August 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: SOF: intel: hda: Clean up link DMA for IPC3 during stop
by Peter Ujfalusi 18 Aug '23

18 Aug '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> With IPC3, we reset hw_params during the stop trigger, so we should also clean up the link DMA during the stop trigger. Fixes: 1bf83fa6654c ("ASoC: SOF: Intel: hda-dai: Do not perform DMA cleanup during stop") Closes: https://github.com/thesofproject/linux/issues/4455 Closes: https://github.com/thesofproject/linux/issues/4482 Closes: https://bugzilla.kernel.org/show_bug.cgi?id=217673 Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/hda-dai-ops.c | 11 ++++++++++- sound/soc/sof/intel/hda-dai.c | 5 ++--- sound/soc/sof/intel/hda.h | 2 ++ 3 files changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index f3513796c189..f33051eac1c0 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -372,6 +372,7 @@ static const struct hda_dai_widget_dma_ops hda_ipc4_chain_dma_ops = { static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, struct snd_pcm_substream *substream, int cmd) { + struct hdac_ext_stream *hext_stream = hda_get_hext_stream(sdev, cpu_dai, substream); struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream); switch (cmd) { @@ -379,9 +380,17 @@ static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *c case SNDRV_PCM_TRIGGER_STOP: { struct snd_sof_dai_config_data data = { 0 }; + int ret; data.dai_data = DMA_CHAN_INVALID; - return hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_HW_FREE, &data); + ret = hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_HW_FREE, &data); + if (ret < 0) + return ret; + + if (cmd == SNDRV_PCM_TRIGGER_STOP) + return hda_link_dma_cleanup(substream, hext_stream, cpu_dai); + + break; } case SNDRV_PCM_TRIGGER_PAUSE_PUSH: return hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_PAUSE, NULL); diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 3297dea493aa..863865f3d77e 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -107,9 +107,8 @@ hda_dai_get_ops(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai return sdai->platform_private; } -static int hda_link_dma_cleanup(struct snd_pcm_substream *substream, - struct hdac_ext_stream *hext_stream, - struct snd_soc_dai *cpu_dai) +int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream, + struct snd_soc_dai *cpu_dai) { const struct hda_dai_widget_dma_ops *ops = hda_dai_get_ops(substream, cpu_dai); struct sof_intel_hda_stream *hda_stream; diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 3f7c6fb05e5d..5b9e4ebcc18b 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -963,5 +963,7 @@ const struct hda_dai_widget_dma_ops * hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget); int hda_dai_config(struct snd_soc_dapm_widget *w, unsigned int flags, struct snd_sof_dai_config_data *data); +int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream, + struct snd_soc_dai *cpu_dai); #endif -- 2.41.0

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[PATCH -next] ALSA: asihpi: Remove unused declarations
by Yue Haibing 18 Aug '23

18 Aug '23

These are not implemented, so can remove them. Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- sound/pci/asihpi/hpi.h | 16 ---------------- sound/pci/asihpi/hpi_internal.h | 5 ----- 2 files changed, 21 deletions(-) diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index 3aebec763fb8..04a5cf6572cd 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -1191,19 +1191,6 @@ u16 hpi_adapter_set_mode_ex(u16 adapter_index, u32 adapter_mode, u16 hpi_adapter_get_mode(u16 adapter_index, u32 *padapter_mode); -u16 hpi_adapter_get_assert2(u16 adapter_index, u16 *p_assert_count, - char *psz_assert, u32 *p_param1, u32 *p_param2, - u32 *p_dsp_string_addr, u16 *p_processor_id); - -u16 hpi_adapter_test_assert(u16 adapter_index, u16 assert_id); - -u16 hpi_adapter_enable_capability(u16 adapter_index, u16 capability, u32 key); - -u16 hpi_adapter_self_test(u16 adapter_index); - -u16 hpi_adapter_debug_read(u16 adapter_index, u32 dsp_address, char *p_bytes, - int *count_bytes); - u16 hpi_adapter_set_property(u16 adapter_index, u16 property, u16 paramter1, u16 paramter2); @@ -1488,9 +1475,6 @@ u16 hpi_pad_get_program_type(u32 h_control, u32 *ppTY); u16 hpi_pad_get_rdsPI(u32 h_control, u32 *ppI); -u16 hpi_pad_get_program_type_string(u32 h_control, const u32 data_type, - const u32 pTY, char *psz_string, const u32 string_length); - /****************************/ /* AES/EBU Receiver control */ /****************************/ diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index 6859d51389f5..e569e3b33b8e 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -1394,17 +1394,12 @@ u32 hpi_indexes_to_handle(const char c_object, const u16 adapter_index, void hpi_send_recv(struct hpi_message *phm, struct hpi_response *phr); /* used in PnP OS/driver */ -u16 hpi_subsys_create_adapter(const struct hpi_resource *p_resource, - u16 *pw_adapter_index); - u16 hpi_outstream_host_buffer_get_info(u32 h_outstream, u8 **pp_buffer, struct hpi_hostbuffer_status **pp_status); u16 hpi_instream_host_buffer_get_info(u32 h_instream, u8 **pp_buffer, struct hpi_hostbuffer_status **pp_status); -u16 hpi_adapter_restart(u16 adapter_index); - /* The following 3 functions were last declared in header files for driver 3.10. HPI_ControlQuery() used to be the recommended way -- 2.34.1

