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[alsa-devel] Support for Compaq Presario F700 - model 755US - NVidia MCP67 Audio - Codec Conexant 5051
by Philippe Paravicini 21 Jan '08

21 Jan '08

I recently purchased the laptop listed above and installed kubuntu-7.10-desktop-amd64 Feisty on it. The sound card works, but, much to my chagrin, plugging in the headphone jack does not mute the speaker. <wining> This is less than ideal for me because I use Skype quite a bit, and work in a no-walls small office with other people. Having the ability to use my headset for talking on the phone and drowning out ambient noise is important. </wining><deepbreath/> I have googled high and low and have tried various suggested solutions, including changing: options snd-hda-intel model=laptop in /etc/modprobe.d/alsa-base to various settings (laptop-hp, laptop-eahd, test). I have also downloaded, compiled and installed the 1.0.15 driver and utils according to the instructions at: https://help.ubuntu.com/community/HdaIntelSoundHowto without success. I consulted the page at: http://www.mjmwired.net/kernel/Documentation/sound/alsa/ALSA-Configuration.… dated 2007-10-11 and could note see support for the Conexant 5051 Codec or NVidia MCP67 Audio. Has anyone had experience with this soundcard/codec and gotten the headphones jack to mute the speakers correctly ? Any help would be greatly appreciated. I am a java programmer with very high reverence for linux C-programmers and would be happy to help test a patch for this device (I may need a little help to get going), or trade for java programming/consulting. Please let me know if you additional information other than the one listed below. Thank you kindly for any help. Sincerely, Philippe Paravicini lspci -v outputs: ... 00:07.0 Audio device: nVidia Corporation MCP67 High Definition Audio (rev a1) Subsystem: Hewlett-Packard Company Unknown device 30ea ... lspci -vn ... 00:07.0 0403: 10de:055c (rev a1) Subsystem: 103c:30ea Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 21 Memory at f6480000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask+ 64bit+ Queue=0/0 Enable- Capabilities: [6c] HyperTransport: MSI Mapping ... cat /proc/asound/card0/codec#0 Codec: Conexant ID 5051 Address: 0 Vendor Id: 0x14f15051 Subsystem Id: 0x103c30ea Revision Id: 0x100000 Modem Function Group: 0x2 Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=0 Amp-Out vals: [0x3a 0x3a] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: 0x0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=0 Amp-Out vals: [0x4a 0x4a] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: 0x0 Node 0x12 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=0 Amp-In vals: [0x06 0x06] [0x35 0x35] PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: 0x0 Connection: 2 0x1d 0x17* Node 0x15 [Audio Input] wcaps 0x100d1b: Stereo Amp-In Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=0 Amp-In vals: [0x4a 0x4a] PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: 0x0 Connection: 1 0x18 Node 0x16 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x081c: OUT HP Detect Pin Default 0x02211030: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Power: 0x0 Connection: 2 0x10* 0x11 Node 0x17 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: Pincap 0x081224: IN Detect Pin Default 0x02a19020: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Power: 0x0 Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: Pincap 0x081224: IN Detect Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Power: 0x0 Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x0814: OUT Detect Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Power: 0x0 Connection: 2 0x10 0x11* Node 0x1a [Pin Complex] wcaps 0x400501: Stereo Pincap 0x0810010: OUT EAPD Pin Default 0x95170110: [Fixed] Speaker at Int Top Conn = Analog, Color = Unknown Pin-ctls: 0x40: OUT Power: 0x0 Connection: 2 0x10* 0x11 Node 0x1b [Pin Complex] wcaps 0x400500: Mono Pincap 0x0810010: OUT EAPD Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x40: OUT Power: 0x0 Connection: 2 0x10* 0x11 Node 0x1c [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x0810: OUT Pin Default 0x224511f0: [Jack] SPDIF Out at Sep Front Conn = Optical, Color = Black Pin-ctls: 0x00: Power: 0x0 Connection: 1 0x12 Node 0x1d [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: Pincap 0x0820: IN Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Power: 0x0 Node 0x1e [Vendor Defined Widget] wcaps 0xf00000: Mono

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[alsa-devel] Patch for ALC882
by Kip Warner 21 Jan '08

