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[alsa-devel] Microphone dead: Gateway MX6543, hda-intel, STAC925x
by Zoran Rilak 27 Jan '08

27 Jan '08

Hello, I've searched the archives and haven't found a solution for this particular model (STAC9250). Front mic does not work. I have tried other models, aside from m2-2, and all four possible values for position_fix as well, but to no avail. If anyone is willing to tackle this, please point me to what other information I should provide in order to get to the bottom of this. I am using Ubuntu kernel 2.6.22-14 and ALSA 1.0.15 compiled from source. Here are my /proc/asound/card0/codec#* files: --- /proc/asound/card0/codec#0 --- Codec: SigmaTel STAC9250 Address: 0 Vendor Id: 0x83847634 Subsystem Id: 0x107b0367 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Node 0x02 [Audio Output] wcaps 0xd0401: Stereo Power: 0x0 Node 0x03 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x14 Node 0x04 [Audio Input] wcaps 0x140311: Stereo Digital PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Connection: 1 0x07 Node 0x05 [Audio Output] wcaps 0x40211: Stereo Digital PCM: rates [0x1e0]: 44100 48000 88200 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Node 0x06 [Audio Selector] wcaps 0x300901: Stereo Connection: 3 0x02* 0x07 0x14 Node 0x07 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x0810024: IN EAPD Detect Pin Default 0x40c003f3: [N/A] SPDIF In at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Power: 0x0 Node 0x08 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x424503f2: [N/A] SPDIF Out at Ext Front Conn = Optical, Color = Unknown Pin-ctls: 0x00: Connection: 2 0x05* 0x14 Node 0x09 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=0 Amp-Out vals: [0x0f 0x0f] Connection: 1 0x0f Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x083f: IN OUT HP Detect Pin Default 0x04180011: [Jack] Speaker at Ext Right Conn = DIN, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x02a19020: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 1 0x0e Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x50a103f0: [N/A] Mic at Int N/A Conn = 1/8, Color = Unknown Pin-ctls: 0x20: IN Connection: 1 0x0e Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x083f: IN OUT HP Detect Pin Default 0x90100212: [Fixed] Speaker at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x0e [Audio Selector] wcaps 0x300105: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x0d 0x0d] Connection: 1 0x06 Node 0x0f [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x04 0x04] Connection: 5 0x0b* 0x0c 0x0d 0x0a 0x11 Node 0x10 [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x00] Pincap 0x0810: OUT Pin Default 0x400003f1: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Connection: 1 0x12 Node 0x11 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x9033032e: [Fixed] CD at Int N/A Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x12 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x06 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 Amp-Out vals: [0x00] Node 0x14 [Audio Selector] wcaps 0x30090d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x09 --- end codec#0 --- --- /proc/asound/card0/codec#1 --- Codec: Motorola Si3054 Address: 1 Vendor Id: 0x10573057 Subsystem Id: 0x10001 Revision Id: 0x100100 Modem Function Group: 0x1 --- end codec#1 ---

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[alsa-devel] [PATCH] Add missing additional verbs for SPDIF with AD1984 & thinkpad
by Jerone Young 26 Jan '08

26 Jan '08

Additional verbs are missing for IEC958 code for the AD1984 in thinkpads. This is identical to the verbs in AD19884. I missed these in my patch last month. Signed-off-by: Jerone Young <jerone(a)gmail.com> diff -r dcc1b9d533d3 alsa-kernel/pci/hda/patch_analog.c --- a/alsa-kernel/pci/hda/patch_analog.c Fri Jan 25 19:49:19 2008 -0600 +++ b/alsa-kernel/pci/hda/patch_analog.c Fri Jan 25 19:51:05 2008 -0600 @@ -3188,6 +3188,9 @@ static struct hda_verb ad1984_thinkpad_i {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* enable EAPD bit */ {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + /* SPDIF output selector */ + {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ }; /* Digial MIC ADC NID 0x05 + 0x06 */

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[alsa-devel] [PATCH] Remove psuedo master from pci/hda/analog_patch.c . It does not function well!
by Jerone Young 26 Jan '08

