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

[PATCH] MAINTAINERS: Add Shengjiu to maintainer list of sound/soc/fsl
by Nicolin Chen 18 Mar '22

18 Mar '22

Shengjiu has been actively working on latest FSL platforms and keeping upstream effort as well, while I have been working on other subsystem lately and cannot guarantee audio patch review in the near term. So replacing with him in the maintainer list. Cc: Shengjiu Wang <shengjiu.wang(a)gmail.com> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- MAINTAINERS | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 46ffe05eaeb7..bf7a0ae10d06 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7779,10 +7779,10 @@ F: drivers/net/ethernet/freescale/fs_enet/ F: include/linux/fs_enet_pd.h FREESCALE SOC SOUND DRIVERS -M: Nicolin Chen <nicoleotsuka(a)gmail.com> +M: Shengjiu Wang <shengjiu.wang(a)gmail.com> M: Xiubo Li <Xiubo.Lee(a)gmail.com> R: Fabio Estevam <festevam(a)gmail.com> -R: Shengjiu Wang <shengjiu.wang(a)gmail.com> +R: Nicolin Chen <nicoleotsuka(a)gmail.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) L: linuxppc-dev(a)lists.ozlabs.org S: Maintained -- 2.17.1

3 2

[PATCH 00/19] Make the SOF control, PCM and PM code IPC agnostic
by Ranjani Sridharan 18 Mar '22

18 Mar '22

This series is a continuation to the SOF IPC abstraction work to support the new IPC version introduced in the SOF firmware. It makes the top-level control IO, PCM and PM code IPC-agnostic. Other than the first patch, the rest are purely for abstraction and include no changes in functionality. Ranjani Sridharan (19): ASoC: SOF: set up scheduler widget before all other widgets in the pipeline ASoC: SOF: Make sof_widget_setup/free IPC agnostic ASoC: SOF: Make sof_suspend/resume IPC agnostic ASoC: SOF: Introduce IPC ops for kcontrol IO ASoC: SOF: Add IPC3 topology control ops ASoC: SOF: Add volume_get/put IPC3 ops ASoC: SOF: Add switch get/put IPC3 ops ASoC: SOF: Add enum_get/put control ops for IPC3 ASoC: SOF: Add bytes_get/put control IPC ops for IPC3 ASoC: SOF: Add bytes_ext control IPC ops for IPC3 ASoC: SOF: Introduce IPC-specific PCM ops ASoC: SOF: pcm: expose the sof_pcm_setup_connected_widgets() function ASoC: SOF: Introduce IPC3 PCM hw_free op ASoC: SOF: Define hw_params PCM op for IPC3 ASoC: SOF: Add trigger PCM op for IPC3 ASoC: SOF: Add dai_link_fixup PCM op for IPC3 ASoC: SOF: expose sof_route_setup() ASoC: SOF: topology: Add ops for setting up and tearing down pipelines ASoC: SOF: Add a new dai_get_clk topology IPC op sound/soc/sof/Makefile | 2 +- sound/soc/sof/control.c | 545 ++++--------------------------- sound/soc/sof/intel/hda-dai.c | 97 +----- sound/soc/sof/intel/hda.c | 169 +++------- sound/soc/sof/intel/hda.h | 7 +- sound/soc/sof/ipc.c | 10 +- sound/soc/sof/ipc3-control.c | 594 ++++++++++++++++++++++++++++++++++ sound/soc/sof/ipc3-ops.h | 21 ++ sound/soc/sof/ipc3-pcm.c | 372 +++++++++++++++++++++ sound/soc/sof/ipc3-topology.c | 384 +++++++++++++++++++++- sound/soc/sof/ipc3.c | 45 +++ sound/soc/sof/pcm.c | 354 +++----------------- sound/soc/sof/pm.c | 80 ++--- sound/soc/sof/sof-audio.c | 442 +++++-------------------- sound/soc/sof/sof-audio.h | 70 +++- sound/soc/sof/sof-priv.h | 17 +- sound/soc/sof/topology.c | 55 ++-- 17 files changed, 1805 insertions(+), 1459 deletions(-) create mode 100644 sound/soc/sof/ipc3-control.c create mode 100644 sound/soc/sof/ipc3-ops.h create mode 100644 sound/soc/sof/ipc3-pcm.c create mode 100644 sound/soc/sof/ipc3.c -- 2.25.1

