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author | Takashi Iwai <tiwai@suse.de> | 2012-07-19 06:03:20 (GMT) |
---|---|---|
committer | Takashi Iwai <tiwai@suse.de> | 2012-07-19 06:03:20 (GMT) |
commit | 4609ed6b1f0ab9f11a9d0361573b53d9d057c440 (patch) | |
tree | 802119cc6ddea286bc03d56431286ac52166352e /sound/soc/pxa | |
parent | 639aa4bd58582f3015ae5621b7e9e754dcb58e6b (diff) | |
parent | 409b78cc17a4a3d07a541037575da648ced99437 (diff) | |
download | linux-4609ed6b1f0ab9f11a9d0361573b53d9d057c440.tar.xz |
Merge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for 3.6
This has been a pretty quiet release - very little activity in framework
terms, mostly just a few new drivers and updates:
- Added the ability to add and remove DAPM paths dynamically, mostly for
reparenting on clock changes.
- New machine drivers for Marvell Brownstone, ST-Ericsson Ux500
reference platform and ttc-dkp.
- New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP,
Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF
- New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI
Isabelle and Wolfson Microelectronics WM5102 and WM5110
Diffstat (limited to 'sound/soc/pxa')
-rw-r--r-- | sound/soc/pxa/Kconfig | 42 | ||||
-rw-r--r-- | sound/soc/pxa/Makefile | 8 | ||||
-rw-r--r-- | sound/soc/pxa/brownstone.c | 174 | ||||
-rw-r--r-- | sound/soc/pxa/mmp-pcm.c | 297 | ||||
-rw-r--r-- | sound/soc/pxa/mmp-sspa.c | 480 | ||||
-rw-r--r-- | sound/soc/pxa/mmp-sspa.h | 92 | ||||
-rw-r--r-- | sound/soc/pxa/ttc-dkb.c | 173 |
7 files changed, 1266 insertions, 0 deletions
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index a0f7d3c..4d2e46f 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -8,6 +8,15 @@ 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_SOC_DMAENGINE_PCM + 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 @@ -26,6 +35,9 @@ config SND_PXA_SOC_SSP tristate select PXA_SSP +config SND_MMP_SOC_SSPA + tristate + config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx @@ -138,6 +150,26 @@ config SND_SOC_TAVOREVB3 Say Y if you want to add support for SoC audio on the Marvell Saarb reference platform. +config SND_PXA910_SOC + tristate "SoC Audio for Marvell PXA910 chip" + depends on ARCH_MMP && SND + select SND_PCM + help + Say Y if you want to add support for SoC audio on the + Marvell PXA910 reference platform. + +config SND_SOC_TTC_DKB + bool "SoC Audio support for TTC DKB" + depends on SND_PXA910_SOC && MACH_TTC_DKB + select PXA_SSP + select SND_PXA_SOC_SSP + select SND_MMP_SOC + select MFD_88PM860X + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on TTC DKB + + config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE @@ -194,3 +226,13 @@ config SND_PXA2XX_SOC_IMOTE2 help Say Y if you want to add support for SoC audio on the IMote 2. + +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 MFD_WM8994 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on the + Marvell Brownstone reference platform. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index af35762..d8a265d 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -3,11 +3,15 @@ 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 +snd-soc-mmp-sspa-objs := mmp-sspa.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-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o +obj-$(CONFIG_SND_MMP_SOC_SSPA) += snd-soc-mmp-sspa.o # PXA Machine Support snd-soc-corgi-objs := corgi.o @@ -28,6 +32,8 @@ snd-soc-mioa701-objs := mioa701_wm9713.o snd-soc-z2-objs := z2.o snd-soc-imote2-objs := imote2.o snd-soc-raumfeld-objs := raumfeld.o +snd-soc-brownstone-objs := brownstone.o +snd-soc-ttc-dkb-objs := ttc-dkb.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -47,3 +53,5 @@ obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o +obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o +obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c new file mode 100644 index 0000000..5e666e0 --- /dev/null +++ b/sound/soc/pxa/brownstone.c @@ -0,0 +1,174 @@ +/* + * linux/sound/soc/pxa/brownstone.