MERGE-via-pending-tracking-hist-MERGE-via-stable-tracking-MERGE-via-mokopatches-tracking-fix-stray-endmenu-patch-1232632040-1232632141

pending-tracking-hist top was MERGE-via-stable-tracking-MERGE-via-mokopatches-tracking-fix-stray-endmenu-patch-1232632040-1232632141 / fdf777a63bcb59e0dfd78bfe2c6242e01f6d4eb9 ... parent commitmessage:
From: merge <null@invalid>
MERGE-via-stable-tracking-hist-MERGE-via-mokopatches-tracking-fix-stray-endmenu-patch-1232632040

stable-tracking-hist top was MERGE-via-mokopatches-tracking-fix-stray-endmenu-patch-1232632040 / 90463bfd2d5a3c8b52f6e6d71024a00e052b0ced ... parent commitmessage:
From: merge <null@invalid>
MERGE-via-mokopatches-tracking-hist-fix-stray-endmenu-patch

mokopatches-tracking-hist top was fix-stray-endmenu-patch / 3630e0be570de8057e7f8d2fe501ed353cdf34e6 ... parent commitmessage:
From: Andy Green <andy@openmoko.com>
fix-stray-endmenu.patch

Signed-off-by: Andy Green <andy@openmoko.com>

6 files changed, 57 insertions, 26 deletions

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index bba9546ba5f..95c12b26fe3 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -20,7 +20,8 @@ config SND_SOC_MPC8610_HPCD
 
 config SND_SOC_MPC5200_I2S
 	tristate "Freescale MPC5200 PSC in I2S mode driver"
+	depends on PPC_MPC52xx && PPC_BESTCOMM
 	select SND_SOC_OF_SIMPLE
-	depends on SND_SOC && PPC_MPC52xx
+	select PPC_BESTCOMM_GEN_BD
 	help
 	  Say Y here to support the MPC5200 PSCs in I2S mode.
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index d2d3da9729f..64993eda567 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -284,7 +284,7 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id)
  * fsl_dma_new: initialize this PCM driver.
  *
  * This function is called when the codec driver calls snd_soc_new_pcms(),
- * once for each .dai_link in the machine driver's snd_soc_machine
+ * once for each .dai_link in the machine driver's snd_soc_card
  * structure.
  */
 static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
@@ -853,6 +853,18 @@ int fsl_dma_configure(struct fsl_dma_info *dma_info)
 }
 EXPORT_SYMBOL_GPL(fsl_dma_configure);
 
+static int __init fsl_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&fsl_soc_platform);
+}
+module_init(fsl_soc_platform_init);
+
+static void __exit fsl_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&fsl_soc_platform);
+}
+module_exit(fsl_soc_platform_exit);
+
 MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
 MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 157a7895ffa..c6d6eb71dc1 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -266,7 +266,8 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
  * If this is the first stream open, then grab the IRQ and program most of
  * the SSI registers.
  */
-static int fsl_ssi_startup(struct snd_pcm_substream *substream)
+static int fsl_ssi_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
@@ -411,7 +412,8 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream)
  * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the
  * clock master.
  */
-static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
+static int fsl_ssi_prepare(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -441,7 +443,8 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
  * The DMA channel is in external master start and pause mode, which
  * means the SSI completely controls the flow of data.
  */
-static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
+static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
@@ -490,7 +493,8 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
  *
  * Shutdown the SSI if there are no other substreams open.
  */
-static void fsl_ssi_shutdown(struct snd_pcm_substream *substream)
+static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
@@ -578,8 +582,6 @@ static struct snd_soc_dai fsl_ssi_dai_template = {
 		.prepare = fsl_ssi_prepare,
 		.shutdown = fsl_ssi_shutdown,
 		.trigger = fsl_ssi_trigger,
-	},
-	.dai_ops = {
 		.set_sysclk = fsl_ssi_set_sysclk,
 		.set_fmt = fsl_ssi_set_fmt,
 	},
@@ -671,6 +673,14 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
 	fsl_ssi_dai->private_data = ssi_private;
 	fsl_ssi_dai->name = ssi_private->name;
 	fsl_ssi_dai->id = ssi_info->id;
+	fsl_ssi_dai->dev = ssi_info->dev;
+
+	ret = snd_soc_register_dai(fsl_ssi_dai);
+	if (ret != 0) {
+		dev_err(ssi_info->dev, "failed to register DAI: %d\n", ret);
+		kfree(fsl_ssi_dai);
+		return NULL;
+	}
 
 	return fsl_ssi_dai;
 }
@@ -688,6 +698,8 @@ void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai)
 
 	device_remove_file(ssi_private->dev, &ssi_private->dev_attr);
 
