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

[alsa-devel] [PATCH v4 3/7] ASoC: da7210: Add support for mute and zero cross controls
by Ashish Chavan 18 Oct '11

18 Oct '11

This patch adds support for below set of controls, (1) Mute controls for MIC, AUX and ADC (2) Zero cross controls for head phone, AUX, INPGA and line out (3) Head phone mode selection - class H or G It also adds digital_mute() call back. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v2: - Added digital_mute() call back and removed control for "DAC Mute Switch" - Renamed mute control switches Changes since v1: - Changed Mic mute control to a double control - Fixed a typo --- sound/soc/codecs/da7210.c | 45 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 45 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 480fc6d..cac4f26 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -30,6 +30,7 @@ #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX2 0x0B #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -41,6 +42,7 @@ #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_SOFTMUTE 0x18 #define DA7210_DAC_EQ1_2 0x19 #define DA7210_DAC_EQ3_4 0x1A #define DA7210_DAC_EQ5 0x1B @@ -49,6 +51,7 @@ #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 +#define DA7210_ZERO_CROSS 0x24 #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 @@ -144,6 +147,9 @@ #define DA7210_PLL_FS_96000 (0xF << 0) #define DA7210_PLL_EN (0x1 << 7) +/* SOFTMUTE bit fields */ +#define DA7210_RAMP_EN (1 << 6) + #define DA7210_VERSION "0.0.1" /* @@ -189,6 +195,13 @@ static const struct soc_enum da7210_dac_vf_cutoff = static const struct soc_enum da7210_adc_vf_cutoff = SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); +static const char *da7210_hp_mode_txt[] = { + "Class H", "Class G" +}; + +static const struct soc_enum da7210_hp_mode_sel = + SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -232,6 +245,22 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff), SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0), SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Capture Switch", DA7210_MIC_L, DA7210_MIC_R, 3, 1, 0), + SOC_SINGLE("Aux2 Capture Switch", DA7210_AUX2, 2, 1, 0), + SOC_SINGLE("ADC Left Capture Switch", DA7210_ADC, 2, 1, 0), + SOC_SINGLE("ADC Right Capture Switch", DA7210_ADC, 6, 1, 0), + SOC_SINGLE("Digital Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0), + SOC_SINGLE("Digital Soft Mute Rate", DA7210_SOFTMUTE, 0, 0x7, 0), + + /* Zero cross controls */ + SOC_DOUBLE("Aux1 ZC Switch", DA7210_ZERO_CROSS, 0, 1, 1, 0), + SOC_DOUBLE("In PGA ZC Switch", DA7210_ZERO_CROSS, 2, 3, 1, 0), + SOC_DOUBLE("Lineout ZC Switch", DA7210_ZERO_CROSS, 4, 5, 1, 0), + SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), + + SOC_ENUM("Headphone Class", da7210_hp_mode_sel), }; /* Codec private data */ @@ -448,6 +477,18 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return 0; } +static int da7210_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = snd_soc_read(codec, DA7210_DAC_HPF) & 0xFB; + + if (mute) + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg | 0x4); + else + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg); + return 0; +} + #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -456,6 +497,7 @@ static struct snd_soc_dai_ops da7210_dai_ops = { .startup = da7210_startup, .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .digital_mute = da7210_mute, }; static struct snd_soc_dai_driver da7210_dai = { @@ -545,6 +587,9 @@ static int da7210_probe(struct snd_soc_codec *codec) DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + /* Enable ramp mode for DAC gain update */ + snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

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Re: [alsa-devel] [PATCH] ALSA: hda/conexant - Add quirks for lenovo notebooks
by Takashi Iwai 18 Oct '11

