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

[PATCH 0/3] ASoC: mt6358: Fixes from an initial glance at a kselftest run
by Mark Brown 28 Feb '23

28 Feb '23

This is a collection of fixes I came up after glancing through an initial test run with the snappily named Kukui Jacuzzi SKU16 Chromebook on KernelCI. There are more issues flagged, this is just what I fixed thus far. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (3): ASoC: mt6358: Fix event generation for wake on voice stage 2 switch ASoC: mt6358: Validate Wake on Voice 2 writes ASoC: mt6358: Remove undefined HPx Mux enumeration values sound/soc/codecs/mt6358.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) --- base-commit: d2980d8d826554fa6981d621e569a453787472f8 change-id: 20230224-asoc-mt6358-quick-fixes-71b4ea6b8f13 Best regards, -- Mark Brown <broonie(a)kernel.org>

2 7

[PATCH 0/4] ASoC: mt8192: Fixes from initial glance at kselftest run
by Mark Brown 28 Feb '23

28 Feb '23

This is a collection of fixes I came up after glancing through an initial test run with the Spherion Chromebook on KernelCI. There are more issues flagged, this is just what I fixed thus far - the volume controls on the MT6359 have issues for example, and a lot of controls aren't marked as Switches like they should be. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (4): ASoC: mt8192: Remove spammy log messages ASoC: mt8192: Fix event generation for controls ASoC: mt8192: Report an error if when an invalid sidetone gain is written ASoC: mt8192: Fix range for sidetone positive gain sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 58 ++++++++++------------------- 1 file changed, 19 insertions(+), 39 deletions(-) --- base-commit: b361d5d2464a88184f6e17a6462719ba79180b1a change-id: 20230223-asoc-mt8192-quick-fixes-8f3e0a187226 Best regards, -- Mark Brown <broonie(a)kernel.org>

3 15

[RFC PATCH v2 00/22] Introduce QC USB SND audio offloading support
by Wesley Cheng 28 Feb '23

28 Feb '23

Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. USB SND: - Added a platform ops during usb_audio_suspend(). This allows for the USB offload driver to halt the audio stream when system enters PM suspend. This ensures the audio DSP is not issuing transfers on the USB bus. - Do not override platform ops if they are already populated. - Introduce a shared status variable between the USB offload and USB SND layers, to ensure that only one path is active at a time. If the USB bus is occupied, then userspace is notified that the path is busy. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Mathias Nyman (4): xhci: fix event ring segment table related masks and variables in header xhci: remove xhci_test_trb_in_td_math early development check xhci: Refactor interrupter code for initial multi interrupter support. xhci: Add support to allocate several interrupters Wesley Cheng (18): usb: xhci: Add XHCI APIs to support USB offloading usb: host: xhci-mem: Cleanup pending secondary event ring events ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: Add USB backend ASoC driver for Q6 sound: usb: card: Introduce USB SND platform op callbacks sound: usb: Export USB SND APIs for modules dt-bindings: usb: dwc3: Add snps,num-hc-interrupters definition usb: dwc3: Add DT parameter to specify maximum number of interrupters sound: usb: Introduce QC USB SND offloading support sound: usb: card: Check for support for requested audio format sound: soc: soc-usb: Add PCM format check API for USB backend sound: soc: qcom: qusb6: Ensure PCM format is supported by USB audio device sound: usb: Prevent starting of audio stream if in use ASoC: dt-bindings: Add Q6USB backend bindings ASoC: dt-bindings: Update example for enabling USB offload on SM8250 .../bindings/sound/qcom,q6usb-dais.yaml | 55 + .../bindings/sound/qcom,sm8250.yaml | 13 + .../devicetree/bindings/usb/snps,dwc3.yaml | 12 + drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 5 +- drivers/usb/host/xhci-debugfs.c | 2 +- drivers/usb/host/xhci-mem.c | 471 +++-- drivers/usb/host/xhci-ring.c | 68 +- drivers/usb/host/xhci.c | 199 +- drivers/usb/host/xhci.h | 81 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-intr.h | 94 + include/sound/pcm_params.h | 4 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 36 + sound/core/oss/pcm_oss.c | 58 - sound/core/pcm_lib.c | 65 + sound/soc/Makefile | 2 +- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 48 + sound/soc/qcom/qdsp6/q6afe.c | 185 +- sound/soc/qcom/qdsp6/q6afe.h | 46 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 8 + sound/soc/qcom/qdsp6/q6usb.c | 234 +++ sound/soc/soc-usb.c | 215 ++ sound/usb/Kconfig | 14 + sound/usb/Makefile | 2 +- sound/usb/card.c | 58 + sound/usb/card.h | 29 + sound/usb/endpoint.c | 2 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 28 +- sound/usb/pcm.h | 12 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1789 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 41 files changed, 4546 insertions(+), 410 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6usb-dais.yaml create mode 100644 include/linux/usb/xhci-intr.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

