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

[alsa-devel] [PATCH 1/2] ALSA: hda/realtek - Remove depop delay for suspend and resume if applicable
by mengdong.lin＠intel.com 29 Nov '13

29 Nov '13

From: Kailang Yang <kailang(a)realtek.com> This patch defines a flag "no_depop_delay" in alc_spec. If this flag is set, delay in alc_eapd_shutup and alc_resume will be skipped. Machine-specific fixup can set this flag to reduce suspend/resume time, if the codec and hardware analog design can avoid pop noise without this delay. Signed-off-by: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6366a66..503cf78 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -118,7 +118,8 @@ struct alc_spec { int init_amp; int codec_variant; /* flag for other variants */ - bool has_alc5505_dsp; + unsigned int has_alc5505_dsp:1; + unsigned int no_depop_delay:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -280,8 +281,11 @@ static void alc_auto_setup_eapd(struct hda_codec *codec, bool on) */ static void alc_eapd_shutup(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; + alc_auto_setup_eapd(codec, false); - msleep(200); + if (!spec->no_depop_delay) + msleep(200); snd_hda_shutup_pins(codec); } @@ -863,7 +867,10 @@ static int alc_suspend(struct hda_codec *codec) #ifdef CONFIG_PM static int alc_resume(struct hda_codec *codec) { - msleep(150); /* to avoid pop noise */ + struct alc_spec *spec = codec->spec; + + if (!spec->no_depop_delay) + msleep(150); /* to avoid pop noise */ codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); @@ -4448,7 +4455,7 @@ static int patch_alc269(struct hda_codec *codec) } if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { - spec->has_alc5505_dsp = true; + spec->has_alc5505_dsp = 1; spec->init_hook = alc5505_dsp_init; } -- 1.8.1.2

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[alsa-devel] [PATCH 0/6] ASoC: rcar: add Gen2 support
by Kuninori Morimoto 29 Nov '13

29 Nov '13

Hi Mark These patches add Gen2 PIO transfer support for Renesas sound driver. DMA transfer will be supported. Kuninori Morimoto (6): ASoC: rcar: remove unused register settings ASoC: rcar: separate regmap init common field ASoC: rcar: add rsnd_is_accessible_reg() ASoC: rcar: remove .path_init/exit from rsnd_gen_ops ASoC: rcar: remove rcar_gen_ops ASoC: rcar: add Gen2 sound support sound/soc/sh/rcar/gen.c | 248 +++++++++++++++++++++++++++++----------------- sound/soc/sh/rcar/rsnd.h | 6 +- sound/soc/sh/rcar/ssi.c | 3 + 3 files changed, 162 insertions(+), 95 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: wm_adsp: Stop region iteration when the desired region is found
by Charles Keepax 29 Nov '13

29 Nov '13

When locating the memory region relating to a coefficient block written through a bin file we keep processing the list of regions even after we have found the region we require. This patch adds a break, so we don't process redundant list items. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 46ec0e9..b42f9af 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1286,6 +1286,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) reg = wm_adsp_region_to_reg(mem, reg); reg += offset; + break; } } -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: cs42l52: Fix build
by Mark Brown 29 Nov '13

29 Nov '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 4a47a634f376..0bac6d5a4ac8 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1232,7 +1232,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, if (of_property_read_u32(i2c_client->dev.of_node, "cirrus,chgfreq-divisor", &val32) >= 0) - pdata->chgfreq_divisor = val32; + pdata->chgfreq = val32; pdata->reset_gpio = of_get_named_gpio(i2c_client->dev.of_node, -- 1.8.4.4

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[alsa-devel] [PATCH V2] ASoC: tegra: use reset framework
by Stephen Warren 29 Nov '13

