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authorLinus Torvalds <torvalds@linux-foundation.org>2012-07-24 22:37:37 +0200
committerLinus Torvalds <torvalds@linux-foundation.org>2012-07-24 22:37:37 +0200
commitdbf7b5915b39bfff548e4c6a3a753fc291a60e25 (patch)
tree55c457a22aa869d2ab558317877138369ae5f9bb /sound/soc/codecs/sta529.c
parentMerge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jik... (diff)
parentMerge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie... (diff)
downloadlinux-dbf7b5915b39bfff548e4c6a3a753fc291a60e25.tar.xz
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Merge tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound update from Takashi Iwai: "This is a fairly quiet release in all sound area. Only a little bit of changes in the core side while most of changes are seen in the drivers. HD-audio: - A few new codec additions for Nvidia, Realtek and VIA - Intel Haswell audio support - Support for "phantom" jacks for consistent jack reporting - Major clean-ups in HDMI/DP driver codes - A workaround for inverted digital-mic pins with Realtek codecs - Removal of beep_mode=2 option ASoC: - Added the ability to add and remove DAPM paths dynamically, mostly for reparenting on clock changes - New machine drivers for Marvell Brownstone, ST-Ericsson Ux500 reference platform and ttc-dkp - New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP, Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF - New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI Isabelle and Wolfson Microelectronics WM5102 and WM5110 - DAPM fixes for the recent locking changes - Fix for _PRE and _POST widgets (which have been broken for a few releases now) - A couple of minor driver updates Misc - Conversion to new dev_pm_ops in platform and PCI drivers - LTC support and some fixes in PCXHR driver - A few fixes and PM support for ISA OPti9xx and WSS cards - Some TLV code cleanup - Move driver-specific headers from include/sound to local dirs" * tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (212 commits) ASoC: dapm: Fix _PRE and _POST events for DAPM performance improvements ALSA: hda - add dock support for Thinkpad X230 Tablet ALSA: hda - Turn on PIN_OUT from hdmi playback prepare. ASoC imx-audmux: add MX31_AUDMUX_PORT7_SSI_PINS_7 define ASoC: littlemill: Add userspace control of the WM1250 I/O ASoC: wm8994: Update micdet for irqdomain conversion ALSA: hda - make sure alc268 does not OOPS on codec parse ALSA: hda - Add support for Realtek ALC282 ALSA: hda - Fix index number conflicts of phantom jacks ALSA: opti9xx: Fix section mismatch by PM support ALSA: snd-opti9xx: Implement suspend/resume ALSA: hda - Add new GPU codec ID to snd-hda ALSA: hda - Fix driver type of Haswell controller to AZX_DRIVER_SCH ALSA: hda - add Haswell HDMI codec id ALSA: hda - Add DeviceID for Haswell HDA ALSA: wss_lib: Fix resume on Yamaha OPL3-SAx ALSA: wss_lib: fix suspend/resume ALSA: es1938: replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ALSA: tlv: add DECLARE_TLV_DB_RANGE() ALSA: tlv: add DECLARE_TLV_CONTAINER() ...
Diffstat (limited to 'sound/soc/codecs/sta529.c')
-rw-r--r--sound/soc/codecs/sta529.c442
1 files changed, 442 insertions, 0 deletions
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
new file mode 100644
index 000000000000..0c225cd569d2
--- /dev/null
+++ b/sound/soc/codecs/sta529.c
@@ -0,0 +1,442 @@
+/*
+ * ASoC codec driver for spear platform
+ *
+ * sound/soc/codecs/sta529.c -- spear ALSA Soc codec driver
+ *
+ * Copyright (C) 2012 ST Microelectronics
+ * Rajeev Kumar <rajeev-dlh.kumar@st.com>
+ *
+ * This file is licensed under the terms of the GNU General Public
+ * License version 2. This program is licensed "as is" without any
+ * warranty of any kind, whether express or implied.
