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

[alsa-devel] [PATCH] hda: add bounds checking for the codec command fields
by Wu Fengguang 17 Jul '09

17 Jul '09

A recent bug involves passing auto detected >0x7f NID to codec command, creating an invalid codec addr field, and finally lead to cmd timeout and fall back into single command mode. Jaroslav fixed that bug in alc880_parse_auto_config(). It would be safer to further check the bounds of all cmd fields. Cc: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 462e2ce..7d09650 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -150,6 +150,16 @@ make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, { u32 val; + if ((direct & ~1) || (nid & ~0x7f) || + (verb & ~0xfff) || (parm & ~0xff)) { + printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x:%x\n", + codec->addr, direct, nid, verb, parm); + direct &= 1; + nid &= 0x7f; + verb &= 0xfff; + parm &= 0xff; + } + val = (u32)(codec->addr & 0x0f) << 28; val |= (u32)direct << 27; val |= (u32)nid << 20;

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Re: [alsa-devel] [PATCH]HDA: Add support for new AMD HD audio devices
by Takashi Iwai 17 Jul '09

17 Jul '09

At Fri, 17 Jul 2009 11:32:32 +0800, Andiry Brienza wrote: > > Add support for new AMD HD audio devices. Use generic driver to detect HD audio > devices with Vendor ID AMD. > > Signed-off-by: Andiry Xu <andiry.xu(a)amd.com> Thanks, applied now. Takashi > diff -ruN a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > --- a/sound/pci/hda/hda_intel.c 2009-07-17 10:46:24.000000000 +0800 > +++ b/sound/pci/hda/hda_intel.c 2009-07-17 11:04:45.000000000 +0800 > @@ -2604,11 +2604,15 @@ > /* this entry seems still valid -- i.e. without emu20kx chip */ > { PCI_DEVICE(0x1102, 0x0009), .driver_data = AZX_DRIVER_GENERIC }, > #endif > - /* AMD Generic, PCI class code and Vendor ID for HD Audio */ > + /* AMD/ATI Generic, PCI class code and Vendor ID for HD Audio */ > { PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID), > .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, > .class_mask = 0xffffff, > .driver_data = AZX_DRIVER_GENERIC }, > + { PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_ANY_ID), > + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, > + .class_mask = 0xffffff, > + .driver_data = AZX_DRIVER_GENERIC }, > { 0, } > }; > MODULE_DEVICE_TABLE(pci, azx_ids); > >

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[alsa-devel] Callbacks from opening or closing alsa streams
by Baek Chang 17 Jul '09

17 Jul '09

Hi, I was wondering if it was possible to get callbacks from opening and closing of alsa streams with the alsa sink name that was opened. So if I have an asound.conf configured like so: pcm.music { type hw card 0 } Is it possible to get a callback saying that a stream on the "music" sink was opened? Thanks Baek -- Baek Chang baeksan(a)ccrma.stanford.edu

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[alsa-devel] ANYONE have spdif working with the ADI AD1986A chipset? (Asus M2NPV-VM mainboard)
by VDR User 17 Jul '09

17 Jul '09

This mainboard isn't that new but from what I've read spdif worked in alsa some time in the past. I get the IEC958/IEC958 D options in alsamixer but can not get any sound. Analog works fine. If anyone has spdif working on this mainboard then PLEASE post your exact setup & settings! The only other option would be to buy new hardware and that's not an option right now. Thanks for any/all help! -Derek

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[alsa-devel] While recording from mono codec twice as much data is received
by Louis Lee 16 Jul '09

16 Jul '09

Hi everyone, I am able to do playback from 8 kHz samples to 48 kHz samples properly. However, when recording using the exact same setup for the data interface I am not getting favorable results. For instance, if I record for 5 seconds I get output in a file for 10 seconds of audio (arecord or sox). This results in my recording being stretched. I change the DAI to stereo capture and record two channels and the recorded output comes back correctly (5 seconds recorded is 5 seconds). I'm using a wolfsom WM8974. I'm developing using ASoC framework (can I call it that?) reusing the WM8974 codec from the Linux GIT repository. The development is using a i.MX27. Data interface is a SSI bus in DSP mode. I believe my PCM code is correct since it works for playback. I'm stuck now. From my understanding the clocks are setup the same way for both playback and capture. Has anyone ran into this problem before? The WM8974 is a mono codec so I should not have to set the DAI to stereo in order to capture data properly. Any help is appreciated. Best regards,Louis _________________________________________________________________ Attention all humans. We are your photos. Free us. http://go.microsoft.com/?linkid=9666046

