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[alsa-devel] [PATCH 0/2] Add support for i.MX8QM
by Daniel Baluta 21 Aug '19

21 Aug '19

i.MX8QM SOC integrates 6 SAI instances. Register map is similar with i.MX6 series. Daniel Baluta (2): ASoC: fsl_sai: Add support for imx8qm ASoC: dt-bindings: Introduce compatible string for imx8qm Documentation/devicetree/bindings/sound/fsl-sai.txt | 3 ++- sound/soc/fsl/fsl_sai.c | 7 +++++++ 2 files changed, 9 insertions(+), 1 deletion(-) -- 2.17.1

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[alsa-devel] [PATCH] ASoC: SOF: topology: fix get control data return type and arguments
by Jaska Uimonen 21 Aug '19

21 Aug '19

sof_get_control_data returns negative values even though the return value is defined unsigned (size_t). So change the return value type to int. Add the data size as pointer argument to sof_get_control_data to avoid ambiquity in the meaning of the return type. Also make the sof_get_control_data to return -EINVAL if data pointer is valid but the size is for some reason zero. Fixes: cac974a51ebb ("ASoC: SOF: topology: use set_get_data in process load") Reported by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Jaska Uimonen <jaska.uimonen(a)intel.com> --- sound/soc/sof/topology.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 28a7a6e06a53..8288b1758542 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1752,15 +1752,15 @@ static int sof_widget_load_siggen(struct snd_soc_component *scomp, int index, return ret; } -static size_t sof_get_control_data(struct snd_sof_dev *sdev, - struct snd_soc_dapm_widget *widget, - struct sof_widget_data *wdata) +static int sof_get_control_data(struct snd_sof_dev *sdev, + struct snd_soc_dapm_widget *widget, + struct sof_widget_data *wdata, + size_t *size) { const struct snd_kcontrol_new *kc; struct soc_mixer_control *sm; struct soc_bytes_ext *sbe; struct soc_enum *se; - size_t size = 0; int i; for (i = 0; i < widget->num_kcontrols; i++) { @@ -1800,7 +1800,11 @@ static size_t sof_get_control_data(struct snd_sof_dev *sdev, if (wdata[i].pdata->magic != SOF_ABI_MAGIC) return -EINVAL; - size += wdata[i].pdata->size; + /* don't accept 0 size for data */ + if (!wdata[i].pdata->size) + return -EINVAL; + + *size += wdata[i].pdata->size; /* get data type */ switch (wdata[i].control->cmd) { @@ -1819,7 +1823,7 @@ static size_t sof_get_control_data(struct snd_sof_dev *sdev, } } - return size; + return 0; } static int sof_process_load(struct snd_soc_component *scomp, int index, @@ -1855,12 +1859,11 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, return -ENOMEM; /* get possible component controls and get size of all pdata */ - ipc_data_size = sof_get_control_data(sdev, widget, wdata); + ret = sof_get_control_data(sdev, widget, wdata, + &ipc_data_size); - if (ipc_data_size <= 0) { - ret = ipc_data_size; + if (ret < 0) goto out; - } } ipc_size = sizeof(struct sof_ipc_comp_process) + -- 2.13.6 --------------------------------------------------------------------- Intel Finland Oy Registered Address: PL 281, 00181 Helsinki Business Identity Code: 0357606 - 4 Domiciled in Helsinki This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

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[alsa-devel] [PATCH v5 00/15] ASoC: sun4i-i2s: Updates to the driver
by codekipper＠gmail.com 21 Aug '19

21 Aug '19

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, here is a patch series to add some improvements to the sun4i-i2s driver found whilst getting slave clocking and hdmi audio working on the newer SoCs. As the LibreELEC project is progressing extremely well then there has been some activity getting H6 SoC support with surround sound working and these changes are also included. The functionality included with the new patch set has been extended to cover more sample resolutions, multi-lane data output for HDMI audio and some bug fixes that have been discovered along the way. I have changed some of the original reg fields into function calls as this made it easier to setup for multi-channel audio especially across different SoCs. I can see more usage of the tdm property since I last attempted to push these patches and the examples currently in mainline sort of the opposite to what I'm trying to achieve. When we first started looking at the i2s driver, the codecs that we were using allowed for the frame width to be determined based on the sampling resolution but in most use cases it seems that a fixed width is required(my highest priority should be to get HDMI audio support in). We're using the tdm property to override the old way to calculate the frame width. What I've seen in what has already been mainlined is that the i2s driver has a frame width that is fixed to 32 bits and this can be overridden using the tdm property. My test branch for this can be found at https://github.com/codekipper/linux-sunxi/commits/upstream-i2s , I've been using a Pine64 to test with; validating the new SoC block with HDMI audio and ensuring that I've not broken the old block by making sure that the audio codec still works. If we able to get the first three patches delivered then that is enough for HDMI audio support on the newer SoCs(H3, A64 etc). I still need to investigate the FIFO syncing issues which i've not had a chance to change or address the concerns that broonie and wens brought up. This change has been moved to the top of the patch stack. I would also like to make the multi-channel audio and audio mapping more configurable via the device tree. Currently what is implemented suites our current needs. BR, CK --- v5 changes compared to v4 are: - removed delivered patches. - Added more details to commit messages. - replaced some reg fields with function calls. - Added DSP_A and DSP_B support for H3 and later SoCs. - Added support for the Allwinner H6. v4 changes compared to v3 are: - Moved patches around so that the more controversial of patches are at the top of the stack. - Added more details to commit messages. - Fixed 20bit audio PCM format to use 4 bytes. - Reduced number of flags used to indicate a new SoC. v3 changes compared to v2 are: - added back slave mode changes - added back the use of tdm properties - changes to regmap and caching - removed loopback functionality - fixes to the channel offset mask v2 changes compared to v1 are: - removed slave mode changes which didn't set mclk and bclk div. - removed use of tdm and now use a dedicated property. - fix commit message to better explain reason for sign extending - add divider calculations for newer SoCs. - add support for multi-lane i2s data output. - add support for 20, 24 and 32 bit samples. - add loopback property so blocks can be tested without a codec. --- Jernej Skrabec (3): clk: sunxi-ng: h6: Allow I2S to change parent rate dt-bindings: ASoC: sun4i-i2s: Add H6 compatible ASoC: sun4i-i2s: Add support for H6 I2S Marcus Cooper (12): ASoC: sun4i-i2s: Add regmap field to sign extend sample ASoC: sun4i-i2s: Add set_tdm_slot functionality ASoC: sun4i-i2s: Correct divider calculations ASoC: sun4i-i2s: Support more formats on newer SoCs ASoC: sun4i-i2s: Add functions for RX and TX channel offsets ASoC: sun4i-i2s: Add functions for RX and TX channel enables ASoC: sun4i-i2s: Add functions for RX and TX channel selects ASoC: sun4i-i2s: Add functions for channel mapping ASoC: sun4i-i2s: Add multi-lane functionality ASoC: sun4i-i2s: Add multichannel functionality ASoc: sun4i-i2s: Add 20, 24 and 32 bit support ASoC: sun4i-i2s: Adjust regmap settings .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + drivers/clk/sunxi-ng/ccu-sun50i-h6.c | 8 +- sound/soc/sunxi/sun4i-i2s.c | 690 ++++++++++++++---- 3 files changed, 542 insertions(+), 158 deletions(-) -- 2.22.0

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[alsa-devel] [PATCH] ALSA: usb-audio: Fix invalid NULL check in snd_emuusb_set_samplerate()
by Takashi Iwai 21 Aug '19

21 Aug '19

The quirk function snd_emuusb_set_samplerate() has a NULL check for the mixer element, but this is useless in the current code. It used to be a check against mixer->id_elems[unitid] but it was changed later to the value after mixer_eleme_list_to_info() which is always non-NULL due to the container_of() usage. This patch fixes the check before the conversion. While we're at it, correct a typo in the comment in the function, too. Fixes: 8c558076c740 ("ALSA: usb-audio: Clean up mixer element list traverse") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_quirks.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199fa157a411..27dcb3743690 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1155,17 +1155,17 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, { struct usb_mixer_interface *mixer; struct usb_mixer_elem_info *cval; - int unitid = 12; /* SamleRate ExtensionUnit ID */ + int unitid = 12; /* SampleRate ExtensionUnit ID */ list_for_each_entry(mixer, &chip->mixer_list, list) { - cval = mixer_elem_list_to_info(mixer->id_elems[unitid]); - if (cval) { + if (mixer->id_elems[unitid]) { + cval = mixer_elem_list_to_info(mixer->id_elems[unitid]); snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, cval->control << 8, samplerate_id); snd_usb_mixer_notify_id(mixer, unitid); + break; } - break; } } -- 2.16.4

1 0

[alsa-devel] Master volume not controlling applications' volume.
by Martín Martínez Rivera 21 Aug '19

21 Aug '19

Changing the master volume in pavucontrol (or alsamixer) does not have any effect. The only way to change the volume is to go into the playback tab and change it for each application. The volume only changes when I set the master volume to 0 or when I increase it over 100%. In between there aren't any changes. Running arch on a Lenovo X1 Carbon if that's of any use. I am using the arch default configuration files. I posted somewhere else and was told to email this list as it could be a kernel bug. kernel: 5.2.9-arch1-1-ARCH pactl info output: Server String: /run/user/1000/pulse/native Library Protocol Version: 32 Server Protocol Version: 32 Is Local: yes Client Index: 13 Tile Size: 65472 User Name: martinmr Host Name: martinmr-laptop Server Name: pulseaudio Server Version: 12.2 Default Sample Specification: s16le 2ch 44100Hz Default Channel Map: front-left,front-right Default Sink: alsa_output.pci-0000_00_1f.3.analog-stereo Default Source: alsa_output.pci-0000_00_1f.3.analog-stereo.monitor Cookie: 1afa:f937 Let me know if the output from some other command is needed. Best, Martin

1 0

[alsa-devel] Applied "ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk" to the asoc tree
by Mark Brown 20 Aug '19

20 Aug '19

The patch ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4 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 d5e120422db8808e1c8b1507900ca393a877c58f Mon Sep 17 00:00:00 2001 From: Daniel Stuart <daniel.stuart14(a)gmail.com> Date: Thu, 15 Aug 2019 14:12:55 -0300 Subject: [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk Every single baytrail chromebook sets PMC to 0, as can be seeing below by searching through coreboot source code: $ grep -rl "PMC_PLT_CLK\[0\]" . ./rambi/variants/glimmer/devicetree.cb ./rambi/variants/clapper/devicetree.cb ./rambi/variants/swanky/devicetree.cb ./rambi/variants/enguarde/devicetree.cb ./rambi/variants/winky/devicetree.cb ./rambi/variants/kip/devicetree.cb ./rambi/variants/squawks/devicetree.cb ./rambi/variants/orco/devicetree.cb ./rambi/variants/ninja/devicetree.cb ./rambi/variants/heli/devicetree.cb ./rambi/variants/sumo/devicetree.cb ./rambi/variants/banjo/devicetree.cb ./rambi/variants/candy/devicetree.cb ./rambi/variants/gnawty/devicetree.cb ./rambi/variants/rambi/devicetree.cb ./rambi/variants/quawks/devicetree.cb Plus, Cyan (only non-baytrail chromebook with max98090) also needs this patch for audio to work. Thus, this commit adds all the missing devices to bsw_max98090 quirk table, implemented by commit a182ecd3809c ("ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0"). Signed-off-by: Daniel Stuart <daniel.stuart14(a)gmail.com> Link: https://lore.kernel.org/r/20190815171300.30126-1-daniel.stuart14@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 98 ++++++++++++++++++++ 1 file changed, 98 insertions(+) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 1db9a95e6a79..eaf3e2208a06 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -398,6 +398,20 @@ static struct snd_soc_card snd_soc_card_cht = { }; static const struct dmi_system_id cht_max98090_quirk_table[] = { + { + /* Banjo model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Banjo"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Candy model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Candy"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Clapper model Chromebook */ .matches = { @@ -405,6 +419,27 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { }, .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, }, + { + /* Cyan model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Cyan"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Enguarde model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Enguarde"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Glimmer model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Glimmer"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Gnawty model Chromebook (Acer Chromebook CB3-111) */ .matches = { @@ -412,6 +447,62 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { }, .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, }, + { + /* Heli model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Heli"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Kip model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Kip"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Ninja model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Ninja"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Orco model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Orco"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Quawks model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Quawks"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Rambi model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Rambi"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Squawks model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Squawks"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Sumo model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Sumo"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Swanky model Chromebook (Toshiba Chromebook 2) */ .matches = { @@ -419,6 +510,13 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { }, .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, }, + { + /* Winky model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Winky"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, {} }; -- 2.20.1

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[alsa-devel] [PATCH 0/3] ASoC: mchp-i2s-mcc: Several fixes
by Codrin Ciubotariu 20 Aug '19

20 Aug '19

This pathset fixes some issues detected while testing some more the Microchip I2S multichannel controller. The first two patches fix some issues that appear mostly when hw_free() and hw_params() callbacks are called multiple times. The third patch fixes a problem caused when the controller is in master mode and both capture and playback are played at the same time. All three patches have a "Fixes" tag. Although they are independent, some conflicts might appear if they are not applied in the order presented in this patchset. If so, please let me know so I can rebase them. Codrin Ciubotariu (3): ASoC: mchp-i2s-mcc: Wait for RX/TX RDY only if controller is running ASoC: mchp-i2s-mcc: Fix unprepare of GCLK ASoC: mchp-i2s-mcc: Fix simultaneous capture and playback in master mode sound/soc/atmel/mchp-i2s-mcc.c | 111 +++++++++++++++++++-------------- 1 file changed, 64 insertions(+), 47 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] ASoC: meson: axg-tdm-formatter: free reset on device removal
by Jerome Brunet 20 Aug '19

20 Aug '19

Use the devm variant to get the formatter reset so it is properly freed on device removal Fixes: 751bd5db5260 ("ASoC: meson: axg-tdm-formatter: add reset") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdm-formatter.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index 21c735afab35..358c8c0d861c 100644 --- a/sound/soc/meson/axg-tdm-formatter.c +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -325,7 +325,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) } /* Formatter dedicated reset line */ - formatter->reset = reset_control_get_optional_exclusive(dev, NULL); + formatter->reset = devm_reset_control_get_optional_exclusive(dev, NULL); if (IS_ERR(formatter->reset)) { ret = PTR_ERR(formatter->reset); if (ret != -EPROBE_DEFER) -- 2.21.0

2 1

[alsa-devel] [PATCH] ASoC: meson: g12a-tohdmitx: require regmap mmio
by Jerome Brunet 20 Aug '19

20 Aug '19

The tohdmitx glue uses regmap MMIO so it should require it. Fixes: c8609f3870f7 ("ASoC: meson: add g12a tohdmitx control") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 63b38c123103..2e3676147cea 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -87,6 +87,7 @@ config SND_MESON_AXG_PDM config SND_MESON_G12A_TOHDMITX tristate "Amlogic G12A To HDMI TX Control Support" + select REGMAP_MMIO imply SND_SOC_HDMI_CODEC help Select Y or M to add support for HDMI audio on the g12a SoC -- 2.21.0

2 1

[alsa-devel] Applied "ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk" to the asoc tree
by Mark Brown 20 Aug '19

20 Aug '19

The patch ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk 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 404be07f4ed27697f5fa69162f67a94555738595 Mon Sep 17 00:00:00 2001 From: Daniel Stuart <daniel.stuart14(a)gmail.com> Date: Thu, 15 Aug 2019 14:12:55 -0300 Subject: [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk Every single baytrail chromebook sets PMC to 0, as can be seeing below by searching through coreboot source code: $ grep -rl "PMC_PLT_CLK\[0\]" . ./rambi/variants/glimmer/devicetree.cb ./rambi/variants/clapper/devicetree.cb ./rambi/variants/swanky/devicetree.cb ./rambi/variants/enguarde/devicetree.cb ./rambi/variants/winky/devicetree.cb ./rambi/variants/kip/devicetree.cb ./rambi/variants/squawks/devicetree.cb ./rambi/variants/orco/devicetree.cb ./rambi/variants/ninja/devicetree.cb ./rambi/variants/heli/devicetree.cb ./rambi/variants/sumo/devicetree.cb ./rambi/variants/banjo/devicetree.cb ./rambi/variants/candy/devicetree.cb ./rambi/variants/gnawty/devicetree.cb ./rambi/variants/rambi/devicetree.cb ./rambi/variants/quawks/devicetree.cb Plus, Cyan (only non-baytrail chromebook with max98090) also needs this patch for audio to work. Thus, this commit adds all the missing devices to bsw_max98090 quirk table, implemented by commit a182ecd3809c ("ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0"). Signed-off-by: Daniel Stuart <daniel.stuart14(a)gmail.com> Link: https://lore.kernel.org/r/20190815171300.30126-1-daniel.stuart14@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 98 ++++++++++++++++++++ 1 file changed, 98 insertions(+) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 33eb72545be6..83b978e7b4c4 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -399,6 +399,20 @@ static struct snd_soc_card snd_soc_card_cht = { }; static const struct dmi_system_id cht_max98090_quirk_table[] = { + { + /* Banjo model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Banjo"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Candy model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Candy"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Clapper model Chromebook */ .matches = { @@ -406,6 +420,27 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { }, .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, }, + { + /* Cyan model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Cyan"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Enguarde model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Enguarde"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Glimmer model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Glimmer"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Gnawty model Chromebook (Acer Chromebook CB3-111) */ .matches = { @@ -413,6 +448,62 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { }, .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, }, + { + /* Heli model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Heli"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Kip model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Kip"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Ninja model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Ninja"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Orco model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Orco"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Quawks model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Quawks"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Rambi model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Rambi"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Squawks model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Squawks"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Sumo model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Sumo"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Swanky model Chromebook (Toshiba Chromebook 2) */ .matches = { @@ -420,6 +511,13 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { }, .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, }, + { + /* Winky model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Winky"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, {} }; -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: uniphier: Fix double reset assersion when transitioning to suspend state
by Kunihiko Hayashi 20 Aug '19

20 Aug '19

When transitioning to supend state, uniphier_aio_dai_suspend() is called and asserts reset lines and disables clocks. However, if there are two or more DAIs, uniphier_aio_dai_suspend() are called multiple times, and double reset assersion will cause. This patch defines the counter that has the number of DAIs at first, and whenever uniphier_aio_dai_suspend() are called, it decrements the counter. And only if the counter is zero, it asserts reset lines and disables clocks. In the same way, uniphier_aio_dai_resume() are called, it increments the counter after deasserting reset lines and enabling clocks. Fixes: 139a34200233 ("ASoC: uniphier: add support for UniPhier AIO CPU DAI driver") Signed-off-by: Kunihiko Hayashi <hayashi.kunihiko(a)socionext.com> --- sound/soc/uniphier/aio-cpu.c | 31 +++++++++++++++++++++---------- sound/soc/uniphier/aio.h | 1 + 2 files changed, 22 insertions(+), 10 deletions(-) diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c index ee90e6c..2ae582a 100644 --- a/sound/soc/uniphier/aio-cpu.c +++ b/sound/soc/uniphier/aio-cpu.c @@ -424,8 +424,11 @@ int uniphier_aio_dai_suspend(struct snd_soc_dai *dai) { struct uniphier_aio *aio = uniphier_priv(dai); - reset_control_assert(aio->chip->rst); - clk_disable_unprepare(aio->chip->clk); + aio->chip->num_wup_aios--; + if (!aio->chip->num_wup_aios) { + reset_control_assert(aio->chip->rst); + clk_disable_unprepare(aio->chip->clk); + } return 0; } @@ -439,13 +442,15 @@ int uniphier_aio_dai_resume(struct snd_soc_dai *dai) if (!aio->chip->active) return 0; - ret = clk_prepare_enable(aio->chip->clk); - if (ret) - return ret; + if (!aio->chip->num_wup_aios) { + ret = clk_prepare_enable(aio->chip->clk); + if (ret) + return ret; - ret = reset_control_deassert(aio->chip->rst); - if (ret) - goto err_out_clock; + ret = reset_control_deassert(aio->chip->rst); + if (ret) + goto err_out_clock; + } aio_iecout_set_enable(aio->chip, true); aio_chip_init(aio->chip); @@ -458,7 +463,7 @@ int uniphier_aio_dai_resume(struct snd_soc_dai *dai) ret = aio_init(sub); if (ret) - goto err_out_clock; + goto err_out_reset; if (!sub->setting) continue; @@ -466,11 +471,16 @@ int uniphier_aio_dai_resume(struct snd_soc_dai *dai) aio_port_reset(sub); aio_src_reset(sub); } + aio->chip->num_wup_aios++; return 0; +err_out_reset: + if (!aio->chip->num_wup_aios) + reset_control_assert(aio->chip->rst); err_out_clock: - clk_disable_unprepare(aio->chip->clk); + if (!aio->chip->num_wup_aios) + clk_disable_unprepare(aio->chip->clk); return ret; } @@ -619,6 +629,7 @@ int uniphier_aio_probe(struct platform_device *pdev) return PTR_ERR(chip->rst); chip->num_aios = chip->chip_spec->num_dais; + chip->num_wup_aios = chip->num_aios; chip->aios = devm_kcalloc(dev, chip->num_aios, sizeof(struct uniphier_aio), GFP_KERNEL); diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h index ca6ccba..a7ff7e5 100644 --- a/sound/soc/uniphier/aio.h +++ b/sound/soc/uniphier/aio.h @@ -285,6 +285,7 @@ struct uniphier_aio_chip { struct uniphier_aio *aios; int num_aios; + int num_wup_aios; struct uniphier_aio_pll *plls; int num_plls; -- 2.7.4

3 2

[alsa-devel] [PATCH v2 1/2] ASoC: wm8904: fix typo in DAPM kcontrol name
by Michał Mirosław 20 Aug '19

20 Aug '19

Trivial fix for typo in "Capture Inverting Mux"es' name. Signed-off-by: Michał Mirosław <mirq-linux(a)rere.qmqm.pl> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- v2: no changes --- sound/soc/codecs/wm8904.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 5ebdd1d9afde..525e4ef654a1 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -866,7 +866,7 @@ static SOC_ENUM_SINGLE_DECL(lin_inv_enum, WM8904_ANALOGUE_LEFT_INPUT_1, 4, lin_text); static const struct snd_kcontrol_new lin_inv_mux = - SOC_DAPM_ENUM("Left Capture Inveting Mux", lin_inv_enum); + SOC_DAPM_ENUM("Left Capture Inverting Mux", lin_inv_enum); static const char *rin_text[] = { "IN1R", "IN2R", "IN3R" @@ -882,7 +882,7 @@ static SOC_ENUM_SINGLE_DECL(rin_inv_enum, WM8904_ANALOGUE_RIGHT_INPUT_1, 4, rin_text); static const struct snd_kcontrol_new rin_inv_mux = - SOC_DAPM_ENUM("Right Capture Inveting Mux", rin_inv_enum); + SOC_DAPM_ENUM("Right Capture Inverting Mux", rin_inv_enum); static const char *aif_text[] = { "Left", "Right" -- 2.20.1

1 1

[alsa-devel] [PATCH] fix odd_ptr_err.cocci warnings
by Julia Lawall 20 Aug '19

20 Aug '19

From: kbuild test robot <lkp(a)intel.com> PTR_ERR should normally access the value just tested by IS_ERR Generated by: scripts/coccinelle/tests/odd_ptr_err.cocci Fixes: 62c9c1442c8f ("j721e new machine driver wip") Signed-off-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Julia Lawall <julia.lawall(a)lip6.fr> --- tree: https://github.com/omap-audio/linux-audio peter/ti-linux-4.19.y/wip head: 62c9c1442c8f61ca93e62e1a9d8318be0abd9d9a commit: 62c9c1442c8f61ca93e62e1a9d8318be0abd9d9a [34/34] j721e new machine driver wip :::::: branch date: 20 hours ago :::::: commit date: 20 hours ago j721e-evm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -283,7 +283,7 @@ static int j721e_get_clocks(struct platf parent = devm_clk_get(&pdev->dev, clk_name); kfree(clk_name); if (IS_ERR(parent)) { - ret = PTR_ERR(clocks->parent); + ret = PTR_ERR(parent); if (ret != -EPROBE_DEFER) dev_err(&pdev->dev, "failed to acquire %s': %d\n", prefix, ret); @@ -299,7 +299,7 @@ static int j721e_get_clocks(struct platf parent = devm_clk_get(&pdev->dev, clk_name); kfree(clk_name); if (IS_ERR(parent)) { - ret = PTR_ERR(clocks->parent); + ret = PTR_ERR(parent); if (ret != -EPROBE_DEFER) dev_err(&pdev->dev, "failed to acquire %s': %d\n", prefix, ret);

4 5

[alsa-devel] [PATCH 1/2] ASoC: wm8904: fix typo in DAPM kcontrol name
by Michał Mirosław 20 Aug '19

20 Aug '19

Trivial fix for typo in "Capture Inverting Mux"es' name. Signed-off-by: Michał Mirosław <mirq-linux(a)rere.qmqm.pl> --- sound/soc/codecs/wm8904.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 5ebdd1d9afde..525e4ef654a1 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -866,7 +866,7 @@ static SOC_ENUM_SINGLE_DECL(lin_inv_enum, WM8904_ANALOGUE_LEFT_INPUT_1, 4, lin_text); static const struct snd_kcontrol_new lin_inv_mux = - SOC_DAPM_ENUM("Left Capture Inveting Mux", lin_inv_enum); + SOC_DAPM_ENUM("Left Capture Inverting Mux", lin_inv_enum); static const char *rin_text[] = { "IN1R", "IN2R", "IN3R" @@ -882,7 +882,7 @@ static SOC_ENUM_SINGLE_DECL(rin_inv_enum, WM8904_ANALOGUE_RIGHT_INPUT_1, 4, rin_text); static const struct snd_kcontrol_new rin_inv_mux = - SOC_DAPM_ENUM("Right Capture Inveting Mux", rin_inv_enum); + SOC_DAPM_ENUM("Right Capture Inverting Mux", rin_inv_enum); static const char *aif_text[] = { "Left", "Right" -- 2.20.1

2 3

[alsa-devel] [PATCH v2] ALSA: hda/ca0132 - Add new SBZ quirk
by Paweł Rekowski 20 Aug '19

20 Aug '19

This patch adds a new PCI subsys ID for the SBZ, as found and tested by me and some reddit users. Signed-off-by: Paweł Rekowski <p.rekowski(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 c3096796ee05..c41865e1222c 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1175,6 +1175,7 @@ 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, 0x0027, "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), -- 2.23.0

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Add implicit fb quirk for Behringer UFX1604
by Takashi Iwai 20 Aug '19

20 Aug '19

Behringer UFX1604 requires the similar quirk to apply implicit fb like another Behringer model UFX1204 in order to fix the noisy playback. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204631 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 75b96929f76c..e4bbf79de956 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -339,6 +339,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 2; goto add_sync_ep_from_ifnum; + case USB_ID(0x1397, 0x0001): /* Behringer UFX1604 */ case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */ ep = 0x81; ifnum = 1; -- 2.16.4

1 0

[alsa-devel] [PATCH 00/15] ASoC: use modern style for aux_dev
by Kuninori Morimoto 20 Aug '19

20 Aug '19

Hi Mark, aux_dev users I'm sorry to bother you. I know I'm posting soc-core.c cleanup patches, and these are still under reviewing, but I want to post next patches before my summer vacation. These patches support modern style for aux_dev. I think it should work, but I can't test, because my platform doesn't use aux_dev. Please test and review these. These exchange aux_dev "style" only, do nothing from "technical" point of view. Kuninori Morimoto (15): ASoC: soc-core: support snd_soc_dai_link_component for aux_dev ASoC: simple-card: use snd_soc_dai_link_component for aux_dev ASoC: intel: cht_bsw_max98090_ti: use snd_soc_dai_link_component for aux_dev ASoC: mediatek: mt8183-da7219-max98357: use snd_soc_dai_link_component for aux_dev ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: use snd_soc_dai_link_component for aux_dev ASoC: meson: axg-card: use snd_soc_dai_link_component for aux_dev ASoC: rockchip: rockchip_max98090: use snd_soc_dai_link_component for aux_dev ASoC: samsung: neo1973_wm8753: use snd_soc_dai_link_component for aux_dev ASoC: samsung: speyside: use snd_soc_dai_link_component for aux_dev ASoC: samsung: tm2_wm5110: use snd_soc_dai_link_component for aux_dev ASoC: sunxi: sun4i-codec: use snd_soc_dai_link_component for aux_dev ASoC: ti: rx51: use snd_soc_dai_link_component for aux_dev ASoC: soc-core: remove legacy style of aux_dev ASoC: soc-core: add for_each_xxx macro for aux_dev ASoC: soc-core: add soc_unbind_aux_dev() include/sound/soc.h | 16 +++++-- sound/soc/generic/simple-card.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 3 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 6 +-- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 6 +-- sound/soc/meson/axg-card.c | 11 ++--- sound/soc/rockchip/rockchip_max98090.c | 6 +-- sound/soc/samsung/neo1973_wm8753.c | 3 +- sound/soc/samsung/speyside.c | 3 +- sound/soc/samsung/tm2_wm5110.c | 10 ++--- sound/soc/soc-core.c | 52 ++++++++-------------- sound/soc/sunxi/sun4i-codec.c | 14 +++--- sound/soc/ti/rx51.c | 14 +++--- 13 files changed, 66 insertions(+), 80 deletions(-) -- 2.7.4

7 41

[alsa-devel] [PATCH] Fix an OOB bug in uac_mixer_unit_bmControls
by Hui Peng 20 Aug '19

20 Aug '19

`uac_mixer_unit_get_channels` calls `uac_mixer_unit_bmControls` to get pointer to bmControls field. The current implementation of `uac_mixer_unit_get_channels` does properly check the size of uac_mixer_unit_descriptor descriptor and may allow OOB access in `uac_mixer_unit_bmControls`. Reported-by: Hui Peng <benquike(a)gmail.com> Reported-by: Mathias Payer <mathias.payer(a)nebelwelt.net> Signed-off-by: Hui Peng <benquike(a)gmail.com> --- sound/usb/mixer.c | 25 ++++++++++++++++++------- 1 file changed, 18 insertions(+), 7 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index b5927c3d5bc0..00e6274a63c3 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -738,28 +738,39 @@ static int get_cluster_channels_v3(struct mixer_build *state, unsigned int clust static int uac_mixer_unit_get_channels(struct mixer_build *state, struct uac_mixer_unit_descriptor *desc) { - int mu_channels; + int mu_channels = 0; void *c; - if (desc->bLength < sizeof(*desc)) - return -EINVAL; if (!desc->bNrInPins) return -EINVAL; - if (desc->bLength < sizeof(*desc) + desc->bNrInPins) - return -EINVAL; switch (state->mixer->protocol) { case UAC_VERSION_1: + // limit derived from uac_mixer_unit_bmControls + if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 4) + return 0; + + mu_channels = uac_mixer_unit_bNrChannels(desc); + break; + case UAC_VERSION_2: - default: - if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 1) + // limit derived from uac_mixer_unit_bmControls + if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 6) return 0; /* no bmControls -> skip */ + mu_channels = uac_mixer_unit_bNrChannels(desc); break; case UAC_VERSION_3: + // limit derived from uac_mixer_unit_bmControls + if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 2) + return 0; /* no bmControls -> skip */ + mu_channels = get_cluster_channels_v3(state, uac3_mixer_unit_wClusterDescrID(desc)); break; + + default: + break; } if (!mu_channels) -- 2.22.1

1 1

[alsa-devel] [PATCH 0/3] ASoC: SOF: initial support for i.MX8
by Pierre-Louis Bossart 19 Aug '19

19 Aug '19

The reviews for these patches took place already on mailing lists and GitHub, and the code is already integrated in the SOF tree (along with dependencies already accepted in the i.MX tree) As agreed with Daniel, I am sending these patches as part of my weekly updates so that we avoid conflicts with other contributions (the next series adds Intel ALH/SoundWire suport which changes the same files) Daniel Baluta (3): ASoC: SOF: Add OF DSP device support ASoC: SOF: imx: Add i.MX8 HW support ASoC: SOF: topology: Add dummy support for i.MX8 DAIs include/sound/sof/dai.h | 2 + include/uapi/sound/sof/tokens.h | 8 + sound/soc/sof/Kconfig | 11 + sound/soc/sof/Makefile | 4 + sound/soc/sof/imx/Kconfig | 22 ++ sound/soc/sof/imx/Makefile | 4 + sound/soc/sof/imx/imx8.c | 394 ++++++++++++++++++++++++++++++++ sound/soc/sof/sof-of-dev.c | 143 ++++++++++++ sound/soc/sof/topology.c | 30 +++ 9 files changed, 618 insertions(+) create mode 100644 sound/soc/sof/imx/Kconfig create mode 100644 sound/soc/sof/imx/Makefile create mode 100644 sound/soc/sof/imx/imx8.c create mode 100644 sound/soc/sof/sof-of-dev.c -- 2.20.1

5 13

[alsa-devel] [PATCH] ALSA: hda - Fixes inverted Conexant GPIO mic mute led
by jeronimo＠borque.com.ar 19 Aug '19

19 Aug '19

From: Jeronimo Borque <jeronimo(a)borque.com.ar> "enabled" parameter historically referred to the device input or output, not to the led indicator. After the changes added with the led helper functions the mic mute led logic refers to the led and not to the mic input which caused led indicator to be negated (Mic mute led was on when the input enabled) Fixing it in the call to cxt_update_gpio_led at the cxt_gpio_micmute_update hook. Maybe more changes are required to be consistent everywhere. Signed-off-by: Jeronimo Borque <jeronimo(a)borque.com.ar> --- sound/pci/hda/patch_conexant.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f299f137eaea..8edf0d1290b5 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -636,6 +636,10 @@ static void cxt_update_gpio_led(struct hda_codec *codec, unsigned int mask, spec->gpio_led &= ~mask; else spec->gpio_led |= mask; + + codec_dbg(codec, "mask:%d enabled:%d gpio_led:%d\n", + mask, enabled, spec->gpio_led); + if (spec->gpio_led != oldval) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_led); @@ -656,7 +660,7 @@ static void cxt_gpio_micmute_update(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; cxt_update_gpio_led(codec, spec->gpio_mic_led_mask, - spec->gen.micmute_led.led_value); + !spec->gen.micmute_led.led_value); } @@ -669,7 +673,6 @@ static void cxt_fixup_mute_led_gpio(struct hda_codec *codec, { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03 }, {} }; - codec_info(codec, "action: %d gpio_led: %d\n", action, spec->gpio_led); if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = cxt_fixup_gpio_mute_hook; -- 2.21.0

3 9

[alsa-devel] [bug report] ASoC: SOF: topology: use set_get_data in process load
by Dan Carpenter 19 Aug '19

19 Aug '19

Hello Jaska Uimonen, The patch cac974a51ebb: "ASoC: SOF: topology: use set_get_data in process load" from Aug 9, 2019, leads to the following static checker warning: sound/soc/sof/topology.c:1857 sof_process_load() warn: 'ipc_data_size' unsigned <= 0 sound/soc/sof/topology.c 1825 static int sof_process_load(struct snd_soc_component *scomp, int index, 1826 struct snd_sof_widget *swidget, 1827 struct snd_soc_tplg_dapm_widget *tw, 1828 struct sof_ipc_comp_reply *r, 1829 int type) 1830 { 1831 struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); 1832 struct snd_soc_dapm_widget *widget = swidget->widget; 1833 struct snd_soc_tplg_private *private = &tw->priv; 1834 struct sof_ipc_comp_process *process = NULL; 1835 struct sof_widget_data *wdata = NULL; 1836 size_t ipc_data_size = 0; ^^^^^^^^^^^^^^^^^^^^ Unsigned. 1837 size_t ipc_size; 1838 int offset = 0; 1839 int ret = 0; 1840 int i; 1841 1842 if (type == SOF_COMP_NONE) { 1843 dev_err(sdev->dev, "error: invalid process comp type %d\n", 1844 type); 1845 return -EINVAL; 1846 } 1847 1848 /* allocate struct for widget control data sizes and types */ 1849 if (widget->num_kcontrols) { 1850 wdata = kcalloc(widget->num_kcontrols, 1851 sizeof(*wdata), 1852 GFP_KERNEL); 1853 1854 if (!wdata) 1855 return -ENOMEM; 1856 1857 /* get possible component controls and get size of all pdata */ 1858 ipc_data_size = sof_get_control_data(sdev, widget, wdata); 1859 1860 if (ipc_data_size <= 0) { The sof_get_control_data() returns negative error codes, but it's also declared as unsigned. But the main reason I'm reporting this instead of fixing it myself (besides laziness) is that I was wondering if it was really handling zero returns correctly. Should we really return success in that situation? 1861 ret = ipc_data_size; 1862 goto out; 1863 } 1864 } 1865 regards, dan carpenter

1 0

[alsa-devel] es8316 multiple issues
by H. C. Dornelas 19 Aug '19

19 Aug '19

Hi. Using only ALSA on my Positivo 2 em 1 Q432A (motherboard CHT12CP, es8316 sound card, kernel 5.2.9), speakers only work with headphones plugged in. Using Pulseaudio, speakers work without headphones, but many other problems remain: no microphones work; speakers are muted every boot and anytime headphones are unplugged, have no volume control and are only a mute/unmute switch; there are more than 160 controls in alsamixer's bytcht-es8316 card, most having no effect on sound output; and I still couldn't test hdmi audio. Here's the alsa-info file: http://alsa-project.org/db/?f=ee6dd9bdcf11a192fcccba9ab1b7abb2f3999b5d UCM files might solve these issues, but I have no experience with those, so any help would be dearly appreciated. If any other info is needed, just tell me and I'll add it. Thank you very much.

1 0

[alsa-devel] [PATCH] Fix an OOB bug in uac_mixer_unit_bmControls
by Hui Peng 17 Aug '19

17 Aug '19

`uac_mixer_unit_get_channels` calls `uac_mixer_unit_bmControls` to get pointer to bmControls field. The current implementation of `uac_mixer_unit_get_channels` does properly check the size of uac_mixer_unit_descriptor descriptor and may allow OOB access in `uac_mixer_unit_bmControls`. Reported-by: Hui Peng <benquike(a)gmail.com> Reported-by: Mathias Payer <mathias.payer(a)nebelwelt.net> Signed-off-by: Hui Peng <benquike(a)gmail.com> --- sound/usb/mixer.c | 25 ++++++++++++++++++------- 1 file changed, 18 insertions(+), 7 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index b5927c3d5bc0..00e6274a63c3 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -738,28 +738,39 @@ static int get_cluster_channels_v3(struct mixer_build *state, unsigned int clust static int uac_mixer_unit_get_channels(struct mixer_build *state, struct uac_mixer_unit_descriptor *desc) { - int mu_channels; + int mu_channels = 0; void *c; - if (desc->bLength < sizeof(*desc)) - return -EINVAL; if (!desc->bNrInPins) return -EINVAL; - if (desc->bLength < sizeof(*desc) + desc->bNrInPins) - return -EINVAL; switch (state->mixer->protocol) { case UAC_VERSION_1: + // limit derived from uac_mixer_unit_bmControls + if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 4) + return 0; + + mu_channels = uac_mixer_unit_bNrChannels(desc); + break; + case UAC_VERSION_2: - default: - if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 1) + // limit derived from uac_mixer_unit_bmControls + if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 6) return 0; /* no bmControls -> skip */ + mu_channels = uac_mixer_unit_bNrChannels(desc); break; case UAC_VERSION_3: + // limit derived from uac_mixer_unit_bmControls + if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 2) + return 0; /* no bmControls -> skip */ + mu_channels = get_cluster_channels_v3(state, uac3_mixer_unit_wClusterDescrID(desc)); break; + + default: + break; } if (!mu_channels) -- 2.22.1

2 5

Re: [alsa-devel] INFO: rcu detected stall in __do_softirq
by Srinivas Kandagatla 17 Aug '19

17 Aug '19

On 16/08/2019 01:10, syzbot wrote: > syzbot has bisected this bug to: > > commit 2aeac95d1a4cc85aae57ab842d5c3340df0f817f > Author: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> > Date: Tue Jun 11 10:40:41 2019 +0000 > > soundwire: add module_sdw_driver helper macro Not sure how adding a macro with no users triggers this rcu stall. BTW this was in mainline since rc1. --srini > > bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=114b45ee600000 > start commit: 882e8691 Add linux-next specific files for 20190801 > git tree: linux-next > final crash: https://syzkaller.appspot.com/x/report.txt?x=134b45ee600000 > console output: https://syzkaller.appspot.com/x/log.txt?x=154b45ee600000 > kernel config: https://syzkaller.appspot.com/x/.config?x=466b331af3f34e94 > dashboard link: > https://syzkaller.appspot.com/bug?extid=6593c6b8c8b66a07cd98 > syz repro: https://syzkaller.appspot.com/x/repro.syz?x=16b216b2600000 > > Reported-by: syzbot+6593c6b8c8b66a07cd98(a)syzkaller.appspotmail.com > Fixes: 2aeac95d1a4c ("soundwire: add module_sdw_driver helper macro") > > For information about bisection process see: > https://goo.gl/tpsmEJ#bisection

2 1

[alsa-devel] ALSA HW PARAMS GET BUFFER TIME ERROR
by 李松鞠 17 Aug '19

17 Aug '19

Hello! Dear Developers, I am from HNU University (Changsha, China), a fresh man in ALSA sound developing. Recently, a alsa bug( snd_pcm_hw_params_get_buffer_time snd_pcm_hw_params_get_period_time ) occured to me which made me confused! Firstly (Step 1) I set hardware params normally(rate: 44100, channel: 2, format: S16_LE, access: raw_interleaved, period size: not set, buffer size: not set), It was OK. Laterly (Step 2) I get hw_params and print it, it was not surprising that they were the same as what I set (buffer_time == period_time * 1024, buffer_size == period_size * 1024). But! In another code ( Add Step 3)I used snd_pcm_format_name((snd_pcm_format_t)val), snd_pcm_format_description( (snd_pcm_format_t)val) to display the format information, It got wrong buffer size(buffer_time == period_time, buffer_size == period_size * 1024 ). the Attachments are code pcm_setting.cpp(With Step 1, 2), pcm_open_seeting.cpp(With Step 1, 2, 3). My alsa-lib version is about 1.1.0 . pcm_setting.cpp running result: get rate: 44100 get channles: 1 period time: 46439 buffer time is 47554467 period size: 2048 frames buffer size default: 2097152 frames periods per buffer = 1024 frames pcm_open_setting running result: PCM handle name = 'default' PCM state = PREPARED access type = RW_INTERLEAVED format(length: 2) = 'S16_LE' (Signed 16 bit Little Endian) channels = 1 rate = 44100 bps period time = 46438 us buffer time = 46438 us period size = 2048 frames buffer size = 2097152 frames periods per buffer = 1024 frames

1 0

Re: [alsa-devel] [PATCH] ASoC: imx-audmux: Add driver suspend and resume to support MEGA Fast
by S.j. Wang 17 Aug '19

17 Aug '19

Hi Mark > > On Fri, Aug 16, 2019 at 01:03:14AM -0400, Shengjiu Wang wrote: > > > + for (i = 0; i < reg_max; i++) > > + regcache[i] = readl(audmux_base + i * 4); > > If only there were some framework which provided a register cache! 😝 Yes, next step I can refine this driver to use the regmap. Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH] ASoC: imx-audmux: Add driver suspend and resume to support MEGA Fast
by Shengjiu Wang 16 Aug '19

16 Aug '19

For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of modules during system suspend and resume procedure. Thus, AUDMUX needs to save all the values of registers before the system suspend and restore them after the system resume. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/imx-audmux.c | 54 +++++++++++++++++++++++++++++++++++++- 1 file changed, 53 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 7595f24a006e..3ce85a43e08f 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -23,6 +23,8 @@ static struct clk *audmux_clk; static void __iomem *audmux_base; +static u32 *regcache; +static u32 reg_max; #define IMX_AUDMUX_V2_PTCR(x) ((x) * 8) #define IMX_AUDMUX_V2_PDCR(x) ((x) * 8 + 4) @@ -315,8 +317,23 @@ static int imx_audmux_probe(struct platform_device *pdev) if (of_id) pdev->id_entry = of_id->data; audmux_type = pdev->id_entry->driver_data; - if (audmux_type == IMX31_AUDMUX) + + switch (audmux_type) { + case IMX31_AUDMUX: audmux_debugfs_init(); + reg_max = 14; + break; + case IMX21_AUDMUX: + reg_max = 6; + break; + default: + dev_err(&pdev->dev, "unsupported version!\n"); + return -EINVAL; + } + + regcache = devm_kzalloc(&pdev->dev, sizeof(u32) * reg_max, GFP_KERNEL); + if (!regcache) + return -ENOMEM; if (of_id) imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); @@ -332,12 +349,47 @@ static int imx_audmux_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int imx_audmux_suspend(struct device *dev) +{ + int i; + + clk_prepare_enable(audmux_clk); + + for (i = 0; i < reg_max; i++) + regcache[i] = readl(audmux_base + i * 4); + + clk_disable_unprepare(audmux_clk); + + return 0; +} + +static int imx_audmux_resume(struct device *dev) +{ + int i; + + clk_prepare_enable(audmux_clk); + + for (i = 0; i < reg_max; i++) + writel(regcache[i], audmux_base + i * 4); + + clk_disable_unprepare(audmux_clk); + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops imx_audmux_pm = { + SET_SYSTEM_SLEEP_PM_OPS(imx_audmux_suspend, imx_audmux_resume) +}; + static struct platform_driver imx_audmux_driver = { .probe = imx_audmux_probe, .remove = imx_audmux_remove, .id_table = imx_audmux_ids, .driver = { .name = DRIVER_NAME, + .pm = &imx_audmux_pm, .of_match_table = imx_audmux_dt_ids, } }; -- 2.21.0

2 2

[alsa-devel] [PATCH 0/4] ASoC: SOF: i.MX8 SAI and Intel ALH/SoundWire DAI support
by Pierre-Louis Bossart 16 Aug '19

16 Aug '19

This patchset adds the basic functionality to extract DAI information from the topology and send the information to firmware. The i.MX8 SAI/ESAI definitions are minimal placeholders for now, they are included to keep the alignment with the firmware/topology. For the Intel ALH, there is no specific configuration in topology file, all the programming takes place based on the dai index and the related ALH stream ID. The renaming from SoundWire to ALH (Audio Link Hub) is a better representation of the hardware, the ALH acts as a proxy between the DMA transfers and the Cadence IP. Bard liao (1): ASoC: SOF: rename SOUNDWIRE to ALH Daniel Baluta (1): ASoC: SOF: topology: Add dummy support for i.MX8 DAIs Pierre-Louis Bossart (2): ASoC: SOF: topology: initial support for Intel ALH DAI type ASoC: SOF: pcm: add ALH support include/sound/sof/dai.h | 4 +- include/uapi/sound/sof/tokens.h | 8 ++++ sound/soc/sof/pcm.c | 3 ++ sound/soc/sof/topology.c | 72 ++++++++++++++++++++++++++++++++- 4 files changed, 85 insertions(+), 2 deletions(-) -- 2.20.1

2 8

[alsa-devel] [PATCH v2 0/3] soundwire: debugfs support for 5.4
by Pierre-Louis Bossart 16 Aug '19

16 Aug '19

This patchset enables debugfs support and corrects all the feedback provided on an earlier RFC ('soundwire: updates for 5.4') There is one remaining hard-coded value in intel.c that will need to be fixed in a follow-up patchset not specific to debugfs: we need to remove hard-coded Intel-specific configurations from cadence_master.c (PDI offsets, etc). Changes since v1 (Feedback from GKH) Handle debugfs in a more self-contained way (no dentry as return or parameter) Used CONFIG_DEBUG_FS in structures and code to make it easier to remove if need be. No functional change for register dumps. Changes since RFC (Feedback from GKH, Vinod, Guennadi, Cezary, Sanyog): removed error checks used DEFINE_SHOW_ATTRIBUTE and seq_file fixed copyright dates fixed SPDX license info to use GPL2.0 only fixed Makefile to include debugfs only if CONFIG_DEBUG_FS is selected used static inlines for fallback compilation removed intermediate variables removed hard-coded constants in loops (used registers offsets and hardware capabilities) squashed patch 3 Pierre-Louis Bossart (3): soundwire: add debugfs support soundwire: cadence_master: add debugfs register dump soundwire: intel: add debugfs register dump drivers/soundwire/Makefile | 4 + drivers/soundwire/bus.c | 6 ++ drivers/soundwire/bus.h | 16 +++ drivers/soundwire/bus_type.c | 3 + drivers/soundwire/cadence_master.c | 107 ++++++++++++++++++++ drivers/soundwire/cadence_master.h | 4 + drivers/soundwire/debugfs.c | 151 +++++++++++++++++++++++++++++ drivers/soundwire/intel.c | 121 +++++++++++++++++++++++ drivers/soundwire/slave.c | 1 + include/linux/soundwire/sdw.h | 8 ++ 10 files changed, 421 insertions(+) create mode 100644 drivers/soundwire/debugfs.c -- 2.20.1

4 7

[alsa-devel] [PATCH] ASoC: core: Move pcm_mutex up to card level from snd_soc_pcm_runtime
by Peter Ujfalusi 16 Aug '19

16 Aug '19

The pcm_mutex is used to prevent concurrent execution of snd_pcm_ops callbacks. This works fine most of the cases but it can not handle setups when the same DAI is used by different rtd, for example: pcm3168a have two DAIs: one for Playback and one for Capture. If the codec is connected to a single CPU DAI we need to have two dai_link to support both playback and capture. In this case the snd_pcm_ops callbacks can be executed in parallel causing unexpected races in DAI drivers. By moving the pcm_mutex up to card level this can be solved while - hopefully - not breaking other setups. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, I have tested the patch on several boards. it fixes the issue with single card with multiple dai_links where the CPU side (mcasp) is the same. However I can not test with anything which use DPCM. It would be great if the patch would gather few tested-by from folks having access to more complicated setups. The backround of this patch: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-August/153864.ht… Regards, Peter include/sound/soc.h | 6 +++-- sound/soc/soc-compress.c | 48 ++++++++++++++++++++-------------------- sound/soc/soc-core.c | 2 +- sound/soc/soc-pcm.c | 36 +++++++++++++++--------------- 4 files changed, 47 insertions(+), 45 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6ac6481b4882..ec2dbe0ac2aa 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -990,6 +990,10 @@ struct snd_soc_card { struct mutex mutex; struct mutex dapm_mutex; + /* Mutex for PCM operations */ + struct mutex pcm_mutex; + enum snd_soc_pcm_subclass pcm_subclass; + spinlock_t dpcm_lock; bool instantiated; @@ -1107,8 +1111,6 @@ struct snd_soc_pcm_runtime { struct device *dev; struct snd_soc_card *card; struct snd_soc_dai_link *dai_link; - struct mutex pcm_mutex; - enum snd_soc_pcm_subclass pcm_subclass; struct snd_pcm_ops ops; unsigned int params_select; /* currently selected param for dai link */ diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 289211069a1e..9e54d8ae6d2c 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -80,7 +80,7 @@ static int soc_compr_open(struct snd_compr_stream *cstream) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (cpu_dai->driver->cops && cpu_dai->driver->cops->startup) { ret = cpu_dai->driver->cops->startup(cstream, cpu_dai); @@ -108,7 +108,7 @@ static int soc_compr_open(struct snd_compr_stream *cstream) snd_soc_runtime_activate(rtd, cstream->direction); - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return 0; @@ -118,7 +118,7 @@ static int soc_compr_open(struct snd_compr_stream *cstream) if (cpu_dai->driver->cops && cpu_dai->driver->cops->shutdown) cpu_dai->driver->cops->shutdown(cstream, cpu_dai); out: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -224,7 +224,7 @@ static void close_delayed_work(struct work_struct *work) container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); struct snd_soc_dai *codec_dai = rtd->codec_dai; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); dev_dbg(rtd->dev, "Compress ASoC: pop wq checking: %s status: %s waiting: %s\n", @@ -239,7 +239,7 @@ static void close_delayed_work(struct work_struct *work) SND_SOC_DAPM_STREAM_STOP); } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); } static int soc_compr_free(struct snd_compr_stream *cstream) @@ -249,7 +249,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) struct snd_soc_dai *codec_dai = rtd->codec_dai; int stream; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (cstream->direction == SND_COMPRESS_PLAYBACK) stream = SNDRV_PCM_STREAM_PLAYBACK; @@ -292,7 +292,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) SND_SOC_DAPM_STREAM_STOP); } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return 0; } @@ -375,7 +375,7 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); ret = soc_compr_components_trigger(cstream, cmd); if (ret < 0) @@ -394,7 +394,7 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) } out: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -480,7 +480,7 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); /* * First we call set_params for the CPU DAI, then the component @@ -514,14 +514,14 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, /* cancel any delayed stream shutdown that is pending */ rtd->pop_wait = 0; - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); cancel_delayed_work_sync(&rtd->delayed_work); return 0; err: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -593,7 +593,7 @@ static int soc_compr_get_params(struct snd_compr_stream *cstream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (cpu_dai->driver->cops && cpu_dai->driver->cops->get_params) { ret = cpu_dai->driver->cops->get_params(cstream, params, cpu_dai); @@ -613,7 +613,7 @@ static int soc_compr_get_params(struct snd_compr_stream *cstream, } err: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -625,7 +625,7 @@ static int soc_compr_get_caps(struct snd_compr_stream *cstream, struct snd_soc_rtdcom_list *rtdcom; int ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -638,7 +638,7 @@ static int soc_compr_get_caps(struct snd_compr_stream *cstream, break; } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -650,7 +650,7 @@ static int soc_compr_get_codec_caps(struct snd_compr_stream *cstream, struct snd_soc_rtdcom_list *rtdcom; int ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -664,7 +664,7 @@ static int soc_compr_get_codec_caps(struct snd_compr_stream *cstream, break; } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -676,7 +676,7 @@ static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (cpu_dai->driver->cops && cpu_dai->driver->cops->ack) { ret = cpu_dai->driver->cops->ack(cstream, bytes, cpu_dai); @@ -697,7 +697,7 @@ static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes) } err: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -710,7 +710,7 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, int ret = 0; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (cpu_dai->driver->cops && cpu_dai->driver->cops->pointer) cpu_dai->driver->cops->pointer(cstream, tstamp, cpu_dai); @@ -726,7 +726,7 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, break; } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -738,7 +738,7 @@ static int soc_compr_copy(struct snd_compr_stream *cstream, struct snd_soc_rtdcom_list *rtdcom; int ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -751,7 +751,7 @@ static int soc_compr_copy(struct snd_compr_stream *cstream, break; } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bb1e9e2c4ff4..7f2064889a23 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1344,7 +1344,6 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, rtd->dev->groups = soc_dev_attr_groups; dev_set_name(rtd->dev, "%s", name); dev_set_drvdata(rtd->dev, rtd); - mutex_init(&rtd->pcm_mutex); INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients); INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients); @@ -2395,6 +2394,7 @@ int snd_soc_register_card(struct snd_soc_card *card) card->instantiated = 0; mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); + mutex_init(&card->pcm_mutex); spin_lock_init(&card->dpcm_lock); return snd_soc_bind_card(card); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 77c986fe08d0..93bb4a347779 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -36,7 +36,7 @@ * Increments the active count for all the DAIs and components attached to a PCM * runtime. Should typically be called when a stream is opened. * - * Must be called with the rtd->pcm_mutex being held + * Must be called with the rtd->card->pcm_mutex being held */ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) { @@ -44,7 +44,7 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) struct snd_soc_dai *codec_dai; int i; - lockdep_assert_held(&rtd->pcm_mutex); + lockdep_assert_held(&rtd->card->pcm_mutex); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active++; @@ -72,7 +72,7 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) * Decrements the active count for all the DAIs and components attached to a PCM * runtime. Should typically be called when a stream is closed. * - * Must be called with the rtd->pcm_mutex being held + * Must be called with the rtd->card->pcm_mutex being held */ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream) { @@ -80,7 +80,7 @@ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream) struct snd_soc_dai *codec_dai; int i; - lockdep_assert_held(&rtd->pcm_mutex); + lockdep_assert_held(&rtd->card->pcm_mutex); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active--; @@ -506,7 +506,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) pm_runtime_get_sync(component->dev); } - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); /* startup the audio subsystem */ ret = snd_soc_dai_startup(cpu_dai, substream); @@ -604,7 +604,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) snd_soc_runtime_activate(rtd, substream->stream); - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return 0; config_err: @@ -623,7 +623,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) snd_soc_dai_shutdown(cpu_dai, substream); out: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -653,7 +653,7 @@ static void close_delayed_work(struct work_struct *work) container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); struct snd_soc_dai *codec_dai = rtd->codec_dais[0]; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); dev_dbg(rtd->dev, "ASoC: pop wq checking: %s status: %s waiting: %s\n", codec_dai->driver->playback.stream_name, @@ -667,7 +667,7 @@ static void close_delayed_work(struct work_struct *work) SND_SOC_DAPM_STREAM_STOP); } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); } static void codec2codec_close_delayed_work(struct work_struct *work) @@ -694,7 +694,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai; int i; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); snd_soc_runtime_deactivate(rtd, substream->stream); @@ -738,7 +738,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -771,7 +771,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai; int i, ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (rtd->dai_link->ops->prepare) { ret = rtd->dai_link->ops->prepare(substream); @@ -826,7 +826,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream); out: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -876,7 +876,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai; int i, ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { @@ -962,7 +962,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (ret) goto component_err; out: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; component_err: @@ -986,7 +986,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream); - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -1001,7 +1001,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int i; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); /* clear the corresponding DAIs parameters when going to be inactive */ if (cpu_dai->active == 1) { @@ -1043,7 +1043,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) snd_soc_dai_hw_free(cpu_dai, substream); - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return 0; } -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 5

[alsa-devel] [GIT PULL] soundwire fixes for v5.4-rc5
by Vinod Koul 16 Aug '19

16 Aug '19

Hi Greg, We have couple of fixes queued up, please pull. Some more are in review, will send them later. These fixes are in linux-next as well. The following changes since commit 5f9e832c137075045d15cd6899ab0505cfb2ca4b: Linus 5.3-rc1 (2019-07-21 14:05:38 -0700) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git tags/soundwire-5.3-rc5 for you to fetch changes up to 8676b3ca4673517650fd509d7fa586aff87b3c28: soundwire: fix regmap dependencies and align with other serial links (2019-08-09 10:20:40 +0530) ---------------------------------------------------------------- soundwire fixes for v5.3-rc5 Pierre sent fixes which are queued now for v5.3-rc5 are: - regmap dependecy - cadence register definitions ---------------------------------------------------------------- Pierre-Louis Bossart (3): soundwire: cadence_master: fix register definition for SLAVE_STATE soundwire: cadence_master: fix definitions for INTSTAT0/1 soundwire: fix regmap dependencies and align with other serial links drivers/base/regmap/Kconfig | 2 +- drivers/soundwire/Kconfig | 7 +------ drivers/soundwire/Makefile | 2 +- drivers/soundwire/cadence_master.c | 8 ++++---- 4 files changed, 7 insertions(+), 12 deletions(-) Thanks -- ~Vinod

2 1

[alsa-devel] [PATCH v3 1/2] ALSA: hda - Expand pin_match function to match upcoming new tbls
by Hui Wang 16 Aug '19

