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

[alsa-devel] [PATCH V2] sound/soc/fsl: remove leftover release_mem_region
by Wolfram Sang 14 Oct '13

14 Oct '13

When converting this driver to devm_ioremap_resource, the removal of this now unneeded function has been forgotten. Signed-off-by: Wolfram Sang <wsa(a)the-dreams.de> --- V2: Remove now unused variable. Thanks to Lars-Peter for pointing out! sound/soc/fsl/imx-ssi.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index f58bcd8..0272289 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -615,7 +615,6 @@ failed_pcm_dma: failed_pcm_fiq: snd_soc_unregister_component(&pdev->dev); failed_register: - release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); failed_clk: snd_soc_set_ac97_ops(NULL); @@ -625,7 +624,6 @@ failed_clk: static int imx_ssi_remove(struct platform_device *pdev) { - struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); struct imx_ssi *ssi = platform_get_drvdata(pdev); imx_pcm_dma_exit(pdev); @@ -636,7 +634,6 @@ static int imx_ssi_remove(struct platform_device *pdev) if (ssi->flags & IMX_SSI_USE_AC97) ac97_ssi = NULL; - release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); snd_soc_set_ac97_ops(NULL); -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH v2] sound: soc: fsl: Fix memory leak in imx-audmux.c
by Felipe Pena 14 Oct '13

14 Oct '13

When audmux_clk is used and clk_prepare_enable function succeed, the memory alloc'd to buf variable is leaked Signed-off-by: Felipe Pena <felipensp(a)gmail.com> --- sound/soc/fsl/imx-audmux.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index d3bf71a..ac86993 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -66,13 +66,10 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) { ssize_t ret; - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + char *buf; int port = (int)file->private_data; u32 pdcr, ptcr; - if (!buf) - return -ENOMEM; - if (audmux_clk) { ret = clk_prepare_enable(audmux_clk); if (ret) @@ -85,6 +82,10 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (audmux_clk) clk_disable_unprepare(audmux_clk); + buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", pdcr, ptcr); -- 1.7.10.4

3 4

[alsa-devel] [Trivial PATCH] sound: Remove unnecessary semicolons
by Joe Perches 14 Oct '13

14 Oct '13

These aren't necessary after switch and if blocks. Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/oss/sb_ess.c | 2 +- sound/pci/au88x0/au88x0_pcm.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/codecs/ak4641.c | 2 +- sound/soc/codecs/mc13783.c | 2 +- sound/soc/codecs/tas5086.c | 2 +- sound/soc/codecs/twl6040.c | 4 ++-- sound/soc/fsl/fsl_spdif.c | 4 ++-- sound/soc/tegra/tegra20_i2s.c | 6 +++--- sound/soc/tegra/tegra20_spdif.c | 6 +++--- sound/soc/tegra/tegra30_ahub.c | 4 ++-- sound/soc/tegra/tegra30_i2s.c | 4 ++-- 12 files changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index c0be085..0e7254b 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -1544,7 +1544,7 @@ static int ess_has_rec_mixer (int submodel) return 1; default: return 0; - }; + } }; #ifdef FKS_LOGGING diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index b46dc9b..9fb03b4 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -671,7 +671,7 @@ static int snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) return err; break; #endif - }; + } if (VORTEX_PCM_TYPE(pcm) == VORTEX_PCM_SPDIF) { for (i = 0; i < ARRAY_SIZE(snd_vortex_mixer_spdif); i++) { diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index bb53dea..8697ced 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -777,7 +777,7 @@ static int asoc_ssc_init(struct device *dev) if (ret) { dev_err(dev, "Could not register PCM: %d\n", ret); goto err_unregister_dai; - }; + } return 0; diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 5f9af1f..49cc5f6 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -328,7 +328,7 @@ static int ak4641_i2s_hw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ak4641->playback_fs = rate; ak4641_set_deemph(codec); - }; + } return 0; } diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index ea141e1..1544330 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -382,7 +382,7 @@ static int mc13783_set_tdm_slot_dac(struct snd_soc_dai *dai, break; default: return -EINVAL; - }; + } snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val); diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 6d31d88..e29cdb7 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -429,7 +429,7 @@ static int tas5086_hw_params(struct snd_pcm_substream *substream, default: dev_err(codec->dev, "Invalid bit width\n"); return -EINVAL; - }; + } ret = regmap_write(priv->regmap, TAS5086_SERIAL_DATA_IF, val); if (ret < 0) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 3c79dbb..35059a2 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -246,7 +246,7 @@ static bool twl6040_is_path_unmuted(struct snd_soc_codec *codec, return priv->dl2_unmuted; default: return 1; - }; + } } /* @@ -1100,7 +1100,7 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i break; default: break; - }; + } } static int twl6040_digital_mute(struct snd_soc_dai *dai, int mute) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 3920c3e..c0fea02 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -963,7 +963,7 @@ static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) @@ -982,7 +982,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config fsl_spdif_regmap_config = { diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 52af7f6..364bf6a 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -297,7 +297,7 @@ static bool tegra20_i2s_wr_rd_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra20_i2s_volatile_reg(struct device *dev, unsigned int reg) @@ -310,7 +310,7 @@ static bool tegra20_i2s_volatile_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra20_i2s_precious_reg(struct device *dev, unsigned int reg) @@ -321,7 +321,7 @@ static bool tegra20_i2s_precious_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config tegra20_i2s_regmap_config = { diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 551b3c9..08bc693 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -213,7 +213,7 @@ static bool tegra20_spdif_wr_rd_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra20_spdif_volatile_reg(struct device *dev, unsigned int reg) @@ -234,7 +234,7 @@ static bool tegra20_spdif_volatile_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra20_spdif_precious_reg(struct device *dev, unsigned int reg) @@ -247,7 +247,7 @@ static bool tegra20_spdif_precious_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config tegra20_spdif_regmap_config = { diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d554d46..805b1f3 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -346,7 +346,7 @@ static bool tegra30_ahub_apbif_wr_rd_reg(struct device *dev, unsigned int reg) return true; default: break; - }; + } if (REG_IN_ARRAY(reg, CHANNEL_CTRL) || REG_IN_ARRAY(reg, CHANNEL_CLEAR) || @@ -381,7 +381,7 @@ static bool tegra30_ahub_apbif_volatile_reg(struct device *dev, return true; default: break; - }; + } if (REG_IN_ARRAY(reg, CHANNEL_CLEAR) || REG_IN_ARRAY(reg, CHANNEL_STATUS) || diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 47565fd04..bd71145 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -369,7 +369,7 @@ static bool tegra30_i2s_wr_rd_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra30_i2s_volatile_reg(struct device *dev, unsigned int reg) @@ -382,7 +382,7 @@ static bool tegra30_i2s_volatile_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config tegra30_i2s_regmap_config = {

