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Re: [alsa-devel] WM9712 monoout
by Mike Rapoport 22 Oct '07

22 Oct '07

Liam Girdwood wrote: > On Sun, 2007-10-21 at 09:35 +0200, Mike Rapoport wrote: >> Hello, >> >> I'm trying to enable phone on a board with WM9715 codec. >> I have not found a control responsible for mute/unmute the MONOUT. >> My question is whether I should add this control as platform specific control, >> or it would be correct to patch sound/soc/codecs/wm9712.c? > > It should probably go in the codec driver. It looks like an omission. > >> Also, "Mono Playback Volume" control is not inverted, which seems to be a bug. >> > > Could you send a patch ? Sure. A also found that treble and bass need to be inverted. > Many Thanks > > Liam > Signed-off-by: Mike Rapoport <mike(a)compulab.co.il> sound/soc/codecs/wm9712.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 986b5d5..427cb61 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -102,7 +102,8 @@ SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0), SOC_SINGLE("Speaker Playback Invert Switch", AC97_MASTER, 6, 1, 0), SOC_SINGLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 7, 1, 0), SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_MONO, 7, 1, 0), -SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 0), +SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), +SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), @@ -145,8 +146,8 @@ SOC_ENUM("Bass Control", wm9712_enum[5]), SOC_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1), SOC_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1), SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0), -SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 0), -SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 0), +SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 1), +SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1), SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1), SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),

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[alsa-devel] snd-pcsp doesn't accept the index parameter
by Alexander E. Patrakov 22 Oct '07

22 Oct '07

It is impossible to put pcsp at a stable index without the trivial patch below. Signed-Off-By: Alexander E. Patrakov <patrakov(a)ums.usu.ru> --- pcsp.c +++ pcsp.c @@ -27,6 +27,8 @@ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for pcsp soundcard."); module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for pcsp soundcard."); module_param(enable, bool, 0444); -- Alexander E. Patrakov

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[alsa-devel] [PATCH] hda: STAC9228 updated DMUX nid.
by Matthew Ranostay 22 Oct '07

22 Oct '07

Changed the dmux for STAC9228 from ADC1MUX to ADC0MUX to avoid confusion. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 7cf5e23f804e pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Oct 19 08:23:00 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Fri Oct 19 13:47:35 2007 -0400 @@ -2777,7 +2777,7 @@ static int patch_stac927x(struct hda_cod case 0x10280209: /* STAC 9228 */ spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; - spec->dmux_nid = 0x1c; + spec->dmux_nid = 0x1b; break; default: spec->num_dmics = 0;

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[alsa-devel] SALSA Lib - TLV Deprecated warnings
by J. Scott Merritt 22 Oct '07

22 Oct '07

It appears that if SALSA Lib 0.0.12 is built -without- TLV enabled in the configuration options, then simply -including- asoundlib.h from an application program will trigger deprecated warnings on some of the TLV read/write function definitions (even though they are not called from the application). Is there something we can do to the header files to eliminate these warnings ? Thanks, Scott.

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[alsa-devel] alsa-lib not installing plugins
by Larry D. Pyeatt 22 Oct '07

22 Oct '07

Hi, I wrote a new plugin for alsa-lib. After running make, there are .o and .lo files in src/pcm. However, when I run make install, nothing gets put into /usr/lib/alsa-lib. I first tried installing alsa-lib-1.0.15rc3. Then I tried alsa-lib-1.0.13rc3 and alsa-lib-1.0.14rc4. All of them did the same thing. Am I missing something obvious? Since they are all doing the same thing, it must be either that I just don't know what I'm doing, or something else in my system is not working. I would appreciate any suggestions on what could cause this behavior. Thanks, -- Larry D. Pyeatt, PhD. pyeatt(a)cs.ttu.edu http://www.cs.ttu.edu/~pyeatt

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Re: [alsa-devel] hda-intel: no soundcard with current linus' git tree
by Maxim Levitsky 21 Oct '07

21 Oct '07

Hi, I understand why this happens. Your sound card has a single "headphone" output, and since it is single, it is called "Master" the stac you have also has a real master volume control, that controls all the DACs, not just the headphones. I added support for it, thus two controls collided. The following patch will rename the "Master" output to "Headphone" Note that on the stac 9205/9204, the only one that has no master volume (the volume knob) but has a "Master Volume" amplifier widget, this will probably rename the "Master Volume" to Headphones. This chip is quite different, thus I think this should be handled in its patch function. I see what I can do later. Meanwhile the rename from "Master" to Headphones" is the only regression. Best regards, Maxim Levitsky PS: I reply to this mail with the patch.