16 Aug '19

With the existing pintbl, we already have many entries in it. it is better to figure out a new way to reduce the size of the pintbl. We plan to define a new tbl which will match more machines with a single tbl, To do that, this function doesn't need to match all valid pins between machine and tbl, it just needs to match all pins defined in the tbl with the machine. And the plan is to move some tbls from pin_fixup_tbl to fallback_pin_fixup_tbl gradually. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 12 ++++++++---- sound/pci/hda/hda_local.h | 3 ++- sound/pci/hda/patch_realtek.c | 4 ++-- 3 files changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 92390d457567..8a134f750bae 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -884,7 +884,8 @@ EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); #define IGNORE_SEQ_ASSOC (~(AC_DEFCFG_SEQUENCE | AC_DEFCFG_DEF_ASSOC)) static bool pin_config_match(struct hda_codec *codec, - const struct hda_pintbl *pins) + const struct hda_pintbl *pins, + bool match_all_pins) { const struct hda_pincfg *pin; int i; @@ -908,7 +909,8 @@ static bool pin_config_match(struct hda_codec *codec, return false; } } - if (!found && (cfg & 0xf0000000) != 0x40000000) + if (match_all_pins && + !found && (cfg & 0xf0000000) != 0x40000000) return false; } @@ -920,10 +922,12 @@ static bool pin_config_match(struct hda_codec *codec, * @codec: the HDA codec * @pin_quirk: zero-terminated pin quirk list * @fixlist: the fixup list + * @match_all_pins: all valid pins must match with the table entries */ void snd_hda_pick_pin_fixup(struct hda_codec *codec, const struct snd_hda_pin_quirk *pin_quirk, - const struct hda_fixup *fixlist) + const struct hda_fixup *fixlist, + bool match_all_pins) { const struct snd_hda_pin_quirk *pq; @@ -935,7 +939,7 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, continue; if (codec->core.vendor_id != pq->codec) continue; - if (pin_config_match(codec, pq->pins)) { + if (pin_config_match(codec, pq->pins, match_all_pins)) { codec->fixup_id = pq->value; #ifdef CONFIG_SND_DEBUG_VERBOSE codec->fixup_name = pq->name; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 349a8312d06a..3942e1b528d8 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -361,7 +361,8 @@ void snd_hda_pick_fixup(struct hda_codec *codec, const struct hda_fixup *fixlist); void snd_hda_pick_pin_fixup(struct hda_codec *codec, const struct snd_hda_pin_quirk *pin_quirk, - const struct hda_fixup *fixlist); + const struct hda_fixup *fixlist, + bool match_all_pins); /* helper macros to retrieve pin default-config values */ #define get_defcfg_connect(cfg) \ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ac7177ce67eb..2fa03fb6bd9e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7950,7 +7950,7 @@ static int patch_alc269(struct hda_codec *codec) snd_hda_pick_fixup(codec, alc269_fixup_models, alc269_fixup_tbl, alc269_fixups); - snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups); + snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups, true); snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl, alc269_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -8935,7 +8935,7 @@ static int patch_alc662(struct hda_codec *codec) snd_hda_pick_fixup(codec, alc662_fixup_models, alc662_fixup_tbl, alc662_fixups); - snd_hda_pick_pin_fixup(codec, alc662_pin_fixup_tbl, alc662_fixups); + snd_hda_pick_pin_fixup(codec, alc662_pin_fixup_tbl, alc662_fixups, true); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); alc_auto_parse_customize_define(codec); -- 2.17.1

2 3

[alsa-devel] [PATCH] Fix a stack buffer overflow bug check_input_term
by Hui Peng 15 Aug '19

15 Aug '19

`check_input_term` recursively calls itself with input from device side (e.g., uac_input_terminal_descriptor.bCSourceID) as argument (id). In `check_input_term`, if `check_input_term` is called with the same `id` argument as the caller, it triggers endless recursive call, resulting kernel space stack overflow. This patch fixes the bug by adding a bitmap to `struct mixer_build` to keep track of the checked ids by `check_input_term` and stop the execution if some id has been checked (similar to how parse_audio_unit handles unitid argument). Reported-by: Hui Peng <benquike(a)gmail.com> Reported-by: Mathias Payer <mathias.payer(a)nebelwelt.net> Signed-off-by: Hui Peng <benquike(a)gmail.com> --- sound/usb/mixer.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ea487378be17..1f6c8213df82 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -68,6 +68,7 @@ struct mixer_build { unsigned char *buffer; unsigned int buflen; DECLARE_BITMAP(unitbitmap, MAX_ID_ELEMS); + DECLARE_BITMAP(termbitmap, MAX_ID_ELEMS); struct usb_audio_term oterm; const struct usbmix_name_map *map; const struct usbmix_selector_map *selector_map; @@ -782,6 +783,8 @@ static int check_input_term(struct mixer_build *state, int id, int err; void *p1; + if (test_and_set_bit(id, state->termbitmap)) + return 0; memset(term, 0, sizeof(*term)); while ((p1 = find_audio_control_unit(state, id)) != NULL) { unsigned char *hdr = p1; -- 2.22.1

2 7

[alsa-devel] [PATCH] Revert "ALSA: hda: Add codec on bus address table lately"
by Takashi Iwai 15 Aug '19

15 Aug '19

This reverts commit ee5f85d9290f ("ALSA: hda: Add codec on bus address table lately"). The commit caused several regression since I've overlooked that the function doesn't manage only the caddr_tbl but also the codec linked list that is referred indirectly in the other drivers. Revert for now to make everything back to work. Fixes: ee5f85d9290f ("ALSA: hda: Add codec on bus address table lately") Reported-by: Chris Wilson <chris(a)chris-wilson.co.uk> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_device.c | 21 +++++++++------------ 1 file changed, 9 insertions(+), 12 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index bf83d7062ef6..9f3e37511408 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -61,6 +61,10 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, pm_runtime_get_noresume(&codec->dev); atomic_set(&codec->in_pm, 0); + err = snd_hdac_bus_add_device(bus, codec); + if (err < 0) + goto error; + /* fill parameters */ codec->vendor_id = snd_hdac_read_parm(codec, AC_NODE_ROOT, AC_PAR_VENDOR_ID); @@ -139,22 +143,15 @@ int snd_hdac_device_register(struct hdac_device *codec) err = device_add(&codec->dev); if (err < 0) return err; - err = snd_hdac_bus_add_device(codec->bus, codec); - if (err < 0) - goto error; mutex_lock(&codec->widget_lock); err = hda_widget_sysfs_init(codec); mutex_unlock(&codec->widget_lock); - if (err < 0) - goto error_remove; + if (err < 0) { + device_del(&codec->dev); + return err; + } return 0; - - error_remove: - snd_hdac_bus_remove_device(codec->bus, codec); - error: - device_del(&codec->dev); - return err; } EXPORT_SYMBOL_GPL(snd_hdac_device_register); @@ -168,8 +165,8 @@ void snd_hdac_device_unregister(struct hdac_device *codec) mutex_lock(&codec->widget_lock); hda_widget_sysfs_exit(codec); mutex_unlock(&codec->widget_lock); - snd_hdac_bus_remove_device(codec->bus, codec); device_del(&codec->dev); + snd_hdac_bus_remove_device(codec->bus, codec); } } EXPORT_SYMBOL_GPL(snd_hdac_device_unregister); -- 2.16.4

1 0

[alsa-devel] [PATCH 0/4] ASoC: SOF: initial support for Intel ALH/SoundWire DAIs
by Pierre-Louis Bossart 15 Aug '19

15 Aug '19

This patchset adds the basic functionality to extract DAI information from the topology and send the information to firmware. At the moment there is no specific configuration in topology file, all the programming takes place based on the dai index and the related ALH stream ID. The renaming from SoundWire to ALH (Audio Link Hub) is a better representation of the hardware, the ALH acts as a proxy between the DMA transfers and the Cadence IP. Bard liao (1): ASoC: SOF: rename SOUNDWIRE to ALH Pierre-Louis Bossart (3): ASoC: SOF: ipc: add ALH parameters ASoC: SOF: topology: initial support for Intel ALH DAI type ASoC: SOF: pcm: add ALH support include/sound/sof/dai-intel.h | 9 ++++++++ include/sound/sof/dai.h | 3 ++- include/uapi/sound/sof/abi.h | 2 +- sound/soc/sof/pcm.c | 3 +++ sound/soc/sof/topology.c | 42 ++++++++++++++++++++++++++++++++++- 5 files changed, 56 insertions(+), 3 deletions(-) -- 2.20.1

2 8

Re: [alsa-devel] [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk
by Hans de Goede 15 Aug '19

15 Aug '19

Hi, On 15-08-19 19:12, Daniel Stuart wrote: > Every single baytrail chromebook sets PMC to 0, as can be seeing > below by searching through coreboot source code: > $ grep -rl "PMC_PLT_CLK\[0\]" . > ./rambi/variants/glimmer/devicetree.cb > ./rambi/variants/clapper/devicetree.cb > ./rambi/variants/swanky/devicetree.cb > ./rambi/variants/enguarde/devicetree.cb > ./rambi/variants/winky/devicetree.cb > ./rambi/variants/kip/devicetree.cb > ./rambi/variants/squawks/devicetree.cb > ./rambi/variants/orco/devicetree.cb > ./rambi/variants/ninja/devicetree.cb > ./rambi/variants/heli/devicetree.cb > ./rambi/variants/sumo/devicetree.cb > ./rambi/variants/banjo/devicetree.cb > ./rambi/variants/candy/devicetree.cb > ./rambi/variants/gnawty/devicetree.cb > ./rambi/variants/rambi/devicetree.cb > ./rambi/variants/quawks/devicetree.cb > > Plus, Cyan (only non-baytrail chromebook with max98090) also needs > this patch for audio to work. > > Thus, this commit adds all the missing devices to bsw_max98090 quirk > table, implemented by commit a182ecd3809c ("ASoC: intel: > cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0"). > > Signed-off-by: Daniel Stuart <daniel.stuart14(a)gmail.com> Thank you for catching this, this patch looks good to me: Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> Regards, Hans > --- > sound/soc/intel/boards/cht_bsw_max98090_ti.c | 98 ++++++++++++++++++++ > 1 file changed, 98 insertions(+) > > diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c > index 33eb72545be6..83b978e7b4c4 100644 > --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c > +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c > @@ -399,6 +399,20 @@ static struct snd_soc_card snd_soc_card_cht = { > }; > > static const struct dmi_system_id cht_max98090_quirk_table[] = { > + { > + /* Banjo model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Banjo"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Candy model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Candy"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > { > /* Clapper model Chromebook */ > .matches = { > @@ -406,6 +420,27 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { > }, > .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > }, > + { > + /* Cyan model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Cyan"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Enguarde model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Enguarde"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Glimmer model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Glimmer"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > { > /* Gnawty model Chromebook (Acer Chromebook CB3-111) */ > .matches = { > @@ -413,6 +448,62 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { > }, > .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > }, > + { > + /* Heli model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Heli"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Kip model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Kip"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Ninja model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Ninja"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Orco model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Orco"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Quawks model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Quawks"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Rambi model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Rambi"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Squawks model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Squawks"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Sumo model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Sumo"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > { > /* Swanky model Chromebook (Toshiba Chromebook 2) */ > .matches = { > @@ -420,6 +511,13 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { > }, > .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > }, > + { > + /* Winky model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Winky"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > {} > }; > >

1 0

[alsa-devel] [PATCH] ASoC: rockchip: rockchip_max98090: Set period size to 240
by Cheng-Yi Chiang 15 Aug '19

15 Aug '19

>From stress testing of arecord, we found that period size greater than ~900 will bring pl330 to DYING state and can not recover within 100 iterations. The result is that arecord will stuck and get I/O error, and issue can not be recovered until reboot. This issue does not happen when period size is small. Set constraint of period size to 240 to prevent such issue. With the constraint, there will be no issue after 2000 iterations. We can revert this patch once the root cause is found in rockchip's pl330 implementation. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/rockchip/rockchip_max98090.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c index 7b0c21fa6dca..0097df1fae66 100644 --- a/sound/soc/rockchip/rockchip_max98090.c +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -137,8 +137,19 @@ static int rk_aif1_hw_params(struct snd_pcm_substream *substream, return ret; } +static int rk_aif1_startup(struct snd_pcm_substream *substream) +{ + /* + * Set period size to 240 because pl330 has issue + * dealing with larger period in stress testing. + */ + return snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 240, 240); +} + static const struct snd_soc_ops rk_aif1_ops = { .hw_params = rk_aif1_hw_params, + .startup = rk_aif1_startup, }; SND_SOC_DAILINK_DEFS(hifi, -- 2.23.0.rc1.153.gdeed80330f-goog

2 1

[alsa-devel] [PATCH] ASoC: Intel: hsw: remove i386 build warning w/ size_t argument
by Pierre-Louis Bossart 15 Aug '19

15 Aug '19

Recent changes in the common IPC code introduced a build warning with size_t fields, use the correct %zu format. include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t' [-Wformat=] Fixes: abf31feea26c0 ('ASoC: Intel: Update request-reply IPC model') Reported-by: kbuild test robot <lkp(a)intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/haswell/sst-haswell-ipc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index 5c73b11375e3..0ff89ea96ccf 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -2015,11 +2015,11 @@ int sst_hsw_module_set_param(struct sst_hsw *hsw, if (payload_size <= SST_HSW_IPC_MAX_SHORT_PARAMETER_SIZE) { /* short parameter, mailbox can contain data */ - dev_dbg(dev, "transfer parameter size : %lu\n", + dev_dbg(dev, "transfer parameter size : %zu\n", request.size); request.size = ALIGN(payload_size, 4); - dev_dbg(dev, "transfer parameter aligned size : %lu\n", + dev_dbg(dev, "transfer parameter aligned size : %zu\n", request.size); parameter = kzalloc(request.size, GFP_KERNEL); -- 2.20.1

3 2

[alsa-devel] [PATCH] ASoC: SOF: fix HDA direct MMIO access
by Pierre-Louis Bossart 15 Aug '19

15 Aug '19

The recent change to remove the bus->io_ops callbacks used an older version of the SOF code base, and when merged into Mark's for-next it invalidated changes, resulting in broken compilation identified by kbuild and reproduced during the weekly SOF rebase. Restore SOF code overridden by git merge and apply Takashi's intended change in the 'right' location. Fixes: c2f16a94a8049 ("Merge branch 'topic/hda-bus-ops-cleanup'") Reported-by: kbuild test robot <lkp(a)intel.com> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-ctrl.c | 2 +- sound/soc/sof/intel/hda-dsp.c | 39 ---------------------------------- 2 files changed, 1 insertion(+), 40 deletions(-) diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index a7fee403cb90..bc41028a7a01 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -254,7 +254,7 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset) #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) /* Reset stream-to-link mapping */ list_for_each_entry(hlink, &bus->hlink_list, list) - bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); + writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); #endif bus->chip_init = true; diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 097727cda5cb..fb55a3c5afd0 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -354,45 +354,6 @@ static int hda_resume(struct snd_sof_dev *sdev, bool runtime_resume) return ret; } - hda_dsp_ctrl_misc_clock_gating(sdev, false); - - /* Reset stream-to-link mapping */ - list_for_each_entry(hlink, &bus->hlink_list, list) - writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); - - hda_dsp_ctrl_misc_clock_gating(sdev, true); -#else - - hda_dsp_ctrl_misc_clock_gating(sdev, false); - - /* reset controller */ - ret = hda_dsp_ctrl_link_reset(sdev, true); - if (ret < 0) { - dev_err(sdev->dev, - "error: failed to reset controller during resume\n"); - return ret; - } - - /* take controller out of reset */ - ret = hda_dsp_ctrl_link_reset(sdev, false); - if (ret < 0) { - dev_err(sdev->dev, - "error: failed to ready controller during resume\n"); - return ret; - } - - /* enable hda bus irq */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, - SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN, - SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN); - - hda_dsp_ctrl_misc_clock_gating(sdev, true); -#endif - - /* enable ppcap interrupt */ - hda_dsp_ctrl_ppcap_enable(sdev, true); - hda_dsp_ctrl_ppcap_int_enable(sdev, true); - #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) /* check jack status */ if (runtime_resume) -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: ti: Fix typos in ti/Kconfig
by Masanari Iida 15 Aug '19

15 Aug '19

This patch fixes some spelling typo in Kconfig. Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> --- sound/soc/ti/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 2197f3e1eaed..87a9b9dd4e98 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -12,7 +12,7 @@ config SND_SOC_TI_SDMA_PCM comment "Texas Instruments DAI support for:" config SND_SOC_DAVINCI_ASP - tristate "daVinci Audio Serial Port (ASP) or McBSP suport" + tristate "daVinci Audio Serial Port (ASP) or McBSP support" depends on ARCH_DAVINCI || COMPILE_TEST select SND_SOC_TI_EDMA_PCM help @@ -33,7 +33,7 @@ config SND_SOC_DAVINCI_MCASP - Keystone devices config SND_SOC_DAVINCI_VCIF - tristate "daVinci Voice Interface (VCIF) suport" + tristate "daVinci Voice Interface (VCIF) support" depends on ARCH_DAVINCI || COMPILE_TEST select SND_SOC_TI_EDMA_PCM help -- 2.23.0.rc2

3 2

[alsa-devel] [PATCH -next] ASoC: es8328: Fix copy-paste error in es8328_right_line_controls
by YueHaibing 15 Aug '19

15 Aug '19

It seems 'es8328_rline_enum' should be used in es8328_right_line_controls Fixes: 567e4f98922c ("ASoC: add es8328 codec driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/es8328.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 4a3d303..fdf64c2 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -227,7 +227,7 @@ static const struct soc_enum es8328_rline_enum = ARRAY_SIZE(es8328_line_texts), es8328_line_texts); static const struct snd_kcontrol_new es8328_right_line_controls = - SOC_DAPM_ENUM("Route", es8328_lline_enum); + SOC_DAPM_ENUM("Route", es8328_rline_enum); /* Left Mixer */ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { -- 2.7.4

2 2

[alsa-devel] [PATCH -next] ASoC: soc-core: Fix -Wunused-const-variable warning
by YueHaibing 15 Aug '19

15 Aug '19

If CONFIG_DMI is not set, gcc warns: sound/soc/soc-core.c:81:27: warning: dmi_blacklist defined but not used [-Wunused-const-variable=] Add #ifdef guard around it. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bb1e9e2..dcf39eb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -73,6 +73,7 @@ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); +#ifdef CONFIG_DMI /* * If a DMI filed contain strings in this blacklist (e.g. * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken @@ -87,6 +88,7 @@ static const char * const dmi_blacklist[] = { "Board Product Name", NULL, /* terminator */ }; +#endif static ssize_t pmdown_time_show(struct device *dev, struct device_attribute *attr, char *buf) -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: mt2701: remove unused variables
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/mediatek/mt2701/mt2701-afe-pcm.c:799:38: warning: mt2701_afe_o23_mix defined but not used [-Wunused-const-variable=] sound/soc/mediatek/mt2701/mt2701-afe-pcm.c:803:38: warning: mt2701_afe_o24_mix defined but not used [-Wunused-const-variable=] sound/soc/mediatek/mt2701/mt2701-afe-pcm.c:835:38: warning: mt2701_afe_multi_ch_out_i2s4 defined but not used [-Wunused-const-variable=] They are never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index 9af76ae..d7f5def 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -796,14 +796,6 @@ static const struct snd_kcontrol_new mt2701_afe_o22_mix[] = { SOC_DAPM_SINGLE_AUTODISABLE("I19 Switch", AFE_CONN22, 19, 1, 0), }; -static const struct snd_kcontrol_new mt2701_afe_o23_mix[] = { - SOC_DAPM_SINGLE_AUTODISABLE("I20 Switch", AFE_CONN23, 20, 1, 0), -}; - -static const struct snd_kcontrol_new mt2701_afe_o24_mix[] = { - SOC_DAPM_SINGLE_AUTODISABLE("I21 Switch", AFE_CONN24, 21, 1, 0), -}; - static const struct snd_kcontrol_new mt2701_afe_o31_mix[] = { SOC_DAPM_SINGLE_AUTODISABLE("I35 Switch", AFE_CONN41, 9, 1, 0), }; @@ -832,11 +824,6 @@ static const struct snd_kcontrol_new mt2701_afe_multi_ch_out_i2s3[] = { PWR2_TOP_CON, 18, 1, 0), }; -static const struct snd_kcontrol_new mt2701_afe_multi_ch_out_i2s4[] = { - SOC_DAPM_SINGLE_AUTODISABLE("Multich I2S4 Out Switch", - PWR2_TOP_CON, 19, 1, 0), -}; - static const struct snd_soc_dapm_widget mt2701_afe_pcm_widgets[] = { /* inter-connections */ SND_SOC_DAPM_MIXER("I00", SND_SOC_NOPM, 0, 0, NULL, 0), -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: cs4349: Use PM ops 'cs4349_runtime_pm'
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/codecs/cs4349.c:358:32: warning: cs4349_runtime_pm defined but not used [-Wunused-const-variable=] cs4349_runtime_pm ops already defined, it seems we should enable it. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: e40da86 ("ASoC: cs4349: Add support for Cirrus Logic CS4349") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/cs4349.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 09716fa..e0d44f6 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -378,6 +378,7 @@ static struct i2c_driver cs4349_i2c_driver = { .driver = { .name = "cs4349", .of_match_table = cs4349_of_match, + .pm = &cs4349_runtime_pm, }, .id_table = cs4349_i2c_id, .probe = cs4349_i2c_probe, -- 2.7.4

2 2

[alsa-devel] [PATCH -next] ASoC: wm8737: Fix copy-paste error in wm8737_snd_controls
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/codecs/wm8737.c:112:29: warning: high_3d defined but not used [-Wunused-const-variable=] 'high_3d' should be used for 3D High Cut-off. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 2a9ae13a2641 ("ASoC: Add initial WM8737 driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/wm8737.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 0c246fb..7a3f9fb 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -167,7 +167,7 @@ SOC_DOUBLE("Polarity Invert Switch", WM8737_ADC_CONTROL, 5, 6, 1, 0), SOC_SINGLE("3D Switch", WM8737_3D_ENHANCE, 0, 1, 0), SOC_SINGLE("3D Depth", WM8737_3D_ENHANCE, 1, 15, 0), SOC_ENUM("3D Low Cut-off", low_3d), -SOC_ENUM("3D High Cut-off", low_3d), +SOC_ENUM("3D High Cut-off", high_3d), SOC_SINGLE_TLV("3D ADC Volume", WM8737_3D_ENHANCE, 7, 1, 1, adc_tlv), SOC_SINGLE("Noise Gate Switch", WM8737_NOISE_GATE, 0, 1, 0), -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: mediatek: mt6358: add delay after dmic clock on
by Jiaxin Yu 15 Aug '19

15 Aug '19

Most dmics produce a high level when they receive clock. The difference between power-on and memory record time is about 10ms, but the dmic needs 50ms to output normal data. This commit add 100ms delay after SoC output clock so that we can cut off the pop noise at the beginning. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- Hi, This patch is based on for-next and discussed with tzungbi(a)google.com. --- --- sound/soc/codecs/mt6358.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c index c17250a..bb737fd 100644 --- a/sound/soc/codecs/mt6358.c +++ b/sound/soc/codecs/mt6358.c @@ -1730,6 +1730,10 @@ static int mt6358_dmic_enable(struct mt6358_priv *priv) /* UL turn on */ regmap_write(priv->regmap, MT6358_AFE_UL_SRC_CON0_L, 0x0003); + + /* Prevent pop noise form dmic hw */ + msleep(100); + return 0; } -- 1.8.1.1.dirty

2 1

[alsa-devel] [PATCH -next] ASoC: mediatek: mt8183-da7219-max98357: remove unused variable
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c:120:1: warning: mt8183_da7219_max98357_dapm_widgets defined but not used [-Wunused-const-variable=] sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c:124:40: warning: mt8183_da7219_max98357_dapm_routes defined but not used [-Wunused-const-variable=] They are never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 59076e2..b0db8a5 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -116,15 +116,6 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static const struct snd_soc_dapm_widget -mt8183_da7219_max98357_dapm_widgets[] = { - SND_SOC_DAPM_OUTPUT("IT6505_8CH"), -}; - -static const struct snd_soc_dapm_route mt8183_da7219_max98357_dapm_routes[] = { - {"IT6505_8CH", NULL, "TDM"}, -}; - /* FE */ SND_SOC_DAILINK_DEFS(playback1, DAILINK_COMP_ARRAY(COMP_CPU("DL1")), -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: remove unused variables
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c:50:1: warning: mt8183_mt6358_ts3a227_max98357_dapm_widgets defined but not used [-Wunused-const-variable=] sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c:55:1: warning: mt8183_mt6358_ts3a227_max98357_dapm_routes defined but not used [-Wunused-const-variable=] They are never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 4c816c8..dcd47b9 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -46,16 +46,6 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static const struct snd_soc_dapm_widget -mt8183_mt6358_ts3a227_max98357_dapm_widgets[] = { - SND_SOC_DAPM_OUTPUT("IT6505_8CH"), -}; - -static const struct snd_soc_dapm_route -mt8183_mt6358_ts3a227_max98357_dapm_routes[] = { - {"IT6505_8CH", NULL, "TDM"}, -}; - static int mt8183_mt6358_ts3a227_max98357_bt_sco_startup( struct snd_pcm_substream *substream) -- 2.7.4

2 1

[alsa-devel] [PATCH v1] ASoC: Intel: Skylake: Print constant literals from format specifier
by Andy Shevchenko 15 Aug '19

15 Aug '19

Instead of using two additional "%s" specifiers, put the constant string literals directly to the format specifier. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/skylake/skl-sst.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 13c636dece56..f3e1399450db 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -421,8 +421,7 @@ static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid) int ret = 0; char mod_name[64]; /* guid str = 32 chars + 4 hyphens */ - snprintf(mod_name, sizeof(mod_name), "%s%pUL%s", - "intel/dsp_fw_", guid, ".bin"); + snprintf(mod_name, sizeof(mod_name), "intel/dsp_fw_%pUL.bin", guid); module_entry = skl_module_get_from_id(ctx, mod_id); if (module_entry == NULL) { -- 2.20.1

4 4

[alsa-devel] [PATCH -next] ASoC: cs42l73: remove unused variables 'vsp_output_mux' and 'xsp_output_mux'
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/codecs/cs42l73.c:276:38: warning: vsp_output_mux defined but not used [-Wunused-const-variable=] sound/soc/codecs/cs42l73.c:279:38: warning: xsp_output_mux defined but not used [-Wunused-const-variable=] They are never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/cs42l73.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index a817393..36089f8 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -273,12 +273,6 @@ static SOC_ENUM_SINGLE_DECL(xsp_output_mux_enum, CS42L73_MIXERCTL, 4, cs42l73_spo_mixer_text); -static const struct snd_kcontrol_new vsp_output_mux = - SOC_DAPM_ENUM("Route", vsp_output_mux_enum); - -static const struct snd_kcontrol_new xsp_output_mux = - SOC_DAPM_ENUM("Route", xsp_output_mux_enum); - static const struct snd_kcontrol_new hp_amp_ctl = SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 0, 1, 1); -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: max98371: remove unused variable 'max98371_noload_gain_tlv'
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/codecs/max98371.c:157:35: warning: max98371_noload_gain_tlv defined but not used [-Wunused-const-variable=] It is never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/max98371.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c index ce80148..dfee05f 100644 --- a/sound/soc/codecs/max98371.c +++ b/sound/soc/codecs/max98371.c @@ -154,10 +154,6 @@ static const DECLARE_TLV_DB_RANGE(max98371_gain_tlv, 8, 10, TLV_DB_SCALE_ITEM(400, 100, 0) ); -static const DECLARE_TLV_DB_RANGE(max98371_noload_gain_tlv, - 0, 11, TLV_DB_SCALE_ITEM(950, 100, 0), -); - static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1); static const struct snd_kcontrol_new max98371_snd_controls[] = { -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: rt1011: remove unused variable 'dac_vol_tlv' and 'adc_vol_tlv'
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/codecs/rt1011.c:981:35: warning: dac_vol_tlv defined but not used [-Wunused-const-variable=] sound/soc/codecs/rt1011.c:982:35: warning: adc_vol_tlv defined but not used [-Wunused-const-variable=] They are never used, so can be removed. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/rt1011.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 638abca..fa34565 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -978,9 +978,6 @@ static bool rt1011_readable_register(struct device *dev, unsigned int reg) } } -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9435, 37, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1739, 37, 0); - static const char * const rt1011_din_source_select[] = { "Left", "Right", -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: tlv320aic23: remove unused variable 'tlv320aic23_rec_src'
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/codecs/tlv320aic23.c:70:29: warning: tlv320aic23_rec_src defined but not used [-Wunused-const-variable=] It is never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/tlv320aic23.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 080a840..f8e2f4b 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -67,8 +67,6 @@ static SOC_ENUM_SINGLE_DECL(rec_src_enum, static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls = SOC_DAPM_ENUM("Input Select", rec_src_enum); -static SOC_ENUM_SINGLE_DECL(tlv320aic23_rec_src, - TLV320AIC23_ANLG, 2, rec_src_text); static SOC_ENUM_SINGLE_DECL(tlv320aic23_deemph, TLV320AIC23_DIGT, 1, deemph_text); -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: tlv320aic31xx: remove unused variable 'cm_m_enum'
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/codecs/tlv320aic31xx.c:261:29: warning: cm_m_enum defined but not used [-Wunused-const-variable=] It is never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/tlv320aic31xx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 26a4f6c..df627a0 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -258,7 +258,6 @@ static SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, static SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, mic_select_text); -static SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text); static SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, mic_select_text); -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: es8328: remove unused variable 'pga_tlv'
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/codecs/es8328.c:102:35: warning: pga_tlv defined but not used [-Wunused-const-variable=] They are never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/es8328.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 822a25a..4a3d303 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -99,7 +99,6 @@ static SOC_ENUM_SINGLE_DECL(adcpol, static const DECLARE_TLV_DB_SCALE(play_tlv, -3000, 100, 0); static const DECLARE_TLV_DB_SCALE(dac_adc_tlv, -9600, 50, 0); -static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0); static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 300, 0); -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: cs42l56: remove unused variable 'adc_swap_enum'
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/codecs/cs42l56.c:206:30: warning: adc_swap_enum defined but not used [-Wunused-const-variable=] It is never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/cs42l56.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index b4d7627..ac569ab 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -199,14 +199,6 @@ static const struct soc_enum beep_bass_enum = SOC_ENUM_SINGLE(CS42L56_BEEP_TONE_CFG, 1, ARRAY_SIZE(beep_bass_text), beep_bass_text); -static const char * const adc_swap_text[] = { - "None", "A+B/2", "A-B/2", "Swap" -}; - -static const struct soc_enum adc_swap_enum = - SOC_ENUM_SINGLE(CS42L56_MISC_ADC_CTL, 3, - ARRAY_SIZE(adc_swap_text), adc_swap_text); - static const char * const pgaa_mux_text[] = { "AIN1A", "AIN2A", "AIN3A"}; -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: 88pm860x: remove unused variables 'pcm_switch_controls' and 'aif1_mux'
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/codecs/88pm860x-codec.c:533:38: warning: pcm_switch_controls defined but not used [-Wunused-const-variable=] sound/soc/codecs/88pm860x-codec.c:560:38: warning: aif1_mux defined but not used [-Wunused-const-variable=] They are never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/88pm860x-codec.c | 15 --------------- 1 file changed, 15 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index e982722..00b2c43 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -529,10 +529,6 @@ static const struct snd_kcontrol_new pm860x_snd_controls[] = { * DAPM Controls */ -/* PCM Switch / PCM Interface */ -static const struct snd_kcontrol_new pcm_switch_controls = - SOC_DAPM_SINGLE("Switch", PM860X_ADC_EN_2, 0, 1, 0); - /* AUX1 Switch */ static const struct snd_kcontrol_new aux1_switch_controls = SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 4, 1, 0); @@ -549,17 +545,6 @@ static const struct snd_kcontrol_new lepa_switch_controls = static const struct snd_kcontrol_new repa_switch_controls = SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 1, 1, 0); -/* PCM Mux / Mux7 */ -static const char *aif1_text[] = { - "PCM L", "PCM R", -}; - -static SOC_ENUM_SINGLE_DECL(aif1_enum, - PM860X_PCM_IFACE_3, 6, aif1_text); - -static const struct snd_kcontrol_new aif1_mux = - SOC_DAPM_ENUM("PCM Mux", aif1_enum); - /* I2S Mux / Mux9 */ static const char *i2s_din_text[] = { "DIN", "DIN1", -- 2.7.4

2 1

[alsa-devel] [PATCH] [v2] ASoC: Intel: boards: Add Cometlake machine driver support
by mac.chiang＠intel.com 15 Aug '19

15 Aug '19

From: Mac Chiang <mac.chiang(a)intel.com> reuse and add Cometlake support with: SSP0 for DA7219 headphone codec SSP1 for MAX98357a speaker amp codec Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- Changelog: v2: - included GENERIC config on APL/KBL platforms as well - reuse the same pcm_id and dai_id, modify in tplg support - replace with soc_intel_is_cml() instead --- --- sound/soc/intel/boards/Kconfig | 27 ++++++++++++++------ sound/soc/intel/boards/bxt_da7219_max98357a.c | 31 ++++++++++++++++++++++- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 12 +++++++++ sound/soc/intel/common/soc-intel-quirks.h | 5 ++++ 4 files changed, 66 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 50bf149..f3d88c87 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -256,16 +256,20 @@ config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH endif ## SND_SOC_INTEL_SKL +config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC + tristate + select SND_SOC_DA7219 + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + if SND_SOC_INTEL_APL config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH tristate "Broxton with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_DA7219 - select SND_SOC_MAX98357A - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI + select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC select SND_HDA_DSP_LOADER help This adds support for ASoC machine driver for Broxton-P platforms @@ -326,10 +330,7 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH tristate "KBL with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_DA7219 - select SND_SOC_MAX98357A - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI + select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC help This adds support for ASoC Onboard Codec I2S machine driver. This will create an alsa sound card for DA7219 + MAX98357A I2S audio codec. @@ -412,4 +413,14 @@ config SND_SOC_INTEL_SOF_RT5682_MACH If unsure select "N". endif ## SND_SOC_SOF_HDA_COMMON || SND_SOC_SOF_BAYTRAIL +if (SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK) + +config SND_SOC_INTEL_CML_LP_DA7219_MAX98357A_MACH + tristate "CML_LP with DA7219 and MAX98357A in I2S Mode" + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC + +endif ## SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK + endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index c0d865a..ac1dea5 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -179,10 +179,17 @@ static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) int ret; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_component *component = rtd->codec_dai->component; + int clk_freq; /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 19200000, + if (soc_intel_is_cml()) + clk_freq = 24000000; + else + clk_freq = 19200000; + + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, clk_freq, SND_SOC_CLOCK_IN); + if (ret) { dev_err(rtd->dev, "can't set codec sysclk configuration\n"); return ret; @@ -683,6 +690,25 @@ static int broxton_audio_probe(struct platform_device *pdev) broxton_dais[i].cpus->dai_name = "SSP2 Pin"; } } + } else if (soc_intel_is_cml()) { + unsigned int i; + + broxton_audio_card.name = "cmlda7219max"; + + for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) { + /* MAXIM_CODEC is connected to SSP1. */ + if (!strcmp(broxton_dais[i].codecs->dai_name, + BXT_MAXIM_CODEC_DAI)) { + broxton_dais[i].name = "SSP1-Codec"; + broxton_dais[i].cpus->dai_name = "SSP1 Pin"; + } + /* DIALOG_CODEC is connected to SSP0 */ + else if (!strcmp(broxton_dais[i].codecs->dai_name, + BXT_DIALOG_CODEC_DAI)) { + broxton_dais[i].name = "SSP0-Codec"; + broxton_dais[i].cpus->dai_name = "SSP0 Pin"; + } + } } /* override plaform name, if required */ @@ -700,6 +726,7 @@ static int broxton_audio_probe(struct platform_device *pdev) static const struct platform_device_id bxt_board_ids[] = { { .name = "bxt_da7219_max98357a" }, { .name = "glk_da7219_max98357a" }, + { .name = "cml_da7219_max98357a" }, { } }; @@ -720,6 +747,8 @@ MODULE_AUTHOR("Rohit Ainapure <rohit.m.ainapure(a)intel.com>"); MODULE_AUTHOR("Harsha Priya <harshapriya.n(a)intel.com>"); MODULE_AUTHOR("Conrad Cooke <conrad.cooke(a)intel.com>"); MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:bxt_da7219_max98357a"); MODULE_ALIAS("platform:glk_da7219_max98357a"); +MODULE_ALIAS("platform:cml_da7219_max98357a"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c index c36c0aa..4ea32b2 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -19,6 +19,11 @@ static struct snd_soc_acpi_codecs cml_codecs = { .codecs = {"10EC5682"} }; +static struct snd_soc_acpi_codecs cml_spk_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { { .id = "INT34C2", @@ -29,6 +34,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { .sof_tplg_filename = "sof-cnl-rt274.tplg", }, { + .id = "DLGS7219", + .drv_name = "cml_da7219_max98357a", + .quirk_data = &cml_spk_codecs, + .sof_fw_filename = "sof-cnl.ri", + .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", + }, + { .id = "MX98357A", .drv_name = "sof_rt5682", .quirk_data = &cml_codecs, diff --git a/sound/soc/intel/common/soc-intel-quirks.h b/sound/soc/intel/common/soc-intel-quirks.h index 4718fd3..e6357d3 100644 --- a/sound/soc/intel/common/soc-intel-quirks.h +++ b/sound/soc/intel/common/soc-intel-quirks.h @@ -36,6 +36,7 @@ SOC_INTEL_IS_CPU(byt, INTEL_FAM6_ATOM_SILVERMONT); SOC_INTEL_IS_CPU(cht, INTEL_FAM6_ATOM_AIRMONT); SOC_INTEL_IS_CPU(apl, INTEL_FAM6_ATOM_GOLDMONT); SOC_INTEL_IS_CPU(glk, INTEL_FAM6_ATOM_GOLDMONT_PLUS); +SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_MOBILE); static inline bool soc_intel_is_byt_cr(struct platform_device *pdev) { @@ -110,6 +111,10 @@ static inline bool soc_intel_is_glk(void) return false; } +static inline bool soc_intel_is_cml(void) +{ + return false; +} #endif #endif /* _SND_SOC_INTEL_QUIRKS_H */ -- 2.7.4

3 2

[alsa-devel] [PATCH 0/4] ASoC: Intel/SOF: add initial support for Tiger Lake and Elhkart Lake
by Pierre-Louis Bossart 15 Aug '19

15 Aug '19

Add initial/minimal support for PCI enumeration, hardware descriptions and ACPI match tables. Follow-up patches will be provided to enable multi-core support (requires new IPC calls). Machine drivers will be provided at a later point as well when there are fewer configurations. And last HDMI work is required to handle the 4 converters (which may be combined with the transition from hdac_hdmi to patch_hdmi already mentioned by Kai). Pan Xiuli (4): ASoC: Intel: common: add ACPI matching tables for Tiger Lake ASoC: SOF: Intel: initial support for Tiger Lake. ASoC: Intel: common: add ACPI matching tables for EHL ASoC: SOF: Intel: initial support for Elkhart Lake include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-ehl-match.c | 18 ++++++++ .../intel/common/soc-acpi-intel-tgl-match.c | 24 ++++++++++ sound/soc/sof/intel/Kconfig | 32 ++++++++++++++ sound/soc/sof/intel/cnl.c | 32 ++++++++++++++ sound/soc/sof/intel/hda.h | 2 + sound/soc/sof/sof-pci-dev.c | 44 +++++++++++++++++++ 8 files changed, 155 insertions(+) create mode 100644 sound/soc/intel/common/soc-acpi-intel-ehl-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-tgl-match.c -- 2.20.1

2 10

[alsa-devel] [PATCH v2 1/2] ALSA: hda - Add a new match function for only undef configurations
by Hui Wang 15 Aug '19

15 Aug '19

With the existing pintbl, we already have many entries in it. it is better to figure out a new match to reduce the size of the pintbl. For example, there are over 10 entries in the pintbl for: 0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE If we define a new tbl like below, and with the new adding match function, we can remove those over 10 entries: SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x19, 0x40000000}, {0x1a, 0x40000000},), Here we put 0x19 and 0x1a in the tbl just because these two pins are undefined on Dell laptops with the codec alc255, and these two pins will be overwritten by ALC255_FIXUP_DELL1_MIC_NO_PRESENCE. In summary: the new match will check vendor id and codec id first, then check the pin_cfg defined in the tbl, only all pin_cfgs in the tbl are undef and the corresponding pin_cfgs on the laptop are undef too, this match returns true. This new match function has lower priority than existing match functions, so the existing tbls still work as before after applying this patch. And for the same vendor and same codec, we could define at most one undef tbl for it, for example for the Dell laptops with alc255 codec, some of them need to apply DELL1_MIC, the rest need to apply DELL2_MIC, here we could only define undef tbl for DELL1_MIC or DELL2_MIC, we could not define undef tbls for both of them. My plan is to change the existing tbl to undef tbl for MIC_NO_PRESENCE fixups gradually. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 62 ++++++++++++++++++++++++++++----- 1 file changed, 54 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 92390d457567..9bc2bfa0128c 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -915,6 +915,53 @@ static bool pin_config_match(struct hda_codec *codec, return true; } +/* Match the undef tbl with the initial verb table on the machine. + * Two nitices here: + * - define 1 undef tbl at most for the same vendor and same codec + * - the undef tbl has lower priority than normal tbls for matching + */ +static bool pin_config_match_undef(struct hda_codec *codec, + const struct hda_pintbl *pins) +{ + bool match = false; + + for (; pins->nid; pins++) { + const struct hda_pincfg *pin; + int i; + + if ((pins->val & 0xf0000000) != 0x40000000) + return false; + + match = false; + snd_array_for_each(&codec->init_pins, i, pin) { + if (pin->nid != pins->nid) + continue; + if ((pin->cfg & 0xf0000000) != 0x40000000) + return false; + match = true; + break; + } + + if (match == false) + return false; + } + + return match; +} + +static void snd_hda_set_found_fixup(struct hda_codec *codec, + const struct snd_hda_pin_quirk *pq, + const struct hda_fixup *fixlist) +{ + codec->fixup_id = pq->value; +#ifdef CONFIG_SND_DEBUG_VERBOSE + codec->fixup_name = pq->name; + codec_dbg(codec, "%s: picked fixup %s (pin match)\n", + codec->core.chip_name, codec->fixup_name); +#endif + codec->fixup_list = fixlist; +} + /** * snd_hda_pick_pin_fixup - Pick up a fixup matching with the pin quirk list * @codec: the HDA codec @@ -925,7 +972,7 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, const struct snd_hda_pin_quirk *pin_quirk, const struct hda_fixup *fixlist) { - const struct snd_hda_pin_quirk *pq; + const struct snd_hda_pin_quirk *pq, *undef_bak_pq = NULL; if (codec->fixup_id != HDA_FIXUP_ID_NOT_SET) return; @@ -936,16 +983,15 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, if (codec->core.vendor_id != pq->codec) continue; if (pin_config_match(codec, pq->pins)) { - codec->fixup_id = pq->value; -#ifdef CONFIG_SND_DEBUG_VERBOSE - codec->fixup_name = pq->name; - codec_dbg(codec, "%s: picked fixup %s (pin match)\n", - codec->core.chip_name, codec->fixup_name); -#endif - codec->fixup_list = fixlist; + snd_hda_set_found_fixup(codec, pq, fixlist); return; } + if (pin_config_match_undef(codec, pq->pins)) + undef_bak_pq = pq; } + + if (undef_bak_pq != NULL) + snd_hda_set_found_fixup(codec, undef_bak_pq, fixlist); } EXPORT_SYMBOL_GPL(snd_hda_pick_pin_fixup); -- 2.17.1

2 2

[alsa-devel] [PATCH 1/2] ALSA: hda - Add a new match function for only undef configurations
by Hui Wang 15 Aug '19

15 Aug '19

With the existing pintbl, we already have many entries in it. it is better to figure out a new match to reduce the size of the pintbl. For example, there are over 10 entries in the pintbl for: 0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE If we define a new tbl like below, and with the new adding match function, we can remove those over 10 entries: SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x19, 0x40000000}, {0x1a, 0x40000000},), Here we put 0x19 and 0x1a in the tbl just because these two pins are undefined on Dell laptops with the codec alc255, and these two pins will be overwritten by ALC255_FIXUP_DELL1_MIC_NO_PRESENCE. In summary: the new match will check vendor id and codec id first, then check the pin_cfg defined in the tbl, only all pin_cfgs in the tbl are undef and the corresponding pin_cfgs on the laptop are undef too, this match returns true. This new match function has lower priority than existing match functions, so the existing tbls still work as before after applying this patch. My plan is to change the existing tbl to undef tbl for MIC_NO_PRESENCE fixups gradually. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 32 +++++++++++++++++++++++++++++++- 1 file changed, 31 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 92390d457567..cfada7401b86 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -915,6 +915,36 @@ static bool pin_config_match(struct hda_codec *codec, return true; } +/* match the pintbl which only contains specific pins with undef configuration */ +static bool pin_config_match_undef(struct hda_codec *codec, + const struct hda_pintbl *pins) +{ + bool match = false; + + for (; pins->nid; pins++) { + const struct hda_pincfg *pin; + int i; + + if ((pins->val & 0xf0000000) != 0x40000000) + return false; + + match = false; + snd_array_for_each(&codec->init_pins, i, pin) { + if (pin->nid != pins->nid) + continue; + if ((pin->cfg & 0xf0000000) != 0x40000000) + return false; + match = true; + break; + } + + if (match == false) + return false; + } + + return match; +} + /** * snd_hda_pick_pin_fixup - Pick up a fixup matching with the pin quirk list * @codec: the HDA codec @@ -935,7 +965,7 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, continue; if (codec->core.vendor_id != pq->codec) continue; - if (pin_config_match(codec, pq->pins)) { + if (pin_config_match(codec, pq->pins) || pin_config_match_undef(codec, pq->pins)) { codec->fixup_id = pq->value; #ifdef CONFIG_SND_DEBUG_VERBOSE codec->fixup_name = pq->name; -- 2.17.1

2 4

[alsa-devel] [PATCH 0/4] A few more HD-audio core fixes / cleanups
by Takashi Iwai 14 Aug '19

14 Aug '19

Hi, this is a series of yet another patches for HD-audio core stuff. The first two are workarounds for possible races in the driver registration and other two are merely cleanups. Takashi === Takashi Iwai (4): ALSA: hda - Drop unsol event handler for Intel HDMI codecs ALSA: hda: Add codec on bus address table lately ALSA: hda: Drop export of snd_hdac_bus_add/remove_device() ALSA: hda: Unexport a few more stuff include/sound/hdaudio.h | 9 --------- sound/hda/hdac_bus.c | 8 ++++---- sound/hda/hdac_device.c | 27 ++++++++++++++------------- sound/hda/hdac_regmap.c | 1 + sound/hda/local.h | 7 +++++++ sound/pci/hda/patch_hdmi.c | 9 ++++++++- 6 files changed, 34 insertions(+), 27 deletions(-) -- 2.16.4

1 4

[alsa-devel] [PATCH] Fix an OOB bug in parse_audio_mixer_unit
by Hui Peng 14 Aug '19

14 Aug '19

The `uac_mixer_unit_descriptor` shown as below is read from the device side. In `parse_audio_mixer_unit`, `baSourceID` field is accessed from index 0 to `bNrInPins` - 1, the current implementation assumes that descriptor is always valid (the length of descriptor is no shorter than 5 + `bNrInPins`). If a descriptor read from the device side is invalid, it may trigger out-of-bound memory access. ``` struct uac_mixer_unit_descriptor { __u8 bLength; __u8 bDescriptorType; __u8 bDescriptorSubtype; __u8 bUnitID; __u8 bNrInPins; __u8 baSourceID[]; } ``` This patch fixes the bug by add a sanity check on the length of the descriptor. Signed-off-by: Hui Peng <benquike(a)gmail.com> Reported-by: Hui Peng <benquike(a)gmail.com> Reported-by: Mathias Payer <mathias.payer(a)nebelwelt.net> --- sound/usb/mixer.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 7498b5191b68..38202ce67237 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2091,6 +2091,15 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, struct usb_audio_term iterm; int input_pins, num_ins, num_outs; int pin, ich, err; + int desc_len = (int) ((unsigned long) state->buffer + + state->buflen - (unsigned long) raw_desc); + + if (desc_len < sizeof(*desc) + desc->bNrInPins) { + usb_audio_err(state->chip, + "descriptor %d too short\n", + unitid); + return -EINVAL; + } err = uac_mixer_unit_get_channels(state, desc); if (err < 0) { -- 2.22.1

4 8

[alsa-devel] [PATCH] ALSA: sb: remove redundant assignment to variable result
by Colin King 14 Aug '19

14 Aug '19

From: Colin Ian King <colin.king(a)canonical.com> Variable result is initialized to a value that is never read and it is re-assigned later. The initialization is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/isa/sb/sb_common.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 162338f1b68a..ff031d670400 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -80,7 +80,7 @@ int snd_sbdsp_reset(struct snd_sb *chip) static int snd_sbdsp_version(struct snd_sb * chip) { - unsigned int result = -ENODEV; + unsigned int result; snd_sbdsp_command(chip, SB_DSP_GET_VERSION); result = (short) snd_sbdsp_get_byte(chip) << 8; -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: fsl_sai: Handle slave mode per TX/RX direction
by Daniel Baluta 14 Aug '19

14 Aug '19

From: Viorel Suman <viorel.suman(a)nxp.com> The SAI interface can be a clock supplier or consumer as a function of stream direction. e.g SAI can be master for Tx and slave for Rx. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 18 +++++++++--------- sound/soc/fsl/fsl_sai.h | 2 +- 2 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 4a346fcb5630..69cf3678c859 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -273,18 +273,18 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_CBS_CFS: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; val_cr4 |= FSL_SAI_CR4_FSD_MSTR; - sai->is_slave_mode = false; + sai->is_slave_mode[tx] = false; break; case SND_SOC_DAIFMT_CBM_CFM: - sai->is_slave_mode = true; + sai->is_slave_mode[tx] = true; break; case SND_SOC_DAIFMT_CBS_CFM: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; - sai->is_slave_mode = false; + sai->is_slave_mode[tx] = false; break; case SND_SOC_DAIFMT_CBM_CFS: val_cr4 |= FSL_SAI_CR4_FSD_MSTR; - sai->is_slave_mode = true; + sai->is_slave_mode[tx] = true; break; default: return -EINVAL; @@ -326,7 +326,7 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) int ret = 0; /* Don't apply to slave mode */ - if (sai->is_slave_mode) + if (sai->is_slave_mode[tx]) return 0; for (id = 0; id < FSL_SAI_MCLK_MAX; id++) { @@ -422,7 +422,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, if (sai->slot_width) slot_width = sai->slot_width; - if (!sai->is_slave_mode) { + if (!sai->is_slave_mode[tx]) { ret = fsl_sai_set_bclk(cpu_dai, tx, slots * slot_width * params_rate(params)); if (ret) @@ -458,7 +458,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, * error. */ - if (!sai->is_slave_mode) { + if (!sai->is_slave_mode[tx]) { if (!sai->synchronous[TX] && sai->synchronous[RX] && !tx) { regmap_update_bits(sai->regmap, FSL_SAI_TCR4(ofs), FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, @@ -497,7 +497,7 @@ static int fsl_sai_hw_free(struct snd_pcm_substream *substream, struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - if (!sai->is_slave_mode && + if (!sai->is_slave_mode[tx] && sai->mclk_streams & BIT(substream->stream)) { clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[tx]]); sai->mclk_streams &= ~BIT(substream->stream); @@ -581,7 +581,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, * This is a hardware bug, and will be fix in the * next sai version. */ - if (!sai->is_slave_mode) { + if (!sai->is_slave_mode[tx]) { /* Software Reset for both Tx and Rx */ regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), FSL_SAI_CSR_SR); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index b89b0ca26053..c2c43a7d9ba1 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -167,7 +167,7 @@ struct fsl_sai { struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; - bool is_slave_mode; + bool is_slave_mode[2]; bool is_lsb_first; bool is_dsp_mode; bool synchronous[2]; -- 2.17.1

3 2

[alsa-devel] [RFC PATCH] ASoC: fsl_sai: Enable data lines based on input channels
by Daniel Baluta 14 Aug '19

14 Aug '19

An audio data frame consists of a number of slots one for each channel. In the case of I2S there are 2 data slots / frame. The maximum number of SAI slots / frame is configurable at IP integration time. This affects the width of Mask Register. SAI supports up to 32 slots per frame. The number of datalines is also configurable (up to 8 datalines) and affects TCE/RCE and the number of data/FIFO registers. The number of needed data lines (pins) is computed as follows: * pins = channels / slots. This can be computed in hw_params function so lets move TRCE bits seting from startup to hw_params. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 34 +++++++++++++--------------------- sound/soc/fsl/fsl_sai.h | 2 +- 2 files changed, 14 insertions(+), 22 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 69cf3678c859..b70032c82fe2 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -414,6 +414,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, u32 val_cr4 = 0, val_cr5 = 0; u32 slots = (channels == 1) ? 2 : channels; u32 slot_width = word_width; + u32 pins; int ret; if (sai->slots) @@ -422,6 +423,8 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, if (sai->slot_width) slot_width = sai->slot_width; + pins = DIV_ROUND_UP(channels, slots); + if (!sai->is_slave_mode[tx]) { ret = fsl_sai_set_bclk(cpu_dai, tx, slots * slot_width * params_rate(params)); @@ -480,13 +483,17 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, } } + regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), + FSL_SAI_CR3_TRCE_MASK, + FSL_SAI_CR3_TRCE((1 << pins) - 1)); + regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | FSL_SAI_CR5_FBT_MASK, val_cr5); - regmap_write(sai->regmap, FSL_SAI_xMR(tx), ~0UL - ((1 << channels) - 1)); + regmap_write(sai->regmap, FSL_SAI_xMR(tx), ~0UL - ((1 << slots) - 1)); return 0; } @@ -496,6 +503,10 @@ static int fsl_sai_hw_free(struct snd_pcm_substream *substream, { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int ofs = sai->soc_data->reg_offset; + + regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), + FSL_SAI_CR3_TRCE_MASK, 0); if (!sai->is_slave_mode[tx] && sai->mclk_streams & BIT(substream->stream)) { @@ -603,32 +614,13 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { - struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - unsigned int ofs = sai->soc_data->reg_offset; - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int ret; - regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), - FSL_SAI_CR3_TRCE_MASK, - FSL_SAI_CR3_TRCE); - ret = snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &fsl_sai_rate_constraints); - return ret; } -static void fsl_sai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - unsigned int ofs = sai->soc_data->reg_offset; - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - - regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), - FSL_SAI_CR3_TRCE_MASK, 0); -} - static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { .set_sysclk = fsl_sai_set_dai_sysclk, .set_fmt = fsl_sai_set_dai_fmt, @@ -637,7 +629,6 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { .hw_free = fsl_sai_hw_free, .trigger = fsl_sai_trigger, .startup = fsl_sai_startup, - .shutdown = fsl_sai_shutdown, }; static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) @@ -881,6 +872,7 @@ static int fsl_sai_probe(struct platform_device *pdev) return -ENOMEM; sai->pdev = pdev; + sai->soc_data = of_device_get_match_data(&pdev->dev); sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index c2c43a7d9ba1..3c5383139dc2 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -109,7 +109,7 @@ #define FSL_SAI_CR2_DIV_MASK 0xff /* SAI Transmit and Receive Configuration 3 Register */ -#define FSL_SAI_CR3_TRCE BIT(16) +#define FSL_SAI_CR3_TRCE(x) ((x) << 16) #define FSL_SAI_CR3_TRCE_MASK GENMASK(23, 16) #define FSL_SAI_CR3_WDFL(x) (x) #define FSL_SAI_CR3_WDFL_MASK 0x1f -- 2.17.1

3 2

[alsa-devel] [PATCH] ALSA: hda: Retry codec read on first time failure
by vani.ganji＠intel.com 14 Aug '19

14 Aug '19

From: Vani Ganji <vani.ganji(a)intel.com> Sometimes HDMI audio fails while fast switching between HDMI and Speaker due to codec Get Power state verb read failure. Verb is codec driver to access codec HW. Since issue scenario involves fast switching between HDMI and speaker, there might be chance to fail read from HW and same happened in the error scenario. Based on Display/Graphics architect suggestion, added a retry mechanism for failure on first attempt. Signed-off-by: Vani Ganji <vani.ganji(a)intel.com> --- sound/hda/hdac_device.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index b26cc93e7e10..225bd1bfe156 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -984,10 +984,17 @@ static unsigned int codec_read(struct hdac_device *hdac, hda_nid_t nid, { unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm); unsigned int res; - - if (snd_hdac_exec_verb(hdac, cmd, flags, &res)) - return -1; - + unsigned int count; + + /* Retry verb once more if it fails first time */ + /* recommended to retry since it is HW related */ + for (count = 0; count <= 1; count++) { + if (snd_hdac_exec_verb(hdac, cmd, flags, &res)) + if (count == 1) + return -1; + } else + break; + } return res; } -- 2.17.1

1 0

[alsa-devel] [PATCH] Add support for Sabre HiFi on System76 and Clevo laptops
by Dennis Mungai 14 Aug '19

14 Aug '19

Hello there, The patch attached below is based on Jeremy Soller's prior work on: https://patchwork.kernel.org/patch/9552671/ , copied herein. Cleaned up to pass checkpatch.pl tests. I can confirm that this patch does indeed fix the audio DAC init on the Clevo based P775TM1-R chassis with no issues. Warm regards, Dennis Mungai

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Let all conexant codec enter D3 when rebooting
by Hui Wang 14 Aug '19

14 Aug '19

We have 3 new lenovo laptops which have conexant codec 0x14f11f86, these 3 laptops also have the noise issue when rebooting, after letting the codec enter D3 before rebooting or poweroff, the noise disappers. Instead of adding a new ID again in the reboot_notify(), let us make this function apply to all conexant codec. In theory make codec enter D3 before rebooting or poweroff is harmless, and I tested this change on a couple of other Lenovo laptops which have different conexant codecs, there is no side effect so far. Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f299f137eaea..93a303676aea 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -163,15 +163,6 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - switch (codec->core.vendor_id) { - case 0x14f12008: /* CX8200 */ - case 0x14f150f2: /* CX20722 */ - case 0x14f150f4: /* CX20724 */ - break; - default: - return; - } - /* Turn the problematic codec into D3 to avoid spurious noises from the internal speaker during (and after) reboot */ cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false); -- 2.17.1

2 3

[alsa-devel] [PATCH] ALSA: hda: Retry codec read on first time failure
by vani.ganji＠intel.com 14 Aug '19

14 Aug '19

From: Vani Ganji <vani.ganji(a)intel.com> Sometimes HDMI audio fails while fast switching between HDMI and Speaker due to codec Get Power state verb read failure. Verb is codec driver to access codec HW. Since issue scenario involves fast switching between HDMI and speaker, there might be chance to fail read from HW and same happened in the error scenario. Based on Display/Graphics architect suggestion, added a retry mechanism for failure on first attempt. Signed-off-by: Vani Ganji <vani.ganji(a)intel.com> --- sound/hda/hdac_device.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index b26cc93e7e10..225bd1bfe156 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -984,10 +984,17 @@ static unsigned int codec_read(struct hdac_device *hdac, hda_nid_t nid, { unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm); unsigned int res; - - if (snd_hdac_exec_verb(hdac, cmd, flags, &res)) - return -1; - + unsigned int count; + + /* Retry verb once more if it fails first time */ + /* recommended to retry since it is HW related */ + for (count = 0; count <= 1; count++) { + if (snd_hdac_exec_verb(hdac, cmd, flags, &res)) + if (count == 1) + return -1; + } else + break; + } return res; } -- 2.17.1

2 1

[alsa-devel] [PATCH] soundwire: Don't build sound.o without CONFIG_ACPI
by Nathan Chancellor 14 Aug '19

