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

[alsa-devel] [PATCH 0/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 14 Feb '10

14 Feb '10

The following patch set adds suspend/resume support to all echoaudio drivers. It has been tested on an IndigoIOx and a Gina3G. Rawmidi device has been tested partially. PCM and controls interfaces have been tested more and I think it's reasonably safe. I also have a Gina24 and a Gina3G, but I didn't test suspend/resume with them because I couldn't manage to make PM test system work on my pmac. Echoaudio soundcards do not support power management and Dx PCI states. Anyway, the card can enter a "comatose" state by sending a command to the DSP that is supposed to turn parts of the hardware off and reduce power consumption. Doing that causes the loss of the card's internal state, thus it is impossible, upon resume, to make the card continue from the exact point where it was suspended. The driver restores all controls and PCM settings and the applications have to call prepare() to restart the substreams. Rawmidi is restarted automatically, but some data in the i/o buffers may be lost. -- Giuliano.

1 0

[alsa-devel] dmix and dsnoop
by Raymond Yau 14 Feb '10

14 Feb '10

why do dmix and dsnoop 's slave alway open subdevice 0 ? for those hardware mixing sound cards which has two or more playback subdevices or hda-intel which may have 3 recording subdevices when subdevice 0 is opened by another application , dmix / dsnoop always fail even when there are available subdevices aplay -v -Dplug:dmix test.wav ALSA lib pcm_dmix.c:1010:(snd_pcm_dmix_open) unable to open slave aplay: main:654: audio open error: Device or resource busy arecord -v -Dplug:dsnoop:1 -f cd -d20 tt.wav ALSA lib pcm_dsnoop.c:593:(snd_pcm_dsnoop_open) unable to open slave arecord: main:654: audio open error: Device or resource busy

1 0

[alsa-devel] [PATCH 1/2] ASoC: Make pmdown_time a per-card setting
by Mark Brown 13 Feb '10

13 Feb '10

Make the pmdown_time a per-card setting rather than a global one, initialised before the card initialisation runs. This allows cards to override the default setting if it makes sense to do so (for example, due to an unavoidable pop). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 4 +++- 2 files changed, 5 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e6a6d10..d9d88dd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -521,6 +521,8 @@ struct snd_soc_card { int (*set_bias_level)(struct snd_soc_card *, enum snd_soc_bias_level level); + int pmdown_time; + /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; int num_links; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ca89c78..94b9cde 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -542,7 +542,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream) /* start delayed pop wq here for playback streams */ codec_dai->pop_wait = 1; schedule_delayed_work(&card->delayed_work, - msecs_to_jiffies(pmdown_time)); + msecs_to_jiffies(card->pmdown_time)); } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(codec, @@ -1039,6 +1039,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) dev_dbg(card->dev, "All components present, instantiating\n"); /* Found everything, bring it up */ + card->pmdown_time = pmdown_time; + if (card->probe) { ret = card->probe(pdev); if (ret < 0) -- 1.6.6.1

3 4

Re: [alsa-devel] HP 6930p mute button/led out of sync; fixed by suspend/resume cycle
by drago01 12 Feb '10

12 Feb '10

On Sun, Jan 10, 2010 at 3:31 PM, drago01 <drago01(a)gmail.com> wrote: > Hi, > > I experience an odd issue with my HP6930p laptop. > > After boot pressing the mute button/led causes the audio to be muted > but the sate is not reflected in software (i.e the mixer does not show > the control as muted). > However pressing after a suspend/resume cycle it works as expected. > The hardware mute button/led and the software control are in sync > (pressing the hw button toggles the mixer to mute/unmute; muting the > software mixer results into the led turning red). > > This was with 2.6.31.9 and 2.6.32.1; also installing alsa-drivers from > source does not change this. > > It seems something is not initialized correctly during boot up but is > done on resume, any pointers on how to debug this? > (Resending to correct list)

2 5

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 12 Feb '10

12 Feb '10

The following changes since commit c6848bf566c7217a6090693ff5cc47091fa772f5: Paul Menzel (1): ASoC: Typo. s/Freecale/Freescale/ are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Guennadi Liakhovetski (1): ASoC: fix compile breakage if CONFIG_SH_DMA_API=y && CONFIG_SND_SIU_MIGOR!=n Mark Brown (3): ASoC: Add WM2000 driver ASoC: Make pmdown_time a per-card setting ASoC: Make pmdown_time runtime configurable Thomas Weber (1): ASoC: Add ASoC support for Devkit8000 include/sound/soc.h | 2 + include/sound/wm2000.h | 26 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm2000.c | 888 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm2000.h | 79 ++++ sound/soc/omap/Kconfig | 8 +- sound/soc/omap/omap3beagle.c | 6 +- sound/soc/sh/Kconfig | 1 + sound/soc/soc-core.c | 31 ++- 10 files changed, 1040 insertions(+), 7 deletions(-) create mode 100644 include/sound/wm2000.h create mode 100644 sound/soc/codecs/wm2000.c create mode 100644 sound/soc/codecs/wm2000.h

