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-rw-r--r--sound/aoa/fabrics/layout.c2
-rw-r--r--sound/oss/au1550_ac97.c16
-rw-r--r--sound/ppc/awacs.c24
-rw-r--r--sound/ppc/burgundy.c4
-rw-r--r--sound/ppc/pmac.c18
-rw-r--r--sound/soc/au1x/Kconfig10
-rw-r--r--sound/soc/au1x/Makefile4
-rw-r--r--sound/soc/au1x/db1200.c141
-rw-r--r--sound/soc/au1x/dbdma2.c14
-rw-r--r--sound/soc/au1x/sample-ac97.c144
-rw-r--r--sound/soc/codecs/uda1380.c2
-rw-r--r--sound/soc/codecs/wm8350.c8
-rw-r--r--sound/soc/fsl/efika-audio-fabric.c2
-rw-r--r--sound/soc/fsl/pcm030-audio-fabric.c2
-rw-r--r--sound/soc/omap/omap-mcbsp.c144
-rw-r--r--sound/soc/omap/omap-mcbsp.h4
-rw-r--r--sound/soc/sh/siu.h2
-rw-r--r--sound/soc/sh/siu_pcm.c2
18 files changed, 342 insertions, 201 deletions
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index 586965f9605..7a437da0564 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -768,7 +768,7 @@ static int check_codec(struct aoa_codec *codec,
"required property %s not present\n", propname);
return -ENODEV;
}
- if (*ref != codec->node->linux_phandle) {
+ if (*ref != codec->node->phandle) {
printk(KERN_INFO "snd-aoa-fabric-layout: "
"%s doesn't match!\n", propname);
return -ENODEV;
diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c
index 4191acccbcd..c1070e33b32 100644
--- a/sound/oss/au1550_ac97.c
+++ b/sound/oss/au1550_ac97.c
@@ -614,7 +614,8 @@ start_adc(struct au1550_state *s)
/* Put two buffers on the ring to get things started.
*/
for (i=0; i<2; i++) {
- au1xxx_dbdma_put_dest(db->dmanr, db->nextIn, db->dma_fragsize);
+ au1xxx_dbdma_put_dest(db->dmanr, virt_to_phys(db->nextIn),
+ db->dma_fragsize, DDMA_FLAGS_IE);
db->nextIn += db->dma_fragsize;
if (db->nextIn >= db->rawbuf + db->dmasize)
@@ -732,8 +733,9 @@ static void dac_dma_interrupt(int irq, void *dev_id)
db->dma_qcount--;
if (db->count >= db->fragsize) {
- if (au1xxx_dbdma_put_source(db->dmanr, db->nextOut,
- db->fragsize) == 0) {
+ if (au1xxx_dbdma_put_source(db->dmanr,
+ virt_to_phys(db->nextOut), db->fragsize,
+ DDMA_FLAGS_IE) == 0) {
err("qcount < 2 and no ring room!");
}
db->nextOut += db->fragsize;
@@ -777,7 +779,8 @@ static void adc_dma_interrupt(int irq, void *dev_id)
/* Put a new empty buffer on the destination DMA.
*/
- au1xxx_dbdma_put_dest(dp->dmanr, dp->nextIn, dp->dma_fragsize);
+ au1xxx_dbdma_put_dest(dp->dmanr, virt_to_phys(dp->nextIn),
+ dp->dma_fragsize, DDMA_FLAGS_IE);
dp->nextIn += dp->dma_fragsize;
if (dp->nextIn >= dp->rawbuf + dp->dmasize)
@@ -1177,8 +1180,9 @@ au1550_write(struct file *file, const char *buffer, size_t count, loff_t * ppos)
* we know the dma has stopped.
