June 2011 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [Fwd: [kde] Low maximum volume with pulseaudio on kde]
by Sérgio Basto 18 Jun '11

18 Jun '11

On Sat, 2011-06-18 at 13:12 -0400, Daniel Chen wrote: > > On Jun 18, 2011 1:05 PM, "Sérgio Basto" <sergio(a)serjux.com> wrote: > > > > On Sat, 2011-06-18 at 10:19 -0400, Daniel Chen wrote: > > > > > > Does the GNOME volume control in Sound Preferences show that the > > > slider is to the right of 100%? > > > > yes go to 150% > > > > -- > > Sérgio M. B. > > > > You should probably get in touch with Colin Guthrie, who made > modifications to Kmix to support PulseAudio. This isn't an alsa-devel > issue per se. so how I put kmix also control "software volume" since system use pulseaudio ? http://alsa.opensrc.org/How_to_use_softvol_to_control_the_master_volume and if I disable pulseaudio , how I configure softvols with kmix or AlsaMixer ? > Cheers, > -Dan > -- Sérgio M. B.

1 0

Re: [alsa-devel] [PATCH v4 11/18] input: Add initial support for TWL6040 vibrator
by Péter Ujfalusi 18 Jun '11

18 Jun '11

On Wednesday 15 June 2011 10:23:01 Tejun Heo wrote: > On Wed, Jun 15, 2011 at 10:18:58AM +0200, Tejun Heo wrote: > > No human being can feel 120usec difference and I can't see how using > > HIGHPRI is justified here (which is what the code is doing > > _accidentally_ by using singlethread_workqueue). > > Ooh, one more thing, and even if you insist on using HIGHPRI (please > don't), you don't need to create workqueue for each device. You can > just create one for the whole driver in init and destroy it from exit. > What matters is the HIGHPRI attribute of the workqueue. The number of > workqueues is completely irrelevant. Fair enough. I'll move to create_workqueue. If we later find issues with this (in a 'live' system), we can figure out a way to fix it. Thank you for your time on this. I'll make the changes accordingly. Regards, Péter

3 7

[alsa-devel] [PATCH] ASoC: Remove adau1701 from SND_SOC_ALL_CODECS due to Sigma dependency
by Mark Brown 18 Jun '11

18 Jun '11

The Sigma code is in drivers/firmware which is only included on a very small subset of architectures and so ends up breaking the build on others. There's a pending patch to make the directory build as standard but it's not merged yet. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- I already applied this due to the build breakage. sound/soc/codecs/Kconfig | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f745e55..7a2e426 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,7 +17,6 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 - select SND_SOC_ADAU1701 if I2C select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C -- 1.7.5.4

1 0

[alsa-devel] [PATCH 1/2] ASoC: Fix ADAU1701 build in SND_SOC_ALL_CODECS
by Mark Brown 18 Jun '11

18 Jun '11

Since select doesn't obey dependencies we need ALL_CODECS to select SIGMA directly. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bb90642..33ceb06 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" + select SIGMA if I2C # Analog devices DSP select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS -- 1.7.5.4

3 5

[alsa-devel] [PATCH] ALSA: HDA: Remove redundant LPIB quirks for ATI chipset
by David Henningsson 17 Jun '11

17 Jun '11

Now that we have changed the position_fix default for ATI and AMD to be LPIB (see commit 50e3bbf989), we can remove the quirks that were added for ATI chipsets. I went through the commits, most of them pointing to launchpad, where the PCI ID is easily available. I'm also attaching my notes in case it is helpful for anyone. Seeing that note list, we should probably also set "VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10)" to use LPIB as well. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] ALSA Setta Mixer Values
by Alfs Kurmis 17 Jun '11

17 Jun '11

Hi Experts ! How with C program i do List/Check available mixer channels for ALSA snd card X, how to determine is it input or output , how to set [and read] value(s) for any channel(s) , [if necessary] [ different for Left-Right if necessary ] Or simplest case - How to just setta Master and PCM to given values ? Which include(s)--Structures--Functions must be used ? Is somewhere any C copy-paste example or template available ? Tnx for examples in advance @ all. Alfs Kurmis ----

1 0

[alsa-devel] [PATCH] ASoC: Blackfin: allow SPI for SSM2602 parts
by Mike Frysinger 17 Jun '11

