August 2010 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] ALC892 support for multiple spdif outputs
by Takashi Iwai 17 Aug '10

17 Aug '10

At Mon, 16 Aug 2010 09:52:31 +0200, Sander Knopper wrote: > > [1 <text/plain; iso-8859-1 (7bit)>] > On Monday 16 August 2010 08:42:07 you wrote: > > At Sun, 15 Aug 2010 14:57:51 +0200, > > > > Sander Knopper wrote: > > > Dear developers, > > > > > > I've opened a bug about this problem which can be viewed here: > > > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5101 > > > > > > Raymond mentions in his last comment that I should ask you to, if I > > > understand correctly, do the same trick as in the commit he refers to > > > also for the ALC892 chip. Which obviously is the one I have... > > > > > > You can find all the relevant information in the bug entry. > > > > Could you post the exact problem and attach alsa-info.sh output to ML? > > It'll help much for communication on ML. > > > > > > Takashi > > Ok, I can do that... > > The problem is that I have an Asus P7H55 M Pro motherboard which uses the > ALC892 codec. I'm using this system as a HTPC and thus I'm trying to get sound > from the HTPC to my receiver. The motherboard only has an optical output which > shouldn't be an problem, be it that my receiver only has a coaxial input. > > So I bought an spdif bracket made by Asus which has both a coaxial and optical > output and connected it to the internal spdif pins on the motherboard. So far > so good, but the real problem is that I can't get the bracket to work. When I > unmute all the spdif related entries with alsamixer only the optical output on > the motherboard itself "gives light" and the one on the bracket doesn't. > > I've searched the internet a bit and I tried switching the audio output in the > bios from HDMI to SPDIF and vice versa, but that doesn't make any difference. > > alsa-info is attached. OK. Did you test the latest alsa-driver-snapshot? ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… The multi-SPDIF support is generic for all Realtek codecs. Takashi

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[PATCH 3/3] ASoC: add saarb machine driver for 88pm860x
by Haojian Zhuang 17 Aug '10

17 Aug '10

88PM860x codec is used in Marvell saarb development board. 88PM860x codec is used as master mode for SSP communication. Only I2S format is supported. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/saarb.c | 200 ++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 211 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/saarb.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 04ddc7b..37f191b 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -117,6 +117,15 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. +config SND_SOC_SAARB + tristate "SoC Audio support for Marvell Saarb" + depends on SND_PXA2XX_SOC && MACH_SAARB + select SND_PXA_SOC_SSP + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on the + Marvell Saarb reference platform. + config SND_SOC_TAVOREVB3 tristate "SoC Audio support for Marvell Tavor EVB3" depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 315941f..0766016 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -19,6 +19,7 @@ snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o +snd-soc-saarb-objs := saarb.o snd-soc-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o @@ -39,6 +40,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o +obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o obj-$(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 diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c new file mode 100644 index 0000000..d63cb47 --- /dev/null +++ b/sound/soc/pxa/saarb.c @@ -0,0 +1,200 @@ +/* + * saarb.c -- SoC audio for saarb + * + * Copyright (C) 2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * 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 <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/i2c.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/jack.h> + +#include <asm/mach-types.h> + +#include "../codecs/88pm860x-codec.h" +#include "pxa-ssp.h" + +static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd); + +static struct platform_device *saarb_snd_device; + +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, }, +}; + +/* saarb machine dapm widgets */ +static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone 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", NULL), + SND_SOC_DAPM_MIC("Ext Mic 3", NULL), +}; + +/* saarb machine audio map */ +static const struct snd_soc_dapm_route 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 saarb_i2s_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 width = snd_pcm_format_physical_width(params_format(params)); + int ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, + PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); + + return ret; +} + +static struct snd_soc_ops saarb_i2s_ops = { + .hw_params = saarb_i2s_hw_params, +}; + +static struct snd_soc_dai_link saarb_dai[] = { + { + .name = "88PM860x I2S", + .stream_name = "I2S Audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .platform_name = "pxa-pcm-audio", + .codec_name = "88pm860x-codec", + .init = saarb_pm860x_init, + .ops = &saarb_i2s_ops, + }, +}; + +static struct snd_soc_card snd_soc_card_saarb = { + .name = "Saarb", + .dai_link = saarb_dai, + .num_links = ARRAY_SIZE(saarb_dai), +}; + +static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + snd_soc_dapm_new_controls(codec, saarb_dapm_widgets, + ARRAY_SIZE(saarb_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Speaker"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 1"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 3"); + snd_soc_dapm_disable_pin(codec, "Headset Mic 2"); + snd_soc_dapm_disable_pin(codec, "Headset Stereophone"); + + ret = snd_soc_dapm_sync(codec); + if (ret) + return ret; + + /* 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; +} + +static int __init saarb_init(void) +{ + int ret; + + if (!machine_is_saarb()) + return -ENODEV; + saarb_snd_device = platform_device_alloc("soc-audio", -1); + if (!saarb_snd_device) + return -ENOMEM; + + platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb); + + ret = platform_device_add(saarb_snd_device); + if (ret) + platform_device_put(saarb_snd_device); + + return ret; +} + +static void __exit saarb_exit(void) +{ + platform_device_unregister(saarb_snd_device); +} + +module_init(saarb_init); +module_exit(saarb_exit); + +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb"); +MODULE_LICENSE("GPL"); -- 1.5.6.5

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[PATCH 3/3] ASoC: add saarb machine driver for 88pm860x
by Haojian Zhuang 17 Aug '10