29 Nov '13

From: Stephen Warren <swarren(a)nvidia.com> Tegra's clock driver now provides an implementation of the common reset API (include/linux/reset.h). Use this instead of the old Tegra- specific API; that will soon be removed. This change also renames "clock"/"clk" to "modules"/"mod" in symbols related to entries in configlink_clocks[], since: - We don't care about clock handles any more, but rather reset handles, so the old name isn't applicable. - It really is a list of modules on the bus, about which we currently only care about reset handles. If we start caring about any other aspect of the modules in the future, we won't have to rename all these symbols again. Note: The addition of "depends COMMON_CLOCK" is something that was missing before, not a new requirement. Cc: treding(a)nvidia.com Cc: pdeschrijver(a)nvidia.com Cc: linux-tegra(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Added note about COMMON_CLOCK to commit description. This patch is part of a series with strong internal depdendencies. I'm looking for an ack so that I can take the entire series through the Tegra and arm-soc trees. The series will be part of a stable branch that can be merged into other subsystems if needed to avoid/resolve dependencies. --- sound/soc/tegra/Kconfig | 2 ++ sound/soc/tegra/tegra30_ahub.c | 70 +++++++++++++++++++++++------------------- sound/soc/tegra/tegra30_ahub.h | 2 +- 3 files changed, 41 insertions(+), 33 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 8fc653ca3ab4..896292bb853f 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -1,6 +1,8 @@ config SND_SOC_TEGRA tristate "SoC Audio for the Tegra System-on-Chip" depends on (ARCH_TEGRA && TEGRA20_APB_DMA) || COMPILE_TEST + depends on COMMON_CLK + depends on RESET_CONTROLLER select REGMAP_MMIO select SND_SOC_GENERIC_DMAENGINE_PCM help diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 31154338c1eb..5ce00dc48c44 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -24,6 +24,7 @@ #include <linux/platform_device.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <linux/slab.h> #include <linux/clk/tegra.h> #include <sound/soc.h> @@ -301,27 +302,27 @@ int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif) } EXPORT_SYMBOL_GPL(tegra30_ahub_unset_rx_cif_source); -#define CLK_LIST_MASK_TEGRA30 BIT(0) -#define CLK_LIST_MASK_TEGRA114 BIT(1) +#define MOD_LIST_MASK_TEGRA30 BIT(0) +#define MOD_LIST_MASK_TEGRA114 BIT(1) -#define CLK_LIST_MASK_TEGRA30_OR_LATER \ - (CLK_LIST_MASK_TEGRA30 | CLK_LIST_MASK_TEGRA114) +#define MOD_LIST_MASK_TEGRA30_OR_LATER \ + (MOD_LIST_MASK_TEGRA30 | MOD_LIST_MASK_TEGRA114) static const struct { - const char *clk_name; - u32 clk_list_mask; -} configlink_clocks[] = { - { "i2s0", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "i2s1", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "i2s2", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "i2s3", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "i2s4", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "dam0", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "dam1", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "dam2", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "spdif_in", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "amx", CLK_LIST_MASK_TEGRA114 }, - { "adx", CLK_LIST_MASK_TEGRA114 }, + const char *rst_name; + u32 mod_list_mask; +} configlink_mods[] = { + { "i2s0", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "i2s1", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "i2s2", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "i2s3", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "i2s4", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "dam0", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "dam1", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "dam2", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "spdif", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "amx", MOD_LIST_MASK_TEGRA114 }, + { "adx", MOD_LIST_MASK_TEGRA114 }, }; #define LAST_REG(name) \ @@ -450,17 +451,17 @@ static const struct regmap_config tegra30_ahub_ahub_regmap_config = { }; static struct tegra30_ahub_soc_data soc_data_tegra30 = { - .clk_list_mask = CLK_LIST_MASK_TEGRA30, + .mod_list_mask = MOD_LIST_MASK_TEGRA30, .set_audio_cif = tegra30_ahub_set_cif, }; static struct tegra30_ahub_soc_data soc_data_tegra114 = { - .clk_list_mask = CLK_LIST_MASK_TEGRA114, + .mod_list_mask = MOD_LIST_MASK_TEGRA114, .set_audio_cif = tegra30_ahub_set_cif, }; static struct tegra30_ahub_soc_data soc_data_tegra124 = { - .clk_list_mask = CLK_LIST_MASK_TEGRA114, + .mod_list_mask = MOD_LIST_MASK_TEGRA114, .set_audio_cif = tegra124_ahub_set_cif, }; @@ -475,7 +476,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) { const struct of_device_id *match; const struct tegra30_ahub_soc_data *soc_data; - struct clk *clk; + struct reset_control *rst; int i; struct resource *res0, *res1, *region; u32 of_dma[2]; @@ -495,19 +496,24 @@ static int tegra30_ahub_probe(struct platform_device *pdev) * operate correctly, all devices on this bus must be out of reset. * Ensure that here. */ - for (i = 0; i < ARRAY_SIZE(configlink_clocks); i++) { - if (!(configlink_clocks[i].clk_list_mask & - soc_data->clk_list_mask)) + for (i = 0; i < ARRAY_SIZE(configlink_mods); i++) { + if (!(configlink_mods[i].mod_list_mask & + soc_data->mod_list_mask)) continue; - clk = clk_get(&pdev->dev, configlink_clocks[i].clk_name); - if (IS_ERR(clk)) { - dev_err(&pdev->dev, "Can't get clock %s\n", - configlink_clocks[i].clk_name); - ret = PTR_ERR(clk); + + rst = reset_control_get(&pdev->dev, + configlink_mods[i].rst_name); + if (IS_ERR(rst)) { + dev_err(&pdev->dev, "Can't get reset %s\n", + configlink_mods[i].rst_name); + ret = PTR_ERR(rst); goto err; } - tegra_periph_reset_deassert(clk); - clk_put(clk); + + ret = reset_control_deassert(rst); + reset_control_put(rst); + if (ret) + goto err; } ahub = devm_kzalloc(&pdev->dev, sizeof(struct tegra30_ahub), diff --git a/sound/soc/tegra/tegra30_ahub.h b/sound/soc/tegra/tegra30_ahub.h index d67321d90faa..1383f8cd3572 100644 --- a/sound/soc/tegra/tegra30_ahub.h +++ b/sound/soc/tegra/tegra30_ahub.h @@ -502,7 +502,7 @@ void tegra124_ahub_set_cif(struct regmap *regmap, unsigned int reg, struct tegra30_ahub_cif_conf *conf); struct tegra30_ahub_soc_data { - u32 clk_list_mask; + u32 mod_list_mask; void (*set_audio_cif)(struct regmap *regmap, unsigned int reg, struct tegra30_ahub_cif_conf *conf); -- 1.8.1.5

