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[alsa-devel] [RFC PATCH v8 0/4] Media Device Allocator API
by shuah＠kernel.org 09 Dec '18

09 Dec '18

From: Shuah Khan <shuah(a)kernel.org> This patch series has been on the ice for the last couple of years. Mauro asked me to restart the discussion on this and see if we can make progress. I rebased the series to Linux 4.19. It has been surprisingly well preserved with one small merge conflict resolved in 0004. Media Device Allocator API to allows multiple drivers share a media device. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. - No changes to 0001,0002 code since the v7 referenced below. - 0003 is a new patch to enable ALSA defines that have been disabled for kernel between 4.9 and 4.19. - Minor merge conflict resolution in 0004. - Added SPDX to new files. References: https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Please review. I am sending this as RFC even though it has been tested several times prior to this non-event rebased v8. I ran sanity tests. Shuah Khan (4): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources Documentation/media/kapi/mc-core.rst | 37 +++ drivers/media/Makefile | 3 +- drivers/media/media-dev-allocator.c | 132 ++++++++++ drivers/media/usb/au0828/au0828-core.c | 12 +- drivers/media/usb/au0828/au0828.h | 1 + include/media/media-dev-allocator.h | 53 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 320 +++++++++++++++++++++++++ sound/usb/media.h | 73 ++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 18 files changed, 692 insertions(+), 28 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.17.0

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[alsa-devel] [PATCH] ALSA: fireface: fix reference to wrong register for clock configuration
by Takashi Sakamoto 09 Dec '18

09 Dec '18

In an initial commit, 'SYNC_STATUS' register is referred to get clock configuration, however this is wrong, according to my local note at hand for reverse-engineering about packet dump. It should be 'CLOCK_CONFIG' register. Actually, ff400_dump_clock_config() is correctly programmed. This commit fixes the bug. Cc: <stable(a)vger.kernel.org> # v4.12+ Fixes: 76fdb3a9e13a ('ALSA: fireface: add support for Fireface 400') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireface/ff-protocol-ff400.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/fireface/ff-protocol-ff400.c b/sound/firewire/fireface/ff-protocol-ff400.c index 64c3cb0fb926..654a50319198 100644 --- a/sound/firewire/fireface/ff-protocol-ff400.c +++ b/sound/firewire/fireface/ff-protocol-ff400.c @@ -30,7 +30,7 @@ static int ff400_get_clock(struct snd_ff *ff, unsigned int *rate, int err; err = snd_fw_transaction(ff->unit, TCODE_READ_QUADLET_REQUEST, - FF400_SYNC_STATUS, &reg, sizeof(reg), 0); + FF400_CLOCK_CONFIG, &reg, sizeof(reg), 0); if (err < 0) return err; data = le32_to_cpu(reg); -- 2.19.1

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix the mute LED regresion on Lenovo X1 Carbon
by Hui Wang 09 Dec '18

09 Dec '18

Users reported a mute LED regression on Lenovo X1 Carbon, the root cause is we applied the fixup of ALC285_FIXUP_LENOVO_HEADPHONE_NOISE to this machine, then the machine can't apply the fixup of ALC269_FIXUP_THINKPAD_ACPI anymore. To fix it, we chain two fixup together. Fixes: c4cfcf6f4297 ("ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops") Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 56882d9b1e43..5b13d9e8d6c7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6450,6 +6450,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC285_FIXUP_LENOVO_HEADPHONE_NOISE] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_invalidate_dacs, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI }, [ALC295_FIXUP_HP_AUTO_MUTE] = { .type = HDA_FIXUP_FUNC, -- 2.17.1

2 1

[alsa-devel] [PATCH 0/6] Introduce audio-mute LED trigger (and conversions to it)
by Takashi Iwai 08 Dec '18

08 Dec '18

Hi, this is patch series I've hacked after useful conversation with Pavel. Basically this adds a new LED trigger audio-mute and audio-micmute, and convert the HD-audio driver and the platform drivers to use the LED trigger instead of the ugly direct dynamic symbol binding. The latest version of patches are found in topic/leds-trigger branch in my sound git tree. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git As these are cross-tree patches, the branch above is based cleanly on v4.20-rc3, so that it can be merged well to multiple trees. Once after getting the ACK's, I'll add tags and fixate for merges. This patch series don't include huawei-wmi stuff; so Huawei patches need rework. I already have some piece of changes for huawei-wmi, so please ping me if needed. I checked briefly on my Dell laptop, and a Thinkpad model. Wider tests are appreciated, of course. thanks, Takashi === Takashi Iwai (6): leds: trigger: Introduce audio mute LED trigger platform/x86: dell-laptop: Add micmute LED trigger support platform/x86: thinkpad_acpi: Add audio mute LED classdev support ALSA: hda - Support led audio trigger platform/x86: dell-laptop: Drop superfluous exported function platform/x86: thinkpad_acpi: Drop superfluous exported function drivers/leds/trigger/Kconfig | 7 +++ drivers/leds/trigger/Makefile | 1 + drivers/leds/trigger/ledtrig-audio.c | 45 +++++++++++++++++++ drivers/platform/x86/Kconfig | 4 ++ drivers/platform/x86/dell-laptop.c | 27 +++++++++--- drivers/platform/x86/thinkpad_acpi.c | 66 +++++++++++++++++++++------- include/linux/dell-led.h | 7 --- include/linux/leds.h | 20 +++++++++ include/linux/thinkpad_acpi.h | 16 ------- sound/pci/hda/dell_wmi_helper.c | 48 -------------------- sound/pci/hda/hda_generic.c | 31 +++++++++++++ sound/pci/hda/hda_generic.h | 2 + sound/pci/hda/patch_realtek.c | 17 +++---- sound/pci/hda/thinkpad_helper.c | 43 +++--------------- 14 files changed, 194 insertions(+), 140 deletions(-) create mode 100644 drivers/leds/trigger/ledtrig-audio.c delete mode 100644 include/linux/dell-led.h delete mode 100644 include/linux/thinkpad_acpi.h delete mode 100644 sound/pci/hda/dell_wmi_helper.c -- 2.19.1

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Re: [alsa-devel] [PATCH v9 2/3] x86: add support for Huawei WMI hotkeys.
by ayman.bagabas＠gmail.com 08 Dec '18

08 Dec '18

On Fri, 2018-12-07 at 00:52 -0500, ayman.bagabas(a)gmail.com wrote: On Mon, 2018-12-03 at 21:17 +0200, Andy Shevchenko wrote: > On Mon, Dec 3, 2018 at 9:04 PM Takashi Iwai <tiwai(a)suse.de> wrote: > > On Mon, 03 Dec 2018 19:53:39 +0100, > > Ayman Bagabas wrote: > > > + if (code == 0x80) { > > > + acpi_status status; > > > + acpi_handle handle; > > > + unsigned long long result; > > > + union acpi_object args[1]; > > > + struct acpi_object_list arg_list = { > > > + .pointer = args, > > > + .count = ARRAY_SIZE(args), > > > + }; > > > + > > > + args[0].type = ACPI_TYPE_INTEGER; > > > + args[0].integer.value = 0; > > > + > > > + status = acpi_get_handle(NULL, "\\WMI0", &handle); > > > + if (ACPI_FAILURE(status)) { > > > + dev_err(&wdev->dev, "Unable to get ACPI > > > handle\n"); > > > + return; > > > + } > > > + > > > + status = acpi_evaluate_integer(NULL, "WQ00", > > > &arg_list, &result); > > > > I guess you need to pass handle here? In the earlier version, you > > passed \\WMI0.WQ00, so it worked with NULL handle. But now it's no > > longer so... > > I think in this case we don't need to have a separate call to get > handle and try to get integer directly. In either we will have an > error case if method / namespace / etc is not found. > > I was digging further into the DSDT table of the laptop that has WQ00 method and it turned out that this method is a ACPI_WMI_EXPENSIVE. It has its own GUID of "39142400-C6A3-40fa-BADB-8A2652834100". I believe wmi_query_block, which is deprecated BTW, can be used to access this method instead of acpi_evaluate. However, I don't have a machine to test it on. Any ideas?

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[alsa-devel] [PATCH - extplug without conversion 0/1] pcm: extplug: Use same format for slave
by twischer＠de.adit-jv.com 07 Dec '18

07 Dec '18

Hi Takashi, I have the following new idea for [PATCH - extplug refinement from hw_params 1/1] pcm: extplug: Update slave PCM if extplug slave params were changed May be you could give a short feedback. I will try to provide a reproduction scenario. Do you have an idea how to handle this additional function parameter without breaking the API. Best regards Timo

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[alsa-devel] [PATCH - extplug refinement from hw_params 1/1] pcm: extplug: Update slave PCM if extplug slave params were changed
by twischer＠de.adit-jv.com 07 Dec '18

07 Dec '18

From: Timo Wischer <twischer(a)de.adit-jv.com> from extplug hw_params() callback function. This feature is for example required in the following use case: - A filter plugin supports only 1-8 channels. Therefore it calls snd_pcm_extplug_set_slave_param_minmax(SND_PCM_EXTPLUG_HW_CHANNELS, 1, 8) to provide the salve PCM parameters limited to 1-8 channels to the user application - The user application requests 2 channels. But in this case the slave PCM will be configured with 1 channel because it is the first valid configuration - To update the salve PCM snd_pcm_extplug_set_slave_param_minmax(SND_PCM_EXTPLUG_HW_CHANNELS, 2, 2) could be called from the extplug hw_params() callback function Without this patch the call to snd_pcm_extplug_set_slave_param_minmax() would not have any effect. With this patch the slave device will also be configured with 2 channels and no down mixing has to be done by the extplug. This new feature can also be used for some specific dependencies. For example a down mixing extplug supports only 2to1 and 4to2 channel down mixing. Therefore the salve PCM has to be opened with 2 channels if the user application requests 4 channels. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_extplug.c b/src/pcm/pcm_extplug.c index 1f887c5..44afadb 100644 --- a/src/pcm/pcm_extplug.c +++ b/src/pcm/pcm_extplug.c @@ -26,6 +26,7 @@ * */ +#include <stdbool.h> #include "pcm_local.h" #include "pcm_plugin.h" #include "pcm_extplug.h" @@ -43,6 +44,7 @@ typedef struct snd_pcm_extplug_priv { snd_pcm_extplug_t *data; struct snd_ext_parm params[SND_PCM_EXTPLUG_HW_PARAMS]; struct snd_ext_parm sparams[SND_PCM_EXTPLUG_HW_PARAMS]; + bool sparams_changed; } extplug_priv_t; static const int hw_params_type[SND_PCM_EXTPLUG_HW_PARAMS] = { @@ -60,18 +62,49 @@ static const unsigned int excl_parbits[SND_PCM_EXTPLUG_HW_PARAMS] = { SND_PCM_HW_PARBIT_FRAME_BITS), }; +static bool snd_ext_parm_equal(const struct snd_ext_parm * const a, + const struct snd_ext_parm * const b) +{ + if (a != b || a->active != b->active || a->num_list != b->num_list) + return false; + + if (a->num_list > 0) { + for (unsigned int i = 0; i++; i < a->num_list) { + if (a->list[i] != b->list[i]) + return false; + } + } else { + if (a->min != b->min || a->max != b->max) + return false; + } + + return true; +} + +static int snd_ext_parm_set(struct snd_ext_parm *parm, unsigned int min, + unsigned int max, unsigned int num_list, + unsigned int *list) +{ + const struct snd_ext_parm new_parm = { + .min = min, + .max = max, + .num_list = num_list, + .list = list, + .active = 1, + }; + const int changed = snd_ext_parm_equal(parm, &new_parm) ? 0 : 1; + + free(parm->list); + *parm = new_parm; + return changed; +} + /* * set min/max values for the given parameter */ int snd_ext_parm_set_minmax(struct snd_ext_parm *parm, unsigned int min, unsigned int max) { - parm->num_list = 0; - free(parm->list); - parm->list = NULL; - parm->min = min; - parm->max = max; - parm->active = 1; - return 0; + return snd_ext_parm_set(parm, min, max, 0, NULL); } /* @@ -92,11 +125,7 @@ int snd_ext_parm_set_list(struct snd_ext_parm *parm, unsigned int num_list, cons memcpy(new_list, list, sizeof(*new_list) * num_list); qsort(new_list, num_list, sizeof(*new_list), val_compar); - free(parm->list); - parm->num_list = num_list; - parm->list = new_list; - parm->active = 1; - return 0; + return snd_ext_parm_set(parm, 0, 0, num_list, new_list); } void snd_ext_parm_clear(struct snd_ext_parm *parm) @@ -291,29 +320,37 @@ static int snd_pcm_extplug_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params */ static int snd_pcm_extplug_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) { - extplug_priv_t *ext = pcm->private_data; snd_pcm_t *slave = ext->plug.gen.slave; - int err = snd_pcm_hw_params_slave(pcm, params, - snd_pcm_extplug_hw_refine_cchange, - snd_pcm_extplug_hw_refine_sprepare, - snd_pcm_extplug_hw_refine_schange, - snd_pcm_generic_hw_params); - if (err < 0) - return err; - ext->data->slave_format = slave->format; - ext->data->slave_subformat = slave->subformat; - ext->data->slave_channels = slave->channels; - ext->data->rate = slave->rate; - INTERNAL(snd_pcm_hw_params_get_format)(params, &ext->data->format); - INTERNAL(snd_pcm_hw_params_get_subformat)(params, &ext->data->subformat); - INTERNAL(snd_pcm_hw_params_get_channels)(params, &ext->data->channels); - - if (ext->data->callback->hw_params) { - err = ext->data->callback->hw_params(ext->data, params); + + do { + ext->sparams_changed = false; + + int err = snd_pcm_hw_params_slave(pcm, params, + snd_pcm_extplug_hw_refine_cchange, + snd_pcm_extplug_hw_refine_sprepare, + snd_pcm_extplug_hw_refine_schange, + snd_pcm_generic_hw_params); if (err < 0) return err; - } + ext->data->slave_format = slave->format; + ext->data->slave_subformat = slave->subformat; + ext->data->slave_channels = slave->channels; + ext->data->rate = slave->rate; + INTERNAL(snd_pcm_hw_params_get_format)( + params, &ext->data->format); + INTERNAL(snd_pcm_hw_params_get_subformat)( + params, &ext->data->subformat); + INTERNAL(snd_pcm_hw_params_get_channels)( + params, &ext->data->channels); + + if (ext->data->callback->hw_params) { + err = ext->data->callback->hw_params(ext->data, params); + if (err < 0) + return err; + } + } while (ext->sparams_changed); + return 0; } @@ -411,6 +448,7 @@ static void clear_ext_params(extplug_priv_t *ext) int i; for (i = 0; i < SND_PCM_EXTPLUG_HW_PARAMS; i++) { snd_ext_parm_clear(&ext->params[i]); + ext->sparams_changed |= ext->sparams[i].active; snd_ext_parm_clear(&ext->sparams[i]); } } @@ -637,7 +675,9 @@ parameters are sample format and channels. To define the constraints of the slave PCM configuration, use either #snd_pcm_extplug_set_slave_param_minmax() and #snd_pcm_extplug_set_slave_param_list(). The arguments are as same -as former functions. +as former functions. Both functions can also be called from the hw_params +callback. This can be used to further limit the configuration of the slave +device depending on the configuration of the user device. To clear the parameter constraints, call #snd_pcm_extplug_params_reset() function. @@ -768,11 +808,17 @@ void snd_pcm_extplug_params_reset(snd_pcm_extplug_t *extplug) int snd_pcm_extplug_set_slave_param_list(snd_pcm_extplug_t *extplug, int type, unsigned int num_list, const unsigned int *list) { extplug_priv_t *ext = extplug->pcm->private_data; + int changed = 0; + if (type < 0 || type >= SND_PCM_EXTPLUG_HW_PARAMS) { SNDERR("EXTPLUG: invalid parameter type %d", type); return -EINVAL; } - return snd_ext_parm_set_list(&ext->sparams[type], num_list, list); + changed = snd_ext_parm_set_list(&ext->sparams[type], num_list, list); + if (changed > 0) + ext->sparams_changed = true; + + return changed; } /** @@ -790,6 +836,8 @@ int snd_pcm_extplug_set_slave_param_list(snd_pcm_extplug_t *extplug, int type, u int snd_pcm_extplug_set_slave_param_minmax(snd_pcm_extplug_t *extplug, int type, unsigned int min, unsigned int max) { extplug_priv_t *ext = extplug->pcm->private_data; + int changed = 0; + if (type < 0 || type >= SND_PCM_EXTPLUG_HW_PARAMS) { SNDERR("EXTPLUG: invalid parameter type %d", type); return -EINVAL; @@ -798,7 +846,11 @@ int snd_pcm_extplug_set_slave_param_minmax(snd_pcm_extplug_t *extplug, int type, SNDERR("EXTPLUG: invalid parameter type %d", type); return -EINVAL; } - return snd_ext_parm_set_minmax(&ext->sparams[type], min, max); + changed = snd_ext_parm_set_minmax(&ext->sparams[type], min, max); + if (changed > 0) + ext->sparams_changed = true; + + return changed; } /** -- 2.7.4

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[alsa-devel] Applied "ASoC: max9867: Remove useless assignment" to the asoc tree
by Mark Brown 07 Dec '18

07 Dec '18

The patch ASoC: max9867: Remove useless assignment 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 8efc1afd7e0570d9bb37f5127f0742304a8ad2fc Mon Sep 17 00:00:00 2001 From: Ladislav Michl <ladis(a)linux-mips.org> Date: Tue, 4 Dec 2018 19:18:17 +0100 Subject: [PATCH] ASoC: max9867: Remove useless assignment ret is assigned later, no need to initialize it. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max9867.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 1cda54b59854..1b155f40b106 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -476,7 +476,7 @@ static int max9867_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct max9867_priv *max9867; - int ret = 0, reg; + int ret, reg; max9867 = devm_kzalloc(&i2c->dev, sizeof(*max9867), GFP_KERNEL); if (!max9867) @@ -497,10 +497,8 @@ static int max9867_i2c_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "device revision: %x\n", reg); ret = devm_snd_soc_register_component(&i2c->dev, &max9867_component, max9867_dai, ARRAY_SIZE(max9867_dai)); - if (ret < 0) { + if (ret < 0) dev_err(&i2c->dev, "Failed to register component: %d\n", ret); - return ret; - } return ret; } -- 2.19.0.rc2

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[alsa-devel] [PATCH v2 0/5] max9867 driver update
by Ladislav Michl 07 Dec '18

07 Dec '18

Hi Mark, this is yet another round of driver update. Tested on AT91SAM9G20 based board with codec in master mode. All implemeted controls do work now. Comments from previous serie taken into account. Ladislav Michl (5): ASoC: max9867: Remove useless assignment ASoC: max9867: Fix power management ASoC: max9867: Calculate LRCLK divider ASoC: max9867: Fix signal paths ASoC: max9867: Add copyright and module author

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[alsa-devel] Applied "ASoC: max9867: Calculate LRCLK divider" to the asoc tree
by Mark Brown 07 Dec '18

07 Dec '18

The patch ASoC: max9867: Calculate LRCLK divider 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 715ee1916b95bfd0773487397a0354ad9a4660e6 Mon Sep 17 00:00:00 2001 From: Ladislav Michl <ladis(a)linux-mips.org> Date: Tue, 4 Dec 2018 19:21:04 +0100 Subject: [PATCH] ASoC: max9867: Calculate LRCLK divider Drop "Common NI Values Table" and calculate LRCLK divider, then add allowed rate constraints based on master clock frequency. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max9867.c | 207 ++++++++++++++++++------------------- sound/soc/codecs/max9867.h | 31 ++---- 2 files changed, 110 insertions(+), 128 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index c454de434d76..faec094161c1 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -126,124 +126,106 @@ static const struct snd_soc_dapm_route max9867_audio_map[] = { {"LINE_IN", NULL, "Right Line"}, }; -enum rates { - pcm_rate_8, pcm_rate_16, pcm_rate_24, - pcm_rate_32, pcm_rate_44, - pcm_rate_48, max_pcm_rate, +static const unsigned int max9867_rates_44k1[] = { + 11025, 22050, 44100, }; -static const struct ni_div_rates { - u32 mclk; - u16 ni[max_pcm_rate]; -} ni_div[] = { - {11289600, {0x116A, 0x22D4, 0x343F, 0x45A9, 0x6000, 0x687D} }, - {12000000, {0x1062, 0x20C5, 0x3127, 0x4189, 0x5A51, 0x624E} }, - {12288000, {0x1000, 0x2000, 0x3000, 0x4000, 0x5833, 0x6000} }, - {13000000, {0x0F20, 0x1E3F, 0x2D5F, 0x3C7F, 0x535F, 0x5ABE} }, - {19200000, {0x0A3D, 0x147B, 0x1EB8, 0x28F6, 0x3873, 0x3D71} }, - {24000000, {0x1062, 0x20C5, 0x1893, 0x4189, 0x5A51, 0x624E} }, - {26000000, {0x0F20, 0x1E3F, 0x16AF, 0x3C7F, 0x535F, 0x5ABE} }, - {27000000, {0x0E90, 0x1D21, 0x15D8, 0x3A41, 0x5048, 0x5762} }, +static const struct snd_pcm_hw_constraint_list max9867_constraints_44k1 = { + .list = max9867_rates_44k1, + .count = ARRAY_SIZE(max9867_rates_44k1), }; -static inline int get_ni_value(int mclk, int rate) +static const unsigned int max9867_rates_48k[] = { + 8000, 16000, 32000, 48000, +}; + +static const struct snd_pcm_hw_constraint_list max9867_constraints_48k = { + .list = max9867_rates_48k, + .count = ARRAY_SIZE(max9867_rates_48k), +}; + +struct max9867_priv { + struct regmap *regmap; + const struct snd_pcm_hw_constraint_list *constraints; + unsigned int sysclk, pclk; + bool master, dsp_a; +}; + +static int max9867_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { - int i, ret = 0; + struct max9867_priv *max9867 = + snd_soc_component_get_drvdata(dai->component); - /* find the closest rate index*/ - for (i = 0; i < ARRAY_SIZE(ni_div); i++) { - if (ni_div[i].mclk >= mclk) - break; - } - if (i == ARRAY_SIZE(ni_div)) - return -EINVAL; + if (max9867->constraints) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, max9867->constraints); - switch (rate) { - case 8000: - return ni_div[i].ni[pcm_rate_8]; - case 16000: - return ni_div[i].ni[pcm_rate_16]; - case 32000: - return ni_div[i].ni[pcm_rate_32]; - case 44100: - return ni_div[i].ni[pcm_rate_44]; - case 48000: - return ni_div[i].ni[pcm_rate_48]; - default: - pr_err("%s wrong rate %d\n", __func__, rate); - ret = -EINVAL; - } - return ret; + return 0; } static int max9867_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + int value; + unsigned long int rate, ratio; struct snd_soc_component *component = dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - unsigned int ni_h, ni_l; - int value; + unsigned int ni = DIV_ROUND_CLOSEST_ULL(96ULL * 0x10000 * params_rate(params), + max9867->pclk); - value = get_ni_value(max9867->sysclk, params_rate(params)); - if (value < 0) - return value; - - ni_h = (0xFF00 & value) >> 8; - ni_l = 0x00FF & value; /* set up the ni value */ regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, - MAX9867_NI_HIGH_MASK, ni_h); + MAX9867_NI_HIGH_MASK, (0xFF00 & ni) >> 8); regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, - MAX9867_NI_LOW_MASK, ni_l); - if (!max9867->master) { - /* - * digital pll locks on to any externally supplied LRCLK signal - * and also enable rapid lock mode. - */ - regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, - MAX9867_RAPID_LOCK, MAX9867_RAPID_LOCK); - regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, - MAX9867_PLL, MAX9867_PLL); - } else { - unsigned long int bclk_rate, pclk_bclk_ratio; - int bclk_value; - - bclk_rate = params_rate(params) * 2 * params_width(params); - pclk_bclk_ratio = max9867->pclk/bclk_rate; - switch (params_width(params)) { - case 8: - case 16: - switch (pclk_bclk_ratio) { - case 2: - bclk_value = MAX9867_IFC1B_PCLK_2; - break; - case 4: - bclk_value = MAX9867_IFC1B_PCLK_4; - break; + MAX9867_NI_LOW_MASK, 0x00FF & ni); + if (max9867->master) { + if (max9867->dsp_a) { + value = MAX9867_IFC1B_48X; + } else { + rate = params_rate(params) * 2 * params_width(params); + ratio = max9867->pclk / rate; + switch (params_width(params)) { case 8: - bclk_value = MAX9867_IFC1B_PCLK_8; - break; case 16: - bclk_value = MAX9867_IFC1B_PCLK_16; + switch (ratio) { + case 2: + value = MAX9867_IFC1B_PCLK_2; + break; + case 4: + value = MAX9867_IFC1B_PCLK_4; + break; + case 8: + value = MAX9867_IFC1B_PCLK_8; + break; + case 16: + value = MAX9867_IFC1B_PCLK_16; + break; + default: + return -EINVAL; + } + break; + case 24: + value = MAX9867_IFC1B_48X; + break; + case 32: + value = MAX9867_IFC1B_64X; break; default: - dev_err(component->dev, - "unsupported sampling rate\n"); return -EINVAL; } - break; - case 24: - bclk_value = MAX9867_IFC1B_24BIT; - break; - case 32: - bclk_value = MAX9867_IFC1B_32BIT; - break; - default: - dev_err(component->dev, "unsupported sampling rate\n"); - return -EINVAL; } regmap_update_bits(max9867->regmap, MAX9867_IFC1B, - MAX9867_IFC1B_BCLK_MASK, bclk_value); + MAX9867_IFC1B_BCLK_MASK, value); + } else { + /* + * digital pll locks on to any externally supplied LRCLK signal + * and also enable rapid lock mode. + */ + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, + MAX9867_RAPID_LOCK, MAX9867_RAPID_LOCK); + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, + MAX9867_PLL, MAX9867_PLL); } return 0; } @@ -285,8 +267,16 @@ static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, freq); return -EINVAL; } - value = value << MAX9867_PSCLK_SHIFT; + if (freq % 48000 == 0) + max9867->constraints = &max9867_constraints_48k; + else if (freq % 44100 == 0) + max9867->constraints = &max9867_constraints_44k1; + else + dev_warn(component->dev, + "Unable to set exact rate with %uHz clock frequency\n", + freq); max9867->sysclk = freq; + value = value << MAX9867_PSCLK_SHIFT; /* exact integer mode is not supported */ value &= ~MAX9867_FREQ_MASK; regmap_update_bits(max9867->regmap, MAX9867_SYSCLK, @@ -299,16 +289,17 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - u8 iface1A = 0, iface1B = 0; + u8 iface1A, iface1B; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - max9867->master = 1; - iface1A |= MAX9867_MASTER; + max9867->master = true; + iface1A = MAX9867_MASTER; + iface1B = MAX9867_IFC1B_48X; break; case SND_SOC_DAIFMT_CBS_CFS: - max9867->master = 0; - iface1A &= ~MAX9867_MASTER; + max9867->master = false; + iface1A = iface1B = 0; break; default: return -EINVAL; @@ -316,9 +307,11 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + max9867->dsp_a = false; iface1A |= MAX9867_I2S_DLY; break; case SND_SOC_DAIFMT_DSP_A: + max9867->dsp_a = true; iface1A |= MAX9867_TDM_MODE | MAX9867_SDOUT_HIZ; break; default: @@ -344,20 +337,18 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, regmap_write(max9867->regmap, MAX9867_IFC1A, iface1A); regmap_write(max9867->regmap, MAX9867_IFC1B, iface1B); + return 0; } static const struct snd_soc_dai_ops max9867_dai_ops = { - .set_fmt = max9867_dai_set_fmt, .set_sysclk = max9867_set_dai_sysclk, + .set_fmt = max9867_dai_set_fmt, .digital_mute = max9867_mute, - .hw_params = max9867_dai_hw_params, + .startup = max9867_startup, + .hw_params = max9867_dai_hw_params, }; -#define MAX9867_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -#define MAX9867_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) - static struct snd_soc_dai_driver max9867_dai[] = { { .name = "max9867-aif1", @@ -365,15 +356,15 @@ static struct snd_soc_dai_driver max9867_dai[] = { .stream_name = "HiFi Playback", .channels_min = 2, .channels_max = 2, - .rates = MAX9867_RATES, - .formats = MAX9867_FORMATS, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "HiFi Capture", .channels_min = 2, .channels_max = 2, - .rates = MAX9867_RATES, - .formats = MAX9867_FORMATS, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &max9867_dai_ops, .symmetric_rates = 1, diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index d9170850c96e..7f037ab701a5 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -26,13 +26,11 @@ #define MAX9867_PSCLK_10_20 0x1 #define MAX9867_PSCLK_20_40 0x2 #define MAX9867_PSCLK_40_60 0x3 -#define MAX9867_AUDIOCLKHIGH 0x06 -#define MAX9867_NI_HIGH_WIDTH 0x7 -#define MAX9867_NI_HIGH_MASK 0x7F -#define MAX9867_NI_LOW_MASK 0x7F -#define MAX9867_NI_LOW_SHIFT 0x1 -#define MAX9867_PLL (1<<7) -#define MAX9867_AUDIOCLKLOW 0x07 +#define MAX9867_AUDIOCLKHIGH 0x06 +#define MAX9867_NI_HIGH_MASK 0x7F +#define MAX9867_NI_LOW_MASK 0xFE +#define MAX9867_PLL (1<<7) +#define MAX9867_AUDIOCLKLOW 0x07 #define MAX9867_RAPID_LOCK 0x01 #define MAX9867_IFC1A 0x08 #define MAX9867_MASTER (1<<7) @@ -43,12 +41,12 @@ #define MAX9867_BCI_MODE (1<<5) #define MAX9867_IFC1B 0x09 #define MAX9867_IFC1B_BCLK_MASK 7 -#define MAX9867_IFC1B_32BIT 0x01 -#define MAX9867_IFC1B_24BIT 0x02 -#define MAX9867_IFC1B_PCLK_2 4 -#define MAX9867_IFC1B_PCLK_4 5 -#define MAX9867_IFC1B_PCLK_8 6 -#define MAX9867_IFC1B_PCLK_16 7 +#define MAX9867_IFC1B_64X 0x01 +#define MAX9867_IFC1B_48X 0x02 +#define MAX9867_IFC1B_PCLK_2 0x04 +#define MAX9867_IFC1B_PCLK_4 0x05 +#define MAX9867_IFC1B_PCLK_8 0x06 +#define MAX9867_IFC1B_PCLK_16 0x07 #define MAX9867_CODECFLTR 0x0a #define MAX9867_DACGAIN 0x0b #define MAX9867_DACLEVEL 0x0c @@ -72,11 +70,4 @@ #define MAX9867_CACHEREGNUM 10 -/* codec private data */ -struct max9867_priv { - struct regmap *regmap; - unsigned int sysclk; - unsigned int pclk; - unsigned int master; -}; #endif -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: max9867: Fix signal paths" to the asoc tree
by Mark Brown 07 Dec '18

07 Dec '18

The patch ASoC: max9867: Fix signal paths 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 bc2610a6c4c92f0bf6bbf4c6641407536f30daaf Mon Sep 17 00:00:00 2001 From: Ladislav Michl <ladis(a)linux-mips.org> Date: Tue, 4 Dec 2018 19:21:33 +0100 Subject: [PATCH] ASoC: max9867: Fix signal paths Sound capture and line bypass currently do not work as well as some mixer controls. Fix that by building proper audio paths and adjusting volume controls to match datasheet. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max9867.c | 188 ++++++++++++++++++++----------------- sound/soc/codecs/max9867.h | 8 +- 2 files changed, 102 insertions(+), 94 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index faec094161c1..7a08e2c70e17 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -23,107 +23,124 @@ static const char *const max9867_spmode[] = { "Stereo Single", "Mono Single", "Stereo Single Fast", "Mono Single Fast" }; -static const char *const max9867_sidetone_text[] = { - "None", "Left", "Right", "LeftRight", "LeftRightDiv2", -}; static const char *const max9867_filter_text[] = {"IIR", "FIR"}; static SOC_ENUM_SINGLE_DECL(max9867_filter, MAX9867_CODECFLTR, 7, max9867_filter_text); static SOC_ENUM_SINGLE_DECL(max9867_spkmode, MAX9867_MODECONFIG, 0, max9867_spmode); -static SOC_ENUM_SINGLE_DECL(max9867_sidetone, MAX9867_DACGAIN, 6, - max9867_sidetone_text); -static DECLARE_TLV_DB_SCALE(max9860_capture_tlv, -600, 200, 0); -static DECLARE_TLV_DB_SCALE(max9860_mic_tlv, 2000, 100, 1); -static DECLARE_TLV_DB_SCALE(max9860_adc_left_tlv, -1200, 100, 1); -static DECLARE_TLV_DB_SCALE(max9860_adc_right_tlv, -1200, 100, 1); -static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max98088_micboost_tlv, - 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), - 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max9867_master_tlv, + 0, 2, TLV_DB_SCALE_ITEM(-8600, 200, 1), + 3, 17, TLV_DB_SCALE_ITEM(-7800, 400, 0), + 18, 25, TLV_DB_SCALE_ITEM(-2000, 200, 0), + 26, 34, TLV_DB_SCALE_ITEM( -500, 100, 0), + 35, 40, TLV_DB_SCALE_ITEM( 350, 50, 0), +); +static DECLARE_TLV_DB_SCALE(max9867_mic_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_line_tlv, -600, 200, 0); +static DECLARE_TLV_DB_SCALE(max9867_adc_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_dac_tlv, -1500, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_dacboost_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max9867_micboost_tlv, + 0, 2, TLV_DB_SCALE_ITEM(-2000, 2000, 1), + 3, 3, TLV_DB_SCALE_ITEM(3000, 0, 0), ); static const struct snd_kcontrol_new max9867_snd_controls[] = { - SOC_DOUBLE_R("Master Playback Volume", MAX9867_LEFTVOL, - MAX9867_RIGHTVOL, 0, 63, 1), - SOC_DOUBLE_R_TLV("Capture Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, - 0, 15, 1, max9860_capture_tlv), - SOC_DOUBLE_R_TLV("Mic Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, 0, 31, 1, max9860_mic_tlv), - SOC_DOUBLE_R_TLV("Mic Boost Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, 5, 3, 0, max98088_micboost_tlv), - SOC_ENUM("Digital Sidetone Src", max9867_sidetone), - SOC_SINGLE("Sidetone Volume", MAX9867_DACGAIN, 0, 31, 1), - SOC_SINGLE("DAC Volume", MAX9867_DACLEVEL, 4, 3, 0), - SOC_SINGLE("DAC Attenuation", MAX9867_DACLEVEL, 0, 15, 1), - SOC_SINGLE_TLV("ADC Left Volume", MAX9867_ADCLEVEL, - 4, 15, 1, max9860_adc_left_tlv), - SOC_SINGLE_TLV("ADC Right Volume", MAX9867_ADCLEVEL, - 0, 15, 1, max9860_adc_right_tlv), + SOC_DOUBLE_R_TLV("Master Playback Volume", MAX9867_LEFTVOL, + MAX9867_RIGHTVOL, 0, 41, 1, max9867_master_tlv), + SOC_DOUBLE_R_TLV("Line Capture Volume", MAX9867_LEFTLINELVL, + MAX9867_RIGHTLINELVL, 0, 15, 1, max9867_line_tlv), + SOC_DOUBLE_R_TLV("Mic Capture Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 0, 20, 1, max9867_mic_tlv), + SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 5, 4, 0, max9867_micboost_tlv), + SOC_SINGLE("Digital Sidetone Volume", MAX9867_SIDETONE, 0, 31, 1), + SOC_SINGLE_TLV("Digital Playback Volume", MAX9867_DACLEVEL, 0, 15, 1, + max9867_dac_tlv), + SOC_SINGLE_TLV("Digital Boost Playback Volume", MAX9867_DACLEVEL, 4, 3, 0, + max9867_dacboost_tlv), + SOC_DOUBLE_TLV("Digital Capture Volume", MAX9867_ADCLEVEL, 0, 4, 15, 1, + max9867_adc_tlv), SOC_ENUM("Speaker Mode", max9867_spkmode), SOC_SINGLE("Volume Smoothing Switch", MAX9867_MODECONFIG, 6, 1, 0), - SOC_SINGLE("ZCD Switch", MAX9867_MODECONFIG, 5, 1, 0), + SOC_SINGLE("Line ZC Switch", MAX9867_MODECONFIG, 5, 1, 0), SOC_ENUM("DSP Filter", max9867_filter), }; -static const char *const max9867_mux[] = {"None", "Mic", "Line", "Mic_Line"}; +/* Input mixer */ +static const struct snd_kcontrol_new max9867_input_mixer_controls[] = { + SOC_DAPM_DOUBLE("Line Capture Switch", MAX9867_INPUTCONFIG, 7, 5, 1, 0), + SOC_DAPM_DOUBLE("Mic Capture Switch", MAX9867_INPUTCONFIG, 6, 4, 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), +}; -static SOC_ENUM_SINGLE_DECL(max9867_mux_enum, - MAX9867_INPUTCONFIG, MAX9867_INPUT_SHIFT, - max9867_mux); +/* Sidetone mixer */ +static const struct snd_kcontrol_new max9867_sidetone_mixer_controls[] = { + SOC_DAPM_DOUBLE("Sidetone Switch", MAX9867_SIDETONE, 6, 7, 1, 0), +}; -static const struct snd_kcontrol_new max9867_dapm_mux_controls = - SOC_DAPM_ENUM("Route", max9867_mux_enum); +/* 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_kcontrol_new max9867_left_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 6, 1, 0); -static const struct snd_kcontrol_new max9867_right_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 5, 1, 0); -static const struct snd_kcontrol_new max9867_line_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_LEFTLINELVL, 6, 1, 1); static const struct snd_soc_dapm_widget max9867_dapm_widgets[] = { - SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("Left DAC", NULL, MAX9867_PWRMAN, 3, 0), - SND_SOC_DAPM_DAC("Right DAC", NULL, MAX9867_PWRMAN, 2, 0), - SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_OUTPUT("HPOUT"), - - SND_SOC_DAPM_AIF_IN("DAI_IN", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture", MAX9867_PWRMAN, 1, 0), - SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", MAX9867_PWRMAN, 0, 0), - SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, - &max9867_dapm_mux_controls), - - SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("Left Line", MAX9867_LEFTLINELVL, 6, 1, - &max9867_left_dapm_control), - SND_SOC_DAPM_SWITCH("Right Line", MAX9867_RIGTHLINELVL, 6, 1, - &max9867_right_dapm_control), - SND_SOC_DAPM_SWITCH("Line Mixer", SND_SOC_NOPM, 0, 0, - &max9867_line_dapm_control), - SND_SOC_DAPM_INPUT("LINE_IN"), + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("LINL"), + SND_SOC_DAPM_INPUT("LINR"), + + SND_SOC_DAPM_PGA("Left Line Input", MAX9867_PWRMAN, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Line Input", MAX9867_PWRMAN, 5, 0, NULL, 0), + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Mixer", SND_SOC_NOPM, 0, 0, + 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_NAMED_CTL("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 DAC", NULL, "DAI_OUT"}, - {"Right DAC", NULL, "DAI_OUT"}, - {"Output Mixer", NULL, "Left DAC"}, - {"Output Mixer", NULL, "Right DAC"}, - {"HPOUT", NULL, "Output Mixer"}, - - {"Left ADC", NULL, "DAI_IN"}, - {"Right ADC", NULL, "DAI_IN"}, - {"Input Mixer", NULL, "Left ADC"}, - {"Input Mixer", NULL, "Right ADC"}, - {"Input Mux", "Line", "Input Mixer"}, - {"Input Mux", "Mic", "Input Mixer"}, - {"Input Mux", "Mic_Line", "Input Mixer"}, - {"Right Line", "Switch", "Input Mux"}, - {"Left Line", "Switch", "Input Mux"}, - {"LINE_IN", NULL, "Left Line"}, - {"LINE_IN", NULL, "Right Line"}, + {"Left Line Input", NULL, "LINL"}, + {"Right Line Input", NULL, "LINR"}, + {"Input Mixer", "Mic Capture Switch", "MICL"}, + {"Input Mixer", "Mic Capture Switch", "MICR"}, + {"Input Mixer", "Line Capture Switch", "Left Line Input"}, + {"Input Mixer", "Line Capture Switch", "Right Line Input"}, + {"ADCL", NULL, "Input Mixer"}, + {"ADCR", NULL, "Input Mixer"}, + + {"Digital", "Sidetone Switch", "ADCL"}, + {"Digital", "Sidetone Switch", "ADCR"}, + {"DACL", NULL, "Digital"}, + {"DACR", NULL, "Digital"}, + + {"Output Mixer", "Line Bypass Switch", "Left Line Input"}, + {"Output Mixer", "Line Bypass Switch", "Right Line Input"}, + {"Output Mixer", NULL, "DACL"}, + {"Output Mixer", NULL, "DACR"}, + {"Master Playback", "Switch", "Output Mixer"}, + {"LOUT", NULL, "Master Playback"}, + {"ROUT", NULL, "Master Playback"}, }; static const unsigned int max9867_rates_44k1[] = { @@ -235,13 +252,8 @@ static int max9867_mute(struct snd_soc_dai *dai, int mute) struct snd_soc_component *component = dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - if (mute) - regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, - MAX9867_DAC_MUTE_MASK, MAX9867_DAC_MUTE_MASK); - else - regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, - MAX9867_DAC_MUTE_MASK, 0); - return 0; + return regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, + 1 << 6, !!mute << 6); } static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, @@ -464,8 +476,8 @@ static const struct reg_default max9867_reg[] = { { 0x0B, 0x00 }, { 0x0C, 0x00 }, { 0x0D, 0x00 }, - { 0x0E, 0x00 }, - { 0x0F, 0x00 }, + { 0x0E, 0x40 }, + { 0x0F, 0x40 }, { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index 7f037ab701a5..2277798291a1 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -48,20 +48,16 @@ #define MAX9867_IFC1B_PCLK_8 0x06 #define MAX9867_IFC1B_PCLK_16 0x07 #define MAX9867_CODECFLTR 0x0a -#define MAX9867_DACGAIN 0x0b +#define MAX9867_SIDETONE 0x0b #define MAX9867_DACLEVEL 0x0c -#define MAX9867_DAC_MUTE_SHIFT 0x6 -#define MAX9867_DAC_MUTE_WIDTH 0x1 -#define MAX9867_DAC_MUTE_MASK (0x1<<MAX9867_DAC_MUTE_SHIFT) #define MAX9867_ADCLEVEL 0x0d #define MAX9867_LEFTLINELVL 0x0e -#define MAX9867_RIGTHLINELVL 0x0f +#define MAX9867_RIGHTLINELVL 0x0f #define MAX9867_LEFTVOL 0x10 #define MAX9867_RIGHTVOL 0x11 #define MAX9867_LEFTMICGAIN 0x12 #define MAX9867_RIGHTMICGAIN 0x13 #define MAX9867_INPUTCONFIG 0x14 -#define MAX9867_INPUT_SHIFT 0x6 #define MAX9867_MICCONFIG 0x15 #define MAX9867_MODECONFIG 0x16 #define MAX9867_PWRMAN 0x17 -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH v2] ASoC: max9867: Add copyright and module author
by Ladislav Michl 07 Dec '18

07 Dec '18

Driver rewritten, assign copyright notice and change module author as original one remains silent and I want to be notified about bugs. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> --- CHANGES: -v2: convert entire comment block to C++ style sound/soc/codecs/max9867.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 7a08e2c70e17..8600c5439e1e 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -1,12 +1,10 @@ -/* - * max9867.c -- max9867 ALSA SoC Audio driver - * - * Copyright 2013-15 Maxim Integrated Products - * - * 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. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// MAX9867 ALSA SoC codec driver +// +// Copyright 2013-2015 Maxim Integrated Products +// Copyright 2018 Ladislav Michl <ladis(a)linux-mips.org> +// #include <linux/delay.h> #include <linux/i2c.h> @@ -551,6 +549,6 @@ static struct i2c_driver max9867_i2c_driver = { module_i2c_driver(max9867_i2c_driver); -MODULE_AUTHOR("anish kumar <yesanishhere(a)gmail.com>"); -MODULE_DESCRIPTION("ALSA SoC MAX9867 driver"); +MODULE_AUTHOR("Ladislav Michl <ladis(a)linux-mips.org>"); +MODULE_DESCRIPTION("ASoC MAX9867 driver"); MODULE_LICENSE("GPL"); -- 2.20.0.rc2

2 1

Re: [alsa-devel] HDA headset button support
by Takashi Iwai 07 Dec '18

07 Dec '18

On Wed, 28 Nov 2018 13:03:58 +0100, Kailang wrote: > > Hi Takashi, > > >I'd rather like to extend snd_hda_jack_add_kctl() with the key > >support (maybe adding a new function). You can pass there the > >combination of SND_JACK_BTN_X with KEY_XXX in an array. > > Do you mean create new function like below? > snd_hda_jack_add_key_kctl() > > It couldn't use snd_hda_jack_add_kctl() directly. > Because it had report pin state and get type from NID. > > type = get_input_jack_type(codec, nid); > ..... > ..... > state = snd_hda_jack_detect(codec, nid); > snd_jack_report(jack->jack, state ? jack->type : 0); > > So, to create a new function in hda_jack.c was better. Well, this seems more deeper than I thought. Below is my quick attempt to add the infrastructure. This adds more flexibility but still easiness for other possible button implementations, hopefully. Could you check the series below? thanks, Takashi

2 10

[alsa-devel] [PATCH 0/4] alsa-utils: axfer: add help texts
by GitHub pull_request - opened 07 Dec '18

07 Dec '18

alsa-project/alsa-utils pull request #7 was opened from takaswie: This patchset adds help texts for command entry, list and transfer subcommands. In this time, contents for each backend of transfer subcommand is not added yet. Unfortunately, execution of aliases (arecord/aplay) can print help text for command text only. This bug will be fixed in future commits. Additionally, due to program design, execution of the aliases cannot print help text of list subcommand. This is unavoidable loss of backward compatibility. ``` Takashi Sakamoto (4): axfer: print help text of command entry axfer: print help text of list subcommand axfer: print help text of transfer subcommand just for common options axfer: enable each backend to print own help axfer/main.c | 12 +++++++++++- axfer/subcmd-list.c | 9 ++++++++- axfer/xfer-libasound.c | 6 ++++++ axfer/xfer-libffado.c | 6 ++++++ axfer/xfer-options.c | 35 +++++++++++++++++++++++++++++++++++ axfer/xfer.c | 5 +++++ axfer/xfer.h | 2 ++ 7 files changed, 73 insertions(+), 2 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/7 Patch URL : https://github.com/alsa-project/alsa-utils/pull/7.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] ALC700 headphone no sound issue
by Kailang 07 Dec '18

07 Dec '18

Hi Takashi, In Intel ENG board with ALC700. It had headphone no sound issue. If it plugged headphone or headset into the jack, then do the reboot, it will have a chance to cause headphone no sound. It just need to run the headphone mode procedure after boot time. The issue will be fixed. It also suitable for ALC234 ALC274 and ALC294. BR, Kailang

2 5

[alsa-devel] [GIT PULL] soundwire updates
by Vinod Koul 07 Dec '18

07 Dec '18

Hey Greg, Please pull to receive a single commit for soundwire tree for this cycle. Am off next week so sending this bit early The following changes since commit 651022382c7f8da46cb4872a545ee1da6d097d2a: Linux 4.20-rc1 (2018-11-04 15:37:52 -0800) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git tags/soundwire-4.21-rc1 for you to fetch changes up to b1635596860dd3229645e41276f6c83ca788e4c1: soundwire: intel: constify snd_soc_dai_ops structures (2018-11-12 15:36:06 +0530) ---------------------------------------------------------------- soundwire updates for 4.21-rc1 A single change: - constify snd_soc_dai_ops struct by Julia ---------------------------------------------------------------- Julia Lawall (1): soundwire: intel: constify snd_soc_dai_ops structures drivers/soundwire/intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) -- ~Vinod

2 1

[alsa-devel] [PATCH] tinycompress: Fix missing '}' around extern C
by Charles Keepax 07 Dec '18

07 Dec '18

From: Piotr Stankiewicz <piotrs(a)opensource.cirrus.com> Signed-off-by: Piotr Stankiewicz <piotrs(a)opensource.cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/tinycompress/tinycompress.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 780dce6..f0fae30 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -276,4 +276,8 @@ int is_compress_ready(struct compress *compress); /* Returns a human readable reason for the last error */ const char *compress_get_error(struct compress *compress); +#if defined(__cplusplus) +} // extern "C" +#endif + #endif -- 2.11.0

2 1

[alsa-devel] [PATCH 1/3] ASoC: rt5660: Add a new ACPI match ID
by Hui Wang 07 Dec '18

07 Dec '18

The Realtek codec ALC3277 is 100% compatible with the codec RT5660 in I2S mode. And on the Dell IoT platform, the codec is ALC3277, and the HID of the codec in the BIOS is 10EC3277, so adding this ID to the ACPI match table. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/codecs/rt5660.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index 27f7445b2432..e74b2e8cd423 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -1246,6 +1246,7 @@ MODULE_DEVICE_TABLE(of, rt5660_of_match); static const struct acpi_device_id rt5660_acpi_match[] = { { "10EC5660", 0 }, + { "10EC3277", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5660_acpi_match); -- 2.17.1

3 7

[alsa-devel] [PATCH v8 0/7] Add support for audiocodec in Allwinner A64
by Vasily Khoruzhick 07 Dec '18

07 Dec '18

v2: - Use simple-amplifier for speaker amp on Pinebook - Rename sun50i-a64-i2s to sun50i-a64-codec-i2s to preserve compatible string for other 3 I2S modules in A64 in case if there's any incompatibility with H3 v3: - renamed sunxi-adda-pr-regmap to sun8i-adda-pr-regmap - use ilog2() to calculate reg value for LRCK div instead of using a table v4: - dts: don't use 'Mic' and 'Headset Mic' widgets from sun8i-codec, define our board-level widgets instead. v5: - collect all the tags v6: - driver patches has been merged through ASoC tree - rebase onto 4.20-rc1 - Drop 'Speaker' from routes on sopine and pine64, they don't have speaker. v7: - drop 'Mic' and 'Headset Mic' from dtsi - move AIF1 Slot 0 ADC to ADC route to sun8i-codec - drop BIAS routes for Pine64 and SoPine v8 - drop merged patches - add regulator property for simple amplifier, make enable-gpio optional - add regulator to speaker amp, introduce headphone amp Vasily Khoruzhick (7): ASoC: dt-bindings: add regulator property to simple amplifier ASoC: simple-amplifier: add VCC regulator widget ASoC: dt-bindings: make enable-gpio optional for simple amplifier ASoC: simple-amplifier: make enable-gpio optional arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../bindings/sound/simple-amplifier.txt | 5 ++ .../boot/dts/allwinner/sun50i-a64-pine64.dts | 35 ++++++++++++ .../dts/allwinner/sun50i-a64-pinebook.dts | 53 ++++++++++++++++++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 35 ++++++++++++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 54 +++++++++++++++++++ sound/soc/codecs/simple-amplifier.c | 7 ++- 6 files changed, 188 insertions(+), 1 deletion(-) -- 2.19.1

4 10

[alsa-devel] [PATCH v2] ASoC: max9867: Remove useless assignment
by Ladislav Michl 07 Dec '18

07 Dec '18

ret is assigned later, no need to initialize it. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> --- CHANGES: -v2: add missing Signed-off-by (I'm sorry for that...) sound/soc/codecs/max9867.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 1cda54b59854..1b155f40b106 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -476,7 +476,7 @@ static int max9867_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct max9867_priv *max9867; - int ret = 0, reg; + int ret, reg; max9867 = devm_kzalloc(&i2c->dev, sizeof(*max9867), GFP_KERNEL); if (!max9867) @@ -497,10 +497,8 @@ static int max9867_i2c_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "device revision: %x\n", reg); ret = devm_snd_soc_register_component(&i2c->dev, &max9867_component, max9867_dai, ARRAY_SIZE(max9867_dai)); - if (ret < 0) { + if (ret < 0) dev_err(&i2c->dev, "Failed to register component: %d\n", ret); - return ret; - } return ret; } -- 2.20.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add pmc_plt_clk_0 quirk for Chromebook Gnawty
by Hans de Goede 06 Dec '18

06 Dec '18

The Gnawty model Chromebook uses pmc_plt_clk_0 instead of pmc_plt_clk_3 for the mclk, just like the Clapper and Swanky models. This commit adds a DMI based quirk for this. This fixing audio no longer working on these devices after commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") that commit fixes us unnecessary keeping unused clocks on, but in case of the Gnawty that was breaking audio support since we were not using the right clock in the cht_bsw_max98090_ti machine driver. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=201787 Cc: stable(a)vger.kernel.org Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Reported-and-tested-by: Jaime Pérez <19.jaime.91(a)gmail.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index ad0c98383853..08a5152e635a 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -396,6 +396,13 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { }, .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, }, + { + /* Gnawty model Chromebook (Acer Chromebook CB3-111) */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Gnawty"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Swanky model Chromebook (Toshiba Chromebook 2) */ .matches = { -- 2.19.2

3 2

[alsa-devel] [PATCH 1/2] ASoC: use DMA addr rather than CPU pa for acp_audio_dma
by Yu Zhao 06 Dec '18

06 Dec '18

We shouldn't assume CPU physical address we get from page_to_phys() is same as DMA address we get from dma_alloc_coherent(). On x86_64, we won't run into any problem with the assumption when dma_ops is nommu_dma_ops. However, DMA address is IOVA when IOMMU is enabled. And it's most likely different from CPU physical address when AMD IOMMU is not in passthrough mode. Signed-off-by: Yu Zhao <yuzhao(a)google.com> --- sound/soc/amd/acp-pcm-dma.c | 15 +++++---------- sound/soc/amd/acp.h | 2 +- 2 files changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index cdebab2f8ce5..fd3db4c37882 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -303,11 +303,10 @@ static void set_acp_to_i2s_dma_descriptors(void __iomem *acp_mmio, u32 size, } /* Create page table entries in ACP SRAM for the allocated memory */ -static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, +static void acp_pte_config(void __iomem *acp_mmio, dma_addr_t addr, u16 num_of_pages, u32 pte_offset) { u16 page_idx; - u64 addr; u32 low; u32 high; u32 offset; @@ -317,7 +316,6 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, /* Load the low address of page int ACP SRAM through SRBM */ acp_reg_write((offset + (page_idx * 8)), acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - addr = page_to_phys(pg); low = lower_32_bits(addr); high = upper_32_bits(addr); @@ -333,7 +331,7 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, acp_reg_write(high, acp_mmio, mmACP_SRBM_Targ_Idx_Data); /* Move to next physically contiguos page */ - pg++; + addr += PAGE_SIZE; } } @@ -343,7 +341,7 @@ static void config_acp_dma(void __iomem *acp_mmio, { u16 ch_acp_sysmem, ch_acp_i2s; - acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages, + acp_pte_config(acp_mmio, rtd->dma_addr, rtd->num_of_pages, rtd->pte_offset); if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { @@ -850,7 +848,6 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, int status; uint64_t size; u32 val = 0; - struct page *pg; struct snd_pcm_runtime *runtime; struct audio_substream_data *rtd; struct snd_soc_pcm_runtime *prtd = substream->private_data; @@ -986,16 +983,14 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, return status; memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); - pg = virt_to_page(substream->dma_buffer.area); - if (pg) { + if (substream->dma_buffer.area) { acp_set_sram_bank_state(rtd->acp_mmio, 0, true); /* Save for runtime private data */ - rtd->pg = pg; + rtd->dma_addr = substream->dma_buffer.addr; rtd->order = get_order(size); /* Fill the page table entries in ACP SRAM */ - rtd->pg = pg; rtd->size = size; rtd->num_of_pages = PAGE_ALIGN(size) >> PAGE_SHIFT; rtd->direction = substream->stream; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index dbbb1a85638d..e5ab6c6040a6 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -123,7 +123,7 @@ enum acp_dma_priority_level { }; struct audio_substream_data { - struct page *pg; + dma_addr_t dma_addr; unsigned int order; u16 num_of_pages; u16 i2s_instance; -- 2.20.0.rc1.387.gf8505762e3-goog

3 5

[alsa-devel] [PATCH] ASoC: Use of_node_name_eq for node name comparisons
by Rob Herring 06 Dec '18

06 Dec '18

Convert string compares of DT node names to use of_node_name_eq helper instead. This removes direct access to the node name pointer. For the FSL ASoC card, the full node names appear to be "ssi", "esai", and "sai", so there's not any reason to use strstr and of_node_name_eq can be used instead. Cc: Timur Tabi <timur(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 6 +++--- sound/soc/generic/simple-scu-card.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 44433b20435c..81f2fe2c6d23 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -571,17 +571,17 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) } /* Common settings for corresponding Freescale CPU DAI driver */ - if (strstr(cpu_np->name, "ssi")) { + if (of_node_name_eq(cpu_np, "ssi")) { /* Only SSI needs to configure AUDMUX */ ret = fsl_asoc_card_audmux_init(np, priv); if (ret) { dev_err(&pdev->dev, "failed to init audmux\n"); goto asrc_fail; } - } else if (strstr(cpu_np->name, "esai")) { + } else if (of_node_name_eq(cpu_np, "esai")) { priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL; - } else if (strstr(cpu_np->name, "sai")) { + } else if (of_node_name_eq(cpu_np, "sai")) { priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1; priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 85b46f0eae0f..e9a1acffcf5b 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -216,7 +216,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) i = 0; for_each_child_of_node(node, np) { is_fe = false; - if (strcmp(np->name, PREFIX "cpu") == 0) + if (of_node_name_eq(np, PREFIX "cpu")) is_fe = true; ret = asoc_simple_card_dai_link_of(np, priv, daifmt, i, is_fe); -- 2.19.1

4 4

[alsa-devel] [PATCH 1/2] ASoC: dt-bindings: cs4270: use 'reset-gpios' rather than 'reset-gpio'
by Daniel Mack 06 Dec '18

06 Dec '18

Bindings should use 'reset-gpios', not 'reset-gpio'. The driver needs to switch to the gpiod consume API to handle this correctly. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- Documentation/devicetree/bindings/sound/cs4270.txt | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/cs4270.txt b/Documentation/devicetree/bindings/sound/cs4270.txt index 6b222f9b8ef5..c33770ec4c3c 100644 --- a/Documentation/devicetree/bindings/sound/cs4270.txt +++ b/Documentation/devicetree/bindings/sound/cs4270.txt @@ -10,8 +10,8 @@ Required properties: Optional properties: - - reset-gpio : a GPIO spec for the reset pin. If specified, it will be - deasserted before communication to the codec starts. + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. Example: -- 2.19.2

2 3

[alsa-devel] [PATCH 1/2] ASoC: dt-bindings: ak4104: use 'reset-gpios' rather than 'reset-gpio'
by Daniel Mack 06 Dec '18

06 Dec '18

Bindings should use 'reset-gpios', not 'reset-gpio'. The driver needs to switch to the gpiod consume API to handle this correctly. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- Documentation/devicetree/bindings/sound/ak4104.txt | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/ak4104.txt b/Documentation/devicetree/bindings/sound/ak4104.txt index deca5e18f304..ae5f7f057dc3 100644 --- a/Documentation/devicetree/bindings/sound/ak4104.txt +++ b/Documentation/devicetree/bindings/sound/ak4104.txt @@ -12,8 +12,8 @@ Required properties: Optional properties: - - reset-gpio : a GPIO spec for the reset pin. If specified, it will be - deasserted before communication to the device starts. + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the device starts. Example: -- 2.19.2

2 3

[alsa-devel] Applied "ASoC: max9867: Calculate LRCLK divider" to the asoc tree
by Mark Brown 06 Dec '18

06 Dec '18

The patch ASoC: max9867: Calculate LRCLK divider 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 127f8a2145957fb53f0d65444720a583b54a58f3 Mon Sep 17 00:00:00 2001 From: Ladislav Michl <ladis(a)linux-mips.org> Date: Tue, 4 Dec 2018 19:21:04 +0100 Subject: [PATCH] ASoC: max9867: Calculate LRCLK divider Drop "Common NI Values Table" and calculate LRCLK divider, then add allowed rate constraints based on master clock frequency. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max9867.c | 207 ++++++++++++++++++------------------- sound/soc/codecs/max9867.h | 31 ++---- 2 files changed, 110 insertions(+), 128 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index c454de434d76..faec094161c1 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -126,124 +126,106 @@ static const struct snd_soc_dapm_route max9867_audio_map[] = { {"LINE_IN", NULL, "Right Line"}, }; -enum rates { - pcm_rate_8, pcm_rate_16, pcm_rate_24, - pcm_rate_32, pcm_rate_44, - pcm_rate_48, max_pcm_rate, +static const unsigned int max9867_rates_44k1[] = { + 11025, 22050, 44100, }; -static const struct ni_div_rates { - u32 mclk; - u16 ni[max_pcm_rate]; -} ni_div[] = { - {11289600, {0x116A, 0x22D4, 0x343F, 0x45A9, 0x6000, 0x687D} }, - {12000000, {0x1062, 0x20C5, 0x3127, 0x4189, 0x5A51, 0x624E} }, - {12288000, {0x1000, 0x2000, 0x3000, 0x4000, 0x5833, 0x6000} }, - {13000000, {0x0F20, 0x1E3F, 0x2D5F, 0x3C7F, 0x535F, 0x5ABE} }, - {19200000, {0x0A3D, 0x147B, 0x1EB8, 0x28F6, 0x3873, 0x3D71} }, - {24000000, {0x1062, 0x20C5, 0x1893, 0x4189, 0x5A51, 0x624E} }, - {26000000, {0x0F20, 0x1E3F, 0x16AF, 0x3C7F, 0x535F, 0x5ABE} }, - {27000000, {0x0E90, 0x1D21, 0x15D8, 0x3A41, 0x5048, 0x5762} }, +static const struct snd_pcm_hw_constraint_list max9867_constraints_44k1 = { + .list = max9867_rates_44k1, + .count = ARRAY_SIZE(max9867_rates_44k1), }; -static inline int get_ni_value(int mclk, int rate) +static const unsigned int max9867_rates_48k[] = { + 8000, 16000, 32000, 48000, +}; + +static const struct snd_pcm_hw_constraint_list max9867_constraints_48k = { + .list = max9867_rates_48k, + .count = ARRAY_SIZE(max9867_rates_48k), +}; + +struct max9867_priv { + struct regmap *regmap; + const struct snd_pcm_hw_constraint_list *constraints; + unsigned int sysclk, pclk; + bool master, dsp_a; +}; + +static int max9867_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { - int i, ret = 0; + struct max9867_priv *max9867 = + snd_soc_component_get_drvdata(dai->component); - /* find the closest rate index*/ - for (i = 0; i < ARRAY_SIZE(ni_div); i++) { - if (ni_div[i].mclk >= mclk) - break; - } - if (i == ARRAY_SIZE(ni_div)) - return -EINVAL; + if (max9867->constraints) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, max9867->constraints); - switch (rate) { - case 8000: - return ni_div[i].ni[pcm_rate_8]; - case 16000: - return ni_div[i].ni[pcm_rate_16]; - case 32000: - return ni_div[i].ni[pcm_rate_32]; - case 44100: - return ni_div[i].ni[pcm_rate_44]; - case 48000: - return ni_div[i].ni[pcm_rate_48]; - default: - pr_err("%s wrong rate %d\n", __func__, rate); - ret = -EINVAL; - } - return ret; + return 0; } static int max9867_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + int value; + unsigned long int rate, ratio; struct snd_soc_component *component = dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - unsigned int ni_h, ni_l; - int value; + unsigned int ni = DIV_ROUND_CLOSEST_ULL(96ULL * 0x10000 * params_rate(params), + max9867->pclk); - value = get_ni_value(max9867->sysclk, params_rate(params)); - if (value < 0) - return value; - - ni_h = (0xFF00 & value) >> 8; - ni_l = 0x00FF & value; /* set up the ni value */ regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, - MAX9867_NI_HIGH_MASK, ni_h); + MAX9867_NI_HIGH_MASK, (0xFF00 & ni) >> 8); regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, - MAX9867_NI_LOW_MASK, ni_l); - if (!max9867->master) { - /* - * digital pll locks on to any externally supplied LRCLK signal - * and also enable rapid lock mode. - */ - regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, - MAX9867_RAPID_LOCK, MAX9867_RAPID_LOCK); - regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, - MAX9867_PLL, MAX9867_PLL); - } else { - unsigned long int bclk_rate, pclk_bclk_ratio; - int bclk_value; - - bclk_rate = params_rate(params) * 2 * params_width(params); - pclk_bclk_ratio = max9867->pclk/bclk_rate; - switch (params_width(params)) { - case 8: - case 16: - switch (pclk_bclk_ratio) { - case 2: - bclk_value = MAX9867_IFC1B_PCLK_2; - break; - case 4: - bclk_value = MAX9867_IFC1B_PCLK_4; - break; + MAX9867_NI_LOW_MASK, 0x00FF & ni); + if (max9867->master) { + if (max9867->dsp_a) { + value = MAX9867_IFC1B_48X; + } else { + rate = params_rate(params) * 2 * params_width(params); + ratio = max9867->pclk / rate; + switch (params_width(params)) { case 8: - bclk_value = MAX9867_IFC1B_PCLK_8; - break; case 16: - bclk_value = MAX9867_IFC1B_PCLK_16; + switch (ratio) { + case 2: + value = MAX9867_IFC1B_PCLK_2; + break; + case 4: + value = MAX9867_IFC1B_PCLK_4; + break; + case 8: + value = MAX9867_IFC1B_PCLK_8; + break; + case 16: + value = MAX9867_IFC1B_PCLK_16; + break; + default: + return -EINVAL; + } + break; + case 24: + value = MAX9867_IFC1B_48X; + break; + case 32: + value = MAX9867_IFC1B_64X; break; default: - dev_err(component->dev, - "unsupported sampling rate\n"); return -EINVAL; } - break; - case 24: - bclk_value = MAX9867_IFC1B_24BIT; - break; - case 32: - bclk_value = MAX9867_IFC1B_32BIT; - break; - default: - dev_err(component->dev, "unsupported sampling rate\n"); - return -EINVAL; } regmap_update_bits(max9867->regmap, MAX9867_IFC1B, - MAX9867_IFC1B_BCLK_MASK, bclk_value); + MAX9867_IFC1B_BCLK_MASK, value); + } else { + /* + * digital pll locks on to any externally supplied LRCLK signal + * and also enable rapid lock mode. + */ + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, + MAX9867_RAPID_LOCK, MAX9867_RAPID_LOCK); + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, + MAX9867_PLL, MAX9867_PLL); } return 0; } @@ -285,8 +267,16 @@ static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, freq); return -EINVAL; } - value = value << MAX9867_PSCLK_SHIFT; + if (freq % 48000 == 0) + max9867->constraints = &max9867_constraints_48k; + else if (freq % 44100 == 0) + max9867->constraints = &max9867_constraints_44k1; + else + dev_warn(component->dev, + "Unable to set exact rate with %uHz clock frequency\n", + freq); max9867->sysclk = freq; + value = value << MAX9867_PSCLK_SHIFT; /* exact integer mode is not supported */ value &= ~MAX9867_FREQ_MASK; regmap_update_bits(max9867->regmap, MAX9867_SYSCLK, @@ -299,16 +289,17 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - u8 iface1A = 0, iface1B = 0; + u8 iface1A, iface1B; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - max9867->master = 1; - iface1A |= MAX9867_MASTER; + max9867->master = true; + iface1A = MAX9867_MASTER; + iface1B = MAX9867_IFC1B_48X; break; case SND_SOC_DAIFMT_CBS_CFS: - max9867->master = 0; - iface1A &= ~MAX9867_MASTER; + max9867->master = false; + iface1A = iface1B = 0; break; default: return -EINVAL; @@ -316,9 +307,11 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + max9867->dsp_a = false; iface1A |= MAX9867_I2S_DLY; break; case SND_SOC_DAIFMT_DSP_A: + max9867->dsp_a = true; iface1A |= MAX9867_TDM_MODE | MAX9867_SDOUT_HIZ; break; default: @@ -344,20 +337,18 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, regmap_write(max9867->regmap, MAX9867_IFC1A, iface1A); regmap_write(max9867->regmap, MAX9867_IFC1B, iface1B); + return 0; } static const struct snd_soc_dai_ops max9867_dai_ops = { - .set_fmt = max9867_dai_set_fmt, .set_sysclk = max9867_set_dai_sysclk, + .set_fmt = max9867_dai_set_fmt, .digital_mute = max9867_mute, - .hw_params = max9867_dai_hw_params, + .startup = max9867_startup, + .hw_params = max9867_dai_hw_params, }; -#define MAX9867_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -#define MAX9867_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) - static struct snd_soc_dai_driver max9867_dai[] = { { .name = "max9867-aif1", @@ -365,15 +356,15 @@ static struct snd_soc_dai_driver max9867_dai[] = { .stream_name = "HiFi Playback", .channels_min = 2, .channels_max = 2, - .rates = MAX9867_RATES, - .formats = MAX9867_FORMATS, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "HiFi Capture", .channels_min = 2, .channels_max = 2, - .rates = MAX9867_RATES, - .formats = MAX9867_FORMATS, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &max9867_dai_ops, .symmetric_rates = 1, diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index d9170850c96e..7f037ab701a5 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -26,13 +26,11 @@ #define MAX9867_PSCLK_10_20 0x1 #define MAX9867_PSCLK_20_40 0x2 #define MAX9867_PSCLK_40_60 0x3 -#define MAX9867_AUDIOCLKHIGH 0x06 -#define MAX9867_NI_HIGH_WIDTH 0x7 -#define MAX9867_NI_HIGH_MASK 0x7F -#define MAX9867_NI_LOW_MASK 0x7F -#define MAX9867_NI_LOW_SHIFT 0x1 -#define MAX9867_PLL (1<<7) -#define MAX9867_AUDIOCLKLOW 0x07 +#define MAX9867_AUDIOCLKHIGH 0x06 +#define MAX9867_NI_HIGH_MASK 0x7F +#define MAX9867_NI_LOW_MASK 0xFE +#define MAX9867_PLL (1<<7) +#define MAX9867_AUDIOCLKLOW 0x07 #define MAX9867_RAPID_LOCK 0x01 #define MAX9867_IFC1A 0x08 #define MAX9867_MASTER (1<<7) @@ -43,12 +41,12 @@ #define MAX9867_BCI_MODE (1<<5) #define MAX9867_IFC1B 0x09 #define MAX9867_IFC1B_BCLK_MASK 7 -#define MAX9867_IFC1B_32BIT 0x01 -#define MAX9867_IFC1B_24BIT 0x02 -#define MAX9867_IFC1B_PCLK_2 4 -#define MAX9867_IFC1B_PCLK_4 5 -#define MAX9867_IFC1B_PCLK_8 6 -#define MAX9867_IFC1B_PCLK_16 7 +#define MAX9867_IFC1B_64X 0x01 +#define MAX9867_IFC1B_48X 0x02 +#define MAX9867_IFC1B_PCLK_2 0x04 +#define MAX9867_IFC1B_PCLK_4 0x05 +#define MAX9867_IFC1B_PCLK_8 0x06 +#define MAX9867_IFC1B_PCLK_16 0x07 #define MAX9867_CODECFLTR 0x0a #define MAX9867_DACGAIN 0x0b #define MAX9867_DACLEVEL 0x0c @@ -72,11 +70,4 @@ #define MAX9867_CACHEREGNUM 10 -/* codec private data */ -struct max9867_priv { - struct regmap *regmap; - unsigned int sysclk; - unsigned int pclk; - unsigned int master; -}; #endif -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: max9867: Fix signal paths" to the asoc tree
by Mark Brown 06 Dec '18

06 Dec '18

The patch ASoC: max9867: Fix signal paths 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 2f388effac683ee34016f915f7ff1d50b3f9df81 Mon Sep 17 00:00:00 2001 From: Ladislav Michl <ladis(a)linux-mips.org> Date: Tue, 4 Dec 2018 19:21:33 +0100 Subject: [PATCH] ASoC: max9867: Fix signal paths Sound capture and line bypass currently do not work as well as some mixer controls. Fix that by building proper audio paths and adjusting volume controls to match datasheet. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max9867.c | 188 ++++++++++++++++++++----------------- sound/soc/codecs/max9867.h | 8 +- 2 files changed, 102 insertions(+), 94 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index faec094161c1..7a08e2c70e17 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -23,107 +23,124 @@ static const char *const max9867_spmode[] = { "Stereo Single", "Mono Single", "Stereo Single Fast", "Mono Single Fast" }; -static const char *const max9867_sidetone_text[] = { - "None", "Left", "Right", "LeftRight", "LeftRightDiv2", -}; static const char *const max9867_filter_text[] = {"IIR", "FIR"}; static SOC_ENUM_SINGLE_DECL(max9867_filter, MAX9867_CODECFLTR, 7, max9867_filter_text); static SOC_ENUM_SINGLE_DECL(max9867_spkmode, MAX9867_MODECONFIG, 0, max9867_spmode); -static SOC_ENUM_SINGLE_DECL(max9867_sidetone, MAX9867_DACGAIN, 6, - max9867_sidetone_text); -static DECLARE_TLV_DB_SCALE(max9860_capture_tlv, -600, 200, 0); -static DECLARE_TLV_DB_SCALE(max9860_mic_tlv, 2000, 100, 1); -static DECLARE_TLV_DB_SCALE(max9860_adc_left_tlv, -1200, 100, 1); -static DECLARE_TLV_DB_SCALE(max9860_adc_right_tlv, -1200, 100, 1); -static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max98088_micboost_tlv, - 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), - 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max9867_master_tlv, + 0, 2, TLV_DB_SCALE_ITEM(-8600, 200, 1), + 3, 17, TLV_DB_SCALE_ITEM(-7800, 400, 0), + 18, 25, TLV_DB_SCALE_ITEM(-2000, 200, 0), + 26, 34, TLV_DB_SCALE_ITEM( -500, 100, 0), + 35, 40, TLV_DB_SCALE_ITEM( 350, 50, 0), +); +static DECLARE_TLV_DB_SCALE(max9867_mic_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_line_tlv, -600, 200, 0); +static DECLARE_TLV_DB_SCALE(max9867_adc_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_dac_tlv, -1500, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_dacboost_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max9867_micboost_tlv, + 0, 2, TLV_DB_SCALE_ITEM(-2000, 2000, 1), + 3, 3, TLV_DB_SCALE_ITEM(3000, 0, 0), ); static const struct snd_kcontrol_new max9867_snd_controls[] = { - SOC_DOUBLE_R("Master Playback Volume", MAX9867_LEFTVOL, - MAX9867_RIGHTVOL, 0, 63, 1), - SOC_DOUBLE_R_TLV("Capture Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, - 0, 15, 1, max9860_capture_tlv), - SOC_DOUBLE_R_TLV("Mic Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, 0, 31, 1, max9860_mic_tlv), - SOC_DOUBLE_R_TLV("Mic Boost Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, 5, 3, 0, max98088_micboost_tlv), - SOC_ENUM("Digital Sidetone Src", max9867_sidetone), - SOC_SINGLE("Sidetone Volume", MAX9867_DACGAIN, 0, 31, 1), - SOC_SINGLE("DAC Volume", MAX9867_DACLEVEL, 4, 3, 0), - SOC_SINGLE("DAC Attenuation", MAX9867_DACLEVEL, 0, 15, 1), - SOC_SINGLE_TLV("ADC Left Volume", MAX9867_ADCLEVEL, - 4, 15, 1, max9860_adc_left_tlv), - SOC_SINGLE_TLV("ADC Right Volume", MAX9867_ADCLEVEL, - 0, 15, 1, max9860_adc_right_tlv), + SOC_DOUBLE_R_TLV("Master Playback Volume", MAX9867_LEFTVOL, + MAX9867_RIGHTVOL, 0, 41, 1, max9867_master_tlv), + SOC_DOUBLE_R_TLV("Line Capture Volume", MAX9867_LEFTLINELVL, + MAX9867_RIGHTLINELVL, 0, 15, 1, max9867_line_tlv), + SOC_DOUBLE_R_TLV("Mic Capture Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 0, 20, 1, max9867_mic_tlv), + SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 5, 4, 0, max9867_micboost_tlv), + SOC_SINGLE("Digital Sidetone Volume", MAX9867_SIDETONE, 0, 31, 1), + SOC_SINGLE_TLV("Digital Playback Volume", MAX9867_DACLEVEL, 0, 15, 1, + max9867_dac_tlv), + SOC_SINGLE_TLV("Digital Boost Playback Volume", MAX9867_DACLEVEL, 4, 3, 0, + max9867_dacboost_tlv), + SOC_DOUBLE_TLV("Digital Capture Volume", MAX9867_ADCLEVEL, 0, 4, 15, 1, + max9867_adc_tlv), SOC_ENUM("Speaker Mode", max9867_spkmode), SOC_SINGLE("Volume Smoothing Switch", MAX9867_MODECONFIG, 6, 1, 0), - SOC_SINGLE("ZCD Switch", MAX9867_MODECONFIG, 5, 1, 0), + SOC_SINGLE("Line ZC Switch", MAX9867_MODECONFIG, 5, 1, 0), SOC_ENUM("DSP Filter", max9867_filter), }; -static const char *const max9867_mux[] = {"None", "Mic", "Line", "Mic_Line"}; +/* Input mixer */ +static const struct snd_kcontrol_new max9867_input_mixer_controls[] = { + SOC_DAPM_DOUBLE("Line Capture Switch", MAX9867_INPUTCONFIG, 7, 5, 1, 0), + SOC_DAPM_DOUBLE("Mic Capture Switch", MAX9867_INPUTCONFIG, 6, 4, 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), +}; -static SOC_ENUM_SINGLE_DECL(max9867_mux_enum, - MAX9867_INPUTCONFIG, MAX9867_INPUT_SHIFT, - max9867_mux); +/* Sidetone mixer */ +static const struct snd_kcontrol_new max9867_sidetone_mixer_controls[] = { + SOC_DAPM_DOUBLE("Sidetone Switch", MAX9867_SIDETONE, 6, 7, 1, 0), +}; -static const struct snd_kcontrol_new max9867_dapm_mux_controls = - SOC_DAPM_ENUM("Route", max9867_mux_enum); +/* 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_kcontrol_new max9867_left_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 6, 1, 0); -static const struct snd_kcontrol_new max9867_right_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 5, 1, 0); -static const struct snd_kcontrol_new max9867_line_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_LEFTLINELVL, 6, 1, 1); static const struct snd_soc_dapm_widget max9867_dapm_widgets[] = { - SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("Left DAC", NULL, MAX9867_PWRMAN, 3, 0), - SND_SOC_DAPM_DAC("Right DAC", NULL, MAX9867_PWRMAN, 2, 0), - SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_OUTPUT("HPOUT"), - - SND_SOC_DAPM_AIF_IN("DAI_IN", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture", MAX9867_PWRMAN, 1, 0), - SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", MAX9867_PWRMAN, 0, 0), - SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, - &max9867_dapm_mux_controls), - - SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("Left Line", MAX9867_LEFTLINELVL, 6, 1, - &max9867_left_dapm_control), - SND_SOC_DAPM_SWITCH("Right Line", MAX9867_RIGTHLINELVL, 6, 1, - &max9867_right_dapm_control), - SND_SOC_DAPM_SWITCH("Line Mixer", SND_SOC_NOPM, 0, 0, - &max9867_line_dapm_control), - SND_SOC_DAPM_INPUT("LINE_IN"), + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("LINL"), + SND_SOC_DAPM_INPUT("LINR"), + + SND_SOC_DAPM_PGA("Left Line Input", MAX9867_PWRMAN, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Line Input", MAX9867_PWRMAN, 5, 0, NULL, 0), + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Mixer", SND_SOC_NOPM, 0, 0, + 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_NAMED_CTL("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 DAC", NULL, "DAI_OUT"}, - {"Right DAC", NULL, "DAI_OUT"}, - {"Output Mixer", NULL, "Left DAC"}, - {"Output Mixer", NULL, "Right DAC"}, - {"HPOUT", NULL, "Output Mixer"}, - - {"Left ADC", NULL, "DAI_IN"}, - {"Right ADC", NULL, "DAI_IN"}, - {"Input Mixer", NULL, "Left ADC"}, - {"Input Mixer", NULL, "Right ADC"}, - {"Input Mux", "Line", "Input Mixer"}, - {"Input Mux", "Mic", "Input Mixer"}, - {"Input Mux", "Mic_Line", "Input Mixer"}, - {"Right Line", "Switch", "Input Mux"}, - {"Left Line", "Switch", "Input Mux"}, - {"LINE_IN", NULL, "Left Line"}, - {"LINE_IN", NULL, "Right Line"}, + {"Left Line Input", NULL, "LINL"}, + {"Right Line Input", NULL, "LINR"}, + {"Input Mixer", "Mic Capture Switch", "MICL"}, + {"Input Mixer", "Mic Capture Switch", "MICR"}, + {"Input Mixer", "Line Capture Switch", "Left Line Input"}, + {"Input Mixer", "Line Capture Switch", "Right Line Input"}, + {"ADCL", NULL, "Input Mixer"}, + {"ADCR", NULL, "Input Mixer"}, + + {"Digital", "Sidetone Switch", "ADCL"}, + {"Digital", "Sidetone Switch", "ADCR"}, + {"DACL", NULL, "Digital"}, + {"DACR", NULL, "Digital"}, + + {"Output Mixer", "Line Bypass Switch", "Left Line Input"}, + {"Output Mixer", "Line Bypass Switch", "Right Line Input"}, + {"Output Mixer", NULL, "DACL"}, + {"Output Mixer", NULL, "DACR"}, + {"Master Playback", "Switch", "Output Mixer"}, + {"LOUT", NULL, "Master Playback"}, + {"ROUT", NULL, "Master Playback"}, }; static const unsigned int max9867_rates_44k1[] = { @@ -235,13 +252,8 @@ static int max9867_mute(struct snd_soc_dai *dai, int mute) struct snd_soc_component *component = dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - if (mute) - regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, - MAX9867_DAC_MUTE_MASK, MAX9867_DAC_MUTE_MASK); - else - regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, - MAX9867_DAC_MUTE_MASK, 0); - return 0; + return regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, + 1 << 6, !!mute << 6); } static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, @@ -464,8 +476,8 @@ static const struct reg_default max9867_reg[] = { { 0x0B, 0x00 }, { 0x0C, 0x00 }, { 0x0D, 0x00 }, - { 0x0E, 0x00 }, - { 0x0F, 0x00 }, + { 0x0E, 0x40 }, + { 0x0F, 0x40 }, { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index 7f037ab701a5..2277798291a1 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -48,20 +48,16 @@ #define MAX9867_IFC1B_PCLK_8 0x06 #define MAX9867_IFC1B_PCLK_16 0x07 #define MAX9867_CODECFLTR 0x0a -#define MAX9867_DACGAIN 0x0b +#define MAX9867_SIDETONE 0x0b #define MAX9867_DACLEVEL 0x0c -#define MAX9867_DAC_MUTE_SHIFT 0x6 -#define MAX9867_DAC_MUTE_WIDTH 0x1 -#define MAX9867_DAC_MUTE_MASK (0x1<<MAX9867_DAC_MUTE_SHIFT) #define MAX9867_ADCLEVEL 0x0d #define MAX9867_LEFTLINELVL 0x0e -#define MAX9867_RIGTHLINELVL 0x0f +#define MAX9867_RIGHTLINELVL 0x0f #define MAX9867_LEFTVOL 0x10 #define MAX9867_RIGHTVOL 0x11 #define MAX9867_LEFTMICGAIN 0x12 #define MAX9867_RIGHTMICGAIN 0x13 #define MAX9867_INPUTCONFIG 0x14 -#define MAX9867_INPUT_SHIFT 0x6 #define MAX9867_MICCONFIG 0x15 #define MAX9867_MODECONFIG 0x16 #define MAX9867_PWRMAN 0x17 -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH 1/2] ASoC: Fix function return type
by Agrawal, Akshu 06 Dec '18

06 Dec '18

Function returns -1 on error and the return type should be int. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- include/sound/soc.h | 2 +- sound/soc/soc-io.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8ec1de8..591d743 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1341,7 +1341,7 @@ static inline int snd_soc_component_cache_sync( /* component IO */ int snd_soc_component_read(struct snd_soc_component *component, unsigned int reg, unsigned int *val); -unsigned int snd_soc_component_read32(struct snd_soc_component *component, +int snd_soc_component_read32(struct snd_soc_component *component, unsigned int reg); int snd_soc_component_write(struct snd_soc_component *component, unsigned int reg, unsigned int val); diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 1ff9175..d08f5b1 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -38,7 +38,7 @@ int snd_soc_component_read(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(snd_soc_component_read); -unsigned int snd_soc_component_read32(struct snd_soc_component *component, +int snd_soc_component_read32(struct snd_soc_component *component, unsigned int reg) { unsigned int val; -- 1.9.1

4 7

[alsa-devel] [PATCH] ASoC: adau1761: Ensure DSP_RUN and DSP_ENABLE are disabled
by Niklas Carlsson 06 Dec '18

06 Dec '18

DSP_RUN and DSP_ENABLE needs to be disabled during FW load. This is not a problem after power-cycling but for soft reboots. Signed-off-by: Niklas Carlsson <niklasc(a)axis.com> --- sound/soc/codecs/adau1761.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..3075c13bcbbc 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -460,8 +460,35 @@ static int adau1761_set_bias_level(struct snd_soc_component *component, regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN); - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + /* + * When going OFF -> STANDBY, the device can be in two + * states: + * 1) Power-cycled and reset + * 2) Soft reboot + * In the case of soft reboot, we need to sync the HW + * registers even if our regmap reports default values. + * Marking the cache as dirty ensures that both cases + * are handled. + */ + regcache_mark_dirty(adau->regmap); regcache_sync(adau->regmap); + /* + * In order for FW to load correctly, the device needs + * DSP_RUN and DSP_ENABLE to be 0. This might not be the + * case for soft reboots. Ensure that DSP_RUN and + * DSP_ENABLE is 0 by bypassing the cache and write + * directly to HW when going OFF -> STANDBY. + */ + if (adau17x1_has_dsp(adau)) { + regcache_cache_bypass(adau->regmap, true); + regmap_write(adau->regmap, ADAU17X1_DSP_RUN, 0); + regmap_write(adau->regmap, + ADAU17X1_DSP_ENABLE, 0); + regcache_cache_bypass(adau->regmap, false); + } + } break; case SND_SOC_BIAS_OFF: regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, -- 2.11.0

2 2

[alsa-devel] [PATCH] MAINTAINERS: add ASoC maintainers for sound dt-bindings
by Clément Péron 06 Dec '18

06 Dec '18

Sound dt-bindings are applied by ASoC maintainers and should be submit to ASoC list in addition to the devicetree list. Hence, add this information into the MAINTAINERS file. Signed-off-by: Clément Péron <peron.clem(a)gmail.com> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 0abecc528dac..62701e0135af 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -13930,6 +13930,7 @@ S: Supported F: Documentation/devicetree/bindings/sound/ F: Documentation/sound/soc/ F: sound/soc/ +F: include/dt-bindings/sound/ F: include/sound/soc* SOUNDWIRE SUBSYSTEM -- 2.19.1

2 2

[alsa-devel] [PATCH v2] ASoC: imx-audmux: complete dt-bindings for i.MX6
by Clément Péron 06 Dec '18

06 Dec '18

From: Colin Didier <colin.didier(a)devialet.com> The i.MX6 audmux differs from i.MX51. This patch adds the audmux for i.MX6 family. Signed-off-by: Colin Didier <colin.didier(a)devialet.com> Signed-off-by: Clément Péron <peron.clem(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- v2: Repost with ASoC maintainer and list include/dt-bindings/sound/fsl-imx-audmux.h | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/include/dt-bindings/sound/fsl-imx-audmux.h b/include/dt-bindings/sound/fsl-imx-audmux.h index 15f138bebe16..a1d0741d9ed1 100644 --- a/include/dt-bindings/sound/fsl-imx-audmux.h +++ b/include/dt-bindings/sound/fsl-imx-audmux.h @@ -25,6 +25,14 @@ #define MX51_AUDMUX_PORT6 5 #define MX51_AUDMUX_PORT7 6 +#define MX6_AUDMUX_PORT1_SSI1 0 +#define MX6_AUDMUX_PORT2_SSI2 1 +#define MX6_AUDMUX_PORT3 2 +#define MX6_AUDMUX_PORT4 3 +#define MX6_AUDMUX_PORT5 4 +#define MX6_AUDMUX_PORT6 5 +#define MX6_AUDMUX_PORT7_SSI3 6 + /* * TFCSEL/RFCSEL (i.MX27) or TFSEL/TCSEL/RFSEL/RCSEL (i.MX31/51/53/6Q) * can be sourced from Rx/Tx. -- 2.19.1

1 1

[alsa-devel] [PATCH] ALSA: Use of_node_name_eq for node name comparisons
by Rob Herring 06 Dec '18

06 Dec '18

Convert string compares of DT node names to use of_node_name_eq helper instead. This removes direct access to the node name pointer. A couple of open coded iterating thru the child node names are converted to use for_each_child_of_node() instead. Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Benjamin Herrenschmidt <benh(a)kernel.crashing.org> Cc: Paul Mackerras <paulus(a)samba.org> Cc: Michael Ellerman <mpe(a)ellerman.id.au> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/aoa/fabrics/layout.c | 2 +- sound/aoa/soundbus/i2sbus/core.c | 6 +++--- sound/ppc/pmac.c | 4 ++-- sound/ppc/tumbler.c | 4 ++-- 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 3a23fd032611..8797d42e2b76 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -776,7 +776,7 @@ static int check_codec(struct aoa_codec *codec, struct codec_connection *cc; /* if the codec has a 'codec' node, we require a reference */ - if (codec->node && (strcmp(codec->node->name, "codec") == 0)) { + if (of_node_name_eq(codec->node, "codec")) { snprintf(propname, sizeof(propname), "platform-%s-codec-ref", codec->name); ref = of_get_property(ldev->sound, propname, NULL); diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index bd7c5029fc59..c3f57a3fb1a5 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -154,7 +154,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct device_node *np) { struct i2sbus_dev *dev; - struct device_node *child = NULL, *sound = NULL; + struct device_node *child, *sound = NULL; struct resource *r; int i, layout = 0, rlen, ok = force; char node_name[6]; @@ -177,8 +177,8 @@ static int i2sbus_add_dev(struct macio_dev *macio, return 0; i = 0; - while ((child = of_get_next_child(np, child))) { - if (strcmp(child->name, "sound") == 0) { + for_each_child_of_node(np, child) { + if (of_node_name_eq(child, "sound")) { i++; sound = child; } diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index 48dd44f8e914..d692e4070167 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -908,7 +908,7 @@ static void detect_byte_swap(struct snd_pmac *chip) /* if seems that Keylargo can't byte-swap */ for (mio = chip->node->parent; mio; mio = mio->parent) { - if (strcmp(mio->name, "mac-io") == 0) { + if (of_node_name_eq(mio, "mac-io")) { if (of_device_is_compatible(mio, "Keylargo")) chip->can_byte_swap = 0; break; @@ -1313,7 +1313,7 @@ int snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return) } else if (chip->is_pbook_G3) { struct device_node* mio; for (mio = chip->node->parent; mio; mio = mio->parent) { - if (strcmp(mio->name, "mac-io") == 0) { + if (of_node_name_eq(mio, "mac-io")) { struct resource r; if (of_address_to_resource(mio, 0, &r) == 0) chip->macio_base = diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 0779a2912237..6d7ffffcce95 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -1365,8 +1365,8 @@ int snd_pmac_tumbler_init(struct snd_pmac *chip) mix->anded_reset = 0; mix->reset_on_sleep = 1; - for (np = chip->node->child; np; np = np->sibling) { - if (!strcmp(np->name, "sound")) { + for_each_child_of_node(chip->node, np) { + if (of_node_name_eq(np, "sound")) { if (of_get_property(np, "has-anded-reset", NULL)) mix->anded_reset = 1; if (of_get_property(np, "layout-id", NULL)) -- 2.19.1

2 1

[alsa-devel] [PATCH] ALSA: soundbus: Remove direct OF name and type accesses
by Rob Herring 06 Dec '18

06 Dec '18

Convert soundbus uevent and sysfs OF node name and device type usage to use printf specifier and helper functions instead of directly accessing the name and type pointers. This will allow the eventual removal of the pointers. Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/aoa/soundbus/core.c | 2 +- sound/aoa/soundbus/sysfs.c | 34 +++++++++++++++++++++------------- 2 files changed, 22 insertions(+), 14 deletions(-) diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c index f3076a3f7f2f..065d3a55725e 100644 --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -74,7 +74,7 @@ static int soundbus_uevent(struct device *dev, struct kobj_uevent_env *env) of = &soundbus_dev->ofdev; /* stuff we want to pass to /sbin/hotplug */ - retval = add_uevent_var(env, "OF_NAME=%s", of->dev.of_node->name); + retval = add_uevent_var(env, "OF_NAME=%pOFn", of->dev.of_node); if (retval) return retval; diff --git a/sound/aoa/soundbus/sysfs.c b/sound/aoa/soundbus/sysfs.c index 81da020bddef..a2d55e15afbb 100644 --- a/sound/aoa/soundbus/sysfs.c +++ b/sound/aoa/soundbus/sysfs.c @@ -1,18 +1,10 @@ // SPDX-License-Identifier: GPL-2.0 #include <linux/kernel.h> +#include <linux/of.h> #include <linux/stat.h> /* FIX UP */ #include "soundbus.h" -#define soundbus_config_of_attr(field, format_string) \ -static ssize_t \ -field##_show (struct device *dev, struct device_attribute *attr, \ - char *buf) \ -{ \ - struct soundbus_dev *mdev = to_soundbus_device (dev); \ - return sprintf (buf, format_string, mdev->ofdev.dev.of_node->field); \ -} - static ssize_t modalias_show(struct device *dev, struct device_attribute *attr, char *buf) { @@ -25,17 +17,33 @@ static ssize_t modalias_show(struct device *dev, struct device_attribute *attr, strcat(buf, "\n"); length = strlen(buf); } else { - length = sprintf(buf, "of:N%sT%s\n", - of->dev.of_node->name, of->dev.of_node->type); + length = sprintf(buf, "of:N%pOFn%c%s\n", + of->dev.of_node, 'T', + of_node_get_device_type(of->dev.of_node)); } return length; } static DEVICE_ATTR_RO(modalias); -soundbus_config_of_attr (name, "%s\n"); +static ssize_t name_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct soundbus_dev *sdev = to_soundbus_device(dev); + struct platform_device *of = &sdev->ofdev; + + return sprintf(buf, "%pOFn\n", of->dev.of_node); +} static DEVICE_ATTR_RO(name); -soundbus_config_of_attr (type, "%s\n"); + +static ssize_t type_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct soundbus_dev *sdev = to_soundbus_device(dev); + struct platform_device *of = &sdev->ofdev; + + return sprintf(buf, "%s\n", of_node_get_device_type(of->dev.of_node)); +} static DEVICE_ATTR_RO(type); struct attribute *soundbus_dev_attrs[] = { -- 2.19.1

2 1

[alsa-devel] Invalid TLS certificate for mailman.alsa-project.org
by Antonio Ospite 06 Dec '18

06 Dec '18

Hi, when logging in to mailman.alsa-project.org[1] to change my subscription options I noticed that the TLS certificate used for HTTPS connections does not cover said domain, firefox says: ----------------------------------------------------------------------- mailman.alsa-project.org uses an invalid security certificate. The certificate is only valid for the following names: alsa-project.org, alsa0.perex.cz, bugtrack.alsa-project.org, git.alsa-project.org, mail.alsa-project.org, www.alsa-project.org Error code: SSL_ERROR_BAD_CERT_DOMAIN ----------------------------------------------------------------------- Basically, mailman.alsa-project.org is not in Subject Alternative Names for the certificate. This is not a huge deal as: 1. mail.alsa-project.org works fine with HTTPS; 2. the password is sent by email in the monthly reminder anyways. However it would still be handy to have this fixed as mailman.alsa-project.org URLs are used in the Wiki[2] pages and in the footer of messages coming from the mailing list, and it can be surprising for users that HTTPS does not work. Thank you, Antonio [1] https://mailman.alsa-project.org/mailman/options/alsa-devel [2] https://www.alsa-project.org/main/index.php/Mailing-lists -- Antonio Ospite https://ao2.it https://twitter.com/ao2it A: Because it messes up the order in which people normally read text. See http://en.wikipedia.org/wiki/Posting_style Q: Why is top-posting such a bad thing?