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix inverted internal mic not indicated on some machines
by David Henningsson 14 Oct '13

14 Oct '13

The create_bind_cap_vol_ctl does not create any control indicating that an inverted dmic is present. Therefore, create multiple capture volumes in this scenario, so we always have some indication that the internal mic is inverted. This happens on the Lenovo Ideapad U310 as well as the Lenovo Yoga 13 (both are based on the CX20590 codec), but the fix is generic and could be needed for other codecs/machines too. Thanks to Szymon Acedański for the pointer and a draft patch. BugLink: https://bugs.launchpad.net/bugs/1239392 BugLink: https://bugs.launchpad.net/bugs/1227491 Reported-by: Szymon Acedański <accek(a)mimuw.edu.pl> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) I think this fix is safe enough to send to Linus and stable, but will leave the final judgement to Takashi. diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ac41e9c..26ad4f0 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3531,7 +3531,7 @@ static int create_capture_mixers(struct hda_codec *codec) if (!multi) err = create_single_cap_vol_ctl(codec, n, vol, sw, inv_dmic); - else if (!multi_cap_vol) + else if (!multi_cap_vol && !inv_dmic) err = create_bind_cap_vol_ctl(codec, n, vol, sw); else err = create_multi_cap_vol_ctl(codec); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] sound/soc/fsl: remove leftover release_mem_region
by Wolfram Sang 13 Oct '13

13 Oct '13

When converting this driver to devm_ioremap_resource, the removal of this now unneeded function has been forgotten. Signed-off-by: Wolfram Sang <wsa(a)the-dreams.de> --- sound/soc/fsl/imx-ssi.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index f58bcd8..b2cc743 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -615,7 +615,6 @@ failed_pcm_dma: failed_pcm_fiq: snd_soc_unregister_component(&pdev->dev); failed_register: - release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); failed_clk: snd_soc_set_ac97_ops(NULL); @@ -636,7 +635,6 @@ static int imx_ssi_remove(struct platform_device *pdev) if (ssi->flags & IMX_SSI_USE_AC97) ac97_ssi = NULL; - release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); snd_soc_set_ac97_ops(NULL); -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH] sound: soc: fsl: Fix memory leak in imx-audmux.c
by Felipe Pena 13 Oct '13