14 Aug '19

clang warns when CONFIG_ACPI is unset: ../drivers/soundwire/slave.c:16:12: warning: unused function 'sdw_slave_add' [-Wunused-function] static int sdw_slave_add(struct sdw_bus *bus, ^ 1 warning generated. Before commit 8676b3ca4673 ("soundwire: fix regmap dependencies and align with other serial links"), this code would only be compiled when ACPI was set because it was only selected by SOUNDWIRE_INTEL, which depends on ACPI. Now, this code can be compiled without CONFIG_ACPI, which causes the above warning. The IS_ENABLED(CONFIG_ACPI) guard could be moved to avoid compiling the function; however, slave.c only contains three functions, two of which are static. Only compile slave.o when CONFIG_ACPI is set, where it will actually be used. bus.h contains a stub for sdw_acpi_find_slaves so there will be no issues with an undefined function. This has been build tested with CONFIG_ACPI set and unset in combination with CONFIG_SOUNDWIRE unset, built in, and a module. Fixes: 8676b3ca4673 ("soundwire: fix regmap dependencies and align with other serial links") Link: https://github.com/ClangBuiltLinux/linux/issues/637 Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> --- drivers/soundwire/Makefile | 6 +++++- drivers/soundwire/slave.c | 3 --- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/Makefile b/drivers/soundwire/Makefile index 45b7e5001653..226090902716 100644 --- a/drivers/soundwire/Makefile +++ b/drivers/soundwire/Makefile @@ -4,9 +4,13 @@ # #Bus Objs -soundwire-bus-objs := bus_type.o bus.o slave.o mipi_disco.o stream.o +soundwire-bus-objs := bus_type.o bus.o mipi_disco.o stream.o obj-$(CONFIG_SOUNDWIRE) += soundwire-bus.o +ifdef CONFIG_ACPI +soundwire-bus-objs += slave.o +endif + #Cadence Objs soundwire-cadence-objs := cadence_master.o obj-$(CONFIG_SOUNDWIRE_CADENCE) += soundwire-cadence.o diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index f39a5815e25d..0dc188e6873b 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -60,7 +60,6 @@ static int sdw_slave_add(struct sdw_bus *bus, return ret; } -#if IS_ENABLED(CONFIG_ACPI) /* * sdw_acpi_find_slaves() - Find Slave devices in Master ACPI node * @bus: SDW bus instance @@ -110,5 +109,3 @@ int sdw_acpi_find_slaves(struct sdw_bus *bus) return 0; } - -#endif -- 2.23.0.rc2

4 8

[alsa-devel] [PATCH AUTOSEL 5.2 014/123] ASoC: samsung: odroid: fix a double-free issue for cpu_dai
by Sasha Levin 14 Aug '19

14 Aug '19

From: Wen Yang <wen.yang99(a)zte.com.cn> [ Upstream commit 2abee12c0ab1924a69993d2c063a39a952e7d836 ] The cpu_dai variable is still being used after the of_node_put() call, which may result in double-free: of_node_put(cpu_dai); ---> released here ret = devm_snd_soc_register_card(dev, card); if (ret < 0) { ... goto err_put_clk_i2s; --> jump to err_put_clk_i2s ... err_put_clk_i2s: clk_put(priv->clk_i2s_bus); err_put_sclk: clk_put(priv->sclk_i2s); err_put_cpu_dai: of_node_put(cpu_dai); --> double-free here Fixes: d832d2b246c5 ("ASoC: samsung: odroid: Fix of_node refcount unbalance") Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Krzysztof Kozlowski <krzk(a)kernel.org> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Sylwester Nawrocki <s.nawrocki(a)samsung.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: linux-kernel(a)vger.kernel.org Link: https://lore.kernel.org/r/1562989575-33785-3-git-send-email-wen.yang99@zte.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/samsung/odroid.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 95c35e3ff3303..d606e48fe551a 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -299,7 +299,6 @@ static int odroid_audio_probe(struct platform_device *pdev) ret = PTR_ERR(priv->clk_i2s_bus); goto err_put_sclk; } - of_node_put(cpu_dai); ret = devm_snd_soc_register_card(dev, card); if (ret < 0) { @@ -307,6 +306,7 @@ static int odroid_audio_probe(struct platform_device *pdev) goto err_put_clk_i2s; } + of_node_put(cpu_dai); of_node_put(codec); return 0; -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.2 013/123] ASoC: samsung: odroid: fix an use-after-free issue for codec
by Sasha Levin 14 Aug '19

14 Aug '19

From: Wen Yang <wen.yang99(a)zte.com.cn> [ Upstream commit 9b6d104a6b150bd4d3e5b039340e1f6b20c2e3c1 ] The codec variable is still being used after the of_node_put() call, which may result in use-after-free. Fixes: bc3cf17b575a ("ASoC: samsung: odroid: Add support for secondary CPU DAI") Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Krzysztof Kozlowski <krzk(a)kernel.org> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Sylwester Nawrocki <s.nawrocki(a)samsung.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: linux-kernel(a)vger.kernel.org Link: https://lore.kernel.org/r/1562989575-33785-2-git-send-email-wen.yang99@zte.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/samsung/odroid.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index e688169ff12ab..95c35e3ff3303 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -275,9 +275,8 @@ static int odroid_audio_probe(struct platform_device *pdev) } of_node_put(cpu); - of_node_put(codec); if (ret < 0) - return ret; + goto err_put_node; ret = snd_soc_of_get_dai_link_codecs(dev, codec, codec_link); if (ret < 0) @@ -308,6 +307,7 @@ static int odroid_audio_probe(struct platform_device *pdev) goto err_put_clk_i2s; } + of_node_put(codec); return 0; err_put_clk_i2s: @@ -317,6 +317,8 @@ static int odroid_audio_probe(struct platform_device *pdev) err_put_cpu_dai: of_node_put(cpu_dai); snd_soc_of_put_dai_link_codecs(codec_link); +err_put_node: + of_node_put(codec); return ret; } -- 2.20.1

1 0

[alsa-devel] Sound Blaster RX does not play sound in side channels
by GitHub issues - opened 13 Aug '19

13 Aug '19

alsa-project/alsa-lib issue #9 was opened from logan001: I don't have the card anymore but maybe someone else have it and can provide some help cause i like the card bass/treble control and if the bug is fixed i will buy it. The card does not output side channels at all. 5.1 works but 7.1 side channels gives no sound with speaker-test -c 8 pulseaudio also gives no sound for side channels in speaker tests. i don't know if its the right place to report this bug but i've tried on kernel bugzilla and there is no movement there. the kernels i've tried 1 month ago are 4.18 and 5.1.15 here is the [alsainfo](https://bugzilla.kernel.org/attachment.cgi?id=283467) at that time what i found strange is that alsamixer when i change the sound card had the side channels volume set at 0. its like by default was build intentionally without side channels support. and yes i have that special cable with 3 jacks (sound card) to 4 jack (for speakers). i have it from an old audigy 2 zs if i remeber corectly. and the cable works since i tested 7.1 in windows without any issues. Issue URL : https://github.com/alsa-project/alsa-lib/issues/9 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH -next] ALSA: hda/sigmatel - remove unused variable 'stac9200_core_init'
by YueHaibing 13 Aug '19

13 Aug '19

sound/pci/hda/patch_sigmatel.c:981:30: warning: stac9200_core_init defined but not used [-Wunused-const-variable=] It is never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/pci/hda/patch_sigmatel.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0d9b627..894f3f5 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -975,15 +975,6 @@ static int stac_create_spdif_mux_ctls(struct hda_codec *codec) return 0; } -/* - */ - -static const struct hda_verb stac9200_core_init[] = { - /* set dac0mux for dac converter */ - { 0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - {} -}; - static const struct hda_verb stac9200_eapd_init[] = { /* set dac0mux for dac converter */ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Add quirk for HP Envy x360
by Takashi Iwai 13 Aug '19

13 Aug '19

HP Envy x360 (AMD Ryzen-based model) with 103c:8497 needs the same quirk like HP Spectre x360 for enabling the mute LED over Mic3 pin. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204373 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index de224cbea7a0..8aaf1d9c55cf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6987,6 +6987,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), 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), -- 2.16.4

1 0

[alsa-devel] Regression in next with codec unload and snd_soc_component_get/put
by Tony Lindgren 13 Aug '19

13 Aug '19

Hi, Looks like commit 4a81e8f30d0b ("ASoC: soc-component: add snd_soc_component_get/put()") causes a regression where trying to rmmod a codec driver fails with module is in use error after rmmod of snd-soc-audio-graph-card for example. Any ideas what goes wrong there? BTW, looks like the lore.kernel.org link in that commit also is unreachable? Regards, Tony

4 6

[alsa-devel] [PATCH 00/11] JZ4740 SoC cleanup
by Paul Cercueil 13 Aug '19

13 Aug '19

Hi, This patchset converts the Qi LB60 MIPS board to devicetree and makes it use all the shiny new drivers that have been developed or updated recently. All the crappy old drivers and custom code can be dropped since they have been replaced by better alternatives. Some of these alternatives are not yet in 5.3-rc1 but have already been accepted by their respective maintainer for inclusion in 5.4-rc1. To upstream this patchset, I think that as soon as MIPS maintainers agree to take patches 01-03/11 and 11/11, the other patches can go through their respective maintainer's tree. Note for MIPS maintainers: Patch 11/11 may conflict with the TCU patchset v15, should this one be accepted upstream, but the conflict is tiny and easy to fix. Should this case appear, don't hesitate to bother me about it. Thanks, -Paul

12 38

[alsa-devel] [asoc:for-5.4 195/232] sound/soc/sof/intel/hda-dsp.c:360:22: error: 'hlink' undeclared
by kbuild test robot 12 Aug '19

12 Aug '19

Hi Mark, FYI, the error/warning still remains. tree: https://kernel.googlesource.com/pub/scm/linux/kernel/git/broonie/sound.git for-5.4 head: 6fa4e0cae684d268d309d1b1f929d52e3df5649c commit: c2f16a94a80497e4b28c27f9ca2cd6cd60706fb6 [195/232] Merge branch 'topic/hda-bus-ops-cleanup' of https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into asoc-5.4 config: x86_64-randconfig-s0-08121238 (attached as .config) compiler: gcc-7 (Debian 7.4.0-10) 7.4.0 reproduce: git checkout c2f16a94a80497e4b28c27f9ca2cd6cd60706fb6 # save the attached .config to linux build tree make ARCH=x86_64 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): In file included from include/linux/kobject.h:19:0, from include/linux/device.h:16, from include/sound/hdaudio.h:9, from include/sound/hdaudio_ext.h:5, from sound/soc/sof/intel/hda-dsp.c:18: sound/soc/sof/intel/hda-dsp.c: In function 'hda_resume': >> sound/soc/sof/intel/hda-dsp.c:360:22: error: 'hlink' undeclared (first use in this function) list_for_each_entry(hlink, &bus->hlink_list, list) ^ include/linux/list.h:577:7: note: in definition of macro 'list_for_each_entry' for (pos = list_first_entry(head, typeof(*pos), member); \ ^~~ sound/soc/sof/intel/hda-dsp.c:360:22: note: each undeclared identifier is reported only once for each function it appears in list_for_each_entry(hlink, &bus->hlink_list, list) ^ include/linux/list.h:577:7: note: in definition of macro 'list_for_each_entry' for (pos = list_first_entry(head, typeof(*pos), member); \ ^~~ In file included from include/linux/list.h:9:0, from include/linux/kobject.h:19, from include/linux/device.h:16, from include/sound/hdaudio.h:9, from include/sound/hdaudio_ext.h:5, from sound/soc/sof/intel/hda-dsp.c:18: >> sound/soc/sof/intel/hda-dsp.c:360:30: error: 'bus' undeclared (first use in this function) list_for_each_entry(hlink, &bus->hlink_list, list) ^ include/linux/kernel.h:972:26: note: in definition of macro 'container_of' void *__mptr = (void *)(ptr); \ ^~~ include/linux/list.h:490:2: note: in expansion of macro 'list_entry' list_entry((ptr)->next, type, member) ^~~~~~~~~~ include/linux/list.h:577:13: note: in expansion of macro 'list_first_entry' for (pos = list_first_entry(head, typeof(*pos), member); \ ^~~~~~~~~~~~~~~~ sound/soc/sof/intel/hda-dsp.c:360:2: note: in expansion of macro 'list_for_each_entry' list_for_each_entry(hlink, &bus->hlink_list, list) ^~~~~~~~~~~~~~~~~~~ >> sound/soc/sof/intel/hda-dsp.c:364:2: error: #else without #if #else ^~~~ >> sound/soc/sof/intel/hda-dsp.c:390:2: error: #endif without #if #endif ^~~~~ vim +/hlink +360 sound/soc/sof/intel/hda-dsp.c 747503b1813a3e Liam Girdwood 2019-04-12 334 fd15f2f5e27214 Rander Wang 2019-07-22 335 static int hda_resume(struct snd_sof_dev *sdev, bool runtime_resume) 747503b1813a3e Liam Girdwood 2019-04-12 336 { 747503b1813a3e Liam Girdwood 2019-04-12 337 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) 747503b1813a3e Liam Girdwood 2019-04-12 338 struct hdac_bus *bus = sof_to_bus(sdev); 747503b1813a3e Liam Girdwood 2019-04-12 339 struct hdac_ext_link *hlink = NULL; 747503b1813a3e Liam Girdwood 2019-04-12 340 #endif 747503b1813a3e Liam Girdwood 2019-04-12 341 int ret; 747503b1813a3e Liam Girdwood 2019-04-12 342 747503b1813a3e Liam Girdwood 2019-04-12 343 /* 747503b1813a3e Liam Girdwood 2019-04-12 344 * clear TCSEL to clear playback on some HD Audio 747503b1813a3e Liam Girdwood 2019-04-12 345 * codecs. PCI TCSEL is defined in the Intel manuals. 747503b1813a3e Liam Girdwood 2019-04-12 346 */ 747503b1813a3e Liam Girdwood 2019-04-12 347 snd_sof_pci_update_bits(sdev, PCI_TCSEL, 0x07, 0); 747503b1813a3e Liam Girdwood 2019-04-12 348 747503b1813a3e Liam Girdwood 2019-04-12 349 /* reset and start hda controller */ 747503b1813a3e Liam Girdwood 2019-04-12 350 ret = hda_dsp_ctrl_init_chip(sdev, true); 747503b1813a3e Liam Girdwood 2019-04-12 351 if (ret < 0) { 747503b1813a3e Liam Girdwood 2019-04-12 352 dev_err(sdev->dev, 747503b1813a3e Liam Girdwood 2019-04-12 353 "error: failed to start controller after resume\n"); 747503b1813a3e Liam Girdwood 2019-04-12 354 return ret; 747503b1813a3e Liam Girdwood 2019-04-12 355 } 747503b1813a3e Liam Girdwood 2019-04-12 356 747503b1813a3e Liam Girdwood 2019-04-12 357 hda_dsp_ctrl_misc_clock_gating(sdev, false); 747503b1813a3e Liam Girdwood 2019-04-12 358 747503b1813a3e Liam Girdwood 2019-04-12 359 /* Reset stream-to-link mapping */ 747503b1813a3e Liam Girdwood 2019-04-12 @360 list_for_each_entry(hlink, &bus->hlink_list, list) 19abfefd4c7604 Takashi Iwai 2019-08-07 361 writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); 747503b1813a3e Liam Girdwood 2019-04-12 362 747503b1813a3e Liam Girdwood 2019-04-12 363 hda_dsp_ctrl_misc_clock_gating(sdev, true); 747503b1813a3e Liam Girdwood 2019-04-12 @364 #else 747503b1813a3e Liam Girdwood 2019-04-12 365 747503b1813a3e Liam Girdwood 2019-04-12 366 hda_dsp_ctrl_misc_clock_gating(sdev, false); 747503b1813a3e Liam Girdwood 2019-04-12 367 747503b1813a3e Liam Girdwood 2019-04-12 368 /* reset controller */ 747503b1813a3e Liam Girdwood 2019-04-12 369 ret = hda_dsp_ctrl_link_reset(sdev, true); 747503b1813a3e Liam Girdwood 2019-04-12 370 if (ret < 0) { 747503b1813a3e Liam Girdwood 2019-04-12 371 dev_err(sdev->dev, 747503b1813a3e Liam Girdwood 2019-04-12 372 "error: failed to reset controller during resume\n"); 747503b1813a3e Liam Girdwood 2019-04-12 373 return ret; 747503b1813a3e Liam Girdwood 2019-04-12 374 } 747503b1813a3e Liam Girdwood 2019-04-12 375 747503b1813a3e Liam Girdwood 2019-04-12 376 /* take controller out of reset */ 747503b1813a3e Liam Girdwood 2019-04-12 377 ret = hda_dsp_ctrl_link_reset(sdev, false); 747503b1813a3e Liam Girdwood 2019-04-12 378 if (ret < 0) { 747503b1813a3e Liam Girdwood 2019-04-12 379 dev_err(sdev->dev, 747503b1813a3e Liam Girdwood 2019-04-12 380 "error: failed to ready controller during resume\n"); 747503b1813a3e Liam Girdwood 2019-04-12 381 return ret; 747503b1813a3e Liam Girdwood 2019-04-12 382 } 747503b1813a3e Liam Girdwood 2019-04-12 383 747503b1813a3e Liam Girdwood 2019-04-12 384 /* enable hda bus irq */ 747503b1813a3e Liam Girdwood 2019-04-12 385 snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, 747503b1813a3e Liam Girdwood 2019-04-12 386 SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN, 747503b1813a3e Liam Girdwood 2019-04-12 387 SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN); 747503b1813a3e Liam Girdwood 2019-04-12 388 747503b1813a3e Liam Girdwood 2019-04-12 389 hda_dsp_ctrl_misc_clock_gating(sdev, true); 24b6ff686fce80 Zhu Yingjiang 2019-06-12 @390 #endif 747503b1813a3e Liam Girdwood 2019-04-12 391 747503b1813a3e Liam Girdwood 2019-04-12 392 /* enable ppcap interrupt */ 747503b1813a3e Liam Girdwood 2019-04-12 393 hda_dsp_ctrl_ppcap_enable(sdev, true); 747503b1813a3e Liam Girdwood 2019-04-12 394 hda_dsp_ctrl_ppcap_int_enable(sdev, true); 747503b1813a3e Liam Girdwood 2019-04-12 395 :::::: The code at line 360 was first introduced by commit :::::: 747503b1813a3e6e4c52d9c0b4bd462b64940940 ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations :::::: TO: Liam Girdwood <liam.r.girdwood(a)linux.intel.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] ASoC: SOF: Intel: hda: fixup HDaudio topology name with DMIC number
by Pierre-Louis Bossart 12 Aug '19

12 Aug '19

The SOF project maintains 6 topologies for HDaudio (iDisp or HDaudio+iDisp, no DMIC, 2 DMICs, 4 DMICs). The user is currently required to manually rename the topology file used in /lib/firmware/intel/sof-tplg. We can do better to avoid such renames and use logic to select the relevant file. The NHLT information can be used to figure out which topology file should be used. Alternatively, when NHLT is not present in ACPI tables or is possibly incorrect, a module parameter can provide that information, e.g. on Up^2 board with the test DMIC kit. Tested on Up^2 board and Acer Swift-SF314-55 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/Kconfig | 1 + sound/soc/sof/intel/hda.c | 75 +++++++++++++++++++++++++++++++------ 2 files changed, 64 insertions(+), 12 deletions(-) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index dd14ce92fe10..545071afbe18 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -254,6 +254,7 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC + select SND_INTEL_NHLT help This option is not user-selectable but automagically handled by 'select' statements at a higher level diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index ebf277754901..c72e9a09eee1 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -19,6 +19,7 @@ #include <sound/hda_register.h> #include <linux/module.h> +#include <sound/intel-nhlt.h> #include <sound/sof.h> #include <sound/sof/xtensa.h> #include "../ops.h" @@ -49,6 +50,12 @@ module_param_named(use_msi, hda_use_msi, bool, 0444); MODULE_PARM_DESC(use_msi, "SOF HDA use PCI MSI mode"); #endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) +static int hda_dmic_num = -1; +module_param_named(dmic_num, hda_dmic_num, int, 0444); +MODULE_PARM_DESC(dmic_num, "SOF HDA DMIC number"); +#endif + static const struct hda_dsp_msg_code hda_dsp_rom_msg[] = { {HDA_DSP_ROM_FW_MANIFEST_LOADED, "status: manifest loaded"}, {HDA_DSP_ROM_FW_FW_LOADED, "status: fw loaded"}, @@ -283,8 +290,26 @@ static int hda_init(struct snd_sof_dev *sdev) #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) +static int check_nhlt_dmic(struct snd_sof_dev *sdev) +{ + struct nhlt_acpi_table *nhlt; + int dmic_num; + + nhlt = intel_nhlt_init(sdev->dev); + if (nhlt) { + dmic_num = intel_nhlt_get_dmic_geo(sdev->dev, nhlt); + intel_nhlt_free(nhlt); + if (dmic_num == 2 || dmic_num == 4) + return dmic_num; + } + + return 0; +} + static const char *fixup_tplg_name(struct snd_sof_dev *sdev, - const char *sof_tplg_filename) + const char *sof_tplg_filename, + const char *idisp_str, + const char *dmic_str) { const char *tplg_filename = NULL; char *filename; @@ -298,7 +323,8 @@ static const char *fixup_tplg_name(struct snd_sof_dev *sdev, split_ext = strsep(&filename, "."); if (split_ext) { tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL, - "%s-idisp.tplg", split_ext); + "%s%s%s.tplg", + split_ext, idisp_str, dmic_str); if (!tplg_filename) return NULL; } @@ -317,6 +343,9 @@ static int hda_init_caps(struct snd_sof_dev *sdev) struct snd_sof_pdata *pdata = sdev->pdata; struct snd_soc_acpi_mach *mach; const char *tplg_filename; + const char *idisp_str; + const char *dmic_str; + int dmic_num; int codec_num = 0; int i; #endif @@ -387,17 +416,39 @@ static int hda_init_caps(struct snd_sof_dev *sdev) dev_info(bus->dev, "using HDA machine driver %s now\n", hda_mach->drv_name); - /* fixup topology file for HDMI only platforms */ - if (codec_num == 1) { - /* use local variable for readability */ - tplg_filename = pdata->tplg_filename; - tplg_filename = fixup_tplg_name(sdev, tplg_filename); - if (!tplg_filename) { - hda_codec_i915_exit(sdev); - return ret; - } - pdata->tplg_filename = tplg_filename; + if (codec_num == 1) + idisp_str = "-idisp"; + else + idisp_str = ""; + + /* first check NHLT for DMICs */ + dmic_num = check_nhlt_dmic(sdev); + + /* allow for module parameter override */ + if (hda_dmic_num != -1) + dmic_num = hda_dmic_num; + + switch (dmic_num) { + case 2: + dmic_str = "-2ch"; + break; + case 4: + dmic_str = "-4ch"; + break; + default: + dmic_num = 0; + dmic_str = ""; + break; + } + + tplg_filename = pdata->tplg_filename; + tplg_filename = fixup_tplg_name(sdev, tplg_filename, + idisp_str, dmic_str); + if (!tplg_filename) { + hda_codec_i915_exit(sdev); + return ret; } + pdata->tplg_filename = tplg_filename; } } -- 2.20.1

2 1

[alsa-devel] asound snd_pcm_start question
by Eric Desjardins 12 Aug '19

12 Aug '19

Hi, This is a simple question. I call: snd_pcm_prepare snd_pcm_writei and it works fine but I would like to delay playback start. There is snd_pcm_start for that. There is a method that is deprecated to specify a start mode but I was wondering what was the correct new way to do that? Thanks, Eric

2 1

[alsa-devel] [PATCH] ALSA: hda: Set fifo_size for both playback and capture streams
by Takashi Iwai 12 Aug '19

12 Aug '19

Currently we set hdac_stream.fifo_size field only for the playback stream by some odd reason I forgot, while this field isn't referred in any places. Actually this fifo_size field would have been required in the position report correction for VIA chipset, but due to the lack of the fifo_size set for capture streams, snd-hda-intel driver fetches the register by itself. This patch straightens and simplifies the code by setting the fifo_size field for both playback and capture streams, and use it in the HD-audio controller driver. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_stream.c | 6 +----- sound/pci/hda/hda_intel.c | 8 +------- 2 files changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index fc68d4ce0a37..d8fe7ff0cd58 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -229,11 +229,7 @@ int snd_hdac_stream_setup(struct hdac_stream *azx_dev) /* set the interrupt enable bits in the descriptor control register */ snd_hdac_stream_updatel(azx_dev, SD_CTL, 0, SD_INT_MASK); - if (azx_dev->direction == SNDRV_PCM_STREAM_PLAYBACK) - azx_dev->fifo_size = - snd_hdac_stream_readw(azx_dev, SD_FIFOSIZE) + 1; - else - azx_dev->fifo_size = 0; + azx_dev->fifo_size = snd_hdac_stream_readw(azx_dev, SD_FIFOSIZE) + 1; /* when LPIB delay correction gives a small negative value, * we ignore it; currently set the threshold statically to diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3a209e07d5d8..dc8a83cb7393 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -84,8 +84,6 @@ enum { #define INTEL_SCH_HDA_DEVC 0x78 #define INTEL_SCH_HDA_DEVC_NOSNOOP (0x1<<11) -/* Define IN stream 0 FIFO size offset in VIA controller */ -#define VIA_IN_STREAM0_FIFO_SIZE_OFFSET 0x90 /* Define VIA HD Audio Device ID*/ #define VIA_HDAC_DEVICE_ID 0x3288 @@ -811,11 +809,7 @@ static unsigned int azx_via_get_position(struct azx *chip, mod_dma_pos = le32_to_cpu(*azx_dev->core.posbuf); mod_dma_pos %= azx_dev->core.period_bytes; - /* azx_dev->fifo_size can't get FIFO size of in stream. - * Get from base address + offset. - */ - fifo_size = readw(azx_bus(chip)->remap_addr + - VIA_IN_STREAM0_FIFO_SIZE_OFFSET); + fifo_size = azx_stream(azx_dev)->fifo_size - 1; if (azx_dev->insufficient) { /* Link position never gather than FIFO size */ -- 2.16.4

1 0

[alsa-devel] [PATCH 0/3] soundwire: debugfs support for 5.4
by Pierre-Louis Bossart 12 Aug '19

12 Aug '19

This patchset enables debugfs support and corrects all the feedback provided on an earlier RFC ('soundwire: updates for 5.4') There is one remaining hard-coded value in intel.c that will need to be fixed in a follow-up patchset not specific to debugfs: we need to remove hard-coded Intel-specific configurations from cadence_master.c (PDI offsets, etc). Changes since RFC (Feedback from GKH, Vinod, Guennadi, Cezary, Sanyog): removed error checks used DEFINE_SHOW_ATTRIBUTE and seq_file fixed copyright dates fixed SPDX license info to use GPL2.0 only fixed Makefile to include debugfs only if CONFIG_DEBUG_FS is selected used static inlines for fallback compilation removed intermediate variables removed hard-coded constants in loops (used registers offsets and hardware capabilities) squashed patch 3 Pierre-Louis Bossart (3): soundwire: add debugfs support soundwire: cadence_master: add debugfs register dump soundwire: intel: add debugfs register dump drivers/soundwire/Makefile | 5 + drivers/soundwire/bus.c | 6 ++ drivers/soundwire/bus.h | 24 +++++ drivers/soundwire/bus_type.c | 3 + drivers/soundwire/cadence_master.c | 104 ++++++++++++++++++++ drivers/soundwire/cadence_master.h | 2 + drivers/soundwire/debugfs.c | 151 +++++++++++++++++++++++++++++ drivers/soundwire/intel.c | 114 ++++++++++++++++++++++ drivers/soundwire/slave.c | 1 + include/linux/soundwire/sdw.h | 4 + 10 files changed, 414 insertions(+) create mode 100644 drivers/soundwire/debugfs.c -- 2.20.1

2 8

[alsa-devel] [PATCH -next] ASoC: max98926: remove two unused variables
by YueHaibing 12 Aug '19

12 Aug '19

sound/soc/codecs/max98926.c:28:26: warning: max98926_dai_txt defined but not used [-Wunused-const-variable=] sound/soc/codecs/max98926.c:23:27: warning: max98926_boost_current_txt defined but not used [-Wunused-const-variable=] They are never used, so can be removd. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/max98926.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c index 818c030..c4dfa8a 100644 --- a/sound/soc/codecs/max98926.c +++ b/sound/soc/codecs/max98926.c @@ -20,15 +20,6 @@ static const char * const max98926_boost_voltage_txt[] = { "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V" }; -static const char * const max98926_boost_current_txt[] = { - "0.6", "0.8", "1.0", "1.2", "1.4", "1.6", "1.8", "2.0", - "2.2", "2.4", "2.6", "2.8", "3.2", "3.6", "4.0", "4.4" -}; - -static const char *const max98926_dai_txt[] = { - "Left", "Right", "LeftRight", "LeftRightDiv2", -}; - static const char *const max98926_pdm_ch_text[] = { "Current", "Voltage", }; -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: max9850: remove unused variable 'max9850_reg'
by YueHaibing 12 Aug '19

12 Aug '19

sound/soc/codecs/max9850.c:31:33: warning: max9850_reg defined but not used [-Wunused-const-variable=] It is not used since commit 068416620c0d ("ASoC: max9850: Convert to direct regmap API usage") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/max9850.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index f50ee8f..6f43748 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -27,19 +27,6 @@ struct max9850_priv { unsigned int sysclk; }; -/* max9850 register cache */ -static const struct reg_default max9850_reg[] = { - { 2, 0x0c }, - { 3, 0x00 }, - { 4, 0x00 }, - { 5, 0x00 }, - { 6, 0x00 }, - { 7, 0x00 }, - { 8, 0x00 }, - { 9, 0x00 }, - { 10, 0x00 }, -}; - /* these registers are not used at the moment but provided for the sake of * completeness */ static bool max9850_volatile_register(struct device *dev, unsigned int reg) -- 2.7.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: fsl_esai: Add compatible string for imx6ull
by Shengjiu Wang 12 Aug '19

12 Aug '19

Add compatible string for imx6ull, from imx6ull platform, the issue of channel swap after xrun is fixed in hardware. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_esai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 10d2210c91ef..4b4a8e831e9e 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -920,6 +920,7 @@ static int fsl_esai_remove(struct platform_device *pdev) static const struct of_device_id fsl_esai_dt_ids[] = { { .compatible = "fsl,imx35-esai", }, { .compatible = "fsl,vf610-esai", }, + { .compatible = "fsl,imx6ull-esai", }, {} }; MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids); -- 2.21.0

3 5

[alsa-devel] [PATCH 00/16] ARM: remove ks8695 and w90x900 platforms
by Arnd Bergmann 12 Aug '19

12 Aug '19

As discussed previously, these two ARM platforms have no known remaining users, let's remove them completely. Subsystem maintainers: feel free to take the driver removals through your respective trees, they are all independent of one another. We can merge any remaining patches through the soc tree. Arnd Arnd Bergmann (16): ARM: remove ks8695 platform serial: remove ks8695 driver gpio: remove ks8695 driver watchdog: remove ks8695 driver net: remove ks8695 driver watchdog: remove w90x900 driver spi: remove w90x900 driver ASoC: remove w90x900/nuc900 platform drivers fbdev: remove w90x900/nuc900 platform drivers Input: remove w90x900 keyboard driver Input: remove w90x900 touchscreen driver mtd: rawnand: remove w90x900 driver net: remove w90p910-ether driver rtc: remove w90x900/nuc900 driver usb: remove ehci-w90x900 driver ARM: remove w90x900 platform Cc: "Wanzongshun (Vincent)" <wanzongshun(a)huawei.com> Cc: Greg Ungerer <gerg(a)kernel.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: linux-serial(a)vger.kernel.org Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: linux-input(a)vger.kernel.org Cc: Linus Walleij <linus.walleij(a)linaro.org> Cc: linux-gpio(a)vger.kernel.org Cc: "David S. Miller" <davem(a)davemloft.net> Cc: netdev(a)vger.kernel.org Cc: Guenter Roeck <linux(a)roeck-us.net> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: linux-spi(a)vger.kernel.org Cc: Bartlomiej Zolnierkiewicz <b.zolnierkie(a)samsung.com> Cc: linux-fbdev(a)vger.kernel.org Cc: Miquel Raynal <miquel.raynal(a)bootlin.com> Cc: linux-mtd(a)lists.infradead.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-kernel(a)vger.kernel.org .../watchdog/watchdog-parameters.rst | 19 - MAINTAINERS | 22 - arch/arm/Kconfig | 34 +- arch/arm/Kconfig.debug | 8 - arch/arm/Makefile | 2 - arch/arm/configs/acs5k_defconfig | 77 - arch/arm/configs/acs5k_tiny_defconfig | 69 - arch/arm/configs/ks8695_defconfig | 67 - arch/arm/configs/nuc910_defconfig | 51 - arch/arm/configs/nuc950_defconfig | 67 - arch/arm/configs/nuc960_defconfig | 57 - arch/arm/include/debug/ks8695.S | 37 - arch/arm/mach-ks8695/Kconfig | 88 - arch/arm/mach-ks8695/Makefile | 23 - arch/arm/mach-ks8695/Makefile.boot | 9 - arch/arm/mach-ks8695/board-acs5k.c | 238 --- arch/arm/mach-ks8695/board-dsm320.c | 127 -- arch/arm/mach-ks8695/board-micrel.c | 59 - arch/arm/mach-ks8695/board-og.c | 197 -- arch/arm/mach-ks8695/board-sg.c | 118 -- arch/arm/mach-ks8695/cpu.c | 60 - arch/arm/mach-ks8695/devices.c | 197 -- arch/arm/mach-ks8695/devices.h | 29 - arch/arm/mach-ks8695/generic.h | 12 - .../mach-ks8695/include/mach/entry-macro.S | 47 - .../mach-ks8695/include/mach/gpio-ks8695.h | 36 - arch/arm/mach-ks8695/include/mach/hardware.h | 42 - arch/arm/mach-ks8695/include/mach/irqs.h | 51 - arch/arm/mach-ks8695/include/mach/memory.h | 51 - arch/arm/mach-ks8695/include/mach/regs-gpio.h | 55 - arch/arm/mach-ks8695/include/mach/regs-irq.h | 41 - arch/arm/mach-ks8695/include/mach/regs-misc.h | 97 - .../mach-ks8695/include/mach/regs-switch.h | 66 - arch/arm/mach-ks8695/include/mach/regs-uart.h | 89 - .../arm/mach-ks8695/include/mach/uncompress.h | 33 - arch/arm/mach-ks8695/irq.c | 164 -- arch/arm/mach-ks8695/pci.c | 247 --- arch/arm/mach-ks8695/regs-hpna.h | 25 - arch/arm/mach-ks8695/regs-lan.h | 65 - arch/arm/mach-ks8695/regs-mem.h | 89 - arch/arm/mach-ks8695/regs-pci.h | 53 - arch/arm/mach-ks8695/regs-sys.h | 34 - arch/arm/mach-ks8695/regs-wan.h | 65 - arch/arm/mach-ks8695/time.c | 159 -- arch/arm/mach-w90x900/Kconfig | 50 - arch/arm/mach-w90x900/Makefile | 20 - arch/arm/mach-w90x900/Makefile.boot | 4 - arch/arm/mach-w90x900/clksel.c | 88 - arch/arm/mach-w90x900/clock.c | 121 -- arch/arm/mach-w90x900/clock.h | 40 - arch/arm/mach-w90x900/cpu.c | 238 --- arch/arm/mach-w90x900/cpu.h | 56 - arch/arm/mach-w90x900/dev.c | 537 ------ arch/arm/mach-w90x900/gpio.c | 150 -- .../mach-w90x900/include/mach/entry-macro.S | 26 - arch/arm/mach-w90x900/include/mach/hardware.h | 19 - arch/arm/mach-w90x900/include/mach/irqs.h | 82 - arch/arm/mach-w90x900/include/mach/map.h | 153 -- arch/arm/mach-w90x900/include/mach/mfp.h | 21 - .../mach-w90x900/include/mach/regs-clock.h | 49 - arch/arm/mach-w90x900/include/mach/regs-irq.h | 46 - arch/arm/mach-w90x900/include/mach/regs-ldm.h | 248 --- .../mach-w90x900/include/mach/regs-serial.h | 54 - .../mach-w90x900/include/mach/uncompress.h | 43 - arch/arm/mach-w90x900/irq.c | 212 --- arch/arm/mach-w90x900/mach-nuc910evb.c | 38 - arch/arm/mach-w90x900/mach-nuc950evb.c | 42 - arch/arm/mach-w90x900/mach-nuc960evb.c | 38 - arch/arm/mach-w90x900/mfp.c | 197 -- arch/arm/mach-w90x900/nuc910.c | 58 - arch/arm/mach-w90x900/nuc910.h | 17 - arch/arm/mach-w90x900/nuc950.c | 52 - arch/arm/mach-w90x900/nuc950.h | 17 - arch/arm/mach-w90x900/nuc960.c | 50 - arch/arm/mach-w90x900/nuc960.h | 17 - arch/arm/mach-w90x900/nuc9xx.h | 22 - arch/arm/mach-w90x900/regs-ebi.h | 29 - arch/arm/mach-w90x900/regs-gcr.h | 34 - arch/arm/mach-w90x900/regs-timer.h | 37 - arch/arm/mach-w90x900/regs-usb.h | 31 - arch/arm/mach-w90x900/time.c | 168 -- arch/arm/mm/Kconfig | 2 +- drivers/gpio/Makefile | 1 - drivers/gpio/gpio-ks8695.c | 284 --- drivers/input/keyboard/Kconfig | 11 - drivers/input/keyboard/Makefile | 1 - drivers/input/keyboard/w90p910_keypad.c | 264 --- drivers/input/touchscreen/Kconfig | 9 - drivers/input/touchscreen/Makefile | 1 - drivers/input/touchscreen/w90p910_ts.c | 331 ---- drivers/mtd/nand/raw/Kconfig | 8 - drivers/mtd/nand/raw/Makefile | 1 - drivers/mtd/nand/raw/nuc900_nand.c | 304 --- drivers/net/ethernet/Kconfig | 1 - drivers/net/ethernet/Makefile | 1 - drivers/net/ethernet/micrel/Kconfig | 11 +- drivers/net/ethernet/micrel/Makefile | 1 - drivers/net/ethernet/micrel/ks8695net.c | 1632 ----------------- drivers/net/ethernet/micrel/ks8695net.h | 108 -- drivers/net/ethernet/nuvoton/Kconfig | 29 - drivers/net/ethernet/nuvoton/Makefile | 6 - drivers/net/ethernet/nuvoton/w90p910_ether.c | 1082 ----------- drivers/rtc/Kconfig | 7 - drivers/rtc/Makefile | 1 - drivers/rtc/rtc-nuc900.c | 271 --- drivers/spi/Kconfig | 7 - drivers/spi/Makefile | 1 - drivers/spi/spi-nuc900.c | 429 ----- drivers/tty/serial/Kconfig | 17 - drivers/tty/serial/Makefile | 1 - drivers/tty/serial/serial_ks8695.c | 698 ------- drivers/usb/host/Kconfig | 6 - drivers/usb/host/Makefile | 1 - drivers/usb/host/ehci-w90x900.c | 130 -- drivers/video/fbdev/Kconfig | 14 - drivers/video/fbdev/Makefile | 1 - drivers/video/fbdev/nuc900fb.c | 760 -------- drivers/video/fbdev/nuc900fb.h | 51 - drivers/watchdog/Kconfig | 16 - drivers/watchdog/Makefile | 2 - drivers/watchdog/ks8695_wdt.c | 319 ---- drivers/watchdog/nuc900_wdt.c | 302 --- include/Kbuild | 2 - include/linux/platform_data/keypad-w90p910.h | 16 - include/linux/platform_data/spi-nuc900.h | 29 - include/linux/platform_data/video-nuc900fb.h | 79 - include/uapi/linux/serial_core.h | 3 - sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - sound/soc/nuc900/Kconfig | 29 - sound/soc/nuc900/Makefile | 12 - sound/soc/nuc900/nuc900-ac97.c | 391 ---- sound/soc/nuc900/nuc900-audio.c | 73 - sound/soc/nuc900/nuc900-audio.h | 108 -- sound/soc/nuc900/nuc900-pcm.c | 321 ---- 135 files changed, 6 insertions(+), 14461 deletions(-) delete mode 100644 arch/arm/configs/acs5k_defconfig delete mode 100644 arch/arm/configs/acs5k_tiny_defconfig delete mode 100644 arch/arm/configs/ks8695_defconfig delete mode 100644 arch/arm/configs/nuc910_defconfig delete mode 100644 arch/arm/configs/nuc950_defconfig delete mode 100644 arch/arm/configs/nuc960_defconfig delete mode 100644 arch/arm/include/debug/ks8695.S delete mode 100644 arch/arm/mach-ks8695/Kconfig delete mode 100644 arch/arm/mach-ks8695/Makefile delete mode 100644 arch/arm/mach-ks8695/Makefile.boot delete mode 100644 arch/arm/mach-ks8695/board-acs5k.c delete mode 100644 arch/arm/mach-ks8695/board-dsm320.c delete mode 100644 arch/arm/mach-ks8695/board-micrel.c delete mode 100644 arch/arm/mach-ks8695/board-og.c delete mode 100644 arch/arm/mach-ks8695/board-sg.c delete mode 100644 arch/arm/mach-ks8695/cpu.c delete mode 100644 arch/arm/mach-ks8695/devices.c delete mode 100644 arch/arm/mach-ks8695/devices.h delete mode 100644 arch/arm/mach-ks8695/generic.h delete mode 100644 arch/arm/mach-ks8695/include/mach/entry-macro.S delete mode 100644 arch/arm/mach-ks8695/include/mach/gpio-ks8695.h delete mode 100644 arch/arm/mach-ks8695/include/mach/hardware.h delete mode 100644 arch/arm/mach-ks8695/include/mach/irqs.h delete mode 100644 arch/arm/mach-ks8695/include/mach/memory.h delete mode 100644 arch/arm/mach-ks8695/include/mach/regs-gpio.h delete mode 100644 arch/arm/mach-ks8695/include/mach/regs-irq.h delete mode 100644 arch/arm/mach-ks8695/include/mach/regs-misc.h delete mode 100644 arch/arm/mach-ks8695/include/mach/regs-switch.h delete mode 100644 arch/arm/mach-ks8695/include/mach/regs-uart.h delete mode 100644 arch/arm/mach-ks8695/include/mach/uncompress.h delete mode 100644 arch/arm/mach-ks8695/irq.c delete mode 100644 arch/arm/mach-ks8695/pci.c delete mode 100644 arch/arm/mach-ks8695/regs-hpna.h delete mode 100644 arch/arm/mach-ks8695/regs-lan.h delete mode 100644 arch/arm/mach-ks8695/regs-mem.h delete mode 100644 arch/arm/mach-ks8695/regs-pci.h delete mode 100644 arch/arm/mach-ks8695/regs-sys.h delete mode 100644 arch/arm/mach-ks8695/regs-wan.h delete mode 100644 arch/arm/mach-ks8695/time.c delete mode 100644 arch/arm/mach-w90x900/Kconfig delete mode 100644 arch/arm/mach-w90x900/Makefile delete mode 100644 arch/arm/mach-w90x900/Makefile.boot delete mode 100644 arch/arm/mach-w90x900/clksel.c delete mode 100644 arch/arm/mach-w90x900/clock.c delete mode 100644 arch/arm/mach-w90x900/clock.h delete mode 100644 arch/arm/mach-w90x900/cpu.c delete mode 100644 arch/arm/mach-w90x900/cpu.h delete mode 100644 arch/arm/mach-w90x900/dev.c delete mode 100644 arch/arm/mach-w90x900/gpio.c delete mode 100644 arch/arm/mach-w90x900/include/mach/entry-macro.S delete mode 100644 arch/arm/mach-w90x900/include/mach/hardware.h delete mode 100644 arch/arm/mach-w90x900/include/mach/irqs.h delete mode 100644 arch/arm/mach-w90x900/include/mach/map.h delete mode 100644 arch/arm/mach-w90x900/include/mach/mfp.h delete mode 100644 arch/arm/mach-w90x900/include/mach/regs-clock.h delete mode 100644 arch/arm/mach-w90x900/include/mach/regs-irq.h delete mode 100644 arch/arm/mach-w90x900/include/mach/regs-ldm.h delete mode 100644 arch/arm/mach-w90x900/include/mach/regs-serial.h delete mode 100644 arch/arm/mach-w90x900/include/mach/uncompress.h delete mode 100644 arch/arm/mach-w90x900/irq.c delete mode 100644 arch/arm/mach-w90x900/mach-nuc910evb.c delete mode 100644 arch/arm/mach-w90x900/mach-nuc950evb.c delete mode 100644 arch/arm/mach-w90x900/mach-nuc960evb.c delete mode 100644 arch/arm/mach-w90x900/mfp.c delete mode 100644 arch/arm/mach-w90x900/nuc910.c delete mode 100644 arch/arm/mach-w90x900/nuc910.h delete mode 100644 arch/arm/mach-w90x900/nuc950.c delete mode 100644 arch/arm/mach-w90x900/nuc950.h delete mode 100644 arch/arm/mach-w90x900/nuc960.c delete mode 100644 arch/arm/mach-w90x900/nuc960.h delete mode 100644 arch/arm/mach-w90x900/nuc9xx.h delete mode 100644 arch/arm/mach-w90x900/regs-ebi.h delete mode 100644 arch/arm/mach-w90x900/regs-gcr.h delete mode 100644 arch/arm/mach-w90x900/regs-timer.h delete mode 100644 arch/arm/mach-w90x900/regs-usb.h delete mode 100644 arch/arm/mach-w90x900/time.c delete mode 100644 drivers/gpio/gpio-ks8695.c delete mode 100644 drivers/input/keyboard/w90p910_keypad.c delete mode 100644 drivers/input/touchscreen/w90p910_ts.c delete mode 100644 drivers/mtd/nand/raw/nuc900_nand.c delete mode 100644 drivers/net/ethernet/micrel/ks8695net.c delete mode 100644 drivers/net/ethernet/micrel/ks8695net.h delete mode 100644 drivers/net/ethernet/nuvoton/Kconfig delete mode 100644 drivers/net/ethernet/nuvoton/Makefile delete mode 100644 drivers/net/ethernet/nuvoton/w90p910_ether.c delete mode 100644 drivers/rtc/rtc-nuc900.c delete mode 100644 drivers/spi/spi-nuc900.c delete mode 100644 drivers/tty/serial/serial_ks8695.c delete mode 100644 drivers/usb/host/ehci-w90x900.c delete mode 100644 drivers/video/fbdev/nuc900fb.c delete mode 100644 drivers/video/fbdev/nuc900fb.h delete mode 100644 drivers/watchdog/ks8695_wdt.c delete mode 100644 drivers/watchdog/nuc900_wdt.c delete mode 100644 include/linux/platform_data/keypad-w90p910.h delete mode 100644 include/linux/platform_data/spi-nuc900.h delete mode 100644 include/linux/platform_data/video-nuc900fb.h delete mode 100644 sound/soc/nuc900/Kconfig delete mode 100644 sound/soc/nuc900/Makefile delete mode 100644 sound/soc/nuc900/nuc900-ac97.c delete mode 100644 sound/soc/nuc900/nuc900-audio.c delete mode 100644 sound/soc/nuc900/nuc900-audio.h delete mode 100644 sound/soc/nuc900/nuc900-pcm.c -- 2.20.0

3 4

[alsa-devel] [PATCH] ASoC: rt5677: Revert remove superfluous set
by Curtis Malainey 12 Aug '19

12 Aug '19

Commit 4dc057a786dc23 ("ASoC: rt5677: remove superfluous set") was an attempted code cleanup but was incorrectly tested before sent and actually breaks the interrupt since it never resets the value on each loop now. The breakage is most testable when hotwording code is added and also uses the interrupt. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index deaed5132dc95..c779dc3474f9e 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5098,6 +5098,7 @@ static irqreturn_t rt5677_irq(int unused, void *data) goto exit; } + irq_fired = false; for (i = 0; i < RT5677_IRQ_NUM; i++) { if (reg_irq & rt5677_irq_descs[i].status_mask) { irq_fired = true; -- 2.23.0.rc1.153.gdeed80330f-goog

2 1

[alsa-devel] [RFC PATCH] ASoC: dapm: Invalidate only paths reachable for a given stream
by Szymon Mielczarek 12 Aug '19

12 Aug '19

By resetting the cached number of endpoints for all card's widgets we may overwrite previously cached values for other streams. The situation may happen especially when running streams simultaneously. Signed-off-by: Szymon Mielczarek <szymonx.mielczarek(a)linux.intel.com> --- sound/soc/soc-dapm.c | 50 ++++++++++++++++++++++++++++++++------------ 1 file changed, 37 insertions(+), 13 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d09bdca63c62..10819b3e0b98 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1128,6 +1128,34 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, return 0; } +/* + * Recursively reset the cached number of inputs or outputs for the specified + * widget and all widgets that can be reached via incoming or outcoming paths + * from the widget. + */ +static void invalidate_paths_ep(struct snd_soc_dapm_widget *widget, + enum snd_soc_dapm_direction dir) +{ + enum snd_soc_dapm_direction rdir = SND_SOC_DAPM_DIR_REVERSE(dir); + struct snd_soc_dapm_path *path; + + widget->endpoints[dir] = -1; + + snd_soc_dapm_widget_for_each_path(widget, rdir, path) { + if (path->weak || path->is_supply) + continue; + + if (path->walking) + return; + + if (path->connect) { + path->walking = 1; + invalidate_paths_ep(path->node[dir], dir); + path->walking = 0; + } + } +} + /* * Common implementation for is_connected_output_ep() and * is_connected_input_ep(). The function is inlined since the combined size of @@ -1257,21 +1285,17 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - /* - * For is_connected_{output,input}_ep fully discover the graph we need - * to reset the cached number of inputs and outputs. - */ - list_for_each_entry(w, &card->widgets, list) { - w->endpoints[SND_SOC_DAPM_DIR_IN] = -1; - w->endpoints[SND_SOC_DAPM_DIR_OUT] = -1; - } - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - paths = is_connected_output_ep(dai->playback_widget, &widgets, + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + w = dai->playback_widget; + invalidate_paths_ep(w, SND_SOC_DAPM_DIR_OUT); + paths = is_connected_output_ep(w, &widgets, custom_stop_condition); - else - paths = is_connected_input_ep(dai->capture_widget, &widgets, + } else { + w = dai->capture_widget; + invalidate_paths_ep(w, SND_SOC_DAPM_DIR_IN); + paths = is_connected_input_ep(w, &widgets, custom_stop_condition); + } /* Drop starting point */ list_del(widgets.next); -- 2.17.1

2 1

[alsa-devel] [PATCH v2] ASoC: SOF: topology: use set_get_data in process load
by Pierre-Louis Bossart 12 Aug '19

12 Aug '19

From: Jaska Uimonen <jaska.uimonen(a)intel.com> Currently when loading sof process components there's a check if binary control data is associated with it. If found the data is extracted to be part of component loading and initialization. If binary data exceeds the ipc max size, loading fails with error as large message support is only implemented in set_get_data method. So make the process loading use set_get_data to enable large parameters in component initialization. Also refactor the process component loading function as it digs out 3 times almost identical information of related controls. This is redundant, looks ugly and makes it difficult to understand the mechanism. So make a function out of fetching the control data and use it in process loading. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Jaska Uimonen <jaska.uimonen(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2: simplify error handling with gotos as suggested by Cezary. sound/soc/sof/topology.c | 201 +++++++++++++++++++++++++-------------- 1 file changed, 127 insertions(+), 74 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 12b7d900b9c2..9cffea142395 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -42,6 +42,13 @@ /* size of tplg abi in byte */ #define SOF_TPLG_ABI_SIZE 3 +struct sof_widget_data { + int ctrl_type; + int ipc_cmd; + struct sof_abi_hdr *pdata; + struct snd_sof_control *control; +}; + /* send pcm params ipc */ static int ipc_pcm_params(struct snd_sof_widget *swidget, int dir) { @@ -1742,51 +1749,32 @@ static int sof_widget_load_siggen(struct snd_soc_component *scomp, int index, return ret; } -static int sof_process_load(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, - struct snd_soc_tplg_dapm_widget *tw, - struct sof_ipc_comp_reply *r, - int type) +static size_t sof_get_control_data(struct snd_sof_dev *sdev, + struct snd_soc_dapm_widget *widget, + struct sof_widget_data *wdata) { - struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); - struct snd_soc_tplg_private *private = &tw->priv; - struct snd_soc_dapm_widget *widget = swidget->widget; const struct snd_kcontrol_new *kc; - struct soc_bytes_ext *sbe; struct soc_mixer_control *sm; + struct soc_bytes_ext *sbe; struct soc_enum *se; - struct snd_sof_control *scontrol = NULL; - struct sof_abi_hdr *pdata = NULL; - struct sof_ipc_comp_process *process; - size_t ipc_size, ipc_data_size = 0; - int ret, i, offset = 0; - - if (type == SOF_COMP_NONE) { - dev_err(sdev->dev, "error: invalid process comp type %d\n", - type); - return -EINVAL; - } + size_t size = 0; + int i; - /* - * get possible component controls - get size of all pdata, - * then memcpy with headers - */ for (i = 0; i < widget->num_kcontrols; i++) { - kc = &widget->kcontrol_news[i]; switch (widget->dobj.widget.kcontrol_type) { case SND_SOC_TPLG_TYPE_MIXER: sm = (struct soc_mixer_control *)kc->private_value; - scontrol = sm->dobj.private; + wdata[i].control = sm->dobj.private; break; case SND_SOC_TPLG_TYPE_BYTES: sbe = (struct soc_bytes_ext *)kc->private_value; - scontrol = sbe->dobj.private; + wdata[i].control = sbe->dobj.private; break; case SND_SOC_TPLG_TYPE_ENUM: se = (struct soc_enum *)kc->private_value; - scontrol = se->dobj.private; + wdata[i].control = se->dobj.private; break; default: dev_err(sdev->dev, "error: unknown kcontrol type %d in widget %s\n", @@ -1795,31 +1783,98 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, return -EINVAL; } - if (!scontrol) { + if (!wdata[i].control) { dev_err(sdev->dev, "error: no scontrol for widget %s\n", widget->name); return -EINVAL; } - /* don't include if no private data */ - pdata = scontrol->control_data->data; - if (!pdata) - continue; + wdata[i].pdata = wdata[i].control->control_data->data; + if (!wdata[i].pdata) + return -EINVAL; /* make sure data is valid - data can be updated at runtime */ - if (pdata->magic != SOF_ABI_MAGIC) - continue; + if (wdata[i].pdata->magic != SOF_ABI_MAGIC) + return -EINVAL; + + size += wdata[i].pdata->size; + + /* get data type */ + switch (wdata[i].control->cmd) { + case SOF_CTRL_CMD_VOLUME: + case SOF_CTRL_CMD_ENUM: + case SOF_CTRL_CMD_SWITCH: + wdata[i].ipc_cmd = SOF_IPC_COMP_SET_VALUE; + wdata[i].ctrl_type = SOF_CTRL_TYPE_VALUE_CHAN_SET; + break; + case SOF_CTRL_CMD_BINARY: + wdata[i].ipc_cmd = SOF_IPC_COMP_SET_DATA; + wdata[i].ctrl_type = SOF_CTRL_TYPE_DATA_SET; + break; + default: + break; + } + } + + return size; +} + +static int sof_process_load(struct snd_soc_component *scomp, int index, + struct snd_sof_widget *swidget, + struct snd_soc_tplg_dapm_widget *tw, + struct sof_ipc_comp_reply *r, + int type) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_dapm_widget *widget = swidget->widget; + struct snd_soc_tplg_private *private = &tw->priv; + struct sof_ipc_comp_process *process = NULL; + struct sof_widget_data *wdata = NULL; + size_t ipc_data_size = 0; + size_t ipc_size; + int offset = 0; + int ret = 0; + int i; + + if (type == SOF_COMP_NONE) { + dev_err(sdev->dev, "error: invalid process comp type %d\n", + type); + return -EINVAL; + } + + /* allocate struct for widget control data sizes and types */ + if (widget->num_kcontrols) { + wdata = kcalloc(widget->num_kcontrols, + sizeof(*wdata), + GFP_KERNEL); + + if (!wdata) + return -ENOMEM; - ipc_data_size += pdata->size; + /* get possible component controls and get size of all pdata */ + ipc_data_size = sof_get_control_data(sdev, widget, wdata); + + if (ipc_data_size <= 0) { + ret = ipc_data_size; + goto out; + } } ipc_size = sizeof(struct sof_ipc_comp_process) + le32_to_cpu(private->size) + ipc_data_size; + /* we are exceeding max ipc size, config needs to be sent separately */ + if (ipc_size > SOF_IPC_MSG_MAX_SIZE) { + ipc_size -= ipc_data_size; + ipc_data_size = 0; + } + process = kzalloc(ipc_size, GFP_KERNEL); - if (!process) - return -ENOMEM; + if (!process) { + ret = -ENOMEM; + goto out; + } /* configure iir IPC message */ process->comp.hdr.size = ipc_size; @@ -1845,40 +1900,13 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, * get possible component controls - get size of all pdata, * then memcpy with headers */ - for (i = 0; i < widget->num_kcontrols; i++) { - kc = &widget->kcontrol_news[i]; - - switch (widget->dobj.widget.kcontrol_type) { - case SND_SOC_TPLG_TYPE_MIXER: - sm = (struct soc_mixer_control *)kc->private_value; - scontrol = sm->dobj.private; - break; - case SND_SOC_TPLG_TYPE_BYTES: - sbe = (struct soc_bytes_ext *)kc->private_value; - scontrol = sbe->dobj.private; - break; - case SND_SOC_TPLG_TYPE_ENUM: - se = (struct soc_enum *)kc->private_value; - scontrol = se->dobj.private; - break; - default: - dev_err(sdev->dev, "error: unknown kcontrol type %d in widget %s\n", - widget->dobj.widget.kcontrol_type, - widget->name); - return -EINVAL; + if (ipc_data_size) { + for (i = 0; i < widget->num_kcontrols; i++) { + memcpy(&process->data + offset, + wdata[i].pdata->data, + wdata[i].pdata->size); + offset += wdata[i].pdata->size; } - - /* don't include if no private data */ - pdata = scontrol->control_data->data; - if (!pdata) - continue; - - /* make sure data is valid - data can be updated at runtime */ - if (pdata->magic != SOF_ABI_MAGIC) - continue; - - memcpy(&process->data + offset, pdata->data, pdata->size); - offset += pdata->size; } process->size = ipc_data_size; @@ -1886,10 +1914,35 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, ret = sof_ipc_tx_message(sdev->ipc, process->comp.hdr.cmd, process, ipc_size, r, sizeof(*r)); - if (ret >= 0) - return ret; + + if (ret < 0) { + dev_err(sdev->dev, "error: create process failed\n"); + goto err; + } + + /* we sent the data in single message so return */ + if (ipc_data_size) + goto out; + + /* send control data with large message supported method */ + for (i = 0; i < widget->num_kcontrols; i++) { + wdata[i].control->readback_offset = 0; + ret = snd_sof_ipc_set_get_comp_data(sdev->ipc, wdata[i].control, + wdata[i].ipc_cmd, + wdata[i].ctrl_type, + wdata[i].control->cmd, + true); + if (ret != 0) { + dev_err(sdev->dev, "error: send control failed\n"); + break; + } + } + err: - kfree(process); + if (ret < 0) + kfree(process); +out: + kfree(wdata); return ret; } -- 2.20.1

