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[PATCH v1] ASoC: meson: axg-card: fix 'use-after-free'
by Arseniy Krasnov 11 Sep '24

11 Sep '24

Buffer 'card->dai_link' is reallocated in 'meson_card_reallocate_links()', so move 'pad' pointer initialization after this function when memory is already reallocated. Kasan bug report: ================================================================== BUG: KASAN: slab-use-after-free in axg_card_add_link+0x76c/0x9bc Read of size 8 at addr ffff000000e8b260 by task modprobe/356 CPU: 0 PID: 356 Comm: modprobe Tainted: G O 6.9.12-sdkernel #1 Call trace: dump_backtrace+0x94/0xec show_stack+0x18/0x24 dump_stack_lvl+0x78/0x90 print_report+0xfc/0x5c0 kasan_report+0xb8/0xfc __asan_load8+0x9c/0xb8 axg_card_add_link+0x76c/0x9bc [snd_soc_meson_axg_sound_card] meson_card_probe+0x344/0x3b8 [snd_soc_meson_card_utils] platform_probe+0x8c/0xf4 really_probe+0x110/0x39c __driver_probe_device+0xb8/0x18c driver_probe_device+0x108/0x1d8 __driver_attach+0xd0/0x25c bus_for_each_dev+0xe0/0x154 driver_attach+0x34/0x44 bus_add_driver+0x134/0x294 driver_register+0xa8/0x1e8 __platform_driver_register+0x44/0x54 axg_card_pdrv_init+0x20/0x1000 [snd_soc_meson_axg_sound_card] do_one_initcall+0xdc/0x25c do_init_module+0x10c/0x334 load_module+0x24c4/0x26cc init_module_from_file+0xd4/0x128 __arm64_sys_finit_module+0x1f4/0x41c invoke_syscall+0x60/0x188 el0_svc_common.constprop.0+0x78/0x13c do_el0_svc+0x30/0x40 el0_svc+0x38/0x78 el0t_64_sync_handler+0x100/0x12c el0t_64_sync+0x190/0x194 Fixes: 7864a79f37b5 ("ASoC: meson: add axg sound card support") Cc: <Stable(a)vger.kernel.org> Signed-off-by: Arseniy Krasnov <avkrasnov(a)salutedevices.com> --- sound/soc/meson/axg-card.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 8c5605c1e34e8..eb0302f207407 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -104,7 +104,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card, int *index) { struct meson_card *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_dai_link *pad = &card->dai_link[*index]; + struct snd_soc_dai_link *pad; struct snd_soc_dai_link *lb; struct snd_soc_dai_link_component *dlc; int ret; @@ -114,6 +114,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card, if (ret) return ret; + pad = &card->dai_link[*index]; lb = &card->dai_link[*index + 1]; lb->name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-lb", pad->name); -- 2.30.1

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[PATCH v3] ASoC: dt-bindings: renesas,rsnd: add post-init-providers property
by Kuninori Morimoto 11 Sep '24

11 Sep '24

At least if rsnd is using DPCM connection on Audio-Graph-Card2, fw_devlink might doesn't have enough information to break the cycle (Same problem might occur with Multi-CPU/Codec or Codec2Codec). In such case, rsnd driver will not be probed. Add post-init-providers support to break the link cycle. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v2 -> v3 - remove "|" from description - removed common post-init-providers description - 100 char to 80 char for 1 line Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 07ec6247d9def..3bc93c59535e9 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -112,6 +112,12 @@ properties: description: List of necessary clock names. # details are defined below + post-init-providers: + description: At least if rsnd is using DPCM connection on Audio-Graph-Card2, + fw_devlink might doesn't have enough information to break the cycle. rsnd + driver will not be probed in such case. Same problem might occur with + Multi-CPU/Codec or Codec2Codec. + # ports is below port: $ref: audio-graph-port.yaml#/definitions/port-base -- 2.43.0

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[PATCH] ASoC: dt-bindings: microchip,sama7g5-spdifrx: Add common DAI reference
by Andrei Simion 11 Sep '24