21 Jan '08

So after many months of wrestling with Realtek's technical support, I finally managed to get all four of my speakers to work in my ASUS A7J laptop. The patch was intended to replace alsa-kernel/pci/hda/patch_realtek.c in 1.0.14rc4 and that was the last version I successfully tested it on. Since then, I haven't managed to get all four speaker to work. I am not sure why Realtek never posted this patch, but I figured that perhaps some of the ALSA brains might be able to make sense of it and integrate properly into HEAD. -- Kip Warner -- Software Engineer OpenPGP encrypted/signed mail preferred http://www.thevertigo.com

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[alsa-devel] [PATCH 1/2] hda: STAC92xx mono mixer controls
by Matthew Ranostay 21 Jan '08

21 Jan '08

Added mono mixer volume/switch controls for STAC9205 and STAC925x codecs. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 4177ec41bbef pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Mon Jan 21 14:33:37 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Mon Jan 21 11:37:49 2008 -0500 @@ -572,6 +572,7 @@ static struct hda_verb stac925x_core_ini static struct hda_verb stac925x_core_init[] = { /* set dac0mux for dac converter */ { 0x06, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {} }; @@ -600,6 +601,7 @@ static struct hda_verb stac9205_core_ini static struct hda_verb stac9205_core_init[] = { /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {} }; @@ -748,6 +750,9 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac925x_mixer[] = { STAC_INPUT_SOURCE(1), + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x10, 0x1, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x10, 0x1, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT), @@ -757,6 +762,9 @@ static struct snd_kcontrol_new stac9205_ static struct snd_kcontrol_new stac9205_mixer[] = { STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), + + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x14, 0x1, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x14, 0x1, 0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT),

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[alsa-devel] [PATCH 01/17] Update MAINTAINERS for ALSA SoC
by Mark Brown 21 Jan '08

21 Jan '08

Add myself as a point of contact for the ALSA SoC subsystem and add a reference to the development GIT tree. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- MAINTAINERS | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index b4f611c..10c3af3 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -3514,6 +3514,9 @@ S: Maintained SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT P: Liam Girdwood M: liam.girdwood(a)wolfsonmicro.com +P: Mark Brown +M: broonie(a)opensource.wolfsonmicro.com +T: git opensource.wolfsonmicro.com/linux-2.6-asoc L: alsa-devel(a)alsa-project.org (subscribers-only) S: Supported -- 1.5.3.8

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[alsa-devel] [PATCH] hda: Added mono mux mixer for some STAC92HD7XXX family codecs
by Matthew Ranostay 21 Jan '08

21 Jan '08

Some 92HD7XXX family codecs have a mono mux, added support for selecting the input path to the mono out. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r a035d5e4801c pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Jan 18 15:32:32 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Fri Jan 18 14:53:36 2008 -0500 @@ -131,6 +131,8 @@ struct sigmatel_spec { hda_nid_t *pwr_nids; /* playback */ + struct hda_input_mux *mono_mux; + unsigned int cur_mmux; struct hda_multi_out multiout; hda_nid_t dac_nids[5]; @@ -144,6 +146,7 @@ struct sigmatel_spec { hda_nid_t *dmux_nids; unsigned int num_dmuxes; hda_nid_t dig_in_nid; + hda_nid_t mono_nid; /* pin widgets */ hda_nid_t *pin_nids; @@ -399,6 +402,31 @@ static int stac92xx_mux_enum_put(struct return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); +} + +static int stac92xx_mono_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + return snd_hda_input_mux_info(spec->mono_mux, uinfo); +} + +static int stac92xx_mono_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_mmux; + return 0; +} + +static int stac92xx_mono_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + return snd_hda_input_mux_put(codec, spec->mono_mux, ucontrol, + spec->mono_nid, &spec->cur_mmux); } #define stac92xx_aloopback_info snd_ctl_boolean_mono_info @@ -603,6 +631,16 @@ static struct hda_verb stac9205_core_ini {} }; +#define STAC_MONO_MUX \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Mono Mux", \ + .count = 1, \ + .info = stac92xx_mono_mux_enum_info, \ + .get = stac92xx_mono_mux_enum_get, \ + .put = stac92xx_mono_mux_enum_put, \ + } + #define STAC_INPUT_SOURCE(cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -714,6 +752,7 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), + STAC_MONO_MUX, HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -733,6 +772,7 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), + STAC_MONO_MUX, HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1180,7 +1220,7 @@ static struct snd_pci_quirk stac92hd73xx static unsigned int ref92hd71bxx_pin_configs[10] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, - 0x0181302e, 0x01114010, 0x01a19020, 0x90a000f0, + 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, 0x90a000f0, 0x01452050, }; @@ -3188,6 +3228,16 @@ again: return 0; } +static struct hda_input_mux stac92hd71bxx_mono_mux = { + /* count set later */ + .items = { + { "DAC0", 0x0 }, + { "DAC1", 0x1 }, + /* not an option on all cards */ + { "Mixer", 0x2 }, + }, +}; + static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -3224,10 +3274,12 @@ again: case 0x111d76b7: case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: + stac92hd71bxx_mono_mux.num_items = 2, spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; break; default: + stac92hd71bxx_mono_mux.num_items = 3, spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; } @@ -3237,6 +3289,8 @@ again: spec->gpio_mask = spec->gpio_data = 0x00000001; /* GPIO0 High = EAPD */ + spec->mono_mux = &stac92hd71bxx_mono_mux; + spec->mono_nid = 0x15; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids;