1 0

[alsa-devel] [PATCH linux-next v3 6/7] ASoC: rsnd: add avb clocks
by Jiada Wang 06 Dec '18

06 Dec '18

There are AVB Counter Clocks in ADG, each clock has 12bits integral and 8 bits fractional dividers which operates with S0D1ϕ clock. This patch registers 8 AVB Counter Clocks when clock-cells of rcar_sound node is 2, Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/adg.c | 316 +++++++++++++++++++++++++++++++++++++-- sound/soc/sh/rcar/gen.c | 9 ++ sound/soc/sh/rcar/rsnd.h | 9 ++ 3 files changed, 325 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 6768a66588eb..0708be1f38d9 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -5,6 +5,7 @@ // Copyright (C) 2013 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> #include <linux/clk-provider.h> +#include <linux/of_address.h> #include "rsnd.h" #define CLKA 0 @@ -21,13 +22,30 @@ #define BRRx_MASK(x) (0x3FF & x) +#define AVB_CLK_NUM 8 +#define AVB_MAX_RATE 25000000 +#define AVB_DIV_EN_COM BIT(31) +#define AVB_MAX_DIV 0x3ffc0 + static struct rsnd_mod_ops adg_ops = { .name = "adg", }; +struct clk_avb { + struct clk_hw hw; + unsigned int idx; + struct rsnd_adg *adg; + /* lock reg access */ + spinlock_t *lock; +}; + +#define hw_to_avb(_hw) container_of(_hw, struct clk_avb, hw) + struct rsnd_adg { struct clk *clk[CLKMAX]; struct clk *clkout[CLKOUTMAX]; + struct clk *clkavb[AVB_CLK_NUM]; + struct clk *clkadg; struct clk_onecell_data onecell; struct rsnd_mod mod; u32 flags; @@ -37,6 +55,7 @@ struct rsnd_adg { int rbga_rate_for_441khz; /* RBGA */ int rbgb_rate_for_48khz; /* RBGB */ + spinlock_t avb_lock; }; #define LRCLK_ASYNC (1 << 0) @@ -408,6 +427,248 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) } } +static struct clk *rsnd_adg_clk_src_twocell_get(struct of_phandle_args *clkspec, + void *data) +{ + unsigned int clktype = clkspec->args[0]; + unsigned int clkidx = clkspec->args[1]; + struct rsnd_adg *adg = data; + struct rsnd_mod *adg_mod = rsnd_mod_get(adg); + struct rsnd_priv *priv = rsnd_mod_to_priv(adg_mod); + struct device *dev = rsnd_priv_to_dev(priv); + struct clk *clk; + + switch (clktype) { + case 0: + if (clkidx >= CLKOUTMAX) { + dev_err(dev, "Invalid fixed clock index %u\n", clkidx); + return ERR_PTR(-EINVAL); + } + clk = adg->clkout[clkidx]; + break; + case 1: + if (clkidx >= AVB_CLK_NUM) { + dev_err(dev, "Invalid avb clock index %u\n", clkidx); + return ERR_PTR(-EINVAL); + } + clk = adg->clkavb[clkidx]; + break; + default: + dev_err(dev, "Invalid ADG clock type %u\n", clktype); + return ERR_PTR(-EINVAL); + } + + return clk; +} + +static void clk_avb_div_write(struct rsnd_adg *adg, u32 data, unsigned int idx) +{ + struct rsnd_mod *mod = rsnd_mod_get(adg); + + switch (idx) { + case 0: + rsnd_mod_write(mod, AVB_CLK_DIV0, data); + break; + case 1: + rsnd_mod_write(mod, AVB_CLK_DIV1, data); + break; + case 2: + rsnd_mod_write(mod, AVB_CLK_DIV2, data); + break; + case 3: + rsnd_mod_write(mod, AVB_CLK_DIV3, data); + break; + case 4: + rsnd_mod_write(mod, AVB_CLK_DIV4, data); + break; + case 5: + rsnd_mod_write(mod, AVB_CLK_DIV5, data); + break; + case 6: + rsnd_mod_write(mod, AVB_CLK_DIV6, data); + break; + case 7: + rsnd_mod_write(mod, AVB_CLK_DIV7, data); + break; + } +} + +static u32 clk_avb_div_read(struct rsnd_adg *adg, unsigned int idx) +{ + struct rsnd_mod *mod = rsnd_mod_get(adg); + u32 val = 0; + + switch (idx) { + case 0: + val = rsnd_mod_read(mod, AVB_CLK_DIV0); + break; + case 1: + val = rsnd_mod_read(mod, AVB_CLK_DIV1); + break; + case 2: + val = rsnd_mod_read(mod, AVB_CLK_DIV2); + break; + case 3: + val = rsnd_mod_read(mod, AVB_CLK_DIV3); + break; + case 4: + val = rsnd_mod_read(mod, AVB_CLK_DIV4); + break; + case 5: + val = rsnd_mod_read(mod, AVB_CLK_DIV5); + break; + case 6: + val = rsnd_mod_read(mod, AVB_CLK_DIV6); + break; + case 7: + val = rsnd_mod_read(mod, AVB_CLK_DIV7); + break; + } + + return val; +} + +static int clk_avb_is_enabled(struct clk_hw *hw) +{ + struct clk_avb *avb = hw_to_avb(hw); + struct rsnd_mod *mod = rsnd_mod_get(avb->adg); + + return rsnd_mod_read(mod, AVB_CLK_CONFIG) & BIT(avb->idx); +} + +static int clk_avb_enabledisable(struct clk_hw *hw, int enable) +{ + struct clk_avb *avb = hw_to_avb(hw); + u32 val; + struct rsnd_mod *mod = rsnd_mod_get(avb->adg); + + spin_lock(avb->lock); + + val = rsnd_mod_read(mod, AVB_CLK_CONFIG); + if (enable) + val |= BIT(avb->idx); + else + val &= ~BIT(avb->idx); + rsnd_mod_write(mod, AVB_CLK_CONFIG, val); + + spin_unlock(avb->lock); + + return 0; +} + +static int clk_avb_enable(struct clk_hw *hw) +{ + return clk_avb_enabledisable(hw, 1); +} + +static void clk_avb_disable(struct clk_hw *hw) +{ + clk_avb_enabledisable(hw, 0); +} + +static unsigned long clk_avb_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct clk_avb *avb = hw_to_avb(hw); + u32 div; + + div = clk_avb_div_read(avb->adg, avb->idx); + if (!div) + return parent_rate; + + return parent_rate * 32 / div; +} + +static unsigned int clk_avb_calc_div(unsigned long rate, + unsigned long parent_rate) +{ + unsigned int div; + + if (!rate) + rate = 1; + + if (rate > AVB_MAX_RATE) + rate = AVB_MAX_RATE; + + div = DIV_ROUND_CLOSEST(parent_rate * 32, rate); + + if (div > AVB_MAX_DIV) + div = AVB_MAX_DIV; + + return div; +} + +static long clk_avb_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + unsigned int div = clk_avb_calc_div(rate, *parent_rate); + + return (*parent_rate * 32) / div; +} + +static int clk_avb_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct clk_avb *avb = hw_to_avb(hw); + unsigned int div = clk_avb_calc_div(rate, parent_rate); + u32 val; + + val = clk_avb_div_read(avb->adg, avb->idx); + clk_avb_div_write(avb->adg, val | div, avb->idx); + + return 0; +} + +static const struct clk_ops clk_avb_ops = { + .enable = clk_avb_enable, + .disable = clk_avb_disable, + .is_enabled = clk_avb_is_enabled, + .recalc_rate = clk_avb_recalc_rate, + .round_rate = clk_avb_round_rate, + .set_rate = clk_avb_set_rate, +}; + +static struct clk *clk_register_avb(struct device *dev, struct rsnd_adg *adg, + unsigned int id, spinlock_t *lock) +{ + struct clk_init_data init; + struct clk_avb *avb; + struct clk *clk; + char name[8]; + const char *parent_name; + + if (IS_ERR(adg->clkadg)) + return ERR_PTR(-ENODEV); + + avb = devm_kzalloc(dev, sizeof(*avb), GFP_KERNEL); + if (!avb) + return ERR_PTR(-ENOMEM); + + parent_name = __clk_get_name(adg->clkadg); + + snprintf(name, sizeof(name), "%s%u", "avb", id); + + avb->idx = id; + avb->lock = lock; + avb->adg = adg; + + /* Register the clock. */ + init.name = name; + init.ops = &clk_avb_ops; + init.flags = CLK_IS_BASIC; + init.parent_names = &parent_name; + init.num_parents = 1; + + avb->hw.init = &init; + + /* init DIV to a valid state */ + clk_avb_div_write(avb->adg, AVB_MAX_DIV, avb->idx); + + clk = devm_clk_register(dev, &avb->hw); + + return clk; +} + static void rsnd_adg_get_clkin(struct rsnd_priv *priv, struct rsnd_adg *adg) { @@ -436,6 +697,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, unsigned long req_48kHz_rate, req_441kHz_rate; int i, req_size; const char *parent_clk_name = NULL; + struct rsnd_mod *mod = rsnd_mod_get(adg); static const char * const clkout_name[] = { [CLKOUT] = "audio_clkout", [CLKOUT1] = "audio_clkout1", @@ -540,21 +802,23 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, */ of_property_read_u32(np, "#clock-cells", &count); - /* - * for clkout - */ - if (!count) { + + switch (count) { + case 0: + /* + * for clkout + */ clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], parent_clk_name, 0, req_rate[0]); if (!IS_ERR(clk)) { adg->clkout[CLKOUT] = clk; of_clk_add_provider(np, of_clk_src_simple_get, clk); } - } - /* - * for clkout0/1/2/3 - */ - else { + break; + case 1: + /* + * for clkout0/1/2/3 + */ for (i = 0; i < CLKOUTMAX; i++) { clk = clk_register_fixed_rate(dev, clkout_name[i], parent_clk_name, 0, @@ -566,6 +830,33 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, adg->onecell.clk_num = CLKOUTMAX; of_clk_add_provider(np, of_clk_src_onecell_get, &adg->onecell); + break; + case 2: + /* + * for clkout0/1/2/3 and avb clocks + */ + for (i = 0; i < CLKOUTMAX; i++) { + clk = clk_register_fixed_rate(dev, clkout_name[i], + parent_clk_name, 0, + req_rate[0]); + if (!IS_ERR(clk)) + adg->clkout[i] = clk; + } + + for (i = 0; i < AVB_CLK_NUM; i++) { + clk = clk_register_avb(dev, adg, i, &adg->avb_lock); + if (!IS_ERR(clk)) + adg->clkavb[i] = clk; + } + + of_clk_add_provider(np, rsnd_adg_clk_src_twocell_get, adg); + + rsnd_mod_write(mod, AVB_CLK_CONFIG, AVB_DIV_EN_COM); + + break; + default: + dev_err(dev, "Invalid clock-cell %d\n", count); + break; } rsnd_adg_get_clkout_end: @@ -606,17 +897,24 @@ int rsnd_adg_probe(struct rsnd_priv *priv) { struct rsnd_adg *adg; struct device *dev = rsnd_priv_to_dev(priv); + struct clk *clk; int ret; adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); if (!adg) return -ENOMEM; + spin_lock_init(&adg->avb_lock); + ret = rsnd_mod_init(priv, &adg->mod, &adg_ops, NULL, 0, 0); if (ret) return ret; + clk = devm_clk_get(dev, "adg"); + if (!IS_ERR(clk)) + adg->clkadg = clk; + rsnd_adg_get_clkin(priv, adg); rsnd_adg_get_clkout(priv, adg); rsnd_adg_clk_dbg_info(priv, adg); diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index ca639404f2cd..1b000d03b76e 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -355,6 +355,15 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_S_REG(SRCOUT_TIMSEL3, 0x54), RSND_GEN_S_REG(SRCOUT_TIMSEL4, 0x58), RSND_GEN_S_REG(CMDOUT_TIMSEL, 0x5c), + RSND_GEN_S_REG(AVB_CLK_DIV0, 0x11c), + RSND_GEN_S_REG(AVB_CLK_DIV1, 0x120), + RSND_GEN_S_REG(AVB_CLK_DIV2, 0x124), + RSND_GEN_S_REG(AVB_CLK_DIV3, 0x128), + RSND_GEN_S_REG(AVB_CLK_DIV4, 0x12c), + RSND_GEN_S_REG(AVB_CLK_DIV5, 0x130), + RSND_GEN_S_REG(AVB_CLK_DIV6, 0x134), + RSND_GEN_S_REG(AVB_CLK_DIV7, 0x138), + RSND_GEN_S_REG(AVB_CLK_CONFIG, 0x13c), }; static const struct rsnd_regmap_field_conf conf_ssi[] = { RSND_GEN_M_REG(SSICR, 0x00, 0x40), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 3c57129af6d1..d31b8a65985f 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -148,6 +148,15 @@ enum rsnd_reg { RSND_REG_AUDIO_CLK_SEL0, RSND_REG_AUDIO_CLK_SEL1, RSND_REG_AUDIO_CLK_SEL2, + RSND_REG_AVB_CLK_DIV0, + RSND_REG_AVB_CLK_DIV1, + RSND_REG_AVB_CLK_DIV2, + RSND_REG_AVB_CLK_DIV3, + RSND_REG_AVB_CLK_DIV4, + RSND_REG_AVB_CLK_DIV5, + RSND_REG_AVB_CLK_DIV6, + RSND_REG_AVB_CLK_DIV7, + RSND_REG_AVB_CLK_CONFIG, /* SSIU */ RSND_REG_SSI_MODE, -- 2.19.2

2 6

[alsa-devel] alsa-utils: axfer: add manuals
by GitHub pull_request - opened 05 Dec '18

05 Dec '18

alsa-project/alsa-utils pull request #6 was opened from takaswie: alsa-utils: axfer: add manuals This patchset adds manuals. As a result, users can refer to documents by man(1): ``` $ man 1 axfer $ man 1 axfer-list $ man 1 axfer-transfer ``` The manuals are written according to groff_man(7) and man-pages(7), however in some points they don't follow the guideline; e.g. usage of hyphen-minus ('-') without backslash '\' in pre-formatted text. I'll welcome any fixes. In this time, description of each backend in transfer subcommand is not yet added. It will be added in later work. ``` Takashi Sakamoto (5): axfer: add a manual for entry point axfer: add a manual for list subcommand axfer: add a manual for transfer subcommand axfer: add a section to describe compatibility to aplay(1) axfer: add a section about design of transfer subcommand axfer/Makefile.am | 10 + axfer/axfer-list.1 | 86 +++++++ axfer/axfer-transfer.1 | 500 +++++++++++++++++++++++++++++++++++++++++ axfer/axfer.1 | 132 +++++++++++ 4 files changed, 728 insertions(+) create mode 100644 axfer/axfer-list.1 create mode 100644 axfer/axfer-transfer.1 create mode 100644 axfer/axfer.1 ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/6 Patch URL : https://github.com/alsa-project/alsa-utils/pull/6.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH 1/2] ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier
by James Schulman 05 Dec '18

05 Dec '18

Add driver support for Cirrus Logic CS35L36 boosted speaker amplifier Signed-off-by: James Schulman <james.schulman(a)cirrus.com> --- include/sound/cs35l36.h | 58 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l36-tables.c | 315 +++++++ sound/soc/codecs/cs35l36.c | 1736 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l36.h | 451 ++++++++++ 6 files changed, 2567 insertions(+) create mode 100644 include/sound/cs35l36.h create mode 100644 sound/soc/codecs/cs35l36-tables.c create mode 100644 sound/soc/codecs/cs35l36.c create mode 100644 sound/soc/codecs/cs35l36.h diff --git a/include/sound/cs35l36.h b/include/sound/cs35l36.h new file mode 100644 index 0000000..6061e40 --- /dev/null +++ b/include/sound/cs35l36.h @@ -0,0 +1,58 @@ +/* + * linux/sound/cs35l36.h -- Platform data for CS35L36 + * + * Copyright (c) 2018 Cirrus Logic Inc. + * + * 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 __CS35L36_H +#define __CS35L36_H + +/* INT/GPIO pin configuration */ +struct irq_cfg { + bool is_present; + int irq_drv_sel; + int irq_pol; + int irq_gpio_sel; + int irq_out_en; + int irq_src_sel; +}; + +struct vpbr_cfg { + bool is_present; + bool vpbr_en; + int vpbr_thld; + int vpbr_atk_rate; + int vpbr_atk_vol; + int vpbr_max_attn; + int vpbr_wait; + int vpbr_rel_rate; + int vpbr_mute_en; +}; + +struct cs35l36_platform_data { + bool sclk_frc; + bool lrclk_frc; + bool multi_amp_mode; + bool dcm_mode; + int ldm_mode_sel; + bool amp_gain_zc; + bool amp_pcm_inv; + bool pdm_ldm_exit; + bool pdm_ldm_enter; + bool imon_pol_inv; + bool vmon_pol_inv; + int boost_ind; + int bst_vctl; + int bst_vctl_sel; + int bst_ipk; + bool extern_boost; + int temp_warn_thld; + struct vpbr_cfg vpbr_config; + struct irq_cfg irq_config; +}; + +#endif /* __CS35L36_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9cc4f18..77cb861 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -53,6 +53,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS35L33 if I2C select SND_SOC_CS35L34 if I2C select SND_SOC_CS35L35 if I2C + select SND_SOC_CS35L36 if I2C select SND_SOC_CS42L42 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT @@ -465,6 +466,10 @@ config SND_SOC_CS35L35 tristate "Cirrus Logic CS35L35 CODEC" depends on I2C +config SND_SOC_CS35L36 + tristate "Cirrus Logic CS35L36 CODEC" + depends on I2C + config SND_SOC_CS42L42 tristate "Cirrus Logic CS42L42 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8ffab8c..7a6009a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -45,6 +45,7 @@ snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o snd-soc-cs35l35-objs := cs35l35.o +snd-soc-cs35l36-objs := cs35l36.o cs35l36-tables.o snd-soc-cs42l42-objs := cs42l42.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o @@ -310,6 +311,7 @@ obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o obj-$(CONFIG_SND_SOC_CS35L35) += snd-soc-cs35l35.o +obj-$(CONFIG_SND_SOC_CS35L36) += snd-soc-cs35l36.o obj-$(CONFIG_SND_SOC_CS42L42) += snd-soc-cs42l42.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o diff --git a/sound/soc/codecs/cs35l36-tables.c b/sound/soc/codecs/cs35l36-tables.c new file mode 100644 index 0000000..4a93182 --- /dev/null +++ b/sound/soc/codecs/cs35l36-tables.c @@ -0,0 +1,315 @@ +/* + * cs35l36-tables.c -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.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. + * + */ + +#include "cs35l36.h" + +struct reg_default cs35l36_reg[CS35L36_MAX_CACHE_REG] = { + {CS35L36_TESTKEY_CTRL, 0x00000000}, + {CS35L36_USERKEY_CTL, 0x00000000}, + {CS35L36_OTP_CTRL1, 0x00002460}, + {CS35L36_OTP_CTRL2, 0x00000000}, + {CS35L36_OTP_CTRL3, 0x00000000}, + {CS35L36_OTP_CTRL4, 0x00000000}, + {CS35L36_OTP_CTRL5, 0x00000000}, + {CS35L36_PAC_CTL1, 0x00000004}, + {CS35L36_PAC_CTL2, 0x00000000}, + {CS35L36_PAC_CTL3, 0x00000000}, + {CS35L36_PWR_CTRL1, 0x00000000}, + {CS35L36_PWR_CTRL2, 0x00003321}, + {CS35L36_PWR_CTRL3, 0x01000010}, + {CS35L36_CTRL_OVRRIDE, 0x00000002}, + {CS35L36_AMP_OUT_MUTE, 0x00000000}, + {CS35L36_OTP_TRIM_STATUS, 0x00000000}, + {CS35L36_DISCH_FILT, 0x00000000}, + {CS35L36_PROTECT_REL_ERR, 0x00000000}, + {CS35L36_PAD_INTERFACE, 0x00000038}, + {CS35L36_PLL_CLK_CTRL, 0x00000010}, + {CS35L36_GLOBAL_CLK_CTRL, 0x00000003}, + {CS35L36_ADC_CLK_CTRL, 0x00000000}, + {CS35L36_SWIRE_CLK_CTRL, 0x00000000}, + {CS35L36_SP_SCLK_CLK_CTRL, 0x00000000}, + {CS35L36_MDSYNC_EN, 0x00000000}, + {CS35L36_MDSYNC_TX_ID, 0x00000000}, + {CS35L36_MDSYNC_PWR_CTRL, 0x00000000}, + {CS35L36_MDSYNC_DATA_TX, 0x00000000}, + {CS35L36_MDSYNC_TX_STATUS, 0x00000002}, + {CS35L36_MDSYNC_RX_STATUS, 0x00000000}, + {CS35L36_MDSYNC_ERR_STATUS, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL1, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL2, 0x00000001}, + {CS35L36_BSTCVRT_PEAK_CUR, 0x0000004A}, + {CS35L36_BSTCVRT_SFT_RAMP, 0x00000003}, + {CS35L36_BSTCVRT_COEFF, 0x00002424}, + {CS35L36_BSTCVRT_SLOPE_LBST, 0x00005800}, + {CS35L36_BSTCVRT_SW_FREQ, 0x00010000}, + {CS35L36_BSTCVRT_DCM_CTRL, 0x00002001}, + {CS35L36_BSTCVRT_DCM_MODE_FORCE, 0x00000000}, + {CS35L36_BSTCVRT_OVERVOLT_CTRL, 0x00000130}, + {CS35L36_VPI_LIMIT_MODE, 0x00000000}, + {CS35L36_VPI_LIMIT_MINMAX, 0x00003000}, + {CS35L36_VPI_VP_THLD, 0x00101010}, + {CS35L36_VPI_TRACK_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_MODE_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_STEPS, 0x00000000}, + {CS35L36_VI_SPKMON_FILT, 0x00000003}, + {CS35L36_VI_SPKMON_GAIN, 0x00000909}, + {CS35L36_VI_SPKMON_IP_SEL, 0x00000000}, + {CS35L36_DTEMP_WARN_THLD, 0x00000002}, + {CS35L36_DTEMP_STATUS, 0x00000000}, + {CS35L36_VPVBST_FS_SEL, 0x00000001}, + {CS35L36_VPVBST_VP_CTRL, 0x000001C0}, + {CS35L36_VPVBST_VBST_CTRL, 0x000001C0}, + {CS35L36_ASP_TX_PIN_CTRL, 0x00000028}, + {CS35L36_ASP_RATE_CTRL, 0x00090000}, + {CS35L36_ASP_FORMAT, 0x00000002}, + {CS35L36_ASP_FRAME_CTRL, 0x00180018}, + {CS35L36_ASP_TX1_TX2_SLOT, 0x00010000}, + {CS35L36_ASP_TX3_TX4_SLOT, 0x00030002}, + {CS35L36_ASP_TX5_TX6_SLOT, 0x00050004}, + {CS35L36_ASP_TX7_TX8_SLOT, 0x00070006}, + {CS35L36_ASP_RX1_SLOT, 0x00000000}, + {CS35L36_ASP_RX_TX_EN, 0x00000000}, + {CS35L36_ASP_RX1_SEL, 0x00000008}, + {CS35L36_ASP_TX1_SEL, 0x00000018}, + {CS35L36_ASP_TX2_SEL, 0x00000019}, + {CS35L36_ASP_TX3_SEL, 0x00000028}, + {CS35L36_ASP_TX4_SEL, 0x00000029}, + {CS35L36_ASP_TX5_SEL, 0x00000020}, + {CS35L36_ASP_TX6_SEL, 0x00000000}, + {CS35L36_SWIRE_P1_TX1_SEL, 0x00000018}, + {CS35L36_SWIRE_P1_TX2_SEL, 0x00000019}, + {CS35L36_SWIRE_P2_TX1_SEL, 0x00000028}, + {CS35L36_SWIRE_P2_TX2_SEL, 0x00000029}, + {CS35L36_SWIRE_P2_TX3_SEL, 0x00000020}, + {CS35L36_SWIRE_DP1_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP2_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP3_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_PCM_RX_DATA, 0x00000000}, + {CS35L36_SWIRE_FS_SEL, 0x00000001}, + {CS35L36_AMP_DIG_VOL_CTRL, 0x00008000}, + {CS35L36_VPBR_CFG, 0x02AA1905}, + {CS35L36_VBBR_CFG, 0x02AA1905}, + {CS35L36_VPBR_STATUS, 0x00000000}, + {CS35L36_VBBR_STATUS, 0x00000000}, + {CS35L36_OVERTEMP_CFG, 0x00000001}, + {CS35L36_AMP_ERR_VOL, 0x00000000}, + {CS35L36_CLASSH_CFG, 0x000B0405}, + {CS35L36_CLASSH_FET_DRV_CFG, 0x00000111}, + {CS35L36_NG_CFG, 0x00000033}, + {CS35L36_AMP_GAIN_CTRL, 0x00000273}, + {CS35L36_PWM_MOD_IO_CTRL, 0x00000000}, + {CS35L36_PWM_MOD_STATUS, 0x00000000}, + {CS35L36_DAC_MSM_CFG, 0x00000000}, + {CS35L36_AMP_SLOPE_CTRL, 0x00000B00}, + {CS35L36_AMP_PDM_VOLUME, 0x00000000}, + {CS35L36_AMP_PDM_RATE_CTRL, 0x00000000}, + {CS35L36_PDM_CH_SEL, 0x00000000}, + {CS35L36_AMP_NG_CTRL, 0x0000212F}, + {CS35L36_PDM_HIGHFILT_CTRL, 0x00000000}, + {CS35L36_PAC_INT0_CTRL, 0x00000001}, + {CS35L36_PAC_INT1_CTRL, 0x00000001}, + {CS35L36_PAC_INT2_CTRL, 0x00000001}, + {CS35L36_PAC_INT3_CTRL, 0x00000001}, + {CS35L36_PAC_INT4_CTRL, 0x00000001}, + {CS35L36_PAC_INT5_CTRL, 0x00000001}, + {CS35L36_PAC_INT6_CTRL, 0x00000001}, + {CS35L36_PAC_INT7_CTRL, 0x00000001}, +}; +EXPORT_SYMBOL_GPL(cs35l36_reg); + +bool cs35l36_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_OTP_MEM30: + case CS35L36_OTP_CTRL1: + case CS35L36_OTP_CTRL2: + case CS35L36_OTP_CTRL3: + case CS35L36_OTP_CTRL4: + case CS35L36_OTP_CTRL5: + case CS35L36_PAC_CTL1: + case CS35L36_PAC_CTL2: + case CS35L36_PAC_CTL3: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_PWR_CTRL1: + case CS35L36_PWR_CTRL2: + case CS35L36_PWR_CTRL3: + case CS35L36_CTRL_OVRRIDE: + case CS35L36_AMP_OUT_MUTE: + case CS35L36_OTP_TRIM_STATUS: + case CS35L36_DISCH_FILT: + case CS35L36_PROTECT_REL_ERR: + case CS35L36_PAD_INTERFACE: + case CS35L36_PLL_CLK_CTRL: + case CS35L36_GLOBAL_CLK_CTRL: + case CS35L36_ADC_CLK_CTRL: + case CS35L36_SWIRE_CLK_CTRL: + case CS35L36_SP_SCLK_CLK_CTRL: + case CS35L36_MDSYNC_EN: + case CS35L36_MDSYNC_TX_ID: + case CS35L36_MDSYNC_PWR_CTRL: + case CS35L36_MDSYNC_DATA_TX: + case CS35L36_MDSYNC_TX_STATUS: + case CS35L36_MDSYNC_RX_STATUS: + case CS35L36_MDSYNC_ERR_STATUS: + case CS35L36_BSTCVRT_VCTRL1: + case CS35L36_BSTCVRT_VCTRL2: + case CS35L36_BSTCVRT_PEAK_CUR: + case CS35L36_BSTCVRT_SFT_RAMP: + case CS35L36_BSTCVRT_COEFF: + case CS35L36_BSTCVRT_SLOPE_LBST: + case CS35L36_BSTCVRT_SW_FREQ: + case CS35L36_BSTCVRT_DCM_CTRL: + case CS35L36_BSTCVRT_DCM_MODE_FORCE: + case CS35L36_BSTCVRT_OVERVOLT_CTRL: + case CS35L36_BST_TST_MANUAL: + case CS35L36_BST_ANA2_TEST: + case CS35L36_VPI_LIMIT_MODE: + case CS35L36_VPI_LIMIT_MINMAX: + case CS35L36_VPI_VP_THLD: + case CS35L36_VPI_TRACK_CTRL: + case CS35L36_VPI_TRIG_MODE_CTRL: + case CS35L36_VPI_TRIG_STEPS: + case CS35L36_VI_SPKMON_FILT: + case CS35L36_VI_SPKMON_GAIN: + case CS35L36_VI_SPKMON_IP_SEL: + case CS35L36_DTEMP_WARN_THLD: + case CS35L36_DTEMP_STATUS: + case CS35L36_VPVBST_FS_SEL: + case CS35L36_VPVBST_VP_CTRL: + case CS35L36_VPVBST_VBST_CTRL: + case CS35L36_ASP_TX_PIN_CTRL: + case CS35L36_ASP_RATE_CTRL: + case CS35L36_ASP_FORMAT: + case CS35L36_ASP_FRAME_CTRL: + case CS35L36_ASP_TX1_TX2_SLOT: + case CS35L36_ASP_TX3_TX4_SLOT: + case CS35L36_ASP_TX5_TX6_SLOT: + case CS35L36_ASP_TX7_TX8_SLOT: + case CS35L36_ASP_RX1_SLOT: + case CS35L36_ASP_RX_TX_EN: + case CS35L36_ASP_RX1_SEL: + case CS35L36_ASP_TX1_SEL: + case CS35L36_ASP_TX2_SEL: + case CS35L36_ASP_TX3_SEL: + case CS35L36_ASP_TX4_SEL: + case CS35L36_ASP_TX5_SEL: + case CS35L36_ASP_TX6_SEL: + case CS35L36_SWIRE_P1_TX1_SEL: + case CS35L36_SWIRE_P1_TX2_SEL: + case CS35L36_SWIRE_P2_TX1_SEL: + case CS35L36_SWIRE_P2_TX2_SEL: + case CS35L36_SWIRE_P2_TX3_SEL: + case CS35L36_SWIRE_DP1_FIFO_CFG: + case CS35L36_SWIRE_DP2_FIFO_CFG: + case CS35L36_SWIRE_DP3_FIFO_CFG: + case CS35L36_SWIRE_PCM_RX_DATA: + case CS35L36_SWIRE_FS_SEL: + case CS35L36_AMP_DIG_VOL_CTRL: + case CS35L36_VPBR_CFG: + case CS35L36_VBBR_CFG: + case CS35L36_VPBR_STATUS: + case CS35L36_VBBR_STATUS: + case CS35L36_OVERTEMP_CFG: + case CS35L36_AMP_ERR_VOL: + case CS35L36_CLASSH_CFG: + case CS35L36_CLASSH_FET_DRV_CFG: + case CS35L36_NG_CFG: + case CS35L36_AMP_GAIN_CTRL: + case CS35L36_PWM_MOD_IO_CTRL: + case CS35L36_PWM_MOD_STATUS: + case CS35L36_DAC_MSM_CFG: + case CS35L36_AMP_SLOPE_CTRL: + case CS35L36_AMP_PDM_VOLUME: + case CS35L36_AMP_PDM_RATE_CTRL: + case CS35L36_PDM_CH_SEL: + case CS35L36_AMP_NG_CTRL: + case CS35L36_PDM_HIGHFILT_CTRL: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + case CS35L36_PAC_INT0_CTRL: + case CS35L36_PAC_INT1_CTRL: + case CS35L36_PAC_INT2_CTRL: + case CS35L36_PAC_INT3_CTRL: + case CS35L36_PAC_INT4_CTRL: + case CS35L36_PAC_INT5_CTRL: + case CS35L36_PAC_INT6_CTRL: + case CS35L36_PAC_INT7_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} +EXPORT_SYMBOL_GPL(cs35l36_readable_reg); + +bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} +EXPORT_SYMBOL_GPL(cs35l36_volatile_reg); diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c new file mode 100644 index 0000000..4fa1531 --- /dev/null +++ b/sound/soc/codecs/cs35l36.c @@ -0,0 +1,1736 @@ +/* + * cs35l36.c -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.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. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs35l36.h> +#include <linux/of_irq.h> +#include <linux/completion.h> + +#include "cs35l36.h" + +/* + * Some fields take zero as a valid value so use a high bit flag that won't + * get written to the device to mark those. + */ +#define CS35L36_VALID_PDATA 0x80000000 + + +static const char * const cs35l36_supplies[] = { + "VA", + "VP", +}; + +struct cs35l36_private { + struct device *dev; + struct cs35l36_platform_data pdata; + struct regmap *regmap; + struct regulator_bulk_data supplies[2]; + int num_supplies; + int clksrc; + int prev_clksrc; + int extclk_freq; + int extclk_cfg; + int fll_igain; + int sclk; + int chip_version; + int rev_id; + struct gpio_desc *reset_gpio; + struct completion global_pup_done; + struct completion global_pdn_done; +}; + +struct cs35l36_pll_sysclk_config { + int freq; + int clk_cfg; + int fll_igain; +}; + +static const struct cs35l36_pll_sysclk_config cs35l36_pll_sysclk[] = { + {32768, 0x00, 0x05}, + {8000, 0x01, 0x03}, + {11025, 0x02, 0x03}, + {12000, 0x03, 0x03}, + {16000, 0x04, 0x04}, + {22050, 0x05, 0x04}, + {24000, 0x06, 0x04}, + {32000, 0x07, 0x05}, + {44100, 0x08, 0x05}, + {48000, 0x09, 0x05}, + {88200, 0x0A, 0x06}, + {96000, 0x0B, 0x06}, + {128000, 0x0C, 0x07}, + {176400, 0x0D, 0x07}, + {192000, 0x0E, 0x07}, + {256000, 0x0F, 0x08}, + {352800, 0x10, 0x08}, + {384000, 0x11, 0x08}, + {512000, 0x12, 0x09}, + {705600, 0x13, 0x09}, + {750000, 0x14, 0x09}, + {768000, 0x15, 0x09}, + {1000000, 0x16, 0x0A}, + {1024000, 0x17, 0x0A}, + {1200000, 0x18, 0x0A}, + {1411200, 0x19, 0x0A}, + {1500000, 0x1A, 0x0A}, + {1536000, 0x1B, 0x0A}, + {2000000, 0x1C, 0x0A}, + {2048000, 0x1D, 0x0A}, + {2400000, 0x1E, 0x0A}, + {2822400, 0x1F, 0x0A}, + {3000000, 0x20, 0x0A}, + {3072000, 0x21, 0x0A}, + {3200000, 0x22, 0x0A}, + {4000000, 0x23, 0x0A}, + {4096000, 0x24, 0x0A}, + {4800000, 0x25, 0x0A}, + {5644800, 0x26, 0x0A}, + {6000000, 0x27, 0x0A}, + {6144000, 0x28, 0x0A}, + {6250000, 0x29, 0x08}, + {6400000, 0x2A, 0x0A}, + {6500000, 0x2B, 0x08}, + {6750000, 0x2C, 0x09}, + {7526400, 0x2D, 0x0A}, + {8000000, 0x2E, 0x0A}, + {8192000, 0x2F, 0x0A}, + {9600000, 0x30, 0x0A}, + {11289600, 0x31, 0x0A}, + {12000000, 0x32, 0x0A}, + {12288000, 0x33, 0x0A}, + {12500000, 0x34, 0x08}, + {12800000, 0x35, 0x0A}, + {13000000, 0x36, 0x0A}, + {13500000, 0x37, 0x0A}, + {19200000, 0x38, 0x0A}, + {22579200, 0x39, 0x0A}, + {24000000, 0x3A, 0x0A}, + {24576000, 0x3B, 0x0A}, + {25000000, 0x3C, 0x0A}, + {25600000, 0x3D, 0x0A}, + {26000000, 0x3E, 0x0A}, + {27000000, 0x3F, 0x0A}, +}; + +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); + +static const char * const cs35l36_pcm_sftramp_text[] = { + "Off", ".5ms", "1ms", "2ms", "4ms", "8ms", "15ms", "30ms"}; + +static SOC_ENUM_SINGLE_DECL(pcm_sft_ramp, + CS35L36_AMP_DIG_VOL_CTRL, 0, + cs35l36_pcm_sftramp_text); + +static const struct snd_kcontrol_new cs35l36_aud_controls[] = { + SOC_SINGLE_SX_TLV("Digital PCM Volume", CS35L36_AMP_DIG_VOL_CTRL, + 3, 0x4D0, 0x390, dig_vol_tlv), + SOC_SINGLE_TLV("AMP PCM Gain", CS35L36_AMP_GAIN_CTRL, 5, 0x13, 0, + amp_gain_tlv), + SOC_ENUM("PCM Soft Ramp", pcm_sft_ramp), +}; + +static int cs35l36_main_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + u32 reg; + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 1 << CS35L36_GLOBAL_EN_SHIFT); + usleep_range(2000, 2100); + + regmap_read(cs35l36->regmap, CS35L36_INT4_RAW_STATUS, &reg); + if (reg & CS35L36_PLL_UNLOCK_MASK) + dev_crit(cs35l36->dev, "PLL Unlocked\n"); + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_PCM); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 0 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_ZERO); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 1 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 0 << CS35L36_GLOBAL_EN_SHIFT); + usleep_range(2000, 2100); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + } + return ret; +} + +static int cs35l36_boost_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_EN << + CS35L36_BST_EN_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_DIS_VP << + CS35L36_BST_EN_SHIFT); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + } + return 0; +} +static const char * const cs35l36_chan_text[] = { + "RX1", + "RX2", +}; + +static SOC_ENUM_SINGLE_DECL(chansel_enum, CS35L36_ASP_RX1_SLOT, 0, + cs35l36_chan_text); + +static const struct snd_kcontrol_new cs35l36_chan_mux[] = { + SOC_DAPM_ENUM("Input Mux", chansel_enum), +}; + +static const char * const cs35l36_boost_text[] = { + "On", + "Off", +}; + +static SOC_ENUM_SINGLE_DECL(boost_enum, SND_SOC_NOPM, 0, + cs35l36_boost_text); + +static const struct snd_kcontrol_new cs35l36_boost_mux[] = { + SOC_DAPM_ENUM("Boost Enable", boost_enum), +}; + +static const struct snd_kcontrol_new amp_enable_ctrl = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_SHIFT, 1, 1); + +static const char * const asp_tx_src_text[] = { + "Zero Fill", "ASPRX1", "VMON", "IMON", + "ERRVOL", "VPMON", "VBSTMON" +}; + +static const unsigned int asp_tx_src_values[] = { + 0x00, 0x08, 0x18, 0x19, 0x20, 0x28, 0x29 +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx1_src_enum, + CS35L36_ASP_TX1_SEL, 0, + CS35L36_APS_TX_SEL_MASK, + asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx1_src = + SOC_DAPM_ENUM("ASPTX1SRC", asp_tx1_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx2_src_enum, + CS35L36_ASP_TX2_SEL, 0, + CS35L36_APS_TX_SEL_MASK, + asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx2_src = + SOC_DAPM_ENUM("ASPTX2SRC", asp_tx2_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx3_src_enum, + CS35L36_ASP_TX3_SEL, 0, + CS35L36_APS_TX_SEL_MASK, + asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx3_src = + SOC_DAPM_ENUM("ASPTX3SRC", asp_tx3_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx4_src_enum, + CS35L36_ASP_TX4_SEL, 0, + CS35L36_APS_TX_SEL_MASK, + asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx4_src = + SOC_DAPM_ENUM("ASPTX4SRC", asp_tx4_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx5_src_enum, + CS35L36_ASP_TX5_SEL, 0, + CS35L36_APS_TX_SEL_MASK, + asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx5_src = + SOC_DAPM_ENUM("ASPTX5SRC", asp_tx5_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx6_src_enum, + CS35L36_ASP_TX6_SEL, 0, + CS35L36_APS_TX_SEL_MASK, + asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx6_src = + SOC_DAPM_ENUM("ASPTX6SRC", asp_tx6_src_enum); + +static const struct snd_soc_dapm_widget cs35l36_dapm_widgets[] = { + + SND_SOC_DAPM_MUX("Channel Mux", SND_SOC_NOPM, 0, 0, cs35l36_chan_mux), + SND_SOC_DAPM_AIF_IN("SDIN", NULL, 0, CS35L36_ASP_RX_TX_EN, 16, 0), + + SND_SOC_DAPM_OUT_DRV_E("Main AMP", CS35L36_PWR_CTRL2, 0, 0, NULL, 0, + cs35l36_main_amp_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_SWITCH("AMP Enable", SND_SOC_NOPM, 0, 1, &amp_enable_ctrl), + SND_SOC_DAPM_MIXER("CLASS H", CS35L36_PWR_CTRL3, 4, 0, NULL, 0), + SND_SOC_DAPM_MUX_E("BOOST Mux", SND_SOC_NOPM, 0, 0, cs35l36_boost_mux, + cs35l36_boost_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_AIF_OUT("ASPTX1", NULL, 0, CS35L36_ASP_RX_TX_EN, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX2", NULL, 1, CS35L36_ASP_RX_TX_EN, 1, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX3", NULL, 2, CS35L36_ASP_RX_TX_EN, 2, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX4", NULL, 3, CS35L36_ASP_RX_TX_EN, 3, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX5", NULL, 4, CS35L36_ASP_RX_TX_EN, 4, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX6", NULL, 5, CS35L36_ASP_RX_TX_EN, 5, 0), + + SND_SOC_DAPM_MUX("ASPTX1SRC", SND_SOC_NOPM, 0, 0, &asp_tx1_src), + SND_SOC_DAPM_MUX("ASPTX2SRC", SND_SOC_NOPM, 0, 0, &asp_tx2_src), + SND_SOC_DAPM_MUX("ASPTX3SRC", SND_SOC_NOPM, 0, 0, &asp_tx3_src), + SND_SOC_DAPM_MUX("ASPTX4SRC", SND_SOC_NOPM, 0, 0, &asp_tx4_src), + SND_SOC_DAPM_MUX("ASPTX5SRC", SND_SOC_NOPM, 0, 0, &asp_tx5_src), + SND_SOC_DAPM_MUX("ASPTX6SRC", SND_SOC_NOPM, 0, 0, &asp_tx6_src), + + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L36_PWR_CTRL2, 12, 0), + SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L36_PWR_CTRL2, 13, 0), + SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L36_PWR_CTRL2, 8, 0), + SND_SOC_DAPM_ADC("VBSTMON ADC", NULL, CS35L36_PWR_CTRL2, 9, 0), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("VBST"), + SND_SOC_DAPM_INPUT("VSENSE"), +}; + +static const struct snd_soc_dapm_route cs35l36_audio_map[] = { + + {"VPMON ADC", NULL, "VP"}, + {"VBSTMON ADC", NULL, "VBST"}, + {"IMON ADC", NULL, "VSENSE"}, + {"VMON ADC", NULL, "VSENSE"}, + + {"ASPTX1SRC", "IMON", "IMON ADC"}, + {"ASPTX1SRC", "VMON", "VMON ADC"}, + {"ASPTX1SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX1SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX2SRC", "IMON", "IMON ADC"}, + {"ASPTX2SRC", "VMON", "VMON ADC"}, + {"ASPTX2SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX2SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX3SRC", "IMON", "IMON ADC"}, + {"ASPTX3SRC", "VMON", "VMON ADC"}, + {"ASPTX3SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX3SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX4SRC", "IMON", "IMON ADC"}, + {"ASPTX4SRC", "VMON", "VMON ADC"}, + {"ASPTX4SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX4SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX5SRC", "IMON", "IMON ADC"}, + {"ASPTX5SRC", "VMON", "VMON ADC"}, + {"ASPTX5SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX5SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX6SRC", "IMON", "IMON ADC"}, + {"ASPTX6SRC", "VMON", "VMON ADC"}, + {"ASPTX6SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX6SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX1", NULL, "ASPTX1SRC"}, + {"ASPTX2", NULL, "ASPTX2SRC"}, + {"ASPTX3", NULL, "ASPTX3SRC"}, + {"ASPTX4", NULL, "ASPTX4SRC"}, + {"ASPTX5", NULL, "ASPTX5SRC"}, + {"ASPTX6", NULL, "ASPTX6SRC"}, + + {"AMP Capture", NULL, "ASPTX1"}, + {"AMP Capture", NULL, "ASPTX2"}, + {"AMP Capture", NULL, "ASPTX3"}, + {"AMP Capture", NULL, "ASPTX4"}, + {"AMP Capture", NULL, "ASPTX5"}, + {"AMP Capture", NULL, "ASPTX6"}, + + {"AMP Enable", "Switch", "AMP Playback"}, + {"SDIN", NULL, "AMP Enable"}, + {"Channel Mux", "RX1", "SDIN"}, + {"Channel Mux", "RX2", "SDIN"}, + {"BOOST Mux", "On", "Channel Mux"}, + {"CLASS H", NULL, "BOOST Mux"}, + {"Main AMP", NULL, "Channel Mux"}, + {"Main AMP", NULL, "CLASS H"}, + {"SPK", NULL, "Main AMP"}, +}; + +static int cs35l36_set_dai_fmt(struct snd_soc_dai *component_dai, + unsigned int fmt) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component_dai->component); + unsigned int asp_fmt, lrclk_fmt, sclk_fmt, slave_mode; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + slave_mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + slave_mode = 0; + break; + default: + return -EINVAL; + } + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_MSTR_MASK, + slave_mode << CS35L36_SCLK_MSTR_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_MSTR_MASK, + slave_mode << CS35L36_LRCLK_MSTR_SHIFT); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + asp_fmt = 0; + break; + case SND_SOC_DAIFMT_I2S: + asp_fmt = 2; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + lrclk_fmt = 1; + sclk_fmt = 0; + break; + case SND_SOC_DAIFMT_IB_NF: + lrclk_fmt = 0; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + lrclk_fmt = 1; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_NB_NF: + lrclk_fmt = 0; + sclk_fmt = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_INV_MASK, + lrclk_fmt << CS35L36_LRCLK_INV_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_INV_MASK, + sclk_fmt << CS35L36_SCLK_INV_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FORMAT, + CS35L36_ASP_FMT_MASK, asp_fmt); + + return 0; +} + +struct cs35l36_global_fs_config { + int rate; + int fs_cfg; +}; + +static const struct cs35l36_global_fs_config cs35l36_fs_rates[] = { + {12000, 0x01}, + {24000, 0x02}, + {48000, 0x03}, + {96000, 0x04}, + {192000, 0x05}, + {384000, 0x06}, + {11025, 0x09}, + {22050, 0x0A}, + {44100, 0x0B}, + {88200, 0x0C}, + {176400, 0x0D}, + {8000, 0x11}, + {16000, 0x12}, + {32000, 0x13}, +}; + +static int cs35l36_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(dai->component); + int i; + unsigned int global_fs = params_rate(params); + unsigned int asp_width; + + for (i = 0; i < ARRAY_SIZE(cs35l36_fs_rates); i++) { + if (global_fs == cs35l36_fs_rates[i].rate) + regmap_update_bits(cs35l36->regmap, + CS35L36_GLOBAL_CLK_CTRL, + CS35L36_GLOBAL_FS_MASK, + cs35l36_fs_rates[i].fs_cfg << + CS35L36_GLOBAL_FS_SHIFT); + } + + switch (params_width(params)) { + case 16: + asp_width = CS35L36_ASP_WIDTH_16; + break; + case 24: + asp_width = CS35L36_ASP_WIDTH_24; + break; + case 32: + asp_width = CS35L36_ASP_WIDTH_32; + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_RX_WIDTH_MASK, + asp_width << CS35L36_ASP_RX_WIDTH_SHIFT); + } else { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_TX_WIDTH_MASK, + asp_width << CS35L36_ASP_TX_WIDTH_SHIFT); + } + + return 0; +} + +static int cs35l36_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int fs1_val = 0; + int fs2_val = 0; + + /* Need the SCLK Frequency regardless of sysclk source */ + cs35l36->sclk = freq; + + if (cs35l36->sclk > 6000000) { + fs1_val = 3 * 4 + 4; + fs2_val = 8 * 4 + 4; + } + + if (cs35l36->sclk <= 6000000) { + fs1_val = 3 * ((24000000 + cs35l36->sclk - 1) / cs35l36->sclk) + + 4; + fs2_val = 5 * ((24000000 + cs35l36->sclk - 1) / cs35l36->sclk) + + 4; + } + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + regmap_update_bits(cs35l36->regmap, CS35L36_TST_FS_MON0, + CS35L36_FS1_WINDOW_MASK, fs1_val); + regmap_update_bits(cs35l36->regmap, CS35L36_TST_FS_MON0, + CS35L36_FS2_WINDOW_MASK, fs2_val << + CS35L36_FS2_WINDOW_SHIFT); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + return 0; +} + +static int cs35l36_get_clk_config(struct cs35l36_private *cs35l36, int freq) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l36_pll_sysclk); i++) { + if (cs35l36_pll_sysclk[i].freq == freq) { + cs35l36->extclk_cfg = cs35l36_pll_sysclk[i].clk_cfg; + cs35l36->fll_igain = cs35l36_pll_sysclk[i].fll_igain; + return i; + } + } + + return -EINVAL; +} + +static const unsigned int cs35l36_src_rates[] = { + 8000, 12000, 11025, 16000, 22050, 24000, 32000, + 44100, 48000, 88200, 96000, 176400, 192000, 384000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l36_constraints = { + .count = ARRAY_SIZE(cs35l36_src_rates), + .list = cs35l36_src_rates, +}; + +static int cs35l36_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + if (!substream->runtime) + return 0; + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs35l36_constraints); + return 0; +} + +static const struct snd_soc_dai_ops cs35l36_ops = { + .startup = cs35l36_pcm_startup, + .set_fmt = cs35l36_set_dai_fmt, + .hw_params = cs35l36_pcm_hw_params, + .set_sysclk = cs35l36_dai_set_sysclk, +}; + +static struct snd_soc_dai_driver cs35l36_dai[] = { + { + .name = "cs35l36-pcm", + .id = 0, + .playback = { + .stream_name = "AMP Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_RX_FORMATS, + }, + .capture = { + .stream_name = "AMP Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_TX_FORMATS, + }, + .ops = &cs35l36_ops, + .symmetric_rates = 1, + }, +}; + +static int cs35l36_component_set_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, + int dir) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int ret; + + cs35l36->extclk_freq = freq; + cs35l36->prev_clksrc = cs35l36->clksrc; + + switch (clk_id) { + case 0: + cs35l36->clksrc = CS35L36_PLLSRC_SCLK; + break; + case 1: + cs35l36->clksrc = CS35L36_PLLSRC_LRCLK; + break; + case 2: + cs35l36->clksrc = CS35L36_PLLSRC_PDMCLK; + break; + case 3: + cs35l36->clksrc = CS35L36_PLLSRC_SELF; + break; + case 4: + cs35l36->clksrc = CS35L36_PLLSRC_MCLK; + break; + default: + return -EINVAL; + } + + ret = cs35l36_get_clk_config(cs35l36, freq); + + if (ret < 0) { + dev_err(component->dev, + "Invalid CLK Config Freq: %d\n", freq); + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 1 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_REFCLK_FREQ_MASK, + cs35l36->extclk_cfg << CS35L36_REFCLK_FREQ_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 0 << CS35L36_PLL_REFCLK_EN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_CLK_SEL_MASK, cs35l36->clksrc); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 0 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 1 << CS35L36_PLL_REFCLK_EN_SHIFT); + + if (cs35l36->rev_id == CS35L36_REV_A0) { + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + regmap_write(cs35l36->regmap, CS35L36_DCO_CTRL, 0x00036DA8); + regmap_write(cs35l36->regmap, CS35L36_MISC_CTRL, 0x0100EE0E); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_IGAIN_MASK, + CS35L36_PLL_IGAIN << + CS35L36_PLL_IGAIN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_FFL_IGAIN_MASK, + cs35l36->fll_igain); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + if (cs35l36->clksrc == CS35L36_PLLSRC_PDMCLK) { + if (cs35l36->pdata.ldm_mode_sel) { + if (cs35l36->prev_clksrc != CS35L36_PLLSRC_PDMCLK) + regmap_update_bits(cs35l36->regmap, + CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + } + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 1 << CS35L36_PDM_MODE_SHIFT); + if (cs35l36->pdata.ldm_mode_sel) { + if (cs35l36->prev_clksrc != CS35L36_PLLSRC_PDMCLK) { + regmap_update_bits(cs35l36->regmap, + CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } + } + } else { + if (cs35l36->pdata.ldm_mode_sel) { + if (cs35l36->prev_clksrc == CS35L36_PLLSRC_PDMCLK) + regmap_update_bits(cs35l36->regmap, + CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + } + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 0 << CS35L36_PDM_MODE_SHIFT); + if (cs35l36->pdata.ldm_mode_sel) { + if (cs35l36->prev_clksrc == CS35L36_PLLSRC_PDMCLK) { + regmap_update_bits(cs35l36->regmap, + CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } + } + } + + return 0; +} + +static int cs35l36_boost_inductor(struct cs35l36_private *cs35l36, int inductor) +{ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K1_MASK, 0x3C); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K2_MASK, + 0x3C << CS35L36_BSTCVRT_K2_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SW_FREQ, + CS35L36_BSTCVRT_CCMFREQ_MASK, 0x00); + + switch (inductor) { + case 1000: /* 1 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x75 << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x00); + break; + case 1200: /* 1.2 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x6B << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x01); + break; + default: + dev_err(cs35l36->dev, "%s Invalid Inductor Value %d uH\n", + __func__, inductor); + return -EINVAL; + } + return 0; +} + +static int cs35l36_component_probe(struct snd_soc_component *component) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + if (cs35l36->pdata.sclk_frc) + regmap_update_bits(cs35l36->regmap, + CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_FRC_MASK, + cs35l36->pdata.sclk_frc << + CS35L36_SCLK_FRC_SHIFT); + + if (cs35l36->pdata.lrclk_frc) + regmap_update_bits(cs35l36->regmap, + CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_FRC_MASK, + cs35l36->pdata.lrclk_frc << + CS35L36_LRCLK_FRC_SHIFT); + + if (cs35l36->rev_id == CS35L36_REV_A0) { + if (cs35l36->pdata.dcm_mode) { + regmap_update_bits(cs35l36->regmap, + CS35L36_BSTCVRT_DCM_CTRL, + CS35L36_DCM_AUTO_MASK, + CS35L36_DCM_AUTO_MASK); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + regmap_update_bits(cs35l36->regmap, + CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_MASK, + 0 << CS35L36_BST_MAN_IPKCOMP_SHIFT); + regmap_update_bits(cs35l36->regmap, + CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_EN_MASK, + CS35L36_BST_MAN_IPKCOMP_EN_MASK); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + } + + if (cs35l36->pdata.amp_gain_zc) + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_GAIN_CTRL, + CS35L36_AMP_ZC_MASK, + CS35L36_AMP_ZC_MASK); + + if (cs35l36->pdata.amp_pcm_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_DIG_VOL_CTRL, + CS35L36_AMP_PCM_INV_MASK, + CS35L36_AMP_PCM_INV_MASK); + + if (cs35l36->pdata.ldm_mode_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_AMP_EN_MASK, + CS35L36_NG_AMP_EN_MASK); + + if (cs35l36->pdata.multi_amp_mode) + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_ASP_TX_HIZ_MASK, + CS35L36_ASP_TX_HIZ_MASK); + + if (cs35l36->pdata.pdm_ldm_enter) + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_ENTER_MASK, + CS35L36_PDM_LDM_ENTER_MASK); + + if (cs35l36->pdata.pdm_ldm_exit) + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_EXIT_MASK, + CS35L36_PDM_LDM_EXIT_MASK); + + if (cs35l36->pdata.imon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_IMON_POL_MASK, 0); + + if (cs35l36->pdata.vmon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_VMON_POL_MASK, 0); + + if (cs35l36->pdata.bst_vctl) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + cs35l36->pdata.bst_vctl); + + if (cs35l36->pdata.bst_vctl_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L35_BSTCVRT_CTL_SEL_MASK, + cs35l36->pdata.bst_vctl_sel); + + if (cs35l36->pdata.bst_ipk) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_PEAK_CUR, + CS35L36_BST_IPK_MASK, + cs35l36->pdata.bst_ipk); + + if (cs35l36->pdata.boost_ind) + ret = cs35l36_boost_inductor(cs35l36, cs35l36->pdata.boost_ind); + + if (cs35l36->pdata.temp_warn_thld) + regmap_update_bits(cs35l36->regmap, CS35L36_DTEMP_WARN_THLD, + CS35L36_TEMP_THLD_MASK, + cs35l36->pdata.temp_warn_thld); + + /* + * Rev B0 has 2 versions + * L36 is 10V + * L37 is 12V + * If L36 we need to clamp some values for safety + * after probe has setup dt values. We want to make + * sure we dont miss any values set in probe + */ + if (cs35l36->chip_version == CS35L36_10V_L36) { + regmap_update_bits(cs35l36->regmap, + CS35L36_BSTCVRT_OVERVOLT_CTRL, + CS35L36_BST_OVP_THLD_MASK, + CS35L36_BST_OVP_THLD_11V); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + regmap_update_bits(cs35l36->regmap, CS35L36_BST_ANA2_TEST, + CS35L36_BST_OVP_TRIM_MASK, + CS35L36_BST_OVP_TRIM_11V << + CS35L36_BST_OVP_TRIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L36_BST_CTRL_LIM_MASK, + 1 << CS35L36_BST_CTRL_LIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + CS35L36_BST_CTRL_10V_CLAMP); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + /* + * RevA and B require the disabling of + * SYNC_GLOBAL_OVR when GLOBAL_EN = 0. + * Just turn it off from default + */ + regmap_update_bits(cs35l36->regmap, CS35L36_CTRL_OVRRIDE, + CS35L36_SYNC_GLOBAL_OVR_MASK, + 0 << CS35L36_SYNC_GLOBAL_OVR_SHIFT); + + + return 0; +} + +static const struct snd_soc_component_driver soc_component_dev_cs35l36 = { + .probe = &cs35l36_component_probe, + .set_sysclk = cs35l36_component_set_sysclk, + .dapm_widgets = cs35l36_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l36_dapm_widgets), + .dapm_routes = cs35l36_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l36_audio_map), + .controls = cs35l36_aud_controls, + .num_controls = ARRAY_SIZE(cs35l36_aud_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static struct regmap_config cs35l36_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = CS35L36_PAC_PMEM_WORD1023, + .reg_defaults = cs35l36_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l36_reg), + .volatile_reg = cs35l36_volatile_reg, + .readable_reg = cs35l36_readable_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static irqreturn_t cs35l36_irq(int irq, void *data) +{ + struct cs35l36_private *cs35l36 = data; + unsigned int status[4]; + unsigned int masks[4]; + + /* ack the irq by reading all status registers */ + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_STATUS, + status, ARRAY_SIZE(status)); + + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_MASK, + masks, ARRAY_SIZE(masks)); + + /* Check to see if unmasked bits are active */ + if (!(status[0] & ~masks[0]) && !(status[1] & ~masks[1]) && + !(status[2] & ~masks[2]) && !(status[3] & ~masks[3])) { + return IRQ_NONE; + } + + /* + * The following interrupts require a + * protection release cycle to get the + * speaker out of Safe-Mode. + */ + if (status[2] & CS35L36_AMP_SHORT_ERR) { + dev_crit(cs35l36->dev, "Amp short error\n"); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, + CS35L36_AMP_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_INT3_STATUS, + CS35L36_AMP_SHORT_ERR, + CS35L36_AMP_SHORT_ERR); + } + + if (status[0] & CS35L36_TEMP_WARN) { + dev_crit(cs35l36->dev, "Over temperature warning\n"); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, + CS35L36_TEMP_WARN_ERR_RLS); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_INT1_STATUS, + CS35L36_TEMP_WARN, + CS35L36_TEMP_WARN); + } + + if (status[0] & CS35L36_TEMP_ERR) { + dev_crit(cs35l36->dev, "Over temperature error\n"); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, + CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_INT1_STATUS, + CS35L36_TEMP_ERR, + CS35L36_TEMP_ERR); + } + + if (status[0] & CS35L36_BST_OVP_ERR) { + dev_crit(cs35l36->dev, "VBST Over Voltage error\n"); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, + CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_INT1_STATUS, + CS35L36_BST_OVP_ERR, + CS35L36_BST_OVP_ERR); + } + + if (status[0] & CS35L36_BST_DCM_UVP_ERR) { + dev_crit(cs35l36->dev, "DCM VBST Under Voltage Error\n"); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, + CS35L36_BST_UVP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_INT1_STATUS, + CS35L36_BST_DCM_UVP_ERR, + CS35L36_BST_DCM_UVP_ERR); + } + + if (status[0] & CS35L36_BST_SHORT_ERR) { + dev_crit(cs35l36->dev, "LBST SHORT error!\n"); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, + CS35L36_BST_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_INT1_STATUS, + CS35L36_BST_SHORT_ERR, + CS35L36_BST_SHORT_ERR); + } + + return IRQ_HANDLED; +} + +static int cs35l36_handle_of_data(struct i2c_client *i2c_client, + struct cs35l36_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + struct irq_cfg *irq_gpio_config = &pdata->irq_config; + struct vpbr_cfg *vpbr_config = &pdata->vpbr_config; + struct device_node *irq_gpio, *vpbr_node; + unsigned int val; + int ret; + + if (!np) + return 0; + + ret = of_property_read_u32(np, "cirrus,boost-ctl-millivolt", &val); + if (!ret) { + if (val < 2550 || val > 12000) { + dev_err(&i2c_client->dev, + "Invalid Boost Voltage %d mV\n", val); + return -EINVAL; + } + pdata->bst_vctl = (((val - 2550) / 100) + 1) << 1; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-ctl-millivolt'"); + return -EINVAL; + } + + ret = of_property_read_u32(np, "cirrus,boost-ctl-select", &val); + if (!ret) + pdata->bst_vctl_sel = val | CS35L36_VALID_PDATA; + + ret = of_property_read_u32(np, "cirrus,boost-peak-milliamp", &val); + if (!ret) { + if (val < 1600 || val > 4500) { + dev_err(&i2c_client->dev, + "Invalid Boost Peak Current %u mA\n", val); + return -EINVAL; + } + + pdata->bst_ipk = (val - 1600) / 50; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-peak-milliamp'"); + return -EINVAL; + } + + pdata->multi_amp_mode = of_property_read_bool(np, + "cirrus,multi-amp-mode"); + + pdata->sclk_frc = of_property_read_bool(np, + "cirrus,sclk-force-output"); + + pdata->lrclk_frc = of_property_read_bool(np, + "cirrus,lrclk-force-output"); + + pdata->dcm_mode = of_property_read_bool(np, + "cirrus,dcm-mode-enable"); + + pdata->amp_gain_zc = of_property_read_bool(np, + "cirrus,amp-gain-zc"); + + pdata->amp_pcm_inv = of_property_read_bool(np, + "cirrus,amp-pcm-inv"); + + ret = of_property_read_u32(np, "cirrus,ldm-mode-select", &val); + if (!ret) + pdata->ldm_mode_sel = val; + + pdata->pdm_ldm_exit = of_property_read_bool(np, + "cirrus,pdm-ldm-exit"); + + pdata->pdm_ldm_enter = of_property_read_bool(np, + "cirrus,pdm-ldm-enter"); + + pdata->imon_pol_inv = of_property_read_bool(np, + "cirrus,imon-pol-inv"); + + pdata->vmon_pol_inv = of_property_read_bool(np, + "cirrus,vmon-pol-inv"); + + if (of_property_read_u32(np, "cirrus,temp-warn-threshold", &val) >= 0) + pdata->temp_warn_thld = val | CS35L36_VALID_PDATA; + + if (of_property_read_u32(np, "cirrus,boost-ind-nanohenry", &val) >= 0) { + pdata->boost_ind = val; + } else { + dev_err(&i2c_client->dev, "Inductor not specified.\n"); + return -EINVAL; + } + + /* VPBR Config */ + vpbr_node = of_get_child_by_name(np, "cirrus,vpbr-config"); + vpbr_config->is_present = vpbr_node ? true : false; + if (vpbr_config->is_present) { + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-en", + &val) >= 0) + vpbr_config->vpbr_en = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-thld", + &val) >= 0) + vpbr_config->vpbr_thld = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-rate", + &val) >= 0) + vpbr_config->vpbr_atk_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-vol", + &val) >= 0) + vpbr_config->vpbr_atk_vol = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-max-attn", + &val) >= 0) + vpbr_config->vpbr_max_attn = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-wait", + &val) >= 0) + vpbr_config->vpbr_wait = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-rel-rate", + &val) >= 0) + vpbr_config->vpbr_rel_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-mute-en", + &val) >= 0) + vpbr_config->vpbr_mute_en = val; + } + of_node_put(vpbr_node); + + /* INT/GPIO Pin Config */ + irq_gpio = of_get_child_by_name(np, "cirrus,irq-config"); + irq_gpio_config->is_present = irq_gpio ? true : false; + if (irq_gpio_config->is_present) { + if (of_property_read_u32(irq_gpio, "cirrus,irq-drive-select", + &val) >= 0) + irq_gpio_config->irq_drv_sel = val; + if (of_property_read_u32(irq_gpio, "cirrus,irq-polarity", + &val) >= 0) + irq_gpio_config->irq_pol = val; + if (of_property_read_u32(irq_gpio, "cirrus,irq-gpio-select", + &val) >= 0) + irq_gpio_config->irq_gpio_sel = val; + if (of_property_read_u32(irq_gpio, "cirrus,irq-output-enable", + &val) >= 0) + irq_gpio_config->irq_out_en = val; + if (of_property_read_u32(irq_gpio, "cirrus,irq-src-select", + &val) >= 0) + irq_gpio_config->irq_src_sel = val; + } + of_node_put(irq_gpio); + + return 0; +} + +static int cs35l36_pac(struct cs35l36_private *cs35l36) +{ + int ret, count; + unsigned int val; + + if (cs35l36->rev_id == CS35L36_REV_B0) { + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + usleep_range(9500, 10500); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_RESET); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS); + regmap_write(cs35l36->regmap, CS35L36_PAC_PMEM_WORD0, + CS35L36_B0_PAC_PATCH); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS_CLR); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK); + + usleep_range(9500, 10500); + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", + ret); + return ret; + } + + count = 0; + while (!(val & CS35L36_MCU_CONFIG_CLR)) { + usleep_range(100, 200); + count++; + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, + &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", + ret); + return ret; + } + + if (count >= 100) + return -EINVAL; + } + + regmap_write(cs35l36->regmap, CS35L36_INT4_STATUS, + CS35L36_MCU_CONFIG_CLR); + regmap_update_bits(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK, 0); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + return 0; +} + +static int cs35l36_vpbr_config(struct cs35l36_private *cs35l36) +{ + struct cs35l36_platform_data *pdata = &cs35l36->pdata; + struct vpbr_cfg *vpbr_config = &pdata->vpbr_config; + + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL3, + CS35L36_VPBR_EN_MASK, + vpbr_config->vpbr_en << + CS35L36_VPBR_EN_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_THLD_MASK, + vpbr_config->vpbr_thld << + CS35L36_VPBR_THLD_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MAX_ATTN_MASK, + vpbr_config->vpbr_max_attn << + CS35L36_VPBR_MAX_ATTN_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_VOL_MASK, + vpbr_config->vpbr_atk_vol << + CS35L36_VPBR_ATK_VOL_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_RATE_MASK, + vpbr_config->vpbr_atk_rate << + CS35L36_VPBR_ATK_RATE_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_WAIT_MASK, + vpbr_config->vpbr_wait << + CS35L36_VPBR_WAIT_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_REL_RATE_MASK, + vpbr_config->vpbr_rel_rate << + CS35L36_VPBR_REL_RATE_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MUTE_EN_MASK, + vpbr_config->vpbr_mute_en << + CS35L36_VPBR_MUTE_EN_SHIFT); + + return 0; +} + +static int cs35l36_irq_gpio_config(struct cs35l36_private *cs35l36) +{ + struct cs35l36_platform_data *pdata = &cs35l36->pdata; + struct irq_cfg *irq_config = &pdata->irq_config; + int irq_polarity; + + /* setup the Interrupt Pin (INT | GPIO) */ + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_OUTPUT_EN_MASK, + irq_config->irq_out_en); + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_GPIO_SEL_MASK, + irq_config->irq_gpio_sel << + CS35L36_INT_GPIO_SEL_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_POL_SEL_MASK, + irq_config->irq_pol << + CS35L36_INT_POL_SEL_SHIFT); + if (cs35l36->rev_id == CS35L36_REV_A0) + regmap_update_bits(cs35l36->regmap, + CS35L36_PAD_INTERFACE, + CS35L36_IRQ_SRC_MASK, + irq_config->irq_src_sel << + CS35L36_IRQ_SRC_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_DRV_SEL_MASK, + irq_config->irq_drv_sel << + CS35L36_INT_DRV_SEL_SHIFT); + + if (irq_config->irq_pol) + irq_polarity = IRQF_TRIGGER_HIGH; + else + irq_polarity = IRQF_TRIGGER_LOW; + + return irq_polarity; +} + +static const struct reg_sequence cs35l36_pac_int_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + { CS35L36_CTRL_OVRRIDE, 0x00000000 }, + { CS35L36_PAC_INT0_CTRL, 0x00860001 }, + { CS35L36_PAC_INT1_CTRL, 0x00860001 }, + { CS35L36_PAC_INT2_CTRL, 0x00860001 }, + { CS35L36_PAC_INT3_CTRL, 0x00860001 }, + { CS35L36_PAC_INT4_CTRL, 0x00860001 }, + { CS35L36_PAC_INT5_CTRL, 0x00860001 }, + { CS35L36_PAC_INT6_CTRL, 0x00860001 }, + { CS35L36_PAC_INT7_CTRL, 0x00860001 }, + { CS35L36_PAC_INT_FLUSH_CTRL, 0x000000FF }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static const struct reg_sequence cs35l36_reva0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + { CS35L36_OTP_CTRL1, 0x00002060 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { CS35L36_OTP_CTRL1, 0x00002460 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { 0x00002088, 0x012A1838 }, + { 0x00003014, 0x0100EE0E }, + { 0x00003008, 0x0008184A }, + { 0x00007418, 0x509001C8 }, + { 0x00007064, 0x0929A800 }, + { 0x00002D10, 0x0002C01C }, + { 0x0000410C, 0x00000A11 }, + { 0x00006E08, 0x8B19140C }, + { 0x00006454, 0x0300000A }, + { CS35L36_AMP_NG_CTRL, 0x000020EF }, + { 0x00007E34, 0x0000000E }, + { 0x0000410C, 0x00000A11 }, + { 0x00007410, 0x20514B00 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static const struct reg_sequence cs35l36_revb0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + { 0x00007064, 0x0929A800 }, + { 0x00007850, 0x00002FA9 }, + { 0x00007854, 0x0003F1D5 }, + { 0x00007858, 0x0003F5E3 }, + { 0x0000785C, 0x00001137 }, + { 0x00007860, 0x0001A7A5 }, + { 0x00007864, 0x0002F16A }, + { 0x00007868, 0x00003E21 }, + { 0x00007848, 0x00000001 }, + { 0x00003854, 0x05180240 }, + { 0x00007418, 0x509001C8 }, + { 0x0000394C, 0x028764BD }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static int cs35l36_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l36_private *cs35l36; + struct device *dev = &i2c_client->dev; + struct cs35l36_platform_data *pdata = dev_get_platdata(dev); + + int i; + int ret; + u32 reg_id, reg_revid, l37_id_reg; + int irq_pol = IRQF_TRIGGER_HIGH; + + cs35l36 = devm_kzalloc(dev, sizeof(struct cs35l36_private), GFP_KERNEL); + if (!cs35l36) + return -ENOMEM; + + cs35l36->dev = dev; + + i2c_set_clientdata(i2c_client, cs35l36); + cs35l36->regmap = devm_regmap_init_i2c(i2c_client, &cs35l36_regmap); + if (IS_ERR(cs35l36->regmap)) { + ret = PTR_ERR(cs35l36->regmap); + dev_err(dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(cs35l36_supplies); i++) + cs35l36->supplies[i].supply = cs35l36_supplies[i]; + cs35l36->num_supplies = ARRAY_SIZE(cs35l36_supplies); + + ret = devm_regulator_bulk_get(dev, cs35l36->num_supplies, + cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to request core supplies: %d\n", ret); + return ret; + } + + if (pdata) + cs35l36->pdata = *pdata; + else { + pdata = devm_kzalloc(dev, sizeof(struct cs35l36_platform_data), + GFP_KERNEL); + if (!pdata) + return -ENOMEM; + if (i2c_client->dev.of_node) { + ret = cs35l36_handle_of_data(i2c_client, pdata); + if (ret != 0) + return ret; + + } + cs35l36->pdata = *pdata; + } + + ret = regulator_bulk_enable(cs35l36->num_supplies, + cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + /* returning NULL can be an option if in stereo mode */ + cs35l36->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(cs35l36->reset_gpio)) { + ret = PTR_ERR(cs35l36->reset_gpio); + cs35l36->reset_gpio = NULL; + if (ret == -EBUSY) + dev_info(dev, + "Reset line busy, assuming shared reset\n"); + else { + dev_err(dev, "Failed to get reset GPIO: %d\n", ret); + goto err; + } + } + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 1); + + usleep_range(2000, 2100); + + /* initialize amplifier */ + ret = regmap_read(cs35l36->regmap, CS35L36_SW_RESET, &reg_id); + if (ret < 0) { + dev_err(dev, "Get Device ID failed %d\n", ret); + goto err; + } + + if (reg_id != CS35L36_CHIP_ID) { + dev_err(dev, "Device ID (%X). Expected ID %X\n", + reg_id, CS35L36_CHIP_ID); + ret = -ENODEV; + goto err; + } + + ret = regmap_read(cs35l36->regmap, CS35L36_REV_ID, &reg_revid); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed %d\n", ret); + goto err; + } + + cs35l36->rev_id = reg_revid >> 8; + + ret = regmap_read(cs35l36->regmap, CS35L36_OTP_MEM30, &l37_id_reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to read otp_id Register %d\n", + ret); + return ret; + } + + if ((l37_id_reg & CS35L36_OTP_REV_MASK) == CS35L36_OTP_REV_L37) + cs35l36->chip_version = CS35L36_12V_L37; + else + cs35l36->chip_version = CS35L36_10V_L36; + + switch (cs35l36->rev_id) { + case CS35L36_REV_A0: + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_reva0_errata_patch, + ARRAY_SIZE(cs35l36_reva0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply A0 errata patch %d\n", + ret); + goto err; + } + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_pac_int_patch, + ARRAY_SIZE(cs35l36_pac_int_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply A0PAC errata patch %d\n", + ret); + goto err; + } + break; + case CS35L36_REV_B0: + ret = cs35l36_pac(cs35l36); + if (ret < 0) { + dev_err(dev, "Failed to Trim OTP %d\n", ret); + goto err; + } + + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_revb0_errata_patch, + ARRAY_SIZE(cs35l36_revb0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply B0 errata patch %d\n", + ret); + goto err; + } + break; + } + + if (pdata->vpbr_config.is_present) + cs35l36_vpbr_config(cs35l36); + + if (pdata->irq_config.is_present) + irq_pol = cs35l36_irq_gpio_config(cs35l36); + + ret = devm_request_threaded_irq(dev, i2c_client->irq, NULL, cs35l36_irq, + IRQF_ONESHOT | + irq_pol, + "cs35l36", cs35l36); + + if (ret != 0) { + dev_err(dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + + /* Set interrupt masks for critical errors */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_DEFAULT); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_DEFAULT); + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L%d, Revision: %02X\n", + cs35l36->chip_version, reg_revid >> 8); + + ret = snd_soc_register_component(dev, &soc_component_dev_cs35l36, + cs35l36_dai, ARRAY_SIZE(cs35l36_dai)); + if (ret < 0) { + dev_err(dev, "%s: Register component failed %d\n", __func__, + ret); + goto err; + } + + return 0; + +err: + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + return ret; +} + +static int cs35l36_i2c_remove(struct i2c_client *client) +{ + struct cs35l36_private *cs35l36 = i2c_get_clientdata(client); + + /* Reset interrupt masks for device removal */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_RESET); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_RESET); + + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + + snd_soc_unregister_component(&client->dev); + + return 0; +} +static const struct of_device_id cs35l36_of_match[] = { + {.compatible = "cirrus,cs35l36"}, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l36_of_match); + +static const struct i2c_device_id cs35l36_id[] = { + {"cs35l36", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l36_id); + +static struct i2c_driver cs35l36_i2c_driver = { + .driver = { + .name = "cs35l36", + .of_match_table = cs35l36_of_match, + }, + .id_table = cs35l36_id, + .probe = cs35l36_i2c_probe, + .remove = cs35l36_i2c_remove, +}; +module_i2c_driver(cs35l36_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L36 driver"); +MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, <james.schulman(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l36.h b/sound/soc/codecs/cs35l36.h new file mode 100644 index 0000000..5e1dc84 --- /dev/null +++ b/sound/soc/codecs/cs35l36.h @@ -0,0 +1,451 @@ +/* + * cs35l36.h -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.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 __CS35L36_H__ +#define __CS35L36_H__ + +#include <linux/regmap.h> + +#define CS35L36_FIRSTREG 0x00000000 +#define CS35L36_LASTREG 0x00E037FC +#define CS35L36_SW_RESET 0x00000000 +#define CS35L36_SW_REV 0x00000004 +#define CS35L36_HW_REV 0x00000008 +#define CS35L36_TESTKEY_CTRL 0x00000020 +#define CS35L36_USERKEY_CTL 0x00000024 +#define CS35L36_OTP_MEM30 0x00000478 +#define CS35L36_OTP_CTRL1 0x00000500 +#define CS35L36_OTP_CTRL2 0x00000504 +#define CS35L36_OTP_CTRL3 0x00000508 +#define CS35L36_OTP_CTRL4 0x0000050C +#define CS35L36_OTP_CTRL5 0x00000510 +#define CS35L36_PAC_CTL1 0x00000C00 +#define CS35L36_PAC_CTL2 0x00000C04 +#define CS35L36_PAC_CTL3 0x00000C08 +#define CS35L36_DEVICE_ID 0x00002004 +#define CS35L36_FAB_ID 0x00002008 +#define CS35L36_REV_ID 0x0000200C +#define CS35L36_PWR_CTRL1 0x00002014 +#define CS35L36_PWR_CTRL2 0x00002018 +#define CS35L36_PWR_CTRL3 0x0000201C +#define CS35L36_CTRL_OVRRIDE 0x00002020 +#define CS35L36_AMP_OUT_MUTE 0x00002024 +#define CS35L36_OTP_TRIM_STATUS 0x00002028 +#define CS35L36_DISCH_FILT 0x0000202C +#define CS35L36_OSC_TRIM 0x00002030 +#define CS35L36_PROTECT_REL_ERR 0x00002034 +#define CS35L36_PAD_INTERFACE 0x00002400 +#define CS35L36_PLL_CLK_CTRL 0x00002C04 +#define CS35L36_GLOBAL_CLK_CTRL 0x00002C0C +#define CS35L36_ADC_CLK_CTRL 0x00002C10 +#define CS35L36_SWIRE_CLK_CTRL 0x00002C14 +#define CS35L36_SP_SCLK_CLK_CTRL 0x00002D00 +#define CS35L36_TST_FS_MON0 0x00002D10 +#define CS35L36_PLL_LOOP_PARAMS 0x00003008 +#define CS35L36_DCO_CTRL 0x00003010 +#define CS35L36_MISC_CTRL 0x00003014 +#define CS35L36_MDSYNC_EN 0x00003404 +#define CS35L36_MDSYNC_TX_ID 0x00003408 +#define CS35L36_MDSYNC_PWR_CTRL 0x0000340C +#define CS35L36_MDSYNC_DATA_TX 0x00003410 +#define CS35L36_MDSYNC_TX_STATUS 0x0000341C +#define CS35L36_MDSYNC_RX_STATUS 0x00003420 +#define CS35L36_MDSYNC_ERR_STATUS 0x00003424 +#define CS35L36_BSTCVRT_VCTRL1 0x00003800 +#define CS35L36_BSTCVRT_VCTRL2 0x00003804 +#define CS35L36_BSTCVRT_PEAK_CUR 0x00003808 +#define CS35L36_BSTCVRT_SFT_RAMP 0x0000380C +#define CS35L36_BSTCVRT_COEFF 0x00003810 +#define CS35L36_BSTCVRT_SLOPE_LBST 0x00003814 +#define CS35L36_BSTCVRT_SW_FREQ 0x00003818 +#define CS35L36_BSTCVRT_DCM_CTRL 0x0000381C +#define CS35L36_BSTCVRT_DCM_MODE_FORCE 0x00003820 +#define CS35L36_BSTCVRT_OVERVOLT_CTRL 0x00003830 +#define CS35L36_BST_TST_MANUAL 0x0000393C +#define CS35L36_BST_ANA2_TEST 0x0000394C +#define CS35L36_VPI_LIMIT_MODE 0x00003C04 +#define CS35L36_VPI_LIMIT_MINMAX 0x00003C08 +#define CS35L36_VPI_VP_THLD 0x00003C0C +#define CS35L36_VPI_TRACK_CTRL 0x00003C10 +#define CS35L36_VPI_TRIG_MODE_CTRL 0x00003C14 +#define CS35L36_VPI_TRIG_STEPS 0x00003C18 +#define CS35L36_VI_SPKMON_FILT 0x00004004 +#define CS35L36_VI_SPKMON_GAIN 0x00004008 +#define CS35L36_VI_SPKMON_IP_SEL 0x00004100 +#define CS35L36_DTEMP_WARN_THLD 0x00004220 +#define CS35L36_DTEMP_STATUS 0x00004300 +#define CS35L36_VPVBST_FS_SEL 0x00004400 +#define CS35L36_VPVBST_VP_CTRL 0x00004440 +#define CS35L36_VPVBST_VBST_CTRL 0x00004444 +#define CS35L36_ASP_TX_PIN_CTRL 0x00004800 +#define CS35L36_ASP_RATE_CTRL 0x00004804 +#define CS35L36_ASP_FORMAT 0x00004808 +#define CS35L36_ASP_FRAME_CTRL 0x00004818 +#define CS35L36_ASP_TX1_TX2_SLOT 0x0000481C +#define CS35L36_ASP_TX3_TX4_SLOT 0x00004820 +#define CS35L36_ASP_TX5_TX6_SLOT 0x00004824 +#define CS35L36_ASP_TX7_TX8_SLOT 0x00004828 +#define CS35L36_ASP_RX1_SLOT 0x0000482C +#define CS35L36_ASP_RX_TX_EN 0x0000483C +#define CS35L36_ASP_RX1_SEL 0x00004C00 +#define CS35L36_ASP_TX1_SEL 0x00004C20 +#define CS35L36_ASP_TX2_SEL 0x00004C24 +#define CS35L36_ASP_TX3_SEL 0x00004C28 +#define CS35L36_ASP_TX4_SEL 0x00004C2C +#define CS35L36_ASP_TX5_SEL 0x00004C30 +#define CS35L36_ASP_TX6_SEL 0x00004C34 +#define CS35L36_SWIRE_P1_TX1_SEL 0x00004C40 +#define CS35L36_SWIRE_P1_TX2_SEL 0x00004C44 +#define CS35L36_SWIRE_P2_TX1_SEL 0x00004C60 +#define CS35L36_SWIRE_P2_TX2_SEL 0x00004C64 +#define CS35L36_SWIRE_P2_TX3_SEL 0x00004C68 +#define CS35L36_SWIRE_DP1_FIFO_CFG 0x00005000 +#define CS35L36_SWIRE_DP2_FIFO_CFG 0x00005004 +#define CS35L36_SWIRE_DP3_FIFO_CFG 0x00005008 +#define CS35L36_SWIRE_PCM_RX_DATA 0x0000500C +#define CS35L36_SWIRE_FS_SEL 0x00005010 +#define CS35L36_SPARE_CP_BITS 0x00005C00 +#define CS35L36_AMP_DIG_VOL_CTRL 0x00006000 +#define CS35L36_VPBR_CFG 0x00006404 +#define CS35L36_VBBR_CFG 0x00006408 +#define CS35L36_VPBR_STATUS 0x0000640C +#define CS35L36_VBBR_STATUS 0x00006410 +#define CS35L36_OVERTEMP_CFG 0x00006414 +#define CS35L36_AMP_ERR_VOL 0x00006418 +#define CS35L36_CLASSH_CFG 0x00006800 +#define CS35L36_CLASSH_FET_DRV_CFG 0x00006804 +#define CS35L36_NG_CFG 0x00006808 +#define CS35L36_AMP_GAIN_CTRL 0x00006C04 +#define CS35L36_PWM_MOD_IO_CTRL 0x0000706C +#define CS35L36_PWM_MOD_STATUS 0x00007070 +#define CS35L36_DAC_MSM_CFG 0x00007400 +#define CS35L36_AMP_SLOPE_CTRL 0x00007410 +#define CS35L36_AMP_PDM_VOLUME 0x00007E04 +#define CS35L36_AMP_PDM_RATE_CTRL 0x00007E08 +#define CS35L36_PDM_CH_SEL 0x00007E10 +#define CS35L36_AMP_NG_CTRL 0x00007E14 +#define CS35L36_PDM_HIGHFILT_CTRL 0x00007E3C +#define CS35L36_INT1_STATUS 0x00D00000 +#define CS35L36_INT2_STATUS 0x00D00004 +#define CS35L36_INT3_STATUS 0x00D00008 +#define CS35L36_INT4_STATUS 0x00D0000C +#define CS35L36_INT1_RAW_STATUS 0x00D00020 +#define CS35L36_INT2_RAW_STATUS 0x00D00024 +#define CS35L36_INT3_RAW_STATUS 0x00D00028 +#define CS35L36_INT4_RAW_STATUS 0x00D0002C +#define CS35L36_INT1_MASK 0x00D00040 +#define CS35L36_INT2_MASK 0x00D00044 +#define CS35L36_INT3_MASK 0x00D00048 +#define CS35L36_INT4_MASK 0x00D0004C +#define CS35L36_INT1_EDGE_LVL_CTRL 0x00D00060 +#define CS35L36_INT3_EDGE_LVL_CTRL 0x00D00068 +#define CS35L36_PAC_INT_STATUS 0x00D00200 +#define CS35L36_PAC_INT_RAW_STATUS 0x00D00210 +#define CS35L36_PAC_INT_FLUSH_CTRL 0x00D00218 +#define CS35L36_PAC_INT0_CTRL 0x00D00220 +#define CS35L36_PAC_INT1_CTRL 0x00D00224 +#define CS35L36_PAC_INT2_CTRL 0x00D00228 +#define CS35L36_PAC_INT3_CTRL 0x00D0022C +#define CS35L36_PAC_INT4_CTRL 0x00D00230 +#define CS35L36_PAC_INT5_CTRL 0x00D00234 +#define CS35L36_PAC_INT6_CTRL 0x00D00238 +#define CS35L36_PAC_INT7_CTRL 0x00D0023C +#define CS35L36_PAC_PMEM_WORD0 0x00E02800 +#define CS35L36_PAC_PMEM_WORD1 0x00E02804 +#define CS35L36_PAC_PMEM_WORD1023 0x00E037FC + +#define CS35L36_MAX_CACHE_REG 0x00007E44 +#define CS35L36_INTPAC_REG_COUNT 25 +#define CS35L36_CHIP_ID 0x00035A36 + +#define CS35L36_INT_OUTPUT_EN_MASK 0x01 +#define CS35L36_INT_GPIO_SEL_MASK 0x02 +#define CS35L36_INT_GPIO_SEL_SHIFT 1 +#define CS35L36_INT_POL_SEL_MASK 0x04 +#define CS35L36_INT_POL_SEL_SHIFT 2 +#define CS35L36_INT_DRV_SEL_MASK 0x20 +#define CS35L36_INT_DRV_SEL_SHIFT 5 +#define CS35L36_IRQ_SRC_MASK 0x08 +#define CS35L36_IRQ_SRC_SHIFT 3 + +#define CS35L36_SCLK_MSTR_MASK 0x40 +#define CS35L36_SCLK_MSTR_SHIFT 6 +#define CS35L36_LRCLK_MSTR_MASK 0x01 +#define CS35L36_LRCLK_MSTR_SHIFT 0 +#define CS35L36_SCLK_INV_MASK 0x100 +#define CS35L36_SCLK_INV_SHIFT 8 +#define CS35L36_LRCLK_INV_MASK 0x04 +#define CS35L36_LRCLK_INV_SHIFT 2 +#define CS35L36_SCLK_FRC_MASK 0x80 +#define CS35L36_SCLK_FRC_SHIFT 7 +#define CS35L36_LRCLK_FRC_MASK 0x02 +#define CS35L36_LRCLK_FRC_SHIFT 1 + +#define CS35L36_PDM_MODE_MASK 0x01 +#define CS35L36_PDM_MODE_SHIFT 0 + +#define CS35L36_ASP_FMT_MASK 0x07 +#define CS35L36_ASP_FMT_SHIFT 0 + +#define CS35L36_ASP_RX_WIDTH_MASK 0xFF0000 +#define CS35L36_ASP_RX_WIDTH_SHIFT 16 +#define CS35L36_ASP_TX_WIDTH_MASK 0xFF +#define CS35L36_ASP_TX_WIDTH_SHIFT 0 +#define CS35L36_ASP_WIDTH_16 0x10 +#define CS35L36_ASP_WIDTH_24 0x18 +#define CS35L36_ASP_WIDTH_32 0x20 + +#define CS35L36_ASP_RX1_SLOT_MASK 0x3F +#define CS35L36_ASP_RX1_EN_MASK 0x00010000 +#define CS35L36_ASP_RX1_EN_SHIFT 16 + +#define CS35L36_ASP_TX1_SLOT_MASK 0x3F +#define CS35L36_ASP_TX2_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX2_SLOT_SHIFT 16 +#define CS35L36_ASP_TX3_SLOT_MASK 0x3F +#define CS35L36_ASP_TX4_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX4_SLOT_SHIFT 16 +#define CS35L36_ASP_TX5_SLOT_MASK 0x3F +#define CS35L36_ASP_TX6_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX6_SLOT_SHIFT 16 +#define CS35L36_ASP_TX7_SLOT_MASK 0x3F +#define CS35L36_ASP_TX8_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX8_SLOT_SHIFT 16 +#define CS35L36_ASP_TX_HIZ_MASK 0x200000 + +#define CS35L36_APS_TX_SEL_MASK 0x7F + +#define CS35L36_ASP_TX1_EN_MASK 0x01 +#define CS35L36_ASP_TX2_EN_MASK 0x02 +#define CS35L36_ASP_TX2_EN_SHIFT 1 +#define CS35L36_ASP_TX3_EN_MASK 0x04 +#define CS35L36_ASP_TX3_EN_SHIFT 2 +#define CS35L36_ASP_TX4_EN_MASK 0x08 +#define CS35L36_ASP_TX4_EN_SHIFT 3 +#define CS35L36_ASP_TX5_EN_MASK 0x10 +#define CS35L36_ASP_TX5_EN_SHIFT 4 +#define CS35L36_ASP_TX6_EN_MASK 0x20 +#define CS35L36_ASP_TX6_EN_SHIFT 5 +#define CS35L36_ASP_TX7_EN_MASK 0x40 +#define CS35L36_ASP_TX7_EN_SHIFT 6 +#define CS35L36_ASP_TX8_EN_MASK 0x80 +#define CS35L36_ASP_TX8_EN_SHIFT 7 + + +#define CS35L36_PLL_CLK_SEL_MASK 0x07 +#define CS35L36_PLL_CLK_SEL_SHIFT 0 +#define CS35L36_PLLSRC_SCLK 0 +#define CS35L36_PLLSRC_LRCLK 1 +#define CS35L36_PLLSRC_SELF 3 +#define CS35L36_PLLSRC_PDMCLK 4 +#define CS35L36_PLLSRC_MCLK 5 +#define CS35L36_PLLSRC_SWIRE 7 +#define CS35L36_REFCLK_FREQ_MASK 0x7E0 +#define CS35L36_REFCLK_FREQ_SHIFT 5 +#define CS35L36_PLL_OPENLOOP_MASK 0x800 +#define CS35L36_PLL_OPENLOOP_SHIFT 11 +#define CS35L36_PLL_REFCLK_EN_MASK 0x10 +#define CS35L36_PLL_REFCLK_EN_SHIFT 4 + + +#define CS35L36_GLOBAL_FS_MASK 0x1F +#define CS35L36_GLOBAL_FS_SHIFT 0 + +#define CS35L36_HPF_PCM_EN_MASK 0x800 +#define CS35L36_HPF_PCM_EN_SHIFT 15 +#define CS35L36_PCM_RX_SEL_MASK 0x7F +#define CS35L36_PCM_RX_SEL_SHIFT 0 + +#define CS35L36_PCM_RX_SEL_ZERO 0x00 +#define CS35L36_PCM_RX_SEL_PCM 0x08 +#define CS35L36_PCM_RX_SEL_SWIRE 0x10 +#define CS35L36_PCM_RX_SEL_DIAG 0x04 + +#define CS35L36_GLOBAL_EN_MASK 0x01 +#define CS35L36_GLOBAL_EN_SHIFT 0x00 + +#define CS35L36_AMP_PCM_INV_MASK 0x4000 +#define CS35L36_AMP_PCM_INV_SHIFT 14 + +#define CS35L36_AMP_VOL_PCM_MASK 0x3FF8 +#define CS35L36_AMP_VOL_PCM_SHIFT 3 +#define CS35L36_DIGITAL_MUTE 0x04CF + +#define CS35L36_AMP_RAMP_MASK 0x0007 +#define CS35L36_AMP_RAMP_SHIFT 0 + +#define CS35L36_AMP_MUTE_MASK 0x0010 +#define CS35L36_AMP_MUTE_SHIFT 4 + +#define CS35L36_GLOBAL_RESYNC_FS1_MASK 0x00000200 +#define CS35L36_GLOBAL_RESYNC_FS2_MASK 0x00000400 +#define CS35L36_SYNC_GLOBAL_OVR_MASK 0x00000002 +#define CS35L36_SYNC_GLOBAL_OVR_SHIFT 1 + +#define CS35L36_REFCLK_IN_MASK 0x00100000 +#define CS35L36_PLL_UNLOCK_MASK 0x00002000 + +#define CS35L36_ASP_RX_UDF_MASK 0x00000040 +#define CS35L36_ASP_RX_OVF_MASK 0x00000080 + +#define CS35L36_IMON_POL_MASK 0x02 +#define CS35L36_IMON_POL_SHIFT 1 + +#define CS35L36_VMON_POL_MASK 0x01 +#define CS35L36_VMON_POL_SHIFT 0 + +#define CS35L36_PDN_DONE 0x40 +#define CS35L36_PDN_DONE_SHIFT 6 +#define CS35L36_PUP_DONE 0x80 +#define CS35L36_PUP_DONE_SHIFT 7 +#define CS35L36_GLOBAL_EN_ASSRT 0x20 +#define CS35L36_PUP_DONE_IRQ_UNMASK 0x7F +#define CS35L36_PUP_DONE_IRQ_MASK 0xBF + +#define CS35L36_FS1_WINDOW_MASK 0x000007FF +#define CS35L36_FS2_WINDOW_MASK 0x00FFF800 +#define CS35L36_FS2_WINDOW_SHIFT 12 + +#define CS35L36_PLL_FFL_IGAIN_MASK 0x0F +#define CS35L36_PLL_IGAIN_MASK 0x3F0 +#define CS35L36_PLL_IGAIN_SHIFT 4 +#define CS35L36_PLL_IGAIN 0x04 + +#define CS35L36_BST_EN_MASK 0x30 +#define CS35L36_BST_EN 0x02 +#define CS35L36_BST_DIS_VP 0x01 +#define CS35L36_BST_DIS_EXTN 0x00 +#define CS35L36_BST_EN_SHIFT 4 +#define CS35L36_BST_MAN_IPKCOMP_MASK 0x200 +#define CS35L36_BST_MAN_IPKCOMP_SHIFT 9 + +#define CS35L36_BST_MAN_IPKCOMP_EN_MASK 0x100 +#define CS35L36_BST_MAN_IPKCOMP_EN_SHIFT 8 + +#define CS35L36_BST_IPK_MASK 0x7F +#define CS35L36_BST_OVP_THLD_MASK 0x3F +#define CS35L36_BST_OVP_THLD_11V 0x10 +#define CS35L36_BST_OVP_TRIM_MASK 0x00078000 +#define CS35L36_BST_OVP_TRIM_SHIFT 15 +#define CS35L36_BST_OVP_TRIM_11V 0x0C +#define CS35L36_BST_CTRL_LIM_MASK 0x04 +#define CS35L36_BST_CTRL_LIM_SHIFT 2 +#define CS35L36_BST_CTRL_10V_CLAMP 0x96 + +#define CS35L36_NG_AMP_EN_MASK 0x3F00 +#define CS35L36_NG_AMP_EN_SHIFT 8 +#define CS35L36_NG_DELAY_MASK 0x70 +#define CS35L36_NG_DELAY_SHIFT 4 +#define CS35L36_AMP_ZC_MASK 0x400 +#define CS35L36_PDM_LDM_ENTER_MASK 0x08 +#define CS35L36_PDM_LDM_EXIT_MASK 0x10 + +#define CS35L36_BSTCVRT_K1_MASK 0xFF +#define CS35L36_BSTCVRT_K2_MASK 0xFF00 +#define CS35L36_BSTCVRT_K2_SHIFT 8 +#define CS35L36_BSTCVRT_SLOPE_MASK 0xFF00 +#define CS35L36_BSTCVRT_SLOPE_SHIFT 8 +#define CS35L36_BSTCVRT_CCMFREQ_MASK 0x0F +#define CS35L36_BSTCVRT_LBSTVAL_MASK 0x03 +#define CS35L35_BSTCVRT_CTL_MASK 0xFF +#define CS35L35_BSTCVRT_CTL_SEL_MASK 0x03 +#define CS35L36_DCM_AUTO_MASK 0x01 + +#define CS35L36_INT1_MASK_DEFAULT 0xF9BA7FFF +#define CS35L36_INT1_MASK_RESET 0xFFFFFFFF +#define CS35L36_INT3_MASK_DEFAULT 0xFFFFEFFF +#define CS35L36_INT3_MASK_RESET 0xFFFFFFFF + + +#define CS35L36_AMP_SHORT_ERR 0x1000 +#define CS35L36_BST_SHORT_ERR 0x40000 +#define CS35L36_TEMP_WARN 0x2000000 +#define CS35L36_TEMP_ERR 0x4000000 +#define CS35L36_BST_OVP_ERR 0x10000 +#define CS35L36_BST_DCM_UVP_ERR 0x20000 + +#define CS35L36_AMP_SHORT_ERR_RLS 0x02 +#define CS35L36_BST_SHORT_ERR_RLS 0x04 +#define CS35L36_BST_OVP_ERR_RLS 0x08 +#define CS35L36_BST_UVP_ERR_RLS 0x10 +#define CS35L36_TEMP_WARN_ERR_RLS 0x20 +#define CS35L36_TEMP_ERR_RLS 0x40 +#define CS35L36_TEMP_THLD_MASK 0x03 + +#define CS35L36_REV_B0 0xb0 +#define CS35L36_REV_A0 0xa0 +#define CS35L36_B0_PAC_PATCH 0x00DD0102 + +#define CS35L36_OTP_ECC_EN_MASK 0x400 +#define CS35L36_OTP_ECC_EN_SHIFT 10 +#define CS35L36_OTP_RUN_BOOT_MASK 0x01 +#define CS35L36_OTP_BOOT_DONE 0x2000000 +#define CS35L36_PAC_RESET_MASK 0x04 +#define CS35L36_PAC_RESET_SHIFT 2 +#define CS35L36_PAC_STALL_MASK 0x02 +#define CS35L36_PAC_STALL_SHIFT 1 +#define CS35L36_PAC_ENABLE_MASK 0x00000001 +#define CS35L36_PAC_MEM_ACCESS 0x01 +#define CS35L36_PAC_MEM_ACCESS_CLR 0 +#define CS35L36_SOFT_RESET 0x5AAA +#define CS35L36_MCU_BOOT_COMPLETE 0x02 +#define CS35L36_MCU_CONFIG_UNMASK 0x00FEFFFF +#define CS35L36_MCU_CONFIG_CLR 0x00010000 +#define CS35L36_MCU_CONFIG_MASK 0x00FFFFFF +#define CS35L36_GPIO_INT_SEL_MASK 0x0000003B +#define CS35L36_GPIO_INT_SEL_UNMASK 0x0000003A +#define CS35L36_PAC_RESET 0x00000000 +#define CS35L36_OTP_REV_MASK 0x00FF0000 +#define CS35L36_OTP_REV_L37 0x00CC0000 +#define CS35L36_12V_L37 37 +#define CS35L36_10V_L36 36 + +#define CS35L36_VPBR_EN_MASK 0x00001000 +#define CS35L36_VPBR_EN_SHIFT 12 + +#define CS35L36_VPBR_THLD_MASK 0x0000001F +#define CS35L36_VPBR_THLD_SHIFT 0 +#define CS35L36_VPBR_MAX_ATTN_MASK 0x00000F00 +#define CS35L36_VPBR_MAX_ATTN_SHIFT 8 +#define CS35L36_VPBR_ATK_VOL_MASK 0x0000F000 +#define CS35L36_VPBR_ATK_VOL_SHIFT 12 +#define CS35L36_VPBR_ATK_RATE_MASK 0x00070000 +#define CS35L36_VPBR_ATK_RATE_SHIFT 16 +#define CS35L36_VPBR_WAIT_MASK 0x00180000 +#define CS35L36_VPBR_WAIT_SHIFT 19 +#define CS35L36_VPBR_REL_RATE_MASK 0x00E00000 +#define CS35L36_VPBR_REL_RATE_SHIFT 21 +#define CS35L36_VPBR_MUTE_EN_MASK 0x01000000 +#define CS35L36_VPBR_MUTE_EN_SHIFT 24 + +#define CS35L36_OSC_FREQ_TRIM_MASK 0x070 +#define CS35L36_OSC_TRIM_DONE 0x08 + +#define CS35L36_TEST_UNLOCK1 0x00005555 +#define CS35L36_TEST_UNLOCK2 0x0000AAAA +#define CS35L36_TEST_LOCK1 0x0000CCCC +#define CS35L36_TEST_LOCK2 0x00003333 + +#define CS35L36_PAC_PROG_MEM 512 + +#define CS35L36_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define CS35L36_TX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +bool cs35l36_readable_reg(struct device *dev, unsigned int reg); +bool cs35l36_volatile_reg(struct device *dev, unsigned int reg); + +extern struct reg_default cs35l36_reg[CS35L36_MAX_CACHE_REG]; +extern const int cs35l36_a0_pac_patch[CS35L36_PAC_PROG_MEM]; + +#endif -- 1.9.1