17 Jun '11

This board has hardware switches for selecting SPI or I2C, so don't require I2C for this driver. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/Kconfig | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 4d361bd..871308f 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -10,10 +10,9 @@ config SND_BF5XX_I2S config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio support for BF52x ezkit" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) select SND_BF5XX_SOC_I2S select SND_SOC_SSM2602 - select I2C help Say Y if you want to add support for SoC audio on BF527-EZKIT. -- 1.7.5.3

3 3

[PATCH] ALSA: HDA: Remove redundant LPIB quirks for ATI chipset
by David Henningsson 17 Jun '11

17 Jun '11

Now that we have changed the position_fix default for ATI and AMD to be LPIB (see commit 50e3bbf989), we can remove the quirks that were added for ATI chipsets. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 8 -------- 1 files changed, 0 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fdfd748..10819c8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2361,28 +2361,20 @@ static int azx_dev_free(struct snd_device *device) * white/black-listing for position_fix */ static struct snd_pci_quirk position_fix_list[] __devinitdata = { - SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1025, 0x026f, "Acer Aspire 5538", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1028, 0x0470, "Dell Inspiron 1120", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB), SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB), SND_PCI_QUIRK(0x1849, 0x0888, "775Dual-VSTA", POS_FIX_LPIB), SND_PCI_QUIRK(0x8086, 0x2503, "DG965OT AAD63733-203", POS_FIX_LPIB), - SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB), {} }; -- 1.7.4.1 --------------070903030003090708030002 Content-Type: text/plain; name="lpibcommits2.txt" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename="lpibcommits2.txt" commit 7a68be94e22e7643038726ebc14360752a91800b BugLink: https://launchpad.net/bugs/583983 Audio device [0403]: ATI Technologies Inc IXP SB4x0 High Definition Audio Controller [1002:437b] (rev 01) Subsystem: Acer Incorporated [ALI] Device [1025:009f] commit dd5a089edfa51a74692604b4b427953d8e16bc35 BugLink: https://launchpad.net/bugs/685161 Audio device [0403]: ATI Technologies Inc SBx00 Azalia (Intel HDA) [1002:4383] Subsystem: Acer Incorporated [ALI] Device [1025:026f] commit d2e1c97331d37a477ef7e9a272b4de4e53b893be Unknown hw commit 9919c7619c52d01e89103bca405cc3d4a2b1ac31 BugLink: https://launchpad.net/bugs/530346 Audio device [0403]: ATI Technologies Inc SBx00 Azalia (Intel HDA) [1002:4383] Subsystem: Dell Device [1028:01f6] commit e03fa055bc126e536c7f65862e08a9b143138ea9 Hardware info not present in buglink, but I know for sure it's ATI chipset. commit 2f703e7a2ea5f6d5ea14a7b2cd0d31be07839ac6 Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) 0403: 8086:293e (rev 03) Subsystem: 103c:306d commit dd37f8e8659bc617c3f2a84e007a4824ccdac458 BugLink: https://launchpad.net/bugs/465942 VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10) Subsystem: ASUSTeK Computer Inc. Device [1043:81b3] commit 9f75c1b12c5ef392ddcea575b13560842c28b1b3 BugLink: https://launchpad.net/bugs/587546 VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10) Subsystem: ASUSTeK Computer Inc. Device [1043:81e7] commit b540afc2b3d6e4cd1d1f137ef6d9e9c78d67fecd BugLink: http://bugs.launchpad.net/bugs/718402 Audio device [0403]: ATI Technologies Inc SBx00 Azalia (Intel HDA) [1002:4383] (rev 40) Subsystem: ASUSTeK Computer Inc. Device [1043:8410] commit e96d3127760a2fc509bca6bf7e61e8bc61497aeb BugLink: https://launchpad.net/bugs/586347 Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 05) 0403: 8086:3b56 (rev 05) Subsystem: 104d:9069 commit 4e0938dba7fccf37a4aecba4d937da7f312b5d55 BugLink: https://launchpad.net/bugs/549560 ATI Technologies Inc IXP SB4x0 High Definition Audio Controller [1002:437b] (rev 01) Subsystem: Toshiba America Info Systems Device [1179:ff10] commit 61bb42c37dfa9016dcacc86bcd41362ab2457d4a BugLink: https://launchpad.net/bugs/551949 Intel Corporation N10/ICH 7 Family High Definition Audio Controller [8086:27d8] (rev 01) Subsystem: Holco Enterprise Co, Ltd/Shuttle Computer Device [1297:3166] commit 9ec8ddad59fadd8021adfea4cb716a49b0e232e9 BugLink: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=575669 Unknown hw commit 45d4ebf1a6255f2234a041685789cbecac3453f1 unknown hw commit 0321b69569eadbc13242922925a4316754c5f744 BugLink: https://launchpad.net/bugs/523953 ATI Technologies Inc SBx00 Azalia (Intel HDA) [1002:4383] Subsystem: Biostar Microtech Int'l Corp Device [1565:820f] commit 8815cd030fdd73932a791d1f06194c8db807cde7 unknown hw commit b90c076424da8166797bdc34187660fd0124f530 BugLink: https://launchpad.net/bugs/580749 VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10) Subsystem: ASRock Incorporation Device [1849:0888] commit 0e0280dc2b0c7395a880d25544b47f3e3e3f79db BugLink: https://launchpad.net/bugs/459083 Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 02) Subsystem: Intel Corporation Device [8086:2503] commit 572c0e3c73341755f3e7dfaaef6b26df12bd709c BugLink: https://bugs.launchpad.net/bugs/538895 ATI Technologies Inc IXP SB4x0 High Definition Audio Controller [1002:437b] (rev 01) Subsystem: Intel Corporation Device [8086:d601] --------------070903030003090708030002--