*/
while ((db->dma_qcount < 2) && (db->count >= db->fragsize)) {
- if (au1xxx_dbdma_put_source(db->dmanr, db->nextOut,
- db->fragsize) == 0) {
+ if (au1xxx_dbdma_put_source(db->dmanr,
+ virt_to_phys(db->nextOut), db->fragsize,
+ DDMA_FLAGS_IE) == 0) {
err("qcount < 2 and no ring room!");
}
db->nextOut += db->fragsize;
diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c
index 2e156467b81..b36679384b2 100644
--- a/sound/ppc/awacs.c
+++ b/sound/ppc/awacs.c
@@ -751,8 +751,8 @@ static void snd_pmac_awacs_suspend(struct snd_pmac *chip)
static void snd_pmac_awacs_resume(struct snd_pmac *chip)
{
- if (machine_is_compatible("PowerBook3,1")
- || machine_is_compatible("PowerBook3,2")) {
+ if (of_machine_is_compatible("PowerBook3,1")
+ || of_machine_is_compatible("PowerBook3,2")) {
msleep(100);
snd_pmac_awacs_write_reg(chip, 1,
chip->awacs_reg[1] & ~MASK_PAROUT);
@@ -780,16 +780,16 @@ static void snd_pmac_awacs_resume(struct snd_pmac *chip)
}
#endif /* CONFIG_PM */
-#define IS_PM7500 (machine_is_compatible("AAPL,7500") \
- || machine_is_compatible("AAPL,8500") \
- || machine_is_compatible("AAPL,9500"))
-#define IS_PM5500 (machine_is_compatible("AAPL,e411"))
-#define IS_BEIGE (machine_is_compatible("AAPL,Gossamer"))
-#define IS_IMAC1 (machine_is_compatible("PowerMac2,1"))
-#define IS_IMAC2 (machine_is_compatible("PowerMac2,2") \
- || machine_is_compatible("PowerMac4,1"))
-#define IS_G4AGP (machine_is_compatible("PowerMac3,1"))
-#define IS_LOMBARD (machine_is_compatible("PowerBook1,1"))
+#define IS_PM7500 (of_machine_is_compatible("AAPL,7500") \
+ || of_machine_is_compatible("AAPL,8500") \
+ || of_machine_is_compatible("AAPL,9500"))
+#define IS_PM5500 (of_machine_is_compatible("AAPL,e411"))
+#define IS_BEIGE (of_machine_is_compatible("AAPL,Gossamer"))
+#define IS_IMAC1 (of_machine_is_compatible("PowerMac2,1"))
+#define IS_IMAC2 (of_machine_is_compatible("PowerMac2,2") \
+ || of_machine_is_compatible("PowerMac4,1"))
+#define IS_G4AGP (of_machine_is_compatible("PowerMac3,1"))
+#define IS_LOMBARD (of_machine_is_compatible("PowerBook1,1"))
static int imac1, imac2;
diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c
index 0accfe49735..1f72e1c786b 100644
--- a/sound/ppc/burgundy.c
+++ b/sound/ppc/burgundy.c
@@ -582,7 +582,7 @@ static int snd_pmac_burgundy_detect_headphone(struct snd_pmac *chip)
static void snd_pmac_burgundy_update_automute(struct snd_pmac *chip, int do_notify)
{
if (chip->auto_mute) {
- int imac = machine_is_compatible("iMac");
+ int imac = of_machine_is_compatible("iMac");
int reg, oreg;
reg = oreg = snd_pmac_burgundy_rcb(chip,
MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES);
@@ -620,7 +620,7 @@ static void snd_pmac_burgundy_update_automute(struct snd_pmac *chip, int do_noti
*/
int __devinit snd_pmac_burgundy_init(struct snd_pmac *chip)
{
- int imac = machine_is_compatible("iMac");
+ int imac = of_machine_is_compatible("iMac");
int i, err;
/* Checks to see the chip is alive and kicking */
diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c
index 7bc492ee77e..85081172403 100644
--- a/sound/ppc/pmac.c
