September 2010 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: HDA: Enable internal speaker on Dell M101z
by David Henningsson 16 Sep '10

16 Sep '10

BugLink: http://launchpad.net/bugs/640254 In some cases a magic processing coefficient is needed to enable the internal speaker on Dell M101z. According to Realtek, this processing coefficient is only present on ALC269vb. Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 files changed, 10 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ab2947d..2063d0f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14671,6 +14671,7 @@ static int alc269_resume(struct hda_codec *codec) enum { ALC269_FIXUP_SONY_VAIO, + ALC269_FIXUP_DELL_M101Z, }; static const struct alc_fixup alc269_fixups[] = { @@ -14680,11 +14681,20 @@ static const struct alc_fixup alc269_fixups[] = { {} } }, + [ALC269_FIXUP_DELL_M101Z] = { + .verbs = (const struct hda_verb[]) { + /* Enables internal speaker */ + {0x20, AC_VERB_SET_COEF_INDEX, 13}, + {0x20, AC_VERB_SET_PROC_COEF, 0x4040}, + {} + } + }, }; static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x104d, 0x9077, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), + SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), {} }; -- 1.7.1 --------------000100000009090600000802--

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[alsa-devel] [PATCH] ASoC: da7210: code clean up
by Kuninori Morimoto 16 Sep '10

16 Sep '10

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/da7210.c | 18 +++++++++--------- 1 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index eabf3c0..58bb9b9 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -454,7 +454,7 @@ static int da7210_probe(struct snd_soc_codec *codec) dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); - codec->control_data = da7210->control_data; + codec->control_data = da7210->control_data; codec->hw_write = (hw_write_t)i2c_master_send; /* FIXME @@ -547,11 +547,11 @@ static int da7210_probe(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_da7210 = { - .probe = da7210_probe, - .read = da7210_read, - .write = da7210_write, - .reg_cache_size = ARRAY_SIZE(da7210_reg), - .reg_word_size = sizeof(u8), + .probe = da7210_probe, + .read = da7210_read, + .write = da7210_write, + .reg_cache_size = ARRAY_SIZE(da7210_reg), + .reg_word_size = sizeof(u8), .reg_cache_default = da7210_reg, }; @@ -597,9 +597,9 @@ static struct i2c_driver da7210_i2c_driver = { .name = "da7210-codec", .owner = THIS_MODULE, }, - .probe = da7210_i2c_probe, - .remove = __devexit_p(da7210_i2c_remove), - .id_table = da7210_i2c_id, + .probe = da7210_i2c_probe, + .remove = __devexit_p(da7210_i2c_remove), + .id_table = da7210_i2c_id, }; #endif -- 1.7.0.4

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[alsa-devel] [PATCH] ALSA nvhdmi: Add missing codec IDs
by Stephen Warren 15 Sep '10

15 Sep '10

Adds codec IDs for various other NVIDIA GPUs to the NVIDIA HDMI audio driver. This is a backport of 9cf2657d05f9f9e04c3e113d68bf7cef5a942070 suitable for inclusion into 2.6.34 and 2.6.35. ... but subset so that it solely adds new IDs, and doesn't reformat nor rename any existing IDs. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/pci/hda/patch_nvhdmi.c | 30 ++++++++++++++++++++++++++++++ 1 files changed, 30 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index b0652ac..dccfcf4 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -559,6 +559,21 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de000d, .name = "GT240 HDMI", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0010, .name = "GPU 10 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0011, .name = "GPU 11 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, {} /* terminator */ @@ -573,6 +588,21 @@ MODULE_ALIAS("snd-hda-codec-id:10de000a"); MODULE_ALIAS("snd-hda-codec-id:10de000b"); MODULE_ALIAS("snd-hda-codec-id:10de000c"); MODULE_ALIAS("snd-hda-codec-id:10de000d"); +MODULE_ALIAS("snd-hda-codec-id:10de0010"); +MODULE_ALIAS("snd-hda-codec-id:10de0011"); +MODULE_ALIAS("snd-hda-codec-id:10de0012"); +MODULE_ALIAS("snd-hda-codec-id:10de0013"); +MODULE_ALIAS("snd-hda-codec-id:10de0014"); +MODULE_ALIAS("snd-hda-codec-id:10de0018"); +MODULE_ALIAS("snd-hda-codec-id:10de0019"); +MODULE_ALIAS("snd-hda-codec-id:10de001a"); +MODULE_ALIAS("snd-hda-codec-id:10de001b"); +MODULE_ALIAS("snd-hda-codec-id:10de001c"); +MODULE_ALIAS("snd-hda-codec-id:10de0040"); +MODULE_ALIAS("snd-hda-codec-id:10de0041"); +MODULE_ALIAS("snd-hda-codec-id:10de0042"); +MODULE_ALIAS("snd-hda-codec-id:10de0043"); +MODULE_ALIAS("snd-hda-codec-id:10de0044"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); -- 1.6.0.6 nvpublic

