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

[alsa-devel] [PATCH] ASoC: Make SND_SOC_MX27VIS_AIC32X4 depend on I2C
by Axel Lin 05 Dec '11

05 Dec '11

SND_SOC_MX27VIS_AIC32X4 selects SND_SOC_TLV320AIC32X4, but SND_SOC_TLV320AIC32X4 needs CONFIG_I2C. So we need to make SND_SOC_MX27VIS_AIC32X4 depend on I2C. otherwise I got below build error if CONFIG_I2C is not selected. CC sound/soc/codecs/tlv320aic32x4.o sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_read': sound/soc/codecs/tlv320aic32x4.c:323: error: implicit declaration of function 'i2c_smbus_read_byte_data' sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_probe': sound/soc/codecs/tlv320aic32x4.c:641: error: 'i2c_master_send' undeclared (first use in this function) sound/soc/codecs/tlv320aic32x4.c:641: error: (Each undeclared identifier is reported only once sound/soc/codecs/tlv320aic32x4.c:641: error: for each function it appears in.) sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_modinit': sound/soc/codecs/tlv320aic32x4.c:763: error: implicit declaration of function 'i2c_add_driver' sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_exit': sound/soc/codecs/tlv320aic32x4.c:774: error: implicit declaration of function 'i2c_del_driver' make[3]: *** [sound/soc/codecs/tlv320aic32x4.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/imx/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index b133bfc..7383917 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -28,7 +28,7 @@ config SND_MXC_SOC_WM1133_EV1 config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" - depends on MACH_IMX27_VISSTRIM_M10 + depends on MACH_IMX27_VISSTRIM_M10 && I2C select SND_SOC_TLV320AIC32X4 select SND_MXC_SOC_MX2 help -- 1.7.5.4

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[alsa-devel] Clevo P170HM / Sager NP8170 audio
by Andrew Mahone 04 Dec '11

04 Dec '11

This laptop is detected as ALC892 audio (and by the codec ID this seems to be valid), but none of the configurations for it quite match this device. I've been working on documenting the connections with HDA Analyzer. alsa-info output: http://www.alsa-project.org/db/?f=b86e4c6447867fbd129a36d8a63e2f5caa836c52 The laptop has built-in 5.1, and 5.1 or 7.1 output via jacks. DAC NIDs: 0x02 front 0x03 rear 0x04 clfe 0x05 side internal mixers: 0x0c front 0x0d rear 0x0e clfe 0x0f side internal speaker pins: 0x14 front 0x15 "rear" (front of laptop) 0x16 clfe external ports: fixed route: 0x17 side can select any output: 0x18 mic / output 0x1a line-in / output 0x1b headphone I've been experimenting with patch_realtek.c to add the extra DAC NID, and have had some success with a 6-channel mode for the built-in speakers and an 8-channel mode that mutes all internal speakers and sets the jacks up for output. I still need to manually connect pins 0x18, 0x1a, 0x1b to the correct internal mixers for 7.1 to work, and headphone plugin does not mute internal speakers in 5.1 mode. I have not added a plain stereo mode for internal speakers, although it appears that i can probably configure the codec to mix the front channels to center and LFE. Are there any suggestions for what to do next? My current state as a patch against the linux-3.0.y repository is at http://pastie.org/2400649 -- Andrew Mahone andrew DOT mahone AT gmail DOT com Q: How many voters does it take to change a lightbulb? A: None - if voting could change anything they'd make it illegal

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Re: [alsa-devel] snd-usb-audio - NI Traktor 10 - phono settings - partly solved
by Peter Dunker 04 Dec '11