13 Oct '13

When audmux_clk is used and clk_prepare_enable function succeed, the memory alloc'd to buf variable is leaked Signed-off-by: Felipe Pena <felipensp(a)gmail.com> --- sound/soc/fsl/imx-audmux.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index d3bf71a..977759a 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -66,19 +66,20 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) { ssize_t ret; - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + char *buf; int port = (int)file->private_data; u32 pdcr, ptcr; - if (!buf) - return -ENOMEM; - if (audmux_clk) { ret = clk_prepare_enable(audmux_clk); if (ret) return ret; } + buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); -- 1.7.10.4

1 0

[alsa-devel] Recording devices opened mono still sending stereo causing corrupt audio
by Sean M. Pappalardo - D.J. Pegasus 12 Oct '13

12 Oct '13

(Re-sending including alsa-devel and more information.) Hello there. I'm a developer for Mixxx, the cross-platform open-source DJ software which uses PortAudio and we've been trying for a long time to track down the cause of "horrible mic sound" as documented in our bug: https://bugs.launchpad.net/mixxx/+bug/900364 The resulting recorded audio sounds about 50-60% lower pitch with crackles and it only happens with the ALSA API on Linux and only with certain (mostly Intel-based) audio devices. I have just discovered by testing in Audacity (which also uses PA) that the problem occurs when PA (and therefore ALSA) is asked to open just one channel of a sound input device yet the device (or ALSA) seems to stream two anyway. I have been able to reproduce the problem in Audacity (which also uses PA) when selecting the ALSA API and mono recording, but it works correctly when I change it to stereo recording. I'm suspecting there's a miscommunication between PA and ALSA or ALSA and the sound hardware where either the lower level is saying its input(s) can only be opened in stereo and the higher level (or the application above it) ignores that, or the lower level says it's fine opening mono but then misbehaves. How can I find out which, whether it's ALSA, PA, or the sound hardware at fault? Sincerely, Sean M. Pappalardo "D.J. Pegasus" Mixxx Developer - Controller Specialist

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[alsa-devel] [PATCH] ASoC: rcar: add ID check on rsnd_dai_get()
by Kuninori Morimoto 11 Oct '13

11 Oct '13

checking id in rsnd_dai_get() is good idea Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 28c24fc..b234ed6 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -318,6 +318,9 @@ int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai) struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id) { + if ((id < 0) || (id >= rsnd_dai_nr(priv))) + return NULL; + return priv->rdai + id; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rcar: fixup generation checker
by Kuninori Morimoto 11 Oct '13

11 Oct '13

Current rcar is using rsnd_is_gen1/gen2() to checking its IP generation, but it needs data mask. This patch fixes it up. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/rcar_snd.h | 1 + sound/soc/sh/rcar/rsnd.h | 4 ++-- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index fe66533..fb0a312 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -68,6 +68,7 @@ struct rsnd_scu_platform_info { * * A : generation */ +#define RSND_GEN_MASK (0xF << 0) #define RSND_GEN1 (1 << 0) /* fixme */ #define RSND_GEN2 (2 << 0) /* fixme */ diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 3868aaf..3b71c3f 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -216,8 +216,8 @@ int rsnd_gen_path_exit(struct rsnd_priv *priv, void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); -#define rsnd_is_gen1(s) ((s)->info->flags & RSND_GEN1) -#define rsnd_is_gen2(s) ((s)->info->flags & RSND_GEN2) +#define rsnd_is_gen1(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN1) +#define rsnd_is_gen2(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN2) /* * R-Car ADG -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rcar: fixup rsnd_platform_call() return value
by Kuninori Morimoto 11 Oct '13

11 Oct '13

Un-implemented platform callback is not error. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index fc83f0f..28c24fc 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -103,7 +103,7 @@ * rsnd_platform functions */ #define rsnd_platform_call(priv, dai, func, param...) \ - (!(priv->info->func) ? -ENODEV : \ + (!(priv->info->func) ? 0 : \ priv->info->func(param)) /* -- 1.7.9.5

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Re: [alsa-devel] [PATCH v4 0/5] dma: mxs-dma bugfixes and cleanup
by Fabio Estevam 11 Oct '13

