parameter -K. When this is set, the capture() loop will close the old file before opening the new file. This is an issue for me for two reasons: 1) When will i run out of file handles? I capture one hour files continuously. After 10 hours, I see that all the capture files are still open? 2) I need to be notified when the file is closed so that I can do post-processing of rht capture. I've tried inotifywait in a script, and in another script I tried lsof. lsof works great, but the files are open,. See, the main issue is that I need to know when each file completes so that I can post process. Here is my post process script outline: #!/bin/bash for f in $HOME/vboxshare/jkirby/newarc/*.raw do filename=$(basename $f) open=$(lsof -t $f | wc -w) if [ $open -lt 1 ] then echo "$filename is closed" echo "Do post processing here" else echo "$filename is open by $open processe(s)" fi done If I could close the file before or after the previous file is closed, I can do my work. Otherwise, I have no way to know if the file is open and being processed or not. So, I've made the changes, I just need to build the darn thing. Thanks for any advice or help. -- Jamey Kirby Disrupting the establishment since 1964 *This is a personal email account and as such, emails are not subject to archiving. Nothing else really matters.*

1 0

[alsa-devel] [PATCH v2] ASoC: da7219: Allow pdata to specify a VDDIO
by Daniel Kurtz 24 Jul '18

24 Jul '18

Some systems do not have software controllable regulators driving the DA7219's supplies, nor can they use device tree to create "always-on fixed regulators" to easily pretend like they do. On these systems the call to devm_regulator_bulk_get() just creates a set of dummy registers. Calling regulator_get_voltage() on a dummy regulator just returns -EINVAL, in which case the DA7219 is always set up to use the default VDDIO voltage range of 2.5-3.6V. Provide a new device property to let such systems specify a different VDDIO if needed (e.g., 1.8V). Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> --- Changes for v2: - fix to use device_property_read_u32() include/sound/da7219.h | 2 ++ sound/soc/codecs/da7219.c | 19 +++++++++++++------ 2 files changed, 15 insertions(+), 6 deletions(-) diff --git a/include/sound/da7219.h b/include/sound/da7219.h index 1bfcb16f2d10ab..16ab125ad4adbf 100644 --- a/include/sound/da7219.h +++ b/include/sound/da7219.h @@ -38,6 +38,8 @@ struct da7219_pdata { const char *dai_clks_name; + u32 vddio; + /* Mic */ enum da7219_micbias_voltage micbias_lvl; enum da7219_mic_amp_in_sel mic_amp_in_sel; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 980a6a8bf56d38..9893920b26f41f 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1634,6 +1634,9 @@ static struct da7219_pdata *da7219_fw_to_pdata(struct snd_soc_component *compone else pdata->mic_amp_in_sel = DA7219_MIC_AMP_IN_SEL_DIFF; + if (device_property_read_u32(dev, "dlg,vddio", &of_val32) >= 0) + pdata->vddio = of_val32; + return pdata; } @@ -1717,8 +1720,12 @@ static int da7219_handle_supplies(struct snd_soc_component *component) /* Determine VDDIO voltage provided */ vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer; ret = regulator_get_voltage(vddio); + /* If regulator_get_voltage() fails, try to use vddio from pdata. */ + if (ret < 0 && da7219->pdata) + ret = da7219->pdata->vddio; if (ret < 1200000) - dev_warn(component->dev, "Invalid VDDIO voltage\n"); + dev_warn(component->dev, "Invalid VDDIO voltage: %d mV\n", + ret); else if (ret < 2800000) io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V; @@ -1872,6 +1879,11 @@ static int da7219_probe(struct snd_soc_component *component) mutex_init(&da7219->ctrl_lock); mutex_init(&da7219->pll_lock); + /* Handle DT/ACPI/Platform data */ + da7219->pdata = dev_get_platdata(component->dev); + if (!da7219->pdata) + da7219->pdata = da7219_fw_to_pdata(component); + /* Regulator configuration */ ret = da7219_handle_supplies(component); if (ret) @@ -1897,11 +1909,6 @@ static int da7219_probe(struct snd_soc_component *component) break; } - /* Handle DT/ACPI/Platform data */ - da7219->pdata = dev_get_platdata(component->dev); - if (!da7219->pdata) - da7219->pdata = da7219_fw_to_pdata(component); - da7219_handle_pdata(component); /* Check if MCLK provided */ -- 2.18.0.233.g985f88cf7e-goog

3 2

[alsa-devel] [PATCH 1/4] ASoC: fsl-asoc-card: Switch to SPDX identifier
by Fabio Estevam 24 Jul '18

24 Jul '18

Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 18 +++++++----------- 1 file changed, 7 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 4a6750a..07808c6 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -1,14 +1,10 @@ -/* - * Freescale Generic ASoC Sound Card driver with ASRC - * - * Copyright (C) 2014 Freescale Semiconductor, Inc. - * - * Author: Nicolin Chen <nicoleotsuka(a)gmail.com> - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale Generic ASoC Sound Card driver with ASRC +// +// Copyright (C) 2014 Freescale Semiconductor, Inc. +// +// Author: Nicolin Chen <nicoleotsuka(a)gmail.com> #include <linux/clk.h> #include <linux/i2c.h> -- 2.7.4

2 7

[alsa-devel] Incorrect name of PCM
by Christopher Head 24 Jul '18

24 Jul '18

Hi folks, I have an Asus P8H77-I motherboard. On board is an Intel HDA connected to a Via VT1708S codec. There are a number of analogue ports. In addition to them, on the back panel are an HDMI port (from an integrated GPU) and a TOSLINK port. The motherboard manual isn’t totally clear on whether the HDMI port can also carry sound or only video, but Windows claims to be able to send sound there, so probably yes. I don’t use the integrated GPU, though. Here’s the problem: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: VT1708S Analog [VT1708S Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 2: VT1708S Alt Analog [VT1708S Alt Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: VT1708S Digital [VT1708S Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 OK, a little bit unexpected that there is only one digital device listed rather than two. The codec datasheet says it should have two digital output widgets. Maybe it’s due to the integrated GPU being disabled though. $ aplay -L null Discard all samples (playback) or generate zero samples (capture) default:CARD=PCH HDA Intel PCH, VT1708S Analog Default Audio Device sysdefault:CARD=PCH HDA Intel PCH, VT1708S Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog Front speakers surround21:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Digital HDMI Audio Output hdmi:CARD=NVidia,DEV=0 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=1 HDA NVidia, HDMI 1 HDMI Audio Output hdmi:CARD=NVidia,DEV=2 HDA NVidia, HDMI 2 HDMI Audio Output hdmi:CARD=NVidia,DEV=3 HDA NVidia, HDMI 3 HDMI Audio Output Why is this wrong? Because when I play to the “hdmi” PCM, at least with the S/PDIF switch turned on in mixer controls, the sound comes out the TOSLINK port! It seems that the PCM naming is wrong. This is all without any configuration file. Everything works fine, just the name is wrong. I don’t know whether turning off the S/PDIF switch results in the sound coming out the HDMI port instead; maybe it does (I don’t have anything that speaks HDMI and can also play sound to test with). Either way, seems like it should be called either “spdif” (if it’s only the TOSLINK port) or some combination of the two (if it’s both and switchable by the mixer switch). If replying, please note that I’m not subscribed. Thanks! -- Christopher Head

3 10

Re: [alsa-devel] [PATCH 1/3] ASoC: rt5651: add Volume to the name of volume control
by Mark Brown 24 Jul '18

24 Jul '18

On Tue, Jul 24, 2018 at 01:49:48AM +0000, Bard Liao wrote: > Please ignore these patches. There is no significant advantage of > adding "Volume" to the control name. To prevent breaking the > configuration files on user space, we should find out all projects > which are using these controls and modify them. It will take time. > So I think it is better to leave it as it is. The RT5631 doesn't appear to have any users except Arndale, certainly no x86 users?

1 0

[alsa-devel] Applied "ASoC: doc: replace codec to component" to the asoc tree
by Mark Brown 24 Jul '18

24 Jul '18

The patch ASoC: doc: replace codec to component has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a3ac7727f3a226b0569431de62a4e9bb4c770b2b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 28 Mar 2018 02:14:22 +0000 Subject: [PATCH] ASoC: doc: replace codec to component Now we can replace Codec to Component. Let's do it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/sound/soc/codec.rst | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/sound/soc/codec.rst b/Documentation/sound/soc/codec.rst index f87612b94812..58f625fe3d39 100644 --- a/Documentation/sound/soc/codec.rst +++ b/Documentation/sound/soc/codec.rst @@ -179,12 +179,12 @@ i.e. static int wm8974_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf; + struct snd_soc_component *component = dai->component; + u16 mute_reg = snd_soc_component_read32(component, WM8974_DAC) & 0xffbf; if (mute) - snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40); + snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40); else - snd_soc_write(codec, WM8974_DAC, mute_reg); + snd_soc_component_write(component, WM8974_DAC, mute_reg); return 0; } -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: trace: remove snd_soc_codec" to the asoc tree
by Mark Brown 24 Jul '18

24 Jul '18

The patch ASoC: trace: remove snd_soc_codec has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 211b4415ecb254e167837206756dea123342689c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 28 Mar 2018 02:18:13 +0000 Subject: [PATCH] ASoC: trace: remove snd_soc_codec snd_soc_codec is replaced to snd_soc_component, and it is not used in this file. Let's remove it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/trace/events/asoc.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index ccd1a3bdff46..40c300fe704d 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -12,7 +12,6 @@ #define DAPM_ARROW(dir) (((dir) == SND_SOC_DAPM_DIR_OUT) ? "->" : "<-") struct snd_soc_jack; -struct snd_soc_codec; struct snd_soc_card; struct snd_soc_dapm_widget; struct snd_soc_dapm_path; -- 2.18.0

1 0

[alsa-devel] [PATCH] ASoC: meson: axg-spdifout: select SND_PCM_IEC958
by Arnd Bergmann 24 Jul '18

24 Jul '18

When CONFIG_SND_PCM_IEC958 is disabled, we get a link error for the new driver: sound/soc/meson/axg-spdifout.o: In function `axg_spdifout_hw_params': axg-spdifout.c:(.text+0x650): undefined reference to `snd_pcm_create_iec958_consumer_hw_params' The other users use 'select', so we should do the same here. Fixes: 53eb4b7aaa04 ("ASoC: meson: add axg spdif output") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/meson/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 4cf93c05a982..8af8bc358a90 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -56,6 +56,7 @@ config SND_MESON_AXG_SOUND_CARD config SND_MESON_AXG_SPDIFOUT tristate "Amlogic AXG SPDIF Output Support" + select SND_PCM_IEC958 imply SND_SOC_SPDIF help Select Y or M to add support for SPDIF output serializer embedded -- 2.18.0

3 2

[alsa-devel] [PATCH] ASoC: rt5514: Fix the issue of the delay volume applied
by Oder Chiou 24 Jul '18

24 Jul '18

The patch fixes the issue of the delay volume applied. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 1570b91..dca82dd 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -64,8 +64,8 @@ static const struct reg_sequence rt5514_patch[] = { {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_ADCFED, 0x00000800}, {RT5514_ASRC_IN_CTRL1, 0x00000003}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, }; static const struct reg_default rt5514_reg[] = { @@ -92,10 +92,10 @@ static const struct reg_default rt5514_reg[] = { {RT5514_ASRC_IN_CTRL1, 0x00000003}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, {RT5514_DOWNFILTER1_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER1_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_LDO18_16, 0x02000345}, {RT5514_ANA_CTRL_ADC12, 0x0000a2a8}, -- 2.7.4

2 1

[alsa-devel] WARNING in port_delete
by Dae R. Jeong 24 Jul '18

24 Jul '18

Reporting the crash: WARNING in port_delete This crash has been found in v4.18-rc3 using RaceFuzzer (a modified version of Syzkaller), which we descrbie more at the end of this report. Our analysis shows that the race occurs when invoking two close syscalls concurrently. The executed program is as follows: r0 = open("/dev/snd/seq", 0x0, 0x0) ioctl(r0, SNDRV_SEQ_IOCTL_CREATE_PORT, {{0x80}, "706f72943000000000000000000000000000000000000000000000000000000000233f770800000000000000000000000000000000000000001000", 0xffffffffffffffff, 0xfffffffffffffbff}) r1 = openat(AT_FDCWD, "/dev/sequencer2", 0x8402, 0x0) close(r0) close(r1) and two threads executed the program as follows: Thread0 Thread1 open("/dev/snd/seq") ioctl(r0, SNDRV_SEQ_IOCTL_CREATE_PORT) openat("/dev/sequencer2") close(r0) close(r1) (Note that two close() syscalls were exeuted after openat() syscall, and two close() syscalls were executed concurrently.) Crash log: ================================================================== WARNING: CPU: 1 PID: 32519 at /home/daeryong/workspace/race-fuzzer/kernels_repo/kernel_v4.18-rc3/sound/core/seq/seq_ports.c:275 port_delete+0xde/0xf0 sound/core/seq/seq_ports.c:275 Kernel panic - not syncing: panic_on_warn set ... CPU: 1 PID: 32519 Comm: syz-executor0 Not tainted 4.18.0-rc3 #1 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS rel-1.8.2-0-g33fbe13 by qemu-project.org 04/01/2014 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x16e/0x22c lib/dump_stack.c:113 panic+0x1a8/0x3a7 kernel/panic.c:184 __warn+0x191/0x1a0 kernel/panic.c:536 report_bug+0x132/0x1b0 lib/bug.c:186 fixup_bug.part.10+0x28/0x50 arch/x86/kernel/traps.c:178 fixup_bug arch/x86/kernel/traps.c:247 [inline] do_error_trap+0x284/0x2c0 arch/x86/kernel/traps.c:296 do_invalid_op+0x1b/0x20 arch/x86/kernel/traps.c:316 invalid_op+0x14/0x20 arch/x86/entry/entry_64.S:992 RIP: 0010:port_delete+0xde/0xf0 sound/core/seq/seq_ports.c:275 Code: 00 00 e8 b5 68 d2 fd 8b 83 58 01 00 00 85 c0 75 1d e8 86 5c ae fd 48 89 df e8 3e 52 d2 fd 31 c0 5b 41 5c 5d c3 e8 72 5c ae fd <0f> 0b eb c8 e8 69 5c ae fd 0f 0b eb da 0f 1f 44 00 00 55 48 89 e5 RSP: 0018:ffff8801d635f900 EFLAGS: 00010216 RAX: 0000000000040000 RBX: ffff8801da196900 RCX: ffffffff839b86ee RDX: 0000000000005e94 RSI: ffffc90002640000 RDI: ffff8801da196978 RBP: ffff8801d635f910 R08: 0000000000000098 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: ffffffff839ba820 R13: ffff8801da196950 R14: ffff8801ed5bc080 R15: ffff8801da196958 snd_seq_delete_port+0x2b0/0x2d0 sound/core/seq/seq_ports.c:303 snd_seq_ioctl_delete_port+0x5b/0xa0 sound/core/seq/seq_clientmgr.c:1325 snd_seq_kernel_client_ctl+0xd4/0xf0 sound/core/seq/seq_clientmgr.c:2361 snd_seq_event_port_detach+0xcb/0x120 sound/core/seq/seq_ports.c:705 delete_port+0x3a/0x70 sound/core/seq/oss/seq_oss_init.c:354 snd_seq_oss_release+0x7c/0x90 sound/core/seq/oss/seq_oss_init.c:433 odev_release+0x40/0x60 sound/core/seq/oss/seq_oss.c:153 __fput+0x234/0x470 fs/file_table.c:209 ____fput+0x15/0x20 fs/file_table.c:243 task_work_run+0x15a/0x1c0 kernel/task_work.c:113 tracehook_notify_resume include/linux/tracehook.h:192 [inline] exit_to_usermode_loop+0x2a3/0x2b0 arch/x86/entry/common.c:166 prepare_exit_to_usermode arch/x86/entry/common.c:197 [inline] syscall_return_slowpath arch/x86/entry/common.c:268 [inline] do_syscall_64+0x485/0x4b0 arch/x86/entry/common.c:293 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x456469 Code: 1d ba fb ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 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 eb b9 fb ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007fe9a8936b28 EFLAGS: 00000246 ORIG_RAX: 0000000000000003 RAX: 0000000000000000 RBX: 000000000072bfa0 RCX: 0000000000456469 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 0000000000000014 RBP: 0000000000000054 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007fe9a89376d4 R13: 00000000ffffffff R14: 00000000006f5880 R15: 0000000000000000 Dumping ftrace buffer: (ftrace buffer empty) Kernel Offset: disabled Rebooting in 86400 seconds.. = About RaceFuzzer RaceFuzzer is a customized version of Syzkaller, specifically tailored to find race condition bugs in the Linux kernel. While we leverage many different technique, the notable feature of RaceFuzzer is in leveraging a custom hypervisor (QEMU/KVM) to interleave the scheduling. In particular, we modified the hypervisor to intentionally stall a per-core execution, which is similar to supporting per-core breakpoint functionality. This allows RaceFuzzer to force the kernel to deterministically trigger racy condition (which may rarely happen in practice due to randomness in scheduling). RaceFuzzer's C repro always pinpoints two racy syscalls. Since C repro's scheduling synchronization should be performed at the user space, its reproducibility is limited (reproduction may take from 1 second to 10 minutes (or even more), depending on a bug). This is because, while RaceFuzzer precisely interleaves the scheduling at the kernel's instruction level when finding this bug, C repro cannot fully utilize such a feature. Please disregard all code related to "should_hypercall" in the C repro, as this is only for our debugging purposes using our own hypervisor.

3 4

[alsa-devel] [PATCH 0/4] drm_audio_component support for AMD/ATI HDMI codecs
by Takashi Iwai 24 Jul '18

24 Jul '18

Hi, this is a patch set to add the support of drm_audio_component for AMD/ATI HDMI codecs. With the drm_audio_component, the HDMI/DP audio hotplug and ELD read-out can be achieved directly without the hardware access. The best point by that is that it makes the hotplug notification working even during runtime suspend. The support is totally optional and dynamic, hence it still works even if either HD-audio or DRM side isn't patched, and it'll fall back to the existing method. The current patch supports only radeon and a part of amdgpu; the DC support isn't included yet. Takashi === Takashi Iwai (4): ALSA: hda/hdmi: Use single mutex unlock in error paths ALSA: hda/hdmi: Allow audio component for AMD/ATI HDMI drm/radeon: Add audio component support drm/amdgpu: Add audio component support drivers/gpu/drm/Kconfig | 2 + drivers/gpu/drm/amd/amdgpu/Makefile | 2 +- drivers/gpu/drm/amd/amdgpu/amdgpu.h | 4 + drivers/gpu/drm/amd/amdgpu/amdgpu_audio.c | 97 ++++++++++ drivers/gpu/drm/amd/amdgpu/amdgpu_mode.h | 3 + drivers/gpu/drm/amd/amdgpu/dce_v10_0.c | 6 + drivers/gpu/drm/amd/amdgpu/dce_v11_0.c | 6 + drivers/gpu/drm/amd/amdgpu/dce_v6_0.c | 6 + drivers/gpu/drm/amd/amdgpu/dce_v8_0.c | 6 + drivers/gpu/drm/radeon/radeon.h | 3 + drivers/gpu/drm/radeon/radeon_audio.c | 79 ++++++++ sound/pci/hda/patch_hdmi.c | 209 +++++++++++++++++----- 12 files changed, 374 insertions(+), 49 deletions(-) create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_audio.c -- 2.18.0

2 6

[alsa-devel] Applied "ASoC: rockchip-i2s: add description for px30" to the asoc tree
by Mark Brown 23 Jul '18

23 Jul '18

The patch ASoC: rockchip-i2s: add description for px30 has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 2f562a4739602d96928de0b040ce46b00d8d3adc Mon Sep 17 00:00:00 2001 From: Liang Chen <cl(a)rock-chips.com> Date: Mon, 23 Jul 2018 17:25:21 +0800 Subject: [PATCH] ASoC: rockchip-i2s: add description for px30 Add "rockchip,px30-i2s", "rockchip,rk3066-i2s" for i2s on px30 platform. Acked-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Liang Chen <cl(a)rock-chips.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt index b208a752576c..54aefab71f2c 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -7,6 +7,7 @@ Required properties: - compatible: should be one of the following: - "rockchip,rk3066-i2s": for rk3066 + - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30 - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036 - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228 -- 2.18.0

1 0

[alsa-devel] [PATCH] alsa/usb-audio: Generic DSD detection for Thesycon-based implementations
by Yue Wang 23 Jul '18

23 Jul '18

Thesycon provides solutions to XMOS chips, and has its own device vendor id. In this patch, we use generic method to detect DSD capability of Thesycon-based UAC2 implementations in order to support a wide range of current and future devices. The patch will enable the SNDRV_PCM_FMTBIT_DSD_U32_BE bit for the DAC hence enable native DSD playback up to DSD512 format. Signed-off-by: Yue Wang <yuleopen(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc02767f..06ae3f5adf4b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1443,6 +1443,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, */ switch (USB_ID_VENDOR(chip->usb_id)) { case 0x20b1: /* XMOS based devices */ + case 0x152a: /* Thesycon devices */ case 0x25ce: /* Mytek devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; -- 2.18.0

2 1

[alsa-devel] [PATCH] alsa/usb-audio: Generic DSD detection for Thesycon-based implementations
by Yue Wang 23 Jul '18

23 Jul '18

Thesycon provides solutions to XMOS chips, and has its own device vendor id. In this patch, we use generic method to detect DSD capability of Thesycon-based UAC2 implementations in order to support a wide range of current and future devices. The patch will enable the SNDRV_PCM_FMTBIT_DSD_U32_BE bit for the DAC hence enable native DSD playback up to DSD512 format. Signed-off-by: Yue Wang <yuleopen(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc02767f..c51e2dee3075 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1374,6 +1374,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, break; case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ + case USB_ID(0x152a, 0x8750): /* Topping DX7s */ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ -- 2.18.0

2 2

[alsa-devel] [PATCH] alsa/usb-audio: Topping DX7s quirk for DSD interface
by Yue Wang 23 Jul '18

23 Jul '18

This patch adds the support for native DSD for Topping DX7s DAC. The product description is on http://en.tpdz.net/products_detail/productId=32.html and it's available on https://www.amazon.com/Balanced-Headphone-Amplifier-Topping-2ES9038Q2M/dp/B… The DAC uses XMOS chip but uses a special vendor id and product id. The patch will enable the SNDRV_PCM_FMTBIT_DSD_U32_BE bit for the DAC hence enable native DSD playback up to DSD512 format. Signed-off-by: Yue Wang <yuleopen(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc02767f..c51e2dee3075 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1374,6 +1374,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, break; case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ + case USB_ID(0x152a, 0x8750): /* Topping DX7s */ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ -- 2.18.0

2 2

Re: [alsa-devel] [PATCH] alsa/usb-audio: Topping DX7s quirk for DSD interface
by Takashi Iwai 23 Jul '18

23 Jul '18

On Mon, 23 Jul 2018 09:25:34 +0200, <yuleopen(a)gmail.com> wrote: > > From: Yue Wang <yuleopen(a)gmail.com> > > Signed-off-by: Yue Wang <yuleopen(a)gmail.com> Please give more explanation. No commit message already indicates that something is very wrong, no matter what the change is. thanks, Takashi

2 2

[alsa-devel] [PATCH] ALSA: core: fix unsigned int pages overflow when comapred
by He, Bo 23 Jul '18

23 Jul '18

we see the below kernel panic on stress suspend resume test in snd_malloc_sgbuf_pages(), snd_dma_alloc_pages_fallback() alloc chunk maybe larger than the left pages due to the pages alignment, which will cause the pages overflow. while (pages > 0) { ... pages -= chunk; } the patch is change the pages from unsigned int to int to fix the issue. BUG: unable to handle kernel paging request at ffff88000deb4000 IP: [<ffffffff81404fa9>] memset_erms+0x9/0x10 Call Trace: [<ffffffff818f222f>] snd_dma_alloc_pages+0xff/0x210 [<ffffffff818f23af>] snd_dma_alloc_pages_fallback+0x6f/0x90 [<ffffffff818f2b85>] snd_malloc_sgbuf_pages+0x145/0x370 [<ffffffff818f229e>] snd_dma_alloc_pages+0x16e/0x210 [<ffffffffc011930d>] hdac_ext_dma_alloc_pages+0x1d/0x40 [snd_hda_ext_core] [<ffffffffc010729a>] snd_hdac_dsp_prepare+0xca/0x1c0 [snd_hda_core] [<ffffffffc01880f9>] skl_dsp_prepare+0x99/0xf0 [snd_soc_skl] [<ffffffffc0162a7e>] bxt_load_base_firmware+0x9e/0x5c0 [snd_soc_skl_ipc] [<ffffffffc01630ec>] bxt_set_dsp_D0+0x14c/0x300 [snd_soc_skl_ipc] [<ffffffffc015f9c3>] skl_dsp_get_core+0x43/0xd0 [snd_soc_skl_ipc] [<ffffffffc015fa60>] skl_dsp_wake+0x10/0x20 [snd_soc_skl_ipc] [<ffffffffc0188e3e>] skl_resume_dsp+0x7e/0x140 [snd_soc_skl] [<ffffffffc0183c4a>] skl_resume+0xda/0x170 [snd_soc_skl] [<ffffffff81452726>] pci_pm_resume+0x76/0xe0 [<ffffffff816616da>] dpm_run_callback+0x5a/0x180 [<ffffffff81661e3c>] device_resume+0xdc/0x2c0 [<ffffffff81663818>] dpm_resume+0x118/0x310 [<ffffffff81663e11>] dpm_resume_end+0x11/0x20 [<ffffffff810f8bcc>] suspend_devices_and_enter+0x11c/0x2b0 [<ffffffff810f90bd>] pm_suspend+0x35d/0x3d0 [<ffffffff810f78a6>] state_store+0x66/0x90 [<ffffffff813f80e2>] kobj_attr_store+0x12/0x20 [<ffffffff812a37bc>] sysfs_kf_write+0x3c/0x50 [<ffffffff812a2cbd>] kernfs_fop_write+0x11d/0x1a0 [<ffffffff8121dfaa>] __vfs_write+0x3a/0x150 [<ffffffff8121f2b1>] vfs_write+0xb1/0x1a0 [<ffffffff81220898>] SyS_write+0x58/0xc0 [<ffffffff81001fca>] do_syscall_64+0x6a/0xe0 [<ffffffff81b06560>] entry_SYSCALL_64_after_swapgs+0x5d/0xd7 Signed-off-by: he, bo <bo.he(a)intel.com> Signed-off-by: zhang jun <jun.zhang(a)intel.com> --- sound/core/sgbuf.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index 84fffab..33449ee 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -68,7 +68,8 @@ void *snd_malloc_sgbuf_pages(struct device *device, size_t *res_size) { struct snd_sg_buf *sgbuf; - unsigned int i, pages, chunk, maxpages; + unsigned int i, chunk, maxpages; + int pages; struct snd_dma_buffer tmpb; struct snd_sg_page *table; struct page **pgtable; -- 2.7.4

3 7

[alsa-devel] [PATCH v6 00/10] soundwire: Add multi link support
by Sanyog Kale 23 Jul '18

23 Jul '18

Currently, in the SoundWire subsystem, the concept of stream is limited to a Master and one or more Slaves(Codecs). This series aims to add support for multiple Master(s) sharing the same reference clock and synchronized in the hardware. This patch series adds: - Fix to avoid duplicate stream state assignment - Avoid incorrect stream release after configuring Master - Helpers to lock bus instances part of the stream - Boiler plate conversion of code to support a list of Master runtime - Support multi link bank switch to support synchronization between multiple masters - Add Intel platform ops for pre/post bank switch - Fix to avoid bus lock acquired twice changes in v6: - Add patch to fix acquiring bus lock twice during master release - In sdw_stream_add_master(), perform multi link check at start of function as suggested by Pierre - Minor comment fixes changes in v5: - Add patch to fix duplicate stream state assignment - fix incorrect exit which was releasing the stream after configuring master - In sdw_release_bus_lock(), release the mutex in reverse order as suggested by Takashi changes in v4: - Added changes in sdw_stream_runtime structure to track Masters added to stream - Add changes to support ml_bankswitch only if both the hardware supports and the stream is handled by multiple masters Sanyog Kale (3): Documentation: soundwire: Add documentation for multi link soundwire: Add support to lock across bus instances soundwire: Fix acquiring bus lock twice during master release Shreyas NC (6): soundwire: Initialize completion for defer messages soundwire: Fix duplicate stream state assignment soundwire: fix incorrect exit after configuring stream soundwire: keep track of Masters in a stream soundwire: Add support for multi link bank switch soundwire: intel: Add pre/post bank switch ops Vinod Koul (1): soundwire: Handle multiple master instances in a stream Documentation/driver-api/soundwire/stream.rst | 28 ++ drivers/soundwire/bus.c | 6 + drivers/soundwire/bus.h | 4 + drivers/soundwire/intel.c | 65 ++++ drivers/soundwire/stream.c | 511 +++++++++++++++++++------- include/linux/soundwire/sdw.h | 12 +- 6 files changed, 489 insertions(+), 137 deletions(-) -- 2.13.0

3 14

[alsa-devel] [PATCH] ASoC: wm9712: fix replace codec to component
by Marcel Ziswiler 23 Jul '18

23 Jul '18

From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Since commit 143b44845d87 ("ASoC: wm9712: replace codec to component") "wm9712-codec" got renamed to "wm9712-component", however, this change never got propagated down to the actual board/platform drivers. E.g. on Colibri T20 this lead to the following spew upon boot with sound/touch being broken: [ 2.214121] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 2.222137] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 2.344384] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 2.351885] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 2.668339] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 2.675811] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 3.208408] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 3.216312] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 3.235397] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 3.248938] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 14.970443] ALSA device list: [ 14.996628] No soundcards found. This commit finally fixes this again. Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> --- arch/arm/mach-pxa/tosa.c | 2 +- arch/mips/alchemy/devboards/db1300.c | 2 +- drivers/mfd/wm97xx-core.c | 6 +++--- sound/soc/au1x/db1200.c | 4 ++-- sound/soc/fsl/pcm030-audio-fabric.c | 6 +++--- sound/soc/fsl/phycore-ac97.c | 6 +++--- sound/soc/pxa/e800_wm9712.c | 4 ++-- sound/soc/pxa/em-x270.c | 4 ++-- sound/soc/pxa/palm27x.c | 4 ++-- sound/soc/pxa/tosa.c | 4 ++-- sound/soc/tegra/tegra_wm9712.c | 4 ++-- 11 files changed, 23 insertions(+), 23 deletions(-) diff --git a/arch/arm/mach-pxa/tosa.c b/arch/arm/mach-pxa/tosa.c index cb5cd8e78c94..32011e600496 100644 --- a/arch/arm/mach-pxa/tosa.c +++ b/arch/arm/mach-pxa/tosa.c @@ -848,7 +848,7 @@ static struct platform_device sharpsl_rom_device = { }; static struct platform_device wm9712_device = { - .name = "wm9712-codec", + .name = "wm9712-component", .id = -1, }; diff --git a/arch/mips/alchemy/devboards/db1300.c b/arch/mips/alchemy/devboards/db1300.c index efb318e03e0a..841a27736eb1 100644 --- a/arch/mips/alchemy/devboards/db1300.c +++ b/arch/mips/alchemy/devboards/db1300.c @@ -638,7 +638,7 @@ static struct platform_device db1300_sd0_dev = { /**********************************************************************/ static struct platform_device db1300_wm9715_dev = { - .name = "wm9712-codec", + .name = "wm9712-component", .id = 1, /* ID of PSC for AC97 audio, see asoc glue! */ }; diff --git a/drivers/mfd/wm97xx-core.c b/drivers/mfd/wm97xx-core.c index f5a8347f837f..9fec11fc0410 100644 --- a/drivers/mfd/wm97xx-core.c +++ b/drivers/mfd/wm97xx-core.c @@ -109,7 +109,7 @@ static const struct regmap_config wm9705_regmap_config = { }; static struct mfd_cell wm9705_cells[] = { - { .name = "wm9705-codec", }, + { .name = "wm9705-component", }, { .name = "wm97xx-ts", }, }; @@ -177,7 +177,7 @@ static const struct regmap_config wm9712_regmap_config = { }; static struct mfd_cell wm9712_cells[] = { - { .name = "wm9712-codec", }, + { .name = "wm9712-component", }, { .name = "wm97xx-ts", }, }; @@ -248,7 +248,7 @@ static const struct regmap_config wm9713_regmap_config = { }; static struct mfd_cell wm9713_cells[] = { - { .name = "wm9713-codec", }, + { .name = "wm9713-component", }, { .name = "wm97xx-ts", }, }; diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 301e1fc9a377..d452478d62b4 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -52,7 +52,7 @@ static struct snd_soc_dai_link db1200_ac97_dai = { .codec_dai_name = "ac97-hifi", .cpu_dai_name = "au1xpsc_ac97.1", .platform_name = "au1xpsc-pcm.1", - .codec_name = "ac97-codec.1", + .codec_name = "ac97-component.1", }; static struct snd_soc_card db1200_ac97_machine = { @@ -68,7 +68,7 @@ static struct snd_soc_dai_link db1300_ac97_dai = { .codec_dai_name = "wm9712-hifi", .cpu_dai_name = "au1xpsc_ac97.1", .platform_name = "au1xpsc-pcm.1", - .codec_name = "wm9712-codec.1", + .codec_name = "wm9712-component.1", }; static struct snd_soc_card db1300_ac97_machine = { diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index ec731223cab3..869f4b226ccd 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -33,14 +33,14 @@ static struct snd_soc_dai_link pcm030_fabric_dai[] = { .stream_name = "AC97 Analog", .codec_dai_name = "wm9712-hifi", .cpu_dai_name = "mpc5200-psc-ac97.0", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, { .name = "AC97.1", .stream_name = "AC97 IEC958", .codec_dai_name = "wm9712-aux", .cpu_dai_name = "mpc5200-psc-ac97.1", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, }; @@ -85,7 +85,7 @@ static int pcm030_fabric_probe(struct platform_device *op) if (ret) dev_err(&op->dev, "request_module returned: %d\n", ret); - pdata->codec_device = platform_device_alloc("wm9712-codec", -1); + pdata->codec_device = platform_device_alloc("wm9712-component", -1); if (!pdata->codec_device) dev_err(&op->dev, "platform_device_alloc() failed\n"); diff --git a/sound/soc/fsl/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c index 66fb6c4614d2..dbd81d3b7c2c 100644 --- a/sound/soc/fsl/phycore-ac97.c +++ b/sound/soc/fsl/phycore-ac97.c @@ -30,8 +30,8 @@ static struct snd_soc_dai_link imx_phycore_dai_ac97[] = { { .name = "HiFi", .stream_name = "HiFi", - .codec_dai_name = "wm9712-hifi", - .codec_name = "wm9712-codec", + .codec_dai_name = "wm9712-hifi", + .codec_name = "wm9712-component", .cpu_dai_name = "imx-ssi.0", .platform_name = "imx-ssi.0", .ops = &imx_phycore_hifi_ops, @@ -88,7 +88,7 @@ static int __init imx_phycore_init(void) if (ret) goto fail1; - imx_phycore_snd_device = platform_device_alloc("wm9712-codec", -1); + imx_phycore_snd_device = platform_device_alloc("wm9712-component", -1); if (!imx_phycore_snd_device) { ret = -ENOMEM; goto fail2; diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 1ed8aa2348f1..937746d1e23e 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -75,7 +75,7 @@ static struct snd_soc_dai_link e800_dai[] = { .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, { .name = "AC97 Aux", @@ -83,7 +83,7 @@ static struct snd_soc_dai_link e800_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, }; diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index e046770ce70e..e958c4c241f6 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -37,7 +37,7 @@ static struct snd_soc_dai_link em_x270_dai[] = { .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, { .name = "AC97 Aux", @@ -45,7 +45,7 @@ static struct snd_soc_dai_link em_x270_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, }; diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 97167048572d..040db695270e 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -93,7 +93,7 @@ static struct snd_soc_dai_link palm27x_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .platform_name = "pxa-pcm-audio", .init = palm27x_ac97_init, }, @@ -102,7 +102,7 @@ static struct snd_soc_dai_link palm27x_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .platform_name = "pxa-pcm-audio", }, }; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index ae9c12e1ea2a..af42417ccec5 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -189,7 +189,7 @@ static struct snd_soc_dai_link tosa_dai[] = { .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .ops = &tosa_ops, }, { @@ -198,7 +198,7 @@ static struct snd_soc_dai_link tosa_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .ops = &tosa_ops, }, }; diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 864a3345972e..aa12be76d15a 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -53,7 +53,7 @@ static struct snd_soc_dai_link tegra_wm9712_dai = { .name = "AC97 HiFi", .stream_name = "AC97 HiFi", .codec_dai_name = "wm9712-hifi", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .init = tegra_wm9712_init, }; @@ -83,7 +83,7 @@ static int tegra_wm9712_driver_probe(struct platform_device *pdev) card->dev = &pdev->dev; snd_soc_card_set_drvdata(card, machine); - machine->codec = platform_device_alloc("wm9712-codec", -1); + machine->codec = platform_device_alloc("wm9712-component", -1); if (!machine->codec) { dev_err(&pdev->dev, "Can't allocate wm9712 platform device\n"); return -ENOMEM; -- 2.14.4

3 4

[alsa-devel] [PATCH] ALSA: hda/i915: Allow delayed i915 audio component binding
by Takashi Iwai 22 Jul '18

22 Jul '18

Currently HD-audio i915 audio binding doesn't support any delayed binding, and supposes that the i915 driver registers the component immediately. This has been OK, so far, but the work-in-progress change in i915 may introduce the asynchronous binding, which effectively delays the component registration. For addressing it, implement a completion to be synced with the master binding. The timeout is set to 10 seconds which should be long enough and hopefully be not too annoying if anyone boots up a debugging session with i915 KMS turned off. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is applied on top of the current for-next or topic/drm_audio_component branch of sound git tree. sound/hda/hdac_i915.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 8f2aa8bc1185..b5282cbbe489 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -20,6 +20,8 @@ #include <sound/hda_i915.h> #include <sound/hda_register.h> +static struct completion bind_complete; + #define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ @@ -97,6 +99,19 @@ static bool i915_gfx_present(void) return pci_dev_present(ids); } +static int i915_master_bind(struct device *dev, + struct drm_audio_component *acomp) +{ + complete_all(&bind_complete); + /* clear audio_ops here as it was needed only for completion call */ + acomp->audio_ops = NULL; + return 0; +} + +static const struct drm_audio_component_audio_ops i915_init_ops = { + .master_bind = i915_master_bind +}; + /** * snd_hdac_i915_init - Initialize i915 audio component * @bus: HDA core bus @@ -117,7 +132,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (!i915_gfx_present()) return -ENODEV; - err = snd_hdac_acomp_init(bus, NULL, + init_completion(&bind_complete); + + err = snd_hdac_acomp_init(bus, &i915_init_ops, i915_component_master_match, sizeof(struct i915_audio_component) - sizeof(*acomp)); if (err < 0) @@ -125,8 +142,11 @@ int snd_hdac_i915_init(struct hdac_bus *bus) acomp = bus->audio_component; if (!acomp) return -ENODEV; - if (!acomp->ops) + if (!acomp->ops) { request_module("i915"); + /* 10s timeout */ + wait_for_completion_timeout(&bind_complete, 10 * 1000); + } if (!acomp->ops) { snd_hdac_acomp_exit(bus); return -ENODEV; -- 2.18.0

1 0

[alsa-devel] [PATCH] ASoC: da7219: Allow pdata to specify a VDDIO
by Daniel Kurtz 21 Jul '18

21 Jul '18

Some systems do not have software controllable regulators driving the DA7219's supplies, nor can they use device tree to create "always-on fixed regulators" to easily pretend like they do. On these systems the call to devm_regulator_bulk_get() just creates a set of dummy registers. Calling regulator_get_voltage() on a dummy regulator just returns -EINVAL, in which case the DA7219 is always set up to use the default VDDIO voltage range of 2.5-3.6V. Provide a new device property to let such systems specify a different VDDIO if needed (e.g., 1.8V). Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> --- This patch tries to solve the same problem as in the following patches, but does so in a way that doesn't require registering fixed regulators in the audio machine-driver: https://patchwork.kernel.org/patch/10531097 & 10531099 https://patchwork.kernel.org/patch/10536023/ include/sound/da7219.h | 2 ++ sound/soc/codecs/da7219.c | 19 +++++++++++++------ 2 files changed, 15 insertions(+), 6 deletions(-) diff --git a/include/sound/da7219.h b/include/sound/da7219.h index 1bfcb16f2d10ab..16ab125ad4adbf 100644 --- a/include/sound/da7219.h +++ b/include/sound/da7219.h @@ -38,6 +38,8 @@ struct da7219_pdata { const char *dai_clks_name; + u32 vddio; + /* Mic */ enum da7219_micbias_voltage micbias_lvl; enum da7219_mic_amp_in_sel mic_amp_in_sel; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 980a6a8bf56d38..d8f202c57f958d 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1634,6 +1634,9 @@ static struct da7219_pdata *da7219_fw_to_pdata(struct snd_soc_component *compone else pdata->mic_amp_in_sel = DA7219_MIC_AMP_IN_SEL_DIFF; + if (!device_property_read_string(dev, "dlg,vddio", &of_val32)) + pdata->vddio = of_val32; + return pdata; } @@ -1717,8 +1720,12 @@ static int da7219_handle_supplies(struct snd_soc_component *component) /* Determine VDDIO voltage provided */ vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer; ret = regulator_get_voltage(vddio); + /* If regulator_get_voltage() fails, try to use vddio from pdata. */ + if (ret < 0 && da7219->pdata) + ret = da7219->pdata->vddio; if (ret < 1200000) - dev_warn(component->dev, "Invalid VDDIO voltage\n"); + dev_warn(component->dev, "Invalid VDDIO voltage: %d mV\n", + ret); else if (ret < 2800000) io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V; @@ -1872,6 +1879,11 @@ static int da7219_probe(struct snd_soc_component *component) mutex_init(&da7219->ctrl_lock); mutex_init(&da7219->pll_lock); + /* Handle DT/ACPI/Platform data */ + da7219->pdata = dev_get_platdata(component->dev); + if (!da7219->pdata) + da7219->pdata = da7219_fw_to_pdata(component); + /* Regulator configuration */ ret = da7219_handle_supplies(component); if (ret) @@ -1897,11 +1909,6 @@ static int da7219_probe(struct snd_soc_component *component) break; } - /* Handle DT/ACPI/Platform data */ - da7219->pdata = dev_get_platdata(component->dev); - if (!da7219->pdata) - da7219->pdata = da7219_fw_to_pdata(component); - da7219_handle_pdata(component); /* Check if MCLK provided */ -- 2.18.0.233.g985f88cf7e-goog

2 3

[alsa-devel] [asoc:for-4.19 199/204] sound/soc/meson/axg-tdm-interface.c:225:8: error: implicit declaration of function 'clk_set_duty_cycle'
by kbuild test robot 21 Jul '18

21 Jul '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.19 head: a8e43c21a8a32a3af4abc605b6ebcab039f28e00 commit: c41c2a355b86368608377eaf3df442ec0f342f1e [199/204] ASoC: meson: add tdm output driver config: openrisc-allmodconfig (attached as .config) compiler: or1k-linux-gcc (GCC) 6.0.0 20160327 (experimental) reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout c41c2a355b86368608377eaf3df442ec0f342f1e # save the attached .config to linux build tree make.cross ARCH=openrisc All errors (new ones prefixed by >>): sound/soc/meson/axg-tdm-interface.c: In function 'axg_tdm_iface_set_lrclk': >> sound/soc/meson/axg-tdm-interface.c:225:8: error: implicit declaration of function 'clk_set_duty_cycle' [-Werror=implicit-function-declaration] ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); ^~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/clk_set_duty_cycle +225 sound/soc/meson/axg-tdm-interface.c d60e4f1e Jerome Brunet 2018-07-17 188 d60e4f1e Jerome Brunet 2018-07-17 189 static int axg_tdm_iface_set_lrclk(struct snd_soc_dai *dai, d60e4f1e Jerome Brunet 2018-07-17 190 struct snd_pcm_hw_params *params) d60e4f1e Jerome Brunet 2018-07-17 191 { d60e4f1e Jerome Brunet 2018-07-17 192 struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); d60e4f1e Jerome Brunet 2018-07-17 193 unsigned int ratio_num; d60e4f1e Jerome Brunet 2018-07-17 194 int ret; d60e4f1e Jerome Brunet 2018-07-17 195 d60e4f1e Jerome Brunet 2018-07-17 196 ret = clk_set_rate(iface->lrclk, params_rate(params)); d60e4f1e Jerome Brunet 2018-07-17 197 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 198 dev_err(dai->dev, "setting sample clock failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 199 return ret; d60e4f1e Jerome Brunet 2018-07-17 200 } d60e4f1e Jerome Brunet 2018-07-17 201 d60e4f1e Jerome Brunet 2018-07-17 202 switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { d60e4f1e Jerome Brunet 2018-07-17 203 case SND_SOC_DAIFMT_I2S: d60e4f1e Jerome Brunet 2018-07-17 204 case SND_SOC_DAIFMT_LEFT_J: d60e4f1e Jerome Brunet 2018-07-17 205 case SND_SOC_DAIFMT_RIGHT_J: d60e4f1e Jerome Brunet 2018-07-17 206 /* 50% duty cycle ratio */ d60e4f1e Jerome Brunet 2018-07-17 207 ratio_num = 1; d60e4f1e Jerome Brunet 2018-07-17 208 break; d60e4f1e Jerome Brunet 2018-07-17 209 d60e4f1e Jerome Brunet 2018-07-17 210 case SND_SOC_DAIFMT_DSP_A: d60e4f1e Jerome Brunet 2018-07-17 211 case SND_SOC_DAIFMT_DSP_B: d60e4f1e Jerome Brunet 2018-07-17 212 /* d60e4f1e Jerome Brunet 2018-07-17 213 * A zero duty cycle ratio will result in setting the mininum d60e4f1e Jerome Brunet 2018-07-17 214 * ratio possible which, for this clock, is 1 cycle of the d60e4f1e Jerome Brunet 2018-07-17 215 * parent bclk clock high and the rest low, This is exactly d60e4f1e Jerome Brunet 2018-07-17 216 * what we want here. d60e4f1e Jerome Brunet 2018-07-17 217 */ d60e4f1e Jerome Brunet 2018-07-17 218 ratio_num = 0; d60e4f1e Jerome Brunet 2018-07-17 219 break; d60e4f1e Jerome Brunet 2018-07-17 220 d60e4f1e Jerome Brunet 2018-07-17 221 default: d60e4f1e Jerome Brunet 2018-07-17 222 return -EINVAL; d60e4f1e Jerome Brunet 2018-07-17 223 } d60e4f1e Jerome Brunet 2018-07-17 224 d60e4f1e Jerome Brunet 2018-07-17 @225 ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); d60e4f1e Jerome Brunet 2018-07-17 226 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 227 dev_err(dai->dev, d60e4f1e Jerome Brunet 2018-07-17 228 "setting sample clock duty cycle failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 229 return ret; d60e4f1e Jerome Brunet 2018-07-17 230 } d60e4f1e Jerome Brunet 2018-07-17 231 d60e4f1e Jerome Brunet 2018-07-17 232 /* Set sample clock inversion */ d60e4f1e Jerome Brunet 2018-07-17 233 ret = clk_set_phase(iface->lrclk, d60e4f1e Jerome Brunet 2018-07-17 234 axg_tdm_lrclk_invert(iface->fmt) ? 180 : 0); d60e4f1e Jerome Brunet 2018-07-17 235 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 236 dev_err(dai->dev, d60e4f1e Jerome Brunet 2018-07-17 237 "setting sample clock phase failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 238 return ret; d60e4f1e Jerome Brunet 2018-07-17 239 } d60e4f1e Jerome Brunet 2018-07-17 240 d60e4f1e Jerome Brunet 2018-07-17 241 return 0; d60e4f1e Jerome Brunet 2018-07-17 242 } d60e4f1e Jerome Brunet 2018-07-17 243 :::::: The code at line 225 was first introduced by commit :::::: d60e4f1e4be5e2dfb55fb084b119aed094227a35 ASoC: meson: add tdm interface driver :::::: TO: Jerome Brunet <jbrunet(a)baylibre.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v1 0/7] arm64: dts: rockchip: add basic dts file for PX30 SoCs
by cl＠rock-chips.com 20 Jul '18

20 Jul '18

From: Liang Chen <cl(a)rock-chips.com> Liang Chen (7): Documentation: usb: dwc2: add description for px30 Documentation: rockchip-dw-mshc: add description for px30 Documentation: spi-rockchip: add description for px30 Documentation: rockchip-i2s: add description for px30 Documentation: rockchip: grf: add grf and pmugrf description for px30 arm64: dts: rockchip: add core dtsi file for PX30 SoCs arm64: dts: rockchip: add PX30 eavluation board devicetree Documentation/devicetree/bindings/arm/rockchip.txt | 4 + .../devicetree/bindings/mmc/rockchip-dw-mshc.txt | 1 + .../devicetree/bindings/soc/rockchip/grf.txt | 2 + .../devicetree/bindings/sound/rockchip-i2s.txt | 1 + .../devicetree/bindings/spi/spi-rockchip.txt | 1 + Documentation/devicetree/bindings/usb/dwc2.txt | 1 + arch/arm64/boot/dts/rockchip/Makefile | 1 + arch/arm64/boot/dts/rockchip/px30-evb.dts | 539 ++++++ arch/arm64/boot/dts/rockchip/px30.dtsi | 2044 ++++++++++++++++++++ 9 files changed, 2594 insertions(+) create mode 100644 arch/arm64/boot/dts/rockchip/px30-evb.dts create mode 100644 arch/arm64/boot/dts/rockchip/px30.dtsi -- 1.9.1

3 14

[alsa-devel] [PATCH v2] ASoC: pxa: remove clock divider and pll setup from zylonite and magician
by Daniel Mack 20 Jul '18

20 Jul '18

The SSP DAI now handles the clocking setup itself, all it needs is the master clock frequency. Remove the code from Zylonite and Magician platforms. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/magician.c | 106 +-------------------------------------- sound/soc/pxa/zylonite.c | 9 ---- 2 files changed, 2 insertions(+), 113 deletions(-) diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 2fc012b06c43..935a248e5bf6 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -90,95 +90,9 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int acps, acds, width; - unsigned int div4 = PXA_SSP_CLK_SCDB_4; + unsigned int width; int ret = 0; - width = snd_pcm_format_physical_width(params_format(params)); - - /* - * rate = SSPSCLK / (2 * width(16 or 32)) - * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1) - */ - switch (params_rate(params)) { - case 8000: - /* off by a factor of 2: bug in the PXA27x audio clock? */ - acps = 32842000; - switch (width) { - case 16: - /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_16; - break; - default: /* 32 */ - /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_8; - } - break; - case 11025: - acps = 5622000; - switch (width) { - case 16: - /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_4; - break; - default: /* 32 */ - /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - } - break; - case 22050: - acps = 5622000; - switch (width) { - case 16: - /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 44100: - acps = 5622000; - switch (width) { - case 16: - /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 48000: - acps = 12235000; - switch (width) { - case 16: - /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 96000: - default: - acps = 12235000; - switch (width) { - case 16: - /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - break; - default: /* 32 */ - /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - div4 = PXA_SSP_CLK_SCDB_1; - break; - } - break; - } - /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -191,6 +105,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + width = snd_pcm_format_physical_width(params_format(params)); ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; @@ -201,23 +116,6 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* set the SSP audio system clock ACDS divider */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_ACDS, acds); - if (ret < 0) - return ret; - - /* set the SSP audio system clock SCDB divider4 */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_SCDB, div4); - if (ret < 0) - return ret; - - /* set SSP audio pll clock */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps); - if (ret < 0) - return ret; - return 0; } diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ba468e560dd2..230eee450f45 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -83,11 +83,9 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int pll_out = 0; unsigned int wm9713_div = 0; int ret = 0; int rate = params_rate(params); - int width = snd_pcm_format_physical_width(params_format(params)); /* Only support ratios that we can generate neatly from the AC97 * based master clock - in particular, this excludes 44.1kHz. @@ -109,17 +107,10 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Add 1 to the width for the leading clock cycle */ - pll_out = rate * (width + 1) * 8; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out); - if (ret < 0) - return ret; - if (clk_pout) ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV, WM9713_PCMDIV(wm9713_div)); -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: topology: Increase SND_SOC_TPLG_NUM_TEXTS to 32
by Puneeth Prabhu 20 Jul '18

20 Jul '18

Topology supports maximum of 32 configs for a path whereas kernel can support only 16 configs of a path. So, increase SND_SOC_TPLG_NUM_TEXTS to 32 to support 32 configs for a path. Signed-off-by: Puneeth Prabhu <puneethx.prabhu(a)intel.com> Signed-off-by: Pardha Saradhi K <pardha.saradhi.kesapragada(a)intel.com> --- include/uapi/sound/asoc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index a74ca23..5b1411b 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -88,7 +88,7 @@ #define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ /* string sizes */ -#define SND_SOC_TPLG_NUM_TEXTS 16 +#define SND_SOC_TPLG_NUM_TEXTS 32 /* ABI version */ #define SND_SOC_TPLG_ABI_VERSION 0x5 /* current version */ -- 2.1.4

3 6

[alsa-devel] No controls in alsamixer
by Riccardo Casagrande 20 Jul '18

20 Jul '18

Hello all, following this thread http://mailman.alsa-project.org/pipermail/alsa-devel/2016-May/108068.html I added a new sound card without codec. On my board the i.MX6Q chip puts audio data directly to some ST FDA2100 power amplifiers. The sound card is listed correctly in alsa but in alsamixer there are no audio controls, I need a volume control. I tried to edit asound.conf file with this part pcm.softvol { type softvol slave { pcm "hw:1,0" } control { name "SoftVol" card 1 } } ctl.softvol { type hw card 1 } This is part of logs during the boot --> ITEL imx_itel_probe --> ITEL Configuro audmux --> ITEL imx-audmux-confic ext-port port=0x00000001ptcr=0x9ce73800 pdcr=0x00006000 --> ITEL imx-audmux-config int-port port=0x00000001ptcr=0x00000800 pdcr=0x00002000 --> ITEL imx-probe registro card imx-itel sound-audio-itel: snd-soc-dummy-dai <-> 202c000.ssi mapping ok --> ITEL imx-probe of ssi --> ITEL imx-probe of codec [...] ALSA device list: #0: cs42888-audio #1: imx-audio-itel #2: imx-spdif #3: imx-hdmi-soc [...] ALSA: Restoring mixer settings... No state is present for card imxaudioitel Found hardware: "imx-audio-itel" "" "" "" "" Hardware is initialized using a generic method No state is present for card imxaudioitel No state is present for card imxspdif Found hardware: "imx-spdif" "" "" "" "" Hardware is initialized using a generic method No state is present for card imxspdif Sorry but I'm new in alsa and linux in general. Riccardo

1 0

[alsa-devel] [PATCH] ASoC: sgtl5000: fix spelling in devicetree binding document
by Marcel Ziswiler 20 Jul '18

20 Jul '18

From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> This fixes a spelling mistake. Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 0f214457476f..9c58f724396a 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -17,7 +17,7 @@ Optional properties: - VDDD-supply : the regulator provider of VDDD -- micbias-resistor-k-ohms : the bias resistor to be used in kOmhs +- micbias-resistor-k-ohms : the bias resistor to be used in kOhms The resistor can take values of 2k, 4k or 8k. If set to 0 it will be off. If this node is not mentioned or if the value is unknown, then -- 2.14.4

2 1

[alsa-devel] [PATCH] MAINTAINERS: add entry for STI audio drivers
by Arnaud Pouliquen 20 Jul '18

20 Jul '18

Add sound/soc/sti drivers entry for STI audio drivers from ST Microelectronics. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- MAINTAINERS | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 192d7f7..50a7a53 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -13574,6 +13574,13 @@ L: linux-block(a)vger.kernel.org S: Maintained F: drivers/block/skd*[ch] +STI AUDIO (ASoC) DRIVERS +M: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +F: sound/soc/sti/ + STI CEC DRIVER M: Benjamin Gaignard <benjamin.gaignard(a)linaro.org> S: Maintained -- 2.7.4

2 3

[alsa-devel] [PATCH 00/11] Sound Open Firmware Core
by Liam Girdwood 20 Jul '18

20 Jul '18

Sound Open Firmware (SOF) is a host and DSP architecture agnostic audio DSP firmware. SOF is not tied to any specific host architecture or any specific physical IO communication type (it will work with on SoC DSPs, or DSP connected via SPI/I2C). SOF is also not coupled to any particular DSP architecture and has abstraction similar to Linux to allow porting to other DSP architectures. https://www.sofproject.org/ This patch series introduces the SOF core, later series will add host platform support for SOF. The SOF core manages all the core DSP services and ALSA/ASoC IO. The core is responsible for loading firmware, parsing topology, exposing PCMs and kcontrols, providing debug and trace mechanisms and performing IPC between host and DSP. The SOF core also has logic to allow reuse of existing machine drivers for other platforms/machines without any code modification. i.e. DAI links can be modified at runtime to bind with SOF and SOF topologies instead of existing hard coded DAI links and topology. Liam Girdwood (11): ASoC: SOF: Add Sound Open Firmware driver core ASoC: SOF: Add Sound Open Firmware KControl support ASoC: SOF: Add driver debug support. ASoC: SOF: Add support for IPC IO between DSP and Host ASoC: SOF: Add PCM operations support ASoC: SOF: Add support for loading topologies ASoC: SOF: Add DSP firmware trace event support ASoC: SOF: Add DSP HW abstraction operations ASoC: SOF: Add firmware loader support ASoC: SOF: Add userspace ABI support ASoC: SOF: Add Build support for SOF core include/sound/soc.h | 1 + include/sound/sof.h | 92 ++ include/uapi/sound/sof-abi.h | 29 + include/uapi/sound/sof-eq.h | 102 ++ include/uapi/sound/sof-fw.h | 69 + include/uapi/sound/sof-ipc.h | 915 +++++++++++++ include/uapi/sound/sof-tone.h | 28 + include/uapi/sound/sof-topology.h | 92 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sof/Kconfig | 38 + sound/soc/sof/Makefile | 6 + sound/soc/sof/control.c | 218 +++ sound/soc/sof/core.c | 373 ++++++ sound/soc/sof/debug.c | 147 +++ sound/soc/sof/ipc.c | 659 ++++++++++ sound/soc/sof/loader.c | 287 ++++ sound/soc/sof/ops.c | 210 +++ sound/soc/sof/ops.h | 298 +++++ sound/soc/sof/pcm.c | 641 +++++++++ sound/soc/sof/sof-priv.h | 504 +++++++ sound/soc/sof/topology.c | 2036 +++++++++++++++++++++++++++++ sound/soc/sof/trace.c | 279 ++++ 23 files changed, 7026 insertions(+) create mode 100644 include/sound/sof.h create mode 100644 include/uapi/sound/sof-abi.h create mode 100644 include/uapi/sound/sof-eq.h create mode 100644 include/uapi/sound/sof-fw.h create mode 100644 include/uapi/sound/sof-ipc.h create mode 100644 include/uapi/sound/sof-tone.h create mode 100644 include/uapi/sound/sof-topology.h create mode 100644 sound/soc/sof/Kconfig create mode 100644 sound/soc/sof/Makefile create mode 100644 sound/soc/sof/control.c create mode 100644 sound/soc/sof/core.c create mode 100644 sound/soc/sof/debug.c create mode 100644 sound/soc/sof/ipc.c create mode 100644 sound/soc/sof/loader.c create mode 100644 sound/soc/sof/ops.c create mode 100644 sound/soc/sof/ops.h create mode 100644 sound/soc/sof/pcm.c create mode 100644 sound/soc/sof/sof-priv.h create mode 100644 sound/soc/sof/topology.c create mode 100644 sound/soc/sof/trace.c -- 2.17.1

