September 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] alsa on Apple Mac Pro Tower ( macPro3, 1, quad core, xenon)
by S. Aguinaga 15 Sep '08

15 Sep '08

I downloaded the latest development drivers, lib, and utils (1.0.18rc3) and compiled and installed it again. Next I made sure that the /etc/modprobe.d/sound had the following line: options snd-hda-intel model=imac24 and first I checked the gui tool called "PulseAudio Volume Control" from my "multimedia" section, and it crashes! Next I tried the aplay somesound.wav file from the command line, it my Mac Pro sings!!! I don't know what the PulseAudio issues are, but ... oh well!! Next I checked that Amrok works using the alsa engine, and it does!! If I plug in a headphone set, and I play music again it comes out of the MacPro speaker only, but no from the headphones. Do you have a handle this, if you guys have some links for me to read, please send them my way! Thank you. // Sal Aguinaga // Northwestern University // Evanston, IL ----- Original Message ---- From: Takashi Iwai <tiwai(a)suse.de> To: S. Aguinaga <sa_jr(a)ieee.org> Cc: alsa-devel(a)alsa-project.org Sent: Tuesday, August 19, 2008 5:48:47 AM Subject: Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45 At Mon, 18 Aug 2008 10:26:42 -0700 (PDT), S. Aguinaga wrote: > > ============================================== > Trying to figure out where to add this "model=macpro" > and it looks like it should be in a file called modprobe.conf or an options file under /etc/modprobe.d/ > > but I don't have (with my installation -- fedora {9} ) those files, do you have any input on this? You can create any file, typically, /etc/modprobe.d/sound or so. Put the following line there options snd-hda-intel model=macpro Then reload the module, check the sysfs file /sys/modules/snd_hda_intel/parameters/macpro to see whether the specified option was really passed properly. Takashi > below are the only files I have. > > /etc/modprobe.d/ > > anaconda > blacklist-visor > i915modeset > blacklist > floppy-pnp > modprobe.conf.dist > > > ============================================== > Subject: Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45 > > At Sat, 16 Aug 2008 06:15:46 -0700 (PDT), > S. Aguinaga wrote: > > > > S. Aguinaga wrote: > > > > > > Hello Stan, > > > > > > This is awesome I did find the problem it's listed in the /var/log/messages under ALSA. > > > > > > 1. When I bring up the "Pulse Volume Control" gui, the first tab "Playback" comes up empty, and grayed-out is "No Streams Available" > > > 2. aplay does not produce sound, period! > > > 3. Amrok's welcome sample does not produce sound out of the std tower speaker or the headphone jack. > > > > > > I didn't notice anything strange on the output of the alsa-info.sh file, but when I checked the var/log/messages, this is what I found: > > > > > > sudo cat /var/log/messages | grep ALSA > > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 > > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x1 > > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Volume, skipped > (snip) > > These messages are harmless. It's just information from the driver, > enabled only when you build it with CONFIG_SND_DEBUG_VERBOSE=y. > > > I think your problem is a misidentification of your sound chip. The > > model being used for your 889A appears to be incompatible, thus all > > the errors above. It appears that the macpro ALC885 is being selected > > before it gets to the choice for the ALC889A. Perhaps a developer can > > look into this. > > Well, ALC882, ALC885 and ALC889A are almost compatible. In the > driver, they are all handled in the same routine. > > There are a few models for intel macs with ALC88x. Try to specify > either model=macpro, model=mbp3 or model=imac24 as the module option > of snd-hda-intel driver. > > > Takashi > > > > >

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[alsa-devel] Dev: Alsa sequencer problem (Why ???)
by Gilbux (free) 15 Sep '08