2 1

[alsa-devel] [PATCH 0/4] ASoC: Intel: boards: quirks and DMIC corrections
by Pierre-Louis Bossart 12 Aug '19

12 Aug '19

Extend quirk for GeminiLake-based Chromebooks and correct missing dependencies for DMIC support (align with all other machine drivers). Keyon Jie (3): ASoC: Intel: sof-rt5682: add dmic dapm widget to support dmic PCM ASoC: Intel: skl-hda-dsp-generic: add dependency to dmic driver ASoC: Intel: skl-hda-dsp-generic: add dmic dapm widget and route Sathya Prakash M R (1): ASoC: Intel: boards: Match Product Family instead of product sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 4 ++++ sound/soc/intel/boards/sof_rt5682.c | 7 +++++-- 3 files changed, 10 insertions(+), 2 deletions(-) -- 2.20.1

2 8

[alsa-devel] Applied "ASoC: pcm3168a: Retain the independence of DAC and ADC side of the codec" to the asoc tree
by Mark Brown 12 Aug '19

12 Aug '19

The patch ASoC: pcm3168a: Retain the independence of DAC and ADC side of the codec has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From abe51c351827e0086dad079dfe02918fecdf4830 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Mon, 12 Aug 2019 12:52:25 +0300 Subject: [PATCH] ASoC: pcm3168a: Retain the independence of DAC and ADC side of the codec The DAC and ADC path of the codec is independent, have dedicated LRCK (FS) and BCK for DAC/ADC. They can be configured to use different format, TDM slots and slot_width if needed. Move these parameters under dedicated io_params structure and manage them independently based on the dai. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Link: https://lore.kernel.org/r/20190812095226.18870-2-peter.ujfalusi@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm3168a.c | 125 ++++++++++++++++-------------------- 1 file changed, 54 insertions(+), 71 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index e84a1509fe65..75fa8e9ee894 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -44,18 +44,25 @@ static const char *const pcm3168a_supply_names[PCM3168A_NUM_SUPPLIES] = { "VCCDA2" }; +#define PCM3168A_DAI_DAC 0 +#define PCM3168A_DAI_ADC 1 + +/* ADC/DAC side parameters */ +struct pcm3168a_io_params { + bool master_mode; + unsigned int fmt; + int tdm_slots; + u32 tdm_mask; + int slot_width; +}; + struct pcm3168a_priv { struct regulator_bulk_data supplies[PCM3168A_NUM_SUPPLIES]; struct regmap *regmap; struct clk *scki; - bool adc_master_mode; - bool dac_master_mode; unsigned long sysclk; - unsigned int adc_fmt; - unsigned int dac_fmt; - int tdm_slots; - u32 tdm_mask[2]; - int slot_width; + + struct pcm3168a_io_params io_params[2]; }; static const char *const pcm3168a_roll_off[] = { "Sharp", "Slow" }; @@ -308,8 +315,7 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai, return 0; } -static int pcm3168a_set_dai_fmt(struct snd_soc_dai *dai, - unsigned int format, bool dac) +static int pcm3168a_set_dai_fmt(struct snd_soc_dai *dai, unsigned int format) { struct snd_soc_component *component = dai->component; struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); @@ -356,43 +362,31 @@ static int pcm3168a_set_dai_fmt(struct snd_soc_dai *dai, return -EINVAL; } - if (dac) { + if (dai->id == PCM3168A_DAI_DAC) { reg = PCM3168A_DAC_PWR_MST_FMT; mask = PCM3168A_DAC_FMT_MASK; shift = PCM3168A_DAC_FMT_SHIFT; - pcm3168a->dac_master_mode = master_mode; - pcm3168a->dac_fmt = fmt; } else { reg = PCM3168A_ADC_MST_FMT; mask = PCM3168A_ADC_FMTAD_MASK; shift = PCM3168A_ADC_FMTAD_SHIFT; - pcm3168a->adc_master_mode = master_mode; - pcm3168a->adc_fmt = fmt; } + pcm3168a->io_params[dai->id].master_mode = master_mode; + pcm3168a->io_params[dai->id].fmt = fmt; + regmap_update_bits(pcm3168a->regmap, reg, mask, fmt << shift); return 0; } -static int pcm3168a_set_dai_fmt_dac(struct snd_soc_dai *dai, - unsigned int format) -{ - return pcm3168a_set_dai_fmt(dai, format, true); -} - -static int pcm3168a_set_dai_fmt_adc(struct snd_soc_dai *dai, - unsigned int format) -{ - return pcm3168a_set_dai_fmt(dai, format, false); -} - static int pcm3168a_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_component *component = dai->component; struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); + struct pcm3168a_io_params *io_params = &pcm3168a->io_params[dai->id]; if (tx_mask >= (1<<slots) || rx_mask >= (1<<slots)) { dev_err(component->dev, @@ -408,22 +402,25 @@ static int pcm3168a_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, return -EINVAL; } - if (pcm3168a->tdm_slots && pcm3168a->tdm_slots != slots) { + if (io_params->tdm_slots && io_params->tdm_slots != slots) { dev_err(component->dev, "Not matching slots %d vs %d\n", - pcm3168a->tdm_slots, slots); + io_params->tdm_slots, slots); return -EINVAL; } - if (pcm3168a->slot_width && pcm3168a->slot_width != slot_width) { + if (io_params->slot_width && io_params->slot_width != slot_width) { dev_err(component->dev, "Not matching slot_width %d vs %d\n", - pcm3168a->slot_width, slot_width); + io_params->slot_width, slot_width); return -EINVAL; } - pcm3168a->tdm_slots = slots; - pcm3168a->slot_width = slot_width; - pcm3168a->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask; - pcm3168a->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = rx_mask; + io_params->tdm_slots = slots; + io_params->slot_width = slot_width; + /* Ignore the not relevant mask for the DAI/direction */ + if (dai->id == PCM3168A_DAI_DAC) + io_params->tdm_mask = tx_mask; + else + io_params->tdm_mask = rx_mask; return 0; } @@ -434,7 +431,8 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); - bool tx, master_mode; + struct pcm3168a_io_params *io_params = &pcm3168a->io_params[dai->id]; + bool master_mode; u32 val, mask, shift, reg; unsigned int rate, fmt, ratio, max_ratio; unsigned int tdm_slots; @@ -444,23 +442,21 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, ratio = pcm3168a->sysclk / rate; - tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - if (tx) { + if (dai->id == PCM3168A_DAI_DAC) { max_ratio = PCM3168A_NUM_SCKI_RATIOS_DAC; reg = PCM3168A_DAC_PWR_MST_FMT; mask = PCM3168A_DAC_MSDA_MASK; shift = PCM3168A_DAC_MSDA_SHIFT; - master_mode = pcm3168a->dac_master_mode; - fmt = pcm3168a->dac_fmt; } else { max_ratio = PCM3168A_NUM_SCKI_RATIOS_ADC; reg = PCM3168A_ADC_MST_FMT; mask = PCM3168A_ADC_MSAD_MASK; shift = PCM3168A_ADC_MSAD_SHIFT; - master_mode = pcm3168a->adc_master_mode; - fmt = pcm3168a->adc_fmt; } + master_mode = io_params->master_mode; + fmt = io_params->fmt; + for (i = 0; i < max_ratio; i++) { if (pcm3168a_scki_ratios[i] == ratio) break; @@ -471,8 +467,8 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (pcm3168a->slot_width) - slot_width = pcm3168a->slot_width; + if (io_params->slot_width) + slot_width = io_params->slot_width; else slot_width = params_width(params); @@ -497,8 +493,8 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (pcm3168a->tdm_slots) - tdm_slots = pcm3168a->tdm_slots; + if (io_params->tdm_slots) + tdm_slots = io_params->tdm_slots; else tdm_slots = params_channels(params); @@ -534,7 +530,7 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(pcm3168a->regmap, reg, mask, val); - if (tx) { + if (dai->id == PCM3168A_DAI_DAC) { mask = PCM3168A_DAC_FMT_MASK; shift = PCM3168A_DAC_FMT_SHIFT; } else { @@ -552,20 +548,13 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - unsigned int fmt; unsigned int sample_min; unsigned int channel_max; unsigned int channel_maxs[] = { - 6, /* rx */ - 8 /* tx */ + 8, /* DAC */ + 6 /* ADC */ }; - if (tx) - fmt = pcm3168a->dac_fmt; - else - fmt = pcm3168a->adc_fmt; - /* * Available Data Bits * @@ -578,7 +567,7 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, * I2S * LEFT_J */ - switch (fmt) { + switch (pcm3168a->io_params[dai->id].fmt) { case PCM3168A_FMT_RIGHT_J: sample_min = 16; channel_max = 2; @@ -588,7 +577,7 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, case PCM3168A_FMT_DSP_A: case PCM3168A_FMT_DSP_B: sample_min = 24; - channel_max = channel_maxs[tx]; + channel_max = channel_maxs[dai->id]; break; default: sample_min = 24; @@ -600,8 +589,8 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, sample_min, 32); /* Allow all channels in multi DIN/DOUT mode */ - if (pcm3168a->tdm_slots == 2) - channel_max = channel_maxs[tx]; + if (pcm3168a->io_params[dai->id].tdm_slots == 2) + channel_max = channel_maxs[dai->id]; snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, @@ -609,26 +598,19 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, return 0; } -static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = { +static const struct snd_soc_dai_ops pcm3168a_dai_ops = { .startup = pcm3168a_startup, - .set_fmt = pcm3168a_set_dai_fmt_dac, + .set_fmt = pcm3168a_set_dai_fmt, .set_sysclk = pcm3168a_set_dai_sysclk, .hw_params = pcm3168a_hw_params, .digital_mute = pcm3168a_digital_mute, .set_tdm_slot = pcm3168a_set_tdm_slot, }; -static const struct snd_soc_dai_ops pcm3168a_adc_dai_ops = { - .startup = pcm3168a_startup, - .set_fmt = pcm3168a_set_dai_fmt_adc, - .set_sysclk = pcm3168a_set_dai_sysclk, - .hw_params = pcm3168a_hw_params, - .set_tdm_slot = pcm3168a_set_tdm_slot, -}; - static struct snd_soc_dai_driver pcm3168a_dais[] = { { .name = "pcm3168a-dac", + .id = PCM3168A_DAI_DAC, .playback = { .stream_name = "Playback", .channels_min = 1, @@ -636,10 +618,11 @@ static struct snd_soc_dai_driver pcm3168a_dais[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = PCM3168A_FORMATS }, - .ops = &pcm3168a_dac_dai_ops + .ops = &pcm3168a_dai_ops }, { .name = "pcm3168a-adc", + .id = PCM3168A_DAI_ADC, .capture = { .stream_name = "Capture", .channels_min = 1, @@ -647,7 +630,7 @@ static struct snd_soc_dai_driver pcm3168a_dais[] = { .rates = SNDRV_PCM_RATE_8000_96000, .formats = PCM3168A_FORMATS }, - .ops = &pcm3168a_adc_dai_ops + .ops = &pcm3168a_dai_ops }, }; -- 2.20.1

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[alsa-devel] [PATCH 0/2] ASoC: pcm3168a: DAC/ADC independence and right_j fix
by Peter Ujfalusi 12 Aug '19

12 Aug '19

Hi, The first patch makes sure that the DAC and ADC side of the codec can be used and configured independently from each other. The second patch allows the reconfiguration of tdm_slots and slot_width. It is needed in case the codec is configured as slave and right_j whisch is the only mode when it can support 16 and 24 bit audio, but 16 bit requires 16 slot_width while 24 bit can use 48 or 32 slot_width. The codec driver must allow runtime reconfiguration of the tdm_slots and slot_width for this purpose. Regards, Peter --- Peter Ujfalusi (2): ASoC: pcm3168a: Retain the independence of DAC and ADC side of the codec ASoC: pcm3168a: Allow reconfiguration of tdm_slots and slot_width sound/soc/codecs/pcm3168a.c | 129 ++++++++++++++---------------------- 1 file changed, 50 insertions(+), 79 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] [PATCH] ASoC: ti: davinci-mcasp: Protect hw_params callback against race
by Peter Ujfalusi 12 Aug '19

12 Aug '19

If the playback and capture of the same McASP is connected to different dai link (non duplex PCM links, like with pcm3168a codec) then there is a high probability of race between the two direction leaving McASP in a confused state. Protect the hw_params() with a mutex to make sure that this is not happening. The concurrent execution of hw_params for capture and playback can be easily triggered with custom .asoundrc file (for pcm3168a codec): pcm.dmixed8 { type dmix ipc_key 2048 ipc_perm 0666 slave { pcm "hw:0,0" format S24_LE channels 8 rate 96000 } bindings { 0 0 1 1 2 2 3 3 4 4 5 5 6 6 7 7 } } pcm.cpb-headset-1 { type plug slave.pcm dmixed8 hint { show on description "Headset 1 jack" } ttable.0.0 1 ttable.1.4 1 } pcm.dsnooped6 { type dsnoop ipc_key 2049 ipc_perm 0666 slave { pcm "hw:0,1" format S24_LE channels 6 rate 96000 } bindings { 0 0 1 1 2 2 3 3 4 4 5 5 } } pcm.cpb-mic-1 { type plug slave.pcm "dsnooped6" hint { show on description "Microphone 1 jack" } ttable.0.0 1 ttable.1.3 1 } Then running: arecord -D cpb-mic-1 -f S24_LE -c2 -r48000 | aplay -D cpb-headset-1 -f S24_LE -c2 -r48000 Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/ti/davinci-mcasp.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 7aa3c32e4a49..fe7a0b3572e2 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -111,6 +111,7 @@ struct davinci_mcasp { u32 channels; int max_format_width; u8 active_serializers[2]; + struct mutex mutex; #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; @@ -1169,6 +1170,8 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int period_size = params_period_size(params); int ret; + mutex_lock(&mcasp->mutex); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: case SNDRV_PCM_FORMAT_S8: @@ -1197,12 +1200,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, default: printk(KERN_WARNING "davinci-mcasp: unsupported PCM format"); - return -EINVAL; + ret = -EINVAL; + goto out; } ret = davinci_mcasp_set_dai_fmt(cpu_dai, mcasp->dai_fmt); if (ret) - return ret; + goto out; /* * If mcasp is BCLK master, and a BCLK divider was not provided by @@ -1223,7 +1227,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, ret = mcasp_common_hw_param(mcasp, substream->stream, period_size * channels, channels); if (ret) - return ret; + goto out; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) ret = mcasp_dit_hw_param(mcasp, params_rate(params)); @@ -1232,7 +1236,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, channels); if (ret) - return ret; + goto out; davinci_config_channel_size(mcasp, word_length); @@ -1242,7 +1246,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, mcasp->max_format_width = word_length; } - return 0; +out: + mutex_unlock(&mcasp->mutex); + + return ret; } static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, @@ -2335,6 +2342,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (ret) return -EINVAL; + mutex_init(&mcasp->mutex); + ret = devm_snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, &davinci_mcasp_dai[pdata->op_mode], 1); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH v4 0/7] Allwinner H6 SPDIF support
by Clément Péron 12 Aug '19

12 Aug '19

*H6 DMA support IS REQUIRED* Allwinner H6 SoC has a SPDIF controller called One Wire Audio (OWA) which is different from the previous H3 generation and not compatible. Difference are an increase of fifo sizes, some memory mapping are different and there is now the possibility to output the master clock on a pin. Actually all these features are unused and only a bit for flushing the TX fifo is required. Also this series requires the DMA working on H6, a first version has been submitted by Jernej Škrabec[1] but has not been accepted yet. [1] https://patchwork.kernel.org/project/linux-arm-kernel/list/?series=89011 Changes since v3: - rename reg_fctl_ftx to val_fctl_ftx - rebase this series on sound-next - fix dt-bindings due to change in sound-next - change node name sound_spdif to sound-spdif Changes since v2: - Split quirks and H6 support patch - Add specific section for quirks comment Changes since v1: - Remove H3 compatible - Add TX fifo bit flush quirks - Add H6 bindings in SPDIF driver Clément Péron (7): dt-bindings: sound: sun4i-spdif: Add Allwinner H6 compatible ASoC: sun4i-spdif: Move quirks to the top ASoC: sun4i-spdif: Add TX fifo bit flush quirks ASoC: sun4i-spdif: Add support for H6 SoC arm64: dts: allwinner: Add SPDIF node for Allwinner H6 arm64: dts: allwinner: h6: Enable SPDIF for Beelink GS1 arm64: defconfig: Enable Sun4i SPDIF module .../sound/allwinner,sun4i-a10-spdif.yaml | 1 + .../dts/allwinner/sun50i-h6-beelink-gs1.dts | 4 ++ arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 38 ++++++++++++++ arch/arm64/configs/defconfig | 1 + sound/soc/sunxi/sun4i-spdif.c | 49 ++++++++++++++++--- 5 files changed, 87 insertions(+), 6 deletions(-) -- 2.20.1

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[alsa-devel] Is period_size bytes per period or frames per period?
by GitHub issues - edited 12 Aug '19

12 Aug '19

alsa-project/alsa-lib issue #8 was edited from crocket: According to https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html period_size in dmix is bytes per period. According to aplay --dump-hw-params -D dmix ... period_size is frames per period. Which one is right? Issue URL : https://github.com/alsa-project/alsa-lib/issues/8 Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] Is period_size bytes per period or frames per period?
by GitHub issues - opened 12 Aug '19

12 Aug '19

alsa-project/alsa-lib issue #8 was opened from crocket: According to https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html period_size in dmix is bytes per period. According to aplay --dump-hw-params period_size is frames per period. Which one is right? Issue URL : https://github.com/alsa-project/alsa-lib/issues/8 Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] [PATCH v4 0/9]ASoC: sun4i-i2s: Updates to the driver
by codekipper＠gmail.com 12 Aug '19

12 Aug '19

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, here is a patch series to add some improvements to the sun4i-i2s driver found whilst getting slave clocking and hdmi audio working on the newer SoCs. As the LibreELEC project is progressing extremely well then there has been some activity getting surround sound working and this is included. The functionality included with the new patch set has been extended to cover more sample resolutions, multi-lane data output for HDMI audio and some bug fixes that have been discovered along the way. I can see more usage of the tdm property since I last attempted to push these patches and the examples currently in mainline sort of the opposite to what I'm trying to achieve. When we first started looking at the i2s driver, the codecs that we were using allowed for the frame width to be determined based on the sampling resolution but in most use cases it seems that a fixed width is required(my highest priority should be to get HDMI audio support in). We're using the tdm property to override the old way to calculate the frame width. What I've seen in what has already been mainlined is that the i2s driver has a frame width that is fixed to 32 bits and this can be overridden using the tdm property. I still need to investigate the FIFO syncing issues which i've not had a chance to change or address the concerns that broonie and wens brought up. This change has been moved to the top of the patch stack. BR, CK --- v4 changes compared to v3 are: - Moved patches around so that the more controversial of patches are at the top of the stack. - Added more details to commit messages. - Fixed 20bit audio PCM format to use 4 bytes. - Reduced number of flags used to indicate a new SoC. v3 changes compared to v2 are: - added back slave mode changes - added back the use of tdm properties - changes to regmap and caching - removed loopback functionality - fixes to the channel offset mask v2 changes compared to v1 are: - removed slave mode changes which didn't set mclk and bclk div. - removed use of tdm and now use a dedicated property. - fix commit message to better explain reason for sign extending - add divider calculations for newer SoCs. - add support for multi-lane i2s data output. - add support for 20, 24 and 32 bit samples. - add loopback property so blocks can be tested without a codec. Marcus Cooper (9): ASoC: sun4i-i2s: Fix sun8i tx channel offset mask ASoC: sun4i-i2s: Add offset to RX channel select ASoC: sun4i-i2s: Add regmap field to sign extend sample ASoC: sun4i-i2s: Reduce quirks for sun8i-h3 ASoC: sun4i-i2s: Add set_tdm_slot functionality ASoC: sun4i-i2s: Add multi-lane functionality ASoC: sun4i-i2s: Add multichannel functionality ASoc: sun4i-i2s: Add 20, 24 and 32 bit support ASoC: sun4i-i2s: Adjust regmap settings sound/soc/sunxi/sun4i-i2s.c | 242 ++++++++++++++++++++++++------------ 1 file changed, 164 insertions(+), 78 deletions(-) -- 2.21.0

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[alsa-devel] [asoc:for-5.4 175/205] include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t'
by kbuild test robot 11 Aug '19

11 Aug '19

tree: https://kernel.googlesource.com/pub/scm/linux/kernel/git/broonie/sound.git for-5.4 head: 83ee7ab1627b7597bbb9acc26b37e9f96fff40c9 commit: abf31feea26c0f412a191c83f408311a0de7435c [175/205] ASoC: Intel: Update request-reply IPC model config: i386-randconfig-a004-201932 (attached as .config) compiler: gcc-4.9 (Debian 4.9.2-10+deb8u1) 4.9.2 reproduce: git checkout abf31feea26c0f412a191c83f408311a0de7435c # save the attached .config to linux build tree make ARCH=i386 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All warnings (new ones prefixed by >>): In file included from include/linux/printk.h:332:0, from include/linux/kernel.h:15, from sound/soc/intel/haswell/sst-haswell-ipc.c:9: sound/soc/intel/haswell/sst-haswell-ipc.c: In function 'sst_hsw_module_set_param': >> include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t' [-Wformat=] static struct _ddebug __aligned(8) \ ^ include/linux/dynamic_debug.h:123:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^ include/linux/dynamic_debug.h:143:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^ include/linux/dynamic_debug.h:157:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^ include/linux/device.h:1503:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^ sound/soc/intel/haswell/sst-haswell-ipc.c:2018:3: note: in expansion of macro 'dev_dbg' dev_dbg(dev, "transfer parameter size : %lu\n", ^ >> include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t' [-Wformat=] static struct _ddebug __aligned(8) \ ^ include/linux/dynamic_debug.h:123:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^ include/linux/dynamic_debug.h:143:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^ include/linux/dynamic_debug.h:157:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^ include/linux/device.h:1503:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^ sound/soc/intel/haswell/sst-haswell-ipc.c:2022:3: note: in expansion of macro 'dev_dbg' dev_dbg(dev, "transfer parameter aligned size : %lu\n", ^ -- In file included from include/linux/printk.h:332:0, from include/linux/kernel.h:15, from sound/soc//intel/haswell/sst-haswell-ipc.c:9: sound/soc//intel/haswell/sst-haswell-ipc.c: In function 'sst_hsw_module_set_param': >> include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t' [-Wformat=] static struct _ddebug __aligned(8) \ ^ include/linux/dynamic_debug.h:123:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^ include/linux/dynamic_debug.h:143:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^ include/linux/dynamic_debug.h:157:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^ include/linux/device.h:1503:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^ sound/soc//intel/haswell/sst-haswell-ipc.c:2018:3: note: in expansion of macro 'dev_dbg' dev_dbg(dev, "transfer parameter size : %lu\n", ^ >> include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t' [-Wformat=] static struct _ddebug __aligned(8) \ ^ include/linux/dynamic_debug.h:123:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^ include/linux/dynamic_debug.h:143:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^ include/linux/dynamic_debug.h:157:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^ include/linux/device.h:1503:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^ sound/soc//intel/haswell/sst-haswell-ipc.c:2022:3: note: in expansion of macro 'dev_dbg' dev_dbg(dev, "transfer parameter aligned size : %lu\n", ^ vim +82 include/linux/dynamic_debug.h 923abb9d797ba0 Gal Pressman 2019-05-01 75 923abb9d797ba0 Gal Pressman 2019-05-01 76 extern __printf(3, 4) 923abb9d797ba0 Gal Pressman 2019-05-01 77 void __dynamic_ibdev_dbg(struct _ddebug *descriptor, 923abb9d797ba0 Gal Pressman 2019-05-01 78 const struct ib_device *ibdev, 923abb9d797ba0 Gal Pressman 2019-05-01 79 const char *fmt, ...); 923abb9d797ba0 Gal Pressman 2019-05-01 80 2bdde670beedf7 Rasmus Villemoes 2019-03-07 81 #define DEFINE_DYNAMIC_DEBUG_METADATA(name, fmt) \ c0d2af63786394 Joe Perches 2012-10-18 @82 static struct _ddebug __aligned(8) \ 07613b0b5ef857 Jason Baron 2011-10-04 83 __attribute__((section("__verbose"))) name = { \ 07613b0b5ef857 Jason Baron 2011-10-04 84 .modname = KBUILD_MODNAME, \ 07613b0b5ef857 Jason Baron 2011-10-04 85 .function = __func__, \ 07613b0b5ef857 Jason Baron 2011-10-04 86 .filename = __FILE__, \ 07613b0b5ef857 Jason Baron 2011-10-04 87 .format = (fmt), \ 07613b0b5ef857 Jason Baron 2011-10-04 88 .lineno = __LINE__, \ 07613b0b5ef857 Jason Baron 2011-10-04 89 .flags = _DPRINTK_FLAGS_DEFAULT, \ 2bdde670beedf7 Rasmus Villemoes 2019-03-07 90 _DPRINTK_KEY_INIT \ 07613b0b5ef857 Jason Baron 2011-10-04 91 } 07613b0b5ef857 Jason Baron 2011-10-04 92 :::::: The code at line 82 was first introduced by commit :::::: c0d2af637863940b1a4fb208224ca7acb905c39f dynamic_debug: Remove unnecessary __used :::::: TO: Joe Perches <joe(a)perches.com> :::::: CC: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] snd_pcm_start question
by Eric Desjardins 11 Aug '19

11 Aug '19

Hi, This is a simple question. I call: snd_pcm_prepare snd_pcm_writei and it works fine but I would like to delay playback start. There is snd_pcm_start for that. There is a method that is deprecated to specify a start mode but I was wondering what was the correct new way to do that? Thanks, Eric

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[alsa-devel] bcm2835.conf for RPi v. 4 incl. HDMI
by Pavel Hofman 10 Aug '19

10 Aug '19

Hi, may I ask for help with cards/bcm2835.conf for the new RPi? No such config exists yet: pi@raspberrypi:~ $ strace -f aplay -L 2>&1 | grep -i bcm2835 access("/usr/share/alsa/cards/bcm2835_alsa.conf", R_OK) = -1 ENOENT (No such file or directory) I understand the _alsa suffix is handled in aliases (e.g. https://www.raspberrypi.org/forums/viewtopic.php?t=221734#p1361104 ) The reason I am looking for this file are Pulseaudio profiles. PA requires front:X, surroundX, hdmi:X, etc. PCM devices, yet non of these are created due to the missing card config: pi@raspberrypi:~ $ aplay -L default Playback/recording through the PulseAudio sound server null Discard all samples (playback) or generate zero samples (capture) jack JACK Audio Connection Kit pulse PulseAudio Sound Server sysdefault:CARD=ALSA bcm2835 ALSA, bcm2835 ALSA Default Audio Device dmix:CARD=ALSA,DEV=0 bcm2835 ALSA, bcm2835 ALSA Direct sample mixing device dmix:CARD=ALSA,DEV=1 bcm2835 ALSA, bcm2835 IEC958/HDMI Direct sample mixing device dsnoop:CARD=ALSA,DEV=0 bcm2835 ALSA, bcm2835 ALSA Direct sample snooping device dsnoop:CARD=ALSA,DEV=1 bcm2835 ALSA, bcm2835 IEC958/HDMI Direct sample snooping device hw:CARD=ALSA,DEV=0 bcm2835 ALSA, bcm2835 ALSA Direct hardware device without any conversions hw:CARD=ALSA,DEV=1 bcm2835 ALSA, bcm2835 IEC958/HDMI Direct hardware device without any conversions plughw:CARD=ALSA,DEV=0 bcm2835 ALSA, bcm2835 ALSA Hardware device with all software conversions plughw:CARD=ALSA,DEV=1 bcm2835 ALSA, bcm2835 IEC958/HDMI Hardware device with all software conversions usbstream:CARD=ALSA bcm2835 ALSA USB Stream Output As a result, only the Analog Mono PA profile (based on the only existing hw:0) is available in PA: D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Mono (analog-mono) D: [pulseaudio] alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... D: [pulseaudio] alsa-util.c: Managed to open hw:0 ... D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Stereo (analog-stereo) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... I: [pulseaudio] (alsa-lib)conf.c: Unknown parameters 0 I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM front:0 --- D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo) D: [pulseaudio] alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ... I: [pulseaudio] (alsa-lib)conf.c: Unknown parameters 0 I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM hdmi:0 ... D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 7.1 (HDMI) (hdmi-surround71) D: [pulseaudio] alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ... I: [pulseaudio] (alsa-lib)conf.c: Unknown parameters 0 I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM hdmi:0 RPi v.4 has 2 HDMI outputs: pi@raspberrypi:~ $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: ALSA [bcm2835 ALSA], device 0: bcm2835 ALSA [bcm2835 ALSA] Subdevices: 6/7 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 card 0: ALSA [bcm2835 ALSA], device 1: bcm2835 IEC958/HDMI [bcm2835 IEC958/HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: ALSA [bcm2835 ALSA], device 2: bcm2835 IEC958/HDMI1 [bcm2835 IEC958/HDMI1] Subdevices: 1/1 Subdevice #0: subdevice #0 Each is 2 - 8 channel: pi@raspberrypi:~ $ aplay -v --dump-hw-params -D hw:0,1 /dev/zero Playing raw data '/dev/zero' : Unsigned 8 bit, Rate 8000 Hz, Mono HW Params of device "hw:0,1": -------------------- ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: [32 128] CHANNELS: [2 8] RATE: [44100 192000] PERIOD_TIME: [10000 743039) PERIOD_SIZE: [441 32768] PERIOD_BYTES: [1776 524288] PERIODS: [1 75) BUFFER_TIME: (2296 743039) BUFFER_SIZE: [441 32768] BUFFER_BYTES: [1764 131072] TICK_TIME: ALL I looked at hdmi section of cards/Intel-HDA.conf but that one is very complicated. May I ask for help how the config should look like to work both for multichannel PCM and digital formats (i.e. support for the AES vars)? I very much appreciate any help. Best regards, Pavel.

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[alsa-devel] Unable to get sound out of Laptop
by Nilesh Kumar 10 Aug '19

10 Aug '19

Hello all, I am unable to get sound out of my "HP Notebook - 15-bs180tx" laptop, the output of alsa-info,sh is been uploaded at http://www.alsa-project.org/db/?f=5b015c625cef0cc14b0c0551cb57bc1e35d4b40a so, I would really appreciate if someone could see the log and help me out in configuring my sound card -- Nilesh Kumar

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[alsa-devel] Writing /usr/share/alsa/cards/ files
by Paul Pawlowski 10 Aug '19

10 Aug '19

Hello, Is there any documentation on writing /usr/share/alsa/cards/ files? I painfully tried to create one in order to enable 2 separate audio outputs, but I ultimately failed to due to the lack of any sort of error reporting. Is there a way to enable any error reporting when parsing these files? It seems that the error output when parsing any hooks goes to nowhere. Also, is this the correct mailing list to get help regarding alsa driver development and configuration? Thank you, Paul Pawlowski

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[alsa-devel] [PATCH] ALSA: hda - Fix a memory leak bug
by Wenwen Wang 10 Aug '19

10 Aug '19

In snd_hda_parse_generic_codec(), 'spec' is allocated through kzalloc(). Then, the pin widgets in 'codec' are parsed. However, if the parsing process fails, 'spec' is not deallocated, leading to a memory leak. To fix the above issue, free 'spec' before returning the error. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 485edab..8f2beb1 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -6100,7 +6100,7 @@ static int snd_hda_parse_generic_codec(struct hda_codec *codec) err = snd_hda_parse_pin_defcfg(codec, &spec->autocfg, NULL, 0); if (err < 0) - return err; + goto error; err = snd_hda_gen_parse_auto_config(codec, &spec->autocfg); if (err < 0) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: SOF: Intel: Add missing include file hdac_hda.h
by YueHaibing 09 Aug '19

09 Aug '19

Building with SND_SOC_SOF_HDA_AUDIO_CODEC fails: sound/soc/sof/intel/hda-bus.c: In function sof_hda_bus_init: sound/soc/sof/intel/hda-bus.c:16:25: error: implicit declaration of function snd_soc_hdac_hda_get_ops; did you mean snd_soc_jack_add_gpiods? [-Werror=implicit-function-declaration] #define sof_hda_ext_ops snd_soc_hdac_hda_get_ops() Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: d4ff1b3917a5 ('ASoC: SOF: Intel: Initialize hdaudio bus properly") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sof/intel/hda-bus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c index 0caec3a..e28a9c4 100644 --- a/sound/soc/sof/intel/hda-bus.c +++ b/sound/soc/sof/intel/hda-bus.c @@ -13,6 +13,7 @@ #include "hda.h" #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) +#include "../../../soc/codecs/hdac_hda.h" #define sof_hda_ext_ops snd_soc_hdac_hda_get_ops() #else #define sof_hda_ext_ops NULL -- 2.7.4

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[alsa-devel] [PATCH] ASoC: hdac_hdmi: Offload dapm update at jack detection
by Takashi Iwai 09 Aug '19

09 Aug '19

hdac_hdmi_present_sense() calls the audio component to get ELD update, then it reports the jack status change and updates DAPM graph accordingly. This works when it's called from the normal code paths. However, it may lead to a dead lock when it's called from the audio component notifier. Namely, the DAPM update involves with the runtime PM, and it eventually calls again the audio component get_power() ops. Since i915 driver already takes a mutex around the audio component ops calls, we'll eventually get the mutex doubly. As a workaround, in this patch, only the jack state is updated in the code path from hdac_hdmi_eld_notify_cb(), and the DAPM update is deferred to a work so that it's processed in another context. Reported-by: Imre Deak <imre.deak(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is a bug fix I did in the last year when we received some bug report, but forgotten in my archive. I believe the fix is still valid although the bug may trigger rarely. sound/soc/codecs/hdac_hdmi.c | 57 +++++++++++++++++++++++++++++++++++++------- 1 file changed, 48 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 29918954e740..18c173e6a13b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -88,8 +88,10 @@ struct hdac_hdmi_port { hda_nid_t mux_nids[HDA_MAX_CONNECTIONS]; struct hdac_hdmi_eld eld; const char *jack_pin; + bool is_connect; struct snd_soc_dapm_context *dapm; const char *output_pin; + struct work_struct dapm_work; }; struct hdac_hdmi_pcm { @@ -163,11 +165,7 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, { struct hdac_device *hdev = port->pin->hdev; - if (is_connect) - snd_soc_dapm_enable_pin(port->dapm, port->jack_pin); - else - snd_soc_dapm_disable_pin(port->dapm, port->jack_pin); - + port->is_connect = is_connect; if (is_connect) { /* * Report Jack connect event when a device is connected @@ -193,10 +191,32 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, if (pcm->jack_event > 0) pcm->jack_event--; } +} +static void hdac_hdmi_port_dapm_update(struct hdac_hdmi_port *port) +{ + if (port->is_connect) + snd_soc_dapm_enable_pin(port->dapm, port->jack_pin); + else + snd_soc_dapm_disable_pin(port->dapm, port->jack_pin); snd_soc_dapm_sync(port->dapm); } +static void hdac_hdmi_jack_dapm_work(struct work_struct *work) +{ + struct hdac_hdmi_port *port; + + port = container_of(work, struct hdac_hdmi_port, dapm_work); + hdac_hdmi_port_dapm_update(port); +} + +static void hdac_hdmi_jack_report_sync(struct hdac_hdmi_pcm *pcm, + struct hdac_hdmi_port *port, bool is_connect) +{ + hdac_hdmi_jack_report(pcm, port, is_connect); + hdac_hdmi_port_dapm_update(port); +} + /* MST supported verbs */ /* * Get the no devices that can be connected to a port on the Pin widget. @@ -904,7 +924,7 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, list_for_each_entry_safe(p, p_next, &pcm->port_list, head) { if (p == port && p->id == port->id && p->pin == port->pin) { - hdac_hdmi_jack_report(pcm, port, false); + hdac_hdmi_jack_report_sync(pcm, port, false); list_del(&p->head); } } @@ -918,7 +938,7 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, if (!strcmp(cvt_name, pcm->cvt->name)) { list_add_tail(&port->head, &pcm->port_list); if (port->eld.monitor_present && port->eld.eld_valid) { - hdac_hdmi_jack_report(pcm, port, true); + hdac_hdmi_jack_report_sync(pcm, port, true); mutex_unlock(&hdmi->pin_mutex); return ret; } @@ -1281,16 +1301,20 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, * report jack here. It will be done in usermode mux * control select. */ - if (pcm) + if (pcm) { hdac_hdmi_jack_report(pcm, port, false); + schedule_work(&port->dapm_work); + } mutex_unlock(&hdmi->pin_mutex); return; } if (port->eld.monitor_present && port->eld.eld_valid) { - if (pcm) + if (pcm) { hdac_hdmi_jack_report(pcm, port, true); + schedule_work(&port->dapm_work); + } print_hex_dump_debug("ELD: ", DUMP_PREFIX_OFFSET, 16, 1, port->eld.eld_buffer, port->eld.eld_size, false); @@ -1319,6 +1343,7 @@ static int hdac_hdmi_add_ports(struct hdac_device *hdev, for (i = 0; i < max_ports; i++) { ports[i].id = i; ports[i].pin = pin; + INIT_WORK(&ports[i].dapm_work, hdac_hdmi_jack_dapm_work); } pin->ports = ports; pin->num_ports = max_ports; @@ -2083,8 +2108,20 @@ static int hdac_hdmi_dev_probe(struct hdac_device *hdev) return ret; } +static void clear_dapm_works(struct hdac_device *hdev) +{ + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); + struct hdac_hdmi_pin *pin; + int i; + + list_for_each_entry(pin, &hdmi->pin_list, head) + for (i = 0; i < pin->num_ports; i++) + cancel_work_sync(&pin->ports[i].dapm_work); +} + static int hdac_hdmi_dev_remove(struct hdac_device *hdev) { + clear_dapm_works(hdev); snd_hdac_display_power(hdev->bus, hdev->addr, false); return 0; @@ -2103,6 +2140,8 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) if (!bus) return 0; + clear_dapm_works(hdev); + /* * Power down afg. * codec_read is preferred over codec_write to set the power state. -- 2.16.4

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[alsa-devel] [PATCH] ALSA: hda - Add a quirk model for fixing Huawei Matebook X right speaker
by Takashi Iwai 09 Aug '19

09 Aug '19

From: Tomas Espeleta <tomas.espeleta(a)gmail.com> [ This is rather a revival of the patch Tomas sent in months ago, but applying only with the quirk model option -- tiwai ] Hard coded coefficients to make Huawuei Matebook X right speaker work. The Matebook X has a ALC298, please refer to bug 197801 on how these numbers were reverse engineered from the Windows driver The reversed engineered sequence represents a repeating pattern of verbs, and the only values that are changing periodically are written on indexes 0x23 and 0x25: 0x500, 0x23 0x400, VALUE1 0x500, 0x25 0x400, VALUE2 * skipped reading sequences (0x500 - 0xc00 sequences are ignored) * static values from reverse engineering are used NOTE: since a significant risk is still considered, this is provided as an experimental fix that isn't applied as default for now. For enabling the fix, you'll have to choose huawei-mbx-stereo via model option of snd-hda-intel module. If we get feedback from users that this works stably, we may apply it per default. [ Some coding style fixes and replacement with AC_VERB_* by tiwai ] BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=197801 Signed-off-by: Tomas Espeleta <tomas.espeleta(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/hd-audio/models.rst | 3 ++ sound/pci/hda/patch_realtek.c | 74 +++++++++++++++++++++++++++++++++ 2 files changed, 77 insertions(+) diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index 7d7c191102a7..11298f0ce44d 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -260,6 +260,9 @@ alc295-hp-x360 HP Spectre X360 fixups alc-sense-combo Headset button support for Chrome platform +huawei-mbx-stereo + Enable initialization verbs for Huawei MBX stereo speakers; + might be risky, try this at your own risk ALC66x/67x/892 ============== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index de224cbea7a0..cb7baa65b298 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3762,6 +3762,72 @@ static void alc269_x101_hp_automute_hook(struct hda_codec *codec, vref); } +/* + * Magic sequence to make Huawei Matebook X right speaker working (bko#197801) + */ +struct hda_alc298_mbxinit { + unsigned char value_0x23; + unsigned char value_0x25; +}; + +static void alc298_huawei_mbx_stereo_seq(struct hda_codec *codec, + const struct hda_alc298_mbxinit *initval, + bool first) +{ + snd_hda_codec_write(codec, 0x06, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x0); + alc_write_coef_idx(codec, 0x26, 0xb000); + + if (first) + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_GET_PIN_SENSE, 0x0); + + snd_hda_codec_write(codec, 0x6, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x80); + alc_write_coef_idx(codec, 0x26, 0xf000); + alc_write_coef_idx(codec, 0x23, initval->value_0x23); + + if (initval->value_0x23 != 0x1e) + alc_write_coef_idx(codec, 0x25, initval->value_0x25); + + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x26); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, 0xb010); +} + +static void alc298_fixup_huawei_mbx_stereo(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* Initialization magic */ + static const struct hda_alc298_mbxinit dac_init[] = { + {0x0c, 0x00}, {0x0d, 0x00}, {0x0e, 0x00}, {0x0f, 0x00}, + {0x10, 0x00}, {0x1a, 0x40}, {0x1b, 0x82}, {0x1c, 0x00}, + {0x1d, 0x00}, {0x1e, 0x00}, {0x1f, 0x00}, + {0x20, 0xc2}, {0x21, 0xc8}, {0x22, 0x26}, {0x23, 0x24}, + {0x27, 0xff}, {0x28, 0xff}, {0x29, 0xff}, {0x2a, 0x8f}, + {0x2b, 0x02}, {0x2c, 0x48}, {0x2d, 0x34}, {0x2e, 0x00}, + {0x2f, 0x00}, + {0x30, 0x00}, {0x31, 0x00}, {0x32, 0x00}, {0x33, 0x00}, + {0x34, 0x00}, {0x35, 0x01}, {0x36, 0x93}, {0x37, 0x0c}, + {0x38, 0x00}, {0x39, 0x00}, {0x3a, 0xf8}, {0x38, 0x80}, + {} + }; + const struct hda_alc298_mbxinit *seq; + + if (action != HDA_FIXUP_ACT_INIT) + return; + + /* Start */ + snd_hda_codec_write(codec, 0x06, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x00); + snd_hda_codec_write(codec, 0x06, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x80); + alc_write_coef_idx(codec, 0x26, 0xf000); + alc_write_coef_idx(codec, 0x22, 0x31); + alc_write_coef_idx(codec, 0x23, 0x0b); + alc_write_coef_idx(codec, 0x25, 0x00); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x26); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, 0xb010); + + for (seq = dac_init; seq->value_0x23; seq++) + alc298_huawei_mbx_stereo_seq(codec, seq, seq == dac_init); +} + static void alc269_fixup_x101_headset_mic(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -5787,6 +5853,7 @@ enum { ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, ALC256_FIXUP_HUAWEI_MACH_WX9_PINS, + ALC298_FIXUP_HUAWEI_MBX_STEREO, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, @@ -6095,6 +6162,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_MIC_MUTE_LED }, + [ALC298_FIXUP_HUAWEI_MBX_STEREO] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_huawei_mbx_stereo, + .chained = true, + .chain_id = ALC255_FIXUP_MIC_MUTE_LED + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -7272,6 +7345,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC225_FIXUP_HEADSET_JACK, .name = "alc-headset-jack"}, {.id = ALC295_FIXUP_CHROME_BOOK, .name = "alc-chrome-book"}, {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"}, + {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.16.4

1 0

[alsa-devel] [PATCH] ALSA: hda - Inform too slow responses
by Takashi Iwai 09 Aug '19

09 Aug '19

For the sake of debugging, show each response that took too long time and its corresponding last verb. Since such an error happens in series once when started, use the ratelimited variant for suppressing the flood. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 462559442f7a..6387c7e90918 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -794,6 +794,7 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, unsigned long timeout; unsigned long loopcounter; int do_poll = 0; + bool warned = false; again: timeout = jiffies + msecs_to_jiffies(1000); @@ -813,9 +814,17 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, spin_unlock_irq(&bus->reg_lock); if (time_after(jiffies, timeout)) break; - if (hbus->needs_damn_long_delay || loopcounter > 3000) +#define LOOP_COUNT_MAX 3000 + if (hbus->needs_damn_long_delay || + loopcounter > LOOP_COUNT_MAX) { + if (loopcounter > LOOP_COUNT_MAX && !warned) { + dev_dbg_ratelimited(chip->card->dev, + "too slow response, last cmd=%#08x\n", + bus->last_cmd[addr]); + warned = true; + } msleep(2); /* temporary workaround */ - else { + } else { udelay(10); cond_resched(); } -- 2.16.4

1 0

[alsa-devel] [PATCH -next] ASoC: mt6351: remove unused variable 'mt_lineout_control'
by YueHaibing 09 Aug '19

09 Aug '19

sound/soc/codecs/mt6351.c:1070:38: warning: mt_lineout_control defined but not used [-Wunused-const-variable=] It is never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/mt6351.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/mt6351.c b/sound/soc/codecs/mt6351.c index 4b3ce01..5c0536e 100644 --- a/sound/soc/codecs/mt6351.c +++ b/sound/soc/codecs/mt6351.c @@ -1066,11 +1066,6 @@ static int mt_mic_bias_2_event(struct snd_soc_dapm_widget *w, return 0; } -/* DAPM Kcontrols */ -static const struct snd_kcontrol_new mt_lineout_control = - SOC_DAPM_SINGLE("Switch", MT6351_AUDDEC_ANA_CON3, - RG_AUDLOLPWRUP_VAUDP32_BIT, 1, 0); - /* DAPM Widgets */ static const struct snd_soc_dapm_widget mt6351_dapm_widgets[] = { /* Digital Clock */ -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: ml26124: remove unused variable 'ngth'
by YueHaibing 09 Aug '19

09 Aug '19

In file included from ./include/sound/tlv.h:10:0, from sound/soc/codecs/ml26124.c:19: sound/soc/codecs/ml26124.c:59:35: warning: ngth defined but not used [-Wunused-const-variable=] static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); ^ ./include/uapi/sound/tlv.h:64:15: note: in definition of macro SNDRV_CTL_TLVD_DECLARE_DB_SCALE unsigned int name[] = { \ ^~~~ sound/soc/codecs/ml26124.c:59:14: note: in expansion of macro DECLARE_TLV_DB_SCALE static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); ^~~~~~~~~~~~~~~~~~~~ It is never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/ml26124.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 3abd278..55823bc 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -56,7 +56,6 @@ static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0); static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0); static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0); static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0); -static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); static const char * const ml26124_companding[] = {"16bit PCM", "u-law", "A-law"}; -- 2.7.4

2 1

[alsa-devel] [PATCH 0/6] ASoC: SOF: refactor duplicated code between Intel and i.MX8
by Pierre-Louis Bossart 09 Aug '19

09 Aug '19

During the initial i.MX patch reviews on GitHub, we realized a lot of the i.MX mailbox/window code was common with the Intel loader code except for a couple of offsets. Most of the credit goes to Daniel Baluta for working on this series which adds a better abstraction and reduces code duplication. Comments from Keyon and Ranjani helped remove unnecessary parts kept for historical reasons and improve the code overall. Daniel Baluta (6): ASoC: SOF: Remove call to snd_sof_dsp_mailbox_init ASoC: SOF: Introduce snd_sof_dsp_get_mailbox_offset ASoC: SOF: Introduce snd_sof_dsp_get_window_offset ASoC: SOF: Intel: byt: Refactor fw ready / mem windows creation ASoC: SOF: Intel: bdw: Use generic function for fw ready / mem windows creation ASoC: SOF: Intel: hda: Use generic function for fw ready / mem windows creation sound/soc/sof/intel/apl.c | 4 +- sound/soc/sof/intel/bdw.c | 161 +++----------------------------- sound/soc/sof/intel/byt.c | 169 ++++------------------------------ sound/soc/sof/intel/cnl.c | 4 +- sound/soc/sof/intel/hda-ipc.c | 150 +----------------------------- sound/soc/sof/intel/hda.h | 4 +- sound/soc/sof/loader.c | 168 +++++++++++++++++++++++++++++++++ sound/soc/sof/ops.h | 18 ++++ sound/soc/sof/sof-priv.h | 6 ++ 9 files changed, 237 insertions(+), 447 deletions(-) -- 2.20.1

3 14

[alsa-devel] [PATCH 0/3] ALSA: hda: bus cleanup
by Takashi Iwai 09 Aug '19

09 Aug '19

Hi, this is a few patches to simplify and cleanup the HD-audio bus ops. The first two patches translate the indirect calls of DMA page allocation and MMIO accesses with the direct ones, as well as eliminating the whole bus->io_ops. The last one is SOF-specific, and fixes/cleans up by calling the proper hdaudio bus init function, as formerly discussed. Takashi === Takashi Iwai (3): ALSA: hda: Remove page allocation redirection ALSA: hda: Direct MMIO accesses ASoC: SOF: Intel: Initialize hdaudio bus properly include/sound/hdaudio.h | 69 +++++++++++++-------------- include/sound/hdaudio_ext.h | 1 - sound/hda/Kconfig | 3 ++ sound/hda/ext/hdac_ext_bus.c | 60 +----------------------- sound/hda/hdac_bus.c | 36 ++++++++++++-- sound/hda/hdac_controller.c | 18 +++---- sound/hda/hdac_stream.c | 8 ++-- sound/pci/hda/Kconfig | 1 + sound/pci/hda/hda_controller.c | 6 +-- sound/pci/hda/hda_controller.h | 3 +- sound/pci/hda/hda_intel.c | 71 ++-------------------------- sound/pci/hda/hda_tegra.c | 84 +-------------------------------- sound/soc/intel/skylake/skl-messages.c | 15 +----- sound/soc/intel/skylake/skl.c | 7 ++- sound/soc/sof/intel/hda-bus.c | 85 ++++------------------------------ sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sof/intel/hda.c | 6 +-- sound/soc/sof/intel/hda.h | 3 +- 18 files changed, 107 insertions(+), 371 deletions(-) -- 2.16.4

3 12

[alsa-devel] [PATCH -next] ALSA: Au88x0 - remove some unused const variables
by YueHaibing 09 Aug '19

09 Aug '19

sound/pci/au88x0/au88x0_xtalk.c:121:28: warning: asXtalkWideCoefsRightXt defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:152:28: warning: asXtalkNarrowCoefsRightXt defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:175:28: warning: asXtalkCoefsNegPipe defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:183:28: warning: asXtalkCoefsNumTest defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:191:28: warning: asXtalkCoefsDenTest defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:199:28: warning: asXtalkOutStateTest defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:20:20: warning: sXtalkWideKRightXt defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:231:28: warning: asDiamondCoefsRightXt defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:24:20: warning: sXtalkWideShiftRightXt defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:30:20: warning: sXtalkNarrowKRightXt defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:34:20: warning: sXtalkNarrowShiftRightXt defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:38:28: warning: asXtalkGainsDefault defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:43:28: warning: asXtalkGainsTest defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:48:28: warning: asXtalkGains1Chan defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:67:28: warning: alXtalkDlineTest defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:77:30: warning: asXtalkInStateTest defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:92:20: warning: sDiamondKRightXt defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:96:20: warning: sDiamondShiftRightXt defined but not used [-Wunused-const-variable=] They are never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/pci/au88x0/au88x0_xtalk.c | 85 ----------------------------------------- 1 file changed, 85 deletions(-) diff --git a/sound/pci/au88x0/au88x0_xtalk.c b/sound/pci/au88x0/au88x0_xtalk.c index 084fcbf..143646c 100644 --- a/sound/pci/au88x0/au88x0_xtalk.c +++ b/sound/pci/au88x0/au88x0_xtalk.c @@ -17,39 +17,20 @@ static short const sXtalkWideKLeftEq = 0x269C; static short const sXtalkWideKRightEq = 0x269C; static short const sXtalkWideKLeftXt = 0xF25E; -static short const sXtalkWideKRightXt = 0xF25E; static short const sXtalkWideShiftLeftEq = 1; static short const sXtalkWideShiftRightEq = 1; static short const sXtalkWideShiftLeftXt = 0; -static short const sXtalkWideShiftRightXt = 0; static unsigned short const wXtalkWideLeftDelay = 0xd; static unsigned short const wXtalkWideRightDelay = 0xd; static short const sXtalkNarrowKLeftEq = 0x468D; static short const sXtalkNarrowKRightEq = 0x468D; static short const sXtalkNarrowKLeftXt = 0xF82E; -static short const sXtalkNarrowKRightXt = 0xF82E; static short const sXtalkNarrowShiftLeftEq = 0x3; static short const sXtalkNarrowShiftRightEq = 0x3; static short const sXtalkNarrowShiftLeftXt = 0; -static short const sXtalkNarrowShiftRightXt = 0; static unsigned short const wXtalkNarrowLeftDelay = 0x7; static unsigned short const wXtalkNarrowRightDelay = 0x7; -static xtalk_gains_t const asXtalkGainsDefault = { - 0x4000, 0x4000, 0x4000, 0x4000, 0x4000, - 0x4000, 0x4000, 0x4000, 0x4000, 0x4000 -}; - -static xtalk_gains_t const asXtalkGainsTest = { - 0x7fff, 0x8000, 0x0000, 0x0000, 0x0001, - 0xffff, 0x4000, 0xc000, 0x0002, 0xfffe -}; - -static xtalk_gains_t const asXtalkGains1Chan = { - 0x7FFF, 0, 0, 0, 0, - 0x7FFF, 0, 0, 0, 0, -}; - // Input gain for 4 A3D slices. One possible input pair is left zero. static xtalk_gains_t const asXtalkGainsAllChan = { 0x7FFF, 0x7FFF, 0x7FFF, 0x7FFF, 0, @@ -64,20 +45,11 @@ static xtalk_dline_t const alXtalkDlineZeros = { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }; -static xtalk_dline_t const alXtalkDlineTest = { - 0x0000fc18, 0xfff03e8, 0x000186a0, 0xfffe7960, 1, 0xffffffff, 0, 0, - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0, 0, 0, 0, 0, 0, 0 -}; static xtalk_instate_t const asXtalkInStateZeros = { 0, 0, 0, 0 }; -static xtalk_instate_t const asXtalkInStateTest = { - 0x0080, 0xff80, 0x0001, 0xffff -}; - static xtalk_state_t const asXtalkOutStateZeros = { {0, 0, 0, 0}, {0, 0, 0, 0}, @@ -89,11 +61,9 @@ static xtalk_state_t const asXtalkOutStateZeros = { static short const sDiamondKLeftEq = 0x401d; static short const sDiamondKRightEq = 0x401d; static short const sDiamondKLeftXt = 0xF90E; -static short const sDiamondKRightXt = 0xF90E; static short const sDiamondShiftLeftEq = 1; static short const sDiamondShiftRightEq = 1; static short const sDiamondShiftLeftXt = 0; -static short const sDiamondShiftRightXt = 0; static unsigned short const wDiamondLeftDelay = 0xb; static unsigned short const wDiamondRightDelay = 0xb; @@ -118,13 +88,6 @@ static xtalk_coefs_t const asXtalkWideCoefsLeftXt = { {0x77dc, 0xc79e, 0xffb8, 0x000a, 0}, {0, 0, 0, 0, 0} }; -static xtalk_coefs_t const asXtalkWideCoefsRightXt = { - {0x55c6, 0xc97b, 0x005b, 0x0047, 0}, - {0x6a60, 0xca20, 0xffc6, 0x0040, 0}, - {0x6411, 0xd711, 0xfca1, 0x0190, 0}, - {0x77dc, 0xc79e, 0xffb8, 0x000a, 0}, - {0, 0, 0, 0, 0} -}; static xtalk_coefs_t const asXtalkNarrowCoefsLeftEq = { {0x50B5, 0xD07C, 0x026D, 0xFD21, 0}, {0x460F, 0xE44F, 0xF75E, 0xEFA6, 0}, @@ -149,14 +112,6 @@ static xtalk_coefs_t const asXtalkNarrowCoefsLeftXt = { {0, 0, 0, 0, 0} }; -static xtalk_coefs_t const asXtalkNarrowCoefsRightXt = { - {0x3CB2, 0xDF49, 0xF6EA, 0x095B, 0}, - {0x6777, 0xC915, 0xFEAF, 0x00B1, 0}, - {0x7762, 0xC7D9, 0x025B, 0xFDA6, 0}, - {0x6B7A, 0xD2AA, 0xF2FB, 0x0B64, 0}, - {0, 0, 0, 0, 0} -}; - static xtalk_coefs_t const asXtalkCoefsZeros = { {0, 0, 0, 0, 0}, {0, 0, 0, 0, 0}, @@ -172,37 +127,6 @@ static xtalk_coefs_t const asXtalkCoefsPipe = { {0, 0, 0x0FA0, 0, 0}, {0, 0, 0x1180, 0, 0}, }; -static xtalk_coefs_t const asXtalkCoefsNegPipe = { - {0, 0, 0xF380, 0, 0}, - {0, 0, 0xF380, 0, 0}, - {0, 0, 0xF380, 0, 0}, - {0, 0, 0xF380, 0, 0}, - {0, 0, 0xF200, 0, 0} -}; - -static xtalk_coefs_t const asXtalkCoefsNumTest = { - {0, 0, 0xF380, 0x8000, 0x6D60}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0} -}; - -static xtalk_coefs_t const asXtalkCoefsDenTest = { - {0xC000, 0x2000, 0x4000, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0} -}; - -static xtalk_state_t const asXtalkOutStateTest = { - {0x7FFF, 0x0004, 0xFFFC, 0}, - {0xFE00, 0x0008, 0xFFF8, 0x4000}, - {0x0200, 0x0010, 0xFFF0, 0xC000}, - {0x8000, 0x0020, 0xFFE0, 0}, - {0, 0, 0, 0} -}; static xtalk_coefs_t const asDiamondCoefsLeftEq = { {0x0F1E, 0x2D05, 0xF8E3, 0x07C8, 0}, @@ -228,15 +152,6 @@ static xtalk_coefs_t const asDiamondCoefsLeftXt = { {0, 0, 0, 0, 0} }; -static xtalk_coefs_t const asDiamondCoefsRightXt = { - {0x3B50, 0xFE08, 0xF959, 0x0060, 0}, - {0x9FCB, 0xD8F1, 0x00A2, 0x003A, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0} -}; - - /**/ /* XTalk EQ and XT */ static void vortex_XtalkHw_SetLeftEQ(vortex_t * vortex, short arg_0, short arg_4, -- 2.7.4

2 2

[alsa-devel] [PATCH -next] ALSA: ac97: remove unused variable 'snd_ac97_controls_master_mono'
by YueHaibing 09 Aug '19

09 Aug '19

sound/pci/ac97/ac97_codec.c:599:38: warning: snd_ac97_controls_master_mono defined but not used [-Wunused-const-variable=] It is never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/pci/ac97/ac97_codec.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 96b4601..66f6c3b 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -596,11 +596,6 @@ static int snd_ac97_put_volsw(struct snd_kcontrol *kcontrol, return err; } -static const struct snd_kcontrol_new snd_ac97_controls_master_mono[2] = { -AC97_SINGLE("Master Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), -AC97_SINGLE("Master Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1) -}; - static const struct snd_kcontrol_new snd_ac97_controls_tone[2] = { AC97_SINGLE("Tone Control - Bass", AC97_MASTER_TONE, 8, 15, 1), AC97_SINGLE("Tone Control - Treble", AC97_MASTER_TONE, 0, 15, 1) -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: hda - Apply workaround for another AMD chip 1022:1487
by Takashi Iwai 09 Aug '19