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[alsa-devel] Building for 2.4 kernel..
by marcus hall 21 Jan '08

21 Jan '08

I support a product that is running the 2.4.18 linux kernel, and was trying to install alsa-driver-1.0.15, and ran into a problem. The alsa-kernel/pci/cs5535audio/cs5535audio_pm.c contains: if (pci_save_state(pci)) { ... and a similar line for pci_restore_state(). But, in include/adriver.h, pci_save_state is #defined to snd_pci_compat_save_state, which is declared to return void, so the compiler complains that the void return isn't being ignored. Now, pci_save_state() appears to be supposed to return status, although in the 2.4 kernel at least, it returns a constant 0. Nevertheless, to get things to work, I changed snd_pci_compat_save_state(), snd_pci_orig_save_state(), and the corresponding restore functions to return an int with the following patch. (The other obvious solution would be to remove the test in cs5535audio_pm.c, but pci_save_state() is *supposed* to return an int...) -- Marcus Hall marcus(a)tuells.org --------- Index: alsa-driver/acore/misc_driver.c =================================================================== RCS file: /usr/local/cvs_repos/almond/alsa/alsa-driver/acore/misc_driver.c,v retrieving revision 1.1 retrieving revision 1.2 diff -u -r1.1 -r1.2 --- alsa-driver/acore/misc_driver.c 20 Jan 2008 16:52:35 -0000 1.1 +++ alsa-driver/acore/misc_driver.c 20 Jan 2008 17:14:23 -0000 1.2 @@ -105,14 +105,14 @@ #ifdef CONFIG_PCI #ifndef CONFIG_HAVE_NEW_PCI_SAVE_STATE #ifdef CONFIG_HAVE_PCI_SAVED_CONFIG -void snd_pci_compat_save_state(struct pci_dev *pci) +int snd_pci_compat_save_state(struct pci_dev *pci) { - snd_pci_orig_save_state(pci, pci->saved_config_space); + return snd_pci_orig_save_state(pci, pci->saved_config_space); } EXPORT_SYMBOL(snd_pci_compat_save_state); -void snd_pci_compat_restore_state(struct pci_dev *pci) +int snd_pci_compat_restore_state(struct pci_dev *pci) { - snd_pci_orig_restore_state(pci, pci->saved_config_space); + return snd_pci_orig_restore_state(pci, pci->saved_config_space); } EXPORT_SYMBOL(snd_pci_compat_restore_state); #else /* !CONFIG_HAVE_PCI_SAVED_CONFIG */ @@ -122,7 +122,7 @@ }; static struct saved_config_tbl saved_tbl[16]; -void snd_pci_compat_save_state(struct pci_dev *pci) +int snd_pci_compat_save_state(struct pci_dev *pci) { int i; /* FIXME: mutex needed for race? */ @@ -130,14 +130,15 @@ if (! saved_tbl[i].pci) { saved_tbl[i].pci = pci; snd_pci_orig_save_state(pci, saved_tbl[i].config); - return; + return 0; } } printk(KERN_DEBUG "snd: no pci config space found!\n"); + return 1; } EXPORT_SYMBOL(snd_pci_compat_save_state); -void snd_pci_compat_restore_state(struct pci_dev *pci) +int snd_pci_compat_restore_state(struct pci_dev *pci) { int i; /* FIXME: mutex needed for race? */ @@ -145,10 +146,11 @@ if (saved_tbl[i].pci == pci) { saved_tbl[i].pci = NULL; snd_pci_orig_restore_state(pci, saved_tbl[i].config); - return; + return 0; } } printk(KERN_DEBUG "snd: no saved pci config!\n"); + return 1; } EXPORT_SYMBOL(snd_pci_compat_restore_state); #endif /* CONFIG_HAVE_PCI_SAVED_CONFIG */ Index: alsa-driver/include/adriver.h =================================================================== RCS file: /usr/local/cvs_repos/almond/alsa/alsa-driver/include/adriver.h,v retrieving revision 1.1 retrieving revision 1.2 diff -u -r1.1 -r1.2 --- alsa-driver/include/adriver.h 20 Jan 2008 16:52:37 -0000 1.1 +++ alsa-driver/include/adriver.h 20 Jan 2008 17:14:23 -0000 1.2 @@ -1165,15 +1165,15 @@ #if LINUX_VERSION_CODE >= KERNEL_VERSION(2, 4, 0) #ifdef CONFIG_PCI #ifndef CONFIG_HAVE_NEW_PCI_SAVE_STATE -void snd_pci_compat_save_state(struct pci_dev *pci); -void snd_pci_compat_restore_state(struct pci_dev *pci); -static inline void snd_pci_orig_save_state(struct pci_dev *pci, u32 *buffer) +int snd_pci_compat_save_state(struct pci_dev *pci); +int snd_pci_compat_restore_state(struct pci_dev *pci); +static inline int snd_pci_orig_save_state(struct pci_dev *pci, u32 *buffer) { - pci_save_state(pci, buffer); + return pci_save_state(pci, buffer); } -static inline void snd_pci_orig_restore_state(struct pci_dev *pci, u32 *buffer) +static inline int snd_pci_orig_restore_state(struct pci_dev *pci, u32 *buffer) { - pci_restore_state(pci, buffer); + return pci_restore_state(pci, buffer); } #undef pci_save_state

