May 2012 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH 2/2 V2] ASoC: fix pxa-ssp compiling issue under mach-mmp
by Eric Miao 30 May '12

30 May '12

On Fri, May 25, 2012 at 11:12 AM, Qiao Zhou <zhouqiao(a)marvell.com> wrote: > pxa-ssp.c uses API like cpu_is_pxa3xx(), cpu_is_pxa2xx(), which is > defined under arch-pxa architecture, and drivers under mach-mmp > can't find it. so just use ssp->type to replace that API. > > Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> Part of job of #ifdef ... #endif, and cpu_is_pxa*() is to reduce code size when the support for these SoCs are not compiled, but providing the overall maintenance burden and ugliness, I'm fine with the changes here. Acked-by: Eric Miao <eric.y.miao(a)gmail.com> > --- > include/linux/pxa2xx_ssp.h | 1 + > sound/soc/pxa/pxa-ssp.c | 38 +++++++++++--------------------------- > 2 files changed, 12 insertions(+), 27 deletions(-) > > diff --git a/include/linux/pxa2xx_ssp.h b/include/linux/pxa2xx_ssp.h > index f9fe15e..f366320 100644 > --- a/include/linux/pxa2xx_ssp.h > +++ b/include/linux/pxa2xx_ssp.h > @@ -160,6 +160,7 @@ enum pxa_ssp_type { > PXA25x_SSP, /* pxa 210, 250, 255, 26x */ > PXA25x_NSSP, /* pxa 255, 26x (including ASSP) */ > PXA27x_SSP, > + PXA3xx_SSP, > PXA168_SSP, > PXA910_SSP, > CE4100_SSP, > diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c > index fd04ce1..3c89162 100644 > --- a/sound/soc/pxa/pxa-ssp.c > +++ b/sound/soc/pxa/pxa-ssp.c > @@ -33,7 +33,6 @@ > > #include <mach/hardware.h> > #include <mach/dma.h> > -#include <mach/audio.h> > > #include "../../arm/pxa2xx-pcm.h" > #include "pxa-ssp.h" > @@ -195,7 +194,7 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) > { > u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); > > - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { > + if (ssp->type == PXA25x_SSP) { > sscr0 &= ~0x0000ff00; > sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ > } else { > @@ -213,7 +212,7 @@ static u32 pxa_ssp_get_scr(struct ssp_device *ssp) > u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); > u32 div; > > - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) > + if (ssp->type == PXA25x_SSP) > div = ((sscr0 >> 8) & 0xff) * 2 + 2; > else > div = ((sscr0 >> 8) & 0xfff) + 1; > @@ -243,7 +242,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, > break; > case PXA_SSP_CLK_PLL: > /* Internal PLL is fixed */ > - if (cpu_is_pxa25x()) > + if (ssp->type == PXA25x_SSP) > priv->sysclk = 1843200; > else > priv->sysclk = 13000000; > @@ -267,11 +266,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, > > /* The SSP clock must be disabled when changing SSP clock mode > * on PXA2xx. On PXA3xx it must be enabled when doing so. */ > - if (!cpu_is_pxa3xx()) > + if (ssp->type != PXA3xx_SSP) > clk_disable(ssp->clk); > val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; > pxa_ssp_write_reg(ssp, SSCR0, val); > - if (!cpu_is_pxa3xx()) > + if (ssp->type != PXA3xx_SSP) > clk_enable(ssp->clk); > > return 0; > @@ -295,24 +294,20 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, > case PXA_SSP_AUDIO_DIV_SCDB: > val = pxa_ssp_read_reg(ssp, SSACD); > val &= ~SSACD_SCDB; > -#if defined(CONFIG_PXA3xx) > - if (cpu_is_pxa3xx()) > + if (ssp->type == PXA3xx_SSP) > val &= ~SSACD_SCDX8; > -#endif > switch (div) { > case PXA_SSP_CLK_SCDB_1: > val |= SSACD_SCDB; > break; > case PXA_SSP_CLK_SCDB_4: > break; > -#if defined(CONFIG_PXA3xx) > case PXA_SSP_CLK_SCDB_8: > - if (cpu_is_pxa3xx()) > + if (ssp->type == PXA3xx_SSP) > val |= SSACD_SCDX8; > else > return -EINVAL; > break; > -#endif > default: > return -EINVAL; > } > @@ -338,10 +333,8 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, > struct ssp_device *ssp = priv->ssp; > u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; > > -#if defined(CONFIG_PXA3xx) > - if (cpu_is_pxa3xx()) > + if (ssp->type == PXA3xx_SSP) > pxa_ssp_write_reg(ssp, SSACDD, 0); > -#endif > > switch (freq_out) { > case 5622000: > @@ -366,11 +359,10 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, > break; > > default: > -#ifdef CONFIG_PXA3xx > /* PXA3xx has a clock ditherer which can be used to generate > * a wider range of frequencies - calculate a value for it. > */ > - if (cpu_is_pxa3xx()) { > + if (ssp->type == PXA3xx_SSP) { > u32 val; > u64 tmp = 19968; > tmp *= 1000000; > @@ -387,7 +379,6 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, > val, freq_out); > break; > } > -#endif > > return -EINVAL; > } > @@ -596,10 +587,8 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, > /* bit size */ > switch (params_format(params)) { > case SNDRV_PCM_FORMAT_S16_LE: > -#ifdef CONFIG_PXA3xx > - if (cpu_is_pxa3xx()) > + if (ssp->type == PXA3xx_SSP) > sscr0 |= SSCR0_FPCKE; > -#endif > sscr0 |= SSCR0_DataSize(16); > break; > case SNDRV_PCM_FORMAT_S24_LE: > @@ -624,9 +613,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, > * trying and failing a lot; some of the registers > * needed for that mode are only available on PXA3xx. > */ > - > -#ifdef CONFIG_PXA3xx > - if (!cpu_is_pxa3xx()) > + if (ssp->type != PXA3xx_SSP) > return -EINVAL; > > sspsp |= SSPSP_SFRMWDTH(width * 2); > @@ -634,9 +621,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, > sspsp |= SSPSP_EDMYSTOP(3); > sspsp |= SSPSP_DMYSTOP(3); > sspsp |= SSPSP_DMYSTRT(1); > -#else > - return -EINVAL; > -#endif > } else { > /* The frame width is the width the LRCLK is > * asserted for; the delay is expressed in > -- > 1.7.4.1 >

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[alsa-devel] [PATCH 0/2] ASoC: fsi: bugfix for DMAEngine
by Kuninori Morimoto 30 May '12

30 May '12

Hi Mark, Liam These are bugfix for FSI DMAEngine. Without these patches, FSI playback includes noise. These are for v3.5 kernel Kuninori Morimoto (2): ASoC: fsi: bugfix: correct dma area ASoC: fsi: bugfix: ensure dma is terminated Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: fsi: use PIO handler if DMA handler was invalid
by Kuninori Morimoto 30 May '12

30 May '12

PIO handler is not good performance, but works on all platform. So, switch to PIO handler if DMA handler was invalid case. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 29 ++++++++++++++++++++--------- 1 files changed, 20 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 05582c1..0da021a 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -247,7 +247,7 @@ struct fsi_priv { struct fsi_stream_handler { int (*init)(struct fsi_priv *fsi, struct fsi_stream *io); int (*quit)(struct fsi_priv *fsi, struct fsi_stream *io); - int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io); + int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev); int (*transfer)(struct fsi_priv *fsi, struct fsi_stream *io); int (*remove)(struct fsi_priv *fsi, struct fsi_stream *io); void (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io, @@ -571,16 +571,16 @@ static int fsi_stream_transfer(struct fsi_stream *io) #define fsi_stream_stop(fsi, io)\ fsi_stream_handler_call(io, start_stop, fsi, io, 0) -static int fsi_stream_probe(struct fsi_priv *fsi) +static int fsi_stream_probe(struct fsi_priv *fsi, struct device *dev) { struct fsi_stream *io; int ret1, ret2; io = &fsi->playback; - ret1 = fsi_stream_handler_call(io, probe, fsi, io); + ret1 = fsi_stream_handler_call(io, probe, fsi, io, dev); io = &fsi->capture; - ret2 = fsi_stream_handler_call(io, probe, fsi, io); + ret2 = fsi_stream_handler_call(io, probe, fsi, io, dev); if (ret1 < 0) return ret1; @@ -1182,7 +1182,7 @@ static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); } -static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io) +static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev) { dma_cap_mask_t mask; @@ -1190,8 +1190,19 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io) dma_cap_set(DMA_SLAVE, mask); io->chan = dma_request_channel(mask, fsi_dma_filter, &io->slave); - if (!io->chan) - return -EIO; + if (!io->chan) { + + /* switch to PIO handler */ + if (fsi_stream_is_play(fsi, io)) + fsi->playback.handler = &fsi_pio_push_handler; + else + fsi->capture.handler = &fsi_pio_pop_handler; + + dev_info(dev, "switch handler (dma => pio)\n"); + + /* probe again */ + return fsi_stream_probe(fsi, dev); + } tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io); @@ -1681,7 +1692,7 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.master = master; master->fsia.info = &info->port_a; fsi_handler_init(&master->fsia); - ret = fsi_stream_probe(&master->fsia); + ret = fsi_stream_probe(&master->fsia, &pdev->dev); if (ret < 0) { dev_err(&pdev->dev, "FSIA stream probe failed\n"); goto exit_iounmap; @@ -1692,7 +1703,7 @@ static int fsi_probe(struct platform_device *pdev) master->fsib.master = master; master->fsib.info = &info->port_b; fsi_handler_init(&master->fsib); - ret = fsi_stream_probe(&master->fsib); + ret = fsi_stream_probe(&master->fsib, &pdev->dev); if (ret < 0) { dev_err(&pdev->dev, "FSIB stream probe failed\n"); goto exit_fsia; -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: fsi: bugfix: enable master clock control on DMA stream
by Kuninori Morimoto 30 May '12

30 May '12

DMA stream handler didn't care about master clock. This patch fixes it up. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 7cee225..05582c1 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1172,9 +1172,14 @@ static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, int start) { + struct fsi_master *master = fsi_get_master(fsi); + u32 clk = fsi_is_port_a(fsi) ? CRA : CRB; u32 enable = start ? DMA_ON : 0; fsi_reg_mask_set(fsi, OUT_DMAC, DMA_ON, enable); + + if (fsi_is_clk_master(fsi)) + fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); } static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io) -- 1.7.5.4

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Re: [alsa-devel] Help requested: new HSS1394 MIDI back-end
by Clemens Ladisch 30 May '12

30 May '12

Sean M. Pappalardo - D.J. Pegasus wrote: >It works perfectly with the SCS.1m!! The .1d works, but some of >the non-standard data it sends appears to be getting corrupted. In >particular, the device sends four-byte platter messages that start with >0xF9 and the remaining three can range from 0x00 to 0xFF. This is not MIDI. Into what MIDI messages should the driver convert this? Is this documented anywhere? Regards, Clemens

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Re: [alsa-devel] [PATCH 2/4] ASoC: mmp: add audio dma support
by Vinod Koul 30 May '12

