From ec8f53fb693dda095ad3342b927a074e7c4dddfa Mon Sep 17 00:00:00 2001 From: Masanari Iida Date: Fri, 2 Nov 2012 00:28:50 +0900 Subject: ALSA: Fix typo in drivers sound Correct spelling typo in debug messages within drivers/sound Signed-off-by: Masanari Iida Signed-off-by: Takashi Iwai diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c index ef68d71..e04e750 100644 --- a/sound/i2c/other/ak4113.c +++ b/sound/i2c/other/ak4113.c @@ -426,7 +426,7 @@ static struct snd_kcontrol_new snd_ak4113_iec958_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "IEC958 Preample Capture Default", + .name = "IEC958 Preamble Capture Default", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ak4113_spdif_pinfo, diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index 816e7d2..5bf4fca 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -401,7 +401,7 @@ static struct snd_kcontrol_new snd_ak4114_iec958_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "IEC958 Preample Capture Default", + .name = "IEC958 Preamble Capture Default", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ak4114_spdif_pinfo, .get = snd_ak4114_spdif_pget, diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c index b4b2a51..40e33c9 100644 --- a/sound/i2c/other/ak4117.c +++ b/sound/i2c/other/ak4117.c @@ -380,7 +380,7 @@ static struct snd_kcontrol_new snd_ak4117_iec958_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "IEC958 Preample Capture Default", + .name = "IEC958 Preamble Capture Default", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ak4117_spdif_pinfo, .get = snd_ak4117_spdif_pget, diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index f1cd1e3..9a8d5ce 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -4899,7 +4899,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, insel = "Coaxial"; break; default: - insel = "Unkown"; + insel = "Unknown"; } snd_iprintf(buffer, diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 6159929..4d8db36 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -763,7 +763,7 @@ static int cs42l52_set_sysclk(struct snd_soc_dai *codec_dai, if ((freq >= CS42L52_MIN_CLK) && (freq <= CS42L52_MAX_CLK)) { cs42l52->sysclk = freq; } else { - dev_err(codec->dev, "Invalid freq paramter\n"); + dev_err(codec->dev, "Invalid freq parameter\n"); return -EINVAL; } return 0; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3fddc7a..b2b2b37 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3722,7 +3722,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) } while (count--); if (count == 0) - dev_warn(codec->dev, "No impedence range reported for jack\n"); + dev_warn(codec->dev, "No impedance range reported for jack\n"); #ifndef CONFIG_SND_SOC_WM8994_MODULE trace_snd_soc_jack_irq(dev_name(codec->dev)); -- cgit v0.10.2 From f0b3da98434589a5665d70041f8e1a5600b84fe8 Mon Sep 17 00:00:00 2001 From: "Lars R. Damerow" Date: Fri, 2 Nov 2012 13:10:39 -0700 Subject: ALSA: hda - support Teradici 2200 host card audio The audio chipset used in Teradici's Tera2 host cards is the same as that in the 1200 host cards. This patch allows ALSA to recognize the Tera2 cards. Signed-off-by: Lars R. Damerow Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 72b085a..cd2dbaf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3563,6 +3563,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA | AZX_DCAPS_NO_64BIT }, + { PCI_DEVICE(0x6549, 0x2200), + .driver_data = AZX_DRIVER_TERA | AZX_DCAPS_NO_64BIT }, /* Creative X-Fi (CA0110-IBG) */ /* CTHDA chips */ { PCI_DEVICE(0x1102, 0x0010), -- cgit v0.10.2 From 5a83b4b5a391f07141b157ac9daa51c409e71ab5 Mon Sep 17 00:00:00 2001 From: Alexander Stein Date: Thu, 1 Nov 2012 13:42:37 +0100 Subject: ALSA: hda: Cirrus: Fix coefficient index for beep configuration Signed-off-by: Alexander