+	snd_soc_unregister_dai(&ssi_private->cpu_dai);
+
 	kfree(ssi_private);
 }
 EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai);
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 94a02eaa482..9eb1ce185bd 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -187,7 +187,8 @@ static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
  * If this is the first stream open, then grab the IRQ and program most of
  * the PSC registers.
  */
-static int psc_i2s_startup(struct snd_pcm_substream *substream)
+static int psc_i2s_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -220,7 +221,8 @@ static int psc_i2s_startup(struct snd_pcm_substream *substream)
 }
 
 static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -256,7 +258,8 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
+static int psc_i2s_hw_free(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	snd_pcm_set_runtime_buffer(substream, NULL);
 	return 0;
@@ -268,7 +271,8 @@ static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
  * This function is called by ALSA to start, stop, pause, and resume the DMA
  * transfer of data.
  */
-static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -383,7 +387,8 @@ static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
  *
  * Shutdown the PSC if there are no other substreams open.
  */
-static void psc_i2s_shutdown(struct snd_pcm_substream *substream)
+static void psc_i2s_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -464,7 +469,6 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
  * psc_i2s_dai_template: template CPU Digital Audio Interface
  */
 static struct snd_soc_dai psc_i2s_dai_template = {
-	.type = SND_SOC_DAI_I2S,
 	.playback = {
 		.channels_min = 2,
 		.channels_max = 2,
@@ -483,8 +487,6 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 		.hw_free = psc_i2s_hw_free,
 		.shutdown = psc_i2s_shutdown,
 		.trigger = psc_i2s_trigger,
-	},
-	.dai_ops = {
 		.set_sysclk = psc_i2s_set_sysclk,
 		.set_fmt = psc_i2s_set_fmt,
 	},
@@ -826,6 +828,8 @@ static int __devinit psc_i2s_of_probe(struct of_device *op,
 	if (rc)
 		dev_info(psc_i2s->dev, "error creating sysfs files\n");
 
+	snd_soc_register_platform(&psc_i2s_pcm_soc_platform);
+
 	/* Tell the ASoC OF helpers about it */
 	of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
 				     &psc_i2s->dai);
@@ -839,6 +843,8 @@ static int __devexit psc_i2s_of_remove(struct of_device *op)
 
 	dev_dbg(&op->dev, "psc_i2s_remove()\n");
 
+	snd_soc_unregister_platform(&psc_i2s_pcm_soc_platform);
+
 	bcom_gen_bd_rx_release(psc_i2s->capture.bcom_task);
 	bcom_gen_bd_tx_release(psc_i2s->playback.bcom_task);
 
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 94f89debde1..bcec3f60bad 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -29,7 +29,7 @@
 struct mpc8610_hpcd_data {
 	struct snd_soc_device sound_devdata;
 	struct snd_soc_dai_link dai;
-	struct snd_soc_machine machine;
+	struct snd_soc_card machine;
 	unsigned int dai_format;
 	unsigned int codec_clk_direction;
 	unsigned int cpu_clk_direction;
@@ -185,7 +185,7 @@ static struct snd_soc_ops mpc8610_hpcd_ops = {
 /**
  * mpc8610_hpcd_machine: ASoC machine data
  */
-static struct snd_soc_machine mpc8610_hpcd_machine = {
+static struct snd_soc_card mpc8610_hpcd_machine = {
 	.probe = mpc8610_hpcd_machine_probe,
 	.remove = mpc8610_hpcd_machine_remove,
 	.name = "MPC8610 HPCD",
@@ -465,9 +465,9 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev,
 		goto error;
 	}
 
-	machine_data->sound_devdata.machine = &mpc8610_hpcd_machine;
+	machine_data->sound_devdata.card = &mpc8610_hpcd_machine;
 	machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270;
-	machine_data->sound_devdata.platform = &fsl_soc_platform;
+	machine_data->machine.platform = &fsl_soc_platform;
 
 	sound_device->dev.platform_data = machine_data;
 
diff --git a/sound/soc/fsl/soc-of-simple.c b/sound/soc/fsl/soc-of-simple.c
index 0382fdac51c..8bc5cd9e972 100644
--- a/sound/soc/fsl/soc-of-simple.c
+++ b/sound/soc/fsl/soc-of-simple.c
@@ -31,7 +31,7 @@ struct of_snd_soc_device {
 	int id;
 	struct list_head list;
 	struct snd_soc_device device;
-	struct snd_soc_machine machine;
+	struct snd_soc_card card;
 	struct snd_soc_dai_link dai_link;
 	struct platform_device *pdev;
 	struct device_node *platform_node;
@@ -58,9 +58,9 @@ of_snd_soc_get_device(struct device_node *codec_node)
 	/* Initialize the structure and add it to the global list */
 	of_soc->codec_node = codec_node;
 	of_soc->id = of_snd_soc_next_index++;
-	of_soc->machine.dai_link = &of_soc->dai_link;
-	of_soc->machine.num_links = 1;
-	of_soc->device.machine = &of_soc->machine;
+	of_soc->card.dai_link = &of_soc->dai_link;
+	of_soc->card.num_links = 1;
+	of_soc->device.card = &of_soc->card;
 	of_soc->dai_link.ops = &of_snd_soc_ops;
 	list_add(&of_soc->list, &of_snd_soc_device_list);
 
@@ -158,8 +158,8 @@ int of_snd_soc_register_platform(struct snd_soc_platform *platform,
 
 	of_soc->platform_node = node;
 	of_soc->dai_link.cpu_dai = cpu_dai;
-	of_soc->device.platform = platform;
-	of_soc->machine.name = of_soc->dai_link.cpu_dai->name;
+	of_soc->card.platform = platform;
+	of_soc->card.name = of_soc->dai_link.cpu_dai->name;
 
 	/* Now try to register the SoC device */
 	of_snd_soc_register_device(of_soc);