May 2011 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: HDA: Add jack detection for HDMI
by David Henningsson 17 May '11

17 May '11

Just as for headphones and microphone jacks, this patch adds reporting of HDMI jack status through the input layer. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 2 ++ sound/pci/hda/patch_hdmi.c | 15 +++++++++++++++ 2 files changed, 17 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c63f376..8edd998 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5055,6 +5055,8 @@ static const char *get_jack_default_name(struct hda_codec *codec, hda_nid_t nid, return "Line-out"; case SND_JACK_HEADSET: return "Headset"; + case SND_JACK_VIDEOOUT: + return "HDMI/DP"; default: return "Misc"; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 6eb209d..16c1505 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -33,6 +33,7 @@ #include <linux/slab.h> #include <linux/moduleparam.h> #include <sound/core.h> +#include <sound/jack.h> #include "hda_codec.h" #include "hda_local.h" @@ -720,6 +721,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) &spec->sink_eld[index]); /* TODO: do real things about ELD */ } + + snd_hda_input_jack_report(codec, tag); } static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) @@ -919,6 +922,17 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) return -E2BIG; } +#ifdef CONFIG_SND_HDA_INPUT_JACK + { + int err; + err = snd_hda_input_jack_add(codec, pin_nid, + SND_JACK_VIDEOOUT, NULL); + if (err < 0) + return err; + snd_hda_input_jack_report(codec, pin_nid); + } +#endif + hdmi_present_sense(codec, pin_nid, &spec->sink_eld[spec->num_pins]); spec->pin[spec->num_pins] = pin_nid; @@ -1120,6 +1134,7 @@ static void generic_hdmi_free(struct hda_codec *codec) for (i = 0; i < spec->num_pins; i++) snd_hda_eld_proc_free(codec, &spec->sink_eld[i]); + snd_hda_input_jack_free(codec); kfree(spec); } -- 1.7.4.1 --------------030905020802080904050601 Content-Type: text/x-patch; name="0001-hda-emu-update-jack.h-to-match-latest-kernel-version.patch" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename*0="0001-hda-emu-update-jack.h-to-match-latest-kernel-version.pa"; filename*1="tch"

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[alsa-devel] [PATCH v2 0/3] ASoC: add support for audio on iPaq hx4700
by Dmitry Artamonow 17 May '11

17 May '11

Ok, here's patches adding audio support for good ole iPAQ hx4700: ak4641 codec, hx4700 machine sound driver and some glue platform code. It took quite longer to make v2 than I expected, but it's finally here. Sorry for such late submission in this release cycle, but if it's not ready yet, no big deal - it certainly can wait for 2.6.41. v1 of these patches can be found here: http://lkml.org/lkml/2010/11/20/50 Current patchset is based on next-20110513 (it also can be applied to 2.6.39-rc7 after slight adjustment of Kconfig/Makefile chunks) Latest version of these patches were only slightly tested on top of 2.6.39-rc7 (-next is tested compile-only, as it doesn't boot due to some unrelated bugs), but all basic operations like playback, recording, EQ, Speaker, Mic Sidetone seem to work. Changes from v1: AK4641 codec: * '-codec' is removed from 'ak4641-codec' platform device name * SNDRV_PCM_RATE_8000_48000 macro now used instead of explicit freqs listing * Unneeded ifdef's on CONFIG_I2C are removed * Missing "Volume" added to names of volume controls, ditto for switches * TLV information is added to various volume controls * Enum control doesn't use indexed array of strings anymore * Enum controls like 'Mic Boost +20dB/0dB' are turned into volume controls * Deemphasis control is converted from enum to switch with automatic detection of optimal deemphasis value depending on current playback frequency * Check for wrong fs value is added to hw_params for HiFi DAI (that actually uncovered bug in hx4700 machine driver - it never set sysclk for the codec. Sound worked properly this way due to strange coincidence: both default fs in codec driver and divider in PXA i2s hardware were equal 256) * empty hw_params() function for Voice DAI is removed. Looks like that samplerate of Voice ADC/DAC can't be controlled by software at all, only format of samples. This Voice DAC/ADC are designed to be connected to Bluetooth chip, so I suspect codec gets sampling frequency directly from it. * Names of mic sidetone controls are adjusted * 'Left/Right Out Enable' DAPM controls are turned into ordinary ones * Mute code readability is slightly improved (based on Axel Lin's patch for ak4535 - commit cca67a3668) * My copyright is added to file, so people won't blame Harald for all the bugs I added during multicomponent conversion and other cleanup :) * Power/reset GPIOs are now handled in set_bias_level(), so codec is powered only when needed. HX4700 sound: * Setting of codec's sysclk is added to hw_params() HX4700 platform: * No changes Dmitry Artamonow (2): ASoC: Asahi Kasei AK4641 codec driver ASoC: add iPAQ hx4700 machine driver Philipp Zabel (1): pxa/hx4700: add I2C board info for AK4641 codec arch/arm/mach-pxa/hx4700.c | 18 + arch/arm/mach-pxa/include/mach/hx4700.h | 4 +- include/sound/ak4641.h | 26 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4641.c | 687 +++++++++++++++++++++++++++++++ sound/soc/codecs/ak4641.h | 47 +++ sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/hx4700.c | 260 ++++++++++++ 10 files changed, 1057 insertions(+), 2 deletions(-) create mode 100644 include/sound/ak4641.h create mode 100644 sound/soc/codecs/ak4641.c create mode 100644 sound/soc/codecs/ak4641.h create mode 100644 sound/soc/pxa/hx4700.c -- 1.7.4.rc1