09 Aug '19

MSI MPG X570 board is with another AMD HD-audio controller (PCI ID 1022:1487) and it requires the same workaround applied for X370, etc (PCI ID 1022:1457). BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a6d8c0d77b84..99fc0917339b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2508,6 +2508,9 @@ static const struct pci_device_id azx_ids[] = { /* AMD, X370 & co */ { PCI_DEVICE(0x1022, 0x1457), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, + /* AMD, X570 & co */ + { PCI_DEVICE(0x1022, 0x1487), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, /* AMD Stoney */ { PCI_DEVICE(0x1022, 0x157a), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | -- 2.16.4

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[alsa-devel] [PATCH v2 0/4] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 09 Aug '19

09 Aug '19

This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This also adds support to some missing bits in SoundWire bus layer like Device Tree support and module_sdw_driver macro. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 speakers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. This patchset also depends on the soundwire Kconfig patch https://lkml.org/lkml/2019/7/18/834 from Pierre Thanks, srini Changes since v1 RFC: - bindings document renamed to slave.txt - fix error code from dt slave parsing Srinivas Kandagatla (4): dt-bindings: soundwire: add slave bindings soundwire: core: add device tree support for slave devices dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.txt | 27 + .../devicetree/bindings/soundwire/slave.txt | 46 + drivers/soundwire/bus.c | 2 + drivers/soundwire/bus.h | 1 + drivers/soundwire/slave.c | 47 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1160 +++++++++++++++++ 8 files changed, 1295 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.txt create mode 100644 Documentation/devicetree/bindings/soundwire/slave.txt create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

4 20

[alsa-devel] jack plugin freezes mpv.
by GitHub issues - opened 09 Aug '19

09 Aug '19

alsa-project/alsa-plugins issue #1 was opened from crocket: I'm using alsa-plugin-1.1.8 `mpv --audio-device=alsa/plug:rawjack /path/to/musicfile` freezes mpv. mpv doesn't freeze only when it plays a wav file. I have to press Ctrl+C twice to kill mpv. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/1 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[alsa-devel] [PATCH] soundwire: fix regmap dependencies and align with other serial links
by Pierre-Louis Bossart 09 Aug '19

09 Aug '19

The existing code has a mixed select/depend usage which makes no sense. config SOUNDWIRE_BUS tristate select REGMAP_SOUNDWIRE config REGMAP_SOUNDWIRE tristate depends on SOUNDWIRE_BUS Let's remove one layer of Kconfig definitions and align with the solutions used by all other serial links. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/base/regmap/Kconfig | 2 +- drivers/soundwire/Kconfig | 7 +------ drivers/soundwire/Makefile | 2 +- 3 files changed, 3 insertions(+), 8 deletions(-) diff --git a/drivers/base/regmap/Kconfig b/drivers/base/regmap/Kconfig index 6ad5ef48b61e..8cd2ac650b50 100644 --- a/drivers/base/regmap/Kconfig +++ b/drivers/base/regmap/Kconfig @@ -44,7 +44,7 @@ config REGMAP_IRQ config REGMAP_SOUNDWIRE tristate - depends on SOUNDWIRE_BUS + depends on SOUNDWIRE config REGMAP_SCCB tristate diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig index 3a01cfd70fdc..f518273cfbe3 100644 --- a/drivers/soundwire/Kconfig +++ b/drivers/soundwire/Kconfig @@ -4,7 +4,7 @@ # menuconfig SOUNDWIRE - bool "SoundWire support" + tristate "SoundWire support" help SoundWire is a 2-Pin interface with data and clock line ratified by the MIPI Alliance. SoundWire is used for transporting data @@ -17,17 +17,12 @@ if SOUNDWIRE comment "SoundWire Devices" -config SOUNDWIRE_BUS - tristate - select REGMAP_SOUNDWIRE - config SOUNDWIRE_CADENCE tristate config SOUNDWIRE_INTEL tristate "Intel SoundWire Master driver" select SOUNDWIRE_CADENCE - select SOUNDWIRE_BUS depends on X86 && ACPI && SND_SOC help SoundWire Intel Master driver. diff --git a/drivers/soundwire/Makefile b/drivers/soundwire/Makefile index fd99a831b92a..45b7e5001653 100644 --- a/drivers/soundwire/Makefile +++ b/drivers/soundwire/Makefile @@ -5,7 +5,7 @@ #Bus Objs soundwire-bus-objs := bus_type.o bus.o slave.o mipi_disco.o stream.o -obj-$(CONFIG_SOUNDWIRE_BUS) += soundwire-bus.o +obj-$(CONFIG_SOUNDWIRE) += soundwire-bus.o #Cadence Objs soundwire-cadence-objs := cadence_master.o -- 2.20.1

5 7

[alsa-devel] [asoc:for-5.4 195/195] sound/soc/sof/intel/hda-ctrl.c:257:8: error: 'struct hdac_bus' has no member named 'io_ops'; did you mean 'ops'?
by kbuild test robot 09 Aug '19

09 Aug '19

tree: https://kernel.googlesource.com/pub/scm/linux/kernel/git/broonie/sound.git for-5.4 head: c2f16a94a80497e4b28c27f9ca2cd6cd60706fb6 commit: c2f16a94a80497e4b28c27f9ca2cd6cd60706fb6 [195/195] Merge branch 'topic/hda-bus-ops-cleanup' of https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into asoc-5.4 config: i386-randconfig-b003-201931 (attached as .config) compiler: gcc-7 (Debian 7.4.0-10) 7.4.0 reproduce: git checkout c2f16a94a80497e4b28c27f9ca2cd6cd60706fb6 # save the attached .config to linux build tree make ARCH=i386 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): sound/soc/sof/intel/hda-ctrl.c: In function 'hda_dsp_ctrl_init_chip': >> sound/soc/sof/intel/hda-ctrl.c:257:8: error: 'struct hdac_bus' has no member named 'io_ops'; did you mean 'ops'? bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); ^~~~~~ ops vim +257 sound/soc/sof/intel/hda-ctrl.c be1b577d01787c Zhu Yingjiang 2019-05-24 240 be1b577d01787c Zhu Yingjiang 2019-05-24 241 /* enable CIE and GIE interrupts */ be1b577d01787c Zhu Yingjiang 2019-05-24 242 snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, be1b577d01787c Zhu Yingjiang 2019-05-24 243 SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN, be1b577d01787c Zhu Yingjiang 2019-05-24 244 SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN); be1b577d01787c Zhu Yingjiang 2019-05-24 245 be1b577d01787c Zhu Yingjiang 2019-05-24 246 /* program the position buffer */ be1b577d01787c Zhu Yingjiang 2019-05-24 247 if (bus->use_posbuf && bus->posbuf.addr) { 061edb23250579 Zhu Yingjiang 2019-07-22 248 snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, SOF_HDA_ADSP_DPLBASE, 061edb23250579 Zhu Yingjiang 2019-07-22 249 (u32)bus->posbuf.addr); 061edb23250579 Zhu Yingjiang 2019-07-22 250 snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, SOF_HDA_ADSP_DPUBASE, be1b577d01787c Zhu Yingjiang 2019-05-24 251 upper_32_bits(bus->posbuf.addr)); be1b577d01787c Zhu Yingjiang 2019-05-24 252 } be1b577d01787c Zhu Yingjiang 2019-05-24 253 d06973515f95ac Zhu Yingjiang 2019-07-22 254 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) d06973515f95ac Zhu Yingjiang 2019-07-22 255 /* Reset stream-to-link mapping */ d06973515f95ac Zhu Yingjiang 2019-07-22 256 list_for_each_entry(hlink, &bus->hlink_list, list) d06973515f95ac Zhu Yingjiang 2019-07-22 @257 bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); d06973515f95ac Zhu Yingjiang 2019-07-22 258 #endif d06973515f95ac Zhu Yingjiang 2019-07-22 259 be1b577d01787c Zhu Yingjiang 2019-05-24 260 bus->chip_init = true; be1b577d01787c Zhu Yingjiang 2019-05-24 261 be1b577d01787c Zhu Yingjiang 2019-05-24 262 hda_dsp_ctrl_misc_clock_gating(sdev, true); be1b577d01787c Zhu Yingjiang 2019-05-24 263 be1b577d01787c Zhu Yingjiang 2019-05-24 264 return ret; be1b577d01787c Zhu Yingjiang 2019-05-24 265 } 13063a2ccf79d2 Zhu Yingjiang 2019-06-12 266 :::::: The code at line 257 was first introduced by commit :::::: d06973515f95ac74d68fb3e1f1fb4b055531296c ASoC: SOF: Intel: hda: use SOF defined init chip in resume :::::: TO: Zhu Yingjiang <yingjiang.zhu(a)linux.intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [asoc:for-5.4 195/195] sound/soc/sof/intel/hda-dsp.c:360:22: error: 'hlink' undeclared
by kbuild test robot 09 Aug '19

09 Aug '19

tree: https://kernel.googlesource.com/pub/scm/linux/kernel/git/broonie/sound.git for-5.4 head: c2f16a94a80497e4b28c27f9ca2cd6cd60706fb6 commit: c2f16a94a80497e4b28c27f9ca2cd6cd60706fb6 [195/195] Merge branch 'topic/hda-bus-ops-cleanup' of https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into asoc-5.4 config: sparc64-allmodconfig (attached as .config) compiler: sparc64-linux-gcc (GCC) 7.4.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout c2f16a94a80497e4b28c27f9ca2cd6cd60706fb6 # save the attached .config to linux build tree GCC_VERSION=7.4.0 make.cross ARCH=sparc64 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All error/warnings (new ones prefixed by >>): In file included from include/linux/kobject.h:19:0, from include/linux/device.h:16, from include/sound/hdaudio.h:9, from include/sound/hdaudio_ext.h:5, from sound/soc/sof/intel/hda-dsp.c:18: sound/soc/sof/intel/hda-dsp.c: In function 'hda_resume': >> sound/soc/sof/intel/hda-dsp.c:360:22: error: 'hlink' undeclared (first use in this function) list_for_each_entry(hlink, &bus->hlink_list, list) ^ include/linux/list.h:577:7: note: in definition of macro 'list_for_each_entry' for (pos = list_first_entry(head, typeof(*pos), member); \ ^~~ sound/soc/sof/intel/hda-dsp.c:360:22: note: each undeclared identifier is reported only once for each function it appears in list_for_each_entry(hlink, &bus->hlink_list, list) ^ include/linux/list.h:577:7: note: in definition of macro 'list_for_each_entry' for (pos = list_first_entry(head, typeof(*pos), member); \ ^~~ In file included from include/linux/list.h:9:0, from include/linux/kobject.h:19, from include/linux/device.h:16, from include/sound/hdaudio.h:9, from include/sound/hdaudio_ext.h:5, from sound/soc/sof/intel/hda-dsp.c:18: >> sound/soc/sof/intel/hda-dsp.c:360:30: error: 'bus' undeclared (first use in this function) list_for_each_entry(hlink, &bus->hlink_list, list) ^ include/linux/kernel.h:972:26: note: in definition of macro 'container_of' void *__mptr = (void *)(ptr); \ ^~~ include/linux/list.h:490:2: note: in expansion of macro 'list_entry' list_entry((ptr)->next, type, member) ^~~~~~~~~~ include/linux/list.h:577:13: note: in expansion of macro 'list_first_entry' for (pos = list_first_entry(head, typeof(*pos), member); \ ^~~~~~~~~~~~~~~~ >> sound/soc/sof/intel/hda-dsp.c:360:2: note: in expansion of macro 'list_for_each_entry' list_for_each_entry(hlink, &bus->hlink_list, list) ^~~~~~~~~~~~~~~~~~~ >> sound/soc/sof/intel/hda-dsp.c:364:2: error: #else without #if #else ^~~~ >> sound/soc/sof/intel/hda-dsp.c:390:2: error: #endif without #if #endif ^~~~~ vim +/hlink +360 sound/soc/sof/intel/hda-dsp.c 747503b1813a3e Liam Girdwood 2019-04-12 334 fd15f2f5e27214 Rander Wang 2019-07-22 335 static int hda_resume(struct snd_sof_dev *sdev, bool runtime_resume) 747503b1813a3e Liam Girdwood 2019-04-12 336 { 747503b1813a3e Liam Girdwood 2019-04-12 337 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) 747503b1813a3e Liam Girdwood 2019-04-12 338 struct hdac_bus *bus = sof_to_bus(sdev); 747503b1813a3e Liam Girdwood 2019-04-12 339 struct hdac_ext_link *hlink = NULL; 747503b1813a3e Liam Girdwood 2019-04-12 340 #endif 747503b1813a3e Liam Girdwood 2019-04-12 341 int ret; 747503b1813a3e Liam Girdwood 2019-04-12 342 747503b1813a3e Liam Girdwood 2019-04-12 343 /* 747503b1813a3e Liam Girdwood 2019-04-12 344 * clear TCSEL to clear playback on some HD Audio 747503b1813a3e Liam Girdwood 2019-04-12 345 * codecs. PCI TCSEL is defined in the Intel manuals. 747503b1813a3e Liam Girdwood 2019-04-12 346 */ 747503b1813a3e Liam Girdwood 2019-04-12 347 snd_sof_pci_update_bits(sdev, PCI_TCSEL, 0x07, 0); 747503b1813a3e Liam Girdwood 2019-04-12 348 747503b1813a3e Liam Girdwood 2019-04-12 349 /* reset and start hda controller */ 747503b1813a3e Liam Girdwood 2019-04-12 350 ret = hda_dsp_ctrl_init_chip(sdev, true); 747503b1813a3e Liam Girdwood 2019-04-12 351 if (ret < 0) { 747503b1813a3e Liam Girdwood 2019-04-12 352 dev_err(sdev->dev, 747503b1813a3e Liam Girdwood 2019-04-12 353 "error: failed to start controller after resume\n"); 747503b1813a3e Liam Girdwood 2019-04-12 354 return ret; 747503b1813a3e Liam Girdwood 2019-04-12 355 } 747503b1813a3e Liam Girdwood 2019-04-12 356 747503b1813a3e Liam Girdwood 2019-04-12 357 hda_dsp_ctrl_misc_clock_gating(sdev, false); 747503b1813a3e Liam Girdwood 2019-04-12 358 747503b1813a3e Liam Girdwood 2019-04-12 359 /* Reset stream-to-link mapping */ 747503b1813a3e Liam Girdwood 2019-04-12 @360 list_for_each_entry(hlink, &bus->hlink_list, list) 19abfefd4c7604 Takashi Iwai 2019-08-07 361 writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); 747503b1813a3e Liam Girdwood 2019-04-12 362 747503b1813a3e Liam Girdwood 2019-04-12 363 hda_dsp_ctrl_misc_clock_gating(sdev, true); 747503b1813a3e Liam Girdwood 2019-04-12 @364 #else 747503b1813a3e Liam Girdwood 2019-04-12 365 747503b1813a3e Liam Girdwood 2019-04-12 366 hda_dsp_ctrl_misc_clock_gating(sdev, false); 747503b1813a3e Liam Girdwood 2019-04-12 367 747503b1813a3e Liam Girdwood 2019-04-12 368 /* reset controller */ 747503b1813a3e Liam Girdwood 2019-04-12 369 ret = hda_dsp_ctrl_link_reset(sdev, true); 747503b1813a3e Liam Girdwood 2019-04-12 370 if (ret < 0) { 747503b1813a3e Liam Girdwood 2019-04-12 371 dev_err(sdev->dev, 747503b1813a3e Liam Girdwood 2019-04-12 372 "error: failed to reset controller during resume\n"); 747503b1813a3e Liam Girdwood 2019-04-12 373 return ret; 747503b1813a3e Liam Girdwood 2019-04-12 374 } 747503b1813a3e Liam Girdwood 2019-04-12 375 747503b1813a3e Liam Girdwood 2019-04-12 376 /* take controller out of reset */ 747503b1813a3e Liam Girdwood 2019-04-12 377 ret = hda_dsp_ctrl_link_reset(sdev, false); 747503b1813a3e Liam Girdwood 2019-04-12 378 if (ret < 0) { 747503b1813a3e Liam Girdwood 2019-04-12 379 dev_err(sdev->dev, 747503b1813a3e Liam Girdwood 2019-04-12 380 "error: failed to ready controller during resume\n"); 747503b1813a3e Liam Girdwood 2019-04-12 381 return ret; 747503b1813a3e Liam Girdwood 2019-04-12 382 } 747503b1813a3e Liam Girdwood 2019-04-12 383 747503b1813a3e Liam Girdwood 2019-04-12 384 /* enable hda bus irq */ 747503b1813a3e Liam Girdwood 2019-04-12 385 snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, 747503b1813a3e Liam Girdwood 2019-04-12 386 SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN, 747503b1813a3e Liam Girdwood 2019-04-12 387 SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN); 747503b1813a3e Liam Girdwood 2019-04-12 388 747503b1813a3e Liam Girdwood 2019-04-12 389 hda_dsp_ctrl_misc_clock_gating(sdev, true); 24b6ff686fce80 Zhu Yingjiang 2019-06-12 @390 #endif 747503b1813a3e Liam Girdwood 2019-04-12 391 747503b1813a3e Liam Girdwood 2019-04-12 392 /* enable ppcap interrupt */ 747503b1813a3e Liam Girdwood 2019-04-12 393 hda_dsp_ctrl_ppcap_enable(sdev, true); 747503b1813a3e Liam Girdwood 2019-04-12 394 hda_dsp_ctrl_ppcap_int_enable(sdev, true); 747503b1813a3e Liam Girdwood 2019-04-12 395 :::::: The code at line 360 was first introduced by commit :::::: 747503b1813a3e6e4c52d9c0b4bd462b64940940 ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations :::::: TO: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [sound:topic/hda-bus-ops-cleanup 2/3] sound/hda/hdac_bus.c:228:6: error: implicit declaration of function 'readl'; did you mean 'd_real'?
by kbuild test robot 09 Aug '19

09 Aug '19

tree: https://kernel.googlesource.com/pub/scm/linux/kernel/git/tiwai/sound.git topic/hda-bus-ops-cleanup head: d4ff1b3917a529bdc75592af6b1504ad6c4029f7 commit: 19abfefd4c7604993d1c31e098a3f48bdafe334d [2/3] ALSA: hda: Direct MMIO accesses config: arm-defconfig (attached as .config) compiler: arm-linux-gnueabi-gcc (GCC) 7.4.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 19abfefd4c7604993d1c31e098a3f48bdafe334d # save the attached .config to linux build tree GCC_VERSION=7.4.0 make.cross ARCH=arm If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): sound/hda/hdac_bus.c: In function 'snd_hdac_aligned_read': >> sound/hda/hdac_bus.c:228:6: error: implicit declaration of function 'readl'; did you mean 'd_real'? [-Werror=implicit-function-declaration] v = readl(aligned_addr); ^~~~~ d_real sound/hda/hdac_bus.c: In function 'snd_hdac_aligned_write': >> sound/hda/hdac_bus.c:244:2: error: implicit declaration of function 'writel' [-Werror=implicit-function-declaration] writel(v, aligned_addr); ^~~~~~ cc1: some warnings being treated as errors vim +228 sound/hda/hdac_bus.c 218 219 #ifdef CONFIG_SND_HDA_ALIGNED_MMIO 220 /* Helpers for aligned read/write of mmio space, for Tegra */ 221 unsigned int snd_hdac_aligned_read(void __iomem *addr, unsigned int mask) 222 { 223 void __iomem *aligned_addr = 224 (void __iomem *)((unsigned long)(addr) & ~0x3); 225 unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; 226 unsigned int v; 227 > 228 v = readl(aligned_addr); 229 return (v >> shift) & mask; 230 } 231 EXPORT_SYMBOL_GPL(snd_hdac_aligned_read); 232 233 void snd_hdac_aligned_write(unsigned int val, void __iomem *addr, 234 unsigned int mask) 235 { 236 void __iomem *aligned_addr = 237 (void __iomem *)((unsigned long)(addr) & ~0x3); 238 unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; 239 unsigned int v; 240 241 v = readl(aligned_addr); 242 v &= ~(mask << shift); 243 v |= val << shift; > 244 writel(v, aligned_addr); --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: sof: Fix warning when IPC flood test is not enabled
by Mark Brown 09 Aug '19

09 Aug '19

dentry is only used when the flood test is done so move the declaration of the variable inside the ifdef for the flood test. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/debug.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index 40940b2fe9d5..54cd431faab7 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -221,7 +221,6 @@ static ssize_t sof_dfsentry_read(struct file *file, char __user *buffer, { struct snd_sof_dfsentry *dfse = file->private_data; struct snd_sof_dev *sdev = dfse->sdev; - struct dentry *dentry; loff_t pos = *ppos; size_t size_ret; int skip = 0; @@ -229,6 +228,8 @@ static ssize_t sof_dfsentry_read(struct file *file, char __user *buffer, u8 *buf; #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST) + struct dentry *dentry; + dentry = file->f_path.dentry; if ((!strcmp(dentry->d_name.name, "ipc_flood_count") || !strcmp(dentry->d_name.name, "ipc_flood_duration_ms")) && -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: hdac_hda: fix page fault issue by removing race
by Pierre-Louis Bossart 08 Aug '19

08 Aug '19

From: Keyon Jie <yang.jie(a)linux.intel.com> There is a race between hda codec device removing and the jack-detecting work, which will lead to a page fault issue as the latter work is accessing codec device which could be already removed. Here add the cancellation of jack-detecting work before codecs are actually removed to avoid the race and fix the issue. Bug: https://github.com/thesofproject/linux/issues/1067 Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/hdac_hda.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 7d4940256914..91242b6f8ea7 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -495,6 +495,10 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) static int hdac_hda_dev_remove(struct hdac_device *hdev) { + struct hdac_hda_priv *hda_pvt; + + hda_pvt = dev_get_drvdata(&hdev->dev); + cancel_delayed_work_sync(&hda_pvt->codec.jackpoll_work); return 0; } -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: soc-core: fix module_put() warning in soc_cleanup_component
by Pierre-Louis Bossart 08 Aug '19

08 Aug '19

The recent changes introduce warnings in the SOF load/unload module tests. The code does not seem balanced with a confusion between _close() and _remove() macros. Using _remove() fixes the issue and removes the warning. Suggested-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Fixes: 4a81e8f30d0b4 ('ASoC: soc-component: add snd_soc_component_get/put()') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- Full trace for reference: [ 98.735393] WARNING: CPU: 3 PID: 1397 at kernel/module.c:1137 module_put+0x83/0x90 [ 98.735393] Modules linked in: snd_soc_cnl_rt700 soundwire_intel soundwire_cadence snd_soc_dmic snd_sof_pci(-) snd_sof_intel_hda_common soundwire_intel_init snd_sof_acpi snd_sof_intel_byt snd_soc_acpi_intel_match snd_sof_intel_bdw snd_sof_intel_ipc snd_sof snd_sof_xtensa_dsp snd_soc_acpi snd_soc_rt700 regmap_sdw soundwire_bus snd_soc_max98357a snd_soc_wm8804_i2c snd_soc_wm8804 snd_soc_pcm512x_i2c snd_soc_pcm512x snd_soc_rt5640 snd_soc_rl6231 snd_soc_rt286 snd_soc_rl6347a snd_soc_core snd_pcm snd_seq snd_seq_device snd_timer snd soundcore wmi_bmof asix usbnet x86_pkg_temp_thermal intel_powerclamp i915 i2c_algo_bit mei_me mei drm_kms_helper syscopyarea sysfillrect sysimgblt fb_sys_fops iwlwifi drm wmi efivarfs sdhci_pci intel_lpss_pci xhci_pci cqhci intel_lpss sdhci mfd_core xhci_hcd [ 98.735410] CPU: 3 PID: 1397 Comm: rmmod Not tainted 5.3.0-rc3-test+ #122 [ 98.735411] Hardware name: Intel Corporation Ice Lake Client Platform/IceLake<DETAILS REMOVED> [ 98.735413] RIP: 0010:module_put+0x83/0x90 [ 98.735414] Code: 48 83 c5 18 4c 89 e2 48 89 de e8 68 1c b1 00 48 8b 45 00 48 85 c0 75 e4 5b 5d 41 5c c3 c3 48 c7 c7 20 f9 7b 9d e8 d6 97 fc ff <0f> 0b eb 9c 66 0f 1f 84 00 00 00 00 00 53 48 89 f3 e8 67 ff ff ff [ 98.735415] RSP: 0018:ffffa88200a33c80 EFLAGS: 00010246 [ 98.735416] RAX: 0000000000000024 RBX: ffffffffc0754280 RCX: 0000000000000000 [ 98.735417] RDX: 0000000000000000 RSI: ffff8cc167ed63d8 RDI: ffff8cc167ed63d8 [ 98.735417] RBP: ffff8cc162cc0028 R08: 000000000000041d R09: 0000000000000004 [ 98.735418] R10: 0000000000000000 R11: 0000000000000001 R12: ffff8cc1456fceb0 [ 98.735419] R13: 0000000000000000 R14: 0000000000000001 R15: ffffffff9d7d85c5 [ 98.735420] FS: 00007fe72a461740(0000) GS:ffff8cc167ec0000(0000) knlGS:0000000000000000 [ 98.735420] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 98.735421] CR2: 000056509f6f2fe4 CR3: 0000000268d8a005 CR4: 0000000000760ee0 [ 98.735422] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [ 98.735422] DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 [ 98.735423] PKRU: 55555554 [ 98.735423] Call Trace: [ 98.735430] soc_remove_link_components.isra.50+0x56/0x70 [snd_soc_core] [ 98.735434] snd_soc_unbind_card.part.52+0x75/0xf0 [snd_soc_core] [ 98.735437] snd_soc_unregister_card+0x8d/0x90 [snd_soc_core] [ 98.735441] release_nodes+0x1ae/0x200 [ 98.735443] device_release_driver_internal+0xe3/0x1b0 [ 98.735445] bus_remove_device+0xd6/0x140 [ 98.735447] device_del+0x167/0x380 [ 98.735449] platform_device_del.part.16+0xe/0x60 [ 98.735451] platform_device_unregister+0x17/0x30 [ 98.735453] snd_sof_device_remove+0x44/0x80 [snd_sof] [ 98.735455] sof_pci_remove+0x10/0x30 [snd_sof_pci] [ 98.735457] pci_device_remove+0x36/0xb0 [ 98.735459] device_release_driver_internal+0xd3/0x1b0 [ 98.735460] driver_detach+0x37/0x6b [ 98.735461] bus_remove_driver+0x6f/0xc1 [ 98.735462] pci_unregister_driver+0x31/0x90 sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bb1e9e2c4ff4..00e8ad122ce7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -924,7 +924,7 @@ static void soc_cleanup_component(struct snd_soc_component *component) snd_soc_dapm_free(snd_soc_component_get_dapm(component)); soc_cleanup_component_debugfs(component); component->card = NULL; - snd_soc_component_module_put_when_close(component); + snd_soc_component_module_put_when_remove(component); } static void soc_remove_component(struct snd_soc_component *component) -- 2.20.1

3 2

[alsa-devel] [PATCH -next] ASoC: tscs454: remove unused variable 'PLL_48K_RATE'
by YueHaibing 08 Aug '19

08 Aug '19

The global variable 'PLL_48K_RATE' is never used so just remove it. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/tscs454.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index 93d84e5..c3587af 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -22,7 +22,6 @@ #include "tscs454.h" -static const unsigned int PLL_48K_RATE = (48000 * 256); static const unsigned int PLL_44_1K_RATE = (44100 * 256); #define COEFF_SIZE 3 -- 2.7.4

2 1

[alsa-devel] [PATCH v5 0/2] ASoC: Intel: Skylake: large_config_get overhaul
by Cezary Rojewski 08 Aug '19

08 Aug '19

LARGE_CONFIG_GET is among the most crucial IPCs. Host is expected to send single request first to obtain total payload size from received header's data_off_size. From then on, it loops for each frame exceeding inbox size until entire payload is read. If entire payload is contained within the very first frame, no looping is performed. LARGE_CONFIG_GET is a flexible IPC, it not only receives payload but may carry one with them to provide list of params DSP module should return info for. This behavior is usually reserved for vendors and IPC handler should not touch that content. To achieve that, simply pass provided payload and bytes to sst_ipc_tx_message_wait as a part of request. In current state, LARGE_CONFIG_GET message handler does nothing of that, in consequence making it dysfunctional. Overhaul said handler allowing rightful king of IPCs to return back on his throne - kingdom he shares with his twin brother: LARGE_CONFIG_SET. The looping has not been added in this update as payloads of such size do not exist in practice. Will need to create custom module specifically for that very case and test against it before that feature can be added. Changes since v4: - Addressed Pierre's review: updated function's declaration and dropped unrelated log changes Changes since v3: - Split "large_config_get overhaul" patch into two segments as suggested by Pierre: first remove looping, then adjust function's behavior Changes since v2: - None, just for-next rebase Changes since v1: - None, just for-next rebase Cezary Rojewski (2): ASoC: Intel: Skylake: Limit large_config_get to single frame ASoC: Intel: Skylake: large_config_get overhaul sound/soc/intel/skylake/skl-messages.c | 3 +- sound/soc/intel/skylake/skl-sst-ipc.c | 54 +++++++++++--------------- sound/soc/intel/skylake/skl-sst-ipc.h | 3 +- 3 files changed, 27 insertions(+), 33 deletions(-) -- 2.17.1

3 5

[alsa-devel] [PATCH] ASoC: intel: skl_hda_dsp_common: create HDMI jack kctl
by Pierre-Louis Bossart 08 Aug '19

08 Aug '19

From: Libin Yang <libin.yang(a)intel.com> This patch call snd_jack_add_new_kctl() to create the HDMI jack kctls. Userspace needs these kctls to detect the hdmi monitor hotplug. In /usr/share/alsa/ucm, the config file needs to assign a jack kctl to "JackControl" to let PA get the jack hotplug status. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- This patch has been used by SOF-based solutions delivered in Q2 but due to a confusion on my side was not upstreamed. Both Libin and Kai confirmed it's required for PulseAudio to track jack events. sound/soc/intel/boards/skl_hda_dsp_common.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c index 55fd82e05e2c..58409b6e476e 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.c +++ b/sound/soc/intel/boards/skl_hda_dsp_common.c @@ -147,6 +147,11 @@ int skl_hda_hdmi_jack_init(struct snd_soc_card *card) if (err) return err; + err = snd_jack_add_new_kctl(pcm->hdmi_jack.jack, + jack_name, SND_JACK_AVOUT); + if (err) + dev_warn(component->dev, "failed creating Jack kctl\n"); + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, &pcm->hdmi_jack); if (err < 0) -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: soc-core: remove error due to probe deferral
by Stefan Agner 08 Aug '19

08 Aug '19

From: Stefan Agner <stefan.agner(a)toradex.com> Deferred probes shouldn't cause error messages in the boot log. Avoid printing with dev_err() in case EPROBE_DEFER is the return value. Signed-off-by: Stefan Agner <stefan.agner(a)toradex.com> --- sound/soc/soc-core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fd6eaae6c0ed..98e1e80b5493 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1985,9 +1985,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); for_each_card_prelinks(card, i, dai_link) { ret = soc_init_dai_link(card, dai_link); - if (ret) { + if (ret && ret != -EPROBE_DEFER) { dev_err(card->dev, "ASoC: failed to init link %s: %d\n", dai_link->name, ret); + } + if (ret) { mutex_unlock(&client_mutex); return ret; } -- 2.22.0

3 7

[alsa-devel] [PATCH] ASoC: Intel: boards: Add Cometlake machine driver support
by mac.chiang＠intel.com 08 Aug '19

08 Aug '19

From: Mac Chiang <mac.chiang(a)intel.com> reuse and add Cometlake support with: SSP0 for DA7219 headphone codec SSP1 for MAX98357a speaker amp codec Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/Kconfig | 15 +++++++++ sound/soc/intel/boards/bxt_da7219_max98357a.c | 37 ++++++++++++++++++++++- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 12 ++++++++ sound/soc/intel/common/soc-intel-quirks.h | 5 +++ 4 files changed, 68 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 50bf149..617d63c 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -256,6 +256,21 @@ config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH endif ## SND_SOC_INTEL_SKL +config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC + tristate + select SND_SOC_DA7219 + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + +if SND_SOC_SOF_COMETLAKE_LP +config SND_SOC_INTEL_CML_LP_DA7219_MAX98357A_MACH + tristate "CML_LP with DA7219 and MAX98357A in I2S Mode" + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC +endif + if SND_SOC_INTEL_APL config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index c0d865a..e5941ff 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -24,6 +24,8 @@ #define BXT_DIALOG_CODEC_DAI "da7219-hifi" #define BXT_MAXIM_CODEC_DAI "HiFi" +#define MCLK_19M 19200000 +#define MCLK_24M 24576000 #define DUAL_CHANNEL 2 #define QUAD_CHANNEL 4 @@ -181,8 +183,13 @@ static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_component *component = rtd->codec_dai->component; /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 19200000, +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, MCLK_19M, SND_SOC_CLOCK_IN); +#else + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, MCLK_24M, + SND_SOC_CLOCK_IN); +#endif if (ret) { dev_err(rtd->dev, "can't set codec sysclk configuration\n"); return ret; @@ -224,7 +231,11 @@ static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd) if (!pcm) return -ENOMEM; +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) pcm->device = BXT_DPCM_AUDIO_HDMI1_PB + dai->id; +#else + pcm->device = dai->id; +#endif pcm->codec_dai = dai; list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); @@ -683,6 +694,27 @@ static int broxton_audio_probe(struct platform_device *pdev) broxton_dais[i].cpus->dai_name = "SSP2 Pin"; } } + } else if (soc_intel_is_cml()) { + unsigned int i; + + broxton_audio_card.name = "cmlda7219max"; + + for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) { + /* MAXIM_CODEC is connected to SSP1. */ + if (!strcmp(broxton_dais[i].codecs->dai_name, + BXT_MAXIM_CODEC_DAI)) { + broxton_dais[i].id = 1; + broxton_dais[i].name = "SSP1-Codec"; + broxton_dais[i].cpus->dai_name = "SSP1 Pin"; + } + /* DIALOG_CODEC is connected to SSP0 */ + else if (!strcmp(broxton_dais[i].codecs->dai_name, + BXT_DIALOG_CODEC_DAI)) { + broxton_dais[i].id = 0; + broxton_dais[i].name = "SSP0-Codec"; + broxton_dais[i].cpus->dai_name = "SSP0 Pin"; + } + } } /* override plaform name, if required */ @@ -700,6 +732,7 @@ static int broxton_audio_probe(struct platform_device *pdev) static const struct platform_device_id bxt_board_ids[] = { { .name = "bxt_da7219_max98357a" }, { .name = "glk_da7219_max98357a" }, + { .name = "cml_da7219_max98357a" }, { } }; @@ -720,6 +753,8 @@ MODULE_AUTHOR("Rohit Ainapure <rohit.m.ainapure(a)intel.com>"); MODULE_AUTHOR("Harsha Priya <harshapriya.n(a)intel.com>"); MODULE_AUTHOR("Conrad Cooke <conrad.cooke(a)intel.com>"); MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:bxt_da7219_max98357a"); MODULE_ALIAS("platform:glk_da7219_max98357a"); +MODULE_ALIAS("platform:cml_da7219_max98357a"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c index c36c0aa..4ea32b2 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -19,6 +19,11 @@ static struct snd_soc_acpi_codecs cml_codecs = { .codecs = {"10EC5682"} }; +static struct snd_soc_acpi_codecs cml_spk_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { { .id = "INT34C2", @@ -29,6 +34,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { .sof_tplg_filename = "sof-cnl-rt274.tplg", }, { + .id = "DLGS7219", + .drv_name = "cml_da7219_max98357a", + .quirk_data = &cml_spk_codecs, + .sof_fw_filename = "sof-cnl.ri", + .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", + }, + { .id = "MX98357A", .drv_name = "sof_rt5682", .quirk_data = &cml_codecs, diff --git a/sound/soc/intel/common/soc-intel-quirks.h b/sound/soc/intel/common/soc-intel-quirks.h index 4718fd3..e6357d3 100644 --- a/sound/soc/intel/common/soc-intel-quirks.h +++ b/sound/soc/intel/common/soc-intel-quirks.h @@ -36,6 +36,7 @@ SOC_INTEL_IS_CPU(byt, INTEL_FAM6_ATOM_SILVERMONT); SOC_INTEL_IS_CPU(cht, INTEL_FAM6_ATOM_AIRMONT); SOC_INTEL_IS_CPU(apl, INTEL_FAM6_ATOM_GOLDMONT); SOC_INTEL_IS_CPU(glk, INTEL_FAM6_ATOM_GOLDMONT_PLUS); +SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_MOBILE); static inline bool soc_intel_is_byt_cr(struct platform_device *pdev) { @@ -110,6 +111,10 @@ static inline bool soc_intel_is_glk(void) return false; } +static inline bool soc_intel_is_cml(void) +{ + return false; +} #endif #endif /* _SND_SOC_INTEL_QUIRKS_H */ -- 2.7.4

3 5

[alsa-devel] [PATCH v4 2/2] ASoC: Intel: Skylake: large_config_get overhaul
by Cezary Rojewski 08 Aug '19

08 Aug '19

LARGE_CONFIG_GET is mainly used to retrieve requested module parameters but it may also carry TX payload with them. Update its implementation to account for both TX and RX data. First reply.header carries total payload size within data_off_sizefield. Make use of reply.header to realloc returned buffer with correct size. Failure of IPC request is permissive - error-payload may be returned, an informative data why GET for given param failed - and thus function should not collapse before entire processing is finished. Caller is responsible for checking returned payload and bytes parameters. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/skylake/skl-messages.c | 3 ++- sound/soc/intel/skylake/skl-sst-ipc.c | 27 ++++++++++++++++++++------ sound/soc/intel/skylake/skl-sst-ipc.h | 3 ++- 3 files changed, 25 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index e8cc710f092b..84f0e6f58eb5 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -1379,11 +1379,12 @@ int skl_get_module_params(struct skl_dev *skl, u32 *params, int size, u32 param_id, struct skl_module_cfg *mcfg) { struct skl_ipc_large_config_msg msg; + size_t bytes = size; msg.module_id = mcfg->id.module_id; msg.instance_id = mcfg->id.pvt_id; msg.param_data_size = size; msg.large_param_id = param_id; - return skl_ipc_get_large_config(&skl->ipc, &msg, params); + return skl_ipc_get_large_config(&skl->ipc, &msg, &params, &bytes); } diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 196c80dadb1f..9d269a5f8bd9 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -969,12 +969,18 @@ int skl_ipc_set_large_config(struct sst_generic_ipc *ipc, EXPORT_SYMBOL_GPL(skl_ipc_set_large_config); int skl_ipc_get_large_config(struct sst_generic_ipc *ipc, - struct skl_ipc_large_config_msg *msg, u32 *param) + struct skl_ipc_large_config_msg *msg, + unsigned int **payload, size_t *bytes) { struct skl_ipc_header header = {0}; - struct sst_ipc_message request = {0}, reply = {0}; + struct sst_ipc_message request, reply = {0}; + unsigned int *buf; int ret; + reply.data = kzalloc(SKL_ADSP_W1_SZ, GFP_KERNEL); + if (!reply.data) + return -ENOMEM; + header.primary = IPC_MSG_TARGET(IPC_MOD_MSG); header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); header.primary |= IPC_GLB_TYPE(IPC_MOD_LARGE_CONFIG_GET); @@ -986,12 +992,21 @@ int skl_ipc_get_large_config(struct sst_generic_ipc *ipc, header.extension |= IPC_FINAL_BLOCK(1); header.extension |= IPC_INITIAL_BLOCK(1); - request.header = *(u64 *)(&header); - reply.data = param; - reply.size = msg->param_data_size; + request.header = *(u64 *)&header; + request.data = *payload; + request.size = *bytes; + reply.size = SKL_ADSP_W1_SZ; + ret = sst_ipc_tx_message_wait(ipc, request, &reply); if (ret < 0) - dev_err(ipc->dev, "ipc: get large config fail, err: %d\n", ret); + dev_err(ipc->dev, "ipc: get large config fail: %d\n", ret); + + reply.size = (reply.header >> 32) & IPC_DATA_OFFSET_SZ_MASK; + buf = krealloc(reply.data, reply.size, GFP_KERNEL); + if (!buf) + return -ENOMEM; + *payload = buf; + *bytes = reply.size; return ret; } diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h index 93af08cf41d2..a7ab2c589cc5 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.h +++ b/sound/soc/intel/skylake/skl-sst-ipc.h @@ -139,7 +139,8 @@ int skl_ipc_set_large_config(struct sst_generic_ipc *ipc, struct skl_ipc_large_config_msg *msg, u32 *param); int skl_ipc_get_large_config(struct sst_generic_ipc *ipc, - struct skl_ipc_large_config_msg *msg, u32 *param); + struct skl_ipc_large_config_msg *msg, + unsigned int **payload, size_t *bytes); int skl_sst_ipc_load_library(struct sst_generic_ipc *ipc, u8 dma_id, u8 table_id, bool wait); -- 2.17.1

2 3

[alsa-devel] [RESEND PATCH v3] ASoC: Intel: Skylake: large_config_get overhaul
by Cezary Rojewski 08 Aug '19

08 Aug '19

LARGE_CONFIG_GET is among the most crucial IPCs. Host is expected to send single request first to obtain total payload size from received header's data_off_size. From then on, it loops for each frame exceeding inbox size until entire payload is read. If entire payload is contained within the very first frame, no looping is performed. LARGE_CONFIG_GET is a flexible IPC, it not only receives payload but may carry one with them to provide list of params DSP module should return info for. This behavior is usually reserved for vendors and IPC handler should not touch that content. To achieve that, simply pass provided payload and bytes to sst_ipc_tx_message_wait as a part of request. In current state, LARGE_CONFIG_GET message handler does nothing of that, in consequence making it dysfunctional. Overhaul said handler allowing rightful king of IPCs to return back on his throne - kingdom he shares with his twin brother: LARGE_CONFIG_SET. The looping has not been added in this update as payloads of such size do not exist in practice. Will need to create custom module specifically for that very case and test against it before that feature can be added. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/skylake/skl-messages.c | 3 +- sound/soc/intel/skylake/skl-sst-ipc.c | 54 +++++++++++--------------- sound/soc/intel/skylake/skl-sst-ipc.h | 3 +- 3 files changed, 27 insertions(+), 33 deletions(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index e8cc710f092b..84f0e6f58eb5 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -1379,11 +1379,12 @@ int skl_get_module_params(struct skl_dev *skl, u32 *params, int size, u32 param_id, struct skl_module_cfg *mcfg) { struct skl_ipc_large_config_msg msg; + size_t bytes = size; msg.module_id = mcfg->id.module_id; msg.instance_id = mcfg->id.pvt_id; msg.param_data_size = size; msg.large_param_id = param_id; - return skl_ipc_get_large_config(&skl->ipc, &msg, params); + return skl_ipc_get_large_config(&skl->ipc, &msg, &params, &bytes); } diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index a2b69a02aab2..9d269a5f8bd9 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -969,12 +969,17 @@ int skl_ipc_set_large_config(struct sst_generic_ipc *ipc, EXPORT_SYMBOL_GPL(skl_ipc_set_large_config); int skl_ipc_get_large_config(struct sst_generic_ipc *ipc, - struct skl_ipc_large_config_msg *msg, u32 *param) + struct skl_ipc_large_config_msg *msg, + unsigned int **payload, size_t *bytes) { struct skl_ipc_header header = {0}; - struct sst_ipc_message request = {0}, reply = {0}; - int ret = 0; - size_t sz_remaining, rx_size, data_offset; + struct sst_ipc_message request, reply = {0}; + unsigned int *buf; + int ret; + + reply.data = kzalloc(SKL_ADSP_W1_SZ, GFP_KERNEL); + if (!reply.data) + return -ENOMEM; header.primary = IPC_MSG_TARGET(IPC_MOD_MSG); header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); @@ -987,34 +992,21 @@ int skl_ipc_get_large_config(struct sst_generic_ipc *ipc, header.extension |= IPC_FINAL_BLOCK(1); header.extension |= IPC_INITIAL_BLOCK(1); - sz_remaining = msg->param_data_size; - data_offset = 0; - - while (sz_remaining != 0) { - rx_size = sz_remaining > SKL_ADSP_W1_SZ - ? SKL_ADSP_W1_SZ : sz_remaining; - if (rx_size == sz_remaining) - header.extension |= IPC_FINAL_BLOCK(1); - - request.header = *(u64 *)(&header); - reply.data = ((char *)param) + data_offset; - reply.size = msg->param_data_size; - ret = sst_ipc_tx_message_wait(ipc, request, &reply); - if (ret < 0) { - dev_err(ipc->dev, - "ipc: get large config fail, err: %d\n", ret); - return ret; - } - sz_remaining -= rx_size; - data_offset = msg->param_data_size - sz_remaining; + request.header = *(u64 *)&header; + request.data = *payload; + request.size = *bytes; + reply.size = SKL_ADSP_W1_SZ; - /* clear the fields */ - header.extension &= IPC_INITIAL_BLOCK_CLEAR; - header.extension &= IPC_DATA_OFFSET_SZ_CLEAR; - /* fill the fields */ - header.extension |= IPC_INITIAL_BLOCK(1); - header.extension |= IPC_DATA_OFFSET_SZ(data_offset); - } + ret = sst_ipc_tx_message_wait(ipc, request, &reply); + if (ret < 0) + dev_err(ipc->dev, "ipc: get large config fail: %d\n", ret); + + reply.size = (reply.header >> 32) & IPC_DATA_OFFSET_SZ_MASK; + buf = krealloc(reply.data, reply.size, GFP_KERNEL); + if (!buf) + return -ENOMEM; + *payload = buf; + *bytes = reply.size; return ret; } diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h index 93af08cf41d2..a7ab2c589cc5 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.h +++ b/sound/soc/intel/skylake/skl-sst-ipc.h @@ -139,7 +139,8 @@ int skl_ipc_set_large_config(struct sst_generic_ipc *ipc, struct skl_ipc_large_config_msg *msg, u32 *param); int skl_ipc_get_large_config(struct sst_generic_ipc *ipc, - struct skl_ipc_large_config_msg *msg, u32 *param); + struct skl_ipc_large_config_msg *msg, + unsigned int **payload, size_t *bytes); int skl_sst_ipc_load_library(struct sst_generic_ipc *ipc, u8 dma_id, u8 table_id, bool wait); -- 2.17.1

2 4

[alsa-devel] [PATCH v4 1/2] ASoC: Intel: Skylake: Limit large_config_get to single frame
by Cezary Rojewski 08 Aug '19

08 Aug '19

Reply for the very first LARGE_CONFIG_GET request contains total size of payload to be retrieved by host. >From then on, each subsequent reply carries buffer offset instead. As looping is not covered by any real-life example, remove it and cleanup the function for followup overhaul. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/skylake/skl-sst-ipc.c | 37 +++++---------------------- 1 file changed, 7 insertions(+), 30 deletions(-) diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index a2b69a02aab2..196c80dadb1f 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -973,8 +973,7 @@ int skl_ipc_get_large_config(struct sst_generic_ipc *ipc, { struct skl_ipc_header header = {0}; struct sst_ipc_message request = {0}, reply = {0}; - int ret = 0; - size_t sz_remaining, rx_size, data_offset; + int ret; header.primary = IPC_MSG_TARGET(IPC_MOD_MSG); header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); @@ -987,34 +986,12 @@ int skl_ipc_get_large_config(struct sst_generic_ipc *ipc, header.extension |= IPC_FINAL_BLOCK(1); header.extension |= IPC_INITIAL_BLOCK(1); - sz_remaining = msg->param_data_size; - data_offset = 0; - - while (sz_remaining != 0) { - rx_size = sz_remaining > SKL_ADSP_W1_SZ - ? SKL_ADSP_W1_SZ : sz_remaining; - if (rx_size == sz_remaining) - header.extension |= IPC_FINAL_BLOCK(1); - - request.header = *(u64 *)(&header); - reply.data = ((char *)param) + data_offset; - reply.size = msg->param_data_size; - ret = sst_ipc_tx_message_wait(ipc, request, &reply); - if (ret < 0) { - dev_err(ipc->dev, - "ipc: get large config fail, err: %d\n", ret); - return ret; - } - sz_remaining -= rx_size; - data_offset = msg->param_data_size - sz_remaining; - - /* clear the fields */ - header.extension &= IPC_INITIAL_BLOCK_CLEAR; - header.extension &= IPC_DATA_OFFSET_SZ_CLEAR; - /* fill the fields */ - header.extension |= IPC_INITIAL_BLOCK(1); - header.extension |= IPC_DATA_OFFSET_SZ(data_offset); - } + request.header = *(u64 *)(&header); + reply.data = param; + reply.size = msg->param_data_size; + ret = sst_ipc_tx_message_wait(ipc, request, &reply); + if (ret < 0) + dev_err(ipc->dev, "ipc: get large config fail, err: %d\n", ret); return ret; } -- 2.17.1

1 0

[alsa-devel] [PATCH v4 0/2] ASoC: Intel: Skylake: large_config_get overhaul
by Cezary Rojewski 08 Aug '19

08 Aug '19

LARGE_CONFIG_GET is among the most crucial IPCs. Host is expected to send single request first to obtain total payload size from received header's data_off_size. From then on, it loops for each frame exceeding inbox size until entire payload is read. If entire payload is contained within the very first frame, no looping is performed. LARGE_CONFIG_GET is a flexible IPC, it not only receives payload but may carry one with them to provide list of params DSP module should return info for. This behavior is usually reserved for vendors and IPC handler should not touch that content. To achieve that, simply pass provided payload and bytes to sst_ipc_tx_message_wait as a part of request. In current state, LARGE_CONFIG_GET message handler does nothing of that, in consequence making it dysfunctional. Overhaul said handler allowing rightful king of IPCs to return back on his throne - kingdom he shares with his twin brother: LARGE_CONFIG_SET. The looping has not been added in this update as payloads of such size do not exist in practice. Will need to create custom module specifically for that very case and test against it before that feature can be added. Changes since v3: - Split "large_config_get overhaul" patch into two segments as suggested by Pierre: first remove looping, then adjust function's behavior Changes since v2: - None, just for-next rebase Changes since v1: - None, just for-next rebase Cezary Rojewski (2): ASoC: Intel: Skylake: Limit large_config_get to single frame ASoC: Intel: Skylake: large_config_get overhaul sound/soc/intel/skylake/skl-messages.c | 3 +- sound/soc/intel/skylake/skl-sst-ipc.c | 54 +++++++++++--------------- sound/soc/intel/skylake/skl-sst-ipc.h | 3 +- 3 files changed, 27 insertions(+), 33 deletions(-) -- 2.17.1

1 0

[alsa-devel] Regarding alsa snd_pcm_rewind()
by Aishwarya M 08 Aug '19

08 Aug '19

Greetings, I am currently working on an AudioPlayer which can play, pause, seek forward, seek backward the audio using ALSA. For seek forward and backward i would like to know if we can employ alsa's snd_pcm_rewind() and snd_pcm_forward() ? Please could you explain how exactly i can use it for my AudioPlayer application. Looking forward to hearing from you. Thank you. Aishwarya M aishdhan986(a)gmail.com

1 0

[alsa-devel] [PATCH] ALSA: firewire: fix a memory leak bug
by Wenwen Wang 08 Aug '19

08 Aug '19

In iso_packets_buffer_init(), 'b->packets' is allocated through kmalloc_array(). Then, the aligned packet size is checked. If it is larger than PAGE_SIZE, -EINVAL will be returned to indicate the error. However, the allocated 'b->packets' is not deallocated on this path, leading to a memory leak. To fix the above issue, free 'b->packets' before returning the error code. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/firewire/packets-buffer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/packets-buffer.c b/sound/firewire/packets-buffer.c index 0d35359..0ecafd0 100644 --- a/sound/firewire/packets-buffer.c +++ b/sound/firewire/packets-buffer.c @@ -37,7 +37,7 @@ int iso_packets_buffer_init(struct iso_packets_buffer *b, struct fw_unit *unit, packets_per_page = PAGE_SIZE / packet_size; if (WARN_ON(!packets_per_page)) { err = -EINVAL; - goto error; + goto err_packets; } pages = DIV_ROUND_UP(count, packets_per_page); -- 2.7.4

3 2

[alsa-devel] [PATCH] sound: fix a memory leak bug
by Wenwen Wang 08 Aug '19

08 Aug '19

In sound_insert_unit(), the controlling structure 's' is allocated through kmalloc(). Then it is added to the sound driver list by invoking __sound_insert_unit(). Later on, if __register_chrdev() fails, 's' is removed from the list through __sound_remove_unit(). If 'index' is not less than 0, -EBUSY is returned to indicate the error. However, 's' is not deallocated on this execution path, leading to a memory leak bug. To fix the above issue, free 's' before -EBUSY is returned. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/sound_core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index b730d97..90d118c 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -275,7 +275,8 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati goto retry; } spin_unlock(&sound_loader_lock); - return -EBUSY; + r = -EBUSY; + goto fail; } } -- 2.7.4

2 1

[alsa-devel] [asoc:for-5.4 175/179] sound/soc/intel/haswell/sst-haswell-ipc.c:2018:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t {aka unsigned int}'
by kbuild test robot 08 Aug '19

08 Aug '19

tree: https://kernel.googlesource.com/pub/scm/linux/kernel/git/broonie/sound.git for-5.4 head: a860fac420971c5a90d4f78959b44ead793aee4f commit: abf31feea26c0f412a191c83f408311a0de7435c [175/179] ASoC: Intel: Update request-reply IPC model config: i386-allmodconfig (attached as .config) compiler: gcc-7 (Debian 7.4.0-10) 7.4.0 reproduce: git checkout abf31feea26c0f412a191c83f408311a0de7435c # save the attached .config to linux build tree make ARCH=i386 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All warnings (new ones prefixed by >>): In file included from include/linux/printk.h:332:0, from include/linux/kernel.h:15, from sound/soc/intel/haswell/sst-haswell-ipc.c:9: sound/soc/intel/haswell/sst-haswell-ipc.c: In function 'sst_hsw_module_set_param': >> sound/soc/intel/haswell/sst-haswell-ipc.c:2018:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t {aka unsigned int}' [-Wformat=] dev_dbg(dev, "transfer parameter size : %lu\n", ^ include/linux/dynamic_debug.h:125:15: note: in definition of macro '__dynamic_func_call' func(&id, ##__VA_ARGS__); \ ^~~~~~~~~~~ include/linux/dynamic_debug.h:157:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^~~~~~~~~~~~~~~~~~ include/linux/device.h:1503:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^~~~~~~~~~~~~~~ include/linux/device.h:1503:23: note: in expansion of macro 'dev_fmt' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^~~~~~~ >> sound/soc/intel/haswell/sst-haswell-ipc.c:2018:3: note: in expansion of macro 'dev_dbg' dev_dbg(dev, "transfer parameter size : %lu\n", ^~~~~~~ sound/soc/intel/haswell/sst-haswell-ipc.c:2022:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t {aka unsigned int}' [-Wformat=] dev_dbg(dev, "transfer parameter aligned size : %lu\n", ^ include/linux/dynamic_debug.h:125:15: note: in definition of macro '__dynamic_func_call' func(&id, ##__VA_ARGS__); \ ^~~~~~~~~~~ include/linux/dynamic_debug.h:157:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^~~~~~~~~~~~~~~~~~ include/linux/device.h:1503:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^~~~~~~~~~~~~~~ include/linux/device.h:1503:23: note: in expansion of macro 'dev_fmt' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^~~~~~~ sound/soc/intel/haswell/sst-haswell-ipc.c:2022:3: note: in expansion of macro 'dev_dbg' dev_dbg(dev, "transfer parameter aligned size : %lu\n", ^~~~~~~ vim +2018 sound/soc/intel/haswell/sst-haswell-ipc.c 1993 1994 int sst_hsw_module_set_param(struct sst_hsw *hsw, 1995 u32 module_id, u32 instance_id, u32 parameter_id, 1996 u32 param_size, char *param) 1997 { 1998 int ret; 1999 struct sst_ipc_message request = {0}; 2000 u32 payload_size = 0; 2001 struct sst_hsw_transfer_parameter *parameter; 2002 struct device *dev = hsw->dev; 2003 2004 request.header = IPC_GLB_TYPE(IPC_GLB_MODULE_OPERATION) | 2005 IPC_MODULE_OPERATION(IPC_MODULE_SET_PARAMETER) | 2006 IPC_MODULE_ID(module_id); 2007 dev_dbg(dev, "sst_hsw_module_set_param header=%x\n", 2008 (u32)request.header); 2009 2010 payload_size = param_size + 2011 sizeof(struct sst_hsw_transfer_parameter) - 2012 sizeof(struct sst_hsw_transfer_list); 2013 dev_dbg(dev, "parameter size : %d\n", param_size); 2014 dev_dbg(dev, "payload size : %d\n", payload_size); 2015 2016 if (payload_size <= SST_HSW_IPC_MAX_SHORT_PARAMETER_SIZE) { 2017 /* short parameter, mailbox can contain data */ > 2018 dev_dbg(dev, "transfer parameter size : %lu\n", 2019 request.size); 2020 2021 request.size = ALIGN(payload_size, 4); 2022 dev_dbg(dev, "transfer parameter aligned size : %lu\n", 2023 request.size); 2024 2025 parameter = kzalloc(request.size, GFP_KERNEL); 2026 if (parameter == NULL) 2027 return -ENOMEM; 2028 2029 memcpy(parameter->data, param, param_size); 2030 } else { 2031 dev_warn(dev, "transfer parameter size too large!"); 2032 return 0; 2033 } 2034 2035 parameter->parameter_id = parameter_id; 2036 parameter->data_size = param_size; 2037 request.data = parameter; 2038 2039 ret = sst_ipc_tx_message_wait(&hsw->ipc, request, NULL); 2040 if (ret < 0) 2041 dev_err(dev, "ipc: module set parameter failed - %d\n", ret); 2042 2043 kfree(parameter); 2044 2045 return ret; 2046 } 2047 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] Alsa-Plugin 1.0.25 License
by karina filer 08 Aug '19

08 Aug '19

Hi Team, Greetings !! I am facing difficulty to understand the Alsa-Plugin Licensing, Could you please help me to make understand?, Is it under GPL 2.0 or LGPL 2.1 because after extracting the folder I can see 2 copying file one for GPL and other for LGPL. https://repology.org/project/alsa-plugins/packages Regards K. Filer