2 20

[PATCH 0/6] Add new HW variants and fix few minor issues
by Vitaly Rodionov 18 Mar '22

18 Mar '22

This series adds new HW variants for Dell's Dolphin and Warlock, and also fixies few very minor issues. Stefan Binding (5): ALSA: hda/cs8409: Fix Warlock using dual mic configuration ALSA: hda/cs8409: Re-order quirk table into ascending order ALSA: hda/cs8409: Fix Full Scale Volume setting for all variants ALSA: hda/cs8409: Support new Warlock MLK Variants ALSA: hda/cs8409: Disable HSBIAS_SENSE_EN for Cyborg Vitaly Rodionov (1): ALSA: hda/cs8409: Add new Dolphin HW variants sound/pci/hda/patch_cs8409-tables.c | 68 +++++++++++++++++++++-------- sound/pci/hda/patch_cs8409.c | 47 +++++++++++--------- sound/pci/hda/patch_cs8409.h | 5 +++ 3 files changed, 83 insertions(+), 37 deletions(-) -- 2.25.1

2 7

[RESEND v6 0/1] firmware: mtk: add adsp ipc protocol for sof
by Allen-KH Cheng 18 Mar '22

18 Mar '22

This patch provides mtk adsp ipc support for sof. ADSP IPC protocol offers (send/recv) interfaces using mediatek-mailbox APIs. This was tested and confirmed on MT8195 cherry board with sof. Based on linux-next changes since v5: - fix WARNING: modpost: missing MODULE_LICENSE() in drivers/mailbox /mtk-adsp-mailbox.o. Add MODULE_LICENSE in the last line. - Due to WARNING: Missing or malformed SPDX-License-Identifier tag in line 1 in checkpatch, we don't remove SPDX-License in line 1. changes since v4: - add error message for wrong mbox chan changes since v3: - rebase on v5.16-rc8 - update reviewers changes since v2: - add out tag for two memory free phases changes since v1: - add comments for mtk_adsp_ipc_send and mtk_adsp_ipc_recv - remove useless MODULE_LICENSE - change label name to out_free Allen-KH Cheng (1): firmware: mediatek: add adsp ipc protocol interface drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/mediatek/Kconfig | 9 + drivers/firmware/mediatek/Makefile | 2 + drivers/firmware/mediatek/mtk-adsp-ipc.c | 161 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 65 +++++++ 6 files changed, 239 insertions(+) create mode 100644 drivers/firmware/mediatek/Kconfig create mode 100644 drivers/firmware/mediatek/Makefile create mode 100644 drivers/firmware/mediatek/mtk-adsp-ipc.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

2 2

[PATCH] ALSA: oss: Release temporary buffers upon errors
by Takashi Iwai 18 Mar '22

18 Mar '22

When the parameter changes fails, we don't need to keep the old temporary buffers. Release those (and plugin instances) upon errors for reducing dead memory footprint. Since we always call it at the exit of snd_pcm_oss_changes_params_locked(), the explicit calls of snd_pcm_oss_plugin_clear() can be dropped, too. Along with it, unify the buffer-free calls to a single helper and call it from the needed places. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_oss.c | 28 +++++++++++++++------------- 1 file changed, 15 insertions(+), 13 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index f158f0abd25d..90c3a367d7de 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -842,6 +842,17 @@ static void unlock_params(struct snd_pcm_runtime *runtime) mutex_unlock(&runtime->oss.params_lock); } +static void snd_pcm_oss_release_buffers(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + kvfree(runtime->oss.buffer); + runtime->oss.buffer = NULL; +#ifdef CONFIG_SND_PCM_OSS_PLUGINS + snd_pcm_oss_plugin_clear(substream); +#endif +} + /* call with params_lock held */ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) { @@ -972,12 +983,10 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) snd_pcm_oss_plugin_clear(substream); if (!direct) { /* add necessary plugins */ - snd_pcm_oss_plugin_clear(substream); err = snd_pcm_plug_format_plugins(substream, params, sparams); if (err < 0) { pcm_dbg(substream->pcm, "snd_pcm_plug_format_plugins failed: %i\n", err); - snd_pcm_oss_plugin_clear(substream); goto failure; } if (runtime->oss.plugin_first) { @@ -986,7 +995,6 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) if (err < 0) { pcm_dbg(substream->pcm, "snd_pcm_plugin_build_io failed: %i\n", err); - snd_pcm_oss_plugin_clear(substream); goto failure; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -994,10 +1002,8 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) } else { err = snd_pcm_plugin_insert(plugin); } - if (err < 0) { - snd_pcm_oss_plugin_clear(substream); + if (err < 0) goto failure; - } } } #endif @@ -1086,6 +1092,8 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) err = 0; failure: + if (err) + snd_pcm_oss_release_buffers(substream); kfree(sw_params); kfree(params); kfree(sparams); @@ -2355,13 +2363,7 @@ static void snd_pcm_oss_look_for_setup(struct snd_pcm *pcm, int stream, static void snd_pcm_oss_release_substream(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime; - runtime = substream->runtime; - kvfree(runtime->oss.buffer); - runtime->oss.buffer = NULL; -#ifdef CONFIG_SND_PCM_OSS_PLUGINS - snd_pcm_oss_plugin_clear(substream); -#endif + snd_pcm_oss_release_buffers(substream); substream->oss.oss = 0; } -- 2.34.1

