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-rw-r--r--sound/soc/Kconfig32
-rw-r--r--sound/soc/Makefile4
-rw-r--r--sound/soc/at91/Kconfig32
-rw-r--r--sound/soc/at91/Makefile11
-rw-r--r--sound/soc/at91/at91-i2s.c720
-rw-r--r--sound/soc/at91/at91-i2s.h27
-rw-r--r--sound/soc/at91/at91-pcm.c432
-rw-r--r--sound/soc/at91/at91-pcm.h72
-rw-r--r--sound/soc/at91/eti_b1_wm8731.c375
-rw-r--r--sound/soc/codecs/Kconfig15
-rw-r--r--sound/soc/codecs/Makefile9
-rw-r--r--sound/soc/codecs/ac97.c156
-rw-r--r--sound/soc/codecs/ac97.h18
-rw-r--r--sound/soc/codecs/wm8731.c758
-rw-r--r--sound/soc/codecs/wm8731.h44
-rw-r--r--sound/soc/codecs/wm8750.c1049
-rw-r--r--sound/soc/codecs/wm8750.h67
-rw-r--r--sound/soc/codecs/wm9712.c771
-rw-r--r--sound/soc/codecs/wm9712.h14
-rw-r--r--sound/soc/pxa/Kconfig60
-rw-r--r--sound/soc/pxa/Makefile20
-rw-r--r--sound/soc/pxa/corgi.c383
-rw-r--r--sound/soc/pxa/poodle.c352
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c431
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.h22
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c318
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.h20
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.c372
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.h34
-rw-r--r--sound/soc/pxa/spitz.c394
-rw-r--r--sound/soc/pxa/tosa.c289
-rw-r--r--sound/soc/soc-core.c1587
-rw-r--r--sound/soc/soc-dapm.c1323
33 files changed, 10211 insertions, 0 deletions
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
new file mode 100644
index 00000000000..ec821a57f84
--- /dev/null
+++ b/sound/soc/Kconfig
@@ -0,0 +1,32 @@
+#
+# SoC audio configuration
+#
+
+menu "SoC audio support"
+ depends on SND!=n
+
+config SND_SOC_AC97_BUS
+ bool
+
+config SND_SOC
+ tristate "SoC audio support"
+ ---help---
+
+ If you want SoC support, you should say Y here and also to the
+ specific driver for your SoC below. You will also need to select the
+ specific codec(s) attached to the SoC
+
+ This SoC audio support can also be built as a module. If so, the module
+ will be called snd-soc-core.
+
+# All the supported Soc's
+menu "SoC Platforms"
+depends on SND_SOC
+source "sound/soc/at91/Kconfig"
+source "sound/soc/pxa/Kconfig"
+endmenu
+
+# Supported codecs
+source "sound/soc/codecs/Kconfig"
+
+endmenu
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
new file mode 100644
index 00000000000..98e6f49dafc
--- /dev/null
+++ b/sound/soc/Makefile
@@ -0,0 +1,4 @@
+snd-soc-core-objs := soc-core.o soc-dapm.o
+
+obj-$(CONFIG_SND_SOC) += snd-soc-core.o
+obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/
diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig
new file mode 100644
index 00000000000..5bcf08b728b
--- /dev/null
+++ b/sound/soc/at91/Kconfig
@@ -0,0 +1,32 @@
+menu "SoC Audio for the Atmel AT91"
+
+config SND_AT91_SOC
+ tristate "SoC Audio for the Atmel AT91 System-on-Chip"
+ depends on ARCH_AT91 && SND
+ select SND_PCM
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the AT91 SSC interface. You will also need
+ to select the audio interfaces to support below.
+
+config SND_AT91_SOC_I2S
+ tristate
+
+config SND_AT91_SOC_ETI_B1_WM8731
+ tristate "SoC I2S Audio support for WM8731-based Endrelia ETI-B1 boards"
+ depends on SND_AT91_SOC && (MACH_ETI_B1 || MACH_ETI_C1)
+ select SND_AT91_SOC_I2S
+ select SND_SOC_WM8731
+ help
+ Say Y if you want to add support for SoC audio on WM8731-based
+ Endrelia Technologies Inc ETI-B1 or ETI-C1 boards.
+
+config SND_AT91_SOC_ETI_SLAVE
+ bool "Run codec in slave Mode on Endrelia boards"
+ depends on SND_AT91_SOC_ETI_B1_WM8731
+ default n
+ help
+ Say Y if you want to run with the AT91 SSC generating the BCLK
+ and LRC signals on Endrelia boards.
+
+endmenu
diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile
new file mode 100644
index 00000000000..b77b01ab202
--- /dev/null
+++ b/sound/soc/at91/Makefile
@@ -0,0 +1,11 @@
+# AT91 Platform Support
+snd-soc-at91-objs := at91-pcm.o
+snd-soc-at91-i2s-objs := at91-i2s.o
+
+obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o
+obj-$(CONFIG_SND_AT91_SOC_I2S) += snd-soc-at91-i2s.o
+
+# AT91 Machine Support
+snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o
+
+obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o
diff --git a/sound/soc/at91/at91-i2s.c b/sound/soc/at91/at91-i2s.c
new file mode 100644
index 00000000000..fcc544a96ba
--- /dev/null
+++ b/sound/soc/at91/at91-i2s.c
@@ -0,0 +1,720 @@
+/*
+ * at91-i2s.c -- ALSA SoC I2S Audio Layer Platform driver
+ *
+ * Author: Frank Mandarino <fmandarino@endrelia.com>
+ * Endrelia Technologies Inc.
+ *
+ * Based on pxa2xx Platform drivers by
+ * Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <asm/arch/hardware.h>
+#include <asm/arch/at91_pmc.h>
+#include <asm/arch/at91_ssc.h>
+#include <asm/arch/at91_pdc.h>
+
+#include "at91-pcm.h"
+#include "at91-i2s.h"
+
+#if 0
+#define DBG(x...) printk(KERN_DEBUG "at91-i2s:" x)
+#else
+#define DBG(x...)
+#endif
+
+#if defined(CONFIG_ARCH_AT91SAM9260)
+#define NUM_SSC_DEVICES 1
+#else
+#define NUM_SSC_DEVICES 3
+#endif
+
+
+/*
+ * SSC PDC registers required by the PCM DMA engine.
+ */
+static struct at91_pdc_regs pdc_tx_reg = {
+ .xpr = AT91_PDC_TPR,
+ .xcr = AT91_PDC_TCR,
+ .xnpr = AT91_PDC_TNPR,
+ .xncr = AT91_PDC_TNCR,
+};
+
+static struct at91_pdc_regs pdc_rx_reg = {
+ .xpr = AT91_PDC_RPR,
+ .xcr = AT91_PDC_RCR,
+ .xnpr = AT91_PDC_RNPR,
+ .xncr = AT91_PDC_RNCR,
+};
+
+/*
+ * SSC & PDC status bits for transmit and receive.
+ */
+static struct at91_ssc_mask ssc_tx_mask = {
+ .ssc_enable = AT91_SSC_TXEN,
+ .ssc_disable = AT91_SSC_TXDIS,
+ .ssc_endx = AT91_SSC_ENDTX,
+ .ssc_endbuf = AT91_SSC_TXBUFE,
+ .pdc_enable = AT91_PDC_TXTEN,
+ .pdc_disable = AT91_PDC_TXTDIS,
+};
+
+static struct at91_ssc_mask ssc_rx_mask = {
+ .ssc_enable = AT91_SSC_RXEN,
+ .ssc_disable = AT91_SSC_RXDIS,
+ .ssc_endx = AT91_SSC_ENDRX,
+ .ssc_endbuf = AT91_SSC_RXBUFF,
+ .pdc_enable = AT91_PDC_RXTEN,
+ .pdc_disable = AT91_PDC_RXTDIS,
+};
+
+
+/*
+ * DMA parameters.
+ */
+static struct at91_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
+ {{
+ .name = "SSC0/I2S PCM Stereo out",
+ .pdc = &pdc_tx_reg,
+ .mask = &ssc_tx_mask,
+ },
+ {
+ .name = "SSC0/I2S PCM Stereo in",
+ .pdc = &pdc_rx_reg,
+ .mask = &ssc_rx_mask,
+ }},
+#if NUM_SSC_DEVICES == 3
+ {{
+ .name = "SSC1/I2S PCM Stereo out",
+ .pdc = &pdc_tx_reg,
+ .mask = &ssc_tx_mask,
+ },
+ {
+ .name = "SSC1/I2S PCM Stereo in",
+ .pdc = &pdc_rx_reg,
+ .mask = &ssc_rx_mask,
+ }},
+ {{
+ .name = "SSC2/I2S PCM Stereo out",
+ .pdc = &pdc_tx_reg,
+ .mask = &ssc_tx_mask,
+ },
+ {
+ .name = "SSC1/I2S PCM Stereo in",
+ .pdc = &pdc_rx_reg,
+ .mask = &ssc_rx_mask,
+ }},
+#endif
+};
+
+struct at91_ssc_state {
+ u32 ssc_cmr;
+ u32 ssc_rcmr;
+ u32 ssc_rfmr;
+ u32 ssc_tcmr;
+ u32 ssc_tfmr;
+ u32 ssc_sr;
+ u32 ssc_imr;
+};
+
+static struct at91_ssc_info {
+ char *name;
+ struct at91_ssc_periph ssc;
+ spinlock_t lock; /* lock for dir_mask */
+ unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */
+ unsigned short initialized; /* 1=SSC has been initialized */
+ unsigned short daifmt;
+ unsigned short cmr_div;
+ unsigned short tcmr_period;
+ unsigned short rcmr_period;
+ struct at91_pcm_dma_params *dma_params[2];
+ struct at91_ssc_state ssc_state;
+
+} ssc_info[NUM_SSC_DEVICES] = {
+ {
+ .name = "ssc0",
+ .lock = SPIN_LOCK_UNLOCKED,
+ .dir_mask = 0,
+ .initialized = 0,
+ },
+#if NUM_SSC_DEVICES == 3
+ {
+ .name = "ssc1",
+ .lock = SPIN_LOCK_UNLOCKED,
+ .dir_mask = 0,
+ .initialized = 0,
+ },
+ {
+ .name = "ssc2",
+ .lock = SPIN_LOCK_UNLOCKED,
+ .dir_mask = 0,
+ .initialized = 0,
+ },
+#endif
+};
+
+static unsigned int at91_i2s_sysclk;
+
+/*
+ * SSC interrupt handler. Passes PDC interrupts to the DMA
+ * interrupt handler in the PCM driver.
+ */
+static irqreturn_t at91_i2s_interrupt(int irq, void *dev_id)
+{
+ struct at91_ssc_info *ssc_p = dev_id;
+ struct at91_pcm_dma_params *dma_params;
+ u32 ssc_sr;
+ int i;
+
+ ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)
+ & at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR);
+
+ /*
+ * Loop through the substreams attached to this SSC. If
+ * a DMA-related interrupt occurred on that substream, call
+ * the DMA interrupt handler function, if one has been
+ * registered in the dma_params structure by the PCM driver.
+ */
+ for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) {
+ dma_params = ssc_p->dma_params[i];
+
+ if (dma_params != NULL && dma_params->dma_intr_handler != NULL &&
+ (ssc_sr &
+ (dma_params->mask->ssc_endx | dma_params->mask->ssc_endbuf)))
+
+ dma_params->dma_intr_handler(ssc_sr, dma_params->substream);
+ }
+
+ return IRQ_HANDLED;
+}
+
+/*
+ * Startup. Only that one substream allowed in each direction.
+ */
+static int at91_i2s_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+ int dir_mask;
+
+ DBG("i2s_startup: SSC_SR=0x%08lx\n",
+ at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR));
+ dir_mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0x1 : 0x2;
+
+ spin_lock_irq(&ssc_p->lock);
+ if (ssc_p->dir_mask & dir_mask) {
+ spin_unlock_irq(&ssc_p->lock);
+ return -EBUSY;
+ }
+ ssc_p->dir_mask |= dir_mask;
+ spin_unlock_irq(&ssc_p->lock);
+
+ return 0;
+}
+
+/*
+ * Shutdown. Clear DMA parameters and shutdown the SSC if there
+ * are no other substreams open.
+ */
+static void at91_i2s_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+ struct at91_pcm_dma_params *dma_params;
+ int dir, dir_mask;
+
+ dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+ dma_params = ssc_p->dma_params[dir];
+
+ if (dma_params != NULL) {
+ at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR,
+ dma_params->mask->ssc_disable);
+ DBG("%s disabled SSC_SR=0x%08lx\n", (dir ? "receive" : "transmit"),
+ at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR));
+
+ dma_params->ssc_base = NULL;
+ dma_params->substream = NULL;
+ ssc_p->dma_params[dir] = NULL;
+ }
+
+ dir_mask = 1 << dir;
+
+ spin_lock_irq(&ssc_p->lock);
+ ssc_p->dir_mask &= ~dir_mask;
+ if (!ssc_p->dir_mask) {
+ /* Shutdown the SSC clock. */
+ DBG("Stopping pid %d clock\n", ssc_p->ssc.pid);
+ at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid);
+
+ if (ssc_p->initialized) {
+ free_irq(ssc_p->ssc.pid, ssc_p);
+ ssc_p->initialized = 0;
+ }
+
+ /* Reset the SSC */
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST);
+
+ /* Clear the SSC dividers */
+ ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0;
+ }
+ spin_unlock_irq(&ssc_p->lock);
+}
+
+/*
+ * Record the SSC system clock rate.
+ */
+static int at91_i2s_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ /*
+ * The only clock supplied to the SSC is the AT91 master clock,
+ * which is only used if the SSC is generating BCLK and/or
+ * LRC clocks.
+ */
+ switch (clk_id) {
+ case AT91_SYSCLK_MCK:
+ at91_i2s_sysclk = freq;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+/*
+ * Record the DAI format for use in hw_params().
+ */
+static int at91_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
+
+ if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S)
+ return -EINVAL;
+
+ ssc_p->daifmt = fmt;
+ return 0;
+}
+
+/*
+ * Record SSC clock dividers for use in hw_params().
+ */
+static int at91_i2s_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
+ int div_id, int div)
+{
+ struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
+
+ switch (div_id) {
+ case AT91SSC_CMR_DIV:
+ /*
+ * The same master clock divider is used for both
+ * transmit and receive, so if a value has already
+ * been set, it must match this value.
+ */
+ if (ssc_p->cmr_div == 0)
+ ssc_p->cmr_div = div;
+ else
+ if (div != ssc_p->cmr_div)
+ return -EBUSY;
+ break;
+
+ case AT91SSC_TCMR_PERIOD:
+ ssc_p->tcmr_period = div;
+ break;
+
+ case AT91SSC_RCMR_PERIOD:
+ ssc_p->rcmr_period = div;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+/*
+ * Configure the SSC.
+ */
+static int at91_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int id = rtd->dai->cpu_dai->id;
+ struct at91_ssc_info *ssc_p = &ssc_info[id];
+ struct at91_pcm_dma_params *dma_params;
+ int dir, channels, bits;
+ u32 tfmr, rfmr, tcmr, rcmr;
+ int start_event;
+ int ret;
+
+ /*
+ * Currently, there is only one set of dma params for
+ * each direction. If more are added, this code will
+ * have to be changed to select the proper set.
+ */
+ dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+
+ dma_params = &ssc_dma_params[id][dir];
+ dma_params->ssc_base = ssc_p->ssc.base;
+ dma_params->substream = substream;
+
+ ssc_p->dma_params[dir] = dma_params;
+
+ /*
+ * The cpu_dai->dma_data field is only used to communicate the
+ * appropriate DMA parameters to the pcm driver hw_params()
+ * function. It should not be used for other purposes
+ * as it is common to all substreams.
+ */
+ rtd->dai->cpu_dai->dma_data = dma_params;
+
+ channels = params_channels(params);
+
+ /*
+ * The SSC only supports up to 16-bit samples in I2S format, due
+ * to the size of the Frame Mode Register FSLEN field. Also, I2S
+ * implies signed data.
+ */
+ bits = 16;
+ dma_params->pdc_xfer_size = 2;
+
+ /*
+ * Compute SSC register settings.
+ */
+ switch (ssc_p->daifmt) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ /*
+ * SSC provides BCLK and LRC clocks.
+ *
+ * The SSC transmit and receive clocks are generated from the
+ * MCK divider, and the BCLK signal is output on the SSC TK line.
+ */
+ rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD)
+ | (( 1 << 16) & AT91_SSC_STTDLY)
+ | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START)
+ | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI)
+ | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO)
+ | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS);
+
+ rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE)
+ | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS)
+ | (((bits - 1) << 16) & AT91_SSC_FSLEN)
+ | (((channels - 1) << 8) & AT91_SSC_DATNB)
+ | (( 1 << 7) & AT91_SSC_MSBF)
+ | (( 0 << 5) & AT91_SSC_LOOP)
+ | (((bits - 1) << 0) & AT91_SSC_DATALEN);
+
+ tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD)
+ | (( 1 << 16) & AT91_SSC_STTDLY)
+ | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START)
+ | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI)
+ | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO)
+ | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS);
+
+ tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE)
+ | (( 0 << 23) & AT91_SSC_FSDEN)
+ | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS)
+ | (((bits - 1) << 16) & AT91_SSC_FSLEN)
+ | (((channels - 1) << 8) & AT91_SSC_DATNB)
+ | (( 1 << 7) & AT91_SSC_MSBF)
+ | (( 0 << 5) & AT91_SSC_DATDEF)
+ | (((bits - 1) << 0) & AT91_SSC_DATALEN);
+ break;
+
+ case SND_SOC_DAIFMT_CBM_CFM:
+
+ /*
+ * CODEC supplies BCLK and LRC clocks.
+ *
+ * The SSC transmit clock is obtained from the BCLK signal on
+ * on the TK line, and the SSC receive clock is generated from the
+ * transmit clock.
+ *
+ * For single channel data, one sample is transferred on the falling
+ * edge of the LRC clock. For two channel data, one sample is
+ * transferred on both edges of the LRC clock.
+ */
+ start_event = channels == 1
+ ? AT91_SSC_START_FALLING_RF
+ : AT91_SSC_START_EDGE_RF;
+
+ rcmr = (( 0 << 24) & AT91_SSC_PERIOD)
+ | (( 1 << 16) & AT91_SSC_STTDLY)
+ | (( start_event ) & AT91_SSC_START)
+ | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI)
+ | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO)
+ | (( AT91_SSC_CKS_CLOCK ) & AT91_SSC_CKS);
+
+ rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE)
+ | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS)
+ | (( 0 << 16) & AT91_SSC_FSLEN)
+ | (( 0 << 8) & AT91_SSC_DATNB)
+ | (( 1 << 7) & AT91_SSC_MSBF)
+ | (( 0 << 5) & AT91_SSC_LOOP)
+ | (((bits - 1) << 0) & AT91_SSC_DATALEN);
+
+ tcmr = (( 0 << 24) & AT91_SSC_PERIOD)
+ | (( 1 << 16) & AT91_SSC_STTDLY)
+ | (( start_event ) & AT91_SSC_START)
+ | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI)
+ | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO)
+ | (( AT91_SSC_CKS_PIN ) & AT91_SSC_CKS);
+
+ tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE)
+ | (( 0 << 23) & AT91_SSC_FSDEN)
+ | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS)
+ | (( 0 << 16) & AT91_SSC_FSLEN)
+ | (( 0 << 8) & AT91_SSC_DATNB)
+ | (( 1 << 7) & AT91_SSC_MSBF)
+ | (( 0 << 5) & AT91_SSC_DATDEF)
+ | (((bits - 1) << 0) & AT91_SSC_DATALEN);
+ break;
+
+ case SND_SOC_DAIFMT_CBS_CFM:
+ case SND_SOC_DAIFMT_CBM_CFS:
+ default:
+ printk(KERN_WARNING "at91-i2s: unsupported DAI format 0x%x.\n",
+ ssc_p->daifmt);
+ return -EINVAL;
+ break;
+ }
+ DBG("RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", rcmr, rfmr, tcmr, tfmr);
+
+ if (!ssc_p->initialized) {
+
+ /* Enable PMC peripheral clock for this SSC */
+ DBG("Starting pid %d clock\n", ssc_p->ssc.pid);
+ at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid);
+
+ /* Reset the SSC and its PDC registers */
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST);
+
+ at91_ssc_write(ssc_p->ssc.base + AT91_PDC_RPR, 0);
+ at91_ssc_write(ssc_p->ssc.base + AT91_PDC_RCR, 0);
+ at91_ssc_write(ssc_p->ssc.base + AT91_PDC_RNPR, 0);
+ at91_ssc_write(ssc_p->ssc.base + AT91_PDC_RNCR, 0);
+ at91_ssc_write(ssc_p->ssc.base + AT91_PDC_TPR, 0);
+ at91_ssc_write(ssc_p->ssc.base + AT91_PDC_TCR, 0);
+ at91_ssc_write(ssc_p->ssc.base + AT91_PDC_TNPR, 0);
+ at91_ssc_write(ssc_p->ssc.base + AT91_PDC_TNCR, 0);
+
+ if ((ret = request_irq(ssc_p->ssc.pid, at91_i2s_interrupt,
+ 0, ssc_p->name, ssc_p)) < 0) {
+ printk(KERN_WARNING "at91-i2s: request_irq failure\n");
+
+ DBG("Stopping pid %d clock\n", ssc_p->ssc.pid);
+ at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid);
+ return ret;
+ }
+
+ ssc_p->initialized = 1;
+ }
+
+ /* set SSC clock mode register */
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->cmr_div);
+
+ /* set receive clock mode and format */
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, rcmr);
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, rfmr);
+
+ /* set transmit clock mode and format */
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, tcmr);
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, tfmr);
+
+ DBG("hw_params: SSC initialized\n");
+ return 0;
+}
+
+
+static int at91_i2s_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+ struct at91_pcm_dma_params *dma_params;
+ int dir;
+
+ dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+ dma_params = ssc_p->dma_params[dir];
+
+ at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR,
+ dma_params->mask->ssc_enable);
+
+ DBG("%s enabled SSC_SR=0x%08lx\n", dir ? "receive" : "transmit",
+ at91_ssc_read(dma_params->ssc_base + AT91_SSC_SR));
+ return 0;
+}
+
+
+#ifdef CONFIG_PM
+static int at91_i2s_suspend(struct platform_device *pdev,
+ struct snd_soc_cpu_dai *cpu_dai)
+{
+ struct at91_ssc_info *ssc_p;
+
+ if(!cpu_dai->active)
+ return 0;
+
+ ssc_p = &ssc_info[cpu_dai->id];
+
+ /* Save the status register before disabling transmit and receive. */
+ ssc_p->ssc_state.ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR);
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR,
+ AT91_SSC_TXDIS | AT91_SSC_RXDIS);
+
+ /* Save the current interrupt mask, then disable unmasked interrupts. */
+ ssc_p->ssc_state.ssc_imr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR);
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IDR, ssc_p->ssc_state.ssc_imr);
+
+ ssc_p->ssc_state.ssc_cmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_CMR);
+ ssc_p->ssc_state.ssc_rcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RCMR);
+ ssc_p->ssc_state.ssc_rfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RFMR);
+ ssc_p->ssc_state.ssc_tcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TCMR);
+ ssc_p->ssc_state.ssc_tfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TFMR);
+
+ return 0;
+}
+
+static int at91_i2s_resume(struct platform_device *pdev,
+ struct snd_soc_cpu_dai *cpu_dai)
+{
+ struct at91_ssc_info *ssc_p;
+
+ if(!cpu_dai->active)
+ return 0;
+
+ ssc_p = &ssc_info[cpu_dai->id];
+
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, ssc_p->ssc_state.ssc_tfmr);
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, ssc_p->ssc_state.ssc_tcmr);
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, ssc_p->ssc_state.ssc_rfmr);
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, ssc_p->ssc_state.ssc_rcmr);
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->ssc_state.ssc_cmr);
+
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IER, ssc_p->ssc_state.ssc_imr);
+
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR,
+ ((ssc_p->ssc_state.ssc_sr & AT91_SSC_RXENA) ? AT91_SSC_RXEN : 0) |
+ ((ssc_p->ssc_state.ssc_sr & AT91_SSC_TXENA) ? AT91_SSC_TXEN : 0));
+
+ return 0;
+}
+
+#else
+#define at91_i2s_suspend NULL
+#define at91_i2s_resume NULL
+#endif
+
+#define AT91_I2S_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 | SNDRV_PCM_RATE_88200 |\
+ SNDRV_PCM_RATE_96000)
+
+struct snd_soc_cpu_dai at91_i2s_dai[NUM_SSC_DEVICES] = {
+ { .name = "at91_ssc0/i2s",
+ .id = 0,
+ .type = SND_SOC_DAI_I2S,
+ .suspend = at91_i2s_suspend,
+ .resume = at91_i2s_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT91_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT91_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .startup = at91_i2s_startup,
+ .shutdown = at91_i2s_shutdown,
+ .prepare = at91_i2s_prepare,
+ .hw_params = at91_i2s_hw_params,},
+ .dai_ops = {
+ .set_sysclk = at91_i2s_set_dai_sysclk,
+ .set_fmt = at91_i2s_set_dai_fmt,
+ .set_clkdiv = at91_i2s_set_dai_clkdiv,},
+ .private_data = &ssc_info[0].ssc,
+ },
+#if NUM_SSC_DEVICES == 3
+ { .name = "at91_ssc1/i2s",
+ .id = 1,
+ .type = SND_SOC_DAI_I2S,
+ .suspend = at91_i2s_suspend,
+ .resume = at91_i2s_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT91_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT91_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .startup = at91_i2s_startup,
+ .shutdown = at91_i2s_shutdown,
+ .prepare = at91_i2s_prepare,
+ .hw_params = at91_i2s_hw_params,},
+ .dai_ops = {
+ .set_sysclk = at91_i2s_set_dai_sysclk,
+ .set_fmt = at91_i2s_set_dai_fmt,
+ .set_clkdiv = at91_i2s_set_dai_clkdiv,},
+ .private_data = &ssc_info[1].ssc,
+ },
+ { .name = "at91_ssc2/i2s",
+ .id = 2,
+ .type = SND_SOC_DAI_I2S,
+ .suspend = at91_i2s_suspend,
+ .resume = at91_i2s_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT91_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT91_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .startup = at91_i2s_startup,
+ .shutdown = at91_i2s_shutdown,
+ .prepare = at91_i2s_prepare,
+ .hw_params = at91_i2s_hw_params,},
+ .dai_ops = {
+ .set_sysclk = at91_i2s_set_dai_sysclk,
+ .set_fmt = at91_i2s_set_dai_fmt,
+ .set_clkdiv = at91_i2s_set_dai_clkdiv,},
+ .private_data = &ssc_info[2].ssc,
+ },
+#endif
+};
+
+EXPORT_SYMBOL_GPL(at91_i2s_dai);
+
+/* Module information */
+MODULE_AUTHOR("Frank Mandarino, fmandarino@endrelia.com, www.endrelia.com");
+MODULE_DESCRIPTION("AT91 I2S ASoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/at91/at91-i2s.h b/sound/soc/at91/at91-i2s.h
new file mode 100644
index 00000000000..f8a875ba0cc
--- /dev/null
+++ b/sound/soc/at91/at91-i2s.h
@@ -0,0 +1,27 @@
+/*
+ * at91-i2s.h - ALSA I2S interface for the Atmel AT91 SoC
+ *
+ * Author: Frank Mandarino <fmandarino@endrelia.com>
+ * Endrelia Technologies Inc.
+ * Created: Jan 9, 2007
+ *
+ * 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 _AT91_I2S_H
+#define _AT91_I2S_H
+
+/* I2S system clock ids */
+#define AT91_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */
+
+/* I2S divider ids */
+#define AT91SSC_CMR_DIV 0 /* MCK divider for BCLK */
+#define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */
+#define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */
+
+extern struct snd_soc_cpu_dai at91_i2s_dai[];
+
+#endif /* _AT91_I2S_H */
+
diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c
new file mode 100644
index 00000000000..e88b12e7cc4
--- /dev/null
+++ b/sound/soc/at91/at91-pcm.c
@@ -0,0 +1,432 @@
+/*
+ * at91-pcm.c -- ALSA PCM interface for the Atmel AT91 SoC
+ *
+ * Author: Frank Mandarino <fmandarino@endrelia.com>
+ * Endrelia Technologies Inc.
+ * Created: Mar 3, 2006
+ *
+ * Based on pxa2xx-pcm.c by:
+ *
+ * Author: Nicolas Pitre
+ * Created: Nov 30, 2004
+ * Copyright: (C) 2004 MontaVista Software, Inc.
+ *
+ * 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 <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/arch/hardware.h>
+#include <asm/arch/at91_ssc.h>
+#include <asm/arch/at91_pdc.h>
+
+#include "at91-pcm.h"
+
+#if 0
+#define DBG(x...) printk(KERN_INFO "at91-pcm: " x)
+#else
+#define DBG(x...)
