August 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ASoC updates for 3.2
by Mark Brown 25 Aug '11

25 Aug '11

The following changes since commit e3d73c1bbf08b9abd3f56293796ba7b5c15008f5: ASoC: sta32x: Move resource allocation and release to the corresponding callback functions (2011-08-19 01:25:56 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.2 Axel Lin (1): ASoC: soc-core: use GFP_KERNEL flag for kmalloc in snd_soc_cnew Dong Aisheng (2): ASoC: mxs-saif: clear clk gate first before register setting ASoC: mxs-sgtl5000: add record function Joseph Pentland (1): ASoC: Add Springbank I/O card to Speyside Kconfig Julia Lawall (5): sound/soc/kirkwood/kirkwood-i2s.c: add missing kfree sound/soc/ep93xx/ep93xx-i2s.c: add missing kfree sound/soc/fsl/p1022_ds.c: add missing of_node_put sound/soc/fsl/mpc8610_hpcd.c: add missing of_node_put sound/soc/mxs/mxs-saif.c: add missing kfree Mark Brown (18): ASoC: WM8996 record paths need AIFCLK ASoC: Fix configuration of WM8996 input enables ASoC: Automatically manage WM8996 MICBIAS regulating mode ASoC: Optimise WM8996 no interrupt path ASoC: Add basic WM8962 capture low/high pass filter control ASoC: Ensure we only run Speyside WM8962 bias level callbacks once ASoC: Initial WM8962 DSP2 support ASoC: Report IRQ_NONE when we don't see an interrupt from WM8962 ASoC: Clear any outstanding WM8962 FLL lock completions before waiting ASoC: Provide more detail on WM8962 thermal shutdown status ASoC: Clear completions from late WM8996 FLL lock IRQs Merge branch 'for-3.1' into for-3.2 ASoC: Convert WM8523 to table based control and DAPM initialization ASoC: Allow idle_bias_off to be specified in CODEC drivers Merge branch 'for-3.1' into for-3.2 Merge branch 'for-3.1' into for-3.2 ASoC: Correct element count for WM8996 sidetone HPF Merge branch 'for-3.1' into for-3.2 Sangbeom Kim (2): ASoC: Add samsung maintainer ASoC: SAMSUNG: Add Kconfig to support SMDK4212 Stephen Warren (1): ASoC: Tegra: wm8903 machine driver: Drop Ventana support Timur Tabi (2): sound/soc/fsl/fsl_dma.c: add missing of_node_put ASoC: MPC5200: replace of_device with platform_device MAINTAINERS | 1 + include/sound/soc.h | 1 + sound/soc/codecs/wm8523.c | 26 ++--- sound/soc/codecs/wm8962.c | 191 ++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm8996.c | 40 +++++--- sound/soc/ep93xx/ep93xx-i2s.c | 5 +- sound/soc/fsl/fsl_dma.c | 2 + sound/soc/fsl/mpc5200_dma.c | 6 +- sound/soc/fsl/mpc8610_hpcd.c | 18 ++-- sound/soc/fsl/p1022_ds.c | 4 +- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/mxs/mxs-saif.c | 27 +++-- sound/soc/mxs/mxs-sgtl5000.c | 10 ++- sound/soc/samsung/Kconfig | 7 +- sound/soc/samsung/speyside_wm8962.c | 6 + sound/soc/soc-core.c | 4 +- sound/soc/tegra/tegra_wm8903.c | 4 +- 17 files changed, 285 insertions(+), 69 deletions(-)

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[alsa-devel] ASoC updates for 3.1
by Mark Brown 25 Aug '11