17 Aug '10

88PM860x codec is used in Marvell saarb development board. 88PM860x codec is used as master mode for SSP communication. Only I2S format is supported. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/saarb.c | 192 ++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 203 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/saarb.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 04ddc7b..37f191b 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -117,6 +117,15 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. +config SND_SOC_SAARB + tristate "SoC Audio support for Marvell Saarb" + depends on SND_PXA2XX_SOC && MACH_SAARB + select SND_PXA_SOC_SSP + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on the + Marvell Saarb reference platform. + config SND_SOC_TAVOREVB3 tristate "SoC Audio support for Marvell Tavor EVB3" depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 315941f..0766016 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -19,6 +19,7 @@ snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o +snd-soc-saarb-objs := saarb.o snd-soc-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o @@ -39,6 +40,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o +obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o obj-$(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 diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c new file mode 100644 index 0000000..b347b54 --- /dev/null +++ b/sound/soc/pxa/saarb.c @@ -0,0 +1,192 @@ +/* + * saarb.c -- SoC audio for saarb + * + * Copyright (C) 2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * 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 <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/i2c.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/jack.h> + +#include <asm/mach-types.h> + +#include "../codecs/88pm860x-codec.h" +#include "pxa-ssp.h" + +static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd); + +static struct platform_device *saarb_snd_device; + +static struct snd_soc_jack hs_jack; + +static struct snd_soc_jack_pin hs_jack_pins[] = { + { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, + { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, +}; + +/* saarb machine dapm widgets */ +static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone 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", NULL), + SND_SOC_DAPM_MIC("Ext Mic 3", NULL), +}; + +/* saarb machine audio map */ +static const struct snd_soc_dapm_route 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 saarb_i2s_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 width = snd_pcm_format_physical_width(params_format(params)); + int ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, + PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); + + return ret; +} + +static struct snd_soc_ops saarb_i2s_ops = { + .hw_params = saarb_i2s_hw_params, +}; + +static struct snd_soc_dai_link saarb_dai[] = { + { + .name = "88PM860x I2S", + .stream_name = "I2S Audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .platform_name = "pxa-pcm-audio", + .codec_name = "88pm860x-codec", + .init = saarb_pm860x_init, + .ops = &saarb_i2s_ops, + }, +}; + +static struct snd_soc_card snd_soc_card_saarb = { + .name = "Saarb", + .dai_link = saarb_dai, + .num_links = ARRAY_SIZE(saarb_dai), +}; + +static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + snd_soc_dapm_new_controls(codec, saarb_dapm_widgets, + ARRAY_SIZE(saarb_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Speaker"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 1"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 3"); + snd_soc_dapm_disable_pin(codec, "Headset Mic 2"); + snd_soc_dapm_disable_pin(codec, "Headset Stereophone"); + + ret = snd_soc_dapm_sync(codec); + if (ret) + return ret; + + /* Headset jack detection */ + snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET + | 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); + /* headphone, microphone detection & headset short detection */ + pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET, + SND_JACK_BTN_0); + pm860x_hook_detect(codec, &hs_jack, SND_JACK_BTN_1, SND_JACK_BTN_2); + return 0; +} + +static int __init saarb_init(void) +{ + int ret; + + if (!machine_is_saarb()) + return -ENODEV; + saarb_snd_device = platform_device_alloc("soc-audio", -1); + if (!saarb_snd_device) + return -ENOMEM; + + platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb); + + ret = platform_device_add(saarb_snd_device); + if (ret) + platform_device_put(saarb_snd_device); + + return ret; +} + +static void __exit saarb_exit(void) +{ + platform_device_unregister(saarb_snd_device); +} + +module_init(saarb_init); +module_exit(saarb_exit); + +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb"); +MODULE_LICENSE("GPL"); -- 1.5.6.5

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[PATCH 3/3] ASoC: add saarb machine driver for 88pm860x
by Haojian Zhuang 17 Aug '10

17 Aug '10

88PM860x codec is used in Marvell saarb development board. 88PM860x codec is used as master mode for SSP communication. Only I2S format is supported. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/saarb.c | 192 ++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 203 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/saarb.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 04ddc7b..37f191b 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -117,6 +117,15 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. +config SND_SOC_SAARB + tristate "SoC Audio support for Marvell Saarb" + depends on SND_PXA2XX_SOC && MACH_SAARB + select SND_PXA_SOC_SSP + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on the + Marvell Saarb reference platform. + config SND_SOC_TAVOREVB3 tristate "SoC Audio support for Marvell Tavor EVB3" depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 315941f..0766016 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -19,6 +19,7 @@ snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o +snd-soc-saarb-objs := saarb.o snd-soc-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o @@ -39,6 +40,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o +obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o obj-$(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 diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c new file mode 100644 index 0000000..b347b54 --- /dev/null +++ b/sound/soc/pxa/saarb.c @@ -0,0 +1,192 @@ +/* + * saarb.c -- SoC audio for saarb + * + * Copyright (C) 2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * 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 <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/i2c.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/jack.h> + +#include <asm/mach-types.h> + +#include "../codecs/88pm860x-codec.h" +#include "pxa-ssp.h" + +static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd); + +static struct platform_device *saarb_snd_device; + +static struct snd_soc_jack hs_jack; + +static struct snd_soc_jack_pin hs_jack_pins[] = { + { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, + { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, +}; + +/* saarb machine dapm widgets */ +static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone 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", NULL), + SND_SOC_DAPM_MIC("Ext Mic 3", NULL), +}; + +/* saarb machine audio map */ +static const struct snd_soc_dapm_route 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 saarb_i2s_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 width = snd_pcm_format_physical_width(params_format(params)); + int ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, + PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); + + return ret; +} + +static struct snd_soc_ops saarb_i2s_ops = { + .hw_params = saarb_i2s_hw_params, +}; + +static struct snd_soc_dai_link saarb_dai[] = { + { + .name = "88PM860x I2S", + .stream_name = "I2S Audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .platform_name = "pxa-pcm-audio", + .codec_name = "88pm860x-codec", + .init = saarb_pm860x_init, + .ops = &saarb_i2s_ops, + }, +}; + +static struct snd_soc_card snd_soc_card_saarb = { + .name = "Saarb", + .dai_link = saarb_dai, + .num_links = ARRAY_SIZE(saarb_dai), +}; + +static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + snd_soc_dapm_new_controls(codec, saarb_dapm_widgets, + ARRAY_SIZE(saarb_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Speaker"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 1"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 3"); + snd_soc_dapm_disable_pin(codec, "Headset Mic 2"); + snd_soc_dapm_disable_pin(codec, "Headset Stereophone"); + + ret = snd_soc_dapm_sync(codec); + if (ret) + return ret; + + /* Headset jack detection */ + snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET + | 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); + /* headphone, microphone detection & headset short detection */ + pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET, + SND_JACK_BTN_0); + pm860x_hook_detect(codec, &hs_jack, SND_JACK_BTN_1, SND_JACK_BTN_2); + return 0; +} + +static int __init saarb_init(void) +{ + int ret; + + if (!machine_is_saarb()) + return -ENODEV; + saarb_snd_device = platform_device_alloc("soc-audio", -1); + if (!saarb_snd_device) + return -ENOMEM; + + platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb); + + ret = platform_device_add(saarb_snd_device); + if (ret) + platform_device_put(saarb_snd_device); + + return ret; +} + +static void __exit saarb_exit(void) +{ + platform_device_unregister(saarb_snd_device); +} + +module_init(saarb_init); +module_exit(saarb_exit); + +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb"); +MODULE_LICENSE("GPL"); -- 1.5.6.5

1 0

[PATCH 3/3] ASoC: add saarb machine driver for 88pm860x
by Haojian Zhuang 17 Aug '10

