Merge tag 'sound-5.0-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai...

Pull sound fixes from Takashi Iwai:
 "It's a bit of surprising that we've got more changes than hoped at
  this late stage, but they all don't look too scary but small fixes.

  One change in ALSA core side is again the PCM regression fix that was
  partially addressed for OSS, but now the all relevant change is
  reverted instead. Also, a few ASoC core fixes for UAF and OOB are
  included, while the rest are usual random device-specific fixes"

* tag 'sound-5.0-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: pcm: Revert capture stream behavior change in blocking mode
  ALSA: usb-audio: Fix implicit fb endpoint setup by quirk
  ALSA: hda - Add quirk for HP EliteBook 840 G5
  ASoC: samsung: Prevent clk_get_rate() calls in atomic context
  ASoC: rsnd: ssiu: correct shift bit for ssiu9
  ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count check
  ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables
  ASoC: topology: fix oops/use-after-free case with dai driver
  ASoC: rsnd: fixup MIX kctrl registration
  ASoC: core: Allow soc_find_component lookups to match parent of_node
  ASoC: rt5682: Correct the setting while select ASRC clk for AD/DA filter
  ASoC: MAINTAINERS: fsl: Change Fabio's email address
  ASoC: hdmi-codec: fix oops on re-probe

diff --git a/MAINTAINERS b/MAINTAINERS
index 9919840d54cde80896e5f6f18bc03a0b9b8f8ffc..41ce5f4ad838784909b02f5d377d2987d8ad0e9e 100644 (file)
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -6151,7 +6151,7 @@ FREESCALE SOC SOUND DRIVERS
 M:     Timur Tabi <timur@kernel.org>
 M:     Nicolin Chen <nicoleotsuka@gmail.com>
 M:     Xiubo Li <Xiubo.Lee@gmail.com>
-R:     Fabio Estevam <fabio.estevam@nxp.com>
+R:     Fabio Estevam <festevam@gmail.com>
 L:     alsa-devel@alsa-project.org (moderated for non-subscribers)
 L:     linuxppc-dev@lists.ozlabs.org
 S:     Maintained
@@ -10898,7 +10898,7 @@ F:      include/linux/nvmem-consumer.h
 F:     include/linux/nvmem-provider.h
 
 NXP SGTL5000 DRIVER
-M:     Fabio Estevam <fabio.estevam@nxp.com>
+M:     Fabio Estevam <festevam@gmail.com>
 L:     alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:     Maintained
 F:     Documentation/devicetree/bindings/sound/sgtl5000.txt

diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 6c99fa8ac5fa183c1b7241366e9e75d7d413b7b9..6c0b30391ba99ced198ee7f7c943083899c4b93d 100644 (file)
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -2112,13 +2112,6 @@ int pcm_lib_apply_appl_ptr(struct snd_pcm_substream *substream,
        return 0;
 }
 
-/* allow waiting for a capture stream that hasn't been started */
-#if IS_ENABLED(CONFIG_SND_PCM_OSS)
-#define wait_capture_start(substream)  ((substream)->oss.oss)
-#else
-#define wait_capture_start(substream)  false
-#endif
-
 /* the common loop for read/write data */
 snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream,
                                     void *data, bool interleaved,
@@ -2184,16 +2177,11 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream,
                snd_pcm_update_hw_ptr(substream);
 
        if (!is_playback &&
-           runtime->status->state == SNDRV_PCM_STATE_PREPARED) {
-               if (size >= runtime->start_threshold) {
-                       err = snd_pcm_start(substream);
-                       if (err < 0)
-                               goto _end_unlock;
-               } else if (!wait_capture_start(substream)) {
-                       /* nothing to do */
-                       err = 0;
+           runtime->status->state == SNDRV_PCM_STATE_PREPARED &&
+           size >= runtime->start_threshold) {
+               err = snd_pcm_start(substream);
+               if (err < 0)
                        goto _end_unlock;
-               }
        }
 
        avail = snd_pcm_avail(substream);

