September 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [3.6-rc7] switcheroo race with Intel HDA...
by Daniel J Blueman 08 Oct '12

08 Oct '12

On my Macbook with a discrete Nvidia GPU, there is a race between selecting the integrated GPU and putting the discrete GPU into D3 [1], reliably causing a kernel oops [2]. Introducing a delay of ~1s between the calls prevents this. When the second 'OFF' write path executes, it looks like struct azx at card->private_data hasn't yet been allocated yet [3], so there is likely some locking missing. I'm happy to perform further testing and debug of course... Thanks, Daniel --- [1] echo IGD > /sys/kernel/debug/vgaswitcheroo/switch echo OFF > /sys/kernel/debug/vgaswitcheroo/switch --- [2] BUG: unable to handle kernel NULL pointer dereference at 0000000000000170 IP: [<ffffffffa01ba936>] azx_vs_set_state+0x26/0x178 [snd_hda_intel] PGD 259c26067 PUD 25a0fd067 PMD 0 Oops: 0000 [#1] SMP DEBUG_PAGEALLOC Modules linked in: snd_hda_codec_hdmi bnep rfcomm b43 joydev nfsd ssb nfs_acl auth_rpcgss binfmt_misc nfs lockd sunrpc uvcvideo bcm5974 videobuf2_core videobuf2_vmalloc videobuf2_memops coretemp kvm_intel snd_hda_codec_cirrus kvm applesmc input_polldev microcode bcma lpc_ich mfd_core mei snd_hda_intel(+) snd_hda_codec snd_hwdep snd_pcm snd_timer snd snd_page_alloc nls_iso8859_1 apple_gmux mac_hid apple_bl btrfs hid_apple sdhci_pci ghash_clmulni_intel tg3 sdhci i915 nouveau ttm drm_kms_helper hwmon mxm_wmi video CPU 2 Pid: 961, comm: sh Not tainted 3.6.0-rc7 #2 Apple Inc. MacBookPro10,1/Mac-C3EC7CD22292981F RIP: 0010:[<ffffffffa01ba936>] [<ffffffffa01ba936>] azx_vs_set_state+0x26/0x178 [snd_hda_intel] RSP: 0018:ffff880264271e48 EFLAGS: 00010286 RAX: 0000000000000000 RBX: ffff88025a2f5280 RCX: 0000000000000000 RDX: 0000000000000006 RSI: 0000000000000000 RDI: ffff880265479098 RBP: ffff880264271e68 R08: 2222222222222222 R09: 2222222222222222 R10: 0000000000000000 R11: 0000000000000000 R12: ffff880265479098 R13: 0000000000000000 R14: ffff880264271f50 R15: 0000000000000000 FS: 00007fa4fe183700(0000) GS:ffff88026f280000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000000000170 CR3: 00000002641a7000 CR4: 00000000001407e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 Process sh (pid: 961, threadinfo ffff880264270000, task ffff880264503a00) Stack: 2222222222222222 ffff88025a2f5280 0000000000000000 ffff880264271e98 ffff880264271e88 ffffffff812e83a7 ffff8802622835c0 0000000000000004 ffff880264271ef8 ffffffff812e89ac ffff88020a46464f ffff880264503a00 Call Trace: [<ffffffff812e83a7>] set_audio_state+0x67/0x70 [<ffffffff812e89ac>] vga_switcheroo_debugfs_write+0xbc/0x380 [<ffffffff81108773>] vfs_write+0xa3/0x160 [<ffffffff81108a75>] sys_write+0x45/0xa0 [<ffffffff815231a6>] system_call_fastpath+0x1a/0x1f Code: 00 00 00 00 00 55 48 89 e5 48 83 ec 20 4c 89 65 f0 4c 8d a7 98 00 00 00 4c 89 e7 48 89 5d e8 4c 89 6d f8 41 89 f5 e8 2a 35 13 e1 <48> 8b 98 70 01 00 00 0f b6 83 55 02 00 00 a8 08 75 34 45 85 ed RIP [<ffffffffa01ba936>] azx_vs_set_state+0x26/0x178 [snd_hda_intel] RSP <ffff880264271e48> CR2: 0000000000000170 --- [3] (gdb) list *(azx_vs_set_state+0x26) 0x2936 is in azx_vs_set_state (sound/pci/hda/hda_intel.c:2505). 2500 2501 static void azx_vs_set_state(struct pci_dev *pci, 2502 enum vga_switcheroo_state state) 2503 { 2504 struct snd_card *card = pci_get_drvdata(pci); 2505 struct azx *chip = card->private_data; 2506 bool disabled; 2507 2508 if (chip->init_failed) 2509 return; -- Daniel J Blueman

