diff options
Diffstat (limited to 'ANDROID_3.4.5/sound/soc/davinci')
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/davinci/Kconfig | 85 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/davinci/Makefile | 20 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/davinci/davinci-evm.c | 339 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/davinci/davinci-i2s.c | 768 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/davinci/davinci-i2s.h | 20 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/davinci/davinci-mcasp.c | 983 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/davinci/davinci-mcasp.h | 59 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/davinci/davinci-pcm.c | 892 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/davinci/davinci-pcm.h | 31 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/davinci/davinci-sffsdr.c | 181 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/davinci/davinci-vcif.c | 266 |
11 files changed, 3644 insertions, 0 deletions
diff --git a/ANDROID_3.4.5/sound/soc/davinci/Kconfig b/ANDROID_3.4.5/sound/soc/davinci/Kconfig new file mode 100644 index 00000000..9e11a14d --- /dev/null +++ b/ANDROID_3.4.5/sound/soc/davinci/Kconfig @@ -0,0 +1,85 @@ +config SND_DAVINCI_SOC + tristate "SoC Audio for the TI DAVINCI chip" + depends on ARCH_DAVINCI + help + Say Y or M if you want to add support for codecs attached to + the DAVINCI AC97 or I2S interface. You will also need + to select the audio interfaces to support below. + +config SND_DAVINCI_SOC_I2S + tristate + +config SND_DAVINCI_SOC_MCASP + tristate + +config SND_DAVINCI_SOC_VCIF + tristate + +config SND_DAVINCI_SOC_EVM + tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" + depends on SND_DAVINCI_SOC + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM + select SND_DAVINCI_SOC_I2S + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on TI + DaVinci DM6446, DM355 or DM365 EVM platforms. + +choice + prompt "DM365 codec select" + depends on SND_DAVINCI_SOC_EVM + depends on MACH_DAVINCI_DM365_EVM + +config SND_DM365_AIC3X_CODEC + bool "Audio Codec - AIC3101" + help + Say Y if you want to add support for AIC3101 audio codec + +config SND_DM365_VOICE_CODEC + bool "Voice Codec - CQ93VC" + select MFD_DAVINCI_VOICECODEC + select SND_DAVINCI_SOC_VCIF + select SND_SOC_CQ0093VC + help + Say Y if you want to add support for SoC On-chip voice codec +endchoice + +config SND_DM6467_SOC_EVM + tristate "SoC Audio support for DaVinci DM6467 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM + select SND_DAVINCI_SOC_MCASP + select SND_SOC_TLV320AIC3X + select SND_SOC_SPDIF + + help + Say Y if you want to add support for SoC audio on TI + +config SND_DAVINCI_SOC_SFFSDR + tristate "SoC Audio support for SFFSDR" + depends on SND_DAVINCI_SOC && MACH_SFFSDR + select SND_DAVINCI_SOC_I2S + select SND_SOC_PCM3008 + select SFFSDR_FPGA + help + Say Y if you want to add support for SoC audio on + Lyrtech SFFSDR board. + +config SND_DA830_SOC_EVM + tristate "SoC Audio support for DA830/OMAP-L137 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM + select SND_DAVINCI_SOC_MCASP + select SND_SOC_TLV320AIC3X + + help + Say Y if you want to add support for SoC audio on TI + DA830/OMAP-L137 EVM + +config SND_DA850_SOC_EVM + tristate "SoC Audio support for DA850/OMAP-L138 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA850_EVM + select SND_DAVINCI_SOC_MCASP + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on TI + DA850/OMAP-L138 EVM + diff --git a/ANDROID_3.4.5/sound/soc/davinci/Makefile b/ANDROID_3.4.5/sound/soc/davinci/Makefile new file mode 100644 index 00000000..a93679d6 --- /dev/null +++ b/ANDROID_3.4.5/sound/soc/davinci/Makefile @@ -0,0 +1,20 @@ +# DAVINCI Platform Support +snd-soc-davinci-objs := davinci-pcm.o +snd-soc-davinci-i2s-objs := davinci-i2s.o +snd-soc-davinci-mcasp-objs:= davinci-mcasp.o +snd-soc-davinci-vcif-objs:= davinci-vcif.o + +obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o +obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o +obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o +obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o + +# DAVINCI Machine Support +snd-soc-evm-objs := davinci-evm.o +snd-soc-sffsdr-objs := davinci-sffsdr.o + +obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DA850_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/ANDROID_3.4.5/sound/soc/davinci/davinci-evm.c b/ANDROID_3.4.5/sound/soc/davinci/davinci-evm.c new file mode 100644 index 00000000..10a2d8c7 --- /dev/null +++ b/ANDROID_3.4.5/sound/soc/davinci/davinci-evm.c @@ -0,0 +1,339 @@ +/* + * ASoC driver for TI DAVINCI EVM platform + * + * Author: Vladimir Barinov, <vbarinov@embeddedalley.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.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/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> + +#include <asm/dma.h> +#include <asm/mach-types.h> + +#include <mach/asp.h> +#include <mach/edma.h> +#include <mach/mux.h> + +#include "davinci-pcm.h" +#include "davinci-i2s.h" +#include "davinci-mcasp.h" + +#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ + SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) +static int evm_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 ret = 0; + unsigned sysclk; + + /* ASP1 on DM355 EVM is clocked by an external oscillator */ + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() || + machine_is_davinci_dm365_evm()) + sysclk = 27000000; + + /* ASP0 in DM6446 EVM is clocked by U55, as configured by + * board-dm644x-evm.c using GPIOs from U18. There are six + * options; here we "know" we use a 48 KHz sample rate. + */ + else if (machine_is_davinci_evm()) + sysclk = 12288000; + + else if (machine_is_davinci_da830_evm() || + machine_is_davinci_da850_evm()) + sysclk = 24576000; + + else + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static int evm_spdif_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 *cpu_dai = rtd->cpu_dai; + + /* set cpu DAI configuration */ + return snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); +} + +static struct snd_soc_ops evm_ops = { + .hw_params = evm_hw_params, +}; + +static struct snd_soc_ops evm_spdif_ops = { + .hw_params = evm_spdif_hw_params, +}; + +/* davinci-evm machine dapm widgets */ +static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +/* davinci-evm machine audio_mapnections to the codec pins */ +static const struct snd_soc_dapm_route audio_map[] = { + /* Headphone connected to HPLOUT, HPROUT */ + {"Headphone Jack", NULL, "HPLOUT"}, + {"Headphone Jack", NULL, "HPROUT"}, + + /* Line Out connected to LLOUT, RLOUT */ + {"Line Out", NULL, "LLOUT"}, + {"Line Out", NULL, "RLOUT"}, + + /* Mic connected to (MIC3L | MIC3R) */ + {"MIC3L", NULL, "Mic Bias 2V"}, + {"MIC3R", NULL, "Mic Bias 2V"}, + {"Mic Bias 2V", NULL, "Mic Jack"}, + + /* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */ + {"LINE1L", NULL, "Line In"}, + {"LINE2L", NULL, "Line In"}, + {"LINE1R", NULL, "Line In"}, + {"LINE2R", NULL, "Line In"}, +}; + +/* Logic for a aic3x as connected on a davinci-evm */ +static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + /* Add davinci-evm specific widgets */ + snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, + ARRAY_SIZE(aic3x_dapm_widgets)); + + /* Set up davinci-evm specific audio path audio_map */ + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + + /* not connected */ + snd_soc_dapm_disable_pin(dapm, "MONO_LOUT"); + snd_soc_dapm_disable_pin(dapm, "HPLCOM"); + snd_soc_dapm_disable_pin(dapm, "HPRCOM"); + + /* always connected */ + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Line Out"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Line In"); + + return 0; +} + +/* davinci-evm digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link dm6446_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai_name = "davinci-mcbsp", + .codec_dai_name = "tlv320aic3x-hifi", + .codec_name = "tlv320aic3x-codec.1-001b", + .platform_name = "davinci-pcm-audio", + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +static struct snd_soc_dai_link dm355_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai_name = "davinci-mcbsp.1", + .codec_dai_name = "tlv320aic3x-hifi", + .codec_name = "tlv320aic3x-codec.1-001b", + .platform_name = "davinci-pcm-audio", + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +static struct snd_soc_dai_link dm365_evm_dai = { +#ifdef CONFIG_SND_DM365_AIC3X_CODEC + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai_name = "davinci-mcbsp", + .codec_dai_name = "tlv320aic3x-hifi", + .init = evm_aic3x_init, + .codec_name = "tlv320aic3x-codec.1-0018", + .ops = &evm_ops, +#elif defined(CONFIG_SND_DM365_VOICE_CODEC) + .name = "Voice Codec - CQ93VC", + .stream_name = "CQ93", + .cpu_dai_name = "davinci-vcif", + .codec_dai_name = "cq93vc-hifi", + .codec_name = "cq93vc-codec", +#endif + .platform_name = "davinci-pcm-audio", +}; + +static struct snd_soc_dai_link dm6467_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai_name= "davinci-mcasp.0", + .codec_dai_name = "tlv320aic3x-hifi", + .platform_name ="davinci-pcm-audio", + .codec_name = "tlv320aic3x-codec.0-001a", + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "McASP", + .stream_name = "spdif", + .cpu_dai_name= "davinci-mcasp.1", + .codec_dai_name = "dit-hifi", + .codec_name = "spdif_dit", + .platform_name = "davinci-pcm-audio", + .ops = &evm_spdif_ops, + }, +}; + +static struct snd_soc_dai_link da830_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai_name = "davinci-mcasp.1", + .codec_dai_name = "tlv320aic3x-hifi", + .codec_name = "tlv320aic3x-codec.1-0018", + .platform_name = "davinci-pcm-audio", + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +static struct snd_soc_dai_link da850_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai_name= "davinci-mcasp.0", + .codec_dai_name = "tlv320aic3x-hifi", + .codec_name = "tlv320aic3x-codec.1-0018", + .platform_name = "davinci-pcm-audio", + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +/* davinci dm6446 evm audio machine driver */ +static struct snd_soc_card dm6446_snd_soc_card_evm = { + .name = "DaVinci DM6446 EVM", + .owner = THIS_MODULE, + .dai_link = &dm6446_evm_dai, + .num_links = 1, +}; + +/* davinci dm355 evm audio machine driver */ +static struct snd_soc_card dm355_snd_soc_card_evm = { + .name = "DaVinci DM355 EVM", + .owner = THIS_MODULE, + .dai_link = &dm355_evm_dai, + .num_links = 1, +}; + +/* davinci dm365 evm audio machine driver */ +static struct snd_soc_card dm365_snd_soc_card_evm = { + .name = "DaVinci DM365 EVM", + .owner = THIS_MODULE, + .dai_link = &dm365_evm_dai, + .num_links = 1, +}; + +/* davinci dm6467 evm audio machine driver */ +static struct snd_soc_card dm6467_snd_soc_card_evm = { + .name = "DaVinci DM6467 EVM", + .owner = THIS_MODULE, + .dai_link = dm6467_evm_dai, + .num_links = ARRAY_SIZE(dm6467_evm_dai), +}; + +static struct snd_soc_card da830_snd_soc_card = { + .name = "DA830/OMAP-L137 EVM", + .owner = THIS_MODULE, + .dai_link = &da830_evm_dai, + .num_links = 1, +}; + +static struct snd_soc_card da850_snd_soc_card = { + .name = "DA850/OMAP-L138 EVM", + .owner = THIS_MODULE, + .dai_link = &da850_evm_dai, + .num_links = 1, +}; + +static struct platform_device *evm_snd_device; + +static int __init evm_init(void) +{ + struct snd_soc_card *evm_snd_dev_data; + int index; + int ret; + + if (machine_is_davinci_evm()) { + evm_snd_dev_data = &dm6446_snd_soc_card_evm; + index = 0; + } else if (machine_is_davinci_dm355_evm()) { + evm_snd_dev_data = &dm355_snd_soc_card_evm; + index = 1; + } else if (machine_is_davinci_dm365_evm()) { + evm_snd_dev_data = &dm365_snd_soc_card_evm; + index = 0; + } else if (machine_is_davinci_dm6467_evm()) { + evm_snd_dev_data = &dm6467_snd_soc_card_evm; + index = 0; + } else if (machine_is_davinci_da830_evm()) { + evm_snd_dev_data = &da830_snd_soc_card; + index = 1; + } else if (machine_is_davinci_da850_evm()) { + evm_snd_dev_data = &da850_snd_soc_card; + index = 0; + } else + return -EINVAL; + + evm_snd_device = platform_device_alloc("soc-audio", index); + if (!evm_snd_device) + return -ENOMEM; + + platform_set_drvdata(evm_snd_device, evm_snd_dev_data); + ret = platform_device_add(evm_snd_device); + if (ret) + platform_device_put(evm_snd_device); + + return ret; +} + +static void __exit evm_exit(void) +{ + platform_device_unregister(evm_snd_device); +} + +module_init(evm_init); +module_exit(evm_exit); + +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_DESCRIPTION("TI DAVINCI EVM ASoC driver"); +MODULE_LICENSE("GPL"); diff --git a/ANDROID_3.4.5/sound/soc/davinci/davinci-i2s.c b/ANDROID_3.4.5/sound/soc/davinci/davinci-i2s.c new file mode 100644 index 00000000..0a74b958 --- /dev/null +++ b/ANDROID_3.4.5/sound/soc/davinci/davinci-i2s.c @@ -0,0 +1,768 @@ +/* + * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor + * + * Author: Vladimir Barinov, <vbarinov@embeddedalley.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.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/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/asp.h> + +#include "davinci-pcm.h" +#include "davinci-i2s.h" + + +/* + * NOTE: terminology here is confusing. + * + * - This driver supports the "Audio Serial Port" (ASP), + * found on dm6446, dm355, and other DaVinci chips. + * + * - But it labels it a "Multi-channel Buffered Serial Port" + * (McBSP) as on older chips like the dm642 ... which was + * backward-compatible, possibly explaining that confusion. + * + * - OMAP chips have a controller called McBSP, which is + * incompatible with the DaVinci flavor of McBSP. + * + * - Newer DaVinci chips have a controller called McASP, + * incompatible with ASP and with either McBSP. + * + * In short: this uses ASP to implement I2S, not McBSP. + * And it won't be the only DaVinci implemention of I2S. + */ +#define DAVINCI_MCBSP_DRR_REG 0x00 +#define DAVINCI_MCBSP_DXR_REG 0x04 +#define DAVINCI_MCBSP_SPCR_REG 0x08 +#define DAVINCI_MCBSP_RCR_REG 0x0c +#define DAVINCI_MCBSP_XCR_REG 0x10 +#define DAVINCI_MCBSP_SRGR_REG 0x14 +#define DAVINCI_MCBSP_PCR_REG 0x24 + +#define DAVINCI_MCBSP_SPCR_RRST (1 << 0) +#define DAVINCI_MCBSP_SPCR_RINTM(v) ((v) << 4) +#define DAVINCI_MCBSP_SPCR_XRST (1 << 16) +#define DAVINCI_MCBSP_SPCR_XINTM(v) ((v) << 20) +#define DAVINCI_MCBSP_SPCR_GRST (1 << 22) +#define DAVINCI_MCBSP_SPCR_FRST (1 << 23) +#define DAVINCI_MCBSP_SPCR_FREE (1 << 25) + +#define DAVINCI_MCBSP_RCR_RWDLEN1(v) ((v) << 5) +#define DAVINCI_MCBSP_RCR_RFRLEN1(v) ((v) << 8) +#define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) +#define DAVINCI_MCBSP_RCR_RFIG (1 << 18) +#define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_RCR_RFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_RCR_RPHASE BIT(31) + +#define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) +#define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) +#define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) +#define DAVINCI_MCBSP_XCR_XFIG (1 << 18) +#define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_XCR_XFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_XCR_XPHASE BIT(31) + +#define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) +#define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) +#define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) +#define DAVINCI_MCBSP_SRGR_CLKSM BIT(29) + +#define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) +#define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) +#define DAVINCI_MCBSP_PCR_FSRP (1 << 2) +#define DAVINCI_MCBSP_PCR_FSXP (1 << 3) +#define DAVINCI_MCBSP_PCR_SCLKME (1 << 7) +#define DAVINCI_MCBSP_PCR_CLKRM (1 << 8) +#define DAVINCI_MCBSP_PCR_CLKXM (1 << 9) +#define DAVINCI_MCBSP_PCR_FSRM (1 << 10) +#define DAVINCI_MCBSP_PCR_FSXM (1 << 11) + +enum { + DAVINCI_MCBSP_WORD_8 = 0, + DAVINCI_MCBSP_WORD_12, + DAVINCI_MCBSP_WORD_16, + DAVINCI_MCBSP_WORD_20, + DAVINCI_MCBSP_WORD_24, + DAVINCI_MCBSP_WORD_32, +}; + +static const unsigned char data_type[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = 1, + [SNDRV_PCM_FORMAT_S16_LE] = 2, + [SNDRV_PCM_FORMAT_S32_LE] = 4, +}; + +static const unsigned char asp_word_length[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = DAVINCI_MCBSP_WORD_8, + [SNDRV_PCM_FORMAT_S16_LE] = DAVINCI_MCBSP_WORD_16, + [SNDRV_PCM_FORMAT_S32_LE] = DAVINCI_MCBSP_WORD_32, +}; + +static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = SNDRV_PCM_FORMAT_S16_LE, + [SNDRV_PCM_FORMAT_S16_LE] = SNDRV_PCM_FORMAT_S32_LE, +}; + +struct davinci_mcbsp_dev { + struct device *dev; + struct davinci_pcm_dma_params dma_params[2]; + void __iomem *base; +#define MOD_DSP_A 0 +#define MOD_DSP_B 1 + int mode; + u32 pcr; + struct clk *clk; + /* + * Combining both channels into 1 element will at least double the + * amount of time between servicing the dma channel, increase + * effiency, and reduce the chance of overrun/underrun. But, + * it will result in the left & right channels being swapped. + * + * If relabeling the left and right channels is not possible, + * you may want to let the codec know to swap them back. + * + * It may allow x10 the amount of time to service dma requests, + * if the codec is master and is using an unnecessarily fast bit clock + * (ie. tlvaic23b), independent of the sample rate. So, having an + * entire frame at once means it can be serviced at the sample rate + * instead of the bit clock rate. + * + * In the now unlikely case that an underrun still + * occurs, both the left and right samples will be repeated + * so that no pops are heard, and the left and right channels + * won't end up being swapped because of the underrun. + */ + unsigned enable_channel_combine:1; + + unsigned int fmt; + int clk_div; + int clk_input_pin; + bool i2s_accurate_sck; +}; + +static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, + int reg, u32 val) +{ + __raw_writel(val, dev->base + reg); +} + +static inline u32 davinci_mcbsp_read_reg(struct davinci_mcbsp_dev *dev, int reg) +{ + return __raw_readl(dev->base + reg); +} + +static void toggle_clock(struct davinci_mcbsp_dev *dev, int playback) +{ + u32 m = playback ? DAVINCI_MCBSP_PCR_CLKXP : DAVINCI_MCBSP_PCR_CLKRP; + /* The clock needs to toggle to complete reset. + * So, fake it by toggling the clk polarity. + */ + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, dev->pcr ^ m); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, dev->pcr); +} + +static void davinci_mcbsp_start(struct davinci_mcbsp_dev *dev, + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + u32 spcr; + u32 mask = playback ? DAVINCI_MCBSP_SPCR_XRST : DAVINCI_MCBSP_SPCR_RRST; + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + if (spcr & mask) { + /* start off disabled */ + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, + spcr & ~mask); + toggle_clock(dev, playback); + } + if (dev->pcr & (DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM | + DAVINCI_MCBSP_PCR_CLKXM | DAVINCI_MCBSP_PCR_CLKRM)) { + /* Start the sample generator */ + spcr |= DAVINCI_MCBSP_SPCR_GRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + } + + if (playback) { + /* Stop the DMA to avoid data loss */ + /* while the transmitter is out of reset to handle XSYNCERR */ + if (platform->driver->ops->trigger) { + int ret = platform->driver->ops->trigger(substream, + SNDRV_PCM_TRIGGER_STOP); + if (ret < 0) + printk(KERN_DEBUG "Playback DMA stop failed\n"); + } + + /* Enable the transmitter */ + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + spcr |= DAVINCI_MCBSP_SPCR_XRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + + /* wait for any unexpected frame sync error to occur */ + udelay(100); + + /* Disable the transmitter to clear any outstanding XSYNCERR */ + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + spcr &= ~DAVINCI_MCBSP_SPCR_XRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + toggle_clock(dev, playback); + + /* Restart the DMA */ + if (platform->driver->ops->trigger) { + int ret = platform->driver->ops->trigger(substream, + SNDRV_PCM_TRIGGER_START); + if (ret < 0) + printk(KERN_DEBUG "Playback DMA start failed\n"); + } + } + + /* Enable transmitter or receiver */ + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + spcr |= mask; + + if (dev->pcr & (DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM)) { + /* Start frame sync */ + spcr |= DAVINCI_MCBSP_SPCR_FRST; + } + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); +} + +static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback) +{ + u32 spcr; + + /* Reset transmitter/receiver and sample rate/frame sync generators */ + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + spcr &= ~(DAVINCI_MCBSP_SPCR_GRST | DAVINCI_MCBSP_SPCR_FRST); + spcr &= playback ? ~DAVINCI_MCBSP_SPCR_XRST : ~DAVINCI_MCBSP_SPCR_RRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + toggle_clock(dev, playback); +} + +#define DEFAULT_BITPERSAMPLE 16 + +static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + unsigned int pcr; + unsigned int srgr; + bool inv_fs = false; + /* Attention srgr is updated by hw_params! */ + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | + DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + + dev->fmt = fmt; + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* cpu is master */ + pcr = DAVINCI_MCBSP_PCR_FSXM | + DAVINCI_MCBSP_PCR_FSRM | + DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; + break; + case SND_SOC_DAIFMT_CBM_CFS: + pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM; + /* + * Selection of the clock input pin that is the + * input for the Sample Rate Generator. + * McBSP FSR and FSX are driven by the Sample Rate + * Generator. + */ + switch (dev->clk_input_pin) { + case MCBSP_CLKS: + pcr |= DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; + break; + case MCBSP_CLKR: + pcr |= DAVINCI_MCBSP_PCR_SCLKME; + break; + default: + dev_err(dev->dev, "bad clk_input_pin\n"); + return -EINVAL; + } + + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* codec