1 0

[alsa-devel] [PATCH] ASoC: fix compile breakage if CONFIG_SH_DMA_API=y && CONFIG_SND_SIU_MIGOR!=n
by Guennadi Liakhovetski 12 Feb '10

12 Feb '10

Audio on Migo-R cannot work if CONFIG_SH_DMA_API=y, but compilation should not break anyway. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- sound/soc/sh/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index a86696b..1066749 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_SH4_FSI config SND_SOC_SH4_SIU tristate depends on (SUPERH || ARCH_SHMOBILE) && HAVE_CLK + select DMA_ENGINE select DMADEVICES select SH_DMAE -- 1.6.2.4

2 1

[alsa-devel] Jack Detection ??
by Johnson, Charles F 12 Feb '10

12 Feb '10

I'm trying to understand the level of jack detection implemented in ALSA. I admit some amount of ignorance here - is this documented somewhere that I haven't been able to find yet ?? Thanks. Charles Johnson Ultra-Mobility Group UPSG Architecture Intel Corporation charles.f.johnson(a)intel.com

2 1

[alsa-devel] IEC958 / SPDIF raw capture
by Donald Tournier 12 Feb '10

12 Feb '10

Hi, We have an application in our institute that will require recording of a digital signal via an optical cable. We have to be able to guarantee perfect bitwise reproduction of the signal (the odd bit error is not a problem). We're thinking of using SPDIF and ALSA to capture the signal (obviously we'll have to make sure the signal is properly encoded to begin with). However, after some initial attempts, it's becoming clear that ALSA can't necessarily guarantee bitwise perfect capture. ALSA might change the format, or resample the signal after capture (although I've heard that can be turned off with "snd_pcm_hw_params_set_rate_resample (capture_handle, hw_params, 0)"). Also, the card that we've been playing with so far (SB Audigy2 ZS) apparently performs some low-pass digital filtering on the input stream anyway, and is therefore no use to us. So my question is: can this be done? Is there a way of setting up ALSA to produce an unaltered version of the SPDIF input stream? This will undoubtedly require a particular combination of hardware and driver. If so, which card/driver would be recommended? Ideally, we would like to capture the raw SPDIF subframes. Is that possible, and if so, how? Any help would be greatly appreciated! Thanks, Donald. -- Jacques-Donald Tournier (PhD) Brain Research Institute, Melbourne, Australia Tel: +61 (0)3 9496 4078

3 4

[alsa-devel] [PATCH] ALSA: hda - Add Macmini 3,1 support
by Luke Yelavich 12 Feb '10

12 Feb '10

From: Luke Yelavich <luke.yelavich(a)canonical.com> BugLink: https://bugs.edge.launchpad.net/ubuntu/+source/linux/+bug/343989 Add a model quirk for the NVIDIA based Macmini hardware, aka Macmini 3,1. The pinout is almost identical to the mb5 quirk, except for no microphone and the line-in mixer controls being on a different index. Everything works in 2ch mode, but as I am not sure what needs to be changed for 6ch mode, or whether the Mac Mini's chip supports 6ch mode, I have simply duplicated the code from the mb5 quirk for the mac mini chmode management. The new model parameter for this quirk is "macmini3". Signed-off-by: Luke Yelavich <luke.yelavich(a)canonical.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 136 ++++++++++++++++++++++++++ 2 files changed, 137 insertions(+), 0 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 8f06f20..0bbdbb8 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -124,6 +124,7 @@ ALC882/883/885/888/889 asus-a7m ASUS A7M macpro MacPro support mb5 Macbook 5,1 + macmini3 Macmini 3,1 mbp3 Macbook Pro rev3 imac24 iMac 24'' with jack detection imac91 iMac 9,1 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0c22497..b5a6ba0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -211,6 +211,7 @@ enum { ALC885_MACPRO, ALC885_MBP3, ALC885_MB5, + ALC885_MACMINI3, ALC885_IMAC24, ALC885_IMAC91, ALC883_3ST_2ch_DIG, @@ -6751,6 +6752,14 @@ static struct hda_input_mux mb5_capture_source = { }, }; +static struct hda_input_mux macmini3_capture_source = { + .num_items = 2, + .items = { + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + static struct hda_input_mux alc883_3stack_6ch_intel = { .num_items = 4, .items = { @@ -6999,6 +7008,35 @@ static struct hda_channel_mode alc885_mb5_6ch_modes[2] = { { 6, alc885_mb5_ch6_init }, }; +/* + * 2ch + * Speakers/Woofer/HP = Front + * LineIn = Input + */ +static struct hda_verb alc885_macmini3_ch2_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + { } /* end */ +}; + +/* + * 6ch mode + * Speakers/HP = Front + * Woofer = LFE + * LineIn = Surround + */ +static struct hda_verb alc885_macmini3_ch6_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + { } /* end */ +}; + +static struct hda_channel_mode alc885_macmini3_6ch_modes[2] = { + { 2, alc885_mb5_ch2_init }, + { 6, alc885_mb5_ch6_init }, +}; + /* * 2ch mode @@ -7243,6 +7281,21 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc885_macmini3_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc885_imac91_mixer[] = { HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT), @@ -7617,6 +7670,53 @@ static struct hda_verb alc885_mb5_init_verbs[] = { { } }; +/* Macmini 3,1 */ +static struct hda_verb alc885_macmini3_init_verbs[] = { + /* DACs */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Front mixer */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Surround mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* LFE mixer */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* HP mixer */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Front Pin (0x0c) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* LFE Pin (0x0e) */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* HP Pin (0x0f) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* Line In pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + { } +}; + /* Macbook Pro rev3 */ static struct hda_verb alc885_mbp3_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ @@ -7800,6 +7900,18 @@ static void alc885_mb5_automute(struct hda_codec *codec) } +static void alc885_macmini3_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + static void alc885_mb5_unsol_event(struct hda_codec *codec, unsigned int res) { @@ -7808,6 +7920,14 @@ static void alc885_mb5_unsol_event(struct hda_codec *codec, alc885_mb5_automute(codec); } +static void alc885_macmini3_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Headphone insertion or removal. */ + if ((res >> 26) == ALC880_HP_EVENT) + alc885_mb5_automute(codec); +} + static void alc885_imac91_automute(struct hda_codec *codec) { unsigned int present; @@ -8974,6 +9094,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC882_ASUS_A7M] = "asus-a7m", [ALC885_MACPRO] = "macpro", [ALC885_MB5] = "mb5", + [ALC885_MACMINI3] = "macmini3", [ALC885_MBP3] = "mbp3", [ALC885_IMAC24] = "imac24", [ALC885_IMAC91] = "imac91", @@ -9157,6 +9278,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { */ SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), + SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC885_MACMINI3), {} /* terminator */ }; @@ -9238,6 +9360,20 @@ static struct alc_config_preset alc882_presets[] = { .unsol_event = alc885_mb5_unsol_event, .init_hook = alc885_mb5_automute, }, + [ALC885_MACMINI3] = { + .mixers = { alc885_macmini3_mixer, alc882_chmode_mixer }, + .init_verbs = { alc885_macmini3_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_macmini3_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_macmini3_6ch_modes), + .input_mux = &macmini3_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc885_macmini3_unsol_event, + .init_hook = alc885_macmini3_automute, + }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, .init_verbs = { alc882_macpro_init_verbs }, -- 1.6.6.1