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[alsa-devel] Constant noise on HDA ALC275
by Julien Danjou 08 Oct '12

08 Oct '12

Hi there, I've an HDA based chipset in a new Sony Vaio Z (Ivy Bridge). As sound as the system starts and the sound module are loaded, there's a constant noise (shhhhhhht) in the speakers. The noise change a bit if I toggle the mute button, but it's still there. This noise stops as sound as I plug something in the jack port. I've tried to play with HDA Analyzer¹, but I've no clue what to look for, so any hint appreciated. Some information: 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller (rev 04) % aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC275 Analog [ALC275 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 % lsmod | grep snd snd_hda_codec_hdmi 30783 1 snd_hda_codec_realtek 50906 1 snd_hda_intel 26504 3 snd_hda_codec 83533 3 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_intel snd_hwdep 13186 1 snd_hda_codec snd_pcm 64080 3 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel snd_page_alloc 12969 2 snd_pcm,snd_hda_intel snd_seq 45130 0 snd_seq_device 13176 1 snd_seq snd_timer 22917 2 snd_pcm,snd_seq snd 53077 15 snd_hda_codec_realtek,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_seq,snd_hda_codec,snd_hda_intel,snd_seq_device soundcore 13026 1 snd /proc/asound/card0/codec#0 ========================== /proc/asound/card0/codec#3 ========================== ¹ http://www.alsa-project.org/main/index.php/HDA_Analyzer -- Julien Danjou # Free Software hacker & freelance # http://julien.danjou.info

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[alsa-devel] [PATCH] ALSA: aloop - add locking to timer access
by Omair Mohammed Abdullah 08 Oct '12

08 Oct '12

When the loopback timer handler is running, calling del_timer() (for STOP trigger) will not wait for the handler to complete before deactivating the timer. The timer gets rescheduled in the handler as usual. Then a subsequent START trigger will try to start the timer using add_timer() with a timer pending leading to a kernel panic. Serialize the calls to add_timer() and del_timer() using a spin lock to avoid this. Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- sound/drivers/aloop.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 5a34355..0fe6d64 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -120,6 +120,7 @@ struct loopback_pcm { unsigned int last_drift; unsigned long last_jiffies; struct timer_list timer; + spinlock_t timer_lock; }; static struct platform_device *devices[SNDRV_CARDS]; @@ -170,6 +171,7 @@ static void loopback_timer_start(struct loopback_pcm *dpcm) unsigned long tick; unsigned int rate_shift = get_rate_shift(dpcm); + spin_lock(&dpcm->timer_lock); if (rate_shift != dpcm->pcm_rate_shift) { dpcm->pcm_rate_shift = rate_shift; dpcm->period_size_frac = frac_pos(dpcm, dpcm->pcm_period_size); @@ -182,12 +184,15 @@ static void loopback_timer_start(struct loopback_pcm *dpcm) tick = (tick + dpcm->pcm_bps - 1) / dpcm->pcm_bps; dpcm->timer.expires = jiffies + tick; add_timer(&dpcm->timer); + spin_unlock(&dpcm->timer_lock); } static inline void loopback_timer_stop(struct loopback_pcm *dpcm) { + spin_lock(&dpcm->timer_lock); del_timer(&dpcm->timer); dpcm->timer.expires = 0; + spin_unlock(&dpcm->timer_lock); } #define CABLE_VALID_PLAYBACK (1 << SNDRV_PCM_STREAM_PLAYBACK) @@ -667,6 +672,7 @@ static int loopback_open(struct snd_pcm_substream *substream) dpcm->substream = substream; setup_timer(&dpcm->timer, loopback_timer_function, (unsigned long)dpcm); + spin_lock_init(&dpcm->timer_lock); cable = loopback->cables[substream->number][dev]; if (!cable) { -- 1.7.4.1

