From 7679e42ec833ed70aa34790a5f39dcb7e5bda4fe Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Feb 2012 15:52:56 +0000 Subject: ASoC: dapm: Check for bias level when powering down Recent enhancements in the bias management means that we might not be in standby when the CODEC is idle and can have active widgets without being in full power mode but the shutdown functionality assumes these things. Add checks for the bias level at each stage so that we don't do transitions other than the ON->PREPARE->STANDBY->OFF ones that the drivers are expecting. Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/soc-dapm.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1f55ded..1315663 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3068,9 +3068,13 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) * standby. */ if (powerdown) { - snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE); + if (dapm->bias_level == SND_SOC_BIAS_ON) + snd_soc_dapm_set_bias_level(dapm, + SND_SOC_BIAS_PREPARE); dapm_seq_run(dapm, &down_list, 0, false); - snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY); + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) + snd_soc_dapm_set_bias_level(dapm, + SND_SOC_BIAS_STANDBY); } } @@ -3083,7 +3087,9 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card) list_for_each_entry(codec, &card->codec_dev_list, list) { soc_dapm_shutdown_codec(&codec->dapm); - snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF); + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + snd_soc_dapm_set_bias_level(&codec->dapm, + SND_SOC_BIAS_OFF); } } -- cgit v1.1 From 5ed80a75b248bfaf840ea6b38f941edcf6ee7dc7 Mon Sep 17 00:00:00 2001 From: Javier Martin Date: Thu, 23 Feb 2012 15:43:18 +0100 Subject: ASoC: i.MX SSI: Fix DSP_A format. According to i.MX27 Reference Manual (p 1593) TXBIT0 bit selects whether the most significant or the less significant part of the data word written to the FIFO is transmitted. As DSP_A is the same as DSP_B with a data offset of 1 bit, it doesn't make any sense to remove TXBIT0 bit here. Signed-off-by: Javier Martin Acked-by: Sascha Hauer Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/imx/imx-ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 01d1f74..b6adbed 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -112,7 +112,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) break; case SND_SOC_DAIFMT_DSP_A: /* data on rising edge of bclk, frame high 1clk before data */ - strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS; + strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS; break; } -- cgit v1.1 From 068b939431486f524438330b0848a8222e33d421 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 25 Feb 2012 11:13:16 +0100 Subject: ALSA: hda/realtek - Fix resume of multiple input sources When there are multiple input sources, the driver wrongly overwrites with the value of the last input source on other slots at resume. Thus the primary input source may be shown wrongly. Reported-and-tested-by: Julian Sikorski Cc: [v3.1+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3647baa..4fe2d59 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3797,7 +3797,7 @@ static void alc_auto_init_input_src(struct hda_codec *codec) else nums = spec->num_adc_nids; for (c = 0; c < nums; c++) - alc_mux_select(codec, 0, spec->cur_mux[c], true); + alc_mux_select(codec, c, spec->cur_mux[c], true); } /* add mic boosts if needed */ -- cgit v1.1 From 87c9e7d7027643bf248b396c15c804456e967fcd Mon Sep 17 00:00:00 2001 From: Alban Bedel Date: Sat, 25 Feb 2012 16:15:57 +0100 Subject: ALSA: azt3328 - Fix NULL ptr dereference on cards without OPL3 opl3->private_data was set even if opl3 could not be created. Signed-off-by: Alban Bedel Signed-off-by: Takashi Iwai --- sound/pci/azt3328.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 95ffa6a..496f14c 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2684,10 +2684,9 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (err < 0) goto out_err; + opl3->private_data = chip; } - opl3->private_data = chip; - sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, chip->ctrl_io, chip->irq); -- cgit v1.1 From 7bff172a352a2fbe9856bba517d71a2072aab041 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 Feb 2012 09:41:17 +0100 Subject: ALSA: hda - Always set HP pin in unsol handler for STAC/IDT codecs A bug report with an old Sony laptop showed that we can't rely on BIOS setting the pins of headphones but the driver should set always by itself. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6345df1..9dbb573 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4629,7 +4629,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec) unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN; if (no_hp_sensing(spec, i)) continue; - if (presence) + if (1 /*presence*/) stac92xx_set_pinctl(codec, cfg->hp_pins[i], val); #if 0 /* FIXME */ /* Resetting the pinctl like below may lead to (a sort of) regressions -- cgit v1.1 From 3868137ea41866773e75d9ac4b9988dcc361ff1d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 Feb 2012 15:00:58 +0100 Subject: ALSA: hda - Add a fake mute feature Some codecs don't supply the mute amp-capabilities although the lowest volume gives the mute. It'd be handy if the parser provides the mute mixers in such a case. This patch adds an extension amp-cap bit (which is used only in the driver) to represent the min volume = mute state. Also modified the amp cache code to support the fake mute feature when this bit is set but the real mute bit is unset. In addition, conexant cx5051 parser uses this new feature to implement the missing mute controls. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42825 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 8 ++++++-- sound/pci/hda/hda_codec.h | 3 +++ sound/pci/hda/patch_conexant.c | 22 +++++++++++++++++++++- 3 files changed, 30 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c2c65f6..0ae6eb20 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1759,7 +1759,11 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info, parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT; parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT; parm |= index << AC_AMP_SET_INDEX_SHIFT; - parm |= val; + if ((val & HDA_AMP_MUTE) && !(info->amp_caps & AC_AMPCAP_MUTE) && + (info->amp_caps & AC_AMPCAP_MIN_MUTE)) + ; /* set the zero value as a fake mute */ + else + parm |= val; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm); info->vol[ch] = val; } @@ -2026,7 +2030,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT); val1 += ofs; val1 = ((int)val1) * ((int)val2); - if (min_mute) + if (min_mute || (caps & AC_AMPCAP_MIN_MUTE)) val2 |= TLV_DB_SCALE_MUTE; if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv)) return -EFAULT; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index e9f71dc..f0f1943 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -298,6 +298,9 @@ enum { #define AC_AMPCAP_MUTE (1<<31) /* mute capable */ #define AC_AMPCAP_MUTE_SHIFT 31 +/* driver-specific amp-caps: using bits 24-30 */ +#define AC_AMPCAP_MIN_MUTE (1 << 30) /* min-volume = mute */ + /* Connection list */ #define AC_CLIST_LENGTH (0x7f<<0) #define AC_CLIST_LONG (1<<7) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a7a5733..ca117cf 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4079,7 +4079,8 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, err = snd_hda_ctl_add(codec, nid, kctl); if (err < 0) return err; - if (!(query_amp_caps(codec, nid, hda_dir) & AC_AMPCAP_MUTE)) + if (!(query_amp_caps(codec, nid, hda_dir) & + (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE))) break; } return 0; @@ -4379,6 +4380,22 @@ static const struct snd_pci_quirk cxt_fixups[] = { {} }; +/* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches + * can be created (bko#42825) + */ +static void add_cx5051_fake_mutes(struct hda_codec *codec) +{ + static hda_nid_t out_nids[] = { + 0x10, 0x11, 0 + }; + hda_nid_t *p; + + for (p = out_nids; *p; p++) + snd_hda_override_amp_caps(codec, *p, HDA_OUTPUT, + AC_AMPCAP_MIN_MUTE | + query_amp_caps(codec, *p, HDA_OUTPUT)); +} + static int patch_conexant_auto(struct hda_codec *codec) { struct conexant_spec *spec; @@ -4397,6 +4414,9 @@ static int patch_conexant_auto(struct hda_codec *codec) case 0x14f15045: spec->single_adc_amp = 1; break; + case 0x14f15051: + add_cx5051_fake_mutes(codec); + break; } apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); -- cgit v1.1 From e49a3434f1bc64dc49ff3a56e416bb5894868dde Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Mar 2012 18:14:41 +0100 Subject: ALSA: hda - Kill hyphenated names Kill hyphens from "Line-Out" name strings, as suggested by Mark Brown. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 ++-- sound/pci/hda/patch_cirrus.c | 4 ++-- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_realtek.c | 6 +++--- 4 files changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0ae6eb20..6843073 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5118,7 +5118,7 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, const char *pfx = "", *sfx = ""; /* handle as a speaker if it's a fixed line-out */ - if (!strcmp(name, "Line-Out") && attr == INPUT_PIN_ATTR_INT) + if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT) name = "Speaker"; /* check the location */ switch (attr) { @@ -5177,7 +5177,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, switch (get_defcfg_device(def_conf)) { case AC_JACK_LINE_OUT: - return fill_audio_out_name(codec, nid, cfg, "Line-Out", + return fill_audio_out_name(codec, nid, cfg, "Line Out", label, maxlen, indexp); case AC_JACK_SPEAKER: return fill_audio_out_name(codec, nid, cfg, "Speaker", diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index bc5a993..c83ccdb 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -609,7 +609,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, "Front Speaker", "Surround Speaker", "Bass Speaker" }; static const char * const line_outs[] = { - "Front Line-Out", "Surround Line-Out", "Bass Line-Out" + "Front Line Out", "Surround Line Out", "Bass Line Out" }; fix_volume_caps(codec, dac); @@ -635,7 +635,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, if (num_ctls > 1) name = line_outs[idx]; else - name = "Line-Out"; + name = "Line Out"; break; } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ca117cf..d29d6d3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3482,7 +3482,7 @@ static int cx_automute_mode_info(struct snd_kcontrol *kcontrol, "Disabled", "Enabled" }; static const char * const texts3[] = { - "Disabled", "Speaker Only", "Line-Out+Speaker" + "Disabled", "Speaker