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-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/ak4104.c6
-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/imx/imx-pcm-fiq.c40
-rw-r--r--sound/soc/omap/Kconfig3
-rw-r--r--sound/soc/omap/Makefile2
-rw-r--r--sound/soc/omap/mcpdm.c484
-rw-r--r--sound/soc/omap/mcpdm.h151
-rw-r--r--sound/soc/omap/omap-mcbsp.c146
-rw-r--r--sound/soc/omap/omap-mcbsp.h4
-rw-r--r--sound/soc/omap/omap-mcpdm.c251
-rw-r--r--sound/soc/omap/omap-mcpdm.h29
-rw-r--r--sound/soc/omap/omap-pcm.c15
-rw-r--r--sound/soc/omap/omap-pcm.h4
-rw-r--r--sound/soc/sh/fsi.c46
-rw-r--r--sound/soc/sh/siu.h2
-rw-r--r--sound/soc/sh/siu_pcm.c2
-rw-r--r--sound/soc/soc-core.c20
25 files changed, 1313 insertions, 219 deletions
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/ak4104.c b/sound/soc/codecs/ak4104.c
index b9ef7e45891..b68d99fb6af 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -90,12 +90,10 @@ static int ak4104_spi_write(struct snd_soc_codec *codec, unsigned int reg,
if (reg >= codec->reg_cache_size)
return -EINVAL;
- reg &= AK4104_REG_MASK;
- reg |= AK4104_WRITE;
-
/* only write to the hardware if value has changed */
if (cache[reg] != value) {
- u8 tmp[2] = { reg, value };
+ u8 tmp[2] = { (reg & AK4104_REG_MASK) | AK4104_WRITE, value };
+
if (spi_write(spi, tmp, sizeof(tmp))) {
dev_err(&spi->dev, "SPI write failed\n");
return -EIO;
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/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c
index 5532579ece4..d9cb9849b03 100644
--- a/sound/soc/imx/imx-pcm-fiq.c
+++ b/sound/soc/imx/imx-pcm-fiq.c
@@ -35,22 +35,25 @@
struct imx_pcm_runtime_data {
int period;
int periods;
- unsigned long dma_addr;
- int dma;
unsigned long offset;
+ unsigned long last_offset;
unsigned long size;
- unsigned long period_cnt;
- void *buf;
struct timer_list timer;
- int period_time;
+ int poll_time;
};
+static inline void imx_ssi_set_next_poll(struct imx_pcm_runtime_data *iprtd)
+{
+ iprtd->timer.expires = jiffies + iprtd->poll_time;
+}
+
static void imx_ssi_timer_callback(unsigned long data)
{
struct snd_pcm_substream *substream = (void *)data;
struct snd_pcm_runtime *runtime = substream->runtime;
struct imx_pcm_runtime_data *iprtd = runtime->private_data;
struct pt_regs regs;
+ unsigned long delta;
get_fiq_regs(&regs);
@@ -59,9 +62,25 @@ static void imx_ssi_timer_callback(unsigned long data)
else
iprtd->offset = regs.ARM_r9 & 0xffff;
- iprtd->timer.expires = jiffies + iprtd->period_time;
+ /* How much data have we transferred since the last period report? */
+ if (iprtd->offset >= iprtd->last_offset)
+ delta = iprtd->offset - iprtd->last_offset;
+ else
+ delta = runtime->buffer_size + iprtd->offset
+ - iprtd->last_offset;
+
+ /* If we've transferred at least a period then report it and
+ * reset our poll time */
+ if (delta >= runtime->period_size) {
+ snd_pcm_period_elapsed(substream);
+ iprtd->last_offset = iprtd->offset;
+
+ imx_ssi_set_next_poll(iprtd);
+ }
+
+ /* Restart the timer; if we didn't report we'll run on the next tick */
add_timer(&iprtd->timer);
- snd_pcm_period_elapsed(substream);
+
}
static struct fiq_handler fh = {
@@ -76,9 +95,10 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream,
iprtd->size = params_buffer_bytes(params);
iprtd->periods = params_periods(params);
- iprtd->period = params_period_bytes(params);
+ iprtd->period = params_period_bytes(params) ;
iprtd->offset = 0;
- iprtd->period_time = HZ / (params_rate(params) / params_period_size(params));
+ iprtd->last_offset = 0;
+ iprtd->poll_time = HZ / (params_rate(params) / params_period_size(params));
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
@@ -114,7 +134,7 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- iprtd->timer.expires = jiffies + iprtd->period_time;
+ imx_ssi_set_next_poll(iprtd);
add_timer(&iprtd->timer);
if (++fiq_enable == 1)
enable_fiq(imx_pcm_fiq);
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 18ebdc7d0a5..f11963c2187 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -6,6 +6,9 @@ config SND_OMAP_SOC_MCBSP
tristate
select OMAP_MCBSP
+config SND_OMAP_SOC_MCPDM
+ tristate
+
config SND_OMAP_SOC_N810
tristate "SoC Audio support for Nokia N810"
depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 19283e5edfb..0bc00ca14b3 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -1,9 +1,11 @@
# OMAP Platform Support
snd-soc-omap-objs := omap-pcm.o
snd-soc-omap-mcbsp-objs := omap-mcbsp.o
+snd-soc-omap-mcpdm-objs := omap-mcpdm.o mcpdm.o
obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o
obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
+obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o
# OMAP Machine Support
snd-soc-n810-objs := n810.o
diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c
new file mode 100644
index 00000000000..ad8df6cfae8
--- /dev/null
+++ b/sound/soc/omap/mcpdm.c
@@ -0,0 +1,484 @@
+/*
+ * mcpdm.c -- McPDM interface driver
+ *
+ * Author: Jorge Eduardo Candelaria <x0107209@ti.com>
+ * Copyright (C) 2009 - Texas Instruments, 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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <linux/wait.h>
+#include <linux/interrupt.h>
+#include <linux/err.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <linux/irq.h>
+
+#include "mcpdm.h"
+
+static struct omap_mcpdm *mcpdm;
+
+static inline void omap_mcpdm_write(u16 reg, u32 val)
+{
+ __raw_writel(val, mcpdm->io_base + reg);
+}
+
+static inline int omap_mcpdm_read(u16 reg)
+{
+ return __raw_readl(mcpdm->io_base + reg);
+}
+
+static void omap_mcpdm_reg_dump(void)
+{
+ dev_dbg(mcpdm->dev, "***********************\n");
+ dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n",
+ omap_mcpdm_read(MCPDM_IRQSTATUS_RAW));
+ dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n",
+ omap_mcpdm_read(MCPDM_IRQSTATUS));
+ dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n",
+ omap_mcpdm_read(MCPDM_IRQENABLE_SET));
+ dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n",
+ omap_mcpdm_read(MCPDM_IRQENABLE_CLR));
+ dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n",
+ omap_mcpdm_read(MCPDM_IRQWAKE_EN));
+ dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n",
+ omap_mcpdm_read(MCPDM_DMAENABLE_SET));
+ dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n",
+ omap_mcpdm_read(MCPDM_DMAENABLE_CLR));
+ dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n",
+ omap_mcpdm_read(MCPDM_DMAWAKEEN));
+ dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n",
+ omap_mcpdm_read(MCPDM_CTRL));
+ dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n",
+ omap_mcpdm_read(MCPDM_DN_DATA));
+ dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n",
+ omap_mcpdm_read(MCPDM_UP_DATA));
+ dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n",
+ omap_mcpdm_read(MCPDM_FIFO_CTRL_DN));
+ dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n",
+ omap_mcpdm_read(MCPDM_FIFO_CTRL_UP));
+ dev_dbg(mcpdm->dev, "DN_OFFSET: 0x%04x\n",
+ omap_mcpdm_read(MCPDM_DN_OFFSET));
+ dev_dbg(mcpdm->dev, "***********************\n");
+}
+
+/*
+ * Takes the McPDM module in and out of reset state.
+ * Uplink and downlink can be reset individually.
+ */
+static void omap_mcpdm_reset_capture(int reset)
+{
+ int ctrl = omap_mcpdm_read(MCPDM_CTRL);
+
+ if (reset)
+ ctrl |= SW_UP_RST;
+ else
+ ctrl &= ~SW_UP_RST;
+
+ omap_mcpdm_write(MCPDM_CTRL, ctrl);
+}
+
+static void omap_mcpdm_reset_playback(int reset)
+{
+ int ctrl = omap_mcpdm_read(MCPDM_CTRL);
+
+ if (reset)
+ ctrl |= SW_DN_RST;
+ else
+ ctrl &= ~SW_DN_RST;
+
+ omap_mcpdm_write(MCPDM_CTRL, ctrl);
+}
+
+/*
+ * Enables the transfer through the PDM interface to/from the Phoenix
+ * codec by enabling the corresponding UP or DN channels.
+ */
+void omap_mcpdm_start(int stream)
+{
+ int ctrl = omap_mcpdm_read(MCPDM_CTRL);
+
+ if (stream)
+ ctrl |= mcpdm->up_channels;
+ else
+ ctrl |= mcpdm->dn_channels;
+
+ omap_mcpdm_write(MCPDM_CTRL, ctrl);
+}
+
+/*
+ * Disables the transfer through the PDM interface to/from the Phoenix
+ * codec by disabling the corresponding UP or DN channels.
+ */
+void omap_mcpdm_stop(int stream)
+{
+ int ctrl = omap_mcpdm_read(MCPDM_CTRL);
+
+ if (stream)
+ ctrl &= ~mcpdm->up_channels;
+ else
+ ctrl &= ~mcpdm->dn_channels;
+
+ omap_mcpdm_write(MCPDM_CTRL, ctrl);
+}
+
+/*
+ * Configures McPDM uplink for audio recording.
+ * This function should be called before omap_mcpdm_start.