4 6

[alsa-devel] multi pcm plugin issue
by Giuliano Zannetti - ART S.p.A. 08 Aug '19

08 Aug '19

Hi, I have a performance related problem using the multi PCM plugin. --------------------------------------------------------------------- THE ISSUE --------------------------------------------------------------------- The following scenario triggers the issue. The CPU load is up to 90%. plug - - > upmix - - > multi - - > forward_dmix - - > loopback_dmix pcm.giuliano_plug { type plug slave.pcm giuliano_upmix slave.channels 2 } pcm.giuliano_upmix { type route slave.pcm giuliano_multi slave.channels 4 ttable.0.0 1 ttable.0.1 1 ttable.1.2 1 ttable.1.3 1 } pcm.giuliano_multi { type multi slaves { slave0 { pcm forward_dmix channels 2 } slave1 { pcm loopback_dmix channels 2 } } bindings { 0 {slave slave0 channel 0} 1 {slave slave0 channel 1} 2 {slave slave1 channel 0} 3 {slave slave1 channel 1} } master 0 } pcm.forward_dmix { type dmix ipc_key 1000 ipc_key_add_uid 1 ipc_perm 0600 slave.period_time 10000 slave.period_size 480 slave.buffer_size 1920 slave.rate 48000 slave.pcm "hw:0,0" } pcm.loopback_dmix { type dmix ipc_key 1001 ipc_key_add_uid 1 ipc_perm 0600 slave.period_time 10000 slave.period_size 480 slave.buffer_size 1920 slave.rate 48000 slave.pcm loopback_out } pcm.loopback_out { type hw card Loopback device 0 subdevice 0 } pcm.loopback_dsnoop { type dsnoop ipc_key 1002 ipc_key_add_uid 1 ipc_perm 0600 slave.period_time 10000 slave.period_size 480 slave.buffer_size 1920 slave.rate 48000 slave.pcm loopback_in } pcm.loopback_in { type hw card Loopback device 1 subdevice 0 } The following scenario does not trigger the issue. The CPU load is at most 2%. plug - - > upmix - - > downmix - - > forward_dmix pcm.giuliano_plug { type plug slave.pcm giuliano_upmix slave.channels 2 } pcm.giuliano_upmix { type route slave.pcm giuliano_downmix slave.channels 4 ttable.0.0 1 ttable.0.1 1 ttable.1.2 1 ttable.1.3 1 } pcm.giuliano_downmix { type route slave.pcm forward_dmix slave.channels 2 ttable.0.0 1 ttable.1.1 1 ttable.2.0 1 ttable.3.1 1 } --------------------------------------------------------------------- DEBUG PRINTS IN FUNCTION snd_pcm_write_areas IN FILE pcm.c --------------------------------------------------------------------- For each while iteration the log prints the available frames (avail) and the frames to be written (size). In the multi scenario the several while iterations decrease the performances and the CPU load is high. In the normal scenario i see that at most two iterations of the while are needed. In other words, in the multi scenario the avail frames are not enough to stop the loop. You can see below the prints: avail < size for 28 iterations. Note that this behaviour does not occurs only at startup, but every time during the playback. snd_pcm_sframes_t snd_pcm_write_areas(snd_pcm_t *pcm, const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, snd_pcm_uframes_t size, snd_pcm_xfer_areas_func_t func) { snd_pcm_uframes_t xfer = 0; snd_pcm_sframes_t err = 0; snd_pcm_state_t state; if (size == 0) return 0; int giuliano_i = 0; __snd_pcm_lock(pcm); /* forced lock */ while (size > 0) { snd_pcm_uframes_t frames; snd_pcm_sframes_t avail; _again: state = __snd_pcm_state(pcm); switch (state) { case SND_PCM_STATE_PREPARED: case SND_PCM_STATE_PAUSED: break; case SND_PCM_STATE_RUNNING: err = __snd_pcm_hwsync(pcm); if (err < 0) goto _end; break; default: err = pcm_state_to_error(state); if (!err) err = -EBADFD; goto _end; } avail = __snd_pcm_avail_update(pcm); fprintf( stderr, "giuliano | pcm.c | snd_pcm_write_areas | (%d) avail=%ld, size=%ld\n", giuliano_i, avail, size ); giuliano_i ++; if (avail < 0) { err = avail; goto _end; } if (state == SND_PCM_STATE_RUNNING && size > (snd_pcm_uframes_t)avail) { if (snd_pcm_may_wait_for_avail_min(pcm, avail)) { if (pcm->mode & SND_PCM_NONBLOCK) { err = -EAGAIN; goto _end; } err = snd_pcm_wait_nocheck(pcm, -1); if (err < 0) break; goto _again; } /* the snd_pcm_may_wait_for_avail_min may check against the * updated hw.ptr (slaves), get the avail again here */ avail = __snd_pcm_avail_update(pcm); if (avail < 0) { err = avail; goto _end; } } frames = size; if (frames > (snd_pcm_uframes_t) avail) frames = avail; if (! frames) break; err = func(pcm, areas, offset, frames); if (err < 0) break; frames = err; if (state == SND_PCM_STATE_PREPARED) { snd_pcm_sframes_t hw_avail = pcm->buffer_size - avail; hw_avail += frames; /* some plugins might automatically start the stream */ state = __snd_pcm_state(pcm); if (state == SND_PCM_STATE_PREPARED && hw_avail >= (snd_pcm_sframes_t) pcm->start_threshold) { err = __snd_pcm_start(pcm); if (err < 0) goto _end; } } offset += frames; size -= frames; xfer += frames; } _end: __snd_pcm_unlock(pcm); return xfer > 0 ? (snd_pcm_sframes_t) xfer : snd_pcm_check_error(pcm, err); } --------------------------------------------------------------------- DEBUG PRINTS OUTPUT --------------------------------------------------------------------- ## Multi scenario ## giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=1920 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=1920 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=1920, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=1440 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=1440 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=1440, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=960 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=960, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=480, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (2) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (3) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (4) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (5) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (6) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (7) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (8) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (9) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (10) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (11) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (12) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (13) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (14) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (15) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (16) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (17) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (18) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (19) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (20) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (21) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (22) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (23) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (24) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (25) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (26) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (27) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=1440 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=576 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=1440 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=576 giuliano | pcm.c | snd_pcm_write_areas | (28) avail=576, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=96 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=96, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=96 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 96 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=96 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=96, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=96 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 96 ## Normal scenario ## giuliano | pcm.c | snd_pcm_write_areas | (0) avail=1920, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=1440, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=960, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 --------------------------------------------------------------------- DEBUG PRINTS IN FUNCTION snd_pcm_wait_nocheck IN FILE pcm.c --------------------------------------------------------------------- As you can see in the debug prints above i have enabled the error message in the function snd_pcm_wait_nocheck. That message lead me to think that the issue is a poll related one, may be? int snd_pcm_wait_nocheck(snd_pcm_t *pcm, int timeout) { struct pollfd *pfd; unsigned short revents = 0; int npfds, err, err_poll; npfds = __snd_pcm_poll_descriptors_count(pcm); if (npfds <= 0 || npfds >= 16) { SNDERR("Invalid poll_fds %d\n", npfds); return -EIO; } pfd = alloca(sizeof(*pfd) * npfds); err = __snd_pcm_poll_descriptors(pcm, pfd, npfds); if (err < 0) return err; if (err != npfds) { SNDMSG("invalid poll descriptors %d\n", err); return -EIO; } do { __snd_pcm_unlock(pcm); err_poll = poll(pfd, npfds, timeout); __snd_pcm_lock(pcm); if (err_poll < 0) { if (errno == EINTR && !PCMINABORT(pcm)) continue; return -errno; } if (! err_poll) break; err = __snd_pcm_poll_revents(pcm, pfd, npfds, &revents); if (err < 0) return err; if (revents & (POLLERR | POLLNVAL)) { /* check more precisely */ err = pcm_state_to_error(__snd_pcm_state(pcm)); return err < 0 ? err : -EIO; } } while (!(revents & (POLLIN | POLLOUT))); #if 1 /* very useful code to test poll related problems */ { snd_pcm_sframes_t avail_update; __snd_pcm_hwsync(pcm); avail_update = __snd_pcm_avail_update(pcm); if (avail_update < (snd_pcm_sframes_t)pcm->avail_min) { printf("*** snd_pcm_wait() FATAL ERROR!!!\n"); printf("avail_min = %li, avail_update = %li\n", pcm->avail_min, avail_update); } } #endif return err_poll > 0 ? 1 : 0; } #endif --------------------------------------------------------------------- DEBUG PRINTS IN FUNCTION __snd_pcm_avail_update IN pcm_multi.c --------------------------------------------------------------------- As you can see in the prints above, i put some debugs logs in the function __snd_pcm_avail_update. I can read there the number of avail frames for each slave of the multi. I see also that the minimum numbers of frames is taken among the slaves. static snd_pcm_sframes_t snd_pcm_multi_avail_update(snd_pcm_t *pcm) { snd_pcm_multi_t *multi = pcm->private_data; snd_pcm_sframes_t ret = LONG_MAX; snd_pcm_sframes_t giuliano_temp = 0; unsigned int i; for (i = 0; i < multi->slaves_count; ++i) { snd_pcm_sframes_t avail; avail = snd_pcm_avail_update(multi->slaves[i].pcm); fprintf( stderr, "giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=%d, avail=%d\n", i, avail ); if (avail < 0) return avail; if (ret > avail) ret = avail; } snd_pcm_multi_hwptr_update(pcm); return ret; } I hope that my debug prints are useful to find the issue. Thanks Giuliano Zannetti

1 0

[alsa-devel] (no subject)
by Giuliano Zannetti - ART S.p.A. 08 Aug '19

08 Aug '19

Hi, I have a performance related problem using the multi PCM plugin. --------------------------------------------------------------------- THE ISSUE --------------------------------------------------------------------- The following scenario triggers the issue. The CPU load is up to 90%. plug - - > upmix - - > multi - - > forward_dmix - - > loopback_dmix pcm.giuliano_plug { type plug slave.pcm giuliano_upmix slave.channels 2 } pcm.giuliano_upmix { type route slave.pcm giuliano_multi slave.channels 4 ttable.0.0 1 ttable.0.1 1 ttable.1.2 1 ttable.1.3 1 } pcm.giuliano_multi { type multi slaves { slave0 { pcm forward_dmix channels 2 } slave1 { pcm loopback_dmix channels 2 } } bindings { 0 {slave slave0 channel 0} 1 {slave slave0 channel 1} 2 {slave slave1 channel 0} 3 {slave slave1 channel 1} } master 0 } pcm.forward_dmix { type dmix ipc_key 1000 ipc_key_add_uid 1 ipc_perm 0600 slave.period_time 10000 slave.period_size 480 slave.buffer_size 1920 slave.rate 48000 slave.pcm "hw:0,0" } pcm.loopback_dmix { type dmix ipc_key 1001 ipc_key_add_uid 1 ipc_perm 0600 slave.period_time 10000 slave.period_size 480 slave.buffer_size 1920 slave.rate 48000 slave.pcm loopback_out } pcm.loopback_out { type hw card Loopback device 0 subdevice 0 } pcm.loopback_dsnoop { type dsnoop ipc_key 1002 ipc_key_add_uid 1 ipc_perm 0600 slave.period_time 10000 slave.period_size 480 slave.buffer_size 1920 slave.rate 48000 slave.pcm loopback_in } pcm.loopback_in { type hw card Loopback device 1 subdevice 0 } The following scenario does not trigger the issue. The CPU load is at most 2%. plug - - > upmix - - > downmix - - > forward_dmix pcm.giuliano_plug { type plug slave.pcm giuliano_upmix slave.channels 2 } pcm.giuliano_upmix { type route slave.pcm giuliano_downmix slave.channels 4 ttable.0.0 1 ttable.0.1 1 ttable.1.2 1 ttable.1.3 1 } pcm.giuliano_downmix { type route slave.pcm forward_dmix slave.channels 2 ttable.0.0 1 ttable.1.1 1 ttable.2.0 1 ttable.3.1 1 } --------------------------------------------------------------------- DEBUG PRINTS IN FUNCTION snd_pcm_write_areas IN FILE pcm.c --------------------------------------------------------------------- For each while iteration the log prints the available frames (avail) and the frames to be written (size). In the multi scenario the several while iterations decrease the performances and the CPU load is high. In the normal scenario i see that at most two iterations of the while are needed. In other words, in the multi scenario the avail frames are not enough to stop the loop. You can see below the prints: avail < size for 28 iterations. Note that this behaviour does not occurs only at startup, but every time during the playback. snd_pcm_sframes_t snd_pcm_write_areas(snd_pcm_t *pcm, const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, snd_pcm_uframes_t size, snd_pcm_xfer_areas_func_t func) { snd_pcm_uframes_t xfer = 0; snd_pcm_sframes_t err = 0; snd_pcm_state_t state; if (size == 0) return 0; int giuliano_i = 0; __snd_pcm_lock(pcm); /* forced lock */ while (size > 0) { snd_pcm_uframes_t frames; snd_pcm_sframes_t avail; _again: state = __snd_pcm_state(pcm); switch (state) { case SND_PCM_STATE_PREPARED: case SND_PCM_STATE_PAUSED: break; case SND_PCM_STATE_RUNNING: err = __snd_pcm_hwsync(pcm); if (err < 0) goto _end; break; default: err = pcm_state_to_error(state); if (!err) err = -EBADFD; goto _end; } avail = __snd_pcm_avail_update(pcm); fprintf( stderr, "giuliano | pcm.c | snd_pcm_write_areas | (%d) avail=%ld, size=%ld\n", giuliano_i, avail, size ); giuliano_i ++; if (avail < 0) { err = avail; goto _end; } if (state == SND_PCM_STATE_RUNNING && size > (snd_pcm_uframes_t)avail) { if (snd_pcm_may_wait_for_avail_min(pcm, avail)) { if (pcm->mode & SND_PCM_NONBLOCK) { err = -EAGAIN; goto _end; } err = snd_pcm_wait_nocheck(pcm, -1); if (err < 0) break; goto _again; } /* the snd_pcm_may_wait_for_avail_min may check against the * updated hw.ptr (slaves), get the avail again here */ avail = __snd_pcm_avail_update(pcm); if (avail < 0) { err = avail; goto _end; } } frames = size; if (frames > (snd_pcm_uframes_t) avail) frames = avail; if (! frames) break; err = func(pcm, areas, offset, frames); if (err < 0) break; frames = err; if (state == SND_PCM_STATE_PREPARED) { snd_pcm_sframes_t hw_avail = pcm->buffer_size - avail; hw_avail += frames; /* some plugins might automatically start the stream */ state = __snd_pcm_state(pcm); if (state == SND_PCM_STATE_PREPARED && hw_avail >= (snd_pcm_sframes_t) pcm->start_threshold) { err = __snd_pcm_start(pcm); if (err < 0) goto _end; } } offset += frames; size -= frames; xfer += frames; } _end: __snd_pcm_unlock(pcm); return xfer > 0 ? (snd_pcm_sframes_t) xfer : snd_pcm_check_error(pcm, err); } --------------------------------------------------------------------- DEBUG PRINTS OUTPUT --------------------------------------------------------------------- ## Multi scenario ## giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=1920 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=1920 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=1920, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=1440 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=1440 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=1440, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=960 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=960, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=480, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (2) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (3) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (4) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (5) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (6) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (7) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (8) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (9) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (10) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (11) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (12) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (13) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (14) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 giuliano | pcm.c | snd_pcm_write_areas | (15) avail=0, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=0 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 0 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (16) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (17) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (18) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (19) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (20) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (21) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (22) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (23) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (24) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (25) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (26) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 384 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=384 giuliano | pcm.c | snd_pcm_write_areas | (27) avail=384, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=1440 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=576 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=1440 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=576 giuliano | pcm.c | snd_pcm_write_areas | (28) avail=576, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=96 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=96, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=96 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 96 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=96 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=96, size=480 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=0, avail=960 giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=1, avail=96 *** snd_pcm_wait() FATAL ERROR!!! avail_min = 480, avail_update = 96 ## Normal scenario ## giuliano | pcm.c | snd_pcm_write_areas | (0) avail=1920, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=1440, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=960, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 giuliano | pcm.c | snd_pcm_write_areas | (0) avail=0, size=480 giuliano | pcm.c | snd_pcm_write_areas | (1) avail=480, size=480 --------------------------------------------------------------------- DEBUG PRINTS IN FUNCTION snd_pcm_wait_nocheck IN FILE pcm.c --------------------------------------------------------------------- As you can see in the debug prints above i have enabled the error message in the function snd_pcm_wait_nocheck. That message lead me to think that the issue is a poll related one, may be? int snd_pcm_wait_nocheck(snd_pcm_t *pcm, int timeout) { struct pollfd *pfd; unsigned short revents = 0; int npfds, err, err_poll; npfds = __snd_pcm_poll_descriptors_count(pcm); if (npfds <= 0 || npfds >= 16) { SNDERR("Invalid poll_fds %d\n", npfds); return -EIO; } pfd = alloca(sizeof(*pfd) * npfds); err = __snd_pcm_poll_descriptors(pcm, pfd, npfds); if (err < 0) return err; if (err != npfds) { SNDMSG("invalid poll descriptors %d\n", err); return -EIO; } do { __snd_pcm_unlock(pcm); err_poll = poll(pfd, npfds, timeout); __snd_pcm_lock(pcm); if (err_poll < 0) { if (errno == EINTR && !PCMINABORT(pcm)) continue; return -errno; } if (! err_poll) break; err = __snd_pcm_poll_revents(pcm, pfd, npfds, &revents); if (err < 0) return err; if (revents & (POLLERR | POLLNVAL)) { /* check more precisely */ err = pcm_state_to_error(__snd_pcm_state(pcm)); return err < 0 ? err : -EIO; } } while (!(revents & (POLLIN | POLLOUT))); #if 1 /* very useful code to test poll related problems */ { snd_pcm_sframes_t avail_update; __snd_pcm_hwsync(pcm); avail_update = __snd_pcm_avail_update(pcm); if (avail_update < (snd_pcm_sframes_t)pcm->avail_min) { printf("*** snd_pcm_wait() FATAL ERROR!!!\n"); printf("avail_min = %li, avail_update = %li\n", pcm->avail_min, avail_update); } } #endif return err_poll > 0 ? 1 : 0; } #endif --------------------------------------------------------------------- DEBUG PRINTS IN FUNCTION __snd_pcm_avail_update IN pcm_multi.c --------------------------------------------------------------------- As you can see in the prints above, i put some debugs logs in the function __snd_pcm_avail_update. I can read there the number of avail frames for each slave of the multi. I see also that the minimum numbers of frames is taken among the slaves. static snd_pcm_sframes_t snd_pcm_multi_avail_update(snd_pcm_t *pcm) { snd_pcm_multi_t *multi = pcm->private_data; snd_pcm_sframes_t ret = LONG_MAX; snd_pcm_sframes_t giuliano_temp = 0; unsigned int i; for (i = 0; i < multi->slaves_count; ++i) { snd_pcm_sframes_t avail; avail = snd_pcm_avail_update(multi->slaves[i].pcm); fprintf( stderr, "giuliano | pcm_multi.c | snd_pcm_multi_avail_update | slave=%d, avail=%d\n", i, avail ); if (avail < 0) return avail; if (ret > avail) ret = avail; } snd_pcm_multi_hwptr_update(pcm); return ret; } I hope that my debug prints are useful to find the issue. Thanks Giuliano Zannetti

1 0

[alsa-devel] [PATCH] ASoC: SOF: topology: use set_get_data in process load
by Pierre-Louis Bossart 07 Aug '19

07 Aug '19

From: Jaska Uimonen <jaska.uimonen(a)intel.com> Currently when loading sof process components there's a check if binary control data is associated with it. If found the data is extracted to be part of component loading and initialization. If binary data exceeds the ipc max size, loading fails with error as large message support is only implemented in set_get_data method. So make the process loading use set_get_data to enable large parameters in component initialization. Also refactor the process component loading function as it digs out 3 times almost identical information of related controls. This is redundant, looks ugly and makes it difficult to understand the mechanism. So make a function out of fetching the control data and use it in process loading. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/topology.c | 204 +++++++++++++++++++++++++-------------- 1 file changed, 130 insertions(+), 74 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 12b7d900b9c2..30b5638622dd 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -42,6 +42,13 @@ /* size of tplg abi in byte */ #define SOF_TPLG_ABI_SIZE 3 +struct sof_widget_data { + int ctrl_type; + int ipc_cmd; + struct sof_abi_hdr *pdata; + struct snd_sof_control *control; +}; + /* send pcm params ipc */ static int ipc_pcm_params(struct snd_sof_widget *swidget, int dir) { @@ -1742,51 +1749,32 @@ static int sof_widget_load_siggen(struct snd_soc_component *scomp, int index, return ret; } -static int sof_process_load(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, - struct snd_soc_tplg_dapm_widget *tw, - struct sof_ipc_comp_reply *r, - int type) +static size_t sof_get_control_data(struct snd_sof_dev *sdev, + struct snd_soc_dapm_widget *widget, + struct sof_widget_data *wdata) { - struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); - struct snd_soc_tplg_private *private = &tw->priv; - struct snd_soc_dapm_widget *widget = swidget->widget; const struct snd_kcontrol_new *kc; - struct soc_bytes_ext *sbe; struct soc_mixer_control *sm; + struct soc_bytes_ext *sbe; struct soc_enum *se; - struct snd_sof_control *scontrol = NULL; - struct sof_abi_hdr *pdata = NULL; - struct sof_ipc_comp_process *process; - size_t ipc_size, ipc_data_size = 0; - int ret, i, offset = 0; - - if (type == SOF_COMP_NONE) { - dev_err(sdev->dev, "error: invalid process comp type %d\n", - type); - return -EINVAL; - } + size_t size = 0; + int i; - /* - * get possible component controls - get size of all pdata, - * then memcpy with headers - */ for (i = 0; i < widget->num_kcontrols; i++) { - kc = &widget->kcontrol_news[i]; switch (widget->dobj.widget.kcontrol_type) { case SND_SOC_TPLG_TYPE_MIXER: sm = (struct soc_mixer_control *)kc->private_value; - scontrol = sm->dobj.private; + wdata[i].control = sm->dobj.private; break; case SND_SOC_TPLG_TYPE_BYTES: sbe = (struct soc_bytes_ext *)kc->private_value; - scontrol = sbe->dobj.private; + wdata[i].control = sbe->dobj.private; break; case SND_SOC_TPLG_TYPE_ENUM: se = (struct soc_enum *)kc->private_value; - scontrol = se->dobj.private; + wdata[i].control = se->dobj.private; break; default: dev_err(sdev->dev, "error: unknown kcontrol type %d in widget %s\n", @@ -1795,31 +1783,98 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, return -EINVAL; } - if (!scontrol) { + if (!wdata[i].control) { dev_err(sdev->dev, "error: no scontrol for widget %s\n", widget->name); return -EINVAL; } - /* don't include if no private data */ - pdata = scontrol->control_data->data; - if (!pdata) - continue; + wdata[i].pdata = wdata[i].control->control_data->data; + if (!wdata[i].pdata) + return -EINVAL; /* make sure data is valid - data can be updated at runtime */ - if (pdata->magic != SOF_ABI_MAGIC) - continue; + if (wdata[i].pdata->magic != SOF_ABI_MAGIC) + return -EINVAL; + + size += wdata[i].pdata->size; - ipc_data_size += pdata->size; + /* get data type */ + switch (wdata[i].control->cmd) { + case SOF_CTRL_CMD_VOLUME: + case SOF_CTRL_CMD_ENUM: + case SOF_CTRL_CMD_SWITCH: + wdata[i].ipc_cmd = SOF_IPC_COMP_SET_VALUE; + wdata[i].ctrl_type = SOF_CTRL_TYPE_VALUE_CHAN_SET; + break; + case SOF_CTRL_CMD_BINARY: + wdata[i].ipc_cmd = SOF_IPC_COMP_SET_DATA; + wdata[i].ctrl_type = SOF_CTRL_TYPE_DATA_SET; + break; + default: + break; + } + } + + return size; +} + +static int sof_process_load(struct snd_soc_component *scomp, int index, + struct snd_sof_widget *swidget, + struct snd_soc_tplg_dapm_widget *tw, + struct sof_ipc_comp_reply *r, + int type) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_dapm_widget *widget = swidget->widget; + struct snd_soc_tplg_private *private = &tw->priv; + struct sof_ipc_comp_process *process = NULL; + struct sof_widget_data *wdata = NULL; + size_t ipc_data_size = 0; + size_t ipc_size; + int offset = 0; + int ret = 0; + int i; + + if (type == SOF_COMP_NONE) { + dev_err(sdev->dev, "error: invalid process comp type %d\n", + type); + return -EINVAL; + } + + /* allocate struct for widget control data sizes and types */ + if (widget->num_kcontrols) { + wdata = kcalloc(widget->num_kcontrols, + sizeof(*wdata), + GFP_KERNEL); + + if (!wdata) + return -ENOMEM; + + /* get possible component controls and get size of all pdata */ + ipc_data_size = sof_get_control_data(sdev, widget, wdata); + + if (ipc_data_size <= 0) { + kfree(wdata); + return ipc_data_size; + } } ipc_size = sizeof(struct sof_ipc_comp_process) + le32_to_cpu(private->size) + ipc_data_size; + /* we are exceeding max ipc size, config needs to be sent separately */ + if (ipc_size > SOF_IPC_MSG_MAX_SIZE) { + ipc_size -= ipc_data_size; + ipc_data_size = 0; + } + process = kzalloc(ipc_size, GFP_KERNEL); - if (!process) + if (!process) { + kfree(wdata); return -ENOMEM; + } /* configure iir IPC message */ process->comp.hdr.size = ipc_size; @@ -1835,7 +1890,9 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, if (ret != 0) { dev_err(sdev->dev, "error: parse process.cfg tokens failed %d\n", le32_to_cpu(private->size)); - goto err; + kfree(wdata); + kfree(process); + return ret; } sof_dbg_comp_config(scomp, &process->config); @@ -1845,40 +1902,13 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, * get possible component controls - get size of all pdata, * then memcpy with headers */ - for (i = 0; i < widget->num_kcontrols; i++) { - kc = &widget->kcontrol_news[i]; - - switch (widget->dobj.widget.kcontrol_type) { - case SND_SOC_TPLG_TYPE_MIXER: - sm = (struct soc_mixer_control *)kc->private_value; - scontrol = sm->dobj.private; - break; - case SND_SOC_TPLG_TYPE_BYTES: - sbe = (struct soc_bytes_ext *)kc->private_value; - scontrol = sbe->dobj.private; - break; - case SND_SOC_TPLG_TYPE_ENUM: - se = (struct soc_enum *)kc->private_value; - scontrol = se->dobj.private; - break; - default: - dev_err(sdev->dev, "error: unknown kcontrol type %d in widget %s\n", - widget->dobj.widget.kcontrol_type, - widget->name); - return -EINVAL; + if (ipc_data_size) { + for (i = 0; i < widget->num_kcontrols; i++) { + memcpy(&process->data + offset, + wdata[i].pdata->data, + wdata[i].pdata->size); + offset += wdata[i].pdata->size; } - - /* don't include if no private data */ - pdata = scontrol->control_data->data; - if (!pdata) - continue; - - /* make sure data is valid - data can be updated at runtime */ - if (pdata->magic != SOF_ABI_MAGIC) - continue; - - memcpy(&process->data + offset, pdata->data, pdata->size); - offset += pdata->size; } process->size = ipc_data_size; @@ -1886,10 +1916,36 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, ret = sof_ipc_tx_message(sdev->ipc, process->comp.hdr.cmd, process, ipc_size, r, sizeof(*r)); - if (ret >= 0) + + if (ret < 0) { + dev_err(sdev->dev, "error: create process failed\n"); + kfree(wdata); + kfree(process); return ret; -err: - kfree(process); + } + + /* we sent the data in single message so return */ + if (ipc_data_size) { + kfree(wdata); + return ret; + } + + /* send control data with large message supported method */ + for (i = 0; i < widget->num_kcontrols; i++) { + wdata[i].control->readback_offset = 0; + ret = snd_sof_ipc_set_get_comp_data(sdev->ipc, wdata[i].control, + wdata[i].ipc_cmd, + wdata[i].ctrl_type, + wdata[i].control->cmd, + true); + if (ret != 0) { + dev_err(sdev->dev, "error: send control failed\n"); + kfree(process); + break; + } + } + + kfree(wdata); return ret; } -- 2.20.1

3 3

[alsa-devel] [PATCH] ASoC: hdmi-codec: reorder channel allocation list
by Jonas Karlman 07 Aug '19

07 Aug '19

Wrong channel allocation is selected by hdmi_codec_get_ch_alloc_table_idx(). E.g when ELD reports FL|FR|LFE|FC|RL|RR or FL|FR|LFE|FC|RL|RR|RC|RLC|RRC ca_id 0x01 with speaker mask FL|FR|LFE gets selected instead of ca_id 0x03 with speaker mask FL|FR|LFE|FC for 4 channels and ca_id 0x04 with speaker mask FL|FR|RC gets selected instead of ca_id 0x0b with speaker mask FL|FR|LFE|FC|RL|RR for 6 channels Fix this by reorder the channel allocation list with most specific speaker mask at the top. Signed-off-by: Jonas Karlman <jonas(a)kwiboo.se> --- sound/soc/codecs/hdmi-codec.c | 115 ++++++++++++++++------------------ 1 file changed, 53 insertions(+), 62 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 0bf1c8cad108..9b5a0a419f86 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -188,84 +188,75 @@ static const struct snd_pcm_chmap_elem hdmi_codec_8ch_chmaps[] = { /* * hdmi_codec_channel_alloc: speaker configuration available for CEA * - * This is an ordered list that must match with hdmi_codec_8ch_chmaps struct + * This is an ordered list where ca_id must exist in hdmi_codec_8ch_chmaps * The preceding ones have better chances to be selected by * hdmi_codec_get_ch_alloc_table_idx(). */ static const struct hdmi_codec_cea_spk_alloc hdmi_codec_channel_alloc[] = { { .ca_id = 0x00, .n_ch = 2, - .mask = FL | FR}, - /* 2.1 */ - { .ca_id = 0x01, .n_ch = 4, - .mask = FL | FR | LFE}, - /* Dolby Surround */ + .mask = FL | FR }, + { .ca_id = 0x03, .n_ch = 4, + .mask = FL | FR | LFE | FC }, { .ca_id = 0x02, .n_ch = 4, .mask = FL | FR | FC }, - /* surround51 */ + { .ca_id = 0x01, .n_ch = 4, + .mask = FL | FR | LFE }, { .ca_id = 0x0b, .n_ch = 6, - .mask = FL | FR | LFE | FC | RL | RR}, - /* surround40 */ - { .ca_id = 0x08, .n_ch = 6, - .mask = FL | FR | RL | RR }, - /* surround41 */ - { .ca_id = 0x09, .n_ch = 6, - .mask = FL | FR | LFE | RL | RR }, - /* surround50 */ + .mask = FL | FR | LFE | FC | RL | RR }, { .ca_id = 0x0a, .n_ch = 6, .mask = FL | FR | FC | RL | RR }, - /* 6.1 */ - { .ca_id = 0x0f, .n_ch = 8, - .mask = FL | FR | LFE | FC | RL | RR | RC }, - /* surround71 */ + { .ca_id = 0x09, .n_ch = 6, + .mask = FL | FR | LFE | RL | RR }, + { .ca_id = 0x08, .n_ch = 6, + .mask = FL | FR | RL | RR }, + { .ca_id = 0x07, .n_ch = 6, + .mask = FL | FR | LFE | FC | RC }, + { .ca_id = 0x06, .n_ch = 6, + .mask = FL | FR | FC | RC }, + { .ca_id = 0x05, .n_ch = 6, + .mask = FL | FR | LFE | RC }, + { .ca_id = 0x04, .n_ch = 6, + .mask = FL | FR | RC }, { .ca_id = 0x13, .n_ch = 8, .mask = FL | FR | LFE | FC | RL | RR | RLC | RRC }, - /* others */ - { .ca_id = 0x03, .n_ch = 8, - .mask = FL | FR | LFE | FC }, - { .ca_id = 0x04, .n_ch = 8, - .mask = FL | FR | RC}, - { .ca_id = 0x05, .n_ch = 8, - .mask = FL | FR | LFE | RC }, - { .ca_id = 0x06, .n_ch = 8, - .mask = FL | FR | FC | RC }, - { .ca_id = 0x07, .n_ch = 8, - .mask = FL | FR | LFE | FC | RC }, - { .ca_id = 0x0c, .n_ch = 8, - .mask = FL | FR | RC | RL | RR }, - { .ca_id = 0x0d, .n_ch = 8, - .mask = FL | FR | LFE | RL | RR | RC }, - { .ca_id = 0x0e, .n_ch = 8, - .mask = FL | FR | FC | RL | RR | RC }, - { .ca_id = 0x10, .n_ch = 8, - .mask = FL | FR | RL | RR | RLC | RRC }, - { .ca_id = 0x11, .n_ch = 8, - .mask = FL | FR | LFE | RL | RR | RLC | RRC }, + { .ca_id = 0x1f, .n_ch = 8, + .mask = FL | FR | LFE | FC | RL | RR | FLC | FRC }, { .ca_id = 0x12, .n_ch = 8, .mask = FL | FR | FC | RL | RR | RLC | RRC }, - { .ca_id = 0x14, .n_ch = 8, - .mask = FL | FR | FLC | FRC }, - { .ca_id = 0x15, .n_ch = 8, - .mask = FL | FR | LFE | FLC | FRC }, - { .ca_id = 0x16, .n_ch = 8, - .mask = FL | FR | FC | FLC | FRC }, - { .ca_id = 0x17, .n_ch = 8, - .mask = FL | FR | LFE | FC | FLC | FRC }, - { .ca_id = 0x18, .n_ch = 8, - .mask = FL | FR | RC | FLC | FRC }, - { .ca_id = 0x19, .n_ch = 8, - .mask = FL | FR | LFE | RC | FLC | FRC }, - { .ca_id = 0x1a, .n_ch = 8, - .mask = FL | FR | RC | FC | FLC | FRC }, - { .ca_id = 0x1b, .n_ch = 8, - .mask = FL | FR | LFE | RC | FC | FLC | FRC }, - { .ca_id = 0x1c, .n_ch = 8, - .mask = FL | FR | RL | RR | FLC | FRC }, - { .ca_id = 0x1d, .n_ch = 8, - .mask = FL | FR | LFE | RL | RR | FLC | FRC }, { .ca_id = 0x1e, .n_ch = 8, .mask = FL | FR | FC | RL | RR | FLC | FRC }, - { .ca_id = 0x1f, .n_ch = 8, - .mask = FL | FR | LFE | FC | RL | RR | FLC | FRC }, + { .ca_id = 0x11, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR | RLC | RRC }, + { .ca_id = 0x1d, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR | FLC | FRC }, + { .ca_id = 0x10, .n_ch = 8, + .mask = FL | FR | RL | RR | RLC | RRC }, + { .ca_id = 0x1c, .n_ch = 8, + .mask = FL | FR | RL | RR | FLC | FRC }, + { .ca_id = 0x0f, .n_ch = 8, + .mask = FL | FR | LFE | FC | RL | RR | RC }, + { .ca_id = 0x1b, .n_ch = 8, + .mask = FL | FR | LFE | RC | FC | FLC | FRC }, + { .ca_id = 0x0e, .n_ch = 8, + .mask = FL | FR | FC | RL | RR | RC }, + { .ca_id = 0x1a, .n_ch = 8, + .mask = FL | FR | RC | FC | FLC | FRC }, + { .ca_id = 0x0d, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR | RC }, + { .ca_id = 0x19, .n_ch = 8, + .mask = FL | FR | LFE | RC | FLC | FRC }, + { .ca_id = 0x0c, .n_ch = 8, + .mask = FL | FR | RC | RL | RR }, + { .ca_id = 0x18, .n_ch = 8, + .mask = FL | FR | RC | FLC | FRC }, + { .ca_id = 0x17, .n_ch = 8, + .mask = FL | FR | LFE | FC | FLC | FRC }, + { .ca_id = 0x16, .n_ch = 8, + .mask = FL | FR | FC | FLC | FRC }, + { .ca_id = 0x15, .n_ch = 8, + .mask = FL | FR | LFE | FLC | FRC }, + { .ca_id = 0x14, .n_ch = 8, + .mask = FL | FR | FLC | FRC }, }; struct hdmi_codec_priv { -- 2.17.1

1 0

[alsa-devel] [PATCH] ASoC: rsnd: don't call clk_get_rate() under atomic context
by Kuninori Morimoto 07 Aug '19

07 Aug '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> ADG is using clk_get_rate() under atomic context, thus, we might have scheduling issue. To avoid this issue, we need to get/keep clk rate under non atomic context. We need to handle ADG as special device at Renesas Sound driver. From SW point of view, we want to impletent it as rsnd_mod_ops :: prepare, but it makes code just complicate. To avoid complicated code/patch, this patch adds new clk_rate[] array, and keep clk IN rate when rsnd_adg_clk_enable() was called. Reported-by: Leon Kong <Leon.KONG(a)cn.bosch.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Leon Kong <Leon.KONG(a)cn.bosch.com> --- sound/soc/sh/rcar/adg.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index fce4e05..b9aacf3 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -30,6 +30,7 @@ struct rsnd_adg { struct clk *clkout[CLKOUTMAX]; struct clk_onecell_data onecell; struct rsnd_mod mod; + int clk_rate[CLKMAX]; u32 flags; u32 ckr; u32 rbga; @@ -114,9 +115,9 @@ static void __rsnd_adg_get_timesel_ratio(struct rsnd_priv *priv, unsigned int val, en; unsigned int min, diff; unsigned int sel_rate[] = { - clk_get_rate(adg->clk[CLKA]), /* 0000: CLKA */ - clk_get_rate(adg->clk[CLKB]), /* 0001: CLKB */ - clk_get_rate(adg->clk[CLKC]), /* 0010: CLKC */ + adg->clk_rate[CLKA], /* 0000: CLKA */ + adg->clk_rate[CLKB], /* 0001: CLKB */ + adg->clk_rate[CLKC], /* 0010: CLKC */ adg->rbga_rate_for_441khz, /* 0011: RBGA */ adg->rbgb_rate_for_48khz, /* 0100: RBGB */ }; @@ -302,7 +303,7 @@ int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate) * AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC/AUDIO_CLKI. */ for_each_rsnd_clk(clk, adg, i) { - if (rate == clk_get_rate(clk)) + if (rate == adg->clk_rate[i]) return sel_table[i]; } @@ -369,10 +370,18 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) for_each_rsnd_clk(clk, adg, i) { ret = 0; - if (enable) + if (enable) { ret = clk_prepare_enable(clk); - else + + /* + * We shouldn't use clk_get_rate() under + * atomic context. Let's keep it when + * rsnd_adg_clk_enable() was called + */ + adg->clk_rate[i] = clk_get_rate(adg->clk[i]); + } else { clk_disable_unprepare(clk); + } if (ret < 0) dev_warn(dev, "can't use clk %d\n", i); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: SOF: Intel: hda: Initialize HDA controller after i915 init
by Ranjani Sridharan 07 Aug '19

07 Aug '19

On some platforms, sound card registration fails when a HDMI monitor is not connected. This is caused by a recent commit that switched the order in which the HDA controller and the i915 are initialized. Initializing the i915 before initializing the HDA controller fixes the problem. Fixes: be1b577d01787c ("ASoC: SOF: Intel: hda: fix the hda init chip" Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda.c | 20 +++++++++++++------- 1 file changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 79cce20666b6..724a494702fa 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -335,10 +335,23 @@ static int hda_init_caps(struct snd_sof_dev *sdev) if (bus->ppcap) dev_dbg(sdev->dev, "PP capability, will probe DSP later.\n"); +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + /* init i915 and HDMI codecs */ + ret = hda_codec_i915_init(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: init i915 and HDMI codec failed\n"); + return ret; + } +#endif + + /* Init HDA controller after i915 init */ ret = hda_dsp_ctrl_init_chip(sdev, true); if (ret < 0) { dev_err(bus->dev, "error: init chip failed with ret: %d\n", ret); +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + hda_codec_i915_exit(sdev); +#endif return ret; } @@ -346,13 +359,6 @@ static int hda_init_caps(struct snd_sof_dev *sdev) if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); - /* init i915 and HDMI codecs */ - ret = hda_codec_i915_init(sdev); - if (ret < 0) { - dev_err(sdev->dev, "error: no HDMI audio devices found\n"); - return ret; - } - /* codec detection */ if (!bus->codec_mask) { dev_info(bus->dev, "no hda codecs found!\n"); -- 2.17.1

3 2

[alsa-devel] [PATCH RESEND] ASoC: rt1011: Add R0 temperature and TDM1 ADC2DAT Swap control
by shumingf＠realtek.com 07 Aug '19

07 Aug '19

From: Shuming Fan <shumingf(a)realtek.com> - The user level application could set the R0 temperature after booting system. The degree Celsius of R0 temperature store in the non-volatile space when doing R0 calibration. - TDM1 ADC2DAT Swap controls use to control TDM slot2/3 data Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 9 ++++++++- sound/soc/codecs/rt1011.h | 1 + 2 files changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 0a6ff13d76e1..b25b4c1127ca 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1029,6 +1029,8 @@ static const char * const rt1011_tdm_adc_swap_select[] = { static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc1_1_enum, RT1011_TDM1_SET_3, 6, rt1011_tdm_adc_swap_select); +static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc2_1_enum, RT1011_TDM1_SET_3, 4, + rt1011_tdm_adc_swap_select); static void rt1011_reset(struct regmap *regmap) { @@ -1333,7 +1335,8 @@ static const struct snd_kcontrol_new rt1011_snd_controls[] = { /* TDM1 Data Out Selection */ SOC_ENUM("TDM1 DOUT Source", rt1011_tdm1_adc1_dat_enum), SOC_ENUM("TDM1 DOUT Location", rt1011_tdm1_adc1_loc_enum), - SOC_ENUM("TDM1 ADCDAT Swap Select", rt1011_tdm_adc1_1_enum), + SOC_ENUM("TDM1 ADC1DAT Swap Select", rt1011_tdm_adc1_1_enum), + SOC_ENUM("TDM1 ADC2DAT Swap Select", rt1011_tdm_adc2_1_enum), /* Data Out Mode */ SOC_ENUM("I2S ADC DOUT Mode", rt1011_adc_dout_mode_enum), @@ -1355,6 +1358,10 @@ static const struct snd_kcontrol_new rt1011_snd_controls[] = { SOC_SINGLE_EXT("R0 Calibration", SND_SOC_NOPM, 0, 1, 0, rt1011_r0_cali_get, rt1011_r0_cali_put), RT1011_R0_LOAD("R0 Load Mode"), + + /* R0 temperature */ + SOC_SINGLE("R0 Temperature", RT1011_STP_INITIAL_RESISTANCE_TEMP, + 2, 255, 0), }; static int rt1011_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 98a38800c4df..5d431d7f1d16 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -227,6 +227,7 @@ #define RT1011_STP_CALIB_RS_TEMP 0x152a #define RT1011_INIT_RECIPROCAL_REG_24_16 0x1538 #define RT1011_INIT_RECIPROCAL_REG_15_0 0x1539 +#define RT1011_STP_INITIAL_RESISTANCE_TEMP 0x153c #define RT1011_STP_ALPHA_RECIPROCAL_MSB 0x153e #define RT1011_SPK_RESISTANCE_1 0x1544 #define RT1011_SPK_RESISTANCE_2 0x1546 -- 2.22.0

2 1

[alsa-devel] [PATCH] ASoC: rt1011: Add a flag for the R0 calibration test
by shumingf＠realtek.com 07 Aug '19

07 Aug '19

From: Shuming Fan <shumingf(a)realtek.com> The factory test needs to know whether the calibration completed. This flag helps to confirm the calibration completed or not. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 14 +++++++++++++- sound/soc/codecs/rt1011.h | 2 +- 2 files changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index b25b4c1127ca..638abcaf52b3 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1225,7 +1225,10 @@ static int rt1011_bq_drc_info(struct snd_kcontrol *kcontrol, static int rt1011_r0_cali_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = 0; + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = rt1011->cali_done; return 0; } @@ -1239,6 +1242,7 @@ static int rt1011_r0_cali_put(struct snd_kcontrol *kcontrol, if (!component->card->instantiated) return 0; + rt1011->cali_done = 0; if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF && ucontrol->value.integer.value[0]) rt1011_calibrate(rt1011, 1); @@ -2135,6 +2139,7 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) r0_factor = ((format / r0[0] * 100) / 128) - (r0_integer * 100); rt1011->r0_reg = r0[0]; + rt1011->cali_done = 1; dev_info(dev, "r0 resistance about %d.%02d ohm, reg=0x%X\n", r0_integer, r0_factor, r0[0]); } @@ -2185,6 +2190,13 @@ static void rt1011_calibration_work(struct work_struct *work) rt1011_calibrate(rt1011, 1); + /* + * This flag should reset after booting. + * The factory test will do calibration again and use this flag to check + * whether the calibration completed + */ + rt1011->cali_done = 0; + /* initial */ rt1011_reg_init(component); } diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 5d431d7f1d16..2d65983f3d0f 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -666,7 +666,7 @@ struct rt1011_priv { int pll_out; int bq_drc_set; - unsigned int r0_reg; + unsigned int r0_reg, cali_done; int recv_spk_mode; }; -- 2.22.0

2 1

[alsa-devel] [PATCH] ASoC: SOF: Intel: hda: fix MSI handling
by Pierre-Louis Bossart 07 Aug '19

07 Aug '19

The addition of a kernel module parameter to optionally disable MSI had the side effect of permanently disabling it. The return value of pci_alloc_irq_vectors() is the number of allocated vectors or a negative number on error, so testing with the ! operator is not quite right. It was one optimization too far. Restore previous behavior to use MSI by default, unless the user selects not to do so or the allocation of irq_vectors fails. Fixes: 672ff5e3596ee ('ASoC: SOF: Intel: hda: add a parameter to disable MSI') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 724a494702fa..c7de5065b961 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -543,7 +543,7 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) * if it fails, use legacy interrupt mode * TODO: support msi multiple vectors */ - if (hda_use_msi && !pci_alloc_irq_vectors(pci, 1, 1, PCI_IRQ_MSI)) { + if (hda_use_msi && pci_alloc_irq_vectors(pci, 1, 1, PCI_IRQ_MSI) > 0) { dev_info(sdev->dev, "use msi interrupt mode\n"); hdev->irq = pci_irq_vector(pci, 0); /* ipc irq number is the same of hda irq */ -- 2.20.1

2 1

[alsa-devel] [PATCH v3 0/5] Add support for new SAI IP version
by Daniel Baluta 07 Aug '19

07 Aug '19

So far SAI IPs integrated with imx6 only supported one data line. Starting with imx7 and imx8 SAI integration support up to 8 data lines. First patch introduce register definition to support this. New SAI IP version introduces two new registers (Version and Parmeter registers) which are placed at the beginning of register address space. For this reason we need to fix the register's address. Support for this is introduced in patch 3. Changes since v2: - removed patches regarding data line mask because I need to find a better way to describe to model data lines. Perhaps,we only need to specify how many datalines a specific SAI instance supports and then let SAI driver to activate datalines based on the number of channels. Will open the discussion on this on a separate thread. - fixed devicetree documentation as per Nicolin comments and will send a separate patch to convert it to yaml. Changes since v1: - removed patches from Lucas as they were already accepted - addressed comments from Lucas and Nicolin regarding device tree property naming - removed comment saying that "datalines" must be always consecutively enabled (this is not true, checked with IP owner) - added new patch to document newly introduced compatbile strings - removed patch introducing combined mode as I will still need some time to figure out how to properly allow users to set it. Daniel Baluta (5): ASoC: fsl_sai: Add registers definition for multiple datalines ASoC: fsl_sai: Update Tx/Rx channel enable mask ASoC: fsl_sai: Add support for SAI new version ASoC: fsl_sai: Add support for imx7ulp/imx8mq ASoC: dt-bindings: Introduce compatible strings for 7ULP and 8MQ .../devicetree/bindings/sound/fsl-sai.txt | 3 +- sound/soc/fsl/fsl_sai.c | 320 ++++++++++++------ sound/soc/fsl/fsl_sai.h | 78 +++-- 3 files changed, 273 insertions(+), 128 deletions(-) -- 2.17.1

3 14

[alsa-devel] [PATCH v2 0/2] ASoC: Intel: IPC framework updates
by Cezary Rojewski 07 Aug '19

07 Aug '19

Existing IPC framework omits crucial part of the entire communication: reply header. Some IPCs cannot function at all without the access to said header. Update the sst-ipc with new sst_ipc_message structure to represent both request and reply allowing reply-processing handlers to save received responses. Despite the range of changes required for model to be updated, no functional changes are made for core hanswell, baytrail and skylake message handlers. Reply-processing handlers now save received response header yet no usage is added by default. To allow for future changes, righful kings of IPC kingdom need to be put back on the throne. This update addresses one of them: LARGE_CONFIG_GET. Changes since v1: - Combined first four patches into single one - Rebased onto v5.4 Cezary Rojewski (2): ASoC: Intel: Update request-reply IPC model ASoC: Intel: Skylake: large_config_get overhaul sound/soc/intel/baytrail/sst-baytrail-ipc.c | 65 ++++---- sound/soc/intel/common/sst-ipc.c | 69 ++++---- sound/soc/intel/common/sst-ipc.h | 27 ++-- sound/soc/intel/haswell/sst-haswell-ipc.c | 164 +++++++++++--------- sound/soc/intel/skylake/cnl-sst.c | 6 +- sound/soc/intel/skylake/skl-messages.c | 3 +- sound/soc/intel/skylake/skl-sst-ipc.c | 152 ++++++++++-------- sound/soc/intel/skylake/skl-sst-ipc.h | 3 +- 8 files changed, 260 insertions(+), 229 deletions(-) -- 2.17.1

2 4

[alsa-devel] [PATCH v3] ALSA: hiface: fix multiple memory leak bugs
by Wenwen Wang 07 Aug '19

07 Aug '19

In hiface_pcm_init(), 'rt' is firstly allocated through kzalloc(). Later on, hiface_pcm_init_urb() is invoked to initialize 'rt->out_urbs[i]'. In hiface_pcm_init_urb(), 'rt->out_urbs[i].buffer' is allocated through kzalloc(). However, if hiface_pcm_init_urb() fails, both 'rt' and 'rt->out_urbs[i].buffer' are not deallocated, leading to memory leak bugs. Also, 'rt->out_urbs[i].buffer' is not deallocated if snd_pcm_new() fails. To fix the above issues, free 'rt' and 'rt->out_urbs[i].buffer'. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/usb/hiface/pcm.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 14fc1e1..c406497 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -600,14 +600,13 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) ret = hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP, hiface_pcm_out_urb_handler); if (ret < 0) - return ret; + goto error; } ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm); if (ret < 0) { - kfree(rt); dev_err(&chip->dev->dev, "Cannot create pcm instance\n"); - return ret; + goto error; } pcm->private_data = rt; @@ -620,4 +619,10 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) chip->pcm = rt; return 0; + +error: + for (i = 0; i < PCM_N_URBS; i++) + kfree(rt->out_urbs[i].buffer); + kfree(rt); + return ret; } -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ALSA: pcm: fix multiple memory leak bugs
by Wenwen Wang 07 Aug '19

07 Aug '19

In hiface_pcm_init(), 'rt' is firstly allocated through kzalloc(). Later on, hiface_pcm_init_urb() is invoked to initialize 'rt->out_urbs[i]'. In hiface_pcm_init_urb(), 'rt->out_urbs[i].buffer' is allocated through kzalloc(). However, if hiface_pcm_init_urb() fails, both 'rt' and 'rt->out_urbs[i].buffer' are not deallocated, leading to memory leak bugs. Also, 'rt->out_urbs[i].buffer' is not deallocated if snd_pcm_new() fails. To fix the above issues, free 'rt' and 'rt->out_urbs[i].buffer'. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/usb/hiface/pcm.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 14fc1e1..9b132aa 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -599,12 +599,18 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) for (i = 0; i < PCM_N_URBS; i++) { ret = hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP, hiface_pcm_out_urb_handler); - if (ret < 0) + if (ret < 0) { + for (; i >= 0; i--) + kfree(rt->out_urbs[i].buffer); + kfree(rt); return ret; + } } ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm); if (ret < 0) { + for (i = 0; i < PCM_N_URBS; i++) + kfree(rt->out_urbs[i].buffer); kfree(rt); dev_err(&chip->dev->dev, "Cannot create pcm instance\n"); return ret; -- 2.7.4

2 2

[alsa-devel] [PATCH] ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)
by Takashi Iwai 07 Aug '19

07 Aug '19

A long-time problem on the recent AMD chip (1022:1457) with Realtek codecs (X370, X470, B450, etc) was the crackled or distorted sound for capture streams, as well as occasional playback hiccups. After lengthy debugging sessions, the workaround we've found is like below: - Set up the proper driver caps for this controller, similar as the other AMD controller. - Correct the DMA position reporting with the fixed FIFO size, which is similar like as workaround used for VIA chip set. - Even after the position correction, PA still shows mysterious stalls of playback streams when a capture is triggered in timer-scheduled mode. Since we have no clear way to eliminate the stall, pass the BATCH PCM flag for PA to suppress the tsched mode as a temporary workaround. This patch implements the workarounds. For the driver caps, it defines a new preset, AXZ_DCAPS_PRESET_AMD_SB. It enables the FIFO- corrected position reporting (corresponding to the new position_fix=6) and enforces the SNDRV_PCM_INFO_BATCH flag. Note that the current implementation is merely a workaround. Hopefully we'll find a better alternative in future, especially about removing the BATCH flag hack again. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.c | 7 +++++ sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 63 +++++++++++++++++++++++++++++++++++++++++- 3 files changed, 70 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 2fbdde239936..48d863736b3c 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -613,6 +613,13 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) 20, 178000000); + /* by some reason, the playback stream stalls on PulseAudio with + * tsched=1 when a capture stream triggers. Until we figure out the + * real cause, disable tsched mode by telling the PCM info flag. + */ + if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) + runtime->hw.info |= SNDRV_PCM_INFO_BATCH; + if (chip->align_buffer_size) /* constrain buffer sizes to be multiple of 128 bytes. This is more efficient in terms of memory diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index baa15374fbcb..f2a6df5e6bcb 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -31,7 +31,7 @@ /* 14 unused */ #define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ #define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ -/* 17 unused */ +#define AZX_DCAPS_AMD_WORKAROUND (1 << 17) /* AMD-specific workaround */ #define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ #define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ #define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1e14d7270adf..a6d8c0d77b84 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -64,6 +64,7 @@ enum { POS_FIX_VIACOMBO, POS_FIX_COMBO, POS_FIX_SKL, + POS_FIX_FIFO, }; /* Defines for ATI HD Audio support in SB450 south bridge */ @@ -135,7 +136,7 @@ module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "DMA pointer read method." - "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+)."); + "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+, 6 = FIFO)."); module_param_array(bdl_pos_adj, int, NULL, 0644); MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); @@ -332,6 +333,11 @@ enum { #define AZX_DCAPS_PRESET_ATI_HDMI_NS \ (AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_SNOOP_OFF) +/* quirks for AMD SB */ +#define AZX_DCAPS_PRESET_AMD_SB \ + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_AMD_WORKAROUND |\ + AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME) + /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ (AZX_DCAPS_NO_MSI | AZX_DCAPS_CORBRP_SELF_CLEAR |\ @@ -841,6 +847,49 @@ static unsigned int azx_via_get_position(struct azx *chip, return bound_pos + mod_dma_pos; } +#define AMD_FIFO_SIZE 32 + +/* get the current DMA position with FIFO size correction */ +static unsigned int azx_get_pos_fifo(struct azx *chip, struct azx_dev *azx_dev) +{ + struct snd_pcm_substream *substream = azx_dev->core.substream; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int pos, delay; + + pos = snd_hdac_stream_get_pos_lpib(azx_stream(azx_dev)); + if (!runtime) + return pos; + + runtime->delay = AMD_FIFO_SIZE; + delay = frames_to_bytes(runtime, AMD_FIFO_SIZE); + if (azx_dev->insufficient) { + if (pos < delay) { + delay = pos; + runtime->delay = bytes_to_frames(runtime, pos); + } else { + azx_dev->insufficient = 0; + } + } + + /* correct the DMA position for capture stream */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (pos < delay) + pos += azx_dev->core.bufsize; + pos -= delay; + } + + return pos; +} + +static int azx_get_delay_from_fifo(struct azx *chip, struct azx_dev *azx_dev, + unsigned int pos) +{ + struct snd_pcm_substream *substream = azx_dev->core.substream; + + /* just read back the calculated value in the above */ + return substream->runtime->delay; +} + static unsigned int azx_skl_get_dpib_pos(struct azx *chip, struct azx_dev *azx_dev) { @@ -1417,6 +1466,7 @@ static int check_position_fix(struct azx *chip, int fix) case POS_FIX_VIACOMBO: case POS_FIX_COMBO: case POS_FIX_SKL: + case POS_FIX_FIFO: return fix; } @@ -1433,6 +1483,10 @@ static int check_position_fix(struct azx *chip, int fix) dev_dbg(chip->card->dev, "Using VIACOMBO position fix\n"); return POS_FIX_VIACOMBO; } + if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) { + dev_dbg(chip->card->dev, "Using FIFO position fix\n"); + return POS_FIX_FIFO; + } if (chip->driver_caps & AZX_DCAPS_POSFIX_LPIB) { dev_dbg(chip->card->dev, "Using LPIB position fix\n"); return POS_FIX_LPIB; @@ -1453,6 +1507,7 @@ static void assign_position_fix(struct azx *chip, int fix) [POS_FIX_VIACOMBO] = azx_via_get_position, [POS_FIX_COMBO] = azx_get_pos_lpib, [POS_FIX_SKL] = azx_get_pos_skl, + [POS_FIX_FIFO] = azx_get_pos_fifo, }; chip->get_position[0] = chip->get_position[1] = callbacks[fix]; @@ -1467,6 +1522,9 @@ static void assign_position_fix(struct azx *chip, int fix) azx_get_delay_from_lpib; } + if (fix == POS_FIX_FIFO) + chip->get_delay[0] = chip->get_delay[1] = + azx_get_delay_from_fifo; } /* @@ -2447,6 +2505,9 @@ 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, X370 & co */ + { PCI_DEVICE(0x1022, 0x1457), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, /* AMD Stoney */ { PCI_DEVICE(0x1022, 0x157a), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | -- 2.16.4

1 0

[alsa-devel] [PATCH] ALSA: pcm: fix a memory leak bug
by Wenwen Wang 07 Aug '19

07 Aug '19

In hiface_pcm_init(), 'rt' is firstly allocated through kzalloc(). Later on, hiface_pcm_init_urb() is invoked to initialize 'rt->out_urbs[i]'. However, if the initialization fails, 'rt' is not deallocated, leading to a memory leak bug. To fix the above issue, free 'rt' before returning the error. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/usb/hiface/pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 14fc1e1..5dbcd0d 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -599,8 +599,10 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) for (i = 0; i < PCM_N_URBS; i++) { ret = hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP, hiface_pcm_out_urb_handler); - if (ret < 0) + if (ret < 0) { + kfree(rt); return ret; + } } ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm); -- 2.7.4

2 2

[alsa-devel] [PATCH] ALSA: usb-midi: fix a memory leak bug
by Wenwen Wang 07 Aug '19

07 Aug '19

In __snd_usbmidi_create(), a MIDI streaming interface structure is allocated through kzalloc() and the pointer is saved to 'umidi'. Later on, the endpoint structures are created by invoking snd_usbmidi_create_endpoints_midiman() or snd_usbmidi_create_endpoints(), depending on the type of the audio quirk type. However, if the creation fails, the allocated 'umidi' is not deallocated, leading to a memory leak bug. To fix the above issue, free 'umidi' before returning the error. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/usb/midi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b737f0e..22db37f 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -2476,7 +2476,7 @@ int __snd_usbmidi_create(struct snd_card *card, else err = snd_usbmidi_create_endpoints(umidi, endpoints); if (err < 0) - goto exit; + goto free_midi; usb_autopm_get_interface_no_resume(umidi->iface); -- 2.7.4

2 2

[alsa-devel] [PATCH v2 0/7] Add support for new SAI IP version
by Daniel Baluta 07 Aug '19

07 Aug '19

So far SAI IPs integrated with imx6 only supported one data line. Starting with imx7 and imx8 SAI integration support up to 8 data lines. New SAI IP version introduces two new registers (Version and Parmeter registers) which are placed at the beginning of register address space. For this reason we need to fix the register's address. Changes since v1: - removed patches from Lucas as they were already accepted - addressed comments from Lucas and Nicolin regarding device tree property naming - removed comment saying that "datalines" must be always consecutively enabled (this is not true, checked with IP owner) - added new patch to document newly introduced compatbile strings - removed patch introducing combined mode as I will still need some time to figure out how to properly allow users to set it. Nicolin, Unfortunately I couldn't find any clean solution on handling registers address shifts. As mentioned in patch 5/7 Tx/Rx data registers and Tx/Rx FIFO registers keep their addresses while others are shifted by 8 bytes. Even if I could create two regmaps as suggested I will still need to update each call of regmap_functions. Daniel Baluta (7): ASoC: fsl_sai: Add registers definition for multiple datalines ASoC: fsl_sai: Update Tx/Rx channel enable mask ASoC: fsl_sai: Add support to enable multiple data lines ASoC: dt-bindings: Document dl-mask property ASoC: fsl_sai: Add support for SAI new version ASoC: fsl_sai: Add support for imx7ulp/imx8mq ASoC: dt-bindings: Introduce compatible strings for 7ULP and 8MQ .../devicetree/bindings/sound/fsl-sai.txt | 10 +- sound/soc/fsl/fsl_sai.c | 331 ++++++++++++------ sound/soc/fsl/fsl_sai.h | 82 +++-- 3 files changed, 293 insertions(+), 130 deletions(-) -- 2.17.1