17 Aug '10

88PM860x codec is used in Marvell saarb development board. 88PM860x codec is used as master mode for SSP communication. Only I2S format is supported. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 9 +++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/saarb.c | 178 ++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 189 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/saarb.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 04ddc7b..37f191b 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -117,6 +117,15 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. +config SND_SOC_SAARB + tristate "SoC Audio support for Marvell Saarb" + depends on SND_PXA2XX_SOC && MACH_SAARB + select SND_PXA_SOC_SSP + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on the + Marvell Saarb reference platform. + config SND_SOC_TAVOREVB3 tristate "SoC Audio support for Marvell Tavor EVB3" depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 315941f..0766016 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -19,6 +19,7 @@ snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o +snd-soc-saarb-objs := saarb.o snd-soc-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o @@ -39,6 +40,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o +obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o obj-$(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 diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c new file mode 100644 index 0000000..3f6ec57 --- /dev/null +++ b/sound/soc/pxa/saarb.c @@ -0,0 +1,178 @@ +/* + * saarb.c -- SoC audio for saarb + * + * Copyright (C) 2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * 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 <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "../codecs/88pm860x-codec.h" +#include "pxa-ssp.h" + +static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd); + +static struct platform_device *saarb_snd_device; +/* saarb machine dapm widgets */ +static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_LINE("Lineout Out 1", NULL), + SND_SOC_DAPM_LINE("Lineout Out 2", NULL), + SND_SOC_DAPM_SPK("Board Speaker", NULL), + SND_SOC_DAPM_MIC("Board Mic 1", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Board Mic 3", NULL), +}; + +/* saarb machine audio map */ +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone", NULL, "HS1"}, + {"Headphone", NULL, "HS2"}, + + {"Board Speaker", NULL, "LSP"}, + {"Board Speaker", NULL, "LSN"}, + + {"Lineout Out 1", NULL, "LINEOUT1"}, + {"Lineout Out 2", NULL, "LINEOUT2"}, + + {"MIC1P", NULL, "Mic1 Bias"}, + {"MIC1N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Board Mic 1"}, + + {"MIC2P", NULL, "Mic1 Bias"}, + {"MIC2N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Headset Mic"}, + + {"MIC3P", NULL, "Mic3 Bias"}, + {"MIC3N", NULL, "Mic3 Bias"}, + {"Mic3 Bias", NULL, "Board Mic 3"}, +}; + +static const struct snd_kcontrol_new pm860x_saarb_controls[] = { + SOC_DAPM_PIN_SWITCH("Board Speaker"), + SOC_DAPM_PIN_SWITCH("Board Mic 1"), + SOC_DAPM_PIN_SWITCH("Board Mic 3"), +}; + +static int saarb_i2s_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 width = snd_pcm_format_physical_width(params_format(params)); + int ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, + PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); + + return ret; +} + +static struct snd_soc_ops saarb_i2s_ops = { + .hw_params = saarb_i2s_hw_params, +}; + +static struct snd_soc_dai_link saarb_dai[] = { + { + .name = "88PM860x I2S", + .stream_name = "I2S Audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .platform_name = "pxa-pcm-audio", + .codec_name = "88pm860x-codec", + .init = saarb_pm860x_init, + .ops = &saarb_i2s_ops, + }, +}; + +static struct snd_soc_card snd_soc_card_saarb = { + .name = "Saarb", + .dai_link = saarb_dai, + .num_links = ARRAY_SIZE(saarb_dai), +}; + +static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + ret = snd_soc_add_controls(codec, pm860x_saarb_controls, + ARRAY_SIZE(pm860x_saarb_controls)); + if (ret < 0) + return ret; + + snd_soc_dapm_new_controls(codec, saarb_dapm_widgets, + ARRAY_SIZE(saarb_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_disable_pin(codec, "Board Speaker"); + snd_soc_dapm_disable_pin(codec, "Board Mic 1"); + snd_soc_dapm_disable_pin(codec, "Board Mic 3"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + + return snd_soc_dapm_sync(codec); +} + +static int __init saarb_init(void) +{ + int ret; + + if (!machine_is_saarb()) + return -ENODEV; + saarb_snd_device = platform_device_alloc("soc-audio", -1); + if (!saarb_snd_device) + return -ENOMEM; + + platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb); + + ret = platform_device_add(saarb_snd_device); + if (ret) + platform_device_put(saarb_snd_device); + + return ret; +} + +static void __exit saarb_exit(void) +{ + platform_device_unregister(saarb_snd_device); +} + +module_init(saarb_init); +module_exit(saarb_exit); + +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb"); +MODULE_LICENSE("GPL"); -- 1.5.6.5 --0016e65aeed62e9cc9048e002c76 Content-Type: text/x-patch; charset=US-ASCII; 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[PATCH 2/3] ASoC: add tavorevb3 machine driver for 88pm860x
by Haojian Zhuang 17 Aug '10

17 Aug '10

88PM860x codec is used in Marvell tavorevb3 development board. 88PM860x codec is used as master mode of SSP communication. Only I2S format is supported. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/tavorevb3.c | 200 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 211 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/tavorevb3.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index e30c832..04ddc7b 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -117,6 +117,15 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. +config SND_SOC_TAVOREVB3 + tristate "SoC Audio support for Marvell Tavor EVB3" + depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 + select SND_PXA_SOC_SSP + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on the + Marvell Saarb reference platform. + config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index caa03d8..315941f 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -19,6 +19,7 @@ snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o +snd-soc-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o @@ -38,6 +39,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o +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 diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c new file mode 100644 index 0000000..248c283 --- /dev/null +++ b/sound/soc/pxa/tavorevb3.c @@ -0,0 +1,200 @@ +/* + * tavorevb3.c -- SoC audio for Tavor EVB3 + * + * Copyright (C) 2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * 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 <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/i2c.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/jack.h> + +#include <asm/mach-types.h> + +#include "../codecs/88pm860x-codec.h" +#include "pxa-ssp.h" + +static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd); + +static struct platform_device *evb3_snd_device; + +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, }, +}; + +/* tavorevb3 machine dapm widgets */ +static const struct snd_soc_dapm_widget evb3_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), +}; + +/* tavorevb3 machine audio map */ +static const struct snd_soc_dapm_route 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 evb3_i2s_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 width = snd_pcm_format_physical_width(params_format(params)); + int ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, + PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); + return ret; +} + +static struct snd_soc_ops evb3_i2s_ops = { + .hw_params = evb3_i2s_hw_params, +}; + +static struct snd_soc_dai_link evb3_dai[] = { + { + .name = "88PM860x I2S", + .stream_name = "I2S Audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .platform_name = "pxa-pcm-audio", + .codec_name = "88pm860x-codec", + .init = evb3_pm860x_init, + .ops = &evb3_i2s_ops, + }, +}; + +static struct snd_soc_card snd_soc_card_evb3 = { + .name = "Tavor EVB3", + .dai_link = evb3_dai, + .num_links = ARRAY_SIZE(evb3_dai), +}; + +static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + snd_soc_dapm_new_controls(codec, evb3_dapm_widgets, + ARRAY_SIZE(evb3_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Speaker"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 1"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 3"); + snd_soc_dapm_disable_pin(codec, "Headset Mic 2"); + snd_soc_dapm_disable_pin(codec, "Headset Stereophone"); + + ret = snd_soc_dapm_sync(codec); + if (ret) + return ret; + + /* 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; +} + +static int __init tavorevb3_init(void) +{ + int ret; + + if (!machine_is_tavorevb3()) + return -ENODEV; + evb3_snd_device = platform_device_alloc("soc-audio", -1); + if (!evb3_snd_device) + return -ENOMEM; + + platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3); + + ret = platform_device_add(evb3_snd_device); + if (ret) + platform_device_put(evb3_snd_device); + + return ret; +} + +static void __exit tavorevb3_exit(void) +{ + platform_device_unregister(evb3_snd_device); +} + +module_init(tavorevb3_init); +module_exit(tavorevb3_exit); + +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3"); +MODULE_LICENSE("GPL"); -- 1.5.6.5

1 0

[PATCH 2/3] ASoC: add tavorevb3 machine driver for 88pm860x
by Haojian Zhuang 17 Aug '10

