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

[alsa-devel] ASoC updates for 3.2
by Mark Brown 22 Oct '11

22 Oct '11

The following changes since commit 24441e17feb57668e4c5013750cef549bf3c4861: ASoC: ak4641: Use SND_SOC_DAPM_DAC for Voice Playback stream widget (2011-10-19 18:01:43 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 Ashish Chavan (2): ASoC: da7210: Add support for DAPM ASoC: da7210: Add support for line out and DAC Axel Lin (10): ASoC: wm9081: Fix setting soft VMID ramp enable with VMID 2*240k ASoC: Fix reg_word_size for ak4104 ASoC: ak4671: Use snd_soc_update_bits for read-modify-write ASoC: rt5631: Remove unused codec field from struct rt5631_priv ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture ASoC: Set sgtl5000->ldo in ldo_regulator_register ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2 ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls Julia Lawall (1): ASoC: keep pointer to resource so it can be freed sound/soc/au1x/ac97c.c | 33 +++--- sound/soc/au1x/i2sc.c | 33 +++--- sound/soc/au1x/psc-ac97.c | 25 +++-- sound/soc/au1x/psc-i2s.c | 25 +++-- sound/soc/codecs/ak4104.c | 2 +- sound/soc/codecs/ak4671.c | 18 +-- sound/soc/codecs/da7210.c | 241 ++++++++++++++++++++++++++++++++++++++----- sound/soc/codecs/rt5631.c | 2 - sound/soc/codecs/sgtl5000.c | 20 ++-- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8996.c | 6 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/mxs/mxs-saif.c | 19 ++-- 13 files changed, 312 insertions(+), 116 deletions(-)

2 1

[alsa-devel] [PATCH v2] ASoC: max98095: Convert codec->hw_write to snd_soc_write
by Axel Lin 22 Oct '11

22 Oct '11

codec->hw_write is broken now, convert codec->hw_write to snd_soc_write. The hardware has 2 banks of registers sharing a section in I2C register space. The 1st bank is the primary one and is cached. The 2nd bank is for loading coefficients only and they do not need cache. These coefficients registers are therefore direct writes. Thus we set cache_bypass flag to deal with this before calling snd_soc_write. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Hi, I don't have this hardware, I'd appreciate if someone can test this patch. Thanks, Axel sound/soc/codecs/max98095.c | 13 ++++++------- 1 files changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 6982f74..26d7b08 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -617,14 +617,13 @@ static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) static int max98095_hw_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u8 data[2]; + int ret; - data[0] = reg; - data[1] = value; - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; + codec->cache_bypass = 1; + ret = snd_soc_write(codec, reg, value); + codec->cache_bypass = 0; + + return ret ? -EIO : 0; } /* -- 1.7.4.1

2 1

[alsa-devel] [PATCH] sound/soc: keep pointer to resource so it can be freed
by Julia Lawall 22 Oct '11