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 <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "../codecs/wm8994.h" +#include "mmp-sspa.h" + +static const struct snd_kcontrol_new brownstone_dapm_control[] = { + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static const struct snd_soc_dapm_widget brownstone_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), +}; + +static const struct snd_soc_dapm_route brownstone_audio_map[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + {"Ext Spk", NULL, "SPKOUTRP"}, + {"Ext Spk", NULL, "SPKOUTRN"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + + {"DMIC1DAT", NULL, "MICBIAS1"}, + {"MICBIAS1", NULL, "Main Mic"}, +}; + +static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_enable_pin(dapm, "Ext 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"); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(dapm, "HPOUT2P"); + snd_soc_dapm_nc_pin(dapm, "HPOUT2N"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT1N"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT1P"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT2N"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT2P"); + snd_soc_dapm_nc_pin(dapm, "IN1LN"); + snd_soc_dapm_nc_pin(dapm, "IN1LP"); + snd_soc_dapm_nc_pin(dapm, "IN1RP"); + snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(dapm, "IN2RN"); + snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(dapm, "IN2LN"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int freq_out, sspa_mclk, sysclk; + int sspa_div; + + if (params_rate(params) > 11025) { + freq_out = params_rate(params) * 512; + sysclk = params_rate(params) * 256; + sspa_mclk = params_rate(params) * 64; + } else { + freq_out = params_rate(params) * 1024; + sysclk = params_rate(params) * 512; + sspa_mclk = params_rate(params) * 64; + } + sspa_div = freq_out; + do_div(sspa_div, sspa_mclk); + + snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0); + snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk); + snd_soc_dai_set_pll(cpu_dai, MMP_SSPA_CLK, 0, freq_out, sspa_mclk); + + /* set wm8994 sysclk */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, sysclk, 0); + + return 0; +} + +/* machine stream operations */ +static struct snd_soc_ops brownstone_ops = { + .hw_params = brownstone_wm8994_hw_params, +}; + +static struct snd_soc_dai_link brownstone_wm8994_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_name = "mmp-sspa-dai.0", + .codec_dai_name = "wm8994-aif1", + .platform_name = "mmp-pcm-audio", + .codec_name = "wm8994-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ops = &brownstone_ops, + .init = brownstone_wm8994_init, +}, +}; + +/* audio machine driver */ +static struct snd_soc_card brownstone = { + .name = "brownstone", + .dai_link = brownstone_wm8994_dai, + .num_links = ARRAY_SIZE(brownstone_wm8994_dai), + + .controls = brownstone_dapm_control, + .num_controls = ARRAY_SIZE(brownstone_dapm_control), + .dapm_widgets = brownstone_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets), + .dapm_routes = brownstone_audio_map, + .num_dapm_routes = ARRAY_SIZE(brownstone_audio_map), +}; + +static int __devinit brownstone_probe(struct platform_device *pdev) +{ + int ret; + + brownstone.dev = &pdev->dev; + ret = snd_soc_register_card(&brownstone); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; +} + +static int __devexit brownstone_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&brownstone); + return 0; +} + +static struct platform_driver mmp_driver = { + .driver = { + .name = "brownstone-audio", + .owner = THIS_MODULE, + }, + .probe = brownstone_probe, + .remove = __devexit_p(brownstone_remove), +}; + +module_platform_driver(mmp_driver); + +MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC Brownstone"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c new file mode 100644 index 0000000..73ac546 --- /dev/null +++ b/sound/soc/pxa/mmp-pcm.c @@ -0,0 +1,297 @@ +/* + * 