8 72

[PATCH V4 0/8] Add SoundWire support for AMD platforms
by Vijendar Mukunda 28 Feb '23

28 Feb '23

ACP IP(v6.x) block has two SoundWire manager instance support. This patchset adds support for AMD SoundWire manager driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> changes since v3: - add usleep_range() in command/response implementation. - add usleep_range() in clock stop sequence. - modify usleep_range() values throughout the code. - remove unncessary debug statement from code. - update comment in probe() call. - fix timeout condition checks in the code. - drop "ret" variable in amd_disable_sdw_manager(). Changes since v2: - Remove useless variable initializations. - Add helper function to interpret peripheral status. - Move runtime pm sequence to probe_work workqueue. - Use string "SoundWire" instead of "soundwire" in code. - Update comments in interrupt handler and probe sequence. - Rename "sdw_lock" as "acp_sdw_lock". - Remove __func__ from dev_dbg statements. Changes since v1: - Drop asoc tree based patches. will send asoc patches as a separate series. - Fixed double space errors. - Use dev instead of pci->dev. - Use SoundWire manager terminology. - Remove amd_sdw_compute_slave_ports() function and use exported sdw_compute_slave_ports() function. - Remove unused variable "num_ports" from amd_manager structure. - Drop startup and shutdown dai callbacks. - Drop reset_page_addr callback. - Use relative address offset to program SoundWire manager registers throughout the code. - Separate wake enable interrupt handling from slave status handling logic. - Use acp_mmio to program ACP common registers. - Use dai_runtime_array implementation in dai_ops. - Refactor port_ops callbacks. - Add comments in port_ops callbacks. - Add retry count logic in irq thread to address faulty case. - Add helper function to interpret command response. - Add generic bandwidth allocation dependency in Kconfig options. - Add comments for AMD SoundWire power modes. - Add missing timeout check in amd_init_sdw_manager callback. - Declare frameshape parameters in probe call. - Handle error case in clock stop sequence. - Add comments in pm_prepare and pm_ops callbacks. Vijendar Mukunda (8): soundwire: export sdw_compute_slave_ports() function soundwire: amd: Add support for AMD Manager driver soundwire: amd: register SoundWire manager dai ops soundwire: amd: enable build for AMD SoundWire manager driver soundwire: amd: add SoundWire manager interrupt handling soundwire: amd: add runtime pm ops for AMD SoundWire manager driver soundwire: amd: handle SoundWire wake enable interrupt soundwire: amd: add pm_prepare callback and pm ops support drivers/soundwire/Kconfig | 10 + drivers/soundwire/Makefile | 4 + drivers/soundwire/amd_manager.c | 1321 +++++++++++++++++ drivers/soundwire/amd_manager.h | 275 ++++ drivers/soundwire/bus.h | 9 + .../soundwire/generic_bandwidth_allocation.c | 12 +- include/linux/soundwire/sdw_amd.h | 109 ++ 7 files changed, 1731 insertions(+), 9 deletions(-) create mode 100644 drivers/soundwire/amd_manager.c create mode 100644 drivers/soundwire/amd_manager.h create mode 100644 include/linux/soundwire/sdw_amd.h -- 2.34.1

5 20

[PATCH v2 0/3] Fixes and Improvements for CS35L41 HDA
by Stefan Binding 28 Feb '23

28 Feb '23

Fixes issue in calibration, where return codes were misinterpreted. Enable High Pass filter to reduce pops and clicks. Add improvement to ensure firmware and tuning files are always loaded together. This ensure the firmware is alsways running with valid coefficients. Stefan Binding (2): ALSA: hda: cs35l41: Ensure firmware/tuning pairs are always loaded ALSA: hda: cs35l41: Enable Amp High Pass Filter Vitaly Rodionov (1): ALSA: hda: cs35l41: Correct error condition handling sound/pci/hda/cs35l41_hda.c | 109 ++++++++++++++++----------------- sound/pci/hda/hda_cs_dsp_ctl.c | 4 +- 2 files changed, 56 insertions(+), 57 deletions(-) -- 2.34.1

3 6

[PATCH] soundwire: bus: Fix unbalanced pm_runtime_put() causing usage count underflow
by Richard Fitzgerald 27 Feb '23

