[RFC PATCH] ASoC: au1x: platform driver restructuring

[RFC PATCH] ASoC: au1x: platform driver restructuring

[Manuel Lauss]

Hi Mark!

Please have a look at the patch below and verify it does what you
suggested earlier, and if you have no objections, add it to your
patchqueue.

Both the AC97 and I2S modules are converted to full platform drivers;
the PCM (DMA) interface is initialized during platform probe.
Run-tested on DB1200 board.

Thank you!
	Manuel Lauss

---

From: Manuel Lauss <manuel.lauss@gmail.com>
Subject: [PATCH] ASoC: au1x: platform driver restructuring

Make the AC97/I2S modules full platform drivers and update the
sample machine code.

Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
---
 sound/soc/au1x/dbdma2.c      |   24 +++++-----
 sound/soc/au1x/psc-ac97.c    |  103 +++++++++++++++++++++++++-----------------
 sound/soc/au1x/psc-i2s.c     |  103 ++++++++++++++++++++++++------------------
 sound/soc/au1x/psc.h         |    7 ++-
 sound/soc/au1x/sample-ac97.c |   48 +++++++++++--------
 5 files changed, 165 insertions(+), 120 deletions(-)

diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 594c6c5..10d21ec 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -1,8 +1,8 @@
 /*
  * Au12x0/Au1550 PSC ALSA ASoC audio support.
  *
- * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
- *	Manuel Lauss <mano@roarinelk.homelinux.net>
+ * (c) 2007-2009 MSC Vertriebsges.m.b.H.,
+ *	Manuel Lauss <manuel.lauss@gmail.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
@@ -331,7 +331,7 @@ static int au1xpsc_pcm_new(struct snd_card *card,
 	return 0;
 }

-static int au1xpsc_pcm_probe(struct platform_device *pdev)
+int au1xpsc_asoc_dbdma_init(struct platform_device *pdev)
 {
 	struct resource *r;
 	int ret;
@@ -375,8 +375,9 @@ out1:
 	au1xpsc_audio_pcmdma[PCM_TX] = NULL;
 	return ret;
 }
+EXPORT_SYMBOL_GPL(au1xpsc_asoc_dbdma_init);

-static int au1xpsc_pcm_remove(struct platform_device *pdev)
+void au1xpsc_asoc_dbdma_exit(void)
 {
 	int i;

@@ -387,36 +388,33 @@ static int au1xpsc_pcm_remove(struct platform_device *pdev)
 			au1xpsc_audio_pcmdma[i] = NULL;
 		}
 	}
-
-	return 0;
 }
+EXPORT_SYMBOL_GPL(au1xpsc_asoc_dbdma_exit);

 /* au1xpsc audio platform */
 struct snd_soc_platform au1xpsc_soc_platform = {
 	.name		= "au1xpsc-pcm-dbdma",
-	.probe		= au1xpsc_pcm_probe,
-	.remove		= au1xpsc_pcm_remove,
 	.pcm_ops 	= &au1xpsc_pcm_ops,
 	.pcm_new	= au1xpsc_pcm_new,
 	.pcm_free	= au1xpsc_pcm_free_dma_buffers,
 };
 EXPORT_SYMBOL_GPL(au1xpsc_soc_platform);

-static int __init au1xpsc_audio_dbdma_init(void)
+static int __init au1xpsc_audio_dbdma_load(void)
 {
 	au1xpsc_audio_pcmdma[PCM_TX] = NULL;
 	au1xpsc_audio_pcmdma[PCM_RX] = NULL;
 	return snd_soc_register_platform(&au1xpsc_soc_platform);
 }

-static void __exit au1xpsc_audio_dbdma_exit(void)
+static void __exit au1xpsc_audio_dbdma_unload(void)
 {
 	snd_soc_unregister_platform(&au1xpsc_soc_platform);
 }

-module_init(au1xpsc_audio_dbdma_init);
-module_exit(au1xpsc_audio_dbdma_exit);
+module_init(au1xpsc_audio_dbdma_load);
+module_exit(au1xpsc_audio_dbdma_unload);

