From a9d14bc0b188a822e42787d01e56c06fe9750162 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 2 Oct 2013 17:49:50 +0200 Subject: ALSA: snd-usb-usx2y: remove bogus frame checks The frame check in i_usX2Y_urb_complete() and i_usX2Y_usbpcm_urb_complete() is bogus and produces false positives as described in this LAU thread: http://linuxaudio.org/mailarchive/lau/2013/5/20/200177 This patch removes the check code entirely. Cc: fzu@wemgehoertderstaat.de Reported-by: Dr Nicholas J Bailey Suggested-by: Takashi Iwai Signed-off-by: Daniel Mack Cc: Signed-off-by: Takashi Iwai --- sound/usb/usx2y/usbusx2yaudio.c | 22 +++------------------- sound/usb/usx2y/usx2yhwdeppcm.c | 7 +------ 2 files changed, 4 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 63fb5219f0f8..6234a51625b1 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -299,19 +299,6 @@ static void usX2Y_error_urb_status(struct usX2Ydev *usX2Y, usX2Y_clients_stop(usX2Y); } -static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, - struct snd_usX2Y_substream *subs, struct urb *urb) -{ - snd_printk(KERN_ERR -"Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" -"Most probably some urb of usb-frame %i is still missing.\n" -"Cause could be too long delays in usb-hcd interrupt handling.\n", - usb_get_current_frame_number(usX2Y->dev), - subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", - usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame); - usX2Y_clients_stop(usX2Y); -} - static void i_usX2Y_urb_complete(struct urb *urb) { struct snd_usX2Y_substream *subs = urb->context; @@ -328,12 +315,9 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + + subs->completed_urb = urb; + { struct snd_usX2Y_substream *capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE], *playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index f2a1acdc4d83..814d0e887c62 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -244,13 +244,8 @@ static void i_usX2Y_usbpcm_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + subs->completed_urb = urb; capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE]; capsubs2 = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE + 2]; playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; -- cgit v1.2.3 From db8d3af33f7f6e1388a65e847f90bbc8d1ba66ce Mon Sep 17 00:00:00 2001 From: Guenter Roeck Date: Wed, 2 Oct 2013 21:15:22 -0700 Subject: ASoC: fsl_ssi: Fix irq_of_parse_and_map() return value check irq_of_parse_and_map() returns 0 on error, not NO_IRQ. Fix the following xtensa:allmodconfig build error. sound/soc/fsl/fsl_ssi.c:705:26: error: 'NO_IRQ' undeclared (first use in this function) make[4]: *** [sound/soc/fsl/fsl_ssi.o] Error 1 Cc: Geert Uytterhoeven Cc: Grant Likely Signed-off-by: Guenter Roeck Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c6b743978d5e..6b81d0ce2c44 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -936,7 +936,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) ssi_private->ssi_phys = res.start; ssi_private->irq = irq_of_parse_and_map(np, 0); - if (ssi_private->irq == NO_IRQ) { + if (ssi_private->irq == 0) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); return -ENXIO; } -- cgit v1.2.3 From 1d73ad298d1bfeee5d77c19e5cd667c551e30632 Mon Sep 17 00:00:00 2001 From: Philippe Rétornaz Date: Tue, 1 Oct 2013 14:36:11 +0200 Subject: ASoC: fsl: Fix sound on mx31moboard MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Commit 42810d (ASoC: imx-mc13783: Add audmux settings for mx27pdk) broke the sound on mx31moboard. Restore back the audmux setting on such boards. Signed-off-by: Philippe Rétornaz Signed-off-by: Mark Brown --- sound/soc/fsl/imx-mc13783.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index a3d60d4bea4c..a2fd7321b5a9 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -112,7 +112,7 @@ static int imx_mc13783_probe(struct platform_device *pdev) return ret; } - if (machine_is_mx31_3ds()) { + if (machine_is_mx31_3ds() || machine_is_mx31moboard()) { imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4, IMX_AUDMUX_V2_PTCR_SYN, IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) | -- cgit v1.2.3 From 2a577a7569182cc9a7fb0c91b3a5d031839806c8 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Tue, 1 Oct 2013 21:20:50 +0300 Subject: ASoC: omap: Fix incorrect ARM dependency Commit b0e0a4d ("ASoC: omap: Enable COMPILE_TEST build for DT platforms") added two incorrect CONFIG_ARCH_ARM dependencies making impossible to select audio support for Nokia RX-51. Fix this by using correct CONFIG_ARM. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index daa78a0095fa..4a07f7179690 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,6 +1,6 @@ config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" - depends on (ARCH_OMAP && DMA_OMAP) || (ARCH_ARM && COMPILE_TEST) + depends on (ARCH_OMAP && DMA_OMAP) || (ARM && COMPILE_TEST) select SND_DMAENGINE_PCM config SND_OMAP_SOC_DMIC @@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_RX51 tristate "SoC Audio support for Nokia RX-51" - depends on SND_OMAP_SOC && ARCH_ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) + depends on SND_OMAP_SOC && ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 -- cgit v1.2.3 From 338cae565c53755de9f87d6a801517940d2d56f7 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 7 Oct 2013 10:39:59 +0200 Subject: ALSA: hda - Fix mono speakers and headset mic on Dell Vostro 5470 On this machine, DAC on node 0x03 seems to give mono output. Also, it needs additional patches for headset mic support. It supports CTIA style headsets only. Alsa-info available at the bug link below. Cc: stable@kernel.org (v3.10+) BugLink: https://bugs.launchpad.net/bugs/1236228 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0e303b99a47c..52c26d3a61d4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3496,6 +3496,15 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec, } } +static void alc290_fixup_mono_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + /* Remove DAC node 0x03, as it seems to be + giving mono output */ + snd_hda_override_wcaps(codec, 0x03, 0); +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -3522,6 +3531,7 @@ enum { ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, + ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, ALC269_FIXUP_HEADSET_MODE, ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC269_FIXUP_ASUS_X101_FUNC, @@ -3535,6 +3545,7 @@ enum { ALC283_FIXUP_CHROME_BOOK, ALC282_FIXUP_ASUS_TX300, ALC283_FIXUP_INT_MIC, + ALC290_FIXUP_MONO_SPEAKERS, }; static const struct hda_fixup alc269_fixups[] = { @@ -3712,6 +3723,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC269_FIXUP_DELL3_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, [ALC269_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode, @@ -3804,6 +3824,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, + [ALC290_FIXUP_MONO_SPEAKERS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc290_fixup_mono_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3845,6 +3871,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- cgit v1.2.3 From 5a6e19bedb13522924f5ee72c1f65b0fb5d33bc0 Mon Sep 17 00:00:00 2001 From: Philippe Rétornaz Date: Tue, 1 Oct 2013 14:36:10 +0200 Subject: ASoC: fsl: imx-ssi: fix probe on imx31 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit On imx31 with mc13783 codec the FIQ is not necessary and not enabled as DMA transfer is available. Change the probe() function to fail only if both FIQ and DMA are not available. Signed-off-by: Philippe Rétornaz Signed-off-by: Mark Brown --- sound/soc/fsl/imx-ssi.c | 23 ++++++++++++----------- sound/soc/fsl/imx-ssi.h | 2 ++ 2 files changed, 14 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index f58bcd85c07f..57d6941676ff 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -600,19 +600,17 @@ static int imx_ssi_probe(struct platform_device *pdev) ssi->fiq_params.dma_params_rx = &ssi->dma_params_rx; ssi->fiq_params.dma_params_tx = &ssi->dma_params_tx; - ret = imx_pcm_fiq_init(pdev, &ssi->fiq_params); - if (ret) - goto failed_pcm_fiq; + ssi->fiq_init = imx_pcm_fiq_init(pdev, &ssi->fiq_params); + ssi->dma_init = imx_pcm_dma_init(pdev); - ret = imx_pcm_dma_init(pdev); - if (ret) - goto failed_pcm_dma; + if (ssi->fiq_init && ssi->dma_init) { + ret = ssi->fiq_init; + goto failed_pcm; + } return 0; -failed_pcm_dma: - imx_pcm_fiq_exit(pdev); -failed_pcm_fiq: +failed_pcm: snd_soc_unregister_component(&pdev->dev); failed_register: release_mem_region(res->start, resource_size(res)); @@ -628,8 +626,11 @@ static int imx_ssi_remove(struct platform_device *pdev) struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); struct imx_ssi *ssi = platform_get_drvdata(pdev); - imx_pcm_dma_exit(pdev); - imx_pcm_fiq_exit(pdev); + if (!ssi->dma_init) + imx_pcm_dma_exit(pdev); + + if (!ssi->fiq_init) + imx_pcm_fiq_exit(pdev); snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index fb1616ba8c59..560c40fc9ebb 100644 --- a/sound/soc/fsl/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h @@ -211,6 +211,8 @@ struct imx_ssi { struct imx_dma_data filter_data_rx; struct imx_pcm_fiq_params fiq_params; + int fiq_init; + int dma_init; int enabled; }; -- cgit v1.2.3 From 6b2afee11a05dfc2bb75bc6bb709c61130df37ae Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 7 Oct 2013 11:59:19 +0300 Subject: ASoC: tlv320aic3x: Connect 'Left Line1R Mux' and 'Right Line1L Mux' The two paths were not connected in the DAPM route causing the associated routes to be non working and the following warnings printed in the logs: tlv320aic3x-codec 0-001b: ASoC: mux Right Line1L Mux has no paths tlv320aic3x-codec 0-001b: ASoC: mux Left Line1R Mux has no paths Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6e3f269243e0..64ad84d8a306 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -674,6 +674,8 @@ static const struct snd_soc_dapm_route intercon[] = { /* Left Input */ {"Left Line1L Mux", "single-ended", "LINE1L"}, {"Left Line1L Mux", "differential", "LINE1L"}, + {"Left Line1R Mux", "single-ended", "LINE1R"}, + {"Left Line1R Mux", "differential", "LINE1R"}, {"Left Line2L Mux", "single-ended", "LINE2L"}, {"Left Line2L Mux", "differential", "LINE2L"}, @@ -690,6 +692,8 @@ static const struct snd_soc_dapm_route intercon[] = { /* Right Input */ {"Right Line1R Mux", "single-ended", "LINE1R"}, {"Right Line1R Mux", "differential", "LINE1R"}, + {"Right Line1L Mux", "single-ended", "LINE1L"}, + {"Right Line1L Mux", "differential", "LINE1L"}, {"Right Line2R Mux", "single-ended", "LINE2R"}, {"Right Line2R Mux", "differential", "LINE2R"}, -- cgit v1.2.3 From 39edac70e9aedf451fccaa851b273ace9fcca0bd Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Mon, 7 Oct 2013 19:24:52 +0300 Subject: ALSA: hda - hdmi: Fix channel map switch not taking effect Currently hdmi_setup_audio_infoframe() reprograms the HDA channel mapping only when the infoframe is not up-to-date or the non-PCM flag has changed. However, when just the channel map has been changed, the infoframe may still be up-to-date and non-PCM flag may not have changed, so the new channel map is not actually programmed into the HDA codec. Notably, this failing case is also always triggered when the device is already in a prepared state and a new channel map is configured while changing only the channel positions (for example, plain "speaker-test -c2 -m FR,FL"). Fix that by always programming the channel map in hdmi_setup_audio_infoframe(). Tested on Intel HDMI. Signed-off-by: Anssi Hannula Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 18 ++++++++---------- 1 file changed, 8 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7ea0245fc6bd..50173d412ac5 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -936,6 +936,14 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, return; } + /* + * always configure channel mapping, it may have been changed by the + * user in the meantime + */ + hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, + channels, per_pin->chmap, + per_pin->chmap_set); + /* * sizeof(ai) is used instead of sizeof(*hdmi_ai) or * sizeof(*dp_ai) to avoid partial match/update problems when @@ -947,20 +955,10 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, "pin=%d channels=%d\n", pin_nid, channels); - hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, - channels, per_pin->chmap, - per_pin->chmap_set); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, ai.bytes, sizeof(ai)); hdmi_start_infoframe_trans(codec, pin_nid); - } else { - /* For non-pcm audio switch, setup new channel mapping - * accordingly */ - if (per_pin->non_pcm != non_pcm) - hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, - channels, per_pin->chmap, - per_pin->chmap_set); } per_pin->non_pcm = non_pcm; -- cgit v1.2.3 From c6cc3d58b4042f5cadae653ff8d3df26af1a0169 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Oct 2013 19:57:50 +0200 Subject: ALSA: hda - Add fixup for ASUS N56VZ ASUS N56VZ needs a fixup for the bass speaker pin, which was already provided via model=asus-mode4. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=841645 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 52c26d3a61d4..ed9deb66f593 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4596,6 +4596,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_ASUS_MODE4), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), -- cgit v1.2.3 From 88cfcf86aa3ada84d97195bcad74f4dadb4ae23b Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 11 Oct 2013 10:18:45 +0200 Subject: ALSA: hda - Fix microphone for Sony VAIO Pro 13 (Haswell model) The external mic showed up with a precense detect of "always present", essentially disabling the internal mic. Therefore turn off presence detection for this pin. Note: The external mic seems not yet working, but an internal mic is certainly better than no mic at all. Cc: stable@vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1227093 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ed9deb66f593..ae847fe006c8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3528,6 +3528,7 @@ enum { ALC269_FIXUP_HP_GPIO_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, + ALC286_FIXUP_SONY_MIC_NO_PRESENCE, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, @@ -3740,6 +3741,13 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_no_hp_mic, }, + [ALC286_FIXUP_SONY_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -3894,6 +3902,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101), + SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), -- cgit v1.2.3 From 7c478f03372ad2cf434fde62082895bfcb6e6e89 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 11 Oct 2013 10:18:46 +0200 Subject: ALSA: hda - Add a headset mic model for ALC269 and friends Using the headset mic model will cause the headset mic to be labeled "headset mic" instead of just "mic". Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 15 +++++++++++++++ 2 files changed, 16 insertions(+) (limited to 'sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index f911e3656209..85c362d8ea34 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -28,6 +28,7 @@ ALC269/270/275/276/28x/29x alc269-dmic Enable ALC269(VA) digital mic workaround alc271-dmic Enable ALC271X digital mic workaround inv-dmic Inverted internal mic workaround + headset-mic Indicates a combined headset (headphone+mic) jack lenovo-dock Enables docking station I/O for some Lenovos dell-headset-multi Headset jack, which can also be used as mic-in dell-headset-dock Headset jack (without mic-in), and also dock I/O diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ae847fe006c8..79e6fe7a863a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2819,6 +2819,15 @@ static void alc269_fixup_hweq(struct hda_codec *codec, alc_write_coef_idx(codec, 0x1e, coef | 0x80); } +static void alc269_fixup_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; +} + static void alc271_fixup_dmic(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3516,6 +3525,7 @@ enum { ALC271_FIXUP_DMIC, ALC269_FIXUP_PCM_44K, ALC269_FIXUP_STEREO_DMIC, + ALC269_FIXUP_HEADSET_MIC, ALC269_FIXUP_QUANTA_MUTE, ALC269_FIXUP_LIFEBOOK, ALC269_FIXUP_AMIC, @@ -3615,6 +3625,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_stereo_dmic, }, + [ALC269_FIXUP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_headset_mic, + }, [ALC269_FIXUP_QUANTA_MUTE] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_quanta_mute, @@ -3988,6 +4002,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_STEREO_DMIC, .name = "alc269-dmic"}, {.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"}, {.