September 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Problems with Intel HDA driver
by Andrius Štikonas 27 Sep '09

27 Sep '09

On my new laptop there were no sound with snd-hda-intel driver, so I tried to change model parameter. It seems that with model=hp-m4 my soundcard mostly works (except that main speakers still play when I plug headphones). I've attached my alsa-info.txt script. -- Andrius Štikonas

4 11

[PATCH 1/1] ALSA: hda - Analog Devices AD1984A add HP Touchsmart model
by Miguel de Barros 27 Sep '09

27 Sep '09

Reference: ALSA bug #0004614 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4614 port-A (0x11) - front hp-out port-D (0x12) - rear line out port-E (0x1c) - front mic-in port-F (0x16) - Internal speakers digital-mic (0x17) - Internal mic init verbs, mixers, jack sensing and PCI_QUIRK to support this hardware Signed-off-by: Miguel de Barros <miguel.de.barros(a)bluewin.ch> diff --git a/Documentation/HD-Audio-Models.txt b/Documentation/HD-Audio-Models.txt index abffd41..9e850ed 100644 --- a/Documentation/HD-Audio-Models.txt +++ b/Documentation/HD-Audio-Models.txt @@ -208,6 +208,7 @@ AD1884A / AD1883 / AD1984A / AD1984B laptop laptop with HP jack sensing mobile mobile devices with HP jack sensing thinkpad Lenovo Thinkpad X300 + touchsmart HP Touchsmart AD1884 ====== diff --git a/pci/hda/patch_analog.c b/pci/hda/patch_analog.c index 215e72a..2d603f6 100644 --- a/pci/hda/patch_analog.c +++ b/pci/hda/patch_analog.c @@ -4032,6 +4032,127 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec) } /* + * HP Touchsmart + * port-A (0x11) - front hp-out + * port-B (0x14) - unused + * port-C (0x15) - unused + * port-D (0x12) - rear line out + * port-E (0x1c) - front mic-in + * port-F (0x16) - Internal speakers + * digital-mic (0x17) - Internal mic + */ + +static struct hda_verb ad1984a_touchsmart_verbs[] = { + /* DACs; unmute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + /* Port-A (HP) mixer - route only from analog mixer */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Port-A (HP) pin - always unmuted */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Port-E (int speaker) mixer - route only from analog mixer */ + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03}, + /* Port-E pin */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /* Port-F (int speaker) mixer - route only from analog mixer */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-F pin */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* capture sources */ + /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* unsolicited event for pin-sense */ + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, + /* allow to touch GPIO1 (for mute control) */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ + /* internal mic - dmic */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* set magic COEFs for dmic */ + {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, + {0x01, AC_VERB_SET_PROC_COEF, 0x08}, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), +/* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = ad1884a_mobile_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), + { } /* end */ +}; + +/* switch to external mic if plugged */ +static void ad1984a_touchsmart_automic(struct hda_codec *codec) +{ + if (snd_hda_codec_read(codec, 0x1c, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000) { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x4); + } else { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x5); + } +} + + +/* unsolicited event for HP jack sensing */ +static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case AD1884A_HP_EVENT: + ad1884a_hp_automute(codec); + break; + case AD1884A_MIC_EVENT: + ad1984a_touchsmart_automic(codec); + break; + } +} + +/* initialize jack-sensing, too */ +static int ad1984a_touchsmart_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1884a_hp_automute(codec); + ad1984a_touchsmart_automic(codec); + return 0; +} + + +/* */ enum { @@ -4039,6 +4160,7 @@ enum { AD1884A_LAPTOP, AD1884A_MOBILE, AD1884A_THINKPAD, + AD1984A_TOUCHSMART, AD1884A_MODELS }; @@ -4047,6 +4169,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = { [AD1884A_LAPTOP] = "laptop", [AD1884A_MOBILE] = "mobile", [AD1884A_THINKPAD] = "thinkpad", + [AD1984A_TOUCHSMART] = "touchsmart", }; static struct snd_pci_quirk ad1884a_cfg_tbl[] = { @@ -4059,6 +4182,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), + SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART), {} }; @@ -4142,6 +4266,21 @@ static int patch_ad1884a(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event; codec->patch_ops.init = ad1984a_thinkpad_init; break; + case AD1984A_TOUCHSMART: + spec->mixers[0] = ad1984a_touchsmart_mixers; + spec->init_verbs[0] = ad1984a_touchsmart_verbs; + spec->multiout.dig_out_nid = 0; + codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event; + codec->patch_ops.init = ad1984a_touchsmart_init; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); + break; } return 0; -- 1.6.0.4 --=-J56U1/hvhPtbcxv8DR5c--