+++ b/sound/ppc/pmac.c
@@ -922,11 +922,11 @@ static void __devinit detect_byte_swap(struct snd_pmac *chip)
}
/* it seems the Pismo & iBook can't byte-swap in hardware. */
- if (machine_is_compatible("PowerBook3,1") ||
- machine_is_compatible("PowerBook2,1"))
+ if (of_machine_is_compatible("PowerBook3,1") ||
+ of_machine_is_compatible("PowerBook2,1"))
chip->can_byte_swap = 0 ;
- if (machine_is_compatible("PowerBook2,1"))
+ if (of_machine_is_compatible("PowerBook2,1"))
chip->can_duplex = 0;
}
@@ -959,11 +959,11 @@ static int __devinit snd_pmac_detect(struct snd_pmac *chip)
chip->control_mask = MASK_IEPC | MASK_IEE | 0x11; /* default */
/* check machine type */
- if (machine_is_compatible("AAPL,3400/2400")
- || machine_is_compatible("AAPL,3500"))
+ if (of_machine_is_compatible("AAPL,3400/2400")
+ || of_machine_is_compatible("AAPL,3500"))
chip->is_pbook_3400 = 1;
- else if (machine_is_compatible("PowerBook1,1")
- || machine_is_compatible("AAPL,PowerBook1998"))
+ else if (of_machine_is_compatible("PowerBook1,1")
+ || of_machine_is_compatible("AAPL,PowerBook1998"))
chip->is_pbook_G3 = 1;
chip->node = of_find_node_by_name(NULL, "awacs");
sound = of_node_get(chip->node);
@@ -1033,8 +1033,8 @@ static int __devinit snd_pmac_detect(struct snd_pmac *chip)
}
if (of_device_is_compatible(sound, "tumbler")) {
chip->model = PMAC_TUMBLER;
- chip->can_capture = machine_is_compatible("PowerMac4,2")
- || machine_is_compatible("PowerBook4,1");
+ chip->can_capture = of_machine_is_compatible("PowerMac4,2")
+ || of_machine_is_compatible("PowerBook4,1");
chip->can_duplex = 0;
// chip->can_byte_swap = 0; /* FIXME: check this */
chip->num_freqs = ARRAY_SIZE(tumbler_freqs);
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
index 410a893aa66..4b67140fdec 100644
--- a/sound/soc/au1x/Kconfig
+++ b/sound/soc/au1x/Kconfig
@@ -22,11 +22,13 @@ config SND_SOC_AU1XPSC_AC97
##
## Boards
##
-config SND_SOC_SAMPLE_PSC_AC97
- tristate "Sample Au12x0/Au1550 PSC AC97 sound machine"
+config SND_SOC_DB1200
+ tristate "DB1200 AC97+I2S audio support"
depends on SND_SOC_AU1XPSC
select SND_SOC_AU1XPSC_AC97
select SND_SOC_AC97_CODEC
+ select SND_SOC_AU1XPSC_I2S
+ select SND_SOC_WM8731
help
- This is a sample AC97 sound machine for use in Au12x0/Au1550
- based systems which have audio on PSC1 (e.g. Db1200 demoboard).
+ Select this option to enable audio (AC97 or I2S) on the
+ Alchemy/AMD/RMI DB1200 demoboard.
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
index 6c6950b8003..16873076e8c 100644
--- a/sound/soc/au1x/Makefile
+++ b/sound/soc/au1x/Makefile
@@ -8,6 +8,6 @@ obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o
obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o
# Boards
-snd-soc-sample-ac97-objs := sample-ac97.o
+snd-soc-db1200-objs := db1200.o
-obj-$(CONFIG_SND_SOC_SAMPLE_PSC_AC97) += snd-soc-sample-ac97.o
+obj-$(CONFIG_SND_SOC_DB1200) += snd-soc-db1200.o
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
new file mode 100644
index 00000000000..cdf7be1b9b9
--- /dev/null
+++ b/sound/soc/au1x/db1200.c
@@ -0,0 +1,141 @@
+/*
+ * DB1200 ASoC audio fabric support code.