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[alsa-devel] [PATCH] ALSA: hda - Fix hang caused by race during suspend.
by Dylan Reid 08 Oct '12

08 Oct '12

There was a race condition when the system suspends while hda_power_work is running in the work queue. If system suspend (snd_hda_suspend) happens after the work queue releases power_lock but before it calls hda_call_codec_suspend, codec_suspend runs with power_on=0, causing the codec to power up for register reads, and hanging when it calls cancel_delayed_work_sync from the running work queue. The call chain from the work queue will look like this: hda_power_work <<- power_on = 1, unlock, then power_on cleard by suspend hda_call_codec_suspend hda_set_power_state snd_hda_codec_read codec_exec_verb snd_hda_power_up snd_hda_power_save __snd_hda_power_up cancel_delayed_work_sync <<-- cancelling executing wq Fix this by waiting for the work queue to finish before starting suspend if suspend is not happening on the work queue. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_codec.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 960800b..c0ab72c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3612,7 +3612,7 @@ static inline void hda_exec_init_verbs(struct hda_codec *codec) {} * call suspend and power-down; used both from PM and power-save * this function returns the power state in the end */ -static unsigned int hda_call_codec_suspend(struct hda_codec *codec) +static unsigned int hda_call_codec_suspend(struct hda_codec *codec, bool in_wq) { unsigned int state; @@ -3620,7 +3620,9 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec) codec->patch_ops.suspend(codec); hda_cleanup_all_streams(codec); state = hda_set_power_state(codec, AC_PWRST_D3); - cancel_delayed_work(&codec->power_work); + /* Cancel delayed work if we aren't currently running from it. */ + if (!in_wq) + cancel_delayed_work_sync(&codec->power_work); spin_lock(&codec->power_lock); snd_hda_update_power_acct(codec); trace_hda_power_down(codec); @@ -4478,7 +4480,7 @@ static void hda_power_work(struct work_struct *work) } spin_unlock(&codec->power_lock); - state = hda_call_codec_suspend(codec); + state = hda_call_codec_suspend(codec, true); codec->pm_down_notified = 0; if (!bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) { codec->pm_down_notified = 1; @@ -5127,7 +5129,7 @@ int snd_hda_suspend(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { if (hda_codec_is_power_on(codec)) - hda_call_codec_suspend(codec); + hda_call_codec_suspend(codec, false); } return 0; } -- 1.7.12.146.g16d26b1

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[alsa-devel] [PATCH] ALSA: hda/realtek - Fix detection of ALC271X codec
by Herton Ronaldo Krzesinski 08 Oct '12

08 Oct '12

In commit af741c1 ("ALSA: hda/realtek - Call alc_auto_parse_customize_define() always after fixup"), alc_auto_parse_customize_define was moved after detection of ALC271X. The problem is that detection of ALC271X relies on spec->cdefine.platform_type, and it's set on alc_auto_parse_customize_define. Move the alc_auto_parse_customize_define and its required fixup setup before the block doing the ALC271X and other codec setup. BugLink: https://bugs.launchpad.net/bugs/1006690 Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) It looks safe to move the functions before the block "if (codec->vendor_id == 0x10ec0269) {", as nothing on the fixups seems to depend on what's done there. But if wanted we could just move the detection/codec rename of ALC271X below too. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c4c4b01..3b187e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6273,6 +6273,12 @@ static int patch_alc269(struct hda_codec *codec) spec = codec->spec; + alc_pick_fixup(codec, alc269_fixup_models, + alc269_fixup_tbl, alc269_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + + alc_auto_parse_customize_define(codec); + if (codec->vendor_id == 0x10ec0269) { spec->codec_variant = ALC269_TYPE_ALC269VA; switch (alc_get_coef0(codec) & 0x00f0) { @@ -6300,12 +6306,6 @@ static int patch_alc269(struct hda_codec *codec) alc269_fill_coef(codec); } - alc_pick_fixup(codec, alc269_fixup_models, - alc269_fixup_tbl, alc269_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - - alc_auto_parse_customize_define(codec); - /* automatic parse from the BIOS config */ err = alc269_parse_auto_config(codec); if (err < 0) -- 1.7.10.4

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[alsa-devel] [PATCH 1/1] ALSA: CA0132 - Ensure SCP packet is little-endian format
by Ian Minett 08 Oct '12