+#endif
+
+static const struct snd_pcm_hardware at91_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8192,
+ .periods_min = 2,
+ .periods_max = 1024,
+ .buffer_bytes_max = 32 * 1024,
+};
+
+struct at91_runtime_data {
+ struct at91_pcm_dma_params *params;
+ dma_addr_t dma_buffer; /* physical address of dma buffer */
+ dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */
+ size_t period_size;
+ dma_addr_t period_ptr; /* physical address of next period */
+ u32 pdc_xpr_save; /* PDC register save */
+ u32 pdc_xcr_save;
+ u32 pdc_xnpr_save;
+ u32 pdc_xncr_save;
+};
+
+static void at91_pcm_dma_irq(u32 ssc_sr,
+ struct snd_pcm_substream *substream)
+{
+ struct at91_runtime_data *prtd = substream->runtime->private_data;
+ struct at91_pcm_dma_params *params = prtd->params;
+ static int count = 0;
+
+ count++;
+
+ if (ssc_sr & params->mask->ssc_endbuf) {
+
+ printk(KERN_WARNING
+ "at91-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n",
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK
+ ? "underrun" : "overrun",
+ params->name, ssc_sr, count);
+
+ /* re-start the PDC */
+ at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_disable);
+
+ prtd->period_ptr += prtd->period_size;
+ if (prtd->period_ptr >= prtd->dma_buffer_end) {
+ prtd->period_ptr = prtd->dma_buffer;
+ }
+
+ at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr);
+ at91_ssc_write(params->ssc_base + params->pdc->xcr,
+ prtd->period_size / params->pdc_xfer_size);
+
+ at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_enable);
+ }
+
+ if (ssc_sr & params->mask->ssc_endx) {
+
+ /* Load the PDC next pointer and counter registers */
+ prtd->period_ptr += prtd->period_size;
+ if (prtd->period_ptr >= prtd->dma_buffer_end) {
+ prtd->period_ptr = prtd->dma_buffer;
+ }
+ at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr);
+ at91_ssc_write(params->ssc_base + params->pdc->xncr,
+ prtd->period_size / params->pdc_xfer_size);
+ }
+
+ snd_pcm_period_elapsed(substream);
+}
+
+static int at91_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct at91_runtime_data *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ /* this may get called several times by oss emulation
+ * with different params */
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ runtime->dma_bytes = params_buffer_bytes(params);
+
+ prtd->params = rtd->dai->cpu_dai->dma_data;
+ prtd->params->dma_intr_handler = at91_pcm_dma_irq;
+
+ prtd->dma_buffer = runtime->dma_addr;
+ prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes;
+ prtd->period_size = params_period_bytes(params);
+
+ DBG("hw_params: DMA for %s initialized (dma_bytes=%d, period_size=%d)\n",
+ prtd->params->name, runtime->dma_bytes, prtd->period_size);
+ return 0;
+}
+
+static int at91_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct at91_runtime_data *prtd = substream->runtime->private_data;
+ struct at91_pcm_dma_params *params = prtd->params;
+
+ if (params != NULL) {
+ at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_disable);
+ prtd->params->dma_intr_handler = NULL;
+ }
+
+ return 0;
+}
+
+static int at91_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct at91_runtime_data *prtd = substream->runtime->private_data;
+ struct at91_pcm_dma_params *params = prtd->params;
+
+ at91_ssc_write(params->ssc_base + AT91_SSC_IDR,
+ params->mask->ssc_endx | params->mask->ssc_endbuf);
+
+ at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_disable);
+ return 0;
+}
+
+static int at91_pcm_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct at91_runtime_data *prtd = substream->runtime->private_data;
+ struct at91_pcm_dma_params *params = prtd->params;
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ prtd->period_ptr = prtd->dma_buffer;
+
+ at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr);
+ at91_ssc_write(params->ssc_base + params->pdc->xcr,
+ prtd->period_size / params->pdc_xfer_size);
+
+ prtd->period_ptr += prtd->period_size;
+ at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr);
+ at91_ssc_write(params->ssc_base + params->pdc->xncr,
+ prtd->period_size / params->pdc_xfer_size);
+
+ DBG("trigger: period_ptr=%lx, xpr=%lx, xcr=%ld, xnpr=%lx, xncr=%ld\n",
+ (unsigned long) prtd->period_ptr,
+ at91_ssc_read(params->ssc_base + params->pdc->xpr),
+ at91_ssc_read(params->ssc_base + params->pdc->xcr),
+ at91_ssc_read(params->ssc_base + params->pdc->xnpr),
+ at91_ssc_read(params->ssc_base + params->pdc->xncr));
+
+ at91_ssc_write(params->ssc_base + AT91_SSC_IER,
+ params->mask->ssc_endx | params->mask->ssc_endbuf);
+
+ at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_enable);
+
+ DBG("sr=%lx imr=%lx\n", at91_ssc_read(params->ssc_base + AT91_SSC_SR),
+ at91_ssc_read(params->ssc_base + AT91_SSC_IER));
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_disable);
+ break;
+
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_enable);
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static snd_pcm_uframes_t at91_pcm_pointer(
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct at91_runtime_data *prtd = runtime->private_data;
+ struct at91_pcm_dma_params *params = prtd->params;
+ dma_addr_t ptr;
+ snd_pcm_uframes_t x;
+
+ ptr = (dma_addr_t) at91_ssc_read(params->ssc_base + params->pdc->xpr);
+ x = bytes_to_frames(runtime, ptr - prtd->dma_buffer);
+
+ if (x == runtime->buffer_size)
+ x = 0;
+ return x;
+}
+
+static int at91_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct at91_runtime_data *prtd;
+ int ret = 0;
+
+ snd_soc_set_runtime_hwparams(substream, &at91_pcm_hardware);
+
+ /* 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)
+ goto out;
+
+ prtd = kzalloc(sizeof(struct at91_runtime_data), GFP_KERNEL);
+ if (prtd == NULL) {
+ ret = -ENOMEM;
+ goto out;
+ }
+ runtime->private_data = prtd;
+
+ out:
+ return ret;
+}
+
+static int at91_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct at91_runtime_data *prtd = substream->runtime->private_data;
+
+ kfree(prtd);
+ return 0;
+}
+
+static int at91_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);
+}
+
+struct snd_pcm_ops at91_pcm_ops = {
+ .open = at91_pcm_open,
+ .close = at91_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = at91_pcm_hw_params,
+ .hw_free = at91_pcm_hw_free,
+ .prepare = at91_pcm_prepare,
+ .trigger = at91_pcm_trigger,
+ .pointer = at91_pcm_pointer,
+ .mmap = at91_pcm_mmap,
+};
+
+static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
+ int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = at91_pcm_hardware.buffer_bytes_max;
+
+ 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);
+
+ DBG("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 u64 at91_pcm_dmamask = 0xffffffff;
+
+static int at91_pcm_new(struct snd_card *card,
+ struct snd_soc_codec_dai *dai, struct snd_pcm *pcm)
+{
+ int ret = 0;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &at91_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = 0xffffffff;
+
+ if (dai->playback.channels_min) {
+ ret = at91_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+ }
+
+ if (dai->capture.channels_min) {
+ ret = at91_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ }
+ out:
+ return ret;
+}
+
+static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ 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;
+ }
+}
+
+#ifdef CONFIG_PM
+static int at91_pcm_suspend(struct platform_device *pdev,
+ struct snd_soc_cpu_dai *dai)
+{
+ struct snd_pcm_runtime *runtime = dai->runtime;
+ struct at91_runtime_data *prtd;
+ struct at91_pcm_dma_params *params;
+
+ if (!runtime)
+ return 0;
+
+ prtd = runtime->private_data;
+ params = prtd->params;
+
+ /* disable the PDC and save the PDC registers */
+
+ at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_disable);
+
+ prtd->pdc_xpr_save = at91_ssc_read(params->ssc_base + params->pdc->xpr);
+ prtd->pdc_xcr_save = at91_ssc_read(params->ssc_base + params->pdc->xcr);
+ prtd->pdc_xnpr_save = at91_ssc_read(params->ssc_base + params->pdc->xnpr);
+ prtd->pdc_xncr_save = at91_ssc_read(params->ssc_base + params->pdc->xncr);
+
+ return 0;
+}
+
+static int at91_pcm_resume(struct platform_device *pdev,
+ struct snd_soc_cpu_dai *dai)
+{
+ struct snd_pcm_runtime *runtime = dai->runtime;
+ struct at91_runtime_data *prtd;
+ struct at91_pcm_dma_params *params;
+
+ if (!runtime)
+ return 0;
+
+ prtd = runtime->private_data;
+ params = prtd->params;
+
+ /* restore the PDC registers and enable the PDC */
+ at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->pdc_xpr_save);
+ at91_ssc_write(params->ssc_base + params->pdc->xcr, prtd->pdc_xcr_save);
+ at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->pdc_xnpr_save);
+ at91_ssc_write(params->ssc_base + params->pdc->xncr, prtd->pdc_xncr_save);
+
+ at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_enable);
+ return 0;
+}
+#else
+#define at91_pcm_suspend NULL
+#define at91_pcm_resume NULL
+#endif
+
+struct snd_soc_platform at91_soc_platform = {
+ .name = "at91-audio",
+ .pcm_ops = &at91_pcm_ops,
+ .pcm_new = at91_pcm_new,
+ .pcm_free = at91_pcm_free_dma_buffers,
+ .suspend = at91_pcm_suspend,
+ .resume = at91_pcm_resume,
+};
+
+EXPORT_SYMBOL_GPL(at91_soc_platform);
+
+MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>");
+MODULE_DESCRIPTION("Atmel AT91 PCM module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h
new file mode 100644
index 00000000000..58d0f00a07b
--- /dev/null
+++ b/sound/soc/at91/at91-pcm.h
@@ -0,0 +1,72 @@
+/*
+ * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC
+ *
+ * Author: Frank Mandarino <fmandarino@endrelia.com>
+ * Endrelia Technologies Inc.
+ * Created: Mar 3, 2006
+ *
+ * Based on pxa2xx-pcm.h by:
+ *
+ * Author: Nicolas Pitre
+ * Created: Nov 30, 2004
+ * Copyright: MontaVista Software, Inc.
+ *
+ * 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 _AT91_PCM_H
+#define _AT91_PCM_H
+
+#include <asm/arch/hardware.h>
+
+struct at91_ssc_periph {
+ void __iomem *base;
+ u32 pid;
+};
+
+/*
+ * Registers and status bits that are required by the PCM driver.
+ */
+struct at91_pdc_regs {
+ unsigned int xpr; /* PDC recv/trans pointer */
+ unsigned int xcr; /* PDC recv/trans counter */
+ unsigned int xnpr; /* PDC next recv/trans pointer */
+ unsigned int xncr; /* PDC next recv/trans counter */
+ unsigned int ptcr; /* PDC transfer control */
+};
+
+struct at91_ssc_mask {
+ u32 ssc_enable; /* SSC recv/trans enable */
+ u32 ssc_disable; /* SSC recv/trans disable */
+ u32 ssc_endx; /* SSC ENDTX or ENDRX */
+ u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */
+ u32 pdc_enable; /* PDC recv/trans enable */
+ u32 pdc_disable; /* PDC recv/trans disable */
+};
+
+/*
+ * This structure, shared between the PCM driver and the interface,
+ * contains all information required by the PCM driver to perform the
+ * PDC DMA operation. All fields except dma_intr_handler() are initialized
+ * by the interface. The dms_intr_handler() pointer is set by the PCM
+ * driver and called by the interface SSC interrupt handler if it is
+ * non-NULL.
+ */
+struct at91_pcm_dma_params {
+ char *name; /* stream identifier */
+ int pdc_xfer_size; /* PDC counter increment in bytes */
+ void __iomem *ssc_base; /* SSC base address */
+ struct at91_pdc_regs *pdc; /* PDC receive or transmit registers */
+ struct at91_ssc_mask *mask;/* SSC & PDC status bits */
+ struct snd_pcm_substream *substream;
+ void (*dma_intr_handler)(u32, struct snd_pcm_substream *);
+};
+
+extern struct snd_soc_platform at91_soc_platform;
+
+#define at91_ssc_read(a) ((unsigned long) __raw_readl(a))
+#define at91_ssc_write(a,v) __raw_writel((v),(a))
+
+#endif /* _AT91_PCM_H */
diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c
new file mode 100644
index 00000000000..8179df3bb2f
--- /dev/null
+++ b/sound/soc/at91/eti_b1_wm8731.c
@@ -0,0 +1,375 @@
+/*
+ * eti_b1_wm8731 -- SoC audio for AT91RM9200-based Endrelia ETI_B1 board.
+ *
+ * Author: Frank Mandarino <fmandarino@endrelia.com>
+ * Endrelia Technologies Inc.
+ * Created: Mar 29, 2006
+ *
+ * Based on corgi.c by:
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Richard Purdie <richard@openedhand.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/clk.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/arch/hardware.h>
+#include <asm/arch/at91_pio.h>
+#include <asm/arch/gpio.h>
+
+#include "../codecs/wm8731.h"
+#include "at91-pcm.h"
+#include "at91-i2s.h"
+
+#if 0
+#define DBG(x...) printk(KERN_INFO "eti_b1_wm8731: " x)
+#else
+#define DBG(x...)
+#endif
+
+#define AT91_PIO_TF1 (1 << (AT91_PIN_PB6 - PIN_BASE) % 32)
+#define AT91_PIO_TK1 (1 << (AT91_PIN_PB7 - PIN_BASE) % 32)
+#define AT91_PIO_TD1 (1 << (AT91_PIN_PB8 - PIN_BASE) % 32)
+#define AT91_PIO_RD1 (1 << (AT91_PIN_PB9 - PIN_BASE) % 32)
+#define AT91_PIO_RK1 (1 << (AT91_PIN_PB10 - PIN_BASE) % 32)
+#define AT91_PIO_RF1 (1 << (AT91_PIN_PB11 - PIN_BASE) % 32)
+
+static struct clk *pck1_clk;
+static struct clk *pllb_clk;
+
+
+static int eti_b1_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ /* cpu clock is the AT91 master clock sent to the SSC */
+ ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, AT91_SYSCLK_MCK,
+ 60000000, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* codec system clock is supplied by PCK1, set to 12MHz */
+ ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK,
+ 12000000, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* Start PCK1 clock. */
+ clk_enable(pck1_clk);
+ DBG("pck1 started\n");
+
+ return 0;
+}
+
+static void eti_b1_shutdown(struct snd_pcm_substream *substream)
+{
+ /* Stop PCK1 clock. */
+ clk_disable(pck1_clk);
+ DBG("pck1 stopped\n");
+}
+
+static int eti_b1_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_codec_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE
+ unsigned int rate;
+ int cmr_div, period;
+
+ /* set codec DAI configuration */
+ ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /*
+ * The SSC clock dividers depend on the sample rate. The CMR.DIV
+ * field divides the system master clock MCK to drive the SSC TK
+ * signal which provides the codec BCLK. The TCMR.PERIOD and
+ * RCMR.PERIOD fields further divide the BCLK signal to drive
+ * the SSC TF and RF signals which provide the codec DACLRC and
+ * ADCLRC clocks.
+ *
+ * The dividers were determined through trial and error, where a
+ * CMR.DIV value is chosen such that the resulting BCLK value is
+ * divisible, or almost divisible, by (2 * sample rate), and then
+ * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1.
+ */
+ rate = params_rate(params);
+
+ switch (rate) {
+ case 8000:
+ cmr_div = 25; /* BCLK = 60MHz/(2*25) = 1.2MHz */
+ period = 74; /* LRC = BCLK/(2*(74+1)) = 8000Hz */
+ break;
+ case 32000:
+ cmr_div = 7; /* BCLK = 60MHz/(2*7) ~= 4.28571428MHz */
+ period = 66; /* LRC = BCLK/(2*(66+1)) = 31982.942Hz */
+ break;
+ case 48000:
+ cmr_div = 13; /* BCLK = 60MHz/(2*13) ~= 2.3076923MHz */
+ period = 23; /* LRC = BCLK/(2*(23+1)) = 48076.923Hz */
+ break;
+ default:
+ printk(KERN_WARNING "unsupported rate %d on ETI-B1 board\n", rate);
+ return -EINVAL;
+ }
+
+ /* set the MCK divider for BCLK */
+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div);
+ if (ret < 0)
+ return ret;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* set the BCLK divider for DACLRC */
+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,
+ AT91SSC_TCMR_PERIOD, period);
+ } else {
+ /* set the BCLK divider for ADCLRC */
+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,
+ AT91SSC_RCMR_PERIOD, period);
+ }
+ if (ret < 0)
+ return ret;
+
+#else /* CONFIG_SND_AT91_SOC_ETI_SLAVE */
+ /*
+ * Codec in Master Mode.
+ */
+
+ /* set codec DAI configuration */
+ ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+#endif /* CONFIG_SND_AT91_SOC_ETI_SLAVE */
+
+ return 0;
+}
+
+static struct snd_soc_ops eti_b1_ops = {
+ .startup = eti_b1_startup,
+ .hw_params = eti_b1_hw_params,
+ .shutdown = eti_b1_shutdown,
+};
+
+
+static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = {
+ SND_SOC_DAPM_MIC("Int Mic", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static const char *intercon[][3] = {
+
+ /* speaker connected to LHPOUT */
+ {"Ext Spk", NULL, "LHPOUT"},
+
+ /* mic is connected to Mic Jack, with WM8731 Mic Bias */
+ {"MICIN", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Int Mic"},
+
+ /* terminator */
+ {NULL, NULL, NULL},
+};
+
+/*
+ * Logic for a wm8731 as connected on a Endrelia ETI-B1 board.
+ */
+static int eti_b1_wm8731_init(struct snd_soc_codec *codec)
+{
+ int i;
+
+ DBG("eti_b1_wm8731_init() called\n");
+
+ /* Add specific widgets */
+ for(i = 0; i < ARRAY_SIZE(eti_b1_dapm_widgets); i++) {
+ snd_soc_dapm_new_control(codec, &eti_b1_dapm_widgets[i]);
+ }
+
+ /* Set up specific audio path interconnects */
+ for(i = 0; intercon[i][0] != NULL; i++) {
+ snd_soc_dapm_connect_input(codec, intercon[i][0],
+ intercon[i][1], intercon[i][2]);
+ }
+
+ /* not connected */
+ snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
+ snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
+
+ /* always connected */
+ snd_soc_dapm_set_endpoint(codec, "Int Mic", 1);
+ snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1);
+
+ snd_soc_dapm_sync_endpoints(codec);
+
+ return 0;
+}
+
+static struct snd_soc_dai_link eti_b1_dai = {
+ .name = "WM8731",
+ .stream_name = "WM8731",
+ .cpu_dai = &at91_i2s_dai[1],
+ .codec_dai = &wm8731_dai,
+ .init = eti_b1_wm8731_init,
+ .ops = &eti_b1_ops,
+};
+
+static struct snd_soc_machine snd_soc_machine_eti_b1 = {
+ .name = "ETI_B1",
+ .dai_link = &eti_b1_dai,
+ .num_links = 1,
+};
+
+static struct wm8731_setup_data eti_b1_wm8731_setup = {
+ .i2c_address = 0x1a,
+};
+
+static struct snd_soc_device eti_b1_snd_devdata = {
+ .machine = &snd_soc_machine_eti_b1,
+ .platform = &at91_soc_platform,
+ .codec_dev = &soc_codec_dev_wm8731,
+ .codec_data = &eti_b1_wm8731_setup,
+};
+
+static struct platform_device *eti_b1_snd_device;
+
+static int __init eti_b1_init(void)
+{
+ int ret;
+ u32 ssc_pio_lines;
+ struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
+
+ if (!request_mem_region(AT91RM9200_BASE_SSC1, SZ_16K, "soc-audio")) {
+ DBG("SSC1 memory region is busy\n");
+ return -EBUSY;
+ }
+
+ ssc->base = ioremap(AT91RM9200_BASE_SSC1, SZ_16K);
+ if (!ssc->base) {
+ DBG("SSC1 memory ioremap failed\n");
+ ret = -ENOMEM;
+ goto fail_release_mem;
+ }
+
+ ssc->pid = AT91RM9200_ID_SSC1;
+
+ eti_b1_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!eti_b1_snd_device) {
+ DBG("platform device allocation failed\n");
+ ret = -ENOMEM;
+ goto fail_io_unmap;
+ }
+
+ platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata);
+ eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev;
+
+ ret = platform_device_add(eti_b1_snd_device);
+ if (ret) {
+ DBG("platform device add failed\n");
+ platform_device_put(eti_b1_snd_device);
+ goto fail_io_unmap;
+ }
+
+ ssc_pio_lines = AT91_PIO_TF1 | AT91_PIO_TK1 | AT91_PIO_TD1
+ | AT91_PIO_RD1 /* | AT91_PIO_RK1 */ | AT91_PIO_RF1;
+
+ /* Reset all PIO registers and assign lines to peripheral A */
+ at91_sys_write(AT91_PIOB + PIO_PDR, ssc_pio_lines);
+ at91_sys_write(AT91_PIOB + PIO_ODR, ssc_pio_lines);
+ at91_sys_write(AT91_PIOB + PIO_IFDR, ssc_pio_lines);
+ at91_sys_write(AT91_PIOB + PIO_CODR, ssc_pio_lines);
+ at91_sys_write(AT91_PIOB + PIO_IDR, ssc_pio_lines);
+ at91_sys_write(AT91_PIOB + PIO_MDDR, ssc_pio_lines);
+ at91_sys_write(AT91_PIOB + PIO_PUDR, ssc_pio_lines);
+ at91_sys_write(AT91_PIOB + PIO_ASR, ssc_pio_lines);
+ at91_sys_write(AT91_PIOB + PIO_OWDR, ssc_pio_lines);
+
+ /*
+ * Set PCK1 parent to PLLB and its rate to 12 Mhz.
+ */
+ pllb_clk = clk_get(NULL, "pllb");
+ pck1_clk = clk_get(NULL, "pck1");
+
+ clk_set_parent(pck1_clk, pllb_clk);
+ clk_set_rate(pck1_clk, 12000000);
+
+ DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk));
+
+ /* assign the GPIO pin to PCK1 */
+ at91_set_B_periph(AT91_PIN_PA24, 0);
+
+#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE
+ printk(KERN_INFO "eti_b1_wm8731: Codec in Slave Mode\n");
+#else
+ printk(KERN_INFO "eti_b1_wm8731: Codec in Master Mode\n");
+#endif
+ return ret;
+
+fail_io_unmap:
+ iounmap(ssc->base);
+fail_release_mem:
+ release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K);
+ return ret;
+}
+
+static void __exit eti_b1_exit(void)
+{
+ struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
+
+ clk_put(pck1_clk);
+ clk_put(pllb_clk);
+
+ platform_device_unregister(eti_b1_snd_device);
+
+ iounmap(ssc->base);
+ release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K);
+}
+
+module_init(eti_b1_init);
+module_exit(eti_b1_exit);
+
+/* Module information */
+MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>");
+MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
new file mode 100644
index 00000000000..78ac2688e12
--- /dev/null
+++ b/sound/soc/codecs/Kconfig
@@ -0,0 +1,15 @@
+config SND_SOC_AC97_CODEC
+ tristate
+ depends SND_SOC
+
+config SND_SOC_WM8731
+ tristate
+ depends SND_SOC
+
+config SND_SOC_WM8750
+ tristate
+ depends SND_SOC
+
+config SND_SOC_WM9712
+ tristate
+ depends SND_SOC
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
new file mode 100644
index 00000000000..3249a6e4f1d
--- /dev/null
+++ b/sound/soc/codecs/Makefile
@@ -0,0 +1,9 @@
+snd-soc-ac97-objs := ac97.o
+snd-soc-wm8731-objs := wm8731.o
+snd-soc-wm8750-objs := wm8750.o
+snd-soc-wm9712-objs := wm9712.o
+
+obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
+obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o
+obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o
+obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
new file mode 100644
index 00000000000..55bc55eb6e2
--- /dev/null
+++ b/sound/soc/codecs/ac97.c
@@ -0,0 +1,156 @@
+/*
+ * ac97.c -- ALSA Soc AC97 codec support
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.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.
+ *
+ * Revision history
+ * 17th Oct 2005 Initial version.
+ *
+ * Generic AC97 support.
+ */
+
+#include <linux/init.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#define AC97_VERSION "0.6"
+
+static int ac97_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+
+ int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE;
+ return snd_ac97_set_rate(codec->ac97, reg, runtime->rate);
+}
+
+#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+static struct snd_soc_codec_dai ac97_dai = {
+ .name = "AC97 HiFi",
+ .playback = {
+ .stream_name = "AC97 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = STD_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "AC97 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = STD_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .prepare = ac97_prepare,},
+};
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ return soc_ac97_ops.read(codec->ac97, reg);
+}
+
+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int val)
+{
+ soc_ac97_ops.write(codec->ac97, reg, val);
+ return 0;
+}
+
+static int ac97_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ struct snd_ac97_bus *ac97_bus;
+ struct snd_ac97_template ac97_template;
+ int ret = 0;
+
+ printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION);
+
+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (socdev->codec == NULL)
+ return -ENOMEM;
+ codec = socdev->codec;
+ mutex_init(&codec->mutex);
+
+ codec->name = "AC97";
+ codec->owner = THIS_MODULE;
+ codec->dai = &ac97_dai;
+ codec->num_dai = 1;
+ codec->write = ac97_write;
+ codec->read = ac97_read;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if(ret < 0)
+ goto err;
+
+ /* add codec as bus device for standard ac97 */
+ ret = snd_ac97_bus(codec->card, 0, &soc_ac97_ops, NULL, &ac97_bus);
+ if(ret < 0)
+ goto bus_err;
+
+ memset(&ac97_template, 0, sizeof(struct snd_ac97_template));
+ ret = snd_ac97_mixer(ac97_bus, &ac97_template, &codec->ac97);
+ if(ret < 0)
+ goto bus_err;
+
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0)
+ goto bus_err;
+ return 0;
+
+bus_err:
+ snd_soc_free_pcms(socdev);
+
+err:
+ kfree(socdev->codec->reg_cache);
+ kfree(socdev->codec);
+ socdev->codec = NULL;
+ return ret;
+}
+
+static int ac97_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if(codec == NULL)
+ return 0;
+
+ snd_soc_free_pcms(socdev);
+ kfree(socdev->codec->reg_cache);
+ kfree(socdev->codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ac97= {
+ .probe = ac97_soc_probe,
+ .remove = ac97_soc_remove,
+};
+
+EXPORT_SYMBOL_GPL(soc_codec_dev_ac97);
+
+MODULE_DESCRIPTION("Soc Generic AC97 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ac97.h b/sound/soc/codecs/ac97.h
new file mode 100644
index 00000000000..930ddfc2321
--- /dev/null
+++ b/sound/soc/codecs/ac97.h
@@ -0,0 +1,18 @@
+/*
+ * linux/sound/codecs/ac97.h -- ALSA SoC Layer
+ *
+ * Author: Liam Girdwood
+ * Created: Dec 1st 2005
+ * Copyright: Wolfson Microelectronics. PLC.
+ *
+ * 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 __LINUX_SND_SOC_AC97_H
+#define __LINUX_SND_SOC_AC97_H
+
+extern struct snd_soc_codec_device soc_codec_dev_ac97;
+
+#endif
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
new file mode 100644
index 00000000000..7ca0b526828
--- /dev/null
+++ b/sound/soc/codecs/wm8731.c
@@ -0,0 +1,758 @@
+/*
+ * wm8731.c -- WM8731 ALSA SoC Audio driver
+ *
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on wm8753.c by Liam Girdwood
+ *
+ * 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/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "wm8731.h"
+
+#define AUDIO_NAME "wm8731"
+#define WM8731_VERSION "0.13"
+
+/*
+ * Debug
+ */
+
+#define WM8731_DEBUG 0
+
+#ifdef WM8731_DEBUG
+#define dbg(format, arg...) \
+ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
+#else
+#define dbg(format, arg...) do {} while (0)
+#endif
+#define err(format, arg...) \
+ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
+#define info(format, arg...) \
+ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
+#define warn(format, arg...) \
+ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
+
+struct snd_soc_codec_device soc_codec_dev_wm8731;
+
+/* codec private data */
+struct wm8731_priv {
+ unsigned int sysclk;
+};
+
+/*
+ * wm8731 register cache
+ * We can't read the WM8731 register space when we are
+ * using 2 wire for device control, so we cache them instead.
+ * There is no point in caching the reset register
+ */
+static const u16 wm8731_reg[WM8731_CACHEREGNUM] = {
+ 0x0097, 0x0097, 0x0079, 0x0079,
+ 0x000a, 0x0008, 0x009f, 0x000a,
+ 0x0000, 0x0000
+};
+
+/*
+ * read wm8731 register cache
+ */
+static inline unsigned int wm8731_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg == WM8731_RESET)
+ return 0;
+ if (reg >= WM8731_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write wm8731 register cache
+ */
+static inline void wm8731_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= WM8731_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the WM8731 register space
+ */
+static int wm8731_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D9 WM8731 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ wm8731_write_reg_cache (codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+#define wm8731_reset(c) wm8731_write(c, WM8731_RESET, 0)
+
+static const char *wm8731_input_select[] = {"Line In", "Mic"};
+static const char *wm8731_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+
+static const struct soc_enum wm8731_enum[] = {
+ SOC_ENUM_SINGLE(WM8731_APANA, 2, 2, wm8731_input_select),
+ SOC_ENUM_SINGLE(WM8731_APDIGI, 1, 4, wm8731_deemph),
+};
+
+static const struct snd_kcontrol_new wm8731_snd_controls[] = {
+
+SOC_DOUBLE_R("Master Playback Volume", WM8731_LOUT1V, WM8731_ROUT1V,
+ 0, 127, 0),
+SOC_DOUBLE_R("Master Playback ZC Switch", WM8731_LOUT1V, WM8731_ROUT1V,
+ 7, 1, 0),
+
+SOC_DOUBLE_R("Capture Volume", WM8731_LINVOL, WM8731_RINVOL, 0, 31, 0),
+SOC_DOUBLE_R("Line Capture Switch", WM8731_LINVOL, WM8731_RINVOL, 7, 1, 1),
+
+SOC_SINGLE("Mic Boost (+20dB)", WM8731_APANA, 0, 1, 0),
+SOC_SINGLE("Capture Mic Switch", WM8731_APANA, 1, 1, 1),
+
+SOC_SINGLE("Sidetone Playback Volume", WM8731_APANA, 6, 3, 1),
+
+SOC_SINGLE("ADC High Pass Filter Switch", WM8731_APDIGI, 0, 1, 1),
+SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0),
+
+SOC_ENUM("Playback De-emphasis", wm8731_enum[1]),
+};
+
+/* add non dapm controls */
+static int wm8731_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) {
+ if ((err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8731_snd_controls[i],codec, NULL))) < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/* Output Mixer */
+static const struct snd_kcontrol_new wm8731_output_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
+SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0),
+SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
+};
+
+/* Input mux */
+static const struct snd_kcontrol_new wm8731_input_mux_controls =
+SOC_DAPM_ENUM("Input Select", wm8731_enum[0]);
+
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1,
+ &wm8731_output_mixer_controls[0],
+ ARRAY_SIZE(wm8731_output_mixer_controls)),
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8731_PWR, 3, 1),
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("LHPOUT"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+SND_SOC_DAPM_OUTPUT("RHPOUT"),
+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8731_PWR, 2, 1),
+SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &wm8731_input_mux_controls),
+SND_SOC_DAPM_PGA("Line Input", WM8731_PWR, 0, 1, NULL, 0),
+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8731_PWR, 1, 1),
+SND_SOC_DAPM_INPUT("MICIN"),
+SND_SOC_DAPM_INPUT("RLINEIN"),
+SND_SOC_DAPM_INPUT("LLINEIN"),
+};
+
+static const char *intercon[][3] = {
+ /* output mixer */
+ {"Output Mixer", "Line Bypass Switch", "Line Input"},
+ {"Output Mixer", "HiFi Playback Switch", "DAC"},
+ {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
+
+ /* outputs */
+ {"RHPOUT", NULL, "Output Mixer"},
+ {"ROUT", NULL, "Output Mixer"},
+ {"LHPOUT", NULL, "Output Mixer"},
+ {"LOUT", NULL, "Output Mixer"},
+
+ /* input mux */
+ {"Input Mux", "Line In", "Line Input"},
+ {"Input Mux", "Mic", "Mic Bias"},
+ {"ADC", NULL, "Input Mux"},
+
+ /* inputs */
+ {"Line Input", NULL, "LLINEIN"},
+ {"Line Input", NULL, "RLINEIN"},
+ {"Mic Bias", NULL, "MICIN"},
+
+ /* terminator */
+ {NULL, NULL, NULL},
+};
+
+static int wm8731_add_widgets(struct snd_soc_codec *codec)
+{
+ int i;
+
+ for(i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) {
+ snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
+ }
+
+ /* set up audio path interconnects */
+ for(i = 0; intercon[i][0] != NULL; i++) {
+ snd_soc_dapm_connect_input(codec, intercon[i][0],
+ intercon[i][1], intercon[i][2]);
+ }
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+struct _coeff_div {
+ u32 mclk;
+ u32 rate;
+ u16 fs;
+ u8 sr:4;
+ u8 bosr:1;
+ u8 usb:1;
+};
+
+/* codec mclk clock divider coefficients */
+static const struct _coeff_div coeff_div[] = {
+ /* 48k */
+ {12288000, 48000, 256, 0x0, 0x0, 0x0},
+ {18432000, 48000, 384, 0x0, 0x1, 0x0},
+ {12000000, 48000, 250, 0x0, 0x0, 0x1},
+
+ /* 32k */
+ {12288000, 32000, 384, 0x6, 0x0, 0x0},
+ {18432000, 32000, 576, 0x6, 0x1, 0x0},
+ {12000000, 32000, 375, 0x6, 0x0, 0x1},
+
+ /* 8k */
+ {12288000, 8000, 1536, 0x3, 0x0, 0x0},
+ {18432000, 8000, 2304, 0x3, 0x1, 0x0},
+ {11289600, 8000, 1408, 0xb, 0x0, 0x0},
+ {16934400, 8000, 2112, 0xb, 0x1, 0x0},
+ {12000000, 8000, 1500, 0x3, 0x0, 0x1},
+
+ /* 96k */
+ {12288000, 96000, 128, 0x7, 0x0, 0x0},
+ {18432000, 96000, 192, 0x7, 0x1, 0x0},
+ {12000000, 96000, 125, 0x7, 0x0, 0x1},
+
+ /* 44.1k */
+ {11289600, 44100, 256, 0x8, 0x0, 0x0},
+ {16934400, 44100, 384, 0x8, 0x1, 0x0},
+ {12000000, 44100, 272, 0x8, 0x1, 0x1},
+
+ /* 88.2k */
+ {11289600, 88200, 128, 0xf, 0x0, 0x0},
+ {16934400, 88200, 192, 0xf, 0x1, 0x0},
+ {12000000, 88200, 136, 0xf, 0x1, 0x1},
+};
+
+static inline int get_coeff(int mclk, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+ return i;
+ }
+ return 0;
+}
+
+static int wm8731_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_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm8731_priv *wm8731 = codec->private_data;
+ u16 iface = wm8731_read_reg_cache(codec, WM8731_IFACE) & 0xfff3;
+ int i = get_coeff(wm8731->sysclk, params_rate(params));
+ u16 srate = (coeff_div[i].sr << 2) |
+ (coeff_div[i].bosr << 1) | coeff_div[i].usb;
+
+ wm8731_write(codec, WM8731_SRATE, srate);
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x0004;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x0008;
+ break;
+ }
+
+ wm8731_write(codec, WM8731_IFACE, iface);
+ return 0;
+}
+
+static int wm8731_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+
+ /* set active */
+ wm8731_write(codec, WM8731_ACTIVE, 0x0001);
+
+ return 0;
+}
+
+static void wm8731_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+
+ /* deactivate */
+ if (!codec->active) {
+ udelay(50);
+ wm8731_write(codec, WM8731_ACTIVE, 0x0);
+ }
+}
+
+static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7;
+
+ if (mute)
+ wm8731_write(codec, WM8731_APDIGI, mute_reg | 0x8);
+ else
+ wm8731_write(codec, WM8731_APDIGI, mute_reg);
+ return 0;
+}
+
+static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8731_priv *wm8731 = codec->private_data;
+
+ switch (freq) {
+ case 11289600:
+ case 12000000:
+ case 12288000:
+ case 16934400:
+ case 18432000:
+ wm8731->sysclk = freq;
+ return 0;
+ }
+ return -EINVAL;
+}
+
+
+static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = 0;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface |= 0x0040;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= 0x0013;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x0090;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x0010;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set iface */
+ wm8731_write(codec, WM8731_IFACE, iface);
+ return 0;
+}
+
+static int wm8731_dapm_event(struct snd_soc_codec *codec, int event)
+{
+ u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f;
+
+ switch (event) {
+ case SNDRV_CTL_POWER_D0: /* full On */
+ /* vref/mid, osc on, dac unmute */
+ wm8731_write(codec, WM8731_PWR, reg);
+ break;
+ case SNDRV_CTL_POWER_D1: /* partial On */
+ case SNDRV_CTL_POWER_D2: /* partial On */
+ break;
+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+ /* everything off except vref/vmid, */
+ wm8731_write(codec, WM8731_PWR, reg | 0x0040);
+ break;
+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+ /* everything off, dac mute, inactive */
+ wm8731_write(codec, WM8731_ACTIVE, 0x0);
+ wm8731_write(codec, WM8731_PWR, 0xffff);
+ break;
+ }
+ codec->dapm_state = event;
+ return 0;
+}
+
+#define WM8731_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 | SNDRV_PCM_RATE_88200 |\
+ SNDRV_PCM_RATE_96000)
+
+#define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_codec_dai wm8731_dai = {
+ .name = "WM8731",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8731_RATES,
+ .formats = WM8731_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8731_RATES,
+ .formats = WM8731_FORMATS,},
+ .ops = {
+ .prepare = wm8731_pcm_prepare,
+ .hw_params = wm8731_hw_params,
+ .shutdown = wm8731_shutdown,
+ },
+ .dai_ops = {
+ .digital_mute = wm8731_mute,
+ .set_sysclk = wm8731_set_dai_sysclk,
+ .set_fmt = wm8731_set_dai_fmt,
+ }
+};
+EXPORT_SYMBOL_GPL(wm8731_dai);
+
+static int wm8731_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm8731_write(codec, WM8731_ACTIVE, 0x0);
+ wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+ return 0;
+}
+
+static int wm8731_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) {
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+ wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+ wm8731_dapm_event(codec, codec->suspend_dapm_state);
+ return 0;
+}
+
+/*
+ * initialise the WM8731 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8731_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int reg, ret = 0;
+
+ codec->name = "WM8731";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8731_read_reg_cache;
+ codec->write = wm8731_write;
+ codec->dapm_event = wm8731_dapm_event;
+ codec->dai = &wm8731_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = sizeof(wm8731_reg);
+ codec->reg_cache = kmemdup(wm8731_reg, sizeof(wm8731_reg), GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ wm8731_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8731: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* power on device */
+ wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+
+ /* set the update bits */
+ reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V);
+ wm8731_write(codec, WM8731_LOUT1V, reg | 0x0100);
+ reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V);
+ wm8731_write(codec, WM8731_ROUT1V, reg | 0x0100);
+ reg = wm8731_read_reg_cache(codec, WM8731_LINVOL);
+ wm8731_write(codec, WM8731_LINVOL, reg | 0x0100);
+ reg = wm8731_read_reg_cache(codec, WM8731_RINVOL);
+ wm8731_write(codec, WM8731_RINVOL, reg | 0x0100);
+
+ wm8731_add_controls(codec);
+ wm8731_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8731: failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+static struct snd_soc_device *wm8731_socdev;
+
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+
+/*
+ * WM8731 2 wire address is determined by GPIO5
+ * state during powerup.
