December 2010 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: HDA: Add auto-mute for Thinkpad SL410/SL510
by David Henningsson 17 Dec '10

17 Dec '10

BugLink: http://launchpad.net/bugs/580006 SKU turns off auto-mute for these machines, so ignore the SKU. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda/patch_realtek.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3aeda3a..cc2b8d7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14808,7 +14808,7 @@ static int alc269_resume(struct hda_codec *codec) enum { ALC269_FIXUP_SONY_VAIO, ALC269_FIXUP_DELL_M101Z, - ALC269_FIXUP_LENOVO_EDGE14, + ALC269_FIXUP_SKU_IGNORE, }; static const struct alc_fixup alc269_fixups[] = { @@ -14826,7 +14826,7 @@ static const struct alc_fixup alc269_fixups[] = { {} } }, - [ALC269_FIXUP_LENOVO_EDGE14] = { + [ALC269_FIXUP_SKU_IGNORE] = { .sku = ALC_FIXUP_SKU_IGNORE, }, }; @@ -14834,7 +14834,8 @@ static const struct alc_fixup alc269_fixups[] = { static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), - SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_LENOVO_EDGE14), + SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), {} }; -- 1.7.1 --------------020109060006030901040804--

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[alsa-devel] [PATCH] ASoC: soc-dapm: Introduce the new snd_soc_dapm_virt_mux type
by Dimitris Papastamos 17 Dec '10

17 Dec '10

This new type is a virtual version of snd_soc_dapm_mux. It is used when a backing register value is not necessary for deciding which input path to connect. A simple virtual enumeration control e.g. SOC_DAPM_ENUM_VIRT() can be exposed to userspace which will be used to choose which path to connect. The snd_soc_dapm_virt_mux type ensures that during the initial path setup, the first (which is also the default) input path will be connected. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 9 +++++++++ sound/soc/soc-dapm.c | 19 +++++++++++++++++++ 2 files changed, 28 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 2f76a51..57aab55 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -93,6 +93,9 @@ #define SND_SOC_DAPM_MUX(wname, wreg, wshift, winvert, wcontrols) \ { .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, .shift = wshift, \ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1} +#define SND_SOC_DAPM_VIRT_MUX(wname, wreg, wshift, winvert, wcontrols) \ +{ .id = snd_soc_dapm_virt_mux, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1} #define SND_SOC_DAPM_VALUE_MUX(wname, wreg, wshift, winvert, wcontrols) \ { .id = snd_soc_dapm_value_mux, .name = wname, .reg = wreg, \ .shift = wshift, .invert = winvert, .kcontrols = wcontrols, \ @@ -148,6 +151,11 @@ { .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, .shift = wshift, \ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1, \ .event = wevent, .event_flags = wflags} +#define SND_SOC_DAPM_VIRT_MUX_E(wname, wreg, wshift, winvert, wcontrols, \ + wevent, wflags) \ +{ .id = snd_soc_dapm_virt_mux, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1, \ + .event = wevent, .event_flags = wflags} /* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */ #define SOC_PGA_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \ @@ -373,6 +381,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ snd_soc_dapm_output, /* output pin */ snd_soc_dapm_mux, /* selects 1 analog signal from many inputs */ + snd_soc_dapm_virt_mux, /* virtual version of snd_soc_dapm_mux */ snd_soc_dapm_value_mux, /* selects 1 analog signal from many inputs */ snd_soc_dapm_mixer, /* mixes several analog signals together */ snd_soc_dapm_mixer_named_ctl, /* mixer with named controls */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3d310af..035cdf4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -56,6 +56,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_aif_out] = 3, [snd_soc_dapm_mic] = 4, [snd_soc_dapm_mux] = 5, + [snd_soc_dapm_virt_mux] = 5, [snd_soc_dapm_value_mux] = 5, [snd_soc_dapm_dac] = 6, [snd_soc_dapm_mixer] = 7, @@ -81,6 +82,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_mic] = 7, [snd_soc_dapm_micbias] = 8, [snd_soc_dapm_mux] = 9, + [snd_soc_dapm_virt_mux] = 9, [snd_soc_dapm_value_mux] = 9, [snd_soc_dapm_aif_in] = 10, [snd_soc_dapm_aif_out] = 10, @@ -216,6 +218,20 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, } } break; + case snd_soc_dapm_virt_mux: { + struct soc_enum *e = (struct soc_enum *)w->kcontrols[i].private_value; + + p->connect = 0; + /* since a virtual mux has no backing registers to + * decide which path to connect, it will try to match + * with the first enumeration. This is to ensure + * that the default mux choice (the first) will be + * correctly powered up during initialization. + */ + if (!strcmp(p->name, e->texts[0])) + p->connect = 1; + } + break; case snd_soc_dapm_value_mux: { struct soc_enum *e = (struct soc_enum *) w->kcontrols[i].private_value; @@ -1173,6 +1189,7 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, int found = 0; if (widget->id != snd_soc_dapm_mux && + widget->id != snd_soc_dapm_virt_mux && widget->id != snd_soc_dapm_value_mux) return -ENODEV; @@ -1448,6 +1465,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, path->connect = 1; return 0; case snd_soc_dapm_mux: + case snd_soc_dapm_virt_mux: case snd_soc_dapm_value_mux: ret = dapm_connect_mux(dapm, wsource, wsink, path, control, &wsink->kcontrols[0]); @@ -1537,6 +1555,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) dapm_new_mixer(dapm, w); break; case snd_soc_dapm_mux: + case snd_soc_dapm_virt_mux: case snd_soc_dapm_value_mux: w->power_check = dapm_generic_check_power; dapm_new_mux(dapm, w); -- 1.7.3.3

