From 9a66598014db7e94193044b3a9a4a79629db2be2 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 9 Mar 2017 13:29:13 +0100 Subject: ALSA: hda - add support for docking station for HP 820 G2 [ Upstream commit 04d5466a976b096364a39a63ac264c1b3a5f8fa1 ] This tested patch adds missing initialization for Line-In/Out PINs for the docking station for HP 820 G2. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Signed-off-by: Sasha Levin Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/patch_realtek.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e5730a7d0480..2159b18f76bf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4839,6 +4839,7 @@ enum { ALC286_FIXUP_HP_GPIO_LED, ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, ALC280_FIXUP_HP_DOCK_PINS, + ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, ALC280_FIXUP_HP_9480M, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, @@ -5377,6 +5378,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC280_FIXUP_HP_GPIO4 }, + [ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x21011020 }, /* line-out */ + { 0x18, 0x2181103f }, /* line-in */ + { }, + }, + .chained = true, + .chain_id = ALC269_FIXUP_HP_GPIO_MIC1_LED + }, [ALC280_FIXUP_HP_9480M] = { .type = HDA_FIXUP_FUNC, .v.func = alc280_fixup_hp_9480m, @@ -5629,7 +5640,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2256, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2257, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2259, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x225a, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x225a, "HP", ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2260, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2263, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2264, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), @@ -5794,6 +5805,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, .name = "headset-mode-no-hp-mic"}, {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, + {.id = ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, .name = "hp-dock-gpio-mic1-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"}, {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"}, -- cgit v1.2.3 From 779214d0eaca81c7ba85d423865a42c4c4dacf75 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 9 Mar 2017 13:30:09 +0100 Subject: ALSA: hda - add support for docking station for HP 840 G3 [ Upstream commit cc3a47a248d7791ef0d2c81a35c46769e55e4c6c ] This tested patch adds missing initialization for Line-In/Out PINs for the docking station for HP 840 G3. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Signed-off-by: Sasha Levin Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/patch_conexant.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ac5de4365e15..c92b7ba344ef 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -261,6 +261,7 @@ enum { CXT_FIXUP_HP_530, CXT_FIXUP_CAP_MIX_AMP_5047, CXT_FIXUP_MUTE_LED_EAPD, + CXT_FIXUP_HP_DOCK, CXT_FIXUP_HP_SPECTRE, CXT_FIXUP_HP_GATE_MIC, }; @@ -778,6 +779,14 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_mute_led_eapd, }, + [CXT_FIXUP_HP_DOCK] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x16, 0x21011020 }, /* line-out */ + { 0x18, 0x2181103f }, /* line-in */ + { } + } + }, [CXT_FIXUP_HP_SPECTRE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -839,6 +848,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x1025, 0x054f, "Acer Aspire 4830T", CXT_FIXUP_ASPIRE_DMIC), + SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), @@ -872,6 +882,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_PINCFG_LEMOTE_A1205, .name = "lemote-a1205" }, { .id = CXT_FIXUP_OLPC_XO, .name = "olpc-xo" }, { .id = CXT_FIXUP_MUTE_LED_EAPD, .name = "mute-led-eapd" }, + { .id = CXT_FIXUP_HP_DOCK, .name = "hp-dock" }, {} }; -- cgit v1.2.3 From ef24d642e92ad2216bb0f70a4307c582f9400732 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Mar 2016 12:29:24 +0200 Subject: ALSA: hda - Clear the leftover component assignment at snd_hdac_i915_exit() commit faafd03d23c913633d2ef7e6ffebdce01b164409 upstream. The commit [d745f5e7b8b2: ALSA: hda - Add the pin / port mapping on Intel ILK and VLV] introduced a WARN_ON() to check the pointer for avoiding the double initializations. But hdac_acomp pointer wasn't cleared at snd_hdac_i915_exit(), thus after reloading the HD-audio driver, it may result in the false positive warning. This patch makes sure to clear the leftover pointer at exit. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=94736 Reported-by: Daniela Doras-prodan Signed-off-by: Takashi Iwai Cc: Kouta Okamoto Signed-off-by: Greg Kroah-Hartman --- sound/hda/hdac_i915.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 8fef1b8d1fd8..166d2eb8fd6b 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -273,6 +273,7 @@ int snd_hdac_i915_exit(struct hdac_bus *bus) kfree(acomp); bus->audio_component = NULL; + hdac_acomp = NULL; return 0; } -- cgit v1.2.3 From 3b67b56ea7703b6c5253c42d036e2aaae7aede73 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Jan 2016 15:00:26 +0100 Subject: ALSA: hda - Degrade i915 binding failure message commit bed2e98e1f4db8b827df507abc30be7b11b0613d upstream. Currently HD-audio driver on Intel Skylake or Broxteon gives an error message when binding with i915 audio component fails. However, this isn't any serious error on a system without Intel graphics. Indeed there are such systems, where a third-party codec (e.g. Creative) is put on the mobo while using other discrete GPU (e.g. Nvidia). Printing a kernel "error" message is overreaction in such a case. This patch downgrades the print level for that message. For systems that mandate the i915 binding (e.g. Haswell or Broadwell HDMI/DP), another kernel error message is shown in addition to make clear what went wrong. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=111021 Signed-off-by: Takashi Iwai Cc: Kouta Okamoto Signed-off-by: Greg Kroah-Hartman --- sound/hda/hdac_i915.c | 2 +- sound/pci/hda/hda_intel.c | 6 ++++-- 2 files changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 166d2eb8fd6b..a90322a71f86 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -240,7 +240,7 @@ out_master_del: out_err: kfree(acomp); bus->audio_component = NULL; - dev_err(dev, "failed to add i915 component master (%d)\n", ret); + dev_info(dev, "failed to add i915 component master (%d)\n", ret); return ret; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e2e08fc73b50..20512fe32a97 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2088,9 +2088,11 @@ static int azx_probe_continue(struct azx *chip) * for other chips, still continue probing as other * codecs can be on the same link. */ - if (CONTROLLER_IN_GPU(pci)) + if (CONTROLLER_IN_GPU(pci)) { + dev_err(chip->card->dev, + "HSW/BDW HD-audio HDMI/DP requires binding with gfx driver\n"); goto out_free; - else + } else goto skip_i915; } -- cgit v1.2.3 From d126c47656da7ca0c7921eea931e871642c90fb7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Mar 2016 18:48:07 +0200 Subject: ALSA: hda - Fix yet another i915 pointer leftover in error path commit 97cc2ed27e5a168cf423f67c3bc7c6cc41d12f82 upstream. The hdac_acomp object in hdac_i915.c is left as assigned even after binding with i915 actually fails, and this leads to the WARN_ON() at the next load of the module. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=94736 Signed-off-by: Takashi Iwai Cc: Kouta Okamoto Signed-off-by: Greg Kroah-Hartman --- sound/hda/hdac_i915.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index a90322a71f86..cce9ae5ec93b 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -240,6 +240,7 @@ out_master_del: out_err: kfree(acomp); bus->audio_component = NULL; + hdac_acomp = NULL; dev_info(dev, "failed to add i915 component master (%d)\n", ret); return ret; -- cgit v1.2.3 From af119535435526f4d7f0ad02863a590be641c76f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Dec 2017 16:44:12 +0100 Subject: ALSA: rawmidi: Avoid racy info ioctl via ctl device commit c1cfd9025cc394fd137a01159d74335c5ac978ce upstream. The rawmidi also allows to obtaining the information via ioctl of ctl API. It means that user can issue an ioctl to the rawmidi device even when it's being removed as long as the control device is present. Although the code has some protection via the global register_mutex, its range is limited to the search of the corresponding rawmidi object, and the mutex is already unlocked at accessing the rawmidi object. This may lead to a use-after-free. For avoiding it, this patch widens the application of register_mutex to the whole snd_rawmidi_info_select() function. We have another mutex per rawmidi object, but this operation isn't very hot path, so it shouldn't matter from the performance POV. Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/core/rawmidi.