+ * low = 0x1a
+ * high = 0x1b
+ */
+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static struct i2c_driver wm8731_i2c_driver;
+static struct i2c_client client_template;
+
+/* If the i2c layer weren't so broken, we could pass this kind of data
+ around */
+
+static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+ struct snd_soc_device *socdev = wm8731_socdev;
+ struct wm8731_setup_data *setup = socdev->codec_data;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct i2c_client *i2c;
+ int ret;
+
+ if (addr != setup->i2c_address)
+ return -ENODEV;
+
+ client_template.adapter = adap;
+ client_template.addr = addr;
+
+ i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
+ if (i2c == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = i2c_attach_client(i2c);
+ if (ret < 0) {
+ err("failed to attach codec at addr %x\n", addr);
+ goto err;
+ }
+
+ ret = wm8731_init(socdev);
+ if (ret < 0) {
+ err("failed to initialise WM8731\n");
+ goto err;
+ }
+ return ret;
+
+err:
+ kfree(codec);
+ kfree(i2c);
+ return ret;
+}
+
+static int wm8731_i2c_detach(struct i2c_client *client)
+{
+ struct snd_soc_codec* codec = i2c_get_clientdata(client);
+ i2c_detach_client(client);
+ kfree(codec->reg_cache);
+ kfree(client);
+ return 0;
+}
+
+static int wm8731_i2c_attach(struct i2c_adapter *adap)
+{
+ return i2c_probe(adap, &addr_data, wm8731_codec_probe);
+}
+
+/* corgi i2c codec control layer */
+static struct i2c_driver wm8731_i2c_driver = {
+ .driver = {
+ .name = "WM8731 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .id = I2C_DRIVERID_WM8731,
+ .attach_adapter = wm8731_i2c_attach,
+ .detach_client = wm8731_i2c_detach,
+ .command = NULL,
+};
+
+static struct i2c_client client_template = {
+ .name = "WM8731",
+ .driver = &wm8731_i2c_driver,
+};
+#endif
+
+static int wm8731_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct wm8731_setup_data *setup;
+ struct snd_soc_codec *codec;
+ struct wm8731_priv *wm8731;
+ int ret = 0;
+
+ info("WM8731 Audio Codec %s", WM8731_VERSION);
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL);
+ if (wm8731 == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+
+ codec->private_data = wm8731;
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ wm8731_socdev = socdev;
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ normal_i2c[0] = setup->i2c_address;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = i2c_add_driver(&wm8731_i2c_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add i2c driver");
+ }
+#else
+ /* Add other interfaces here */
+#endif
+ return ret;
+}
+
+/* power down chip */
+static int wm8731_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+ i2c_del_driver(&wm8731_i2c_driver);
+#endif
+ kfree(codec->private_data);
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8731 = {
+ .probe = wm8731_probe,
+ .remove = wm8731_remove,
+ .suspend = wm8731_suspend,
+ .resume = wm8731_resume,
+};
+
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
+
+MODULE_DESCRIPTION("ASoC WM8731 driver");
+MODULE_AUTHOR("Richard Purdie");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h
new file mode 100644
index 00000000000..5bcab6a7afb
--- /dev/null
+++ b/sound/soc/codecs/wm8731.h
@@ -0,0 +1,44 @@
+/*
+ * wm8731.h -- WM8731 Soc Audio driver
+ *
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on wm8753.h
+ *
+ * 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 _WM8731_H
+#define _WM8731_H
+
+/* WM8731 register space */
+
+#define WM8731_LINVOL 0x00
+#define WM8731_RINVOL 0x01
+#define WM8731_LOUT1V 0x02
+#define WM8731_ROUT1V 0x03
+#define WM8731_APANA 0x04
+#define WM8731_APDIGI 0x05
+#define WM8731_PWR 0x06
+#define WM8731_IFACE 0x07
+#define WM8731_SRATE 0x08
+#define WM8731_ACTIVE 0x09
+#define WM8731_RESET 0x0f
+
+#define WM8731_CACHEREGNUM 10
+
+#define WM8731_SYSCLK 0
+#define WM8731_DAI 0
+
+struct wm8731_setup_data {
+ unsigned short i2c_address;
+};
+
+extern struct snd_soc_codec_dai wm8731_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8731;
+
+#endif
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
new file mode 100644
index 00000000000..7073e8e294f
--- /dev/null
+++ b/sound/soc/codecs/wm8750.c
@@ -0,0 +1,1049 @@
+/*
+ * wm8750.c -- WM8750 ALSA SoC audio driver
+ *
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on WM8753.c
+ *
+ * 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/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "wm8750.h"
+
+#define AUDIO_NAME "WM8750"
+#define WM8750_VERSION "0.12"
+
+/*
+ * Debug
+ */
+
+#define WM8750_DEBUG 0
+
+#ifdef WM8750_DEBUG
+#define dbg(format, arg...) \
+ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
+#else
+#define dbg(format, arg...) do {} while (0)
+#endif
+#define err(format, arg...) \
+ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
+#define info(format, arg...) \
+ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
+#define warn(format, arg...) \
+ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
+
+/* codec private data */
+struct wm8750_priv {
+ unsigned int sysclk;
+};
+
+/*
+ * wm8750 register cache
+ * We can't read the WM8750 register space when we
+ * are using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8750_reg[] = {
+ 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */
+ 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */
+ 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */
+ 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */
+ 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */
+ 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */
+ 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */
+ 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */
+ 0x0079, 0x0079, 0x0079, /* 40 */
+};
+
+/*
+ * read wm8750 register cache
+ */
+static inline unsigned int wm8750_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg > WM8750_CACHE_REGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write wm8750 register cache
+ */
+static inline void wm8750_write_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg > WM8750_CACHE_REGNUM)
+ return;
+ cache[reg] = value;
+}
+
+static int wm8750_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D9 WM8753 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ wm8750_write_reg_cache (codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+#define wm8750_reset(c) wm8750_write(c, WM8750_RESET, 0)
+
+/*
+ * WM8750 Controls
+ */
+static const char *wm8750_bass[] = {"Linear Control", "Adaptive Boost"};
+static const char *wm8750_bass_filter[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" };
+static const char *wm8750_treble[] = {"8kHz", "4kHz"};
+static const char *wm8750_3d_lc[] = {"200Hz", "500Hz"};
+static const char *wm8750_3d_uc[] = {"2.2kHz", "1.5kHz"};
+static const char *wm8750_3d_func[] = {"Capture", "Playback"};
+static const char *wm8750_alc_func[] = {"Off", "Right", "Left", "Stereo"};
+static const char *wm8750_ng_type[] = {"Constant PGA Gain",
+ "Mute ADC Output"};
+static const char *wm8750_line_mux[] = {"Line 1", "Line 2", "Line 3", "PGA",
+ "Differential"};
+static const char *wm8750_pga_sel[] = {"Line 1", "Line 2", "Line 3",
+ "Differential"};
+static const char *wm8750_out3[] = {"VREF", "ROUT1 + Vol", "MonoOut",
+ "ROUT1"};
+static const char *wm8750_diff_sel[] = {"Line 1", "Line 2"};
+static const char *wm8750_adcpol[] = {"Normal", "L Invert", "R Invert",
+ "L + R Invert"};
+static const char *wm8750_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const char *wm8750_mono_mux[] = {"Stereo", "Mono (Left)",
+ "Mono (Right)", "Digital Mono"};
+
+static const struct soc_enum wm8750_enum[] = {
+SOC_ENUM_SINGLE(WM8750_BASS, 7, 2, wm8750_bass),
+SOC_ENUM_SINGLE(WM8750_BASS, 6, 2, wm8750_bass_filter),
+SOC_ENUM_SINGLE(WM8750_TREBLE, 6, 2, wm8750_treble),
+SOC_ENUM_SINGLE(WM8750_3D, 5, 2, wm8750_3d_lc),
+SOC_ENUM_SINGLE(WM8750_3D, 6, 2, wm8750_3d_uc),
+SOC_ENUM_SINGLE(WM8750_3D, 7, 2, wm8750_3d_func),
+SOC_ENUM_SINGLE(WM8750_ALC1, 7, 4, wm8750_alc_func),
+SOC_ENUM_SINGLE(WM8750_NGATE, 1, 2, wm8750_ng_type),
+SOC_ENUM_SINGLE(WM8750_LOUTM1, 0, 5, wm8750_line_mux),
+SOC_ENUM_SINGLE(WM8750_ROUTM1, 0, 5, wm8750_line_mux),
+SOC_ENUM_SINGLE(WM8750_LADCIN, 6, 4, wm8750_pga_sel), /* 10 */
+SOC_ENUM_SINGLE(WM8750_RADCIN, 6, 4, wm8750_pga_sel),
+SOC_ENUM_SINGLE(WM8750_ADCTL2, 7, 4, wm8750_out3),
+SOC_ENUM_SINGLE(WM8750_ADCIN, 8, 2, wm8750_diff_sel),
+SOC_ENUM_SINGLE(WM8750_ADCDAC, 5, 4, wm8750_adcpol),
+SOC_ENUM_SINGLE(WM8750_ADCDAC, 1, 4, wm8750_deemph),
+SOC_ENUM_SINGLE(WM8750_ADCIN, 6, 4, wm8750_mono_mux), /* 16 */
+
+};
+
+static const struct snd_kcontrol_new wm8750_snd_controls[] = {
+
+SOC_DOUBLE_R("Capture Volume", WM8750_LINVOL, WM8750_RINVOL, 0, 63, 0),
+SOC_DOUBLE_R("Capture ZC Switch", WM8750_LINVOL, WM8750_RINVOL, 6, 1, 0),
+SOC_DOUBLE_R("Capture Switch", WM8750_LINVOL, WM8750_RINVOL, 7, 1, 1),
+
+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8750_LOUT1V,
+ WM8750_ROUT1V, 7, 1, 0),
+SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8750_LOUT2V,
+ WM8750_ROUT2V, 7, 1, 0),
+
+SOC_ENUM("Playback De-emphasis", wm8750_enum[15]),
+
+SOC_ENUM("Capture Polarity", wm8750_enum[14]),
+SOC_SINGLE("Playback 6dB Attenuate", WM8750_ADCDAC, 7, 1, 0),
+SOC_SINGLE("Capture 6dB Attenuate", WM8750_ADCDAC, 8, 1, 0),
+
+SOC_DOUBLE_R("PCM Volume", WM8750_LDAC, WM8750_RDAC, 0, 255, 0),
+
+SOC_ENUM("Bass Boost", wm8750_enum[0]),
+SOC_ENUM("Bass Filter", wm8750_enum[1]),
+SOC_SINGLE("Bass Volume", WM8750_BASS, 0, 15, 1),
+
+SOC_SINGLE("Treble Volume", WM8750_TREBLE, 0, 15, 0),
+SOC_ENUM("Treble Cut-off", wm8750_enum[2]),
+
+SOC_SINGLE("3D Switch", WM8750_3D, 0, 1, 0),
+SOC_SINGLE("3D Volume", WM8750_3D, 1, 15, 0),
+SOC_ENUM("3D Lower Cut-off", wm8750_enum[3]),
+SOC_ENUM("3D Upper Cut-off", wm8750_enum[4]),
+SOC_ENUM("3D Mode", wm8750_enum[5]),
+
+SOC_SINGLE("ALC Capture Target Volume", WM8750_ALC1, 0, 7, 0),
+SOC_SINGLE("ALC Capture Max Volume", WM8750_ALC1, 4, 7, 0),
+SOC_ENUM("ALC Capture Function", wm8750_enum[6]),
+SOC_SINGLE("ALC Capture ZC Switch", WM8750_ALC2, 7, 1, 0),
+SOC_SINGLE("ALC Capture Hold Time", WM8750_ALC2, 0, 15, 0),
+SOC_SINGLE("ALC Capture Decay Time", WM8750_ALC3, 4, 15, 0),
+SOC_SINGLE("ALC Capture Attack Time", WM8750_ALC3, 0, 15, 0),
+SOC_SINGLE("ALC Capture NG Threshold", WM8750_NGATE, 3, 31, 0),
+SOC_ENUM("ALC Capture NG Type", wm8750_enum[4]),
+SOC_SINGLE("ALC Capture NG Switch", WM8750_NGATE, 0, 1, 0),
+
+SOC_SINGLE("Left ADC Capture Volume", WM8750_LADC, 0, 255, 0),
+SOC_SINGLE("Right ADC Capture Volume", WM8750_RADC, 0, 255, 0),
+
+SOC_SINGLE("ZC Timeout Switch", WM8750_ADCTL1, 0, 1, 0),
+SOC_SINGLE("Playback Invert Switch", WM8750_ADCTL1, 1, 1, 0),
+
+SOC_SINGLE("Right Speaker Playback Invert Switch", WM8750_ADCTL2, 4, 1, 0),
+
+/* Unimplemented */
+/* ADCDAC Bit 0 - ADCHPD */
+/* ADCDAC Bit 4 - HPOR */
+/* ADCTL1 Bit 2,3 - DATSEL */
+/* ADCTL1 Bit 4,5 - DMONOMIX */
+/* ADCTL1 Bit 6,7 - VSEL */
+/* ADCTL2 Bit 2 - LRCM */
+/* ADCTL2 Bit 3 - TRI */
+/* ADCTL3 Bit 5 - HPFLREN */
+/* ADCTL3 Bit 6 - VROI */
+/* ADCTL3 Bit 7,8 - ADCLRM */
+/* ADCIN Bit 4 - LDCM */
+/* ADCIN Bit 5 - RDCM */
+
+SOC_DOUBLE_R("Mic Boost", WM8750_LADCIN, WM8750_RADCIN, 4, 3, 0),
+
+SOC_DOUBLE_R("Bypass Left Playback Volume", WM8750_LOUTM1,
+ WM8750_LOUTM2, 4, 7, 1),
+SOC_DOUBLE_R("Bypass Right Playback Volume", WM8750_ROUTM1,
+ WM8750_ROUTM2, 4, 7, 1),
+SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8750_MOUTM1,
+ WM8750_MOUTM2, 4, 7, 1),
+
+SOC_SINGLE("Mono Playback ZC Switch", WM8750_MOUTV, 7, 1, 0),
+
+SOC_DOUBLE_R("Headphone Playback Volume", WM8750_LOUT1V, WM8750_ROUT1V,
+ 0, 127, 0),
+SOC_DOUBLE_R("Speaker Playback Volume", WM8750_LOUT2V, WM8750_ROUT2V,
+ 0, 127, 0),
+
+SOC_SINGLE("Mono Playback Volume", WM8750_MOUTV, 0, 127, 0),
+
+};
+
+/* add non dapm controls */
+static int wm8750_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8750_snd_controls[i],codec, NULL));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/*
+ * DAPM Controls
+ */
+
+/* Left Mixer */
+static const struct snd_kcontrol_new wm8750_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("Playback Switch", WM8750_LOUTM1, 8, 1, 0),
+SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_LOUTM1, 7, 1, 0),
+SOC_DAPM_SINGLE("Right Playback Switch", WM8750_LOUTM2, 8, 1, 0),
+SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_LOUTM2, 7, 1, 0),
+};
+
+/* Right Mixer */
+static const struct snd_kcontrol_new wm8750_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("Left Playback Switch", WM8750_ROUTM1, 8, 1, 0),
+SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_ROUTM1, 7, 1, 0),
+SOC_DAPM_SINGLE("Playback Switch", WM8750_ROUTM2, 8, 1, 0),
+SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_ROUTM2, 7, 1, 0),
+};
+
+/* Mono Mixer */
+static const struct snd_kcontrol_new wm8750_mono_mixer_controls[] = {
+SOC_DAPM_SINGLE("Left Playback Switch", WM8750_MOUTM1, 8, 1, 0),
+SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_MOUTM1, 7, 1, 0),
+SOC_DAPM_SINGLE("Right Playback Switch", WM8750_MOUTM2, 8, 1, 0),
+SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_MOUTM2, 7, 1, 0),
+};
+
+/* Left Line Mux */
+static const struct snd_kcontrol_new wm8750_left_line_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[8]);
+
+/* Right Line Mux */
+static const struct snd_kcontrol_new wm8750_right_line_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[9]);
+
+/* Left PGA Mux */
+static const struct snd_kcontrol_new wm8750_left_pga_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[10]);
+
+/* Right PGA Mux */
+static const struct snd_kcontrol_new wm8750_right_pga_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[11]);
+
+/* Out 3 Mux */
+static const struct snd_kcontrol_new wm8750_out3_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[12]);
+
+/* Differential Mux */
+static const struct snd_kcontrol_new wm8750_diffmux_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[13]);
+
+/* Mono ADC Mux */
+static const struct snd_kcontrol_new wm8750_monomux_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[16]);
+
+static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
+ SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+ &wm8750_left_mixer_controls[0],
+ ARRAY_SIZE(wm8750_left_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+ &wm8750_right_mixer_controls[0],
+ ARRAY_SIZE(wm8750_right_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Mono Mixer", WM8750_PWR2, 2, 0,
+ &wm8750_mono_mixer_controls[0],
+ ARRAY_SIZE(wm8750_mono_mixer_controls)),
+
+ SND_SOC_DAPM_PGA("Right Out 2", WM8750_PWR2, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Left Out 2", WM8750_PWR2, 4, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Right Out 1", WM8750_PWR2, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Left Out 1", WM8750_PWR2, 6, 0, NULL, 0),
+ SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8750_PWR2, 7, 0),
+ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8750_PWR2, 8, 0),
+
+ SND_SOC_DAPM_MICBIAS("Mic Bias", WM8750_PWR1, 1, 0),
+ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8750_PWR1, 2, 0),
+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8750_PWR1, 3, 0),
+
+ SND_SOC_DAPM_MUX("Left PGA Mux", WM8750_PWR1, 5, 0,
+ &wm8750_left_pga_controls),
+ SND_SOC_DAPM_MUX("Right PGA Mux", WM8750_PWR1, 4, 0,
+ &wm8750_right_pga_controls),
+ SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0,
+ &wm8750_left_line_controls),
+ SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0,
+ &wm8750_right_line_controls),
+
+ SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, &wm8750_out3_controls),
+ SND_SOC_DAPM_PGA("Out 3", WM8750_PWR2, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Mono Out 1", WM8750_PWR2, 2, 0, NULL, 0),
+
+ SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0,
+ &wm8750_diffmux_controls),
+ SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0,
+ &wm8750_monomux_controls),
+ SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0,
+ &wm8750_monomux_controls),
+
+ SND_SOC_DAPM_OUTPUT("LOUT1"),
+ SND_SOC_DAPM_OUTPUT("ROUT1"),
+ SND_SOC_DAPM_OUTPUT("LOUT2"),
+ SND_SOC_DAPM_OUTPUT("ROUT2"),
+ SND_SOC_DAPM_OUTPUT("MONO"),
+ SND_SOC_DAPM_OUTPUT("OUT3"),
+
+ SND_SOC_DAPM_INPUT("LINPUT1"),
+ SND_SOC_DAPM_INPUT("LINPUT2"),
+ SND_SOC_DAPM_INPUT("LINPUT3"),
+ SND_SOC_DAPM_INPUT("RINPUT1"),
+ SND_SOC_DAPM_INPUT("RINPUT2"),
+ SND_SOC_DAPM_INPUT("RINPUT3"),
+};
+
+static const char *audio_map[][3] = {
+ /* left mixer */
+ {"Left Mixer", "Playback Switch", "Left DAC"},
+ {"Left Mixer", "Left Bypass Switch", "Left Line Mux"},
+ {"Left Mixer", "Right Playback Switch", "Right DAC"},
+ {"Left Mixer", "Right Bypass Switch", "Right Line Mux"},
+
+ /* right mixer */
+ {"Right Mixer", "Left Playback Switch", "Left DAC"},
+ {"Right Mixer", "Left Bypass Switch", "Left Line Mux"},
+ {"Right Mixer", "Playback Switch", "Right DAC"},
+ {"Right Mixer", "Right Bypass Switch", "Right Line Mux"},
+
+ /* left out 1 */
+ {"Left Out 1", NULL, "Left Mixer"},
+ {"LOUT1", NULL, "Left Out 1"},
+
+ /* left out 2 */
+ {"Left Out 2", NULL, "Left Mixer"},
+ {"LOUT2", NULL, "Left Out 2"},
+
+ /* right out 1 */
+ {"Right Out 1", NULL, "Right Mixer"},
+ {"ROUT1", NULL, "Right Out 1"},
+
+ /* right out 2 */
+ {"Right Out 2", NULL, "Right Mixer"},
+ {"ROUT2", NULL, "Right Out 2"},
+
+ /* mono mixer */
+ {"Mono Mixer", "Left Playback Switch", "Left DAC"},
+ {"Mono Mixer", "Left Bypass Switch", "Left Line Mux"},
+ {"Mono Mixer", "Right Playback Switch", "Right DAC"},
+ {"Mono Mixer", "Right Bypass Switch", "Right Line Mux"},
+
+ /* mono out */
+ {"Mono Out 1", NULL, "Mono Mixer"},
+ {"MONO1", NULL, "Mono Out 1"},
+
+ /* out 3 */
+ {"Out3 Mux", "VREF", "VREF"},
+ {"Out3 Mux", "ROUT1 + Vol", "ROUT1"},
+ {"Out3 Mux", "ROUT1", "Right Mixer"},
+ {"Out3 Mux", "MonoOut", "MONO1"},
+ {"Out 3", NULL, "Out3 Mux"},
+ {"OUT3", NULL, "Out 3"},
+
+ /* Left Line Mux */
+ {"Left Line Mux", "Line 1", "LINPUT1"},
+ {"Left Line Mux", "Line 2", "LINPUT2"},
+ {"Left Line Mux", "Line 3", "LINPUT3"},
+ {"Left Line Mux", "PGA", "Left PGA Mux"},
+ {"Left Line Mux", "Differential", "Differential Mux"},
+
+ /* Right Line Mux */
+ {"Right Line Mux", "Line 1", "RINPUT1"},
+ {"Right Line Mux", "Line 2", "RINPUT2"},
+ {"Right Line Mux", "Line 3", "RINPUT3"},
+ {"Right Line Mux", "PGA", "Right PGA Mux"},
+ {"Right Line Mux", "Differential", "Differential Mux"},
+
+ /* Left PGA Mux */
+ {"Left PGA Mux", "Line 1", "LINPUT1"},
+ {"Left PGA Mux", "Line 2", "LINPUT2"},
+ {"Left PGA Mux", "Line 3", "LINPUT3"},
+ {"Left PGA Mux", "Differential", "Differential Mux"},
+
+ /* Right PGA Mux */
+ {"Right PGA Mux", "Line 1", "RINPUT1"},
+ {"Right PGA Mux", "Line 2", "RINPUT2"},
+ {"Right PGA Mux", "Line 3", "RINPUT3"},
+ {"Right PGA Mux", "Differential", "Differential Mux"},
+
+ /* Differential Mux */
+ {"Differential Mux", "Line 1", "LINPUT1"},
+ {"Differential Mux", "Line 1", "RINPUT1"},
+ {"Differential Mux", "Line 2", "LINPUT2"},
+ {"Differential Mux", "Line 2", "RINPUT2"},
+
+ /* Left ADC Mux */
+ {"Left ADC Mux", "Stereo", "Left PGA Mux"},
+ {"Left ADC Mux", "Mono (Left)", "Left PGA Mux"},
+ {"Left ADC Mux", "Digital Mono", "Left PGA Mux"},
+
+ /* Right ADC Mux */
+ {"Right ADC Mux", "Stereo", "Right PGA Mux"},
+ {"Right ADC Mux", "Mono (Right)", "Right PGA Mux"},
+ {"Right ADC Mux", "Digital Mono", "Right PGA Mux"},
+
+ /* ADC */
+ {"Left ADC", NULL, "Left ADC Mux"},
+ {"Right ADC", NULL, "Right ADC Mux"},
+
+ /* terminator */
+ {NULL, NULL, NULL},
+};
+
+static int wm8750_add_widgets(struct snd_soc_codec *codec)
+{
+ int i;
+
+ for(i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) {
+ snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]);
+ }
+
+ /* set up audio path audio_mapnects */
+ for(i = 0; audio_map[i][0] != NULL; i++) {
+ snd_soc_dapm_connect_input(codec, audio_map[i][0],
+ audio_map[i][1], audio_map[i][2]);
+ }
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+struct _coeff_div {
+ u32 mclk;
+ u32 rate;
+ u16 fs;
+ u8 sr:5;
+ u8 usb:1;
+};
+
+/* codec hifi mclk clock divider coefficients */
+static const struct _coeff_div coeff_div[] = {
+ /* 8k */
+ {12288000, 8000, 1536, 0x6, 0x0},
+ {11289600, 8000, 1408, 0x16, 0x0},
+ {18432000, 8000, 2304, 0x7, 0x0},
+ {16934400, 8000, 2112, 0x17, 0x0},
+ {12000000, 8000, 1500, 0x6, 0x1},
+
+ /* 11.025k */
+ {11289600, 11025, 1024, 0x18, 0x0},
+ {16934400, 11025, 1536, 0x19, 0x0},
+ {12000000, 11025, 1088, 0x19, 0x1},
+
+ /* 16k */
+ {12288000, 16000, 768, 0xa, 0x0},
+ {18432000, 16000, 1152, 0xb, 0x0},
+ {12000000, 16000, 750, 0xa, 0x1},
+
+ /* 22.05k */
+ {11289600, 22050, 512, 0x1a, 0x0},
+ {16934400, 22050, 768, 0x1b, 0x0},
+ {12000000, 22050, 544, 0x1b, 0x1},
+
+ /* 32k */
+ {12288000, 32000, 384, 0xc, 0x0},
+ {18432000, 32000, 576, 0xd, 0x0},
+ {12000000, 32000, 375, 0xa, 0x1},
+
+ /* 44.1k */
+ {11289600, 44100, 256, 0x10, 0x0},
+ {16934400, 44100, 384, 0x11, 0x0},
+ {12000000, 44100, 272, 0x11, 0x1},
+
+ /* 48k */
+ {12288000, 48000, 256, 0x0, 0x0},
+ {18432000, 48000, 384, 0x1, 0x0},
+ {12000000, 48000, 250, 0x0, 0x1},
+
+ /* 88.2k */
+ {11289600, 88200, 128, 0x1e, 0x0},
+ {16934400, 88200, 192, 0x1f, 0x0},
+ {12000000, 88200, 136, 0x1f, 0x1},
+
+ /* 96k */
+ {12288000, 96000, 128, 0xe, 0x0},
+ {18432000, 96000, 192, 0xf, 0x0},
+ {12000000, 96000, 125, 0xe, 0x1},
+};
+
+static inline int get_coeff(int mclk, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+ return i;
+ }
+
+ printk(KERN_ERR "wm8750: could not get coeff for mclk %d @ rate %d\n",
+ mclk, rate);
+ return -EINVAL;
+}
+
+static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8750_priv *wm8750 = codec->private_data;
+
+ switch (freq) {
+ case 11289600:
+ case 12000000:
+ case 12288000:
+ case 16934400:
+ case 18432000:
+ wm8750->sysclk = freq;
+ return 0;
+ }
+ return -EINVAL;
+}
+
+static int wm8750_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = 0;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface = 0x0040;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= 0x0013;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x0090;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x0010;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8750_write(codec, WM8750_IFACE, iface);
+ return 0;
+}
+
+static int wm8750_pcm_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_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm8750_priv *wm8750 = codec->private_data;
+ u16 iface = wm8750_read_reg_cache(codec, WM8750_IFACE) & 0x1f3;
+ u16 srate = wm8750_read_reg_cache(codec, WM8750_SRATE) & 0x1c0;
+ int coeff = get_coeff(wm8750->sysclk, params_rate(params));
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x0004;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x0008;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= 0x000c;
+ break;
+ }
+
+ /* set iface & srate */
+ wm8750_write(codec, WM8750_IFACE, iface);
+ if (coeff >= 0)
+ wm8750_write(codec, WM8750_SRATE, srate |
+ (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb);
+
+ return 0;
+}
+
+static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7;
+
+ if (mute)
+ wm8750_write(codec, WM8750_ADCDAC, mute_reg | 0x8);
+ else
+ wm8750_write(codec, WM8750_ADCDAC, mute_reg);
+ return 0;
+}
+
+static int wm8750_dapm_event(struct snd_soc_codec *codec, int event)
+{
+ u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e;
+
+ switch (event) {
+ case SNDRV_CTL_POWER_D0: /* full On */
+ /* set vmid to 50k and unmute dac */
+ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0);
+ break;
+ case SNDRV_CTL_POWER_D1: /* partial On */
+ case SNDRV_CTL_POWER_D2: /* partial On */
+ /* set vmid to 5k for quick power up */
+ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1);
+ break;
+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+ /* mute dac and set vmid to 500k, enable VREF */
+ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141);
+ break;
+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+ wm8750_write(codec, WM8750_PWR1, 0x0001);
+ break;
+ }
+ codec->dapm_state = event;
+ return 0;
+}
+
+#define WM8750_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_codec_dai wm8750_dai = {
+ .name = "WM8750",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8750_RATES,
+ .formats = WM8750_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8750_RATES,
+ .formats = WM8750_FORMATS,},
+ .ops = {
+ .hw_params = wm8750_pcm_hw_params,
+ },
+ .dai_ops = {
+ .digital_mute = wm8750_mute,
+ .set_fmt = wm8750_set_dai_fmt,
+ .set_sysclk = wm8750_set_dai_sysclk,
+ },
+};
+EXPORT_SYMBOL_GPL(wm8750_dai);
+
+static void wm8750_work(struct work_struct *work)
+{
+ struct snd_soc_codec *codec =
+ container_of(work, struct snd_soc_codec, delayed_work.work);
+ wm8750_dapm_event(codec, codec->dapm_state);
+}
+
+static int wm8750_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+ return 0;
+}
+
+static int wm8750_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(wm8750_reg); i++) {
+ if (i == WM8750_RESET)
+ continue;
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+
+ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+
+ /* charge wm8750 caps */
+ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
+ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2);
+ codec->dapm_state = SNDRV_CTL_POWER_D0;
+ schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000));
+ }
+
+ return 0;
+}
+
+/*
+ * initialise the WM8750 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8750_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int reg, ret = 0;
+
+ codec->name = "WM8750";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8750_read_reg_cache;
+ codec->write = wm8750_write;
+ codec->dapm_event = wm8750_dapm_event;
+ codec->dai = &wm8750_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = sizeof(wm8750_reg);
+ codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KRENEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ wm8750_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8750: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* charge output caps */
+ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2);
+ codec->dapm_state = SNDRV_CTL_POWER_D3hot;
+ schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000));
+
+ /* set the update bits */
+ reg = wm8750_read_reg_cache(codec, WM8750_LDAC);
+ wm8750_write(codec, WM8750_LDAC, reg | 0x0100);
+ reg = wm8750_read_reg_cache(codec, WM8750_RDAC);
+ wm8750_write(codec, WM8750_RDAC, reg | 0x0100);
+ reg = wm8750_read_reg_cache(codec, WM8750_LOUT1V);
+ wm8750_write(codec, WM8750_LOUT1V, reg | 0x0100);
+ reg = wm8750_read_reg_cache(codec, WM8750_ROUT1V);
+ wm8750_write(codec, WM8750_ROUT1V, reg | 0x0100);
+ reg = wm8750_read_reg_cache(codec, WM8750_LOUT2V);
+ wm8750_write(codec, WM8750_LOUT2V, reg | 0x0100);
+ reg = wm8750_read_reg_cache(codec, WM8750_ROUT2V);
+ wm8750_write(codec, WM8750_ROUT2V, reg | 0x0100);
+ reg = wm8750_read_reg_cache(codec, WM8750_LINVOL);
+ wm8750_write(codec, WM8750_LINVOL, reg | 0x0100);
+ reg = wm8750_read_reg_cache(codec, WM8750_RINVOL);
+ wm8750_write(codec, WM8750_RINVOL, reg | 0x0100);
+
+ wm8750_add_controls(codec);
+ wm8750_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8750: failed to register card\n");
+ goto card_err;
+ }
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+/* If the i2c layer weren't so broken, we could pass this kind of data
+ around */
+static struct snd_soc_device *wm8750_socdev;
+
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+
+/*
+ * WM8731 2 wire address is determined by GPIO5
+ * state during powerup.
