September 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH V2] ASoC: Davinci: McASP: add support new McASP IP Variant
by Hebbar, Gururaja 06 Sep '12

06 Sep '12

The OMAP2+ variant of McASP is different from Davinci variant w.r.to some register offset. Changes - Add new MCASP_VERSION_3 to identify new variant. New DT compatible "ti,omap2-mcasp-audio" to identify version 3 controller. - The register offsets are handled depending on the version. Note: DMA parameters (dma fifo offset) are not updated and will be done later. Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- Changes in V2: Use "switch" instead of "if" statement to allow future revisions :100644 100644 e6148ec... 374e145... M Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt :100644 100644 79c26aa... d0c5825... M include/linux/platform_data/davinci_asp.h :100644 100644 c3eae1d... 714e51e... M sound/soc/davinci/davinci-mcasp.c .../bindings/sound/davinci-mcasp-audio.txt | 1 + include/linux/platform_data/davinci_asp.h | 1 + sound/soc/davinci/davinci-mcasp.c | 86 +++++++++++++++++--- 3 files changed, 75 insertions(+), 13 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index e6148ec..374e145 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -4,6 +4,7 @@ Required properties: - compatible : "ti,dm646x-mcasp-audio" : for DM646x platforms "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms + "ti,omap2-mcasp-audio" : for OMAP2 platforms (TI81xx, AM33xx) - reg : Should contain McASP registers offset and length - interrupts : Interrupt number for McASP diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index 79c26aa..d0c5825 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -87,6 +87,7 @@ struct snd_platform_data { enum { MCASP_VERSION_1 = 0, /* DM646x */ MCASP_VERSION_2, /* DA8xx/OMAPL1x */ + MCASP_VERSION_3, /* TI81xx/AM33xx */ }; enum mcbsp_clk_input_pin { diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c3eae1d..714e51e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -111,6 +111,10 @@ #define DAVINCI_MCASP_WFIFOSTS (0x1014) #define DAVINCI_MCASP_RFIFOCTL (0x1018) #define DAVINCI_MCASP_RFIFOSTS (0x101C) +#define MCASP_VER3_WFIFOCTL (0x1000) +#define MCASP_VER3_WFIFOSTS (0x1004) +#define MCASP_VER3_RFIFOCTL (0x1008) +#define MCASP_VER3_RFIFOSTS (0x100C) /* * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management @@ -384,18 +388,36 @@ static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (dev->txnumevt) { /* enable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + mcasp_set_bits(dev->base + MCASP_VER3_WFIFOCTL, FIFO_ENABLE); + break; + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + mcasp_set_bits(dev->base + + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } } mcasp_start_tx(dev); } else { if (dev->rxnumevt) { /* enable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + mcasp_set_bits(dev->base + MCASP_VER3_RFIFOCTL, FIFO_ENABLE); + break; + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + mcasp_set_bits(dev->base + + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } } mcasp_start_rx(dev); } @@ -416,14 +438,31 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt) /* disable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + if (dev->txnumevt) { /* disable FIFO */ + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL, FIFO_ENABLE); + break; + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } + } mcasp_stop_tx(dev); } else { - if (dev->rxnumevt) /* disable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + if (dev->rxnumevt) { /* disable FIFO */ + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL, FIFO_ENABLE); + break; + + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } + } mcasp_stop_rx(dev); } } @@ -622,20 +661,37 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) if (dev->txnumevt * tx_ser > 64) dev->txnumevt = 1; - mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, tx_ser, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, tx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + break; + default: + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + tx_ser, NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + } } if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { if (dev->rxnumevt * rx_ser > 64) dev->rxnumevt = 1; - - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, rx_ser, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL, rx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL, + ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + break; + default: + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + rx_ser, NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + } } } @@ -874,6 +930,10 @@ static const struct of_device_id mcasp_dt_ids[] = { .compatible = "ti,da830-mcasp-audio", .data = (void *)MCASP_VERSION_2, }, + { + .compatible = "ti,omap2-mcasp-audio", + .data = (void *)MCASP_VERSION_3, + }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, mcasp_dt_ids); -- 1.7.1

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Re: [alsa-devel] [ASoC] Fix: Revert 'ASoC: imx-ssi: Remove mono support'.
by Mark Brown 06 Sep '12

06 Sep '12

On Tue, Sep 04, 2012 at 11:09:20AM +0200, javier Martin wrote: > could you pick up the fixed patch sent by Gaëtan? It's mangled, can someone resend please (ideally with an appropriate subject line and so on).