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Re: [alsa-devel] [PATCH] ASoC: WM8985: Initial driver
by Mark Brown 15 Sep '10

15 Sep '10

On Wed, Sep 15, 2010 at 03:23:20PM +0100, Dimitris Papastamos wrote: > The WM8985 is a low power, high quality, feature-rich stereo > CODEC designed for portable multimedia applications that > require low power consumption and high quality audio. > > Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> Applied, thanks. > sound/soc/codecs/wm8985.c | 1195 +++++++++++++++++++++++++++++++++++++++++++++ > sound/soc/codecs/wm8985.h | 1046 +++++++++++++++++++++++++++++++++++++++ > 2 files changed, 2241 insertions(+), 0 deletions(-) As discussed I added the Makefile and Kconfig entries as I did so.

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Re: [alsa-devel] [PATCH] ASoC: WM8985: Initial driver
by Liam Girdwood 15 Sep '10

15 Sep '10

On Wed, 2010-09-15 at 15:23 +0100, Dimitris Papastamos wrote: > The WM8985 is a low power, high quality, feature-rich stereo > CODEC designed for portable multimedia applications that > require low power consumption and high quality audio. > > Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> > --- > sound/soc/codecs/wm8985.c | 1195 +++++++++++++++++++++++++++++++++++++++++++++ > sound/soc/codecs/wm8985.h | 1046 +++++++++++++++++++++++++++++++++++++++ > 2 files changed, 2241 insertions(+), 0 deletions(-) > create mode 100644 sound/soc/codecs/wm8985.c > create mode 100644 sound/soc/codecs/wm8985.h Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

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[alsa-devel] [PATCH 1/3] ASoC: tlv320aic3x: Use snd_soc_read and snd_soc_write
by Jarkko Nikula 15 Sep '10