26 Jan '08

Hi I noticed after installing 1.0.16rc-1 today that my Thinkpad T61 with AD1984 had a Master mixer now. Problem is it's not very good in that it 1) does not follow with PCM (the real master) volume well. It causes a volume disparity and does not follow or even boost PCM signal. So it just a bad mister. 2) it does not mute the volume (since PCM is the real master mixer you have to go to PCM to mute the volume!) I have fixed a problem in Gstreamer so that Gnome (gnome volume mixer) does finally work correctly for the AD1984 which does not have a Master. When it looks for a master mixer it now will fallback to PCM as a master mixer (the code is upsteam too). You can find this here: http://bugzilla.gnome.org/show_bug.cgi?id=506928 I have a feeling this is why this patch was put in. I have fixed the problem in Gnome, KDE has allways worked. just fine. But I would honestly love it if this code was removed!!! If you need a Master can you rename PCM to master for the AD1984! I have attached a patch to remove this code .. because it is so does not function well! Signed-off-by : Jerone Young <jerone(a)gmail.com> diff -r 5dfc8324ae17 alsa-kernel/pci/hda/patch_analog.c --- a/alsa-kernel/pci/hda/patch_analog.c Fri Jan 25 00:12:14 2008 -0600 +++ b/alsa-kernel/pci/hda/patch_analog.c Fri Jan 25 01:04:13 2008 -0600 @@ -170,26 +170,6 @@ static int ad198x_build_controls(struct } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); - if (err < 0) - return err; - } - - /* if we have no master control, let's create it */ - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { - snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, - HDA_OUTPUT, spec->vmaster_tlv); - err = snd_hda_add_vmaster(codec, "Master Playback Volume", - spec->vmaster_tlv, - (spec->slave_vols ? - spec->slave_vols : ad_slave_vols)); - if (err < 0) - return err; - } - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { - err = snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, - (spec->slave_sws ? - spec->slave_sws : ad_slave_sws)); if (err < 0) return err; }

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Re: [alsa-devel] Error when compiling from HG on Ubuntu 7.10
by Pavel Hofman 25 Jan '08

25 Jan '08

I am sorry for the false alarm. Clean hg clone and hgcompile works flawlessly. Pavel Hofman. > Hello, > > While trying to compile the latest alsa-kernel + alsa-drivers from HG > using ./hgcompile, I am hitting the following error: > > CC [M] /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.o > In file included from > /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:23: > /home/pavel/projects/alsa/alsa-driver/include/sound/driver.h:1:2: > warning: #warning "This file is deprecated" > /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:60: > error: redefinition of ‘video_get_drvdata’ > include/media/v4l2-dev.h:383: error: previous definition of > ‘video_get_drvdata’ was here

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[alsa-devel] [PATCH] hda: In-Amp support for 92HD7xxx codecs.
by Matthew Ranostay 25 Jan '08