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index b450a27588c8..16f8124b1150 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -579,15 +579,14 @@ static int snd_rawmidi_info_user(struct snd_rawmidi_substream *substream, return 0; } -int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info) +static int __snd_rawmidi_info_select(struct snd_card *card, + struct snd_rawmidi_info *info) { struct snd_rawmidi *rmidi; struct snd_rawmidi_str *pstr; struct snd_rawmidi_substream *substream; - mutex_lock(®ister_mutex); rmidi = snd_rawmidi_search(card, info->device); - mutex_unlock(®ister_mutex); if (!rmidi) return -ENXIO; if (info->stream < 0 || info->stream > 1) @@ -603,6 +602,16 @@ int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info } return -ENXIO; } + +int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info) +{ + int ret; + + mutex_lock(®ister_mutex); + ret = __snd_rawmidi_info_select(card, info); + mutex_unlock(®ister_mutex); + return ret; +} EXPORT_SYMBOL(snd_rawmidi_info_select); static int snd_rawmidi_info_select_user(struct snd_card *card, -- cgit v1.2.3 From 6b08ff879603eb320d20234e0413f5627df5c629 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Dec 2017 23:36:57 +0100 Subject: ALSA: usb-audio: Fix the missing ctl name suffix at parsing SU commit 5a15f289ee87eaf33f13f08a4909ec99d837ec5f upstream. The commit 89b89d121ffc ("ALSA: usb-audio: Add check return value for usb_string()") added the check of the return value from snd_usb_copy_string_desc(), which is correct per se, but it introduced a regression. In the original code, either the "Clock Source", "Playback Source" or "Capture Source" suffix is added after the terminal string, while the commit changed it to add the suffix only when get_term_name() is failing. It ended up with an incorrect ctl name like "PCM" instead of "PCM Capture Source". Also, even the original code has a similar bug: when the ctl name is generated from snd_usb_copy_string_desc() for the given iSelector, it also doesn't put the suffix. This patch addresses these issues: the suffix is added always when no static mapping is found. Also the patch tries to put more comments and cleans up the if/else block for better readability in order to avoid the same pitfall again. Fixes: 89b89d121ffc ("ALSA: usb-audio: Add check return value for usb_string()") Reported-and-tested-by: Mauro Santos Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/usb/mixer.c | 27 ++++++++++++++++----------- 1 file changed, 16 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 0ed9ae030ce1..c9ae29068c7c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2101,20 +2101,25 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, kctl->private_value = (unsigned long)namelist; kctl->private_free = usb_mixer_selector_elem_free; - nameid = uac_selector_unit_iSelector(desc); + /* check the static mapping table at first */ len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); - if (len) - ; - else if (nameid) - len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, - sizeof(kctl->id.name)); - else - len = get_term_name(state, &state->oterm, - kctl->id.name, sizeof(kctl->id.name), 0); - if (!len) { - strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); + /* no mapping ? */ + /* if iSelector is given, use it */ + nameid = uac_selector_unit_iSelector(desc); + if (nameid) + len = snd_usb_copy_string_desc(state, nameid, + kctl->id.name, + sizeof(kctl->id.name)); + /* ... or pick up the terminal name at next */ + if (!len) + len = get_term_name(state, &state->oterm, + kctl->id.name, sizeof(kctl->id.name), 0); + /* ... or use the fixed string "USB" as the last resort */ + if (!len) + strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); + /* and add the proper suffix */ if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) append_ctl_name(kctl, " Clock Source"); else if ((state->oterm.type & 0xff00) == 0x0100) -- cgit v1.2.3 From 3096ced5a93969a7d045ee06a14197a630415af4 Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Mon, 20 Nov 2017 23:14:55 +0100 Subject: ASoC: fsl_ssi: AC'97 ops need regmap, clock and cleaning up on failure commit 695b78b548d8a26288f041e907ff17758df9e1d5 upstream. AC'97 ops (register read / write) need SSI regmap and clock, so they have to be set after them. We also need to set these ops back to NULL if we fail the probe. Signed-off-by: Maciej S. Szmigiero Acked-by: Nicolin Chen Signed-off-by: Mark Brown Signed-off-by: Greg Kroah-Hartman --- sound/soc/fsl/fsl_ssi.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 95d2392303eb..7ca67613e0d4 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1408,12 +1408,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) sizeof(fsl_ssi_ac97_dai)); fsl_ac97_data = ssi_private; - - ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); - if (ret) { - dev_err(&pdev->dev, "could not set AC'97 ops\n"); - return ret; - } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1473,6 +1467,14 @@ static int fsl_ssi_probe(struct platform_device *pdev) return ret; } + if (fsl_ssi_is_ac97(ssi_private)) { + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + goto error_ac97_ops; + } + } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_ssi_component, &ssi_private->cpu_dai_drv, 1); if (ret) { @@ -1556,6 +1558,10 @@ error_sound_card: fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); error_asoc_register: + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + +error_ac97_ops: if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); -- cgit v1.2.3 From 5251932b974d23e36bd0a5d1d05fcaa7e94d8302 Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Mon, 13 Nov 2017 12:12:56 +0100 Subject: ASoC: twl4030: fix child-node lookup commit 15f8c5f2415bfac73f33a14bcd83422bcbfb5298 upstream. Fix child-node lookup during probe, which ended up searching the whole device tree depth-first starting at the parent rather than just matching on its children. To make things worse, the parent codec node was also prematurely freed, while the child node was leaked. Fixes: 2d6d649a2e0f ("ASoC: twl4030: Support for DT booted kernel") Signed-off-by: Johan Hovold Signed-off-by: Mark Brown Signed-off-by: Greg Kroah-Hartman --- sound/soc/codecs/twl4030.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index a5a4e9f75c57..a06395507225 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -232,7 +232,7 @@ static struct twl4030_codec_data *twl4030_get_pdata(struct snd_soc_codec *codec) struct twl4030_codec_data *pdata = dev_get_platdata(codec->dev); struct device_node *twl4030_codec_node = NULL; - twl4030_codec_node = of_find_node_by_name(codec->dev->parent->of_node, + twl4030_codec_node = of_get_child_by_name(codec->dev->parent->of_node, "codec"); if (!pdata && twl4030_codec_node) { @@ -241,9 +241,11 @@ static struct twl4030_codec_data *twl4030_get_pdata(struct snd_soc_codec *codec) GFP_KERNEL); if (!pdata) { dev_err(codec->dev, "Can not allocate memory\n"); + of_node_put(twl4030_codec_node); return NULL; } twl4030_setup_pdata_of(pdata, twl4030_codec_node); + of_node_put(twl4030_codec_node); } return pdata; -- cgit v1.2.3 From 0ba2ebc9f355c87c13e37ad136e9c4508a8a5029 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 22 Dec 2017 10:45:07 +0100 Subject: ALSA: hda: Drop useless WARN_ON() commit a36c2638380c0a4676647a1f553b70b20d3ebce1 upstream. Since the commit 97cc2ed27e5a ("ALSA: hda - Fix yet another i915 pointer leftover in error path") cleared hdac_acomp pointer, the WARN_ON() non-NULL check in snd_hdac_i915_register_notifier() may give a false-positive warning, as the function gets called no matter whether the component is registered or not. For fixing it, let's get rid of the spurious WARN_ON(). Fixes: 97cc2ed27e5a ("ALSA: hda - Fix yet another i915 pointer leftover in error path") Reported-by: Kouta Okamoto Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/hda/hdac_i915.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index cce9ae5ec93b..bd7bcf428bcf 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -183,7 +183,7 @@ static int hdac_component_master_match(struct device *dev, void *data) */ int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *aops) { - if (WARN_ON(!hdac_acomp)) + if (!hdac_acomp) return -ENODEV; hdac_acomp->audio_ops = aops; -- cgit v1.2.3 From c04ed3a849616f2c67ff016f50935ff1553c6751 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Fri, 22 Dec 2017 11:17:45 +0800 Subject: ALSA: hda - fix headset mic detection issue on a Dell machine commit 285d5ddcffafa5d5e68c586f4c9eaa8b24a2897d upstream. It has the codec alc256, and add its pin definition to pin quirk table to let it apply ALC255_FIXUP_DELL1_MIC_NO_PRESENCE. Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2159b18f76bf..5875a08d555e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5953,6 +5953,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x1b, 0x01011020}, {0x21, 0x02211010}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x14, 0x90170110}, + {0x1b, 0x01011020}, + {0x21, 0x0221101f}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, -- cgit v1.2.3 From 3074fe070a89bd1c14f8d70e5a5db7f78b84496e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 1 Jan 2018 09:50:50 +0100 Subject: ALSA: pcm: Remove incorrect snd_BUG_ON() usages commit fe08f34d066f4404934a509b6806db1a4f700c86 upstream. syzkaller triggered kernel warnings through PCM OSS emulation at closing a stream: WARNING: CPU: 0 PID: 3502 at sound/core/pcm_lib.c:1635 snd_pcm_hw_param_first+0x289/0x690 sound/core/pcm_lib.c:1635 Call Trace: .... snd_pcm_hw_param_near.constprop.27+0x78d/0x9a0 sound/core/oss/pcm_oss.c:457 snd_pcm_oss_change_params+0x17d3/0x3720 sound/core/oss/pcm_oss.c:969 snd_pcm_oss_make_ready+0xaa/0x130 sound/core/oss/pcm_oss.c:1128 snd_pcm_oss_sync+0x257/0x830 sound/core/oss/pcm_oss.c:1638 snd_pcm_oss_release+0x20b/0x280 sound/core/oss/pcm_oss.c:2431 __fput+0x327/0x7e0 fs/file_table.c:210 .... This happens while it tries to open and set up the aloop device concurrently. The warning above (invoked from snd_BUG_ON() macro) is to detect the unexpected logical error where