From 0962bb217ac74c4b8fae34c5367ebc63131c962c Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Wed, 2 Feb 2011 21:11:41 +0100 Subject: [PATCH 01/12] ASoC: fill in snd_soc_pcm_runtime.card before calling snd_soc_dai_link.init() The .card member of the snd_soc_pcm_runtime structure pointed to by the snd_soc_dai_link.init() argument used to be initialized before the function being called. This has changed, probably unintentionally, after recent refactorings. Since the function implementations are free to make use of this pointer, move its assignment back before the function is called to avoid NULL pointer dereferences. Created and tested on Amstrad Delta againts linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c4b60610beb0..c3f6f1e72790 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1449,6 +1449,7 @@ static int soc_post_component_init(struct snd_soc_card *card, rtd = &card->rtd_aux[num]; name = aux_dev->name; } + rtd->card = card; /* machine controls, routes and widgets are not prefixed */ temp = codec->name_prefix; @@ -1471,7 +1472,6 @@ static int soc_post_component_init(struct snd_soc_card *card, /* register the rtd device */ rtd->codec = codec; - rtd->card = card; rtd->dev.parent = card->dev; rtd->dev.release = rtd_release; rtd->dev.init_name = name; From f9eb9dd14c2ca2a1f8d979637fb651512d16ad22 Mon Sep 17 00:00:00 2001 From: Vaibhav Bedia Date: Thu, 3 Feb 2011 16:42:25 +0530 Subject: [PATCH 02/12] asoc: davinci: da830/omap-l137: correct cpu_dai_name McASP1 is used on the DA830/OMAP-L137 platform for the codec. This is different from the DA850/OMAP-L138 platform which uses McASP0. This is fixed by adding a new snd_soc_dai_link struct. Signed-off-by: Vaibhav Bedia Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-evm.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index b36f0b39b090..fe7984221eb9 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -218,7 +218,19 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .ops = &evm_spdif_ops, }, }; -static struct snd_soc_dai_link da8xx_evm_dai = { + +static struct snd_soc_dai_link da830_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai_name = "davinci-mcasp.1", + .codec_dai_name = "tlv320aic3x-hifi", + .codec_name = "tlv320aic3x-codec.1-0018", + .platform_name = "davinci-pcm-audio", + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +static struct snd_soc_dai_link da850_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", @@ -259,13 +271,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { static struct snd_soc_card da830_snd_soc_card = { .name = "DA830/OMAP-L137 EVM", - .dai_link = &da8xx_evm_dai, + .dai_link = &da830_evm_dai, .num_links = 1, }; static struct snd_soc_card da850_snd_soc_card = { .name = "DA850/OMAP-L138 EVM", - .dai_link = &da8xx_evm_dai, + .dai_link = &da850_evm_dai, .num_links = 1, }; From 7f94de483f4e37e14d646ad6e85a3c82f66fb487 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 3 Feb 2011 16:27:34 +0000 Subject: [PATCH 03/12] ASoC: Create an AIF1ADCDAT signal widget to match AIF2 Due to the different routing for AIF1 and AIF2 we weren't using a single widget to represent the ADCDAT signal. For consistency add one. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3351f77607b3..3e308ad97ddf 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1287,9 +1287,9 @@ SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 9, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, 0, WM8994_POWER_MANAGEMENT_4, 8, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0, WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev, @@ -1298,9 +1298,9 @@ SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0, WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), -SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 11, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL, 0, WM8994_POWER_MANAGEMENT_4, 10, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0, WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev, @@ -1345,6 +1345,7 @@ SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux), @@ -1546,6 +1547,11 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" }, { "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" }, + { "AIF1ADCDAT", NULL, "AIF1ADC1L" }, + { "AIF1ADCDAT", NULL, "AIF1ADC1R" }, + { "AIF1ADCDAT", NULL, "AIF1ADC2L" }, + { "AIF1ADCDAT", NULL, "AIF1ADC2R" }, + { "AIF2ADCDAT", NULL, "AIF2ADC Mux" }, /* AIF3 output */ From 6ed8f1485fc82d44ac464bc84a7dcdddd1fa096f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 3 Feb 2011 16:27:35 +0000 Subject: [PATCH 04/12] ASoC: Improve WM8994 digital power sequencing On WM8994 revision D and earlier ensure optimal sequencing with simultaneous usage of AIF1 and AIF2 by tying the signals together so if paths through both are connected the streams are started simultaneously. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3e308ad97ddf..37b8aa8a680f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1584,6 +1584,13 @@ static const struct snd_soc_dapm_route intercon[] = { { "Right Headphone Mux", "DAC", "DAC1R" }, }; +static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { + { "AIF1DACDAT", NULL, "AIF2DACDAT" }, + { "AIF2DACDAT", NULL, "AIF1DACDAT" }, + { "AIF1ADCDAT", NULL, "AIF2ADCDAT" }, + { "AIF2ADCDAT", NULL, "AIF1ADCDAT" }, +}; + static const struct snd_soc_dapm_route wm8994_intercon[] = { { "AIF2DACL", NULL, "AIF2DAC Mux" }, { "AIF2DACR", NULL, "AIF2DAC Mux" }, @@ -3135,6 +3142,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8994: snd_soc_dapm_add_routes(dapm, wm8994_intercon, ARRAY_SIZE(wm8994_intercon)); + + if (wm8994->revision < 4) + snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, + ARRAY_SIZE(wm8994_revd_intercon)); + break; case WM8958: snd_soc_dapm_add_routes(dapm, wm8958_intercon, From 460c92fa38ff140f83c269e948e2aaab071d0af0 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?=C5=81ukasz=20Wojni=C5=82owicz?= Date: Mon, 7 Feb 2011 13:13:27 +0100 Subject: [PATCH 05/12] ALSA: hda - switch lfe with side in mixer for 4930g MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Built-in sub-woofer can now be controlled by lfe slider instead of side slider on Acer Aspire 5930g Signed-off-by: Łukasz Wojniłowicz Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2fa9ed99c32f..2571d977df22 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2290,6 +2290,29 @@ static struct snd_kcontrol_new alc888_base_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0f, 2, 0x0, + HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0f, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0f, 1, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0f, 1, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Side Playback Volume", 0x0e, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Side Playback Switch", 0x0e, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + + static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -10359,7 +10382,7 @@ static struct alc_config_preset alc882_presets[] = { .init_hook = alc_automute_amp, }, [ALC888_ACER_ASPIRE_4930G] = { - .mixers = { alc888_base_mixer, + .mixers = { alc888_acer_aspire_4930g_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, alc888_acer_aspire_4930g_verbs }, From 1cdfa9f34acb9780e0fe7b8a41fb1a885ab94735 Mon Sep 17 00:00:00 2001 From: Joseph Teichman Date: Tue, 8 Feb 2011 01:22:36 -0500 Subject: [PATCH 06/12] ALSA: usbaudio - Enable the E-MU 0204 USB Signed-off-by: Joseph Teichman Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 4 ++-- sound/usb/quirks-table.h | 7 +++++++ sound/usb/quirks.c | 3 ++- 3 files changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 7df89b3d7ded..85af6051b52d 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -95,7 +95,7 @@ enum { }; -/*E-mu 0202(0404) eXtension Unit(XU) control*/ +/*E-mu 0202/0404/0204 eXtension Unit(XU) control*/ enum { USB_XU_CLOCK_RATE = 0xe301, USB_XU_CLOCK_SOURCE = 0xe302, @@ -1566,7 +1566,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw cval->initialized = 1; } else { if (type == USB_XU_CLOCK_RATE) { - /* E-Mu USB 0404/0202/TrackerPre + /* E-Mu USB 0404/0202/TrackerPre/0204 * samplerate control quirk */ cval->min = 0; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 35999874d301..921a86fd9884 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -79,6 +79,13 @@ .idProduct = 0x3f0a, .bInterfaceClass = USB_CLASS_AUDIO, }, +{ + /* E-Mu 0204 USB */ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x041e, + .idProduct = 0x3f19, + .bInterfaceClass = USB_CLASS_AUDIO, +}, /* * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index cf8bf088394b..e314cdb85003 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -532,7 +532,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat } /* - * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device, + * For E-Mu 0404USB/0202USB/TrackerPre/0204 sample rate should be set for device, * not for interface. */ @@ -589,6 +589,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ + case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; } From 11839aed21881d7edd65dd79f22a8eb18426f672 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Feb 2011 17:25:49 +0100 Subject: [PATCH 07/12] ALSA: hda - Fix missing CA initialization for HDMI/DP The commit 53d7d69d8ffdfa60c5b66cc2e9ee0774aaaef5c0 ALSA: hdmi - support infoframe for DisplayPort dropped the initialization of CA field accidentally. This resulted in only two-channel LPCM mode on Nvidia machines. Reference: kernel bug 28592 https://bugzilla.kernel.org/show_bug.cgi?id=28592 Signed-off-by: Takashi Iwai Cc: --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2d5b83fa8d24..a58767736727 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -642,6 +642,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, hdmi_ai->ver = 0x01; hdmi_ai->len = 0x0a; hdmi_ai->CC02_CT47 = channels - 1; + hdmi_ai->CA = ca; hdmi_checksum_audio_infoframe(hdmi_ai); } else if (spec->sink_eld[i].conn_type == 1) { /* DisplayPort */ struct dp_audio_infoframe *dp_ai; @@ -651,6 +652,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, dp_ai->len = 0x1b; dp_ai->ver = 0x11 << 2; dp_ai->CC02_CT47 = channels - 1; + dp_ai->CA = ca; } else { snd_printd("HDMI: unknown connection type at pin %d\n", pin_nid); From 41a63f18d339ae6aefe73d45a8147f63f3439b30 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Feb 2011 17:39:20 +0100 Subject: [PATCH 08/12] ALSA: hda - Don't handle empty patch files When an empty string is passed to patch option, the driver should ignore it. Otherwise it gets an error by trying to load it. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2e91a991eb15..0baffcdee8f9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2703,7 +2703,7 @@ static int __devinit azx_probe(struct pci_dev *pci, if (err < 0) goto out_free; #ifdef CONFIG_SND_HDA_PATCH_LOADER - if (patch[dev]) { + if (patch[dev] && *patch[dev]) { snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n", patch[dev]); err = snd_hda_load_patch(chip->bus, patch[dev]); From a6c47a85b8e7e4a8c47394607c5e5c43224b0892 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 10 Feb 2011 15:39:19 +0100 Subject: [PATCH 09/12] ALSA: HDA: Add subwoofer quirk for Acer Aspire 8942G According to the reporter, node 0x15 needs to be muted for subwoofer to stop sounding. This pin is marked as unused by BIOS, so fix that. BugLink: http://bugs.launchpad.net/bugs/715877 Cc: stable@kernel.org (2.6.37+) Reported-by: Hans Peter Signed-off-by: David Henningsson 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 2571d977df22..089a7de2439e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -19517,6 +19517,7 @@ static const struct alc_fixup alc662_fixups[] = { }; static struct snd_pci_quirk alc662_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), From b1d4f7f4bdcf9915c41ff8cfc4425c84dabb1fde Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 10 Feb 2011 16:15:44 +0100 Subject: [PATCH 10/12] ALSA: hrtimer: handle delayed timer interrupts If a timer interrupt was delayed too much, hrtimer_forward_now() will forward the timer expiry more than once. When this happens, the additional number of elapsed ALSA timer ticks must be passed to snd_timer_interrupt() to prevent the ALSA timer from falling behind. This mostly fixes MIDI slowdown problems on highly-loaded systems with badly behaved interrupt handlers. Signed-off-by: Clemens Ladisch Reported-and-tested-by: Arthur Marsh Cc: Signed-off-by: Takashi Iwai --- sound/core/hrtimer.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index 7730575bfadd..07efa29dfd4a 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -45,12 +45,13 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) { struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt); struct snd_timer *t = stime->timer; + unsigned long oruns; if (!atomic_read(&stime->running)) return HRTIMER_NORESTART; - hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution)); - snd_timer_interrupt(stime->timer, t->sticks); + oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution)); + snd_timer_interrupt(stime->timer, t->sticks * oruns); if (!atomic_read(&stime->running)) return HRTIMER_NORESTART; From 2243c4d0727ad85aff3f54be9d178632cc9234b2 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 10 Feb 2011 16:16:32 +0100 Subject: [PATCH 11/12] ALSA: hrtimer: remove superfluous tasklet invocation Commit bb758e9637e5ddc removed snd_hrtimer_callback() from the hardware interrupt handler, thus moving it into a tasklet, but did not tell the ALSA timer framework about this, so the timer handling would now be done in the ALSA timer tasklet scheduled from another tasklet. To fix this, add the flag to tell the ALSA timer framework that the timer handler is already being invoked in a tasklet. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/core/hrtimer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index 07efa29dfd4a..b8b31c433d64 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -105,7 +105,7 @@ static int snd_hrtimer_stop(struct snd_timer *t) } static struct snd_timer_hardware hrtimer_hw = { - .flags = SNDRV_TIMER_HW_AUTO, + .flags = SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_TASKLET, .open = snd_hrtimer_open, .close = snd_hrtimer_close, .start = snd_hrtimer_start, From 965b76d23ea354848dea8d34059d04e150dcd464 Mon Sep 17 00:00:00 2001 From: Anisse Astier Date: Thu, 10 Feb 2011 13:14:44 +0100 Subject: [PATCH 12/12] ALSA: hda - add quirk for Ordissimo EVE using a realtek ALC662 This netbook has a only one jack output and an internal mic. By default, mic and jack sense aren't working. Using lenovo-101e parameters makes both work. The device seems based on a Sharetronic Q70, so this should fix audio for this model too. Signed-off-by: Anisse Astier 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 089a7de2439e..3328a259a242 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -18825,6 +18825,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { ALC662_3ST_6ch_DIG), SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E), {} };