30 May '12

On Fri, 2012-05-25 at 15:11 +0800, Zhangfei Gao wrote: > mmp-pcm handle audio dma based on dmaengine > Support mmp and pxa910 Looks like this is *not* using soc-dmaengine library, why? > > Signed-off-by: Zhangfei Gao <zhangfei.gao(a)marvell.com> > Signed-off-by: Leo Yan <leoy(a)marvell.com> > Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> > --- > include/linux/platform_data/mmp_audio.h | 22 ++ > sound/soc/pxa/Kconfig | 8 + > sound/soc/pxa/Makefile | 2 + > sound/soc/pxa/mmp-pcm.c | 448 +++++++++++++++++++++++++++++++ > 4 files changed, 480 insertions(+), 0 deletions(-) > create mode 100644 include/linux/platform_data/mmp_audio.h > create mode 100644 sound/soc/pxa/mmp-pcm.c > > diff --git a/include/linux/platform_data/mmp_audio.h b/include/linux/platform_data/mmp_audio.h > new file mode 100644 > index 0000000..0f25d16 > --- /dev/null > +++ b/include/linux/platform_data/mmp_audio.h > @@ -0,0 +1,22 @@ > +/* > + * MMP Platform AUDIO Management > + * > + * Copyright (c) 2011 Marvell Semiconductors Inc. > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License version 2 as > + * published by the Free Software Foundation. > + * > + */ > + > +#ifndef MMP_AUDIO_H > +#define MMP_AUDIO_H > + > +struct mmp_audio_platdata { > + u32 period_max_capture; > + u32 buffer_max_capture; > + u32 period_max_playback; > + u32 buffer_max_playback; > +}; > + > +#endif /* MMP_AUDIO_H */ > diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig > index a0f7d3c..a516068 100644 > --- a/sound/soc/pxa/Kconfig > +++ b/sound/soc/pxa/Kconfig > @@ -8,6 +8,14 @@ config SND_PXA2XX_SOC > the PXA2xx AC97, I2S or SSP interface. You will also need > to select the audio interfaces to support below. > > +config SND_MMP_SOC > + bool "Soc Audio for Marvell MMP chips" > + depends on ARCH_MMP > + select SND_ARM > + help > + Say Y if you want to add support for codecs attached to > + the MMP SSPA interface. > + > config SND_PXA2XX_AC97 > tristate > select SND_AC97_CODEC > diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile > index af35762..f913e9b 100644 > --- a/sound/soc/pxa/Makefile > +++ b/sound/soc/pxa/Makefile > @@ -3,11 +3,13 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o > snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o > snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o > snd-soc-pxa-ssp-objs := pxa-ssp.o > +snd-soc-mmp-objs := mmp-pcm.o > > obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o > obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o > obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o > obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o > +obj-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o > > # PXA Machine Support > snd-soc-corgi-objs := corgi.o > diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c > new file mode 100644 > index 0000000..abafbb9 > --- /dev/null > +++ b/sound/soc/pxa/mmp-pcm.c > @@ -0,0 +1,448 @@ > +/* > + * linux/sound/soc/pxa/mmp-pcm.c > + * > + * Copyright (C) 2011 Marvell International Ltd. > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License as published by > + * the Free Software Foundation; either version 2 of the License, or > + * (at your option) any later version. > + * > + */ > +#include <linux/module.h> > +#include <linux/init.h> > +#include <linux/platform_device.h> > +#include <linux/slab.h> > +#include <linux/dma-mapping.h> > +#include <linux/dmaengine.h> > +#include <linux/platform_data/mmp_dma.h> > +#include <linux/platform_data/mmp_audio.h> > +#include <sound/pxa2xx-lib.h> > +#include <sound/core.h> > +#include <sound/pcm.h> > +#include <sound/pcm_params.h> > +#include <sound/soc.h> > +#include <mach/sram.h> > + > +struct mmp_runtime_data { > + int ssp_id; > + u32 period_size; > + u32 pointer; > + struct resource *dma_res; > + struct mmp_tdma_data tdma_data; > + struct gen_pool *gpool; > + struct snd_pcm_substream *substream; > + struct dma_chan *dma_chan; > + struct dma_async_tx_descriptor *desc; > +}; > + > +#define MMP_PCM_INFO (SNDRV_PCM_INFO_MMAP | \ > + SNDRV_PCM_INFO_MMAP_VALID | \ > + SNDRV_PCM_INFO_INTERLEAVED | \ > + SNDRV_PCM_INFO_PAUSE | \ > + SNDRV_PCM_INFO_RESUME) > + > +#define MMP_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ > + SNDRV_PCM_FMTBIT_S24_LE | \ > + SNDRV_PCM_FMTBIT_S32_LE) > + > +static struct snd_pcm_hardware mmp_pcm_hardware[] = { > + { > + .info = MMP_PCM_INFO, > + .formats = MMP_PCM_FORMATS, > + .period_bytes_min = 1024, > + .period_bytes_max = 2048, > + .periods_min = 2, > + .periods_max = 32, > + .buffer_bytes_max = 4096, > + .fifo_size = 32, > + }, > + { > + .info = MMP_PCM_INFO, > + .formats = MMP_PCM_FORMATS, > + .period_bytes_min = 1024, > + .period_bytes_max = 2048, > + .periods_min = 2, > + .periods_max = 32, > + .buffer_bytes_max = 4096, > + .fifo_size = 32, > + }, > +}; > + > +static void mmp_pcm_adma_irq(void *data) > +{ > + struct snd_pcm_substream *substream = data; > + struct mmp_runtime_data *prtd = substream->runtime->private_data; > + size_t dma_bytes = substream->runtime->dma_bytes; > + > + prtd->pointer = (prtd->pointer + prtd->period_size) % dma_bytes; > + snd_pcm_period_elapsed(substream); > + return; > +} > + > +static bool filter(struct dma_chan *chan, void *param) > +{ > + struct mmp_runtime_data *prtd = param; > + bool found = false; > + char *devname; > + > + devname = kasprintf(GFP_KERNEL, "%s.%d", prtd->dma_res->name, > + prtd->ssp_id); > + if ((strcmp(dev_name(chan->device->dev), devname) == 0) && > + (chan->chan_id == prtd->dma_res->start)) { > + chan->private = &prtd->tdma_data; > + found = true; > + } > + > + kfree(devname); > + return found; > +} > + > +static int mmp_pcm_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct mmp_runtime_data *prtd = runtime->private_data; > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct pxa2xx_pcm_dma_params *dma_params; > + size_t totsize = params_buffer_bytes(params); > + size_t period = params_period_bytes(params); > + int ret; > + struct dma_slave_config slave_config; > + dma_cap_mask_t mask; > + > + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); > + if (!dma_params) > + return 0; > + > + switch (params_format(params)) { > + case SNDRV_PCM_FORMAT_S8: > + prtd->tdma_data.bus_size = 8; > + break; > + case SNDRV_PCM_FORMAT_S16_LE: > + prtd->tdma_data.bus_size = 16; > + break; > + case SNDRV_PCM_FORMAT_S24_LE: > + prtd->tdma_data.bus_size = 24; > + break; > + case SNDRV_PCM_FORMAT_S32_LE: > + prtd->tdma_data.bus_size = 32; > + break; > + default: > + return -EINVAL; > + } > + prtd->tdma_data.pack_mod = true; > + > + dma_cap_zero(mask); > + dma_cap_set(DMA_CYCLIC, mask); > + prtd->dma_chan = dma_request_channel(mask, filter, prtd); > + if (!prtd->dma_chan) > + return -EINVAL; > + > + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { > + slave_config.direction = DMA_TO_DEVICE; > + slave_config.dst_addr = dma_params->dev_addr; > + slave_config.dst_maxburst = 4; > + } else { > + slave_config.direction = DMA_FROM_DEVICE; > + slave_config.src_addr = dma_params->dev_addr; > + slave_config.src_maxburst = 4; > + } > + > + ret = dmaengine_slave_config(prtd->dma_chan, &slave_config); > + if (ret) > + return ret; > + > + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); > + > + prtd->period_size = period; > + runtime->dma_bytes = totsize; > + > + prtd->desc = prtd->dma_chan->device->device_prep_dma_cyclic( > + prtd->dma_chan, runtime->dma_addr, totsize, period, > + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? > + DMA_TO_DEVICE : DMA_FROM_DEVICE, NULL); > + if (!prtd->desc) { > + dev_err(&prtd->dma_chan->dev->device, "cannot prepare slave dma\n"); > + return -EINVAL; > + } > + > + prtd->desc->callback = mmp_pcm_adma_irq; > + prtd->desc->callback_param = substream; > + dmaengine_submit(prtd->desc); > + return 0; > +} > + > +static int mmp_pcm_hw_free(struct snd_pcm_substream *substream) > +{ > + struct mmp_runtime_data *prtd = substream->runtime->private_data; > + > + if (prtd->dma_chan) { > + dma_release_channel(prtd->dma_chan); > + prtd->dma_chan = NULL; > + } > + > + return 0; > +} > + > +static int mmp_pcm_trigger(struct snd_pcm_substream *substream, int cmd) > +{ > + struct mmp_runtime_data *prtd = substream->runtime->private_data; > + int ret = 0; > + > + switch (cmd) { > + case SNDRV_PCM_TRIGGER_START: > + prtd->pointer = 0; > + dma_async_issue_pending(prtd->dma_chan); > + break; > + > + case SNDRV_PCM_TRIGGER_STOP: > + case SNDRV_PCM_TRIGGER_SUSPEND: > + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: > + dmaengine_terminate_all(prtd->dma_chan); > + break; > + > + case SNDRV_PCM_TRIGGER_RESUME: > + dma_async_issue_pending(prtd->dma_chan); > + break; > + > + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > + dma_async_issue_pending(prtd->dma_chan); > + break; > + > + default: > + ret = -EINVAL; > + break; > + } > + > + return ret; > +} > + > +static snd_pcm_uframes_t mmp_pcm_pointer(struct snd_pcm_substream *substream) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct mmp_runtime_data *prtd = runtime->private_data; > + snd_pcm_uframes_t x; > + > + x = bytes_to_frames(runtime, prtd->pointer); > + if (x == runtime->buffer_size) > + x = 0; > + > + return x; > +} > + > +static int mmp_pcm_open(struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct platform_device *pdev = to_platform_device(rtd->platform->dev); > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; > + struct mmp_runtime_data *prtd; > + struct resource *r; > + int ret; > + > + r = platform_get_resource(pdev, IORESOURCE_DMA, substream->stream); > + if (!r) > + return -EBUSY; > + > + snd_soc_set_runtime_hwparams(substream, > + &mmp_pcm_hardware[substream->stream]); > + > + /* > + * For mysterious reasons (and despite what the manual says) > + * playback samples are lost if the DMA count is not a multiple > + * of the DMA burst size. Let's add a rule to enforce that. > + */ > + ret = snd_pcm_hw_constraint_step(runtime, 0, > + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); > + if (ret) > + goto out; > + > + ret = snd_pcm_hw_constraint_step(runtime, 0, > + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); > + if (ret) > + goto out; > + > + ret = snd_pcm_hw_constraint_integer(runtime, > + SNDRV_PCM_HW_PARAM_PERIODS); > + if (ret < 0) > + goto out; > + > + prtd = devm_kzalloc(&pdev->dev, > + sizeof(struct mmp_runtime_data), GFP_KERNEL); > + if (prtd == NULL) { > + ret = -ENOMEM; > + goto out; > + } > + > + prtd->substream = substream; > + runtime->private_data = prtd; > + prtd->dma_res = r; > + prtd->ssp_id = cpu_dai->id; > + return 0; > + > +out: > + return ret; > +} > + > +static int mmp_pcm_close(struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct platform_device *pdev = to_platform_device(rtd->platform->dev); > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct mmp_runtime_data *prtd = runtime->private_data; > + > + devm_kfree(&pdev->dev, prtd); > + runtime->private_data = NULL; > + return 0; > +} > + > +static int mmp_pcm_mmap(struct snd_pcm_substream *substream, > + struct vm_area_struct *vma) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + unsigned long off = vma->vm_pgoff; > + > + vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); > + return remap_pfn_range(vma, vma->vm_start, > + __phys_to_pfn(runtime->dma_addr) + off, > + vma->vm_end - vma->vm_start, vma->vm_page_prot); > +} > + > +struct snd_pcm_ops mmp_pcm_ops = { > + .open = mmp_pcm_open, > + .close = mmp_pcm_close, > + .ioctl = snd_pcm_lib_ioctl, > + .hw_params = mmp_pcm_hw_params, > + .hw_free = mmp_pcm_hw_free, > + .trigger = mmp_pcm_trigger, > + .pointer = mmp_pcm_pointer, > + .mmap = mmp_pcm_mmap, > +}; > + > +static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) > +{ > + struct snd_pcm_substream *substream = pcm->streams[stream].substream; > + struct snd_dma_buffer *buf = &substream->dma_buffer; > + size_t size = mmp_pcm_hardware[stream].buffer_bytes_max; > + struct gen_pool *gpool; > + > + buf->dev.type = SNDRV_DMA_TYPE_DEV; > + buf->dev.dev = pcm->card->dev; > + buf->private_data = NULL; > + > + gpool = sram_get_gpool("asram"); > + if (!gpool) > + return -ENOMEM; > + > + buf->area = (unsigned char *)gen_pool_alloc(gpool, size); > + if (!buf->area) > + return -ENOMEM; > + buf->addr = gen_pool_virt_to_phys(gpool, (unsigned long)buf->area); > + buf->bytes = size; > + return 0; > +} > + > +static void mmp_pcm_free_dma_buffers(struct snd_pcm *pcm) > +{ > + struct snd_pcm_substream *substream; > + struct snd_dma_buffer *buf; > + int stream; > + struct gen_pool *gpool; > + > + gpool = sram_get_gpool("asram"); > + if (!gpool) > + return; > + > + for (stream = 0; stream < 2; stream++) { > + size_t size = mmp_pcm_hardware[stream].buffer_bytes_max; > + > + substream = pcm->streams[stream].substream; > + if (!substream) > + continue; > + > + buf = &substream->dma_buffer; > + if (!buf->area) > + continue; > + gen_pool_free(gpool, (unsigned long)buf->area, size); > + buf->area = NULL; > + } > + > + return; > +} > + > +static u64 mmp_pcm_dmamask = DMA_BIT_MASK(64); > + > +int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd) > +{ > + struct snd_card *card = rtd->card->snd_card; > + struct snd_pcm *pcm = rtd->pcm; > + int ret = 0; > + > + if (!card->dev->dma_mask) > + card->dev->dma_mask = &mmp_pcm_dmamask; > + > + if (!card->dev->coherent_dma_mask) > + card->dev->coherent_dma_mask = DMA_BIT_MASK(64); > + > + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > + ret = mmp_pcm_preallocate_dma_buffer(pcm, > + SNDRV_PCM_STREAM_PLAYBACK); > + if (ret) > + goto out; > + } > + > + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { > + ret = mmp_pcm_preallocate_dma_buffer(pcm, > + SNDRV_PCM_STREAM_CAPTURE); > + if (ret) > + goto out; > + } > + out: > + return ret; > +} > + > +struct snd_soc_platform_driver mmp_soc_platform = { > + .ops = &mmp_pcm_ops, > + .pcm_new = mmp_pcm_new, > + .pcm_free = mmp_pcm_free_dma_buffers, > +}; > + > +static __devinit int mmp_pcm_probe(struct platform_device *pdev) > +{ > + struct mmp_audio_platdata *pdata = pdev->dev.platform_data; > + > + if (pdata) { > + mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].buffer_bytes_max = > + pdata->buffer_max_playback; > + mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].period_bytes_max = > + pdata->period_max_playback; > + mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].buffer_bytes_max = > + pdata->buffer_max_capture; > + mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].period_bytes_max = > + pdata->period_max_capture; > + } > + return snd_soc_register_platform(&pdev->dev, &mmp_soc_platform); > +} > + > +static int __devexit mmp_pcm_remove(struct platform_device *pdev) > +{ > + snd_soc_unregister_platform(&pdev->dev); > + return 0; > +} > + > +static struct platform_driver mmp_pcm_driver = { > + .driver = { > + .name = "mmp-pcm-audio", > + .owner = THIS_MODULE, > + }, > + > + .probe = mmp_pcm_probe, > + .remove = __devexit_p(mmp_pcm_remove), > +}; > + > +module_platform_driver(mmp_pcm_driver); > + > +MODULE_AUTHOR("Leo Yan <leoy(a)marvell.com>"); > +MODULE_DESCRIPTION("MMP Soc Audio DMA module"); > +MODULE_LICENSE("GPL"); -- ~Vinod -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

5 14

[alsa-devel] Dynamic PCM and Tegra AHUB
by Stephen Warren 29 May '12

29 May '12

Mark, Liam, Tegra30's AHUB is structured as follows: > +-------------+ +-----+ +--------------+ +-----+ +------+ > | FIFO pair 0 |<->| CIF |<->| Cross-bar |<->| CIF |<->| I2S0 |<-> External IO > +-------------+ +-----+ | (the "AHUB") | +-----+ +------+ > . . . | | . . . > +-------------+ +-----+ | | +-----+ +------+ > | FIFO pair 3 |<->| CIF |<->| |<->| CIF |<->| I2S4 |<-> External IO > +-------------+ +-----+ | | +-----+ +------+ > | | > | | +-----+ +-------+ > | |<->| CIF |<->| SPDIF |<-> External IO > | | +-----+ +-------+ > | | > | | +-----+ +-------+ > | |<->| CIF |<->| DAM | > | | +-----+ +-------+ > +--------------+ Notes on the diagram: Each FIFO above is a separate TX and RX FIFO. I merged them in the drawing for simplicity, but they operate completely independently; different memory packing formats, data flow rates, ... The CIF can convert the audio format, e.g. mono<->stereo conversion and change the # of bits in the data in pretty arbitrary combinations. This is true for all CIFs; those that join the AHUB core to either the DMA FIFOs or the I2S/SPDIF/DAM controllers. The AHUB core is a complete cross-bar; each output selects 1 of the n inputs. The I2S and SPDIF controllers take audio from the AHUB, format it to the appropriate protocol, and send to external IO (or the other way around). The I2S and SPDIF modules don't perform any additional data rate/width/channel conversion; the CIF must do whatever conversions are needed. The DAMs take 2 input channels from the AHUB, optionally perform some sample rate conversion and/or bit size conversion beyond what the CIF does, mix them together, and send them back into the AHUB cross-bar. Questions: It seems like the whole point of ASoC dynamic PCM is to represent the AHUB core, and at least some of the surrounding boxes above, as an ASoC CODEC. Initially, I would assume the AHUB CODEC's DAIs would be the DMA FIFOs and the I2S* controllers themselves, i.e. include everything in the above diagram. If I do that, then the dai_links that the ASoC machine driver registers for I2S->WM8903 would represent a codec DAI <-> codec DAI link rather than a regular CPU DAI <-> codec DAI link. The difference here is the lack of symmetry in struct snd_soc_dai_link: The CPU-side of each DAI link only has a "cpu_dai_name" to identify it, whereas the codec side has both a codec_name (or of node) /plus/ a DAI name. This is important at least in the context of device tree, where to avoid hard-coding specific device names into the dai_link structure's .codec_name field in the machine driver, an of_node is looked up at run-time instead, and then the codec_dai_name interpreted relative to that. Without a CPU-side separation between name/node and dai_name, that's not possible. So, I sent a patch for that in case this is the right approach. An alternative to expanding struct snd_soc_dai_link might be to add an API so the machine driver could ask the AHUB driver for the globally-unique name of each DAI it exposes, and write that into dai_link.cpu_dai_name. But if we have that API, we could use it for the codec-side of the link to, and just have .codec_dai_name and remove the .codec_name and .codec_of_node fields. Alternatively, perhaps the DMA FIFOs should be registered as CPU DAIs completely separately from the AHUB CODEC. The AHUB CIFs would then be the DAIs registered by the AHUB CODEC. But then, the machine driver would need to include dai_links for all the DMA FIFO <-> CIF connections, which would end up duplicating that list of links into all Tegra machine drives (or perhaps sharing it in common code). Very curious for your thoughts...