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[alsa-devel] [GIT PULL] ASoC updates for 2.6.38
by Liam Girdwood 17 Dec '10

17 Dec '10

More updates for 2.6.38, mainly wrt TI CODECs. Thanks Liam --- The following changes since commit 649e5fb033bbadbfa78aba5e2711f1faf2affaf0: Merge branch 'for-2.6.37' into for-2.6.38 (2010-11-29 18:33:47 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.38 Axel Lin (1): ASoC: Remove unused aic3x_i2c_init and aic3x_i2c_exit functions Francois Mazard (1): ASoC: twl6040: Fix analog Mic L & R mux controls Jorge Eduardo Candelaria (6): ASoC: sdp4430: Enable FM stereo pins ASoC: twl6040: Add jack support for headset and handset ASoC: sdp4430: Add Jack support ASoC: twl6040: Add headset and handset mux controls ASoC: twl6040: Clear interrupt status at boot time ASoC: twl6040: Increase timeout for power up Margarita Olaya Cabrera (1): ASoC: twl6040: Add ramp up/down volume for HS and HF Misael Lopez Cruz (2): ASoC: twl6040: Enable plug detection interrupts ASoC: twl6040: Use correct offset for LineInAmp Right Olaya, Margarita (8): ASoC: twl6040: Update twl IO macro ASoC: twl6040: Modify the IRQ handler ASoC: twl6040: Restore bias level at resume ASoC: twl6040: Fix PCM error handling ops ASoC: twl6040: Support other sample rates. ASoC: twl6040: Enable automatic power for phoenix 1.1 ASoC: twl6040: Fix TLV dB step values for gains ASoC: twl6040: Set default gains to minimun value Peter Ujfalusi (12): ASoC: tlv320dac33: Do not enable the codec in init_chip ASoC: tlv320dac33: Avoid multiple soft power up ASoC: tpa6130a2: Simplify power state management ASoC: tpa6130a2: Defer SW enable from power enable ASoC: tpa6130a2: Use one event handler for PGA_E ASoC: tpa6130a2: Add stereo DAPM path ASoC: tpa6130a2: Replace DAPM code with direct interface ASoC: tlv320dac33: Rename outpup amplifier widget ASoC: tlv320dac33: Move DAC LR power on to a supply widget ASoC: tlv320dac33: Fix compillation error ASoC: tlv320dac33: Power down digital parts, when not needed ASoC: TWL4030: Fix 24bit support sound/soc/codecs/tlv320aic3x.c | 15 - sound/soc/codecs/tlv320dac33.c | 57 ++- sound/soc/codecs/tpa6130a2.c | 92 +---- sound/soc/codecs/tpa6130a2.h | 1 + sound/soc/codecs/twl4030.c | 5 +- sound/soc/codecs/twl6040.c | 842 ++++++++++++++++++++++++++++++++++------ sound/soc/codecs/twl6040.h | 8 + sound/soc/omap/sdp4430.c | 43 ++- 8 files changed, 844 insertions(+), 219 deletions(-)

