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-rw-r--r--sound/arm/pxa2xx-ac97.c8
-rw-r--r--sound/arm/pxa2xx-pcm.c4
-rw-r--r--sound/arm/sa11xx-uda1341.c4
-rw-r--r--sound/core/seq/oss/seq_oss_synth.c3
-rw-r--r--sound/oss/vidc.c2
-rw-r--r--sound/oss/vidc_fill.S2
-rw-r--r--sound/oss/waveartist.c2
-rw-r--r--sound/pci/hda/hda_intel.c4
-rw-r--r--sound/sh/aica.c2
-rw-r--r--sound/soc/at32/playpaq_wm8510.c4
-rw-r--r--sound/soc/at91/at91-pcm.c4
-rw-r--r--sound/soc/at91/at91-pcm.h2
-rw-r--r--sound/soc/at91/at91-ssc.c6
-rw-r--r--sound/soc/at91/eti_b1_wm8731.c4
-rw-r--r--sound/soc/codecs/wm8750.c3
-rw-r--r--sound/soc/davinci/davinci-evm.c3
-rw-r--r--sound/soc/fsl/fsl_dma.c242
-rw-r--r--sound/soc/fsl/fsl_ssi.c74
-rw-r--r--sound/soc/omap/n810.c4
-rw-r--r--sound/soc/omap/omap-mcbsp.c6
-rw-r--r--sound/soc/omap/omap-pcm.c2
-rw-r--r--sound/soc/pxa/corgi.c8
-rw-r--r--sound/soc/pxa/e800_wm9712.c6
-rw-r--r--sound/soc/pxa/em-x270.c6
-rw-r--r--sound/soc/pxa/poodle.c16
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c8
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c8
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.c6
-rw-r--r--sound/soc/pxa/spitz.c10
-rw-r--r--sound/soc/pxa/tosa.c9
-rw-r--r--sound/soc/s3c24xx/neo1973_wm8753.c11
-rw-r--r--sound/soc/s3c24xx/s3c2412-i2s.c8
-rw-r--r--sound/soc/s3c24xx/s3c2443-ac97.c10
-rw-r--r--sound/soc/s3c24xx/s3c24xx-i2s.c10
-rw-r--r--sound/soc/s3c24xx/s3c24xx-pcm.c6
-rw-r--r--sound/soc/soc-dapm.c1
36 files changed, 291 insertions, 217 deletions
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 5b3274b465e..199cca3366d 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -26,10 +26,10 @@
#include <asm/irq.h>
#include <linux/mutex.h>
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/pxa2xx-gpio.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/pxa2xx-gpio.h>
+#include <mach/audio.h>
#include "pxa2xx-pcm.h"
diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c
index 0ede9e4656a..381094aab23 100644
--- a/sound/arm/pxa2xx-pcm.c
+++ b/sound/arm/pxa2xx-pcm.c
@@ -21,8 +21,8 @@
#include <sound/pcm_params.h>
#include <asm/dma.h>
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
#include "pxa2xx-pcm.h"
diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c
index faeddf3eced..b9c51bf8cd7 100644
--- a/sound/arm/sa11xx-uda1341.c
+++ b/sound/arm/sa11xx-uda1341.c
@@ -71,8 +71,8 @@
#include <linux/pm.h>
#endif
-#include <asm/hardware.h>
-#include <asm/arch/h3600.h>
+#include <mach/hardware.h>
+#include <mach/h3600.h>
#include <asm/mach-types.h>
#include <asm/dma.h>
diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c
index 558dadbf45f..e024e4588b8 100644
--- a/sound/core/seq/oss/seq_oss_synth.c
+++ b/sound/core/seq/oss/seq_oss_synth.c
@@ -604,6 +604,9 @@ snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_in
{
struct seq_oss_synth *rec;
+ if (dev < 0 || dev >= dp->max_synthdev)
+ return -ENXIO;
+
if (dp->synths[dev].is_midi) {
struct midi_info minf;
snd_seq_oss_midi_make_info(dp, dp->synths[dev].midi_mapped, &minf);
diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c