2 2

[alsa-devel] [PATCH] ASoC: Support TI Isabelle Audio driver
by MR.Swami.Reddy＠ti.com 29 May '12

29 May '12

From: M R Swami Reddy <MR.Swami.Reddy(a)ti.com> Updated the patch by removing the startup(), as per the Liam's comments. Plz review the below Isabelle audio codec driver patch. Latest datasheet at below url: https://sps07.itg.ti.com/sites/MIS/PAIS_DESIGN/ISABELLE/CHIP_DESIGN/Shared%… ----------------- ASoC: Support TI ISABELLE Audio driver The Isabelle Audio IC is a complete low power high fidelity CODEC with integrated ADCs, DACs, decimation and interpolation filters, PLL, and power providers. This device supports 2 analog and 2 digital microphone channels, a mono earpiece driver, stereo class G headphone drivers with ultra low power and best SNR in the industry, stereo Class D speaker drivers, and 2 high performance Line drivers. Signed-off-by: Vishwas A Deshpande <vishwas.a.deshpande(a)ti.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/isabelle.c | 1262 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/isabelle.h | 143 ++++ 4 files changed, 1411 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ed5f66c..4ebf9e8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -30,6 +30,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_DFBMCS320 + select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C @@ -185,6 +186,9 @@ config SND_SOC_DFBMCS320 config SND_SOC_DMIC tristate +config SND_SOC_ISABELLE + tristate + config SND_SOC_LM49453 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7bbfdb2..fdc1ff6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -17,6 +17,7 @@ snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o +snd-soc-isabelle-objs := isabelle.o snd-soc-l3-objs := l3.o snd-soc-lm49453-objs := lm49453.o snd-soc-max98088-objs := max98088.o @@ -106,6 +107,7 @@ obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o +obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c new file mode 100644 index 0000000..ac33d4f --- /dev/null +++ b/sound/soc/codecs/isabelle.c @@ -0,0 +1,1262 @@ +/* + * isabelle.c - Low power high fidelity audio codec driver + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * + * Initially based on sound/soc/codecs/twl6040.c + * + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <asm/div64.h> +#include "isabelle.h" + + +/* Register default values for ISABELLE driver. */ +static struct reg_default isabelle_reg_defs[] = { + { 0, 0x00 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x00 }, + { 5, 0x00 }, + { 6, 0x00 }, + { 7, 0x00 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x00 }, + { 11, 0x00 }, + { 12, 0x00 }, + { 13, 0x00 }, + { 14, 0x00 }, + { 15, 0x00 }, + { 16, 0x00 }, + { 17, 0x00 }, + { 18, 0x00 }, + { 19, 0x00 }, + { 20, 0x00 }, + { 21, 0x02 }, + { 22, 0x02 }, + { 23, 0x02 }, + { 24, 0x02 }, + { 25, 0x0F }, + { 26, 0x8F }, + { 27, 0x0F }, + { 28, 0x8F }, + { 29, 0x00 }, + { 30, 0x00 }, + { 31, 0x00 }, + { 32, 0x00 }, + { 33, 0x00 }, + { 34, 0x00 }, + { 35, 0x00 }, + { 36, 0x00 }, + { 37, 0x00 }, + { 38, 0x00 }, + { 39, 0x00 }, + { 40, 0x00 }, + { 41, 0x00 }, + { 42, 0x00 }, + { 43, 0x00 }, + { 44, 0x00 }, + { 45, 0x00 }, + { 46, 0x00 }, + { 47, 0x00 }, + { 48, 0x00 }, + { 49, 0x00 }, + { 50, 0x00 }, + { 51, 0x00 }, + { 52, 0x00 }, + { 53, 0x00 }, + { 54, 0x00 }, + { 55, 0x00 }, + { 56, 0x00 }, + { 57, 0x00 }, + { 58, 0x00 }, + { 59, 0x00 }, + { 60, 0x00 }, + { 61, 0x00 }, + { 62, 0x00 }, + { 63, 0x00 }, + { 64, 0x00 }, + { 65, 0x00 }, + { 66, 0x00 }, + { 67, 0x00 }, + { 68, 0x00 }, + { 69, 0x90 }, + { 70, 0x90 }, + { 71, 0x90 }, + { 72, 0x00 }, + { 73, 0x00 }, + { 74, 0x00 }, + { 75, 0x00 }, + { 76, 0x00 }, + { 77, 0x00 }, + { 78, 0x00 }, + { 79, 0x00 }, + { 80, 0x00 }, + { 81, 0x00 }, + { 82, 0x00 }, + { 83, 0x00 }, + { 84, 0x00 }, + { 85, 0x07 }, + { 86, 0x00 }, + { 87, 0x00 }, + { 88, 0x00 }, + { 89, 0x07 }, + { 90, 0x80 }, + { 91, 0x07 }, + { 92, 0x07 }, + { 93, 0x00 }, + { 94, 0x00 }, + { 95, 0x00 }, + { 96, 0x00 }, + { 97, 0x00 }, + { 98, 0x00 }, + { 99, 0x00 }, +}; + +/* codec private data */ +struct isabelle_priv { + struct regmap *regmap; +}; + +static const char *isabelle_rx1_texts[] = {"VRX1", "ARX1"}; +static const char *isabelle_rx2_texts[] = {"VRX2", "ARX2"}; + +static const struct soc_enum isabelle_rx1_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, 1, isabelle_rx1_texts), + SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, 1, isabelle_rx1_texts), +}; + +static const struct soc_enum isabelle_rx2_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, 1, isabelle_rx2_texts), + SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, 1, isabelle_rx2_texts), +}; + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new rx1_mux_controls = + SOC_DAPM_ENUM("Route", isabelle_rx1_enum); + +static const struct snd_kcontrol_new rx2_mux_controls = + SOC_DAPM_ENUM("Route", isabelle_rx2_enum); + +/* TX input selection */ +static const char *isabelle_atx_texts[] = {"AMIC1", "DMIC"}; +static const char *isabelle_vtx_texts[] = {"AMIC2", "DMIC"}; + +static const struct soc_enum isabelle_atx_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, 1, isabelle_atx_texts), + SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_atx_texts), +}; + +static const struct soc_enum isabelle_vtx_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, 1, isabelle_vtx_texts), + SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_vtx_texts), +}; + +static const struct snd_kcontrol_new atx_mux_controls = + SOC_DAPM_ENUM("Route", isabelle_atx_enum); + +static const struct snd_kcontrol_new vtx_mux_controls = + SOC_DAPM_ENUM("Route", isabelle_vtx_enum); + +/* Left analog microphone selection */ +static const char *isabelle_amic1_texts[] = { + "Main Mic", "Headset Mic", "Aux/FM Left"}; + +/* Left analog microphone selection */ +static const char *isabelle_amic2_texts[] = {"Sub Mic", "Aux/FM Right"}; + +static const struct soc_enum isabelle_amic1_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 5, + ARRAY_SIZE(isabelle_amic1_texts), + isabelle_amic1_texts), +}; + +static const struct soc_enum isabelle_amic2_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 4, + ARRAY_SIZE(isabelle_amic2_texts), + isabelle_amic2_texts), +}; + +static const struct snd_kcontrol_new amic1_control = + SOC_DAPM_ENUM("Route", isabelle_amic1_enum); + +static const struct snd_kcontrol_new amic2_control = + SOC_DAPM_ENUM("Route", isabelle_amic2_enum); + +static const char *isabelle_st_audio_texts[] = {"ATX1", "ATX2"}; + +static const char *isabelle_st_voice_texts[] = {"VTX1", "VTX2"}; + +static const struct soc_enum isabelle_st_audio_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, 1, + isabelle_st_audio_texts), + SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, 1, + isabelle_st_audio_texts), +}; + +static const struct soc_enum isabelle_st_voice_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, 1, + isabelle_st_voice_texts), + SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, 1, + isabelle_st_voice_texts), +}; + +static const struct snd_kcontrol_new st_audio_control = + SOC_DAPM_ENUM("Route", isabelle_st_audio_enum); + +static const struct snd_kcontrol_new st_voice_control = + SOC_DAPM_ENUM("Route", isabelle_st_voice_enum); + +/* Mixer controls */ +static const struct snd_kcontrol_new isabelle_hs_left_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC1L Playback Switch", ISABELLE_HSDRV_CFG1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_HSDRV_CFG1_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_hs_right_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC1R Playback Switch", ISABELLE_HSDRV_CFG1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_HSDRV_CFG1_REG, 4, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_hf_left_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC2L Playback Switch", ISABELLE_HFLPGA_CFG_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_HFLPGA_CFG_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_hf_right_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC2R Playback Switch", ISABELLE_HFRPGA_CFG_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_HFRPGA_CFG_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_ep_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC2L Playback Switch", ISABELLE_EARDRV_CFG1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_EARDRV_CFG1_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_aux_left_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC3L Playback Switch", ISABELLE_LINEAMP_CFG_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_LINEAMP_CFG_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_aux_right_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC3R Playback Switch", ISABELLE_LINEAMP_CFG_REG, 5, 1, 0), +SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_LINEAMP_CFG_REG, 4, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga1_left_mixer_controls[] = { +SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 7, 1, 0), +SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 6, 1, 0), +SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 5, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga1_right_mixer_controls[] = { +SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 3, 1, 0), +SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 2, 1, 0), +SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 1, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga2_left_mixer_controls[] = { +SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 7, 1, 0), +SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 6, 1, 0), +SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 5, 1, 0), +SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 4, 1, 0), +SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 3, 1, 0), +SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 2, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga2_right_mixer_controls[] = { +SOC_DAPM_SINGLE("USNC Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 7, 1, 0), +SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 3, 1, 0), +SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 2, 1, 0), +SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 1, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga3_left_mixer_controls[] = { +SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 7, 1, 0), +SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 6, 1, 0), +SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 5, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga3_right_mixer_controls[] = { +SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 3, 1, 0), +SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 2, 1, 0), +SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 1, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx1_mixer_controls[] = { +SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DL1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx2_mixer_controls[] = { +SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 5, 1, 0), +SOC_DAPM_SINGLE("DL2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 4, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx3_mixer_controls[] = { +SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 3, 1, 0), +SOC_DAPM_SINGLE("DL3 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 2, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx4_mixer_controls[] = { +SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DL4 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 0, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx5_mixer_controls[] = { +SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DL5 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx6_mixer_controls[] = { +SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("DL6 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 4, 1, 0), +}; + +static const struct snd_kcontrol_new ep_path_enable_control = + SOC_DAPM_SINGLE("Switch", ISABELLE_EARDRV_CFG2_REG, 0, 1, 0); + +/* TLV Declarations */ + +/* + * MICGAIN volume control: + * from 0 to 30 dB in 1 dB steps + */ +static const DECLARE_TLV_DB_SCALE(mic_amp_tlv, 0, 100, 0); + +/* + * AFMGAIN volume control: + * from -33 to 12 dB in 3 dB steps + */ +static const DECLARE_TLV_DB_SCALE(afm_amp_tlv, -3300, 300, 0); + +/* + * DACGAIN (HS, EP, LINE) volume control: + * from -12 to 6 dB in 2 dB steps + */ +static const DECLARE_TLV_DB_SCALE(dac_tlv, -1200, 200, 0); + +/* + * HFGAIN volume control: + * from -50 to 6 dB in 2 dB steps + */ +static const DECLARE_TLV_DB_SCALE(hf_tlv, -5000, 200, 0); + +/* + * DPGAGAIN volume control: + * from -63 to 0 dB in 1 dB steps + */ +static const DECLARE_TLV_DB_SCALE(dpga_tlv, -6300, 100, 1); + +/* + * STGAIN volume control: + * from -27 to 0 dB in 3 dB steps + */ +static const DECLARE_TLV_DB_SCALE(st_tlv, -2700, 300, 1); + +/* + * RXGAIN volume control: + * from -63 to 9 dB in 1 dB steps + */ +static const DECLARE_TLV_DB_SCALE(rx_tlv, -6300, 100, 1); + +/* + * TXGAIN volume control: + * from 9 to -6 dB in 1 dB steps + */ +static const DECLARE_TLV_DB_SCALE(tx_tlv, -600, 100, 0); + +static const struct snd_kcontrol_new isabelle_snd_controls[] = { + SOC_DOUBLE_TLV("Headset Playback Volume", ISABELLE_HSDRV_GAIN_REG, + 4, 0, 0xF, 0, dac_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + ISABELLE_HFLPGA_CFG_REG, ISABELLE_HFRPGA_CFG_REG, + 0, 0x1F, 0, hf_tlv), + SOC_DOUBLE_TLV("Aux Playback Volume", ISABELLE_LINEAMP_GAIN_REG, + 4, 0, 0xF, 0, dac_tlv), + SOC_SINGLE_TLV("Earpiece Playback Volume", ISABELLE_EARDRV_CFG1_REG, + 0, 0xF, 0, dac_tlv), + + SOC_DOUBLE_TLV("Aux FM Volume", ISABELLE_APGA_GAIN_REG, 4, 0, 0xF, 0, + afm_amp_tlv), + SOC_SINGLE_TLV("Mic1 Capture Volume", ISABELLE_MIC1_GAIN_REG, 3, 0x1F, + 0, mic_amp_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", ISABELLE_MIC2_GAIN_REG, 3, 0x1F, + 0, mic_amp_tlv), + + SOC_DOUBLE_R_TLV("DPGA1 Volume", ISABELLE_DPGA1L_GAIN_REG, + ISABELLE_DPGA1R_GAIN_REG, 0, 0x3F, 0, dpga_tlv), + SOC_DOUBLE_R_TLV("DPGA2 Volume", ISABELLE_DPGA2L_GAIN_REG, + ISABELLE_DPGA2R_GAIN_REG, 0, 0x3F, 0, dpga_tlv), + SOC_DOUBLE_R_TLV("DPGA3 Volume", ISABELLE_DPGA3L_GAIN_REG, + ISABELLE_DPGA3R_GAIN_REG, 0, 0x3F, 0, dpga_tlv), + + SOC_SINGLE_TLV("Sidetone Audio TX1 Volume", + ISABELLE_ATX_STPGA1_CFG_REG, 0, 0xF, 0, st_tlv), + SOC_SINGLE_TLV("Sidetone Audio TX2 Volume", + ISABELLE_ATX_STPGA2_CFG_REG, 0, 0xF, 0, st_tlv), + SOC_SINGLE_TLV("Sidetone Voice TX1 Volume", + ISABELLE_VTX_STPGA1_CFG_REG, 0, 0xF, 0, st_tlv), + SOC_SINGLE_TLV("Sidetone Voice TX2 Volume", + ISABELLE_VTX2_STPGA2_CFG_REG, 0, 0xF, 0, st_tlv), + + SOC_SINGLE_TLV("Audio TX1 Volume", ISABELLE_ATX1_DPGA_REG, 4, 0xF, 0, + tx_tlv), + SOC_SINGLE_TLV("Audio TX2 Volume", ISABELLE_ATX2_DPGA_REG, 4, 0xF, 0, + tx_tlv), + SOC_SINGLE_TLV("Voice TX1 Volume", ISABELLE_VTX1_DPGA_REG, 4, 0xF, 0, + tx_tlv), + SOC_SINGLE_TLV("Voice TX2 Volume", ISABELLE_VTX2_DPGA_REG, 4, 0xF, 0, + tx_tlv), + + SOC_SINGLE_TLV("RX1 DPGA Volume", ISABELLE_RX1_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX2 DPGA Volume", ISABELLE_RX2_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX3 