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_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> +#include <sound/dmaengine_pcm.h> + +struct mmp_dma_data { + int ssp_id; + struct resource *dma_res; +}; + +#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 int mmp_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct pxa2xx_pcm_dma_params *dma_params; + struct dma_slave_config slave_config; + int ret; + + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dma_params) + return 0; + + ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config); + if (ret) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + slave_config.dst_addr = dma_params->dev_addr; + slave_config.dst_maxburst = 4; + } else { + slave_config.src_addr = dma_params->dev_addr; + slave_config.src_maxburst = 4; + } + + ret = dmaengine_slave_config(chan, &slave_config); + if (ret) + return ret; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static bool filter(struct dma_chan *chan, void *param) +{ + struct mmp_dma_data *dma_data = param; + bool found = false; + char *devname; + + devname = kasprintf(GFP_KERNEL, "%s.%d", dma_data->dma_res->name, + dma_data->ssp_id); + if ((strcmp(dev_name(chan->device->dev), devname) == 0) && + (chan->chan_id == dma_data->dma_res->start)) { + found = true; + } + + kfree(devname); + return found; +} + +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_soc_dai *cpu_dai = rtd->cpu_dai; + struct mmp_dma_data *dma_data; + 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]); + dma_data = devm_kzalloc(&pdev->dev, + sizeof(struct mmp_dma_data), GFP_KERNEL); + if (dma_data == NULL) + return -ENOMEM; + + dma_data->dma_res = r; + dma_data->ssp_id = cpu_dai->id; + + ret = snd_dmaengine_pcm_open(substream, filter, dma_data); + if (ret) { + devm_kfree(&pdev->dev, dma_data); + return ret; + } + + snd_dmaengine_pcm_set_data(substream, dma_data); + return 0; +} + +static int mmp_pcm_close(struct snd_pcm_substream *substream) +{ + struct mmp_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct platform_device *pdev = to_platform_device(rtd->platform->dev); + + snd_dmaengine_pcm_close(substream); + devm_kfree(&pdev->dev, dma_data); + 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, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = mmp_pcm_mmap, +}; + +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 int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream, + int stream) +{ + 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 = substream->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; +} + +int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm_substream *substream; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0, stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + + ret = mmp_pcm_preallocate_dma_buffer(substream, stream); + if (ret) + goto err; + } + + return 0; + +err: + mmp_pcm_free_dma_buffers(pcm); + 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@marvell.com>"); +MODULE_DESCRIPTION("MMP Soc Audio DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c new file mode 100644 index 0000000..4d6cb8a --- /dev/null +++ b/sound/soc/pxa/mmp-sspa.c @@ -0,0 +1,480 @@ +/* + * linux/sound/soc/pxa/mmp-sspa.c + * Base on pxa2xx-ssp.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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/slab.h> +#include <linux/pxa2xx_ssp.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/pxa2xx-lib.h> +#include "mmp-sspa.h" + +/* + * SSPA audio private data + */ +struct sspa_priv { + struct ssp_device *sspa; + struct pxa2xx_pcm_dma_params *dma_params; + struct clk *audio_clk; + struct clk *sysclk; + int dai_fmt; + int running_cnt; +}; + +static void mmp_sspa_write_reg(struct ssp_device *sspa, u32 reg, u32 val) +{ + __raw_writel(val, sspa->mmio_base + reg); +} + +static u32 mmp_sspa_read_reg(struct ssp_device *sspa, u32 reg) +{ + return __raw_readl(sspa->mmio_base + reg); +} + +static void mmp_sspa_tx_enable(struct ssp_device *sspa) +{ + unsigned int