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[alsa-devel] [PATCH] ALSA: hda - Add CX20582 and OLPC XO-1.5 support
by Daniel Drake 16 Jul '09

16 Jul '09

This adds support for the Conexant CX20582 codec, based on code from http://www.linuxant.com/alsa-driver/alsa-driver-linuxant-1.0.19ppch12-1.noa… This is the codec to be shipped in the OLPC XO-1.5, so this patch also includes an XO-specific profile. Resultant configuration: http://dev.laptop.org/~dsd/20090713/codec0.txt http://dev.laptop.org/~dsd/20090713/codec0.svg As the Linuxant code is structured differently to the other codecs, I was unable to cleanly reimplement everything in the generic and Dell profiles as more info is needed (e.g. codec graphs). I simplified those profiles so that hopefully it will not break anyone's audio. If it does, it may be worth returning -ENODEV from patch_cx5066 on non-OLPC systems, and then fixing snd_hda_codec_configure() to fall back on the generic parser, at least until support for other systems is figured out. --- sound/pci/hda/patch_conexant.c | 479 ++++++++++++++++++++++++++++++++++++++++ 1 files changed, 479 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4fcbe21..77b4f28 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -108,6 +108,8 @@ struct conexant_spec { struct hda_input_mux private_imux; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; + unsigned int dell_automute; + unsigned int port_d_mode; }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -1908,6 +1910,480 @@ static int patch_cxt5051(struct hda_codec *codec) return 0; } +/* Conexant 5066 specific */ + +static hda_nid_t cxt5066_dac_nids[1] = { 0x10 }; +static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 }; +static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 }; +#define CXT5066_SPDIF_OUT 0x21 + +static struct hda_channel_mode cxt5066_modes[1] = { + { 2, NULL }, +}; + +static void cxt5066_update_speaker(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + unsigned int pinctl; + + snd_printdd("CXT5066: update speaker, hp_present=%d\n", + spec->hp_present); + + /* Port A (HP) */ + pinctl = ((spec->hp_present & 1) && spec->cur_eapd) ? PIN_HP : 0; + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); + + /* Port D (HP/LO) */ + pinctl = ((spec->hp_present & 2) && spec->cur_eapd) + ? spec->port_d_mode : 0; + snd_hda_codec_write(codec, 0x1c, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); + + /* CLASS_D AMP */ + pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); + + if (spec->dell_automute) { + /* DELL AIO Port Rule: PortA > PortD > IntSpk */ + pinctl = (!(spec->hp_present & 1) && spec->cur_eapd) + ? PIN_OUT : 0; + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); + } +} + +/* turn on/off EAPD (+ mute HP) as a master switch */ +static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + if (!cxt_eapd_put(kcontrol, ucontrol)) + return 0; + + cxt5066_update_speaker(codec); + return 1; +} + +/* toggle input of built-in and mic jack appropriately */ +static void cxt5066_automic(struct hda_codec *codec) +{ + static struct hda_verb ext_mic_present[] = { + /* enable external mic, port B */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* switch to external mic input */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0}, + + /* disable internal mic, port C */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + static struct hda_verb ext_mic_absent[] = { + /* enable internal mic, port C */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* switch to internal mic input */ + {0x17, AC_VERB_SET_CONNECT_SEL, 1}, + + /* disable external mic, port B */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + unsigned int present; + + present = snd_hda_codec_read(codec, 0x1a, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + if (present) { + snd_printdd("CXT5066: external microphone detected\n"); + snd_hda_sequence_write(codec, ext_mic_present); + } else { + snd_printdd("CXT5066: external microphone absent\n"); + snd_hda_sequence_write(codec, ext_mic_absent); + } +} + +/* mute internal speaker if HP is plugged */ +static void cxt5066_hp_automute(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + unsigned int portA, portD; + + /* Port A */ + portA = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + + /* Port D */ + portD = (snd_hda_codec_read(codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE) << 1; + + spec->hp_present = !!(portA | portD); + snd_printdd("CXT5066: hp automute portA=%x portD=%x present=%d\n", + portA, portD, spec->hp_present); + cxt5066_update_speaker(codec); +} + +/* unsolicited event for jack sensing */ +static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_automic(codec); + break; + } +} + +static const struct hda_input_mux cxt5066_analog_mic_boost = { + .num_items = 5, + .items = { + { "0dB", 0 }, + { "10dB", 1 }, + { "20dB", 2 }, + { "30dB", 3 }, + { "40dB", 4 }, + }, +}; + +static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return snd_hda_input_mux_info(&cxt5066_analog_mic_boost, uinfo); +} + +static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + int val; + + val = snd_hda_codec_read(codec, 0x17, 0, + AC_VERB_GET_AMP_GAIN_MUTE, AC_AMP_GET_OUTPUT); + + ucontrol->value.enumerated.item[0] = val & AC_AMP_GAIN; + return 0; +} + +static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; + unsigned int idx; + + if (!imux->num_items) + return 0; + idx = ucontrol->value.enumerated.item[0]; + if (idx >= imux->num_items) + idx = imux->num_items - 1; + + snd_hda_codec_write_cache(codec, 0x17, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | + imux->items[idx].index); + + return 1; +} + +static struct hda_input_mux cxt5066_capture_source = { + .num_items = 4, + .items = { + { "Mic B", 0 }, + { "Mic C", 1 }, + { "Mic E", 2 }, + { "Mic F", 3 }, + }, +}; + +static struct hda_bind_ctls cxt5066_bind_capture_vol_others = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT), + 0 + }, +}; + +static