27 Feb '23

This reverts commit 443a98e649b4 ("soundwire: bus: use pm_runtime_resume_and_get()") Change calls to pm_runtime_resume_and_get() back to pm_runtime_get_sync(). This fixes a usage count underrun caused by doing a pm_runtime_put() even though pm_runtime_resume_and_get() returned an error. The three affected functions ignore -EACCES error from trying to get pm_runtime, and carry on, including a put at the end of the function. But pm_runtime_resume_and_get() does not increment the usage count if it returns an error. So in the -EACCES case you must not call pm_runtime_put(). The documentation for pm_runtime_get_sync() says: "Consider using pm_runtime_resume_and_get() ... as this is likely to result in cleaner code." In this case I don't think it results in cleaner code because the pm_runtime_put() at the end of the function would have to be conditional on the return value from pm_runtime_resume_and_get() at the top of the function. pm_runtime_get_sync() doesn't have this problem because it always increments the count, so always needs a put. The code can just flow through and do the pm_runtime_put() unconditionally. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/soundwire/bus.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 76515c33e639..4fd221d0cc81 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -571,9 +571,11 @@ int sdw_nread(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) { int ret; - ret = pm_runtime_resume_and_get(&slave->dev); - if (ret < 0 && ret != -EACCES) + ret = pm_runtime_get_sync(&slave->dev); + if (ret < 0 && ret != -EACCES) { + pm_runtime_put_noidle(&slave->dev); return ret; + } ret = sdw_nread_no_pm(slave, addr, count, val); @@ -595,9 +597,11 @@ int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) { int ret; - ret = pm_runtime_resume_and_get(&slave->dev); - if (ret < 0 && ret != -EACCES) + ret = pm_runtime_get_sync(&slave->dev); + if (ret < 0 && ret != -EACCES) { + pm_runtime_put_noidle(&slave->dev); return ret; + } ret = sdw_nwrite_no_pm(slave, addr, count, val); @@ -1565,9 +1569,10 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) sdw_modify_slave_status(slave, SDW_SLAVE_ALERT); - ret = pm_runtime_resume_and_get(&slave->dev); + ret = pm_runtime_get_sync(&slave->dev); if (ret < 0 && ret != -EACCES) { dev_err(&slave->dev, "Failed to resume device: %d\n", ret); + pm_runtime_put_noidle(&slave->dev); return ret; } -- 2.30.2

2 1

[PATCH] Fix snprintf format warnings during 'alsa' kselftest compilation
by Ivan Orlov 27 Feb '23

27 Feb '23

Fix 'alsa' kselftest compilation warnings by making snprintf format correspond the actual parameters types. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- tools/testing/selftests/alsa/pcm-test.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/tools/testing/selftests/alsa/pcm-test.c b/tools/testing/selftests/alsa/pcm-test.c index f293c7d81009..52b109a162c6 100644 --- a/tools/testing/selftests/alsa/pcm-test.c +++ b/tools/testing/selftests/alsa/pcm-test.c @@ -318,7 +318,7 @@ static void test_pcm_time1(struct pcm_data *data, goto __close; } if (rchannels != channels) { - snprintf(msg, sizeof(msg), "channels unsupported %ld != %ld", channels, rchannels); + snprintf(msg, sizeof(msg), "channels unsupported %ld != %u", channels, rchannels); skip = true; goto __close; } @@ -329,7 +329,7 @@ static void test_pcm_time1(struct pcm_data *data, goto __close; } if (rrate != rate) { - snprintf(msg, sizeof(msg), "rate unsupported %ld != %ld", rate, rrate); + snprintf(msg, sizeof(msg), "rate unsupported %ld != %u", rate, rrate); skip = true; goto __close; } @@ -393,24 +393,24 @@ static void test_pcm_time1(struct pcm_data *data, frames = snd_pcm_writei(handle, samples, rate); if (frames < 0) { snprintf(msg, sizeof(msg), - "Write failed: expected %d, wrote %li", rate, frames); + "Write failed: expected %ld, wrote %li", rate, frames); goto __close; } if (frames < rate) { snprintf(msg, sizeof(msg), - "expected %d, wrote %li", rate, frames); + "expected %ld, wrote %li", rate, frames); goto __close; } } else { frames = snd_pcm_readi(handle, samples, rate); if (frames < 0) { snprintf(msg, sizeof(msg), - "expected %d, wrote %li", rate, frames); + "expected %ld, wrote %li", rate, frames); goto __close; } if (frames < rate) { snprintf(msg, sizeof(msg), - "expected %d, wrote %li", rate, frames); + "expected %ld, wrote %li", rate, frames); goto __close; } } -- 2.34.1

2 3

Re: [bug report] firewire: cdev: obsolete NULL check to detect IEC 61883-1 FCP region
by Takashi Sakamoto 27 Feb '23