1 0

[alsa-devel] [PATCH] ASoC: nau8822: new codec driver
by David Lin 20 Jul '18

20 Jul '18

Add driver for NAU88C22. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8822.txt | 16 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8822.c | 1132 +++++++++++++++++ sound/soc/codecs/nau8822.h | 204 +++ 5 files changed, 1359 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8822.txt create mode 100644 sound/soc/codecs/nau8822.c create mode 100644 sound/soc/codecs/nau8822.h diff --git a/Documentation/devicetree/bindings/sound/nau8822.txt b/Documentation/devicetree/bindings/sound/nau8822.txt new file mode 100644 index 000000000000..a471d162d4e5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8822.txt @@ -0,0 +1,16 @@ +NAU8822 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "nuvoton,nau8822" + + - reg : the I2C address of the device. + +Example: + +codec: nau8822@1a { + compatible = "nuvoton,nau8822"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index efb095dbcd71..57365dbc1581 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -109,6 +109,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C + select SND_SOC_NAU8822 if I2C select SND_SOC_NAU8824 if I2C select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC @@ -1299,6 +1300,10 @@ config SND_SOC_NAU8810 tristate "Nuvoton Technology Corporation NAU88C10 CODEC" depends on I2C +config SND_SOC_NAU8822 + tristate "Nuvoton Technology Corporation NAU88C22 CODEC" + depends on I2C + config SND_SOC_NAU8824 tristate "Nuvoton Technology Corporation NAU88L24 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7ae7c85e8219..181a7d222fd7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -106,6 +106,7 @@ snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o +snd-soc-nau8822-objs := nau8822.o snd-soc-nau8824-objs := nau8824.o snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi-codec.o @@ -366,6 +367,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o +obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c new file mode 100644 index 000000000000..bdde5393cad9 --- /dev/null +++ b/sound/soc/codecs/nau8822.c @@ -0,0 +1,1132 @@ +/* + * nau8822.c -- NAU8822 ALSA Soc Audio Codec driver + * + * Copyright 2017 Nuvoton Technology Corp. + * + * Author: David Lin <ctlin0(a)nuvoton.com> + * Co-author: John Hsu <kchsu0(a)nuvoton.com> + * Co-author: Seven Li <wtli(a)nuvoton.com> + * + * Based on WM8974.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <asm/div64.h> +#include "nau8822.h" + +#define NAU_PLL_FREQ_MAX 100000000 +#define NAU_PLL_FREQ_MIN 90000000 +#define NAU_PLL_REF_MAX 33000000 +#define NAU_PLL_REF_MIN 8000000 +#define NAU_PLL_OPTOP_MIN 6 + +static const int nau8822_mclk_scaler[] = { 10, 15, 20, 30, 40, 60, 80, 120 }; + +static const struct reg_default nau8822_reg_defaults[] = { + { NAU8822_REG_POWER_MANAGEMENT_1, 0x0000 }, + { NAU8822_REG_POWER_MANAGEMENT_2, 0x0000 }, + { NAU8822_REG_POWER_MANAGEMENT_3, 0x0000 }, + { NAU8822_REG_AUDIO_INTERFACE, 0x0050 }, + { NAU8822_REG_COMPANDING_CONTROL, 0x0000 }, + { NAU8822_REG_CLOCKING, 0x0140 }, + { NAU8822_REG_ADDITIONAL_CONTROL, 0x0000 }, + { NAU8822_REG_GPIO_CONTROL, 0x0000 }, + { NAU8822_REG_JACK_DETECT_CONTROL_1, 0x0000 }, + { NAU8822_REG_DAC_CONTROL, 0x0000 }, + { NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_JACK_DETECT_CONTROL_2, 0x0000 }, + { NAU8822_REG_ADC_CONTROL, 0x0100 }, + { NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_EQ1, 0x012c }, + { NAU8822_REG_EQ2, 0x002c }, + { NAU8822_REG_EQ3, 0x002c }, + { NAU8822_REG_EQ4, 0x002c }, + { NAU8822_REG_EQ5, 0x002c }, + { NAU8822_REG_DAC_LIMITER_1, 0x0032 }, + { NAU8822_REG_DAC_LIMITER_2, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_1, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_2, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_3, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_4, 0x0000 }, + { NAU8822_REG_ALC_CONTROL_1, 0x0038 }, + { NAU8822_REG_ALC_CONTROL_2, 0x000b }, + { NAU8822_REG_ALC_CONTROL_3, 0x0032 }, + { NAU8822_REG_NOISE_GATE, 0x0010 }, + { NAU8822_REG_PLL_N, 0x0008 }, + { NAU8822_REG_PLL_K1, 0x000c }, + { NAU8822_REG_PLL_K2, 0x0093 }, + { NAU8822_REG_PLL_K3, 0x00e9 }, + { NAU8822_REG_3D_CONTROL, 0x0000 }, + { NAU8822_REG_RIGHT_SPEAKER_CONTROL, 0x0000 }, + { NAU8822_REG_INPUT_CONTROL, 0x0033 }, + { NAU8822_REG_LEFT_INP_PGA_CONTROL, 0x0010 }, + { NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0x0010 }, + { NAU8822_REG_LEFT_ADC_BOOST_CONTROL, 0x0100 }, + { NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0x0100 }, + { NAU8822_REG_OUTPUT_CONTROL, 0x0002 }, + { NAU8822_REG_LEFT_MIXER_CONTROL, 0x0001 }, + { NAU8822_REG_RIGHT_MIXER_CONTROL, 0x0001 }, + { NAU8822_REG_LHP_VOLUME, 0x0039 }, + { NAU8822_REG_RHP_VOLUME, 0x0039 }, + { NAU8822_REG_LSPKOUT_VOLUME, 0x0039 }, + { NAU8822_REG_RSPKOUT_VOLUME, 0x0039 }, + { NAU8822_REG_AUX2_MIXER, 0x0001 }, + { NAU8822_REG_AUX1_MIXER, 0x0001 }, + { NAU8822_REG_POWER_MANAGEMENT_4, 0x0000 }, + { NAU8822_REG_LEFT_TIME_SLOT, 0x0000 }, + { NAU8822_REG_MISC, 0x0020 }, + { NAU8822_REG_RIGHT_TIME_SLOT, 0x0000 }, + { NAU8822_REG_DEVICE_REVISION, 0x007f }, + { NAU8822_REG_DEVICE_ID, 0x001a }, + { NAU8822_REG_DAC_DITHER, 0x0114 }, + { NAU8822_REG_ALC_ENHANCE_1, 0x0000 }, + { NAU8822_REG_ALC_ENHANCE_2, 0x0000 }, + { NAU8822_REG_192KHZ_SAMPLING, 0x0008 }, + { NAU8822_REG_MISC_CONTROL, 0x0000 }, + { NAU8822_REG_INPUT_TIEOFF, 0x0000 }, + { NAU8822_REG_POWER_REDUCTION, 0x0000 }, + { NAU8822_REG_AGC_PEAK2PEAK, 0x0000 }, + { NAU8822_REG_AGC_PEAK_DETECT, 0x0000 }, + { NAU8822_REG_AUTOMUTE_CONTROL, 0x0000 }, + { NAU8822_REG_OUTPUT_TIEOFF, 0x0000 }, +}; + +static bool nau8822_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1: + case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5: + case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2: + case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4: + case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3: + case NAU8822_REG_3D_CONTROL: + case NAU8822_REG_RIGHT_SPEAKER_CONTROL: + case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL: + case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER: + case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID: + case NAU8822_REG_DAC_DITHER: + case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL: + case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF: + return true; + default: + return false; + } +} + +static bool nau8822_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1: + case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5: + case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2: + case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4: + case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3: + case NAU8822_REG_3D_CONTROL: + case NAU8822_REG_RIGHT_SPEAKER_CONTROL: + case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL: + case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER: + case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID: + case NAU8822_REG_DAC_DITHER: + case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL: + case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF: + return true; + default: + return false; + } +} + +static bool nau8822_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET: + case NAU8822_REG_DEVICE_REVISION: + case NAU8822_REG_DEVICE_ID: + case NAU8822_REG_AGC_PEAK2PEAK: + case NAU8822_REG_AGC_PEAK_DETECT: + case NAU8822_REG_AUTOMUTE_CONTROL: + return true; + default: + return false; + } +} + +/* The EQ parameters get function is to get the 5 band equalizer control. + * The regmap raw read can't work here because regmap doesn't provide + * value format for value width of 9 bits. Therefore, the driver reads data + * from cache and makes value format according to the endianness of + * bytes type control element. + */ +static int nau8822_eq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + int i, reg; + u16 reg_val, *val; + + val = (u16 *)ucontrol->value.bytes.data; + reg = NAU8822_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + reg_val = snd_soc_component_read32(component, reg + i); + /* conversion of 16-bit integers between native CPU format + * and big endian format + */ + reg_val = cpu_to_be16(reg_val); + memcpy(val + i, &reg_val, sizeof(reg_val)); + } + + return 0; +} + +/* The EQ parameters put function is to make configuration of 5 band equalizer + * control. These configuration includes central frequency, equalizer gain, + * cut-off frequency, bandwidth control, and equalizer path. + * The regmap raw write can't work here because regmap doesn't provide + * register and value format for register with address 7 bits and value 9 bits. + * Therefore, the driver makes value format according to the endianness of + * bytes type control element and writes data to codec. + */ +static int nau8822_eq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + void *data; + u16 *val, value; + int i, reg, ret; + + data = kmemdup(ucontrol->value.bytes.data, + params->max, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + val = (u16 *)data; + reg = NAU8822_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + /* conversion of 16-bit integers between native CPU format + * and big endian format + */ + value = be16_to_cpu(*(val + i)); + ret = snd_soc_component_write(component, reg + i, value); + if (ret) { + dev_err(component->dev, + "EQ configuration fail, register: %x ret: %d\n", + reg + i, ret); + kfree(data); + return ret; + } + } + kfree(data); + + return 0; +} + +static const char * const nau8822_companding[] = { + "Off", "NC", "u-law", "A-law"}; + +static const struct soc_enum nau8822_companding_adc_enum = + SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_ADCCM_SFT, + ARRAY_SIZE(nau8822_companding), nau8822_companding); + +static const struct soc_enum nau8822_companding_dac_enum = + SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_DACCM_SFT, + ARRAY_SIZE(nau8822_companding), nau8822_companding); + +static const char * const nau8822_eqmode[] = {"Capture", "Playback"}; + +static const struct soc_enum nau8822_eqmode_enum = + SOC_ENUM_SINGLE(NAU8822_REG_EQ1, NAU8822_EQM_SFT, + ARRAY_SIZE(nau8822_eqmode), nau8822_eqmode); + +static const char * const nau8822_alc1[] = {"Off", "Right", "Left", "Both"}; +static const char * const nau8822_alc3[] = {"Normal", "Limiter"}; + +static const struct soc_enum nau8822_alc_enable_enum = + SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_1, NAU8822_ALCEN_SFT, + ARRAY_SIZE(nau8822_alc1), nau8822_alc1); + +static const struct soc_enum nau8822_alc_mode_enum = + SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_3, NAU8822_ALCM_SFT, + ARRAY_SIZE(nau8822_alc3), nau8822_alc3); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1); +static const DECLARE_TLV_DB_SCALE(limiter_tlv, 0, 100, 0); + +static const struct snd_kcontrol_new nau8822_snd_controls[] = { + SOC_ENUM("ADC Companding", nau8822_companding_adc_enum), + SOC_ENUM("DAC Companding", nau8822_companding_dac_enum), + + SOC_ENUM("EQ Function", nau8822_eqmode_enum), + SND_SOC_BYTES_EXT("EQ Parameters", 10, + nau8822_eq_get, nau8822_eq_put), + + SOC_DOUBLE("DAC Inversion Switch", + NAU8822_REG_DAC_CONTROL, 0, 1, 1, 0), + SOC_DOUBLE_R_TLV("PCM Volume", + NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", + NAU8822_REG_ADC_CONTROL, 8, 1, 0), + SOC_SINGLE("High Pass Cut Off", + NAU8822_REG_ADC_CONTROL, 4, 7, 0), + + SOC_DOUBLE("ADC Inversion Switch", + NAU8822_REG_ADC_CONTROL, 0, 1, 1, 0), + SOC_DOUBLE_R_TLV("ADC Volume", + NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv), + + SOC_SINGLE("DAC Limiter Switch", + NAU8822_REG_DAC_LIMITER_1, 8, 1, 0), + SOC_SINGLE("DAC Limiter Decay", + NAU8822_REG_DAC_LIMITER_1, 4, 15, 0), + SOC_SINGLE("DAC Limiter Attack", + NAU8822_REG_DAC_LIMITER_1, 0, 15, 0), + SOC_SINGLE("DAC Limiter Threshold", + NAU8822_REG_DAC_LIMITER_2, 4, 7, 0), + SOC_SINGLE_TLV("DAC Limiter Volume", + NAU8822_REG_DAC_LIMITER_2, 0, 12, 0, limiter_tlv), + + SOC_ENUM("ALC Mode", nau8822_alc_mode_enum), + SOC_ENUM("ALC Enable Switch", nau8822_alc_enable_enum), + SOC_SINGLE("ALC Min Gain", + NAU8822_REG_ALC_CONTROL_1, 0, 7, 0), + SOC_SINGLE("ALC Max Gain", + NAU8822_REG_ALC_CONTROL_1, 3, 7, 0), + SOC_SINGLE("ALC Hold", + NAU8822_REG_ALC_CONTROL_2, 4, 10, 0), + SOC_SINGLE("ALC Target", + NAU8822_REG_ALC_CONTROL_2, 0, 15, 0), + SOC_SINGLE("ALC Decay", + NAU8822_REG_ALC_CONTROL_3, 4, 10, 0), + SOC_SINGLE("ALC Attack", + NAU8822_REG_ALC_CONTROL_3, 0, 10, 0), + SOC_SINGLE("ALC Noise Gate Switch", + NAU8822_REG_NOISE_GATE, 3, 1, 0), + SOC_SINGLE("ALC Noise Gate Threshold", + NAU8822_REG_NOISE_GATE, 0, 7, 0), + + SOC_DOUBLE_R("PGA ZC Switch", + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, + 7, 1, 0), + SOC_DOUBLE_R_TLV("PGA Volume", + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0, 63, 0, inpga_tlv), + + SOC_DOUBLE_R("Headphone ZC Switch", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 7, 1, 0), + SOC_DOUBLE_R("Headphone Mute Switch", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 6, 1, 0), + SOC_DOUBLE_R_TLV("Headphone Volume", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 0, 63, 0, spk_tlv), + + SOC_DOUBLE_R("Speaker ZC Switch", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 7, 1, 0), + SOC_DOUBLE_R("Speaker Mute Switch", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 6, 1, 0), + SOC_DOUBLE_R_TLV("Speaker Volume", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 0, 63, 0, spk_tlv), + + SOC_DOUBLE_R("AUXOUT Mute Switch", + NAU8822_REG_AUX2_MIXER, + NAU8822_REG_AUX1_MIXER, 6, 1, 0), + + SOC_DOUBLE_R("PGA Boost (+20dB)", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 8, 1, 0), + SOC_DOUBLE_R_TLV("L2/R2 Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 4, 7, 0, boost_tlv), + SOC_DOUBLE_R_TLV("Aux Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0, 7, 0, boost_tlv), + + SOC_SINGLE("DAC 128x Oversampling Switch", + NAU8822_REG_DAC_CONTROL, 5, 1, 0), + SOC_SINGLE("ADC 128x Oversampling Switch", + NAU8822_REG_ADC_CONTROL, 5, 1, 0), +}; + +/* LMAIN and RMAIN Mixer */ +static const struct snd_kcontrol_new nau8822_left_out_mixer[] = { + SOC_DAPM_SINGLE("LINMIX Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("LAUX Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("RDAC Switch", + NAU8822_REG_OUTPUT_CONTROL, 5, 1, 0), +}; + +static const struct snd_kcontrol_new nau8822_right_out_mixer[] = { + SOC_DAPM_SINGLE("RINMIX Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("RAUX Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("RDAC Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", + NAU8822_REG_OUTPUT_CONTROL, 6, 1, 0), +}; + +/* AUX1 and AUX2 Mixer */ +static const struct snd_kcontrol_new nau8822_auxout1_mixer[] = { + SOC_DAPM_SINGLE("RDAC Switch", NAU8822_REG_AUX1_MIXER, 0, 1, 0), + SOC_DAPM_SINGLE("RMIX Switch", NAU8822_REG_AUX1_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("RINMIX Switch", NAU8822_REG_AUX1_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX1_MIXER, 3, 1, 0), + SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX1_MIXER, 4, 1, 0), +}; + +static const struct snd_kcontrol_new nau8822_auxout2_mixer[] = { + SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX2_MIXER, 0, 1, 0), + SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX2_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("LINMIX Switch", NAU8822_REG_AUX2_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("AUX1MIX Output Switch", + NAU8822_REG_AUX1_MIXER, 3, 1, 0), +}; + +/* Input PGA */ +static const struct snd_kcontrol_new nau8822_left_input_mixer[] = { + SOC_DAPM_SINGLE("L2 Switch", NAU8822_REG_INPUT_CONTROL, 2, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 0, 1, 0), +}; +static const struct snd_kcontrol_new nau8822_right_input_mixer[] = { + SOC_DAPM_SINGLE("R2 Switch", NAU8822_REG_INPUT_CONTROL, 6, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 4, 1, 0), +}; + +/* Loopback Switch */ +static const struct snd_kcontrol_new nau8822_loopback = + SOC_DAPM_SINGLE("Switch", NAU8822_REG_COMPANDING_CONTROL, + NAU8822_ADDAP_SFT, 1, 0); + +static int check_mclk_select_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(source->dapm); + unsigned int value; + + value = snd_soc_component_read32(component, NAU8822_REG_CLOCKING); + + return (value & NAU8822_CLKM_MASK); +} + +static const struct snd_soc_dapm_widget nau8822_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + NAU8822_REG_POWER_MANAGEMENT_3, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + NAU8822_REG_POWER_MANAGEMENT_3, 1, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", + NAU8822_REG_POWER_MANAGEMENT_2, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", + NAU8822_REG_POWER_MANAGEMENT_2, 1, 0), + + SOC_MIXER_ARRAY("Left Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_3, 2, 0, nau8822_left_out_mixer), + SOC_MIXER_ARRAY("Right Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_3, 3, 0, nau8822_right_out_mixer), + SOC_MIXER_ARRAY("AUX1 Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_1, 7, 0, nau8822_auxout1_mixer), + SOC_MIXER_ARRAY("AUX2 Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_1, 6, 0, nau8822_auxout2_mixer), + + SOC_MIXER_ARRAY("Left Input Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, + 2, 0, nau8822_left_input_mixer), + SOC_MIXER_ARRAY("Right Input Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, + 3, 0, nau8822_right_input_mixer), + + SND_SOC_DAPM_PGA("Left Boost Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Boost Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Capture PGA", + NAU8822_REG_LEFT_INP_PGA_CONTROL, 6, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right Capture PGA", + NAU8822_REG_RIGHT_INP_PGA_CONTROL, 6, 1, NULL, 0), + + SND_SOC_DAPM_PGA("Left Headphone Out", + NAU8822_REG_POWER_MANAGEMENT_2, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Headphone Out", + NAU8822_REG_POWER_MANAGEMENT_2, 8, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Speaker Out", + NAU8822_REG_POWER_MANAGEMENT_3, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Speaker Out", + NAU8822_REG_POWER_MANAGEMENT_3, 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("AUX1 Out", + NAU8822_REG_POWER_MANAGEMENT_3, 8, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX2 Out", + NAU8822_REG_POWER_MANAGEMENT_3, 7, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Mic Bias", + NAU8822_REG_POWER_MANAGEMENT_1, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", + NAU8822_REG_POWER_MANAGEMENT_1, 5, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0, + &nau8822_loopback), + + SND_SOC_DAPM_INPUT("LMICN"), + SND_SOC_DAPM_INPUT("LMICP"), + SND_SOC_DAPM_INPUT("RMICN"), + SND_SOC_DAPM_INPUT("RMICP"), + SND_SOC_DAPM_INPUT("LAUX"), + SND_SOC_DAPM_INPUT("RAUX"), + SND_SOC_DAPM_INPUT("L2"), + SND_SOC_DAPM_INPUT("R2"), + SND_SOC_DAPM_OUTPUT("LHP"), + SND_SOC_DAPM_OUTPUT("RHP"), + SND_SOC_DAPM_OUTPUT("LSPK"), + SND_SOC_DAPM_OUTPUT("RSPK"), + SND_SOC_DAPM_OUTPUT("AUXOUT1"), + SND_SOC_DAPM_OUTPUT("AUXOUT2"), +}; + +static const struct snd_soc_dapm_route nau8822_dapm_routes[] = { + {"Right DAC", NULL, "PLL", check_mclk_select_pll}, + {"Left DAC", NULL, "PLL", check_mclk_select_pll}, + + /* LMAIN and RMAIN Mixer */ + {"Right Output Mixer", "LDAC Switch", "Left DAC"}, + {"Right Output Mixer", "RDAC Switch", "Right DAC"}, + {"Right Output Mixer", "RAUX Switch", "RAUX"}, + {"Right Output Mixer", "RINMIX Switch", "Right Boost Mixer"}, + + {"Left Output Mixer", "LDAC Switch", "Left DAC"}, + {"Left Output Mixer", "RDAC Switch", "Right DAC"}, + {"Left Output Mixer", "LAUX Switch", "LAUX"}, + {"Left Output Mixer", "LINMIX Switch", "Left Boost Mixer"}, + + /* AUX1 and AUX2 Mixer */ + {"AUX1 Output Mixer", "RDAC Switch", "Right DAC"}, + {"AUX1 Output Mixer", "RMIX Switch", "Right Output Mixer"}, + {"AUX1 Output Mixer", "RINMIX Switch", "Right Boost Mixer"}, + {"AUX1 Output Mixer", "LDAC Switch", "Left DAC"}, + {"AUX1 Output Mixer", "LMIX Switch", "Left Output Mixer"}, + + {"AUX2 Output Mixer", "LDAC Switch", "Left DAC"}, + {"AUX2 Output Mixer", "LMIX Switch", "Left Output Mixer"}, + {"AUX2 Output Mixer", "LINMIX Switch", "Left Boost Mixer"}, + {"AUX2 Output Mixer", "AUX1MIX Output Switch", "AUX1 Output Mixer"}, + + /* Outputs */ + {"Right Headphone Out", NULL, "Right Output Mixer"}, + {"RHP", NULL, "Right Headphone Out"}, + + {"Left Headphone Out", NULL, "Left Output Mixer"}, + {"LHP", NULL, "Left Headphone Out"}, + + {"Right Speaker Out", NULL, "Right Output Mixer"}, + {"RSPK", NULL, "Right Speaker Out"}, + + {"Left Speaker Out", NULL, "Left Output Mixer"}, + {"LSPK", NULL, "Left Speaker Out"}, + + {"AUX1 Out", NULL, "AUX1 Output Mixer"}, + {"AUX2 Out", NULL, "AUX2 Output Mixer"}, + {"AUXOUT1", NULL, "AUX1 Out"}, + {"AUXOUT2", NULL, "AUX2 Out"}, + + /* Boost Mixer */ + {"Right ADC", NULL, "PLL", check_mclk_select_pll}, + {"Left ADC", NULL, "PLL", check_mclk_select_pll}, + + {"Right ADC", NULL, "Right Boost Mixer"}, + + {"Right Boost Mixer", NULL, "RAUX"}, + {"Right Boost Mixer", NULL, "Right Capture PGA"}, + {"Right Boost Mixer", NULL, "R2"}, + + {"Left ADC", NULL, "Left Boost Mixer"}, + + {"Left Boost Mixer", NULL, "LAUX"}, + {"Left Boost Mixer", NULL, "Left Capture PGA"}, + {"Left Boost Mixer", NULL, "L2"}, + + /* Input PGA */ + {"Right Capture PGA", NULL, "Right Input Mixer"}, + {"Left Capture PGA", NULL, "Left Input Mixer"}, + + /* Enable Microphone Power */ + {"Right Capture PGA", NULL, "Mic Bias"}, + {"Left Capture PGA", NULL, "Mic Bias"}, + + {"Right Input Mixer", "R2 Switch", "R2"}, + {"Right Input Mixer", "MicN Switch", "RMICN"}, + {"Right Input Mixer", "MicP Switch", "RMICP"}, + + {"Left Input Mixer", "L2 Switch", "L2"}, + {"Left Input Mixer", "MicN Switch", "LMICN"}, + {"Left Input Mixer", "MicP Switch", "LMICP"}, + + /* Digital Loopback */ + {"Digital Loopback", "Switch", "Left ADC"}, + {"Digital Loopback", "Switch", "Right ADC"}, + {"Left DAC", NULL, "Digital Loopback"}, + {"Right DAC", NULL, "Digital Loopback"}, +}; + +static int nau8822_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + nau8822->div_id = clk_id; + nau8822->sysclk = freq; + dev_dbg(component->dev, "master sysclk %dHz, source %s\n", freq, + clk_id == NAU8822_CLK_PLL ? "PLL" : "MCLK"); + + return 0; +} + +static int nau8822_calc_pll(unsigned int pll_in, unsigned int fs, + struct nau8822_pll *pll_param) +{ + u64 f2, f2_max, pll_ratio; + int i, scal_sel; + + if (pll_in > NAU_PLL_REF_MAX || pll_in < NAU_PLL_REF_MIN) + return -EINVAL; + f2_max = 0; + scal_sel = ARRAY_SIZE(nau8822_mclk_scaler); + + for (i = 0; i < scal_sel; i++) { + f2 = 256 * fs * 4 * nau8822_mclk_scaler[i] / 10; + if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && + f2_max < f2) { + f2_max = f2; + scal_sel = i; + } + } + + if (ARRAY_SIZE(nau8822_mclk_scaler) == scal_sel) + return -EINVAL; + pll_param->mclk_scaler = scal_sel; + f2 = f2_max; + + /* Calculate the PLL 4-bit integer input and the PLL 24-bit fractional + * input; round up the 24+4bit. + */ + pll_ratio = div_u64(f2 << 28, pll_in); + pll_param->pre_factor = 0; + if (((pll_ratio >> 28) & 0xF) < NAU_PLL_OPTOP_MIN) { + pll_ratio <<= 1; + pll_param->pre_factor = 1; + } + pll_param->pll_int = (pll_ratio >> 28) & 0xF; + pll_param->pll_frac = ((pll_ratio & 0xFFFFFFF) >> 4); + + return 0; +} + +static int nau8822_config_clkdiv(struct snd_soc_dai *dai, int div, int rate) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + struct nau8822_pll *pll = &nau8822->pll; + int i, sclk, imclk; + + switch (nau8822->div_id) { + case NAU8822_CLK_MCLK: + /* Configure the master clock prescaler div to make system + * clock to approximate the internal master clock (IMCLK); + * and large or equal to IMCLK. + */ + div = 0; + imclk = rate * 256; + for (i = 1; i < ARRAY_SIZE(nau8822_mclk_scaler); i++) { + sclk = (nau8822->sysclk * 10) / nau8822_mclk_scaler[i]; + if (sclk < imclk) + break; + div = i; + } + dev_dbg(component->dev, "master clock prescaler %x for fs %d\n", + div, rate); + + /* master clock from MCLK and disable PLL */ + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + (div << NAU8822_MCLKSEL_SFT)); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, + NAU8822_CLKM_MCLK); + break; + + case NAU8822_CLK_PLL: + /* master clock from PLL and enable PLL */ + if (pll->mclk_scaler != div) { + dev_err(component->dev, + "master clock prescaler not meet PLL parameters\n"); + return -EINVAL; + } + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + (div << NAU8822_MCLKSEL_SFT)); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, + NAU8822_CLKM_PLL); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + struct nau8822_pll *pll_param = &nau8822->pll; + int ret, fs; + + fs = freq_out / 256; + + ret = nau8822_calc_pll(freq_in, fs, pll_param); + if (ret < 0) { + dev_err(component->dev, "Unsupported input clock %d\n", + freq_in); + return ret; + } + + dev_info(component->dev, + "pll_int=%x pll_frac=%x mclk_scaler=%x pre_factor=%x\n", + pll_param->pll_int, pll_param->pll_frac, + pll_param->mclk_scaler, pll_param->pre_factor); + + snd_soc_component_update_bits(component, + NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, + (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | + pll_param->pll_int); + snd_soc_component_write(component, + NAU8822_REG_PLL_K1, (pll_param->pll_frac >> NAU8822_PLLK1_SFT) & + NAU8822_PLLK1_MASK); + snd_soc_component_write(component, + NAU8822_REG_PLL_K2, (pll_param->pll_frac >> NAU8822_PLLK2_SFT) & + NAU8822_PLLK2_MASK); + snd_soc_component_write(component, + NAU8822_REG_PLL_K3, pll_param->pll_frac & NAU8822_PLLK3_MASK); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + + return 0; +} + +static int nau8822_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + u16 ctrl1_val = 0, ctrl2_val = 0; + + dev_dbg(component->dev, "%s\n", __func__); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl2_val &= ~1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= 0x10; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= 0x8; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= 0x18; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl1_val |= 0x180; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= 0x100; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl1_val |= 0x80; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, + NAU8822_REG_AUDIO_INTERFACE, + NAU8822_AIFMT_MASK | NAU8822_LRP_MASK | NAU8822_BCLKP_MASK, + ctrl1_val); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKIOEN_MASK, ctrl2_val); + + return 0; +} + +static int nau8822_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + int val_len = 0, val_rate = 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= NAU8822_WLEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= NAU8822_WLEN_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + val_len |= NAU8822_WLEN_32; + break; + } + + switch (params_rate(params)) { + case 8000: + val_rate |= NAU8822_SMPLR_8K; + break; + case 11025: + val_rate |= NAU8822_SMPLR_12K; + break; + case 16000: + val_rate |= NAU8822_SMPLR_16K; + break; + case 22050: + val_rate |= NAU8822_SMPLR_24K; + break; + case 32000: + val_rate |= NAU8822_SMPLR_32K; + break; + case 44100: + case 48000: + break; + } + + snd_soc_component_update_bits(component, + NAU8822_REG_AUDIO_INTERFACE, NAU8822_WLEN_MASK, val_len); + snd_soc_component_update_bits(component, + NAU8822_REG_ADDITIONAL_CONTROL, NAU8822_SMPLR_MASK, val_rate); + + /* If the master clock is from MCLK, provide the runtime FS for driver + * to get the master clock prescaler configuration. + */ + if (nau8822->div_id == NAU8822_CLK_MCLK) + nau8822_config_clkdiv(dai, 0, params_rate(params)); + + return 0; +} + +static int nau8822_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + + dev_dbg(component->dev, "%s: %d\n", __func__, mute); + + if (mute) + snd_soc_component_update_bits(component, + NAU8822_REG_DAC_CONTROL, 0x40, 0x40); + else + snd_soc_component_update_bits(component, + NAU8822_REG_DAC_CONTROL, 0x40, 0); + + return 0; +} + +static int nau8822_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_80K); + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_IOBUF_EN | NAU8822_ABIAS_EN, + NAU8822_IOBUF_EN | NAU8822_ABIAS_EN); + + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_3K); + mdelay(100); + } + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_300K); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_1, 0); + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_2, 0); + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_3, 0); + break; + } + + dev_dbg(component->dev, "%s: %d\n", __func__, level); + + return 0; +} + +#define NAU8822_RATES (SNDRV_PCM_RATE_8000_48000) + +#define NAU8822_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops nau8822_dai_ops = { + .hw_params = nau8822_hw_params, + .digital_mute = nau8822_mute, + .set_fmt = nau8822_set_dai_fmt, + .set_sysclk = nau8822_set_dai_sysclk, + .set_pll = nau8822_set_pll, +}; + +static struct snd_soc_dai_driver nau8822_dai = { + .name = "nau8822-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8822_RATES, + .formats = NAU8822_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8822_RATES, + .formats = NAU8822_FORMATS, + }, + .ops = &nau8822_dai_ops, + .symmetric_rates = 1, +}; + +static int nau8822_suspend(struct snd_soc_component *component) +{ + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); + + regcache_mark_dirty(nau8822->regmap); + + return 0; +} + +static int nau8822_resume(struct snd_soc_component *component) +{ + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + regcache_sync(nau8822->regmap); + + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* + * These registers contain an "update" bit - bit 8. This means, for example, + * that one can write new DAC digital volume for both channels, but only when + * the update bit is set, will also the volume be updated - simultaneously for + * both channels. + */ +static const int update_reg[] = { + NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, + NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, +}; + +static int nau8822_probe(struct snd_soc_component *component) +{ + int i; + + /* + * Set the update bit in all registers, that have one. This way all + * writes to those registers will also cause the update bit to be + * written. + */ + for (i = 0; i < ARRAY_SIZE(update_reg); i++) + snd_soc_component_update_bits(component, + update_reg[i], 0x100, 0x100); + + return 0; +} + +static const struct snd_soc_component_driver soc_component_dev_nau8822 = { + .probe = nau8822_probe, + .suspend = nau8822_suspend, + .resume = nau8822_resume, + .set_bias_level = nau8822_set_bias_level, + .controls = nau8822_snd_controls, + .num_controls = ARRAY_SIZE(nau8822_snd_controls), + .dapm_widgets = nau8822_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8822_dapm_widgets), + .dapm_routes = nau8822_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8822_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config nau8822_regmap_config = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = NAU8822_REG_MAX_REGISTER, + .volatile_reg = nau8822_volatile, + + .readable_reg = nau8822_readable_reg, + .writeable_reg = nau8822_writeable_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8822_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8822_reg_defaults), +}; + +static int nau8822_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8822 *nau8822 = dev_get_platdata(dev); + int ret; + + if (!nau8822) { + nau8822 = devm_kzalloc(dev, sizeof(*nau8822), GFP_KERNEL); + if (nau8822 == NULL) + return -ENOMEM; + } + i2c_set_clientdata(i2c, nau8822); + + nau8822->regmap = devm_regmap_init_i2c(i2c, &nau8822_regmap_config); + if (IS_ERR(nau8822->regmap)) { + ret = PTR_ERR(nau8822->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + nau8822->dev = dev; + + /* Reset the codec */ + ret = regmap_write(nau8822->regmap, NAU8822_REG_RESET, 0x00); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); + return ret; + } + + ret = devm_snd_soc_register_component(dev, &soc_component_dev_nau8822, + &nau8822_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct i2c_device_id nau8822_i2c_id[] = { + { "nau8822", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8822_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id nau8822_of_match[] = { + { .compatible = "nuvoton,nau8822", }, + { } +}; +MODULE_DEVICE_TABLE(of, nau8822_of_match); +#endif + +static struct i2c_driver nau8822_i2c_driver = { + .driver = { + .name = "nau8822", + .of_match_table = of_match_ptr(nau8822_of_match), + }, + .probe = nau8822_i2c_probe, + .id_table = nau8822_i2c_id, +}; +module_i2c_driver(nau8822_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8822 codec driver"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h new file mode 100644 index 000000000000..aa79c969cd44 --- /dev/null +++ b/sound/soc/codecs/nau8822.h @@ -0,0 +1,204 @@ +/* + * nau8822.h -- NAU8822 Soc Audio Codec driver + * + * Author: David Lin <ctlin0(a)nuvoton.com> + * Co-author: John Hsu <kchsu0(a)nuvoton.com> + * Co-author: Seven Li <wtli(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __NAU8822_H__ +#define __NAU8822_H__ + +#define NAU8822_REG_RESET 0x00 +#define NAU8822_REG_POWER_MANAGEMENT_1 0x01 +#define NAU8822_REG_POWER_MANAGEMENT_2 0x02 +#define NAU8822_REG_POWER_MANAGEMENT_3 0x03 +#define NAU8822_REG_AUDIO_INTERFACE 0x04 +#define NAU8822_REG_COMPANDING_CONTROL 0x05 +#define NAU8822_REG_CLOCKING 0x06 +#define NAU8822_REG_ADDITIONAL_CONTROL 0x07 +#define NAU8822_REG_GPIO_CONTROL 0x08 +#define NAU8822_REG_JACK_DETECT_CONTROL_1 0x09 +#define NAU8822_REG_DAC_CONTROL 0x0A +#define NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define NAU8822_REG_JACK_DETECT_CONTROL_2 0x0D +#define NAU8822_REG_ADC_CONTROL 0x0E +#define NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define NAU8822_REG_EQ1 0x12 +#define NAU8822_REG_EQ2 0x13 +#define NAU8822_REG_EQ3 0x14 +#define NAU8822_REG_EQ4 0x15 +#define NAU8822_REG_EQ5 0x16 +#define NAU8822_REG_DAC_LIMITER_1 0x18 +#define NAU8822_REG_DAC_LIMITER_2 0x19 +#define NAU8822_REG_NOTCH_FILTER_1 0x1B +#define NAU8822_REG_NOTCH_FILTER_2 0x1C +#define NAU8822_REG_NOTCH_FILTER_3 0x1D +#define NAU8822_REG_NOTCH_FILTER_4 0x1E +#define NAU8822_REG_ALC_CONTROL_1 0x20 +#define NAU8822_REG_ALC_CONTROL_2 0x21 +#define NAU8822_REG_ALC_CONTROL_3 0x22 +#define NAU8822_REG_NOISE_GATE 0x23 +#define NAU8822_REG_PLL_N 0x24 +#define NAU8822_REG_PLL_K1 0x25 +#define NAU8822_REG_PLL_K2 0x26 +#define NAU8822_REG_PLL_K3 0x27 +#define NAU8822_REG_3D_CONTROL 0x29 +#define NAU8822_REG_RIGHT_SPEAKER_CONTROL 0x2B +#define NAU8822_REG_INPUT_CONTROL 0x2C +#define NAU8822_REG_LEFT_INP_PGA_CONTROL 0x2D +#define NAU8822_REG_RIGHT_INP_PGA_CONTROL 0x2E +#define NAU8822_REG_LEFT_ADC_BOOST_CONTROL 0x2F +#define NAU8822_REG_RIGHT_ADC_BOOST_CONTROL 0x30 +#define NAU8822_REG_OUTPUT_CONTROL 0x31 +#define NAU8822_REG_LEFT_MIXER_CONTROL 0x32 +#define NAU8822_REG_RIGHT_MIXER_CONTROL 0x33 +#define NAU8822_REG_LHP_VOLUME 0x34 +#define NAU8822_REG_RHP_VOLUME 0x35 +#define NAU8822_REG_LSPKOUT_VOLUME 0x36 +#define NAU8822_REG_RSPKOUT_VOLUME 0x37 +#define NAU8822_REG_AUX2_MIXER 0x38 +#define NAU8822_REG_AUX1_MIXER 0x39 +#define NAU8822_REG_POWER_MANAGEMENT_4 0x3A +#define NAU8822_REG_LEFT_TIME_SLOT 0x3B +#define NAU8822_REG_MISC 0x3C +#define NAU8822_REG_RIGHT_TIME_SLOT 0x3D +#define NAU8822_REG_DEVICE_REVISION 0x3E +#define NAU8822_REG_DEVICE_ID 0x3F +#define NAU8822_REG_DAC_DITHER 0x41 +#define NAU8822_REG_ALC_ENHANCE_1 0x46 +#define NAU8822_REG_ALC_ENHANCE_2 0x47 +#define NAU8822_REG_192KHZ_SAMPLING 0x48 +#define NAU8822_REG_MISC_CONTROL 0x49 +#define NAU8822_REG_INPUT_TIEOFF 0x4A +#define NAU8822_REG_POWER_REDUCTION 0x4B +#define NAU8822_REG_AGC_PEAK2PEAK 0x4C +#define NAU8822_REG_AGC_PEAK_DETECT 0x4D +#define NAU8822_REG_AUTOMUTE_CONTROL 0x4E +#define NAU8822_REG_OUTPUT_TIEOFF 0x4F +#define NAU8822_REG_MAX_REGISTER NAU8822_REG_OUTPUT_TIEOFF + +/* NAU8822_REG_POWER_MANAGEMENT_1 (0x1) */ +#define NAU8822_REFIMP_MASK 0x3 +#define NAU8822_REFIMP_80K 0x1 +#define NAU8822_REFIMP_300K 0x2 +#define NAU8822_REFIMP_3K 0x3 +#define NAU8822_IOBUF_EN (0x1 << 2) +#define NAU8822_ABIAS_EN (0x1 << 3) + +/* NAU8822_REG_AUDIO_INTERFACE (0x4) */ +#define NAU8822_AIFMT_MASK (0x3 << 3) +#define NAU8822_WLEN_MASK (0x3 << 5) +#define NAU8822_WLEN_20 (0x1 << 5) +#define NAU8822_WLEN_24 (0x2 << 5) +#define NAU8822_WLEN_32 (0x3 << 5) +#define NAU8822_LRP_MASK (0x1 << 7) +#define NAU8822_BCLKP_MASK (0x1 << 8) + +/* NAU8822_REG_COMPANDING_CONTROL (0x5) */ +#define NAU8822_ADDAP_SFT 0 +#define NAU8822_ADCCM_SFT 1 +#define NAU8822_DACCM_SFT 3 + +/* NAU8822_REG_CLOCKING (0x6) */ +#define NAU8822_CLKIOEN_MASK 0x1 +#define NAU8822_MCLKSEL_SFT 5 +#define NAU8822_MCLKSEL_MASK (0x7 << 5) +#define NAU8822_BCLKSEL_SFT 2 +#define NAU8822_BCLKSEL_MASK (0x7 << 2) +#define NAU8822_CLKM_MASK (0x1 << 8) +#define NAU8822_CLKM_MCLK (0x0 << 8) +#define NAU8822_CLKM_PLL (0x1 << 8) + +/* NAU8822_REG_ADDITIONAL_CONTROL (0x08) */ +#define NAU8822_SMPLR_SFT 1 +#define NAU8822_SMPLR_MASK (0x7 << 1) +#define NAU8822_SMPLR_48K (0x0 << 1) +#define NAU8822_SMPLR_32K (0x1 << 1) +#define NAU8822_SMPLR_24K (0x2 << 1) +#define NAU8822_SMPLR_16K (0x3 << 1) +#define NAU8822_SMPLR_12K (0x4 << 1) +#define NAU8822_SMPLR_8K (0x5 << 1) + +/* NAU8822_REG_EQ1 (0x12) */ +#define NAU8822_EQ1GC_SFT 0 +#define NAU8822_EQ1CF_SFT 5 +#define NAU8822_EQM_SFT 8 + +/* NAU8822_REG_EQ2 (0x13) */ +#define NAU8822_EQ2GC_SFT 0 +#define NAU8822_EQ2CF_SFT 5 +#define NAU8822_EQ2BW_SFT 8 + +/* NAU8822_REG_EQ3 (0x14) */ +#define NAU8822_EQ3GC_SFT 0 +#define NAU8822_EQ3CF_SFT 5 +#define NAU8822_EQ3BW_SFT 8 + +/* NAU8822_REG_EQ4 (0x15) */ +#define NAU8822_EQ4GC_SFT 0 +#define NAU8822_EQ4CF_SFT 5 +#define NAU8822_EQ4BW_SFT 8 + +/* NAU8822_REG_EQ5 (0x16) */ +#define NAU8822_EQ5GC_SFT 0 +#define NAU8822_EQ5CF_SFT 5 + +/* NAU8822_REG_ALC_CONTROL_1 (0x20) */ +#define NAU8822_ALCMINGAIN_SFT 0 +#define NAU8822_ALCMXGAIN_SFT 3 +#define NAU8822_ALCEN_SFT 7 + +/* NAU8822_REG_ALC_CONTROL_2 (0x21) */ +#define NAU8822_ALCSL_SFT 0 +#define NAU8822_ALCHT_SFT 4 + +/* NAU8822_REG_ALC_CONTROL_3 (0x22) */ +#define NAU8822_ALCATK_SFT 0 +#define NAU8822_ALCDCY_SFT 4 +#define NAU8822_ALCM_SFT 8 + +/* NAU8822_REG_PLL_N (0x24) */ +#define NAU8822_PLLMCLK_DIV2 (0x1 << 4) +#define NAU8822_PLLN_MASK 0xF + +#define NAU8822_PLLK1_SFT 18 +#define NAU8822_PLLK1_MASK 0x3F + +/* NAU8822_REG_PLL_K2 (0x26) */ +#define NAU8822_PLLK2_SFT 9 +#define NAU8822_PLLK2_MASK 0x1FF + +/* NAU8822_REG_PLL_K3 (0x27) */ +#define NAU8822_PLLK3_MASK 0x1FF + +/* System Clock Source */ +enum { + NAU8822_CLK_MCLK, + NAU8822_CLK_PLL, +}; + +struct nau8822_pll { + int pre_factor; + int mclk_scaler; + int pll_frac; + int pll_int; +}; + +/* Codec Private Data */ +struct nau8822 { + struct device *dev; + struct regmap *regmap; + int mclk_idx; + struct nau8822_pll pll; + int sysclk; + int div_id; +}; + +#endif /* __NAU8822_H__ */ -- 2.18.0

1 0

[alsa-devel] [PATCH v2 0/3] Make the audio component binding more generic
by Takashi Iwai 20 Jul '18

20 Jul '18

Hi, this is a preliminiary patch set to convert the existing i915 / HD-audio component binding to be applicable to other drivers like radeon / amdgpu. This patchset itself doesn't change the functionality but only renames and split to a new drm_audio_component stuff from i915_audio_component. The actual usage of the new API will follow once after this one gets reviewed / accepted. The whole patches (including this patchset) are found in topic/hda-acomp branch of sound.git tree. BTW, since the whole stuff is about the audio binding, I suppose these will go through sound git tree. Let me know if anyone has concerns. Thanks! Takashi === v1->v2: * Change to SPDX for the new drm_audio_component.h * Fix remaining i915 word in drm_audio_component.h comments * Fix NULL dereference in master_bind / _unbind ops === Takashi Iwai (3): drm/i915: Split audio component to a generic type ALSA: hda/i915: Associate audio component with devres ALSA: hda: Make audio component support more generic drivers/gpu/drm/i915/Kconfig | 1 + drivers/gpu/drm/i915/intel_audio.c | 22 +- include/drm/drm_audio_component.h | 118 ++++++++++ include/drm/i915_component.h | 85 +------- include/sound/hda_component.h | 61 ++++++ include/sound/hda_i915.h | 37 +--- include/sound/hdaudio.h | 6 +- sound/hda/Kconfig | 7 +- sound/hda/Makefile | 1 + sound/hda/hdac_component.c | 335 +++++++++++++++++++++++++++++ sound/hda/hdac_i915.c | 335 ++--------------------------- sound/pci/hda/patch_hdmi.c | 57 +++-- sound/soc/codecs/hdac_hdmi.c | 10 +- 13 files changed, 607 insertions(+), 468 deletions(-) create mode 100644 include/drm/drm_audio_component.h create mode 100644 include/sound/hda_component.h create mode 100644 sound/hda/hdac_component.c -- 2.18.0

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[alsa-devel] [PATCH - ALSA JACK plugin 1/1] jack: Only allow ALSA periods multiple of JACKd period
by twischer＠de.adit-jv.com 19 Jul '18

19 Jul '18

From: Laxmi Devi <ldevi(a)de.adit-jv.com> There is a higher Xrun probability whenever the ALSA period is not a multiple of the JACKd period and the ALSA buffer is only twice the period. Allowing ALSA buffers of min. 3x period is not a good solution because this would increase the latency. As an example in case of ALSA is using a period of 11 frames and JACK a period of 12 frames and a buffer of 22 frames (2x ALSA period) is used: - The buffer is filled and contains 22 frames - JACK is reading 12 frames. Buffer is only containing 10 frames - Now, ALSA has to be scheduled exactly after the last JACK read and before the next read. Otherwise we will get an Xrun In case of ALSA is using a multiple period of JACK, JACK can always read 2 periods before an Xrun if the buffer was full. In case of ALSA is using a period of 12 frames and JACK a period of 11 frames and a buffer of 24 frames (2x ALSA period) is used: - The buffer is filled and contains 24 frames - JACK is reading 11 frames. Buffer is now containing 13 frames. But ALSA can still not write to the buffer - JACK reads 11 additional frames. Buffer contains 2 frames - Now, ALSA has to be scheduled exactly after the last JACK read and before the next read. Otherwise we will get an Xrun In case of ALSA is using a multiple period of JACK, ALSA can always write to the buffer as long as the buffer is filled with less or equal of half of buffer. Therefore this patch enforces an ALSA period which is always a multiple of the JACKd period. Signed-off-by: Laxmi Devi <ldevi(a)de.adit-jv.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index fbc9daf..b39835e 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -29,6 +29,8 @@ #include <alsa/asoundlib.h> #include <alsa/pcm_external.h> +#define MAX_PERIODS_MULTIPLE 64 + typedef enum _jack_format { SND_PCM_JACK_FORMAT_RAW } snd_pcm_jack_format_t; @@ -364,8 +366,20 @@ static int jack_set_hw_constraint(snd_pcm_jack_t *jack) }; unsigned int format = SND_PCM_FORMAT_FLOAT; unsigned int rate = jack_get_sample_rate(jack->client); + unsigned int psize_list[MAX_PERIODS_MULTIPLE]; + unsigned int nframes = jack_get_buffer_size(jack->client); + unsigned int jack_buffer_bytes = (snd_pcm_format_size(format, nframes) * + jack->channels); + unsigned int i; int err; + if (!jack_buffer_bytes) { + SNDERR("Buffer size is zero"); + return -EINVAL; + } + for (i = 1; i <= ARRAY_SIZE(psize_list); i++) + psize_list[i-1] = jack_buffer_bytes * i; + jack->sample_bits = snd_pcm_format_physical_width(format); if ((err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_ACCESS, ARRAY_SIZE(access_list), access_list)) < 0 || @@ -375,8 +389,8 @@ static int jack_set_hw_constraint(snd_pcm_jack_t *jack) jack->channels, jack->channels)) < 0 || (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_RATE, rate, rate)) < 0 || - (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, - 128, 64*1024)) < 0 || + (err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, + ARRAY_SIZE(psize_list), psize_list)) < 0 || (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_PERIODS, 2, 64)) < 0) return err; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: Intel: bytcr_rt5640: Add quirk for the "Connect Tablet 9" tablet
by Hans de Goede 19 Jul '18

19 Jul '18

Add a quirk for the "Connect Tablet 9" tablet, this tablet has a mono-speaker. Otherwise it works fine with the defaults. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 7456566c5648..718f022b3aa0 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -486,6 +486,16 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { }, .driver_data = (void *)(BYT_RT5640_DMIC1_MAP), }, + { /* Connect Tablet 9 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Connect"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "Tablet 9"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Dell Inc."), -- 2.17.1

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[alsa-devel] [PATCH 1/5] ASoC: wm_adsp: Correct algorithm list allocation size
by Charles Keepax 19 Jul '18

19 Jul '18

Commit 6396bb221514 ("treewide: kzalloc() -> kcalloc()") was overlooked when doing some refactoring to the algorithm list handling, which lead to twice as much buffer being allocated as required for reading the algorithm list. A kcalloc is no longer appropriate since the allocation size is now in bytes not registers, as such change back to kzalloc. Fixes: 7f7cca08abf4 ("ASoC: wm_adsp: Simplify handling of alg offset and length") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 07c17acc8a4f..99108d18de8d 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1906,7 +1906,7 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, /* Convert length from DSP words to bytes */ len *= sizeof(u32); - alg = kcalloc(len, 2, GFP_KERNEL | GFP_DMA); + alg = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!alg) return ERR_PTR(-ENOMEM); -- 2.11.0

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[alsa-devel] [PATCH RESEND 00/15] ASoC: meson: add axg audio subsystem support
by Jerome Brunet 19 Jul '18

19 Jul '18

This patchset adds support for the audio subsystem found on Amlogic's axg SoC family. The first SoC of this family is the A113D/X targeted at smart speaker application. The G12a SoC family, Amlogic next-gen STB chipset, will reuse this audio subsystem with a few changes. This patchset implements the DPCM machine driver and the different components it made of. Several components are still missing, such spdif input and pdm, and will submitted later on. The fancier part of this patchset is probably around the TDM. Properly modeling the SoC architecture and its capatilities made things a little more complex than usual (details in the related commit descriptions) If necessary, this patchset could broken into smaller series. For the intial submission, I wanted to give a global view of the subsystem. Hopefully, it will make it a bit easier to understand. Jerome Brunet (15): ASoC: meson: add axg fifos DT binding documentation ASoC: meson: add axg fifo base driver ASoC: meson: add axg frddr driver ASoC: meson: add axg toddr driver ASoC: meson: add axg spdif output DT bindings documentation ASoC: meson: add axg spdif output ASoC: meson: add axg tdm formatters DT bindings documentation ASoC: meson: add axg tdm interface DT bindings documentation ASoC: meson: add tdm formatter base driver ASoC: meson: add tdm interface driver ASoC: meson: add tdm output driver ASoC: meson: add tdm input driver ASoC: export snd_soc_of_get_slot_mask ASoC: meson: add axg sound card DT bindings documentation ASoC: meson: add axg sound card support .../devicetree/bindings/sound/amlogic,axg-fifo.txt | 23 + .../bindings/sound/amlogic,axg-sound-card.txt | 124 ++++ .../bindings/sound/amlogic,axg-spdifout.txt | 20 + .../bindings/sound/amlogic,axg-tdm-formatters.txt | 28 + .../bindings/sound/amlogic,axg-tdm-iface.txt | 22 + include/sound/soc.h | 3 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/meson/Kconfig | 64 ++ sound/soc/meson/Makefile | 21 + sound/soc/meson/axg-card.c | 671 +++++++++++++++++++++ sound/soc/meson/axg-fifo.c | 341 +++++++++++ sound/soc/meson/axg-fifo.h | 80 +++ sound/soc/meson/axg-frddr.c | 141 +++++ sound/soc/meson/axg-spdifout.c | 456 ++++++++++++++ sound/soc/meson/axg-tdm-formatter.c | 381 ++++++++++++ sound/soc/meson/axg-tdm-formatter.h | 39 ++ sound/soc/meson/axg-tdm-interface.c | 542 +++++++++++++++++ sound/soc/meson/axg-tdm.h | 78 +++ sound/soc/meson/axg-tdmin.c | 229 +++++++ sound/soc/meson/axg-tdmout.c | 259 ++++++++ sound/soc/meson/axg-toddr.c | 199 ++++++ sound/soc/soc-core.c | 7 +- 23 files changed, 3727 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt create mode 100644 sound/soc/meson/Kconfig create mode 100644 sound/soc/meson/Makefile create mode 100644 sound/soc/meson/axg-card.c create mode 100644 sound/soc/meson/axg-fifo.c create mode 100644 sound/soc/meson/axg-fifo.h create mode 100644 sound/soc/meson/axg-frddr.c create mode 100644 sound/soc/meson/axg-spdifout.c create mode 100644 sound/soc/meson/axg-tdm-formatter.c create mode 100644 sound/soc/meson/axg-tdm-formatter.h create mode 100644 sound/soc/meson/axg-tdm-interface.c create mode 100644 sound/soc/meson/axg-tdm.h create mode 100644 sound/soc/meson/axg-tdmin.c create mode 100644 sound/soc/meson/axg-tdmout.c create mode 100644 sound/soc/meson/axg-toddr.c -- 2.14.4

3 18

[alsa-devel] [PATCH - usb-audio v2 1/1] ALSA: usb-audio: Allow changing from a bad sample rate
by Adam Goode 19 Jul '18

19 Jul '18

If the audio device is externally clocked and set to a rate that does not match the external clock, the clock will never be valid and we cannot set the rate successfully. To fix this, allow a rate change even if the clock is initially invalid, and validate again after the rate is changed. This fixes problems with MOTU UltraLite AVB hardware over USB. Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/sound/usb/clock.c b/sound/usb/clock.c index c79749613fa6..db5e39d67a90 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -513,14 +513,28 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, bool writeable; u32 bmControls; + /* First, try to find a valid clock. This may trigger + * automatic clock selection if the current clock is not + * valid. + */ clock = snd_usb_clock_find_source(chip, fmt->protocol, fmt->clock, true); - if (clock < 0) - return clock; + if (clock < 0) { + /* We did not find a valid clock, but that might be + * because the current sample rate does not match an + * external clock source. Try again without validation + * and we will do another validation after setting the + * rate. + */ + clock = snd_usb_clock_find_source(chip, fmt->protocol, + fmt->clock, false); + if (clock < 0) + return clock; + } prev_rate = get_sample_rate_v2v3(chip, iface, fmt->altsetting, clock); if (prev_rate == rate) - return 0; + goto validation; if (fmt->protocol == UAC_VERSION_3) { struct uac3_clock_source_descriptor *cs_desc; @@ -577,6 +591,10 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, snd_usb_set_interface_quirk(dev); } +validation: + /* validate clock after rate change */ + if (!uac_clock_source_is_valid(chip, fmt->protocol, clock)) + return -ENXIO; return 0; } -- 2.17.1

2 1

[alsa-devel] [PATCH 1/1] Preventive fix in sound module
by Srikanth K H 19 Jul '18

19 Jul '18

Signed-off-by: Srikanth K H <srikanth.h(a)samsung.com> --- sound/core/timer.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index b6f076bb..c7be4f1 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1192,10 +1192,12 @@ static void snd_timer_proc_read(struct snd_info_entry *entry, break; case SNDRV_TIMER_CLASS_CARD: snd_iprintf(buffer, "C%i-%i: ", - timer->card->number, timer->tmr_device); + timer->card ? timer->card->number : -1, + timer->tmr_device); break; case SNDRV_TIMER_CLASS_PCM: - snd_iprintf(buffer, "P%i-%i-%i: ", timer->card->number, + snd_iprintf(buffer, "P%i-%i-%i: ", + timer->card ? timer->card->number : -1, timer->tmr_device, timer->tmr_subdevice); break; default: -- 1.9.1

3 6

Re: [alsa-devel] i.MX6Q no audio with wm8962 Question
by Fabio Estevam 18 Jul '18

18 Jul '18

Hi Adam, On Wed, Jul 18, 2018 at 1:07 PM, Adam Ford <aford173(a)gmail.com> wrote: > On Tue, Jul 17, 2018 at 2:40 PM Adam Ford <aford173(a)gmail.com> wrote: >> >> I have an iMX6Q with a wm8962 attached using AUDMUX4 and SSI2, but I >> am having difficulty getting the sound to generate from the headphone >> jack. > > sorry for the noise. It turns out to be a faulty pull-up resistor on > the I2C bus > > it's working Glad you fixed the problem :-)

1 0

[alsa-devel] [PATCH v2 0/5] ASoC: core: add support to card re-bind using component framework
by Srinivas Kandagatla 18 Jul '18

18 Jul '18

During discussion regarding card re-binding when components unregister and register back at https://lkml.org/lkml/2018/7/9/785 it was suggested that component framework can be added into core to provide this feature. With this new changes the card will re-bind once the dependent component re-registers after unregistering. This works based on the match done from component name using component framework. I have tested this patch with qdsp start-stop and module-load-unload usecase for more than 10000 times in loop on Qcom platforms. Thanks, srini Changes since v1: - Added new flag to enable/disable auto bind at card level, It is useful if machine driver/module is really removed/unloaded. Also to help cleanup the master component. Without this we will endup with a memory leak if the machine driver and memory associated with card are removed. - removed unregister from unbind callback. - updated match function as suggested by Vinod. - Added qdsp6 cleanup patches to this series. Srinivas Kandagatla (5): ASoC: core: add support to card re-bind using component framework ASoC: qdsp6: q6afe-dai: remove component fw related code ASoC: qdsp6: q6asm-dai: remove component framework related code ASoC: qdsp6: q6routing: remove component framework related code ASoC: qcom: apq8096: remove component framework related code include/sound/soc.h | 7 ++++ sound/soc/qcom/apq8096.c | 75 ++++------------------------------------ sound/soc/qcom/qdsp6/q6afe-dai.c | 34 +++--------------- sound/soc/qcom/qdsp6/q6asm-dai.c | 35 +++---------------- sound/soc/qcom/qdsp6/q6routing.c | 32 ++++------------- sound/soc/soc-core.c | 62 +++++++++++++++++++++++++++++++++ 6 files changed, 89 insertions(+), 156 deletions(-) -- 2.16.2

3 11

[alsa-devel] [PATCH] ASoC: hdac_hdmi: Add documentation for power management
by Sriram Periyasamy 18 Jul '18

18 Jul '18

Add documentation for power management of HDAC HDMI codec device for various scenarios such as S0/S3, probe and playback use case. Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 69 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 69 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 84f7a7a36e4b..30ccc902e5cf 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -2103,6 +2103,75 @@ static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) } #ifdef CONFIG_PM +/* + * Power management sequences + * ========================== + * + * The following explains the PM handling of HDAC HDMI with its parent + * device SKL and display power usage + * + * Probe + * ----- + * In SKL probe, + * 1. skl_probe_work() powers up the display (refcount++ -> 1) + * 2. enumerates the codecs on the link + * 3. powers down the display (refcount-- -> 0) + * + * In HDAC HDMI probe, + * 1. hdac_hdmi_dev_probe() powers up the display (refcount++ -> 1) + * 2. probe the codec + * 3. put the HDAC HDMI device to runtime suspend + * 4. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * + * Once children are runtime suspended, SKL device also goes to runtime + * suspend + * + * HDMI Playback + * ------------- + * Open HDMI device, + * 1. skl_runtime_resume() invoked + * 2. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) + * + * Close HDMI device, + * 1. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * 2. skl_runtime_suspend() invoked + * + * S0/S3 Cycle with playback in progress + * ------------------------------------- + * When the device is opened for playback, the device is runtime active + * already and the display refcount is 1 as explained above. + * + * Entering to S3, + * 1. hdmi_codec_prepare() invoke the runtime resume of codec which just + * increments the PM runtime usage count of the codec since the device + * is in use already + * 2. skl_suspend() powers down the display (refcount-- -> 0) + * + * Wakeup from S3, + * 1. skl_resume() powers up the display (refcount++ -> 1) + * 2. hdmi_codec_complete() invokes the runtime suspend of codec which just + * decrements the PM runtime usage count of the codec since the device + * is in use already + * + * Once playback is stopped, the display refcount is set to 0 as explained + * above in the HDMI playback sequence. The PM handlings are designed in + * such way that to balance the refcount of display power when the codec + * device put to S3 while playback is going on. + * + * S0/S3 Cycle without playback in progress + * ---------------------------------------- + * Entering to S3, + * 1. hdmi_codec_prepare() invoke the runtime resume of codec + * 2. skl_runtime_resume() invoked + * 3. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) + * 4. skl_suspend() powers down the display (refcount-- -> 0) + * + * Wakeup from S3, + * 1. skl_resume() powers up the display (refcount++ -> 1) + * 2. hdmi_codec_complete() invokes the runtime suspend of codec + * 3. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * 4. skl_runtime_suspend() invoked + */ static int hdac_hdmi_runtime_suspend(struct device *dev) { struct hdac_ext_device *edev = to_hda_ext_device(dev); -- 2.7.4

2 1

[alsa-devel] [PATCH 2/2] ASoC: AMD: Add an always on fixed 1.8V regulator for da7219 VDDIO
by Daniel Kurtz 18 Jul '18

18 Jul '18

From: Akshu Agrawal <akshu.agrawal(a)amd.com> DA7219's VDDIO for our platform need to be configured for 1.8V. Hence we register a fixed 1.8V voltage regulator in the machine driver. Change-Id: I65fd93e1dd37c3e0d38265b4b1492ea53b93afd4 Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> --- sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp-da7219-max98357a.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 6cbf9cf4d1a4c2..58c1dcb4d2550f 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -8,6 +8,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_ADAU7002 + select REGULATOR depends on SND_SOC_AMD_ACP && I2C help This option enables machine driver for DA7219 and MAX9835. diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index ccddc6650b9c79..6ea0f42f5b7692 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -32,6 +32,8 @@ #include <linux/clk.h> #include <linux/gpio.h> #include <linux/module.h> +#include <linux/regulator/fixed.h> +#include <linux/regulator/machine.h> #include <linux/i2c.h> #include <linux/input.h> #include <linux/acpi.h> @@ -278,11 +280,23 @@ static struct snd_soc_card cz_card = { .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static struct regulator_consumer_supply acp_da7219_supplies[] = { + REGULATOR_SUPPLY("VDDIO", "i2c-DLGS7219:00") +}; + static int cz_probe(struct platform_device *pdev) { int ret; struct snd_soc_card *card; struct acp_platform_info *machine; + struct platform_device *reg; + + reg = regulator_register_always_on(0, "fixed-1.8V", + acp_da7219_supplies, + ARRAY_SIZE(acp_da7219_supplies), + 1800000); + if (!reg) + return -ENOMEM; machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); -- 2.18.0.203.gfac676dfb9-goog

