July 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda - Fix error accessing non-existent node for Realtek codecs
by David Henningsson 15 Jul '13

15 Jul '13

A recent commit caused my hda-emu based test tool to complain about accessing nodes that don't exist. This simple test works around this. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) I don't have ALC5505 hardware here, no alsa-info to look at, and I don't know what it is. Therefore it would be good with some confirmation that this patch does not break such hardware, which could be the case if the Realtek codec breaks the HDA spec by having "hidden" nodes. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8bd2261..ca73f10 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3844,7 +3844,8 @@ static int patch_alc269(struct hda_codec *codec) break; } - if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { + if (codec->num_nodes > 0x51 && + snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { spec->has_alc5505_dsp = true; spec->init_hook = alc5505_dsp_init; } -- 1.7.9.5

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[alsa-devel] [PATCH RESEND] ASoC: max98088 - fix element type of the register cache.
by Dylan Reid 15 Jul '13

15 Jul '13

From: Chih-Chung Chang <chihchung(a)chromium.org> The registers of max98088 are 8 bits, not 16 bits. This bug causes the contents of registers to be overwritten with bad values when the codec is suspended and then resumed. Signed-off-by: Chih-Chung Chang <chihchung(a)chromium.org> Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/max98088.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 3eeada5..566a367 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1612,7 +1612,7 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) static void max98088_sync_cache(struct snd_soc_codec *codec) { - u16 *reg_cache = codec->reg_cache; + u8 *reg_cache = codec->reg_cache; int i; if (!codec->cache_sync) -- 1.8.1.3.605.g02339dd

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[alsa-devel] [PATCH] ASoC: sgtl5000: defer the probe if clock is not found
by Shawn Guo 15 Jul '13

15 Jul '13

It's not always the case that clock is already available when sgtl5000 get probed at the first time, e.g. the clock is provided by CPU DAI which may be probed after sgtl5000. So let's defer the probe when devm_clk_get() call fails and give it chance to try later. It fixes the following problem on imx28. [ 1.927637] sgtl5000 0-000a: Failed to get mclock: -2 [ 1.934280] sgtl5000: probe of 0-000a failed with error -2 Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- Mark, This is a fix for 3.11. Shawn sound/soc/codecs/sgtl5000.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d441559..7ebf923 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1527,7 +1527,8 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, if (IS_ERR(sgtl5000->mclk)) { ret = PTR_ERR(sgtl5000->mclk); dev_err(&client->dev, "Failed to get mclock: %d\n", ret); - return ret; + /* Defer the probe to see if the clk will be provided later */ + return ret == -ENOENT ? -EPROBE_DEFER : ret; } ret = clk_prepare_enable(sgtl5000->mclk); -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: omap: Enable COMPILE_TEST build for DT platforms
by Mark Brown 15 Jul '13

15 Jul '13

From: Mark Brown <broonie(a)linaro.org> The DT platforms don't have any source dependency on any OMAP stuff so allow them to be built when COMPILE_TEST is enabled. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/omap/Kconfig | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 9f5d55e..68dfc0a 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,6 +1,6 @@ config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" - depends on ARCH_OMAP && DMA_OMAP + depends on (ARCH_OMAP && DMA_OMAP) || COMPILE_TEST select SND_SOC_DMAENGINE_PCM config SND_OMAP_SOC_DMIC @@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_RX51 tristate "SoC Audio support for Nokia RX-51" - depends on SND_OMAP_SOC && MACH_NOKIA_RX51 + depends on SND_OMAP_SOC && (MACH_NOKIA_RX51 || COMPILE_TEST) select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 @@ -87,7 +87,7 @@ config SND_OMAP_SOC_OMAP_TWL4030 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && SND_OMAP_SOC && ARCH_OMAP4 + depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST) select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 -- 1.8.3.2

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Re: [alsa-devel] [PATCH 1/1] ASoC: max98088 - fix element type of the register cache.
by Dylan Reid 15 Jul '13

15 Jul '13

+Maintainers and Maxim, Chih-Chung, please CC maintainers on patches. On Tue, Jun 25, 2013 at 12:47 AM, <me(a)google.com> wrote: > From: Chih-Chung Chang <chihchung(a)chromium.org> > > The registers of max98088 are 8 bits, not 16 bits. This bug causes the > contents of registers to be overwritten with bad values when the codec > is suspended and then resumed. > > Signed-off-by: Chih-Chung Chang <chihchung(a)chromium.org> > > diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c > index 3eeada5..566a367 100644 > --- a/sound/soc/codecs/max98088.c > +++ b/sound/soc/codecs/max98088.c > @@ -1612,7 +1612,7 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) > > static void max98088_sync_cache(struct snd_soc_codec *codec) > { > - u16 *reg_cache = codec->reg_cache; > + u8 *reg_cache = codec->reg_cache; > int i; > > if (!codec->cache_sync) > -- > 1.8.3 >