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[alsa-devel] [PATCH 2/2] ASoC: sh: fsi: Add over/under run counter
by Kuninori Morimoto 17 Dec '10

17 Dec '10

Current FSI driver had printed under/over run error if status register have its error bit. But runtime print cause the next error because print out is slow. This patch add error counter and print error when sound stop Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 76 +++++++++++++++++++++++++++++++++------------------ 1 files changed, 49 insertions(+), 27 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 9ab8593..2b06402 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -134,6 +134,9 @@ struct fsi_stream { int buff_len; int period_len; int period_num; + + int uerr_num; + int oerr_num; }; struct fsi_priv { @@ -326,17 +329,29 @@ static void fsi_stream_push(struct fsi_priv *fsi, io->buff_offset = 0; io->period_len = period_len; io->period_num = 0; + io->oerr_num = -1; /* ignore 1st err */ + io->uerr_num = -1; /* ignore 1st err */ } static void fsi_stream_pop(struct fsi_priv *fsi, int is_play) { struct fsi_stream *io = fsi_get_stream(fsi, is_play); + struct snd_soc_dai *dai = fsi_get_dai(io->substream); + + + if (io->oerr_num > 0) + dev_err(dai->dev, "over_run = %d\n", io->oerr_num); + + if (io->uerr_num > 0) + dev_err(dai->dev, "under_run = %d\n", io->uerr_num); io->substream = NULL; io->buff_len = 0; io->buff_offset = 0; io->period_len = 0; io->period_num = 0; + io->oerr_num = 0; + io->uerr_num = 0; } static int fsi_get_fifo_data_num(struct fsi_priv *fsi, int is_play) @@ -375,6 +390,27 @@ static int fsi_get_frame_width(struct fsi_priv *fsi, int is_play) return frames_to_bytes(runtime, 1) / io->chan_num; } +static void fsi_count_fifo_err(struct fsi_priv *fsi) +{ + u32 ostatus = fsi_reg_read(fsi, DOFF_ST); + u32 istatus = fsi_reg_read(fsi, DIFF_ST); + + if (ostatus & ERR_OVER) + fsi->playback.oerr_num++; + + if (ostatus & ERR_UNDER) + fsi->playback.uerr_num++; + + if (istatus & ERR_OVER) + fsi->capture.oerr_num++; + + if (istatus & ERR_UNDER) + fsi->capture.uerr_num++; + + fsi_reg_write(fsi, DOFF_ST, 0); + fsi_reg_write(fsi, DIFF_ST, 0); +} + /* * dma function */ @@ -574,7 +610,7 @@ static void fsi_soft_all_reset(struct fsi_master *master) mdelay(10); } -static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int startup, int stream) +static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream) { struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream = NULL; @@ -667,37 +703,20 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int startup, int stream) /* update buff_offset */ io->buff_offset += fsi_num2offset(data_num, ch_width); - /* check fifo status */ - if (!startup) { - struct snd_soc_dai *dai = fsi_get_dai(substream); - u32 status = is_play ? - fsi_reg_read(fsi, DOFF_ST) : - fsi_reg_read(fsi, DIFF_ST); - - if (status & ERR_OVER) - dev_err(dai->dev, "over run\n"); - if (status & ERR_UNDER) - dev_err(dai->dev, "under run\n"); - } - - is_play ? - fsi_reg_write(fsi, DOFF_ST, 0) : - fsi_reg_write(fsi, DIFF_ST, 0); - if (over_period) snd_pcm_period_elapsed(substream); return 0; } -static int fsi_data_pop(struct fsi_priv *fsi, int startup) +static int fsi_data_pop(struct fsi_priv *fsi) { - return fsi_fifo_data_ctrl(fsi, startup, SNDRV_PCM_STREAM_CAPTURE); + return fsi_fifo_data_ctrl(fsi, SNDRV_PCM_STREAM_CAPTURE); } -static int fsi_data_push(struct fsi_priv *fsi, int startup) +static int fsi_data_push(struct fsi_priv *fsi) { - return fsi_fifo_data_ctrl(fsi, startup, SNDRV_PCM_STREAM_PLAYBACK); + return fsi_fifo_data_ctrl(fsi, SNDRV_PCM_STREAM_PLAYBACK); } static irqreturn_t fsi_interrupt(int irq, void *data) @@ -710,13 +729,16 @@ static irqreturn_t fsi_interrupt(int irq, void *data) fsi_master_mask_set(master, SOFT_RST, IR, IR); if (int_st & AB_IO(1, AO_SHIFT)) - fsi_data_push(&master->fsia, 0); + fsi_data_push(&master->fsia); if (int_st & AB_IO(1, BO_SHIFT)) - fsi_data_push(&master->fsib, 0); + fsi_data_push(&master->fsib); if (int_st & AB_IO(1, AI_SHIFT)) - fsi_data_pop(&master->fsia, 0); + fsi_data_pop(&master->fsia); if (int_st & AB_IO(1, BI_SHIFT)) - fsi_data_pop(&master->fsib, 0); + fsi_data_pop(&master->fsib); + + fsi_count_fifo_err(&master->fsia); + fsi_count_fifo_err(&master->fsib); fsi_irq_clear_status(&master->fsia); fsi_irq_clear_status(&master->fsib); @@ -855,7 +877,7 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, fsi_stream_push(fsi, is_play, substream, frames_to_bytes(runtime, runtime->buffer_size), frames_to_bytes(runtime, runtime->period_size)); - ret = is_play ? fsi_data_push(fsi, 1) : fsi_data_pop(fsi, 1); + ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi); fsi_irq_enable(fsi, is_play); break; case SNDRV_PCM_TRIGGER_STOP: -- 1.7.1