15 Sep '10

Start phasing out aic3x_read_reg_cache, aic3x_write_reg_cache, aic3x_read and aic3x_write calls in order to switch to soc-cache helpers. This patch replaces aic3x_read_reg_cache and aic3x_write with snd_soc_read and snd_soc_write. This is basically null-op since .read and .write in soc_codec_dev_aic3x points to aic3x_read_reg_cache and aic3x_write. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/codecs/tlv320aic3x.c | 194 ++++++++++++++++++++-------------------- 1 files changed, 98 insertions(+), 96 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 94dc707..138165c 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -839,8 +839,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, int clk; /* select data word length */ - data = - aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4)); + data = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4)); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: break; @@ -854,7 +853,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, data |= (0x03 << 4); break; } - aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, data); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLB, data); /* Fsref can be 44100 or 48000 */ fsref = (params_rate(params) % 11025 == 0) ? 44100 : 48000; @@ -869,17 +868,17 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, if (bypass_pll) { pll_q &= 0xf; - aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT); - aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV); + snd_soc_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT); + snd_soc_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV); /* disable PLL if it is bypassed */ - reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); - aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE); + reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + snd_soc_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE); } else { - aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV); + snd_soc_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV); /* enable PLL when it is used */ - reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); - aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE); + reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + snd_soc_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE); } /* Route Left DAC to left channel input and @@ -888,7 +887,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000; if (params_rate(params) >= 64000) data |= DUAL_RATE_MODE; - aic3x_write(codec, AIC3X_CODEC_DATAPATH_REG, data); + snd_soc_write(codec, AIC3X_CODEC_DATAPATH_REG, data); /* codec sample rate select */ data = (fsref * 20) / params_rate(params); @@ -897,7 +896,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, data /= 5; data -= 2; data |= (data << 4); - aic3x_write(codec, AIC3X_SAMPLE_RATE_SEL_REG, data); + snd_soc_write(codec, AIC3X_SAMPLE_RATE_SEL_REG, data); if (bypass_pll) return 0; @@ -966,13 +965,16 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, } found: - data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); - aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); - aic3x_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT); - aic3x_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT); - aic3x_write(codec, AIC3X_PLL_PROGC_REG, (pll_d >> 6) << PLLD_MSB_SHIFT); - aic3x_write(codec, AIC3X_PLL_PROGD_REG, - (pll_d & 0x3F) << PLLD_LSB_SHIFT); + data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + snd_soc_write(codec, AIC3X_PLL_PROGA_REG, + data | (pll_p << PLLP_SHIFT)); + snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, + pll_r << PLLR_SHIFT); + snd_soc_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT); + snd_soc_write(codec, AIC3X_PLL_PROGC_REG, + (pll_d >> 6) << PLLD_MSB_SHIFT); + snd_soc_write(codec, AIC3X_PLL_PROGD_REG, + (pll_d & 0x3F) << PLLD_LSB_SHIFT); return 0; } @@ -980,15 +982,15 @@ found: static int aic3x_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u8 ldac_reg = aic3x_read_reg_cache(codec, LDAC_VOL) & ~MUTE_ON; - u8 rdac_reg = aic3x_read_reg_cache(codec, RDAC_VOL) & ~MUTE_ON; + u8 ldac_reg = snd_soc_read(codec, LDAC_VOL) & ~MUTE_ON; + u8 rdac_reg = snd_soc_read(codec, RDAC_VOL) & ~MUTE_ON; if (mute) { - aic3x_write(codec, LDAC_VOL, ldac_reg | MUTE_ON); - aic3x_write(codec, RDAC_VOL, rdac_reg | MUTE_ON); + snd_soc_write(codec, LDAC_VOL, ldac_reg | MUTE_ON); + snd_soc_write(codec, RDAC_VOL, rdac_reg | MUTE_ON); } else { - aic3x_write(codec, LDAC_VOL, ldac_reg); - aic3x_write(codec, RDAC_VOL, rdac_reg); + snd_soc_write(codec, LDAC_VOL, ldac_reg); + snd_soc_write(codec, RDAC_VOL, rdac_reg); } return 0; @@ -1012,8 +1014,8 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, u8 iface_areg, iface_breg; int delay = 0; - iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f; - iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f; + iface_areg = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f; + iface_breg = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -1052,9 +1054,9 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, } /* set iface */ - aic3x_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg); - aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg); - aic3x_write(codec, AIC3X_ASD_INTF_CTRLC, delay); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay); return 0; } @@ -1072,18 +1074,18 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, if (codec->bias_level == SND_SOC_BIAS_STANDBY && aic3x->master) { /* enable pll */ - reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); - aic3x_write(codec, AIC3X_PLL_PROGA_REG, - reg | PLL_ENABLE); + reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + snd_soc_write(codec, AIC3X_PLL_PROGA_REG, + reg | PLL_ENABLE); } break; case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_PREPARE && aic3x->master) { /* disable pll */ - reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); - aic3x_write(codec, AIC3X_PLL_PROGA_REG, - reg & ~PLL_ENABLE); + reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + snd_soc_write(codec, AIC3X_PLL_PROGA_REG, + reg & ~PLL_ENABLE); } break; case SND_SOC_BIAS_OFF: @@ -1098,8 +1100,8 @@ void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state) { u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG; u8 bit = gpio ? 3: 0; - u8 val = aic3x_read_reg_cache(codec, reg) & ~(1 << bit); - aic3x_write(codec, reg, val | (!!state << bit)); + u8 val = snd_soc_read(codec, reg) & ~(1 << bit); + snd_soc_write(codec, reg, val | (!!state << bit)); } EXPORT_SYMBOL_GPL(aic3x_set_gpio); @@ -1128,7 +1130,7 @@ void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect, if (detect & AIC3X_HEADSET_DETECT_MASK) val |= AIC3X_HEADSET_DETECT_ENABLED; - aic3x_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val); + snd_soc_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val); } EXPORT_SYMBOL_GPL(aic3x_set_headset_detection); @@ -1211,81 +1213,81 @@ static int aic3x_init(struct snd_soc_codec *codec) struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); int reg; - aic3x_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); - aic3x_write(codec, AIC3X_RESET, SOFT_RESET); + snd_soc_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); + snd_soc_write(codec, AIC3X_RESET, SOFT_RESET); /* DAC default volume and mute */ - aic3x_write(codec, LDAC_VOL, DEFAULT_VOL | MUTE_ON); - aic3x_write(codec, RDAC_VOL, DEFAULT_VOL | MUTE_ON); + snd_soc_write(codec, LDAC_VOL, DEFAULT_VOL | MUTE_ON); + snd_soc_write(codec, RDAC_VOL, DEFAULT_VOL | MUTE_ON); /* DAC to HP default volume and route to Output mixer */ - aic3x_write(codec, DACL1_2_HPLOUT_VOL, DEFAULT_VOL | ROUTE_ON); - aic3x_write(codec, DACR1_2_HPROUT_VOL, DEFAULT_VOL | ROUTE_ON); - aic3x_write(codec, DACL1_2_HPLCOM_VOL, DEFAULT_VOL | ROUTE_ON); - aic3x_write(codec, DACR1_2_HPRCOM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACL1_2_HPLOUT_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_HPROUT_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACL1_2_HPLCOM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_HPRCOM_VOL, DEFAULT_VOL | ROUTE_ON); /* DAC to Line Out default volume and route to Output mixer */ - aic3x_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - aic3x_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON); /* DAC to Mono Line Out default volume and route to Output mixer */ - aic3x_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - aic3x_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); /* unmute all outputs */ - reg = aic3x_read_reg_cache(codec, LLOPM_CTRL); - aic3x_write(codec, LLOPM_CTRL, reg | UNMUTE); - reg = aic3x_read_reg_cache(codec, RLOPM_CTRL); - aic3x_write(codec, RLOPM_CTRL, reg | UNMUTE); - reg = aic3x_read_reg_cache(codec, MONOLOPM_CTRL); - aic3x_write(codec, MONOLOPM_CTRL, reg | UNMUTE); - reg = aic3x_read_reg_cache(codec, HPLOUT_CTRL); - aic3x_write(codec, HPLOUT_CTRL, reg | UNMUTE); - reg = aic3x_read_reg_cache(codec, HPROUT_CTRL); - aic3x_write(codec, HPROUT_CTRL, reg | UNMUTE); - reg = aic3x_read_reg_cache(codec, HPLCOM_CTRL); - aic3x_write(codec, HPLCOM_CTRL, reg | UNMUTE); - reg = aic3x_read_reg_cache(codec, HPRCOM_CTRL); - aic3x_write(codec, HPRCOM_CTRL, reg | UNMUTE); + reg = snd_soc_read(codec, LLOPM_CTRL); + snd_soc_write(codec, LLOPM_CTRL, reg | UNMUTE); + reg = snd_soc_read(codec, RLOPM_CTRL); + snd_soc_write(codec, RLOPM_CTRL, reg | UNMUTE); + reg = snd_soc_read(codec, MONOLOPM_CTRL); + snd_soc_write(codec, MONOLOPM_CTRL, reg | UNMUTE); + reg = snd_soc_read(codec, HPLOUT_CTRL); + snd_soc_write(codec, HPLOUT_CTRL, reg | UNMUTE); + reg = snd_soc_read(codec, HPROUT_CTRL); + snd_soc_write(codec, HPROUT_CTRL, reg | UNMUTE); + reg = snd_soc_read(codec, HPLCOM_CTRL); + snd_soc_write(codec, HPLCOM_CTRL, reg | UNMUTE); + reg = snd_soc_read(codec, HPRCOM_CTRL); + snd_soc_write(codec, HPRCOM_CTRL, reg | UNMUTE); /* ADC default volume and unmute */ - aic3x_write(codec, LADC_VOL, DEFAULT_GAIN); - aic3x_write(codec, RADC_VOL, DEFAULT_GAIN); + snd_soc_write(codec, LADC_VOL, DEFAULT_GAIN); + snd_soc_write(codec, RADC_VOL, DEFAULT_GAIN); /* By default route Line1 to ADC PGA mixer */ - aic3x_write(codec, LINE1L_2_LADC_CTRL, 0x0); - aic3x_write(codec, LINE1R_2_RADC_CTRL, 0x0); + snd_soc_write(codec, LINE1L_2_LADC_CTRL, 0x0); + snd_soc_write(codec, LINE1R_2_RADC_CTRL, 0x0); /* PGA to HP Bypass default volume, disconnect from Output Mixer */ - aic3x_write(codec, PGAL_2_HPLOUT_VOL, DEFAULT_VOL); - aic3x_write(codec, PGAR_2_HPROUT_VOL, DEFAULT_VOL); - aic3x_write(codec, PGAL_2_HPLCOM_VOL, DEFAULT_VOL); - aic3x_write(codec, PGAR_2_HPRCOM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAL_2_HPLOUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_HPROUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAL_2_HPLCOM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_HPRCOM_VOL, DEFAULT_VOL); /* PGA to Line Out default volume, disconnect from Output Mixer */ - aic3x_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); - aic3x_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); /* PGA to Mono Line Out default volume, disconnect from Output Mixer */ - aic3x_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL); - aic3x_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL); /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ - aic3x_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); - aic3x_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); - aic3x_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); - aic3x_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); /* Line2 Line Out default volume, disconnect from Output Mixer */ - aic3x_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); - aic3x_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); /* Line2 to Mono Out default volume, disconnect from Output Mixer */ - aic3x_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); - aic3x_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); if (aic3x->model == AIC3X_MODEL_3007) { /* Class-D speaker driver init; datasheet p. 46 */ - aic3x_write(codec, AIC3X_PAGE_SELECT, 0x0D); - aic3x_write(codec, 0xD, 0x0D); - aic3x_write(codec, 0x8, 0x5C); - aic3x_write(codec, 0x8, 0x5D); - aic3x_write(codec, 0x8, 0x5C); - aic3x_write(codec, AIC3X_PAGE_SELECT, 0x00); - aic3x_write(codec, CLASSD_CTRL, 0); + snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x0D); + snd_soc_write(codec, 0xD, 0x0D); + snd_soc_write(codec, 0x8, 0x5C); + snd_soc_write(codec, 0x8, 0x5D); + snd_soc_write(codec, 0x8, 0x5C); + snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x00); + snd_soc_write(codec, CLASSD_CTRL, 0); } /* off, with power on */ @@ -1305,10 +1307,10 @@ static int aic3x_probe(struct snd_soc_codec *codec) if (aic3x->setup) { /* setup GPIO functions */ - aic3x_write(codec, AIC3X_GPIO1_REG, - (aic3x->setup->gpio_func[0] & 0xf) << 4); - aic3x_write(codec, AIC3X_GPIO2_REG, - (aic3x->setup->gpio_func[1] & 0xf) << 4); + snd_soc_write(codec, AIC3X_GPIO1_REG, + (aic3x->setup->gpio_func[0] & 0xf) << 4); + snd_soc_write(codec, AIC3X_GPIO2_REG, + (aic3x->setup->gpio_func[1] & 0xf) << 4); } snd_soc_add_controls(codec, aic3x_snd_controls, -- 1.7.1