22 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 resource variable names are also changed to be more descriptive. This code is also missing some calls to iounmap. 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/ac97c.c | 33 ++++++++++++++++++--------------- sound/soc/au1x/i2sc.c | 33 ++++++++++++++++++--------------- sound/soc/au1x/psc-ac97.c | 25 +++++++++++++------------ sound/soc/au1x/psc-i2s.c | 25 +++++++++++++------------ sound/soc/mxs/mxs-saif.c | 19 ++++++++++--------- 5 files changed, 72 insertions(+), 63 deletions(-) diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index 13802ff..726bd65 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -226,7 +226,7 @@ static struct snd_soc_dai_driver au1xac97c_dai_driver = { static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) { int ret; - struct resource *r; + struct resource *iores, *dmares; struct au1xpsc_audio_data *ctx; ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); @@ -235,29 +235,30 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) mutex_init(&ctx->lock); - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; goto out0; } ret = -EBUSY; - if (!request_mem_region(r->start, resource_size(r), pdev->name)) + if (!request_mem_region(iores->start, resource_size(iores), + pdev->name)) goto out0; - ctx->mmio = ioremap_nocache(r->start, resource_size(r)); + ctx->mmio = ioremap_nocache(iores->start, resource_size(iores)); if (!ctx->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; /* switch it on */ WR(ctx, AC97_ENABLE, EN_D | EN_CE); @@ -270,13 +271,15 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &au1xac97c_dai_driver); if (ret) - goto out1; + goto out2; ac97c_workdata = ctx; return 0; +out2: + iounmap(ctx->mmio); out1: - release_mem_region(r->start, resource_size(r)); + release_mem_region(iores->start, resource_size(iores)); out0: kfree(ctx); return ret; diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 19e0d2a..6bcf48f 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -228,47 +228,50 @@ static struct snd_soc_dai_driver au1xi2s_dai_driver = { static int __devinit au1xi2s_drvprobe(struct platform_device *pdev) { int ret; - struct resource *r; + struct resource *iores, *dmares; struct au1xpsc_audio_data *ctx; ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; goto out0; } ret = -EBUSY; - if (!request_mem_region(r->start, resource_size(r), pdev->name)) + if (!request_mem_region(iores->start, resource_size(iores), + pdev->name)) goto out0; - ctx->mmio = ioremap_nocache(r->start, resource_size(r)); + ctx->mmio = ioremap_nocache(iores->start, resource_size(iores)); if (!ctx->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; platform_set_drvdata(pdev, ctx); ret = snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver); if (ret) - goto out1; + goto out2; return 0; +out2: + iounmap(ctx->mmio); out1: - release_mem_region(r->start, resource_size(r)); + release_mem_region(iores->start, resource_size(iores)); out0: kfree(ctx); return ret; diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 172eefd..cab74ab 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 *iores, *dmares; unsigned long sel; struct au1xpsc_audio_data *wd; @@ -374,29 +374,30 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) mutex_init(&wd->lock); - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; goto out0; } ret = -EBUSY; - if (!request_mem_region(r->start, resource_size(r), pdev->name)) + if (!request_mem_region(iores->start, resource_size(iores), + pdev->name)) goto out0; - wd->mmio = ioremap(r->start, resource_size(r)); + wd->mmio = ioremap(iores->start, resource_size(iores)); if (!wd->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; /* configuration: max dma trigger threshold, enable ac97 */ wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 | @@ -428,7 +429,7 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) out2: iounmap(wd->mmio); out1: - release_mem_region(r->start, resource_size(r)); + release_mem_region(iores->start, resource_size(iores)); out0: kfree(wd); return ret; diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 7c5ae92..d792bc0 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 *iores, *dmares; unsigned long sel; int ret; struct au1xpsc_audio_data *wd; @@ -299,29 +299,30 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) if (!wd) return -ENOMEM; - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; goto out0; } ret = -EBUSY; - if (!request_mem_region(r->start, resource_size(r), pdev->name)) + if (!request_mem_region(iores->start, resource_size(iores), + pdev->name)) goto out0; - wd->mmio = ioremap(r->start, resource_size(r)); + wd->mmio = ioremap(iores->start, resource_size(iores)); if (!wd->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) @@ -355,7 +356,7 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) out2: iounmap(wd->mmio); out1: - release_mem_region(r->start, resource_size(r)); + release_mem_region(iores->start, resource_size(iores)); out0: kfree(wd); return ret; diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 401944c..76dc74d 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -617,7 +617,7 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id) static int mxs_saif_probe(struct platform_device *pdev) { - struct resource *res; + struct resource *iores, *dmares; struct mxs_saif *saif; struct mxs_saif_platform_data *pdata; int ret = 0; @@ -655,35 +655,36 @@ static int mxs_saif_probe(struct platform_device *pdev) goto failed_clk; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; dev_err(&pdev->dev, "failed to get io resource: %d\n", ret); goto failed_get_resource; } - if (!request_mem_region(res->start, resource_size(res), "mxs-saif")) { + if (!request_mem_region(iores->start, resource_size(iores), + "mxs-saif")) { dev_err(&pdev->dev, "request_mem_region failed\n"); ret = -EBUSY; goto failed_get_resource; } - saif->base = ioremap(res->start, resource_size(res)); + saif->base = ioremap(iores->start, resource_size(iores)); if (!saif->base) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENODEV; goto failed_ioremap; } - res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) { ret = -ENODEV; dev_err(&pdev->dev, "failed to get dma resource: %d\n", ret); goto failed_ioremap; } - saif->dma_param.chan_num = res->start; + saif->dma_param.chan_num = dmares->start; saif->irq = platform_get_irq(pdev, 0); if (saif->irq < 0) { @@ -742,7 +743,7 @@ failed_get_irq2: failed_get_irq1: iounmap(saif->base); failed_ioremap: - release_mem_region(res->start, resource_size(res)); + release_mem_region(iores->start, resource_size(iores)); failed_get_resource: clk_put(saif->clk); failed_clk:

2 1

[alsa-devel] [PATCH] ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2
by Axel Lin 22 Oct '11

22 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.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 1d7b174..645c980 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1920,7 +1920,7 @@ static int wm8996_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, lrclk_reg, WM8996_AIF1RX_RATE_MASK, lrclk); snd_soc_update_bits(codec, WM8996_AIF_CLOCKING_2, - WM8996_DSP1_DIV_SHIFT << dsp_shift, dsp); + WM8996_DSP1_DIV_MASK << dsp_shift, dsp); return 0; } -- 1.7.5.4

3 2

[alsa-devel] [PATCH v8 2/3] ASoC: da7210: Add support for line out and DAC
by Ashish Chavan 22 Oct '11

22 Oct '11

DA7210 has three line outputs. OUT1 Left, OUT1 Right and OUT2 (mono). This patch adds support for gain controls for these three line outs. It also adds support for overall DAC gain control. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v7: - Added DAPM support for OUT1 and OUT2 (Moved here from DAPM patch) - Added comment explaining rational behind usage of STANDBY bits Changes since v1: - Removed explicit setting of default gains - Renamed "DAC Playback" to "Digital Playback" --- sound/soc/codecs/da7210.c | 96 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 96 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 7a4b952..0ebcbd5 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -54,6 +54,9 @@ #define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT1_L 0x1E +#define DA7210_OUT1_R 0x1F +#define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 @@ -186,6 +189,17 @@ #define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ #define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ +/* OUT1_L bit fields */ +#define DA7210_OUT1_L_EN (1 << 7) + +/* OUT1_R bit fields */ +#define DA7210_OUT1_R_EN (1 << 7) + +/* OUT2 bit fields */ +#define DA7210_OUT2_OUTMIX_R (1 << 5) +#define DA7210_OUT2_OUTMIX_L (1 << 6) +#define DA7210_OUT2_EN (1 << 7) + #define DA7210_VERSION "0.0.1" /* @@ -206,8 +220,23 @@ static const unsigned int hp_out_tlv[] = { 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), }; +static const unsigned int lineout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int mono_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x2, TLV_DB_SCALE_ITEM(-1800, 0, 1), + /* -18dB to 6dB */ + 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) +}; + static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); +static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); /* ADC and DAC high pass filter f0 value */ static const char const *da7210_hpf_cutoff_txt[] = { @@ -306,6 +335,14 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", DA7210_HP_L_VOL, DA7210_HP_R_VOL, 0, 0x3F, 0, hp_out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume", + DA7210_DAC_L, DA7210_DAC_R, + 0, 0x77, 1, dac_gain_tlv), + SOC_DOUBLE_R_TLV("Lineout Playback Volume", + DA7210_OUT1_L, DA7210_OUT1_R, + 0, 0x3f, 0, lineout_vol_tlv), + SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, + mono_vol_tlv), /* DAC Equalizer controls */ SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), @@ -402,6 +439,12 @@ static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), }; +/* Mono Mixer */ +static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), +}; + /* DAPM widgets */ static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { /* Input Side */ @@ -443,16 +486,26 @@ static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { &da7210_dapm_outmixr_controls[0], ARRAY_SIZE(da7210_dapm_outmixr_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, + &da7210_dapm_monomix_controls[0], + ARRAY_SIZE(da7210_dapm_monomix_controls)), + /* Output PGAs */ SND_SOC_DAPM_PGA("OUTPGA Left Enable", DA7210_OUTMIX_L, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("OUTPGA Right Enable", DA7210_OUTMIX_R, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Out1 Left", DA7210_STARTUP2, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out1 Right", DA7210_STARTUP2, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out2 Mono", DA7210_STARTUP2, 2, 1, NULL, 0), SND_SOC_DAPM_PGA("Headphone Left", DA7210_STARTUP2, 3, 1, NULL, 0), SND_SOC_DAPM_PGA("Headphone Right", DA7210_STARTUP2, 4, 1, NULL, 0), /* Output Lines */ + SND_SOC_DAPM_OUTPUT("OUT1L"), + SND_SOC_DAPM_OUTPUT("OUT1R"), SND_SOC_DAPM_OUTPUT("HPL"), SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("OUT2"), }; /* DAPM audio route definition */ @@ -478,14 +531,26 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, + {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, + {"OUTPGA Left Enable", NULL, "Out Mixer Left"}, {"OUTPGA Right Enable", NULL, "Out Mixer Right"}, + {"Out1 Left", NULL, "OUTPGA Left Enable"}, + {"OUT1L", NULL, "Out1 Left"}, + + {"Out1 Right", NULL, "OUTPGA Right Enable"}, + {"OUT1R", NULL, "Out1 Right"}, + {"Headphone Left", NULL, "OUTPGA Left Enable"}, {"HPL", NULL, "Headphone Left"}, {"Headphone Right", NULL, "OUTPGA Right Enable"}, {"HPR", NULL, "Headphone Right"}, + + {"Out2 Mono", NULL, "Mono Mixer"}, + {"OUT2", NULL, "Out2 Mono"}, }; /* Codec private data */ @@ -791,6 +856,37 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable ramp mode for DAC gain update */ snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + /* + * For DA7210 codec, there are two ways to enable/disable analog IOs + * and ADC/DAC, + * (1) Using "Enable Bit" of register associated with that IO + * (or ADC/DAC) + * e.g. Mic Left can be enabled using bit 7 of MIC_L(0x7) reg + * + * (2) Using "Standby Bit" of STARTUP2 or STARTUP3 register + * e.g. Mic left can be put to STANDBY using bit 0 of STARTUP3(0x5) + * + * Out of these two methods, the one using STANDBY bits is preferred + * way to enable/disable individual blocks. This is because STANDBY + * registers are part of system controller which allows system power + * up/down in a controlled, pop-free manner. Also, as per application + * note of DA7210, STANDBY register bits are only effective if a + * particular IO (or ADC/DAC) is already enabled using enable/disable + * register bits. Keeping these things in mind, current DAPM + * implementation manipulates only STANDBY bits. + * + * Overall implementation can be outlined as below, + * + * - "Enable bit" of an IO or ADC/DAC is used to enable it in probe() + * - "STANDBY bit" is controlled by DAPM + */ + + /* Enable Line out amplifiers */ + snd_soc_write(codec, DA7210_OUT1_L, DA7210_OUT1_L_EN); + snd_soc_write(codec, DA7210_OUT1_R, DA7210_OUT1_R_EN); + snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | + DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

