From 95491d902b4ed1bfd8f602aada793d74cc85428b Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 8 Nov 2009 19:03:55 -0500 Subject: ALSA: hda: Use model=auto quirk for Sony VAIO VGN-FW170J using ALC262 BugLink: https://bugs.launchpad.net/bugs/478309 The internal microphone on this VAIO model does not work unless the "auto" quirk is used. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ff20048504b..9bb4f75ca43 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11460,6 +11460,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */ SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06), + SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO), SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", -- cgit v1.2.3 From dbaccc0cca830efe9bb3c9e4a1cfcd6503790079 Mon Sep 17 00:00:00 2001 From: Daniel Drake Date: Mon, 9 Nov 2009 15:17:24 +0000 Subject: ALSA: hda - Tweak OLPC XO-1.5 microphone bias Our contacts at Conexant suggested that we reduce the external microphone bias to 50% in order to center the input signal with the DC input range of the codec. This is because the microphone port is DC coupled for potential use with sensors. Signed-off-by: Daniel Drake Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6479e65858d..905859d4f4d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -110,6 +110,7 @@ struct conexant_spec { unsigned int dell_automute; unsigned int port_d_mode; + unsigned char ext_mic_bias; }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -1927,6 +1928,11 @@ static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 }; static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 }; #define CXT5066_SPDIF_OUT 0x21 +/* OLPC's microphone port is DC coupled for use with external sensors, + * therefore we use a 50% mic bias in order to center the input signal with + * the DC input range of the codec. */ +#define CXT5066_OLPC_EXT_MIC_BIAS PIN_VREF50 + static struct hda_channel_mode cxt5066_modes[1] = { { 2, NULL }, }; @@ -1980,9 +1986,10 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, /* toggle input of built-in and mic jack appropriately */ static void cxt5066_automic(struct hda_codec *codec) { - static struct hda_verb ext_mic_present[] = { + struct conexant_spec *spec = codec->spec; + struct hda_verb ext_mic_present[] = { /* enable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias}, /* switch to external mic input */ {0x17, AC_VERB_SET_CONNECT_SEL, 0}, @@ -2235,7 +2242,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = { {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ /* Port B: external microphone */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, CXT5066_OLPC_EXT_MIC_BIAS}, /* Port C: internal microphone */ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, @@ -2353,6 +2360,7 @@ static int patch_cxt5066(struct hda_codec *codec) spec->input_mux = &cxt5066_capture_source; spec->port_d_mode = PIN_HP; + spec->ext_mic_bias = PIN_VREF80; spec->num_init_verbs = 1; spec->init_verbs[0] = cxt5066_init_verbs; @@ -2384,6 +2392,7 @@ static int patch_cxt5066(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->port_d_mode = 0; + spec->ext_mic_bias = CXT5066_OLPC_EXT_MIC_BIAS; /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; -- cgit v1.2.3 From 4ac55982907e1d48e64feaa56be91b9b52d3714d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Nov 2009 16:08:45 +0100 Subject: ALSA: hda - Avoid quirk for HP dc5750 The present quirk for HP dc5750 seems broken and maps the pins wrongly. Since the auto-parser works well for this device, set the default entry to use model=auto. Reference: Novell bnc#552154 https://bugzilla.novell.com/show_bug.cgi?id=552154 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9bb4f75ca43..d1ccb6eaf9f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6249,7 +6249,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER), SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100), SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013), - SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013), + SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_AUTO), /* no quirk */ SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600), -- cgit v1.2.3 From 71121d9fcc494453b9311992de220abb47dde3f1 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Tue, 10 Nov 2009 20:11:55 +0100 Subject: ALSA: hda - possible read past array alc88[02]_parse_auto_config() The test of index `i' is after the read - too late - and unsafe: if snd_hda_get_connections() fails in the last iteration a read beyond the array is possible. Signed-off-by: Roel