2 1

[alsa-devel] [PATCH] Fix lockdep warning in ASoC machine probe
by Mark Brown 21 Jan '08

21 Jan '08

Don't take the codec mutex during machine probe until we have registered with ALSA, fixing a lockdep warning reported by Dmitry Baryshkov. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Dmitry Baryshkov <dbaryshkov(a)gmail.com> --- sound/soc/soc-core.c | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2eb9b20..78d1662 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1098,7 +1098,6 @@ int snd_soc_register_card(struct snd_soc_device *socdev) struct snd_soc_machine *machine = socdev->machine; int ret = 0, i, ac97 = 0, err = 0; - mutex_lock(&codec->mutex); for(i = 0; i < machine->num_links; i++) { if (socdev->machine->dai_link[i].init) { err = socdev->machine->dai_link[i].init(codec); @@ -1124,12 +1123,14 @@ int snd_soc_register_card(struct snd_soc_device *socdev) goto out; } + mutex_lock(&codec->mutex); #ifdef CONFIG_SND_SOC_AC97_BUS if (ac97) { ret = soc_ac97_dev_register(codec); if (ret < 0) { printk(KERN_ERR "asoc: AC97 device register failed\n"); snd_card_free(codec->card); + mutex_unlock(&codec->mutex); goto out; } } @@ -1142,8 +1143,10 @@ int snd_soc_register_card(struct snd_soc_device *socdev) err = device_create_file(socdev->dev, &dev_attr_codec_reg); if (err < 0) printk(KERN_WARNING "asoc: failed to add codec sysfs entries\n"); -out: + mutex_unlock(&codec->mutex); + +out: return ret; } EXPORT_SYMBOL_GPL(snd_soc_register_card); -- 1.5.3.8

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Re: [alsa-devel] [alsa-cvslog] alsa-driver: Added CONFIG_HAVE_INIT_UTSNAME test
by Takashi Iwai 21 Jan '08