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Yet another Clevo P950 quirk entry
by Takashi Iwai 18 Jul '18

18 Jul '18

The PCI SSID 1558:95e1 needs the same quirk for other Clevo P950 models, too. Otherwise no sound comes out of speakers. Bugzilla: https://bugzilla.opensuse.org/show_bug.cgi?id=1101143 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b8a21ff8e68c..f6af3e1c2b93 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2366,6 +2366,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK(0x1558, 0x9501, "Clevo P950HR", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1558, 0x95e1, "Clevo P95xER", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x95e2, "Clevo P950ER", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), -- 2.18.0

1 0

[alsa-devel] [PATCH] ALSA: pcm: Nuke snd_pcm_lib_mmap_vmalloc()
by Takashi Iwai 18 Jul '18

18 Jul '18

snd_pcm_lib_mmap_vmalloc() was supposed to be implemented with somewhat special for vmalloc handling, but in the end, this turned to just the default handler, i.e. NULL. As the situation has never changed over decades, let's rip it off. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/staging/most/sound/sound.c | 1 - include/sound/pcm.h | 2 -- sound/drivers/aloop.c | 1 - sound/drivers/vx/vx_pcm.c | 2 -- sound/firewire/bebob/bebob_pcm.c | 1 - sound/firewire/dice/dice-pcm.c | 2 -- sound/firewire/digi00x/digi00x-pcm.c | 1 - sound/firewire/fireface/ff-pcm.c | 1 - sound/firewire/fireworks/fireworks_pcm.c | 1 - sound/firewire/isight.c | 1 - sound/firewire/motu/motu-pcm.c | 2 -- sound/firewire/oxfw/oxfw-pcm.c | 2 -- sound/firewire/tascam/tascam-pcm.c | 1 - sound/mips/sgio2audio.c | 3 --- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 1 - sound/soc/codecs/rt5514-spi.c | 1 - sound/usb/6fire/pcm.c | 1 - sound/usb/caiaq/audio.c | 1 - sound/usb/hiface/pcm.c | 1 - sound/usb/misc/ua101.c | 2 -- sound/usb/pcm.c | 2 -- 21 files changed, 30 deletions(-) diff --git a/drivers/staging/most/sound/sound.c b/drivers/staging/most/sound/sound.c index 04c18323c2ea..89b02fc305b8 100644 --- a/drivers/staging/most/sound/sound.c +++ b/drivers/staging/most/sound/sound.c @@ -457,7 +457,6 @@ static const struct snd_pcm_ops pcm_ops = { .trigger = pcm_trigger, .pointer = pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static int split_arg_list(char *buf, char **card_name, u16 *ch_num, diff --git a/include/sound/pcm.h b/include/sound/pcm.h index fcdf358a25f0..1206045ccf03 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1342,8 +1342,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s #define snd_pcm_lib_mmap_iomem NULL #endif -#define snd_pcm_lib_mmap_vmalloc NULL - /** * snd_pcm_limit_isa_dma_size - Get the max size fitting with ISA DMA transfer * @dma: DMA number diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 78a2fdc38531..1e34e6381baa 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -778,7 +778,6 @@ static const struct snd_pcm_ops loopback_pcm_ops = { .trigger = loopback_trigger, .pointer = loopback_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static int loopback_pcm_new(struct loopback *loopback, diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index 380a028469c4..ba80f459bdc5 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -883,7 +883,6 @@ static const struct snd_pcm_ops vx_pcm_playback_ops = { .trigger = vx_pcm_trigger, .pointer = vx_pcm_playback_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; @@ -1105,7 +1104,6 @@ static const struct snd_pcm_ops vx_pcm_capture_ops = { .trigger = vx_pcm_trigger, .pointer = vx_pcm_capture_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index e6adab3ef42e..ea9b86450580 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -373,7 +373,6 @@ int snd_bebob_create_pcm_devices(struct snd_bebob *bebob) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 80351b29fe0d..bb3ef5ff3488 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -412,7 +412,6 @@ int snd_dice_create_pcm(struct snd_dice *dice) .pointer = capture_pointer, .ack = capture_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops playback_ops = { .open = pcm_open, @@ -425,7 +424,6 @@ int snd_dice_create_pcm(struct snd_dice *dice) .pointer = playback_pointer, .ack = playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; unsigned int capture, playback; diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c index 796f4b4645f5..fdcff0460c53 100644 --- a/sound/firewire/digi00x/digi00x-pcm.c +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -352,7 +352,6 @@ int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index e3c16308363d..bf47f9ec8703 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -383,7 +383,6 @@ int snd_ff_create_pcm_devices(struct snd_ff *ff) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index 40faed5e6968..aed566d82726 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -397,7 +397,6 @@ int snd_efw_create_pcm_devices(struct snd_efw *efw) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 3919e186a30b..30957477e005 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -454,7 +454,6 @@ static int isight_create_pcm(struct isight *isight) .trigger = isight_trigger, .pointer = isight_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index 4330220890e8..ab69d7e6ac05 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -363,7 +363,6 @@ int snd_motu_create_pcm_devices(struct snd_motu *motu) .pointer = capture_pointer, .ack = capture_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops playback_ops = { .open = pcm_open, @@ -376,7 +375,6 @@ int snd_motu_create_pcm_devices(struct snd_motu *motu) .pointer = playback_pointer, .ack = playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 3dd46285c0e2..b3f6503dd34d 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -389,7 +389,6 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw) .pointer = pcm_capture_pointer, .ack = pcm_capture_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops playback_ops = { .open = pcm_open, @@ -402,7 +401,6 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; unsigned int cap = 0; diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index 6ec8ec634d4d..e4cc8990e195 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -279,7 +279,6 @@ int snd_tscm_create_pcm_devices(struct snd_tscm *tscm) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 9fb68b35de5a..3ec9391a4736 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -685,7 +685,6 @@ static const struct snd_pcm_ops snd_sgio2audio_playback1_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = { @@ -698,7 +697,6 @@ static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops snd_sgio2audio_capture_ops = { @@ -711,7 +709,6 @@ static const struct snd_pcm_ops snd_sgio2audio_capture_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; /* diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 4a2354b5ae00..98a6863e933c 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -276,7 +276,6 @@ static const struct snd_pcm_ops pdacf_pcm_capture_ops = { .trigger = pdacf_pcm_trigger, .pointer = pdacf_pcm_capture_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 18686ffb0cd5..6478d10c4f4a 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -268,7 +268,6 @@ static const struct snd_pcm_ops rt5514_spi_pcm_ops = { .hw_params = rt5514_spi_hw_params, .hw_free = rt5514_spi_hw_free, .pointer = rt5514_spi_pcm_pointer, - .mmap = snd_pcm_lib_mmap_vmalloc, .page = snd_pcm_lib_get_vmalloc_page, }; diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 2dd2518a71d3..f8ef3e2a8ca0 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -565,7 +565,6 @@ static const struct snd_pcm_ops pcm_ops = { .trigger = usb6fire_pcm_trigger, .pointer = usb6fire_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static void usb6fire_pcm_init_urb(struct pcm_urb *urb, diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 15344d39a6cd..c6108a3d7f8f 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -348,7 +348,6 @@ static const struct snd_pcm_ops snd_usb_caiaq_ops = { .trigger = snd_usb_caiaq_pcm_trigger, .pointer = snd_usb_caiaq_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static void check_for_elapsed_periods(struct snd_usb_caiaqdev *cdev, diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 396c317115b1..e1fbb9cc9ea7 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -523,7 +523,6 @@ static const struct snd_pcm_ops pcm_ops = { .trigger = hiface_pcm_trigger, .pointer = hiface_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static int hiface_pcm_init_urb(struct pcm_urb *urb, diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 386fbfd5c617..a0b6d039017f 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -900,7 +900,6 @@ static const struct snd_pcm_ops capture_pcm_ops = { .trigger = capture_pcm_trigger, .pointer = capture_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops playback_pcm_ops = { @@ -913,7 +912,6 @@ static const struct snd_pcm_ops playback_pcm_ops = { .trigger = playback_pcm_trigger, .pointer = playback_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct uac_format_type_i_discrete_descriptor * diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 160f52c4871b..4b930fa47277 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1694,7 +1694,6 @@ static const struct snd_pcm_ops snd_usb_playback_ops = { .trigger = snd_usb_substream_playback_trigger, .pointer = snd_usb_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops snd_usb_capture_ops = { @@ -1707,7 +1706,6 @@ static const struct snd_pcm_ops snd_usb_capture_ops = { .trigger = snd_usb_substream_capture_trigger, .pointer = snd_usb_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops snd_usb_playback_dev_ops = { -- 2.18.0

2 1

[alsa-devel] [PATCH 0/4] ALSA: rawmidi cleanups
by Takashi Iwai 17 Jul '18

17 Jul '18

Hi, this is a patchset mainly as a cleanup of rawmidi code, and most of them are just coding style or readability fixes without functional behavior changes. The last patch replaces kmalloc() with kvmalloc() for buffer allocations, which may fail frequently as is. Takashi === Takashi Iwai (4): ALSA: rawmidi: Tidy up coding styles ALSA: rawmidi: Simplify error paths ALSA: rawmidi: Minor code refactoring ALSA: rawmidi: Use kvmalloc() for buffers sound/core/rawmidi.c | 248 ++++++++++++++++++++++--------------------- 1 file changed, 128 insertions(+), 120 deletions(-) -- 2.18.0

1 4

[alsa-devel] i.MX6Q no audio with wm8962 Question
by Adam Ford 17 Jul '18

17 Jul '18

I have an iMX6Q with a wm8962 attached using AUDMUX4 and SSI2, but I am having difficulty getting the sound to generate from the headphone jack. I have verified the power is working, and we're getting the 24MHz clock signal from CK0 to the codec, and we've played with the alsamixer to un-mute a variety of options, yet we're still unable to get sound from the headphone port. sound { compatible = "fsl,imx-audio-wm8962"; model = "wm8962-audio"; ssi-controller = <&ssi2>; audio-codec = <&codec>; audio-routing = "Headphone Jack", "HPOUTL", "Headphone Jack", "HPOUTR", "Ext Spk", "SPKOUTL", "Ext Spk", "SPKOUTR", "AMIC", "MICBIAS", "IN3R", "AMIC"; mux-int-port = <2>; mux-ext-port = <4>; }; I beleive the mux-int-port points to ssi2 which should match the ssi-controller = <&ssi2> option I also beleive mux-ext-port = <4> should match the audmux 4 port. &i2c1 { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c1>; clock-frequency = <400000>; status = "okay"; codec: wm8962@1a { compatible = "wlf,wm8962"; reg = <0x1a>; clocks = <&clks IMX6QDL_CLK_CKO>; clock-names = "xclk"; DCVDD-supply = <&reg_audio>; DBVDD-supply = <&reg_audio>; AVDD-supply = <&reg_audio>; CPVDD-supply = <&reg_audio>; MICVDD-supply = <&reg_audio>; PLLVDD-supply = <&reg_audio>; SPKVDD1-supply = <&reg_audio>; SPKVDD2-supply = <&reg_audio>; gpio-cfg = < 0x0000 /* 0:Default */ 0x0000 /* 1:Default */ 0x0013 /* 2:FN_DMICCLK */ 0x0000 /* 3:Default */ 0x8014 /* 4:FN_DMICCDAT */ 0x0000 /* 5:Default */ >; }; }; The snippet of my boot log are as follows: [ 4.359592] snd-soc-dummy snd-soc-dummy: codec register snd-soc-dummy [ 4.359615] snd-soc-dummy snd-soc-dummy: ASoC: dai register snd-soc-dummy #1 [ 4.359625] snd-soc-dummy snd-soc-dummy: ASoC: dynamically register DAI snd-soc-dummy [ 4.359637] snd-soc-dummy snd-soc-dummy: ASoC: Registered DAI 'snd-soc-dummy-dai' [ 4.359744] snd-soc-dummy snd-soc-dummy: ASoC: Registered codec 'snd-soc-dummy' [ 4.359753] snd-soc-dummy snd-soc-dummy: ASoC: platform register snd-soc-dummy [ 4.359775] snd-soc-dummy snd-soc-dummy: ASoC: Registered platform 'snd-soc-dummy' [ 4.365610] wm8962 0-001a: customer id 0 revision F [ 4.382144] wm8962 0-001a: codec register 0-001a [ 4.382170] wm8962 0-001a: ASoC: dai register 0-001a #1 [ 4.382178] wm8962 0-001a: ASoC: dynamically register DAI 0-001a [ 4.382190] wm8962 0-001a: ASoC: Registered DAI 'wm8962' [ 4.382202] wm8962 0-001a: ASoC: Registered codec 'wm8962.0-001a' [ 4.383977] fsl-asoc-card sound: ASoC: binding HiFi [ 4.384007] fsl-asoc-card sound: ASoC: CPU DAI (null) not registered This seems like an issue,but it looks like it potentially gets resolved later. [ 4.386043] fsl-asrc 2034000.asrc: ASoC: dai register 2034000.asrc #1 [ 4.386092] fsl-asrc 2034000.asrc: ASoC: dynamically register DAI 2034000.asrc [ 4.386108] fsl-asrc 2034000.asrc: ASoC: Registered DAI '2034000.asrc' [ 4.386126] fsl-asrc 2034000.asrc: ASoC: platform register 2034000.asrc [ 4.386144] fsl-asrc 2034000.asrc: ASoC: Registered platform '2034000.asrc' [ 4.387439] fsl-ssi-dai 202c000.ssi: No cache defaults, reading back from HW [ 4.388567] fsl-ssi-dai 202c000.ssi: ASoC: Registered platform '202c000.ssi' [ 4.388589] fsl-ssi-dai 202c000.ssi: ASoC: dai register 202c000.ssi #1 [ 4.388597] fsl-ssi-dai 202c000.ssi: ASoC: dynamically register DAI 202c000.ssi [ 4.388610] fsl-ssi-dai 202c000.ssi: ASoC: Registered DAI '202c000.ssi' [ 4.393581] imx-wm8962 sound: ASoC: binding HiFi [ 4.396233] input: WM8962 Beep Generator as /devices/soc0/soc/2100000.aips-bus/21a0000.i2c/i2c-0/0-001a/input/input1 [ 4.442347] imx-wm8962 sound: ASoC: probe wm8962-audio dai link 0 late -2 [ 4.442380] imx-wm8962 sound: ASoC: probe wm8962-audio dai link 0 late -1 [ 4.442392] imx-wm8962 sound: ASoC: probe wm8962-audio dai link 0 late 0 [ 4.442403] imx-wm8962 sound: ASoC: probe wm8962-audio dai link 0 late 1 [ 4.442412] imx-wm8962 sound: ASoC: probe wm8962-audio dai link 0 late 2 [ 4.458756] imx-wm8962 sound: wm8962 <-> 202c000.ssi mapping ok When trying to play audio through the headphone jack, I unmute the headphones in alsamixer, but when I query the interface, it appears to remain muted. # amixer -c0 cget numid=24 numid=24,iface=MIXER,name='Headphone Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=127,step=0 : values=107,107 | dBscale-min=-121.00dB,step=1.00dB,mute=1 I have tried unmute with a simiar instruction without any luck. I also noticed that when I run speaker-test -c2 -t wav, I get the following message: [ 1467.051679] wm8962 0-001a: Failed to read DCS status: -1 Rate set to 48000Hz (requested 48000Hz) Buffer size range from 3840 to 17280 Period size range from 1920 to 1920 Using max buffer size 17280 Periods = 4 was set period_size = 1920 was set buffer_size = 17280 0 - Front Left Time per period = 1.138101 0 - Front Left ^CTransfer failed: Bad address # [ 1474.553646] wm8962 0-001a: Failed to read DSPCLK: -1 I am assuming that being unable to read the DCS -1 is a bad thing. Unfortunatly, after this instruction. we can no longer use alsamixer or the any alsa tool except for the speaker-test app. When running the speaker test a second time I get even more errors: Stream parameters are 48000Hz, S16_LE, 2 channels WAV file(s) [ 1700.659314] wm8962 0-001a: Unable to sync registers 0x0-0x3. -6 [ 1700.833594] wm8962 0-001a: Failed to read DSPCLK: -1 [ 1700.862662] wm8962 0-001a: ASoC: POST_PMU: HPOUTL PGA event failed: -6 [ 1700.869709] wm8962 0-001a: ASoC: POST_PMU: SPKOUTL PGA event failed: -6 [ 1700.879766] wm8962 0-001a: ASoC: POST_PMU: HPOUTR PGA event failed: -6 [ 1700.888565] wm8962 0-001a: ASoC: POST_PMU: SPKOUTR PGA event failed: -6 [ 1700.922471] wm8962 0-001a: Failed to read DCS status: -1 Rate set to 48000Hz (requested 48000Hz) Buffer size range from 3840 to 17280 Period size range from 1920 to 1920 Using max buffer size 17280 Periods = 4 was set period_size = 1920 was set buffer_size = 17280 0 - Front Left 1 - Front Right This worked one day, and for some reason, it seems to have stopped, and its repeatable on multiple boards, so I think it's a software configuration somewhere. Might someone have any suggestions? thank you, adam

1 0

[alsa-devel] [PATCH] ALSA: rawmidi: Change resized buffers atomically
by Takashi Iwai 17 Jul '18

17 Jul '18

The SNDRV_RAWMIDI_IOCTL_PARAMS ioctl may resize the buffers and the current code is racy. For example, the sequencer client may write to buffer while it being resized. As a simple workaround, let's switch to the resized buffer inside the stream runtime lock. Reported-by: syzbot+52f83f0ea8df16932f7f(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/rawmidi.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 69616d00481c..b53026a72e73 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -635,7 +635,7 @@ static int snd_rawmidi_info_select_user(struct snd_card *card, int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params * params) { - char *newbuf; + char *newbuf, *oldbuf; struct snd_rawmidi_runtime *runtime = substream->runtime; if (substream->append && substream->use_count > 1) @@ -648,13 +648,17 @@ int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream, return -EINVAL; } if (params->buffer_size != runtime->buffer_size) { - newbuf = krealloc(runtime->buffer, params->buffer_size, - GFP_KERNEL); + newbuf = kmalloc(params->buffer_size, GFP_KERNEL); if (!newbuf) return -ENOMEM; + spin_lock_irq(&runtime->lock); + oldbuf = runtime->buffer; runtime->buffer = newbuf; runtime->buffer_size = params->buffer_size; runtime->avail = runtime->buffer_size; + runtime->appl_ptr = runtime->hw_ptr = 0; + spin_unlock_irq(&runtime->lock); + kfree(oldbuf); } runtime->avail_min = params->avail_min; substream->active_sensing = !params->no_active_sensing; @@ -665,7 +669,7 @@ EXPORT_SYMBOL(snd_rawmidi_output_params); int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params * params) { - char *newbuf; + char *newbuf, *oldbuf; struct snd_rawmidi_runtime *runtime = substream->runtime; snd_rawmidi_drain_input(substream); @@ -676,12 +680,16 @@ int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, return -EINVAL; } if (params->buffer_size != runtime->buffer_size) { - newbuf = krealloc(runtime->buffer, params->buffer_size, - GFP_KERNEL); + newbuf = kmalloc(params->buffer_size, GFP_KERNEL); if (!newbuf) return -ENOMEM; + spin_lock_irq(&runtime->lock); + oldbuf = runtime->buffer; runtime->buffer = newbuf; runtime->buffer_size = params->buffer_size; + runtime->appl_ptr = runtime->hw_ptr = 0; + spin_unlock_irq(&runtime->lock); + kfree(oldbuf); } runtime->avail_min = params->avail_min; return 0; -- 2.18.0

1 0

[alsa-devel] [PATCH] [RFC] y2038: globally rename compat_time to old_time32
by Arnd Bergmann 17 Jul '18

17 Jul '18

Christoph Hellwig suggested a slightly different path for handling backwards compatibility with the 32-bit time_t based system calls: Rather than simply reusing the compat_sys_* entry points on 32-bit architectures unchanged, we get rid of those entry points and the compat_time types by renaming them to something that makes more sense on 32-bit architectures (which don't have a compat mode otherwise), and then share the entry points under the new name with the 64-bit architectures that use them for implementing the compatibility. The following types and interfaces are renamed here, and moved from linux/compat_time.h to linux/time32.h: old new --- --- compat_time_t old_time32_t struct compat_timeval struct old_timeval32 struct compat_timespec struct old_timespec32 struct compat_itimerspec struct old_itimerspec32 ns_to_compat_timeval() ns_to_old_timeval32() get_compat_itimerspec64() get_old_itimerspec32() put_compat_itimerspec64() put_old_itimerspec32() compat_get_timespec64() get_old_timespec32() compat_put_timespec64() put_old_timespec32() I considered doing this only for those instances that are actually shared between 32-bit and 64-bit kernels, but doing it everywhere avoids duplication. This should make more sense for the 32-bit implementation, and the naming suggests that these interfaces are meant for 'old' syscalls only, and that they always operate on 32-bit time_t on both 32-bit and 64-bit architectures, something that may not be obvious from the compat_* naming. I'm not renaming the timex, rusage and itimerval structures, as we are still debating what the new interface will look like, and whether we will need a replacement at all. This also doesn't change the names of the syscall entry points, which can be done more easily when we actually switch over the 32-bit architectures to use them, at that point we need to change COMPAT_SYSCALL_DEFINEx to SYSCALL_DEFINEx with a new name, e.g. with a _time32 suffix. Suggested-by: Christoph Hellwig <hch(a)infradead.org> Link: https://lore.kernel.org/lkml/20180705222110.GA5698@infradead.org/ Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- Both Deepa and I have pending syscall conversion patches, but if we can get consensus on the rename, they should be easy to rebase on top of this patch. The patch would be easy to split up in various ways, but since it is a trivial rename, I went for a big patch with the first attempt. --- arch/arm64/include/asm/compat.h | 6 +- arch/arm64/include/asm/stat.h | 2 +- arch/mips/include/asm/compat.h | 6 +- arch/mips/kernel/binfmt_elfn32.c | 14 ++-- arch/mips/kernel/binfmt_elfo32.c | 14 ++-- arch/parisc/include/asm/compat.h | 6 +- arch/powerpc/include/asm/compat.h | 6 +- arch/powerpc/kernel/asm-offsets.c | 8 +-- arch/powerpc/oprofile/backtrace.c | 2 +- arch/sparc/include/asm/compat.h | 6 +- drivers/media/v4l2-core/v4l2-compat-ioctl32.c | 4 +- fs/aio.c | 8 +-- fs/compat_binfmt_elf.c | 2 +- fs/compat_ioctl.c | 6 +- fs/select.c | 20 +++--- fs/timerfd.c | 12 ++-- fs/utimes.c | 12 ++-- fs/xfs/xfs_ioctl32.c | 2 +- fs/xfs/xfs_ioctl32.h | 2 +- include/linux/compat.h | 96 +++++++++++++-------------- include/linux/compat_time.h | 32 --------- include/linux/elfcore-compat.h | 8 +-- include/linux/restart_block.h | 4 +- include/linux/syscalls.h | 2 +- include/linux/time32.h | 25 +++++++ ipc/mqueue.c | 8 +-- ipc/msg.c | 6 +- ipc/sem.c | 10 +-- ipc/shm.c | 6 +- ipc/syscall.c | 2 +- ipc/util.h | 2 +- kernel/compat.c | 8 +-- kernel/futex_compat.c | 2 +- kernel/sched/core.c | 4 +- kernel/signal.c | 2 +- kernel/time/hrtimer.c | 8 +-- kernel/time/posix-stubs.c | 18 ++--- kernel/time/posix-timers.c | 30 ++++----- kernel/time/time.c | 58 ++++++++-------- net/bluetooth/hci_sock.c | 2 +- net/compat.c | 22 +++--- sound/core/pcm_compat.c | 14 ++-- sound/core/rawmidi_compat.c | 2 +- sound/core/timer_compat.c | 2 +- 44 files changed, 252 insertions(+), 259 deletions(-) delete mode 100644 include/linux/compat_time.h diff --git a/arch/arm64/include/asm/compat.h b/arch/arm64/include/asm/compat.h index 1a037b94eba1..4a89007db14a 100644 --- a/arch/arm64/include/asm/compat.h +++ b/arch/arm64/include/asm/compat.h @@ -86,11 +86,11 @@ struct compat_stat { compat_off_t st_size; compat_off_t st_blksize; compat_off_t st_blocks; - compat_time_t st_atime; + old_time32_t st_atime; compat_ulong_t st_atime_nsec; - compat_time_t st_mtime; + old_time32_t st_mtime; compat_ulong_t st_mtime_nsec; - compat_time_t st_ctime; + old_time32_t st_ctime; compat_ulong_t st_ctime_nsec; compat_ulong_t __unused4[2]; }; diff --git a/arch/arm64/include/asm/stat.h b/arch/arm64/include/asm/stat.h index eab738019707..397c6ccd04e7 100644 --- a/arch/arm64/include/asm/stat.h +++ b/arch/arm64/include/asm/stat.h @@ -20,7 +20,7 @@ #ifdef CONFIG_COMPAT -#include <linux/compat_time.h> +#include <linux/time.h> #include <asm/compat.h> /* diff --git a/arch/mips/include/asm/compat.h b/arch/mips/include/asm/compat.h index 78675f19440f..7dcbd855814e 100644 --- a/arch/mips/include/asm/compat.h +++ b/arch/mips/include/asm/compat.h @@ -59,11 +59,11 @@ struct compat_stat { s32 st_pad2[2]; compat_off_t st_size; s32 st_pad3; - compat_time_t st_atime; + old_time32_t st_atime; s32 st_atime_nsec; - compat_time_t st_mtime; + old_time32_t st_mtime; s32 st_mtime_nsec; - compat_time_t st_ctime; + old_time32_t st_ctime; s32 st_ctime_nsec; s32 st_blksize; s32 st_blocks; diff --git a/arch/mips/kernel/binfmt_elfn32.c b/arch/mips/kernel/binfmt_elfn32.c index 89b234844534..7a12763d553a 100644 --- a/arch/mips/kernel/binfmt_elfn32.c +++ b/arch/mips/kernel/binfmt_elfn32.c @@ -54,10 +54,10 @@ struct elf_prstatus32 pid_t pr_ppid; pid_t pr_pgrp; pid_t pr_sid; - struct compat_timeval pr_utime; /* User time */ - struct compat_timeval pr_stime; /* System time */ - struct compat_timeval pr_cutime;/* Cumulative user time */ - struct compat_timeval pr_cstime;/* Cumulative system time */ + struct old_timeval32 pr_utime; /* User time */ + struct old_timeval32 pr_stime; /* System time */ + struct old_timeval32 pr_cutime;/* Cumulative user time */ + struct old_timeval32 pr_cstime;/* Cumulative system time */ elf_gregset_t pr_reg; /* GP registers */ int pr_fpvalid; /* True if math co-processor being used. */ }; @@ -81,9 +81,9 @@ struct elf_prpsinfo32 #define elf_caddr_t u32 #define init_elf_binfmt init_elfn32_binfmt -#define jiffies_to_timeval jiffies_to_compat_timeval +#define jiffies_to_timeval jiffies_to_old_timeval32 static __inline__ void -jiffies_to_compat_timeval(unsigned long jiffies, struct compat_timeval *value) +jiffies_to_old_timeval32(unsigned long jiffies, struct old_timeval32 *value) { /* * Convert jiffies to nanoseconds and separate with @@ -101,6 +101,6 @@ jiffies_to_compat_timeval(unsigned long jiffies, struct compat_timeval *value) #define TASK_SIZE TASK_SIZE32 #undef ns_to_timeval -#define ns_to_timeval ns_to_compat_timeval +#define ns_to_timeval ns_to_old_timeval32 #include "../../../fs/binfmt_elf.c" diff --git a/arch/mips/kernel/binfmt_elfo32.c b/arch/mips/kernel/binfmt_elfo32.c index a88c59db3d48..e6db06a1d31a 100644 --- a/arch/mips/kernel/binfmt_elfo32.c +++ b/arch/mips/kernel/binfmt_elfo32.c @@ -59,10 +59,10 @@ struct elf_prstatus32 pid_t pr_ppid; pid_t pr_pgrp; pid_t pr_sid; - struct compat_timeval pr_utime; /* User time */ - struct compat_timeval pr_stime; /* System time */ - struct compat_timeval pr_cutime;/* Cumulative user time */ - struct compat_timeval pr_cstime;/* Cumulative system time */ + struct old_timeval32 pr_utime; /* User time */ + struct old_timeval32 pr_stime; /* System time */ + struct old_timeval32 pr_cutime;/* Cumulative user time */ + struct old_timeval32 pr_cstime;/* Cumulative system time */ elf_gregset_t pr_reg; /* GP registers */ int pr_fpvalid; /* True if math co-processor being used. */ }; @@ -86,9 +86,9 @@ struct elf_prpsinfo32 #define elf_caddr_t u32 #define init_elf_binfmt init_elf32_binfmt -#define jiffies_to_timeval jiffies_to_compat_timeval +#define jiffies_to_timeval jiffies_to_old_timeval32 static inline void -jiffies_to_compat_timeval(unsigned long jiffies, struct compat_timeval *value) +jiffies_to_old_timeval32(unsigned long jiffies, struct old_timeval32 *value) { /* * Convert jiffies to nanoseconds and separate with @@ -104,6 +104,6 @@ jiffies_to_compat_timeval(unsigned long jiffies, struct compat_timeval *value) #define TASK_SIZE TASK_SIZE32 #undef ns_to_timeval -#define ns_to_timeval ns_to_compat_timeval +#define ns_to_timeval ns_to_old_timeval32 #include "../../../fs/binfmt_elf.c" diff --git a/arch/parisc/include/asm/compat.h b/arch/parisc/include/asm/compat.h index ab8a54771507..32dadaf2b534 100644 --- a/arch/parisc/include/asm/compat.h +++ b/arch/parisc/include/asm/compat.h @@ -48,11 +48,11 @@ struct compat_stat { u16 st_reserved2; /* old st_gid */ compat_dev_t st_rdev; compat_off_t st_size; - compat_time_t st_atime; + old_time32_t st_atime; u32 st_atime_nsec; - compat_time_t st_mtime; + old_time32_t st_mtime; u32 st_mtime_nsec; - compat_time_t st_ctime; + old_time32_t st_ctime; u32 st_ctime_nsec; s32 st_blksize; s32 st_blocks; diff --git a/arch/powerpc/include/asm/compat.h b/arch/powerpc/include/asm/compat.h index 85c8af2bb272..93f79d1a03c3 100644 --- a/arch/powerpc/include/asm/compat.h +++ b/arch/powerpc/include/asm/compat.h @@ -55,11 +55,11 @@ struct compat_stat { compat_off_t st_size; compat_off_t st_blksize; compat_off_t st_blocks; - compat_time_t st_atime; + old_time32_t st_atime; u32 st_atime_nsec; - compat_time_t st_mtime; + old_time32_t st_mtime; u32 st_mtime_nsec; - compat_time_t st_ctime; + old_time32_t st_ctime; u32 st_ctime_nsec; u32 __unused4[2]; }; diff --git a/arch/powerpc/kernel/asm-offsets.c b/arch/powerpc/kernel/asm-offsets.c index 0a0544335950..83e54cad564b 100644 --- a/arch/powerpc/kernel/asm-offsets.c +++ b/arch/powerpc/kernel/asm-offsets.c @@ -387,12 +387,12 @@ int main(void) OFFSET(CFG_SYSCALL_MAP64, vdso_data, syscall_map_64); OFFSET(TVAL64_TV_SEC, timeval, tv_sec); OFFSET(TVAL64_TV_USEC, timeval, tv_usec); - OFFSET(TVAL32_TV_SEC, compat_timeval, tv_sec); - OFFSET(TVAL32_TV_USEC, compat_timeval, tv_usec); + OFFSET(TVAL32_TV_SEC, old_timeval32, tv_sec); + OFFSET(TVAL32_TV_USEC, old_timeval32, tv_usec); OFFSET(TSPC64_TV_SEC, timespec, tv_sec); OFFSET(TSPC64_TV_NSEC, timespec, tv_nsec); - OFFSET(TSPC32_TV_SEC, compat_timespec, tv_sec); - OFFSET(TSPC32_TV_NSEC, compat_timespec, tv_nsec); + OFFSET(TSPC32_TV_SEC, old_timespec32, tv_sec); + OFFSET(TSPC32_TV_NSEC, old_timespec32, tv_nsec); #else OFFSET(TVAL32_TV_SEC, timeval, tv_sec); OFFSET(TVAL32_TV_USEC, timeval, tv_usec); diff --git a/arch/powerpc/oprofile/backtrace.c b/arch/powerpc/oprofile/backtrace.c index ad054dd0d666..5df6290d1ccc 100644 --- a/arch/powerpc/oprofile/backtrace.c +++ b/arch/powerpc/oprofile/backtrace.c @@ -7,7 +7,7 @@ * 2 of the License, or (at your option) any later version. **/ -#include <linux/compat_time.h> +#include <linux/time.h> #include <linux/oprofile.h> #include <linux/sched.h> #include <asm/processor.h> diff --git a/arch/sparc/include/asm/compat.h b/arch/sparc/include/asm/compat.h index 4eb51d2dae98..7018cb60beef 100644 --- a/arch/sparc/include/asm/compat.h +++ b/arch/sparc/include/asm/compat.h @@ -47,11 +47,11 @@ struct compat_stat { __compat_gid_t st_gid; compat_dev_t st_rdev; compat_off_t st_size; - compat_time_t st_atime; + old_time32_t st_atime; compat_ulong_t st_atime_nsec; - compat_time_t st_mtime; + old_time32_t st_mtime; compat_ulong_t st_mtime_nsec; - compat_time_t st_ctime; + old_time32_t st_ctime; compat_ulong_t st_ctime_nsec; compat_off_t st_blksize; compat_off_t st_blocks; diff --git a/drivers/media/v4l2-core/v4l2-compat-ioctl32.c b/drivers/media/v4l2-core/v4l2-compat-ioctl32.c index 6481212fda77..a57cbfaf332a 100644 --- a/drivers/media/v4l2-core/v4l2-compat-ioctl32.c +++ b/drivers/media/v4l2-core/v4l2-compat-ioctl32.c @@ -468,7 +468,7 @@ struct v4l2_buffer32 { __u32 bytesused; __u32 flags; __u32 field; /* enum v4l2_field */ - struct compat_timeval timestamp; + struct old_timeval32 timestamp; struct v4l2_timecode timecode; __u32 sequence; @@ -1024,7 +1024,7 @@ struct v4l2_event32 { } u; __u32 pending; __u32 sequence; - struct compat_timespec timestamp; + struct old_timespec32 timestamp; __u32 id; __u32 reserved[8]; }; diff --git a/fs/aio.c b/fs/aio.c index 76def7523ece..19513b880a06 100644 --- a/fs/aio.c +++ b/fs/aio.c @@ -1933,12 +1933,12 @@ COMPAT_SYSCALL_DEFINE5(io_getevents, compat_aio_context_t, ctx_id, compat_long_t, min_nr, compat_long_t, nr, struct io_event __user *, events, - struct compat_timespec __user *, timeout) + struct old_timespec32 __user *, timeout) { struct timespec64 t; int ret; - if (timeout && compat_get_timespec64(&t, timeout)) + if (timeout && get_old_timespec32(&t, timeout)) return -EFAULT; ret = do_io_getevents(ctx_id, min_nr, nr, events, timeout ? &t : NULL); @@ -1958,7 +1958,7 @@ COMPAT_SYSCALL_DEFINE6(io_pgetevents, compat_long_t, min_nr, compat_long_t, nr, struct io_event __user *, events, - struct compat_timespec __user *, timeout, + struct old_timespec32 __user *, timeout, const struct __compat_aio_sigset __user *, usig) { struct __compat_aio_sigset ksig = { NULL, }; @@ -1966,7 +1966,7 @@ COMPAT_SYSCALL_DEFINE6(io_pgetevents, struct timespec64 t; int ret; - if (timeout && compat_get_timespec64(&t, timeout)) + if (timeout && get_old_timespec32(&t, timeout)) return -EFAULT; if (usig && copy_from_user(&ksig, usig, sizeof(ksig))) diff --git a/fs/compat_binfmt_elf.c b/fs/compat_binfmt_elf.c index 504b3c3539dc..15f6e96b3bd9 100644 --- a/fs/compat_binfmt_elf.c +++ b/fs/compat_binfmt_elf.c @@ -52,7 +52,7 @@ #define elf_prpsinfo compat_elf_prpsinfo #undef ns_to_timeval -#define ns_to_timeval ns_to_compat_timeval +#define ns_to_timeval ns_to_old_timeval32 /* * To use this file, asm/elf.h must define compat_elf_check_arch. diff --git a/fs/compat_ioctl.c b/fs/compat_ioctl.c index 9907475b4226..f79871246b1e 100644 --- a/fs/compat_ioctl.c +++ b/fs/compat_ioctl.c @@ -135,7 +135,7 @@ static int do_ioctl(struct file *file, unsigned int cmd, unsigned long arg) struct compat_video_event { int32_t type; - compat_time_t timestamp; + old_time32_t timestamp; union { video_size_t size; unsigned int frame_rate; @@ -449,8 +449,8 @@ struct ppp_option_data32 { #define PPPIOCSCOMPRESS32 _IOW('t', 77, struct ppp_option_data32) struct ppp_idle32 { - compat_time_t xmit_idle; - compat_time_t recv_idle; + old_time32_t xmit_idle; + old_time32_t recv_idle; }; #define PPPIOCGIDLE32 _IOR('t', 63, struct ppp_idle32) diff --git a/fs/select.c b/fs/select.c index 4a6b6e4b21cb..22b3bf89f051 100644 --- a/fs/select.c +++ b/fs/select.c @@ -1120,7 +1120,7 @@ int compat_poll_select_copy_remaining(struct timespec64 *end_time, void __user * ts.tv_sec = ts.tv_nsec = 0; if (timeval) { - struct compat_timeval rtv; + struct old_timeval32 rtv; rtv.tv_sec = ts.tv_sec; rtv.tv_usec = ts.tv_nsec / NSEC_PER_USEC; @@ -1128,7 +1128,7 @@ int compat_poll_select_copy_remaining(struct timespec64 *end_time, void __user * if (!copy_to_user(p, &rtv, sizeof(rtv))) return ret; } else { - if (!compat_put_timespec64(&ts, p)) + if (!put_old_timespec32(&ts, p)) return ret; } /* @@ -1257,10 +1257,10 @@ static int compat_core_sys_select(int n, compat_ulong_t __user *inp, static int do_compat_select(int n, compat_ulong_t __user *inp, compat_ulong_t __user *outp, compat_ulong_t __user *exp, - struct compat_timeval __user *tvp) + struct old_timeval32 __user *tvp) { struct timespec64 end_time, *to = NULL; - struct compat_timeval tv; + struct old_timeval32 tv; int ret; if (tvp) { @@ -1282,7 +1282,7 @@ static int do_compat_select(int n, compat_ulong_t __user *inp, COMPAT_SYSCALL_DEFINE5(select, int, n, compat_ulong_t __user *, inp, compat_ulong_t __user *, outp, compat_ulong_t __user *, exp, - struct compat_timeval __user *, tvp) + struct old_timeval32 __user *, tvp) { return do_compat_select(n, inp, outp, exp, tvp); } @@ -1307,7 +1307,7 @@ COMPAT_SYSCALL_DEFINE1(old_select, struct compat_sel_arg_struct __user *, arg) static long do_compat_pselect(int n, compat_ulong_t __user *inp, compat_ulong_t __user *outp, compat_ulong_t __user *exp, - struct compat_timespec __user *tsp, compat_sigset_t __user *sigmask, + struct old_timespec32 __user *tsp, compat_sigset_t __user *sigmask, compat_size_t sigsetsize) { sigset_t ksigmask, sigsaved; @@ -1315,7 +1315,7 @@ static long do_compat_pselect(int n, compat_ulong_t __user *inp, int ret; if (tsp) { - if (compat_get_timespec64(&ts, tsp)) + if (get_old_timespec32(&ts, tsp)) return -EFAULT; to = &end_time; @@ -1355,7 +1355,7 @@ static long do_compat_pselect(int n, compat_ulong_t __user *inp, COMPAT_SYSCALL_DEFINE6(pselect6, int, n, compat_ulong_t __user *, inp, compat_ulong_t __user *, outp, compat_ulong_t __user *, exp, - struct compat_timespec __user *, tsp, void __user *, sig) + struct old_timespec32 __user *, tsp, void __user *, sig) { compat_size_t sigsetsize = 0; compat_uptr_t up = 0; @@ -1373,7 +1373,7 @@ COMPAT_SYSCALL_DEFINE6(pselect6, int, n, compat_ulong_t __user *, inp, } COMPAT_SYSCALL_DEFINE5(ppoll, struct pollfd __user *, ufds, - unsigned int, nfds, struct compat_timespec __user *, tsp, + unsigned int, nfds, struct old_timespec32 __user *, tsp, const compat_sigset_t __user *, sigmask, compat_size_t, sigsetsize) { sigset_t ksigmask, sigsaved; @@ -1381,7 +1381,7 @@ COMPAT_SYSCALL_DEFINE5(ppoll, struct pollfd __user *, ufds, int ret; if (tsp) { - if (compat_get_timespec64(&ts, tsp)) + if (get_old_timespec32(&ts, tsp)) return -EFAULT; to = &end_time; diff --git a/fs/timerfd.c b/fs/timerfd.c index 38c695ce385b..cd055bce24e0 100644 --- a/fs/timerfd.c +++ b/fs/timerfd.c @@ -561,29 +561,29 @@ SYSCALL_DEFINE2(timerfd_gettime, int, ufd, struct __kernel_itimerspec __user *, #ifdef CONFIG_COMPAT_32BIT_TIME COMPAT_SYSCALL_DEFINE4(timerfd_settime, int, ufd, int, flags, - const struct compat_itimerspec __user *, utmr, - struct compat_itimerspec __user *, otmr) + const struct old_itimerspec32 __user *, utmr, + struct old_itimerspec32 __user *, otmr) { struct itimerspec64 new, old; int ret; - if (get_compat_itimerspec64(&new, utmr)) + if (get_old_itimerspec32(&new, utmr)) return -EFAULT; ret = do_timerfd_settime(ufd, flags, &new, &old); if (ret) return ret; - if (otmr && put_compat_itimerspec64(&old, otmr)) + if (otmr && put_old_itimerspec32(&old, otmr)) return -EFAULT; return ret; } COMPAT_SYSCALL_DEFINE2(timerfd_gettime, int, ufd, - struct compat_itimerspec __user *, otmr) + struct old_itimerspec32 __user *, otmr) { struct itimerspec64 kotmr; int ret = do_timerfd_gettime(ufd, &kotmr); if (ret) return ret; - return put_compat_itimerspec64(&kotmr, otmr) ? -EFAULT : 0; + return put_old_itimerspec32(&kotmr, otmr) ? -EFAULT : 0; } #endif diff --git a/fs/utimes.c b/fs/utimes.c index 69d4b6ba1bfb..79a65c163f40 100644 --- a/fs/utimes.c +++ b/fs/utimes.c @@ -245,13 +245,13 @@ COMPAT_SYSCALL_DEFINE2(utime, const char __user *, filename, return do_utimes(AT_FDCWD, filename, t ? tv : NULL, 0); } -COMPAT_SYSCALL_DEFINE4(utimensat, unsigned int, dfd, const char __user *, filename, struct compat_timespec __user *, t, int, flags) +COMPAT_SYSCALL_DEFINE4(utimensat, unsigned int, dfd, const char __user *, filename, struct old_timespec32 __user *, t, int, flags) { struct timespec64 tv[2]; if (t) { - if (compat_get_timespec64(&tv[0], &t[0]) || - compat_get_timespec64(&tv[1], &t[1])) + if (get_old_timespec32(&tv[0], &t[0]) || + get_old_timespec32(&tv[1], &t[1])) return -EFAULT; if (tv[0].tv_nsec == UTIME_OMIT && tv[1].tv_nsec == UTIME_OMIT) @@ -261,7 +261,7 @@ COMPAT_SYSCALL_DEFINE4(utimensat, unsigned int, dfd, const char __user *, filena } static long do_compat_futimesat(unsigned int dfd, const char __user *filename, - struct compat_timeval __user *t) + struct old_timeval32 __user *t) { struct timespec64 tv[2]; @@ -282,12 +282,12 @@ static long do_compat_futimesat(unsigned int dfd, const char __user *filename, COMPAT_SYSCALL_DEFINE3(futimesat, unsigned int, dfd, const char __user *, filename, - struct compat_timeval __user *, t) + struct old_timeval32 __user *, t) { return do_compat_futimesat(dfd, filename, t); } -COMPAT_SYSCALL_DEFINE2(utimes, const char __user *, filename, struct compat_timeval __user *, t) +COMPAT_SYSCALL_DEFINE2(utimes, const char __user *, filename, struct old_timeval32 __user *, t) { return do_compat_futimesat(AT_FDCWD, filename, t); } diff --git a/fs/xfs/xfs_ioctl32.c b/fs/xfs/xfs_ioctl32.c index fba115f4103a..2689d3f4c65c 100644 --- a/fs/xfs/xfs_ioctl32.c +++ b/fs/xfs/xfs_ioctl32.c @@ -115,7 +115,7 @@ xfs_ioctl32_bstime_copyin( xfs_bstime_t *bstime, compat_xfs_bstime_t __user *bstime32) { - compat_time_t sec32; /* tv_sec differs on 64 vs. 32 */ + old_time32_t sec32; /* tv_sec differs on 64 vs. 32 */ if (get_user(sec32, &bstime32->tv_sec) || get_user(bstime->tv_nsec, &bstime32->tv_nsec)) diff --git a/fs/xfs/xfs_ioctl32.h b/fs/xfs/xfs_ioctl32.h index d28fa824284a..4324b96a6d95 100644 --- a/fs/xfs/xfs_ioctl32.h +++ b/fs/xfs/xfs_ioctl32.h @@ -32,7 +32,7 @@ #endif typedef struct compat_xfs_bstime { - compat_time_t tv_sec; /* seconds */ + old_time32_t tv_sec; /* seconds */ __s32 tv_nsec; /* and nanoseconds */ } compat_xfs_bstime_t; diff --git a/include/linux/compat.h b/include/linux/compat.h index 1a3c4f37e908..0e058792ecf6 100644 --- a/include/linux/compat.h +++ b/include/linux/compat.h @@ -7,7 +7,7 @@ */ #include <linux/types.h> -#include <linux/compat_time.h> +#include <linux/time.h> #include <linux/stat.h> #include <linux/param.h> /* for HZ */ @@ -116,13 +116,13 @@ struct compat_sel_arg_struct; struct rusage; struct compat_utimbuf { - compat_time_t actime; - compat_time_t modtime; + old_time32_t actime; + old_time32_t modtime; }; struct compat_itimerval { - struct compat_timeval it_interval; - struct compat_timeval it_value; + struct old_timeval32 it_interval; + struct old_timeval32 it_value; }; struct itimerval; @@ -146,7 +146,7 @@ struct compat_timex { compat_long_t constant; compat_long_t precision; compat_long_t tolerance; - struct compat_timeval time; + struct old_timeval32 time; compat_long_t tick; compat_long_t ppsfreq; compat_long_t jitter; @@ -307,8 +307,8 @@ struct compat_rlimit { }; struct compat_rusage { - struct compat_timeval ru_utime; - struct compat_timeval ru_stime; + struct old_timeval32 ru_utime; + struct old_timeval32 ru_stime; compat_long_t ru_maxrss; compat_long_t ru_ixrss; compat_long_t ru_idrss; @@ -457,8 +457,8 @@ int copy_siginfo_to_user32(struct compat_siginfo __user *to, const siginfo_t *fr int get_compat_sigevent(struct sigevent *event, const struct compat_sigevent __user *u_event); -static inline int compat_timeval_compare(struct compat_timeval *lhs, - struct compat_timeval *rhs) +static inline int old_timeval32_compare(struct old_timeval32 *lhs, + struct old_timeval32 *rhs) { if (lhs->tv_sec < rhs->tv_sec) return -1; @@ -467,8 +467,8 @@ static inline int compat_timeval_compare(struct compat_timeval *lhs, return lhs->tv_usec - rhs->tv_usec; } -static inline int compat_timespec_compare(struct compat_timespec *lhs, - struct compat_timespec *rhs) +static inline int old_timespec32_compare(struct old_timespec32 *lhs, + struct old_timespec32 *rhs) { if (lhs->tv_sec < rhs->tv_sec) return -1; @@ -552,12 +552,12 @@ asmlinkage long compat_sys_io_getevents(compat_aio_context_t ctx_id, compat_long_t min_nr, compat_long_t nr, struct io_event __user *events, - struct compat_timespec __user *timeout); + struct old_timespec32 __user *timeout); asmlinkage long compat_sys_io_pgetevents(compat_aio_context_t ctx_id, compat_long_t min_nr, compat_long_t nr, struct io_event __user *events, - struct compat_timespec __user *timeout, + struct old_timespec32 __user *timeout, const struct __compat_aio_sigset __user *usig); /* fs/cookies.c */ @@ -642,11 +642,11 @@ asmlinkage long compat_sys_sendfile64(int out_fd, int in_fd, asmlinkage long compat_sys_pselect6(int n, compat_ulong_t __user *inp, compat_ulong_t __user *outp, compat_ulong_t __user *exp, - struct compat_timespec __user *tsp, + struct old_timespec32 __user *tsp, void __user *sig); asmlinkage long compat_sys_ppoll(struct pollfd __user *ufds, unsigned int nfds, - struct compat_timespec __user *tsp, + struct old_timespec32 __user *tsp, const compat_sigset_t __user *sigmask, compat_size_t sigsetsize); @@ -671,15 +671,15 @@ asmlinkage long compat_sys_newfstat(unsigned int fd, /* fs/timerfd.c */ asmlinkage long compat_sys_timerfd_gettime(int ufd, - struct compat_itimerspec __user *otmr); + struct old_itimerspec32 __user *otmr); asmlinkage long compat_sys_timerfd_settime(int ufd, int flags, - const struct compat_itimerspec __user *utmr, - struct compat_itimerspec __user *otmr); + const struct old_itimerspec32 __user *utmr, + struct old_itimerspec32 __user *otmr); /* fs/utimes.c */ asmlinkage long compat_sys_utimensat(unsigned int dfd, const char __user *filename, - struct compat_timespec __user *t, + struct old_timespec32 __user *t, int flags); /* kernel/exit.c */ @@ -691,7 +691,7 @@ asmlinkage long compat_sys_waitid(int, compat_pid_t, /* kernel/futex.c */ asmlinkage long compat_sys_futex(u32 __user *uaddr, int op, u32 val, - struct compat_timespec __user *utime, u32 __user *uaddr2, + struct old_timespec32 __user *utime, u32 __user *uaddr2, u32 val3); asmlinkage long compat_sys_set_robust_list(struct compat_robust_list_head __user *head, @@ -701,8 +701,8 @@ compat_sys_get_robust_list(int pid, compat_uptr_t __user *head_ptr, compat_size_t __user *len_ptr); /* kernel/hrtimer.c */ -asmlinkage long compat_sys_nanosleep(struct compat_timespec __user *rqtp, - struct compat_timespec __user *rmtp); +asmlinkage long compat_sys_nanosleep(struct old_timespec32 __user *rqtp, + struct old_timespec32 __user *rmtp); /* kernel/itimer.c */ asmlinkage long compat_sys_getitimer(int which, @@ -722,19 +722,19 @@ asmlinkage long compat_sys_timer_create(clockid_t which_clock, struct compat_sigevent __user *timer_event_spec, timer_t __user *created_timer_id); asmlinkage long compat_sys_timer_gettime(timer_t timer_id, - struct compat_itimerspec __user *setting); + struct old_itimerspec32 __user *setting); asmlinkage long compat_sys_timer_settime(timer_t timer_id, int flags, - struct compat_itimerspec __user *new, - struct compat_itimerspec __user *old); + struct old_itimerspec32 __user *new, + struct old_itimerspec32 __user *old); asmlinkage long compat_sys_clock_settime(clockid_t which_clock, - struct compat_timespec __user *tp); + struct old_timespec32 __user *tp); asmlinkage long compat_sys_clock_gettime(clockid_t which_clock, - struct compat_timespec __user *tp); + struct old_timespec32 __user *tp); asmlinkage long compat_sys_clock_getres(clockid_t which_clock, - struct compat_timespec __user *tp); + struct old_timespec32 __user *tp); asmlinkage long compat_sys_clock_nanosleep(clockid_t which_clock, int flags, - struct compat_timespec __user *rqtp, - struct compat_timespec __user *rmtp); + struct old_timespec32 __user *rqtp, + struct old_timespec32 __user *rmtp); /* kernel/ptrace.c */ asmlinkage long compat_sys_ptrace(compat_long_t request, compat_long_t pid, @@ -748,7 +748,7 @@ asmlinkage long compat_sys_sched_getaffinity(compat_pid_t pid, unsigned int len, compat_ulong_t __user *user_mask_ptr); asmlinkage long compat_sys_sched_rr_get_interval(compat_pid_t pid, - struct compat_timespec __user *interval); + struct old_timespec32 __user *interval); /* kernel/signal.c */ asmlinkage long compat_sys_sigaltstack(const compat_stack_t __user *uss_ptr, @@ -768,7 +768,7 @@ asmlinkage long compat_sys_rt_sigpending(compat_sigset_t __user *uset, compat_size_t sigsetsize); asmlinkage long compat_sys_rt_sigtimedwait(compat_sigset_t __user *uthese, struct compat_siginfo __user *uinfo, - struct compat_timespec __user *uts, compat_size_t sigsetsize); + struct old_timespec32 __user *uts, compat_size_t sigsetsize); asmlinkage long compat_sys_rt_sigqueueinfo(compat_pid_t pid, int sig, struct compat_siginfo __user *uinfo); /* No generic prototype for rt_sigreturn */ @@ -782,9 +782,9 @@ asmlinkage long compat_sys_setrlimit(unsigned int resource, asmlinkage long compat_sys_getrusage(int who, struct compat_rusage __user *ru); /* kernel/time.c */ -asmlinkage long compat_sys_gettimeofday(struct compat_timeval __user *tv, +asmlinkage long compat_sys_gettimeofday(struct old_timeval32 __user *tv, struct timezone __user *tz); -asmlinkage long compat_sys_settimeofday(struct compat_timeval __user *tv, +asmlinkage long compat_sys_settimeofday(struct old_timeval32 __user *tv, struct timezone __user *tz); asmlinkage long compat_sys_adjtimex(struct compat_timex __user *utp); @@ -798,11 +798,11 @@ asmlinkage long compat_sys_mq_open(const char __user *u_name, asmlinkage long compat_sys_mq_timedsend(mqd_t mqdes, const char __user *u_msg_ptr, compat_size_t msg_len, unsigned int msg_prio, - const struct compat_timespec __user *u_abs_timeout); + const struct old_timespec32 __user *u_abs_timeout); asmlinkage ssize_t compat_sys_mq_timedreceive(mqd_t mqdes, char __user *u_msg_ptr, compat_size_t msg_len, unsigned int __user *u_msg_prio, - const struct compat_timespec __user *u_abs_timeout); + const struct old_timespec32 __user *u_abs_timeout); asmlinkage long compat_sys_mq_notify(mqd_t mqdes, const struct compat_sigevent __user *u_notification); asmlinkage long compat_sys_mq_getsetattr(mqd_t mqdes, @@ -819,7 +819,7 @@ asmlinkage long compat_sys_msgsnd(int msqid, compat_uptr_t msgp, /* ipc/sem.c */ asmlinkage long compat_sys_semctl(int semid, int semnum, int cmd, int arg); asmlinkage long compat_sys_semtimedop(int semid, struct sembuf __user *tsems, - unsigned nsems, const struct compat_timespec __user *timeout); + unsigned nsems, const struct old_timespec32 __user *timeout); /* ipc/shm.c */ asmlinkage long compat_sys_shmctl(int first, int second, void __user *uptr); @@ -876,7 +876,7 @@ asmlinkage long compat_sys_rt_tgsigqueueinfo(compat_pid_t tgid, struct compat_siginfo __user *uinfo); asmlinkage long compat_sys_recvmmsg(int fd, struct compat_mmsghdr __user *mmsg, unsigned vlen, unsigned int flags, - struct compat_timespec __user *timeout); + struct old_timespec32 __user *timeout); asmlinkage long compat_sys_wait4(compat_pid_t pid, compat_uint_t __user *stat_addr, int options, struct compat_rusage __user *ru); @@ -928,7 +928,7 @@ asmlinkage long compat_sys_pwritev64v2(unsigned long fd, asmlinkage long compat_sys_open(const char __user *filename, int flags, umode_t mode); asmlinkage long compat_sys_utimes(const char __user *filename, - struct compat_timeval __user *t); + struct old_timeval32 __user *t); /* __ARCH_WANT_SYSCALL_NO_FLAGS */ asmlinkage long compat_sys_signalfd(int ufd, @@ -942,15 +942,15 @@ asmlinkage long compat_sys_newlstat(const char __user *filename, struct compat_stat __user *statbuf); /* __ARCH_WANT_SYSCALL_DEPRECATED */ -asmlinkage long compat_sys_time(compat_time_t __user *tloc); +asmlinkage long compat_sys_time(old_time32_t __user *tloc); asmlinkage long compat_sys_utime(const char __user *filename, struct compat_utimbuf __user *t); asmlinkage long compat_sys_futimesat(unsigned int dfd, const char __user *filename, - struct compat_timeval __user *t); + struct old_timeval32 __user *t); asmlinkage long compat_sys_select(int n, compat_ulong_t __user *inp, compat_ulong_t __user *outp, compat_ulong_t __user *exp, - struct compat_timeval __user *tvp); + struct old_timeval32 __user *tvp); asmlinkage long compat_sys_ustat(unsigned dev, struct compat_ustat __user *u32); asmlinkage long compat_sys_recv(int fd, void __user *buf, compat_size_t len, unsigned flags); @@ -983,7 +983,7 @@ asmlinkage long compat_sys_sigaction(int sig, #endif /* obsolete: kernel/time/time.c */ -asmlinkage long compat_sys_stime(compat_time_t __user *tptr); +asmlinkage long compat_sys_stime(old_time32_t __user *tptr); /* obsolete: net/socket.c */ asmlinkage long compat_sys_socketcall(int call, u32 __user *args); @@ -1002,15 +1002,15 @@ static inline bool in_compat_syscall(void) { return is_compat_task(); } #endif /** - * ns_to_compat_timeval - Compat version of ns_to_timeval + * ns_to_old_timeval32 - Compat version of ns_to_timeval * @nsec: the nanoseconds value to be converted * - * Returns the compat_timeval representation of the nsec parameter. + * Returns the old_timeval32 representation of the nsec parameter. */ -static inline struct compat_timeval ns_to_compat_timeval(s64 nsec) +static inline struct old_timeval32 ns_to_old_timeval32(s64 nsec) { struct timeval tv; - struct compat_timeval ctv; + struct old_timeval32 ctv; tv = ns_to_timeval(nsec); ctv.tv_sec = tv.tv_sec; diff --git a/include/linux/compat_time.h b/include/linux/compat_time.h deleted file mode 100644 index e70bfd1d2c3f..000000000000 --- a/include/linux/compat_time.h +++ /dev/null @@ -1,32 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ -#ifndef _LINUX_COMPAT_TIME_H -#define _LINUX_COMPAT_TIME_H - -#include <linux/types.h> -#include <linux/time64.h> - -typedef s32 compat_time_t; - -struct compat_timespec { - compat_time_t tv_sec; - s32 tv_nsec; -}; - -struct compat_timeval { - compat_time_t tv_sec; - s32 tv_usec; -}; - -struct compat_itimerspec { - struct compat_timespec it_interval; - struct compat_timespec it_value; -}; - -extern int compat_get_timespec64(struct timespec64 *, const void __user *); -extern int compat_put_timespec64(const struct timespec64 *, void __user *); -extern int get_compat_itimerspec64(struct itimerspec64 *its, - const struct compat_itimerspec __user *uits); -extern int put_compat_itimerspec64(const struct itimerspec64 *its, - struct compat_itimerspec __user *uits); - -#endif /* _LINUX_COMPAT_TIME_H */ diff --git a/include/linux/elfcore-compat.h b/include/linux/elfcore-compat.h index b5f2efdd05e0..7a37f4ce9fd2 100644 --- a/include/linux/elfcore-compat.h +++ b/include/linux/elfcore-compat.h @@ -27,10 +27,10 @@ struct compat_elf_prstatus compat_pid_t pr_ppid; compat_pid_t pr_pgrp; compat_pid_t pr_sid; - struct compat_timeval pr_utime; - struct compat_timeval pr_stime; - struct compat_timeval pr_cutime; - struct compat_timeval pr_cstime; + struct old_timeval32 pr_utime; + struct old_timeval32 pr_stime; + struct old_timeval32 pr_cutime; + struct old_timeval32 pr_cstime; compat_elf_gregset_t pr_reg; #ifdef CONFIG_BINFMT_ELF_FDPIC compat_ulong_t pr_exec_fdpic_loadmap; diff --git a/include/linux/restart_block.h b/include/linux/restart_block.h index 5d83d0c1d06c..bba2920e9c05 100644 --- a/include/linux/restart_block.h +++ b/include/linux/restart_block.h @@ -10,7 +10,7 @@ #include <linux/time64.h> struct timespec; -struct compat_timespec; +struct old_timespec32; struct pollfd; enum timespec_type { @@ -40,7 +40,7 @@ struct restart_block { enum timespec_type type; union { struct __kernel_timespec __user *rmtp; - struct compat_timespec __user *compat_rmtp; + struct old_timespec32 __user *compat_rmtp; }; u64 expires; } nanosleep; diff --git a/include/linux/syscalls.h b/include/linux/syscalls.h index 615cc9b374cf..a4f69cf75470 100644 --- a/include/linux/syscalls.h +++ b/include/linux/syscalls.h @@ -60,7 +60,7 @@ struct tms; struct utimbuf; struct mq_attr; struct compat_stat; -struct compat_timeval; +struct old_timeval32; struct robust_list_head; struct getcpu_cache; struct old_linux_dirent; diff --git a/include/linux/time32.h b/include/linux/time32.h index 0b14f936100a..206508967405 100644 --- a/include/linux/time32.h +++ b/include/linux/time32.h @@ -13,6 +13,31 @@ #define TIME_T_MAX (time_t)((1UL << ((sizeof(time_t) << 3) - 1)) - 1) +typedef s32 old_time32_t; + +struct old_timespec32 { + old_time32_t tv_sec; + s32 tv_nsec; +}; + +struct old_timeval32 { + old_time32_t tv_sec; + s32 tv_usec; +}; + +struct old_itimerspec32 { + struct old_timespec32 it_interval; + struct old_timespec32 it_value; +}; + +extern int get_old_timespec32(struct timespec64 *, const void __user *); +extern int put_old_timespec32(const struct timespec64 *, void __user *); +extern int get_old_itimerspec32(struct itimerspec64 *its, + const struct old_itimerspec32 __user *uits); +extern int put_old_itimerspec32(const struct itimerspec64 *its, + struct old_itimerspec32 __user *uits); + + #if __BITS_PER_LONG == 64 /* timespec64 is defined as timespec here */ diff --git a/ipc/mqueue.c b/ipc/mqueue.c index 0f102210f89e..a210340062da 100644 --- a/ipc/mqueue.c +++ b/ipc/mqueue.c @@ -1534,10 +1534,10 @@ COMPAT_SYSCALL_DEFINE3(mq_getsetattr, mqd_t, mqdes, #endif #ifdef CONFIG_COMPAT_32BIT_TIME -static int compat_prepare_timeout(const struct compat_timespec __user *p, +static int compat_prepare_timeout(const struct old_timespec32 __user *p, struct timespec64 *ts) { - if (compat_get_timespec64(ts, p)) + if (get_old_timespec32(ts, p)) return -EFAULT; if (!timespec64_valid(ts)) return -EINVAL; @@ -1547,7 +1547,7 @@ static int compat_prepare_timeout(const struct compat_timespec __user *p, COMPAT_SYSCALL_DEFINE5(mq_timedsend, mqd_t, mqdes, const char __user *, u_msg_ptr, compat_size_t, msg_len, unsigned int, msg_prio, - const struct compat_timespec __user *, u_abs_timeout) + const struct old_timespec32 __user *, u_abs_timeout) { struct timespec64 ts, *p = NULL; if (u_abs_timeout) { @@ -1562,7 +1562,7 @@ COMPAT_SYSCALL_DEFINE5(mq_timedsend, mqd_t, mqdes, COMPAT_SYSCALL_DEFINE5(mq_timedreceive, mqd_t, mqdes, char __user *, u_msg_ptr, compat_size_t, msg_len, unsigned int __user *, u_msg_prio, - const struct compat_timespec __user *, u_abs_timeout) + const struct old_timespec32 __user *, u_abs_timeout) { struct timespec64 ts, *p = NULL; if (u_abs_timeout) { diff --git a/ipc/msg.c b/ipc/msg.c index decefbd062a8..fc85d852296f 100644 --- a/ipc/msg.c +++ b/ipc/msg.c @@ -622,9 +622,9 @@ struct compat_msqid_ds { struct compat_ipc_perm msg_perm; compat_uptr_t msg_first; compat_uptr_t msg_last; - compat_time_t msg_stime; - compat_time_t msg_rtime; - compat_time_t msg_ctime; + old_time32_t msg_stime; + old_time32_t msg_rtime; + old_time32_t msg_ctime; compat_ulong_t msg_lcbytes; compat_ulong_t msg_lqbytes; unsigned short msg_cbytes; diff --git a/ipc/sem.c b/ipc/sem.c index 6cbbf34a44ac..47ba7a1fde54 100644 --- a/ipc/sem.c +++ b/ipc/sem.c @@ -1698,8 +1698,8 @@ SYSCALL_DEFINE4(semctl, int, semid, int, semnum, int, cmd, unsigned long, arg) struct compat_semid_ds { struct compat_ipc_perm sem_perm; - compat_time_t sem_otime; - compat_time_t sem_ctime; + old_time32_t sem_otime; + old_time32_t sem_ctime; compat_uptr_t sem_base; compat_uptr_t sem_pending; compat_uptr_t sem_pending_last; @@ -2214,11 +2214,11 @@ SYSCALL_DEFINE4(semtimedop, int, semid, struct sembuf __user *, tsops, #ifdef CONFIG_COMPAT_32BIT_TIME long compat_ksys_semtimedop(int semid, struct sembuf __user *tsems, unsigned int nsops, - const struct compat_timespec __user *timeout) + const struct old_timespec32 __user *timeout) { if (timeout) { struct timespec64 ts; - if (compat_get_timespec64(&ts, timeout)) + if (get_old_timespec32(&ts, timeout)) return -EFAULT; return do_semtimedop(semid, tsems, nsops, &ts); } @@ -2227,7 +2227,7 @@ long compat_ksys_semtimedop(int semid, struct sembuf __user *tsems, COMPAT_SYSCALL_DEFINE4(semtimedop, int, semid, struct sembuf __user *, tsems, unsigned int, nsops, - const struct compat_timespec __user *, timeout) + const struct old_timespec32 __user *, timeout) { return compat_ksys_semtimedop(semid, tsems, nsops, timeout); } diff --git a/ipc/shm.c b/ipc/shm.c index 584eddfb88fc..7a60a542fe79 100644 --- a/ipc/shm.c +++ b/ipc/shm.c @@ -1172,9 +1172,9 @@ SYSCALL_DEFINE3(shmctl, int, shmid, int, cmd, struct shmid_ds __user *, buf) struct compat_shmid_ds { struct compat_ipc_perm shm_perm; int shm_segsz; - compat_time_t shm_atime; - compat_time_t shm_dtime; - compat_time_t shm_ctime; + old_time32_t shm_atime; + old_time32_t shm_dtime; + old_time32_t shm_ctime; compat_ipc_pid_t shm_cpid; compat_ipc_pid_t shm_lpid; unsigned short shm_nattch; diff --git a/ipc/syscall.c b/ipc/syscall.c index 65d405f1ba0c..1ac06e3983c0 100644 --- a/ipc/syscall.c +++ b/ipc/syscall.c @@ -35,7 +35,7 @@ SYSCALL_DEFINE6(ipc, unsigned int, call, int, first, unsigned long, second, (const struct __kernel_timespec __user *)fifth); else if (IS_ENABLED(CONFIG_COMPAT_32BIT_TIME)) return compat_ksys_semtimedop(first, ptr, second, - (const struct compat_timespec __user *)fifth); + (const struct old_timespec32 __user *)fifth); else return -ENOSYS; diff --git a/ipc/util.h b/ipc/util.h index 9a8a40de3682..b4d0794513af 100644 --- a/ipc/util.h +++ b/ipc/util.h @@ -267,7 +267,7 @@ long ksys_shmctl(int shmid, int cmd, struct shmid_ds __user *buf); /* for CONFIG_ARCH_WANT_OLD_COMPAT_IPC */ long compat_ksys_semtimedop(int semid, struct sembuf __user *tsems, unsigned int nsops, - const struct compat_timespec __user *timeout); + const struct old_timespec32 __user *timeout); #ifdef CONFIG_COMPAT long compat_ksys_semctl(int semid, int semnum, int cmd, int arg); long compat_ksys_msgctl(int msqid, int cmd, void __user *uptr); diff --git a/kernel/compat.c b/kernel/compat.c index 8e40efc2928a..089d00d0da9c 100644 --- a/kernel/compat.c +++ b/kernel/compat.c @@ -93,28 +93,28 @@ int compat_put_timex(struct compat_timex __user *utp, const struct timex *txc) return 0; } -static int __compat_get_timeval(struct timeval *tv, const struct compat_timeval __user *ctv) +static int __compat_get_timeval(struct timeval *tv, const struct old_timeval32 __user *ctv) { return (!access_ok(VERIFY_READ, ctv, sizeof(*ctv)) || __get_user(tv->tv_sec, &ctv->tv_sec) || __get_user(tv->tv_usec, &ctv->tv_usec)) ? -EFAULT : 0; } -static int __compat_put_timeval(const struct timeval *tv, struct compat_timeval __user *ctv) +static int __compat_put_timeval(const struct timeval *tv, struct old_timeval32 __user *ctv) { return (!access_ok(VERIFY_WRITE, ctv, sizeof(*ctv)) || __put_user(tv->tv_sec, &ctv->tv_sec) || __put_user(tv->tv_usec, &ctv->tv_usec)) ? -EFAULT : 0; } -static int __compat_get_timespec(struct timespec *ts, const struct compat_timespec __user *cts) +static int __compat_get_timespec(struct timespec *ts, const struct old_timespec32 __user *cts) { return (!access_ok(VERIFY_READ, cts, sizeof(*cts)) || __get_user(ts->tv_sec, &cts->tv_sec) || __get_user(ts->tv_nsec, &cts->tv_nsec)) ? -EFAULT : 0; } -static int __compat_put_timespec(const struct timespec *ts, struct compat_timespec __user *cts) +static int __compat_put_timespec(const struct timespec *ts, struct old_timespec32 __user *cts) { return (!access_ok(VERIFY_WRITE, cts, sizeof(*cts)) || __put_user(ts->tv_sec, &cts->tv_sec) || diff --git a/kernel/futex_compat.c b/kernel/futex_compat.c index 83f830acbb5f..410a77a8f6e2 100644 --- a/kernel/futex_compat.c +++ b/kernel/futex_compat.c @@ -173,7 +173,7 @@ COMPAT_SYSCALL_DEFINE3(get_robust_list, int, pid, } COMPAT_SYSCALL_DEFINE6(futex, u32 __user *, uaddr, int, op, u32, val, - struct compat_timespec __user *, utime, u32 __user *, uaddr2, + struct old_timespec32 __user *, utime, u32 __user *, uaddr2, u32, val3) { struct timespec ts; diff --git a/kernel/sched/core.c b/kernel/sched/core.c index a0065c84e73f..756aba643a28 100644 --- a/kernel/sched/core.c +++ b/kernel/sched/core.c @@ -5280,13 +5280,13 @@ SYSCALL_DEFINE2(sched_rr_get_interval, pid_t, pid, #ifdef CONFIG_COMPAT COMPAT_SYSCALL_DEFINE2(sched_rr_get_interval, compat_pid_t, pid, - struct compat_timespec __user *, interval) + struct old_timespec32 __user *, interval) { struct timespec64 t; int retval = sched_rr_get_interval(pid, &t); if (retval == 0) - retval = compat_put_timespec64(&t, interval); + retval = put_old_timespec32(&t, interval); return retval; } #endif diff --git a/kernel/signal.c b/kernel/signal.c index 786bacc60649..bc87ed06cbae 100644 --- a/kernel/signal.c +++ b/kernel/signal.c @@ -3127,7 +3127,7 @@ SYSCALL_DEFINE4(rt_sigtimedwait, const sigset_t __user *, uthese, #ifdef CONFIG_COMPAT COMPAT_SYSCALL_DEFINE4(rt_sigtimedwait, compat_sigset_t __user *, uthese, struct compat_siginfo __user *, uinfo, - struct compat_timespec __user *, uts, compat_size_t, sigsetsize) + struct old_timespec32 __user *, uts, compat_size_t, sigsetsize) { sigset_t s; struct timespec t; diff --git a/kernel/time/hrtimer.c b/kernel/time/hrtimer.c index 3e93c54bd3a1..d1190e9fe3f8 100644 --- a/kernel/time/hrtimer.c +++ b/kernel/time/hrtimer.c @@ -1661,7 +1661,7 @@ int nanosleep_copyout(struct restart_block *restart, struct timespec64 *ts) switch(restart->nanosleep.type) { #ifdef CONFIG_COMPAT_32BIT_TIME case TT_COMPAT: - if (compat_put_timespec64(ts, restart->nanosleep.compat_rmtp)) + if (put_old_timespec32(ts, restart->nanosleep.compat_rmtp)) return -EFAULT; break; #endif @@ -1781,12 +1781,12 @@ SYSCALL_DEFINE2(nanosleep, struct __kernel_timespec __user *, rqtp, #ifdef CONFIG_COMPAT_32BIT_TIME -COMPAT_SYSCALL_DEFINE2(nanosleep, struct compat_timespec __user *, rqtp, - struct compat_timespec __user *, rmtp) +COMPAT_SYSCALL_DEFINE2(nanosleep, struct old_timespec32 __user *, rqtp, + struct old_timespec32 __user *, rmtp) { struct timespec64 tu; - if (compat_get_timespec64(&tu, rqtp)) + if (get_old_timespec32(&tu, rqtp)) return -EFAULT; if (!timespec64_valid(&tu)) diff --git a/kernel/time/posix-stubs.c b/kernel/time/posix-stubs.c index 2c6847d5d69b..989ccf028bde 100644 --- a/kernel/time/posix-stubs.c +++ b/kernel/time/posix-stubs.c @@ -162,20 +162,20 @@ COMPAT_SYS_NI(setitimer); #ifdef CONFIG_COMPAT_32BIT_TIME COMPAT_SYSCALL_DEFINE2(clock_settime, const clockid_t, which_clock, - struct compat_timespec __user *, tp) + struct old_timespec32 __user *, tp) { struct timespec64 new_tp; if (which_clock != CLOCK_REALTIME) return -EINVAL; - if (compat_get_timespec64(&new_tp, tp)) + if (get_old_timespec32(&new_tp, tp)) return -EFAULT; return do_sys_settimeofday64(&new_tp, NULL); } COMPAT_SYSCALL_DEFINE2(clock_gettime, clockid_t, which_clock, - struct compat_timespec __user *, tp) + struct old_timespec32 __user *, tp) { int ret; struct timespec64 kernel_tp; @@ -184,13 +184,13 @@ COMPAT_SYSCALL_DEFINE2(clock_gettime, clockid_t, which_clock, if (ret) return ret; - if (compat_put_timespec64(&kernel_tp, tp)) + if (put_old_timespec32(&kernel_tp, tp)) return -EFAULT; return 0; } COMPAT_SYSCALL_DEFINE2(clock_getres, clockid_t, which_clock, - struct compat_timespec __user *, tp) + struct old_timespec32 __user *, tp) { struct timespec64 rtn_tp = { .tv_sec = 0, @@ -201,7 +201,7 @@ COMPAT_SYSCALL_DEFINE2(clock_getres, clockid_t, which_clock, case CLOCK_REALTIME: case CLOCK_MONOTONIC: case CLOCK_BOOTTIME: - if (compat_put_timespec64(&rtn_tp, tp)) + if (put_old_timespec32(&rtn_tp, tp)) return -EFAULT; return 0; default: @@ -210,8 +210,8 @@ COMPAT_SYSCALL_DEFINE2(clock_getres, clockid_t, which_clock, } COMPAT_SYSCALL_DEFINE4(clock_nanosleep, clockid_t, which_clock, int, flags, - struct compat_timespec __user *, rqtp, - struct compat_timespec __user *, rmtp) + struct old_timespec32 __user *, rqtp, + struct old_timespec32 __user *, rmtp) { struct timespec64 t; @@ -224,7 +224,7 @@ COMPAT_SYSCALL_DEFINE4(clock_nanosleep, clockid_t, which_clock, int, flags, return -EINVAL; } - if (compat_get_timespec64(&t, rqtp)) + if (get_old_timespec32(&t, rqtp)) return -EFAULT; if (!timespec64_valid(&t)) return -EINVAL; diff --git a/kernel/time/posix-timers.c b/kernel/time/posix-timers.c index 3ac7295306dc..8695d444d3b7 100644 --- a/kernel/time/posix-timers.c +++ b/kernel/time/posix-timers.c @@ -767,13 +767,13 @@ SYSCALL_DEFINE2(timer_gettime, timer_t, timer_id, #ifdef CONFIG_COMPAT_32BIT_TIME COMPAT_SYSCALL_DEFINE2(timer_gettime, timer_t, timer_id, - struct compat_itimerspec __user *, setting) + struct old_itimerspec32 __user *, setting) { struct itimerspec64 cur_setting; int ret = do_timer_gettime(timer_id, &cur_setting); if (!ret) { - if (put_compat_itimerspec64(&cur_setting, setting)) + if (put_old_itimerspec32(&cur_setting, setting)) ret = -EFAULT; } return ret; @@ -940,8 +940,8 @@ SYSCALL_DEFINE4(timer_settime, timer_t, timer_id, int, flags, #ifdef CONFIG_COMPAT_32BIT_TIME COMPAT_SYSCALL_DEFINE4(timer_settime, timer_t, timer_id, int, flags, - struct compat_itimerspec __user *, new, - struct compat_itimerspec __user *, old) + struct old_itimerspec32 __user *, new, + struct old_itimerspec32 __user *, old) { struct itimerspec64 new_spec, old_spec; struct itimerspec64 *rtn = old ? &old_spec : NULL; @@ -949,12 +949,12 @@ COMPAT_SYSCALL_DEFINE4(timer_settime, timer_t, timer_id, int, flags, if (!new) return -EINVAL; - if (get_compat_itimerspec64(&new_spec, new)) + if (get_old_itimerspec32(&new_spec, new)) return -EFAULT; error = do_timer_settime(timer_id, flags, &new_spec, rtn); if (!error && old) { - if (put_compat_itimerspec64(&old_spec, old)) + if (put_old_itimerspec32(&old_spec, old)) error = -EFAULT; } return error; @@ -1127,7 +1127,7 @@ SYSCALL_DEFINE2(clock_getres, const clockid_t, which_clock, #ifdef CONFIG_COMPAT_32BIT_TIME COMPAT_SYSCALL_DEFINE2(clock_settime, clockid_t, which_clock, - struct compat_timespec __user *, tp) + struct old_timespec32 __user *, tp) { const struct k_clock *kc = clockid_to_kclock(which_clock); struct timespec64 ts; @@ -1135,14 +1135,14 @@ COMPAT_SYSCALL_DEFINE2(clock_settime, clockid_t, which_clock, if (!kc || !kc->clock_set) return -EINVAL; - if (compat_get_timespec64(&ts, tp)) + if (get_old_timespec32(&ts, tp)) return -EFAULT; return kc->clock_set(which_clock, &ts); } COMPAT_SYSCALL_DEFINE2(clock_gettime, clockid_t, which_clock, - struct compat_timespec __user *, tp) + struct old_timespec32 __user *, tp) { const struct k_clock *kc = clockid_to_kclock(which_clock); struct timespec64 ts; @@ -1153,7 +1153,7 @@ COMPAT_SYSCALL_DEFINE2(clock_gettime, clockid_t, which_clock, err = kc->clock_get(which_clock, &ts); - if (!err && compat_put_timespec64(&ts, tp)) + if (!err && put_old_timespec32(&ts, tp)) err = -EFAULT; return err; @@ -1192,7 +1192,7 @@ COMPAT_SYSCALL_DEFINE2(clock_adjtime, clockid_t, which_clock, #ifdef CONFIG_COMPAT_32BIT_TIME COMPAT_SYSCALL_DEFINE2(clock_getres, clockid_t, which_clock, - struct compat_timespec __user *, tp) + struct old_timespec32 __user *, tp) { const struct k_clock *kc = clockid_to_kclock(which_clock); struct timespec64 ts; @@ -1202,7 +1202,7 @@ COMPAT_SYSCALL_DEFINE2(clock_getres, clockid_t, which_clock, return -EINVAL; err = kc->clock_getres(which_clock, &ts); - if (!err && tp && compat_put_timespec64(&ts, tp)) + if (!err && tp && put_old_timespec32(&ts, tp)) return -EFAULT; return err; @@ -1249,8 +1249,8 @@ SYSCALL_DEFINE4(clock_nanosleep, const clockid_t, which_clock, int, flags, #ifdef CONFIG_COMPAT_32BIT_TIME COMPAT_SYSCALL_DEFINE4(clock_nanosleep, clockid_t, which_clock, int, flags, - struct compat_timespec __user *, rqtp, - struct compat_timespec __user *, rmtp) + struct old_timespec32 __user *, rqtp, + struct old_timespec32 __user *, rmtp) { const struct k_clock *kc = clockid_to_kclock(which_clock); struct timespec64 t; @@ -1260,7 +1260,7 @@ COMPAT_SYSCALL_DEFINE4(clock_nanosleep, clockid_t, which_clock, int, flags, if (!kc->nsleep) return -ENANOSLEEP_NOTSUP; - if (compat_get_timespec64(&t, rqtp)) + if (get_old_timespec32(&t, rqtp)) return -EFAULT; if (!timespec64_valid(&t)) diff --git a/kernel/time/time.c b/kernel/time/time.c index ccdb351277ee..a6215551333a 100644 --- a/kernel/time/time.c +++ b/kernel/time/time.c @@ -104,12 +104,12 @@ SYSCALL_DEFINE1(stime, time_t __user *, tptr) #ifdef CONFIG_COMPAT #ifdef __ARCH_WANT_COMPAT_SYS_TIME -/* compat_time_t is a 32 bit "long" and needs to get converted. */ -COMPAT_SYSCALL_DEFINE1(time, compat_time_t __user *, tloc) +/* old_time32_t is a 32 bit "long" and needs to get converted. */ +COMPAT_SYSCALL_DEFINE1(time, old_time32_t __user *, tloc) { - compat_time_t i; + old_time32_t i; - i = (compat_time_t)ktime_get_real_seconds(); + i = (old_time32_t)ktime_get_real_seconds(); if (tloc) { if (put_user(i,tloc)) @@ -119,7 +119,7 @@ COMPAT_SYSCALL_DEFINE1(time, compat_time_t __user *, tloc) return i; } -COMPAT_SYSCALL_DEFINE1(stime, compat_time_t __user *, tptr) +COMPAT_SYSCALL_DEFINE1(stime, old_time32_t __user *, tptr) { struct timespec64 tv; int err; @@ -223,7 +223,7 @@ SYSCALL_DEFINE2(settimeofday, struct timeval __user *, tv, } #ifdef CONFIG_COMPAT -COMPAT_SYSCALL_DEFINE2(gettimeofday, struct compat_timeval __user *, tv, +COMPAT_SYSCALL_DEFINE2(gettimeofday, struct old_timeval32 __user *, tv, struct timezone __user *, tz) { if (tv) { @@ -241,7 +241,7 @@ COMPAT_SYSCALL_DEFINE2(gettimeofday, struct compat_timeval __user *, tv, return 0; } -COMPAT_SYSCALL_DEFINE2(settimeofday, struct compat_timeval __user *, tv, +COMPAT_SYSCALL_DEFINE2(settimeofday, struct old_timeval32 __user *, tv, struct timezone __user *, tz) { struct timespec64 new_ts; @@ -884,10 +884,10 @@ int put_timespec64(const struct timespec64 *ts, } EXPORT_SYMBOL_GPL(put_timespec64); -int __compat_get_timespec64(struct timespec64 *ts64, - const struct compat_timespec __user *cts) +int __get_old_timespec32(struct timespec64 *ts64, + const struct old_timespec32 __user *cts) { - struct compat_timespec ts; + struct old_timespec32 ts; int ret; ret = copy_from_user(&ts, cts, sizeof(ts)); @@ -900,33 +900,33 @@ int __compat_get_timespec64(struct timespec64 *ts64, return 0; } -int __compat_put_timespec64(const struct timespec64 *ts64, - struct compat_timespec __user *cts) +int __put_old_timespec32(const struct timespec64 *ts64, + struct old_timespec32 __user *cts) { - struct compat_timespec ts = { + struct old_timespec32 ts = { .tv_sec = ts64->tv_sec, .tv_nsec = ts64->tv_nsec }; return copy_to_user(cts, &ts, sizeof(ts)) ? -EFAULT : 0; } -int compat_get_timespec64(struct timespec64 *ts, const void __user *uts) +int get_old_timespec32(struct timespec64 *ts, const void __user *uts) { if (COMPAT_USE_64BIT_TIME) return copy_from_user(ts, uts, sizeof(*ts)) ? -EFAULT : 0; else - return __compat_get_timespec64(ts, uts); + return __get_old_timespec32(ts, uts); } -EXPORT_SYMBOL_GPL(compat_get_timespec64); +EXPORT_SYMBOL_GPL(get_old_timespec32); -int compat_put_timespec64(const struct timespec64 *ts, void __user *uts) +int put_old_timespec32(const struct timespec64 *ts, void __user *uts) { if (COMPAT_USE_64BIT_TIME) return copy_to_user(uts, ts, sizeof(*ts)) ? -EFAULT : 0; else - return __compat_put_timespec64(ts, uts); + return __put_old_timespec32(ts, uts); } -EXPORT_SYMBOL_GPL(compat_put_timespec64); +EXPORT_SYMBOL_GPL(put_old_timespec32); int get_itimerspec64(struct itimerspec64 *it, const struct __kernel_itimerspec __user *uit) @@ -958,23 +958,23 @@ int put_itimerspec64(const struct itimerspec64 *it, } EXPORT_SYMBOL_GPL(put_itimerspec64); -int get_compat_itimerspec64(struct itimerspec64 *its, - const struct compat_itimerspec __user *uits) +int get_old_itimerspec32(struct itimerspec64 *its, + const struct old_itimerspec32 __user *uits) { - if (__compat_get_timespec64(&its->it_interval, &uits->it_interval) || - __compat_get_timespec64(&its->it_value, &uits->it_value)) + if (__get_old_timespec32(&its->it_interval, &uits->it_interval) || + __get_old_timespec32(&its->it_value, &uits->it_value)) return -EFAULT; return 0; } -EXPORT_SYMBOL_GPL(get_compat_itimerspec64); +EXPORT_SYMBOL_GPL(get_old_itimerspec32); -int put_compat_itimerspec64(const struct itimerspec64 *its, - struct compat_itimerspec __user *uits) +int put_old_itimerspec32(const struct itimerspec64 *its, + struct old_itimerspec32 __user *uits) { - if (__compat_put_timespec64(&its->it_interval, &uits->it_interval) || - __compat_put_timespec64(&its->it_value, &uits->it_value)) + if (__put_old_timespec32(&its->it_interval, &uits->it_interval) || + __put_old_timespec32(&its->it_value, &uits->it_value)) return -EFAULT; return 0; } -EXPORT_SYMBOL_GPL(put_compat_itimerspec64); +EXPORT_SYMBOL_GPL(put_old_itimerspec32); diff --git a/net/bluetooth/hci_sock.c b/net/bluetooth/hci_sock.c index 1506e1632394..4f4e20ffbd0b 100644 --- a/net/bluetooth/hci_sock.c +++ b/net/bluetooth/hci_sock.c @@ -1383,7 +1383,7 @@ static void hci_sock_cmsg(struct sock *sk, struct msghdr *msg, if (mask & HCI_CMSG_TSTAMP) { #ifdef CONFIG_COMPAT - struct compat_timeval ctv; + struct old_timeval32 ctv; #endif struct timeval tv; void *data; diff --git a/net/compat.c b/net/compat.c index 7242cce5631b..c07d6539d7c7 100644 --- a/net/compat.c +++ b/net/compat.c @@ -209,8 +209,8 @@ int put_cmsg_compat(struct msghdr *kmsg, int level, int type, int len, void *dat { struct compat_cmsghdr __user *cm = (struct compat_cmsghdr __user *) kmsg->msg_control; struct compat_cmsghdr cmhdr; - struct compat_timeval ctv; - struct compat_timespec cts[3]; + struct old_timeval32 ctv; + struct old_timespec32 cts[3]; int cmlen; if (cm == NULL || kmsg->msg_controllen < sizeof(*cm)) { @@ -351,7 +351,7 @@ static int do_set_attach_filter(struct socket *sock, int level, int optname, static int do_set_sock_timeout(struct socket *sock, int level, int optname, char __user *optval, unsigned int optlen) { - struct compat_timeval __user *up = (struct compat_timeval __user *)optval; + struct old_timeval32 __user *up = (struct old_timeval32 __user *)optval; struct timeval ktime; mm_segment_t old_fs; int err; @@ -420,12 +420,12 @@ COMPAT_SYSCALL_DEFINE5(setsockopt, int, fd, int, level, int, optname, static int do_get_sock_timeout(struct socket *sock, int level, int optname, char __user *optval, int __user *optlen) { - struct compat_timeval __user *up; + struct old_timeval32 __user *up; struct timeval ktime; mm_segment_t old_fs; int len, err; - up = (struct compat_timeval __user *) optval; + up = (struct old_timeval32 __user *) optval; if (get_user(len, optlen)) return -EFAULT; if (len < sizeof(*up)) @@ -457,14 +457,14 @@ static int compat_sock_getsockopt(struct socket *sock, int level, int optname, int compat_sock_get_timestamp(struct sock *sk, struct timeval __user *userstamp) { - struct compat_timeval __user *ctv; + struct old_timeval32 __user *ctv; int err; struct timeval tv; if (COMPAT_USE_64BIT_TIME) return sock_get_timestamp(sk, userstamp); - ctv = (struct compat_timeval __user *) userstamp; + ctv = (struct old_timeval32 __user *) userstamp; err = -ENOENT; if (!sock_flag(sk, SOCK_TIMESTAMP)) sock_enable_timestamp(sk, SOCK_TIMESTAMP); @@ -485,14 +485,14 @@ EXPORT_SYMBOL(compat_sock_get_timestamp); int compat_sock_get_timestampns(struct sock *sk, struct timespec __user *userstamp) { - struct compat_timespec __user *ctv; + struct old_timespec32 __user *ctv; int err; struct timespec ts; if (COMPAT_USE_64BIT_TIME) return sock_get_timestampns (sk, userstamp); - ctv = (struct compat_timespec __user *) userstamp; + ctv = (struct old_timespec32 __user *) userstamp; err = -ENOENT; if (!sock_flag(sk, SOCK_TIMESTAMP)) sock_enable_timestamp(sk, SOCK_TIMESTAMP); @@ -814,7 +814,7 @@ COMPAT_SYSCALL_DEFINE6(recvfrom, int, fd, void __user *, buf, compat_size_t, len static int __compat_sys_recvmmsg(int fd, struct compat_mmsghdr __user *mmsg, unsigned int vlen, unsigned int flags, - struct compat_timespec __user *timeout) + struct old_timespec32 __user *timeout) { int datagrams; struct timespec ktspec; @@ -836,7 +836,7 @@ static int __compat_sys_recvmmsg(int fd, struct compat_mmsghdr __user *mmsg, COMPAT_SYSCALL_DEFINE5(recvmmsg, int, fd, struct compat_mmsghdr __user *, mmsg, unsigned int, vlen, unsigned int, flags, - struct compat_timespec __user *, timeout) + struct old_timespec32 __user *, timeout) { return __compat_sys_recvmmsg(fd, mmsg, vlen, flags, timeout); } diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 946ab080ac00..58b3d9626d86 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -184,8 +184,8 @@ static int snd_pcm_channel_info_user(struct snd_pcm_substream *substream, struct snd_pcm_status32 { s32 state; - struct compat_timespec trigger_tstamp; - struct compat_timespec tstamp; + struct old_timespec32 trigger_tstamp; + struct old_timespec32 tstamp; u32 appl_ptr; u32 hw_ptr; s32 delay; @@ -194,10 +194,10 @@ struct snd_pcm_status32 { u32 overrange; s32 suspended_state; u32 audio_tstamp_data; - struct compat_timespec audio_tstamp; - struct compat_timespec driver_tstamp; + struct old_timespec32 audio_tstamp; + struct old_timespec32 driver_tstamp; u32 audio_tstamp_accuracy; - unsigned char reserved[52-2*sizeof(struct compat_timespec)]; + unsigned char reserved[52-2*sizeof(struct old_timespec32)]; } __attribute__((packed)); @@ -455,9 +455,9 @@ struct snd_pcm_mmap_status32 { s32 state; s32 pad1; u32 hw_ptr; - struct compat_timespec tstamp; + struct old_timespec32 tstamp; s32 suspended_state; - struct compat_timespec audio_tstamp; + struct old_timespec32 audio_tstamp; } __attribute__((packed)); struct snd_pcm_mmap_control32 { diff --git a/sound/core/rawmidi_compat.c b/sound/core/rawmidi_compat.c index e30e30ba6e39..0b3ac672dc41 100644 --- a/sound/core/rawmidi_compat.c +++ b/sound/core/rawmidi_compat.c @@ -57,7 +57,7 @@ static int snd_rawmidi_ioctl_params_compat(struct snd_rawmidi_file *rfile, struct snd_rawmidi_status32 { s32 stream; - struct compat_timespec tstamp; + struct old_timespec32 tstamp; u32 avail; u32 xruns; unsigned char reserved[16]; diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c index e00f7e399e46..212cb7287b66 100644 --- a/sound/core/timer_compat.c +++ b/sound/core/timer_compat.c @@ -84,7 +84,7 @@ static int snd_timer_user_info_compat(struct file *file, } struct snd_timer_status32 { - struct compat_timespec tstamp; + struct old_timespec32 tstamp; u32 resolution; u32 lost; u32 overrun; -- 2.9.0