snd_pcm_hw_refine() call shouldn't fail. The theory is true for the case where the hw_params config rules are static. But for an aloop device, the hw_params rule condition does vary dynamically depending on the connected target; when another device is opened and changes the parameters, the device connected in another side is also affected, and it caused the error from snd_pcm_hw_refine(). That is, the simplest "solution" for this is to remove the incorrect assumption of static rules, and treat such an error as a normal error path. As there are a couple of other places using snd_BUG_ON() incorrectly, this patch removes these spurious snd_BUG_ON() calls. Reported-by: syzbot+6f11c7e2a1b91d466432@syzkaller.appspotmail.com Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/core/oss/pcm_oss.c | 1 - sound/core/pcm_lib.c | 4 ++-- 2 files changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 33e72c809e50..4a5bcf178982 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -465,7 +465,6 @@ static int snd_pcm_hw_param_near(struct snd_pcm_substream *pcm, v = snd_pcm_hw_param_last(pcm, params, var, dir); else v = snd_pcm_hw_param_first(pcm, params, var, dir); - snd_BUG_ON(v < 0); return v; } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index cd20f91326fe..7b805766306e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1664,7 +1664,7 @@ int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, return changed; if (params->rmask) { int err = snd_pcm_hw_refine(pcm, params); - if (snd_BUG_ON(err < 0)) + if (err < 0) return err; } return snd_pcm_hw_param_value(params, var, dir); @@ -1711,7 +1711,7 @@ int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm, return changed; if (params->rmask) { int err = snd_pcm_hw_refine(pcm, params); - if (snd_BUG_ON(err < 0)) + if (err < 0) return err; } return snd_pcm_hw_param_value(params, var, dir); -- cgit v1.2.3 From 1ee7bc5526d8056b23dfd65948f4c7b57c883fa5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Jan 2018 16:39:27 +0100 Subject: ALSA: pcm: Add missing error checks in OSS emulation plugin builder commit 6708913750344a900f2e73bfe4a4d6dbbce4fe8d upstream. In the OSS emulation plugin builder where the frame size is parsed in the plugin chain, some places miss the possible errors returned from the plugin src_ or dst_frames callback. This patch papers over such places. Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/core/oss/pcm_plugin.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 727ac44d39f4..a84a1d3d23e5 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -591,18 +591,26 @@ snd_pcm_sframes_t snd_pcm_plug_write_transfer(struct snd_pcm_substream *plug, st snd_pcm_sframes_t frames = size; plugin = snd_pcm_plug_first(plug); - while (plugin && frames > 0) { + while (plugin) { + if (frames <= 0) + return frames; if ((next = plugin->next) != NULL) { snd_pcm_sframes_t frames1 = frames; - if (plugin->dst_frames) + if (plugin->dst_frames) { frames1 = plugin->dst_frames(plugin, frames); + if (frames1 <= 0) + return frames1; + } if ((err = next->client_channels(next, frames1, &dst_channels)) < 0) { return err; } if (err != frames1) { frames = err; - if (plugin->src_frames) + if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames1); + if (frames <= 0) + return frames; + } } } else dst_channels = NULL; -- cgit v1.2.3 From 9bb4bb18ccffc4dc4a1f1038a5dc0fb3a4020c05 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Jan 2018 13:58:31 +0100 Subject: ALSA: pcm: Abort properly at pending signal in OSS read/write loops commit 29159a4ed7044c52e3e2cf1a9fb55cec4745c60b upstream. The loops for read and write in PCM OSS emulation have no proper check of pending signals, and they keep processing even after user tries to break. This results in a very long delay, often seen as RCU stall when a huge unprocessed bytes remain queued. The bug could be easily triggered by syzkaller. As a simple workaround, this patch adds the proper check of pending signals and aborts the loop appropriately. Reported-by: syzbot+993cb4cfcbbff3947c21@syzkaller.appspotmail.com Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/core/oss/pcm_oss.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 4a5bcf178982..d2a9e0fd46b0 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1416,6 +1416,10 @@ static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const cha tmp != runtime->oss.period_bytes) break; } + if (signal_pending(current)) { + tmp = -ERESTARTSYS; + goto err; + } } mutex_unlock(&runtime->oss.params_lock); return xfer; @@ -1501,6 +1505,10 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use bytes -= tmp; xfer += tmp; } + if (signal_pending(current)) { + tmp = -ERESTARTSYS; + goto err; + } } mutex_unlock(&runtime->oss.params_lock); return xfer; -- cgit v1.2.3 From fa6c1876ecf724f4bb77842770941db41d35c3f9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Jan 2018 14:03:53 +0100 Subject: ALSA: pcm: Allow aborting mutex lock at OSS read/write loops commit 900498a34a3ac9c611e9b425094c8106bdd7dc1c upstream. PCM OSS read/write loops keep taking the mutex lock for the whole read/write, and this might take very long when the exceptionally high amount of data is given. Also, since it invokes with mutex_lock(), the concurrent read/write becomes unbreakable. This patch tries to address these issues by replacing mutex_lock() with mutex_lock_interruptible(), and also splits / re-takes the lock at each read/write period chunk, so that it can switch the context more finely if requested. Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/core/oss/pcm_oss.c | 36 +++++++++++++++++++++--------------- 1 file changed, 21 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index d2a9e0fd46b0..494b7b533366 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1369,8 +1369,11 @@ static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const cha if ((tmp = snd_pcm_oss_make_ready(substream)) < 0) return tmp; - mutex_lock(&runtime->oss.params_lock); while (bytes > 0) { + if (mutex_lock_interruptible(&runtime->oss.params_lock)) { + tmp = -ERESTARTSYS; + break; + } if (bytes < runtime->oss.period_bytes || runtime->oss.buffer_used > 0) { tmp = bytes; if (tmp + runtime->oss.buffer_used > runtime->oss.period_bytes) @@ -1414,18 +1417,18 @@ static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const cha xfer += tmp; if ((substream->f_flags & O_NONBLOCK) != 0 && tmp != runtime->oss.period_bytes) - break; + tmp = -EAGAIN; } + err: + mutex_unlock(&runtime->oss.params_lock); + if (tmp < 0) + break; if (signal_pending(current)) { tmp = -ERESTARTSYS; - goto err; + break; } + tmp = 0; } - mutex_unlock(&runtime->oss.params_lock); - return xfer; - - err: - mutex_unlock(&runtime->oss.params_lock); return xfer > 0 ? (snd_pcm_sframes_t)xfer : tmp; } @@ -1473,8 +1476,11 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use if ((tmp = snd_pcm_oss_make_ready(substream)) < 0) return tmp; - mutex_lock(&runtime->oss.params_lock); while (bytes > 0) { + if (mutex_lock_interruptible(&runtime->oss.params_lock)) { + tmp = -ERESTARTSYS; + break; + } if (bytes < runtime->oss.period_bytes || runtime->oss.buffer_used > 0) { if (runtime->oss.buffer_used == 0) { tmp = snd_pcm_oss_read2(substream, runtime->oss.buffer, runtime->oss.period_bytes, 1); @@ -1505,16 +1511,16 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use bytes -= tmp; xfer += tmp; } + err: + mutex_unlock(&runtime->oss.params_lock); + if (tmp < 0) + break; if (signal_pending(current)) { tmp = -ERESTARTSYS; - goto err; + break; } + tmp = 0; } - mutex_unlock(&runtime->oss.params_lock); - return xfer; - - err: - mutex_unlock(&runtime->oss.params_lock); return xfer > 0 ? (snd_pcm_sframes_t)xfer : tmp; } -- cgit v1.2.3 From 3d3b2c61e1bea49bdbf42169e9f28ca889f4d707 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 5 Jan 2018 16:09:47 +0100 Subject: ALSA: aloop: Release cable upon open error path commit 9685347aa0a5c2869058ca6ab79fd8e93084a67f upstream. The aloop runtime object and its assignment in the cable are left even when opening a substream fails. This doesn't mean any memory leak, but it still keeps the invalid pointer that may be referred by the another side of the cable spontaneously, which is a potential Oops cause. Clean up the cable assignment and the empty cable upon the error path properly. Fixes: 597603d615d2 ("ALSA: introduce the snd-aloop module for the PCM loopback") Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/drivers/aloop.c | 38 +++++++++++++++++++++++++------------- 1 file changed, 25 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 54f348a4fb78..2adc88d6d507 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -658,12 +658,31 @@ static int rule_channels(struct snd_pcm_hw_params *params, return snd_interval_refine(hw_param_interval(params, rule->var), &t); } +static void free_cable(struct snd_pcm_substream *substream) +{ + struct loopback *loopback = substream->private_data; + int dev = get_cable_index(substream); + struct loopback_cable *cable; + + cable = loopback->cables[substream->number][dev]; + if (!cable) + return; + if (cable->streams[!substream->stream]) { + /* other stream is still alive */ + cable->streams[substream->stream] = NULL; + } else { + /* free the cable */ + loopback->cables[substream->number][dev] = NULL; + kfree(cable); + } +} + static int loopback_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct loopback *loopback = substream->private_data; struct loopback_pcm *dpcm; - struct