From a1f3f1ca66bd12c339b17a0c2ef93a093f90a277 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 8 Mar 2015 18:29:50 +0100 Subject: [PATCH 1/9] ALSA: hda - Fix regression of HD-audio controller fallback modes The commit [63e51fd708f5: ALSA: hda - Don't take unresponsive D3 transition too serious] introduced a conditional fallback behavior to the HD-audio controller depending on the flag set. However, it introduced a silly bug, too, that the flag was evaluated in a reverse way. This resulted in a regression of HD-audio controller driver where it can't go to the fallback mode at communication errors. Unfortunately (or fortunately?) this didn't come up until recently because the affected code path is an error handling that happens only on an unstable hardware chip. Most of recent chips work stably, thus they didn't hit this problem. Now, we've got a regression report with a VIA chip, and this seems indeed requiring the fallback to the polling mode, and finally the bug was revealed. The fix is a oneliner to remove the wrong logical NOT in the check. (Lesson learned - be careful about double negation.) The bug should be backported to stable, but the patch won't be applicable to 3.13 or earlier because of the code splits. The stable fix patches for earlier kernels will be posted later manually. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=94021 Fixes: 63e51fd708f5 ('ALSA: hda - Don't take unresponsive D3 transition too serious') Cc: # v3.14+ Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index a2ce773bdc62..17c2637d842c 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1164,7 +1164,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } } - if (!bus->no_response_fallback) + if (bus->no_response_fallback) return -1; if (!chip->polling_mode && chip->poll_count < 2) { From 5b1274efe2a24eb5a85a00cc48c334b1cdfc75aa Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 10 Mar 2015 21:58:48 +0900 Subject: [PATCH 2/9] Revert "ALSA: dice: fix wrong offsets for Dice interface" This reverts commit 8cdebf71098c07168ef6335e2f1f35d85dbe3049. The reverted commit breaks out-stream functionality of Dice driver. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/dice/dice-interface.h | 18 +++++++++--------- sound/firewire/dice/dice-proc.c | 4 ++-- 2 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h index de7602bd69b5..27b044f84c81 100644 --- a/sound/firewire/dice/dice-interface.h +++ b/sound/firewire/dice/dice-interface.h @@ -298,24 +298,24 @@ */ #define RX_ISOCHRONOUS 0x008 +/* + * Index of first quadlet to be interpreted; read/write. If > 0, that many + * quadlets at the beginning of each data block will be ignored, and all the + * audio and MIDI quadlets will follow. + */ +#define RX_SEQ_START 0x00c + /* * The number of audio channels; read-only. There will be one quadlet per * channel. */ -#define RX_NUMBER_AUDIO 0x00c +#define RX_NUMBER_AUDIO 0x010 /* * The number of MIDI ports, 0-8; read-only. If > 0, there will be one * additional quadlet in each data block, following the audio quadlets. */ -#define RX_NUMBER_MIDI 0x010 - -/* - * Index of first quadlet to be interpreted; read/write. If > 0, that many - * quadlets at the beginning of each data block will be ignored, and all the - * audio and MIDI quadlets will follow. - */ -#define RX_SEQ_START 0x014 +#define RX_NUMBER_MIDI 0x014 /* * Names of all audio channels; read-only. Quadlets are byte-swapped. Names diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c index ecfe20fd4de5..f5c1d1bced59 100644 --- a/sound/firewire/dice/dice-proc.c +++ b/sound/firewire/dice/dice-proc.c @@ -99,9 +99,9 @@ static void dice_proc_read(struct snd_info_entry *entry, } tx; struct { u32 iso; + u32 seq_start; u32 number_audio; u32 number_midi; - u32 seq_start; char names[RX_NAMES_SIZE]; u32 ac3_caps; u32 ac3_enable; @@ -204,10 +204,10 @@ static void dice_proc_read(struct snd_info_entry *entry, break; snd_iprintf(buffer, "rx %u:\n", stream); snd_iprintf(buffer, " iso channel: %d\n", (int)buf.rx.iso); + snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); snd_iprintf(buffer, " audio channels: %u\n", buf.rx.number_audio); snd_iprintf(buffer, " midi ports: %u\n", buf.rx.number_midi); - snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); if (quadlets >= 68) { dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE); snd_iprintf(buffer, " names: %s\n", buf.rx.names); From 59294a01d7037f63fb8bf994af10ce63c618770a Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 10 Mar 2015 21:54:35 +0900 Subject: [PATCH 3/9] ALSA: firewire-lib: leave unit reference counting completely With previous commit, this module managed to leave the counting to each drivers, but the isochronous resources functionality still increment/decrement the count. This commit purge such codes to leave the responsibility to each drivers. Fix: c6f224dc20ad ('ALSA: firewire-lib: remove reference counting') Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/iso-resources.