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[alsa-devel] [PATCH 1/2] ASoC: sh: fsi: move fsi_irq_enable function to fsi_dai_trigger
by Kuninori Morimoto 17 Dec '10

17 Dec '10

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 4 +--- 1 files changed, 1 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 97c5394..9ab8593 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -684,9 +684,6 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int startup, int stream) fsi_reg_write(fsi, DOFF_ST, 0) : fsi_reg_write(fsi, DIFF_ST, 0); - /* re-enable irq */ - fsi_irq_enable(fsi, is_play); - if (over_period) snd_pcm_period_elapsed(substream); @@ -859,6 +856,7 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, frames_to_bytes(runtime, runtime->buffer_size), frames_to_bytes(runtime, runtime->period_size)); ret = is_play ? fsi_data_push(fsi, 1) : fsi_data_pop(fsi, 1); + fsi_irq_enable(fsi, is_play); break; case SNDRV_PCM_TRIGGER_STOP: fsi_irq_disable(fsi, is_play); -- 1.7.1

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[alsa-devel] [PATCH] ASoC: sh: fsi: remove unnecessary dependent from Kconfig
by Kuninori Morimoto 17 Dec '10

17 Dec '10

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/Kconfig | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 7f0a496..91170ee 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -48,7 +48,7 @@ config SND_SH7760_AC97 config SND_FSI_AK4642 tristate "FSI-AK4642 sound support" - depends on SND_SOC_SH4_FSI && I2C_SH_MOBILE + depends on SND_SOC_SH4_FSI select SND_SOC_AK4642 help This option enables generic sound support for the @@ -56,7 +56,7 @@ config SND_FSI_AK4642 config SND_FSI_DA7210 tristate "FSI-DA7210 sound support" - depends on SND_SOC_SH4_FSI && I2C_SH_MOBILE + depends on SND_SOC_SH4_FSI select SND_SOC_DA7210 help This option enables generic sound support for the -- 1.7.1

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[alsa-devel] [PATCH 1/2] ASoC: Remove unused DAPM_DOUBLE control types
by Mark Brown 17 Dec '10

17 Dec '10

There are no users of these and it's not clear what they would do given the mono flow modelling which DAPM does. If need arises we can add them again. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- May as well do this anyway, regardless of the fate of patch 2/2 but patch 2/2 will remove the implementation of stereo controls so depends on this one. include/sound/soc-dapm.h | 16 ---------------- 1 files changed, 0 insertions(+), 16 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 2f76a51..aef975f 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -227,13 +227,6 @@ .info = snd_soc_info_volsw, \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } -#define SOC_DAPM_DOUBLE(xname, reg, shift_left, shift_right, max, invert, \ - power) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ - .private_value = (reg) | ((shift_left) << 8) | ((shift_right) << 12) |\ - ((max) << 16) | ((invert) << 24) } #define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ @@ -241,15 +234,6 @@ .tlv.p = (tlv_array), \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } -#define SOC_DAPM_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, \ - power, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ - .private_value = (reg) | ((shift_left) << 8) | ((shift_right) << 12) |\ - ((max) << 16) | ((invert) << 24) } #define SOC_DAPM_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ -- 1.7.1