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[alsa-devel] [PATCH] ASoC: Remove needless codec->bias_level assignment to SND_SOC_BIAS_OFF
by Jarkko Nikula 15 Sep '10

15 Sep '10

This assignment is done by the snd_soc_register_codec so there is no need to redo it in probe function of a codec driver. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- I can separate this patch between TI and other codecs but I think this is trivial enough to be carried either through Mark's or Liam's tree. --- sound/soc/codecs/ak4671.c | 1 - sound/soc/codecs/ssm2602.c | 1 - sound/soc/codecs/tlv320dac33.c | 1 - sound/soc/codecs/twl4030.c | 1 - sound/soc/codecs/wm8510.c | 1 - sound/soc/codecs/wm8731.c | 2 -- sound/soc/codecs/wm8753.c | 1 - sound/soc/codecs/wm8990.c | 1 - 8 files changed, 0 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 239f056..24f5f49 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -642,7 +642,6 @@ static int ak4671_probe(struct snd_soc_codec *codec) int ret; codec->hw_write = (hw_write_t)i2c_master_send; - codec->bias_level = SND_SOC_BIAS_OFF; ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4671->control_type); if (ret < 0) { diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 67d8c04..6f38d61 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -560,7 +560,6 @@ static int ssm2602_probe(struct snd_soc_codec *codec) pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION); - codec->bias_level = SND_SOC_BIAS_OFF, codec->control_data = ssm2602->control_data; ssm2602_reset(codec); diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index a3c5b52..bf6d01f 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1385,7 +1385,6 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) codec->control_data = dac33->control_data; codec->hw_write = (hw_write_t) i2c_master_send; - codec->bias_level = SND_SOC_BIAS_OFF; codec->idle_bias_off = 1; dac33->codec = codec; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c7ee1a4..cbebec6 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2245,7 +2245,6 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, twl4030); /* Set the defaults, and power up the codec */ twl4030->sysclk = twl4030_codec_get_mclk() / 1000; - codec->bias_level = SND_SOC_BIAS_OFF; codec->idle_bias_off = 1; twl4030_init_chip(codec); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index d00da20..02ecf54 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -569,7 +569,6 @@ static int wm8510_probe(struct snd_soc_codec *codec) wm8510_reset(codec); /* power on device */ - codec->bias_level = SND_SOC_BIAS_OFF; wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); snd_soc_add_controls(codec, wm8510_snd_controls, ARRAY_SIZE(wm8510_snd_controls)); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 56f5408..6980ff8 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -488,8 +488,6 @@ static int wm8731_probe(struct snd_soc_codec *codec) struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); int ret = 0, i; - codec->bias_level = SND_SOC_BIAS_OFF, - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8731->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 4a945d3..6c0ff35 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1550,7 +1550,6 @@ static int wm8753_probe(struct snd_soc_codec *codec) struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); int ret = 0, reg; - codec->bias_level = SND_SOC_BIAS_OFF; INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work); ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8753->control_type); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 0ffecbd..264828e 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1354,7 +1354,6 @@ static int wm8990_probe(struct snd_soc_codec *codec) wm8990_reset(codec); /* charge output caps */ - codec->bias_level = SND_SOC_BIAS_OFF; wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); reg = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_4); -- 1.7.1

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[alsa-devel] need help diagnosing USB MIDI latency problem with MPD16
by Krzysztof Foltman 15 Sep '10