25 Aug '11

The following changes since commit e574044acbad7421879270a80acd337459c94cc8: ASoC: omap: Fix build errors in ams-delta (2011-08-19 01:19:45 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 A fairly large collection of small, local patches that mostly only affect individual drivers - should be safe. Axel Lin (1): ASoC: soc-core: use GFP_KERNEL flag for kmalloc in snd_soc_cnew Joseph Pentland (1): ASoC: Add Springbank I/O card to Speyside Kconfig Julia Lawall (4): sound/soc/kirkwood/kirkwood-i2s.c: add missing kfree sound/soc/ep93xx/ep93xx-i2s.c: add missing kfree sound/soc/fsl/p1022_ds.c: add missing of_node_put sound/soc/fsl/mpc8610_hpcd.c: add missing of_node_put Mark Brown (6): ASoC: WM8996 record paths need AIFCLK ASoC: Fix configuration of WM8996 input enables ASoC: Ensure we only run Speyside WM8962 bias level callbacks once ASoC: Clear any outstanding WM8962 FLL lock completions before waiting ASoC: Clear completions from late WM8996 FLL lock IRQs ASoC: Correct element count for WM8996 sidetone HPF Sangbeom Kim (1): ASoC: Add samsung maintainer Stephen Warren (1): ASoC: Tegra: wm8903 machine driver: Drop Ventana support Timur Tabi (1): sound/soc/fsl/fsl_dma.c: add missing of_node_put MAINTAINERS | 1 + sound/soc/codecs/wm8962.c | 4 ++++ sound/soc/codecs/wm8996.c | 28 ++++++++++++++++++---------- sound/soc/ep93xx/ep93xx-i2s.c | 5 +++-- sound/soc/fsl/fsl_dma.c | 2 ++ sound/soc/fsl/mpc8610_hpcd.c | 18 +++++++++--------- sound/soc/fsl/p1022_ds.c | 4 +++- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/samsung/Kconfig | 1 + sound/soc/samsung/speyside_wm8962.c | 6 ++++++ sound/soc/soc-core.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 4 ++-- 12 files changed, 51 insertions(+), 26 deletions(-)

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[PATCH] ALSA: hda: Conexant: Allow different output types to share DAC
by David Henningsson 25 Aug '11

25 Aug '11

Headphones has stopped working for the original reported (a regression compared to 2.6.38). This is because Speaker and Headphones share the same DAC, in which case no Headphones volume control was created. This patch fixes so that both Speaker and Headphones volume controls are created in such scenario. BugLink: http://bugs.launchpad.net/bugs/817943 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 46 +++++++++++++++++++++++---------------- 1 files changed, 27 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 197ad93..aa0e4b9 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3329,18 +3329,26 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs) /* fill pin_dac_pair list from the pin and dac list */ static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins, int num_pins, hda_nid_t *dacs, int *rest, - struct pin_dac_pair *filled, int type) + struct pin_dac_pair *filled, int nums, + int type) { - int i, nums; + int i, start = nums; - nums = 0; - for (i = 0; i < num_pins; i++) { + for (i = 0; i < num_pins; i++, nums++) { filled[nums].pin = pins[i]; filled[nums].type = type; filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest); - if (!filled[nums].dac && i > 0 && filled[0].dac) + if (filled[nums].dac) + continue; + if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) { + filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG; + continue; + } + if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) { filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG; - nums++; + continue; + } + snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]); } return nums; } @@ -3356,14 +3364,14 @@ static void cx_auto_parse_output(struct hda_codec *codec) rest = fill_cx_auto_dacs(codec, dacs); /* parse all analog output pins */ nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs, - dacs, &rest, spec->dac_info, - AUTO_PIN_LINE_OUT); - nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, - dacs, &rest, spec->dac_info + nums, - AUTO_PIN_HP_OUT); - nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, - dacs, &rest, spec->dac_info + nums, - AUTO_PIN_SPEAKER_OUT); + dacs, &rest, spec->dac_info, 0, + AUTO_PIN_LINE_OUT); + nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, + dacs, &rest, spec->dac_info, nums, + AUTO_PIN_HP_OUT); + nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, + dacs, &rest, spec->dac_info, nums, + AUTO_PIN_SPEAKER_OUT); spec->dac_info_filled = nums; /* fill multiout struct */ for (i = 0; i < nums; i++) { @@ -4130,9 +4138,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac, hda_nid_t pin, const char *name, int idx) { unsigned int caps; - caps = query_amp_caps(codec, dac, HDA_OUTPUT); - if (caps & AC_AMPCAP_NUM_STEPS) - return cx_auto_add_pb_volume(codec, dac, name, idx); + if (dac && !(dac & DAC_SLAVE_FLAG)) { + caps = query_amp_caps(codec, dac, HDA_OUTPUT); + if (caps & AC_AMPCAP_NUM_STEPS) + return cx_auto_add_pb_volume(codec, dac, name, idx); + } caps = query_amp_caps(codec, pin, HDA_OUTPUT); if (caps & AC_AMPCAP_NUM_STEPS) return cx_auto_add_pb_volume(codec, pin, name, idx); @@ -4155,8 +4165,6 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) const char *label; int idx, type; hda_nid_t dac = spec->dac_info[i].dac; - if (!dac || (dac & DAC_SLAVE_FLAG)) - continue; type = spec->dac_info[i].type; if (type == AUTO_PIN_LINE_OUT) type = spec->autocfg.line_out_type; -- 1.7.4.1 --------------030805020208050803070801--