Stein Cc: Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 61a7113..3b7d67a 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1107,7 +1107,7 @@ static const struct hda_verb cs_coef_init_verbs[] = { | 0x0400 /* Disable Coefficient Auto increment */ )}, /* Beep */ - {0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG}, + {0x11, AC_VERB_SET_COEF_INDEX, IDX_BEEP_CFG}, {0x11, AC_VERB_SET_PROC_COEF, 0x0007}, /* Enable Beep thru DAC1/2/3 */ {} /* terminator */ -- cgit v0.10.2 From 16337e028a6dae9fbdd718c0d42161540a668ff3 Mon Sep 17 00:00:00 2001 From: Daniel J Blueman Date: Sun, 4 Nov 2012 13:19:03 +0800 Subject: ALSA: HDA: Fix digital microphone on CS420x Correctly enable the digital microphones with the right bits in the right coeffecient registers on Cirrus CS4206/7 codecs. It also prevents misconfiguring ADC1/2. This fixes the digital mic on the Macbook Pro 10,1/Retina. Based-on-patch-by: Alexander Stein Signed-off-by: Daniel J Blueman Cc: Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 3b7d67a..859a119 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -101,8 +101,8 @@ enum { #define CS420X_VENDOR_NID 0x11 #define CS_DIG_OUT1_PIN_NID 0x10 #define CS_DIG_OUT2_PIN_NID 0x15 -#define CS_DMIC1_PIN_NID 0x12 -#define CS_DMIC2_PIN_NID 0x0e +#define CS_DMIC1_PIN_NID 0x0e +#define CS_DMIC2_PIN_NID 0x12 /* coef indices */ #define IDX_SPDIF_STAT 0x0000 @@ -1079,14 +1079,18 @@ static void init_input(struct hda_codec *codec) cs_automic(codec, NULL); coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */ + cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + + coef = cs_vendor_coef_get(codec, IDX_BEEP_CFG); if (is_active_pin(codec, CS_DMIC2_PIN_NID)) - coef |= 0x0500; /* DMIC2 2 chan on, GPIO1 off */ + coef |= 1 << 4; /* DMIC2 2 chan on, GPIO1 off */ if (is_active_pin(codec, CS_DMIC1_PIN_NID)) - coef |= 0x1800; /* DMIC1 2 chan on, GPIO0 off + coef |= 1 << 3; /* DMIC1 2 chan on, GPIO0 off * No effect if SPDIF_OUT2 is * selected in IDX_SPDIF_CTL. */ - cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + + cs_vendor_coef_set(codec, IDX_BEEP_CFG, coef); } else { if (spec->mic_detect) cs_automic(codec, NULL); -- cgit v0.10.2 From 00e17f767e3e8d42b83a12af3ed16e3129e4feb0 Mon Sep 17 00:00:00 2001 From: Daniel J Blueman Date: Sun, 4 Nov 2012 13:19:04 +0800 Subject: ALSA: HDA: Mark CS260x immutable structures const Mark structures that won't change const. Signed-off-by: Daniel J Blueman Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 859a119..d5f3a26 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1732,8 +1732,7 @@ static int cs421x_mux_enum_put(struct snd_kcontrol *kcontrol, } -static struct snd_kcontrol_new cs421x_capture_source = { - +static const struct snd_kcontrol_new cs421x_capture_source = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -1950,7 +1949,7 @@ static int cs421x_suspend(struct hda_codec *codec) } #endif -static struct hda_codec_ops cs421x_patch_ops = { +static const struct hda_codec_ops cs421x_patch_ops = { .build_controls = cs421x_build_controls, .build_pcms = cs_build_pcms, .init = cs421x_init, -- cgit v0.10.2 From 5c0ee9497b33cde3e57460efe4f73313dc0b57a3 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Sun, 4 Nov 2012 23:34:58 +0100 Subject: ALSA: es1968: Add ESS vendor ID to pm_whitelist Add generic ESS vendor ID to pm_whitelist. This should fix suspend on all Maestro-2 and Maestro-2E based PCI cards. Tested on Terratec DMX and SF64-PCE2. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 5d0e568..50169bc 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2655,6 +2655,8 @@ static struct ess_device_list pm_whitelist[] __devinitdata = { { TYPE_MAESTRO2E, 0x1179 }, { TYPE_MAESTRO2E, 0x14c0 }, /* HP omnibook 4150 */ { TYPE_MAESTRO2E, 0x1558 }, + { TYPE_MAESTRO2E, 0x125d }, /* a PCI card, e.g. Terratec DMX */ + { TYPE_MAESTRO2, 0x125d }, /* a PCI card, e.g. SF64-PCE2 */ }; static struct ess_device_list mpu_blacklist[] __devinitdata = { -- cgit v0.10.2 From ae24c3191ba2ab03ec6b4be323e730e00404b4b6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 5 Nov 2012 12:32:46 +0100 Subject: ALSA: hda - Force to reset IEC958 status bits for AD codecs Several bug reports suggest that the forcibly resetting IEC958 status bits is required for AD codecs to get the SPDIF output working properly after changing streams. Original fix credit to Javeed Shaikh. BugLink: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/359361 Reported-by: Robin Kreis Cc: Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index cdd43ea..1eeba73 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -545,6 +545,7 @@ static int ad198x_build_pcms(struct hda_codec *codec) if (spec->multiout.dig_out_nid) { info++; codec->num_pcms++; + codec->spdif_status_reset = 1; info->name = "AD198x Digital"; info->pcm_type = HDA_PCM_TYPE_SPDIF; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback; -- cgit v0.10.2 From 5b3761954dac2d1393beef8210eb8cee81d16b8d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Nov 2012 10:32:47 +0100 Subject: ALSA: hda - Fix empty DAC filling in patch_via.c In via_auto_fill_adc_nids(), the parser tries to fill dac_nids[] at the point of the current line-out (i). When no valid path is found for this output, this results in dac = 0, thus it creates a hole in dac_nids[]. This confuses is_empty_dac() and trims the detected DAC in later reference. This patch fixes the bug by appending DAC properly to dac_nids[] in via_auto_fill_adc_nids(). Reported-by: Massimo Del Fedele Cc: Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 72a2f60..bf57fa6 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1809,11 +1809,11 @@ static int via_auto_fill_dac_nids(struct hda_codec *codec) { struct via_spec *spec = codec->spec; const struct auto_pin_cfg *cfg = &spec->autocfg; - int i, dac_num; + int i; hda_nid_t nid; + spec->multiout.num_dacs = 0; spec->multiout.dac_nids = spec->private_dac_nids; - dac_num = 0; for (i = 0; i < cfg->line_outs; i++) { hda_nid_t dac = 0; nid = cfg->line_out_pins[i]; @@ -1824,16 +1824,13 @@ static int via_auto_fill_dac_nids(struct hda_codec *codec) if (!i && parse_output_path(codec, nid, dac, 1, &spec->out_mix_path)) dac = spec->out_mix_path.path[0]; - if (dac) { - spec->private_dac_nids[i] = dac; - dac_num++; - } + if (dac) + spec->private_dac_nids[spec->multiout.num_dacs++] = dac; } if (!spec->out_path[0].depth && spec->out_mix_path.depth) { spec->out_path[0] = spec->out_mix_path; spec->out_mix_path.depth = 0; } - spec->multiout.num_dacs = dac_num; return 0; } -- cgit v0.10.2 From ef4da45828603df57e5e21b8aa21a66ce309f79b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Nov 2012 10:37:48 +0100 Subject: ALSA: hda - Fix invalid connections in VT1802 codec VT1802 codec provides the invalid connection lists of NID 0x24 and 0x33 containing the routes to a non-exist widget 0x3e. This confuses the auto-parser. Fix it up in the driver by overriding these connections. Reported-by: Massimo Del Fedele Cc: Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index bf57fa6..c2eef5c 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3646,6 +3646,18 @@ static const struct snd_pci_quirk vt2002p_fixups[] = { {} }; +/* NIDs 0x24 and 0x33 on VT1802 have connections to non-existing NID 0x3e + * Replace this with mixer NID 0x1c + */ +static void fix_vt1802_connections(struct hda_codec *codec) +{ + static hda_nid_t conn_24[] = { 0x14, 0x1c }; + static hda_nid_t conn_33[] = { 0x1c }; + + snd_hda_override_conn_list(codec, 0x24, ARRAY_SIZE(conn_24), conn_24); + snd_hda_override_conn_list(codec, 