+ */
+int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink)
+{
+ int irq_mask = 0;
+ int ctrl;
+
+ if (!uplink)
+ return -EINVAL;
+
+ mcpdm->uplink = uplink;
+
+ /* Enable irq request generation */
+ irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK;
+ omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask);
+
+ /* Configure uplink threshold */
+ if (uplink->threshold > UP_THRES_MAX)
+ uplink->threshold = UP_THRES_MAX;
+
+ omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold);
+
+ /* Configure DMA controller */
+ omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE);
+
+ /* Set pdm out format */
+ ctrl = omap_mcpdm_read(MCPDM_CTRL);
+ ctrl &= ~PDMOUTFORMAT;
+ ctrl |= uplink->format & PDMOUTFORMAT;
+
+ /* Uplink channels */
+ mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK);
+
+ omap_mcpdm_write(MCPDM_CTRL, ctrl);
+
+ return 0;
+}
+
+/*
+ * Configures McPDM downlink for audio playback.
+ * This function should be called before omap_mcpdm_start.
+ */
+int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink)
+{
+ int irq_mask = 0;
+ int ctrl;
+
+ if (!downlink)
+ return -EINVAL;
+
+ mcpdm->downlink = downlink;
+
+ /* Enable irq request generation */
+ irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK;
+ omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask);
+
+ /* Configure uplink threshold */
+ if (downlink->threshold > DN_THRES_MAX)
+ downlink->threshold = DN_THRES_MAX;
+
+ omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold);
+
+ /* Enable DMA request generation */
+ omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE);
+
+ /* Set pdm out format */
+ ctrl = omap_mcpdm_read(MCPDM_CTRL);
+ ctrl &= ~PDMOUTFORMAT;
+ ctrl |= downlink->format & PDMOUTFORMAT;
+
+ /* Downlink channels */
+ mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK);
+
+ omap_mcpdm_write(MCPDM_CTRL, ctrl);
+
+ return 0;
+}
+
+/*
+ * Cleans McPDM uplink configuration.
+ * This function should be called when the stream is closed.
+ */
+int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink)
+{
+ int irq_mask = 0;
+
+ if (!uplink)
+ return -EINVAL;
+
+ /* Disable irq request generation */
+ irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK;
+ omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask);
+
+ /* Disable DMA request generation */
+ omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE);
+
+ /* Clear Downlink channels */
+ mcpdm->up_channels = 0;
+
+ mcpdm->uplink = NULL;
+
+ return 0;
+}
+
+/*
+ * Cleans McPDM downlink configuration.
+ * This function should be called when the stream is closed.
+ */
+int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink)
+{
+ int irq_mask = 0;
+
+ if (!downlink)
+ return -EINVAL;
+
+ /* Disable irq request generation */
+ irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK;
+ omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask);
+
+ /* Disable DMA request generation */
+ omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE);
+
+ /* clear Downlink channels */
+ mcpdm->dn_channels = 0;
+
+ mcpdm->downlink = NULL;
+
+ return 0;
+}
+
+static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id)
+{
+ struct omap_mcpdm *mcpdm_irq = dev_id;
+ int irq_status;
+
+ irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS);
+
+ /* Acknowledge irq event */
+ omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status);
+
+ if (irq & MCPDM_DN_IRQ_FULL) {
+ dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status);
+ omap_mcpdm_reset_playback(1);
+ omap_mcpdm_playback_open(mcpdm_irq->downlink);
+ omap_mcpdm_reset_playback(0);
+ }
+
+ if (irq & MCPDM_DN_IRQ_EMPTY) {
+ dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status);
+ omap_mcpdm_reset_playback(1);
+ omap_mcpdm_playback_open(mcpdm_irq->downlink);
+ omap_mcpdm_reset_playback(0);
+ }
+
+ if (irq & MCPDM_DN_IRQ) {
+ dev_dbg(mcpdm_irq->dev, "DN write request\n");
+ }
+
+ if (irq & MCPDM_UP_IRQ_FULL) {
+ dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status);
+ omap_mcpdm_reset_capture(1);
+ omap_mcpdm_capture_open(mcpdm_irq->uplink);
+ omap_mcpdm_reset_capture(0);
+ }
+
+ if (irq & MCPDM_UP_IRQ_EMPTY) {
+ dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status);
+ omap_mcpdm_reset_capture(1);
+ omap_mcpdm_capture_open(mcpdm_irq->uplink);
+ omap_mcpdm_reset_capture(0);
+ }
+
+ if (irq & MCPDM_UP_IRQ) {
+ dev_dbg(mcpdm_irq->dev, "UP write request\n");
+ }
+
+ return IRQ_HANDLED;
+}
+
+int omap_mcpdm_request(void)
+{
+ int ret;
+
+ clk_enable(mcpdm->clk);
+
+ spin_lock(&mcpdm->lock);
+
+ if (!mcpdm->free) {
+ dev_err(mcpdm->dev, "McPDM interface is in use\n");
+ spin_unlock(&mcpdm->lock);
+ ret = -EBUSY;
+ goto err;
+ }
+ mcpdm->free = 0;
+
+ spin_unlock(&mcpdm->lock);
+
+ /* Disable lines while request is ongoing */
+ omap_mcpdm_write(MCPDM_CTRL, 0x00);
+
+ ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler,
+ 0, "McPDM", (void *)mcpdm);
+ if (ret) {
+ dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n");
+ goto err;
+ }
+
+ return 0;
+
+err:
+ clk_disable(mcpdm->clk);
+ return ret;
+}
+
+void omap_mcpdm_free(void)
+{
+ spin_lock(&mcpdm->lock);
+ if (mcpdm->free) {
+ dev_err(mcpdm->dev, "McPDM interface is already free\n");
+ spin_unlock(&mcpdm->lock);
+ return;
+ }
+ mcpdm->free = 1;
+ spin_unlock(&mcpdm->lock);
+
+ clk_disable(mcpdm->clk);
+
+ free_irq(mcpdm->irq, (void *)mcpdm);
+}
+
+/* Enable/disable DC offset cancelation for the analog
+ * headset path (PDM channels 1 and 2).