1 0

[PATCH 1/1] ALSA: hda - Analog Devices AD1984A add HP Touchsmart model modified: Documentation/HD-Audio-Models.txt modified: pci/hda/patch_analog.c
by Miguel de Barros 27 Sep '09

27 Sep '09

Signed-off-by: Miguel de Barros <miguel.de.barros(a)bluewin.ch> diff --git a/Documentation/HD-Audio-Models.txt b/Documentation/HD-Audio-Models.txt index abffd41..9e850ed 100644 --- a/Documentation/HD-Audio-Models.txt +++ b/Documentation/HD-Audio-Models.txt @@ -208,6 +208,7 @@ AD1884A / AD1883 / AD1984A / AD1984B laptop laptop with HP jack sensing mobile mobile devices with HP jack sensing thinkpad Lenovo Thinkpad X300 + touchsmart HP Touchsmart AD1884 ====== diff --git a/pci/hda/patch_analog.c b/pci/hda/patch_analog.c index 215e72a..7df8a08 100644 --- a/pci/hda/patch_analog.c +++ b/pci/hda/patch_analog.c @@ -4032,6 +4032,143 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec) } /* + * HP Touchsmart + * port-A (0x11) - front hp-out + * port-B (0x14) - unused + * port-C (0x15) - unused + * port-D (0x12) - rear line out + * port-E (0x1c) - front mic-in + * port-F (0x16) - Internal speakers + * digital-mic (0x17) - Internal mic + */ + +static struct hda_verb ad1984a_touchsmart_verbs[] = { + /* DACs; unmute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + /* Port-A (HP) mixer - route only from analog mixer */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Port-A (HP) pin - always unmuted */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Port-E (int speaker) mixer - route only from analog mixer */ + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03}, + /* Port-E pin */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /* Port-F (int speaker) mixer - route only from analog mixer */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-F pin */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* capture sources */ + /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* unsolicited event for pin-sense */ + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, + /* allow to touch GPIO1 (for mute control) */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ + /* internal mic - dmic */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* set magic COEFs for dmic */ + {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, + {0x01, AC_VERB_SET_PROC_COEF, 0x08}, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), +/* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = ad1884a_mobile_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_input_mux ad1984a_touchsmart_capture_source = { + .num_items = 3, + .items = { + { "Mix", 0x3 }, /* Mixer */ + { "Mic", 0x4 }, /* port-E */ + { "Internal Mic", 0x5 }, /* Digital Mic */ + }, +}; + +/* switch to external mic if plugged */ +static void ad1984a_touchsmart_automic(struct hda_codec *codec) +{ + if (snd_hda_codec_read(codec, 0x1c, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000) { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x4); + } else { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x5); + } +} + + +/* unsolicited event for HP jack sensing */ +static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case AD1884A_HP_EVENT: + ad1884a_hp_automute(codec); + break; + case AD1884A_MIC_EVENT: + ad1984a_touchsmart_automic(codec); + break; + } +} + +/* initialize jack-sensing, too */ +static int ad1984a_touchsmart_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1884a_hp_automute(codec); + ad1984a_touchsmart_automic(codec); + return 0; +} + + +/* */ enum { @@ -4039,6 +4176,7 @@ enum { AD1884A_LAPTOP, AD1884A_MOBILE, AD1884A_THINKPAD, + AD1984A_TOUCHSMART, AD1884A_MODELS }; @@ -4047,6 +4185,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = { [AD1884A_LAPTOP] = "laptop", [AD1884A_MOBILE] = "mobile", [AD1884A_THINKPAD] = "thinkpad", + [AD1984A_TOUCHSMART] = "touchsmart", }; static struct snd_pci_quirk ad1884a_cfg_tbl[] = { @@ -4059,6 +4198,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), + SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART), {} }; @@ -4142,6 +4282,22 @@ static int patch_ad1884a(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event; codec->patch_ops.init = ad1984a_thinkpad_init; break; + case AD1984A_TOUCHSMART: + spec->mixers[0] = ad1984a_touchsmart_mixers; + spec->init_verbs[0] = ad1984a_touchsmart_verbs; + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1984a_touchsmart_capture_source; + codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event; + codec->patch_ops.init = ad1984a_touchsmart_init; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); + break; } return 0; -- 1.6.0.4 --=-w/QltLfzo6NHdpL3XZCU--