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[alsa-devel] [PATCH] ASoC mc13783 codec: Add 'loudspeaker amp' and 'lineout amp'
by Lambrecht Jürgen 29 Nov '13

29 Nov '13

Add Alsp (loudspeaker amp) and Arxout (lineout amp) in the same way as Asp (speaker amp) and Ahs (headphone amp) - Remove "LSPL": all output controls are added now; this one doesn't exist - Fix: "LSP" is loudspeaker amp; they indeed share the same mux according to the datasheet figure but they have separate registers. And testing shows the Asp and Alsp muxes are separated. - Add "CODEC Capture Switch" control and move "CODEC Capture Mixin Switch" to its correct place Signed-off-by: Jürgen Lambrecht <J.Lambrecht(a)televic.com> --- sound/soc/codecs/mc13783.c | 29 ++++++++++++++++++++++++----- 1 file changed, 24 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index af7bc55..08fb022 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -437,6 +437,10 @@ static const SOC_ENUM_SINGLE_DECL(speaker_amp_source, MC13783_AUDIO_RX0, 4, speaker_amp_source_text); static const struct snd_kcontrol_new speaker_amp_source_mux = SOC_DAPM_ENUM("Speaker Amp Source MUX", speaker_amp_source); +static const SOC_ENUM_SINGLE_DECL(loudspeaker_amp_source, MC13783_AUDIO_RX0, 7, + speaker_amp_source_text); +static const struct snd_kcontrol_new loudspeaker_amp_source_mux = + SOC_DAPM_ENUM("Loudspeaker Amp Source MUX", loudspeaker_amp_source); static const char * const headset_amp_source_text[] = { "CODEC", "Mixer" @@ -447,6 +451,11 @@ static const SOC_ENUM_SINGLE_DECL(headset_amp_source, MC13783_AUDIO_RX0, 11, static const struct snd_kcontrol_new headset_amp_source_mux = SOC_DAPM_ENUM("Headset Amp Source MUX", headset_amp_source); +static const SOC_ENUM_SINGLE_DECL(lineout_amp_source, MC13783_AUDIO_RX0, 17, + headset_amp_source_text); +static const struct snd_kcontrol_new lineout_amp_source_mux = + SOC_DAPM_ENUM("Lineout Amp Source MUX", lineout_amp_source); + static const struct snd_kcontrol_new cdcout_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 18, 1, 0); @@ -493,10 +502,15 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_MUX("Speaker Amp Source MUX", SND_SOC_NOPM, 0, 0, &speaker_amp_source_mux), + SND_SOC_DAPM_MUX("Loudspeaker Amp Source MUX", SND_SOC_NOPM, 0, 0, + &loudspeaker_amp_source_mux), SND_SOC_DAPM_MUX("Headset Amp Source MUX", SND_SOC_NOPM, 0, 0, &headset_amp_source_mux), + SND_SOC_DAPM_MUX("Lineout Amp Source MUX", SND_SOC_NOPM, 0, 0, + &lineout_amp_source_mux), + 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), @@ -514,7 +528,6 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("RXOUTR"), SND_SOC_DAPM_OUTPUT("HSL"), SND_SOC_DAPM_OUTPUT("HSR"), - SND_SOC_DAPM_OUTPUT("LSPL"), SND_SOC_DAPM_OUTPUT("LSP"), SND_SOC_DAPM_OUTPUT("SP"), SND_SOC_DAPM_OUTPUT("CDCOUT"), @@ -563,21 +576,26 @@ static struct snd_soc_dapm_route mc13783_routes[] = { { "Speaker Amp Source MUX", "CODEC", "Voice CODEC PGA"}, { "Speaker Amp Source MUX", "Right", "DAC PGA"}, + { "Loudspeaker Amp Source MUX", "CODEC", "Voice CODEC PGA"}, + { "Loudspeaker Amp Source MUX", "Right", "DAC PGA"}, { "Headset Amp Source MUX", "CODEC", "Voice CODEC PGA"}, { "Headset Amp Source MUX", "Mixer", "DAC PGA"}, + { "Lineout Amp Source MUX", "CODEC", "Voice CODEC PGA"}, + { "Lineout Amp Source MUX", "Mixer", "DAC PGA"}, + /* Output */ { "HSL", NULL, "Headset Amp Left" }, { "HSR", NULL, "Headset Amp Right"}, { "RXOUTL", NULL, "Line out Amp Left"}, { "RXOUTR", NULL, "Line out Amp Right"}, { "SP", "Speaker Amp Switch", "Speaker Amp Source MUX"}, - { "LSP", "Loudspeaker Amp", "Speaker Amp Source MUX"}, + { "LSP", "Loudspeaker Amp", "Loudspeaker Amp Source MUX"}, { "HSL", "Headset Amp Left", "Headset Amp Source MUX"}, { "HSR", "Headset Amp Right", "Headset Amp Source MUX"}, - { "Line out Amp Left", NULL, "DAC PGA"}, - { "Line out Amp Right", NULL, "DAC PGA"}, + { "RXOUTL", "Line out Amp Left", "Lineout Amp Source MUX"}, + { "RXOUTR", "Line out Amp Right", "Lineout Amp Source MUX"}, { "DAC PGA", NULL, "DAC"}, { "DAC", NULL, "DAC_E"}, { "CDCOUT", "CDCOUT Switch", "Voice CODEC PGA"}, @@ -602,11 +620,12 @@ static struct snd_kcontrol_new mc13783_control_list[] = { 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("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 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("CODEC Capture Switch", MC13783_AUDIO_RX1, 0, 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), -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC mc13783 codec: complete capture mux
by Lambrecht Jürgen 29 Nov '13