+ */
+int omap_mcpdm_set_offset(int offset1, int offset2)
+{
+ int offset;
+
+ if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX))
+ return -EINVAL;
+
+ offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2);
+
+ /* offset cancellation for channel 1 */
+ if (offset1)
+ offset |= DN_OFST_RX1_EN;
+ else
+ offset &= ~DN_OFST_RX1_EN;
+
+ /* offset cancellation for channel 2 */
+ if (offset2)
+ offset |= DN_OFST_RX2_EN;
+ else
+ offset &= ~DN_OFST_RX2_EN;
+
+ omap_mcpdm_write(MCPDM_DN_OFFSET, offset);
+
+ return 0;
+}
+
+static int __devinit omap_mcpdm_probe(struct platform_device *pdev)
+{
+ struct resource *res;
+ int ret = 0;
+
+ mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL);
+ if (!mcpdm) {
+ ret = -ENOMEM;
+ goto exit;
+ }
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (res == NULL) {
+ dev_err(&pdev->dev, "no resource\n");
+ goto err_resource;
+ }
+
+ spin_lock_init(&mcpdm->lock);
+ mcpdm->free = 1;
+ mcpdm->io_base = ioremap(res->start, resource_size(res));
+ if (!mcpdm->io_base) {
+ ret = -ENOMEM;
+ goto err_resource;
+ }
+
+ mcpdm->irq = platform_get_irq(pdev, 0);
+
+ mcpdm->clk = clk_get(&pdev->dev, "pdm_ck");
+ if (IS_ERR(mcpdm->clk)) {
+ ret = PTR_ERR(mcpdm->clk);
+ dev_err(&pdev->dev, "unable to get pdm_ck: %d\n", ret);
+ goto err_clk;
+ }
+
+ mcpdm->dev = &pdev->dev;
+ platform_set_drvdata(pdev, mcpdm);
+
+ return 0;
+
+err_clk:
+ iounmap(mcpdm->io_base);
+err_resource:
+ kfree(mcpdm);
+exit:
+ return ret;
+}
+
+static int __devexit omap_mcpdm_remove(struct platform_device *pdev)
+{
+ struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev);
+
+ platform_set_drvdata(pdev, NULL);
+
+ clk_put(mcpdm_ptr->clk);
+
+ iounmap(mcpdm_ptr->io_base);
+
+ mcpdm_ptr->clk = NULL;
+ mcpdm_ptr->free = 0;
+ mcpdm_ptr->dev = NULL;
+
+ kfree(mcpdm_ptr);
+
+ return 0;
+}
+
+static struct platform_driver omap_mcpdm_driver = {
+ .probe = omap_mcpdm_probe,
+ .remove = __devexit_p(omap_mcpdm_remove),
+ .driver = {
+ .name = "omap-mcpdm",
+ },
+};
+
+static struct platform_device *omap_mcpdm_device;
+
+static int __init omap_mcpdm_init(void)
+{
+ return platform_driver_register(&omap_mcpdm_driver);
+}
+arch_initcall(omap_mcpdm_init);
diff --git a/sound/soc/omap/mcpdm.h b/sound/soc/omap/mcpdm.h
new file mode 100644
index 00000000000..7bb326ef088
--- /dev/null
+++ b/sound/soc/omap/mcpdm.h
@@ -0,0 +1,151 @@
+/*
+ * mcpdm.h -- Defines for McPDM driver
+ *
+ * Author: Jorge Eduardo Candelaria <x0107209@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+/* McPDM registers */
+
+#define MCPDM_REVISION 0x00
+#define MCPDM_SYSCONFIG 0x10
+#define MCPDM_IRQSTATUS_RAW 0x24
+#define MCPDM_IRQSTATUS 0x28
+#define MCPDM_IRQENABLE_SET 0x2C
+#define MCPDM_IRQENABLE_CLR 0x30
+#define MCPDM_IRQWAKE_EN 0x34
+#define MCPDM_DMAENABLE_SET 0x38
+#define MCPDM_DMAENABLE_CLR 0x3C
+#define MCPDM_DMAWAKEEN 0x40
+#define MCPDM_CTRL 0x44
+#define MCPDM_DN_DATA 0x48
+#define MCPDM_UP_DATA 0x4C
+#define MCPDM_FIFO_CTRL_DN 0x50
+#define MCPDM_FIFO_CTRL_UP 0x54
+#define MCPDM_DN_OFFSET 0x58
+
+/*
+ * MCPDM_IRQ bit fields
+ * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR
+ */
+
+#define MCPDM_DN_IRQ (1 << 0)
+#define MCPDM_DN_IRQ_EMPTY (1 << 1)
+#define MCPDM_DN_IRQ_ALMST_EMPTY (1 << 2)
+#define MCPDM_DN_IRQ_FULL (1 << 3)
+
+#define MCPDM_UP_IRQ (1 << 8)
+#define MCPDM_UP_IRQ_EMPTY (1 << 9)
+#define MCPDM_UP_IRQ_ALMST_FULL (1 << 10)
+#define MCPDM_UP_IRQ_FULL (1 << 11)
+
+#define MCPDM_DOWNLINK_IRQ_MASK 0x00F
+#define MCPDM_UPLINK_IRQ_MASK 0xF00
+
+/*
+ * MCPDM_DMAENABLE bit fields
+ */
+
+#define DMA_DN_ENABLE 0x1
+#define DMA_UP_ENABLE 0x2
+
+/*
+ * MCPDM_CTRL bit fields
+ */
+
+#define PDM_UP1_EN 0x0001
+#define PDM_UP2_EN 0x0002
+#define PDM_UP3_EN 0x0004
+#define PDM_DN1_EN 0x0008
+#define PDM_DN2_EN 0x0010
+#define PDM_DN3_EN 0x0020
+#define PDM_DN4_EN 0x0040
+#define PDM_DN5_EN 0x0080
+#define PDMOUTFORMAT 0x0100
+#define CMD_INT 0x0200