4 23

[alsa-devel] [PATCH 00/28] ASoC: cleanup patches for soc-core
by Kuninori Morimoto 07 Aug '19

07 Aug '19

Hi Mark I want to add multi CPU DAI support to ALSA SoC. But I noticed that we want to cleanup ALSA SoC before that. These patches will do nothing from "technical" point of view. Just for cleanup Kuninori Morimoto (28): ASoC: soc-core: use device_register() ASoC: soc-core: set component->debugfs_root NULL ASoC: soc-core: add comment for for_each_xxx ASoC: soc-core: check return value of snd_soc_add_dai_link() ASoC: soc-core: don't use for_each_card_links_safe() at snd_soc_find_dai_link() ASoC: soc-core: reuse rtdcom at snd_soc_rtdcom_add() ASoC: soc-core: don't check widget for snd_soc_dapm_new_controls() ASoC: soc-core: don't check controls for snd_soc_add_component_controls() ASoC: soc-core: don't check routes for snd_soc_dapm_add_routes() ASoC: soc-core: don't check controls for snd_soc_add_card_controls() ASoC: soc-core: call snd_soc_dapm_debugfs_init() at soc_init_card_debugfs() ASoC: soc-core: remove unneeded list_empty() check for snd_soc_try_rebind_card() ASoC: soc-core: remove verbose debug message from soc_bind_dai_link() ASoC: soc-core: remove duplicated soc_is_dai_link_bound() ASoC: soc-core: tidyup for card->deferred_resume_work ASoC: soc-core: initialize rtd->list ASoC: soc-core: define soc_dpcm_debugfs_add() for non CONFIG_DEBUG_FS ASoC: soc-core: dai_link check under soc_dpcm_debugfs_add() ASoC: soc-core: merge snd_soc_initialize_card_lists() ASoC: soc-core: remove unneeded dai_link check from snd_soc_remove_dai_link() ASoC: soc-core: add NOTE to snd_soc_rtdcom_lookup() ASoC: soc-core: don't call snd_soc_component_set_jack() ASoC: soc-core: don't alloc memory carelessly when creating debugfs ASoC: soc-core: soc_cleanup_card_resources() become void ASoC: soc-core: initialize component list ASoC: soc-core: initialize list at one place ASoC: soc-dai: use bit field for bus_control ASoC: soc-topology: use for_each_component_dais() at remove_dai() include/sound/soc-dai.h | 3 +- include/sound/soc-dpcm.h | 9 ++- include/sound/soc.h | 15 +--- sound/soc/soc-core.c | 196 ++++++++++++++++++++++------------------------- sound/soc/soc-pcm.c | 3 + sound/soc/soc-topology.c | 2 +- 6 files changed, 108 insertions(+), 120 deletions(-) -- 2.7.4

4 45

[alsa-devel] [PATCH] ASoC: Intel: boards: Add Cometlake Dialog Maxim machine driver
by mac.chiang＠intel.com 06 Aug '19

06 Aug '19

From: Mac Chiang <mac.chiang(a)intel.com> enable Cometlake support with: SSP0 for DA7219 headphone codec SSP1 for MAX98357a speaker amp codec Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/Kconfig | 31 +++++++------ sound/soc/intel/boards/kbl_da7219_max98357a.c | 55 ++++++++++++++++++++--- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 12 +++++ 3 files changed, 80 insertions(+), 18 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 50bf149..545aef8 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -322,19 +322,6 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH Say Y or m if you have such a device. This is a recommended option. If unsure select "N". -config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH - tristate "KBL with DA7219 and MAX98357A in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_DA7219 - select SND_SOC_MAX98357A - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI - help - This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for DA7219 + MAX98357A I2S audio codec. - Say Y if you have such a device. - config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH tristate "KBL with DA7219 and MAX98927 in I2S Mode" depends on I2C && ACPI @@ -363,6 +350,24 @@ config SND_SOC_INTEL_KBL_RT5660_MACH endif ## SND_SOC_INTEL_KBL +if SND_SOC_INTEL_KBL || SND_SOC_SOF_COMETLAKE_LP + +config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH + tristate "KBL with DA7219 and MAX98357A in I2S Mode" + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_DA7219 + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC Onboard Codec I2S machine driver. This will + create an alsa sound card for DA7219 + MAX98357A I2S audio codec for + Kabylake/Cometlake platforms. + Say Y if you have such a device. + +endif ## SND_SOC_INTEL_KBL || SND_SOC_SOF_COMETLAKE_LP + if SND_SOC_INTEL_GLK || (SND_SOC_SOF_GEMINILAKE && SND_SOC_SOF_HDA_LINK) config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 537a889..fe3ac70 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -9,6 +9,7 @@ * RT5663 codecs */ +#include <asm/cpu_device_id.h> #include <linux/input.h> #include <linux/module.h> #include <linux/platform_device.h> @@ -17,6 +18,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include "../../codecs/da7219.h" #include "../../codecs/hdac_hdmi.h" #include "../../codecs/da7219-aad.h" @@ -210,7 +212,11 @@ static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) if (!pcm) return -ENOMEM; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + pcm->device = dai->id; +#else pcm->device = device; +#endif pcm->codec_dai = dai; list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); @@ -587,9 +593,17 @@ static struct snd_soc_card kabylake_audio_card_da7219_m98357a = { .late_probe = kabylake_card_late_probe, }; +static const struct x86_cpu_id cml_ids[] = { + { X86_VENDOR_INTEL, 6, 0x8E }, /* Cometlake CPU_ID */ + {} +}; + static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; + struct snd_soc_acpi_mach *mach; + const char *platform_name; + int ret; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -600,17 +614,46 @@ static int kabylake_audio_probe(struct platform_device *pdev) kabylake_audio_card = (struct snd_soc_card *)pdev->id_entry->driver_data; + kabylake_audio_card = &kabylake_audio_card_da7219_m98357a; + kabylake_audio_card->dev = &pdev->dev; snd_soc_card_set_drvdata(kabylake_audio_card, ctx); + + if (x86_match_cpu(cml_ids)) { + unsigned int i; + + kabylake_audio_card->name = "cmlda7219max"; + + for (i = 0; i < ARRAY_SIZE(kabylake_dais); i++) { + /* MAXIM_CODEC is connected to SSP1. */ + if (!strcmp(kabylake_dais[i].cpus->dai_name, + KBL_MAXIM_CODEC_DAI)) { + kabylake_dais[i].name = "SSP1-Codec"; + kabylake_dais[i].cpus->dai_name = "SSP1 Pin"; + } + /* DIALOG_CODEC is connected to SSP0 */ + else if (!strcmp(kabylake_dais[i].cpus->dai_name, + KBL_DIALOG_CODEC_DAI)) { + kabylake_dais[i].name = "SSP0-Codec"; + kabylake_dais[i].cpus->dai_name = "SSP0 Pin"; + } + } + } + + mach = (&pdev->dev)->platform_data; + platform_name = mach->mach_params.platform; + + ret = snd_soc_fixup_dai_links_platform_name(kabylake_audio_card, + platform_name); + if (ret) + return ret; + return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card); } static const struct platform_device_id kbl_board_ids[] = { - { - .name = "kbl_da7219_max98357a", - .driver_data = - (kernel_ulong_t)&kabylake_audio_card_da7219_m98357a, - }, + {.name = "kbl_da7219_max98357a",}, + {.name = "cml_da7219_max98357a",}, { } }; @@ -628,5 +671,7 @@ module_platform_driver(kabylake_audio) /* Module information */ MODULE_DESCRIPTION("Audio Machine driver-DA7219 & MAX98357A in I2S mode"); MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:kbl_da7219_max98357a"); +MODULE_ALIAS("platform:cml_da7219_max98357a"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c index c36c0aa..4ea32b2 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -19,6 +19,11 @@ static struct snd_soc_acpi_codecs cml_codecs = { .codecs = {"10EC5682"} }; +static struct snd_soc_acpi_codecs cml_spk_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { { .id = "INT34C2", @@ -29,6 +34,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { .sof_tplg_filename = "sof-cnl-rt274.tplg", }, { + .id = "DLGS7219", + .drv_name = "cml_da7219_max98357a", + .quirk_data = &cml_spk_codecs, + .sof_fw_filename = "sof-cnl.ri", + .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", + }, + { .id = "MX98357A", .drv_name = "sof_rt5682", .quirk_data = &cml_codecs, -- 2.7.4

3 3

[alsa-devel] [PATCH] ALSA: hda - Don't override global PCM hw info flag
by Takashi Iwai 06 Aug '19

06 Aug '19

The commit bfcba288b97f ("ALSA - hda: Add support for link audio time reporting") introduced the conditional PCM hw info setup, but it overwrites the global azx_pcm_hw object. This will cause a problem if any other HD-audio controller, as it'll inherit the same bit flag although another controller doesn't support that feature. Fix the bug by setting the PCM hw info flag locally. Fixes: bfcba288b97f ("ALSA - hda: Add support for link audio time reporting") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index c8d1b4316245..2fbdde239936 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -598,11 +598,9 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) } runtime->private_data = azx_dev; - if (chip->gts_present) - azx_pcm_hw.info = azx_pcm_hw.info | - SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME; - runtime->hw = azx_pcm_hw; + if (chip->gts_present) + runtime->hw.info |= SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME; runtime->hw.channels_min = hinfo->channels_min; runtime->hw.channels_max = hinfo->channels_max; runtime->hw.formats = hinfo->formats; -- 2.16.4

1 0

[alsa-devel] [GIT PULL] ASoC fixes for v5.3
by Mark Brown 06 Aug '19

06 Aug '19

The following changes since commit 87a6fe80d56be85bda093c4dd8a3e6535bda7e4a: ASoC: SOF: Intel: implement runtime idle for CNL/APL (2019-07-06 12:23:47 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-fix-v5.3-rc3-2 for you to fetch changes up to 7c0767643f3b6b0dd2cda923ae37a18590d431cf: ASoC: max98373: Remove executable bits (2019-08-06 11:15:06 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.3 Incremental fix removing executable bits added in a prior patch accidentally. ---------------------------------------------------------------- Cezary Rojewski (1): MAINTAINERS: Update Intel ASoC drivers maintainers Charles Keepax (1): ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks Cheng-Yi Chiang (1): ASoC: rockchip: Fix mono capture Christophe JAILLET (1): ASoC: Intel: Fix some acpi vs apci typo in somme comments Enric Balletbo i Serra (1): SoC: rockchip: rockchip_max98090: Enable MICBIAS for headset keypress detection Hans de Goede (1): ASoC: Intel: bytcht_es8316: Add quirk for Irbis NB41 netbook Joe Perches (1): ASoC: rt1308: Remove executable attribute from source files Kai Vehmanen (1): ASoC: SOF: Intel: hda: remove misleading error trace from IRQ thread Kirill Marinushkin (1): ASoC: Relocate my e-mail to .com domain zone Kuninori Morimoto (3): ASoC: simple_card_utils.h: care NULL dai at asoc_simple_debug_dai() ASoC: simple-card-utils: care no Platform for DPCM ASoC: audio-graph-card: add missing const at graph_get_dai_id() Marcus Cooper (1): ASoC: sun4i-i2s: Incorrect SR and WSS computation Mark Brown (1): ASoC: max98373: Remove executable bits Masahiro Yamada (1): ASoC: SOF: use __u32 instead of uint32_t in uapi headers Peter Ujfalusi (2): ASoC: ti: davinci-mcasp: Fix clk PDIR handling for i2s master mode ASoC: ti: davinci-mcasp: Correct slot_width posed constraint Ricard Wanderlof (1): ASoC: Fail card instantiation if DAI format setup fails Shengjiu Wang (1): ASoC: cs42xx8: Fix MFREQ selection issue for async mode Shuming Fan (1): ASoC: rt1011: fix DC calibration offset not applying Stephan Gerhold (1): ASoC: qcom: apq8016_sbc: Fix oops with multiple DAI links Tzung-Bi Shih (1): ASoC: max98357a: use mdelay for sdmode-delay Vijendar Mukunda (2): ASoC: amd: acp3x: use dma_ops of parent device for acp3x dma driver ASoC: amd: acp3x: use dma address for acp3x dma driver Wen Yang (6): ASoC: simple-card: fix an use-after-free in simple_dai_link_of_dpcm() ASoC: simple-card: fix an use-after-free in simple_for_each_link() ASoC: audio-graph-card: fix use-after-free in graph_dai_link_of_dpcm() ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id() ASoC: samsung: odroid: fix an use-after-free issue for codec ASoC: samsung: odroid: fix a double-free issue for cpu_dai Wenwen Wang (1): ASoC: dapm: fix a memory leak bug fengchunguo (1): ASoC: max98373: add 88200 and 96000 sampling rate support MAINTAINERS | 3 +- include/sound/simple_card_utils.h | 4 + include/uapi/sound/sof/fw.h | 16 +-- include/uapi/sound/sof/header.h | 14 +-- sound/soc/amd/raven/acp3x-pcm-dma.c | 20 ++-- sound/soc/codecs/cs42xx8.c | 116 +++++++++++++++++---- sound/soc/codecs/max98357a.c | 25 +---- sound/soc/codecs/max98373.c | 6 ++ sound/soc/codecs/max98373.h | 2 + sound/soc/codecs/pcm3060-i2c.c | 4 +- sound/soc/codecs/pcm3060-spi.c | 4 +- sound/soc/codecs/pcm3060.c | 4 +- sound/soc/codecs/pcm3060.h | 2 +- sound/soc/codecs/rt1011.c | 4 +- sound/soc/codecs/rt1308.c | 0 sound/soc/codecs/rt1308.h | 0 sound/soc/generic/audio-graph-card.c | 30 +++--- sound/soc/generic/simple-card-utils.c | 7 ++ sound/soc/generic/simple-card.c | 26 ++--- sound/soc/intel/boards/bytcht_es8316.c | 8 ++ sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-glk-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-hda-match.c | 2 +- .../intel/common/soc-acpi-intel-hsw-bdw-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-skl-match.c | 2 +- sound/soc/qcom/apq8016_sbc.c | 16 +-- sound/soc/rockchip/rockchip_i2s.c | 5 +- sound/soc/rockchip/rockchip_max98090.c | 32 ++++++ sound/soc/samsung/odroid.c | 8 +- sound/soc/soc-core.c | 7 +- sound/soc/soc-dapm.c | 10 +- sound/soc/sof/intel/cnl.c | 4 +- sound/soc/sof/intel/hda-ipc.c | 4 +- sound/soc/sunxi/sun4i-i2s.c | 4 +- sound/soc/ti/davinci-mcasp.c | 46 ++++++-- 40 files changed, 304 insertions(+), 147 deletions(-) mode change 100755 => 100644 sound/soc/codecs/rt1308.c mode change 100755 => 100644 sound/soc/codecs/rt1308.h

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[alsa-devel] Possible bug report
by Harry Bloomberg 06 Aug '19

06 Aug '19

While running an amateur radio application on a Raspberry Pi 3B+ with the stretch version of Raspberrian, I see the following in my syslog: Aug 1 17:31:06 raspberrypi pulseaudio[853]: [alsa-sink-USB Audio] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Aug 1 17:31:06 raspberrypi pulseaudio[853]: [alsa-sink-USB Audio] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Aug 1 17:31:06 raspberrypi pulseaudio[853]: [alsa-sink-USB Audio] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. So...I am reporting this to the developers as instructed by the error message. What additional information do you need? I'm not exactly sure what I was doing to trigger this error. I may have been exiting the application. Thanks, Harry Bloomberg

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[alsa-devel] [PATCH 0/2] ALSA snd-usb-audio: Add Pioneer DDJ-SX3 PCM support
by Ard van Breemen 06 Aug '19

06 Aug '19

The Pioneer lacks the UAC descriptors (added to quircks). It also needs a fix in synchronisation handling for those cases that bSynchAddress == 0 Ard van Breemen (2): Skip bSynchAddress endpoint check if it is invalid. Add Pioneer DDJ-SX3 PCM quirck. sound/usb/pcm.c | 1 + sound/usb/quirks-table.h | 57 ++++++++++++++++++++++++++++++++++++++++ 2 files changed, 58 insertions(+) -- 2.17.1

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[alsa-devel] [PATCH] topology: add support to parse private data for pcm
by Keyon Jie 06 Aug '19

06 Aug '19

We have private data section in struct snd_soc_tplg_pcm, but alsatplg doesn't support handling it yet, here add handling in tplg_parse_pcm() to enable it. Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> --- src/topology/data.c | 3 +++ src/topology/pcm.c | 8 ++++++++ 2 files changed, 11 insertions(+) diff --git a/src/topology/data.c b/src/topology/data.c index b3f4421f..729ce1f4 100644 --- a/src/topology/data.c +++ b/src/topology/data.c @@ -53,6 +53,9 @@ struct snd_soc_tplg_private *get_priv_data(struct tplg_elem *elem) case SND_TPLG_TYPE_BE: priv = &elem->link->priv; break; + case SND_TPLG_TYPE_PCM: + priv = &elem->pcm->priv; + break; default: SNDERR("error: '%s': no support for private data for type %d\n", elem->id, elem->type); diff --git a/src/topology/pcm.c b/src/topology/pcm.c index 5f586dc1..c533ee68 100644 --- a/src/topology/pcm.c +++ b/src/topology/pcm.c @@ -732,6 +732,14 @@ int tplg_parse_pcm(snd_tplg_t *tplg, return err; continue; } + + /* private data */ + if (strcmp(id, "data") == 0) { + err = tplg_parse_data_refs(n, elem); + if (err < 0) + return err; + continue; + } } return 0; -- 2.20.1

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[alsa-devel] [PATCH] ALSA: usb-audio: fix a memory leak bug
by Wenwen Wang 06 Aug '19

06 Aug '19

In snd_usb_get_audioformat_uac3(), a structure for channel maps 'chmap' is allocated through kzalloc() before the execution goto 'found_clock'. However, this structure is not deallocated if the memory allocation for 'pd' fails, leading to a memory leak bug. To fix the above issue, free 'fp->chmap' before returning NULL. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/usb/stream.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 7ee9d17..e852c7f 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1043,6 +1043,7 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, pd = kzalloc(sizeof(*pd), GFP_KERNEL); if (!pd) { + kfree(fp->chmap); kfree(fp->rate_table); kfree(fp); return NULL; -- 2.7.4

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[alsa-devel] [PATCH] slimbus: qcom-ngd-ctrl: Add of_node_put() before return
by Nishka Dasgupta 06 Aug '19

06 Aug '19

Each iteration of for_each_available_child_of_node puts the previous node, but in the case of a return from the middle of the loop, there is no put, thus causing a memory leak. Hence add an of_node_put before the return in two places. Issue found with Coccinelle. Signed-off-by: Nishka Dasgupta <nishkadg.linux(a)gmail.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index f3585777324c..29fbab55c3b3 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1338,12 +1338,15 @@ static int of_qcom_slim_ngd_register(struct device *parent, continue; ngd = kzalloc(sizeof(*ngd), GFP_KERNEL); - if (!ngd) + if (!ngd) { + of_node_put(node); return -ENOMEM; + } ngd->pdev = platform_device_alloc(QCOM_SLIM_NGD_DRV_NAME, id); if (!ngd->pdev) { kfree(ngd); + of_node_put(node); return -ENOMEM; } ngd->id = id; -- 2.19.1

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[alsa-devel] [PATCH v1 0/3] Allow reconfiguration of clock rate
by Jiada Wang 06 Aug '19

06 Aug '19

There is use case in Gstreamer ALSA sink, in case of changed caps Gsreatmer reconfigs and it calls snd_pcm_hw_free() before snd_pcm_prepre(). See gstreamer1.0-plugins-base/ gst-libs/gst/audio/gstaudiobasesink.c: gst_audio_base_sink_setcaps(): - gst_audio_ring_buffer_release() - gst_audio_sink_ring_buffer_release() - gst_alsasink_unprepare() - snd_pcm_hw_free() is called before - gst_audio_ring_buffer_acquire() - gst_audio_sink_ring_buffer_acquire() - gst_alsasink_prepare() - set_hwparams() - snd_pcm_hw_params() - snd_pcm_prepare() But with current implementation after clock rate is started in .prepare reconfiguration of clock rate is not allowed, unless the stream is stopped. This patch set by move stop of clock to .hw_free callback, to allow reconfiguration of clock rate. Jiada Wang (1): ASoC: rsnd: call .hw_{params,free} in pair for same stream Timo Wischer (2): ASoC: rsnd: Support hw_free() callback at DAI level ASoC: rsnd: Allow reconfiguration of clock rate sound/soc/sh/rcar/core.c | 22 +++++++++++++-- sound/soc/sh/rcar/rsnd.h | 36 ++++++++++++++++++++---- sound/soc/sh/rcar/ssi.c | 61 +++++++++++++++++++++++++++++----------- sound/soc/sh/rcar/ssiu.c | 3 +- 4 files changed, 96 insertions(+), 26 deletions(-) -- 2.19.2

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[alsa-devel] [PATCH] ALSA: usb-audio: fix a memory leak bug
by Wenwen Wang 06 Aug '19

06 Aug '19

In add_new_ctl(), a mixer element structure is allocated through kzalloc() and the pointer is saved to 'elem'. Later on, a new alsa control element is created and added to this structure. In case the add process fails, i.e., the return value of snd_usb_mixer_add_control() is less than 0, the allocated structure is not freed, leading to a memory leak. To fix the above issue, free 'elem' before returning the error. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/usb/mixer_scarlett.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index 83715fd..a6c028a 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -562,8 +562,10 @@ static int add_new_ctl(struct usb_mixer_interface *mixer, strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); err = snd_usb_mixer_add_control(&elem->head, kctl); - if (err < 0) + if (err < 0) { + kfree(elem); return err; + } if (elem_ret) *elem_ret = elem; -- 2.7.4

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[alsa-devel] [PATCH v2 00/29] ASoC: add soc-component.c
by Kuninori Morimoto 06 Aug '19

06 Aug '19

Hi Mark These are v2 of soc-component.c patch-set. I want to add multi CPU DAI support to ALSA SoC. But I noticed that we want to cleanup ALSA SoC before that. These patches will do nothing from "technical" point of view, but, will makes code readable. These are focusing to "component". Actually, soc-io.c is also for component. I'm thinking that we can merge soc-io.c into soc-component.c, but do nothing by these patch-set. 1) - 4) : new patch. it removes un-used component functions 5) - 19) : small bug fixed from v1 20) - 22) : more cleanuped 23) - 29) : cares for_each_rtdcom() loop Kuninori Morimoto (29): 1) ASoC: Intel: skl-pcm: disable skl_get_time_info 2) ASoC: soc-pcm: remove soc_rtdcom_ack() 3) ASoC: soc-pcm: remove soc_rtdcom_copy_kernel() 4) ASoC: soc-pcm: remove soc_fill_silence() 5) ASoC: add soc-component.c 6) ASoC: soc-component: add snd_soc_component_get/put() 7) ASoC: soc-component: add snd_soc_component_open() 8) ASoC: soc-component: add snd_soc_component_close() 9) ASoC: soc-component: add snd_soc_component_prepare() 10) ASoC: soc-component: add snd_soc_component_hw_params() 11) ASoC: soc-component: add snd_soc_component_hw_free() 12) ASoC: soc-component: add snd_soc_component_trigger() 13) ASoC: soc-component: add snd_soc_component_suspend() 14) ASoC: soc-component: add snd_soc_component_resume() 15) ASoC: soc-component: add snd_soc_component_is_suspended() 16) ASoC: soc-component: add snd_soc_component_probe() 17) ASoC: soc-component: add snd_soc_component_remove() 18) ASoC: soc-component: add snd_soc_component_of_xlate_dai_id() 19) ASoC: soc-component: add snd_soc_component_of_xlate_dai_name() 20) ASoC: soc-component: move snd_soc_component_seq_notifier() 21) ASoC: soc-component: move snd_soc_component_stream_event() 22) ASoC: soc-component: move snd_soc_component_set_bias_level() 23) ASoC: soc-component: add snd_soc_pcm_component_pointer() 24) ASoC: soc-component: add snd_soc_pcm_component_ioctrl() 25) ASoC: soc-component: add snd_soc_pcm_component_copy_user() 26) ASoC: soc-component: add snd_soc_pcm_component_page() 27) ASoC: soc-component: add snd_soc_pcm_component_mmap() 28) ASoC: soc-component: add snd_soc_pcm_component_pcm_new() 29) ASoC: soc-component: add snd_soc_pcm_component_pcm_free() include/sound/soc-component.h | 387 ++++++++++++++++++++++++++ include/sound/soc-dapm.h | 6 - include/sound/soc.h | 306 +-------------------- sound/soc/Makefile | 2 +- sound/soc/intel/skylake/skl-pcm.c | 3 +- sound/soc/soc-component.c | 561 ++++++++++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 123 ++------- sound/soc/soc-dapm.c | 28 +- sound/soc/soc-jack.c | 18 -- sound/soc/soc-pcm.c | 260 ++---------------- sound/soc/soc-utils.c | 199 -------------- 11 files changed, 1012 insertions(+), 881 deletions(-) create mode 100644 include/sound/soc-component.h create mode 100644 sound/soc/soc-component.c -- 2.7.4

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[alsa-devel] [GIT PULL] ASoC updates for v5.3
by Mark Brown 05 Aug '19

05 Aug '19

The following changes since commit 87a6fe80d56be85bda093c4dd8a3e6535bda7e4a: ASoC: SOF: Intel: implement runtime idle for CNL/APL (2019-07-06 12:23:47 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.3-rc3 for you to fetch changes up to 30c21734d853dae99d05a5295a59b7e26ccd5135: ASoC: amd: acp3x: use dma address for acp3x dma driver (2019-08-02 19:04:43 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.3 A relatively large batch of mostly unremarkable fixes here, a couple of small core fixes for fairly obscure issues, more comment/email updates with no code impact than usual and a bunch of small driver fixes. The support for new sample rates in the max98373 driver is a fix for the fact that the driver declared support for those rates but would in fact return an error if these rates were selected. ---------------------------------------------------------------- Cezary Rojewski (1): MAINTAINERS: Update Intel ASoC drivers maintainers Charles Keepax (1): ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks Cheng-Yi Chiang (1): ASoC: rockchip: Fix mono capture Christophe JAILLET (1): ASoC: Intel: Fix some acpi vs apci typo in somme comments Enric Balletbo i Serra (1): SoC: rockchip: rockchip_max98090: Enable MICBIAS for headset keypress detection Hans de Goede (1): ASoC: Intel: bytcht_es8316: Add quirk for Irbis NB41 netbook Joe Perches (1): ASoC: rt1308: Remove executable attribute from source files Kai Vehmanen (1): ASoC: SOF: Intel: hda: remove misleading error trace from IRQ thread Kirill Marinushkin (1): ASoC: Relocate my e-mail to .com domain zone Kuninori Morimoto (3): ASoC: simple_card_utils.h: care NULL dai at asoc_simple_debug_dai() ASoC: simple-card-utils: care no Platform for DPCM ASoC: audio-graph-card: add missing const at graph_get_dai_id() Marcus Cooper (1): ASoC: sun4i-i2s: Incorrect SR and WSS computation Masahiro Yamada (1): ASoC: SOF: use __u32 instead of uint32_t in uapi headers Peter Ujfalusi (2): ASoC: ti: davinci-mcasp: Fix clk PDIR handling for i2s master mode ASoC: ti: davinci-mcasp: Correct slot_width posed constraint Ricard Wanderlof (1): ASoC: Fail card instantiation if DAI format setup fails Shengjiu Wang (1): ASoC: cs42xx8: Fix MFREQ selection issue for async mode Shuming Fan (1): ASoC: rt1011: fix DC calibration offset not applying Stephan Gerhold (1): ASoC: qcom: apq8016_sbc: Fix oops with multiple DAI links Tzung-Bi Shih (1): ASoC: max98357a: use mdelay for sdmode-delay Vijendar Mukunda (2): ASoC: amd: acp3x: use dma_ops of parent device for acp3x dma driver ASoC: amd: acp3x: use dma address for acp3x dma driver Wen Yang (6): ASoC: simple-card: fix an use-after-free in simple_dai_link_of_dpcm() ASoC: simple-card: fix an use-after-free in simple_for_each_link() ASoC: audio-graph-card: fix use-after-free in graph_dai_link_of_dpcm() ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id() ASoC: samsung: odroid: fix an use-after-free issue for codec ASoC: samsung: odroid: fix a double-free issue for cpu_dai Wenwen Wang (1): ASoC: dapm: fix a memory leak bug fengchunguo (1): ASoC: max98373: add 88200 and 96000 sampling rate support MAINTAINERS | 3 +- include/sound/simple_card_utils.h | 4 + include/uapi/sound/sof/fw.h | 16 +-- include/uapi/sound/sof/header.h | 14 +-- sound/soc/amd/raven/acp3x-pcm-dma.c | 20 ++-- sound/soc/codecs/cs42xx8.c | 116 +++++++++++++++++---- sound/soc/codecs/max98357a.c | 25 +---- sound/soc/codecs/max98373.c | 6 ++ sound/soc/codecs/max98373.h | 2 + sound/soc/codecs/pcm3060-i2c.c | 4 +- sound/soc/codecs/pcm3060-spi.c | 4 +- sound/soc/codecs/pcm3060.c | 4 +- sound/soc/codecs/pcm3060.h | 2 +- sound/soc/codecs/rt1011.c | 4 +- sound/soc/codecs/rt1308.c | 0 sound/soc/codecs/rt1308.h | 0 sound/soc/generic/audio-graph-card.c | 30 +++--- sound/soc/generic/simple-card-utils.c | 7 ++ sound/soc/generic/simple-card.c | 26 ++--- sound/soc/intel/boards/bytcht_es8316.c | 8 ++ sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-glk-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-hda-match.c | 2 +- .../intel/common/soc-acpi-intel-hsw-bdw-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-skl-match.c | 2 +- sound/soc/qcom/apq8016_sbc.c | 16 +-- sound/soc/rockchip/rockchip_i2s.c | 5 +- sound/soc/rockchip/rockchip_max98090.c | 32 ++++++ sound/soc/samsung/odroid.c | 8 +- sound/soc/soc-core.c | 7 +- sound/soc/soc-dapm.c | 10 +- sound/soc/sof/intel/cnl.c | 4 +- sound/soc/sof/intel/hda-ipc.c | 4 +- sound/soc/sunxi/sun4i-i2s.c | 4 +- sound/soc/ti/davinci-mcasp.c | 46 ++++++-- 40 files changed, 304 insertions(+), 147 deletions(-) mode change 100644 => 100755 sound/soc/codecs/max98373.c mode change 100644 => 100755 sound/soc/codecs/max98373.h mode change 100755 => 100644 sound/soc/codecs/rt1308.c mode change 100755 => 100644 sound/soc/codecs/rt1308.h

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[alsa-devel] [PATCH 00/19] ALSA: firewire: introduce AMDTP domain
by Takashi Sakamoto 05 Aug '19

05 Aug '19

Hi, This patchset is to implement AMDTP domain I addressed in my previous patchset: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-July/152430.html My work for libhinoko[1][2] allows me to sniff actual packet transmission between devices and drivers in Windows/Mac OS for long period (e.g. 1 hour). As a result, some supported devices don't follow packet sequences transferred by the drivers. They expect drivers to parse isoc packets in tx stream and recover frequency of data block then transfer isoc packets as rx stream. (I note that no specification describes this mechanism as long as I know. In the specification, clock recovery is one-way from transmitter to receivers, thus recovered clock is not necessarily used for transmission from the receiver to the transmitter.) For the clock recovery in driver side, several isoc contexts should be handled in one time to parse packets in incoming stream and to build packets in outgoing stream. The AMDTP domain is designed for this purpose. In this time, the AMDTP domain implements the functionalities to start/stop a couple of isoc contexts for some AMDTP streams. Each context still runs in own handler for scheduled hardware IRQs. I'll post further work after merged. [1] https://mailman.alsa-project.org/pipermail/alsa-devel/2019-April/147862.html [2] https://github.com/takaswie/libhinoko Takashi Sakamoto (19): ALSA: firewire-lib: add AMDTP domain structure to handle several isoc context in one interrupt callback ALSA: firewire-lib: add a kernel API to stop a couple of AMDTP streams in AMDTP domain ALSA: firewire-lib: add a kernel API to add AMDTP stream into AMDTP domain ALSA: firewire-lib: add a kernel API to start AMDTP streams in AMDTP domain ALSA: fireworks: code refactoring for initialization/destruction of AMDTP streams ALSA: fireworks: code refactoring for bus reset handler ALSA: firewire-digi00x: code refactoring for initialization/destruction of AMDTP stream ALSA: firewire-tascam: code refactoring for initialization/destruction of AMDTP stream ALSA: firewire-motu: code refactoring for initialization/destruction of AMDTP stream ALSA: fireface: code refactoring for initialization/destruction of AMDTP stream ALSA: bebob: support AMDTP domain ALSA: fireworks: support AMDTP domain ALSA: oxfw: support AMDTP domain ALSA: dice: support AMDTP domain ALSA: firewire-digi00x: support AMDTP domain ALSA: firewire-tascam: support AMDTP domain ALSA: firewire-motu: support AMDTP domain ALSA: fireface: support AMDTP domain ALSA: firewire-lib: localize kernel APIs to start/stop each AMDTP stream sound/firewire/amdtp-stream.c | 94 ++++++++++- sound/firewire/amdtp-stream.h | 20 ++- sound/firewire/bebob/bebob.h | 2 + sound/firewire/bebob/bebob_stream.c | 62 +++---- sound/firewire/dice/dice-stream.c | 32 +++- sound/firewire/dice/dice.h | 2 + sound/firewire/digi00x/digi00x-stream.c | 106 +++++++----- sound/firewire/digi00x/digi00x.h | 2 + sound/firewire/fireface/ff-stream.c | 96 ++++++----- sound/firewire/fireface/ff.h | 2 + sound/firewire/fireworks/fireworks.h | 2 + sound/firewire/fireworks/fireworks_stream.c | 176 ++++++++++---------- sound/firewire/motu/motu-stream.c | 131 +++++++-------- sound/firewire/motu/motu.h | 2 + sound/firewire/oxfw/oxfw-stream.c | 72 ++++---- sound/firewire/oxfw/oxfw.h | 2 + sound/firewire/tascam/tascam-stream.c | 123 ++++++++------ sound/firewire/tascam/tascam.h | 2 + 18 files changed, 558 insertions(+), 370 deletions(-) -- 2.20.1

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Re: [alsa-devel] [Sound-open-firmware] Signed firmware availability for kbl/cnl
by Jaroslav Kysela 05 Aug '19

05 Aug '19

Dne 24. 07. 19 v 18:09 Liam Girdwood napsal(a): > On Wed, 2019-07-24 at 18:00 +0200, Jaroslav Kysela wrote: >> Dne 24. 07. 19 v 16:41 Pierre-Louis Bossart napsal(a): >>> On 7/24/19 8:59 AM, Jaroslav Kysela wrote: >>>> Dne 23. 07. 19 v 16:22 Liam Girdwood napsal(a): >>>>> On Thu, 2019-07-18 at 16:43 +0800, Daniel Drake >>>>> wrote:_______________________________________________ >>>>>> Sound-open-firmware mailing list >>>>>> Sound-open-firmware(a)alsa-project.org >>>>>> >>>>> https://mailman.alsa-project.org/mailman/listinfo/sound-open-firmware >>>>> >>>>>> On Wed, Jul 17, 2019 at 8:09 AM Pierre-Louis Bossart >>>>>> <pierre-louis.bossart(a)linux.intel.com> >>>>>> wrote:_______________________________________________ >>>>>>> Sound-open-firmware mailing list >>>>>>> Sound-open-firmware(a)alsa-project.org >>>>>>> >>>>>> https://mailman.alsa-project.org/mailman/listinfo/sound-open-firmware >>>>>> >>>>>>> I was indeed told a while ago that there was a limited >>>>>>> number of >>>>>>> KBL-based devices with DMIC, but mistakenly assumed we >>>>>>> could avoid >>>>>>> dealing with this configuration and Murphy's Law applied of >>>>>>> course. >>>>>>> we'll have to huddle with our Intel colleagues to figure >>>>>>> this one >>>>>>> out. >>>>>> >>>>>> OK, thanks. >>>>>> >>>>>> Is there any update on the release of signed firmware files >>>>>> for the >>>>>> other platforms? We are under pressure to return the other >>>>>> unit we >>>>>> have to the vendor (which needs the cnl files), but we would >>>>>> like to >>>>>> try SOF first. >>>>>> >>>>> >>>>> Apologies for the delay, I hurt my back and was off work for a >>>>> few >>>>> weeks. Signed binaries now on v1.3 github release tag. Will now >>>>> be >>>>> upstreaming into Linux FW repo. >>>> >>>> Liam, the sizes of signed firmware binaries are a lot different >>>> than the >>>> unsigned ones (v1.3 tag) which I can build in docker: >>>> >>>> -rw-rw-r--. 1 perex perex 270336 Jul 24 15:44 sof-apl-signed- >>>> intel.ri >>>> -rw-r--r--. 1 perex perex 167936 Jul 24 15:44 sof-apl.ri >>>> -rw-rw-r--. 1 perex perex 278528 Jul 24 15:44 sof-cnl-signed- >>>> intel.ri >>>> -rw-r--r--. 1 perex perex 172032 Jul 24 15:44 sof-cnl.ri >>>> -rw-rw-r--. 1 perex perex 278528 Jul 24 15:44 sof-icl-signed- >>>> intel.ri >>>> -rw-r--r--. 1 perex perex 172032 Jul 24 15:44 sof-icl.ri >>>> >>>> Is that ok? >>> >>> The firmware used for production is typically built with the >>> Cadence >>> tools, which unfortunately are not available publicly (but can be >>> made >>> available to Intel partners). It wouldn't be surprising if the code >>> size >>> was different due to the use of intrinsics (though 100K seems like >>> a lot >>> indeed). >>> >>> Liam, I think we ought to release binaries with the community key >>> as >>> well so that people can use them as is, e.g. on the Up2 board which >>> does >>> not require the Intel production key. Same for GLK Chromebooks. >> >> It would be probably more nice to create a tar ball with all firmware >> and >> topology files bundled with the proper (usual) filesystem location >> (/lib/firmware/intel/sof/... and /lib/firmware/intel/sof-tplg/...). >> So we >> (users/distribution packagers) can just use it. > > Yeah, been thinking about this atm. It may be better to package the > binaries (firmware and topologies) as part of Linux firmware repo > (since the driver expects to load them all from lib/firmware) and > package the sources (firmware and topology) via sof tarball ? It looks good in my eyes, because topology files are another pieces of the driver from the user space perspective. The unanswered question is the UCM configuration which is linked to the topology configuration (if I understand this correctly). I proposed to place an unique identifier/version to the topology file and propagate this identifier to the user space, so the alsa-lib can pick the right UCM configuration when topology changes. The component string (snd_component_add function / struct snd_ctl_card_info -> components) can be used for this identification. Jaroslav > > Would this be OK for the distros ? > > Liam > >> >> BTW: Do we need UCM config files for SOF, too? >> >> Jaroslav >> > -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH 1/2] ASoC: amd: acp3x: use dma_ops of parent device for acp3x dma driver
by Vijendar Mukunda 02 Aug '19

02 Aug '19

AMD platform device acp3x_rv_i2s created by parent PCI device driver. Pass struct device of the parent to snd_pcm_lib_preallocate_pages() so dma_alloc_coherent() can use correct dma_ops. Otherwise, it will use default dma_ops which is nommu_dma_ops on x86_64 even when IOMMU is enabled and set to non passthrough mode. Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index a4ade6b..905ed2f 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -385,9 +385,11 @@ static snd_pcm_uframes_t acp3x_dma_pointer(struct snd_pcm_substream *substream) static int acp3x_dma_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, + DRV_NAME); + struct device *parent = component->dev->parent; snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, SNDRV_DMA_TYPE_DEV, - rtd->pcm->card->dev, - MIN_BUFFER, MAX_BUFFER); + parent, MIN_BUFFER, MAX_BUFFER); return 0; } -- 2.7.4

2 3

[alsa-devel] [PATCH v2 0/3] ASoC: Codecs: Add TDA7802 codec
by Thomas Preston 02 Aug '19

02 Aug '19

Hi, This patch series adds a driver for the ST TDA7802 amplifier. Thank you Mark Brown, Charles Keepax, Cezary Rojewski and Kirill Marinushkin for your time and useful feedback (on IRC too). Sorry for taking so long in getting back to you, there were quite a lot of changes! Please let me know if there's anything else which needs changing. Many thanks, Thomas Thomas Preston (3): dt-bindings: ASoC: Add TDA7802 amplifier ASoC: Add codec driver for ST TDA7802 ASoC: TDA7802: Add turn-on diagnostic routine .../devicetree/bindings/sound/tda7802.txt | 26 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda7802.c | 693 ++++++++++++++++++ 4 files changed, 727 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tda7802.txt create mode 100644 sound/soc/codecs/tda7802.c -- 2.21.0

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[alsa-devel] Audio issue with chtnau8824
by lukasbf＠tutanota.com 02 Aug '19

02 Aug '19

Hello ALSA devs, I am sorry to annoy but my new Laptop seems to make a strange noise everytime my sound system tries to do something. This is the sound, Warning, probably loud! https://youtu.be/W77nBnDb4dE <https://youtu.be/W77nBnDb4dE> https://youtu.be/lkrz2sDnUcA <https://youtu.be/lkrz2sDnUcA> I suspect it has to do something with the driver/alsa. i have the whole weekend free so i can send diagnostics like this, http://alsa-project.org/db/?f=4aadfe9275a9dc6e4669c8be4301eec11064b7f0 <http://alsa-project.org/db/?f=4aadfe9275a9dc6e4669c8be4301eec11064b7f0> It would be nice to get my soundcard working, maybe i found something important. Thank you for your Work, Lukas Friedrich

1 1

[alsa-devel] [PATCH v1] soundwire: mipi_disco: Switch to use fwnode_property_count_uXX()
by Andy Shevchenko 02 Aug '19

02 Aug '19

Use use fwnode_property_count_uXX() directly, that makes code neater. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- drivers/soundwire/mipi_disco.c | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) diff --git a/drivers/soundwire/mipi_disco.c b/drivers/soundwire/mipi_disco.c index 79fee1b21ab6..844e6b22974f 100644 --- a/drivers/soundwire/mipi_disco.c +++ b/drivers/soundwire/mipi_disco.c @@ -60,8 +60,7 @@ int sdw_master_read_prop(struct sdw_bus *bus) "mipi-sdw-max-clock-frequency", &prop->max_clk_freq); - nval = fwnode_property_read_u32_array(link, - "mipi-sdw-clock-frequencies-supported", NULL, 0); + nval = fwnode_property_count_u32(link, "mipi-sdw-clock-frequencies-supported"); if (nval > 0) { prop->num_clk_freq = nval; prop->clk_freq = devm_kcalloc(bus->dev, prop->num_clk_freq, @@ -87,8 +86,7 @@ int sdw_master_read_prop(struct sdw_bus *bus) } } - nval = fwnode_property_read_u32_array(link, - "mipi-sdw-supported-clock-gears", NULL, 0); + nval = fwnode_property_count_u32(link, "mipi-sdw-supported-clock-gears"); if (nval > 0) { prop->num_clk_gears = nval; prop->clk_gears = devm_kcalloc(bus->dev, prop->num_clk_gears, @@ -134,8 +132,7 @@ static int sdw_slave_read_dp0(struct sdw_slave *slave, fwnode_property_read_u32(port, "mipi-sdw-port-min-wordlength", &dp0->min_word); - nval = fwnode_property_read_u32_array(port, - "mipi-sdw-port-wordlength-configs", NULL, 0); + nval = fwnode_property_count_u32(port, "mipi-sdw-port-wordlength-configs"); if (nval > 0) { dp0->num_words = nval; @@ -193,8 +190,7 @@ static int sdw_slave_read_dpn(struct sdw_slave *slave, fwnode_property_read_u32(node, "mipi-sdw-port-min-wordlength", &dpn[i].min_word); - nval = fwnode_property_read_u32_array(node, - "mipi-sdw-port-wordlength-configs", NULL, 0); + nval = fwnode_property_count_u32(node, "mipi-sdw-port-wordlength-configs"); if (nval > 0) { dpn[i].num_words = nval; dpn[i].words = devm_kcalloc(&slave->dev, @@ -233,8 +229,7 @@ static int sdw_slave_read_dpn(struct sdw_slave *slave, fwnode_property_read_u32(node, "mipi-sdw-max-channel-number", &dpn[i].max_ch); - nval = fwnode_property_read_u32_array(node, - "mipi-sdw-channel-number-list", NULL, 0); + nval = fwnode_property_count_u32(node, "mipi-sdw-channel-number-list"); if (nval > 0) { dpn[i].num_ch = nval; dpn[i].ch = devm_kcalloc(&slave->dev, dpn[i].num_ch, @@ -248,8 +243,7 @@ static int sdw_slave_read_dpn(struct sdw_slave *slave, dpn[i].ch, dpn[i].num_ch); } - nval = fwnode_property_read_u32_array(node, - "mipi-sdw-channel-combination-list", NULL, 0); + nval = fwnode_property_count_u32(node, "mipi-sdw-channel-combination-list"); if (nval > 0) { dpn[i].num_ch_combinations = nval; dpn[i].ch_combinations = devm_kcalloc(&slave->dev, -- 2.20.1

3 2

[alsa-devel] [PATCH v6 49/57] ASoC: Remove dev_err() usage after platform_get_irq()
by Stephen Boyd 02 Aug '19

02 Aug '19

We don't need dev_err() messages when platform_get_irq() fails now that platform_get_irq() prints an error message itself when something goes wrong. Let's remove these prints with a simple semantic patch. // <smpl> @@ expression ret; struct platform_device *E; @@ ret = ( platform_get_irq(E, ...) | platform_get_irq_byname(E, ...) ); if ( $ ret < 0 \| ret <= 0 $ ) { ( -if (ret != -EPROBE_DEFER) -{ ... -dev_err(...); -... } | ... -dev_err(...); ) ... } // </smpl> While we're here, remove braces on if statements that only have one statement (manually). 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: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Stephen Boyd <swboyd(a)chromium.org> --- Please apply directly to subsystem trees sound/soc/atmel/atmel-classd.c | 7 ++----- sound/soc/atmel/atmel-pdmic.c | 7 ++----- sound/soc/bcm/cygnus-ssp.c | 7 ++----- sound/soc/codecs/msm8916-wcd-analog.c | 12 +++--------- sound/soc/codecs/twl6040.c | 4 +--- sound/soc/fsl/fsl_asrc.c | 4 +--- sound/soc/fsl/fsl_esai.c | 4 +--- sound/soc/fsl/fsl_sai.c | 4 +--- sound/soc/fsl/fsl_spdif.c | 4 +--- sound/soc/fsl/fsl_ssi.c | 4 +--- sound/soc/fsl/imx-ssi.c | 4 +--- sound/soc/kirkwood/kirkwood-i2s.c | 4 +--- sound/soc/mediatek/common/mtk-btcvsd.c | 4 +--- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 4 +--- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 4 +--- sound/soc/mxs/mxs-saif.c | 8 ++------ sound/soc/qcom/lpass-platform.c | 5 +---- sound/soc/sof/intel/bdw.c | 5 +---- sound/soc/sof/intel/byt.c | 5 +---- sound/soc/sprd/sprd-mcdt.c | 4 +--- sound/soc/sti/sti_uniperif.c | 4 +--- sound/soc/stm/stm32_i2s.c | 5 +---- sound/soc/stm/stm32_sai.c | 4 +--- sound/soc/stm/stm32_spdifrx.c | 4 +--- sound/soc/sunxi/sun4i-i2s.c | 4 +--- sound/soc/uniphier/aio-dma.c | 4 +--- sound/soc/xilinx/xlnx_formatter_pcm.c | 2 -- sound/soc/xtensa/xtfpga-i2s.c | 1 - 28 files changed, 32 insertions(+), 100 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index 0f2c574f27f1..e98601eccfa3 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -571,11 +571,8 @@ static int atmel_classd_probe(struct platform_device *pdev) dd->pdata = pdata; dd->irq = platform_get_irq(pdev, 0); - if (dd->irq < 0) { - ret = dd->irq; - dev_err(dev, "failed to could not get irq: %d\n", ret); - return ret; - } + if (dd->irq < 0) + return dd->irq; dd->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(dd->pclk)) { diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c index e09c28349e0d..04ec6f0af179 100644 --- a/sound/soc/atmel/atmel-pdmic.c +++ b/sound/soc/atmel/atmel-pdmic.c @@ -612,11 +612,8 @@ static int atmel_pdmic_probe(struct platform_device *pdev) dd->dev = dev; dd->irq = platform_get_irq(pdev, 0); - if (dd->irq < 0) { - ret = dd->irq; - dev_err(dev, "failed to get irq: %d\n", ret); - return ret; - } + if (dd->irq < 0) + return dd->irq; dd->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(dd->pclk)) { diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index b7c358b48d8d..2f9357d7da96 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -1342,11 +1342,8 @@ static int cygnus_ssp_probe(struct platform_device *pdev) } cygaud->irq_num = platform_get_irq(pdev, 0); - if (cygaud->irq_num <= 0) { - dev_err(dev, "platform_get_irq failed\n"); - err = cygaud->irq_num; - return err; - } + if (cygaud->irq_num <= 0) + return cygaud->irq_num; err = audio_clk_init(pdev, cygaud); if (err) { diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 368b6c09474b..667e9f73aba3 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -1185,10 +1185,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) } irq = platform_get_irq_byname(pdev, "mbhc_switch_int"); - if (irq < 0) { - dev_err(dev, "failed to get mbhc switch irq\n"); + if (irq < 0) return irq; - } ret = devm_request_threaded_irq(dev, irq, NULL, pm8916_mbhc_switch_irq_handler, @@ -1200,10 +1198,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) if (priv->mbhc_btn_enabled) { irq = platform_get_irq_byname(pdev, "mbhc_but_press_det"); - if (irq < 0) { - dev_err(dev, "failed to get button press irq\n"); + if (irq < 0) return irq; - } ret = devm_request_threaded_irq(dev, irq, NULL, mbhc_btn_press_irq_handler, @@ -1214,10 +1210,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) dev_err(dev, "cannot request mbhc button press irq\n"); irq = platform_get_irq_byname(pdev, "mbhc_but_rel_det"); - if (irq < 0) { - dev_err(dev, "failed to get button release irq\n"); + if (irq < 0) return irq; - } ret = devm_request_threaded_irq(dev, irq, NULL, mbhc_btn_release_irq_handler, diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 472c2fff34a8..f34637afee51 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1108,10 +1108,8 @@ static int twl6040_probe(struct snd_soc_component *component) priv->component = component; priv->plug_irq = platform_get_irq(pdev, 0); - if (priv->plug_irq < 0) { - dev_err(component->dev, "invalid irq: %d\n", priv->plug_irq); + if (priv->plug_irq < 0) return priv->plug_irq; - } INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work); diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index cbbf6257f08a..cfa40ef6b1ca 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -885,10 +885,8 @@ static int fsl_asrc_probe(struct platform_device *pdev) } irq = platform_get_irq(pdev, 0); - if (irq < 0) { - dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); + if (irq < 0) return irq; - } ret = devm_request_irq(&pdev->dev, irq, fsl_asrc_isr, 0, dev_name(&pdev->dev), asrc_priv); diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 10d2210c91ef..19524ca9b4f1 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -824,10 +824,8 @@ static int fsl_esai_probe(struct platform_device *pdev) PTR_ERR(esai_priv->spbaclk)); irq = platform_get_irq(pdev, 0); - if (irq < 0) { - dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); + if (irq < 0) return irq; - } ret = devm_request_irq(&pdev->dev, irq, esai_isr, 0, esai_priv->name, esai_priv); diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d58cc3ae90d8..6faf222add2b 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -832,10 +832,8 @@ static int fsl_sai_probe(struct platform_device *pdev) } irq = platform_get_irq(pdev, 0); - if (irq < 0) { - dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); + if (irq < 0) return irq; - } ret = devm_request_irq(&pdev->dev, irq, fsl_sai_isr, 0, np->name, sai); if (ret) { diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 4842e6df9a2d..7858a5499ac5 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1248,10 +1248,8 @@ static int fsl_spdif_probe(struct platform_device *pdev) } irq = platform_get_irq(pdev, 0); - if (irq < 0) { - dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); + if (irq < 0) return irq; - } ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0, dev_name(&pdev->dev), spdif_priv); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index fa862af25c1a..b0a6fead1a6a 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1510,10 +1510,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) } ssi->irq = platform_get_irq(pdev, 0); - if (ssi->irq < 0) { - dev_err(dev, "no irq for node %s\n", pdev->name); + if (ssi->irq < 0) return ssi->irq; - } /* Set software limitations for synchronous mode except AC97 */ if (ssi->synchronous && !fsl_ssi_is_ac97(ssi)) { diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 9038b61317be..42031ba7da31 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -520,10 +520,8 @@ static int imx_ssi_probe(struct platform_device *pdev) } ssi->irq = platform_get_irq(pdev, 0); - if (ssi->irq < 0) { - dev_err(&pdev->dev, "Failed to get IRQ: %d\n", ssi->irq); + if (ssi->irq < 0) return ssi->irq; - } ssi->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ssi->clk)) { diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 3446a113f482..bf9aa58b31c6 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -539,10 +539,8 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return PTR_ERR(priv->io); priv->irq = platform_get_irq(pdev, 0); - if (priv->irq < 0) { - dev_err(&pdev->dev, "platform_get_irq failed: %d\n", priv->irq); + if (priv->irq < 0) return priv->irq; - } if (np) { priv->burst = 128; /* might be 32 or 128 */ diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index c7a81c4be068..d00608c73c6e 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -1335,10 +1335,8 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) /* irq */ irq_id = platform_get_irq(pdev, 0); - if (irq_id <= 0) { - dev_err(dev, "%pOFn no irq found\n", dev->of_node); + if (irq_id <= 0) return irq_id < 0 ? irq_id : -ENXIO; - } ret = devm_request_irq(dev, irq_id, mtk_btcvsd_snd_irq_handler, IRQF_TRIGGER_LOW, "BTCVSD_ISR_Handle", diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index 7064a9fd6f74..9af76ae315a5 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -1342,10 +1342,8 @@ static int mt2701_afe_pcm_dev_probe(struct platform_device *pdev) return -ENOMEM; irq_id = platform_get_irq_byname(pdev, "asys"); - if (irq_id < 0) { - dev_err(dev, "unable to get ASYS IRQ\n"); + if (irq_id < 0) return irq_id; - } ret = devm_request_irq(dev, irq_id, mt2701_asys_isr, IRQF_TRIGGER_NONE, "asys-isr", (void *)afe); diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index 0382896c162e..2d45d44ee97b 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -1075,10 +1075,8 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) afe->dev = &pdev->dev; irq_id = platform_get_irq(pdev, 0); - if (irq_id <= 0) { - dev_err(afe->dev, "np %pOFn no irq\n", afe->dev->of_node); + if (irq_id <= 0) return irq_id < 0 ? irq_id : -ENXIO; - } ret = devm_request_irq(afe->dev, irq_id, mt8173_afe_irq_handler, 0, "Afe_ISR_Handle", (void *)afe); if (ret) { diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 269b6d6df250..ccfec47f63e8 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -793,12 +793,8 @@ static int mxs_saif_probe(struct platform_device *pdev) return PTR_ERR(saif->base); irq = platform_get_irq(pdev, 0); - if (irq < 0) { - ret = irq; - dev_err(&pdev->dev, "failed to get irq resource: %d\n", - ret); - return ret; - } + if (irq < 0) + return irq; saif->dev = &pdev->dev; ret = devm_request_irq(&pdev->dev, irq, mxs_saif_irq, 0, diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index cf7a299f4547..4c745baa39f7 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -564,11 +564,8 @@ int asoc_qcom_lpass_platform_register(struct platform_device *pdev) int ret; drvdata->lpaif_irq = platform_get_irq_byname(pdev, "lpass-irq-lpaif"); - if (drvdata->lpaif_irq < 0) { - dev_err(&pdev->dev, "error getting irq handle: %d\n", - drvdata->lpaif_irq); + if (drvdata->lpaif_irq < 0) return -ENODEV; - } /* ensure audio hardware is disabled */ ret = regmap_write(drvdata->lpaif_map, diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index 70d524ef9bc0..4bb9636da990 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -613,11 +613,8 @@ static int bdw_probe(struct snd_sof_dev *sdev) /* register our IRQ */ sdev->ipc_irq = platform_get_irq(pdev, desc->irqindex_host_ipc); - if (sdev->ipc_irq < 0) { - dev_err(sdev->dev, "error: failed to get IRQ at index %d\n", - desc->irqindex_host_ipc); + if (sdev->ipc_irq < 0) return sdev->ipc_irq; - } dev_dbg(sdev->dev, "using IRQ %d\n", sdev->ipc_irq); ret = devm_request_threaded_irq(sdev->dev, sdev->ipc_irq, diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index 107d711efc3f..000d576f6a8d 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -728,11 +728,8 @@ static int byt_acpi_probe(struct snd_sof_dev *sdev) irq: /* register our IRQ */ sdev->ipc_irq = platform_get_irq(pdev, desc->irqindex_host_ipc); - if (sdev->ipc_irq < 0) { - dev_err(sdev->dev, "error: failed to get IRQ at index %d\n", - desc->irqindex_host_ipc); + if (sdev->ipc_irq < 0) return sdev->ipc_irq; - } dev_dbg(sdev->dev, "using IRQ %d\n", sdev->ipc_irq); ret = devm_request_threaded_irq(sdev->dev, sdev->ipc_irq, diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c index 7448015a4935..f439e5503a3c 100644 --- a/sound/soc/sprd/sprd-mcdt.c +++ b/sound/soc/sprd/sprd-mcdt.c @@ -959,10 +959,8 @@ static int sprd_mcdt_probe(struct platform_device *pdev) platform_set_drvdata(pdev, mcdt); irq = platform_get_irq(pdev, 0); - if (irq < 0) { - dev_err(&pdev->dev, "Failed to get MCDT interrupt\n"); + if (irq < 0) return irq; - } ret = devm_request_irq(&pdev->dev, irq, sprd_mcdt_irq_handler, 0, "sprd-mcdt", mcdt); diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index 645bcbe91601..ee4a0151e63e 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -426,10 +426,8 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node, UNIPERIF_FIFO_DATA_OFFSET(uni); uni->irq = platform_get_irq(priv->pdev, 0); - if (uni->irq < 0) { - dev_err(dev, "Failed to get IRQ resource\n"); + if (uni->irq < 0) return -ENXIO; - } uni->type = dev_data->type; diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index ba6452dab69b..3e7226a53e53 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -855,11 +855,8 @@ static int stm32_i2s_parse_dt(struct platform_device *pdev, /* Get irqs */ irq = platform_get_irq(pdev, 0); - if (irq < 0) { - if (irq != -EPROBE_DEFER) - dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); + if (irq < 0) return irq; - } ret = devm_request_irq(&pdev->dev, irq, stm32_i2s_isr, IRQF_ONESHOT, dev_name(&pdev->dev), i2s); diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index 63f68e663676..1c4cc1ca1eaa 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -195,10 +195,8 @@ static int stm32_sai_probe(struct platform_device *pdev) /* init irqs */ sai->irq = platform_get_irq(pdev, 0); - if (sai->irq < 0) { - dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); + if (sai->irq < 0) return sai->irq; - } /* reset */ rst = devm_reset_control_get_exclusive(&pdev->dev, NULL); diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index ee71b898897b..cd4b235fce57 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -909,10 +909,8 @@ static int stm32_spdifrx_parse_of(struct platform_device *pdev, } spdifrx->irq = platform_get_irq(pdev, 0); - if (spdifrx->irq < 0) { - dev_err(&pdev->dev, "No irq for node %s\n", pdev->name); + if (spdifrx->irq < 0) return spdifrx->irq; - } return 0; } diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 9b2232908b65..d97d694c48df 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1087,10 +1087,8 @@ static int sun4i_i2s_probe(struct platform_device *pdev) return PTR_ERR(regs); irq = platform_get_irq(pdev, 0); - if (irq < 0) { - dev_err(&pdev->dev, "Can't retrieve our interrupt\n"); + if (irq < 0) return irq; - } i2s->variant = of_device_get_match_data(&pdev->dev); if (!i2s->variant) { diff --git a/sound/soc/uniphier/aio-dma.c b/sound/soc/uniphier/aio-dma.c index fa001d3c1a88..2261655be10d 100644 --- a/sound/soc/uniphier/aio-dma.c +++ b/sound/soc/uniphier/aio-dma.c @@ -291,10 +291,8 @@ int uniphier_aiodma_soc_register_platform(struct platform_device *pdev) return PTR_ERR(chip->regmap); irq = platform_get_irq(pdev, 0); - if (irq < 0) { - dev_err(dev, "Could not get irq.\n"); + if (irq < 0) return irq; - } ret = devm_request_irq(dev, irq, aiodma_irq, IRQF_SHARED, dev_name(dev), pdev); diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index dc8721f4f56b..48970efe7838 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -613,7 +613,6 @@ static int xlnx_formatter_pcm_probe(struct platform_device *pdev) aud_drv_data->mm2s_irq = platform_get_irq_byname(pdev, "irq_mm2s"); if (aud_drv_data->mm2s_irq < 0) { - dev_err(dev, "xlnx audio mm2s irq resource failed\n"); ret = aud_drv_data->mm2s_irq; goto clk_err; } @@ -640,7 +639,6 @@ static int xlnx_formatter_pcm_probe(struct platform_device *pdev) aud_drv_data->s2mm_irq = platform_get_irq_byname(pdev, "irq_s2mm"); if (aud_drv_data->s2mm_irq < 0) { - dev_err(dev, "xlnx audio s2mm irq resource failed\n"); ret = aud_drv_data->s2mm_irq; goto clk_err; } diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c index 9ce2c75186b9..e635bf08d935 100644 --- a/sound/soc/xtensa/xtfpga-i2s.c +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -572,7 +572,6 @@ static int xtfpga_i2s_probe(struct platform_device *pdev) irq = platform_get_irq(pdev, 0); if (irq < 0) { - dev_err(&pdev->dev, "No IRQ resource\n"); err = irq; goto err; } -- Sent by a computer through tubes