loopback_cable *cable; + struct loopback_cable *cable = NULL; int err = 0; int dev = get_cable_index(substream); @@ -682,7 +701,6 @@ static int loopback_open(struct snd_pcm_substream *substream) if (!cable) { cable = kzalloc(sizeof(*cable), GFP_KERNEL); if (!cable) { - kfree(dpcm); err = -ENOMEM; goto unlock; } @@ -724,6 +742,10 @@ static int loopback_open(struct snd_pcm_substream *substream) else runtime->hw = cable->hw; unlock: + if (err < 0) { + free_cable(substream); + kfree(dpcm); + } mutex_unlock(&loopback->cable_lock); return err; } @@ -732,20 +754,10 @@ static int loopback_close(struct snd_pcm_substream *substream) { struct loopback *loopback = substream->private_data; struct loopback_pcm *dpcm = substream->runtime->private_data; - struct loopback_cable *cable; - int dev = get_cable_index(substream); loopback_timer_stop(dpcm); mutex_lock(&loopback->cable_lock); - cable = loopback->cables[substream->number][dev]; - if (cable->streams[!substream->stream]) { - /* other stream is still alive */ - cable->streams[substream->stream] = NULL; - } else { - /* free the cable */ - loopback->cables[substream->number][dev] = NULL; - kfree(cable); - } + free_cable(substream); mutex_unlock(&loopback->cable_lock); return 0; } -- cgit v1.2.3 From a9cad56436f43c781239a58c4904dc2c34fe0921 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 5 Jan 2018 16:15:33 +0100 Subject: ALSA: aloop: Fix inconsistent format due to incomplete rule commit b088b53e20c7d09b5ab84c5688e609f478e5c417 upstream. The extra hw constraint rule for the formats the aloop driver introduced has a slight flaw, where it doesn't return a positive value when the mask got changed. It came from the fact that it's basically a copy&paste from snd_hw_constraint_mask64(). The original code is supposed to be a single-shot and it modifies the mask bits only once and never after, while what we need for aloop is the dynamic hw rule that limits the mask bits. This difference results in the inconsistent state, as the hw_refine doesn't apply the dependencies fully. The worse and surprisingly result is that it causes a crash in OSS emulation when multiple full-duplex reads/writes are performed concurrently (I leave why it triggers Oops to readers as a homework). For fixing this, replace a few open-codes with the standard snd_mask_*() macros. Reported-by: syzbot+3902b5220e8ca27889ca@syzkaller.appspotmail.com Fixes: b1c73fc8e697 ("ALSA: snd-aloop: Fix hw_params restrictions and checking") Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/drivers/aloop.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 2adc88d6d507..59e4a88757b1 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -39,6 +39,7 @@ #include #include #include +#include #include #include @@ -622,14 +623,12 @@ static int rule_format(struct snd_pcm_hw_params *params, { struct snd_pcm_hardware *hw = rule->private; - struct snd_mask *maskp = hw_param_mask(params, rule->var); + struct snd_mask m; - maskp->bits[0] &= (u_int32_t)hw->formats; - maskp->bits[1] &= (u_int32_t)(hw->formats >> 32); - memset(maskp->bits + 2, 0, (SNDRV_MASK_MAX-64) / 8); /* clear rest */ - if (! maskp->bits[0] && ! maskp->bits[1]) - return -EINVAL; - return 0; + snd_mask_none(&m); + m.bits[0] = (u_int32_t)hw->formats; + m.bits[1] = (u_int32_t)(hw->formats >> 32); + return snd_mask_refine(hw_param_mask(params, rule->var), &m); } static int rule_rate(struct snd_pcm_hw_params *params, -- cgit v1.2.3 From d091a2bb8c2e9801875531b6cb14e1df1729045c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Jan 2018 17:38:54 +0100 Subject: ALSA: aloop: Fix racy hw constraints adjustment commit 898dfe4687f460ba337a01c11549f87269a13fa2 upstream. The aloop driver tries to update the hw constraints of the connected target on the cable of the opened PCM substream. This is done by adding the extra hw constraints rules referring to the substream runtime->hw fields, while the other substream may update the runtime hw of another side on the fly. This is, however, racy and may result in the inconsistent values when both PCM streams perform the prepare concurrently. One of the reason is that it overwrites the other's runtime->hw field; which is not only racy but also broken when it's called before the open of another side finishes. And, since the reference to runtime->hw isn't protected, the concurrent write may give the partial value update and become inconsistent. This patch is an attempt to fix and clean up: - The prepare doesn't change the runtime->hw of other side any longer, but only update the cable->hw that is referred commonly. - The extra rules refer to the loopback_pcm object instead of the runtime->hw. The actual hw is deduced from cable->hw. - The extra rules take the cable_lock to protect against the race. Fixes: b1c73fc8e697 ("ALSA: snd-aloop: Fix hw_params restrictions and checking") Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/drivers/aloop.c | 51 +++++++++++++++++++++------------------------------ 1 file changed, 21 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 59e4a88757b1..cbd20cb8ca11 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -306,19 +306,6 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } -static void params_change_substream(struct loopback_pcm *dpcm, - struct snd_pcm_runtime *runtime) -{ - struct snd_pcm_runtime *dst_runtime; - - if (dpcm == NULL || dpcm->substream == NULL) - return; - dst_runtime = dpcm->substream->runtime; - if (dst_runtime == NULL) - return; - dst_runtime->hw = dpcm->cable->hw; -} - static void params_change(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -330,10 +317,6 @@ static void params_change(struct snd_pcm_substream *substream) cable->hw.rate_max = runtime->rate; cable->hw.channels_min = runtime->channels; cable->hw.channels_max = runtime->channels; - params_change_substream(cable->streams[SNDRV_PCM_STREAM_PLAYBACK], - runtime); - params_change_substream(cable->streams[SNDRV_PCM_STREAM_CAPTURE], - runtime); } static int loopback_prepare(struct snd_pcm_substream *substream) @@ -621,24 +604,29 @@ static unsigned int get_cable_index(struct snd_pcm_substream *substream) static int rule_format(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - - struct snd_pcm_hardware *hw = rule->private; + struct loopback_pcm *dpcm = rule->private; + struct loopback_cable *cable = dpcm->cable; struct snd_mask m; snd_mask_none(&m); - m.bits[0] = (u_int32_t)hw->formats; - m.bits[1] = (u_int32_t)(hw->formats >> 32); + mutex_lock(&dpcm->loopback->cable_lock); + m.bits[0] = (u_int32_t)cable->hw.formats; + m.bits[1] = (u_int32_t)(cable->hw.formats >> 32); + mutex_unlock(&dpcm->loopback->cable_lock); return snd_mask_refine(hw_param_mask(params, rule->var), &m); } static int rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hardware *hw = rule->private; + struct loopback_pcm *dpcm = rule->private; + struct loopback_cable *cable = dpcm->cable; struct snd_interval t; - t.min = hw->rate_min; - t.max = hw->rate_max; + mutex_lock(&dpcm->loopback->cable_lock); + t.min = cable->hw.rate_min; + t.max = cable->hw.rate_max; + mutex_unlock(&dpcm->loopback->cable_lock); t.openmin = t.openmax = 0; t.integer = 0; return snd_interval_refine(hw_param_interval(params, rule->var), &t); @@ -647,11 +635,14 @@ static int rule_rate(struct snd_pcm_hw_params *params, static int rule_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hardware *hw = rule->private; + struct loopback_pcm *dpcm = rule->private; + struct loopback_cable *cable = dpcm->cable; struct snd_interval t; - t.min = hw->channels_min; - t.max = hw->channels_max; + mutex_lock(&dpcm->loopback->cable_lock); + t.min = cable->hw.channels_min; + t.max = cable->hw.channels_max; + mutex_unlock(&dpcm->loopback->cable_lock); t.openmin = t.openmax = 0; t.integer = 0; return snd_interval_refine(hw_param_interval(params, rule->var), &t); @@ -717,19 +708,19 @@ static int loopback_open(struct snd_pcm_substream *substream) /* are cached -> they do not reflect the actual state */ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, - rule_format, &runtime->hw, + rule_format, dpcm, SNDRV_PCM_HW_PARAM_FORMAT, -1); if (err < 0) goto unlock; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - rule_rate, &runtime->hw, + rule_rate, dpcm, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) goto unlock; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - rule_channels, &runtime->hw, + rule_channels, dpcm, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) goto unlock; -- cgit v1.2.3 From 80547bb6154d02279e446526a7d3daf7d5aa15db Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 10 Jan 2018 23:48:05 +0100 Subject: ALSA: pcm: Remove yet superfluous WARN_ON() commit 23b19b7b50fe1867da8d431eea9cd3e4b6328c2c upstream. muldiv32() contains a snd_BUG_ON() (which is morphed as WARN_ON() with debug option) for checking the case of 0 / 0. This would be helpful if this happens only as a logical error; however, since the hw refine is performed with any data set provided by user, the inconsistent values that can trigger such a condition might be passed easily. Actually, syzbot caught this by passing some zero'ed old hw_params ioctl. So, having snd_BUG_ON() there is simply superfluous and rather harmful to give unnecessary confusions. Let's get rid of it. Reported-by: syzbot+7e6ee55011deeebce15d@syzkaller.appspotmail.com Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/core/pcm_lib.