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

Re: [alsa-devel] [PATCH 2/2] USB: Gadget: Add Audio Class 2.0 Driver
by Jassi Brar 22 Aug '11

22 Aug '11

On Mon, Aug 22, 2011 at 7:16 PM, Felipe Balbi <balbi(a)ti.com> wrote: > >> But that would mean having two drivers in one file, because they are >> so different. > > then make only a differnt function driver, f_uac2.c, which gets added as > another interface (or configuration) to audio.c. Yup, by audio.c I meant {audio.c + f_audio.c} and audio2.c {audio2.c + f_audio2.c} >> Otherwise, please have a look at audio.c and audio2.c and suggest what >> exactly you mean. > > audio.c doesn't do anything audio related. It's just adding the > configuration to composite layer. The real specification is implemented > in f_audio.c (should probably be called f_uac1.c, though). Agreed. Reusing audio.c is quite reasonable. So, I'll :- * Merge audio2.c into audio.c * Rename f_audio.c -> f_uac1.c * Add new f_audio2.c as f_uac2.c * Add Kconfig switch to compile UAC_1 (audio.c + f_uac1.c) or UAC_2 (audio.c + f_uac2.c) I'll wait for other comments and revise the patch after a few days. Thanks.

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Re: [alsa-devel] [PATCH 2/2] USB: Gadget: Add Audio Class 2.0 Driver
by Jassi Brar 22 Aug '11

22 Aug '11

On Mon, Aug 22, 2011 at 6:14 PM, Felipe Balbi <balbi(a)ti.com> wrote: >> Functionality wise, audio.c hardcodes the data-path to the physical >> audio-codec onboard >> whereas audio2.c provides user-space with a virtual sound-card to fool around >> with audio-data the way the user wants. The functionality that has >> lion's share of code. >> Also, although the functionality of audio.c can be achieved by audio2.c (but not >> vice-versa) we'll break user-space by changing the behavior of audio.c >> to audio2.c >> >> Besides fundamental differences in functionality, they differ by >> Usb-Audio-Class >> compliance as well. audio.c is UAC_1 compliant, whereas audio2.c is UAC_2 >> complaint. >> One can not replace the other. For ex, MS-Windows doesn't support >> UAC_2 by default >> yet, Linux does reasonably well. And by definition having >> Linux-USB-Gadget support >> UAC_2 also, is only better. So by having both we cover more ground than by >> having just one. > > I understand your point, but then we will have two gadget drivers (and > two function drivers) for the same thing (USB Audio), it's the same > thing with the storage gadgets and we have decided on dropping one of > them recently, so I'm not keen on accepting another gadget driver which > will essentially do the same thing. No dear, not the same thing. As I said, they differ by features as well as USB-Spec... only the 'audio' thing is common. > If audio2.c can achieve audio.c's functionality, then you can start by > re-factoring audio.c so that it come closer to what audio2.c is today. audio.c and audio2.c have almost nothing in common. Please have a look at the code. Making audio.c closer to audio2.c would mean :- * Changing all of the descriptors of audio.c {uac_1 -> uac_2} * Removing all alsa related code and adding virtual alsa sound card. At the end, audio.c would have been changed completely and Linux-usb-gadget lost the UAC_1 compliance. > WRT UAC_1/UAC_2 compliance, you can use a Kconfig entry, or two separate > USB configurations (one with UAC_1 and another UAC_2). So there are ways > to make this all work while still keeping backwards compatibility. IIUC, you mean adding UAC_2 compliance and virtual sound-card i/f to audio.c, while preserving the it's original functionality ? But that would mean having two drivers in one file, because they are so different. Otherwise, please have a look at audio.c and audio2.c and suggest what exactly you mean. thnx

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Re: [alsa-devel] [PATCH] ASoC: soc-core: use GFP_KERNEL flag for kmalloc in snd_soc_cnew
by Liam Girdwood 22 Aug '11

22 Aug '11

On 20/08/11 04:03, Axel Lin wrote: > GFP_ATOMIC is not needed here, use GFP_KERNEL instead. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/soc-core.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c > index ae93aa8..b6d8716 100644 > --- a/sound/soc/soc-core.c > +++ b/sound/soc/soc-core.c > @@ -1913,7 +1913,7 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, > > if (prefix) { > name_len = strlen(long_name) + strlen(prefix) + 2; > - name = kmalloc(name_len, GFP_ATOMIC); > + name = kmalloc(name_len, GFP_KERNEL); > if (!name) > return NULL; > Acked-by: Liam Girdwood <lrg(a)ti.com>

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Re: [alsa-devel] [PATCH] ASoC: Provide more detail on WM8962 thermal shutdown status
by Liam Girdwood 22 Aug '11