+ */
+
+#include <linux/clk.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/pm.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+/* STA529 Register offsets */
+#define STA529_FFXCFG0 0x00
+#define STA529_FFXCFG1 0x01
+#define STA529_MVOL 0x02
+#define STA529_LVOL 0x03
+#define STA529_RVOL 0x04
+#define STA529_TTF0 0x05
+#define STA529_TTF1 0x06
+#define STA529_TTP0 0x07
+#define STA529_TTP1 0x08
+#define STA529_S2PCFG0 0x0A
+#define STA529_S2PCFG1 0x0B
+#define STA529_P2SCFG0 0x0C
+#define STA529_P2SCFG1 0x0D
+#define STA529_PLLCFG0 0x14
+#define STA529_PLLCFG1 0x15
+#define STA529_PLLCFG2 0x16
+#define STA529_PLLCFG3 0x17
+#define STA529_PLLPFE 0x18
+#define STA529_PLLST 0x19
+#define STA529_ADCCFG 0x1E /*mic_select*/
+#define STA529_CKOCFG 0x1F
+#define STA529_MISC 0x20
+#define STA529_PADST0 0x21
+#define STA529_PADST1 0x22
+#define STA529_FFXST 0x23
+#define STA529_PWMIN1 0x2D
+#define STA529_PWMIN2 0x2E
+#define STA529_POWST 0x32
+
+#define STA529_MAX_REGISTER 0x32
+
+#define STA529_RATES (SNDRV_PCM_RATE_8000 | \
+ SNDRV_PCM_RATE_11025 | \
+ SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+#define STA529_FORMAT (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+#define S2PC_VALUE 0x98
+#define CLOCK_OUT 0x60
+#define LEFT_J_DATA_FORMAT 0x10
+#define I2S_DATA_FORMAT 0x12
+#define RIGHT_J_DATA_FORMAT 0x14
+#define CODEC_MUTE_VAL 0x80
+
+#define POWER_CNTLMSAK 0x40
+#define POWER_STDBY 0x40
+#define FFX_MASK 0x80
+#define FFX_OFF 0x80
+#define POWER_UP 0x00
+#define FFX_CLK_ENB 0x01
+#define FFX_CLK_DIS 0x00
+#define FFX_CLK_MSK 0x01
+#define PLAY_FREQ_RANGE_MSK 0x70
+#define CAP_FREQ_RANGE_MSK 0x0C
+#define PDATA_LEN_MSK 0xC0
+#define BCLK_TO_FS_MSK 0x30
+#define AUDIO_MUTE_MSK 0x80
+
+static const struct reg_default sta529_reg_defaults[] = {
+ { 0, 0x35 }, /* R0 - FFX Configuration reg 0 */
+ { 1, 0xc8 }, /* R1 - FFX Configuration reg 1 */
+ { 2, 0x50 }, /* R2 - Master Volume */
+ { 3, 0x00 }, /* R3 - Left Volume */
+ { 4, 0x00 }, /* R4 - Right Volume */
+ { 10, 0xb2 }, /* R10 - S2P Config Reg 0 */
+ { 11, 0x41 }, /* R11 - S2P Config Reg 1 */
+ { 12, 0x92 }, /* R12 - P2S Config Reg 0 */
+ { 13, 0x41 }, /* R13 - P2S Config Reg 1 */
+ { 30, 0xd2 }, /* R30 - ADC Config Reg */
+ { 31, 0x40 }, /* R31 - clock Out Reg */
+ { 32, 0x21 }, /* R32 - Misc Register */
+};
+
+struct sta529 {
+ struct regmap *regmap;
+};
+
+static bool sta529_readable(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+
+ case STA529_FFXCFG0:
+ case STA529_FFXCFG1:
+ case STA529_MVOL:
+ case STA529_LVOL:
+ case STA529_RVOL:
+ case STA529_S2PCFG0:
+ case STA529_S2PCFG1:
+ case STA529_P2SCFG0:
+ case STA529_P2SCFG1:
+ case STA529_ADCCFG:
+ case STA529_CKOCFG:
+ case STA529_MISC:
+ return true;
+ default:
+ return false;
+ }
+}
+
+
+static const char *pwm_mode_text[] = { "Binary", "Headphone", "Ternary",
+ "Phase-shift"};
+
+static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -9150, 50, 0);
+static const DECLARE_TLV_DB_SCALE(master_vol_tlv, -12750, 50, 0);
+static const SOC_ENUM_SINGLE_DECL(pwm_src, STA529_FFXCFG1, 4, pwm_mode_text);
+
+static const struct snd_kcontrol_new sta529_snd_controls[] = {