04 Dec '11

Hi Daniel, I tried amixer contents, which looks like more info, but it was not obvious: numid=1,iface=MIXER,name='Direct Thru Channel A' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=2,iface=MIXER,name='Direct Thru Channel B' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=3,iface=MIXER,name='Direct Thru Channel C' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=4,iface=MIXER,name='Direct Thru Channel D' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=5,iface=MIXER,name='Phono Input Channel A' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=6,iface=MIXER,name='Phono Input Channel B' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=7,iface=MIXER,name='Phono Input Channel C' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=8,iface=MIXER,name='Phono Input Channel D' ; type=BOOLEAN,access=rw------,values=1 : values=on I used now amixer -c 1 numid=<1..8> off and it worked! The channels are on line level now: numid=1,iface=MIXER,name='Direct Thru Channel A' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=2,iface=MIXER,name='Direct Thru Channel B' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=3,iface=MIXER,name='Direct Thru Channel C' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=4,iface=MIXER,name='Direct Thru Channel D' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=5,iface=MIXER,name='Phono Input Channel A' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=6,iface=MIXER,name='Phono Input Channel B' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=7,iface=MIXER,name='Phono Input Channel C' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=8,iface=MIXER,name='Phono Input Channel D' ; type=BOOLEAN,access=rw------,values=1 : values=off I have 2 problems left: 1) The amixer shows only 8, but the card has 10 channels. It might be no problem, as 2 channels were always line, but I have to double check, as soon as I have the card in hand (currently remote). Any idea, in case the 9. and 10. channel are still phono, where and how to look to set these channels too. 2) When I connect 2 cards, how can I for sure discriminate which is which and assign certain plugins in asoundrc to a specific one. I plan to use more then one T10. I read people use the name of the card to avoid confusion, but they have no wall the same name. The problem is that the order of card numbers is given by the order the cards are connected and its not clear what happens when reboot, maybe every time something different. Any idea, how I could define for sure, which one is which one? Maybe with a serial number, but I haven't seen one. Thanks, Peter Message: 5 Date: Sat, 03 Dec 2011 17:37:05 +0100 From: Daniel Mack <zonque(a)gmail.com> Subject: Re: [alsa-devel] snd-usb-audio - NI Traktor 10 - phono settings To: alsa-devel(a)alsa-project.org Message-ID: <4EDA5031.1050502(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1 On 12/03/2011 05:02 AM, Peter Dunker wrote: > Hi, I am new to this list, so please excuse if my questions are not > at the right target group. No, this is the perfectly right place for asking such questions :) For the record, please make sure you send your follow-ups to the list as well. > I am using since a couple days the NI Traktor 10 USB audio sound card > that has 10 channels. I use Ubuntu 11.10. and I was able to capture > with arecord from the device out of the box all 10 channels and I was > able to make an asoundrc to make 10 devices each 1 channel of it > too. > > My problem is now that this device enables by default 4 of the 10 > channels as phono level input, which leads to clipping and corrupt > audio on these channels when I feed line level. NI provides an config > app for Mac and Windows to set phono or non phono to the channels > that you like. Unfortunately, these settings do not stay as you set > under win or mac. There is no such tool, but controls for all 4 analog stereo pairs to configure the "direct thru" modes and the phono pre-amps should be exported as generic amixer controls. > My question, is there anything in this snd-usb-audio driver module > that could help me set this phono on/off values or can you guys think > of any other Linux tool that I could use to set these fields? I also > noticed that some channels actually changed the phono setting while > connecting to win and back to linux. So if you guys have an idea how > to set these values so that they might stay even when connect to > Linux, this would be helpful too. Post the output of "amixer controls", please. If there is more than one card connected and the TA10 is not your default, you might need to pass the "-D" option to amixer as well. See the output of "cat /proc/asound/cards" for the right value to pass. With this output, it should be easy to find the correct ALSA control to use with "amixer cset". Daniel

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[alsa-devel] [PATCH] ASoC: Make SND_SOC_LITTLEMILL select MFD_WM8994
by Axel Lin 04 Dec '11