1 0

[PATCH] ALSA: oss: Fix PCM OSS buffer allocation overflow
by Takashi Iwai 18 Mar '22

18 Mar '22

We've got syzbot reports hitting INT_MAX overflow at vmalloc() allocation that is called from snd_pcm_plug_alloc(). Although we apply the restrictions to input parameters, it's based only on the hw_params of the underlying PCM device. Since the PCM OSS layer allocates a temporary buffer for the data conversion, the size may become unexpectedly large when more channels or higher rates is given; in the reported case, it went over INT_MAX, hence it hits WARN_ON(). This patch is an attempt to avoid such an overflow and an allocation for too large buffers. First off, it adds the limit of 1MB as the upper bound for period bytes. This must be large enough for all use cases, and we really don't want to handle a larger temporary buffer than this size. The size check is performed at two places, where the original period bytes is calculated and where the plugin buffer size is calculated. In addition, the driver uses array_size() and array3_size() for multiplications to catch overflows for the converted period size and buffer bytes. Reported-by: syzbot+72732c532ac1454eeee9(a)syzkaller.appspotmail.com Suggested-by: Linus Torvalds <torvalds(a)linux-foundation.org> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/00000000000085b1b305da5a66f3@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_oss.c | 12 ++++++++---- sound/core/oss/pcm_plugin.c | 5 ++++- 2 files changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 3ee9edf85815..f158f0abd25d 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -774,6 +774,11 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, if (oss_period_size < 16) return -EINVAL; + + /* don't allocate too large period; 1MB period must be enough */ + if (oss_period_size > 1024 * 1024) + return -ENOMEM; + runtime->oss.period_bytes = oss_period_size; runtime->oss.period_frames = 1; runtime->oss.periods = oss_periods; @@ -1043,10 +1048,9 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) goto failure; } #endif - oss_period_size *= oss_frame_size; - - oss_buffer_size = oss_period_size * runtime->oss.periods; - if (oss_buffer_size < 0) { + oss_period_size = array_size(oss_period_size, oss_frame_size); + oss_buffer_size = array_size(oss_period_size, runtime->oss.periods); + if (oss_buffer_size <= 0) { err = -EINVAL; goto failure; } diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 061ba06bc926..82e180c776ae 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -62,7 +62,10 @@ static int snd_pcm_plugin_alloc(struct snd_pcm_plugin *plugin, snd_pcm_uframes_t width = snd_pcm_format_physical_width(format->format); if (width < 0) return width; - size = frames * format->channels * width; + size = array3_size(frames, format->channels, width); + /* check for too large period size once again */ + if (size > 1024 * 1024) + return -ENOMEM; if (snd_BUG_ON(size % 8)) return -ENXIO; size /= 8; -- 2.34.1

1 0

[PATCH] ALSA: hda/realtek: Fix LED on Zbook Studio G9
by Kai-Heng Feng 18 Mar '22