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/firewire/iso-resources.c b/sound/firewire/iso-resources.c index 5f17b77ee152..f0e4d502d604 100644 --- a/sound/firewire/iso-resources.c +++ b/sound/firewire/iso-resources.c @@ -26,7 +26,7 @@ int fw_iso_resources_init(struct fw_iso_resources *r, struct fw_unit *unit) { r->channels_mask = ~0uLL; - r->unit = fw_unit_get(unit); + r->unit = unit; mutex_init(&r->mutex); r->allocated = false; @@ -42,7 +42,6 @@ void fw_iso_resources_destroy(struct fw_iso_resources *r) { WARN_ON(r->allocated); mutex_destroy(&r->mutex); - fw_unit_put(r->unit); } EXPORT_SYMBOL(fw_iso_resources_destroy); From ddb6ca75b5671b8fbf1909bc588c449ee74b34f9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Mar 2015 16:05:19 +0100 Subject: [PATCH 4/9] ALSA: hda - Fix built-in mic on Compaq Presario CQ60 Compaq Presario CQ60 laptop with CX20561 gives a wrong pin for the built-in mic NID 0x17 instead of NID 0x1d, and it results in the non-working mic. This patch just remaps the pin correctly via fixup. Bugzilla: https://bugzilla.opensuse.org/show_bug.cgi?id=920604 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index fd3ed18670e9..da67ea8645a6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -223,6 +223,7 @@ enum { CXT_PINCFG_LENOVO_TP410, CXT_PINCFG_LEMOTE_A1004, CXT_PINCFG_LEMOTE_A1205, + CXT_PINCFG_COMPAQ_CQ60, CXT_FIXUP_STEREO_DMIC, CXT_FIXUP_INC_MIC_BOOST, CXT_FIXUP_HEADPHONE_MIC_PIN, @@ -660,6 +661,15 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_PINS, .v.pins = cxt_pincfg_lemote, }, + [CXT_PINCFG_COMPAQ_CQ60] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* 0x17 was falsely set up as a mic, it should 0x1d */ + { 0x17, 0x400001f0 }, + { 0x1d, 0x97a70120 }, + { } + } + }, [CXT_FIXUP_STEREO_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_stereo_dmic, @@ -769,6 +779,7 @@ static const struct hda_model_fixup cxt5047_fixup_models[] = { }; static const struct snd_pci_quirk cxt5051_fixups[] = { + SND_PCI_QUIRK(0x103c, 0x360b, "Compaq CQ60", CXT_PINCFG_COMPAQ_CQ60), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), {} }; From be3bb8236db2d0fcd705062ae2e2a9d75131222f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Mar 2015 18:12:49 +0100 Subject: [PATCH 5/9] ALSA: control: Add sanity checks for user ctl id name string There was no check about the id string of user control elements, so we accepted even a control element with an empty string, which is obviously bogus. This patch adds more sanity checks of id strings. Cc: Signed-off-by: Takashi Iwai --- sound/core/control.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/control.c b/sound/core/control.c index 35324a8e83c8..eeb691d1911f 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1170,6 +1170,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (info->count < 1) return -EINVAL; + if (!*info->id.name) + return -EINVAL; + if (strnlen(info->id.name, sizeof(info->id.name)) >= sizeof(info->id.name)) + return -EINVAL; access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : (info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| SNDRV_CTL_ELEM_ACCESS_INACTIVE| From fcdcd1dec6d2c7b718385ec743ae5a9a233edad4 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 12 Mar 2015 09:41:32 +0100 Subject: [PATCH 6/9] ALSA: snd-usb: add quirks for Roland UA-22 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The device complies to the UAC1 standard but hides that fact with proprietary descriptors. The autodetect quirk for Roland devices catches the audio interface but misses the MIDI part, so a specific quirk is needed. Signed-off-by: Daniel Mack Reported-by: Rafa Lafuente Tested-by: Raphaël Doursenaud Cc: Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 67d476548dcf..07f984d5f516 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1773,6 +1773,36 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0582, 0x0159), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Roland", */ + /* .product_name = "UA-22", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* this catches most recent vendor-specific Roland devices */ { .match_flags = USB_DEVICE_ID_MATCH_VENDOR | From bad994f5b4ab57eec8d56c180edca00505c3eeb2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2015 20:28:04 +0100 Subject: [PATCH 7/9] ALSA: hda - Set