+ *
+ * (c) 2008-9 Manuel Lauss <manuel.lauss@gmail.com>
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1xxx_psc.h>
+#include <asm/mach-au1x00/au1xxx_dbdma.h>
+#include <asm/mach-db1x00/bcsr.h>
+
+#include "../codecs/ac97.h"
+#include "../codecs/wm8731.h"
+#include "psc.h"
+
+/*------------------------- AC97 PART ---------------------------*/
+
+static struct snd_soc_dai_link db1200_ac97_dai = {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &au1xpsc_ac97_dai,
+ .codec_dai = &ac97_dai,
+};
+
+static struct snd_soc_card db1200_ac97_machine = {
+ .name = "DB1200_AC97",
+ .dai_link = &db1200_ac97_dai,
+ .num_links = 1,
+ .platform = &au1xpsc_soc_platform,
+};
+
+static struct snd_soc_device db1200_ac97_devdata = {
+ .card = &db1200_ac97_machine,
+ .codec_dev = &soc_codec_dev_ac97,
+};
+
+/*------------------------- I2S PART ---------------------------*/
+
+static int db1200_i2s_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ /* WM8731 has its own 12MHz crystal */
+ snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
+ 12000000, SND_SOC_CLOCK_IN);
+
+ /* codec is bitclock and lrclk master */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ goto out;
+
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_LEFT_J |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ goto out;
+
+ ret = 0;
+out:
+ return ret;
+}
+
+static struct snd_soc_ops db1200_i2s_wm8731_ops = {
+ .startup = db1200_i2s_startup,
+};
+
+static struct snd_soc_dai_link db1200_i2s_dai = {
+ .name = "WM8731",
+ .stream_name = "WM8731 PCM",
+ .cpu_dai = &au1xpsc_i2s_dai,
+ .codec_dai = &wm8731_dai,
+ .ops = &db1200_i2s_wm8731_ops,
+};
+
+static struct snd_soc_card db1200_i2s_machine = {
+ .name = "DB1200_I2S",
+ .dai_link = &db1200_i2s_dai,
+ .num_links = 1,
+ .platform = &au1xpsc_soc_platform,
+};
+
+static struct snd_soc_device db1200_i2s_devdata = {
+ .card = &db1200_i2s_machine,
+ .codec_dev = &soc_codec_dev_wm8731,
+};
+
+/*------------------------- COMMON PART ---------------------------*/
+
+static struct platform_device *db1200_asoc_dev;
+
+static int __init db1200_audio_load(void)
+{
+ int ret;
+
+ ret = -ENOMEM;
+ db1200_asoc_dev = platform_device_alloc("soc-audio", -1);
+ if (!db1200_asoc_dev)
+ goto out;
+
+ /* DB1200 board setup set PSC1MUX to preferred audio device */
+ if (bcsr_read(BCSR_RESETS) & BCSR_RESETS_PSC1MUX)
+ platform_set_drvdata(db1200_asoc_dev, &db1200_i2s_devdata);
+ else
+ platform_set_drvdata(db1200_asoc_dev, &db1200_ac97_devdata);
+
+ db1200_ac97_devdata.dev = &db1200_asoc_dev->dev;
+ db1200_i2s_devdata.dev = &db1200_asoc_dev->dev;
+ ret = platform_device_add(db1200_asoc_dev);
+
+ if (ret) {
+ platform_device_put(db1200_asoc_dev);
+ db1200_asoc_dev = NULL;
+ }
+out:
+ return ret;
+}
+
+static void __exit db1200_audio_unload(void)
+{
+ platform_device_unregister(db1200_asoc_dev);
+}
+
+module_init(db1200_audio_load);
+module_exit(db1200_audio_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("DB1200 ASoC audio support");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 19e4d37eba1..6d9f4c62494 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -51,8 +51,8 @@ struct au1xpsc_audio_dmadata {
struct snd_pcm_substream *substream;
unsigned long curr_period; /* current segment DDMA is working on */
unsigned long q_period; /* queue period(s) */
- unsigned long dma_area; /* address of queued DMA area */
- unsigned long dma_area_s; /* start address of DMA area */
+ dma_addr_t dma_area; /* address of queued DMA area */