+ * low = 0x1a
+ * high = 0x1b
+ */
+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static struct i2c_driver wm8750_i2c_driver;
+static struct i2c_client client_template;
+
+static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+ struct snd_soc_device *socdev = wm8750_socdev;
+ struct wm8750_setup_data *setup = socdev->codec_data;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct i2c_client *i2c;
+ int ret;
+
+ if (addr != setup->i2c_address)
+ return -ENODEV;
+
+ client_template.adapter = adap;
+ client_template.addr = addr;
+
+ i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
+ if (i2c == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = i2c_attach_client(i2c);
+ if (ret < 0) {
+ err("failed to attach codec at addr %x\n", addr);
+ goto err;
+ }
+
+ ret = wm8750_init(socdev);
+ if (ret < 0) {
+ err("failed to initialise WM8750\n");
+ goto err;
+ }
+ return ret;
+
+err:
+ kfree(codec);
+ kfree(i2c);
+ return ret;
+}
+
+static int wm8750_i2c_detach(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ i2c_detach_client(client);
+ kfree(codec->reg_cache);
+ kfree(client);
+ return 0;
+}
+
+static int wm8750_i2c_attach(struct i2c_adapter *adap)
+{
+ return i2c_probe(adap, &addr_data, wm8750_codec_probe);
+}
+
+/* corgi i2c codec control layer */
+static struct i2c_driver wm8750_i2c_driver = {
+ .driver = {
+ .name = "WM8750 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .id = I2C_DRIVERID_WM8750,
+ .attach_adapter = wm8750_i2c_attach,
+ .detach_client = wm8750_i2c_detach,
+ .command = NULL,
+};
+
+static struct i2c_client client_template = {
+ .name = "WM8750",
+ .driver = &wm8750_i2c_driver,
+};
+#endif
+
+static int wm8750_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct wm8750_setup_data *setup = socdev->codec_data;
+ struct snd_soc_codec *codec;
+ struct wm8750_priv *wm8750;
+ int ret = 0;
+
+ info("WM8750 Audio Codec %s", WM8750_VERSION);
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL);
+ if (wm8750 == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+
+ codec->private_data = wm8750;
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+ wm8750_socdev = socdev;
+ INIT_DELAYED_WORK(&codec->delayed_work, wm8750_work);
+
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ normal_i2c[0] = setup->i2c_address;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = i2c_add_driver(&wm8750_i2c_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add i2c driver");
+ }
+#else
+ /* Add other interfaces here */
+#endif
+
+ return ret;
+}
+
+/*
+ * This function forces any delayed work to be queued and run.
+ */
+static int run_delayed_work(struct delayed_work *dwork)
+{
+ int ret;
+
+ /* cancel any work waiting to be queued. */
+ ret = cancel_delayed_work(dwork);
+
+ /* if there was any work waiting then we run it now and
+ * wait for it's completion */
+ if (ret) {
+ schedule_delayed_work(dwork, 0);
+ flush_scheduled_work();
+ }
+ return ret;
+}
+
+/* power down chip */
+static int wm8750_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+ run_delayed_work(&codec->delayed_work);
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+ i2c_del_driver(&wm8750_i2c_driver);
+#endif
+ kfree(codec->private_data);
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8750 = {
+ .probe = wm8750_probe,
+ .remove = wm8750_remove,
+ .suspend = wm8750_suspend,
+ .resume = wm8750_resume,
+};
+
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750);
+
+MODULE_DESCRIPTION("ASoC WM8750 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8750.h b/sound/soc/codecs/wm8750.h
new file mode 100644
index 00000000000..a97a54a6348
--- /dev/null
+++ b/sound/soc/codecs/wm8750.h
@@ -0,0 +1,67 @@
+/*
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on WM8753.h
+ *
+ * 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 _WM8750_H
+#define _WM8750_H
+
+/* WM8750 register space */
+
+#define WM8750_LINVOL 0x00
+#define WM8750_RINVOL 0x01
+#define WM8750_LOUT1V 0x02
+#define WM8750_ROUT1V 0x03
+#define WM8750_ADCDAC 0x05
+#define WM8750_IFACE 0x07
+#define WM8750_SRATE 0x08
+#define WM8750_LDAC 0x0a
+#define WM8750_RDAC 0x0b
+#define WM8750_BASS 0x0c
+#define WM8750_TREBLE 0x0d
+#define WM8750_RESET 0x0f
+#define WM8750_3D 0x10
+#define WM8750_ALC1 0x11
+#define WM8750_ALC2 0x12
+#define WM8750_ALC3 0x13
+#define WM8750_NGATE 0x14
+#define WM8750_LADC 0x15
+#define WM8750_RADC 0x16
+#define WM8750_ADCTL1 0x17
+#define WM8750_ADCTL2 0x18
+#define WM8750_PWR1 0x19
+#define WM8750_PWR2 0x1a
+#define WM8750_ADCTL3 0x1b
+#define WM8750_ADCIN 0x1f
+#define WM8750_LADCIN 0x20
+#define WM8750_RADCIN 0x21
+#define WM8750_LOUTM1 0x22
+#define WM8750_LOUTM2 0x23
+#define WM8750_ROUTM1 0x24
+#define WM8750_ROUTM2 0x25
+#define WM8750_MOUTM1 0x26
+#define WM8750_MOUTM2 0x27
+#define WM8750_LOUT2V 0x28
+#define WM8750_ROUT2V 0x29
+#define WM8750_MOUTV 0x2a
+
+#define WM8750_CACHE_REGNUM 0x2a
+
+#define WM8750_SYSCLK 0
+
+struct wm8750_setup_data {
+ unsigned short i2c_address;
+};
+
+extern struct snd_soc_codec_dai wm8750_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8750;
+
+#endif
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
new file mode 100644
index 00000000000..92a64871bcd
--- /dev/null
+++ b/sound/soc/codecs/wm9712.c
@@ -0,0 +1,771 @@
+/*
+ * wm9712.c -- ALSA Soc WM9712 codec support
+ *
+ * Copyright 2006 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.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.
+ *
+ * Revision history
+ * 4th Feb 2006 Initial version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#define WM9712_VERSION "0.4"
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+ unsigned int reg);
+static int ac97_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int val);
+
+/*
+ * WM9712 register cache
+ */
+static const u16 wm9712_reg[] = {
+ 0x6174, 0x8000, 0x8000, 0x8000, // 6
+ 0xf0f0, 0xaaa0, 0xc008, 0x6808, // e
+ 0xe808, 0xaaa0, 0xad00, 0x8000, // 16
+ 0xe808, 0x3000, 0x8000, 0x0000, // 1e
+ 0x0000, 0x0000, 0x0000, 0x000f, // 26
+ 0x0405, 0x0410, 0xbb80, 0xbb80, // 2e
+ 0x0000, 0xbb80, 0x0000, 0x0000, // 36
+ 0x0000, 0x2000, 0x0000, 0x0000, // 3e
+ 0x0000, 0x0000, 0x0000, 0x0000, // 46
+ 0x0000, 0x0000, 0xf83e, 0xffff, // 4e
+ 0x0000, 0x0000, 0x0000, 0xf83e, // 56
+ 0x0008, 0x0000, 0x0000, 0x0000, // 5e
+ 0xb032, 0x3e00, 0x0000, 0x0000, // 66
+ 0x0000, 0x0000, 0x0000, 0x0000, // 6e
+ 0x0000, 0x0000, 0x0000, 0x0006, // 76
+ 0x0001, 0x0000, 0x574d, 0x4c12, // 7e
+ 0x0000, 0x0000 // virtual hp mixers
+};
+
+/* virtual HP mixers regs */
+#define HPL_MIXER 0x80
+#define HPR_MIXER 0x82
+
+static const char *wm9712_alc_select[] = {"None", "Left", "Right", "Stereo"};
+static const char *wm9712_alc_mux[] = {"Stereo", "Left", "Right", "None"};
+static const char *wm9712_out3_src[] = {"Left", "VREF", "Left + Right",
+ "Mono"};
+static const char *wm9712_spk_src[] = {"Speaker Mix", "Headphone Mix"};
+static const char *wm9712_rec_adc[] = {"Stereo", "Left", "Right", "Mute"};
+static const char *wm9712_base[] = {"Linear Control", "Adaptive Boost"};
+static const char *wm9712_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"};
+static const char *wm9712_mic[] = {"Mic 1", "Differential", "Mic 2",
+ "Stereo"};
+static const char *wm9712_rec_sel[] = {"Mic", "NC", "NC", "Speaker Mixer",
+ "Line", "Headphone Mixer", "Phone Mixer", "Phone"};
+static const char *wm9712_ng_type[] = {"Constant Gain", "Mute"};
+static const char *wm9712_diff_sel[] = {"Mic", "Line"};
+
+static const struct soc_enum wm9712_enum[] = {
+SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9712_alc_select),
+SOC_ENUM_SINGLE(AC97_VIDEO, 12, 4, wm9712_alc_mux),
+SOC_ENUM_SINGLE(AC97_AUX, 9, 4, wm9712_out3_src),
+SOC_ENUM_SINGLE(AC97_AUX, 8, 2, wm9712_spk_src),
+SOC_ENUM_SINGLE(AC97_REC_SEL, 12, 4, wm9712_rec_adc),
+SOC_ENUM_SINGLE(AC97_MASTER_TONE, 15, 2, wm9712_base),
+SOC_ENUM_DOUBLE(AC97_REC_GAIN, 14, 6, 2, wm9712_rec_gain),
+SOC_ENUM_SINGLE(AC97_MIC, 5, 4, wm9712_mic),
+SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9712_rec_sel),
+SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9712_rec_sel),
+SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9712_ng_type),
+SOC_ENUM_SINGLE(0x5c, 8, 2, wm9712_diff_sel),
+};
+
+static const struct snd_kcontrol_new wm9712_snd_ac97_controls[] = {
+SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
+SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1),
+SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
+SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE,15, 1, 1),
+
+SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0),
+SOC_SINGLE("Speaker Playback Invert Switch", AC97_MASTER, 6, 1, 0),
+SOC_SINGLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 7, 1, 0),
+SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_MONO, 7, 1, 0),
+SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 0),
+
+SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
+SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0),
+SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0),
+SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0),
+SOC_ENUM("ALC Function", wm9712_enum[0]),
+SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),
+SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 1),
+SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0),
+SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0),
+SOC_ENUM("ALC NG Type", wm9712_enum[10]),
+SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 1),
+
+SOC_SINGLE("Mic Headphone Volume", AC97_VIDEO, 12, 7, 1),
+SOC_SINGLE("ALC Headphone Volume", AC97_VIDEO, 7, 7, 1),
+
+SOC_SINGLE("Out3 Switch", AC97_AUX, 15, 1, 1),
+SOC_SINGLE("Out3 ZC Switch", AC97_AUX, 7, 1, 1),
+SOC_SINGLE("Out3 Volume", AC97_AUX, 0, 31, 1),
+
+SOC_SINGLE("PCBeep Bypass Headphone Volume", AC97_PC_BEEP, 12, 7, 1),
+SOC_SINGLE("PCBeep Bypass Speaker Volume", AC97_PC_BEEP, 8, 7, 1),
+SOC_SINGLE("PCBeep Bypass Phone Volume", AC97_PC_BEEP, 4, 7, 1),
+
+SOC_SINGLE("Aux Playback Headphone Volume", AC97_CD, 12, 7, 1),
+SOC_SINGLE("Aux Playback Speaker Volume", AC97_CD, 8, 7, 1),
+SOC_SINGLE("Aux Playback Phone Volume", AC97_CD, 4, 7, 1),
+
+SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 0),
+SOC_DOUBLE("Line Capture Volume", AC97_LINE, 8, 0, 31, 1),
+
+SOC_SINGLE("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0),
+SOC_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1),
+
+SOC_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1),
+SOC_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1),
+SOC_SINGLE("3D Playback Volume", AC97_3D_CONTROL, 0, 15, 0),
+
+SOC_ENUM("Bass Control", wm9712_enum[5]),
+SOC_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1),
+SOC_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1),
+SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0),
+SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 0),
+SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 0),
+
+SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1),
+SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),
+SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1),
+SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
+
+SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1),
+SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
+SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0),
+};
+
+/* add non dapm controls */
+static int wm9712_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm9712_snd_ac97_controls[i],codec, NULL));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/* We have to create a fake left and right HP mixers because
+ * the codec only has a single control that is shared by both channels.
+ * This makes it impossible to determine the audio path.
+ */
+static int mixer_event (struct snd_soc_dapm_widget *w, int event)
+{
+ u16 l, r, beep, line, phone, mic, pcm, aux;
+
+ l = ac97_read(w->codec, HPL_MIXER);
+ r = ac97_read(w->codec, HPR_MIXER);
+ beep = ac97_read(w->codec, AC97_PC_BEEP);
+ mic = ac97_read(w->codec, AC97_VIDEO);
+ phone = ac97_read(w->codec, AC97_PHONE);
+ line = ac97_read(w->codec, AC97_LINE);
+ pcm = ac97_read(w->codec, AC97_PCM);
+ aux = ac97_read(w->codec, AC97_CD);
+
+ if (l & 0x1 || r & 0x1)
+ ac97_write(w->codec, AC97_VIDEO, mic & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_VIDEO, mic | 0x8000);
+
+ if (l & 0x2 || r & 0x2)
+ ac97_write(w->codec, AC97_PCM, pcm & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_PCM, pcm | 0x8000);
+
+ if (l & 0x4 || r & 0x4)
+ ac97_write(w->codec, AC97_LINE, line & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_LINE, line | 0x8000);
+
+ if (l & 0x8 || r & 0x8)
+ ac97_write(w->codec, AC97_PHONE, phone & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_PHONE, phone | 0x8000);
+
+ if (l & 0x10 || r & 0x10)
+ ac97_write(w->codec, AC97_CD, aux & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_CD, aux | 0x8000);
+
+ if (l & 0x20 || r & 0x20)
+ ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000);
+
+ return 0;
+}
+
+/* Left Headphone Mixers */
+static const struct snd_kcontrol_new wm9712_hpl_mixer_controls[] = {
+ SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPL_MIXER, 5, 1, 0),
+ SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 4, 1, 0),
+ SOC_DAPM_SINGLE("Phone Bypass Switch", HPL_MIXER, 3, 1, 0),
+ SOC_DAPM_SINGLE("Line Bypass Switch", HPL_MIXER, 2, 1, 0),
+ SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 1, 1, 0),
+ SOC_DAPM_SINGLE("Mic Sidetone Switch", HPL_MIXER, 0, 1, 0),
+};
+
+/* Right Headphone Mixers */
+static const struct snd_kcontrol_new wm9712_hpr_mixer_controls[] = {
+ SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPR_MIXER, 5, 1, 0),
+ SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 4, 1, 0),
+ SOC_DAPM_SINGLE("Phone Bypass Switch", HPR_MIXER, 3, 1, 0),
+ SOC_DAPM_SINGLE("Line Bypass Switch", HPR_MIXER, 2, 1, 0),
+ SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 1, 1, 0),
+ SOC_DAPM_SINGLE("Mic Sidetone Switch", HPR_MIXER, 0, 1, 0),
+};
+
+/* Speaker Mixer */
+static const struct snd_kcontrol_new wm9712_speaker_mixer_controls[] = {
+ SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 11, 1, 1),
+ SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 11, 1, 1),
+ SOC_DAPM_SINGLE("Phone Bypass Switch", AC97_PHONE, 14, 1, 1),
+ SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 14, 1, 1),
+ SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 14, 1, 1),
+};
+
+/* Phone Mixer */
+static const struct snd_kcontrol_new wm9712_phone_mixer_controls[] = {
+ SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 7, 1, 1),
+ SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 7, 1, 1),
+ SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 13, 1, 1),
+ SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 13, 1, 1),
+ SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_MIC, 14, 1, 1),
+ SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_MIC, 13, 1, 1),
+};
+
+/* ALC headphone mux */
+static const struct snd_kcontrol_new wm9712_alc_mux_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[1]);
+
+/* out 3 mux */
+static const struct snd_kcontrol_new wm9712_out3_mux_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[2]);
+
+/* spk mux */
+static const struct snd_kcontrol_new wm9712_spk_mux_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[3]);
+
+/* Capture to Phone mux */
+static const struct snd_kcontrol_new wm9712_capture_phone_mux_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[4]);
+
+/* Capture left select */
+static const struct snd_kcontrol_new wm9712_capture_selectl_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[8]);
+
+/* Capture right select */
+static const struct snd_kcontrol_new wm9712_capture_selectr_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[9]);
+
+/* Mic select */
+static const struct snd_kcontrol_new wm9712_mic_src_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[7]);
+
+/* diff select */
+static const struct snd_kcontrol_new wm9712_diff_sel_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[11]);
+
+static const struct snd_soc_dapm_widget wm9712_dapm_widgets[] = {
+SND_SOC_DAPM_MUX("ALC Sidetone Mux", SND_SOC_NOPM, 0, 0,
+ &wm9712_alc_mux_controls),
+SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0,
+ &wm9712_out3_mux_controls),
+SND_SOC_DAPM_MUX("Speaker Mux", SND_SOC_NOPM, 0, 0,
+ &wm9712_spk_mux_controls),
+SND_SOC_DAPM_MUX("Capture Phone Mux", SND_SOC_NOPM, 0, 0,
+ &wm9712_capture_phone_mux_controls),
+SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0,
+ &wm9712_capture_selectl_controls),
+SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0,
+ &wm9712_capture_selectr_controls),
+SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0,
+ &wm9712_mic_src_controls),
+SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0,
+ &wm9712_diff_sel_controls),
+SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_INT_PAGING, 9, 1,
+ &wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls),
+ mixer_event, SND_SOC_DAPM_POST_REG),
+SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_INT_PAGING, 8, 1,
+ &wm9712_hpr_mixer_controls[0], ARRAY_SIZE(wm9712_hpr_mixer_controls),
+ mixer_event, SND_SOC_DAPM_POST_REG),
+SND_SOC_DAPM_MIXER("Phone Mixer", AC97_INT_PAGING, 6, 1,
+ &wm9712_phone_mixer_controls[0], ARRAY_SIZE(wm9712_phone_mixer_controls)),
+SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_INT_PAGING, 7, 1,
+ &wm9712_speaker_mixer_controls[0],
+ ARRAY_SIZE(wm9712_speaker_mixer_controls)),
+SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_INT_PAGING, 14, 1),
+SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_INT_PAGING, 13, 1),
+SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_INT_PAGING, 12, 1),
+SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_INT_PAGING, 11, 1),
+SND_SOC_DAPM_PGA("Headphone PGA", AC97_INT_PAGING, 4, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Speaker PGA", AC97_INT_PAGING, 3, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0),
+SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1),
+SND_SOC_DAPM_OUTPUT("MONOOUT"),
+SND_SOC_DAPM_OUTPUT("HPOUTL"),
+SND_SOC_DAPM_OUTPUT("HPOUTR"),
+SND_SOC_DAPM_OUTPUT("LOUT2"),
+SND_SOC_DAPM_OUTPUT("ROUT2"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+SND_SOC_DAPM_INPUT("LINEINL"),
+SND_SOC_DAPM_INPUT("LINEINR"),
+SND_SOC_DAPM_INPUT("PHONE"),
+SND_SOC_DAPM_INPUT("PCBEEP"),
+SND_SOC_DAPM_INPUT("MIC1"),
+SND_SOC_DAPM_INPUT("MIC2"),
+};
+
+static const char *audio_map[][3] = {
+ /* virtual mixer - mixes left & right channels for spk and mono */
+ {"AC97 Mixer", NULL, "Left DAC"},
+ {"AC97 Mixer", NULL, "Right DAC"},
+
+ /* Left HP mixer */
+ {"Left HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
+ {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Left HP Mixer", "Phone Bypass Switch", "Phone PGA"},
+ {"Left HP Mixer", "Line Bypass Switch", "Line PGA"},
+ {"Left HP Mixer", "PCM Playback Switch", "Left DAC"},
+ {"Left HP Mixer", "Mic Sidetone Switch", "Mic PGA"},
+ {"Left HP Mixer", NULL, "ALC Sidetone Mux"},
+ //{"Right HP Mixer", NULL, "HP Mixer"},
+
+ /* Right HP mixer */
+ {"Right HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
+ {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Right HP Mixer", "Phone Bypass Switch", "Phone PGA"},
+ {"Right HP Mixer", "Line Bypass Switch", "Line PGA"},
+ {"Right HP Mixer", "PCM Playback Switch", "Right DAC"},
+ {"Right HP Mixer", "Mic Sidetone Switch", "Mic PGA"},
+ {"Right HP Mixer", NULL, "ALC Sidetone Mux"},
+
+ /* speaker mixer */
+ {"Speaker Mixer", "PCBeep Bypass Switch", "PCBEEP"},
+ {"Speaker Mixer", "Line Bypass Switch", "Line PGA"},
+ {"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"},
+ {"Speaker Mixer", "Phone Bypass Switch", "Phone PGA"},
+ {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"},
+
+ /* Phone mixer */
+ {"Phone Mixer", "PCBeep Bypass Switch", "PCBEEP"},
+ {"Phone Mixer", "Line Bypass Switch", "Line PGA"},
+ {"Phone Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Phone Mixer", "PCM Playback Switch", "AC97 Mixer"},
+ {"Phone Mixer", "Mic 1 Sidetone Switch", "Mic PGA"},
+ {"Phone Mixer", "Mic 2 Sidetone Switch", "Mic PGA"},
+
+ /* inputs */
+ {"Line PGA", NULL, "LINEINL"},
+ {"Line PGA", NULL, "LINEINR"},
+ {"Phone PGA", NULL, "PHONE"},
+ {"Mic PGA", NULL, "MIC1"},
+ {"Mic PGA", NULL, "MIC2"},
+
+ /* left capture selector */
+ {"Left Capture Select", "Mic", "MIC1"},
+ {"Left Capture Select", "Speaker Mixer", "Speaker Mixer"},
+ {"Left Capture Select", "Line", "LINEINL"},
+ {"Left Capture Select", "Headphone Mixer", "Left HP Mixer"},
+ {"Left Capture Select", "Phone Mixer", "Phone Mixer"},
+ {"Left Capture Select", "Phone", "PHONE"},
+
+ /* right capture selector */
+ {"Right Capture Select", "Mic", "MIC2"},
+ {"Right Capture Select", "Speaker Mixer", "Speaker Mixer"},
+ {"Right Capture Select", "Line", "LINEINR"},
+ {"Right Capture Select", "Headphone Mixer", "Right HP Mixer"},
+ {"Right Capture Select", "Phone Mixer", "Phone Mixer"},
+ {"Right Capture Select", "Phone", "PHONE"},
+
+ /* ALC Sidetone */
+ {"ALC Sidetone Mux", "Stereo", "Left Capture Select"},
+ {"ALC Sidetone Mux", "Stereo", "Right Capture Select"},
+ {"ALC Sidetone Mux", "Left", "Left Capture Select"},
+ {"ALC Sidetone Mux", "Right", "Right Capture Select"},
+
+ /* ADC's */
+ {"Left ADC", NULL, "Left Capture Select"},
+ {"Right ADC", NULL, "Right Capture Select"},
+
+ /* outputs */
+ {"MONOOUT", NULL, "Phone Mixer"},
+ {"HPOUTL", NULL, "Headphone PGA"},
+ {"Headphone PGA", NULL, "Left HP Mixer"},
+ {"HPOUTR", NULL, "Headphone PGA"},
+ {"Headphone PGA", NULL, "Right HP Mixer"},
+
+ /* mono hp mixer */
+ {"Mono HP Mixer", NULL, "Left HP Mixer"},
+ {"Mono HP Mixer", NULL, "Right HP Mixer"},
+
+ /* Out3 Mux */
+ {"Out3 Mux", "Left", "Left HP Mixer"},
+ {"Out3 Mux", "Mono", "Phone Mixer"},
+ {"Out3 Mux", "Left + Right", "Mono HP Mixer"},
+ {"Out 3 PGA", NULL, "Out3 Mux"},
+ {"OUT3", NULL, "Out 3 PGA"},
+
+ /* speaker Mux */
+ {"Speaker Mux", "Speaker Mix", "Speaker Mixer"},
+ {"Speaker Mux", "Headphone Mix", "Mono HP Mixer"},
+ {"Speaker PGA", NULL, "Speaker Mux"},
+ {"LOUT2", NULL, "Speaker PGA"},
+ {"ROUT2", NULL, "Speaker PGA"},
+
+ {NULL, NULL, NULL},
+};
+
+static int wm9712_add_widgets(struct snd_soc_codec *codec)
+{
+ int i;
+
+ for(i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++) {
+ snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]);
+ }
+
+ /* set up audio path audio_mapnects */
+ for(i = 0; audio_map[i][0] != NULL; i++) {
+ snd_soc_dapm_connect_input(codec, audio_map[i][0],
+ audio_map[i][1], audio_map[i][2]);
+ }
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+
+ if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
+ reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 ||
+ reg == AC97_REC_GAIN)
+ return soc_ac97_ops.read(codec->ac97, reg);
+ else {
+ reg = reg >> 1;
+
+ if (reg > (ARRAY_SIZE(wm9712_reg)))
+ return -EIO;
+
+ return cache[reg];
+ }
+}
+
+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int val)
+{
+ u16 *cache = codec->reg_cache;
+
+ soc_ac97_ops.write(codec->ac97, reg, val);
+ reg = reg >> 1;
+ if (reg <= (ARRAY_SIZE(wm9712_reg)))
+ cache[reg] = val;
+
+ return 0;
+}
+
+static int ac97_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int reg;
+ u16 vra;
+
+ vra = ac97_read(codec, AC97_EXTENDED_STATUS);
+ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ reg = AC97_PCM_FRONT_DAC_RATE;
+ else
+ reg = AC97_PCM_LR_ADC_RATE;
+
+ return ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_aux_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 vra, xsle;
+
+ vra = ac97_read(codec, AC97_EXTENDED_STATUS);
+ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
+ xsle = ac97_read(codec, AC97_PCI_SID);
+ ac97_write(codec, AC97_PCI_SID, xsle | 0x8000);
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENODEV;
+
+ return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate);
+}
+
+#define WM9712_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+struct snd_soc_codec_dai wm9712_dai[] = {
+{
+ .name = "AC97 HiFi",
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM9712_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "HiFi Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM9712_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .prepare = ac97_prepare,},
+},
+{
+ .name = "AC97 Aux",
+ .playback = {
+ .stream_name = "Aux Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = WM9712_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .prepare = ac97_aux_prepare,},
+}
+};
+EXPORT_SYMBOL_GPL(wm9712_dai);
+
+static int wm9712_dapm_event(struct snd_soc_codec *codec, int event)
+{
+ u16 reg;
+
+ switch (event) {
+ case SNDRV_CTL_POWER_D0: /* full On */
+ /* liam - maybe enable thermal shutdown */
+ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xdfff;
+ ac97_write(codec, AC97_EXTENDED_MID, reg);
+ break;
+ case SNDRV_CTL_POWER_D1: /* partial On */
+ case SNDRV_CTL_POWER_D2: /* partial On */
+ break;
+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+ /* enable master bias and vmid */
+ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xbbff;
+ ac97_write(codec, AC97_EXTENDED_MID, reg);
+ ac97_write(codec, AC97_POWERDOWN, 0x0000);
+ break;
+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+ /* disable everything including AC link */
+ ac97_write(codec, AC97_EXTENDED_MID, 0xffff);
+ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
+ ac97_write(codec, AC97_POWERDOWN, 0xffff);
+ break;
+ }
+ codec->dapm_state = event;
+ return 0;
+}
+
+static int wm9712_reset(struct snd_soc_codec *codec, int try_warm)
+{
+ if (try_warm && soc_ac97_ops.warm_reset) {
+ soc_ac97_ops.warm_reset(codec->ac97);
+ if (!(ac97_read(codec, 0) & 0x8000))
+ return 1;
+ }
+
+ soc_ac97_ops.reset(codec->ac97);
+ if (ac97_read(codec, 0) & 0x8000)
+ goto err;
+ return 0;
+
+err:
+ printk(KERN_ERR "WM9712 AC97 reset failed\n");
+ return -EIO;
+}
+
+static int wm9712_soc_suspend(struct platform_device *pdev,
+ pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+ return 0;
+}
+
+static int wm9712_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i, ret;
+ u16 *cache = codec->reg_cache;
+
+ ret = wm9712_reset(codec, 1);
+ if (ret < 0){
+ printk(KERN_ERR "could not reset AC97 codec\n");
+ return ret;
+ }
+
+ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+
+ if (ret == 0) {
+ /* Sync reg_cache with the hardware after cold reset */
+ for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i+=2) {
+ if (i == AC97_INT_PAGING || i == AC97_POWERDOWN ||
+ (i > 0x58 && i != 0x5c))
+ continue;
+ soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
+ }
+ }
+
+ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0)
+ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D0);
+
+ return ret;
+}
+
+static int wm9712_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION);
+
+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (socdev->codec == NULL)
+ return -ENOMEM;
+ codec = socdev->codec;
+ mutex_init(&codec->mutex);
+
+ codec->reg_cache =
+ kzalloc(sizeof(u16) * ARRAY_SIZE(wm9712_reg), GFP_KERNEL);
+ if (codec->reg_cache == NULL) {
+ ret = -ENOMEM;
+ goto cache_err;
+ }
+ memcpy(codec->reg_cache, wm9712_reg, sizeof(u16) * ARRAY_SIZE(wm9712_reg));
+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm9712_reg);
+ codec->reg_cache_step = 2;
+
+ codec->name = "WM9712";
+ codec->owner = THIS_MODULE;
+ codec->dai = wm9712_dai;
+ codec->num_dai = ARRAY_SIZE(wm9712_dai);
+ codec->write = ac97_write;
+ codec->read = ac97_read;
+ codec->dapm_event = wm9712_dapm_event;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+ if (ret < 0) {
+ printk(KERN_ERR "wm9712: failed to register AC97 codec\n");
+ goto codec_err;
+ }
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0)
+ goto pcm_err;
+
+ ret = wm9712_reset(codec, 0);
+ if (ret < 0) {
+ printk(KERN_ERR "AC97 link error\n");
+ goto reset_err;
+ }
+
+ /* set alc mux to none */
+ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000);
+
+ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+ wm9712_add_controls(codec);
+ wm9712_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "wm9712: failed to register card\n");
+ goto reset_err;
+ }
+
+ return 0;
+
+reset_err:
+ snd_soc_free_pcms(socdev);
+
+pcm_err:
+ snd_soc_free_ac97_codec(codec);
+
+codec_err:
+ kfree(codec->reg_cache);
+
+cache_err:
+ kfree(socdev->codec);
+ socdev->codec = NULL;
+ return ret;
+}
+
+static int wm9712_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec == NULL)
+ return 0;
+
+ snd_soc_dapm_free(socdev);
+ snd_soc_free_pcms(socdev);
+ snd_soc_free_ac97_codec(codec);
+ kfree(codec->reg_cache);
+ kfree(codec);
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm9712 = {
+ .probe = wm9712_soc_probe,
+ .remove = wm9712_soc_remove,
+ .suspend = wm9712_soc_suspend,
+ .resume = wm9712_soc_resume,
+};
+
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9712);
+
+MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm9712.h b/sound/soc/codecs/wm9712.h
new file mode 100644
index 00000000000..719105d61e6
--- /dev/null
+++ b/sound/soc/codecs/wm9712.h
@@ -0,0 +1,14 @@
+/*
+ * wm9712.h -- WM9712 Soc Audio driver
+ */
+
+#ifndef _WM9712_H
+#define _WM9712_H
+
+#define WM9712_DAI_AC97_HIFI 0
+#define WM9712_DAI_AC97_AUX 1
+
+extern struct snd_soc_codec_dai wm9712_dai[2];
+extern struct snd_soc_codec_device soc_codec_dev_wm9712;
+
+#endif
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
new file mode 100644
index 00000000000..579e1c8d2b2
--- /dev/null
+++ b/sound/soc/pxa/Kconfig
@@ -0,0 +1,60 @@
+menu "SoC Audio for the Intel PXA2xx"
+
+config SND_PXA2XX_SOC
+ tristate "SoC Audio for the Intel PXA2xx chip"
+ depends on ARCH_PXA && SND
+ select SND_PCM
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the PXA2xx AC97, I2S or SSP interface. You will also need
+ to select the audio interfaces to support below.
+
+config SND_PXA2XX_AC97
+ tristate
+ select SND_AC97_CODEC
+
+config SND_PXA2XX_SOC_AC97
+ tristate
+ select AC97_BUS
+ select SND_SOC_AC97_BUS
+
+config SND_PXA2XX_SOC_I2S
+ tristate
+
+config SND_PXA2XX_SOC_CORGI
+ tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
+ depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx
+ select SND_PXA2XX_SOC_I2S
+ select SND_SOC_WM8731
+ help
+ Say Y if you want to add support for SoC audio on Sharp
+ Zaurus SL-C7x0 models (Corgi, Shepherd, Husky).