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Re: [alsa-devel] [PATCH] ASoC: mc13783: Remove mono support
by Mark Brown 06 Sep '12

06 Sep '12

On Mon, Sep 03, 2012 at 01:04:13PM -0300, Fabio Estevam wrote: > Playing a mono track on a mc13783 codec results in incorrect playback rate. > > Remove mono support so that a mono track can be played correctly. Applied, thanks.

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[alsa-devel] [PATCH] ASoC: Remove unused 'saved_value' field from snd_soc_dapm_widget struct
by Lars-Peter Clausen 06 Sep '12

06 Sep '12

The only user was removed over two years ago in commit a6c65736 ("ASoC: Remove current PGA control handling"). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dapm.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index abe373d..0a15537 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -510,7 +510,6 @@ struct snd_soc_dapm_widget { /* dapm control */ int reg; /* negative reg = no direct dapm */ unsigned char shift; /* bits to shift */ - unsigned int saved_value; /* widget saved value */ unsigned int value; /* widget current value */ unsigned int mask; /* non-shifted mask */ unsigned int on_val; /* on state value */ -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: imx-mc13783: use defines instead of numerical address of register
by Gaëtan Carlier 06 Sep '12

06 Sep '12

This uses already defined name of registers and makes code more readable. Signed-off-by: Gaëtan Carlier <gcembed(a)gmail.com> --- sound/soc/codecs/mc13783.c | 58 +++++++++++++++++++++++-------------------- 1 files changed, 31 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 8f726c0..bbfa553 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -426,16 +426,16 @@ static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai, } static const struct snd_kcontrol_new mc1l_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 7, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 7, 1, 0); static const struct snd_kcontrol_new mc1r_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 5, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 5, 1, 0); static const struct snd_kcontrol_new mc2_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 9, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 9, 1, 0); static const struct snd_kcontrol_new atx_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 11, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 11, 1, 0); /* Virtual mux. The chip does the input selection automatically @@ -461,22 +461,22 @@ static const struct snd_kcontrol_new right_input_mux = SOC_DAPM_ENUM_VIRT("Route", adcr_enum); static const struct snd_kcontrol_new samp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 3, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 3, 1, 0); static const struct snd_kcontrol_new lamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 5, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 5, 1, 0); static const struct snd_kcontrol_new hlamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 10, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 10, 1, 0); static const struct snd_kcontrol_new hramp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 9, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 9, 1, 0); static const struct snd_kcontrol_new llamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 16, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 16, 1, 0); static const struct snd_kcontrol_new lramp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 15, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 15, 1, 0); static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { /* Input */ @@ -487,13 +487,13 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RXINL"), SND_SOC_DAPM_INPUT("TXIN"), - SND_SOC_DAPM_SUPPLY("MC1 Bias", 38, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("MC2 Bias", 38, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC1 Bias", MC13783_AUDIO_TX, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC2 Bias", MC13783_AUDIO_TX, 1, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("MC1L Amp", 38, 7, 0, &mc1l_amp_ctl), - SND_SOC_DAPM_SWITCH("MC1R Amp", 38, 5, 0, &mc1r_amp_ctl), - SND_SOC_DAPM_SWITCH("MC2 Amp", 38, 9, 0, &mc2_amp_ctl), - SND_SOC_DAPM_SWITCH("TXIN Amp", 38, 11, 0, &atx_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1L Amp", MC13783_AUDIO_TX, 7, 0, &mc1l_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1R Amp", MC13783_AUDIO_TX, 5, 0, &mc1r_amp_ctl), + SND_SOC_DAPM_SWITCH("MC2 Amp", MC13783_AUDIO_TX, 9, 0, &mc2_amp_ctl), + SND_SOC_DAPM_SWITCH("TXIN Amp", MC13783_AUDIO_TX, 11, 0, &atx_amp_ctl), SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0, &left_input_mux), @@ -503,12 +503,12 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", 40, 11, 0), - SND_SOC_DAPM_SUPPLY("ADC_Reset", 40, 15, 0, NULL, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", MC13783_AUDIO_CODEC, 11, 0), + SND_SOC_DAPM_SUPPLY("ADC_Reset", MC13783_AUDIO_CODEC, 15, 0, NULL, 0), /* Output */ - SND_SOC_DAPM_SUPPLY("DAC_E", 41, 11, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DAC_Reset", 41, 15, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_E", MC13783_AUDIO_DAC, 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_Reset", MC13783_AUDIO_DAC, 15, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("RXOUTL"), SND_SOC_DAPM_OUTPUT("RXOUTR"), SND_SOC_DAPM_OUTPUT("HSL"), @@ -516,14 +516,18 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LSP"), SND_SOC_DAPM_OUTPUT("SP"), - SND_SOC_DAPM_SWITCH("Speaker Amp", 36, 3, 0, &samp_ctl), + SND_SOC_DAPM_SWITCH("Speaker Amp", MC13783_AUDIO_RX0, 3, 0, &samp_ctl), SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), - SND_SOC_DAPM_SWITCH("Headset Amp Left", 36, 10, 0, &hlamp_ctl), - SND_SOC_DAPM_SWITCH("Headset Amp Right", 36, 9, 0, &hramp_ctl), - SND_SOC_DAPM_SWITCH("Line out Amp Left", 36, 16, 0, &llamp_ctl), - SND_SOC_DAPM_SWITCH("Line out Amp Right", 36, 15, 0, &lramp_ctl), - SND_SOC_DAPM_DAC("DAC", "Playback", 36, 22, 0), - SND_SOC_DAPM_PGA("DAC PGA", 37, 5, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0, + &hlamp_ctl), + SND_SOC_DAPM_SWITCH("Headset Amp Right", MC13783_AUDIO_RX0, 9, 0, + &hramp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Left", MC13783_AUDIO_RX0, 16, 0, + &llamp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Right", MC13783_AUDIO_RX0, 15, 0, + &lramp_ctl), + SND_SOC_DAPM_DAC("DAC", "Playback", MC13783_AUDIO_RX0, 22, 0), + SND_SOC_DAPM_PGA("DAC PGA", MC13783_AUDIO_RX1, 5, 0, NULL, 0), }; static struct snd_soc_dapm_route mc13783_routes[] = { -- 1.7.7.4

