From 8eb22214b7cb0c0a28be6caf3b81201629d8ea7c Mon Sep 17 00:00:00 2001 From: Maruthi Srinivas Bayyavarapu Date: Thu, 31 Mar 2016 18:10:03 +0530 Subject: ALSA: hda: add AMD Polaris-10/11 AZ PCI IDs with proper driver caps MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This commit fixes garbled audio on Polaris-10/11 variants Signed-off-by: Maruthi Bayyavarapu Reviewed-by: Alex Deucher Acked-by: Christian König Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2624cfe98884..b680be0e937d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2361,6 +2361,10 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaae8), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaae0), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaaf0), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA }, /* VIA GFX VT7122/VX900 */ -- cgit v1.2.3 From 836b34a935abc91e13e63053d0a83b24dfb5ea78 Mon Sep 17 00:00:00 2001 From: Vladis Dronov Date: Thu, 31 Mar 2016 12:05:43 -0400 Subject: ALSA: usb-audio: Fix double-free in error paths after snd_usb_add_audio_stream() call create_fixed_stream_quirk(), snd_usb_parse_audio_interface() and create_uaxx_quirk() functions allocate the audioformat object by themselves and free it upon error before returning. However, once the object is linked to a stream, it's freed again in snd_usb_audio_pcm_free(), thus it'll be double-freed, eventually resulting in a memory corruption. This patch fixes these failures in the error paths by unlinking the audioformat object before freeing it. Based on a patch by Takashi Iwai [Note for stable backports: this patch requires the commit 902eb7fd1e4a ('ALSA: usb-audio: Minor code cleanup in create_fixed_stream_quirk()')] Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1283358 Reported-by: Ralf Spenneberg Cc: # see the note above Signed-off-by: Vladis Dronov Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 4 ++++ sound/usb/stream.c | 6 +++++- 2 files changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index fb62bce2435c..6178bb5d0731 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -150,6 +150,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, usb_audio_err(chip, "cannot memdup\n"); return -ENOMEM; } + INIT_LIST_HEAD(&fp->list); if (fp->nr_rates > MAX_NR_RATES) { kfree(fp); return -EINVAL; @@ -193,6 +194,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return 0; error: + list_del(&fp->list); /* unlink for avoiding double-free */ kfree(fp); kfree(rate_table); return err; @@ -469,6 +471,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; fp->datainterval = 0; fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + INIT_LIST_HEAD(&fp->list); switch (fp->maxpacksize) { case 0x120: @@ -492,6 +495,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; err = snd_usb_add_audio_stream(chip, stream, fp); if (err < 0) { + list_del(&fp->list); /* unlink for avoiding double-free */ kfree(fp); return err; } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index c4dc577ab1bd..8e9548bc1f1a 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -314,7 +314,9 @@ static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits, /* * add this endpoint to the chip instance. * if a stream with the same endpoint already exists, append to it. - * if not, create a new pcm stream. + * if not, create a new pcm stream. note, fp is added to the substream + * fmt_list and will be freed on the chip instance release. do not free + * fp or do remove it from the substream fmt_list to avoid double-free. */ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, int stream, @@ -675,6 +677,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) * (fp->maxpacksize & 0x7ff); fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no); fp->clock = clock; + INIT_LIST_HEAD(&fp->list); /* some quirks for attributes here */ @@ -723,6 +726,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) dev_dbg(&dev->dev, "%u:%d: add audio endpoint %#x\n", iface_no, altno, fp->endpoint); err = snd_usb_add_audio_stream(chip, stream, fp); if (err < 0) { + list_del(&fp->list); /* unlink for avoiding double-free */ kfree(fp->rate_table); kfree(fp->chmap); kfree(fp); -- cgit v1.2.3 From e549d190f7b5f94e9ab36bd965028112914d010d Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Fri, 1 Apr 2016 11:00:15 +0800 Subject: ALSA: hda - fix front mic problem for a HP desktop The front mic jack (pink color) can't detect any plug or unplug. After applying this fix, both detecting function and recording function work well. BugLink: https://bugs.launchpad.net/bugs/1564712 Cc: stable@vger.kernel.org Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- 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 73978c79981f..fefe83f2beab 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4759,6 +4759,7 @@ enum { ALC255_FIXUP_DELL_SPK_NOISE, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC280_FIXUP_HP_HEADSET_MIC, + ALC221_FIXUP_HP_FRONT_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -5401,6 +5402,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC, }, + [ALC221_FIXUP_HP_FRONT_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x02a19020 }, /* Front Mic */ + { } + }, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5506,6 +5514,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), + SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- cgit v1.2.3 From 4a07083ed613644c96c34a7dd2853dc5d7c70902 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 1 Apr 2016 12:28:16 +0200 Subject: ALSA: timer: Use mod_timer() for rearming the system timer ALSA system timer backend stops the timer via del_timer() without sync and leaves del_timer_sync() at the close instead. This is because of the restriction by the design of ALSA timer: namely, the stop callback may be called from the timer handler, and calling the sync shall lead to a hangup. However, this also triggers a kernel BUG() when the timer is rearmed immediately after stopping without sync: kernel BUG at kernel/time/timer.c:966! Call Trace: [] snd_timer_s_start+0x13e/0x1a0 [] snd_timer_interrupt+0x504/0xec0 [] ? debug_check_no_locks_freed+0x290/0x290 [] snd_timer_s_function+0xb4/0x120 [] call_timer_fn+0x162/0x520 [] ? call_timer_fn+0xcd/0x520 [] ? snd_timer_interrupt+0xec0/0xec0 .... It's the place where add_timer() checks the pending timer. It's clear that this may happen after the immediate restart without sync in our cases. So, the workaround here is just to use mod_timer() instead of add_timer(). This looks like a band-aid fix, but it's a right move, as snd_timer_interrupt() takes care of the continuous rearm of timer. Reported-by: Jiri Slaby Cc: Signed-off-by: Takashi Iwai --- sound/core/timer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index ea4d999113ef..6469bedda2f3 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1019,8 +1019,8 @@ static int snd_timer_s_start(struct snd_timer * timer) njiff += timer->sticks - priv->correction; priv->correction = 0; } - priv->last_expires = priv->tlist.expires = njiff; - add_timer(&priv->tlist); + priv->last_expires = njiff; + mod_timer(&priv->tlist, njiff); return 0; } -- cgit v1.2.3 From bfa5fb14fb9e698ae2d9429a82ef0ab67a17df37 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Mar 2016 15:03:06 +0200 Subject: ALSA: hda - Bind with i915 only when Intel graphics is present On Skylake and onwards, the HD-audio controller driver needs to bind with i915 for having the control of power well audio domain before actually probing the codec. This leads to the load of i915 driver from the audio driver side. But, there are systems that have no Intel graphics but Nvidia or AMD GPU, although they still use HD-audio bus for the onboard audio codecs. On these, loading the i915 driver is nothing but a useless memory and CPU consumption. A simple way to avoid it is just to look for the Intel graphics PCI entry beforehand, and try to bind with i915 only when such an entry is found. Currently, it assumes the PCI display class. If another class appears, this needs to be extended (although it's very unlikely). Signed-off-by: Takashi Iwai --- sound/hda/hdac_i915.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index fb96aead8257..54babe1c0b16 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -267,6 +267,18 @@ int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops } EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier); +/* check whether intel graphics is present */ +static bool i915_gfx_present(void) +{ + static struct pci_device_id ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), + .class = PCI_BASE_CLASS_DISPLAY << 16, + .class_mask = 0xff << 16 }, + {} + }; + return pci_dev_present(ids); +} + /** * snd_hdac_i915_init - Initialize i915 audio component * @bus: HDA core bus @@ -286,6 +298,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus) struct i915_audio_component *acomp; int ret; + if (!i915_gfx_present()) + return -ENODEV; + acomp = kzalloc(sizeof(*acomp), GFP_KERNEL); if (!acomp) return -ENOMEM; -- cgit v1.2.3 From f03b24a851d32ca85dacab01785b24a7ee717d37 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Apr 2016 11:47:50 +0200 Subject: ALSA: usb-audio: Add a sample rate quirk for Phoenix Audio TMX320 Phoenix Audio TMX320 gives the similar error when the sample rate is asked: usb 2-1.3: 2:1: cannot get freq at ep 0x85 usb 2-1.3: 1:1: cannot get freq at ep 0x2 .... Add the corresponding USB-device ID (1de7:0014) to snd_usb_get_sample_rate_quirk() list. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221 Cc: Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6178bb5d0731..24c7c2311b47 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1137,6 +1137,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) 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(0x1de7, 0x0014): /* Phoenix Audio TMX320 */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ return true; } -- cgit v1.2.3