sspa_sp; + + sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP); + sspa_sp |= SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); +} + +static void mmp_sspa_tx_disable(struct ssp_device *sspa) +{ + unsigned int sspa_sp; + + sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP); + sspa_sp &= ~SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); +} + +static void mmp_sspa_rx_enable(struct ssp_device *sspa) +{ + unsigned int sspa_sp; + + sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP); + sspa_sp |= SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp); +} + +static void mmp_sspa_rx_disable(struct ssp_device *sspa) +{ + unsigned int sspa_sp; + + sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP); + sspa_sp &= ~SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp); +} + +static int mmp_sspa_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai); + + clk_enable(priv->sysclk); + clk_enable(priv->sspa->clk); + + return 0; +} + +static void mmp_sspa_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai); + + clk_disable(priv->sspa->clk); + clk_disable(priv->sysclk); + + return; +} + +/* + * Set the SSP ports SYSCLK. + */ +static int mmp_sspa_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + + switch (clk_id) { + case MMP_SSPA_CLK_AUDIO: + ret = clk_set_rate(priv->audio_clk, freq); + if (ret) + return ret; + break; + case MMP_SSPA_CLK_PLL: + case MMP_SSPA_CLK_VCXO: + /* not support yet */ + return -EINVAL; + default: + return -EINVAL; + } + + return 0; +} + +static int mmp_sspa_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + + switch (pll_id) { + case MMP_SYSCLK: + ret = clk_set_rate(priv->sysclk, freq_out); + if (ret) + return ret; + break; + case MMP_SSPA_CLK: + ret = clk_set_rate(priv->sspa->clk, freq_out); + if (ret) + return ret; + break; + default: + return -ENODEV; + } + + return 0; +} + +/* + * Set up the sspa dai format. The sspa port must be inactive + * before calling this function as the physical + * interface format is changed. + */ +static int mmp_sspa_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct ssp_device *sspa = sspa_priv->sspa; + u32 sspa_sp, sspa_ctrl; + + /* check if we need to change anything at all */ + if (sspa_priv->dai_fmt == fmt) + return 0; + + /* we can only change the settings if the port is not in use */ + if ((mmp_sspa_read_reg(sspa, SSPA_TXSP) & SSPA_SP_S_EN) || + (mmp_sspa_read_reg(sspa, SSPA_RXSP) & SSPA_SP_S_EN)) { + dev_err(&sspa->pdev->dev, + "can't change hardware dai format: stream is in use\n"); + return -EINVAL; + } + + /* reset port settings */ + sspa_sp = SSPA_SP_WEN | SSPA_SP_S_RST | SSPA_SP_FFLUSH; + sspa_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + sspa_sp |= SSPA_SP_MSL; + break; + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspa_sp |= SSPA_SP_FSP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + sspa_sp |= SSPA_TXSP_FPER(63); + sspa_sp |= SSPA_SP_FWID(31); + sspa_ctrl |= SSPA_CTL_XDATDLY(1); + break; + default: + return -EINVAL; + } + + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); + mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp); + + sspa_sp &= ~(SSPA_SP_S_RST | SSPA_SP_FFLUSH); + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); + mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp); + + /* + * FIXME: hw issue, for the tx serial port, + * can not config the master/slave mode; + * so must clean this bit. + * The master/slave mode has been set in the + * rx port. + */ + sspa_sp &= ~SSPA_SP_MSL; + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); + + mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl); + mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl); + + /* Since we are configuring the timings for the format by hand + * we have to defer some things until hw_params() where we + * know parameters like the sample size. + */ + sspa_priv->dai_fmt = fmt; + return 0; +} + +/* + * Set the SSPA audio DMA parameters and sample size. + * Can be called multiple times by oss