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[PATCH v3 0/9] sound: Use -EPROBE_DEFER instead of i915 module loading.
by Maarten Lankhorst 18 Aug '23

18 Aug '23

From: Maarten Lankhorst <dev(a)lankhorst.se> Explicitly loading i915 becomes a problem when upstreaming the new intel driver for Tiger Lake and higher graphics (xe). By loading i915, it doesn't wait for driver load of xe, and will fail completely before it loads. -EPROBE_DEFER has to be returned before any device is created in probe(), otherwise the removal of the device will cause EPROBE_DEFER to try again in an infinite loop. The conversion is done in gradual steps. First I add an argument to snd_hdac_i915_init to allow for -EPROBE_DEFER so I can convert each driver separately. Then I convert each driver to move snd_hdac_i915_init out of the workqueue. Finally I drop the ability to choose modprobe behavior after the last user is converted. I suspect the avs and skylake drivers used snd_hdac_i915_init purely for the modprobe, but I don't have the hardware to test if it can be safely removed. It can still be done easily in a followup patch to simplify probing. --- Changes since previous version: - Keep the workqueue for soc/sof/intel, instead only move out snd_hdac_i915_init. (Patch 8/9) Rest of patches unchanged. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Daniel Baluta <daniel.baluta(a)nxp.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Cc: sound-open-firmware(a)alsa-project.org Maarten Lankhorst (9): ALSA: hda/intel: Fix error handling in azx_probe() ALSA: hda/i915: Allow override of gpu binding. ALSA: hda/i915: Add an allow_modprobe argument to snd_hdac_i915_init ALSA: hda/i915: Allow xe as match for i915_component_master_match ASoC: Intel: avs: Move snd_hdac_i915_init to before probe_work. ASoC: Intel: Skylake: Move snd_hdac_i915_init to before probe_work. ALSA: hda/intel: Move snd_hdac_i915_init to before probe_work. ASoC: SOF: Intel: Move binding to display driver outside of deferred probe ALSA: hda/i915: Remove extra argument from snd_hdac_i915_init sound/hda/hdac_i915.c | 25 ++++++++------- sound/pci/hda/hda_intel.c | 60 ++++++++++++++++++----------------- sound/soc/intel/avs/core.c | 13 +++++--- sound/soc/intel/skylake/skl.c | 31 ++++++------------ sound/soc/sof/core.c | 19 ++++------- 5 files changed, 70 insertions(+), 78 deletions(-) -- 2.39.2

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Re: [PATCH 3/4] ASoC: amd: acp: Add machine driver that enables sound for systems with a ES8336 codec
by Zhu Ning 18 Aug '23