11 Sep '24

Update the spdifrx yaml file to reference the dai-common.yaml schema, enabling the use of the 'sound-name-prefix' property Signed-off-by: Andrei Simion <andrei.simion(a)microchip.com> --- .../devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml index 2f43c684ab88..7fbab5871be4 100644 --- a/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml @@ -13,6 +13,9 @@ description: The Microchip Sony/Philips Digital Interface Receiver is a serial port compliant with the IEC-60958 standard. +allOf: + - $ref: dai-common.yaml# + properties: "#sound-dai-cells": const: 0 @@ -53,7 +56,7 @@ required: - dmas - dma-names -additionalProperties: false +unevaluatedProperties: false examples: - | base-commit: 6708132e80a2ced620bde9b9c36e426183544a23 -- 2.34.1

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[PATCH] ASoC: codecs: avoid possible garbage value in peb2466_reg_read()
by Su Hui 11 Sep '24

11 Sep '24

Clang static checker (scan-build) warning: sound/soc/codecs/peb2466.c:232:8: Assigned value is garbage or undefined [core.uninitialized.Assign] 232 | *val = tmp; | ^ ~~~ When peb2466_read_byte() fails, 'tmp' will have a garbage value. Add a judgemnet to avoid this problem. Fixes: 227f609c7c0e ("ASoC: codecs: Add support for the Infineon PEB2466 codec") Signed-off-by: Su Hui <suhui(a)nfschina.com> --- sound/soc/codecs/peb2466.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/peb2466.c b/sound/soc/codecs/peb2466.c index 76ee7e3f4d9b..67ea70cef0c7 100644 --- a/sound/soc/codecs/peb2466.c +++ b/sound/soc/codecs/peb2466.c @@ -229,7 +229,8 @@ static int peb2466_reg_read(void *context, unsigned int reg, unsigned int *val) case PEB2466_CMD_XOP: case PEB2466_CMD_SOP: ret = peb2466_read_byte(peb2466, reg, &tmp); - *val = tmp; + if (!ret) + *val = tmp; break; default: dev_err(&peb2466->spi->dev, "Not a XOP or SOP command\n"); -- 2.30.2

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[PATCH 0/1] Add support for primary mi2s on SM8250
by Jens Reidel 11 Sep '24

11 Sep '24

This patch adds support for the primary mi2s interface on devices using SM8250 audio drivers. Tested on SM7150 (xiaomi-davinci). SM7150 sound is close to SM8250 and we intend to use it as a fallback in the future. To: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> To: Liam Girdwood <lgirdwood(a)gmail.com> To: Mark Brown <broonie(a)kernel.org> To: Jaroslav Kysela <perex(a)perex.cz> To: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-arm-msm(a)vger.kernel.org Cc: linux-sound(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: linux(a)mainlining.org Jens Reidel (1): ASoC: qcom: sm8250: enable primary mi2s sound/soc/qcom/sm8250.c | 8 ++++++++ 1 file changed, 8 insertions(+) -- 2.46.0

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[PATCH 00/13] ALSA: update sample rate definitions
by Jerome Brunet 11 Sep '24