DPGA Volume", ISABELLE_RX3_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX4 DPGA Volume", ISABELLE_RX4_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX5 DPGA Volume", ISABELLE_RX5_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX6 DPGA Volume", ISABELLE_RX6_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + + SOC_SINGLE("Headset Noise Gate", ISABELLE_HS_NG_CFG1_REG, 7, 1, 0), + SOC_SINGLE("Handsfree Noise Gate", ISABELLE_HF_NG_CFG1_REG, 7, 1, 0), + + SOC_SINGLE("ATX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 7, 1, 0), + SOC_SINGLE("ATX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 6, 1, 0), + SOC_SINGLE("ARX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 5, 1, 0), + SOC_SINGLE("ARX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 4, 1, 0), + SOC_SINGLE("ARX3 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 3, 1, 0), + SOC_SINGLE("ARX4 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 2, 1, 0), + SOC_SINGLE("ARX5 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 1, 1, 0), + SOC_SINGLE("ARX6 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 0, 1, 0), + SOC_SINGLE("VRX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 3, 1, 0), + SOC_SINGLE("VRX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 2, 1, 0), + + SOC_SINGLE("ATX1 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG, + 7, 1, 0), + SOC_SINGLE("ATX2 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG, + 6, 1, 0), + SOC_SINGLE("VTX1 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG, + 5, 1, 0), + SOC_SINGLE("VTX2 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG, + 4, 1, 0), + SOC_SINGLE("RX1 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 5, 1, 0), + SOC_SINGLE("RX2 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 4, 1, 0), + SOC_SINGLE("RX3 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 3, 1, 0), + SOC_SINGLE("RX4 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 2, 1, 0), + SOC_SINGLE("RX5 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 1, 1, 0), + SOC_SINGLE("RX6 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 0, 1, 0), + + SOC_SINGLE("ULATX12 Capture Switch", ISABELLE_ULATX12_INTF_CFG_REG, + 7, 1, 0), + + SOC_SINGLE("DL12 Playback Switch", ISABELLE_DL12_INTF_CFG_REG, + 7, 1, 0), + SOC_SINGLE("DL34 Playback Switch", ISABELLE_DL34_INTF_CFG_REG, + 7, 1, 0), + SOC_SINGLE("DL56 Playback Switch", ISABELLE_DL56_INTF_CFG_REG, + 7, 1, 0), + + /* DMIC Switch */ + SOC_SINGLE("DMIC Switch", ISABELLE_DMIC_CFG_REG, 0, 1, 0), +}; + +static const struct snd_soc_dapm_widget isabelle_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MAINMIC"), + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("SUBMIC"), + SND_SOC_DAPM_INPUT("LINEIN1"), + SND_SOC_DAPM_INPUT("LINEIN2"), + SND_SOC_DAPM_INPUT("DMICDAT"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), + SND_SOC_DAPM_OUTPUT("EP"), + SND_SOC_DAPM_OUTPUT("LINEOUT1"), + SND_SOC_DAPM_OUTPUT("LINEOUT2"), + + SND_SOC_DAPM_PGA("DL1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL4", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL5", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL6", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Analog input muxes for the capture amplifiers */ + SND_SOC_DAPM_MUX("Analog Left Capture Route", + SND_SOC_NOPM, 0, 0, &amic1_control), + SND_SOC_DAPM_MUX("Analog Right Capture Route", + SND_SOC_NOPM, 0, 0, &amic2_control), + + SND_SOC_DAPM_MUX("Sidetone Audio Playback", SND_SOC_NOPM, 0, 0, + &st_audio_control), + SND_SOC_DAPM_MUX("Sidetone Voice Playback", SND_SOC_NOPM, 0, 0, + &st_voice_control), + + /* AIF */ + SND_SOC_DAPM_AIF_IN("INTF1_SDI", NULL, 0, ISABELLE_INTF_EN_REG, 7, 0), + SND_SOC_DAPM_AIF_IN("INTF2_SDI", NULL, 0, ISABELLE_INTF_EN_REG, 6, 0), + + SND_SOC_DAPM_AIF_OUT("INTF1_SDO", NULL, 0, ISABELLE_INTF_EN_REG, 5, 0), + SND_SOC_DAPM_AIF_OUT("INTF2_SDO", NULL, 0, ISABELLE_INTF_EN_REG, 4, 0), + + SND_SOC_DAPM_OUT_DRV("ULATX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("ULATX2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("ULVTX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("ULVTX2", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Analog Capture PGAs */ + SND_SOC_DAPM_PGA("MicAmp1", ISABELLE_AMIC_CFG_REG, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmp2", ISABELLE_AMIC_CFG_REG, 4, 0, NULL, 0), + + /* Auxiliary FM PGAs */ + SND_SOC_DAPM_PGA("APGA1", ISABELLE_APGA_CFG_REG, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("APGA2", ISABELLE_APGA_CFG_REG, 6, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC1", "Left Front Capture", + ISABELLE_AMIC_CFG_REG, 7, 0), + SND_SOC_DAPM_ADC("ADC2", "Right Front Capture", + ISABELLE_AMIC_CFG_REG, 6, 0), + + /* Microphone Bias */ + SND_SOC_DAPM_SUPPLY("Headset Mic Bias", ISABELLE_ABIAS_CFG_REG, + 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Main Mic Bias", ISABELLE_ABIAS_CFG_REG, + 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Digital Mic1 Bias", + ISABELLE_DBIAS_CFG_REG, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Digital Mic2 Bias", + ISABELLE_DBIAS_CFG_REG, 2, 0, NULL, 0), + + /* Mixers */ + SND_SOC_DAPM_MIXER("Headset Left Mixer", SND_SOC_NOPM, 0, 0, + isabelle_hs_left_mixer_controls, + ARRAY_SIZE(isabelle_hs_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Headset Right Mixer", SND_SOC_NOPM, 0, 0, + isabelle_hs_right_mixer_controls, + ARRAY_SIZE(isabelle_hs_right_mixer_controls)), + SND_SOC_DAPM_MIXER("Handsfree Left Mixer", SND_SOC_NOPM, 0, 0, + isabelle_hf_left_mixer_controls, + ARRAY_SIZE(isabelle_hf_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Handsfree Right Mixer", SND_SOC_NOPM, 0, 0, + isabelle_hf_right_mixer_controls, + ARRAY_SIZE(isabelle_hf_right_mixer_controls)), + SND_SOC_DAPM_MIXER("LINEOUT1 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_aux_left_mixer_controls, + ARRAY_SIZE(isabelle_aux_left_mixer_controls)), + SND_SOC_DAPM_MIXER("LINEOUT2 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_aux_right_mixer_controls, + ARRAY_SIZE(isabelle_aux_right_mixer_controls)), + SND_SOC_DAPM_MIXER("Earphone Mixer", SND_SOC_NOPM, 0, 0, + isabelle_ep_mixer_controls, + ARRAY_SIZE(isabelle_ep_mixer_controls)), + + SND_SOC_DAPM_MIXER("DPGA1L Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga1_left_mixer_controls, + ARRAY_SIZE(isabelle_dpga1_left_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA1R Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga1_right_mixer_controls, + ARRAY_SIZE(isabelle_dpga1_right_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA2L Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga2_left_mixer_controls, + ARRAY_SIZE(isabelle_dpga2_left_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA2R Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga2_right_mixer_controls, + ARRAY_SIZE(isabelle_dpga2_right_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA3L Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga3_left_mixer_controls, + ARRAY_SIZE(isabelle_dpga3_left_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA3R Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga3_right_mixer_controls, + ARRAY_SIZE(isabelle_dpga3_right_mixer_controls)), + + SND_SOC_DAPM_MIXER("RX1 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx1_mixer_controls, + ARRAY_SIZE(isabelle_rx1_mixer_controls)), + SND_SOC_DAPM_MIXER("RX2 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx2_mixer_controls, + ARRAY_SIZE(isabelle_rx2_mixer_controls)), + SND_SOC_DAPM_MIXER("RX3 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx3_mixer_controls, + ARRAY_SIZE(isabelle_rx3_mixer_controls)), + SND_SOC_DAPM_MIXER("RX4 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx4_mixer_controls, + ARRAY_SIZE(isabelle_rx4_mixer_controls)), + SND_SOC_DAPM_MIXER("RX5 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx5_mixer_controls, + ARRAY_SIZE(isabelle_rx5_mixer_controls)), + SND_SOC_DAPM_MIXER("RX6 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx6_mixer_controls, + ARRAY_SIZE(isabelle_rx6_mixer_controls)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC1L", "Headset Playback", ISABELLE_DAC_CFG_REG, + 5, 0), + SND_SOC_DAPM_DAC("DAC1R", "Headset Playback", ISABELLE_DAC_CFG_REG, + 4, 0), + SND_SOC_DAPM_DAC("DAC2L", "Handsfree Playback", ISABELLE_DAC_CFG_REG, + 3, 0), + SND_SOC_DAPM_DAC("DAC2R", "Handsfree Playback", ISABELLE_DAC_CFG_REG, + 2, 0), + SND_SOC_DAPM_DAC("DAC3L", "Lineout Playback", ISABELLE_DAC_CFG_REG, + 1, 0), + SND_SOC_DAPM_DAC("DAC3R", "Lineout Playback", ISABELLE_DAC_CFG_REG, + 0, 0), + + /* Analog Playback PGAs */ + SND_SOC_DAPM_PGA("Sidetone Audio PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Sidetone Voice PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HF Left PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HF Right PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA1L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA1R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA2L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA2R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA3L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA3R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Analog Playback Mux */ + SND_SOC_DAPM_MUX("RX1 Playback", ISABELLE_ALU_RX_EN_REG, 5, 0, + &rx1_mux_controls), + SND_SOC_DAPM_MUX("RX2 Playback", ISABELLE_ALU_RX_EN_REG, 4, 0, + &rx2_mux_controls), + + /* TX Select */ + SND_SOC_DAPM_MUX("ATX Select", ISABELLE_TX_INPUT_CFG_REG, + 7, 0, &atx_mux_controls), + SND_SOC_DAPM_MUX("VTX Select", ISABELLE_TX_INPUT_CFG_REG, + 6, 0, &vtx_mux_controls), + + SND_SOC_DAPM_SWITCH("Earphone Playback", SND_SOC_NOPM, 0, 0, + &ep_path_enable_control), + + /* Output Drivers */ + SND_SOC_DAPM_OUT_DRV("HS Left Driver", ISABELLE_HSDRV_CFG2_REG, + 1, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HS Right Driver", ISABELLE_HSDRV_CFG2_REG, + 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("LINEOUT1 Left Driver", ISABELLE_LINEAMP_CFG_REG, + 1, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("LINEOUT2 Right Driver", ISABELLE_LINEAMP_CFG_REG, + 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Earphone Driver", ISABELLE_EARDRV_CFG2_REG, + 1, 0, NULL, 0), + + SND_SOC_DAPM_OUT_DRV("HF Left Driver", ISABELLE_HFDRV_CFG_REG, + 1, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HF Right Driver", ISABELLE_HFDRV_CFG_REG, + 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route isabelle_intercon[] = { + /* Interface mapping */ + { "DL1", "DL12 Playback Switch", "INTF1_SDI" }, + { "DL2", "DL12 Playback Switch", "INTF1_SDI" }, + { "DL3", "DL34 Playback Switch", "INTF1_SDI" }, + { "DL4", "DL34 Playback Switch", "INTF1_SDI" }, + { "DL5", "DL56 Playback Switch", "INTF1_SDI" }, + { "DL6", "DL56 Playback Switch", "INTF1_SDI" }, + + { "DL1", "DL12 Playback Switch", "INTF2_SDI" }, + { "DL2", "DL12 Playback Switch", "INTF2_SDI" }, + { "DL3", "DL34 Playback Switch", "INTF2_SDI" }, + { "DL4", "DL34 Playback Switch", "INTF2_SDI" }, + { "DL5", "DL56 Playback Switch", "INTF2_SDI" }, + { "DL6", "DL56 Playback Switch", "INTF2_SDI" }, + + /* Input side mapping */ + { "Sidetone Audio PGA", NULL, "Sidetone Audio Playback" }, + { "Sidetone Voice PGA", NULL, "Sidetone Voice Playback" }, + + { "RX1 Mixer", "ST1 Playback Switch", "Sidetone Audio PGA" }, + + { "RX1 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" }, + { "RX1 Mixer", "DL1 Playback Switch", "DL1" }, + + { "RX2 Mixer", "ST2 Playback Switch", "Sidetone Audio PGA" }, + + { "RX2 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" }, + { "RX2 Mixer", "DL2 Playback Switch", "DL2" }, + + { "RX3 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" }, + { "RX3 Mixer", "DL3 Playback Switch", "DL3" }, + + { "RX4 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" }, + { "RX4 Mixer", "DL4 Playback Switch", "DL4" }, + + { "RX5 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" }, + { "RX5 Mixer", "DL5 Playback Switch", "DL5" }, + + { "RX6 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" }, + { "RX6 Mixer", "DL6 Playback Switch", "DL6" }, + + /* Capture path */ + { "Analog Left Capture Route", "Headset Mic", "HSMIC" }, + { "Analog Left Capture Route", "Main Mic", "MAINMIC" }, + { "Analog Left Capture Route", "Aux/FM Left", "LINEIN1" }, + + { "Analog Right Capture Route", "Sub Mic", "SUBMIC" }, + { "Analog Right Capture Route", "Aux/FM Right", "LINEIN2" }, + + { "MicAmp1", NULL, "Analog Left Capture Route" }, + { "MicAmp2", NULL, "Analog Right Capture Route" }, + + { "ADC1", NULL, "MicAmp1" }, + { "ADC2", NULL, "MicAmp2" }, + + { "ATX Select", "AMIC1", "ADC1" }, + { "ATX Select", "DMIC", "DMICDAT" }, + { "ATX Select", "AMIC2", "ADC2" }, + + { "VTX Select", "AMIC1", "ADC1" }, + { "VTX Select", "DMIC", "DMICDAT" }, + { "VTX Select", "AMIC2", "ADC2" }, + + { "ULATX1", "ATX1 Filter Enable Switch", "ATX Select" }, + { "ULATX1", "ATX1 Filter Bypass Switch", "ATX Select" }, + { "ULATX2", "ATX2 Filter Enable Switch", "ATX Select" }, + { "ULATX2", "ATX2 Filter Bypass Switch", "ATX Select" }, + + { "ULVTX1", "VTX1 Filter Enable Switch", "VTX Select" }, + { "ULVTX1", "VTX1 Filter Bypass Switch", "VTX Select" }, + { "ULVTX2", "VTX2 Filter Enable Switch", "VTX Select" }, + { "ULVTX2", "VTX2 Filter Bypass Switch", "VTX Select" }, + + { "INTF1_SDO", "ULATX12 Capture Switch", "ULATX1" }, + { "INTF1_SDO", "ULATX12 Capture Switch", "ULATX2" }, + { "INTF2_SDO", "ULATX12 Capture Switch", "ULATX1" }, + { "INTF2_SDO", "ULATX12 Capture Switch", "ULATX2" }, + + { "INTF1_SDO", NULL, "ULVTX1" }, + { "INTF1_SDO", NULL, "ULVTX2" }, + { "INTF2_SDO", NULL, "ULVTX1" }, + { "INTF2_SDO", NULL, "ULVTX2" }, + + /* AFM Path */ + { "APGA1", NULL, "LINEIN1" }, + { "APGA2", NULL, "LINEIN2" }, + + { "RX1 Playback", "VRX1 Filter Bypass Switch", "RX1 Mixer" }, + { "RX1 Playback", "ARX1 Filter Bypass Switch", "RX1 Mixer" }, + { "RX1 Playback", "RX1 Filter Enable Switch", "RX1 Mixer" }, + + { "RX2 Playback", "VRX2 Filter Bypass Switch", "RX2 Mixer" }, + { "RX2 Playback", "ARX2 Filter Bypass Switch", "RX2 Mixer" }, + { "RX2 Playback", "RX2 Filter Enable Switch", "RX2 Mixer" }, + + { "RX3 Playback", "ARX3 Filter Bypass Switch", "RX3 Mixer" }, + { "RX3 Playback", "RX3 Filter Enable Switch", "RX3 Mixer" }, + + { "RX4 Playback", "ARX4 Filter Bypass Switch", "RX4 Mixer" }, + { "RX4 Playback", "RX4 Filter Enable Switch", "RX4 Mixer" }, + + { "RX5 Playback", "ARX5 Filter Bypass Switch", "RX5 Mixer" }, + { "RX5 Playback", "RX5 Filter Enable Switch", "RX5 Mixer" }, + + { "RX6 Playback", "ARX6 Filter Bypass Switch", "RX6 Mixer" }, + { "RX6 Playback", "RX6 Filter Enable Switch", "RX6 Mixer" }, + + { "DPGA1L Mixer", "RX1 Playback Switch", "RX1 Playback" }, + { "DPGA1L Mixer", "RX3 Playback Switch", "RX3 Playback" }, + { "DPGA1L Mixer", "RX5 Playback Switch", "RX5 Playback" }, + + { "DPGA1R Mixer", "RX2 Playback Switch", "RX2 Playback" }, + { "DPGA1R Mixer", "RX4 Playback Switch", "RX4 Playback" }, + { "DPGA1R Mixer", "RX6 Playback Switch", "RX6 Playback" }, + + { "DPGA1L", NULL, "DPGA1L Mixer" }, + { "DPGA1R", NULL, "DPGA1R