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Re: [alsa-devel] [PATCH] ASoC: Davinci: Replace use of IO_ADDRESS with ioremap()
by Bedia, Vaibhav 17 Dec '10

17 Dec '10

On Thursday, December 16, 2010 6:19 PM, Sergei Shtylyov wrote: > Hello. > > On 16-12-2010 14:41, Bedia, Vaibhav wrote: > >>>> err_free_mem: > >>> Don't you forget to call clk_disable()/clk_put()? > >> Will fix this issue also as part of the patch. > > No, this is different issue -- please submit another patch. > >>> release_mem_region(mem->start, (mem->end - mem->start) + 1); >>> err_release_data: > Seems like multiple issues can be fixed in the driver. Will send patches for those also. Regards, Vaibhav

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[alsa-devel] [PATCH] ALSA: hda - factorize an automute_mic realtek quirk function
by Anisse Astier 16 Dec '10

16 Dec '10

Multiple quirk functions were using the exact same code to verify if the Mic jack was plugged and mute the Mic accordingly Signed-off-by: Anisse Astier <anisse(a)astier.eu> --- Hi, I'm not sure that's the way to go since we loose track of why these functions were written, but sending anyway, it might help see how to have a better quirk management (or best: no quirk at all). Regards, Anisse --- sound/pci/hda/patch_realtek.c | 35 +++++++---------------------------- 1 files changed, 7 insertions(+), 28 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 371f698..d58a181 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3356,7 +3356,7 @@ static struct hda_verb alc880_beep_init_verbs[] = { }; /* auto-toggle front mic */ -static void alc880_uniwill_mic_automute(struct hda_codec *codec) +static void alc88x_simple_mic_automute(struct hda_codec *codec) { unsigned int present; unsigned char bits; @@ -3378,7 +3378,7 @@ static void alc880_uniwill_setup(struct hda_codec *codec) static void alc880_uniwill_init_hook(struct hda_codec *codec) { alc_automute_amp(codec); - alc880_uniwill_mic_automute(codec); + alc88x_simple_mic_automute(codec); } static void alc880_uniwill_unsol_event(struct hda_codec *codec, @@ -3389,7 +3389,7 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec, */ switch (res >> 28) { case ALC880_MIC_EVENT: - alc880_uniwill_mic_automute(codec); + alc88x_simple_mic_automute(codec); break; default: alc_automute_amp_unsol_event(codec, res); @@ -9522,15 +9522,6 @@ static void alc883_lenovo_nb0763_setup(struct hda_codec *codec) #define alc883_targa_init_hook alc882_targa_init_hook #define alc883_targa_unsol_event alc882_targa_unsol_event -static void alc883_clevo_m720_mic_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x18); - snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} - static void alc883_clevo_m720_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -9542,7 +9533,7 @@ static void alc883_clevo_m720_setup(struct hda_codec *codec) static void alc883_clevo_m720_init_hook(struct hda_codec *codec) { alc_automute_amp(codec); - alc883_clevo_m720_mic_automute(codec); + alc88x_simple_mic_automute(codec); } static void alc883_clevo_m720_unsol_event(struct hda_codec *codec, @@ -9550,7 +9541,7 @@ static void alc883_clevo_m720_unsol_event(struct hda_codec *codec, { switch (res >> 26) { case ALC880_MIC_EVENT: - alc883_clevo_m720_mic_automute(codec); + alc88x_simple_mic_automute(codec); break; default: alc_automute_amp_unsol_event(codec, res); @@ -16806,18 +16797,6 @@ static struct hda_verb alc861vd_lenovo_unsol_verbs[] = { {} }; -static void alc861vd_lenovo_mic_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_jack_detect(codec, 0x18); - bits = present ? HDA_AMP_MUTE : 0; - - snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, - HDA_AMP_MUTE, bits); -} - static void alc861vd_lenovo_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -16828,7 +16807,7 @@ static void alc861vd_lenovo_setup(struct hda_codec *codec) static void alc861vd_lenovo_init_hook(struct hda_codec *codec) { alc_automute_amp(codec); - alc861vd_lenovo_mic_automute(codec); + alc88x_simple_mic_automute(codec); } static void alc861vd_lenovo_unsol_event(struct hda_codec *codec, @@ -16836,7 +16815,7 @@ static void alc861vd_lenovo_unsol_event(struct hda_codec *codec, { switch (res >> 26) { case ALC880_MIC_EVENT: - alc861vd_lenovo_mic_automute(codec); + alc88x_simple_mic_automute(codec); break; default: alc_automute_amp_unsol_event(codec, res); -- 1.7.3.2