08 Oct '12

From: Ian Minett <ian_minett(a)creativelabs.com> Hi Takashi, Great - good to hear the previous DSP loader patches are now living in the hda-ca0132-dsp branch. This patch is based on that branch, and addresses the SCP packet endianness. >> The firmware does need to be reloaded when the chip loses power, so it >> doesn't survive S3. >OK, we need suspend/resume hooks, too. Do we need to do anything specific to handle this? The firmware is already restored upon S3 by the current caching code. Thanks, - Ian --- The DSP firmware expects the SCP data to be in little-endian format regardless of the host architecture. Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> --- diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 2fd3121..4d7cda4 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -28,6 +28,7 @@ #include <linux/mutex.h> #include <linux/module.h> #include <linux/firmware.h> +#include <asm/unaligned.h> #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" @@ -701,9 +702,13 @@ static int dspio_send(struct hda_codec *codec, unsigned int reg, unsigned int res; int retry = 50; + /* Handling for Big Endian machine */ + unsigned int val; + put_unaligned_le32(data, &val); + /* send bits of data specified by reg to dsp */ do { - res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, data); + res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, val); if ((res >= 0) && (res != VENDOR_STATUS_DSPIO_BUSY)) return res; } while (--retry); -- 1.7.4.1

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[alsa-devel] [PATCH 0/10] ALSA: Add DSP firmware loader (v3)
by Ian Minett 08 Oct '12

08 Oct '12

From: Ian Minett <ian_minett(a)creativelabs.com> Hi, thanks for the response. I've resubmitted the patch series with updates to the ones that needed altering based on the feedback. The MODULE_FIRMWARE line has been shifted to the firmware loading patch, DSP download is now guarded by the config switch and code changes that were bundled in with the previous comments patch have moved to a new patch. Also, hda_stream_format has been removed. Regarding the SCP packet endianness, the DSP image requires data to be sent as little-endian, so we may need to swizzle data being sent to the DSP before the transfer. Is there a recommended or standard way of detecting and handling the case of running on big-endian architectures? >Does the firmware on chip survive after S3? The firmware does need to be reloaded when the chip loses power, so it doesn't survive S3. Thanks very much, - Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> --- 1: - memalloc.h - pcm.h - pcm_memory.c - sgbuf.c Include Takashi's patch: Make snd_sgbuf_get_{ptr|addr}() available for non-SG cases. Passing struct snd_dma_buffer pointer instead, so that they work no matter whether real SG buffer is used or not. 2: - hda_intel.c - hda_codec.h Include Takashi's code: Pass DMA buffer pointers in calls to setup_bdle(). Add DSP loader callback routines to controller. Add new DSP loader switch to Kconfig to enable DSP firmware loading. 3: - patch_ca0132.c - ca0132_regs.h Add DSP register definitions header file 4 (updated): - patch_ca0132.c Add DSP firmware enums and defs to CA0132 codec MODULE_FIRMWARE not declared in this patch, moved into patch #5 5 (updated): - patch_ca0132.c Add calls to new DSP loader system to transfer firmware binary to the hardware. Add chip read/write routines, DSP I/O, SCP packet format helper functions and transfer DMA management. Add MODULE_FIRMWARE line for DSP firmware binary Protect DSP download using CONFIG_SND_HDA_DSP_LOADER switch 6: - patch_ca0132.c Add DSP firmware caching to CA0132 codec 7 (updated): - patch_ca0132.c Add comments and descriptions to functions. Previous code changes moved to patch #9 8: - hda_codec.h Change return value for load_dsp_prepare to -ENOSYS in case where DSP loader routines are not available. 9 (new): - patch_ca0132.c Merge chipio write address functions and update timeout mechanism in dsp_write_wait(). 10 (new): - patch_ca0132.c Remove unnecessary struct hda_stream_format and references to it.