18 Aug '23

> > This is saying that the machine driver should disable jack detection > > over suspend and restart it during resume. The machine driver should > > suspend before the rest of the card which should mean that the CODEC > > gets powered off then. The core can't tell if jack detection is > > supposed to work over suspend, it is a standard wake event on systems > > like phones, but it sounds like on this system the power gets removed > > from the device so that can't work. > > > > Sorry, I don't understand what you are trying to say here. My intention > is to find a way to have sound working when suspending/resuming > with jack inserted by not fudging the jack in the machine driver but > fixing it in the CODEC. > > > I don't understand why that would be expected to help? The main effect > > of keeping the bias on all the time would be to consume more power. > > I don't fully understand the whole bias thing (I did try hard to > understand it from the code), but in this specific instance it helps for > suspending the CODEC. If idle_bias_on is true then idle_bias_off will be > false, that would mean that during suspend (with jack inserted) > in snd_soc_suspend() in switch (snd_soc_dapm_get_bias_level(dapm)) > the flow goes in case SND_SOC_BIAS_STANDBY and since dapm->idle_bias_off > is false it doesn't break from the case but falls through to > SND_SOC_BIAS_OFF case which in turn calls the CODEC component's suspend > callback. > > The reason the sound stops working with jack inserted is that the CODEC > suspend callback never gets called. It only gets called when the jack > is not inserted. Hi Postevca, I just reproduced the bug on my side. You are right that when jack inserted, the codec driver will never enter SND_SOC_BIAS_OFF or suspend or resume. This makes sense because the codec needs micbias power to detect button. Problem is that ALSA or dapm thought the codec power is on during suspend, but the system power is turned off and the registers are reset. The system power is controlled by EC and neither dapm or codec driver can do anything about it. The system power is just turned off. Disabling the jack detection before suspend is ok because the system stops working during suspend anyway. Maybe a detailed comment is needed to explain why?

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[PATCH] ALSA: aoa: Fix typos in PCM fix patch
by Takashi Iwai 18 Aug '23

18 Aug '23

There was typos in the previous fix for PCM to detach the struct device that caused build errors. Corrected here. Fixes: bc41a7228ced ("ALSA: pcm: Don't embed device") Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202308181347.q3XPr3Lm-lkp@intel.com/ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/aoa/soundbus/i2sbus/pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index 3680eb6eabc9..07df5cc0f2d7 100644 --- a/sound/aoa/soundbus/i2sbus/pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -972,7 +972,7 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, goto out_put_ci_module; snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, &i2sbus_playback_ops); - dev->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK]->dev.parent = + dev->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].dev->parent = &dev->ofdev.dev; i2sdev->out.created = 1; } @@ -989,7 +989,7 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, goto out_put_ci_module; snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, &i2sbus_record_ops); - dev->pcm->streams[SNDRV_PCM_STREAM_CAPTURE]->dev.parent = + dev->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].dev->parent = &dev->ofdev.dev; i2sdev->in.created = 1; } -- 2.35.3

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[tiwai-sound:topic/dev-split-kref 3/11] sound/aoa/soundbus/i2sbus/pcm.c:975:47: error: member reference type 'struct snd_pcm_str' is not a pointer; did you mean to use '.'?
by kernel test robot 18 Aug '23