index bb4a0969f46..725fef0f59a 100644
--- a/sound/oss/vidc.c
+++ b/sound/oss/vidc.c
@@ -22,7 +22,7 @@
#include <linux/kernel.h>
#include <linux/interrupt.h>
-#include <asm/hardware.h>
+#include <mach/hardware.h>
#include <asm/dma.h>
#include <asm/io.h>
#include <asm/hardware/iomd.h>
diff --git a/sound/oss/vidc_fill.S b/sound/oss/vidc_fill.S
index 01ccc074cc1..bed34921d17 100644
--- a/sound/oss/vidc_fill.S
+++ b/sound/oss/vidc_fill.S
@@ -11,7 +11,7 @@
*/
#include <linux/linkage.h>
#include <asm/assembler.h>
-#include <asm/hardware.h>
+#include <mach/hardware.h>
#include <asm/hardware/iomd.h>
.text
diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c
index 88490418f93..c47842fad65 100644
--- a/sound/oss/waveartist.c
+++ b/sound/oss/waveartist.c
@@ -47,7 +47,7 @@
#include "waveartist.h"
#ifdef CONFIG_ARM
-#include <asm/hardware.h>
+#include <mach/hardware.h>
#include <asm/mach-types.h>
#endif
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ef9f072b47f..a73d6ca0a90 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -101,6 +101,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
"{Intel, ICH8},"
"{Intel, ICH9},"
"{Intel, ICH10},"
+ "{Intel, PCH},"
"{Intel, SCH},"
"{ATI, SB450},"
"{ATI, SB600},"
@@ -2263,6 +2264,8 @@ static struct pci_device_id azx_ids[] = {
{ PCI_DEVICE(0x8086, 0x293f), .driver_data = AZX_DRIVER_ICH },
{ PCI_DEVICE(0x8086, 0x3a3e), .driver_data = AZX_DRIVER_ICH },
{ PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH },
+ /* PCH */
+ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH },
/* SCH */
{ PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH },
/* ATI SB 450/600 */
@@ -2272,6 +2275,7 @@ static struct pci_device_id azx_ids[] = {
{ PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI },
{ PCI_DEVICE(0x1002, 0x7919), .driver_data = AZX_DRIVER_ATIHDMI },
{ PCI_DEVICE(0x1002, 0x960f), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0x970f), .driver_data = AZX_DRIVER_ATIHDMI },
{ PCI_DEVICE(0x1002, 0xaa00), .driver_data = AZX_DRIVER_ATIHDMI },
{ PCI_DEVICE(0x1002, 0xaa08), .driver_data = AZX_DRIVER_ATIHDMI },
{ PCI_DEVICE(0x1002, 0xaa10), .driver_data = AZX_DRIVER_ATIHDMI },
diff --git a/sound/sh/aica.c b/sound/sh/aica.c
index 9ca11332614..54df8baf916 100644
--- a/sound/sh/aica.c
+++ b/sound/sh/aica.c
@@ -42,7 +42,7 @@
#include <sound/info.h>
#include <asm/io.h>
#include <asm/dma.h>
-#include <asm/dreamcast/sysasic.h>
+#include <mach/sysasic.h>
#include "aica.h"
MODULE_AUTHOR("Adrian McMenamin <adrian@mcmen.demon.co.uk>");
diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c
index fee5f8e5895..3f326219f1e 100644
--- a/sound/soc/at32/playpaq_wm8510.c
+++ b/sound/soc/at32/playpaq_wm8510.c
@@ -36,8 +36,8 @@
#include <sound/soc.h>
#include <sound/soc-dapm.h>
-#include <asm/arch/at32ap700x.h>
-#include <asm/arch/portmux.h>
+#include <mach/at32ap700x.h>
+#include <mach/portmux.h>
#include "../codecs/wm8510.h"
#include "at32-pcm.h"
diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c
index d47492b2b6e..7ab48bd25e4 100644