1 0

[alsa-devel] ASoC: "No soundcards found" but 'snd_soc_instantiate_card' succeeds afterwards
by Lambrecht Jürgen 17 Jun '11

17 Jun '11

Hello, I have a imx27 ARM9 with MC13783 codec (all from Freescale). alsa init does not find a sound card, but finally there is a sound card. cat /proc/asound/cards: 0 [iMXmc13783audio]: - iMX-mc13783-audio iMX-mc13783-audio Maybe the inits are done in a wrong order, but how to fix? Here some parts of the prints: [start] bus: 'platform': add driver mc13783-codec bus: 'platform': driver_probe_device: matched device mc13783-codec with driver mc13783-codec bus: 'platform': really_probe: probing driver mc13783-codec with device mc13783-codec mc13783-codec mc13783-codec: codec register mc13783-codec mc13783-codec mc13783-codec: dai register mc13783-codec #1 Registered DAI 'mc13783-hifi' Registered codec 'mc13783-codec' driver: 'mc13783-codec': driver_bound: bound to device 'mc13783-codec' bus: 'platform': really_probe: bound device mc13783-codec to driver mc13783-codec bus: 'platform': add driver imx-ssi bus: 'platform': driver_probe_device: matched device imx-ssi.0 with driver imx-ssi bus: 'platform': really_probe: probing driver imx-ssi with device imx-ssi.0 imx-ssi imx-ssi.0: dai register imx-ssi.0 Registered DAI 'imx-ssi.0' Registering platform device 'imx-fiq-pcm-audio.0'. Parent at platform device: 'imx-fiq-pcm-audio.0': device_add bus: 'platform': add device imx-fiq-pcm-audio.0 Registering platform device 'imx-pcm-audio.0'. Parent at platform device: 'imx-pcm-audio.0': device_add bus: 'platform': add device imx-pcm-audio.0 driver: 'imx-ssi.0': driver_bound: bound to device 'imx-ssi' bus: 'platform': really_probe: bound device imx-ssi.0 to driver imx-ssi bus: 'platform': add driver imx-pcm-audio bus: 'platform': driver_probe_device: matched device imx-pcm-audio.0 with driver imx-pcm-audio bus: 'platform': really_probe: probing driver imx-pcm-audio with device imx-pcm-audio.0 imx-pcm-audio imx-pcm-audio.0: platform register imx-pcm-audio.0 asoc: snd_soc_instantiate_cards Registered platform 'imx-pcm-audio.0' driver: 'imx-pcm-audio.0': driver_bound: bound to device 'imx-pcm-audio' bus: 'platform': really_probe: bound device imx-pcm-audio.0 to driver imx-pcm-audio ALSA device list: No soundcards found. [end] -> then a sound card is registered by calling 'late_initcall(imx_mc13783_audio_init);' [start] Registering platform device 'soc-audio'. Parent at platform device: 'soc-audio': device_add bus: 'platform': add device soc-audio bus: 'platform': driver_probe_device: matched device soc-audio with driver soc-audio bus: 'platform': really_probe: probing driver soc-audio with device soc-audio asoc: snd_soc_instantiate_card iMX-mc13783-audio soc-audio soc-audio: binding MC13783 Playback at idx 0 soc-audio soc-audio: binding MC13783 Capture at idx 1 asoc: created sound card for card iMX-mc13783-audio asoc: sound card iMX-mc13783-audio probed soc-audio soc-audio: probe iMX-mc13783-audio dai link 0 device: 'MC13783 Playback': device_add soc-audio soc-audio: registered pcm #0 Playback mc13783-hifi-0 asoc: mc13783-hifi <-> imx-ssi.0 mapping ok asoc: sound card iMX-mc13783-audio instantiated soc-audio soc-audio: probe iMX-mc13783-audio dai link 1 device: 'MC13783 Capture': device_add soc-audio soc-audio: registered pcm #1 Capture mc13783-hifi-1 asoc: mc13783-hifi <-> imx-ssi.0 mapping ok asoc: sound card iMX-mc13783-audio instantiated device: 'card0': device_add device: 'pcmC0D1p': device_add device: 'pcmC0D1c': device_add device: 'adsp': device_add device: 'pcmC0D0p': device_add device: 'pcmC0D0c': device_add device: 'dsp': device_add device: 'audio': device_add device: 'controlC0': device_add device: 'mixer': device_add asoc: sound card iMX-mc13783-audio registered [end] Regards, Jürgen P.S.: to enable pr_debug and dev_dbg I had to add them myself in soc-core.c (see my previous mail) -- Jürgen Lambrecht R&D Associate Tel: +32 (0)51 303045 Fax: +32 (0)51 310670 http://www.televic-rail.com Televic Rail NV - Leo Bekaertlaan 1 - 8870 Izegem - Belgium Company number 0825.539.581 - RPR Kortrijk