+
+config SND_PXA2XX_SOC_SPITZ
+ tristate "SoC Audio support for Sharp Zaurus SL-Cxx00"
+ depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00
+ select SND_PXA2XX_SOC_I2S
+ select SND_SOC_WM8750
+ help
+ Say Y if you want to add support for SoC audio on Sharp
+ Zaurus SL-Cxx00 models (Spitz, Borzoi and Akita).
+
+config SND_PXA2XX_SOC_POODLE
+ tristate "SoC Audio support for Poodle"
+ depends on SND_PXA2XX_SOC && MACH_POODLE
+ select SND_PXA2XX_SOC_I2S
+ select SND_SOC_WM8731
+ help
+ Say Y if you want to add support for SoC audio on Sharp
+ Zaurus SL-5600 model (Poodle).
+
+config SND_PXA2XX_SOC_TOSA
+ tristate "SoC AC97 Audio support for Tosa"
+ depends on SND_PXA2XX_SOC && MACH_TOSA
+ select SND_PXA2XX_SOC_AC97
+ select SND_SOC_WM9712
+ help
+ Say Y if you want to add support for SoC audio on Sharp
+ Zaurus SL-C6000x models (Tosa).
+
+endmenu
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
new file mode 100644
index 00000000000..78e0d6b07d1
--- /dev/null
+++ b/sound/soc/pxa/Makefile
@@ -0,0 +1,20 @@
+# PXA Platform Support
+snd-soc-pxa2xx-objs := pxa2xx-pcm.o
+snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
+snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
+
+obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
+obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
+obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o
+
+# PXA Machine Support
+snd-soc-corgi-objs := corgi.o
+snd-soc-poodle-objs := poodle.o
+snd-soc-tosa-objs := tosa.o
+snd-soc-spitz-objs := spitz.o
+
+obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
+obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
+obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
+obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
+
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
new file mode 100644
index 00000000000..5ee51a994ac
--- /dev/null
+++ b/sound/soc/pxa/corgi.c
@@ -0,0 +1,383 @@
+/*
+ * corgi.c -- SoC audio for Corgi
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Richard Purdie <richard@openedhand.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.
+ *
+ * Revision history
+ * 30th Nov 2005 Initial version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <asm/hardware/scoop.h>
+#include <asm/arch/pxa-regs.h>
+#include <asm/arch/hardware.h>
+#include <asm/arch/corgi.h>
+#include <asm/arch/audio.h>
+
+#include "../codecs/wm8731.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+
+#define CORGI_HP 0
+#define CORGI_MIC 1
+#define CORGI_LINE 2
+#define CORGI_HEADSET 3
+#define CORGI_HP_OFF 4
+#define CORGI_SPK_ON 0
+#define CORGI_SPK_OFF 1
+
+ /* audio clock in Hz - rounded from 12.235MHz */
+#define CORGI_AUDIO_CLOCK 12288000
+
+static int corgi_jack_func;
+static int corgi_spk_func;
+
+static void corgi_ext_control(struct snd_soc_codec *codec)
+{
+ int spk = 0, mic = 0, line = 0, hp = 0, hs = 0;
+
+ /* set up jack connection */
+ switch (corgi_jack_func) {
+ case CORGI_HP:
+ hp = 1;
+ /* set = unmute headphone */
+ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
+ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ break;
+ case CORGI_MIC:
+ mic = 1;
+ /* reset = mute headphone */
+ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
+ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ break;
+ case CORGI_LINE:
+ line = 1;
+ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
+ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ break;
+ case CORGI_HEADSET:
+ hs = 1;
+ mic = 1;
+ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
+ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ break;
+ }
+
+ if (corgi_spk_func == CORGI_SPK_ON)
+ spk = 1;
+
+ /* set the enpoints to their new connetion states */
+ snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk);
+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", mic);
+ snd_soc_dapm_set_endpoint(codec, "Line Jack", line);
+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp);
+ snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs);
+
+ /* signal a DAPM event */
+ snd_soc_dapm_sync_endpoints(codec);
+}
+
+static int corgi_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->codec;
+
+ /* check the jack status at stream startup */
+ corgi_ext_control(codec);
+ return 0;
+}
+
+/* we need to unmute the HP at shutdown as the mute burns power on corgi */
+static int corgi_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->codec;
+
+ /* set = unmute headphone */
+ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
+ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ return 0;
+}
+
+static int corgi_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_codec_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int clk = 0;
+ int ret = 0;
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ case 48000:
+ case 96000:
+ clk = 12288000;
+ break;
+ case 11025:
+ case 22050:
+ case 44100:
+ clk = 11289600;
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as input (unused) */
+ ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops corgi_ops = {
+ .startup = corgi_startup,
+ .hw_params = corgi_hw_params,
+ .shutdown = corgi_shutdown,
+};
+
+static int corgi_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = corgi_jack_func;
+ return 0;
+}
+
+static int corgi_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (corgi_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ corgi_jack_func = ucontrol->value.integer.value[0];
+ corgi_ext_control(codec);
+ return 1;
+}
+
+static int corgi_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = corgi_spk_func;
+ return 0;
+}
+
+static int corgi_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (corgi_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ corgi_spk_func = ucontrol->value.integer.value[0];
+ corgi_ext_control(codec);
+ return 1;
+}
+
+static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
+ else
+ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
+
+ return 0;
+}
+
+static int corgi_mic_event(struct snd_soc_dapm_widget *w, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS);
+ else
+ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS);
+
+ return 0;
+}
+
+/* corgi machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_HP("Headphone Jack", NULL),
+SND_SOC_DAPM_MIC("Mic Jack", corgi_mic_event),
+SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event),
+SND_SOC_DAPM_LINE("Line Jack", NULL),
+SND_SOC_DAPM_HP("Headset Jack", NULL),
+};
+
+/* Corgi machine audio map (connections to the codec pins) */
+static const char *audio_map[][3] = {
+
+ /* headset Jack - in = micin, out = LHPOUT*/
+ {"Headset Jack", NULL, "LHPOUT"},
+
+ /* headphone connected to LHPOUT1, RHPOUT1 */
+ {"Headphone Jack", NULL, "LHPOUT"},
+ {"Headphone Jack", NULL, "RHPOUT"},
+
+ /* speaker connected to LOUT, ROUT */
+ {"Ext Spk", NULL, "ROUT"},
+ {"Ext Spk", NULL, "LOUT"},
+
+ /* mic is connected to MICIN (via right channel of headphone jack) */
+ {"MICIN", NULL, "Mic Jack"},
+
+ /* Same as the above but no mic bias for line signals */
+ {"MICIN", NULL, "Line Jack"},
+
+ {NULL, NULL, NULL},
+};
+
+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
+ "Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum corgi_enum[] = {
+ SOC_ENUM_SINGLE_EXT(5, jack_function),
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new wm8731_corgi_controls[] = {
+ SOC_ENUM_EXT("Jack Function", corgi_enum[0], corgi_get_jack,
+ corgi_set_jack),
+ SOC_ENUM_EXT("Speaker Function", corgi_enum[1], corgi_get_spk,
+ corgi_set_spk),
+};
+
+/*
+ * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device
+ */
+static int corgi_wm8731_init(struct snd_soc_codec *codec)
+{
+ int i, err;
+
+ snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
+ snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
+
+ /* Add corgi specific controls */
+ for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8731_corgi_controls[i],codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ /* Add corgi specific widgets */
+ for(i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) {
+ snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
+ }
+
+ /* Set up corgi specific audio path audio_map */
+ for(i = 0; audio_map[i][0] != NULL; i++) {
+ snd_soc_dapm_connect_input(codec, audio_map[i][0],
+ audio_map[i][1], audio_map[i][2]);
+ }
+
+ snd_soc_dapm_sync_endpoints(codec);
+ return 0;
+}
+
+/* corgi digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link corgi_dai = {
+ .name = "WM8731",
+ .stream_name = "WM8731",
+ .cpu_dai = &pxa_i2s_dai,
+ .codec_dai = &wm8731_dai,
+ .init = corgi_wm8731_init,
+ .ops = &corgi_ops,
+};
+
+/* corgi audio machine driver */
+static struct snd_soc_machine snd_soc_machine_corgi = {
+ .name = "Corgi",
+ .dai_link = &corgi_dai,
+ .num_links = 1,
+};
+
+/* corgi audio private data */
+static struct wm8731_setup_data corgi_wm8731_setup = {
+ .i2c_address = 0x1b,
+};
+
+/* corgi audio subsystem */
+static struct snd_soc_device corgi_snd_devdata = {
+ .machine = &snd_soc_machine_corgi,
+ .platform = &pxa2xx_soc_platform,
+ .codec_dev = &soc_codec_dev_wm8731,
+ .codec_data = &corgi_wm8731_setup,
+};
+
+static struct platform_device *corgi_snd_device;
+
+static int __init corgi_init(void)
+{
+ int ret;
+
+ if (!(machine_is_corgi() || machine_is_shepherd() || machine_is_husky()))
+ return -ENODEV;
+
+ corgi_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!corgi_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(corgi_snd_device, &corgi_snd_devdata);
+ corgi_snd_devdata.dev = &corgi_snd_device->dev;
+ ret = platform_device_add(corgi_snd_device);
+
+ if (ret)
+ platform_device_put(corgi_snd_device);
+
+ return ret;
+}
+
+static void __exit corgi_exit(void)
+{
+ platform_device_unregister(corgi_snd_device);
+}
+
+module_init(corgi_init);
+module_exit(corgi_exit);
+
+/* Module information */
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Corgi");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
new file mode 100644
index 00000000000..0915cf74042
--- /dev/null
+++ b/sound/soc/pxa/poodle.c
@@ -0,0 +1,352 @@
+/*
+ * poodle.c -- SoC audio for Poodle
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Richard Purdie <richard@openedhand.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <asm/hardware/locomo.h>
+#include <asm/arch/pxa-regs.h>
+#include <asm/arch/hardware.h>
+#include <asm/arch/poodle.h>
+#include <asm/arch/audio.h>
+
+#include "../codecs/wm8731.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+
+#define POODLE_HP 1
+#define POODLE_HP_OFF 0
+#define POODLE_SPK_ON 1
+#define POODLE_SPK_OFF 0
+
+ /* audio clock in Hz - rounded from 12.235MHz */
+#define POODLE_AUDIO_CLOCK 12288000
+
+static int poodle_jack_func;
+static int poodle_spk_func;
+
+static void poodle_ext_control(struct snd_soc_codec *codec)
+{
+ int spk = 0;
+
+ /* set up jack connection */
+ if (poodle_jack_func == POODLE_HP) {
+ /* set = unmute headphone */
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_L, 1);
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_R, 1);
+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
+ } else {
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_L, 0);
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_R, 0);
+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
+ }
+
+ if (poodle_spk_func == POODLE_SPK_ON)
+ spk = 1;
+
+ /* set the enpoints to their new connetion states */
+ snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk);
+
+ /* signal a DAPM event */
+ snd_soc_dapm_sync_endpoints(codec);
+}
+
+static int poodle_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->codec;
+
+ /* check the jack status at stream startup */
+ poodle_ext_control(codec);
+ return 0;
+}
+
+/* we need to unmute the HP at shutdown as the mute burns power on poodle */
+static int poodle_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->codec;
+
+ /* set = unmute headphone */
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_L, 1);
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_R, 1);
+ return 0;
+}
+
+static int poodle_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_codec_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int clk = 0;
+ int ret = 0;
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ case 48000:
+ case 96000:
+ clk = 12288000;
+ break;
+ case 11025:
+ case 22050:
+ case 44100:
+ clk = 11289600;
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as input (unused) */
+ ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops poodle_ops = {
+ .startup = poodle_startup,
+ .hw_params = poodle_hw_params,
+ .shutdown = poodle_shutdown,
+};
+
+static int poodle_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = poodle_jack_func;
+ return 0;
+}
+
+static int poodle_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (poodle_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ poodle_jack_func = ucontrol->value.integer.value[0];
+ poodle_ext_control(codec);
+ return 1;
+}
+
+static int poodle_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = poodle_spk_func;
+ return 0;
+}
+
+static int poodle_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (poodle_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ poodle_spk_func = ucontrol->value.integer.value[0];
+ poodle_ext_control(codec);
+ return 1;
+}
+
+static int poodle_amp_event(struct snd_soc_dapm_widget *w, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_AMP_ON, 0);
+ else
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_AMP_ON, 1);
+
+ return 0;
+}
+
+/* poodle machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_HP("Headphone Jack", NULL),
+SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event),
+};
+
+/* Corgi machine audio_mapnections to the codec pins */
+static const char *audio_map[][3] = {
+
+ /* headphone connected to LHPOUT1, RHPOUT1 */
+ {"Headphone Jack", NULL, "LHPOUT"},
+ {"Headphone Jack", NULL, "RHPOUT"},
+
+ /* speaker connected to LOUT, ROUT */
+ {"Ext Spk", NULL, "ROUT"},
+ {"Ext Spk", NULL, "LOUT"},
+
+ {NULL, NULL, NULL},
+};
+
+static const char *jack_function[] = {"Off", "Headphone"};
+static const char *spk_function[] = {"Off", "On"};
+static const struct soc_enum poodle_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, jack_function),
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const snd_kcontrol_new_t wm8731_poodle_controls[] = {
+ SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack,
+ poodle_set_jack),
+ SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk,
+ poodle_set_spk),
+};
+
+/*
+ * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device
+ */
+static int poodle_wm8731_init(struct snd_soc_codec *codec)
+{
+ int i, err;
+
+ snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
+ snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
+ snd_soc_dapm_set_endpoint(codec, "MICIN", 1);
+
+ /* Add poodle specific controls */
+ for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8731_poodle_controls[i],codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ /* Add poodle specific widgets */
+ for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) {
+ snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
+ }
+
+ /* Set up poodle specific audio path audio_map */
+ for (i = 0; audio_map[i][0] != NULL; i++) {
+ snd_soc_dapm_connect_input(codec, audio_map[i][0],
+ audio_map[i][1], audio_map[i][2]);
+ }
+
+ snd_soc_dapm_sync_endpoints(codec);
+ return 0;
+}
+
+/* poodle digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link poodle_dai = {
+ .name = "WM8731",
+ .stream_name = "WM8731",
+ .cpu_dai = &pxa_i2s_dai,
+ .codec_dai = &wm8731_dai,
+ .init = poodle_wm8731_init,
+ .ops = &poodle_ops,
+};
+
+/* poodle audio machine driver */
+static struct snd_soc_machine snd_soc_machine_poodle = {
+ .name = "Poodle",
+ .dai_link = &poodle_dai,
+ .num_links = 1,
+};
+
+/* poodle audio private data */
+static struct wm8731_setup_data poodle_wm8731_setup = {
+ .i2c_address = 0x1b,
+};
+
+/* poodle audio subsystem */
+static struct snd_soc_device poodle_snd_devdata = {
+ .machine = &snd_soc_machine_poodle,
+ .platform = &pxa2xx_soc_platform,
+ .codec_dev = &soc_codec_dev_wm8731,
+ .codec_data = &poodle_wm8731_setup,
+};
+
+static struct platform_device *poodle_snd_device;
+
+static int __init poodle_init(void)
+{
+ int ret;
+
+ if (!machine_is_poodle())
+ return -ENODEV;
+
+ locomo_gpio_set_dir(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_AMP_ON, 0);
+ /* should we mute HP at startup - burning power ?*/
+ locomo_gpio_set_dir(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_L, 0);
+ locomo_gpio_set_dir(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_R, 0);
+
+ poodle_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!poodle_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(poodle_snd_device, &poodle_snd_devdata);
+ poodle_snd_devdata.dev = &poodle_snd_device->dev;
+ ret = platform_device_add(poodle_snd_device);
+
+ if (ret)
+ platform_device_put(poodle_snd_device);
+
+ return ret;
+}
+
+static void __exit poodle_exit(void)
+{
+ platform_device_unregister(poodle_snd_device);
+}
+
+module_init(poodle_init);
+module_exit(poodle_exit);
+
+/* Module information */
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Poodle");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
new file mode 100644
index 00000000000..1bbbeff84ef
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -0,0 +1,431 @@
+/*
+ * linux/sound/pxa2xx-ac97.c -- AC97 support for the Intel PXA2xx chip.
+ *
+ * Author: Nicolas Pitre
+ * Created: Dec 02, 2004
+ * Copyright: MontaVista Software Inc.
+ *
+ * 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/platform_device.h>
+#include <linux/interrupt.h>
+#include <linux/wait.h>
+#include <linux/delay.h>
+
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <asm/irq.h>
+#include <linux/mutex.h>
+#include <asm/hardware.h>
+#include <asm/arch/pxa-regs.h>
+#include <asm/arch/audio.h>
+
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static DEFINE_MUTEX(car_mutex);
+static DECLARE_WAIT_QUEUE_HEAD(gsr_wq);
+static volatile long gsr_bits;
+
+/*
+ * Beware PXA27x bugs:
+ *
+ * o Slot 12 read from modem space will hang controller.
+ * o CDONE, SDONE interrupt fails after any slot 12 IO.
+ *
+ * We therefore have an hybrid approach for waiting on SDONE (interrupt or
+ * 1 jiffy timeout if interrupt never comes).
+ */
+
+static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ unsigned short val = -1;
+ volatile u32 *reg_addr;
+
+ mutex_lock(&car_mutex);
+
+ /* set up primary or secondary codec/modem space */
+#ifdef CONFIG_PXA27x
+ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
+#else
+ if (reg == AC97_GPIO_STATUS)
+ reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE;
+ else
+ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
+#endif
+ reg_addr += (reg >> 1);
+
+#ifndef CONFIG_PXA27x
+ if (reg == AC97_GPIO_STATUS) {
+ /* read from controller cache */
+ val = *reg_addr;
+ goto out;
+ }
+#endif
+
+ /* start read access across the ac97 link */
+ GSR = GSR_CDONE | GSR_SDONE;
+ gsr_bits = 0;
+ val = *reg_addr;
+
+ wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1);
+ if (!((GSR | gsr_bits) & GSR_SDONE)) {
+ printk(KERN_ERR "%s: read error (ac97_reg=%x GSR=%#lx)\n",
+ __FUNCTION__, reg, GSR | gsr_bits);
+ val = -1;
+ goto out;
+ }
+
+ /* valid data now */
+ GSR = GSR_CDONE | GSR_SDONE;
+ gsr_bits = 0;
+ val = *reg_addr;
+ /* but we've just started another cycle... */
+ wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1);
+
+out: mutex_unlock(&car_mutex);
+ return val;
+}
+
+static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ volatile u32 *reg_addr;
+
+ mutex_lock(&car_mutex);
+
+ /* set up primary or secondary codec/modem space */
+#ifdef CONFIG_PXA27x
+ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
+#else
+ if (reg == AC97_GPIO_STATUS)
+ reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE;
+ else
+ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
+#endif
+ reg_addr += (reg >> 1);
+
+ GSR = GSR_CDONE | GSR_SDONE;
+ gsr_bits = 0;
+ *reg_addr = val;
+ wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_CDONE, 1);
+ if (!((GSR | gsr_bits) & GSR_CDONE))
+ printk(KERN_ERR "%s: write error (ac97_reg=%x GSR=%#lx)\n",
+ __FUNCTION__, reg, GSR | gsr_bits);
+
+ mutex_unlock(&car_mutex);
+}
+
+static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+ gsr_bits = 0;
+
+#ifdef CONFIG_PXA27x
+ /* warm reset broken on Bulverde,
+ so manually keep AC97 reset high */
+ pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH);
+ udelay(10);
+ GCR |= GCR_WARM_RST;
+ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
+ udelay(500);
+#else
+ GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN;
+ wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1);
+#endif
+
+ if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)))
+ printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n",
+ __FUNCTION__, gsr_bits);
+
+ GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN);
+ GCR |= GCR_SDONE_IE|GCR_CDONE_IE;
+}
+
+static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+ GCR &= GCR_COLD_RST; /* clear everything but nCRST */
+ GCR &= ~GCR_COLD_RST; /* then assert nCRST */
+
+ gsr_bits = 0;
+#ifdef CONFIG_PXA27x
+ /* PXA27x Developers Manual section 13.5.2.2.1 */
+ pxa_set_cken(1 << 31, 1);
+ udelay(5);
+ pxa_set_cken(1 << 31, 0);
+ GCR = GCR_COLD_RST;
+ udelay(50);
+#else
+ GCR = GCR_COLD_RST;
+ GCR |= GCR_CDONE_IE|GCR_SDONE_IE;
+ wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1);
+#endif
+
+ if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)))
+ printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n",
+ __FUNCTION__, gsr_bits);
+
+ GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN);
+ GCR |= GCR_SDONE_IE|GCR_CDONE_IE;
+}
+
+static irqreturn_t pxa2xx_ac97_irq(int irq, void *dev_id)
+{
+ long status;
+
+ status = GSR;
+ if (status) {
+ GSR = status;
+ gsr_bits |= status;
+ wake_up(&gsr_wq);
+
+#ifdef CONFIG_PXA27x
+ /* Although we don't use those we still need to clear them
+ since they tend to spuriously trigger when MMC is used
+ (hardware bug? go figure)... */
+ MISR = MISR_EOC;
+ PISR = PISR_EOC;
+ MCSR = MCSR_EOC;
+#endif
+
+ return IRQ_HANDLED;
+ }
+
+ return IRQ_NONE;
+}
+
+struct snd_ac97_bus_ops soc_ac97_ops = {
+ .read = pxa2xx_ac97_read,
+ .write = pxa2xx_ac97_write,
+ .warm_reset = pxa2xx_ac97_warm_reset,
+ .reset = pxa2xx_ac97_cold_reset,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = {
+ .name = "AC97 PCM Stereo out",
+ .dev_addr = __PREG(PCDR),
+ .drcmr = &DRCMRTXPCDR,
+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
+ DCMD_BURST32 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = {
+ .name = "AC97 PCM Stereo in",
+ .dev_addr = __PREG(PCDR),
+ .drcmr = &DRCMRRXPCDR,
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST32 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = {
+ .name = "AC97 Aux PCM (Slot 5) Mono out",
+ .dev_addr = __PREG(MODR),
+ .drcmr = &DRCMRTXMODR,
+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
+ DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = {
+ .name = "AC97 Aux PCM (Slot 5) Mono in",
+ .dev_addr = __PREG(MODR),
+ .drcmr = &DRCMRRXMODR,
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = {
+ .name = "AC97 Mic PCM (Slot 6) Mono in",
+ .dev_addr = __PREG(MCDR),
+ .drcmr = &DRCMRRXMCDR,
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+#ifdef CONFIG_PM
+static int pxa2xx_ac97_suspend(struct platform_device *pdev,
+ struct snd_soc_cpu_dai *dai)
+{
+ GCR |= GCR_ACLINK_OFF;
+ pxa_set_cken(CKEN2_AC97, 0);
+ return 0;
+}
+
+static int pxa2xx_ac97_resume(struct platform_device *pdev,
+ struct snd_soc_cpu_dai *dai)
+{
+ pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
+ pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD);
+ pxa_gpio_mode(GPIO28_BITCLK_AC97_MD);
+ pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD);
+#ifdef CONFIG_PXA27x
+ /* Use GPIO 113 as AC97 Reset on Bulverde */
+ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
+#endif
+ pxa_set_cken(CKEN2_AC97, 1);
+ return 0;
+}
+
+#else
+#define pxa2xx_ac97_suspend NULL
+#define pxa2xx_ac97_resume NULL
+#endif
+
+static int pxa2xx_ac97_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL);
+ if (ret < 0)
+ goto err;
+
+ pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
+ pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD);
+ pxa_gpio_mode(GPIO28_BITCLK_AC97_MD);
+ pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD);
+#ifdef CONFIG_PXA27x
+ /* Use GPIO 113 as AC97 Reset on Bulverde */
+ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
+#endif
+ pxa_set_cken(CKEN2_AC97, 1);
+ return 0;
+
+ err:
+ if (CKEN & CKEN2_AC97) {
+ GCR |= GCR_ACLINK_OFF;
+ free_irq(IRQ_AC97, NULL);
+ pxa_set_cken(CKEN2_AC97, 0);
+ }
+ return ret;
+}
+
+static void pxa2xx_ac97_remove(struct platform_device *pdev)
+{
+ GCR |= GCR_ACLINK_OFF;
+ free_irq(IRQ_AC97, NULL);
+ pxa_set_cken(CKEN2_AC97, 0);
+}
+
+static int pxa2xx_ac97_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_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out;
+ else
+ cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_in;
+
+ return 0;
+}
+
+static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
+ else
+ cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_in;
+
+ return 0;
+}
+
+static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENODEV;
+ else
+ cpu_dai->dma_data = &pxa2xx_ac97_pcm_mic_mono_in;
+
+ return 0;
+}
+
+#define PXA2XX_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+/*
+ * There is only 1 physical AC97 interface for pxa2xx, but it
+ * has extra fifo's that can be used for aux DACs and ADCs.
+ */
+struct snd_soc_cpu_dai pxa_ac97_dai[] = {
+{
+ .name = "pxa2xx-ac97",
+ .id = 0,
+ .type = SND_SOC_DAI_AC97,
+ .probe = pxa2xx_ac97_probe,
+ .remove = pxa2xx_ac97_remove,
+ .suspend = pxa2xx_ac97_suspend,
+ .resume = pxa2xx_ac97_resume,
+ .playback = {
+ .stream_name = "AC97 Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PXA2XX_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "AC97 Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PXA2XX_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .hw_params = pxa2xx_ac97_hw_params,},
+},
+{
+ .name = "pxa2xx-ac97-aux",
+ .id = 1,
+ .type = SND_SOC_DAI_AC97,
+ .playback = {
+ .stream_name = "AC97 Aux Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = PXA2XX_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "AC97 Aux Capture",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = PXA2XX_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .hw_params = pxa2xx_ac97_hw_aux_params,},
+},
+{
+ .name = "pxa2xx-ac97-mic",
+ .id = 2,
+ .type = SND_SOC_DAI_AC97,
+ .capture = {
+ .stream_name = "AC97 Mic Capture",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = PXA2XX_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .hw_params = pxa2xx_ac97_hw_mic_params,},
+},
+};
+
+EXPORT_SYMBOL_GPL(pxa_ac97_dai);
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+MODULE_AUTHOR("Nicolas Pitre");
+MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h
new file mode 100644
index 00000000000..4c4b882316a
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-ac97.h
@@ -0,0 +1,22 @@
+/*
+ * linux/sound/arm/pxa2xx-ac97.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _PXA2XX_AC97_H
+#define _PXA2XX_AC97_H
+
+/* pxa2xx DAI ID's */
+#define PXA2XX_DAI_AC97_HIFI 0
+#define PXA2XX_DAI_AC97_AUX 1
+#define PXA2XX_DAI_AC97_MIC 2
+
+extern struct snd_soc_cpu_dai pxa_ac97_dai[3];
+
+/* platform data */
+extern struct snd_ac97_bus_ops pxa2xx_ac97_ops;
+
+#endif
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
new file mode 100644
index 00000000000..575a6137c04
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -0,0 +1,318 @@
+/*
+ * pxa2xx-i2s.c -- ALSA Soc Audio Layer
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.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.
+ *
+ * Revision history
+ * 12th Aug 2005 Initial version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <asm/hardware.h>
+#include <asm/arch/pxa-regs.h>
+#include <asm/arch/audio.h>
+
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+
+struct pxa_i2s_port {
+ u32 sadiv;
+ u32 sacr0;
+ u32 sacr1;
+ u32 saimr;
+ int master;
+ u32 fmt;
+};
+static struct pxa_i2s_port pxa_i2s;
+
+static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = {
+ .name = "I2S PCM Stereo out",
+ .dev_addr = __PREG(SADR),
+ .drcmr = &DRCMRTXSADR,
+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
+ DCMD_BURST32 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = {
+ .name = "I2S PCM Stereo in",
+ .dev_addr = __PREG(SADR),
+ .drcmr = &DRCMRRXSADR,
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST32 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_gpio gpio_bus[] = {
+ { /* I2S SoC Slave */
+ .rx = GPIO29_SDATA_IN_I2S_MD,
+ .tx = GPIO30_SDATA_OUT_I2S_MD,
+ .clk = GPIO28_BITCLK_IN_I2S_MD,
+ .frm = GPIO31_SYNC_I2S_MD,
+ },
+ { /* I2S SoC Master */
+#ifdef CONFIG_PXA27x
+ .sys = GPIO113_I2S_SYSCLK_MD,
+#else
+ .sys = GPIO32_SYSCLK_I2S_MD,
+#endif
+ .rx = GPIO29_SDATA_IN_I2S_MD,
+ .tx = GPIO30_SDATA_OUT_I2S_MD,
+ .clk = GPIO28_BITCLK_OUT_I2S_MD,
+ .frm = GPIO31_SYNC_I2S_MD,
+ },
+};
+
+static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ if (!cpu_dai->active) {
+ SACR0 |= SACR0_RST;
+ SACR0 = 0;
+ }
+
+ return 0;
+}
+
+/* wait for I2S controller to be ready */
+static int pxa_i2s_wait(void)
+{
+ int i;
+
+ /* flush the Rx FIFO */
+ for(i = 0; i < 16; i++)
+ SADR;
+ return 0;
+}
+
+static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
+ unsigned int fmt)
+{
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ pxa_i2s.fmt = 0;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ pxa_i2s.fmt = SACR1_AMSL;
+ break;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ pxa_i2s.master = 1;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ pxa_i2s.master = 0;
+ break;
+ default:
+ break;
+ }
+ return 0;
+}
+
+static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ if (clk_id != PXA2XX_I2S_SYSCLK)
+ return -ENODEV;
+
+ if (pxa_i2s.master && dir == SND_SOC_CLOCK_OUT)
+ pxa_gpio_mode(gpio_bus[pxa_i2s.master].sys);
+
+ return 0;
+}
+
+static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx);
+ pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx);
+ pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm);
+ pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk);
+ pxa_set_cken(CKEN8_I2S, 1);
+ pxa_i2s_wait();
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out;
+ else
+ cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_in;
+
+ /* is port used by another stream */
+ if (!(SACR0 & SACR0_ENB)) {
+
+ SACR0 = 0;
+ SACR1 = 0;
+ if (pxa_i2s.master)
+ SACR0 |= SACR0_BCKD;
+
+ SACR0 |= SACR0_RFTH(14) | SACR0_TFTH(1);
+ SACR1 |= pxa_i2s.fmt;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ SAIMR |= SAIMR_TFS;
+ else
+ SAIMR |= SAIMR_RFS;
+
+ switch (params_rate(params)) {
+ case 8000:
+ SADIV = 0x48;
+ break;
+ case 11025:
+ SADIV = 0x34;
+ break;
+ case 16000:
+ SADIV = 0x24;
+ break;
+ case 22050:
+ SADIV = 0x1a;
+ break;
+ case 44100:
+ SADIV = 0xd;
+ break;
+ case 48000:
+ SADIV = 0xc;
+ break;
+ case 96000: /* not in manual and possibly slightly inaccurate */
+ SADIV = 0x6;
+ break;
+ }
+
+ return 0;
+}
+
+static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ SACR0 |= SACR0_ENB;
+ break;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream)
+{
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ SACR1 |= SACR1_DRPL;
+ SAIMR &= ~SAIMR_TFS;
+ } else {
+ SACR1 |= SACR1_DREC;
+ SAIMR &= ~SAIMR_RFS;
+ }
+
+ if (SACR1 & (SACR1_DREC | SACR1_DRPL)) {
+ SACR0 &= ~SACR0_ENB;
+ pxa_i2s_wait();
+ pxa_set_cken(CKEN8_I2S, 0);
+ }
+}
+
+#ifdef CONFIG_PM
+static int pxa2xx_i2s_suspend(struct platform_device *dev,
+ struct snd_soc_cpu_dai *dai)
+{
+ if (!dai->active)
+ return 0;
+
+ /* store registers */
+ pxa_i2s.sacr0 = SACR0;
+ pxa_i2s.sacr1 = SACR1;
+ pxa_i2s.saimr = SAIMR;
+ pxa_i2s.sadiv = SADIV;
+
+ /* deactivate link */
+ SACR0 &= ~SACR0_ENB;
+ pxa_i2s_wait();
+ return 0;
+}
+
+static int pxa2xx_i2s_resume(struct platform_device *pdev,
+ struct snd_soc_cpu_dai *dai)
+{
+ if (!dai->active)
+ return 0;
+
+ pxa_i2s_wait();
+
+ SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB;
+ SACR1 = pxa_i2s.sacr1;
+ SAIMR = pxa_i2s.saimr;
+ SADIV = pxa_i2s.sadiv;
+ SACR0 |= SACR0_ENB;
+
+ return 0;
+}
+
+#else
+#define pxa2xx_i2s_suspend NULL
+#define pxa2xx_i2s_resume NULL
+#endif
+
+#define PXA2XX_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
+
+struct snd_soc_cpu_dai pxa_i2s_dai = {
+ .name = "pxa2xx-i2s",
+ .id = 0,
+ .type = SND_SOC_DAI_I2S,
+ .suspend = pxa2xx_i2s_suspend,
+ .resume = pxa2xx_i2s_resume,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PXA2XX_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PXA2XX_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .startup = pxa2xx_i2s_startup,
+ .shutdown = pxa2xx_i2s_shutdown,
+ .trigger = pxa2xx_i2s_trigger,
+ .hw_params = pxa2xx_i2s_hw_params,},
+ .dai_ops = {
+ .set_fmt = pxa2xx_i2s_set_dai_fmt,
+ .set_sysclk = pxa2xx_i2s_set_dai_sysclk,
+ },
+};
+
+EXPORT_SYMBOL_GPL(pxa_i2s_dai);
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_DESCRIPTION("pxa2xx I2S SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-i2s.h b/sound/soc/pxa/pxa2xx-i2s.h
new file mode 100644
index 00000000000..a2484f0881f
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-i2s.h
@@ -0,0 +1,20 @@
+/*
+ * linux/sound/arm/pxa2xx-i2s.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _PXA2XX_I2S_H
+#define _PXA2XX_I2S_H
+
+/* pxa2xx DAI ID's */
+#define PXA2XX_DAI_I2S 0
+
+/* I2S clock */
+#define PXA2XX_I2S_SYSCLK 0
+
+extern struct snd_soc_cpu_dai pxa_i2s_dai;
+
+#endif
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
new file mode 100644
index 00000000000..35e8fa3a469
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -0,0 +1,372 @@
+/*
+ * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip
+ *
+ * Author: Nicolas Pitre
+ * Created: Nov 30, 2004
+ * Copyright: (C) 2004 MontaVista Software, Inc.
