February 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 24/45] hdspm.h: include stdint.h in userspace
by Mikko Rapeli 11 Mar '15

11 Mar '15

Fixes compilation error: sound/hdspm.h:43:2: error: unknown type name ‘uint32_t’ Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/hdspm.h | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index d956c35..f799828 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -20,6 +20,12 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#ifdef __KERNEL__ +#include <linux/types.h> +#else +#include <stdint.h> +#endif + /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */ #define HDSPM_MAX_CHANNELS 64 -- 2.1.4

2 3

[alsa-devel] [PATCH 0/8] ALSA: control: refactoring core codes
by Takashi Sakamoto 10 Mar '15

10 Mar '15

This patchset is for refactoring core control codes. This is also a preparation for my further work for ALSA control functionality. Takashi Sakamoto (8): ALSA: control: change type from long due to the definition of sizeof operator ALSA: control: obsolete switch statement with const value table ALSA: control: gathering evaluations to access ALSA: control: add a comment about locking values after creating ALSA: control: rename loop index to i ALSA: control: fix over 80 characters lines ALSA: control: rename to appropriate macro name ALSA: control: arrange snd_ctl_new() as a local function sound/core/control.c | 223 ++++++++++++++++++++++++++------------------------- 1 file changed, 113 insertions(+), 110 deletions(-) -- 2.1.0

4 31

[alsa-devel] [baohua:sirf-3.18 2573/2782] sound/soc/sirf/atlas7-iacc.c:168:24: sparse: symbol 'atlas7_iacc_dai_ops' was not declared. Should it be static?
by kbuild test robot 10 Mar '15

10 Mar '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/baohua/linux sirf-3.18 head: d61ca44a1d96f134d35d31522ca8efb19a6e1451 commit: 81aba96393135459ac811ce78fe8f91a8b767b82 [2573/2782] ASoC: sirf: atlas7: Add internal audio codec controller driver reproduce: # apt-get install sparse git checkout 81aba96393135459ac811ce78fe8f91a8b767b82 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/sirf/atlas7-iacc.c:168:24: sparse: symbol 'atlas7_iacc_dai_ops' was not declared. Should it be static? sound/soc/sirf/atlas7-iacc.c:482:27: sparse: incorrect type in initializer (incompatible argument 4 (different address spaces)) sound/soc/sirf/atlas7-iacc.c:482:27: expected int ( *copy )( ... ) sound/soc/sirf/atlas7-iacc.c:482:27: got int ( static [toplevel] *<noident> )( ... ) Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

4 4

[alsa-devel] [PATCH 0/24 v2] ASoC: rsnd: move 2nd DMAC in rsnd driver and subnode dma-name support
by Kuninori Morimoto 10 Mar '15

10 Mar '15

Hi Mark, Arnd, Vinod Cc: Laurent, SH-ML These patches add Audio DMAC peri peri implementation in rsnd driver. I will remove current DMAEngine side Audio DMAC peri peri implement if these are accepted. http://thread.gmane.org/gmane.linux.ports.sh.devel/41063/focus=43829 Arnd, Vinod This series adds subnode DT support for 1st DMAC, and it is based on 1) patch. can you please check it ? Mark Actually, 1) patch has no relationship until 20). I guess this patch-set has merge issue. Can you check 2) - 19) as 1st step ? 1) : export of_dma_request_slave_channel() for 20) 2) - 6) : cleanup / tidyup 7) - 12) : Audio DMAC peri peri implementation 13) - 14) : cleanup. these are no longer needed now 15) - 19) : prepare for 1st DMAC dma-names subnode support 20) : 1st DMAC dma-names subnode subnode 21) - 24) : DT bindings text update Kuninori Morimoto (24): 1) dmaengine: export symbol of of_dma_request_slave_channel() 2) ASoC: rsnd: remove SH-DMA-BASE specific implementation 3) ASoC: rsnd: remove un-needed parameter from rsnd_dma_init() 4) ASoC: rsnd: remove unused rsnd_dma_available() 5) ASoC: rsnd: remove un-needed parameter from rsnd_dma_quit() 6) ASoC: rsnd: tidyup rsnd_dma_to_mod() macro declaration position 7) ASoC: rsnd: enable to get resource by name 8) ASoC: rsnd: add rsnd_gen_get_phy_addr() to get physical address 9) ASoC: rsnd: add dma.c for Audio DMAC / Audio DMAC peri peri 10) ASoC: rsnd: move rsnd_gen_dma_addr() from gen.c to dma.c 11) ASoC: rsnd: enable to care 1st / 2nd DMAC on rsnd_dma_xxx() 12) ASoC: rsnd: add Audio DMAC peri peri support rework 13) ASoC: rsnd: remove rsnd_dma::addr 14) ASoC: rsnd: remove rsnd_dma::dir 15) ASoC: rsnd: use union with rsnd_dmaen / rsnd_dmapp 16) ASoC: rsnd: enable rsnd_ssi_use_busif() for DMA 17) ASoC: rsnd: ssi: add rsnd_ssi_of_node() 18) ASoC: rsnd: src: add rsnd_src_of_node() 19) ASoC: rsnd: dvc: add rsnd_dvc_of_node() 20) ASoC: rsnd: 1st DMAC dma-names cares subnode 21) ASoC: rsnd: add sample code of rcar_sound,src irq 22) ASoC: rsnd: add sample code of missing clocks 23) ASoC: rsnd: add sample code of dma entry 24) ASoC: rsnd: add sample code of reg-names .../devicetree/bindings/sound/renesas,rsnd.txt | 125 +++- drivers/dma/of-dma.c | 1 + include/linux/of_dma.h | 4 +- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/core.c | 235 +------- sound/soc/sh/rcar/dma.c | 603 ++++++++++++++++++++ sound/soc/sh/rcar/dvc.c | 12 + sound/soc/sh/rcar/gen.c | 145 +---- sound/soc/sh/rcar/rsnd.h | 72 ++- sound/soc/sh/rcar/src.c | 23 +- sound/soc/sh/rcar/ssi.c | 28 +- 11 files changed, 838 insertions(+), 412 deletions(-) Best regards --- Kuninori Morimoto _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