is master */ + pcr = 0; + break; + default: + printk(KERN_ERR "%s:bad master\n", __func__); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* Davinci doesn't support TRUE I2S, but some codecs will have + * the left and right channels contiguous. This allows + * dsp_a mode to be used with an inverted normal frame clk. + * If your codec is master and does not have contiguous + * channels, then you will have sound on only one channel. + * Try using a different mode, or codec as slave. + * + * The TLV320AIC33 is an example of a codec where this works. + * It has a variable bit clock frequency allowing it to have + * valid data on every bit clock. + * + * The TLV320AIC23 is an example of a codec where this does not + * work. It has a fixed bit clock frequency with progressively + * more empty bit clock slots between channels as the sample + * rate is lowered. + */ + inv_fs = true; + case SND_SOC_DAIFMT_DSP_A: + dev->mode = MOD_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + dev->mode = MOD_DSP_B; + break; + default: + printk(KERN_ERR "%s:bad format\n", __func__); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* CLKRP Receive clock polarity, + * 1 - sampled on rising edge of CLKR + * valid on rising edge + * CLKXP Transmit clock polarity, + * 1 - clocked on falling edge of CLKX + * valid on rising edge + * FSRP Receive frame sync pol, 0 - active high + * FSXP Transmit frame sync pol, 0 - active high + */ + pcr |= (DAVINCI_MCBSP_PCR_CLKXP | DAVINCI_MCBSP_PCR_CLKRP); + break; + case SND_SOC_DAIFMT_IB_IF: + /* CLKRP Receive clock polarity, + * 0 - sampled on falling edge of CLKR + * valid on falling edge + * CLKXP Transmit clock polarity, + * 0 - clocked on rising edge of CLKX + * valid on falling edge + * FSRP Receive frame sync pol, 1 - active low + * FSXP Transmit frame sync pol, 1 - active low + */ + pcr |= (DAVINCI_MCBSP_PCR_FSXP | DAVINCI_MCBSP_PCR_FSRP); + break; + case SND_SOC_DAIFMT_NB_IF: + /* CLKRP Receive clock polarity, + * 1 - sampled on rising edge of CLKR + * valid on rising edge + * CLKXP Transmit clock polarity, + * 1 - clocked on falling edge of CLKX + * valid on rising edge + * FSRP Receive frame sync pol, 1 - active low + * FSXP Transmit frame sync pol, 1 - active low + */ + pcr |= (DAVINCI_MCBSP_PCR_CLKXP | DAVINCI_MCBSP_PCR_CLKRP | + DAVINCI_MCBSP_PCR_FSXP | DAVINCI_MCBSP_PCR_FSRP); + break; + case SND_SOC_DAIFMT_IB_NF: + /* CLKRP Receive clock polarity, + * 0 - sampled on falling edge of CLKR + * valid on falling edge + * CLKXP Transmit clock polarity, + * 0 - clocked on rising edge of CLKX + * valid on falling edge + * FSRP Receive frame sync pol, 0 - active high + * FSXP Transmit frame sync pol, 0 - active high + */ + break; + default: + return -EINVAL; + } + if (inv_fs == true) + pcr ^= (DAVINCI_MCBSP_PCR_FSXP | DAVINCI_MCBSP_PCR_FSRP); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); + dev->pcr = pcr; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, pcr); + return 0; +} + +static int davinci_i2s_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + + if (div_id != DAVINCI_MCBSP_CLKGDV) + return -ENODEV; + + dev->clk_div = div; + return 0; +} + +static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); + struct davinci_pcm_dma_params *dma_params = + &dev->dma_params[substream->stream]; + struct snd_interval *i = NULL; + int mcbsp_word_length, master; + unsigned int rcr, xcr, srgr, clk_div, freq, framesize; + u32 spcr; + snd_pcm_format_t fmt; + unsigned element_cnt = 1; + + /* general line settings */ + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + spcr |= DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_FREE; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + } else { + spcr |= DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + } + + master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + fmt = params_format(params); + mcbsp_word_length = asp_word_length[fmt]; + + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + freq = clk_get_rate(dev->clk); + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + if (dev->i2s_accurate_sck) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + } + clk_div &= 0xFF; + srgr |= clk_div; + break; + case SND_SOC_DAIFMT_CBM_CFS: + srgr = DAVINCI_MCBSP_SRGR_FSGM; + clk_div = dev->clk_div - 1; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* Clock and frame sync given from external sources */ + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + srgr = DAVINCI_MCBSP_SRGR_FSGM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + pr_debug("%s - %d FWID set: re-read srgr = %X\n", + __func__, __LINE__, snd_interval_value(i) - 1); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + break; + default: + return -EINVAL; + } + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); + + rcr = DAVINCI_MCBSP_RCR_RFIG; + xcr = DAVINCI_MCBSP_XCR_XFIG; + if (dev->mode == MOD_DSP_B) { + rcr |= DAVINCI_MCBSP_RCR_RDATDLY(0); + xcr |= DAVINCI_MCBSP_XCR_XDATDLY(0); + } else { + rcr |= DAVINCI_MCBSP_RCR_RDATDLY(1); + xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1); + } + /* Determine xfer data type */ + fmt = params_format(params); + if ((fmt > SNDRV_PCM_FORMAT_S32_LE) || !data_type[fmt]) { + printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n"); + return -EINVAL; + } + + if (params_channels(params) == 2) { + element_cnt = 2; + if (double_fmt[fmt] && dev->enable_channel_combine) { + element_cnt = 1; + fmt = double_fmt[fmt]; + } + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0); + rcr |= DAVINCI_MCBSP_RCR_RPHASE; + xcr |= DAVINCI_MCBSP_XCR_XPHASE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1); + break; + default: + return -EINVAL; + } + } + dma_params->acnt = dma_params->data_type = data_type[fmt]; + dma_params->fifo_level = 0; + mcbsp_word_length = asp_word_length[fmt]; + + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0); + break; + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + break; + default: + return -EINVAL; + } + + rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); + xcr |= DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); + else + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); + + pr_debug("%s - %d srgr=%X\n", __func__, __LINE__, srgr); + pr_debug("%s - %d xcr=%X\n", __func__, __LINE__, xcr); + pr_debug("%s - %d rcr=%X\n", __func__, __LINE__, rcr); + return 0; +} + +static int davinci_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); + int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + davinci_mcbsp_stop(dev, playback); + return 0; +} + +static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); + int ret = 0; + int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_mcbsp_start(dev, substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcbsp_stop(dev, playback); + break; + default: + ret = -EINVAL; + } + return ret; +} + +static int davinci_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); + return 0; +} + +static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); + int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + davinci_mcbsp_stop(dev, playback); +} + +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +static const struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .shutdown = davinci_i2s_shutdown, + .prepare = davinci_i2s_prepare, + .trigger = davinci_i2s_trigger, + .hw_params = davinci_i2s_hw_params, + .set_fmt = davinci_i2s_set_dai_fmt, + .set_clkdiv = davinci_i2s_dai_set_clkdiv, + +}; + +static struct snd_soc_dai_driver davinci_i2s_dai = { + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_i2s_dai_ops, + +}; + +static int davinci_i2s_probe(struct platform_device *pdev) +{ + struct snd_platform_data *pdata = pdev->dev.platform_data; + struct davinci_mcbsp_dev *dev; + struct resource *mem, *ioarea, *res; + enum dma_event_q asp_chan_q = EVENTQ_0; + enum dma_event_q ram_chan_q = EVENTQ_1; + int ret; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + return -ENODEV; + } + + ioarea = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), + pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "McBSP region already claimed\n"); + return -EBUSY; + } + + dev = devm_kzalloc(&pdev->dev, sizeof(struct davinci_mcbsp_dev), + GFP_KERNEL); + if (!dev) + return -ENOMEM; + if (pdata) { + dev->enable_channel_combine = pdata->enable_channel_combine; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].sram_size = + pdata->sram_size_playback; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = + pdata->sram_size_capture; + dev->clk_input_pin = pdata->clk_input_pin; + dev->i2s_accurate_sck = pdata->i2s_accurate_sck; + asp_chan_q = pdata->asp_chan_q; + ram_chan_q = pdata->ram_chan_q; + } + + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].asp_chan_q = asp_chan_q; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].ram_chan_q = ram_chan_q; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].asp_chan_q = asp_chan_q; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].ram_chan_q = ram_chan_q; + + dev->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) + return -ENODEV; + clk_enable(dev->clk); + + dev->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); + if (!dev->base) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_release_clk; + } + + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = + (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); + + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = + (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG); + + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + ret = -ENXIO; + goto err_release_clk; + } + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + ret = -ENXIO; + goto err_release_clk; + } + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; + dev->dev = &pdev->dev; + + dev_set_drvdata(&pdev->dev, dev); + + ret = snd_soc_register_dai(&pdev->dev, &davinci_i2s_dai); + if (ret != 0) + goto err_release_clk; + + return 0; + +err_release_clk: + clk_disable(dev->clk); + clk_put(dev->clk); + return ret; +} + +static int davinci_i2s_remove(struct platform_device *pdev) +{ + struct davinci_mcbsp_dev *dev = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_dai(&pdev->dev); + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + return 0; +} + +static struct platform_driver davinci_mcbsp_driver = { + .probe = davinci_i2s_probe, + .remove = davinci_i2s_remove, + .driver = { + .name = "davinci-mcbsp", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(davinci_mcbsp_driver); + +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_DESCRIPTION("TI DAVINCI I2S (McBSP) SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/ANDROID_3.4.5/sound/soc/davinci/davinci-i2s.h b/ANDROID_3.4.5/sound/soc/davinci/davinci-i2s.h new file mode 100644 index 00000000..48dac3e2 --- /dev/null +++ b/ANDROID_3.4.5/sound/soc/davinci/davinci-i2s.h @@ -0,0 +1,20 @@ +/* + * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor + * + * Author: Vladimir Barinov, <vbarinov@embeddedalley.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.