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[alsa-devel] [PATCH] ASoC: omap-mcbsp: Fix compilation error due to leftover code
by Peter Ujfalusi 06 Sep '12

06 Sep '12

Part of commit (which patches sound/soc/omap/mcbsp.c file): 8fef626 ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code since the tree where it has been applied did not had the earlier patch: d0db84e ASoC: omap-mcbsp: Fix 6pin mux configuration which changed code around omap_mcbsp_6pin_src_mux(). Because of the missing part from 8fef626 the sound/soc/omap/mcbsp.c does not compile in linux-next. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, This patch has been generated against linux-next as of today. The reason for this is that that is the only place where we have the upstream patch (in 3.6-rc3): d0db84e ASoC: omap-mcbsp: Fix 6pin mux configuration and the 8fef626 ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code commit in the same branch. When you applied 8fef626 you mentioned that it did not applied cleanly. The reason was that the branch did not had commit d0db84e applied prior to the series: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-August/054668.html Now I can not find the d0db84e commit in the asoc tree (not in for-3.6, for-3.7 and not in for-next branch) anymore. I don't really know what has happened, but as of now this patch is needed to fix the OMAP audio in linux-next. I have also prepared a branch with the OMAP patches [1]: topic/omap rebased on top of today's mainline and fixed up the 8fef626 to it's original content. But I think it is already late to revert the OMAP McBSP patches from for-next and from for-3.7 branches... [1] git://gitorious.org/omap-audio/linux-audio.git peter/topic/for-mark/omap-3.7 CC-ing linux-next as well. Regards, Peter sound/soc/omap/mcbsp.c | 31 ------------------------------- 1 file changed, 31 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 935ccf6..bc06175 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -762,37 +762,6 @@ int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) } -int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) -{ - const char *signal, *src; - - if (!mcbsp->pdata->mux_signal) - return -EINVAL; - - switch (mux) { - case CLKR_SRC_CLKR: - signal = "clkr"; - src = "clkr"; - break; - case CLKR_SRC_CLKX: - signal = "clkr"; - src = "clkx"; - break; - case FSR_SRC_FSR: - signal = "fsr"; - src = "fsr"; - break; - case FSR_SRC_FSX: - signal = "fsr"; - src = "fsx"; - break; - default: - return -EINVAL; - } - - return mcbsp->pdata->mux_signal(mcbsp->dev, signal, src); -} - #define max_thres(m) (mcbsp->pdata->buffer_size) #define valid_threshold(m, val) ((val) <= max_thres(m)) #define THRESHOLD_PROP_BUILDER(prop) \ -- 1.7.12