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7b805766306e..4c145d6bccd4 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -578,7 +578,6 @@ static inline unsigned int muldiv32(unsigned int a, unsigned int b, { u_int64_t n = (u_int64_t) a * b; if (c == 0) { - snd_BUG_ON(!n); *r = 0; return UINT_MAX; } -- cgit v1.2.3 From a4d7639d5fb65070e43f679493341f1ec4212e6e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 10 Jan 2018 08:34:28 +0100 Subject: ALSA: hda - Apply headphone noise quirk for another Dell XPS 13 variant commit e4c9fd10eb21376f44723c40ad12395089251c28 upstream. There is another Dell XPS 13 variant (SSID 1028:082a) that requires the existing fixup for reducing the headphone noise. This patch adds the quirk entry for that. BugLink: http://lkml.kernel.org/r/CAHXyb9ZCZJzVisuBARa+UORcjRERV8yokez=DP1_5O5isTz0ZA@mail.gmail.com Reported-and-tested-by: Francisco G. Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5875a08d555e..f14c1f288443 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5600,6 +5600,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), + SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- cgit v1.2.3 From 478a7fa82ff78be3b7aa94a9994fe0544b931f53 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 10 Jan 2018 10:53:18 +0100 Subject: ALSA: hda - Apply the existing quirk to iMac 14,1 commit 031f335cda879450095873003abb03ae8ed3b74a upstream. iMac 14,1 requires the same quirk as iMac 12,2, using GPIO 2 and 3 for headphone and speaker output amps. Add the codec SSID quirk entry (106b:0600) accordingly. BugLink: http://lkml.kernel.org/r/CAEw6Zyteav09VGHRfD5QwsfuWv5a43r0tFBNbfcHXoNrxVz7ew@mail.gmail.com Reported-by: Freaky Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 80bbadc83721..d6e079f4ec09 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -408,6 +408,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = { /*SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),*/ /* codec SSID */ + SND_PCI_QUIRK(0x106b, 0x0600, "iMac 14,1", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81), SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101), -- cgit v1.2.3 From 623e5c8ae32b39cc8baea83478695dc624935318 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 9 Jan 2018 23:11:03 +0100 Subject: ALSA: seq: Make ioctls race-free commit b3defb791b26ea0683a93a4f49c77ec45ec96f10 upstream. The ALSA sequencer ioctls have no protection against racy calls while the concurrent operations may lead to interfere with each other. As reported recently, for example, the concurrent calls of setting client pool with a combination of write calls may lead to either the unkillable dead-lock or UAF. As a slightly big hammer solution, this patch introduces the mutex to make each ioctl exclusive. Although this may reduce performance via parallel ioctl calls, usually it's not demanded for sequencer usages, hence it should be negligible. Reported-by: Luo Quan Reviewed-by: Kees Cook Reviewed-by: Greg Kroah-Hartman Signed-off-by: Takashi Iwai [bwh: Backported to 4.4: ioctl dispatch is done from snd_seq_do_ioctl(); take the mutex and add ret variable there.] Signed-off-by: Ben Hutchings Signed-off-by: Greg Kroah-Hartman --- sound/core/seq/seq_clientmgr.c | 10 ++++++++-- sound/core/seq/seq_clientmgr.h | 1 + 2 files changed, 9 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index b36de76f24e2..7bb9fe7a2c8e 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -236,6 +236,7 @@ static struct snd_seq_client *seq_create_client1(int client_index, int poolsize) rwlock_init(&client->ports_lock); mutex_init(&client->ports_mutex); INIT_LIST_HEAD(&client->ports_list_head); + mutex_init(&client->ioctl_mutex); /* find free slot in the client table */ spin_lock_irqsave(&clients_lock, flags); @@ -2195,6 +2196,7 @@ static int snd_seq_do_ioctl(struct snd_seq_client *client, unsigned int cmd, void __user *arg) { struct seq_ioctl_table *p; + int ret; switch (cmd) { case SNDRV_SEQ_IOCTL_PVERSION: @@ -2208,8 +2210,12 @@ static int snd_seq_do_ioctl(struct snd_seq_client *client, unsigned int cmd, if (! arg) return -EFAULT; for (p = ioctl_tables; p->cmd; p++) { - if (p->cmd == cmd) - return p->func(client, arg); + if (p->cmd == cmd) { + mutex_lock(&client->ioctl_mutex); + ret = p->func(client, arg); + mutex_unlock(&client->ioctl_mutex); + return ret; + } } pr_debug("ALSA: seq unknown ioctl() 0x%x (type='%c', number=0x%02x)\n", cmd, _IOC_TYPE(cmd), _IOC_NR(cmd)); diff --git a/sound/core/seq/seq_clientmgr.h b/sound/core/seq/seq_clientmgr.h index 20f0a725ec7d..91f8f165bfdc 100644 --- a/sound/core/seq/seq_clientmgr.h +++ b/sound/core/seq/seq_clientmgr.h @@ -59,6 +59,7 @@ struct snd_seq_client { struct list_head ports_list_head; rwlock_t ports_lock; struct mutex ports_mutex; + struct mutex ioctl_mutex; int convert32; /* convert 32->64bit */ /* output pool */ -- cgit v1.2.3 From f056ba2f25857c55433341ccf473944082819f12 Mon Sep 17 00:00:00 2001 From: Jesse Chan Date: Sun, 19 Nov 2017 23:45:49 -0800 Subject: ASoC: pcm512x: add missing MODULE_DESCRIPTION/AUTHOR/LICENSE commit 0cab20cec0b663b7be8e2be5998d5a4113647f86 upstream. This change resolves a new compile-time warning when built as a loadable module: WARNING: modpost: missing MODULE_LICENSE() in sound/soc/codecs/snd-soc-pcm512x-spi.o see include/linux/module.h for more information This adds the license as "GPL v2", which matches the header of the file. MODULE_DESCRIPTION and MODULE_AUTHOR are also added. Signed-off-by: Jesse Chan Signed-off-by: Mark Brown Signed-off-by: Greg Kroah-Hartman --- sound/soc/codecs/pcm512x-spi.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/pcm512x-spi.c b/sound/soc/codecs/pcm512x-spi.c index 712ed6598c48..ebdf9bd5a64c 100644 --- a/sound/soc/codecs/pcm512x-spi.c +++ b/sound/soc/codecs/pcm512x-spi.c @@ -70,3 +70,7 @@ static struct spi_driver pcm512x_spi_driver = { }; module_spi_driver(pcm512x_spi_driver); + +MODULE_DESCRIPTION("ASoC PCM512x codec driver - SPI"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From 69fcbf02d56f3bb08d661b21429f7fcf5374e439 Mon Sep 17 00:00:00 2001 From: Julian Scheel Date: Wed, 24 May 2017 12:28:23 +0200 Subject: ASoC: simple-card: Fix misleading error message commit 7ac45d1635a4cd2e99a4b11903d4a2815ca1b27b upstream. In case cpu could not be found the error message would always refer to /codec/ not being found in DT. Fix this by catching the cpu node not found case explicitly. Signed-off-by: Julian Scheel Signed-off-by: Mark Brown Signed-off-by: thongsyho Signed-off-by: Nhan Nguyen Signed-off-by: Greg Kroah-Hartman --- sound/soc/generic/simple-card.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index ff6fcd9f92f7..0b1b6fcb7500 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -343,13 +343,19 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, snprintf(prop, sizeof(prop), "%scpu", prefix); cpu = of_get_child_by_name(node, prop); + if (!cpu) { + ret = -EINVAL; + dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop); + goto dai_link_of_err; + } + snprintf(prop, sizeof(prop), "%splat", prefix); plat = of_get_child_by_name(node, prop); snprintf(prop, sizeof(prop), "%scodec", prefix); codec = of_get_child_by_name(node, prop); - if (!cpu || !codec) { + if (!codec) { ret = -EINVAL; dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop); goto dai_link_of_err; -- cgit v1.2.3 From e09eea9417d7c57546f12a5678440d3932960f71 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 16 May 2017 01:48:24 +0000 Subject: ASoC: rsnd: don't call free_irq() on Parent SSI commit 1f8754d4daea5f257370a52a30fcb22798c54516 upstream. If SSI uses shared pin, some SSI will be used as parent SSI. Then, normal SSI's remove and Parent SSI's remove (these are same SSI) will be called when unbind or remove timing. In this case, free_irq() will be called twice. This patch solve this issue. Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Reported-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown Signed-off-by: thongsyho Signed-off-by: Nhan Nguyen Signed-off-by: Greg Kroah-Hartman --- sound/soc/sh/rcar/rsnd.h | 2 ++ sound/soc/sh/rcar/ssi.c | 5 +++++ 2 files changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 085329878525..5976e3992dd1 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -235,6 +235,7 @@ enum rsnd_mod_type { RSND_MOD_MIX, RSND_MOD_CTU, RSND_MOD_SRC, + RSND_MOD_SSIP, /* SSI parent */ RSND_MOD_SSI, RSND_MOD_MAX, }; @@ -365,6 +366,7 @@ struct rsnd_dai_stream { }; #define rsnd_io_to_mod(io, i) ((i) < RSND_MOD_MAX ? (io)->mod[(i)] : NULL) #define rsnd_io_to_mod_ssi(io) rsnd_io_to_mod((io), RSND_MOD_SSI) +#define rsnd_io_to_mod_ssip(io) rsnd_io_to_mod((io), RSND_MOD_SSIP) #define rsnd_io_to_mod_src(io) rsnd_io_to_mod((io), RSND_MOD_SRC) #define rsnd_io_to_mod_ctu(io) rsnd_io_to_mod((io), RSND_MOD_CTU) #define rsnd_io_to_mod_mix(io) rsnd_io_to_mod((io), RSND_MOD_MIX) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index c62a2947ac14..94739b7aca77 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -550,11 +550,16 @@ static int rsnd_ssi_dma_remove(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct rsnd_mod *ssi_parent_mod = rsnd_io_to_mod_ssip(io); struct device *dev = rsnd_priv_to_dev(priv); int irq = ssi->info->irq; rsnd_dma_quit(io, rsnd_mod_to_dma(mod)); + /* Do nothing for SSI parent mod */ + if (ssi_parent_mod == mod) + return 0; + /* PIO will request IRQ again */ devm_free_irq(dev, irq, mod); -- cgit v1.2.3 From 1e5ed917dc65aee4c8840686bf83c4a55ad4210d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 9 Aug 2017 02:16:20 +0000 Subject: ASoC: rsnd: avoid duplicate free_irq() commit e0936c3471a8411a5df327641fa3ffe12a2fb07b upstream. commit 1f8754d4daea5f ("ASoC: rsnd: don't call free_irq() on Parent SSI") fixed Parent SSI duplicate free_irq(). But on Renesas Sound, not only Parent SSI but also Multi SSI have same issue. This patch avoid duplicate free_irq() if it was not pure SSI. Fixes: 1f8754d4daea5f ("ASoC: rsnd: don't call free_irq() on Parent SSI") Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown Signed-off-by: thongsyho Signed-off-by: Nhan Nguyen Signed-off-by: Greg Kroah-Hartman --- sound/soc/sh/rcar/ssi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 94739b7aca77..38aae96267c9 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -550,14 +550,14 @@ static int rsnd_ssi_dma_remove(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - struct rsnd_mod *ssi_parent_mod = rsnd_io_to_mod_ssip(io); + struct rsnd_mod *pure_ssi_mod = rsnd_io_to_mod_ssi(io); struct device *dev = rsnd_priv_to_dev(priv); int irq = ssi->info->irq; rsnd_dma_quit(io, rsnd_mod_to_dma(mod)); - /* Do nothing for SSI parent mod */ - if (ssi_parent_mod == mod) + /* Do nothing if non SSI (= SSI parent, multi SSI) mod */ + if (pure_ssi_mod != mod) return 0; /* PIO will request IRQ again */ -- cgit v1.2.3 From 5ff8af891df048c57462cabc42bb03a55a42196e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 19 Feb 2018 17:16:01 +0100 Subject: ALSA: seq: Fix regression by incorrect ioctl_mutex usages This is the revised backport of the upstream commit b3defb791b26ea0683a93a4f49c77ec45ec96f10 We had another backport (e.g. 623e5c8ae32b in 4.4.115), but it applies the new mutex also to the code paths that are invoked via faked kernel-to-kernel ioctls. As reported recently, this leads to a deadlock at suspend (or other scenarios triggering the kernel sequencer client). This patch addresses the issue by taking the mutex only in the code paths invoked by user-space, just like the original fix patch does. Reported-and-tested-by: Andres Bertens Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/core/seq/seq_clientmgr.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 7bb9fe7a2c8e..dacc62fe5a58 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -2196,7 +2196,6 @@ static int snd_seq_do_ioctl(struct snd_seq_client *client, unsigned int cmd, void __user *arg) { struct seq_ioctl_table *p; - int ret; switch (cmd) { case SNDRV_SEQ_IOCTL_PVERSION: @@ -2210,12 +2209,8 @@ static int snd_seq_do_ioctl(struct snd_seq_client *client, unsigned int cmd, if (! arg) return -EFAULT; for (p = ioctl_tables; p->cmd; p++) { - if (p->cmd == cmd) { - mutex_lock(&client->ioctl_mutex); - ret = p->func(client, arg); - mutex_unlock(&client->ioctl_mutex); - return ret; - } + if (p->cmd == cmd) + return p->func(client, arg); } pr_debug("ALSA: seq unknown ioctl() 0x%x (type='%c', number=0x%02x)\n", cmd, _IOC_TYPE(cmd), _IOC_NR(cmd)); @@ -2226,11 +2221,15 @@ static int snd_seq_do_ioctl(struct snd_seq_client *client, unsigned int cmd, static long snd_seq_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { struct snd_seq_client *client = file->private_data; + long ret; if (snd_BUG_ON(!client)) return -ENXIO; - return snd_seq_do_ioctl(client, cmd, (void __user *) arg); + mutex_lock(&client->ioctl_mutex); + ret = snd_seq_do_ioctl(client, cmd, (void __user *) arg); + mutex_unlock(&client->ioctl_mutex); + return ret; } #ifdef CONFIG_COMPAT -- cgit v1.2.3 From bc4c681fcaff86d73fe2dbda87be835d640dfa6f Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Mon, 29 Jan 2018 14:23:15 +0800 Subject: ALSA: hda - Fix headset mic detection problem for two Dell machines commit 3f2f7c553d077be6a30cb96b2976a2c940bf5335 upstream. One of them has the codec of alc256 and the other one has the codec of alc289. Cc: Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f14c1f288443..b6427ff9a211 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5975,6 +5975,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0xb7a60130}, {0x14, 0x90170110}, {0x21, 0x02211020}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x14, 0x90170110}, + {0x14, 0x01011020}, + {0x21, 0x0221101f}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC256_STANDARD_PINS), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, @@ -6031,6 +6036,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60120}, {0x14, 0x90170110}, {0x21, 0x0321101f}), + SND_HDA_PIN_QUIRK(0x10ec0289, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0xb7a60130}, + {0x14, 0x90170110}, + {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, {0x15, 0x04211040}, -- cgit v1.2.3 From 4c6e8dd5d2670c74669b1627eb18f6a4226993a3 Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin Date: Mon, 29 Jan 2018 06:37:55 +0100 Subject: ALSA: usb-audio: Fix UAC2 get_ctl request with a RANGE attribute commit 447cae58cecd69392b74a4a42cd0ab9cabd816af upstream. The layout of the UAC2 Control request and response varies depending on the request type. With the current implementation, only the Layout 2 Parameter Block (with the 2-byte sized RANGE attribute) is handled properly. For the Control requests with the 1-byte sized RANGE attribute (Bass Control, Mid Control, Tremble Control), the response is parsed incorrectly. This commit: * fixes the wLength field value in the request * fixes parsing the range values from the response Fixes: 23caaf19b11e ("ALSA: usb-mixer: Add support for Audio Class v2.0") Signed-off-by: Kirill Marinushkin Cc: Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/usb/mixer.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c9ae29068c7c..c5447ff078b3 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -343,17 +343,20 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[4 + 3 * sizeof(__u32)]; /* enough space for one range */ + /* enough space for one range */ + unsigned char buf[sizeof(__u16) + 3 * sizeof(__u32)]; unsigned char *val; - int idx = 0, ret, size; + int idx = 0, ret, val_size, size; __u8 bRequest; + val_size = uac2_ctl_value_size(cval->val_type); + if (request == UAC_GET_CUR) { bRequest = UAC2_CS_CUR; - size = uac2_ctl_value_size(cval->val_type); + size = val_size; } else { bRequest = UAC2_CS_RANGE; - size = sizeof(buf); + size = sizeof(__u16) + 3 * val_size; } memset(buf, 0, sizeof(buf)); @@ -386,16 +389,17 @@ error: val = buf + sizeof(__u16); break; case UAC_GET_MAX: - val = buf + sizeof(__u16) * 2; + val = buf + sizeof(__u16) + val_size; break; case UAC_GET_RES: - val = buf + sizeof(__u16) * 3; + val = buf + sizeof(__u16) + val_size * 2; break; default: return -EINVAL; } - *value_ret = convert_signed_value(cval, snd_usb_combine_bytes(val, sizeof(__u16))); + *value_ret = convert_signed_value(cval, + snd_usb_combine_bytes(val, val_size)); return 0; } -- cgit v1.2.3 From d84b8a33526b29d7e39ae2f6e8ce0e7d520fa5cd Mon Sep 17 00:00:00 2001 From: Jan-Marek Glogowski Date: Wed, 14 Feb 2018 11:29:15 +0100 Subject: ALSA: hda/realtek: PCI quirk for Fujitsu U7x7 commit fdcc968a3b290407bcba9d4c90e2fba6d8d928f1 upstream. These laptops have a combined jack to attach headsets, the U727 on the left, the U757 on the right, but a headsets microphone doesn't work. Using hdajacksensetest I found that pin 0x19 changed the present state when plugging the headset, in addition to 0x21, but didn't have the correct configuration (shown as "Not connected"). So this sets the configuration to the same values as the headphone pin 0x21 except for the device type microphone, which makes it work correctly. With the patch the configured pins for U727 are Pin 0x12 (Internal Mic, Mobile-In): present = No Pin 0x14 (Internal Speaker): present = No Pin 0x19 (Black Mic, Left side): present = No Pin 0x1d (Internal Aux): present = No Pin 0x21 (Black Headphone, Left side): present = No Signed-off-by: Jan-Marek Glogowski Cc: Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/patch_realtek.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b6427ff9a211..b302d056e5d3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3130,6 +3130,19 @@ static void alc269_fixup_pincfg_no_hp_to_lineout(struct hda_codec *codec, spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; } +static void alc269_fixup_pincfg_U7x7_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + unsigned int cfg_headphone = snd_hda_codec_get_pincfg(codec, 0x21); + unsigned int cfg_headset_mic = snd_hda_codec_get_pincfg(codec, 0x19); + + if (cfg_headphone && cfg_headset_mic == 0x411111f0) + snd_hda_codec_set_pincfg(codec, 0x19, + (cfg_headphone & ~AC_DEFCFG_DEVICE) | + (AC_JACK_MIC_IN << AC_DEFCFG_DEVICE_SHIFT)); +} + static void alc269_fixup_hweq(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4782,6 +4795,7 @@ enum { ALC269_FIXUP_LIFEBOOK_EXTMIC, ALC269_FIXUP_LIFEBOOK_HP_PIN, ALC269_FIXUP_LIFEBOOK_NO_HP_TO_LINEOUT, + ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC, ALC269_FIXUP_AMIC, ALC269_FIXUP_DMIC, ALC269VB_FIXUP_AMIC, @@ -4972,6 +4986,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_no_hp_to_lineout, }, + [ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_pincfg_U7x7_headset_mic, + }, [ALC269_FIXUP_AMIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5687,6 +5705,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x159f, "Lifebook E780", ALC269_FIXUP_LIFEBOOK_NO_HP_TO_LINEOUT), SND_PCI_QUIRK(0x10cf, 0x15dc, "Lifebook T731", ALC269_FIXUP_LIFEBOOK_HP_PIN), SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN), + SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_BXBT2807_MIC), -- cgit v1.2.3 From 5e5d1372ba7cfa0cf040a4e038e689f6f16e6470 Mon Sep 17 00:00:00 2001 From: Lassi Ylikojola Date: Fri, 9 Feb 2018 16:51:36 +0200 Subject: ALSA: usb-audio: add implicit fb quirk for Behringer UFX1204 commit 5e35dc0338d85ccebacf3f77eca1e5dea73155e8 upstream. Add quirk to ensure a sync endpoint is properly configured. This patch is a fix for same symptoms on Behringer UFX1204 as patch from Albertto Aquirre on Dec 8 2016 for Axe-Fx II. Signed-off-by: Lassi Ylikojola Cc: Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/usb/pcm.