3 4

[alsa-devel] Bug in snd_pcm_ioplug_avail_update()?
by Rob Duncan 17 Jul '18

17 Jul '18

It seems to me that there's something wrong in snd_pcm_ioplug_avail_update() for capture IO plugins. Once a bunch of conditions are met it makes this call: __snd_pcm_mmap_begin(pcm, &areas, &offset, &size); result = io->data->callback->transfer(io->data, areas, offset, size); if (result < 0) return result; This transfers size frames from the IO plugin, which in general is not all the available data: it's limited by the amount of contiguous space in the mmap. However, the function returns like this: avail = snd_pcm_mmap_avail(pcm); if (avail > io->avail_max) io->avail_max = avail; return (snd_pcm_sframes_t)avail; But this is all the available data in the IO plugin, without considering the contiguous space limitation. This means that there is now uninitialized data in the mmap, and some data still in the IO plugin that has yet to be transferred. I'm running into this misbehaviour with a rate conversion plugin pulling data from an IO plugin. I get chunks of old data showing up later in the stream, I think because the mmap buffer is not being fully written to. I suspect that this avail mismatch tricks snd_pcm_rate_avail_update() into thinking that it can grab an integral number of periods of data from the mmap, which is not the case. But I'm quite tangled up trying to follow the control flow here so I could certainly be mistaken. I'd appreciate any insight into what might be going wrong. Thanks, Rob.

2 5

[alsa-devel] [PATCH] ALSA: emu8000: Use swap macro in snd_emu8000_sample_new
by Gustavo A. R. Silva 17 Jul '18

17 Jul '18

Make use of the swap macro and remove unnecessary variable *tmp*. This makes the code easier to read and maintain. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/isa/sb/emu8000_patch.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c index c2e41d2..d45a6b9 100644 --- a/sound/isa/sb/emu8000_patch.c +++ b/sound/isa/sb/emu8000_patch.c @@ -165,11 +165,8 @@ snd_emu8000_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, return 0; /* be sure loop points start < end */ - if (sp->v.loopstart > sp->v.loopend) { - int tmp = sp->v.loopstart; - sp->v.loopstart = sp->v.loopend; - sp->v.loopend = tmp; - } + if (sp->v.loopstart > sp->v.loopend) + swap(sp->v.loopstart, sp->v.loopend); /* compute true data size to be loaded */ truesize = sp->v.size; -- 2.7.4

2 2

[alsa-devel] Boundary calculation
by Ricardo Biehl Pasquali 17 Jul '18

17 Jul '18

Have you addressed your reply to pasqualirb at gmail.com? I have not received in my mail neither yours nor Takashi's reply. I had to copy manually the headers and the message. Thanks! Clemens Ladisch wrote: > To ensure that the factor is a power of two. Ok, but why LONG_MAX - buffer_size?

2 1

[alsa-devel] [PATCH] ALSA: emu10k1_patch: Use swap macro in snd_emu10k1_sample_new
by Gustavo A. R. Silva 17 Jul '18

17 Jul '18

Make use of the swap macro and remove unnecessary variable *tmp*. This makes the code easier to read and maintain. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/emu10k1/emu10k1_patch.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 0e069ae..c32eb70 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -70,11 +70,8 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, loopend = sampleend; /* be sure loop points start < end */ - if (sp->v.loopstart >= sp->v.loopend) { - int tmp = sp->v.loopstart; - sp->v.loopstart = sp->v.loopend; - sp->v.loopend = tmp; - } + if (sp->v.loopstart >= sp->v.loopend) + swap(sp->v.loopstart, sp->v.loopend); /* compute true data size to be loaded */ truesize = sp->v.size + BLANK_HEAD_SIZE; -- 2.7.4

2 1

[alsa-devel] [PATCH v3] ASoC: hdac_hdmi: Program pin-port-mux during jack report event
by Sriram Periyasamy 17 Jul '18

17 Jul '18

During d3/d0 cycle, the connection selection index of all pins points to the default value. This needs to be restored to ensure audio is restored after d3/d0 cycle. So store the connection selection index and program it during jack report event which gets invoked in cases like d3/d0 cycle, hot plug detection when multiple displays are connected. Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- v2 -> v3: - Balance PM usage count in error path - Add more explanation to the patch v1 -> v2: - Runtime resume the device explicitly in order to use snd_hdac_codec_write() sound/soc/codecs/hdac_hdmi.c | 39 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 39 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 84f7a7a36e4b..39224f01e59b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -83,6 +83,7 @@ struct hdac_hdmi_pin { struct list_head head; hda_nid_t nid; bool mst_capable; + int conn_index; struct hdac_hdmi_port *ports; int num_ports; struct hdac_ext_device *edev; @@ -141,6 +142,9 @@ struct hdac_hdmi_priv { #define hdev_to_hdmi_priv(_hdev) ((to_ehdac_device(_hdev))->private_data) +static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev, + struct hdac_hdmi_port *port); + static struct hdac_hdmi_pcm * hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, struct hdac_hdmi_cvt *cvt) @@ -167,6 +171,38 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, if (is_connect) { /* + * Restore the connection selection index in + * hdac_hdmi_jack_report() which will restore all ports + * irrespective of their usage in all use cases like D0/D3 + * cycles, Hot plug detection. + * + * Increment the usage count to ensure that the device + * is runtime active to send verb across. + */ + pm_runtime_get_sync(&edev->hdev.dev); + + /* set the device if pin is mst_capable */ + if (hdac_hdmi_port_select_set(edev, port) < 0) { + dev_err(&edev->hdev.dev, + "port %d device select fail\n", port->id); + pm_runtime_put_sync(&edev->hdev.dev); + return; + } + + /* + * Ensure that valid connection selection index of the + * respective pin is updated since writing invalid will + * lead to undefined HW behaviour in such cases like if + * mux is not set for ports from userland. + */ + if (port->pin->conn_index > 0) + snd_hdac_codec_write(&edev->hdev, port->pin->nid, + 0, AC_VERB_SET_CONNECT_SEL, + port->pin->conn_index - 1); + + pm_runtime_put_sync(&edev->hdev.dev); + + /* * Report Jack connect event when a device is connected * for the first time where same PCM is attached to multiple * ports. @@ -903,6 +939,9 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, } } + if (ucontrol->value.enumerated.item[0] > 0) + port->pin->conn_index = ucontrol->value.enumerated.item[0]; + /* * Jack status is not reported during device probe as the * PCMs are not registered by then. So report it here. -- 2.7.4

1 0

[alsa-devel] [V3] vga_switcheroo: set audio client id according to bound GPU id
by Jim Qu 17 Jul '18

17 Jul '18

On modern laptop, there are more and more platforms have two GPUs, and each of them maybe have audio codec for HDMP/DP output. For some dGPU which is no output, audio codec usually is disabled. In currect HDA audio driver, it will set all codec as VGA_SWITCHEROO_DIS, the audio which is binded to UMA will be suspended if user use debugfs to contorl power In HDA driver side, it is difficult to know which GPU the audio has binded to. So set the bound gpu pci dev to vga_switcheroo. if the audio client is not the third registration, audio id will set in vga_switcheroo enable function. if the audio client is the last registration when vga_switcheroo _ready() get true, we should get audio client id from bound GPU directly. Signed-off-by: Jim Qu <Jim.Qu(a)amd.com> --- drivers/gpu/vga/vga_switcheroo.c | 63 +++++++++++++++++++++++++++++++++------- include/linux/vga_switcheroo.h | 8 ++--- sound/pci/hda/hda_intel.c | 11 +++---- 3 files changed, 62 insertions(+), 20 deletions(-) diff --git a/drivers/gpu/vga/vga_switcheroo.c b/drivers/gpu/vga/vga_switcheroo.c index fc4adf3..af04e3c 100644 --- a/drivers/gpu/vga/vga_switcheroo.c +++ b/drivers/gpu/vga/vga_switcheroo.c @@ -103,9 +103,11 @@ * runtime pm. If true, writing ON and OFF to the vga_switcheroo debugfs * interface is a no-op so as not to interfere with runtime pm * @list: client list + * @vga_dev: pci device, indicate which GPU is bound to current audio client * * Registered client. A client can be either a GPU or an audio device on a GPU. - * For audio clients, the @fb_info and @active members are bogus. + * For audio clients, the @fb_info and @active members are bogus. For GPU + * clients, the @vga_dev is bogus. */ struct vga_switcheroo_client { struct pci_dev *pdev; @@ -116,6 +118,7 @@ struct vga_switcheroo_client { bool active; bool driver_power_control; struct list_head list; + struct pci_dev *vga_dev; }; /* @@ -161,9 +164,8 @@ struct vgasr_priv { }; #define ID_BIT_AUDIO 0x100 -#define client_is_audio(c) ((c)->id & ID_BIT_AUDIO) -#define client_is_vga(c) ((c)->id == VGA_SWITCHEROO_UNKNOWN_ID || \ - !client_is_audio(c)) +#define client_is_audio(c) ((c)->id & ID_BIT_AUDIO) +#define client_is_vga(c) (!client_is_audio(c)) #define client_id(c) ((c)->id & ~ID_BIT_AUDIO) static int vga_switcheroo_debugfs_init(struct vgasr_priv *priv); @@ -192,14 +194,29 @@ static void vga_switcheroo_enable(void) vgasr_priv.handler->init(); list_for_each_entry(client, &vgasr_priv.clients, list) { - if (client->id != VGA_SWITCHEROO_UNKNOWN_ID) + if (!client_is_vga(client) || + client_id(client) != VGA_SWITCHEROO_UNKNOWN_ID) continue; + ret = vgasr_priv.handler->get_client_id(client->pdev); if (ret < 0) return; client->id = ret; } + + list_for_each_entry(client, &vgasr_priv.clients, list) { + if (!client_is_audio(client) || + client_id(client) != VGA_SWITCHEROO_UNKNOWN_ID) + continue; + + ret = vgasr_priv.handler->get_client_id(client->vga_dev); + if (ret < 0) + return; + + client->id = ret | ID_BIT_AUDIO; + } + vga_switcheroo_debugfs_init(&vgasr_priv); vgasr_priv.active = true; } @@ -272,7 +289,9 @@ EXPORT_SYMBOL(vga_switcheroo_handler_flags); static int register_client(struct pci_dev *pdev, const struct vga_switcheroo_client_ops *ops, - enum vga_switcheroo_client_id id, bool active, + enum vga_switcheroo_client_id id, + struct pci_dev *vga_dev, + bool active, bool driver_power_control) { struct vga_switcheroo_client *client; @@ -287,6 +306,7 @@ static int register_client(struct pci_dev *pdev, client->id = id; client->active = active; client->driver_power_control = driver_power_control; + client->vga_dev = vga_dev; mutex_lock(&vgasr_mutex); list_add_tail(&client->list, &vgasr_priv.clients); @@ -319,7 +339,7 @@ int vga_switcheroo_register_client(struct pci_dev *pdev, const struct vga_switcheroo_client_ops *ops, bool driver_power_control) { - return register_client(pdev, ops, VGA_SWITCHEROO_UNKNOWN_ID, + return register_client(pdev, ops, VGA_SWITCHEROO_UNKNOWN_ID, NULL, pdev == vga_default_device(), driver_power_control); } @@ -329,19 +349,40 @@ EXPORT_SYMBOL(vga_switcheroo_register_client); * vga_switcheroo_register_audio_client - register audio client * @pdev: client pci device * @ops: client callbacks - * @id: client identifier + * @vga_dev: pci device which is bound to current audio client * * Register audio client (audio device on a GPU). The client is assumed * to use runtime PM. Beforehand, vga_switcheroo_client_probe_defer() * shall be called to ensure that all prerequisites are met. * - * Return: 0 on success, -ENOMEM on memory allocation error. + * Return: 0 on success, -ENOMEM on memory allocation error, -EINVAL on getting + * client id error. */ int vga_switcheroo_register_audio_client(struct pci_dev *pdev, const struct vga_switcheroo_client_ops *ops, - enum vga_switcheroo_client_id id) + struct pci_dev *vga_dev) { - return register_client(pdev, ops, id | ID_BIT_AUDIO, false, true); + enum vga_switcheroo_client_id id = VGA_SWITCHEROO_UNKNOWN_ID; + + /* + * if vga_switcheroo has enabled, that mean two GPU clients and also + * handler are registered. Get audio client id from bound GPU client + * id directly, otherwise, set it as VGA_SWITCHEROO_UNKNOWN_ID, + * it will set to correct id in later when vga_switcheroo_enable() + * is called. + */ + mutex_lock(&vgasr_mutex); + if (vgasr_priv.active) { + id = vgasr_priv.handler->get_client_id(vga_dev); + if (id < 0) { + mutex_unlock(&vgasr_mutex); + return -EINVAL; + } + } + mutex_unlock(&vgasr_mutex); + + return register_client(pdev, ops, id | ID_BIT_AUDIO, vga_dev, + false, true); } EXPORT_SYMBOL(vga_switcheroo_register_audio_client); diff --git a/include/linux/vga_switcheroo.h b/include/linux/vga_switcheroo.h index 77f0f0a..a34539b 100644 --- a/include/linux/vga_switcheroo.h +++ b/include/linux/vga_switcheroo.h @@ -84,8 +84,8 @@ enum vga_switcheroo_state { * Client identifier. Audio clients use the same identifier & 0x100. */ enum vga_switcheroo_client_id { - VGA_SWITCHEROO_UNKNOWN_ID = -1, - VGA_SWITCHEROO_IGD, + VGA_SWITCHEROO_UNKNOWN_ID = 0x1000, + VGA_SWITCHEROO_IGD = 0, VGA_SWITCHEROO_DIS, VGA_SWITCHEROO_MAX_CLIENTS, }; @@ -151,7 +151,7 @@ int vga_switcheroo_register_client(struct pci_dev *dev, bool driver_power_control); int vga_switcheroo_register_audio_client(struct pci_dev *pdev, const struct vga_switcheroo_client_ops *ops, - enum vga_switcheroo_client_id id); + struct pci_dev *vga_dev); void vga_switcheroo_client_fb_set(struct pci_dev *dev, struct fb_info *info); @@ -180,7 +180,7 @@ static inline int vga_switcheroo_register_handler(const struct vga_switcheroo_ha enum vga_switcheroo_handler_flags_t handler_flags) { return 0; } static inline int vga_switcheroo_register_audio_client(struct pci_dev *pdev, const struct vga_switcheroo_client_ops *ops, - enum vga_switcheroo_client_id id) { return 0; } + struct pci_dev *vga_dev) { return 0; } static inline void vga_switcheroo_unregister_handler(void) {} static inline enum vga_switcheroo_handler_flags_t vga_switcheroo_handler_flags(void) { return 0; } static inline int vga_switcheroo_lock_ddc(struct pci_dev *pdev) { return -ENODEV; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 14733ff..ead6b7a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1319,15 +1319,16 @@ static const struct vga_switcheroo_client_ops azx_vs_ops = { static int register_vga_switcheroo(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct pci_dev *p; int err; if (!hda->use_vga_switcheroo) return 0; - /* FIXME: currently only handling DIS controller - * is there any machine with two switchable HDMI audio controllers? - */ - err = vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, - VGA_SWITCHEROO_DIS); + + p = get_bound_vga(chip->pci); + err = vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, p); + pci_dev_put(p); + if (err < 0) return err; hda->vga_switcheroo_registered = 1; -- 2.7.4

4 4

[alsa-devel] Boundary calculation
by Ricardo Biehl Pasquali 17 Jul '18

17 Jul '18

Hello. I can't get the tricky boundary calculation. Perhaps I'm missing some obvious thing, or maybe it's really some circumvent for a specific case that was not commented. The current calculation for boundary is: boundary = buffer_size; while (boundary * 2 <= LONG_MAX - buffer_size) boundary *= 2; I've checked out some repositories, specially alsa-driver in ALSA git, and I realized this calculation exists at least since 2001. I guess it was Jaroslav who wrote. Visualization of the calculation (B = buffer_size): LONG_MAX ------------------------------- \______________/ \__/ B * 4 B \______/ B * 2 \__/ B Why isn't a simpler calculation used instead? e.g.: /* closest multiple of buffer size less or equal LONG_MAX */ boundary = LONG_MAX - LONG_MAX % buffer_size; LONG_MAX ------------------------------- \__________________________/ B * 6 Cheers! pasquali

3 2

Re: [alsa-devel] [PATCH RFC] usb: add usb_fill_iso_urb()
by Clemens Ladisch 17 Jul '18

17 Jul '18

Sebastian Andrzej Siewior wrote: > sound/ is (almost) the only part where struct usb_iso_packet_descriptor > init does not fit. It looks like it is done just before urb_submit() and > could be avoided there (moved to the init funtcion instead). For playback, the packet lengths change dynamically, because the nominal sample rate (let alone the actual sample rate) often is not an integer multiple of the 1 kHz/8 kHz USB frame rate. Regards, Clemens

1 0

[alsa-devel] [PATCH 0/3] Make the audio component binding more generic
by Takashi Iwai 16 Jul '18

16 Jul '18

Hi, this is a preliminiary patch set to convert the existing i915 / HD-audio component binding to be applicable to other drivers like radeon / amdgpu. This patchset itself doesn't change the functionality but only renames and split to a new drm_audio_component stuff from i915_audio_component. The actual usage of the new API will follow once after this one gets reviewed / accepted. The whole patches (including this patchset) are found in topic/hda-acomp branch of sound.git tree. BTW, since the whole stuff is about the audio binding, I suppose these will go through sound git tree. Let me know if anyone has concerns. Thanks! Takashi === Takashi Iwai (3): drm/i915: Split audio component to a generic type ALSA: hda/i915: Associate audio component with devres ALSA: hda: Make audio component support more generic drivers/gpu/drm/i915/Kconfig | 1 + drivers/gpu/drm/i915/intel_audio.c | 22 +- include/drm/drm_audio_component.h | 138 ++++++++++++ include/drm/i915_component.h | 85 +------- include/sound/hda_component.h | 61 ++++++ include/sound/hda_i915.h | 37 +--- include/sound/hdaudio.h | 6 +- sound/hda/Kconfig | 7 +- sound/hda/Makefile | 1 + sound/hda/hdac_component.c | 332 ++++++++++++++++++++++++++++ sound/hda/hdac_i915.c | 335 ++--------------------------- sound/pci/hda/patch_hdmi.c | 57 +++-- sound/soc/codecs/hdac_hdmi.c | 10 +- 13 files changed, 624 insertions(+), 468 deletions(-) create mode 100644 include/drm/drm_audio_component.h create mode 100644 include/sound/hda_component.h create mode 100644 sound/hda/hdac_component.c -- 2.18.0

2 7

Re: [alsa-devel] 答复: 答复: 答复: [PATCH] vgaswitchroo: set audio client id according to bound gpu client id
by Takashi Iwai 16 Jul '18

16 Jul '18

On Tue, 10 Jul 2018 09:44:42 +0200, Qu, Jim wrote: > > Hi Takashi, > > I am not expert in audio driver, so I just update some test result, please help triage the issue. > > With audio runtime pm: What does this mean? Is it the stock 4.17.x (or 4.18-rc)? Please clarify your test environment at first: which kernel, which hardware setup. > 1. ubuntu audio setting use pactl to query audio card/output sink. Attachment is pactl output with audio runtime pm. > 2. cat /proc/asound/card0/eld* can get nothing. > > Without audio runtime pm: > 1. pactl can get available audio output/sink > 2. can get eld info from eld#0.1 > > IMO, the issue should be: > > Current operations like HDMI hotplug in/out, pactl command, they do > not access audio HW, so the audio could not resume back at this > period. Sorry, not understood. Why they don't access the audio hardware? > Therefore, upper software could not get HDMI ELD info, can select a available card/output sink. > I am also confuse that if there is no ELD info, how driver to steam audio device to wake up itself? It's sort of a chicken-or-egg question. As long as it's i915 and HD-audio, the hotplug and ELD info transfer doesn't trigger the runtime PM since it's done via the direct callback. And ELD value is cached in HD-audio side. The exception is that if the notification is done during the PM operation. But, the connection and ELD query is performed at the end of codec resume, hence it'll be covered there. So in theory, ELD information should be passed from the GPU to HD-audio no matter whether runtime PM is enabled or not. BTW, please stop top-posting. It makes the thread readability awfully worse. thanks, Takashi > > Thanks > JimQu > > ________________________________________ > 发件人: Takashi Iwai <tiwai(a)suse.de> > 发送时间: 2018年7月10日 1:09 > 收件人: Qu, Jim > 抄送: Daniel Vetter; Alex Deucher; alsa-devel(a)alsa-project.org; amd-gfx(a)lists.freedesktop.org; dri-devel(a)lists.freedesktop.org; Deucher, Alexander > 主题: Re: 答复: [alsa-devel] 答复: [PATCH] vgaswitchroo: set audio client id according to bound gpu client id > > On Mon, 09 Jul 2018 18:05:09 +0200, > Qu, Jim wrote: > > > > Hi Takashi, > > > > Not intel, but it is AMD APU+ AMD GFX, the APU has a local HDMI port for extension. And dGPU is only for offloading render via PRIME. > > > > Originally, the HDA driver before v4.17, there is a bug, that all the audio is set to CLIENT_DIS, so the when the dGPU suspend, the audio which is bound to iGPU will also be suspend. > > > > Now, after Lukas' patches. The audio will auto suspend. It make ubuntu audio setting could not detect HDMI audio, even if HDMI has light up. > > OK, and it has all necessary patches including the one Lukas > suggested, right? > > If so, figure out what you're actually seeing as "PA could not detect > HDMI audio". For example, is it the HDMI (jack) detection (giving > false even if it's connected)? Or is it an error at opening the given > device? Does the driver give the proper ELD bytes as well as the jack > state? > > > Takashi > > > > > Thanks > > JimQu > > > > -----邮件原件----- > > 发件人: Takashi Iwai <tiwai(a)suse.de> > > 发送时间: 2018年7月9日 23:58 > > 收件人: Daniel Vetter <daniel(a)ffwll.ch> > > 抄送: Alex Deucher <alexdeucher(a)gmail.com>; alsa-devel(a)alsa-project.org; amd-gfx(a)lists.freedesktop.org; Qu, Jim <Jim.Qu(a)amd.com>; dri-devel(a)lists.freedesktop.org; Deucher, Alexander <Alexander.Deucher(a)amd.com> > > 主题: Re: [alsa-devel] 答复: [PATCH] vgaswitchroo: set audio client id according to bound gpu client id > > > > On Mon, 09 Jul 2018 17:56:43 +0200, > > Daniel Vetter wrote: > > > > > > On Mon, Jul 09, 2018 at 05:04:22PM +0200, Takashi Iwai wrote: > > > > On Mon, 09 Jul 2018 15:58:51 +0200, > > > > Alex Deucher wrote: > > > > > > > > > > On Mon, Jul 9, 2018 at 6:16 AM, Qu, Jim <Jim.Qu(a)amd.com> wrote: > > > > > > Hi Lukas, > > > > > > > > > > > > Thanks to your explanation, and see comments in line. > > > > > > > > > > > > > > > > > > Do you need to runtime resume the HDA controller even if user > > > > > > space isn't streaming audio? Why, and in which situation exactly? > > > > > > > > > > > > Jim: OEM system uses pactl to queiry audio card and audio output sink, since the audio has power down by runtime pm, so the audio card and output sink are all unavailable. they could not select the available HDMI audio for audio playing. > > > > > > > > > > > > You're saying above that the HDA controller isn't runtime > > > > > > resumed on hotplug of a display. Is that necessary to retrieve ELD or something? > > > > > > I'm not sure if there's code in the HDA driver to acquire a > > > > > > runtime PM ref on HPD, but maybe it's necessary. Or maybe the > > > > > > code is there but somehow no HPD interrupt is received by the HDA driver? > > > > > > > > > > > > Jim: So far, I do not find any code about audio response HPD in kernel. the HPD interrupt will sent to user mode via uevent, not sure whether audio user mode driver can receive the event or not. > > > > > > > > > > On the gfx side at least, we can get a hotplug event via ACPI > > > > > (depending on the OEM design) if displays are attached/detached > > > > > while the dGPU is powered down. I suppose the gfx driver could > > > > > call into the audio driver during one of those events. On the gfx > > > > > side at least we just generate the gfx hotplug event and let userspace deal with it. > > > > > > > > IMO, a more proper way would be to have the direct communication > > > > between the graphics driver and the audio driver like i915 driver > > > > does. Then the audio driver can get plug/unplug event at more > > > > accurate timing without races. > > > > > > > > (Though, it will cause another mess wrt the weak module dependency, > > > > but it's another story :) > > > > > > Just don't do what snd-hda-intel did and instead implement the > > > component stuff properly :-) > > > > A patch is welcome :) > > > > > > thanks, > > > > Takashi > > > > > -Daniel > > > > > > > > > > > > > > > thanks, > > > > > > > > Takashi > > > > > > > > > > > > > > > > > > Alex > > > > > > > > > > > > > > > > > Thanks > > > > > > JimQu > > > > > > > > > > > > ________________________________________ > > > > > > 发件人: Lukas Wunner <lukas(a)wunner.de> > > > > > > 发送时间: 2018年7月9日 17:27 > > > > > > 收件人: Qu, Jim > > > > > > 抄送: alsa-devel(a)alsa-project.org; > > > > > > dri-devel(a)lists.freedesktop.org; Deucher, Alexander; > > > > > > amd-gfx(a)lists.freedesktop.org > > > > > > 主题: Re: [PATCH] vgaswitchroo: set audio client id according to > > > > > > bound gpu client id > > > > > > > > > > > > On Mon, Jul 09, 2018 at 08:52:33AM +0000, Qu, Jim wrote: > > > > > >> Now, I found the audio device will auto suspend even if the gpu > > > > > >> is active, and if I plug in a HDMI device it also do not resume back. > > > > > >> > > > > > >> 1. Did you encounter similar issue before? > > > > > >> 2. audio will auto suspend as default at beginning of boot, is > > > > > >> it expect behaviour? > > > > > > > > > > > > Yes, that's expected. Once you start streaming audio to > > > > > > attached displays, user space opens the codec device and this > > > > > > causes the HDA controller to runtime resume. If the discrete > > > > > > GPU is suspended at that moment, it will be powered on and kept powered on as long as user space is streaming audio. > > > > > > > > > > > > Do you need to runtime resume the HDA controller even if user > > > > > > space isn't streaming audio? Why, and in which situation exactly? > > > > > > > > > > > > You're saying above that the HDA controller isn't runtime > > > > > > resumed on hotplug of a display. Is that necessary to retrieve ELD or something? > > > > > > I'm not sure if there's code in the HDA driver to acquire a > > > > > > runtime PM ref on HPD, but maybe it's necessary. Or maybe the > > > > > > code is there but somehow no HPD interrupt is received by the HDA driver? > > > > > > > > > > > > Thanks, > > > > > > > > > > > > Lukas > > > > > > _______________________________________________ > > > > > > dri-devel mailing list > > > > > > dri-devel(a)lists.freedesktop.org > > > > > > https://lists.freedesktop.org/mailman/listinfo/dri-devel > > > > > _______________________________________________ > > > > > Alsa-devel mailing list > > > > > Alsa-devel(a)alsa-project.org > > > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > _______________________________________________ > > > > dri-devel mailing list > > > > dri-devel(a)lists.freedesktop.org > > > > https://lists.freedesktop.org/mailman/listinfo/dri-devel > > > > > > -- > > > Daniel Vetter > > > Software Engineer, Intel Corporation > > > http://blog.ffwll.ch > > > > [2 pactl_with_runtimepm <application/octet-stream (base64)>] >

4 18

[alsa-devel] [V2 1/2] ALSA: HDA: use PCI_BASE_CLASS_DISPLAY to replace PCI_CLASS_DISPLAY_VGA
by Jim Qu 16 Jul '18

16 Jul '18

Except PCI_CLASS_DISPLAY_VGA, some PCI class is sometimes PCI_CLASS_DISPLAY_3D or PCI_CLASS_DISPLAY_OTHER. Signed-off-by: Jim Qu <Jim.Qu(a)amd.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1ae1850..14733ff 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1429,7 +1429,7 @@ static struct pci_dev *get_bound_vga(struct pci_dev *pci) p = pci_get_domain_bus_and_slot(pci_domain_nr(pci->bus), pci->bus->number, 0); if (p) { - if ((p->class >> 8) == PCI_CLASS_DISPLAY_VGA) + if ((p->class >> 16) == PCI_BASE_CLASS_DISPLAY) return p; pci_dev_put(p); } -- 2.7.4

5 9

[alsa-devel] [PATCH] ALSA: hda/realtek - Add Panasonic CF-SZ6 headset jack quirk
by yokota.hgml＠gmail.com 16 Jul '18

16 Jul '18

This adds some required quirk when uses headset or headphone on Panasonic CF-SZ6. Signed-off-by: YOKOTA Hiroshi <yokota.hgml(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+)

2 1

[alsa-devel] [PATCH] ALSA: hda: add mute led support for HP ProBook 455 G5
by Po-Hsu Lin 16 Jul '18