04 Dec '11

SND_SOC_LITTLEMILL selects SND_SOC_WM8994, but SND_SOC_WM8994 needs MFD_WM8994. Thus we need to select MFD_WM8994 to fix below build error: LD .tmp_vmlinux1 sound/built-in.o: In function `wm8994_write': sound/soc/codecs/wm8994.c:201: undefined reference to `wm8994_reg_write' sound/built-in.o: In function `wm8994_read': sound/soc/codecs/wm8994.c:222: undefined reference to `wm8994_reg_read' sound/built-in.o: In function `wm8994_resume': sound/soc/codecs/wm8994.c:2847: undefined reference to `wm8994_reg_read' sound/built-in.o: In function `wm8994_codec_probe': sound/soc/codecs/wm8994.c:3501: undefined reference to `wm8994_reg_read' sound/soc/codecs/wm8994.c:3660: undefined reference to `wm8994_reg_read' sound/soc/codecs/wm8994.c:3672: undefined reference to `wm8994_reg_read' sound/built-in.o: In function `wm8958_dsp2_fw': sound/soc/codecs/wm8958-dsp2.c:154: undefined reference to `wm8994_bulk_write' make: *** [.tmp_vmlinux1] Error 1 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/samsung/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 09d636c..f3417f2 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -210,4 +210,5 @@ config SND_SOC_LITTLEMILL tristate "Audio support for Wolfson Littlemill" depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 select SND_SAMSUNG_I2S + select MFD_WM8994 select SND_SOC_WM8994 -- 1.7.5.4

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Re: [alsa-devel] [patch][saa7134] do not change mute state for capturing audio
by Mauro Carvalho Chehab 03 Dec '11

03 Dec '11

Em 24-07-2011 14:45, Stas Sergeev escreveu: > 23.07.2011 19:09, Mauro Carvalho Chehab wrote: >> > In this case, it will not be autounmuted after tuning. >> Hard to tell about your solution without seeing a patch. > OK, it turns out the automute code is already there, > but it doesn't work. The driver for some reasons > starts the scan on initialization, finds the carrier: > --- > saa7134[0]/audio: found PAL main sound carrier @ 6.000 MHz [3969/324] > --- > and, because of that, disables the automute. If the > real mute is not enabled at that point, you get the > white noise right away. The automute code works fine. Maybe you have a strong interference at the default tuning frequency, leading into saa7134 miss-detection. For saa7134 driver: 1) There is an audio carrier; 2) an application wants to listen audio; 3) the device is at automute mode. So, when there's an audio carrier, it will unmute the audio at streaming start. The driver is doing the _right_ thing by letting the audio to flow to PA, when it starts the capture. Unfortunately, on your specific device, starting audio capture also enables audio at the audio output pin. So, you're noticing a problem. People with a saa7134 device without alsa stream won't notice it (old devices). People with alsa stream without anything connected to the LINE OUT people aren't noticing it, if PA is not copying the saa7134 PCM IN stream to the sound card PCM out device (the default, for PA). So, only people that has saa7134 with alsa stream that opted to wire the saa7134 device to the sound card, and with a strong interference at the default tunning frequency (400 MHz) would notice a problem. > Since I have no idea why it finds some carrier, I can't > fix that in any way. Or, maybe, not to call the scan > on driver init? What will that break? Analog tuners need to be tuned at the device init on a high frequency according with their datasheets, otherwise the PLL may fail. > Anyway, as long as the automute code is broken, > we should either start fixing it, or fix PA, or fix mplayer... As I always said, the fix should be at PA, as it makes no sense to start capturing at saa7134 without first configuring it. So, or PA should be converted into a V4L2-aware application, in order to properly init the device (with seems to be the wrong approach) or it _SHOULD_NOT_ automatically enable capture on those devices, as this may cause undesired side effects, on the devices that have the capture pin directly wired to the output pin, witch seems to be the case of your device. PS.: it seems that you've removed Lennart/alsa people from the c/c. I'm re-adding them, as I'm expecting a fix from their side. > Dunno. I wonder how come so many bugs left unfixed > for so long, resulting in a white noise to people... You're the first one that reported it, and the code is there for _years_. So, this is not a commonly noticed problem at all. Mauro.