18 Mar '22

Commit 07bcab93946c ("ALSA: hda/realtek: Add support for HP Laptops") breaks mute and micmute LEDs because it changed the LED quirk from ALC285_FIXUP_HP_GPIO_LED to ALC245_FIXUP_HP_GPIO_LED, so change it back here. Also reorder the chain of quirks to ensure LED quirk is the last one being applied. Fixes: 07bcab93946c ("ALSA: hda/realtek: Add support for HP Laptops") Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4650ef9110d62..1d14be4ee31d8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8773,9 +8773,9 @@ static const struct hda_fixup alc269_fixups[] = { }, [ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, - .v.func = alc245_fixup_hp_gpio_led, + .v.func = cs35l41_fixup_spi_four, .chained = true, - .chain_id = ALC245_FIXUP_CS35L41_SPI_4, + .chain_id = ALC285_FIXUP_HP_GPIO_LED, }, [ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED] = { .type = HDA_FIXUP_VERBS, -- 2.34.1

2 1

Re: [PATCH 10/19] dma: imx-sdma: Add multi fifo support
by Sascha Hauer 18 Mar '22

18 Mar '22

On Fri, Mar 18, 2022 at 01:42:51PM +0800, Shengjiu Wang wrote: > Hi > On Thu, Mar 17, 2022 at 4:28 PM Sascha Hauer <[1]s.hauer(a)pengutronix.de> > wrote: > > Signed-off-by: Sascha Hauer <[2]s.hauer(a)pengutronix.de> > --- > drivers/dma/imx-sdma.c | 54 +++++++++++++++++++++++++++ > include/linux/platform_data/dma-imx.h | 7 ++++ > 2 files changed, 61 insertions(+) > > diff --git a/drivers/dma/imx-sdma.c b/drivers/dma/imx-sdma.c > index 1038f6bc7f846..21e1cec2ffde9 100644 > --- a/drivers/dma/imx-sdma.c > +++ b/drivers/dma/imx-sdma.c > @@ -14,6 +14,7 @@ > #include <linux/iopoll.h> > #include <linux/module.h> > #include <linux/types.h> > +#include <linux/bitfield.h> > #include <linux/bitops.h> > #include <linux/mm.h> > #include <linux/interrupt.h> > @@ -73,6 +74,7 @@ > #define SDMA_CHNENBL0_IMX35 0x200 > #define SDMA_CHNENBL0_IMX31 0x080 > #define SDMA_CHNPRI_0 0x100 > +#define SDMA_DONE0_CONFIG 0x1000 > > /* > * Buffer descriptor status values. > @@ -180,6 +182,12 @@ > BIT(DMA_MEM_TO_DEV) | \ > BIT(DMA_DEV_TO_DEV)) > > +#define SDMA_WATERMARK_LEVEL_N_FIFOS GENMASK(15, 12) > +#define SDMA_WATERMARK_LEVEL_SW_DONE BIT(23) > + > +#define SDMA_DONE0_CONFIG_DONE_SEL BIT(7) > +#define SDMA_DONE0_CONFIG_DONE_DIS BIT(6) > + > /** > * struct sdma_script_start_addrs - SDMA script start pointers > * > @@ -441,6 +449,11 @@ struct sdma_channel { > struct work_struct terminate_worker; > struct list_head terminated; > bool is_ram_script; > + unsigned int n_fifos; > + unsigned int n_fifos_src; > + unsigned int n_fifos_dst; > + bool sw_done; > + u32 sw_done_sel; > > "sw_done_sel" is not used, and may not be needed. Ok, will drop. > And can we just add 'struct sdma_peripheral_config *pconfig' > to replace each item here ('n_fifos_src', 'n_fifos_dst', > 'sw_done')? I rather do not access the pointer to the peripheral_config outside of sdma_config because I know nothing about the lifetime of that structure. > the pconfig can point to the struct in dma_slave_config. > And 'n_fifos' can be moved to locally function in > sdma_set_watermarklevel_for_sais(), then use sdmac->direction > to select 'n_fifos_dst' or 'n_fifos_src'. Ok. Sascha -- Pengutronix e.K. | | Steuerwalder Str. 21 | http://www.pengutronix.de/ | 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

1 0

[PATCH] ASoC: fsl_spdif: Add new registers included on i.MX8ULP
by Shengjiu Wang 17 Mar '22