emulation. + */ +static int mmp_sspa_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_dai *cpu_dai = rtd->cpu_dai; + struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai); + struct ssp_device *sspa = sspa_priv->sspa; + struct pxa2xx_pcm_dma_params *dma_params; + u32 sspa_ctrl; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_TXCTL); + else + sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_RXCTL); + + sspa_ctrl &= ~SSPA_CTL_XFRLEN1_MASK; + sspa_ctrl |= SSPA_CTL_XFRLEN1(params_channels(params) - 1); + sspa_ctrl &= ~SSPA_CTL_XWDLEN1_MASK; + sspa_ctrl |= SSPA_CTL_XWDLEN1(SSPA_CTL_32_BITS); + sspa_ctrl &= ~SSPA_CTL_XSSZ1_MASK; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_8_BITS); + break; + case SNDRV_PCM_FORMAT_S16_LE: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_16_BITS); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_20_BITS); + break; + case SNDRV_PCM_FORMAT_S24_3LE: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_24_BITS); + break; + case SNDRV_PCM_FORMAT_S32_LE: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_32_BITS); + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl); + mmp_sspa_write_reg(sspa, SSPA_TXFIFO_LL, 0x1); + } else { + mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl); + mmp_sspa_write_reg(sspa, SSPA_RXFIFO_UL, 0x0); + } + + dma_params = &sspa_priv->dma_params[substream->stream]; + dma_params->dev_addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + (sspa->phys_base + SSPA_TXD) : + (sspa->phys_base + SSPA_RXD); + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_params); + return 0; +} + +static int mmp_sspa_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai); + struct ssp_device *sspa = sspa_priv->sspa; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * whatever playback or capture, must enable rx. + * this is a hw issue, so need check if rx has been + * enabled or not; if has been enabled by another + * stream, do not enable again. + */ + if (!sspa_priv->running_cnt) + mmp_sspa_rx_enable(sspa); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mmp_sspa_tx_enable(sspa); + + sspa_priv->running_cnt++; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sspa_priv->running_cnt--; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mmp_sspa_tx_disable(sspa); + + /* have no capture stream, disable rx port */ + if (!sspa_priv->running_cnt) + mmp_sspa_rx_disable(sspa); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static int mmp_sspa_probe(struct snd_soc_dai *dai) +{ + struct sspa_priv *priv = dev_get_drvdata(dai->dev); + + snd_soc_dai_set_drvdata(dai, priv); + return 0; + +} + +#define MMP_SSPA_RATES SNDRV_PCM_RATE_8000_192000 +#define MMP_SSPA_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops mmp_sspa_dai_ops = { + .startup = mmp_sspa_startup, + .shutdown = mmp_sspa_shutdown, + .trigger = mmp_sspa_trigger, + .hw_params = mmp_sspa_hw_params, + .set_sysclk = mmp_sspa_set_dai_sysclk, + .set_pll = mmp_sspa_set_dai_pll, + .set_fmt = mmp_sspa_set_dai_fmt, +}; + +struct snd_soc_dai_driver mmp_sspa_dai = { + .probe = mmp_sspa_probe, + .playback = { + .channels_min = 1, + .channels_max = 128, + .rates = MMP_SSPA_RATES, + .formats = MMP_SSPA_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = MMP_SSPA_RATES, + .formats = MMP_SSPA_FORMATS, + }, + .ops = &mmp_sspa_dai_ops, +}; + +static __devinit int asoc_mmp_sspa_probe(struct platform_device *pdev) +{ + struct sspa_priv *priv; + struct resource *res; + + priv = devm_kzalloc(&pdev->dev, + sizeof(struct sspa_priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->sspa = devm_kzalloc(&pdev->dev, + sizeof(struct ssp_device), GFP_KERNEL); + if (priv->sspa == NULL) + return -ENOMEM; + + priv->dma_params = devm_kzalloc(&pdev->dev, + 2 * sizeof(struct pxa2xx_pcm_dma_params), GFP_KERNEL); + if (priv->dma_params == NULL) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) + return -ENOMEM; + + priv->sspa->mmio_base = devm_request_and_ioremap(&pdev->dev, res); + if (priv->sspa->mmio_base == NULL) + return -ENODEV; + + priv->sspa->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(priv->sspa->clk)) + return PTR_ERR(priv->sspa->clk); + + priv->audio_clk = clk_get(NULL, "mmp-audio"); + if (IS_ERR(priv->audio_clk)) + return PTR_ERR(priv->audio_clk); + + priv->sysclk = clk_get(NULL, "mmp-sysclk"); + if (IS_ERR(priv->sysclk)) { + clk_put(priv->audio_clk); + return PTR_ERR(priv->sysclk); + } + clk_enable(priv->audio_clk); + priv->dai_fmt = (unsigned int) -1; + platform_set_drvdata(pdev, priv); + + return snd_soc_register_dai(&pdev->dev, &mmp_sspa_dai); +} + +static int __devexit asoc_mmp_sspa_remove(struct platform_device *pdev) +{ + struct sspa_priv *priv = platform_get_drvdata(pdev); + + clk_disable(priv->audio_clk); + clk_put(priv->audio_clk); + clk_put(priv->sysclk); + snd_soc_unregister_dai(&pdev->dev); + return 0; +} + +static struct platform_driver asoc_mmp_sspa_driver = { + .driver = { + .name = "mmp-sspa-dai", + .owner = THIS_MODULE, + }, + .probe = asoc_mmp_sspa_probe, + .remove = __devexit_p(asoc_mmp_sspa_remove), +}; + +module_platform_driver(asoc_mmp_sspa_driver); + +MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); +MODULE_DESCRIPTION("MMP SSPA SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/mmp-sspa.h b/sound/soc/pxa/mmp-sspa.h new file mode 100644 index 0000000..ea365cb --- /dev/null +++ b/sound/soc/pxa/mmp-sspa.h @@ -0,0 +1,92 @@ +/* + * linux/sound/soc/pxa/mmp-sspa.h + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#ifndef _MMP_SSPA_H +#define _MMP_SSPA_H + +/* + * SSPA Registers + */ +#define SSPA_RXD (0x00) +#define SSPA_RXID (0x04) +#define SSPA_RXCTL (0x08) +#define SSPA_RXSP (0x0c) +#define SSPA_RXFIFO_UL (0x10) +#define SSPA_RXINT_MASK (0x14) +#define SSPA_RXC (0x18) +#define SSPA_RXFIFO_NOFS (0x1c) +#define SSPA_RXFIFO_SIZE (0x20) + +#define SSPA_TXD (0x80) +#define SSPA_TXID (0x84) +#define SSPA_TXCTL (0x88) +#define SSPA_TXSP (0x8c) +#define SSPA_TXFIFO_LL (0x90) +#define SSPA_TXINT_MASK (0x94) +#define SSPA_TXC (0x98) +#define SSPA_TXFIFO_NOFS (0x9c) +#define SSPA_TXFIFO_SIZE (0xa0) + +/* SSPA Control Register */ +#define SSPA_CTL_XPH (1 << 31) /* Read Phase */ +#define SSPA_CTL_XFIG (1 << 15) /* Transmit Zeros when FIFO Empty */ +#define SSPA_CTL_JST (1 << 3) /* Audio Sample Justification */ +#define SSPA_CTL_XFRLEN2_MASK (7 << 24) +#define SSPA_CTL_XFRLEN2(x) ((x) << 24) /* Transmit Frame Length in Phase 2 */ +#define SSPA_CTL_XWDLEN2_MASK (7 << 21) +#define SSPA_CTL_XWDLEN2(x) ((x) << 21) /* Transmit Word Length in Phase 2 */ +#define SSPA_CTL_XDATDLY(x) ((x) << 19) /* Tansmit Data Delay */ +#define SSPA_CTL_XSSZ2_MASK (7 << 16) +#define SSPA_CTL_XSSZ2(x) ((x) << 16) /* Transmit Sample Audio Size */ +#define SSPA_CTL_XFRLEN1_MASK (7 << 8) +#define SSPA_CTL_XFRLEN1(x) ((x) << 8) /* Transmit Frame Length in Phase 1 */ +#define SSPA_CTL_XWDLEN1_MASK (7 << 5) +#define SSPA_CTL_XWDLEN1(x) ((x) << 5) /* Transmit Word Length in Phase 1 */ +#define SSPA_CTL_XSSZ1_MASK (7 << 0) +#define SSPA_CTL_XSSZ1(x) ((x) << 0) /* XSSZ1 */ + +#define SSPA_CTL_8_BITS (0x0) /* Sample Size */ +#define SSPA_CTL_12_BITS (0x1) +#define SSPA_CTL_16_BITS (0x2) +#define SSPA_CTL_20_BITS (0x3) +#define SSPA_CTL_24_BITS (0x4) +#define SSPA_CTL_32_BITS (0x5) + +/* SSPA Serial Port Register */ +#define SSPA_SP_WEN (1 << 31) /* Write Configuration Enable */ +#define SSPA_SP_MSL (1 << 18) /* Master Slave Configuration */ +#define SSPA_SP_CLKP (1 << 17) /* CLKP Polarity Clock Edge Select */ +#define SSPA_SP_FSP (1 << 16) /* FSP Polarity Clock Edge Select */ +#define SSPA_SP_FFLUSH (1 << 2) /* FIFO Flush */ +#define SSPA_SP_S_RST (1 << 1) /* Active High Reset Signal */ +#define SSPA_SP_S_EN (1 << 0) /* Serial Clock Domain Enable */ +#define SSPA_SP_FWID(x) ((x) << 20) /* Frame-Sync Width */ +#define SSPA_TXSP_FPER(x) ((x) << 4) /* Frame-Sync Active */ + +/* sspa clock sources */ +#define MMP_SSPA_CLK_PLL 0 +#define MMP_SSPA_CLK_VCXO 1 +#define MMP_SSPA_CLK_AUDIO 3 + +/* sspa pll id */ +#define MMP_SYSCLK 0 +#define MMP_SSPA_CLK 1 + +#endif /* _MMP_SSPA_H */ diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c new file mode 100644 index 0000000..935491a --- /dev/null +++ b/sound/soc/pxa/ttc-dkb.c @@ -0,0 +1,173 @@ +/* + * linux/sound/soc/pxa/ttc_dkb.c + * + * Copyright (C) 2012 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <sound/pcm_params.h> +#include "../codecs/88pm860x-codec.h" + +static struct snd_soc_jack hs_jack, mic_jack; + +static struct snd_soc_jack_pin hs_jack_pins[] = { + { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, +}; + +static struct snd_soc_jack_pin mic_jack_pins[] = { + { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, +}; + +/* ttc machine dapm widgets */ +static const struct snd_soc_dapm_widget ttc_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_LINE("Lineout Out 1", NULL), + SND_SOC_DAPM_LINE("Lineout Out 2", NULL), + SND_SOC_DAPM_SPK("Ext Speaker", NULL), + SND_SOC_DAPM_MIC("Ext Mic 1", NULL), + SND_SOC_DAPM_MIC("Headset Mic 2", NULL), + SND_SOC_DAPM_MIC("Ext Mic 3", NULL), +}; + +/* ttc machine audio map */ +static const struct snd_soc_dapm_route ttc_audio_map[] = { + {"Headset Stereophone", NULL, "HS1"}, + {"Headset Stereophone", NULL, "HS2"}, + + {"Ext Speaker", NULL, "LSP"}, + {"Ext Speaker", NULL, "LSN"}, + + {"Lineout Out 1", NULL, "LINEOUT1"}, + {"Lineout Out 2", NULL, "LINEOUT2"}, + + {"MIC1P", NULL, "Mic1 Bias"}, + {"MIC1N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Ext Mic 1"}, + + {"MIC2P", NULL, "Mic1 Bias"}, + {"MIC2N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Headset Mic 2"}, + + {"MIC3P", NULL, "Mic3 Bias"}, + {"MIC3N", NULL, "Mic3 Bias"}, + {"Mic3 Bias", NULL, "Ext Mic 3"}, +}; + +static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + /* connected pins */ + snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); + snd_soc_dapm_enable_pin(dapm, "Ext Mic 1"); + snd_soc_dapm_enable_pin(dapm, "Ext Mic 3"); + snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); + snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); + + /* Headset jack detection */ + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE + | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, + &hs_jack); + snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, + &mic_jack); + snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), + mic_jack_pins); + + /* headphone, microphone detection & headset short detection */ + pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, + SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2); + pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE); + + return 0; +} + +/* ttc/td-dkb digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = { +{ + .name = "88pm860x i2s", + .stream_name = "audio playback", + .codec_name = "88pm860x-codec", + .platform_name = "mmp-pcm-audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + .init = ttc_pm860x_init, +}, +}; + +/* ttc/td audio machine driver */ +static struct snd_soc_card ttc_dkb_card = { + .name = "ttc-dkb-hifi", + .dai_link = ttc_pm860x_hifi_dai, + .num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai), + + .dapm_widgets = ttc_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ttc_dapm_widgets), + .dapm_routes = ttc_audio_map, + .num_dapm_routes = ARRAY_SIZE(ttc_audio_map), +}; + +static int __devinit ttc_dkb_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &ttc_dkb_card; + int ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + + return ret; +} + +static int __devexit ttc_dkb_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver ttc_dkb_driver = { + .driver = { + .name = "ttc-dkb-audio", + .owner = THIS_MODULE, + }, + .probe = ttc_dkb_probe, + .remove = __devexit_p(ttc_dkb_remove), +}; + +module_platform_driver(ttc_dkb_driver); + +/* Module information */ +MODULE_AUTHOR("Qiao Zhou, <zhouqiao@marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC TTC DKB"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:ttc-dkb-audio"); |