4 34

[alsa-devel] [PATCH 1/1] ASoC: Intel: remove conflicts when load/unload multiple firmware images
by han.lu＠intel.com 06 Mar '15

06 Mar '15

From: "Lu, Han" <han.lu(a)intel.com> Details: 1. Unload all modules on fw_list of dsp when suspend, and reload all modules on fw_list when resume. 2. A DSP expects only one scratch, but hsw_parse_fw_image() allocates scratch blocks for each firmware image it parses. Move the allocate function sst_block_alloc_scratch() out of hsw_parse_fw_image() to make sure a scratch be allocated only after all firmware images be parsed. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index c42ffae..402b728 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -207,9 +207,6 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) module = (void *)module + sizeof(*module) + module->mod_size; } - /* allocate scratch mem regions */ - sst_block_alloc_scratch(dsp); - return 0; } diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 394af56..863a9ca 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1732,6 +1732,7 @@ static void sst_hsw_drop_all(struct sst_hsw *hsw) int sst_hsw_dsp_load(struct sst_hsw *hsw) { struct sst_dsp *dsp = hsw->dsp; + struct sst_fw *sst_fw, *t; int ret; dev_dbg(hsw->dev, "loading audio DSP...."); @@ -1748,12 +1749,17 @@ int sst_hsw_dsp_load(struct sst_hsw *hsw) return ret; } - ret = sst_fw_reload(hsw->sst_fw); - if (ret < 0) { - dev_err(hsw->dev, "error: SST FW reload failed\n"); - sst_dsp_dma_put_channel(dsp); - return -ENOMEM; + list_for_each_entry_safe_reverse(sst_fw, t, &dsp->fw_list, list) { + ret = sst_fw_reload(sst_fw); + if (ret < 0) { + dev_err(hsw->dev, "error: SST FW reload failed\n"); + sst_dsp_dma_put_channel(dsp); + return -ENOMEM; + } } + ret = sst_block_alloc_scratch(hsw->dsp); + if (ret < 0) + return -EINVAL; sst_dsp_dma_put_channel(dsp); return 0; @@ -1809,12 +1815,17 @@ int sst_hsw_dsp_runtime_suspend(struct sst_hsw *hsw) int sst_hsw_dsp_runtime_sleep(struct sst_hsw *hsw) { - sst_fw_unload(hsw->sst_fw); - sst_block_free_scratch(hsw->dsp); + struct sst_fw *sst_fw, *t; + struct sst_dsp *dsp = hsw->dsp; + + list_for_each_entry_safe(sst_fw, t, &dsp->fw_list, list) { + sst_fw_unload(sst_fw); + } + sst_block_free_scratch(dsp); hsw->boot_complete = false; - sst_dsp_sleep(hsw->dsp); + sst_dsp_sleep(dsp); return 0; } @@ -1943,6 +1954,11 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) goto fw_err; } + /* allocate scratch mem regions */ + ret = sst_block_alloc_scratch(hsw->dsp); + if (ret < 0) + goto boot_err; + /* wait for DSP boot completion */ sst_dsp_boot(hsw->dsp); ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: remove useless register write clearing CHRGPUMP_POWERUP
by Eric Nelson 06 Mar '15