Mixer" }, + + { "DAC1L", NULL, "DPGA1L" }, + { "DAC1R", NULL, "DPGA1R" }, + + { "DPGA2L Mixer", "RX1 Playback Switch", "RX1 Playback" }, + { "DPGA2L Mixer", "RX2 Playback Switch", "RX2 Playback" }, + { "DPGA2L Mixer", "RX3 Playback Switch", "RX3 Playback" }, + { "DPGA2L Mixer", "RX4 Playback Switch", "RX4 Playback" }, + { "DPGA2L Mixer", "RX5 Playback Switch", "RX5 Playback" }, + { "DPGA2L Mixer", "RX6 Playback Switch", "RX6 Playback" }, + + { "DPGA2R Mixer", "RX2 Playback Switch", "RX2 Playback" }, + { "DPGA2R Mixer", "RX4 Playback Switch", "RX4 Playback" }, + { "DPGA2R Mixer", "RX6 Playback Switch", "RX6 Playback" }, + + { "DPGA2L", NULL, "DPGA2L Mixer" }, + { "DPGA2R", NULL, "DPGA2R Mixer" }, + + { "DAC2L", NULL, "DPGA2L" }, + { "DAC2R", NULL, "DPGA2R" }, + + { "DPGA3L Mixer", "RX1 Playback Switch", "RX1 Playback" }, + { "DPGA3L Mixer", "RX3 Playback Switch", "RX3 Playback" }, + { "DPGA3L Mixer", "RX5 Playback Switch", "RX5 Playback" }, + + { "DPGA3R Mixer", "RX2 Playback Switch", "RX2 Playback" }, + { "DPGA3R Mixer", "RX4 Playback Switch", "RX4 Playback" }, + { "DPGA3R Mixer", "RX6 Playback Switch", "RX6 Playback" }, + + { "DPGA3L", NULL, "DPGA3L Mixer" }, + { "DPGA3R", NULL, "DPGA3R Mixer" }, + + { "DAC3L", NULL, "DPGA3L" }, + { "DAC3R", NULL, "DPGA3R" }, + + { "Headset Left Mixer", "DAC1L Playback Switch", "DAC1L" }, + { "Headset Left Mixer", "APGA1 Playback Switch", "APGA1" }, + + { "Headset Right Mixer", "DAC1R Playback Switch", "DAC1R" }, + { "Headset Right Mixer", "APGA2 Playback Switch", "APGA2" }, + + { "HS Left Driver", NULL, "Headset Left Mixer" }, + { "HS Right Driver", NULL, "Headset Right Mixer" }, + + { "HSOL", NULL, "HS Left Driver" }, + { "HSOR", NULL, "HS Right Driver" }, + + /* Earphone playback path */ + { "Earphone Mixer", "DAC2L Playback Switch", "DAC2L" }, + { "Earphone Mixer", "APGA1 Playback Switch", "APGA1" }, + + { "Earphone Playback", "Switch", "Earphone Mixer" }, + { "Earphone Driver", NULL, "Earphone Playback" }, + { "EP", NULL, "Earphone Driver" }, + + { "Handsfree Left Mixer", "DAC2L Playback Switch", "DAC2L" }, + { "Handsfree Left Mixer", "APGA1 Playback Switch", "APGA1" }, + + { "Handsfree Right Mixer", "DAC2R Playback Switch", "DAC2R" }, + { "Handsfree Right Mixer", "APGA2 Playback Switch", "APGA2" }, + + { "HF Left PGA", NULL, "Handsfree Left Mixer" }, + { "HF Right PGA", NULL, "Handsfree Right Mixer" }, + + { "HF Left Driver", NULL, "HF Left PGA" }, + { "HF Right Driver", NULL, "HF Right PGA" }, + + { "HFL", NULL, "HF Left Driver" }, + { "HFR", NULL, "HF Right Driver" }, + + { "LINEOUT1 Mixer", "DAC3L Playback Switch", "DAC3L" }, + { "LINEOUT1 Mixer", "APGA1 Playback Switch", "APGA1" }, + + { "LINEOUT2 Mixer", "DAC3R Playback Switch", "DAC3R" }, + { "LINEOUT2 Mixer", "APGA2 Playback Switch", "APGA2" }, + + { "LINEOUT1 Driver", NULL, "LINEOUT1 Mixer" }, + { "LINEOUT2 Driver", NULL, "LINEOUT2 Mixer" }, + + { "LINEOUT1", NULL, "LINEOUT1 Driver" }, + { "LINEOUT2", NULL, "LINEOUT2 Driver" }, +}; + +static int isabelle_hs_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, ISABELLE_DAC1_SOFTRAMP_REG, + BIT(4), (mute ? BIT(4) : 0)); + + return 0; +} + +static int isabelle_hf_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, ISABELLE_DAC2_SOFTRAMP_REG, + BIT(4), (mute ? BIT(4) : 0)); + + return 0; +} + +static int isabelle_line_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, ISABELLE_DAC3_SOFTRAMP_REG, + BIT(4), (mute ? BIT(4) : 0)); + + return 0; +} + +static int isabelle_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct isabelle_priv *isabelle = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(isabelle->regmap); + + snd_soc_update_bits(codec, ISABELLE_PWR_EN_REG, + ISABELLE_CHIP_EN, BIT(0)); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, ISABELLE_PWR_EN_REG, + ISABELLE_CHIP_EN, 0); + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static int isabelle_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + u16 aif = 0; + unsigned int fs_val = 0; + + switch (params_rate(params)) { + case 8000: + fs_val = ISABELLE_FS_RATE_8; + break; + case 11025: + fs_val = ISABELLE_FS_RATE_11; + break; + case 12000: + fs_val = ISABELLE_FS_RATE_12; + break; + case 16000: + fs_val = ISABELLE_FS_RATE_16; + break; + case 22050: + fs_val = ISABELLE_FS_RATE_22; + break; + case 24000: + fs_val = ISABELLE_FS_RATE_24; + break; + case 32000: + fs_val = ISABELLE_FS_RATE_32; + break; + case 44100: + fs_val = ISABELLE_FS_RATE_44; + break; + case 48000: + fs_val = ISABELLE_FS_RATE_48; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ISABELLE_FS_RATE_CFG_REG, + ISABELLE_FS_RATE_MASK, fs_val); + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S20_3LE: + aif |= ISABELLE_AIF_LENGTH_20; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aif |= ISABELLE_AIF_LENGTH_32; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ISABELLE_INTF_CFG_REG, + ISABELLE_AIF_LENGTH_MASK, aif); + + return 0; +} + +static int isabelle_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int aif_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + aif_val &= ~ISABELLE_AIF_MS; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif_val |= ISABELLE_AIF_MS; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aif_val |= ISABELLE_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + aif_val |= ISABELLE_LEFT_J_MODE; + break; + case SND_SOC_DAIFMT_PDM: + aif_val |= ISABELLE_PDM_MODE; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ISABELLE_INTF_CFG_REG, + (ISABELLE_AIF_MS | ISABELLE_AIF_FMT_MASK), aif_val); + + return 0; +} + +/* Rates supported by Isabelle driver */ +#define ISABELLE_RATES SNDRV_PCM_RATE_8000_48000 + +/* Formates supported by Isabelle driver. */ +#define ISABELLE_FORMATS (SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops isabelle_hs_dai_ops = { + .hw_params = isabelle_hw_params, + .set_fmt = isabelle_set_dai_fmt, + .digital_mute = isabelle_hs_mute, +}; + +static struct snd_soc_dai_ops isabelle_hf_dai_ops = { + .hw_params = isabelle_hw_params, + .set_fmt = isabelle_set_dai_fmt, + .digital_mute = isabelle_hf_mute, +}; + +static struct snd_soc_dai_ops isabelle_line_dai_ops = { + .hw_params = isabelle_hw_params, + .set_fmt = isabelle_set_dai_fmt, + .digital_mute = isabelle_line_mute, +}; + +static struct snd_soc_dai_ops isabelle_ul_dai_ops = { + .hw_params = isabelle_hw_params, + .set_fmt = isabelle_set_dai_fmt, +}; + +/* ISABELLE dai structure */ +struct snd_soc_dai_driver isabelle_dai[] = { + { + .name = "isabelle-dl1", + .playback = { + .stream_name = "Headset Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ISABELLE_RATES, + .formats = ISABELLE_FORMATS, + }, + .ops = &isabelle_hs_dai_ops, + }, + { + .name = "isabelle-dl2", + .playback = { + .stream_name = "Handsfree Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ISABELLE_RATES, + .formats = ISABELLE_FORMATS, + }, + .ops = &isabelle_hf_dai_ops, + }, + { + .name = "isabelle-lineout", + .playback = { + .stream_name = "Lineout Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ISABELLE_RATES, + .formats = ISABELLE_FORMATS, + }, + .ops = &isabelle_line_dai_ops, + }, + { + .name = "isabelle-ul", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ISABELLE_RATES, + .formats = ISABELLE_FORMATS, + }, + .ops = &isabelle_ul_dai_ops, + }, +}; + +#ifdef CONFIG_PM +static int isabelle_suspend(struct snd_soc_codec *codec) +{ + isabelle_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int isabelle_resume(struct snd_soc_codec *codec) +{ + isabelle_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + isabelle_set_bias_level(codec, codec->dapm.suspend_bias_level); + + return 0; +} +#else +#define isabelle_suspend NULL +#define isabelle_resume NULL +#endif + +static int isabelle_probe(struct snd_soc_codec *codec) +{ + struct isabelle_priv *isabelle = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + codec->control_data = isabelle->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + return 0; +} + +/* power down chip */ +static int isabelle_remove(struct snd_soc_codec *codec) +{ + isabelle_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_isabelle = { + .probe = isabelle_probe, + .remove = isabelle_remove, + .suspend = isabelle_suspend, + .resume = isabelle_resume, + .set_bias_level = isabelle_set_bias_level, + .controls = isabelle_snd_controls, + .num_controls = ARRAY_SIZE(isabelle_snd_controls), + .dapm_widgets = isabelle_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(isabelle_dapm_widgets), + .dapm_routes = isabelle_intercon, + .num_dapm_routes = ARRAY_SIZE(isabelle_intercon), + .idle_bias_off = true, +}; + +static const struct regmap_config isabelle_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = ISABELLE_MAX_REGISTER, + .reg_defaults = isabelle_reg_defs, + .num_reg_defaults = ARRAY_SIZE(isabelle_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit isabelle_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct isabelle_priv *isabelle; + int ret = 0; + + isabelle = devm_kzalloc(&i2c->dev, sizeof(struct isabelle_priv), + GFP_KERNEL); + + if (isabelle == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, isabelle); + + isabelle->regmap = regmap_init_i2c(i2c, &isabelle_regmap_config); + if (IS_ERR(isabelle->regmap)) { + ret = PTR_ERR(isabelle->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_isabelle, isabelle_dai, + ARRAY_SIZE(isabelle_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + regmap_exit(isabelle->regmap); + return ret; + } + + return ret; +} + +static int __devexit isabelle_i2c_remove(struct i2c_client *client) +{ + struct isabelle_priv *isabelle = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(isabelle->regmap); + return 0; +} + +static const struct i2c_device_id isabelle_i2c_id[] = { + { "isabelle", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, isabelle_i2c_id); + +static struct i2c_driver isabelle_i2c_driver = { + .driver = { + .name = "isabelle", + .owner = THIS_MODULE, + }, + .probe = isabelle_i2c_probe, + .remove = __devexit_p(isabelle_i2c_remove), + .id_table = isabelle_i2c_id, +}; + +module_i2c_driver(isabelle_i2c_driver); + +MODULE_DESCRIPTION("ASoC ISABELLE driver"); +MODULE_AUTHOR("Vishwas A Deshpande <vishwas.a.deshpande(a)ti.com>"); +MODULE_AUTHOR("M R Swami Reddy <MR.Swami.Reddy(a)ti.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/isabelle.h b/sound/soc/codecs/isabelle.h new file mode 100644 index 0000000..96d839a --- /dev/null +++ b/sound/soc/codecs/isabelle.h @@ -0,0 +1,143 @@ +/* + * isabelle.h - Low power high fidelity audio codec driver header file + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + */ + +#ifndef _ISABELLE_H +#define _ISABELLE_H + +#include <linux/bitops.h> + +/* ISABELLE REGISTERS */ + +#define ISABELLE_PWR_CFG_REG 0x01 +#define ISABELLE_PWR_EN_REG 0x02 +#define ISABELLE_PS_EN1_REG 0x03 +#define ISABELLE_INT1_STATUS_REG 0x04 +#define ISABELLE_INT1_MASK_REG 0x05 +#define ISABELLE_INT2_STATUS_REG 0x06 +#define ISABELLE_INT2_MASK_REG 0x07 +#define ISABELLE_HKCTL1_REG 0x08 +#define ISABELLE_HKCTL2_REG 0x09 +#define ISABELLE_HKCTL3_REG 0x0A +#define ISABELLE_ACCDET_STATUS_REG 0x0B +#define ISABELLE_BUTTON_ID_REG 0x0C +#define ISABELLE_PLL_CFG_REG 0x10 +#define ISABELLE_PLL_EN_REG 0x11 +#define ISABELLE_FS_RATE_CFG_REG 0x12 +#define ISABELLE_INTF_CFG_REG 0x13 +#define ISABELLE_INTF_EN_REG 0x14 +#define ISABELLE_ULATX12_INTF_CFG_REG 0x15 +#define ISABELLE_DL12_INTF_CFG_REG 0x16 +#define ISABELLE_DL34_INTF_CFG_REG 0x17 +#define ISABELLE_DL56_INTF_CFG_REG 0x18 +#define ISABELLE_ATX_STPGA1_CFG_REG 0x19 +#define ISABELLE_ATX_STPGA2_CFG_REG 0x1A +#define ISABELLE_VTX_STPGA1_CFG_REG 0x1B +#define ISABELLE_VTX2_STPGA2_CFG_REG 0x1C +#define ISABELLE_ATX1_DPGA_REG 0x1D +#define ISABELLE_ATX2_DPGA_REG 0x1E +#define ISABELLE_VTX1_DPGA_REG 0x1F +#define ISABELLE_VTX2_DPGA_REG 0x20 +#define ISABELLE_TX_INPUT_CFG_REG 0x21 +#define ISABELLE_RX_INPUT_CFG_REG 0x22 +#define ISABELLE_RX_INPUT_CFG2_REG 0x23 +#define ISABELLE_VOICE_HPF_CFG_REG 0x24 +#define ISABELLE_AUDIO_HPF_CFG_REG 0x25 +#define ISABELLE_RX1_DPGA_REG 0x26 +#define ISABELLE_RX2_DPGA_REG 0x27 +#define ISABELLE_RX3_DPGA_REG 0x28 +#define ISABELLE_RX4_DPGA_REG 0x29 +#define ISABELLE_RX5_DPGA_REG 0x2A +#define ISABELLE_RX6_DPGA_REG 0x2B +#define ISABELLE_ALU_TX_EN_REG 0x2C +#define ISABELLE_ALU_RX_EN_REG 0x2D +#define ISABELLE_IIR_RESYNC_REG 0x2E +#define ISABELLE_ABIAS_CFG_REG 0x30 +#define ISABELLE_DBIAS_CFG_REG 0x31 +#define ISABELLE_MIC1_GAIN_REG 0x32 +#define ISABELLE_MIC2_GAIN_REG 0x33 +#define ISABELLE_AMIC_CFG_REG 0x34 +#define ISABELLE_DMIC_CFG_REG 0x35 +#define ISABELLE_APGA_GAIN_REG 0x36 +#define ISABELLE_APGA_CFG_REG 0x37 +#define ISABELLE_TX_GAIN_DLY_REG 0x38 +#define ISABELLE_RX_GAIN_DLY_REG 0x39 +#define ISABELLE_RX_PWR_CTRL_REG 0x3A +#define ISABELLE_DPGA1LR_IN_SEL_REG 0x3B +#define ISABELLE_DPGA1L_GAIN_REG 0x3C +#define ISABELLE_DPGA1R_GAIN_REG 0x3D +#define ISABELLE_DPGA2L_IN_SEL_REG 0x3E +#define ISABELLE_DPGA2R_IN_SEL_REG 0x3F +#define ISABELLE_DPGA2L_GAIN_REG 0x40 +#define ISABELLE_DPGA2R_GAIN_REG 0x41 +#define ISABELLE_DPGA3LR_IN_SEL_REG 0x42 +#define ISABELLE_DPGA3L_GAIN_REG 0x43 +#define ISABELLE_DPGA3R_GAIN_REG 0x44 +#define ISABELLE_DAC1_SOFTRAMP_REG 0x45 +#define ISABELLE_DAC2_SOFTRAMP_REG 0x46 +#define ISABELLE_DAC3_SOFTRAMP_REG 0x47 +#define ISABELLE_DAC_CFG_REG 0x48 +#define ISABELLE_EARDRV_CFG1_REG 0x49 +#define ISABELLE_EARDRV_CFG2_REG 0x4A +#define ISABELLE_HSDRV_GAIN_REG 0x4B +#define ISABELLE_HSDRV_CFG1_REG 0x4C +#define ISABELLE_HSDRV_CFG2_REG 0x4D +#define ISABELLE_HS_NG_CFG1_REG 0x4E +#define ISABELLE_HS_NG_CFG2_REG 0x4F +#define ISABELLE_LINEAMP_GAIN_REG 0x50 +#define ISABELLE_LINEAMP_CFG_REG 0x51 +#define ISABELLE_HFL_VOL_CTRL_REG 0x52 +#define ISABELLE_HFL_SFTVOL_CTRL_REG 0x53 +#define ISABELLE_HFL_LIM_CTRL_1_REG 0x54 +#define ISABELLE_HFL_LIM_CTRL_2_REG 0x55 +#define ISABELLE_HFR_VOL_CTRL_REG 0x56 +#define ISABELLE_HFR_SFTVOL_CTRL_REG 0x57 +#define ISABELLE_HFR_LIM_CTRL_1_REG 0x58 +#define ISABELLE_HFR_LIM_CTRL_2_REG 0x59 +#define ISABELLE_HF_MODE_REG 0x5A +#define ISABELLE_HFLPGA_CFG_REG 0x5B +#define ISABELLE_HFRPGA_CFG_REG 0x5C +#define ISABELLE_HFDRV_CFG_REG 0x5D +#define ISABELLE_PDMOUT_CFG1_REG 0x5E +#define ISABELLE_PDMOUT_CFG2_REG 0x5F +#define ISABELLE_PDMOUT_L_WM_REG 0x60 +#define ISABELLE_PDMOUT_R_WM_REG 0x61 +#define ISABELLE_HF_NG_CFG1_REG 0x62 +#define ISABELLE_HF_NG_CFG2_REG 0x63 + +/* ISABELLE_PWR_EN_REG (0x02h) */ +#define ISABELLE_CHIP_EN BIT(0) + +/* ISABELLE DAI FORMATS */ +#define ISABELLE_AIF_FMT_MASK 0x70 +#define ISABELLE_I2S_MODE 0x0 +#define ISABELLE_LEFT_J_MODE 0x1 +#define ISABELLE_PDM_MODE 0x2 + +#define ISABELLE_AIF_LENGTH_MASK 0x30 +#define ISABELLE_AIF_LENGTH_20 0x00 +#define ISABELLE_AIF_LENGTH_32 0x10 + +#define ISABELLE_AIF_MS 0x80 + +#define ISABELLE_FS_RATE_MASK 0xF +#define ISABELLE_FS_RATE_8 0x0 +#define ISABELLE_FS_RATE_11 0x1 +#define ISABELLE_FS_RATE_12 0x2 +#define ISABELLE_FS_RATE_16 0x4 +#define ISABELLE_FS_RATE_22 0x5 +#define ISABELLE_FS_RATE_24 0x6 +#define ISABELLE_FS_RATE_32 0x8 +#define ISABELLE_FS_RATE_44 0x9 +#define ISABELLE_FS_RATE_48 0xA + +#define ISABELLE_MAX_REGISTER 0xFF + +#endif