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {.id = ALC269_FIXUP_HEADSET_MIC, .name = "headset-mic"}, {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, -- cgit v1.2.3 From fbc78ad62471c54ca5c10c6a7d440d1ca64d74e7 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 11 Oct 2013 13:46:04 +0200 Subject: ALSA: hda - Sony VAIO Pro 13 (haswell) now has a working headset jack Just got the positive confirmation from a tester: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1227093/comments/28 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 79e6fe7a863a..bf313bea7085 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3761,6 +3761,8 @@ static const struct hda_fixup alc269_fixups[] = { { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ { } }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, -- cgit v1.2.3 From c5d5a58d7ff977289c4bba8eae447c9afa66516b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 11 Oct 2013 00:07:01 -0700 Subject: ASoC: rcar: fixup generation checker Current rcar is using rsnd_is_gen1/gen2() to checking its IP generation, but it needs data mask. This patch fixes it up. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 1 + sound/soc/sh/rcar/rsnd.h | 4 ++-- 2 files changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index fe66533e9b7a..fb0a312bcb81 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -68,6 +68,7 @@ struct rsnd_scu_platform_info { * * A : generation */ +#define RSND_GEN_MASK (0xF << 0) #define RSND_GEN1 (1 << 0) /* fixme */ #define RSND_GEN2 (2 << 0) /* fixme */ diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 9cc6986a8cfb..5dd87f4c919e 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -220,8 +220,8 @@ int rsnd_gen_path_exit(struct rsnd_priv *priv, void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); -#define rsnd_is_gen1(s) ((s)->info->flags & RSND_GEN1) -#define rsnd_is_gen2(s) ((s)->info->flags & RSND_GEN2) +#define rsnd_is_gen1(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN1) +#define rsnd_is_gen2(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN2) /* * R-Car ADG -- cgit v1.2.3 From ccb041571b73888785ef7828a276e380125891a4 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 14 Oct 2013 10:16:22 +0200 Subject: ALSA: hda - Fix inverted internal mic not indicated on some machines MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The create_bind_cap_vol_ctl does not create any control indicating that an inverted dmic is present. Therefore, create multiple capture volumes in this scenario, so we always have some indication that the internal mic is inverted. This happens on the Lenovo Ideapad U310 as well as the Lenovo Yoga 13 (both are based on the CX20590 codec), but the fix is generic and could be needed for other codecs/machines too. Thanks to Szymon Acedański for the pointer and a draft patch. BugLink: https://bugs.launchpad.net/bugs/1239392 BugLink: https://bugs.launchpad.net/bugs/1227491 Reported-by: Szymon Acedański Signed-off-by: David Henningsson Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ac41e9cdc976..26ad4f0aade3 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3531,7 +3531,7 @@ static int create_capture_mixers(struct hda_codec *codec) if (!multi) err = create_single_cap_vol_ctl(codec, n, vol, sw, inv_dmic); - else if (!multi_cap_vol) + else if (!multi_cap_vol && !inv_dmic) err = create_bind_cap_vol_ctl(codec, n, vol, sw); else err = create_multi_cap_vol_ctl(codec); -- cgit v1.2.3 From 64256ac6c2b6fb598fbe187a5503fd9dbb810374 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 12 Oct 2013 17:24:25 +0800 Subject: ASoC: pcm1681: Fix max_register setting According to the datasheet, the max_register is 13h. ARRAY_SIZE(pcm1681_reg_defaults) + 1 is 18 which is wrong. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1681.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 651ce0923675..c91eba504f92 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -270,7 +270,7 @@ MODULE_DEVICE_TABLE(of, pcm1681_dt_ids); static const struct regmap_config pcm1681_regmap = { .reg_bits = 8, .val_bits = 8, - .max_register = ARRAY_SIZE(pcm1681_reg_defaults) + 1, + .max_register = 0x13, .reg_defaults = pcm1681_reg_defaults, .num_reg_defaults = ARRAY_SIZE(pcm1681_reg_defaults), .writeable_reg = pcm1681_writeable_reg, -- cgit v1.2.3 From acc8da7642c8d8dc408d9713de61273950c20714 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 12 Oct 2013 17:24:25 +0800 Subject: ASoC: pcm1681: Fix max_register setting According to the datasheet, the max_register is 13h. ARRAY_SIZE(pcm1681_reg_defaults) + 1 is 18 which is wrong. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1681.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 651ce0923675..c91eba504f92 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -270,7 +270,7 @@ MODULE_DEVICE_TABLE(of, pcm1681_dt_ids); static const struct regmap_config pcm1681_regmap = { .reg_bits = 8, .val_bits = 8, - .max_register = ARRAY_SIZE(pcm1681_reg_defaults) + 1, + .max_register = 0x13, .reg_defaults = pcm1681_reg_defaults, .num_reg_defaults = ARRAY_SIZE(pcm1681_reg_defaults), .writeable_reg = pcm1681_writeable_reg, -- cgit v1.2.3 From c92f66e2809dc46f834678329d7f744193557db6 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 12 Oct 2013 17:26:49 +0800 Subject: ASoC: pcm1792a: Fix max_register setting According to the datasheet, the max_register is register 23. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1792a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 2a8eccf64c76..7613181123fe 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -188,7 +188,7 @@ MODULE_DEVICE_TABLE(of, pcm1792a_of_match); static const struct regmap_config pcm1792a_regmap = { .reg_bits = 8, .val_bits = 8, - .max_register = 24, + .max_register = 23, .reg_defaults = pcm1792a_reg_defaults, .num_reg_defaults = ARRAY_SIZE(pcm1792a_reg_defaults), .writeable_reg = pcm1792a_writeable_reg, -- cgit v1.2.3 From ac536a848a1643e4b87e8fbd376a63091afc2ccc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 14 Oct 2013 16:02:15 +0200 Subject: ALSA: us122l: Fix pcm_usb_stream mmapping regression The pcm_usb_stream plugin requires the mremap explicitly for the read buffer, as it expands itself once after reading the required size. But the commit [314e51b9: mm: kill vma flag VM_RESERVED and mm->reserved_vm counter] converted blindly to a combination of VM_DONTEXPAND | VM_DONTDUMP like other normal drivers, and this resulted in the failure of mremap(). For fixing this regression, we need to remove VM_DONTEXPAND for the read-buffer mmap. Reported-and-tested-by: James Miller Cc: Signed-off-by: Takashi Iwai --- sound/usb/usx2y/us122l.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index d0323a693ba2..999550bbad40 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -262,7 +262,9 @@ static int usb_stream_hwdep_mmap(struct snd_hwdep *hw, } area->vm_ops = &usb_stream_hwdep_vm_ops; - area->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP; + area->vm_flags |= VM_DONTDUMP; + if (!read) + area->vm_flags |= VM_DONTEXPAND; area->vm_private_data = us122l; atomic_inc(&us122l->mmap_count); out: -- cgit v1.2.3 From d14df339c72b6efbba4eddd1d1f3f4b173273f74 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 16 Oct 2013 11:44:25 +0300 Subject: ALSA: hdsp - info leak in snd_hdsp_hwdep_ioctl() In GCC the sizeof(hdsp_version) is 8 because there is a 2 byte hole at the end of the struct after ->firmware_rev. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 4f255dfee450..f59a321a6d6a 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -4845,6 +4845,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne if ((err = hdsp_get_iobox_version(hdsp)) < 0) return err; } + memset(&hdsp_version, 0, sizeof(hdsp_version)); hdsp_version.io_type = hdsp->io_type; hdsp_version.firmware_rev = hdsp->firmware_rev; if ((err = copy_to_user(argp, &hdsp_version, sizeof(hdsp_version)))) -- cgit v1.2.3 From e6bbe666673ab044a3d39ddb74e4d9a401cf1d6f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 24 Oct 2013 01:20:24 +0200 Subject: ALSA: hda - Fix unbalanced runtime PM refcount after S3/S4 When a machine goes to S3/S4 after power-save is enabled, the runtime PM refcount might be incorrectly decreased because the power-down triggered soon after resume assumes that the controller was already powered up, and issues the pm_notify down. This patch fixes the incorrect pm_notify call simply by checking the current value properly. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5b6c4e3c92ca..748c6a941963 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4864,8 +4864,8 @@ static void hda_power_work(struct work_struct *work) spin_unlock(&codec->power_lock); state = hda_call_codec_suspend(codec, true); - codec->pm_down_notified = 0; - if (!bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) { + if (!codec->pm_down_notified && + !bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) { codec->pm_down_notified = 1; hda_call_pm_notify(bus, false); } -- cgit v1.2.3 From b63eae0a6c84839275a4638a7baa391be965cd0e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 25 Oct 2013 23:43:10 +0200 Subject: ALSA: hda - Add missing initial vmaster hook at build_controls callback The generic parser has a support of vmaster hook, but this is initialized only in the init callback with the check of the presence of the corresponding kctl. However, since kctl is NULL at the very first init callback that is called before build_controls callback, the vmaster hook sync is skipped there. Eventually this leads to the uninitialized state depending on the hook implementation. This patch adds a simple workaround, just calling the sync function explicitly at build_controls callback. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 26ad4f0aade3..b7c89dff7066 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4475,9 +4475,11 @@ int snd_hda_gen_build_controls(struct hda_codec *codec) true, &spec->vmaster_mute.sw_kctl); if (err < 0) return err; - if (spec->vmaster_mute.hook) + if (spec->vmaster_mute.hook) { snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, spec->vmaster_mute_enum); + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + } } free_kctls(spec); /* no longer needed */ -- cgit v1.2.3 From 1ac3293095deb01ccc491f3c171e12722ebd0bc9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 26 Oct 2013 00:24:14 +0200 Subject: ALSA: hda - Fix silent headphone on Thinkpads with AD1984A codec AD1984A codec has a couple of pins with EAPD controls, and the generic codec driver tries to turn each of them on/off depending on the pin active state. However, Thinkpads seem to use EAPD of the speaker pin as a master EAPD for controlling the mute of all outputs, including the headphone. This results in the dead headphone output via the headphone plugging because it mutes the speaker and turns off EAPD. The fix is to simply add spec->gen.keep_on_eapd flag. [This is a regression fix on 3.12 where we moved the AD codec parser to the generic parser. 3.11 and earlier didn't show this problem because still static quirks have been used.] Reported-and-tested-by: Vito Caputo Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0cbdd87dde6d..2aa2f579b4d6 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -968,6 +968,15 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec, } } +static void ad1884_fixup_thinkpad(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct ad198x_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->gen.keep_eapd_on = 1; +} + /* set magic COEFs for dmic */ static const struct hda_verb ad1884_dmic_init_verbs[] = { {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, @@ -979,6 +988,7 @@ enum { AD1884_FIXUP_AMP_OVERRIDE, AD1884_FIXUP_HP_EAPD, AD1884_FIXUP_DMIC_COEF, + AD1884_FIXUP_THINKPAD, AD1884_FIXUP_HP_TOUCHSMART, }; @@ -997,6 +1007,12 @@ static const struct hda_fixup ad1884_fixups[] = { .type = HDA_FIXUP_VERBS, .v.verbs = ad1884_dmic_init_verbs, }, + [AD1884_FIXUP_THINKPAD] = { + .type = HDA_FIXUP_FUNC, + .v.func = ad1884_fixup_thinkpad, + .chained = true, + .chain_id = AD1884_FIXUP_DMIC_COEF, + }, [AD1884_FIXUP_HP_TOUCHSMART] = { .type = HDA_FIXUP_VERBS, .v.verbs = ad1884_dmic_init_verbs, @@ -1008,7 +1024,7 @@ static const struct hda_fixup ad1884_fixups[] = { static const struct snd_pci_quirk ad1884_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a82, "HP Touchsmart", AD1884_FIXUP_HP_TOUCHSMART), SND_PCI_QUIRK_VENDOR(0x103c, "HP", AD1884_FIXUP_HP_EAPD), - SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_DMIC_COEF), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_THINKPAD), {} }; -- cgit v1.2.3