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. + */ + +#ifndef _DAVINCI_I2S_H +#define _DAVINCI_I2S_H + +/* McBSP dividers */ +enum davinci_mcbsp_div { + DAVINCI_MCBSP_CLKGDV, /* Sample rate generator divider */ +}; + +#endif diff --git a/ANDROID_3.4.5/sound/soc/davinci/davinci-mcasp.c b/ANDROID_3.4.5/sound/soc/davinci/davinci-mcasp.c new file mode 100644 index 00000000..95441bfc --- /dev/null +++ b/ANDROID_3.4.5/sound/soc/davinci/davinci-mcasp.c @@ -0,0 +1,983 @@ +/* + * ALSA SoC McASP Audio Layer for TI DAVINCI processor + * + * Multi-channel Audio Serial Port Driver + * + * Author: Nirmal Pandey <n-pandey@ti.com>, + * Suresh Rajashekara <suresh.r@ti.com> + * Steve Chen <schen@.mvista.com> + * + * Copyright: (C) 2009 MontaVista Software, Inc., <source@mvista.com> + * Copyright: (C) 2009 Texas Instruments, India + * + * 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/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-mcasp.h" + +/* + * McASP register definitions + */ +#define DAVINCI_MCASP_PID_REG 0x00 +#define DAVINCI_MCASP_PWREMUMGT_REG 0x04 + +#define DAVINCI_MCASP_PFUNC_REG 0x10 +#define DAVINCI_MCASP_PDIR_REG 0x14 +#define DAVINCI_MCASP_PDOUT_REG 0x18 +#define DAVINCI_MCASP_PDSET_REG 0x1c + +#define DAVINCI_MCASP_PDCLR_REG 0x20 + +#define DAVINCI_MCASP_TLGC_REG 0x30 +#define DAVINCI_MCASP_TLMR_REG 0x34 + +#define DAVINCI_MCASP_GBLCTL_REG 0x44 +#define DAVINCI_MCASP_AMUTE_REG 0x48 +#define DAVINCI_MCASP_LBCTL_REG 0x4c + +#define DAVINCI_MCASP_TXDITCTL_REG 0x50 + +#define DAVINCI_MCASP_GBLCTLR_REG 0x60 +#define DAVINCI_MCASP_RXMASK_REG 0x64 +#define DAVINCI_MCASP_RXFMT_REG 0x68 +#define DAVINCI_MCASP_RXFMCTL_REG 0x6c + +#define DAVINCI_MCASP_ACLKRCTL_REG 0x70 +#define DAVINCI_MCASP_AHCLKRCTL_REG 0x74 +#define DAVINCI_MCASP_RXTDM_REG 0x78 +#define DAVINCI_MCASP_EVTCTLR_REG 0x7c + +#define DAVINCI_MCASP_RXSTAT_REG 0x80 +#define DAVINCI_MCASP_RXTDMSLOT_REG 0x84 +#define DAVINCI_MCASP_RXCLKCHK_REG 0x88 +#define DAVINCI_MCASP_REVTCTL_REG 0x8c + +#define DAVINCI_MCASP_GBLCTLX_REG 0xa0 +#define DAVINCI_MCASP_TXMASK_REG 0xa4 +#define DAVINCI_MCASP_TXFMT_REG 0xa8 +#define DAVINCI_MCASP_TXFMCTL_REG 0xac + +#define DAVINCI_MCASP_ACLKXCTL_REG 0xb0 +#define DAVINCI_MCASP_AHCLKXCTL_REG 0xb4 +#define DAVINCI_MCASP_TXTDM_REG 0xb8 +#define DAVINCI_MCASP_EVTCTLX_REG 0xbc + +#define DAVINCI_MCASP_TXSTAT_REG 0xc0 +#define DAVINCI_MCASP_TXTDMSLOT_REG 0xc4 +#define DAVINCI_MCASP_TXCLKCHK_REG 0xc8 +#define DAVINCI_MCASP_XEVTCTL_REG 0xcc + +/* Left(even TDM Slot) Channel Status Register File */ +#define DAVINCI_MCASP_DITCSRA_REG 0x100 +/* Right(odd TDM slot) Channel Status Register File */ +#define DAVINCI_MCASP_DITCSRB_REG 0x118 +/* Left(even TDM slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRA_REG 0x130 +/* Right(odd TDM Slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRB_REG 0x148 + +/* Serializer n Control Register */ +#define DAVINCI_MCASP_XRSRCTL_BASE_REG 0x180 +#define DAVINCI_MCASP_XRSRCTL_REG(n) (DAVINCI_MCASP_XRSRCTL_BASE_REG + \ + (n << 2)) + +/* Transmit Buffer for Serializer n */ +#define DAVINCI_MCASP_TXBUF_REG 0x200 +/* Receive Buffer for Serializer n */ +#define DAVINCI_MCASP_RXBUF_REG 0x280 + +/* McASP FIFO Registers */ +#define DAVINCI_MCASP_WFIFOCTL (0x1010) +#define DAVINCI_MCASP_WFIFOSTS (0x1014) +#define DAVINCI_MCASP_RFIFOCTL (0x1018) +#define DAVINCI_MCASP_RFIFOSTS (0x101C) + +/* + * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management + * Register Bits + */ +#define MCASP_FREE BIT(0) +#define MCASP_SOFT BIT(1) + +/* + * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits + */ +#define AXR(n) (1<<n) +#define PFUNC_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits + */ +#define AXR(n) (1<<n) +#define PDIR_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits + */ +#define DITEN BIT(0) /* Transmit DIT mode enable/disable */ +#define VA BIT(2) +#define VB BIT(3) + +/* + * DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits + */ +#define TXROT(val) (val) +#define TXSEL BIT(3) +#define TXSSZ(val) (val<<4) +#define TXPBIT(val) (val<<8) +#define TXPAD(val) (val<<13) +#define TXORD BIT(15) +#define FSXDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits + */ +#define RXROT(val) (val) +#define RXSEL BIT(3) +#define RXSSZ(val) (val<<4) +#define RXPBIT(val) (val<<8) +#define RXPAD(val) (val<<13) +#define RXORD BIT(15) +#define FSRDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_TXFMCTL_REG - Transmit Frame Control Register Bits + */ +#define FSXPOL BIT(0) +#define AFSXE BIT(1) +#define FSXDUR BIT(4) +#define FSXMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits + */ +#define FSRPOL BIT(0) +#define AFSRE BIT(1) +#define FSRDUR BIT(4) +#define FSRMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits + */ +#define ACLKXDIV(val) (val) +#define ACLKXE BIT(5) +#define TX_ASYNC BIT(6) +#define ACLKXPOL BIT(7) + +/* + * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits + */ +#define ACLKRDIV(val) (val) +#define ACLKRE BIT(5) +#define RX_ASYNC BIT(6) +#define ACLKRPOL BIT(7) + +/* + * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control + * Register Bits + */ +#define AHCLKXDIV(val) (val) +#define AHCLKXPOL BIT(14) +#define AHCLKXE BIT(15) + +/* + * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control + * Register Bits + */ +#define AHCLKRDIV(val) (val) +#define AHCLKRPOL BIT(14) +#define AHCLKRE BIT(15) + +/* + * DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits + */ +#define MODE(val) (val) +#define DISMOD (val)(val<<2) +#define TXSTATE BIT(4) +#define RXSTATE BIT(5) + +/* + * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits + */ +#define LBEN BIT(0) +#define LBORD BIT(1) +#define LBGENMODE(val) (val<<2) + +/* + * DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration + */ +#define TXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration + */ +#define RXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_GBLCTL_REG - Global Control Register Bits + */ +#define RXCLKRST BIT(0) /* Receiver Clock Divider Reset */ +#define RXHCLKRST BIT(1) /* Receiver High Frequency Clock Divider */ +#define RXSERCLR BIT(2) /* Receiver Serializer Clear */ +#define RXSMRST BIT(3) /* Receiver State Machine Reset */ +#define RXFSRST BIT(4) /* Frame Sync Generator Reset */ +#define TXCLKRST BIT(8) /* Transmitter Clock Divider Reset */ +#define TXHCLKRST BIT(9) /* Transmitter High Frequency Clock Divider*/ +#define TXSERCLR BIT(10) /* Transmit Serializer Clear */ +#define TXSMRST BIT(11) /* Transmitter State Machine Reset */ +#define TXFSRST BIT(12) /* Frame Sync Generator Reset */ + +/* + * DAVINCI_MCASP_AMUTE_REG - Mute Control Register Bits + */ +#define MUTENA(val) (val) +#define MUTEINPOL BIT(2) +#define MUTEINENA BIT(3) +#define MUTEIN BIT(4) +#define MUTER BIT(5) +#define MUTEX BIT(6) +#define MUTEFSR BIT(7) +#define MUTEFSX BIT(8) +#define MUTEBADCLKR BIT(9) +#define MUTEBADCLKX BIT(10) +#define MUTERXDMAERR BIT(11) +#define MUTETXDMAERR BIT(12) + +/* + * DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits + */ +#define RXDATADMADIS BIT(0) + +/* + * DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits + */ +#define TXDATADMADIS BIT(0) + +/* + * DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits + */ +#define FIFO_ENABLE BIT(16) +#define NUMEVT_MASK (0xFF << 8) +#define NUMDMA_MASK (0xFF) + +#define DAVINCI_MCASP_NUM_SERIALIZER 16 + +static inline void mcasp_set_bits(void __iomem *reg, u32 val) +{ + __raw_writel(__raw_readl(reg) | val, reg); +} + +static inline void mcasp_clr_bits(void __iomem *reg, u32 val) +{ + __raw_writel((__raw_readl(reg) & ~(val)), reg); +} + +static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask) +{ + __raw_writel((__raw_readl(reg) & ~mask) | val, reg); +} + +static inline void mcasp_set_reg(void __iomem *reg, u32 val) +{ + __raw_writel(val, reg); +} + +static inline u32 mcasp_get_reg(void __iomem *reg) +{ + return (unsigned int)__raw_readl(reg); +} + +static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +{ + int i = 0; + + mcasp_set_bits(regs, val); + + /* programming GBLCTL needs to read back from GBLCTL and verfiy */ + /* loop count is to avoid the lock-up */ + for (i = 0; i < 1000; i++) { + if ((mcasp_get_reg(regs) & val) == val) + break; + } + + if (i == 1000 && ((mcasp_get_reg(regs) & val) != val)) + printk(KERN_ERR "GBLCTL write error\n"); +} + +static void mcasp_start_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); +} + +static void mcasp_start_tx(struct davinci_audio_dev *dev) +{ + u8 offset = 0, i; + u32 cnt; + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) { + offset = i; + break; + } + } + + /* wait for TX ready */ + cnt = 0; + while (!(mcasp_get_reg(dev->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) & + TXSTATE) && (cnt < 100000)) + cnt++; + + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); +} + +static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt) /* enable FIFO */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); + mcasp_start_tx(dev); + } else { + if (dev->rxnumevt) /* enable FIFO */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); + mcasp_start_rx(dev); + } +} + +static void mcasp_stop_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); +} + +static void mcasp_stop_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); +} + +static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt) /* disable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); + mcasp_stop_tx(dev); + } else { + if (dev->rxnumevt) /* disable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); + mcasp_stop_rx(dev); + } +} + +static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + void __iomem *base = dev->base; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* codec is clock and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, + ACLKX | AHCLKX | AFSX); + break; + case SND_SOC_DAIFMT_CBM_CFS: + /* codec is clock master and frame slave */ + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, + ACLKX | ACLKR); + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, + AFSX | AFSR); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* codec is clock and frame master */ + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, + ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR); + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_NB_IF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_IB_IF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_NB_NF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int davinci_config_channel_size(struct davinci_audio_dev *dev, + int channel_size) +{ + u32 fmt = 0; + u32 mask, rotate; + + switch (channel_size) { + case DAVINCI_AUDIO_WORD_8: + fmt = 0x03; + rotate = 6; + mask = 0x000000ff; + break; + + case DAVINCI_AUDIO_WORD_12: + fmt = 0x05; + rotate = 5; + mask = 0x00000fff; + break; + + case DAVINCI_AUDIO_WORD_16: + fmt = 0x07; + rotate = 4; + mask = 0x0000ffff; + break; + + case DAVINCI_AUDIO_WORD_20: + fmt = 0x09; + rotate = 3; + mask = 0x000fffff; + break; + + case DAVINCI_AUDIO_WORD_24: + fmt = 0x0B; + rotate = 2; + mask = 0x00ffffff; + break; + + case DAVINCI_AUDIO_WORD_28: + fmt = 0x0D; + rotate = 1; + mask = 0x0fffffff; + break; + + case DAVINCI_AUDIO_WORD_32: + fmt = 0x0F; + rotate = 0; + mask = 0xffffffff; + break; + + default: + return -EINVAL; + } + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), + TXROT(7)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), + RXROT(7)); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); + + return 0; +} + +static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +{ + int i; + u8 tx_ser = 0; + u8 rx_ser = 0; + + /* Default configuration */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); + + /* All PINS as McASP */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, + TXDATADMADIS); + } else { + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG, + RXDATADMADIS); + } + + for (i = 0; i < dev->num_serializer; i++) { + mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + dev->serial_dir[i]); + if (dev->serial_dir[i] == TX_MODE) { + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + tx_ser++; + } else if (dev->serial_dir[i] == RX_MODE) { + mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + rx_ser++; + } + } + + if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt * tx_ser > 64) + dev->txnumevt = 1; + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, tx_ser, + NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + } + + if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { + if (dev->rxnumevt * rx_ser > 64) + dev->rxnumevt = 1; + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, rx_ser, + NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + } +} + +static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) +{ + int i, active_slots; + u32 mask = 0; + + active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots; + for (i = 0; i < active_slots; i++) + mask |= (1 << i); + + mcasp_clr_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* bit stream is MSB first with no delay */ + /* DSP_B mode */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD); + + if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(dev->tdm_slots), FSXMOD(0x1FF)); + else + printk(KERN_ERR "playback tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + } else { + /* bit stream is MSB first with no delay */ + /* DSP_B mode */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXORD); + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, + AHCLKRE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask); + + if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(dev->tdm_slots), FSRMOD(0x1FF)); + else + printk(KERN_ERR "capture tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + } +} + +/* S/PDIF */ +static void davinci_hw_dit_param(struct davinci_audio_dev *dev) +{ + /* Set the PDIR for Serialiser as output */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); + + /* TXMASK for 24 bits */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); + + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 + and LSB first */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(6) | TXSSZ(15)); + + /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + AFSXE | FSXMOD(0x180)); + + /* Set the TX tdm : for all the slots */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); + + /* Set the TX clock controls : div = 1 and internal */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE | TX_ASYNC); + + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + + /* Only 44100 and 48000 are valid, both have the same setting */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); + + /* Enable the DIT */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); +} + +static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + struct davinci_pcm_dma_params *dma_params = + &dev->dma_params[substream->stream]; + int word_length; + u8 fifo_level; + + davinci_hw_common_param(dev, substream->stream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + fifo_level = dev->txnumevt; + else + fifo_level = dev->rxnumevt; + + if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) + davinci_hw_dit_param(dev); + else + davinci_hw_param(dev, substream->stream); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + word_length = DAVINCI_AUDIO_WORD_8; + break; + + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + word_length = DAVINCI_AUDIO_WORD_16; + break; + + case SNDRV_PCM_FORMAT_U32_LE: + case SNDRV_PCM_FORMAT_S32_LE: + dma_params->data_type = 4; + word_length = DAVINCI_AUDIO_WORD_32; + break; + + default: + printk(KERN_WARNING "davinci-mcasp: unsupported PCM format"); + return -EINVAL; + } + + if (dev->version == MCASP_VERSION_2 && !fifo_level) + dma_params->acnt = 4; + else + dma_params->acnt = dma_params->data_type; + + dma_params->fifo_level = fifo_level; + davinci_config_channel_size(dev, word_length); + + return 0; +} + +static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) +{ + struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!dev->clk_active) { + clk_enable(dev->clk); + dev->clk_active = 1; + } + davinci_mcasp_start(dev, substream->stream); + break; + + case SNDRV_PCM_TRIGGER_SUSPEND: + davinci_mcasp_stop(dev, substream->stream); + if (dev->clk_active) { + clk_disable(dev->clk); + dev->clk_active = 0; + } + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcasp_stop(dev, substream->stream); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static int davinci_mcasp_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); + return 0; +} + +static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { + .startup = davinci_mcasp_startup, + .trigger = davinci_mcasp_trigger, + .hw_params = davinci_mcasp_hw_params, + .set_fmt = davinci_mcasp_set_dai_fmt, + +}; + +#define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_soc_dai_driver davinci_mcasp_dai[] = { + { + .name = "davinci-mcasp.0", + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_MCASP_RATES, + .formats = DAVINCI_MCASP_PCM_FMTS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_MCASP_RATES, + .formats = DAVINCI_MCASP_PCM_FMTS, + }, + .ops = &davinci_mcasp_dai_ops, + + }, + { + "davinci-mcasp.1", + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_MCASP_RATES, + .formats = DAVINCI_MCASP_PCM_FMTS, + }, + .ops = &davinci_mcasp_dai_ops, + }, + +}; + +static int davinci_mcasp_probe(struct platform_device *pdev) +{ + struct davinci_pcm_dma_params *dma_data; + struct resource *mem, *ioarea, *res; + struct snd_platform_data *pdata; + struct davinci_audio_dev *dev; + int ret; + + dev = devm_kzalloc(&pdev->dev, sizeof(struct davinci_audio_dev), + GFP_KERNEL); + if (!dev) + return -ENOMEM; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + return -ENODEV; + } + + ioarea = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "Audio region already claimed\n"); + return -EBUSY; + } + + pdata = pdev->dev.platform_data; + dev->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) + return -ENODEV; + + clk_enable(dev->clk); + dev->clk_active = 1; + + dev->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); + if (!dev->base) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_release_clk; + } + + dev->op_mode = pdata->op_mode; + dev->tdm_slots = pdata->tdm_slots; + dev->num_serializer = pdata->num_serializer; + dev->serial_dir = pdata->serial_dir; + dev->codec_fmt = pdata->codec_fmt; + dev->version = pdata->version; + dev->txnumevt = pdata->txnumevt; + dev->rxnumevt = pdata->rxnumevt; + + dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + dma_data->asp_chan_q = pdata->asp_chan_q; + dma_data->ram_chan_q = pdata->ram_chan_q; + dma_data->sram_size = pdata->sram_size_playback; + dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset + + mem->start); + + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + ret = -ENODEV; + goto err_release_clk; + } + + dma_data->channel = res->start; + + dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]; + dma_data->asp_chan_q = pdata->asp_chan_q; + dma_data->ram_chan_q = pdata->ram_chan_q; + dma_data->sram_size = pdata->sram_size_capture; + dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset + + mem->start); + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + ret = -ENODEV; + goto err_release_clk; + } + + dma_data->channel = res->start; + dev_set_drvdata(&pdev->dev, dev); + ret = snd_soc_register_dai(&pdev->dev, &davinci_mcasp_dai[pdata->op_mode]); + + if (ret != 0) + goto err_release_clk; + return 0; + +err_release_clk: + clk_disable(dev->clk); + clk_put(dev->clk); + return ret; +} + +static int davinci_mcasp_remove(struct platform_device *pdev) +{ + struct davinci_audio_dev *dev = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_dai(&pdev->dev); + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + return 0; +} + +static struct platform_driver davinci_mcasp_driver = { + .probe = davinci_mcasp_probe, + .remove = davinci_mcasp_remove, + .driver = { + .name = "davinci-mcasp", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(davinci_mcasp_driver); + +MODULE_AUTHOR("Steve Chen"); +MODULE_DESCRIPTION("TI DAVINCI McASP SoC Interface"); +MODULE_LICENSE("GPL"); + diff --git a/ANDROID_3.4.5/sound/soc/davinci/davinci-mcasp.h b/ANDROID_3.4.5/sound/soc/davinci/davinci-mcasp.h new file mode 100644 index 00000000..4681acc6 --- /dev/null +++ b/ANDROID_3.4.5/sound/soc/davinci/davinci-mcasp.h @@ -0,0 +1,59 @@ +/* + * ALSA SoC McASP Audio Layer for TI DAVINCI processor + * + * MCASP related definitions + * + * Author: Nirmal Pandey <n-pandey@ti.com>, + * Suresh Rajashekara <suresh.r@ti.com> + * Steve Chen <schen@.mvista.com> + * + * Copyright: (C) 2009 MontaVista Software, Inc., <source@mvista.com> + * Copyright: (C) 2009 Texas Instruments, India + * + * 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 DAVINCI_MCASP_H +#define DAVINCI_MCASP_H + +#include <linux/io.h> +#include <mach/asp.h> +#include "davinci-pcm.h" + +#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_96000 +#define DAVINCI_MCASP_I2S_DAI 0 +#define DAVINCI_MCASP_DIT_DAI 1 + +enum { + DAVINCI_AUDIO_WORD_8 = 0, + DAVINCI_AUDIO_WORD_12, + DAVINCI_AUDIO_WORD_16, + DAVINCI_AUDIO_WORD_20, + DAVINCI_AUDIO_WORD_24, + DAVINCI_AUDIO_WORD_32, + DAVINCI_AUDIO_WORD_28, /* This is only valid for McASP */ +}; + +struct davinci_audio_dev { + struct davinci_pcm_dma_params dma_params[2]; + void __iomem *base; + int sample_rate; + struct clk *clk; + unsigned int codec_fmt; + u8 clk_active; + + /* McASP specific data */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; + u8 version; + + /* McASP FIFO related */ + u8 txnumevt; + u8 rxnumevt; +}; + +#endif /* DAVINCI_MCASP_H */ diff --git a/ANDROID_3.4.5/sound/soc/davinci/davinci-pcm.c b/ANDROID_3.4.5/sound/soc/davinci/davinci-pcm.c new file mode 100644 index 00000000..97d77b29 --- /dev/null +++ b/ANDROID_3.4.5/sound/soc/davinci/davinci-pcm.c @@ -0,0 +1,892 @@ +/* + * ALSA PCM interface for the TI DAVINCI processor + * + * Author: Vladimir Barinov, <vbarinov@embeddedalley.