11 Oct '13

Hi Markus, [Also including alsa-devel and some other folks involved with mxs audio] On Fri, Oct 11, 2013 at 9:12 AM, Markus Pargmann <mpa(a)pengutronix.de> wrote: > Hi, > > v4 adds another patch to increase sound stability. Patch 3 ("dma: mxs-dma: Fix > channel reset hardware bug") decreases the number of channel stalls, but after > hours of repeated playback the bug still appears. The new patch ("dma: mxs: Use > semaphores for cyclic DMA") replaces reset channel usage with hardware > semaphore counter mechanism. How is the bug you are trying to solve manifest and how to reproduce it? I have been investigating an issue about sound stability on mx28 as well. The usecase I am doing is: while true do arecord -D hw:0,1 -d 10 -f S16_LE -r 44100 -c 2 | aplay sleep 2 done (Record 10 seconds, sleep and keep in this loop forever). The problem I see is that randomly the captured audio becomes extremely noisy and loud (throughout the whole 10 seconds) and then in the subsequent loop it becomes normal. This is very random and it happens at a 1 or 2% failure rate. I tested your v3, but it did not help to fix this particular issue. I also tried experimenting with the SAIF IOMUX settings: index cda19c8..463fa92 100644 --- a/arch/arm/boot/dts/imx28.dtsi +++ b/arch/arm/boot/dts/imx28.dtsi @@ -545,9 +545,9 @@ MX28_PAD_SAIF0_BITCLK__SAIF0_BITCLK MX28_PAD_SAIF0_SDATA0__SAIF0_SDATA0 >; - fsl,drive-strength = <MXS_DRIVE_12mA>; + fsl,drive-strength = <MXS_DRIVE_4mA>; fsl,voltage = <MXS_VOLTAGE_HIGH>; - fsl,pull-up = <MXS_PULL_ENABLE>; + fsl,pull-up = <MXS_PULL_DISABLE>; }; saif0_pins_b: saif0@1 { @@ -557,9 +557,9 @@ MX28_PAD_SAIF0_BITCLK__SAIF0_BITCLK MX28_PAD_SAIF0_SDATA0__SAIF0_SDATA0 >; - fsl,drive-strength = <MXS_DRIVE_12mA>; + fsl,drive-strength = <MXS_DRIVE_4mA>; fsl,voltage = <MXS_VOLTAGE_HIGH>; - fsl,pull-up = <MXS_PULL_ENABLE>; + fsl,pull-up = <MXS_PULL_DISABLE>; }; saif1_pins_a: saif1@0 { @@ -567,9 +567,9 @@ fsl,pinmux-ids = < MX28_PAD_SAIF1_SDATA0__SAIF1_SDATA0 >; - fsl,drive-strength = <MXS_DRIVE_12mA>; + fsl,drive-strength = <MXS_DRIVE_4mA>; fsl,voltage = <MXS_VOLTAGE_HIGH>; - fsl,pull-up = <MXS_PULL_ENABLE>; + fsl,pull-up = <MXS_PULL_DISABLE>; }; and my tests are showing a much better stability. Not able to get this problem after several hours. Regards, Fabio Estevam

3 2

[alsa-devel] [PATCH 1/5] ARM: Kconfig: Move if ARCH_S3C64XX statement to mach-s3c64xx/Kconfig
by Tomasz Figa 11 Oct '13

11 Oct '13

All other platforms have this condition checked inside their own Kconfig files, so for consistency this patch makes it this way for mach-s3c64xx as well. Signed-off-by: Tomasz Figa <tomasz.figa(a)gmail.com> --- arch/arm/Kconfig | 2 -- arch/arm/mach-s3c64xx/Kconfig | 4 ++++ 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/arch/arm/Kconfig b/arch/arm/Kconfig index b766dad..dc51f8a 100644 --- a/arch/arm/Kconfig +++ b/arch/arm/Kconfig @@ -995,9 +995,7 @@ source "arch/arm/mach-sti/Kconfig" source "arch/arm/mach-s3c24xx/Kconfig" -if ARCH_S3C64XX source "arch/arm/mach-s3c64xx/Kconfig" -endif source "arch/arm/mach-s5p64x0/Kconfig" diff --git a/arch/arm/mach-s3c64xx/Kconfig b/arch/arm/mach-s3c64xx/Kconfig index bd14e3a..0e23910 100644 --- a/arch/arm/mach-s3c64xx/Kconfig +++ b/arch/arm/mach-s3c64xx/Kconfig @@ -3,6 +3,8 @@ # # Licensed under GPLv2 +if ARCH_S3C64XX + # temporary until we can eliminate all drivers using it. config PLAT_S3C64XX bool @@ -322,3 +324,5 @@ config MACH_S3C64XX_DT board. Note: This is under development and not all peripherals can be supported with this machine file. + +endif -- 1.8.3.2

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[alsa-devel] [PATCH] pcm: use the same set of uint_XX types everywhere
by John Spencer 11 Oct '13