Only", "Line Out+Speaker" }; const char * const *texts; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4fe2d59..f286bb8f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -802,7 +802,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol, "Disabled", "Enabled" }; static const char * const texts3[] = { - "Disabled", "Speaker Only", "Line-Out+Speaker" + "Disabled", "Speaker Only", "Line Out+Speaker" }; const char * const *texts; @@ -1856,7 +1856,7 @@ static const char * const alc_slave_vols[] = { "Headphone Playback Volume", "Speaker Playback Volume", "Mono Playback Volume", - "Line-Out Playback Volume", + "Line Out Playback Volume", "CLFE Playback Volume", "Bass Speaker Playback Volume", "PCM Playback Volume", @@ -1873,7 +1873,7 @@ static const char * const alc_slave_sws[] = { "Speaker Playback Switch", "Mono Playback Switch", "IEC958 Playback Switch", - "Line-Out Playback Switch", + "Line Out Playback Switch", "CLFE Playback Switch", "Bass Speaker Playback Switch", "PCM Playback Switch", -- cgit v1.1 From b2ccf065f7b23147ed135a41b01d05a332ca6b7e Mon Sep 17 00:00:00 2001 From: Denis 'GNUtoo' Carikli Date: Sun, 26 Feb 2012 19:21:54 +0100 Subject: ASoC: neo1973: fix neo1973 wm8753 initialization The neo1973 driver had wrong codec name which prevented the "sound card" from appearing. Signed-off-by: Denis 'GNUtoo' Carikli Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/samsung/neo1973_wm8753.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index c6012ff..d23b19a 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -367,7 +367,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .platform_name = "samsung-audio", .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "wm8753-hifi", - .codec_name = "wm8753-codec.0-001a", + .codec_name = "wm8753.0-001a", .init = neo1973_wm8753_init, .ops = &neo1973_hifi_ops, }, @@ -376,7 +376,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .stream_name = "Voice", .cpu_dai_name = "dfbmcs320-pcm", .codec_dai_name = "wm8753-voice", - .codec_name = "wm8753-codec.0-001a", + .codec_name = "wm8753.0-001a", .ops = &neo1973_voice_ops, }, }; -- cgit v1.1 From 8f2392142346f2754c8292a94cc62a157ed1e093 Mon Sep 17 00:00:00 2001 From: Marton Balint Date: Mon, 5 Mar 2012 21:33:23 +0100 Subject: ALSA: hda - add quirk to detect CD input on Gigabyte EP45-DS3 My CD input got lost in commit 68ef0561efe494143516df38c03a16b837b8e79c. Raymond helped me to add the necessary pin fixup to make it appear again. In fact, this is basically his patch. It fixes alsa bug #5541. Signed-off-by: Marton Balint 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 f286bb8f..5e53020 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4367,6 +4367,7 @@ enum { ALC882_FIXUP_PB_M5210, ALC882_FIXUP_ACER_ASPIRE_7736, ALC882_FIXUP_ASUS_W90V, + ALC889_FIXUP_CD, ALC889_FIXUP_VAIO_TT, ALC888_FIXUP_EEE1601, ALC882_FIXUP_EAPD, @@ -4494,6 +4495,13 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [ALC889_FIXUP_CD] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1c, 0x993301f0 }, /* CD */ + { } + } + }, [ALC889_FIXUP_VAIO_TT] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { @@ -4650,6 +4658,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), -- cgit v1.1 From 526af6eb4dc71302f59806e2ccac7793963a7fe0 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 7 Mar 2012 08:25:20 +0100 Subject: ALSA: hda/realtek - Apply the coef-setup only to ALC269VB The coef setup in alc269_fill_coef() was designed only for ALC269VB model, and this has some bad effects for other ALC269 variants, such as turning off the external mic input. Apply it only to ALC269VB. Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5e53020..22c73b7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2068,12 +2068,16 @@ static int alc_build_controls(struct hda_codec *codec) */ static void alc_init_special_input_src(struct hda_codec *codec); +static int alc269_fill_coef(struct hda_codec *codec); static int alc_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; unsigned int i; + if (codec->vendor_id == 0x10ec0269) + alc269_fill_coef(codec); + alc_fix_pll(codec); alc_auto_init_amp(codec, spec->init_amp); @@ -5476,8 +5480,12 @@ static const struct alc_model_fixup alc269_fixup_models[] = { static int alc269_fill_coef(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; int val; + if (spec->codec_variant != ALC269_TYPE_ALC269VB) + return 0; + if ((alc_get_coef0(codec) & 0x00ff) < 0x015) { alc_write_coef_idx(codec, 0xf, 0x960b); alc_write_coef_idx(codec, 0xe, 0x8817); -- cgit v1.1 From 8de5d6f19bbe7c77676a62ab52be901aa10d6b54 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Thu, 8 Mar 2012 15:38:04 +0100 Subject: ALSA: hdspm - Provide ioctl_compat snd_hdspm uses its own ioctls to acquire config- and status information. Expose the corresponding ioctl handler via ioctl_compat, so that 32bit applications can use it on 64bit kernels. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index cc9f6c8..bc030a2 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6333,6 +6333,7 @@ static int __devinit snd_hdspm_create_hwdep(struct snd_card *card, hw->ops.open = snd_hdspm_hwdep_dummy_op; hw->ops.ioctl = snd_hdspm_hwdep_ioctl; + hw->ops.ioctl_compat = snd_hdspm_hwdep_ioctl; hw->ops.release = snd_hdspm_hwdep_dummy_op; return 0; -- cgit v1.1