+#define STATUS_INT 0x0400
+#define SW_UP_RST 0x0800
+#define SW_DN_RST 0x1000
+#define PDM_UP_MASK 0x007
+#define PDM_DN_MASK 0x0F8
+#define PDM_CMD_MASK 0x200
+#define PDM_STATUS_MASK 0x400
+
+
+#define PDMOUTFORMAT_LJUST (0 << 8)
+#define PDMOUTFORMAT_RJUST (1 << 8)
+
+/*
+ * MCPDM_FIFO_CTRL bit fields
+ */
+
+#define UP_THRES_MAX 0xF
+#define DN_THRES_MAX 0xF
+
+/*
+ * MCPDM_DN_OFFSET bit fields
+ */
+
+#define DN_OFST_RX1_EN 0x0001
+#define DN_OFST_RX2_EN 0x0100
+
+#define DN_OFST_RX1 1
+#define DN_OFST_RX2 9
+#define DN_OFST_MAX 0x1F
+
+#define MCPDM_UPLINK 1
+#define MCPDM_DOWNLINK 2
+
+struct omap_mcpdm_link {
+ int irq_mask;
+ int threshold;
+ int format;
+ int channels;
+};
+
+struct omap_mcpdm_platform_data {
+ unsigned long phys_base;
+ u16 irq;
+};
+
+struct omap_mcpdm {
+ struct device *dev;
+ unsigned long phys_base;
+ void __iomem *io_base;
+ u8 free;
+ int irq;
+
+ spinlock_t lock;
+ struct omap_mcpdm_platform_data *pdata;
+ struct clk *clk;
+ struct omap_mcpdm_link *downlink;
+ struct omap_mcpdm_link *uplink;
+ struct completion irq_completion;
+
+ int dn_channels;
+ int up_channels;
+};
+
+extern void omap_mcpdm_start(int stream);
+extern void omap_mcpdm_stop(int stream);
+extern int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink);
+extern int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink);
+extern int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink);
+extern int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink);
+extern int omap_mcpdm_request(void);
+extern void omap_mcpdm_free(void);
+extern int omap_mcpdm_set_offset(int offset1, int offset2);
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 6bbbd2ab0ee..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 },
@@ -287,6 +295,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode;
+ omap_mcbsp_dai_dma_params[id][substream->stream].data_type =
+ OMAP_DMA_DATA_TYPE_S16;
cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
if (mcbsp_data->configured) {
@@ -637,6 +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/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
new file mode 100644
index 00000000000..25f19e4728b
--- /dev/null
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -0,0 +1,251 @@
+/*
+ * omap-mcpdm.c -- OMAP ALSA SoC DAI driver using McPDM port
+ *
+ * Copyright (C) 2009 Texas Instruments
+ *
+ * Author: Misael Lopez Cruz <x0052729@ti.com>
+ * Contact: Jorge Eduardo Candelaria <x0107209@ti.com>
+ * Margarita Olaya <magi.olaya@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <plat/control.h>
+#include <plat/dma.h>
+#include <plat/mcbsp.h>
+#include "mcpdm.h"
+#include "omap-mcpdm.h"
+#include "omap-pcm.h"
+
+struct omap_mcpdm_data {
+ struct omap_mcpdm_link *links;
+ int active;
+};
+
+static struct omap_mcpdm_link omap_mcpdm_links[] = {
+ /* downlink */
+ {
+ .irq_mask = MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL,
+ .threshold = 1,
+ .format = PDMOUTFORMAT_LJUST,
+ },
+ /* uplink */
+ {
+ .irq_mask = MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL,
+ .threshold = 1,
+ .format = PDMOUTFORMAT_LJUST,
+ },
+};
+
+static struct omap_mcpdm_data mcpdm_data = {
+ .links = omap_mcpdm_links,
+ .active = 0,
+};
+
+/*
+ * Stream DMA parameters
+ */
+static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = {
+ {
+ .name = "Audio playback",
+ .dma_req = OMAP44XX_DMA_MCPDM_DL,
+ .data_type = OMAP_DMA_DATA_TYPE_S32,
+ .sync_mode = OMAP_DMA_SYNC_PACKET,
+ .packet_size = 16,
+ .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_DN_DATA,
+ },
+ {
+ .name = "Audio capture",
+ .dma_req = OMAP44XX_DMA_MCPDM_UP,
+ .data_type = OMAP_DMA_DATA_TYPE_S32,
+ .sync_mode = OMAP_DMA_SYNC_PACKET,
+ .packet_size = 16,
+ .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_UP_DATA,
+ },
+};
+
+static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int err = 0;
+
+ if (!cpu_dai->active)
