zte's code,first commit
Change-Id: I9a04da59e459a9bc0d67f101f700d9d7dc8d681b
diff --git a/ap/os/linux/linux-3.4.x/sound/soc/davinci/Kconfig b/ap/os/linux/linux-3.4.x/sound/soc/davinci/Kconfig
new file mode 100644
index 0000000..9e11a14
--- /dev/null
+++ b/ap/os/linux/linux-3.4.x/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/ap/os/linux/linux-3.4.x/sound/soc/davinci/Makefile b/ap/os/linux/linux-3.4.x/sound/soc/davinci/Makefile
new file mode 100644
index 0000000..a93679d
--- /dev/null
+++ b/ap/os/linux/linux-3.4.x/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/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-evm.c b/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-evm.c
new file mode 100644
index 0000000..10a2d8c
--- /dev/null
+++ b/ap/os/linux/linux-3.4.x/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/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-i2s.c b/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-i2s.c
new file mode 100644
index 0000000..0a74b95
--- /dev/null
+++ b/ap/os/linux/linux-3.4.x/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/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-i2s.h b/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-i2s.h
new file mode 100644
index 0000000..48dac3e
--- /dev/null
+++ b/ap/os/linux/linux-3.4.x/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/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-mcasp.c b/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-mcasp.c
new file mode 100644
index 0000000..95441bf
--- /dev/null
+++ b/ap/os/linux/linux-3.4.x/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/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-mcasp.h b/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-mcasp.h
new file mode 100644
index 0000000..4681acc
--- /dev/null
+++ b/ap/os/linux/linux-3.4.x/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/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-pcm.c b/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-pcm.c
new file mode 100644
index 0000000..97d77b2
--- /dev/null
+++ b/ap/os/linux/linux-3.4.x/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/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-pcm.h b/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-pcm.h
new file mode 100644
index 0000000..c0d6c9b
--- /dev/null
+++ b/ap/os/linux/linux-3.4.x/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/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-sffsdr.c b/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-sffsdr.c
new file mode 100644
index 0000000..f71175b
--- /dev/null
+++ b/ap/os/linux/linux-3.4.x/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/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-vcif.c b/ap/os/linux/linux-3.4.x/sound/soc/davinci/davinci-vcif.c
new file mode 100644
index 0000000..da030ff
--- /dev/null
+++ b/ap/os/linux/linux-3.4.x/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");