25 Jan '08

Some 92HD7xxx codecs have amps on the ports to volume control and/or mute certain ports. Also this makes stac92hd71bxx unmute amps lines in the init not needed. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 5bf4c5d02f4b pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Jan 24 15:32:15 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Jan 24 11:25:36 2008 -0500 @@ -577,10 +577,6 @@ static struct hda_verb stac92hd71bxx_cor /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, }; static struct hda_verb stac92hd71bxx_analog_core_init[] = { @@ -594,11 +590,6 @@ static struct hda_verb stac92hd71bxx_ana { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute dac0 input in audio mixer */ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {} }; static struct hda_verb stac925x_core_init[] = { @@ -2215,6 +2206,37 @@ static int create_controls(struct sigmat return 0; } +/* add playback controls for ports that have amps */ +static int stac92xx_create_amp_ctls(struct hda_codec *codec, + hda_nid_t nid, char *pfx, int idx) +{ + struct sigmatel_spec *spec = codec->spec; + int err; + char name[48]; + u32 caps = query_amp_caps(codec, nid, HDA_INPUT); + if (idx) + sprintf(name, "%s %d", pfx, idx); + else + strcpy(name, pfx); + + if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { + sprintf(name, "%s Playback Volume", name); + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + } + + if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { + sprintf(name, "%s Playback Switch", name); + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + } + return 0; +} + /* add playback controls from the parsed DAC table */ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) @@ -2262,13 +2284,39 @@ static int stac92xx_auto_create_multi_ou } } - if (spec->line_switch) - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Line In as Output Switch", cfg->input_pins[AUTO_PIN_LINE] << 8)) < 0) + if (spec->line_switch) { + int val = cfg->input_pins[AUTO_PIN_LINE] << 8; + wid_caps = get_wcaps(codec, val >> 8); + + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, + "Line In as Output Switch", val); + if (err < 0) return err; - if (spec->mic_switch) - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Mic as Output Switch", (cfg->input_pins[AUTO_PIN_MIC] << 8) | 1)) < 0) + if (wid_caps & AC_WCAP_IN_AMP) { + err = stac92xx_create_amp_ctls(codec, val >> 8, + "Line In as Output Gain", 0); + if (err < 0) + return err; + } + } + + if (spec->mic_switch) { + int val = cfg->input_pins[AUTO_PIN_MIC] << 8; + wid_caps = get_wcaps(codec, val >> 8); + + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, + "Mic as Output Switch", val | 1); + if (err < 0) return err; + + if (wid_caps & AC_WCAP_IN_AMP) { + err = stac92xx_create_amp_ctls(codec, val >> 8, + "Mic as Output Gain", 0); + if (err < 0) + return err; + } + } return 0; } @@ -2311,6 +2359,13 @@ static int stac92xx_auto_create_hp_ctls( spec->hp_detect = 1; nid = snd_hda_codec_read(codec, cfg->hp_pins[i], 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (wid_caps & AC_WCAP_IN_AMP) { + err = stac92xx_create_amp_ctls(codec, + cfg->hp_pins[i], + "Headphone Gain", i); + if (err < 0) + return err; + } if (check_in_dac_nids(spec, nid)) nid = 0; if (! nid) @@ -2320,6 +2375,14 @@ static int stac92xx_auto_create_hp_ctls( for (i = 0; i < cfg->speaker_outs; i++) { nid = snd_hda_codec_read(codec, cfg->speaker_pins[i], 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (get_wcaps(codec, cfg->speaker_pins[i]) & AC_WCAP_IN_AMP) { + err = stac92xx_create_amp_ctls(codec, + cfg->speaker_pins[i], + "Speaker Gain", i); + if (err < 0) + return err; + } + if (check_in_dac_nids(spec, nid)) nid = 0; if (! nid) @@ -2329,6 +2392,13 @@ static int stac92xx_auto_create_hp_ctls( for (i = 0; i < cfg->line_outs; i++) { nid = snd_hda_codec_read(codec, cfg->line_out_pins[i], 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (get_wcaps(codec, cfg->line_out_pins[i]) & AC_WCAP_IN_AMP) { + err = stac92xx_create_amp_ctls(codec, + cfg->line_out_pins[i], + "Line Out Gain", i); + if (err < 0) + return err; + } if (check_in_dac_nids(spec, nid)) nid = 0; if (! nid)

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[alsa-devel] [PATCH] hda: STAC92xx Line In/Mic as output check
by Matthew Ranostay 25 Jan '08

25 Jan '08

This patch checks to see the Line In/Mic port have the ability to do output before creating the the control switches. The 92hd71bxx series of codecs has this issue with the port 0xe, which only allows input. Signed-off-by: Matthew Ranostay <mranostay(a)embededalley.com> --- diff -r 303f894ac2a8 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Jan 24 18:46:42 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Jan 24 22:41:27 2008 -0500 @@ -2226,7 +2226,7 @@ static int stac92xx_auto_create_multi_ou int i, err; struct sigmatel_spec *spec = codec->spec; - unsigned int wid_caps; + unsigned int wid_caps, pincap; for (i = 0; i < cfg->line_outs; i++) { @@ -2262,13 +2262,31 @@ static int stac92xx_auto_create_multi_ou } } - if (spec->line_switch) - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Line In as Output Switch", cfg->input_pins[AUTO_PIN_LINE] << 8)) < 0) - return err; - - if (spec->mic_switch) - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Mic as Output Switch", (cfg->input_pins[AUTO_PIN_MIC] << 8) | 1)) < 0) - return err; + if (spec->line_switch) { + nid = cfg->input_pins[AUTO_PIN_LINE]; + pincap = snd_hda_param_read(codec, nid, + AC_PAR_PIN_CAP); + if (pincap & AC_PINCAP_OUT) { + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_IO_SWITCH, + "Line In as Output Switch", nid << 8); + if (err < 0) + return err; + } + } + + if (spec->mic_switch) { + nid = cfg->input_pins[AUTO_PIN_MIC]; + pincap = snd_hda_param_read(codec, nid, + AC_PAR_PIN_CAP); + if (pincap & AC_PINCAP_OUT) { + ` err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_IO_SWITCH, + "Mic as Output Switch", (nid << 8) | 1); + if (err < 0) + return err; + } + } return 0; }