1 0

[alsa-devel] [PATCH] ASoC: TWL4030: Add supply for audio serial interface control
by Peter Ujfalusi 11 Feb '10

11 Feb '10

The serial interface (TDM/I2S) for the audio block have been constantly enabled. Introduce a new DAPM_SUPPLY for handling the AIF_EN bit, so the interface is only enabled, when there is a need for it. For example when only the analog loopback is enabled, there is no need to keep the serial interface active. I have added the persons who contributed to the Voice path of twl4030 codec driver, so they might have the ability to test this patch, and send an update for the Voice path, if it is necessary Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- This patch has been tested in OPTION 1 mode, where the Voice path is not in use. I have looked for dependencies in the Voice path regarding to the Audio interface enable bit, but I have not found any. If someone could test, if the Voice path also needs the AIF_EN bit set, than please send a patch, which adds this new supply to the DAPM rule for the Voice path. sound/soc/codecs/twl4030.c | 14 +++++++++++++- 1 files changed, 13 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index b32aeb3..277862e 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -55,7 +55,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x0c, /* REG_ATXR1PGA (0xB) */ 0x00, /* REG_AVTXL2PGA (0xC) */ 0x00, /* REG_AVTXR2PGA (0xD) */ - 0x01, /* REG_AUDIO_IF (0xE) */ + 0x00, /* REG_AUDIO_IF (0xE) */ 0x00, /* REG_VOICE_IF (0xF) */ 0x00, /* REG_ARXR1PGA (0x10) */ 0x00, /* REG_ARXL1PGA (0x11) */ @@ -1203,6 +1203,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event, SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("AIF Enable", TWL4030_REG_AUDIO_IF, 0, 0, NULL, 0), + /* Output MIXER controls */ /* Earpiece */ SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, @@ -1337,6 +1339,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"Digital L2 Playback Mixer", NULL, "APLL Enable"}, {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, + {"Digital R1 Playback Mixer", NULL, "AIF Enable"}, + {"Digital L1 Playback Mixer", NULL, "AIF Enable"}, + {"Digital R2 Playback Mixer", NULL, "AIF Enable"}, + {"Digital L2 Playback Mixer", NULL, "AIF Enable"}, + {"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"}, {"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"}, {"Analog L2 Playback Mixer", NULL, "Digital L2 Playback Mixer"}, @@ -1455,6 +1462,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Left2", NULL, "APLL Enable"}, {"ADC Virtual Right2", NULL, "APLL Enable"}, + {"ADC Virtual Left1", NULL, "AIF Enable"}, + {"ADC Virtual Right1", NULL, "AIF Enable"}, + {"ADC Virtual Left2", NULL, "AIF Enable"}, + {"ADC Virtual Right2", NULL, "AIF Enable"}, + /* Analog bypass routes */ {"Right1 Analog Loopback", "Switch", "Analog Right"}, {"Left1 Analog Loopback", "Switch", "Analog Left"}, -- 1.6.6.1

4 5