16 Jul '18

Audio mute led does not work on HP ProBook 455 G5, this can be fixed by using CXT_FIXUP_MUTE_LED_GPIO to support it. BugLink: https://bugs.launchpad.net/bugs/1781763 Reported-by: James Buren Signed-off-by: Po-Hsu Lin <po-hsu.lin(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 e7fcfc3..f641c20 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -964,6 +964,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { 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(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), -- 2.7.4

2 1

[alsa-devel] [PATCH 1/2] ALSA: HDA: use PCI_BASE_CLASS_DISPLAY to cover more display adapters
by Jim Qu 16 Jul '18

16 Jul '18

Signed-off-by: Jim Qu <Jim.Qu(a)amd.com> --- sound/pci/hda/hda_intel.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1ae1850..e0064567 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -31,7 +31,7 @@ * CHANGES: * * 2004.12.01 Major rewrite by tiwai, merged the work of pshou - * + * */ #include <linux/delay.h> @@ -390,8 +390,8 @@ static char *driver_short_names[] = { [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", [AZX_DRIVER_NVIDIA] = "HDA NVidia", - [AZX_DRIVER_TERA] = "HDA Teradici", - [AZX_DRIVER_CTX] = "HDA Creative", + [AZX_DRIVER_TERA] = "HDA Teradici", + [AZX_DRIVER_CTX] = "HDA Creative", [AZX_DRIVER_CTHDA] = "HDA Creative", [AZX_DRIVER_CMEDIA] = "HDA C-Media", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", @@ -1429,7 +1429,7 @@ static struct pci_dev *get_bound_vga(struct pci_dev *pci) p = pci_get_domain_bus_and_slot(pci_domain_nr(pci->bus), pci->bus->number, 0); if (p) { - if ((p->class >> 8) == PCI_CLASS_DISPLAY_VGA) + if ((p->class >> 16) == PCI_BASE_CLASS_DISPLAY) return p; pci_dev_put(p); } -- 2.7.4

5 13

[alsa-devel] [PATCH 0/5] UAC3: Add Selectors and Processing Units.
by Jorge Sanjuan 16 Jul '18

16 Jul '18

This patchset is motivated by the addition of Multi Function Processing Units (MFPU) to an UAC3 topology where there could be signal processing algorithims applied to the audio signal. The MFPUs themself don't provide any useful control, they offer a description of what algorithims it supports and it is for Selector Units to take the bypass control of the dry (unprocessed) signal and the modified one. Moreover, Up/Down mixers may be needed in this topologies to control which outputs of the logical output cluster from the MFPU are to be passed as the final modified audio signal. These patches add support for Selector Units and Processing units for UAC3 and adds a couple fixes that I found while implemeting them: 1) Defualt naming of the virtual terminals was not accurate due to codes overlap between the three UAC standards. 2) UAC2 parsing of processing units was using UAC1 controls bitmap. Based on: next-20180711 Jorge Sanjuan (5): ALSA: usb-audio: Add support for Selector Units in UAC3 ALSA: usb-audio: Processing Unit controls parsing in UAC2 ALSA: usb-audio: Add support for Processing Units in UAC3 ALSA: usb-audio: Unify virtual type units type to UAC3 values ALSA: usb-audio: Tidy up logic for Processing Unit min/max values include/linux/usb/audio-v3.h | 15 +++ include/uapi/linux/usb/audio.h | 49 ++++++++-- sound/usb/mixer.c | 212 +++++++++++++++++++++++++++++++++-------- 3 files changed, 225 insertions(+), 51 deletions(-) -- 2.11.0

2 6

[alsa-devel] Applied "ASoC: hdmi-codec: fix routing" to the asoc tree
by Mark Brown 16 Jul '18

16 Jul '18

The patch ASoC: hdmi-codec: fix routing has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d30e23d69981a4b665f5ce8711335df986576389 Mon Sep 17 00:00:00 2001 From: Russell King <rmk+kernel(a)armlinux.org.uk> Date: Sat, 14 Jul 2018 16:01:06 +0100 Subject: [PATCH] ASoC: hdmi-codec: fix routing Commit 943fa0228252 ("ASoC: hdmi-codec: Use different name for playback streams") broke hdmi-codec's routing between it's output "TX" widget and the S/PDIF or I2S streams by renaming the streams. Whether an error occurs or not is dependent on whether there is another widget called "Playback" registered by some other component - if there is, that widget will be (incorrectly) bound to the HDMI codec's "TX" output widget. If we end up connecting "TX" incorrectly, it can result in components not being started, causing no audio output. Since the I2S and S/PDIF streams now have different names, we can't use a static route at component level to describe the relationship, so arrange to dynamically create the route when the DAI driver is probed. Fixes: 943fa0228252 ("ASoC: hdmi-codec: Use different name for playback streams") Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/hdmi-codec.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 6fa11888672d..3e5b12de71bb 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -291,10 +291,6 @@ static const struct snd_soc_dapm_widget hdmi_widgets[] = { SND_SOC_DAPM_OUTPUT("TX"), }; -static const struct snd_soc_dapm_route hdmi_routes[] = { - { "TX", NULL, "Playback" }, -}; - enum { DAI_ID_I2S = 0, DAI_ID_SPDIF, @@ -689,9 +685,23 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, return snd_ctl_add(rtd->card->snd_card, kctl); } +static int hdmi_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_dapm_context *dapm; + struct snd_soc_dapm_route route = { + .sink = "TX", + .source = dai->driver->playback.stream_name, + }; + + dapm = snd_soc_component_get_dapm(dai->component); + + return snd_soc_dapm_add_routes(dapm, &route, 1); +} + static const struct snd_soc_dai_driver hdmi_i2s_dai = { .name = "i2s-hifi", .id = DAI_ID_I2S, + .probe = hdmi_dai_probe, .playback = { .stream_name = "I2S Playback", .channels_min = 2, @@ -707,6 +717,7 @@ static const struct snd_soc_dai_driver hdmi_i2s_dai = { static const struct snd_soc_dai_driver hdmi_spdif_dai = { .name = "spdif-hifi", .id = DAI_ID_SPDIF, + .probe = hdmi_dai_probe, .playback = { .stream_name = "SPDIF Playback", .channels_min = 2, @@ -733,8 +744,6 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, static const struct snd_soc_component_driver hdmi_driver = { .dapm_widgets = hdmi_widgets, .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), - .dapm_routes = hdmi_routes, - .num_dapm_routes = ARRAY_SIZE(hdmi_routes), .of_xlate_dai_id = hdmi_of_xlate_dai_id, .idle_bias_on = 1, .use_pmdown_time = 1, -- 2.18.0.rc2

1 0

[alsa-devel] [PATCH v4] ASoC: Intel: Boards: Add GLK Realtek Maxim I2S machine driver
by Naveen Manohar 16 Jul '18

16 Jul '18

Patch adds Geminilake I2S machine driver which uses following codecs: RT5682 and MAX98357A. Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> --- v1->v2: reverse xmas tree alignment of var declaration, Add Acked-by. v2->v3: corrected: platform-name for echo-ref dai & change-log position. v3->v4: turn-off sys_clk instead of PLL, when use-case is not ran to save power sound/soc/intel/boards/Kconfig | 14 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 643 ++++++++++++++++++++++++++ 3 files changed, 659 insertions(+) create mode 100644 sound/soc/intel/boards/glk_rt5682_max98357a.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 2479748..cccda87 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -281,6 +281,20 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH + tristate "GLK with RT5682 and MAX98357A in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI + select SND_SOC_RT5682 + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + select SND_HDA_DSP_LOADER + help + This adds support for ASoC machine driver for Geminilake platforms + with RT5682 + MAX98357A I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_INTEL_SKYLAKE endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 92b5507..87ef8b4 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -6,6 +6,7 @@ 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 snd-soc-sst-bxt-rt298-objs := bxt_rt298.o +snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o @@ -27,6 +28,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(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-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c new file mode 100644 index 0000000..c4b94e2 --- /dev/null +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -0,0 +1,643 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018 Intel Corporation. + +/* + * Intel Geminilake I2S Machine Driver with MAX98357A & RT5682 Codecs + * + * Modified from: + * Intel Apollolake I2S Machine driver + */ + +#include <linux/input.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 "../skylake/skl.h" +#include "../../codecs/rt5682.h" +#include "../../codecs/hdac_hdmi.h" + +/* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */ +#define GLK_PLAT_CLK_FREQ 19200000 +#define RT5682_PLL_FREQ (48000 * 512) +#define GLK_REALTEK_CODEC_DAI "rt5682-aif1" +#define GLK_MAXIM_CODEC_DAI "HiFi" +#define MAXIM_DEV0_NAME "MX98357A:00" +#define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 +#define NAME_SIZE 32 + +static struct snd_soc_jack geminilake_hdmi[3]; + +struct glk_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct glk_card_private { + struct snd_soc_jack geminilake_headset; + struct list_head hdmi_pcm_list; +}; + +enum { + GLK_DPCM_AUDIO_PB = 0, + GLK_DPCM_AUDIO_CP, + GLK_DPCM_AUDIO_HS_PB, + GLK_DPCM_AUDIO_ECHO_REF_CP, + GLK_DPCM_AUDIO_REF_CP, + GLK_DPCM_AUDIO_DMIC_CP, + GLK_DPCM_AUDIO_HDMI1_PB, + GLK_DPCM_AUDIO_HDMI2_PB, + GLK_DPCM_AUDIO_HDMI3_PB, +}; + +static int platform_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; + int ret = 0; + + codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop sysclk: %d\n", ret); + } else if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(card->dev, "can't set codec pll: %d\n", ret); + return ret; + } + } + + if (ret) + dev_err(card->dev, "failed to start internal clk: %d\n", ret); + + return ret; +} + +static const struct snd_kcontrol_new geminilake_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget geminilake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Spk", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SPK("HDMI1", NULL), + SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SPK("HDMI3", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route geminilake_map[] = { + /* HP jack connectors - unknown if we have jack detection */ + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* speaker */ + { "Spk", NULL, "Speaker" }, + + /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, + { "IN1P", NULL, "Headset Mic" }, + + /* digital mics */ + { "DMic", NULL, "SoC DMIC" }, + + /* CODEC BE connections */ + { "HiFi Playback", NULL, "ssp1 Tx" }, + { "ssp1 Tx", NULL, "codec0_out" }, + + { "AIF1 Playback", NULL, "ssp2 Tx" }, + { "ssp2 Tx", NULL, "codec1_out" }, + + { "codec0_in", NULL, "ssp2 Rx" }, + { "ssp2 Rx", NULL, "AIF1 Capture" }, + + { "HDMI1", NULL, "hif5-0 Output" }, + { "HDMI2", NULL, "hif6-0 Output" }, + { "HDMI2", NULL, "hif7-0 Output" }, + + { "hifi3", NULL, "iDisp3 Tx" }, + { "iDisp3 Tx", NULL, "iDisp3_out" }, + { "hifi2", NULL, "iDisp2 Tx" }, + { "iDisp2 Tx", NULL, "iDisp2_out" }, + { "hifi1", NULL, "iDisp1 Tx" }, + { "iDisp1 Tx", NULL, "iDisp1_out" }, + + /* DMIC */ + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "DMIC AIF" }, +}; + +static int geminilake_ssp_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); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = DUAL_CHANNEL; + + /* set SSP to 24 bit */ + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct glk_card_private *ctx = 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; + struct snd_soc_jack *jack; + int ret; + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->geminilake_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->geminilake_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +}; + +static int geminilake_rt5682_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; + + /* Set valid bitmask & configuration for I2S in 24 bit */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, 24); + if (ret < 0) { + dev_err(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + return ret; +} + +static struct snd_soc_ops geminilake_rt5682_ops = { + .hw_params = geminilake_rt5682_hw_params, +}; + +static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct glk_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = GLK_DPCM_AUDIO_HDMI1_PB + dai->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_dapm_context *dapm; + int ret; + + dapm = snd_soc_component_get_dapm(component); + ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); + if (ret) { + dev_err(rtd->dev, "Ref Cap ignore suspend failed %d\n", ret); + return ret; + } + + return ret; +} + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + +static int geminilake_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); + + /* + * set BE channel constraint as user FE channels + */ + channels->min = channels->max = 4; + + return 0; +} + +static int geminilake_dmic_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels_quad); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); +} + +static const struct snd_soc_ops geminilake_dmic_ops = { + .startup = geminilake_dmic_startup, +}; + +static const unsigned int rates_16000[] = { + 16000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_16000 = { + .count = ARRAY_SIZE(rates_16000), + .list = rates_16000, +}; + +static int geminilake_refcap_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +}; + +static const struct snd_soc_ops geminilake_refcap_ops = { + .startup = geminilake_refcap_startup, +}; + +/* geminilake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link geminilake_dais[] = { + /* Front End DAI links */ + [GLK_DPCM_AUDIO_PB] = { + .name = "Glk Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .init = geminilake_rt5682_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + [GLK_DPCM_AUDIO_CP] = { + .name = "Glk Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + [GLK_DPCM_AUDIO_HS_PB] = { + .name = "Glk Audio Headset Playback", + .stream_name = "Headset Audio", + .cpu_dai_name = "System Pin2", + .platform_name = "0000:00:0e.0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dpcm_playback = 1, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_ECHO_REF_CP] = { + .name = "Glk Audio Echo Reference cap", + .stream_name = "Echoreference Capture", + .cpu_dai_name = "Echoref Pin", + .platform_name = "0000:00:0e.0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = NULL, + .capture_only = 1, + .nonatomic = 1, + }, + [GLK_DPCM_AUDIO_REF_CP] = { + .name = "Glk Audio Reference cap", + .stream_name = "Refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &geminilake_refcap_ops, + }, + [GLK_DPCM_AUDIO_DMIC_CP] = { + .name = "Glk Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &geminilake_dmic_ops, + }, + [GLK_DPCM_AUDIO_HDMI1_PB] = { + .name = "Glk HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_HDMI2_PB] = { + .name = "Glk HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_HDMI3_PB] = { + .name = "Glk HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + /* Back End DAI links */ + { + /* SSP1 - Codec */ + .name = "SSP1-Codec", + .id = 0, + .cpu_dai_name = "SSP1 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = MAXIM_DEV0_NAME, + .codec_dai_name = GLK_MAXIM_CODEC_DAI, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = geminilake_ssp_fixup, + .dpcm_playback = 1, + }, + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .id = 1, + .cpu_dai_name = "SSP2 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = "i2c-10EC5682:00", + .codec_dai_name = GLK_REALTEK_CODEC_DAI, + .init = geminilake_rt5682_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = geminilake_ssp_fixup, + .ops = &geminilake_rt5682_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .id = 2, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:0e.0", + .ignore_suspend = 1, + .be_hw_params_fixup = geminilake_dmic_fixup, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 3, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 4, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 5, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +static int glk_card_late_probe(struct snd_soc_card *card) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = NULL; + char jack_name[NAME_SIZE]; + struct glk_hdmi_pcm *pcm; + int err = 0; + int i = 0; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &geminilake_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &geminilake_hdmi[i]); + if (err < 0) + return err; + + i++; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} + +/* geminilake audio machine driver for SPT + RT5682 */ +static struct snd_soc_card glk_audio_card_rt5682_m98357a = { + .name = "glkrt5682max", + .owner = THIS_MODULE, + .dai_link = geminilake_dais, + .num_links = ARRAY_SIZE(geminilake_dais), + .controls = geminilake_controls, + .num_controls = ARRAY_SIZE(geminilake_controls), + .dapm_widgets = geminilake_widgets, + .num_dapm_widgets = ARRAY_SIZE(geminilake_widgets), + .dapm_routes = geminilake_map, + .num_dapm_routes = ARRAY_SIZE(geminilake_map), + .fully_routed = true, + .late_probe = glk_card_late_probe, +}; + +static int geminilake_audio_probe(struct platform_device *pdev) +{ + struct glk_card_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + glk_audio_card_rt5682_m98357a.dev = &pdev->dev; + snd_soc_card_set_drvdata(&glk_audio_card_rt5682_m98357a, ctx); + + return devm_snd_soc_register_card(&pdev->dev, + &glk_audio_card_rt5682_m98357a); +} + +static const struct platform_device_id glk_board_ids[] = { + { + .name = "glk_rt5682_max98357a", + .driver_data = + (kernel_ulong_t)&glk_audio_card_rt5682_m98357a, + }, + { } +}; + +static struct platform_driver geminilake_audio = { + .probe = geminilake_audio_probe, + .driver = { + .name = "glk_rt5682_max98357a", + .pm = &snd_soc_pm_ops, + }, + .id_table = glk_board_ids, +}; +module_platform_driver(geminilake_audio) + +/* Module information */ +MODULE_DESCRIPTION("Geminilake Audio Machine driver-RT5682 & MAX98357A in I2S mode"); +MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); +MODULE_AUTHOR("Harsha Priya <harshapriya.n(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:glk_rt5682_max98357a"); -- 2.7.4

3 2

[alsa-devel] [PATCH 1/2] ASoC: AMD: Send correct channel for configuring DMA descriptors
by Akshu Agrawal 16 Jul '18

16 Jul '18

Earlier, ch1 was used to define ACP-SYSMEM transfer and ch2 for ACP-I2S transfer. With recent patches ch1 is used to define channel order number 1 and ch2 as channel order number 2. Thus, Playback: ch1:SYSMEM->ACP ch2:ACP->I2S Capture: ch1:I2S->ACP ch1:ACP->SYSMEM Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 26 ++++++++++++++++++++++---- 1 file changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 65c1033..eeb8677 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -322,17 +322,27 @@ static void config_acp_dma(void __iomem *acp_mmio, struct audio_substream_data *rtd, u32 asic_type) { + u16 ch_acp_sysmem, ch_acp_i2s; + acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages, rtd->pte_offset); + + if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { + ch_acp_sysmem = rtd->ch1; + ch_acp_i2s = rtd->ch2; + } else { + ch_acp_i2s = rtd->ch1; + ch_acp_sysmem = rtd->ch2; + } /* Configure System memory <-> ACP SRAM DMA descriptors */ set_acp_sysmem_dma_descriptors(acp_mmio, rtd->size, rtd->direction, rtd->pte_offset, - rtd->ch1, rtd->sram_bank, + ch_acp_sysmem, rtd->sram_bank, rtd->dma_dscr_idx_1, asic_type); /* Configure ACP SRAM <-> I2S DMA descriptors */ set_acp_to_i2s_dma_descriptors(acp_mmio, rtd->size, rtd->direction, rtd->sram_bank, - rtd->destination, rtd->ch2, + rtd->destination, ch_acp_i2s, rtd->dma_dscr_idx_2, asic_type); } @@ -995,16 +1005,24 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + u16 ch_acp_sysmem, ch_acp_i2s; if (!rtd) return -EINVAL; + if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { + ch_acp_sysmem = rtd->ch1; + ch_acp_i2s = rtd->ch2; + } else { + ch_acp_i2s = rtd->ch1; + ch_acp_sysmem = rtd->ch2; + } config_acp_dma_channel(rtd->acp_mmio, - rtd->ch1, + ch_acp_sysmem, rtd->dma_dscr_idx_1, NUM_DSCRS_PER_CHANNEL, 0); config_acp_dma_channel(rtd->acp_mmio, - rtd->ch2, + ch_acp_i2s, rtd->dma_dscr_idx_2, NUM_DSCRS_PER_CHANNEL, 0); return 0; -- 1.9.1

2 3

[alsa-devel] [PATCH v5 0/5] ASoC: add driver for Atmel I2S controller
by Codrin Ciubotariu 16 Jul '18

16 Jul '18

This is a rework of Cyrille's patches named: [PATCH v3 0/2] ASoC: add driver for Atmel I2S controller https://lkml.org/lkml/2015/9/29/454 This is the version 5 of the series, and addresses the received feedback on the mailing lists. This series applies on top of clk-next branch of clk/linux.git. Based on received feedback, I created a mux clock driver to select the I2S clock source, that also includes proper devicetree bindings and nodes. Also, I added the I2S nodes in sama5d2's devicetree, with the missing pin muxing for the second I2S controller. Alexandre Belloni plans to change the at91 fine grained clock nodes and he recommended to leave the bindings for this clock driver in the same (wrong) manner, so he could fix them all at once. This series of patches adds support to the new Atmel I2S controller embedded on sama5d2 SoCs. ChangeLog v4 -> v5 - removed patches: [PATCH v4 4/7] ASoC: atmel-i2s: dt-bindings: add DT bindings for I2S controller [PATCH v4 5/7] ASoC: atmel-i2s: add driver for the new Atmel I2S controller from this series since they were applied to the asoc tree at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git - addressed comments received on the mux clock; v3 -> v4 - as suggested by Rob Herring: - added a clock mux driver for I2S's clock control bit; - more precise description of I2S's devicetree bindings; - removed SoC and internal detalls from bindings; - addressed comments from Mark Brown; - added devicetree nodes and pin muxing for I2S; v2 -> v3 - fix the coding style, add some more comments and add a section dedicated to sama5d2 SoCs in the DT binding documentation as suggested by Mark Brown. v1 -> v2 - initialize dev->dev before calling dev->caps->mck_init() Codrin Ciubotariu (3): dt-bindings: clk: at91: add an I2S mux clock clk: at91: add I2S clock mux driver ARM: dts: at91: sama5d2: add I2S clock muxing nodes Cyrille Pitchen (2): ARM: dts: at91: sama5d2: add nodes for I2S controllers ARM: dts: at91: sama5d2 Xplained: add pin muxing for I2S .../devicetree/bindings/clock/at91-clock.txt | 35 ++++++ arch/arm/boot/dts/at91-sama5d2_xplained.dts | 28 +++++ arch/arm/boot/dts/sama5d2.dtsi | 52 ++++++++ arch/arm/mach-at91/Kconfig | 4 + drivers/clk/at91/Makefile | 1 + drivers/clk/at91/clk-i2s-mux.c | 116 ++++++++++++++++++ 6 files changed, 236 insertions(+) create mode 100644 drivers/clk/at91/clk-i2s-mux.c -- 2.17.1

4 13

[alsa-devel] [PATCH] ALSA: cs46xx: remove redundant pointer 'ins'
by Colin King 16 Jul '18

16 Jul '18

From: Colin Ian King <colin.king(a)canonical.com> Pointer 'ins' is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'ins' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/cs46xx/dsp_spos_scb_lib.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index abb01ce66983..8d0a3d357345 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -73,13 +73,10 @@ static void cs46xx_dsp_proc_scb_info_read (struct snd_info_entry *entry, { struct proc_scb_info * scb_info = entry->private_data; struct dsp_scb_descriptor * scb = scb_info->scb_desc; - struct dsp_spos_instance * ins; struct snd_cs46xx *chip = scb_info->chip; int j,col; void __iomem *dst = chip->region.idx[1].remap_addr + DSP_PARAMETER_BYTE_OFFSET; - ins = chip->dsp_spos_instance; - mutex_lock(&chip->spos_mutex); snd_iprintf(buffer,"%04x %s:\n",scb->address,scb->scb_name); -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: ali5451: remove redundant pointer 'codec'
by Colin King 16 Jul '18

16 Jul '18

From: Colin Ian King <colin.king(a)canonical.com> Pointer 'codec' is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'codec' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/ali5451/ali5451.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 39547e32e584..9f569379b77e 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1484,12 +1484,9 @@ static struct snd_pcm_hardware snd_ali_capture = static void snd_ali_pcm_free_substream(struct snd_pcm_runtime *runtime) { struct snd_ali_voice *pvoice = runtime->private_data; - struct snd_ali *codec; - if (pvoice) { - codec = pvoice->codec; + if (pvoice) snd_ali_free_voice(pvoice->codec, pvoice); - } } static int snd_ali_open(struct snd_pcm_substream *substream, int rec, -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: sb8: remove redundant pointer runtime
by Colin King 16 Jul '18

16 Jul '18

From: Colin Ian King <colin.king(a)canonical.com> Pointer runtime is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'runtime' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/isa/sb/sb8_main.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index d45df5c54423..80e7dcaa551f 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -381,7 +381,6 @@ static int snd_sb8_capture_trigger(struct snd_pcm_substream *substream, irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) { struct snd_pcm_substream *substream; - struct snd_pcm_runtime *runtime; snd_sb_ack_8bit(chip); switch (chip->mode) { @@ -391,7 +390,6 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) /* fallthru */ case SB_MODE_PLAYBACK_8: substream = chip->playback_substream; - runtime = substream->runtime; if (chip->playback_format == SB_DSP_OUTPUT) snd_sb8_playback_trigger(substream, SNDRV_PCM_TRIGGER_START); snd_pcm_period_elapsed(substream); @@ -402,7 +400,6 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) /* fallthru */ case SB_MODE_CAPTURE_8: substream = chip->capture_substream; - runtime = substream->runtime; if (chip->capture_format == SB_DSP_INPUT) snd_sb8_capture_trigger(substream, SNDRV_PCM_TRIGGER_START); snd_pcm_period_elapsed(substream); -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: gus: remove redundant pointer private_data
by Colin King 16 Jul '18

16 Jul '18

From: Colin Ian King <colin.king(a)canonical.com> Pointer private_data is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'private_data' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/isa/gus/gus_reset.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/isa/gus/gus_reset.c b/sound/isa/gus/gus_reset.c index 3d1fed0c2620..59b3f683d49b 100644 --- a/sound/isa/gus/gus_reset.c +++ b/sound/isa/gus/gus_reset.c @@ -292,7 +292,6 @@ void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice) { unsigned long flags; void (*private_free)(struct snd_gus_voice *voice); - void *private_data; if (voice == NULL || !voice->use) return; @@ -300,7 +299,6 @@ void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice) snd_gf1_clear_voices(gus, voice->number, voice->number); spin_lock_irqsave(&gus->voice_alloc, flags); private_free = voice->private_free; - private_data = voice->private_data; voice->private_free = NULL; voice->private_data = NULL; if (voice->pcm) -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: es1688: remove redundant pointer chip
by Colin King 16 Jul '18

16 Jul '18

From: Colin Ian King <colin.king(a)canonical.com> Pointer chip is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'chip' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/isa/es1688/es1688.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index a826c138e7f5..3dfe7e592c25 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -260,7 +260,6 @@ static int snd_es968_pnp_detect(struct pnp_card_link *pcard, struct snd_card *card; static unsigned int dev; int error; - struct snd_es1688 *chip; if (snd_es968_pnp_is_probed) return -EBUSY; @@ -276,7 +275,6 @@ static int snd_es968_pnp_detect(struct pnp_card_link *pcard, sizeof(struct snd_es1688), &card); if (error < 0) return error; - chip = card->private_data; error = snd_card_es968_pnp(card, dev, pcard, pid); if (error < 0) { -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: opl3: remove redundant pointer opl3
by Colin King 16 Jul '18

16 Jul '18

From: Colin Ian King <colin.king(a)canonical.com> Variable opl3 is being assigned but is never used hence it is redundant and can be removed. Cleans up several clang warnings: warning: variable 'opl3' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/drivers/opl3/opl3_midi.c | 12 ------------ sound/drivers/opl3/opl3_oss.c | 6 ------ 2 files changed, 18 deletions(-) diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index bb3f3a5a6951..71cd5a2fbe82 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -721,9 +721,6 @@ void snd_opl3_note_off(void *p, int note, int vel, */ void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *chan) { - struct snd_opl3 *opl3; - - opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "Key pressure, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); @@ -735,9 +732,6 @@ void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *cha */ void snd_opl3_terminate_note(void *p, int note, struct snd_midi_channel *chan) { - struct snd_opl3 *opl3; - - opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "Terminate note, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); @@ -861,9 +855,6 @@ void snd_opl3_control(void *p, int type, struct snd_midi_channel *chan) void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, struct snd_midi_channel_set *chset) { - struct snd_opl3 *opl3; - - opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "NRPN, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); @@ -876,9 +867,6 @@ void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, void snd_opl3_sysex(void *p, unsigned char *buf, int len, int parsed, struct snd_midi_channel_set *chset) { - struct snd_opl3 *opl3; - - opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "SYSEX\n"); #endif diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c index 22c3e4bca220..8a0ce3f43f42 100644 --- a/sound/drivers/opl3/opl3_oss.c +++ b/sound/drivers/opl3/opl3_oss.c @@ -233,11 +233,8 @@ static int snd_opl3_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format, static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd, unsigned long ioarg) { - struct snd_opl3 *opl3; - if (snd_BUG_ON(!arg)) return -ENXIO; - opl3 = arg->private_data; switch (cmd) { case SNDCTL_FM_LOAD_INSTR: snd_printk(KERN_ERR "OPL3: " @@ -261,11 +258,8 @@ static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd, /* reset device */ static int snd_opl3_reset_seq_oss(struct snd_seq_oss_arg *arg) { - struct snd_opl3 *opl3; - if (snd_BUG_ON(!arg)) return -ENXIO; - opl3 = arg->private_data; return 0; } -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: pxa: remove clock divider and pll setup from zylonite and magician
by Daniel Mack 16 Jul '18

16 Jul '18

The SSP DAI now handles the clocking setup itself, all it needs is the master clock frequency. Remove the code from Zylonite and Magician platforms. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/magician.c | 105 +-------------------------------------- sound/soc/pxa/zylonite.c | 9 ---- 2 files changed, 1 insertion(+), 113 deletions(-) diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 2fc012b06c43..2746725a898c 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -90,95 +90,9 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int acps, acds, width; - unsigned int div4 = PXA_SSP_CLK_SCDB_4; + unsigned int width; int ret = 0; - width = snd_pcm_format_physical_width(params_format(params)); - - /* - * rate = SSPSCLK / (2 * width(16 or 32)) - * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1) - */ - switch (params_rate(params)) { - case 8000: - /* off by a factor of 2: bug in the PXA27x audio clock? */ - acps = 32842000; - switch (width) { - case 16: - /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_16; - break; - default: /* 32 */ - /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_8; - } - break; - case 11025: - acps = 5622000; - switch (width) { - case 16: - /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_4; - break; - default: /* 32 */ - /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - } - break; - case 22050: - acps = 5622000; - switch (width) { - case 16: - /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 44100: - acps = 5622000; - switch (width) { - case 16: - /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 48000: - acps = 12235000; - switch (width) { - case 16: - /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 96000: - default: - acps = 12235000; - switch (width) { - case 16: - /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - break; - default: /* 32 */ - /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - div4 = PXA_SSP_CLK_SCDB_1; - break; - } - break; - } - /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -201,23 +115,6 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* set the SSP audio system clock ACDS divider */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_ACDS, acds); - if (ret < 0) - return ret; - - /* set the SSP audio system clock SCDB divider4 */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_SCDB, div4); - if (ret < 0) - return ret; - - /* set SSP audio pll clock */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps); - if (ret < 0) - return ret; - return 0; } diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ba468e560dd2..230eee450f45 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -83,11 +83,9 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int pll_out = 0; unsigned int wm9713_div = 0; int ret = 0; int rate = params_rate(params); - int width = snd_pcm_format_physical_width(params_format(params)); /* Only support ratios that we can generate neatly from the AC97 * based master clock - in particular, this excludes 44.1kHz. @@ -109,17 +107,10 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Add 1 to the width for the leading clock cycle */ - pll_out = rate * (width + 1) * 8; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out); - if (ret < 0) - return ret; - if (clk_pout) ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV, WM9713_PCMDIV(wm9713_div)); -- 2.17.1

2 2

[alsa-devel] WARNING in __snd_rawmidi_transmit_ack
by syzbot 16 Jul '18

16 Jul '18

Hello, syzbot found the following crash on: HEAD commit: 37b5dca2898d Merge tag 'rtc-4.18-2' of git://git.kernel.or.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=12204d62400000 kernel config: https://syzkaller.appspot.com/x/.config?x=25856fac4e580aa7 dashboard link: https://syzkaller.appspot.com/bug?extid=52f83f0ea8df16932f7f compiler: gcc (GCC) 8.0.1 20180413 (experimental) userspace arch: i386 syzkaller repro:https://syzkaller.appspot.com/x/repro.syz?x=148ea9a4400000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+52f83f0ea8df16932f7f(a)syzkaller.appspotmail.com random: sshd: uninitialized urandom read (32 bytes read) random: sshd: uninitialized urandom read (32 bytes read) random: sshd: uninitialized urandom read (32 bytes read) random: cc1: uninitialized urandom read (8 bytes read) IPVS: ftp: loaded support on port[0] = 21 WARNING: CPU: 0 PID: 4608 at sound/core/rawmidi.c:1163 __snd_rawmidi_transmit_ack+0x263/0x2f0 sound/core/rawmidi.c:1163 Kernel panic - not syncing: panic_on_warn set ... CPU: 0 PID: 4608 Comm: syz-executor0 Not tainted 4.18.0-rc4+ #51 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+0x1c9/0x2b4 lib/dump_stack.c:113 panic+0x238/0x4e7 kernel/panic.c:184 __warn.cold.8+0x163/0x1ba kernel/panic.c:536 report_bug+0x252/0x2d0 lib/bug.c:186 fixup_bug arch/x86/kernel/traps.c:178 [inline] do_error_trap+0x1fc/0x4d0 arch/x86/kernel/traps.c:296 do_invalid_op+0x1b/0x20 arch/x86/kernel/traps.c:316 invalid_op+0x14/0x20 arch/x86/entry/entry_64.S:992 RIP: 0010:__snd_rawmidi_transmit_ack+0x263/0x2f0 sound/core/rawmidi.c:1163 Code: ba 01 00 00 00 be 03 00 00 00 e8 88 4a 83 fb e8 43 fe a2 fb 89 d8 48 83 c4 18 5b 41 5c 41 5d 41 5e 41 5f 5d c3 e8 2d fe a2 fb <0f> 0b e9 72 fe ff ff e8 e1 dc e0 fb e9 1b ff ff ff bb ea ff ff ff RSP: 0018:ffff8801abb8f720 EFLAGS: 00010093 RAX: ffff8801d8b500c0 RBX: 0000000000000002 RCX: ffffffff85d90b82 RDX: 0000000000000000 RSI: ffffffff85d90d13 RDI: 0000000000000006 RBP: ffff8801abb8f760 R08: ffff8801d8b500c0 R09: ffffed003b128cb4 R10: ffffed003b128cb4 R11: ffff8801d89465a3 R12: ffff8801cd646240 R13: 0000000000000802 R14: 0000000000000002 R15: ffff8801ab48e5c0 snd_virmidi_output_trigger+0x3ae/0x6b0 sound/core/seq/seq_virmidi.c:194 snd_rawmidi_output_trigger sound/core/rawmidi.c:150 [inline] snd_rawmidi_kernel_write1+0x519/0x700 sound/core/rawmidi.c:1288 snd_rawmidi_write+0x2ea/0xde0 sound/core/rawmidi.c:1338 __vfs_write+0x117/0x9f0 fs/read_write.c:485 vfs_write+0x1f8/0x560 fs/read_write.c:549 ksys_write+0x101/0x260 fs/read_write.c:598 __do_sys_write fs/read_write.c:610 [inline] __se_sys_write fs/read_write.c:607 [inline] __ia32_sys_write+0x71/0xb0 fs/read_write.c:607 do_syscall_32_irqs_on arch/x86/entry/common.c:326 [inline] do_fast_syscall_32+0x34d/0xfb2 arch/x86/entry/common.c:397 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 RIP: 0023:0xf7f5fcb9 Code: 55 08 8b 88 64 cd ff ff 8b 98 68 cd ff ff 89 c8 85 d2 74 02 89 0a 5b 5d c3 8b 04 24 c3 8b 1c 24 c3 51 52 55 89 e5 0f 34 cd 80 <5d> 5a 59 c3 90 90 90 90 eb 0d 90 90 90 90 90 90 90 90 90 90 90 90 RSP: 002b:00000000f7f190ac EFLAGS: 00000282 ORIG_RAX: 0000000000000004 RAX: ffffffffffffffda RBX: 0000000000000006 RCX: 00000000200000c0 RDX: 000000000000ff4c RSI: 0000000000000000 RDI: 0000000000000000 RBP: 0000000000000000 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 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#bug-status-tracking for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

1 0

[alsa-devel] [PATCH] ASoC: hdmi-codec: fix routing
by Russell King 14 Jul '18

14 Jul '18

Commit 943fa0228252 ("ASoC: hdmi-codec: Use different name for playback streams") broke hdmi-codec's routing between it's output "TX" widget and the S/PDIF or I2S streams by renaming the streams. Whether an error occurs or not is dependent on whether there is another widget called "Playback" registered by some other component - if there is, that widget will be (incorrectly) bound to the HDMI codec's "TX" output widget. If we end up connecting "TX" incorrectly, it can result in components not being started, causing no audio output. Since the I2S and S/PDIF streams now have different names, we can't use a static route at component level to describe the relationship, so arrange to dynamically create the route when the DAI driver is probed. Fixes: 943fa0228252 ("ASoC: hdmi-codec: Use different name for playback streams") Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- sound/soc/codecs/hdmi-codec.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 6fa11888672d..3e5b12de71bb 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -291,10 +291,6 @@ static const struct snd_soc_dapm_widget hdmi_widgets[] = { SND_SOC_DAPM_OUTPUT("TX"), }; -static const struct snd_soc_dapm_route hdmi_routes[] = { - { "TX", NULL, "Playback" }, -}; - enum { DAI_ID_I2S = 0, DAI_ID_SPDIF, @@ -689,9 +685,23 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, return snd_ctl_add(rtd->card->snd_card, kctl); } +static int hdmi_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_dapm_context *dapm; + struct snd_soc_dapm_route route = { + .sink = "TX", + .source = dai->driver->playback.stream_name, + }; + + dapm = snd_soc_component_get_dapm(dai->component); + + return snd_soc_dapm_add_routes(dapm, &route, 1); +} + static const struct snd_soc_dai_driver hdmi_i2s_dai = { .name = "i2s-hifi", .id = DAI_ID_I2S, + .probe = hdmi_dai_probe, .playback = { .stream_name = "I2S Playback", .channels_min = 2, @@ -707,6 +717,7 @@ static const struct snd_soc_dai_driver hdmi_i2s_dai = { static const struct snd_soc_dai_driver hdmi_spdif_dai = { .name = "spdif-hifi", .id = DAI_ID_SPDIF, + .probe = hdmi_dai_probe, .playback = { .stream_name = "SPDIF Playback", .channels_min = 2, @@ -733,8 +744,6 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, static const struct snd_soc_component_driver hdmi_driver = { .dapm_widgets = hdmi_widgets, .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), - .dapm_routes = hdmi_routes, - .num_dapm_routes = ARRAY_SIZE(hdmi_routes), .of_xlate_dai_id = hdmi_of_xlate_dai_id, .idle_bias_on = 1, .use_pmdown_time = 1, -- 2.7.4

1 0

[alsa-devel] [sound:topic/hda-acomp 20/27] htmldocs: include/drm/i915_component.h:45: warning: Function parameter or member 'base' not described in 'i915_audio_component'
by kbuild test robot 13 Jul '18

13 Jul '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/hda-acomp head: db43f0e9b348f78c88362d28c835d479730fbdb0 commit: 292c80bd71755c9f07fd14aee517121c72e13328 [20/27] drm/i915: Split audio component to a generic type reproduce: make htmldocs All warnings (new ones prefixed by >>): WARNING: convert(1) not found, for SVG to PDF conversion install ImageMagick (https://www.imagemagick.org) mm/mempool.c:228: warning: Function parameter or member 'pool' not described in 'mempool_init' Error: Cannot open file arch/x86/kernel/cpu/mtrr/main.c Error: Cannot open file arch/x86/kernel/cpu/mtrr/main.c WARNING: kernel-doc 'scripts/kernel-doc -rst -enable-lineno -export arch/x86/kernel/cpu/mtrr/main.c' failed with return code 2 include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.ibss' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.connect' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.keys' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.ie' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.ie_len' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.bssid' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.ssid' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.default_key' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.default_mgmt_key' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.prev_bssid_valid' not described in 'wireless_dev' include/net/mac80211.h:2282: warning: Function parameter or member 'radiotap_timestamp.units_pos' not described in 'ieee80211_hw' include/net/mac80211.h:2282: warning: Function parameter or member 'radiotap_timestamp.accuracy' not described in 'ieee80211_hw' include/net/mac80211.h:955: warning: Function parameter or member 'control.rates' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.rts_cts_rate_idx' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.use_rts' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.use_cts_prot' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.short_preamble' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.skip_table' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.jiffies' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.vif' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.hw_key' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.flags' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.enqueue_time' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'ack' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'ack.cookie' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'status.rates' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'status.ack_signal' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'status.ampdu_ack_len' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'status.ampdu_len' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'status.antenna' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'status.tx_time' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'status.is_valid_ack_signal' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'status.status_driver_data' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'driver_rates' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'pad' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'rate_driver_data' not described in 'ieee80211_tx_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'rx_stats_avg' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'rx_stats_avg.signal' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'rx_stats_avg.chain_signal' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.filtered' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.retry_failed' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.retry_count' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.lost_packets' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.last_tdls_pkt_time' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.msdu_retries' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.msdu_failed' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.last_ack' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.last_ack_signal' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.ack_signal_filled' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.avg_ack_signal' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'tx_stats.packets' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'tx_stats.bytes' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'tx_stats.last_rate' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'tx_stats.msdu' not described in 'sta_info' kernel/sched/fair.c:3760: warning: Function parameter or member 'flags' not described in 'attach_entity_load_avg' include/linux/device.h:93: warning: bad line: this bus. include/linux/dma-buf.h:307: warning: Function parameter or member 'cb_excl.cb' not described in 'dma_buf' include/linux/dma-buf.h:307: warning: Function parameter or member 'cb_excl.poll' not described in 'dma_buf' include/linux/dma-buf.h:307: warning: Function parameter or member 'cb_excl.active' not described in 'dma_buf' include/linux/dma-buf.h:307: warning: Function parameter or member 'cb_shared.cb' not described in 'dma_buf' include/linux/dma-buf.h:307: warning: Function parameter or member 'cb_shared.poll' not described in 'dma_buf' include/linux/dma-buf.h:307: warning: Function parameter or member 'cb_shared.active' not described in 'dma_buf' include/linux/dma-fence-array.h:54: warning: Function parameter or member 'work' not described in 'dma_fence_array' include/linux/gpio/driver.h:142: warning: Function parameter or member 'request_key' not described in 'gpio_irq_chip' include/linux/iio/hw-consumer.h:1: warning: no structured comments found include/linux/device.h:94: warning: bad line: this bus. include/linux/input/sparse-keymap.h:46: warning: Function parameter or member 'sw' not described in 'key_entry' include/linux/regulator/driver.h:227: warning: Function parameter or member 'resume_early' not described in 'regulator_ops' drivers/regulator/core.c:4465: warning: Excess function parameter 'state' description in 'regulator_suspend_late' arch/s390/include/asm/cio.h:245: warning: Function parameter or member 'esw.esw0' not described in 'irb' arch/s390/include/asm/cio.h:245: warning: Function parameter or member 'esw.esw1' not described in 'irb' arch/s390/include/asm/cio.h:245: warning: Function parameter or member 'esw.esw2' not described in 'irb' arch/s390/include/asm/cio.h:245: warning: Function parameter or member 'esw.esw3' not described in 'irb' arch/s390/include/asm/cio.h:245: warning: Function parameter or member 'esw.eadm' not described in 'irb' drivers/usb/dwc3/gadget.c:510: warning: Excess function parameter 'dwc' description in 'dwc3_gadget_start_config' include/drm/drm_drv.h:610: warning: Function parameter or member 'gem_prime_pin' not described in 'drm_driver' include/drm/drm_drv.h:610: warning: Function parameter or member 'gem_prime_unpin' not described in 'drm_driver' include/drm/drm_drv.h:610: warning: Function parameter or member 'gem_prime_res_obj' not described in 'drm_driver' include/drm/drm_drv.h:610: warning: Function parameter or member 'gem_prime_get_sg_table' not described in 'drm_driver' include/drm/drm_drv.h:610: warning: Function parameter or member 'gem_prime_import_sg_table' not described in 'drm_driver' include/drm/drm_drv.h:610: warning: Function parameter or member 'gem_prime_vmap' not described in 'drm_driver' include/drm/drm_drv.h:610: warning: Function parameter or member 'gem_prime_vunmap' not described in 'drm_driver' include/drm/drm_drv.h:610: warning: Function parameter or member 'gem_prime_mmap' not described in 'drm_driver' >> include/drm/i915_component.h:45: warning: Function parameter or member 'base' not described in 'i915_audio_component' drivers/gpu/drm/i915/i915_vma.h:48: warning: cannot understand function prototype: 'struct i915_vma ' drivers/gpu/drm/i915/i915_vma.h:1: warning: no structured comments found include/drm/tinydrm/tinydrm.h:34: warning: Function parameter or member 'fb_dirty' not described in 'tinydrm_device' drivers/gpu/drm/tinydrm/mipi-dbi.c:272: warning: Function parameter or member 'crtc_state' not described in 'mipi_dbi_enable_flush' drivers/gpu/drm/tinydrm/mipi-dbi.c:272: warning: Function parameter or member 'plane_state' not described in 'mipi_dbi_enable_flush' vim +45 include/drm/i915_component.h 58fddc28 Imre Deak 2015-01-08 @45 :::::: The code at line 45 was first introduced by commit :::::: 58fddc288b5cec192ad9eb9221da7ed14d974a27 drm/i915: add component support :::::: TO: Imre Deak <imre.deak(a)intel.com> :::::: CC: Daniel Vetter <daniel.vetter(a)ffwll.ch> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: qdsp6: q6afe-dai: fix a range check in of_q6afe_parse_dai_data()
by Dan Carpenter 13 Jul '18

13 Jul '18

The main thing is that the data->priv[] array has AFE_PORT_MAX elements so the > condition should be >=. But we may as well check for negative values as well just to be safe. Fixes: 24c4cbcfac09 ("ASoC: qdsp6: q6afe: Add q6afe dai driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index e7f809e3ee1f..89df25e711c5 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1329,7 +1329,7 @@ static void of_q6afe_parse_dai_data(struct device *dev, int id, i, num_lines; ret = of_property_read_u32(node, "reg", &id); - if (ret || id > AFE_PORT_MAX) { + if (ret || id < 0 || id >= AFE_PORT_MAX) { dev_err(dev, "valid dai id not found:%d\n", ret); continue; }

3 2

[alsa-devel] [PATCH] ASoC: qdsp6: q6routing: off by one in routing_hw_params()
by Dan Carpenter 13 Jul '18

13 Jul '18

The data->port_data[] array has AFE_MAX_PORTS elements so the check should be >= instead of > or we write one element beyond the end of the array. Fixes: e3a33673e845 ("ASoC: qdsp6: q6routing: Add q6routing driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 35269b492761..0e7cf9e35d51 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -927,7 +927,7 @@ static int routing_hw_params(struct snd_pcm_substream *substream, else path_type = ADM_PATH_LIVE_REC; - if (be_id > AFE_MAX_PORTS) + if (be_id >= AFE_MAX_PORTS) return -EINVAL; session = &data->port_data[be_id];

3 2

[alsa-devel] [PATCH] ASoC: doc: replace codec to component
by Kuninori Morimoto 13 Jul '18

13 Jul '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Now we can replace Codec to Component. Let's do it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/sound/soc/codec.rst | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/sound/soc/codec.rst b/Documentation/sound/soc/codec.rst index f87612b..58f625f 100644 --- a/Documentation/sound/soc/codec.rst +++ b/Documentation/sound/soc/codec.rst @@ -179,12 +179,12 @@ i.e. static int wm8974_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf; + struct snd_soc_component *component = dai->component; + u16 mute_reg = snd_soc_component_read32(component, WM8974_DAC) & 0xffbf; if (mute) - snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40); + snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40); else - snd_soc_write(codec, WM8974_DAC, mute_reg); + snd_soc_component_write(component, WM8974_DAC, mute_reg); return 0; } -- 1.9.1

2 7

[alsa-devel] [PATCH] ASoC: trace: remove snd_soc_codec
by Kuninori Morimoto 13 Jul '18

13 Jul '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_codec is replaced to snd_soc_component, and it is not used in this file. Let's remove it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/trace/events/asoc.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index ccd1a3b..40c300f 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -12,7 +12,6 @@ #define DAPM_ARROW(dir) (((dir) == SND_SOC_DAPM_DIR_OUT) ? "->" : "<-") struct snd_soc_jack; -struct snd_soc_codec; struct snd_soc_card; struct snd_soc_dapm_widget; struct snd_soc_dapm_path; -- 1.9.1

2 7

[alsa-devel] [PATCH 0/2] ASoC: add sound-name-prefix property
by Jerome Brunet 13 Jul '18

13 Jul '18

This patchset add support in ASoC to pick up name prefix from the component nodes when the component does not match the configuration table provided to the card. The need for this feature came from the platform I'm working on (Amlogic A113). On this platform, the card is made of several instances of each components. It relies heavily on prefixes to describe the routing between these components. The proposed solution benefits every other DT based sound cards, such as the simple card family. It allows to express the routing easily even when the card uses several instances of the same component, or different components with the same sink/source names (Playback, Capture, Output, etc ...) Jerome Brunet (2): ASoC: add DT documentation for the sound-name-prefix property ASoC: allow soc-core to pick up name prefixes from component nodes .../devicetree/bindings/sound/name-prefix.txt | 24 +++++++++++++++++++ sound/soc/soc-core.c | 28 ++++++++++++++++++---- 2 files changed, 47 insertions(+), 5 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/name-prefix.txt -- 2.14.4

2 4

[alsa-devel] [PATCH] ASoC: core: add support to card re-bind/unbind using component framework
by Srinivas Kandagatla 12 Jul '18

12 Jul '18

This patch aims at add achieving dynamic behaviour of audio card when the dependent components disappear and reappear. With this patch the card is removed if any of the dependent component is removed and card is added back if the dependent component comes back. All this is done using component framework and matching based on component name. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- During discussion regarding card re-binding when components unregister and register back at https://lkml.org/lkml/2018/7/9/785 it was suggested that component framework can be added into core to provide this feature. With this new changes the card will re-bind once the dependent component re-registers after unregistering. This works based on the match done from component name using component framework. I have tested this patch with qdsp start-stop usecase for more than 10000 times in loop on Qcom platforms. I will send qdsp side cleanup patches once I get some feedback on this patch. include/sound/soc.h | 5 ++++ sound/soc/soc-core.c | 70 ++++++++++++++++++++++++++++++++++++++++++++++++---- 2 files changed, 70 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 870ba6b64817..b94149d29c0d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -17,6 +17,7 @@ #include <linux/workqueue.h> #include <linux/interrupt.h> #include <linux/kernel.h> +#include <linux/component.h> #include <linux/regmap.h> #include <linux/log2.h> #include <sound/core.h> @@ -1088,6 +1089,10 @@ struct snd_soc_card { struct work_struct deferred_resume_work; + /* component framework related */ + bool components_added; + struct component_match *match; + /* lists of probed devices belonging to this card */ struct list_head component_dev_list; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6d33634b934b..377ed8e67686 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -279,11 +279,31 @@ static inline void snd_soc_debugfs_exit(void) #endif +static int snd_soc_card_comp_compare(struct device *dev, void *data) +{ + struct snd_soc_component *component = NULL; + struct snd_soc_component *t; + + lockdep_assert_held(&client_mutex); + list_for_each_entry(t, &component_list, list) { + if (dev == t->dev) { + component = t; + break; + } + } + + if (component && !strcmp(component->name, data)) + return 1; + + return 0; +} + static int snd_soc_rtdcom_add(struct snd_soc_pcm_runtime *rtd, struct snd_soc_component *component) { struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_rtdcom_list *new_rtdcom; + char *cname; for_each_rtdcom(rtd, rtdcom) { /* already connected */ @@ -300,6 +320,13 @@ static int snd_soc_rtdcom_add(struct snd_soc_pcm_runtime *rtd, list_add_tail(&new_rtdcom->list, &rtd->component_list); + if (!rtd->card->components_added) { + cname = devm_kasprintf(rtd->card->dev, GFP_KERNEL, + "%s", component->name); + component_match_add(rtd->card->dev, &rtd->card->match, + snd_soc_card_comp_compare, cname); + } + return 0; } @@ -835,6 +862,28 @@ static bool soc_is_dai_link_bound(struct snd_soc_card *card, return false; } +static int snd_soc_card_comp_bind(struct device *dev) +{ + struct snd_soc_card *card = dev_get_drvdata(dev); + + if (card->instantiated) + return 0; + + return snd_soc_register_card(card); +} + +static void snd_soc_card_comp_unbind(struct device *dev) +{ + struct snd_soc_card *card = dev_get_drvdata(dev); + + snd_soc_unregister_card(card); +} + +static const struct component_master_ops snd_soc_card_comp_ops = { + .bind = snd_soc_card_comp_bind, + .unbind = snd_soc_card_comp_unbind, +}; + static int soc_bind_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -2108,6 +2157,12 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) card->instantiated = 1; snd_soc_dapm_sync(&card->dapm); + if (!card->components_added) { + component_master_add_with_match(card->dev, + &snd_soc_card_comp_ops, + card->match); + card->components_added = true; + } mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); @@ -3098,11 +3153,6 @@ static void snd_soc_component_cleanup(struct snd_soc_component *component) static void snd_soc_component_del_unlocked(struct snd_soc_component *component) { - struct snd_soc_card *card = component->card; - - if (card) - snd_soc_unregister_card(card); - list_del(&component->list); } @@ -3169,8 +3219,17 @@ int snd_soc_add_component(struct device *dev, snd_soc_component_add(component); + ret = component_add(dev, NULL); + if (ret < 0) { + dev_err(dev, "ASoC: Failed to add Component: %d\n", ret); + goto err_comp; + } + return 0; +err_comp: + soc_remove_component(component); + snd_soc_unregister_dais(component); err_cleanup: snd_soc_component_cleanup(component); err_free: @@ -3218,6 +3277,7 @@ static int __snd_soc_unregister_component(struct device *dev) mutex_unlock(&client_mutex); if (found) { + component_del(dev, NULL); snd_soc_component_cleanup(component); } -- 2.16.2

5 9

[alsa-devel] [PATCH 1/2] ALSA: hda/ca0132: Add Recon3Di quirk for Gigabyte G1.Sniper Z97
by Alastair Bridgewater 12 Jul '18

12 Jul '18

These motherboards have Sound Core3D and apparently "support" Recon3Di. Added to the quirk list as QUIRK_R3DI. Issue report, PCI Subsystem ID, and testing by a contributor on IRC who wished to remain anonymous. Signed-off-by: Alastair Bridgewater <alastair.bridgewater(a)gmail.com> diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index d62c56feaf7d..c803cca1c236 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1048,6 +1048,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ), SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ), SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), + SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA036, "Recon3Di", QUIRK_R3DI), {} }; -- 2.16.1

3 4

[alsa-devel] Alsa_development
by Srinivas Reddy 12 Jul '18

12 Jul '18

Hi ALSA Team,

1 0

[alsa-devel] [PATCH v3 0/5] Add support for audio on SDM845 SoC
by Rohit kumar 11 Jul '18

11 Jul '18

This provides initial patchset to support audio on Qualcomm Techonologies Inc. SDM845 SoC. Currently, it supports audio playback/capture over Primary MI2S and Quaternary TDM ports. Changes since v2: - Added new file to parse dai-links from device tree and use it in sdm845 and apq8096 machine driver. - Removed qcom prefix from device tree properties as suggested by Robb. - Updated apq8096 machine driver to use common APIs. Rohit kumar (5): ASoC: qcom: dt-bindings: Add sdm845 machine bindings ASoC: qcom: Add support to parse common audio device nodes ASoC: qcom: add sdm845 sound card support ASoC: dt-bindings: Update dt binding name for apq8096 ASoC: qcom: apq8096: Use common APIs to parse device nodes .../devicetree/bindings/sound/qcom,apq8096.txt | 15 +- .../devicetree/bindings/sound/qcom,sdm845.txt | 80 +++++ sound/soc/qcom/Kconfig | 14 + sound/soc/qcom/Makefile | 3 + sound/soc/qcom/apq8096.c | 151 +------- sound/soc/qcom/common.c | 150 ++++++++ sound/soc/qcom/common.h | 14 + sound/soc/qcom/sdm845.c | 390 +++++++++++++++++++++ 8 files changed, 674 insertions(+), 143 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sdm845.txt create mode 100644 sound/soc/qcom/common.c create mode 100644 sound/soc/qcom/common.h create mode 100644 sound/soc/qcom/sdm845.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

5 23

[alsa-devel] [PATCH v2] ASoC: hdac_hdmi: Program pin-port-mux during jack report event
by Sriram Periyasamy 11 Jul '18

11 Jul '18

During d3/d0 cycle, the connection selection index of all pins points to the default value. This needs to be restored to ensure audio is restored after d3/d0 cycle. So store the connection selection index and program it during jack report event which gets invoked in cases like d3/d0 cycle, hot plug detection when multiple displays are connected. Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- v1 -> v2: - Runtime resume the device explicitly in order to use snd_hdac_codec_write() sound/soc/codecs/hdac_hdmi.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 84f7a7a36e4b..dd5863830eef 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -83,6 +83,7 @@ struct hdac_hdmi_pin { struct list_head head; hda_nid_t nid; bool mst_capable; + int conn_index; struct hdac_hdmi_port *ports; int num_ports; struct hdac_ext_device *edev; @@ -141,6 +142,9 @@ struct hdac_hdmi_priv { #define hdev_to_hdmi_priv(_hdev) ((to_ehdac_device(_hdev))->private_data) +static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev, + struct hdac_hdmi_port *port); + static struct hdac_hdmi_pcm * hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, struct hdac_hdmi_cvt *cvt) @@ -167,6 +171,30 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, if (is_connect) { /* + * Increment the usage count to ensure that the device + * is runtime active to send verb across. + */ + pm_runtime_get_sync(&edev->hdev.dev); + + /* set the device if pin is mst_capable */ + if (hdac_hdmi_port_select_set(edev, port) < 0) { + dev_err(&edev->hdev.dev, + "port %d device select fail\n", port->id); + return; + } + + /* + * Restore the connection selection index of the + * respective pin. + */ + if (port->pin->conn_index > 0) + snd_hdac_codec_write(&edev->hdev, port->pin->nid, + 0, AC_VERB_SET_CONNECT_SEL, + port->pin->conn_index - 1); + + pm_runtime_put_sync(&edev->hdev.dev); + + /* * Report Jack connect event when a device is connected * for the first time where same PCM is attached to multiple * ports. @@ -903,6 +931,9 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, } } + if (ucontrol->value.enumerated.item[0] > 0) + port->pin->conn_index = ucontrol->value.enumerated.item[0]; + /* * Jack status is not reported during device probe as the * PCMs are not registered by then. So report it here. -- 2.7.4

2 5

[alsa-devel] [PATCH v2] regmap: slimbus: add support to multi read/write
by Srinivas Kandagatla 11 Jul '18

11 Jul '18

SLIMbus supports upto 16 bytes in value management messages, so add support to read/writes upto 16 bytes. This also removes redundant single register reg_read/reg_write. Also useful for paged register access on SLIMbus interfaced codecs. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/base/regmap/regmap-slimbus.c | 23 +++++++---------------- 1 file changed, 7 insertions(+), 16 deletions(-) diff --git a/drivers/base/regmap/regmap-slimbus.c b/drivers/base/regmap/regmap-slimbus.c index 91d501eda8a9..0968059f1ef5 100644 --- a/drivers/base/regmap/regmap-slimbus.c +++ b/drivers/base/regmap/regmap-slimbus.c @@ -7,33 +7,24 @@ #include "internal.h" -static int regmap_slimbus_byte_reg_read(void *context, unsigned int reg, - unsigned int *val) +static int regmap_slimbus_write(void *context, const void *data, size_t count) { struct slim_device *sdev = context; - int v; - v = slim_readb(sdev, reg); - - if (v < 0) - return v; - - *val = v; - - return 0; + return slim_write(sdev, *(u16 *)data, count - 2, (u8 *)data + 2); } -static int regmap_slimbus_byte_reg_write(void *context, unsigned int reg, - unsigned int val) +static int regmap_slimbus_read(void *context, const void *reg, size_t reg_size, + void *val, size_t val_size) { struct slim_device *sdev = context; - return slim_writeb(sdev, reg, val); + return slim_read(sdev, *(u16 *)reg, val_size, val); } static struct regmap_bus regmap_slimbus_bus = { - .reg_write = regmap_slimbus_byte_reg_write, - .reg_read = regmap_slimbus_byte_reg_read, + .write = regmap_slimbus_write, + .read = regmap_slimbus_read, .reg_format_endian_default = REGMAP_ENDIAN_LITTLE, .val_format_endian_default = REGMAP_ENDIAN_LITTLE, }; -- 2.16.2

2 1

[alsa-devel] [PATCH 0/2] ASoC: dpcm: add runtime rate merge of BE.
by Jerome Brunet 11 Jul '18

11 Jul '18

This patchset adds the possibility to merge BE rates in the dpcm runtime hw, like it is done for formats and channels. This first patch is a simple rework of format and channel merge to ease the addition of rate merge. Jerome Brunet (2): ASoC: dpcm: rework runtime stream merge ASoC: dpcm: add rate merge to the BE stream merge include/sound/soc.h | 2 ++ sound/soc/soc-pcm.c | 99 ++++++++++++++++++++++++++++++++++++++++------------- 2 files changed, 77 insertions(+), 24 deletions(-) -- 2.14.4

2 4

[alsa-devel] poll blocked after prepare state
by Harald Schuster 11 Jul '18

11 Jul '18

Hi I have a problem when I got an underrun that my playback path is not working correctly any longer. I debugged the problem and see that the snd_pcm_direct_prepare function is called and than the snd_pcm_dshare_start function. After this the snd_pcm_write_areas is called and the snd_pcm_wait_nocheck never returns. He is blocked in the poll function. Is there any solution for this problem?! best regards Harald

2 14

[alsa-devel] [PATCH - usb-audio 1/1] ALSA: usb-audio: Allow changing from a bad sample rate
by Adam Goode 11 Jul '18