--- a/sound/soc/at91/at91-pcm.c
+++ b/sound/soc/at91/at91-pcm.c
@@ -28,8 +28,8 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/at91_ssc.h>
+#include <mach/hardware.h>
+#include <mach/at91_ssc.h>
#include "at91-pcm.h"
diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h
index 58d0f00a07b..e5aada2cb10 100644
--- a/sound/soc/at91/at91-pcm.h
+++ b/sound/soc/at91/at91-pcm.h
@@ -19,7 +19,7 @@
#ifndef _AT91_PCM_H
#define _AT91_PCM_H
-#include <asm/arch/hardware.h>
+#include <mach/hardware.h>
struct at91_ssc_periph {
void __iomem *base;
diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c
index 090e607f869..5d44515e62e 100644
--- a/sound/soc/at91/at91-ssc.c
+++ b/sound/soc/at91/at91-ssc.c
@@ -28,9 +28,9 @@
#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 <mach/hardware.h>
+#include <mach/at91_pmc.h>
+#include <mach/at91_ssc.h>
#include "at91-pcm.h"
#include "at91-ssc.h"
diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c
index d532de95424..b081e83766b 100644
--- a/sound/soc/at91/eti_b1_wm8731.c
+++ b/sound/soc/at91/eti_b1_wm8731.c
@@ -33,8 +33,8 @@
#include <sound/soc.h>
#include <sound/soc-dapm.h>
-#include <asm/hardware.h>
-#include <asm/arch/gpio.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
#include "../codecs/wm8731.h"
#include "at91-pcm.h"
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index e23cb09f0d1..c6a8edf302a 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -348,8 +348,9 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("ROUT1"),
SND_SOC_DAPM_OUTPUT("LOUT2"),
SND_SOC_DAPM_OUTPUT("ROUT2"),
- SND_SOC_DAPM_OUTPUT("MONO"),
+ SND_SOC_DAPM_OUTPUT("MONO1"),
SND_SOC_DAPM_OUTPUT("OUT3"),
+ SND_SOC_DAPM_OUTPUT("VREF"),
SND_SOC_DAPM_INPUT("LINPUT1"),
SND_SOC_DAPM_INPUT("LINPUT2"),
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 5e2c306399e..65fdbd81a37 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -19,9 +19,8 @@
#include <sound/soc.h>
#include <sound/soc-dapm.h>
-#include <asm/mach-types.h>
#include <asm/dma.h>
-#include <asm/arch/hardware.h>
+#include <mach/hardware.h>
#include "../codecs/tlv320aic3x.h"
#include "davinci-pcm.h"
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index da2bc590286..d2d3da9729f 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -132,12 +132,17 @@ struct fsl_dma_private {
* Since each link descriptor has a 32-bit byte count field, we set
* period_bytes_max to the largest 32-bit number. We also have no maximum
* number of periods.
+ *
+ * Note that we specify SNDRV_PCM_INFO_JOINT_DUPLEX here, but only because a
+ * limitation in the SSI driver requires the sample rates for playback and
+ * capture to be the same.
*/
static const struct snd_pcm_hardware fsl_dma_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_MMAP_VALID,
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_JOINT_DUPLEX,
.formats = FSLDMA_PCM_FORMATS,
.rates = FSLDMA_PCM_RATES,
.rate_min = 5512,
@@ -322,14 +327,75 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
* fsl_dma_open: open a new substream.
*
* Each substream has its own DMA buffer.