0x33, ARRAY_SIZE(conn_33), conn_33); +} + /* patch for vt2002P */ static int patch_vt2002P(struct hda_codec *codec) { @@ -3660,6 +3672,8 @@ static int patch_vt2002P(struct hda_codec *codec) spec->aa_mix_nid = 0x21; override_mic_boost(codec, 0x2b, 0, 3, 40); override_mic_boost(codec, 0x29, 0, 3, 40); + if (spec->codec_type == VT1802) + fix_vt1802_connections(codec); add_secret_dac_path(codec); snd_hda_pick_fixup(codec, NULL, vt2002p_fixups, via_fixups); -- cgit v0.10.2 From d5266125fb439a5dfa4edd548d888fda47414ac5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Nov 2012 10:40:36 +0100 Subject: ALSA: hda - Add pin fixups for ASUS G75 To parse properly the subwoofer outputs on ASUS G75 laptop with VT1802 codec, correct the default configurations of speaker pins 0x24 and 0x33. Reported-by: Massimo Del Fedele Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c2eef5c..019e1a0 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3625,6 +3625,7 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) */ enum { VIA_FIXUP_INTMIC_BOOST, + VIA_FIXUP_ASUS_G75, }; static void via_fixup_intmic_boost(struct hda_codec *codec, @@ -3639,9 +3640,19 @@ static const struct hda_fixup via_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = via_fixup_intmic_boost, }, + [VIA_FIXUP_ASUS_G75] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* set 0x24 and 0x33 as speakers */ + { 0x24, 0x991301f0 }, + { 0x33, 0x991301f1 }, /* subwoofer */ + { } + } + }, }; static const struct snd_pci_quirk vt2002p_fixups[] = { + SND_PCI_QUIRK(0x1043, 0x1487, "Asus G75", VIA_FIXUP_ASUS_G75), SND_PCI_QUIRK(0x1043, 0x8532, "Asus X202E", VIA_FIXUP_INTMIC_BOOST), {} }; -- cgit v0.10.2 From d1a3c98d50731c627909029bb653a0557946f0f5 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Wed, 7 Nov 2012 18:00:09 +0100 Subject: ALSA: hdspm - Fix sync check reporting on RME RayDAT The RayDAT reports the sync status of its inputs in consecutive bit positions, so all we do in hdspm_s1_sync_check is to iterate over idx: status = hdspm_read(hdspm, HDSPM_RD_STATUS_1); lock = (status & (0x1<private_value: HDSPM_SYNC_CHECK("WC SyncCheck", 0), HDSPM_SYNC_CHECK("AES SyncCheck", 1), HDSPM_SYNC_CHECK("SPDIF SyncCheck", 2), HDSPM_SYNC_CHECK("ADAT1 SyncCheck", 3), HDSPM_SYNC_CHECK("ADAT2 SyncCheck", 4), HDSPM_SYNC_CHECK("ADAT3 SyncCheck", 5), HDSPM_SYNC_CHECK("ADAT4 SyncCheck", 6), HDSPM_SYNC_CHECK("TCO SyncCheck", 7), HDSPM_SYNC_CHECK("SYNC IN SyncCheck", 8), The patch corrects the indicated sync flags by passing the proper index value to hdspm_s1_sync_check(). Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 9a8d5ce..748e36c 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3979,7 +3979,8 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol, case 8: /* SYNC IN */ val = hdspm_sync_in_sync_check(hdspm); break; default: - val = hdspm_s1_sync_check(hdspm, ucontrol->id.index-1); + val = hdspm_s1_sync_check(hdspm, + kcontrol->private_value-1); } break; -- cgit v0.10.2 From f58161ba1b05a968e5136824b5a16b714b6a5317 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Nov 2012 08:52:45 +0100 Subject: ALSA: usb-audio: Fix crash at re-preparing the PCM stream There are bug reports of a crash with USB-audio devices when PCM prepare is performed immediately after the stream is stopped via trigger callback. It turned out that the problem is that we don't wait until all URBs are killed. This patch adds a new function to synchronize the pending stop operation on an endpoint, and calls in the prepare callback for avoiding the crash above. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181 Reported-and-tested-by: Artem S. Tashkinov Cc: [v3.6] Signed-off-by: Takashi Iwai diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 7f78c6d..34de6f2 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -35,6 +35,7 @@ #define EP_FLAG_ACTIVATED 0 #define EP_FLAG_RUNNING 1 +#define EP_FLAG_STOPPING 2 /* * snd_usb_endpoint is a model that abstracts everything related to an @@ -502,10 +503,20 @@ static int wait_clear_urbs(struct snd_usb_endpoint *ep) if (alive) snd_printk(KERN_ERR "timeout: still %d active urbs on EP #%x\n", alive, ep->ep_num); + clear_bit(EP_FLAG_STOPPING, &ep->flags); return 0; } +/* sync the pending stop operation; + * this function itself doesn't trigger the stop operation + */ +void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep) +{ + if (ep && test_bit(EP_FLAG_STOPPING, &ep->flags)) + wait_clear_urbs(ep); +} + /* * unlink active urbs. */ @@ -918,6 +929,8 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, if (wait) wait_clear_urbs(ep); + else + set_bit(EP_FLAG_STOPPING, &ep->flags); } } diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 6376ccf..3d4c970 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -19,6 +19,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep); void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, int force, int can_sleep, int wait); +void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep); int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct list_head *head); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 37428f7..5c12a3f 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -568,6 +568,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } + snd_usb_endpoint_sync_pending_stop(subs->sync_endpoint); + snd_usb_endpoint_sync_pending_stop(subs->data_endpoint); + ret = set_format(subs, subs->cur_audiofmt); if (ret < 0) goto unlock; -- cgit v0.10.2 From 1387e2d12799e554df2f60e7ae7fe01384bcb96f Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 8 Nov 2012 10:23:18 +0100 Subject: ALSA: hda - Improve HP depop when system enter to S3 alc269_toggle_power_output() was only use in ALC269VB. I rename it to alc269vb_toggle_power_output(). Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f7397ad..b25e9b2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5840,7 +5840,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc269_ignore, ssids); } -static void alc269_toggle_power_output(struct hda_codec *codec, int power_up) +static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { int val = alc_read_coef_idx(codec, 0x04); if (power_up) @@ -5857,10 +5857,10 @@ static void alc269_shutup(struct hda_codec *codec) if (spec->codec_variant != ALC269_TYPE_ALC269VB) return; - if ((alc_get_coef0(codec) & 0x00ff) == 0x017) - alc269_toggle_power_output(codec, 0); - if ((alc_get_coef0(codec) & 0x00ff) == 0x018) { - alc269_toggle_power_output(codec, 0); + if (spec->codec_variant == ALC269_TYPE_ALC269VB) + alc269vb_toggle_power_output(codec, 0); + if (spec->codec_variant == ALC269_TYPE_ALC269VB && + (alc_get_coef0(codec) & 0x00ff) == 0x018) { msleep(150); } } @@ -5870,24 +5870,22 @@ static int alc269_resume(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (spec->codec_variant == ALC269_TYPE_ALC269VB || + if (spec->codec_variant == ALC269_TYPE_ALC269VB) + alc269vb_toggle_power_output(codec, 0); + if (spec->codec_variant == ALC269_TYPE_ALC269VB && (alc_get_coef0(codec) & 0x00ff) == 0x018) { - alc269_toggle_power_output(codec, 0); msleep(150); } codec->patch_ops.init(codec); - if (spec->codec_variant == ALC269_TYPE_ALC269VB || + if (spec->codec_variant == ALC269_TYPE_ALC269VB) + alc269vb_toggle_power_output(codec, 1); + if (spec->codec_variant == ALC269_TYPE_ALC269VB && (alc_get_coef0(codec) & 0x00ff) == 0x017) { - alc269_toggle_power_output(codec, 1); msleep(200); } - if (spec->codec_variant == ALC269_TYPE_ALC269VB || - (alc_get_coef0(codec) & 0x00ff) == 0x018) - alc269_toggle_power_output(codec, 1); - snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); hda_call_check_power_status(codec, 0x01); -- cgit v0.10.2 From 19a62823eae453619604636082085812c14ee391 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 8 Nov 2012 10:25:37 +0100 Subject: ALSA: hda - Add new codec ALC668 and ALC900 (default name ALC1150) Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b25e9b2..c0ce3b1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7077,6 +7077,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { .patch = patch_alc662 }, { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, + { .