struct hda_bind_ctls cxt5066_bind_capture_sw_others = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT), + 0 + }, +}; + +static struct snd_kcontrol_new cxt5066_mixer_master[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + {} +}; + +static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = snd_hda_mixer_amp_volume_put, + .tlv = { .c = snd_hda_mixer_amp_tlv }, + /* offset by 28 volume steps to limit minimum gain to -46dB */ + .private_value = + HDA_COMPOSE_AMP_VAL_OFS(0x10, 3, 0, HDA_OUTPUT, 28), + }, + {} +}; + +static struct snd_kcontrol_new cxt5066_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5066_hp_master_sw_put, + .private_value = 0x1d, + }, + + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Mic Boost Capture Enum", + .info = cxt5066_mic_boost_mux_enum_info, + .get = cxt5066_mic_boost_mux_enum_get, + .put = cxt5066_mic_boost_mux_enum_put, + }, + + HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others), + HDA_BIND_SW("Capture Switch", &cxt5066_bind_capture_sw_others), + {} +}; + +static struct hda_verb cxt5066_init_verbs[] = { + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ + + /* Speakers */ + {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* HP, Amp */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* no digital microphone support yet */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Audio input selector */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3}, + + /* SPDIF route: PCM */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x22, AC_VERB_SET_CONNECT_SEL, 0x0}, + + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* EAPD */ + {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + + /* not handling these yet */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + { } /* end */ +}; + +static struct hda_verb cxt5066_init_verbs_olpc[] = { + /* Port A: headphones */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* Port B: external microphone */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* Port C: internal microphone */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* Port D: unused */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port E: unused, but has primary EAPD */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + + /* Port F: unused */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port G: internal speakers */ + {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* DAC2: unused */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + + /* Disable digital microphone port */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Audio input selectors */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + + /* Disable SPDIF */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* enable unsolicited events for Port A and B */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + { } /* end */ +}; + +static struct hda_verb cxt5066_init_verbs_portd_lo[] = { + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { } /* end */ +}; + +/* initialize jack-sensing, too */ +static int cxt5066_init(struct hda_codec *codec) +{ + snd_printdd("CXT5066: init\n"); + conexant_init(codec); + if (codec->patch_ops.unsol_event) { + cxt5066_hp_automute(codec); + cxt5066_automic(codec); + } + return 0; +} + +enum { + CXT5066_LAPTOP, /* Laptops w/ EAPD support */ + CXT5066_DELL_LAPTOP, /* Dell Laptop */ + CXT5066_OLPC_XO_1_5, /* OLPC XO 1.5 */ + CXT5066_MODELS +}; + +static const char *cxt5066_models[CXT5066_MODELS] = { + [CXT5066_LAPTOP] = "laptop", + [CXT5066_DELL_LAPTOP] = "dell-laptop", + [CXT5066_OLPC_XO_1_5] = "olpc-xo-1_5", +}; + +static struct snd_pci_quirk cxt5066_cfg_tbl[] = { + SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", + CXT5066_LAPTOP), + SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", + CXT5066_DELL_LAPTOP), + {} +}; + +static int patch_cxt5066(struct hda_codec *codec) +{ + struct conexant_spec *spec; + int board_config; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + codec->patch_ops = conexant_patch_ops; + codec->patch_ops.init = cxt5066_init; + + spec->dell_automute = 0; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids); + spec->multiout.dac_nids = cxt5066_dac_nids; + spec->multiout.dig_out_nid = CXT5066_SPDIF_OUT; + spec->num_adc_nids = 1; + spec->adc_nids = cxt5066_adc_nids; + spec->capsrc_nids = cxt5066_capsrc_nids; + spec->input_mux = &cxt5066_capture_source; + + spec->port_d_mode = PIN_HP; + + spec->num_init_verbs = 1; + spec->init_verbs[0] = cxt5066_init_verbs; + spec->num_channel_mode = ARRAY_SIZE(cxt5066_modes); + spec->channel_mode = cxt5066_modes; + spec->cur_adc = 0; + spec->cur_adc_idx = 0; + + board_config = snd_hda_check_board_config(codec, CXT5066_MODELS, + cxt5066_models, cxt5066_cfg_tbl); + switch (board_config) { + default: + case CXT5066_LAPTOP: + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + break; + case CXT5066_DELL_LAPTOP: + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + + spec->port_d_mode = PIN_OUT; + spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_portd_lo; + spec->num_init_verbs++; + spec->dell_automute = 1; + break; + case CXT5066_OLPC_XO_1_5: + codec->patch_ops.unsol_event = cxt5066_unsol_event; + spec->init_verbs[0] = cxt5066_init_verbs_olpc; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + spec->port_d_mode = 0; + + /* no S/PDIF out */ + spec->multiout.dig_out_nid = 0; + + /* input source automatically selected */ + spec->input_mux = NULL; + break; + } + + return 0; +} /* */ @@ -1919,12 +2395,15 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5047 }, { .id = 0x14f15051, .name = "CX20561 (Hermosa)", .patch = patch_cxt5051 }, + { .id = 0x14f15066, .name = "CX20582 (Pebble)", + .patch = patch_cxt5066 }, {} /* terminator */ }; MODULE_ALIAS("snd-hda-codec-id:14f15045"); MODULE_ALIAS("snd-hda-codec-id:14f15047"); MODULE_ALIAS("snd-hda-codec-id:14f15051"); +MODULE_ALIAS("snd-hda-codec-id:14f15066"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); -- 1.6.2.5