+ *
+ * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link
+ * descriptors that ping-pong from one period to the next. For example, if
+ * there are six periods and two link descriptors, this is how they look
+ * before playback starts:
+ *
+ * The last link descriptor
+ * ____________ points back to the first
+ * | |
+ * V |
+ * ___ ___ |
+ * | |->| |->|
+ * |___| |___|
+ * | |
+ * | |
+ * V V
+ * _________________________________________
+ * | | | | | | | The DMA buffer is
+ * | | | | | | | divided into 6 parts
+ * |______|______|______|______|______|______|
+ *
+ * and here's how they look after the first period is finished playing:
+ *
+ * ____________
+ * | |
+ * V |
+ * ___ ___ |
+ * | |->| |->|
+ * |___| |___|
+ * | |
+ * |______________
+ * | |
+ * V V
+ * _________________________________________
+ * | | | | | | |
+ * | | | | | | |
+ * |______|______|______|______|______|______|
+ *
+ * The first link descriptor now points to the third period. The DMA
+ * controller is currently playing the second period. When it finishes, it
+ * will jump back to the first descriptor and play the third period.
+ *
+ * There are four reasons we do this:
+ *
+ * 1. The only way to get the DMA controller to automatically restart the
+ * transfer when it gets to the end of the buffer is to use chaining
+ * mode. Basic direct mode doesn't offer that feature.
+ * 2. We need to receive an interrupt at the end of every period. The DMA
+ * controller can generate an interrupt at the end of every link transfer
+ * (aka segment). Making each period into a DMA segment will give us the
+ * interrupts we need.
+ * 3. By creating only two link descriptors, regardless of the number of
+ * periods, we do not need to reallocate the link descriptors if the
+ * number of periods changes.
+ * 4. All of the audio data is still stored in a single, contiguous DMA
+ * buffer, which is what ALSA expects. We're just dividing it into
+ * contiguous parts, and creating a link descriptor for each one.
*/
static int fsl_dma_open(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct fsl_dma_private *dma_private;
+ struct ccsr_dma_channel __iomem *dma_channel;
dma_addr_t ld_buf_phys;
+ u64 temp_link; /* Pointer to next link descriptor */
+ u32 mr;
unsigned int channel;
int ret = 0;
+ unsigned int i;
/*
* Reject any DMA buffer whose size is not a multiple of the period
@@ -390,68 +456,74 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware);
runtime->private_data = dma_private;
+ /* Program the fixed DMA controller parameters */
+
+ dma_channel = dma_private->dma_channel;
+
+ temp_link = dma_private->ld_buf_phys +
+ sizeof(struct fsl_dma_link_descriptor);
+
+ for (i = 0; i < NUM_DMA_LINKS; i++) {
+ struct fsl_dma_link_descriptor *link = &dma_private->link[i];
+
+ link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
+ link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
+ link->next = cpu_to_be64(temp_link);
+
+ temp_link += sizeof(struct fsl_dma_link_descriptor);
+ }
+ /* The last link descriptor points to the first */
+ dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys);
+
+ /* Tell the DMA controller where the first link descriptor is */
+ out_be32(&dma_channel->clndar,
+ CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys));
+ out_be32(&dma_channel->eclndar,
+ CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys));
+
+ /* The manual says the BCR must be clear before enabling EMP */
+ out_be32(&dma_channel->bcr, 0);
+
+ /*
+ * Program the mode register for interrupts, external master control,
+ * and source/destination hold. Also clear the Channel Abort bit.
+ */
+ mr = in_be32(&dma_channel->mr) &
+ ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE);
+
+ /*
+ * We want External Master Start and External Master Pause enabled,
+ * because the SSI is controlling the DMA controller. We want the DMA
+ * controller to be set up in advance, and then we signal only the SSI
+ * to start transferring.
+ *
+ * We want End-Of-Segment Interrupts enabled, because this will generate
+ * an interrupt at the end of each segment (each link descriptor
+ * represents one segment). Each DMA segment is the same thing as an
+ * ALSA period, so this is how we get an interrupt at the end of every
+ * period.
+ *
+ * We want Error Interrupt enabled, so that we can get an error if
+ * the DMA controller is mis-programmed somehow.