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com> + * added SRAM ping/pong (C) 2008 Troy Kisky <troy.kisky@boundarydevices.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/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/kernel.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/dma.h> +#include <mach/edma.h> +#include <mach/sram.h> + +#include "davinci-pcm.h" + +#ifdef DEBUG +static void print_buf_info(int slot, char *name) +{ + struct edmacc_param p; + if (slot < 0) + return; + edma_read_slot(slot, &p); + printk(KERN_DEBUG "%s: 0x%x, opt=%x, src=%x, a_b_cnt=%x dst=%x\n", + name, slot, p.opt, p.src, p.a_b_cnt, p.dst); + printk(KERN_DEBUG " src_dst_bidx=%x link_bcntrld=%x src_dst_cidx=%x ccnt=%x\n", + p.src_dst_bidx, p.link_bcntrld, p.src_dst_cidx, p.ccnt); +} +#else +static void print_buf_info(int slot, char *name) +{ +} +#endif + +#define DAVINCI_PCM_FMTBITS (\ + SNDRV_PCM_FMTBIT_S8 |\ + SNDRV_PCM_FMTBIT_U8 |\ + SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S16_BE |\ + SNDRV_PCM_FMTBIT_U16_LE |\ + SNDRV_PCM_FMTBIT_U16_BE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S24_BE |\ + SNDRV_PCM_FMTBIT_U24_LE |\ + SNDRV_PCM_FMTBIT_U24_BE |\ + SNDRV_PCM_FMTBIT_S32_LE |\ + SNDRV_PCM_FMTBIT_S32_BE |\ + SNDRV_PCM_FMTBIT_U32_LE |\ + SNDRV_PCM_FMTBIT_U32_BE) + +static struct snd_pcm_hardware pcm_hardware_playback = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME| + SNDRV_PCM_INFO_BATCH), + .formats = DAVINCI_PCM_FMTBITS, + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_KNOT), + .rate_min = 8000, + .rate_max = 96000, + .channels_min = 2, + .channels_max = 384, + .buffer_bytes_max = 128 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8 * 1024, + .periods_min = 16, + .periods_max = 255, + .fifo_size = 0, +}; + +static struct snd_pcm_hardware pcm_hardware_capture = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_BATCH), + .formats = DAVINCI_PCM_FMTBITS, + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_KNOT), + .rate_min = 8000, + .rate_max = 96000, + .channels_min = 2, + .channels_max = 384, + .buffer_bytes_max = 128 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8 * 1024, + .periods_min = 16, + .periods_max = 255, + .fifo_size = 0, +}; + +/* + * How ping/pong works.... + * + * Playback: + * ram_params - copys 2*ping_size from start of SDRAM to iram, + * links to ram_link2 + * ram_link2 - copys rest of SDRAM to iram in ping_size units, + * links to ram_link + * ram_link - copys entire SDRAM to iram in ping_size uints, + * links to self + * + * asp_params - same as asp_link[0] + * asp_link[0] - copys from lower half of iram to asp port + * links to asp_link[1], triggers iram copy event on completion + * asp_link[1] - copys from upper half of iram to asp port + * links to asp_link[0], triggers iram copy event on completion + * triggers interrupt only needed to let upper SOC levels update position + * in stream on completion + * + * When playback is started: + * ram_params started + * asp_params started + * + * Capture: + * ram_params - same as ram_link, + * links to ram_link + * ram_link - same as playback + * links to self + * + * asp_params - same as playback + * asp_link[0] - same as playback + * asp_link[1] - same as playback + * + * When capture is started: + * asp_params started + */ +struct davinci_runtime_data { + spinlock_t lock; + int period; /* current DMA period */ + int asp_channel; /* Master DMA channel */ + int asp_link[2]; /* asp parameter link channel, ping/pong */ + struct davinci_pcm_dma_params *params; /* DMA params */ + int ram_channel; + int ram_link; + int ram_link2; + struct edmacc_param asp_params; + struct edmacc_param ram_params; +}; + +static void davinci_pcm_period_elapsed(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + + prtd->period++; + if (unlikely(prtd->period >= runtime->periods)) + prtd->period = 0; +} + +static void davinci_pcm_period_reset(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + + prtd->period = 0; +} +/* + * Not used with ping/pong + */ +static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int period_size; + unsigned int dma_offset; + dma_addr_t dma_pos; + dma_addr_t src, dst; + unsigned short src_bidx, dst_bidx; + unsigned short src_cidx, dst_cidx; + unsigned int data_type; + unsigned short acnt; + unsigned int count; + unsigned int fifo_level; + + period_size = snd_pcm_lib_period_bytes(substream); + dma_offset = prtd->period * period_size; + dma_pos = runtime->dma_addr + dma_offset; + fifo_level = prtd->params->fifo_level; + + pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " + "dma_ptr = %x period_size=%x\n", prtd->asp_link[0], dma_pos, + period_size); + + data_type = prtd->params->data_type; + count = period_size / data_type; + if (fifo_level) + count /= fifo_level; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src = dma_pos; + dst = prtd->params->dma_addr; + src_bidx = data_type; + dst_bidx = 0; + src_cidx = data_type * fifo_level; + dst_cidx = 0; + } else { + src = prtd->params->dma_addr; + dst = dma_pos; + src_bidx = 0; + dst_bidx = data_type; + src_cidx = 0; + dst_cidx = data_type * fifo_level; + } + + acnt = prtd->params->acnt; + edma_set_src(prtd->asp_link[0], src, INCR, W8BIT); + edma_set_dest(prtd->asp_link[0], dst, INCR, W8BIT); + + edma_set_src_index(prtd->asp_link[0], src_bidx, src_cidx); + edma_set_dest_index(prtd->asp_link[0], dst_bidx, dst_cidx); + + if (!fifo_level) + edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, + ASYNC); + else + edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, + count, fifo_level, + ABSYNC); +} + +static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) +{ + struct snd_pcm_substream *substream = data; + struct davinci_runtime_data *prtd = substream->runtime->private_data; + + print_buf_info(prtd->ram_channel, "i ram_channel"); + pr_debug("davinci_pcm: link=%d, status=0x%x\n", link, ch_status); + + if (unlikely(ch_status != DMA_COMPLETE)) + return; + + if (snd_pcm_running(substream)) { + spin_lock(&prtd->lock); + if (prtd->ram_channel < 0) { + /* No ping/pong must fix up link dma data*/ + davinci_pcm_enqueue_dma(substream); + } + davinci_pcm_period_elapsed(substream); + spin_unlock(&prtd->lock); + snd_pcm_period_elapsed(substream); + } +} + +static int allocate_sram(struct snd_pcm_substream *substream, unsigned size, + struct snd_pcm_hardware *ppcm) +{ + struct snd_dma_buffer *buf = &substream->dma_buffer; + struct snd_dma_buffer *iram_dma = NULL; + dma_addr_t iram_phys = 0; + void *iram_virt = NULL; + + if (buf->private_data || !size) + return 0; + + ppcm->period_bytes_max = size; + iram_virt = sram_alloc(size, &iram_phys); + if (!iram_virt) + goto exit1; + iram_dma = kzalloc(sizeof(*iram_dma), GFP_KERNEL); + if (!iram_dma) + goto exit2; + iram_dma->area = iram_virt; + iram_dma->addr = iram_phys; + memset(iram_dma->area, 0, size); + iram_dma->bytes = size; + buf->private_data = iram_dma; + return 0; +exit2: + if (iram_virt) + sram_free(iram_virt, size); +exit1: + return -ENOMEM; +} + +/* + * Only used with ping/pong. + * This is called after runtime->dma_addr, period_bytes and data_type are valid + */ +static int ping_pong_dma_setup(struct snd_pcm_substream *substream) +{ + unsigned short ram_src_cidx, ram_dst_cidx; + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd = runtime->private_data; + struct snd_dma_buffer *iram_dma = + (struct snd_dma_buffer *)substream->dma_buffer.private_data; + struct davinci_pcm_dma_params *params = prtd->params; + unsigned int data_type = params->data_type; + unsigned int acnt = params->acnt; + /* divide by 2 for ping/pong */ + unsigned int ping_size = snd_pcm_lib_period_bytes(substream) >> 1; + unsigned int fifo_level = prtd->params->fifo_level; + unsigned int count; + if ((data_type == 0) || (data_type > 4)) { + printk(KERN_ERR "%s: data_type=%i\n", __func__, data_type); + return -EINVAL; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dma_addr_t asp_src_pong = iram_dma->addr + ping_size; + ram_src_cidx = ping_size; + ram_dst_cidx = -ping_size; + edma_set_src(prtd->asp_link[1], asp_src_pong, INCR, W8BIT); + + edma_set_src_index(prtd->asp_link[0], data_type, + data_type * fifo_level); + edma_set_src_index(prtd->asp_link[1], data_type, + data_type * fifo_level); + + edma_set_src(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); + } else { + dma_addr_t asp_dst_pong = iram_dma->addr + ping_size; + ram_src_cidx = -ping_size; + ram_dst_cidx = ping_size; + edma_set_dest(prtd->asp_link[1], asp_dst_pong, INCR, W8BIT); + + edma_set_dest_index(prtd->asp_link[0], data_type, + data_type * fifo_level); + edma_set_dest_index(prtd->asp_link[1], data_type, + data_type * fifo_level); + + edma_set_dest(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); + } + + if (!fifo_level) { + count = ping_size / data_type; + edma_set_transfer_params(prtd->asp_link[0], acnt, count, + 1, 0, ASYNC); + edma_set_transfer_params(prtd->asp_link[1], acnt, count, + 1, 0, ASYNC); + } else { + count = ping_size / (data_type * fifo_level); + edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, + count, fifo_level, ABSYNC); + edma_set_transfer_params(prtd->asp_link[1], acnt, fifo_level, + count, fifo_level, ABSYNC); + } + + edma_set_src_index(prtd->ram_link, ping_size, ram_src_cidx); + edma_set_dest_index(prtd->ram_link, ping_size, ram_dst_cidx); + edma_set_transfer_params(prtd->ram_link, ping_size, 2, + runtime->periods, 2, ASYNC); + + /* init master params */ + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); + edma_read_slot(prtd->ram_link, &prtd->ram_params); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + struct edmacc_param p_ram; + /* Copy entire iram buffer before playback started */ + prtd->ram_params.a_b_cnt = (1 << 16) | (ping_size << 1); + /* 0 dst_bidx */ + prtd->ram_params.src_dst_bidx = (ping_size << 1); + /* 0 dst_cidx */ + prtd->ram_params.src_dst_cidx = (ping_size << 1); + prtd->ram_params.ccnt = 1; + + /* Skip 1st period */ + edma_read_slot(prtd->ram_link, &p_ram); + p_ram.src += (ping_size << 1); + p_ram.ccnt -= 1; + edma_write_slot(prtd->ram_link2, &p_ram); + /* + * When 1st started, ram -> iram dma channel will fill the + * entire iram. Then, whenever a ping/pong asp buffer finishes, + * 1/2 iram will be filled. + */ + prtd->ram_params.link_bcntrld = + EDMA_CHAN_SLOT(prtd->ram_link2) << 5; + } + return 0; +} + +/* 1 asp tx or rx channel using 2 parameter channels + * 1 ram to/from iram channel using 1 parameter channel + * + * Playback + * ram copy channel kicks off first, + * 1st ram copy of entire iram buffer completion kicks off asp channel + * asp tcc always kicks off ram copy of 1/2 iram buffer + * + * Record + * asp channel starts, tcc kicks off ram copy + */ +static int request_ping_pong(struct snd_pcm_substream *substream, + struct davinci_runtime_data *prtd, + struct snd_dma_buffer *iram_dma) +{ + dma_addr_t asp_src_ping; + dma_addr_t