11 Oct '13

previously, pcm.h prototyped the functions snd_pcm_format_silence* with u_int_XX types (legacy BSD types from sys/types.h), while the implementation in pcm_misc.c used the C99 stdint.h types uint_XX. this caused compilation errors on systems where the typedef for u_int_XX and uint_XX is not exactly the same. we fix it by using the C99 stdint.h types everywhere. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- include/pcm.h | 9 +++++---- src/pcm/pcm_misc.c | 42 +++++++++++++++++++++--------------------- 2 files changed, 26 insertions(+), 25 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 95b8aed..cd847e4 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -33,6 +33,7 @@ extern "C" { #endif +#include <stdint.h> /** * \defgroup PCM PCM Interface * See the \ref pcm page for more details. @@ -1048,10 +1049,10 @@ int snd_pcm_format_width(snd_pcm_format_t format); /* in bits */ int snd_pcm_format_physical_width(snd_pcm_format_t format); /* in bits */ snd_pcm_format_t snd_pcm_build_linear_format(int width, int pwidth, int unsignd, int big_endian); ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples); -u_int8_t snd_pcm_format_silence(snd_pcm_format_t format); -u_int16_t snd_pcm_format_silence_16(snd_pcm_format_t format); -u_int32_t snd_pcm_format_silence_32(snd_pcm_format_t format); -u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format); +uint8_t snd_pcm_format_silence(snd_pcm_format_t format); +uint16_t snd_pcm_format_silence_16(snd_pcm_format_t format); +uint32_t snd_pcm_format_silence_32(snd_pcm_format_t format); +uint64_t snd_pcm_format_silence_64(snd_pcm_format_t format); int snd_pcm_format_set_silence(snd_pcm_format_t format, void *buf, unsigned int samples); snd_pcm_sframes_t snd_pcm_bytes_to_frames(snd_pcm_t *pcm, ssize_t bytes); diff --git a/src/pcm/pcm_misc.c b/src/pcm/pcm_misc.c index d52160c..2a8be21 100644 --- a/src/pcm/pcm_misc.c +++ b/src/pcm/pcm_misc.c @@ -363,7 +363,7 @@ ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples) * \param format Sample format * \return silence 64 bit word */ -u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) +uint64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { switch (format) { case SNDRV_PCM_FORMAT_S8: @@ -437,7 +437,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { union { float f[2]; - u_int64_t i; + uint64_t i; } u; u.f[0] = u.f[1] = 0.0; #ifdef SNDRV_LITTLE_ENDIAN @@ -450,7 +450,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { union { double f; - u_int64_t i; + uint64_t i; } u; u.f = 0.0; #ifdef SNDRV_LITTLE_ENDIAN @@ -463,7 +463,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { union { float f[2]; - u_int64_t i; + uint64_t i; } u; u.f[0] = u.f[1] = 0.0; #ifdef SNDRV_LITTLE_ENDIAN @@ -476,7 +476,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { union { double f; - u_int64_t i; + uint64_t i; } u; u.f = 0.0; #ifdef SNDRV_LITTLE_ENDIAN @@ -509,10 +509,10 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) * \param format Sample format * \return silence 32 bit word */ -u_int32_t snd_pcm_format_silence_32(snd_pcm_format_t format) +uint32_t snd_pcm_format_silence_32(snd_pcm_format_t format) { assert(snd_pcm_format_physical_width(format) <= 32); - return (u_int32_t)snd_pcm_format_silence_64(format); + return (uint32_t)snd_pcm_format_silence_64(format); } /** @@ -520,10 +520,10 @@ u_int32_t snd_pcm_format_silence_32(snd_pcm_format_t format) * \param format Sample format * \return silence 16 bit word */ -u_int16_t snd_pcm_format_silence_16(snd_pcm_format_t format) +uint16_t snd_pcm_format_silence_16(snd_pcm_format_t format) { assert(snd_pcm_format_physical_width(format) <= 16); - return (u_int16_t)snd_pcm_format_silence_64(format); + return (uint16_t)snd_pcm_format_silence_64(format); } /** @@ -531,10 +531,10 @@ u_int16_t snd_pcm_format_silence_16(snd_pcm_format_t format) * \param format Sample format * \return silence 8 bit word */ -u_int8_t snd_pcm_format_silence(snd_pcm_format_t format) +uint8_t snd_pcm_format_silence(snd_pcm_format_t format) { assert(snd_pcm_format_physical_width(format) <= 8); - return (u_int8_t)snd_pcm_format_silence_64(format); + return (uint8_t)snd_pcm_format_silence_64(format); } /** @@ -550,7 +550,7 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int return 0; switch (snd_pcm_format_physical_width(format)) { case 4: { - u_int8_t silence = snd_pcm_format_silence_64(format); + uint8_t silence = snd_pcm_format_silence_64(format); unsigned int samples1; if (samples % 2 != 0) return -EINVAL; @@ -559,13 +559,13 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int break; } case 8: { - u_int8_t silence = snd_pcm_format_silence_64(format); + uint8_t silence = snd_pcm_format_silence_64(format); memset(data, silence, samples); break; } case 16: { - u_int16_t silence = snd_pcm_format_silence_64(format); - u_int16_t *pdata = (u_int16_t *)data; + uint16_t silence = snd_pcm_format_silence_64(format); + uint16_t *pdata = (uint16_t *)data; if (! silence) memset(data, 0, samples * 2); else { @@ -575,8 +575,8 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int break; } case 24: { - u_int32_t silence = snd_pcm_format_silence_64(format); - u_int8_t *pdata = (u_int8_t *)data; + uint32_t silence = snd_pcm_format_silence_64(format); + uint8_t *pdata = (uint8_t *)data; if (! silence) memset(data, 0, samples * 3); else { @@ -595,8 +595,8 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int break; } case 32: { - u_int32_t silence = snd_pcm_format_silence_64(format); - u_int32_t *pdata = (u_int32_t *)data; + uint32_t silence = snd_pcm_format_silence_64(format); + uint32_t *pdata = (uint32_t *)data; if (! silence) memset(data, 0, samples * 4); else { @@ -606,8 +606,8 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int break; } case 64: { - u_int64_t silence = snd_pcm_format_silence_64(format); - u_int64_t *pdata = (u_int64_t *)data; + uint64_t silence = snd_pcm_format_silence_64(format); + uint64_t *pdata = (uint64_t *)data; if (! silence) memset(data, 0, samples * 8); else { -- 1.7.3.4