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Re: [alsa-devel] hda-intel: no soundcard with current linus' git tree
by Jan-Simon Möller 21 Oct '07

21 Oct '07

I've also no sound. Had it working using the alsa.git tree _before_ the last merge (>48h). Needed to use model=asus for this fujitsu v5505 here. Now it doesn't work, even with other model-options (fujitsu, laptop). legolas:/proc/asound # cat cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfc400000 irq 22 legolas:/proc/asound # cat pcm 00-01: Conexant Digital : Conexant Digital : playback 1 00-00: CONEXANT Analog : CONEXANT Analog : playback 1 : capture 1 legolas:/proc/asound # cat hwdep 00-00: HDA Codec 0 legolas:/proc/asound # cat devices 2: : timer 3: [ 0- 1]: digital audio playback 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0- 0]: hardware dependent 7: [ 0] : control 8: : sequencer legolas:/proc/asound # cat card0/codec#0 Codec: Conexant CX20549 (Venice) Address: 0 Vendor Id: 0x14f15045 Subsystem Id: 0x1734110e Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x140]: 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x10 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x2b 0x2b] Pincap 0x0810014: OUT EAPD Detect Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown Pin-ctls: 0x40: OUT Power: 0x0 Connection: 2 0x19 0x17* Node 0x11 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x2b 0x2b] Pincap 0x08113c: IN OUT HP Detect Pin Default 0x0121401f: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Power: 0x0 Connection: 2 0x19 0x17* Node 0x12 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x2b 0x2b] Pincap 0x08113c: IN OUT HP Detect Pin Default 0x01a19020: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x20: IN Power: 0x0 Connection: 2 0x19 0x17* Node 0x13 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x214571f0: [Jack] SPDIF Out at Sep Rear Conn = Optical, Color = Yellow Pin-ctls: 0x00: Connection: 1 0x18 Node 0x14 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081124: IN Detect Pin Default 0x97a7012e: [Fixed] Mic at Int Riser Conn = Analog, Color = Unknown Pin-ctls: 0x24: IN Node 0x15 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Node 0x16 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0b, mute=1 Amp-Out vals: [0x06] Node 0x17 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x14, nsteps=0x2b, stepsize=0x05, mute=1 Amp-In vals: [0x94 0x94] [0x94 0x94] [0x94 0x94] [0x94 0x94] [0x94 0x94] Power: 0x0 Connection: 5 0x19 0x14 0x12 0x11 0x15 Node 0x18 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x40]: 48000 bits [0x6]: 16 20 formats [0x5]: PCM AC3 Node 0x19 [Audio Output] wcaps 0xc11: Stereo PCM: rates [0x540]: 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: 0x0 Node 0x1a [Audio Input] wcaps 0x100d0b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x17, stepsize=0x05, mute=1 Amp-In vals: [0x17 0x17] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Power: 0x0 Connection: 5 0x17 0x14* 0x12 0x11 0x15 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Best regards Jan-Simon

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Re: [alsa-devel] [Jackit-devel] RME HDSP driver enhancements for HDSPe cards
by John Rigg 19 Oct '07

19 Oct '07

On Fri, Oct 19, 2007 at 08:58:00AM +0100, Ed W wrote: > RME are also helpful to the extent that you ask them specific questions > and in the name of writing drivers. They should give you hardware > details, etc > > > As an aside, apart from CPU load I will be suprised if turning floats > into 32bit ints makes any difference to the audio, but I agree that it > would be a nice feature to have! IIRC the new RME cards convert 24 bit integers to 32 bit floats in hardware (and vice versa), thus saving apps like jackd (and the CPU) the trouble. jackd doesn't support that yet, but AFAIK it shouldn't be too much effort to change that. John

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[alsa-devel] [PATCH] hda: Add dmux to STAC 9228
by Matthew Ranostay 19 Oct '07

19 Oct '07

Added a dmux to the STAC9228 cards with DMIC support. And added a STAC_DIGITAL_INPUT_SOURCE macro for repeating mixer code. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 821a4558dd47 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Oct 18 17:38:17 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Thu Oct 18 17:35:56 2007 -0400 @@ -430,6 +430,16 @@ static struct hda_verb stac9205_core_ini {} }; +#define STAC_DIGITAL_INPUT_SOURCE(cnt) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Digital Input Source", \ + .count = cnt, \ + .info = stac92xx_dmux_enum_info, \ + .get = stac92xx_dmux_enum_get, \ + .put = stac92xx_dmux_enum_put,\ + } + #define STAC_INPUT_SOURCE(cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -482,14 +492,7 @@ static struct snd_kcontrol_new stac925x_ }; static struct snd_kcontrol_new stac9205_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Digital Input Source", - .count = 1, - .info = stac92xx_dmux_enum_info, - .get = stac92xx_dmux_enum_get, - .put = stac92xx_dmux_enum_put, - }, + STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0), STAC_VOLKNOB(0x24), @@ -521,6 +524,7 @@ static struct snd_kcontrol_new stac922x_ static struct snd_kcontrol_new stac927x_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(3), STAC_VOLKNOB(0x24), STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB), @@ -2773,6 +2777,7 @@ static int patch_stac927x(struct hda_cod case 0x10280209: /* STAC 9228 */ spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; + spec->dmux_nid = 0x1c; break; default: spec->num_dmics = 0;