3 2

[alsa-devel] [PATCH v8 1/3] ASoC: da7210: Add support for DAPM
by Ashish Chavan 22 Oct '11

22 Oct '11

This patch adds support for DAPM covering all inputs and outputs as well as ADC and DAC. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v7: - Re-factored the patch to only include DAPM for existing IOs. (i.e. removed DAPM support for AUX, line outs etc. This will be added via other patches which add actual support for those IOs) Changes since v3: - Added missing routes for In-mixer left and right - Added comment explaining rational behind use of STANDBY registers Changes since v1: - used supply widget for mic bias - removed mic bias hookup - assigned dapm widgets and routes in card struct - removed explicit setting of default mixer settings - removed explicit setting of ios to standby mode --- sound/soc/codecs/da7210.c | 145 +++++++++++++++++++++++++++++++++++++-------- 1 files changed, 121 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index e9ee6a4..7a4b952 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -29,6 +29,8 @@ #define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 +#define DA7210_STARTUP2 0x04 +#define DA7210_STARTUP3 0x05 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 #define DA7210_AUX1_L 0x09 @@ -372,6 +374,120 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), }; +/* + * DAPM Controls + * + * Current DAPM implementation covers almost all codec components e.g. IOs, + * mixers, PGAs,ADC and DAC. + */ +/* In Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), +}; + +/* Out Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + + SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixl_controls[0], + ARRAY_SIZE(da7210_dapm_inmixl_controls)), + + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixr_controls[0], + ARRAY_SIZE(da7210_dapm_inmixr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA7210_STARTUP3, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA7210_STARTUP3, 6, 1), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA7210_STARTUP2, 5, 1), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA7210_STARTUP2, 6, 1), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixl_controls[0], + ARRAY_SIZE(da7210_dapm_outmixl_controls)), + + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixr_controls[0], + ARRAY_SIZE(da7210_dapm_outmixr_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("OUTPGA Left Enable", DA7210_OUTMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTPGA Right Enable", DA7210_OUTMIX_R, 7, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Headphone Left", DA7210_STARTUP2, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right", DA7210_STARTUP2, 4, 1, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route da7210_audio_map[] = { + /* Dest Connecting Widget source */ + /* Input path */ + {"Mic Left", NULL, "MICL"}, + {"Mic Right", NULL, "MICR"}, + + {"In Mixer Left", "Mic Left Switch", "Mic Left"}, + {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + + {"In Mixer Right", "Mic Right Switch", "Mic Right"}, + {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + + {"INPGA Left", NULL, "In Mixer Left"}, + {"ADC Left", NULL, "INPGA Left"}, + + {"INPGA Right", NULL, "In Mixer Right"}, + {"ADC Right", NULL, "INPGA Right"}, + + /* Output path */ + {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + + {"OUTPGA Left Enable", NULL, "Out Mixer Left"}, + {"OUTPGA Right Enable", NULL, "Out Mixer Right"}, + + {"Headphone Left", NULL, "OUTPGA Left Enable"}, + {"HPL", NULL, "Headphone Left"}, + + {"Headphone Right", NULL, "OUTPGA Right Enable"}, + {"HPR", NULL, "Headphone Right"}, +}; + /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; @@ -411,29 +527,6 @@ static int da7210_volatile_register(struct snd_soc_codec *codec, return 0; } } -static int da7210_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - struct snd_soc_codec *codec = dai->codec; - - if (is_play) { - /* Enable Out */ - snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); - snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); - - } else { - /* Volume 7 */ - snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7); - snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7); - - /* Enable Mic */ - snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1); - snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1); - } - - return 0; -} /* * Set PCM DAI word length. @@ -603,7 +696,6 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute) /* DAI operations */ 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, @@ -742,6 +834,11 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .controls = da7210_snd_controls, .num_controls = ARRAY_SIZE(da7210_snd_controls), + + .dapm_widgets = da7210_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da7210_dapm_widgets), + .dapm_routes = da7210_audio_map, + .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls
by Axel Lin 22 Oct '11