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[alsa-devel] USB speaker with CM103+ chip has left and right audio inverted
by Helge Deller 05 Dec '18

05 Dec '18

Hello, When using my Logitech LogiLink USB Speaker with Linux, the left and right audio channel is inverted. lsusb shows Bus 001 Device 016: ID 0d8c:0104 C-Media Electronics, Inc. CM103+ Audio Controller dmesg gives: kernel: usb 1-3: new full-speed USB device number 16 using xhci_hcd kernel: usb 1-3: New USB device found, idVendor=0d8c, idProduct=0104, bcdDevice= 0.10 kernel: usb 1-3: New USB device strings: Mfr=1, Product=2, SerialNumber=0 kernel: usb 1-3: Product: USB Sound Device kernel: usb 1-3: Manufacturer: C-Media INC. kernel: usb 1-3: Warning! Unlikely big volume range (=9472), cval->res is probably wrong. kernel: usb 1-3: [13] FU [PCM Playback Volume] ch = 2, val = -9473/-1/1 kernel: input: C-Media INC. USB Sound Device as /devices/pci0000:00/0000:00:14.0/usb1/1-3/1-3:1.2/0003:0D8C:0104.0034/input/input59 kernel: hid-generic 0003:0D8C:0104.0034: input,hidraw4: USB HID v1.00 Device [C-Media INC. USB Sound Device ] on usb-0000:00:14.0-3/input2 Is there a quirk possible? Any idea/patch what I could test? This is on Fedora29 with kernel 4.19.5-300.fc29.x86_64. Thanks, Helge

1 0

[alsa-devel] [resend PATCH] dt-bindings: sound: Add documentation for pcm3060 property out-single-ended
by Kirill Marinushkin 05 Dec '18

05 Dec '18

Output of pcm3060 codec may be configured as single-ended or differential Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Cc: devicetree(a)vger.kernel.org --- Hello Mark, yesterday there was a misunderstanding: when I wrote you > I think you forgot one patch in the series you accidently applied the already-applied patch instead of the forgotten one. To avoid the confusion, in this e-mail I resend you the proper forgotten patch in a separate mailing thread. Please apply this patch for-next. Best Regards, Kirill --- Documentation/devicetree/bindings/sound/pcm3060.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/pcm3060.txt b/Documentation/devicetree/bindings/sound/pcm3060.txt index 90fcb8523099..97de66932d44 100644 --- a/Documentation/devicetree/bindings/sound/pcm3060.txt +++ b/Documentation/devicetree/bindings/sound/pcm3060.txt @@ -9,9 +9,15 @@ Required properties: - reg : the I2C address of the device for I2C, the chip select number for SPI. +Optional properties: + +- ti,out-single-ended: "true" if output is single-ended; + "false" or not specified if output is differential. + Examples: pcm3060: pcm3060@46 { compatible = "ti,pcm3060"; reg = <0x46>; + ti,out-single-ended = "true"; }; -- 2.13.6

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Re: [alsa-devel] [PATCH 1/4] ALSA: hda/realtek: ALC286 mic and headset-mode fixups for Acer Aspire U27-880
by Takashi Iwai 05 Dec '18

05 Dec '18

On Wed, 05 Dec 2018 13:37:43 +0100, Kailang wrote: > > Hi Takashi, > > You should apply alc286 platform first. > I think they had test passed already. > ALC294 platform just wait for test via change model. OK, now applied this series for ALC286. For ALC294, I'll wait for the test result. thanks, Takashi > > BR, > Kailang > ________________________________________ > 從: Takashi Iwai [tiwai(a)suse.de] > 寄件日期: 2018年12月5日下午 05:32 > 至: Kailang > 副本: Jian-Hong Pan; Jaroslav Kysela; alsa-devel(a)alsa-project.org; Hui Wang; Chris Chiu; Daniel Drake; linux(a)endlessm.com; linux-kernel(a)vger.kernel.org > 主旨: Re: [PATCH 1/4] ALSA: hda/realtek: ALC286 mic and headset-mode fixups for Acer Aspire U27-880 > > On Wed, 05 Dec 2018 10:06:14 +0100, > Kailang wrote: > > > > Sorry!! > > Don't change this. > > Hm, should I apply these series or not? > > > thanks, > > Takashi > > > > > -----Original Message----- > > From: Kailang > > Sent: Wednesday, December 5, 2018 4:52 PM > > To: 'Jian-Hong Pan' <jian-hong(a)endlessm.com>; Jaroslav Kysela <perex(a)perex.cz>; Takashi Iwai <tiwai(a)suse.com> > > Cc: Hui Wang <hui.wang(a)canonical.com>; alsa-devel(a)alsa-project.org; linux-kernel(a)vger.kernel.org; linux(a)endlessm.com; Chris Chiu <chiu(a)endlessm.com>; Daniel Drake <drake(a)endlessm.com> > > Subject: RE: [PATCH 1/4] ALSA: hda/realtek: ALC286 mic and headset-mode fixups for Acer Aspire U27-880 > > > > Hi Jian Hong, > > > > Please also try to change model to ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC. > > > > + .chain_id = ALC269_FIXUP_HEADSET_MIC ==> change to ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC > > > > BR, > > Kailang > > -----Original Message----- > > From: Jian-Hong Pan <jian-hong(a)endlessm.com> > > Sent: Wednesday, December 5, 2018 2:49 PM > > To: Jaroslav Kysela <perex(a)perex.cz>; Takashi Iwai <tiwai(a)suse.com> > > Cc: Kailang <kailang(a)realtek.com>; Hui Wang <hui.wang(a)canonical.com>; alsa-devel(a)alsa-project.org; linux-kernel(a)vger.kernel.org; linux(a)endlessm.com; Chris Chiu <chiu(a)endlessm.com>; Jian-Hong Pan <jian-hong(a)endlessm.com>; Daniel Drake <drake(a)endlessm.com> > > Subject: [PATCH 1/4] ALSA: hda/realtek: ALC286 mic and headset-mode fixups for Acer Aspire U27-880 > > > > From: Chris Chiu <chiu(a)endlessm.com> > > > > Acer Aspire U27-880(AIO) with ALC286 codec can not detect headset mic and internal mic not working either. It needs the similar quirk like Sony laptops to fix headphone jack sensing and enables use of the internal microphone. > > > > Unfortunately jack sensing for the headset mic is still not working. > > > > Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> > > Signed-off-by: Daniel Drake <drake(a)endlessm.com> > > Signed-off-by: Chris Chiu <chiu(a)endlessm.com> > > --- > > sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ > > 1 file changed, 14 insertions(+) > > > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c0b289ba397f..f21d52eb2ed3 100644 > > --- a/sound/pci/hda/patch_realtek.c > > +++ b/sound/pci/hda/patch_realtek.c > > @@ -5516,6 +5516,7 @@ enum { > > ALC221_FIXUP_HP_HEADSET_MIC, > > ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, > > ALC295_FIXUP_HP_AUTO_MUTE, > > + ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, > > }; > > > > static const struct hda_fixup alc269_fixups[] = { @@ -6393,6 +6394,15 @@ static const struct hda_fixup alc269_fixups[] = { > > .type = HDA_FIXUP_FUNC, > > .v.func = alc_fixup_auto_mute_via_amp, > > }, > > + [ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE] = { > > + .type = HDA_FIXUP_PINS, > > + .v.pins = (const struct hda_pintbl[]) { > > + { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ > > + { } > > + }, > > + .chained = true, > > + .chain_id = ALC269_FIXUP_HEADSET_MIC > > + }, > > }; > > > > static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7071,6 +7081,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { > > {0x14, 0x90170110}, > > {0x19, 0x04a11040}, > > {0x21, 0x04211020}), > > + SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, > > + {0x12, 0x90a60130}, > > + {0x17, 0x90170110}, > > + {0x21, 0x02211020}), > > SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, > > {0x12, 0x90a60120}, > > {0x14, 0x90170110}, > > -- > > 2.11.0 > > > > > > ------Please consider the environment before printing this e-mail.

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[alsa-devel] [PATCH linux-next v3 7/7] clk: renesas: Add binding document for ADG
by Jiada Wang 05 Dec '18

05 Dec '18

Add device tree bindings for Audio Clock Generator (ADG) of R-Car Socs. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- .../clock/renesas,rcar-adg-clocks.txt | 24 +++++++++++++++++++ 1 file changed, 24 insertions(+) create mode 100644 Documentation/devicetree/bindings/clock/renesas,rcar-adg-clocks.txt diff --git a/Documentation/devicetree/bindings/clock/renesas,rcar-adg-clocks.txt b/Documentation/devicetree/bindings/clock/renesas,rcar-adg-clocks.txt new file mode 100644 index 000000000000..76fc4f8964e9 --- /dev/null +++ b/Documentation/devicetree/bindings/clock/renesas,rcar-adg-clocks.txt @@ -0,0 +1,24 @@ +* Renesas R-Car Audio Clock Generator (ADG) + +The Audio Clock Generator (ADG) is part of R-Car Audio module, it +selects and supplies the necessary clock for the SSIU, EAVB-IF, SCU, +ADSP or DTCP module. It also divides the frequency of the selected +clock and sends it outside the chip. + +Required Properties: + + - compatible: Must be one of + - "renesas,rcar_sound-gen1" for the R-Car GEN1 SoCs + - "renesas,rcar_sound-gen2" for the R-Car GEN2 SoCs + - "renesas,rcar_sound-gen3" for the R-Car GEN3 SoCs + + - reg: Base address and length of the memory resource used by the ADG + + - clocks: References to the parent clock S0D1. + - clock-names: ADG refer to its parent clock by name "adg". + - #clock-cells: Can be 0, 1 or 2 + - When clock-cells = 0, ADG registers one fixed clock out + - When clock-cells = 1, ADG registers 3 fixed clock out + - When clock-cells = 2, ADG registers 3 fixed clock out and 8 AVB clocks + second clock specifier need to be 0 to refer to fixed clock out, need + to be 1 to refer to AVB clocks -- 2.19.2

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Re: [alsa-devel] No audio on iMac18,1 (A1418) with kernel 4.15.14
by Andrey L 05 Dec '18

05 Dec '18

Good day. I'm yet another owner of 2018 iMac who can't get sound to work. I was wondering if someone with the expertise can comment whether it is possible at all. Relevant earlier threads: https://www.spinics.net/lists/alsa-devel/msg76308.html https://bbs.archlinux.org/viewtopic.php?id=233449 Like others before I tried pretty much all options of snd-hda-intel with no success. The sound however works on windows, so I decided to look there. The codec chip is Cirrus CS4208, and without installing chip drivers, there was no audio in windows as well. It turns out that windows driver is a generic chip driver, where all hardware configuration is stored in registry (binary config file). These settings are configured in installation inf file (text config file) , and what is important – they are heavily commented. Unfortunately I do lack domain knowledge about hda-intel / codec / endpoint or alsa architecture, nor have I any desire to learn this area. I would appreciate if anyone can look at my findings and tell whether current linux drivers can be hacked to support this configuration. I’ve copied and sorted all relevant sections from inf file here - https://pastebin.com/p74qSdzc Here is what INF file defines: The chip is CS4208, but it is also referenced as "Cirrus Logic CS8409 (AB 16)" . There are 4 interfaces: speaker, headphones, mic and d-mic. Headphones are stereo. Speakers are quad(!!!) “quad (FL/FR - woofers, RL/RR – tweeters)”. The INF file contains following sections: CONF_0910.InitVerbs – apparently configuration initialization sequence, lists register-value pairs written by driver at startup. SettingsCS8409Exit – same for shutdown CONF_0900.Gpio, CONF_0910.Gpio – configuration of GPIO with comments CONF_0900.TXRX – Widgets,pins and amps caps (I have no clue what it is), but from comments this section seems to define all chip connections. CONF_09xx.I2C – raw i2c commands for various scenarios (chip power up, start/stop streaming, etc). So the question is, is this data of any help? Can this be somehow plugged into existing hda-intel or cs4207 drivers to get the sound? Thanks.

1 0

[alsa-devel] remove bogus GFP_DMA32 flags for dma allocations
by Christoph Hellwig 05 Dec '18

05 Dec '18

dma_alloc_* internally selects the zone to allocate from based on the DMA mask. Remove all explicit uses of GFP_DMA32 with dma_alloc_*.

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[alsa-devel] [PATCH v2] ASoC: DA7219: Implement error check on reg read and write
by Agrawal, Akshu 05 Dec '18

05 Dec '18

Failed i2c transaction can lead to failure in reg read or write. Need to have error check for each read write operation. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: replaced snd_soc_component_read32 by snd_soc_component_read Since, snd_soc_component_read32 implementation has error, in coming patches we can remove snd_soc_component_read32 from soc-io and replace all codecs by snd_soc_component_read sound/soc/codecs/da7219.c | 349 +++++++++++++++++++++++++++++++++++----------- sound/soc/codecs/da7219.h | 2 +- 2 files changed, 265 insertions(+), 86 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index e46e9f4..c65e734 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -312,69 +312,105 @@ static int da7219_enum_locked_put(struct snd_kcontrol *kcontrol, } /* ALC */ -static void da7219_alc_calib(struct snd_soc_component *component) +static int da7219_alc_calib(struct snd_soc_component *component) { - u8 mic_ctrl, mixin_ctrl, adc_ctrl, calib_ctrl; + unsigned int mic_ctrl, mixin_ctrl, adc_ctrl, calib_ctrl; + int ret; /* Save current state of mic control register */ - mic_ctrl = snd_soc_component_read32(component, DA7219_MIC_1_CTRL); + ret = snd_soc_component_read(component, DA7219_MIC_1_CTRL, &mic_ctrl); + if (ret) + return ret; /* Save current state of input mixer control register */ - mixin_ctrl = snd_soc_component_read32(component, DA7219_MIXIN_L_CTRL); + ret = snd_soc_component_read(component, DA7219_MIXIN_L_CTRL, + &mixin_ctrl); + if (ret) + return ret; /* Save current state of input ADC control register */ - adc_ctrl = snd_soc_component_read32(component, DA7219_ADC_L_CTRL); + ret = snd_soc_component_read(component, DA7219_ADC_L_CTRL, &adc_ctrl); + if (ret) + return ret; /* Enable then Mute MIC PGAs */ - snd_soc_component_update_bits(component, DA7219_MIC_1_CTRL, DA7219_MIC_1_AMP_EN_MASK, + ret = snd_soc_component_update_bits(component, DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_EN_MASK, DA7219_MIC_1_AMP_EN_MASK); - snd_soc_component_update_bits(component, DA7219_MIC_1_CTRL, + if (ret < 0) + return ret; + + ret = snd_soc_component_update_bits(component, DA7219_MIC_1_CTRL, DA7219_MIC_1_AMP_MUTE_EN_MASK, DA7219_MIC_1_AMP_MUTE_EN_MASK); + if (ret < 0) + return ret; /* Enable input mixers unmuted */ - snd_soc_component_update_bits(component, DA7219_MIXIN_L_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_MIXIN_L_CTRL, DA7219_MIXIN_L_AMP_EN_MASK | DA7219_MIXIN_L_AMP_MUTE_EN_MASK, DA7219_MIXIN_L_AMP_EN_MASK); + if (ret < 0) + return ret; /* Enable input filters unmuted */ - snd_soc_component_update_bits(component, DA7219_ADC_L_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_ADC_L_CTRL, DA7219_ADC_L_MUTE_EN_MASK | DA7219_ADC_L_EN_MASK, DA7219_ADC_L_EN_MASK); + if (ret < 0) + return ret; /* Perform auto calibration */ - snd_soc_component_update_bits(component, DA7219_ALC_CTRL1, + ret = snd_soc_component_update_bits(component, DA7219_ALC_CTRL1, DA7219_ALC_AUTO_CALIB_EN_MASK, DA7219_ALC_AUTO_CALIB_EN_MASK); + if (ret < 0) + return ret; do { - calib_ctrl = snd_soc_component_read32(component, DA7219_ALC_CTRL1); + ret = snd_soc_component_read(component, DA7219_ALC_CTRL1, + &calib_ctrl); + if (ret) + return ret; } while (calib_ctrl & DA7219_ALC_AUTO_CALIB_EN_MASK); /* If auto calibration fails, disable DC offset, hybrid ALC */ if (calib_ctrl & DA7219_ALC_CALIB_OVERFLOW_MASK) { dev_warn(component->dev, "ALC auto calibration failed with overflow\n"); - snd_soc_component_update_bits(component, DA7219_ALC_CTRL1, + ret = snd_soc_component_update_bits(component, DA7219_ALC_CTRL1, DA7219_ALC_OFFSET_EN_MASK | DA7219_ALC_SYNC_MODE_MASK, 0); + if (ret < 0) + return ret; } else { /* Enable DC offset cancellation, hybrid mode */ - snd_soc_component_update_bits(component, DA7219_ALC_CTRL1, + ret = snd_soc_component_update_bits(component, DA7219_ALC_CTRL1, DA7219_ALC_OFFSET_EN_MASK | DA7219_ALC_SYNC_MODE_MASK, DA7219_ALC_OFFSET_EN_MASK | DA7219_ALC_SYNC_MODE_MASK); + if (ret < 0) + return ret; } /* Restore input filter control register to original state */ - snd_soc_component_write(component, DA7219_ADC_L_CTRL, adc_ctrl); + ret = snd_soc_component_write(component, DA7219_ADC_L_CTRL, adc_ctrl); + if (ret) + return ret; /* Restore input mixer control registers to original state */ - snd_soc_component_write(component, DA7219_MIXIN_L_CTRL, mixin_ctrl); + ret = snd_soc_component_write(component, DA7219_MIXIN_L_CTRL, + mixin_ctrl); + if (ret) + return ret; /* Restore MIC control registers to original states */ - snd_soc_component_write(component, DA7219_MIC_1_CTRL, mic_ctrl); + ret = snd_soc_component_write(component, DA7219_MIC_1_CTRL, mic_ctrl); + if (ret) + return ret; + + return 0; } static int da7219_mixin_gain_put(struct snd_kcontrol *kcontrol, @@ -391,7 +427,7 @@ static int da7219_mixin_gain_put(struct snd_kcontrol *kcontrol, * make sure calibrated offsets are updated. */ if ((ret == 1) && (da7219->alc_en)) - da7219_alc_calib(component); + ret = da7219_alc_calib(component); return ret; } @@ -401,11 +437,14 @@ static int da7219_alc_sw_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); + int ret; /* Force ALC offset calibration if enabling ALC */ if ((ucontrol->value.integer.value[0]) && (!da7219->alc_en)) { - da7219_alc_calib(component); + ret = da7219_alc_calib(component); + if (ret) + return ret; da7219->alc_en = true; } else { da7219->alc_en = false; @@ -797,7 +836,7 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); - u8 pll_ctrl, pll_status; + unsigned int pll_ctrl, pll_status; int i = 0, ret; bool srm_lock = false; @@ -813,25 +852,35 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, return ret; } } else { - snd_soc_component_update_bits(component, + ret = snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, DA7219_DAI_CLK_EN_MASK, DA7219_DAI_CLK_EN_MASK); + if (ret < 0) + return ret; } } /* PC synchronised to DAI */ - snd_soc_component_update_bits(component, DA7219_PC_COUNT, + ret = snd_soc_component_update_bits(component, DA7219_PC_COUNT, DA7219_PC_FREERUN_MASK, 0); + if (ret < 0) + return ret; /* Slave mode, if SRM not enabled no need for status checks */ - pll_ctrl = snd_soc_component_read32(component, DA7219_PLL_CTRL); + ret = snd_soc_component_read(component, DA7219_PLL_CTRL, + &pll_ctrl); + if (ret) + return ret; if ((pll_ctrl & DA7219_PLL_MODE_MASK) != DA7219_PLL_MODE_SRM) return 0; /* Check SRM has locked */ do { - pll_status = snd_soc_component_read32(component, DA7219_PLL_SRM_STS); + ret = snd_soc_component_read(component, + DA7219_PLL_SRM_STS, &pll_status); + if (ret) + return ret; if (pll_status & DA7219_PLL_SRM_STS_SRM_LOCK) { srm_lock = true; } else { @@ -846,19 +895,23 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, return 0; case SND_SOC_DAPM_POST_PMD: /* PC free-running */ - snd_soc_component_update_bits(component, DA7219_PC_COUNT, + ret = snd_soc_component_update_bits(component, DA7219_PC_COUNT, DA7219_PC_FREERUN_MASK, DA7219_PC_FREERUN_MASK); + if (ret < 0) + return ret; /* Disable DAI clks if in master mode */ if (da7219->master) { if (da7219->dai_clks) clk_disable_unprepare(da7219->dai_clks); else - snd_soc_component_update_bits(component, + ret = snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, DA7219_DAI_CLK_EN_MASK, 0); + if (ret < 0) + return ret; } return 0; @@ -887,6 +940,7 @@ static int da7219_mixout_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); u8 hp_ctrl, min_gain_mask; + int ret; switch (w->reg) { case DA7219_MIXOUT_L_CTRL: @@ -904,15 +958,20 @@ static int da7219_mixout_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMD: /* Enable minimum gain on HP to avoid pops */ - snd_soc_component_update_bits(component, hp_ctrl, min_gain_mask, - min_gain_mask); + ret = snd_soc_component_update_bits(component, hp_ctrl, + min_gain_mask, min_gain_mask); + if (ret < 0) + return ret; msleep(DA7219_MIN_GAIN_DELAY); break; case SND_SOC_DAPM_POST_PMU: /* Remove minimum gain on HP */ - snd_soc_component_update_bits(component, hp_ctrl, min_gain_mask, 0); + ret = snd_soc_component_update_bits(component, hp_ctrl, + min_gain_mask, 0); + if (ret < 0) + return ret; break; } @@ -925,21 +984,30 @@ static int da7219_gain_ramp_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); + int ret; switch (event) { case SND_SOC_DAPM_PRE_PMU: case SND_SOC_DAPM_PRE_PMD: /* Ensure nominal gain ramping for DAPM sequence */ - da7219->gain_ramp_ctrl = - snd_soc_component_read32(component, DA7219_GAIN_RAMP_CTRL); - snd_soc_component_write(component, DA7219_GAIN_RAMP_CTRL, - DA7219_GAIN_RAMP_RATE_NOMINAL); + ret = snd_soc_component_read(component, + DA7219_GAIN_RAMP_CTRL, + &da7219->gain_ramp_ctrl); + if (ret) + return ret; + ret = snd_soc_component_write(component, + DA7219_GAIN_RAMP_CTRL, + DA7219_GAIN_RAMP_RATE_NOMINAL); + if (ret) + return ret; break; case SND_SOC_DAPM_POST_PMU: case SND_SOC_DAPM_POST_PMD: /* Restore previous gain ramp settings */ - snd_soc_component_write(component, DA7219_GAIN_RAMP_CTRL, + ret = snd_soc_component_write(component, DA7219_GAIN_RAMP_CTRL, da7219->gain_ramp_ctrl); + if (ret) + return ret; break; } @@ -1177,13 +1245,17 @@ static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (clk_id) { case DA7219_CLKSRC_MCLK_SQR: - snd_soc_component_update_bits(component, DA7219_PLL_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_PLL_CTRL, DA7219_PLL_MCLK_SQR_EN_MASK, DA7219_PLL_MCLK_SQR_EN_MASK); + if (ret < 0) + return ret; break; case DA7219_CLKSRC_MCLK: - snd_soc_component_update_bits(component, DA7219_PLL_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_PLL_CTRL, DA7219_PLL_MCLK_SQR_EN_MASK, 0); + if (ret < 0) + return ret; break; default: dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); @@ -1219,6 +1291,7 @@ int da7219_set_pll(struct snd_soc_component *component, int source, unsigned int u8 pll_frac_top, pll_frac_bot, pll_integer; u32 freq_ref; u64 frac_div; + int ret; /* Verify 2MHz - 54MHz MCLK provided, and set input divider */ if (da7219->mclk_rate < 2000000) { @@ -1252,9 +1325,11 @@ int da7219_set_pll(struct snd_soc_component *component, int source, unsigned int switch (source) { case DA7219_SYSCLK_MCLK: pll_ctrl |= DA7219_PLL_MODE_BYPASS; - snd_soc_component_update_bits(component, DA7219_PLL_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_PLL_CTRL, DA7219_PLL_INDIV_MASK | DA7219_PLL_MODE_MASK, pll_ctrl); + if (ret < 0) + return ret; return 0; case DA7219_SYSCLK_PLL: pll_ctrl |= DA7219_PLL_MODE_NORMAL; @@ -1275,12 +1350,23 @@ int da7219_set_pll(struct snd_soc_component *component, int source, unsigned int pll_frac_bot = (frac_div) & DA7219_BYTE_MASK; /* Write PLL config & dividers */ - snd_soc_component_write(component, DA7219_PLL_FRAC_TOP, pll_frac_top); - snd_soc_component_write(component, DA7219_PLL_FRAC_BOT, pll_frac_bot); - snd_soc_component_write(component, DA7219_PLL_INTEGER, pll_integer); - snd_soc_component_update_bits(component, DA7219_PLL_CTRL, + ret = snd_soc_component_write(component, DA7219_PLL_FRAC_TOP, + pll_frac_top); + if (ret) + return ret; + ret = snd_soc_component_write(component, DA7219_PLL_FRAC_BOT, + pll_frac_bot); + if (ret) + return ret; + ret = snd_soc_component_write(component, DA7219_PLL_INTEGER, + pll_integer); + if (ret) + return ret; + ret = snd_soc_component_update_bits(component, DA7219_PLL_CTRL, DA7219_PLL_INDIV_MASK | DA7219_PLL_MODE_MASK, pll_ctrl); + if (ret < 0) + return ret; return 0; } @@ -1304,6 +1390,7 @@ static int da7219_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct snd_soc_component *component = codec_dai->component; struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 dai_clk_mode = 0, dai_ctrl = 0; + int ret; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -1379,12 +1466,16 @@ static int da7219_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* By default 64 BCLKs per WCLK is supported */ dai_clk_mode |= DA7219_DAI_BCLKS_PER_WCLK_64; - snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, + ret = snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, DA7219_DAI_BCLKS_PER_WCLK_MASK | DA7219_DAI_CLK_POL_MASK | DA7219_DAI_WCLK_POL_MASK, dai_clk_mode); - snd_soc_component_update_bits(component, DA7219_DAI_CTRL, DA7219_DAI_FORMAT_MASK, - dai_ctrl); + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, DA7219_DAI_CTRL, + DA7219_DAI_FORMAT_MASK, dai_ctrl); + if (ret < 0) + return ret; return 0; } @@ -1398,15 +1489,22 @@ static int da7219_set_dai_tdm_slot(struct snd_soc_dai *dai, u8 dai_bclks_per_wclk; u16 offset; u32 frame_size; + int ret; /* No channels enabled so disable TDM, revert to 64-bit frames */ if (!tx_mask) { - snd_soc_component_update_bits(component, DA7219_DAI_TDM_CTRL, + ret = snd_soc_component_update_bits(component, + DA7219_DAI_TDM_CTRL, DA7219_DAI_TDM_CH_EN_MASK | DA7219_DAI_TDM_MODE_EN_MASK, 0); - snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, + DA7219_DAI_CLK_MODE, DA7219_DAI_BCLKS_PER_WCLK_MASK, DA7219_DAI_BCLKS_PER_WCLK_64); + if (ret < 0) + return ret; return 0; } @@ -1444,19 +1542,25 @@ static int da7219_set_dai_tdm_slot(struct snd_soc_dai *dai, return -EINVAL; } - snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, + ret = snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, DA7219_DAI_BCLKS_PER_WCLK_MASK, dai_bclks_per_wclk); + if (ret < 0) + return ret; offset = cpu_to_le16(rx_mask); - regmap_bulk_write(da7219->regmap, DA7219_DAI_OFFSET_LOWER, + ret = regmap_bulk_write(da7219->regmap, DA7219_DAI_OFFSET_LOWER, &offset, sizeof(offset)); + if (ret) + return ret; - snd_soc_component_update_bits(component, DA7219_DAI_TDM_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_DAI_TDM_CTRL, DA7219_DAI_TDM_CH_EN_MASK | DA7219_DAI_TDM_MODE_EN_MASK, (tx_mask << DA7219_DAI_TDM_CH_EN_SHIFT) | DA7219_DAI_TDM_MODE_EN_MASK); + if (ret < 0) + return ret; return 0; } @@ -1468,6 +1572,7 @@ static int da7219_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; u8 dai_ctrl = 0, fs; unsigned int channels; + int ret; switch (params_width(params)) { case 16: @@ -1533,11 +1638,15 @@ static int da7219_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - snd_soc_component_update_bits(component, DA7219_DAI_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_DAI_CTRL, DA7219_DAI_WORD_LENGTH_MASK | DA7219_DAI_CH_NUM_MASK, dai_ctrl); - snd_soc_component_write(component, DA7219_SR, fs); + if (ret < 0) + return ret; + ret = snd_soc_component_write(component, DA7219_SR, fs); + if (ret) + return ret; return 0; } @@ -1692,9 +1801,12 @@ static int da7219_set_bias_level(struct snd_soc_component *component, case SND_SOC_BIAS_STANDBY: if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) /* Master bias */ - snd_soc_component_update_bits(component, DA7219_REFERENCES, + ret = snd_soc_component_update_bits(component, + DA7219_REFERENCES, DA7219_BIAS_EN_MASK, DA7219_BIAS_EN_MASK); + if (ret < 0) + return ret; if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_PREPARE) { /* Remove MCLK */ @@ -1705,8 +1817,11 @@ static int da7219_set_bias_level(struct snd_soc_component *component, case SND_SOC_BIAS_OFF: /* Only disable master bias if we're not a wake-up source */ if (!da7219->wakeup_source) - snd_soc_component_update_bits(component, DA7219_REFERENCES, + ret = snd_soc_component_update_bits(component, + DA7219_REFERENCES, DA7219_BIAS_EN_MASK, 0); + if (ret < 0) + return ret; break; } @@ -1754,10 +1869,16 @@ static int da7219_handle_supplies(struct snd_soc_component *component) } /* Ensure device in active mode */ - snd_soc_component_write(component, DA7219_SYSTEM_ACTIVE, DA7219_SYSTEM_ACTIVE_MASK); + ret = snd_soc_component_write(component, DA7219_SYSTEM_ACTIVE, + DA7219_SYSTEM_ACTIVE_MASK); + if (ret) + return ret; /* Update IO voltage level range */ - snd_soc_component_write(component, DA7219_IO_CTRL, io_voltage_lvl); + ret = snd_soc_component_write(component, DA7219_IO_CTRL, + io_voltage_lvl); + if (ret) + return ret; return 0; } @@ -1768,10 +1889,13 @@ static int da7219_dai_clks_prepare(struct clk_hw *hw) struct da7219_priv *da7219 = container_of(hw, struct da7219_priv, dai_clks_hw); struct snd_soc_component *component = da7219->aad->component; + int ret; - snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, + ret = snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, DA7219_DAI_CLK_EN_MASK, DA7219_DAI_CLK_EN_MASK); + if (ret < 0) + return ret; return 0; } @@ -1781,9 +1905,12 @@ static void da7219_dai_clks_unprepare(struct clk_hw *hw) struct da7219_priv *da7219 = container_of(hw, struct da7219_priv, dai_clks_hw); struct snd_soc_component *component = da7219->aad->component; + int ret; - snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, + ret = snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, DA7219_DAI_CLK_EN_MASK, 0); + if (ret < 0) + dev_err(component->dev, "Failed to disable clk: %d\n", ret); } static int da7219_dai_clks_is_prepared(struct clk_hw *hw) @@ -1791,9 +1918,12 @@ static int da7219_dai_clks_is_prepared(struct clk_hw *hw) struct da7219_priv *da7219 = container_of(hw, struct da7219_priv, dai_clks_hw); struct snd_soc_component *component = da7219->aad->component; - u8 clk_reg; + unsigned int clk_reg; + int ret; - clk_reg = snd_soc_component_read32(component, DA7219_DAI_CLK_MODE); + ret = snd_soc_component_read(component, DA7219_DAI_CLK_MODE, &clk_reg); + if (ret) + return ret; return !!(clk_reg & DA7219_DAI_CLK_EN_MASK); } @@ -1844,10 +1974,11 @@ static void da7219_register_dai_clks(struct snd_soc_component *component) static inline void da7219_register_dai_clks(struct snd_soc_component *component) {} #endif /* CONFIG_COMMON_CLK */ -static void da7219_handle_pdata(struct snd_soc_component *component) +static int da7219_handle_pdata(struct snd_soc_component *component) { struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); struct da7219_pdata *pdata = da7219->pdata; + int ret; if (pdata) { u8 micbias_lvl = 0; @@ -1869,7 +2000,10 @@ static void da7219_handle_pdata(struct snd_soc_component *component) break; } - snd_soc_component_write(component, DA7219_MICBIAS_CTRL, micbias_lvl); + ret = snd_soc_component_write(component, DA7219_MICBIAS_CTRL, + micbias_lvl); + if (ret) + return ret; /* * Calculate delay required to compensate for DC offset in @@ -1884,11 +2018,15 @@ static void da7219_handle_pdata(struct snd_soc_component *component) case DA7219_MIC_AMP_IN_SEL_DIFF: case DA7219_MIC_AMP_IN_SEL_SE_P: case DA7219_MIC_AMP_IN_SEL_SE_N: - snd_soc_component_write(component, DA7219_MIC_1_SELECT, - pdata->mic_amp_in_sel); + ret = snd_soc_component_write(component, + DA7219_MIC_1_SELECT, + pdata->mic_amp_in_sel); + if (ret) + return ret; break; } } + return 0; } static struct reg_sequence da7219_rev_aa_patch[] = { @@ -1934,7 +2072,9 @@ static int da7219_probe(struct snd_soc_component *component) if (!da7219->pdata) da7219->pdata = da7219_fw_to_pdata(component); - da7219_handle_pdata(component); + ret = da7219_handle_pdata(component); + if (ret) + return ret; /* Check if MCLK provided */ da7219->mclk = devm_clk_get(component->dev, "mclk"); @@ -1948,36 +2088,57 @@ static int da7219_probe(struct snd_soc_component *component) } /* Default PC counter to free-running */ - snd_soc_component_update_bits(component, DA7219_PC_COUNT, DA7219_PC_FREERUN_MASK, - DA7219_PC_FREERUN_MASK); + ret = snd_soc_component_update_bits(component, DA7219_PC_COUNT, + DA7219_PC_FREERUN_MASK, DA7219_PC_FREERUN_MASK); + if (ret < 0) + return ret; /* Default gain ramping */ - snd_soc_component_update_bits(component, DA7219_MIXIN_L_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_MIXIN_L_CTRL, DA7219_MIXIN_L_AMP_RAMP_EN_MASK, DA7219_MIXIN_L_AMP_RAMP_EN_MASK); - snd_soc_component_update_bits(component, DA7219_ADC_L_CTRL, DA7219_ADC_L_RAMP_EN_MASK, - DA7219_ADC_L_RAMP_EN_MASK); - snd_soc_component_update_bits(component, DA7219_DAC_L_CTRL, DA7219_DAC_L_RAMP_EN_MASK, - DA7219_DAC_L_RAMP_EN_MASK); - snd_soc_component_update_bits(component, DA7219_DAC_R_CTRL, DA7219_DAC_R_RAMP_EN_MASK, - DA7219_DAC_R_RAMP_EN_MASK); - snd_soc_component_update_bits(component, DA7219_HP_L_CTRL, + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, DA7219_ADC_L_CTRL, + DA7219_ADC_L_RAMP_EN_MASK, DA7219_ADC_L_RAMP_EN_MASK); + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, DA7219_DAC_L_CTRL, + DA7219_DAC_L_RAMP_EN_MASK, DA7219_DAC_L_RAMP_EN_MASK); + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, DA7219_DAC_R_CTRL, + DA7219_DAC_R_RAMP_EN_MASK, DA7219_DAC_R_RAMP_EN_MASK); + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, DA7219_HP_L_CTRL, DA7219_HP_L_AMP_RAMP_EN_MASK, DA7219_HP_L_AMP_RAMP_EN_MASK); - snd_soc_component_update_bits(component, DA7219_HP_R_CTRL, + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, DA7219_HP_R_CTRL, DA7219_HP_R_AMP_RAMP_EN_MASK, DA7219_HP_R_AMP_RAMP_EN_MASK); + if (ret < 0) + return ret; /* Default minimum gain on HP to avoid pops during DAPM sequencing */ - snd_soc_component_update_bits(component, DA7219_HP_L_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_HP_L_CTRL, DA7219_HP_L_AMP_MIN_GAIN_EN_MASK, DA7219_HP_L_AMP_MIN_GAIN_EN_MASK); - snd_soc_component_update_bits(component, DA7219_HP_R_CTRL, + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, DA7219_HP_R_CTRL, DA7219_HP_R_AMP_MIN_GAIN_EN_MASK, DA7219_HP_R_AMP_MIN_GAIN_EN_MASK); + if (ret < 0) + return ret; /* Default infinite tone gen, start/stop by Kcontrol */ - snd_soc_component_write(component, DA7219_TONE_GEN_CYCLES, DA7219_BEEP_CYCLES_MASK); + ret = snd_soc_component_write(component, DA7219_TONE_GEN_CYCLES, + DA7219_BEEP_CYCLES_MASK); + if (ret) + return ret; /* Initialise AAD block */ ret = da7219_aad_init(component); @@ -2221,16 +2382,28 @@ static int da7219_i2c_probe(struct i2c_client *i2c, regcache_cache_bypass(da7219->regmap, true); /* Disable audio paths if still active from previous start */ - regmap_read(da7219->regmap, DA7219_SYSTEM_ACTIVE, &system_active); + ret = regmap_read(da7219->regmap, DA7219_SYSTEM_ACTIVE, &system_active); + if (ret) + return ret; if (system_active) { - regmap_write(da7219->regmap, DA7219_GAIN_RAMP_CTRL, + ret = regmap_write(da7219->regmap, DA7219_GAIN_RAMP_CTRL, DA7219_GAIN_RAMP_RATE_NOMINAL); - regmap_write(da7219->regmap, DA7219_SYSTEM_MODES_INPUT, 0x00); - regmap_write(da7219->regmap, DA7219_SYSTEM_MODES_OUTPUT, 0x01); + if (ret) + return ret; + ret = regmap_write(da7219->regmap, DA7219_SYSTEM_MODES_INPUT, + 0x00); + if (ret) + return ret; + ret = regmap_write(da7219->regmap, DA7219_SYSTEM_MODES_OUTPUT, + 0x01); + if (ret) + return ret; for (i = 0; i < DA7219_SYS_STAT_CHECK_RETRIES; ++i) { - regmap_read(da7219->regmap, DA7219_SYSTEM_STATUS, + ret = regmap_read(da7219->regmap, DA7219_SYSTEM_STATUS, &system_status); + if (ret) + return ret; if (!system_status) break; @@ -2239,13 +2412,19 @@ static int da7219_i2c_probe(struct i2c_client *i2c, } /* Soft reset component */ - regmap_write_bits(da7219->regmap, DA7219_ACCDET_CONFIG_1, + ret = regmap_write_bits(da7219->regmap, DA7219_ACCDET_CONFIG_1, DA7219_ACCDET_EN_MASK, 0); - regmap_write_bits(da7219->regmap, DA7219_CIF_CTRL, + if (ret) + return ret; + ret = regmap_write_bits(da7219->regmap, DA7219_CIF_CTRL, DA7219_CIF_REG_SOFT_RESET_MASK, DA7219_CIF_REG_SOFT_RESET_MASK); - regmap_write_bits(da7219->regmap, DA7219_SYSTEM_ACTIVE, + if (ret) + return ret; + ret = regmap_write_bits(da7219->regmap, DA7219_SYSTEM_ACTIVE, DA7219_SYSTEM_ACTIVE_MASK, 0); + if (ret) + return ret; regcache_cache_bypass(da7219->regmap, false); diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index 3a00686..78d2194 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -832,7 +832,7 @@ struct da7219_priv { bool alc_en; bool micbias_on_event; unsigned int mic_pga_delay; - u8 gain_ramp_ctrl; + unsigned int gain_ramp_ctrl; }; int da7219_set_pll(struct snd_soc_component *component, int source, unsigned int fout); -- 1.9.1

1 0

[alsa-devel] [PATCH 0/3] Add ALC294 quirks for ASUS laptops
by Jian-Hong Pan 05 Dec '18

05 Dec '18

Add Realtek ALC294 quirks for ASUS X542UN, UX533FD and UX433FN laptops. Chris Chiu (1): ALSA: hda/realtek: ALC294 mic and headset-mode fixups for ASUS X542UN Jian-Hong Pan (2): ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294 ALSA: hda/realtek: Enable audio jacks of ASUS UX433FN with ALC294 sound/pci/hda/patch_realtek.c | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) -- 2.11.0

2 5

[alsa-devel] [PATCH 1/4] ALSA: hda/realtek: ALC286 mic and headset-mode fixups for Acer Aspire U27-880
by Jian-Hong Pan 05 Dec '18

05 Dec '18

From: Chris Chiu <chiu(a)endlessm.com> Acer Aspire U27-880(AIO) with ALC286 codec can not detect headset mic and internal mic not working either. It needs the similar quirk like Sony laptops to fix headphone jack sensing and enables use of the internal microphone. Unfortunately jack sensing for the headset mic is still not working. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Signed-off-by: Daniel Drake <drake(a)endlessm.com> Signed-off-by: Chris Chiu <chiu(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c0b289ba397f..f21d52eb2ed3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5516,6 +5516,7 @@ enum { ALC221_FIXUP_HP_HEADSET_MIC, ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, ALC295_FIXUP_HP_AUTO_MUTE, + ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6393,6 +6394,15 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_auto_mute_via_amp, }, + [ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7071,6 +7081,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x19, 0x04a11040}, {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x17, 0x90170110}, + {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60120}, {0x14, 0x90170110}, -- 2.11.0

3 6

[alsa-devel] [PATCH linux-next v3 5/7] clk: renesas: r8a77965: Add ADG clock
by Jiada Wang 05 Dec '18

05 Dec '18

From: Takeshi Kihara <takeshi.kihara.df(a)renesas.com> This patch adds ADG clock to the R8A77965 SoC. Signed-off-by: Takeshi Kihara <takeshi.kihara.df(a)renesas.com> Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- drivers/clk/renesas/r8a77965-cpg-mssr.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/clk/renesas/r8a77965-cpg-mssr.c b/drivers/clk/renesas/r8a77965-cpg-mssr.c index 1fcc411502da..3ce8b0b31647 100644 --- a/drivers/clk/renesas/r8a77965-cpg-mssr.c +++ b/drivers/clk/renesas/r8a77965-cpg-mssr.c @@ -211,6 +211,7 @@ static const struct mssr_mod_clk r8a77965_mod_clks[] __initconst = { DEF_MOD("can-if0", 916, R8A77965_CLK_S3D4), DEF_MOD("i2c6", 918, R8A77965_CLK_S0D6), DEF_MOD("i2c5", 919, R8A77965_CLK_S0D6), + DEF_MOD("adg", 922, R8A77965_CLK_S0D1), DEF_MOD("i2c-dvfs", 926, R8A77965_CLK_CP), DEF_MOD("i2c4", 927, R8A77965_CLK_S0D6), DEF_MOD("i2c3", 928, R8A77965_CLK_S0D6), -- 2.19.2

1 0

[alsa-devel] [PATCH linux-next v3 0/7] clk: renesas: adg: add AVB Clock
by Jiada Wang 05 Dec '18

05 Dec '18

on R-Car SoCs there are AVB Counter Clocks, each clock has 12bits integral and 8 bits fractional dividers which operates with S0D1ϕ clock. This patch-set adds 'adg' clock to R-Car Soc, and changes adg driver to register avb clocks when clock-cells of rcar_sound node is 2. --- v3: - Removed clock id header file, added device tree bindings documentation instead to describe clock id - Instead of hardcode parent clock name in adg driver, refer to parent clock via clock specifier in device tree with 'adg' name - Added patch to add ADG in r8a77965 - Some other fixes v2: - expends adg register size and register avb clocks instead of add new clk-avb driver - Add adg clock v1: initial version Jiada Wang (2): ASoC: rsnd: add avb clocks clk: renesas: Add binding document for ADG Takeshi Kihara (5): clk: renesas: r8a7795: Add ADG clock clk: renesas: r8a7796: Add ADG clock clk: renesas: r8a77990: Add ADG clock clk: renesas: r8a77995: Add ADG clock clk: renesas: r8a77965: Add ADG clock .../clock/renesas,rcar-adg-clocks.txt | 24 ++ drivers/clk/renesas/r8a7795-cpg-mssr.c | 1 + drivers/clk/renesas/r8a7796-cpg-mssr.c | 1 + drivers/clk/renesas/r8a77965-cpg-mssr.c | 1 + drivers/clk/renesas/r8a77990-cpg-mssr.c | 1 + drivers/clk/renesas/r8a77995-cpg-mssr.c | 1 + sound/soc/sh/rcar/adg.c | 316 +++++++++++++++++- sound/soc/sh/rcar/gen.c | 9 + sound/soc/sh/rcar/rsnd.h | 9 + 9 files changed, 354 insertions(+), 9 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/renesas,rcar-adg-clocks.txt -- 2.19.2

1 4

[alsa-devel] axfer: check whether a terminal is referred for stdio
by GitHub pull_request - opened 05 Dec '18

05 Dec '18

alsa-project/alsa-utils pull request #5 was opened from takaswie: A reference to a terminal for standard input/output brings some troubles. For capture transmission, it can bring some terminal control codes and it's possible to make the terminal in disorder. For playback transmission, it can bring endless loop to read data for detection of type of container. This commit checks whether a terminal is referred for the standard input/output by a call isatty(3). When detecting a terminal, axfer run time prints message and goes to finish. Suggested-by: Takashi Iwai <tiwai(a)suse.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Request URL : https://github.com/alsa-project/alsa-utils/pull/5 Patch URL : https://github.com/alsa-project/alsa-utils/pull/5.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH] ASoC: pxa: remove raumfeld machine driver
by Daniel Mack 04 Dec '18

04 Dec '18

These boards are now fully ported to devicetree and make use of the simple-card driver, so the platform specific machine driver can be removed. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/Kconfig | 10 -- sound/soc/pxa/Makefile | 1 - sound/soc/pxa/raumfeld.c | 318 --------------------------------------- 3 files changed, 329 deletions(-) delete mode 100644 sound/soc/pxa/raumfeld.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 943b44de1464..a6968bfe1d2d 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -169,16 +169,6 @@ config SND_SOC_ZYLONITE Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. -config SND_SOC_RAUMFELD - tristate "SoC Audio support Raumfeld audio adapter" - depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) - depends on I2C && SPI_MASTER - select SND_PXA_SOC_SSP - select SND_SOC_CS4270 - select SND_SOC_AK4104 - help - Say Y if you want to add support for SoC audio on Raumfeld devices - config SND_PXA2XX_SOC_HX4700 tristate "SoC Audio support for HP iPAQ hx4700" depends on SND_PXA2XX_SOC && MACH_H4700 && I2C diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 5b265662f04f..0ab2a9dcb720 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -49,6 +49,5 @@ obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o -obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c deleted file mode 100644 index 111a907c4eb9..000000000000 --- a/sound/soc/pxa/raumfeld.c +++ /dev/null @@ -1,318 +0,0 @@ -/* - * raumfeld_audio.c -- SoC audio for Raumfeld audio devices - * - * Copyright (c) 2009 Daniel Mack <daniel(a)caiaq.de> - * - * based on code from: - * - * Wolfson Microelectronics PLC. - * Openedhand Ltd. - * Liam Girdwood <lrg(a)slimlogic.co.uk> - * 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. - */ - -#include <linux/module.h> -#include <linux/i2c.h> -#include <linux/delay.h> -#include <linux/gpio.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> - -#include "pxa-ssp.h" - -#define GPIO_SPDIF_RESET (38) -#define GPIO_MCLK_RESET (111) -#define GPIO_CODEC_RESET (120) - -static struct i2c_client *max9486_client; -static struct i2c_board_info max9486_hwmon_info = { - I2C_BOARD_INFO("max9485", 0x63), -}; - -#define MAX9485_MCLK_FREQ_112896 0x22 -#define MAX9485_MCLK_FREQ_122880 0x23 -#define MAX9485_MCLK_FREQ_225792 0x32 -#define MAX9485_MCLK_FREQ_245760 0x33 - -static void set_max9485_clk(char clk) -{ - i2c_master_send(max9486_client, &clk, 1); -} - -static void raumfeld_enable_audio(bool en) -{ - if (en) { - gpio_set_value(GPIO_MCLK_RESET, 1); - - /* wait some time to let the clocks become stable */ - msleep(100); - - gpio_set_value(GPIO_SPDIF_RESET, 1); - gpio_set_value(GPIO_CODEC_RESET, 1); - } else { - gpio_set_value(GPIO_MCLK_RESET, 0); - gpio_set_value(GPIO_SPDIF_RESET, 0); - gpio_set_value(GPIO_CODEC_RESET, 0); - } -} - -/* CS4270 */ -static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - /* set freq to 0 to enable all possible codec sample rates */ - return snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); -} - -static void raumfeld_cs4270_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - /* set freq to 0 to enable all possible codec sample rates */ - snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); -} - -static int raumfeld_cs4270_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; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int clk = 0; - int ret = 0; - - switch (params_rate(params)) { - case 44100: - set_max9485_clk(MAX9485_MCLK_FREQ_112896); - clk = 11289600; - break; - case 48000: - set_max9485_clk(MAX9485_MCLK_FREQ_122880); - clk = 12288000; - break; - case 88200: - set_max9485_clk(MAX9485_MCLK_FREQ_225792); - clk = 22579200; - break; - case 96000: - set_max9485_clk(MAX9485_MCLK_FREQ_245760); - clk = 24576000; - break; - default: - return -EINVAL; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); - if (ret < 0) - return ret; - - /* setup the CPU DAI */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); - if (ret < 0) - return ret; - - return 0; -} - -static const struct snd_soc_ops raumfeld_cs4270_ops = { - .startup = raumfeld_cs4270_startup, - .shutdown = raumfeld_cs4270_shutdown, - .hw_params = raumfeld_cs4270_hw_params, -}; - -static int raumfeld_analog_suspend(struct snd_soc_card *card) -{ - raumfeld_enable_audio(false); - return 0; -} - -static int raumfeld_analog_resume(struct snd_soc_card *card) -{ - raumfeld_enable_audio(true); - return 0; -} - -/* AK4104 */ - -static int raumfeld_ak4104_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 *cpu_dai = rtd->cpu_dai; - int ret = 0, clk = 0; - - switch (params_rate(params)) { - case 44100: - set_max9485_clk(MAX9485_MCLK_FREQ_112896); - clk = 11289600; - break; - case 48000: - set_max9485_clk(MAX9485_MCLK_FREQ_122880); - clk = 12288000; - break; - case 88200: - set_max9485_clk(MAX9485_MCLK_FREQ_225792); - clk = 22579200; - break; - case 96000: - set_max9485_clk(MAX9485_MCLK_FREQ_245760); - clk = 24576000; - break; - default: - return -EINVAL; - } - - /* setup the CPU DAI */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops raumfeld_ak4104_ops = { - .hw_params = raumfeld_ak4104_hw_params, -}; - -#define DAI_LINK_CS4270 \ -{ \ - .name = "CS4270", \ - .stream_name = "CS4270", \ - .cpu_dai_name = "pxa-ssp-dai.0", \ - .platform_name = "pxa-pcm-audio", \ - .codec_dai_name = "cs4270-hifi", \ - .codec_name = "cs4270.0-0048", \ - .dai_fmt = SND_SOC_DAIFMT_I2S | \ - SND_SOC_DAIFMT_NB_NF | \ - SND_SOC_DAIFMT_CBS_CFS, \ - .ops = &raumfeld_cs4270_ops, \ -} - -#define DAI_LINK_AK4104 \ -{ \ - .name = "ak4104", \ - .stream_name = "Playback", \ - .cpu_dai_name = "pxa-ssp-dai.1", \ - .codec_dai_name = "ak4104-hifi", \ - .platform_name = "pxa-pcm-audio", \ - .dai_fmt = SND_SOC_DAIFMT_I2S | \ - SND_SOC_DAIFMT_NB_NF | \ - SND_SOC_DAIFMT_CBS_CFS, \ - .ops = &raumfeld_ak4104_ops, \ - .codec_name = "spi0.0", \ -} - -static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] = { - DAI_LINK_CS4270, - DAI_LINK_AK4104, -}; - -static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = { - DAI_LINK_CS4270, -}; - -static struct snd_soc_card snd_soc_raumfeld_connector = { - .name = "Raumfeld Connector", - .owner = THIS_MODULE, - .dai_link = snd_soc_raumfeld_connector_dai, - .num_links = ARRAY_SIZE(snd_soc_raumfeld_connector_dai), - .suspend_post = raumfeld_analog_suspend, - .resume_pre = raumfeld_analog_resume, -}; - -static struct snd_soc_card snd_soc_raumfeld_speaker = { - .name = "Raumfeld Speaker", - .owner = THIS_MODULE, - .dai_link = snd_soc_raumfeld_speaker_dai, - .num_links = ARRAY_SIZE(snd_soc_raumfeld_speaker_dai), - .suspend_post = raumfeld_analog_suspend, - .resume_pre = raumfeld_analog_resume, -}; - -static struct platform_device *raumfeld_audio_device; - -static int __init raumfeld_audio_init(void) -{ - int ret; - - if (!machine_is_raumfeld_speaker() && - !machine_is_raumfeld_connector()) - return 0; - - max9486_client = i2c_new_device(i2c_get_adapter(0), - &max9486_hwmon_info); - - if (!max9486_client) - return -ENOMEM; - - set_max9485_clk(MAX9485_MCLK_FREQ_122880); - - /* Register analog device */ - raumfeld_audio_device = platform_device_alloc("soc-audio", 0); - if (!raumfeld_audio_device) - return -ENOMEM; - - if (machine_is_raumfeld_speaker()) - platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld_speaker); - - if (machine_is_raumfeld_connector()) - platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld_connector); - - ret = platform_device_add(raumfeld_audio_device); - if (ret < 0) { - platform_device_put(raumfeld_audio_device); - return ret; - } - - raumfeld_enable_audio(true); - return 0; -} - -static void __exit raumfeld_audio_exit(void) -{ - raumfeld_enable_audio(false); - - platform_device_unregister(raumfeld_audio_device); - - i2c_unregister_device(max9486_client); - - gpio_free(GPIO_MCLK_RESET); - gpio_free(GPIO_CODEC_RESET); - gpio_free(GPIO_SPDIF_RESET); -} - -module_init(raumfeld_audio_init); -module_exit(raumfeld_audio_exit); - -/* Module information */ -MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("Raumfeld audio SoC"); -MODULE_LICENSE("GPL"); -- 2.19.2

2 1

[alsa-devel] [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add pmc_plt_clk_0 quirk for Chromebook Clapper
by Hans de Goede 04 Dec '18

04 Dec '18

The Clapper model Chromebook uses pmc_plt_clk_0 instead of pmc_plt_clk_3 for the mclk, just like the Swanky model. This commit adds a DMI based quirk for this. This fixing audio no longer working on these devices after commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") that commit fixes us unnecessary keeping unused clocks on, but in case of the Clapper that was breaking audio support since we were not using the right clock in the cht_bsw_max98090_ti machine driver. Cc: stable(a)vger.kernel.org Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 9d9f6e41d81c..ad0c98383853 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -389,6 +389,13 @@ static struct snd_soc_card snd_soc_card_cht = { }; static const struct dmi_system_id cht_max98090_quirk_table[] = { + { + /* Clapper model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Clapper"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Swanky model Chromebook (Toshiba Chromebook 2) */ .matches = { -- 2.19.1

3 2

[alsa-devel] [PATCH] ASoC: rsnd: indicates Channel and Mode for debug
by Kuninori Morimoto 04 Dec '18

04 Dec '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> For TDM debug purpose, indicating Channel and Mode is very useful. This patch indicate it if it has #define DEBUG Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 6439a74..45ef295 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -329,7 +329,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, ssi->rate = rate; ssi->chan = chan; - dev_dbg(dev, "%s outputs %u Hz\n", rsnd_mod_name(mod), rate); + dev_dbg(dev, "%s outputs %d chan %u Hz\n", + rsnd_mod_name(mod), chan, rate); return 0; } @@ -360,6 +361,8 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { struct rsnd_dai *rdai = rsnd_io_to_rdai(io); + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); u32 cr_own = ssi->cr_own; @@ -371,6 +374,11 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, is_tdm = rsnd_runtime_is_tdm(io); is_tdm_split = rsnd_runtime_is_tdm_split(io); + if (is_tdm) + dev_dbg(dev, "TDM mode\n"); + if (is_tdm_split) + dev_dbg(dev, "TDM Split mode\n"); + cr_own |= FORCE | rsnd_rdai_width_to_swl(rdai); if (rdai->bit_clk_inv) -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rsnd: fixup mod ID for CTU regmap read/write
by Kuninori Morimoto 04 Dec '18

04 Dec '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit c16015f36cc12824 ("ASoC: rsnd: add .get_id/.get_id_sub") add new .get_id/.get_id_sub to indicate module ID/subID. It is used for SSIU and CTU. In SSIU case, subID indicates BUSIF, but register settings is based on SSIU ID. OTOH, in CTU case, subID indicates CTU channel, and register settings is based on it. This means regmap read/write function needs to care it. This patch fixup this issue. It can't play MIXed sound without this patch. Fixes: c16015f36cc12824 ("ASoC: rsnd: add .get_id/.get_id_sub") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ctu.c | 1 + sound/soc/sh/rcar/gen.c | 14 +++++++++++--- sound/soc/sh/rcar/rsnd.h | 1 + 3 files changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 4dda7d9..ff41a5e 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -317,6 +317,7 @@ static struct rsnd_mod_ops rsnd_ctu_ops = { .get_status = rsnd_mod_get_status, .id = rsnd_ctu_id, .id_sub = rsnd_ctu_id_sub, + .id_cmd = rsnd_mod_id_raw, }; struct rsnd_mod *rsnd_ctu_mod_get(struct rsnd_priv *priv, int id) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index ca63940..3283d67 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -71,6 +71,14 @@ static int rsnd_is_accessible_reg(struct rsnd_priv *priv, return 1; } +static int rsnd_mod_id_cmd(struct rsnd_mod *mod) +{ + if (mod->ops->id_cmd) + return mod->ops->id_cmd(mod); + + return rsnd_mod_id(mod); +} + u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg) { @@ -81,7 +89,7 @@ u32 rsnd_read(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return 0; - regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); + regmap_fields_read(gen->regs[reg], rsnd_mod_id_cmd(mod), &val); dev_dbg(dev, "r %s - %-18s (%4d) : %08x\n", rsnd_mod_name(mod), @@ -100,7 +108,7 @@ void rsnd_write(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return; - regmap_fields_force_write(gen->regs[reg], rsnd_mod_id(mod), data); + regmap_fields_force_write(gen->regs[reg], rsnd_mod_id_cmd(mod), data); dev_dbg(dev, "w %s - %-18s (%4d) : %08x\n", rsnd_mod_name(mod), @@ -117,7 +125,7 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, return; regmap_fields_force_update_bits(gen->regs[reg], - rsnd_mod_id(mod), mask, data); + rsnd_mod_id_cmd(mod), mask, data); dev_dbg(dev, "b %s - %-18s (%4d) : %08x/%08x\n", rsnd_mod_name(mod), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 3c57129..94d51fe 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -306,6 +306,7 @@ struct rsnd_mod_ops { enum rsnd_mod_type type); int (*id)(struct rsnd_mod *mod); int (*id_sub)(struct rsnd_mod *mod); + int (*id_cmd)(struct rsnd_mod *mod); }; struct rsnd_dai_stream; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rsnd: add missing TDM Split mode support for simple-card
by Kuninori Morimoto 04 Dec '18

04 Dec '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit f69f452243e4e1 ("ASoC: rsnd: add TDM Split mode support") added TDM Split mode support for rsnd driver. But, it cares audio-graph-card style only. We can't use TDM Split mode on simple-card style now. This patch fixup this issue. Fixes: f69f452243e4e1 ("ASoC: rsnd: add TDM Split mode support") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 53 ++++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 51 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 031cd5b..59e250c 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1031,6 +1031,49 @@ static const struct snd_soc_dai_ops rsnd_soc_dai_ops = { .prepare = rsnd_soc_dai_prepare, }; +static void rsnd_parse_connect_simple(struct rsnd_priv *priv, + struct device_node *dai_np, + int dai_i, int is_play) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_dai *rdai = rsnd_rdai_get(priv, dai_i); + struct rsnd_dai_stream *io = is_play ? + &rdai->playback : + &rdai->capture; + struct device_node *ssiu_np = rsnd_ssiu_of_node(priv); + struct device_node *np; + int i, j; + + if (!ssiu_np) + return; + + if (!rsnd_io_to_mod_ssi(io)) + return; + + /* + * This driver assumes that it is TDM Split mode + * if it includes ssiu node + */ + for (i = 0;; i++) { + struct device_node *node = is_play ? + of_parse_phandle(dai_np, "playback", i) : + of_parse_phandle(dai_np, "capture", i); + + if (!node) + break; + + j = 0; + for_each_child_of_node(ssiu_np, np) { + if (np == node) { + rsnd_flags_set(io, RSND_STREAM_TDM_SPLIT); + dev_dbg(dev, "%s is part of TDM Split\n", io->name); + } + j++; + } + + } +} + static void rsnd_parse_connect_graph(struct rsnd_priv *priv, struct rsnd_dai_stream *io, struct device_node *endpoint) @@ -1246,8 +1289,14 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) dai_i++; } } else { - for_each_child_of_node(dai_node, dai_np) - __rsnd_dai_probe(priv, dai_np, dai_i++); + for_each_child_of_node(dai_node, dai_np) { + __rsnd_dai_probe(priv, dai_np, dai_i); + if (rsnd_is_gen3(priv)) { + rsnd_parse_connect_simple(priv, dai_np, dai_i, 1); + rsnd_parse_connect_simple(priv, dai_np, dai_i, 0); + } + dai_i++; + } } return 0; -- 2.7.4

2 1

[alsa-devel] [PATCH v2 0/5] ASoC: merge audio-graph-scu into audio-graph - step3
by Kuninori Morimoto 04 Dec '18

04 Dec '18

Hi Mark audio-graph card can handle normal sound card, and audio-graph-scu card can handle DPCM sound card. But, we can't use these feature in same time now. This is v2 of step3 patches. I mentioned that step4 patch-set will be based on OF-graph posted patch, but, I could solve it without posted patch. There is no blocker anymore. Thus, I included blocked patch on step3. Kuninori Morimoto (5): ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting ASoC: audio-graph-scu-card: care link / dai count ASoC: audio-graph-scu-card: use cpu/codec pointer on graph_dai_props ASoC: audio-graph-scu-card: care multi DPCM codec_conf ASoC: audio-graph-card: use cpu/codec pointer on graph_dai_props sound/soc/generic/audio-graph-card.c | 57 ++++++---- sound/soc/generic/audio-graph-scu-card.c | 180 +++++++++++++++++++++++-------- sound/soc/generic/simple-card-utils.c | 22 +--- 3 files changed, 173 insertions(+), 86 deletions(-) -- 2.7.4

2 7

[alsa-devel] Applied "ASoC: audio-graph-scu-card: use cpu/codec pointer on graph_dai_props" to the asoc tree
by Mark Brown 04 Dec '18

04 Dec '18

The patch ASoC: audio-graph-scu-card: use cpu/codec pointer on graph_dai_props 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 a1af9858edae12eca8f77b152d17ee049835e57d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 30 Nov 2018 02:07:15 +0000 Subject: [PATCH] ASoC: audio-graph-scu-card: use cpu/codec pointer on graph_dai_props In DPCM case, it uses CPU-dummy / dummy-Codec dai links, and non DPCM case, it uses CPU-Codec dai links. Now, we want to merge audio-graph-card and audio-graph-scu-card. These sound cards are using silimar but not same logic on each functions. Then, of course we want to share same logic. To compromise, this patch uses cpu/codec pointer on audio-graph-scu-card. It is same logic with audio-graph-card, thus easy merging. This is prepare for merging audio card Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-scu-card.c | 72 ++++++++++++++++-------- 1 file changed, 49 insertions(+), 23 deletions(-) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index a0a28671948a..ed128d430060 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -26,12 +26,14 @@ struct graph_card_data { struct snd_soc_card snd_card; struct snd_soc_codec_conf codec_conf; struct graph_dai_props { - struct asoc_simple_dai dai; + struct asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; struct snd_soc_dai_link_component platform; struct asoc_simple_card_data adata; } *dai_props; struct snd_soc_dai_link *dai_link; + struct asoc_simple_dai *dais; struct asoc_simple_card_data adata; }; @@ -47,8 +49,17 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + int ret = 0; - return asoc_simple_card_clk_enable(&dai_props->dai); + ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); + if (ret) + return ret; + + ret = asoc_simple_card_clk_enable(dai_props->codec_dai); + if (ret) + asoc_simple_card_clk_disable(dai_props->cpu_dai); + + return ret; } static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) @@ -57,7 +68,9 @@ static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - asoc_simple_card_clk_disable(&dai_props->dai); + asoc_simple_card_clk_disable(dai_props->cpu_dai); + + asoc_simple_card_clk_disable(dai_props->codec_dai); } static const struct snd_soc_ops asoc_graph_card_ops = { @@ -68,18 +81,20 @@ static const struct snd_soc_ops asoc_graph_card_ops = { static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) { struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai; - struct snd_soc_dai_link *dai_link; - struct graph_dai_props *dai_props; - int num = rtd->num; + struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + int ret = 0; - dai_link = graph_priv_to_link(priv, num); - dai_props = graph_priv_to_props(priv, num); - dai = dai_link->dynamic ? - rtd->cpu_dai : - rtd->codec_dai; + ret = asoc_simple_card_init_dai(rtd->codec_dai, + dai_props->codec_dai); + if (ret < 0) + return ret; - return asoc_simple_card_init_dai(dai, &dai_props->dai); + ret = asoc_simple_card_init_dai(rtd->cpu_dai, + dai_props->cpu_dai); + if (ret < 0) + return ret; + + return 0; } static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -99,14 +114,15 @@ static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, struct device_node *codec_ep, struct graph_card_data *priv, - int idx, int is_fe) + int *dai_idx, int link_idx, int is_fe) { struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, idx); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, idx); + struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); struct snd_soc_card *card = graph_priv_to_card(priv); struct device_node *ep = is_fe ? cpu_ep : codec_ep; struct device_node *node = of_graph_get_port_parent(ep); + struct asoc_simple_dai *dai; int ret; if (is_fe) { @@ -122,11 +138,14 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; + dai = + dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + ret = asoc_simple_card_parse_graph_cpu(ep, dai_link); if (ret) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, &dai_props->dai); + ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai); if (ret < 0) return ret; @@ -149,11 +168,14 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_graph_card_be_hw_params_fixup; + dai = + dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + ret = asoc_simple_card_parse_graph_codec(ep, dai_link); if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, &dai_props->dai); + ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai); if (ret < 0) return ret; @@ -178,7 +200,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); - ret = asoc_simple_card_of_parse_tdm(ep, &dai_props->dai); + ret = asoc_simple_card_of_parse_tdm(ep, dai); if (ret) return ret; @@ -210,7 +232,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device_node *codec_ep; struct device_node *codec_port; struct device_node *codec_port_old; - int dai_idx, ret; + int dai_idx, link_idx, ret; int rc, codec; if (!node) @@ -232,6 +254,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) * see asoc_graph_get_dais_count */ + link_idx = 0; dai_idx = 0; codec_port_old = NULL; for (codec = 0; codec < 2; codec++) { @@ -260,8 +283,8 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) } ret = asoc_graph_card_dai_link_of(cpu_ep, codec_ep, - priv, dai_idx++, - !codec); + priv, &dai_idx, + link_idx++, !codec); if (ret < 0) goto parse_of_err; } @@ -364,6 +387,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) struct graph_card_data *priv; struct snd_soc_dai_link *dai_link; struct graph_dai_props *dai_props; + struct asoc_simple_dai *dais; struct device *dev = &pdev->dev; struct snd_soc_card *card; int lnum = 0, dnum = 0, cnum = 0; @@ -380,7 +404,8 @@ static int asoc_graph_card_probe(struct platform_device *pdev) dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); - if (!dai_props || !dai_link) + dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); + if (!dai_props || !dai_link || !dais) return -ENOMEM; /* @@ -397,6 +422,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) priv->dai_props = dai_props; priv->dai_link = dai_link; + priv->dais = dais; /* Init snd_soc_card */ card = graph_priv_to_card(priv); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: audio-graph-scu-card: care multi DPCM codec_conf" to the asoc tree
by Mark Brown 04 Dec '18

04 Dec '18

The patch ASoC: audio-graph-scu-card: care multi DPCM codec_conf 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 7c2ae7981d73f9e4f10ed9d1dcb5bc6056189b30 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 30 Nov 2018 02:07:32 +0000 Subject: [PATCH] ASoC: audio-graph-scu-card: care multi DPCM codec_conf Current audio-graph-scu-card didn't care about codec_conf for multi DPCM case. This patch cares it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-scu-card.c | 40 +++++++++++++++++------- 1 file changed, 29 insertions(+), 11 deletions(-) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index ed128d430060..e1b192ea147b 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -24,17 +24,18 @@ struct graph_card_data { struct snd_soc_card snd_card; - struct snd_soc_codec_conf codec_conf; struct graph_dai_props { struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; struct snd_soc_dai_link_component platform; struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_dai *dais; struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; }; #define graph_priv_to_card(priv) (&(priv)->snd_card) @@ -114,7 +115,8 @@ static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, struct device_node *codec_ep, struct graph_card_data *priv, - int *dai_idx, int link_idx, int is_fe) + int *dai_idx, int link_idx, + int *conf_idx, int is_fe) { struct device *dev = graph_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); @@ -159,6 +161,8 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, asoc_simple_card_canonicalize_cpu(dai_link, of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); } else { + struct snd_soc_codec_conf *cconf; + /* FE is dummy */ dai_link->cpu_of_node = NULL; dai_link->cpu_dai_name = "snd-soc-dummy-dai"; @@ -171,6 +175,9 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, dai = dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + cconf = + dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; + ret = asoc_simple_card_parse_graph_codec(ep, dai_link); if (ret < 0) return ret; @@ -186,14 +193,12 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, return ret; /* check "prefix" from top node */ - snd_soc_of_parse_audio_prefix(card, - &priv->codec_conf, + snd_soc_of_parse_audio_prefix(card, cconf, dai_link->codecs->of_node, "prefix"); /* check "prefix" from each node if top doesn't have */ - if (!priv->codec_conf.of_node) - snd_soc_of_parse_node_prefix(node, - &priv->codec_conf, + if (!cconf->of_node) + snd_soc_of_parse_node_prefix(node, cconf, dai_link->codecs->of_node, PREFIX "prefix"); } @@ -232,7 +237,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device_node *codec_ep; struct device_node *codec_port; struct device_node *codec_port_old; - int dai_idx, link_idx, ret; + int dai_idx, link_idx, conf_idx, ret; int rc, codec; if (!node) @@ -256,6 +261,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) link_idx = 0; dai_idx = 0; + conf_idx = 0; codec_port_old = NULL; for (codec = 0; codec < 2; codec++) { /* @@ -284,7 +290,8 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) ret = asoc_graph_card_dai_link_of(cpu_ep, codec_ep, priv, &dai_idx, - link_idx++, !codec); + link_idx++, &conf_idx, + !codec); if (ret < 0) goto parse_of_err; } @@ -294,6 +301,14 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) if (ret) goto parse_of_err; + if ((card->num_links != link_idx) || + (card->num_configs != conf_idx)) { + dev_err(dev, "dai_link or codec_config wrong (%d/%d, %d/%d)\n", + card->num_links, link_idx, card->num_configs, conf_idx); + ret = -EINVAL; + goto parse_of_err; + } + ret = 0; parse_of_err: @@ -390,6 +405,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) struct asoc_simple_dai *dais; struct device *dev = &pdev->dev; struct snd_soc_card *card; + struct snd_soc_codec_conf *cconf; int lnum = 0, dnum = 0, cnum = 0; int ret, i; @@ -405,6 +421,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); + cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); if (!dai_props || !dai_link || !dais) return -ENOMEM; @@ -423,6 +440,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) priv->dai_props = dai_props; priv->dai_link = dai_link; priv->dais = dais; + priv->codec_conf = cconf; /* Init snd_soc_card */ card = graph_priv_to_card(priv); @@ -430,8 +448,8 @@ static int asoc_graph_card_probe(struct platform_device *pdev) card->dev = dev; card->dai_link = priv->dai_link; card->num_links = lnum; - card->codec_conf = &priv->codec_conf; - card->num_configs = 1; + card->codec_conf = cconf; + card->num_configs = cnum; ret = asoc_graph_card_parse_of(priv); if (ret < 0) { -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: audio-graph-card: use cpu/codec pointer on graph_dai_props" to the asoc tree
by Mark Brown 04 Dec '18

04 Dec '18

The patch ASoC: audio-graph-card: use cpu/codec pointer on graph_dai_props 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 0e3460bceedc98176684602a34659d51cd8bd4ce Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 30 Nov 2018 02:07:48 +0000 Subject: [PATCH] ASoC: audio-graph-card: use cpu/codec pointer on graph_dai_props In DPCM case, it uses CPU-dummy / dummy-Codec dai links, and non DPCM case, it uses CPU-Codec dai links. Now, we want to merge audio-graph-card and audio-graph-scu-card. These sound cards are using silimar but not same logic on each functions. Then, of course we want to share same logic. To compromise, this patch uses cpu/codec pointer on audio-graph-card. It is same logic with audio-graph-scu-card, thus easy merging. This is prepare for merging audio card Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 57 +++++++++++++++++----------- 1 file changed, 34 insertions(+), 23 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 1b158cc2df88..1da9532ef897 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -23,8 +23,8 @@ struct graph_card_data { struct snd_soc_card snd_card; struct graph_dai_props { - struct asoc_simple_dai cpu_dai; - struct asoc_simple_dai codec_dai; + struct asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; /* single codec */ struct snd_soc_dai_link_component platform; unsigned int mclk_fs; @@ -33,6 +33,7 @@ struct graph_card_data { struct asoc_simple_jack hp_jack; struct asoc_simple_jack mic_jack; struct snd_soc_dai_link *dai_link; + struct asoc_simple_dai *dais; struct gpio_desc *pa_gpio; }; @@ -75,13 +76,13 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream) struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); int ret; - ret = asoc_simple_card_clk_enable(&dai_props->cpu_dai); + ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); if (ret) return ret; - ret = asoc_simple_card_clk_enable(&dai_props->codec_dai); + ret = asoc_simple_card_clk_enable(dai_props->codec_dai); if (ret) - asoc_simple_card_clk_disable(&dai_props->cpu_dai); + asoc_simple_card_clk_disable(dai_props->cpu_dai); return ret; } @@ -92,9 +93,9 @@ static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - asoc_simple_card_clk_disable(&dai_props->cpu_dai); + asoc_simple_card_clk_disable(dai_props->cpu_dai); - asoc_simple_card_clk_disable(&dai_props->codec_dai); + asoc_simple_card_clk_disable(dai_props->codec_dai); } static int asoc_graph_card_hw_params(struct snd_pcm_substream *substream, @@ -139,17 +140,16 @@ static const struct snd_soc_ops asoc_graph_card_ops = { static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) { struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *codec = rtd->codec_dai; - struct snd_soc_dai *cpu = rtd->cpu_dai; - struct graph_dai_props *dai_props = - graph_priv_to_props(priv, rtd->num); - int ret; + struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + int ret = 0; - ret = asoc_simple_card_init_dai(codec, &dai_props->codec_dai); + ret = asoc_simple_card_init_dai(rtd->codec_dai, + dai_props->codec_dai); if (ret < 0) return ret; - ret = asoc_simple_card_init_dai(cpu, &dai_props->cpu_dai); + ret = asoc_simple_card_init_dai(rtd->cpu_dai, + dai_props->cpu_dai); if (ret < 0) return ret; @@ -158,17 +158,22 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, struct graph_card_data *priv, - int idx) + int *dai_idx, int link_idx) { struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, idx); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, idx); - struct asoc_simple_dai *cpu_dai = &dai_props->cpu_dai; - struct asoc_simple_dai *codec_dai = &dai_props->codec_dai; + struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); + struct asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; struct device_node *cpu_ep = of_get_next_child(cpu_port, NULL); struct device_node *codec_ep = of_graph_get_remote_endpoint(cpu_ep); int ret; + cpu_dai = + dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + codec_dai = + dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, NULL, &dai_link->dai_fmt); if (ret < 0) @@ -231,8 +236,8 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device *dev = graph_priv_to_dev(priv); struct snd_soc_card *card = graph_priv_to_card(priv); struct device_node *node = dev->of_node; - int rc, idx = 0; - int ret; + int rc, ret; + int link_idx, dai_idx; ret = asoc_simple_card_of_parse_widgets(card, NULL); if (ret < 0) @@ -245,8 +250,11 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) /* Factor to mclk, used in hw_params() */ of_property_read_u32(node, "mclk-fs", &priv->mclk_fs); + link_idx = 0; + dai_idx = 0; of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { - ret = asoc_graph_card_dai_link_of(it.node, priv, idx++); + ret = asoc_graph_card_dai_link_of(it.node, priv, + &dai_idx, link_idx++); if (ret < 0) { of_node_put(it.node); @@ -291,6 +299,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) struct graph_card_data *priv; struct snd_soc_dai_link *dai_link; struct graph_dai_props *dai_props; + struct asoc_simple_dai *dais; struct device *dev = &pdev->dev; struct snd_soc_card *card; int num, ret, i; @@ -306,7 +315,8 @@ static int asoc_graph_card_probe(struct platform_device *pdev) dai_props = devm_kcalloc(dev, num, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, num, sizeof(*dai_link), GFP_KERNEL); - if (!dai_props || !dai_link) + dais = devm_kcalloc(dev, num * 2, sizeof(*dais), GFP_KERNEL); + if (!dai_props || !dai_link || !dais) return -ENOMEM; /* @@ -330,6 +340,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) priv->dai_props = dai_props; priv->dai_link = dai_link; + priv->dais = dais; /* Init snd_soc_card */ card = graph_priv_to_card(priv); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: max98373: Added max98373_reset for stable amp reset" to the asoc tree
by Mark Brown 04 Dec '18

04 Dec '18

The patch ASoC: max98373: Added max98373_reset for stable amp reset 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 20f2ab247d3b787af91c1aa5eb27c5061744c154 Mon Sep 17 00:00:00 2001 From: Ryan Lee <RyanS.Lee(a)maximintegrated.com> Date: Fri, 30 Nov 2018 03:21:09 +0000 Subject: [PATCH] ASoC: max98373: Added max98373_reset for stable amp reset This patch added max98373_reset function to avoid amp software reset failure and code duplication. Reset verification step has been added for stable amp reset and it repeats verification maximum 3 times when it is failed. Chip revision ID is available when the amp is in the idle state which means software reset is completed well. Additional 10ms delay was added for every retrial and maximum 30ms delay can be applied. Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98373.c | 35 +++++++++++++++++++++++++++++------ 1 file changed, 29 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index a09d01318f79..9c8616a7b61c 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -724,14 +724,39 @@ static struct snd_soc_dai_driver max98373_dai[] = { } }; +static void max98373_reset(struct max98373_priv *max98373, struct device *dev) +{ + int ret, reg, count; + + /* Software Reset */ + ret = regmap_update_bits(max98373->regmap, + MAX98373_R2000_SW_RESET, + MAX98373_SOFT_RESET, + MAX98373_SOFT_RESET); + if (ret) + dev_err(dev, "Reset command failed. (ret:%d)\n", ret); + + count = 0; + while (count < 3) { + usleep_range(10000, 11000); + /* Software Reset Verification */ + ret = regmap_read(max98373->regmap, + MAX98373_R21FF_REV_ID, &reg); + if (!ret) { + dev_info(dev, "Reset completed (retry:%d)\n", count); + return; + } + count++; + } + dev_err(dev, "Reset failed. (ret:%d)\n", ret); +} + static int max98373_probe(struct snd_soc_component *component) { struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); /* Software Reset */ - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, component->dev); /* IV default slot configuration */ regmap_write(max98373->regmap, @@ -818,9 +843,7 @@ static int max98373_resume(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, dev); regcache_cache_only(max98373->regmap, false); regcache_sync(max98373->regmap); return 0; -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH v9 0/3] Huawei laptops
by Ayman Bagabas 04 Dec '18