22 Aug '11

+ alsa-devel On 21/08/11 18:08, Mark Brown wrote: > Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > --- Acked-by: Liam Girdwood <lrg(a)ti.com> > sound/soc/codecs/wm8962.c | 17 +++++++++++++++-- > 1 files changed, 15 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c > index d2db512..ae7a263 100644 > --- a/sound/soc/codecs/wm8962.c > +++ b/sound/soc/codecs/wm8962.c > @@ -839,7 +839,7 @@ static const struct wm8962_reg_access { > [40] = { 0x00FF, 0x01FF, 0x0000 }, /* R40 - SPKOUTL volume */ > [41] = { 0x00FF, 0x01FF, 0x0000 }, /* R41 - SPKOUTR volume */ > > - [47] = { 0x000F, 0x0000, 0x0000 }, /* R47 - Thermal Shutdown Status */ > + [47] = { 0x000F, 0x0000, 0xFFFF }, /* R47 - Thermal Shutdown Status */ > [48] = { 0x7EC7, 0x7E07, 0xFFFF }, /* R48 - Additional Control (4) */ > [49] = { 0x00D3, 0x00D7, 0xFFFF }, /* R49 - Class D Control 1 */ > [51] = { 0x0047, 0x0047, 0x0000 }, /* R51 - Class D Control 2 */ > @@ -3575,6 +3575,7 @@ static irqreturn_t wm8962_irq(int irq, void *data) > struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); > int mask; > int active; > + int reg; > > mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2_MASK); > > @@ -3595,9 +3596,21 @@ static irqreturn_t wm8962_irq(int irq, void *data) > if (active & WM8962_FIFOS_ERR_EINT) > dev_err(codec->dev, "FIFO error\n"); > > - if (active & WM8962_TEMP_SHUT_EINT) > + if (active & WM8962_TEMP_SHUT_EINT) { > dev_crit(codec->dev, "Thermal shutdown\n"); > > + reg = snd_soc_read(codec, WM8962_THERMAL_SHUTDOWN_STATUS); > + > + if (reg & WM8962_TEMP_ERR_HP) > + dev_crit(codec->dev, "Headphone thermal error\n"); > + if (reg & WM8962_TEMP_WARN_HP) > + dev_crit(codec->dev, "Headphone thermal warning\n"); > + if (reg & WM8962_TEMP_ERR_SPK) > + dev_crit(codec->dev, "Speaker thermal error\n"); > + if (reg & WM8962_TEMP_WARN_SPK) > + dev_crit(codec->dev, "Speaker thermal warning\n"); > + } > + > if (active & (WM8962_MICSCD_EINT | WM8962_MICD_EINT)) { > dev_dbg(codec->dev, "Microphone event detected\n"); >

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Re: [alsa-devel] asoc: no valid backend routes for PCM
by Liam Girdwood 22 Aug '11

22 Aug '11

Please don't drop the list from replies. On 22/08/11 06:38, sarika baby wrote: > Hello, > Thanks for your information, but can you tell that if i have to add the audio route should i have to do it at kernel drivers level or else at ALSA (mydroid level) ? > The routes are all configured in userspace with either UCM, alsamixer, amixer or by calling the alsa-lib C API. Liam > Thanks and Regards > Soujanya > > On Fri, Aug 19, 2011 at 9:50 PM, Liam Girdwood <lrg(a)ti.com <mailto:lrg@ti.com>> wrote: > > On 19/08/11 08:53, Sarika wrote: > > Hello, > > I am not able to hear audio on my headset when it connect it to panda board. > > What mite be the issue? > > This means you have no valid audio route from your pcm device to your headphone. > > Please use the supplied UCM configuration files (with Ubuntu) to set the route or use alsamixer to do this manually. > > http://www.omappedia.org/wiki/Audio_Drive_Arch > > > Also panda does not has DMIC does this make any difference ? > > > > No. > > Liam > > > Regards > > Sarika > > > > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org <mailto:Alsa-devel@alsa-project.org> > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > > -- > SarikA >

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[alsa-devel] [PATCH] ucm: change msleep to usleep
by Lu Guanqun 22 Aug '11