+ *
+ * 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 <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/dma.h>
+#include <asm/hardware.h>
+#include <asm/arch/pxa-regs.h>
+#include <asm/arch/audio.h>
+
+#include "pxa2xx-pcm.h"
+
+static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8192 - 32,
+ .periods_min = 1,
+ .periods_max = PAGE_SIZE/sizeof(pxa_dma_desc),
+ .buffer_bytes_max = 128 * 1024,
+ .fifo_size = 32,
+};
+
+struct pxa2xx_runtime_data {
+ int dma_ch;
+ struct pxa2xx_pcm_dma_params *params;
+ pxa_dma_desc *dma_desc_array;
+ dma_addr_t dma_desc_array_phys;
+};
+
+static void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id)
+{
+ struct snd_pcm_substream *substream = dev_id;
+ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+ int dcsr;
+
+ dcsr = DCSR(dma_ch);
+ DCSR(dma_ch) = dcsr & ~DCSR_STOPIRQEN;
+
+ if (dcsr & DCSR_ENDINTR) {
+ snd_pcm_period_elapsed(substream);
+ } else {
+ printk( KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n",
+ prtd->params->name, dma_ch, dcsr );
+ }
+}
+
+static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pxa2xx_runtime_data *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct pxa2xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data;
+ size_t totsize = params_buffer_bytes(params);
+ size_t period = params_period_bytes(params);
+ pxa_dma_desc *dma_desc;
+ dma_addr_t dma_buff_phys, next_desc_phys;
+ int ret;
+
+ /* return if this is a bufferless transfer e.g.
+ * codec <--> BT codec or GSM modem -- lg FIXME */
+ if (!dma)
+ return 0;
+
+ /* this may get called several times by oss emulation
+ * with different params */
+ if (prtd->params == NULL) {
+ prtd->params = dma;
+ ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW,
+ pxa2xx_pcm_dma_irq, substream);
+ if (ret < 0)
+ return ret;
+ prtd->dma_ch = ret;
+ } else if (prtd->params != dma) {
+ pxa_free_dma(prtd->dma_ch);
+ prtd->params = dma;
+ ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW,
+ pxa2xx_pcm_dma_irq, substream);
+ if (ret < 0)
+ return ret;
+ prtd->dma_ch = ret;
+ }
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ runtime->dma_bytes = totsize;
+
+ dma_desc = prtd->dma_desc_array;
+ next_desc_phys = prtd->dma_desc_array_phys;
+ dma_buff_phys = runtime->dma_addr;
+ do {
+ next_desc_phys += sizeof(pxa_dma_desc);
+ dma_desc->ddadr = next_desc_phys;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ dma_desc->dsadr = dma_buff_phys;
+ dma_desc->dtadr = prtd->params->dev_addr;
+ } else {
+ dma_desc->dsadr = prtd->params->dev_addr;
+ dma_desc->dtadr = dma_buff_phys;
+ }
+ if (period > totsize)
+ period = totsize;
+ dma_desc->dcmd = prtd->params->dcmd | period | DCMD_ENDIRQEN;
+ dma_desc++;
+ dma_buff_phys += period;
+ } while (totsize -= period);
+ dma_desc[-1].ddadr = prtd->dma_desc_array_phys;
+
+ return 0;
+}
+
+static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+
+ if (prtd && prtd->params)
+ *prtd->params->drcmr = 0;
+
+ if (prtd->dma_ch) {
+ snd_pcm_set_runtime_buffer(substream, NULL);
+ pxa_free_dma(prtd->dma_ch);
+ prtd->dma_ch = 0;
+ }
+
+ return 0;
+}
+
+static int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+
+ DCSR(prtd->dma_ch) &= ~DCSR_RUN;
+ DCSR(prtd->dma_ch) = 0;
+ DCMD(prtd->dma_ch) = 0;
+ *prtd->params->drcmr = prtd->dma_ch | DRCMR_MAPVLD;
+
+ return 0;
+}
+
+static int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys;
+ DCSR(prtd->dma_ch) = DCSR_RUN;
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ DCSR(prtd->dma_ch) &= ~DCSR_RUN;
+ break;
+
+ case SNDRV_PCM_TRIGGER_RESUME:
+ DCSR(prtd->dma_ch) |= DCSR_RUN;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys;
+ DCSR(prtd->dma_ch) |= DCSR_RUN;
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static snd_pcm_uframes_t
+pxa2xx_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pxa2xx_runtime_data *prtd = runtime->private_data;
+
+ dma_addr_t ptr = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ DSADR(prtd->dma_ch) : DTADR(prtd->dma_ch);
+ snd_pcm_uframes_t x = bytes_to_frames(runtime, ptr - runtime->dma_addr);
+
+ if (x == runtime->buffer_size)
+ x = 0;
+ return x;
+}
+
+static int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pxa2xx_runtime_data *prtd;
+ int ret;
+
+ snd_soc_set_runtime_hwparams(substream, &pxa2xx_pcm_hardware);
+
+ /*
+ * For mysterious reasons (and despite what the manual says)
+ * playback samples are lost if the DMA count is not a multiple
+ * of the DMA burst size. Let's add a rule to enforce that.
+ */
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+ if (ret)
+ goto out;
+
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
+ if (ret)
+ goto out;
+
+ ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ goto out;
+
+ prtd = kzalloc(sizeof(struct pxa2xx_runtime_data), GFP_KERNEL);
+ if (prtd == NULL) {
+ ret = -ENOMEM;
+ goto out;
+ }
+
+ prtd->dma_desc_array =
+ dma_alloc_writecombine(substream->pcm->card->dev, PAGE_SIZE,
+ &prtd->dma_desc_array_phys, GFP_KERNEL);
+ if (!prtd->dma_desc_array) {
+ ret = -ENOMEM;
+ goto err1;
+ }
+
+ runtime->private_data = prtd;
+ return 0;
+
+ err1:
+ kfree(prtd);
+ out:
+ return ret;
+}
+
+static int pxa2xx_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pxa2xx_runtime_data *prtd = runtime->private_data;
+
+ dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE,
+ prtd->dma_desc_array, prtd->dma_desc_array_phys);
+ kfree(prtd);
+ return 0;
+}
+
+static int pxa2xx_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);
+}
+
+struct snd_pcm_ops pxa2xx_pcm_ops = {
+ .open = pxa2xx_pcm_open,
+ .close = pxa2xx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pxa2xx_pcm_hw_params,
+ .hw_free = pxa2xx_pcm_hw_free,
+ .prepare = pxa2xx_pcm_prepare,
+ .trigger = pxa2xx_pcm_trigger,
+ .pointer = pxa2xx_pcm_pointer,
+ .mmap = pxa2xx_pcm_mmap,
+};
+
+static int pxa2xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = pxa2xx_pcm_hardware.buffer_bytes_max;
+ 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);
+ if (!buf->area)
+ return -ENOMEM;
+ buf->bytes = size;
+ return 0;
+}
+
+static void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ 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;
+ }
+}
+
+static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK;
+
+int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
+ struct snd_pcm *pcm)
+{
+ int ret = 0;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &pxa2xx_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_32BIT_MASK;
+
+ if (dai->playback.channels_min) {
+ ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+ }
+
+ if (dai->capture.channels_min) {
+ ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ }
+ out:
+ return ret;
+}
+
+struct snd_soc_platform pxa2xx_soc_platform = {
+ .name = "pxa2xx-audio",
+ .pcm_ops = &pxa2xx_pcm_ops,
+ .pcm_new = pxa2xx_pcm_new,
+ .pcm_free = pxa2xx_pcm_free_dma_buffers,
+};
+
+EXPORT_SYMBOL_GPL(pxa2xx_soc_platform);
+
+MODULE_AUTHOR("Nicolas Pitre");
+MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-pcm.h b/sound/soc/pxa/pxa2xx-pcm.h
new file mode 100644
index 00000000000..54c9c755e50
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-pcm.h
@@ -0,0 +1,34 @@
+/*
+ * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip
+ *
+ * Author: Nicolas Pitre
+ * Created: Nov 30, 2004
+ * Copyright: MontaVista Software, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _PXA2XX_PCM_H
+#define _PXA2XX_PCM_H
+
+struct pxa2xx_pcm_dma_params {
+ char *name; /* stream identifier */
+ u32 dcmd; /* DMA descriptor dcmd field */
+ volatile u32 *drcmr; /* the DMA request channel to use */
+ u32 dev_addr; /* device physical address for DMA */
+};
+
+struct pxa2xx_gpio {
+ u32 sys;
+ u32 rx;
+ u32 tx;
+ u32 clk;
+ u32 frm;
+};
+
+/* platform data */
+extern struct snd_soc_platform pxa2xx_soc_platform;
+
+#endif
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
new file mode 100644
index 00000000000..80e82109fef
--- /dev/null
+++ b/sound/soc/pxa/spitz.c
@@ -0,0 +1,394 @@
+/*
+ * spitz.c -- SoC audio for Sharp SL-Cxx00 models Spitz, Borzoi and Akita
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Richard Purdie <richard@openedhand.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.
+ *
+ * Revision history
+ * 30th Nov 2005 Initial version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <asm/hardware/scoop.h>
+#include <asm/arch/pxa-regs.h>
+#include <asm/arch/hardware.h>
+#include <asm/arch/akita.h>
+#include <asm/arch/spitz.h>
+#include <asm/mach-types.h>
+#include "../codecs/wm8750.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+
+#define SPITZ_HP 0
+#define SPITZ_MIC 1
+#define SPITZ_LINE 2
+#define SPITZ_HEADSET 3
+#define SPITZ_HP_OFF 4
+#define SPITZ_SPK_ON 0
+#define SPITZ_SPK_OFF 1
+
+ /* audio clock in Hz - rounded from 12.235MHz */
+#define SPITZ_AUDIO_CLOCK 12288000
+
+static int spitz_jack_func;
+static int spitz_spk_func;
+
+static void spitz_ext_control(struct snd_soc_codec *codec)
+{
+ if (spitz_spk_func == SPITZ_SPK_ON)
+ snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1);
+ else
+ snd_soc_dapm_set_endpoint(codec, "Ext Spk", 0);
+
+ /* set up jack connection */
+ switch (spitz_jack_func) {
+ case SPITZ_HP:
+ /* enable and unmute hp jack, disable mic bias */
+ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
+ snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
+ set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
+ set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+ break;
+ case SPITZ_MIC:
+ /* enable mic jack and bias, mute hp */
+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
+ snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
+ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
+ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+ break;
+ case SPITZ_LINE:
+ /* enable line jack, disable mic bias and mute hp */
+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
+ snd_soc_dapm_set_endpoint(codec, "Line Jack", 1);
+ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
+ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+ break;
+ case SPITZ_HEADSET:
+ /* enable and unmute headset jack enable mic bias, mute L hp */
+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
+ snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 1);
+ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
+ set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+ break;
+ case SPITZ_HP_OFF:
+
+ /* jack removed, everything off */
+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
+ snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
+ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
+ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+ break;
+ }
+ snd_soc_dapm_sync_endpoints(codec);
+}
+
+static int spitz_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->codec;
+
+ /* check the jack status at stream startup */
+ spitz_ext_control(codec);
+ return 0;
+}
+
+static int spitz_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_codec_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int clk = 0;
+ int ret = 0;
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ case 48000:
+ case 96000:
+ clk = 12288000;
+ break;
+ case 11025:
+ case 22050:
+ case 44100:
+ clk = 11289600;
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8750_SYSCLK, clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as input (unused) */
+ ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops spitz_ops = {
+ .startup = spitz_startup,
+ .hw_params = spitz_hw_params,
+};
+
+static int spitz_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = spitz_jack_func;
+ return 0;
+}
+
+static int spitz_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (spitz_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ spitz_jack_func = ucontrol->value.integer.value[0];
+ spitz_ext_control(codec);
+ return 1;
+}
+
+static int spitz_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = spitz_spk_func;
+ return 0;
+}
+
+static int spitz_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (spitz_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ spitz_spk_func = ucontrol->value.integer.value[0];
+ spitz_ext_control(codec);
+ return 1;
+}
+
+static int spitz_mic_bias(struct snd_soc_dapm_widget *w, int event)
+{
+ if (machine_is_borzoi() || machine_is_spitz()) {
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ set_scoop_gpio(&spitzscoop2_device.dev,
+ SPITZ_SCP2_MIC_BIAS);
+ else
+ reset_scoop_gpio(&spitzscoop2_device.dev,
+ SPITZ_SCP2_MIC_BIAS);
+ }
+
+ if (machine_is_akita()) {
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ akita_set_ioexp(&akitaioexp_device.dev,
+ AKITA_IOEXP_MIC_BIAS);
+ else
+ akita_reset_ioexp(&akitaioexp_device.dev,
+ AKITA_IOEXP_MIC_BIAS);
+ }
+ return 0;
+}
+
+/* spitz machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_LINE("Line Jack", NULL),
+
+ /* headset is a mic and mono headphone */
+ SND_SOC_DAPM_HP("Headset Jack", NULL),
+};
+
+/* Spitz machine audio_map */
+static const char *audio_map[][3] = {
+
+ /* headphone connected to LOUT1, ROUT1 */
+ {"Headphone Jack", NULL, "LOUT1"},
+ {"Headphone Jack", NULL, "ROUT1"},
+
+ /* headset connected to ROUT1 and LINPUT1 with bias (def below) */
+ {"Headset Jack", NULL, "ROUT1"},
+
+ /* ext speaker connected to LOUT2, ROUT2 */
+ {"Ext Spk", NULL , "ROUT2"},
+ {"Ext Spk", NULL , "LOUT2"},
+
+ /* mic is connected to input 1 - with bias */
+ {"LINPUT1", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Mic Jack"},
+
+ /* line is connected to input 1 - no bias */
+ {"LINPUT1", NULL, "Line Jack"},
+
+ {NULL, NULL, NULL},
+};
+
+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
+ "Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum spitz_enum[] = {
+ SOC_ENUM_SINGLE_EXT(5, jack_function),
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new wm8750_spitz_controls[] = {
+ SOC_ENUM_EXT("Jack Function", spitz_enum[0], spitz_get_jack,
+ spitz_set_jack),
+ SOC_ENUM_EXT("Speaker Function", spitz_enum[1], spitz_get_spk,
+ spitz_set_spk),
+};
+
+/*
+ * Logic for a wm8750 as connected on a Sharp SL-Cxx00 Device
+ */
+static int spitz_wm8750_init(struct snd_soc_codec *codec)
+{
+ int i, err;
+
+ /* NC codec pins */
+ snd_soc_dapm_set_endpoint(codec, "RINPUT1", 0);
+ snd_soc_dapm_set_endpoint(codec, "LINPUT2", 0);
+ snd_soc_dapm_set_endpoint(codec, "RINPUT2", 0);
+ snd_soc_dapm_set_endpoint(codec, "LINPUT3", 0);
+ snd_soc_dapm_set_endpoint(codec, "RINPUT3", 0);
+ snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
+ snd_soc_dapm_set_endpoint(codec, "MONO", 0);
+
+ /* Add spitz specific controls */
+ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8750_spitz_controls[i], codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ /* Add spitz specific widgets */
+ for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) {
+ snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]);
+ }
+
+ /* Set up spitz specific audio path audio_map */
+ for (i = 0; audio_map[i][0] != NULL; i++) {
+ snd_soc_dapm_connect_input(codec, audio_map[i][0],
+ audio_map[i][1], audio_map[i][2]);
+ }
+
+ snd_soc_dapm_sync_endpoints(codec);
+ return 0;
+}
+
+/* spitz digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link spitz_dai = {
+ .name = "wm8750",
+ .stream_name = "WM8750",
+ .cpu_dai = &pxa_i2s_dai,
+ .codec_dai = &wm8750_dai,
+ .init = spitz_wm8750_init,
+ .ops = &spitz_ops,
+};
+
+/* spitz audio machine driver */
+static struct snd_soc_machine snd_soc_machine_spitz = {
+ .name = "Spitz",
+ .dai_link = &spitz_dai,
+ .num_links = 1,
+};
+
+/* spitz audio private data */
+static struct wm8750_setup_data spitz_wm8750_setup = {
+ .i2c_address = 0x1b,
+};
+
+/* spitz audio subsystem */
+static struct snd_soc_device spitz_snd_devdata = {
+ .machine = &snd_soc_machine_spitz,
+ .platform = &pxa2xx_soc_platform,
+ .codec_dev = &soc_codec_dev_wm8750,
+ .codec_data = &spitz_wm8750_setup,
+};
+
+static struct platform_device *spitz_snd_device;
+
+static int __init spitz_init(void)
+{
+ int ret;
+
+ if (!(machine_is_spitz() || machine_is_borzoi() || machine_is_akita()))
+ return -ENODEV;
+
+ spitz_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!spitz_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(spitz_snd_device, &spitz_snd_devdata);
+ spitz_snd_devdata.dev = &spitz_snd_device->dev;
+ ret = platform_device_add(spitz_snd_device);
+
+ if (ret)
+ platform_device_put(spitz_snd_device);
+
+ return ret;
+}
+
+static void __exit spitz_exit(void)
+{
+ platform_device_unregister(spitz_snd_device);
+}
+
+module_init(spitz_init);
+module_exit(spitz_exit);
+
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Spitz");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
new file mode 100644
index 00000000000..5504e30acf1
--- /dev/null
+++ b/sound/soc/pxa/tosa.c
@@ -0,0 +1,289 @@
+/*
+ * tosa.c -- SoC audio for Tosa
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Richard Purdie <richard@openedhand.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.
+ *
+ * Revision history
+ * 30th Nov 2005 Initial version.
+ *
+ * GPIO's
+ * 1 - Jack Insertion
+ * 5 - Hookswitch (headset answer/hang up switch)
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <asm/hardware/tmio.h>
+#include <asm/arch/pxa-regs.h>
+#include <asm/arch/hardware.h>
+#include <asm/arch/audio.h>
+#include <asm/arch/tosa.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static struct snd_soc_machine tosa;
+
+#define TOSA_HP 0
+#define TOSA_MIC_INT 1
+#define TOSA_HEADSET 2
+#define TOSA_HP_OFF 3
+#define TOSA_SPK_ON 0
+#define TOSA_SPK_OFF 1
+
+static int tosa_jack_func;
+static int tosa_spk_func;
+
+static void tosa_ext_control(struct snd_soc_codec *codec)
+{
+ int spk = 0, mic_int = 0, hp = 0, hs = 0;
+
+ /* set up jack connection */
+ switch (tosa_jack_func) {
+ case TOSA_HP:
+ hp = 1;
+ break;
+ case TOSA_MIC_INT:
+ mic_int = 1;
+ break;
+ case TOSA_HEADSET:
+ hs = 1;
+ break;
+ }
+
+ if (tosa_spk_func == TOSA_SPK_ON)
+ spk = 1;
+
+ snd_soc_dapm_set_endpoint(codec, "Speaker", spk);
+ snd_soc_dapm_set_endpoint(codec, "Mic (Internal)", mic_int);
+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp);
+ snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs);
+ snd_soc_dapm_sync_endpoints(codec);
+}
+
+static int tosa_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->codec;
+
+ /* check the jack status at stream startup */
+ tosa_ext_control(codec);
+ return 0;
+}
+
+static struct snd_soc_ops tosa_ops = {
+ .startup = tosa_startup,
+};
+
+static int tosa_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = tosa_jack_func;
+ return 0;
+}
+
+static int tosa_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (tosa_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ tosa_jack_func = ucontrol->value.integer.value[0];
+ tosa_ext_control(codec);
+ return 1;
+}
+
+static int tosa_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = tosa_spk_func;
+ return 0;
+}
+
+static int tosa_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (tosa_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ tosa_spk_func = ucontrol->value.integer.value[0];
+ tosa_ext_control(codec);
+ return 1;
+}
+
+/* tosa dapm event handlers */
+static int tosa_hp_event(struct snd_soc_dapm_widget *w, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ set_tc6393_gpio(&tc6393_device.dev,TOSA_TC6393_L_MUTE);
+ else
+ reset_tc6393_gpio(&tc6393_device.dev,TOSA_TC6393_L_MUTE);
+ return 0;
+}
+
+/* tosa machine dapm widgets */
+static const struct snd_soc_dapm_widget tosa_dapm_widgets[] = {
+SND_SOC_DAPM_HP("Headphone Jack", tosa_hp_event),
+SND_SOC_DAPM_HP("Headset Jack", NULL),
+SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/* tosa audio map */
+static const char *audio_map[][3] = {
+
+ /* headphone connected to HPOUTL, HPOUTR */
+ {"Headphone Jack", NULL, "HPOUTL"},
+ {"Headphone Jack", NULL, "HPOUTR"},
+
+ /* ext speaker connected to LOUT2, ROUT2 */
+ {"Speaker", NULL, "LOUT2"},
+ {"Speaker", NULL, "ROUT2"},
+
+ /* internal mic is connected to mic1, mic2 differential - with bias */
+ {"MIC1", NULL, "Mic Bias"},
+ {"MIC2", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Mic (Internal)"},
+
+ /* headset is connected to HPOUTR, and LINEINR with bias */
+ {"Headset Jack", NULL, "HPOUTR"},
+ {"LINEINR", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Headset Jack"},
+
+ {NULL, NULL, NULL},
+};
+
+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
+ "Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum tosa_enum[] = {
+ SOC_ENUM_SINGLE_EXT(5, jack_function),
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new tosa_controls[] = {
+ SOC_ENUM_EXT("Jack Function", tosa_enum[0], tosa_get_jack,
+ tosa_set_jack),
+ SOC_ENUM_EXT("Speaker Function", tosa_enum[1], tosa_get_spk,
+ tosa_set_spk),
+};
+
+static int tosa_ac97_init(struct snd_soc_codec *codec)
+{
+ int i, err;
+
+ snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
+ snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0);
+
+ /* add tosa specific controls */
+ for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&tosa_controls[i],codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ /* add tosa specific widgets */
+ for (i = 0; i < ARRAY_SIZE(tosa_dapm_widgets); i++) {
+ snd_soc_dapm_new_control(codec, &tosa_dapm_widgets[i]);
+ }
+
+ /* set up tosa specific audio path audio_map */
+ for (i = 0; audio_map[i][0] != NULL; i++) {
+ snd_soc_dapm_connect_input(codec, audio_map[i][0],
+ audio_map[i][1], audio_map[i][2]);
+ }
+
+ snd_soc_dapm_sync_endpoints(codec);
+ return 0;
+}
+
+static struct snd_soc_dai_link tosa_dai[] = {
+{
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+ .init = tosa_ac97_init,
+ .ops = &tosa_ops,
+},
+{
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+ .ops = &tosa_ops,
+},
+};
+
+static struct snd_soc_machine tosa = {
+ .name = "Tosa",
+ .dai_link = tosa_dai,
+ .num_links = ARRAY_SIZE(tosa_dai),
+};
+
+static struct snd_soc_device tosa_snd_devdata = {
+ .machine = &tosa,
+ .platform = &pxa2xx_soc_platform,
+ .codec_dev = &soc_codec_dev_wm9712,
+};
+
+static struct platform_device *tosa_snd_device;
+
+static int __init tosa_init(void)
+{
+ int ret;
+
+ if (!machine_is_tosa())
+ return -ENODEV;
+
+ tosa_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!tosa_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(tosa_snd_device, &tosa_snd_devdata);
+ tosa_snd_devdata.dev = &tosa_snd_device->dev;
+ ret = platform_device_add(tosa_snd_device);
+
+ if (ret)
+ platform_device_put(tosa_snd_device);
+
+ return ret;
+}
+
+static void __exit tosa_exit(void)
+{
+ platform_device_unregister(tosa_snd_device);
+}
+
+module_init(tosa_init);
+module_exit(tosa_exit);
+
+/* Module information */
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Tosa");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
new file mode 100644
index 00000000000..36519aef55d
--- /dev/null
+++ b/sound/soc/soc-core.c
@@ -0,0 +1,1587 @@
+/*
+ * soc-core.c -- ALSA SoC Audio Layer
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Liam Girdwood
+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * with code, comments and ideas from :-
+ * Richard Purdie <richard@openedhand.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.
+ *
+ * Revision history
+ * 12th Aug 2005 Initial version.
+ * 25th Oct 2005 Working Codec, Interface and Platform registration.
+ *
+ * TODO:
+ * o Add hw rules to enforce rates, etc.
+ * o More testing with other codecs/machines.
+ * o Add more codecs and platforms to ensure good API coverage.
+ * o Support TDM on PCM and I2S
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/bitops.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+/* debug */
+#define SOC_DEBUG 0
+#if SOC_DEBUG
+#define dbg(format, arg...) printk(format, ## arg)
+#else
+#define dbg(format, arg...)
+#endif
+
+static DEFINE_MUTEX(pcm_mutex);
+static DEFINE_MUTEX(io_mutex);
+static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
+
+/*
+ * This is a timeout to do a DAPM powerdown after a stream is closed().
+ * It can be used to eliminate pops between different playback streams, e.g.
+ * between two audio tracks.
+ */
+static int pmdown_time = 5000;
+module_param(pmdown_time, int, 0);
+MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
+
+/*
+ * This function forces any delayed work to be queued and run.
+ */
+static int run_delayed_work(struct delayed_work *dwork)
+{
+ int ret;
+
+ /* cancel any work waiting to be queued. */
+ ret = cancel_delayed_work(dwork);
+
+ /* if there was any work waiting then we run it now and
+ * wait for it's completion */
+ if (ret) {
+ schedule_delayed_work(dwork, 0);
+ flush_scheduled_work();
+ }
+ return ret;
+}
+
+#ifdef CONFIG_SND_SOC_AC97_BUS
+/* unregister ac97 codec */
+static int soc_ac97_dev_unregister(struct snd_soc_codec *codec)
+{
+ if (codec->ac97->dev.bus)
+ device_unregister(&codec->ac97->dev);
+ return 0;
+}
+
+/* stop no dev release warning */
+static void soc_ac97_device_release(struct device *dev){}
+
+/* register ac97 codec to bus */
+static int soc_ac97_dev_register(struct snd_soc_codec *codec)
+{
+ int err;
+
+ codec->ac97->dev.bus = &ac97_bus_type;
+ codec->ac97->dev.parent = NULL;
+ codec->ac97->dev.release = soc_ac97_device_release;
+
+ snprintf(codec->ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s",
+ codec->card->number, 0, codec->name);
+ err = device_register(&codec->ac97->dev);
+ if (err < 0) {
+ snd_printk(KERN_ERR "Can't register ac97 bus\n");
+ codec->ac97->dev.bus = NULL;
+ return err;
+ }
+ return 0;
+}
+#endif
+
+static inline const char* get_dai_name(int type)
+{
+ switch(type) {
+ case SND_SOC_DAI_AC97:
+ return "AC97";
+ case SND_SOC_DAI_I2S:
+ return "I2S";
+ case SND_SOC_DAI_PCM:
+ return "PCM";
+ }
+ return NULL;
+}
+
+/*
+ * Called by ALSA when a PCM substream is opened, the runtime->hw record is
+ * then initialized and any private data can be allocated. This also calls
+ * startup for the cpu DAI, platform, machine and codec DAI.
+ */
+static int soc_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_dai_link *machine = rtd->dai;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+ int ret = 0;
+
+ mutex_lock(&pcm_mutex);
+
+ /* startup the audio subsystem */
+ if (cpu_dai->ops.startup) {
+ ret = cpu_dai->ops.startup(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't open interface %s\n",
+ cpu_dai->name);
+ goto out;
+ }
+ }
+
+ if (platform->pcm_ops->open) {
+ ret = platform->pcm_ops->open(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't open platform %s\n", platform->name);
+ goto platform_err;
+ }
+ }
+
+ if (codec_dai->ops.startup) {
+ ret = codec_dai->ops.startup(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't open codec %s\n",
+ codec_dai->name);
+ goto codec_dai_err;
+ }
+ }
+
+ if (machine->ops && machine->ops->startup) {
+ ret = machine->ops->startup(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: %s startup failed\n", machine->name);
+ goto machine_err;
+ }
+ }
+
+ /* Check that the codec and cpu DAI's are compatible */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ runtime->hw.rate_min =
+ max(codec_dai->playback.rate_min, cpu_dai->playback.rate_min);
+ runtime->hw.rate_max =
+ min(codec_dai->playback.rate_max, cpu_dai->playback.rate_max);
+ runtime->hw.channels_min =
+ max(codec_dai->playback.channels_min,
+ cpu_dai->playback.channels_min);
+ runtime->hw.channels_max =
+ min(codec_dai->playback.channels_max,
+ cpu_dai->playback.channels_max);
+ runtime->hw.formats =
+ codec_dai->playback.formats & cpu_dai->playback.formats;
+ runtime->hw.rates =
+ codec_dai->playback.rates & cpu_dai->playback.rates;
+ } else {
+ runtime->hw.rate_min =
+ max(codec_dai->capture.rate_min, cpu_dai->capture.rate_min);
+ runtime->hw.rate_max =
+ min(codec_dai->capture.rate_max, cpu_dai->capture.rate_max);
+ runtime->hw.channels_min =
+ max(codec_dai->capture.channels_min,
+ cpu_dai->capture.channels_min);
+ runtime->hw.channels_max =
+ min(codec_dai->capture.channels_max,
+ cpu_dai->capture.channels_max);
+ runtime->hw.formats =
+ codec_dai->capture.formats & cpu_dai->capture.formats;
+ runtime->hw.rates =
+ codec_dai->capture.rates & cpu_dai->capture.rates;
+ }
+
+ snd_pcm_limit_hw_rates(runtime);
+ if (!runtime->hw.rates) {
+ printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
+ codec_dai->name, cpu_dai->name);
+ goto machine_err;
+ }
+ if (!runtime->hw.formats) {
+ printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
+ codec_dai->name, cpu_dai->name);
+ goto machine_err;
+ }
+ if (!runtime->hw.channels_min || !runtime->hw.channels_max) {
+ printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
+ codec_dai->name, cpu_dai->name);
+ goto machine_err;
+ }
+
+ dbg("asoc: %s <-> %s info:\n",codec_dai->name, cpu_dai->name);
+ dbg("asoc: rate mask 0x%x\n", runtime->hw.rates);
+ dbg("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
+ runtime->hw.channels_max);
+ dbg("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
+ runtime->hw.rate_max);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cpu_dai->playback.active = codec_dai->playback.active = 1;
+ else
+ cpu_dai->capture.active = codec_dai->capture.active = 1;
+ cpu_dai->active = codec_dai->active = 1;
+ cpu_dai->runtime = runtime;
+ socdev->codec->active++;
+ mutex_unlock(&pcm_mutex);
+ return 0;
+
+machine_err:
+ if (machine->ops && machine->ops->shutdown)
+ machine->ops->shutdown(substream);
+
+codec_dai_err:
+ if (platform->pcm_ops->close)
+ platform->pcm_ops->close(substream);
+
+platform_err:
+ if (cpu_dai->ops.shutdown)
+ cpu_dai->ops.shutdown(substream);
+out:
+ mutex_unlock(&pcm_mutex);
+ return ret;
+}
+
+/*
+ * Power down the audio subsytem pmdown_time msecs after close is called.
+ * This is to ensure there are no pops or clicks in between any music tracks
+ * due to DAPM power cycling.
+ */
+static void close_delayed_work(struct work_struct *work)
+{
+ struct snd_soc_device *socdev =
+ container_of(work, struct snd_soc_device, delayed_work.work);
+ struct snd_soc_codec *codec = socdev->codec;
+ struct snd_soc_codec_dai *codec_dai;
+ int i;
+
+ mutex_lock(&pcm_mutex);
+ for(i = 0; i < codec->num_dai; i++) {
+ codec_dai = &codec->dai[i];
+
+ dbg("pop wq checking: %s status: %s waiting: %s\n",
+ codec_dai->playback.stream_name,
+ codec_dai->playback.active ? "active" : "inactive",
+ codec_dai->pop_wait ? "yes" : "no");
+
+ /* are we waiting on this codec DAI stream */
+ if (codec_dai->pop_wait == 1) {
+
+ codec_dai->pop_wait = 0;
+ snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name,
+ SND_SOC_DAPM_STREAM_STOP);
+
+ /* power down the codec power domain if no longer active */
+ if (codec->active == 0) {
+ dbg("pop wq D3 %s %s\n", codec->name,
+ codec_dai->playback.stream_name);
+ if (codec->dapm_event)
+ codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+ }
+ }
+ }
+ mutex_unlock(&pcm_mutex);
+}
+
+/*
+ * Called by ALSA when a PCM substream is closed. Private data can be
+ * freed here. The cpu DAI, codec DAI, machine and platform are also
+ * shutdown.