18 Aug '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/dev-split-kref head: 6e1b44385310dd465cae48213eae65d0b8095d13 commit: 7a3dce3fb37c25de3f3c347f4e69ba358cb81a3d [3/11] ALSA: pcm: Don't embed device config: powerpc-g5_defconfig (https://download.01.org/0day-ci/archive/20230818/202308181347.q3XPr3Lm-lkp@…) compiler: clang version 15.0.7 (https://github.com/llvm/llvm-project.git 8dfdcc7b7bf66834a761bd8de445840ef68e4d1a) reproduce: (https://download.01.org/0day-ci/archive/20230818/202308181347.q3XPr3Lm-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202308181347.q3XPr3Lm-lkp@intel.com/ All errors (new ones prefixed by >>): >> sound/aoa/soundbus/i2sbus/pcm.c:975:47: error: member reference type 'struct snd_pcm_str' is not a pointer; did you mean to use '.'? dev->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK]->dev.parent = ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~ . >> sound/aoa/soundbus/i2sbus/pcm.c:975:52: error: member reference type 'struct device *' is a pointer; did you mean to use '->'? dev->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK]->dev.parent = ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^ -> sound/aoa/soundbus/i2sbus/pcm.c:992:46: error: member reference type 'struct snd_pcm_str' is not a pointer; did you mean to use '.'? dev->pcm->streams[SNDRV_PCM_STREAM_CAPTURE]->dev.parent = ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~ . sound/aoa/soundbus/i2sbus/pcm.c:992:51: error: member reference type 'struct device *' is a pointer; did you mean to use '->'? dev->pcm->streams[SNDRV_PCM_STREAM_CAPTURE]->dev.parent = ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^ -> 4 errors generated. vim +975 sound/aoa/soundbus/i2sbus/pcm.c 866 867 int 868 i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, 869 struct codec_info *ci, void *data) 870 { 871 int err, in = 0, out = 0; 872 struct transfer_info *tmp; 873 struct i2sbus_dev *i2sdev = soundbus_dev_to_i2sbus_dev(dev); 874 struct codec_info_item *cii; 875 876 if (!dev->pcmname || dev->pcmid == -1) { 877 printk(KERN_ERR "i2sbus: pcm name and id must be set!\n"); 878 return -EINVAL; 879 } 880 881 list_for_each_entry(cii, &dev->codec_list, list) { 882 if (cii->codec_data == data) 883 return -EALREADY; 884 } 885 886 if (!ci->transfers || !ci->transfers->formats 887 || !ci->transfers->rates || !ci->usable) 888 return -EINVAL; 889 890 /* we currently code the i2s transfer on the clock, and support only 891 * 32 and 64 */ 892 if (ci->bus_factor != 32 && ci->bus_factor != 64) 893 return -EINVAL; 894 895 /* If you want to fix this, you need to keep track of what transport infos 896 * are to be used, which codecs they belong to, and then fix all the 897 * sysclock/busclock stuff above to depend on which is usable */ 898 list_for_each_entry(cii, &dev->codec_list, list) { 899 if (cii->codec->sysclock_factor != ci->sysclock_factor) { 900 printk(KERN_DEBUG 901 "cannot yet handle multiple different sysclocks!\n"); 902 return -EINVAL; 903 } 904 if (cii->codec->bus_factor != ci->bus_factor) { 905 printk(KERN_DEBUG 906 "cannot yet handle multiple different bus clocks!\n"); 907 return -EINVAL; 908 } 909 } 910 911 tmp = ci->transfers; 912 while (tmp->formats && tmp->rates) { 913 if (tmp->transfer_in) 914 in = 1; 915 else 916 out = 1; 917 tmp++; 918 } 919 920 cii = kzalloc(sizeof(struct codec_info_item), GFP_KERNEL); 921 if (!cii) 922 return -ENOMEM; 923 924 /* use the private data to point to the codec info */ 925 cii->sdev = soundbus_dev_get(dev); 926 cii->codec = ci; 927 cii->codec_data = data; 928 929 if (!cii->sdev) { 930 printk(KERN_DEBUG 931 "i2sbus: failed to get soundbus dev reference\n"); 932 err = -ENODEV; 933 goto out_free_cii; 934 } 935 936 if (!try_module_get(THIS_MODULE)) { 937 printk(KERN_DEBUG "i2sbus: failed to get module reference!\n"); 938 err = -EBUSY; 939 goto out_put_sdev; 940 } 941 942 if (!try_module_get(ci->owner)) { 943 printk(KERN_DEBUG 944 "i2sbus: failed to get module reference to codec owner!\n"); 945 err = -EBUSY; 946 goto out_put_this_module; 947 } 948 949 if (!dev->pcm) { 950 err = snd_pcm_new(card, dev->pcmname, dev->pcmid, 0, 0, 951 &dev->pcm); 952 if (err) { 953 printk(KERN_DEBUG "i2sbus: failed to create pcm\n"); 954 goto out_put_ci_module; 955 } 956 } 957 958 /* ALSA yet again sucks. 959 * If it is ever fixed, remove this line. See below. */ 960 out = in = 1; 961 962 if (!i2sdev->out.created && out) { 963 if (dev->pcm->card != card) { 964 /* eh? */ 965 printk(KERN_ERR 966 "Can't attach same bus to different cards!\n"); 967 err = -EINVAL; 968 goto out_put_ci_module; 969 } 970 err = snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, 1); 971 if (err) 972 goto out_put_ci_module; 973 snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, 974 &i2sbus_playback_ops); > 975 dev->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK]->dev.parent = 976 &dev->ofdev.dev; 977 i2sdev->out.created = 1; 978 } 979 980 if (!i2sdev->in.created && in) { 981 if (dev->pcm->card != card) { 982 printk(KERN_ERR 983 "Can't attach same bus to different cards!\n"); 984 err = -EINVAL; 985 goto out_put_ci_module; 986 } 987 err = snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, 1); 988 if (err) 989 goto out_put_ci_module; 990 snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, 991 &i2sbus_record_ops); 992 dev->pcm->streams[SNDRV_PCM_STREAM_CAPTURE]->dev.parent = 993 &dev->ofdev.dev; 994 i2sdev->in.created = 1; 995 } 996 997 /* so we have to register the pcm after adding any substream 998 * to it because alsa doesn't create the devices for the 999 * substreams when we add them later. 1000 * Therefore, force in and out on both busses (above) and 1001 * register the pcm now instead of just after creating it. 1002 */ 1003 err = snd_device_register(card, dev->pcm); 1004 if (err) { 1005 printk(KERN_ERR "i2sbus: error registering new pcm\n"); 1006 goto out_put_ci_module; 1007 } 1008 /* no errors any more, so let's add this to our list */ 1009 list_add(&cii->list, &dev->codec_list); 1010 1011 dev->pcm->private_data = i2sdev; 1012 dev->pcm->private_free = i2sbus_private_free; 1013 1014 /* well, we really should support scatter/gather DMA */ 1015 snd_pcm_set_managed_buffer_all( 1016 dev->pcm, SNDRV_DMA_TYPE_DEV, 1017 &macio_get_pci_dev(i2sdev->macio)->dev, 1018 64 * 1024, 64 * 1024); 1019 1020 return 0; 1021 out_put_ci_module: 1022 module_put(ci->owner); 1023 out_put_this_module: 1024 module_put(THIS_MODULE); 1025 out_put_sdev: 1026 soundbus_dev_put(dev); 1027 out_free_cii: 1028 kfree(cii); 1029 return err; 1030 } 1031 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[PATCH RFC 0/6] ALSA: Fix UAF with delayed kobj release
by Takashi Iwai 18 Aug '23