17 Aug '10

88PM860x codec is used in Marvell tavorevb3 development board. 88PM860x codec is used as master mode of SSP communication. Only I2S format is supported. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/tavorevb3.c | 193 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 204 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/tavorevb3.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index e30c832..04ddc7b 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -117,6 +117,15 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. +config SND_SOC_TAVOREVB3 + tristate "SoC Audio support for Marvell Tavor EVB3" + depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 + select SND_PXA_SOC_SSP + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on the + Marvell Saarb reference platform. + config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index caa03d8..315941f 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -19,6 +19,7 @@ snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o +snd-soc-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o @@ -38,6 +39,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o +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 diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c new file mode 100644 index 0000000..3ee39d4 --- /dev/null +++ b/sound/soc/pxa/tavorevb3.c @@ -0,0 +1,193 @@ +/* + * tavorevb3.c -- SoC audio for Tavor EVB3 + * + * Copyright (C) 2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * 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 <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/i2c.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/jack.h> + +#include <asm/mach-types.h> + +#include "../codecs/88pm860x-codec.h" +#include "pxa-ssp.h" + +static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd); + +static struct platform_device *evb3_snd_device; + +static struct snd_soc_jack hs_jack; + +static struct snd_soc_jack_pin hs_jack_pins[] = { + { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, + { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, +}; + +/* tavorevb3 machine dapm widgets */ +static const struct snd_soc_dapm_widget evb3_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), +}; + +/* tavorevb3 machine audio map */ +static const struct snd_soc_dapm_route 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 evb3_i2s_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 width = snd_pcm_format_physical_width(params_format(params)); + int ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, + PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); + return ret; +} + +static struct snd_soc_ops evb3_i2s_ops = { + .hw_params = evb3_i2s_hw_params, +}; + +static struct snd_soc_dai_link evb3_dai[] = { + { + .name = "88PM860x I2S", + .stream_name = "I2S Audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .platform_name = "pxa-pcm-audio", + .codec_name = "88pm860x-codec", + .init = evb3_pm860x_init, + .ops = &evb3_i2s_ops, + }, +}; + +static struct snd_soc_card snd_soc_card_evb3 = { + .name = "Tavor EVB3", + .dai_link = evb3_dai, + .num_links = ARRAY_SIZE(evb3_dai), +}; + +static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + snd_soc_dapm_new_controls(codec, evb3_dapm_widgets, + ARRAY_SIZE(evb3_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Speaker"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 1"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 3"); + snd_soc_dapm_disable_pin(codec, "Headset Mic 2"); + snd_soc_dapm_disable_pin(codec, "Headset Stereophone"); + + ret = snd_soc_dapm_sync(codec); + if (ret) + return ret; + + /* Headset jack detection */ + snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET + | 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); + /* headphone, microphone detection & headset short detection */ + pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET, + SND_JACK_BTN_0); + pm860x_hook_detect(codec, &hs_jack, SND_JACK_BTN_1, SND_JACK_BTN_2); + return 0; +} + +static int __init tavorevb3_init(void) +{ + int ret; + + if (!machine_is_tavorevb3()) + return -ENODEV; + evb3_snd_device = platform_device_alloc("soc-audio", -1); + if (!evb3_snd_device) + return -ENOMEM; + + platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3); + + ret = platform_device_add(evb3_snd_device); + if (ret) + platform_device_put(evb3_snd_device); + + return ret; +} + +static void __exit tavorevb3_exit(void) +{ + platform_device_unregister(evb3_snd_device); +} + +module_init(tavorevb3_init); +module_exit(tavorevb3_exit); + +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3"); +MODULE_LICENSE("GPL"); + -- 1.5.6.5

1 0

[PATCH 2/3] ASoC: add tavorevb3 machine driver for 88pm860x
by Haojian Zhuang 17 Aug '10