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 48afae053c56..8e8db4ddf365 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -343,6 +343,15 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; iface = usb_ifnum_to_if(dev, 2); + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; + case USB_ID(0x1397, 0x0002): + ep = 0x81; + iface = usb_ifnum_to_if(dev, 1); + if (!iface || iface->num_altsetting == 0) return -EINVAL; -- cgit v1.2.3 From b374197df2deb08fec55d48763711ea1df8efde7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Feb 2018 15:20:51 +0100 Subject: ALSA: seq: Fix racy pool initializations MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit commit d15d662e89fc667b90cd294b0eb45694e33144da upstream. ALSA sequencer core initializes the event pool on demand by invoking snd_seq_pool_init() when the first write happens and the pool is empty. Meanwhile user can reset the pool size manually via ioctl concurrently, and this may lead to UAF or out-of-bound accesses since the function tries to vmalloc / vfree the buffer. A simple fix is to just wrap the snd_seq_pool_init() call with the recently introduced client->ioctl_mutex; as the calls for snd_seq_pool_init() from other side are always protected with this mutex, we can avoid the race. Reported-by: 范龙飞 Cc: Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/core/seq/seq_clientmgr.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index dacc62fe5a58..167b943469ab 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1012,7 +1012,7 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf, { struct snd_seq_client *client = file->private_data; int written = 0, len; - int err = -EINVAL; + int err; struct snd_seq_event event; if (!(snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_OUTPUT)) @@ -1027,11 +1027,15 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf, /* allocate the pool now if the pool is not allocated yet */ if (client->pool->size > 0 && !snd_seq_write_pool_allocated(client)) { - if (snd_seq_pool_init(client->pool) < 0) + mutex_lock(&client->ioctl_mutex); + err = snd_seq_pool_init(client->pool); + mutex_unlock(&client->ioctl_mutex); + if (err < 0) return -ENOMEM; } /* only process whole events */ + err = -EINVAL; while (count >= sizeof(struct snd_seq_event)) { /* Read in the event header from the user */ len = sizeof(event); -- cgit v1.2.3 From 33180fe1d84a4c1da7124ba9b6f31c114ccac124 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 10 Jan 2018 17:34:45 +0100 Subject: ASoC: ux500: add MODULE_LICENSE tag commit 1783c9d7cb7bc3181b9271665959b87280d98d8e upstream. This adds MODULE_LICENSE/AUTHOR/DESCRIPTION tags to the ux500 platform drivers, to avoid these build warnings: WARNING: modpost: missing MODULE_LICENSE() in sound/soc/ux500/snd-soc-ux500-plat-dma.o WARNING: modpost: missing MODULE_LICENSE() in sound/soc/ux500/snd-soc-ux500-mach-mop500.o The company no longer exists, so the email addresses of the authors don't work any more, but I've added them anyway for consistency. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown Signed-off-by: Greg Kroah-Hartman --- sound/soc/ux500/mop500.c | 4 ++++ sound/soc/ux500/ux500_pcm.c | 5 +++++ 2 files changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index ba9fc099cf67..503aef8fcde2 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -164,3 +164,7 @@ static struct platform_driver snd_soc_mop500_driver = { }; module_platform_driver(snd_soc_mop500_driver); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("ASoC MOP500 board driver"); +MODULE_AUTHOR("Ola Lilja"); diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index f12c01dddc8d..d35ba7700f46 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -165,3 +165,8 @@ int ux500_pcm_unregister_platform(struct platform_device *pdev) return 0; } EXPORT_SYMBOL_GPL(ux500_pcm_unregister_platform); + +MODULE_AUTHOR("Ola Lilja"); +MODULE_AUTHOR("Roger Nilsson"); +MODULE_DESCRIPTION("ASoC UX500 driver"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From 253e3a668b273546530436a294b0107878883d25 Mon Sep 17 00:00:00 2001 From: Stefan Potyra Date: Wed, 6 Dec 2017 16:03:24 +0100 Subject: ASoC: rockchip: disable clock on error [ Upstream commit c7b92172a61b91936be985cb9bc499a4ebc6489b ] Disable the clocks in rk_spdif_probe when an error occurs after one of the clocks has been enabled previously. Found by Linux Driver Verification project (linuxtesting.org). Fixes: f874b80e1571 ASoC: rockchip: Add rockchip SPDIF transceiver driver Signed-off-by: Stefan Potyra Signed-off-by: Mark Brown Signed-off-by: Sasha Levin Signed-off-by: Greg Kroah-Hartman --- sound/soc/rockchip/rockchip_spdif.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index 5a806da89f42..dae80be6ca71 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -316,26 +316,30 @@ static int rk_spdif_probe(struct platform_device *pdev) spdif->mclk = devm_clk_get(&pdev->dev, "mclk"); if (IS_ERR(spdif->mclk)) { dev_err(&pdev->dev, "Can't retrieve rk_spdif master clock\n"); - return PTR_ERR(spdif->mclk); + ret = PTR_ERR(spdif->mclk); + goto err_disable_hclk; } ret = clk_prepare_enable(spdif->mclk); if (ret) { dev_err(spdif->dev, "clock enable failed %d\n", ret); - return ret; + goto err_disable_clocks; } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(regs)) - return PTR_ERR(regs); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); + goto err_disable_clocks; + } spdif->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "hclk", regs, &rk_spdif_regmap_config); if (IS_ERR(spdif->regmap)) { dev_err(&pdev->dev, "Failed to initialise managed register map\n"); - return PTR_ERR(spdif->regmap); + ret = PTR_ERR(spdif->regmap); + goto err_disable_clocks; } spdif->playback_dma_data.addr = res->start + SPDIF_SMPDR; @@ -367,6 +371,10 @@ static int rk_spdif_probe(struct platform_device *pdev) err_pm_runtime: pm_runtime_disable(&pdev->dev); +err_disable_clocks: + clk_disable_unprepare(spdif->mclk); +err_disable_hclk: + clk_disable_unprepare(spdif->hclk); return ret; } -- cgit v1.2.3 From 3f0754639bac017520524034cf75d11c889c0b2f Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 11 Jan 2017 14:39:44 +0100 Subject: ALSA: hda/ca0132 - fix possible NULL pointer use commit 46a049dae771b95e77ac6c823330f4a60f600236 upstream. gcc-7 caught what it considers a NULL pointer dereference: sound/pci/hda/patch_ca0132.c: In function 'dspio_scp.constprop': sound/pci/hda/patch_ca0132.c:1487:4: error: argument 1 null where non-null expected [-Werror=nonnull] This is plausible from looking at the function, as we compare 'reply' to NULL earlier in it. I have not tried to analyze if there are constraints that make it impossible to hit the bug, but adding another NULL check in the end kills the warning and makes the function more robust. Signed-off-by: Arnd Bergmann Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/patch_ca0132.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index c146d0de53d8..29e1ce2263bc 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1482,6 +1482,9 @@ static int dspio_scp(struct hda_codec *codec, } else if (ret_size != reply_data_size) { codec_dbg(codec, "RetLen and HdrLen .NE.\n"); return -EINVAL; + } else if (!reply) { + codec_dbg(codec, "NULL reply\n"); + return -EINVAL; } else { *reply_len = ret_size*sizeof(unsigned int); memcpy(reply, scp_reply.data, *reply_len); -- cgit v1.2.3 From c7284a1114f07e866d9c7fe27d6f491262b8d0d0 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 13 Jan 2016 23:14:54 +0100 Subject: ASoC: mediatek: add i2c dependency commit ec3995da27e782cc407ce48101c98c19c9ce738d upstream. The newly added mediatek drivers for mt8173 select codes that depend on I2C, which cuases a build failure if I2C is disabled: warning: (SND_SOC_ADAU1761_I2C && SND_SOC_ADAU1781_I2C && SND_SOC_ADAU1977_I2C && SND_SOC_RT5677 && EXTCON_MAX14577 && EXTCON_MAX77693 && EXTCON_MAX77843 && BMC150_ACCEL_I2C && BMG160_I2C) selects REGMAP_I2C which has unmet direct dependencies (I2C) codecs/rt5645.c:3854:1: warning: data definition has no type or storage class codecs/rt5645.c:3854:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] codecs/rt5677.c:5270:1: warning: data definition has no type or storage class 77_i2c_driver); codecs/rt5677.c:5270:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] This adds an explicit dependency. Signed-off-by: Arnd Bergmann Acked-by: Koro Chen Signed-off-by: Mark Brown Signed-off-by: Greg Kroah-Hartman --- sound/soc/mediatek/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 15c04e2eae34..976967675387 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -9,7 +9,7 @@ config SND_SOC_MEDIATEK config SND_SOC_MT8173_MAX98090 tristate "ASoC Audio driver for MT8173 with MAX98090 codec" - depends on SND_SOC_MEDIATEK + depends on SND_SOC_MEDIATEK && I2C select SND_SOC_MAX98090 help This adds ASoC driver for Mediatek MT8173 boards @@ -19,7 +19,7 @@ config SND_SOC_MT8173_MAX98090 config SND_SOC_MT8173_RT5650_RT5676 tristate "ASoC Audio driver for MT8173 with RT5650 RT5676 codecs" - depends on SND_SOC_MEDIATEK + depends on SND_SOC_MEDIATEK && I2C select SND_SOC_RT5645 select SND_SOC_RT5677 help -- cgit v1.2.3 From 8caadd7bf286336f5542f2302b960419190b60fe Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 2 Mar 2016 16:59:06 +0100 Subject: ASoC: rockchip: use __maybe_unused to hide st_irq_syscfg_resume commit d8fc2198aab117a4bc16ee305caef19c4c7e7f5c upstream. The rockchip spdif driver uses SIMPLE_DEV_PM_OPS to conditionally set its power management functions, but we get a warning about rk_spdif_runtime_resume being unused when CONFIG_PM is not set: sound/soc/rockchip/rockchip_spdif.c:67:12: error: 'rk_spdif_runtime_resume' defined but not used [-Werror=unused-function] This adds a __maybe_unused annotation so the compiler knows it can silently drop it instead of warning. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown Signed-off-by: Greg Kroah-Hartman --- sound/soc/rockchip/rockchip_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index dae80be6ca71..5e2eb4cc5cf1 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -54,7 +54,7 @@ static const struct of_device_id rk_spdif_match[] = { }; MODULE_DEVICE_TABLE(of, rk_spdif_match); -static int rk_spdif_runtime_suspend(struct device *dev) +static int __maybe_unused rk_spdif_runtime_suspend(struct device *dev) { struct rk_spdif_dev *spdif = dev_get_drvdata(dev); @@ -64,7 +64,7 @@ static int rk_spdif_runtime_suspend(struct device *dev) return 0; } -static int rk_spdif_runtime_resume(struct device *dev) +static int __maybe_unused rk_spdif_runtime_resume(struct device *dev) { struct rk_spdif_dev *spdif = dev_get_drvdata(dev); int ret; -- cgit v1.2.3 From bd84055a3748ca0b11c466a252d46af6cf8e765f Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 20 Feb 2018 12:54:57 +0100 Subject: ASoC: Intel: Kconfig: fix build when ACPI is not enabled commit 3493d4a86457c7de9f1e602b4267c9b0f9ec1c9f upstream. Randy reported following error when ACPI is not enabled: warning: (SND_SOC_INTEL_BYTCR_RT5640_MACH && SND_SOC_INTEL_BYTCR_RT5651_MACH && SND_SOC_INTEL_CHT_BSW_RT5672_MACH && SND_SOC_INTEL_CHT_BSW_RT5645_MACH && SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) selects SND_SST_IPC_ACPI +which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && ACPI) causing these build errors: In file included from ../sound/soc/intel/atom/sst/sst_acpi.c:40:0: ../include/acpi/acpi_bus.h:65:20: error: conflicting types for 'acpi_evaluate_dsm' union acpi_object *acpi_evaluate_dsm(acpi_handle handle, const u8 *uuid, In file included from ../sound/soc/intel/atom/sst/sst_acpi.c:31:0: ../include/linux/acpi.h:676:34: note: previous definition of 'acpi_evaluate_dsm' was here static inline union acpi_object *acpi_evaluate_dsm(acpi_handle handle, CONFIG_SND_SST_IPC_ACPI was already dependent upon ACPI, but that was not solving it. So move the depends up to machine drivers and remove from CONFIG_SND_SST_IPC_ACPI. Reported-by: Randy Dunlap Signed-off-by: Vinod Koul Signed-off-by: Mark Brown [arnd: rebased to PATCH kernel] Signed-off-by: Arnd Bergmann Signed-off-by: Greg Kroah-Hartman --- sound/soc/intel/Kconfig | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index d430ef5a4f38..79c29330c56a 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -24,7 +24,6 @@ config SND_SST_IPC_PCI config SND_SST_IPC_ACPI tristate select SND_SST_IPC - depends on ACPI config SND_SOC_INTEL_SST tristate @@ -91,7 +90,7 @@ config SND_SOC_INTEL_BROADWELL_MACH config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "ASoC Audio DSP Support for MID BYT Platform" - depends on X86 && I2C + depends on X86 && I2C && ACPI select SND_SOC_RT5640 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -103,7 +102,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH config SND_SOC_INTEL_CHT_BSW_RT5672_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" - depends on X86_INTEL_LPSS && I2C + depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_RT5670 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -115,7 +114,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645/5650 codec" - depends on X86_INTEL_LPSS && I2C + depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_RT5645 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI -- cgit v1.2.3 From b43e8110ca018ecb92e95ff9048df7ab71327558 Mon Sep 17 00:00:00 2001 From: Erik Veijola Date: Fri, 23 Feb 2018 14:06:52 +0200 Subject: ALSA: usb-audio: Add a quirck for B&W PX headphones commit 240a8af929c7c57dcde28682725b29cf8474e8e5 upstream. The capture interface doesn't work and the playback interface only supports 48 kHz sampling rate even though it advertises more rates. Signed-off-by: Erik Veijola Cc: Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/usb/quirks-table.h | 47 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 47 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8a59d4782a0f..69bf5cf1e91e 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3277,4 +3277,51 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +{ + /* + * Bower's & Wilkins PX headphones only support the 48 kHz sample rate + * even though it advertises more. The capture interface doesn't work + * even on windows. + */ + USB_DEVICE(0x19b5, 0x0021), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + /* Capture */ + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* Playback */ + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_FILL_MAX | + UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x03, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + } + } +}, + #undef USB_DEVICE_VENDOR_SPEC -- cgit v1.2.3 From 4fdc12f6d818b50b459be27692072e5d6447404c Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 22 Feb 2018 14:20:35 +0100 Subject: ALSA: hda: Add a power_save blacklist commit 1ba8f9d308174e647b864c36209b4d7934d99888 upstream. On some boards setting power_save to a non 0 value leads to clicking / popping sounds when ever we enter/leave powersaving mode. Ideally we would figure out how to avoid these sounds, but that is not always feasible. This commit adds a blacklist for devices where powersaving is known to cause problems and disables it on these devices. Note I tried to put this blacklist in userspace first: https://github.com/systemd/systemd/pull/8128 But the systemd maintainers rightfully pointed out that it would be impossible to then later remove entries once we actually find a way to make power-saving work on listed boards without issues. Having this list in the kernel will allow removal of the blacklist entry in the same commit which fixes the clicks / plops. The blacklist only applies to the default power_save module-option value, if a user explicitly sets the module-option then the blacklist is not used. [ added an ifdef CONFIG_PM for the build error -- tiwai] BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=198611 Cc: stable@vger.kernel.org Signed-off-by: Hans de Goede Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/hda_intel.c | 38 ++++++++++++++++++++++++++++++++++++-- 1 file changed, 36 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 20512fe32a97..e2212830df0c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -179,7 +179,7 @@ static const struct kernel_param_ops param_ops_xint = { }; #define param_check_xint param_check_int -static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; +static int power_save = -1; module_param(power_save, xint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " "(in second, 0 = disable)."); @@ -2055,6 +2055,24 @@ out_free: return err; } +#ifdef CONFIG_PM +/* On some boards setting power_save to a non 0 value leads to clicking / + * popping sounds when ever we enter/leave powersaving mode. Ideally we would + * figure out how to avoid these sounds, but that is not always feasible. + * So we keep a list of devices where we disable powersaving as its known + * to causes problems on these devices. + */ +static struct snd_pci_quirk power_save_blacklist[] = { + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1849, 0x0c0c, "Asrock B85M-ITX", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), + /* https://bugzilla.kernel.org/show_bug.cgi?id=198611 */ + SND_PCI_QUIRK(0x17aa, 0x2227, "Lenovo X1 Carbon 3rd Gen", 0), + {} +}; +#endif /* CONFIG_PM */ + /* number of codec slots for each chipset: 0 = default slots (i.e. 4) */ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] = { [AZX_DRIVER_NVIDIA] = 8, @@ -2067,6 +2085,7 @@ static int azx_probe_continue(struct azx *chip) struct hdac_bus *bus = azx_bus(chip); struct pci_dev *pci = chip->pci; int dev = chip->dev_index; + int val; int err; hda->probe_continued = 1; @@ -2142,7 +2161,22 @@ static int azx_probe_continue(struct azx *chip) chip->running = 1; azx_add_card_list(chip); - snd_hda_set_power_save(&chip->bus, power_save * 1000); + + val = power_save; +#ifdef CONFIG_PM + if (val == -1) { + const struct snd_pci_quirk *q; + + val = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; + q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist); + if (q && val) { + dev_info(chip->card->dev, "device %04x:%04x is on the power_save blacklist, forcing power_save to 0\n", + q->subvendor, q->subdevice); + val = 0; + } + } +#endif /* CONFIG_PM */ + snd_hda_set_power_save(&chip->bus, val * 1000); if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) pm_runtime_put_noidle(&pci->dev); -- cgit v1.2.3