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[PATCH] ALSA: hda: Conexant: Allow different output types to share DAC
by David Henningsson 25 Aug '11

25 Aug '11

Headphones has stopped working for the original reported (a regression compared to 2.6.38). This is because Speaker and Headphones share the same DAC, in which case no Headphones volume control was created. This patch fixes so that both Speaker and Headphones volume controls are created in such scenario. Buglink: http://bugs.launchpad.net/bugs/817943 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 40 ++++++++++++++++++++++------------------ 1 files changed, 22 insertions(+), 18 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 197ad93..a29b88d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3327,22 +3327,23 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs) } /* fill pin_dac_pair list from the pin and dac list */ -static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins, +static void fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins, int num_pins, hda_nid_t *dacs, int *rest, - struct pin_dac_pair *filled, int type) + struct pin_dac_pair *filled, int *saved_nums, + int type) { int i, nums; - nums = 0; + nums = *saved_nums; for (i = 0; i < num_pins; i++) { filled[nums].pin = pins[i]; filled[nums].type = type; filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest); - if (!filled[nums].dac && i > 0 && filled[0].dac) + if (!filled[nums].dac && filled[0].dac) filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG; nums++; } - return nums; + *saved_nums = nums; } /* parse analog output paths */ @@ -3355,15 +3356,16 @@ static void cx_auto_parse_output(struct hda_codec *codec) rest = fill_cx_auto_dacs(codec, dacs); /* parse all analog output pins */ - nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs, - dacs, &rest, spec->dac_info, - AUTO_PIN_LINE_OUT); - nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, - dacs, &rest, spec->dac_info + nums, - AUTO_PIN_HP_OUT); - nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, - dacs, &rest, spec->dac_info + nums, - AUTO_PIN_SPEAKER_OUT); + nums = 0; + fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs, + dacs, &rest, spec->dac_info, &nums, + AUTO_PIN_LINE_OUT); + fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, + dacs, &rest, spec->dac_info, &nums, + AUTO_PIN_HP_OUT); + fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, + dacs, &rest, spec->dac_info, &nums, + AUTO_PIN_SPEAKER_OUT); spec->dac_info_filled = nums; /* fill multiout struct */ for (i = 0; i < nums; i++) { @@ -4130,9 +4132,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac, hda_nid_t pin, const char *name, int idx) { unsigned int caps; - caps = query_amp_caps(codec, dac, HDA_OUTPUT); - if (caps & AC_AMPCAP_NUM_STEPS) - return cx_auto_add_pb_volume(codec, dac, name, idx); + if (!(dac & DAC_SLAVE_FLAG)) { + caps = query_amp_caps(codec, dac, HDA_OUTPUT); + if (caps & AC_AMPCAP_NUM_STEPS) + return cx_auto_add_pb_volume(codec, dac, name, idx); + } caps = query_amp_caps(codec, pin, HDA_OUTPUT); if (caps & AC_AMPCAP_NUM_STEPS) return cx_auto_add_pb_volume(codec, pin, name, idx); @@ -4155,7 +4159,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) const char *label; int idx, type; hda_nid_t dac = spec->dac_info[i].dac; - if (!dac || (dac & DAC_SLAVE_FLAG)) + if (!dac) continue; type = spec->dac_info[i].type; if (type == AUTO_PIN_LINE_OUT) -- 1.7.4.1 --------------060508080506000304070004--

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[alsa-devel] How to handle special SPI read/write for ASoC codec driver?
by Ashish Chavan 25 Aug '11