1 0

Re: [alsa-devel] [PATCH] netwinder: nw_gpio_lock is a raw_spinlock_t
by David Woodhouse 29 May '12

29 May '12

On Tue, 2012-05-29 at 12:06 +0200, Christian Dietrich wrote: > --- a/drivers/char/ds1620.c > +++ b/drivers/char/ds1620.c > @@ -74,21 +74,21 @@ static inline void netwinder_ds1620_reset(void) > > static inline void netwinder_lock(unsigned long *flags) > { > - spin_lock_irqsave(&nw_gpio_lock, *flags); > + raw_spin_lock_irqsave(&nw_gpio_lock, *flags); > } > > static inline void netwinder_unlock(unsigned long *flags) > { > - spin_unlock_irqrestore(&nw_gpio_lock, *flags); > + raw_spin_unlock_irqrestore(&nw_gpio_lock, *flags); > } If you were to make these functions public by shifting them into arch/arm/mach-footbridge/include/mach/hardware.h that would be a lot nicer, and other places wouldn't have to touch the raw spinlock directly. Also... while we're thinking about preemption and netwinder, note that the write enable is valid for only 2ms or so. So all the functions in dc21285.c that you just touched should probably *also* be disabling preemption when they're run on a netwinder, to ensure that that time doesn't expire before they actually get to run. -- dwmw2

1 0

Re: [alsa-devel] [PATCH] netwinder: nw_gpio_lock is a raw_spinlock_t
by David Woodhouse 29 May '12

29 May '12

On Tue, 2012-05-29 at 12:06 +0200, Christian Dietrich wrote: > Since nw_gpio_lock is a raw_spinlock_t it should be used with the > raw_spinlock_* functions and not the spinlock_* variants. Functionally > this is equivalent at the moment, because the raw_spinlock_t is the > first field of spinlock_t, and therefore &nw_gpio_lock == > &(nw_gpio_lock->rlock). But when other spinlock_t functions use other > field they read and write random memory. Hm, why are we exposing a raw spinlock to drivers? Should we export a helper function (or macro, I suppose) which does the appropriate locking *and* the GPIO operation? -- dwmw2

1 0

[alsa-devel] [PATCH] ucm: Handle repeat device and modifier enables/disables
by Arun Raghavan 29 May '12

29 May '12

Currently, enabling a device twice can cause it to be added to snd_use_case_mgr_t->active_devices twice, causing the list to become a loop and subsequent uses to result in an infinite loop. This patch makes sure we don't enable/disable a device twice, and avoid doing the same for modifiers. --- src/ucm/main.c | 62 ++++++++++++++++++++++++++++++------------------------- 1 files changed, 34 insertions(+), 28 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 76ca151..79901f8 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -565,6 +565,34 @@ static struct use_case_modifier * return NULL; } +long device_status(snd_use_case_mgr_t *uc_mgr, + const char *device_name) +{ + struct use_case_device *dev; + struct list_head *pos; + + list_for_each(pos, &uc_mgr->active_devices) { + dev = list_entry(pos, struct use_case_device, active_list); + if (strcmp(dev->name, device_name) == 0) + return 1; + } + return 0; +} + +long modifier_status(snd_use_case_mgr_t *uc_mgr, + const char *modifier_name) +{ + struct use_case_modifier *mod; + struct list_head *pos; + + list_for_each(pos, &uc_mgr->active_modifiers) { + mod = list_entry(pos, struct use_case_modifier, active_list); + if (strcmp(mod->name, modifier_name) == 0) + return 1; + } + return 0; +} + /** * \brief Set verb * \param uc_mgr Use case manager @@ -607,6 +635,9 @@ static int set_modifier(snd_use_case_mgr_t *uc_mgr, struct list_head *seq; int err; + if (modifier_status(uc_mgr, modifier->name) == enable) + return 0; + if (enable) { seq = &modifier->enable_list; } else { @@ -638,6 +669,9 @@ static int set_device(snd_use_case_mgr_t *uc_mgr, struct list_head *seq; int err; + if (device_status(uc_mgr, device->name) == enable) + return 0; + if (enable) { seq = &device->enable_list; } else { @@ -1316,34 +1350,6 @@ int snd_use_case_get(snd_use_case_mgr_t *uc_mgr, return err; } -long device_status(snd_use_case_mgr_t *uc_mgr, - const char *device_name) -{ - struct use_case_device *dev; - struct list_head *pos; - - list_for_each(pos, &uc_mgr->active_devices) { - dev = list_entry(pos, struct use_case_device, active_list); - if (strcmp(dev->name, device_name) == 0) - return 1; - } - return 0; -} - -long modifier_status(snd_use_case_mgr_t *uc_mgr, - const char *modifier_name) -{ - struct use_case_modifier *mod; - struct list_head *pos; - - list_for_each(pos, &uc_mgr->active_modifiers) { - mod = list_entry(pos, struct use_case_modifier, active_list); - if (strcmp(mod->name, modifier_name) == 0) - return 1; - } - return 0; -} - /** * \brief Get current - integer -- 1.7.8.6

3 4

[alsa-devel] [PATCH] ASoC: fsi: use dmaengine helper functions
by Kuninori Morimoto 29 May '12

29 May '12

This patch used dmaengine helper functions instead of using hand setting. And reduced local variables Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This is for v3.6 sound/soc/sh/fsi.c | 19 ++++--------------- 1 files changed, 4 insertions(+), 15 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 18c920f..53486ff 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1089,13 +1089,10 @@ static void fsi_dma_do_tasklet(unsigned long data) { struct fsi_stream *io = (struct fsi_stream *)data; struct fsi_priv *fsi = fsi_stream_to_priv(io); - struct dma_chan *chan; struct snd_soc_dai *dai; struct dma_async_tx_descriptor *desc; - struct scatterlist sg; struct snd_pcm_runtime *runtime; enum dma_data_direction dir; - dma_cookie_t cookie; int is_play = fsi_stream_is_play(fsi, io); int len; dma_addr_t buf; @@ -1104,7 +1101,6 @@ static void fsi_dma_do_tasklet(unsigned long data) return; dai = fsi_get_dai(io->substream); - chan = io->chan; runtime = io->substream->runtime; dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; len = samples_to_bytes(runtime, io->period_samples); @@ -1112,14 +1108,8 @@ static void fsi_dma_do_tasklet(unsigned long data) dma_sync_single_for_device(dai->dev, buf, len, dir); - sg_init_table(&sg, 1); - sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf)), - len , offset_in_page(buf)); - sg_dma_address(&sg) = buf; - sg_dma_len(&sg) = len; - - desc = dmaengine_prep_slave_sg(chan, &sg, 1, dir, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + desc = dmaengine_prep_slave_single(io->chan, buf, len, dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); return; @@ -1128,13 +1118,12 @@ static void fsi_dma_do_tasklet(unsigned long data) desc->callback = fsi_dma_complete; desc->callback_param = io; - cookie = desc->tx_submit(desc); - if (cookie < 0) { + if (dmaengine_submit(desc) < 0) { dev_err(dai->dev, "tx_submit() fail\n"); return; } - dma_async_issue_pending(chan); + dma_async_issue_pending(io->chan); /* * FIXME -- 1.7.5.4

1 0

Re: [alsa-devel] [PATCH 3/4] ASOC: mmp: add sspa support
by Mark Brown 29 May '12

29 May '12

On Fri, May 25, 2012 at 03:11:02PM +0800, Zhangfei Gao wrote: > The SSPA is a configurable multi-channel audio serial (TDM) interface. > It's configurable at runtime to support up to 128 channels and the > number of bits per sample: 8, 12, 16, 20, 24 and 32 bits. It also > support stereo format: I2S, left-justified or right-justified. Mostly looks good. A few fairly minor things... > +static int mmp_sspa_startup(struct snd_pcm_substream *substream, > + struct snd_soc_dai *dai) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; > + struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai); > + struct ssp_device *sspa = sspa_priv->sspa; > + int ret = 0; > + > + if (!cpu_dai->active) { > + clk_enable(sysclk); > + clk_enable(sspa->clk); > + } The clock API is refcounted so you shouldn't need to worry about multiple enables, you should just be able to unconditionally enable and disable. If this is needed we probably have a problem we should fix. > + switch (pll_id) { > + case MMP_SYSCLK: > + clk_set_rate(sysclk, freq_out); > + break; > + case MMP_SSPA_CLK: > + clk_set_rate(sspa->clk, freq_out); > + break; You're ignoring the return values here. > + priv->sspa->clk = clk_get(&pdev->dev, NULL); > + if (IS_ERR(priv->sspa->clk)) { > + ret = PTR_ERR(priv->sspa->clk); > + goto err_free_clk; > + } devm_clk_get(). > + res = request_mem_region(res->start, resource_size(res), > + pdev->name); > + if (res == NULL) { > + dev_err(&pdev->dev, "failed to request memory resource\n"); > + ret = -EBUSY; > + goto err_get_res; > + } > + > + priv->sspa->mmio_base = ioremap(res->start, resource_size(res)); > + if (priv->sspa->mmio_base == NULL) { > + dev_err(&pdev->dev, "failed to ioremap() registers\n"); > + ret = -ENODEV; > + goto err_free_mem; > + } devm_request_and_ioremap(). > +err_free_clk: > + devm_kfree(&pdev->dev, priv->dma_params); > +err_priv_dma_params: > + devm_kfree(&pdev->dev, priv->sspa); > +err_priv_sspa: > + devm_kfree(&pdev->dev, priv); The whole point of the devm_ stuff is that you don't need to do stuff like this.