22 Oct '11

Ensure all mask bits are clear before setting new value. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- I don't have this hardware, I'd appreciate if someone can test this patch. Thanks, Axel sound/soc/codecs/sgtl5000.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 32b5bbd..d15695d 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -723,7 +723,9 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, i2s_ctl, i2s_ctl); + snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, + SGTL5000_I2S_DLEN_MASK | SGTL5000_I2S_SCLKFREQ_MASK, + i2s_ctl); return 0; } @@ -1146,8 +1148,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) vag = (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP; snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, - vag << SGTL5000_ANA_GND_SHIFT, - vag << SGTL5000_ANA_GND_SHIFT); + SGTL5000_ANA_GND_MASK, vag << SGTL5000_ANA_GND_SHIFT); /* set line out VAG to vddio / 2, in range (0.8v, 1.675v) */ vag = vddio / 2; @@ -1161,9 +1162,8 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) SGTL5000_LINE_OUT_GND_STP; snd_soc_update_bits(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - vag << SGTL5000_LINE_OUT_GND_SHIFT | - SGTL5000_LINE_OUT_CURRENT_360u << - SGTL5000_LINE_OUT_CURRENT_SHIFT, + SGTL5000_LINE_OUT_CURRENT_MASK | + SGTL5000_LINE_OUT_GND_MASK, vag << SGTL5000_LINE_OUT_GND_SHIFT | SGTL5000_LINE_OUT_CURRENT_360u << SGTL5000_LINE_OUT_CURRENT_SHIFT); -- 1.7.5.4

4 3

Re: [alsa-devel] [PATCH v5] ASoC:Add support for cs42l73 codec
by Mark Brown 21 Oct '11

21 Oct '11

On Thu, Oct 20, 2011 at 09:31:50PM +0300, Georgi Vlaev wrote: > Hello, Don't top post. > The codec has 2 basic groups of power control bits: You're describing some features that don't sound at all unusual here... I'm not sure what point you're trying to make here?