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[alsa-devel] [PATCH] ASoC: mxs: Enable COMPILE_TEST builds
by Mark Brown 15 Jul '13

15 Jul '13

From: Mark Brown <broonie(a)linaro.org> Since DT based boards don't have any dependency on arch/arm enable them if COMPILE_TEST is enabled. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/mxs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index 78d321c..7daf860 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -1,6 +1,6 @@ menuconfig SND_MXS_SOC tristate "SoC Audio for Freescale MXS CPUs" - depends on ARCH_MXS + depends on ARCH_MXS || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to -- 1.8.3.2

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[alsa-devel] [PATCH 1/3] ASoC: kirkwood: Remove unused headers
by Mark Brown 15 Jul '13

15 Jul '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/kirkwood/kirkwood-openrd.c | 2 -- sound/soc/kirkwood/kirkwood-t5325.c | 2 -- 2 files changed, 4 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index b979c71..addbebc 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -16,9 +16,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <sound/soc.h> -#include <mach/kirkwood.h> #include <linux/platform_data/asoc-kirkwood.h> -#include <asm/mach-types.h> #include "../codecs/cs42l51.h" static int openrd_client_hw_params(struct snd_pcm_substream *substream, diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 1d0ed6f..4f4cb56 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -15,9 +15,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <sound/soc.h> -#include <mach/kirkwood.h> #include <linux/platform_data/asoc-kirkwood.h> -#include <asm/mach-types.h> #include "../codecs/alc5623.h" static int t5325_hw_params(struct snd_pcm_substream *substream, -- 1.8.3.2

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[alsa-devel] [PATCH v5 0/7] Sound support for at91sam9x5-wm8731 based boards
by Richard Genoud 15 Jul '13

15 Jul '13

Hi, This patchset add sound on the at91sam9x5ek board. It's based on Nicolas Ferre's work in the 2.6.39 atmel patch for sam9x5: https://github.com/linux4sam/linux-at91/commit/0aa157c9e71ccccf3abc30fa37eb… It needs Bo Chen's patchset on the generic dmaengine (topic/atmel branch on Mark's sound tree) and the SSC interrupts disabled (cf http://www.spinics.net/lists/arm-kernel/msg256901.html) Outstanding issue: If this machine driver will support more SoCs maybe we should name it differently. It has been tested on at91sam9g35ek, and should work on g15, g25, x25 and x35 also. patches applies on next-20130711 [I let the original signed-off from Nicolas and Uwe in place, I don't know if I should replace them by something like "originaly-signed-off-by" since the code has been changed.] Changes from v4: * Move the rates constraints into the codec * Hard code the DAI format since it cannot be changed anyway (and remove the DT bindings accordingly) * Add some consistency in the functions naming (sam9x5ek/at91sam9x5ek/sam9x5) * Retrieve the SSC id from the SSC controller phandle * Clarify the atmel-ssc.txt documentation * Develop audio-routing documentation * Tidy up the includes in the machine driver Changes from v3: * Remove the use of macros in device tree documentation. Changes from v2: * Change atmel,sam9x5-audio-wm8731 to atmel,sam9x5-wm8731-audio for consistency with sam9g20 machine driver. * Use the snd_pcm_hw_constraint() API as suggested by Mark and Lars-Peter. * Remove the unnecessary snd_soc_dapm_sync() call as Lars-Peter suggested. * Add documentation for machine driver DT binding. * Update atmel-ssc DT binding documentation. * Add GPL license instead of just the "GPL" word (hope that's ok !) Changes from v1-RFC: * drop patches on SSC since Bo Chen already did the work * reorder patches to have the machine driver first * drop code already handled by the ASoC framework * drop unneeded SND_ATMEL_SOC_PDC config selection * use static snd_soc_card and snd_soc_dai_link structures. Best regards, Richard. Nicolas Ferre (1): sound: sam9x5_wm8731: machine driver for at91sam9x5 wm8731 boards Richard Genoud (6): sound: codec: wm8731: add rates constraints Documentation: DT: update atmel SSC with DMA binding ARM: AT91: DTS: sam9x5: add SSC DMA parameters ARM: AT91: DTS: sam9x5ek: add WM8731 codec ARM: AT91: DTS: sam9x5ek: enable SSC ARM: AT91: DTS: sam9x5ek: add sound configuration .../devicetree/bindings/misc/atmel-ssc.txt | 23 ++- .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 41 ++++ arch/arm/boot/dts/at91sam9x5.dtsi | 3 + arch/arm/boot/dts/at91sam9x5ek.dtsi | 24 +++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/sam9x5_wm8731.c | 207 ++++++++++++++++++++ sound/soc/codecs/wm8731.c | 57 +++++- 8 files changed, 363 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt create mode 100644 sound/soc/atmel/sam9x5_wm8731.c -- 1.7.10.4