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 152f54137082321f66dea8bb0563a39f299e61d6..a4ee7656d9ee904e6f6de6a048cfbbe3a52ce027 100644 (file)
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -924,6 +924,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
        SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK),
        SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK),
        SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK),
+       SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK),
        SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK),
        SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK),
        SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE),

diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index d00734d31e0429bf3df8c12b0bcc118ccefb3ec2..e5b6769b9797724ceef38f3b8132200876f227e8 100644 (file)
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -795,6 +795,8 @@ static int hdmi_codec_probe(struct platform_device *pdev)
        if (hcd->spdif)
                hcp->daidrv[i] = hdmi_spdif_dai;
 
+       dev_set_drvdata(dev, hcp);
+
        ret = devm_snd_soc_register_component(dev, &hdmi_driver, hcp->daidrv,
                                     dai_count);
        if (ret) {
@@ -802,8 +804,6 @@ static int hdmi_codec_probe(struct platform_device *pdev)
                        __func__, ret);
                return ret;
        }
-
-       dev_set_drvdata(dev, hcp);
        return 0;
 }
 

diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index 89c43b26c3790814645172e8165e1ac971972319..a9b91bcfcc0967a3d8db46d7e96be88806a93f0e 100644 (file)
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -1778,7 +1778,9 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = {
        {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc},
        {"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc},
        {"ADC STO1 ASRC", NULL, "AD ASRC"},
+       {"ADC STO1 ASRC", NULL, "DA ASRC"},
        {"ADC STO1 ASRC", NULL, "CLKDET"},
+       {"DAC STO1 ASRC", NULL, "AD ASRC"},
        {"DAC STO1 ASRC", NULL, "DA ASRC"},
        {"DAC STO1 ASRC", NULL, "CLKDET"},
 

diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index d6c62aa130414369f729bbc5333c82052bdcc9d2..ce00fe2f6aae32d2eb777c2ac7533f3ecdd47a0a 100644 (file)
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -700,6 +700,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream,
 {
        struct i2s_dai *i2s = to_info(dai);
        u32 mod, mask = 0, val = 0;
+       struct clk *rclksrc;
        unsigned long flags;
 
        WARN_ON(!pm_runtime_active(dai->dev));
@@ -782,6 +783,10 @@ static int i2s_hw_params(struct snd_pcm_substream *substream,
 
        i2s->frmclk = params_rate(params);
 
+       rclksrc = i2s->clk_table[CLK_I2S_RCLK_SRC];
+       if (rclksrc && !IS_ERR(rclksrc))
+               i2s->rclk_srcrate = clk_get_rate(rclksrc);
+
        return 0;
 }
 
@@ -886,11 +891,6 @@ static int config_setup(struct i2s_dai *i2s)
                return 0;
 
        if (!(i2s->quirks & QUIRK_NO_MUXPSR)) {
-               struct clk *rclksrc = i2s->clk_table[CLK_I2S_RCLK_SRC];
-
-               if (rclksrc && !IS_ERR(rclksrc))
-                       i2s->rclk_srcrate = clk_get_rate(rclksrc);
-
                psr = i2s->rclk_srcrate / i2s->frmclk / rfs;
                writel(((psr - 1) << 8) | PSR_PSREN, i2s->addr + I2SPSR);
                dev_dbg(&i2s->pdev->dev,

diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 59e250cc2e9df285d778d7939a4812a3c90219df..e819e965e1dbe9254137dfea8c2fd0350542f510 100644 (file)
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1526,14 +1526,14 @@ int rsnd_kctrl_new(struct rsnd_mod *mod,
        int ret;
 
        /*
-        * 1) Avoid duplicate register (ex. MIXer case)
-        * 2) re-register if card was rebinded
+        * 1) Avoid duplicate register for DVC with MIX case
+        * 2) Allow duplicate register for MIX
+        * 3) re-register if card was rebinded
         */
        list_for_each_entry(kctrl, &card->controls, list) {
                struct rsnd_kctrl_cfg *c = kctrl->private_data;
 