+ */
+ mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN |
+ CCSR_DMA_MR_EMS_EN;
+
+ /* For playback, we want the destination address to be held. For
+ capture, set the source address to be held. */
+ mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE;
+
+ out_be32(&dma_channel->mr, mr);
+
return 0;
}
/**
- * fsl_dma_hw_params: allocate the DMA buffer and the DMA link descriptors.
+ * fsl_dma_hw_params: continue initializing the DMA links
*
- * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link
- * descriptors that ping-pong from one period to the next. For example, if
- * there are six periods and two link descriptors, this is how they look
- * before playback starts:
- *
- * The last link descriptor
- * ____________ points back to the first
- * | |
- * V |
- * ___ ___ |
- * | |->| |->|
- * |___| |___|
- * | |
- * | |
- * V V
- * _________________________________________
- * | | | | | | | The DMA buffer is
- * | | | | | | | divided into 6 parts
- * |______|______|______|______|______|______|
- *
- * and here's how they look after the first period is finished playing:
- *
- * ____________
- * | |
- * V |
- * ___ ___ |
- * | |->| |->|
- * |___| |___|
- * | |
- * |______________
- * | |
- * V V
- * _________________________________________
- * | | | | | | |
- * | | | | | | |
- * |______|______|______|______|______|______|
- *
- * The first link descriptor now points to the third period. The DMA
- * controller is currently playing the second period. When it finishes, it
- * will jump back to the first descriptor and play the third period.
- *
- * There are four reasons we do this:
- *
- * 1. The only way to get the DMA controller to automatically restart the
- * transfer when it gets to the end of the buffer is to use chaining
- * mode. Basic direct mode doesn't offer that feature.
- * 2. We need to receive an interrupt at the end of every period. The DMA
- * controller can generate an interrupt at the end of every link transfer
- * (aka segment). Making each period into a DMA segment will give us the
- * interrupts we need.
- * 3. By creating only two link descriptors, regardless of the number of
- * periods, we do not need to reallocate the link descriptors if the
- * number of periods changes.
- * 4. All of the audio data is still stored in a single, contiguous DMA
- * buffer, which is what ALSA expects. We're just dividing it into
- * contiguous parts, and creating a link descriptor for each one.
+ * This function obtains hardware parameters about the opened stream and
+ * programs the DMA controller accordingly.
*
* Note that due to a quirk of the SSI's STX register, the target address
* for the DMA operations depends on the sample size. So we don't program
@@ -463,11 +535,8 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct fsl_dma_private *dma_private = runtime->private_data;
- struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
dma_addr_t temp_addr; /* Pointer to next period */
- u64 temp_link; /* Pointer to next link descriptor */
- u32 mr; /* Temporary variable for MR register */
unsigned int i;
@@ -485,8 +554,6 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
dma_private->dma_buf_next = dma_private->dma_buf_phys;
/*
- * Initialize each link descriptor.
- *
* The actual address in STX0 (destination for playback, source for
* capture) is based on the sample size, but we don't know the sample
* size in this function, so we'll have to adjust that later. See
@@ -502,16 +569,11 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
* buffer itself.
*/
temp_addr = substream->dma_buffer.addr;
- temp_link = dma_private->ld_buf_phys +
- sizeof(struct fsl_dma_link_descriptor);
for (i = 0; i < NUM_DMA_LINKS; i++) {
struct fsl_dma_link_descriptor *link = &dma_private->link[i];
link->count = cpu_to_be32(period_size);
- link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
- link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
- link->next = cpu_to_be64(temp_link);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
link->source_addr = cpu_to_be32(temp_addr);
@@ -519,51 +581,7 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
link->dest_addr = cpu_to_be32(temp_addr);
temp_addr += period_size;
- temp_link += sizeof(struct fsl_dma_link_descriptor);
}
- /* The last link descriptor points to the first */
- dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys);
-
- /* Tell the DMA controller where the first link descriptor is */
- out_be32(&dma_channel->clndar,
- CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys));
- out_be32(&dma_channel->eclndar,
- CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys));
-
- /* The manual says the BCR must be clear before enabling EMP */
- out_be32(&dma_channel->bcr, 0);
-
- /*
- * Program the mode register for interrupts, external master control,
- * and source/destination hold. Also clear the Channel Abort bit.