17 Mar '22

There are some new registers added on i.MX8ULP, they are the SPDIF transmit Professional C channel registers, 192bit SPDIF receive C channel registers, and 192bit SPDIF transmit C channel registers. There are two output lines, SPDIF_OUT1 and SPDIF_OUT2, the original REG_SPDIF_STCSCH and REG_SPDIF_STCSCL are used for SPDIF_OUT1, the new REG_SPDIF_STCSPH and REG_SPDIF_STCSPL are used for SPDIF_OUT2, the 192bit SPDIF C channel registers are used for both. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 52 ++++++++++++++++++++++++++++++++++++++- sound/soc/fsl/fsl_spdif.h | 14 +++++++++++ 2 files changed, 65 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index b502e7c3c04d..42d11aca38a1 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -50,6 +50,7 @@ static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb }; * @shared_root_clock: flag of sharing a clock source with others; * so the driver shouldn't set root clock rate * @raw_capture_mode: if raw capture mode support + * @cchannel_192b: if there are registers for 192bits C channel data * @interrupts: interrupt number * @tx_burst: tx maxburst size * @rx_burst: rx maxburst size @@ -59,6 +60,7 @@ struct fsl_spdif_soc_data { bool imx; bool shared_root_clock; bool raw_capture_mode; + bool cchannel_192b; u32 interrupts; u32 tx_burst; u32 rx_burst; @@ -196,6 +198,7 @@ static struct fsl_spdif_soc_data fsl_spdif_imx8ulp = { .tx_burst = 2, /* Applied for EDMA */ .rx_burst = 2, /* Applied for EDMA */ .tx_formats = SNDRV_PCM_FMTBIT_S24_LE, /* Applied for EDMA */ + .cchannel_192b = true, }; /* Check if clk is a root clock that does not share clock source with others */ @@ -441,6 +444,23 @@ static void spdif_write_channel_status(struct fsl_spdif_priv *spdif_priv) regmap_write(regmap, REG_SPDIF_STCSCL, ch_status); dev_dbg(&pdev->dev, "STCSCL: 0x%06x\n", ch_status); + + if (spdif_priv->soc->cchannel_192b) { + ch_status = (bitrev8(ctrl->ch_status[0]) << 24) | + (bitrev8(ctrl->ch_status[1]) << 16) | + (bitrev8(ctrl->ch_status[2]) << 8) | + bitrev8(ctrl->ch_status[3]); + + regmap_update_bits(regmap, REG_SPDIF_SCR, 0x1000000, 0x1000000); + + /* + * The first 32bit should be in REG_SPDIF_STCCA_31_0 register, + * but here we need to set REG_SPDIF_STCCA_191_160 on 8ULP + * then can get correct result with HDMI analyzer capture. + * There is a hardware bug here. + */ + regmap_write(regmap, REG_SPDIF_STCCA_191_160, ch_status); + } } /* Set SPDIF PhaseConfig register for rx clock */ @@ -1229,6 +1249,8 @@ static const struct reg_default fsl_spdif_reg_defaults[] = { {REG_SPDIF_STR, 0x00000000}, {REG_SPDIF_STCSCH, 0x00000000}, {REG_SPDIF_STCSCL, 0x00000000}, + {REG_SPDIF_STCSPH, 0x00000000}, + {REG_SPDIF_STCSPL, 0x00000000}, {REG_SPDIF_STC, 0x00020f00}, }; @@ -1248,8 +1270,22 @@ static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg) case REG_SPDIF_SRQ: case REG_SPDIF_STCSCH: case REG_SPDIF_STCSCL: + case REG_SPDIF_STCSPH: + case REG_SPDIF_STCSPL: case REG_SPDIF_SRFM: case REG_SPDIF_STC: + case REG_SPDIF_SRCCA_31_0: + case REG_SPDIF_SRCCA_63_32: + case REG_SPDIF_SRCCA_95_64: + case REG_SPDIF_SRCCA_127_96: + case REG_SPDIF_SRCCA_159_128: + case REG_SPDIF_SRCCA_191_160: + case REG_SPDIF_STCCA_31_0: + case REG_SPDIF_STCCA_63_32: + case REG_SPDIF_STCCA_95_64: + case REG_SPDIF_STCCA_127_96: + case REG_SPDIF_STCCA_159_128: + case REG_SPDIF_STCCA_191_160: return true; default: return false; @@ -1268,6 +1304,12 @@ static bool fsl_spdif_volatile_reg(struct device *dev, unsigned int reg) case REG_SPDIF_SRU: case REG_SPDIF_SRQ: case REG_SPDIF_SRFM: + case REG_SPDIF_SRCCA_31_0: + case REG_SPDIF_SRCCA_63_32: + case REG_SPDIF_SRCCA_95_64: + case REG_SPDIF_SRCCA_127_96: + case REG_SPDIF_SRCCA_159_128: + case REG_SPDIF_SRCCA_191_160: return true; default: return false; @@ -1286,7 +1328,15 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) case REG_SPDIF_STR: case REG_SPDIF_STCSCH: case REG_SPDIF_STCSCL: + case REG_SPDIF_STCSPH: + case REG_SPDIF_STCSPL: case REG_SPDIF_STC: + case REG_SPDIF_STCCA_31_0: + case REG_SPDIF_STCCA_63_32: + case REG_SPDIF_STCCA_95_64: + case REG_SPDIF_STCCA_127_96: + case REG_SPDIF_STCCA_159_128: + case REG_SPDIF_STCCA_191_160: return true; default: return false; @@ -1298,7 +1348,7 @@ static const struct regmap_config fsl_spdif_regmap_config = { .reg_stride = 4, .val_bits = 32, - .max_register = REG_SPDIF_STC, + .max_register = REG_SPDIF_STCCA_191_160, .reg_defaults = fsl_spdif_reg_defaults, .num_reg_defaults = ARRAY_SIZE(fsl_spdif_reg_defaults), .readable_reg = fsl_spdif_readable_reg, diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h index bff8290e71f2..75b42a692c90 100644 --- a/sound/soc/fsl/fsl_spdif.h +++ b/sound/soc/fsl/fsl_spdif.h @@ -31,9 +31,23 @@ #define REG_SPDIF_STR 0x30 /* SPDIFTxRight Register */ #define REG_SPDIF_STCSCH 0x34 /* SPDIFTxCChannelCons_h Register */ #define REG_SPDIF_STCSCL 0x38 /* SPDIFTxCChannelCons_l Register */ +#define REG_SPDIF_STCSPH 0x3C /* SPDIFTxCChannel_Prof_h Register */ +#define REG_SPDIF_STCSPL 0x40 /* SPDIFTxCChannel_Prof_l Register */ #define REG_SPDIF_SRFM 0x44 /* FreqMeas Register */ #define REG_SPDIF_STC 0x50 /* SPDIFTxClk Register */ +#define REG_SPDIF_SRCCA_31_0 0x60 /* SPDIF receive C channel register, bits 31-0 */ +#define REG_SPDIF_SRCCA_63_32 0x64 /* SPDIF receive C channel register, bits 63-32 */ +#define REG_SPDIF_SRCCA_95_64 0x68 /* SPDIF receive C channel register, bits 95-64 */ +#define REG_SPDIF_SRCCA_127_96 0x6C /* SPDIF receive C channel register, bits 127-96 */ +#define REG_SPDIF_SRCCA_159_128 0x70 /* SPDIF receive C channel register, bits 159-128 */ +#define REG_SPDIF_SRCCA_191_160 0x74 /* SPDIF receive C channel register, bits 191-160 */ +#define REG_SPDIF_STCCA_31_0 0x78 /* SPDIF transmit C channel register, bits 31-0 */ +#define REG_SPDIF_STCCA_63_32 0x7C /* SPDIF transmit C channel register, bits 63-32 */ +#define REG_SPDIF_STCCA_95_64 0x80 /* SPDIF transmit C channel register, bits 95-64 */ +#define REG_SPDIF_STCCA_127_96 0x84 /* SPDIF transmit C channel register, bits 127-96 */ +#define REG_SPDIF_STCCA_159_128 0x88 /* SPDIF transmit C channel register, bits 159-128 */ +#define REG_SPDIF_STCCA_191_160 0x8C /* SPDIF transmit C channel register, bits 191-160 */ /* SPDIF Configuration register */ #define SCR_RXFIFO_CTL_OFFSET 23 -- 2.17.1

2 1

Re: [PATCH] ASoC: SOF: topology: Use kmemdup() to replace kzalloc + memcpy
by Mark Brown 17 Mar '22

17 Mar '22

On Thu, 17 Mar 2022 02:38:41 -0700, Yihao Han wrote: > fix memdup.cocci warning: > sound/soc/sof/topology.c:876:19-26: WARNING opportunity for kmemdup > > Generated by: scripts/coccinelle/api/memdup.cocci > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: SOF: topology: Use kmemdup() to replace kzalloc + memcpy commit: b26f965f7c83464f60d5af411ef9a03079e41800 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

1 0