1 0

Re: [alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 26 Sep '09

26 Sep '09

Hi Peter, I forgot to tell you that if I deliberately misconfigure the McBXP_CLKS pin so that it's an output, the FSX -> DACLRC/ADCLRC signal stops. This should indicate that the CLKOUT signal is actually making it to the CLKS pin on the OMAP. Right? Rick

2 1

Re: [alsa-devel] [Pkg-alsa-devel] Bug#527074: /etc/init.d/alsa-utils: `/etc/init.d/alsa-utils stop` triggers amixer error
by Elimar Riesebieter 26 Sep '09

26 Sep '09

* Piotr Lewandowski [090925 20:54 +0200] > Hi alsa-devel@! > > One of the maintainers of alsa-utils package in Debian advised me to > ask for your help. I've encountered following problem: > > #v+ > $ amixer -c 0 -q set Mic 0% mute > amixer: Invalid command! > #v- > > This command is invoked from /etc/init.d/alsa-utils script during > system shutdown. Do you have any ideas how to fix that? > > Here is some basic info about my hardware: > > #v+ > $ lspci -v | sed -ne '/Audio/,/^$/p' > 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High > Definition Audio Controller (rev 01) > Subsystem: Hewlett-Packard Company Device 30a2 > Flags: bus master, fast devsel, latency 0, IRQ 16 > Memory at f4580000 (64-bit, non-prefetchable) [size=16K] > Capabilities: <access denied> > Kernel driver in use: HDA Intel > > $ cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.18a. 1. It would be an option to insert: 'echo option "snd-hda-intel enable_msi=1"' >> /etc/modprobe.d/local.conf 2. It is a good idea to update to alsa-source 1.0.21-2 and build biuld your driver up from there. Elimar -- Never make anything simple and efficient when a way can be found to make it complex and wonderful ;-)

1 0

[alsa-devel] [PATCH] [PATCH] [ASOC] pxa-ssp increase max_channels to 8
by Graeme Gregory 25 Sep '09

25 Sep '09

When running in TDM mode there can be more than 2 channels used. Datasheet has figures for upto 8 channels so increase max_channels on all SSP interfaces to this figure. Signed-off-by: Graeme Gregory <dp(a)xora.org.uk> --- sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 19c4540..e8a5629 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -767,13 +767,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -787,13 +787,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -808,13 +808,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -829,13 +829,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, -- 1.6.3.3

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Re: [alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 25 Sep '09

25 Sep '09

Hi Peter, Thanks very much for the help. > I think the problem is here. > The MCBSP_CLKS pin is connected to CLK256FS pin of TWL4030 (or TPA). > Since you are not using the TWL codec, there will be no clock on the MCBSP_CLKS > pin, thus no functional clock for McBSP, thus no bits are shifted in or out. > You can try to use the internal functional clock: > > ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_FCLK, 0, > SND_SOC_CLOCK_IN); I have checked the CLKOUT (WM8731) -> McBSP_CLKS (OMAP3) pin and it has a 12288000 clock on it. I've tried using the internal clock as you show above, just to see if I get anything out on DX but I get the same thing: DX just goes low for the "aplay" duration. Any idea why I'm not getting DMA interrupts? This would indeed cause no data at DX. omap2_dma_irq_handler() never gets called but omap_init_dma() and omap_start_dma() get called. Do you think there may be any issues with my DMA code (specifically arch/arm/plat-omap/dma.c). As I mentioned, I'm at "2.6.29-rc3-omap1" Thanks, Rick

1 0

[alsa-devel] [PATCH] [ASOC] pxa-ssp increase max_channels to 8
by Graeme Gregory 25 Sep '09

25 Sep '09

When running in TDM mode there can be more than 2 channels used. Datasheet has figures for upto 8 channels so increase max_channels on all SSP interfaces to this figure. --- sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 19c4540..e8a5629 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -767,13 +767,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -787,13 +787,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -808,13 +808,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -829,13 +829,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, -- 1.6.3.3

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[alsa-devel] Mic stops working after installation (ALC861)
by remrot＠gmail.com 25 Sep '09