-               if (strcmp(kctrl->id.name, name) == 0 &&
-                   c->mod == mod)
+               if (c == cfg)
                        return 0;
        }
 

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 45ef295743ecdc2f6042e36b2c2f31edd619a3f9..f5afab631abbc15516fa50427c908a9617e8d82f 100644 (file)
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -286,7 +286,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
        if (rsnd_ssi_is_multi_slave(mod, io))
                return 0;
 
-       if (ssi->usrcnt > 1) {
+       if (ssi->usrcnt > 0) {
                if (ssi->rate != rate) {
                        dev_err(dev, "SSI parent/child should use same rate\n");
                        return -EINVAL;

diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c
index c5934adcfd0155e5a410ecc1d5ea171268f16e8d..c74991dd18abbb459f125529e831e2bbf2d9139c 100644 (file)
--- a/sound/soc/sh/rcar/ssiu.c
+++ b/sound/soc/sh/rcar/ssiu.c
@@ -79,7 +79,7 @@ static int rsnd_ssiu_init(struct rsnd_mod *mod,
                break;
        case 9:
                for (i = 0; i < 4; i++)
-                       rsnd_mod_write(mod, SSI_SYS_STATUS((i * 2) + 1), 0xf << (id * 4));
+                       rsnd_mod_write(mod, SSI_SYS_STATUS((i * 2) + 1), 0xf << 4);
                break;
        }
 

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index aae450ba4f08f3c02c2b23f88b50ef3e07c8e817..50617db05c46b8a5fce3a6553c672e54996c7dfd 100644 (file)
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -735,12 +735,17 @@ static struct snd_soc_component *soc_find_component(
        const struct device_node *of_node, const char *name)
 {
        struct snd_soc_component *component;
+       struct device_node *component_of_node;
 
        lockdep_assert_held(&client_mutex);
 
        for_each_component(component) {
                if (of_node) {
-                       if (component->dev->of_node == of_node)
+                       component_of_node = component->dev->of_node;
+                       if (!component_of_node && component->dev->parent)
+                               component_of_node = component->dev->parent->of_node;
+
+                       if (component_of_node == of_node)
                                return component;
                } else if (name && strcmp(component->name, name) == 0) {
                        return component;
@@ -951,7 +956,7 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order)
 {
        int err;
 
-       if (!dai || !dai->probed ||
+       if (!dai || !dai->probed || !dai->driver ||
            dai->driver->remove_order != order)
                return;
 