2 1

[alsa-devel] [PATCH] ALSA: hda: Add support of Zhaoxin controller
by Tony W Wang-oc 02 Aug '19

02 Aug '19

Add the new PCI ID 0x1d17 0x3288 Zhaoxin controller support Signed-off-by: Tony W Wang-oc <TonyWWang-oc(a)zhaoxin.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 324a4b2..d08da0e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -266,6 +266,7 @@ enum { AZX_DRIVER_CTX, AZX_DRIVER_CTHDA, AZX_DRIVER_CMEDIA, + AZX_DRIVER_ZHAOXIN, AZX_DRIVER_GENERIC, AZX_NUM_DRIVERS, /* keep this as last entry */ }; @@ -379,6 +380,7 @@ static char *driver_short_names[] = { [AZX_DRIVER_CTX] = "HDA Creative", [AZX_DRIVER_CTHDA] = "HDA Creative", [AZX_DRIVER_CMEDIA] = "HDA C-Media", + [AZX_DRIVER_ZHAOXIN] = "HDA Zhaoxin", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; @@ -2589,6 +2591,8 @@ static const struct pci_device_id azx_ids[] = { .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, .class_mask = 0xffffff, .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI }, + /* Zhaoxin */ + { PCI_DEVICE(0x1d17, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: isa/wavefront: remove redundant assignment to pointer bptr
by Colin King 02 Aug '19

02 Aug '19

From: Colin Ian King <colin.king(a)canonical.com> The pointer bptr is being assigned a value that is never read and it is being updated in the next statement with a new value. The initialization is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/isa/wavefront/wavefront_synth.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index aec1c46e6697..c5b1d5900eed 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -788,7 +788,6 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header) dev->patch_status[header->number] |= WF_SLOT_FILLED; - bptr = buf; bptr = munge_int32 (header->number, buf, 2); munge_buf ((unsigned char *)&header->hdr.p, bptr, WF_PATCH_BYTES); -- 2.20.1

2 1

[alsa-devel] [PATCH v2 0/6] ASoC: improve codec to codec link support
by Jerome Brunet 01 Aug '19

01 Aug '19

As explained in this previous series [0], on Amlogic, we are using codec to codec links to deal with the glue which is between the i2s backends and the synopsys hdmi controller. This worked well until I tried to .get_eld() support in the dw-hdmi-i2s driver. Doing so adds channel mapping controls to the codec. This shown several problem 1) .pcm_new() is not called on codec to codec links. struct snd_soc_pcm_runtime do not even have a valid .pcm 2) struct snd_pcm_substream and struct snd_pcm_runtime are ephemeral This is a problem if a control needs to access them The goal of this patchset is to resolve the above issues and improve the codec to codec link support enough to correctly handle the hdmi-codec The support of these codec to codec links could probably be improved in the future to behave like any other link and use soc_pcm_open(), soc_pcm_hw_params(), etc... The challenge lies in the dapm mutex. The soc_pcm call dapm function locking this mutex but the dapm mutex is already held in snd_soc_dai_link_event() Changes since v1 [1]: * Fix rebase on Murimoto-san's patches * Allocate params dynamically again and refactor the pre_pmu code to simplify the error handling and rollback a bit [0]: https://lkml.kernel.org/r/20190515131858.32130-1-jbrunet@baylibre.com [1]: https://lkml.kernel.org/r/20190724162405.6574-1-jbrunet@baylibre.com Jerome Brunet (6): ASoC: codec2codec: run callbacks in order ASoC: codec2codec: name link using stream direction ASoC: codec2codec: deal with params when necessary ASoC: create pcm for codec2codec links as well ASoC: codec2codec: remove ephemeral variables ASoC: codec2codec: fill some of the runtime stream parameters sound/soc/soc-core.c | 42 +++------ sound/soc/soc-dapm.c | 220 +++++++++++++++++++++++++++---------------- sound/soc/soc-pcm.c | 35 ++++++- 3 files changed, 182 insertions(+), 115 deletions(-) -- 2.21.0

2 12

[alsa-devel] [PATCH v2 1/3] ASoC: Intel: SoC: skylake: no need to check return value of debugfs_create functions
by Greg Kroah-Hartman 01 Aug '19

01 Aug '19

When calling debugfs functions, there is no need to ever check the return value. The function can work or not, but the code logic should never do something different based on this. Also, if a debugfs call fails, userspace is notified with an error in the log, so no need to log the error again. Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Jie Yang <yang.jie(a)linux.intel.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 Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- v2: rebase on 5.3-rc1 change Subject line to match the subsystem better rework based on debugfs core now reporting errors. sound/soc/intel/skylake/skl-debug.c | 28 +++++----------------------- 1 file changed, 5 insertions(+), 23 deletions(-) diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c index b9b4a72a4334..1d6240ad6112 100644 --- a/sound/soc/intel/skylake/skl-debug.c +++ b/sound/soc/intel/skylake/skl-debug.c @@ -162,10 +162,8 @@ void skl_debug_init_module(struct skl_debug *d, struct snd_soc_dapm_widget *w, struct skl_module_cfg *mconfig) { - if (!debugfs_create_file(w->name, 0444, - d->modules, mconfig, - &mcfg_fops)) - dev_err(d->dev, "%s: module debugfs init failed\n", w->name); + debugfs_create_file(w->name, 0444, d->modules, mconfig, + &mcfg_fops); } static ssize_t fw_softreg_read(struct file *file, char __user *user_buf, @@ -222,34 +220,18 @@ struct skl_debug *skl_debugfs_init(struct skl *skl) return NULL; /* create the debugfs dir with platform component's debugfs as parent */ - d->fs = debugfs_create_dir("dsp", - skl->component->debugfs_root); - if (IS_ERR(d->fs) || !d->fs) { - dev_err(&skl->pci->dev, "debugfs root creation failed\n"); - return NULL; - } + d->fs = debugfs_create_dir("dsp", skl->component->debugfs_root); d->skl = skl; d->dev = &skl->pci->dev; /* now create the module dir */ d->modules = debugfs_create_dir("modules", d->fs); - if (IS_ERR(d->modules) || !d->modules) { - dev_err(&skl->pci->dev, "modules debugfs create failed\n"); - goto err; - } - if (!debugfs_create_file("fw_soft_regs_rd", 0444, d->fs, d, - &soft_regs_ctrl_fops)) { - dev_err(d->dev, "fw soft regs control debugfs init failed\n"); - goto err; - } + debugfs_create_file("fw_soft_regs_rd", 0444, d->fs, d, + &soft_regs_ctrl_fops); return d; - -err: - debugfs_remove_recursive(d->fs); - return NULL; } void skl_debugfs_exit(struct skl *skl) -- 2.22.0

3 6

[alsa-devel] [PATCH] ASoC: zx-tdm: remove redundant assignment to ts_width on error return path
by Colin King 01 Aug '19

01 Aug '19

From: Colin Ian King <colin.king(a)canonical.com> The value assigned to ts_width is never read on the error return path so the assignment is redundant and can be removed. Remove it. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/zte/zx-tdm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c index 5e877fe9ba7b..0e5a05b25a77 100644 --- a/sound/soc/zte/zx-tdm.c +++ b/sound/soc/zte/zx-tdm.c @@ -211,7 +211,6 @@ static int zx_tdm_hw_params(struct snd_pcm_substream *substream, ts_width = 1; break; default: - ts_width = 0; dev_err(socdai->dev, "Unknown data format\n"); return -EINVAL; } -- 2.20.1

2 1

[alsa-devel] [PATCH 1/3] ASoC: amd: use DMA addr rather than CPU pa for audio buffer
by Ravulapati Vishnu vardhan rao 01 Aug '19

01 Aug '19

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: Vijendar Mukunda <vijendar.mukunda(a)amd.com> Tested-by: RAVULAPATI VISHNU VARDHAN RAO <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index a4ade6b..49c4872 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -31,7 +31,7 @@ struct i2s_stream_instance { u16 num_pages; u16 channels; u32 xfer_resolution; - struct page *pg; + dma_addr_t dma_addr; u64 bytescount; void __iomem *acp3x_base; }; @@ -211,11 +211,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) { u16 page_idx; - u64 addr; + dma_addr_t addr; u32 low, high, val, acp_fifo_addr; - struct page *pg = rtd->pg; - /* 8 scratch registers used to map one 64 bit address */ + addr = rtd->dma_addr; + /* 8 scratch registers used to map one 64 bit address. + * For 2 pages (8192 * 2 bytes), it will be 16 registers. + */ if (direction == SNDRV_PCM_STREAM_PLAYBACK) val = 0; else @@ -229,7 +231,6 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) for (page_idx = 0; page_idx < rtd->num_pages; page_idx++) { /* Load the low address of page int ACP SRAM through SRBM */ - addr = page_to_phys(pg); low = lower_32_bits(addr); high = upper_32_bits(addr); @@ -239,7 +240,7 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) + 4); /* Move to next physically contiguos page */ val += 8; - pg++; + addr += PAGE_SIZE; } if (direction == SNDRV_PCM_STREAM_PLAYBACK) { @@ -340,10 +341,8 @@ static int acp3x_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { int status; - u64 size; - struct page *pg; - struct snd_pcm_runtime *runtime = substream->runtime; - struct i2s_stream_instance *rtd = runtime->private_data; + uint64_t size; + struct i2s_stream_instance *rtd = substream->runtime->private_data; if (!rtd) return -EINVAL; @@ -355,8 +354,8 @@ static int acp3x_dma_hw_params(struct snd_pcm_substream *substream, memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); pg = virt_to_page(substream->dma_buffer.area); - if (pg) { - rtd->pg = pg; + if (substream->dma_buffer.area) { + rtd->dma_addr = substream->dma_buffer.addr; rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); config_acp3x_dma(rtd, substream->stream); status = 0; -- 2.7.4

2 4

[alsa-devel] standard read/write vs event waiting routines (poll) in ALSA
by Diego Buffa 01 Aug '19

01 Aug '19

Hi all, I am trying to learn more about ALSA, I always used example found in the howtos but now I would go further. I went though ALSA documentation ( https://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html) and in the specific the transfer methods. Basically I don't understand the advantages of using poll instead of traditional blocking read/write. Generally I know that using poll I can multiplex I/O on several files, but in ALSA I cannot create the poll descriptor array as I want, as I must use snd_pcm_poll_descriptors() which requires that the poll descriptor array should have the size returned by snd_pcm_poll_descriptors_count(). Am I supposed to realloc the poll array descriptor once I set it up with ALSA, and add my own files? Should I then always check poll array descriptor with snd_pcm_poll_descriptors_revents() before checking my own files? And in this case it is safe to call snd_pcm_poll_descriptors_revents() with a poll array descriptor bigger than the size returned by snd_pcm_poll_descriptors_count()? Many thanks in advance.

2 2

[alsa-devel] snd_pcm_delay in snd_usb_audio
by Roeland Jansen 01 Aug '19

01 Aug '19

just subscribed to the list (I hope). while troubleshooting a different thing in audio I stumbled across the following and it's in the logs more than once. what do you need from me? this is opensuse 15.1 leap. warn:2019-07-31T19:12:58.207893+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. warn:2019-07-31T19:12:58.208183+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: snd_pcm_dump(): warn:2019-07-31T19:12:58.208473+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: Hardware PCM card 2 'USB Audio CODEC' device 0 subdevice 0 warn:2019-07-31T19:12:58.208760+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: Its setup is: warn:2019-07-31T19:12:58.209071+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: stream : CAPTURE warn:2019-07-31T19:12:58.209366+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: access : MMAP_INTERLEAVED warn:2019-07-31T19:12:58.209671+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: format : S16_LE warn:2019-07-31T19:12:58.209958+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: subformat : STD warn:2019-07-31T19:12:58.210245+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: channels : 2 warn:2019-07-31T19:12:58.210588+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: rate : 48000 warn:2019-07-31T19:12:58.210880+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: exact rate : 48000 (48000/1) warn:2019-07-31T19:12:58.211198+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: msbits : 16 warn:2019-07-31T19:12:58.211746+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: buffer_size : 88200 warn:2019-07-31T19:12:58.212028+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: period_size : 44100 warn:2019-07-31T19:12:58.212261+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: period_time : 918750 warn:2019-07-31T19:12:58.212548+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: tstamp_mode : ENABLE warn:2019-07-31T19:12:58.212785+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: tstamp_type : MONOTONIC warn:2019-07-31T19:12:58.213014+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: period_step : 1 warn:2019-07-31T19:12:58.213238+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: avail_min : 44100 warn:2019-07-31T19:12:58.213470+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: period_event : 0 warn:2019-07-31T19:12:58.213722+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: start_threshold : -1 warn:2019-07-31T19:12:58.213995+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: stop_threshold : 6206523236469964800 warn:2019-07-31T19:12:58.214465+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: silence_threshold: 0 warn:2019-07-31T19:12:58.214730+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: silence_size : 0 warn:2019-07-31T19:12:58.214971+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: boundary : 6206523236469964800 warn:2019-07-31T19:12:58.215228+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: appl_ptr : 45284903 warn:2019-07-31T19:12:58.215510+02:00 taplop pulseaudio[28164]: [alsa-source-USB Audio] alsa-util.c: hw_ptr : 55414415 warn:2019-07-31T21:55:16.513279+02:00 taplop pulseaudio[28164]: [alsa-sink-USB Audio] alsa-util.c: Got POLLNVAL from ALSA warn:2019-07-31T21:55:16.513748+02:00 taplop pulseaudio[28164]: [alsa-sink-USB Audio] alsa-util.c: Could not recover from POLLERR|POLLNVAL|POLLHUP with snd_pcm_prepare(): No such device

1 0

[alsa-devel] [PATCH 0/6] ReST conversion patches not applied yet
by Mauro Carvalho Chehab 31 Jul '19

31 Jul '19

Hi Jon, As promised, this is the rebased version of the patches that were not applied from the /26 patch series because you had merge conflicts. They're all based on your docs-next branch, so should apply fine. The first one fixes all but one error with a broken reference. The only broken reference right now is due to a DT patch with was not accepted (no idea why), but whose driver is upstream. Mauro Carvalho Chehab (6): docs: fix a couple of new broken references docs: writing-schema.md: convert from markdown to ReST spi: docs: convert to ReST and add it to the kABI bookset docs: fs: cifs: convert to ReST and add to admin-guide book docs: fs: convert porting to ReST docs: w1: convert to ReST and add to the kAPI group of docs Documentation/ABI/stable/sysfs-bus-w1 | 2 +- .../ABI/stable/sysfs-driver-w1_ds28e04 | 4 +- .../ABI/stable/sysfs-driver-w1_ds28ea00 | 2 +- .../AUTHORS => admin-guide/cifs/authors.rst} | 64 +- .../CHANGES => admin-guide/cifs/changes.rst} | 4 + Documentation/admin-guide/cifs/index.rst | 21 + .../cifs/introduction.rst} | 8 + .../cifs/TODO => admin-guide/cifs/todo.rst} | 87 +-- .../README => admin-guide/cifs/usage.rst} | 560 +++++++++++------- .../cifs/winucase_convert.pl | 0 Documentation/admin-guide/index.rst | 1 + .../bindings/sound/sun8i-a33-codec.txt | 2 +- Documentation/devicetree/writing-schema.md | 130 ---- Documentation/devicetree/writing-schema.rst | 153 +++++ Documentation/filesystems/porting.rst | 2 +- Documentation/index.rst | 2 + .../spi/{butterfly => butterfly.rst} | 44 +- Documentation/spi/index.rst | 22 + Documentation/spi/{pxa2xx => pxa2xx.rst} | 95 +-- .../spi/{spi-lm70llp => spi-lm70llp.rst} | 17 +- .../spi/{spi-sc18is602 => spi-sc18is602.rst} | 3 + .../spi/{spi-summary => spi-summary.rst} | 105 ++-- Documentation/spi/{spidev => spidev.rst} | 30 +- Documentation/w1/index.rst | 21 + .../w1/masters/{ds2482 => ds2482.rst} | 16 +- .../w1/masters/{ds2490 => ds2490.rst} | 6 +- Documentation/w1/masters/index.rst | 14 + Documentation/w1/masters/mxc-w1 | 12 - Documentation/w1/masters/mxc-w1.rst | 17 + .../w1/masters/{omap-hdq => omap-hdq.rst} | 12 +- .../w1/masters/{w1-gpio => w1-gpio.rst} | 21 +- Documentation/w1/slaves/index.rst | 16 + .../w1/slaves/{w1_ds2406 => w1_ds2406.rst} | 4 +- .../w1/slaves/{w1_ds2413 => w1_ds2413.rst} | 9 + Documentation/w1/slaves/w1_ds2423 | 47 -- Documentation/w1/slaves/w1_ds2423.rst | 54 ++ .../w1/slaves/{w1_ds2438 => w1_ds2438.rst} | 10 +- .../w1/slaves/{w1_ds28e04 => w1_ds28e04.rst} | 5 + .../w1/slaves/{w1_ds28e17 => w1_ds28e17.rst} | 16 +- .../w1/slaves/{w1_therm => w1_therm.rst} | 11 +- .../w1/{w1.generic => w1-generic.rst} | 88 +-- .../w1/{w1.netlink => w1-netlink.rst} | 89 +-- MAINTAINERS | 6 +- drivers/hwtracing/coresight/Kconfig | 2 +- drivers/iio/dummy/iio_simple_dummy.c | 2 +- drivers/spi/Kconfig | 2 +- drivers/spi/spi-butterfly.c | 2 +- drivers/spi/spi-lm70llp.c | 2 +- fs/jfs/Kconfig | 2 +- fs/ufs/Kconfig | 2 +- include/linux/platform_data/sc18is602.h | 2 +- 51 files changed, 1126 insertions(+), 722 deletions(-) rename Documentation/{filesystems/cifs/AUTHORS => admin-guide/cifs/authors.rst} (60%) rename Documentation/{filesystems/cifs/CHANGES => admin-guide/cifs/changes.rst} (91%) create mode 100644 Documentation/admin-guide/cifs/index.rst rename Documentation/{filesystems/cifs/cifs.txt => admin-guide/cifs/introduction.rst} (98%) rename Documentation/{filesystems/cifs/TODO => admin-guide/cifs/todo.rst} (58%) rename Documentation/{filesystems/cifs/README => admin-guide/cifs/usage.rst} (72%) rename Documentation/{filesystems => admin-guide}/cifs/winucase_convert.pl (100%) delete mode 100644 Documentation/devicetree/writing-schema.md create mode 100644 Documentation/devicetree/writing-schema.rst rename Documentation/spi/{butterfly => butterfly.rst} (71%) create mode 100644 Documentation/spi/index.rst rename Documentation/spi/{pxa2xx => pxa2xx.rst} (83%) rename Documentation/spi/{spi-lm70llp => spi-lm70llp.rst} (88%) rename Documentation/spi/{spi-sc18is602 => spi-sc18is602.rst} (97%) rename Documentation/spi/{spi-summary => spi-summary.rst} (93%) rename Documentation/spi/{spidev => spidev.rst} (90%) create mode 100644 Documentation/w1/index.rst rename Documentation/w1/masters/{ds2482 => ds2482.rst} (71%) rename Documentation/w1/masters/{ds2490 => ds2490.rst} (98%) create mode 100644 Documentation/w1/masters/index.rst delete mode 100644 Documentation/w1/masters/mxc-w1 create mode 100644 Documentation/w1/masters/mxc-w1.rst rename Documentation/w1/masters/{omap-hdq => omap-hdq.rst} (90%) rename Documentation/w1/masters/{w1-gpio => w1-gpio.rst} (75%) create mode 100644 Documentation/w1/slaves/index.rst rename Documentation/w1/slaves/{w1_ds2406 => w1_ds2406.rst} (96%) rename Documentation/w1/slaves/{w1_ds2413 => w1_ds2413.rst} (81%) delete mode 100644 Documentation/w1/slaves/w1_ds2423 create mode 100644 Documentation/w1/slaves/w1_ds2423.rst rename Documentation/w1/slaves/{w1_ds2438 => w1_ds2438.rst} (93%) rename Documentation/w1/slaves/{w1_ds28e04 => w1_ds28e04.rst} (93%) rename Documentation/w1/slaves/{w1_ds28e17 => w1_ds28e17.rst} (88%) rename Documentation/w1/slaves/{w1_therm => w1_therm.rst} (95%) rename Documentation/w1/{w1.generic => w1-generic.rst} (59%) rename Documentation/w1/{w1.netlink => w1-netlink.rst} (77%) -- 2.21.0

5 9

[alsa-devel] [PATCH v4 0/5] ALSA/HDA: abort probe when DMICs are detected
by Pierre-Louis Bossart 31 Jul '19

31 Jul '19

This is the second take on same problem of detecting when the HDaudio legacy driver can be used and when the SST or SOF drivers are required. The previous attempt based on a the PCI Class information was a resounding failure and broke audio on Linus' laptop, so we need additional information to avoid enabling a DSP-based driver without a good reason to do so. This patchset suggests the use of the NHLT information which *in theory* exposes DMIC endpoints. The legacy HDaudio driver cannot handle DMICs and will not provide any capture capabilities. Since it's typically the first one to probe due to the Makefile order, aborting the probe will let the PCI subsystem look for the next driver which hopefully will support this capability. I tested this patch on 5 devices (SKL, KBL, APL, GLK, WHL), three without DMICs and two with, and the detection seems to work as planned. Additional testing by Canonical and Endless folks did not expose any issues. Changes since v3 (Feedback from Cezary) Code cleanups (spaces and unnecessary inits) Flipped test statement to return on errors and reduce indentation Removed ACPI leak by adding missing ACPI_FREE() Changes since v2 (Feedback from Takahi and Cezary) Added comment in Kconfig to alert the user to the dependency on ACPI Changes since v1 (Feedback from Takashi): Squashed patch3 in patch2 Changed log to dbg_info Fixed module parameter handling Changes since RFC: Cosmetic code improvements Moved intel-nhlt.h to include/sound Moved intel-nhlt.c to sound/hda Removed SOC prefixes Added full-support for vendor-defined geometries Added Kconfig and kernel module parameter to opt-in Pierre-Louis Bossart (5): ASoC: Intel: Skylake: move NHLT header to common directory ALSA: hda: move parts of NHLT code to new module ALSA: hda: intel-nhlt: handle NHLT VENDOR_DEFINED DMIC geometry ASoC: Intel: Skylake: use common NHLT module ALSA: hda/intel: stop probe if DMICS are detected on Skylake+ platforms .../skl-nhlt.h => include/sound/intel-nhlt.h | 51 +++++++-- sound/hda/Kconfig | 5 + sound/hda/Makefile | 3 + sound/hda/intel-nhlt.c | 107 ++++++++++++++++++ sound/pci/hda/Kconfig | 10 ++ sound/pci/hda/hda_intel.c | 34 ++++++ sound/soc/intel/Kconfig | 1 + sound/soc/intel/skylake/skl-nhlt.c | 91 +-------------- sound/soc/intel/skylake/skl-ssp-clk.c | 1 + sound/soc/intel/skylake/skl-topology.c | 1 + sound/soc/intel/skylake/skl.c | 12 +- sound/soc/intel/skylake/skl.h | 4 - 12 files changed, 212 insertions(+), 108 deletions(-) rename sound/soc/intel/skylake/skl-nhlt.h => include/sound/intel-nhlt.h (65%) create mode 100644 sound/hda/intel-nhlt.c -- 2.20.1

2 7

[alsa-devel] Audio miniconf + sof meeting at ELCE, Lyon
by Daniel Baluta 31 Jul '19

31 Jul '19

Hi Mark, Pierre, Liam, Ranjani, Will you be in Lyon for ELCE this year? I'm very interested in participating to an (eventual) audio miniconf meeting. Maybe we can also do a separate SOF meeting if they are enough people interested.

3 2

[alsa-devel] [PATCH][next] ASoC: codec2codec: fix missing return of error return code
by Colin King 31 Jul '19

31 Jul '19

From: Colin Ian King <colin.king(a)canonical.com> Currently in function snd_soc_dai_link_event_pre_pmu the error return code in variable err is being set but this is not actually being returned, the function just returns zero even when there are failures. Fix this by returning the error return code. Addresses-Coverity: ("Unused value") Fixes: 3dcfb397dad2 ("ASoC: codec2codec: deal with params when necessary") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index be9bb05..2d183e2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3776,7 +3776,7 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w, struct snd_pcm_hw_params *params = NULL; const struct snd_soc_pcm_stream *config = NULL; unsigned int fmt; - int ret; + int ret = 0; params = kzalloc(sizeof(*params), GFP_KERNEL); if (!params) @@ -3865,7 +3865,7 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w, out: kfree(params); - return 0; + return ret; } static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: meson: g12a-tohdmitx: override codec2codec params
by Jerome Brunet 31 Jul '19

31 Jul '19

So far, forwarding the hw_params of the input to output relied on the .hw_params() callback of the cpu side of the codec2codec link to be called first. This is a bit weak. Instead, override the stream params of the codec2codec to link to set it up correctly. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/g12a-tohdmitx.c | 34 ++++++++++++++++----------------- 1 file changed, 16 insertions(+), 18 deletions(-) diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c index 707ccb192e4c..9943c807ec5d 100644 --- a/sound/soc/meson/g12a-tohdmitx.c +++ b/sound/soc/meson/g12a-tohdmitx.c @@ -28,7 +28,7 @@ #define CTRL0_SPDIF_CLK_SEL BIT(0) struct g12a_tohdmitx_input { - struct snd_pcm_hw_params params; + struct snd_soc_pcm_stream params; unsigned int fmt; }; @@ -225,26 +225,17 @@ static int g12a_tohdmitx_input_hw_params(struct snd_pcm_substream *substream, { struct g12a_tohdmitx_input *data = dai->playback_dma_data; - /* Save the stream params for the downstream link */ - memcpy(&data->params, params, sizeof(*params)); + data->params.rates = snd_pcm_rate_to_rate_bit(params_rate(params)); + data->params.rate_min = params_rate(params); + data->params.rate_max = params_rate(params); + data->params.formats = 1 << params_format(params); + data->params.channels_min = params_channels(params); + data->params.channels_max = params_channels(params); + data->params.sig_bits = dai->driver->playback.sig_bits; return 0; } -static int g12a_tohdmitx_output_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct g12a_tohdmitx_input *in_data = - g12a_tohdmitx_get_input_data(dai->capture_widget); - - if (!in_data) - return -ENODEV; - - memcpy(params, &in_data->params, sizeof(*params)); - - return 0; -} static int g12a_tohdmitx_input_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) @@ -266,6 +257,14 @@ static int g12a_tohdmitx_output_startup(struct snd_pcm_substream *substream, if (!in_data) return -ENODEV; + if (WARN_ON(!rtd->dai_link->params)) { + dev_warn(dai->dev, "codec2codec link expected\n"); + return -EINVAL; + } + + /* Replace link params with the input params */ + rtd->dai_link->params = &in_data->params; + if (!in_data->fmt) return 0; @@ -278,7 +277,6 @@ static const struct snd_soc_dai_ops g12a_tohdmitx_input_ops = { }; static const struct snd_soc_dai_ops g12a_tohdmitx_output_ops = { - .hw_params = g12a_tohdmitx_output_hw_params, .startup = g12a_tohdmitx_output_startup, }; -- 2.21.0

2 1

[alsa-devel] [PATCH -next 00/34] ASoC: use devm_platform_ioremap_resource() to simplify code
by YueHaibing 31 Jul '19

31 Jul '19

devm_platform_ioremap_resource() internally have platform_get_resource() and devm_ioremap_resource() in it. So instead of calling them separately use devm_platform_ioremap_resource() directly. YueHaibing (34): ASoC: tegra20_das: use devm_platform_ioremap_resource() to simplify code ASoC: tegra: use devm_platform_ioremap_resource() to simplify code ASoC: rockchip: use devm_platform_ioremap_resource() to simplify code ASoC: ep93xx-i2s: use devm_platform_ioremap_resource() to simplify code ASoC: mt8173: use devm_platform_ioremap_resource() to simplify code ASoC: mt6797: use devm_platform_ioremap_resource() to simplify code ASoC: imx-audmux: use devm_platform_ioremap_resource() to simplify code ASoC: fsl_audmix: use devm_platform_ioremap_resource() to simplify code ASoC: bcm2835-i2s: use devm_platform_ioremap_resource() to simplify code ASoC: sirf: use devm_platform_ioremap_resource() to simplify code ASoC: mxs-saif: use devm_platform_ioremap_resource() to simplify code ASoC: spear: use devm_platform_ioremap_resource() to simplify code ASoC: kirkwood-i2s: use devm_platform_ioremap_resource() to simplify code ASoC: xtfpga-i2s: use devm_platform_ioremap_resource() to simplify code ASoC: stm32: sai: use devm_platform_ioremap_resource() to simplify code ASoC: codecs: msm8916-wcd: use devm_platform_ioremap_resource() to simplify code ASoC: codecs: jz4725b: use devm_platform_ioremap_resource() to simplify code ASoC: mmp-sspa: use devm_platform_ioremap_resource() to simplify code ASoC: jz4740: use devm_platform_ioremap_resource() to simplify code ASoC: inno_rk3036: use devm_platform_ioremap_resource() to simplify code ASoC: uniphier: evea: use devm_platform_ioremap_resource() to simplify code ASoC: uniphier: aio-dma: use devm_platform_ioremap_resource() to simplify code ASoC: psc-ac97: use devm_platform_ioremap_resource() to simplify code ASoC: au1x: psc-i2s: use devm_platform_ioremap_resource() to simplify code ASoC: meson: g12a-tohdmitx: use devm_platform_ioremap_resource() to simplify code ASoC: meson: axg-tdm-formatter: use devm_platform_ioremap_resource() to simplify code ASoC: meson: axg-pdm: use devm_platform_ioremap_resource() to simplify code ASoC: meson: axg-spdifin: use devm_platform_ioremap_resource() to simplify code ASoC: meson: axg-spdifout: use devm_platform_ioremap_resource() to simplify code ASoC: meson: axg-fifo: use devm_platform_ioremap_resource() to simplify code ASoC: xlnx: use devm_platform_ioremap_resource() to simplify code ASoC: sun8i-codec-analog: use devm_platform_ioremap_resource() to simplify code ASoC: sunxi: sun50i-codec-analog: use devm_platform_ioremap_resource() to simplify code ASoC: sun8i-codec: use devm_platform_ioremap_resource() to simplify code sound/soc/au1x/psc-ac97.c | 5 ++--- sound/soc/au1x/psc-i2s.c | 5 ++--- sound/soc/bcm/bcm2835-i2s.c | 4 +--- sound/soc/cirrus/ep93xx-i2s.c | 4 +--- sound/soc/codecs/inno_rk3036.c | 4 +--- sound/soc/codecs/jz4725b.c | 4 +--- sound/soc/codecs/jz4740.c | 4 +--- sound/soc/codecs/msm8916-wcd-digital.c | 4 +--- sound/soc/codecs/rk3328_codec.c | 4 +--- sound/soc/fsl/fsl_audmix.c | 4 +--- sound/soc/fsl/imx-audmux.c | 4 +--- sound/soc/kirkwood/kirkwood-i2s.c | 4 +--- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 5 +---- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 4 +--- sound/soc/meson/axg-fifo.c | 4 +--- sound/soc/meson/axg-pdm.c | 4 +--- sound/soc/meson/axg-spdifin.c | 4 +--- sound/soc/meson/axg-spdifout.c | 4 +--- sound/soc/meson/axg-tdm-formatter.c | 4 +--- sound/soc/meson/g12a-tohdmitx.c | 4 +--- sound/soc/mxs/mxs-saif.c | 5 +---- sound/soc/pxa/mmp-sspa.c | 4 +--- sound/soc/sirf/sirf-usp.c | 4 +--- sound/soc/spear/spdif_in.c | 5 ++--- sound/soc/stm/stm32_sai.c | 4 +--- sound/soc/sunxi/sun50i-codec-analog.c | 4 +--- sound/soc/sunxi/sun8i-codec-analog.c | 4 +--- sound/soc/sunxi/sun8i-codec.c | 4 +--- sound/soc/tegra/tegra20_das.c | 4 +--- sound/soc/tegra/tegra30_i2s.c | 4 +--- sound/soc/uniphier/aio-dma.c | 4 +--- sound/soc/uniphier/evea.c | 4 +--- sound/soc/xilinx/xlnx_i2s.c | 4 +--- sound/soc/xilinx/xlnx_spdif.c | 3 +-- sound/soc/xtensa/xtfpga-i2s.c | 4 +--- 35 files changed, 38 insertions(+), 106 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: max98373: add 88200 and 96000 sampling rate support
by chunguo feng 31 Jul '19

31 Jul '19

From: fengchunguo <chunguo.feng(a)amlogic.com> 88200 and 96000 sampling rate was not enabled on driver, so can't be played. The error information: max98373 3-0031：rate 96000 not supported max98373 3-0031：ASoC: can't set max98373-aif1 hw params: -22 Signed-off-by: fengchunguo <chunguo.feng(a)amlogic.com> --- sound/soc/codecs/max98373.c | 6 ++++++ sound/soc/codecs/max98373.h | 2 ++ 2 files changed, 8 insertions(+) mode change 100644 => 100755 sound/soc/codecs/max98373.c mode change 100644 => 100755 sound/soc/codecs/max98373.h diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c old mode 100644 new mode 100755 index 528695cd6a1c..8c601a3ebc27 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -267,6 +267,12 @@ static int max98373_dai_hw_params(struct snd_pcm_substream *substream, case 48000: sampling_rate = MAX98373_PCM_SR_SET1_SR_48000; break; + case 88200: + sampling_rate = MAX98373_PCM_SR_SET1_SR_88200; + break; + case 96000: + sampling_rate = MAX98373_PCM_SR_SET1_SR_96000; + break; default: dev_err(component->dev, "rate %d not supported\n", params_rate(params)); diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h old mode 100644 new mode 100755 index f6a37aa02f26..a59e51355a84 --- a/sound/soc/codecs/max98373.h +++ b/sound/soc/codecs/max98373.h @@ -130,6 +130,8 @@ #define MAX98373_PCM_SR_SET1_SR_32000 (0x6 << 0) #define MAX98373_PCM_SR_SET1_SR_44100 (0x7 << 0) #define MAX98373_PCM_SR_SET1_SR_48000 (0x8 << 0) +#define MAX98373_PCM_SR_SET1_SR_88200 (0x9 << 0) +#define MAX98373_PCM_SR_SET1_SR_96000 (0xA << 0) /* MAX98373_R2028_PCM_SR_SETUP_2 */ #define MAX98373_PCM_SR_SET2_SR_MASK (0xF << 4) -- 2.22.0

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[alsa-devel] Applied "ASoC: sun4i-i2s: Incorrect SR and WSS computation" to the asoc tree
by Mark Brown 31 Jul '19

31 Jul '19

The patch ASoC: sun4i-i2s: Incorrect SR and WSS computation has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.3 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 52f87f3ca251f5e43b42e78ab9816b2b07718bfe Mon Sep 17 00:00:00 2001 From: Marcus Cooper <codekipper(a)gmail.com> Date: Mon, 29 Jul 2019 17:21:30 +0200 Subject: [PATCH] ASoC: sun4i-i2s: Incorrect SR and WSS computation The A64 audio codec uses the original I2S block but the SR and WSS computation currently assigned is for the newer block. Fixes: 619c15f7fac9 (ASoC: sun4i-i2s: Change SR and WSS computation) Signed-off-by: Marcus Cooper <codekipper(a)gmail.com> Link: https://lore.kernel.org/r/20190729152130.27955-1-codekipper@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sunxi/sun4i-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 9b2232908b65..7fa5c61169db 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1002,8 +1002,8 @@ static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = { .field_rxchanmap = REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31), .field_txchansel = REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2), .field_rxchansel = REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2), - .get_sr = sun8i_i2s_get_sr_wss, - .get_wss = sun8i_i2s_get_sr_wss, + .get_sr = sun4i_i2s_get_sr, + .get_wss = sun4i_i2s_get_wss, }; static int sun4i_i2s_init_regmap_fields(struct device *dev, -- 2.20.1

1 0

[alsa-devel] [PATCH] MAINTAINERS: Update Intel ASoC drivers maintainers
by Cezary Rojewski 31 Jul '19

31 Jul '19

Adding myself to Intel ASoC drivers maintainers list. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 783569e3c4b4..0db421933d0c 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -8044,6 +8044,7 @@ S: Maintained F: drivers/video/fbdev/i810/ INTEL ASoC DRIVERS +M: Cezary Rojewski <cezary.rojewski(a)intel.com> M: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> M: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> M: Jie Yang <yang.jie(a)linux.intel.com> -- 2.17.1

3 2

[alsa-devel] [PATCH] ASoC: ti: Mark expected switch fall-throughs
by Gustavo A. R. Silva 31 Jul '19

31 Jul '19

Mark switch cases where we are expecting to fall through. This patch fixes the following warning (Building: arm): sound/soc/ti/n810.c: In function ‘n810_ext_control’: sound/soc/ti/n810.c:48:10: warning: this statement may fall through [-Wimplicit-fallthrough=] line1l = 1; ~~~~~~~^~~ sound/soc/ti/n810.c:49:2: note: here case N810_JACK_HP: ^~~~ sound/soc/ti/rx51.c: In function ‘rx51_ext_control’: sound/soc/ti/rx51.c:57:6: warning: this statement may fall through [-Wimplicit-fallthrough=] hs = 1; ~~~^~~ sound/soc/ti/rx51.c:58:2: note: here case RX51_JACK_HP: ^~~~ Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/ti/n810.c | 1 + sound/soc/ti/rx51.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/ti/n810.c b/sound/soc/ti/n810.c index 2c3f2a4c1700..3ad2b6daf31e 100644 --- a/sound/soc/ti/n810.c +++ b/sound/soc/ti/n810.c @@ -46,6 +46,7 @@ static void n810_ext_control(struct snd_soc_dapm_context *dapm) switch (n810_jack_func) { case N810_JACK_HS: line1l = 1; + /* fall through */ case N810_JACK_HP: hp = 1; break; diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c index bc6046534fa5..ccd0e8a07dd1 100644 --- a/sound/soc/ti/rx51.c +++ b/sound/soc/ti/rx51.c @@ -55,6 +55,7 @@ static void rx51_ext_control(struct snd_soc_dapm_context *dapm) break; case RX51_JACK_HS: hs = 1; + /* fall through */ case RX51_JACK_HP: hp = 1; break; -- 2.22.0

3 2

[alsa-devel] [PATCH] ASoC: cs4271: Fix a typo in the CS4171_NR_RATIOS
by Christophe JAILLET 31 Jul '19

31 Jul '19

This should be CS4271_NR_RATIOS. Fix it and use it. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/cs4271.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 1d03a1348162..04b86a51e055 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -334,7 +334,7 @@ static struct cs4271_clk_cfg cs4271_clk_tab[] = { {0, CS4271_MODE1_MODE_4X, 256, CS4271_MODE1_DIV_2}, }; -#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) +#define CS4271_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) static int cs4271_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -383,13 +383,13 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, val = CS4271_MODE1_MODE_4X; ratio = cs4271->mclk / cs4271->rate; - for (i = 0; i < CS4171_NR_RATIOS; i++) + for (i = 0; i < CS4271_NR_RATIOS; i++) if ((cs4271_clk_tab[i].master == cs4271->master) && (cs4271_clk_tab[i].speed_mode == val) && (cs4271_clk_tab[i].ratio == ratio)) break; - if (i == CS4171_NR_RATIOS) { + if (i == CS4271_NR_RATIOS) { dev_err(component->dev, "Invalid sample rate\n"); return -EINVAL; } -- 2.20.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: Add uda1334 codec driver
by Andra Danciu 31 Jul '19

31 Jul '19

The first patch contains uda1334 documentation. The second patch contains the codec driver. Andra Danciu (2): dt-bindings: sound: Add bindings for UDA1334 codec ASoC: codecs: Add uda1334 codec driver .../devicetree/bindings/sound/uda1334.txt | 17 ++ sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/uda1334.c | 295 +++++++++++++++++++++ 4 files changed, 323 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/uda1334.txt create mode 100644 sound/soc/codecs/uda1334.c -- 2.11.0

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Re: [alsa-devel] [Sound-open-firmware] [GIT PULL] SOF v1.3 firmware binaries
by Pierre-Louis Bossart 31 Jul '19

31 Jul '19

[fixed alsa-devel email] On 7/30/19 10:33 AM, Liam Girdwood wrote: > The following changes since commit dff98c6c57383fe343407bcb7b6e775e0b87274f: > > Merge branch 'master' of git://github.com/skeggsb/linux-firmware (2019-07-26 07:32:37 -0400) > > are available in the Git repository at: > > https://github.com/thesofproject/linux-firmware.git sof-v1.3 > > for you to fetch changes up to cde3a116cea96976125b9215b303edfda85c9b54: > > sof: Add Intel SOF V1.3 release firmware binaries. (2019-07-30 16:06:41 +0100) > > ---------------------------------------------------------------- > Liam Girdwood (1): > sof: Add Intel SOF V1.3 release firmware binaries. > > LICENCE.sof | 1090 ++++++++++++++++++++++++++ Humm, that LICENSE file needs to be double-checked. Is there any reason why the text of this LICENSE.sof is different the usual text, e.g. from the LICENSE.adsp_sst? You are missing both the first part: ***** INTEL BINARY FIRMWARE RELEASE LICENCE ******************************** Copyright (c) 2014-15 Intel Corporation. All rights reserved. Redistribution. Redistribution and use in binary form, without modification, are permitted provided that the following conditions are met: * Redistributions must reproduce the above copyright notice and the following disclaimer in the documentation and/or other materials provided with the distribution. * Neither the name of Intel Corporation nor the names of its suppliers may be used to endorse or promote products derived from this software without specific prior written permission. * No reverse engineering, decompilation, or disassembly of this software is permitted. and the DISCLAIMER part, both of which seem pretty important to me. IANAL, but seeing only a patent clause looks odd. There should be a mention of redistribution and a clear disclaimer (not sure about the reverse engineering parts since the code is available it makes no sense). > WHENCE | 33 + > intel/sof/apl/intel/sof-apl-v1.3.ri | Bin 0 -> 270336 bytes > intel/sof/bdw/sof-bdw-v1.3.ri | Bin 0 -> 100144 bytes > intel/sof/byt/sof-byt-v1.3.ri | Bin 0 -> 89668 bytes > intel/sof/cht/sof-cht-v1.3.ri | Bin 0 -> 90484 bytes > intel/sof/cnl/intel/sof-cnl-v1.3-6cc8da10.ri | Bin 0 -> 274432 bytes > intel/sof/icl/intel/sof-icl-v1.3.ri | Bin 0 -> 278528 bytes There are two types of platforms, the ones which require the Intel firmware to be signed with a private production key and the ones that are signed with the SOF community key. if we have a single directory, then how do we deal with the two cases? It's not even clear to me which of the two cases are handled here. > intel/sof/sof-apl.ri | 1 + > intel/sof/sof-bdw.ri | 1 + > intel/sof/sof-byt.ri | 1 + > intel/sof/sof-cht.ri | 1 + > intel/sof/sof-cml.ri | 1 + > intel/sof/sof-cnl.ri | 1 + > intel/sof/sof-glk.ri | 1 + > intel/sof/sof-icl.ri | 1 + > intel/sof/sof-whl.ri | 1 + unless I am missing something, we don't have any tables in the Linux kernel for the WHL and CML configurations, and IIRC we only generate sof-cnl.ri. Is there actually a user for sof-whl.ri and sof-cml.ri? > 17 files changed, 1132 insertions(+) > create mode 100644 LICENCE.sof > create mode 100644 intel/sof/apl/intel/sof-apl-v1.3.ri > create mode 100644 intel/sof/bdw/sof-bdw-v1.3.ri > create mode 100644 intel/sof/byt/sof-byt-v1.3.ri > create mode 100644 intel/sof/cht/sof-cht-v1.3.ri > create mode 100644 intel/sof/cnl/intel/sof-cnl-v1.3-6cc8da10.ri > create mode 100644 intel/sof/icl/intel/sof-icl-v1.3.ri > create mode 120000 intel/sof/sof-apl.ri > create mode 120000 intel/sof/sof-bdw.ri > create mode 120000 intel/sof/sof-byt.ri > create mode 120000 intel/sof/sof-cht.ri > create mode 120000 intel/sof/sof-cml.ri > create mode 120000 intel/sof/sof-cnl.ri > create mode 120000 intel/sof/sof-glk.ri > create mode 120000 intel/sof/sof-icl.ri > create mode 120000 intel/sof/sof-whl.ri > > _______________________________________________ > Sound-open-firmware mailing list > Sound-open-firmware(a)alsa-project.org > https://mailman.alsa-project.org/mailman/listinfo/sound-open-firmware >

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[alsa-devel] Patch Update for OpenFrame Sigmatel STAC9202
by Andy Davison 31 Jul '19

31 Jul '19

Hi all, Would anybody on this list have a little time to investigate this patch and bring it up to date for application against the latest 5.1 kernel, or indeed anything being actively developed? https://gist.github.com/andydvsn/c0c159f99bf19d5b30b5e5e156dcac3e This applies correctly against the 2.6.33 kernel, but I I lack the necessary ALSA knowledge and programming skills to make the appropriate corrections for anything more recent. Alternatively, any other advice on where I could seek help with this would be much appreciated. It is the only outstanding kernel issue for this device and would love to have it fixed. All the best, Andy.

3 5

[alsa-devel] [PATCH] ASoC: max98383: add 88200 and 96000 sampling rate support
by chunguo feng 31 Jul '19

31 Jul '19

From: fengchunguo <chunguo.feng(a)amlogic.com> 88200 and 96000 sampling rate was not enabled on driver, so can't be played. The error information: max98373 3-0031：rate 96000 not supported max98373 3-0031：ASoC: can't set max98373-aif1 hw params: -22 Signed-off-by: fengchunguo <chunguo.feng(a)amlogic.com> --- sound/soc/codecs/max98373.c | 6 ++++++ sound/soc/codecs/max98373.h | 2 ++ 2 files changed, 8 insertions(+) mode change 100644 => 100755 sound/soc/codecs/max98373.c mode change 100644 => 100755 sound/soc/codecs/max98373.h diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c old mode 100644 new mode 100755 index 528695cd6a1c..8c601a3ebc27 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -267,6 +267,12 @@ static int max98373_dai_hw_params(struct snd_pcm_substream *substream, case 48000: sampling_rate = MAX98373_PCM_SR_SET1_SR_48000; break; + case 88200: + sampling_rate = MAX98373_PCM_SR_SET1_SR_88200; + break; + case 96000: + sampling_rate = MAX98373_PCM_SR_SET1_SR_96000; + break; default: dev_err(component->dev, "rate %d not supported\n", params_rate(params)); diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h old mode 100644 new mode 100755 index f6a37aa02f26..a59e51355a84 --- a/sound/soc/codecs/max98373.h +++ b/sound/soc/codecs/max98373.h @@ -130,6 +130,8 @@ #define MAX98373_PCM_SR_SET1_SR_32000 (0x6 << 0) #define MAX98373_PCM_SR_SET1_SR_44100 (0x7 << 0) #define MAX98373_PCM_SR_SET1_SR_48000 (0x8 << 0) +#define MAX98373_PCM_SR_SET1_SR_88200 (0x9 << 0) +#define MAX98373_PCM_SR_SET1_SR_96000 (0xA << 0) /* MAX98373_R2028_PCM_SR_SETUP_2 */ #define MAX98373_PCM_SR_SET2_SR_MASK (0xF << 4) -- 2.22.0

2 1

[alsa-devel] [PATCH v4 1/2] ASoC: SOF: introduce no_stream_position so host_period_bytes preserves its data
by Rajwa, Marcin 31 Jul '19

31 Jul '19

From: Marcin Rajwa <marcin.rajwa(a)linux.intel.com> Change the use of host_period_bytes. So far this field was used as an bool value indicating whether FW should send stream position update. With this patch we use host_period_bytes to provide firmware information about the frequency of host interrupts aimed to read its input buffer. This is accoring to ALSA definition of 'FramePeriod'. Knowing this firmware can safely copy large/irregular chunks of data (like data comming from i.e draining task) without the risk of buffer overflow. Signed-off-by: Marcin Rajwa <marcin.rajwa(a)linux.intel.com> Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- include/sound/sof/stream.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/sof/stream.h b/include/sound/sof/stream.h index 643f175cb479..06af4ecb2584 100644 --- a/include/sound/sof/stream.h +++ b/include/sound/sof/stream.h @@ -83,10 +83,10 @@ struct sof_ipc_stream_params { uint16_t sample_valid_bytes; uint16_t sample_container_bytes; - /* for notifying host period has completed - 0 means no period IRQ */ uint32_t host_period_bytes; + uint16_t no_stream_position; /* 1 means no IPC for position update */ - uint32_t reserved[2]; + uint16_t reserved[3]; uint16_t chmap[SOF_IPC_MAX_CHANNELS]; /**< channel map - SOF_CHMAP_ */ } __packed;

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[alsa-devel] [PATCH v2 3/3] gpiolib-acpi: Move acpi_dev_add_driver_gpios() et al to consumer.h
by Andy Shevchenko 31 Jul '19

31 Jul '19

The API, which belongs to GPIO library, is foreign to ACPI headers. Earlier we moved out I²C out of the latter, and now it's time for acpi_dev_add_driver_gpios() et al. For time being the acpi_gpio_get_irq_resource() and acpi_dev_gpio_irq_get() are left untouched as they need more thought about. Note, it requires uninline acpi_dev_remove_driver_gpios() to keep purity of consumer.h. Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org (moderated list:INTEL ASoC DRIVERS) Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- v2: new patch drivers/gpio/gpiolib-acpi.c | 7 +++ include/linux/acpi.h | 51 ------------------ include/linux/gpio/consumer.h | 57 ++++++++++++++++++++ sound/soc/intel/boards/bytcht_cx2072x.c | 1 + sound/soc/intel/boards/cht_bsw_max98090_ti.c | 1 + sound/soc/intel/boards/cht_bsw_rt5672.c | 1 + 6 files changed, 67 insertions(+), 51 deletions(-) diff --git a/drivers/gpio/gpiolib-acpi.c b/drivers/gpio/gpiolib-acpi.c index d54fc6e7c8a9..fdee8afa5339 100644 --- a/drivers/gpio/gpiolib-acpi.c +++ b/drivers/gpio/gpiolib-acpi.c @@ -383,6 +383,13 @@ int acpi_dev_add_driver_gpios(struct acpi_device *adev, } EXPORT_SYMBOL_GPL(acpi_dev_add_driver_gpios); +void acpi_dev_remove_driver_gpios(struct acpi_device *adev) +{ + if (adev) + adev->driver_gpios = NULL; +} +EXPORT_SYMBOL_GPL(acpi_dev_remove_driver_gpios); + static void devm_acpi_dev_release_driver_gpios(struct device *dev, void *res) { acpi_dev_remove_driver_gpios(ACPI_COMPANION(dev)); diff --git a/include/linux/acpi.h b/include/linux/acpi.h index 9426b9aaed86..e40e1e27ed8e 100644 --- a/include/linux/acpi.h +++ b/include/linux/acpi.h @@ -994,62 +994,11 @@ void __acpi_handle_debug(struct _ddebug *descriptor, acpi_handle handle, const c #endif #endif -struct acpi_gpio_params { - unsigned int crs_entry_index; - unsigned int line_index; - bool active_low; -}; - -struct acpi_gpio_mapping { - const char *name; - const struct acpi_gpio_params *data; - unsigned int size; - -/* Ignore IoRestriction field */ -#define ACPI_GPIO_QUIRK_NO_IO_RESTRICTION BIT(0) -/* - * When ACPI GPIO mapping table is in use the index parameter inside it - * refers to the GPIO resource in _CRS method. That index has no - * distinction of actual type of the resource. When consumer wants to - * get GpioIo type explicitly, this quirk may be used. - */ -#define ACPI_GPIO_QUIRK_ONLY_GPIOIO BIT(1) - - unsigned int quirks; -}; - #if defined(CONFIG_ACPI) && defined(CONFIG_GPIOLIB) -int acpi_dev_add_driver_gpios(struct acpi_device *adev, - const struct acpi_gpio_mapping *gpios); - -static inline void acpi_dev_remove_driver_gpios(struct acpi_device *adev) -{ - if (adev) - adev->driver_gpios = NULL; -} - -int devm_acpi_dev_add_driver_gpios(struct device *dev, - const struct acpi_gpio_mapping *gpios); -void devm_acpi_dev_remove_driver_gpios(struct device *dev); - bool acpi_gpio_get_irq_resource(struct acpi_resource *ares, struct acpi_resource_gpio **agpio); int acpi_dev_gpio_irq_get(struct acpi_device *adev, int index); #else -static inline int acpi_dev_add_driver_gpios(struct acpi_device *adev, - const struct acpi_gpio_mapping *gpios) -{ - return -ENXIO; -} -static inline void acpi_dev_remove_driver_gpios(struct acpi_device *adev) {} - -static inline int devm_acpi_dev_add_driver_gpios(struct device *dev, - const struct acpi_gpio_mapping *gpios) -{ - return -ENXIO; -} -static inline void devm_acpi_dev_remove_driver_gpios(struct device *dev) {} - static inline bool acpi_gpio_get_irq_resource(struct acpi_resource *ares, struct acpi_resource_gpio **agpio) { diff --git a/include/linux/gpio/consumer.h b/include/linux/gpio/consumer.h index b548a3fb5eee..b70af921c614 100644 --- a/include/linux/gpio/consumer.h +++ b/include/linux/gpio/consumer.h @@ -602,6 +602,63 @@ struct gpio_desc *devm_gpiod_get_from_of_node(struct device *dev, #endif /* CONFIG_GPIOLIB */ +struct acpi_gpio_params { + unsigned int crs_entry_index; + unsigned int line_index; + bool active_low; +}; + +struct acpi_gpio_mapping { + const char *name; + const struct acpi_gpio_params *data; + unsigned int size; + +/* Ignore IoRestriction field */ +#define ACPI_GPIO_QUIRK_NO_IO_RESTRICTION BIT(0) +/* + * When ACPI GPIO mapping table is in use the index parameter inside it + * refers to the GPIO resource in _CRS method. That index has no + * distinction of actual type of the resource. When consumer wants to + * get GpioIo type explicitly, this quirk may be used. + */ +#define ACPI_GPIO_QUIRK_ONLY_GPIOIO BIT(1) + + unsigned int quirks; +}; + +#if IS_ENABLED(CONFIG_GPIOLIB) && IS_ENABLED(CONFIG_ACPI) + +struct acpi_device; + +int acpi_dev_add_driver_gpios(struct acpi_device *adev, + const struct acpi_gpio_mapping *gpios); +void acpi_dev_remove_driver_gpios(struct acpi_device *adev); + +int devm_acpi_dev_add_driver_gpios(struct device *dev, + const struct acpi_gpio_mapping *gpios); +void devm_acpi_dev_remove_driver_gpios(struct device *dev); + +#else /* CONFIG_GPIOLIB && CONFIG_ACPI */ + +struct acpi_device; + +static inline int acpi_dev_add_driver_gpios(struct acpi_device *adev, + const struct acpi_gpio_mapping *gpios) +{ + return -ENXIO; +} +static inline void acpi_dev_remove_driver_gpios(struct acpi_device *adev) {} + +static inline int devm_acpi_dev_add_driver_gpios(struct device *dev, + const struct acpi_gpio_mapping *gpios) +{ + return -ENXIO; +} +static inline void devm_acpi_dev_remove_driver_gpios(struct device *dev) {} + +#endif /* CONFIG_GPIOLIB && CONFIG_ACPI */ + + #if IS_ENABLED(CONFIG_GPIOLIB) && IS_ENABLED(CONFIG_GPIO_SYSFS) int gpiod_export(struct gpio_desc *desc, bool direction_may_change); diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c index 54ac2fd41925..67f06c95eec5 100644 --- a/sound/soc/intel/boards/bytcht_cx2072x.c +++ b/sound/soc/intel/boards/bytcht_cx2072x.c @@ -6,6 +6,7 @@ #include <linux/acpi.h> #include <linux/device.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 33eb72545be6..05db311b579e 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -12,6 +12,7 @@ */ #include <linux/dmi.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 4977b5a65eb8..9d657421730a 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -8,6 +8,7 @@ * Mengdong Lin <mengdong.lin(a)intel.com> */ +#include <linux/gpio/consumer.h> #include <linux/input.h> #include <linux/module.h> #include <linux/platform_device.h> -- 2.20.1

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