- */
- mr = in_be32(&dma_channel->mr) &
- ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE);
-
- /*
- * We want External Master Start and External Master Pause enabled,
- * because the SSI is controlling the DMA controller. We want the DMA
- * controller to be set up in advance, and then we signal only the SSI
- * to start transfering.
- *
- * We want End-Of-Segment Interrupts enabled, because this will generate
- * an interrupt at the end of each segment (each link descriptor
- * represents one segment). Each DMA segment is the same thing as an
- * ALSA period, so this is how we get an interrupt at the end of every
- * period.
- *
- * We want Error Interrupt enabled, so that we can get an error if
- * the DMA controller is mis-programmed somehow.
- */
- mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN |
- CCSR_DMA_MR_EMS_EN;
-
- /* For playback, we want the destination address to be held. For
- capture, set the source address to be held. */
- mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
- CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE;
-
- out_be32(&dma_channel->mr, mr);
return 0;
}
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 71bff33f552..157a7895ffa 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -67,6 +67,8 @@
* @ssi: pointer to the SSI's registers
* @ssi_phys: physical address of the SSI registers
* @irq: IRQ of this SSI
+ * @first_stream: pointer to the stream that was opened first
+ * @second_stream: pointer to second stream
* @dev: struct device pointer
* @playback: the number of playback streams opened
* @capture: the number of capture streams opened
@@ -79,6 +81,8 @@ struct fsl_ssi_private {
struct ccsr_ssi __iomem *ssi;
dma_addr_t ssi_phys;
unsigned int irq;
+ struct snd_pcm_substream *first_stream;
+ struct snd_pcm_substream *second_stream;
struct device *dev;
unsigned int playback;
unsigned int capture;
@@ -342,6 +346,49 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream)
*/
}
+ if (!ssi_private->first_stream)
+ ssi_private->first_stream = substream;
+ else {
+ /* This is the second stream open, so we need to impose sample
+ * rate and maybe sample size constraints. Note that this can
+ * cause a race condition if the second stream is opened before
+ * the first stream is fully initialized.
+ *
+ * We provide some protection by checking to make sure the first
+ * stream is initialized, but it's not perfect. ALSA sometimes
+ * re-initializes the driver with a different sample rate or
+ * size. If the second stream is opened before the first stream
+ * has received its final parameters, then the second stream may
+ * be constrained to the wrong sample rate or size.
+ *
+ * FIXME: This code does not handle opening and closing streams
+ * repeatedly. If you open two streams and then close the first
+ * one, you may not be able to open another stream until you
+ * close the second one as well.
+ */
+ struct snd_pcm_runtime *first_runtime =
+ ssi_private->first_stream->runtime;
+
+ if (!first_runtime->rate || !first_runtime->sample_bits) {
+ dev_err(substream->pcm->card->dev,
+ "set sample rate and size in %s stream first\n",
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK
+ ? "capture" : "playback");
+ return -EAGAIN;
+ }
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ first_runtime->rate, first_runtime->rate);
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ first_runtime->sample_bits,
+ first_runtime->sample_bits);
+
+ ssi_private->second_stream = substream;
+ }
+
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
ssi_private->playback++;
@@ -371,18 +418,16 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
- u32 wl;
- wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format));
+ if (substream == ssi_private->first_stream) {
+ u32 wl;
- clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+ /* The SSI should always be disabled at this points (SSIEN=0) */
+ wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format));
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ /* In synchronous mode, the SSI uses STCCR for capture */
clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
- else
- clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl);
-
- setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+ }
return 0;
}
@@ -407,9 +452,13 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- setbits32(&ssi->scr, CCSR_SSI_SCR_TE);
+ clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+ setbits32(&ssi->scr,
+ CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
} else {
- setbits32(&ssi->scr, CCSR_SSI_SCR_RE);
+ clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+ setbits32(&ssi->scr,
+ CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
/*
* I think we need this delay to allow time for the SSI
@@ -452,6 +501,11 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream)
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
ssi_private->capture--;
+ if (ssi_private->first_stream == substream)
+ ssi_private->first_stream = ssi_private->second_stream;
+
+ ssi_private->second_stream = NULL;
+
/*
* If this is the last active substream, disable the SSI and release
* the IRQ.