+ err = omap_mcpdm_request();
+
+ return err;
+}
+
+static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ if (!cpu_dai->active)
+ omap_mcpdm_free();
+}
+
+static int omap_mcpdm_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data;
+ int stream = substream->stream;
+ int err = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (!mcpdm_priv->active++)
+ omap_mcpdm_start(stream);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (!--mcpdm_priv->active)
+ omap_mcpdm_stop(stream);
+ break;
+ default:
+ err = -EINVAL;
+ }
+
+ return err;
+}
+
+static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data;
+ struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links;
+ int stream = substream->stream;
+ int channels, err, link_mask = 0;
+
+ cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream];
+
+ channels = params_channels(params);
+ switch (channels) {
+ case 4:
+ if (stream == SNDRV_PCM_STREAM_CAPTURE)
+ /* up to 2 channels for capture */
+ return -EINVAL;
+ link_mask |= 1 << 3;
+ case 3:
+ if (stream == SNDRV_PCM_STREAM_CAPTURE)
+ /* up to 2 channels for capture */
+ return -EINVAL;
+ link_mask |= 1 << 2;
+ case 2:
+ link_mask |= 1 << 1;
+ case 1:
+ link_mask |= 1 << 0;
+ break;
+ default:
+ /* unsupported number of channels */
+ return -EINVAL;
+ }
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ mcpdm_links[stream].channels = link_mask << 3;
+ err = omap_mcpdm_playback_open(&mcpdm_links[stream]);
+ } else {
+ mcpdm_links[stream].channels = link_mask << 0;
+ err = omap_mcpdm_capture_open(&mcpdm_links[stream]);
+ }
+
+ return err;
+}
+
+static int omap_mcpdm_dai_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data;
+ struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links;
+ int stream = substream->stream;
+ int err;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ err = omap_mcpdm_playback_close(&mcpdm_links[stream]);
+ else
+ err = omap_mcpdm_capture_close(&mcpdm_links[stream]);
+
+ return err;
+}
+
+static struct snd_soc_dai_ops omap_mcpdm_dai_ops = {
+ .startup = omap_mcpdm_dai_startup,
+ .shutdown = omap_mcpdm_dai_shutdown,
+ .trigger = omap_mcpdm_dai_trigger,
+ .hw_params = omap_mcpdm_dai_hw_params,
+ .hw_free = omap_mcpdm_dai_hw_free,
+};
+
+#define OMAP_MCPDM_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+#define OMAP_MCPDM_FORMATS (SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai omap_mcpdm_dai = {
+ .name = "omap-mcpdm",
+ .id = -1,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 4,
+ .rates = OMAP_MCPDM_RATES,
+ .formats = OMAP_MCPDM_FORMATS,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = OMAP_MCPDM_RATES,
+ .formats = OMAP_MCPDM_FORMATS,
+ },
+ .ops = &omap_mcpdm_dai_ops,
+ .private_data = &mcpdm_data,
+};
+EXPORT_SYMBOL_GPL(omap_mcpdm_dai);
+
+static int __init snd_omap_mcpdm_init(void)
+{
+ return snd_soc_register_dai(&omap_mcpdm_dai);
+}
+module_init(snd_omap_mcpdm_init);
+
+static void __exit snd_omap_mcpdm_exit(void)
+{
+ snd_soc_unregister_dai(&omap_mcpdm_dai);
+}
+module_exit(snd_omap_mcpdm_exit);
+
+MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
+MODULE_DESCRIPTION("OMAP PDM SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcpdm.h b/sound/soc/omap/omap-mcpdm.h
new file mode 100644
index 00000000000..73b80d55934
--- /dev/null
+++ b/sound/soc/omap/omap-mcpdm.h
@@ -0,0 +1,29 @@
+/*
+ * omap-mcpdm.h
+ *
+ * Copyright (C) 2009 Texas Instruments
+ *
+ * Contact: Misael Lopez Cruz <x0052729@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __OMAP_MCPDM_H__
+#define __OMAP_MCPDM_H__
+
+extern struct snd_soc_dai omap_mcpdm_dai;
+
+#endif /* End of __OMAP_MCPDM_H__ */
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 9db2770e964..825db385f01 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -37,7 +37,8 @@ static const struct snd_pcm_hardware omap_pcm_hardware = {
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE |
SNDRV_PCM_INFO_RESUME,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
.period_bytes_min = 32,
.period_bytes_max = 64 * 1024,
.periods_min = 2,
@@ -149,6 +150,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
struct omap_runtime_data *prtd = runtime->private_data;
struct omap_pcm_dma_data *dma_data = prtd->dma_data;
struct omap_dma_channel_params dma_params;
+ int bytes;
/* return if this is a bufferless transfer e.g.
* codec <--> BT codec or GSM modem -- lg FIXME */
@@ -156,11 +158,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
return 0;
memset(&dma_params, 0, sizeof(dma_params));
- /*
- * Note: Regardless of interface data formats supported by OMAP McBSP
- * or EAC blocks, internal representation is always fixed 16-bit/sample
- */
- dma_params.data_type = OMAP_DMA_DATA_TYPE_S16;
+ dma_params.data_type = dma_data->data_type;
dma_params.trigger = dma_data->dma_req;
dma_params.sync_mode = dma_data->sync_mode;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -170,6 +168,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
dma_params.src_start = runtime->dma_addr;
dma_params.dst_start = dma_data->port_addr;
dma_params.dst_port = OMAP_DMA_PORT_MPUI;
+ dma_params.dst_fi = dma_data->packet_size;
} else {
dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT;
dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC;
@@ -177,6 +176,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
dma_params.src_start = dma_data->port_addr;
dma_params.dst_start = runtime->dma_addr;
dma_params.src_port = OMAP_DMA_PORT_MPUI;
+ dma_params.src_fi = dma_data->packet_size;
}
/*
* Set DMA transfer frame size equal to ALSA period size and frame
@@ -184,7 +184,8 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
* we can transfer the whole ALSA buffer with single DMA transfer but
* still can get an interrupt at each period bounary
*/
- dma_params.elem_count = snd_pcm_lib_period_bytes(substream) / 2;
+ bytes = snd_pcm_lib_period_bytes(substream);
+ dma_params.elem_count = bytes >> dma_data->data_type;
dma_params.frame_count = runtime->periods;
omap_set_dma_params(prtd->dma_ch, &dma_params);
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
index 38a821dd411..b19975d2690 100644
--- a/sound/soc/omap/omap-pcm.h
+++ b/sound/soc/omap/omap-pcm.h
@@ -29,8 +29,10 @@ struct omap_pcm_dma_data {
char *name; /* stream identifier */
int dma_req; /* DMA request line */
unsigned long port_addr; /* transmit/receive register */
- int sync_mode; /* DMA sync mode */
void (*set_threshold)(struct snd_pcm_substream *substream);
+ int data_type; /* data type 8,16,32 */
+ int sync_mode; /* DMA sync mode */
+ int packet_size; /* packet size only in PACKET mode */
};
extern struct snd_soc_platform omap_soc_platform;
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 3c36d24a6c2..993abb730df 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -388,7 +388,7 @@ static void fsi_soft_all_reset(struct fsi_master *master)
}
/* playback interrupt */
-static int fsi_data_push(struct fsi_priv *fsi)
+static int fsi_data_push(struct fsi_priv *fsi, int startup)
{
struct snd_pcm_runtime *runtime;
struct snd_pcm_substream *substream = NULL;
@@ -397,7 +397,7 @@ static int fsi_data_push(struct fsi_priv *fsi)
int fifo_free;
int width;
u8 *start;
- int i, ret, over_period;
+ int i, over_period;
if (!fsi ||
!fsi->substream ||
@@ -453,24 +453,26 @@ static int fsi_data_push(struct fsi_priv *fsi)
fsi->byte_offset += send * width;
- ret = 0;
status = fsi_reg_read(fsi, DOFF_ST);
- if (status & ERR_OVER) {
+ if (!startup) {
struct snd_soc_dai *dai = fsi_get_dai(substream);
- dev_err(dai->dev, "over run error\n");
- fsi_reg_write(fsi, DOFF_ST, status & ~ST_ERR);
- ret = -EIO;
+
+ if (status & ERR_OVER)
+ dev_err(dai->dev, "over run\n");
+ if (status & ERR_UNDER)
+ dev_err(dai->dev, "under run\n");
}
+ fsi_reg_write(fsi, DOFF_ST, 0);
fsi_irq_enable(fsi, 1);
if (over_period)
snd_pcm_period_elapsed(substream);