3 5

[alsa-devel] [PATCH v7 2/3] ASoC: da7210: Add support for line out and DAC
by Ashish Chavan 21 Oct '11

21 Oct '11

DA7210 has three line outputs. OUT1 Left, OUT1 Right and OUT2 (mono). This patch adds support for gain controls for these three line outs. It also adds support for overall DAC gain control. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - Removed explicit setting of default gains - Renamed "DAC Playback" to "Digital Playback" --- sound/soc/codecs/da7210.c | 42 ++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 42 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 2187db9..eaec60a 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -54,6 +54,8 @@ #define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT1_L 0x1E +#define DA7210_OUT1_R 0x1F #define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 @@ -187,6 +189,17 @@ #define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ #define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ +/* OUT1_L bit fields */ +#define DA7210_OUT1_L_EN (1 << 7) + +/* OUT1_R bit fields */ +#define DA7210_OUT1_R_EN (1 << 7) + +/* OUT2 bit fields */ +#define DA7210_OUT2_OUTMIX_R (1 << 5) +#define DA7210_OUT2_OUTMIX_L (1 << 6) +#define DA7210_OUT2_EN (1 << 7) + #define DA7210_VERSION "0.0.1" /* @@ -207,8 +220,23 @@ static const unsigned int hp_out_tlv[] = { 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), }; +static const unsigned int lineout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int mono_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x2, TLV_DB_SCALE_ITEM(-1800, 0, 1), + /* -18dB to 6dB */ + 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) +}; + static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); +static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); /* ADC and DAC high pass filter f0 value */ static const char const *da7210_hpf_cutoff_txt[] = { @@ -307,6 +335,14 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", DA7210_HP_L_VOL, DA7210_HP_R_VOL, 0, 0x3F, 0, hp_out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume", + DA7210_DAC_L, DA7210_DAC_R, + 0, 0x77, 1, dac_gain_tlv), + SOC_DOUBLE_R_TLV("Lineout Playback Volume", + DA7210_OUT1_L, DA7210_OUT1_R, + 0, 0x3f, 0, lineout_vol_tlv), + SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, + mono_vol_tlv), /* DAC Equalizer controls */ SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), @@ -886,6 +922,12 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable ramp mode for DAC gain update */ snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + /* Enable Line out amplifiers */ + snd_soc_write(codec, DA7210_OUT1_L, DA7210_OUT1_L_EN); + snd_soc_write(codec, DA7210_OUT1_R, DA7210_OUT1_R_EN); + snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | + DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

3 8

[alsa-devel] [PATCH v7 1/3] ASoC: da7210: Add support for DAPM
by Ashish Chavan 21 Oct '11