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 02cec96859b..7694621ec40 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -29,9 +29,9 @@
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
-#include <asm/arch/hardware.h>
+#include <mach/hardware.h>
#include <linux/gpio.h>
-#include <asm/arch/mcbsp.h>
+#include <mach/mcbsp.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 00b0c9d73cd..35310e16d7f 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -30,9 +30,9 @@
#include <sound/initval.h>
#include <sound/soc.h>
-#include <asm/arch/control.h>
-#include <asm/arch/dma.h>
-#include <asm/arch/mcbsp.h>
+#include <mach/control.h>
+#include <mach/dma.h>
+#include <mach/mcbsp.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index e092f3d836d..690bfeaec4a 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -27,7 +27,7 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
-#include <asm/arch/dma.h>
+#include <mach/dma.h>
#include "omap-pcm.h"
static const struct snd_pcm_hardware omap_pcm_hardware = {
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index c0294464a23..0a53f72077f 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -25,10 +25,10 @@
#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 <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/corgi.h>
+#include <mach/audio.h>
#include "../codecs/wm8731.h"
#include "pxa2xx-pcm.h"
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 06e8afb2527..6781c5be242 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -21,9 +21,9 @@
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/audio.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
#include "../codecs/wm9712.h"
#include "pxa2xx-pcm.h"
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index 02dcac39cdf..d9c3f7b28be 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -30,9 +30,9 @@
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/audio.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
#include "../codecs/wm9712.h"
#include "pxa2xx-pcm.h"
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 65a4e9a8c39..a4697f7e292 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -26,10 +26,10 @@
#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 <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/poodle.h>
+#include <mach/audio.h>
#include "../codecs/wm8731.h"
#include "pxa2xx-pcm.h"
@@ -85,17 +85,13 @@ static int poodle_startup(struct snd_pcm_substream *substream)
}
/* we need to unmute the HP at shutdown as the mute burns power on poodle */
-static int poodle_shutdown(struct snd_pcm_substream *substream)
+static void 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,
@@ -232,7 +228,7 @@ static const struct soc_enum poodle_enum[] = {
SOC_ENUM_SINGLE_EXT(2, spk_function),
};
-static const snd_kcontrol_new_t wm8731_poodle_controls[] = {
+static const struct snd_kcontrol_new 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,
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 059af815ea0..d94a495bd6b 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -26,10 +26,10 @@
#include <asm/irq.h>
#include <linux/mutex.h>
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/pxa2xx-gpio.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/pxa2xx-gpio.h>
+#include <mach/audio.h>
#include "pxa2xx-pcm.h"
#include "pxa2xx-ac97.h"
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 8f96d87f7b4..8548818eea0 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -21,10 +21,10 @@
#include <sound/initval.h>
#include <sound/soc.h>
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/pxa2xx-gpio.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/pxa2xx-gpio.h>
+#include <mach/audio.h>
#include "pxa2xx-pcm.h"
#include "pxa2xx-i2s.h"
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 2df03ee5819..4345f387fe4 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -22,9 +22,9 @@
#include <sound/soc.h>
#include <asm/dma.h>
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/audio.h>
#include "pxa2xx-pcm.h"