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[alsa-devel] [PATCH RFC 0/2] da850 mcasp playback ping-pong buffers don't work
by Ben Gardiner 16 May '11

16 May '11

Hi All, I've been testing playback on the da850evm. The good news is on 2.6.39-rc7 playback works fine. The bad news is that if the patches in the series are applied -- playback does not work. The first patch copies the sram-size platform data to the dma parameters used by davinci-pcm the same way that the existing davinci-i2s driver does. The second patch assigns a buffer size to the playback sram platform data and sets the queues as per Troy's introduction of ping-pong buffers. I was hoping this would result in playback of audio; but instead there is soft click and then silence. I added pointer debug and observed that the period_update and hwptr_update events progress; but that's as far as I have probed. Does anyone have any clues why ping-pong buffers on da850 to the McASP would work? Or any clues why the patches are broken? Best Regards, Ben Gardiner Ben Gardiner (2): [RFC] ASoC: davinci-mcasp: enable ping-pong SRAM buffers [RFC] da850evm: enable mcasp ping-pong buffers arch/arm/configs/da8xx_omapl_defconfig | 1 + arch/arm/mach-davinci/board-da850-evm.c | 4 +++- sound/soc/davinci/davinci-mcasp.c | 2 ++ 3 files changed, 6 insertions(+), 1 deletions(-) -- 1.7.4.1

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Re: [alsa-devel] [PATCH 7/9] ASoC: Silence DEBUG_STRICT_USER_COPY_CHECKS=y warning
by Liam Girdwood 16 May '11

16 May '11

On Thu, 2011-05-12 at 16:50 -0700, Stephen Boyd wrote: > Enabling DEBUG_STRICT_USER_COPY_CHECKS causes the following > warning: > > In file included from arch/x86/include/asm/uaccess.h:573, > from include/linux/poll.h:14, > from include/sound/pcm.h:29, > from include/sound/ac97_codec.h:31, > from sound/soc/soc-core.c:34: > In function 'copy_from_user', > inlined from 'codec_reg_write_file' at > sound/soc/soc-core.c:252: > arch/x86/include/asm/uaccess_64.h:65: > warning: call to 'copy_from_user_overflow' declared with > attribute warning: copy_from_user() buffer size is not provably > correct > > presumably due to buf_size being signed causing GCC to fail to > see that buf_size can't become negative. > > Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> > Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > Signed-off-by: Stephen Boyd <sboyd(a)codeaurora.org> Acked-by: Liam Girdwood <lrg(a)ti.com>

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[alsa-devel] [PATCHv2] ASoC: Fix wrong data type access in a few codec drivers
by Jarkko Nikula 16 May '11

16 May '11

Commit fafd217 ("ASoC: Store a list of widgets in a DAPM mux/mixer kcontrol") changed the control private data type that is passed to snd_soc_cnew when creating dapm mixer and mux controls. Commit did not update a few codec drivers that are using their own put callbacks and thus are accessing a wrong data type. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Tested-by: Stephen Warren <swarren(a)nvidia.com> --- v2: - wm8994 and wm8995 needs to be changed too. Thanks to Stephen Warren <swarren(a)nvidia.com> for noticing. - "Otherwise, on Tegra+WM8903, I repro'd the bug, and tested the fix, so: Tested-by: Stephen Warren <swarren(a)nvidia.com>" --- sound/soc/codecs/tlv320aic3x.c | 3 ++- sound/soc/codecs/wm8903.c | 3 ++- sound/soc/codecs/wm8993.c | 3 ++- sound/soc/codecs/wm8994.c | 6 ++++-- sound/soc/codecs/wm8995.c | 4 ++-- 5 files changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6c43c13..c3d96fc 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -157,7 +157,8 @@ static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg, static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 957cd66..43e3d76 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -382,7 +382,8 @@ static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, static int wm8903_class_w_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); u16 reg; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 056aef9..9e5ff78 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -718,7 +718,8 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, static int class_w_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); int ret; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e6dfa10..970a95c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -877,7 +877,8 @@ static const char *hp_mux_text[] = { static int wm8994_put_hp_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *w = wlist->widgets[0]; struct snd_soc_codec *codec = w->codec; int ret; @@ -1004,7 +1005,8 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *w = wlist->widgets[0]; struct snd_soc_codec *codec = w->codec; int ret; diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 67eaaec..5ad873f 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -305,11 +305,11 @@ static int check_clk_sys(struct snd_soc_dapm_widget *source, static int wm8995_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *w = wlist->widgets[0]; struct snd_soc_codec *codec; int ret; - w = snd_kcontrol_chip(kcontrol); codec = w->codec; ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); wm8995_update_class_w(codec); -- 1.7.0.4