2 2

[alsa-devel] [PATCH] sound: codec: tlv320aic3x: remove mono controls
by Jan Weitzel 17 Jun '11

17 Jun '11

if codec driver is used for AIC3X_MODEL_3007 the mono iout controls overwrite registers for class-d amplifier. classd amplifier controls are only used for AIC3X_MODEL_3007. Removing all mono snd_kcontrol_new, snd_soc_dapm_widget, snd_soc_dapm_route and aic3x_init stuff from common code and call only for not AIC3X_MODEL_3007 codecs. Testet only with AIC3X_MODEL_3007 Signed-off-by: Jan Weitzel <j.weitzel(a)phytec.de> --- sound/soc/codecs/tlv320aic3x.c | 106 ++++++++++++++++++++++++--------------- 1 files changed, 65 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index c3d96fc..7146832 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -324,16 +324,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume", - LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, - 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("Mono PGA Bypass Volume", - PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, - 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("Mono DAC Playback Volume", - DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, - 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("HP Line2 Bypass Volume", LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), @@ -357,7 +347,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Output pin mute controls */ SOC_DOUBLE_R("Line Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3, 0x01, 0), - SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("HPCOM Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, @@ -377,6 +366,19 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), }; +static const struct snd_kcontrol_new aic3x_mono_controls[] = { + SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume", + LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("Mono PGA Bypass Volume", + PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("Mono DAC Playback Volume", + DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), +}; + /* * Class-D amplifier gain. From 0 to 18 dB in 6 dB steps */ @@ -526,9 +528,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_PGA("Right HP Out", HPROUT_CTRL, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Right HP Com", HPRCOM_CTRL, 0, 0, NULL, 0), - /* Mono Output */ - SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0), - /* Inputs to Left ADC */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0), SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, @@ -590,9 +589,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Right Line Mixer", SND_SOC_NOPM, 0, 0, &aic3x_right_line_mixer_controls[0], ARRAY_SIZE(aic3x_right_line_mixer_controls)), - SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, - &aic3x_mono_mixer_controls[0], - ARRAY_SIZE(aic3x_mono_mixer_controls)), SND_SOC_DAPM_MIXER("Left HP Mixer", SND_SOC_NOPM, 0, 0, &aic3x_left_hp_mixer_controls[0], ARRAY_SIZE(aic3x_left_hp_mixer_controls)), @@ -608,7 +604,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LLOUT"), SND_SOC_DAPM_OUTPUT("RLOUT"), - SND_SOC_DAPM_OUTPUT("MONO_LOUT"), SND_SOC_DAPM_OUTPUT("HPLOUT"), SND_SOC_DAPM_OUTPUT("HPROUT"), SND_SOC_DAPM_OUTPUT("HPLCOM"), @@ -630,6 +625,18 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Detection"), }; +static const struct snd_soc_dapm_widget aic3x_dapm_mono_widgets[] = { + /* Mono Output */ + SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0), + + /* Output mixers */ + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_mono_mixer_controls[0], + ARRAY_SIZE(aic3x_mono_mixer_controls)), + + SND_SOC_DAPM_OUTPUT("MONO_LOUT"), +}; + static const struct snd_soc_dapm_widget aic3007_dapm_widgets[] = { /* Class-D outputs */ SND_SOC_DAPM_PGA("Left Class-D Out", CLASSD_CTRL, 3, 0, NULL, 0), @@ -714,17 +721,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right Line Out", NULL, "Right DAC Mux"}, {"RLOUT", NULL, "Right Line Out"}, - /* Mono Output */ - {"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, - {"Mono Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, - {"Mono Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Mono Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, - {"Mono Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, - {"Mono Mixer", "DACR1 Switch", "Right DAC Mux"}, - - {"Mono Out", NULL, "Mono Mixer"}, - {"MONO_LOUT", NULL, "Mono Out"}, - /* Left HP Output */ {"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, @@ -780,6 +776,19 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPRCOM", NULL, "Right HP Com"}, }; +static const struct snd_soc_dapm_route intercon_mono[] = { + /* Mono Output */ + {"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Mono Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, + {"Mono Mixer", "DACL1 Switch", "Left DAC Mux"}, + {"Mono Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + {"Mono Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, + {"Mono Mixer", "DACR1 Switch", "Right DAC Mux"}, + + {"Mono Out", NULL, "Mono Mixer"}, + {"MONO_LOUT", NULL, "Mono Out"}, +}; + static const struct snd_soc_dapm_route intercon_3007[] = { /* Class-D outputs */ {"Left Class-D Out", NULL, "Left Line Out"}, @@ -804,6 +813,11 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) ARRAY_SIZE(aic3007_dapm_widgets)); snd_soc_dapm_add_routes(dapm, intercon_3007, ARRAY_SIZE(intercon_3007)); + } else { + snd_soc_dapm_new_controls(dapm, aic3x_dapm_mono_widgets, + ARRAY_SIZE(aic3x_dapm_mono_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_mono, + ARRAY_SIZE(intercon_mono)); } return 0; @@ -1290,17 +1304,12 @@ static int aic3x_init(struct snd_soc_codec *codec) /* DAC to Line Out default volume and route to Output mixer */ snd_soc_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON); snd_soc_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - /* DAC to Mono Line Out default volume and route to Output mixer */ - snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); /* unmute all outputs */ reg = snd_soc_read(codec, LLOPM_CTRL); snd_soc_write(codec, LLOPM_CTRL, reg | UNMUTE); reg = snd_soc_read(codec, RLOPM_CTRL); snd_soc_write(codec, RLOPM_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, MONOLOPM_CTRL); - snd_soc_write(codec, MONOLOPM_CTRL, reg | UNMUTE); reg = snd_soc_read(codec, HPLOUT_CTRL); snd_soc_write(codec, HPLOUT_CTRL, reg | UNMUTE); reg = snd_soc_read(codec, HPROUT_CTRL); @@ -1325,9 +1334,6 @@ static int aic3x_init(struct snd_soc_codec *codec) /* PGA to Line Out default volume, disconnect from Output Mixer */ snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); - /* PGA to Mono Line Out default volume, disconnect from Output Mixer */ - snd_soc_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL); - snd_soc_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL); /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); @@ -1337,14 +1343,30 @@ static int aic3x_init(struct snd_soc_codec *codec) /* Line2 Line Out default volume, disconnect from Output Mixer */ snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); - /* Line2 to Mono Out default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); if (aic3x->model == AIC3X_MODEL_3007) { aic3x_init_3007(codec); snd_soc_write(codec, CLASSD_CTRL, 0); + + return 0; } + /* Not on AIC3X_MODEL_3007 init */ + + /* DAC to Mono Line Out default volume and route to Output mixer */ + snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + + /* unmute outputs */ + reg = snd_soc_read(codec, MONOLOPM_CTRL); + snd_soc_write(codec, MONOLOPM_CTRL, reg | UNMUTE); + + /* PGA to Mono Line Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL); + + /* Line2 to Mono Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); return 0; } @@ -1423,7 +1445,9 @@ static int aic3x_probe(struct snd_soc_codec *codec) ARRAY_SIZE(aic3x_snd_controls)); if (aic3x->model == AIC3X_MODEL_3007) snd_soc_add_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); - + else + snd_soc_add_controls(codec, aic3x_mono_controls, + ARRAY_SIZE(aic3x_mono_controls)); aic3x_add_widgets(codec); list_add(&aic3x->list, &reset_list); -- 1.7.0.4

3 4