21 Jan '08

At Mon, 21 Jan 2008 11:48:06 +0100 (CET), Jaroslav Kysela wrote: > > changeset: 3491:94a0c6eeb244 > tag: tip > user: perex > date: Mon Jan 21 11:48:04 2008 +0100 > files: acore/info_oss.c configure.in include/config.h.in > description: > Added CONFIG_HAVE_INIT_UTSNAME test > > > diff -r f8ead0afa48d -r 94a0c6eeb244 acore/info_oss.c > --- a/acore/info_oss.c Mon Jan 21 10:42:45 2008 +0100 > +++ b/acore/info_oss.c Mon Jan 21 11:48:04 2008 +0100 > @@ -1,6 +1,6 @@ > #define __NO_VERSION__ > #include <linux/version.h> > -#if LINUX_VERSION_CODE < KERNEL_VERSION(2, 6, 19) > +#ifdef CONFIG_HAVE_INIT_UTSNAME > #define init_utsname() (&system_utsname) > #endif > #include "adriver.h" > diff -r f8ead0afa48d -r 94a0c6eeb244 configure.in > --- a/configure.in Mon Jan 21 10:42:45 2008 +0100 > +++ b/configure.in Mon Jan 21 11:48:04 2008 +0100 > @@ -2566,6 +2566,32 @@ if test "$CONFIG_HAVE_GFP_T" = "1"; then > AC_DEFINE(CONFIG_HAVE_GFP_T) > fi > > +dnl Check for init_utsname... > +AC_MSG_CHECKING(for init_utsname) > +gfp_t="0" > +ac_save_CFLAGS="$CFLAGS" > +ac_save_CC="$CC" > +CFLAGS="$KERNEL_CHECK_CFLAGS" > +CC=$KCC > +AC_TRY_COMPILE([ > +#define __KERNEL__ > +#include <linux/config.h> > +#include <linux/utsname.h> > +],[ > + void *x = init_utsname(); > +], I'm afraid it won't work. In C, it's fine to call undeclared functions. It's supposed to be an int function. Takashi

2 1

[alsa-devel] [PATCH] dapm_new_mixer and 32 bytes limitation
by Robert Jarzmik 21 Jan '08

21 Jan '08

Dapm mixer names are truncated to 32 bytes, while alsa control names are 44 bytes wide. Subsequent usage of functions like snd_ctl_find_id fail because of the mismatch. Unless there's a limitation I don't understand, the following patch fixes the limit back to 44 bytes. -- Robert

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[alsa-devel] alsaconf usefulness
by Alexander E. Patrakov 21 Jan '08

21 Jan '08

Hello developers, Debian applies the following patch to alsaconf: Index: alsa-utils-1.0.15/alsaconf/alsaconf.in =================================================================== --- alsa-utils-1.0.15.orig/alsaconf/alsaconf.in 2007-10-21 14:57:02.000000000 +0200 +++ alsa-utils-1.0.15/alsaconf/alsaconf.in 2007-10-21 14:57:02.000000000 +0200 @@ -458,8 +458,16 @@ Advanced Linux Sound Architecture (ALSA) driver. - You should stop all sound applications now." $version) - $DIALOG --msgbox "$msg" 20 63 || acex 0 + You should stop all sound applications now. + + Please notice, that sound configuration with this script + is not needed if you are running testing/sid with an up + to date kernel and your sound device is a supported PCI + one. In that case *udev* is predestinated to load your + driver ;) + But any way, you can finish this procedure with a working + PCI-Sound at the end." $version) + $DIALOG --msgbox "$msg" 24 63 || acex 0 } # FAREWELL The added note is completely valid: in most cases, there is nothing to do, even for ISA-PnP cards (because all self-respecting distributions already contain a rule for loading ISA-PnP drivers). However, the very presence of alsaconf generates a lot of support traffic on forums in the form "I have tried alsaconf and it didn't help" (or worse, ruined a working setup). Also, it cannot deal with more than one sound card. Could you please either apply the above patch (of course, replacing the wording with a distro-neutral one), or just drop this obsolete program and replace it with the "alsa-unmute" script which essentially contains the same as the set_mixers() function (but allows to specify the card to unmute)? -- Alexander E. Patrakov

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