22 Aug '11

In file src/ucm/parser.c: if (strcmp(cmd, "usleep") == 0) { string `usleep' is compared, however, in the comment and example conf file, `msleep' is used, it's better to unify them all. Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- src/ucm/parser.c | 8 ++++---- test/ucm/TestHDA/TestHDA.conf | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 23b67bc..a56f1e6 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -792,7 +792,7 @@ static int parse_modifier_name(snd_use_case_mgr_t *uc_mgr, * EnableSequence [ * cset "name='Master Playback Switch',index=2 0,0" * cset "name='Master Playback Volume',index=2 25,25" - * msleep 50 + * usleep 50 * cset "name='Master Playback Switch',index=2 1,1" * cset "name='Master Playback Volume',index=2 50,50" * ] @@ -800,14 +800,14 @@ static int parse_modifier_name(snd_use_case_mgr_t *uc_mgr, * DisableSequence [ * cset "name='Master Playback Switch',index=2 0,0" * cset "name='Master Playback Volume',index=2 25,25" - * msleep 50 + * usleep 50 * cset "name='Master Playback Switch',index=2 1,1" * cset "name='Master Playback Volume',index=2 50,50" * ] * * # Optional transition verb * TransitionSequence."ToCaseName" [ - * msleep 1 + * usleep 1 * ] * * # Optional TQ and ALSA PCMs @@ -1106,7 +1106,7 @@ static int parse_controls(snd_use_case_mgr_t *uc_mgr, snd_config_t *cfg) * cset "name='Master Mono Playback Volume',index=1 0" * cset "name='PCM Switch',index=2 1,1" * exec "some binary here" - * msleep 50 + * usleep 50 * ........ * ] * diff --git a/test/ucm/TestHDA/TestHDA.conf b/test/ucm/TestHDA/TestHDA.conf index 41dd74c..0a75e21 100644 --- a/test/ucm/TestHDA/TestHDA.conf +++ b/test/ucm/TestHDA/TestHDA.conf @@ -7,7 +7,7 @@ SectionUseCase."Case1" { SectionDefaults [ exec "my prg" - msleep 1 + usleep 1 cdev "hw:0" cset "name='PCM Playback Volume' 50%" ]

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[alsa-devel] [PATCH] ASoC: Clear any outstanding WM8962 FLL lock completions before waiting
by Mark Brown 22 Aug '11

22 Aug '11

Ensure that we don't spuriously trigger early. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index c9e07aa..d2db512 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2356,6 +2356,8 @@ static int sysclk_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (fll) { + try_wait_for_completion(&wm8962->fll_lock); + snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_ENA, WM8962_FLL_ENA); if (wm8962->irq) { @@ -3456,6 +3458,8 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda); snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n); + try_wait_for_completion(&wm8962->fll_lock); + snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK | WM8962_FLL_ENA, fll1); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Ensure we get an impedence reported for WM8958 jack detect
by Mark Brown 22 Aug '11

22 Aug '11

Occasionally we may see an accessory reported before we have a stable impedance for the accessory. If this happens then reread the status in order to ensure that the handler can take the appropriate action for the status change. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/mfd/wm8994/registers.h | 15 +++++++++++++ sound/soc/codecs/wm8994.c | 37 +++++++++++++++++++++++---------- 2 files changed, 41 insertions(+), 11 deletions(-) diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h index d7be463..10c483f 100644 --- a/include/linux/mfd/wm8994/registers.h +++ b/include/linux/mfd/wm8994/registers.h @@ -1967,6 +1967,21 @@ #define WM8958_MICB2_DISCH_WIDTH 1 /* MICB2_DISCH */ /* + * R210 (0xD2) - Mic Detect 3 + */ +#define WM8958_MICD_LVL_MASK 0x07FC /* MICD_LVL - [10:2] */ +#define WM8958_MICD_LVL_SHIFT 2 /* MICD_LVL - [10:2] */ +#define WM8958_MICD_LVL_WIDTH 9 /* MICD_LVL - [10:2] */ +#define WM8958_MICD_VALID 0x0002 /* MICD_VALID */ +#define WM8958_MICD_VALID_MASK 0x0002 /* MICD_VALID */ +#define WM8958_MICD_VALID_SHIFT 1 /* MICD_VALID */ +#define WM8958_MICD_VALID_WIDTH 1 /* MICD_VALID */ +#define WM8958_MICD_STS 0x0001 /* MICD_STS */ +#define WM8958_MICD_STS_MASK 0x0001 /* MICD_STS */ +#define WM8958_MICD_STS_SHIFT 0 /* MICD_STS */ +#define WM8958_MICD_STS_WIDTH 1 /* MICD_STS */ + +/* * R76 (0x4C) - Charge Pump (1) */ #define WM8994_CP_ENA 0x8000 /* CP_ENA */ diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 5e8d66d..a26cee1 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3043,19 +3043,34 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) { struct wm8994_priv *wm8994 = data; struct snd_soc_codec *codec = wm8994->codec; - int reg; + int reg, count; - reg = snd_soc_read(codec, WM8958_MIC_DETECT_3); - if (reg < 0) { - dev_err(codec->dev, "Failed to read mic detect status: %d\n", - reg); - return IRQ_NONE; - } + /* We may occasionally read a detection without an impedence + * range being provided - if that happens loop again. + */ + count = 10; + do { + reg = snd_soc_read(codec, WM8958_MIC_DETECT_3); + if (reg < 0) { + dev_err(codec->dev, + "Failed to read mic detect status: %d\n", + reg); + return IRQ_NONE; + } - if (!(reg & WM8958_MICD_VALID)) { - dev_dbg(codec->dev, "Mic detect data not valid\n"); - goto out; - } + if (!(reg & WM8958_MICD_VALID)) { + dev_dbg(codec->dev, "Mic detect data not valid\n"); + goto out; + } + + if (!(reg & WM8958_MICD_STS) || (reg & WM8958_MICD_LVL_MASK)) + break; + + msleep(1); + } while (count--); + + if (count == 0) + dev_warn(codec->dev, "No impedence range reported for jack\n"); #ifndef CONFIG_SND_SOC_WM8994_MODULE trace_snd_soc_jack_irq(dev_name(codec->dev)); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Report IRQ_NONE when we don't see an interrupt from WM8962
by Mark Brown 22 Aug '11