21 Oct '11

This patch adds support for DAPM covering all inputs and outputs as well as ADC and DAC. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v3: - Added missing routes for In-mixer left and right - Added comment explaining rational behind use of STANDBY registers Changes since v1: - used supply widget for mic bias - removed mic bias hookup - assigned dapm widgets and routes in card struct - removed explicit setting of default mixer settings - removed explicit setting of ios to standby mode --- sound/soc/codecs/da7210.c | 240 ++++++++++++++++++++++++++++++++++++++++----- 1 files changed, 216 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index f0a1999..2187db9 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -29,6 +29,8 @@ #define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 +#define DA7210_STARTUP2 0x04 +#define DA7210_STARTUP3 0x05 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 #define DA7210_AUX1_L 0x09 @@ -52,6 +54,7 @@ #define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 @@ -372,6 +375,214 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), }; +/* + * DAPM Controls + * + * Current DAPM implementation covers almost all codec components e.g. IOs, + * mixers, PGAs,ADC and DAC. + * + * For DA7210 codec, there are two ways to enable/disable analog IOs and + * ADC/DAC, + * (1) Using "Enable Bit" of register associated with that IO(or ADC/DAC) + * e.g. Mic Left can be enabled using bit 7 of MIC_L(0x7) reg + * + * (2) Using "Standby Bit" of STARTUP2 or STARTUP3 register + * e.g. Mic left can be put to STANDBY using bit 0 of STARTUP3(0x5) reg + * + * Out of these two methods, the one using STANDBY bits is preferred way to + * enable/disable individual blocks. This is because STANDBY registers are part + * of system controller which allows system power up/down in a controlled, + * pop-free manner. Also, as per application note of DA7210, STANDBY register + * bits are only effective if a particular IO (or ADC/DAC) is already enabled + * using enable/disable register bits. Keeping these things in mind, current + * DAPM implementation manipulates only STANDBY bits. + * + * Overall implementation can be outlined as below, + * + * - "Enable bit" of an IO or ADC/DAC is used to enable it in probe() function + * - "STANDBY bit" is controlled by DAPM + */ +/* In Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_INMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_INMIX_L, 4, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_INMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_INMIX_R, 4, 1, 0), +}; + +/* Out Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_OUTMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_OUTMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), +}; + +/* Mono Mixer */ +static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUT2, 3, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUT2, 4, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("AUX1L"), + SND_SOC_DAPM_INPUT("AUX1R"), + SND_SOC_DAPM_INPUT("AUX2"), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Left", DA7210_STARTUP3, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Right", DA7210_STARTUP3, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux2 Mono", DA7210_STARTUP3, 4, 1, NULL, 0), + + SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + + /* MICBIAS */ + SND_SOC_DAPM_SUPPLY("Mic Bias", DA7210_MIC_L, 6, 0, NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixl_controls[0], + ARRAY_SIZE(da7210_dapm_inmixl_controls)), + + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixr_controls[0], + ARRAY_SIZE(da7210_dapm_inmixr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA7210_STARTUP3, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA7210_STARTUP3, 6, 1), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA7210_STARTUP2, 5, 1), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA7210_STARTUP2, 6, 1), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixl_controls[0], + ARRAY_SIZE(da7210_dapm_outmixl_controls)), + + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixr_controls[0], + ARRAY_SIZE(da7210_dapm_outmixr_controls)), + + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, + &da7210_dapm_monomix_controls[0], + ARRAY_SIZE(da7210_dapm_monomix_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("OUTPGA Left Enable", DA7210_OUTMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTPGA Right Enable", DA7210_OUTMIX_R, 7, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Out1 Left", DA7210_STARTUP2, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out1 Right", DA7210_STARTUP2, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out2 Mono", DA7210_STARTUP2, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left", DA7210_STARTUP2, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right", DA7210_STARTUP2, 4, 1, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("OUT1L"), + SND_SOC_DAPM_OUTPUT("OUT1R"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("OUT2"), +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route da7210_audio_map[] = { + /* Dest Connecting Widget source */ + /* Input path */ + {"Mic Left", NULL, "MICL"}, + {"Mic Right", NULL, "MICR"}, + {"Aux1 Left", NULL, "AUX1L"}, + {"Aux1 Right", NULL, "AUX1R"}, + {"Aux2 Mono", NULL, "AUX2"}, + + {"In Mixer Left", "Mic Left Switch", "Mic Left"}, + {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + {"In Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"In Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Left", "Outmix Left Switch", "Out Mixer Left"}, + + {"In Mixer Right", "Mic Right Switch", "Mic Right"}, + {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + {"In Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"In Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Right", "Outmix Right Switch", "Out Mixer Right"}, + + {"INPGA Left", NULL, "In Mixer Left"}, + {"ADC Left", NULL, "INPGA Left"}, + + {"INPGA Right", NULL, "In Mixer Right"}, + {"ADC Right", NULL, "INPGA Right"}, + + /* Output path */ + {"Out Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"Out Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Left", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Left", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"Out Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Right", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Right", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + + {"Mono Mixer", "INPGA Right Switch", "INPGA Right"}, + {"Mono Mixer", "INPGA Left Switch", "INPGA Left"}, + {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, + {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, + + {"OUTPGA Left Enable", NULL, "Out Mixer Left"}, + {"OUTPGA Right Enable", NULL, "Out Mixer Right"}, + + {"Out1 Left", NULL, "OUTPGA Left Enable"}, + {"OUT1L", NULL, "Out1 Left"}, + + {"Out1 Right", NULL, "OUTPGA Right Enable"}, + {"OUT1R", NULL, "Out1 Right"}, + + {"Headphone Left", NULL, "OUTPGA Left Enable"}, + {"HPL", NULL, "Headphone Left"}, + + {"Headphone Right", NULL, "OUTPGA Right Enable"}, + {"HPR", NULL, "Headphone Right"}, + + {"Out2 Mono", NULL, "Mono Mixer"}, + {"OUT2", NULL, "Out2 Mono"}, +}; + /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; @@ -411,29 +622,6 @@ static int da7210_volatile_register(struct snd_soc_codec *codec, return 0; } } -static int da7210_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - struct snd_soc_codec *codec = dai->codec; - - if (is_play) { - /* Enable Out */ - snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); - snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); - - } else { - /* Volume 7 */ - snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7); - snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7); - - /* Enable Mic */ - snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1); - snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1); - } - - return 0; -} /* * Set PCM DAI word length. @@ -603,7 +791,6 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute) /* DAI operations */ 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, @@ -742,6 +929,11 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .controls = da7210_snd_controls, .num_controls = ARRAY_SIZE(da7210_snd_controls), + + .dapm_widgets = da7210_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da7210_dapm_widgets), + .dapm_routes = da7210_audio_map, + .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- 1.7.1

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