+ SOC_DOUBLE_R_TLV("Digital Playback Volume", STA529_LVOL, STA529_RVOL, 0,
+ 127, 0, out_gain_tlv),
+ SOC_SINGLE_TLV("Master Playback Volume", STA529_MVOL, 0, 127, 1,
+ master_vol_tlv),
+ SOC_ENUM("PWM Select", pwm_src),
+};
+
+static int sta529_set_bias_level(struct snd_soc_codec *codec, enum
+ snd_soc_bias_level level)
+{
+ struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ snd_soc_update_bits(codec, STA529_FFXCFG0, POWER_CNTLMSAK,
+ POWER_UP);
+ snd_soc_update_bits(codec, STA529_MISC, FFX_CLK_MSK,
+ FFX_CLK_ENB);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+ regcache_sync(sta529->regmap);
+ snd_soc_update_bits(codec, STA529_FFXCFG0,
+ POWER_CNTLMSAK, POWER_STDBY);
+ /* Making FFX output to zero */
+ snd_soc_update_bits(codec, STA529_FFXCFG0, FFX_MASK,
+ FFX_OFF);
+ snd_soc_update_bits(codec, STA529_MISC, FFX_CLK_MSK,
+ FFX_CLK_DIS);
+ break;
+ case SND_SOC_BIAS_OFF:
+ break;
+ }
+
+ /*
+ * store the label for powers down audio subsystem for suspend.This is
+ * used by soc core layer
+ */
+ codec->dapm.bias_level = level;
+
+ return 0;
+
+}
+
+static int sta529_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->codec;
+ int pdata, play_freq_val, record_freq_val;
+ int bclk_to_fs_ratio;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ pdata = 1;
+ bclk_to_fs_ratio = 0;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ pdata = 2;
+ bclk_to_fs_ratio = 1;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ pdata = 3;
+ bclk_to_fs_ratio = 2;
+ break;
+ default:
+ dev_err(codec->dev, "Unsupported format\n");
+ return -EINVAL;
+ }
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 11025:
+ play_freq_val = 0;
+ record_freq_val = 2;
+ break;
+ case 16000:
+ case 22050:
+ play_freq_val = 1;
+ record_freq_val = 0;
+ break;
+
+ case 32000:
+ case 44100:
+ case 48000:
+ play_freq_val = 2;
+ record_freq_val = 0;
+ break;
+ default:
+ dev_err(codec->dev, "Unsupported rate\n");
+ return -EINVAL;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ snd_soc_update_bits(codec, STA529_S2PCFG1, PDATA_LEN_MSK,
+ pdata << 6);
+ snd_soc_update_bits(codec, STA529_S2PCFG1, BCLK_TO_FS_MSK,
+ bclk_to_fs_ratio << 4);
+ snd_soc_update_bits(codec, STA529_MISC, PLAY_FREQ_RANGE_MSK,
+ play_freq_val << 4);
+ } else {
+ snd_soc_update_bits(codec, STA529_P2SCFG1, PDATA_LEN_MSK,
+ pdata << 6);
+ snd_soc_update_bits(codec, STA529_P2SCFG1, BCLK_TO_FS_MSK,
+ bclk_to_fs_ratio << 4);
+ snd_soc_update_bits(codec, STA529_MISC, CAP_FREQ_RANGE_MSK,
+ record_freq_val << 2);
+ }
+
+ return 0;
+}
+
+static int sta529_mute(struct snd_soc_dai *dai, int mute)
+{
+ u8 val = 0;
+
+ if (mute)
+ val |= CODEC_MUTE_VAL;
+
+ snd_soc_update_bits(dai->codec, STA529_FFXCFG0, AUDIO_MUTE_MSK, val);
+
+ return 0;
+}
+
+static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u8 mode = 0;
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_LEFT_J:
+ mode = LEFT_J_DATA_FORMAT;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ mode = I2S_DATA_FORMAT;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ mode = RIGHT_J_DATA_FORMAT;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops sta529_dai_ops = {