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Re: [alsa-devel] [PATCH 01/06] Fix compilation warning for sound/soc/codecs/wm8400.c
by Mark Brown 16 Jul '09

16 Jul '09

On Thu, Jul 16, 2009 at 06:40:01PM +0530, Subrata Modak wrote: > How about the following brutal shutdown ? No, this is the patch you originally submitted. You are missing the point here - your patches are all just papering over the warning and won't help at all if there is an actual problem. > 'factors' will get initialized here, as, 'freq_out' will probably never have > a '0' value. So, 'fll_factors()' will actually overwrite values in 'factors' > even after the initial brutal assignment: > "struct fll_factors factors = {}" This is pretty much the point of the warning when it's valid - it's trying to catch situations where there is a code path where the variable is used without being initialised. If you just blindly initialise the variable as you are doing then the warning goes away but any problematic code remains so the situation is worse. In this case I suspect that whatever compiler you are using (none of those I've tried with seem to be doing this) isn't able to figure out that if we skip initialising the variable then we use exactly the same condition to return from the function before we try to use the variable. For something like this the warning can normally be worked around by changing the control flow so that the compiler is able to figure out that things are safe.

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Re: [alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Takashi Iwai 16 Jul '09

16 Jul '09

At Wed, 8 Jul 2009 18:11:29 +0200, Andreas Nüßlein wrote: > > On Wednesday 08 July 2009 16:49:34 you wrote: > > At Tue, 07 Jul 2009 12:53:48 +0200, > > > > I wrote: > > > At Tue, 7 Jul 2009 11:49:57 +0200, > > > > > > Andreas Nüßlein wrote: > > > > On Tuesday 07 July 2009 08:00:19 you wrote: > > > > > At Mon, 6 Jul 2009 22:14:56 +0200, > > > > > > > > > > Andreas Nüßlein wrote: > > > > > > On Monday 06 July 2009 21:26:18 you wrote: > > > > > > > At Mon, 06 Jul 2009 11:16:35 -0600, > > > > > > > > > > > > > > Sean Burke wrote: > > > > > > > > Scríobh Takashi Iwai: > > > > > > > > > At Mon, 6 Jul 2009 17:53:46 +0200, > > > > > > > > > > > > > > > > > > Andreas Nüßlein wrote: > > > > > > > > >>> The missing pin configuration initialization was already > > > > > > > > >>> fixed by the driver overriding it after checking PCI SSID > > > > > > > > >>> (which is different from the codec SSID). So, this should > > > > > > > > >>> be no problem. > > > > > > > > >>> > > > > > > > > >>> However, the reason why the analog output doesn't work > > > > > > > > >>> might be different from that. There might be something > > > > > > > > >>> else missing, but I don't know. > > > > > > > > >>> > > > > > > > > >>> > > > > > > > > >>> Takashi > > > > > > > > >> > > > > > > > > >> oh =( > > > > > > > > >> > > > > > > > > >> > > > > > > > > >> hmm.. anything i can do? would it help if i tried changing > > > > > > > > >> values randomly with hda-analyzer.py? > > > > > > > > > > > > > > > > > > Well, did the driver without my change work more or less with > > > > > > > > > the analog audio, or have you never gotten the analog output? > > > > > > > > > You can use the generic parser (i.e. the state without cirrus > > > > > > > > > patch) by passing model=generic option to snd-hda-intel. > > > > > > > > > > > > > > > > For my part, nothing worked with the generic driver. I can't > > > > > > > > confirm digital out, but I can confirm that the kfree error is > > > > > > > > gone. What options are open for figuring out what remains? > > > > > > > > > > > > > > Easy things to test are GPIO bits. Run hda-verb like > > > > > > > > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > > or > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > > > > > > > > > etc. CS4206 seems to have 4 GPIO lines, and each bit (0-3) > > > > > > > corresponds to each GPIO. In many case, GPIO0 or GPIO1 > > > > > > > corresponds to the amplifier (EAPD) bit. > > > > > > > Define the GPIO direction of each GPIO bit by SET_GPIO_DIR, and > > > > > > > turn on/off the GPIO bits by SET_GPIO_DATA. Running > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 