+ dma_addr_t dma_area_s; /* start address of DMA area */
unsigned long pos; /* current byte position being played */
unsigned long periods; /* number of SG segments in total */
unsigned long period_bytes; /* size in bytes of one SG segment */
@@ -94,8 +94,7 @@ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd)
{
- au1xxx_dbdma_put_source_flags(cd->ddma_chan,
- (void *)phys_to_virt(cd->dma_area),
+ au1xxx_dbdma_put_source(cd->ddma_chan, cd->dma_area,
cd->period_bytes, DDMA_FLAGS_IE);
/* update next-to-queue period */
@@ -109,9 +108,8 @@ static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd)
static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd)
{
- au1xxx_dbdma_put_dest_flags(cd->ddma_chan,
- (void *)phys_to_virt(cd->dma_area),
- cd->period_bytes, DDMA_FLAGS_IE);
+ au1xxx_dbdma_put_dest(cd->ddma_chan, cd->dma_area,
+ cd->period_bytes, DDMA_FLAGS_IE);
/* update next-to-queue period */
++cd->q_period;
@@ -233,7 +231,7 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
pcd->substream = substream;
pcd->period_bytes = params_period_bytes(params);
pcd->periods = params_periods(params);
- pcd->dma_area_s = pcd->dma_area = (unsigned long)runtime->dma_addr;
+ pcd->dma_area_s = pcd->dma_area = runtime->dma_addr;
pcd->q_period = 0;
pcd->curr_period = 0;
pcd->pos = 0;
diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
deleted file mode 100644
index 27683eb7905..00000000000
--- a/sound/soc/au1x/sample-ac97.c
+++ /dev/null
@@ -1,144 +0,0 @@
-/*
- * Sample Au12x0/Au1550 PSC AC97 sound machine.
- *
- * Copyright (c) 2007-2008 Manuel Lauss <mano@roarinelk.homelinux.net>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms outlined in the file COPYING at the root of this
- * source archive.
- *
- * This is a very generic AC97 sound machine driver for boards which
- * have (AC97) audio at PSC1 (e.g. DB1200 demoboards).
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/soc-dapm.h>
-#include <asm/mach-au1x00/au1000.h>
-#include <asm/mach-au1x00/au1xxx_psc.h>
-#include <asm/mach-au1x00/au1xxx_dbdma.h>
-
-#include "../codecs/ac97.h"
-#include "psc.h"
-
-static int au1xpsc_sample_ac97_init(struct snd_soc_codec *codec)
-{
- snd_soc_dapm_sync(codec);
- return 0;
-}
-
-static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = {
- .name = "AC97",
- .stream_name = "AC97 HiFi",
- .cpu_dai = &au1xpsc_ac97_dai, /* see psc-ac97.c */
- .codec_dai = &ac97_dai, /* see codecs/ac97.c */
- .init = au1xpsc_sample_ac97_init,
- .ops = NULL,
-};
-
-static struct snd_soc_card au1xpsc_sample_ac97_machine = {
- .name = "Au1xxx PSC AC97 Audio",
- .dai_link = &au1xpsc_sample_ac97_dai,
- .num_links = 1,
-};
-
-static struct snd_soc_device au1xpsc_sample_ac97_devdata = {
- .card = &au1xpsc_sample_ac97_machine,
- .platform = &au1xpsc_soc_platform, /* see dbdma2.c */
- .codec_dev = &soc_codec_dev_ac97,
-};
-
-static struct resource au1xpsc_psc1_res[] = {
- [0] = {
- .start = CPHYSADDR(PSC1_BASE_ADDR),
- .end = CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff,
- .flags = IORESOURCE_MEM,
- },
- [1] = {
-#ifdef CONFIG_SOC_AU1200
- .start = AU1200_PSC1_INT,
- .end = AU1200_PSC1_INT,
-#elif defined(CONFIG_SOC_AU1550)
- .start = AU1550_PSC1_INT,
- .end = AU1550_PSC1_INT,
-#endif
- .flags = IORESOURCE_IRQ,
- },
- [2] = {
- .start = DSCR_CMD0_PSC1_TX,
- .end = DSCR_CMD0_PSC1_TX,
- .flags = IORESOURCE_DMA,
- },
- [3] = {
- .start = DSCR_CMD0_PSC1_RX,
- .end = DSCR_CMD0_PSC1_RX,
- .flags = IORESOURCE_DMA,
- },
-};
-