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Re: [alsa-devel] 1.0.16rc1 release -- error: 'system_utsname' undeclared
by Takashi Iwai 25 Jan '08

25 Jan '08

At Fri, 25 Jan 2008 22:48:43 +1100, Rick Measham wrote: > > > Rick Measham wrote: > >> dmesg tells me there are zero speaker_outs. So I figure it didn't > >> auto-detect them, and there are no pins specified for this exact setup. > > Takashi Iwai wrote: > > OK, it means that either the BIOS set up is broken or the preset > > pincfg is broken. Could you check which board_config is chosen for > > your device? It's selected via PCI SSID (see lspci -nv output). > > > > If it's a bugin preset pincfg, it'd be realtivelyeasy to fix (once > > after we figure out which pin corresponds to the spekaer output). > > I'm certainly hoping for the latter .. and it looks hopeful > > 00:1b.0 0403: 8086:27d8 (rev 01) > Subsystem: 1028:01bf > Flags: bus master, fast devsel, latency 0, IRQ 21 > Memory at dfffc000 (64-bit, non-prefetchable) [size=16K] > Capabilities: [50] Power Management version 2 > Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ > Queue=0/0 Enable- > Capabilities: [70] Express Unknown type IRQ 0 > > Lines up with: > /* > STAC 922X pin configs for > 102801BF > */ > static unsigned int dell_922x_m81_pin_configs[10] = { > 0x0321101f, 0x01112024, 0x01111222, 0x91174220, > 0x03a11050, 0x01116221, 0x90a70330, 0x01452340, > 0x40C003f1, 0x405003f0, > }; > > > So it certainly looks like we might have some hope? > > It's listed as "unknown Dell" .. so let me fill that blank: It's the XPS > M2010. > > Now for the billion dollar question: How do I work out what the > pin_configs SHOULD be? I can play guess-the-number, but that's going to > take some time and I'd guess a lot of crashing! That pin config actually looks strange. It defines 4 "speaker" outputs. The autocfg routine in hda-intel driver switches these speaker outputs to line-outs (just for compatibility issue). That's why you don't see any speaker there. The patch below is a quick fix to change the jack types for 3 outputs. Give it a try. Now more questions: - Does your machine have 3 output jacks in addition to HP? - What is the content of /proc/asound/card0/codec#* when you load the driver with model=auto? After setting it, reboot so that BIOS resets the configuration. - Show both codec#* files for model=auto and without model option. thanks, Takashi --- diff -r 19029da61288 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Jan 25 11:56:57 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Fri Jan 25 13:04:20 2008 +0100 @@ -1273,8 +1273,8 @@ static unsigned int dell_922x_d82_pin_co 102801BF */ static unsigned int dell_922x_m81_pin_configs[10] = { - 0x0321101f, 0x01112024, 0x01111222, 0x91174220, - 0x03a11050, 0x01116221, 0x90a70330, 0x01452340, + 0x0321101f, 0x01012024, 0x01011222, 0x91174220, + 0x03a11050, 0x01016221, 0x90a70330, 0x01452340, 0x40C003f1, 0x405003f0, };

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[alsa-devel] [PATCH] Add model for Acer Aspire 5310
by Andy Shevchenko 25 Jan '08