18 Oct '11

At Tue, 18 Oct 2011 14:05:53 +0200, Torsten Hilbrich wrote: > > Am 18.10.2011 12:57, schrieb Takashi Iwai: > > At Tue, 18 Oct 2011 12:20:28 +0200, > > Torsten Hilbrich wrote: > >> > >> The lenovo sandy bridge notebooks (T420, T520, L520) didn't work with > >> the external sound input and output connectors on the dockig unit > >> (tested with P/N 45N5887). > >> > >> Loading snd-hda-intel with model=thinkpad solved the problem so I > >> added the accordings quirks to perform that decision automatically > >> for these specific models. > >> > >> Signed-off-by: Torsten Hilbrich <torsten.hilbrich(a)secunet.com> > > > > Could you check whether model=auto with the latest kernel work? > > Basically I won't take any more the model-specific quirk but > > move all to the auto-parser as much as possible. > > I just checked with upstream v3.1-rc10, model=auto didn't solve the > problem on the Lenovo T520. OK, so it's a bug of BIOS. BIOS doesn't provide the pin for docking-station. The right fix in this case is to provide the pin config table. In sound git tree, there is a fix for Lenovo X200. A similar entry would work for you. git://github.com/tiwai/sound.git Check either topic/hda, for-next or master branch. > Here are some dmesg output with model=auto: > > snd_hda_intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 > snd_hda_intel 0000:00:1b.0: irq 45 for MSI/MSI-X > snd_hda_intel 0000:00:1b.0: setting latency timer to 64 > hda-intel: chipset global capabilities = 0x4401 > hda-intel: Clearing TCSEL > hda-intel: HDA snoop disabled, enabling ... OK > hda-intel: codec_mask = 0xb > hda-intel: codec #0 probed OK > hda-intel: codec #1 probed OK > hda-intel: codec #3 probed OK > IBM TrackPoint firmware: 0x0e, buttons: 3/3 > hda_codec: model 'auto' is selected > hda_codec: CX20590: BIOS auto-probing. > autoconfig: line_outs=1 (0x1f/0x0/0x0/0x0/0x0) type:speaker > speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > hp_outs=1 (0x19/0x0/0x0/0x0/0x0) > mono: mono_out=0x0 > inputs: Mic=0x1b Internal Mic=0x23 > > when playing some audio: > > hda-intel: azx_pcm_prepare: bufsize=0x10000, format=0x31 > hda_codec_setup_stream: NID=0x10, stream=0x5, channel=0, format=0x31 > hda_codec_setup_stream: NID=0x11, stream=0x5, channel=0, format=0x31 > hda_codec_cleanup_stream: NID=0x10 > hda_codec_cleanup_stream: NID=0x11 > > I will attach the codec#x files from /proc/asound/PCH if you need this > information. At best, please give alsa-info.sh output (run with --no-upload option). thanks, Takashi

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[alsa-devel] ASoC: Amplifier behind codec?
by Wolfram Sang 18 Oct '11

18 Oct '11

Hi, I have here an amplifier (MAX9768) between codec output and the speaker, all controlled via I2C. I noticed that other amplifier drivers export a function called *_add_controls(), so the platform glue code can add the amplifier controls to the codec. Now, all drivers working like this use a single static instance of the i2c client, because there doesn't seem to be a connection between the added control elements and the additional i2c client. Is the above approach still the preferred way of doing this? Thanks, Wolfram -- Pengutronix e.K. | Wolfram Sang | Industrial Linux Solutions | http://www.pengutronix.de/ |

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[PATCH] ALSA: HDA: Add new revision for ALC662
by David Henningsson 18 Oct '11

18 Oct '11

The revision 0x100300 was found for ALC662. It seems to work well with patch_alc662. Cc: stable(a)kernel.org BugLink: http://bugs.launchpad.net/bugs/877373 Tested-by: Shengyao Xue <Shengyao.xue(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f9d24c3..86ecc72 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5774,6 +5774,8 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { .patch = patch_alc882 }, { .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1", .patch = patch_alc662 }, + { .id = 0x10ec0662, .rev = 0x100300, .name = "ALC662 rev3", + .patch = patch_alc662 }, { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, -- 1.7.5.4 --------------070301030905080206060603--

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Re: [alsa-devel] [PATCH] ALSA: hda/conexant - Add quirks for lenovo notebooks
by Takashi Iwai 18 Oct '11

18 Oct '11

At Tue, 18 Oct 2011 12:20:28 +0200, Torsten Hilbrich wrote: > > The lenovo sandy bridge notebooks (T420, T520, L520) didn't work with > the external sound input and output connectors on the dockig unit > (tested with P/N 45N5887). > > Loading snd-hda-intel with model=thinkpad solved the problem so I > added the accordings quirks to perform that decision automatically > for these specific models. > > Signed-off-by: Torsten Hilbrich <torsten.hilbrich(a)secunet.com> Could you check whether model=auto with the latest kernel work? Basically I won't take any more the model-specific quirk but move all to the auto-parser as much as possible. thanks, Takashi > --- > sound/pci/hda/patch_conexant.c | 3 +++ > 1 files changed, 3 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index 502fc94..90a6b98 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -3110,8 +3110,11 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { > SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), > SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), > SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), > + SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT5066_THINKPAD), > + SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT5066_THINKPAD), > SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), > SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), > + SND_PCI_QUIRK(0x17aa, 0x21de, "Lenovo L520", CXT5066_THINKPAD), > SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), > SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), > SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), > -- > 1.7.5.4 >

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Re: [alsa-devel] [REGRESSION] Sound goes too fast due to 798cb7e897210
by Takashi Iwai 18 Oct '11

18 Oct '11

At Tue, 18 Oct 2011 10:10:14 +0200, Éric Piel wrote: > > Dear Takashi, > > I have a regression here on 3.1-rc9+ causing the sound "going too fast" > with hda. Reverting 798cb7e897210 (ALSA: hda - Fix a regression of the > position-buffer check) fixes this regression for me. > > Of course, as this commit is itself a regression fix, my goat feeling is > that it's not going to be possible to directly revert it. Attached if > the output of alsa-info.sh (on the fixed kernel). Please let me know if > I can provide you any additional information or test any patch. If that commit affects, the best fix would be to give a quirk specific to your device. Try to pass either position_fix=1 or position_fix=2. Only one of them should work (likely 2). After checking it, you can add it to position_fix_list[] in sound/pci/hda/hda_intel.c together with PCI SSID. thanks, Takashi