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[alsa-devel] hda-jack-sense-test
by David Henningsson 16 May '11

16 May '11

Just a little something I wrote on the train back home yesterday. I hope someone finds it a nice way to figure out if your BIOS is lying to you about what pin is used for what. It (re)uses some of the classes in hda-analyzer. For Ubuntu 11.04 users, this small tool is now packaged together with codecgraph, hda-verb, hda-emu and hda-analyzer in my snd-hda-tools package, which is available in ppa:diwic/ppa. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] [PATCH 1/2] ASoC: Skip noop reconfiguration of WM8958 DSP2 algorithms
by Mark Brown 16 May '11

16 May '11

If we're setting the currently applied value for one of the DSP algorithm configurations we can just skip all the handling as the control set is a noop. This ensures we do not disrupt a running DSP. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8958-dsp2.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 5d4bc7a..ca26779 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -508,6 +508,9 @@ static int wm8958_mbc_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (wm8994->mbc_ena[mbc] == ucontrol->value.integer.value[0]) + return 0; + if (ucontrol->value.integer.value[0] > 1) return -EINVAL; @@ -628,6 +631,9 @@ static int wm8958_vss_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (wm8994->vss_ena[vss] == ucontrol->value.integer.value[0]) + return 0; + if (ucontrol->value.integer.value[0] > 1) return -EINVAL; @@ -689,6 +695,16 @@ static int wm8958_hpf_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (hpf < 3) { + if (wm8994->hpf1_ena[hpf % 3] == + ucontrol->value.integer.value[0]) + return 0; + } else { + if (wm8994->hpf2_ena[hpf % 3] == + ucontrol->value.integer.value[0]) + return 0; + } + if (ucontrol->value.integer.value[0] > 1) return -EINVAL; @@ -782,6 +798,9 @@ static int wm8958_enh_eq_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (wm8994->enh_eq_ena[eq] == ucontrol->value.integer.value[0]) + return 0; + if (ucontrol->value.integer.value[0] > 1) return -EINVAL; -- 1.7.5.1

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[alsa-devel] [PATCH] ASoC: Add some missing volume update bit sets for wm_hubs devices
by Mark Brown 16 May '11

16 May '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index c7c2fec..e55b298 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -836,17 +836,21 @@ int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, WM8993_IN2_VU, WM8993_IN2_VU); + snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_LEFT, + WM8993_SPKOUT_VU, WM8993_SPKOUT_VU); snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_RIGHT, WM8993_SPKOUT_VU, WM8993_SPKOUT_VU); snd_soc_update_bits(codec, WM8993_LEFT_OUTPUT_VOLUME, - WM8993_HPOUT1L_ZC, WM8993_HPOUT1L_ZC); + WM8993_HPOUT1_VU | WM8993_HPOUT1L_ZC, + WM8993_HPOUT1_VU | WM8993_HPOUT1L_ZC); snd_soc_update_bits(codec, WM8993_RIGHT_OUTPUT_VOLUME, WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC, WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC); snd_soc_update_bits(codec, WM8993_LEFT_OPGA_VOLUME, - WM8993_MIXOUTL_ZC, WM8993_MIXOUTL_ZC); + WM8993_MIXOUTL_ZC | WM8993_MIXOUT_VU, + WM8993_MIXOUTL_ZC | WM8993_MIXOUT_VU); snd_soc_update_bits(codec, WM8993_RIGHT_OPGA_VOLUME, WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU, WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU); -- 1.7.5.1

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[alsa-devel] [PATCH] alsa-lib - emu10k1.conf - Fix no sound problem when using SB Live! Platinum
by Raymond Yau 16 May '11

16 May '11

This patch allow pulseaudio to use "front" device for playback and capture , also allow user to use "rear" device of SB Live! for playback by adding type asym to front, rear and center_lfe After commit 16f716eadd61be80b28839396105c020e401294c which remove the lock If application using "hw", "front" , "rear" , "surround40" and "surround51" for playback , right channel or both left and right channels suddenly lost sound beause the route of the first stereo channel is changed when other application using "front", "rear" or "center_lfe" to record After applying this patch, "rear" , "center_lfe", "surround40" and "surround51" can no longer be used to record arecord -v -Drear:CARD=Live -f cd /dev/null ALSA lib pcm_asym.c:106:(_snd_pcm_asym_open) capture slave is not defined arecord: main:660: audio open error: Invalid argument

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[alsa-devel] [PATCH] ALSA: HDA: Use one dmic only for Dell Studio 1558
by David Henningsson 16 May '11

16 May '11

There are no signs of a dmic at node 0x0b, so the user is left with an additional internal mic which does not exist. This commit removes that non-existing mic. Codec info at http://bugs.launchpad.net/bugs/731706. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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