Kluin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d1ccb6eaf9f..daf6975b0c2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4684,9 +4684,9 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->multiout.dig_out_nid = dig_nid; else { spec->multiout.slave_dig_outs = spec->slave_dig_outs; - spec->slave_dig_outs[i - 1] = dig_nid; - if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1) + if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1) break; + spec->slave_dig_outs[i - 1] = dig_nid; } } if (spec->autocfg.dig_in_pin) @@ -9813,9 +9813,9 @@ static int alc882_parse_auto_config(struct hda_codec *codec) spec->multiout.dig_out_nid = dig_nid; else { spec->multiout.slave_dig_outs = spec->slave_dig_outs; - spec->slave_dig_outs[i - 1] = dig_nid; - if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1) + if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1) break; + spec->slave_dig_outs[i - 1] = dig_nid; } } if (spec->autocfg.dig_in_pin) -- cgit v1.2.3 From 46ef6ec9da420b298b1f197e445bf5b06fe01ef4 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Wed, 11 Nov 2009 14:32:10 -0500 Subject: ALSA: hda: Use model=mb5 for MacBookPro 5,2 BugLink: https://bugs.launchpad.net/bugs/462098 Until we can look closer at the verbs, let's use ALC885_MB5 for codec SSID 0x106b4600 to enable playback and capture for MacBookPro 5,2s. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index daf6975b0c2..84a52efdb2d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8911,10 +8911,11 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3), SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24), SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5), - /* FIXME: HP jack sense seems not working for MBP 5,1, so apparently - * no perfect solution yet + /* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2, + * so apparently no perfect solution yet */ SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), + SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), {} /* terminator */ }; -- cgit v1.2.3 From e2e527ae7fb07caa58f8fa8fa7e90ada0b175dd7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Nov 2009 08:28:03 +0100 Subject: ALSA: hda - Add another Nvidia HDMI codec id (10de:0005) Found on Nvidia 9800M GTS. Reported-by: Chris Balcum Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_nvhdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 9fb60276f5c..6afdab09bab 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -397,6 +397,7 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec) static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { { .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, { .id = 0x10de0003, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, + { .id = 0x10de0005, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, { .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, { .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi_8ch }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, @@ -406,6 +407,7 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { MODULE_ALIAS("snd-hda-codec-id:10de0002"); MODULE_ALIAS("snd-hda-codec-id:10de0003"); +MODULE_ALIAS("snd-hda-codec-id:10de0005"); MODULE_ALIAS("snd-hda-codec-id:10de0006"); MODULE_ALIAS("snd-hda-codec-id:10de0007"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); -- cgit v1.2.3 From 8ef5837a47f73faee18fa7ce2f9a9eb7675be8de Mon Sep 17 00:00:00 2001 From: Daniel J Blueman Date: Sat, 14 Nov 2009 18:20:04 +0000 Subject: ALSA: hda - Dell Studio 1557 hd-audio quirk Add the Dell Studio 15 (model 1557, Core i7) laptop to the hd-audio quirk list, enabling audio. Signed-off-by: Daniel J Blueman Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8eb6508cd99..86de305fc9f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1590,6 +1590,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { "Dell Studio 17", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02be, "Dell Studio 1555", STAC_DELL_M6_DMIC), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd, + "Dell Studio 1557", STAC_DELL_M6_DMIC), {} /* terminator */ }; -- cgit v1.2.3 From bf97402052483c125a9ea7bf13df0dd9b4134078 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 16 Nov 2009 11:07:17 +0200 Subject: ALSA: ice1724 - make some bitfields unsigned This is a clean up and doesn't change the behavior. Bit fields should always be unsigned. Otherwise pm_suspend_enabled will be -1 when you want it to be 1. The other bad thing is that the sparse checker will complain 36 times if they aren't unsigned. The other bitfields in that struct are unsigned already. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 9da2dae64c5..d063149e704 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -382,8 +382,8 @@ struct snd_ice1712 { #ifdef CONFIG_PM int (*pm_suspend)(struct snd_ice1712 *); int (*pm_resume)(struct snd_ice1712 *); - int pm_suspend_enabled:1; - int pm_saved_is_spdif_master:1; + unsigned int pm_suspend_enabled:1; + unsigned int pm_saved_is_spdif_master:1; unsigned int pm_saved_spdif_ctrl; unsigned char pm_saved_spdif_cfg; unsigned int pm_saved_route; -- cgit v1.2.3 From 0c3cee57efcb1c79d62b1238c0d22afef4599247 Mon Sep 17 00:00:00 2001 From: Javier Kohen Date: Tue, 17 Nov 2009 15:36:13 +0100 Subject: ALSA: usb - Quirk to disable master volume control in PCM2702 Disable the master volume control in the PCM2702 chipset. The datasheet documents two independent channel volume controls, one master mute control and one master volume control. All controls are fully functional except for the master volume control, which returns USB stalls on all GET requests. Signed-off-by: Javier Kohen Signed-off-by: Takashi Iwai --- sound/usb/usbmixer.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 9efcfd08d74..c998220b99c 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -1071,6 +1071,15 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsig channels = (ftr[0] - 7) / csize - 1; master_bits = snd_usb_combine_bytes(ftr + 6, csize); + /* master configuration quirks */ + switch (state->chip->usb_id) { + case USB_ID(0x08bb, 0x2702): + snd_printk(KERN_INFO + "usbmixer: master volume quirk for PCM2702 chip\n"); + /* disable non-functional volume control */ + master_bits &= ~(1 << (USB_FEATURE_VOLUME - 1)); + break; + } if (channels > 0) first_ch_bits = snd_usb_combine_bytes(ftr + 6 + csize, csize); else -- cgit v1.2.3 From 12929baea4b29d70525f764034b3dac771dd69e5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Nov 2009 15:58:35 +0100 Subject: ALSA: hda - Fix quirk for VAIO type G Vaio type G laptop doesn't work with the current quirk setup. After some tests, it turned out that it should be model=auto as default. Reported-by: Mattia Dongili Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 84a52efdb2d..70583719282 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11462,6 +11462,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */ SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06), SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO), + SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO), SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", -- cgit v1.2.3 From bd6ddcb41d5fbdcbc1486f48d8023f234b4a7f8d Mon Sep 17 00:00:00 2001 From: Anuj Aggarwal Date: Tue, 17 Nov 2009 21:43:42 +0530 Subject: ASoC: Modifying the license string GPLv2 for OMAP3 EVM Correcting the license string from GPLv2 -> GPL v2. Found the problem while building OMAP3 ASoC driver as module. Signed-off-by: Anuj Aggarwal Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/omap3evm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 9114c263077..13aa380de16 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -144,4 +144,4 @@ module_exit(omap3evm_soc_exit); MODULE_AUTHOR("Anuj Aggarwal "); MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM"); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From f3dd70414cdc0203ca63eef83ca130c2d1903b30 Mon Sep 17 00:00:00 2001 From: Grazvydas Ignotas Date: Sat, 7 Nov 2009 23:16:12 +0200 Subject: ASoC: OMAP3 Pandora: update for TWL4030 codec changes A while ago TWL4030 had it's playback stream name changed, but pandora needs it for it's playback path. Update to correct stream name so that playback works again. Also mark VIBRA output as not connected. Signed-off-by: Grazvydas Ignotas Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/omap3pandora.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index ad219aaf7cb..0cd06f5dd35 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -134,7 +134,7 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, * |P| <--- TWL4030 <--------- Line In and MICs */ static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { - SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("PCM DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM, 0, 0, NULL, 0, omap3pandora_hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -181,6 +181,7 @@ static int omap3pandora_out_init(struct snd_soc_codec *codec) snd_soc_dapm_nc_pin(codec, "CARKITR"); snd_soc_dapm_nc_pin(codec, "HFL"); snd_soc_dapm_nc_pin(codec, "HFR"); + snd_soc_dapm_nc_pin(codec, "VIBRA"); ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets, ARRAY_SIZE(omap3pandora_out_dapm_widgets)); -- cgit