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[alsa-devel] snd-usb-audio - NI Traktor 10 - phono settings
by Peter Dunker 03 Dec '11

03 Dec '11

Hi, I am new to this list, so please excuse if my questions are not at the right target group. First of all, awesome work of the alsa developer to provide these system and modules! I am using since a couple days the NI Traktor 10 USB audio sound card that has 10 channels. I use Ubuntu 11.10. and I was able to capture with arecord from the device out of the box all 10 channels and I was able to make an asoundrc to make 10 devices each 1 channel of it too. My problem is now that this device enables by default 4 of the 10 channels as phono level input, which leads to clipping and corrupt audio on these channels when I feed line level. NI provides an config app for Mac and Windows to set phono or non phono to the channels that you like. Unfortunately, these settings do not stay as you set under win or mac. My question, is there anything in this snd-usb-audio driver module that could help me set this phono on/off values or can you guys think of any other Linux tool that I could use to set these fields? I also noticed that some channels actually changed the phono setting while connecting to win and back to linux. So if you guys have an idea how to set these values so that they might stay even when connect to Linux, this would be helpful too. Thanks in advance for reading, Peter

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[alsa-devel] [PATCH] ASoC: kirkwood: Make SND_KIRKWOOD_SOC_OPENRD and SND_KIRKWOOD_SOC_T5325 depend on I2C
by Axel Lin 03 Dec '11

03 Dec '11

SND_KIRKWOOD_SOC_T5325 selects SND_SOC_ALC5623, but SND_SOC_ALC5623 needs CONFIG_I2C. So we need to make SND_KIRKWOOD_SOC_T5325 depend on I2C, otherwise I got below build error if CONFIG_I2C is not selected. CC sound/soc/codecs/alc5623.o sound/soc/codecs/alc5623.c: In function 'alc5623_i2c_probe': sound/soc/codecs/alc5623.c:1002: error: implicit declaration of function 'i2c_smbus_read_word_data' sound/soc/codecs/alc5623.c:1009: error: implicit declaration of function 'i2c_smbus_read_byte_data' sound/soc/codecs/alc5623.c: In function 'alc5623_modinit': sound/soc/codecs/alc5623.c:1096: error: implicit declaration of function 'i2c_add_driver' sound/soc/codecs/alc5623.c: In function 'alc5623_modexit': sound/soc/codecs/alc5623.c:1108: error: implicit declaration of function 'i2c_del_driver' make[3]: *** [sound/soc/codecs/alc5623.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Also fix the same issue for SND_KIRKWOOD_SOC_OPENRD. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/kirkwood/Kconfig | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 8f49e16..c62d715 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -12,6 +12,7 @@ config SND_KIRKWOOD_SOC_I2S config SND_KIRKWOOD_SOC_OPENRD tristate "SoC Audio support for Kirkwood Openrd Client" depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE) + depends on I2C select SND_KIRKWOOD_SOC_I2S select SND_SOC_CS42L51 help @@ -20,7 +21,7 @@ config SND_KIRKWOOD_SOC_OPENRD config SND_KIRKWOOD_SOC_T5325 tristate "SoC Audio support for HP t5325" - depends on SND_KIRKWOOD_SOC && MACH_T5325 + depends on SND_KIRKWOOD_SOC && MACH_T5325 && I2C select SND_KIRKWOOD_SOC_I2S select SND_SOC_ALC5623 help -- 1.7.5.4

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[alsa-devel] [PATCH 0/6] Random WM8903 updates
by Mark Brown 02 Dec '11

02 Dec '11

This is a random collection of WM8903 updates that I started coding because I was looking at the driver and spotted a few style issues but haven't actually tested yet. I'm pushing them out now mostly because Stephen is also working on the driver and I want to avoid collisions. There's also a patch to convert to devm_kzalloc() that I posted earlier but that's trivial enough. It's pretty nice that even though I added a readable register function for regmap listing every register in the chip the overall diffstat is neutral, mostly as a result of the more efficient cache specification format that regmap has. Mark Brown (6): ASoC: Use table based control init for WM8903 ASoC: WM8903 only supports I2C so don't ifdef it ASoC: Use a normal cache sync for WM8903 ASoC: Don't resync WM8903 register cache on reset ASoC: Convert WM8903 to direct regmap API usage ASoC: Move initial WM8903 identification and reset to I2C probe sound/soc/codecs/wm8903.c | 449 +++++++++++++++++++++++---------------------- 1 files changed, 225 insertions(+), 224 deletions(-)