04 Dec '18

This patch set is based on new audio LED triggers in branch topic/leds-trigger from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Changes from v8: * Explicitly use NULL in acpi_get_handle * Drop __initconst from huawei_wmi_keymap Changes from v7: * Switch to wmi_driver * Use devm to allocate and manage devices * Skip registering LED subsystem if a ACPI controller method was not found Changes from v6: * Use audio LED triggers patch set * Use KEY_CONFIG (XF86Tools) instead of KEY_PROG1. In Windows, the key is used to launch Huawei PC manager. XF86Tools is used by default on most desktop environments i.e. Gnome. Changes from v5: * Review tags Changes from v4: * Consistency in file names * How module would be enabled (Kconfig) * Match license in SPDX and MODULE_LICENSE Changes from v3: * Code formatting Changes from v2: * Support for Huawei MBX * Style and formatting issues Ayman Bagabas (3): ALSA: hda: fix front speakers on Huawei MBXP. x86: add support for Huawei WMI hotkeys. ALSA: hda: add support for Huawei WMI micmute LED drivers/platform/x86/Kconfig | 16 +++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/huawei-wmi.c | 226 ++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 22 +++ 4 files changed, 265 insertions(+) create mode 100644 drivers/platform/x86/huawei-wmi.c -- 2.19.2

5 8

[alsa-devel] [PATCH linux-next v2 6/6] ASoC: rsnd: add avb clocks
by jiada_wang＠mentor.com 04 Dec '18

04 Dec '18

From: Jiada Wang <jiada_wang(a)mentor.com> There are AVB Counter Clocks in ADG, each clock has 12bits integral and 8 bits fractional dividers which operates with S0D1ϕ clock. This patch registers 8 AVB Counter Clocks when clock-cells of rcar_sound node is 2, Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/adg.c | 306 +++++++++++++++++++++++++++++++++++++-- sound/soc/sh/rcar/gen.c | 9 ++ sound/soc/sh/rcar/rsnd.h | 9 ++ 3 files changed, 315 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 6768a66588eb..2c03d420ae76 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -5,6 +5,8 @@ // Copyright (C) 2013 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> #include <linux/clk-provider.h> +#include <linux/of_address.h> +#include <dt-bindings/clock/renesas-adg.h> #include "rsnd.h" #define CLKA 0 @@ -21,13 +23,33 @@ #define BRRx_MASK(x) (0x3FF & x) +#define ADG_CLK_NAME "adg" +#define AVB_CLK_NAME "avb" +#define AVB_CLK_NUM 8 +#define AVB_CLK_NAME_SIZE 10 +#define AVB_MAX_RATE 25000000 +#define AVB_DIV_EN_COM BIT(31) +#define AVB_DIV_MASK 0x3ffff +#define AVB_MAX_DIV 0x3ffc0 + static struct rsnd_mod_ops adg_ops = { .name = "adg", }; +struct clk_avb { + struct clk_hw hw; + unsigned int idx; + struct rsnd_mod *mod; + /* lock reg access */ + spinlock_t *lock; +}; + +#define to_clk_avb(_hw) container_of(_hw, struct clk_avb, hw) + struct rsnd_adg { struct clk *clk[CLKMAX]; struct clk *clkout[CLKOUTMAX]; + struct clk *clkavb[AVB_CLK_NUM]; struct clk_onecell_data onecell; struct rsnd_mod mod; u32 flags; @@ -37,6 +59,7 @@ struct rsnd_adg { int rbga_rate_for_441khz; /* RBGA */ int rbgb_rate_for_48khz; /* RBGB */ + spinlock_t avb_lock; }; #define LRCLK_ASYNC (1 << 0) @@ -408,6 +431,239 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) } } +static struct clk *rsnd_adg_clk_src_twocell_get(struct of_phandle_args *clkspec, + void *data) +{ + unsigned int clkidx = clkspec->args[1]; + struct rsnd_adg *adg = data; + const char *type; + struct clk *clk; + + switch (clkspec->args[0]) { + case ADG_FIX: + type = "fixed"; + if (clkidx >= CLKOUTMAX) { + pr_err("Invalid %s clock index %u\n", type, + clkidx); + return ERR_PTR(-EINVAL); + } + clk = adg->clkout[clkidx]; + break; + case ADG_AVB: + type = "avb"; + if (clkidx >= AVB_CLK_NUM) { + pr_err("Invalid %s clock index %u\n", type, + clkidx); + return ERR_PTR(-EINVAL); + } + clk = adg->clkavb[clkidx]; + break; + default: + pr_err("Invalid ADG clock type %u\n", clkspec->args[0]); + return ERR_PTR(-EINVAL); + } + + return clk; +} + +static void clk_avb_div_write(struct rsnd_mod *mod, u32 data, int idx) +{ + switch (idx) { + case 0: + rsnd_mod_write(mod, AVB_CLK_DIV0, data); + break; + case 1: + rsnd_mod_write(mod, AVB_CLK_DIV1, data); + break; + case 2: + rsnd_mod_write(mod, AVB_CLK_DIV2, data); + break; + case 3: + rsnd_mod_write(mod, AVB_CLK_DIV3, data); + break; + case 4: + rsnd_mod_write(mod, AVB_CLK_DIV4, data); + break; + case 5: + rsnd_mod_write(mod, AVB_CLK_DIV5, data); + break; + case 6: + rsnd_mod_write(mod, AVB_CLK_DIV6, data); + break; + case 7: + rsnd_mod_write(mod, AVB_CLK_DIV7, data); + break; + } +} + +static u32 clk_avb_div_read(struct rsnd_mod *mod, int idx) +{ + u32 val = 0; + + switch (idx) { + case 0: + val = rsnd_mod_read(mod, AVB_CLK_DIV0); + break; + case 1: + val = rsnd_mod_read(mod, AVB_CLK_DIV1); + break; + case 2: + val = rsnd_mod_read(mod, AVB_CLK_DIV2); + break; + case 3: + val = rsnd_mod_read(mod, AVB_CLK_DIV3); + break; + case 4: + val = rsnd_mod_read(mod, AVB_CLK_DIV4); + break; + case 5: + val = rsnd_mod_read(mod, AVB_CLK_DIV5); + break; + case 6: + val = rsnd_mod_read(mod, AVB_CLK_DIV6); + break; + case 7: + val = rsnd_mod_read(mod, AVB_CLK_DIV7); + break; + } + + return val; +} + +static int clk_avb_is_enabled(struct clk_hw *hw) +{ + struct clk_avb *avb = to_clk_avb(hw); + + return rsnd_mod_read(avb->mod, AVB_CLK_CONFIG) & BIT(avb->idx); +} + +static int clk_avb_enabledisable(struct clk_hw *hw, int enable) +{ + struct clk_avb *avb = to_clk_avb(hw); + u32 val; + + spin_lock(avb->lock); + + val = rsnd_mod_read(avb->mod, AVB_CLK_CONFIG); + if (enable) + val |= BIT(avb->idx); + else + val &= ~BIT(avb->idx); + rsnd_mod_write(avb->mod, AVB_CLK_CONFIG, val); + + spin_unlock(avb->lock); + + return 0; +} + +static int clk_avb_enable(struct clk_hw *hw) +{ + return clk_avb_enabledisable(hw, 1); +} + +static void clk_avb_disable(struct clk_hw *hw) +{ + clk_avb_enabledisable(hw, 0); +} + +static unsigned long clk_avb_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct clk_avb *avb = to_clk_avb(hw); + u32 div; + + div = clk_avb_div_read(avb->mod, avb->idx) & AVB_DIV_MASK; + if (!div) + return parent_rate; + + return parent_rate * 32 / div; +} + +static unsigned int clk_avb_calc_div(unsigned long rate, + unsigned long parent_rate) +{ + unsigned int div; + + if (!rate) + rate = 1; + + if (rate > AVB_MAX_RATE) + rate = AVB_MAX_RATE; + + div = DIV_ROUND_CLOSEST(parent_rate * 32, rate); + + if (div > AVB_MAX_DIV) + div = AVB_MAX_DIV; + + return div; +} + +static long clk_avb_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + unsigned int div = clk_avb_calc_div(rate, *parent_rate); + + return (*parent_rate * 32) / div; +} + +static int clk_avb_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct clk_avb *avb = to_clk_avb(hw); + unsigned int div = clk_avb_calc_div(rate, parent_rate); + u32 val; + + val = clk_avb_div_read(avb->mod, avb->idx) & ~AVB_DIV_MASK; + clk_avb_div_write(avb->mod, val | div, avb->idx); + + return 0; +} + +static const struct clk_ops clk_avb_ops = { + .enable = clk_avb_enable, + .disable = clk_avb_disable, + .is_enabled = clk_avb_is_enabled, + .recalc_rate = clk_avb_recalc_rate, + .round_rate = clk_avb_round_rate, + .set_rate = clk_avb_set_rate, +}; + +static struct clk *clk_register_avb(struct device *dev, struct rsnd_mod *mod, + unsigned int id, spinlock_t *lock) +{ + struct clk_init_data init; + struct clk_avb *avb; + struct clk *clk; + char name[AVB_CLK_NAME_SIZE]; + const char *parent_name = ADG_CLK_NAME; + + avb = devm_kzalloc(dev, sizeof(*avb), GFP_KERNEL); + if (!avb) + return ERR_PTR(-ENOMEM); + + snprintf(name, AVB_CLK_NAME_SIZE, "%s%u", AVB_CLK_NAME, id); + + avb->idx = id; + avb->lock = lock; + avb->mod = mod; + + /* Register the clock. */ + init.name = name; + init.ops = &clk_avb_ops; + init.flags = CLK_IS_BASIC; + init.parent_names = &parent_name; + init.num_parents = 1; + + avb->hw.init = &init; + + /* init DIV to a valid state */ + clk_avb_div_write(avb->mod, avb->idx, AVB_MAX_DIV); + + clk = devm_clk_register(dev, &avb->hw); + + return clk; +} + static void rsnd_adg_get_clkin(struct rsnd_priv *priv, struct rsnd_adg *adg) { @@ -436,6 +692,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, unsigned long req_48kHz_rate, req_441kHz_rate; int i, req_size; const char *parent_clk_name = NULL; + struct rsnd_mod *mod = rsnd_mod_get(adg); static const char * const clkout_name[] = { [CLKOUT] = "audio_clkout", [CLKOUT1] = "audio_clkout1", @@ -540,21 +797,23 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, */ of_property_read_u32(np, "#clock-cells", &count); - /* - * for clkout - */ - if (!count) { + + switch (count) { + case 0: + /* + * for clkout + */ clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], parent_clk_name, 0, req_rate[0]); if (!IS_ERR(clk)) { adg->clkout[CLKOUT] = clk; of_clk_add_provider(np, of_clk_src_simple_get, clk); } - } - /* - * for clkout0/1/2/3 - */ - else { + break; + case 1: + /* + * for clkout0/1/2/3 + */ for (i = 0; i < CLKOUTMAX; i++) { clk = clk_register_fixed_rate(dev, clkout_name[i], parent_clk_name, 0, @@ -566,6 +825,33 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, adg->onecell.clk_num = CLKOUTMAX; of_clk_add_provider(np, of_clk_src_onecell_get, &adg->onecell); + break; + case 2: + /* + * for clkout0/1/2/3 and avb clocks + */ + for (i = 0; i < CLKOUTMAX; i++) { + clk = clk_register_fixed_rate(dev, clkout_name[i], + parent_clk_name, 0, + req_rate[0]); + if (!IS_ERR(clk)) + adg->clkout[i] = clk; + } + + for (i = 0; i < AVB_CLK_NUM; i++) { + clk = clk_register_avb(dev, mod, i, &adg->avb_lock); + if (!IS_ERR(clk)) + adg->clkavb[i] = clk; + } + + of_clk_add_provider(np, rsnd_adg_clk_src_twocell_get, adg); + + rsnd_mod_write(mod, AVB_CLK_CONFIG, AVB_DIV_EN_COM); + + break; + default: + dev_err(dev, "Invalid clock-cell %d\n", count); + break; } rsnd_adg_get_clkout_end: @@ -612,6 +898,8 @@ int rsnd_adg_probe(struct rsnd_priv *priv) if (!adg) return -ENOMEM; + spin_lock_init(&adg->avb_lock); + ret = rsnd_mod_init(priv, &adg->mod, &adg_ops, NULL, 0, 0); if (ret) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index ca639404f2cd..1b000d03b76e 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -355,6 +355,15 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_S_REG(SRCOUT_TIMSEL3, 0x54), RSND_GEN_S_REG(SRCOUT_TIMSEL4, 0x58), RSND_GEN_S_REG(CMDOUT_TIMSEL, 0x5c), + RSND_GEN_S_REG(AVB_CLK_DIV0, 0x11c), + RSND_GEN_S_REG(AVB_CLK_DIV1, 0x120), + RSND_GEN_S_REG(AVB_CLK_DIV2, 0x124), + RSND_GEN_S_REG(AVB_CLK_DIV3, 0x128), + RSND_GEN_S_REG(AVB_CLK_DIV4, 0x12c), + RSND_GEN_S_REG(AVB_CLK_DIV5, 0x130), + RSND_GEN_S_REG(AVB_CLK_DIV6, 0x134), + RSND_GEN_S_REG(AVB_CLK_DIV7, 0x138), + RSND_GEN_S_REG(AVB_CLK_CONFIG, 0x13c), }; static const struct rsnd_regmap_field_conf conf_ssi[] = { RSND_GEN_M_REG(SSICR, 0x00, 0x40), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 3c57129af6d1..d31b8a65985f 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -148,6 +148,15 @@ enum rsnd_reg { RSND_REG_AUDIO_CLK_SEL0, RSND_REG_AUDIO_CLK_SEL1, RSND_REG_AUDIO_CLK_SEL2, + RSND_REG_AVB_CLK_DIV0, + RSND_REG_AVB_CLK_DIV1, + RSND_REG_AVB_CLK_DIV2, + RSND_REG_AVB_CLK_DIV3, + RSND_REG_AVB_CLK_DIV4, + RSND_REG_AVB_CLK_DIV5, + RSND_REG_AVB_CLK_DIV6, + RSND_REG_AVB_CLK_DIV7, + RSND_REG_AVB_CLK_CONFIG, /* SSIU */ RSND_REG_SSI_MODE, -- 2.17.0

4 5

[alsa-devel] [PATCH linux-next v2 0/6] clk: renesas: adg: add AVB Clock
by jiada_wang＠mentor.com 04 Dec '18

04 Dec '18

From: Jiada Wang <jiada_wang(a)mentor.com> on R-Car SoCs there are AVB Counter Clocks, each clock has 12bits integral and 8 bits fractional dividers which operates with S0D1ϕ clock. This patch-set adds 'adg' clock to R-Car Soc, and changes adg driver to register avb clocks when clock-cells of rcar_sound node is 2. --- v2: - expends adg register size and register avb clocks instead of add new clk-avb driver - Add adg clock v1: initial version Jiada Wang (2): dt-bindings: clock: add clock id for renesas adg clocks ASoC: rsnd: add avb clocks Takeshi Kihara (4): clk: renesas: r8a7795: Add ADG clock clk: renesas: r8a7796: Add ADG clock clk: renesas: r8a77990: Add ADG clocks clk: renesas: r8a77995: Add ADG clock drivers/clk/renesas/r8a7795-cpg-mssr.c | 1 + drivers/clk/renesas/r8a7796-cpg-mssr.c | 1 + drivers/clk/renesas/r8a77990-cpg-mssr.c | 1 + drivers/clk/renesas/r8a77995-cpg-mssr.c | 1 + include/dt-bindings/clock/renesas-adg.h | 11 + sound/soc/sh/rcar/adg.c | 306 +++++++++++++++++++++++- sound/soc/sh/rcar/gen.c | 9 + sound/soc/sh/rcar/rsnd.h | 9 + 8 files changed, 330 insertions(+), 9 deletions(-) create mode 100644 include/dt-bindings/clock/renesas-adg.h -- 2.17.0

3 6

[alsa-devel] How to inform application about sample rate changed from device.
by Chakravarthi Pradeep 04 Dec '18

04 Dec '18

Hi All, I'm working in FPGA device, which can provide raw video and pcm data. I'm using v4l2 and alsa framework for video and audio respectively. Audio flow path is using ALSA framework and video flow path is using v4l2 framework. Device can generate different audio sample rate such as 44.1KHz, 48KHz and 96KHz. VLC can render video and audio smoothly with 48KHz of sample rate, 2 channel and video with FHD(1920x1080). When device is set with 44.1KHz and 96KHz then vlc audio is breaking.I'm using vlc version as 2.2.8 and there is no way to inform how to set the sample rate for audio device. Below are queries: 1) When device is generating audio sample rate of 44.1KHz then how to inform VLC that use 44.1KHz sample rate(same is for 96KHz) 2) what is command for vlc with video and audio device setting for width=1920 height= 1080 colorspace = RV24, channel =2,sample rate 44.1KHz. Assume video device is (/dev/video0 and audio device is hw:1.0) -- Thanks and Regards Chakravarthi Pradeep.K Ph: 91 9980434900

3 3

[alsa-devel] [PATCH 1/2] ALSA: hda/tegra - Probe up to 8 codecs
by Thierry Reding 03 Dec '18

03 Dec '18

From: Thierry Reding <treding(a)nvidia.com> Recent devices support more than the 4 codecs that the AZX core will probe by default. Probe up to 8 codecs to make sure all of them are enumerated. Suggested-by: Sameer Pujar <spujar(a)nvidia.com> Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 84ea6069d28b..83befd8d43e8 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -531,7 +531,7 @@ static void hda_tegra_probe_work(struct work_struct *work) goto out_free; /* create codec instances */ - err = azx_probe_codecs(chip, 0); + err = azx_probe_codecs(chip, 8); if (err < 0) goto out_free; -- 2.19.1

2 3

[alsa-devel] [PATCH v8 0/3] Huawei laptops
by Ayman Bagabas 03 Dec '18

03 Dec '18

This patch set is based on new audio LED triggers in branch topic/leds-trigger from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Changes from v7: * Switch to wmi_driver * Use devm to allocate and register devices * Skip registering LED subsystem if a ACPI controller method was not found Changes from v6: * Use audio LED triggers patch set * Use KEY_CONFIG (XF86Tools) instead of KEY_PROG1. In Windows, the key is used to launch Huawei PC manager. XF86Tools is used by default on most desktop environments i.e. Gnome. Changes from v5: * Review tags Changes from v4: * Consistency in file names * How module would be enabled (Kconfig) * Match license in SPDX and MODULE_LICENSE Changes from v3: * Code formatting Changes from v2: * Support for Huawei MBX * Style and formatting issues Ayman Bagabas (3): ALSA: hda: fix front speakers on Huawei MBXP. x86: add support for Huawei WMI hotkeys. ALSA: hda: add support for Huawei WMI micmute LED drivers/platform/x86/Kconfig | 16 ++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/huawei-wmi.c | 238 ++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 22 +++ 4 files changed, 277 insertions(+) create mode 100644 drivers/platform/x86/huawei-wmi.c -- 2.19.1

4 12

[alsa-devel] UAF write in usb_audio_probe
by Mathias Payer 03 Dec '18

03 Dec '18

Hi there, We are submitting a patch for an UAF write in usb_audio_probe. A malicious USB device can send a broken device configuration that will trigger a free of the underlying object before it is decremented. I'm sending the patch and KASAN report after discussing with Takashi on the security mailing list so that he can merge. The attacker needs local access to plug in a malicious USB device that replays the trace (e.g., through FaceDancer) to get read/write primitives in the kernel. For, e.g., Android or locked Desktops this becomes security critical. This bug is present in 4.14.81 through to HEAD (i.e., all versions we tested). The place where UAF is triggered is in `usb_audio_probe` in `sound/usb/card.c`. In the error handling code in `usb_audio_probe`, if the number of interfaces read from the device side is zero, `chip->card` object is freed. Unfortunately, the `chip` object is part of the `card` object, thus, when `chip->card` is freed, `chip` is also freed. However, in the error handling code, the `active` field will be written to by an `atomic_dec` operation, resulting in an UAF write. ``` static int usb_audio_probe(struct usb_interface *intf, const struct usb_device_id *usb_id) ..... __error: if (chip) { if (!chip->num_interfaces) snd_card_free(chip->card); // UAF write atomic_dec(&chip->active); } mutex_unlock(&register_mutex); return err; ``` The attacker can race the snd_card_free and the atomic_dec by attaching new USB devices (the attacker can time when what parts of the first malicious device are read then play with attaching the second device; the race is arbitrarily repeatable). The new USB device descriptors will use the recently freed memory which is then modified by the atomic_dec, resulting in an attacker-controlled decrement operation to, e.g., a USB data structure under the attacker's control. We have also attached the KASAN report from when the bug was triggered. The patch for card.c is attached. It moves the atomic_dec to above the free of the chip memory object, ensuring that, if no cards are left, the decrement happens before the free. We've also added a comment to clarify that the two objects are dependent. Thanks, Hui Peng Mathias Payer

2 1

[alsa-devel] cplay: Always write whole buffer requested by user
by GitHub pull_request - opened 03 Dec '18

03 Dec '18

alsa-project/tinycompress pull request #1 was opened from dbaluta: cplay first writes frag * fragment_size and then it only writes one fragment at a time. This means for example than if the user supplied a buffer_size it will only be used for the first write. Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Request URL : https://github.com/alsa-project/tinycompress/pull/1 Patch URL : https://github.com/alsa-project/tinycompress/pull/1.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

[alsa-devel] [PATCH linux-next v2 5/6] dt-bindings: clock: add clock id for renesas adg clocks
by jiada_wang＠mentor.com 03 Dec '18

03 Dec '18

From: Jiada Wang <jiada_wang(a)mentor.com> This patch adds clock ID for renesas adg clocks Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- include/dt-bindings/clock/renesas-adg.h | 11 +++++++++++ 1 file changed, 11 insertions(+) create mode 100644 include/dt-bindings/clock/renesas-adg.h diff --git a/include/dt-bindings/clock/renesas-adg.h b/include/dt-bindings/clock/renesas-adg.h new file mode 100644 index 000000000000..fe30c186cef7 --- /dev/null +++ b/include/dt-bindings/clock/renesas-adg.h @@ -0,0 +1,11 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (C) 2018 Mentor Graphics inc. + */ +#ifndef __DT_BINDINGS_RENESAS_ADG_H__ +#define __DT_BINDINGS_RENESAS_ADG_H__ + +#define ADG_FIX 0 +#define ADG_AVB 1 + +#endif /* __DT_BINDINGS_RENESAS_ADG_H__ */ -- 2.17.0

2 1

[alsa-devel] [PATCH] ASoC: dt-bindings: use a generic node name for rt5631
by Lubomir Rintel 03 Dec '18

03 Dec '18

The example should follow the practice or using a generic node name instead of the precise programming model, as recommended by the DTSpec. Signed-off-by: Lubomir Rintel <lkundrak(a)v3.sk> --- Documentation/devicetree/bindings/sound/rt5631.txt | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5631.txt b/Documentation/devicetree/bindings/sound/rt5631.txt index 92b986ca337b..56bc85232c49 100644 --- a/Documentation/devicetree/bindings/sound/rt5631.txt +++ b/Documentation/devicetree/bindings/sound/rt5631.txt @@ -35,14 +35,14 @@ Pins on the device (for linking into audio routes): Example: -alc5631: alc5631@1a { +alc5631: audio-codec@1a { compatible = "realtek,alc5631"; reg = <0x1a>; }; or -rt5631: rt5631@1a { +rt5631: audio-codec@1a { compatible = "realtek,rt5631"; reg = <0x1a>; }; -- 2.19.1

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[alsa-devel] [PATCH 0/2] SOF dependencies
by Pierre-Louis Bossart 03 Dec '18

03 Dec '18

Two minor patches that are needed to build the rest of the SOF patches which I will post next week after the last round of internal reviews. Submitting them first. Keyon Jie (1): ALSA: HDA: export process_unsol_events() Pierre-Louis Bossart (1): ASoC: Intel: common: add ACPI matching tables for ICL include/sound/hdaudio.h | 1 + include/sound/soc-acpi-intel-match.h | 1 + sound/hda/hdac_bus.c | 7 ++-- sound/soc/intel/common/Makefile | 2 +- .../intel/common/soc-acpi-intel-icl-match.c | 32 +++++++++++++++++++ 5 files changed, 38 insertions(+), 5 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-icl-match.c -- 2.17.1

2 4

[alsa-devel] alsa-utils: axfer: use second argument for transmission direction
by GitHub pull_request - opened 03 Dec '18

03 Dec '18

alsa-project/alsa-utils pull request #4 was opened from takaswie: This patchset is for axfer command in alsa-utils to use second argument for transmission direction. As a result, syntax of its command line is axfer subcommand direction options For example: $ axfer transfer playback [options] $ axfer list capture [options] This fashion requires more typing to users, however in my taste this looks good because it's easier to understand than long/short options. Additionally, this patchset applies refactoring to list subcommand. Takashi Sakamoto (5): axfer: apply refactoring to list subcommand for backward compatibility to aplay(1) axfer: apply refactoring in list subcommand for new command system axfer: use second argument in command line for transmission direction axfer: use transfer subcommand as a default for compatibility mode to aplay(1) axfer: truncate parsed arguments before operating subcommand ``` axfer/main.c | 100 +++++++++++++++++++++++++++++----------- axfer/subcmd-list.c | 85 +++++++++++++++++++++++----------- axfer/subcmd-transfer.c | 7 --- 3 files changed, 133 insertions(+), 59 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/4 Patch URL : https://github.com/alsa-project/alsa-utils/pull/4.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] Regression: very quiet speakers on Thinkpad T570s
by Jeremy Cline 03 Dec '18

03 Dec '18

Hi, Some folks have reported on the Fedora bug tracker[0] that the laptop speaker volume is very low on the Thinkpad T570 when running a kernel that includes commit 61fcf8ece9b6 ("ALSA: hda/realtek - Enable Thinkpad Dock device for ALC298 platform"). alsa-info.sh from v4.15.4 (just before commit 61fcf8ece9b6 arrived in stable) and v4.19.4 with the issue present are attached to the bugzilla. I've also Cc'd Tim, who uploaded them and has the laptop in question. [0] https://bugzilla.redhat.com/show_bug.cgi?id=1554304 Regards, Jeremy

2 5

[alsa-devel] [Xen-devel][PATCH 1/3] xen: Introduce shared buffer helpers for page directory...
by Oleksandr Andrushchenko 30 Nov '18

30 Nov '18

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> based frontends. Currently the frontends which implement similar code for sharing big buffers between frontend and backend are para-virtualized DRM and sound drivers. Both define the same way to share grant references of a data buffer with the corresponding backend with little differences. Move shared code into a helper module, so there is a single implementation of the same functionality for all. Signed-off-by: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> --- drivers/xen/Kconfig | 3 + drivers/xen/Makefile | 1 + drivers/xen/xen-front-pgdir-shbuf.c | 553 ++++++++++++++++++++++++++++ include/xen/xen-front-pgdir-shbuf.h | 89 +++++ 4 files changed, 646 insertions(+) create mode 100644 drivers/xen/xen-front-pgdir-shbuf.c create mode 100644 include/xen/xen-front-pgdir-shbuf.h diff --git a/drivers/xen/Kconfig b/drivers/xen/Kconfig index 815b9e9bb975..838b66a9a0e7 100644 --- a/drivers/xen/Kconfig +++ b/drivers/xen/Kconfig @@ -340,4 +340,7 @@ config XEN_SYMS config XEN_HAVE_VPMU bool +config XEN_FRONT_PGDIR_SHBUF + tristate + endmenu diff --git a/drivers/xen/Makefile b/drivers/xen/Makefile index 3e542f60f29f..c48927a58e10 100644 --- a/drivers/xen/Makefile +++ b/drivers/xen/Makefile @@ -44,3 +44,4 @@ xen-gntdev-y := gntdev.o xen-gntdev-$(CONFIG_XEN_GNTDEV_DMABUF) += gntdev-dmabuf.o xen-gntalloc-y := gntalloc.o xen-privcmd-y := privcmd.o privcmd-buf.o +obj-$(CONFIG_XEN_FRONT_PGDIR_SHBUF) += xen-front-pgdir-shbuf.o diff --git a/drivers/xen/xen-front-pgdir-shbuf.c b/drivers/xen/xen-front-pgdir-shbuf.c new file mode 100644 index 000000000000..48a658dc7ccf --- /dev/null +++ b/drivers/xen/xen-front-pgdir-shbuf.c @@ -0,0 +1,553 @@ +// SPDX-License-Identifier: GPL-2.0 OR MIT + +/* + * Xen frontend/backend page directory based shared buffer + * helper module. + * + * Copyright (C) 2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> + */ + +#include <linux/module.h> +#include <linux/errno.h> +#include <linux/mm.h> + +#include <asm/xen/hypervisor.h> +#include <xen/balloon.h> +#include <xen/xen.h> +#include <xen/xenbus.h> +#include <xen/interface/io/ring.h> + +#include <xen/xen-front-pgdir-shbuf.h> + +#ifndef GRANT_INVALID_REF +/* + * FIXME: usage of grant reference 0 as invalid grant reference: + * grant reference 0 is valid, but never exposed to a PV driver, + * because of the fact it is already in use/reserved by the PV console. + */ +#define GRANT_INVALID_REF 0 +#endif + +/** + * This structure represents the structure of a shared page + * that contains grant references to the pages of the shared + * buffer. This structure is common to many Xen para-virtualized + * protocols at include/xen/interface/io/ + */ +struct xen_page_directory { + grant_ref_t gref_dir_next_page; + grant_ref_t gref[1]; /* Variable length */ +}; + +/** + * Shared buffer ops which are differently implemented + * depending on the allocation mode, e.g. if the buffer + * is allocated by the corresponding backend or frontend. + * Some of the operations. + */ +struct xen_front_pgdir_shbuf_ops { + /* + * Calculate number of grefs required to handle this buffer, + * e.g. if grefs are required for page directory only or the buffer + * pages as well. + */ + void (*calc_num_grefs)(struct xen_front_pgdir_shbuf *buf); + + /* Fill page directory according to para-virtual display protocol. */ + void (*fill_page_dir)(struct xen_front_pgdir_shbuf *buf); + + /* Claim grant references for the pages of the buffer. */ + int (*grant_refs_for_buffer)(struct xen_front_pgdir_shbuf *buf, + grant_ref_t *priv_gref_head, int gref_idx); + + /* Map grant references of the buffer. */ + int (*map)(struct xen_front_pgdir_shbuf *buf); + + /* Unmap grant references of the buffer. */ + int (*unmap)(struct xen_front_pgdir_shbuf *buf); +}; + +/** + * Get granted reference to the very first page of the + * page directory. Usually this is passed to the backend, + * so it can find/fill the grant references to the buffer's + * pages. + * + * \param buf shared buffer which page directory is of interest. + * \return granted reference to the very first page of the + * page directory. + */ +grant_ref_t +xen_front_pgdir_shbuf_get_dir_start(struct xen_front_pgdir_shbuf *buf) +{ + if (!buf->grefs) + return GRANT_INVALID_REF; + + return buf->grefs[0]; +} +EXPORT_SYMBOL_GPL(xen_front_pgdir_shbuf_get_dir_start); + +/** + * Map granted references of the shared buffer. + * + * Depending on the shared buffer mode of allocation + * (be_alloc flag) this can either do nothing (for buffers + * shared by the frontend itself) or map the provided granted + * references onto the backing storage (buf->pages). + * + * \param buf shared buffer which grants to be maped. + * \return zero on success or a negative number on failure. + */ +int xen_front_pgdir_shbuf_map(struct xen_front_pgdir_shbuf *buf) +{ + if (buf->ops && buf->ops->map) + return buf->ops->map(buf); + + /* No need to map own grant references. */ + return 0; +} +EXPORT_SYMBOL_GPL(xen_front_pgdir_shbuf_map); + +/** + * Unmap granted references of the shared buffer. + * + * Depending on the shared buffer mode of allocation + * (be_alloc flag) this can either do nothing (for buffers + * shared by the frontend itself) or unmap the provided granted + * references. + * + * \param buf shared buffer which grants to be unmaped. + * \return zero on success or a negative number on failure. + */ +int xen_front_pgdir_shbuf_unmap(struct xen_front_pgdir_shbuf *buf) +{ + if (buf->ops && buf->ops->unmap) + return buf->ops->unmap(buf); + + /* No need to unmap own grant references. */ + return 0; +} +EXPORT_SYMBOL_GPL(xen_front_pgdir_shbuf_unmap); + +/** + * Free all the resources of the shared buffer. + * + * \param buf shared buffer which resources to be freed. + */ +void xen_front_pgdir_shbuf_free(struct xen_front_pgdir_shbuf *buf) +{ + if (buf->grefs) { + int i; + + for (i = 0; i < buf->num_grefs; i++) + if (buf->grefs[i] != GRANT_INVALID_REF) + gnttab_end_foreign_access(buf->grefs[i], + 0, 0UL); + } + kfree(buf->grefs); + kfree(buf->directory); +} +EXPORT_SYMBOL_GPL(xen_front_pgdir_shbuf_free); + +/* + * Number of grefs a page can hold with respect to the + * struct xen_page_directory header. + */ +#define XEN_NUM_GREFS_PER_PAGE ((PAGE_SIZE - \ + offsetof(struct xen_page_directory, \ + gref)) / sizeof(grant_ref_t)) + +/** + * Get the number of pages the page directory consumes itself. + * + * \param buf shared buffer. + */ +static int get_num_pages_dir(struct xen_front_pgdir_shbuf *buf) +{ + return DIV_ROUND_UP(buf->num_pages, XEN_NUM_GREFS_PER_PAGE); +} + +/** + * Calculate the number of grant references needed to share the buffer + * and its pages when backend allocates the buffer. + * + * \param buf shared buffer. + */ +static void backend_calc_num_grefs(struct xen_front_pgdir_shbuf *buf) +{ + /* Only for pages the page directory consumes itself. */ + buf->num_grefs = get_num_pages_dir(buf); +} + +/** + * Calculate the number of grant references needed to share the buffer + * and its pages when frontend allocates the buffer. + * + * \param buf shared buffer. + */ +static void guest_calc_num_grefs(struct xen_front_pgdir_shbuf *buf) +{ + /* + * Number of pages the page directory consumes itself + * plus grefs for the buffer pages. + */ + buf->num_grefs = get_num_pages_dir(buf) + buf->num_pages; +} + +#define xen_page_to_vaddr(page) \ + ((uintptr_t)pfn_to_kaddr(page_to_xen_pfn(page))) + +/** + * Unmap the buffer previously mapped with grant references + * provided by the backend. + * + * \param buf shared buffer. + * \return zero on success or a negative number on failure. + */ +static int backend_unmap(struct xen_front_pgdir_shbuf *buf) +{ + struct gnttab_unmap_grant_ref *unmap_ops; + int i, ret; + + if (!buf->pages || !buf->backend_map_handles || !buf->grefs) + return 0; + + unmap_ops = kcalloc(buf->num_pages, sizeof(*unmap_ops), + GFP_KERNEL); + if (!unmap_ops) + return -ENOMEM; + + for (i = 0; i < buf->num_pages; i++) { + phys_addr_t addr; + + addr = xen_page_to_vaddr(buf->pages[i]); + gnttab_set_unmap_op(&unmap_ops[i], addr, GNTMAP_host_map, + buf->backend_map_handles[i]); + } + + ret = gnttab_unmap_refs(unmap_ops, NULL, buf->pages, + buf->num_pages); + + for (i = 0; i < buf->num_pages; i++) { + if (unlikely(unmap_ops[i].status != GNTST_okay)) + dev_err(&buf->xb_dev->dev, + "Failed to unmap page %d: %d\n", + i, unmap_ops[i].status); + } + + if (ret) + dev_err(&buf->xb_dev->dev, + "Failed to unmap grant references, ret %d", ret); + + kfree(unmap_ops); + kfree(buf->backend_map_handles); + buf->backend_map_handles = NULL; + return ret; +} + +/** + * Map the buffer with grant references provided by the backend. + * + * \param buf shared buffer. + * \return zero on success or a negative number on failure. + */ +static int backend_map(struct xen_front_pgdir_shbuf *buf) +{ + struct gnttab_map_grant_ref *map_ops = NULL; + unsigned char *ptr; + int ret, cur_gref, cur_dir_page, cur_page, grefs_left; + + map_ops = kcalloc(buf->num_pages, sizeof(*map_ops), GFP_KERNEL); + if (!map_ops) + return -ENOMEM; + + buf->backend_map_handles = kcalloc(buf->num_pages, + sizeof(*buf->backend_map_handles), + GFP_KERNEL); + if (!buf->backend_map_handles) { + kfree(map_ops); + return -ENOMEM; + } + + /* + * Read page directory to get grefs from the backend: for external + * buffer we only allocate buf->grefs for the page directory, + * so buf->num_grefs has number of pages in the page directory itself. + */ + ptr = buf->directory; + grefs_left = buf->num_pages; + cur_page = 0; + for (cur_dir_page = 0; cur_dir_page < buf->num_grefs; cur_dir_page++) { + struct xen_page_directory *page_dir = + (struct xen_page_directory *)ptr; + int to_copy = XEN_NUM_GREFS_PER_PAGE; + + if (to_copy > grefs_left) + to_copy = grefs_left; + + for (cur_gref = 0; cur_gref < to_copy; cur_gref++) { + phys_addr_t addr; + + addr = xen_page_to_vaddr(buf->pages[cur_page]); + gnttab_set_map_op(&map_ops[cur_page], addr, + GNTMAP_host_map, + page_dir->gref[cur_gref], + buf->xb_dev->otherend_id); + cur_page++; + } + + grefs_left -= to_copy; + ptr += PAGE_SIZE; + } + ret = gnttab_map_refs(map_ops, NULL, buf->pages, buf->num_pages); + + /* Save handles even if error, so we can unmap. */ + for (cur_page = 0; cur_page < buf->num_pages; cur_page++) { + buf->backend_map_handles[cur_page] = map_ops[cur_page].handle; + if (unlikely(map_ops[cur_page].status != GNTST_okay)) + dev_err(&buf->xb_dev->dev, + "Failed to map page %d: %d\n", + cur_page, map_ops[cur_page].status); + } + + if (ret) { + dev_err(&buf->xb_dev->dev, + "Failed to map grant references, ret %d", ret); + backend_unmap(buf); + } + + kfree(map_ops); + return ret; +} + +/** + * Fill page directory with grant references to the pages of the + * page directory itself. + * + * The grant references to the buffer pages are provided by the + * backend in this case. + * + * \param buf shared buffer. + */ +static void backend_fill_page_dir(struct xen_front_pgdir_shbuf *buf) +{ + struct xen_page_directory *page_dir; + unsigned char *ptr; + int i, num_pages_dir; + + ptr = buf->directory; + num_pages_dir = get_num_pages_dir(buf); + + /* Fill only grefs for the page directory itself. */ + for (i = 0; i < num_pages_dir - 1; i++) { + page_dir = (struct xen_page_directory *)ptr; + + page_dir->gref_dir_next_page = buf->grefs[i + 1]; + ptr += PAGE_SIZE; + } + /* Last page must say there is no more pages. */ + page_dir = (struct xen_page_directory *)ptr; + page_dir->gref_dir_next_page = GRANT_INVALID_REF; +} + +/** + * Fill page directory with grant references to the pages of the + * page directory and the buffer we share with the backend. + * + * \param buf shared buffer. + */ +static void guest_fill_page_dir(struct xen_front_pgdir_shbuf *buf) +{ + unsigned char *ptr; + int cur_gref, grefs_left, to_copy, i, num_pages_dir; + + ptr = buf->directory; + num_pages_dir = get_num_pages_dir(buf); + + /* + * While copying, skip grefs at start, they are for pages + * granted for the page directory itself. + */ + cur_gref = num_pages_dir; + grefs_left = buf->num_pages; + for (i = 0; i < num_pages_dir; i++) { + struct xen_page_directory *page_dir = + (struct xen_page_directory *)ptr; + + if (grefs_left <= XEN_NUM_GREFS_PER_PAGE) { + to_copy = grefs_left; + page_dir->gref_dir_next_page = GRANT_INVALID_REF; + } else { + to_copy = XEN_NUM_GREFS_PER_PAGE; + page_dir->gref_dir_next_page = buf->grefs[i + 1]; + } + memcpy(&page_dir->gref, &buf->grefs[cur_gref], + to_copy * sizeof(grant_ref_t)); + ptr += PAGE_SIZE; + grefs_left -= to_copy; + cur_gref += to_copy; + } +} + +/** + * Grant references to the frontend's buffer pages. + * + * These will be shared with the backend, so it can + * access the buffer's data. + * + * \param buf shared buffer. + * \return zero on success or a negative number on failure. + */ +static int guest_grant_refs_for_buffer(struct xen_front_pgdir_shbuf *buf, + grant_ref_t *priv_gref_head, + int gref_idx) +{ + int i, cur_ref, otherend_id; + + otherend_id = buf->xb_dev->otherend_id; + for (i = 0; i < buf->num_pages; i++) { + cur_ref = gnttab_claim_grant_reference(priv_gref_head); + if (cur_ref < 0) + return cur_ref; + + gnttab_grant_foreign_access_ref(cur_ref, otherend_id, + xen_page_to_gfn(buf->pages[i]), + 0); + buf->grefs[gref_idx++] = cur_ref; + } + return 0; +} + +/** + * Grant all the references needed to share the buffer. + * + * Grant references to the page directory pages and, if + * needed, also to the pages of the shared buffer data. + * + * \param buf shared buffer. + * \return zero on success or a negative number on failure. + */ +static int grant_references(struct xen_front_pgdir_shbuf *buf) +{ + grant_ref_t priv_gref_head; + int ret, i, j, cur_ref; + int otherend_id, num_pages_dir; + + ret = gnttab_alloc_grant_references(buf->num_grefs, &priv_gref_head); + if (ret < 0) { + dev_err(&buf->xb_dev->dev, + "Cannot allocate grant references\n"); + return ret; + } + + otherend_id = buf->xb_dev->otherend_id; + j = 0; + num_pages_dir = get_num_pages_dir(buf); + for (i = 0; i < num_pages_dir; i++) { + unsigned long frame; + + cur_ref = gnttab_claim_grant_reference(&priv_gref_head); + if (cur_ref < 0) + return cur_ref; + + frame = xen_page_to_gfn(virt_to_page(buf->directory + + PAGE_SIZE * i)); + gnttab_grant_foreign_access_ref(cur_ref, otherend_id, frame, 0); + buf->grefs[j++] = cur_ref; + } + + if (buf->ops->grant_refs_for_buffer) { + ret = buf->ops->grant_refs_for_buffer(buf, &priv_gref_head, j); + if (ret) + return ret; + } + + gnttab_free_grant_references(priv_gref_head); + return 0; +} + +/** + * Allocate all required structures to mange shared buffer. + * + * \param buf shared buffer. + * \return zero on success or a negative number on failure. + */ +static int alloc_storage(struct xen_front_pgdir_shbuf *buf) +{ + buf->grefs = kcalloc(buf->num_grefs, sizeof(*buf->grefs), GFP_KERNEL); + if (!buf->grefs) + return -ENOMEM; + + buf->directory = kcalloc(get_num_pages_dir(buf), PAGE_SIZE, GFP_KERNEL); + if (!buf->directory) + return -ENOMEM; + + return 0; +} + +/* + * For backend allocated buffers we don't need grant_refs_for_buffer + * as those grant references are allocated at backend side. + */ +static const struct xen_front_pgdir_shbuf_ops backend_ops = { + .calc_num_grefs = backend_calc_num_grefs, + .fill_page_dir = backend_fill_page_dir, + .map = backend_map, + .unmap = backend_unmap +}; + +/* + * For locally granted references we do not need to map/unmap + * the references. + */ +static const struct xen_front_pgdir_shbuf_ops local_ops = { + .calc_num_grefs = guest_calc_num_grefs, + .fill_page_dir = guest_fill_page_dir, + .grant_refs_for_buffer = guest_grant_refs_for_buffer, +}; + +/** + * Allocate a new instance of a shared buffer. + * + * \param cfg configuration to be used while allocating a new shared buffer. + * \return zero on success or a negative number on failure. + */ +int xen_front_pgdir_shbuf_alloc(struct xen_front_pgdir_shbuf_cfg *cfg) +{ + struct xen_front_pgdir_shbuf *buf = cfg->pgdir; + int ret; + + if (cfg->be_alloc) + buf->ops = &backend_ops; + else + buf->ops = &local_ops; + buf->xb_dev = cfg->xb_dev; + buf->num_pages = cfg->num_pages; + buf->pages = cfg->pages; + + buf->ops->calc_num_grefs(buf); + + ret = alloc_storage(buf); + if (ret) + goto fail; + + ret = grant_references(buf); + if (ret) + goto fail; + + buf->ops->fill_page_dir(buf); + + return 0; + +fail: + xen_front_pgdir_shbuf_free(buf); + return ret; +} +EXPORT_SYMBOL_GPL(xen_front_pgdir_shbuf_alloc); + +MODULE_DESCRIPTION("Xen frontend/backend page directory based " + "shared buffer handling"); +MODULE_AUTHOR("Oleksandr Andrushchenko"); +MODULE_LICENSE("GPL"); diff --git a/include/xen/xen-front-pgdir-shbuf.h b/include/xen/xen-front-pgdir-shbuf.h new file mode 100644 index 000000000000..150ef7ec51ec --- /dev/null +++ b/include/xen/xen-front-pgdir-shbuf.h @@ -0,0 +1,89 @@ +/* SPDX-License-Identifier: GPL-2.0 OR MIT */ + +/* + * Xen frontend/backend page directory based shared buffer + * helper module. + * + * Copyright (C) 2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> + */ + +#ifndef __XEN_FRONT_PGDIR_SHBUF_H_ +#define __XEN_FRONT_PGDIR_SHBUF_H_ + +#include <linux/kernel.h> + +#include <xen/grant_table.h> + +struct xen_front_pgdir_shbuf_ops; + +struct xen_front_pgdir_shbuf { + /* + * Number of references granted for the backend use: + * + * - for frontend allocated/imported buffers this holds the number + * of grant references for the page directory and the pages + * of the buffer + * + * - for the buffer provided by the backend this only holds the number + * of grant references for the page directory itself as grant + * references for the buffer will be provided by the backend. + */ + int num_grefs; + grant_ref_t *grefs; + /* Page directory backing storage. */ + u8 *directory; + + /* + * Number of pages for the shared buffer itself (excluding the page + * directory). + */ + int num_pages; + /* + * Backing storage of the shared buffer: these are the pages being + * shared. + */ + struct page **pages; + + struct xenbus_device *xb_dev; + + /* These are the ops used internally depending on be_alloc mode. */ + const struct xen_front_pgdir_shbuf_ops *ops; + + /* Xen map handles for the buffer allocated by the backend. */ + grant_handle_t *backend_map_handles; +}; + +struct xen_front_pgdir_shbuf_cfg { + struct xenbus_device *xb_dev; + + /* Number of pages of the buffer backing storage. */ + int num_pages; + /* Pages of the buffer to be shared. */ + struct page **pages; + + /* + * This is allocated outside because there are use-cases when + * the buffer structure is allocated as a part of a bigger one. + */ + struct xen_front_pgdir_shbuf *pgdir; + /* + * Mode of grant reference sharing: if set then backend will share + * grant references to the buffer with the frontend. + */ + int be_alloc; +}; + +int xen_front_pgdir_shbuf_alloc(struct xen_front_pgdir_shbuf_cfg *cfg); + +grant_ref_t +xen_front_pgdir_shbuf_get_dir_start(struct xen_front_pgdir_shbuf *buf); + +int xen_front_pgdir_shbuf_map(struct xen_front_pgdir_shbuf *buf); + +int xen_front_pgdir_shbuf_unmap(struct xen_front_pgdir_shbuf *buf); + +void xen_front_pgdir_shbuf_free(struct xen_front_pgdir_shbuf *buf); + +#endif /* __XEN_FRONT_PGDIR_SHBUF_H_ */ -- 2.19.1

4 10

[alsa-devel] [V3] ASoC: max98373: Added max98373_reset for stable amp reset
by Ryan Lee 30 Nov '18

30 Nov '18

This patch added max98373_reset function to avoid amp software reset failure and code duplication. Reset verification step has been added for stable amp reset and it repeats verification maximum 3 times when it is failed. Chip revision ID is available when the amp is in the idle state which means software reset is completed well. Additional 10ms delay was added for every retrial and maximum 30ms delay can be applied. Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes since v2: - Added commit message. Changes since v1: - Removed unusual repeat for amp software reset and verification. - Amp software reset will be performed once and it repeats verification maximum 3 times if it is failed. - Wait 10ms before every verification trial. Maximum 30ms delay will be applied to wait AMP idle state. Changes: - Created max98373_reset function to minimize code duplication. - Changed regmap_write to regmap_update_bits. Other bits except LSB need to be masked. - Added reset verification step to make sure software reset is completed well. Software reset is done in 10ms in normal case. - Revision ID is available when the amp is in the idle state which means software reset is completed. - Software reset will be performed maximum 3 times to avoid amp reset failure. Generally it is done in the first trial. - sleep time after software reset is increased + 30ms for every retrial. Maximum possible msleep time is 100 ms (initial 10 ms + 30 ms * 3 times). sound/soc/codecs/max98373.c | 35 +++++++++++++++++++++++++++++------ 1 file changed, 29 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index a09d013..9c8616a 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -724,14 +724,39 @@ static struct snd_soc_dai_driver max98373_dai[] = { } }; +static void max98373_reset(struct max98373_priv *max98373, struct device *dev) +{ + int ret, reg, count; + + /* Software Reset */ + ret = regmap_update_bits(max98373->regmap, + MAX98373_R2000_SW_RESET, + MAX98373_SOFT_RESET, + MAX98373_SOFT_RESET); + if (ret) + dev_err(dev, "Reset command failed. (ret:%d)\n", ret); + + count = 0; + while (count < 3) { + usleep_range(10000, 11000); + /* Software Reset Verification */ + ret = regmap_read(max98373->regmap, + MAX98373_R21FF_REV_ID, &reg); + if (!ret) { + dev_info(dev, "Reset completed (retry:%d)\n", count); + return; + } + count++; + } + dev_err(dev, "Reset failed. (ret:%d)\n", ret); +} + static int max98373_probe(struct snd_soc_component *component) { struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); /* Software Reset */ - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, component->dev); /* IV default slot configuration */ regmap_write(max98373->regmap, @@ -818,9 +843,7 @@ static int max98373_resume(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, dev); regcache_cache_only(max98373->regmap, false); regcache_sync(max98373->regmap); return 0; -- 2.7.4

1 0

[alsa-devel] [PATCH V2] ASoC: max98373: Added max98373_reset for stable amp reset
by Ryan Lee 30 Nov '18

30 Nov '18

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes since v1 : Removed unusual repeat for amp software reset and verification. Amp software reset will be performed once and it repeats verification maximum 3 times if it is failed. Wait 10ms before every verification trial. Maximum 30ms delay will be applied to wait AMP idle state. Changes : Created max98373_reset function to minimize code duplication. Changed regmap_write to regmap_update_bits. Other bits except LSB need to be masked. Added reset verification step to make sure software reset is completed well. Software reset is done in 10ms in normal case. Revision ID is available when the amp is in the idle state which means software reset is completed. Software reset will be performed maximum 3 times to avoid amp reset failure. Generally it is done in the first trial. sleep time after software reset is increased + 30ms for every retrial. Maximum possible msleep time is 100 ms (initial 10 ms + 30 ms * 3 times). sound/soc/codecs/max98373.c | 35 +++++++++++++++++++++++++++++------ 1 file changed, 29 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index a09d013..9c8616a 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -724,14 +724,39 @@ static struct snd_soc_dai_driver max98373_dai[] = { } }; +static void max98373_reset(struct max98373_priv *max98373, struct device *dev) +{ + int ret, reg, count; + + /* Software Reset */ + ret = regmap_update_bits(max98373->regmap, + MAX98373_R2000_SW_RESET, + MAX98373_SOFT_RESET, + MAX98373_SOFT_RESET); + if (ret) + dev_err(dev, "Reset command failed. (ret:%d)\n", ret); + + count = 0; + while (count < 3) { + usleep_range(10000, 11000); + /* Software Reset Verification */ + ret = regmap_read(max98373->regmap, + MAX98373_R21FF_REV_ID, &reg); + if (!ret) { + dev_info(dev, "Reset completed (retry:%d)\n", count); + return; + } + count++; + } + dev_err(dev, "Reset failed. (ret:%d)\n", ret); +} + static int max98373_probe(struct snd_soc_component *component) { struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); /* Software Reset */ - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, component->dev); /* IV default slot configuration */ regmap_write(max98373->regmap, @@ -818,9 +843,7 @@ static int max98373_resume(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, dev); regcache_cache_only(max98373->regmap, false); regcache_sync(max98373->regmap); return 0; -- 2.7.4

3 6

[alsa-devel] [PATCH 0/4] ASoC: merge audio-graph-scu into audio-graph - step3
by Kuninori Morimoto 30 Nov '18

30 Nov '18

Hi Mark audio-graph card can handle normal sound card, and audio-graph-scu card can handle DPCM sound card. But, we can't use these feature in same time now. So, this is the time to merge audio-graph and audio-graph-scu. The patch-set are a little bit huge (and few patches are depend on posted OF patch. I'm now waiting...), thus, I separated these into few steps. This is step3 patches. After step3, the remaining patch is "merge audio-graph and audio-graph-scu" and "remove audio-graph-scu" patches, and, +1 more patch which is based on OF graph patches. I'm posting it to OF ML, but no response so far. Thus, step4 patch will be *after* OF graph patch. Kuninori Morimoto (4): ASoC: audio-graph-scu-card: care link / dai count ASoC: audio-graph-scu-card: use cpu/codec pointer on graph_dai_props ASoC: audio-graph-scu-card: care multi DPCM codec_conf ASoC: audio-graph-card: use cpu/codec pointer on graph_dai_props sound/soc/generic/audio-graph-card.c | 57 ++++++----- sound/soc/generic/audio-graph-scu-card.c | 163 ++++++++++++++++++++++--------- 2 files changed, 153 insertions(+), 67 deletions(-) -- 2.7.4

2 7

[alsa-devel] [PATCH] ALSA: pcm: Fix starvation on down_write_nonblock()
by Chanho Min 30 Nov '18

30 Nov '18

Commit 67ec1072b053 ("ALSA: pcm: Fix rwsem deadlock for non-atomic PCM stream") fixes deadlock for non-atomic PCM stream. But, This patch causes antother stuck. If writer is RT thread and reader is a normal thread, the reader thread will be difficult to get scheduled. It may not give chance to release readlocks and writer gets stuck for a long time if they are pinned to single cpu. The deadlock described in the previous commit is because the linux rwsem queues like a FIFO. So, we might need non-FIFO writelock, not non-block one. My suggestion is that the writer gives reader a chance to be scheduled by using the minimum msleep() instaed of spinning without blocking by writer. Also, The *_nonblock may be changed to *_nonfifo appropriately to this concept. In terms of performance, when trylock is failed, this minimum periodic msleep will have the same performance as the tick-based schedule()/wake_up_q(). Suggested-by: Wonmin Jung <wonmin.jung(a)lge.com> Signed-off-by: Chanho Min <chanho.min(a)lge.com> --- sound/core/pcm_native.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 66c90f4..bdca0e1 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -36,6 +36,7 @@ #include <sound/timer.h> #include <sound/minors.h> #include <linux/uio.h> +#include <linux/delay.h> #include "pcm_local.h" @@ -91,12 +92,12 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem); * and this may lead to a deadlock when the code path takes read sem * twice (e.g. one in snd_pcm_action_nonatomic() and another in * snd_pcm_stream_lock()). As a (suboptimal) workaround, let writer to - * spin until it gets the lock. + * sleep until all the readers are completed without blocking by writer. */ -static inline void down_write_nonblock(struct rw_semaphore *lock) +static inline void down_write_nonfifo(struct rw_semaphore *lock) { while (!down_write_trylock(lock)) - cond_resched(); + msleep(1); } #define PCM_LOCK_DEFAULT 0 @@ -1967,7 +1968,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) res = -ENOMEM; goto _nolock; } - down_write_nonblock(&snd_pcm_link_rwsem); + down_write_nonfifo(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN || substream->runtime->status->state != substream1->runtime->status->state || @@ -2014,7 +2015,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) struct snd_pcm_substream *s; int res = 0; - down_write_nonblock(&snd_pcm_link_rwsem); + down_write_nonfifo(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (!snd_pcm_stream_linked(substream)) { res = -EALREADY; -- 2.1.4

2 7

[alsa-devel] [PATCH] ALSA: pcm: Fix interval evaluation with openmin/max
by Takashi Iwai 29 Nov '18

29 Nov '18

As addressed in alsa-lib (commit b420056604f0), we need to fix the case where the evaluation of PCM interval "(x x+1]" leading to -EINVAL. After applying rules, such an interval may be translated as "(x x+1)". Fixes: ff2d6acdf6f1 ("ALSA: pcm: Fix snd_interval_refine first/last with open min/max") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/pcm_params.h | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 2dd37cada7c0..888a833d3b00 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -254,11 +254,13 @@ static inline int snd_interval_empty(const struct snd_interval *i) static inline int snd_interval_single(const struct snd_interval *i) { return (i->min == i->max || - (i->min + 1 == i->max && i->openmax)); + (i->min + 1 == i->max && (i->openmin || i->openmax))); } static inline int snd_interval_value(const struct snd_interval *i) { + if (i->openmin && !i->openmax) + return i->max; return i->min; } -- 2.19.1

1 0

[alsa-devel] [PATCH v2 0/6] Introduce audio-mute LED trigger (and conversions to it)
by Takashi Iwai 29 Nov '18

29 Nov '18

Hi, this is revised patch series for adding a new LED trigger audio-mute and audio-micmute, and converting the HD-audio driver and the platform drivers to use the LED trigger instead of the ugly direct dynamic symbol binding. v2 series addressed a couple of issues Andy suggested, and changed the led class dev names to "platform::xxx" as discussed in the previous thread. The latest version of patches are found in topic/leds-trigger branch in my sound git tree. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git As these are cross-tree patches, the branch above is based cleanly on v4.20-rc3. From now on, it'll stay as an immutable branch, so feel free to merge into other trees. thanks, Takashi === Takashi Iwai (6): leds: trigger: Introduce audio mute LED trigger platform/x86: dell-laptop: Add micmute LED trigger support platform/x86: thinkpad_acpi: Add audio mute LED classdev support ALSA: hda - Support led audio trigger platform/x86: dell-laptop: Drop superfluous exported function platform/x86: thinkpad_acpi: Drop superfluous exported function drivers/leds/trigger/Kconfig | 7 +++ drivers/leds/trigger/Makefile | 1 + drivers/leds/trigger/ledtrig-audio.c | 44 ++++++++++++++++++ drivers/platform/x86/Kconfig | 4 ++ drivers/platform/x86/dell-laptop.c | 27 ++++++++--- drivers/platform/x86/thinkpad_acpi.c | 67 +++++++++++++++++++++------- include/linux/dell-led.h | 7 --- include/linux/leds.h | 20 +++++++++ include/linux/thinkpad_acpi.h | 16 ------- sound/pci/hda/dell_wmi_helper.c | 48 -------------------- sound/pci/hda/hda_generic.c | 31 +++++++++++++ sound/pci/hda/hda_generic.h | 2 + sound/pci/hda/patch_realtek.c | 17 +++---- sound/pci/hda/thinkpad_helper.c | 43 +++--------------- 14 files changed, 194 insertions(+), 140 deletions(-) create mode 100644 drivers/leds/trigger/ledtrig-audio.c delete mode 100644 include/linux/dell-led.h delete mode 100644 include/linux/thinkpad_acpi.h delete mode 100644 sound/pci/hda/dell_wmi_helper.c -- 2.19.1

1 7

[alsa-devel] [PATCH] ASoc: au8540: use 64-bit arithmetic instead of 32-bit
by Yang Xiao 29 Nov '18

29 Nov '18

From: Young_X <YangX92(a)hotmail.com> 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 Signed-off-by: Young_X <YangX92(a)hotmail.com> --- sound/soc/codecs/nau8540.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index e3c8cd1..4dd1a60 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -585,7 +585,7 @@ static int nau8540_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

2 1

[alsa-devel] [PATCH] ASoC: adau1761: Ensure DSP_RUN and DSP_ENABLE are disabled
by niklas.morberg＠axis.com 29 Nov '18

29 Nov '18

From: Niklas Carlsson <niklasc(a)axis.com> DSP_RUN and DSP_ENABLE needs to be disabled during FW load. This is not a problem after power-cycling but for soft reboots. Signed-off-by: Niklas Carlsson <niklasc(a)axis.com> --- sound/soc/codecs/adau1761.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..3075c13bcbbc 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -460,8 +460,35 @@ static int adau1761_set_bias_level(struct snd_soc_component *component, regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN); - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + /* + * When going OFF -> STANDBY, the device can be in two + * states: + * 1) Power-cycled and reset + * 2) Soft reboot + * In the case of soft reboot, we need to sync the HW + * registers even if our regmap reports default values. + * Marking the cache as dirty ensures that both cases + * are handled. + */ + regcache_mark_dirty(adau->regmap); regcache_sync(adau->regmap); + /* + * In order for FW to load correctly, the device needs + * DSP_RUN and DSP_ENABLE to be 0. This might not be the + * case for soft reboots. Ensure that DSP_RUN and + * DSP_ENABLE is 0 by bypassing the cache and write + * directly to HW when going OFF -> STANDBY. + */ + if (adau17x1_has_dsp(adau)) { + regcache_cache_bypass(adau->regmap, true); + regmap_write(adau->regmap, ADAU17X1_DSP_RUN, 0); + regmap_write(adau->regmap, + ADAU17X1_DSP_ENABLE, 0); + regcache_cache_bypass(adau->regmap, false); + } + } break; case SND_SOC_BIAS_OFF: regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, -- 2.11.0

2 2

[alsa-devel] [PATCH] ALSA: hda: Add support for AMD Stoney Ridge
by Kai-Heng Feng 29 Nov '18

29 Nov '18

It's similar to other AMD audio devices, it also supports D3, which can save some power drain. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d8eb2b5f51ae..3f5659064585 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2496,6 +2496,10 @@ static const struct pci_device_id azx_ids[] = { /* AMD Hudson */ { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + /* AMD Stoney */ + { PCI_DEVICE(0x1022, 0x157a), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | + AZX_DCAPS_PM_RUNTIME }, /* AMD Raven */ { PCI_DEVICE(0x1022, 0x15e3), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | -- 2.17.1

2 1

[alsa-devel] User-friendly dynamic HDA pin reconfiguration
by Daniel Drake 29 Nov '18

29 Nov '18

Hi, We are working with a new consumer All-in-One platform with HDA audio (Realtek codec). Upon plugging in an audio device into one of the audio jacks, under Windows, a prompt appears which says: Which device did you plug in? And you can select from several options: Mic In / Headphone / Speaker Out Screenshot: https://imgur.com/a/mrUolHN The system vendor is asking if we can implement the same under Linux. At first glance, this sounds quite similar to what happens on other models when you plug in a headset under GNOME: a dialog pops up asking if you connected a mic, headphones, or a headset. Under the surface, such headset jacks seems to correspond to 2 separate HDA pins (one input and one output) and this dialog seems to manage which streams are enabled or muted. However, this case is different. We have experimented, and the jack corresponds to just a single HDA pin. We can manually quirk or retask this pin to be an input or an output and it works in the direction that we choose. But that's a far cry from having a nice GUI that automatically comes up asking the user how the pin should be configured. Is there any current solution or work in progress with the goal of offering this functionality? If not, brainstorming, possible solutions could include: 1. Taking the hdajackretask idea but making it user-accessible and user-friendly (currently it goes through debugfs and must be done by root). This would require the creation of a new kernel interface to expose pin reconfiguration in a nicer way, which would have to be plumbed through pulseaudio and GNOME. 2. Could ALSA ignore the pin config here and provide both input and output streams for this pin? Then pulseaudio + GUI could pick which one to use - similar to the existing Headset dialog. This would still require some work on pulseaudio and the corresponding UI, but it might be a less invasive project. Thanks, Daniel

2 2

[alsa-devel] [PATCH] ucm: Set default include path
by Takashi Iwai 29 Nov '18

29 Nov '18

Many UCM profiles include the UCM profile components under ucm/* subdirectories and thusly put <searchdir:ucm> at each place. This is rather cumbersome. This patch makes the UCM parser to set the default include path, so that each profile no longer needs to set searchdir. All the <searchdir:ucm> lines currently found in the profiles are removed gracefully, too. For the needed implementation, a new helper, _snd_config_load_with_include() is introduced. It's not exported, only for the use inside alsa-lib. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/local.h | 3 ++ src/conf.c | 32 +++++++++++++++++-- .../HiFi.conf | 1 - .../bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf | 1 - .../bytcr-rt5640-mono-spk-in1-mic/HiFi.conf | 1 - .../bytcr-rt5640-mono-spk-in3-mic/HiFi.conf | 1 - .../HiFi.conf | 1 - .../bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf | 1 - .../bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf | 1 - src/conf/ucm/bytcr-rt5640/HiFi.conf | 1 - .../bytcr-rt5651-mono-spk-in1-mic/HiFi.conf | 1 - .../HiFi.conf | 1 - .../bytcr-rt5651-mono-spk-in2-mic/HiFi.conf | 1 - .../bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf | 1 - .../HiFi.conf | 1 - .../bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf | 1 - src/conf/ucm/bytcr-rt5651/HiFi.conf | 1 - src/conf/ucm/chtnau8824/HiFi.conf | 1 - src/conf/ucm/chtrt5645/HiFi.conf | 1 - .../HiFi.conf | 1 - src/ucm/parser.c | 3 -- src/ucm/ucm_local.h | 3 ++ src/ucm/utils.c | 7 +++- 23 files changed, 41 insertions(+), 25 deletions(-) diff --git a/include/local.h b/include/local.h index b5e7b6f00ff5..03d8f0ccdc58 100644 --- a/include/local.h +++ b/include/local.h @@ -354,6 +354,9 @@ int snd_config_search_alias_hooks(snd_config_t *config, int _snd_conf_generic_id(const char *id); +int _snd_config_load_with_include(snd_config_t *config, snd_input_t *in, + const char *default_include_path); + /* convenience macros */ #define ARRAY_SIZE(x) (sizeof(x) / sizeof(x[0])) diff --git a/src/conf.c b/src/conf.c index 254c485868b5..1f9bad7b3a97 100644 --- a/src/conf.c +++ b/src/conf.c @@ -1832,7 +1832,8 @@ int snd_config_top(snd_config_t **config) return _snd_config_make(config, 0, SND_CONFIG_TYPE_COMPOUND); } -static int snd_config_load1(snd_config_t *config, snd_input_t *in, int override) +static int snd_config_load1(snd_config_t *config, snd_input_t *in, int override, + char *default_include_path) { int err; input_t input; @@ -1847,6 +1848,11 @@ static int snd_config_load1(snd_config_t *config, snd_input_t *in, int override) fd->column = 0; fd->next = NULL; INIT_LIST_HEAD(&fd->include_paths); + if (default_include_path) { + err = add_include_path(fd, default_include_path); + if (err < 0) + goto _end; + } input.current = fd; input.unget = 0; err = parse_defs(config, &input, 0, override); @@ -1915,9 +1921,29 @@ static int snd_config_load1(snd_config_t *config, snd_input_t *in, int override) */ int snd_config_load(snd_config_t *config, snd_input_t *in) { - return snd_config_load1(config, in, 0); + return snd_config_load1(config, in, 0, NULL); } +#ifndef DOC_HIDDEN +/* load config with the default include path; used internally for UCM parser */ +int _snd_config_load_with_include(snd_config_t *config, snd_input_t *in, + const char *default_include_path) +{ + int err; + char *s = NULL; + + if (default_include_path) { + s = strdup(default_include_path); + if (!s) + return -ENOMEM; + } + err = snd_config_load1(config, in, 0, s); + if (err < 0) + free(s); + return err; +} +#endif + /** * \brief Loads a configuration tree and overrides existing configuration nodes. * \param config Handle to a top level configuration node. @@ -1930,7 +1956,7 @@ int snd_config_load(snd_config_t *config, snd_input_t *in) */ int snd_config_load_override(snd_config_t *config, snd_input_t *in) { - return snd_config_load1(config, in, 1); + return snd_config_load1(config, in, 1, NULL); } /** diff --git a/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf b/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf index facc73ac3b2b..b931b7914ce7 100644 --- a/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf +++ b/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf @@ -1,4 +1,3 @@ -<searchdir:ucm> SectionVerb { diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf index c78cdfb80a44..0e21d6e1abf9 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf index 34a5d53d363c..d8ca499a5799 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf index 0aa0c0266412..e4fb28df546f 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf index e358d4f23f12..bb31bf531633 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf index 097560d02d12..9474a64e84ff 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf index 8a018f63c9c6..478130e67ae4 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640/HiFi.conf b/src/conf/ucm/bytcr-rt5640/HiFi.conf index 20ebe2dd36e2..54dd2517dbd7 100644 --- a/src/conf/ucm/bytcr-rt5640/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf index ca8a7d9078fb..b93e0be33cbd 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf index 60ef6b332ea2..ef048626732e 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf index e75210d4aa7e..c324c14ad542 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf index 2b7675bce831..aa1a1cde7856 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf index 7a8c94bc5d6b..343aed211bb4 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf index 45c7837c6fb9..6f6d74cf2e68 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651/HiFi.conf b/src/conf/ucm/bytcr-rt5651/HiFi.conf index d3928e9b9976..fddd8a3ca19a 100644 --- a/src/conf/ucm/bytcr-rt5651/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/chtnau8824/HiFi.conf b/src/conf/ucm/chtnau8824/HiFi.conf index 97a7ac7ba366..845cbcaffe5f 100644 --- a/src/conf/ucm/chtnau8824/HiFi.conf +++ b/src/conf/ucm/chtnau8824/HiFi.conf @@ -1,4 +1,3 @@ -<searchdir:ucm> SectionVerb { diff --git a/src/conf/ucm/chtrt5645/HiFi.conf b/src/conf/ucm/chtrt5645/HiFi.conf index 6a3c6875f301..422f3b7eef4f 100644 --- a/src/conf/ucm/chtrt5645/HiFi.conf +++ b/src/conf/ucm/chtrt5645/HiFi.conf @@ -1,4 +1,3 @@ -<searchdir:ucm> SectionVerb { # ALSA PCM diff --git a/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf b/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf index facc73ac3b2b..b931b7914ce7 100644 --- a/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf +++ b/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf @@ -1,4 +1,3 @@ -<searchdir:ucm> SectionVerb { diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 2b6f1159aa7d..5e1a8862fac1 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -34,9 +34,6 @@ #include <dirent.h> #include <limits.h> -/** The name of the environment variable containing the UCM directory */ -#define ALSA_CONFIG_UCM_VAR "ALSA_CONFIG_UCM" - /* Directories to store UCM configuration files for components, like * off-soc codecs or embedded DSPs. Components can define their own * devices and sequences, to be reused by sound cards/machines. UCM diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index ce2fc6ea18eb..8e7ac7ea48e9 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -243,3 +243,6 @@ void uc_mgr_free_sequence_element(struct sequence_element *seq); void uc_mgr_free_transition_element(struct transition_sequence *seq); void uc_mgr_free_verb(snd_use_case_mgr_t *uc_mgr); void uc_mgr_free(snd_use_case_mgr_t *uc_mgr); + +/** The name of the environment variable containing the UCM directory */ +#define ALSA_CONFIG_UCM_VAR "ALSA_CONFIG_UCM" diff --git a/src/ucm/utils.c b/src/ucm/utils.c index 18785488db07..ea5ac5dd1b01 100644 --- a/src/ucm/utils.c +++ b/src/ucm/utils.c @@ -54,6 +54,7 @@ int uc_mgr_config_load(const char *file, snd_config_t **cfg) FILE *fp; snd_input_t *in; snd_config_t *top; + const char *default_path; int err; fp = fopen(file, "r"); @@ -70,7 +71,11 @@ int uc_mgr_config_load(const char *file, snd_config_t **cfg) err = snd_config_top(&top); if (err < 0) return err; - err = snd_config_load(top, in); + + default_path = getenv(ALSA_CONFIG_UCM_VAR); + if (!default_path || !*default_path) + default_path = ALSA_CONFIG_DIR "/ucm"; + err = _snd_config_load_with_include(top, in, default_path); if (err < 0) { uc_error("could not load configuration file %s", file); snd_config_delete(top); -- 2.19.1

1 0

[alsa-devel] [PATCH v2] conf: Move UCM profile snippets into components subdirectory
by Takashi Iwai 29 Nov '18

29 Nov '18

We have placed UCM profile snippets to be included by the main config files also in the same directory, src/conf/ucm, it confuses alsaucm program that scans over all subdirectories. It thinks such a file is also the main config file, and spews errors like: % alsaucm ALSA lib utils.c:67:(uc_mgr_config_load) could not open configuration file /usr/share/alsa/ucm/bytcr/bytcr.conf ALSA lib parser.c:1427:(load_master_config) error: could not parse configuration for card bytcr alsaucm: unable to obtain card list: No such file or directory Actually we already defined the subdirectory for such components, and they are skipped at parsing the main configs. So we just need to move the files there -- this is what's done here. One more thing done here is to add a new component subdirectory, platforms, for definitions bytcr/* that don't match with neither the existing ones (codecs nor dsps). Suggested-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- configure.ac | 10 ++++++---- src/conf/ucm/Makefile.am | 6 ++---- .../HiFi.conf | 14 ++++++------- .../bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf | 14 ++++++------- .../bytcr-rt5640-mono-spk-in1-mic/HiFi.conf | 14 ++++++------- .../bytcr-rt5640-mono-spk-in3-mic/HiFi.conf | 14 ++++++------- .../HiFi.conf | 14 ++++++------- .../bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf | 14 ++++++------- .../bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf | 14 ++++++------- src/conf/ucm/bytcr-rt5640/HiFi.conf | 20 +++++++++---------- .../bytcr-rt5651-mono-spk-in1-mic/HiFi.conf | 14 ++++++------- .../HiFi.conf | 14 ++++++------- .../bytcr-rt5651-mono-spk-in2-mic/HiFi.conf | 14 ++++++------- .../bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf | 14 ++++++------- .../HiFi.conf | 14 ++++++------- .../bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf | 14 ++++++------- src/conf/ucm/bytcr-rt5651/HiFi.conf | 20 +++++++++---------- src/conf/ucm/chtnau8824/HiFi.conf | 14 ++++++------- src/conf/ucm/chtrt5645/HiFi.conf | 2 +- src/conf/ucm/codecs/Makefile.am | 4 ++++ .../ucm/{ => codecs}/nau8824/EnableSeq.conf | 0 .../ucm/{ => codecs}/nau8824/HeadPhones.conf | 0 .../ucm/{ => codecs}/nau8824/HeadsetMic.conf | 0 .../ucm/{ => codecs}/nau8824/InternalMic.conf | 0 src/conf/ucm/{ => codecs}/nau8824/Makefile.am | 2 +- .../ucm/{ => codecs}/nau8824/MonoSpeaker.conf | 0 .../ucm/{ => codecs}/nau8824/Speaker.conf | 0 .../ucm/{ => codecs}/rt5640/DigitalMics.conf | 0 .../ucm/{ => codecs}/rt5640/EnableSeq.conf | 0 .../ucm/{ => codecs}/rt5640/HeadPhones.conf | 0 .../ucm/{ => codecs}/rt5640/HeadsetMic.conf | 0 .../{ => codecs}/rt5640/IN1-InternalMic.conf | 0 .../{ => codecs}/rt5640/IN3-InternalMic.conf | 0 src/conf/ucm/{ => codecs}/rt5640/Makefile.am | 2 +- .../ucm/{ => codecs}/rt5640/MonoSpeaker.conf | 0 src/conf/ucm/{ => codecs}/rt5640/Speaker.conf | 0 .../ucm/{ => codecs}/rt5651/EnableSeq.conf | 0 .../rt5651/HeadPhones-swapped.conf | 0 .../ucm/{ => codecs}/rt5651/HeadPhones.conf | 0 .../{ => codecs}/rt5651/IN1-InternalMic.conf | 0 .../{ => codecs}/rt5651/IN12-InternalMic.conf | 0 .../{ => codecs}/rt5651/IN2-InternalMic.conf | 0 .../{ => codecs}/rt5651/IN3-HeadsetMic.conf | 0 src/conf/ucm/{ => codecs}/rt5651/Makefile.am | 2 +- .../ucm/{ => codecs}/rt5651/MonoSpeaker.conf | 0 src/conf/ucm/{ => codecs}/rt5651/Speaker.conf | 0 .../HiFi.conf | 14 ++++++------- src/conf/ucm/platforms/Makefile.am | 2 ++ .../ucm/{ => platforms}/bytcr/Makefile.am | 2 +- .../bytcr/PlatformDisableSeq.conf | 0 .../bytcr/PlatformEnableSeq.conf | 0 src/ucm/parser.c | 1 + 52 files changed, 145 insertions(+), 138 deletions(-) create mode 100644 src/conf/ucm/codecs/Makefile.am rename src/conf/ucm/{ => codecs}/nau8824/EnableSeq.conf (100%) rename src/conf/ucm/{ => codecs}/nau8824/HeadPhones.conf (100%) rename src/conf/ucm/{ => codecs}/nau8824/HeadsetMic.conf (100%) rename src/conf/ucm/{ => codecs}/nau8824/InternalMic.conf (100%) rename src/conf/ucm/{ => codecs}/nau8824/Makefile.am (79%) rename src/conf/ucm/{ => codecs}/nau8824/MonoSpeaker.conf (100%) rename src/conf/ucm/{ => codecs}/nau8824/Speaker.conf (100%) rename src/conf/ucm/{ => codecs}/rt5640/DigitalMics.conf (100%) rename src/conf/ucm/{ => codecs}/rt5640/EnableSeq.conf (100%) rename src/conf/ucm/{ => codecs}/rt5640/HeadPhones.conf (100%) rename src/conf/ucm/{ => codecs}/rt5640/HeadsetMic.conf (100%) rename src/conf/ucm/{ => codecs}/rt5640/IN1-InternalMic.conf (100%) rename src/conf/ucm/{ => codecs}/rt5640/IN3-InternalMic.conf (100%) rename src/conf/ucm/{ => codecs}/rt5640/Makefile.am (82%) rename src/conf/ucm/{ => codecs}/rt5640/MonoSpeaker.conf (100%) rename src/conf/ucm/{ => codecs}/rt5640/Speaker.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/EnableSeq.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/HeadPhones-swapped.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/HeadPhones.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/IN1-InternalMic.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/IN12-InternalMic.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/IN2-InternalMic.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/IN3-HeadsetMic.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/Makefile.am (85%) rename src/conf/ucm/{ => codecs}/rt5651/MonoSpeaker.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/Speaker.conf (100%) create mode 100644 src/conf/ucm/platforms/Makefile.am rename src/conf/ucm/{ => platforms}/bytcr/Makefile.am (71%) rename src/conf/ucm/{ => platforms}/bytcr/PlatformDisableSeq.conf (100%) rename src/conf/ucm/{ => platforms}/bytcr/PlatformEnableSeq.conf (100%) diff --git a/configure.ac b/configure.ac index 4c9d860fd0e5..d0cab2d6cf78 100644 --- a/configure.ac +++ b/configure.ac @@ -720,7 +720,6 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/Makefile \ src/conf/ucm/broadwell-rt286/Makefile \ src/conf/ucm/broxton-rt298/Makefile \ - src/conf/ucm/bytcr/Makefile \ src/conf/ucm/bytcr-rt5640/Makefile \ src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/Makefile \ src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/Makefile \ @@ -747,18 +746,21 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/HDAudio-Gigabyte-ALC1220DualCodecs/Makefile \ src/conf/ucm/HDAudio-Lenovo-DualCodecs/Makefile \ src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/Makefile \ - src/conf/ucm/nau8824/Makefile \ src/conf/ucm/PandaBoard/Makefile \ src/conf/ucm/PandaBoardES/Makefile \ src/conf/ucm/PAZ00/Makefile \ src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/Makefile \ - src/conf/ucm/rt5640/Makefile \ - src/conf/ucm/rt5651/Makefile \ src/conf/ucm/SDP4430/Makefile \ src/conf/ucm/skylake-rt286/Makefile \ src/conf/ucm/TECLAST-X80Pro-Defaultstring-CherryTrailCR/Makefile \ src/conf/ucm/tegraalc5632/Makefile \ src/conf/ucm/VEYRON-I2S/Makefile \ + src/conf/ucm/codecs/Makefile \ + src/conf/ucm/codecs/nau8824/Makefile \ + src/conf/ucm/codecs/rt5640/Makefile \ + src/conf/ucm/codecs/rt5651/Makefile \ + src/conf/ucm/platforms/Makefile \ + src/conf/ucm/platforms/bytcr/Makefile \ src/conf/topology/Makefile \ src/conf/topology/broadwell/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 7cdc6176584a..85c15090951d 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1,8 +1,9 @@ SUBDIRS=\ +codecs \ +platforms \ ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN \ broadwell-rt286 \ broxton-rt298 \ -bytcr \ bytcr-rt5640 \ bytcr-rt5640-mono-spk-dmic1-mic \ bytcr-rt5640-mono-spk-in1-mic \ @@ -29,13 +30,10 @@ gpd-win-pocket-rt5645 \ HDAudio-Gigabyte-ALC1220DualCodecs \ HDAudio-Lenovo-DualCodecs \ LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216 \ -nau8824 \ PandaBoard \ PandaBoardES \ PAZ00 \ PIPO-W2S-Defaultstring-CherryTrailCR \ -rt5640 \ -rt5651 \ SDP4430 \ skylake-rt286 \ TECLAST-X80Pro-Defaultstring-CherryTrailCR \ diff --git a/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf b/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf index 799163dbe481..facc73ac3b2b 100644 --- a/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf +++ b/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf @@ -11,18 +11,18 @@ SectionVerb { EnableSequence [ cdev "hw:chtnau8824" - <bytcr/PlatformEnableSeq.conf> - <nau8824/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/nau8824/EnableSeq.conf> ] DisableSequence [ cdev "hw:chtnau8824" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] } -<nau8824/MonoSpeaker.conf> -<nau8824/HeadPhones.conf> +<codecs/nau8824/MonoSpeaker.conf> +<codecs/nau8824/HeadPhones.conf> -<nau8824/InternalMic.conf> -<nau8824/HeadsetMic.conf> +<codecs/nau8824/InternalMic.conf> +<codecs/nau8824/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf index 938b359290d4..c78cdfb80a44 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf @@ -6,13 +6,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformEnableSeq.conf> - <rt5640/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5640/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -21,8 +21,8 @@ SectionVerb { } } -<rt5640/MonoSpeaker.conf> -<rt5640/HeadPhones.conf> +<codecs/rt5640/MonoSpeaker.conf> +<codecs/rt5640/HeadPhones.conf> -<rt5640/DigitalMics.conf> -<rt5640/HeadsetMic.conf> +<codecs/rt5640/DigitalMics.conf> +<codecs/rt5640/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf index f36b245e5092..34a5d53d363c 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf @@ -6,13 +6,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformEnableSeq.conf> - <rt5640/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5640/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -21,8 +21,8 @@ SectionVerb { } } -<rt5640/MonoSpeaker.conf> -<rt5640/HeadPhones.conf> +<codecs/rt5640/MonoSpeaker.conf> +<codecs/rt5640/HeadPhones.conf> -<rt5640/IN1-InternalMic.conf> -<rt5640/HeadsetMic.conf> +<codecs/rt5640/IN1-InternalMic.conf> +<codecs/rt5640/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf index 1a08ededc40c..0aa0c0266412 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf @@ -6,13 +6,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformEnableSeq.conf> - <rt5640/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5640/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -21,8 +21,8 @@ SectionVerb { } } -<rt5640/MonoSpeaker.conf> -<rt5640/HeadPhones.conf> +<codecs/rt5640/MonoSpeaker.conf> +<codecs/rt5640/HeadPhones.conf> -<rt5640/IN3-InternalMic.conf> -<rt5640/HeadsetMic.conf> +<codecs/rt5640/IN3-InternalMic.conf> +<codecs/rt5640/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf index 4587b2e1294e..e358d4f23f12 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf @@ -6,13 +6,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformEnableSeq.conf> - <rt5640/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5640/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -21,8 +21,8 @@ SectionVerb { } } -<rt5640/Speaker.conf> -<rt5640/HeadPhones.conf> +<codecs/rt5640/Speaker.conf> +<codecs/rt5640/HeadPhones.conf> -<rt5640/DigitalMics.conf> -<rt5640/HeadsetMic.conf> +<codecs/rt5640/DigitalMics.conf> +<codecs/rt5640/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf index decd396047aa..097560d02d12 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf @@ -6,13 +6,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformEnableSeq.conf> - <rt5640/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5640/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -21,8 +21,8 @@ SectionVerb { } } -<rt5640/Speaker.conf> -<rt5640/HeadPhones.conf> +<codecs/rt5640/Speaker.conf> +<codecs/rt5640/HeadPhones.conf> -<rt5640/IN1-InternalMic.conf> -<rt5640/HeadsetMic.conf> +<codecs/rt5640/IN1-InternalMic.conf> +<codecs/rt5640/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf index d243bf823511..8a018f63c9c6 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf @@ -6,13 +6,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformEnableSeq.conf> - <rt5640/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5640/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -21,8 +21,8 @@ SectionVerb { } } -<rt5640/Speaker.conf> -<rt5640/HeadPhones.conf> +<codecs/rt5640/Speaker.conf> +<codecs/rt5640/HeadPhones.conf> -<rt5640/IN3-InternalMic.conf> -<rt5640/HeadsetMic.conf> +<codecs/rt5640/IN3-InternalMic.conf> +<codecs/rt5640/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5640/HiFi.conf b/src/conf/ucm/bytcr-rt5640/HiFi.conf index c833fe74845b..20ebe2dd36e2 100644 --- a/src/conf/ucm/bytcr-rt5640/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640/HiFi.conf @@ -6,13 +6,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformEnableSeq.conf> - <rt5640/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5640/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -21,11 +21,11 @@ SectionVerb { } } -<rt5640/Speaker.conf> -<rt5640/MonoSpeaker.conf> -<rt5640/HeadPhones.conf> +<codecs/rt5640/Speaker.conf> +<codecs/rt5640/MonoSpeaker.conf> +<codecs/rt5640/HeadPhones.conf> -<rt5640/DigitalMics.conf> -<rt5640/IN1-InternalMic.conf> -<rt5640/IN3-InternalMic.conf> -<rt5640/HeadsetMic.conf> +<codecs/rt5640/DigitalMics.conf> +<codecs/rt5640/IN1-InternalMic.conf> +<codecs/rt5640/IN3-InternalMic.conf> +<codecs/rt5640/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf index dd5b35fc02cb..ca8a7d9078fb 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf @@ -5,13 +5,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformEnableSeq.conf> - <rt5651/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5651/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -20,8 +20,8 @@ SectionVerb { } } -<rt5651/MonoSpeaker.conf> -<rt5651/HeadPhones.conf> +<codecs/rt5651/MonoSpeaker.conf> +<codecs/rt5651/HeadPhones.conf> -<rt5651/IN1-InternalMic.conf> -<rt5651/IN3-HeadsetMic.conf> +<codecs/rt5651/IN1-InternalMic.conf> +<codecs/rt5651/IN3-HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf index 777ed7836ee5..60ef6b332ea2 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf @@ -5,13 +5,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformEnableSeq.conf> - <rt5651/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5651/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -20,8 +20,8 @@ SectionVerb { } } -<rt5651/MonoSpeaker.conf> -<rt5651/HeadPhones-swapped.conf> +<codecs/rt5651/MonoSpeaker.conf> +<codecs/rt5651/HeadPhones-swapped.conf> -<rt5651/IN2-InternalMic.conf> -<rt5651/IN3-HeadsetMic.conf> +<codecs/rt5651/IN2-InternalMic.conf> +<codecs/rt5651/IN3-HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf index 9f56c077b65a..e75210d4aa7e 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf @@ -5,13 +5,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformEnableSeq.conf> - <rt5651/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5651/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -20,8 +20,8 @@ SectionVerb { } } -<rt5651/MonoSpeaker.conf> -<rt5651/HeadPhones.conf> +<codecs/rt5651/MonoSpeaker.conf> +<codecs/rt5651/HeadPhones.conf> -<rt5651/IN2-InternalMic.conf> -<rt5651/IN3-HeadsetMic.conf> +<codecs/rt5651/IN2-InternalMic.conf> +<codecs/rt5651/IN3-HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf index e2a579ab2261..2b7675bce831 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf @@ -5,13 +5,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformEnableSeq.conf> - <rt5651/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5651/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -20,8 +20,8 @@ SectionVerb { } } -<rt5651/Speaker.conf> -<rt5651/HeadPhones.conf> +<codecs/rt5651/Speaker.conf> +<codecs/rt5651/HeadPhones.conf> -<rt5651/IN1-InternalMic.conf> -<rt5651/IN3-HeadsetMic.conf> +<codecs/rt5651/IN1-InternalMic.conf> +<codecs/rt5651/IN3-HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf index ae52e7196110..7a8c94bc5d6b 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf @@ -5,13 +5,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformEnableSeq.conf> - <rt5651/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5651/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -20,8 +20,8 @@ SectionVerb { } } -<rt5651/Speaker.conf> -<rt5651/HeadPhones.conf> +<codecs/rt5651/Speaker.conf> +<codecs/rt5651/HeadPhones.conf> -<rt5651/IN12-InternalMic.conf> -<rt5651/IN3-HeadsetMic.conf> +<codecs/rt5651/IN12-InternalMic.conf> +<codecs/rt5651/IN3-HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf index a73ff5f86b8f..45c7837c6fb9 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf @@ -5,13 +5,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformEnableSeq.conf> - <rt5651/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5651/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -20,8 +20,8 @@ SectionVerb { } } -<rt5651/Speaker.conf> -<rt5651/HeadPhones.conf> +<codecs/rt5651/Speaker.conf> +<codecs/rt5651/HeadPhones.conf> -<rt5651/IN2-InternalMic.conf> -<rt5651/IN3-HeadsetMic.conf> +<codecs/rt5651/IN2-InternalMic.conf> +<codecs/rt5651/IN3-HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5651/HiFi.conf b/src/conf/ucm/bytcr-rt5651/HiFi.conf index 8733403cf28d..d3928e9b9976 100644 --- a/src/conf/ucm/bytcr-rt5651/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651/HiFi.conf @@ -5,13 +5,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformEnableSeq.conf> - <rt5651/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5651/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -20,11 +20,11 @@ SectionVerb { } } -<rt5651/Speaker.conf> -<rt5651/MonoSpeaker.conf> -<rt5651/HeadPhones.conf> +<codecs/rt5651/Speaker.conf> +<codecs/rt5651/MonoSpeaker.conf> +<codecs/rt5651/HeadPhones.conf> -<rt5651/IN1-InternalMic.conf> -<rt5651/IN2-InternalMic.conf> -<rt5651/IN12-InternalMic.conf> -<rt5651/IN3-HeadsetMic.conf> +<codecs/rt5651/IN1-InternalMic.conf> +<codecs/rt5651/IN2-InternalMic.conf> +<codecs/rt5651/IN12-InternalMic.conf> +<codecs/rt5651/IN3-HeadsetMic.conf> diff --git a/src/conf/ucm/chtnau8824/HiFi.conf b/src/conf/ucm/chtnau8824/HiFi.conf index dc52800e5eaa..97a7ac7ba366 100644 --- a/src/conf/ucm/chtnau8824/HiFi.conf +++ b/src/conf/ucm/chtnau8824/HiFi.conf @@ -11,18 +11,18 @@ SectionVerb { EnableSequence [ cdev "hw:chtnau8824" - <bytcr/PlatformEnableSeq.conf> - <nau8824/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/nau8824/EnableSeq.conf> ] DisableSequence [ cdev "hw:chtnau8824" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] } -<nau8824/Speaker.conf> -<nau8824/HeadPhones.conf> +<codecs/nau8824/Speaker.conf> +<codecs/nau8824/HeadPhones.conf> -<nau8824/InternalMic.conf> -<nau8824/HeadsetMic.conf> +<codecs/nau8824/InternalMic.conf> +<codecs/nau8824/HeadsetMic.conf> diff --git a/src/conf/ucm/chtrt5645/HiFi.conf b/src/conf/ucm/chtrt5645/HiFi.conf index d993f6ae9929..9a72dab9343c 100644 --- a/src/conf/ucm/chtrt5645/HiFi.conf +++ b/src/conf/ucm/chtrt5645/HiFi.conf @@ -11,7 +11,7 @@ SectionVerb { EnableSequence [ cdev "hw:chtrt5645" - <bytcr/PlatformEnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> # Output Configuration cset "name='DAC1 L Mux' IF1 DAC" diff --git a/src/conf/ucm/codecs/Makefile.am b/src/conf/ucm/codecs/Makefile.am new file mode 100644 index 000000000000..2b374acb09fd --- /dev/null +++ b/src/conf/ucm/codecs/Makefile.am @@ -0,0 +1,4 @@ +SUBDIRS=\ +rt5640 \ +rt5651 \ +nau8824 diff --git a/src/conf/ucm/nau8824/EnableSeq.conf b/src/conf/ucm/codecs/nau8824/EnableSeq.conf similarity index 100% rename from src/conf/ucm/nau8824/EnableSeq.conf rename to src/conf/ucm/codecs/nau8824/EnableSeq.conf diff --git a/src/conf/ucm/nau8824/HeadPhones.conf b/src/conf/ucm/codecs/nau8824/HeadPhones.conf similarity index 100% rename from src/conf/ucm/nau8824/HeadPhones.conf rename to src/conf/ucm/codecs/nau8824/HeadPhones.conf diff --git a/src/conf/ucm/nau8824/HeadsetMic.conf b/src/conf/ucm/codecs/nau8824/HeadsetMic.conf similarity index 100% rename from src/conf/ucm/nau8824/HeadsetMic.conf rename to src/conf/ucm/codecs/nau8824/HeadsetMic.conf diff --git a/src/conf/ucm/nau8824/InternalMic.conf b/src/conf/ucm/codecs/nau8824/InternalMic.conf similarity index 100% rename from src/conf/ucm/nau8824/InternalMic.conf rename to src/conf/ucm/codecs/nau8824/InternalMic.conf diff --git a/src/conf/ucm/nau8824/Makefile.am b/src/conf/ucm/codecs/nau8824/Makefile.am similarity index 79% rename from src/conf/ucm/nau8824/Makefile.am rename to src/conf/ucm/codecs/nau8824/Makefile.am index b39b8bb4c934..fd3bafe135b7 100644 --- a/src/conf/ucm/nau8824/Makefile.am +++ b/src/conf/ucm/codecs/nau8824/Makefile.am @@ -1,5 +1,5 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/nau8824 +ucmdir = $(alsaconfigdir)/ucm/codecs/nau8824 ucm_DATA = EnableSeq.conf HeadPhones.conf HeadsetMic.conf InternalMic.conf \ MonoSpeaker.conf Speaker.conf EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/nau8824/MonoSpeaker.conf b/src/conf/ucm/codecs/nau8824/MonoSpeaker.conf similarity index 100% rename from src/conf/ucm/nau8824/MonoSpeaker.conf rename to src/conf/ucm/codecs/nau8824/MonoSpeaker.conf diff --git a/src/conf/ucm/nau8824/Speaker.conf b/src/conf/ucm/codecs/nau8824/Speaker.conf similarity index 100% rename from src/conf/ucm/nau8824/Speaker.conf rename to src/conf/ucm/codecs/nau8824/Speaker.conf diff --git a/src/conf/ucm/rt5640/DigitalMics.conf b/src/conf/ucm/codecs/rt5640/DigitalMics.conf similarity index 100% rename from src/conf/ucm/rt5640/DigitalMics.conf rename to src/conf/ucm/codecs/rt5640/DigitalMics.conf diff --git a/src/conf/ucm/rt5640/EnableSeq.conf b/src/conf/ucm/codecs/rt5640/EnableSeq.conf similarity index 100% rename from src/conf/ucm/rt5640/EnableSeq.conf rename to src/conf/ucm/codecs/rt5640/EnableSeq.conf diff --git a/src/conf/ucm/rt5640/HeadPhones.conf b/src/conf/ucm/codecs/rt5640/HeadPhones.conf similarity index 100% rename from src/conf/ucm/rt5640/HeadPhones.conf rename to src/conf/ucm/codecs/rt5640/HeadPhones.conf diff --git a/src/conf/ucm/rt5640/HeadsetMic.conf b/src/conf/ucm/codecs/rt5640/HeadsetMic.conf similarity index 100% rename from src/conf/ucm/rt5640/HeadsetMic.conf rename to src/conf/ucm/codecs/rt5640/HeadsetMic.conf diff --git a/src/conf/ucm/rt5640/IN1-InternalMic.conf b/src/conf/ucm/codecs/rt5640/IN1-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5640/IN1-InternalMic.conf rename to src/conf/ucm/codecs/rt5640/IN1-InternalMic.conf diff --git a/src/conf/ucm/rt5640/IN3-InternalMic.conf b/src/conf/ucm/codecs/rt5640/IN3-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5640/IN3-InternalMic.conf rename to src/conf/ucm/codecs/rt5640/IN3-InternalMic.conf diff --git a/src/conf/ucm/rt5640/Makefile.am b/src/conf/ucm/codecs/rt5640/Makefile.am similarity index 82% rename from src/conf/ucm/rt5640/Makefile.am rename to src/conf/ucm/codecs/rt5640/Makefile.am index a6fbdc572529..80fa8cfc758c 100644 --- a/src/conf/ucm/rt5640/Makefile.am +++ b/src/conf/ucm/codecs/rt5640/Makefile.am @@ -1,5 +1,5 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/rt5640 +ucmdir = $(alsaconfigdir)/ucm/codecs/rt5640 ucm_DATA = DigitalMics.conf EnableSeq.conf HeadPhones.conf HeadsetMic.conf \ IN1-InternalMic.conf IN3-InternalMic.conf MonoSpeaker.conf Speaker.conf EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/rt5640/MonoSpeaker.conf b/src/conf/ucm/codecs/rt5640/MonoSpeaker.conf similarity index 100% rename from src/conf/ucm/rt5640/MonoSpeaker.conf rename to src/conf/ucm/codecs/rt5640/MonoSpeaker.conf diff --git a/src/conf/ucm/rt5640/Speaker.conf b/src/conf/ucm/codecs/rt5640/Speaker.conf similarity index 100% rename from src/conf/ucm/rt5640/Speaker.conf rename to src/conf/ucm/codecs/rt5640/Speaker.conf diff --git a/src/conf/ucm/rt5651/EnableSeq.conf b/src/conf/ucm/codecs/rt5651/EnableSeq.conf similarity index 100% rename from src/conf/ucm/rt5651/EnableSeq.conf rename to src/conf/ucm/codecs/rt5651/EnableSeq.conf diff --git a/src/conf/ucm/rt5651/HeadPhones-swapped.conf b/src/conf/ucm/codecs/rt5651/HeadPhones-swapped.conf similarity index 100% rename from src/conf/ucm/rt5651/HeadPhones-swapped.conf rename to src/conf/ucm/codecs/rt5651/HeadPhones-swapped.conf diff --git a/src/conf/ucm/rt5651/HeadPhones.conf b/src/conf/ucm/codecs/rt5651/HeadPhones.conf similarity index 100% rename from src/conf/ucm/rt5651/HeadPhones.conf rename to src/conf/ucm/codecs/rt5651/HeadPhones.conf diff --git a/src/conf/ucm/rt5651/IN1-InternalMic.conf b/src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5651/IN1-InternalMic.conf rename to src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf diff --git a/src/conf/ucm/rt5651/IN12-InternalMic.conf b/src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5651/IN12-InternalMic.conf rename to src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf diff --git a/src/conf/ucm/rt5651/IN2-InternalMic.conf b/src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5651/IN2-InternalMic.conf rename to src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf diff --git a/src/conf/ucm/rt5651/IN3-HeadsetMic.conf b/src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf similarity index 100% rename from src/conf/ucm/rt5651/IN3-HeadsetMic.conf rename to src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf diff --git a/src/conf/ucm/rt5651/Makefile.am b/src/conf/ucm/codecs/rt5651/Makefile.am similarity index 85% rename from src/conf/ucm/rt5651/Makefile.am rename to src/conf/ucm/codecs/rt5651/Makefile.am index acc8ea491043..a68bd7bb9fcf 100644 --- a/src/conf/ucm/rt5651/Makefile.am +++ b/src/conf/ucm/codecs/rt5651/Makefile.am @@ -1,5 +1,5 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/rt5651 +ucmdir = $(alsaconfigdir)/ucm/codecs/rt5651 ucm_DATA = EnableSeq.conf HeadPhones.conf HeadPhones-swapped.conf \ IN1-InternalMic.conf IN2-InternalMic.conf IN12-InternalMic.conf \ IN3-HeadsetMic.conf Speaker.conf MonoSpeaker.conf diff --git a/src/conf/ucm/rt5651/MonoSpeaker.conf b/src/conf/ucm/codecs/rt5651/MonoSpeaker.conf similarity index 100% rename from src/conf/ucm/rt5651/MonoSpeaker.conf rename to src/conf/ucm/codecs/rt5651/MonoSpeaker.conf diff --git a/src/conf/ucm/rt5651/Speaker.conf b/src/conf/ucm/codecs/rt5651/Speaker.conf similarity index 100% rename from src/conf/ucm/rt5651/Speaker.conf rename to src/conf/ucm/codecs/rt5651/Speaker.conf diff --git a/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf b/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf index 799163dbe481..facc73ac3b2b 100644 --- a/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf +++ b/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf @@ -11,18 +11,18 @@ SectionVerb { EnableSequence [ cdev "hw:chtnau8824" - <bytcr/PlatformEnableSeq.conf> - <nau8824/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/nau8824/EnableSeq.conf> ] DisableSequence [ cdev "hw:chtnau8824" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] } -<nau8824/MonoSpeaker.conf> -<nau8824/HeadPhones.conf> +<codecs/nau8824/MonoSpeaker.conf> +<codecs/nau8824/HeadPhones.conf> -<nau8824/InternalMic.conf> -<nau8824/HeadsetMic.conf> +<codecs/nau8824/InternalMic.conf> +<codecs/nau8824/HeadsetMic.conf> diff --git a/src/conf/ucm/platforms/Makefile.am b/src/conf/ucm/platforms/Makefile.am new file mode 100644 index 000000000000..1bf252f0e0e3 --- /dev/null +++ b/src/conf/ucm/platforms/Makefile.am @@ -0,0 +1,2 @@ +SUBDIRS=\ +bytcr diff --git a/src/conf/ucm/bytcr/Makefile.am b/src/conf/ucm/platforms/bytcr/Makefile.am similarity index 71% rename from src/conf/ucm/bytcr/Makefile.am rename to src/conf/ucm/platforms/bytcr/Makefile.am index f7a54cbdedcd..c53b21438ad2 100644 --- a/src/conf/ucm/bytcr/Makefile.am +++ b/src/conf/ucm/platforms/bytcr/Makefile.am @@ -1,4 +1,4 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/bytcr +ucmdir = $(alsaconfigdir)/ucm/platforms/bytcr ucm_DATA = PlatformEnableSeq.conf PlatformDisableSeq.conf EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/bytcr/PlatformDisableSeq.conf b/src/conf/ucm/platforms/bytcr/PlatformDisableSeq.conf similarity index 100% rename from src/conf/ucm/bytcr/PlatformDisableSeq.conf rename to src/conf/ucm/platforms/bytcr/PlatformDisableSeq.conf diff --git a/src/conf/ucm/bytcr/PlatformEnableSeq.conf b/src/conf/ucm/platforms/bytcr/PlatformEnableSeq.conf similarity index 100% rename from src/conf/ucm/bytcr/PlatformEnableSeq.conf rename to src/conf/ucm/platforms/bytcr/PlatformEnableSeq.conf diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 219edb9691a6..2b6f1159aa7d 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -53,6 +53,7 @@ static const char * const component_dir[] = { "codecs", /* for off-soc codecs */ "dsps", /* for DSPs embedded in SoC */ + "platforms", /* for common platform implementations */ NULL, /* terminator */ }; -- 2.19.1