11 Jul '18

If the audio device is externally clocked and set to a rate that does not match the external clock, the clock will never be valid and we cannot set the rate successfully. To fix this, allow a rate change even if the clock is initially invalid, and validate again after the rate is changed. This fixes problems with MOTU UltraLite AVB hardware over USB. Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/sound/usb/clock.c b/sound/usb/clock.c index c79749613fa6..9e9019464f91 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -513,14 +513,26 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, bool writeable; u32 bmControls; + /* First, try to find a valid clock. This may trigger + * automatic clock selection if the current clock is not + * valid. */ clock = snd_usb_clock_find_source(chip, fmt->protocol, fmt->clock, true); - if (clock < 0) - return clock; + if (clock < 0) { + /* We did not find a valid clock, but that might be + * because the current sample rate does not match an + * external clock source. Try again without validation + * and we will do another validation after setting the + * rate. */ + clock = snd_usb_clock_find_source(chip, fmt->protocol, + fmt->clock, false); + if (clock < 0) + return clock; + } prev_rate = get_sample_rate_v2v3(chip, iface, fmt->altsetting, clock); if (prev_rate == rate) - return 0; + return uac_clock_source_is_valid(chip, fmt->protocol, clock) ? 0 : -ENXIO; if (fmt->protocol == UAC_VERSION_3) { struct uac3_clock_source_descriptor *cs_desc; @@ -577,7 +589,8 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, snd_usb_set_interface_quirk(dev); } - return 0; + /* validate clock after rate change */ + return uac_clock_source_is_valid(chip, fmt->protocol, clock) ? 0 : -ENXIO; } int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, -- 2.18.0.203.gfac676dfb9-goog

2 1

[alsa-devel] [PATCH - snd_interval_refine 1/1] pcm: snd_interval_refine_first/last: exclude value only if also excluded before
by twischer＠de.adit-jv.com 11 Jul '18

11 Jul '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this commit the following intervals [x y), (x y) were be replaced to (y-1 y) by snd_interval_refine_last(). This was also done if y-1 is part of the previous interval. With this changes it will be replaced with [y-1 y) in case of y-1 is part of the previous interval. A similar behavior will be used for snd_interval_refine_first(). This commit adapts the changes for alsa-lib of commit 9bb985c ("pcm: snd_interval_refine_first/last: exclude value only if also excluded before") Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- That this fix is also required for kernel was spotted by Takashi in [1]. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-July/137764.html diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index faa6786..729a85a 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -629,27 +629,33 @@ EXPORT_SYMBOL(snd_interval_refine); static int snd_interval_refine_first(struct snd_interval *i) { + const unsigned int last_max = i->max; + if (snd_BUG_ON(snd_interval_empty(i))) return -EINVAL; if (snd_interval_single(i)) return 0; i->max = i->min; - i->openmax = i->openmin; - if (i->openmax) + if (i->openmin) i->max++; + /* only exclude max value if also excluded before refine */ + i->openmax = (i->openmax && i->max >= last_max); return 1; } static int snd_interval_refine_last(struct snd_interval *i) { + const unsigned int last_min = i->min; + if (snd_BUG_ON(snd_interval_empty(i))) return -EINVAL; if (snd_interval_single(i)) return 0; i->min = i->max; - i->openmin = i->openmax; - if (i->openmin) + if (i->openmax) i->min--; + /* only exclude min value if also excluded before refine */ + i->openmin = (i->openmin && i->min <= last_min); return 1; } -- 2.7.4

2 1

[alsa-devel] [PATCH v5 0/9] soundwire: Add multi link support
by Shreyas NC 11 Jul '18

11 Jul '18

Currently, in the SoundWire subsystem, the concept of stream is limited to a Master and one or more Slaves(Codecs). This series aims to add support for multiple Master(s) sharing the same reference clock and synchronized in the hardware. This patch series adds: - Fix to avoid duplicate stream state assignment - Avoid incorrect stream release after configuring Master - Helpers to lock bus instances part of the stream - Boiler plate conversion of code to support a list of Master runtime - Support multi link bank switch to support synchronization between multiple masters - Add Intel platform ops for pre/post bank switch changes in v5: - Add patch to fix duplicate stream state assignment - fix incorrect exit which was releasing the stream after configuring master - In sdw_release_bus_lock(), release the mutex in reverse order as suggested by Takashi changes in v4: - Added changes in sdw_stream_runtime structure to track Masters added to stream - Add changes to support ml_bankswitch only if both the hardware supports and the stream is handled by multiple masters Sanyog Kale (2): Documentation: soundwire: Add documentation for multi link soundwire: Add support to lock across bus instances Shreyas NC (6): soundwire: Fix duplicate stream state assignment soundwire: fix incorrect exit after configuring stream soundwire: Initialize completion for defer messages soundwire: keep track of Masters in a stream soundwire: Add support for multi link bank switch soundwire: intel: Add pre/post bank switch ops Vinod Koul (1): soundwire: Handle multiple master instances in a stream Documentation/driver-api/soundwire/stream.rst | 28 ++ drivers/soundwire/bus.c | 6 + drivers/soundwire/bus.h | 4 + drivers/soundwire/intel.c | 65 ++++ drivers/soundwire/stream.c | 502 +++++++++++++++++++------- include/linux/soundwire/sdw.h | 12 +- 6 files changed, 483 insertions(+), 134 deletions(-) -- 2.7.4

3 18

[alsa-devel] [PATCH] ASoC: Intel: bytcr_rt5640: Add quirk for the Lenovo Miix2 8 tablet
by Hans de Goede 10 Jul '18

10 Jul '18

Add a quirk for the Lenovo Miix2 8 tablet, this tablet uses a digital mic on DMIC1 and has a mono-speaker. The jack-detect uses the default settings.. Reported-and-tested-by: russianneuromancer(a)ya.ru Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 718f022b3aa0..d32844f94d74 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -562,6 +562,19 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Lenovo Miix 2 8 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "20326"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "Hiking"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_MCLK_EN), + }, { /* MSI S100 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."), -- 2.17.1

3 2

[alsa-devel] [PATCH v2 1/3] ASoC: Intel: bytcr_rt5651: Remove is_valleyview helper
by Hans de Goede 10 Jul '18

10 Jul '18

Remove is_valleyview helper, this is not necessary, we can simply call x86_match_cpu() directly instead. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5651.c | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index ba2753e0e12a..80f47a45cb10 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -28,6 +28,7 @@ #include <linux/dmi.h> #include <linux/slab.h> #include <asm/cpu_device_id.h> +#include <asm/intel-family.h> #include <asm/platform_sst_audio.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -680,17 +681,10 @@ static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-mic[-swapped-hp]" */ -static bool is_valleyview(void) -{ - static const struct x86_cpu_id cpu_ids[] = { - { X86_VENDOR_INTEL, 6, 55 }, /* Valleyview, Bay Trail */ - {} - }; - - if (!x86_match_cpu(cpu_ids)) - return false; - return true; -} +static const struct x86_cpu_id baytrail_cpu_ids[] = { + { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_SILVERMONT1 }, /* Valleyview */ + {} +}; struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ u64 aif_value; /* 1: AIF1, 2: AIF2 */ @@ -741,7 +735,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) * swap SSP0 if bytcr is detected * (will be overridden if DMI quirk is detected) */ - if (is_valleyview()) { + if (x86_match_cpu(baytrail_cpu_ids)) { struct sst_platform_info *p_info = mach->pdata; const struct sst_res_info *res_info = p_info->res_info; -- 2.17.1

3 7

[alsa-devel] [PATCH v2] ASoC: Intel: bytcr_rt5651: Add support for externar amplifier enable GPIO
by Hans de Goede 10 Jul '18

10 Jul '18

The rt5651 does not have a built-in speaker amplifier, so it is often used together with an external amplifier. On Cherry Trail boards this external amplifier's enable pin is driven through a GPIO, which is given as the first GPIO in the ACPI resources of the codec fwnode. This commit adds support to the bytcr_rt5651 for this GPIO, fixing the speaker not working on CHT devices with a rt5651 codec. Cc: Carlo Caione <carlo(a)endlessm.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Changes in v2: -Handle the ext amp GPIO entirely in the machine driver, the ext-amp is not part of the codec, so it should not be controlled by the codec driver --- sound/soc/intel/boards/bytcr_rt5651.c | 65 +++++++++++++++++++++++++-- 1 file changed, 62 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index d920725ce603..5301205496be 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -26,6 +26,8 @@ #include <linux/clk.h> #include <linux/device.h> #include <linux/dmi.h> +#include <linux/gpio/consumer.h> +#include <linux/gpio/machine.h> #include <linux/slab.h> #include <asm/cpu_device_id.h> #include <asm/intel-family.h> @@ -86,6 +88,7 @@ enum { struct byt_rt5651_private { struct clk *mclk; + struct gpio_desc *ext_amp_gpio; struct snd_soc_jack jack; }; @@ -208,6 +211,20 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return 0; } +static int rt5651_ext_amp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card); + + if (SND_SOC_DAPM_EVENT_ON(event)) + gpiod_set_value_cansleep(priv->ext_amp_gpio, 1); + else + gpiod_set_value_cansleep(priv->ext_amp_gpio, 0); + + return 0; +} + static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -217,7 +234,9 @@ static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = { SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - + SND_SOC_DAPM_SUPPLY("Ext Amp Power", SND_SOC_NOPM, 0, 0, + rt5651_ext_amp_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), }; static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { @@ -225,6 +244,7 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { {"Headset Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "Platform Clock"}, {"Speaker", NULL, "Platform Clock"}, + {"Speaker", NULL, "Ext Amp Power"}, {"Line In", NULL, "Platform Clock"}, {"Headset Mic", NULL, "micbias1"}, /* lowercase for rt5651 */ @@ -678,6 +698,18 @@ static const struct x86_cpu_id baytrail_cpu_ids[] = { {} }; +static const struct x86_cpu_id cherrytrail_cpu_ids[] = { + { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_AIRMONT }, /* Braswell */ + {} +}; + +static const struct acpi_gpio_params ext_amp_enable_gpios = { 0, 0, false }; + +static const struct acpi_gpio_mapping byt_rt5651_gpios[] = { + { "ext-amp-enable-gpios", &ext_amp_enable_gpios, 1 }, + { }, +}; + struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ u64 aif_value; /* 1: AIF1, 2: AIF2 */ u64 mclock_value; /* usually 25MHz (0x17d7940), ignored */ @@ -793,9 +825,36 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) /* Must be called before register_card, also see declaration comment. */ ret_val = byt_rt5651_add_codec_device_props(codec_dev); - put_device(codec_dev); - if (ret_val) + if (ret_val) { + put_device(codec_dev); return ret_val; + } + + /* Cherry Trail devices use an external amplifier enable gpio */ + if (x86_match_cpu(cherrytrail_cpu_ids)) { + devm_acpi_dev_add_driver_gpios(codec_dev, byt_rt5651_gpios); + priv->ext_amp_gpio = devm_fwnode_get_index_gpiod_from_child( + &pdev->dev, "ext-amp-enable", 0, + codec_dev->fwnode, + GPIOD_OUT_LOW, "speaker-amp"); + if (IS_ERR(priv->ext_amp_gpio)) { + ret_val = PTR_ERR(priv->ext_amp_gpio); + switch (ret_val) { + case -ENOENT: + priv->ext_amp_gpio = NULL; + break; + default: + dev_err(&pdev->dev, "Failed to get ext-amp-enable GPIO: %d\n", + ret_val); + /* fall through */ + case -EPROBE_DEFER: + put_device(codec_dev); + return ret_val; + } + } + } + + put_device(codec_dev); log_quirks(&pdev->dev); -- 2.17.1

2 2

[alsa-devel] [PATCH 1/5] ASoC: rt5651: Fix workqueue cancel vs irq free race on remove
by Hans de Goede 10 Jul '18

10 Jul '18

On removal we must free the IRQ *before* cancelling the jack-detect work, so that the jack-detect work cannot be rescheduled by the IRQ. Before this commit we were cancelling the jack-detect work from the driver remove callback, while relying on devm to free the IRQ, which happens after the remove callback. This is the wrong order. This commit uses a devm-action to register a devm callback which cancels the work, before requesting the IRQ (devm tears things down in reverse order). This also allows us to remove the now empty remove driver callback. Cc: Carlo Caione <carlo(a)endlessm.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/rt5651.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 6b5669f3e85d..39d2c67cd064 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1696,6 +1696,13 @@ static irqreturn_t rt5651_irq(int irq, void *data) return IRQ_HANDLED; } +static void rt5651_cancel_work(void *data) +{ + struct rt5651_priv *rt5651 = data; + + cancel_work_sync(&rt5651->jack_detect_work); +} + static int rt5651_set_jack(struct snd_soc_component *component, struct snd_soc_jack *hp_jack, void *data) { @@ -2036,6 +2043,11 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, INIT_WORK(&rt5651->jack_detect_work, rt5651_jack_detect_work); + /* Make sure work is stopped on probe-error / remove */ + ret = devm_add_action_or_reset(&i2c->dev, rt5651_cancel_work, rt5651); + if (ret) + return ret; + ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5651, rt5651_dai, ARRAY_SIZE(rt5651_dai)); @@ -2043,15 +2055,6 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, return ret; } -static int rt5651_i2c_remove(struct i2c_client *i2c) -{ - struct rt5651_priv *rt5651 = i2c_get_clientdata(i2c); - - cancel_work_sync(&rt5651->jack_detect_work); - - return 0; -} - static struct i2c_driver rt5651_i2c_driver = { .driver = { .name = "rt5651", @@ -2059,7 +2062,6 @@ static struct i2c_driver rt5651_i2c_driver = { .of_match_table = of_match_ptr(rt5651_of_match), }, .probe = rt5651_i2c_probe, - .remove = rt5651_i2c_remove, .id_table = rt5651_i2c_id, }; module_i2c_driver(rt5651_i2c_driver); -- 2.17.1

3 11

[alsa-devel] [PATCH] dma: add support for reporting pause and resume separately
by Marek Szyprowski 09 Jul '18

09 Jul '18

'cmd_pause' DMA channel capability means that respective DMA engine supports both pausing and resuming given DMA channel. However, in some cases it is important to know if DMA channel can be paused without the need to resume it. This is a typical requirement for proper residue reading on transfer timeout in UART drivers. There are also some DMA engines with limited hardware, which doesn't really support resuming. Reporting pause and resume capabilities separately allows UART drivers to properly check for the really required capabilities and operate in DMA mode also in systems with limited DMA hardware. On the other hand drivers, which rely on full channel suspend/resume support, should now check for both 'pause' and 'resume' features. Existing clients of dma_get_slave_caps() have been checked and the only driver which rely on proper channel resuming is soc-generic-dmaengine-pcm driver, which has been updated to check the newly added capability. Existing 'cmd_pause' now only indicates that DMA engine support pausing given DMA channel. Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- This patch is a follow-up of the Vinod's advise from the following discussion: https://www.spinics.net/lists/linux-samsung-soc/msg63166.html --- drivers/dma/dmaengine.c | 8 ++------ include/linux/dmaengine.h | 5 ++++- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 3 files changed, 7 insertions(+), 8 deletions(-) diff --git a/drivers/dma/dmaengine.c b/drivers/dma/dmaengine.c index 08ba8473a284..84ac38dbdb65 100644 --- a/drivers/dma/dmaengine.c +++ b/drivers/dma/dmaengine.c @@ -500,12 +500,8 @@ int dma_get_slave_caps(struct dma_chan *chan, struct dma_slave_caps *caps) caps->max_burst = device->max_burst; caps->residue_granularity = device->residue_granularity; caps->descriptor_reuse = device->descriptor_reuse; - - /* - * Some devices implement only pause (e.g. to get residuum) but no - * resume. However cmd_pause is advertised as pause AND resume. - */ - caps->cmd_pause = !!(device->device_pause && device->device_resume); + caps->cmd_pause = !!device->device_pause; + caps->cmd_resume = !!device->device_resume; caps->cmd_terminate = !!device->device_terminate_all; return 0; diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index 861be5cab1df..c8c3a7a93802 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -415,7 +415,9 @@ enum dma_residue_granularity { * each type, the dma controller should set BIT(<TYPE>) and same * should be checked by controller as well * @max_burst: max burst capability per-transfer - * @cmd_pause: true, if pause and thereby resume is supported + * @cmd_pause: true, if pause is supported (i.e. for reading residue or + * for resume later) + * @cmd_resume: true, if resume is supported * @cmd_terminate: true, if terminate cmd is supported * @residue_granularity: granularity of the reported transfer residue * @descriptor_reuse: if a descriptor can be reused by client and @@ -427,6 +429,7 @@ struct dma_slave_caps { u32 directions; u32 max_burst; bool cmd_pause; + bool cmd_resume; bool cmd_terminate; enum dma_residue_granularity residue_granularity; bool descriptor_reuse; diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 56a541b9ff9e..76c46d793843 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -156,7 +156,7 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea ret = dma_get_slave_caps(chan, &dma_caps); if (ret == 0) { - if (dma_caps.cmd_pause) + if (dma_caps.cmd_pause && dma_caps.cmd_resume) hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT) hw.info |= SNDRV_PCM_INFO_BATCH; -- 2.17.1

4 6

Re: [alsa-devel] 答复: [PATCH] vgaswitchroo: set audio client id according to bound gpu client id
by Alex Deucher 09 Jul '18

09 Jul '18

On Mon, Jul 9, 2018 at 6:16 AM, Qu, Jim <Jim.Qu(a)amd.com> wrote: > Hi Lukas, > > Thanks to your explanation, and see comments in line. > > > Do you need to runtime resume the HDA controller even if user space isn't > streaming audio? Why, and in which situation exactly? > > Jim: OEM system uses pactl to queiry audio card and audio output sink, since the audio has power down by runtime pm, so the audio card and output sink are all unavailable. they could not select the available HDMI audio for audio playing. > > You're saying above that the HDA controller isn't runtime resumed on > hotplug of a display. Is that necessary to retrieve ELD or something? > I'm not sure if there's code in the HDA driver to acquire a runtime PM > ref on HPD, but maybe it's necessary. Or maybe the code is there but > somehow no HPD interrupt is received by the HDA driver? > > Jim: So far, I do not find any code about audio response HPD in kernel. the HPD interrupt will sent to user mode via uevent, not sure whether audio user mode driver can receive the event or not. On the gfx side at least, we can get a hotplug event via ACPI (depending on the OEM design) if displays are attached/detached while the dGPU is powered down. I suppose the gfx driver could call into the audio driver during one of those events. On the gfx side at least we just generate the gfx hotplug event and let userspace deal with it. Alex > > Thanks > JimQu > > ________________________________________ > 发件人: Lukas Wunner <lukas(a)wunner.de> > 发送时间: 2018年7月9日 17:27 > 收件人: Qu, Jim > 抄送: alsa-devel(a)alsa-project.org; dri-devel(a)lists.freedesktop.org; Deucher, Alexander; amd-gfx(a)lists.freedesktop.org > 主题: Re: [PATCH] vgaswitchroo: set audio client id according to bound gpu client id > > On Mon, Jul 09, 2018 at 08:52:33AM +0000, Qu, Jim wrote: >> Now, I found the audio device will auto suspend even if the gpu is active, >> and if I plug in a HDMI device it also do not resume back. >> >> 1. Did you encounter similar issue before? >> 2. audio will auto suspend as default at beginning of boot, is it expect >> behaviour? > > Yes, that's expected. Once you start streaming audio to attached displays, > user space opens the codec device and this causes the HDA controller to > runtime resume. If the discrete GPU is suspended at that moment, it will > be powered on and kept powered on as long as user space is streaming audio. > > Do you need to runtime resume the HDA controller even if user space isn't > streaming audio? Why, and in which situation exactly? > > You're saying above that the HDA controller isn't runtime resumed on > hotplug of a display. Is that necessary to retrieve ELD or something? > I'm not sure if there's code in the HDA driver to acquire a runtime PM > ref on HPD, but maybe it's necessary. Or maybe the code is there but > somehow no HPD interrupt is received by the HDA driver? > > Thanks, > > Lukas > _______________________________________________ > dri-devel mailing list > dri-devel(a)lists.freedesktop.org > https://lists.freedesktop.org/mailman/listinfo/dri-devel

5 12

Re: [alsa-devel] 答复: ??????: [PATCH] vgaswitchroo: set audio client id according to bound gpu client id
by Takashi Iwai 09 Jul '18

09 Jul '18

On Mon, 09 Jul 2018 17:53:19 +0200, Qu, Jim wrote: > > Hi All, > > Here, I want to clarify the audio device is bound to iGPU. There is no audio codec for dGPU. I'm confused. So you mean that the HDMI detection on Intel GPU doesn't work with the hybrid GPUs? And no audio codec on discrete GPU...? Why do we need vga_switcheroo at all, then? Takashi > > Thanks > JimQu > > -----邮件原件----- > 发件人: Lukas Wunner <lukas(a)wunner.de> > 发送时间: 2018年7月9日 23:48 > 收件人: Takashi Iwai <tiwai(a)suse.de> > 抄送: Alex Deucher <alexdeucher(a)gmail.com>; alsa-devel(a)alsa-project.org; amd-gfx(a)lists.freedesktop.org; Qu, Jim <Jim.Qu(a)amd.com>; dri-devel(a)lists.freedesktop.org; Deucher, Alexander <Alexander.Deucher(a)amd.com> > 主题: Re: [alsa-devel] ??????: [PATCH] vgaswitchroo: set audio client id according to bound gpu client id > > On Mon, Jul 09, 2018 at 05:04:22PM +0200, Takashi Iwai wrote: > > On Mon, 09 Jul 2018 15:58:51 +0200, Alex Deucher wrote: > > > On Mon, Jul 9, 2018 at 6:16 AM, Qu, Jim <Jim.Qu(a)amd.com> wrote: > > > > > You're saying above that the HDA controller isn't runtime > > > > > resumed on hotplug of a display. Is that necessary to retrieve ELD or something? > > > > > I'm not sure if there's code in the HDA driver to acquire a > > > > > runtime PM ref on HPD, but maybe it's necessary. Or maybe the > > > > > code is there but somehow no HPD interrupt is received by the HDA driver? > > > > > > > > So far, I do not find any code about audio response HPD in kernel. > > > > the HPD interrupt will sent to user mode via uevent, not sure > > > > whether audio user mode driver can receive the event or not. > > > > > > On the gfx side at least, we can get a hotplug event via ACPI > > > (depending on the OEM design) if displays are attached/detached > > > while the dGPU is powered down. I suppose the gfx driver could call > > > into the audio driver during one of those events. On the gfx side > > > at least we just generate the gfx hotplug event and let userspace deal with it. > > > > IMO, a more proper way would be to have the direct communication > > between the graphics driver and the audio driver like i915 driver > > does. Then the audio driver can get plug/unplug event at more > > accurate timing without races. > > Since v4.17, every time the GPU is powered up, the HDA controller is runtime resumed to PCI_D0. (See the call to pci_wakeup_bus() in > vga_switcheroo_runtime_resume() added by dcac86b7d0.) > > If the HDA controller can't detect presence of an HDMI display even if it's in PCI_D0, then I suppose that needs to be adressed in the HDA driver. Thus so far I don't see the need to call from amdgpu into the HDA driver. > > Thanks, > > Lukas

2 2

Re: [alsa-devel] 答复: [PATCH] vgaswitchroo: set audio client id according to bound gpu client id
by Alex Deucher 09 Jul '18

09 Jul '18

On Mon, Jul 9, 2018 at 6:16 AM, Qu, Jim <Jim.Qu(a)amd.com> wrote: > Hi Lukas, > > Thanks to your explanation, and see comments in line. > > > Do you need to runtime resume the HDA controller even if user space isn't > streaming audio? Why, and in which situation exactly? > > Jim: OEM system uses pactl to queiry audio card and audio output sink, since the audio has power down by runtime pm, so the audio card and output sink are all unavailable. they could not select the available HDMI audio for audio playing. Sounds like a bug in the HDA driver. The HDA driver should cache what it needs or power it back up when there is an ioctl/etc. that requires something that can't be cached. Alex > > You're saying above that the HDA controller isn't runtime resumed on > hotplug of a display. Is that necessary to retrieve ELD or something? > I'm not sure if there's code in the HDA driver to acquire a runtime PM > ref on HPD, but maybe it's necessary. Or maybe the code is there but > somehow no HPD interrupt is received by the HDA driver? > > Jim: So far, I do not find any code about audio response HPD in kernel. the HPD interrupt will sent to user mode via uevent, not sure whether audio user mode driver can receive the event or not. > > Thanks > JimQu > > ________________________________________ > 发件人: Lukas Wunner <lukas(a)wunner.de> > 发送时间: 2018年7月9日 17:27 > 收件人: Qu, Jim > 抄送: alsa-devel(a)alsa-project.org; dri-devel(a)lists.freedesktop.org; Deucher, Alexander; amd-gfx(a)lists.freedesktop.org > 主题: Re: [PATCH] vgaswitchroo: set audio client id according to bound gpu client id > > On Mon, Jul 09, 2018 at 08:52:33AM +0000, Qu, Jim wrote: >> Now, I found the audio device will auto suspend even if the gpu is active, >> and if I plug in a HDMI device it also do not resume back. >> >> 1. Did you encounter similar issue before? >> 2. audio will auto suspend as default at beginning of boot, is it expect >> behaviour? > > Yes, that's expected. Once you start streaming audio to attached displays, > user space opens the codec device and this causes the HDA controller to > runtime resume. If the discrete GPU is suspended at that moment, it will > be powered on and kept powered on as long as user space is streaming audio. > > Do you need to runtime resume the HDA controller even if user space isn't > streaming audio? Why, and in which situation exactly? > > You're saying above that the HDA controller isn't runtime resumed on > hotplug of a display. Is that necessary to retrieve ELD or something? > I'm not sure if there's code in the HDA driver to acquire a runtime PM > ref on HPD, but maybe it's necessary. Or maybe the code is there but > somehow no HPD interrupt is received by the HDA driver? > > Thanks, > > Lukas > _______________________________________________ > dri-devel mailing list > dri-devel(a)lists.freedesktop.org > https://lists.freedesktop.org/mailman/listinfo/dri-devel

2 1

[alsa-devel] [RESEND][PATCH] ASoC: hdac_hdmi: Program pin-port-mux during jack report event
by Sriram Periyasamy 09 Jul '18

09 Jul '18

During d3/d0 cycle, the connection selection index of all pins points to the default value. This needs to be restored to ensure audio is restored after d3/d0 cycle. So store the connection selection index and program it during jack report event which gets invoked in cases like d3/d0 cycle, hot plug detection when multiple displays are connected. Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> Acked-by: Vinod Koul <vkoul(a)kernel.org> --- sound/soc/codecs/hdac_hdmi.c | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 84f7a7a36e4b..7eb5a7138483 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -32,6 +32,7 @@ #include <sound/hda_chmap.h> #include "../../hda/local.h" #include "hdac_hdmi.h" +#include <sound/hda_regmap.h> #define NAME_SIZE 32 @@ -83,6 +84,7 @@ struct hdac_hdmi_pin { struct list_head head; hda_nid_t nid; bool mst_capable; + int conn_index; struct hdac_hdmi_port *ports; int num_ports; struct hdac_ext_device *edev; @@ -141,6 +143,9 @@ struct hdac_hdmi_priv { #define hdev_to_hdmi_priv(_hdev) ((to_ehdac_device(_hdev))->private_data) +static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev, + struct hdac_hdmi_port *port); + static struct hdac_hdmi_pcm * hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, struct hdac_hdmi_cvt *cvt) @@ -159,6 +164,7 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, struct hdac_hdmi_port *port, bool is_connect) { struct hdac_ext_device *edev = port->pin->edev; + int cmd, err; if (is_connect) snd_soc_dapm_enable_pin(port->dapm, port->jack_pin); @@ -166,6 +172,29 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, snd_soc_dapm_disable_pin(port->dapm, port->jack_pin); if (is_connect) { + /* set the device if pin is mst_capable */ + if (hdac_hdmi_port_select_set(edev, port) < 0) { + dev_err(&edev->hdev.dev, + "port %d device select fail\n", port->id); + return; + } + /* + * Restore the connection selection index of the + * respective pin. + */ + if (port->pin->conn_index > 0) { + cmd = snd_hdac_regmap_encode_verb(port->pin->nid, + AC_VERB_SET_CONNECT_SEL); + err = snd_hdac_regmap_write_raw(&edev->hdev, cmd, + port->pin->conn_index - 1); + if (err < 0) { + dev_err(&edev->hdev.dev, + "pin %d conn select index fail %d\n", + port->pin->nid, err); + return; + } + } + /* * Report Jack connect event when a device is connected * for the first time where same PCM is attached to multiple @@ -903,6 +932,9 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, } } + if (ucontrol->value.enumerated.item[0] > 0) + port->pin->conn_index = ucontrol->value.enumerated.item[0]; + /* * Jack status is not reported during device probe as the * PCMs are not registered by then. So report it here. -- 2.7.4

2 7

[alsa-devel] Applied "ASoC: adau171x1: Connect playback DAI to the DSP" to the asoc tree
by Mark Brown 09 Jul '18

09 Jul '18

The patch ASoC: adau171x1: Connect playback DAI to the DSP has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 5bea327962fa296efd16f2d3369dd339ddd7ce6f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen <lars(a)metafoo.de> Date: Fri, 6 Jul 2018 16:19:14 +0300 Subject: [PATCH] ASoC: adau171x1: Connect playback DAI to the DSP The playback DAI is connected to the DSP and the DSP might be sourcing signals from the playback stream. Add a DAPM route between the two to make sure that the playback DAI is powered up, when the DSP is active. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau17x1.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index ae41edd1c406..57169b8ff14e 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -299,6 +299,7 @@ static const struct snd_soc_dapm_route adau17x1_dsp_dapm_routes[] = { { "DSP", NULL, "Left Decimator" }, { "DSP", NULL, "Right Decimator" }, + { "DSP", NULL, "Playback" }, }; static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { -- 2.18.0.rc2

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[alsa-devel] [PATCH] ASoC: pxa: make SND_PXA_SOC_SSP depend on PLAT_PXA
by Arnd Bergmann 09 Jul '18

09 Jul '18

For the moment, we can't enable CONFIG_SND_PXA_SOC_SSP unless we are building for ARM PXA or MMP: WARNING: unmet direct dependencies detected for PXA_SSP Depends on [n]: PLAT_PXA [=n] Selected by [y]: - SND_PXA_SOC_SSP [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] This adds an explicit dependency for it. Fixes: 0a94cf345740 ("ASoC: pxa: make SND_PXA2XX_SOC_I2S selectable") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/pxa/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 2fc02c227f69..776e148b0aa2 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -34,6 +34,7 @@ config SND_PXA2XX_SOC_I2S config SND_PXA_SOC_SSP tristate "Soc Audio via PXA2xx/PXA3xx SSP ports" + depends on PLAT_PXA select PXA_SSP select SND_PXA2XX_LIB -- 2.9.0

2 1

[alsa-devel] [PATCH] sound: stm32: replace "%p" with "%pK"
by Benjamin Gaignard 09 Jul '18

09 Jul '18

The format specifier "%p" can leak kernel addresses. Use "%pK" instead. Signed-off-by: Benjamin Gaignard <benjamin.gaignard(a)st.com> --- sound/soc/stm/stm32_adfsdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index db73fef3e500..0e9373064032 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -149,7 +149,7 @@ static int stm32_afsdm_pcm_cb(const void *data, size_t size, void *private) unsigned int old_pos = priv->pos; unsigned int cur_size = size; - dev_dbg(rtd->dev, "%s: buff_add :%p, pos = %d, size = %zu\n", + dev_dbg(rtd->dev, "%s: buff_add :%pK, pos = %d, size = %zu\n", __func__, &pcm_buff[priv->pos], priv->pos, size); if ((priv->pos + size) > buff_size) { -- 2.15.0

3 3

[alsa-devel] How to add new sound card with ASoC generic driver
by Riccardo Casagrande 09 Jul '18

09 Jul '18

Hello, I'm new to ALSA and linux in general, I need help from who knows well ALSA due to the tight schedule of the project. I'm working on a Yocto project on NXP i.MX6Quad. The hardware architetture is this: the i.MAX6Quad has 6 output connected to 3 ST FDA2100 and 2 input from ADC. In the first step what I need is simply to enable 6 output and send I2S audio to FDAs. In the second step is required a clock for ADC and enable input ports. I spent days/weeks reading docs, examples etc without success. I understood that working on device tree file and setting a simple audio card with generic codec properly is the answer, but my knowledge is almost nothing. I followed these docs/thread https://elixir.bootlin.com/linux/latest/source/Documentation/devicetree/bin… https://community.nxp.com/thread/475428 but compiling I have syntax errors on device tree file. There is someone who can help me step by step? Thanks in advance Riccardo

1 0

[alsa-devel] Binding volume and mute switch
by Ladislav Michl 09 Jul '18

09 Jul '18

Hello, to provide almost self descriptive controls to alsamixer I'd like to achive something like this: http://ladislav.michl.sweb.cz/alsamixer1.png Master volume is muted using switch coupled with master volume control, microphone is muted using switch coupled with microphone volume control, etc... For mic and line input is works as expected, but for sidetone and master volume I had to use "proper" names for mixer controls, see bellow: /* Input mixer */ static const struct snd_kcontrol_new max9867_input_mixer_controls[] = { SOC_DAPM_DOUBLE("Mic Switch", MAX9867_INPUTCONFIG, 6, 4, 1, 0), SOC_DAPM_DOUBLE("Line Switch", MAX9867_INPUTCONFIG, 7, 5, 1, 0), }; /* Output mixer */ static const struct snd_kcontrol_new max9867_output_mixer_controls[] = { SOC_DAPM_DOUBLE_R("Line Bypass Switch", MAX9867_LEFTLINELVL, MAX9867_RIGHTLINELVL, 6, 1, 1), }; /* Sidetone mixer */ static const struct snd_kcontrol_new max9867_sidetone_mixer_controls[] = { SOC_DAPM_DOUBLE("Sidetone Switch", MAX9867_SIDETONE, 6, 7, 1, 0), }; /* Line out switch */ static const struct snd_kcontrol_new max9867_line_out_control = SOC_DAPM_DOUBLE_R("Switch", MAX9867_LEFTVOL, MAX9867_RIGHTVOL, 6, 1, 1); static const struct snd_soc_dapm_widget max9867_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MICL"), SND_SOC_DAPM_INPUT("MICR"), SND_SOC_DAPM_INPUT("LINL"), SND_SOC_DAPM_INPUT("LINR"), SND_SOC_DAPM_MIXER("Left Input Mixer", MAX9867_PWRMAN, 6, 1, max9867_input_mixer_controls, ARRAY_SIZE(max9867_input_mixer_controls)), SND_SOC_DAPM_MIXER("Right Input Mixer", MAX9867_PWRMAN, 5, 1, max9867_input_mixer_controls, ARRAY_SIZE(max9867_input_mixer_controls)), SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", MAX9867_PWRMAN, 1, 0), SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", MAX9867_PWRMAN, 0, 0), SND_SOC_DAPM_MIXER("Digital", SND_SOC_NOPM, 0, 0, max9867_sidetone_mixer_controls, ARRAY_SIZE(max9867_sidetone_mixer_controls)), SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, max9867_output_mixer_controls, ARRAY_SIZE(max9867_output_mixer_controls)), SND_SOC_DAPM_DAC("DACL", "HiFi Playback", MAX9867_PWRMAN, 3, 0), SND_SOC_DAPM_DAC("DACR", "HiFi Playback", MAX9867_PWRMAN, 2, 0), SND_SOC_DAPM_SWITCH("Master Playback", SND_SOC_NOPM, 0, 0, &max9867_line_out_control), SND_SOC_DAPM_OUTPUT("LOUT"), SND_SOC_DAPM_OUTPUT("ROUT"), }; static const struct snd_soc_dapm_route max9867_audio_map[] = { {"Left Input Mixer", "Mic Switch", "MICL"}, {"Right Input Mixer", "Mic Switch", "MICR"}, {"Left Input Mixer", "Line Switch", "LINL"}, {"Right Input Mixer", "Line Switch", "LINR"}, {"ADCL", NULL, "Left Input Mixer"}, {"ADCR", NULL, "Right Input Mixer"}, {"Digital", "Sidetone Switch", "ADCL"}, {"Digital", "Sidetone Switch", "ADCR"}, {"DACL", NULL, "Digital"}, {"DACR", NULL, "Digital"}, {"Output Mixer", "Line Bypass Switch", "LINL"}, {"Output Mixer", "Line Bypass Switch", "LINR"}, {"Output Mixer", NULL, "DACL"}, {"Output Mixer", NULL, "DACR"}, {"Master Playback", "Switch", "Output Mixer"}, {"LOUT", NULL, "Master Playback"}, {"ROUT", NULL, "Master Playback"}, }; With other control names than "Digital" for sidetone digital mixer, sidetone switch is shown separately from sidetone volume. The same for master volume switch, which is bind with master volume only once its name is "Master Playback". Is it intended to work this way or there is some other, more elegant way how to bind mute switch with volume control. (above solution seem too fragile to me) Thank you, ladis

1 0

[alsa-devel] Hiding some of the controls defined in codec driver from machine driver context
by Giedrius Trainavičius 09 Jul '18

09 Jul '18

Hello, Is it possible to 'hide' some of the controls of the codec in the machine driver? The codec I'm using exposes many advanced controls which are useful for board developers, but not for the end user, so I'd like to define my own 'simplified' controls, and take care of board specific logic internally in the machine driver. I found this old discussion on the topic: https://www.spinics.net/lists/alsa-devel/msg36386.html - I'm targeting kernel 4.14, maybe there's already some means to achieve this? Thank you! Giedrius.

2 1

Re: [alsa-devel] [PATCH] vgaswitchroo: set audio client id according to bound gpu client id
by Lukas Wunner 09 Jul '18

09 Jul '18

On Mon, Jul 09, 2018 at 08:52:33AM +0000, Qu, Jim wrote: > Now, I found the audio device will auto suspend even if the gpu is active, > and if I plug in a HDMI device it also do not resume back. > > 1. Did you encounter similar issue before? > 2. audio will auto suspend as default at beginning of boot, is it expect > behaviour? Yes, that's expected. Once you start streaming audio to attached displays, user space opens the codec device and this causes the HDA controller to runtime resume. If the discrete GPU is suspended at that moment, it will be powered on and kept powered on as long as user space is streaming audio. Do you need to runtime resume the HDA controller even if user space isn't streaming audio? Why, and in which situation exactly? You're saying above that the HDA controller isn't runtime resumed on hotplug of a display. Is that necessary to retrieve ELD or something? I'm not sure if there's code in the HDA driver to acquire a runtime PM ref on HPD, but maybe it's necessary. Or maybe the code is there but somehow no HPD interrupt is received by the HDA driver? Thanks, Lukas

1 0

[alsa-devel] Getting a notification of snd_kcontrol value change, defined in codec, in machine driver
by Giedrius Trainavičius 08 Jul '18

08 Jul '18

Hey, What would be the best way in machine driver to learn about changes of alsa mixer control values? The controls are defined in the codec driver. The intended use case would be to use the current volume setting for the codec on the board to forward to a dedicated microcontroller for visual feedback. I am aware that this could be bridged over via a user space program, but I will have to forward the current PCM stream format to the MCU anyway from hw_params, so ideally I would like all of this functionality contained in the machine driver. Thank you, Giedrius.

3 3

[alsa-devel] [PATCH] slimbus: stream: Fix htmldocs warnings
by Srinivas Kandagatla 07 Jul '18

07 Jul '18

This patch fixes below warning during building htmldoc: slimbus.h:352: warning: Function parameter or member 'name' not described in 'slim_stream_runtime' This patch also removes documentation for state variable in struct slim_stream_runtime which was redundant and removed. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/slimbus/slimbus.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/drivers/slimbus/slimbus.h b/drivers/slimbus/slimbus.h index 411880f3fc0a..32ac4046154f 100644 --- a/drivers/slimbus/slimbus.h +++ b/drivers/slimbus/slimbus.h @@ -325,9 +325,8 @@ enum slim_transport_protocol { /** * struct slim_stream_runtime - SLIMbus stream runtime instance * - * @dev: Name of the stream + * @name: Name of the stream * @dev: SLIM Device instance associated with this stream - * @state: state of stream * @direction: direction of stream * @prot: Transport protocol used in this stream * @rate: Data rate of samples * -- 2.16.2

1 0

[alsa-devel] Bad header guard in alsa-lib
by Guillaume Blanc 06 Jul '18

06 Jul '18

Hello, While building alsa-lib, I got a redefinition error caused by a bad header guard in pcm_plugin.h. I assume this is a mistake and not a weird include trick, attached is a patch to fix this. Best regards, Guillaume Blanc

2 1

[alsa-devel] [PATCH] ASoC: nau8810: use 64-bit arithmetic instead of 32-bit
by Gustavo A. R. Silva 06 Jul '18

06 Jul '18

Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: 256 * fs * 4 * nau8810_mclk_scaler[i] / 10 Addresses-Coverity-ID: 1357595 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/nau8810.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index bfd74b8..e7fd0b2 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -505,7 +505,7 @@ static int nau88l0_calc_pll(unsigned int pll_in, f2_max = 0; scal_sel = ARRAY_SIZE(nau8810_mclk_scaler); for (i = 0; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { - f2 = 256 * fs * 4 * nau8810_mclk_scaler[i] / 10; + f2 = 256ULL * fs * 4 * nau8810_mclk_scaler[i] / 10; if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && f2_max < f2) { f2_max = f2; -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - two more lenovo models need fixup of MIC_LOCATION
by Hui Wang 06 Jul '18

06 Jul '18

We have two new lenovo desktop models which need to apply the fixup of ALC294_FIXUP_LENOVO_MIC_LOCATION, and they have the same pin cfg as the machine with subsystem id:0x17aa3136, now use the pincfg table to apply the fixup for them. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 63154b9..666713e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6556,7 +6556,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x312a, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), - SND_PCI_QUIRK(0x17aa, 0x3136, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), @@ -6822,6 +6821,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1a, 0x02a11040}, {0x1b, 0x01014020}, {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC294_FIXUP_LENOVO_MIC_LOCATION, + {0x14, 0x90170110}, + {0x19, 0x02a11020}, + {0x1a, 0x02a11030}, + {0x21, 0x0221101f}), SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60140}, {0x14, 0x90170110}, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: nau8824: use 64-bit arithmetic instead of 32-bit
by Gustavo A. R. Silva 06 Jul '18

06 Jul '18

Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: 256 * fs * 2 * mclk_src_scaling[i].param Addresses-Coverity-ID: 1432039 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/nau8824.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 6bd1445..468d514 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1274,7 +1274,7 @@ static int nau8824_calc_fll_param(unsigned int fll_in, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; -- 2.7.4

3 2

[alsa-devel] [PATCH] regmap: slimbus: add support to raw read/writes
by Srinivas Kandagatla 05 Jul '18

05 Jul '18

SLIMbus supports upto 16 bytes in value management messages, so add support to raw read/writes upto 16 bytes. Also useful for paged register access on SLIMbus interfaced codecs. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/base/regmap/regmap-slimbus.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) diff --git a/drivers/base/regmap/regmap-slimbus.c b/drivers/base/regmap/regmap-slimbus.c index 91d501eda8a9..fc07518666b2 100644 --- a/drivers/base/regmap/regmap-slimbus.c +++ b/drivers/base/regmap/regmap-slimbus.c @@ -31,11 +31,30 @@ static int regmap_slimbus_byte_reg_write(void *context, unsigned int reg, return slim_writeb(sdev, reg, val); } +static int regmap_slimbus_write(void *context, const void *data, size_t count) +{ + struct slim_device *sdev = context; + + return slim_write(sdev, *(u16 *)data, count - 2, (u8 *)data + 2); +} + +static int regmap_slimbus_read(void *context, const void *reg, size_t reg_size, + void *val, size_t val_size) +{ + struct slim_device *sdev = context; + + return slim_read(sdev, *(u16 *)reg, val_size, val); +} + static struct regmap_bus regmap_slimbus_bus = { .reg_write = regmap_slimbus_byte_reg_write, .reg_read = regmap_slimbus_byte_reg_read, - .reg_format_endian_default = REGMAP_ENDIAN_LITTLE, - .val_format_endian_default = REGMAP_ENDIAN_LITTLE, + .max_raw_write = 16, + .max_raw_read = 16, + .write = regmap_slimbus_write, + .read = regmap_slimbus_read, + .reg_format_endian_default = REGMAP_ENDIAN_NATIVE, + .val_format_endian_default = REGMAP_ENDIAN_NATIVE, }; static const struct regmap_bus *regmap_get_slimbus(struct slim_device *slim, -- 2.16.2

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[alsa-devel] Can't detect "XRUNs"
by Raphaël Dingé 05 Jul '18

05 Jul '18

Hi, I don't get audio drop notification "XRUNs" in /proc/kmsg. Not sure if this is the right mailing list as maybe my question is more kernel related, but I'll try a shot here. I'm currently making a proof of concept to see if I can run a quite CPU hungry audio application running without audio drops over time given a set of tests. In particular I want to automate playback buffer underruns detection so that I don't have to sit and listen closely when they happen. With some very extreme tests, I can hear that there are audio drops all over the place. I have 2 audio cards in the system, one is audio over HDMI, and the other one is an USB audio card that supports audio class definition. In both case the sound is working as I would expect. The system itself is an Intel NUC i7. I configured the kernel with CONFIG_SND_DEBUG=y, CONFIG_SND_PCM_XRUN_DEBUG=y and CONFIG_SND_VERBOSE_PROCFS=y as well as echo 5 > /proc/asound/cardX/pcmYp/xrun_debug with X and Y the card and pcm stream of interest, and so hw:X,Y when opening the device with snd_pcm_open. Then I watch the output of the ksyslog interface using cat /proc/kmsg When an audio drop occurs, there is nothing about XRUNs or whatever it is called. I don't even know what kind of log message it should look like, and can't find it on google as well. With audio over HDMI, I have some occurence of ALSA: PCM: [Q] Lost interrupts? that seems more of less related to audio drops, but not always. With the USB audio card, I have nothing. I tried to write to the xrun_injection proc file in the stream, but nothing happens as well. Finally I log any occurence of snd_pcm_writei returning a negative number, but that never happens as well. Am I missing something completely? Thanks for any information on that matter, Raphael

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[alsa-devel] [PATCH v2 0/2] slimbus: Add Stream Support
by Srinivas Kandagatla 05 Jul '18

05 Jul '18

Thanks to Vinod and Stephen for reviewing v1 patchset This v2 patchset adds basic stream support for SLIMbus devices and controllers. Mostly inspired by soundwire stream patches. But slimbus stream is much simpler compared to soundwire >From slim_device side, we have below 6 new apis. slim_stream_allocate() - allocating runtime slim stream slim_stream_prepare() - to configure runtime stream with config slim_stream_enable() - enable channels/ports for data slim_stream_disable() - disable channels/ports. slim_stream_unprepare() - un configure runtime stream slim_stream_free() - free all associated memory. >From Controller side: Data channel Management and reconfiguration messages are applicable for all the controllers which are inline with SLIMbus specs. However light weight controller like NGD which have user specific implementation of some messages need to be invoked instead of standard message commands. For usecases like this simple enable/disable stream apis are provided. Assumptions: 1> Current design assumes that the channel and ports are statically allocated to the device during SoC integration, which is the case with all the Qualcomm SoCs. 2> One-to-One mapping between Port and Channel, SLIMBus versions earlier than 2.0 has only one endpoint per port. Current patchset can be extended to support more than one endpoints per port. 3> Only audio usecase, This patchset only supports Isochronous and Push/Pull transport protocols, which are sufficient for audio use cases. 4> DSP does all the data handling for the allocated channels. Which is true for Qcom SoCs. TODO: Bandwidth management. Dependency: This patchset has dependency on the NGD driver https://patchwork.kernel.org/patch/10474959/ Tested this patchset with WCD9335 codec playback and capture on DB820c using 4.18-rc1 with qdsp6 + slim tx patches merged in sound.git. I have pushed my working branch to [1] incase someone want to try. [1]:https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=slimb… Changes since v1 (https://lwn.net/Articles/758100/): - removed redundant state machine checks for audio usecases, suggested by Vinod. - fixed few typos. - removed irqsave/restore spinlock variant spotted by Vinod. - general cleanups suggested by Stephen and Vinod. thanks, srini Srinivas Kandagatla (2): slimbus: stream: add stream support slimbus: ngd: add stream support Documentation/driver-api/slimbus.rst | 5 + drivers/slimbus/Makefile | 2 +- drivers/slimbus/core.c | 2 + drivers/slimbus/qcom-ngd-ctrl.c | 149 ++++++++++- drivers/slimbus/slimbus.h | 189 ++++++++++++++ drivers/slimbus/stream.c | 477 +++++++++++++++++++++++++++++++++++ include/linux/slimbus.h | 46 ++++ 7 files changed, 867 insertions(+), 3 deletions(-) create mode 100644 drivers/slimbus/stream.c -- 2.16.2

1 2

[alsa-devel] [PATCH - JACK IO plug v2 0/3] snd_pcm_drain support
by twischer＠de.adit-jv.com 05 Jul '18

05 Jul '18

Now in a clean manner as composed patches. pcm: ioplug: Provide avail helper function --------- No changes jack: Avoid call to snd_pcm_avail_update() from JACK thread --------- v2 uses snd_pcm_uframes_t as type for avail and does not check for avail < 0 becasue snd_pcm_ioplug_avail() will never retrun a value < 0 jack: Update poll_fd also in draining state --------- This patch is depending on [1]. Therefore it was rebased on top of [1]. Additionally the commit message was extended. v1 of this patch can be found in [2]. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-July/137775.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-July/137769.html

3 7

[alsa-devel] [PATCH] ASoC: fsl_spdif: Use 64-bit arithmetic instead of 32-bit
by Gustavo A. R. Silva 05 Jul '18

05 Jul '18

Add suffix ULL to constant 64 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: rate[index] * txclk_df * 64 Addresses-Coverity-ID: 1222129 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/fsl/fsl_spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 9b59d87..740b90d 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1118,7 +1118,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) { for (txclk_df = 1; txclk_df <= 128; txclk_df++) { - rate_ideal = rate[index] * txclk_df * 64; + rate_ideal = rate[index] * txclk_df * 64ULL; if (round) rate_actual = clk_round_rate(clk, rate_ideal); else -- 2.7.4

3 2

[alsa-devel] [PATCH 1/1] ASoC: rsnd: cmd: Add missing newline to debug message
by jiada_wang＠mentor.com 05 Jul '18

05 Jul '18

From: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> To comply with the style of all kernel messages, add newline to the end of every message. Fixes: 70fb10529f61 ("ASoC: rsnd: add MIX (Mixer) support") Signed-off-by: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/cmd.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index d8043ad33540..cc191cd5fb82 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -86,7 +86,7 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, cmd_case[rsnd_mod_id(src)] << 16; } - dev_dbg(dev, "ctu/mix path = 0x%08x", data); + dev_dbg(dev, "ctu/mix path = 0x%08x\n", data); rsnd_mod_write(mod, CMD_ROUTE_SLCT, data); rsnd_mod_write(mod, CMD_BUSIF_MODE, rsnd_get_busif_shift(io, mod) | 1); -- 2.17.0

3 2

[alsa-devel] [PATCH] ASoC: pxa-ssp: add support for an external clock in devicetree
by Daniel Mack 05 Jul '18

05 Jul '18

Allow setting a clock called 'extclk' in the device of the ssp-dai device. If specified, this clock will be set to the mclk rate from the DAI's .set_sysclk() callback. The DAI will also configure itself to use that external clock. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- .../bindings/sound/mrvl,pxa-ssp.txt | 8 ++++++ sound/soc/pxa/pxa-ssp.c | 25 +++++++++++++++++++ 2 files changed, 33 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt index efd3fb1f68d2..feef39b4a4fd 100644 --- a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt +++ b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt @@ -5,6 +5,14 @@ Required properties: compatible Must be "mrvl,pxa-ssp-dai" port A phandle reference to a PXA ssp upstream device +Optional properties: + + clock-names + clocks Through "clock-names" and "clocks", external clocks + can be configured. If a clock names "extclk" exists, + it will be set to the mclk rate of the audio stream + and be used as clock provider of the DAI. + Example: /* upstream device */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index f8339bb01251..c43fe3b9c79c 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -41,6 +41,7 @@ */ struct ssp_priv { struct ssp_device *ssp; + struct clk *extclk; unsigned long ssp_clk; unsigned int sysclk; unsigned int dai_fmt; @@ -205,6 +206,21 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + if (priv->extclk) { + int ret; + + /* + * For DT based boards, if an extclk is given, use it + * here and configure PXA_SSP_CLK_EXT. + */ + + ret = clk_set_rate(priv->extclk, freq); + if (ret < 0) + return ret; + + clk_id = PXA_SSP_CLK_EXT; + } + dev_dbg(&ssp->pdev->dev, "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", cpu_dai->id, clk_id, freq); @@ -773,6 +789,15 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai) ret = -ENODEV; goto err_priv; } + + priv->extclk = devm_clk_get(dev, "extclk"); + if (IS_ERR(priv->extclk)) { + ret = PTR_ERR(priv->extclk); + if (ret == -EPROBE_DEFER) + return ret; + + priv->extclk = NULL; + } } else { priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio"); if (priv->ssp == NULL) { -- 2.17.1

2 1

[alsa-devel] [PATCH v3] ALSA: core: Allow drivers to set R/W wait time.
by Liam Girdwood 04 Jul '18

04 Jul '18

Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO. This needs to be configurable for modern hardware like DSPs where no pointer update in milliseconds can indicate terminal DSP errors. Add a substream variable to set the wait time in ms. This allows userspace and drivers to recover more quickly from terminal DSP errors. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Changes since V1 :- o Remove API method and alow drivers to set directly. o Validate time is driver supplied times. V2 :- o Max wait calc now in ms. o checkpatch. include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 12 +++++++++--- 2 files changed, 10 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e054c583d3b3..fcdf358a25f0 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -462,6 +462,7 @@ struct snd_pcm_substream { /* -- timer section -- */ struct snd_timer *timer; /* timer */ unsigned timer_running: 1; /* time is running */ + long wait_time; /* time in ms for R/W to wait for avail */ /* -- next substream -- */ struct snd_pcm_substream *next; /* -- linked substreams -- */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 44b5ae833082..82a41e4d7170 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1832,12 +1832,18 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { - wait_time = 10; + /* use wait time from substream if available */ + if (substream->wait_time) + wait_time = substream->wait_time; + else + wait_time = 10 * 1000; /* 10 secs */ + if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; + long t = runtime->period_size * 2 / + (runtime->rate / 1000); wait_time = max(t, wait_time); } - wait_time = msecs_to_jiffies(wait_time * 1000); + wait_time = msecs_to_jiffies(wait_time); } for (;;) { -- 2.17.1

2 2

[alsa-devel] [PATCH] ASoC: nau8825: use 64-bit arithmetic instead of 32-bit
by Gustavo A. R. Silva 04 Jul '18

04 Jul '18

Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: 256 * fs * 2 * mclk_src_scaling[i].param Addresses-Coverity-ID: 1339616 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/nau8825.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index dc6ea49..b9fed99 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2016,7 +2016,7 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; -- 2.7.4

2 1

[alsa-devel] [PATCH 0/4] ASoC: Use snd_pcm_stop_xrun() helper
by Takashi Iwai 04 Jul '18

04 Jul '18

Hi, this is a series of trivial fixes to replace some open codes with the standard snd_pcm_stop_xrun() helper. Functionally same, but it addresses the potential stream lock issues automagically, too. Takashi === Takashi Iwai (4): ASoC: davinci: Use snd_pcm_stop_xrun() helper ASoC: qcom: Use snd_pcm_stop_xrun() helper ASoC: sti: Use snd_pcm_stop_xrun() helper ASoC: stm32: Use snd_pcm_stop_xrun() helper sound/soc/davinci/davinci-mcasp.c | 16 ++++------------ sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/sti/uniperif_player.c | 6 +++--- sound/soc/sti/uniperif_reader.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 7 ++----- 5 files changed, 11 insertions(+), 22 deletions(-) -- 2.18.0

2 8

[alsa-devel] [PATCH 0/6] ASoC: qdsp6: Add missing slimbus capture support
by Srinivas Kandagatla 04 Jul '18

04 Jul '18

This patchset adds support to slimbus tx ports plus 3 trival fixes to do with closing port when its not really open. slim tx support invoving add dais and routing, rest of the dsp configuration commands are already available in mainline. slim tx is verified with wcd9335 codec on DB820c. Thanks, srini Srinivas Kandagatla (6): ASoC: qdsp6: q6afe: Add missing slimbus capture ports ASoC: qdsp6: q6afe-dai: add support to slim tx dais ASoC: qdsp6: q6routing: add slim rx routings ASoC: qdsp6: q6afe-dai: Do not overwrite slim dai num_channels ASoC: qdsp6: q6afe-dai: do not close port if its not opened ASoC: qdsp6: q6asm-dai: do not close port if its not opened sound/soc/qcom/qdsp6/q6afe-dai.c | 179 +++++++++++++++++++++++++++++++++++---- sound/soc/qcom/qdsp6/q6afe.c | 21 +++++ sound/soc/qcom/qdsp6/q6asm-dai.c | 4 +- sound/soc/qcom/qdsp6/q6routing.c | 28 ++++++ 4 files changed, 216 insertions(+), 16 deletions(-) -- 2.16.2

2 12

[alsa-devel] [PATCH - PCM rate v2 1/1] pcm: rate: Add error check for snd_pcm_avail_update()
by twischer＠de.adit-jv.com 04 Jul '18

04 Jul '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without these changes a negative error code returned by snd_pcm_avail_update() will be not handled correctly. With this patch the returned error code of snd_pcm_avail_update() will be returned by snd_pcm_rate_avail_update(). Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- Compared to v1[1] of this patch only the changelog and the subject has changed. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-July/137838.html diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 4e0c7ca..031a43d 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -970,9 +970,12 @@ static snd_pcm_sframes_t snd_pcm_rate_avail_update(snd_pcm_t *pcm) { snd_pcm_rate_t *rate = pcm->private_data; snd_pcm_t *slave = rate->gen.slave; - snd_pcm_uframes_t slave_size; + snd_pcm_sframes_t slave_size; slave_size = snd_pcm_avail_update(slave); + if (slave_size < 0) + return slave_size; + if (pcm->stream == SND_PCM_STREAM_CAPTURE) goto _capture; snd_pcm_rate_sync_hwptr(pcm); -- 2.7.4

2 1

[alsa-devel] [PATCH 1/2] ALSA: pcm: trace XRUN event at injection, too
by Takashi Iwai 04 Jul '18

04 Jul '18

The PCM xrun injection triggers directly snd_pcm_stop() without the standard xrun handler, hence it's not recorded on the event buffer. Ditto for snd_pcm_stop_xrun() call and SNDRV_PCM_IOCTL_XRUN ioctl. They are inconvenient from the debugging POV. Let's make them to trigger XRUN via the standard helper more consistently. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm.c | 2 +- sound/core/pcm_lib.c | 7 ++++--- sound/core/pcm_local.h | 2 ++ sound/core/pcm_native.c | 8 ++++---- 4 files changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index c352bfb973cc..6f037a4b8b52 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -497,7 +497,7 @@ static void snd_pcm_xrun_injection_write(struct snd_info_entry *entry, snd_pcm_stream_lock_irq(substream); runtime = substream->runtime; if (runtime && runtime->status->state == SNDRV_PCM_STATE_RUNNING) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + __snd_pcm_xrun(substream); snd_pcm_stream_unlock_irq(substream); } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 44b5ae833082..c1d2e8e1fc6b 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -153,7 +153,8 @@ EXPORT_SYMBOL(snd_pcm_debug_name); dump_stack(); \ } while (0) -static void xrun(struct snd_pcm_substream *substream) +/* call with stream lock held */ +void __snd_pcm_xrun(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -201,7 +202,7 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, } } else { if (avail >= runtime->stop_threshold) { - xrun(substream); + __snd_pcm_xrun(substream); return -EPIPE; } } @@ -297,7 +298,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, } if (pos == SNDRV_PCM_POS_XRUN) { - xrun(substream); + __snd_pcm_xrun(substream); return -EPIPE; } if (pos >= runtime->buffer_size) { diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h index 7a499d02df6c..c515612969a4 100644 --- a/sound/core/pcm_local.h +++ b/sound/core/pcm_local.h @@ -65,4 +65,6 @@ static inline void snd_pcm_timer_init(struct snd_pcm_substream *substream) {} static inline void snd_pcm_timer_done(struct snd_pcm_substream *substream) {} #endif +void __snd_pcm_xrun(struct snd_pcm_substream *substream); + #endif /* __SOUND_CORE_PCM_LOCAL_H */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index cecc79772c94..20174d0c0527 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1337,13 +1337,12 @@ int snd_pcm_drain_done(struct snd_pcm_substream *substream) int snd_pcm_stop_xrun(struct snd_pcm_substream *substream) { unsigned long flags; - int ret = 0; snd_pcm_stream_lock_irqsave(substream, flags); if (snd_pcm_running(substream)) - ret = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + __snd_pcm_xrun(substream); snd_pcm_stream_unlock_irqrestore(substream, flags); - return ret; + return 0; } EXPORT_SYMBOL_GPL(snd_pcm_stop_xrun); @@ -1591,7 +1590,8 @@ static int snd_pcm_xrun(struct snd_pcm_substream *substream) result = 0; /* already there */ break; case SNDRV_PCM_STATE_RUNNING: - result = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + __snd_pcm_xrun(substream); + result = 0; break; default: result = -EBADFD; -- 2.18.0

1 1

[alsa-devel] [PATCH - PCM rate 1/1] pcm: rate: Use signed variable for return of snd_pcm_avail_update
by twischer＠de.adit-jv.com 04 Jul '18

04 Jul '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this changes a negative error code returned by snd_pcm_avail_update() will be not handled correctly Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 4e0c7ca..031a43d 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -970,9 +970,12 @@ static snd_pcm_sframes_t snd_pcm_rate_avail_update(snd_pcm_t *pcm) { snd_pcm_rate_t *rate = pcm->private_data; snd_pcm_t *slave = rate->gen.slave; - snd_pcm_uframes_t slave_size; + snd_pcm_sframes_t slave_size; slave_size = snd_pcm_avail_update(slave); + if (slave_size < 0) + return slave_size; + if (pcm->stream == SND_PCM_STREAM_CAPTURE) goto _capture; snd_pcm_rate_sync_hwptr(pcm); -- 2.7.4