11 Sep '24

This patchset adds rate definitions for 12kHz, 24kHz and 128kHz. It is follow-up on the series/discussion [0] about adding 128kHz for spdif/eARC support. The outcome was to add 12kHz and 24kHz as well and clean up the drivers that no longer require custom rules to allow these rates. Identifying these drivers is not straight forward, I tried my best but I may have missed some. Those will continue to work anyway so it is not critical. Some drivers using these rates have not been changed on purpose. The reason for this may be: * The driver used other uncommon rates that still don't have a definition so a custom rule is still required. * The constraint structure is used in some other way by the driver and removing it would not help the readability or maintainability. This is the case the freescale asrc drivers for example. There is one change per driver so, if there is a problem later on, it will easier to properly add Fixes tag. The series has been tested with * ARM64 defconfig - spdif 128kHz at runtime. * x86_64 allmodconfig - compile test only Last, the change adding IEC958 definitions has been dropped for this patchset and will be resent separately [0]: https://lore.kernel.org/all/20240628122429.2018059-1-jbrunet@baylibre.com/ --- Jerome Brunet (13): ALSA: pcm: add more sample rate definitions ALSA: cmipci: drop SNDRV_PCM_RATE_KNOT ALSA: emu10k1: drop SNDRV_PCM_RATE_KNOT ALSA: hdsp: drop SNDRV_PCM_RATE_KNOT ALSA: hdspm: drop SNDRV_PCM_RATE_KNOT ASoC: cs35l36: drop SNDRV_PCM_RATE_KNOT ASoC: cs35l41: drop SNDRV_PCM_RATE_KNOT ASoC: cs53l30: drop SNDRV_PCM_RATE_KNOT ASoC: Intel: avs: drop SNDRV_PCM_RATE_KNOT ASoC: qcom: q6asm-dai: drop SNDRV_PCM_RATE_KNOT ASoC: sunxi: sun4i-codec: drop SNDRV_PCM_RATE_KNOT ASoC: cs35l34: drop useless rate contraint ASoC: spdif: extend supported rates to 768kHz include/sound/pcm.h | 31 +++++++++++++++++-------------- sound/core/pcm_native.c | 6 +++--- sound/pci/cmipci.c | 32 +++++++++----------------------- sound/pci/emu10k1/emupcm.c | 31 +++++-------------------------- sound/pci/rme9652/hdsp.c | 18 ++++++------------ sound/pci/rme9652/hdspm.c | 16 +--------------- sound/soc/codecs/cs35l34.c | 21 --------------------- sound/soc/codecs/cs35l36.c | 34 ++++++++++++---------------------- sound/soc/codecs/cs35l41.c | 34 +++++++++++----------------------- sound/soc/codecs/cs53l30.c | 24 +++--------------------- sound/soc/codecs/spdif_receiver.c | 3 ++- sound/soc/codecs/spdif_transmitter.c | 3 ++- sound/soc/intel/avs/pcm.c | 22 ++++++---------------- sound/soc/qcom/qdsp6/q6asm-dai.c | 31 ++++++++++--------------------- sound/soc/sunxi/sun4i-codec.c | 28 +++++++++------------------- 15 files changed, 96 insertions(+), 238 deletions(-) --- base-commit: 785f4052380684dbcc156203c537c799e2f4be09 change-id: 20240905-alsa-12-24-128-8edab4c08dd4 Best regards, -- Jerome

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[PATCH 1/2] ASoC: amd: acp: remove MODULE_ALIAS for legacy machine driver
by Vijendar Mukunda 11 Sep '24

11 Sep '24

As module device table added for AMD legacy machine driver, MODULE_ALIAS is not required. Remove MODULE_ALIAS for AMD legacy machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-legacy-mach.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-mach.c b/sound/soc/amd/acp/acp-legacy-mach.c index 0d529e32e552..d104f7e8fdcd 100644 --- a/sound/soc/amd/acp/acp-legacy-mach.c +++ b/sound/soc/amd/acp/acp-legacy-mach.c @@ -242,11 +242,4 @@ module_platform_driver(acp_asoc_audio); MODULE_IMPORT_NS(SND_SOC_AMD_MACH); MODULE_DESCRIPTION("ACP chrome audio support"); -MODULE_ALIAS("platform:acp3xalc56821019"); -MODULE_ALIAS("platform:acp3xalc5682sm98360"); -MODULE_ALIAS("platform:acp3xalc5682s1019"); -MODULE_ALIAS("platform:acp3x-es83xx"); -MODULE_ALIAS("platform:rmb-nau8825-max"); -MODULE_ALIAS("platform:rmb-rt5682s-rt1019"); -MODULE_ALIAS("platform:acp-pdm-mach"); MODULE_LICENSE("GPL v2"); -- 2.34.1

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[PATCH v1 0/4] Add static channel mapping between soundwire master and slave
by Mohammad Rafi Shaik 11 Sep '24