GET_GPIO_DATA 0 > > > > > > > will show the current GPIO data bits. Or you can check it in > > > > > > > codec#* proc file. > > > > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > w000000000000000000000000000000000t! =) > > > > > > > > > > > > takashi, thank you _so_ much! > > > > > > > > > > > > after running all 4 of those: > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > > or > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > > > > > > > i suddendly had sound!! :D :D :D :D :D :D :D > > > > > > only via speakers though - there is no sound via headphones right > > > > > > now. > > > > > > > > > > > > mixer channels: > > > > > > - Master (with Mutebutton), PCM and Front (also with Mute) all work > > > > > > =) - i don't know what surround would do (or it's extra switch) - > > > > > > headphones-volumes and mute button don't affect the speakers, which > > > > > > is good =) > > > > > > > > > > > > > > > > > > is there a way to reset what i did with hda-verb, so that i can > > > > > > figure out which combination it was exactly? > > > > > > > > > > You can just change the value 0x0f to a different value. > > > > > At least, you can try commands like > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x04 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 > > > > > and check the speaker output at each time. > > > > > Also, check the GPIO direction, > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x01 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x02 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > > > > > ... > > > > > > > > > > Regarding the headphone: is the speaker muted when you plug in the > > > > > headphone? If not, it's likely an issue of the jack detection. If > > > > > the speaker is muted but no headphone output, it's a missing > > > > > initialization (or wrong GPIO setup). > > > > > > > > > > > > > > > Takashi > > > > > > > > so here are my findings so far: > > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 > > > > > > > > > > > > is sufficient to turn on the right functionality for the speakers; > > > > meaning: - 'front speaker'-channel actually controls the two front > > > > speakers within the macbookpro - even controlling left and right > > > > channel seperately works =) - 'surround' mute-switch and > > > > volume-control (why are those not within on thing btw?) also work and > > > > control a third speaker in the mac, which really seems responsibly for > > > > surround :) > > > > > > > > headphones do not yet work, however i made a diff of > > > > "/proc/asound/card0/codec#0" with and without something plugged in: > > > > > > OK, so actually the driver works as expected, but the hardware doesn't > > > behave as expected. For muting the speaker, something else (like > > > toggling that GPIO) is needed. > > > > > > The mystery about the silent HP output still remains, though. > > > > One another thing we can try is to use only one PCM stream instead of > > multi streams. For example, try the patch below. This will enable > > only the pin for the headphone, thus only one DAC is used. > > > > > > Takashi > > > > --- > > diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c > > index 57251d7..d7f52b1 100644 > > --- a/sound/pci/hda/patch_cirrus.c > > +++ b/sound/pci/hda/patch_cirrus.c > > @@ -1086,13 +1086,17 @@ struct cs_pincfg { > > > > static struct cs_pincfg mbp55_pincfgs[] = { > > { 0x09, 0x012b4030 }, > > - { 0x0a, 0x90100121 }, > > - { 0x0b, 0x90100120 }, > > + // { 0x0a, 0x90100121 }, > > + { 0x0a, 0x400000f0 }, > > + // { 0x0b, 0x90100120 }, > > + { 0x0b, 0x400000f0 }, > > { 0x0c, 0x400000f0 }, > > - { 0x0d, 0x90a00110 }, > > + // { 0x0d, 0x90a00110 }, > > + { 0x0d, 0x400000f0 }, > > { 0x0e, 0x400000f0 }, > > { 0x0f, 0x400000f0 }, > > - { 0x10, 0x014be040 }, > > + // { 0x10, 0x014be040 }, > > + { 0x10, 0x400000f0 }, > > { 0x12, 0x400000f0 }, > > { 0x15, 0x400000f0 }, > > {} /* terminator */ > > > =( no - that doesn't work either > > (there is obivously no sound via built-in speakers either anymore) Hmm, then I have really no clue. At best, you can try again GPIO pins and reverting COEF setup by my earlier patch, and try some more adjustments of widgets amp or pinctl via hda-verb or hda-analyzer... Takashi