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[alsa-devel] [PATCH] ASoC: Convert WM9081 to table based control init
by Mark Brown 18 Oct '11

18 Oct '11

At least for the core controls, the optionally selected controls are still added programatically. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index f4fc03b..93f794c 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1235,8 +1235,6 @@ static int wm9081_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM9081_ANALOGUE_SPEAKER_PGA, WM9081_SPKPGAZC, WM9081_SPKPGAZC); - snd_soc_add_controls(codec, wm9081_snd_controls, - ARRAY_SIZE(wm9081_snd_controls)); if (!wm9081->pdata.num_retune_configs) { dev_dbg(codec->dev, "No ReTune Mobile data, using normal EQ\n"); @@ -1296,6 +1294,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .reg_cache_default = wm9081_reg_defaults, .volatile_register = wm9081_volatile_register, + .controls = wm9081_snd_controls, + .num_controls = ARRAY_SIZE(wm9081_snd_controls), .dapm_widgets = wm9081_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm9081_dapm_widgets), .dapm_routes = wm9081_audio_paths, -- 1.7.6.3

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[alsa-devel] [PATCH] ASoC: Update WM1811 DCS codes for latest evaluation results
by Mark Brown 18 Oct '11

18 Oct '11

Evaluation of larger quantities of material has provided new DCS codes values to be applied for WM1811. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index df55fa3..044350a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3241,8 +3241,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (wm8994->revision) { case 0: case 1: - wm8994->hubs.dcs_codes_l = -7; - wm8994->hubs.dcs_codes_r = -4; + wm8994->hubs.dcs_codes_l = -9; + wm8994->hubs.dcs_codes_r = -5; break; default: break; -- 1.7.6.3

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[alsa-devel] [PATCH 2/3] sound/soc/au1x: keep pointer to resource so it can be freed
by Julia Lawall 18 Oct '11

18 Oct '11

From: Julia Lawall <julia(a)diku.dk> Add a new variable for storing resources accessed subsequent to the one accessed using request_mem_region, so the one accessed using request_mem_region can be released if needed. The semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @r@ expression E, E1; identifier f; statement S1,S2,S3; @@ if (E == NULL) { ... when != if (E == NULL || ...) S1 else S2 when != E = E1 *E->f ... when any return ...; } else S3 // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/au1x/psc-ac97.c | 14 +++++++------- sound/soc/au1x/psc-i2s.c | 14 +++++++------- 2 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 172eefd..62e5640 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -364,7 +364,7 @@ static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = { static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) { int ret; - struct resource *r; + struct resource *r, *r1; unsigned long sel; struct au1xpsc_audio_data *wd; @@ -388,15 +388,15 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) if (!wd->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) + r1 = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r1) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r1->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) + r1 = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r1) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r1->start; /* configuration: max dma trigger threshold, enable ac97 */ wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 | diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 7c5ae92..621e260 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -290,7 +290,7 @@ static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = { static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) { - struct resource *r; + struct resource *r, *r1; unsigned long sel; int ret; struct au1xpsc_audio_data *wd; @@ -313,15 +313,15 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) if (!wd->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) + r1 = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r1) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r1->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) + r1 = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r1) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r1->start; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer)

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[alsa-devel] [PATCH 1/2] ASoC: ad193x: Fix define of AD193X_PLL_INPUT_MASK
by Axel Lin 18 Oct '11

18 Oct '11

Current code defines AD193X_PLL_INPUT_MASK as (~0x6) which is quite different from other MASK defines. To make it consistent with other mask defines, define AD193X_PLL_INPUT_MASK as 0x6 and change the code accordingly. I think this change improves the readability. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ad193x.c | 2 +- sound/soc/codecs/ad193x.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index f934670..39056ce 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -298,7 +298,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, } reg = snd_soc_read(codec, AD193X_PLL_CLK_CTRL0); - reg = (reg & AD193X_PLL_INPUT_MASK) | master_rate; + reg = (reg & (~AD193X_PLL_INPUT_MASK)) | master_rate; snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg); reg = snd_soc_read(codec, AD193X_DAC_CTRL2); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index 536e5f2..1507eaa 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -11,7 +11,7 @@ #define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 -#define AD193X_PLL_INPUT_MASK (~0x6) +#define AD193X_PLL_INPUT_MASK 0x6 #define AD193X_PLL_INPUT_256 (0 << 1) #define AD193X_PLL_INPUT_384 (1 << 1) #define AD193X_PLL_INPUT_512 (2 << 1) -- 1.7.4.1

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