+ */
+static int soc_codec_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_dai_link *machine = rtd->dai;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+ struct snd_soc_codec *codec = socdev->codec;
+
+ mutex_lock(&pcm_mutex);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cpu_dai->playback.active = codec_dai->playback.active = 0;
+ else
+ cpu_dai->capture.active = codec_dai->capture.active = 0;
+
+ if (codec_dai->playback.active == 0 &&
+ codec_dai->capture.active == 0) {
+ cpu_dai->active = codec_dai->active = 0;
+ }
+ codec->active--;
+
+ if (cpu_dai->ops.shutdown)
+ cpu_dai->ops.shutdown(substream);
+
+ if (codec_dai->ops.shutdown)
+ codec_dai->ops.shutdown(substream);
+
+ if (machine->ops && machine->ops->shutdown)
+ machine->ops->shutdown(substream);
+
+ if (platform->pcm_ops->close)
+ platform->pcm_ops->close(substream);
+ cpu_dai->runtime = NULL;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* start delayed pop wq here for playback streams */
+ codec_dai->pop_wait = 1;
+ schedule_delayed_work(&socdev->delayed_work,
+ msecs_to_jiffies(pmdown_time));
+ } else {
+ /* capture streams can be powered down now */
+ snd_soc_dapm_stream_event(codec,
+ codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_STOP);
+
+ if (codec->active == 0 && codec_dai->pop_wait == 0){
+ if (codec->dapm_event)
+ codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+ }
+ }
+
+ mutex_unlock(&pcm_mutex);
+ return 0;
+}
+
+/*
+ * Called by ALSA when the PCM substream is prepared, can set format, sample
+ * rate, etc. This function is non atomic and can be called multiple times,
+ * it can refer to the runtime info.
+ */
+static int soc_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_dai_link *machine = rtd->dai;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+
+ mutex_lock(&pcm_mutex);
+
+ if (machine->ops && machine->ops->prepare) {
+ ret = machine->ops->prepare(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: machine prepare error\n");
+ goto out;
+ }
+ }
+
+ if (platform->pcm_ops->prepare) {
+ ret = platform->pcm_ops->prepare(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: platform prepare error\n");
+ goto out;
+ }
+ }
+
+ if (codec_dai->ops.prepare) {
+ ret = codec_dai->ops.prepare(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: codec DAI prepare error\n");
+ goto out;
+ }
+ }
+
+ if (cpu_dai->ops.prepare) {
+ ret = cpu_dai->ops.prepare(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: cpu DAI prepare error\n");
+ goto out;
+ }
+ }
+
+ /* we only want to start a DAPM playback stream if we are not waiting
+ * on an existing one stopping */
+ if (codec_dai->pop_wait) {
+ /* we are waiting for the delayed work to start */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ snd_soc_dapm_stream_event(socdev->codec,
+ codec_dai->capture.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+ else {
+ codec_dai->pop_wait = 0;
+ cancel_delayed_work(&socdev->delayed_work);
+ if (codec_dai->dai_ops.digital_mute)
+ codec_dai->dai_ops.digital_mute(codec_dai, 0);
+ }
+ } else {
+ /* no delayed work - do we need to power up codec */
+ if (codec->dapm_state != SNDRV_CTL_POWER_D0) {
+
+ if (codec->dapm_event)
+ codec->dapm_event(codec, SNDRV_CTL_POWER_D1);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_dapm_stream_event(codec,
+ codec_dai->playback.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+ else
+ snd_soc_dapm_stream_event(codec,
+ codec_dai->capture.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+
+ if (codec->dapm_event)
+ codec->dapm_event(codec, SNDRV_CTL_POWER_D0);
+ if (codec_dai->dai_ops.digital_mute)
+ codec_dai->dai_ops.digital_mute(codec_dai, 0);
+
+ } else {
+ /* codec already powered - power on widgets */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_dapm_stream_event(codec,
+ codec_dai->playback.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+ else
+ snd_soc_dapm_stream_event(codec,
+ codec_dai->capture.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+ if (codec_dai->dai_ops.digital_mute)
+ codec_dai->dai_ops.digital_mute(codec_dai, 0);
+ }
+ }
+
+out:
+ mutex_unlock(&pcm_mutex);
+ return ret;
+}
+
+/*
+ * Called by ALSA when the hardware params are set by application. This
+ * function can also be called multiple times and can allocate buffers
+ * (using snd_pcm_lib_* ). It's non-atomic.
+ */
+static int soc_pcm_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_device *socdev = rtd->socdev;
+ struct snd_soc_dai_link *machine = rtd->dai;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+ int ret = 0;
+
+ mutex_lock(&pcm_mutex);
+
+ if (machine->ops && machine->ops->hw_params) {
+ ret = machine->ops->hw_params(substream, params);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: machine hw_params failed\n");
+ goto out;
+ }
+ }
+
+ if (codec_dai->ops.hw_params) {
+ ret = codec_dai->ops.hw_params(substream, params);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't set codec %s hw params\n",
+ codec_dai->name);
+ goto codec_err;
+ }
+ }
+
+ if (cpu_dai->ops.hw_params) {
+ ret = cpu_dai->ops.hw_params(substream, params);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't set interface %s hw params\n",
+ cpu_dai->name);
+ goto interface_err;
+ }
+ }
+
+ if (platform->pcm_ops->hw_params) {
+ ret = platform->pcm_ops->hw_params(substream, params);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't set platform %s hw params\n",
+ platform->name);
+ goto platform_err;
+ }
+ }
+
+out:
+ mutex_unlock(&pcm_mutex);
+ return ret;
+
+platform_err:
+ if (cpu_dai->ops.hw_free)
+ cpu_dai->ops.hw_free(substream);
+
+interface_err:
+ if (codec_dai->ops.hw_free)
+ codec_dai->ops.hw_free(substream);
+
+codec_err:
+ if(machine->ops && machine->ops->hw_free)
+ machine->ops->hw_free(substream);
+
+ mutex_unlock(&pcm_mutex);
+ return ret;
+}
+
+/*
+ * Free's resources allocated by hw_params, can be called multiple times
+ */
+static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_dai_link *machine = rtd->dai;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+ struct snd_soc_codec *codec = socdev->codec;
+
+ mutex_lock(&pcm_mutex);
+
+ /* apply codec digital mute */
+ if (!codec->active && codec_dai->dai_ops.digital_mute)
+ codec_dai->dai_ops.digital_mute(codec_dai, 1);
+
+ /* free any machine hw params */
+ if (machine->ops && machine->ops->hw_free)
+ machine->ops->hw_free(substream);
+
+ /* free any DMA resources */
+ if (platform->pcm_ops->hw_free)
+ platform->pcm_ops->hw_free(substream);
+
+ /* now free hw params for the DAI's */
+ if (codec_dai->ops.hw_free)
+ codec_dai->ops.hw_free(substream);
+
+ if (cpu_dai->ops.hw_free)
+ cpu_dai->ops.hw_free(substream);
+
+ mutex_unlock(&pcm_mutex);
+ return 0;
+}
+
+static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_dai_link *machine = rtd->dai;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+ int ret;
+
+ if (codec_dai->ops.trigger) {
+ ret = codec_dai->ops.trigger(substream, cmd);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (platform->pcm_ops->trigger) {
+ ret = platform->pcm_ops->trigger(substream, cmd);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (cpu_dai->ops.trigger) {
+ ret = cpu_dai->ops.trigger(substream, cmd);
+ if (ret < 0)
+ return ret;
+ }
+ return 0;
+}
+
+/* ASoC PCM operations */
+static struct snd_pcm_ops soc_pcm_ops = {
+ .open = soc_pcm_open,
+ .close = soc_codec_close,
+ .hw_params = soc_pcm_hw_params,
+ .hw_free = soc_pcm_hw_free,
+ .prepare = soc_pcm_prepare,
+ .trigger = soc_pcm_trigger,
+};
+
+#ifdef CONFIG_PM
+/* powers down audio subsystem for suspend */
+static int soc_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_machine *machine = socdev->machine;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+
+ /* mute any active DAC's */
+ for(i = 0; i < machine->num_links; i++) {
+ struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai;
+ if (dai->dai_ops.digital_mute && dai->playback.active)
+ dai->dai_ops.digital_mute(dai, 1);
+ }
+
+ if (machine->suspend_pre)
+ machine->suspend_pre(pdev, state);
+
+ for(i = 0; i < machine->num_links; i++) {
+ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97)
+ cpu_dai->suspend(pdev, cpu_dai);
+ if (platform->suspend)
+ platform->suspend(pdev, cpu_dai);
+ }
+
+ /* close any waiting streams and save state */
+ run_delayed_work(&socdev->delayed_work);
+ codec->suspend_dapm_state = codec->dapm_state;
+
+ for(i = 0; i < codec->num_dai; i++) {
+ char *stream = codec->dai[i].playback.stream_name;
+ if (stream != NULL)
+ snd_soc_dapm_stream_event(codec, stream,
+ SND_SOC_DAPM_STREAM_SUSPEND);
+ stream = codec->dai[i].capture.stream_name;
+ if (stream != NULL)
+ snd_soc_dapm_stream_event(codec, stream,
+ SND_SOC_DAPM_STREAM_SUSPEND);
+ }
+
+ if (codec_dev->suspend)
+ codec_dev->suspend(pdev, state);
+
+ for(i = 0; i < machine->num_links; i++) {
+ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97)
+ cpu_dai->suspend(pdev, cpu_dai);
+ }
+
+ if (machine->suspend_post)
+ machine->suspend_post(pdev, state);
+
+ return 0;
+}
+
+/* powers up audio subsystem after a suspend */
+static int soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_machine *machine = socdev->machine;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+
+ if (machine->resume_pre)
+ machine->resume_pre(pdev);
+
+ for(i = 0; i < machine->num_links; i++) {
+ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97)
+ cpu_dai->resume(pdev, cpu_dai);
+ }
+
+ if (codec_dev->resume)
+ codec_dev->resume(pdev);
+
+ for(i = 0; i < codec->num_dai; i++) {
+ char* stream = codec->dai[i].playback.stream_name;
+ if (stream != NULL)
+ snd_soc_dapm_stream_event(codec, stream,
+ SND_SOC_DAPM_STREAM_RESUME);
+ stream = codec->dai[i].capture.stream_name;
+ if (stream != NULL)
+ snd_soc_dapm_stream_event(codec, stream,
+ SND_SOC_DAPM_STREAM_RESUME);
+ }
+
+ /* unmute any active DAC's */
+ for(i = 0; i < machine->num_links; i++) {
+ struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai;
+ if (dai->dai_ops.digital_mute && dai->playback.active)
+ dai->dai_ops.digital_mute(dai, 0);
+ }
+
+ for(i = 0; i < machine->num_links; i++) {
+ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97)
+ cpu_dai->resume(pdev, cpu_dai);
+ if (platform->resume)
+ platform->resume(pdev, cpu_dai);
+ }
+
+ if (machine->resume_post)
+ machine->resume_post(pdev);
+
+ return 0;
+}
+
+#else
+#define soc_suspend NULL
+#define soc_resume NULL
+#endif
+
+/* probes a new socdev */
+static int soc_probe(struct platform_device *pdev)
+{
+ int ret = 0, i;
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_machine *machine = socdev->machine;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+
+ if (machine->probe) {
+ ret = machine->probe(pdev);
+ if(ret < 0)
+ return ret;
+ }
+
+ for (i = 0; i < machine->num_links; i++) {
+ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ if (cpu_dai->probe) {
+ ret = cpu_dai->probe(pdev);
+ if(ret < 0)
+ goto cpu_dai_err;
+ }
+ }
+
+ if (codec_dev->probe) {
+ ret = codec_dev->probe(pdev);
+ if(ret < 0)
+ goto cpu_dai_err;
+ }
+
+ if (platform->probe) {
+ ret = platform->probe(pdev);
+ if(ret < 0)
+ goto platform_err;
+ }
+
+ /* DAPM stream work */
+ INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work);
+ return 0;
+
+platform_err:
+ if (codec_dev->remove)
+ codec_dev->remove(pdev);
+
+cpu_dai_err:
+ for (i--; i >= 0; i--) {
+ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ if (cpu_dai->remove)
+ cpu_dai->remove(pdev);
+ }
+
+ if (machine->remove)
+ machine->remove(pdev);
+
+ return ret;
+}
+
+/* removes a socdev */
+static int soc_remove(struct platform_device *pdev)
+{
+ int i;
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_machine *machine = socdev->machine;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+
+ run_delayed_work(&socdev->delayed_work);
+
+ if (platform->remove)
+ platform->remove(pdev);
+
+ if (codec_dev->remove)
+ codec_dev->remove(pdev);
+
+ for (i = 0; i < machine->num_links; i++) {
+ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ if (cpu_dai->remove)
+ cpu_dai->remove(pdev);
+ }
+
+ if (machine->remove)
+ machine->remove(pdev);
+
+ return 0;
+}
+
+/* ASoC platform driver */
+static struct platform_driver soc_driver = {
+ .driver = {
+ .name = "soc-audio",
+ },
+ .probe = soc_probe,
+ .remove = soc_remove,
+ .suspend = soc_suspend,
+ .resume = soc_resume,
+};
+
+/* create a new pcm */
+static int soc_new_pcm(struct snd_soc_device *socdev,
+ struct snd_soc_dai_link *dai_link, int num)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ struct snd_soc_codec_dai *codec_dai = dai_link->codec_dai;
+ struct snd_soc_cpu_dai *cpu_dai = dai_link->cpu_dai;
+ struct snd_soc_pcm_runtime *rtd;
+ struct snd_pcm *pcm;
+ char new_name[64];
+ int ret = 0, playback = 0, capture = 0;
+
+ rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL);
+ if (rtd == NULL)
+ return -ENOMEM;
+
+ rtd->dai = dai_link;
+ rtd->socdev = socdev;
+ codec_dai->codec = socdev->codec;
+
+ /* check client and interface hw capabilities */
+ sprintf(new_name, "%s %s-%s-%d",dai_link->stream_name, codec_dai->name,
+ get_dai_name(cpu_dai->type), num);
+
+ if (codec_dai->playback.channels_min)
+ playback = 1;
+ if (codec_dai->capture.channels_min)
+ capture = 1;
+
+ ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback,
+ capture, &pcm);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
+ kfree(rtd);
+ return ret;
+ }
+
+ pcm->private_data = rtd;
+ soc_pcm_ops.mmap = socdev->platform->pcm_ops->mmap;
+ soc_pcm_ops.pointer = socdev->platform->pcm_ops->pointer;
+ soc_pcm_ops.ioctl = socdev->platform->pcm_ops->ioctl;
+ soc_pcm_ops.copy = socdev->platform->pcm_ops->copy;
+ soc_pcm_ops.silence = socdev->platform->pcm_ops->silence;
+ soc_pcm_ops.ack = socdev->platform->pcm_ops->ack;
+ soc_pcm_ops.page = socdev->platform->pcm_ops->page;
+
+ if (playback)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
+
+ if (capture)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
+
+ ret = socdev->platform->pcm_new(codec->card, codec_dai, pcm);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: platform pcm constructor failed\n");
+ kfree(rtd);
+ return ret;
+ }
+
+ pcm->private_free = socdev->platform->pcm_free;
+ printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
+ cpu_dai->name);
+ return ret;
+}
+
+/* codec register dump */
+static ssize_t codec_reg_show(struct device *dev,
+ struct device_attribute *attr, char *buf)
+{
+ struct snd_soc_device *devdata = dev_get_drvdata(dev);
+ struct snd_soc_codec *codec = devdata->codec;
+ int i, step = 1, count = 0;
+
+ if (!codec->reg_cache_size)
+ return 0;
+
+ if (codec->reg_cache_step)
+ step = codec->reg_cache_step;
+
+ count += sprintf(buf, "%s registers\n", codec->name);
+ for(i = 0; i < codec->reg_cache_size; i += step)
+ count += sprintf(buf + count, "%2x: %4x\n", i, codec->read(codec, i));
+
+ return count;
+}
+static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
+
+/**
+ * snd_soc_new_ac97_codec - initailise AC97 device
+ * @codec: audio codec
+ * @ops: AC97 bus operations
+ * @num: AC97 codec number
+ *
+ * Initialises AC97 codec resources for use by ad-hoc devices only.
+ */
+int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
+ struct snd_ac97_bus_ops *ops, int num)
+{
+ mutex_lock(&codec->mutex);
+
+ codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL);
+ if (codec->ac97 == NULL) {
+ mutex_unlock(&codec->mutex);
+ return -ENOMEM;
+ }
+
+ codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL);
+ if (codec->ac97->bus == NULL) {
+ kfree(codec->ac97);
+ codec->ac97 = NULL;
+ mutex_unlock(&codec->mutex);
+ return -ENOMEM;
+ }
+
+ codec->ac97->bus->ops = ops;
+ codec->ac97->num = num;
+ mutex_unlock(&codec->mutex);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec);
+
+/**
+ * snd_soc_free_ac97_codec - free AC97 codec device
+ * @codec: audio codec
+ *
+ * Frees AC97 codec device resources.
+ */
+void snd_soc_free_ac97_codec(struct snd_soc_codec *codec)
+{
+ mutex_lock(&codec->mutex);
+ kfree(codec->ac97->bus);
+ kfree(codec->ac97);
+ codec->ac97 = NULL;
+ mutex_unlock(&codec->mutex);
+}
+EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec);
+
+/**
+ * snd_soc_update_bits - update codec register bits
+ * @codec: audio codec
+ * @reg: codec register
+ * @mask: register mask
+ * @value: new value
+ *
+ * Writes new register value.
+ *
+ * Returns 1 for change else 0.
+ */
+int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
+ unsigned short mask, unsigned short value)
+{
+ int change;
+ unsigned short old, new;
+
+ mutex_lock(&io_mutex);
+ old = snd_soc_read(codec, reg);
+ new = (old & ~mask) | value;
+ change = old != new;
+ if (change)
+ snd_soc_write(codec, reg, new);
+
+ mutex_unlock(&io_mutex);
+ return change;
+}
+EXPORT_SYMBOL_GPL(snd_soc_update_bits);
+
+/**
+ * snd_soc_test_bits - test register for change
+ * @codec: audio codec
+ * @reg: codec register
+ * @mask: register mask
+ * @value: new value
+ *
+ * Tests a register with a new value and checks if the new value is
+ * different from the old value.
+ *
+ * Returns 1 for change else 0.
+ */
+int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
+ unsigned short mask, unsigned short value)
+{
+ int change;
+ unsigned short old, new;
+
+ mutex_lock(&io_mutex);
+ old = snd_soc_read(codec, reg);
+ new = (old & ~mask) | value;
+ change = old != new;
+ mutex_unlock(&io_mutex);
+
+ return change;
+}
+EXPORT_SYMBOL_GPL(snd_soc_test_bits);
+
+/**
+ * snd_soc_new_pcms - create new sound card and pcms
+ * @socdev: the SoC audio device
+ *
+ * Create a new sound card based upon the codec and interface pcms.
+ *
+ * Returns 0 for success, else error.
+ */
+int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ struct snd_soc_machine *machine = socdev->machine;
+ int ret = 0, i;
+
+ mutex_lock(&codec->mutex);
+
+ /* register a sound card */
+ codec->card = snd_card_new(idx, xid, codec->owner, 0);
+ if (!codec->card) {
+ printk(KERN_ERR "asoc: can't create sound card for codec %s\n",
+ codec->name);
+ mutex_unlock(&codec->mutex);
+ return -ENODEV;
+ }
+
+ codec->card->dev = socdev->dev;
+ codec->card->private_data = codec;
+ strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));
+
+ /* create the pcms */
+ for(i = 0; i < machine->num_links; i++) {
+ ret = soc_new_pcm(socdev, &machine->dai_link[i], i);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't create pcm %s\n",
+ machine->dai_link[i].stream_name);
+ mutex_unlock(&codec->mutex);
+ return ret;
+ }
+ }
+
+ mutex_unlock(&codec->mutex);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
+
+/**
+ * snd_soc_register_card - register sound card
+ * @socdev: the SoC audio device
+ *
+ * Register a SoC sound card. Also registers an AC97 device if the
+ * codec is AC97 for ad hoc devices.
+ *
+ * Returns 0 for success, else error.
+ */
+int snd_soc_register_card(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ struct snd_soc_machine *machine = socdev->machine;
+ int ret = 0, i, ac97 = 0, err = 0;
+
+ mutex_lock(&codec->mutex);
+ for(i = 0; i < machine->num_links; i++) {
+ if (socdev->machine->dai_link[i].init) {
+ err = socdev->machine->dai_link[i].init(codec);
+ if (err < 0) {
+ printk(KERN_ERR "asoc: failed to init %s\n",
+ socdev->machine->dai_link[i].stream_name);
+ continue;
+ }
+ }
+ if (socdev->machine->dai_link[i].cpu_dai->type == SND_SOC_DAI_AC97)
+ ac97 = 1;
+ }
+ snprintf(codec->card->shortname, sizeof(codec->card->shortname),
+ "%s", machine->name);
+ snprintf(codec->card->longname, sizeof(codec->card->longname),
+ "%s (%s)", machine->name, codec->name);
+
+ ret = snd_card_register(codec->card);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: failed to register soundcard for codec %s\n",
+ codec->name);
+ goto out;
+ }
+
+#ifdef CONFIG_SND_SOC_AC97_BUS
+ if (ac97) {
+ ret = soc_ac97_dev_register(codec);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: AC97 device register failed\n");
+ snd_card_free(codec->card);
+ goto out;
+ }
+ }
+#endif
+
+ err = snd_soc_dapm_sys_add(socdev->dev);
+ if (err < 0)
+ printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n");
+
+ err = device_create_file(socdev->dev, &dev_attr_codec_reg);
+ if (err < 0)
+ printk(KERN_WARNING "asoc: failed to add codec sysfs entries\n");
+out:
+ mutex_unlock(&codec->mutex);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_card);
+
+/**
+ * snd_soc_free_pcms - free sound card and pcms
+ * @socdev: the SoC audio device
+ *
+ * Frees sound card and pcms associated with the socdev.
+ * Also unregister the codec if it is an AC97 device.
+ */
+void snd_soc_free_pcms(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+
+ mutex_lock(&codec->mutex);
+#ifdef CONFIG_SND_SOC_AC97_BUS
+ if (codec->ac97)
+ soc_ac97_dev_unregister(codec);
+#endif
+
+ if (codec->card)
+ snd_card_free(codec->card);
+ device_remove_file(socdev->dev, &dev_attr_codec_reg);
+ mutex_unlock(&codec->mutex);
+}
+EXPORT_SYMBOL_GPL(snd_soc_free_pcms);
+
+/**
+ * snd_soc_set_runtime_hwparams - set the runtime hardware parameters
+ * @substream: the pcm substream
+ * @hw: the hardware parameters
+ *
+ * Sets the substream runtime hardware parameters.
+ */
+int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
+ const struct snd_pcm_hardware *hw)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ runtime->hw.info = hw->info;
+ runtime->hw.formats = hw->formats;
+ runtime->hw.period_bytes_min = hw->period_bytes_min;
+ runtime->hw.period_bytes_max = hw->period_bytes_max;
+ runtime->hw.periods_min = hw->periods_min;
+ runtime->hw.periods_max = hw->periods_max;
+ runtime->hw.buffer_bytes_max = hw->buffer_bytes_max;
+ runtime->hw.fifo_size = hw->fifo_size;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams);
+
+/**
+ * snd_soc_cnew - create new control
+ * @_template: control template
+ * @data: control private data
+ * @lnng_name: control long name
+ *
+ * Create a new mixer control from a template control.
+ *
+ * Returns 0 for success, else error.
+ */
+struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
+ void *data, char *long_name)
+{
+ struct snd_kcontrol_new template;
+
+ memcpy(&template, _template, sizeof(template));
+ if (long_name)
+ template.name = long_name;
+ template.access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
+ template.index = 0;
+
+ return snd_ctl_new1(&template, data);
+}
+EXPORT_SYMBOL_GPL(snd_soc_cnew);
+
+/**
+ * snd_soc_info_enum_double - enumerated double mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a double enumerated
+ * mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = e->shift_l == e->shift_r ? 1 : 2;
+ uinfo->value.enumerated.items = e->mask;
+
+ if (uinfo->value.enumerated.item > e->mask - 1)
+ uinfo->value.enumerated.item = e->mask - 1;
+ strcpy(uinfo->value.enumerated.name,
+ e->texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_enum_double);
+
+/**
+ * snd_soc_get_enum_double - enumerated double mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a double enumerated mixer.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned short val, bitmask;
+
+ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
+ ;
+ val = snd_soc_read(codec, e->reg);
+ ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1);
+ if (e->shift_l != e->shift_r)
+ ucontrol->value.enumerated.item[1] =
+ (val >> e->shift_r) & (bitmask - 1);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_enum_double);
+
+/**
+ * snd_soc_put_enum_double - enumerated double mixer put callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a double enumerated mixer.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned short val;
+ unsigned short mask, bitmask;
+
+ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
+ ;
+ if (ucontrol->value.enumerated.item[0] > e->mask - 1)
+ return -EINVAL;
+ val = ucontrol->value.enumerated.item[0] << e->shift_l;
+ mask = (bitmask - 1) << e->shift_l;
+ if (e->shift_l != e->shift_r) {
+ if (ucontrol->value.enumerated.item[1] > e->mask - 1)
+ return -EINVAL;
+ val |= ucontrol->value.enumerated.item[1] << e->shift_r;
+ mask |= (bitmask - 1) << e->shift_r;
+ }
+
+ return snd_soc_update_bits(codec, e->reg, mask, val);
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_enum_double);
+
+/**
+ * snd_soc_info_enum_ext - external enumerated single mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about an external enumerated
+ * single mixer.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = e->mask;
+
+ if (uinfo->value.enumerated.item > e->mask - 1)
+ uinfo->value.enumerated.item = e->mask - 1;
+ strcpy(uinfo->value.enumerated.name,
+ e->texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext);
+
+/**
+ * snd_soc_info_volsw_ext - external single mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a single external mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask = kcontrol->private_value;
+
+ uinfo->type =
+ mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext);
+
+/**
+ * snd_soc_info_bool_ext - external single boolean mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a single boolean external mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_bool_ext(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_bool_ext);
+
+/**
+ * snd_soc_info_volsw - single mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a single mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0x0f;
+ int rshift = (kcontrol->private_value >> 12) & 0x0f;
+
+ uinfo->type =
+ mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = shift == rshift ? 1 : 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_volsw);
+
+/**
+ * snd_soc_get_volsw - single mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a single mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0x0f;
+ int rshift = (kcontrol->private_value >> 12) & 0x0f;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0x01;
+
+ ucontrol->value.integer.value[0] =
+ (snd_soc_read(codec, reg) >> shift) & mask;
+ if (shift != rshift)
+ ucontrol->value.integer.value[1] =
+ (snd_soc_read(codec, reg) >> rshift) & mask;
+ if (invert) {
+ ucontrol->value.integer.value[0] =
+ mask - ucontrol->value.integer.value[0];
+ if (shift != rshift)
+ ucontrol->value.integer.value[1] =
+ mask - ucontrol->value.integer.value[1];
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_volsw);
+
+/**
+ * snd_soc_put_volsw - single mixer put callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a single mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0x0f;
+ int rshift = (kcontrol->private_value >> 12) & 0x0f;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0x01;
+ int err;
+ unsigned short val, val2, val_mask;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+ if (invert)
+ val = mask - val;
+ val_mask = mask << shift;
+ val = val << shift;
+ if (shift != rshift) {
+ val2 = (ucontrol->value.integer.value[1] & mask);
+ if (invert)
+ val2 = mask - val2;
+ val_mask |= mask << rshift;
+ val |= val2 << rshift;
+ }
+ err = snd_soc_update_bits(codec, reg, val_mask, val);
+ return err;
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_volsw);
+
+/**
+ * snd_soc_info_volsw_2r - double mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a double mixer control that
+ * spans 2 codec registers.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 12) & 0xff;
+
+ uinfo->type =
+ mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r);
+
+/**
+ * snd_soc_get_volsw_2r - double mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a double mixer control that spans 2 registers.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int reg2 = (kcontrol->private_value >> 24) & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0x0f;
+ int mask = (kcontrol->private_value >> 12) & 0xff;
+ int invert = (kcontrol->private_value >> 20) & 0x01;
+
+ ucontrol->value.integer.value[0] =
+ (snd_soc_read(codec, reg) >> shift) & mask;
+ ucontrol->value.integer.value[1] =
+ (snd_soc_read(codec, reg2) >> shift) & mask;
+ if (invert) {
+ ucontrol->value.integer.value[0] =
+ mask - ucontrol->value.integer.value[0];
+ ucontrol->value.integer.value[1] =
+ mask - ucontrol->value.integer.value[1];
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r);
+
+/**
+ * snd_soc_put_volsw_2r - double mixer set callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a double mixer control that spans 2 registers.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int reg2 = (kcontrol->private_value >> 24) & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0x0f;
+ int mask = (kcontrol->private_value >> 12) & 0xff;
+ int invert = (kcontrol->private_value >> 20) & 0x01;
+ int err;
+ unsigned short val, val2, val_mask;
+
+ val_mask = mask << shift;
+ val = (ucontrol->value.integer.value[0] & mask);
+ val2 = (ucontrol->value.integer.value[1] & mask);
+
+ if (invert) {
+ val = mask - val;
+ val2 = mask - val2;
+ }
+
+ val = val << shift;
+ val2 = val2 << shift;
+
+ if ((err = snd_soc_update_bits(codec, reg, val_mask, val)) < 0)
+ return err;
+
+ err = snd_soc_update_bits(codec, reg2, val_mask, val2);
+ return err;
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r);
+
+static int __devinit snd_soc_init(void)
+{
+ printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION);
+ return platform_driver_register(&soc_driver);
+}
+
+static void snd_soc_exit(void)
+{
+ platform_driver_unregister(&soc_driver);
+}
+
+module_init(snd_soc_init);
+module_exit(snd_soc_exit);
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_DESCRIPTION("ALSA SoC Core");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
new file mode 100644
index 00000000000..7caf8c7b0ac
--- /dev/null
+++ b/sound/soc/soc-dapm.c
@@ -0,0 +1,1323 @@
+/*
+ * soc-dapm.c -- ALSA SoC Dynamic Audio Power Management
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.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.
+ *
+ * Revision history
+ * 12th Aug 2005 Initial version.
+ * 25th Oct 2005 Implemented path power domain.
+ * 18th Dec 2005 Implemented machine and stream level power domain.
+ *
+ * Features:
+ * o Changes power status of internal codec blocks depending on the
+ * dynamic configuration of codec internal audio paths and active
+ * DAC's/ADC's.
+ * o Platform power domain - can support external components i.e. amps and
+ * mic/meadphone insertion events.
+ * o Automatic Mic Bias support
+ * o Jack insertion power event initiation - e.g. hp insertion will enable
+ * sinks, dacs, etc
+ * o Delayed powerdown of audio susbsytem to reduce pops between a quick
+ * device reopen.
+ *
+ * Todo:
+ * o DAPM power change sequencing - allow for configurable per
+ * codec sequences.
+ * o Support for analogue bias optimisation.
+ * o Support for reduced codec oversampling rates.
+ * o Support for reduced codec bias currents.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/bitops.h>
+#include <linux/platform_device.h>
+#include <linux/jiffies.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+/* debug */
+#define DAPM_DEBUG 0
+#if DAPM_DEBUG
+#define dump_dapm(codec, action) dbg_dump_dapm(codec, action)
+#define dbg(format, arg...) printk(format, ## arg)
+#else
+#define dump_dapm(codec, action)
+#define dbg(format, arg...)
+#endif
+
+#define POP_DEBUG 0
+#if POP_DEBUG
+#define POP_TIME 500 /* 500 msecs - change if pop debug is too fast */
+#define pop_wait(time) schedule_timeout_interruptible(msecs_to_jiffies(time))
+#define pop_dbg(format, arg...) printk(format, ## arg); pop_wait(POP_TIME)
+#else
+#define pop_dbg(format, arg...)