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index 64385797da5..37cb768fc93 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -26,10 +26,10 @@
#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 <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/akita.h>
+#include <mach/spitz.h>
#include "../codecs/wm8750.h"
#include "pxa2xx-pcm.h"
#include "pxa2xx-i2s.h"
@@ -297,7 +297,7 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
snd_soc_dapm_disable_pin(codec, "LINPUT3");
snd_soc_dapm_disable_pin(codec, "RINPUT3");
snd_soc_dapm_disable_pin(codec, "OUT3");
- snd_soc_dapm_disable_pin(codec, "MONO");
+ snd_soc_dapm_disable_pin(codec, "MONO1");
/* Add spitz specific controls */
for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index fe6cca9c9e7..2baaa750f12 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -29,11 +29,10 @@
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
-#include <asm/arch/tosa.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/audio.h>
-#include <asm/arch/tosa.h>
+#include <mach/tosa.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
#include "../codecs/wm9712.h"
#include "pxa2xx-pcm.h"
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 4d7a9aa15f1..8089f8ee05c 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -24,14 +24,13 @@
#include <sound/soc-dapm.h>
#include <sound/tlv.h>
-#include <asm/mach-types.h>
#include <asm/hardware/scoop.h>
-#include <asm/arch/regs-clock.h>
-#include <asm/arch/regs-gpio.h>
-#include <asm/hardware.h>
-#include <asm/arch/audio.h>
+#include <mach/regs-clock.h>
+#include <mach/regs-gpio.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
#include <linux/io.h>
-#include <asm/arch/spi-gpio.h>
+#include <mach/spi-gpio.h>
#include <asm/plat-s3c24xx/regs-iis.h>
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index ee4676ed128..ded7d995a92 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -28,16 +28,16 @@
#include <sound/pcm_params.h>
#include <sound/initval.h>
#include <sound/soc.h>
-#include <asm/hardware.h>
+#include <mach/hardware.h>
#include <linux/io.h>
#include <asm/dma.h>
#include <asm/plat-s3c24xx/regs-s3c2412-iis.h>
-#include <asm/arch/regs-gpio.h>
-#include <asm/arch/audio.h>
-#include <asm/arch/dma.h>
+#include <mach/regs-gpio.h>
+#include <mach/audio.h>
+#include <mach/dma.h>
#include "s3c24xx-pcm.h"
#include "s3c2412-i2s.h"
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 783349b7fed..19c5c3cf5d8 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -27,13 +27,13 @@
#include <sound/initval.h>
#include <sound/soc.h>
-#include <asm/hardware.h>
+#include <mach/hardware.h>
#include <asm/plat-s3c/regs-ac97.h>
-#include <asm/arch/regs-gpio.h>
-#include <asm/arch/regs-clock.h>
-#include <asm/arch/audio.h>
+#include <mach/regs-gpio.h>
+#include <mach/regs-clock.h>
+#include <mach/audio.h>
#include <asm/dma.h>
-#include <asm/arch/dma.h>
+#include <mach/dma.h>
#include "s3c24xx-pcm.h"
#include "s3c24xx-ac97.h"
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 397524282b5..ba4476b55fb 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -27,12 +27,12 @@
#include <sound/initval.h>
#include <sound/soc.h>
-#include <asm/hardware.h>
-#include <asm/arch/regs-gpio.h>
-#include <asm/arch/regs-clock.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/regs-gpio.h>
+#include <mach/regs-clock.h>
+#include <mach/audio.h>
#include <asm/dma.h>
-#include <asm/arch/dma.h>
+#include <mach/dma.h>
#include <asm/plat-s3c24xx/regs-iis.h>
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index cef79b34dc6..e13e614bada 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -27,9 +27,9 @@
#include <sound/soc.h>
#include <asm/dma.h>
-#include <asm/hardware.h>
-#include <asm/arch/dma.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/dma.h>
+#include <mach/audio.h>
#include "s3c24xx-pcm.h"
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 820347c9ae4..f9d100bc847 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -470,6 +470,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w,
return 0;
}
+EXPORT_SYMBOL_GPL(dapm_reg_event);
/*
* Scan each dapm widget for complete audio path.