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[alsa-devel] [PATCH] ALSA: hda - Sony VAIO Pro 13 (haswell) now has a working headset jack
by David Henningsson 11 Oct '13

11 Oct '13

Just got the positive confirmation from a tester: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1227093/comments/28 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 79e6fe7..bf313be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3761,6 +3761,8 @@ static const struct hda_fixup alc269_fixups[] = { { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ { } }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: mc13783: add mixer controls
by Steffen Trumtrar 11 Oct '13

11 Oct '13

Add more controls to the alsa mixer infrastructure. Signed-off-by: Steffen Trumtrar <s.trumtrar(a)pengutronix.de> --- Changes in v2: - Fix some control names - remove controls that belong to DAPM sound/soc/codecs/mc13783.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index eedbf05..2b62737 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -541,8 +541,26 @@ static const struct soc_enum mc13783_enum_3d_mixer = static struct snd_kcontrol_new mc13783_control_list[] = { SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), + SOC_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0), SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_ENUM("3D Control", mc13783_enum_3d_mixer), + + SOC_SINGLE("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 1, 0), + SOC_SINGLE("Earpiece Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0), + SOC_DOUBLE("Headset Amp Switch", MC13783_AUDIO_RX0, 10, 9, 1, 0), + SOC_DOUBLE("Line out Amp Switch", MC13783_AUDIO_RX0, 16, 15, 1, 0), + + SOC_SINGLE("PCM Capture Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), + SOC_SINGLE("Line in Capture Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), + + SOC_SINGLE("CODEC Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0), + SOC_SINGLE("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), + + SOC_SINGLE("Line in Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0), + SOC_SINGLE("Line in Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0), + + SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0), + SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0), }; static int mc13783_probe(struct snd_soc_codec *codec) -- 1.8.4.rc3

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[alsa-devel] [PATCH v2 0/2] Sony vaio pro 13 (haswell)
by David Henningsson 11 Oct '13

11 Oct '13

Changes since v1: - Mic patch now applies to linux-stable branch linux-3.11.y cleanly - The two patches are now not connected to each other. I'll send a patch that connects them if/when we get some confirmation that this actually is a working headset mic pin. David Henningsson (2): ALSA: hda - Fix microphone for Sony VAIO Pro 13 (Haswell model) ALSA: hda - Add a headset mic model for ALC269 and friends Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 24 ++++++++++++++++++++++++ 2 files changed, 25 insertions(+) -- 1.7.9.5

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[alsa-devel] [PATCH] local.h: include sys/types.h to fix issues with pcm.h
by John Spencer 11 Oct '13

11 Oct '13

sys/types.h is required for the u_int_XX types used by pcm.h. since a change in pcm.h is not desired, we add the inclusion to the header that includes pcm.h during build. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- include/local.h | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/include/local.h b/include/local.h index f0db188..9464efa 100644 --- a/include/local.h +++ b/include/local.h @@ -31,6 +31,7 @@ #include <endian.h> #include <stdarg.h> #include <sys/poll.h> +#include <sys/types.h> #include <errno.h> #include <linux/types.h> #include <linux/ioctl.h> -- 1.7.3.4