18 Aug '23

Hi, this is a test patch set for addressing the UAF problems with delayed kobj releases reported by Curtis: https://lore.kernel.org/r/20230801171928.1460120-1-cujomalainey@chromium.org The patch introduced a simple helper for allocating memory with a refcount, and converts the card object, control, PCM and compress objects with the new type. With the refcount, the actual memory release is delayed until all referrer are gone. It's just a RFC and only lightly tested. I myself am not sure whether this is the best way to go. It might be better to take Curtis' approach, just converting the device to its own allocation, too. (But I don't know whether Curtis' patch set covers all cases -- can still be a UAF of card_dev due to devres vs kobj release?) thanks, Takashi === Takashi Iwai (6): ALSA: core: Introduced referenced memory allocator ALSA: core: Fix potential UAF by delayed kobject release of card_dev ALSA: core: Associate memory reference with device initialization ALSA: pcm: Release memory with reference ALSA: control: Reference card by ctl_dev ALSA: compress: Reference card by the device include/sound/core.h | 7 ++- sound/core/compress_offload.c | 2 +- sound/core/control.c | 2 +- sound/core/hwdep.c | 2 +- sound/core/init.c | 105 +++++++++++++++++++++++++++------ sound/core/pcm.c | 6 +- sound/core/rawmidi.c | 2 +- sound/core/seq/seq_clientmgr.c | 2 +- sound/core/timer.c | 2 +- 9 files changed, 101 insertions(+), 29 deletions(-) -- 2.35.3

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[PATCH v2 0/5] ASoC: Convert remaining Realtek codecs to GPIO descriptors
by Linus Walleij 18 Aug '23

