From bae35992afc7ca2f446108b16670d76de0624e7a Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 3 Sep 2013 15:14:12 -0300 Subject: ASoC: fsl: imx-audmux: Do not call imx_audmux_parse_dt_defaults() on non-dt kernel Booting a mx51babbage board with a non-dt kernel leads to the following crash: Unable to handle kernel NULL pointer dereference at virtual address 0000001c pgd = 80004000 [0000001c] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.11.0-next-20130903 #287 task: 9f860000 ti: 9f862000 task.ti: 9f862000 PC is at of_get_next_available_child+0x5c/0x68 LR is at of_get_next_available_child+0x1c/0x68 pc : [<8043ea58>] lr : [<8043ea18>] psr: 60000193 sp : 9f863d58 ip : 00000000 fp : 9f863d74 r10: 9f89a010 r9 : 9f862000 r8 : 807bb26c r7 : 80615d5c r6 : 00000000 r5 : 60000113 r4 : 00000000 r3 : 00000000 r2 : 808770a4 r1 : 00000011 r0 : 60000113 Flags: nZCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c5387d Table: 90004019 DAC: 00000017 Process swapper/0 (pid: 1, stack limit = 0x9f862240) Stack: (0x9f863d58 to 0x9f864000) This is caused by commit 8548a464b9 (ASoC: imx-audmux: Read default configuration from devicetree). In order to fix this, add a check for 'of_id' so that imx_audmux_parse_dt_defaults() only gets called when a dt kernel is running. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmux.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index ab17381cc981..d3bf71a0ec56 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -335,7 +335,8 @@ static int imx_audmux_probe(struct platform_device *pdev) if (audmux_type == IMX31_AUDMUX) audmux_debugfs_init(); - imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); + if (of_id) + imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); return 0; } -- cgit v1.2.3 From e011143454606de70eba1db5d99454eec0017fdb Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Wed, 4 Sep 2013 15:27:46 +0800 Subject: ASoC: atmel: disable error interrupt As once the error interrupt is triggered, it can not be cleared. So, disable it. No side effect found while testing on sama5d3xek and at91sam9x5ek boards. Signed-off-by: Bo Shen Acked-by: Nicolas Ferre Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 0ecf356027f6..bb53dea85b17 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -649,7 +649,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, dma_params = ssc_p->dma_params[dir]; ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); - ssc_writel(ssc_p->ssc->regs, IER, dma_params->mask->ssc_error); + ssc_writel(ssc_p->ssc->regs, IDR, dma_params->mask->ssc_error); pr_debug("%s enabled SSC_SR=0x%08x\n", dir ? "receive" : "transmit", -- cgit v1.2.3 From d098b2f0cf6280f2b7b0415b48921385fdc1861f Mon Sep 17 00:00:00 2001 From: Thomas Petazzoni Date: Thu, 5 Sep 2013 14:56:31 +0200 Subject: ASoC: kirkwood: change the compatible string of the kirkwood-i2s driver The compatible string of the kirkwood-i2s driver was chosen as "marvell,mvebu-audio". Using such a compatible string is not a good idea, since "mvebu" is the name of a large family of SOCs, in which new, unknown SOCs will be coming in the future. It is therefore impossible to know what will be evolutions of this hardware block in the next generations of the SOCs. For this reason, the recommandation for compatible strings of on-SOCs devices has always been to use the name of the oldest SOC that has the hardware block. New SOCs that have an exactly compatible hardware block can reference it using the same compatible string. See [1], [2] and [3] for various cases were this suggestion was made, including from Rob Herring, a Device Tree binding maintainer. As an example, there are already small differences between current generations: * On Kirkwood, only one interrupt is used for audio. * On Dove, two interrupts are used, one for audio data and one for error reporting. In the near future, I'll be adding audio support to Armada 370, which allows has the same hardware block (but maybe with minor variants). Therefore, this patch changes the driver to accept "marvell,kirkwood-audio" and "marvell,dove-audio" as compatible strings instead of the too-generic "marvell,mvebu-audio". The reason for the two different compatible strings is the difference in the number of interrupts used by the two SOCs for audio. This Device Tree binding has never been part of a Linux kernel stable release so far, so it can be changed now without breaking backward compatibility. [1] http://lists.infradead.org/pipermail/linux-mtd/2012-March/040417.html [2] http://lists.infradead.org/pipermail/linux-arm-kernel/2013-April/161065.html [3] http://lists.infradead.org/pipermail/linux-arm-kernel/2012-March/087702.html Signed-off-by: Thomas Petazzoni Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/mvebu-audio.txt | 12 ++++++++---- sound/soc/kirkwood/kirkwood-i2s.c | 3 ++- 2 files changed, 10 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt index 7e5fd37c1b3f..f0062c5871b4 100644 --- a/Documentation/devicetree/bindings/sound/mvebu-audio.txt +++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt @@ -2,13 +2,17 @@ Required properties: -- compatible: "marvell,mvebu-audio" +- compatible: + "marvell,kirkwood-audio" for Kirkwood platforms + "marvell,dove-audio" for Dove platforms - reg: physical base address of the controller and length of memory mapped region. -- interrupts: list of two irq numbers. - The first irq is used for data flow and the second one is used for errors. +- interrupts: + with "marvell,kirkwood-audio", the audio interrupt + with "marvell,dove-audio", a list of two interrupts, the first for + the data flow, and the second for errors. - clocks: one or two phandles. The first one is mandatory and defines the internal clock. @@ -21,7 +25,7 @@ Required properties: Example: i2s1: audio-controller@b4000 { - compatible = "marvell,mvebu-audio"; + compatible = "marvell,dove-audio"; reg = <0xb4000 0x2210>; interrupts = <21>, <22>; clocks = <&gate_clk 13>; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 7fce340ab3ef..0f3d73d4ef48 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -559,7 +559,8 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) #ifdef CONFIG_OF static struct of_device_id mvebu_audio_of_match[] = { - { .compatible = "marvell,mvebu-audio" }, + { .compatible = "marvell,kirkwood-audio" }, + { .compatible = "marvell,dove-audio" }, { } }; MODULE_DEVICE_TABLE(of, mvebu_audio_of_match); -- cgit v1.2.3 From 7bba2157c5d3ee7f076adfdfa96eec274801da8f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Sep 2013 15:45:38 +0200 Subject: ALSA: hda - Add dock speaker support for ASUS TX300 ASUS TX300 has a built-in speaker in the tablet part and in the dock part, and the tablet speaker is supposed to be unused while the machine is docked. The current HD-audio driver, however, doesn't support the dock speaker, partly because BIOS doesn't set up the pin for the corresponding output. But, not only the missing pin config, also the missing unsol event handling is another issue. Otherwise the automatic switching via dock/undock won't work. Through debugging sessions, we found out that the dock speaker pin is NID 0x1b, and it generates an unsol event at docking/undocking, the docking state can be inquired via the normal pin detection verb. Also, it's turned out that GPIO 2 is needed as an amp. So, all materials are ready to cook. This patch provides the basic dock speaker support with TX300: - The dock speaker is turned on/off via "Dock Speaker" mixer mute. - The dock speaker is automatically muted when docked. This is independently from the mixer mute switch, just like the headphone auto-mute function. The implementation is a bit tricky. Since we want to handle it as a secondary speaker, we set it up a pin as a speaker with a jack detection. Then, the fixup function registers the own unsol callback for this pin because the standard automute can't handle the thing like a "speaker jack". In the own automute hook, we apply the mute of the tablet speaker in addition by checking the dock state. Also, the speaker control names are slightly shuffled because the generic parser doesn't give good names but blindly assumes a bass speaker as a secondary speaker. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59791 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 56 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 56 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4a909170b59e..d38d6a04eb67 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3443,6 +3443,56 @@ static void alc283_fixup_chromebook(struct hda_codec *codec, } } +/* mute tablet speaker pin (0x14) via dock plugging in addition */ +static void asus_tx300_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + snd_hda_gen_update_outputs(codec); + if (snd_hda_jack_detect(codec, 0x1b)) + spec->gen.mute_bits |= (1ULL << 0x14); +} + +static void alc282_fixup_asus_tx300(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + /* TX300 needs to set up GPIO2 for the speaker amp */ + static const struct hda_verb gpio2_verbs[] = { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x04 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 }, + { 0x01, AC_VERB_SET_GPIO_DATA, 0x04 }, + {} + }; + static const struct hda_pintbl dock_pins[] = { + { 0x1b, 0x21114000 }, /* dock speaker pin */ + {} + }; + struct snd_kcontrol *kctl; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_add_verbs(codec, gpio2_verbs); + snd_hda_apply_pincfgs(codec, dock_pins); + spec->gen.auto_mute_via_amp = 1; + spec->gen.automute_hook = asus_tx300_automute; + snd_hda_jack_detect_enable_callback(codec, 0x1b, + HDA_GEN_HP_EVENT, + snd_hda_gen_hp_automute); + break; + case HDA_FIXUP_ACT_BUILD: + /* this is a bit tricky; give more sane names for the main + * (tablet) speaker and the dock speaker, respectively + */ + kctl = snd_hda_find_mixer_ctl(codec, "Speaker Playback Switch"); + if (kctl) + strcpy(kctl->id.name, "Dock Speaker Playback Switch"); + kctl = snd_hda_find_mixer_ctl(codec, "Bass Speaker Playback Switch"); + if (kctl) + strcpy(kctl->id.name, "Speaker Playback Switch"); + break; + } +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -3480,6 +3530,7 @@ enum { ALC269_FIXUP_LIMIT_INT_MIC_BOOST, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, + ALC282_FIXUP_ASUS_TX300, }; static const struct hda_fixup alc269_fixups[] = { @@ -3735,6 +3786,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc283_fixup_chromebook, }, + [ALC282_FIXUP_ASUS_TX300] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc282_fixup_asus_tx300, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3784,6 +3839,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), -- cgit v1.2.3 From fb87fa3a79823c15e70d4fc7653bde59f8556ca1 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Wed, 4 Sep 2013 16:36:57 -0400 Subject: ALSA: hda - define is_haswell() to check if a display audio codec is Haswell To apply Haswell specific fixings, this patch defines is_haswell() to check whether a display audio codec is Haswell, to avoid explicitly checking Haswell vendor ID everywhere. Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 22b50899b151..009bcccbee7a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -44,6 +44,8 @@ static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); +#define is_haswell(codec) ((codec)->vendor_id == 0x80862807) + struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -1087,7 +1089,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, int pinctl; int new_pinctl = 0; - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) haswell_verify_pin_D0(codec, cvt_nid, pin_nid); if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { @@ -1227,7 +1229,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); /* configure unused pins to choose other converters */ - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) haswell_config_cvts(codec, pin_idx, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); @@ -1358,7 +1360,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) /* Haswell-specific workaround: re-setup when the transcoder is * changed during the stream playback */ - if (codec->vendor_id == 0x80862807 && + if (is_haswell(codec) && eld->eld_valid && !old_eld_valid && per_pin->setup) { snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -1405,7 +1407,7 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) return 0; - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) intel_haswell_fixup_connect_list(codec, pin_nid); pin_idx = spec->num_pins; @@ -2011,7 +2013,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->spec = spec; hdmi_array_init(spec, 4); - if (codec->vendor_id == 0x80862807) { + if (is_haswell(codec)) { intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); } @@ -2022,7 +2024,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; - if (codec->vendor_id == 0x80862807) { + if (is_haswell(codec)) { codec->patch_ops.set_power_state = haswell_set_power_state; codec->dp_mst = true; } -- cgit v1.2.3 From 58f7d28da6994e4292ade2ac9eabebb723a9bbe6 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Wed, 4 Sep 2013 16:37:12 -0400 Subject: ALSA: hda - unmute pin amplifier in infoframe setup for Haswell When Gfx driver reconnects a port and transcoder, the pin amplifier will be muted. To enable sound, the pin amp need to be unmuted. This patch - moves pin amp unmuting from stream preparing to hdmi_setup_audio_infoframe(). So if port:transcoder reconnection happens during stream playback, the ELDV unsol event can stil trigger pin's amp unmuting when re-setting up audio info frame. - remove reading pin amp status before unmuting for speed-up, since pin amp should always be unmuted. - rename haswell_verify_pin_D0() to haswell_verify_D0(), since the convertor power state is also fixed here. This patch is mostly based on suggestion of David Henningsson. Cc: David Henningsson Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 36 +++++++++--------------------------- 1 file changed, 9 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 009bcccbee7a..3a15225d6982 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -896,6 +896,11 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (!channels) return; + if (is_haswell(codec)) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + eld = &per_pin->sink_eld; if (!eld->monitor_present) return; @@ -1035,10 +1040,10 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) hdmi_non_intrinsic_event(codec, res); } -static void haswell_verify_pin_D0(struct hda_codec *codec, +static void haswell_verify_D0(struct hda_codec *codec, hda_nid_t cvt_nid, hda_nid_t nid) { - int pwr, lamp, ramp; + int pwr; /* For Haswell, the converter 1/2 may keep in D3 state after bootup, * thus pins could only choose converter 0 for use. Make sure the @@ -1054,25 +1059,6 @@ static void haswell_verify_pin_D0(struct hda_codec *codec, pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; snd_printd("Haswell HDMI audio: Power for pin 0x%x is now D%d\n", nid, pwr); } - - lamp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT); - ramp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT); - if (lamp != ramp) { - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_OUTPUT | lamp); - - lamp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT); - ramp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT); - snd_printd("Haswell HDMI audio: Mute after set on pin 0x%x: [0x%x 0x%x]\n", nid, lamp, ramp); - } } /* @@ -1090,7 +1076,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, int new_pinctl = 0; if (is_haswell(codec)) - haswell_verify_pin_D0(codec, cvt_nid, pin_nid); + haswell_verify_D0(codec, cvt_nid, pin_nid); if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { pinctl = snd_hda_codec_read(codec, pin_nid, 0, @@ -1361,13 +1347,9 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) * changed during the stream playback */ if (is_haswell(codec) && - eld->eld_valid && !old_eld_valid && per_pin->setup) { - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); + eld->eld_valid && !old_eld_valid && per_pin->setup) hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); - } } mutex_unlock(&pin_eld->lock); -- cgit v1.2.3 From 4345adf92db760ca1a54061ce284aaa2e7d0791e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 8 Sep 2013 17:20:37 +0100 Subject: ASoC: fsl_spdif: Select regmap-mmio The S/PDIF driver needs regmap so select it to make sure it gets included in the build. Reported-by: Fengguang Wu Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 704e246f5b1e..b7ab71f2ccc1 