11 Sep '24

Add static channel map support between soundwire master and slave. This patch series will resolve channel mask mismatch between master and slave. Scenario: wcd937x AMIC2 usecase Master Slave (wcd937x) +--------------+ +--------------+ | +--------+ | | +--------+ | AMIC1 ----->| | PORT1 | | | | TX1 | |<-----------AMIC1 AMIC2 ----->| | | | | | | | | +--------+ | | +--------+ | | | | | AMIC3 ----->| +--------+ | | +--------+ | | | PORT2 | | | | TX2 | |<-----------AMIC2 | | | | | | | |<-----------AMIC3 | +--------+ | | +--------+ | | | | | | +--------+ | | +--------+ | DMIC0...DMIC3------>| | PORT3 | | | | TX3 | |<-----------DMIC0...DMIC3 | | | | | | | |<-----------MBHC | +--------+ | | +--------+ | | | | | | +--------+ | | +--------+ | DMIC4...DMIC37----->| | PORT4 | | | | TX4 | |<-----------DMIC4...DMIC7 | | | | | | | | | +--------+ | | +--------+ | | | | | +------------- + +--------------+ For AMIC2 usecase, The Slave need to configure TX2 Port with channel mask 1 and for Master required PORT1 with channel mask 2, In existing design master and slave configured with same channel mask, it will fail AMIC2 usecase. The New design will help to configure channel mapping between master and slave from device tree. Mohammad Rafi Shaik (4): ASoC: dt-bindings: wcd938x-sdw: Add static channel mapping support soundwire: stream: Add set_master_channel_map() to set static channel mapping soundwire: qcom: Add static channel mapping support in soundwire master ASoC: codecs: wcd937x: Add static channel mapping support in wcd937x-sdw .../bindings/sound/qcom,wcd937x-sdw.yaml | 28 ++++++++++ drivers/soundwire/qcom.c | 18 +++++++ drivers/soundwire/stream.c | 16 ++++++ include/linux/soundwire/sdw.h | 5 ++ sound/soc/codecs/wcd937x-sdw.c | 52 ++++++++++++++++--- sound/soc/codecs/wcd937x.c | 12 ++++- sound/soc/codecs/wcd937x.h | 6 ++- 7 files changed, 126 insertions(+), 11 deletions(-) base-commit: 9aaeb87ce1e966169a57f53a02ba05b30880ffb8 -- 2.25.1

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[PATCH RESEND] soundwire: stream: Revert "soundwire: stream: fix programming slave ports for non-continous port maps"
by Krzysztof Kozlowski 11 Sep '24

11 Sep '24

This reverts commit ab8d66d132bc8f1992d3eb6cab8d32dda6733c84 because it breaks codecs using non-continuous masks in source and sink ports. The commit missed the point that port numbers are not used as indices for iterating over prop.sink_ports or prop.source_ports. Soundwire core and existing codecs expect that the array passed as prop.sink_ports and prop.source_ports is continuous. The port mask still might be non-continuous, but that's unrelated. Reported-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Closes: https://lore.kernel.org/all/b6c75eee-761d-44c8-8413-2a5b34ee2f98@linux.inte… Fixes: ab8d66d132bc ("soundwire: stream: fix programming slave ports for non-continous port maps") Acked-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Resending with Ack/Rb tags and missing Cc-stable. --- drivers/soundwire/stream.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index f275143d7b18..7aa4900dcf31 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1291,18 +1291,18 @@ struct sdw_dpn_prop *sdw_get_slave_dpn_prop(struct sdw_slave *slave, unsigned int port_num) { struct sdw_dpn_prop *dpn_prop; - unsigned long mask; + u8 num_ports; int i; if (direction == SDW_DATA_DIR_TX) { - mask = slave->prop.source_ports; + num_ports = hweight32(slave->prop.source_ports); dpn_prop = slave->prop.src_dpn_prop; } else { - mask = slave->prop.sink_ports; + num_ports = hweight32(slave->prop.sink_ports); dpn_prop = slave->prop.sink_dpn_prop; } - for_each_set_bit(i, &mask, 32) { + for (i = 0; i < num_ports; i++) { if (dpn_prop[i].num == port_num) return &dpn_prop[i]; } -- 2.43.0

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[PATCH] ASoC: tlv320aic31xx: Fix typos
by Andrew Kreimer 11 Sep '24

11 Sep '24

Fix typos in comments. Reported-by: Matthew Wilcox <willy(a)infradead.org> Signed-off-by: Andrew Kreimer <algonell(a)gmail.com> --- sound/soc/codecs/tlv320aic31xx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 7e624c4b77b6..187d68e8688c 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -895,7 +895,7 @@ static int aic31xx_setup_pll(struct snd_soc_component *component, dev_err(component->dev, "%s: Sample rate (%u) and format not supported\n", __func__, params_rate(params)); - /* See bellow for details how fix this. */ + /* See below for details on how to fix this. */ return -EINVAL; } if (bclk_score != 0) { -- 2.46.0

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