29 Nov '13

codec adc (capture) mux was not complete: * fix naming of TXIN/TX * reorder RXIN and TXIN to show muxing priority * implement RXIN/RX (R and L are the same control) Signed-off-by: Jürgen Lambrecht <J.Lambrecht(a)televic.com> --- sound/soc/codecs/mc13783.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index bae6016..af7bc55 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -401,11 +401,14 @@ static const struct snd_kcontrol_new mc2_amp_ctl = static const struct snd_kcontrol_new atx_amp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 11, 1, 0); +static const struct snd_kcontrol_new rx_sel_ctl = + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 13, 1, 0); + /* Virtual mux. The chip does the input selection automatically * as soon as we enable one input. */ static const char * const adcl_enum_text[] = { - "MC1L", "RXINL", + "MC1L", "RXL", }; static const struct soc_enum adcl_enum = @@ -415,7 +418,7 @@ static const struct snd_kcontrol_new left_input_mux = SOC_DAPM_ENUM_VIRT("Route", adcl_enum); static const char * const adcr_enum_text[] = { - "MC1R", "MC2", "RXINR", "TXIN", + "MC1R", "MC2", "TX", "RXR", }; static const struct soc_enum adcr_enum = @@ -470,9 +473,9 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MC1LIN"), SND_SOC_DAPM_INPUT("MC1RIN"), SND_SOC_DAPM_INPUT("MC2IN"), + SND_SOC_DAPM_INPUT("TXIN"), SND_SOC_DAPM_INPUT("RXINR"), SND_SOC_DAPM_INPUT("RXINL"), - SND_SOC_DAPM_INPUT("TXIN"), 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), @@ -480,7 +483,8 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { 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_SWITCH("TX Amp", MC13783_AUDIO_TX, 11, 0, &atx_amp_ctl), + SND_SOC_DAPM_SWITCH("RX Sel", MC13783_AUDIO_TX, 13, 0, &rx_sel_ctl), SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0, &left_input_mux), @@ -537,14 +541,16 @@ static struct snd_soc_dapm_route mc13783_routes[] = { { "MC1L Amp", NULL, "MC1LIN"}, { "MC1R Amp", NULL, "MC1RIN" }, { "MC2 Amp", NULL, "MC2IN" }, - { "TXIN Amp", NULL, "TXIN"}, + { "TX Amp", NULL, "TXIN"}, + { "RX Sel", NULL, "RXINR"}, + { "RX Sel", NULL, "RXINL"}, /* indeed same control bit for RXINR and RXINL */ { "PGA Left Input Mux", "MC1L", "MC1L Amp" }, - { "PGA Left Input Mux", "RXINL", "RXINL"}, + { "PGA Left Input Mux", "RXL", "RX Sel"}, { "PGA Right Input Mux", "MC1R", "MC1R Amp" }, { "PGA Right Input Mux", "MC2", "MC2 Amp"}, - { "PGA Right Input Mux", "TXIN", "TXIN Amp"}, - { "PGA Right Input Mux", "RXINR", "RXINR"}, + { "PGA Right Input Mux", "TX", "TX Amp"}, + { "PGA Right Input Mux", "RXR", "RX Sel"}, { "PGA Left Input", NULL, "PGA Left Input Mux"}, { "PGA Right Input", NULL, "PGA Right Input Mux"}, -- 1.7.10.4

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[alsa-devel] [patch] ALSA: dice: fix array limits in dice_proc_read()
by Dan Carpenter 29 Nov '13

29 Nov '13

The array limits are supposed to be in units of u32 instead of in bytes. The current code has a potential array overflow. Fixes: c614475b0ea9 ('ALSA: dice: add a proc file to show device information') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 57bcd31fcc12..c0aa64941cee 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -1019,7 +1019,7 @@ static void dice_proc_read(struct snd_info_entry *entry, if (dice_proc_read_mem(dice, &tx_rx_header, sections[2], 2) < 0) return; - quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx)); + quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx) / 4); for (stream = 0; stream < tx_rx_header.number; ++stream) { if (dice_proc_read_mem(dice, &buf.tx, sections[2] + 2 + stream * tx_rx_header.size, @@ -1045,7 +1045,7 @@ static void dice_proc_read(struct snd_info_entry *entry, if (dice_proc_read_mem(dice, &tx_rx_header, sections[4], 2) < 0) return; - quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx)); + quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx) / 4); for (stream = 0; stream < tx_rx_header.number; ++stream) { if (dice_proc_read_mem(dice, &buf.rx, sections[4] + 2 + stream * tx_rx_header.size,

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[alsa-devel] [PATCH] ALSA: hda - Add mono speaker quirk for Dell Inspiron 5439
by David Henningsson 29 Nov '13

29 Nov '13

This machine also has mono output if run through DAC node 0x03. Cc: stable(a)vger.kernel.org (v3.10+) BugLink: https://bugs.launchpad.net/bugs/1256212 Tested-by: David Chen <david.chen(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2ad568e..8099200 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4243,6 +4243,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), -- 1.7.9.5

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[alsa-devel] [PATCH 00/11] codecs: Use IS_ENABLED() macro
by Fabio Estevam 29 Nov '13

29 Nov '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Using the IS_ENABLED() macro can make the code shorter and simpler. Mark, There are still more ocurrences that can be converted to IS_ENABLED. If you are happy with this series I will send another one with the remaining pieces.

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