17 Aug '10

88PM860x codec is used in Marvell tavorevb3 development board. 88PM860x codec is used as master mode of SSP communication. Only I2S format is supported. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/pxa2xx-ssp.c | 532 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/pxa/pxa2xx-ssp.h | 59 +++++ sound/soc/pxa/ssp.c | 298 +++++++++++++++++++++++++ sound/soc/pxa/ssp.h | 42 ++++ sound/soc/pxa/tavorevb3.c | 193 ++++++++++++++++ 7 files changed, 1135 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/pxa2xx-ssp.c create mode 100644 sound/soc/pxa/pxa2xx-ssp.h create mode 100644 sound/soc/pxa/ssp.c create mode 100644 sound/soc/pxa/ssp.h create mode 100644 sound/soc/pxa/tavorevb3.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index e30c832..04ddc7b 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -117,6 +117,15 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. +config SND_SOC_TAVOREVB3 + tristate "SoC Audio support for Marvell Tavor EVB3" + depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 + select SND_PXA_SOC_SSP + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on the + Marvell Saarb reference platform. + config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index caa03d8..315941f 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -19,6 +19,7 @@ snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o +snd-soc-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o @@ -38,6 +39,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o +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 diff --git a/sound/soc/pxa/pxa2xx-ssp.c b/sound/soc/pxa/pxa2xx-ssp.c new file mode 100644 index 0000000..5eab055 --- /dev/null +++ b/sound/soc/pxa/pxa2xx-ssp.c @@ -0,0 +1,532 @@ +/* + * pxa2xx-ssp.c -- ALSA Soc Audio Layer + * + * Copyright 2005,2008 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * Copyright 2009-2010 Marvell International Ltd. + * Author: Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * 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. + * + * TODO: + * o Test network mode for > 16bit sample size + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <linux/io.h> + +#include <asm/irq.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 <mach/hardware.h> +#include <mach/dma.h> +#include <mach/regs-ssp.h> +#include <mach/audio.h> +#include <plat/ssp.h> + +#include "pxa2xx-pcm.h" +#include "pxa2xx-ssp.h" +#include "ssp.h" + +static void dump_registers(struct ssp_device *ssp) +{ + dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", + ssp_read_reg(ssp, SSCR0), ssp_read_reg(ssp, SSCR1), + ssp_read_reg(ssp, SSTO)); + + dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n", + ssp_read_reg(ssp, SSPSP), ssp_read_reg(ssp, SSSR), + ssp_read_reg(ssp, SSACD)); +} + +/** + * ssp_set_clkdiv - set SSP clock divider + * @div: serial clock rate divider + */ +static void ssp_set_scr(struct ssp_device *ssp, u32 div) +{ + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + sscr0 &= ~0x0000ff00; + sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ + } else { + sscr0 &= ~0x000fff00; + sscr0 |= (div - 1) << 8; /* 1..4096 */ + } + ssp_write_reg(ssp, SSCR0, sscr0); +} + +/** + * ssp_get_clkdiv - get SSP clock divider + */ +static u32 ssp_get_scr(struct ssp_device *ssp) +{ + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 div; + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + div = ((sscr0 >> 8) & 0xff) * 2 + 2; + else + div = ((sscr0 >> 8) & 0xfff) + 1; + return div; +} + +/* + * Set the SSP ports SYSCLK. + */ +static int pxa2xx_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, + unsigned int freq, int dir) +{ + struct ssp_info *info = cpu_dai->private_data; + struct ssp_device *ssp = info->dev.ssp; + int val; + + u32 sscr0 = ssp_read_reg(ssp, SSCR0) & + ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + + dev_dbg(&ssp->pdev->dev, + "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", + cpu_dai->id, clk_id, freq); + + switch (clk_id) { + case PXA2XX_SSP_CLK_NET_PLL: + sscr0 |= SSCR0_MOD; + break; + case PXA2XX_SSP_CLK_PLL: + /* Internal PLL is fixed */ + if (cpu_is_pxa25x()) + info->sysclk = 1843200; + else + info->sysclk = 13000000; + break; + case PXA2XX_SSP_CLK_EXT: + info->sysclk = freq; + sscr0 |= SSCR0_ECS; + break; + case PXA2XX_SSP_CLK_NET: + info->sysclk = freq; + sscr0 |= SSCR0_NCS | SSCR0_MOD; + break; + case PXA2XX_SSP_CLK_AUDIO: + info->sysclk = 0; + ssp_set_scr(ssp, 1); + sscr0 |= SSCR0_ACS; + break; + default: + return -ENODEV; + } + + /* 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()) + clk_disable(info->dev.ssp->clk); + val = ssp_read_reg(ssp, SSCR0) | sscr0; + ssp_write_reg(ssp, SSCR0, val); + if (!cpu_is_pxa3xx()) + clk_enable(info->dev.ssp->clk); + + return 0; +} + +/* + * Set the SSP clock dividers. + */ +static int pxa2xx_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct ssp_info *info = cpu_dai->private_data; + struct ssp_device *ssp = info->dev.ssp; + int val; + + switch (div_id) { + case PXA2XX_SSP_AUDIO_DIV_ACDS: + val = (ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); + ssp_write_reg(ssp, SSACD, val); + break; + case PXA2XX_SSP_AUDIO_DIV_SCDB: + val = ssp_read_reg(ssp, SSACD); + val &= ~SSACD_SCDB; + if (cpu_is_pxa3xx()) + val &= ~SSACD_SCDX8; + switch (div) { + case PXA2XX_SSP_CLK_SCDB_1: + val |= SSACD_SCDB; + break; + case PXA2XX_SSP_CLK_SCDB_4: + break; + case PXA2XX_SSP_CLK_SCDB_8: + if (cpu_is_pxa3xx()) + val |= SSACD_SCDX8; + else + return -EINVAL; + break; + default: + return -EINVAL; + } + ssp_write_reg(ssp, SSACD, val); + break; + case PXA2XX_SSP_DIV_SCR: + ssp_set_scr(ssp, div); + break; + default: + return -ENODEV; + } + + return 0; +} + +/* + * Configure the PLL frequency pxa27x and (afaik - pxa320 only) + */ +static int pxa2xx_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct ssp_info *info = cpu_dai->private_data; + struct ssp_device *ssp = info->dev.ssp; + u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70; + + if (cpu_is_pxa3xx()) + ssp_write_reg(ssp, SSACDD, 0); + + switch (freq_out) { + case 5622000: + break; + case 11345000: + ssacd |= (0x1 << 4); + break; + case 12235000: + ssacd |= (0x2 << 4); + break; + case 14857000: + ssacd |= (0x3 << 4); + break; + case 32842000: + ssacd |= (0x4 << 4); + break; + case 48000000: + ssacd |= (0x5 << 4); + break; + case 0: + /* Disable */ + break; + + default: + /* 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()) { + u32 val; + u64 tmp = 19968; + tmp *= 1000000; + do_div(tmp, freq_out); + val = tmp; + + val = (val << 16) | 64; + ssp_write_reg(ssp, SSACDD, val); + + ssacd |= (0x6 << 4); + + dev_dbg(&ssp->pdev->dev, + "Using SSACDD %x to supply %uHz\n", + val, freq_out); + break; + } + + return -EINVAL; + } + + ssp_write_reg(ssp, SSACD, ssacd); + + return 0; +} + +/* + * Set up the SSP DAI format. + * The SSP Port must be inactive before calling this function as the + * physical interface format is changed. + */ +static int pxa2xx_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct ssp_info *info = cpu_dai->private_data; + struct ssp_device *ssp = info->dev.ssp; + u32 sscr0; + u32 sscr1; + u32 sspsp; + + /* check if we need to change anything at all */ + if (info->dai_fmt == fmt) + return 0; + + /* we can only change the settings if the port is not in use */ + if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { + dev_err(&ssp->pdev->dev, + "can't change hardware dai format: stream is in use"); + return -EINVAL; + } + + /* reset port settings */ + sscr0 = ssp_read_reg(ssp, SSCR0) & + (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); + sspsp = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + sscr1 |= SSCR1_SCLKDIR; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspsp |= SSPSP_SFRMP; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + case SND_SOC_DAIFMT_IB_IF: + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + sscr0 |= SSCR0_PSP; + sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; + /* See hw_params() */ + break; + + case SND_SOC_DAIFMT_DSP_A: + sspsp |= SSPSP_FSRT; + case SND_SOC_DAIFMT_DSP_B: + sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; + break; + + default: + return -EINVAL; + } + + ssp_write_reg(ssp, SSCR0, sscr0); + ssp_write_reg(ssp, SSCR1, sscr1); + ssp_write_reg(ssp, SSPSP, sspsp); + + dump_registers(ssp); + + /* 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. + */ + info->dai_fmt = fmt; + + return 0; +} + +/* + * Set the SSP audio DMA parameters and sample size. + * Can be called multiple times by oss emulation. + */ +static int pxa2xx_ssp_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->dai->cpu_dai; + struct ssp_info *info = cpu_dai->private_data; + struct ssp_device *ssp = info->dev.ssp; + int chn = params_channels(params); + u32 sscr0; + u32 sspsp; + int width = snd_pcm_format_physical_width(params_format(params)); + int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + + /* generate correct DMA params */ + if (cpu_dai->dma_data) + kfree(cpu_dai->dma_data); + + /* Network mode with one active slot (ttsa == 1) can be used + * to force 16-bit frame width on the wire (for S16_LE), even + * with two channels. Use 16-bit DMA transfers for this case. + */ + cpu_dai->dma_data = pxa_ssp_get_dma_params(ssp, + ((chn == 2) && (ttsa != 1)) || (width == 32), + substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + + /* we can only change the settings if the port is not in use */ + if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) + return 0; + + /* clear selected SSP bits */ + sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); + ssp_write_reg(ssp, SSCR0, sscr0); + + /* bit size */ + sscr0 = ssp_read_reg(ssp, SSCR0); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + if (cpu_is_pxa3xx()) + sscr0 |= SSCR0_FPCKE; + sscr0 |= SSCR0_DataSize(16); + break; + case SNDRV_PCM_FORMAT_S24_LE: + sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); + break; + case SNDRV_PCM_FORMAT_S32_LE: + sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); + break; + } + ssp_write_reg(ssp, SSCR0, sscr0); + + switch (info->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + sspsp = ssp_read_reg(ssp, SSPSP); + + if ((ssp_get_scr(ssp) == 4) && (width == 16)) { + /* This is a special case where the bitclk is 64fs + * and we're not dealing with 2*32 bits of audio + * samples. + * + * The SSP values used for that are all found out by + * trying and failing a lot; some of the registers + * needed for that mode are only available on PXA3xx. + */ + + if (!cpu_is_pxa3xx()) + return -EINVAL; + + sspsp |= SSPSP_SFRMWDTH(width * 2); + sspsp |= SSPSP_SFRMDLY(width * 4); + sspsp |= SSPSP_EDMYSTOP(3); + sspsp |= SSPSP_DMYSTOP(3); + sspsp |= SSPSP_DMYSTRT(1); + } else { + /* The frame width is the width the LRCLK is + * asserted for; the delay is expressed in + * half cycle units. We need the extra cycle + * because the data starts clocking out one BCLK + * after LRCLK changes polarity. + */ + sspsp |= SSPSP_SFRMWDTH(width + 1); + sspsp |= SSPSP_SFRMDLY((width + 1) * 2); + sspsp |= SSPSP_DMYSTRT(1); + } + + ssp_write_reg(ssp, SSPSP, sspsp); + break; + default: + break; + } + + /* When we use a network mode, we always require TDM slots + * - complain loudly and fail if they've not been set up yet. + */ + if ((sscr0 & SSCR0_MOD) && !ttsa) { + dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); + return -EINVAL; + } + + dump_registers(ssp); + + return 0; +} + +static struct snd_soc_dai_ops pxa2xx_ssp_dai_ops = { + .hw_params = pxa2xx_ssp_hw_params, + .set_sysclk = pxa2xx_ssp_set_dai_sysclk, + .set_clkdiv = pxa2xx_ssp_set_dai_clkdiv, + .set_pll = pxa2xx_ssp_set_dai_pll, + .set_fmt = pxa2xx_ssp_set_dai_fmt, +}; + +struct snd_soc_dai pxa2xx_ssp_dai[PXA2XX_DAI_SSP_MAX]; +EXPORT_SYMBOL(pxa2xx_ssp_dai); + +static int __devinit pxa2xx_ssp_dev_probe(struct platform_device *pdev) +{ + struct snd_soc_dai *dai; + int ret; + + if (pdev->id >= PXA2XX_DAI_SSP_MAX) { + dev_err(&pdev->dev, "id %d is out of range\n", pdev->id); + return -EINVAL; + } + + dai = &pxa2xx_ssp_dai[pdev->id]; + dai->dev = &pdev->dev; + dai->name = "pxa2xx-ssp"; + dai->id = pdev->id; + dai->playback.channels_min = 1; + dai->playback.channels_max = 8; + dai->playback.rates = PXA2XX_SSP_RATES; + dai->playback.formats = PXA2XX_SSP_FORMATS; + dai->capture.channels_min = 1; + dai->capture.channels_max = 8; + dai->capture.rates = PXA2XX_SSP_RATES; + dai->capture.formats = PXA2XX_SSP_FORMATS; + dai->ops = &pxa2xx_ssp_dai_ops; + + ret = pxa_ssp_register_dai(dai); + return ret; +} + +static int __devexit pxa2xx_ssp_dev_remove(struct platform_device *pdev) +{ + struct snd_soc_dai *dai; + + dai = &pxa2xx_ssp_dai[pdev->id]; + snd_soc_unregister_dai(dai); + return 0; +} + +static struct platform_driver pxa2xx_ssp_driver = { + .probe = pxa2xx_ssp_dev_probe, + .remove = __devexit_p(pxa2xx_ssp_dev_remove), + .driver = { + .name = "pxa2xx-ssp", + .owner = THIS_MODULE, + }, +}; + +static int __init pxa2xx_ssp_init(void) +{ + return platform_driver_register(&pxa2xx_ssp_driver); +} +module_init(pxa2xx_ssp_init); + +static void __exit pxa2xx_ssp_exit(void) +{ + platform_driver_unregister(&pxa2xx_ssp_driver); +} +module_exit(pxa2xx_ssp_exit); + +/* Module information */ +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa2xx-ssp.h b/sound/soc/pxa/pxa2xx-ssp.h new file mode 100644 index 0000000..2455bf4 --- /dev/null +++ b/sound/soc/pxa/pxa2xx-ssp.h @@ -0,0 +1,59 @@ +/* + * ASoC PXA SSP port support + * + * 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 __PXA2XX_SOC_SSP_H +#define __PXA2XX_SOC_SSP_H + +#define PXA2XX_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +#define PXA2XX_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* pxa DAI SSP IDs */ +enum { + PXA2XX_DAI_SSP1, + PXA2XX_DAI_SSP2, + PXA2XX_DAI_SSP3, + PXA2XX_DAI_SSP4, + PXA2XX_DAI_SSP_MAX, +}; + +/* SSP clock sources */ +#define PXA2XX_SSP_CLK_PLL 0 +#define PXA2XX_SSP_CLK_EXT 1 +#define PXA2XX_SSP_CLK_NET 2 +#define PXA2XX_SSP_CLK_AUDIO 3 +#define PXA2XX_SSP_CLK_NET_PLL 4 + +/* SSP audio dividers */ +#define PXA2XX_SSP_AUDIO_DIV_ACDS 0 +#define PXA2XX_SSP_AUDIO_DIV_SCDB 1 +#define PXA2XX_SSP_DIV_SCR 2 + +/* SSP ACDS audio dividers values */ +#define PXA2XX_SSP_CLK_AUDIO_DIV_1 0 +#define PXA2XX_SSP_CLK_AUDIO_DIV_2 1 +#define PXA2XX_SSP_CLK_AUDIO_DIV_4 2 +#define PXA2XX_SSP_CLK_AUDIO_DIV_8 3 +#define PXA2XX_SSP_CLK_AUDIO_DIV_16 4 +#define PXA2XX_SSP_CLK_AUDIO_DIV_32 5 + +/* SSP divider bypass */ +#define PXA2XX_SSP_CLK_SCDB_4 0 +#define PXA2XX_SSP_CLK_SCDB_1 1 +#define PXA2XX_SSP_CLK_SCDB_8 2 + +#define PXA2XX_SSP_PLL_OUT 0 + +extern struct snd_soc_dai pxa2xx_ssp_dai[PXA2XX_DAI_SSP_MAX]; + +#endif /* __PXA2XX_SOC_SSP_H */ diff --git a/sound/soc/pxa/ssp.c b/sound/soc/pxa/ssp.c new file mode 100644 index 0000000..444b643 --- /dev/null +++ b/sound/soc/pxa/ssp.c @@ -0,0 +1,298 @@ +/* + * ssp.c -- ALSA Soc Audio Layer + * + * Copyright 2009-2010 Marvell International Ltd. + * Author: + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * 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/clk.h> +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/pxa2xx-lib.h> + +#include <mach/hardware.h> +#include <mach/dma.h> +#include <mach/regs-ssp.h> + +#include <plat/ssp.h> + +#include "ssp.h" + +struct pxa2xx_pcm_dma_data { + struct pxa2xx_pcm_dma_params params; + char name[20]; +}; + +struct pxa2xx_pcm_dma_params * +pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) +{ + struct pxa2xx_pcm_dma_data *dma; + + dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL); + if (dma == NULL) + return NULL; + + snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id, + width4 ? "32-bit" : "16-bit", out ? "out" : "in"); + + dma->params.name = dma->name; + dma->params.drcmr = &DRCMR(out ? ssp->drcmr_tx : ssp->drcmr_rx); + dma->params.dcmd = (out ? (DCMD_INCSRCADDR | DCMD_FLOWTRG) : + (DCMD_INCTRGADDR | DCMD_FLOWSRC)) | + (width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16; + dma->params.dev_addr = ssp->phys_base + SSDR; + + return &dma->params; +} +EXPORT_SYMBOL(pxa_ssp_get_dma_params); + +static int pxa_ssp_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->dai->cpu_dai; + struct ssp_info *info = cpu_dai->private_data; + int ret = 0; + + if (!cpu_dai->active) { + info->dev.port = cpu_dai->id + 1; + info->dev.irq = NO_IRQ; + clk_enable(info->dev.ssp->clk); + ssp_disable(&info->dev); + } + + if (cpu_dai->dma_data) { + kfree(cpu_dai->dma_data); + cpu_dai->dma_data = NULL; + } + return ret; +} + +static void pxa_ssp_shutdown(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->dai->cpu_dai; + struct ssp_info *info = cpu_dai->private_data; + + if (!cpu_dai->active) { + ssp_disable(&info->dev); + clk_disable(info->dev.ssp->clk); + } + + if (cpu_dai->dma_data) { + kfree(cpu_dai->dma_data); + cpu_dai->dma_data = NULL; + } +} + +#ifdef CONFIG_PM + +static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) +{ + struct ssp_info *info = cpu_dai->private_data; + + if (!cpu_dai->active) + clk_enable(info->dev.ssp->clk); + + ssp_save_state(&info->dev, &info->state); + clk_disable(info->dev.ssp->clk); + + return 0; +} + +static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) +{ + struct ssp_info *info = cpu_dai->private_data; + + clk_enable(info->dev.ssp->clk); + ssp_restore_state(&info->dev, &info->state); + + if (cpu_dai->active) + ssp_enable(&info->dev); + else + clk_disable(info->dev.ssp->clk); + + return 0; +} + +#else +#define pxa_ssp_suspend NULL +#define pxa_ssp_resume NULL +#endif + +/* + * Set the active slots in TDM/Network mode + */ +static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct ssp_info *info = cpu_dai->private_data; + struct ssp_device *ssp = info->dev.ssp; + u32 sscr0; + + sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS); + + /* set slot width */ + if (slot_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(slot_width - 16); + else + sscr0 |= SSCR0_DataSize(slot_width); + + if (slots > 1) { + /* enable network mode */ + sscr0 |= SSCR0_MOD; + + /* set number of active slots */ + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* set active slot mask */ + ssp_write_reg(ssp, SSTSA, tx_mask); + ssp_write_reg(ssp, SSRSA, rx_mask); + } + ssp_write_reg(ssp, SSCR0, sscr0); + + return 0; +} + +/* + * Tristate the SSP DAI lines + */ +static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, + int tristate) +{ + struct ssp_info *info = cpu_dai->private_data; + struct ssp_device *ssp = info->dev.ssp; + u32 sscr1; + + sscr1 = ssp_read_reg(ssp, SSCR1); + if (tristate) + sscr1 &= ~SSCR1_TTE; + else + sscr1 |= SSCR1_TTE; + ssp_write_reg(ssp, SSCR1, sscr1); + + return 0; +} + +static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + struct ssp_info *info = cpu_dai->private_data; + struct ssp_device *ssp = info->dev.ssp; + int val; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + ssp_enable(&info->dev); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= SSCR1_TSRE; + else + val |= SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + val = ssp_read_reg(ssp, SSSR); + ssp_write_reg(ssp, SSSR, val); + break; + case SNDRV_PCM_TRIGGER_START: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= SSCR1_TSRE; + else + val |= SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + ssp_enable(&info->dev); + break; + case SNDRV_PCM_TRIGGER_STOP: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val &= ~SSCR1_TSRE; + else + val &= ~SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + ssp_disable(&info->dev); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val &= ~SSCR1_TSRE; + else + val &= ~SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static int pxa_ssp_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct ssp_info *info; + int ret; + + info = kzalloc(sizeof(struct ssp_info), GFP_KERNEL); + if (!info) + return -ENOMEM; + + info->dev.ssp = ssp_request(dai->id + 1, "SoC audio"); + if (info->dev.ssp == NULL) { + ret = -ENODEV; + goto err; + } + + info->dai_fmt = (unsigned int) -1; + dai->private_data = info; + + return 0; + +err: + kfree(info); + return ret; +} + +static void pxa_ssp_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct ssp_info *info = dai->private_data; + ssp_free(info->dev.ssp); +} + +int pxa_ssp_register_dai(struct snd_soc_dai *dai) +{ + struct snd_soc_dai_ops *ops = dai->ops; + + ops->startup = pxa_ssp_startup; + ops->shutdown = pxa_ssp_shutdown; + ops->trigger = pxa_ssp_trigger; + ops->set_tdm_slot = pxa_ssp_set_dai_tdm_slot; + ops->set_tristate = pxa_ssp_set_dai_tristate; + + dai->probe = pxa_ssp_probe; + dai->remove = pxa_ssp_remove; + dai->suspend = pxa_ssp_suspend; + dai->resume = pxa_ssp_resume; + + return snd_soc_register_dai(dai); +} +EXPORT_SYMBOL(pxa_ssp_register_dai); diff --git a/sound/soc/pxa/ssp.h b/sound/soc/pxa/ssp.h new file mode 100644 index 0000000..314c06d --- /dev/null +++ b/sound/soc/pxa/ssp.h @@ -0,0 +1,42 @@ +/* + * ssp.h -- ALSA Soc Audio Layer Head file + * + * Copyright 2009-2010 Marvell International Ltd. + * Author: + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * 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. + * + */ + +#ifndef __PXA_SSP_H +#define __PXA_SSP_H + +/* + * SSP audio data + */ +struct ssp_info { + struct ssp_dev dev; + unsigned int sysclk; + int dai_fmt; +#ifdef CONFIG_PM + struct ssp_state state; +#endif +}; + +struct dai_ssp { + unsigned int sysclk; + int dai_fmt; +#ifdef CONFIG_PM + struct ssp_state state; +#endif +}; + +extern struct pxa2xx_pcm_dma_params * +pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out); +extern int pxa_ssp_register_dai(struct snd_soc_dai *dai); + +#endif /* __PXA_SSP_H */ diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c new file mode 100644 index 0000000..3ee39d4 --- /dev/null +++ b/sound/soc/pxa/tavorevb3.c @@ -0,0 +1,193 @@ +/* + * tavorevb3.c -- SoC audio for Tavor EVB3 + * + * Copyright (C) 2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * 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 <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/i2c.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/jack.h> + +#include <asm/mach-types.h> + +#include "../codecs/88pm860x-codec.h" +#include "pxa-ssp.h" + +static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd); + +static struct platform_device *evb3_snd_device; + +static struct snd_soc_jack hs_jack; + +static struct snd_soc_jack_pin hs_jack_pins[] = { + { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, + { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, +}; + +/* tavorevb3 machine dapm widgets */ +static const struct snd_soc_dapm_widget evb3_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), +}; + +/* tavorevb3 machine audio map */ +static const struct snd_soc_dapm_route 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 evb3_i2s_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 width = snd_pcm_format_physical_width(params_format(params)); + int ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, + PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); + return ret; +} + +static struct snd_soc_ops evb3_i2s_ops = { + .hw_params = evb3_i2s_hw_params, +}; + +static struct snd_soc_dai_link evb3_dai[] = { + { + .name = "88PM860x I2S", + .stream_name = "I2S Audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .platform_name = "pxa-pcm-audio", + .codec_name = "88pm860x-codec", + .init = evb3_pm860x_init, + .ops = &evb3_i2s_ops, + }, +}; + +static struct snd_soc_card snd_soc_card_evb3 = { + .name = "Tavor EVB3", + .dai_link = evb3_dai, + .num_links = ARRAY_SIZE(evb3_dai), +}; + +static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + snd_soc_dapm_new_controls(codec, evb3_dapm_widgets, + ARRAY_SIZE(evb3_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Speaker"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 1"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 3"); + snd_soc_dapm_disable_pin(codec, "Headset Mic 2"); + snd_soc_dapm_disable_pin(codec, "Headset Stereophone"); + + ret = snd_soc_dapm_sync(codec); + if (ret) + return ret; + + /* Headset jack detection */ + snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET + | 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); + /* headphone, microphone detection & headset short detection */ + pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET, + SND_JACK_BTN_0); + pm860x_hook_detect(codec, &hs_jack, SND_JACK_BTN_1, SND_JACK_BTN_2); + return 0; +} + +static int __init tavorevb3_init(void) +{ + int ret; + + if (!machine_is_tavorevb3()) + return -ENODEV; + evb3_snd_device = platform_device_alloc("soc-audio", -1); + if (!evb3_snd_device) + return -ENOMEM; + + platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3); + + ret = platform_device_add(evb3_snd_device); + if (ret) + platform_device_put(evb3_snd_device); + + return ret; +} + +static void __exit tavorevb3_exit(void) +{ + platform_device_unregister(evb3_snd_device); +} + +module_init(tavorevb3_init); +module_exit(tavorevb3_exit); + +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3"); +MODULE_LICENSE("GPL"); + -- 1.5.6.5