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[alsa-devel] [PATCH] ASoC: Convert WM8903 to devm_kzalloc()
by Mark Brown 02 Dec '11

02 Dec '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 7 +++---- sound/soc/codecs/wm8960.c | 23 ++++++++++++++++------- 2 files changed, 19 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 12dae43..fa44c9a 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2281,7 +2281,8 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, struct wm8903_priv *wm8903; int ret; - wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL); + wm8903 = devm_kzalloc(&i2c->dev, sizeof(struct wm8903_priv), + GFP_KERNEL); if (wm8903 == NULL) return -ENOMEM; @@ -2290,15 +2291,13 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8903, &wm8903_dai, 1); - if (ret < 0) - kfree(wm8903); + return ret; } static __devexit int wm8903_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index ed2773f..4393394 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -14,6 +14,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -71,6 +72,7 @@ static const u16 wm8960_reg[WM8960_CACHEREGNUM] = { struct wm8960_priv { enum snd_soc_control_type control_type; + void *control_data; int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); struct snd_soc_dapm_widget *lout1; @@ -264,7 +266,7 @@ SND_SOC_DAPM_INPUT("RINPUT2"), SND_SOC_DAPM_INPUT("LINPUT3"), SND_SOC_DAPM_INPUT("RINPUT3"), -SND_SOC_DAPM_SUPPLY("MICB", WM8960_POWER1, 1, 0, NULL, 0), +SND_SOC_DAPM_MICBIAS("MICB", WM8960_POWER1, 1, 0), SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8960_POWER1, 5, 0, wm8960_lin_boost, ARRAY_SIZE(wm8960_lin_boost)), @@ -573,8 +575,6 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); - /* Enable anti-pop features */ snd_soc_write(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | @@ -677,9 +677,6 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, WM8960_VREF | WM8960_VMID_MASK, 0); break; - case SND_SOC_BIAS_OFF: - snd_soc_cache_sync(codec); - break; default: break; } @@ -868,7 +865,7 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec, (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) -static const struct snd_soc_dai_ops wm8960_dai_ops = { +static struct snd_soc_dai_ops wm8960_dai_ops = { .hw_params = wm8960_hw_params, .digital_mute = wm8960_mute, .set_fmt = wm8960_set_dai_fmt, @@ -905,6 +902,16 @@ static int wm8960_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8960_resume(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8960_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -918,6 +925,7 @@ static int wm8960_probe(struct snd_soc_codec *codec) u16 reg; wm8960->set_bias_level = wm8960_set_bias_level_out3; + codec->control_data = wm8960->control_data; if (!pdata) { dev_warn(codec->dev, "No platform data supplied\n"); @@ -1007,6 +1015,7 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8960); wm8960->control_type = SND_SOC_I2C; + wm8960->control_data = i2c; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); -- 1.7.7.3

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[alsa-devel] [PATCH] ASoC: Map microphones on Littlemill
by Mark Brown 02 Dec '11

02 Dec '11

Littlemill has one analogue microphone on the board (connected to IN1LN) and an array of four DMICs connected to both DMICDAT lines. The biases can be selected by jumpers but pick the default jumper fit. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/littlemill.c | 10 ++++++++++ 1 files changed, 10 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 5d7680f..5cea59b 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -147,11 +147,21 @@ static struct snd_soc_dai_link littlemill_dai[] = { static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), + + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), }; static struct snd_soc_dapm_route audio_paths[] = { { "Headphone", NULL, "HPOUT1L" }, { "Headphone", NULL, "HPOUT1R" }, + + { "AMIC", NULL, "MICBIAS1" }, /* Default for AMICBIAS jumper */ + { "IN1LN", NULL, "AMIC" }, + + { "DMIC", NULL, "MICBIAS2" }, /* Default for DMICBIAS jumper */ + { "DMIC1DAT", NULL, "DMIC" }, + { "DMIC2DAT", NULL, "DMIC" }, }; static struct snd_soc_jack littlemill_headset; -- 1.7.7.3

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