-static struct platform_device *au1xpsc_sample_ac97_dev;
-
-static int __init au1xpsc_sample_ac97_load(void)
-{
- int ret;
-
-#ifdef CONFIG_SOC_AU1200
- unsigned long io;
-
- /* modify sys_pinfunc for AC97 on PSC1 */
- io = au_readl(SYS_PINFUNC);
- io |= SYS_PINFUNC_P1C;
- io &= ~(SYS_PINFUNC_P1A | SYS_PINFUNC_P1B);
- au_writel(io, SYS_PINFUNC);
- au_sync();
-#endif
-
- ret = -ENOMEM;
-
- /* setup PSC clock source for AC97 part: external clock provided
- * by codec. The psc-ac97.c driver depends on this setting!
- */
- au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET);
- au_sync();
-
- au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1);
- if (!au1xpsc_sample_ac97_dev)
- goto out;
-
- au1xpsc_sample_ac97_dev->resource =
- kmemdup(au1xpsc_psc1_res, sizeof(struct resource) *
- ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL);
- au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res);
- au1xpsc_sample_ac97_dev->id = 1;
-
- platform_set_drvdata(au1xpsc_sample_ac97_dev,
- &au1xpsc_sample_ac97_devdata);
- au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev;
- ret = platform_device_add(au1xpsc_sample_ac97_dev);
-
- if (ret) {
- platform_device_put(au1xpsc_sample_ac97_dev);
- au1xpsc_sample_ac97_dev = NULL;
- }
-
-out:
- return ret;
-}
-
-static void __exit au1xpsc_sample_ac97_exit(void)
-{
- platform_device_unregister(au1xpsc_sample_ac97_dev);
-}
-
-module_init(au1xpsc_sample_ac97_load);
-module_exit(au1xpsc_sample_ac97_exit);
-
-MODULE_LICENSE("GPL");
-MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine");
-MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index a2763c2e734..9cd0a66b766 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -137,7 +137,7 @@ static void uda1380_flush_work(struct work_struct *work)
{
int bit, reg;
- for_each_bit(bit, &uda1380_cache_dirty, UDA1380_CACHEREGNUM - 0x10) {
+ for_each_set_bit(bit, &uda1380_cache_dirty, UDA1380_CACHEREGNUM - 0x10) {
reg = 0x10 + bit;
pr_debug("uda1380: flush reg %x val %x:\n", reg,
uda1380_read_reg_cache(uda1380_codec, reg));
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 718ef912e75..df2c6d9617f 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1349,7 +1349,7 @@ static irqreturn_t wm8350_hp_jack_handler(int irq, void *data)
int mask;
struct wm8350_jack_data *jack = NULL;
- switch (irq) {
+ switch (irq - wm8350->irq_base) {
case WM8350_IRQ_CODEC_JCK_DET_L:
jack = &priv->hpl;
mask = WM8350_JACK_L_LVL;
@@ -1424,7 +1424,7 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena);
/* Sync status */
- wm8350_hp_jack_handler(irq, priv);
+ wm8350_hp_jack_handler(irq + wm8350->irq_base, priv);
return 0;
}
@@ -1521,8 +1521,8 @@ static int wm8350_remove(struct platform_device *pdev)
WM8350_JDL_ENA | WM8350_JDR_ENA);
wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
- wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
- wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+ wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, priv);
+ wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, priv);
priv->hpl.jack = NULL;
priv->hpr.jack = NULL;
diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c
index 3326e2a1e86..1a5b8e0d6a3 100644
--- a/sound/soc/fsl/efika-audio-fabric.c
+++ b/sound/soc/fsl/efika-audio-fabric.c
@@ -55,7 +55,7 @@ static __init int efika_fabric_init(void)
struct platform_device *pdev;
int rc;
- if (!machine_is_compatible("bplan,efika"))
+ if (!of_machine_is_compatible("bplan,efika"))
return -ENODEV;