3 8

[alsa-devel] [PATCH] ALSA: hda/tegra: compatible string as shortname
by Sameer Pujar 29 Nov '18

29 Nov '18

By default HDA sound card is registered with shortname "tegra-hda". Same driver is used across tegra platforms and it is necessary to distinguish between platforms to use platform specific settings from userspace. One such example is, hdmi port on different platforms use different alsa pcm device ID. For hdmi playback to work it should open correct pcm device depending on the platform. This patch applies shortname from first compatible string provided in root node of device tree. Userspace then can use this card name to apply specific settings. Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 20 +++++++++++++++++++- 1 file changed, 19 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index dd7d424..84ea606 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -31,6 +31,7 @@ #include <linux/of_device.h> #include <linux/slab.h> #include <linux/time.h> +#include <linux/string.h> #include <sound/core.h> #include <sound/initval.h> @@ -344,6 +345,8 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) int err; unsigned short gcap; int irq_id = platform_get_irq(pdev, 0); + const char *sname; + struct device_node *root; err = hda_tegra_init_chip(chip, pdev); if (err) @@ -401,8 +404,23 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) return -ENODEV; } + /* driver name */ strcpy(card->driver, "tegra-hda"); - strcpy(card->shortname, "tegra-hda"); + + root = of_find_node_by_path("/"); + sname = of_get_property(root, "compatible", NULL); + of_node_put(root); + if (!sname) { + dev_err(card->dev, + "failed to get compatible property from root node\n"); + return -ENODEV; + } + /* shortname for card */ + if (strlen(sname) > sizeof(card->shortname)) + dev_info(card->dev, "truncating shortname for card\n"); + strncpy(card->shortname, sname, sizeof(card->shortname)); + + /* longname for card */ snprintf(card->longname, sizeof(card->longname), "%s at 0x%lx irq %i", card->shortname, bus->addr, bus->irq); -- 2.7.4 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

2 1

[alsa-devel] Subwoofers aren't working on ALC298 (Lenovo Yoga C930 Laptop)
by Jonathan Reeve 29 Nov '18

29 Nov '18

Hi Alsa Devs, I've been struggling with a sound issue for a while now. I just bought a new laptop that's known for its high-quality audio. It's billed as a Dolby Atmos speaker system, and it appears to be using a ALC298 card (or so I gathered from `aplay -l`). From what I can tell, it has tweeters in the back, and subwoofers on the bottom of the laptop. It sounds really great in Windows, but tinny and flat in Linux. I'm guessing this is because the subwoofers aren't being engaged. Here are some things I've tried so far: - Tweaking what is possible to tweak in GNOME. - Running pavucontrol. No useful options here. - Unmuting everything in alsamixer. Unmuting only some things in alsamixer. - Running hdajackretask, from the alsaTools package. It seems to correctly recognize that there are some unassigned pins, and even gives me the option of assigning them to "internal speaker (LFE)," which sounds like it could be a bass output. But when I click "apply now" or "apply boot options," the scripts fail by saying they can't find tools like md or tee. I can run the scripts it generates manually, but then the changes don't really seem to take effect. - Following the instructions from the ALSA page on the wiki, and trying all kinds of different options - All of the above, and also restarting alsa / snd_hda / pulseaudio all manually. As far as I can tell, I'm running the latest version of ALSA, since I'm on NixOS unstable, and [[https://github.com/NixOS/nixpkgs/blob/master/pkgs/os-specific/linux/alsa-fi… build script]] seems to grab the latest version. [[http://www.alsa-project.org/db/?f=8c072b3fbb139d350cd0e2c4c98cbb7736c6455a]…'s the output from my alsa-info.sh.]] I don't know enough about ALSA, or sound hardware, to know what's going on there. Is it obvious from my alsa-info which pins I should try reassigning to which channels? And does anyone know of a good way of doing this? Preferably one that doesn't use hdajackretask, which doesn't appear to work on NixOS. [[https://unix.stackexchange.com/questions/482885/how-can-i-get-my-laptops-in… also posted a few more details about this on this StackExchange question.]] I hope that getting this to work will not only help me, but others with this same laptop or speaker configuration. Thanks in advance for any help you'd be able to provide! Best, Jonathan

3 5

[alsa-devel] [RFC PATCH 0/6] ASoC:Intel:Skylake: Enable HDaudio legacy fallback
by Pierre-Louis Bossart 28 Nov '18

28 Nov '18

This patchset is provided as an RFC and should not be merged as is (Turkey break in the USA and more validation needed). This is however a good time to gather comments. This work is the result of multiple email discussions to finally provide more flexibility for distributions to chose whether to stick with the legacy HDaudio driver or to enable the SST/Skylake driver for newer platforms (required e.g. for digital microphone support) The patches add support for CoffeeLake, simplify the probe for the Skylake driver, introduce more compile-time granularity so that platforms can be selected individually and last provide a dynamic fallback mechanism when two drivers are registered for the same PCI ID. When the SOF driver is released, the same mechanism will be used to enable the SOF-legacy fallback. There will be no plans to enable an SOF->SST falback. Pierre-Louis Bossart (4): ASoC: Intel: Skylake: stop init/probe if DSP is not present ASoC: Intel: Skylake: remove useless tests on DSP presence ASoC: Intel: Skylake: Add more platform granularity ALSA: hda: add fallback capabilities for SKL+ platforms Takashi Iwai (2): ASoC: Intel: Skylake: Add CFL-S support ALSA: hda: Allow fallback binding with legacy HD-audio for Intel SKL+ include/sound/hdaudio.h | 6 + sound/pci/hda/Kconfig | 40 +++++++ sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 51 ++++++-- sound/soc/intel/Kconfig | 86 ++++++++++++-- sound/soc/intel/boards/Kconfig | 16 ++- sound/soc/intel/skylake/skl-messages.c | 8 ++ sound/soc/intel/skylake/skl.c | 154 +++++++++++++++++++------ 8 files changed, 311 insertions(+), 52 deletions(-) -- 2.17.1

3 19

[alsa-devel] [PATCH 1/1] ASoC: pcm3168a: Don't disable pcm3168a when CONFIG_PM defined
by jiada_wang＠mentor.com 28 Nov '18

28 Nov '18

From: Jiada Wang <jiada_wang(a)mentor.com> pcm3168 codec support runtime_[resume|suspend], whenever it is not active, it enters suspend mode, and it's clock and regulators will be disabled. so there is no need to disable them again in remove callback. Otherwise we got following kernel warnings, when unload pcm3168a driver [ 222.257514] unbalanced disables for amp-en-regulator [ 222.262526] ------------[ cut here ]------------ [ 222.267158] WARNING: CPU: 0 PID: 2423 at drivers/regulator/core.c:2264 _regulator_disable+0x28/0x108 [ 222.276291] Modules linked in: [ 222.279343] snd_soc_pcm3168a_i2c(-) [ 222.282916] snd_aloop [ 222.285272] arc4 [ 222.287194] wl18xx [ 222.289289] wlcore [ 222.291385] mac80211 [ 222.293654] cfg80211 [ 222.295923] aes_ce_blk [ 222.298366] crypto_simd [ 222.300896] cryptd [ 222.302992] aes_ce_cipher [ 222.305696] crc32_ce [ 222.307965] ghash_ce [ 222.310234] aes_arm64 [ 222.312590] gf128mul [ 222.314860] snd_soc_rcar [ 222.317476] sha2_ce [ 222.319658] xhci_plat_hcd [ 222.322362] sha256_arm64 [ 222.324978] xhci_hcd [ 222.327247] sha1_ce [ 222.329430] renesas_usbhs [ 222.332133] evdev [ 222.334142] sha1_generic [ 222.336758] rcar_gen3_thermal [ 222.339810] cpufreq_dt [ 222.342253] ravb_streaming(C) [ 222.345304] wlcore_sdio [ 222.347834] thermal_sys [ 222.350363] udc_core [ 222.352632] mch_core(C) [ 222.355161] usb_dmac [ 222.357430] snd_soc_pcm3168a [ 222.360394] snd_soc_ak4613 [ 222.363184] gpio_keys [ 222.365540] virt_dma [ 222.367809] nfsd [ 222.369730] ipv6 [ 222.371652] autofs4 [ 222.373834] [last unloaded: snd_soc_pcm3168a_i2c] [ 222.378629] CPU: 0 PID: 2423 Comm: rmmod Tainted: G WC 4.14.63-04798-gd456126e4a42-dirty #457 [ 222.388196] Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT) [ 222.396199] task: ffff8006fa8c6200 task.stack: ffff00000a0a0000 [ 222.402117] PC is at _regulator_disable+0x28/0x108 [ 222.406906] LR is at _regulator_disable+0x28/0x108 [ 222.411695] pc : [<ffff0000083bd89c>] lr : [<ffff0000083bd89c>] pstate: 00000145 [ 222.419089] sp : ffff00000a0a3c80 [ 222.422401] x29: ffff00000a0a3c80 [ 222.425799] x28: ffff8006fa8c6200 [ 222.429199] x27: ffff0000086f1000 [ 222.432597] x26: 000000000000006a [ 222.435997] x25: 0000000000000124 [ 222.439395] x24: 0000000000000018 [ 222.442795] x23: 0000000000000006 [ 222.446193] x22: ffff8006f925d490 [ 222.449592] x21: ffff8006f9ac2068 [ 222.452991] x20: ffff8006f9ac2000 [ 222.456390] x19: 0000000000000005 [ 222.459787] x18: 000000000000000a [ 222.463186] x17: 0000000000000000 [ 222.466584] x16: 0000000000000000 [ 222.469984] x15: 000000000d3f616a [ 222.473382] x14: 0720072007200720 [ 222.476781] x13: 0720072007200720 [ 222.480179] x12: 0720072007200720 [ 222.483578] x11: 0720072007200720 [ 222.486975] x10: 0720072007200720 [ 222.490375] x9 : 0720072007200720 [ 222.493773] x8 : 07200772076f0774 [ 222.497172] x7 : 0000000000000000 [ 222.500570] x6 : 0000000000000007 [ 222.503969] x5 : 0000000000000000 [ 222.507367] x4 : 0000000000000000 [ 222.510766] x3 : 0000000000000000 [ 222.514164] x2 : c790b852091e2600 [ 222.517563] x1 : 0000000000000000 [ 222.520961] x0 : 0000000000000028 [ 222.524361] Call trace: [ 222.526805] Exception stack(0xffff00000a0a3b40 to 0xffff00000a0a3c80) [ 222.533245] 3b40: 0000000000000028 0000000000000000 c790b852091e2600 0000000000000000 [ 222.541075] 3b60: 0000000000000000 0000000000000000 0000000000000007 0000000000000000 [ 222.548905] 3b80: 07200772076f0774 0720072007200720 0720072007200720 0720072007200720 [ 222.556735] 3ba0: 0720072007200720 0720072007200720 0720072007200720 000000000d3f616a [ 222.564564] 3bc0: 0000000000000000 0000000000000000 000000000000000a 0000000000000005 [ 222.572394] 3be0: ffff8006f9ac2000 ffff8006f9ac2068 ffff8006f925d490 0000000000000006 [ 222.580224] 3c00: 0000000000000018 0000000000000124 000000000000006a ffff0000086f1000 [ 222.588053] 3c20: ffff8006fa8c6200 ffff00000a0a3c80 ffff0000083bd89c ffff00000a0a3c80 [ 222.595883] 3c40: ffff0000083bd89c 0000000000000145 0000000000000000 0000000000000000 [ 222.603713] 3c60: 0000ffffffffffff ffff00000a0a3c30 ffff00000a0a3c80 ffff0000083bd89c [ 222.611543] [<ffff0000083bd89c>] _regulator_disable+0x28/0x108 [ 222.617375] [<ffff0000083bd9c4>] regulator_disable+0x48/0x68 [ 222.623033] [<ffff0000083be8e4>] regulator_bulk_disable+0x58/0xc0 [ 222.629134] [<ffff0000007d831c>] pcm3168a_remove+0x30/0x50 [snd_soc_pcm3168a] [ 222.636270] [<ffff0000007e5010>] pcm3168a_i2c_remove+0x10/0x1c [snd_soc_pcm3168a_i2c] [ 222.644106] [<ffff0000084b9d9c>] i2c_device_remove+0x38/0x70 [ 222.649766] [<ffff00000843cd5c>] device_release_driver_internal+0xd0/0x1c0 [ 222.656640] [<ffff00000843ced8>] driver_detach+0x70/0x7c [ 222.661951] [<ffff00000843bf68>] bus_remove_driver+0x74/0xa0 [ 222.667609] [<ffff00000843d7e4>] driver_unregister+0x48/0x4c [ 222.673268] [<ffff0000084ba8dc>] i2c_del_driver+0x24/0x30 [ 222.678666] [<ffff0000007e5078>] pcm3168a_i2c_driver_exit+0x10/0xf98 [snd_soc_pcm3168a_i2c] [ 222.687019] [<ffff00000811bd28>] SyS_delete_module+0x198/0x1d4 [ 222.692850] Exception stack(0xffff00000a0a3ec0 to 0xffff00000a0a4000) [ 222.699289] 3ec0: 0000aaaafeb4b268 0000000000000800 14453f6470497100 0000fffffaa520d8 [ 222.707119] 3ee0: 0000fffffaa520d9 000000000000000a 1999999999999999 0000000000000000 [ 222.714948] 3f00: 000000000000006a 0000ffffa8f7d1d8 000000000000000a 0000000000000005 [ 222.722778] 3f20: 0000000000000000 0000000000000000 000000000000002d 0000000000000000 [ 222.730607] 3f40: 0000aaaae19b9f68 0000ffffa8f411f0 0000000000000000 0000aaaae19b9000 [ 222.738436] 3f60: 0000fffffaa533b8 0000fffffaa531f0 0000000000000000 0000000000000001 [ 222.746266] 3f80: 0000fffffaa53ec6 0000000000000000 0000aaaafeb4b200 0000aaaafeb4a010 [ 222.754096] 3fa0: 0000000000000000 0000fffffaa53130 0000aaaae199f36c 0000fffffaa53130 [ 222.761926] 3fc0: 0000ffffa8f411f8 0000000000000000 0000aaaafeb4b268 000000000000006a [ 222.769755] 3fe0: 0000000000000000 0000000000000000 0000000000000000 0000000000000000 [ 222.777589] [<ffff0000080832c0>] el0_svc_naked+0x34/0x38 [ 222.782899] ---[ end trace eaf8939a3698b1a8 ]--- [ 222.787609] Failed to disable VCCDA2: -5 [ 222.791649] ------------[ cut here ]------------ [ 222.796283] WARNING: CPU: 0 PID: 2423 at drivers/clk/clk.c:595 clk_core_disable+0xc/0x1d8 [ 222.804460] Modules linked in: [ 222.807511] snd_soc_pcm3168a_i2c(-) [ 222.811083] snd_aloop [ 222.813439] arc4 [ 222.815360] wl18xx [ 222.817456] wlcore [ 222.819551] mac80211 [ 222.821820] cfg80211 [ 222.824088] aes_ce_blk [ 222.826531] crypto_simd [ 222.829060] cryptd [ 222.831155] aes_ce_cipher [ 222.833859] crc32_ce [ 222.836127] ghash_ce [ 222.838396] aes_arm64 [ 222.840752] gf128mul [ 222.843020] snd_soc_rcar [ 222.845637] sha2_ce [ 222.847818] xhci_plat_hcd [ 222.850522] sha256_arm64 [ 222.853138] xhci_hcd [ 222.855407] sha1_ce [ 222.857589] renesas_usbhs [ 222.860292] evdev [ 222.862300] sha1_generic [ 222.864917] rcar_gen3_thermal [ 222.867968] cpufreq_dt [ 222.870410] ravb_streaming(C) [ 222.873461] wlcore_sdio [ 222.875991] thermal_sys [ 222.878520] udc_core [ 222.880789] mch_core(C) [ 222.883318] usb_dmac [ 222.885587] snd_soc_pcm3168a [ 222.888551] snd_soc_ak4613 [ 222.891341] gpio_keys [ 222.893696] virt_dma [ 222.895965] nfsd [ 222.897886] ipv6 [ 222.899808] autofs4 [ 222.901990] [last unloaded: snd_soc_pcm3168a_i2c] [ 222.906783] CPU: 0 PID: 2423 Comm: rmmod Tainted: G WC 4.14.63-04798-gd456126e4a42-dirty #457 [ 222.916349] Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT) [ 222.924351] task: ffff8006fa8c6200 task.stack: ffff00000a0a0000 [ 222.930270] PC is at clk_core_disable+0xc/0x1d8 [ 222.934799] LR is at clk_core_disable_lock+0x20/0x34 [ 222.939761] pc : [<ffff0000083ab9b8>] lr : [<ffff0000083acd28>] pstate: 800001c5 [ 222.947154] sp : ffff00000a0a3cf0 [ 222.950466] x29: ffff00000a0a3cf0 [ 222.953864] x28: ffff8006fa8c6200 [ 222.957263] x27: ffff0000086f1000 [ 222.960661] x26: 000000000000006a [ 222.964061] x25: 0000000000000124 [ 222.967458] x24: 0000000000000015 [ 222.970858] x23: ffff8006f9ffa8d0 [ 222.974256] x22: ffff8006faf16480 [ 222.977655] x21: ffff0000007e7040 [ 222.981053] x20: ffff8006faadd100 [ 222.984452] x19: 0000000000000140 [ 222.987850] x18: 000000000000000a [ 222.991249] x17: 0000000000000000 [ 222.994647] x16: 0000000000000000 [ 222.998046] x15: 000000000d477819 [ 223.001444] x14: 0720072007200720 [ 223.004843] x13: 0720072007200720 [ 223.008242] x12: 0720072007200720 [ 223.011641] x11: 0720072007200720 [ 223.015039] x10: 0720072007200720 [ 223.018438] x9 : 0720072007200720 [ 223.021837] x8 : 0720072007200720 [ 223.025236] x7 : 0000000000000000 [ 223.028634] x6 : 0000000000000007 [ 223.032034] x5 : 0000000000000000 [ 223.035432] x4 : 0000000000000000 [ 223.038831] x3 : 0000000000000000 [ 223.042229] x2 : 0000000004720471 [ 223.045628] x1 : 0000000000000000 [ 223.049026] x0 : ffff8006faadd100 [ 223.052426] Call trace: [ 223.054870] Exception stack(0xffff00000a0a3bb0 to 0xffff00000a0a3cf0) [ 223.061309] 3ba0: ffff8006faadd100 0000000000000000 [ 223.069139] 3bc0: 0000000004720471 0000000000000000 0000000000000000 0000000000000000 [ 223.076969] 3be0: 0000000000000007 0000000000000000 0720072007200720 0720072007200720 [ 223.084798] 3c00: 0720072007200720 0720072007200720 0720072007200720 0720072007200720 [ 223.092628] 3c20: 0720072007200720 000000000d477819 0000000000000000 0000000000000000 [ 223.100458] 3c40: 000000000000000a 0000000000000140 ffff8006faadd100 ffff0000007e7040 [ 223.108287] 3c60: ffff8006faf16480 ffff8006f9ffa8d0 0000000000000015 0000000000000124 [ 223.116117] 3c80: 000000000000006a ffff0000086f1000 ffff8006fa8c6200 ffff00000a0a3cf0 [ 223.123947] 3ca0: ffff0000083acd28 ffff00000a0a3cf0 ffff0000083ab9b8 00000000800001c5 [ 223.131777] 3cc0: ffff00000a0a3cf0 ffff0000083acd1c 0000ffffffffffff ffff8006faadd100 [ 223.139606] 3ce0: ffff00000a0a3cf0 ffff0000083ab9b8 [ 223.144483] [<ffff0000083ab9b8>] clk_core_disable+0xc/0x1d8 [ 223.150054] [<ffff0000083acd58>] clk_disable+0x1c/0x28 [ 223.155198] [<ffff0000007d8328>] pcm3168a_remove+0x3c/0x50 [snd_soc_pcm3168a] [ 223.162334] [<ffff0000007e5010>] pcm3168a_i2c_remove+0x10/0x1c [snd_soc_pcm3168a_i2c] [ 223.170167] [<ffff0000084b9d9c>] i2c_device_remove+0x38/0x70 [ 223.175826] [<ffff00000843cd5c>] device_release_driver_internal+0xd0/0x1c0 [ 223.182700] [<ffff00000843ced8>] driver_detach+0x70/0x7c [ 223.188012] [<ffff00000843bf68>] bus_remove_driver+0x74/0xa0 [ 223.193669] [<ffff00000843d7e4>] driver_unregister+0x48/0x4c [ 223.199329] [<ffff0000084ba8dc>] i2c_del_driver+0x24/0x30 [ 223.204726] [<ffff0000007e5078>] pcm3168a_i2c_driver_exit+0x10/0xf98 [snd_soc_pcm3168a_i2c] [ 223.213079] [<ffff00000811bd28>] SyS_delete_module+0x198/0x1d4 [ 223.218909] Exception stack(0xffff00000a0a3ec0 to 0xffff00000a0a4000) [ 223.225349] 3ec0: 0000aaaafeb4b268 0000000000000800 14453f6470497100 0000fffffaa520d8 [ 223.233179] 3ee0: 0000fffffaa520d9 000000000000000a 1999999999999999 0000000000000000 [ 223.241008] 3f00: 000000000000006a 0000ffffa8f7d1d8 000000000000000a 0000000000000005 [ 223.248838] 3f20: 0000000000000000 0000000000000000 000000000000002d 0000000000000000 [ 223.256668] 3f40: 0000aaaae19b9f68 0000ffffa8f411f0 0000000000000000 0000aaaae19b9000 [ 223.264497] 3f60: 0000fffffaa533b8 0000fffffaa531f0 0000000000000000 0000000000000001 [ 223.272327] 3f80: 0000fffffaa53ec6 0000000000000000 0000aaaafeb4b200 0000aaaafeb4a010 [ 223.280157] 3fa0: 0000000000000000 0000fffffaa53130 0000aaaae199f36c 0000fffffaa53130 [ 223.287986] 3fc0: 0000ffffa8f411f8 0000000000000000 0000aaaafeb4b268 000000000000006a [ 223.295816] 3fe0: 0000000000000000 0000000000000000 0000000000000000 0000000000000000 [ 223.303648] [<ffff0000080832c0>] el0_svc_naked+0x34/0x38 [ 223.308958] ---[ end trace eaf8939a3698b1a9 ]--- [ 223.313752] ------------[ cut here ]------------ [ 223.318383] WARNING: CPU: 0 PID: 2423 at drivers/clk/clk.c:477 clk_core_unprepare+0xc/0x1ac [ 223.326733] Modules linked in: [ 223.329784] snd_soc_pcm3168a_i2c(-) [ 223.333356] snd_aloop [ 223.335712] arc4 [ 223.337633] wl18xx [ 223.339728] wlcore [ 223.341823] mac80211 [ 223.344092] cfg80211 [ 223.346360] aes_ce_blk [ 223.348803] crypto_simd [ 223.351332] cryptd [ 223.353428] aes_ce_cipher [ 223.356131] crc32_ce [ 223.358400] ghash_ce [ 223.360668] aes_arm64 [ 223.363024] gf128mul [ 223.365293] snd_soc_rcar [ 223.367909] sha2_ce [ 223.370091] xhci_plat_hcd [ 223.372794] sha256_arm64 [ 223.375410] xhci_hcd [ 223.377679] sha1_ce [ 223.379861] renesas_usbhs [ 223.382564] evdev [ 223.384572] sha1_generic [ 223.387188] rcar_gen3_thermal [ 223.390239] cpufreq_dt [ 223.392682] ravb_streaming(C) [ 223.395732] wlcore_sdio [ 223.398261] thermal_sys [ 223.400790] udc_core [ 223.403059] mch_core(C) [ 223.405588] usb_dmac [ 223.407856] snd_soc_pcm3168a [ 223.410820] snd_soc_ak4613 [ 223.413609] gpio_keys [ 223.415965] virt_dma [ 223.418234] nfsd [ 223.420155] ipv6 [ 223.422076] autofs4 [ 223.424258] [last unloaded: snd_soc_pcm3168a_i2c] [ 223.429050] CPU: 0 PID: 2423 Comm: rmmod Tainted: G WC 4.14.63-04798-gd456126e4a42-dirty #457 [ 223.438616] Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT) [ 223.446618] task: ffff8006fa8c6200 task.stack: ffff00000a0a0000 [ 223.452536] PC is at clk_core_unprepare+0xc/0x1ac [ 223.457239] LR is at clk_unprepare+0x28/0x3c [ 223.461506] pc : [<ffff0000083ab5a4>] lr : [<ffff0000083ace4c>] pstate: 60000145 [ 223.468900] sp : ffff00000a0a3d00 [ 223.472211] x29: ffff00000a0a3d00 [ 223.475609] x28: ffff8006fa8c6200 [ 223.479009] x27: ffff0000086f1000 [ 223.482407] x26: 000000000000006a [ 223.485807] x25: 0000000000000124 [ 223.489205] x24: 0000000000000015 [ 223.492604] x23: ffff8006f9ffa8d0 [ 223.496003] x22: ffff8006faf16480 [ 223.499402] x21: ffff0000007e7040 [ 223.502800] x20: ffff8006faf16420 [ 223.506199] x19: ffff8006faadd100 [ 223.509597] x18: 000000000000000a [ 223.512997] x17: 0000000000000000 [ 223.516395] x16: 0000000000000000 [ 223.519794] x15: 0000000000000000 [ 223.523192] x14: 00000033fe89076c [ 223.526591] x13: 0000000000000400 [ 223.529989] x12: 0000000000000400 [ 223.533388] x11: 0000000000000000 [ 223.536786] x10: 00000000000009e0 [ 223.540185] x9 : ffff00000a0a3be0 [ 223.543583] x8 : ffff8006fa8c6c40 [ 223.546982] x7 : ffff8006fa8c6400 [ 223.550380] x6 : 0000000000000001 [ 223.553780] x5 : 0000000000000000 [ 223.557178] x4 : ffff8006fa8c6200 [ 223.560577] x3 : 0000000000000000 [ 223.563975] x2 : ffff8006fa8c6200 [ 223.567374] x1 : 0000000000000000 [ 223.570772] x0 : ffff8006faadd100 [ 223.574170] Call trace: [ 223.576615] Exception stack(0xffff00000a0a3bc0 to 0xffff00000a0a3d00) [ 223.583054] 3bc0: ffff8006faadd100 0000000000000000 ffff8006fa8c6200 0000000000000000 [ 223.590884] 3be0: ffff8006fa8c6200 0000000000000000 0000000000000001 ffff8006fa8c6400 [ 223.598714] 3c00: ffff8006fa8c6c40 ffff00000a0a3be0 00000000000009e0 0000000000000000 [ 223.606544] 3c20: 0000000000000400 0000000000000400 00000033fe89076c 0000000000000000 [ 223.614374] 3c40: 0000000000000000 0000000000000000 000000000000000a ffff8006faadd100 [ 223.622204] 3c60: ffff8006faf16420 ffff0000007e7040 ffff8006faf16480 ffff8006f9ffa8d0 [ 223.630033] 3c80: 0000000000000015 0000000000000124 000000000000006a ffff0000086f1000 [ 223.637863] 3ca0: ffff8006fa8c6200 ffff00000a0a3d00 ffff0000083ace4c ffff00000a0a3d00 [ 223.645693] 3cc0: ffff0000083ab5a4 0000000060000145 0000000000000140 ffff8006faadd100 [ 223.653523] 3ce0: 0000ffffffffffff ffff0000083ace44 ffff00000a0a3d00 ffff0000083ab5a4 [ 223.661353] [<ffff0000083ab5a4>] clk_core_unprepare+0xc/0x1ac [ 223.667103] [<ffff0000007d8330>] pcm3168a_remove+0x44/0x50 [snd_soc_pcm3168a] [ 223.674239] [<ffff0000007e5010>] pcm3168a_i2c_remove+0x10/0x1c [snd_soc_pcm3168a_i2c] [ 223.682070] [<ffff0000084b9d9c>] i2c_device_remove+0x38/0x70 [ 223.687731] [<ffff00000843cd5c>] device_release_driver_internal+0xd0/0x1c0 [ 223.694604] [<ffff00000843ced8>] driver_detach+0x70/0x7c [ 223.699915] [<ffff00000843bf68>] bus_remove_driver+0x74/0xa0 [ 223.705572] [<ffff00000843d7e4>] driver_unregister+0x48/0x4c [ 223.711230] [<ffff0000084ba8dc>] i2c_del_driver+0x24/0x30 [ 223.716628] [<ffff0000007e5078>] pcm3168a_i2c_driver_exit+0x10/0xf98 [snd_soc_pcm3168a_i2c] [ 223.724980] [<ffff00000811bd28>] SyS_delete_module+0x198/0x1d4 [ 223.730811] Exception stack(0xffff00000a0a3ec0 to 0xffff00000a0a4000) [ 223.737250] 3ec0: 0000aaaafeb4b268 0000000000000800 14453f6470497100 0000fffffaa520d8 [ 223.745079] 3ee0: 0000fffffaa520d9 000000000000000a 1999999999999999 0000000000000000 [ 223.752909] 3f00: 000000000000006a 0000ffffa8f7d1d8 000000000000000a 0000000000000005 [ 223.760739] 3f20: 0000000000000000 0000000000000000 000000000000002d 0000000000000000 [ 223.768568] 3f40: 0000aaaae19b9f68 0000ffffa8f411f0 0000000000000000 0000aaaae19b9000 [ 223.776398] 3f60: 0000fffffaa533b8 0000fffffaa531f0 0000000000000000 0000000000000001 [ 223.784227] 3f80: 0000fffffaa53ec6 0000000000000000 0000aaaafeb4b200 0000aaaafeb4a010 [ 223.792057] 3fa0: 0000000000000000 0000fffffaa53130 0000aaaae199f36c 0000fffffaa53130 [ 223.799886] 3fc0: 0000ffffa8f411f8 0000000000000000 0000aaaafeb4b268 000000000000006a [ 223.807715] 3fe0: 0000000000000000 0000000000000000 0000000000000000 0000000000000000 [ 223.815546] [<ffff0000080832c0>] el0_svc_naked+0x34/0x38 [ 223.820855] ---[ end trace eaf8939a3698b1aa ]--- Fix this issue by only disable clock and regulators in remove callback when CONFIG_PM isn't defined Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/codecs/pcm3168a.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 115f21658222..08d3fe192e65 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -758,15 +758,22 @@ int pcm3168a_probe(struct device *dev, struct regmap *regmap) } EXPORT_SYMBOL_GPL(pcm3168a_probe); -void pcm3168a_remove(struct device *dev) +static void pcm3168a_disable(struct device *dev) { struct pcm3168a_priv *pcm3168a = dev_get_drvdata(dev); - pm_runtime_disable(dev); regulator_bulk_disable(ARRAY_SIZE(pcm3168a->supplies), - pcm3168a->supplies); + pcm3168a->supplies); clk_disable_unprepare(pcm3168a->scki); } + +void pcm3168a_remove(struct device *dev) +{ + pm_runtime_disable(dev); +#ifndef CONFIG_PM + pcm3168a_disable(dev); +#endif +} EXPORT_SYMBOL_GPL(pcm3168a_remove); #ifdef CONFIG_PM @@ -821,10 +828,7 @@ static int pcm3168a_rt_suspend(struct device *dev) regcache_cache_only(pcm3168a->regmap, true); - regulator_bulk_disable(ARRAY_SIZE(pcm3168a->supplies), - pcm3168a->supplies); - - clk_disable_unprepare(pcm3168a->scki); + pcm3168a_disable(dev); return 0; } -- 2.17.0

2 2

[alsa-devel] [PATCH] ASoC: rsnd: tidyup registering method for rsnd_kctrl_new()
by Kuninori Morimoto 28 Nov '18

28 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd dvc.c is using flags to avoid duplicating register for MIXer case. OTOH, commit e894efef9ac7 ("ASoC: core: add support to card rebind") allows to rebind sound card without rebinding all drivers. Because of above patch and dvc.c flags, it can't re-register kctrl if only sound card was rebinded, because dvc is keeping old flags. (Of course it will be no problem if rsnd driver also be rebinded, but it is not purpose of above patch). This patch checks current card registered kctrl when registering. In MIXer case, it can avoid duplicate register if card already has same kctrl. In rebind case, it can re-register kctrl because card registered kctl had been removed when unbinding. This patch is updated version of commit b918f1bc7f1ce ("ASoC: rsnd: DVC kctrl sets once") Reported-by: Nguyen Viet Dung <dung.nguyen.aj(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Nguyen Viet Dung <dung.nguyen.aj(a)renesas.com> Cc: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- > Mark This patch is for v4.20-rcX sound/soc/sh/rcar/core.c | 12 ++++++++++++ sound/soc/sh/rcar/dvc.c | 8 -------- 2 files changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 56469ac..031cd5b 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1476,6 +1476,18 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, }; int ret; + /* + * 1) Avoid duplicate register (ex. MIXer case) + * 2) re-register if card was rebinded + */ + list_for_each_entry(kctrl, &card->controls, list) { + struct rsnd_kctrl_cfg *c = kctrl->private_data; + + if (strcmp(kctrl->id.name, name) == 0 && + c->mod == mod) + return 0; + } + if (size > RSND_MAX_CHANNELS) return -EINVAL; diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index d65f24b..27c3761 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -40,11 +40,8 @@ struct rsnd_dvc { struct rsnd_kctrl_cfg_s ren; /* Ramp Enable */ struct rsnd_kctrl_cfg_s rup; /* Ramp Rate Up */ struct rsnd_kctrl_cfg_s rdown; /* Ramp Rate Down */ - u32 flags; }; -#define KCTRL_INITIALIZED (1 << 0) - #define rsnd_dvc_get(priv, id) ((struct rsnd_dvc *)(priv->dvc) + id) #define rsnd_dvc_nr(priv) ((priv)->dvc_nr) @@ -227,9 +224,6 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, int channels = rsnd_rdai_channels_get(rdai); int ret; - if (rsnd_flags_has(dvc, KCTRL_INITIALIZED)) - return 0; - /* Volume */ ret = rsnd_kctrl_new_m(mod, io, rtd, is_play ? @@ -285,8 +279,6 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, if (ret < 0) return ret; - rsnd_flags_set(dvc, KCTRL_INITIALIZED); - return 0; } -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: qcom: sdm845: Add board specific dapm widgets" to the asoc tree
by Mark Brown 28 Nov '18

28 Nov '18

The patch ASoC: qcom: sdm845: Add board specific dapm widgets 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 7f1806ebb08cc28e17dc3320790a01ccdff3d17d Mon Sep 17 00:00:00 2001 From: Cheng-Yi Chiang <cychiang(a)chromium.org> Date: Sat, 24 Nov 2018 19:09:45 +0800 Subject: [PATCH] ASoC: qcom: sdm845: Add board specific dapm widgets Add board specific dapm widgets so these widgets can be used in the route. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Acked-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/sdm845.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 58593db2ab15..95d8d4422dae 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -212,6 +212,14 @@ static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static const struct snd_soc_dapm_widget sdm845_snd_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + static void sdm845_add_be_ops(struct snd_soc_card *card) { struct snd_soc_dai_link *link; @@ -243,6 +251,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) goto data_alloc_fail; } + card->dapm_widgets = sdm845_snd_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: qcom: sdm845: Create and setup jack in init callback" to the asoc tree
by Mark Brown 28 Nov '18

28 Nov '18

The patch ASoC: qcom: sdm845: Create and setup jack in init callback 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 80e733fda42f511b346106351ba7b73d5a8196d4 Mon Sep 17 00:00:00 2001 From: Cheng-Yi Chiang <cychiang(a)chromium.org> Date: Sat, 24 Nov 2018 19:09:46 +0800 Subject: [PATCH] ASoC: qcom: sdm845: Create and setup jack in init callback Add a callback for init ops on dai_link to create and setup jack. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Acked-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/sdm845.c | 57 +++++++++++++++++++++++++++++++++++++++-- 1 file changed, 55 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 95d8d4422dae..43c03f8e8cdc 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -8,6 +8,8 @@ #include <linux/of_device.h> #include <sound/pcm.h> #include <sound/pcm_params.h> +#include <sound/jack.h> +#include <uapi/linux/input-event-codes.h> #include "common.h" #include "qdsp6/q6afe.h" @@ -17,6 +19,8 @@ #define MI2S_BCLK_RATE 1536000 struct sdm845_snd_data { + struct snd_soc_jack jack; + bool jack_setup; struct snd_soc_card *card; uint32_t pri_mi2s_clk_count; uint32_t sec_mi2s_clk_count; @@ -100,6 +104,54 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, return ret; } +static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component; + struct snd_soc_dai_link *dai_link = rtd->dai_link; + struct snd_soc_card *card = rtd->card; + struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card); + int i, rval; + + if (!pdata->jack_setup) { + struct snd_jack *jack; + + rval = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_HEADPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &pdata->jack, NULL, 0); + + if (rval < 0) { + dev_err(card->dev, "Unable to add Headphone Jack\n"); + return rval; + } + + jack = pdata->jack.jack; + + snd_jack_set_key(jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + pdata->jack_setup = true; + } + + for (i = 0 ; i < dai_link->num_codecs; i++) { + struct snd_soc_dai *dai = rtd->codec_dais[i]; + + component = dai->component; + rval = snd_soc_component_set_jack( + component, &pdata->jack, NULL); + if (rval != 0 && rval != -ENOTSUPP) { + dev_warn(card->dev, "Failed to set jack: %d\n", rval); + return rval; + } + } + + return 0; +} + + static int sdm845_snd_startup(struct snd_pcm_substream *substream) { unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS; @@ -220,7 +272,7 @@ static const struct snd_soc_dapm_widget sdm845_snd_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), }; -static void sdm845_add_be_ops(struct snd_soc_card *card) +static void sdm845_add_ops(struct snd_soc_card *card) { struct snd_soc_dai_link *link; int i; @@ -230,6 +282,7 @@ static void sdm845_add_be_ops(struct snd_soc_card *card) link->ops = &sdm845_be_ops; link->be_hw_params_fixup = sdm845_be_hw_params_fixup; } + link->init = sdm845_dai_init; } } @@ -264,7 +317,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) data->card = card; snd_soc_card_set_drvdata(card, data); - sdm845_add_be_ops(card); + sdm845_add_ops(card); ret = snd_soc_register_card(card); if (ret) { dev_err(dev, "Sound card registration failed\n"); -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH 0/2] ASoC: omap-mcpdm: Support for handling pdmclk
by Peter Ujfalusi 28 Nov '18

28 Nov '18

Hi, This series is on top of the pm_qos changes I have sent earlier today. I have sent this originally two years ago: https://patchwork.kernel.org/patch/9119551/ But needed to revert it as the MFD and clk changes were not merged in time: https://patchwork.kernel.org/patch/9287431/ The dts patches are already upstream, only the patch for McPDM is missing. I have separated the binding document update and the driver patch and also added the return value check for the clk_prepare_enable() as per Mark's comment. Regards, Peter --- Peter Ujfalusi (2): dt-bindings: sound: omap-mcpdm: Update documentation for pdmclk ASoC: omap-mcpdm: Add support for pdmclk clock handling .../devicetree/bindings/sound/omap-mcpdm.txt | 10 ++++++++ sound/soc/omap/omap-mcpdm.c | 23 +++++++++++++++++++ 2 files changed, 33 insertions(+) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

4 7

[alsa-devel] [PATCH v5 0/2] Introduce dmic mode switch delay parameter
by Jenny TC 28 Nov '18

28 Nov '18

Some DMICs need clock to be running for a specified duration as per the DMIC spec to complete the mode transition like sleep to normal, off to normal etc. This series add a mode-switch delay which will prevent the clock from being turned off without complying with manufacturer timing specifications. Also introduces a module parameter for the wakeup_delay parameter. v1: First patch set v2: 1). Removed ACPI device entry since a) The DMIC vendor doesn't have an ACPI vendor ID b) The ACPI enumeration creates one more device and conflicts with the existing skl.c dmic platform device registration. 2). Updated commit message v3: Updated commit message and introduced module param for wakeup_delay v4: Addressed parameter name to align with existing parameter. Added dt entry in Documentation v5: Validate modeswitch_delay to ensure sensible value Jenny TC (2): ASoC: dmic: introduce mode switch delay ASoC: dmic: introduce module_param wakeup_delay Documentation/devicetree/bindings/sound/dmic.txt | 2 ++ sound/soc/codecs/dmic.c | 40 ++++++++++++++++++++++++ 2 files changed, 42 insertions(+) -- 1.9.1

2 3

[alsa-devel] Applied "ASoC: dmic: introduce module_param wakeup_delay" to the asoc tree
by Mark Brown 28 Nov '18

28 Nov '18

The patch ASoC: dmic: introduce module_param wakeup_delay 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 f6f30a609c526dbf6d59490a8c85adaf6ac9b0fa Mon Sep 17 00:00:00 2001 From: Jenny TC <jenny.tc(a)intel.com> Date: Wed, 28 Nov 2018 12:22:46 +0530 Subject: [PATCH] ASoC: dmic: introduce module_param wakeup_delay Introducing a module param for wakeup_delay in order to align with modeswitch_delay parameter. With this change, both wakeup_delay and modeswitch_delay parameters can be passed as module parameters. Signed-off-by: Jenny TC <jenny.tc(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/dmic.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index f4eb0a438a3a..da921da50ef0 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -34,6 +34,9 @@ static int modeswitch_delay; module_param(modeswitch_delay, uint, 0644); +static int wakeup_delay; +module_param(wakeup_delay, uint, 0644); + struct dmic { struct gpio_desc *gpio_en; int wakeup_delay; @@ -115,6 +118,8 @@ static int dmic_component_probe(struct snd_soc_component *component) &dmic->wakeup_delay); device_property_read_u32(component->dev, "modeswitch-delay-ms", &dmic->modeswitch_delay); + if (wakeup_delay) + dmic->wakeup_delay = wakeup_delay; if (modeswitch_delay) dmic->modeswitch_delay = modeswitch_delay; -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: Intel: Skylake: Make DSP replies more human readable" to the asoc tree
by Mark Brown 28 Nov '18

28 Nov '18

The patch ASoC: Intel: Skylake: Make DSP replies more human readable 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 aa15679b2dc898049e9117fbe3ddda0b50fa52d2 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" <subhransu.s.prusty(a)intel.com> Date: Mon, 29 Jan 2018 21:45:59 +0530 Subject: [PATCH] ASoC: Intel: Skylake: Make DSP replies more human readable Add more meaning to the IPC replies for easy debugging. Replace the switch case with a lookup table to lookup for the IPC replies and print in human readable form. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-sst-ipc.c | 44 ++++++++++++++++++++------- 1 file changed, 33 insertions(+), 11 deletions(-) diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 5234fafb758a..8708755a8f9a 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -392,18 +392,43 @@ int skl_ipc_process_notification(struct sst_generic_ipc *ipc, return 0; } -static int skl_ipc_set_reply_error_code(u32 reply) +struct skl_ipc_err_map { + const char *msg; + enum skl_ipc_glb_reply reply; + int err; +}; + +static struct skl_ipc_err_map skl_err_map[] = { + {"DSP out of memory", IPC_GLB_REPLY_OUT_OF_MEMORY, -ENOMEM}, + {"DSP busy", IPC_GLB_REPLY_BUSY, -EBUSY}, +}; + +static int skl_ipc_set_reply_error_code(struct sst_generic_ipc *ipc, u32 reply) { - switch (reply) { - case IPC_GLB_REPLY_OUT_OF_MEMORY: - return -ENOMEM; + int i; - case IPC_GLB_REPLY_BUSY: - return -EBUSY; + for (i = 0; i < ARRAY_SIZE(skl_err_map); i++) { + if (skl_err_map[i].reply == reply) + break; + } - default: + if (i == ARRAY_SIZE(skl_err_map)) { + dev_err(ipc->dev, "ipc FW reply: %d FW Error Code: %u\n", + reply, + ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); return -EINVAL; } + + if (skl_err_map[i].err < 0) + dev_err(ipc->dev, "ipc FW reply: %s FW Error Code: %u\n", + skl_err_map[i].msg, + ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); + else + dev_info(ipc->dev, "ipc FW reply: %s FW Error Code: %u\n", + skl_err_map[i].msg, + ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); + + return skl_err_map[i].err; } void skl_ipc_process_reply(struct sst_generic_ipc *ipc, @@ -441,10 +466,7 @@ void skl_ipc_process_reply(struct sst_generic_ipc *ipc, } } else { - msg->errno = skl_ipc_set_reply_error_code(reply); - dev_err(ipc->dev, "ipc FW reply: reply=%d\n", reply); - dev_err(ipc->dev, "FW Error Code: %u\n", - ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); + msg->errno = skl_ipc_set_reply_error_code(ipc, reply); switch (IPC_GLB_NOTIFY_MSG_TYPE(header.primary)) { case IPC_GLB_LOAD_MULTIPLE_MODS: case IPC_GLB_LOAD_LIBRARY: -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: Intel: Skylake: Add FW reply for MCLK/SCLK IPC" to the asoc tree
by Mark Brown 28 Nov '18

28 Nov '18

The patch ASoC: Intel: Skylake: Add FW reply for MCLK/SCLK IPC 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 61f94ee4a7435c35d78b22e4ae6e0551908000ae Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" <subhransu.s.prusty(a)intel.com> Date: Mon, 29 Jan 2018 21:46:00 +0530 Subject: [PATCH] ASoC: Intel: Skylake: Add FW reply for MCLK/SCLK IPC If mclk/sclk is already running, FW responds with IPC reply MCLK/SCLK already running. Add these to the IPC reply lookup table. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-sst-ipc.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 8708755a8f9a..9f3ce73593ae 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -249,6 +249,8 @@ enum skl_ipc_glb_reply { IPC_GLB_REPLY_INVALID_CONFIG_DATA_LEN = 121, IPC_GLB_REPLY_GATEWAY_NOT_INITIALIZED = 140, IPC_GLB_REPLY_GATEWAY_NOT_EXIST = 141, + IPC_GLB_REPLY_SCLK_ALREADY_RUNNING = 150, + IPC_GLB_REPLY_MCLK_ALREADY_RUNNING = 151, IPC_GLB_REPLY_PPL_NOT_INITIALIZED = 160, IPC_GLB_REPLY_PPL_NOT_EXIST = 161, @@ -401,6 +403,10 @@ struct skl_ipc_err_map { static struct skl_ipc_err_map skl_err_map[] = { {"DSP out of memory", IPC_GLB_REPLY_OUT_OF_MEMORY, -ENOMEM}, {"DSP busy", IPC_GLB_REPLY_BUSY, -EBUSY}, + {"SCLK already running", IPC_GLB_REPLY_SCLK_ALREADY_RUNNING, + IPC_GLB_REPLY_SCLK_ALREADY_RUNNING}, + {"MCLK already running", IPC_GLB_REPLY_MCLK_ALREADY_RUNNING, + IPC_GLB_REPLY_MCLK_ALREADY_RUNNING}, }; static int skl_ipc_set_reply_error_code(struct sst_generic_ipc *ipc, u32 reply) -- 2.19.1

1 0

[alsa-devel] [PATCH 0/4] ASoC; davinci-mcasp: Pin handling updates
by Peter Ujfalusi 28 Nov '18

28 Nov '18

Hi, The series will improve the McASP pin handling: - To avoid non configured clocks leaking out from McASP - make sure that the AXR pin (TX) state is correct in all scenarios - Allow user configurable DISMOD for the tx pin instead of hardwiring it in the code to low The DISMOD configuration is needed when the codec requires the TX line to be high during inactive slots (mu-law codecs for example). Regards, Peter --- Peter Ujfalusi (4): ASoC: davinci-mcasp: Clear TXSTAT register before activating serializers ASoC: davinci-mcasp: Update PDIR (pin direction) register handling bindings: sound: davinci-mcasp: Document dismod optional property ASoC: davinci-mcasp: Implement configurable dismod handling .../bindings/sound/davinci-mcasp-audio.txt | 5 + include/linux/platform_data/davinci_asp.h | 1 + sound/soc/davinci/davinci-mcasp.c | 109 +++++++++++++++--- sound/soc/davinci/davinci-mcasp.h | 30 ++--- 4 files changed, 112 insertions(+), 33 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] Pioneer Interface 2 support?
by Daniel James 28 Nov '18

28 Nov '18

Hi folks, A user of the PiDeck project is interested in helping to add ALSA support for this USB audio interface: https://www.pioneerdj.com/en-gb/product/software/interface-2/audio-interfac… Some initial information is here: https://github.com/pideck/pideck/issues/85 Is there anyone willing to hack on this please? Does the info script available at http://www.alsa-project.org/main/index.php/Help_To_Debug provide additional details that would be useful to have? Thanks! Daniel

2 2

[alsa-devel] [PATCH 1/4] ASoC: qcom: sdm845: Add board specific dapm widgets
by Cheng-Yi Chiang 28 Nov '18

28 Nov '18

Add board specific dapm widgets so these widgets can be used in the route. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/qcom/sdm845.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 58593db2ab151..95d8d4422dae0 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -212,6 +212,14 @@ static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static const struct snd_soc_dapm_widget sdm845_snd_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + static void sdm845_add_be_ops(struct snd_soc_card *card) { struct snd_soc_dai_link *link; @@ -243,6 +251,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) goto data_alloc_fail; } + card->dapm_widgets = sdm845_snd_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); -- 2.20.0.rc0.387.gc7a69e6b6c-goog

2 8

[alsa-devel] HDA headset button support
by Kailang 28 Nov '18

28 Nov '18

Hi Takashi, New Chrome platform need to support HS button function. I sent it to Intel chrome team already. Could you take a look for attach patch? I think it maybe could upstream. It just waiting for test by Intel Chrome team. Thanks. BR, Kailang

2 1

[alsa-devel] [alsa-lib][PATCH v2] conf: USB-Audio: Add Dell WD19 Dock in the IEC958 blacklist
by Hui Wang 28 Nov '18

28 Nov '18

This Dock doesn't have IEC958 physical output, so add it to the blacklist to prevent it being opened. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- In the v2, the 2nd patch of adding wd19 ucm is dropped. here only adding wd19 into the blacklist, I don't know if wd15 has this issue or not, if it has, will adding wd15 to the blacklist too. src/conf/cards/USB-Audio.conf | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 05779c47..3a12a729 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -60,6 +60,7 @@ USB-Audio.pcm.iec958_device { "SWTOR Gaming Headset by Razer" 999 "USB Device 0x46d_0x821" 999 "USB Device 0x46d_0x992" 999 + "WD19 Dock" 999 } # Second iec958 device number, if any. -- 2.17.1

2 1

[alsa-devel] [PATCH v2] ALSA: usb-audio: Add vendor and product name for Dell WD19 Dock
by Hui Wang 28 Nov '18

28 Nov '18

Like the Dell WD15 Dock, the WD19 Dock (0bda:402e) doens't provide useful string for the vendor and product names too. In order to share the UCM with WD15, here we keep the profile_name same as the WD15. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/usb/quirks-table.h | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 849953e5775c..37fc0447c071 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3382,5 +3382,15 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = QUIRK_NO_INTERFACE } }, +/* Dell WD19 Dock */ +{ + USB_DEVICE(0x0bda, 0x402e), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Dell", + .product_name = "WD19 Dock", + .profile_name = "Dell-WD15-Dock", + .ifnum = QUIRK_NO_INTERFACE + } +}, #undef USB_DEVICE_VENDOR_SPEC -- 2.17.1

2 1

[alsa-devel] [PATCH] conf: Move UCM profile snippets into ucm-includes subdirectory
by Takashi Iwai 28 Nov '18

28 Nov '18

We have placed UCM profile snippets to be included by the main config files also in the same directory, src/conf/ucm, it confuses alsaucm program that scans over all subdirectories. It thinks such a file is also the main config file, and spews errors like: % alsaucm ALSA lib utils.c:67:(uc_mgr_config_load) could not open configuration file /usr/share/alsa/ucm/bytcr/bytcr.conf ALSA lib parser.c:1427:(load_master_config) error: could not parse configuration for card bytcr alsaucm: unable to obtain card list: No such file or directory For avoiding this, move UCM profile snippets into the own root directory, src/conf/ucm-includes, and include them by changing the include with "<searchdir:ucm-includes>". Suggested-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- configure.ac | 9 +++++---- src/conf/Makefile.am | 2 +- src/conf/ucm-includes/Makefile.am | 5 +++++ src/conf/{ucm => ucm-includes}/bytcr/Makefile.am | 2 +- .../{ucm => ucm-includes}/bytcr/PlatformDisableSeq.conf | 0 .../{ucm => ucm-includes}/bytcr/PlatformEnableSeq.conf | 0 src/conf/{ucm => ucm-includes}/nau8824/EnableSeq.conf | 0 src/conf/{ucm => ucm-includes}/nau8824/HeadPhones.conf | 0 src/conf/{ucm => ucm-includes}/nau8824/HeadsetMic.conf | 0 src/conf/{ucm => ucm-includes}/nau8824/InternalMic.conf | 0 src/conf/{ucm => ucm-includes}/nau8824/Makefile.am | 2 +- src/conf/{ucm => ucm-includes}/nau8824/MonoSpeaker.conf | 0 src/conf/{ucm => ucm-includes}/nau8824/Speaker.conf | 0 src/conf/{ucm => ucm-includes}/rt5640/DigitalMics.conf | 0 src/conf/{ucm => ucm-includes}/rt5640/EnableSeq.conf | 0 src/conf/{ucm => ucm-includes}/rt5640/HeadPhones.conf | 0 src/conf/{ucm => ucm-includes}/rt5640/HeadsetMic.conf | 0 .../{ucm => ucm-includes}/rt5640/IN1-InternalMic.conf | 0 .../{ucm => ucm-includes}/rt5640/IN3-InternalMic.conf | 0 src/conf/{ucm => ucm-includes}/rt5640/Makefile.am | 2 +- src/conf/{ucm => ucm-includes}/rt5640/MonoSpeaker.conf | 0 src/conf/{ucm => ucm-includes}/rt5640/Speaker.conf | 0 src/conf/{ucm => ucm-includes}/rt5651/EnableSeq.conf | 0 .../{ucm => ucm-includes}/rt5651/HeadPhones-swapped.conf | 0 src/conf/{ucm => ucm-includes}/rt5651/HeadPhones.conf | 0 .../{ucm => ucm-includes}/rt5651/IN1-InternalMic.conf | 0 .../{ucm => ucm-includes}/rt5651/IN12-InternalMic.conf | 0 .../{ucm => ucm-includes}/rt5651/IN2-InternalMic.conf | 0 .../{ucm => ucm-includes}/rt5651/IN3-HeadsetMic.conf | 0 src/conf/{ucm => ucm-includes}/rt5651/Makefile.am | 2 +- src/conf/{ucm => ucm-includes}/rt5651/MonoSpeaker.conf | 0 src/conf/{ucm => ucm-includes}/rt5651/Speaker.conf | 0 src/conf/ucm/Makefile.am | 4 ---- .../ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5640/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf | 2 +- .../bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5651/HiFi.conf | 2 +- src/conf/ucm/chtnau8824/HiFi.conf | 2 +- .../ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf | 2 +- 50 files changed, 32 insertions(+), 30 deletions(-) create mode 100644 src/conf/ucm-includes/Makefile.am rename src/conf/{ucm => ucm-includes}/bytcr/Makefile.am (72%) rename src/conf/{ucm => ucm-includes}/bytcr/PlatformDisableSeq.conf (100%) rename src/conf/{ucm => ucm-includes}/bytcr/PlatformEnableSeq.conf (100%) rename src/conf/{ucm => ucm-includes}/nau8824/EnableSeq.conf (100%) rename src/conf/{ucm => ucm-includes}/nau8824/HeadPhones.conf (100%) rename src/conf/{ucm => ucm-includes}/nau8824/HeadsetMic.conf (100%) rename src/conf/{ucm => ucm-includes}/nau8824/InternalMic.conf (100%) rename src/conf/{ucm => ucm-includes}/nau8824/Makefile.am (78%) rename src/conf/{ucm => ucm-includes}/nau8824/MonoSpeaker.conf (100%) rename src/conf/{ucm => ucm-includes}/nau8824/Speaker.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5640/DigitalMics.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5640/EnableSeq.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5640/HeadPhones.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5640/HeadsetMic.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5640/IN1-InternalMic.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5640/IN3-InternalMic.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5640/Makefile.am (81%) rename src/conf/{ucm => ucm-includes}/rt5640/MonoSpeaker.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5640/Speaker.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/EnableSeq.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/HeadPhones-swapped.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/HeadPhones.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/IN1-InternalMic.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/IN12-InternalMic.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/IN2-InternalMic.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/IN3-HeadsetMic.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/Makefile.am (84%) rename src/conf/{ucm => ucm-includes}/rt5651/MonoSpeaker.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/Speaker.conf (100%) diff --git a/configure.ac b/configure.ac index 4c9d860fd0e5..0b7a1bdc42fa 100644 --- a/configure.ac +++ b/configure.ac @@ -720,7 +720,6 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/Makefile \ src/conf/ucm/broadwell-rt286/Makefile \ src/conf/ucm/broxton-rt298/Makefile \ - src/conf/ucm/bytcr/Makefile \ src/conf/ucm/bytcr-rt5640/Makefile \ src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/Makefile \ src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/Makefile \ @@ -747,18 +746,20 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/HDAudio-Gigabyte-ALC1220DualCodecs/Makefile \ src/conf/ucm/HDAudio-Lenovo-DualCodecs/Makefile \ src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/Makefile \ - src/conf/ucm/nau8824/Makefile \ src/conf/ucm/PandaBoard/Makefile \ src/conf/ucm/PandaBoardES/Makefile \ src/conf/ucm/PAZ00/Makefile \ src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/Makefile \ - src/conf/ucm/rt5640/Makefile \ - src/conf/ucm/rt5651/Makefile \ src/conf/ucm/SDP4430/Makefile \ src/conf/ucm/skylake-rt286/Makefile \ src/conf/ucm/TECLAST-X80Pro-Defaultstring-CherryTrailCR/Makefile \ src/conf/ucm/tegraalc5632/Makefile \ src/conf/ucm/VEYRON-I2S/Makefile \ + src/conf/ucm-includes/Makefile \ + src/conf/ucm-includes/bytcr/Makefile \ + src/conf/ucm-includes/nau8824/Makefile \ + src/conf/ucm-includes/rt5640/Makefile \ + src/conf/ucm-includes/rt5651/Makefile \ src/conf/topology/Makefile \ src/conf/topology/broadwell/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ diff --git a/src/conf/Makefile.am b/src/conf/Makefile.am index 2b46f95c2cb7..4a4022e9bc1b 100644 --- a/src/conf/Makefile.am +++ b/src/conf/Makefile.am @@ -1,7 +1,7 @@ SUBDIRS=cards pcm if BUILD_UCM -SUBDIRS += ucm +SUBDIRS += ucm ucm-includes endif if BUILD_TOPOLOGY SUBDIRS += topology diff --git a/src/conf/ucm-includes/Makefile.am b/src/conf/ucm-includes/Makefile.am new file mode 100644 index 000000000000..f94bba7b837f --- /dev/null +++ b/src/conf/ucm-includes/Makefile.am @@ -0,0 +1,5 @@ +SUBDIRS=\ +bytcr \ +rt5640 \ +rt5651 \ +nau8824 diff --git a/src/conf/ucm/bytcr/Makefile.am b/src/conf/ucm-includes/bytcr/Makefile.am similarity index 72% rename from src/conf/ucm/bytcr/Makefile.am rename to src/conf/ucm-includes/bytcr/Makefile.am index f7a54cbdedcd..f9565a53eaf2 100644 --- a/src/conf/ucm/bytcr/Makefile.am +++ b/src/conf/ucm-includes/bytcr/Makefile.am @@ -1,4 +1,4 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/bytcr +ucmdir = $(alsaconfigdir)/ucm-includes/bytcr ucm_DATA = PlatformEnableSeq.conf PlatformDisableSeq.conf EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/bytcr/PlatformDisableSeq.conf b/src/conf/ucm-includes/bytcr/PlatformDisableSeq.conf similarity index 100% rename from src/conf/ucm/bytcr/PlatformDisableSeq.conf rename to src/conf/ucm-includes/bytcr/PlatformDisableSeq.conf diff --git a/src/conf/ucm/bytcr/PlatformEnableSeq.conf b/src/conf/ucm-includes/bytcr/PlatformEnableSeq.conf similarity index 100% rename from src/conf/ucm/bytcr/PlatformEnableSeq.conf rename to src/conf/ucm-includes/bytcr/PlatformEnableSeq.conf diff --git a/src/conf/ucm/nau8824/EnableSeq.conf b/src/conf/ucm-includes/nau8824/EnableSeq.conf similarity index 100% rename from src/conf/ucm/nau8824/EnableSeq.conf rename to src/conf/ucm-includes/nau8824/EnableSeq.conf diff --git a/src/conf/ucm/nau8824/HeadPhones.conf b/src/conf/ucm-includes/nau8824/HeadPhones.conf similarity index 100% rename from src/conf/ucm/nau8824/HeadPhones.conf rename to src/conf/ucm-includes/nau8824/HeadPhones.conf diff --git a/src/conf/ucm/nau8824/HeadsetMic.conf b/src/conf/ucm-includes/nau8824/HeadsetMic.conf similarity index 100% rename from src/conf/ucm/nau8824/HeadsetMic.conf rename to src/conf/ucm-includes/nau8824/HeadsetMic.conf diff --git a/src/conf/ucm/nau8824/InternalMic.conf b/src/conf/ucm-includes/nau8824/InternalMic.conf similarity index 100% rename from src/conf/ucm/nau8824/InternalMic.conf rename to src/conf/ucm-includes/nau8824/InternalMic.conf diff --git a/src/conf/ucm/nau8824/Makefile.am b/src/conf/ucm-includes/nau8824/Makefile.am similarity index 78% rename from src/conf/ucm/nau8824/Makefile.am rename to src/conf/ucm-includes/nau8824/Makefile.am index b39b8bb4c934..aa65f4bd9b53 100644 --- a/src/conf/ucm/nau8824/Makefile.am +++ b/src/conf/ucm-includes/nau8824/Makefile.am @@ -1,5 +1,5 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/nau8824 +ucmdir = $(alsaconfigdir)/ucm-includes/nau8824 ucm_DATA = EnableSeq.conf HeadPhones.conf HeadsetMic.conf InternalMic.conf \ MonoSpeaker.conf Speaker.conf EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/nau8824/MonoSpeaker.conf b/src/conf/ucm-includes/nau8824/MonoSpeaker.conf similarity index 100% rename from src/conf/ucm/nau8824/MonoSpeaker.conf rename to src/conf/ucm-includes/nau8824/MonoSpeaker.conf diff --git a/src/conf/ucm/nau8824/Speaker.conf b/src/conf/ucm-includes/nau8824/Speaker.conf similarity index 100% rename from src/conf/ucm/nau8824/Speaker.conf rename to src/conf/ucm-includes/nau8824/Speaker.conf diff --git a/src/conf/ucm/rt5640/DigitalMics.conf b/src/conf/ucm-includes/rt5640/DigitalMics.conf similarity index 100% rename from src/conf/ucm/rt5640/DigitalMics.conf rename to src/conf/ucm-includes/rt5640/DigitalMics.conf diff --git a/src/conf/ucm/rt5640/EnableSeq.conf b/src/conf/ucm-includes/rt5640/EnableSeq.conf similarity index 100% rename from src/conf/ucm/rt5640/EnableSeq.conf rename to src/conf/ucm-includes/rt5640/EnableSeq.conf diff --git a/src/conf/ucm/rt5640/HeadPhones.conf b/src/conf/ucm-includes/rt5640/HeadPhones.conf similarity index 100% rename from src/conf/ucm/rt5640/HeadPhones.conf rename to src/conf/ucm-includes/rt5640/HeadPhones.conf diff --git a/src/conf/ucm/rt5640/HeadsetMic.conf b/src/conf/ucm-includes/rt5640/HeadsetMic.conf similarity index 100% rename from src/conf/ucm/rt5640/HeadsetMic.conf rename to src/conf/ucm-includes/rt5640/HeadsetMic.conf diff --git a/src/conf/ucm/rt5640/IN1-InternalMic.conf b/src/conf/ucm-includes/rt5640/IN1-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5640/IN1-InternalMic.conf rename to src/conf/ucm-includes/rt5640/IN1-InternalMic.conf diff --git a/src/conf/ucm/rt5640/IN3-InternalMic.conf b/src/conf/ucm-includes/rt5640/IN3-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5640/IN3-InternalMic.conf rename to src/conf/ucm-includes/rt5640/IN3-InternalMic.conf diff --git a/src/conf/ucm/rt5640/Makefile.am b/src/conf/ucm-includes/rt5640/Makefile.am similarity index 81% rename from src/conf/ucm/rt5640/Makefile.am rename to src/conf/ucm-includes/rt5640/Makefile.am index a6fbdc572529..2cbea4da4543 100644 --- a/src/conf/ucm/rt5640/Makefile.am +++ b/src/conf/ucm-includes/rt5640/Makefile.am @@ -1,5 +1,5 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/rt5640 +ucmdir = $(alsaconfigdir)/ucm-includes/rt5640 ucm_DATA = DigitalMics.conf EnableSeq.conf HeadPhones.conf HeadsetMic.conf \ IN1-InternalMic.conf IN3-InternalMic.conf MonoSpeaker.conf Speaker.conf EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/rt5640/MonoSpeaker.conf b/src/conf/ucm-includes/rt5640/MonoSpeaker.conf similarity index 100% rename from src/conf/ucm/rt5640/MonoSpeaker.conf rename to src/conf/ucm-includes/rt5640/MonoSpeaker.conf diff --git a/src/conf/ucm/rt5640/Speaker.conf b/src/conf/ucm-includes/rt5640/Speaker.conf similarity index 100% rename from src/conf/ucm/rt5640/Speaker.conf rename to src/conf/ucm-includes/rt5640/Speaker.conf diff --git a/src/conf/ucm/rt5651/EnableSeq.conf b/src/conf/ucm-includes/rt5651/EnableSeq.conf similarity index 100% rename from src/conf/ucm/rt5651/EnableSeq.conf rename to src/conf/ucm-includes/rt5651/EnableSeq.conf diff --git a/src/conf/ucm/rt5651/HeadPhones-swapped.conf b/src/conf/ucm-includes/rt5651/HeadPhones-swapped.conf similarity index 100% rename from src/conf/ucm/rt5651/HeadPhones-swapped.conf rename to src/conf/ucm-includes/rt5651/HeadPhones-swapped.conf diff --git a/src/conf/ucm/rt5651/HeadPhones.conf b/src/conf/ucm-includes/rt5651/HeadPhones.conf similarity index 100% rename from src/conf/ucm/rt5651/HeadPhones.conf rename to src/conf/ucm-includes/rt5651/HeadPhones.conf diff --git a/src/conf/ucm/rt5651/IN1-InternalMic.conf b/src/conf/ucm-includes/rt5651/IN1-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5651/IN1-InternalMic.conf rename to src/conf/ucm-includes/rt5651/IN1-InternalMic.conf diff --git a/src/conf/ucm/rt5651/IN12-InternalMic.conf b/src/conf/ucm-includes/rt5651/IN12-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5651/IN12-InternalMic.conf rename to src/conf/ucm-includes/rt5651/IN12-InternalMic.conf diff --git a/src/conf/ucm/rt5651/IN2-InternalMic.conf b/src/conf/ucm-includes/rt5651/IN2-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5651/IN2-InternalMic.conf rename to src/conf/ucm-includes/rt5651/IN2-InternalMic.conf diff --git a/src/conf/ucm/rt5651/IN3-HeadsetMic.conf b/src/conf/ucm-includes/rt5651/IN3-HeadsetMic.conf similarity index 100% rename from src/conf/ucm/rt5651/IN3-HeadsetMic.conf rename to src/conf/ucm-includes/rt5651/IN3-HeadsetMic.conf diff --git a/src/conf/ucm/rt5651/Makefile.am b/src/conf/ucm-includes/rt5651/Makefile.am similarity index 84% rename from src/conf/ucm/rt5651/Makefile.am rename to src/conf/ucm-includes/rt5651/Makefile.am index acc8ea491043..f513b8c4d6f6 100644 --- a/src/conf/ucm/rt5651/Makefile.am +++ b/src/conf/ucm-includes/rt5651/Makefile.am @@ -1,5 +1,5 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/rt5651 +ucmdir = $(alsaconfigdir)/ucm-includes/rt5651 ucm_DATA = EnableSeq.conf HeadPhones.conf HeadPhones-swapped.conf \ IN1-InternalMic.conf IN2-InternalMic.conf IN12-InternalMic.conf \ IN3-HeadsetMic.conf Speaker.conf MonoSpeaker.conf diff --git a/src/conf/ucm/rt5651/MonoSpeaker.conf b/src/conf/ucm-includes/rt5651/MonoSpeaker.conf similarity index 100% rename from src/conf/ucm/rt5651/MonoSpeaker.conf rename to src/conf/ucm-includes/rt5651/MonoSpeaker.conf diff --git a/src/conf/ucm/rt5651/Speaker.conf b/src/conf/ucm-includes/rt5651/Speaker.conf similarity index 100% rename from src/conf/ucm/rt5651/Speaker.conf rename to src/conf/ucm-includes/rt5651/Speaker.conf diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 7cdc6176584a..2c98eac0b745 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -2,7 +2,6 @@ SUBDIRS=\ ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN \ broadwell-rt286 \ broxton-rt298 \ -bytcr \ bytcr-rt5640 \ bytcr-rt5640-mono-spk-dmic1-mic \ bytcr-rt5640-mono-spk-in1-mic \ @@ -29,13 +28,10 @@ gpd-win-pocket-rt5645 \ HDAudio-Gigabyte-ALC1220DualCodecs \ HDAudio-Lenovo-DualCodecs \ LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216 \ -nau8824 \ PandaBoard \ PandaBoardES \ PAZ00 \ PIPO-W2S-Defaultstring-CherryTrailCR \ -rt5640 \ -rt5651 \ SDP4430 \ skylake-rt286 \ TECLAST-X80Pro-Defaultstring-CherryTrailCR \ diff --git a/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf b/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf index 799163dbe481..f9c64ea16bd5 100644 --- a/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf +++ b/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf @@ -1,4 +1,4 @@ -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf index 938b359290d4..15b521be02e5 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf @@ -1,6 +1,6 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf index f36b245e5092..f9aa2cb2d2b1 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf @@ -1,6 +1,6 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf index 1a08ededc40c..0bec7f056de1 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf @@ -1,6 +1,6 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf index 4587b2e1294e..7c8388a0e153 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf @@ -1,6 +1,6 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf index decd396047aa..f074576b053b 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf @@ -1,6 +1,6 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf index d243bf823511..b3c2d03a301c 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf @@ -1,6 +1,6 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640/HiFi.conf b/src/conf/ucm/bytcr-rt5640/HiFi.conf index c833fe74845b..97823e3684d2 100644 --- a/src/conf/ucm/bytcr-rt5640/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640/HiFi.conf @@ -1,6 +1,6 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf index dd5b35fc02cb..391e977ddbd0 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf @@ -1,6 +1,6 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf index 777ed7836ee5..f0b80194e39c 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf @@ -1,6 +1,6 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf index 9f56c077b65a..049b2ebec9ad 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf @@ -1,6 +1,6 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf index e2a579ab2261..63570fb45fbb 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf @@ -1,6 +1,6 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf index ae52e7196110..85516cd23856 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf @@ -1,6 +1,6 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf index a73ff5f86b8f..1767906ffd99 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf @@ -1,6 +1,6 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651/HiFi.conf b/src/conf/ucm/bytcr-rt5651/HiFi.conf index 8733403cf28d..f254eac979cb 100644 --- a/src/conf/ucm/bytcr-rt5651/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651/HiFi.conf @@ -1,6 +1,6 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/chtnau8824/HiFi.conf b/src/conf/ucm/chtnau8824/HiFi.conf index dc52800e5eaa..4bff322b33ee 100644 --- a/src/conf/ucm/chtnau8824/HiFi.conf +++ b/src/conf/ucm/chtnau8824/HiFi.conf @@ -1,4 +1,4 @@ -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { diff --git a/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf b/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf index 799163dbe481..f9c64ea16bd5 100644 --- a/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf +++ b/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf @@ -1,4 +1,4 @@ -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { -- 2.19.1

3 3

[alsa-devel] [PATCH v2 01/13] conf/ucm: bytcr-rt5645: Use the generic bytcr/PlatformEnableSeq.conf
by Hans de Goede 28 Nov '18

28 Nov '18

Use the generic Intel SSP bytcr/PlatformEnableSeq.conf file, it is identical to all the cset statements this commit removes. Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- src/conf/ucm/chtrt5645/HiFi.conf | 116 +------------------------------ 1 file changed, 1 insertion(+), 115 deletions(-) diff --git a/src/conf/ucm/chtrt5645/HiFi.conf b/src/conf/ucm/chtrt5645/HiFi.conf index e81866cf..d993f6ae 100644 --- a/src/conf/ucm/chtrt5645/HiFi.conf +++ b/src/conf/ucm/chtrt5645/HiFi.conf @@ -11,121 +11,7 @@ SectionVerb { EnableSequence [ cdev "hw:chtrt5645" - # media mixer settings - # compress - cset "name='media0_in Gain 0 Switch' on" - cset "name='media0_in Gain 0 Volume' 0" - - # normal - cset "name='media1_in Gain 0 Switch' on" - cset "name='media1_in Gain 0 Volume' 0" - # swm loopback - cset "name='media2_in Gain 0 Switch' off" - cset "name='media2_in Gain 0 Volume' 0%" - # deep buffer - cset "name='media3_in Gain 0 Switch' on" - cset "name='media3_in Gain 0 Volume' 0" - - cset "name='media0_out mix 0 media0_in Switch' on" - cset "name='media0_out mix 0 media1_in Switch' on" - cset "name='media0_out mix 0 media2_in Switch' off" - cset "name='media0_out mix 0 media3_in Switch' on" - - cset "name='media1_out mix 0 media0_in Switch' off" - cset "name='media1_out mix 0 media1_in Switch' off" - cset "name='media1_out mix 0 media2_in Switch' off" - cset "name='media1_out mix 0 media3_in Switch' off" - - cset "name='pcm0_in Gain 0 Switch' on" - cset "name='pcm0_in Gain 0 Volume' 0" - - cset "name='pcm1_in Gain 0 Switch' off" - cset "name='pcm1_in Gain 0 Volume' 0%" - - # codec0_out settings (used if ssp2 is connected to aif1) - cset "name='codec_out0 mix 0 codec_in0 Switch' off" - cset "name='codec_out0 mix 0 codec_in1 Switch' off" - cset "name='codec_out0 mix 0 media_loop1_in Switch' off" - cset "name='codec_out0 mix 0 media_loop2_in Switch' off" - cset "name='codec_out0 mix 0 pcm0_in Switch' on" - cset "name='codec_out0 mix 0 pcm1_in Switch' off" - cset "name='codec_out0 mix 0 sprot_loop_in Switch' off" - cset "name='codec_out0 Gain 0 Switch' on" - cset "name='codec_out0 Gain 0 Volume' 0" - - # modem_out settings (used if ssp0 is connected to aif2) - cset "name='modem_out mix 0 codec_in0 Switch' off" - cset "name='modem_out mix 0 codec_in1 Switch' off" - cset "name='modem_out mix 0 media_loop1_in Switch' off" - cset "name='modem_out mix 0 media_loop2_in Switch' off" - cset "name='modem_out mix 0 pcm0_in Switch' on" - cset "name='modem_out mix 0 pcm1_in Switch' off" - cset "name='modem_out mix 0 sprot_loop_in Switch' off" - cset "name='modem_out Gain 0 Switch' on" - cset "name='modem_out Gain 0 Volume' 0" - - # input settings - # pcm1_out settings - - # input used when SSP2 is connected - cset "name='codec_in0 Gain 0 Switch' on" - cset "name='codec_in0 Gain 0 Volume' 0" - - # input used when SSP0 is connected - cset "name='modem_in Gain 0 Switch' on" - cset "name='modem_in Gain 0 Volume' 0" - - cset "name='pcm1_out mix 0 codec_in0 Switch' on" - cset "name='pcm1_out mix 0 modem_in Switch' on" - cset "name='pcm1_out mix 0 codec_in1 Switch' off" - cset "name='pcm1_out mix 0 media_loop1_in Switch' off" - cset "name='pcm1_out mix 0 media_loop2_in Switch' off" - cset "name='pcm1_out mix 0 pcm0_in Switch' off" - cset "name='pcm1_out mix 0 pcm1_in Switch' off" - cset "name='pcm1_out mix 0 sprot_loop_in Switch' off" - - cset "name='pcm1_out Gain 0 Switch' on" - cset "name='pcm1_out Gain 0 Volume' 0" - - # disable codec_out1 - cset "name='codec_out1 mix 0 codec_in0 Switch' off" - cset "name='codec_out1 mix 0 codec_in1 Switch' off" - cset "name='codec_out1 mix 0 media_loop1_in Switch' off" - cset "name='codec_out1 mix 0 media_loop2_in Switch' off" - cset "name='codec_out1 mix 0 pcm0_in Switch' off" - cset "name='codec_out1 mix 0 pcm1_in Switch' off" - cset "name='codec_out1 mix 0 sprot_loop_in Switch' off" - cset "name='codec_out1 Gain 0 Switch' off" - cset "name='codec_out1 Gain 0 Volume' 0%" - - # disable codec_in1 - cset "name='codec_in1 Gain 0 Switch' off" - cset "name='codec_in1 Gain 0 Volume' 0%" - - # disable all loops - cset "name='media_loop1_out mix 0 codec_in0 Switch' off" - cset "name='media_loop1_out mix 0 codec_in1 Switch' off" - cset "name='media_loop1_out mix 0 media_loop1_in Switch' off" - cset "name='media_loop1_out mix 0 media_loop2_in Switch' off" - cset "name='media_loop1_out mix 0 pcm0_in Switch' off" - cset "name='media_loop1_out mix 0 pcm1_in Switch' off" - cset "name='media_loop1_out mix 0 sprot_loop_in Switch' off" - - cset "name='media_loop2_out mix 0 codec_in0 Switch' off" - cset "name='media_loop2_out mix 0 codec_in1 Switch' off" - cset "name='media_loop2_out mix 0 media_loop1_in Switch' off" - cset "name='media_loop2_out mix 0 media_loop2_in Switch' off" - cset "name='media_loop2_out mix 0 pcm0_in Switch' off" - cset "name='media_loop2_out mix 0 pcm1_in Switch' off" - cset "name='media_loop2_out mix 0 sprot_loop_in Switch' off" - - cset "name='sprot_loop_out mix 0 codec_in0 Switch' off" - cset "name='sprot_loop_out mix 0 codec_in1 Switch' off" - cset "name='sprot_loop_out mix 0 media_loop1_in Switch' off" - cset "name='sprot_loop_out mix 0 media_loop2_in Switch' off" - cset "name='sprot_loop_out mix 0 pcm0_in Switch' off" - cset "name='sprot_loop_out mix 0 pcm1_in Switch' off" - cset "name='sprot_loop_out mix 0 sprot_loop_in Switch' off" + <bytcr/PlatformEnableSeq.conf> # Output Configuration cset "name='DAC1 L Mux' IF1 DAC" -- 2.18.0

6 30

[alsa-devel] [PATCH 0/5] max9867 driver update
by Ladislav Michl 28 Nov '18

28 Nov '18

Hi Mark, this is yet another round of driver update. Tested on AT91SAM9G20 based board with codec in master mode. All implemeted controls do work now. Ladislav Michl (5): ASoC: max9867: Fix whitespace ASoC: max9867: Fix power management ASoC: max9867: Calculate LRCLK divider ASoC: max9867: Fix signal paths ASoC: max9867: Add copyright and module author sound/soc/codecs/max9867.c | 479 ++++++++++++++++++------------------- sound/soc/codecs/max9867.h | 41 ++-- 2 files changed, 247 insertions(+), 273 deletions(-) -- 2.19.1

2 9

[alsa-devel] [PATCH v7 0/3] Huawei laptops
by Ayman Bagabas 28 Nov '18

28 Nov '18

This patch set is based on new audio LED triggers in branch topic/leds-trigger from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Changes from v6: * Use audio LED triggers patch set * Use KEY_CONFIG (XF86Tools) instead of KEY_PROG1. In Windows, the key is used to launch Huawei PC manager. XF86Tools is used by default on most desktop environments i.e. Gnome. Ayman Bagabas (3): ALSA: hda: fix front speakers on Huawei MBXP. x86: add support for Huawei WMI hotkeys. ALSA: hda: add support for Huawei WMI micmute LED drivers/platform/x86/Kconfig | 15 ++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/huawei-wmi.c | 247 ++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 22 +++ 4 files changed, 285 insertions(+) create mode 100644 drivers/platform/x86/huawei-wmi.c -- 2.19.1

4 9

[alsa-devel] [PATCH] ALSA: usb-audio: Add vendor and product name for Dell WD19 Dock
by Hui Wang 28 Nov '18

28 Nov '18

Like the Dell WD15 Dock, the WD19 Dock (0bda:402e) doens't provide useful string for the vendor and product names too, and WD19 has a headset Jack and a Speaker out Jack. In order to support UCM for this Dock in the alsa-lib, here we add profile_name too. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/usb/quirks-table.h | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 849953e5775c..4c10015cc0ca 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3382,5 +3382,15 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = QUIRK_NO_INTERFACE } }, +/* Dell WD19 Dock */ +{ + USB_DEVICE(0x0bda, 0x402e), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Dell", + .product_name = "WD19 Dock", + .profile_name = "Dell-WD19-Dock", + .ifnum = QUIRK_NO_INTERFACE + } +}, #undef USB_DEVICE_VENDOR_SPEC -- 2.17.1