+#define pop_wait(time)
+#endif
+
+/* dapm power sequences - make this per codec in the future */
+static int dapm_up_seq[] = {
+ snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic,
+ snd_soc_dapm_mux, snd_soc_dapm_dac, snd_soc_dapm_mixer, snd_soc_dapm_pga,
+ snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post
+};
+static int dapm_down_seq[] = {
+ snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk,
+ snd_soc_dapm_pga, snd_soc_dapm_mixer, snd_soc_dapm_dac, snd_soc_dapm_mic,
+ snd_soc_dapm_micbias, snd_soc_dapm_mux, snd_soc_dapm_post
+};
+
+static int dapm_status = 1;
+module_param(dapm_status, int, 0);
+MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries");
+
+/* create a new dapm widget */
+static inline struct snd_soc_dapm_widget *dapm_cnew_widget(
+ const struct snd_soc_dapm_widget *_widget)
+{
+ return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL);
+}
+
+/* set up initial codec paths */
+static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
+ struct snd_soc_dapm_path *p, int i)
+{
+ switch (w->id) {
+ case snd_soc_dapm_switch:
+ case snd_soc_dapm_mixer: {
+ int val;
+ int reg = w->kcontrols[i].private_value & 0xff;
+ int shift = (w->kcontrols[i].private_value >> 8) & 0x0f;
+ int mask = (w->kcontrols[i].private_value >> 16) & 0xff;
+ int invert = (w->kcontrols[i].private_value >> 24) & 0x01;
+
+ val = snd_soc_read(w->codec, reg);
+ val = (val >> shift) & mask;
+
+ if ((invert && !val) || (!invert && val))
+ p->connect = 1;
+ else
+ p->connect = 0;
+ }
+ break;
+ case snd_soc_dapm_mux: {
+ struct soc_enum *e = (struct soc_enum *)w->kcontrols[i].private_value;
+ int val, item, bitmask;
+
+ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
+ ;
+ val = snd_soc_read(w->codec, e->reg);
+ item = (val >> e->shift_l) & (bitmask - 1);
+
+ p->connect = 0;
+ for (i = 0; i < e->mask; i++) {
+ if (!(strcmp(p->name, e->texts[i])) && item == i)
+ p->connect = 1;
+ }
+ }
+ break;
+ /* does not effect routing - always connected */
+ case snd_soc_dapm_pga:
+ case snd_soc_dapm_output:
+ case snd_soc_dapm_adc:
+ case snd_soc_dapm_input:
+ case snd_soc_dapm_dac:
+ case snd_soc_dapm_micbias:
+ case snd_soc_dapm_vmid:
+ p->connect = 1;
+ break;
+ /* does effect routing - dynamically connected */
+ case snd_soc_dapm_hp:
+ case snd_soc_dapm_mic:
+ case snd_soc_dapm_spk:
+ case snd_soc_dapm_line:
+ case snd_soc_dapm_pre:
+ case snd_soc_dapm_post:
+ p->connect = 0;
+ break;
+ }
+}
+
+/* connect mux widget to it's interconnecting audio paths */
+static int dapm_connect_mux(struct snd_soc_codec *codec,
+ struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
+ struct snd_soc_dapm_path *path, const char *control_name,
+ const struct snd_kcontrol_new *kcontrol)
+{
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ int i;
+
+ for (i = 0; i < e->mask; i++) {
+ if (!(strcmp(control_name, e->texts[i]))) {
+ list_add(&path->list, &codec->dapm_paths);
+ list_add(&path->list_sink, &dest->sources);
+ list_add(&path->list_source, &src->sinks);
+ path->name = (char*)e->texts[i];
+ dapm_set_path_status(dest, path, 0);
+ return 0;
+ }
+ }
+
+ return -ENODEV;
+}
+
+/* connect mixer widget to it's interconnecting audio paths */
+static int dapm_connect_mixer(struct snd_soc_codec *codec,
+ struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
+ struct snd_soc_dapm_path *path, const char *control_name)
+{
+ int i;
+
+ /* search for mixer kcontrol */
+ for (i = 0; i < dest->num_kcontrols; i++) {
+ if (!strcmp(control_name, dest->kcontrols[i].name)) {
+ list_add(&path->list, &codec->dapm_paths);
+ list_add(&path->list_sink, &dest->sources);
+ list_add(&path->list_source, &src->sinks);
+ path->name = dest->kcontrols[i].name;
+ dapm_set_path_status(dest, path, i);
+ return 0;
+ }
+ }
+ return -ENODEV;
+}
+
+/* update dapm codec register bits */
+static int dapm_update_bits(struct snd_soc_dapm_widget *widget)
+{
+ int change, power;
+ unsigned short old, new;
+ struct snd_soc_codec *codec = widget->codec;
+
+ /* check for valid widgets */
+ if (widget->reg < 0 || widget->id == snd_soc_dapm_input ||
+ widget->id == snd_soc_dapm_output ||
+ widget->id == snd_soc_dapm_hp ||
+ widget->id == snd_soc_dapm_mic ||
+ widget->id == snd_soc_dapm_line ||
+ widget->id == snd_soc_dapm_spk)
+ return 0;
+
+ power = widget->power;
+ if (widget->invert)
+ power = (power ? 0:1);
+
+ old = snd_soc_read(codec, widget->reg);
+ new = (old & ~(0x1 << widget->shift)) | (power << widget->shift);
+
+ change = old != new;
+ if (change) {
+ pop_dbg("pop test %s : %s in %d ms\n", widget->name,
+ widget->power ? "on" : "off", POP_TIME);
+ snd_soc_write(codec, widget->reg, new);
+ pop_wait(POP_TIME);
+ }
+ dbg("reg old %x new %x change %d\n", old, new, change);
+ return change;
+}
+
+/* ramps the volume up or down to minimise pops before or after a
+ * DAPM power event */
+static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power)
+{
+ const struct snd_kcontrol_new *k = widget->kcontrols;
+
+ if (widget->muted && !power)
+ return 0;
+ if (!widget->muted && power)
+ return 0;
+
+ if (widget->num_kcontrols && k) {
+ int reg = k->private_value & 0xff;
+ int shift = (k->private_value >> 8) & 0x0f;
+ int mask = (k->private_value >> 16) & 0xff;
+ int invert = (k->private_value >> 24) & 0x01;
+
+ if (power) {
+ int i;
+ /* power up has happended, increase volume to last level */
+ if (invert) {
+ for (i = mask; i > widget->saved_value; i--)
+ snd_soc_update_bits(widget->codec, reg, mask, i);
+ } else {
+ for (i = 0; i < widget->saved_value; i++)
+ snd_soc_update_bits(widget->codec, reg, mask, i);
+ }
+ widget->muted = 0;
+ } else {
+ /* power down is about to occur, decrease volume to mute */
+ int val = snd_soc_read(widget->codec, reg);
+ int i = widget->saved_value = (val >> shift) & mask;
+ if (invert) {
+ for (; i < mask; i++)
+ snd_soc_update_bits(widget->codec, reg, mask, i);
+ } else {
+ for (; i > 0; i--)
+ snd_soc_update_bits(widget->codec, reg, mask, i);
+ }
+ widget->muted = 1;
+ }
+ }
+ return 0;
+}
+
+/* create new dapm mixer control */
+static int dapm_new_mixer(struct snd_soc_codec *codec,
+ struct snd_soc_dapm_widget *w)
+{
+ int i, ret = 0;
+ char name[32];
+ struct snd_soc_dapm_path *path;
+
+ /* add kcontrol */
+ for (i = 0; i < w->num_kcontrols; i++) {
+
+ /* match name */
+ list_for_each_entry(path, &w->sources, list_sink) {
+
+ /* mixer/mux paths name must match control name */
+ if (path->name != (char*)w->kcontrols[i].name)
+ continue;
+
+ /* add dapm control with long name */
+ snprintf(name, 32, "%s %s", w->name, w->kcontrols[i].name);
+ path->long_name = kstrdup (name, GFP_KERNEL);
+ if (path->long_name == NULL)
+ return -ENOMEM;
+
+ path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w,
+ path->long_name);
+ ret = snd_ctl_add(codec->card, path->kcontrol);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: failed to add dapm kcontrol %s\n",
+ path->long_name);
+ kfree(path->long_name);
+ path->long_name = NULL;
+ return ret;
+ }
+ }
+ }
+ return ret;
+}
+
+/* create new dapm mux control */
+static int dapm_new_mux(struct snd_soc_codec *codec,
+ struct snd_soc_dapm_widget *w)
+{
+ struct snd_soc_dapm_path *path = NULL;
+ struct snd_kcontrol *kcontrol;
+ int ret = 0;
+
+ if (!w->num_kcontrols) {
+ printk(KERN_ERR "asoc: mux %s has no controls\n", w->name);
+ return -EINVAL;
+ }
+
+ kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name);
+ ret = snd_ctl_add(codec->card, kcontrol);
+ if (ret < 0)
+ goto err;
+
+ list_for_each_entry(path, &w->sources, list_sink)
+ path->kcontrol = kcontrol;
+
+ return ret;
+
+err:
+ printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name);
+ return ret;
+}
+
+/* create new dapm volume control */
+static int dapm_new_pga(struct snd_soc_codec *codec,
+ struct snd_soc_dapm_widget *w)
+{
+ struct snd_kcontrol *kcontrol;
+ int ret = 0;
+
+ if (!w->num_kcontrols)
+ return -EINVAL;
+
+ kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name);
+ ret = snd_ctl_add(codec->card, kcontrol);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name);
+ return ret;
+ }
+
+ return ret;
+}
+
+/* reset 'walked' bit for each dapm path */
+static inline void dapm_clear_walk(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_path *p;
+
+ list_for_each_entry(p, &codec->dapm_paths, list)
+ p->walked = 0;
+}
+
+/*
+ * Recursively check for a completed path to an active or physically connected
+ * output widget. Returns number of complete paths.
+ */
+static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
+{
+ struct snd_soc_dapm_path *path;
+ int con = 0;
+
+ if (widget->id == snd_soc_dapm_adc && widget->active)
+ return 1;
+
+ if (widget->connected) {
+ /* connected pin ? */
+ if (widget->id == snd_soc_dapm_output && !widget->ext)
+ return 1;
+
+ /* connected jack or spk ? */
+ if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk ||
+ widget->id == snd_soc_dapm_line)
+ return 1;
+ }
+
+ list_for_each_entry(path, &widget->sinks, list_source) {
+ if (path->walked)
+ continue;
+
+ if (path->sink && path->connect) {
+ path->walked = 1;
+ con += is_connected_output_ep(path->sink);
+ }
+ }
+
+ return con;
+}
+
+/*
+ * Recursively check for a completed path to an active or physically connected
+ * input widget. Returns number of complete paths.
+ */
+static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
+{
+ struct snd_soc_dapm_path *path;
+ int con = 0;
+
+ /* active stream ? */
+ if (widget->id == snd_soc_dapm_dac && widget->active)
+ return 1;
+
+ if (widget->connected) {
+ /* connected pin ? */
+ if (widget->id == snd_soc_dapm_input && !widget->ext)
+ return 1;
+
+ /* connected VMID/Bias for lower pops */
+ if (widget->id == snd_soc_dapm_vmid)
+ return 1;
+
+ /* connected jack ? */
+ if (widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line)
+ return 1;
+ }
+
+ list_for_each_entry(path, &widget->sources, list_sink) {
+ if (path->walked)
+ continue;
+
+ if (path->source && path->connect) {
+ path->walked = 1;
+ con += is_connected_input_ep(path->source);
+ }
+ }
+
+ return con;
+}
+
+/*
+ * Scan each dapm widget for complete audio path.
+ * A complete path is a route that has valid endpoints i.e.:-
+ *
+ * o DAC to output pin.
+ * o Input Pin to ADC.
+ * o Input pin to Output pin (bypass, sidetone)
+ * o DAC to ADC (loopback).
+ */
+static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
+{
+ struct snd_soc_dapm_widget *w;
+ int in, out, i, c = 1, *seq = NULL, ret = 0, power_change, power;
+
+ /* do we have a sequenced stream event */
+ if (event == SND_SOC_DAPM_STREAM_START) {
+ c = ARRAY_SIZE(dapm_up_seq);
+ seq = dapm_up_seq;
+ } else if (event == SND_SOC_DAPM_STREAM_STOP) {
+ c = ARRAY_SIZE(dapm_down_seq);
+ seq = dapm_down_seq;
+ }
+
+ for(i = 0; i < c; i++) {
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
+
+ /* is widget in stream order */
+ if (seq && seq[i] && w->id != seq[i])
+ continue;
+
+ /* vmid - no action */
+ if (w->id == snd_soc_dapm_vmid)
+ continue;
+
+ /* active ADC */
+ if (w->id == snd_soc_dapm_adc && w->active) {
+ in = is_connected_input_ep(w);
+ dapm_clear_walk(w->codec);
+ w->power = (in != 0) ? 1 : 0;
+ dapm_update_bits(w);
+ continue;
+ }
+
+ /* active DAC */
+ if (w->id == snd_soc_dapm_dac && w->active) {
+ out = is_connected_output_ep(w);
+ dapm_clear_walk(w->codec);
+ w->power = (out != 0) ? 1 : 0;
+ dapm_update_bits(w);
+ continue;
+ }
+
+ /* programmable gain/attenuation */
+ if (w->id == snd_soc_dapm_pga) {
+ int on;
+ in = is_connected_input_ep(w);
+ dapm_clear_walk(w->codec);
+ out = is_connected_output_ep(w);
+ dapm_clear_walk(w->codec);
+ w->power = on = (out != 0 && in != 0) ? 1 : 0;
+
+ if (!on)
+ dapm_set_pga(w, on); /* lower volume to reduce pops */
+ dapm_update_bits(w);
+ if (on)
+ dapm_set_pga(w, on); /* restore volume from zero */
+
+ continue;
+ }
+
+ /* pre and post event widgets */
+ if (w->id == snd_soc_dapm_pre) {
+ if (!w->event)
+ continue;
+
+ if (event == SND_SOC_DAPM_STREAM_START) {
+ ret = w->event(w, SND_SOC_DAPM_PRE_PMU);
+ if (ret < 0)
+ return ret;
+ } else if (event == SND_SOC_DAPM_STREAM_STOP) {
+ ret = w->event(w, SND_SOC_DAPM_PRE_PMD);
+ if (ret < 0)
+ return ret;
+ }
+ continue;
+ }
+ if (w->id == snd_soc_dapm_post) {
+ if (!w->event)
+ continue;
+
+ if (event == SND_SOC_DAPM_STREAM_START) {
+ ret = w->event(w, SND_SOC_DAPM_POST_PMU);
+ if (ret < 0)
+ return ret;
+ } else if (event == SND_SOC_DAPM_STREAM_STOP) {
+ ret = w->event(w, SND_SOC_DAPM_POST_PMD);
+ if (ret < 0)
+ return ret;
+ }
+ continue;
+ }
+
+ /* all other widgets */
+ in = is_connected_input_ep(w);
+ dapm_clear_walk(w->codec);
+ out = is_connected_output_ep(w);
+ dapm_clear_walk(w->codec);
+ power = (out != 0 && in != 0) ? 1 : 0;
+ power_change = (w->power == power) ? 0: 1;
+ w->power = power;
+
+ /* call any power change event handlers */
+ if (power_change) {
+ if (w->event) {
+ dbg("power %s event for %s flags %x\n",
+ w->power ? "on" : "off", w->name, w->event_flags);
+ if (power) {
+ /* power up event */
+ if (w->event_flags & SND_SOC_DAPM_PRE_PMU) {
+ ret = w->event(w, SND_SOC_DAPM_PRE_PMU);
+ if (ret < 0)
+ return ret;
+ }
+ dapm_update_bits(w);
+ if (w->event_flags & SND_SOC_DAPM_POST_PMU){
+ ret = w->event(w, SND_SOC_DAPM_POST_PMU);
+ if (ret < 0)
+ return ret;
+ }
+ } else {
+ /* power down event */
+ if (w->event_flags & SND_SOC_DAPM_PRE_PMD) {
+ ret = w->event(w, SND_SOC_DAPM_PRE_PMD);
+ if (ret < 0)
+ return ret;
+ }
+ dapm_update_bits(w);
+ if (w->event_flags & SND_SOC_DAPM_POST_PMD) {
+ ret = w->event(w, SND_SOC_DAPM_POST_PMD);
+ if (ret < 0)
+ return ret;
+ }
+ }
+ } else
+ /* no event handler */
+ dapm_update_bits(w);
+ }
+ }
+ }
+
+ return ret;
+}
+
+#if DAPM_DEBUG
+static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action)
+{
+ struct snd_soc_dapm_widget *w;
+ struct snd_soc_dapm_path *p = NULL;
+ int in, out;
+
+ printk("DAPM %s %s\n", codec->name, action);
+
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
+
+ /* only display widgets that effect routing */
+ switch (w->id) {
+ case snd_soc_dapm_pre:
+ case snd_soc_dapm_post:
+ case snd_soc_dapm_vmid:
+ continue;
+ case snd_soc_dapm_mux:
+ case snd_soc_dapm_output:
+ case snd_soc_dapm_input:
+ case snd_soc_dapm_switch:
+ case snd_soc_dapm_hp:
+ case snd_soc_dapm_mic:
+ case snd_soc_dapm_spk:
+ case snd_soc_dapm_line:
+ case snd_soc_dapm_micbias:
+ case snd_soc_dapm_dac:
+ case snd_soc_dapm_adc:
+ case snd_soc_dapm_pga:
+ case snd_soc_dapm_mixer:
+ if (w->name) {
+ in = is_connected_input_ep(w);
+ dapm_clear_walk(w->codec);
+ out = is_connected_output_ep(w);
+ dapm_clear_walk(w->codec);
+ printk("%s: %s in %d out %d\n", w->name,
+ w->power ? "On":"Off",in, out);
+
+ list_for_each_entry(p, &w->sources, list_sink) {
+ if (p->connect)
+ printk(" in %s %s\n", p->name ? p->name : "static",
+ p->source->name);
+ }
+ list_for_each_entry(p, &w->sinks, list_source) {
+ if (p->connect)
+ printk(" out %s %s\n", p->name ? p->name : "static",
+ p->sink->name);
+ }
+ }
+ break;
+ }
+ }
+}
+#endif
+
+/* test and update the power status of a mux widget */
+static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
+ struct snd_kcontrol *kcontrol, int mask,
+ int val, struct soc_enum* e)
+{
+ struct snd_soc_dapm_path *path;
+ int found = 0;
+
+ if (widget->id != snd_soc_dapm_mux)
+ return -ENODEV;
+
+ if (!snd_soc_test_bits(widget->codec, e->reg, mask, val))
+ return 0;
+
+ /* find dapm widget path assoc with kcontrol */
+ list_for_each_entry(path, &widget->codec->dapm_paths, list) {
+ if (path->kcontrol != kcontrol)
+ continue;
+
+ if (!path->name || ! e->texts[val])
+ continue;
+
+ found = 1;
+ /* we now need to match the string in the enum to the path */
+ if (!(strcmp(path->name, e->texts[val])))
+ path->connect = 1; /* new connection */
+ else
+ path->connect = 0; /* old connection must be powered down */
+ }
+
+ if (found)
+ dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
+
+ return 0;
+}
+
+/* test and update the power status of a mixer widget */
+static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
+ struct snd_kcontrol *kcontrol, int reg,
+ int val_mask, int val, int invert)
+{
+ struct snd_soc_dapm_path *path;
+ int found = 0;
+
+ if (widget->id != snd_soc_dapm_mixer)
+ return -ENODEV;
+
+ if (!snd_soc_test_bits(widget->codec, reg, val_mask, val))
+ return 0;
+
+ /* find dapm widget path assoc with kcontrol */
+ list_for_each_entry(path, &widget->codec->dapm_paths, list) {
+ if (path->kcontrol != kcontrol)
+ continue;
+
+ /* found, now check type */
+ found = 1;
+ if (val)
+ /* new connection */
+ path->connect = invert ? 0:1;
+ else
+ /* old connection must be powered down */
+ path->connect = invert ? 1:0;
+ break;
+ }
+
+ if (found)
+ dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
+
+ return 0;
+}
+
+/* show dapm widget status in sys fs */
+static ssize_t dapm_widget_show(struct device *dev,
+ struct device_attribute *attr, char *buf)
+{
+ struct snd_soc_device *devdata = dev_get_drvdata(dev);
+ struct snd_soc_codec *codec = devdata->codec;
+ struct snd_soc_dapm_widget *w;
+ int count = 0;
+ char *state = "not set";
+
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
+
+ /* only display widgets that burnm power */
+ switch (w->id) {
+ case snd_soc_dapm_hp:
+ case snd_soc_dapm_mic:
+ case snd_soc_dapm_spk:
+ case snd_soc_dapm_line:
+ case snd_soc_dapm_micbias:
+ case snd_soc_dapm_dac:
+ case snd_soc_dapm_adc:
+ case snd_soc_dapm_pga:
+ case snd_soc_dapm_mixer:
+ if (w->name)
+ count += sprintf(buf + count, "%s: %s\n",
+ w->name, w->power ? "On":"Off");
+ break;
+ default:
+ break;
+ }
+ }
+
+ switch(codec->dapm_state){
+ case SNDRV_CTL_POWER_D0:
+ state = "D0";
+ break;
+ case SNDRV_CTL_POWER_D1:
+ state = "D1";
+ break;
+ case SNDRV_CTL_POWER_D2:
+ state = "D2";
+ break;
+ case SNDRV_CTL_POWER_D3hot:
+ state = "D3hot";
+ break;
+ case SNDRV_CTL_POWER_D3cold:
+ state = "D3cold";
+ break;
+ }
+ count += sprintf(buf + count, "PM State: %s\n", state);
+
+ return count;
+}
+
+static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL);
+
+int snd_soc_dapm_sys_add(struct device *dev)
+{
+ int ret = 0;
+
+ if (dapm_status)
+ ret = device_create_file(dev, &dev_attr_dapm_widget);
+
+ return ret;
+}
+
+static void snd_soc_dapm_sys_remove(struct device *dev)
+{
+ if (dapm_status)
+ device_remove_file(dev, &dev_attr_dapm_widget);
+}
+
+/* free all dapm widgets and resources */
+static void dapm_free_widgets(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_widget *w, *next_w;
+ struct snd_soc_dapm_path *p, *next_p;
+
+ list_for_each_entry_safe(w, next_w, &codec->dapm_widgets, list) {
+ list_del(&w->list);
+ kfree(w);
+ }
+
+ list_for_each_entry_safe(p, next_p, &codec->dapm_paths, list) {
+ list_del(&p->list);
+ kfree(p->long_name);
+ kfree(p);
+ }
+}
+
+/**
+ * snd_soc_dapm_sync_endpoints - scan and power dapm paths
+ * @codec: audio codec
+ *
+ * Walks all dapm audio paths and powers widgets according to their
+ * stream or path usage.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec)
+{
+ return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_endpoints);
+
+/**
+ * snd_soc_dapm_connect_input - connect dapm widgets
+ * @codec: audio codec
+ * @sink: name of target widget
+ * @control: mixer control name
+ * @source: name of source name
+ *
+ * Connects 2 dapm widgets together via a named audio path. The sink is
+ * the widget receiving the audio signal, whilst the source is the sender
+ * of the audio signal.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink,
+ const char * control, const char *source)
+{
+ struct snd_soc_dapm_path *path;
+ struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w;
+ int ret = 0;
+
+ /* find src and dest widgets */
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
+
+ if (!wsink && !(strcmp(w->name, sink))) {
+ wsink = w;
+ continue;
+ }
+ if (!wsource && !(strcmp(w->name, source))) {
+ wsource = w;
+ }
+ }
+
+ if (wsource == NULL || wsink == NULL)
+ return -ENODEV;
+
+ path = kzalloc(sizeof(struct snd_soc_dapm_path), GFP_KERNEL);
+ if (!path)
+ return -ENOMEM;
+
+ path->source = wsource;
+ path->sink = wsink;
+ INIT_LIST_HEAD(&path->list);
+ INIT_LIST_HEAD(&path->list_source);
+ INIT_LIST_HEAD(&path->list_sink);
+
+ /* check for external widgets */
+ if (wsink->id == snd_soc_dapm_input) {
+ if (wsource->id == snd_soc_dapm_micbias ||
+ wsource->id == snd_soc_dapm_mic ||
+ wsink->id == snd_soc_dapm_line)
+ wsink->ext = 1;
+ }
+ if (wsource->id == snd_soc_dapm_output) {
+ if (wsink->id == snd_soc_dapm_spk ||
+ wsink->id == snd_soc_dapm_hp ||
+ wsink->id == snd_soc_dapm_line)
+ wsource->ext = 1;
+ }
+
+ /* connect static paths */
+ if (control == NULL) {
+ list_add(&path->list, &codec->dapm_paths);
+ list_add(&path->list_sink, &wsink->sources);
+ list_add(&path->list_source, &wsource->sinks);
+ path->connect = 1;
+ return 0;
+ }
+
+ /* connect dynamic paths */
+ switch(wsink->id) {
+ case snd_soc_dapm_adc:
+ case snd_soc_dapm_dac:
+ case snd_soc_dapm_pga:
+ case snd_soc_dapm_input:
+ case snd_soc_dapm_output:
+ case snd_soc_dapm_micbias:
+ case snd_soc_dapm_vmid:
+ case snd_soc_dapm_pre:
+ case snd_soc_dapm_post:
+ list_add(&path->list, &codec->dapm_paths);
+ list_add(&path->list_sink, &wsink->sources);
+ list_add(&path->list_source, &wsource->sinks);
+ path->connect = 1;
+ return 0;
+ case snd_soc_dapm_mux:
+ ret = dapm_connect_mux(codec, wsource, wsink, path, control,
+ &wsink->kcontrols[0]);
+ if (ret != 0)
+ goto err;
+ break;
+ case snd_soc_dapm_switch:
+ case snd_soc_dapm_mixer:
+ ret = dapm_connect_mixer(codec, wsource, wsink, path, control);
+ if (ret != 0)
+ goto err;
+ break;
+ case snd_soc_dapm_hp:
+ case snd_soc_dapm_mic:
+ case snd_soc_dapm_line:
+ case snd_soc_dapm_spk:
+ list_add(&path->list, &codec->dapm_paths);
+ list_add(&path->list_sink, &wsink->sources);
+ list_add(&path->list_source, &wsource->sinks);
+ path->connect = 0;
+ return 0;
+ }
+ return 0;
+
+err:
+ printk(KERN_WARNING "asoc: no dapm match for %s --> %s --> %s\n", source,
+ control, sink);
+ kfree(path);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_connect_input);
+
+/**
+ * snd_soc_dapm_new_widgets - add new dapm widgets
+ * @codec: audio codec
+ *
+ * Checks the codec for any new dapm widgets and creates them if found.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_widget *w;
+
+ mutex_lock(&codec->mutex);
+ list_for_each_entry(w, &codec->dapm_widgets, list)
+ {
+ if (w->new)
+ continue;
+
+ switch(w->id) {
+ case snd_soc_dapm_switch:
+ case snd_soc_dapm_mixer:
+ dapm_new_mixer(codec, w);
+ break;
+ case snd_soc_dapm_mux:
+ dapm_new_mux(codec, w);
+ break;
+ case snd_soc_dapm_adc:
+ case snd_soc_dapm_dac:
+ case snd_soc_dapm_pga:
+ dapm_new_pga(codec, w);
+ break;
+ case snd_soc_dapm_input:
+ case snd_soc_dapm_output:
+ case snd_soc_dapm_micbias:
+ case snd_soc_dapm_spk:
+ case snd_soc_dapm_hp:
+ case snd_soc_dapm_mic:
+ case snd_soc_dapm_line:
+ case snd_soc_dapm_vmid:
+ case snd_soc_dapm_pre:
+ case snd_soc_dapm_post:
+ break;
+ }
+ w->new = 1;
+ }
+
+ dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
+ mutex_unlock(&codec->mutex);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets);
+
+/**
+ * snd_soc_dapm_get_volsw - dapm mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a dapm mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0x0f;
+ int rshift = (kcontrol->private_value >> 12) & 0x0f;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0x01;
+
+ /* return the saved value if we are powered down */
+ if (widget->id == snd_soc_dapm_pga && !widget->power) {
+ ucontrol->value.integer.value[0] = widget->saved_value;
+ return 0;
+ }
+
+ ucontrol->value.integer.value[0] =
+ (snd_soc_read(widget->codec, reg) >> shift) & mask;
+ if (shift != rshift)
+ ucontrol->value.integer.value[1] =
+ (snd_soc_read(widget->codec, reg) >> rshift) & mask;
+ if (invert) {
+ ucontrol->value.integer.value[0] =
+ mask - ucontrol->value.integer.value[0];
+ if (shift != rshift)
+ ucontrol->value.integer.value[1] =
+ mask - ucontrol->value.integer.value[1];
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw);
+
+/**
+ * snd_soc_dapm_put_volsw - dapm mixer set callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a dapm mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0x0f;
+ int rshift = (kcontrol->private_value >> 12) & 0x0f;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0x01;
+ unsigned short val, val2, val_mask;
+ int ret;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+
+ if (invert)
+ val = mask - val;
+ val_mask = mask << shift;
+ val = val << shift;
+ if (shift != rshift) {
+ val2 = (ucontrol->value.integer.value[1] & mask);
+ if (invert)
+ val2 = mask - val2;
+ val_mask |= mask << rshift;
+ val |= val2 << rshift;
+ }
+
+ mutex_lock(&widget->codec->mutex);
+ widget->value = val;
+
+ /* save volume value if the widget is powered down */
+ if (widget->id == snd_soc_dapm_pga && !widget->power) {
+ widget->saved_value = val;
+ mutex_unlock(&widget->codec->mutex);
+ return 1;
+ }
+
+ dapm_mixer_update_power(widget, kcontrol, reg, val_mask, val, invert);
+ if (widget->event) {
+ if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
+ ret = widget->event(widget, SND_SOC_DAPM_PRE_REG);
+ if (ret < 0)
+ goto out;
+ }
+ ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
+ if (widget->event_flags & SND_SOC_DAPM_POST_REG)
+ ret = widget->event(widget, SND_SOC_DAPM_POST_REG);
+ } else
+ ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
+
+out:
+ mutex_unlock(&widget->codec->mutex);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw);
+
+/**
+ * snd_soc_dapm_get_enum_double - dapm enumerated double mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a dapm enumerated double mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned short val, bitmask;
+
+ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
+ ;
+ val = snd_soc_read(widget->codec, e->reg);
+ ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1);
+ if (e->shift_l != e->shift_r)
+ ucontrol->value.enumerated.item[1] =
+ (val >> e->shift_r) & (bitmask - 1);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);
+
+/**
+ * snd_soc_dapm_put_enum_double - dapm enumerated double mixer set callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a dapm enumerated double mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned short val, mux;
+ unsigned short mask, bitmask;
+ int ret = 0;
+
+ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
+ ;
+ if (ucontrol->value.enumerated.item[0] > e->mask - 1)
+ return -EINVAL;
+ mux = ucontrol->value.enumerated.item[0];
+ val = mux << e->shift_l;
+ mask = (bitmask - 1) << e->shift_l;
+ if (e->shift_l != e->shift_r) {
+ if (ucontrol->value.enumerated.item[1] > e->mask - 1)
+ return -EINVAL;
+ val |= ucontrol->value.enumerated.item[1] << e->shift_r;
+ mask |= (bitmask - 1) << e->shift_r;
+ }
+
+ mutex_lock(&widget->codec->mutex);
+ widget->value = val;
+ dapm_mux_update_power(widget, kcontrol, mask, mux, e);
+ if (widget->event) {
+ if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
+ ret = widget->event(widget, SND_SOC_DAPM_PRE_REG);
+ if (ret < 0)
+ goto out;
+ }
+ ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
+ if (widget->event_flags & SND_SOC_DAPM_POST_REG)
+ ret = widget->event(widget, SND_SOC_DAPM_POST_REG);
+ } else
+ ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
+
+out:
+ mutex_unlock(&widget->codec->mutex);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double);
+
+/**
+ * snd_soc_dapm_new_control - create new dapm control
+ * @codec: audio codec
+ * @widget: widget template
+ *
+ * Creates a new dapm control based upon the template.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_new_control(struct snd_soc_codec *codec,
+ const struct snd_soc_dapm_widget *widget)
+{
+ struct snd_soc_dapm_widget *w;
+
+ if ((w = dapm_cnew_widget(widget)) == NULL)
+ return -ENOMEM;
+
+ w->codec = codec;
+ INIT_LIST_HEAD(&w->sources);
+ INIT_LIST_HEAD(&w->sinks);
+ INIT_LIST_HEAD(&w->list);
+ list_add(&w->list, &codec->dapm_widgets);
+
+ /* machine layer set ups unconnected pins and insertions */
+ w->connected = 1;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
+
+/**
+ * snd_soc_dapm_stream_event - send a stream event to the dapm core
+ * @codec: audio codec
+ * @stream: stream name
+ * @event: stream event
+ *
+ * Sends a stream event to the dapm core. The core then makes any
+ * necessary widget power changes.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
+ char *stream, int event)
+{
+ struct snd_soc_dapm_widget *w;
+
+ if (stream == NULL)
+ return 0;
+
+ mutex_lock(&codec->mutex);
+ list_for_each_entry(w, &codec->dapm_widgets, list)
+ {
+ if (!w->sname)
+ continue;
+ dbg("widget %s\n %s stream %s event %d\n", w->name, w->sname,
+ stream, event);
+ if (strstr(w->sname, stream)) {
+ switch(event) {
+ case SND_SOC_DAPM_STREAM_START:
+ w->active = 1;
+ break;
+ case SND_SOC_DAPM_STREAM_STOP:
+ w->active = 0;
+ break;
+ case SND_SOC_DAPM_STREAM_SUSPEND:
+ if (w->active)
+ w->suspend = 1;
+ w->active = 0;
+ break;
+ case SND_SOC_DAPM_STREAM_RESUME:
+ if (w->suspend) {
+ w->active = 1;
+ w->suspend = 0;
+ }
+ break;
+ case SND_SOC_DAPM_STREAM_PAUSE_PUSH:
+ break;
+ case SND_SOC_DAPM_STREAM_PAUSE_RELEASE:
+ break;
+ }
+ }
+ }
+ mutex_unlock(&codec->mutex);
+
+ dapm_power_widgets(codec, event);
+ dump_dapm(codec, __FUNCTION__);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
+
+/**
+ * snd_soc_dapm_set_endpoint - set audio endpoint status
+ * @codec: audio codec
+ * @endpoint: audio signal endpoint (or start point)
+ * @status: point status
+ *
+ * Set audio endpoint status - connected or disconnected.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
+ char *endpoint, int status)
+{
+ struct snd_soc_dapm_widget *w;
+
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
+ if (!strcmp(w->name, endpoint)) {
+ w->connected = status;
+ }
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint);
+
+/**
+ * snd_soc_dapm_free - free dapm resources
+ * @socdev: SoC device
+ *
+ * Free all dapm widgets and resources.
+ */
+void snd_soc_dapm_free(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+
+ snd_soc_dapm_sys_remove(socdev->dev);
+ dapm_free_widgets(codec);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC");
+MODULE_LICENSE("GPL");