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[alsa-devel] [PATCH] regulator: Convert WM835x to use get_voltage_sel()
by Mark Brown 16 Dec '10

16 Dec '10

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- drivers/regulator/wm8350-regulator.c | 16 ++++++---------- 1 files changed, 6 insertions(+), 10 deletions(-) diff --git a/drivers/regulator/wm8350-regulator.c b/drivers/regulator/wm8350-regulator.c index 7e45b0d..1bcb22c 100644 --- a/drivers/regulator/wm8350-regulator.c +++ b/drivers/regulator/wm8350-regulator.c @@ -405,11 +405,10 @@ static int wm8350_dcdc_set_voltage(struct regulator_dev *rdev, int min_uV, return 0; } -static int wm8350_dcdc_get_voltage(struct regulator_dev *rdev) +static int wm8350_dcdc_get_voltage_sel(struct regulator_dev *rdev) { struct wm8350 *wm8350 = rdev_get_drvdata(rdev); int volt_reg, dcdc = rdev_get_id(rdev); - u16 val; switch (dcdc) { case WM8350_DCDC_1: @@ -431,8 +430,7 @@ static int wm8350_dcdc_get_voltage(struct regulator_dev *rdev) } /* all DCDCs have same mV bits */ - val = wm8350_reg_read(wm8350, volt_reg) & WM8350_DC1_VSEL_MASK; - return wm8350_dcdc_val_to_mvolts(val) * 1000; + return wm8350_reg_read(wm8350, volt_reg) & WM8350_DC1_VSEL_MASK; } static int wm8350_dcdc_list_voltage(struct regulator_dev *rdev, @@ -807,11 +805,10 @@ static int wm8350_ldo_set_voltage(struct regulator_dev *rdev, int min_uV, return 0; } -static int wm8350_ldo_get_voltage(struct regulator_dev *rdev) +static int wm8350_ldo_get_voltage_sel(struct regulator_dev *rdev) { struct wm8350 *wm8350 = rdev_get_drvdata(rdev); int volt_reg, ldo = rdev_get_id(rdev); - u16 val; switch (ldo) { case WM8350_LDO_1: @@ -831,8 +828,7 @@ static int wm8350_ldo_get_voltage(struct regulator_dev *rdev) } /* all LDOs have same mV bits */ - val = wm8350_reg_read(wm8350, volt_reg) & WM8350_LDO1_VSEL_MASK; - return wm8350_ldo_val_to_mvolts(val) * 1000; + return wm8350_reg_read(wm8350, volt_reg) & WM8350_LDO1_VSEL_MASK; } static int wm8350_ldo_list_voltage(struct regulator_dev *rdev, @@ -1229,7 +1225,7 @@ static int wm8350_ldo_is_enabled(struct regulator_dev *rdev) static struct regulator_ops wm8350_dcdc_ops = { .set_voltage = wm8350_dcdc_set_voltage, - .get_voltage = wm8350_dcdc_get_voltage, + .get_voltage_sel = wm8350_dcdc_get_voltage_sel, .list_voltage = wm8350_dcdc_list_voltage, .enable = wm8350_dcdc_enable, .disable = wm8350_dcdc_disable, @@ -1253,7 +1249,7 @@ static struct regulator_ops wm8350_dcdc2_5_ops = { static struct regulator_ops wm8350_ldo_ops = { .set_voltage = wm8350_ldo_set_voltage, - .get_voltage = wm8350_ldo_get_voltage, + .get_voltage_sel = wm8350_ldo_get_voltage_sel, .list_voltage = wm8350_ldo_list_voltage, .enable = wm8350_ldo_enable, .disable = wm8350_ldo_disable, -- 1.7.1

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