2 2

[alsa-devel] [sound:for-linus 18/18] drivers/net//ethernet/via/via-rhine.c:1118:2: error: implicit declaration of function 'pci_iounmap'; did you mean 'pcim_iounmap'?
by kbuild test robot 27 Nov '18

27 Nov '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: 22449df3017d056abde1a1366abe6bb6a3e87751 commit: 22449df3017d056abde1a1366abe6bb6a3e87751 [18/18] sh: Always enable CONFIG_GENERIC_PCI_IOMAP config: sh-allyesconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 22449df3017d056abde1a1366abe6bb6a3e87751 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): drivers/net//ethernet/via/via-rhine.c: In function 'rhine_init_one_pci': >> drivers/net//ethernet/via/via-rhine.c:1118:2: error: implicit declaration of function 'pci_iounmap'; did you mean 'pcim_iounmap'? [-Werror=implicit-function-declaration] pci_iounmap(pdev, ioaddr); ^~~~~~~~~~~ pcim_iounmap drivers/net//ethernet/via/via-rhine.c: In function 'rhine_remove_one_platform': drivers/net//ethernet/via/via-rhine.c:2467:24: warning: unused variable 'rp' [-Wunused-variable] struct rhine_private *rp = netdev_priv(dev); ^~ cc1: some warnings being treated as errors vim +1118 drivers/net//ethernet/via/via-rhine.c 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1060 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1061 rc = pci_enable_device(pdev); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1062 if (rc) 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1063 goto err_out; 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1064 ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1065 if (pdev->revision < VTunknown0) { 5b579e212 drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-05-03 1066 quirks |= rqRhineI; ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1067 } else if (pdev->revision >= VT6102) { 5b579e212 drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-05-03 1068 quirks |= rqWOL | rqForceReset; ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1069 if (pdev->revision < VT6105) { ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1070 quirks |= rqStatusWBRace; ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1071 } else { ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1072 quirks |= rqIntPHY; ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1073 if (pdev->revision >= VT6105_B0) ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1074 quirks |= rq6patterns; ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1075 if (pdev->revision >= VT6105M) ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1076 quirks |= rqMgmt; ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1077 } ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1078 } ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1079 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1080 /* sanity check */ 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1081 if ((pci_resource_len(pdev, 0) < io_size) || 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1082 (pci_resource_len(pdev, 1) < io_size)) { 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1083 rc = -EIO; 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1084 dev_err(hwdev, "Insufficient PCI resources, aborting\n"); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1085 goto err_out_pci_disable; 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1086 } 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1087 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1088 pioaddr = pci_resource_start(pdev, 0); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1089 memaddr = pci_resource_start(pdev, 1); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1090 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1091 pci_set_master(pdev); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1092 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1093 rc = pci_request_regions(pdev, DRV_NAME); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1094 if (rc) 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1095 goto err_out_pci_disable; 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1096 5b579e212 drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-05-03 1097 ioaddr = pci_iomap(pdev, (quirks & rqNeedEnMMIO ? 1 : 0), io_size); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1098 if (!ioaddr) { 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1099 rc = -EIO; 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1100 dev_err(hwdev, 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1101 "ioremap failed for device %s, region 0x%X @ 0x%lX\n", 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1102 dev_name(hwdev), io_size, memaddr); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1103 goto err_out_free_res; 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1104 } 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1105 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1106 enable_mmio(pioaddr, quirks); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1107 5b579e212 drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-05-03 1108 rc = verify_mmio(hwdev, pioaddr, ioaddr, quirks); 5b579e212 drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-05-03 1109 if (rc) 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1110 goto err_out_unmap; 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1111 ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1112 rc = rhine_init_one_common(&pdev->dev, quirks, 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1113 pioaddr, ioaddr, pdev->irq); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1114 if (!rc) 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1115 return 0; 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1116 ^1da177e4 drivers/net/via-rhine.c Linus Torvalds 2005-04-16 1117 err_out_unmap: ^1da177e4 drivers/net/via-rhine.c Linus Torvalds 2005-04-16 @1118 pci_iounmap(pdev, ioaddr); ^1da177e4 drivers/net/via-rhine.c Linus Torvalds 2005-04-16 1119 err_out_free_res: ^1da177e4 drivers/net/via-rhine.c Linus Torvalds 2005-04-16 1120 pci_release_regions(pdev); ae996154f drivers/net/ethernet/via/via-rhine.c Roger Luethi 2014-03-18 1121 err_out_pci_disable: ae996154f drivers/net/ethernet/via/via-rhine.c Roger Luethi 2014-03-18 1122 pci_disable_device(pdev); ^1da177e4 drivers/net/via-rhine.c Linus Torvalds 2005-04-16 1123 err_out: ^1da177e4 drivers/net/via-rhine.c Linus Torvalds 2005-04-16 1124 return rc; ^1da177e4 drivers/net/via-rhine.c Linus Torvalds 2005-04-16 1125 } ^1da177e4 drivers/net/via-rhine.c Linus Torvalds 2005-04-16 1126 :::::: The code at line 1118 was first introduced by commit :::::: 1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 Linux-2.6.12-rc2 :::::: TO: Linus Torvalds <torvalds(a)ppc970.osdl.org> :::::: CC: Linus Torvalds <torvalds(a)ppc970.osdl.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: max98373: Added max98373_reset for stable amp reset
by Ryan Lee 27 Nov '18

27 Nov '18

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes : Created max98373_reset function to minimize code duplication. Changed regmap_write to regmap_update_bits. Other bits except LSB need to be masked. Added reset verification step to make sure software reset is completed well. Software reset is done in 10ms in normal case. Revision ID is available when the amp is in the idle state which means software reset is completed. Software reset will be performed maximum 3 times to avoid amp reset failure. Generally it is done in the first trial. sleep time after software reset is increased + 30ms for every retrial. Maximum possible msleep time is 100 ms (initial 10 ms + 30 ms * 3 times). sound/soc/codecs/max98373.c | 41 +++++++++++++++++++++++++++++++++++------ 1 file changed, 35 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index a09d013..55af7f02 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -724,14 +724,45 @@ static struct snd_soc_dai_driver max98373_dai[] = { } }; +static void max98373_reset(struct max98373_priv *max98373, struct device *dev) +{ + int ret, reg, count, delay; + + count = 0; + while (true) { + /* Software Reset */ + ret = regmap_update_bits(max98373->regmap, + MAX98373_R2000_SW_RESET, + MAX98373_SOFT_RESET, + MAX98373_SOFT_RESET); + if (ret) + dev_err(dev, "Reset command failed. (ret:%d)\n", ret); + + delay = 10000 + (count * 30000); + usleep_range(delay, delay + 1000); + + /* Software Reset Verification */ + ret = regmap_read(max98373->regmap, + MAX98373_R21FF_REV_ID, &reg); + if (!ret) { + dev_info(dev, "Reset completed (retry:%d)\n", count); + break; + } + + if (++count > 3) { + dev_err(dev, "Reset failed. (ret:%d)\n", ret); + break; + } + usleep_range(10000, 11000); + } +} + static int max98373_probe(struct snd_soc_component *component) { struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); /* Software Reset */ - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, component->dev); /* IV default slot configuration */ regmap_write(max98373->regmap, @@ -818,9 +849,7 @@ static int max98373_resume(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, dev); regcache_cache_only(max98373->regmap, false); regcache_sync(max98373->regmap); return 0; -- 2.7.4

3 6

[alsa-devel] [PATCH] slimbus: ngd: fix spelling mistake "exeeds" -> "exceeds"
by Colin King 27 Nov '18

27 Nov '18

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_err message, fix this. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index ccf33217236c..71f094c9ec68 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -787,7 +787,7 @@ static int qcom_slim_ngd_xfer_msg(struct slim_controller *sctrl, if (txn->msg->num_bytes > SLIM_MSGQ_BUF_LEN || txn->rl > SLIM_MSGQ_BUF_LEN) { - dev_err(ctrl->dev, "msg exeeds HW limit\n"); + dev_err(ctrl->dev, "msg exceeds HW limit\n"); return -EINVAL; } -- 2.19.1

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v4.20
by Mark Brown 27 Nov '18

27 Nov '18

The following changes since commit 6be0f96d799f487f05eb412d362d5a1747d665c2: ASoC: stm32: sai: fix master clock naming (2018-10-23 10:51:45 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.20-rc4 for you to fetch changes up to ffdcc3638c58d55a6fa68b6e5dfd4fb4109652eb: ASoC: omap-dmic: Add pm_qos handling to avoid overruns with CPU_IDLE (2018-11-23 14:05:30 +0000) ---------------------------------------------------------------- ASoC: Fixes for v4.20 Lots of fixes here, the majority of which are driver specific but there's a couple of core things and one notable driver specific one: - A core fix for a DAPM regression introduced during the component refactoring, we'd lost the code that forced a reevaluation of the DAPM graph after probe (which we suppress during init to save lots of recalcuation) and have now restored it. - A core fix for error handling using the newly added for_each_rtd_codec_dai_rollback() macro. - A fix for the names of widgets in the newly introduced pcm3060 driver, merged as a fix so we don't have a release with legacy names. ---------------------------------------------------------------- Andreas Dannenberg (1): ASoC: pcm186x: Fix device reset-registers trigger value Clément Péron (1): MAINTAINERS: add ASoC maintainers for sound dt-bindings Hans de Goede (1): ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0 Katsuhiro Suzuki (1): ASoC: rockchip: add missing slave_config setting for I2S Keyon Jie (1): ASoC: acpi: fix: continue searching when machine is ignored Kirill Marinushkin (1): ASoC: pcm3060: Rename output widgets Kuninori Morimoto (1): ASoC: rsnd: fixup clock start checker Peter Ujfalusi (4): ASoC: omap-abe-twl6040: Fix missing audio card caused by deferred probing ASoC: omap-mcbsp: Fix latency value calculation for pm_qos ASoC: omap-mcpdm: Add pm_qos handling to avoid under/overruns with CPU_IDLE ASoC: omap-dmic: Add pm_qos handling to avoid overruns with CPU_IDLE Pierre-Louis Bossart (3): ASoC: fix oops w/ for_each_rtd_codec_dai_rollback() macro ASoC: Intel: Power down links before turning off display audio power ASoC: Intel: Skylake: fix Kconfigs, make HDaudio codec optional Richard Fitzgerald (1): ASoC: wm_adsp: Fix dma-unsafe read of scratch registers Rohit kumar (2): ASoC: qdsp6: q6afe: Fix wrong MI2S SD line mask ASoC: qcom: Set dai_link id to each dai_link Srinivas Kandagatla (2): ASoC: qdsp6: q6asm-dai: Only add routing once. ASoC: qdsp6: q6afe-dai: Fix the dai widgets Tzung-Bi Shih (1): ASoC: dapm: Recalculate audio map forcely when card instantiated Vasily Khoruzhick (3): ASoC: sunxi: rename SND_SUNXI_ADDA_PR_REGMAP to SND_SUN8I_ADDA_PR_REGMAP ASoC: sun8i-codec: fix crash on module removal ASoC: sun8i-codec: add missing route for ADC kbuild test robot (1): ASoC: stm32: sai: fix noderef.cocci warnings MAINTAINERS | 1 + include/sound/soc.h | 2 +- sound/soc/codecs/hdac_hdmi.c | 11 +- sound/soc/codecs/pcm186x.h | 2 +- sound/soc/codecs/pcm3060.c | 12 +- sound/soc/codecs/wm_adsp.c | 37 ++--- sound/soc/intel/Kconfig | 26 +++- sound/soc/intel/boards/Kconfig | 24 ++-- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 32 ++++- sound/soc/intel/skylake/skl.c | 32 +++-- sound/soc/omap/omap-abe-twl6040.c | 67 ++++----- sound/soc/omap/omap-dmic.c | 9 ++ sound/soc/omap/omap-mcbsp.c | 6 +- sound/soc/omap/omap-mcpdm.c | 43 +++++- sound/soc/qcom/common.c | 9 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 208 +++++++++++++-------------- sound/soc/qcom/qdsp6/q6afe.c | 16 +-- sound/soc/qcom/qdsp6/q6asm-dai.c | 33 ----- sound/soc/qcom/qdsp6/q6routing.c | 19 +++ sound/soc/rockchip/rockchip_pcm.c | 1 + sound/soc/sh/rcar/ssi.c | 2 +- sound/soc/soc-acpi.c | 10 +- sound/soc/soc-core.c | 1 + sound/soc/stm/stm32_sai_sub.c | 2 +- sound/soc/sunxi/Kconfig | 2 +- sound/soc/sunxi/sun8i-codec.c | 12 +- 26 files changed, 360 insertions(+), 259 deletions(-)

2 1

[alsa-devel] ALC300 enable
by Kailang 27 Nov '18

27 Nov '18

Hi Takashi, Attach patch will enable new codec ALC300. Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Add disable auto-mute quirk for HP Spectre x360 laptop
by Girija Kumar Kasinadhuni 27 Nov '18

27 Nov '18

This device makes a loud buzzing sound when a headphone is inserted while playing audio at full volume through the speaker. Signed-off-by: Girija Kumar Kasinadhuni <gkumar(a)neverware.com> --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fa61674..eeff2ff 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5495,6 +5495,7 @@ enum { ALC255_FIXUP_DELL_HEADSET_MIC, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, + ALC295_FIXUP_HP_DISABLE_AUTOMUTE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6362,6 +6363,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC295_FIXUP_HP_DISABLE_AUTOMUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_auto_mute_via_amp, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6486,6 +6491,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre X360 ", ALC295_FIXUP_HP_DISABLE_AUTOMUTE), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.7.4

2 5

Re: [alsa-devel] CS8409 Macbook Pro 2016 2017
by Takashi Iwai 27 Nov '18

27 Nov '18

Please don't drop Cc to ML. Also avoid top-posting. On Sat, 04 Aug 2018 19:02:53 +0200, David Ulricht wrote: > > How can I do that ? > I tried verifying if /proc/asound/card0/codec#0 changes when i put > handphone jack in and out with "diff ver1 ver2" and it is exactly the same. Just do for each bit as you tried for GPIO output. In this case, the direction bit should be different (set 0), and you need to read. > > One strange thing I note is hda-jack-retask.fw contains: > [pincfg] > 0x24 0x90100080 > 0x25 0x90100082 > I have 0x90100080 for 0x24 but in /proc/asound/card0/codec#0 : > Node 0x24 [Pin Complex] wcaps 0x400101: Stereo > Pincap 0x00000010: OUT > Pin Default 0x90100110: [Fixed] Speaker at Int N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Connection: 1 > 0x02 > > Shouldn't "Pin Default 0x90100110" be 0x90100080 after applying the patch ? No, the codec config value itself won't change, the driver uses the given value only internally. Takashi > I can see that Applying patch firmware 'hda-jack-retask.fw' is before > hdaudioC0D0: autoconfig for Generic: line_outs=2 (0x24/0x25/0x0/0x0/0x0) > type:speaker > in dmesg and this worries me, maybe the pins are overriden by generic > cirrus logic ? > > > On Sat, Aug 4, 2018 at 7:26 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > On Sat, 04 Aug 2018 17:44:59 +0200, > > David Ulricht wrote: > > > > > > I used the following script running as root to verify the 8 GPIOs while > > > playing long youtube video, > > > I have running pulseaudio meanwhile and not restarting it, i think its > > okay > > > like that people on the ubuntu forums say that after using > > > hda-verb and setting a gpio active immediately sound appears, so i guess > > > there is no need to restart pulseaudio. > > > I have done the same test with headphones in. > > > No sound in both scenarios, no input no output detected (i'm watching > > > pavucontrol for input detection). > > > > This codec is fairly unique and doesn't provide the standard jack > > detection on each pin. Maybe it's via either GPIO or any other > > method. You can try to read GPIO pins (set as input) to see whether > > any of them corresponds to the jack detection, for example. > > > > > > Takashi > > > > > > > > #!/usr/bin/env python3 > > > import os,time > > > for x in range(0,256): > > > mhex=("0x%0.2X" % x) > > > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK "+mhex) > > > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIRECTION "+mhex) > > > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA "+mhex) > > > time.sleep(3) > > > > > > Meanwhile script is running I'm watching /proc/asound/card0/codec#0 and > > > IO[0-7] changes its configuration, > > > in comparison Joel Krahemann in his scripts is changing the hex right > > after > > > /dev/snd/hwC0D0 (0x01) e.g. i dont think he is modifiying the GPIOs > > > correctly. > > > > > > I found the following post: > > > https://forum.manjaro.org/t/sound-not-working-on-a-2017- > > imac-27-5k-retina/43638 > > > According to https://bugzilla.kernel.org/show_bug.cgi?id=195671 iMac27 > > has > > > a similar soundcard. > > > How can i setup with command line or any other utility: > > > """select “off” in the top menu that propose to setup the Digital > > > Surround""" > > > this is some setting in xfce4-mixer package which is not available for > > > ubuntu since 2 LTS versions behind. I tried current Manjaro and couldn't > > > find this setting in audio mixer of xfce4 either. > > > > > > On Sat, Aug 4, 2018 at 9:48 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > > > > On Sat, 04 Aug 2018 04:07:43 +0200, > > > > David Ulricht wrote: > > > > > > > > > > Hello, > > > > > > > > > > Macbook pro models containing CS8409 for sure are: > > > > > MBP131 MBP141.I own MBP 14,1 e.g. 2017 model. > > > > > > > > > > I'm attaching working sound configuration from Windows 10 registry > > (note > > > > > that the cs420x might need to be ignored or might be important i > > don't > > > > > really know). I believe what is really used is in the > > PinConfigOverride > > > > > section. > > > > > > > > Comparing between BIOS default (init_pin_cfg in alsa-info.sh output) > > > > and your override (user_pin_cfg), there aren't so many changes. > > > > All changes are pretty minor, and I guess it won't influence on the > > > > driver behavior. > > > > > > > > > Interesting thing to note is that sound in MacOS is much louder than > > > > > Bootcamp Windows, would be nice to hear how loud is on linux. > > > > > > > > This is possibly with some vendor-specific GPIO or COEF verbs. > > > > Or it's some direct I2C control. The INI file mentions it. > > > > And that's above HD-audio driver's responsibility. > > > > > > > > > I did convert PinConfigOverride HEX strings to [pincfg] format in the > > > > > attached hda-jack-retask.fw thanks to Takashi's guidance. > > > > > > > > > > I'm attaching also alsamixer ASCII. Only PCM, no Master ? > > > > > > > > It's because the codec chip has no output amplifier at all. > > > > So there can be neither output volume nor mute control on this chip. > > > > > > > > > What I have tried is execute: > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x@@ > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIRECTION 0x@@ > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x@@ > > > > > putting @@ from 0x00 to 0x50 > > > > > > > > There are eight GPIOs, so you'd need to test each bit, i.e. 0x01, > > > > 0x02, 0x04, 0x08, ... 0x80. > > > > > > > > And how is the current situation? You can't play *and* record > > > > anything from any inputs / outputs? > > > > > > > > > > > > Takashi > > > > > > > [2 <text/html; UTF-8 (quoted-printable)>] > > > > > > [2 <text/html; UTF-8 (quoted-printable)>] >

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[alsa-devel] [alsa-lib][PATCH 1/2] conf/ucm: Add a UCM profile for Dell WD19 Dock USB-audio
by Hui Wang 27 Nov '18

27 Nov '18

USB-audio device on Dell WD19 docking station provides two individual output PCM streams, one for headphone Jack and another for speaker out Jack. A UCM profile gives the proper roles for these. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- .../ucm/Dell-WD19-Dock/Dell-WD19-Dock.conf | 5 ++++ src/conf/ucm/Dell-WD19-Dock/HiFi.conf | 26 +++++++++++++++++++ src/conf/ucm/Dell-WD19-Dock/Makefile.am | 4 +++ 3 files changed, 35 insertions(+) create mode 100644 src/conf/ucm/Dell-WD19-Dock/Dell-WD19-Dock.conf create mode 100644 src/conf/ucm/Dell-WD19-Dock/HiFi.conf create mode 100644 src/conf/ucm/Dell-WD19-Dock/Makefile.am diff --git a/src/conf/ucm/Dell-WD19-Dock/Dell-WD19-Dock.conf b/src/conf/ucm/Dell-WD19-Dock/Dell-WD19-Dock.conf new file mode 100644 index 00000000..465ff550 --- /dev/null +++ b/src/conf/ucm/Dell-WD19-Dock/Dell-WD19-Dock.conf @@ -0,0 +1,5 @@ +Comment "USB-audio on Dell WD19 docking station" +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Default" +} diff --git a/src/conf/ucm/Dell-WD19-Dock/HiFi.conf b/src/conf/ucm/Dell-WD19-Dock/HiFi.conf new file mode 100644 index 00000000..e1427a79 --- /dev/null +++ b/src/conf/ucm/Dell-WD19-Dock/HiFi.conf @@ -0,0 +1,26 @@ +SectionDevice."Headphone" { + Comment "Headphone" + + Value { + PlaybackChannels "2" + PlaybackPCM "hw:Dock,0" + } +} + +SectionDevice."Speaker" { + Comment "Speaker" + + Value { + PlaybackChannels "2" + PlaybackPCM "hw:Dock,1" + } +} + +SectionDevice."Mic" { + Comment "Microphone" + + Value { + CaptureChannels "2" + CapturePCM "hw:Dock,0" + } +} diff --git a/src/conf/ucm/Dell-WD19-Dock/Makefile.am b/src/conf/ucm/Dell-WD19-Dock/Makefile.am new file mode 100644 index 00000000..6549ae1b --- /dev/null +++ b/src/conf/ucm/Dell-WD19-Dock/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/Dell-WD19-Dock +ucm_DATA = Dell-WD19-Dock.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) -- 2.17.1

1 1

[alsa-devel] [PATCH v7 3/3] ALSA: hda: add support for Huawei WMI micmute LED
by Ayman Bagabas 27 Nov '18

27 Nov '18

Some of Huawei laptops come with a LED in the micmute key. This patch enables the use of micmute LED for these devices: 1. Matebook X (19e5:3200), (19e5:3201) 2. Matebook X Pro (19e5:3204) Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.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 1326f32f4574..9766fd249bdf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5776,7 +5776,9 @@ static const struct hda_fixup alc269_fixups[] = { {0x1e, 0x411111f0}, {0x21, 0x04211020}, { } - } + }, + .chained = true, + .chain_id = ALC255_FIXUP_MIC_MUTE_LED }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, @@ -6608,6 +6610,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3200, "Huawei MBX", ALC255_FIXUP_MIC_MUTE_LED), + SND_PCI_QUIRK(0x19e5, 0x3201, "Huawei MBX", ALC255_FIXUP_MIC_MUTE_LED), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 2.19.1

1 0

[alsa-devel] [PATCH v7 1/3] ALSA: hda: fix front speakers on Huawei MBXP.
by Ayman Bagabas 27 Nov '18

27 Nov '18

This patch solves bug 200501 'Only 2 of 4 speakers playing sound.' https://bugzilla.kernel.org/show_bug.cgi?id=200501 It enables the front speakers on Huawei Matebook X Pro laptops. These laptops come with Dolby Atmos sound system and these pins configuration enables the front speakers. Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 993d34c141c2..1326f32f4574 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5490,6 +5490,7 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, + ALC256_FIXUP_HUAWEI_MBXP_PINS, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -5761,6 +5762,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x12, 0x90a60130}, + {0x13, 0x40000000}, + {0x14, 0x90170110}, + {0x18, 0x411111f0}, + {0x19, 0x04a11040}, + {0x1a, 0x411111f0}, + {0x1b, 0x90170112}, + {0x1d, 0x40759a05}, + {0x1e, 0x411111f0}, + {0x21, 0x04211020}, + { } + } + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -6591,6 +6608,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ #if 0 -- 2.19.1

1 0

[alsa-devel] [PATCH v9 0/8] Add support for audiocodec in Allwinner A64
by Chen-Yu Tsai 26 Nov '18

26 Nov '18

Hi everyone, This is v9 of the A64 audio codec series, originally started by Vasily. These are the last changes to make the A64 audio codec work correctly, and be accurately represented in the device tree and in software. Notably, the new driver changes cover regulator supplies for the simple audio amplifier, which is used in the Pinebook for the internal speakers, and the audio codec's internal headphone amp. There's also an extra patch that enables SDM fractional clock support for the audio PLL on the A64, which gives us accurate audio playback. Patch 1 & 2 add the cpvdd regulator supply to the sun50i-codec-analog dt bindings & driver. Patch 3 & 4 add a regulator supply to the simple-audio-amplifier dt bindings & driver. Patch 5 adds SDM fractional clock support for the audio PLL on the A64. Patch 6 ~ 8 add the device tree bits and enables audio on several boards. I have one extra patch for the Bananapi M64 that's not included here, as I'm waiting for an accessory to test the microphone bias. I'll send that one out once it's tested. As always, we'll take the clock and dts patches after Mark has merged the ASoC ones. And a big thanks to Vasily for having worked on this for so long. Regards ChenYu v9: - Dropped simple amplifier headphone amp device tree changes - Dropped "make enable-gpio optional for simple amplifier" - Lowered oversampling rate from 512 to 128 - Add regulator supply for audio codec CPVDD (headphone amp charge pump) - Hook up CPVDD regulator in board device trees - Add patch to enable SDM fractional clock support for audio PLL on A64 v8: - drop merged patches - add regulator property for simple amplifier, make enable-gpio optional - add regulator to speaker amp, introduce headphone amp v7: - drop 'Mic' and 'Headset Mic' from dtsi - move AIF1 Slot 0 ADC to ADC route to sun8i-codec - drop BIAS routes for Pine64 and SoPine v6: - driver patches has been merged through ASoC tree - rebase onto 4.20-rc1 - Drop 'Speaker' from routes on sopine and pine64, they don't have speaker. v5: - collect all the tags v4: - dts: don't use 'Mic' and 'Headset Mic' widgets from sun8i-codec, define our board-level widgets instead. v3: - renamed sunxi-adda-pr-regmap to sun8i-adda-pr-regmap - use ilog2() to calculate reg value for LRCK div instead of using a table v2: - Use simple-amplifier for speaker amp on Pinebook - Rename sun50i-a64-i2s to sun50i-a64-codec-i2s to preserve compatible string for other 3 I2S modules in A64 in case if there's any incompatibility with H3 Chen-Yu Tsai (3): ASoC: dt-bindings: sun50i-codec-analog: Add headphone amp regulator supply ASoC: sunxi: sun50i-codec-analog: Add support for cpvdd regulator supply clk: sunxi-ng: sun50i: a64: Use sigma-delta modulation for audio PLL Vasily Khoruzhick (5): ASoC: dt-bindings: add regulator property to simple amplifier ASoC: simple-amplifier: add VCC regulator widget arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../bindings/sound/simple-amplifier.txt | 5 ++ .../bindings/sound/sun50i-codec-analog.txt | 2 + .../boot/dts/allwinner/sun50i-a64-pine64.dts | 27 ++++++++++ .../dts/allwinner/sun50i-a64-pinebook.dts | 49 +++++++++++++++++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 26 +++++++++ .../boot/dts/allwinner/sun50i-a64-sopine.dtsi | 4 ++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 54 +++++++++++++++++++ drivers/clk/sunxi-ng/ccu-sun50i-a64.c | 37 ++++++++----- sound/soc/codecs/simple-amplifier.c | 4 ++ sound/soc/sunxi/sun50i-codec-analog.c | 2 + 10 files changed, 197 insertions(+), 13 deletions(-) -- 2.19.1

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[alsa-devel] [PATCH] ASoC: amd: Fix a NULL vs IS_ERR() check in probe
by Dan Carpenter 26 Nov '18

26 Nov '18

The platform_device_register_full() function doesn't return NULL, it returns error pointers. Fixes: 7894a7e7ea3d ("ASoC: amd: create ACP3x PCM platform device") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/amd/raven/pci-acp3x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index c28457fd9b81..facec2472b34 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -97,10 +97,10 @@ static int snd_acp3x_probe(struct pci_dev *pci, pdevinfo.size_data = sizeof(irqflags); adata->pdev = platform_device_register_full(&pdevinfo); - if (!adata->pdev) { + if (IS_ERR(adata->pdev)) { dev_err(&pci->dev, "cannot register %s device\n", pdevinfo.name); - ret = -ENODEV; + ret = PTR_ERR(adata->pdev); goto unmap_mmio; } break; -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: rsnd: tidyup for SSIU subnode
by Kuninori Morimoto 26 Nov '18

26 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit da48a6eb82ea2 ("ASoC: rsnd: add SSIU BUSIF support for Document") updated Documentation for SSIU, but 1) we want to keep old/deprecated DMA description, 2) it is missing SSIU subnode properties. This patch tidyup these Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 47dc819..4d19d2b 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -327,8 +327,13 @@ SSI subnode properties: - no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case - dma : Should contain Audio DMAC entry - dma-names : SSI case "rx" (=playback), "tx" (=capture) + Deprecated: see SSIU subnode properties SSIU case "rxu" (=playback), "txu" (=capture) +SSIU subnode properties: +- dma : Should contain Audio DMAC entry +- dma-names : "rx" (=playback), "tx" (=capture) + SRC subnode properties: - dma : Should contain Audio DMAC entry - dma-names : "rx" (=playback), "tx" (=capture) -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: soc.h: makes snd_soc_of_parse_audio_prefix() inline
by Kuninori Morimoto 26 Nov '18

26 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 3b7103562c03c ("ASoC: soc-core: add snd_soc_of_parse_node_prefix()") maked snd_soc_of_parse_audio_prefix() as #define. But it'd be better to make this a static inline rather than a #define. It helps with error messages and type safety. This patch makes it inline. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 37155f9..8ec1de8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1481,9 +1481,15 @@ void snd_soc_of_parse_node_prefix(struct device_node *np, struct snd_soc_codec_conf *codec_conf, struct device_node *of_node, const char *propname); -#define snd_soc_of_parse_audio_prefix(card, conf, node, name) \ - snd_soc_of_parse_node_prefix((card)->dev->of_node, \ - (conf), (node), (name)) +static inline +void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card, + struct snd_soc_codec_conf *codec_conf, + struct device_node *of_node, + const char *propname) +{ + snd_soc_of_parse_node_prefix(card->dev->of_node, + codec_conf, of_node, propname); +} int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); -- 2.7.4

2 1

[alsa-devel] [RFC 0/9] ASoC/ARM: Merge the davinci and omap audio directories
by Peter Ujfalusi 26 Nov '18

26 Nov '18

Hi, This RFC series is on top of the omap-mcbsp cleanup series: http://mailman.alsa-project.org/pipermail/alsa-devel/2018-November/141604.h… For Texas Instruments audio support we have two directories at the moment: davinci omap In reality TI have the following SoC lines supported by Linux: daVinci OMAP Sitara Keystone We already needed to have cross directory 'links' for DRA7x/AM5 since they use McASP (davinci directory) and sDMA (omap directory). Keystone II line of SoCs are using drivers from davinci directory (McASP and EDMA). Which is kind of fine as they are in the same place. With the new AM654 (http://www.ti.com/lit/pdf/spruid7) we will have completely new DMA and there is no place to put that as it does not belong under davinci or omap, but it makes no sense to create yet another TI related directory just for the small udma-pcm. Another issue with the current support is how the Kconfigs are written. The DAI drivers are depending on the ASoC platform drivers and it is a bit of a mess at the moment. For simple-sound-card one need to know what DMA is used and then select the DAI driver (if the Kconfig is right). I have also cleaned up the Kconfig prompts and comments a bit to be more precise. As can be seen in the diffstat I tried to do the merger with minimal code change, resisting the urge to rename the davinci-i2c driver to davinci-asp. With the merger I have created a new Kconfig structure: - Submenu for TI related settings for clarity - Only the DAI drivers are selectable and the platform drivers are selected based on the available (enabled) DMA engines - The remaining ASoC machine drivers are selecting the DAIs they need - The davinci-evm is now a single entry, combining all the various daVinci EVMS it supports and selects the needed DAIs based on the enabled boards The series also includes patches to updated the arch/arm/ files (defconfigs, source files using audio Kconfig symbols). I believe with v1 all of this can be merged via ASoC tree as the chances of conflicts are low for the touched arch/arm files. But Sekhar and Tony can correct me if I'm wrong. I think I have catched all (I hope) issues regarding to Kconfig thanks to the various autobuild machines out there kindly testing my linux-next-wip branch. I would appreciate any testing comments before I send the v1! Regards, Peter --- Peter Ujfalusi (9): ASoC: ti: Merge davinci and omap directories MAINTAINERS: Add entry for sound/soc/ti and update the OMAP audio support ARM: OMAP1: Makefile: Update for new MCBSP Kconfig option ARM: OMAP2: Update for new MCBSP Kconfig option ARM: davinci: dm365-evm: Update for the new ASoC Kcofnig options ARM: omap2plus_defconfig: Update the audio options ARM: omap1_defconfig: Do not select ASoC by default ARM: davinci_all_defconfig: Update the audio options ASoC: ti: Kconfig: Remove the deprecated options MAINTAINERS | 11 +- arch/arm/configs/davinci_all_defconfig | 5 +- arch/arm/configs/omap1_defconfig | 2 - arch/arm/configs/omap2plus_defconfig | 14 +- arch/arm/mach-davinci/board-dm365-evm.c | 4 +- arch/arm/mach-omap1/Makefile | 2 +- arch/arm/mach-omap2/Makefile | 2 +- arch/arm/mach-omap2/pdata-quirks.c | 4 +- sound/soc/Kconfig | 3 +- sound/soc/Makefile | 3 +- sound/soc/davinci/Kconfig | 106 --------- sound/soc/davinci/Makefile | 16 -- sound/soc/omap/Kconfig | 120 ---------- sound/soc/omap/Makefile | 30 --- sound/soc/ti/Kconfig | 209 ++++++++++++++++++ sound/soc/ti/Makefile | 44 ++++ sound/soc/{omap => ti}/ams-delta.c | 0 sound/soc/{davinci => ti}/davinci-evm.c | 4 +- sound/soc/{davinci => ti}/davinci-i2s.c | 0 sound/soc/{davinci => ti}/davinci-i2s.h | 0 sound/soc/{davinci => ti}/davinci-mcasp.c | 14 +- sound/soc/{davinci => ti}/davinci-mcasp.h | 0 sound/soc/{davinci => ti}/davinci-vcif.c | 0 sound/soc/{davinci => ti}/edma-pcm.c | 0 sound/soc/{davinci => ti}/edma-pcm.h | 4 +- sound/soc/{omap => ti}/n810.c | 0 sound/soc/{omap => ti}/omap-abe-twl6040.c | 0 sound/soc/{omap => ti}/omap-dmic.c | 0 sound/soc/{omap => ti}/omap-dmic.h | 0 .../omap-hdmi-audio.c => ti/omap-hdmi.c} | 0 sound/soc/{omap => ti}/omap-mcbsp-priv.h | 0 sound/soc/{omap => ti}/omap-mcbsp-st.c | 0 sound/soc/{omap => ti}/omap-mcbsp.c | 0 sound/soc/{omap => ti}/omap-mcbsp.h | 0 sound/soc/{omap => ti}/omap-mcpdm.c | 0 sound/soc/{omap => ti}/omap-mcpdm.h | 0 sound/soc/{omap => ti}/omap-twl4030.c | 0 sound/soc/{omap => ti}/omap3pandora.c | 0 sound/soc/{omap => ti}/osk5912.c | 0 sound/soc/{omap => ti}/rx51.c | 0 sound/soc/{omap => ti}/sdma-pcm.c | 0 sound/soc/{omap => ti}/sdma-pcm.h | 4 +- 42 files changed, 294 insertions(+), 307 deletions(-) delete mode 100644 sound/soc/davinci/Kconfig delete mode 100644 sound/soc/davinci/Makefile delete mode 100644 sound/soc/omap/Kconfig delete mode 100644 sound/soc/omap/Makefile create mode 100644 sound/soc/ti/Kconfig create mode 100644 sound/soc/ti/Makefile rename sound/soc/{omap => ti}/ams-delta.c (100%) rename sound/soc/{davinci => ti}/davinci-evm.c (99%) rename sound/soc/{davinci => ti}/davinci-i2s.c (100%) rename sound/soc/{davinci => ti}/davinci-i2s.h (100%) rename sound/soc/{davinci => ti}/davinci-mcasp.c (99%) rename sound/soc/{davinci => ti}/davinci-mcasp.h (100%) rename sound/soc/{davinci => ti}/davinci-vcif.c (100%) rename sound/soc/{davinci => ti}/edma-pcm.c (100%) rename sound/soc/{davinci => ti}/edma-pcm.h (91%) rename sound/soc/{omap => ti}/n810.c (100%) rename sound/soc/{omap => ti}/omap-abe-twl6040.c (100%) rename sound/soc/{omap => ti}/omap-dmic.c (100%) rename sound/soc/{omap => ti}/omap-dmic.h (100%) rename sound/soc/{omap/omap-hdmi-audio.c => ti/omap-hdmi.c} (100%) rename sound/soc/{omap => ti}/omap-mcbsp-priv.h (100%) rename sound/soc/{omap => ti}/omap-mcbsp-st.c (100%) rename sound/soc/{omap => ti}/omap-mcbsp.c (100%) rename sound/soc/{omap => ti}/omap-mcbsp.h (100%) rename sound/soc/{omap => ti}/omap-mcpdm.c (100%) rename sound/soc/{omap => ti}/omap-mcpdm.h (100%) rename sound/soc/{omap => ti}/omap-twl4030.c (100%) rename sound/soc/{omap => ti}/omap3pandora.c (100%) rename sound/soc/{omap => ti}/osk5912.c (100%) rename sound/soc/{omap => ti}/rx51.c (100%) rename sound/soc/{omap => ti}/sdma-pcm.c (100%) rename sound/soc/{omap => ti}/sdma-pcm.h (85%) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

5 14

[alsa-devel] Fwd: [BUG] The infamous MOTU Ultralite * bitcrunch issue
by Stanisław Drozd 26 Nov '18

26 Nov '18

[Originally posted in alsa-user, but that list doesn't appear to be very widely used] Hello everyone, I recently updated the firmware on my UltraLite mk4 (IF YOURS IS WORKING OK, DON'T UPGRADE or you'll repeat my mistake). Current firmware on my piece is 1.3.4+232, 07/27/18 17:46:22, BIOS is Aug-04-2017 b568a4c. After the update an issue appeared - a moment after initializing the device and playing normally, it starts sounding as if it was put through a bitcruncher effect. After a longer while the output loses gain only to disappear completely after some more seconds pass (the gain change is not smooth, it seems to go away in steps). What I have tried: * Switching internal clock to LTC on the interface - would hold the sync together for a little longer; I did NOT send the actual timecode because I had trouble finding something to generate it for me * Switching sample rates - No noticeable change in behavior * Testing with speaker-test - No change in behavior, looks like this has to happen on ALSA level or lower * Changing clock source in Cadence (a JACK helper) settings and reinitializing - No effect Possible clues I saw: * dmesg started showing this when I plug the device in: [ 533.419099] usb 1-2.3: new high-speed USB device number 8 using xhci_hcd [ 533.527638] usb 1-2.3: New USB device found, idVendor=07fd, idProduct=0005, bcdDevice= 2.00 [ 533.527646] usb 1-2.3: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 533.527650] usb 1-2.3: Product: UltraLite-mk4 [ 533.527654] usb 1-2.3: Manufacturer: MOTU [ 533.527658] usb 1-2.3: SerialNumber: 0001f2fffe008e19 [ 534.377873] usb 1-2.3: clock source 1 is not valid, cannot use <and about 50 more lines of the clock source message> Questions: * What other places can I go to to report this? * What other settings can I tweak? Thanks, Stanislaw PS: I have a basic understanding of driver work, I think I could help with creating a patch with some guidance

1 0

[alsa-devel] [PATCH] ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops
by Hui Wang 26 Nov '18

26 Nov '18

We have several Lenovo laptops with the codec alc285, when playing sound via headphone, we can hear click/pop noise in the headphone, if we let the headphone share the DAC of NID 0x2 with the speaker, the noise disappears. The Lenovo laptops here include P52, P72, X1 yoda2 and X1 carbon. I have tried to set preferred_dacs and override_conn, but neither of them worked. Thanks for Kailang, he told me to invalidate the NID 0x3 through override_wcaps. BugLink: https://bugs.launchpad.net/bugs/1805079 Cc: <stable(a)vger.kernel.org> Signed-off-by: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1118fd1bbf1a..e66da22272fd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5358,6 +5358,16 @@ static void alc274_fixup_bind_dacs(struct hda_codec *codec, spec->gen.preferred_dacs = preferred_pairs; } +/* The DAC of NID 0x3 will introduce click/pop noise on headphones, so invalidate it */ +static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + snd_hda_override_wcaps(codec, 0x03, 0); +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -5495,6 +5505,7 @@ enum { ALC255_FIXUP_DELL_HEADSET_MIC, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, + ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6362,6 +6373,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC285_FIXUP_LENOVO_HEADPHONE_NOISE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_invalidate_dacs, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7035,6 +7050,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60130}, {0x19, 0x03a11020}, {0x21, 0x0321101f}), + SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, + {0x12, 0x90a60130}, + {0x14, 0x90170110}, + {0x19, 0x04a11040}, + {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60120}, {0x14, 0x90170110}, -- 2.17.1

2 1

Re: [alsa-devel] [LKP] [ALSA] 3a6d1980fe: kernel_BUG_at_mm/page_alloc.c
by Takashi Iwai 26 Nov '18

26 Nov '18

On Mon, 26 Nov 2018 01:28:05 +0100, kernel test robot wrote: > > FYI, we noticed the following commit (built with gcc-7): > > commit: 3a6d1980fe96dbbfe3ae58db0048867f5319cdbf ("ALSA: Replace snd_malloc_pages() and snd_free_pages() with standard helpers") > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git for-next .... > [ 174.830284] kernel BUG at mm/page_alloc.c:2850! OK, this seem to be an issue about using __GFP_COMP and alloc_pages_exact(). I didn't hit this while testing locally maybe just because I used the buffer size in power-of-two casually. I dropped the patch until we sort this out. thanks, Takashi

1 0

[alsa-devel] [PATCH 00/10] ASoC: merge audio-graph-scu into audio-graph - step2
by Kuninori Morimoto 26 Nov '18

26 Nov '18

Hi Mark audio-graph card can handle normal sound card, and audio-graph-scu card can handle DPCM sound card. But, we can't use these feature in same time now. So, this is the time to merge audio-graph and audio-graph-scu. The patch-set are a little bit huge (and few patches are depend on posted OF patch. I'm now waiting...), thus, I separated these into few steps. These are step2 patches Kuninori Morimoto (10): ASoC: soc-core: add snd_soc_of_parse_node_prefix() ASoC: simple-scu-card: tidyup "prefix" parsing on Doc ASoC: simple-scu-card: tidyup "prefix" parsing ASoC: audio-graph-scu-card: tidyup "prefix" parsing on Doc ASoC: audio-graph-scu-card: tidyup "prefix" parsing ASoC: audio-graph-scu-card: tidyup "convert-rate/channels" parsing on Doc ASoC: audio-graph-scu-card: tidyup "convert-rate/channels" parsing ASoC: audio-graph-scu-card: tidyup asoc_simple_card_parse_daifmt() timing ASoC: audio-graph-card: tidyup typo SND_AUDIO_GRAPH_CARD ASoC: audio-graph-card: tidyup define position .../bindings/sound/audio-graph-scu-card.txt | 4 +- .../devicetree/bindings/sound/simple-scu-card.txt | 2 +- include/sound/soc.h | 6 +- sound/soc/generic/Kconfig | 2 +- sound/soc/generic/audio-graph-card.c | 10 +-- sound/soc/generic/audio-graph-scu-card.c | 71 ++++++++++------------ sound/soc/generic/simple-scu-card.c | 7 +++ sound/soc/soc-core.c | 11 ++-- 8 files changed, 59 insertions(+), 54 deletions(-) -- 2.7.4

2 22

[alsa-devel] [PATCH] ALSA: pcm: Fix starvation on down_write_nonblock()
by Chanho Min 25 Nov '18

25 Nov '18

Commit 67ec1072b053 (ALSA: pcm: Fix rwsem deadlock for non-atomic PCM stream) fixes deadlock for non-atomic PCM stream. But, This patch causes antother stuck. If writer is RT thread and reader is a normal thread, the reader thread will be difficult to get scheduled. It may not give chance to release read locks and writer gets stuck for a long time or forever if they are pinned to single cpu. To fix this, The writer gives reader a chance to be scheduled by using the minimum msleep() instaed of spinning. This is for concept, We may need to change the function name and comments or suggest another approach. Signed-off-by: Chanho Min <chanho.min(a)lge.com> --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 66c90f4..88d4aab 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -96,7 +96,7 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem); static inline void down_write_nonblock(struct rw_semaphore *lock) { while (!down_write_trylock(lock)) - cond_resched(); + msleep(1); } #define PCM_LOCK_DEFAULT 0 -- 2.1.4

2 2

[alsa-devel] Fwd: [PATCH v2 01/13] conf/ucm: bytcr-rt5645: Use the generic bytcr/PlatformEnableSeq.conf
by youling 257 25 Nov '18

25 Nov '18

---------- Forwarded message --------- From: youling 257 <youling257(a)gmail.com> Date: 2018年11月24日周六下午6:42 Subject: Re: [PATCH v2 01/13] conf/ucm: bytcr-rt5645: Use the generic bytcr/PlatformEnableSeq.conf To: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>, Takashi Iwai <tiwai(a)suse.de>, Hans de Goede <hdegoede(a)redhat.com>, <alsa-devel(a)vger.kernel.org> alsaucm -c chtrt5645 set _verb HiFi set _enadev Speaker ALSA lib conf.c:823:(get_char_skip_comments) Cannot access file bytcr/PlatformEnableSeq.conf ALSA lib conf.c:1877:(snd_config_load1) _toplevel_:17:18:Unexpected char ALSA lib utils.c:67:(uc_mgr_config_load) could not open configuration file /usr/share/alsa/ucm/ASUSTeKCOMPUTERINC.-X205TA-1.0-X205TA/ASUSTeKCOMPUTERINC.-X205TA-1.0-X205TA.conf ALSA lib parser.c:1427:(load_master_config) error: could not parse configuration for card ASUSTeKCOMPUTERINC.-X205TA-1.0-X205TA ALSA lib conf.c:823:(get_char_skip_comments) Cannot access file bytcr/PlatformEnableSeq.conf ALSA lib conf.c:1877:(snd_config_load1) _toplevel_:17:18:Unexpected char ALSA lib utils.c:75:(uc_mgr_config_load) could not load configuration file /usr/share/alsa/ucm/chtrt5645/HiFi.conf ALSA lib parser.c:1101:(parse_verb_file) error: failed to open verb file /usr/share/alsa/ucm/chtrt5645/HiFi.conf : -2 ALSA lib main.c:946:(snd_use_case_mgr_open) error: failed to import chtrt5645 use case configuration -22 alsaucm: error failed to open sound card chtrt5645: Invalid argument

1 0

[alsa-devel] [PATCH] ALSA: Replace snd_malloc_pages() and snd_free_pages() with standard helpers
by Takashi Iwai 24 Nov '18

24 Nov '18

snd_malloc_pages() and snd_free_pages() are merely thin wrappers of the standard page allocator / free functions. Even the arguments are compatible with some standard helpers, so there is little merit of keeping these wrappers. This patch replaces the all existing callers of snd_malloc_pages() and snd_free_pages() with the direct calls of the standard helper functions. In this version, we use a recently introduced one, alloc_pages_exact(), which suits better than the old snd_malloc_pages() implementation. Then we can avoid the waste of pages by alignment to power-of-two. Also, the __GFP_COMP flag that has been always added in the old implementation is dropped for most of callers now. The only one that may still need it is the PCM buffer allocation which can be quite large. The rest are single page allocations, so no sense to keep that flag. With these conversions, snd_malloc_pages() and snd_free_pages() are dropped altogether from the whole tree. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/memalloc.h | 4 --- sound/core/memalloc.c | 54 +++------------------------------ sound/core/pcm.c | 14 ++++----- sound/usb/usx2y/usX2Yhwdep.c | 3 +- sound/usb/usx2y/usbusx2y.c | 3 +- sound/usb/usx2y/usx2yhwdeppcm.c | 6 ++-- 6 files changed, 19 insertions(+), 65 deletions(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index af3fa577fa06..c56cb2871368 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -152,9 +152,5 @@ int snd_dma_alloc_pages_fallback(int type, struct device *dev, size_t size, struct snd_dma_buffer *dmab); void snd_dma_free_pages(struct snd_dma_buffer *dmab); -/* basic memory allocation functions */ -void *snd_malloc_pages(size_t size, gfp_t gfp_flags); -void snd_free_pages(void *ptr, size_t size); - #endif /* __SOUND_MEMALLOC_H */ diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 59a4adc286ed..d12bf811e751 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -30,53 +30,6 @@ #endif #include <sound/memalloc.h> -/* - * - * Generic memory allocators - * - */ - -/** - * snd_malloc_pages - allocate pages with the given size - * @size: the size to allocate in bytes - * @gfp_flags: the allocation conditions, GFP_XXX - * - * Allocates the physically contiguous pages with the given size. - * - * Return: The pointer of the buffer, or %NULL if no enough memory. - */ -void *snd_malloc_pages(size_t size, gfp_t gfp_flags) -{ - int pg; - - if (WARN_ON(!size)) - return NULL; - if (WARN_ON(!gfp_flags)) - return NULL; - gfp_flags |= __GFP_COMP; /* compound page lets parts be mapped */ - pg = get_order(size); - return (void *) __get_free_pages(gfp_flags, pg); -} -EXPORT_SYMBOL(snd_malloc_pages); - -/** - * snd_free_pages - release the pages - * @ptr: the buffer pointer to release - * @size: the allocated buffer size - * - * Releases the buffer allocated via snd_malloc_pages(). - */ -void snd_free_pages(void *ptr, size_t size) -{ - int pg; - - if (ptr == NULL) - return; - pg = get_order(size); - free_pages((unsigned long) ptr, pg); -} -EXPORT_SYMBOL(snd_free_pages); - /* * * Bus-specific memory allocators @@ -188,8 +141,9 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->bytes = 0; switch (type) { case SNDRV_DMA_TYPE_CONTINUOUS: - dmab->area = snd_malloc_pages(size, - (__force gfp_t)(unsigned long)device); + dmab->area = alloc_pages_exact(size, + (__force gfp_t)(unsigned long)device | + __GFP_COMP); dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA @@ -273,7 +227,7 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) { switch (dmab->dev.type) { case SNDRV_DMA_TYPE_CONTINUOUS: - snd_free_pages(dmab->area, dmab->bytes); + free_pages_exact(dmab->area, dmab->bytes); break; #ifdef CONFIG_HAS_DMA #ifdef CONFIG_GENERIC_ALLOCATOR diff --git a/sound/core/pcm.c b/sound/core/pcm.c index fdb9b92fc8d6..1c0ecb22b1f4 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1004,22 +1004,22 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, return -ENOMEM; size = PAGE_ALIGN(sizeof(struct snd_pcm_mmap_status)); - runtime->status = snd_malloc_pages(size, GFP_KERNEL); + runtime->status = alloc_pages_exact(size, GFP_KERNEL); if (runtime->status == NULL) { kfree(runtime); return -ENOMEM; } - memset((void*)runtime->status, 0, size); + memset(runtime->status, 0, size); size = PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control)); - runtime->control = snd_malloc_pages(size, GFP_KERNEL); + runtime->control = alloc_pages_exact(size, GFP_KERNEL); if (runtime->control == NULL) { - snd_free_pages((void*)runtime->status, + free_pages_exact(runtime->status, PAGE_ALIGN(sizeof(struct snd_pcm_mmap_status))); kfree(runtime); return -ENOMEM; } - memset((void*)runtime->control, 0, size); + memset(runtime->control, 0, size); init_waitqueue_head(&runtime->sleep); init_waitqueue_head(&runtime->tsleep); @@ -1045,9 +1045,9 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) runtime = substream->runtime; if (runtime->private_free != NULL) runtime->private_free(runtime); - snd_free_pages((void*)runtime->status, + free_pages_exact(runtime->status, PAGE_ALIGN(sizeof(struct snd_pcm_mmap_status))); - snd_free_pages((void*)runtime->control, + free_pages_exact(runtime->control, PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control))); kfree(runtime->hw_constraints.rules); /* Avoid concurrent access to runtime via PCM timer interface */ diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index c1dd9a7b48df..bfe1108416cf 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -75,7 +75,8 @@ static int snd_us428ctls_mmap(struct snd_hwdep * hw, struct file *filp, struct v if (!us428->us428ctls_sharedmem) { init_waitqueue_head(&us428->us428ctls_wait_queue_head); - if(!(us428->us428ctls_sharedmem = snd_malloc_pages(sizeof(struct us428ctls_sharedmem), GFP_KERNEL))) + us428->us428ctls_sharedmem = alloc_pages_exact(sizeof(struct us428ctls_sharedmem), GFP_KERNEL); + if (!us428->us428ctls_sharedmem) return -ENOMEM; memset(us428->us428ctls_sharedmem, -1, sizeof(struct us428ctls_sharedmem)); us428->us428ctls_sharedmem->CtlSnapShotLast = -2; diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index da4a5a541512..9f7bbed2c0f0 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -437,7 +437,8 @@ static void snd_usX2Y_card_private_free(struct snd_card *card) kfree(usX2Y(card)->In04Buf); usb_free_urb(usX2Y(card)->In04urb); if (usX2Y(card)->us428ctls_sharedmem) - snd_free_pages(usX2Y(card)->us428ctls_sharedmem, sizeof(*usX2Y(card)->us428ctls_sharedmem)); + free_pages_exact(usX2Y(card)->us428ctls_sharedmem, + sizeof(*usX2Y(card)->us428ctls_sharedmem)); if (usX2Y(card)->card_index >= 0 && usX2Y(card)->card_index < SNDRV_CARDS) snd_usX2Y_card_used[usX2Y(card)->card_index] = 0; } diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 4fd9276b8e50..4453c2b911d0 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -488,7 +488,9 @@ static int snd_usX2Y_usbpcm_prepare(struct snd_pcm_substream *substream) snd_printdd("snd_usX2Y_pcm_prepare(%p)\n", substream); if (NULL == usX2Y->hwdep_pcm_shm) { - if (NULL == (usX2Y->hwdep_pcm_shm = snd_malloc_pages(sizeof(struct snd_usX2Y_hwdep_pcm_shm), GFP_KERNEL))) + usX2Y->hwdep_pcm_shm = alloc_pages_exact(sizeof(struct snd_usX2Y_hwdep_pcm_shm), + GFP_KERNEL); + if (!usX2Y->hwdep_pcm_shm) return -ENOMEM; memset(usX2Y->hwdep_pcm_shm, 0, sizeof(struct snd_usX2Y_hwdep_pcm_shm)); } @@ -700,7 +702,7 @@ static void snd_usX2Y_hwdep_pcm_private_free(struct snd_hwdep *hwdep) { struct usX2Ydev *usX2Y = hwdep->private_data; if (NULL != usX2Y->hwdep_pcm_shm) - snd_free_pages(usX2Y->hwdep_pcm_shm, sizeof(struct snd_usX2Y_hwdep_pcm_shm)); + free_pages_exact(usX2Y->hwdep_pcm_shm, sizeof(struct snd_usX2Y_hwdep_pcm_shm)); } -- 2.19.1

3 3

[alsa-devel] [PATCH 1/2] ALSA: wss: Fix invalid snd_free_pages() at error path
by Takashi Iwai 24 Nov '18

24 Nov '18

Some spurious calls of snd_free_pages() have been overlooked and remain in the error paths of wss driver code. Since runtime->dma_area is managed by the PCM core helper, we shouldn't release manually. Drop the superfluous calls. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/wss/wss_lib.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 32453f81b95a..3a5008837576 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1531,7 +1531,6 @@ static int snd_wss_playback_open(struct snd_pcm_substream *substream) if (err < 0) { if (chip->release_dma) chip->release_dma(chip, chip->dma_private_data, chip->dma1); - snd_free_pages(runtime->dma_area, runtime->dma_bytes); return err; } chip->playback_substream = substream; @@ -1572,7 +1571,6 @@ static int snd_wss_capture_open(struct snd_pcm_substream *substream) if (err < 0) { if (chip->release_dma) chip->release_dma(chip, chip->dma_private_data, chip->dma2); - snd_free_pages(runtime->dma_area, runtime->dma_bytes); return err; } chip->capture_substream = substream; -- 2.19.1

2 3

[alsa-devel] [PATCH v3 0/3] Huawei laptops
by Ayman Bagabas 24 Nov '18

24 Nov '18

Changes from v2: * Support for Huawei MBX * Style and formating issues [PATCH v3 1/3] The first patch adds support for missing hotkeys on some models. [PATCH v3 2/3] This one enables the front speakers on the Huawei Matebook X Pro (MBXP). This solves bug 200501 https://bugzilla.kernel.org/show_bug.cgi?id=200501 It simply uses the pins configurations generated by hdajackretast using the settings posted on the bug page https://imgur.com/a/N1xsCVZ [PATCH v3 3/3] This enables the micmute LED on Huawei laptops using ACPI calls. Ayman Bagabas (3): x86: add support for Huawei WMI hotkeys. ALSA: hda: fix front speakers on Huawei MBXP. ALSA: hda: add support for Huawei WMI micmute LED drivers/platform/x86/Kconfig | 11 + drivers/platform/x86/Makefile | 1 + drivers/platform/x86/huawei_wmi.c | 236 +++++++++++++++++++ include/linux/platform_data/x86/huawei_wmi.h | 9 + sound/pci/hda/huawei_wmi_helper.c | 47 ++++ sound/pci/hda/patch_realtek.c | 29 +++ 6 files changed, 333 insertions(+) create mode 100644 drivers/platform/x86/huawei_wmi.c create mode 100644 include/linux/platform_data/x86/huawei_wmi.h create mode 100644 sound/pci/hda/huawei_wmi_helper.c -- 2.17.2

5 30

[alsa-devel] Undefined behaviour in ac97_codec.c - shift exponent 68 is too large for 32-bit type 'int'
by Meelis Roos 23 Nov '18

23 Nov '18

I updated one of my old laptops (ECS Desknote 532 with Transmeta CPU) to newest kernel (4.20.0-rc3-00145-gedeca3a769ad) and turned on UBSAN checks. Got the following UBSAN warning multiple times per boot. The soundcard: 00:04.0 Multimedia audio controller [0401]: ULi Electronics Inc. M5455 PCI AC-Link Controller Audio Device [10b9:5455] (rev 10) Subsystem: Elitegroup Computer Systems M5455 PCI AC-Link Controller Audio Device [1019:0f56] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (16000ns min), Cache Line Size: 128 bytes Interrupt: pin A routed to IRQ 5 Region 0: I/O ports at e400 [size=256] Region 1: Memory at febfe000 (32-bit, non-prefetchable) [size=4K] Capabilities: <access denied> Kernel driver in use: snd_intel8x0 Kernel modules: snd_intel8x0 /proc/asound/cards: 0 [M5455 ]: ICH - ALi M5455 ALi M5455 with ALC655 at irq 5 gcc version 8.2.0 (Debian 8.2.0-9) [ 15.688683] snd_intel8x0 0000:00:04.0: intel8x0_measure_ac97_clock: measured 58318 usecs (2808 samples) [ 15.689033] snd_intel8x0 0000:00:04.0: clocking to 48000 ... [ 19.667746] ================================================================================ [ 19.668078] UBSAN: Undefined behaviour in sound/pci/ac97/ac97_codec.c:836:7 [ 19.668268] shift exponent 68 is too large for 32-bit type 'int' [ 19.668465] CPU: 0 PID: 199 Comm: alsactl Not tainted 4.20.0-rc3-00145-gedeca3a769ad #2 [ 19.668602] Hardware name: Elitegroup Co. 532/532, BIOS 080010 02/22/2005 [ 19.668602] Call Trace: [ 19.668602] dump_stack+0x16/0x19 [ 19.668602] ubsan_epilogue+0xb/0x29 [ 19.668602] __ubsan_handle_shift_out_of_bounds.cold.15+0x26/0x78 [ 19.668602] snd_ac97_put_spsa.cold.50+0xf/0x24 [snd_ac97_codec] [ 19.668602] ? _copy_from_user+0x33/0xd0 [ 19.668602] snd_ctl_ioctl+0x69e/0x820 [ 19.668602] ? __seccomp_filter+0x60/0x320 [ 19.668602] ? snd_ctl_elem_add_user+0x8a0/0x8a0 [ 19.668602] do_vfs_ioctl+0x90/0x6c0 [ 19.668602] ? __switch_to_asm+0x26/0x4c [ 19.668602] ? __switch_to_asm+0x32/0x4c [ 19.668602] ? __switch_to_asm+0x26/0x4c [ 19.668602] ? __switch_to_asm+0x32/0x4c [ 19.668602] ? __switch_to_asm+0x26/0x4c [ 19.668602] ? __switch_to_asm+0x32/0x4c [ 19.668602] ? __switch_to_asm+0x26/0x4c [ 19.668602] ? __switch_to_asm+0x32/0x4c [ 19.668602] ? __switch_to_asm+0x26/0x4c [ 19.668602] ? __switch_to_asm+0x26/0x4c [ 19.668602] ? __secure_computing+0x2b/0x80 [ 19.668602] ? syscall_trace_enter+0x141/0x1b0 [ 19.668602] ? __switch_to_asm+0x26/0x4c [ 19.668602] ksys_ioctl+0x39/0x70 [ 19.668602] sys_ioctl+0x11/0x13 [ 19.668602] do_fast_syscall_32+0x90/0x1c0 [ 19.668602] entry_SYSENTER_32+0x6b/0xbd [ 19.668602] EIP: 0xb7fd69ad [ 19.668602] Code: 54 cd ff ff 85 d2 8b 98 58 cd ff ff 89 c8 74 02 89 0a 5b 5d c3 8b 04 24 c3 8b 14 24 c3 8b 3c 24 c3 51 52 55 89 e5 0f 34 cd 80 <5d> 5a 59 c3 90 90 90 90 8d 76 00 58 b8 77 00 00 00 cd 80 90 8d 76 [ 19.668602] EAX: ffffffda EBX: 00000003 ECX: c2c45513 EDX: bffff670 [ 19.668602] ESI: 00000000 EDI: 00000001 EBP: bffff9c8 ESP: bffff508 [ 19.668602] DS: 007b ES: 007b FS: 0000 GS: 0033 SS: 007b EFLAGS: 00000292 [ 19.668602] ================================================================================ -- Meelis Roos <mroos(a)linux.ee>

2 3

[alsa-devel] [PATCH] ALSA: hda/realtek - fix headset mic detection for MSI MS-B171
by Anisse Astier 23 Nov '18

23 Nov '18

MSI Cubi N 8GL (MS-B171) needs the same fixup as its older model, the MS-B120, in order for the headset mic to be properly detected. They both use a single 3-way jack for both mic and headset with an ALC283 codec, with the same pins used. Cc: stable(a)vger.kernel.org Signed-off-by: Anisse Astier <anisse(a)astier.eu> --- 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 fa61674a5605..1e789e43a5f4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6531,6 +6531,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), -- 2.11.0

2 1

[alsa-devel] [PATCH] ALSA: hda: Add ASRock N68C-S UCC the power_save blacklist
by Hans de Goede 23 Nov '18

23 Nov '18

Power-saving is causing plops on audio start/stop on the built-in audio of the nForce 430 based ASRock N68C-S UCC motherboard, add this model to the power_save blacklist. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 Cc: <stable(a)vger.kernel.org> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d8eb2b5f51ae..0bbdf1a01e76 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2169,6 +2169,8 @@ static struct snd_pci_quirk power_save_blacklist[] = { /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1849, 0xc892, "Asrock B85M-ITX", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1849, 0x0397, "Asrock N68C-S UCC", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1849, 0x7662, "Asrock H81M-HDS", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), -- 2.19.1

2 1

[alsa-devel] [PATCH 0/5] ALSA: firewire-tascam: support notification of
by Takashi Sakamoto 23 Nov '18

23 Nov '18

Hi, Units of TASCAM FireWire series transfer its state in tx isochronous packets. The way to handle change of the state is a longstanding issue since I firstly reveal the mechanism in this list[1]. In this time, Scott Bahling dedicates for further investigation to seek better implementation[2][3][4]. Finally, we identified meaning of the most of bits in the image of state, and decided which changes are worth for userspace applications to handle as events. This commit adds support notification of the events. Userspace applications can handle the notification via ALSA hwdep interface. Furthermore, they can retrieve image of the latest states via the interface. Changes from the initial proposal[5]: - obsolete usage of mmap(2) due to several disadvantages. Instead, use event queue for change notification, and add ioctl command to retrieve image of states. Changes from the latest work[6]: - To reduce time in an interrupt context in which tx isochronous packets are handled, data in the event and the image is passed to userspace in big-endian order. Additionally, image of the states are passed to userspace in big-endian order as well. Userspace applications are responsible to convert the data in host-endianness order. - improve efficiency to copy batch of events to userspace in hwdep interface - rename structure and macro name. This patchset is also available in my remote branch for backport purpose to Linux v4.17 or later[7]. I've already prepare userspace implementation; libhinwa. A 'topic/firewire-tascam-pr' remote branch includes patchset for the additional feature[8]. This library produces GObject class to handle the event notification and emits GObject signal, and has a method 'HinawaSndTscm.get_state()', with enough consideration to endianness. Developers can process change of state with these two APIs. > Scott As the above, I changed method name from 'HinawaSndTscm.get_status()'. I'd like you to change your local implementation for 'hinawa-utils'. Sorry to add further work to you. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/094817.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140051.… [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-October/140593.ht… [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-November/141395.h… [5] https://github.com/takaswie/snd-firewire-improve/tree/topic/tascam-userspace [6] https://github.com/takaswie/snd-firewire-improve/tree/topic/tascam-userspac… [7] https://github.com/takaswie/snd-firewire-improve/tree/topic/tascam-userspac… [8] https://github.com/takaswie/libhinawa/tree/topic/tascam-userspace-pr Regards Takashi Sakamoto (5): ALSA: firewire-tascam: pick up data of state from tx isochronous pakcets ALSA: firewire-tascam: add new hwdep ioctl command to get state image ALSA: firewire-tascam: queue events for change of control surface ALSA: firewire-tascam: apply small refactoring to handle several type of event for hwdep interface ALSA: firewire-tascam: notify events of change of state for userspace applications include/uapi/sound/firewire.h | 20 +++++ sound/firewire/tascam/amdtp-tascam.c | 51 +++++++++++- sound/firewire/tascam/tascam-hwdep.c | 115 +++++++++++++++++++++++---- sound/firewire/tascam/tascam.h | 9 +++ 4 files changed, 180 insertions(+), 15 deletions(-) -- 2.19.1

2 7

[alsa-devel] Applied "ASoC: dt-bindings: add regulator property to simple amplifier" to the asoc tree
by Mark Brown 23 Nov '18

23 Nov '18

The patch ASoC: dt-bindings: add regulator property to simple amplifier 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 8e84de031a11c4240f3654fd4c380542e358f5f6 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick <anarsoul(a)gmail.com> Date: Thu, 22 Nov 2018 18:23:20 +0800 Subject: [PATCH] ASoC: dt-bindings: add regulator property to simple amplifier Amplifier may have associated regulator, so add a property for it. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Signed-off-by: Chen-Yu Tsai <wens(a)csie.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/simple-amplifier.txt | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-amplifier.txt b/Documentation/devicetree/bindings/sound/simple-amplifier.txt index 8647edae7af0..7182ac4f1e65 100644 --- a/Documentation/devicetree/bindings/sound/simple-amplifier.txt +++ b/Documentation/devicetree/bindings/sound/simple-amplifier.txt @@ -4,9 +4,14 @@ Required properties: - compatible : "dioo,dio2125" or "simple-audio-amplifier" - enable-gpios : the gpio connected to the enable pin of the simple amplifier +Optional properties: +- VCC-supply : power supply for the device, as covered + in Documentation/devicetree/bindings/regulator/regulator.txt + Example: amp: analog-amplifier { compatible = "simple-audio-amplifier"; + VCC-supply = <&regulator>; enable-gpios = <&gpio GPIOH_3 0>; }; -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: simple-amplifier: add VCC regulator widget" to the asoc tree
by Mark Brown 23 Nov '18

23 Nov '18

The patch ASoC: simple-amplifier: add VCC regulator widget 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 6debd01a74b8c15ab3f2a6f7ff4a7dcbada101cb Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick <anarsoul(a)gmail.com> Date: Thu, 22 Nov 2018 18:23:21 +0800 Subject: [PATCH] ASoC: simple-amplifier: add VCC regulator widget Amplifier may have assosicated regulator, so add a widget for it and appropriate route. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Signed-off-by: Chen-Yu Tsai <wens(a)csie.org> Acked-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/simple-amplifier.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/simple-amplifier.c b/sound/soc/codecs/simple-amplifier.c index 85524acf3e9c..c07e8a80b4b7 100644 --- a/sound/soc/codecs/simple-amplifier.c +++ b/sound/soc/codecs/simple-amplifier.c @@ -19,6 +19,7 @@ #include <linux/gpio/consumer.h> #include <linux/module.h> +#include <linux/regulator/consumer.h> #include <sound/soc.h> #define DRV_NAME "simple-amplifier" @@ -58,11 +59,14 @@ static const struct snd_soc_dapm_widget simple_amp_dapm_widgets[] = { (SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD)), SND_SOC_DAPM_OUTPUT("OUTL"), SND_SOC_DAPM_OUTPUT("OUTR"), + SND_SOC_DAPM_REGULATOR_SUPPLY("VCC", 20, 0), }; static const struct snd_soc_dapm_route simple_amp_dapm_routes[] = { { "DRV", NULL, "INL" }, { "DRV", NULL, "INR" }, + { "OUTL", NULL, "VCC" }, + { "OUTR", NULL, "VCC" }, { "OUTL", NULL, "DRV" }, { "OUTR", NULL, "DRV" }, }; -- 2.19.1

1 0

[alsa-devel] [PATCH 0/3] ASoC: omap: Fix and add pm_qos configuration
by Peter Ujfalusi 23 Nov '18

23 Nov '18

Hi, The defconfig for OMAP2+ now have the CPU_IDLE enabled which can cause audio artifacts because we try to enter too low power state from where the wakeup takes longer than the FIFO can tolerate on the dai side. While adding pm_qos to McPDM and DMIC I have noticed that the McBSP calculation was not correct as we need usec for the latency value. Regards, Peter --- Peter Ujfalusi (3): ASoC: omap-mcbsp: Fix latency value calculation for pm_qos ASoC: omap-mcpdm: Add pm_qos handling to avoid under/overruns with CPU_IDLE ASoC: omap-dmic: Add pm_qos handling to avoid overruns with CPU_IDLE sound/soc/omap/omap-dmic.c | 9 ++++++++ sound/soc/omap/omap-mcbsp.c | 6 +++--- sound/soc/omap/omap-mcpdm.c | 43 ++++++++++++++++++++++++++++++++++++- 3 files changed, 54 insertions(+), 4 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

4 14

[alsa-devel] [PATCH] ASoC: sunxi: rename SND_SUN8I_ADDA_PR_REGMAP
by Arnd Bergmann 23 Nov '18

23 Nov '18

The newly added SND_SUN50I_CODEC_ANALOG driver selects a nonexisting symbol SND_SUNXI_ADDA_PR_REGMAP, which was evidently intended to replace SND_SUN8I_ADDA_PR_REGMAP, but since they are now mismatched, we can run into link errors for some configurations: sound/soc/sunxi/sun50i-codec-analog.o: In function `sun50i_codec_analog_probe': sun50i-codec-analog.c:(.text+0x62): undefined reference to `sun8i_adda_pr_regmap_init' The new name appears to be more sensible, and as the symbol is hidden, there are no downsides in the rename, so use that consistently now. Fixes: 42371f327df0 ("ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/sunxi/Kconfig | 4 ++-- sound/soc/sunxi/Makefile | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 66aad0d3f9c7..0f992bdf1a10 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -23,7 +23,7 @@ config SND_SUN8I_CODEC config SND_SUN8I_CODEC_ANALOG tristate "Allwinner sun8i Codec Analog Controls Support" depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST - select SND_SUN8I_ADDA_PR_REGMAP + select SND_SUNXI_ADDA_PR_REGMAP help Say Y or M if you want to add support for the analog controls for the codec embedded in newer Allwinner SoCs. @@ -54,7 +54,7 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. -config SND_SUN8I_ADDA_PR_REGMAP +config SND_SUNXI_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..c4dd2803a011 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -5,4 +5,4 @@ obj-$(CONFIG_SND_SUN4I_SPDIF) += sun4i-spdif.o obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o -obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUNXI_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o -- 2.18.0

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[alsa-devel] [PATCH 0/2] ALSA: control: Race fix and cleanup for user ctls
by Takashi Iwai 22 Nov '18

22 Nov '18

Hi, there seems a possible race between addition and removal of a user ctl element as spotted by syzkaller. This patchset addresses it and also provides a cleanup thereafter. The first one should go to 4.20 while the second one should go for 4.21. Takashi === Takashi Iwai (2): ALSA: control: Fix race between adding and removing a user element ALSA: control: Consolidate helpers for adding and replacing ctl elements sound/core/control.c | 171 ++++++++++++++++++++----------------------- 1 file changed, 81 insertions(+), 90 deletions(-) -- 2.19.1

1 2

[alsa-devel] [PATCH 0/2] ASoC: SDM845: Update MI2S and TDM configuration
by Rohit kumar 21 Nov '18

21 Nov '18

Update bit clock rate, slot width for TDM and MI2S interfaces. Also add support for secondary MI2S TX interface in SDM845 machine driver. Rohit kumar (2): ASoC: sdm845: Update slot_width for Quaternary TDM port ASoC: sdm845: Add support for Secondary MI2S interface sound/soc/qcom/sdm845.c | 27 +++++++++++++++++++++++---- 1 file changed, 23 insertions(+), 4 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 6

[alsa-devel] [PATCH] ASoC: rsnd: Add device tree support for r8a77995
by Kuninori Morimoto 21 Nov '18

21 Nov '18

From: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Simply document new compat strings. There appears to be no need for a driver updates. Signed-off-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 453e8ec..47dc819 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -277,6 +277,7 @@ Required properties: - "renesas,rcar_sound-r8a7796" (R-Car M3-W) - "renesas,rcar_sound-r8a77965" (R-Car M3-N) - "renesas,rcar_sound-r8a77990" (R-Car E3) + - "renesas,rcar_sound-r8a77995" (R-Car D3) - reg : Should contain the register physical address. required register is SRU/ADG/SSI if generation1 -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rsnd: tidyup rsnd_parse_connect_ssiu_compatible()
by Kuninori Morimoto 21 Nov '18

21 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd_parse_connect_ssiu_compatible() is doing - using rsnd_ssiu_id(), but we use it via rsnd_mod_id() - we can break loop if rsnd_dai_connect() was called This patch fixup these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssiu.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 650b14e..04b1a48 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -382,9 +382,11 @@ static void rsnd_parse_connect_ssiu_compatible(struct rsnd_priv *priv, for_each_rsnd_ssiu(ssiu, priv, i) { mod = rsnd_mod_get(ssiu); - if ((rsnd_mod_id(ssi_mod) == rsnd_ssiu_id(mod)) && - (rsnd_mod_id_sub(mod) == 0)) + if ((rsnd_mod_id(ssi_mod) == rsnd_mod_id(mod)) && + (rsnd_mod_id_sub(mod) == 0)) { rsnd_dai_connect(mod, io, mod->type); + return; + } } } -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rsnd: makes rsnd_ssi_is_dma_mode() static
by Kuninori Morimoto 21 Nov '18

21 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> ssi.c only is using rsnd_ssi_is_dma_mode(). Let's move it as static function. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsnd.h | 1 - sound/soc/sh/rcar/ssi.c | 5 +++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 64c3a3b..3c57129 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -726,7 +726,6 @@ extern const char * const volume_ramp_rate[]; int rsnd_ssi_probe(struct rsnd_priv *priv); void rsnd_ssi_remove(struct rsnd_priv *priv); struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id); -int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod); int rsnd_ssi_use_busif(struct rsnd_dai_stream *io); u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 361dc2c..6439a74 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -117,6 +117,8 @@ struct rsnd_ssi { (rsnd_ssi_run_mods(io) & (1 << rsnd_mod_id(mod))) #define rsnd_ssi_can_output_clk(mod) (!__rsnd_ssi_is_pin_sharing(mod)) +static int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod); + int rsnd_ssi_use_busif(struct rsnd_dai_stream *io) { struct rsnd_mod *mod = rsnd_io_to_mod_ssi(io); @@ -1003,12 +1005,11 @@ static struct rsnd_mod_ops rsnd_ssi_dma_ops = { .get_status = rsnd_ssi_get_status, }; -int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod) +static int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod) { return mod->ops == &rsnd_ssi_dma_ops; } - /* * ssi mod function */ -- 2.7.4

2 1

[alsa-devel] [PATCH 0/4] ASoC: merge audio-graph-scu into audio-graph - step1
by Kuninori Morimoto 21 Nov '18

21 Nov '18

Hi Mark audio-graph card can handle normal sound card, and audio-graph-scu card can handle DPCM sound card. But, we can't use these feature in same time now. ex) audio-graph can handle normal sound card CPU1 -- Codec1 CPU2 -- Codec2 audio-graph-scu can handle DPCM sound card CPU1 --- Codec1 CPU2 -/ But, we can't use these in same time CPU1 --- Codec1 CPU2 --- Codec2 CPU3 -/ So, this is the time to merge audio-graph and audio-graph-scu. The patch-set are a little bit huge (and few patches are depend on posted OF patch. I'm now waiting...), thus, I separated these into few steps. This is step1 as prepare merging. I think it will be 4 or more steps. "simple card" and "audio graph card" are similar sound card, and now I'm focusing to "audio graph card" merging only. Basically simple-card and simple-scu-card are similar too, and we can merge it. But I have no plan for it now. Please try it someone ;P Kuninori Morimoto (4): ASoC: simple-card-utils: remove asoc_simple_card_clk_register() ASoC: simple-card-utils: accept NULL parameter on asoc_simple_card_xxx() ASoC: simple-card-utils: tidyup asoc_simple_card_parse_convert() ASoC: simple_card_utils: remove "option" from asoc_simple_card_of_parse_routing() include/sound/simple_card_utils.h | 6 +++--- sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/audio-graph-scu-card.c | 4 ++-- sound/soc/generic/simple-card-utils.c | 34 +++++++++++++++----------------- sound/soc/generic/simple-card.c | 2 +- sound/soc/generic/simple-scu-card.c | 4 ++-- 6 files changed, 25 insertions(+), 27 deletions(-) -- 2.7.4

2 8

[alsa-devel] [PATCH v4 0/2] Introduce dmic mode switch delay parameter
by Jenny TC 21 Nov '18

21 Nov '18

Some DMICs need clock to be running for a specified duration as per the DMIC spec to complete the mode transition like sleep to normal, off to normal etc. This series add a mode-switch delay which will prevent the clock from being turned off without complying with manufacturer timing specifications. Also introduces a module parameter for the wakeup_delay parameter. v1: First patch set v2: 1). Removed ACPI device entry since a) The DMIC vendor doesn't have an ACPI vendor ID b) The ACPI enumeration creates one more device and conflicts with the existing skl.c dmic platform device registration. 2). Updated commit message v3: Updated commit message and introduced module param for wakeup_delay v4: Addressed parameter name to align with existing parameter. Added dt entry in Documentation Jenny TC (2): ASoC: dmic: introduce mode switch delay ASoC: dmic: introduce module_param wakeup_delay Documentation/devicetree/bindings/sound/dmic.txt | 2 ++ sound/soc/codecs/dmic.c | 36 ++++++++++++++++++++++++ 2 files changed, 38 insertions(+) -- 1.9.1

3 5

[alsa-devel] [PATCH alsa-lib] pcm_dshare: Fix segfault when not binding channel 0
by Kirill Marinushkin 21 Nov '18

21 Nov '18

Configuration to reproduce: ~~~~ pcm.share_right { type dshare ipc_key 73 ipc_perm 0666 slave { pcm "hw:0,0" } bindings { # the seagfault happens when we don't bind channel 0 1 1 } } ~~~~ Execute to reproduce: ~~~~ $ aplay -D plug:share_right test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Segmentation fault ~~~~ For channels whithout binding, values are set to UINT_MAX in function `snd_pcm_direct_parse_bindings()`: ~~~~ for (chn = 0; chn < count; chn++) bindings[chn] = UINT_MAX; /* don't route */ ~~~~ But, these values are not checked when playing, which causes the segfault. This commit fixes the issue. Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> --- src/pcm/pcm_dshare.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 54dc9d48..2bb735fe 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -67,7 +67,9 @@ static void do_silence(snd_pcm_t *pcm) format = dshare->shmptr->s.format; for (chn = 0; chn < channels; chn++) { dchn = dshare->bindings ? dshare->bindings[chn] : chn; - snd_pcm_area_silence(&dst_areas[dchn], 0, dshare->shmptr->s.buffer_size, format); + if (dchn != UINT_MAX) + snd_pcm_area_silence(&dst_areas[dchn], 0, + dshare->shmptr->s.buffer_size, format); } } @@ -91,7 +93,9 @@ static void share_areas(snd_pcm_direct_t *dshare, } else { for (chn = 0; chn < channels; chn++) { dchn = dshare->bindings ? dshare->bindings[chn] : chn; - snd_pcm_area_copy(&dst_areas[dchn], dst_ofs, &src_areas[chn], src_ofs, size, format); + if (dchn != UINT_MAX) + snd_pcm_area_copy(&dst_areas[dchn], dst_ofs, + &src_areas[chn], src_ofs, size, format); } } @@ -834,8 +838,11 @@ int snd_pcm_dshare_open(snd_pcm_t **pcmp, const char *name, dshare->spcm = spcm; } - for (chn = 0; chn < dshare->channels; chn++) - dshare->u.dshare.chn_mask |= (1ULL<<dshare->bindings[chn]); + for (chn = 0; chn < dshare->channels; chn++) { + unsigned int dchn = dshare->bindings ? dshare->bindings[chn] : chn; + if (dchn != UINT_MAX) + dshare->u.dshare.chn_mask |= (1ULL << dchn); + } if (dshare->shmptr->u.dshare.chn_mask & dshare->u.dshare.chn_mask) { SNDERR("destination channel specified in bindings is already used"); dshare->u.dshare.chn_mask = 0; -- 2.13.6

2 1

[alsa-devel] [RFC PATCH] ASoC: Intel: Skylake: fix Kconfigs, make HDaudio codec optional
by Pierre-Louis Bossart 20 Nov '18

20 Nov '18

The Skylake driver currently has a set of problems supporting load/unload modules. We need to make the HDaudio codec support optional to help narrow down the issues. Support for HDaudio codecs also leads to a Kconfig issue. We want the hdac_hda codec to be compilable independently of Skylake (e.g. with ALL_CODECS) but when Skylake is selected as built-in the hdac_hda codec needs to use the same option due a a code dependency Solve both problems by adding a user-selectable boolean Kconfig, select HDAC_HDA as needed and make the HDaudio codec support in the Skylake driver optional. Tests on a Chell Chromebook device without HDaudio show no regression for speaker and HDMI playback. This is submitted as an RFC to allow for comments and more validation. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/Kconfig | 26 +++++++++++++++++++++++--- sound/soc/intel/boards/Kconfig | 24 ++++++++++++++---------- sound/soc/intel/skylake/skl.c | 20 ++++++++++++++++++-- 3 files changed, 55 insertions(+), 15 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 0caa1f4eb94d..18e717703685 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -101,22 +101,42 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI codec, then enable this option by saying Y or m. This is a recommended option -config SND_SOC_INTEL_SKYLAKE_SSP_CLK - tristate - config SND_SOC_INTEL_SKYLAKE tristate "SKL/BXT/KBL/GLK/CNL... Platforms" depends on PCI && ACPI + select SND_SOC_INTEL_SKYLAKE_COMMON + help + If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ + GeminiLake or CannonLake platform with the DSP enabled in the BIOS + then enable this option by saying Y or m. + +if SND_SOC_INTEL_SKYLAKE + +config SND_SOC_INTEL_SKYLAKE_SSP_CLK + tristate + +config SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC + bool "HDAudio codec support" + help + If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ + GeminiLake or CannonLake platform with an HDaudio codec + then enable this option by saying Y + +config SND_SOC_INTEL_SKYLAKE_COMMON + tristate select SND_HDA_EXT_CORE select SND_HDA_DSP_LOADER select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST + select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS then enable this option by saying Y or m. +endif ## SND_SOC_INTEL_SKYLAKE + config SND_SOC_ACPI_INTEL_MATCH tristate select SND_SOC_ACPI if ACPI diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 73ca1350aa31..b177db2a0dbb 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -293,16 +293,6 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH Say Y if you have such a device. If unsure select "N". -config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH - tristate "SKL/KBL/BXT/APL with HDA Codecs" - select SND_SOC_HDAC_HDMI - select SND_SOC_HDAC_HDA - help - This adds support for ASoC machine driver for Intel platforms - SKL/KBL/BXT/APL with iDisp, HDA audio codecs. - Say Y or m if you have such a device. This is a recommended option. - 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 @@ -319,4 +309,18 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH endif ## SND_SOC_INTEL_SKYLAKE +if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC + +config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH + tristate "SKL/KBL/BXT/APL with HDA Codecs" + select SND_SOC_HDAC_HDMI + # SND_SOC_HDAC_HDA is already selected + help + This adds support for ASoC machine driver for Intel platforms + SKL/KBL/BXT/APL with iDisp, HDA audio codecs. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC + endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index ac0b4ff21acc..901ef5902762 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -37,7 +37,9 @@ #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) #include "../../../soc/codecs/hdac_hda.h" +#endif /* * initialize the PCI registers @@ -654,6 +656,8 @@ static void skl_clock_device_unregister(struct skl *skl) platform_device_unregister(skl->clk_dev); } +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) + #define IDISP_INTEL_VENDOR_ID 0x80860000 /* @@ -672,6 +676,8 @@ static void load_codec_module(struct hda_codec *codec) #endif } +#endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */ + /* * Probe the given codec address */ @@ -681,9 +687,11 @@ static int probe_codec(struct hdac_bus *bus, int addr) (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; unsigned int res = -1; struct skl *skl = bus_to_skl(bus); +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) struct hdac_hda_priv *hda_codec; - struct hdac_device *hdev; int err; +#endif + struct hdac_device *hdev; mutex_lock(&bus->cmd_mutex); snd_hdac_bus_send_cmd(bus, cmd); @@ -693,6 +701,7 @@ static int probe_codec(struct hdac_bus *bus, int addr) return -EIO; dev_dbg(bus->dev, "codec #%d probed OK: %x\n", addr, res); +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) hda_codec = devm_kzalloc(&skl->pci->dev, sizeof(*hda_codec), GFP_KERNEL); if (!hda_codec) @@ -711,6 +720,13 @@ static int probe_codec(struct hdac_bus *bus, int addr) load_codec_module(&hda_codec->codec); } return 0; +#else + hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL); + if (!hdev) + return -ENOMEM; + + return snd_hdac_ext_bus_device_init(bus, addr, hdev); +#endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */ } /* Codec initialization */ @@ -866,7 +882,7 @@ static int skl_create(struct pci_dev *pci, hbus = skl_to_hbus(skl); bus = skl_to_bus(skl); -#if IS_ENABLED(CONFIG_SND_SOC_HDAC_HDA) +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) ext_ops = snd_soc_hdac_hda_get_ops(); #endif snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, ext_ops); -- 2.17.1

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[alsa-devel] [PATCH 0/3] Intel SST fixes for Chromebooks/HDaudio platorms
by Pierre-Louis Bossart 20 Nov '18

20 Nov '18

While testing Mark's for-next branch on Chromebooks, 3 problems were exposed a. the search loop for ACPI IDs is broken on SKL/KBL which are the most advanced users of quirks. This is embarrassing since it's been broken for a very very long time and those platforms have been stuck in time with 3.18 and 4.4 kernels. b. the change I personally made to use a common interface for machine driver didn't work either for all the Chromebooks using information on dmic number. My bad. c. the change to use the common interface also missed the fact that the Skylake driver passed the information using drv_data instead of platform_data as done by the Atom and SOF drivers. The third patch aligns the Skylake driver with the two others. We should now have no dependencies between machine and platform drivers, which will help reuse the same code. These patches were tested on HDaudio platforms (ApolloLake and Whisky Lake) as well as Chell (SKL) and Soraka (KBL) Chromebooks - for the latter case we are still investigating DSP IPC errors that are unrelated to these changes Keyon Jie (1): ASoC: acpi: fix: continue searching when machine is ignored Pierre-Louis Bossart (2): ASoC: Intel: fix interface for Chromebook machine drivers ASoC: Intel: use platform_data for machine drivers include/sound/soc-acpi.h | 1 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 10 +++++----- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 10 +++++----- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 10 +++++----- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 10 +++++----- sound/soc/intel/skylake/skl.c | 15 +++++++++++---- sound/soc/intel/skylake/skl.h | 3 --- sound/soc/soc-acpi.c | 10 ++++++++-- 9 files changed, 41 insertions(+), 30 deletions(-) -- 2.17.1

2 6

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Add support for CBM_CFS and CBS_CFM clocking modes
by Peter Ujfalusi 20 Nov '18

20 Nov '18

The codec can support any variation of bclk/fs master/slave configuration. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6a271e6e6b8f..6aa0edf8c5ef 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1260,6 +1260,16 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, aic3x->master = 0; iface_areg &= ~(BIT_CLK_MASTER | WORD_CLK_MASTER); break; + case SND_SOC_DAIFMT_CBM_CFS: + aic3x->master = 1; + iface_areg |= BIT_CLK_MASTER; + iface_areg &= ~WORD_CLK_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFM: + aic3x->master = 1; + iface_areg |= WORD_CLK_MASTER; + iface_areg &= ~BIT_CLK_MASTER; + break; default: return -EINVAL; } -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ASoC: Intel: hdac_hdmi: add Icelake support
by Bard liao 20 Nov '18

20 Nov '18

From: Bard liao <bard.liao(a)intel.com> Add Icelake device id. Also, Icelake's pin2port mapping table is complicated. So we use a mapping table to do the pin2port mapping. Signed-off-by: Bard liao <bard.liao(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 63 ++++++++++++++++++++++++++++++------ 1 file changed, 54 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 4e9854889a95..fac397326515 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -121,8 +121,16 @@ struct hdac_hdmi_dai_port_map { struct hdac_hdmi_cvt *cvt; }; +/* + * pin to port mapping table where the value indicate the pin number and + * the index indicate the port number with 1 base. + */ +static const int icl_pin2port_map[] = {0x4, 0x6, 0x8, 0xa, 0xb}; + struct hdac_hdmi_drv_data { unsigned int vendor_nid; + const int *port_map; /* pin to port mapping table */ + int port_num; }; struct hdac_hdmi_priv { @@ -1329,11 +1337,12 @@ static int hdac_hdmi_add_pin(struct hdac_device *hdev, hda_nid_t nid) return 0; } -#define INTEL_VENDOR_NID 0x08 -#define INTEL_GLK_VENDOR_NID 0x0b +#define INTEL_VENDOR_NID_0x2 0x02 +#define INTEL_VENDOR_NID_0x8 0x08 +#define INTEL_VENDOR_NID_0xb 0x0b #define INTEL_GET_VENDOR_VERB 0xf81 #define INTEL_SET_VENDOR_VERB 0x781 -#define INTEL_EN_DP12 0x02 /* enable DP 1.2 features */ +#define INTEL_EN_DP12 0x02 /* enable DP 1.2 features */ #define INTEL_EN_ALL_PIN_CVTS 0x01 /* enable 2nd & 3rd pins and convertors */ static void hdac_hdmi_skl_enable_all_pins(struct hdac_device *hdev) @@ -1538,7 +1547,26 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_device *hdev, static int hdac_hdmi_pin2port(void *aptr, int pin) { - return pin - 4; /* map NID 0x05 -> port #1 */ + struct hdac_device *hdev = aptr; + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); + const int *map = hdmi->drv_data->port_map; + int i; + + if (!hdmi->drv_data->port_num) + return pin - 4; /* map NID 0x05 -> port #1 */ + + /* + * looking for the pin number in the mapping table and return + * the index which indicate the port number + */ + for (i = 0; i < hdmi->drv_data->port_num; i++) { + if (pin == map[i]) + return i + 1; + } + + /* return -1 if pin number exceeds our expectation */ + dev_err(&hdev->dev, "Can't find the port for pin %d\n", pin); + return -1; } static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) @@ -1549,9 +1577,18 @@ static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) struct hdac_hdmi_port *hport = NULL; struct snd_soc_component *component = hdmi->component; int i; - - /* Don't know how this mapping is derived */ - hda_nid_t pin_nid = port + 0x04; + hda_nid_t pin_nid; + + if (!hdmi->drv_data->port_num) { + /* for legacy platforms */ + pin_nid = port + 0x04; + } else if (port < hdmi->drv_data->port_num) { + /* get pin number from the pin2port mapping table */ + pin_nid = hdmi->drv_data->port_map[port - 1]; + } else { + dev_err(&hdev->dev, "Can't find the pin for port %d\n", port); + return; + } dev_dbg(&hdev->dev, "%s: for pin:%d port=%d\n", __func__, pin_nid, pipe); @@ -1973,12 +2010,18 @@ static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdev, int pcm_idx) return port->eld.info.spk_alloc; } +static struct hdac_hdmi_drv_data intel_icl_drv_data = { + .vendor_nid = INTEL_VENDOR_NID_0x2, + .port_map = icl_pin2port_map, + .port_num = ARRAY_SIZE(icl_pin2port_map), +}; + static struct hdac_hdmi_drv_data intel_glk_drv_data = { - .vendor_nid = INTEL_GLK_VENDOR_NID, + .vendor_nid = INTEL_VENDOR_NID_0xb, }; static struct hdac_hdmi_drv_data intel_drv_data = { - .vendor_nid = INTEL_VENDOR_NID, + .vendor_nid = INTEL_VENDOR_NID_0x8, }; static int hdac_hdmi_dev_probe(struct hdac_device *hdev) @@ -2259,6 +2302,8 @@ static const struct hda_device_id hdmi_list[] = { &intel_glk_drv_data), HDA_CODEC_EXT_ENTRY(0x8086280d, 0x100000, "Geminilake HDMI", &intel_glk_drv_data), + HDA_CODEC_EXT_ENTRY(0x8086280f, 0x100000, "Icelake HDMI", + &intel_icl_drv_data), {} }; -- 2.17.1