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[alsa-devel] [PATCH] ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
by Felix Kaechele 08 Oct '12

08 Oct '12

The Lenovo IdeaPad U310 has an internal mic where the right channel is phase inverted. Signed-off-by: Felix Kaechele <felix(a)fetzig.org> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5e22a8f..784017e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4462,6 +4462,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), {} }; -- 1.7.11.4

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[alsa-devel] [PATCH 1/2] ALSA: hda - make a generic unsol event handler
by David Henningsson 08 Oct '12

08 Oct '12

Moving towards less duplication of code between codecs - this patch takes some of the common code of unsol event handling and makes it generic. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_jack.c | 32 ++++++++++++++++++++++++++++++-- sound/pci/hda/hda_jack.h | 9 +++++++++ 2 files changed, 39 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index c9333c9..5c690cb 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -192,8 +192,9 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_detect); /** * snd_hda_jack_detect_enable - enable the jack-detection */ -int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, - unsigned char action) +int snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, + unsigned char action, + hda_jack_callback cb) { struct hda_jack_tbl *jack = snd_hda_jack_tbl_new(codec, nid); if (!jack) @@ -203,10 +204,19 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, jack->jack_detect = 1; if (action) jack->action = action; + if (cb) + jack->callback = cb; return snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | jack->tag); } +EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable_callback); + +int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, + unsigned char action) +{ + return snd_hda_jack_detect_enable_callback(codec, nid, action, NULL); +} EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable); /** @@ -411,3 +421,21 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, return 0; } EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctls); + +void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res) +{ + struct hda_jack_tbl *event; + int tag = (res >> AC_UNSOL_RES_TAG_SHIFT) & 0x7f; + + event = snd_hda_jack_tbl_get_from_tag(codec, tag); + if (!event) + return; + event->jack_dirty = 1; + + if (event->callback) + event->callback(codec, event); + + snd_hda_jack_report_sync(codec); +} +EXPORT_SYMBOL_HDA(snd_hda_jack_unsol_event); + diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index a9803da..af8dd47 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -13,12 +13,16 @@ #define __SOUND_HDA_JACK_H struct auto_pin_cfg; +struct hda_jack_tbl; + +typedef void (*hda_jack_callback) (struct hda_codec *, struct hda_jack_tbl *); struct hda_jack_tbl { hda_nid_t nid; unsigned char action; /* event action (0 = none) */ unsigned char tag; /* unsol event tag */ unsigned int private_data; /* arbitrary data */ + hda_jack_callback callback; /* jack-detection stuff */ unsigned int pin_sense; /* cached pin-sense value */ unsigned int jack_detect:1; /* capable of jack-detection? */ @@ -61,6 +65,10 @@ void snd_hda_jack_set_dirty_all(struct hda_codec *codec); int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, unsigned char action); +int snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, + unsigned char action, + hda_jack_callback cb); + u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); @@ -74,5 +82,6 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, void snd_hda_jack_report_sync(struct hda_codec *codec); +void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res); #endif /* __SOUND_HDA_JACK_H */ -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: ams-delta: fix card initalization failure
by Janusz Krzysztofik 04 Oct '12

04 Oct '12

Since commit 0998d0631001288a5974afc0b2a5f568bcdecb4d, 'device-core: Ensure drvdata = NULL when no driver is bound', the Amstrad Delta sound card no longer initializes correctly due to drvdata reset to NULL by an upper layer before the codec device, required for successful card setup, is registered. Fix this by moving the codec registration bits up, before the card is probed for. Created and tested against linux-3.6-rc3 Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- sound/soc/omap/ams-delta.c | 18 ++++++++++-------- 1 files changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 7d4fa8e..270de9c 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -590,20 +590,22 @@ static int __init ams_delta_module_init(void) if (!ams_delta_audio_platform_device) return -ENOMEM; - platform_set_drvdata(ams_delta_audio_platform_device, - &ams_delta_audio_card); - - ret = platform_device_add(ams_delta_audio_platform_device); - if (ret) - goto err; - /* * Codec platform device could be registered from elsewhere (board?), * but I do it here as it makes sense only if used with the card. + * Moreover, it must be registered before the card is probed for, + * or the card setup fails due to drvdata reset by upper layers. */ cx20442_platform_device = platform_device_register_simple("cx20442-codec", -1, NULL, 0); - return 0; + + platform_set_drvdata(ams_delta_audio_platform_device, + &ams_delta_audio_card); + + ret = platform_device_add(ams_delta_audio_platform_device); + if (!ret) + return ret; + err: platform_device_put(ams_delta_audio_platform_device); return ret; -- 1.7.3.4

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