2 1

[alsa-devel] [PATCH v3 00/13] Enable HDA Codec support on Intel Platforms
by Pierre-Louis Bossart 04 Jul '18

04 Jul '18

Many Intel platforms (SKL, KBL) etc. in the market supports enhanced audio capabilities which also includes DSP processing. The default HDaudio legacy driver does not allow for the use of the DSP, this patch set makes it possible while reusing existing code for HDAudio codecs and without significant changes to the legacy driver. This v3 is not split into two batches as done for v1 and v2, but keeps the same logical progression. The first three patches are mostly data structure changes, the DSP support capability is added then with an ASoC HDA driver and the last patches are fixes required for Skylake+. The changes to the HDAudio legacy driver are minimal. Tests were run successfully on multiple platforms (Dell XPS13, KBL NUC, APL NUC and LeafHill reference board). Credits: all the initial code was written by Rakesh Ughreja, the rebase to broonie/for-next, cleanups and additional tests were done by Pierre Bossart. Changes v3: - port to component model - additional tests on ApolloLake and KabyLake NUC devices - cleanups (alignment, typos, etc) Changes v2: - Resolved review comments and rebased to latest kernel. - added module load support for codec drivers. Rakesh Ughreja (13): ALSA: hdac: Remove usage of struct hdac_ext_device and use hdac_device instead ALSA: hdac: Remove usage of struct hdac_ext_bus and use hdac_bus instead ALSA: hdac: Remove usage of struct hdac_ext_driver, use hdac_driver instead ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs ASoC: Intel: Skylake: Add entry in sst_acpi_mach for HDA codecs ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ALSA: hda: split snd_hda_codec_new function ALSA: hdac: remove memory allocation from snd_hdac_ext_bus_device_init ALSA: hdac: add extended ops in the hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers ALSA: hdac: ext: add wait for codec to respond after link reset ASoC: Intel: Skylake: fix widget handling include/sound/hdaudio.h | 29 ++ include/sound/hdaudio_ext.h | 123 ++----- sound/hda/ext/hdac_ext_bus.c | 80 ++--- sound/hda/ext/hdac_ext_controller.c | 64 ++-- sound/hda/ext/hdac_ext_stream.c | 104 +++--- sound/pci/hda/hda_bind.c | 12 + sound/pci/hda/hda_codec.c | 68 +++- sound/pci/hda/hda_codec.h | 2 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 469 +++++++++++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 24 ++ sound/soc/codecs/hdac_hdmi.c | 416 +++++++++++------------- sound/soc/intel/boards/Kconfig | 9 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 133 ++++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 37 +++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 174 ++++++++++ sound/soc/intel/skylake/skl-messages.c | 50 ++- sound/soc/intel/skylake/skl-nhlt.c | 8 +- sound/soc/intel/skylake/skl-pcm.c | 182 +++++++---- sound/soc/intel/skylake/skl-topology.c | 23 +- sound/soc/intel/skylake/skl-topology.h | 6 +- sound/soc/intel/skylake/skl.c | 296 +++++++++++------ sound/soc/intel/skylake/skl.h | 13 +- 25 files changed, 1640 insertions(+), 691 deletions(-) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c -- 2.14.1

3 22

[alsa-devel] [PATCH v4 0/7] soundwire: Add multi link support
by Shreyas NC 04 Jul '18

04 Jul '18

Currently, in the SoundWire subsystem, the concept of stream is limited to a Master and one or more Slaves(Codecs). This series aims to add support for multiple Master(s) sharing the same reference clock and synchronized in the hardware. This patch series adds: - Helpers to lock bus instances part of the stream - Boiler plate conversion of code to support a list of Master runtime - Support multi link bank switch to support synchronization between multiple masters - Add Intel platform ops for pre/post bank switch changes in v4: - Added changes in sdw_stream_runtime structure to track Masters added to stream - Add changes to support ml_bankswitch only if both the hardware supports and the stream is handled by multiple masters changes in v3: - Added comments to explain the trigger for multi link bank switch - Added comments in the intel_post_bank_switch ops to explain how synchronous trigger of bank switch is handled. Sanyog Kale (2): Documentation: soundwire: Add documentation for multi link soundwire: Add support to lock across bus instances Shreyas NC (4): soundwire: Initialize completion for defer messages soundwire: keep track of Masters in a stream soundwire: Add support for multi link bank switch soundwire: intel: Add pre/post bank switch ops Vinod Koul (1): soundwire: Handle multiple master instances in a stream Documentation/driver-api/soundwire/stream.rst | 28 ++ drivers/soundwire/bus.c | 6 + drivers/soundwire/bus.h | 4 + drivers/soundwire/intel.c | 65 ++++ drivers/soundwire/stream.c | 483 +++++++++++++++++++------- include/linux/soundwire/sdw.h | 12 +- 6 files changed, 467 insertions(+), 131 deletions(-) -- 2.7.4

5 23

[alsa-devel] [PATCH v3] ASoC: Intel: Boards: Add GLK Realtek Maxim I2S machine driver
by Naveen Manohar 03 Jul '18

03 Jul '18

Patch adds Geminilake I2S machine driver which uses following codecs: RT5682 and MAX98357A. Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v1->v2: reverse xmas tree alignment of var declaration, Add Acked-by. v2->v3: corrected: platform-name for echo-ref dai & change-log position. sound/soc/intel/boards/Kconfig | 14 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 644 ++++++++++++++++++++++++++ 3 files changed, 660 insertions(+) create mode 100644 sound/soc/intel/boards/glk_rt5682_max98357a.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 2479748..cccda87 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -281,6 +281,20 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH + tristate "GLK with RT5682 and MAX98357A in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI + select SND_SOC_RT5682 + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + select SND_HDA_DSP_LOADER + help + This adds support for ASoC machine driver for Geminilake platforms + with RT5682 + MAX98357A I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_INTEL_SKYLAKE endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 92b5507..87ef8b4 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -6,6 +6,7 @@ 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 snd-soc-sst-bxt-rt298-objs := bxt_rt298.o +snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o @@ -27,6 +28,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(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-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c new file mode 100644 index 0000000..2c36ff4 --- /dev/null +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -0,0 +1,644 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018 Intel Corporation. + +/* + * Intel Geminilake I2S Machine Driver with MAX98357A & RT5682 Codecs + * + * Modified from: + * Intel Apollolake I2S Machine driver + */ + +#include <linux/input.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 "../skylake/skl.h" +#include "../../codecs/rt5682.h" +#include "../../codecs/hdac_hdmi.h" + +/* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */ +#define GLK_PLAT_CLK_FREQ 19200000 +#define RT5682_PLL_FREQ (48000 * 512) +#define GLK_REALTEK_CODEC_DAI "rt5682-aif1" +#define GLK_MAXIM_CODEC_DAI "HiFi" +#define MAXIM_DEV0_NAME "MX98357A:00" +#define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 +#define NAME_SIZE 32 + +static struct snd_soc_jack geminilake_hdmi[3]; + +struct glk_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct glk_card_private { + struct snd_soc_jack geminilake_headset; + struct list_head hdmi_pcm_list; +}; + +enum { + GLK_DPCM_AUDIO_PB = 0, + GLK_DPCM_AUDIO_CP, + GLK_DPCM_AUDIO_HS_PB, + GLK_DPCM_AUDIO_ECHO_REF_CP, + GLK_DPCM_AUDIO_REF_CP, + GLK_DPCM_AUDIO_DMIC_CP, + GLK_DPCM_AUDIO_HDMI1_PB, + GLK_DPCM_AUDIO_HDMI2_PB, + GLK_DPCM_AUDIO_HDMI3_PB, +}; + +static int platform_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; + int ret = 0; + + codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, + RT5682_PLL1_S_MCLK, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop PLL: %d\n", ret); + } else if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(card->dev, "can't set codec pll: %d\n", ret); + return ret; + } + } + + if (ret) + dev_err(card->dev, "failed to start internal clk: %d\n", ret); + + return ret; +} + +static const struct snd_kcontrol_new geminilake_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget geminilake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Spk", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SPK("HDMI1", NULL), + SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SPK("HDMI3", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route geminilake_map[] = { + /* HP jack connectors - unknown if we have jack detection */ + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* speaker */ + { "Spk", NULL, "Speaker" }, + + /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, + { "IN1P", NULL, "Headset Mic" }, + + /* digital mics */ + { "DMic", NULL, "SoC DMIC" }, + + /* CODEC BE connections */ + { "HiFi Playback", NULL, "ssp1 Tx" }, + { "ssp1 Tx", NULL, "codec0_out" }, + + { "AIF1 Playback", NULL, "ssp2 Tx" }, + { "ssp2 Tx", NULL, "codec1_out" }, + + { "codec0_in", NULL, "ssp2 Rx" }, + { "ssp2 Rx", NULL, "AIF1 Capture" }, + + { "HDMI1", NULL, "hif5-0 Output" }, + { "HDMI2", NULL, "hif6-0 Output" }, + { "HDMI2", NULL, "hif7-0 Output" }, + + { "hifi3", NULL, "iDisp3 Tx" }, + { "iDisp3 Tx", NULL, "iDisp3_out" }, + { "hifi2", NULL, "iDisp2 Tx" }, + { "iDisp2 Tx", NULL, "iDisp2_out" }, + { "hifi1", NULL, "iDisp1 Tx" }, + { "iDisp1 Tx", NULL, "iDisp1_out" }, + + /* DMIC */ + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "DMIC AIF" }, +}; + +static int geminilake_ssp_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); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = DUAL_CHANNEL; + + /* set SSP to 24 bit */ + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct glk_card_private *ctx = 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; + struct snd_soc_jack *jack; + int ret; + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->geminilake_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->geminilake_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +}; + +static int geminilake_rt5682_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; + + /* Set valid bitmask & configuration for I2S in 24 bit */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, 24); + if (ret < 0) { + dev_err(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + return ret; +} + +static struct snd_soc_ops geminilake_rt5682_ops = { + .hw_params = geminilake_rt5682_hw_params, +}; + +static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct glk_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = GLK_DPCM_AUDIO_HDMI1_PB + dai->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_dapm_context *dapm; + int ret; + + dapm = snd_soc_component_get_dapm(component); + ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); + if (ret) { + dev_err(rtd->dev, "Ref Cap ignore suspend failed %d\n", ret); + return ret; + } + + return ret; +} + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + +static int geminilake_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); + + /* + * set BE channel constraint as user FE channels + */ + channels->min = channels->max = 4; + + return 0; +} + +static int geminilake_dmic_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels_quad); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); +} + +static const struct snd_soc_ops geminilake_dmic_ops = { + .startup = geminilake_dmic_startup, +}; + +static const unsigned int rates_16000[] = { + 16000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_16000 = { + .count = ARRAY_SIZE(rates_16000), + .list = rates_16000, +}; + +static int geminilake_refcap_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +}; + +static const struct snd_soc_ops geminilake_refcap_ops = { + .startup = geminilake_refcap_startup, +}; + +/* geminilake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link geminilake_dais[] = { + /* Front End DAI links */ + [GLK_DPCM_AUDIO_PB] = { + .name = "Glk Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .init = geminilake_rt5682_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + [GLK_DPCM_AUDIO_CP] = { + .name = "Glk Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + [GLK_DPCM_AUDIO_HS_PB] = { + .name = "Glk Audio Headset Playback", + .stream_name = "Headset Audio", + .cpu_dai_name = "System Pin2", + .platform_name = "0000:00:0e.0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dpcm_playback = 1, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_ECHO_REF_CP] = { + .name = "Glk Audio Echo Reference cap", + .stream_name = "Echoreference Capture", + .cpu_dai_name = "Echoref Pin", + .platform_name = "0000:00:0e.0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = NULL, + .capture_only = 1, + .nonatomic = 1, + }, + [GLK_DPCM_AUDIO_REF_CP] = { + .name = "Glk Audio Reference cap", + .stream_name = "Refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &geminilake_refcap_ops, + }, + [GLK_DPCM_AUDIO_DMIC_CP] = { + .name = "Glk Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &geminilake_dmic_ops, + }, + [GLK_DPCM_AUDIO_HDMI1_PB] = { + .name = "Glk HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_HDMI2_PB] = { + .name = "Glk HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_HDMI3_PB] = { + .name = "Glk HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + /* Back End DAI links */ + { + /* SSP1 - Codec */ + .name = "SSP1-Codec", + .id = 0, + .cpu_dai_name = "SSP1 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = MAXIM_DEV0_NAME, + .codec_dai_name = GLK_MAXIM_CODEC_DAI, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = geminilake_ssp_fixup, + .dpcm_playback = 1, + }, + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .id = 1, + .cpu_dai_name = "SSP2 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = "i2c-10EC5682:00", + .codec_dai_name = GLK_REALTEK_CODEC_DAI, + .init = geminilake_rt5682_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = geminilake_ssp_fixup, + .ops = &geminilake_rt5682_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .id = 2, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:0e.0", + .ignore_suspend = 1, + .be_hw_params_fixup = geminilake_dmic_fixup, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 3, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 4, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 5, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +static int glk_card_late_probe(struct snd_soc_card *card) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = NULL; + char jack_name[NAME_SIZE]; + struct glk_hdmi_pcm *pcm; + int err = 0; + int i = 0; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &geminilake_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &geminilake_hdmi[i]); + if (err < 0) + return err; + + i++; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} + +/* geminilake audio machine driver for SPT + RT5682 */ +static struct snd_soc_card glk_audio_card_rt5682_m98357a = { + .name = "glkrt5682max", + .owner = THIS_MODULE, + .dai_link = geminilake_dais, + .num_links = ARRAY_SIZE(geminilake_dais), + .controls = geminilake_controls, + .num_controls = ARRAY_SIZE(geminilake_controls), + .dapm_widgets = geminilake_widgets, + .num_dapm_widgets = ARRAY_SIZE(geminilake_widgets), + .dapm_routes = geminilake_map, + .num_dapm_routes = ARRAY_SIZE(geminilake_map), + .fully_routed = true, + .late_probe = glk_card_late_probe, +}; + +static int geminilake_audio_probe(struct platform_device *pdev) +{ + struct glk_card_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + glk_audio_card_rt5682_m98357a.dev = &pdev->dev; + snd_soc_card_set_drvdata(&glk_audio_card_rt5682_m98357a, ctx); + + return devm_snd_soc_register_card(&pdev->dev, + &glk_audio_card_rt5682_m98357a); +} + +static const struct platform_device_id glk_board_ids[] = { + { + .name = "glk_rt5682_max98357a", + .driver_data = + (kernel_ulong_t)&glk_audio_card_rt5682_m98357a, + }, + { } +}; + +static struct platform_driver geminilake_audio = { + .probe = geminilake_audio_probe, + .driver = { + .name = "glk_rt5682_max98357a", + .pm = &snd_soc_pm_ops, + }, + .id_table = glk_board_ids, +}; +module_platform_driver(geminilake_audio) + +/* Module information */ +MODULE_DESCRIPTION("Geminilake Audio Machine driver-RT5682 & MAX98357A in I2S mode"); +MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); +MODULE_AUTHOR("Harsha Priya <harshapriya.n(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:glk_rt5682_max98357a"); -- 2.7.4

2 1

[alsa-devel] [PATCH v3 0/2] ASoC: add es7241 support
by Jerome Brunet 03 Jul '18

03 Jul '18

Add support for the es7241 which is a simple 2 channels analog to digital converter. Changes since v1: [0] * Add missing break as reported by smatch [1] My device is hardwired in slave mode, which probably why this slipped through unnoticed. Changes since v2: [2] * Supply are now required dt properties * Add a dt property to state the serial interface configuration [0]: https://lkml.kernel.org/r/20180629152444.21525-1-jbrunet@baylibre.com [1]: https://lkml.kernel.org/r/20180630082829.qwnhkk4f76n32omm@mwanda [2]: https://lkml.kernel.org/r/20180630172241.10806-1-jbrunet@baylibre.com Jerome Brunet (2): ASoC: es7241: add dt-bindings documentation for the es7241 adc ASoC: es7241: add es7241 codec support .../devicetree/bindings/sound/everest,es7241.txt | 28 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es7241.c | 322 +++++++++++++++++++++ 4 files changed, 356 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/everest,es7241.txt create mode 100644 sound/soc/codecs/es7241.c -- 2.14.4

2 8

[alsa-devel] [PATCH 1/2] jack: Allow build with older alsa-lib
by Takashi Iwai 03 Jul '18

03 Jul '18

The new API snd_pcm_ioplug_avail() doesn't appear prior to alsa-lib 1.1.7. Add an own code for allowing the build with alsa-lib 1.1.6. It has a very slight overhead than the original snd_pcm_ioplug_avail() but should be OK as a workaround. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- jack/pcm_jack.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 98a6f7ebb82e..fbc9dafc594b 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -58,6 +58,15 @@ typedef struct { static int snd_pcm_jack_stop(snd_pcm_ioplug_t *io); +/* snd_pcm_ioplug_avail() was introduced after alsa-lib 1.1.6 */ +#if SND_LIB_VERSION < 0x10107 +static snd_pcm_uframes_t snd_pcm_ioplug_avail(const snd_pcm_ioplug_t *io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + return io->buffer_size - snd_pcm_ioplug_hw_avail(io, hw_ptr, appl_ptr); +} +#endif static int pcm_poll_block_check(snd_pcm_ioplug_t *io) { -- 2.18.0

1 2

[alsa-devel] [PATCH] ASoC: rt5682: add button detection mode control
by Bard Liao 03 Jul '18

03 Jul '18

We are currently using power saving mode for button detection. However, it will impact the headset recording performance. This patch will switch button detection to normal mode in capture and switch to power saving mode in the end of capture. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5682.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index baad177..640d400 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -857,6 +857,8 @@ static int rt5682_button_detect(struct snd_soc_component *component) btn_type = val & 0xfff0; snd_soc_component_write(component, RT5682_4BTN_IL_CMD_1, val); pr_debug("%s btn_type=%x\n", __func__, btn_type); + snd_soc_component_update_bits(component, + RT5682_SAR_IL_CMD_2, 0x10, 0x10); return btn_type; } @@ -1645,6 +1647,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682_STO1_ADC_DIG_VOL, RT5682_R_MUTE_SFT, 1, rt5682_sto1_adc_r_mix, ARRAY_SIZE(rt5682_sto1_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("BTN Detection Mode", RT5682_SAR_IL_CMD_1, + 14, 1, NULL, 0), /* ADC PGA */ SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -1807,6 +1811,8 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"}, {"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"}, + {"ADC Stereo1 Filter", NULL, "BTN Detection Mode"}, + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"}, {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"}, -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: pxa-ssp: mark expected switch fall-through" to the asoc tree
by Mark Brown 03 Jul '18

03 Jul '18

The patch ASoC: pxa-ssp: mark expected switch fall-through has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e0431de301cbb8e3915261dfff4d0b072738de69 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" <gustavo(a)embeddedor.com> Date: Mon, 2 Jul 2018 07:17:07 -0500 Subject: [PATCH] ASoC: pxa-ssp: mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/pxa/pxa-ssp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index f8339bb01251..ff1e0bd8d407 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -470,6 +470,7 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; + /* fall through */ case SND_SOC_DAIFMT_DSP_B: sscr0 |= SSCR0_MOD | SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; -- 2.18.0.rc2

1 0

[alsa-devel] [PATCH 1/6] ASoC: AMD: Always stop ch2 first
by Daniel Kurtz 03 Jul '18

03 Jul '18

Commit 6b116dfb4633a ("ASoC: AMD: make channel 1 dma as circular") made both channels circular, so this comment and logic no longer applies. Always stop ch2 (the channel closest to the output) before ch1. This ensures that the downstream circular DMA channel does not continue to play/capture repeated samples after the upstream circular DMA channel has already stopped. Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> --- sound/soc/amd/acp-pcm-dma.c | 17 ++--------------- 1 file changed, 2 insertions(+), 15 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 3c3d398d0d0b1..4665ae12e74e4 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1067,21 +1067,8 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - /* For playback, non circular dma should be stopped first - * i.e Sysram to acp dma transfer channel(rtd->ch1) should be - * stopped before stopping cirular dma which is acp sram to i2s - * fifo dma transfer channel(rtd->ch2). Where as in Capture - * scenario, i2s fifo to acp sram dma channel(rtd->ch2) stopped - * first before stopping acp sram to sysram which is circular - * dma(rtd->ch1). - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - acp_dma_stop(rtd->acp_mmio, rtd->ch1); - ret = acp_dma_stop(rtd->acp_mmio, rtd->ch2); - } else { - acp_dma_stop(rtd->acp_mmio, rtd->ch2); - ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); - } + acp_dma_stop(rtd->acp_mmio, rtd->ch2); + ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); rtd->bytescount = 0; break; default: -- 2.18.0.399.gad0ab374a1-goog

2 11

[alsa-devel] [PATCH v2 0/3] ASoC: es7134: driver updates
by Jerome Brunet 03 Jul '18

03 Jul '18

This patchset fixes issues of v1 [0] around power supply requirement and add es7154 support. AVDD is renamed to PVDD to better match the component datasheet. v1 assumed optional power supplies, which was a wrong thing do. Now VDD, is required on all chip variant and PVDD is required on the es7154. [0]: https://lkml.kernel.org/r/20180629150924.18197-1-jbrunet@baylibre.com Jerome Brunet (3): ASoC: es7134: update DT binding with new compatible and supplies ASoC: es7134: correct required power supplies ASoC: es7134: add support for the es7154 .../devicetree/bindings/sound/everest,es7134.txt | 7 +- sound/soc/codecs/es7134.c | 104 ++++++++++++++++++++- 2 files changed, 106 insertions(+), 5 deletions(-) -- 2.14.4

2 6

[alsa-devel] [PATCH 0/2] ASoC: atmel-i2s: Remove unnecessary audio PLL clock
by Codrin Ciubotariu 03 Jul '18

03 Jul '18

Thanks to Quentin Schulz's patch: commit 1a1a36d72e3d ("clk: at91: clk-generated: make gclk determine audio_pll rate") the generated clock (gclk) itself is able to configure the audio PLL clock (aclk). For this reason, the I2S driver no longer needs to set aclk, but only gclk. This patchet removes all references to the audio PLL clock from the driver and updates the driver's bindings. Codrin Ciubotariu (2): ASoC: atmel-i2s: Remove unnecessary audio PLL clock (aclk) ASoC: atmel-i2s: dt-bindings: Remove unnecessary phandle to aclk .../devicetree/bindings/sound/atmel-i2s.txt | 11 ++--- sound/soc/atmel/atmel-i2s.c | 46 +++---------------- 2 files changed, 12 insertions(+), 45 deletions(-) -- 2.17.1

2 4

[alsa-devel] [PATCH - JACK IO plug 1/2 v2] jack: Avoid call to snd_pcm_avail_update() from JACK thread
by twischer＠de.adit-jv.com 03 Jul '18

03 Jul '18

From: Timo Wischer <twischer(a)de.adit-jv.com> snd_pcm_avail_update() can call snd_pcm_jack_stop() but snd_pcm_jack_stop() should not be called by the JACK thread. It should only be called by the thread how has called snd_pcm_jack_start(). In addition the execution of snd_pcm_avail_update() can take a while. Therefore it should not be called by the JACK thread to not block this thread. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- v2 uses snd_pcm_uframes_t as type for avail and does not check for avail < 0 becasue snd_pcm_ioplug_avail() will never retrun a value < 0 diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index e3df4d2..5a57e13 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -62,13 +62,13 @@ static int snd_pcm_jack_stop(snd_pcm_ioplug_t *io); static int pcm_poll_block_check(snd_pcm_ioplug_t *io) { static char buf[32]; - snd_pcm_sframes_t avail; + snd_pcm_uframes_t avail; snd_pcm_jack_t *jack = io->private_data; if (io->state == SND_PCM_STATE_RUNNING || (io->state == SND_PCM_STATE_PREPARED && io->stream == SND_PCM_STREAM_CAPTURE)) { - avail = snd_pcm_avail_update(io->pcm); - if (avail >= 0 && avail < jack->min_avail) { + avail = snd_pcm_ioplug_avail(io, jack->hw_ptr, io->appl_ptr); + if (avail < jack->min_avail) { while (read(io->poll_fd, &buf, sizeof(buf)) == sizeof(buf)) ; return 1; @@ -81,11 +81,11 @@ static int pcm_poll_block_check(snd_pcm_ioplug_t *io) static int pcm_poll_unblock_check(snd_pcm_ioplug_t *io) { static char buf[1]; - snd_pcm_sframes_t avail; + snd_pcm_uframes_t avail; snd_pcm_jack_t *jack = io->private_data; - avail = snd_pcm_avail_update(io->pcm); - if (avail < 0 || avail >= jack->min_avail) { + avail = snd_pcm_ioplug_avail(io, jack->hw_ptr, io->appl_ptr); + if (avail >= jack->min_avail) { write(jack->fd, &buf, 1); return 1; } -- 2.7.4

1 0

[alsa-devel] [PATCH - JACK IO plug 0/3] snd_pcm_drain support
by twischer＠de.adit-jv.com 03 Jul '18

03 Jul '18

The following patches are required to support draining with the JACK IO plugin. Without these patches $ speaker-test -D jack -c2 will freeze. It is required to support the draining feature implemented in commit ce2095c41f2 (pcm: ioplug: Implement proper drain behavior)

2 8

[alsa-devel] [PATCH] snd_interval_refine_first/last: exclude value only if also excluded before
by twischer＠de.adit-jv.com 03 Jul '18

03 Jul '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this commit the following intervals [x y), (x y) were be replaced to (y-1 y) by snd_interval_refine_last(). This was also done if y-1 is part of the previous interval. With this changes it will be replaced with [y-1 y) in case of y-1 is part of the previous interval. A similar behavior will be used for snd_interval_refine_first(). This solves the issue reported here: https://bugzilla.opensuse.org/show_bug.cgi?id=1033179 and work arounded with commit e736715 ("pcm: dmix: Disable var_periodsize as default"). I am able to reproduce the issue with a simplified aplay use case using the following configuration: pcm_slave.adr3_tdm_8ch { pcm { type hw card "Loopback" device 0 } rate 48000 period_size 128 buffer_size 1024 channels 2 } pcm.dshare_Playback_3 { type dmix ipc_key 600 ipc_perm 0660 ipc_gid audio var_periodsize true slave adr3_tdm_8ch } pcm.AdevAcousticoutSpeech { type rate slave.pcm dshare_Playback_3 slave.rate 48000 } $ modprobe snd_aloop $ aplay -v --period-size=352 -c2 -fS16_LE -r22500 -DAdevAcousticoutSpeech /dev/urandom ... Rule 9 (0xffff91d1f230): PERIODS=(0 2) -> NONE BUFFER_SIZE=480 PERIOD_SIZE=[240 240] refine_soft 'AdevAcousticoutSpeech' (end--22) ... aplay: ../../alsa-utils-1.1.5/aplay/aplay.c:1390: set_params: Assertion `err >= 0' failed. Aborted by signal Aborted... The following stack trace shows where the -EINVAL will be thrown: __snd_pcm_hw_params_set_period_size_near() snd1_pcm_hw_param_set_near() snd1_pcm_hw_param_set_last() snd1_pcm_hw_refine_slave() snd1_pcm_hw_refine_soft() snd_pcm_hw_rule_div() snd1_interval_refine() This issue exists due to PERIODS does not include 2 Rule 9 (0xffff91d1f230): PERIODS=(0 9) -> (0 2) BUFFER_SIZE=[120 480] PERIOD_SIZE=(240 241) because of an invalid integer inverval of PERIOD_SIZE of (240 241). This interval is set by snd_interval_refine_last(). Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- src/pcm/interval.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/src/pcm/interval.c b/src/pcm/interval.c index 95cedd5..ef4c2ed 100644 --- a/src/pcm/interval.c +++ b/src/pcm/interval.c @@ -200,27 +200,33 @@ int snd_interval_refine(snd_interval_t *i, const snd_interval_t *v) int snd_interval_refine_first(snd_interval_t *i) { + const unsigned int last_max = i->max; + if (snd_interval_empty(i)) return -ENOENT; if (snd_interval_single(i)) return 0; i->max = i->min; - i->openmax = i->openmin; - if (i->openmax) + if (i->openmin) i->max++; + /* only exclude max value if also excluded before refine */ + i->openmax = (i->openmax && i->max >= last_max); return 1; } int snd_interval_refine_last(snd_interval_t *i) { + const unsigned int last_min = i->min; + if (snd_interval_empty(i)) return -ENOENT; if (snd_interval_single(i)) return 0; i->min = i->max; - i->openmin = i->openmax; - if (i->openmin) + if (i->openmax) i->min--; + /* only exclude min value if also excluded before refine */ + i->openmin = (i->openmin && i->min <= last_min); return 1; } -- 2.7.4

2 1

[alsa-devel] [PATCH v3 24/27] devicetree: fix a series of wrong file references
by Mauro Carvalho Chehab 03 Jul '18

03 Jul '18

As files got renamed, their references broke. Manually fix a series of broken refs at the DT bindings. Signed-off-by: Mauro Carvalho Chehab <mchehab+samsung(a)kernel.org> --- .../devicetree/bindings/input/rmi4/rmi_2d_sensor.txt | 2 +- Documentation/devicetree/bindings/mfd/sun6i-prcm.txt | 2 +- .../devicetree/bindings/pci/hisilicon-pcie.txt | 2 +- Documentation/devicetree/bindings/pci/kirin-pcie.txt | 2 +- .../devicetree/bindings/pci/pci-keystone.txt | 4 ++-- .../devicetree/bindings/sound/st,stm32-i2s.txt | 2 +- .../devicetree/bindings/sound/st,stm32-sai.txt | 2 +- MAINTAINERS | 12 ++++++------ 8 files changed, 14 insertions(+), 14 deletions(-) diff --git a/Documentation/devicetree/bindings/input/rmi4/rmi_2d_sensor.txt b/Documentation/devicetree/bindings/input/rmi4/rmi_2d_sensor.txt index f2c30c8b725d..9afffbdf6e28 100644 --- a/Documentation/devicetree/bindings/input/rmi4/rmi_2d_sensor.txt +++ b/Documentation/devicetree/bindings/input/rmi4/rmi_2d_sensor.txt @@ -12,7 +12,7 @@ Additional documentation for F11 can be found at: http://www.synaptics.com/sites/default/files/511-000136-01-Rev-E-RMI4-Inter… Optional Touch Properties: -Description in Documentation/devicetree/bindings/input/touch +Description in Documentation/devicetree/bindings/input/touchscreen - touchscreen-inverted-x - touchscreen-inverted-y - touchscreen-swapped-x-y diff --git a/Documentation/devicetree/bindings/mfd/sun6i-prcm.txt b/Documentation/devicetree/bindings/mfd/sun6i-prcm.txt index 4d21ffdb0fc1..daa091c2e67b 100644 --- a/Documentation/devicetree/bindings/mfd/sun6i-prcm.txt +++ b/Documentation/devicetree/bindings/mfd/sun6i-prcm.txt @@ -8,7 +8,7 @@ Required properties: - reg: The PRCM registers range The prcm node may contain several subdevices definitions: - - see Documentation/devicetree/clk/sunxi.txt for clock devices + - see Documentation/devicetree/bindings/clock/sunxi.txt for clock devices - see Documentation/devicetree/bindings/reset/allwinner,sunxi-clock-reset.txt for reset controller devices diff --git a/Documentation/devicetree/bindings/pci/hisilicon-pcie.txt b/Documentation/devicetree/bindings/pci/hisilicon-pcie.txt index 7bf9df047a1e..0dcb87d6554f 100644 --- a/Documentation/devicetree/bindings/pci/hisilicon-pcie.txt +++ b/Documentation/devicetree/bindings/pci/hisilicon-pcie.txt @@ -3,7 +3,7 @@ HiSilicon Hip05 and Hip06 PCIe host bridge DT description HiSilicon PCIe host controller is based on the Synopsys DesignWare PCI core. It shares common functions with the PCIe DesignWare core driver and inherits common properties defined in -Documentation/devicetree/bindings/pci/designware-pci.txt. +Documentation/devicetree/bindings/pci/designware-pcie.txt. Additional properties are described here: diff --git a/Documentation/devicetree/bindings/pci/kirin-pcie.txt b/Documentation/devicetree/bindings/pci/kirin-pcie.txt index 6e217c63123d..6bbe43818ad5 100644 --- a/Documentation/devicetree/bindings/pci/kirin-pcie.txt +++ b/Documentation/devicetree/bindings/pci/kirin-pcie.txt @@ -3,7 +3,7 @@ HiSilicon Kirin SoCs PCIe host DT description Kirin PCIe host controller is based on the Synopsys DesignWare PCI core. It shares common functions with the PCIe DesignWare core driver and inherits common properties defined in -Documentation/devicetree/bindings/pci/designware-pci.txt. +Documentation/devicetree/bindings/pci/designware-pcie.txt. Additional properties are described here: diff --git a/Documentation/devicetree/bindings/pci/pci-keystone.txt b/Documentation/devicetree/bindings/pci/pci-keystone.txt index 7e05487544ed..3d4a209b0fd0 100644 --- a/Documentation/devicetree/bindings/pci/pci-keystone.txt +++ b/Documentation/devicetree/bindings/pci/pci-keystone.txt @@ -3,9 +3,9 @@ TI Keystone PCIe interface Keystone PCI host Controller is based on the Synopsys DesignWare PCI hardware version 3.65. It shares common functions with the PCIe DesignWare core driver and inherits common properties defined in -Documentation/devicetree/bindings/pci/designware-pci.txt +Documentation/devicetree/bindings/pci/designware-pcie.txt -Please refer to Documentation/devicetree/bindings/pci/designware-pci.txt +Please refer to Documentation/devicetree/bindings/pci/designware-pcie.txt for the details of DesignWare DT bindings. Additional properties are described here as well as properties that are not applicable. diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt index 4bda52042402..58c341300552 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt @@ -18,7 +18,7 @@ Required properties: See Documentation/devicetree/bindings/dma/stm32-dma.txt. - dma-names: Identifier for each DMA request line. Must be "tx" and "rx". - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/pinctrl-stm32.txt + - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt Optional properties: - resets: Reference to a reset controller asserting the reset controller diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt index f301cdf0b7e6..3a3fc506e43a 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt @@ -37,7 +37,7 @@ SAI subnodes required properties: "tx": if sai sub-block is configured as playback DAI "rx": if sai sub-block is configured as capture DAI - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/pinctrl-stm32.txt + - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt SAI subnodes Optional properties: - st,sync: specify synchronization mode. diff --git a/MAINTAINERS b/MAINTAINERS index 67641f5bb373..69c9e9924902 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6965,7 +6965,7 @@ IIO MULTIPLEXER M: Peter Rosin <peda(a)axentia.se> L: linux-iio(a)vger.kernel.org S: Maintained -F: Documentation/devicetree/bindings/iio/multiplexer/iio-mux.txt +F: Documentation/devicetree/bindings/iio/multiplexer/io-channel-mux.txt F: drivers/iio/multiplexer/iio-mux.c IIO SUBSYSTEM AND DRIVERS @@ -9695,7 +9695,7 @@ MXSFB DRM DRIVER M: Marek Vasut <marex(a)denx.de> S: Supported F: drivers/gpu/drm/mxsfb/ -F: Documentation/devicetree/bindings/display/mxsfb-drm.txt +F: Documentation/devicetree/bindings/display/mxsfb.txt MYRICOM MYRI-10G 10GbE DRIVER (MYRI10GE) M: Chris Lee <christopher.lee(a)cspi.com> @@ -10884,7 +10884,7 @@ M: Will Deacon <will.deacon(a)arm.com> L: linux-pci(a)vger.kernel.org L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Maintained -F: Documentation/devicetree/bindings/pci/controller-generic-pci.txt +F: Documentation/devicetree/bindings/pci/host-generic-pci.txt F: drivers/pci/controller/pci-host-common.c F: drivers/pci/controller/pci-host-generic.c @@ -11065,7 +11065,7 @@ M: Xiaowei Song <songxiaowei(a)hisilicon.com> M: Binghui Wang <wangbinghui(a)hisilicon.com> L: linux-pci(a)vger.kernel.org S: Maintained -F: Documentation/devicetree/bindings/pci/pcie-kirin.txt +F: Documentation/devicetree/bindings/pci/kirin-pcie.txt F: drivers/pci/controller/dwc/pcie-kirin.c PCIE DRIVER FOR HISILICON STB @@ -12456,7 +12456,7 @@ L: linux-crypto(a)vger.kernel.org L: linux-samsung-soc(a)vger.kernel.org S: Maintained F: drivers/crypto/exynos-rng.c -F: Documentation/devicetree/bindings/crypto/samsung,exynos-rng4.txt +F: Documentation/devicetree/bindings/rng/samsung,exynos4-rng.txt SAMSUNG EXYNOS TRUE RANDOM NUMBER GENERATOR (TRNG) DRIVER M: Łukasz Stelmach <l.stelmach(a)samsung.com> @@ -13570,7 +13570,7 @@ F: drivers/*/stm32-*timer* F: drivers/pwm/pwm-stm32* F: include/linux/*/stm32-*tim* F: Documentation/ABI/testing/*timer-stm32 -F: Documentation/devicetree/bindings/*/stm32-*timer +F: Documentation/devicetree/bindings/*/stm32-*timer* F: Documentation/devicetree/bindings/pwm/pwm-stm32* STMMAC ETHERNET DRIVER -- 2.17.1

4 3

[alsa-devel] [PATCH 0/5] ASoC: simple/audio card: convert to SPDX identifiers
by Kuninori Morimoto 03 Jul '18

03 Jul '18

Hi Mark These convert to SPDX identifiers for simple/audio cards Kuninori Morimoto (5): ASoC: simple-card: convert to SPDX identifiers ASoC: simple-card-utils: convert to SPDX identifiers ASoC: simple-scu-card.c: convert to SPDX identifiers ASoC: audio-graph-card.c: convert to SPDX identifiers ASoC: audio-graph-scu-card.c: convert to SPDX identifiers include/sound/simple_card.h | 7 ++----- include/sound/simple_card_utils.h | 8 +++----- sound/soc/generic/audio-graph-card.c | 21 +++++++++------------ sound/soc/generic/audio-graph-scu-card.c | 25 +++++++++++-------------- sound/soc/generic/simple-card-utils.c | 15 ++++++--------- sound/soc/generic/simple-card.c | 17 +++++++---------- sound/soc/generic/simple-scu-card.c | 21 +++++++++------------ 7 files changed, 47 insertions(+), 67 deletions(-) -- 2.7.4

3 15

Re: [alsa-devel] linux-next: Tree for Jul 2 (sound/soc/pxa/)
by Randy Dunlap 02 Jul '18

02 Jul '18

[add other maintainers] On 07/02/18 09:09, Randy Dunlap wrote: > On 07/01/18 21:51, Stephen Rothwell wrote: >> Hi all, >> >> Changes since 20180629: >> > > on i386 or x86_64: > > builtin: > sound/soc/pxa/pxa-ssp.o:(.rodata+0x228): undefined reference to `pxa2xx_soc_pcm_new' > sound/soc/pxa/pxa-ssp.o:(.rodata+0x22c): undefined reference to `pxa2xx_pcm_free_dma_buffers' > sound/soc/pxa/pxa-ssp.o:(.rodata+0x250): undefined reference to `pxa2xx_pcm_ops' > /home/rdunlap/lnx/next/linux-next-20180702/Makefile:1023: recipe for target 'vmlinux' failed > > modular: > ERROR: "pxa2xx_pcm_ops" [sound/soc/pxa/snd-soc-pxa-ssp.ko] undefined! > ERROR: "pxa2xx_pcm_free_dma_buffers" [sound/soc/pxa/snd-soc-pxa-ssp.ko] undefined! > ERROR: "pxa2xx_soc_pcm_new" [sound/soc/pxa/snd-soc-pxa-ssp.ko] undefined! > > > x86_64 [modular] randconfig file is attached. > -- ~Randy

2 1

[alsa-devel] [PATCH v2] ALSA: core: Allow drivers to set R/W wait time.
by Liam Girdwood 02 Jul '18

02 Jul '18

Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO. This needs to be configurable for modern hardware like DSPs where no pointer update in milliseconds can indicate terminal DSP errors. Add a substream variable to set the wait time in ms. This allows userspace and drivers to recover more quickly from terminal DSP errors. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Changes since V1 :- o Remove API method and alow drivers to set directly. o Validate time is driver supplied times. include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 10 ++++++++-- 2 files changed, 9 insertions(+), 2 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e054c583d3b3..edad4a506b93 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -462,6 +462,7 @@ struct snd_pcm_substream { /* -- timer section -- */ struct snd_timer *timer; /* timer */ unsigned timer_running: 1; /* time is running */ + unsigned wait_time; /* time in ms for R/W to wait for avail */ /* -- next substream -- */ struct snd_pcm_substream *next; /* -- linked substreams -- */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 44b5ae833082..6909b896a6a1 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1832,12 +1832,18 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { - wait_time = 10; + /* use wait time from substream if available */ + if (substream->wait_time) { + wait_time = substream->wait_time; + } else { + wait_time = 10 * 1000; /* 10 secs */ + } + if (runtime->rate) { long t = runtime->period_size * 2 / runtime->rate; wait_time = max(t, wait_time); } - wait_time = msecs_to_jiffies(wait_time * 1000); + wait_time = msecs_to_jiffies(wait_time); } for (;;) { -- 2.17.1

2 2

[alsa-devel] [sound:for-linus 15/15] sound/pci/hda/patch_hdmi.c:1638:18: error: implicit declaration of function 'pm_runtime_suspended'; did you mean 'pm_generic_suspend'?
by kbuild test robot 02 Jul '18

02 Jul '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: b0526c9f124fa2eada1c262743bf09edc3799a20 commit: b0526c9f124fa2eada1c262743bf09edc3799a20 [15/15] ALSA: hda - Handle pm failure during hotplug config: x86_64-randconfig-x010-201826 (attached as .config) compiler: gcc-7 (Debian 7.3.0-16) 7.3.0 reproduce: git checkout b0526c9f124fa2eada1c262743bf09edc3799a20 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): In file included from include/linux/init.h:5:0, from sound/pci/hda/patch_hdmi.c:32: sound/pci/hda/patch_hdmi.c: In function 'hdmi_present_sense': >> sound/pci/hda/patch_hdmi.c:1638:18: error: implicit declaration of function 'pm_runtime_suspended'; did you mean 'pm_generic_suspend'? [-Werror=implicit-function-declaration] if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) { ^ include/linux/compiler.h:58:30: note: in definition of macro '__trace_if' if (__builtin_constant_p(!!(cond)) ? !!(cond) : \ ^~~~ sound/pci/hda/patch_hdmi.c:1638:3: note: in expansion of macro 'if' if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) { ^~ cc1: some warnings being treated as errors vim +1638 sound/pci/hda/patch_hdmi.c 1629 1630 static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) 1631 { 1632 struct hda_codec *codec = per_pin->codec; 1633 int ret; 1634 1635 /* no temporary power up/down needed for component notifier */ 1636 if (!codec_has_acomp(codec)) { 1637 ret = snd_hda_power_up_pm(codec); > 1638 if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) { 1639 snd_hda_power_down_pm(codec); 1640 return false; 1641 } 1642 } 1643 1644 if (codec_has_acomp(codec)) { 1645 sync_eld_via_acomp(codec, per_pin); 1646 ret = false; /* don't call snd_hda_jack_report_sync() */ 1647 } else { 1648 ret = hdmi_present_sense_via_verbs(per_pin, repoll); 1649 } 1650 1651 if (!codec_has_acomp(codec)) 1652 snd_hda_power_down_pm(codec); 1653 1654 return ret; 1655 } 1656 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] [PATCH REPOST 0/2] ALSA: use irqsave in USB's completion callback
by Sebastian Andrzej Siewior 02 Jul '18

02 Jul '18

This series is about using _irqsave() primitives in the completion callback in order to get rid of local_irq_save() in __usb_hcd_giveback_urb(). Sebastian

4 5

[alsa-devel] alsa-lib-devel sources for centos 7 and backward compatibility
by maxim maxim 02 Jul '18

02 Jul '18

Hello, I have some issues with alsa-lib-devel for centos 7 , I'm working with gcc7.2.1 , I achieved the alsa-lb and compiled with gcc-7.2.1, my question where I can find alsa-lib-devel sources to compile it with my compiler? Additional question that I have it's about backward compatibility I saw that alsa-lib-devel-1.1.4.1-2.el7.x86_64.rpm doesn't have file alsa/iatomic.h as it was for Centos 5, could someone explain me please were I can achieve the functions that were in alsa/iatomic.h Thanks in advance, Max

1 0

[alsa-devel] [PATCH 0/5] ASoC: es7134: driver updates
by Jerome Brunet 02 Jul '18

02 Jul '18

This patchset provides several updates for the es7134 driver It removes unsupported sample rates, add runtime check of the mclk-fs ratio as documented in the datasheet and adds the component power supplies. Finally support for the es7154 is added. Jerome Brunet (5): ASoC: es7134: remove 64kHz rate from the supported rates ASoC: es7134: check if mclk rate is valid ASoC: es7134: update DT binding with new compatible and supplies ASoC: es7134: Add VDD and AVDD power supplies ASoC: es7134: add support for the es7154 .../devicetree/bindings/sound/everest,es7134.txt | 8 +- sound/soc/codecs/es7134.c | 176 ++++++++++++++++++++- 2 files changed, 179 insertions(+), 5 deletions(-) -- 2.14.4

2 13

[alsa-devel] [PATCH v2 0/2] ASoC: add es7241 support
by Jerome Brunet 02 Jul '18

02 Jul '18

Add support for the es7241 which is a simple 2 channels analog to digital converter. Changes since v1: [0] * Add missing break as reported by smatch [1] My device is hardwired in slave mode, which probably why this slipped through unnoticed. [0]: https://lkml.kernel.org/r/20180629152444.21525-1-jbrunet@baylibre.com [1]: https://lkml.kernel.org/r/20180630082829.qwnhkk4f76n32omm@mwanda Jerome Brunet (2): ASoC: es7241: add dt-bindings documentation for the es7241 adc ASoC: es7241: add es7241 codec support .../devicetree/bindings/sound/everest,es7241.txt | 23 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es7241.c | 308 +++++++++++++++++++++ 4 files changed, 337 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/everest,es7241.txt create mode 100644 sound/soc/codecs/es7241.c -- 2.14.4

2 6

[alsa-devel] [PATCH 00/14] ASoC: soc-xxx: convert to SPDX identifiers
by Kuninori Morimoto 02 Jul '18

02 Jul '18

Hi Mark These convert to SPDX identifiers for soc-xxx files Kuninori Morimoto (14): ASoC: soc-io.c: convert to SPDX identifiers ASoC: soc-dai.h: convert to SPDX identifiers ASoC: soc-ops.c: convert to SPDX identifiers ASoC: soc-pcm.c: convert to SPDX identifiers ASoC: soc-dpcm.h: convert to SPDX identifiers ASoC: soc-jack.c: convert to SPDX identifiers ASoC: soc-utils.c: convert to SPDX identifiers ASoC: soc-devres.c: convert to SPDX identifiers ASoC: soc-acpi: convert to SPDX identifiers ASoC: soc-core: convert to SPDX identifiers ASoC: soc-dapm: convert to SPDX identifiers ASoC: soc-topology: convert to SPDX identifiers ASoC: soc-compress: convert to SPDX identifiers ASoC: soc-generic-dmaengine-pcm: convert to SPDX identifiers include/sound/compress_driver.h | 21 +++------------- include/sound/dmaengine_pcm.h | 14 +++-------- include/sound/soc-acpi-intel-match.h | 14 ++--------- include/sound/soc-acpi.h | 13 ++-------- include/sound/soc-dai.h | 7 ++---- include/sound/soc-dapm.h | 11 +++----- include/sound/soc-dpcm.h | 7 ++---- include/sound/soc-topology.h | 7 ++---- include/sound/soc.h | 11 +++----- sound/soc/soc-acpi.c | 20 ++++----------- sound/soc/soc-compress.c | 24 +++++++----------- sound/soc/soc-core.c | 41 ++++++++++++++---------------- sound/soc/soc-dapm.c | 42 ++++++++++++++----------------- sound/soc/soc-devres.c | 15 ++++------- sound/soc/soc-generic-dmaengine-pcm.c | 19 ++++---------- sound/soc/soc-io.c | 19 ++++++-------- sound/soc/soc-jack.c | 19 ++++++-------- sound/soc/soc-ops.c | 29 +++++++++------------ sound/soc/soc-pcm.c | 28 ++++++++------------- sound/soc/soc-topology.c | 47 ++++++++++++++++------------------- sound/soc/soc-utils.c | 22 ++++++---------- 21 files changed, 150 insertions(+), 280 deletions(-) -- 2.7.4

2 28

[alsa-devel] [PATCH] ASoC: ac97: convert to SPDX identifiers
by Kuninori Morimoto 02 Jul '18

02 Jul '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/ac97/codec.h | 8 +++----- include/sound/ac97/compat.h | 9 +++------ include/sound/ac97/controller.h | 8 +++----- include/sound/ac97/regs.h | 20 ++------------------ include/sound/ac97_codec.h | 25 +++++-------------------- sound/soc/soc-ac97.c | 29 ++++++++++++----------------- 6 files changed, 28 insertions(+), 71 deletions(-) diff --git a/include/sound/ac97/codec.h b/include/sound/ac97/codec.h index ec04be9..9792d25 100644 --- a/include/sound/ac97/codec.h +++ b/include/sound/ac97/codec.h @@ -1,10 +1,8 @@ -/* - * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik(a)free.fr> +/* SPDX-License-Identifier: GPL-2.0 * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. + * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik(a)free.fr> */ + #ifndef __SOUND_AC97_CODEC2_H #define __SOUND_AC97_CODEC2_H diff --git a/include/sound/ac97/compat.h b/include/sound/ac97/compat.h index 1351cba..57e19af 100644 --- a/include/sound/ac97/compat.h +++ b/include/sound/ac97/compat.h @@ -1,14 +1,11 @@ -/* - * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik(a)free.fr> +/* SPDX-License-Identifier: GPL-2.0 * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. + * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik(a)free.fr> * * This file is for backward compatibility with snd_ac97 structure and its * multiple usages, such as the snd_ac97_bus and snd_ac97_build_ops. - * */ + #ifndef AC97_COMPAT_H #define AC97_COMPAT_H diff --git a/include/sound/ac97/controller.h b/include/sound/ac97/controller.h index b36ecdd..06b5afb 100644 --- a/include/sound/ac97/controller.h +++ b/include/sound/ac97/controller.h @@ -1,10 +1,8 @@ -/* - * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik(a)free.fr> +/* SPDX-License-Identifier: GPL-2.0 * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. + * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik(a)free.fr> */ + #ifndef AC97_CONTROLLER_H #define AC97_CONTROLLER_H diff --git a/include/sound/ac97/regs.h b/include/sound/ac97/regs.h index 9a4fa0c..843f73f 100644 --- a/include/sound/ac97/regs.h +++ b/include/sound/ac97/regs.h @@ -1,27 +1,11 @@ -/* +/* SPDX-License-Identifier: GPL-2.0+ + * * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.1 * by Intel Corporation (http://developer.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; either version 2 of the License, or - * (at your option) any later version. - * - * 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. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * */ - /* * AC'97 codec registers */ diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 89d311a..cc38399 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -1,30 +1,15 @@ -#ifndef __SOUND_AC97_CODEC_H -#define __SOUND_AC97_CODEC_H - -/* +/* SPDX-License-Identifier: GPL-2.0+ + * * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.1 * by Intel Corporation (http://developer.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; either version 2 of the License, or - * (at your option) any later version. - * - * 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. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * */ +#ifndef __SOUND_AC97_CODEC_H +#define __SOUND_AC97_CODEC_H + #include <linux/bitops.h> #include <linux/device.h> #include <linux/workqueue.h> diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 3f424f2..c086786 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -1,20 +1,15 @@ -/* - * soc-ac97.c -- ALSA SoC Audio Layer AC97 support - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> - * with code, comments and ideas from :- - * Richard Purdie <richard(a)openedhand.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; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-ac97.c -- ALSA SoC Audio Layer AC97 support +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Author: Liam Girdwood <lrg(a)slimlogic.co.uk> +// with code, comments and ideas from :- +// Richard Purdie <richard(a)openedhand.com> #include <linux/ctype.h> #include <linux/delay.h> -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: pxa: select SND_PXA2XX_LIB for drivers that depend on it
by Daniel Mack 02 Jul '18

02 Jul '18

Commit d767d3ce5c48b ("ASoC: pxa: provide PCM ops for ssp, i2s and ac97 components") created a build-time dependency to SND_PXA2XX_LIB but missed to reflect that in Kconfig. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 95dcf97a6271..2fc02c227f69 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -24,15 +24,18 @@ config SND_PXA2XX_AC97 config SND_PXA2XX_SOC_AC97 tristate select AC97_BUS + select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 select SND_SOC_AC97_BUS config SND_PXA2XX_SOC_I2S + select SND_PXA2XX_LIB tristate config SND_PXA_SOC_SSP tristate "Soc Audio via PXA2xx/PXA3xx SSP ports" select PXA_SSP + select SND_PXA2XX_LIB config SND_MMP_SOC_SSPA tristate -- 2.17.1

2 1

[alsa-devel] [PATCH RFC] ASoC: pxa-ssp: remove .set_pll() and .set_clkdiv() callbacks
by Daniel Mack 02 Jul '18

02 Jul '18

The .set_pll() and .set_clkdiv() callbacks are considered legacy and should not be used anymore. In order to support PXA boards on DT platforms, remove them and let the code figure out the correct dividers and PLL base frequencies itself. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- I'd appreciate some review of this one. The only PXA3xx based platform I have access to uses the EXTCLK clocking source, but this patch changes the way the internal audio PLL works. Hence, I could only complile-test it for now. The PLL lookup table is taken from the datasheet, so it should be fine, but I'm not sure whether I got the mclk/bclk/fs calculations right, so a 2nd pair of eyes would help to avoid regressions. Given that the PXA platform has very little active mainline users, another option would be to just merge it and fix regressions if someone reports them. Once this is applied, I'll post another patch that removes the calculations and the calls to snd_soc_dai_set_{clkdiv,pll} from the zylonite and magician machine drivers. Thanks, Daniel include/linux/pxa2xx_ssp.h | 8 ++ sound/soc/pxa/pxa-ssp.c | 146 ++++++++++++++++++------------------- 2 files changed, 81 insertions(+), 73 deletions(-) diff --git a/include/linux/pxa2xx_ssp.h b/include/linux/pxa2xx_ssp.h index 03a7ca46735b..13b4244d44c1 100644 --- a/include/linux/pxa2xx_ssp.h +++ b/include/linux/pxa2xx_ssp.h @@ -171,6 +171,14 @@ #define SSACD_SCDB (1 << 3) /* SSPSYSCLK Divider Bypass */ #define SSACD_ACPS(x) ((x) << 4) /* Audio clock PLL select */ #define SSACD_ACDS(x) ((x) << 0) /* Audio clock divider select */ +#define SSACD_ACDS_1 (0) +#define SSACD_ACDS_2 (1) +#define SSACD_ACDS_4 (2) +#define SSACD_ACDS_8 (3) +#define SSACD_ACDS_16 (4) +#define SSACD_ACDS_32 (5) +#define SSACD_SCDB_4X (0) +#define SSACD_SCDB_1X (1) #define SSACD_SCDX8 (1 << 7) /* SYSCLK division ratio select */ /* LPSS SSP */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 01d54697ede4..f8339bb01251 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -41,6 +41,7 @@ */ struct ssp_priv { struct ssp_device *ssp; + unsigned long ssp_clk; unsigned int sysclk; unsigned int dai_fmt; unsigned int configured_dai_fmt; @@ -192,21 +193,6 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) pxa_ssp_write_reg(ssp, SSCR0, sscr0); } -/** - * pxa_ssp_get_clkdiv - get SSP clock divider - */ -static u32 pxa_ssp_get_scr(struct ssp_device *ssp) -{ - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - u32 div; - - if (ssp->type == PXA25x_SSP) - div = ((sscr0 >> 8) & 0xff) * 2 + 2; - else - div = ((sscr0 >> 8) & 0xfff) + 1; - return div; -} - /* * Set the SSP ports SYSCLK. */ @@ -262,67 +248,18 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return 0; } -/* - * Set the SSP clock dividers. - */ -static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); - struct ssp_device *ssp = priv->ssp; - int val; - - switch (div_id) { - case PXA_SSP_AUDIO_DIV_ACDS: - val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); - pxa_ssp_write_reg(ssp, SSACD, val); - break; - case PXA_SSP_AUDIO_DIV_SCDB: - val = pxa_ssp_read_reg(ssp, SSACD); - val &= ~SSACD_SCDB; - if (ssp->type == PXA3xx_SSP) - val &= ~SSACD_SCDX8; - switch (div) { - case PXA_SSP_CLK_SCDB_1: - val |= SSACD_SCDB; - break; - case PXA_SSP_CLK_SCDB_4: - break; - case PXA_SSP_CLK_SCDB_8: - if (ssp->type == PXA3xx_SSP) - val |= SSACD_SCDX8; - else - return -EINVAL; - break; - default: - return -EINVAL; - } - pxa_ssp_write_reg(ssp, SSACD, val); - break; - case PXA_SSP_DIV_SCR: - pxa_ssp_set_scr(ssp, div); - break; - default: - return -ENODEV; - } - - return 0; -} - /* * Configure the PLL frequency pxa27x and (afaik - pxa320 only) */ -static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) +static int pxa_ssp_set_pll(struct ssp_priv *priv, unsigned int freq) { - struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; if (ssp->type == PXA3xx_SSP) pxa_ssp_write_reg(ssp, SSACDD, 0); - switch (freq_out) { + switch (freq) { case 5622000: break; case 11345000: @@ -353,7 +290,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, u64 tmp = 19968; tmp *= 1000000; - do_div(tmp, freq_out); + do_div(tmp, freq); val = tmp; val = (val << 16) | 64; @@ -363,7 +300,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, dev_dbg(&ssp->pdev->dev, "Using SSACDD %x to supply %uHz\n", - val, freq_out); + val, freq); break; } @@ -568,6 +505,24 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) return 0; } +struct pxa_ssp_clock_mode { + int rate; + int pll; + u8 acds; + u8 scdb; +}; + +static const struct pxa_ssp_clock_mode pxa_ssp_clock_modes[] = { + { .rate = 8000, .pll = 32842000, .acds = SSACD_ACDS_32, .scdb = SSACD_SCDB_4X }, + { .rate = 11025, .pll = 5622000, .acds = SSACD_ACDS_4, .scdb = SSACD_SCDB_4X }, + { .rate = 16000, .pll = 32842000, .acds = SSACD_ACDS_16, .scdb = SSACD_SCDB_4X }, + { .rate = 22050, .pll = 5622000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 44100, .pll = 11345000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 48000, .pll = 12235000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 96000, .pll = 12235000, .acds = SSACD_ACDS_4, .scdb = SSACD_SCDB_1X }, + {} +}; + /* * Set the SSP audio DMA parameters and sample size. * Can be called multiple times by oss emulation. @@ -579,11 +534,12 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; int chn = params_channels(params); - u32 sscr0; - u32 sspsp; + u32 sscr0, sspsp; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; struct snd_dmaengine_dai_dma_data *dma_data; + int rate = params_rate(params); + int bclk = rate * chn * (width / 8); int ret; dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); @@ -623,11 +579,57 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, } pxa_ssp_write_reg(ssp, SSCR0, sscr0); + if (sscr0 & SSCR0_ACS) { + ret = pxa_ssp_set_pll(priv, bclk); + + /* + * If we were able to generate the bclk directly, + * all is fine. Otherwise, look up the closest rate + * from the table and also set the dividers. + */ + + if (ret < 0) { + const struct pxa_ssp_clock_mode *m; + int ssacd, acds; + + for (m = pxa_ssp_clock_modes; m->rate; m++) { + if (m->rate == rate) + break; + } + + if (!m->rate) + return -EINVAL; + + acds = m->acds; + + /* The values in the table are for 16 bits */ + if (width == 32) + acds--; + + ret = pxa_ssp_set_pll(priv, bclk); + if (ret < 0) + return ret; + + ssacd = pxa_ssp_read_reg(ssp, SSACD); + ssacd &= ~(SSACD_ACDS(7) | SSACD_SCDB_1X); + ssacd |= SSACD_ACDS(m->acds); + ssacd |= m->scdb; + pxa_ssp_write_reg(ssp, SSACD, ssacd); + } + } else if (sscr0 & SSCR0_ECS) { + /* + * For setups with external clocking, the PLL and its diviers + * are not active. Instead, the SCR bits in SSCR0 can be used + * to divide the clock. + */ + pxa_ssp_set_scr(ssp, bclk / rate); + } + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: sspsp = pxa_ssp_read_reg(ssp, SSPSP); - if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { + if (((priv->sysclk / bclk) == 64) && (width == 16)) { /* This is a special case where the bitclk is 64fs * and we're not dealing with 2*32 bits of audio * samples. @@ -812,8 +814,6 @@ static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { .trigger = pxa_ssp_trigger, .hw_params = pxa_ssp_hw_params, .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, .set_fmt = pxa_ssp_set_dai_fmt, .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, .set_tristate = pxa_ssp_set_dai_tristate, -- 2.17.1

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[alsa-devel] [asoc:for-4.19 92/94] sound/soc/pxa/pxa-ssp.o:undefined reference to `pxa2xx_soc_pcm_new'
by kbuild test robot 30 Jun '18

30 Jun '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.19 head: c1d9b4196ba6b311bd48a9320cd46aa125e0b034 commit: 0a94cf3457408058f894cc4d95e58d8e18eb7f75 [92/94] ASoC: pxa: make SND_PXA2XX_SOC_I2S selectable config: x86_64-randconfig-x000-201826 (attached as .config) compiler: gcc-7 (Debian 7.3.0-16) 7.3.0 reproduce: git checkout 0a94cf3457408058f894cc4d95e58d8e18eb7f75 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): >> sound/soc/pxa/pxa-ssp.o:(.rodata+0x268): undefined reference to `pxa2xx_soc_pcm_new' >> sound/soc/pxa/pxa-ssp.o:(.rodata+0x270): undefined reference to `pxa2xx_pcm_free_dma_buffers' >> sound/soc/pxa/pxa-ssp.o:(.rodata+0x2b8): undefined reference to `pxa2xx_pcm_ops' --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 0/2] ASoC: add es7241 support
by Jerome Brunet 30 Jun '18