4 18

[alsa-devel] [PATCH 1/2] ASoC: max9892x: Add documentation for reset-gpio support
by Cheng-Yi Chiang 20 Nov '18

20 Nov '18

max98927 codec driver will support reset-gpio binding so it can toggle reset line in its probe function. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Documentation/devicetree/bindings/sound/max9892x.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/max9892x.txt b/Documentation/devicetree/bindings/sound/max9892x.txt index f6171591ddc61..8bf30483eedcd 100644 --- a/Documentation/devicetree/bindings/sound/max9892x.txt +++ b/Documentation/devicetree/bindings/sound/max9892x.txt @@ -29,6 +29,8 @@ Required properties: Range : 0 (off), 1 (on), Default : 0 - reg : the I2C address of the device for I2C + - reset-gpio: reset gpio the chip is connected to. This is supported + on max98927. Example: @@ -38,4 +40,5 @@ codec: max98927@3a { imon-slot-no = <1>; interleave-mode = <0>; reg = <0x3a>; + reset-gpio = <&tlmm 75 GPIO_ACTIVE_LOW>; }; -- 2.16.4

6 13

[alsa-devel] [sound:for-linus 5/7] sound/pci/hda/patch_ca0132.c:8416:3: error: implicit declaration of function 'pci_iounmap'; did you mean 'pcim_iounmap'?
by kbuild test robot 20 Nov '18

20 Nov '18

Hi Takashi, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: a6b0961b39896a9f9f1350d26d202f078a7d9dbc commit: d99501b8575dc1248bacf1b58d2241cb4b265d49 [5/7] ALSA: hda/ca0132 - Call pci_iounmap() instead of iounmap() config: sh-allmodconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout d99501b8575dc1248bacf1b58d2241cb4b265d49 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sound/pci/hda/patch_ca0132.c: In function 'ca0132_free': >> sound/pci/hda/patch_ca0132.c:8416:3: error: implicit declaration of function 'pci_iounmap'; did you mean 'pcim_iounmap'? [-Werror=implicit-function-declaration] pci_iounmap(codec->bus->pci, spec->mem_base); ^~~~~~~~~~~ pcim_iounmap sound/pci/hda/patch_ca0132.c: In function 'patch_ca0132': sound/pci/hda/patch_ca0132.c:8799:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'? [-Werror=implicit-function-declaration] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^~~~~~~~~ pcim_iomap sound/pci/hda/patch_ca0132.c:8799:18: warning: assignment makes pointer from integer without a cast [-Wint-conversion] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^ cc1: some warnings being treated as errors vim +8416 sound/pci/hda/patch_ca0132.c 8386 8387 static void ca0132_free(struct hda_codec *codec) 8388 { 8389 struct ca0132_spec *spec = codec->spec; 8390 8391 cancel_delayed_work_sync(&spec->unsol_hp_work); 8392 snd_hda_power_up(codec); 8393 switch (spec->quirk) { 8394 case QUIRK_SBZ: 8395 sbz_exit_chip(codec); 8396 break; 8397 case QUIRK_ZXR: 8398 zxr_exit_chip(codec); 8399 break; 8400 case QUIRK_R3D: 8401 r3d_exit_chip(codec); 8402 break; 8403 case QUIRK_AE5: 8404 ae5_exit_chip(codec); 8405 break; 8406 case QUIRK_R3DI: 8407 r3di_gpio_shutdown(codec); 8408 break; 8409 } 8410 8411 snd_hda_sequence_write(codec, spec->base_exit_verbs); 8412 ca0132_exit_chip(codec); 8413 8414 snd_hda_power_down(codec); 8415 if (spec->mem_base) > 8416 pci_iounmap(codec->bus->pci, spec->mem_base); 8417 kfree(spec->spec_init_verbs); 8418 kfree(codec->spec); 8419 } 8420 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ALSA: aoa: Use device_type helpers to access the node type
by Rob Herring 19 Nov '18

19 Nov '18

Remove directly accessing device_node.type pointer and use the accessors instead. This will eventually allow removing the type pointer. Replace the open coded iterating over child nodes with for_each_child_of_node() while we're here. Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/aoa/fabrics/layout.c | 4 ++-- sound/aoa/soundbus/core.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 1eddf8fa188f..3a23fd032611 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -1008,8 +1008,8 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) return -ENODEV; /* by breaking out we keep a reference */ - while ((sound = of_get_next_child(sdev->ofdev.dev.of_node, sound))) { - if (sound->type && strcasecmp(sound->type, "soundchip") == 0) + for_each_child_of_node(sdev->ofdev.dev.of_node, sound) { + if (of_node_is_type(sound, "soundchip")) break; } if (!sound) diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c index 70bcaa7f93dd..f3076a3f7f2f 100644 --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -78,7 +78,7 @@ static int soundbus_uevent(struct device *dev, struct kobj_uevent_env *env) if (retval) return retval; - retval = add_uevent_var(env, "OF_TYPE=%s", of->dev.of_node->type); + retval = add_uevent_var(env, "OF_TYPE=%s", of_node_get_device_type(of->dev.of_node)); if (retval) return retval; -- 2.19.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda/ca0132 - Add new ZxR quirk
by Connor McAdams 19 Nov '18

19 Nov '18

This patch adds a new PCI subsys ID for the ZxR, as found and tested by other users. Without a way to know if any Z's use it as well, it keeps the quirk of QUIRK_SBZ and goes through the HDA subsys test function. Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 9c0b94ba786e..4c21bd24bcbd 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1189,6 +1189,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ), SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1102, 0x0033, "Sound Blaster ZxR", 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, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI), -- 2.18.1

2 3

[alsa-devel] [PATCH 1/5] ALSA: asihpi: clean up indentation, replace spaces with tab
by Colin King 19 Nov '18

19 Nov '18

From: Colin Ian King <colin.king(a)canonical.com> The struct declaration is not indented correctly. Fix this by replacing spaces with a tab. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/asihpi/asihpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index a31fe1550903..aad74e809797 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1183,7 +1183,7 @@ static int snd_card_asihpi_capture_prepare(struct snd_pcm_substream *substream) static u64 snd_card_asihpi_capture_formats(struct snd_card_asihpi *asihpi, u32 h_stream) { - struct hpi_format hpi_format; + struct hpi_format hpi_format; u16 format; u16 err; u32 h_control; -- 2.19.1

3 9

[alsa-devel] [PATCH v2 1/2] ASoC: ak4118: Add support for AK4118 S/PDIF transceiver
by Clément Péron 19 Nov '18

19 Nov '18

From: Adrien Charruel <adrien.charruel(a)devialet.com> The AK4118A is a digital audio transceiver supporting 8 input channels at 192kHz and with 24bits resolution. It converts the S/PDIF signal to I2S format and is configurable over I2C. This driver introduce a minimal support of the AK4118, like selecting the input channel, reading input frequency and detecting some errors. Datasheet is available here: https://www.akm.com/akm/en/file/datasheet/AK4118AEQ.pdf Signed-off-by: Adrien Charruel <adrien.charruel(a)devialet.com> Signed-off-by: Clément Péron <peron.clem(a)gmail.com> --- v2: - request threaded_irq in driver probe - change fs_enum from VALUE_SINGLE to VALUE_ENUM_SINGLE sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4118.c | 438 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 446 insertions(+) create mode 100644 sound/soc/codecs/ak4118.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9cc4f1848c9b..62bdb7e333b8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,6 +35,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU7002 select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER + select SND_SOC_AK4118 if I2C select SND_SOC_AK4458 if I2C select SND_SOC_AK4535 if I2C select SND_SOC_AK4554 @@ -392,6 +393,11 @@ config SND_SOC_AK4104 tristate "AKM AK4104 CODEC" depends on SPI_MASTER +config SND_SOC_AK4118 + tristate "AKM AK4118 CODEC" + depends on I2C + select REGMAP_I2C + config SND_SOC_AK4458 tristate "AKM AK4458 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8ffab8c8dbfa..66f55d185620 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -27,6 +27,7 @@ snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o +snd-soc-ak4118-objs := ak4118.o snd-soc-ak4458-objs := ak4458.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4554-objs := ak4554.o @@ -290,6 +291,7 @@ obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o +obj-$(CONFIG_SND_SOC_AK4118) += snd-soc-ak4118.o obj-$(CONFIG_SND_SOC_AK4458) += snd-soc-ak4458.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o diff --git a/sound/soc/codecs/ak4118.c b/sound/soc/codecs/ak4118.c new file mode 100644 index 000000000000..238ab29f2bf4 --- /dev/null +++ b/sound/soc/codecs/ak4118.c @@ -0,0 +1,438 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * ak4118.c -- Asahi Kasei ALSA Soc Audio driver + * + * Copyright 2018 DEVIALET + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/slab.h> + +#include <sound/asoundef.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#define AK4118_REG_CLK_PWR_CTL 0x00 +#define AK4118_REG_FORMAT_CTL 0x01 +#define AK4118_REG_IO_CTL0 0x02 +#define AK4118_REG_IO_CTL1 0x03 +#define AK4118_REG_INT0_MASK 0x04 +#define AK4118_REG_INT1_MASK 0x05 +#define AK4118_REG_RCV_STATUS0 0x06 +#define AK4118_REG_RCV_STATUS1 0x07 +#define AK4118_REG_RXCHAN_STATUS0 0x08 +#define AK4118_REG_RXCHAN_STATUS1 0x09 +#define AK4118_REG_RXCHAN_STATUS2 0x0a +#define AK4118_REG_RXCHAN_STATUS3 0x0b +#define AK4118_REG_RXCHAN_STATUS4 0x0c +#define AK4118_REG_TXCHAN_STATUS0 0x0d +#define AK4118_REG_TXCHAN_STATUS1 0x0e +#define AK4118_REG_TXCHAN_STATUS2 0x0f +#define AK4118_REG_TXCHAN_STATUS3 0x10 +#define AK4118_REG_TXCHAN_STATUS4 0x11 +#define AK4118_REG_BURST_PREAMB_PC0 0x12 +#define AK4118_REG_BURST_PREAMB_PC1 0x13 +#define AK4118_REG_BURST_PREAMB_PD0 0x14 +#define AK4118_REG_BURST_PREAMB_PD1 0x15 +#define AK4118_REG_QSUB_CTL 0x16 +#define AK4118_REG_QSUB_TRACK 0x17 +#define AK4118_REG_QSUB_INDEX 0x18 +#define AK4118_REG_QSUB_MIN 0x19 +#define AK4118_REG_QSUB_SEC 0x1a +#define AK4118_REG_QSUB_FRAME 0x1b +#define AK4118_REG_QSUB_ZERO 0x1c +#define AK4118_REG_QSUB_ABS_MIN 0x1d +#define AK4118_REG_QSUB_ABS_SEC 0x1e +#define AK4118_REG_QSUB_ABS_FRAME 0x1f +#define AK4118_REG_GPE 0x20 +#define AK4118_REG_GPDR 0x21 +#define AK4118_REG_GPSCR 0x22 +#define AK4118_REG_GPLR 0x23 +#define AK4118_REG_DAT_MASK_DTS 0x24 +#define AK4118_REG_RX_DETECT 0x25 +#define AK4118_REG_STC_DAT_DETECT 0x26 +#define AK4118_REG_RXCHAN_STATUS5 0x27 +#define AK4118_REG_TXCHAN_STATUS5 0x28 +#define AK4118_REG_MAX 0x29 + +#define AK4118_REG_FORMAT_CTL_DIF0 (1 << 4) +#define AK4118_REG_FORMAT_CTL_DIF1 (1 << 5) +#define AK4118_REG_FORMAT_CTL_DIF2 (1 << 6) + +struct ak4118_priv { + struct regmap *regmap; + struct gpio_desc *reset; + struct gpio_desc *irq; + struct snd_soc_component *component; +}; + +static const struct reg_default ak4118_reg_defaults[] = { + {AK4118_REG_CLK_PWR_CTL, 0x43}, + {AK4118_REG_FORMAT_CTL, 0x6a}, + {AK4118_REG_IO_CTL0, 0x88}, + {AK4118_REG_IO_CTL1, 0x48}, + {AK4118_REG_INT0_MASK, 0xee}, + {AK4118_REG_INT1_MASK, 0xb5}, + {AK4118_REG_RCV_STATUS0, 0x00}, + {AK4118_REG_RCV_STATUS1, 0x10}, + {AK4118_REG_TXCHAN_STATUS0, 0x00}, + {AK4118_REG_TXCHAN_STATUS1, 0x00}, + {AK4118_REG_TXCHAN_STATUS2, 0x00}, + {AK4118_REG_TXCHAN_STATUS3, 0x00}, + {AK4118_REG_TXCHAN_STATUS4, 0x00}, + {AK4118_REG_GPE, 0x77}, + {AK4118_REG_GPDR, 0x00}, + {AK4118_REG_GPSCR, 0x00}, + {AK4118_REG_GPLR, 0x00}, + {AK4118_REG_DAT_MASK_DTS, 0x3f}, + {AK4118_REG_RX_DETECT, 0x00}, + {AK4118_REG_STC_DAT_DETECT, 0x00}, + {AK4118_REG_TXCHAN_STATUS5, 0x00}, +}; + +static const char * const ak4118_input_select_txt[] = { + "RX0", "RX1", "RX2", "RX3", "RX4", "RX5", "RX6", "RX7", +}; +static SOC_ENUM_SINGLE_DECL(ak4118_insel_enum, AK4118_REG_IO_CTL1, 0x0, + ak4118_input_select_txt); + +static const struct snd_kcontrol_new ak4118_input_mux_controls = + SOC_DAPM_ENUM("Input Select", ak4118_insel_enum); + +static const char * const ak4118_iec958_fs_txt[] = { + "44100", "48000", "32000", "22050", "11025", "24000", "16000", "88200", + "8000", "96000", "64000", "176400", "192000", +}; + +static const int ak4118_iec958_fs_val[] = { + 0x0, 0x2, 0x3, 0x4, 0x5, 0x6, 0x7, 0x8, 0x9, 0xA, 0xB, 0xC, 0xE, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(ak4118_iec958_fs_enum, AK4118_REG_RCV_STATUS1, + 0x4, 0x4, ak4118_iec958_fs_txt, + ak4118_iec958_fs_val); + +static struct snd_kcontrol_new ak4118_iec958_controls[] = { + SOC_SINGLE("IEC958 Parity Errors", AK4118_REG_RCV_STATUS0, 0, 1, 0), + SOC_SINGLE("IEC958 No Audio", AK4118_REG_RCV_STATUS0, 1, 1, 0), + SOC_SINGLE("IEC958 PLL Lock", AK4118_REG_RCV_STATUS0, 4, 1, 1), + SOC_SINGLE("IEC958 Non PCM", AK4118_REG_RCV_STATUS0, 6, 1, 0), + SOC_ENUM("IEC958 Sampling Freq", ak4118_iec958_fs_enum), +}; + +static const struct snd_soc_dapm_widget ak4118_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("INRX0"), + SND_SOC_DAPM_INPUT("INRX1"), + SND_SOC_DAPM_INPUT("INRX2"), + SND_SOC_DAPM_INPUT("INRX3"), + SND_SOC_DAPM_INPUT("INRX4"), + SND_SOC_DAPM_INPUT("INRX5"), + SND_SOC_DAPM_INPUT("INRX6"), + SND_SOC_DAPM_INPUT("INRX7"), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ak4118_input_mux_controls), +}; + +static const struct snd_soc_dapm_route ak4118_dapm_routes[] = { + {"Input Mux", "RX0", "INRX0"}, + {"Input Mux", "RX1", "INRX1"}, + {"Input Mux", "RX2", "INRX2"}, + {"Input Mux", "RX3", "INRX3"}, + {"Input Mux", "RX4", "INRX4"}, + {"Input Mux", "RX5", "INRX5"}, + {"Input Mux", "RX6", "INRX6"}, + {"Input Mux", "RX7", "INRX7"}, +}; + + +static int ak4118_set_dai_fmt_master(struct ak4118_priv *ak4118, + unsigned int format) +{ + int dif; + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dif = AK4118_REG_FORMAT_CTL_DIF0 | AK4118_REG_FORMAT_CTL_DIF2; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dif = AK4118_REG_FORMAT_CTL_DIF0 | AK4118_REG_FORMAT_CTL_DIF1; + break; + case SND_SOC_DAIFMT_LEFT_J: + dif = AK4118_REG_FORMAT_CTL_DIF2; + break; + default: + return -ENOTSUPP; + } + + return dif; +} + +static int ak4118_set_dai_fmt_slave(struct ak4118_priv *ak4118, + unsigned int format) +{ + int dif; + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dif = AK4118_REG_FORMAT_CTL_DIF0 | AK4118_REG_FORMAT_CTL_DIF1 | + AK4118_REG_FORMAT_CTL_DIF2; + break; + case SND_SOC_DAIFMT_LEFT_J: + dif = AK4118_REG_FORMAT_CTL_DIF1 | AK4118_REG_FORMAT_CTL_DIF2; + break; + default: + return -ENOTSUPP; + } + + return dif; +} + +static int ak4118_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int format) +{ + struct snd_soc_component *component = dai->component; + struct ak4118_priv *ak4118 = snd_soc_component_get_drvdata(component); + int dif; + int ret = 0; + + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* component is master */ + dif = ak4118_set_dai_fmt_master(ak4118, format); + break; + case SND_SOC_DAIFMT_CBS_CFS: + /*component is slave */ + dif = ak4118_set_dai_fmt_slave(ak4118, format); + break; + default: + ret = -ENOTSUPP; + goto exit; + } + + /* format not supported */ + if (dif < 0) { + ret = dif; + goto exit; + } + + ret = regmap_update_bits(ak4118->regmap, AK4118_REG_FORMAT_CTL, + AK4118_REG_FORMAT_CTL_DIF0 | + AK4118_REG_FORMAT_CTL_DIF1 | + AK4118_REG_FORMAT_CTL_DIF2, dif); + if (ret < 0) + goto exit; + +exit: + return ret; +} + +static int ak4118_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + return 0; +} + +static const struct snd_soc_dai_ops ak4118_dai_ops = { + .hw_params = ak4118_hw_params, + .set_fmt = ak4118_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ak4118_dai = { + .name = "ak4118-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE + }, + .ops = &ak4118_dai_ops, +}; + +static irqreturn_t ak4118_irq_handler(int irq, void *data) +{ + struct ak4118_priv *ak4118 = data; + struct snd_soc_component *component = ak4118->component; + struct snd_kcontrol_new *kctl_new; + struct snd_kcontrol *kctl; + struct snd_ctl_elem_id *id; + unsigned int i; + + if (!component) + return IRQ_NONE; + + for (i = 0; i < ARRAY_SIZE(ak4118_iec958_controls); i++) { + kctl_new = &ak4118_iec958_controls[i]; + kctl = snd_soc_card_get_kcontrol(component->card, + kctl_new->name); + if (!kctl) + continue; + id = &kctl->id; + snd_ctl_notify(component->card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, id); + } + + return IRQ_HANDLED; +} + +static int ak4118_probe(struct snd_soc_component *component) +{ + struct ak4118_priv *ak4118 = snd_soc_component_get_drvdata(component); + int ret = 0; + + ak4118->component = component; + + /* release reset */ + gpiod_set_value(ak4118->reset, 0); + + /* unmask all int1 sources */ + ret = regmap_write(ak4118->regmap, AK4118_REG_INT1_MASK, 0x00); + if (ret < 0) { + dev_err(component->dev, + "failed to write regmap 0x%x 0x%x: %d\n", + AK4118_REG_INT1_MASK, 0x00, ret); + return ret; + } + + /* rx detect enable on all channels */ + ret = regmap_write(ak4118->regmap, AK4118_REG_RX_DETECT, 0xff); + if (ret < 0) { + dev_err(component->dev, + "failed to write regmap 0x%x 0x%x: %d\n", + AK4118_REG_RX_DETECT, 0xff, ret); + return ret; + } + + ret = snd_soc_add_component_controls(component, ak4118_iec958_controls, + ARRAY_SIZE(ak4118_iec958_controls)); + if (ret) { + dev_err(component->dev, + "failed to add component kcontrols: %d\n", ret); + return ret; + } + + return 0; +} + +static void ak4118_remove(struct snd_soc_component *component) +{ + struct ak4118_priv *ak4118 = snd_soc_component_get_drvdata(component); + + /* hold reset */ + gpiod_set_value(ak4118->reset, 1); +} + +static const struct snd_soc_component_driver soc_component_drv_ak4118 = { + .probe = ak4118_probe, + .remove = ak4118_remove, + .dapm_widgets = ak4118_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4118_dapm_widgets), + .dapm_routes = ak4118_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4118_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config ak4118_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = ak4118_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ak4118_reg_defaults), + + .cache_type = REGCACHE_NONE, + .max_register = AK4118_REG_MAX - 1, +}; + +static int ak4118_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ak4118_priv *ak4118; + int ret; + + ak4118 = devm_kzalloc(&i2c->dev, sizeof(struct ak4118_priv), + GFP_KERNEL); + if (ak4118 == NULL) + return -ENOMEM; + + ak4118->regmap = devm_regmap_init_i2c(i2c, &ak4118_regmap); + if (IS_ERR(ak4118->regmap)) + return PTR_ERR(ak4118->regmap); + + i2c_set_clientdata(i2c, ak4118); + + ak4118->reset = devm_gpiod_get(&i2c->dev, "reset", GPIOD_OUT_HIGH); + if (IS_ERR(ak4118->reset)) { + ret = PTR_ERR(ak4118->reset); + if (ret != -EPROBE_DEFER) + dev_err(&i2c->dev, "Failed to get reset: %d\n", ret); + return ret; + } + + ak4118->irq = devm_gpiod_get(&i2c->dev, "irq", GPIOD_IN); + if (IS_ERR(ak4118->irq)) { + ret = PTR_ERR(ak4118->irq); + if (ret != -EPROBE_DEFER) + dev_err(&i2c->dev, "Failed to get IRQ: %d\n", ret); + return ret; + } + + ret = devm_request_threaded_irq(&i2c->dev, gpiod_to_irq(ak4118->irq), + NULL, ak4118_irq_handler, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, + "ak4118-irq", ak4118); + if (ret < 0) { + dev_err(&i2c->dev, "Fail to request_irq: %d\n", ret); + return ret; + } + + return snd_soc_register_component(&i2c->dev, &soc_component_drv_ak4118, + &ak4118_dai, 1); +} + +static int ak4118_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_component(&i2c->dev); + return 0; +} + +static const struct of_device_id ak4118_of_match[] = { + { .compatible = "asahi-kasei,ak4118", }, + {} +}; +MODULE_DEVICE_TABLE(of, ak4118_of_match); + +static const struct i2c_device_id ak4118_id_table[] = { + { "ak4118", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, ak4118_id_table); + +static struct i2c_driver ak4118_i2c_driver = { + .driver = { + .name = "ak4118", + .of_match_table = of_match_ptr(ak4118_of_match), + }, + .id_table = ak4118_id_table, + .probe = ak4118_i2c_probe, + .remove = ak4118_i2c_remove, +}; + +module_i2c_driver(ak4118_i2c_driver); + +MODULE_DESCRIPTION("Asahi Kasei AK4118 ALSA SoC driver"); +MODULE_AUTHOR("Adrien Charruel <adrien.charruel(a)devialet.com>"); +MODULE_LICENSE("GPL"); -- 2.19.1

3 6

[alsa-devel] [PATCH v2] ASoC: rt5663: Add documentation for power supply support
by Cheng-Yi Chiang 17 Nov '18

17 Nov '18

rt5663 codec driver will support setting CPVDD and AVDD power supply from device tree. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Moved power supply properties to required properties. Documentation/devicetree/bindings/sound/rt5663.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt index 23386446c63d6..2a55e91334082 100644 --- a/Documentation/devicetree/bindings/sound/rt5663.txt +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -10,6 +10,10 @@ Required properties: - interrupts : The CODEC's interrupt output. +- avdd-supply: Power supply for AVDD, providing 1.8V. + +- cpvdd-supply: Power supply for CPVDD, providing 3.5V. + Optional properties: - "realtek,dc_offset_l_manual" @@ -51,4 +55,6 @@ rt5663: codec@12 { compatible = "realtek,rt5663"; reg = <0x12>; interrupts = <7 IRQ_TYPE_EDGE_FALLING>; + avdd-supply = <&pp1800_a_alc5662>; + cpvdd-supply = <&pp3500_a_alc5662>; }; -- 2.19.1.1215.g8438c0b245-goog

2 1

[alsa-devel] Applied "ASoC: Convert a few more users to using %pOFn instead of device_node.name" to the asoc tree
by Mark Brown 17 Nov '18

17 Nov '18

The patch ASoC: Convert a few more users to using %pOFn instead of device_node.name 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 dc43d3aa045c6588f163f4c0f5084e09cb5c0295 Mon Sep 17 00:00:00 2001 From: Rob Herring <robh(a)kernel.org> Date: Fri, 16 Nov 2018 15:43:49 -0600 Subject: [PATCH] ASoC: Convert a few more users to using %pOFn instead of device_node.name In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Olivier Moysan <olivier.moysan(a)st.com> Cc: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Cc: Maxime Coquelin <mcoquelin.stm32(a)gmail.com> Cc: Alexandre Torgue <alexandre.torgue(a)st.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-stm32(a)st-md-mailman.stormreply.com Cc: linux-arm-kernel(a)lists.infradead.org Signed-off-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/stm/stm32_sai.c | 8 ++++---- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 192f4d7b37b6..bff7d71d0742 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -828,7 +828,7 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) /* request irq */ irq_id = platform_get_irq(pdev, 0); if (!irq_id) { - dev_err(dev, "%s no irq found\n", dev->of_node->name); + dev_err(dev, "%pOFn no irq found\n", dev->of_node); return -ENXIO; } ret = devm_request_irq(dev, irq_id, mt6797_afe_irq_handler, diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index c0b6697503fd..166aed28330d 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -1092,7 +1092,7 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) irq_id = platform_get_irq(pdev, 0); if (irq_id <= 0) { - dev_err(afe->dev, "np %s no irq\n", afe->dev->of_node->name); + dev_err(afe->dev, "np %pOFn no irq\n", afe->dev->of_node); return irq_id < 0 ? irq_id : -ENXIO; } ret = devm_request_irq(afe->dev, irq_id, mt8173_afe_irq_handler, diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index d597eba61992..bcb35cae2a2c 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -74,14 +74,14 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) return ret; } - dev_dbg(&sai->pdev->dev, "Set %s%s as synchro provider\n", - sai->pdev->dev.of_node->name, + dev_dbg(&sai->pdev->dev, "Set %pOFn%s as synchro provider\n", + sai->pdev->dev.of_node, synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); prev_synco = FIELD_GET(SAI_GCR_SYNCOUT_MASK, readl_relaxed(sai->base)); if (prev_synco != STM_SAI_SYNC_OUT_NONE && synco != prev_synco) { - dev_err(&sai->pdev->dev, "%s%s already set as sync provider\n", - sai->pdev->dev.of_node->name, + dev_err(&sai->pdev->dev, "%pOFn%s already set as sync provider\n", + sai->pdev->dev.of_node, prev_synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); clk_disable_unprepare(sai->pclk); return -EINVAL; -- 2.19.1

1 0

[alsa-devel] [PATCH][next] ASoC: amd: fix spelling mistake "Inavlid" -> "Invalid"
by Colin King 17 Nov '18

17 Nov '18

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_err message. Fix this. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/raven/pci-acp3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index ef805e71a98f..c28457fd9b81 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -105,7 +105,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, } break; default: - dev_err(&pci->dev, "Inavlid ACP audio mode : %d\n", val); + dev_err(&pci->dev, "Invalid ACP audio mode : %d\n", val); ret = -ENODEV; goto unmap_mmio; } -- 2.19.1

2 1

[alsa-devel] [PATCH] ASoC: rt5663: Fix error handling of regulator_set_load
by Cheng-Yi Chiang 17 Nov '18

17 Nov '18

The default implementation of regulator_set_load returns REGULATOR_MODE_NORMAL, which is positive. rt5663_i2c_probe should only do error handling when return value of regulator_set_load is negative. In this case, rt5663_i2c_probe should return error. Also, consolidate err_irq into err_enable. Fix the missing goto for temporary regmap and rt5663->regmap. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- This patch is the fixup of the patch e81a2a6d12e85 ASoC: rt5663: Add regulator support sound/soc/codecs/rt5663.c | 21 ++++++++++++--------- 1 file changed, 12 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index bb2c1ee4a836f..473a54d1ae2c5 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3522,10 +3522,11 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, for (i = 0; i < ARRAY_SIZE(rt5663->supplies); i++) { ret = regulator_set_load(rt5663->supplies[i].consumer, RT5663_SUPPLY_CURRENT_UA); - if (ret) { + if (ret < 0) { dev_err(&i2c->dev, - "Failed to set regulator %s, ret: %d\n", + "Failed to set regulator load on %s, ret: %d\n", rt5663->supplies[i].supply, ret); + return ret; } } @@ -3543,7 +3544,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, ret = PTR_ERR(regmap); dev_err(&i2c->dev, "Failed to allocate temp register map: %d\n", ret); - return ret; + goto err_enable; } ret = regmap_read(regmap, RT5663_VENDOR_ID_2, &val); @@ -3576,7 +3577,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, ret = PTR_ERR(rt5663->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); - return ret; + goto err_enable; } /* reset and calibrate */ @@ -3686,17 +3687,19 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, rt5663_dai, ARRAY_SIZE(rt5663_dai)); if (ret) - goto err_irq; + goto err_enable; return 0; -err_irq: + + /* + * Error after enabling regulators should goto err_enable + * to disable regulators. + */ +err_enable: if (i2c->irq) free_irq(i2c->irq, rt5663); -err_enable: - dev_err(&i2c->dev, - "%s: Disable regulator after probe error\n", __func__); regulator_bulk_disable(ARRAY_SIZE(rt5663->supplies), rt5663->supplies); return ret; } -- 2.19.1.1215.g8438c0b245-goog

2 2

[alsa-devel] [PATCH v3 0/2] Introduce dmic mode switch delay parameter
by Jenny TC 17 Nov '18

17 Nov '18

Some DMICs need clock to be running for a specified duration as per the DMIC spec to complete the mode transition like sleep to normal, off to normal etc. This series add a mode-switch delay which will prevent the clock from being turned off without complying with manufacturer timing specifications. Also introduces a module parameter for the wakeup_delay parameter. v1: First patch set v2: 1). Removed ACPI device entry since a) The DMIC vendor doesn't have an ACPI vendor ID b) The ACPI enumeration creates one more device and conflicts with the existing skl.c dmic platform device registration. 2). Updated commit message v3: Updated commit message and introduced module param for wakeup_delay Jenny TC (2): ASoC: dmic: introduce mode switch delay ASoC: dmic: introduce module_param wakeup_delay sound/soc/codecs/dmic.c | 36 ++++++++++++++++++++++++++++++++++++ 1 file changed, 36 insertions(+) -- 1.9.1

4 5

[alsa-devel] [PATCH] ASoC: Convert a few more users to using %pOFn instead of device_node.name
by Rob Herring 16 Nov '18

16 Nov '18

In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Olivier Moysan <olivier.moysan(a)st.com> Cc: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Cc: Maxime Coquelin <mcoquelin.stm32(a)gmail.com> Cc: Alexandre Torgue <alexandre.torgue(a)st.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-stm32(a)st-md-mailman.stormreply.com Cc: linux-arm-kernel(a)lists.infradead.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/stm/stm32_sai.c | 8 ++++---- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 192f4d7b37b6..bff7d71d0742 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -828,7 +828,7 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) /* request irq */ irq_id = platform_get_irq(pdev, 0); if (!irq_id) { - dev_err(dev, "%s no irq found\n", dev->of_node->name); + dev_err(dev, "%pOFn no irq found\n", dev->of_node); return -ENXIO; } ret = devm_request_irq(dev, irq_id, mt6797_afe_irq_handler, diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index c0b6697503fd..166aed28330d 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -1092,7 +1092,7 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) irq_id = platform_get_irq(pdev, 0); if (irq_id <= 0) { - dev_err(afe->dev, "np %s no irq\n", afe->dev->of_node->name); + dev_err(afe->dev, "np %pOFn no irq\n", afe->dev->of_node); return irq_id < 0 ? irq_id : -ENXIO; } ret = devm_request_irq(afe->dev, irq_id, mt8173_afe_irq_handler, diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index d597eba61992..bcb35cae2a2c 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -74,14 +74,14 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) return ret; } - dev_dbg(&sai->pdev->dev, "Set %s%s as synchro provider\n", - sai->pdev->dev.of_node->name, + dev_dbg(&sai->pdev->dev, "Set %pOFn%s as synchro provider\n", + sai->pdev->dev.of_node, synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); prev_synco = FIELD_GET(SAI_GCR_SYNCOUT_MASK, readl_relaxed(sai->base)); if (prev_synco != STM_SAI_SYNC_OUT_NONE && synco != prev_synco) { - dev_err(&sai->pdev->dev, "%s%s already set as sync provider\n", - sai->pdev->dev.of_node->name, + dev_err(&sai->pdev->dev, "%pOFn%s already set as sync provider\n", + sai->pdev->dev.of_node, prev_synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); clk_disable_unprepare(sai->pclk); return -EINVAL; -- 2.19.1

1 0

Re: [alsa-devel] Controlling the Tascam FW-1884
by Takashi Sakamoto 16 Nov '18

16 Nov '18

Hi Scott, (C.Ced to alsa-devel, ffado-devel) On Sep 7 2018 23:01, Scott Bahling wrote: > I recently got my hands on a FW-1884 and would like to get it working in > Linux. I'm mostly interested in using the control interface which should > support several midi control interface protocols like Mackie Control or > Mackie HUI. From the user manual it appears that the host computer needs to > set the registers on the FW-1884 to switch modes (they reference Windows > applications). > > I see that you have recently been working on user space support asynchronous > commands to such devices. The hinawa_utils code has some beginnings of > support for setting options on the Tascam devices and I have just started > playing with that. Thanks for your suggestion. At present, the most of features on control surface of FW-1884 are not available, especially for fader, knob and start/stop buttons. In data transmission between system and devices on IEEE 1394 bus, both of asynchronous transactions and isochronous packet streams are available. In a case of TASCAM FireWire series, messages via MIDI physical interface is bypassed to asynchronous transaction, while messages generated on control surface are bypassed to isochronous packets, multiplexed with PCM frames. When enabling multiplexing messages to isochronous packet, demultiplexer should be implemented to ALSA firewire-tascam driver, but not yet. You can see my initial RFC in alsa-devel[1]. This patch enables userspace applications to perform the demultiplexing, however this idea easily brings cache-coherency problems due to page mapping between kernel/userspace and impossible to be applied. We need to investigate another idea to achieve the demultiplexing in kernel land. As another option, according to page 26 in 'TASCAM FW-1884 Owner's Manual', when programming Keys to port 1-4, system could receive MIDI messages from control surface of FW-1884. However, I've not investigated it. > How are you determining the registers and flags? Do you have documentation, > or are you reverse engineering? Do you know the registers and flags for > setting the control protocol? > > If you have any hints on how figure out the proper registers and flags I'm > happy to contribute to the work you have started and help support more > functions of the FW-1884. User manuals are documents to which we can refer. The most vendors licenses their drivers/software with EULA and in most cases it can't allow users to binary-based reverse engineering. All of my work are done with external protocol analyzer unit on IEEE 1394 bus. When working to implement drivers, I read/classified/parsed content of many packets sniffed by the analyzer. An initial idea of libhinawa/hinawa-utils is to assist this my work. I heard FFADO developers uses some ways for this type of work: - Usage of host controller of TI PCILynx and driver module (nosy.ko) - Windows application; bushound[2] However, this type of work takes you long time, so I don't necessarily recommend you. It's 2018 and I think you can find alternative control surfaces on USB. If you take long time to make devices available on GNU/Linux, it's better to select devices on 'live' buses instead of 'legacy' buses such as IEEE 1394 bus, IMO. In a case of USB, wireshark with usbcap is the easiest way for us to sniff packets on Windows. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/094817.html [2] http://perisoft.net/bushound/ Regards Takashi Sakamoto

2 21

[alsa-devel] [PATCH] ASoC: acpi: fix: continue searching when machine is ignored
by Keyon Jie 16 Nov '18

16 Nov '18

The machine_quirk may return NULL which means the acpi entries should be skipped and search for next matched entry is needed, here add return check here and continue for NULL case. Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> --- sound/soc/soc-acpi.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index b8e72b52db30..faf8941363e4 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -15,7 +15,10 @@ snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines) if (acpi_dev_present(mach->id, NULL, -1)) { if (mach->machine_quirk) mach = mach->machine_quirk(mach); - return mach; + + /* return matched machine, continue otherwise */ + if (mach) + return mach; } } return NULL; -- 2.17.1

2 4

[alsa-devel] How to propagate parameters from FE to BE with compressed audio interfaces
by Daniel Baluta 16 Nov '18

16 Nov '18

Hello, soc_compr_set_params_fe calls dpcm_be_dai_hw_params in order to setup BE hw_params. Is there a way to transfer parameters like for example the rate from FE to BE? In soc_compr_set_params_fe there is this code: * * Create an empty hw_params for the BE as the machine driver must * fix this up to match DSP decoder and ASRC configuration. * I.e. machine driver fixup for compressed BE is mandatory. */ memset(&fe->dpcm[fe_substream->stream].hw_params, 0, sizeof(struct snd_pcm_hw_params)); So, information about parameters is just ignored. The only way we can setup hw_params for BE is via be_hw_params_fixup() but this function doesn't have any information from FE. My use case is something like this: FE: - cpu: FSL SAI DAI interface - platform: DSP - codec: dummy codec BE: - cpu: dummy - platform: dummy - codec: cs42888. codec could handle 44100Hz and 48000Hz streams, but the BE doesn't have any way to know what is the rate cplay sets to FE. thanks, Daniel.

3 2

[alsa-devel] Applied "ASoC: Add documentation for pcm3060 property out-single-ended" to the asoc tree
by Mark Brown 16 Nov '18

16 Nov '18

The patch ASoC: Add documentation for pcm3060 property out-single-ended 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 fd7de6370cb62b147185834d60069568a21acacd Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Date: Thu, 15 Nov 2018 08:12:53 +0100 Subject: [PATCH] ASoC: Add documentation for pcm3060 property out-single-ended Output of pcm3060 codec may be configured as single-ended or differential Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/pcm3060.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/pcm3060.txt b/Documentation/devicetree/bindings/sound/pcm3060.txt index 90fcb8523099..97de66932d44 100644 --- a/Documentation/devicetree/bindings/sound/pcm3060.txt +++ b/Documentation/devicetree/bindings/sound/pcm3060.txt @@ -9,9 +9,15 @@ Required properties: - reg : the I2C address of the device for I2C, the chip select number for SPI. +Optional properties: + +- ti,out-single-ended: "true" if output is single-ended; + "false" or not specified if output is differential. + Examples: pcm3060: pcm3060@46 { compatible = "ti,pcm3060"; reg = <0x46>; + ti,out-single-ended = "true"; }; -- 2.19.1

1 0

[alsa-devel] [PATCH] ASoC: nau8822: convert to SPDX identifiers
by David Lin 15 Nov '18

15 Nov '18

This patch updates license to SPDX-License-Identifier instead of verbose license text. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- sound/soc/codecs/nau8822.c | 26 +++++++++++--------------- sound/soc/codecs/nau8822.h | 11 +++++------ 2 files changed, 16 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 622ce947f134..c6152a044416 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -1,18 +1,14 @@ -/* - * 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. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// nau8822.c -- NAU8822 ALSA Soc Audio driver +// +// Copyright 2017 Nuvoton Technology Crop. +// +// 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 #include <linux/module.h> #include <linux/moduleparam.h> diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index aa79c969cd44..d15afd782815 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -1,13 +1,12 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* - * nau8822.h -- NAU8822 Soc Audio Codec driver + * nau8822.h -- NAU8822 ALSA SoC Audio driver + * + * Copyright 2017 Nuvoton Technology Crop. * * Author: David Lin <ctlin0(a)nuvoton.com> - * Co-author: John Hsu <kchsu0(a)nuvoton.com> + * Co-author: John Hsu <kchsu(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__ -- 2.18.0

2 3

[alsa-devel] [PATCH 1/2] ASoC: amd: Fixed build errors
by Vijendar Mukunda 15 Nov '18

15 Nov '18

Fixed build errors. - Implicit declaration of pci_enable_msi() & pci_disable_msi() api's for openrisc architecture. - type defaults to 'int' in declaration of 'module_pci_driver' Enabled build for x86 architecture. Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 5f3b560..33ebec9 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -22,5 +22,6 @@ config SND_SOC_AMD_CZ_RT5645_MACH config SND_SOC_AMD_ACP3x tristate "AMD Audio Coprocessor-v3.x support" + depends on X86 && PCI help This option enables ACP v3.x I2S support on AMD platform -- 2.7.4

2 3

[alsa-devel] Applied "ASoC: amd: remove set but not used variable 'dma_buffer'" to the asoc tree
by Mark Brown 15 Nov '18

15 Nov '18

The patch ASoC: amd: remove set but not used variable 'dma_buffer' 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 424d5b56d020b750803134cc15bbd4d0addaaee7 Mon Sep 17 00:00:00 2001 From: YueHaibing <yuehaibing(a)huawei.com> Date: Thu, 15 Nov 2018 01:56:35 +0000 Subject: [PATCH] ASoC: amd: remove set but not used variable 'dma_buffer' Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/amd/raven/acp3x-pcm-dma.c: In function 'acp3x_dma_hw_params': sound/soc/amd/raven/acp3x-pcm-dma.c:333:25: warning: variable 'dma_buffer' set but not used [-Wunused-but-set-variable] It never used since introduction in commit 8de1b5ed0337 ("ASoC: amd: add acp3x system resume pm op") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 2aa34208ae92..022a8912c8a2 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -322,7 +322,6 @@ static int acp3x_dma_hw_params(struct snd_pcm_substream *substream, { int status; u64 size; - struct snd_dma_buffer *dma_buffer; struct page *pg; struct snd_pcm_runtime *runtime = substream->runtime; struct i2s_stream_instance *rtd = runtime->private_data; @@ -330,7 +329,6 @@ static int acp3x_dma_hw_params(struct snd_pcm_substream *substream, if (!rtd) return -EINVAL; - dma_buffer = &substream->dma_buffer; size = params_buffer_bytes(params); status = snd_pcm_lib_malloc_pages(substream, size); if (status < 0) -- 2.19.1

1 0

[alsa-devel] [PATCH] ASoC: pcm186x: Fix device reset-registers trigger value
by Andreas Dannenberg 15 Nov '18

15 Nov '18

According to the current device datasheet (TI Lit # SLAS831D, revised March 2018) the value written to the device's PAGE register to trigger a complete register reset should be 0xfe, not 0xff. So go ahead and update to the correct value. Reported-by: Stephane Le Provost <stephane.leprovost(a)mediatek.com> Tested-by: Stephane Le Provost <stephane.leprovost(a)mediatek.com> Signed-off-by: Andreas Dannenberg <dannenberg(a)ti.com> --- sound/soc/codecs/pcm186x.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm186x.h b/sound/soc/codecs/pcm186x.h index 2c6ba55bf394..bb3f0c42a1cd 100644 --- a/sound/soc/codecs/pcm186x.h +++ b/sound/soc/codecs/pcm186x.h @@ -139,7 +139,7 @@ enum pcm186x_type { #define PCM186X_MAX_REGISTER PCM186X_CURR_TRIM_CTRL /* PCM186X_PAGE */ -#define PCM186X_RESET 0xff +#define PCM186X_RESET 0xfe /* PCM186X_ADCX_INPUT_SEL_X */ #define PCM186X_ADC_INPUT_SEL_POL BIT(7) -- 2.15.1

3 2

[alsa-devel] Applied "ASoC: rt5663: Add regulator support" to the asoc tree
by Mark Brown 15 Nov '18

15 Nov '18

The patch ASoC: rt5663: Add regulator support 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 e81a2a6d12e85fbd7b19e96ad72b82a3cc8a6b2d Mon Sep 17 00:00:00 2001 From: Cheng-Yi Chiang <cychiang(a)chromium.org> Date: Thu, 15 Nov 2018 12:13:34 +0800 Subject: [PATCH] ASoC: rt5663: Add regulator support Add regulator support to turn on cpvdd and avdd in probe. If a regulator is not given from device tree, a dummy regulator will be used. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5663.c | 68 +++++++++++++++++++++++++++++++++++---- 1 file changed, 61 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 7eb2cbd39d6e..29c059ed0682 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -17,6 +17,7 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/acpi.h> +#include <linux/regulator/consumer.h> #include <linux/workqueue.h> #include <sound/core.h> #include <sound/pcm.h> @@ -33,6 +34,9 @@ #define RT5663_DEVICE_ID_2 0x6451 #define RT5663_DEVICE_ID_1 0x6406 +#define RT5663_POWER_ON_DELAY_MS 300 +#define RT5663_SUPPLY_CURRENT_UA 500000 + enum { CODEC_VER_1, CODEC_VER_0, @@ -48,6 +52,11 @@ struct impedance_mapping_table { unsigned int dc_offset_r_manual_mic; }; +static const char *const rt5663_supply_names[] = { + "avdd", + "cpvdd", +}; + struct rt5663_priv { struct snd_soc_component *component; struct rt5663_platform_data pdata; @@ -56,6 +65,7 @@ struct rt5663_priv { struct snd_soc_jack *hs_jack; struct timer_list btn_check_timer; struct impedance_mapping_table *imp_table; + struct regulator_bulk_data supplies[ARRAY_SIZE(rt5663_supply_names)]; int codec_ver; int sysclk; @@ -3483,7 +3493,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, { struct rt5663_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt5663_priv *rt5663; - int ret; + int ret, i; unsigned int val; struct regmap *regmap; @@ -3500,6 +3510,37 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, else rt5663_parse_dp(rt5663, &i2c->dev); + for (i = 0; i < ARRAY_SIZE(rt5663->supplies); i++) + rt5663->supplies[i].supply = rt5663_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(rt5663->supplies), + rt5663->supplies); + if (ret) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + /* Set load for regulator. */ + for (i = 0; i < ARRAY_SIZE(rt5663->supplies); i++) { + ret = regulator_set_load(rt5663->supplies[i].consumer, + RT5663_SUPPLY_CURRENT_UA); + if (ret) { + dev_err(&i2c->dev, + "Failed to set regulator %s, ret: %d\n", + rt5663->supplies[i].supply, ret); + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(rt5663->supplies), + rt5663->supplies); + + if (ret) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + msleep(RT5663_POWER_ON_DELAY_MS); + regmap = devm_regmap_init_i2c(i2c, &temp_regmap); if (IS_ERR(regmap)) { ret = PTR_ERR(regmap); @@ -3530,7 +3571,8 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Device with ID register %#x is not rt5663\n", val); - return -ENODEV; + ret = -ENODEV; + goto err_enable; } if (IS_ERR(rt5663->regmap)) { @@ -3635,20 +3677,30 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, ret = request_irq(i2c->irq, rt5663_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5663", rt5663); - if (ret) + if (ret) { dev_err(&i2c->dev, "%s Failed to reguest IRQ: %d\n", __func__, ret); + goto err_enable; + } } ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5663, rt5663_dai, ARRAY_SIZE(rt5663_dai)); - if (ret) { - if (i2c->irq) - free_irq(i2c->irq, rt5663); - } + if (ret) + goto err_irq; + return 0; + +err_irq: + if (i2c->irq) + free_irq(i2c->irq, rt5663); + +err_enable: + dev_err(&i2c->dev, + "%s: Disable regulator after probe error\n", __func__); + regulator_bulk_disable(ARRAY_SIZE(rt5663->supplies), rt5663->supplies); return ret; } @@ -3659,6 +3711,8 @@ static int rt5663_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5663); + regulator_bulk_disable(ARRAY_SIZE(rt5663->supplies), rt5663->supplies); + return 0; } -- 2.19.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: rt5663: Add documentation for power supply support
by Cheng-Yi Chiang 15 Nov '18

15 Nov '18

rt5663 codec driver will support setting CPVDD and AVDD power supply from device tree. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Fixed the commit title of v1. Documentation/devicetree/bindings/sound/rt5663.txt | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt index 23386446c63d6..d4058dfde0392 100644 --- a/Documentation/devicetree/bindings/sound/rt5663.txt +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -36,6 +36,9 @@ Optional properties: "realtek,impedance_sensing_num" is 2. It means that there are 2 ranges of impedance in the impedance sensing function. +- avdd-supply: Power supply for AVDD, providing 1.8V. +- cpvdd-supply: Power supply for CPVDD, providing 3.5V. + Pins on the device (for linking into audio routes) for RT5663: * IN1P @@ -51,4 +54,6 @@ rt5663: codec@12 { compatible = "realtek,rt5663"; reg = <0x12>; interrupts = <7 IRQ_TYPE_EDGE_FALLING>; + avdd-supply = <&pp1800_a_alc5662>; + cpvdd-supply = <&pp3500_a_alc5662>; }; -- 2.19.1.930.g4563a0d9d0-goog

2 2

[alsa-devel] [PATCH alsa-lib 1/2] pcm_share: Fix compiler warnings -Wunused-result
by Kirill Marinushkin 15 Nov '18

15 Nov '18

Before this commit, compilation of `pcm_share` causes warnings "ignoring return value" for several `read` and `write` operations: ~~~~ pcm_share.c: In function '_snd_pcm_share_missing': pcm_share.c:293:5: warning: ignoring return value of 'read', declared with attribute warn_unused_result [-Wunused-result] read(share->slave_socket, buf, 1); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ <...> ~~~~ However, ignoring return values in these use-cases is safe and optimal, because `read` and `write` operations are executed upon descriptors, created with `socketpair()` and `pipe()`. This commit fixes the warnings by introducing the return value with the attribute `unused` (for details, see [1]). The macro, used in this commit, is defined at `global.h`: ~~~~ define ATTRIBUTE_UNUSED __attribute__ ((__unused__)) ~~~~ I explicitly checked the objdump: this commit doesn't change the machine code, related to the modified C code lines. The execution performance is not affected. [1] https://gcc.gnu.org/onlinedocs/gcc-8.2.0/gcc/Common-Variable-Attributes.html Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> --- src/pcm/pcm_share.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/src/pcm/pcm_share.c b/src/pcm/pcm_share.c index 5a540c4f..9592e003 100644 --- a/src/pcm/pcm_share.c +++ b/src/pcm/pcm_share.c @@ -287,17 +287,18 @@ static snd_pcm_uframes_t _snd_pcm_share_missing(snd_pcm_t *pcm) update_poll: if (ready != share->ready) { + int un ATTRIBUTE_UNUSED; char buf[1]; if (pcm->stream == SND_PCM_STREAM_PLAYBACK) { if (ready) - read(share->slave_socket, buf, 1); + un = read(share->slave_socket, buf, 1); else - write(share->client_socket, buf, 1); + un = write(share->client_socket, buf, 1); } else { if (ready) - write(share->slave_socket, buf, 1); + un = write(share->slave_socket, buf, 1); else - read(share->client_socket, buf, 1); + un = read(share->client_socket, buf, 1); } share->ready = ready; } @@ -404,8 +405,9 @@ static void *snd_pcm_share_thread(void *data) err = poll(pfd, 2, -1); Pthread_mutex_lock(&slave->mutex); if (pfd[0].revents & POLLIN) { + int un ATTRIBUTE_UNUSED; char buf[1]; - read(pfd[0].fd, buf, 1); + un = read(pfd[0].fd, buf, 1); } } else { slave->polling = 0; -- 2.13.6

2 4

[alsa-devel] [PATCH 1/2] ASoC: ak4118: Add support for AK4118 S/PDIF transceiver
by Clément Péron 15 Nov '18

15 Nov '18

From: Adrien Charruel <adrien.charruel(a)devialet.com> The AK4118A is a digital audio transceiver supporting 8 input channels at 192kHz and with 24bits resolution. It converts the S/PDIF signal to I2S format and is configurable over I2C. This driver introduce a minimal support of the AK4118, like selecting the input channel, reading input frequency and detecting some errors. Datasheet is available here: https://www.akm.com/akm/en/file/datasheet/AK4118AEQ.pdf Signed-off-by: Adrien Charruel <adrien.charruel(a)devialet.com> Signed-off-by: Clément Péron <peron.clem(a)gmail.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4118.c | 449 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 457 insertions(+) create mode 100644 sound/soc/codecs/ak4118.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9cc4f1848c9b..62bdb7e333b8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,6 +35,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU7002 select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER + select SND_SOC_AK4118 if I2C select SND_SOC_AK4458 if I2C select SND_SOC_AK4535 if I2C select SND_SOC_AK4554 @@ -392,6 +393,11 @@ config SND_SOC_AK4104 tristate "AKM AK4104 CODEC" depends on SPI_MASTER +config SND_SOC_AK4118 + tristate "AKM AK4118 CODEC" + depends on I2C + select REGMAP_I2C + config SND_SOC_AK4458 tristate "AKM AK4458 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8ffab8c8dbfa..66f55d185620 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -27,6 +27,7 @@ snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o +snd-soc-ak4118-objs := ak4118.o snd-soc-ak4458-objs := ak4458.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4554-objs := ak4554.o @@ -290,6 +291,7 @@ obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o +obj-$(CONFIG_SND_SOC_AK4118) += snd-soc-ak4118.o obj-$(CONFIG_SND_SOC_AK4458) += snd-soc-ak4458.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o diff --git a/sound/soc/codecs/ak4118.c b/sound/soc/codecs/ak4118.c new file mode 100644 index 000000000000..a88e7d6a65a3 --- /dev/null +++ b/sound/soc/codecs/ak4118.c @@ -0,0 +1,449 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * ak4118.c -- Asahi Kasei ALSA Soc Audio driver + * + * Copyright 2018 DEVIALET + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/slab.h> + +#include <sound/asoundef.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#define AK4118_REG_CLK_PWR_CTL 0x00 +#define AK4118_REG_FORMAT_CTL 0x01 +#define AK4118_REG_IO_CTL0 0x02 +#define AK4118_REG_IO_CTL1 0x03 +#define AK4118_REG_INT0_MASK 0x04 +#define AK4118_REG_INT1_MASK 0x05 +#define AK4118_REG_RCV_STATUS0 0x06 +#define AK4118_REG_RCV_STATUS1 0x07 +#define AK4118_REG_RXCHAN_STATUS0 0x08 +#define AK4118_REG_RXCHAN_STATUS1 0x09 +#define AK4118_REG_RXCHAN_STATUS2 0x0a +#define AK4118_REG_RXCHAN_STATUS3 0x0b +#define AK4118_REG_RXCHAN_STATUS4 0x0c +#define AK4118_REG_TXCHAN_STATUS0 0x0d +#define AK4118_REG_TXCHAN_STATUS1 0x0e +#define AK4118_REG_TXCHAN_STATUS2 0x0f +#define AK4118_REG_TXCHAN_STATUS3 0x10 +#define AK4118_REG_TXCHAN_STATUS4 0x11 +#define AK4118_REG_BURST_PREAMB_PC0 0x12 +#define AK4118_REG_BURST_PREAMB_PC1 0x13 +#define AK4118_REG_BURST_PREAMB_PD0 0x14 +#define AK4118_REG_BURST_PREAMB_PD1 0x15 +#define AK4118_REG_QSUB_CTL 0x16 +#define AK4118_REG_QSUB_TRACK 0x17 +#define AK4118_REG_QSUB_INDEX 0x18 +#define AK4118_REG_QSUB_MIN 0x19 +#define AK4118_REG_QSUB_SEC 0x1a +#define AK4118_REG_QSUB_FRAME 0x1b +#define AK4118_REG_QSUB_ZERO 0x1c +#define AK4118_REG_QSUB_ABS_MIN 0x1d +#define AK4118_REG_QSUB_ABS_SEC 0x1e +#define AK4118_REG_QSUB_ABS_FRAME 0x1f +#define AK4118_REG_GPE 0x20 +#define AK4118_REG_GPDR 0x21 +#define AK4118_REG_GPSCR 0x22 +#define AK4118_REG_GPLR 0x23 +#define AK4118_REG_DAT_MASK_DTS 0x24 +#define AK4118_REG_RX_DETECT 0x25 +#define AK4118_REG_STC_DAT_DETECT 0x26 +#define AK4118_REG_RXCHAN_STATUS5 0x27 +#define AK4118_REG_TXCHAN_STATUS5 0x28 +#define AK4118_REG_MAX 0x29 + +#define AK4118_REG_FORMAT_CTL_DIF0 (1 << 4) +#define AK4118_REG_FORMAT_CTL_DIF1 (1 << 5) +#define AK4118_REG_FORMAT_CTL_DIF2 (1 << 6) + +struct ak4118_priv { + struct regmap *regmap; + struct gpio_desc *reset; + struct gpio_desc *irq; + struct snd_soc_component *component; +}; + +static const struct reg_default ak4118_reg_defaults[] = { + {AK4118_REG_CLK_PWR_CTL, 0x43}, + {AK4118_REG_FORMAT_CTL, 0x6a}, + {AK4118_REG_IO_CTL0, 0x88}, + {AK4118_REG_IO_CTL1, 0x48}, + {AK4118_REG_INT0_MASK, 0xee}, + {AK4118_REG_INT1_MASK, 0xb5}, + {AK4118_REG_RCV_STATUS0, 0x00}, + {AK4118_REG_RCV_STATUS1, 0x10}, + {AK4118_REG_TXCHAN_STATUS0, 0x00}, + {AK4118_REG_TXCHAN_STATUS1, 0x00}, + {AK4118_REG_TXCHAN_STATUS2, 0x00}, + {AK4118_REG_TXCHAN_STATUS3, 0x00}, + {AK4118_REG_TXCHAN_STATUS4, 0x00}, + {AK4118_REG_GPE, 0x77}, + {AK4118_REG_GPDR, 0x00}, + {AK4118_REG_GPSCR, 0x00}, + {AK4118_REG_GPLR, 0x00}, + {AK4118_REG_DAT_MASK_DTS, 0x3f}, + {AK4118_REG_RX_DETECT, 0x00}, + {AK4118_REG_STC_DAT_DETECT, 0x00}, + {AK4118_REG_TXCHAN_STATUS5, 0x00}, +}; + +static const char * const ak4118_input_select_txt[] = { + "RX0", "RX1", "RX2", "RX3", "RX4", "RX5", "RX6", "RX7", +}; +static SOC_ENUM_SINGLE_DECL(ak4118_insel_enum, AK4118_REG_IO_CTL1, 0, + ak4118_input_select_txt); + +static const struct snd_kcontrol_new ak4118_input_mux_controls = + SOC_DAPM_ENUM("Input Select", ak4118_insel_enum); + +static const char * const ak4118_iec958_fs_texts[] = { + "44100", + "Reserved", + "48000", + "32000", + "22050", + "11025", + "24000", + "16000", + "88200", + "8000", + "96000", + "64000", + "176400", + "Unused", + "192000", +}; + +static const struct soc_enum ak4118_iec958_fs_enum = + SOC_ENUM_SINGLE(AK4118_REG_RCV_STATUS1, 4, + ARRAY_SIZE(ak4118_iec958_fs_texts), + ak4118_iec958_fs_texts); + +static struct snd_kcontrol_new ak4118_iec958_controls[] = { + SOC_SINGLE("IEC958 Parity Errors", AK4118_REG_RCV_STATUS0, 0, 1, 0), + SOC_SINGLE("IEC958 No Audio", AK4118_REG_RCV_STATUS0, 1, 1, 0), + SOC_SINGLE("IEC958 PLL Lock", AK4118_REG_RCV_STATUS0, 4, 1, 1), + SOC_SINGLE("IEC958 Non PCM", AK4118_REG_RCV_STATUS0, 6, 1, 0), + SOC_ENUM("IEC958 Sampling Freq", ak4118_iec958_fs_enum), +}; + +static const struct snd_soc_dapm_widget ak4118_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("INRX0"), + SND_SOC_DAPM_INPUT("INRX1"), + SND_SOC_DAPM_INPUT("INRX2"), + SND_SOC_DAPM_INPUT("INRX3"), + SND_SOC_DAPM_INPUT("INRX4"), + SND_SOC_DAPM_INPUT("INRX5"), + SND_SOC_DAPM_INPUT("INRX6"), + SND_SOC_DAPM_INPUT("INRX7"), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ak4118_input_mux_controls), +}; + +static const struct snd_soc_dapm_route ak4118_dapm_routes[] = { + {"Input Mux", "RX0", "INRX0"}, + {"Input Mux", "RX1", "INRX1"}, + {"Input Mux", "RX2", "INRX2"}, + {"Input Mux", "RX3", "INRX3"}, + {"Input Mux", "RX4", "INRX4"}, + {"Input Mux", "RX5", "INRX5"}, + {"Input Mux", "RX6", "INRX6"}, + {"Input Mux", "RX7", "INRX7"}, +}; + + +static int ak4118_set_dai_fmt_master(struct ak4118_priv *ak4118, + unsigned int format) +{ + int dif; + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dif = AK4118_REG_FORMAT_CTL_DIF0 | AK4118_REG_FORMAT_CTL_DIF2; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dif = AK4118_REG_FORMAT_CTL_DIF0 | AK4118_REG_FORMAT_CTL_DIF1; + break; + case SND_SOC_DAIFMT_LEFT_J: + dif = AK4118_REG_FORMAT_CTL_DIF2; + break; + default: + return -ENOTSUPP; + } + + return dif; +} + +static int ak4118_set_dai_fmt_slave(struct ak4118_priv *ak4118, + unsigned int format) +{ + int dif; + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dif = AK4118_REG_FORMAT_CTL_DIF0 | AK4118_REG_FORMAT_CTL_DIF1 | + AK4118_REG_FORMAT_CTL_DIF2; + break; + case SND_SOC_DAIFMT_LEFT_J: + dif = AK4118_REG_FORMAT_CTL_DIF1 | AK4118_REG_FORMAT_CTL_DIF2; + break; + default: + return -ENOTSUPP; + } + + return dif; +} + +static int ak4118_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int format) +{ + struct snd_soc_component *component = dai->component; + struct ak4118_priv *ak4118 = snd_soc_component_get_drvdata(component); + int dif; + int ret = 0; + + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* component is master */ + dif = ak4118_set_dai_fmt_master(ak4118, format); + break; + case SND_SOC_DAIFMT_CBS_CFS: + /*component is slave */ + dif = ak4118_set_dai_fmt_slave(ak4118, format); + break; + default: + ret = -ENOTSUPP; + goto exit; + } + + /* format not supported */ + if (dif < 0) { + ret = dif; + goto exit; + } + + ret = regmap_update_bits(ak4118->regmap, AK4118_REG_FORMAT_CTL, + AK4118_REG_FORMAT_CTL_DIF0 | + AK4118_REG_FORMAT_CTL_DIF1 | + AK4118_REG_FORMAT_CTL_DIF2, dif); + if (ret < 0) + goto exit; + +exit: + return ret; +} + +static int ak4118_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + return 0; +} + +static const struct snd_soc_dai_ops ak4118_dai_ops = { + .hw_params = ak4118_hw_params, + .set_fmt = ak4118_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ak4118_dai = { + .name = "ak4118-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE + }, + .ops = &ak4118_dai_ops, +}; + +static irqreturn_t ak4118_irq_handler(int irq, void *data) +{ + struct ak4118_priv *ak4118 = data; + struct snd_soc_component *component = ak4118->component; + struct snd_kcontrol_new *kctl_new; + struct snd_kcontrol *kctl; + struct snd_ctl_elem_id *id; + unsigned int i; + + if (!component) + return IRQ_NONE; + + for (i = 0; i < ARRAY_SIZE(ak4118_iec958_controls); i++) { + kctl_new = &ak4118_iec958_controls[i]; + kctl = snd_soc_card_get_kcontrol(component->card, + kctl_new->name); + if (!kctl) + continue; + id = &kctl->id; + snd_ctl_notify(component->card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, id); + } + + return IRQ_HANDLED; +} + +static int ak4118_probe(struct snd_soc_component *component) +{ + struct ak4118_priv *ak4118 = snd_soc_component_get_drvdata(component); + int ret = 0; + + ak4118->component = component; + + /* release reset */ + gpiod_set_value(ak4118->reset, 0); + + /* unmask all int1 sources */ + ret = regmap_write(ak4118->regmap, AK4118_REG_INT1_MASK, 0x00); + if (ret < 0) { + dev_err(component->dev, + "failed to write regmap 0x%x 0x%x: %d\n", + AK4118_REG_INT1_MASK, 0x00, ret); + return ret; + } + + /* rx detect enable on all channels */ + ret = regmap_write(ak4118->regmap, AK4118_REG_RX_DETECT, 0xff); + if (ret < 0) { + dev_err(component->dev, + "failed to write regmap 0x%x 0x%x: %d\n", + AK4118_REG_RX_DETECT, 0xff, ret); + return ret; + } + + ret = snd_soc_add_component_controls(component, ak4118_iec958_controls, + ARRAY_SIZE(ak4118_iec958_controls)); + if (ret) { + dev_err(component->dev, + "failed to add component kcontrols: %d\n", ret); + return ret; + } + + ret = request_threaded_irq(gpiod_to_irq(ak4118->irq), NULL, + ak4118_irq_handler, IRQF_TRIGGER_RISING | + IRQF_ONESHOT, "ak4118-irq", ak4118); + if (ret < 0) { + dev_err(component->dev, "Fail to request_irq: %d\n", ret); + return ret; + } + + return 0; +} + +static void ak4118_remove(struct snd_soc_component *component) +{ + struct ak4118_priv *ak4118 = snd_soc_component_get_drvdata(component); + + /* hold reset */ + gpiod_set_value(ak4118->reset, 1); + + free_irq(gpiod_to_irq(ak4118->irq), ak4118); +} + +static const struct snd_soc_component_driver soc_component_drv_ak4118 = { + .probe = ak4118_probe, + .remove = ak4118_remove, + .dapm_widgets = ak4118_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4118_dapm_widgets), + .dapm_routes = ak4118_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4118_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config ak4118_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = ak4118_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ak4118_reg_defaults), + + .cache_type = REGCACHE_NONE, + .max_register = AK4118_REG_MAX - 1, +}; + +static int ak4118_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ak4118_priv *ak4118; + int ret; + + ak4118 = devm_kzalloc(&i2c->dev, sizeof(struct ak4118_priv), + GFP_KERNEL); + if (ak4118 == NULL) + return -ENOMEM; + + ak4118->regmap = devm_regmap_init_i2c(i2c, &ak4118_regmap); + if (IS_ERR(ak4118->regmap)) + return PTR_ERR(ak4118->regmap); + + i2c_set_clientdata(i2c, ak4118); + + ak4118->reset = devm_gpiod_get(&i2c->dev, "reset", GPIOD_OUT_HIGH); + if (IS_ERR(ak4118->reset)) { + ret = PTR_ERR(ak4118->reset); + if (ret != -EPROBE_DEFER) + dev_err(&i2c->dev, "Failed to get reset: %d\n", ret); + return ret; + } + + ak4118->irq = devm_gpiod_get(&i2c->dev, "irq", GPIOD_IN); + if (IS_ERR(ak4118->irq)) { + ret = PTR_ERR(ak4118->irq); + if (ret != -EPROBE_DEFER) + dev_err(&i2c->dev, "Failed to get IRQ: %d\n", ret); + return ret; + } + + return snd_soc_register_component(&i2c->dev, &soc_component_drv_ak4118, + &ak4118_dai, 1); +} + +static int ak4118_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_component(&i2c->dev); + return 0; +} + +static const struct of_device_id ak4118_of_match[] = { + { .compatible = "asahi-kasei,ak4118", }, + {} +}; +MODULE_DEVICE_TABLE(of, ak4118_of_match); + +static const struct i2c_device_id ak4118_id_table[] = { + { "ak4118", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, ak4118_id_table); + +static struct i2c_driver ak4118_i2c_driver = { + .driver = { + .name = "ak4118", + .of_match_table = of_match_ptr(ak4118_of_match), + }, + .id_table = ak4118_id_table, + .probe = ak4118_i2c_probe, + .remove = ak4118_i2c_remove, +}; + +module_i2c_driver(ak4118_i2c_driver); + +MODULE_DESCRIPTION("Asahi Kasei AK4118 ALSA SoC driver"); +MODULE_AUTHOR("Adrien Charruel <adrien.charruel(a)devialet.com>"); +MODULE_LICENSE("GPL"); -- 2.19.1

2 5

[alsa-devel] [RFC][PATCH] ASoC: rsnd: add SSIU BUSIF support
by Kuninori Morimoto 15 Nov '18

15 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Gen2 has BUSIF0-3, Gen3 has BUSIF0-7 on some SSIU. Current driver is assuming it is using BUSIF0 as default. Thus, SSI is attaching SSIU (with BUSIF0) by using rsnd_ssiu_attach(). But, TDM split mode also needs other BUSIF to use it. This patch adds missing SSIU BUSIFx support. BUSIF is handled by SSIU instead of SSI anymore. Thus, its settings no longer needed on SSI node on DT. This patch removes its settings from Document, but driver is still keeping compatibility. Thus, old DT style is still working. But, to avoid confusing, it doesn't indicate old compatiblity things on Document. New SoC should have SSIU on DT from this patch. 1) old style DT is still supported (= no rcar_sound,ssiu node on DT) 2) If ssiu is not indicated on playback/capture, BUSIF0 will be used as default playback = <&ssi3>; /* ssiu30 will be selected */ 3) you can select own ssiu playback = <&ssi32 &ssi3>; /* ssiu32 will be selected */ Not-yet-Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- > Jiada This patch supports BUSIF, you can select it on DT. Is this style OK for you ? For example playback = <&src0 &dvc0 &ssiu02 &ssi0>; ~~~~~~~ It is keeping compatiblity. no <&ssiu> is still OK playback = <&src0 &dvc0 &ssi0>; .../devicetree/bindings/sound/renesas,rsnd.txt | 78 ++++----- sound/soc/sh/rcar/core.c | 1 + sound/soc/sh/rcar/dma.c | 37 ++++- sound/soc/sh/rcar/rsnd.h | 6 +- sound/soc/sh/rcar/ssi.c | 22 +-- sound/soc/sh/rcar/ssiu.c | 178 +++++++++++++++++++-- 6 files changed, 255 insertions(+), 67 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index d92b705..8dfbe5b 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -363,6 +363,9 @@ Required properties: - rcar_sound,ssi : Should contain SSI feature. The number of SSI subnode should be same as HW. see below for detail. +- rcar_sound,ssiu : Should contain SSIU feature. + The number of SSIU subnode should be same as HW. + see below for detail. - rcar_sound,src : Should contain SRC feature. The number of SRC subnode should be same as HW. see below for detail. @@ -532,56 +535,55 @@ rcar_sound: sound@ec500000 { }; }; + rcar_sound,ssiu { + ssiu00: ssiu-0 { + dmas = <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx"; + }; + ssiu01: ssiu-1 { + dmas = <&audma0 0x35>, <&audma1 0x36>; + dma-names = "rx", "tx"; + }; + + ... + + ssiu95: ssiu-49 { + dmas = <&audma0 0xA5>, <&audma1 0xA6>; + dma-names = "rx", "tx"; + }; + ssiu96: ssiu-50 { + dmas = <&audma0 0xA7>, <&audma1 0xA8>; + dma-names = "rx", "tx"; + }; + ssiu97: ssiu-51 { + dmas = <&audma0 0xA9>, <&audma1 0xAA>; + dma-names = "rx", "tx"; + }; + }; + rcar_sound,ssi { ssi0: ssi-0 { interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x01>, <&audma1 0x02>, <&audma0 0x15>, <&audma1 0x16>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x01>, <&audma1 0x02>; + dma-names = "rx", "tx"; }; ssi1: ssi-1 { interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x03>, <&audma1 0x04>, <&audma0 0x49>, <&audma1 0x4a>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi2: ssi-2 { - interrupts = <0 372 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x05>, <&audma1 0x06>, <&audma0 0x63>, <&audma1 0x64>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi3: ssi-3 { - interrupts = <0 373 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x07>, <&audma1 0x08>, <&audma0 0x6f>, <&audma1 0x70>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi4: ssi-4 { - interrupts = <0 374 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x09>, <&audma1 0x0a>, <&audma0 0x71>, <&audma1 0x72>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi5: ssi-5 { - interrupts = <0 375 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0b>, <&audma1 0x0c>, <&audma0 0x73>, <&audma1 0x74>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi6: ssi-6 { - interrupts = <0 376 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0d>, <&audma1 0x0e>, <&audma0 0x75>, <&audma1 0x76>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi7: ssi-7 { - interrupts = <0 377 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0f>, <&audma1 0x10>, <&audma0 0x79>, <&audma1 0x7a>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x03>, <&audma1 0x04>; + dma-names = "rx", "tx"; }; + + ... + ssi8: ssi-8 { interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x11>, <&audma1 0x12>, <&audma0 0x7b>, <&audma1 0x7c>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x11>, <&audma1 0x12>; + dma-names = "rx", "tx"; }; ssi9: ssi-9 { interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x13>, <&audma1 0x14>, <&audma0 0x7d>, <&audma1 0x7e>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x13>, <&audma1 0x14>; + dma-names = "rx", "tx"; }; }; diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 5e10012..fff0a3a 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1121,6 +1121,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, break; rsnd_parse_connect_ssi(rdai, playback, capture); + rsnd_parse_connect_ssiu(rdai, playback, capture); rsnd_parse_connect_src(rdai, playback, capture); rsnd_parse_connect_ctu(rdai, playback, capture); rsnd_parse_connect_mix(rdai, playback, capture); diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 5daa6c9..0324a5c 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -218,7 +218,7 @@ struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, int i = 0; for_each_child_of_node(of_node, np) { - if (i == rsnd_mod_id(mod) && (!chan)) + if (i == rsnd_mod_id_raw(mod) && (!chan)) chan = of_dma_request_slave_channel(np, name); i++; } @@ -344,14 +344,16 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io, struct rsnd_mod *mod) { struct rsnd_mod *ssi = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *ssiu = rsnd_io_to_mod_ssiu(io); struct rsnd_mod *src = rsnd_io_to_mod_src(io); struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io); const u8 *entry = NULL; int id = 255; int size = 0; - if (mod == ssi) { - int busif = rsnd_ssi_get_busif(io); + if ((mod == ssi) || + (mod == ssiu)) { + int busif = rsnd_mod_id_sub(ssiu); entry = gen2_id_table_ssiu; size = ARRAY_SIZE(gen2_id_table_ssiu); @@ -530,13 +532,14 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io, struct device *dev = rsnd_priv_to_dev(priv); phys_addr_t ssi_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SSI); phys_addr_t src_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SCU); - int is_ssi = !!(rsnd_io_to_mod_ssi(io) == mod); + int is_ssi = !!(rsnd_io_to_mod_ssi(io) == mod) || + !!(rsnd_io_to_mod_ssiu(io) == mod); int use_src = !!rsnd_io_to_mod_src(io); int use_cmd = !!rsnd_io_to_mod_dvc(io) || !!rsnd_io_to_mod_mix(io) || !!rsnd_io_to_mod_ctu(io); int id = rsnd_mod_id(mod); - int busif = rsnd_ssi_get_busif(io); + int busif = rsnd_mod_id_sub(rsnd_io_to_mod_ssiu(io)); struct dma_addr { dma_addr_t out_addr; dma_addr_t in_addr; @@ -620,7 +623,7 @@ static void rsnd_dma_of_path(struct rsnd_mod *this, struct rsnd_mod **mod_from, struct rsnd_mod **mod_to) { - struct rsnd_mod *ssi = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *ssi; struct rsnd_mod *src = rsnd_io_to_mod_src(io); struct rsnd_mod *ctu = rsnd_io_to_mod_ctu(io); struct rsnd_mod *mix = rsnd_io_to_mod_mix(io); @@ -631,6 +634,28 @@ static void rsnd_dma_of_path(struct rsnd_mod *this, struct device *dev = rsnd_priv_to_dev(priv); int nr, i, idx; + /* + * It should use "rcar_sound,ssiu" on DT. + * But, we need to keep compatibility for old version. + * + * If it has "rcar_sound.ssiu", it will be used. + * If not, "rcar_sound.ssi" will be used. + * see + * rsnd_ssiu_dma_req() + * rsnd_ssi_dma_req() + */ + if (rsnd_ssiu_of_node(priv)) { + struct rsnd_mod *ssiu = rsnd_io_to_mod_ssiu(io); + + /* use SSIU */ + ssi = ssiu; + if (this == rsnd_io_to_mod_ssi(io)) + this = ssiu; + } else { + /* keep compatible, use SSI */ + ssi = rsnd_io_to_mod_ssi(io); + } + if (!ssi) return; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 28bd90a..1875fd9 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -441,6 +441,7 @@ int rsnd_runtime_is_ssi_tdm(struct rsnd_dai_stream *io); of_get_child_by_name(rsnd_priv_to_dev(priv)->of_node, node) #define RSND_NODE_DAI "rcar_sound,dai" #define RSND_NODE_SSI "rcar_sound,ssi" +#define RSND_NODE_SSIU "rcar_sound,ssiu" #define RSND_NODE_SRC "rcar_sound,src" #define RSND_NODE_CTU "rcar_sound,ctu" #define RSND_NODE_MIX "rcar_sound,mix" @@ -715,7 +716,6 @@ void rsnd_ssi_remove(struct rsnd_priv *priv); struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id); int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod); int rsnd_ssi_use_busif(struct rsnd_dai_stream *io); -int rsnd_ssi_get_busif(struct rsnd_dai_stream *io); u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io); #define RSND_SSI_HDMI_PORT0 0xf0 @@ -743,6 +743,10 @@ int rsnd_ssiu_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod); int rsnd_ssiu_probe(struct rsnd_priv *priv); void rsnd_ssiu_remove(struct rsnd_priv *priv); +void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, + struct device_node *playback, + struct device_node *capture); +#define rsnd_ssiu_of_node(priv) rsnd_parse_of_node(priv, RSND_NODE_SSIU) /* * R-Car SRC diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index a032819..95acc8b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -150,11 +150,6 @@ int rsnd_ssi_use_busif(struct rsnd_dai_stream *io) return use_busif; } -int rsnd_ssi_get_busif(struct rsnd_dai_stream *io) -{ - return 0; /* BUSIF0 only for now */ -} - static void rsnd_ssi_status_clear(struct rsnd_mod *mod) { rsnd_mod_write(mod, SSISR, 0); @@ -761,7 +756,7 @@ static int rsnd_ssi_common_probe(struct rsnd_mod *mod, { struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - int ret; + int ret = 0; /* * SSIP/SSIU/IRQ are not needed on @@ -775,10 +770,6 @@ static int rsnd_ssi_common_probe(struct rsnd_mod *mod, * see rsnd_ssi_pcm_new() */ - ret = rsnd_ssiu_attach(io, mod); - if (ret < 0) - return ret; - /* * SSI might be called again as PIO fallback * It is easy to manual handling for IRQ request/free @@ -972,6 +963,17 @@ static struct dma_chan *rsnd_ssi_dma_req(struct rsnd_dai_stream *io, int is_play = rsnd_io_is_play(io); char *name; + /* + * It should use "rcar_sound,ssiu" on DT. + * But, we need to keep compatibility for old version. + * + * If it has "rcar_sound.ssiu", it will be used. + * If not, "rcar_sound.ssi" will be used. + * see + * rsnd_ssiu_dma_req() + * rsnd_dma_of_path() + */ + if (rsnd_ssi_use_busif(io)) name = is_play ? "rxu" : "txu"; else diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index ebcb826..20f7b04 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -12,6 +12,8 @@ struct rsnd_ssiu { struct rsnd_mod mod; u32 busif_status[8]; /* for BUSIF0 - BUSIF7 */ unsigned int usrcnt; + int id; + int id_sub; }; #define rsnd_ssiu_nr(priv) ((priv)->ssiu_nr) @@ -22,12 +24,29 @@ struct rsnd_ssiu { ((pos) = ((struct rsnd_ssiu *)(priv)->ssiu + i)); \ i++) +/* + * SSI Gen2 Gen3 + * 0 BUSIF0-3 BUSIF0-7 + * 1 BUSIF0-3 BUSIF0-7 + * 2 BUSIF0-3 BUSIF0-7 + * 3 BUSIF0 BUSIF0-7 + * 4 BUSIF0 BUSIF0-7 + * 5 BUSIF0 BUSIF0 + * 6 BUSIF0 BUSIF0 + * 7 BUSIF0 BUSIF0 + * 8 BUSIF0 BUSIF0 + * 9 BUSIF0-3 BUSIF0-7 + * total 22 52 + */ +static const int gen2_id[] = { 0, 4, 8, 12, 13, 14, 15, 16, 17, 18 }; +static const int gen3_id[] = { 0, 8, 16, 24, 32, 40, 41, 42, 43, 44 }; + static u32 *rsnd_ssiu_get_status(struct rsnd_mod *mod, struct rsnd_dai_stream *io, enum rsnd_mod_type type) { struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); - int busif = rsnd_ssi_get_busif(io); + int busif = rsnd_mod_id_sub(mod); return &ssiu->busif_status[busif]; } @@ -158,7 +177,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, if (rsnd_ssi_use_busif(io)) { int id = rsnd_mod_id(mod); - int busif = rsnd_ssi_get_busif(io); + int busif = rsnd_mod_id_sub(mod); /* * FIXME @@ -255,7 +274,7 @@ static int rsnd_ssiu_start_gen2(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - int busif = rsnd_ssi_get_busif(io); + int busif = rsnd_mod_id_sub(mod); if (!rsnd_ssi_use_busif(io)) return 0; @@ -273,7 +292,7 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); - int busif = rsnd_ssi_get_busif(io); + int busif = rsnd_mod_id_sub(mod); if (!rsnd_ssi_use_busif(io)) return 0; @@ -289,8 +308,49 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod, return 0; } +static int rsnd_ssiu_id(struct rsnd_mod *mod) +{ + struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); + + /* see rsnd_ssiu_probe() */ + return ssiu->id; +} + +static int rsnd_ssiu_id_sub(struct rsnd_mod *mod) +{ + struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); + + /* see rsnd_ssiu_probe() */ + return ssiu->id_sub; +} + +static struct dma_chan *rsnd_ssiu_dma_req(struct rsnd_dai_stream *io, + struct rsnd_mod *mod) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + int is_play = rsnd_io_is_play(io); + char *name; + + /* + * It should use "rcar_sound,ssiu" on DT. + * But, we need to keep compatibility for old version. + * + * If it has "rcar_sound.ssiu", it will be used. + * If not, "rcar_sound.ssi" will be used. + * see + * rsnd_ssi_dma_req() + * rsnd_dma_of_path() + */ + + name = is_play ? "rx" : "tx"; + + return rsnd_dma_request_channel(rsnd_ssiu_of_node(priv), + mod, name); +} + static struct rsnd_mod_ops rsnd_ssiu_ops_gen2 = { .name = SSIU_NAME, + .dma_req = rsnd_ssiu_dma_req, .init = rsnd_ssiu_init_gen2, .start = rsnd_ssiu_start_gen2, .stop = rsnd_ssiu_stop_gen2, @@ -305,26 +365,83 @@ static struct rsnd_mod *rsnd_ssiu_mod_get(struct rsnd_priv *priv, int id) return rsnd_mod_get((struct rsnd_ssiu *)(priv->ssiu) + id); } -int rsnd_ssiu_attach(struct rsnd_dai_stream *io, - struct rsnd_mod *ssi_mod) +static void rsnd_parse_connect_ssiu_compatible(struct rsnd_priv *priv, + struct rsnd_dai_stream *io) +{ + struct rsnd_mod *ssi_mod = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *mod; + struct rsnd_ssiu *ssiu; + int i; + + if (!ssi_mod) + return; + + /* select BUSIF0 */ + for_each_rsnd_ssiu(ssiu, priv, i) { + mod = rsnd_mod_get(ssiu); + + if ((rsnd_mod_id(ssi_mod) == rsnd_ssiu_id(mod)) && + (rsnd_mod_id_sub(mod) == 0)) + rsnd_dai_connect(mod, io, mod->type); + } +} + +void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, + struct device_node *playback, + struct device_node *capture) { - struct rsnd_priv *priv = rsnd_io_to_priv(io); - struct rsnd_mod *mod = rsnd_ssiu_mod_get(priv, rsnd_mod_id(ssi_mod)); + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device_node *node = rsnd_ssiu_of_node(priv); + struct device_node *np; + struct rsnd_mod *mod; + struct rsnd_dai_stream *io_p = &rdai->playback; + struct rsnd_dai_stream *io_c = &rdai->capture; + int i; - rsnd_mod_confirm_ssi(ssi_mod); + /* use rcar_sound,ssiu if exist */ + if (node) { + i = 0; + for_each_child_of_node(node, np) { + mod = rsnd_ssiu_mod_get(priv, i); + if (np == playback) + rsnd_dai_connect(mod, io_p, mod->type); + if (np == capture) + rsnd_dai_connect(mod, io_c, mod->type); + i++; + } - return rsnd_dai_connect(mod, io, mod->type); + of_node_put(node); + } + + /* Keep DT compatibility */ + if (!rsnd_io_to_mod_ssiu(io_p)) + rsnd_parse_connect_ssiu_compatible(priv, io_p); + if (!rsnd_io_to_mod_ssiu(io_c)) + rsnd_parse_connect_ssiu_compatible(priv, io_c); } int rsnd_ssiu_probe(struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); + struct device_node *node; struct rsnd_ssiu *ssiu; struct rsnd_mod_ops *ops; + const int *list = NULL; int i, nr, ret; - /* same number to SSI */ - nr = priv->ssi_nr; + /* + * Keep DT compatibility. + * if it has "rcar_sound,ssiu", use it. + * if not, use "rcar_sound,ssi" + * see + * rsnd_ssiu_bufsif_to_id() + */ + node = rsnd_ssiu_of_node(priv); + if (node) + nr = of_get_child_count(node); + else + nr = priv->ssi_nr; + ssiu = devm_kcalloc(dev, nr, sizeof(*ssiu), GFP_KERNEL); if (!ssiu) return -ENOMEM; @@ -337,7 +454,44 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv) else ops = &rsnd_ssiu_ops_gen2; + /* Keep compatibility */ + nr = 0; + if ((node) && + (ops == &rsnd_ssiu_ops_gen2)) { + ops->id = rsnd_ssiu_id; + ops->id_sub = rsnd_ssiu_id_sub; + + if (rsnd_is_gen2(priv)) { + list = gen2_id; + nr = ARRAY_SIZE(gen2_id); + } else if (rsnd_is_gen3(priv)) { + list = gen3_id; + nr = ARRAY_SIZE(gen3_id); + } else { + dev_err(dev, "unknown SSIU\n"); + return -ENODEV; + } + } + for_each_rsnd_ssiu(ssiu, priv, i) { + if (node) { + int j; + + /* + * see + * rsnd_ssiu_get_id() + * rsnd_ssiu_get_id_sub() + */ + for (j = 0; j < nr; j++) { + if (list[j] > i) + break; + ssiu->id = j; + ssiu->id_sub = i - list[ssiu->id]; + } + } else { + ssiu->id = i; + } + ret = rsnd_mod_init(priv, rsnd_mod_get(ssiu), ops, NULL, RSND_MOD_SSIU, i); if (ret) -- 2.7.4

2 2

Re: [alsa-devel] [EXTERNAL] Re: [PATCH 1/2] ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier
by Schulman, James 15 Nov '18

15 Nov '18

On Tue, 13 Nov 2018, Mark Brown wrote: > On Tue, Nov 13, 2018 at 12:49:16PM -0600, James Schulman wrote: > >> +++ b/sound/soc/codecs/cs35l36-tables.c >> @@ -0,0 +1,315 @@ >> +/* >> + * cs35l36-tables.c -- CS35L36 ALSA SoC audio driver >> + * >> + * Copyright 2018 Cirrus Logic, Inc. > > Please use SPDX headers. > Will do. >> + SOC_SINGLE_TLV("AMP PCM Gain", CS35L36_AMP_GAIN_CTRL, 5, 0x13, 0, >> + amp_gain_tlv), > > All volume controls should end in Volume, see control-names.rst. > >> + regmap_read(cs35l36->regmap, CS35L36_INT4_RAW_STATUS, &reg); >> + if (reg & CS35L36_PLL_UNLOCK_MASK) >> + dev_crit(cs35l36->dev, "PLL Unlocked\n"); > > WARN_ON_ONCE? > Will do. >> + } >> + return 0; >> +} >> +static const char * const cs35l36_chan_text[] = { > > Missing blank line. > Will do. >> +static const char * const cs35l36_boost_text[] = { >> + "On", >> + "Off", >> +}; >> + >> +static SOC_ENUM_SINGLE_DECL(boost_enum, SND_SOC_NOPM, 0, >> + cs35l36_boost_text); >> + >> +static const struct snd_kcontrol_new cs35l36_boost_mux[] = { >> + SOC_DAPM_ENUM("Boost Enable", boost_enum), >> +}; > > Simple on/off conttrols should be _SINGLE controls with Switch at the > end of their name. It's not super clear why this is in DAPM as a widget > at all, the routing around it is: > > + {"BOOST Mux", "On", "Channel Mux"}, > + {"CLASS H", NULL, "BOOST Mux"}, > > which is a bit perplexing. Possibly this should be handled in the event > for the main amplifier? > We used _ENUM instead of _SINGLE because we wanted the default value to be "On" instead of "Off". We don't want the default assumption to have boost turned off. However, we're happy to change this if you would rather have consitency with the _SINGLE controls. >> + if (cs35l36->sclk > 6000000) { >> + fs1_val = 3 * 4 + 4; >> + fs2_val = 8 * 4 + 4; >> + } > > These are just constants, why write out the operations? > Will do. >> + >> + if (cs35l36->sclk <= 6000000) { > > if/else? > Will do. >> + /* >> + * Rev B0 has 2 versions >> + * L36 is 10V >> + * L37 is 12V > > \o/ > Yes we had 2 silicon spins for this revision >_< is there anything about this comment you want us to change? >> + } >> + >> + return IRQ_HANDLED; > > This unconditionally reports that it handled the interrupt, even if it > didn't. This will stop the interrupt core handling for faulty interrupt > lines working and will disrupt any support that gets added for handling > shared threaded interrupts. > Earlier in the handler, we have a check to see if any unmasked bits are set. If not we return IRQ_NONE. I figured this would be enough to ensure we don't unconditionally report IRQ_HANDLED. >> + if (pdata) >> + cs35l36->pdata = *pdata; >> + else { > > Use { } on both sides of the if (see coding-style.rst)? > Will do. >> + cs35l36->reset_gpio = devm_gpiod_get_optional(dev, "reset", >> + GPIOD_OUT_LOW); >> + if (IS_ERR(cs35l36->reset_gpio)) { >> + ret = PTR_ERR(cs35l36->reset_gpio); >> + cs35l36->reset_gpio = NULL; >> + if (ret == -EBUSY) >> + dev_info(dev, >> + "Reset line busy, assuming shared reset\n"); >> + else { >> + dev_err(dev, "Failed to get reset GPIO: %d\n", ret); >> + goto err; >> + } >> + } > > This doesn't handle -EPROBE_DEFER. > I think we will add an err_disable_regs jump statement to ensure we don't try to use the uninitialized reset_gpio handle. Once we do that, I think this will handle -EPROBE_DEFER cleanly? Thanks for the comments Mark. - James

2 3

[alsa-devel] [PATCH] ASoC: dapm: Recalculate audio map forcely when card instantiated
by Tzung-Bi Shih 15 Nov '18