25 Sep '09

/* Sorry for posting this to the devel list, but the problem seems to be trickier than I first thought * * and I don't have much time left to fix it (not my machine)! Thanks... */ Hi! I've got a very strange problem with a ALC861 soundchip on a friend's Toshiba A100-507 Laptop. Sound output and input work perfectly within a live session using the Ubuntu Jaunty and Karmic (Alpha6) CDs but immediately stops working after installing the system. Currently I've installed the latest Karmic containing kernel 2.6.31 and ALSA 1.0.20. I've attached the output of the alsa-info script as well as the output of 'pacmd list' for both systems (live and installed). Right after the installation the sound stopped working and I couldn't record anything. Alsamixer showed only output devices, no capture devices at all (everything has been correct within the live session). In 'dmesg' I found the following error messages: [ 279.924019] hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x301f000a [ 280.928019] hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x301f000a After doing some research I found this guide (thx Mr. Iwai): http://www.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.html#_cod… And indeed, adding 'probe_mask=8' fixed the sound output and brought back the capture devices (remember that they were there within the live session without that option). I've also tried the 'model=toshiba' option (combined with 'probe_mask' and standalone) but that didn't help at all (no capture devices and no sound). The current situation: I've already posted to the PulseAudio mailing ist and got some help from Mr. Colin Guthrie but we haven't found a solution (everything seems OK regarding PA). I still can't record anything within the installed system. I tried audacity, alsarec and the gnome-sound-recorder without luck. When recording in audacity, the timeline cursor doesn't move, it only flickers. In gnome-recorder, recording stops right after it has started and the application hangs on exit. PA's volume meter doesn't show anything either, it is empty. I compared the configurations on both systems but I see no differences that are not related to the 'probe_mask' option... I really believe that there is a solution because everything works in a live session. Can anybody help me with this? Regards JK

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[alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 25 Sep '09

25 Sep '09

Hi, I'm working on getting a WM8731 (in SLAVE) working using I2S on omap3 McBSP2 (in MASTER). Here is the hardware: http://www.efn.org/~rick/pub/wm8731.jpg Here are the PADCONF's and the pins they hook to on the WM8731: /*Audio Interface for csb740 */\ MUX_VAL(CP(McBSP2_FSX), (OFF_OUT_PD | IDIS | PTU | EN | M0)) /* McBSP2_FSX -> I2S_FRM -> LRCLK -> DACLRC and ADCLRC*/\ MUX_VAL(CP(McBSP2_CLKX), (OFF_OUT_PD | IDIS | PTU | EN | M0)) /* McBSP2_CLKX -> I2S_CLK -> I2S_BCLK -> BCLK */\ MUX_VAL(CP(McBSP2_DR), (OFF_IN_PD | IEN | PTD | DIS | M0)) /* McBSP2_DR <- I2S_RXD <- ACDDAT */\ MUX_VAL(CP(McBSP2_DX), (OFF_OUT_PD | IDIS | PTU | EN | M0)) /* McBSP2_DX -> I2S_TXD -> DACDAT */\ MUX_VAL(CP(McBSP_CLKS), (OFF_IN_PD | IEN | PTD | DIS | M0)) /* McBSP_CLKS <- I2S_MCLK <- CLKOUT */\ My kernel is "2.6.29-rc3-omap1", here's what I see at bootup: 1 WM8731 Audio Codec 0.13<6>asoc: WM8731 <-> omap-mcbsp-dai-0 mapping ok 2 ALSA device list: 3 #0: omap3csb740 (WM8731) ... 4 beagle:~# aplay test48000.wav 5 Playing WAVE 'test48000.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo 6 aplay: pcm_write:1269: write error: Input/output error I get no audio output, other than a "click" when the 8731 is initialized upon boot. When I look at the signals the McBSP2_DX line just goes low for the duration of the "aplay", it does not toggle. The other lines look fine. The file is at http://www.efn.org/~rick/pub/omap3csb740.c but roughly, here's what I'm doing: static struct snd_soc_dai_link omap3csb740_dai = { ... .cpu_dai = &omap_mcbsp_dai[0], /* use 1st one */ static int __init omap3csb740_soc_init(void) ... *(unsigned int *)omap_mcbsp_dai[0].private_data = 1; /* McBSP2 */ *(unsigned int *)omap3csb740_dai.cpu_dai->private_data = 1; /* McBSP2 debug only? */ static int omap3csb740_hw_params(struct snd_pcm_substream *substream, ... ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); ret = snd_soc_dai_set_sysclk(codec_dai, 0, 12288000, SND_SOC_CLOCK_IN); ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0, SND_SOC_CLOCK_IN); ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 256); Any help is greatly appreciated. Rick Bronson

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