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 }, { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, { .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, @@ -7094,6 +7095,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 }, { .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 }, { .id = 0x10ec0899, .name = "ALC898", .patch = patch_alc882 }, + { .id = 0x10ec0900, .name = "ALC1150", .patch = patch_alc882 }, {} /* terminator */ }; -- cgit v0.10.2 From 8bb4d9ce08b0a92ca174e41d92c180328f86173f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Nov 2012 14:36:18 +0100 Subject: ALSA: Fix card refcount unbalance There are uncovered cases whether the card refcount introduced by the commit a0830dbd isn't properly increased or decreased: - OSS PCM and mixer success paths - When lookup function gets NULL This patch fixes these places. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=50251 Cc: Signed-off-by: Takashi Iwai diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index a9a2e63..e8a1d18 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -76,6 +76,7 @@ static int snd_mixer_oss_open(struct inode *inode, struct file *file) snd_card_unref(card); return -EFAULT; } + snd_card_unref(card); return 0; } diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index f337b66..4c1cc51 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2454,6 +2454,7 @@ static int snd_pcm_oss_open(struct inode *inode, struct file *file) mutex_unlock(&pcm->open_mutex); if (err < 0) goto __error; + snd_card_unref(pcm->card); return err; __error: diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 6e8872d..f9ddecf 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2122,7 +2122,8 @@ static int snd_pcm_playback_open(struct inode *inode, struct file *file) pcm = snd_lookup_minor_data(iminor(inode), SNDRV_DEVICE_TYPE_PCM_PLAYBACK); err = snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_PLAYBACK); - snd_card_unref(pcm->card); + if (pcm) + snd_card_unref(pcm->card); return err; } @@ -2135,7 +2136,8 @@ static int snd_pcm_capture_open(struct inode *inode, struct file *file) pcm = snd_lookup_minor_data(iminor(inode), SNDRV_DEVICE_TYPE_PCM_CAPTURE); err = snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_CAPTURE); - snd_card_unref(pcm->card); + if (pcm) + snd_card_unref(pcm->card); return err; } diff --git a/sound/core/sound.c b/sound/core/sound.c index 89780c3..70ccdab 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -114,7 +114,7 @@ void *snd_lookup_minor_data(unsigned int minor, int type) mreg = snd_minors[minor]; if (mreg && mreg->type == type) { private_data = mreg->private_data; - if (mreg->card_ptr) + if (private_data && mreg->card_ptr) atomic_inc(&mreg->card_ptr->refcount); } else private_data = NULL; diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index e1d79ee..726a49a 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -54,7 +54,7 @@ void *snd_lookup_oss_minor_data(unsigned int minor, int type) mreg = snd_oss_minors[minor]; if (mreg && mreg->type == type) { private_data = mreg->private_data; - if (mreg->card_ptr) + if (private_data && mreg->card_ptr) atomic_inc(&mreg->card_ptr->refcount); } else private_data = NULL; -- cgit v0.10.2