From ec8f53fb693dda095ad3342b927a074e7c4dddfa Mon Sep 17 00:00:00 2001 From: Masanari Iida Date: Fri, 2 Nov 2012 00:28:50 +0900 Subject: [PATCH 01/15] 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 --- sound/i2c/other/ak4113.c | 2 +- sound/i2c/other/ak4114.c | 2 +- sound/i2c/other/ak4117.c | 2 +- sound/pci/rme9652/hdspm.c | 2 +- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c index ef68d710d08cf..e04e750a77ed4 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 816e7d225fb0a..5bf4fca19e486 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 b4b2a51fc117a..40e33c9f2b095 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 f1cd1e387801b..9a8d5cef32c7d 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 61599298fb26c..4d8db3685e961 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 3fddc7ad1127e..b2b2b37131bdd 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)); From f0b3da98434589a5665d70041f8e1a5600b84fe8 Mon Sep 17 00:00:00 2001 From: "Lars R. Damerow" Date: Fri, 2 Nov 2012 13:10:39 -0700 Subject: [PATCH 02/15] 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 --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 72b085ae7d469..cd2dbaf1be786 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), From 5a83b4b5a391f07141b157ac9daa51c409e71ab5 Mon Sep 17 00:00:00 2001 From: Alexander Stein Date: Thu, 1 Nov 2012 13:42:37 +0100 Subject: [PATCH 03/15] ALSA: hda: Cirrus: Fix coefficient index for beep configuration Signed-off-by: Alexander Stein Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 61a71131711c7..3b7d67af14416 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 */ From 16337e028a6dae9fbdd718c0d42161540a668ff3 Mon Sep 17 00:00:00 2001 From: Daniel J Blueman Date: Sun, 4 Nov 2012 13:19:03 +0800 Subject: [PATCH 04/15] 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 --- sound/pci/hda/patch_cirrus.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 3b7d67af14416..859a1197e0801 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); From 00e17f767e3e8d42b83a12af3ed16e3129e4feb0 Mon Sep 17 00:00:00 2001 From: Daniel J Blueman Date: Sun, 4 Nov 2012 13:19:04 +0800 Subject: [PATCH 05/15] 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 --- sound/pci/hda/patch_cirrus.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 859a1197e0801..d5f3a26d608db 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, From 5c0ee9497b33cde3e57460efe4f73313dc0b57a3 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Sun, 4 Nov 2012 23:34:58 +0100 Subject: [PATCH 06/15] 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 --- sound/pci/es1968.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 5d0e568fdea1b..50169bcfd9037 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 = { From ae24c3191ba2ab03ec6b4be323e730e00404b4b6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 5 Nov 2012 12:32:46 +0100 Subject: [PATCH 07/15] 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 --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index cdd43eadbc674..1eeba73866663 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; From 5b3761954dac2d1393beef8210eb8cee81d16b8d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Nov 2012 10:32:47 +0100 Subject: [PATCH 08/15] 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 --- sound/pci/hda/patch_via.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 72a2f60b087c8..bf57fa6a4add9 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; } From ef4da45828603df57e5e21b8aa21a66ce309f79b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Nov 2012 10:37:48 +0100 Subject: [PATCH 09/15] 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 --- sound/pci/hda/patch_via.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index bf57fa6a4add9..c2eef5cb78d86 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); From d5266125fb439a5dfa4edd548d888fda47414ac5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Nov 2012 10:40:36 +0100 Subject: [PATCH 10/15] 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 --- sound/pci/hda/patch_via.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c2eef5cb78d86..019e1a00414a4 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), {} }; From d1a3c98d50731c627909029bb653a0557946f0f5 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Wed, 7 Nov 2012 18:00:09 +0100 Subject: [PATCH 11/15] 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 --- sound/pci/rme9652/hdspm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 9a8d5cef32c7d..748e36c66603a 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; From f58161ba1b05a968e5136824b5a16b714b6a5317 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Nov 2012 08:52:45 +0100 Subject: [PATCH 12/15] 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 --- sound/usb/endpoint.c | 13 +++++++++++++ sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 3 +++ 3 files changed, 17 insertions(+) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 7f78c6d782b07..34de6f2faf612 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 6376ccf10fd47..3d4c9705041ff 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 37428f74dbb69..5c12a3fe8c3e8 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; From 1387e2d12799e554df2f60e7ae7fe01384bcb96f Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 8 Nov 2012 10:23:18 +0100 Subject: [PATCH 13/15] 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 --- sound/pci/hda/patch_realtek.c | 24 +++++++++++------------- 1 file changed, 11 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f7397ad02a0de..b25e9b22cd696 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); From 19a62823eae453619604636082085812c14ee391 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 8 Nov 2012 10:25:37 +0100 Subject: [PATCH 14/15] ALSA: hda - Add new codec ALC668 and ALC900 (default name ALC1150) Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b25e9b22cd696..c0ce3b1f04b4a 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 */ }; From 8bb4d9ce08b0a92ca174e41d92c180328f86173f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Nov 2012 14:36:18 +0100 Subject: [PATCH 15/15] 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 --- sound/core/oss/mixer_oss.c | 1 + sound/core/oss/pcm_oss.c | 1 + sound/core/pcm_native.c | 6 ++++-- sound/core/sound.c | 2 +- sound/core/sound_oss.c | 2 +- 5 files changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index a9a2e63c0222c..e8a1d18774b20 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 f337b66a020b5..4c1cc51772e6f 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 6e8872de5ba05..f9ddecf2f4cd7 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 89780c323f19f..70ccdab741532 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 e1d79ee359065..726a49ac97253 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;