22 Aug '11

This should never happen with level triggered IRQs. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 0c61c2f..b877fa9 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3577,6 +3577,9 @@ static irqreturn_t wm8962_irq(int irq, void *data) active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); active &= ~mask; + if (!active) + return IRQ_NONE; + /* Acknowledge the interrupts */ snd_soc_write(codec, WM8962_INTERRUPT_STATUS_2, active); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Initial WM8962 DSP2 support
by Mark Brown 22 Aug '11

22 Aug '11

The WM8962 features a DSP providing a number of signal processing features including HD Bass and Virtual Surround Sound (VSS). Enable initial support for this, allowing users to enable and disable the algorithms using the default coefficient sets. Further patches will add support for runtime configuration of the DSP coefficients. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 157 ++++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 156 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8526383..c9e07aa 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -63,6 +63,8 @@ struct wm8962_priv { int fll_fref; int fll_fout; + u16 dsp2_ena; + struct delayed_work mic_work; struct snd_soc_jack *jack; @@ -967,7 +969,7 @@ static const struct wm8962_reg_access { [584] = { 0x002D, 0x002D, 0x0000 }, /* R584 - IRQ Debounce */ [586] = { 0xC000, 0xC000, 0x0000 }, /* R586 - MICINT Source Pol */ [768] = { 0x0001, 0x0001, 0x0000 }, /* R768 - DSP2 Power Management */ - [1037] = { 0x0000, 0x003F, 0x0000 }, /* R1037 - DSP2_ExecControl */ + [1037] = { 0x0000, 0x003F, 0xFFFF }, /* R1037 - DSP2_ExecControl */ [4096] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4096 - Write Sequencer 0 */ [4097] = { 0x00FF, 0x00FF, 0x0000 }, /* R4097 - Write Sequencer 1 */ [4098] = { 0x070F, 0x070F, 0x0000 }, /* R4098 - Write Sequencer 2 */ @@ -1988,6 +1990,122 @@ static const unsigned int classd_tlv[] = { }; static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static int wm8962_dsp2_write_config(struct snd_soc_codec *codec) +{ + return 0; +} + +static int wm8962_dsp2_set_enable(struct snd_soc_codec *codec, u16 val) +{ + u16 adcl = snd_soc_read(codec, WM8962_LEFT_ADC_VOLUME); + u16 adcr = snd_soc_read(codec, WM8962_RIGHT_ADC_VOLUME); + u16 dac = snd_soc_read(codec, WM8962_ADC_DAC_CONTROL_1); + + /* Mute the ADCs and DACs */ + snd_soc_write(codec, WM8962_LEFT_ADC_VOLUME, 0); + snd_soc_write(codec, WM8962_RIGHT_ADC_VOLUME, WM8962_ADC_VU); + snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1, + WM8962_DAC_MUTE, WM8962_DAC_MUTE); + + snd_soc_write(codec, WM8962_SOUNDSTAGE_ENABLES_0, val); + + /* Restore the ADCs and DACs */ + snd_soc_write(codec, WM8962_LEFT_ADC_VOLUME, adcl); + snd_soc_write(codec, WM8962_RIGHT_ADC_VOLUME, adcr); + snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1, + WM8962_DAC_MUTE, dac); + + return 0; +} + +static int wm8962_dsp2_start(struct snd_soc_codec *codec) +{ + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + wm8962_dsp2_write_config(codec); + + snd_soc_write(codec, WM8962_DSP2_EXECCONTROL, WM8962_DSP2_RUNR); + + wm8962_dsp2_set_enable(codec, wm8962->dsp2_ena); + + return 0; +} + +static int wm8962_dsp2_stop(struct snd_soc_codec *codec) +{ + wm8962_dsp2_set_enable(codec, 0); + + snd_soc_write(codec, WM8962_DSP2_EXECCONTROL, WM8962_DSP2_STOP); + + return 0; +} + +#define WM8962_DSP2_ENABLE(xname, xshift) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = wm8962_dsp2_ena_info, \ + .get = wm8962_dsp2_ena_get, .put = wm8962_dsp2_ena_put, \ + .private_value = xshift } + +static int wm8962_dsp2_ena_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + return 0; +} + +static int wm8962_dsp2_ena_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int shift = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = !!(wm8962->dsp2_ena & 1 << shift); + + return 0; +} + +static int wm8962_dsp2_ena_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int shift = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + int old = wm8962->dsp2_ena; + int ret = 0; + int dsp2_running = snd_soc_read(codec, WM8962_DSP2_POWER_MANAGEMENT) & + WM8962_DSP2_ENA; + + mutex_lock(&codec->mutex); + + if (ucontrol->value.integer.value[0]) + wm8962->dsp2_ena |= 1 << shift; + else + wm8962->dsp2_ena &= ~(1 << shift); + + if (wm8962->dsp2_ena == old) + goto out; + + ret = 1; + + if (dsp2_running) { + if (wm8962->dsp2_ena) + wm8962_dsp2_set_enable(codec, wm8962->dsp2_ena); + else + wm8962_dsp2_stop(codec); + } + +out: + mutex_unlock(&codec->mutex); + + return ret; +} + /* The VU bits for the headphones are in a different register to the mute * bits and only take effect on the PGA if it is actually powered. */ @@ -2146,6 +2264,11 @@ SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23, WM8962_EQL_B4_GAIN_SHIFT, 31, 0, eq_tlv), SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23, WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv), + +WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT), +WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT), +WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT), +WM8962_DSP2_ENABLE("HD Bass Switch", WM8962_HDBASS_ENA_SHIFT), }; static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = { @@ -2405,6 +2528,31 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, } } +static int dsp2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (wm8962->dsp2_ena) + wm8962_dsp2_start(codec); + break; + + case SND_SOC_DAPM_PRE_PMD: + if (wm8962->dsp2_ena) + wm8962_dsp2_stop(codec); + break; + + default: + BUG(); + return -EINVAL; + } + + return 0; +} + static const char *st_text[] = { "None", "Right", "Left" }; static const struct soc_enum str_enum = @@ -2527,6 +2675,9 @@ SND_SOC_DAPM_SUPPLY("SYSCLK", WM8962_CLOCKING2, 5, 0, sysclk_event, SND_SOC_DAPM_SUPPLY("Charge Pump", WM8962_CHARGE_PUMP_1, 0, 0, cp_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("TOCLK", WM8962_ADDITIONAL_CONTROL_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY_S("DSP2", 1, WM8962_DSP2_POWER_MANAGEMENT, + WM8962_DSP2_ENA_SHIFT, 0, dsp2_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_MIXER("INPGAL", WM8962_LEFT_INPUT_PGA_CONTROL, 4, 0, inpgal, ARRAY_SIZE(inpgal)), @@ -2622,11 +2773,13 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "ADCL", NULL, "TOCLK" }, { "ADCL", NULL, "MIXINL" }, { "ADCL", NULL, "DMIC" }, + { "ADCL", NULL, "DSP2" }, { "ADCR", NULL, "SYSCLK" }, { "ADCR", NULL, "TOCLK" }, { "ADCR", NULL, "MIXINR" }, { "ADCR", NULL, "DMIC" }, + { "ADCR", NULL, "DSP2" }, { "STL", "Left", "ADCL" }, { "STL", "Right", "ADCR" }, @@ -2638,11 +2791,13 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "DACL", NULL, "TOCLK" }, { "DACL", NULL, "Beep" }, { "DACL", NULL, "STL" }, + { "DACL", NULL, "DSP2" }, { "DACR", NULL, "SYSCLK" }, { "DACR", NULL, "TOCLK" }, { "DACR", NULL, "Beep" }, { "DACR", NULL, "STR" }, + { "DACR", NULL, "DSP2" }, { "HPMIXL", "IN4L Switch", "IN4L" }, { "HPMIXL", "IN4R Switch", "IN4R" }, -- 1.7.5.4

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