25 Aug '11

Hi, I am working on updating ASoC codec driver for Dialog's DA7210 codec (sound/soc/codec/da7210.c). I am planning to add SPI support and have a query. DA7210 codec has total 136, 8 bit registers (0x88). It supports 8 bit address and 8 bit data format for SPI. One out of eight address bits is used to indicate read/write operation, so effectively there are only seven bits to represent address. So, directly we can only address 128 (0 to 127) registers. In order to make other upper registers (128 to 136) addressable, DA7210 provides a special PAGE1 register addressed at (0x00 / 0x80). This PAGE1 register basically acts as an offset to normal registers addresses. If PAGE1 register's value is "0x0" then, register address "0" actually maps to 0th register. But if page register's value is "0x80", then register address "0" actually maps to 128th register, register "2" maps to 129th reg, and so on. While adding support for SPI, I need to add this logic of setting correct page while reading and writing. AFAIK, the proper place to add this extra logic is read/write functions on codec. i.e. something similar to, @@ -208,6 +208,38 @@ static int da7210_write(struct snd_soc_codec *codec, u32 reg, u32 value) BUG_ON(codec->driver->volatile_register); +#if defined(CONFIG_SPI_MASTER) + /* + * spi_page member of private data stores current status of page + * register. 0 => page 0 selected, 1 => page 1 selected + */ + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + if(reg >= 0x80){ + /* We are writing register in page 1 */ + if(!da7210->spi_page){ + /* Need to set page register first */ + data[0] = 0x00; + data[1] = 0x80; /* need to set seventh bit */ + if (2 != codec->hw_write(codec->control_data, data, 2)) + return -EIO; + cache[0] = 0x80; /* Update cache */ + da7210->spi_page = 1; /* Update local status */ + } + + }else { + /* We are writing register in page 0 */ + if(da7210->spi_page){ + /* Need to reset page register first */ + data[0] = 0x80; + data[1] = 0x00; /* need to reset seventh bit */ + if (2 != codec->hw_write(codec->control_data, data, 2)) + return -EIO; + cache[0x80] = 0x00; /* Update cache */ + da7210->spi_page = 0; /* Update local status */ + } + } +#endif + data[0] = reg & 0xff; data[1] = value & 0xff; Any comments on this code? Is there any better place to put this extra logic? Any way to accommodate this in to generic soc-io functions? Thanks, -- Ashish

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[alsa-devel] How to handle special SPI read/write for ASoC codec driver?
by Ashish Chavan 25 Aug '11

25 Aug '11

Hi, I am working on updating ASoC codec driver for Dialog's DA7210 codec (sound/soc/codec/da7210.c). I am planning to add SPI support and have a query. DA7210 codec has total 136, 8 bit registers (0x88). It supports 8 bit address and 8 bit data format for SPI. One out of eight address bits is used to indicate read/write operation, so effectively there are only seven bits to represent address. So, directly we can only address 128 (0 to 127) registers. In order to make other upper registers (128 to 136) addressable, DA7210 provides a special PAGE1 register addressed at (0x00 / 0x80). This PAGE1 register basically acts as an offset to normal registers addresses. If PAGE1 register's value is "0x0" then, register address "0" actually maps to 0th register. But if page register's value is "0x80", then register address "0" actually maps to 128th register, register "2" maps to 129th reg, and so on. While adding support for SPI, I need to add this logic of setting correct page while reading and writing. AFAIK, the proper place to add this extra logic is read/write functions on codec. i.e. something similar to, @@ -208,6 +208,38 @@ static int da7210_write(struct snd_soc_codec *codec, u32 reg, u32 value) BUG_ON(codec->driver->volatile_register); +#if defined(CONFIG_SPI_MASTER) + /* + * spi_page member of private data stores current status of page + * register. 0 => page 0 selected, 1 => page 1 selected + */ + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + if(reg >= 80){ + /* We are writing register in page 1 */ + if(!da7210->spi_page){ + /* Need to set page register first */ + data[0] = 0x00; + data[1] = 0x80; /* need to set seventh bit */ + if (2 != codec->hw_write(codec->control_data, data, 2)) + return -EIO; + cache[0] = 0x80; /* Update cache */ + da7210->spi_page = 1; /* Update local status */ + } + + }else { + /* We are writing register in page 0 */ + if(da7210->spi_page){ + /* Need to reset page register first */ + data[0] = 0x80; + data[1] = 0x00; /* need to reset seventh bit */ + if (2 != codec->hw_write(codec->control_data, data, 2)) + return -EIO; + cache[80] = 0x00; /* Update cache */ + da7210->spi_page = 0; /* Update local status */ + } + } +#endif + data[0] = reg & 0xff; data[1] = value & 0xff; Any comments on this code? Is there any better place to put this extra logic? Any way to accommodate this in to generic soc-io functions? Thanks, -- Ashish