27 Feb '23

Hi, (C.C.ed to LKML and alsa-devel) On Mon, Feb 27, 2023 at 02:06:51PM +0300, Dan Carpenter wrote: > Hello Takashi Sakamoto, > > The patch e699600232e0: "firewire: cdev: obsolete NULL check to > detect IEC 61883-1 FCP region" from Jan 20, 2023, leads to the > following Smatch static checker warning: > > drivers/firewire/core-transaction.c:947 handle_fcp_region_request() > warn: passing freed memory 'request' > > drivers/firewire/core-transaction.c > 930 fw_send_response(card, request, RCODE_TYPE_ERROR); > 931 > 932 return; > 933 } > 934 > 935 rcu_read_lock(); > 936 list_for_each_entry_rcu(handler, &address_handler_list, link) { > 937 if (is_enclosing_handler(handler, offset, request->length)) > 938 handler->address_callback(card, request, tcode, > ^^^^^^^ > This warning is because fwnet_receive_packet() has a kfree(r) on the > first return path. > > 939 destination, source, > 940 p->generation, offset, > 941 request->data, > 942 request->length, > 943 handler->callback_data); > 944 } > 945 rcu_read_unlock(); > 946 > --> 947 fw_send_response(card, request, RCODE_COMPLETE); > 948 } Thanks for your report. Fortunately, We can not see the access to the released memory since the fwnet's address handler is registered to high memory region (0x'0001'0000'0000 to 0x'ffff'e000'0000). The region does not overlap IEC 61883-1 FCP region (0x'ffff'f000'0b00 to 0x'ffff'f000'0f00). The handler is called from handle_exclusive_region_request() instead of handle_fcp_region_request(). However, the code in fwnet is against the design of address handler apparently. The callee never release the memory for the request structure directly. It should be done by the call of fw_send_response(). I'll correct it for next merge window; i.e. for v6.4. Thanks Takashi Sakamoto

1 0

[PATCH 1/4] ASoC: dt-bindings: wlf,wm8524: Convert to json-schema
by Chancel Liu 27 Feb '23

27 Feb '23

Convert the Wolfson WM8524 24-bit 192KHz Stereo DAC device tree binding documentation to json-schema. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- .../devicetree/bindings/sound/wlf,wm8524.yaml | 37 +++++++++++++++++++ .../devicetree/bindings/sound/wm8524.txt | 16 -------- 2 files changed, 37 insertions(+), 16 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8524.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8524.txt diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8524.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8524.yaml new file mode 100644 index 000000000000..09c54cc7de95 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8524.yaml @@ -0,0 +1,37 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8524.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8524 24-bit 192KHz Stereo DAC + +maintainers: + - patches(a)opensource.cirrus.com + +properties: + compatible: + const: wlf,wm8524 + + "#sound-dai-cells": + const: 0 + + wlf,mute-gpios: + maxItems: 1 + description: + a GPIO spec for the MUTE pin. + +required: + - compatible + - wlf,mute-gpios + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + wm8524: codec { + compatible = "wlf,wm8524"; + wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8524.txt b/Documentation/devicetree/bindings/sound/wm8524.txt deleted file mode 100644 index f6c0c263b135..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8524.txt +++ /dev/null @@ -1,16 +0,0 @@ -WM8524 audio CODEC - -This device does not use I2C or SPI but a simple Hardware Control Interface. - -Required properties: - - - compatible : "wlf,wm8524" - - - wlf,mute-gpios: a GPIO spec for the MUTE pin. - -Example: - -wm8524: codec { - compatible = "wlf,wm8524"; - wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; -}; -- 2.25.1

3 15

[PATCH AUTOSEL 4.14 4/9] ASoC: kirkwood: Iterate over array indexes instead of using pointer math
by Sasha Levin 27 Feb '23

27 Feb '23

From: Kees Cook <keescook(a)chromium.org> [ Upstream commit b3bcedc0402fcdc5c8624c433562d9d1882749d8 ] Walking the dram->cs array was seen as accesses beyond the first array item by the compiler. Instead, use the array index directly. This allows for run-time bounds checking under CONFIG_UBSAN_BOUNDS as well. Seen with GCC 13 with -fstrict-flex-arrays: ../sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_conf_mbus_windows.constprop': ../sound/soc/kirkwood/kirkwood-dma.c:90:24: warning: array subscript 0 is outside array bounds of 'const struct mbus_dram_window[0]' [-Warray-bounds=] 90 | if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { | ~~^~~~~~ 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 Signed-off-by: Kees Cook <keescook(a)chromium.org> Link: https://lore.kernel.org/r/20230127224128.never.410-kees@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 35ca8e8bb5e52..9736fb36082fb 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -90,7 +90,7 @@ kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, /* try to find matching cs for current dma address */ for (i = 0; i < dram->num_cs; i++) { - const struct mbus_dram_window *cs = dram->cs + i; + const struct mbus_dram_window *cs = &dram->cs[i]; if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { writel(cs->base & 0xffff0000, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); -- 2.39.0

1 0