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[alsa-devel] ALSA Bluetooth connection issue
by 19 Oct '07

19 Oct '07

Hi All: Im using ALSA 1.0.14 and bluez 3.20 and linux 2.6.18.The linux 2.6.18 (montavista) is running on an embeded development board(using S3C2410). When i tried to use 'aplay' to connect my bluetooth headset, i got the following error from ALSA: $ aplay -D bluetooth /1.wav audio[326]: Accepted new client connection on unix socket (fd=8) audio[326]: Package PKT_TYPE_CFG_REQ:0 hcid[318]: /org/bluez: org.bluez.Manager.FindAdapter() audio[326]: Got path /org/bluez/hci0 for adapter with address 00:80:E1:00:00:00 hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceHandles() hcid[318]: no matching session found. creating a new one hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceHandles() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceHandles() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceRecord() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceRecord() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceRecord() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceRecord() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache audio[326]: Audio service discovery completed with success hcid[318]: /org/bluez/hci0: org.bluez.Adapter.FinishRemoteServiceTransaction() hcid[318]: found matching session, removing from list audio[326]: Found Hansfree record audio[326]: Discovered Headset service on RFCOMM channel 1 audio[326]: Found Headset record audio[326]: Found Audio Sink audio[326]: Selecting default device audio[326]: avdtp_ref(0x3b018): ref=2 audio[326]: a2dp_source_request_stream: selected SEP 0x37250 audio[326]: avdtp_ref(0x3b018): ref=3 hcid[318]: link_key_request (sba=00:80:E1:00:00:00, dba=00:0D:3C:EB:53:F0) audio[326]: AVDTP: connected signaling channel to 00:0D:3C:EB:53:F0 audio[326]: session_cb audio[326]: DISCOVER request succeeded audio[326]: seid 1 type 1 media 0 in use 0 audio[326]: seid 2 type 1 media 0 in use 0 audio[326]: session_cb audio[326]: GET_CAPABILITIES request succeeded audio[326]: seid 1 type 1 media 0 hcid[318]: link_key_request (sba=00:80:E1:00:00:00, dba=00:0D:3C:EB:53:F0) hcid[318]: link_key_request (sba=00:80:E1:00:00:00, dba=00:0D:3C:EB:53:F0) audio[326]: session_cb audio[326]: GET_CAPABILITIES request succeeded audio[326]: seid 2 type 1 media 0 audio[326]: Discovery complete audio[326]: avdtp_set_configuration(0x3b018): int_seid=1, acp_seid=1 audio[326]: session_cb audio[326]: SET_CONFIGURATION request succeeded audio[326]: SBC Source: Set_Configuration_Cfm audio[326]: avdtp_ref(0x3b018): ref=4 audio[326]: stream state changed: IDLE -> CONFIGURED audio[326]: session_cb audio[326]: OPEN request succeeded audio[326]: stream state changed: CONFIGURED -> OPEN audio[326]: AVDTP: connected transport channel to 00:0D:3C:EB:53:F0 audio[326]: SBC Source: Open_Cfm audio[326]: session_cb audio[326]: START request succeeded audio[326]: SBC Source: Start_CAlignment trap: bluetoothd-serv (326) PC=0x0001be44 Instr=0xe1c320b0 Address=0xbec85c2d FSR 0x813 fm audio[326]: SBC Source SEP 0x37250 locked hcid[318]: Service owner exited: :1.2 aplay: main:550: audio open error: Input/output error hcid[318]: Audio service (audio) was killed by signal 7 ALSA's aplay reports an "Input/output error". Later i digged into alsa-lib-1.0.14/src/pcm/pcm.c and found that snd_config_search_definition() returns an error. It seems ALSA didn't find the proper conf definition or something like that. My ".asoundrc" is as followed: pcm.a2dpd { type a2dpd } pcm.bluetooth { type bluetooth device 00:0D:3C:EB:53:F0 } pcm.!default { type hw card 0 device 0 } pcm.card0 { type hw card 0 } ctl.card0 { type hw card 0 } Does someone have any idea about what's going wrong in this? Thanks in advance! semiyd _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

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