card.platform = &mpc5200_audio_dma_platform;
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index b928ef7d28e..6644cba7cbf 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -55,7 +55,7 @@ static __init int pcm030_fabric_init(void)
struct platform_device *pdev;
int rc;
- if (!machine_is_compatible("phytec,pcm030"))
+ if (!of_machine_is_compatible("phytec,pcm030"))
return -ENODEV;
card.platform = &mpc5200_audio_dma_platform;
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index d2972566418..e814a9591f7 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -39,6 +39,14 @@
#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000)
+#define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \
+ xhandler_get, xhandler_put) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = omap_mcbsp_st_info_volsw, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = (unsigned long) &(struct soc_mixer_control) \
+ {.min = xmin, .max = xmax} }
+
struct omap_mcbsp_data {
unsigned int bus_id;
struct omap_mcbsp_reg_cfg regs;
@@ -82,11 +90,11 @@ static const int omap1_dma_reqs[][2] = {};
static const unsigned long omap1_mcbsp_port[][2] = {};
#endif
-#if defined(CONFIG_ARCH_OMAP24XX) || defined(CONFIG_ARCH_OMAP34XX)
+#if defined(CONFIG_ARCH_OMAP2) || defined(CONFIG_ARCH_OMAP3)
static const int omap24xx_dma_reqs[][2] = {
{ OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX },
{ OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX },
-#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX)
+#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3)
{ OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX },
{ OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX },
{ OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX },
@@ -124,7 +132,7 @@ static const unsigned long omap2430_mcbsp_port[][2] = {
static const unsigned long omap2430_mcbsp_port[][2] = {};
#endif
-#if defined(CONFIG_ARCH_OMAP34XX)
+#if defined(CONFIG_ARCH_OMAP3)
static const unsigned long omap34xx_mcbsp_port[][2] = {
{ OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
@@ -639,6 +647,136 @@ struct snd_soc_dai omap_mcbsp_dai[] = {
EXPORT_SYMBOL_GPL(omap_mcbsp_dai);
+int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ int max = mc->max;
+ int min = mc->min;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = min;
+ uinfo->value.integer.max = max;
+ return 0;
+}
+
+#define OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(id, channel) \
+static int \
+omap_mcbsp##id##_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \
+ struct snd_ctl_elem_value *uc) \
+{ \
+ struct soc_mixer_control *mc = \
+ (struct soc_mixer_control *)kc->private_value; \
+ int max = mc->max; \
+ int min = mc->min; \
+ int val = uc->value.integer.value[0]; \
+ \
+ if (val < min || val > max) \
+ return -EINVAL; \
+ \
+ /* OMAP McBSP implementation uses index values 0..4 */ \
+ return omap_st_set_chgain((id)-1, channel, val); \
+}
+
+#define OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(id, channel) \
+static int \
+omap_mcbsp##id##_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \
+ struct snd_ctl_elem_value *uc) \
+{ \
+ s16 chgain; \
+ \
+ if (omap_st_get_chgain((id)-1, channel, &chgain)) \
+ return -EAGAIN; \
+ \
+ uc->value.integer.value[0] = chgain; \
+ return 0; \
+}
+
+OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(2, 0)
+OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(2, 1)
+OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(3, 0)
+OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(3, 1)
+OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(2, 0)
+OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(2, 1)
+OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(3, 0)
+OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(3, 1)
+
+static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ u8 value = ucontrol->value.integer.value[0];
+
+ if (value == omap_st_is_enabled(mc->reg))
+ return 0;
+
+ if (value)
+ omap_st_enable(mc->reg);
+ else
+ omap_st_disable(mc->reg);
+
+ return 1;
+}
+
+static int omap_mcbsp_st_get_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ ucontrol->value.integer.value[0] = omap_st_is_enabled(mc->reg);
+ return 0;
+}
+
+static const struct snd_kcontrol_new omap_mcbsp2_st_controls[] = {
+ SOC_SINGLE_EXT("McBSP2 Sidetone Switch", 1, 0, 1, 0,
+ omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode),
+ OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 0 Volume",
+ -32768, 32767,
+ omap_mcbsp2_get_st_ch0_volume,
+ omap_mcbsp2_set_st_ch0_volume),
+ OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 1 Volume",
+ -32768, 32767,
+ omap_mcbsp2_get_st_ch1_volume,
+ omap_mcbsp2_set_st_ch1_volume),
+};
+
+static const struct snd_kcontrol_new omap_mcbsp3_st_controls[] = {
+ SOC_SINGLE_EXT("McBSP3 Sidetone Switch", 2, 0, 1, 0,
+ omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode),
+ OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 0 Volume",
+ -32768, 32767,
+ omap_mcbsp3_get_st_ch0_volume,
+ omap_mcbsp3_set_st_ch0_volume),
+ OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 1 Volume",
+ -32768, 32767,
+ omap_mcbsp3_get_st_ch1_volume,
+ omap_mcbsp3_set_st_ch1_volume),
+};
+
+int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id)
+{
+ if (!cpu_is_omap34xx())
+ return -ENODEV;
+
+ switch (mcbsp_id) {
+ case 1: /* McBSP 2 */
+ return snd_soc_add_controls(codec, omap_mcbsp2_st_controls,
+ ARRAY_SIZE(omap_mcbsp2_st_controls));
+ case 2: /* McBSP 3 */
+ return snd_soc_add_controls(codec, omap_mcbsp3_st_controls,
+ ARRAY_SIZE(omap_mcbsp3_st_controls));
+ default:
+ break;
+ }
+
+ return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls);
+
static int __init snd_omap_mcbsp_init(void)
{
return snd_soc_register_dais(omap_mcbsp_dai,
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index 647d2f981ab..6c363e5f438 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -50,11 +50,13 @@ enum omap_mcbsp_div {
#undef NUM_LINKS
#define NUM_LINKS 3
#endif
-#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX)
+#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3)
#undef NUM_LINKS
#define NUM_LINKS 5
#endif
extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS];
+int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id);
+
#endif
diff --git a/sound/soc/sh/siu.h b/sound/soc/sh/siu.h
index 9cc04ab2bce..c0bfab8fed3 100644
--- a/sound/soc/sh/siu.h
+++ b/sound/soc/sh/siu.h
@@ -72,7 +72,7 @@ struct siu_firmware {
#include <linux/interrupt.h>
#include <linux/io.h>
-#include <asm/dma-sh.h>
+#include <asm/dmaengine.h>
#include <sound/core.h>
#include <sound/pcm.h>
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index c5efc30f013..ba7f8d05d97 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -32,7 +32,7 @@
#include <sound/pcm_params.h>
#include <sound/soc-dai.h>
-#include <asm/dma-sh.h>
+#include <asm/dmaengine.h>
#include <asm/siu.h>
#include "siu.h"
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-09 19:21:30 +0000