From 537e48136295c5860a92138c5ea3959b9542868b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 14:25:16 +0100 Subject: ALSA: hdspm: Fix wrong boolean ctl value accesses snd-hdspm driver accesses enum item values (int) instead of boolean values (long) wrongly for some ctl elements. This patch fixes them. Cc: Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 8bc8016c173d..b047b1ba48fd 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2261,7 +2261,7 @@ static int snd_hdspm_put_system_sample_rate(struct snd_kcontrol *kcontrol, { struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); - hdspm_set_dds_value(hdspm, ucontrol->value.enumerated.item[0]); + hdspm_set_dds_value(hdspm, ucontrol->value.integer.value[0]); return 0; } @@ -4449,7 +4449,7 @@ static int snd_hdspm_get_tco_word_term(struct snd_kcontrol *kcontrol, { struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = hdspm->tco->term; + ucontrol->value.integer.value[0] = hdspm->tco->term; return 0; } @@ -4460,8 +4460,8 @@ static int snd_hdspm_put_tco_word_term(struct snd_kcontrol *kcontrol, { struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); - if (hdspm->tco->term != ucontrol->value.enumerated.item[0]) { - hdspm->tco->term = ucontrol->value.enumerated.item[0]; + if (hdspm->tco->term != ucontrol->value.integer.value[0]) { + hdspm->tco->term = ucontrol->value.integer.value[0]; hdspm_tco_write(hdspm); -- cgit v1.2.3-58-ga151 From c1099c3294c2344110085a38c50e478a5992b368 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 14:32:42 +0100 Subject: ALSA: hdspm: Fix zero-division HDSPM driver contains a code issuing zero-division potentially in system sample rate ctl code. This patch fixes it by not processing a zero or invalid rate value as a divisor, as well as excluding the invalid value to be passed via the given ctl element. Cc: Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index b047b1ba48fd..a4a999a0317e 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1601,6 +1601,9 @@ static void hdspm_set_dds_value(struct hdspm *hdspm, int rate) { u64 n; + if (snd_BUG_ON(rate <= 0)) + return; + if (rate >= 112000) rate /= 4; else if (rate >= 56000) @@ -2215,6 +2218,8 @@ static int hdspm_get_system_sample_rate(struct hdspm *hdspm) } else { /* slave mode, return external sample rate */ rate = hdspm_external_sample_rate(hdspm); + if (!rate) + rate = hdspm->system_sample_rate; } } @@ -2260,7 +2265,10 @@ static int snd_hdspm_put_system_sample_rate(struct snd_kcontrol *kcontrol, ucontrol) { struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); + int rate = ucontrol->value.integer.value[0]; + if (rate < 27000 || rate > 207000) + return -EINVAL; hdspm_set_dds_value(hdspm, ucontrol->value.integer.value[0]); return 0; } -- cgit v1.2.3-58-ga151 From eab3c4db193f5fcccf70e884de9a922ca2c63d80 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Feb 2016 14:26:43 +0100 Subject: ALSA: hdsp: Fix wrong boolean ctl value accesses snd-hdsp driver accesses enum item values (int) instead of boolean values (long) wrongly for some ctl elements. This patch fixes them. Cc: Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 2875b4f6d8c9..7c8941b8b2de 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -2879,7 +2879,7 @@ static int snd_hdsp_get_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = hdsp_dds_offset(hdsp); + ucontrol->value.integer.value[0] = hdsp_dds_offset(hdsp); return 0; } @@ -2891,7 +2891,7 @@ static int snd_hdsp_put_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; - val = ucontrol->value.enumerated.item[0]; + val = ucontrol->value.integer.value[0]; spin_lock_irq(&hdsp->lock); if (val != hdsp_dds_offset(hdsp)) change = (hdsp_set_dds_offset(hdsp, val) == 0) ? 1 : 0; -- cgit v1.2.3-58-ga151 From 17e2df4613be57d0fab68df749f6b8114e453152 Mon Sep 17 00:00:00 2001 From: Dennis Kadioglu Date: Tue, 1 Mar 2016 14:23:29 +0100 Subject: ALSA: usb-audio: Add a quirk for Plantronics DA45 Plantronics DA45 does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x4" and "cannot get freq at ep 0x84". This patch adds the USB ID of the DA45 to quirks.c and avoids those error messages. Signed-off-by: Dennis Kadioglu Cc: Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4f6ce1cac8e2..c458d60d5030 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1124,6 +1124,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ + case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ -- cgit v1.2.3-58-ga151 From 197b958c1e76a575d77038cc98b4bebc2134279f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 1 Mar 2016 18:30:18 +0100 Subject: ALSA: seq: oss: Don't drain at closing a client The OSS sequencer client tries to drain the pending events at releasing. Unfortunately, as spotted by syzkaller fuzzer, this may lead to an unkillable process state when the event has been queued at the far future. Since the process being released can't be signaled any longer, it remains and waits for the echo-back event in that far future. Back to history, the draining feature was implemented at the time we misinterpreted POSIX definition for blocking file operation. Actually, such a behavior is superfluous at release, and we should just release the device as is instead of keeping it up forever. This patch just removes the draining call that may block the release for too long time unexpectedly. BugLink: http://lkml.kernel.org/r/CACT4Y+Y4kD-aBGj37rf-xBw9bH3GMU6P+MYg4W1e-s-paVD2pg@mail.gmail.com Reported-by: Dmitry Vyukov Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss.c | 2 -- sound/core/seq/oss/seq_oss_device.h | 1 - sound/core/seq/oss/seq_oss_init.c | 16 ---------------- 3 files changed, 19 deletions(-) diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index 8db156b207f1..8cdf489df80e 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -149,8 +149,6 @@ odev_release(struct inode *inode, struct file *file) if ((dp = file->private_data) == NULL) return 0; - snd_seq_oss_drain_write(dp); - mutex_lock(®ister_mutex); snd_seq_oss_release(dp); mutex_unlock(®ister_mutex); diff --git a/sound/core/seq/oss/seq_oss_device.h b/sound/core/seq/oss/seq_oss_device.h index b43924325249..d7b4d016b547 100644 --- a/sound/core/seq/oss/seq_oss_device.h +++ b/sound/core/seq/oss/seq_oss_device.h @@ -127,7 +127,6 @@ int snd_seq_oss_write(struct seq_oss_devinfo *dp, const char __user *buf, int co unsigned int snd_seq_oss_poll(struct seq_oss_devinfo *dp, struct file *file, poll_table * wait); void snd_seq_oss_reset(struct seq_oss_devinfo *dp); -void snd_seq_oss_drain_write(struct seq_oss_devinfo *dp); /* */ void snd_seq_oss_process_queue(struct seq_oss_devinfo *dp, abstime_t time); diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index 6779e82b46dd..92c96a95a903 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -435,22 +435,6 @@ snd_seq_oss_release(struct seq_oss_devinfo *dp) } -/* - * Wait until the queue is empty (if we don't have nonblock) - */ -void -snd_seq_oss_drain_write(struct seq_oss_devinfo *dp) -{ - if (! dp->timer->running) - return; - if (is_write_mode(dp->file_mode) && !is_nonblock_mode(dp->file_mode) && - dp->writeq) { - while (snd_seq_oss_writeq_sync(dp->writeq)) - ; - } -} - - /* * reset sequencer devices */ -- cgit v1.2.3-58-ga151 From 02322ac9dee9aff8d8862e8d6660ebe102f492ea Mon Sep 17 00:00:00 2001 From: Simon South Date: Wed, 2 Mar 2016 23:10:44 -0500 Subject: ALSA: hda - Fix mic issues on Acer Aspire E1-472 This patch applies the microphone-related fix created for the Acer Aspire E1-572 to the E1-472 as well, as it uses the same Realtek ALC282 CODEC and demonstrates the same issues. This patch allows an external, headset microphone to be used and limits the gain on the (quite noisy) internal microphone. Signed-off-by: Simon South Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1f357cd72d9c..93d2156b6241 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5412,6 +5412,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), + SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), -- cgit v1.2.3-58-ga151 From ec75a940b1037e877efd9a5a9e94eab1e464f73b Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Fri, 4 Mar 2016 14:33:06 +0800 Subject: ALSA: hda - hdmi add wmb barrier for audio component To make sure audio_ptr is set before intel_audio_codec_enable() or intel_audio_codec_disable() calling pin_eld_notify(), this patch adds wmb barrier to prevent optimizing. Signed-off-by: Libin Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8ee78dbd4c60..6858e88c7326 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2480,6 +2480,11 @@ static int patch_generic_hdmi(struct hda_codec *codec) if (codec_has_acomp(codec)) { codec->depop_delay = 0; spec->i915_audio_ops.audio_ptr = codec; + /* intel_audio_codec_enable() or intel_audio_codec_disable() + * will call pin_eld_notify with using audio_ptr pointer + * We need make sure audio_ptr is really setup + */ + wmb(); spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; snd_hdac_i915_register_notifier(&spec->i915_audio_ops); } -- cgit v1.2.3-58-ga151 From 790b415c98de62602810b0eedce26f0f9d6ddd78 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Fri, 4 Mar 2016 14:33:43 +0800 Subject: ALSA: hda - hdmi defer to register acomp eld notifier Defer to register acomp eld notifier until hdmi audio driver is fully ready. After registering eld notifier, gfx driver can use this callback function to notify audio driver the monitor connection event. However this action may happen when audio driver is adding the pins or doing other initialization. This is not always safe, however. For example, using per_pin->lock before the lock is initialized. Let's register the eld notifier after the initialization is done. Signed-off-by: Libin Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 6858e88c7326..bcbc4ee10130 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2477,18 +2477,6 @@ static int patch_generic_hdmi(struct hda_codec *codec) is_broxton(codec)) codec->core.link_power_control = 1; - if (codec_has_acomp(codec)) { - codec->depop_delay = 0; - spec->i915_audio_ops.audio_ptr = codec; - /* intel_audio_codec_enable() or intel_audio_codec_disable() - * will call pin_eld_notify with using audio_ptr pointer - * We need make sure audio_ptr is really setup - */ - wmb(); - spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; - snd_hdac_i915_register_notifier(&spec->i915_audio_ops); - } - if (hdmi_parse_codec(codec) < 0) { if (spec->i915_bound) snd_hdac_i915_exit(&codec->bus->core); @@ -2510,6 +2498,18 @@ static int patch_generic_hdmi(struct hda_codec *codec) init_channel_allocations(); + if (codec_has_acomp(codec)) { + codec->depop_delay = 0; + spec->i915_audio_ops.audio_ptr = codec; + /* intel_audio_codec_enable() or intel_audio_codec_disable() + * will call pin_eld_notify with using audio_ptr pointer + * We need make sure audio_ptr is really setup + */ + wmb(); + spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; + snd_hdac_i915_register_notifier(&spec->i915_audio_ops); + } + return 0; } -- cgit v1.2.3-58-ga151