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -198,6 +198,7 @@ config SND_SOC_IMX_SPDIF select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF select SND_SOC_SPDIF + select REGMAP_MMIO help SoC Audio support for i.MX boards with S/PDIF Say Y if you want to add support for SoC audio on an i.MX board with -- cgit v1.2.3 From 83f72151352791836a1b9c1542614cc9bf71ac61 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Sep 2013 10:20:48 +0200 Subject: ALSA: hda - Add Toshiba Satellite C870 to MSI blacklist Toshiba Satellite C870 shows interrupt problems occasionally when certain mixer controls like "Mic Switch" is toggled. This seems worked around by not using MSI. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=833585 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c6c98298ac39..7f88dd6aff6e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3400,6 +3400,7 @@ static struct snd_pci_quirk msi_black_list[] = { SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */ + SND_PCI_QUIRK(0x1179, 0xfb44, "Toshiba Satellite C870", 0), /* AMD Hudson */ SND_PCI_QUIRK(0x1849, 0x0888, "ASRock", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0xa0a0, 0x0575, "Aopen MZ915-M", 0), /* ICH6 */ {} -- cgit v1.2.3 From be8cf44526d8972c2dbf6e561162dad924a712a5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Sep 2013 13:57:57 +0200 Subject: ALSA: hda - Add CS4208 codec support for MacBook 6,1 and 6,2 MacBook 6,1 and 6,2 have a CS4208 codec instead of CS4206/CS4207 on the former models. Most of functions work fine as is, except for the silent speaker output. After debugging sessions, it turned out that the machine needs to set GPIO 0 for the speaker amp. This patch adds the basic support for CS4208 and the fixup for these MacBooks. Basically the codec works just with the generic parser. For re-using the existing GPIO amp code and init/free callbacks, a few places have been changed so that CS4206/4207-specific codes (errata, etc) won't hit with CS4208. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=60811 Reported-and-tested-by: Imre Kaloz Reported-and-tested-by: Ian Munsie Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 89 ++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 82 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index cccaf9c7a7bb..b524f89a1f13 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -169,7 +169,7 @@ static void cs_automute(struct hda_codec *codec) snd_hda_gen_update_outputs(codec); - if (spec->gpio_eapd_hp) { + if (spec->gpio_eapd_hp || spec->gpio_eapd_speaker) { spec->gpio_data = spec->gen.hp_jack_present ? spec->gpio_eapd_hp : spec->gpio_eapd_speaker; snd_hda_codec_write(codec, 0x01, 0, @@ -291,10 +291,11 @@ static int cs_init(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; - /* init_verb sequence for C0/C1/C2 errata*/ - snd_hda_sequence_write(codec, cs_errata_init_verbs); - - snd_hda_sequence_write(codec, cs_coef_init_verbs); + if (spec->vendor_nid == CS420X_VENDOR_NID) { + /* init_verb sequence for C0/C1/C2 errata*/ + snd_hda_sequence_write(codec, cs_errata_init_verbs); + snd_hda_sequence_write(codec, cs_coef_init_verbs); + } snd_hda_gen_init(codec); @@ -307,8 +308,10 @@ static int cs_init(struct hda_codec *codec) spec->gpio_data); } - init_input_coef(codec); - init_digital_coef(codec); + if (spec->vendor_nid == CS420X_VENDOR_NID) { + init_input_coef(codec); + init_digital_coef(codec); + } return 0; } @@ -551,6 +554,76 @@ static int patch_cs420x(struct hda_codec *codec) return err; } +/* + * CS4208 support: + * Its layout is no longer compatible with CS4206/CS4207, and the generic + * parser seems working fairly well, except for trivial fixups. + */ +enum { + CS4208_GPIO0, +}; + +static const struct hda_model_fixup cs4208_models[] = { + { .id = CS4208_GPIO0, .name = "gpio0" }, + {} +}; + +static const struct snd_pci_quirk cs4208_fixup_tbl[] = { + /* codec SSID */ + SND_PCI_QUIRK(0x106b, 0x7100, "MacBookPro 6,1", CS4208_GPIO0), + SND_PCI_QUIRK(0x106b, 0x7200, "MacBookPro 6,2", CS4208_GPIO0), + {} /* terminator */ +}; + +static void cs4208_fixup_gpio0(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct cs_spec *spec = codec->spec; + spec->gpio_eapd_hp = 0; + spec->gpio_eapd_speaker = 1; + spec->gpio_mask = spec->gpio_dir = + spec->gpio_eapd_hp | spec->gpio_eapd_speaker; + } +} + +static const struct hda_fixup cs4208_fixups[] = { + [CS4208_GPIO0] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs4208_fixup_gpio0, + }, +}; + +static int patch_cs4208(struct hda_codec *codec) +{ + struct cs_spec *spec; + int err; + + spec = cs_alloc_spec(codec, 0); /* no specific w/a */ + if (!spec) + return -ENOMEM; + + spec->gen.automute_hook = cs_automute; + + snd_hda_pick_fixup(codec, cs4208_models, cs4208_fixup_tbl, + cs4208_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + + err = cs_parse_auto_config(codec); + if (err < 0) + goto error; + + codec->patch_ops = cs_patch_ops; + + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + + return 0; + + error: + cs_free(codec); + return err; +} + /* * Cirrus Logic CS4210 * @@ -991,6 +1064,7 @@ static int patch_cs4213(struct hda_codec *codec) static const struct hda_codec_preset snd_hda_preset_cirrus[] = { { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x }, { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x }, + { .id = 0x10134208, .name = "CS4208", .patch = patch_cs4208 }, { .id = 0x10134210, .name = "CS4210", .patch = patch_cs4210 }, { .id = 0x10134213, .name = "CS4213", .patch = patch_cs4213 }, {} /* terminator */ @@ -998,6 +1072,7 @@ static const struct hda_codec_preset snd_hda_preset_cirrus[] = { MODULE_ALIAS("snd-hda-codec-id:10134206"); MODULE_ALIAS("snd-hda-codec-id:10134207"); +MODULE_ALIAS("snd-hda-codec-id:10134208"); MODULE_ALIAS("snd-hda-codec-id:10134210"); MODULE_ALIAS("snd-hda-codec-id:10134213"); -- cgit v1.2.3 From 34e4447515a18e0602f6df1a08b6a6ea63dea14b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 8 Sep 2013 21:21:41 -0700 Subject: ASoC: rsnd: fixup flag name of rsnd_scu_platform_info it should be *USE*, not *USB* Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 2 +- sound/soc/sh/rcar/scu.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index d35412ae03b3..fe66533e9b7a 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -55,7 +55,7 @@ struct rsnd_ssi_platform_info { /* * flags */ -#define RSND_SCU_USB_HPBIF (1 << 31) /* it needs RSND_SSI_DEPENDENT */ +#define RSND_SCU_USE_HPBIF (1 << 31) /* it needs RSND_SSI_DEPENDENT */ struct rsnd_scu_platform_info { u32 flags; diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 184d9008cecd..2df2e9150b89 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -157,9 +157,9 @@ static int rsnd_scu_start(struct rsnd_mod *mod, int ret; /* - * SCU will be used if it has RSND_SCU_USB_HPBIF flags + * SCU will be used if it has RSND_SCU_USE_HPBIF flags */ - if (!(flags & RSND_SCU_USB_HPBIF)) { + if (!(flags & RSND_SCU_USE_HPBIF)) { /* it use PIO transter */ dev_dbg(dev, "%s%d is not used\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); -- cgit v1.2.3 From 9f6f0afbb9fdabf6dcac642dfec457f28981e3f8 Mon Sep 17 00:00:00 2001 From: Steffen Trumtrar Date: Mon, 9 Sep 2013 18:09:12 +0200 Subject: ASoC: mc13783: add spi errata fix The MC13783 Chip Errata, Rev. 4 says, that depending on SPI clock and main audio clock speed, the Audio Codec or Stereo DAC do sometimes not start when programmed to do so. This is due to an internal clock timing issue related to the loading of the SPI bits into the audio block. On an i.MX27 based system, this issue lead to switched audio channels under certain circumstances: RTC + Touch + Audio are used and loaded at startup. The mentioned workaround of writing registers 40 and 41 two times is implemented here. Signed-off-by: Steffen Trumtrar Cc: stable@vger.kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 4d3c8fd8c5db..ea141e1d6f28 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -125,6 +125,10 @@ static int mc13783_write(struct snd_soc_codec *codec, ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + /* include errata fix for spi audio problems */ + if (reg == MC13783_AUDIO_CODEC || reg == MC13783_AUDIO_DAC) + ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + mc13xxx_unlock(priv->mc13xxx); return ret; -- cgit v1.2.3