+ .hw_params = sta529_hw_params,
+ .set_fmt = sta529_set_dai_fmt,
+ .digital_mute = sta529_mute,
+};
+
+static struct snd_soc_dai_driver sta529_dai = {
+ .name = "sta529-audio",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = STA529_RATES,
+ .formats = STA529_FORMAT,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = STA529_RATES,
+ .formats = STA529_FORMAT,
+ },
+ .ops = &sta529_dai_ops,
+};
+
+static int sta529_probe(struct snd_soc_codec *codec)
+{
+ struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ codec->control_data = sta529->regmap;
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
+
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+ sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+
+/* power down chip */
+static int sta529_remove(struct snd_soc_codec *codec)
+{
+ sta529_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int sta529_suspend(struct snd_soc_codec *codec)
+{
+ sta529_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int sta529_resume(struct snd_soc_codec *codec)
+{
+ sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+
+struct snd_soc_codec_driver sta529_codec_driver = {
+ .probe = sta529_probe,
+ .remove = sta529_remove,
+ .set_bias_level = sta529_set_bias_level,
+ .suspend = sta529_suspend,
+ .resume = sta529_resume,
+ .controls = sta529_snd_controls,
+ .num_controls = ARRAY_SIZE(sta529_snd_controls),
+};
+
+static const struct regmap_config sta529_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = STA529_MAX_REGISTER,
+ .readable_reg = sta529_readable,
+
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults = sta529_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(sta529_reg_defaults),
+};
+
+static __devinit int sta529_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct sta529 *sta529;
+ int ret;
+
+ if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
+ return -EINVAL;
+
+ sta529 = devm_kzalloc(&i2c->dev, sizeof(struct sta529), GFP_KERNEL);
+ if (sta529 == NULL) {
+ dev_err(&i2c->dev, "Can not allocate memory\n");
+ return -ENOMEM;
+ }
+
+ sta529->regmap = devm_regmap_init_i2c(i2c, &sta529_regmap);
+ if (IS_ERR(sta529->regmap)) {
+ ret = PTR_ERR(sta529->regmap);
+ dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret);
+ return ret;
+ }
+
+ i2c_set_clientdata(i2c, sta529);
+
+ ret = snd_soc_register_codec(&i2c->dev,
+ &sta529_codec_driver, &sta529_dai, 1);
+ if (ret != 0)
+ dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
+
+ return ret;
+}
+
+static int __devexit sta529_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+
+ return 0;
+}
+
+static const struct i2c_device_id sta529_i2c_id[] = {
+ { "sta529", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, sta529_i2c_id);
+
+static struct i2c_driver sta529_i2c_driver = {
+ .driver = {
+ .name = "sta529",
+ .owner = THIS_MODULE,
+ },
+ .probe = sta529_i2c_probe,
+ .remove = __devexit_p(sta529_i2c_remove),
+ .id_table = sta529_i2c_id,
+};
+
+module_i2c_driver(sta529_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC STA529 codec driver");
+MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
+MODULE_LICENSE("GPL");