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Re: [alsa-devel] Fast Track Ultra 8r...
by Felix Homann 11 Oct '13

11 Oct '13

Am 10.10.2013 19:41 schrieb "Petri Järvisalo" <petri.jarvisalo(a)gmail.com>: > > When plugging in the device at other OS, the internal routing for inputs is disabled and software routes straight 1 = 1 to 8 = 8, like this, and there is no effects applied. Those defaults are the worst I can think of for *my* use case. That's why I haven't implemented them. I do fully appreciate that those would be sane defaults for many people. I wouldn't mind anyone implementing them in Alsa. It's just, well, I won't ;-) When I started working on the FTU mixer support I didn't think about the defaults at all. The default we have right now are the ones you get without explicitly specifying any. But in fact, now I completely fell in love with these insane defaults: They give me an instant alert that I haven't restored my personal 'right' settings! That's of extreme value for my purposes as I use the device mainly on live occasions and the 'wrong' settings certainly cannot go unnoticed in the soundcheck. So again, anybody interested is invited to implement new defaults for the FTU. I won' t object any choice, how could I? But I won't do it myself since I'm personally very happy with the current state. Regards, Felix

3 4

[alsa-devel] ALSA: HDA Control and node attachment
by Jason Steigler 10 Oct '13

10 Oct '13

Hello, I was wondering if it's possible from user-space to change which node a control is attached to. In older ALSA (1.0.23), I had used the '3stack-6ch-dig' model to get 6 channel audio for an "Realtek ALC662 rev 1" codec. I had configured the front panel mic (node 0x19) to be an output using hda-verb that could be controlled by the Center and LFE mixer controls. The board also has jacks on the back and one of those would match the audio coming out the front. When I change the volume or mute the controls, they both were affected (which I expected/wanted). With newer ALSA (1.0.24 with debian Wheezy's stock kernel), the model '3stack-6ch-dig' doesn't exist, but the codec can be configured to 6 channels. I have made the same adjustments with hda-verb to make the node an output. The volume control works for the 'front' and 'rear' jacks, but only muting works on the 'rear' jack. I noticed that the control nodes for Center and LFE are not the same between the 2 setups. In the older version, they were tied to node 0x0e (Audio Mixer), and with the newer version node 0x18 (which looks like the 'Rear Mic' jack). I am not knowledgeable of the details of the HDA codec (yet), and am mostly curious if they can be changed in user-space or if I'll have to start learning the HDA codec code and try to find a way to move the controls around. I tried using hda-jack-retask and could assign the mic nodes as outputs, but wasn't sure if it would allow me to move controls on the HDA graph. Thanks in advance! Jason

1 0

[alsa-devel] Fast Track Ultra 8r...
by Petri Järvisalo 10 Oct '13

10 Oct '13

Hello, I'm new here and somewhat understand how to script php and python ... and very unskilled with linux... anyway i noticed that the default alsa mixer values for fast track ultra 8r are way off the optimal when the sound card is first time plugged in. What I expected was clear stereo/surround sound, what i got was distorted mono sound with reverb! Far away from optimal settings! So, I stated to seek knowledge on web, and finally stumbled upon a homepage that had some info about the Linux support. What i found out was that the sound card has internal routing system, zero latency and a effect processor.Also i found out that the routing works by controlling the input/output levels at alsa mixer. The problem, by default all alsa mixer values are assigned to 100%, this means the internal routing is routing everything to everything distorting the sound, and making the output mono! The website I found had suggestion and a bash script to correct the internal routing: http://heikki.ketoharju.info/2013/03/linux-and-fasttrackultra/#comment-989 alas, the main article is a good read too :) Well the script did not work for me, so i changed the settings manually to my alsa mixer, which took quite some time, and it works like a charm. Anyway my best wish would be to get a fix for this sound card, so it would "just work" when plugged in. Thanks for any info on how i could get this "patch" to be widely available so others could benefit it too. Best regards, Petri

2 1

[alsa-devel] [PATCH] pcm.h: include sys/types.h, necessary for u_int_XX types
by John Spencer 10 Oct '13

10 Oct '13

fixes compilation on musl libc. --- include/pcm.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 95b8aed..7ea9e84 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -29,6 +29,8 @@ #ifndef __ALSA_PCM_H #define __ALSA_PCM_H +#include <sys/types.h> + #ifdef __cplusplus extern "C" { #endif -- 1.7.3.4

2 4

[alsa-devel] Internal Mic Boost channel is unused
by Nathanael D. Noblet 10 Oct '13