15 Sep '08

Hello, I am trying to write an alsa sequencer. So I read as I can some of your tutorials, and I don't really understand what I am doing wrong, please help me. I want to send scheduled event with the queuing system, but with RTC timer the queue doesn't start, and with the other timer the queue start but no events are sent. I have joined to this email a small code, with the same kind of functions I am using, and some /proc/asound/seq/* outputs. Regards, Gilbux /proc/asound/seq/* before starting the queue : Client info cur clients : 2 peak clients : 3 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Client 128 : "TestQSchedEv" [User] Port 0 : "OutputPort" (R-e-) Output pool : Pool size : 500 Cells in use : 16 Peak cells in use : 16 Alloc success : 16 Alloc failures : 0 queue 0: [TQSE_Queue] owned by client : 128 lock status : Locked queued time events : 0 queued tick events : 16 timer state : Stopped timer PPQ : 192 current tempo : 500000 current BPM : 120 current time : 0.000000000 s current tick : 0 Timer for queue 0 : RTC timer Period time : 0.000000000 Skew : 65536 / 65536 ----- /proc/asound/seq/* after starting the queue : Client info cur clients : 2 peak clients : 3 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Client 128 : "TestQSchedEv" [User] Port 0 : "OutputPort" (R-e-) Output pool : Pool size : 500 Cells in use : 0 Peak cells in use : 16 Alloc success : 16 Alloc failures : 0 queue 0: [TQSE_Queue] owned by client : 128 lock status : Locked queued time events : 0 queued tick events : 0 timer state : Running timer PPQ : 192 current tempo : 500000 current BPM : 120 current time : 0.000000000 s current tick : 0 Timer for queue 0 : RTC timer Period time : 0.000976562 Skew : 65536 / 65536

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[alsa-devel] [PATCH] ALSA: HDA: patch_analog: Fix SPDIF output on AD1989B
by robbat2＠gentoo.org 15 Sep '08

15 Sep '08

From: Robin H. Johnson <robbat2(a)gentoo.org> The SPDIF pins for AD1989 are not enabled by default. Set OUT bit so that they actually work. Also initialize the HDMI SPDIF at the same time. Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index b037324..42b1a23 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2255,8 +2255,12 @@ static struct hda_verb ad1988_spdif_init_verbs[] = { /* AD1989 has no ADC -> SPDIF route */ static struct hda_verb ad1989_spdif_init_verbs[] = { - /* SPDIF out pin */ + /* SPDIF-1 out pin */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + /* SPDIF-2/HDMI out pin */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ { } }; -- 1.6.0.1

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[alsa-devel] [PATCH] hda: 92hd73xx fixes
by Matthew Ranostay 14 Sep '08