15 Sep '10

Hello, Some time ago, I wrote the Akai MPD16 protocol handlers (input/output) for the USB MIDI driver. Unfortunately, the kernel driver seems to suffer from a rather embarrassing input latency, it looks like the MIDI event timing granularity is >100ms (so the latency is variable, making the pad practically useless except if a rhythm is played with the rate matching the polling rate!). At the same time, a simple Python based contraption using libusb and bulkRead in a while loop, gets much better results. The same USB MIDI driver in the same kernel and PC doesn't have noticeable problems with class-compliant E-Mu Xboard25 connected to the same USB port. Both devices use endpoints in bulk mode. I've attached the excerpts of lsusb -v output for both devices - the only differences are: 1) the Akai device isn't class compliant, 2) the E-mu device has one pair of inputs/outputs, the Akai has two (control/configuration and MIDI proper) 3) the maximum packet length is 32 for E-mu, 64 for Akai. When I enabled logging in midi.c (#define DUMP_PACKETS), it looks like the device is returning data packets rather infrequently, as a single packet often contains multiple events (up to 3) when two pads are pressed in rapid succession. So, it looks like a problem outside of my snd_usbmidi_akai_input. One thing that *does* help, though, is disabling the 'control' port by adding endpoints[0].in_cables = 0; in Akai quirk handling in snd_usbmidi_create - when input from the control port is not polled, the choppiness disappears completely. I'm guessing that the device itself (or something in between) has some problems having to handle the URBs for both endpoints, and that perhaps the IN/NAK exchange on a "control" endpoint causes the device's microcontroller (or something else?) to stall for a while. However, in that case, a proper solution (one that doesn't involve sacrificing the ability to use control port for configuration) would involve submitting URBs for the control input endpoint only when control port is open on ALSA side, and cancelling them when the port is closed. However, that sounds pretty invasive. Any hints? The device in question is rather uncommon and hasn't been supported for a while, so refactoring the whole USB MIDI driver to accommodate it doesn't sound like a great idea - but still, the latency thing sucks pretty bad for a drumming-oriented device! Thanks in advance, Krzysztof

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[alsa-devel] Fwd: Microphone not working on PCI ID 8086:3b56
by Jeff Chua 15 Sep '10

15 Sep '10

Hi, I'm having problem getting the microphone to work on the Lenovo X201s, but the audio output is working fine. It's showing up as: 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06) [00:1b.0 0403: 8086:3b56 (rev 06)] # uname -s Linux boston 2.6.36-rc4 #57 SMP PREEMPT Tue Sep 14 05:12:21 EDT 2010 x86_64 GNU/Linux # cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer 160: [ 5] : control # cat /proc/asound/pcm 00-00: CONEXANT Analog : CONEXANT Analog : playback 1 : capture 1 # dmesg thinkpad_acpi: Console audio control enabled, mode: override (read/write) input: ThinkPad Extra Buttons as /devices/platform/thinkpad_acpi/input/input6 HDA Intel 0000:00:1b.0: PCI INT B -> GSI 17 (level, low) -> IRQ 17 HDA Intel 0000:00:1b.0: irq 45 for MSI/MSI-X HDA Intel 0000:00:1b.0: setting latency timer to 64 hda-codec: No codec parser is available ALSA device list: #0: HDA Intel at 0xf2520000 irq 45 #5: ThinkPad Console Audio Control at EC reg 0x30, fw 6QHT29WW-1.10 I've alsa-plugins-1.0.23, alsa-utils-1.0.23, alsa-lib-1.0.23 installed. alsamixer is showing Capture at 100%, Analog Mic Boost at 40db and Digital at 100, but captured using "arecord" or "rec" and then playback is all just dead silence! Any pointers on how to make this work would be very much appreciated. Thanks, Jeff

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[alsa-devel] [PATCH] patch_nvhdmi.c: Fix supported sample rate list.
by Stephen Warren 14 Sep '10

14 Sep '10

22050 isn't a valid HDMI sample rate. 32000 is. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> Acked-By: Wei Ni <wni(a)nvidia.com> --- sound/pci/hda/patch_nvhdmi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 69b950d..baa108b 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -84,7 +84,7 @@ static struct hda_verb nvhdmi_basic_init_7x[] = { #else /* support all rates and formats */ #define SUPPORTED_RATES \ - (SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\ SNDRV_PCM_RATE_192000) #define SUPPORTED_MAXBPS 24 -- 1.7.0.4 nvpublic

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