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[alsa-devel] [PATCH] ASoC: arizona: Fix typo in 44.1kHz rates
by Mark Brown 06 Sep '12

06 Sep '12

From: Heather Lomond <hlomond(a)opensource.wolfsonmicro.com> Signed-off-by: Heather Lomond <hlomond(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5c9caca..1cf7a32 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -426,7 +426,7 @@ static const int arizona_44k1_bclk_rates[] = { 940800, 1411200, 1881600, - 2882400, + 2822400, 3763200, 5644800, 7526400, -- 1.7.10.4

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[alsa-devel] [asoc:for-3.7 95/95] sound/soc/codecs/wm0010.c:850:7-27: ERROR: Threaded IRQ with no primary handler requested without IRQF_ONESHOT
by Fengguang Wu 06 Sep '12

06 Sep '12

[add more CC on request of Dimitris] Hi Dimitris, Mark, FYI, there are new coccinelle warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.7 head: e3523e01869da20fdd12ffd19ae1df7bf492650e commit: e3523e01869da20fdd12ffd19ae1df7bf492650e [95/95] ASoC: wm0010: Add initial wm0010 DSP driver All coccinelle warnings: + sound/soc/codecs/wm0010.c:850:7-27: ERROR: Threaded IRQ with no primary handler requested without IRQF_ONESHOT -- + sound/soc/codecs/wm0010.c:660:1-7: preceding lock on line 359 vim +850 sound/soc/codecs/wm0010.c 847 trigger = IRQF_TRIGGER_FALLING; 848 trigger |= IRQF_ONESHOT; 849 > 850 ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger, 851 "wm0010", wm0010); 852 if (ret) 853 dev_err(wm0010->dev, "Failed to request IRQ %d: %d\n", Please consider folding the attached diff :-) --- 0-DAY kernel build testing backend Open Source Technology Centre Fengguang Wu <wfg(a)linux.intel.com> Intel Corporation

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Re: [alsa-devel] [PATCH] ASoC: wm0010: Fix warning, use format %zu for type size_t
by Mark Brown 06 Sep '12

06 Sep '12

On Wed, Sep 05, 2012 at 10:22:24PM +0200, Emil Goode wrote: > Fix warning by using format specifier %zu for type size_t > > Sparse warning: > sound/soc/codecs/wm0010.c:411:2: warning: > format ‘%d’ expects argument of type ‘int’, > but argument 4 has type ‘size_t’ [-Wformat] Applied, thanks.

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[alsa-devel] [PATCH V2] ASoC: tegra: move platform data header
by Stephen Warren 06 Sep '12

06 Sep '12

From: Stephen Warren <swarren(a)nvidia.com> Move the Tegra+WM8903 ASoC platform data header out of arch/arm/mach-tegra, as a pre-requisite of single zImage. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: * Remove stale filename from comment. * Add include guards --- .../sound/tegra_wm8903.h | 7 +++++-- sound/soc/tegra/tegra_wm8903.c | 3 +-- 2 files changed, 6 insertions(+), 4 deletions(-) rename arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h => include/sound/tegra_wm8903.h (89%) diff --git a/arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h b/include/sound/tegra_wm8903.h similarity index 89% rename from arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h rename to include/sound/tegra_wm8903.h index 9d29334..57b202e 100644 --- a/arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h +++ b/include/sound/tegra_wm8903.h @@ -1,6 +1,4 @@ /* - * arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h - * * Copyright 2011 NVIDIA, Inc. * * This software is licensed under the terms of the GNU General Public @@ -14,6 +12,9 @@ * */ +#ifndef __SOUND_TEGRA_WM38903_H +#define __SOUND_TEGRA_WM38903_H + struct tegra_wm8903_platform_data { int gpio_spkr_en; int gpio_hp_det; @@ -21,3 +22,5 @@ struct tegra_wm8903_platform_data { int gpio_int_mic_en; int gpio_ext_mic_en; }; + +#endif diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index d4f14e4..cee13b7 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -34,13 +34,12 @@ #include <linux/gpio.h> #include <linux/of_gpio.h> -#include <mach/tegra_wm8903_pdata.h> - #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/tegra_wm8903.h> #include "../codecs/wm8903.h" -- 1.7.0.4

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