14 Sep '08

Added support for defining the number of amps and the creation of the private_dimux dynamically for the 92hd73xx codec family. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index f41ba17..fff27cc 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -156,13 +156,11 @@ struct sigmatel_spec { unsigned int num_pwrs; unsigned int *pwr_mapping; hda_nid_t *pwr_nids; - hda_nid_t *amp_nids; hda_nid_t *dac_list; /* playback */ struct hda_input_mux *mono_mux; struct hda_input_mux *amp_mux; - unsigned int cur_amux; unsigned int cur_mmux; struct hda_multi_out multiout; hda_nid_t dac_nids[5]; @@ -201,6 +199,9 @@ struct sigmatel_spec { unsigned int cur_mux[3]; struct hda_input_mux *sinput_mux; unsigned int cur_smux[2]; + unsigned int cur_amux; + hda_nid_t *amp_nids; + unsigned int num_amps; unsigned int powerdown_adcs; /* i/o switches */ @@ -247,8 +248,9 @@ static hda_nid_t stac92hd73xx_adc_nids[2] = { 0x1a, 0x1b }; -static hda_nid_t stac92hd73xx_amp_nids[4] = { - 0x0b, 0x0c, 0x0e, 0 +#define DELL_M6_AMP 2 +static hda_nid_t stac92hd73xx_amp_nids[3] = { + 0x0b, 0x0c, 0x0e }; #define STAC92HD73XX_NUM_DMICS 2 @@ -963,29 +965,33 @@ static struct snd_kcontrol_new stac9200_mixer[] = { { } /* end */ }; +#define DELL_M6_MIXER 6 static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), - + /* start of config #1 */ HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + + /* start of config #2 */ + HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + { } /* end */ }; @@ -2956,17 +2962,19 @@ static int stac92xx_auto_create_amp_input_ctls(struct hda_codec *codec) struct hda_input_mux *amp_mux = &spec->private_amp_mux; int i, err; - for (i = 0; i < ARRAY_SIZE(stac92xx_amp_labels); i++) { + for (i = 0; i < spec->num_amps; i++) { amp_mux->items[amp_mux->num_items].label = stac92xx_amp_labels[i]; amp_mux->items[amp_mux->num_items].index = i; amp_mux->num_items++; } - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_MUX, - "Amp Selector Capture Switch", 0); - if (err < 0) - return err; + if (spec->num_amps > 1) { + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_MUX, + "Amp Selector Capture Switch", 0); + if (err < 0) + return err; + } return stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_VOL, "Amp Capture Volume", HDA_COMPOSE_AMP_VAL(spec->amp_nids[0], 3, 0, HDA_INPUT)); @@ -3342,12 +3350,12 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (err < 0) return err; } - if (spec->amp_nids) { + if (spec->num_amps) { err = stac92xx_auto_create_amp_input_ctls(codec); if (err < 0) return err; } - if (spec->num_dmics > 0) + if (spec->num_dmics > 0 && !spec->dinput_mux) if ((err = stac92xx_auto_create_dmic_input_ctls(codec, &spec->autocfg)) < 0) return err; @@ -3375,8 +3383,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->mixers[spec->num_mixers++] = spec->kctl_alloc; spec->input_mux = &spec->private_imux; - if (!spec->dinput_mux) - spec->dinput_mux = &spec->private_dimux; + spec->dinput_mux = &spec->private_dimux; spec->sinput_mux = &spec->private_smux; spec->mono_mux = &spec->private_mono_mux; spec->amp_mux = &spec->private_amp_mux; @@ -3975,9 +3982,9 @@ static struct hda_input_mux stac92hd73xx_dmux = { .num_items = 4, .items = { { "Analog Inputs", 0x0b }, - { "CD", 0x08 }, { "Digital Mic 1", 0x09 }, { "Digital Mic 2", 0x0a }, + { "CD", 0x08 }, } }; @@ -4048,12 +4055,14 @@ again: spec->dmux_nids = stac92hd73xx_dmux_nids; spec->smux_nids = stac92hd73xx_smux_nids; spec->amp_nids = stac92hd73xx_amp_nids; + spec->num_amps = ARRAY_SIZE(stac92hd73xx_amp_nids); spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); - spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids); - spec->dinput_mux = &stac92hd73xx_dmux; + memcpy(&spec->private_dimux, &stac92hd73xx_dmux, + sizeof(stac92hd73xx_dmux)); + /* GPIO0 High = Enable EAPD */ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; @@ -4061,11 +4070,18 @@ again: switch (spec->board_config) { case STAC_DELL_M6: spec->init = dell_eq_core_init; + spec->num_smuxes = 0; + spec->multiout.hp_nid = + spec->multiout.dac_nids[spec->multiout.num_dacs - 1]; + spec->mixer = &stac92hd73xx_6ch_mixer[DELL_M6_MIXER]; + spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP]; + spec->num_amps = 1; switch (codec->subsystem_id) { case 0x1028025e: /* Analog Mics */ case 0x1028025f: stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); spec->num_dmics = 0; + spec->private_dimux.num_items = 1; break; case 0x10280271: /* Digital Mics */ case 0x10280272: @@ -4075,18 +4091,22 @@ again: case 0x10280255: stac92xx_set_config_reg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; + spec->private_dimux.num_items = 2; break; case 0x10280256: /* Both */ case 0x10280057: stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); stac92xx_set_config_reg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; + spec->private_dimux.num_items = 2; break; } break; default: spec->num_dmics = STAC92HD73XX_NUM_DMICS; + spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids); } + spec->dinput_mux = &spec->private_dimux; spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids;

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[alsa-devel] alsa callback not async - Second post to this list
by William Estrada 14 Sep '08

14 Sep '08

Hi guys, I am cross posting this because I am not sure which list it belongs to. I have a test program I wrote to help me learn ALSA APIs, It work to a degree. It is a simple Capture/Playback interface. My problem is that when the Callback starts, it does not allow any cycles back to the main program. The main program waits for console input to tell the callback when to stop. Well it does not accept any console input when the callback is running. Any ideas?? The source is here: http://64.124.13.3/_ALSA_/Echo/ -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

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[alsa-devel] [PATCH take 2] ALSA: HDA: patch_analog: Quirk for Asus P5Q Premium/Pro boards.
by robbat2＠gentoo.org 14 Sep '08

14 Sep '08

From: Robin H. Johnson <robbat2(a)gentoo.org> Use 6STACK_DIG for the AD2000BX variant of the AD1989B chip used by Asus on their Asus P5Q Premium and Pro boards. Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 42b1a23..2b00c4a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2884,6 +2884,7 @@ static struct snd_pci_quirk ad1988_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG), + SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG), {} }; -- 1.6.0.1