1 0

[alsa-devel] [PATCH] ASoC: MPC5200: replace of_device with platform_device
by Timur Tabi 25 Aug '11

25 Aug '11

'struct of_device' no longer exists, and its functionality has been merged into platform_device. Update the MPC5200 audio DMA driver (mpc5200_dma) accordingly. This fixes a build break. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/mpc5200_dma.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index fd0dc46..5c6c245 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -369,7 +369,7 @@ static struct snd_soc_platform_driver mpc5200_audio_dma_platform = { .pcm_free = &psc_dma_free, }; -static int mpc5200_hpcd_probe(struct of_device *op) +static int mpc5200_hpcd_probe(struct platform_device *op) { phys_addr_t fifo; struct psc_dma *psc_dma; @@ -487,7 +487,7 @@ out_unmap: return ret; } -static int mpc5200_hpcd_remove(struct of_device *op) +static int mpc5200_hpcd_remove(struct platform_device *op) { struct psc_dma *psc_dma = dev_get_drvdata(&op->dev); @@ -519,7 +519,7 @@ MODULE_DEVICE_TABLE(of, mpc5200_hpcd_match); static struct platform_driver mpc5200_hpcd_of_driver = { .probe = mpc5200_hpcd_probe, .remove = mpc5200_hpcd_remove, - .dev = { + .driver = { .owner = THIS_MODULE, .name = "mpc5200-pcm-audio", .of_match_table = mpc5200_hpcd_match, -- 1.7.3.4

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[alsa-devel] Conexant 5051 lenovo-x200 fixes
by Ulo Mets 24 Aug '11

24 Aug '11

Hi, I am not a kernel developer, but occasionally dig into it if something gets in my way. On Lenovo X200 Tablet none of the mixers showed Mic Boost adjustments, but HDA-Analyzer found them and adjusted properly. So I inserted the corresponding controls, and it worked. Then, during testing I found that the labels of External and Docking Mic are swapped, so I exchanged these too. Since I could not test it on other hardware, I put it all on lenovo-x200 branch. The patch applies to 3.1-rc2, I hope it is sufficiently recent. The Mic Switches don't have any effect on my unit (X200T without modem) With best regards, Ulo Mets Signed-off-by: Ulo Mets<umets at osa.pri.ee> ---b/sound/pci/hda/patch_conexant.c 2011-08-01 12:45:32.302640276 +0300 +++ a/sound/pci/hda/patch_conexant.c 2011-08-01 12:50:22.003141176 +0300 @@ -1785,6 +1785,19 @@ {} }; +static const struct snd_kcontrol_new cxt5051_lenovo_x200_mixers[] = { + HDA_CODEC_VOLUME("Int Mic Volume", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Switch", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Volume", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Dock Mic Switch", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Ext Mic Volume", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Ext Mic Switch", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x1d, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Boost", 0x17, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0x00, HDA_INPUT), + {} +}; + static const struct snd_kcontrol_new cxt5051_hp_mixers[] = { HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), @@ -2055,6 +2079,7 @@ break; case CXT5051_LENOVO_X200: spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; + spec->mixers[0] = cxt5051_lenovo_x200_mixers; /* Thinkpad X301 does not have S/PDIF wired and no ability to use a docking station. */ if (codec->subsystem_id == 0x17aa211f)

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[alsa-devel] [PATCH] ASoC: Correct element count for WM8996 sidetone HPF
by Mark Brown 24 Aug '11

24 Aug '11

I can count. Honest. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 73a1588..3c9080f 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -420,7 +420,7 @@ static const char *sidetone_hpf_text[] = { }; static const struct soc_enum sidetone_hpf = - SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 6, sidetone_hpf_text); + SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 7, sidetone_hpf_text); static const char *hpf_mode_text[] = { "HiFi", "Custom", "Voice" -- 1.7.5.4

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[alsa-devel] [PATCH 1/3] ALSA: core: trivial code style fix
by Lu Guanqun 24 Aug '11

24 Aug '11

remove trailing tab on the line. Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- sound/core/control.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index f8c5be4..7f2b3a7 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1072,7 +1072,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, long private_size; struct user_element *ue; int idx, err; - + if (card->user_ctl_count >= MAX_USER_CONTROLS) return -ENOMEM; if (info->count < 1)

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