06 Mar '15

The only way the SGTL5000_VDDC_CHRGPMP_POWERUP bit is set is by execution on a machine with vddio and vdda below 3.1V, and the driver needn't handle the case where the design changes through a soft reboot. Even if that were desired, the register write removed by this patch would get over-written by the cached value. Patch inspired by Russell King's more colorful remarks in this patch: https://github.com/SolidRun/linux-imx6-3.14/commit/dd4bf6a Signed-off-by: Eric Nelson <eric.nelson(a)boundarydevices.com> --- sound/soc/codecs/sgtl5000.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d4d793f..518731a 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -952,13 +952,6 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) /* Enable VDDC charge pump */ ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP; } else if (vddio >= 3100 && vdda >= 3100) { - /* - * if vddio and vddd > 3.1v, - * charge pump should be clean before set ana_pwr - */ - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VDDC_CHRGPMP_POWERUP, 0); - /* VDDC use VDDIO rail */ lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD; lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO << -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: Intel: Add suspend_pre and resume_post for Broadwell snd_soc_card
by Jie Yang 06 Mar '15

06 Mar '15

For broadwell machine, we need do some machine related setting before suspend and after resume, e.g. disable/enable jack detection, here adding snd_soc_card suspend_pre and resume_post for this task. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/broadwell.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index fba2ef5..af5d730 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -225,6 +225,32 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { }, }; +static int broadwell_suspend(struct snd_soc_card *card){ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-INT343A:00")) { + dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n"); + rt286_mic_detect(codec, NULL); + break; + } + } + return 0; +} + +static int broadwell_resume(struct snd_soc_card *card){ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-INT343A:00")) { + dev_dbg(codec->dev, "enabling jack detect for resume.\n"); + rt286_mic_detect(codec, &broadwell_headset); + break; + } + } + return 0; +} + /* broadwell audio machine driver for WPT + RT286S */ static struct snd_soc_card broadwell_rt286 = { .name = "broadwell-rt286", @@ -238,6 +264,8 @@ static struct snd_soc_card broadwell_rt286 = { .dapm_routes = broadwell_rt286_map, .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map), .fully_routed = true, + .suspend_pre = broadwell_suspend, + .resume_post = broadwell_resume, }; static int broadwell_audio_probe(struct platform_device *pdev) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt286: correct the OR to AND
by Jie Yang 06 Mar '15

06 Mar '15

Here it should be AND(&) to check the status. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 16723b1..49c44a7 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -397,7 +397,7 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) if (jack) { /* enable IRQ */ - if (rt286->jack->status | SND_JACK_HEADPHONE) + if (rt286->jack->status & SND_JACK_HEADPHONE) snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO1"); regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x2); /* Send an initial empty report */ -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: omap: fix up SND_OMAP_SOC_OMAP_ABE_TWL6040 dependency
by Arnd Bergmann 06 Mar '15

06 Mar '15

The change to enable OMAP5 support on this platform was a little too eager in adding a 'select' for a particular clock driver that might not be enabled in all configurations, which in turn leads to a build error: warning: (SND_OMAP_SOC_OMAP_ABE_TWL6040) selects COMMON_CLK_PALMAS which has unmet direct dependencies (COMMON_CLK && MFD_PALMAS) drivers/built-in.o: In function `palmas_clks_probe': drivers/clk/clk-palmas.c:228: undefined reference to `palmas_ext_control_req_config' I do not see a strong dependency here, so it's probably better to drop this select and to avoid adding more complexity here. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 5163c1eede8e9 ("ASoC: omap: Kconfig: Support for omap5-uevm analog audio") diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index e7c78b0406b5..e74d8db4619d 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -105,7 +105,6 @@ config SND_OMAP_SOC_OMAP_ABE_TWL6040 select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 select SND_SOC_DMIC - select COMMON_CLK_PALMAS if SOC_OMAP5 help Say Y if you want to add support for SoC audio on OMAP boards using ABE and twl6040 codec. This driver currently supports:

3 6

Re: [alsa-devel] [RESEND PATCH v4] Sound: PXA: Fix settings of PXAxxx DMA
by Petr Cvek 06 Mar '15

06 Mar '15

Fix PXA DMA autoincrementation, so capture will not use playback settings (incrementation in register space instead in target buffer). Regression from: 'commit d65a14587a9be853a887a1407db133df1fb68e29 ("ASoC: pxa: use snd_dmaengine_dai_dma_data")' Signed-off-by: Petr Cvek <petr.cvek(a)tul.cz> Acked-by: Daniel Mack <daniel(a)zonque.org> --- sound/arm/pxa2xx-pcm-lib.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 01f8fdc..f8ca3ca 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -45,7 +45,12 @@ int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, size_t period = params_period_bytes(params); pxa_dma_desc *dma_desc; dma_addr_t dma_buff_phys, next_desc_phys; - u32 dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG; + u32 dcmd = 0; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG; + else + dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC; /* temporary transition hack */ switch (rtd->params->addr_width) { -- 1.7.12.1

3 4