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 2c4c134195392936bc03f9440c013360ac853a31..20bad755888b119d4102fa030766d492675cf4b8 100644 (file)
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -70,12 +70,16 @@ static int dapm_up_seq[] = {
        [snd_soc_dapm_clock_supply] = 1,
        [snd_soc_dapm_supply] = 2,
        [snd_soc_dapm_micbias] = 3,
+       [snd_soc_dapm_vmid] = 3,
        [snd_soc_dapm_dai_link] = 2,
        [snd_soc_dapm_dai_in] = 4,
        [snd_soc_dapm_dai_out] = 4,
        [snd_soc_dapm_aif_in] = 4,
        [snd_soc_dapm_aif_out] = 4,
        [snd_soc_dapm_mic] = 5,
+       [snd_soc_dapm_siggen] = 5,
+       [snd_soc_dapm_input] = 5,
+       [snd_soc_dapm_output] = 5,
        [snd_soc_dapm_mux] = 6,
        [snd_soc_dapm_demux] = 6,
        [snd_soc_dapm_dac] = 7,
@@ -83,11 +87,19 @@ static int dapm_up_seq[] = {
        [snd_soc_dapm_mixer] = 8,
        [snd_soc_dapm_mixer_named_ctl] = 8,
        [snd_soc_dapm_pga] = 9,
+       [snd_soc_dapm_buffer] = 9,
+       [snd_soc_dapm_scheduler] = 9,
+       [snd_soc_dapm_effect] = 9,
+       [snd_soc_dapm_src] = 9,
+       [snd_soc_dapm_asrc] = 9,
+       [snd_soc_dapm_encoder] = 9,
+       [snd_soc_dapm_decoder] = 9,
        [snd_soc_dapm_adc] = 10,
        [snd_soc_dapm_out_drv] = 11,
        [snd_soc_dapm_hp] = 11,
        [snd_soc_dapm_spk] = 11,
        [snd_soc_dapm_line] = 11,
+       [snd_soc_dapm_sink] = 11,
        [snd_soc_dapm_kcontrol] = 12,
        [snd_soc_dapm_post] = 13,
 };
@@ -100,13 +112,25 @@ static int dapm_down_seq[] = {
        [snd_soc_dapm_spk] = 3,
        [snd_soc_dapm_line] = 3,
        [snd_soc_dapm_out_drv] = 3,
+       [snd_soc_dapm_sink] = 3,
        [snd_soc_dapm_pga] = 4,
+       [snd_soc_dapm_buffer] = 4,
+       [snd_soc_dapm_scheduler] = 4,
+       [snd_soc_dapm_effect] = 4,
+       [snd_soc_dapm_src] = 4,
+       [snd_soc_dapm_asrc] = 4,
+       [snd_soc_dapm_encoder] = 4,
+       [snd_soc_dapm_decoder] = 4,
        [snd_soc_dapm_switch] = 5,
        [snd_soc_dapm_mixer_named_ctl] = 5,
        [snd_soc_dapm_mixer] = 5,
        [snd_soc_dapm_dac] = 6,
        [snd_soc_dapm_mic] = 7,
+       [snd_soc_dapm_siggen] = 7,
+       [snd_soc_dapm_input] = 7,
+       [snd_soc_dapm_output] = 7,
        [snd_soc_dapm_micbias] = 8,
+       [snd_soc_dapm_vmid] = 8,
        [snd_soc_dapm_mux] = 9,
        [snd_soc_dapm_demux] = 9,
        [snd_soc_dapm_aif_in] = 10,

diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 045ef136903d63b4a7e3339e137cb5156dedffb7..fc79ec6927e385fbd85ab6182a6788daea610726 100644 (file)
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -502,6 +502,7 @@ static void remove_dai(struct snd_soc_component *comp,
 {
        struct snd_soc_dai_driver *dai_drv =
                container_of(dobj, struct snd_soc_dai_driver, dobj);
+       struct snd_soc_dai *dai;
 
        if (pass != SOC_TPLG_PASS_PCM_DAI)
                return;
@@ -509,6 +510,10 @@ static void remove_dai(struct snd_soc_component *comp,
        if (dobj->ops && dobj->ops->dai_unload)
                dobj->ops->dai_unload(comp, dobj);
 
+       list_for_each_entry(dai, &comp->dai_list, list)
+               if (dai->driver == dai_drv)
+                       dai->driver = NULL;
+
        kfree(dai_drv->name);
        list_del(&dobj->list);
        kfree(dai_drv);

diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 382847154227ef69bb77b477402ed6a3a54f4170..db114f3977e0fb4044e88c028f9dea9bad6382af 100644 (file)
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -314,6 +314,9 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum,
        return 0;
 }
 
+/* Setup an implicit feedback endpoint from a quirk. Returns 0 if no quirk
+ * applies. Returns 1 if a quirk was found.
+ */
 static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
                                         struct usb_device *dev,
                                         struct usb_interface_descriptor *altsd,
@@ -384,7 +387,7 @@ add_sync_ep:
 
        subs->data_endpoint->sync_master = subs->sync_endpoint;
 
-       return 0;
+       return 1;
 }
 
 static int set_sync_endpoint(struct snd_usb_substream *subs,
@@ -423,6 +426,10 @@ static int set_sync_endpoint(struct snd_usb_substream *subs,
        if (err < 0)
                return err;
 
+       /* endpoint set by quirk */
+       if (err > 0)
+               return 0;
+
        if (altsd->bNumEndpoints < 2)
                return 0;