2 1

[alsa-devel] [PATCH 15/15] ASoC: mxs-saif: Use clk_prepare_enable/clk_disable_unprepare
by Fabio Estevam 28 May '12

28 May '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Prepare the clock before enabling it. Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: <alsa-devel(a)alsa-project.org> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index aba71bf..15ad61a 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -482,7 +482,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: dev_dbg(cpu_dai->dev, "start\n"); - clk_enable(master_saif->clk); + clk_prepare_enable(master_saif->clk); if (!master_saif->mclk_in_use) __raw_writel(BM_SAIF_CTRL_RUN, master_saif->base + SAIF_CTRL + MXS_SET_ADDR); @@ -492,7 +492,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, * itself clk for its internal basic logic to work. */ if (saif != master_saif) { - clk_enable(saif->clk); + clk_prepare_enable(saif->clk); __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_SET_ADDR); } @@ -534,13 +534,13 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, master_saif->base + SAIF_CTRL + MXS_CLR_ADDR); udelay(delay); } - clk_disable(master_saif->clk); + clk_disable_unprepare(master_saif->clk); if (saif != master_saif) { __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_CLR_ADDR); udelay(delay); - clk_disable(saif->clk); + clk_disable_unprepare(saif->clk); } master_saif->ongoing = 0; -- 1.7.1

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Re: [alsa-devel] [PATCH 4/4] ASoC: add mmp brownstone support
by Mark Brown 28 May '12

28 May '12

On Fri, May 25, 2012 at 03:11:03PM +0800, Zhangfei Gao wrote: > +config SND_MMP_SOC_BROWNSTONE > + tristate "SoC Audio support for Marvell Brownstone" > + depends on SND_MMP_SOC && MACH_BROWNSTONE > + select SND_MMP_SOC_SSPA > + select SND_SOC_WM8994 Should depend on MFD_WM8994. > +static void brownstone_ext_control(struct snd_soc_dapm_context *dapm) > +{ This stuff is really unexpected in a modern machine driver, it was used on things like spitz due to the odd wiring but for modern stuff I'd not expect to see it. > + if (brownstone_spk_func == BROWNSTONE_SPK_ON) { > + snd_soc_dapm_enable_pin(dapm, "Ext Left Spk"); > + snd_soc_dapm_enable_pin(dapm, "Ext Right Spk"); > + } else { > + snd_soc_dapm_disable_pin(dapm, "Ext Left Spk"); > + snd_soc_dapm_disable_pin(dapm, "Ext Right Spk"); > + } Just define a single widget for the speakers and use a SND_SOC_DAPM_PIN_SWITCH(). > + /* set up jack connection */ > + switch (brownstone_jack_func) { > + case BROWNSTONE_HP: > + snd_soc_dapm_disable_pin(dapm, "Headset Mic"); > + snd_soc_dapm_enable_pin(dapm, "Main Mic"); > + snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); > + break; This should all be autodetectable, users having to manually select it is *very* unusual. Is there really no accessory detection hardware on the board? > + snd_soc_dapm_enable_pin(dapm, "Ext Left Spk"); > + snd_soc_dapm_enable_pin(dapm, "Ext Right Spk"); > + snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); > + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); > + snd_soc_dapm_enable_pin(dapm, "Main Mic"); Everything is enable dby default. > + /* turn on micbias 1/2 always */ > + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, > + WM8994_MICB1_ENA_MASK | > + WM8994_MICB2_ENA_MASK, > + WM8994_MICB1_ENA | > + WM8994_MICB2_ENA); If you need to do this force enable them with DAPM, this won't work anyway as the widgets will be powered off as soon as DAPM notices they're on. > + snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | > + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); > + > + snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | > + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); Set this in the dai_link. > +static struct platform_driver mmp_driver = { > + .driver = { > + .name = "mmp-audio", Should probably be something like "brownstone-audio".

1 0

Re: [alsa-devel] [PATCH] ASoC: imx-ssi: Use clk_prepare_enable/clk_disable_unprepare
by Mark Brown 28 May '12

28 May '12

On Fri, May 25, 2012 at 05:27:20PM -0300, Fabio Estevam wrote: > Fix the following kernel crash: Applied, thanks.

1 0

[alsa-devel] [PATCH 3/5] ASoC: Ux500: Add platform-driver
by Ola Lilja 28 May '12

28 May '12

Add platform-driver handling all DMA-activities. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- sound/soc/ux500/Kconfig | 7 + sound/soc/ux500/Makefile | 3 + sound/soc/ux500/ux500_pcm.c | 318 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/ux500/ux500_pcm.h | 35 +++++ 4 files changed, 363 insertions(+), 0 deletions(-) create mode 100644 sound/soc/ux500/ux500_pcm.c create mode 100644 sound/soc/ux500/ux500_pcm.h diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index 44cf434..1d38515 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -12,3 +12,10 @@ menuconfig SND_SOC_UX500 config SND_SOC_UX500_PLAT_MSP_I2S tristate depends on SND_SOC_UX500 + +config SND_SOC_UX500_PLAT_DMA + tristate "Platform - DB8500 (DMA)" + depends on SND_SOC_UX500 + select SND_SOC_DMAENGINE_PCM + help + Say Y if you want to enable the Ux500 platform-driver. diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile index 19974c5..4634bf0 100644 --- a/sound/soc/ux500/Makefile +++ b/sound/soc/ux500/Makefile @@ -2,3 +2,6 @@ snd-soc-ux500-plat-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o + +snd-soc-ux500-plat-dma-objs := ux500_pcm.o +obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c new file mode 100644 index 0000000..66b080e --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.c @@ -0,0 +1,318 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <asm/page.h> + +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <linux/dmaengine.h> +#include <linux/slab.h> + +#include <plat/ste_dma40.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "ux500_msp_i2s.h" +#include "ux500_pcm.h" + +static struct snd_pcm_hardware ux500_pcm_hw_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_PLAYBACK, + .rate_max = UX500_PLATFORM_MAX_RATE_PLAYBACK, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static struct snd_pcm_hardware ux500_pcm_hw_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_CAPTURE, + .rate_max = UX500_PLATFORM_MAX_RATE_CAPTURE, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static void ux500_pcm_dma_hw_free(struct device *dev, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + + if (runtime->dma_area == NULL) + return; + + if (buf != &substream->dma_buffer) { + dma_free_coherent(buf->dev.dev, buf->bytes, buf->area, + buf->addr); + kfree(runtime->dma_buffer_p); + } + + snd_pcm_set_runtime_buffer(substream, NULL); +} + +static int ux500_pcm_open(struct snd_pcm_substream *substream) +{ + int stream_id = substream->pstr->stream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct device *dev = dai->dev; + int ret; + struct ux500_msp_dma_params *dma_params; + u16 per_data_width, mem_data_width; + struct stedma40_chan_cfg *dma_cfg; + + dev_dbg(dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, + snd_pcm_stream_str(substream)); + + dev_dbg(dev, "%s: Set runtime hwparams.\n", __func__); + if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_set_runtime_hwparams(substream, + &ux500_pcm_hw_playback); + else + snd_soc_set_runtime_hwparams(substream, + &ux500_pcm_hw_capture); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(dev, "%s: Error: snd_pcm_hw_constraints failed (%d)\n", + __func__, ret); + return ret; + } + + dev_dbg(dev, "%s: Set hw-struct for %s.\n", __func__, + snd_pcm_stream_str(substream)); + runtime->hw = (stream_id == SNDRV_PCM_STREAM_PLAYBACK) ? + ux500_pcm_hw_playback : ux500_pcm_hw_capture; + + mem_data_width = STEDMA40_HALFWORD_WIDTH; + + dma_params = snd_soc_dai_get_dma_data(dai, substream); + switch (dma_params->data_size) { + case 32: + per_data_width = STEDMA40_WORD_WIDTH; + break; + case 16: + per_data_width = STEDMA40_HALFWORD_WIDTH; + break; + case 8: + per_data_width = STEDMA40_BYTE_WIDTH; + break; + default: + per_data_width = STEDMA40_WORD_WIDTH; + dev_warn(rtd->platform->dev, + "%s: Unknown data-size (%d)! Assuming 32 bits.\n", + __func__, dma_params->data_size); + } + + dma_cfg = dma_params->dma_cfg; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dma_cfg->src_info.data_width = mem_data_width; + dma_cfg->dst_info.data_width = per_data_width; + } else { + dma_cfg->src_info.data_width = per_data_width; + dma_cfg->dst_info.data_width = mem_data_width; + } + + + ret = snd_dmaengine_pcm_open(substream, stedma40_filter, dma_cfg); + if (ret) { + dev_dbg(dai->dev, + "%s: ERROR: snd_dmaengine_pcm_open failed (%d)!\n", + __func__, ret); + return ret; + } + + snd_dmaengine_pcm_set_data(substream, dma_cfg); + + return 0; +} + +static int ux500_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + + dev_dbg(dai->dev, "%s: Enter\n", __func__); + + snd_dmaengine_pcm_close(substream); + + return 0; +} + +static int ux500_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret = 0; + int size; + + dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__); + + size = params_buffer_bytes(hw_params); + + if (buf) { + if (buf->bytes >= size) + goto out; + ux500_pcm_dma_hw_free(NULL, substream); + } + + if (substream->dma_buffer.area != NULL && + substream->dma_buffer.bytes >= size) { + buf = &substream->dma_buffer; + } else { + buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL); + if (!buf) + goto nomem; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = NULL; + buf->area = dma_alloc_coherent(NULL, size, &buf->addr, + GFP_KERNEL); + buf->bytes = size; + buf->private_data = NULL; + + if (!buf->area) + goto free; + } + snd_pcm_set_runtime_buffer(substream, buf); + ret = 1; + out: + runtime->dma_bytes = size; + return ret; + + free: + kfree(buf); + nomem: + return -ENOMEM; +} + +static int ux500_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__); + + ux500_pcm_dma_hw_free(NULL, substream); + + return 0; +} + +static int ux500_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->platform->dev, "%s: Enter.\n", __func__); + + return dma_mmap_coherent(NULL, vma, runtime->dma_area, + runtime->dma_addr, runtime->dma_bytes); +} + +static struct snd_pcm_ops ux500_pcm_ops = { + .open = ux500_pcm_open, + .close = ux500_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = ux500_pcm_hw_params, + .hw_free = ux500_pcm_hw_free, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = ux500_pcm_mmap +}; + +int ux500_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + + dev_dbg(rtd->platform->dev, "%s: Enter (id = '%s').\n", __func__, + pcm->id); + + pcm->info_flags = 0; + + return 0; +} + +static struct snd_soc_platform_driver ux500_pcm_soc_drv = { + .ops = &ux500_pcm_ops, + .pcm_new = ux500_pcm_new, +}; + +static int __devexit ux500_pcm_drv_probe(struct platform_device *pdev) +{ + int ret; + + ret = snd_soc_register_platform(&pdev->dev, &ux500_pcm_soc_drv); + if (ret < 0) { + dev_err(&pdev->dev, + "%s: ERROR: Failed to register platform '%s' (%d)!\n", + __func__, pdev->name, ret); + return ret; + } + + return 0; +} + +static int __devinit ux500_pcm_drv_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static struct platform_driver ux500_pcm_driver = { + .driver = { + .name = "ux500-pcm", + .owner = THIS_MODULE, + }, + + .probe = ux500_pcm_drv_probe, + .remove = __devexit_p(ux500_pcm_drv_remove), +}; +module_platform_driver(ux500_pcm_driver); + +MODULE_LICENSE("GPLv2"); diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h new file mode 100644 index 0000000..77ed44d --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.h @@ -0,0 +1,35 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ +#ifndef UX500_PCM_H +#define UX500_PCM_H + +#include <asm/page.h> + +#include <linux/workqueue.h> + +#define UX500_PLATFORM_MIN_RATE_PLAYBACK 8000 +#define UX500_PLATFORM_MAX_RATE_PLAYBACK 48000 +#define UX500_PLATFORM_MIN_RATE_CAPTURE 8000 +#define UX500_PLATFORM_MAX_RATE_CAPTURE 48000 + +#define UX500_PLATFORM_MIN_CHANNELS 1 +#define UX500_PLATFORM_MAX_CHANNELS 8 + +#define UX500_PLATFORM_PERIODS_BYTES_MIN 128 +#define UX500_PLATFORM_PERIODS_BYTES_MAX (64 * PAGE_SIZE) +#define UX500_PLATFORM_PERIODS_MIN 2 +#define UX500_PLATFORM_PERIODS_MAX 48 +#define UX500_PLATFORM_BUFFER_BYTES_MAX (2048 * PAGE_SIZE) + +#endif -- 1.7.8.3

2 1

[alsa-devel] [PATCH 2/5] ASoC: core: Add widget SND_SOC_DAPM_CLOCK_SUPPLY
by Ola Lilja 28 May '12

28 May '12

Adds a supply-widget variant for connection to the clock-framework. This widget-type corresponds to the variant for regulators. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- include/sound/soc-dapm.h | 10 ++++++++++ sound/soc/soc-dapm.c | 38 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 48 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e3833d9..05559e5 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -229,6 +229,10 @@ struct device; { .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert, \ .event = wevent, .event_flags = wflags} +#define SND_SOC_DAPM_CLOCK_SUPPLY(wname) \ +{ .id = snd_soc_dapm_clock_supply, .name = wname, \ + .reg = SND_SOC_NOPM, .event = dapm_clock_event, \ + .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } /* generic widgets */ #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \ @@ -245,6 +249,7 @@ struct device; .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } + /* dapm kcontrol types */ #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -327,6 +332,8 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +int dapm_clock_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, @@ -432,6 +439,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_post, /* machine specific post widget - exec last */ snd_soc_dapm_supply, /* power/clock supply */ snd_soc_dapm_regulator_supply, /* external regulator */ + snd_soc_dapm_clock_supply, /* external clock */ snd_soc_dapm_aif_in, /* audio interface input */ snd_soc_dapm_aif_out, /* audio interface output */ snd_soc_dapm_siggen, /* signal generator */ @@ -537,6 +545,8 @@ struct snd_soc_dapm_widget { struct list_head dirty; int inputs; int outputs; + + struct clk *clk; }; struct snd_soc_dapm_update { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 90ee77d..3bb7a6f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -35,6 +35,7 @@ #include <linux/debugfs.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> +#include <linux/clk.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -51,6 +52,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, [snd_soc_dapm_supply] = 1, [snd_soc_dapm_regulator_supply] = 1, + [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_micbias] = 2, [snd_soc_dapm_dai_link] = 2, [snd_soc_dapm_dai] = 3, @@ -92,6 +94,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_aif_out] = 10, [snd_soc_dapm_dai] = 10, [snd_soc_dapm_dai_link] = 11, + [snd_soc_dapm_clock_supply] = 12, [snd_soc_dapm_regulator_supply] = 12, [snd_soc_dapm_supply] = 12, [snd_soc_dapm_post] = 13, @@ -391,6 +394,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_vmid: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: case snd_soc_dapm_dai: @@ -764,6 +768,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: return 0; default: break; @@ -850,6 +855,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: return 0; default: break; @@ -996,6 +1002,24 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(dapm_regulator_event); +/* + * Handler for clock supply widget. + */ +int dapm_clock_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (!w->clk) + return -EIO; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + return clk_enable(w->clk); + } else { + clk_disable(w->clk); + return 0; + } +} +EXPORT_SYMBOL_GPL(dapm_clock_event); + static int dapm_widget_power_check(struct snd_soc_dapm_widget *w) { if (w->power_checked) @@ -1487,6 +1511,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, switch (w->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: /* Supplies can't affect their outputs, only their inputs */ break; default: @@ -1587,6 +1612,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_micbias: if (d->target_bias_level < SND_SOC_BIAS_STANDBY) d->target_bias_level = SND_SOC_BIAS_STANDBY; @@ -1941,6 +1967,7 @@ static ssize_t dapm_widget_show(struct device *dev, case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: if (w->name) count += sprintf(buf + count, "%s: %s\n", w->name, w->power ? "On":"Off"); @@ -2187,6 +2214,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_post: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: case snd_soc_dapm_dai: @@ -2873,6 +2901,15 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, return NULL; } break; + case snd_soc_dapm_clock_supply: + w->clk = clk_get(dapm->dev, w->name); + if (IS_ERR(w->clk)) { + ret = PTR_ERR(w->clk); + dev_err(dapm->dev, "Failed to request %s: %d\n", + w->name, ret); + return NULL; + } + break; default: break; } @@ -2924,6 +2961,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: w->power_check = dapm_supply_check_power; break; case snd_soc_dapm_dai: -- 1.7.8.3