15 Nov '18

Audio map are possible in wrong state before card->instantiated has been set to true. Imaging the following examples: time 1: at the beginning in:-1 in:-1 in:-1 in:-1 out:-1 out:-1 out:-1 out:-1 SIGGEN A B Spk time 2: after someone called snd_soc_dapm_new_widgets() (e.g. create_fill_widget_route_map() in sound/soc/codecs/hdac_hdmi.c) in:1 in:0 in:0 in:0 out:0 out:0 out:0 out:1 SIGGEN A B Spk time 3: routes added in:1 in:0 in:0 in:0 out:0 out:0 out:0 out:1 SIGGEN -----> A -----> B ---> Spk In the end, the path should be powered on but it did not. At time 3, "in" of SIGGEN and "out" of Spk did not propagate to their neighbors because snd_soc_dapm_add_path() will not invalidate the paths if the card has not instantiated (i.e. card->instantiated is false). To correct the state of audio map, recalculate the whole map forcely. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6ddcf12bc030..b29d0f65611e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2131,6 +2131,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } card->instantiated = 1; + dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); -- 2.19.1.930.g4563a0d9d0-goog

2 1

[alsa-devel] Applied "ASoC: pcm3060: Add DT property for single-ended output" to the asoc tree
by Mark Brown 15 Nov '18

15 Nov '18

The patch ASoC: pcm3060: Add DT property for single-ended output 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 9d4b17efa51f3bb0918021090c09b0013039df00 Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Date: Mon, 12 Nov 2018 08:08:33 +0100 Subject: [PATCH] ASoC: pcm3060: Add DT property for single-ended output DAC output may be differential (default) or single-ended. Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm3060.c | 14 ++++++++++++++ sound/soc/codecs/pcm3060.h | 1 + 2 files changed, 15 insertions(+) diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 494d9d662be8..5f479aa61097 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -274,9 +274,23 @@ EXPORT_SYMBOL(pcm3060_regmap); /* device */ +static void pcm3060_parse_dt(const struct device_node *np, + struct pcm3060_priv *priv) +{ + priv->out_se = of_property_read_bool(np, "ti,out-single-ended"); +} + int pcm3060_probe(struct device *dev) { int rc; + struct pcm3060_priv *priv = dev_get_drvdata(dev); + + if (dev->of_node) + pcm3060_parse_dt(dev->of_node, priv); + + if (priv->out_se) + regmap_update_bits(priv->regmap, PCM3060_REG64, + PCM3060_REG_SE, PCM3060_REG_SE); rc = devm_snd_soc_register_component(dev, &pcm3060_soc_comp_driver, pcm3060_dai, diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h index fd89a68aa8a7..c895cf40ee10 100644 --- a/sound/soc/codecs/pcm3060.h +++ b/sound/soc/codecs/pcm3060.h @@ -25,6 +25,7 @@ struct pcm3060_priv_dai { struct pcm3060_priv { struct regmap *regmap; struct pcm3060_priv_dai dai[PCM3060_DAI_IDS_NUM]; + u8 out_se: 1; }; int pcm3060_probe(struct device *dev); -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: amd: fix memory leak of i2s_data on error return" to the asoc tree
by Mark Brown 15 Nov '18

15 Nov '18

The patch ASoC: amd: fix memory leak of i2s_data on error return 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 46dce4042659755a2b9753369d0c2d8ca33e35cf Mon Sep 17 00:00:00 2001 From: Colin Ian King <colin.king(a)canonical.com> Date: Wed, 14 Nov 2018 21:31:48 +0000 Subject: [PATCH] ASoC: amd: fix memory leak of i2s_data on error return Currently when snd_pcm_hw_constraint_integer fails there is a memory leak of i2s_data on the error return path. Fix this by kfree'ing i2s_data before returning. Detected by CoverityScan, CID#1475479 ("Resource leak") Fixes: 0b87d6bcd648 ("ASoC: amd: add acp3x pcm driver dma ops") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 2e61cef00f41..b16cdd187139 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -309,6 +309,7 @@ static int acp3x_dma_open(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) { dev_err(component->dev, "set integer constraint failed\n"); + kfree(i2s_data); return ret; } -- 2.19.1

1 0

[alsa-devel] [PATCH] ASoC: omap-abe-twl6040: Fix missing audio card caused by deferred probing
by Peter Ujfalusi 15 Nov '18

15 Nov '18

The machine driver fails to probe in next-20181113 with: [ 2.539093] omap-abe-twl6040 sound: ASoC: CODEC DAI twl6040-legacy not registered [ 2.546630] omap-abe-twl6040 sound: devm_snd_soc_register_card() failed: -517 ... [ 3.693206] omap-abe-twl6040 sound: ASoC: Both platform name/of_node are set for TWL6040 [ 3.701446] omap-abe-twl6040 sound: ASoC: failed to init link TWL6040 [ 3.708007] omap-abe-twl6040 sound: devm_snd_soc_register_card() failed: -22 [ 3.715148] omap-abe-twl6040: probe of sound failed with error -22 Bisect pointed to a merge commit: first bad commit: [0f688ab20a540aafa984c5dbd68a71debebf4d7f] Merge remote-tracking branch 'net-next/master' and a diff between a working kernel does not reveal anything which would explain the change in behavior. Further investigation showed that on the second try of loading fails because the dai_link->platform is no longer NULL and it might be pointing to uninitialized memory. The fix is to move the snd_soc_dai_link and snd_soc_card inside of the abe_twl6040 struct, which is dynamically allocated every time the driver probes. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-abe-twl6040.c | 67 +++++++++++++------------------ 1 file changed, 29 insertions(+), 38 deletions(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index d5ae9eb8c756..fed45b41f9d3 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -36,6 +36,8 @@ #include "../codecs/twl6040.h" struct abe_twl6040 { + struct snd_soc_card card; + struct snd_soc_dai_link dai_links[2]; int jack_detection; /* board can detect jack events */ int mclk_freq; /* MCLK frequency speed for twl6040 */ }; @@ -208,40 +210,10 @@ static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(dmic_audio_map)); } -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link abe_twl6040_dai_links[] = { - { - .name = "TWL6040", - .stream_name = "TWL6040", - .codec_dai_name = "twl6040-legacy", - .codec_name = "twl6040-codec", - .init = omap_abe_twl6040_init, - .ops = &omap_abe_ops, - }, - { - .name = "DMIC", - .stream_name = "DMIC Capture", - .codec_dai_name = "dmic-hifi", - .codec_name = "dmic-codec", - .init = omap_abe_dmic_init, - .ops = &omap_abe_dmic_ops, - }, -}; - -/* Audio machine driver */ -static struct snd_soc_card omap_abe_card = { - .owner = THIS_MODULE, - - .dapm_widgets = twl6040_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - static int omap_abe_probe(struct platform_device *pdev) { struct device_node *node = pdev->dev.of_node; - struct snd_soc_card *card = &omap_abe_card; + struct snd_soc_card *card; struct device_node *dai_node; struct abe_twl6040 *priv; int num_links = 0; @@ -252,12 +224,18 @@ static int omap_abe_probe(struct platform_device *pdev) return -ENODEV; } - card->dev = &pdev->dev; - priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); if (priv == NULL) return -ENOMEM; + card = &priv->card; + card->dev = &pdev->dev; + card->owner = THIS_MODULE; + card->dapm_widgets = twl6040_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets); + card->dapm_routes = audio_map; + card->num_dapm_routes = ARRAY_SIZE(audio_map); + if (snd_soc_of_parse_card_name(card, "ti,model")) { dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; @@ -274,14 +252,27 @@ static int omap_abe_probe(struct platform_device *pdev) dev_err(&pdev->dev, "McPDM node is not provided\n"); return -EINVAL; } - abe_twl6040_dai_links[0].cpu_of_node = dai_node; - abe_twl6040_dai_links[0].platform_of_node = dai_node; + + priv->dai_links[0].name = "DMIC"; + priv->dai_links[0].stream_name = "TWL6040"; + priv->dai_links[0].cpu_of_node = dai_node; + priv->dai_links[0].platform_of_node = dai_node; + priv->dai_links[0].codec_dai_name = "twl6040-legacy"; + priv->dai_links[0].codec_name = "twl6040-codec"; + priv->dai_links[0].init = omap_abe_twl6040_init; + priv->dai_links[0].ops = &omap_abe_ops; dai_node = of_parse_phandle(node, "ti,dmic", 0); if (dai_node) { num_links = 2; - abe_twl6040_dai_links[1].cpu_of_node = dai_node; - abe_twl6040_dai_links[1].platform_of_node = dai_node; + priv->dai_links[1].name = "TWL6040"; + priv->dai_links[1].stream_name = "DMIC Capture"; + priv->dai_links[1].cpu_of_node = dai_node; + priv->dai_links[1].platform_of_node = dai_node; + priv->dai_links[1].codec_dai_name = "dmic-hifi"; + priv->dai_links[1].codec_name = "dmic-codec"; + priv->dai_links[1].init = omap_abe_dmic_init; + priv->dai_links[1].ops = &omap_abe_dmic_ops; } else { num_links = 1; } @@ -300,7 +291,7 @@ static int omap_abe_probe(struct platform_device *pdev) return -ENODEV; } - card->dai_link = abe_twl6040_dai_links; + card->dai_link = priv->dai_links; card->num_links = num_links; snd_soc_card_set_drvdata(card, priv); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH][next] ASoC: amd: fix memory leak of i2s_data on error return
by Colin King 14 Nov '18

14 Nov '18

From: Colin Ian King <colin.king(a)canonical.com> Currently when snd_pcm_hw_constraint_integer fails there is a memory leak of i2s_data on the error return path. Fix this by kfree'ing i2s_data before returning. Detected by CoverityScan, CID#1475479 ("Resource leak") Fixes: 0b87d6bcd648 ("ASoC: amd: add acp3x pcm driver dma ops") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 2e61cef00f41..b16cdd187139 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -309,6 +309,7 @@ static int acp3x_dma_open(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) { dev_err(component->dev, "set integer constraint failed\n"); + kfree(i2s_data); return ret; } -- 2.19.1

1 0

Re: [alsa-devel] linux-next: Tree for Nov 14 (sound/soc/amd/raven/)
by Randy Dunlap 14 Nov '18

14 Nov '18

On 11/13/18 9:26 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20181113: > on i386 or x86_64: when CONFIG_PCI is not enabled. ../sound/soc/amd/raven/pci-acp3x.c: In function 'snd_acp3x_probe': ../sound/soc/amd/raven/pci-acp3x.c:59:2: error: implicit declaration of function 'pci_enable_msi' [-Werror=implicit-function-declaration] ret = pci_enable_msi(pci); ^ ../sound/soc/amd/raven/pci-acp3x.c:124:2: error: implicit declaration of function 'pci_disable_msi' [-Werror=implicit-function-declaration] pci_disable_msi(pci); ^ ../sound/soc/amd/raven/pci-acp3x.c: At top level: ../sound/soc/amd/raven/pci-acp3x.c:161:1: warning: data definition has no type or storage class [enabled by default] module_pci_driver(acp3x_driver); ^ ../sound/soc/amd/raven/pci-acp3x.c:161:1: error: type defaults to 'int' in declaration of 'module_pci_driver' [-Werror=implicit-int] ../sound/soc/amd/raven/pci-acp3x.c:161:1: warning: parameter names (without types) in function declaration [enabled by default] ../sound/soc/amd/raven/pci-acp3x.c:154:26: warning: 'acp3x_driver' defined but not used [-Wunused-variable] static struct pci_driver acp3x_driver = { ^ config SND_SOC_AMD_ACP3x tristate "AMD Audio Coprocessor-v3.x support" help This option enables ACP v3.x I2S support on AMD platform Needs "depends on PCI"?? -- ~Randy

1 0

[alsa-devel] alsa-utils: axfer: rewrite aplay, adding 'timer-based scheduling' option
by GitHub pull_request - opened 14 Nov '18

14 Nov '18

alsa-project/alsa-utils pull request #3 was opened from takaswie: Hi, This patchset is an updated version of my previous RFCv3, and for merging to upstream. [alsa-devel] [RFCv3][PATCH 00/39] axfer: rewrite aplay, adding 'timer-based scheduling' option http://mailman.alsa-project.org/pipermail/alsa-devel/2017-October/125961.ht… As I noted in RFCv3, some patches are dropped in this patchset, due to some technical issues. In detail, please refer to RFCv3. Changes from RFCv3: - rename scheduling option to 'sched-model' - improve waiter implementation and add support for select(2) - categorize supported options as common and backend-dependent - release alsa-lib configuration space at the end of runtime - valgrind test and fix some memory leaks - add 'frame-cache' internal interface to share implementation by several backends. - add 'libffado' backend just for technical preview (linkage of this library brings problems. Please refet to patch comment.) I apologize but in this time I don't prepare any help document and manuals as well as RFCv3. Let me put them to a part of task for this development period. Regards Takashi Sakamoto (35): axfer: add an entry point for this command axfer: add a sub-command to print list of PCMs/devices axfer: add a common interface to handle a file with audio-specific data format axfer: add support for a container of Microsoft/IBM RIFF/Wave format axfer: add support for a container of Sparc AU format axfer: add support for a container of Creative Tech. voice format axfer: add support for a container of raw data axfer: add unit test for container interface axfer: add a common interface to align data frames on different layout axfer: add support for a mapper for single target axfer: add support for a mapper for multiple target axfer: add a unit test for mapper interface axfer: add a common interface to transfer data frames axfer: add a parser for command-line options axfer: add support to transfer data frames by alsa-lib PCM APIs axfer: add support for blocking data transmission operation of alsa-lib PCM API axfer: add a sub-command to transfer data frames axfer: add informative output and an option to suppress it axfer: add an option to dump available hardware parameters axfer: add options related to duration and obsolete '--max-file-size' option axfer: add an option to finish transmission at XRUN axfer: add support for non-blocking operation axfer: add support for MMAP PCM operation axfer: add an option to suppress event waiting axfer: add options for buffer arrangement axfer: add options for software parameters of PCM substream axfer: add options for plugins in alsa-lib axfer: add a common interface of waiter for I/O event notification axfer: add an option for waiter type axfer: add an implementation of waiter for poll(2) axfer: add an implementation of waiter for select(2) axfer: add an implementation of waiter for epoll(7) axfer: add support for timer-based scheduling model with MMAP operation axfer: obsolete some unimplemented options axfer: add support for libffado transmission backend ``` Makefile.am | 2 +- axfer/Makefile.am | 62 +++ axfer/container-au.c | 203 +++++++ axfer/container-raw.c | 66 +++ axfer/container-riff-wave.c | 575 +++++++++++++++++++ axfer/container-voc.c | 833 +++++++++++++++++++++++++++ axfer/container.c | 457 +++++++++++++++ axfer/container.h | 126 +++++ axfer/frame-cache.c | 111 ++++ axfer/frame-cache.h | 47 ++ axfer/main.c | 229 ++++++++ axfer/mapper-multiple.c | 259 +++++++++ axfer/mapper-single.c | 191 +++++++ axfer/mapper.c | 150 +++++ axfer/mapper.h | 87 +++ axfer/misc.h | 19 + axfer/subcmd-list.c | 234 ++++++++ axfer/subcmd-transfer.c | 467 ++++++++++++++++ axfer/subcmd.h | 18 + axfer/test/Makefile.am | 31 ++ axfer/test/container-test.c | 299 ++++++++++ axfer/test/generator.c | 260 +++++++++ axfer/test/generator.h | 47 ++ axfer/test/mapper-test.c | 491 ++++++++++++++++ axfer/waiter-epoll.c | 107 ++++ axfer/waiter-poll.c | 45 ++ axfer/waiter-select.c | 100 ++++ axfer/waiter.c | 104 ++++ axfer/waiter.h | 61 ++ axfer/xfer-libasound-irq-mmap.c | 282 ++++++++++ axfer/xfer-libasound-irq-rw.c | 427 ++++++++++++++ axfer/xfer-libasound-timer-mmap.c | 455 +++++++++++++++ axfer/xfer-libasound.c | 898 ++++++++++++++++++++++++++++++ axfer/xfer-libasound.h | 92 +++ axfer/xfer-libffado.c | 555 ++++++++++++++++++ axfer/xfer-options.c | 589 ++++++++++++++++++++ axfer/xfer.c | 254 +++++++++ axfer/xfer.h | 115 ++++ configure.ac | 15 +- 39 files changed, 9361 insertions(+), 2 deletions(-) create mode 100644 axfer/Makefile.am create mode 100644 axfer/container-au.c create mode 100644 axfer/container-raw.c create mode 100644 axfer/container-riff-wave.c create mode 100644 axfer/container-voc.c create mode 100644 axfer/container.c create mode 100644 axfer/container.h create mode 100644 axfer/frame-cache.c create mode 100644 axfer/frame-cache.h create mode 100644 axfer/main.c create mode 100644 axfer/mapper-multiple.c create mode 100644 axfer/mapper-single.c create mode 100644 axfer/mapper.c create mode 100644 axfer/mapper.h create mode 100644 axfer/misc.h create mode 100644 axfer/subcmd-list.c create mode 100644 axfer/subcmd-transfer.c create mode 100644 axfer/subcmd.h create mode 100644 axfer/test/Makefile.am create mode 100644 axfer/test/container-test.c create mode 100644 axfer/test/generator.c create mode 100644 axfer/test/generator.h create mode 100644 axfer/test/mapper-test.c create mode 100644 axfer/waiter-epoll.c create mode 100644 axfer/waiter-poll.c create mode 100644 axfer/waiter-select.c create mode 100644 axfer/waiter.c create mode 100644 axfer/waiter.h create mode 100644 axfer/xfer-libasound-irq-mmap.c create mode 100644 axfer/xfer-libasound-irq-rw.c create mode 100644 axfer/xfer-libasound-timer-mmap.c create mode 100644 axfer/xfer-libasound.c create mode 100644 axfer/xfer-libasound.h create mode 100644 axfer/xfer-libffado.c create mode 100644 axfer/xfer-options.c create mode 100644 axfer/xfer.c create mode 100644 axfer/xfer.h ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/3 Patch URL : https://github.com/alsa-project/alsa-utils/pull/3.patch Repository URL: https://github.com/alsa-project/alsa-utils

3 38

[alsa-devel] GitHub - alsa-project - repositories
by Jaroslav Kysela 14 Nov '18

14 Nov '18

Hi all, URL: https://github.com/alsa-project I finally finished the first phase of the integration with GitHub. All repositories are now on GitHub and all repositories are mirrored to alsa-project.org. The github repositories are master (developers should push changes only to those repositories). If you refer the repository as the code source (for packaging or so), please, keep to use the repositories on alsa-project.org (in case when we migrate to other service in future like GitLab or so). The mirroring is realtime, so the changes should be visible on git.alsa-project.org in few seconds after the push. I invited few people to the GitHub team and actually Takashi has full access to all repos, Vinod Koul should have the write access to tinycompress and Takashi Sakamoto should have the write access to alsa-gi. If I omitted someone (or someone is not on github), please, let me know (and register before). Because github adds possibility for the pull requests and issue tracking, I added notifications for them to this (alsa-devel) mailing list. The short notification should be sent when an pull request or an issue is opened or changed. I think that it would be best to handle this per request, so the developer can ask to resend the patch to the mailing list for the wider review or just push the change with the signing. I activated Travis CI for alsa-lib and alsa-utils and I will add other repos soon, too. URL: https://travis-ci.org/alsa-project If you have some ideas which other github applications can be used to improve the code maintenance, let me know. I will probably play with the coverity checker (http://scan.coverity.com), too. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

5 10

[alsa-devel] [PATCH 01/11] ASoC: AMD: add ACP 3.x IP register header
by Vijendar Mukunda 14 Nov '18

14 Nov '18

From: Maruthi Srinivas Bayyavarapu <Maruthi.Bayyavarapu(a)amd.com> ACP 3.x is a new audio block in raven. Added register header of the same. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/chip_offset_byte.h | 655 +++++++++++++++++++++++++++++++++ 1 file changed, 655 insertions(+) create mode 100644 sound/soc/amd/raven/chip_offset_byte.h diff --git a/sound/soc/amd/raven/chip_offset_byte.h b/sound/soc/amd/raven/chip_offset_byte.h new file mode 100644 index 0000000..fceb7bb --- /dev/null +++ b/sound/soc/amd/raven/chip_offset_byte.h @@ -0,0 +1,655 @@ +/* + * ACP 3.0 Register documentation + * + * Copyright (C) 2016 Advanced Micro Devices, Inc. + * + * Permission is hereby granted, free of charge, to any person obtaining a + * copy of this software and associated documentation files (the "Software"), + * to deal in the Software without restriction, including without limitation + * the rights to use, copy, modify, merge, publish, distribute, sublicense, + * and/or sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included + * in all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS + * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE COPYRIGHT HOLDER(S) BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR + * IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + */ + +#ifndef _acp_ip_OFFSET_HEADER +#define _acp_ip_OFFSET_HEADER +// Registers from ACP_DMA block + +#define mmACP_DMA_CNTL_0 0x1240000 +#define mmACP_DMA_CNTL_1 0x1240004 +#define mmACP_DMA_CNTL_2 0x1240008 +#define mmACP_DMA_CNTL_3 0x124000C +#define mmACP_DMA_CNTL_4 0x1240010 +#define mmACP_DMA_CNTL_5 0x1240014 +#define mmACP_DMA_CNTL_6 0x1240018 +#define mmACP_DMA_CNTL_7 0x124001C +#define mmACP_DMA_DSCR_STRT_IDX_0 0x1240020 +#define mmACP_DMA_DSCR_STRT_IDX_1 0x1240024 +#define mmACP_DMA_DSCR_STRT_IDX_2 0x1240028 +#define mmACP_DMA_DSCR_STRT_IDX_3 0x124002C +#define mmACP_DMA_DSCR_STRT_IDX_4 0x1240030 +#define mmACP_DMA_DSCR_STRT_IDX_5 0x1240034 +#define mmACP_DMA_DSCR_STRT_IDX_6 0x1240038 +#define mmACP_DMA_DSCR_STRT_IDX_7 0x124003C +#define mmACP_DMA_DSCR_CNT_0 0x1240040 +#define mmACP_DMA_DSCR_CNT_1 0x1240044 +#define mmACP_DMA_DSCR_CNT_2 0x1240048 +#define mmACP_DMA_DSCR_CNT_3 0x124004C +#define mmACP_DMA_DSCR_CNT_4 0x1240050 +#define mmACP_DMA_DSCR_CNT_5 0x1240054 +#define mmACP_DMA_DSCR_CNT_6 0x1240058 +#define mmACP_DMA_DSCR_CNT_7 0x124005C +#define mmACP_DMA_PRIO_0 0x1240060 +#define mmACP_DMA_PRIO_1 0x1240064 +#define mmACP_DMA_PRIO_2 0x1240068 +#define mmACP_DMA_PRIO_3 0x124006C +#define mmACP_DMA_PRIO_4 0x1240070 +#define mmACP_DMA_PRIO_5 0x1240074 +#define mmACP_DMA_PRIO_6 0x1240078 +#define mmACP_DMA_PRIO_7 0x124007C +#define mmACP_DMA_CUR_DSCR_0 0x1240080 +#define mmACP_DMA_CUR_DSCR_1 0x1240084 +#define mmACP_DMA_CUR_DSCR_2 0x1240088 +#define mmACP_DMA_CUR_DSCR_3 0x124008C +#define mmACP_DMA_CUR_DSCR_4 0x1240090 +#define mmACP_DMA_CUR_DSCR_5 0x1240094 +#define mmACP_DMA_CUR_DSCR_6 0x1240098 +#define mmACP_DMA_CUR_DSCR_7 0x124009C +#define mmACP_DMA_CUR_TRANS_CNT_0 0x12400A0 +#define mmACP_DMA_CUR_TRANS_CNT_1 0x12400A4 +#define mmACP_DMA_CUR_TRANS_CNT_2 0x12400A8 +#define mmACP_DMA_CUR_TRANS_CNT_3 0x12400AC +#define mmACP_DMA_CUR_TRANS_CNT_4 0x12400B0 +#define mmACP_DMA_CUR_TRANS_CNT_5 0x12400B4 +#define mmACP_DMA_CUR_TRANS_CNT_6 0x12400B8 +#define mmACP_DMA_CUR_TRANS_CNT_7 0x12400BC +#define mmACP_DMA_ERR_STS_0 0x12400C0 +#define mmACP_DMA_ERR_STS_1 0x12400C4 +#define mmACP_DMA_ERR_STS_2 0x12400C8 +#define mmACP_DMA_ERR_STS_3 0x12400CC +#define mmACP_DMA_ERR_STS_4 0x12400D0 +#define mmACP_DMA_ERR_STS_5 0x12400D4 +#define mmACP_DMA_ERR_STS_6 0x12400D8 +#define mmACP_DMA_ERR_STS_7 0x12400DC +#define mmACP_DMA_DESC_BASE_ADDR 0x12400E0 +#define mmACP_DMA_DESC_MAX_NUM_DSCR 0x12400E4 +#define mmACP_DMA_CH_STS 0x12400E8 +#define mmACP_DMA_CH_GROUP 0x12400EC +#define mmACP_DMA_CH_RST_STS 0x12400F0 + + +// Registers from ACP_AXI2AXIATU block + +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_1 0x1240C00 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_1 0x1240C04 +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_2 0x1240C08 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_2 0x1240C0C +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_3 0x1240C10 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_3 0x1240C14 +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_4 0x1240C18 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_4 0x1240C1C +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_5 0x1240C20 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_5 0x1240C24 +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_6 0x1240C28 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_6 0x1240C2C +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_7 0x1240C30 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_7 0x1240C34 +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_8 0x1240C38 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_8 0x1240C3C +#define mmACPAXI2AXI_ATU_CTRL 0x1240C40 + + +// Registers from ACP_CLKRST block + +#define mmACP_SOFT_RESET 0x1241000 +#define mmACP_CONTROL 0x1241004 +#define mmACP_STATUS 0x1241008 +#define mmACP_DSP0_OCD_HALT_ON_RST 0x124100C +#define mmACP_DYNAMIC_CG_MASTER_CONTROL 0x1241010 + + +// Registers from ACP_MISC block + +#define mmACP_EXTERNAL_INTR_ENB 0x1241800 +#define mmACP_EXTERNAL_INTR_CNTL 0x1241804 +#define mmACP_EXTERNAL_INTR_STAT 0x1241808 +#define mmACP_DSP0_INTR_CNTL 0x124180C +#define mmACP_DSP0_INTR_STAT 0x1241810 +#define mmACP_DSP_SW_INTR_CNTL 0x1241814 +#define mmACP_DSP_SW_INTR_STAT 0x1241818 +#define mmACP_SW_INTR_TRIG 0x124181C +#define mmACP_SMU_MAILBOX 0x1241820 +#define mmDSP_INTERRUPT_ROUTING_CTRL 0x1241824 +#define mmACP_DSP0_WATCHDOG_TIMER_CNTL 0x1241828 +#define mmACP_DSP0_EXT_TIMER1_CNTL 0x124182C +#define mmACP_DSP0_EXT_TIMER2_CNTL 0x1241830 +#define mmACP_DSP0_EXT_TIMER3_CNTL 0x1241834 +#define mmACP_DSP0_EXT_TIMER4_CNTL 0x1241838 +#define mmACP_DSP0_EXT_TIMER5_CNTL 0x124183C +#define mmACP_DSP0_EXT_TIMER6_CNTL 0x1241840 +#define mmACP_DSP0_EXT_TIMER1_CURR_VALUE 0x1241844 +#define mmACP_DSP0_EXT_TIMER2_CURR_VALUE 0x1241848 +#define mmACP_DSP0_EXT_TIMER3_CURR_VALUE 0x124184C +#define mmACP_DSP0_EXT_TIMER4_CURR_VALUE 0x1241850 +#define mmACP_DSP0_EXT_TIMER5_CURR_VALUE 0x1241854 +#define mmACP_DSP0_EXT_TIMER6_CURR_VALUE 0x1241858 +#define mmACP_FW_STATUS 0x124185C +#define mmACP_TIMER 0x1241874 +#define mmACP_TIMER_CNTL 0x1241878 +#define mmACP_PGMEM_CTRL 0x12418C0 +#define mmACP_ERROR_STATUS 0x12418C4 +#define mmACP_SW_I2S_ERROR_REASON 0x12418C8 +#define mmACP_MEM_PG_STS 0x12418CC + + +// Registers from ACP_PGFSM block + +#define mmACP_I2S_PIN_CONFIG 0x1241400 +#define mmACP_PAD_PULLUP_PULLDOWN_CTRL 0x1241404 +#define mmACP_PAD_DRIVE_STRENGTH_CTRL 0x1241408 +#define mmACP_SW_PAD_KEEPER_EN 0x124140C +#define mmACP_SW_WAKE_EN 0x1241410 +#define mmACP_I2S_WAKE_EN 0x1241414 +#define mmACP_PME_EN 0x1241418 +#define mmACP_PGFSM_CONTROL 0x124141C +#define mmACP_PGFSM_STATUS 0x1241420 + + +// Registers from ACP_SCRATCH block + +#define mmACP_SCRATCH_REG_0 0x1250000 +#define mmACP_SCRATCH_REG_1 0x1250004 +#define mmACP_SCRATCH_REG_2 0x1250008 +#define mmACP_SCRATCH_REG_3 0x125000C +#define mmACP_SCRATCH_REG_4 0x1250010 +#define mmACP_SCRATCH_REG_5 0x1250014 +#define mmACP_SCRATCH_REG_6 0x1250018 +#define mmACP_SCRATCH_REG_7 0x125001C +#define mmACP_SCRATCH_REG_8 0x1250020 +#define mmACP_SCRATCH_REG_9 0x1250024 +#define mmACP_SCRATCH_REG_10 0x1250028 +#define mmACP_SCRATCH_REG_11 0x125002C +#define mmACP_SCRATCH_REG_12 0x1250030 +#define mmACP_SCRATCH_REG_13 0x1250034 +#define mmACP_SCRATCH_REG_14 0x1250038 +#define mmACP_SCRATCH_REG_15 0x125003C +#define mmACP_SCRATCH_REG_16 0x1250040 +#define mmACP_SCRATCH_REG_17 0x1250044 +#define mmACP_SCRATCH_REG_18 0x1250048 +#define mmACP_SCRATCH_REG_19 0x125004C +#define mmACP_SCRATCH_REG_20 0x1250050 +#define mmACP_SCRATCH_REG_21 0x1250054 +#define mmACP_SCRATCH_REG_22 0x1250058 +#define mmACP_SCRATCH_REG_23 0x125005C +#define mmACP_SCRATCH_REG_24 0x1250060 +#define mmACP_SCRATCH_REG_25 0x1250064 +#define mmACP_SCRATCH_REG_26 0x1250068 +#define mmACP_SCRATCH_REG_27 0x125006C +#define mmACP_SCRATCH_REG_28 0x1250070 +#define mmACP_SCRATCH_REG_29 0x1250074 +#define mmACP_SCRATCH_REG_30 0x1250078 +#define mmACP_SCRATCH_REG_31 0x125007C +#define mmACP_SCRATCH_REG_32 0x1250080 +#define mmACP_SCRATCH_REG_33 0x1250084 +#define mmACP_SCRATCH_REG_34 0x1250088 +#define mmACP_SCRATCH_REG_35 0x125008C +#define mmACP_SCRATCH_REG_36 0x1250090 +#define mmACP_SCRATCH_REG_37 0x1250094 +#define mmACP_SCRATCH_REG_38 0x1250098 +#define mmACP_SCRATCH_REG_39 0x125009C +#define mmACP_SCRATCH_REG_40 0x12500A0 +#define mmACP_SCRATCH_REG_41 0x12500A4 +#define mmACP_SCRATCH_REG_42 0x12500A8 +#define mmACP_SCRATCH_REG_43 0x12500AC +#define mmACP_SCRATCH_REG_44 0x12500B0 +#define mmACP_SCRATCH_REG_45 0x12500B4 +#define mmACP_SCRATCH_REG_46 0x12500B8 +#define mmACP_SCRATCH_REG_47 0x12500BC +#define mmACP_SCRATCH_REG_48 0x12500C0 +#define mmACP_SCRATCH_REG_49 0x12500C4 +#define mmACP_SCRATCH_REG_50 0x12500C8 +#define mmACP_SCRATCH_REG_51 0x12500CC +#define mmACP_SCRATCH_REG_52 0x12500D0 +#define mmACP_SCRATCH_REG_53 0x12500D4 +#define mmACP_SCRATCH_REG_54 0x12500D8 +#define mmACP_SCRATCH_REG_55 0x12500DC +#define mmACP_SCRATCH_REG_56 0x12500E0 +#define mmACP_SCRATCH_REG_57 0x12500E4 +#define mmACP_SCRATCH_REG_58 0x12500E8 +#define mmACP_SCRATCH_REG_59 0x12500EC +#define mmACP_SCRATCH_REG_60 0x12500F0 +#define mmACP_SCRATCH_REG_61 0x12500F4 +#define mmACP_SCRATCH_REG_62 0x12500F8 +#define mmACP_SCRATCH_REG_63 0x12500FC +#define mmACP_SCRATCH_REG_64 0x1250100 +#define mmACP_SCRATCH_REG_65 0x1250104 +#define mmACP_SCRATCH_REG_66 0x1250108 +#define mmACP_SCRATCH_REG_67 0x125010C +#define mmACP_SCRATCH_REG_68 0x1250110 +#define mmACP_SCRATCH_REG_69 0x1250114 +#define mmACP_SCRATCH_REG_70 0x1250118 +#define mmACP_SCRATCH_REG_71 0x125011C +#define mmACP_SCRATCH_REG_72 0x1250120 +#define mmACP_SCRATCH_REG_73 0x1250124 +#define mmACP_SCRATCH_REG_74 0x1250128 +#define mmACP_SCRATCH_REG_75 0x125012C +#define mmACP_SCRATCH_REG_76 0x1250130 +#define mmACP_SCRATCH_REG_77 0x1250134 +#define mmACP_SCRATCH_REG_78 0x1250138 +#define mmACP_SCRATCH_REG_79 0x125013C +#define mmACP_SCRATCH_REG_80 0x1250140 +#define mmACP_SCRATCH_REG_81 0x1250144 +#define mmACP_SCRATCH_REG_82 0x1250148 +#define mmACP_SCRATCH_REG_83 0x125014C +#define mmACP_SCRATCH_REG_84 0x1250150 +#define mmACP_SCRATCH_REG_85 0x1250154 +#define mmACP_SCRATCH_REG_86 0x1250158 +#define mmACP_SCRATCH_REG_87 0x125015C +#define mmACP_SCRATCH_REG_88 0x1250160 +#define mmACP_SCRATCH_REG_89 0x1250164 +#define mmACP_SCRATCH_REG_90 0x1250168 +#define mmACP_SCRATCH_REG_91 0x125016C +#define mmACP_SCRATCH_REG_92 0x1250170 +#define mmACP_SCRATCH_REG_93 0x1250174 +#define mmACP_SCRATCH_REG_94 0x1250178 +#define mmACP_SCRATCH_REG_95 0x125017C +#define mmACP_SCRATCH_REG_96 0x1250180 +#define mmACP_SCRATCH_REG_97 0x1250184 +#define mmACP_SCRATCH_REG_98 0x1250188 +#define mmACP_SCRATCH_REG_99 0x125018C +#define mmACP_SCRATCH_REG_100 0x1250190 +#define mmACP_SCRATCH_REG_101 0x1250194 +#define mmACP_SCRATCH_REG_102 0x1250198 +#define mmACP_SCRATCH_REG_103 0x125019C +#define mmACP_SCRATCH_REG_104 0x12501A0 +#define mmACP_SCRATCH_REG_105 0x12501A4 +#define mmACP_SCRATCH_REG_106 0x12501A8 +#define mmACP_SCRATCH_REG_107 0x12501AC +#define mmACP_SCRATCH_REG_108 0x12501B0 +#define mmACP_SCRATCH_REG_109 0x12501B4 +#define mmACP_SCRATCH_REG_110 0x12501B8 +#define mmACP_SCRATCH_REG_111 0x12501BC +#define mmACP_SCRATCH_REG_112 0x12501C0 +#define mmACP_SCRATCH_REG_113 0x12501C4 +#define mmACP_SCRATCH_REG_114 0x12501C8 +#define mmACP_SCRATCH_REG_115 0x12501CC +#define mmACP_SCRATCH_REG_116 0x12501D0 +#define mmACP_SCRATCH_REG_117 0x12501D4 +#define mmACP_SCRATCH_REG_118 0x12501D8 +#define mmACP_SCRATCH_REG_119 0x12501DC +#define mmACP_SCRATCH_REG_120 0x12501E0 +#define mmACP_SCRATCH_REG_121 0x12501E4 +#define mmACP_SCRATCH_REG_122 0x12501E8 +#define mmACP_SCRATCH_REG_123 0x12501EC +#define mmACP_SCRATCH_REG_124 0x12501F0 +#define mmACP_SCRATCH_REG_125 0x12501F4 +#define mmACP_SCRATCH_REG_126 0x12501F8 +#define mmACP_SCRATCH_REG_127 0x12501FC +#define mmACP_SCRATCH_REG_128 0x1250200 + + +// Registers from ACP_SW_ACLK block + +#define mmSW_CORB_Base_Address 0x1243200 +#define mmSW_CORB_Write_Pointer 0x1243204 +#define mmSW_CORB_Read_Pointer 0x1243208 +#define mmSW_CORB_Control 0x124320C +#define mmSW_CORB_Size 0x1243214 +#define mmSW_RIRB_Base_Address 0x1243218 +#define mmSW_RIRB_Write_Pointer 0x124321C +#define mmSW_RIRB_Response_Interrupt_Count 0x1243220 +#define mmSW_RIRB_Control 0x1243224 +#define mmSW_RIRB_Size 0x1243228 +#define mmSW_RIRB_FIFO_MIN_THDL 0x124322C +#define mmSW_imm_cmd_UPPER_WORD 0x1243230 +#define mmSW_imm_cmd_LOWER_QWORD 0x1243234 +#define mmSW_imm_resp_UPPER_WORD 0x1243238 +#define mmSW_imm_resp_LOWER_QWORD 0x124323C +#define mmSW_imm_cmd_sts 0x1243240 +#define mmSW_BRA_BASE_ADDRESS 0x1243244 +#define mmSW_BRA_TRANSFER_SIZE 0x1243248 +#define mmSW_BRA_DMA_BUSY 0x124324C +#define mmSW_BRA_RESP 0x1243250 +#define mmSW_BRA_RESP_FRAME_ADDR 0x1243254 +#define mmSW_BRA_CURRENT_TRANSFER_SIZE 0x1243258 +#define mmSW_STATE_CHANGE_STATUS_0TO7 0x124325C +#define mmSW_STATE_CHANGE_STATUS_8TO11 0x1243260 +#define mmSW_STATE_CHANGE_STATUS_MASK_0to7 0x1243264 +#define mmSW_STATE_CHANGE_STATUS_MASK_8to11 0x1243268 +#define mmSW_CLK_FREQUENCY_CTRL 0x124326C +#define mmSW_ERROR_INTR_MASK 0x1243270 +#define mmSW_PHY_TEST_MODE_DATA_OFF 0x1243274 + + +// Registers from ACP_SW_SWCLK block + +#define mmACP_SW_EN 0x1243000 +#define mmACP_SW_EN_STATUS 0x1243004 +#define mmACP_SW_FRAMESIZE 0x1243008 +#define mmACP_SW_SSP_Counter 0x124300C +#define mmACP_SW_Audio_TX_EN 0x1243010 +#define mmACP_SW_Audio_TX_EN_STATUS 0x1243014 +#define mmACP_SW_Audio_TX_Frame_Format 0x1243018 +#define mmACP_SW_Audio_TX_SampleInterval 0x124301C +#define mmACP_SW_Audio_TX_Hctrl_DP0 0x1243020 +#define mmACP_SW_Audio_TX_Hctrl_DP1 0x1243024 +#define mmACP_SW_Audio_TX_Hctrl_DP2 0x1243028 +#define mmACP_SW_Audio_TX_Hctrl_DP3 0x124302C +#define mmACP_SW_Audio_TX_offset_DP0 0x1243030 +#define mmACP_SW_Audio_TX_offset_DP1 0x1243034 +#define mmACP_SW_Audio_TX_offset_DP2 0x1243038 +#define mmACP_SW_Audio_TX_offset_DP3 0x124303C +#define mmACP_SW_Audio_TX_Channel_Enable_DP0 0x1243040 +#define mmACP_SW_Audio_TX_Channel_Enable_DP1 0x1243044 +#define mmACP_SW_Audio_TX_Channel_Enable_DP2 0x1243048 +#define mmACP_SW_Audio_TX_Channel_Enable_DP3 0x124304C +#define mmACP_SW_BT_TX_EN 0x1243050 +#define mmACP_SW_BT_TX_EN_STATUS 0x1243054 +#define mmACP_SW_BT_TX_Frame_Format 0x1243058 +#define mmACP_SW_BT_TX_SampleInterval 0x124305C +#define mmACP_SW_BT_TX_Hctrl 0x1243060 +#define mmACP_SW_BT_TX_offset 0x1243064 +#define mmACP_SW_BT_TX_Channel_Enable_DP0 0x1243068 +#define mmACP_SW_Headset_TX_EN 0x124306C +#define mmACP_SW_Headset_TX_EN_STATUS 0x1243070 +#define mmACP_SW_Headset_TX_Frame_Format 0x1243074 +#define mmACP_SW_Headset_TX_SampleInterval 0x1243078 +#define mmACP_SW_Headset_TX_Hctrl 0x124307C +#define mmACP_SW_Headset_TX_offset 0x1243080 +#define mmACP_SW_Headset_TX_Channel_Enable_DP0 0x1243084 +#define mmACP_SW_Audio_RX_EN 0x1243088 +#define mmACP_SW_Audio_RX_EN_STATUS 0x124308C +#define mmACP_SW_Audio_RX_Frame_Format 0x1243090 +#define mmACP_SW_Audio_RX_SampleInterval 0x1243094 +#define mmACP_SW_Audio_RX_Hctrl_DP0 0x1243098 +#define mmACP_SW_Audio_RX_Hctrl_DP1 0x124309C +#define mmACP_SW_Audio_RX_Hctrl_DP2 0x1243100 +#define mmACP_SW_Audio_RX_Hctrl_DP3 0x1243104 +#define mmACP_SW_Audio_RX_offset_DP0 0x1243108 +#define mmACP_SW_Audio_RX_offset_DP1 0x124310C +#define mmACP_SW_Audio_RX_offset_DP2 0x1243110 +#define mmACP_SW_Audio_RX_offset_DP3 0x1243114 +#define mmACP_SW_Audio_RX_Channel_Enable_DP0 0x1243118 +#define mmACP_SW_Audio_RX_Channel_Enable_DP1 0x124311C +#define mmACP_SW_Audio_RX_Channel_Enable_DP2 0x1243120 +#define mmACP_SW_Audio_RX_Channel_Enable_DP3 0x1243124 +#define mmACP_SW_BT_RX_EN 0x1243128 +#define mmACP_SW_BT_RX_EN_STATUS 0x124312C +#define mmACP_SW_BT_RX_Frame_Format 0x1243130 +#define mmACP_SW_BT_RX_SampleInterval 0x1243134 +#define mmACP_SW_BT_RX_Hctrl 0x1243138 +#define mmACP_SW_BT_RX_offset 0x124313C +#define mmACP_SW_BT_RX_Channel_Enable_DP0 0x1243140 +#define mmACP_SW_Headset_RX_EN 0x1243144 +#define mmACP_SW_Headset_RX_EN_STATUS 0x1243148 +#define mmACP_SW_Headset_RX_Frame_Format 0x124314C +#define mmACP_SW_Headset_RX_SampleInterval 0x1243150 +#define mmACP_SW_Headset_RX_Hctrl 0x1243154 +#define mmACP_SW_Headset_RX_offset 0x1243158 +#define mmACP_SW_Headset_RX_Channel_Enable_DP0 0x124315C +#define mmACP_SW_BPT_PORT_EN 0x1243160 +#define mmACP_SW_BPT_PORT_EN_STATUS 0x1243164 +#define mmACP_SW_BPT_PORT_Frame_Format 0x1243168 +#define mmACP_SW_BPT_PORT_SampleInterval 0x124316C +#define mmACP_SW_BPT_PORT_Hctrl 0x1243170 +#define mmACP_SW_BPT_PORT_offset 0x1243174 +#define mmACP_SW_BPT_PORT_Channel_Enable 0x1243178 +#define mmACP_SW_BPT_PORT_First_byte_addr 0x124317C +#define mmACP_SW_CLK_RESUME_CTRL 0x1243180 +#define mmACP_SW_CLK_RESUME_Delay_Cntr 0x1243184 +#define mmACP_SW_BUS_RESET_CTRL 0x1243188 +#define mmACP_SW_PRBS_ERR_STATUS 0x124318C + + +// Registers from ACP_AUDIO_BUFFERS block + +#define mmACP_I2S_RX_RINGBUFADDR 0x1242000 +#define mmACP_I2S_RX_RINGBUFSIZE 0x1242004 +#define mmACP_I2S_RX_LINKPOSITIONCNTR 0x1242008 +#define mmACP_I2S_RX_FIFOADDR 0x124200C +#define mmACP_I2S_RX_FIFOSIZE 0x1242010 +#define mmACP_I2S_RX_DMA_SIZE 0x1242014 +#define mmACP_I2S_RX_LINEARPOSITIONCNTR_HIGH 0x1242018 +#define mmACP_I2S_RX_LINEARPOSITIONCNTR_LOW 0x124201C +#define mmACP_I2S_RX_INTR_WATERMARK_SIZE 0x1242020 +#define mmACP_I2S_TX_RINGBUFADDR 0x1242024 +#define mmACP_I2S_TX_RINGBUFSIZE 0x1242028 +#define mmACP_I2S_TX_LINKPOSITIONCNTR 0x124202C +#define mmACP_I2S_TX_FIFOADDR 0x1242030 +#define mmACP_I2S_TX_FIFOSIZE 0x1242034 +#define mmACP_I2S_TX_DMA_SIZE 0x1242038 +#define mmACP_I2S_TX_LINEARPOSITIONCNTR_HIGH 0x124203C +#define mmACP_I2S_TX_LINEARPOSITIONCNTR_LOW 0x1242040 +#define mmACP_I2S_TX_INTR_WATERMARK_SIZE 0x1242044 +#define mmACP_BT_RX_RINGBUFADDR 0x1242048 +#define mmACP_BT_RX_RINGBUFSIZE 0x124204C +#define mmACP_BT_RX_LINKPOSITIONCNTR 0x1242050 +#define mmACP_BT_RX_FIFOADDR 0x1242054 +#define mmACP_BT_RX_FIFOSIZE 0x1242058 +#define mmACP_BT_RX_DMA_SIZE 0x124205C +#define mmACP_BT_RX_LINEARPOSITIONCNTR_HIGH 0x1242060 +#define mmACP_BT_RX_LINEARPOSITIONCNTR_LOW 0x1242064 +#define mmACP_BT_RX_INTR_WATERMARK_SIZE 0x1242068 +#define mmACP_BT_TX_RINGBUFADDR 0x124206C +#define mmACP_BT_TX_RINGBUFSIZE 0x1242070 +#define mmACP_BT_TX_LINKPOSITIONCNTR 0x1242074 +#define mmACP_BT_TX_FIFOADDR 0x1242078 +#define mmACP_BT_TX_FIFOSIZE 0x124207C +#define mmACP_BT_TX_DMA_SIZE 0x1242080 +#define mmACP_BT_TX_LINEARPOSITIONCNTR_HIGH 0x1242084 +#define mmACP_BT_TX_LINEARPOSITIONCNTR_LOW 0x1242088 +#define mmACP_BT_TX_INTR_WATERMARK_SIZE 0x124208C +#define mmACP_HS_RX_RINGBUFADDR 0x1242090 +#define mmACP_HS_RX_RINGBUFSIZE 0x1242094 +#define mmACP_HS_RX_LINKPOSITIONCNTR 0x1242098 +#define mmACP_HS_RX_FIFOADDR 0x124209C +#define mmACP_HS_RX_FIFOSIZE 0x12420A0 +#define mmACP_HS_RX_DMA_SIZE 0x12420A4 +#define mmACP_HS_RX_LINEARPOSITIONCNTR_HIGH 0x12420A8 +#define mmACP_HS_RX_LINEARPOSITIONCNTR_LOW 0x12420AC +#define mmACP_HS_RX_INTR_WATERMARK_SIZE 0x12420B0 +#define mmACP_HS_TX_RINGBUFADDR 0x12420B4 +#define mmACP_HS_TX_RINGBUFSIZE 0x12420B8 +#define mmACP_HS_TX_LINKPOSITIONCNTR 0x12420BC +#define mmACP_HS_TX_FIFOADDR 0x12420C0 +#define mmACP_HS_TX_FIFOSIZE 0x12420C4 +#define mmACP_HS_TX_DMA_SIZE 0x12420C8 +#define mmACP_HS_TX_LINEARPOSITIONCNTR_HIGH 0x12420CC +#define mmACP_HS_TX_LINEARPOSITIONCNTR_LOW 0x12420D0 +#define mmACP_HS_TX_INTR_WATERMARK_SIZE 0x12420D4 + + +// Registers from ACP_I2S_TDM block + +#define mmACP_I2STDM_IER 0x1242400 +#define mmACP_I2STDM_IRER 0x1242404 +#define mmACP_I2STDM_RXFRMT 0x1242408 +#define mmACP_I2STDM_ITER 0x124240C +#define mmACP_I2STDM_TXFRMT 0x1242410 + + +// Registers from ACP_BT_TDM block + +#define mmACP_BTTDM_IER 0x1242800 +#define mmACP_BTTDM_IRER 0x1242804 +#define mmACP_BTTDM_RXFRMT 0x1242808 +#define mmACP_BTTDM_ITER 0x124280C +#define mmACP_BTTDM_TXFRMT 0x1242810 + + +// Registers from AZALIA_IP block + +#define mmAudio_Az_Global_Capabilities 0x1200000 +#define mmAudio_Az_Minor_Version 0x1200002 +#define mmAudio_Az_Major_Version 0x1200003 +#define mmAudio_Az_Output_Payload_Capability 0x1200004 +#define mmAudio_Az_Input_Payload_Capability 0x1200006 +#define mmAudio_Az_Global_Control 0x1200008 +#define mmAudio_Az_Wake_Enable 0x120000C +#define mmAudio_Az_State_Change_Status 0x120000E +#define mmAudio_Az_Global_Status 0x1200010 +#define mmAudio_Az_Linked_List_Capability_Header 0x1200014 +#define mmAudio_Az_Output_Stream_Payload_Capability 0x1200018 +#define mmAudio_Az_Input_Stream_Payload_Capability 0x120001A +#define mmAudio_Az_Interrupt_Control 0x1200020 +#define mmAudio_Az_Interrupt_Status 0x1200024 +#define mmAudio_Az_Wall_Clock_Counter 0x1200030 +#define mmAudio_Az_Stream_Synchronization 0x1200038 +#define mmAudio_Az_CORB_Lower_Base_Address 0x1200040 +#define mmAudio_Az_CORB_Upper_Base_Address 0x1200044 +#define mmAudio_Az_CORB_Write_Pointer 0x1200048 +#define mmAudio_Az_CORB_Read_Pointer 0x120004A +#define mmAudio_Az_CORB_Control 0x120004C +#define mmAudio_Az_CORB_Status 0x120004D +#define mmAudio_Az_CORB_Size 0x120004E +#define mmAudio_Az_RIRB_Lower_Base_Address 0x1200050 +#define mmAudio_Az_RIRB_Upper_Base_Address 0x1200054 +#define mmAudio_Az_RIRB_Write_Pointer 0x1200058 +#define mmAudio_Az_RIRB_Response_Interrupt_Count 0x120005A +#define mmAudio_Az_RIRB_Control 0x120005C +#define mmAudio_Az_RIRB_Status 0x120005D +#define mmAudio_Az_RIRB_Size 0x120005E +#define mmAudio_Az_Immediate_Command_Output_Interface 0x1200060 +#define mmAudio_Az_Immediate_Response_Input_Interface 0x1200064 +#define mmAudio_Az_Immediate_Command_Status 0x1200068 +#define mmAudio_Az_DPLBASE 0x1200070 +#define mmAudio_Az_DPUBASE 0x1200074 +#define mmAudio_Az_Input_SD0CTL_and_STS 0x1200080 +#define mmAudio_Az_Input_SD0LPIB 0x1200084 +#define mmAudio_Az_Input_SD0CBL 0x1200088 +#define mmAudio_Az_Input_SD0LVI 0x120008C +#define mmAudio_Az_Input_SD0FIFOS 0x1200090 +#define mmAudio_Az_Input_SD0FMT 0x1200092 +#define mmAudio_Az_Input_SD0BDPL 0x1200098 +#define mmAudio_Az_Input_SD0BDPU 0x120009C +#define mmAudio_Az_Input_SD1CTL_and_STS 0x12000A0 +#define mmAudio_Az_Input_SD1LPIB 0x12000A4 +#define mmAudio_Az_Input_SD1CBL 0x12000A8 +#define mmAudio_Az_Input_SD1LVI 0x12000AC +#define mmAudio_Az_Input_SD1FIFOS 0x12000B0 +#define mmAudio_Az_Input_SD1FMT 0x12000B2 +#define mmAudio_Az_Input_SD1BDPL 0x12000B8 +#define mmAudio_Az_Input_SD1BDPU 0x12000BC +#define mmAudio_Az_Input_SD2CTL_and_STS 0x12000C0 +#define mmAudio_Az_Input_SD2LPIB 0x12000C4 +#define mmAudio_Az_Input_SD2CBL 0x12000C8 +#define mmAudio_Az_Input_SD2LVI 0x12000CC +#define mmAudio_Az_Input_SD2FIFOS 0x12000D0 +#define mmAudio_Az_Input_SD2FMT 0x12000D2 +#define mmAudio_Az_Input_SD2BDPL 0x12000D8 +#define mmAudio_Az_Input_SD2BDPU 0x12000DC +#define mmAudio_Az_Input_SD3CTL_and_STS 0x12000E0 +#define mmAudio_Az_Input_SD3LPIB 0x12000E4 +#define mmAudio_Az_Input_SD3CBL 0x12000E8 +#define mmAudio_Az_Input_SD3LVI 0x12000EC +#define mmAudio_Az_Input_SD3FIFOS 0x12000F0 +#define mmAudio_Az_Input_SD3FMT 0x12000F2 +#define mmAudio_Az_Input_SD3BDPL 0x12000F8 +#define mmAudio_Az_Input_SD3BDPU 0x12000FC +#define mmAudio_Az_Output_SD0CTL_and_STS 0x1200100 +#define mmAudio_Az_Output_SD0LPIB 0x1200104 +#define mmAudio_Az_Output_SD0CBL 0x1200108 +#define mmAudio_Az_Output_SD0LVI 0x120010C +#define mmAudio_Az_Output_SD0FIFOS 0x1200110 +#define mmAudio_Az_Output_SD0FMT 0x1200112 +#define mmAudio_Az_Output_SD0BDPL 0x1200118 +#define mmAudio_Az_Output_SD0BDPU 0x120011C +#define mmAudio_Az_Output_SD1CTL_and_STS 0x1200120 +#define mmAudio_Az_Output_SD1LPIB 0x1200124 +#define mmAudio_Az_Output_SD1CBL 0x1200128 +#define mmAudio_Az_Output_SD1LVI 0x120012C +#define mmAudio_Az_Output_SD1FIFOS 0x1200130 +#define mmAudio_Az_Output_SD1FMT 0x1200132 +#define mmAudio_Az_Output_SD1BDPL 0x1200138 +#define mmAudio_Az_Output_SD1BDPU 0x120013C +#define mmAudio_Az_Output_SD2CTL_and_STS 0x1200140 +#define mmAudio_Az_Output_SD2LPIB 0x1200144 +#define mmAudio_Az_Output_SD2CBL 0x1200148 +#define mmAudio_Az_Output_SD2LVI 0x120014C +#define mmAudio_Az_Output_SD2FIFOS 0x1200150 +#define mmAudio_Az_Output_SD2FMT 0x1200152 +#define mmAudio_Az_Output_SD2BDPL 0x1200158 +#define mmAudio_Az_Output_SD2BDPU 0x120015C +#define mmAudio_Az_Output_SD3CTL_and_STS 0x1200160 +#define mmAudio_Az_Output_SD3LPIB 0x1200164 +#define mmAudio_Az_Output_SD3CBL 0x1200168 +#define mmAudio_Az_Output_SD3LVI 0x120016C +#define mmAudio_Az_Output_SD3FIFOS 0x1200170 +#define mmAudio_Az_Output_SD3FMT 0x1200172 +#define mmAudio_Az_Output_SD3BDPL 0x1200178 +#define mmAudio_Az_Output_SD3BDPU 0x120017C +#define mmAudioAZ_Misc_Control_Register_1 0x1200180 +#define mmAudioAZ_Misc_Control_Register_2 0x1200182 +#define mmAudioAZ_Misc_Control_Register_3 0x1200183 +#define mmAudio_AZ_Multiple_Links_Capability_Header 0x1200200 +#define mmAudio_AZ_Multiple_Links_Capability_Declaration 0x1200204 +#define mmAudio_AZ_Link0_Capabilities 0x1200240 +#define mmAudio_AZ_Link0_Control 0x1200244 +#define mmAudio_AZ_Link0_Output_Stream_ID 0x1200248 +#define mmAudio_AZ_Link0_SDI_Identifier 0x120024C +#define mmAudio_AZ_Link0_Per_Stream_Overhead 0x1200250 +#define mmAudio_AZ_Link0_Wall_Frame_Counter 0x1200258 +#define mmAudio_AZ_Link0_Output_Payload_Capability_L 0x1200260 +#define mmAudio_AZ_Link0_Output_Payload_Capability_U 0x1200264 +#define mmAudio_AZ_Link0_Input_Payload_Capability_L 0x1200270 +#define mmAudio_AZ_Link0_Input_Payload_Capability_U 0x1200274 +#define mmAudio_Az_Input_SD0LICBA 0x1202084 +#define mmAudio_Az_Input_SD1LICBA 0x12020A4 +#define mmAudio_Az_Input_SD2LICBA 0x12020C4 +#define mmAudio_Az_Input_SD3LICBA 0x12020E4 +#define mmAudio_Az_Output_SD0LICBA 0x1202104 +#define mmAudio_Az_Output_SD1LICBA 0x1202124 +#define mmAudio_Az_Output_SD2LICBA 0x1202144 +#define mmAudio_Az_Output_SD3LICBA 0x1202164 +#define mmAUDIO_AZ_POWER_MANAGEMENT_CONTROL 0x1204000 +#define mmAUDIO_AZ_IOC_SOFTRST_CONTROL 0x1204004 +#define mmAUDIO_AZ_IOC_CLKGATE_CONTROL 0x1204008 + + +// Registers from ACP_AZALIA block + +#define mmACP_AZ_PAGE0_LBASE_ADDR 0x1243800 +#define mmACP_AZ_PAGE0_UBASE_ADDR 0x1243804 +#define mmACP_AZ_PAGE0_PGEN_SIZE 0x1243808 +#define mmACP_AZ_PAGE0_OFFSET 0x124380C +#define mmACP_AZ_PAGE1_LBASE_ADDR 0x1243810 +#define mmACP_AZ_PAGE1_UBASE_ADDR 0x1243814 +#define mmACP_AZ_PAGE1_PGEN_SIZE 0x1243818 +#define mmACP_AZ_PAGE1_OFFSET 0x124381C +#define mmACP_AZ_PAGE2_LBASE_ADDR 0x1243820 +#define mmACP_AZ_PAGE2_UBASE_ADDR 0x1243824 +#define mmACP_AZ_PAGE2_PGEN_SIZE 0x1243828 +#define mmACP_AZ_PAGE2_OFFSET 0x124382C +#define mmACP_AZ_PAGE3_LBASE_ADDR 0x1243830 +#define mmACP_AZ_PAGE3_UBASE_ADDR 0x1243834 +#define mmACP_AZ_PAGE3_PGEN_SIZE 0x1243838 +#define mmACP_AZ_PAGE3_OFFSET 0x124383C +#define mmACP_AZ_PAGE4_LBASE_ADDR 0x1243840 +#define mmACP_AZ_PAGE4_UBASE_ADDR 0x1243844 +#define mmACP_AZ_PAGE4_PGEN_SIZE 0x1243848 +#define mmACP_AZ_PAGE4_OFFSET 0x124384C +#define mmACP_AZ_PAGE5_LBASE_ADDR 0x1243850 +#define mmACP_AZ_PAGE5_UBASE_ADDR 0x1243854 +#define mmACP_AZ_PAGE5_PGEN_SIZE 0x1243858 +#define mmACP_AZ_PAGE5_OFFSET 0x124385C +#define mmACP_AZ_PAGE6_LBASE_ADDR 0x1243860 +#define mmACP_AZ_PAGE6_UBASE_ADDR 0x1243864 +#define mmACP_AZ_PAGE6_PGEN_SIZE 0x1243868 +#define mmACP_AZ_PAGE6_OFFSET 0x124386C +#define mmACP_AZ_PAGE7_LBASE_ADDR 0x1243870 +#define mmACP_AZ_PAGE7_UBASE_ADDR 0x1243874 +#define mmACP_AZ_PAGE7_PGEN_SIZE 0x1243878 +#define mmACP_AZ_PAGE7_OFFSET 0x124387C + + +#endif -- 2.7.4

3 3

[alsa-devel] [PATCH v6 3/3] ALSA: hda: add support for Huawei WMI micmute LED
by Ayman Bagabas 14 Nov '18

14 Nov '18

Some of Huawei laptops come with a LED in the micmute key. This patch enables the use of micmute LED for these devices: 1. Matebook X (19e5:3200), (19e5:3201) 2. Matebook X Pro (19e5:3204) Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/huawei_wmi_helper.c | 47 +++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 14 ++++++++- 2 files changed, 60 insertions(+), 1 deletion(-) create mode 100644 sound/pci/hda/huawei_wmi_helper.c diff --git a/sound/pci/hda/huawei_wmi_helper.c b/sound/pci/hda/huawei_wmi_helper.c new file mode 100644 index 000000000000..7c91f914ffba --- /dev/null +++ b/sound/pci/hda/huawei_wmi_helper.c @@ -0,0 +1,47 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Helper functions for Huawei WMI micmute LED; + * to be included from codec driver + */ + +#if IS_ENABLED(CONFIG_HUAWEI_WMI) +#include <linux/platform_data/x86/huawei-wmi.h> + +static int (*huawei_wmi_micmute_led_set_func)(bool); + +static void update_huawei_wmi_micmute_led(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + huawei_wmi_micmute_led_set_func(spec->micmute_led.led_value); +} + +static void alc_fixup_huawei_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + bool removefunc = false; + + if (action == HDA_FIXUP_ACT_PROBE) { + if (!huawei_wmi_micmute_led_set_func) + huawei_wmi_micmute_led_set_func = symbol_request(huawei_wmi_micmute_led_set); + if (!huawei_wmi_micmute_led_set_func) { + codec_warn(codec, "Failed to find huawei_wmi symbol huawei_wmi_micmute_led_set\n"); + return; + } + + removefunc = (huawei_wmi_micmute_led_set_func(false) < 0) + || (snd_hda_gen_add_micmute_led(codec, update_huawei_wmi_micmute_led) < 0); + } + + if (huawei_wmi_micmute_led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { + symbol_put(huawei_wmi_micmute_led_set); + huawei_wmi_micmute_led_set_func = NULL; + } +} + +#else + +static void alc_fixup_huawei_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ +} + +#endif diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 135c0826859b..5431dee40f4f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5374,6 +5374,9 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" +/* for alc_fixup_huawei_wmi */ +#include "huawei_wmi_helper.c" + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -5494,6 +5497,7 @@ enum { ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, ALC256_FIXUP_HUAWEI_MBXP_PINS, + ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -5765,6 +5769,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_huawei_wmi + }, [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5779,7 +5787,9 @@ static const struct hda_fixup alc269_fixups[] = { {0x1e, 0x411111f0}, {0x21, 0x04211020}, { } - } + }, + .chained = true, + .chain_id = ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, @@ -6611,6 +6621,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3200, "Huawei MBX", ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED), + SND_PCI_QUIRK(0x19e5, 0x3201, "Huawei MBX", ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 2.17.2

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[alsa-devel] [PATCH v6 2/3] ALSA: hda: fix front speakers on Huawei MBXP.
by Ayman Bagabas 14 Nov '18

14 Nov '18

This patch solves bug 200501 'Only 2 of 4 speakers playing sound.' https://bugzilla.kernel.org/show_bug.cgi?id=200501 It enables the front speakers on Huawei Matebook X Pro laptops. These laptops come with Dolby Atmos sound system and these pins configuration enables the front speakers. Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fa61674a5605..135c0826859b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5493,6 +5493,7 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, + ALC256_FIXUP_HUAWEI_MBXP_PINS, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -5764,6 +5765,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x12, 0x90a60130}, + {0x13, 0x40000000}, + {0x14, 0x90170110}, + {0x18, 0x411111f0}, + {0x19, 0x04a11040}, + {0x1a, 0x411111f0}, + {0x1b, 0x90170112}, + {0x1d, 0x40759a05}, + {0x1e, 0x411111f0}, + {0x21, 0x04211020}, + { } + } + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -6594,6 +6611,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ #if 0 -- 2.17.2

1 0

[alsa-devel] [asoc:for-4.21 60/62] sound/soc/amd/raven/pci-acp3x.c:59:8: error: implicit declaration of function 'pci_enable_msi'
by kbuild test robot 14 Nov '18

14 Nov '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.21 head: 9d4b17efa51f3bb0918021090c09b0013039df00 commit: 818ba93b0e9ed4a08cba575dd49c1391ef4b7776 [60/62] ASoC: amd: enable acp3x drivers build 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 818ba93b0e9ed4a08cba575dd49c1391ef4b7776 # save the attached .config to linux build tree make.cross ARCH=openrisc All errors (new ones prefixed by >>): sound/soc/amd/raven/pci-acp3x.c: In function 'snd_acp3x_probe': >> sound/soc/amd/raven/pci-acp3x.c:59:8: error: implicit declaration of function 'pci_enable_msi' [-Werror=implicit-function-declaration] ret = pci_enable_msi(pci); ^~~~~~~~~~~~~~ >> sound/soc/amd/raven/pci-acp3x.c:124:2: error: implicit declaration of function 'pci_disable_msi' [-Werror=implicit-function-declaration] pci_disable_msi(pci); ^~~~~~~~~~~~~~~ sound/soc/amd/raven/pci-acp3x.c: At top level: sound/soc/amd/raven/pci-acp3x.c:161:1: warning: data definition has no type or storage class module_pci_driver(acp3x_driver); ^~~~~~~~~~~~~~~~~ sound/soc/amd/raven/pci-acp3x.c:161:1: error: type defaults to 'int' in declaration of 'module_pci_driver' [-Werror=implicit-int] sound/soc/amd/raven/pci-acp3x.c:161:1: warning: parameter names (without types) in function declaration sound/soc/amd/raven/pci-acp3x.c:154:26: warning: 'acp3x_driver' defined but not used [-Wunused-variable] static struct pci_driver acp3x_driver = { ^~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/pci_enable_msi +59 sound/soc/amd/raven/pci-acp3x.c e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 30 e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 31 static int snd_acp3x_probe(struct pci_dev *pci, e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 32 const struct pci_device_id *pci_id) e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 33 { e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 34 int ret; 7894a7e7 Vijendar Mukunda 2018-11-12 35 u32 addr, val; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 36 struct acp3x_dev_data *adata; 7894a7e7 Vijendar Mukunda 2018-11-12 37 struct platform_device_info pdevinfo; 7894a7e7 Vijendar Mukunda 2018-11-12 38 unsigned int irqflags; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 39 e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 40 if (pci_enable_device(pci)) { e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 41 dev_err(&pci->dev, "pci_enable_device failed\n"); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 42 return -ENODEV; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 43 } e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 44 e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 45 ret = pci_request_regions(pci, "AMD ACP3x audio"); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 46 if (ret < 0) { e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 47 dev_err(&pci->dev, "pci_request_regions failed\n"); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 48 goto disable_pci; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 49 } e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 50 e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 51 adata = devm_kzalloc(&pci->dev, sizeof(struct acp3x_dev_data), e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 52 GFP_KERNEL); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 53 if (!adata) { e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 54 ret = -ENOMEM; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 55 goto release_regions; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 56 } e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 57 7894a7e7 Vijendar Mukunda 2018-11-12 58 /* check for msi interrupt support */ 7894a7e7 Vijendar Mukunda 2018-11-12 @59 ret = pci_enable_msi(pci); 7894a7e7 Vijendar Mukunda 2018-11-12 60 if (ret) 7894a7e7 Vijendar Mukunda 2018-11-12 61 /* msi is not enabled */ 7894a7e7 Vijendar Mukunda 2018-11-12 62 irqflags = IRQF_SHARED; 7894a7e7 Vijendar Mukunda 2018-11-12 63 else 7894a7e7 Vijendar Mukunda 2018-11-12 64 /* msi is enabled */ 7894a7e7 Vijendar Mukunda 2018-11-12 65 irqflags = 0; 7894a7e7 Vijendar Mukunda 2018-11-12 66 e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 67 addr = pci_resource_start(pci, 0); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 68 adata->acp3x_base = ioremap(addr, pci_resource_len(pci, 0)); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 69 if (!adata->acp3x_base) { e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 70 ret = -ENOMEM; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 71 goto release_regions; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 72 } e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 73 pci_set_master(pci); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 74 pci_set_drvdata(pci, adata); 7894a7e7 Vijendar Mukunda 2018-11-12 75 7894a7e7 Vijendar Mukunda 2018-11-12 76 val = rv_readl(adata->acp3x_base + mmACP_I2S_PIN_CONFIG); 7894a7e7 Vijendar Mukunda 2018-11-12 77 switch (val) { 7894a7e7 Vijendar Mukunda 2018-11-12 78 case I2S_MODE: 7894a7e7 Vijendar Mukunda 2018-11-12 79 adata->res = devm_kzalloc(&pci->dev, 7894a7e7 Vijendar Mukunda 2018-11-12 80 sizeof(struct resource) * 2, 7894a7e7 Vijendar Mukunda 2018-11-12 81 GFP_KERNEL); 7894a7e7 Vijendar Mukunda 2018-11-12 82 if (!adata->res) { 7894a7e7 Vijendar Mukunda 2018-11-12 83 ret = -ENOMEM; 7894a7e7 Vijendar Mukunda 2018-11-12 84 goto unmap_mmio; 7894a7e7 Vijendar Mukunda 2018-11-12 85 } 7894a7e7 Vijendar Mukunda 2018-11-12 86 7894a7e7 Vijendar Mukunda 2018-11-12 87 adata->res[0].name = "acp3x_i2s_iomem"; 7894a7e7 Vijendar Mukunda 2018-11-12 88 adata->res[0].flags = IORESOURCE_MEM; 7894a7e7 Vijendar Mukunda 2018-11-12 89 adata->res[0].start = addr; 7894a7e7 Vijendar Mukunda 2018-11-12 90 adata->res[0].end = addr + (ACP3x_REG_END - ACP3x_REG_START); 7894a7e7 Vijendar Mukunda 2018-11-12 91 7894a7e7 Vijendar Mukunda 2018-11-12 92 adata->res[1].name = "acp3x_i2s_irq"; 7894a7e7 Vijendar Mukunda 2018-11-12 93 adata->res[1].flags = IORESOURCE_IRQ; 7894a7e7 Vijendar Mukunda 2018-11-12 94 adata->res[1].start = pci->irq; 7894a7e7 Vijendar Mukunda 2018-11-12 95 adata->res[1].end = pci->irq; 7894a7e7 Vijendar Mukunda 2018-11-12 96 7894a7e7 Vijendar Mukunda 2018-11-12 97 adata->acp3x_audio_mode = ACP3x_I2S_MODE; 7894a7e7 Vijendar Mukunda 2018-11-12 98 7894a7e7 Vijendar Mukunda 2018-11-12 99 memset(&pdevinfo, 0, sizeof(pdevinfo)); 7894a7e7 Vijendar Mukunda 2018-11-12 100 pdevinfo.name = "acp3x_rv_i2s"; 7894a7e7 Vijendar Mukunda 2018-11-12 101 pdevinfo.id = 0; 7894a7e7 Vijendar Mukunda 2018-11-12 102 pdevinfo.parent = &pci->dev; 7894a7e7 Vijendar Mukunda 2018-11-12 103 pdevinfo.num_res = 2; 7894a7e7 Vijendar Mukunda 2018-11-12 104 pdevinfo.res = adata->res; 7894a7e7 Vijendar Mukunda 2018-11-12 105 pdevinfo.data = &irqflags; 7894a7e7 Vijendar Mukunda 2018-11-12 106 pdevinfo.size_data = sizeof(irqflags); 7894a7e7 Vijendar Mukunda 2018-11-12 107 7894a7e7 Vijendar Mukunda 2018-11-12 108 adata->pdev = platform_device_register_full(&pdevinfo); 7894a7e7 Vijendar Mukunda 2018-11-12 109 if (!adata->pdev) { 7894a7e7 Vijendar Mukunda 2018-11-12 110 dev_err(&pci->dev, "cannot register %s device\n", 7894a7e7 Vijendar Mukunda 2018-11-12 111 pdevinfo.name); 7894a7e7 Vijendar Mukunda 2018-11-12 112 ret = -ENODEV; 7894a7e7 Vijendar Mukunda 2018-11-12 113 goto unmap_mmio; 7894a7e7 Vijendar Mukunda 2018-11-12 114 } 7894a7e7 Vijendar Mukunda 2018-11-12 115 break; 7894a7e7 Vijendar Mukunda 2018-11-12 116 default: 7894a7e7 Vijendar Mukunda 2018-11-12 117 dev_err(&pci->dev, "Inavlid ACP audio mode : %d\n", val); 7894a7e7 Vijendar Mukunda 2018-11-12 118 ret = -ENODEV; 7894a7e7 Vijendar Mukunda 2018-11-12 119 goto unmap_mmio; 7894a7e7 Vijendar Mukunda 2018-11-12 120 } e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 121 return 0; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 122 7894a7e7 Vijendar Mukunda 2018-11-12 123 unmap_mmio: 7894a7e7 Vijendar Mukunda 2018-11-12 @124 pci_disable_msi(pci); 7894a7e7 Vijendar Mukunda 2018-11-12 125 iounmap(adata->acp3x_base); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 126 release_regions: e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 127 pci_release_regions(pci); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 128 disable_pci: e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 129 pci_disable_device(pci); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 130 e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 131 return ret; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 132 } e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 133 :::::: The code at line 59 was first introduced by commit :::::: 7894a7e7ea3de6c1fd5b2b43f3a73106c94ef584 ASoC: amd: create ACP3x PCM platform device :::::: TO: Vijendar Mukunda <Vijendar.Mukunda(a)amd.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

2 1

[alsa-devel] [PATCH v5 3/3] ALSA: hda: add support for Huawei WMI micmute LED
by Ayman Bagabas 14 Nov '18

14 Nov '18

Some of Huawei laptops come with a LED in the micmute key. This patch enables and disable this LED accordingly. This enables micmute LED for: 1. Matebook X (19e5:3200), (19e5:3201) 2. Matebook X Pro (19e5:3204) Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> --- sound/pci/hda/huawei_wmi_helper.c | 47 +++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 14 ++++++++- 2 files changed, 60 insertions(+), 1 deletion(-) create mode 100644 sound/pci/hda/huawei_wmi_helper.c diff --git a/sound/pci/hda/huawei_wmi_helper.c b/sound/pci/hda/huawei_wmi_helper.c new file mode 100644 index 000000000000..7c91f914ffba --- /dev/null +++ b/sound/pci/hda/huawei_wmi_helper.c @@ -0,0 +1,47 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Helper functions for Huawei WMI micmute LED; + * to be included from codec driver + */ + +#if IS_ENABLED(CONFIG_HUAWEI_WMI) +#include <linux/platform_data/x86/huawei-wmi.h> + +static int (*huawei_wmi_micmute_led_set_func)(bool); + +static void update_huawei_wmi_micmute_led(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + huawei_wmi_micmute_led_set_func(spec->micmute_led.led_value); +} + +static void alc_fixup_huawei_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + bool removefunc = false; + + if (action == HDA_FIXUP_ACT_PROBE) { + if (!huawei_wmi_micmute_led_set_func) + huawei_wmi_micmute_led_set_func = symbol_request(huawei_wmi_micmute_led_set); + if (!huawei_wmi_micmute_led_set_func) { + codec_warn(codec, "Failed to find huawei_wmi symbol huawei_wmi_micmute_led_set\n"); + return; + } + + removefunc = (huawei_wmi_micmute_led_set_func(false) < 0) + || (snd_hda_gen_add_micmute_led(codec, update_huawei_wmi_micmute_led) < 0); + } + + if (huawei_wmi_micmute_led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { + symbol_put(huawei_wmi_micmute_led_set); + huawei_wmi_micmute_led_set_func = NULL; + } +} + +#else + +static void alc_fixup_huawei_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ +} + +#endif diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 135c0826859b..5431dee40f4f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5374,6 +5374,9 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" +/* for alc_fixup_huawei_micmute_led */ +#include "huawei_wmi_helper.c" + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -5494,6 +5497,7 @@ enum { ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, ALC256_FIXUP_HUAWEI_MBXP_PINS, + ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -5765,6 +5769,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_huawei_wmi + }, [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5779,7 +5787,9 @@ static const struct hda_fixup alc269_fixups[] = { {0x1e, 0x411111f0}, {0x21, 0x04211020}, { } - } + }, + .chained = true, + .chain_id = ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, @@ -6611,6 +6621,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3200, "Huawei MBX", ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED), + SND_PCI_QUIRK(0x19e5, 0x3201, "Huawei MBX", ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 2.17.2

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[alsa-devel] HTTP GIT access to alsa-lib seems to be misaligned
by Timo Wischer 14 Nov '18

14 Nov '18

Hi all, The master branch of $ git clone http://git.alsa-project.org/http/alsa-lib.git is a different one as of $ git clone git://git.alsa-project.org/alsa-lib.git alsa-lib $ git log --graph --oneline http/master * dd2f6f8 update README * 9021f6c samples: make them work well * eec7186 tests: add a test for client instance of ALSATimer * c46b9e6 tests: add a test for port instance of ALSASeq * d6467d9 tests: add a test for client instance of ALSASeq * 03a8b8b tests: add test for public API of client class of ALSACtl * 2f94c2d rawmidi; add GObject enumerator for information flags * ed49bff rawmidi: add GObject enums for stream direction * 25e37bc rawmidi: add support for GObject enumerations * 709c4ce timer: add GObject enums for event type of timer instance * 10c9f2e timer: add GObject flags for paramter flags of timer instance * 3118597 timer: add GObject flags for flag of global timer instance * b28e1d7 timer: add GObject enums for type of global timer instance * 48f904b timer: add GObject enums for slave class of timer instance * 3bc28ec timer: add GObject enums for device class of timer instance * 480b358 timer: add support for GObject enumerations versus $ git log --graph --oneline origin/master * 178fe66 utils/alsa.m4: conditionally enable libdl in AM_PATH_ALSA m4 macro * 9a8d0b0 README.md: add link to www.alsa-project.org * fa72edc initial version of README.md for github * 6fbee19 initial version of .travis.yml file * b420056 pcm: interval: Interpret (x x+1] correctly and return x+1 * 92172ad test/audio_time: remove unused variables * 956bc15 test: rename code to more approriate mixtest * 82ae026 test/code: make it work again * 6852f88 test/latecy: fix typo in tstamp compare * 6eced6b conf/ucm/Dell-WD15-Dock: Fix incorrect device names * 3ec6dce Release v1.1.7 * fd13267 ioplug: Fix comment in struct snd_pcm_ioplug * 60d74ce timer: fix wrong comment to refer to 'SNDRV_TIMER_PSFLG_*' * 7c5c050 seq: Fix signedness in MIDI encoder/decoder * a849163 control_hw: Fix issue when applying seccomp policy * 040cfea conf/ucm: bytcr-rt5640: Improve human readable input/output names @Takashi The latest commit of http/master is yours. May be you have an idea how to solve this issue. -- Best regards *Timo Wischer* Engineering Software Base (ADITG/ESB) Tel. +49 5121 49 6938

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[alsa-devel] [PATCH 1/2] ASoC: pcm3060: Add control for differential output
by Kirill Marinushkin 14 Nov '18

14 Nov '18

DAC may be switched between differential and single-ended output. Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> --- sound/soc/codecs/pcm3060.c | 9 +++++++++ sound/soc/codecs/pcm3060.h | 1 + 2 files changed, 10 insertions(+) diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 494d9d662be8..f792ad637ae4 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -179,6 +179,13 @@ static struct snd_soc_dai_driver pcm3060_dai[] = { /* dapm */ +static const char * const pcm3060_dapm_out_dif_txt[] = { + "Differential", "Single-Ended" +}; + +static SOC_ENUM_SINGLE_DECL(pcm3060_dapm_out_dif_enum, PCM3060_REG64, + PCM3060_REG_SHIFT_SE, pcm3060_dapm_out_dif_txt); + static DECLARE_TLV_DB_SCALE(pcm3060_dapm_tlv, -10050, 50, 1); static const struct snd_kcontrol_new pcm3060_dapm_controls[] = { @@ -195,6 +202,8 @@ static const struct snd_kcontrol_new pcm3060_dapm_controls[] = { 0, pcm3060_dapm_tlv), SOC_DOUBLE("Master Capture Switch", PCM3060_REG73, PCM3060_REG_SHIFT_MUT11, PCM3060_REG_SHIFT_MUT12, 1, 1), + + SOC_ENUM("DAC Out Differential", pcm3060_dapm_out_dif_enum), }; static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = { diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h index fd89a68aa8a7..32080ff00166 100644 --- a/sound/soc/codecs/pcm3060.h +++ b/sound/soc/codecs/pcm3060.h @@ -38,6 +38,7 @@ int pcm3060_remove(struct device *dev); #define PCM3060_REG_ADPSV 0x20 #define PCM3060_REG_DAPSV 0x10 #define PCM3060_REG_SE 0x01 +#define PCM3060_REG_SHIFT_SE 0x00 #define PCM3060_REG65 0x41 #define PCM3060_REG66 0x42 -- 2.13.6

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[alsa-devel] [RFC PATCH] cplay: Always write frag * fragment_size
by Daniel Baluta 14 Nov '18

14 Nov '18

cplay first writes frag * fragment_size and then it only writes one fragment at a time. This means for example than if the user supplied a buffer_size it will only be used for the first write. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- I noticed this while investigating why cplay prints buffer_size as 0 when not specified as command line argument to cplay. I also noticed that cred always reads frag * frament_size, so I think this patch should be OK, but marking it as RFC to get your thoughts. src/utils/cplay.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/src/utils/cplay.c b/src/utils/cplay.c index 2a52b52..b016f52 100644 --- a/src/utils/cplay.c +++ b/src/utils/cplay.c @@ -435,15 +435,15 @@ void play_samples(char *name, unsigned int card, unsigned int device, }; if (verbose) printf("%s: Opened compress device\n", __func__); - size = config.fragment_size; - buffer = malloc(size * config.fragments); + size = config.fragments * config.fragment_size; + buffer = malloc(size); if (!buffer) { fprintf(stderr, "Unable to allocate %d bytes\n", size); goto COMP_EXIT; } /* we will write frag fragment_size and then start */ - num_read = fread(buffer, 1, size * config.fragments, file); + num_read = fread(buffer, 1, size, file); if (num_read > 0) { if (verbose) printf("%s: Doing first buffer write of %d\n", __func__, num_read); @@ -459,7 +459,7 @@ void play_samples(char *name, unsigned int card, unsigned int device, } } printf("Playing file %s On Card %u device %u, with buffer of %lu bytes\n", - name, card, device, buffer_size); + name, card, device, size); printf("Format %u Channels %u, %u Hz, Bit Rate %d\n", codec.id, codec.ch_in, codec.sample_rate, codec.bit_rate); -- 2.7.4

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[alsa-devel] Applied "ASoC: stm32: sai: fix invalid use of sizeof in stm32_sai_add_mclk_provider()" to the asoc tree
by Mark Brown 14 Nov '18

14 Nov '18

The patch ASoC: stm32: sai: fix invalid use of sizeof in stm32_sai_add_mclk_provider() 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 496fa3ba0a66c0192be86a45f44f94827ab8bcb6 Mon Sep 17 00:00:00 2001 From: Wei Yongjun <weiyongjun1(a)huawei.com> Date: Sat, 27 Oct 2018 02:19:59 +0000 Subject: [PATCH] ASoC: stm32: sai: fix invalid use of sizeof in stm32_sai_add_mclk_provider() sizeof() when applied to a pointer typed expression gives the size of the pointer, not that of the pointed data. Fixes: 8307b2afd386 ("ASoC: stm32: sai: set sai as mclk clock provider") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 506423d5861a..d4825700b63f 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -389,7 +389,7 @@ static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai) char *mclk_name, *p, *s = (char *)pname; int ret, i = 0; - mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL); + mclk = devm_kzalloc(dev, sizeof(*mclk), GFP_KERNEL); if (!mclk) return -ENOMEM; -- 2.19.1

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[alsa-devel] Applied "ASoC: pcm3060: Rename output widgets" to the asoc tree
by Mark Brown 14 Nov '18

14 Nov '18

The patch ASoC: pcm3060: Rename output widgets 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 efcf8f9ee39dd4d63fc95d05e460a78605a6cbf2 Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Date: Mon, 12 Nov 2018 08:08:34 +0100 Subject: [PATCH] ASoC: pcm3060: Rename output widgets In the initial commit [1], I added differential output of the codec as separate `+` and `-` widgets: OUTL+ OUTR+ OUTL- OUTR- Later, in the commit [2], I added a device tree property to configure the output as single-ended or differential. Having this property, the `+` and `-` separation in widgets seems for me confusing. There are no functional benefits in such separation, so I find reasonable to get rid of it: OUTL OUTR The new naming is more friendly for sound cards, and is better aligned with other codec drivers in kernel. Renaming the output widgets now should not be a problem from the backwards- compatibility perspective, as the driver for PCM3060 is added into the mainline very recently, and did not yet appear in any releases. [1] commit 6ee47d4a8dac ("ASoC: pcm3060: Add codec driver") [2] commit a78c62de00d5 ("ASoC: pcm3060: Add DT property for single-ended output") Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm3060.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 494d9d662be8..771b46e1974b 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -198,20 +198,16 @@ static const struct snd_kcontrol_new pcm3060_dapm_controls[] = { }; static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = { - SND_SOC_DAPM_OUTPUT("OUTL+"), - SND_SOC_DAPM_OUTPUT("OUTR+"), - SND_SOC_DAPM_OUTPUT("OUTL-"), - SND_SOC_DAPM_OUTPUT("OUTR-"), + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), SND_SOC_DAPM_INPUT("INL"), SND_SOC_DAPM_INPUT("INR"), }; static const struct snd_soc_dapm_route pcm3060_dapm_map[] = { - { "OUTL+", NULL, "Playback" }, - { "OUTR+", NULL, "Playback" }, - { "OUTL-", NULL, "Playback" }, - { "OUTR-", NULL, "Playback" }, + { "OUTL", NULL, "Playback" }, + { "OUTR", NULL, "Playback" }, { "Capture", NULL, "INL" }, { "Capture", NULL, "INR" }, -- 2.19.1

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[alsa-devel] Applied "ASoC: pcm3060: Rename output widgets" to the asoc tree
by Mark Brown 14 Nov '18

14 Nov '18

The patch ASoC: pcm3060: Rename output widgets 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 933a95496e48bde40e366a79e6dd9f9740905573 Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Date: Mon, 12 Nov 2018 08:08:34 +0100 Subject: [PATCH] ASoC: pcm3060: Rename output widgets In the initial commit [1], I added differential output of the codec as separate `+` and `-` widgets: OUTL+ OUTR+ OUTL- OUTR- Later, in the commit [2], I added a device tree property to configure the output as single-ended or differential. Having this property, the `+` and `-` separation in widgets seems for me confusing. There are no functional benefits in such separation, so I find reasonable to get rid of it: OUTL OUTR The new naming is more friendly for sound cards, and is better aligned with other codec drivers in kernel. Renaming the output widgets now should not be a problem from the backwards- compatibility perspective, as the driver for PCM3060 is added into the mainline very recently, and did not yet appear in any releases. [1] commit 6ee47d4a8dac ("ASoC: pcm3060: Add codec driver") [2] commit a78c62de00d5 ("ASoC: pcm3060: Add DT property for single-ended output") Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm3060.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 494d9d662be8..771b46e1974b 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -198,20 +198,16 @@ static const struct snd_kcontrol_new pcm3060_dapm_controls[] = { }; static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = { - SND_SOC_DAPM_OUTPUT("OUTL+"), - SND_SOC_DAPM_OUTPUT("OUTR+"), - SND_SOC_DAPM_OUTPUT("OUTL-"), - SND_SOC_DAPM_OUTPUT("OUTR-"), + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), SND_SOC_DAPM_INPUT("INL"), SND_SOC_DAPM_INPUT("INR"), }; static const struct snd_soc_dapm_route pcm3060_dapm_map[] = { - { "OUTL+", NULL, "Playback" }, - { "OUTR+", NULL, "Playback" }, - { "OUTL-", NULL, "Playback" }, - { "OUTR-", NULL, "Playback" }, + { "OUTL", NULL, "Playback" }, + { "OUTR", NULL, "Playback" }, { "Capture", NULL, "INL" }, { "Capture", NULL, "INR" }, -- 2.19.1

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[alsa-devel] [PATCH] ASoC: dapm: Use less aggressive caching to ensure correctness
by Tzung-Bi Shih 14 Nov '18

14 Nov '18

Commit 92a99ea439c4 ("ASoC: dapm: Use more aggressive caching") only invalidates necessary caches to improve the cache hit rate. However, in snd_soc_dapm_add_path(), the cache is too aggressive which results in wrong state of the audio map. E.g. imaging the following examples during snd_soc_instantiate_card(): time 1: at the beginning in:-1 in:-1 in:-1 out:-1 out:-1 out:-1 A -----> B Spk time 2: after someone called snd_soc_dapm_new_widgets() (e.g. create_fill_widget_route_map() in sound/soc/codecs/hdac_hdmi.c) in:0 in:0 in:0 out:0 out:0 out:1 A ----> B Spk time 3: route added to Spk in:0 in:0 in:0 out:0 out:0 out:1 A ----> B ----> Spk time N: route added from SIGGEN in:1 in:0 in:0 in:0 out:0 out:0 out:0 out:1 SIGGEN ----> A ----> B ----> Spk In the end, the path will not power on but it should be. At time 2, the caches have been polluted. At time 3, it will not get the correct state if snd_soc_dapm_add_path() does not invalidate "out" of B and A. time 3 (expected result): in:0 in:0 in:0 out:1 out:1 out:1 A ----> B ----> Spk Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a5178845065b..2914f3adb333 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2722,7 +2722,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(widgets[dir], "Route added"); } - if (dapm->card->instantiated && path->connect) + if (path->connect) dapm_path_invalidate(path); return 0; -- 2.19.1.930.g4563a0d9d0-goog

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[alsa-devel] [PATCH v7 0/4] Add support for audiocodec in Allwinner A64
by Vasily Khoruzhick 14 Nov '18

14 Nov '18

v2: - Use simple-amplifier for speaker amp on Pinebook - Rename sun50i-a64-i2s to sun50i-a64-codec-i2s to preserve compatible string for other 3 I2S modules in A64 in case if there's any incompatibility with H3 v3: - renamed sunxi-adda-pr-regmap to sun8i-adda-pr-regmap - use ilog2() to calculate reg value for LRCK div instead of using a table v4: - dts: don't use 'Mic' and 'Headset Mic' widgets from sun8i-codec, define our board-level widgets instead. v5: - collect all the tags v6: - driver patches has been merged through ASoC tree - rebase onto 4.20-rc1 - Drop 'Speaker' from routes on sopine and pine64, they don't have speaker. v7: - drop 'Mic' and 'Headset Mic' from dtsi - move AIF1 Slot 0 ADC to ADC route to sun8i-codec - drop BIAS routes for Pine64 and SoPine Vasily Khoruzhick (4): ASoC: sun8i-codec: add missing route for ADC arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../boot/dts/allwinner/sun50i-a64-pine64.dts | 25 +++++++++ .../dts/allwinner/sun50i-a64-pinebook.dts | 40 ++++++++++++++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 25 +++++++++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 54 +++++++++++++++++++ sound/soc/sunxi/sun8i-codec.c | 6 ++- 5 files changed, 149 insertions(+), 1 deletion(-) -- 2.19.1

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[alsa-devel] [PATCH] ASoC: qcom: Set dai_link id to each dai_link
by Rohit kumar 14 Nov '18

14 Nov '18

Frontend dai_link id is used for closing ADM sessions. During concurrent usecase when one session is closed, it closes other ADM session associated with other usecase too. Dai_link->id should always point to Frontend dai id. Set cpu_dai id as dai_link id to fix the issue. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- sound/soc/qcom/common.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index eb1b9da..4715527 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -13,6 +13,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) struct device_node *cpu = NULL; struct device *dev = card->dev; struct snd_soc_dai_link *link; + struct of_phandle_args args; int ret, num_links; ret = snd_soc_of_parse_card_name(card, "model"); @@ -47,12 +48,14 @@ int qcom_snd_parse_of(struct snd_soc_card *card) goto err; } - link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); - if (!link->cpu_of_node) { + ret = of_parse_phandle_with_args(cpu, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { dev_err(card->dev, "error getting cpu phandle\n"); - ret = -EINVAL; goto err; } + link->cpu_of_node = args.np; + link->id = args.args[0]; ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); if (ret) { -- 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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