v1.2.3 From bab0212467e58929470ae3ae32515f17e30c3926 Mon Sep 17 00:00:00 2001 From: Troy Kisky Date: Tue, 17 Nov 2009 13:51:01 -0700 Subject: ASoC: tlv320aic23 fix rate selection Fix the ordering of sr_valid_mask array. The lower bit of the index represents USB not bosr. Reported-by: Anuj Aggarwal Signed-off-by: Troy Kisky Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 0b8dcb5cd72..6b24d8bb02b 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -265,8 +265,8 @@ static const int bosr_usb_divisor_table[] = { #define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15)) static const unsigned short sr_valid_mask[] = { LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/ - LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/ LOWER_GROUP, /* Usb, bosr - 0*/ + LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/ UPPER_GROUP, /* Usb, bosr - 1*/ }; /* -- cgit v1.2.3 From 50b6bce59d154b5db137907a5c0ed45a4e7a3829 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Nov 2009 13:11:53 +0000 Subject: ASoC: Fix suspend with active audio streams When we get a stream suspend event force the power down since otherwise the stream would remain marked as active. In future we'll probably want to make this stream-specific and add an interface to make the power down of other widgets optional in order to support leaving bypass paths active while suspending the processor. Cc: stable@kernel.org Reported-by: Joonyoung Shim Tested-by: Joonyoung Shim Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d89f6dc0090..66d4c165f99 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -973,9 +973,19 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) if (!w->power_check) continue; - power = w->power_check(w); - if (power) - sys_power = 1; + /* If we're suspending then pull down all the + * power. */ + switch (event) { + case SND_SOC_DAPM_STREAM_SUSPEND: + power = 0; + break; + + default: + power = w->power_check(w); + if (power) + sys_power = 1; + break; + } if (w->power == power) continue; @@ -999,8 +1009,12 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) case SND_SOC_DAPM_STREAM_RESUME: sys_power = 1; break; + case SND_SOC_DAPM_STREAM_SUSPEND: + sys_power = 0; + break; case SND_SOC_DAPM_STREAM_NOP: sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY; + break; default: break; } -- cgit v1.2.3 From e9ff5eb2ae018fe2298c68746c873bf828c6b10e Mon Sep 17 00:00:00 2001 From: Anuj Aggarwal Date: Fri, 27 Nov 2009 17:40:58 +0530 Subject: ASoC: AIC23: Fixing infinite loop in resume path This patch fixes two issues: a) Infinite loop in resume function b) Writes to non-existing registers in resume function Cc: stable@kernel.org Signed-off-by: Anuj Aggarwal Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 6b24d8bb02b..90a0264f753 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -625,11 +625,10 @@ static int tlv320aic23_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - int i; u16 reg; /* Sync reg_cache with the hardware */ - for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) { + for (reg = 0; reg < TLV320AIC23_RESET; reg++) { u16 val = tlv320aic23_read_reg_cache(codec, reg); tlv320aic23_write(codec, reg, val); } -- cgit v1.2.3 From 4acd57c3de62374fe5bb52e5cd24538190f4eab2 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sun, 29 Nov 2009 16:39:52 +0000 Subject: ALSA: AACI: fix AC97 multiple-open bug Signed-off-by: Russell King Cc: Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 1f0f8213e2d..1cb7c282a1f 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -504,6 +504,10 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, int err; aaci_pcm_hw_free(substream); + if (aacirun->pcm_open) { + snd_ac97_pcm_close(aacirun->pcm); + aacirun->pcm_open = 0; + } err = devdma_hw_alloc(NULL, substream, params_buffer_bytes(params)); -- cgit v1.2.3 From 8ee763b9c82c6ca0a59a7271ce4fa29d7baf5c09 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sun, 29 Nov 2009 16:39:59 +0000 Subject: ALSA: AACI: fix recording bug pcm->r[1].slots is the double rate slot information, not the capture information. For capture, 'pcm' will already be the capture ac97 pcm structure. Signed-off-by: Russell King Cc: Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 1cb7c282a1f..6c160a038b2 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -521,7 +521,7 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, else err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), params_channels(params), - aacirun->pcm->r[1].slots); + aacirun->pcm->r[0].slots); if (err) goto out; -- cgit v1.2.3