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[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: Let the codec acquire its clock
by Fabio Estevam 15 Jul '13

15 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> On a mx6qsabrelite board the following error happens on probe: sgtl5000: probe of 0-000a failed with error -5 imx-sgtl5000 sound.13: ASoC: CODEC (null) not registered imx-sgtl5000 sound.13: snd_soc_register_card failed (-517) platform sound.13: Driver imx-sgtl5000 requests probe defer Prior to reading the codec ID we need to turn the SYS_MCLK clock, so let's enable the codec clock inside sgtl5000_i2c_probe(). Also remove the codec clock enable/disable functions from the machine driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Shawn, I haven't had a chance to convert mx28 to pass the codec clock via device tree yet. It looks like it is a bit trickier there, as it uses the saif clock. Given that mx28 audio playback is already broken in linux-next, this series does not make things worse on mx28 side. sound/soc/codecs/sgtl5000.c | 34 ++++++++++++++++++++++++++++++---- sound/soc/fsl/imx-sgtl5000.c | 30 +++++++----------------------- 2 files changed, 37 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c8f2afb..2e0227b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -114,6 +114,7 @@ struct sgtl5000_priv { struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM]; struct ldo_regulator *ldo; struct regmap *regmap; + struct clk *mclk; }; /* @@ -1522,16 +1523,28 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, return ret; } + sgtl5000->mclk = devm_clk_get(&client->dev, NULL); + if (IS_ERR(sgtl5000->mclk)) { + ret = PTR_ERR(sgtl5000->mclk); + dev_err(&client->dev, "Failed to get mclock: %d\n", ret); + return ret; + } + + ret = clk_prepare_enable(sgtl5000->mclk); + if (ret) + return ret; + /* read chip information */ ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); if (ret) - return ret; + goto disable_clk; if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != SGTL5000_PARTID_PART_ID) { dev_err(&client->dev, "Device with ID register %x is not a sgtl5000\n", reg); - return -ENODEV; + ret = -ENODEV; + goto disable_clk; } rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; @@ -1542,17 +1555,30 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, /* Ensure sgtl5000 will start with sane register values */ ret = sgtl5000_fill_defaults(sgtl5000); if (ret) - return ret; + goto disable_clk; ret = snd_soc_register_codec(&client->dev, &sgtl5000_driver, &sgtl5000_dai, 1); + if (ret) + goto disable_clk; + + return 0; + +disable_clk: + clk_disable_unprepare(sgtl5000->mclk); return ret; } static int sgtl5000_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + struct sgtl5000_priv *sgtl5000; + sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), + GFP_KERNEL); + if (!sgtl5000) + return -ENOMEM; + snd_soc_unregister_codec(&client->dev); + clk_disable_unprepare(sgtl5000->mclk); return 0; } diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index a60aaa0..823151b 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -129,20 +129,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) } data->codec_clk = clk_get(&codec_dev->dev, NULL); - if (IS_ERR(data->codec_clk)) { - /* assuming clock enabled by default */ - data->codec_clk = NULL; - ret = of_property_read_u32(codec_np, "clock-frequency", - &data->clk_frequency); - if (ret) { - dev_err(&codec_dev->dev, - "clock-frequency missing or invalid\n"); - goto fail; - } - } else { - data->clk_frequency = clk_get_rate(data->codec_clk); - clk_prepare_enable(data->codec_clk); - } + if (IS_ERR(data->codec_clk)) + goto fail; + + data->clk_frequency = clk_get_rate(data->codec_clk); data->dai.name = "HiFi"; data->dai.stream_name = "HiFi"; @@ -157,10 +147,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto clk_fail; + goto fail; ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) - goto clk_fail; + goto fail; data->card.num_links = 1; data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; @@ -170,7 +160,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = snd_soc_register_card(&data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto clk_fail; + goto fail; } platform_set_drvdata(pdev, data); @@ -179,8 +169,6 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; -clk_fail: - clk_put(data->codec_clk); fail: if (ssi_np) of_node_put(ssi_np); @@ -194,10 +182,6 @@ static int imx_sgtl5000_remove(struct platform_device *pdev) { struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); - if (data->codec_clk) { - clk_disable_unprepare(data->codec_clk); - clk_put(data->codec_clk); - } snd_soc_unregister_card(&data->card); return 0; -- 1.8.1.2

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[alsa-devel] [PATCH] hdspm - remove unneeded semicolon
by Fengguang Wu 15 Jul '13

15 Jul '13

sound/pci/rme9652/hdspm.c:1110:2-3: Unneeded semicolon Generated by: coccinelle/misc/semicolon.cocci Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> --- cocci-output-13040-790f4c-hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1107,7 +1107,7 @@ static int hdspm_rate_multiplier(struct else if (hdspm->control_register & HDSPM_DoubleSpeed) return rate * 2; - }; + } return rate; }

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