10 Oct '13

Hello, I have an issue I'd like to help completely solve. Unfortunately I only have a tenuous grasp of what is actually going on so some of this could be wrong/include errors. Here's what I know. 1) I have the following hardware (alsa-info attached). 2) The internal microphone requires that the mic boost channel be something other than 0 to function properly. 3) Changing what pulseaudio expects things to be labelled causes the problem to go away. (via the attached patch to /usr/share/pulseaudio/paths/analog-input-mic.conf). From the discussion with David Henningsson (diwic on IRC). It seems that pulse doesn't expect a Mic Boost channel to be used with internal microphones. As such to fix this particular hardware, the driver would need to make the internal mic boost be labelled "Internal Mic Boost" as opposed to Mic Boost, which is then is used for both external and internal mics. I'd love to be able to help any way I can to get these fixed once and for all. Let me know if you need more information or what not. -- Nathanael d. Noblet t 403.875.4613

3 13

[alsa-devel] [PATCH 1/2] ALSA: hda - Add a headset mic model for ALC269 and friends
by David Henningsson 10 Oct '13

10 Oct '13

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 15 +++++++++++++++ 2 files changed, 16 insertions(+) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index f911e36..ba143cd 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -28,6 +28,7 @@ ALC269/270/275/276/28x/29x alc269-dmic Enable ALC269(VA) digital mic workaround alc271-dmic Enable ALC271X digital mic workaround inv-dmic Inverted internal mic workaround + headset-mic Indicates a combined headset (mic+headphone jack) lenovo-dock Enables docking station I/O for some Lenovos dell-headset-multi Headset jack, which can also be used as mic-in dell-headset-dock Headset jack (without mic-in), and also dock I/O diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ed9deb6..8cb3350 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2819,6 +2819,15 @@ static void alc269_fixup_hweq(struct hda_codec *codec, alc_write_coef_idx(codec, 0x1e, coef | 0x80); } +static void alc269_fixup_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; +} + static void alc271_fixup_dmic(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3516,6 +3525,7 @@ enum { ALC271_FIXUP_DMIC, ALC269_FIXUP_PCM_44K, ALC269_FIXUP_STEREO_DMIC, + ALC269_FIXUP_HEADSET_MIC, ALC269_FIXUP_QUANTA_MUTE, ALC269_FIXUP_LIFEBOOK, ALC269_FIXUP_AMIC, @@ -3614,6 +3624,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_stereo_dmic, }, + [ALC269_FIXUP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_headset_mic, + }, [ALC269_FIXUP_QUANTA_MUTE] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_quanta_mute, @@ -3979,6 +3993,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_STEREO_DMIC, .name = "alc269-dmic"}, {.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"}, {.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {.id = ALC269_FIXUP_HEADSET_MIC, .name = "headset-mic"}, {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - Distinguish between the headset mic pin and the TRS mic
by David Henningsson 10 Oct '13

10 Oct '13

On the Acer AO722, pin 0x1a represents the mic part of the TRRS headset jack, and pin 0x1b represents the TRS mic jack. Mark pin 0x1a as headset mic will help us know which one is which. Reported-by: Nathanael D. Noblet <nathanael(a)gnat.ca> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ec68eae..7f30c9f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3231,6 +3231,8 @@ enum { CXT_FIXUP_INC_MIC_BOOST, CXT_FIXUP_HEADPHONE_MIC_PIN, CXT_FIXUP_HEADPHONE_MIC, + CXT_FIXUP_HEADSET_MIC_PIN, + CXT_FIXUP_HEADSET_MIC, CXT_FIXUP_GPIO1, }; @@ -3306,6 +3308,15 @@ static void cxt_fixup_headphone_mic(struct hda_codec *codec, } } +static void cxt_fixup_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; +} + /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { @@ -3376,6 +3387,19 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_headphone_mic, }, + [CXT_FIXUP_HEADSET_MIC_PIN] = { + .type = HDA_FIXUP_PINS, + .chained = true, + .chain_id = CXT_FIXUP_HEADSET_MIC, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x03a1903c }, /* use as headset mic, with jack detect */ + { } + } + }, + [CXT_FIXUP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_headset_mic, + }, [CXT_FIXUP_GPIO1] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -3395,6 +3419,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1), + SND_PCI_QUIRK(0x1025, 0x0598, "Acer AO722", CXT_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), -- 1.7.9.5

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[alsa-devel] [PATCHv2] pcm.h: include sys/types.h, necessary for u_int_XX types
by John Spencer 10 Oct '13

10 Oct '13

fixes compilation on musl libc. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- include/pcm.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 95b8aed..7ea9e84 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -29,6 +29,8 @@ #ifndef __ALSA_PCM_H #define __ALSA_PCM_H +#include <sys/types.h> + #ifdef __cplusplus extern "C" { #endif -- 1.7.3.4

1 0