1 0

[PATCH 2/3] ASoC: add tavorevb3 machine driver for 88pm860x
by Haojian Zhuang 17 Aug '10

17 Aug '10

88PM860x codec is used in Marvell tavorevb3 development board. 88PM860x codec is used as master mode of SSP communication. Only I2S format is supported. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/tavorevb3.c | 180 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 191 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/tavorevb3.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index e30c832..04ddc7b 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -117,6 +117,15 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. +config SND_SOC_TAVOREVB3 + tristate "SoC Audio support for Marvell Tavor EVB3" + depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 + select SND_PXA_SOC_SSP + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on the + Marvell Saarb reference platform. + config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index caa03d8..315941f 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -19,6 +19,7 @@ snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o +snd-soc-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o @@ -38,6 +39,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o +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 diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c new file mode 100644 index 0000000..194f81a --- /dev/null +++ b/sound/soc/pxa/tavorevb3.c @@ -0,0 +1,180 @@ +/* + * tavorevb3.c -- SoC audio for Tavor EVB3 + * + * Copyright (C) 2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * 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 <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "../codecs/88pm860x-codec.h" +#include "pxa-ssp.h" + +static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd); + +static struct platform_device *evb3_snd_device; + +/* tavorevb3 machine dapm widgets */ +static const struct snd_soc_dapm_widget evb3_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_LINE("Lineout Out 1", NULL), + SND_SOC_DAPM_LINE("Lineout Out 2", NULL), + SND_SOC_DAPM_SPK("Board Speaker", NULL), + SND_SOC_DAPM_MIC("Board Mic 1", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Board Mic 3", NULL), +}; + +/* tavorevb3 machine audio map */ +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone", NULL, "HS1"}, + {"Headphone", NULL, "HS2"}, + + {"Board Speaker", NULL, "LSP"}, + {"Board Speaker", NULL, "LSN"}, + + {"Lineout Out 1", NULL, "LINEOUT1"}, + {"Lineout Out 2", NULL, "LINEOUT2"}, + + {"MIC1P", NULL, "Mic1 Bias"}, + {"MIC1N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Board Mic 1"}, + + {"MIC2P", NULL, "Mic1 Bias"}, + {"MIC2N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Headset Mic"}, + + {"MIC3P", NULL, "Mic3 Bias"}, + {"MIC3N", NULL, "Mic3 Bias"}, + {"Mic3 Bias", NULL, "Board Mic 3"}, +}; + +static const struct snd_kcontrol_new pm860x_evb3_controls[] = { + SOC_DAPM_PIN_SWITCH("Board Speaker"), + SOC_DAPM_PIN_SWITCH("Board Mic 1"), + SOC_DAPM_PIN_SWITCH("Board Mic 3"), +}; + +static int evb3_i2s_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 width = snd_pcm_format_physical_width(params_format(params)); + int ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, + PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); + return ret; +} + +static struct snd_soc_ops evb3_i2s_ops = { + .hw_params = evb3_i2s_hw_params, +}; + +static struct snd_soc_dai_link evb3_dai[] = { + { + .name = "88PM860x I2S", + .stream_name = "I2S Audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .platform_name = "pxa-pcm-audio", + .codec_name = "88pm860x-codec", + .init = evb3_pm860x_init, + .ops = &evb3_i2s_ops, + }, +}; + +static struct snd_soc_card snd_soc_card_evb3 = { + .name = "Tavor EVB3", + .dai_link = evb3_dai, + .num_links = ARRAY_SIZE(evb3_dai), +}; + +static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + ret = snd_soc_add_controls(codec, pm860x_evb3_controls, + ARRAY_SIZE(pm860x_evb3_controls)); + if (ret < 0) + return ret; + + snd_soc_dapm_new_controls(codec, evb3_dapm_widgets, + ARRAY_SIZE(evb3_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_disable_pin(codec, "Board Speaker"); + snd_soc_dapm_disable_pin(codec, "Board Mic 1"); + snd_soc_dapm_disable_pin(codec, "Board Mic 3"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + + return snd_soc_dapm_sync(codec); +} + +static int __init tavorevb3_init(void) +{ + int ret; + + if (!machine_is_tavorevb3()) + return -ENODEV; + evb3_snd_device = platform_device_alloc("soc-audio", -1); + if (!evb3_snd_device) + return -ENOMEM; + + platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3); + + ret = platform_device_add(evb3_snd_device); + if (ret) + platform_device_put(evb3_snd_device); + + return ret; +} + +static void __exit tavorevb3_exit(void) +{ + platform_device_unregister(evb3_snd_device); +} + +module_init(tavorevb3_init); +module_exit(tavorevb3_exit); + +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3"); +MODULE_LICENSE("GPL"); + -- 1.5.6.5 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[alsa-devel] Temporary ASoC tree for next
by Mark Brown 17 Aug '10

17 Aug '10

Hi Stephen, Liam has recently done some work on ASoC to enhance the ability to work with multiple external devices which we're expecting to get in for .37. Since this work required some architecture modifications and we've had a bit of trouble getting testers we're expecting a higher than average degree of cross-tree issues this cycle so in order to avoid affecting all other ALSA work we'd like to have a separate ASoC-specific tree in -next for this release cycle if possible: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-next Please also include Liam as a contact for the tree. Thanks, Mark

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