30 Jun '18

Add support for the es7241 which is a simple 2 channels analog to digital converter. Jerome Brunet (2): ASoC: es7241: add dt-bindings documentation for the es7241 adc ASoC: es7241: add es7241 codec support .../devicetree/bindings/sound/everest,es7241.txt | 23 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es7241.c | 307 +++++++++++++++++++++ 4 files changed, 336 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/everest,es7241.txt create mode 100644 sound/soc/codecs/es7241.c -- 2.14.4

2 3

[alsa-devel] dsd formats in ALSA plugins
by slava 30 Jun '18

30 Jun '18

Yes, patch from previous post works as expected. I try to write external filter plugin for dinamically change slave format ( dsd_xxx -> xxx), unsuccessfully ( It suppose only static config or i mistake? Thanks

1 0

[alsa-devel] [PATCH] ASoC: Intel: kbl_da7219_max98357a: Map BTN_0 to KEY_PLAYPAUSE
by mac.chiang＠intel.com 29 Jun '18

29 Jun '18

From: Mac Chiang <mac.chiang(a)intel.com> On kbl_rt5663_max98927, commit 38a5882e4292 ("ASoC: Intel: kbl_rt5663_max98927: Map BTN_0 to KEY_PLAYPAUSE") This key pair mapping to play/pause when playing Youtube The Android 3.5mm Headset jack specification mentions that BTN_0 should be mapped to KEY_MEDIA, but this is less logical than KEY_PLAYPAUSE, which has much broader userspace support. For example, the Chrome OS userspace now supports KEY_PLAYPAUSE to toggle play/pause of videos and audio, but does not handle KEY_MEDIA. Furthermore, Android itself now supports KEY_PLAYPAUSE equivalently, as the new USB headset spec requires KEY_PLAYPAUSE for BTN_0. https://source.android.com/devices/accessories/headset/usb-headset-spec The same fix is required on Chrome kbl_da7219_max98357a. Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 7961f1f..edafe23 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -188,7 +188,7 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) jack = &ctx->kabylake_headset; - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); -- 2.7.4

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[alsa-devel] [PATCH RESEND] ASoC: pxa: remove the dmaengine compat need
by Daniel Mack 29 Jun '18

29 Jun '18

From: Robert Jarzmik <robert.jarzmik(a)free.fr> As the pxa architecture switched towards the dmaengine slave map, the old compatibility mechanism to acquire the dma requestor line number and priority are not needed anymore. This patch simplifies the dma resource acquisition, using the more generic function dma_request_slave_channel(). Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> Signed-off-by: Daniel Mack <daniel(a)zonque.org> Acked-by: Mark Brown <broonie(a)kernel.org> --- sound/arm/pxa2xx-ac97.c | 14 ++------------ sound/arm/pxa2xx-pcm-lib.c | 6 +++--- sound/soc/pxa/pxa2xx-ac97.c | 32 +++++--------------------------- sound/soc/pxa/pxa2xx-i2s.c | 6 ++---- 4 files changed, 12 insertions(+), 46 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 4bc244c40f80..236a63cdaf9f 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -63,28 +63,18 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_legacy_reset, }; -static struct pxad_param pxa2xx_ac97_pcm_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 12, -}; - static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_out", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_out_req, -}; - -static struct pxad_param pxa2xx_ac97_pcm_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 11, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_in", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_in_req, }; static struct snd_pcm *pxa2xx_ac97_pcm; diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index e8da3b8ee721..dcbe7ecc1835 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -125,9 +125,9 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) return ret; - return snd_dmaengine_pcm_open_request_chan(substream, - pxad_filter_fn, - dma_params->filter_data); + return snd_dmaengine_pcm_open( + substream, dma_request_slave_channel(rtd->cpu_dai->dev, + dma_params->chan_name)); } EXPORT_SYMBOL(__pxa2xx_pcm_open); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 803818aabee9..1b41c0f2a8fb 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -68,61 +68,39 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 11, -}; - static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_in", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, -}; - -static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 12, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_out", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_stereo_out_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 10, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_out", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 9, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_in", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 8, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .addr = __PREG(MCDR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mic_mono", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req, }; static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream, diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 3fb60baf6eab..e7184de0de04 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -82,20 +82,18 @@ static struct pxa_i2s_port pxa_i2s; static struct clk *clk_i2s; static int clk_ena = 0; -static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = { .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "tx", .maxburst = 32, - .filter_data = &pxa2xx_i2s_pcm_stereo_out_req, }; -static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = { .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "rx", .maxburst = 32, - .filter_data = &pxa2xx_i2s_pcm_stereo_in_req, }; static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: snd-aoa: add of_node_put() in error path
by Nicholas Mc Guire 29 Jun '18

29 Jun '18

Both calls to of_find_node_by_name() and of_get_next_child() return a node pointer with refcount incremented thus it must be explicidly decremented here after the last usage. As we are assured to have a refcounted np either from the initial of_find_node_by_name(NULL, name); or from the of_get_next_child(gpio, np) in the while loop if we reached the error code path below, an x of_node_put(np) is needed. Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> Fixes: commit f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa") --- Problem located by an experimental coccinelle script Patch was compiletested with: ppc64_defconfig (implies CONFIG_SND_AOA=m) Patch is against 4.18-rc2 (localversion-next is next-20180629) sound/aoa/core/gpio-feature.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c index 7196008..6555742 100644 --- a/sound/aoa/core/gpio-feature.c +++ b/sound/aoa/core/gpio-feature.c @@ -88,8 +88,10 @@ static struct device_node *get_gpio(char *name, } reg = of_get_property(np, "reg", NULL); - if (!reg) + if (!reg) { + of_node_put(np); return NULL; + } *gpioptr = *reg; -- 2.1.4

2 1

[alsa-devel] snd_pcm_hw_params_set_buffer_time_near fails with "Invalid argument" when var_periodsize is enabled
by Wischer, Timo (ADITG/ESB) 29 Jun '18

29 Jun '18

Hi all, at the moment I am investigating in the issue described in [1]. This issue is already avoided by a workaround [2]. I am able to reproduce the issue with a more simple aplay use case using the following configuration: pcm_slave.adr3_tdm_8ch { pcm { type hw card "Loopback" device 0 } rate 48000 period_size 128 buffer_size 1024 channels 2 } pcm.dshare_Playback_3 { type dmix ipc_key 600 ipc_perm 0660 ipc_gid audio var_periodsize true slave adr3_tdm_8ch } pcm.AdevAcousticoutSpeech { type rate slave.pcm dshare_Playback_3 slave.rate 48000 } $ modprobe snd_aloop $ aplay -v --period-size=352 -c2 -fS16_LE -r22500 -D AdevAcousticoutSpeech /dev/urandom aplay: ../../alsa-utils-1.1.5/aplay/aplay.c:1390: set_params: Assertion `err >= 0' failed. Aborted by signal Aborted... It looks like the refinement fails. Now, I need your help because the intervals seems not to be handled in the same way in each function and I want to understand what is the correct interpretation of an interval. For example snd_interval_checkempty() and snd_interval_single() conflicting with each other. snd_interval_checkempty() interprets (x x] and [x x) as an empty set but snd_interval_single() does not. (x x) seems not to be an empty set as implemented by snd_interval_single(). Therefore could someone please explain me what is the correct interpretation of the snd_interval_t in combination with openmin and openmax. The following seems to be consistent (x x+1) and [x x+1) → x (see snd_interval_single() and snd_interval_value()) but it is different to what I have learnt at university. So it would be very helpful if someone could direct me to the right definition. So I hope I could easier solve this issue. [1] https://bugzilla.opensuse.org/show_bug.cgi?id=1033179 [2] http://git.alsa-project.org/?p=alsa-lib.git;a=commit;h=e736715a8ed4c8690059… Best regards Timo Wischer Advanced Driver Information Technology GmbH Engineering Software Base (ADITG/ESB) Robert-Bosch-Str. 200 31139 Hildesheim Germany Tel. +49 5121 49 6938 Fax +49 5121 49 6999 twischer(a)de.adit-jv.com ADIT is a joint venture company of Robert Bosch GmbH/Robert Bosch Car Multimedia GmbH and DENSO Corporation Sitz: Hildesheim, Registergericht: Amtsgericht Hildesheim HRB 3438 Geschäftsführung: Wilhelm Grabow, Ken Yaguchi

1 0

[alsa-devel] [PATCH v2] ALSA: hda - Handle pm failure during hotplug
by Takashi Iwai 29 Jun '18

29 Jun '18

From: Chris Wilson <chris(a)chris-wilson.co.uk> Obtaining the runtime pm wakeref can fail, especially in a hotplug scenario where i915.ko has been unloaded. If we do not catch the failure, we end up with an unbalanced pm. [ Additional changes: hdmi_present_sense() checks the return value and handle only a negative error case and bails out only if it's really still suspended. Also, snd_hda_power_down() is called at the error path so that the refcount is balanced. Along with it, the spec->pcm_lock is taken outside hdmi_present_sense() in the caller side, so that it won't cause deadlock at reentrace via runtime resume. -- tiwai ] References: 222bde03881c ("ALSA: hda - Fix mutex deadlock at HDMI/DP hotplug") Signed-off-by: Chris Wilson <chris(a)chris-wilson.co.uk> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 98e1c411c56a..9e3916ed2588 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -764,8 +764,10 @@ static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid, if (pin_idx < 0) return; + mutex_lock(&spec->pcm_lock); if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) snd_hda_jack_report_sync(codec); + mutex_unlock(&spec->pcm_lock); } static void jack_callback(struct hda_codec *codec, @@ -1628,21 +1630,23 @@ static void sync_eld_via_acomp(struct hda_codec *codec, static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { struct hda_codec *codec = per_pin->codec; - struct hdmi_spec *spec = codec->spec; int ret; /* no temporary power up/down needed for component notifier */ - if (!codec_has_acomp(codec)) - snd_hda_power_up_pm(codec); + if (!codec_has_acomp(codec)) { + ret = snd_hda_power_up_pm(codec); + if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) { + snd_hda_power_down_pm(codec); + return false; + } + } - mutex_lock(&spec->pcm_lock); if (codec_has_acomp(codec)) { sync_eld_via_acomp(codec, per_pin); ret = false; /* don't call snd_hda_jack_report_sync() */ } else { ret = hdmi_present_sense_via_verbs(per_pin, repoll); } - mutex_unlock(&spec->pcm_lock); if (!codec_has_acomp(codec)) snd_hda_power_down_pm(codec); @@ -1654,12 +1658,16 @@ static void hdmi_repoll_eld(struct work_struct *work) { struct hdmi_spec_per_pin *per_pin = container_of(to_delayed_work(work), struct hdmi_spec_per_pin, work); + struct hda_codec *codec = per_pin->codec; + struct hdmi_spec *spec = codec->spec; if (per_pin->repoll_count++ > 6) per_pin->repoll_count = 0; + mutex_lock(&spec->pcm_lock); if (hdmi_present_sense(per_pin, per_pin->repoll_count)) snd_hda_jack_report_sync(per_pin->codec); + mutex_unlock(&spec->pcm_lock); } static void intel_haswell_fixup_connect_list(struct hda_codec *codec, -- 2.17.1

1 0

[alsa-devel] Applied "ASoC: mediatek: sub dai use list_head" to the asoc tree
by Mark Brown 29 Jun '18

29 Jun '18

The patch ASoC: mediatek: sub dai use list_head has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f11c5db770ab675d270cb4d5a2bb90923066ef49 Mon Sep 17 00:00:00 2001 From: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> Date: Fri, 29 Jun 2018 20:29:44 +0800 Subject: [PATCH] ASoC: mediatek: sub dai use list_head use list_head for sub_dais, since original sub_dais array is sparsely occupied Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../mediatek/common/mtk-afe-platform-driver.c | 64 +++++++------------ sound/soc/mediatek/common/mtk-base-afe.h | 6 +- 2 files changed, 28 insertions(+), 42 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 51ec4ff6ed95..697aa50aff9a 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -15,20 +15,12 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) { - struct snd_soc_dai_driver *sub_dai_drivers; + struct mtk_base_afe_dai *dai; size_t num_dai_drivers = 0, dai_idx = 0; - int i; - - if (!afe->sub_dais) { - dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); - return -EINVAL; - } /* calcualte total dai driver size */ - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].dai_drivers && - afe->sub_dais[i].num_dai_drivers != 0) - num_dai_drivers += afe->sub_dais[i].num_dai_drivers; + list_for_each_entry(dai, &afe->sub_dais, list) { + num_dai_drivers += dai->num_dai_drivers; } dev_info(afe->dev, "%s(), num of dai %zd\n", __func__, num_dai_drivers); @@ -42,19 +34,14 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) if (!afe->dai_drivers) return -ENOMEM; - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].dai_drivers && - afe->sub_dais[i].num_dai_drivers != 0) { - sub_dai_drivers = afe->sub_dais[i].dai_drivers; - /* dai driver */ - memcpy(&afe->dai_drivers[dai_idx], - sub_dai_drivers, - afe->sub_dais[i].num_dai_drivers * - sizeof(struct snd_soc_dai_driver)); - dai_idx += afe->sub_dais[i].num_dai_drivers; - } + list_for_each_entry(dai, &afe->sub_dais, list) { + /* dai driver */ + memcpy(&afe->dai_drivers[dai_idx], + dai->dai_drivers, + dai->num_dai_drivers * + sizeof(struct snd_soc_dai_driver)); + dai_idx += dai->num_dai_drivers; } - return 0; } EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai); @@ -62,28 +49,25 @@ EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai); int mtk_afe_add_sub_dai_control(struct snd_soc_component *component) { struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - int i; + struct mtk_base_afe_dai *dai; - if (!afe->sub_dais) { - dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); - return -EINVAL; - } - - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].controls) + list_for_each_entry(dai, &afe->sub_dais, list) { + if (dai->controls) snd_soc_add_component_controls(component, - afe->sub_dais[i].controls, - afe->sub_dais[i].num_controls); + dai->controls, + dai->num_controls); - if (afe->sub_dais[i].dapm_widgets) + if (dai->dapm_widgets) snd_soc_dapm_new_controls(&component->dapm, - afe->sub_dais[i].dapm_widgets, - afe->sub_dais[i].num_dapm_widgets); - - if (afe->sub_dais[i].dapm_routes) + dai->dapm_widgets, + dai->num_dapm_widgets); + } + /* add routes after all widgets are added */ + list_for_each_entry(dai, &afe->sub_dais, list) { + if (dai->dapm_routes) snd_soc_dapm_add_routes(&component->dapm, - afe->sub_dais[i].dapm_routes, - afe->sub_dais[i].num_dapm_routes); + dai->dapm_routes, + dai->num_dapm_routes); } snd_soc_dapm_new_widgets(component->dapm.card); diff --git a/sound/soc/mediatek/common/mtk-base-afe.h b/sound/soc/mediatek/common/mtk-base-afe.h index bcf562f029b6..bd8d5e0c6843 100644 --- a/sound/soc/mediatek/common/mtk-base-afe.h +++ b/sound/soc/mediatek/common/mtk-base-afe.h @@ -46,6 +46,7 @@ struct mtk_base_irq_data { }; struct device; +struct list_head; struct mtk_base_afe_memif; struct mtk_base_afe_irq; struct mtk_base_afe_dai; @@ -72,8 +73,7 @@ struct mtk_base_afe { struct mtk_base_afe_irq *irqs; int irqs_size; - struct mtk_base_afe_dai *sub_dais; - int num_sub_dais; + struct list_head sub_dais; struct snd_soc_dai_driver *dai_drivers; unsigned int num_dai_drivers; @@ -110,6 +110,8 @@ struct mtk_base_afe_dai { unsigned int num_dapm_widgets; const struct snd_soc_dapm_route *dapm_routes; unsigned int num_dapm_routes; + + struct list_head list; }; #endif -- 2.18.0.rc2

1 0

[alsa-devel] Applied "ASoC: mt6797: sub dai use list_head" to the asoc tree
by Mark Brown 29 Jun '18

29 Jun '18

The patch ASoC: mt6797: sub dai use list_head has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c1d9b4196ba6b311bd48a9320cd46aa125e0b034 Mon Sep 17 00:00:00 2001 From: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> Date: Fri, 29 Jun 2018 20:29:45 +0800 Subject: [PATCH] ASoC: mt6797: sub dai use list_head Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/mediatek/mt6797/mt6797-afe-common.h | 1 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 65 +++++++++++++------ sound/soc/mediatek/mt6797/mt6797-dai-adda.c | 20 ++++-- .../soc/mediatek/mt6797/mt6797-dai-hostless.c | 16 +++-- sound/soc/mediatek/mt6797/mt6797-dai-pcm.c | 19 ++++-- 5 files changed, 81 insertions(+), 40 deletions(-) diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-common.h b/sound/soc/mediatek/mt6797/mt6797-afe-common.h index 22eb7b455cf1..4eac9977b2b0 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-common.h +++ b/sound/soc/mediatek/mt6797/mt6797-afe-common.h @@ -10,6 +10,7 @@ #define _MT_6797_AFE_COMMON_H_ #include <sound/soc.h> +#include <linux/list.h> #include <linux/regmap.h> #include "../common/mtk-base-afe.h" diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 6c5dd9fc9976..192f4d7b37b6 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -733,6 +733,34 @@ static const struct snd_soc_component_driver mt6797_afe_component = { .probe = mt6797_afe_component_probe, }; +static int mt6797_dai_memif_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mt6797_memif_dai_driver; + dai->num_dai_drivers = ARRAY_SIZE(mt6797_memif_dai_driver); + + dai->dapm_widgets = mt6797_memif_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mt6797_memif_widgets); + dai->dapm_routes = mt6797_memif_routes; + dai->num_dapm_routes = ARRAY_SIZE(mt6797_memif_routes); + return 0; +} + +typedef int (*dai_register_cb)(struct mtk_base_afe *); +static const dai_register_cb dai_register_cbs[] = { + mt6797_dai_adda_register, + mt6797_dai_pcm_register, + mt6797_dai_hostless_register, + mt6797_dai_memif_register, +}; + static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) { struct mtk_base_afe *afe; @@ -811,29 +839,24 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) } /* init sub_dais */ - afe->num_sub_dais = MT6797_DAI_NUM; - afe->sub_dais = devm_kcalloc(dev, afe->num_sub_dais, - sizeof(*afe->sub_dais), - GFP_KERNEL); - if (!afe->sub_dais) - return -ENOMEM; - - mt6797_dai_adda_register(afe); - mt6797_dai_pcm_register(afe); - mt6797_dai_hostless_register(afe); - - afe->sub_dais[MT6797_MEMIF_DL1].dai_drivers = mt6797_memif_dai_driver; - afe->sub_dais[MT6797_MEMIF_DL1].num_dai_drivers = - ARRAY_SIZE(mt6797_memif_dai_driver); - afe->sub_dais[MT6797_MEMIF_DL1].dapm_widgets = mt6797_memif_widgets; - afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_widgets = - ARRAY_SIZE(mt6797_memif_widgets); - afe->sub_dais[MT6797_MEMIF_DL1].dapm_routes = mt6797_memif_routes; - afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_routes = - ARRAY_SIZE(mt6797_memif_routes); + INIT_LIST_HEAD(&afe->sub_dais); + + for (i = 0; i < ARRAY_SIZE(dai_register_cbs); i++) { + ret = dai_register_cbs[i](afe); + if (ret) { + dev_warn(afe->dev, "dai register i %d fail, ret %d\n", + i, ret); + return ret; + } + } /* init dai_driver and component_driver */ - mtk_afe_combine_sub_dai(afe); + ret = mtk_afe_combine_sub_dai(afe); + if (ret) { + dev_warn(afe->dev, "mtk_afe_combine_sub_dai fail, ret %d\n", + ret); + return ret; + } afe->mtk_afe_hardware = &mt6797_afe_hardware; afe->memif_fs = mt6797_memif_fs; diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c index ad083265ce94..0ac6409c6d61 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c @@ -383,14 +383,20 @@ static struct snd_soc_dai_driver mtk_dai_adda_driver[] = { int mt6797_dai_adda_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_ADDA; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_adda_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_widgets = mtk_dai_adda_widgets; - afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets); - afe->sub_dais[id].dapm_routes = mtk_dai_adda_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes); + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_adda_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver); + + dai->dapm_widgets = mtk_dai_adda_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets); + dai->dapm_routes = mtk_dai_adda_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes); return 0; } diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c index 4cf985b15a11..ed23e6a53b08 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c @@ -100,13 +100,19 @@ static struct snd_soc_dai_driver mtk_dai_hostless_driver[] = { int mt6797_dai_hostless_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_HOSTLESS_LPBK; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_hostless_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_routes = mtk_dai_hostless_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes); + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_hostless_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver); + + dai->dapm_routes = mtk_dai_hostless_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes); return 0; } diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c index 16d5b5067204..3136f0bc7827 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c @@ -298,15 +298,20 @@ static struct snd_soc_dai_driver mtk_dai_pcm_driver[] = { int mt6797_dai_pcm_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_PCM_1; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_pcm_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_widgets = mtk_dai_pcm_widgets; - afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets); - afe->sub_dais[id].dapm_routes = mtk_dai_pcm_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes); + list_add(&dai->list, &afe->sub_dais); + dai->dai_drivers = mtk_dai_pcm_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver); + + dai->dapm_widgets = mtk_dai_pcm_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets); + dai->dapm_routes = mtk_dai_pcm_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes); return 0; } -- 2.18.0.rc2

1 0

[alsa-devel] [PATCH 1/2] ASoC: mediatek: sub dai use list_head
by KaiChieh Chuang 29 Jun '18

29 Jun '18

use list_head for sub_dais, since original sub_dais array is sparsely occupied Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- .../soc/mediatek/common/mtk-afe-platform-driver.c | 64 ++++++++-------------- sound/soc/mediatek/common/mtk-base-afe.h | 6 +- 2 files changed, 28 insertions(+), 42 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 0061858..3a5bde6 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -15,20 +15,12 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) { - struct snd_soc_dai_driver *sub_dai_drivers; + struct mtk_base_afe_dai *dai; size_t num_dai_drivers = 0, dai_idx = 0; - int i; - - if (!afe->sub_dais) { - dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); - return -EINVAL; - } /* calcualte total dai driver size */ - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].dai_drivers && - afe->sub_dais[i].num_dai_drivers != 0) - num_dai_drivers += afe->sub_dais[i].num_dai_drivers; + list_for_each_entry(dai, &afe->sub_dais, list) { + num_dai_drivers += dai->num_dai_drivers; } dev_info(afe->dev, "%s(), num of dai %zd\n", __func__, num_dai_drivers); @@ -42,47 +34,39 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) if (!afe->dai_drivers) return -ENOMEM; - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].dai_drivers && - afe->sub_dais[i].num_dai_drivers != 0) { - sub_dai_drivers = afe->sub_dais[i].dai_drivers; - /* dai driver */ - memcpy(&afe->dai_drivers[dai_idx], - sub_dai_drivers, - afe->sub_dais[i].num_dai_drivers * - sizeof(struct snd_soc_dai_driver)); - dai_idx += afe->sub_dais[i].num_dai_drivers; - } + list_for_each_entry(dai, &afe->sub_dais, list) { + /* dai driver */ + memcpy(&afe->dai_drivers[dai_idx], + dai->dai_drivers, + dai->num_dai_drivers * + sizeof(struct snd_soc_dai_driver)); + dai_idx += dai->num_dai_drivers; } - return 0; } int mtk_afe_add_sub_dai_control(struct snd_soc_component *component) { struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - int i; + struct mtk_base_afe_dai *dai; - if (!afe->sub_dais) { - dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); - return -EINVAL; - } - - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].controls) + list_for_each_entry(dai, &afe->sub_dais, list) { + if (dai->controls) snd_soc_add_component_controls(component, - afe->sub_dais[i].controls, - afe->sub_dais[i].num_controls); + dai->controls, + dai->num_controls); - if (afe->sub_dais[i].dapm_widgets) + if (dai->dapm_widgets) snd_soc_dapm_new_controls(&component->dapm, - afe->sub_dais[i].dapm_widgets, - afe->sub_dais[i].num_dapm_widgets); - - if (afe->sub_dais[i].dapm_routes) + dai->dapm_widgets, + dai->num_dapm_widgets); + } + /* add routes after all widgets are added */ + list_for_each_entry(dai, &afe->sub_dais, list) { + if (dai->dapm_routes) snd_soc_dapm_add_routes(&component->dapm, - afe->sub_dais[i].dapm_routes, - afe->sub_dais[i].num_dapm_routes); + dai->dapm_routes, + dai->num_dapm_routes); } snd_soc_dapm_new_widgets(component->dapm.card); diff --git a/sound/soc/mediatek/common/mtk-base-afe.h b/sound/soc/mediatek/common/mtk-base-afe.h index bcf562f0..bd8d5e0 100644 --- a/sound/soc/mediatek/common/mtk-base-afe.h +++ b/sound/soc/mediatek/common/mtk-base-afe.h @@ -46,6 +46,7 @@ struct mtk_base_irq_data { }; struct device; +struct list_head; struct mtk_base_afe_memif; struct mtk_base_afe_irq; struct mtk_base_afe_dai; @@ -72,8 +73,7 @@ struct mtk_base_afe { struct mtk_base_afe_irq *irqs; int irqs_size; - struct mtk_base_afe_dai *sub_dais; - int num_sub_dais; + struct list_head sub_dais; struct snd_soc_dai_driver *dai_drivers; unsigned int num_dai_drivers; @@ -110,6 +110,8 @@ struct mtk_base_afe_dai { unsigned int num_dapm_widgets; const struct snd_soc_dapm_route *dapm_routes; unsigned int num_dapm_routes; + + struct list_head list; }; #endif -- 1.9.1

1 1

Re: [alsa-devel] [PATCH] vgaswitchroo: set audio client id according to bound gpu client id
by Lukas Wunner 29 Jun '18

29 Jun '18

On Fri, Jun 29, 2018 at 08:55:40AM +0000, Qu, Jim wrote: > When our dGPU does suspend by runtime pm. amdgpu driver for dgpu will > also call vgaswtichroo to power off its audio. vgaswitchroo driver will > find audio codec by vgaswitchroo dgpu client id(VGA_SWITCHEROO_DIS). That is no longer the case since v4.17. The HDA controller now runtime suspends autonomously, see commit 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller"). Your patch appears to be geared towards an older kernel version. Please retest on the laptop in question with a v4.17+ kernel. > I think the issue should be observed on both Intel+AMD or AMD+AMD > platform which has the same HW configuration. > > 1.if dGPU has no audio codec. the issue should be always observed. > 2.if both iGPU and dGPU has audio codecs, the issue should be random, > it depends on the first audio found by vgaswitchroo driver is on iGPU > or dGPU. On discrete AMD and Nvidia GPUs, the HDA controller is function 1 and the GPU is function 0 in the same PCI slot. On Intel chipsets, the HDA controller and the GPU have completely different PCI device numbers, e.g. the GPU might be 0000:00:02.0 and the HDA controller might be 0000:00:1b.0. get_bound_vga() checks that the HDA controller is function 1 and there's a GPU in function 0 of the same slot. Thus get_bound_vga() always returns NULL for an Intel HDA controller and the controller is never registered with vga_switcheroo (which is fine). Thanks, Lukas

2 2

[alsa-devel] [PATCH v2 00/13] ARM: pxa: switch to DMA slave maps
by Robert Jarzmik 29 Jun '18

29 Jun '18

This v1 cover letter is quoted in [1]. For maintainers the table below should help you focus on the patches targetted at you, and ignore the other noise. The differences since v1 is by maintainers / topic / patch : - Arnd and Daniel / PXA topic / 0002 devices.c split into pxa25x.c, pxa27x.c and pxa3xx.c - Boris and Daniel / MTD topic / 0005 Review and ack of this one - Arnd and netdev / NET topic / 0006 and 0007 Arnd comment taken, review and ack of these one - Mark and alsa-devel / ASoC topic / 0008 Mark, I couldn't keep your former Ack because : - I changed one line so that the cpu device provides the DMA - I added pxa2xx-i2s which was forgotten Therefore I need a new ack - Arnd and Daniel / SSP topic / 0013 Review and ack of this one Happy review. -- Robert Robert Jarzmik (13): dmaengine: pxa: use a dma slave map ARM: pxa: add dma slave map mmc: pxamci: remove the dmaengine compat need media: pxa_camera: remove the dmaengine compat need mtd: rawnand: marvell: remove the dmaengine compat need net: smc911x: remove the dmaengine compat need net: smc91x: remove the dmaengine compat need ASoC: pxa: remove the dmaengine compat need ata: pata_pxa: remove the dmaengine compat need dmaengine: pxa: document pxad_param dmaengine: pxa: make the filter function internal ARM: pxa: remove the DMA IO resources ARM: pxa: change SSP DMA channels allocation arch/arm/mach-pxa/devices.c | 148 +--------------------------------- arch/arm/mach-pxa/devices.h | 6 +- arch/arm/mach-pxa/pxa25x.c | 41 +++++++++- arch/arm/mach-pxa/pxa27x.c | 42 +++++++++- arch/arm/mach-pxa/pxa3xx.c | 44 +++++++++- arch/arm/plat-pxa/ssp.c | 47 ----------- drivers/ata/pata_pxa.c | 10 +-- drivers/dma/pxa_dma.c | 13 ++- drivers/media/platform/pxa_camera.c | 22 +---- drivers/mmc/host/pxamci.c | 29 +------ drivers/mtd/nand/raw/marvell_nand.c | 17 +--- drivers/net/ethernet/smsc/smc911x.c | 16 +--- drivers/net/ethernet/smsc/smc91x.c | 12 +-- drivers/net/ethernet/smsc/smc91x.h | 1 - include/linux/dma/pxa-dma.h | 20 +++-- include/linux/platform_data/mmp_dma.h | 4 + include/linux/pxa2xx_ssp.h | 2 - sound/arm/pxa2xx-ac97.c | 14 +--- sound/arm/pxa2xx-pcm-lib.c | 6 +- sound/soc/pxa/pxa-ssp.c | 5 +- sound/soc/pxa/pxa2xx-ac97.c | 32 ++------ sound/soc/pxa/pxa2xx-i2s.c | 6 +- 22 files changed, 180 insertions(+), 357 deletions(-) -- 2.11.0 --- [1] Former v1 cover letter This serie is aimed at removing the dmaengine slave compat use, and transfer knowledge of the DMA requestors into architecture code. This was discussed/advised by Arnd a couple of years back, it's almost time. The serie is divided in 3 phasees : - phase 1 : patch 1/15 and patch 2/15 => this is the preparation work - phase 2 : patches 3/15 .. 10/15 => this is the switch of all the drivers => this one will require either an Ack of the maintainers or be taken by them once phase 1 is merged - phase 3 : patches 11/15 => this is the last part, cleanup and removal of export of the DMA filter function As this looks like a patch bomb, each maintainer expressing for his tree either an Ack or "I want to take through my tree" will be spared in the next iterations of this serie. Several of these changes have been tested on actual hardware, including : - pxamci - pxa_camera - smc* - ASoC and SSP

5 27

[alsa-devel] [PATCH 0/7] ASoC: pxa: code refactoring
by Daniel Mack 29 Jun '18

29 Jun '18

This is a series to clean up the code base in the ASoC/PXA area a bit. It folds some implementations, removes some files and brings pcm ops to the CPU DAIS, so they can be used on DT platforms without platform drivers. They are a preparation for some more patches that will follow on top. Daniel Mack (7): ASoC: fold pxa2xx-pcm into its only user, pxa2xx-ac97 ASoC: pxa: clean up function names in pxa2xx-lib ASoC: pxa: move some functions to pxa2xx-lib ASoC: pxa2xx-pcm-lib: fix indenting ASoC: pxa: provide PCM ops for ssp, i2s and ac97 components ASoC: pxa: remove bindings from pxa2xx-pcm ASoC: pxa: make SND_PXA2XX_SOC_I2S selectable .../bindings/sound/mrvl,pxa2xx-pcm.txt | 15 -- include/sound/pxa2xx-lib.h | 13 +- sound/arm/Kconfig | 5 - sound/arm/Makefile | 3 - sound/arm/pxa2xx-ac97.c | 114 ++++++++++------ sound/arm/pxa2xx-pcm-lib.c | 69 ++++++++-- sound/arm/pxa2xx-pcm.c | 129 ------------------ sound/arm/pxa2xx-pcm.h | 27 ---- sound/soc/pxa/Kconfig | 2 +- sound/soc/pxa/pxa-ssp.c | 4 +- sound/soc/pxa/pxa2xx-ac97.c | 3 + sound/soc/pxa/pxa2xx-i2s.c | 3 + sound/soc/pxa/pxa2xx-pcm.c | 73 ---------- 13 files changed, 148 insertions(+), 312 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt delete mode 100644 sound/arm/pxa2xx-pcm.c delete mode 100644 sound/arm/pxa2xx-pcm.h -- 2.17.1

2 13

[alsa-devel] Applied "ASoC: pxa: provide PCM ops for ssp, i2s and ac97 components" to the asoc tree
by Mark Brown 29 Jun '18

29 Jun '18

The patch ASoC: pxa: provide PCM ops for ssp, i2s and ac97 components has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d767d3ce5c48b3378e20e8cfd5d5379c4ca6001b Mon Sep 17 00:00:00 2001 From: Daniel Mack <daniel(a)zonque.org> Date: Wed, 27 Jun 2018 21:33:57 +0200 Subject: [PATCH] ASoC: pxa: provide PCM ops for ssp, i2s and ac97 components Now that the functions are now available through pxa2xx-lib, hook them up to pxa-sspi, pxa-ac97 and pxa-i2s. This allows DT platforms to use the DAIs without a platform driver. Signed-off-by: Daniel Mack <daniel(a)zonque.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/pxa/pxa-ssp.c | 3 +++ sound/soc/pxa/pxa2xx-ac97.c | 3 +++ sound/soc/pxa/pxa2xx-i2s.c | 3 +++ 3 files changed, 9 insertions(+) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index c1f4af869289..01d54697ede4 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -841,6 +841,9 @@ static struct snd_soc_dai_driver pxa_ssp_dai = { static const struct snd_soc_component_driver pxa_ssp_component = { .name = "pxa-ssp", + .ops = &pxa2xx_pcm_ops, + .pcm_new = pxa2xx_soc_pcm_new, + .pcm_free = pxa2xx_pcm_free_dma_buffers, }; #ifdef CONFIG_OF diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index c52b33802bf2..9f779657bc86 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -214,6 +214,9 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { static const struct snd_soc_component_driver pxa_ac97_component = { .name = "pxa-ac97", + .ops = &pxa2xx_pcm_ops, + .pcm_new = pxa2xx_soc_pcm_new, + .pcm_free = pxa2xx_pcm_free_dma_buffers, }; #ifdef CONFIG_OF diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index e7184de0de04..42820121e5b9 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -364,6 +364,9 @@ static struct snd_soc_dai_driver pxa_i2s_dai = { static const struct snd_soc_component_driver pxa_i2s_component = { .name = "pxa-i2s", + .ops = &pxa2xx_pcm_ops, + .pcm_new = pxa2xx_soc_pcm_new, + .pcm_free = pxa2xx_pcm_free_dma_buffers, }; static int pxa2xx_i2s_drv_probe(struct platform_device *pdev) -- 2.18.0.rc2

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Re: [alsa-devel] [PATCH] vgaswitchroo: set audio client id according to bound gpu client id
by Lukas Wunner 29 Jun '18

29 Jun '18

> On Thu, Jun 28, 2018 at 2:22 AM, Jim Qu <Jim.Qu(a)amd.com> wrote: > > On modern laptop, there are more and more platforms > > have two GPUs, and each of them maybe have audio codec > > for HDMP/DP output. For some dGPU which is no output, > > audio codec usually is disabled. > > > > In currect HDA audio driver, it will set all codec as > > VGA_SWITCHEROO_DIS, so if system runtime pm is enabled, > > the audio which is binded to UMA will be suspended. Please explain what you mean by UMA here. Are you referring to dual GPU systems where both GPUs are by AMD? And the integrated GPU is incorrectly assigned VGA_SWITCHEROO_DIS? Starting with v4.17, the only reason the audio driver registers with vga_switcheroo is because it can be powered down if manual power control is used. However manual power control is not the default, it's primary use case nowadays is MacBook Pros, and there are no MacBook Pros with dual AMD GPUs. So I don't understand what real world use case you're trying to fix. Please explain. > > Newer dGPUs use PCI_CLASS_DISPLAY_OTHER rather than > > PCI_CLASS_DISPLAY_VGA. Make sure we check for both when > > checking for the dGPU in get_bound_vga() Unlike the other changes in your patch, this one would seem perfectly valid to me. However it's a separate issue, so please put it in a separate patch. Instead of adding a check for PCI_CLASS_DISPLAY_OTHER, I'd prefer you replacing the existing check for PCI_CLASS_DISPLAY_VGA with a check for PCI_BASE_CLASS_DISPLAY, like so: - if ((p->class >> 8) == PCI_CLASS_DISPLAY_VGA) + if ((p->class >> 16) == PCI_BASE_CLASS_DISPLAY) Thanks, Lukas

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[alsa-devel] Driver for Maxim MAX9744 Stereo Amplifier
by Tom Rinehart 29 Jun '18

29 Jun '18

I am trying to develop a driver the the Maxim MAX9744 20W Class D Stereo Amplifier. (see https://www.maximintegrated.com/en/products/analog/audio/MAX9744.html <https://www.maximintegrated.com/en/products/analog/audio/MAX9744.html>). It is very similar to the MAX9768 10W Class D Mono amplifier which has a linux driver. (see https://www.maximintegrated.com/en/products/analog/audio/MAX9768.html <https://www.maximintegrated.com/en/products/analog/audio/MAX9768.html>) From a control perspective, both look to be the same from their data sheets. Volume is controlled through i2c, and it has mute and shutdown controlled through gpio. The MAX9744 has a volume control that changes both channels together. When I try to use the MAX9768 driver it shows up as a mono amplifier in alsamixer. What do I need to change in the driver to have it recognize stereo input and output but with the volume control tied together (not independent for each channel)? It seems like I should be able to modify the MAX9768 driver to work with the MAX9744. I have an Adafruit board with the MAX9744 amplifier on it so it is easy to test: https://www.adafruit.com/product/1752 <https://www.adafruit.com/product/1752>.

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[alsa-devel] [PATCH] ASoC: pxa: remove the dmaengine compat need
by Daniel Mack 28 Jun '18

28 Jun '18

From: Robert Jarzmik <robert.jarzmik(a)free.fr> As the pxa architecture switched towards the dmaengine slave map, the old compatibility mechanism to acquire the dma requestor line number and priority are not needed anymore. This patch simplifies the dma resource acquisition, using the more generic function dma_request_slave_channel(). Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> Reviewed-by: Daniel Mack <daniel(a)zonque.org> Acked-by: Mark Brown <broonie(a)kernel.org> --- Mark, That's the patch that's Robert says is missing in your tree. As a matter of fact, that's the only reason why I pinged you and asked you to pull it in, because the cleanups for ASoC/pxa I've sent build upon it. Not sure what happened and caused this confusion :-/ Sorry. Thanks, Daniel sound/arm/pxa2xx-ac97.c | 14 ++------------ sound/arm/pxa2xx-pcm-lib.c | 6 +++--- sound/soc/pxa/pxa2xx-ac97.c | 32 +++++--------------------------- sound/soc/pxa/pxa2xx-i2s.c | 6 ++---- 4 files changed, 12 insertions(+), 46 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 4bc244c40f80..236a63cdaf9f 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -63,28 +63,18 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_legacy_reset, }; -static struct pxad_param pxa2xx_ac97_pcm_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 12, -}; - static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_out", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_out_req, -}; - -static struct pxad_param pxa2xx_ac97_pcm_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 11, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_in", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_in_req, }; static struct snd_pcm *pxa2xx_ac97_pcm; diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index e8da3b8ee721..dcbe7ecc1835 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -125,9 +125,9 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) return ret; - return snd_dmaengine_pcm_open_request_chan(substream, - pxad_filter_fn, - dma_params->filter_data); + return snd_dmaengine_pcm_open( + substream, dma_request_slave_channel(rtd->cpu_dai->dev, + dma_params->chan_name)); } EXPORT_SYMBOL(__pxa2xx_pcm_open); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 803818aabee9..1b41c0f2a8fb 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -68,61 +68,39 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 11, -}; - static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_in", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, -}; - -static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 12, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_out", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_stereo_out_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 10, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_out", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 9, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_in", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 8, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .addr = __PREG(MCDR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mic_mono", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req, }; static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream, diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 3fb60baf6eab..e7184de0de04 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -82,20 +82,18 @@ static struct pxa_i2s_port pxa_i2s; static struct clk *clk_i2s; static int clk_ena = 0; -static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = { .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "tx", .maxburst = 32, - .filter_data = &pxa2xx_i2s_pcm_stereo_out_req, }; -static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = { .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "rx", .maxburst = 32, - .filter_data = &pxa2xx_i2s_pcm_stereo_in_req, }; static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, -- 2.17.1

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[alsa-devel] [PATCH v2 0/2] Add support for audio on SDM845 SoC
by Rohit kumar 28 Jun '18

28 Jun '18

This provides initial patchset to support audio on Qualcomm Techonologies Inc. SDM845 SoC. Currently, it supports audio playback/capture over Primary MI2S and Quaternary TDM ports. Changes since v1: Addressed comments posted by Vinod and Srinivas. Rohit kumar (2): ASoC: qcom: dt-bindings: Add sdm845 machine bindings ASoC: qcom: add sdm845 sound card support .../devicetree/bindings/sound/qcom,sdm845.txt | 82 ++++ sound/soc/qcom/Kconfig | 9 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sdm845.c | 514 +++++++++++++++++++++ 4 files changed, 607 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sdm845.txt create mode 100644 sound/soc/qcom/sdm845.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[alsa-devel] [PATCH] ASoC: dpcm: extend channel merging to the backend cpu dai
by Jerome Brunet 28 Jun '18

28 Jun '18

Extend dpcm_merge_chan to also check backend cpu dai channels capabilities. Apply the same policy as soc_pcm_init_runtime_hw() for multicodec links and only check cpu dai in this case. Cc: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- Hi Mark, As you've probably noticed, this patch applies on top Jiada's patch pf4c277b817cc ("ASoC: soc-pcm: DPCM cares BE channel constraint") Regards. Jerome sound/soc/soc-pcm.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b82d9ff1f47e..5141409ccaed 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1728,12 +1728,26 @@ static void dpcm_runtime_base_chan(struct snd_pcm_substream *substream, list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; struct snd_soc_dai_driver *codec_dai_drv; struct snd_soc_pcm_stream *codec_stream; - int i; + struct snd_soc_pcm_stream *cpu_stream; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_stream = &cpu_dai_drv->playback; + else + cpu_stream = &cpu_dai_drv->capture; + + *channels_min = max(*channels_min, cpu_stream->channels_min); + *channels_max = min(*channels_max, cpu_stream->channels_max); + + /* + * chan min/max cannot be enforced if there are multiple CODEC + * DAIs connected to a single CPU DAI, use CPU DAI's directly + */ + if (be->num_codecs == 1) { + codec_dai_drv = be->codec_dais[0]->driver; - for (i = 0; i < be->num_codecs; i++) { - codec_dai_drv = be->codec_dais[i]->driver; if (stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; else -- 2.14.4

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Re: [alsa-devel] [PATCH] vgaswitchroo: set audio client id according to bound gpu client id
by Alex Deucher 28 Jun '18

28 Jun '18

On Thu, Jun 28, 2018 at 2:22 AM, Jim Qu <Jim.Qu(a)amd.com> wrote: > On modern laptop, there are more and more platforms > have two GPUs, and each of them maybe have audio codec > for HDMP/DP output. For some dGPU which is no output, > audio codec usually is disabled. > > In currect HDA audio driver, it will set all codec as > VGA_SWITCHEROO_DIS, so if system runtime pm is enabled, > the audio which is binded to UMA will be suspended. > > In HDA driver side, it is difficult to know which GPU > the audio has binded to. So set the bound gpu pci dev > to vgaswitchroo, let vgaswitchroo make decision. > > Newer dGPUs use PCI_CLASS_DISPLAY_OTHER rather than > PCI_CLASS_DISPLAY_VGA. Make sure we check for both when > checking for the dGPU in get_bound_vga() > > The patch also combine the HDA change to avoid break > building. > > Change-Id: I9906c1bd4dd5b36108d7133bb1cf724d13f1cd6d > Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> > Signed-off-by: Jim Qu <Jim.Qu(a)amd.com> > --- > drivers/gpu/vga/vga_switcheroo.c | 11 +++++++++-- > include/linux/vga_switcheroo.h | 4 ++-- > sound/pci/hda/hda_intel.c | 15 +++++++++------ > 3 files changed, 20 insertions(+), 10 deletions(-) > Please resend and send it to dri-devel and alsa-devel(a)alsa-project.org since this patch touches multiple subsystems. I would propose we merge this through the drm tree if possible. Alex > diff --git a/drivers/gpu/vga/vga_switcheroo.c b/drivers/gpu/vga/vga_switcheroo.c > index fc4adf3..7b7b0fd 100644 > --- a/drivers/gpu/vga/vga_switcheroo.c > +++ b/drivers/gpu/vga/vga_switcheroo.c > @@ -329,7 +329,7 @@ EXPORT_SYMBOL(vga_switcheroo_register_client); > * vga_switcheroo_register_audio_client - register audio client > * @pdev: client pci device > * @ops: client callbacks > - * @id: client identifier > + * @bound_vga: client bound vga pci device > * > * Register audio client (audio device on a GPU). The client is assumed > * to use runtime PM. Beforehand, vga_switcheroo_client_probe_defer() > @@ -339,8 +339,15 @@ EXPORT_SYMBOL(vga_switcheroo_register_client); > */ > int vga_switcheroo_register_audio_client(struct pci_dev *pdev, > const struct vga_switcheroo_client_ops *ops, > - enum vga_switcheroo_client_id id) > + struct pci_dev *bound_vga) > { > + enum vga_switcheroo_client_id id = VGA_SWITCHEROO_DIS; > + > + if (bound_vga) { > + if (vgasr_priv.handler->get_client_id) > + id = vgasr_priv.handler->get_client_id(bound_vga); > + } > + > return register_client(pdev, ops, id | ID_BIT_AUDIO, false, true); > } > EXPORT_SYMBOL(vga_switcheroo_register_audio_client); > diff --git a/include/linux/vga_switcheroo.h b/include/linux/vga_switcheroo.h > index 77f0f0a..a0c7b86 100644 > --- a/include/linux/vga_switcheroo.h > +++ b/include/linux/vga_switcheroo.h > @@ -151,7 +151,7 @@ int vga_switcheroo_register_client(struct pci_dev *dev, > bool driver_power_control); > int vga_switcheroo_register_audio_client(struct pci_dev *pdev, > const struct vga_switcheroo_client_ops *ops, > - enum vga_switcheroo_client_id id); > + struct pci_dev *bound_vga); > > void vga_switcheroo_client_fb_set(struct pci_dev *dev, > struct fb_info *info); > @@ -180,7 +180,7 @@ static inline int vga_switcheroo_register_handler(const struct vga_switcheroo_ha > enum vga_switcheroo_handler_flags_t handler_flags) { return 0; } > static inline int vga_switcheroo_register_audio_client(struct pci_dev *pdev, > const struct vga_switcheroo_client_ops *ops, > - enum vga_switcheroo_client_id id) { return 0; } > + struct pci_dev *bound_vga) { return 0; } > static inline void vga_switcheroo_unregister_handler(void) {} > static inline enum vga_switcheroo_handler_flags_t vga_switcheroo_handler_flags(void) { return 0; } > static inline int vga_switcheroo_lock_ddc(struct pci_dev *pdev) { return -ENODEV; } > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index 1ae1850..8f992e6 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -1319,15 +1319,17 @@ static const struct vga_switcheroo_client_ops azx_vs_ops = { > static int register_vga_switcheroo(struct azx *chip) > { > struct hda_intel *hda = container_of(chip, struct hda_intel, chip); > + struct pci_dev *p; > int err; > > if (!hda->use_vga_switcheroo) > return 0; > - /* FIXME: currently only handling DIS controller > - * is there any machine with two switchable HDMI audio controllers? > - */ > - err = vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, > - VGA_SWITCHEROO_DIS); > + > + p = get_bound_vga(chip->pci); > + err = vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, p); > + > + if (p) > + pci_dev_put(p); > if (err < 0) > return err; > hda->vga_switcheroo_registered = 1; > @@ -1429,7 +1431,8 @@ static struct pci_dev *get_bound_vga(struct pci_dev *pci) > p = pci_get_domain_bus_and_slot(pci_domain_nr(pci->bus), > pci->bus->number, 0); > if (p) { > - if ((p->class >> 8) == PCI_CLASS_DISPLAY_VGA) > + if ((p->class >> 8) == PCI_CLASS_DISPLAY_VGA || > + (p->class >> 8) == PCI_CLASS_DISPLAY_OTHER) > return p; > pci_dev_put(p); > } > -- > 2.7.4 > > _______________________________________________ > amd-gfx mailing list > amd-gfx(a)lists.freedesktop.org > https://lists.freedesktop.org/mailman/listinfo/amd-gfx

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[alsa-devel] [PATCH v3 00/14] ARM: pxa: switch to DMA slave maps
by Robert Jarzmik 28 Jun '18

28 Jun '18

As I gathered almost all the required acks, this is an information only post before queuing to the PXA tree. The only missing part is for netdev, for which I'd like an ack from netdev people for patches 0007 and 0008, but that won't prevent me from queuing all the other patches (excepting the patch 0012 which can only be applied once 0007 and 0008 are queued). Cheers. -- Robert Robert Jarzmik (14): dmaengine: pxa: use a dma slave map ARM: pxa: add dma slave map dmaengine: pxa: add a default requestor policy mmc: pxamci: remove the dmaengine compat need media: pxa_camera: remove the dmaengine compat need mtd: rawnand: marvell: remove the dmaengine compat need net: smc911x: remove the dmaengine compat need net: smc91x: remove the dmaengine compat need ASoC: pxa: remove the dmaengine compat need ata: pata_pxa: remove the dmaengine compat need dmaengine: pxa: document pxad_param dmaengine: pxa: make the filter function internal ARM: pxa: remove the DMA IO resources ARM: pxa: change SSP DMA channels allocation arch/arm/mach-pxa/devices.c | 148 +--------------------------------- arch/arm/mach-pxa/devices.h | 6 +- arch/arm/mach-pxa/pxa25x.c | 38 ++++++++- arch/arm/mach-pxa/pxa27x.c | 39 ++++++++- arch/arm/mach-pxa/pxa3xx.c | 41 +++++++++- arch/arm/plat-pxa/ssp.c | 47 ----------- drivers/ata/pata_pxa.c | 10 +-- drivers/dma/pxa_dma.c | 18 ++++- drivers/media/platform/pxa_camera.c | 22 +---- drivers/mmc/host/pxamci.c | 29 +------ drivers/mtd/nand/raw/marvell_nand.c | 17 +--- drivers/net/ethernet/smsc/smc911x.c | 13 +-- drivers/net/ethernet/smsc/smc91x.c | 9 +-- drivers/net/ethernet/smsc/smc91x.h | 1 - include/linux/dma/pxa-dma.h | 20 +++-- include/linux/platform_data/mmp_dma.h | 4 + include/linux/pxa2xx_ssp.h | 2 - sound/arm/pxa2xx-ac97.c | 14 +--- sound/arm/pxa2xx-pcm-lib.c | 6 +- sound/soc/pxa/pxa-ssp.c | 5 +- sound/soc/pxa/pxa2xx-ac97.c | 32 ++------ sound/soc/pxa/pxa2xx-i2s.c | 6 +- 22 files changed, 176 insertions(+), 351 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH v2] ASoC: dpcm: improve runtime update predictability
by Jerome Brunet 28 Jun '18

28 Jun '18

As it is, dpcm_runtime_update() performs the old path and new path update of a frontend before going on to the next frontend DAI. Depending the order of the FEs within the rtd list, the result of the update might be different. For example: * Frontend A connected to backend C, with a 48kHz playback * Frontend B connected to backend D, with a 44.1kHz playback * FE A appears before FE B in the rtd list of the card. If we reparent BE C to FE B (disconnecting BE D): * old path update of FE A will run first, and BE C will get hw_free() and shutdown() * new path update of FE B will run after and BE C, which is stopped, so it will be configured at 44.1kHz, as expected If we reparent BE D to FE A (disconnecting BE C): * new path update of FE A will run first but since BE D is still running at 44.1kHz, it won't be reconfigured (no call to startup() or hw_params()) * old path update of FE B runs after, nothing happens * In this case, we end up with a BE playing at 44.1kHz a stream which is supposed to be played at 48Khz (too slow) To improve this situation, this patch performs all the FE old paths update before going on to update the new paths. With this, the result should no longer depend on the order of the FE within the card rtd list. Please note that there might be a small performance penalty since dpcm_process_paths() is called twice per stream direction. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- Changes since v1 [0]: * Add missing static as reported by kbuild robot. [0]: https://lkml.kernel.org/r/20180625082923.25636-1-jbrunet@baylibre.com sound/soc/soc-pcm.c | 165 +++++++++++++++++++++++++++------------------------- 1 file changed, 86 insertions(+), 79 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5e7ae47a9658..4eda18cfe895 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2543,106 +2543,113 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) return ret; } -/* Called by DAPM mixer/mux changes to update audio routing between PCMs and - * any DAI links. - */ -int soc_dpcm_runtime_update(struct snd_soc_card *card) +static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) { - struct snd_soc_pcm_runtime *fe; - int old, new, paths; + struct snd_soc_dapm_widget_list *list; + int count, paths; - mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - list_for_each_entry(fe, &card->rtd_list, list) { - struct snd_soc_dapm_widget_list *list; + if (!fe->dai_link->dynamic) + return 0; - /* make sure link is FE */ - if (!fe->dai_link->dynamic) - continue; + /* only check active links */ + if (!fe->cpu_dai->active) + return 0; - /* only check active links */ - if (!fe->cpu_dai->active) - continue; + /* DAPM sync will call this to update DSP paths */ + dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n", + new ? "new" : "old", fe->dai_link->name); - /* DAPM sync will call this to update DSP paths */ - dev_dbg(fe->dev, "ASoC: DPCM runtime update for FE %s\n", - fe->dai_link->name); + /* skip if FE doesn't have playback capability */ + if (!fe->cpu_dai->driver->playback.channels_min || + !fe->codec_dai->driver->playback.channels_min) + goto capture; - /* skip if FE doesn't have playback capability */ - if (!fe->cpu_dai->driver->playback.channels_min - || !fe->codec_dai->driver->playback.channels_min) - goto capture; - - /* skip if FE isn't currently playing */ - if (!fe->cpu_dai->playback_active - || !fe->codec_dai->playback_active) - goto capture; - - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "playback"); - mutex_unlock(&card->mutex); - return paths; - } + /* skip if FE isn't currently playing */ + if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active) + goto capture; - /* update any new playback paths */ - new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 1); - if (new) { - dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, "playback"); + return paths; + } - /* update any old playback paths */ - old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 0); - if (old) { + /* update any playback paths */ + count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new); + if (count) { + if (new) + dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); + else dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } - dpcm_path_put(&list); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); + } + + dpcm_path_put(&list); + capture: - /* skip if FE doesn't have capture capability */ - if (!fe->cpu_dai->driver->capture.channels_min - || !fe->codec_dai->driver->capture.channels_min) - continue; + /* skip if FE doesn't have capture capability */ + if (!fe->cpu_dai->driver->capture.channels_min || + !fe->codec_dai->driver->capture.channels_min) + return 0; - /* skip if FE isn't currently capturing */ - if (!fe->cpu_dai->capture_active - || !fe->codec_dai->capture_active) - continue; + /* skip if FE isn't currently capturing */ + if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active) + return 0; - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "capture"); - mutex_unlock(&card->mutex); - return paths; - } + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, "capture"); + return paths; + } - /* update any new capture paths */ - new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 1); - if (new) { + /* update any old capture paths */ + count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new); + if (count) { + if (new) dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); - } - - /* update any old capture paths */ - old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 0); - if (old) { + else dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); - } - dpcm_path_put(&list); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); } - mutex_unlock(&card->mutex); + dpcm_path_put(&list); + return 0; } + +/* Called by DAPM mixer/mux changes to update audio routing between PCMs and + * any DAI links. + */ +int soc_dpcm_runtime_update(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *fe; + int ret = 0; + + mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + /* shutdown all old paths first */ + list_for_each_entry(fe, &card->rtd_list, list) { + ret = soc_dpcm_fe_runtime_update(fe, 0); + if (ret) + goto out; + } + + /* bring new paths up */ + list_for_each_entry(fe, &card->rtd_list, list) { + ret = soc_dpcm_fe_runtime_update(fe, 1); + if (ret) + goto out; + } + +out: + mutex_unlock(&card->mutex); + return ret; +} int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) { struct snd_soc_dpcm *dpcm; -- 2.14.4

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[alsa-devel] [PATCH] ASoC: qdsp6: qdafe: fix some off by one bugs
by Dan Carpenter 28 Jun '18

28 Jun '18

The > should be >= or we could read one element beyond the end of the port_maps[] array. Fixes: 7fa2d70f9766 ("ASoC: qdsp6: q6afe: Add q6afe driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 01f43218984b..69a7896cb713 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -777,7 +777,7 @@ static int q6afe_callback(struct apr_device *adev, struct apr_resp_pkt *data) */ int q6afe_get_port_id(int index) { - if (index < 0 || index > AFE_PORT_MAX) + if (index < 0 || index >= AFE_PORT_MAX) return -EINVAL; return port_maps[index].port_id; @@ -1014,7 +1014,7 @@ int q6afe_port_stop(struct q6afe_port *port) port_id = port->id; index = port->token; - if (index < 0 || index > AFE_PORT_MAX) { + if (index < 0 || index >= AFE_PORT_MAX) { dev_err(afe->dev, "AFE port index[%d] invalid!\n", index); return -EINVAL; } @@ -1355,7 +1355,7 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) unsigned long flags; int cfg_type; - if (id < 0 || id > AFE_PORT_MAX) { + if (id < 0 || id >= AFE_PORT_MAX) { dev_err(dev, "AFE port token[%d] invalid!\n", id); return ERR_PTR(-EINVAL); }

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[alsa-devel] [PATCH] ASoC: dpcm: don't merge format from invalid codec dai
by Jerome Brunet 28 Jun '18

28 Jun '18

When merging codec formats, dpcm_runtime_base_format() should skip the codecs which are not supporting the current stream direction. At the moment, if a BE link has more than one codec, and only one of these codecs has no capture DAI, it becomes impossible to start a capture stream because the merged format would be 0. Skipping invalid codec DAI solves the problem. Fixes: b073ed4e2126 ("ASoC: soc-pcm: DPCM cares BE format") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/soc-pcm.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5141409ccaed..a58d8e7bf52b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1694,6 +1694,14 @@ static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) int i; for (i = 0; i < be->num_codecs; i++) { + /* + * Skip CODECs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(be->codec_dais[i], + stream)) + continue; + codec_dai_drv = be->codec_dais[i]->driver; if (stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; -- 2.14.4

2 1

[alsa-devel] [PATCH 0/3] ALSA: PM-related fixes/cleanups
by Takashi Iwai 28 Jun '18

28 Jun '18

Hi, this is a patch set addressing some minor issues found in PM-related stuff, as well as the Intel HDMI fallback issue. The first patch covers the unusual error paths from runtime PM, the second one is a code refactoring and the last one fixes the superfluous (rather harmful) binding with the generic HDMI driver as fallback of i915 component binding. Takashi === Takashi Iwai (3): ALSA: hda - Handle error from snd_hda_power_up*() ALSA: hda - Move in_pm accessors to HDA core ALSA: hda/hdmi - Don't fall back to generic when i915 binding fails include/sound/hdaudio.h | 27 +++++++++++++++++ sound/pci/hda/hda_beep.c | 3 +- sound/pci/hda/hda_codec.c | 25 ++++++---------- sound/pci/hda/hda_controller.c | 4 ++- sound/pci/hda/hda_proc.c | 3 +- sound/pci/hda/hda_sysfs.c | 4 ++- sound/pci/hda/patch_ca0132.c | 53 +++++++++++++++++++++++++++------- sound/pci/hda/patch_hdmi.c | 4 ++- sound/pci/hda/patch_realtek.c | 3 +- 9 files changed, 93 insertions(+), 33 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH] ALSA: hda: Add AZX_DCAPS_PM_RUNTIME for AMD Raven Ridge
by Kai-Heng Feng 28 Jun '18

28 Jun '18

This patch can make audio controller in AMD Raven Ridge gets runtime suspended to D3, to save ~1W power when it's not in use. Cc: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1ae1850b3bfd..a9b55d65f2bd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2535,7 +2535,8 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, /* AMD Raven */ { PCI_DEVICE(0x1022, 0x15e3), - .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | + AZX_DCAPS_PM_RUNTIME }, /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x0002), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, -- 2.17.1

2 1

[alsa-devel] dsd formats in ALSA plugins
by slava 28 Jun '18

28 Jun '18

Sorry, don't receive my own post, so can't reply Find in alsa-lib-1.1.6 file src/pcm/pcm-params.c line 1945 static const snd_mask_t refine_masks[SND_PCM_HW_PARAM_LAST_MASK - SND_PCM_HW_PARAM_FIRST_MASK + 1] = { [SND_PCM_HW_PARAM_ACCESS - SND_PCM_HW_PARAM_FIRST_MASK] = { .bits = { 0x1f }, }, [SND_PCM_HW_PARAM_FORMAT - SND_PCM_HW_PARAM_FIRST_MASK] = { .bits = { 0x81ffffff, 0xfff}, Change last 0xfff to 0xfffff and now can use DSD formats How i understand, in version 1.0.27 last format = 43 and when new format was added but mask not changed May be i absolutely wrong and what i do has side effects or needed any else? Thanks

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