asp_dst_ping; + int ret; + struct davinci_pcm_dma_params *params = prtd->params; + + /* Request ram master channel */ + ret = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, + davinci_pcm_dma_irq, substream, + prtd->params->ram_chan_q); + if (ret < 0) + goto exit1; + + /* Request ram link channel */ + ret = prtd->ram_link = edma_alloc_slot( + EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); + if (ret < 0) + goto exit2; + + ret = prtd->asp_link[1] = edma_alloc_slot( + EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); + if (ret < 0) + goto exit3; + + prtd->ram_link2 = -1; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = prtd->ram_link2 = edma_alloc_slot( + EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); + if (ret < 0) + goto exit4; + } + /* circle ping-pong buffers */ + edma_link(prtd->asp_link[0], prtd->asp_link[1]); + edma_link(prtd->asp_link[1], prtd->asp_link[0]); + /* circle ram buffers */ + edma_link(prtd->ram_link, prtd->ram_link); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + asp_src_ping = iram_dma->addr; + asp_dst_ping = params->dma_addr; /* fifo */ + } else { + asp_src_ping = params->dma_addr; /* fifo */ + asp_dst_ping = iram_dma->addr; + } + /* ping */ + edma_set_src(prtd->asp_link[0], asp_src_ping, INCR, W16BIT); + edma_set_dest(prtd->asp_link[0], asp_dst_ping, INCR, W16BIT); + edma_set_src_index(prtd->asp_link[0], 0, 0); + edma_set_dest_index(prtd->asp_link[0], 0, 0); + + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); + prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN); + prtd->asp_params.opt |= TCCHEN | + EDMA_TCC(prtd->ram_channel & 0x3f); + edma_write_slot(prtd->asp_link[0], &prtd->asp_params); + + /* pong */ + edma_set_src(prtd->asp_link[1], asp_src_ping, INCR, W16BIT); + edma_set_dest(prtd->asp_link[1], asp_dst_ping, INCR, W16BIT); + edma_set_src_index(prtd->asp_link[1], 0, 0); + edma_set_dest_index(prtd->asp_link[1], 0, 0); + + edma_read_slot(prtd->asp_link[1], &prtd->asp_params); + prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f)); + /* interrupt after every pong completion */ + prtd->asp_params.opt |= TCINTEN | TCCHEN | + EDMA_TCC(prtd->ram_channel & 0x3f); + edma_write_slot(prtd->asp_link[1], &prtd->asp_params); + + /* ram */ + edma_set_src(prtd->ram_link, iram_dma->addr, INCR, W32BIT); + edma_set_dest(prtd->ram_link, iram_dma->addr, INCR, W32BIT); + pr_debug("%s: audio dma channels/slots in use for ram:%u %u %u," + "for asp:%u %u %u\n", __func__, + prtd->ram_channel, prtd->ram_link, prtd->ram_link2, + prtd->asp_channel, prtd->asp_link[0], + prtd->asp_link[1]); + return 0; +exit4: + edma_free_channel(prtd->asp_link[1]); + prtd->asp_link[1] = -1; +exit3: + edma_free_channel(prtd->ram_link); + prtd->ram_link = -1; +exit2: + edma_free_channel(prtd->ram_channel); + prtd->ram_channel = -1; +exit1: + return ret; +} + +static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) +{ + struct snd_dma_buffer *iram_dma; + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct davinci_pcm_dma_params *params = prtd->params; + int ret; + + if (!params) + return -ENODEV; + + /* Request asp master DMA channel */ + ret = prtd->asp_channel = edma_alloc_channel(params->channel, + davinci_pcm_dma_irq, substream, + prtd->params->asp_chan_q); + if (ret < 0) + goto exit1; + + /* Request asp link channels */ + ret = prtd->asp_link[0] = edma_alloc_slot( + EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); + if (ret < 0) + goto exit2; + + iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; + if (iram_dma) { + if (request_ping_pong(substream, prtd, iram_dma) == 0) + return 0; + printk(KERN_WARNING "%s: dma channel allocation failed," + "not using sram\n", __func__); + } + + /* Issue transfer completion IRQ when the channel completes a + * transfer, then always reload from the same slot (by a kind + * of loopback link). The completion IRQ handler will update + * the reload slot with a new buffer. + * + * REVISIT save p_ram here after setting up everything except + * the buffer and its length (ccnt) ... use it as a template + * so davinci_pcm_enqueue_dma() takes less time in IRQ. + */ + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); + prtd->asp_params.opt |= TCINTEN | + EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); + prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(prtd->asp_link[0]) << 5; + edma_write_slot(prtd->asp_link[0], &prtd->asp_params); + return 0; +exit2: + edma_free_channel(prtd->asp_channel); + prtd->asp_channel = -1; +exit1: + return ret; +} + +static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + edma_start(prtd->asp_channel); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + prtd->ram_channel >= 0) { + /* copy 1st iram buffer */ + edma_start(prtd->ram_channel); + } + break; + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + edma_resume(prtd->asp_channel); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + edma_pause(prtd->asp_channel); + break; + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static int davinci_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + + davinci_pcm_period_reset(substream); + if (prtd->ram_channel >= 0) { + int ret = ping_pong_dma_setup(substream); + if (ret < 0) + return ret; + + edma_write_slot(prtd->ram_channel, &prtd->ram_params); + edma_write_slot(prtd->asp_channel, &prtd->asp_params); + + print_buf_info(prtd->ram_channel, "ram_channel"); + print_buf_info(prtd->ram_link, "ram_link"); + print_buf_info(prtd->ram_link2, "ram_link2"); + print_buf_info(prtd->asp_channel, "asp_channel"); + print_buf_info(prtd->asp_link[0], "asp_link[0]"); + print_buf_info(prtd->asp_link[1], "asp_link[1]"); + + /* + * There is a phase offset of 2 periods between the position + * used by dma setup and the position reported in the pointer + * function. + * + * The phase offset, when not using ping-pong buffers, is due to + * the two consecutive calls to davinci_pcm_enqueue_dma() below. + * + * Whereas here, with ping-pong buffers, the phase is due to + * there being an entire buffer transfer complete before the + * first dma completion event triggers davinci_pcm_dma_irq(). + */ + davinci_pcm_period_elapsed(substream); + davinci_pcm_period_elapsed(substream); + + return 0; + } + davinci_pcm_enqueue_dma(substream); + davinci_pcm_period_elapsed(substream); + + /* Copy self-linked parameter RAM entry into master channel */ + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); + edma_write_slot(prtd->asp_channel, &prtd->asp_params); + davinci_pcm_enqueue_dma(substream); + davinci_pcm_period_elapsed(substream); + + return 0; +} + +static snd_pcm_uframes_t +davinci_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd = runtime->private_data; + unsigned int offset; + int asp_count; + unsigned int period_size = snd_pcm_lib_period_bytes(substream); + + /* + * There is a phase offset of 2 periods between the position used by dma + * setup and the position reported in the pointer function. Either +2 in + * the dma setup or -2 here in the pointer function (with wrapping, + * both) accounts for this offset -- choose the latter since it makes + * the first-time setup clearer. + */ + spin_lock(&prtd->lock); + asp_count = prtd->period - 2; + spin_unlock(&prtd->lock); + + if (asp_count < 0) + asp_count += runtime->periods; + asp_count *= period_size; + + offset = bytes_to_frames(runtime, asp_count); + if (offset >= runtime->buffer_size) + offset = 0; + + return offset; +} + +static int davinci_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd; + struct snd_pcm_hardware *ppcm; + int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_pcm_dma_params *pa; + struct davinci_pcm_dma_params *params; + + pa = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!pa) + return -ENODEV; + params = &pa[substream->stream]; + + ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &pcm_hardware_playback : &pcm_hardware_capture; + allocate_sram(substream, params->sram_size, ppcm); + snd_soc_set_runtime_hwparams(substream, ppcm); + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + prtd = kzalloc(sizeof(struct davinci_runtime_data), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + spin_lock_init(&prtd->lock); + prtd->params = params; + prtd->asp_channel = -1; + prtd->asp_link[0] = prtd->asp_link[1] = -1; + prtd->ram_channel = -1; + prtd->ram_link = -1; + prtd->ram_link2 = -1; + + runtime->private_data = prtd; + + ret = davinci_pcm_dma_request(substream); + if (ret) { + printk(KERN_ERR "davinci_pcm: Failed to get dma channels\n"); + kfree(prtd); + } + + return ret; +} + +static int davinci_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd = runtime->private_data; + + if (prtd->ram_channel >= 0) + edma_stop(prtd->ram_channel); + if (prtd->asp_channel >= 0) + edma_stop(prtd->asp_channel); + if (prtd->asp_link[0] >= 0) + edma_unlink(prtd->asp_link[0]); + if (prtd->asp_link[1] >= 0) + edma_unlink(prtd->asp_link[1]); + if (prtd->ram_link >= 0) + edma_unlink(prtd->ram_link); + + if (prtd->asp_link[0] >= 0) + edma_free_slot(prtd->asp_link[0]); + if (prtd->asp_link[1] >= 0) + edma_free_slot(prtd->asp_link[1]); + if (prtd->asp_channel >= 0) + edma_free_channel(prtd->asp_channel); + if (prtd->ram_link >= 0) + edma_free_slot(prtd->ram_link); + if (prtd->ram_link2 >= 0) + edma_free_slot(prtd->ram_link2); + if (prtd->ram_channel >= 0) + edma_free_channel(prtd->ram_channel); + + kfree(prtd); + + return 0; +} + +static int davinci_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int davinci_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int davinci_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops davinci_pcm_ops = { + .open = davinci_pcm_open, + .close = davinci_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = davinci_pcm_hw_params, + .hw_free = davinci_pcm_hw_free, + .prepare = davinci_pcm_prepare, + .trigger = davinci_pcm_trigger, + .pointer = davinci_pcm_pointer, + .mmap = davinci_pcm_mmap, +}; + +static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, + size_t size) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + + pr_debug("davinci_pcm: preallocate_dma_buffer: area=%p, addr=%p, " + "size=%d\n", (void *) buf->area, (void *) buf->addr, size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} + +static void davinci_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + struct snd_dma_buffer *iram_dma; + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + iram_dma = buf->private_data; + if (iram_dma) { + sram_free(iram_dma->area, iram_dma->bytes); + kfree(iram_dma); + } + } +} + +static u64 davinci_pcm_dmamask = DMA_BIT_MASK(32); + +static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &davinci_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = davinci_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK, + pcm_hardware_playback.buffer_bytes_max); + if (ret) + return ret; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = davinci_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE, + pcm_hardware_capture.buffer_bytes_max); + if (ret) + return ret; + } + + return 0; +} + +static struct snd_soc_platform_driver davinci_soc_platform = { + .ops = &davinci_pcm_ops, + .pcm_new = davinci_pcm_new, + .pcm_free = davinci_pcm_free, +}; + +static int __devinit davinci_soc_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &davinci_soc_platform); +} + +static int __devexit davinci_soc_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver davinci_pcm_driver = { + .driver = { + .name = "davinci-pcm-audio", + .owner = THIS_MODULE, + }, + + .probe = davinci_soc_platform_probe, + .remove = __devexit_p(davinci_soc_platform_remove), +}; + +module_platform_driver(davinci_pcm_driver); + +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_DESCRIPTION("TI DAVINCI PCM DMA module"); +MODULE_LICENSE("GPL"); diff --git a/ANDROID_3.4.5/sound/soc/davinci/davinci-pcm.h b/ANDROID_3.4.5/sound/soc/davinci/davinci-pcm.h new file mode 100644 index 00000000..c0d6c9be --- /dev/null +++ b/ANDROID_3.4.5/sound/soc/davinci/davinci-pcm.h @@ -0,0 +1,31 @@ +/* + * ALSA PCM interface for the TI DAVINCI processor + * + * Author: Vladimir Barinov, <vbarinov@embeddedalley.