single_adc_amp flag for CS420x codecs CS420x codecs seem to deal only the single amps of ADC nodes even though the nodes receive multiple inputs. This leads to the inconsistent amp value after S3/S4 resume, for example. The fix is just to set codec->single_adc_amp flag. Then the driver handles these ADC amps as if single connections. Reported-and-tested-by: Vasil Zlatanov Cc: # 3.9+ Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 1589c9bcce3e..ab687ffb28c2 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -584,6 +584,7 @@ static int patch_cs420x(struct hda_codec *codec) return -ENOMEM; spec->gen.automute_hook = cs_automute; + codec->single_adc_amp = 1; snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl, cs420x_fixups); From 2ddee91abe9cc34ddb6294ee14702b46ae07d460 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2015 20:47:15 +0100 Subject: [PATCH 8/9] ALSA: hda - Add workaround for MacBook Air 5,2 built-in mic MacBook Air 5,2 has the same problem as MacBook Pro 8,1 where the built-in mic records only the right channel. Apply the same workaround as MBP8,1 to spread the mono channel via a Cirrus codec vendor-specific COEF setup. Reported-and-tested-by: Vasil Zlatanov Cc: # 3.9+ Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index ab687ffb28c2..dd2b3d92071f 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -393,6 +393,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = { 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), + SND_PCI_QUIRK(0x106b, 0x5600, "MacBookAir 5,2", CS420X_MBP81), SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42), SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE), {} /* terminator */ From ef403edb75580a3ec5d155f5de82155f0419c621 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2015 08:30:11 +0100 Subject: [PATCH 9/9] ALSA: hda - Don't access stereo amps for mono channel widgets The current HDA generic parser initializes / modifies the amp values always in stereo, but this seems causing the problem on ALC3229 codec that has a few mono channel widgets: namely, these mono widgets react to actions for both channels equally. In the driver code, we do care the mono channel and create a control only for the left channel (as defined in HD-audio spec) for such a node. When the control is updated, only the left channel value is changed. However, in the resume, the right channel value is also restored from the initial value we took as stereo, and this overwrites the left channel value. This ends up being the silent output as the right channel has been never touched and remains muted. This patch covers the places where unconditional stereo amp accesses are done and converts to the conditional accesses. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=94581 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 30 ++++++++++++++++++++++-------- 1 file changed, 22 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b680b4ec6331..fe18071bf93a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -692,7 +692,23 @@ static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx) { unsigned int caps = query_amp_caps(codec, nid, dir); int val = get_amp_val_to_activate(codec, nid, dir, caps, false); - snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val); + + if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val); + else + snd_hda_codec_amp_init(codec, nid, 0, dir, idx, 0xff, val); +} + +/* update the amp, doing in stereo or mono depending on NID */ +static int update_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx, + unsigned int mask, unsigned int val) +{ + if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + return snd_hda_codec_amp_stereo(codec, nid, dir, idx, + mask, val); + else + return snd_hda_codec_amp_update(codec, nid, 0, dir, idx, + mask, val); } /* calculate amp value mask we can modify; @@ -732,7 +748,7 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir, return; val &= mask; - snd_hda_codec_amp_stereo(codec, nid, dir, idx, mask, val); + update_amp(codec, nid, dir, idx, mask, val); } static void activate_amp_out(struct hda_codec *codec, struct nid_path *path, @@ -4424,13 +4440,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) has_amp = nid_has_mute(codec, mix, HDA_INPUT); for (i = 0; i < nums; i++) { if (has_amp) - snd_hda_codec_amp_stereo(codec, mix, - HDA_INPUT, i, - 0xff, HDA_AMP_MUTE); + update_amp(codec, mix, HDA_INPUT, i, + 0xff, HDA_AMP_MUTE); else if (nid_has_volume(codec, conn[i], HDA_OUTPUT)) - snd_hda_codec_amp_stereo(codec, conn[i], - HDA_OUTPUT, 0, - 0xff, HDA_AMP_MUTE); + update_amp(codec, conn[i], HDA_OUTPUT, 0, + 0xff, HDA_AMP_MUTE); } }