- return ret;
+ return 0;
}
-static int fsi_data_pop(struct fsi_priv *fsi)
+static int fsi_data_pop(struct fsi_priv *fsi, int startup)
{
struct snd_pcm_runtime *runtime;
struct snd_pcm_substream *substream = NULL;
@@ -479,7 +481,7 @@ static int fsi_data_pop(struct fsi_priv *fsi)
int fifo_fill;
int width;
u8 *start;
- int i, ret, over_period;
+ int i, over_period;
if (!fsi ||
!fsi->substream ||
@@ -534,21 +536,23 @@ static int fsi_data_pop(struct fsi_priv *fsi)
fsi->byte_offset += fifo_fill * width;
- ret = 0;
status = fsi_reg_read(fsi, DIFF_ST);
- if (status & ERR_UNDER) {
+ if (!startup) {
struct snd_soc_dai *dai = fsi_get_dai(substream);
- dev_err(dai->dev, "under run error\n");
- fsi_reg_write(fsi, DIFF_ST, status & ~ST_ERR);
- ret = -EIO;
+
+ if (status & ERR_OVER)
+ dev_err(dai->dev, "over run\n");
+ if (status & ERR_UNDER)
+ dev_err(dai->dev, "under run\n");
}
+ fsi_reg_write(fsi, DIFF_ST, 0);
fsi_irq_enable(fsi, 0);
if (over_period)
snd_pcm_period_elapsed(substream);
- return ret;
+ return 0;
}
static irqreturn_t fsi_interrupt(int irq, void *data)
@@ -562,13 +566,13 @@ static irqreturn_t fsi_interrupt(int irq, void *data)
fsi_master_write(master, SOFT_RST, status | 0x00000010);
if (int_st & INT_A_OUT)
- fsi_data_push(&master->fsia);
+ fsi_data_push(&master->fsia, 0);
if (int_st & INT_B_OUT)
- fsi_data_push(&master->fsib);
+ fsi_data_push(&master->fsib, 0);
if (int_st & INT_A_IN)
- fsi_data_pop(&master->fsia);
+ fsi_data_pop(&master->fsia, 0);
if (int_st & INT_B_IN)
- fsi_data_pop(&master->fsib);
+ fsi_data_pop(&master->fsib, 0);
fsi_master_write(master, INT_ST, 0x0000000);
@@ -726,7 +730,7 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
fsi_stream_push(fsi, substream,
frames_to_bytes(runtime, runtime->buffer_size),
frames_to_bytes(runtime, runtime->period_size));
- ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi);
+ ret = is_play ? fsi_data_push(fsi, 1) : fsi_data_pop(fsi, 1);
break;
case SNDRV_PCM_TRIGGER_STOP:
fsi_irq_disable(fsi, is_play);
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"
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4011ad3dc57..06c38d1502b 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -427,24 +427,24 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
if (!runtime->hw.rates) {
printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
codec_dai->name, cpu_dai->name);
- goto machine_err;
+ goto config_err;
}
if (!runtime->hw.formats) {
printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
codec_dai->name, cpu_dai->name);
- goto machine_err;
+ goto config_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;
+ goto config_err;
}
/* Symmetry only applies if we've already got an active stream. */
if (cpu_dai->active || codec_dai->active) {
ret = soc_pcm_apply_symmetry(substream);
if (ret != 0)
- goto machine_err;
+ goto config_err;
}
pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
@@ -467,10 +467,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
mutex_unlock(&pcm_mutex);
return 0;
-machine_err:
+config_err:
if (machine->ops && machine->ops->shutdown)
machine->ops->shutdown(substream);
+machine_err:
+ if (codec_dai->ops->shutdown)
+ codec_dai->ops->shutdown(substream, codec_dai);
+
codec_dai_err:
if (platform->pcm_ops->close)
platform->pcm_ops->close(substream);
@@ -1002,6 +1006,12 @@ static int soc_resume(struct device *dev)
struct snd_soc_card *card = socdev->card;
struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai;
+ /* If the initialization of this soc device failed, there is no codec
+ * associated with it. Just bail out in this case.
+ */
+ if (!card->codec)
+ return 0;
+
/* AC97 devices might have other drivers hanging off them so
* need to resume immediately. Other drivers don't have that
* problem and may take a substantial amount of time to resume
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-12-30 19:05:57 +0000