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 16 Jul '09

16 Jul '09

The following changes since commit 4b75e9476737dffef44a66818a4a481edd2512a1: Mark Brown (1): ASoC: Error out if we can't determine a suitable WM9081 sysclk are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (1): ASoC: new ad1938 codec driver based on asoc Joonyoung Shim (3): ASoC: add SOC_DOUBLE_EXT_TLV control type ASoC: add SOC_DOUBLE_R_EXT_TLV control type ASoC: MAX9877: add MAX9877 amp driver Kevin Hilman (1): ASoC: davinci: don't use clock names include/sound/soc.h | 22 ++ sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/ad1938.c | 652 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ad1938.h | 100 ++++++ sound/soc/codecs/max9877.c | 270 +++++++++++++++ sound/soc/codecs/max9877.h | 37 ++ sound/soc/davinci/davinci-i2s.c | 2 +- sound/soc/davinci/davinci-mcasp.c | 2 +- 9 files changed, 1100 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/ad1938.c create mode 100644 sound/soc/codecs/ad1938.h create mode 100644 sound/soc/codecs/max9877.c create mode 100644 sound/soc/codecs/max9877.h

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[alsa-devel] [PATCH 0/4] alsa-info.sh updates
by Wu Fengguang 16 Jul '09

16 Jul '09

Hi Takashi, This patchset mainly disables the auto-upload of alsa info by default. Comments are welcome. Thanks, Fengguang --

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