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.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. + */ + +#ifndef _DAVINCI_PCM_H +#define _DAVINCI_PCM_H + +#include <mach/edma.h> +#include <mach/asp.h> + + +struct davinci_pcm_dma_params { + int channel; /* sync dma channel ID */ + unsigned short acnt; + dma_addr_t dma_addr; /* device physical address for DMA */ + unsigned sram_size; + enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ + enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ + unsigned char data_type; /* xfer data type */ + unsigned char convert_mono_stereo; + unsigned int fifo_level; +}; + +#endif diff --git a/ANDROID_3.4.5/sound/soc/davinci/davinci-sffsdr.c b/ANDROID_3.4.5/sound/soc/davinci/davinci-sffsdr.c new file mode 100644 index 00000000..f71175b2 --- /dev/null +++ b/ANDROID_3.4.5/sound/soc/davinci/davinci-sffsdr.c @@ -0,0 +1,181 @@ +/* + * ASoC driver for Lyrtech SFFSDR board. + * + * Author: Hugo Villeneuve + * Copyright (C) 2008 Lyrtech inc + * + * Based on ASoC driver for TI DAVINCI EVM platform, original copyright follow: + * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.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/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> + +#include <asm/dma.h> +#include <asm/mach-types.h> +#ifdef CONFIG_SFFSDR_FPGA +#include <asm/plat-sffsdr/sffsdr-fpga.h> +#endif + +#include <mach/edma.h> + +#include "../codecs/pcm3008.h" +#include "davinci-pcm.h" +#include "davinci-i2s.h" + +/* + * CLKX and CLKR are the inputs for the Sample Rate Generator. + * FSX and FSR are outputs, driven by the sample Rate Generator. + */ +#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ + SND_SOC_DAIFMT_CBM_CFS | \ + SND_SOC_DAIFMT_IB_NF) + +static int sffsdr_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 *cpu_dai = rtd->cpu_dai; + int fs; + int ret = 0; + + /* Fsref can be 32000, 44100 or 48000. */ + fs = params_rate(params); + +#ifndef CONFIG_SFFSDR_FPGA + /* Without the FPGA module, the Fs is fixed at 44100 Hz */ + if (fs != 44100) { + pr_debug("warning: only 44.1 kHz is supported without SFFSDR FPGA module\n"); + return -EINVAL; + } +#endif + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); + if (ret < 0) + return ret; + + pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs); + +#ifndef CONFIG_SFFSDR_FPGA + return 0; +#else + return sffsdr_fpga_set_codec_fs(fs); +#endif +} + +static struct snd_soc_ops sffsdr_ops = { + .hw_params = sffsdr_hw_params, +}; + +/* davinci-sffsdr digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sffsdr_dai = { + .name = "PCM3008", /* Codec name */ + .stream_name = "PCM3008 HiFi", + .cpu_dai_name = "davinci-mcbsp", + .codec_dai_name = "pcm3008-hifi", + .codec_name = "pcm3008-codec", + .platform_name = "davinci-pcm-audio", + .ops = &sffsdr_ops, +}; + +/* davinci-sffsdr audio machine driver */ +static struct snd_soc_card snd_soc_sffsdr = { + .name = "DaVinci SFFSDR", + .owner = THIS_MODULE, + .dai_link = &sffsdr_dai, + .num_links = 1, +}; + +/* sffsdr audio private data */ +static struct pcm3008_setup_data sffsdr_pcm3008_setup = { + .dem0_pin = GPIO(45), + .dem1_pin = GPIO(46), + .pdad_pin = GPIO(47), + .pdda_pin = GPIO(38), +}; + +struct platform_device pcm3008_codec = { + .name = "pcm3008-codec", + .id = 0, + .dev = { + .platform_data = &sffsdr_pcm3008_setup, + }, +}; + +static struct resource sffsdr_snd_resources[] = { + { + .start = DAVINCI_MCBSP_BASE, + .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data sffsdr_snd_data = { + .tx_dma_ch = DAVINCI_DMA_MCBSP_TX, + .rx_dma_ch = DAVINCI_DMA_MCBSP_RX, +}; + +static struct platform_device *sffsdr_snd_device; + +static int __init sffsdr_init(void) +{ + int ret; + + if (!machine_is_sffsdr()) + return -EINVAL; + + platform_device_register(&pcm3008_codec); + + sffsdr_snd_device = platform_device_alloc("soc-audio", 0); + if (!sffsdr_snd_device) { + printk(KERN_ERR "platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(sffsdr_snd_device, &snd_soc_sffsdr); + platform_device_add_data(sffsdr_snd_device, &sffsdr_snd_data, + sizeof(sffsdr_snd_data)); + + ret = platform_device_add_resources(sffsdr_snd_device, + sffsdr_snd_resources, + ARRAY_SIZE(sffsdr_snd_resources)); + if (ret) { + printk(KERN_ERR "platform device add resources failed\n"); + goto error; + } + + ret = platform_device_add(sffsdr_snd_device); + if (ret) + goto error; + + return ret; + +error: + platform_device_put(sffsdr_snd_device); + return ret; +} + +static void __exit sffsdr_exit(void) +{ + platform_device_unregister(sffsdr_snd_device); + platform_device_unregister(&pcm3008_codec); +} + +module_init(sffsdr_init); +module_exit(sffsdr_exit); + +MODULE_AUTHOR("Hugo Villeneuve"); +MODULE_DESCRIPTION("Lyrtech SFFSDR ASoC driver"); +MODULE_LICENSE("GPL"); diff --git a/ANDROID_3.4.5/sound/soc/davinci/davinci-vcif.c b/ANDROID_3.4.5/sound/soc/davinci/davinci-vcif.c new file mode 100644 index 00000000..da030ff8 --- /dev/null +++ b/ANDROID_3.4.5/sound/soc/davinci/davinci-vcif.c @@ -0,0 +1,266 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar@ridgerun.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. + * + * 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/device.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <linux/mfd/davinci_voicecodec.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s.h" + +#define MOD_REG_BIT(val, mask, set) do { \ + if (set) { \ + val |= mask; \ + } else { \ + val &= ~mask; \ + } \ +} while (0) + +struct davinci_vcif_dev { + struct davinci_vc *davinci_vc; + struct davinci_pcm_dma_params dma_params[2]; +}; + +static void davinci_vcif_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *davinci_vcif_dev = + snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + u32 w; + + /* Start the sample generator and enable transmitter/receiver */ + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); + else + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); +} + +static void davinci_vcif_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *davinci_vcif_dev = + snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + u32 w; + + /* Reset transmitter/receiver and sample rate/frame sync generators */ + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); + else + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); +} + +static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct davinci_vcif_dev *davinci_vcif_dev = snd_soc_dai_get_drvdata(dai); + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + struct davinci_pcm_dma_params *dma_params = + &davinci_vcif_dev->dma_params[substream->stream]; + u32 w; + + /* Restart the codec before setup */ + davinci_vcif_stop(substream); + davinci_vcif_start(substream); + + /* General line settings */ + writel(DAVINCI_VC_CTRL_MASK, davinci_vc->base + DAVINCI_VC_CTRL); + + writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTCLR); + + writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTEN); + + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + + /* Determine xfer data type */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + dma_params->data_type = 0; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_BITS_8 | + DAVINCI_VC_CTRL_WD_UNSIGNED, 1); + break; + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_WD_BITS_8, 1); + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_UNSIGNED, 0); + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_BITS_8 | + DAVINCI_VC_CTRL_WD_UNSIGNED, 0); + break; + default: + printk(KERN_WARNING "davinci-vcif: unsupported PCM format"); + return -EINVAL; + } + + dma_params->acnt = dma_params->data_type; + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); + + return 0; +} + +static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_vcif_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_vcif_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int davinci_vcif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); + return 0; +} + +#define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 + +static const struct snd_soc_dai_ops davinci_vcif_dai_ops = { + .startup = davinci_vcif_startup, + .trigger = davinci_vcif_trigger, + .hw_params = davinci_vcif_hw_params, +}; + +static struct snd_soc_dai_driver davinci_vcif_dai = { + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_vcif_dai_ops, + +}; + +static int davinci_vcif_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = pdev->dev.platform_data; + struct davinci_vcif_dev *davinci_vcif_dev; + int ret; + + davinci_vcif_dev = devm_kzalloc(&pdev->dev, + sizeof(struct davinci_vcif_dev), + GFP_KERNEL); + if (!davinci_vcif_dev) { + dev_dbg(&pdev->dev, + "could not allocate memory for private data\n"); + return -ENOMEM; + } + + /* DMA tx params */ + davinci_vcif_dev->davinci_vc = davinci_vc; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = + davinci_vc->davinci_vcif.dma_tx_channel; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = + davinci_vc->davinci_vcif.dma_tx_addr; + + /* DMA rx params */ + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = + davinci_vc->davinci_vcif.dma_rx_channel; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = + davinci_vc->davinci_vcif.dma_rx_addr; + + dev_set_drvdata(&pdev->dev, davinci_vcif_dev); + + ret = snd_soc_register_dai(&pdev->dev, &davinci_vcif_dai); + if (ret != 0) { + dev_err(&pdev->dev, "could not register dai\n"); + return ret; + } + + return 0; +} + +static int davinci_vcif_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&pdev->dev); + + return 0; +} + +static struct platform_driver davinci_vcif_driver = { + .probe = davinci_vcif_probe, + .remove = davinci_vcif_remove, + .driver = { + .name = "davinci-vcif", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(davinci_vcif_driver); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC Voice Codec Interface"); +MODULE_LICENSE("GPL"); |