18 Aug '23

After dropping unused headers a few Realtek devices actually using the GPIO descriptors remain. Converting them to use optional GPIO descriptors is pretty straight-forward. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Changes in v2: - Rebased on asoc-6.6 - Link to v1: https://lore.kernel.org/r/20230817-descriptors-asoc-rt-v1-0-434f5f177cee@li… --- Linus Walleij (5): ASoC: rt5640: Convert to just use GPIO descriptors ASoC: rt5665: Convert to use GPIO descriptors ASoC: rt5668: Convert to use GPIO descriptors ASoC: rt5682: Convert to use GPIO descriptors ASoC: rt5682s: Convert to use GPIO descriptors include/sound/rt5665.h | 2 -- include/sound/rt5668.h | 3 --- include/sound/rt5682.h | 3 --- include/sound/rt5682s.h | 3 --- sound/soc/codecs/rt5640.c | 55 ++++++++++++------------------------------- sound/soc/codecs/rt5640.h | 2 +- sound/soc/codecs/rt5665.c | 17 +++++++------ sound/soc/codecs/rt5668.c | 17 +++++++------ sound/soc/codecs/rt5682-i2c.c | 11 ++++----- sound/soc/codecs/rt5682-sdw.c | 5 ++++ sound/soc/codecs/rt5682.c | 20 ++++++++++++---- sound/soc/codecs/rt5682.h | 3 +++ sound/soc/codecs/rt5682s.c | 16 ++++++------- sound/soc/codecs/rt5682s.h | 2 ++ 14 files changed, 68 insertions(+), 91 deletions(-) --- base-commit: 17b9f4387ebabb19b871bbe2d06562e48e4e7130 change-id: 20230817-descriptors-asoc-rt-9c072934d376 Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

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[PATCH 0/5] ASoC: Convert remaining Realtek codecs to GPIO descriptors
by Linus Walleij 18 Aug '23

18 Aug '23

After dropping unused headers a few Realtek devices actually using the GPIO descriptors remain. Converting them to use optional GPIO descriptors is pretty straight-forward. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Linus Walleij (5): ASoC: rt5640: Convert to just use GPIO descriptors ASoC: rt5665: Convert to use GPIO descriptors ASoC: rt5668: Convert to use GPIO descriptors ASoC: rt5682: Convert to use GPIO descriptors ASoC: rt5682s: Convert to use GPIO descriptors include/sound/rt5665.h | 2 -- include/sound/rt5668.h | 3 --- include/sound/rt5682.h | 3 --- include/sound/rt5682s.h | 3 --- sound/soc/codecs/rt5640.c | 55 ++++++++++++------------------------------- sound/soc/codecs/rt5640.h | 2 +- sound/soc/codecs/rt5665.c | 17 +++++++------ sound/soc/codecs/rt5668.c | 17 +++++++------ sound/soc/codecs/rt5682-i2c.c | 11 ++++----- sound/soc/codecs/rt5682-sdw.c | 4 ++++ sound/soc/codecs/rt5682.c | 20 ++++++++++++---- sound/soc/codecs/rt5682.h | 3 +++ sound/soc/codecs/rt5682s.c | 16 ++++++------- sound/soc/codecs/rt5682s.h | 2 ++ 14 files changed, 67 insertions(+), 91 deletions(-) --- base-commit: c5b06b35a73b80f1af931c2371de084fa77b5c23 change-id: 20230817-descriptors-asoc-rt-9c072934d376 Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

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[PATCH v1] ASoC: tas2781: fixed register access error when switching to other chips
by Shenghao Ding 18 Aug '23

18 Aug '23

fixed register access error when switching to other tas2781 -- refresh the page inside regmap on the switched tas2781 Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Changes in v1: - fixed register access error when switching to other tas2781 --- sound/soc/codecs/tas2781-comlib.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index a88c6c28a394..ffb26e4a7e2f 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -57,16 +57,17 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, if (client->addr != tasdev->dev_addr) { client->addr = tasdev->dev_addr; - if (tasdev->cur_book == book) { - ret = regmap_write(map, - TASDEVICE_PAGE_SELECT, 0); - if (ret < 0) { - dev_err(tas_priv->dev, "%s, E=%d\n", - __func__, ret); - goto out; - } + /* All tas2781s share the same regmap, clear the page + * inside regmap once switching to another tas2781. + * Register 0 at any pages and any books inside tas2781 + * is the same one for page-switching. + */ + ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", + __func__, ret); + goto out; } - goto out; } if (tasdev->cur_book != book) { -- 2.34.1

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