25 Jan '08

Hi! Simplify usage of the Acer Aspire 5310 laptop with the ALC268 based codec sound card via add correct PCI SSID. Signed-off-by: Andy Shevchenko <andy(a)smile.org.ua> --- alsa-driver-hg20080125/alsa-kernel/pci/hda/patch_realtek.c.acer5310 2008-01-25 03:00:07.000000000 +0200 +++ alsa-driver-hg20080125/alsa-kernel/pci/hda/patch_realtek.c 2008-01-25 13:05:39.000000000 +0200 @@ -9925,6 +9925,7 @@ static const char *alc268_models[ALC268_ static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER), + SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER), SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER), SND_PCI_QUIRK(0x1025, 0x0136, "Acer Aspire 5315", ALC268_ACER), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), P.S. Sorry for separate patch. I've been noticed today about this one. -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua

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[alsa-devel] [Fwd: Re: Code relsease to alsa.]
by Tobin Davis 25 Jan '08

25 Jan '08

Forwarded from the good people at Linuxant. Tobin -------- Forwarded Message -------- From: Marc Boucher <marc(a)linuxant.com> To: Tobin Davis <tdavis(a)dsl-only.net> Cc: Linuxant support (Jonathan) <support(a)linuxant.com> Subject: Re: Code relsease to alsa. Date: Mon, 21 Jan 2008 10:33:03 -0500 Tobin, As promised here is our current tentative patch (relative to the current ALSA development tree in hg) for merging with ALSA. It is also available from: http://www.linuxant.com/alsa-hg-20080120.patch Naturally the changes are released under the GPL, so feel free to integrate them. Comments and improvements are welcome. Kind regards, Marc -- Marc Boucher Linuxant inc. http://www.linuxant.com On 19-Jan-08, at 6:40 PM, Marc Boucher wrote: > > Hi Tobin, > > We have isolated the modifications in patch form and are currently > working to merge them with the latest ALSA tree / submit them back > upstream. > > By the way, all changes and code under the modules/GPL/ directory > (including the hda subdirectory) of our modem driver are released > under the GPL, so you would be free to take these fixes if you'd like. > > However you would be also welcome to reviewing the patches that we > are about to submit. I will arrange to send you a copy asap (should > be next week). > > Kind regards, (and many thanks for your great contributions to ALSA) > > Marc > > PS: We could also try to help you getting any missing info on the > CX20561 or other things from Conexant... > > -- > Marc Boucher > Linuxant inc. > http://www.linuxant.com > > On 19-Jan-08, at 1:07 PM, Tobin Davis wrote: > >> Hello, >> >> I'm the developer that put the current audio support for the >> Conexant HD Audio codecs in alsa. I worked many hours on this, >> with testers around the world sending me feedback. Most of the >> work was done prior to Conexant offering me any documentation. I >> have since received documentation for both CX20549 (Venice), and >> CX20551 (Waikiki), but am waiting for the latest info on the new >> CX20561, which you apparently already have. >> >> I have also seen in your mailing list that you are providing >> several fixes to the audio portion of my code, but only making them >> available to users of your code, which requires a license agreement >> that I believe is incompatible with the GPL. At the least, it >> prohibits someone like me taking any of these fixes and putting >> them in the mainstream alsa, where more users would benefit from >> them. >> >> What I'm asking, is if you could at least provide the audio >> modifications back upstream to alsa? It would help out far more >> users (modem usage is on the decline - sorry), and you would have >> some benefit in that your name would be in the source as well. >> >> >> -- >> Tobin Davis <tdavis(a)dsl-only.net> > -- Tobin Davis no maintenance: Impossible to fix.

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[alsa-devel] HDMI Status
by Zsolt Barat 25 Jan '08

25 Jan '08

hi list, since i don't got any response on the users list, i try it here. i own a MSI board, K9AGM2, with north bridge S690G/690V and a as southbridge SB600. the hda-intel driver works perfectly for the most part. i just want to ask, how complete is the support for hdmi-audio. it seems that the hdmi-hardware is at least detected since alsamixer shows me as chip: Generic 1002 ATI RS690/780 HDMI. is it also supposed to put out some audio? i couldn't test it cause of lack of hdmi-sound capable hardware. best regards zsolt

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