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[alsa-devel] [PATCH] ALSA: HDA: patch_analog: Quirk for Asus P5Q Premium/Pro boards.
by robbat2＠gentoo.org 14 Sep '08

14 Sep '08

From: Robin H. Johnson <robbat2(a)orbis-terrarum.net> Use 6STACK_DIG for the AD2000BX variant of the AD1989B chip used by Asus on their Asus P5Q Premium and Pro boards. Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 42b1a23..2b00c4a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2884,6 +2884,7 @@ static struct snd_pci_quirk ad1988_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG), + SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG), {} }; -- 1.6.0.1

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[alsa-devel] [PATCH] ALSA: HDA: hda_local: Less magic numbers.
by robbat2＠gentoo.org 14 Sep '08

14 Sep '08

From: Robin H. Johnson <robbat2(a)gentoo.org> Explain some of the magic numbers I saw while trying to fix the AD1989 SPDIF issue. Maybe should just use the expanded form directly in the verbs? Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index d688f50..7957fef 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -368,12 +368,15 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, #define AMP_OUT_UNMUTE 0xb000 #define AMP_OUT_ZERO 0xb000 /* pinctl values */ -#define PIN_IN 0x20 -#define PIN_VREF80 0x24 -#define PIN_VREF50 0x21 -#define PIN_OUT 0x40 -#define PIN_HP 0xc0 -#define PIN_HP_AMP 0x80 +#define PIN_IN (AC_PINCTL_IN_EN) +#define PIN_VREFHIZ (AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ) +#define PIN_VREF50 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_50) +#define PIN_VREFGRD (AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD) +#define PIN_VREF80 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_80) +#define PIN_VREF100 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_100) +#define PIN_OUT (AC_PINCTL_OUT_EN) +#define PIN_HP (AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN) +#define PIN_HP_AMP (AC_PINCTL_HP_EN) /* * get widget capabilities -- 1.6.0.1

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[alsa-devel] [PATCH] ALSA: HDA: patch_analog: Implement multiple outputs for AD1988
by robbat2＠gentoo.org 14 Sep '08

14 Sep '08

From: Robin H. Johnson <robbat2(a)gentoo.org> Implement the slave_dig_outs for AD1989 cards. Untested due to lack of hardware. Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0899b6b..b037324 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1826,9 +1826,14 @@ static hda_nid_t ad1988_capsrc_nids[3] = { 0x0c, 0x0d, 0x0e }; -#define AD1988_SPDIF_OUT 0x02 +#define AD1988_SPDIF_OUT 0x02 +#define AD1988_SPDIF_OUT_HDMI 0x0b #define AD1988_SPDIF_IN 0x07 +static hda_nid_t ad1989b_slave_dig_outs[2] = { + AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI +}; + static struct hda_input_mux ad1988_6stack_capture_source = { .num_items = 5, .items = { @@ -2143,6 +2148,7 @@ static struct snd_kcontrol_new ad1988_spdif_in_mixers[] = { static struct snd_kcontrol_new ad1989_spdif_out_mixers[] = { HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("HDMI Playback Volume", 0x1d, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -2981,6 +2987,7 @@ static int patch_ad1988(struct hda_codec *codec) ad1989_spdif_out_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1989_spdif_init_verbs; + codec->slave_dig_outs = ad1989b_slave_dig_outs; } else { spec->mixers[spec->num_mixers++] = ad1988_spdif_out_mixers; -- 1.6.0.1

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[alsa-devel] [PATCH] ALSA: HDA: hda_proc: Fix printf format specifier
by robbat2＠gentoo.org 14 Sep '08

14 Sep '08

From: Robin H. Johnson <robbat2(a)gentoo.org> The Pincap output had a typod format specifier, leading to an extraneous "08" in the output, which is a reserved bit of the Vref field, and was really confused :-). Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 4927c4b..743d779 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -216,7 +216,7 @@ static void print_pin_caps(struct snd_info_buffer *buffer, unsigned int caps, val; caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); - snd_iprintf(buffer, " Pincap 0x08%x:", caps); + snd_iprintf(buffer, " Pincap 0x%08x:", caps); if (caps & AC_PINCAP_IN) snd_iprintf(buffer, " IN"); if (caps & AC_PINCAP_OUT) -- 1.6.0.1

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