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[alsa-devel] [PATCH 1/5] ALSA: pcm: Add debug-print helper function
by Ola Lilja 28 May '12

28 May '12

Adds a function getting the stream-name as a string for a specific stream. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- include/sound/pcm.h | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 0d11128..a55d5db 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1073,4 +1073,15 @@ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) const char *snd_pcm_format_name(snd_pcm_format_t format); +/** + * Get a string naming the direction of a stream + */ +static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return "Playback"; + else + return "Capture"; +} + #endif /* __SOUND_PCM_H */ -- 1.7.8.3

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Re: [alsa-devel] Adjust volume may cause audio playback corruption
by cee1 28 May '12

28 May '12

2012/5/12 cee1 <fykcee1(a)gmail.com>: > Hi all, > > We're using PA on our product, and sometimes audio playback may > corrupt after adjusting volume. > > We found a way to (relatively) easily reproduce the problem: > 1. Play sample.mp3 through totem in repeat mode. > 2. In gnome-volume-control, Hardware tap, repeat to shift between "Off > profile" and "Analog Stereo Duplex profile". > > When the problem happens, the corruption continues until I: > * Do one more shift between "Off profile" and "Analog Stereo Duplex profile". > * Close totem (close all PA playback clients). > * Adjust volume. > > When the problem happens, open another PA client doing audio playback, > is also corrupt. > > Some additional information: > * Output of alsa-info.sh on our product: > http://dev.lemote.com/files/upload/software/PA-apc/alsa-info-lemote > * PA daemon log: > http://dev.lemote.com/files/upload/software/PA-apc/pulseaudio.log > * The corrupted sound: > http://dev.lemote.com/files/upload/software/PA-apc/corrupted_sound.ogg > * The sample mp3: http://dev.lemote.com/files/upload/software/PA-apc/sample.mp3 When the problem happens today, I notice: 1) PA complains "alsa-source.c: Overrun!". I doesn't run any recording application then. Curious. 2) Playback becomes good automatically when play sample.mp3 again(note totem is in repeat playing mode). It says sink becomes idle(I didn't touch totem, so totem should keep playing back). Any idea? Following is PA daemon log: I: [alsa-source] ratelimit.c: 1226 events suppressed I: [alsa-source] alsa-source.c: Overrun! I: [alsa-source] alsa-source.c: Increasing minimal latency to 1.00 ms D: [alsa-source] alsa-source.c: latency set to 20.00ms D: [alsa-source] alsa-source.c: hwbuf_unused=62008 D: [alsa-source] alsa-source.c: setting avail_min=442 I: [alsa-source] alsa-source.c: Overrun! I: [alsa-source] alsa-source.c: Increasing minimal latency to 2.00 ms D: [alsa-source] alsa-source.c: latency set to 20.00ms D: [alsa-source] alsa-source.c: hwbuf_unused=62008 D: [alsa-source] alsa-source.c: setting avail_min=442 I: [alsa-source] alsa-source.c: Overrun! I: [alsa-source] alsa-source.c: Increasing minimal latency to 4.00 ms D: [alsa-source] alsa-source.c: latency set to 20.00ms D: [alsa-source] alsa-source.c: hwbuf_unused=62008 D: [alsa-source] alsa-source.c: setting avail_min=442 I: [alsa-source] alsa-source.c: Overrun! I: [alsa-source] alsa-source.c: Increasing minimal latency to 8.00 ms D: [alsa-source] alsa-source.c: latency set to 20.00ms D: [alsa-source] alsa-source.c: hwbuf_unused=62008 D: [alsa-source] alsa-source.c: setting avail_min=442 I: [alsa-source] alsa-source.c: Overrun! I: [alsa-source] alsa-source.c: Increasing minimal latency to 16.00 ms D: [alsa-source] alsa-source.c: latency set to 20.00ms D: [alsa-source] alsa-source.c: hwbuf_unused=62008 D: [alsa-source] alsa-source.c: setting avail_min=442 I: [alsa-source] alsa-source.c: Overrun! I: [alsa-source] alsa-source.c: Increasing minimal latency to 26.00 ms D: [alsa-source] alsa-source.c: latency set to 26.00ms D: [alsa-source] alsa-source.c: hwbuf_unused=60952 D: [alsa-source] alsa-source.c: setting avail_min=706 D: [alsa-sink] protocol-native.c: Underrun on 'Playback Stream', 0 bytes in queue. D: [alsa-sink] protocol-native.c: Requesting rewind due to rewrite. D: [alsa-sink] sink-input.c: Requesting rewind due to corking D: [alsa-sink] alsa-sink.c: Requested to rewind 16208 bytes. D: [alsa-sink] alsa-sink.c: Limited to 15096 bytes. D: [alsa-sink] alsa-sink.c: before: 3774 D: [alsa-sink] alsa-sink.c: after: 3642 D: [alsa-sink] alsa-sink.c: Rewound 14568 bytes. D: [alsa-sink] sink.c: Processing rewind... D: [alsa-sink] sink.c: latency = 4156 D: [alsa-sink] sink-input.c: Have to rewind 14568 bytes on render memblockq. D: [alsa-sink] sink-input.c: Have to rewind 29136 bytes on implementor. D: [alsa-sink] source.c: Processing rewind... D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds. D: [alsa-sink] protocol-native.c: Requesting rewind due to end of underrun. D: [alsa-sink] protocol-native.c: Requesting rewind due to end of underrun. D: [alsa-sink] protocol-native.c: Requesting rewind due to end of underrun. D: [alsa-sink] protocol-native.c: Requesting rewind due to end of underrun. D: [alsa-sink] protocol-native.c: Requesting rewind due to end of underrun. D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0004, suspending I: [alsa-sink] alsa-sink.c: Device suspended... I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 44100 I: [pulseaudio] sink.c: Changed sampling rate successfully D: [alsa-sink] protocol-native.c: Requesting rewind due to end of underrun. D: [alsa-sink] sink-input.c: Requesting rewind due to uncorking D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming I: [alsa-sink] alsa-sink.c: Trying resume... I: [alsa-sink] alsa-util.c: Trying to disable ALSA period wakeups, using timers only D: [alsa-sink] alsa-util.c: Maximum hw buffer size is 371 ms D: [alsa-sink] alsa-util.c: Set buffer size first (to 16384 samples), period size second (to 8192 samples). I: [alsa-sink] alsa-util.c: ALSA period wakeups disabled D: [alsa-sink] alsa-sink.c: Latency set to 90.00ms D: [alsa-sink] alsa-sink.c: hwbuf_unused=49660 D: [alsa-sink] alsa-sink.c: setting avail_min=15503 I: [alsa-sink] alsa-sink.c: Time scheduling watermark is 20.00ms I: [alsa-sink] alsa-sink.c: Resumed successfully... D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. I: [alsa-sink] alsa-sink.c: Starting playback. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy. D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. > > We've tried PA 1.0-0ubuntu3, 1.1-0ubuntu15 and 2.0, the problem continues. > > The problem can also be reproduced on an AMD turion 64 x2 board with > ubuntu 10.10 and ubuntu 12.04: > * Output of alsa-info: > ubuntu 10.10(32bit) > http://dev.lemote.com/files/upload/software/PA-apc/alsa-info-ubuntu-10.10 > ubuntu 12.04(32bit) > http://dev.lemote.com/files/upload/software/PA-apc/alsa-info-ubuntu-12.04 > > Note: > * The AMD turion 64 x2 board uses the same chipset as our product. > * gnome-volume-control is removed in ubuntu 12.04, thus we compiled > it: http://dev.lemote.com/files/upload/software/PA-apc/gnome-volume-control -- Regards, - cee1

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Re: [alsa-devel] [PATCH] ASoC: imx-ssi: Use clk_prepare_enable()
by Sascha Hauer 25 May '12

25 May '12

Hi Fabio, On Fri, May 25, 2012 at 05:06:50PM -0300, Fabio Estevam wrote: > Fix the following kernel crash: > > This applies against linux-next. > > sound/soc/fsl/imx-ssi.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c > index cf3ed03..9027a83 100644 > --- a/sound/soc/fsl/imx-ssi.c > +++ b/sound/soc/fsl/imx-ssi.c > @@ -543,7 +543,7 @@ static int imx_ssi_probe(struct platform_device *pdev) > ret); > goto failed_clk; > } > - clk_enable(ssi->clk); > + clk_prepare_enable(ssi->clk); You should use clk_disable_unprepare then aswell. Sascha -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

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Re: [alsa-devel] [PATCH] [emu10k1]: remove the kcallloc cast (reported from make coccicheck's drop_kmalloc_cast.cocci)
by Takashi Iwai 25 May '12

25 May '12

At Thu, 24 May 2012 19:32:29 +0530, Devendra Naga wrote: > > Signed-off-by: Devendra Naga <devendra.aaru(a)gmail.com> I don't think removing it improves anything much. It's cast to __user pointer, and this rather helps to catch the caution. thanks, Takashi > --- > sound/pci/emu10k1/emufx.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c > index dae4050..272f8ac 100644 > --- a/sound/pci/emu10k1/emufx.c > +++ b/sound/pci/emu10k1/emufx.c > @@ -1815,7 +1815,7 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) > > if ((icode = kzalloc(sizeof(*icode), GFP_KERNEL)) == NULL) > return -ENOMEM; > - if ((icode->gpr_map = (u_int32_t __user *) > + if ((icode->gpr_map = > kcalloc(256 + 160 + 160 + 2 * 512, sizeof(u_int32_t), > GFP_KERNEL)) == NULL || > (controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, > -- > 1.7.9.5 >

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[alsa-devel] [PATCH 1/2] ALSA: update sync header when streams are linked/unlinked
by Pierre-Louis Bossart 25 May '12

25 May '12

Previous code only reported card number and was not updated when devices were linked/unlinked. This makes alsa-lib snd_pcm_info_get_sync totally useless. Add hooks to force update of sync header when devices are linked/unlinked, and provide more information such as number of devices and indices of capture/playback devices linked to Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/core/pcm_lib.c | 37 ++++++++++++++++++++++++++++++++----- sound/core/pcm_native.c | 6 ++++++ 2 files changed, 38 insertions(+), 5 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index faedb14..ae46d75 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -533,11 +533,38 @@ EXPORT_SYMBOL(snd_pcm_set_ops); void snd_pcm_set_sync(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->sync.id32[0] = substream->pcm->card->number; - runtime->sync.id32[1] = -1; - runtime->sync.id32[2] = -1; - runtime->sync.id32[3] = -1; + struct snd_pcm_substream *s; + int dev_c, dev_p; + + if (snd_pcm_stream_linked(substream)) { + runtime->sync.id32[0] = substream->pcm->card->number; + /* save number of devices linked */ + runtime->sync.id32[1] = substream->group->count-1; + dev_c = dev_p = 0; + + snd_pcm_group_for_each_entry(s, substream) { + if (s == substream) + continue; + if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dev_p++; + if (dev_p == 1) + runtime->sync.id32[2] = 0; + /* save mask of linked playback devices */ + runtime->sync.id32[2] |= (1<<s->number); + } else { + dev_c++; + if (dev_c == 1) + runtime->sync.id32[3] = 0; + /* save mask of linked capture devices */ + runtime->sync.id32[3] |= (1<<s->number); + } + } + } else { + runtime->sync.id32[0] = -1; + runtime->sync.id32[1] = -1; + runtime->sync.id32[2] = -1; + runtime->sync.id32[3] = -1; + } } EXPORT_SYMBOL(snd_pcm_set_sync); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 3fe99e6..e92bc62 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1619,6 +1619,9 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) list_add_tail(&substream1->link_list, &substream->group->substreams); substream->group->count++; substream1->group = substream->group; + snd_pcm_set_sync(substream); + snd_pcm_set_sync(substream1); + _end: write_unlock_irq(&snd_pcm_link_rwlock); up_write(&snd_pcm_link_rwsem); @@ -1652,11 +1655,14 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) if (substream->group->count == 1) { /* detach the last stream, too */ snd_pcm_group_for_each_entry(s, substream) { relink_to_local(s); + snd_pcm_set_sync(s); break; } kfree(substream->group); } relink_to_local(substream); + snd_pcm_set_sync(substream); + _end: write_unlock_irq(&snd_pcm_link_rwlock); up_write(&snd_pcm_link_rwsem); -- 1.7.6.5

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[alsa-devel] emu10k1 (Audigy) FXBus routing query
by Dan Swain 25 May '12

25 May '12

Dear all, I am writing to ask how to route certain DSP 'outputs' from my Audigy 2 ZS back into the equivalent 'ASIO inputs' for recording purposes. Specifically, I'd like to route outputs from the Wavetable Synths on my card to inputs that would go straight into a DAW (such as Ardour), so that I can apply effects, record and mix-down as if they were external synth signals. I know that with the kX project drivers in Windows this is possible, by routing the Synth outputs to FXBus outputs 16-63, leaving the ASIO inputs as 0-15 for the audio input, and then patching which channels I require from the Synth to those inputs (e.g. Synth Channel 1 -> FXBus out 16&17 -> FXBus in 0&1). After reading a thread that contained responses from Jaroslav, I believe the answer is in emufx.c, but I'm unsure as to where to start as I need to assign the outputs from the Synths correctly first. Thanks in advance! Any help would be greatly appreciated! Kind Regards, Dan -- Dan Swain <sp4md4n(a)gmail.com>

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[alsa-devel] runtime change in plugin behaviour
by Massimiliano Cialdi 24 May '12

24 May '12

Hi, I have developed a plugin that implements a bi-quad filter. filter's parameters are given into asoundrc file. Now I need to change those parameters during playback. How can I do? what help can gives alsalib to achieve this? best regards Max

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