 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver");
-MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index 479d7bd..ee8e642 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -1,8 +1,8 @@
 /*
  * Au12x0/Au1550 PSC ALSA ASoC audio support.
  *
- * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
- *	Manuel Lauss <mano@roarinelk.homelinux.net>
+ * (c) 2007-2009 MSC Vertriebsges.m.b.H.,
+ *	Manuel Lauss <manuel.lauss@gmail.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
@@ -236,8 +236,32 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }

-static int au1xpsc_ac97_probe(struct platform_device *pdev,
-			      struct snd_soc_dai *dai)
+static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
+	.trigger	= au1xpsc_ac97_trigger,
+	.hw_params	= au1xpsc_ac97_hw_params,
+};
+
+struct snd_soc_dai au1xpsc_ac97_dai = {
+	.name			= "au1xpsc_ac97",
+	.ac97_control		= 1,
+	.playback = {
+		.rates		= AC97_RATES,
+		.formats	= AC97_FMTS,
+		.channels_min	= 2,
+		.channels_max	= 2,
+	},
+	.capture = {
+		.rates		= AC97_RATES,
+		.formats	= AC97_FMTS,
+		.channels_min	= 2,
+		.channels_max	= 2,
+	},
+	.ops = &au1xpsc_ac97_dai_ops,
+};
+EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai);
+
+
+static int __devinit au1xpsc_ac97_probe(struct platform_device *pdev)
 {
 	int ret;
 	struct resource *r;
@@ -287,8 +311,18 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev,
 	 * there may not be any codec clock yet.
 	 */

-	return 0;
+	ret = au1xpsc_asoc_dbdma_init(pdev);
+	if (ret)
+		goto out2;
+
+	ret = snd_soc_register_dai(&au1xpsc_ac97_dai);
+	if (!ret)
+		return 0;
+

+	au1xpsc_asoc_dbdma_exit();
+out2:
+	iounmap(au1xpsc_ac97_workdata->mmio);
 out1:
 	release_resource(au1xpsc_ac97_workdata->ioarea);
 	kfree(au1xpsc_ac97_workdata->ioarea);
@@ -298,9 +332,11 @@ out0:
 	return ret;
 }

-static void au1xpsc_ac97_remove(struct platform_device *pdev,
-				struct snd_soc_dai *dai)
+static int __devexit au1xpsc_ac97_remove(struct platform_device *pdev)
 {
+	snd_soc_unregister_dai(&au1xpsc_ac97_dai);
+	au1xpsc_asoc_dbdma_exit();
+
 	/* disable PSC completely */
 	au_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
 	au_sync();
@@ -312,9 +348,11 @@ static void au1xpsc_ac97_remove(struct platform_device *pdev,
 	kfree(au1xpsc_ac97_workdata->ioarea);
 	kfree(au1xpsc_ac97_workdata);
 	au1xpsc_ac97_workdata = NULL;
+
+	return 0;
 }

-static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai)
+static int au1xpsc_ac97_suspend(struct platform_device *pdev, pm_message_t m)
 {
 	/* save interesting registers and disable PSC */
 	au1xpsc_ac97_workdata->pm[0] =
@@ -328,7 +366,7 @@ static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai)
 	return 0;
 }

-static int au1xpsc_ac97_resume(struct snd_soc_dai *dai)
+static int au1xpsc_ac97_resume(struct platform_device *pdev)
 {
 	/* restore PSC clock config */
 	au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE,
@@ -342,48 +380,31 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai)
 	return 0;
 }

-static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
-	.trigger	= au1xpsc_ac97_trigger,
-	.hw_params	= au1xpsc_ac97_hw_params,
-};
-
-struct snd_soc_dai au1xpsc_ac97_dai = {
-	.name			= "au1xpsc_ac97",
-	.ac97_control		= 1,
-	.probe			= au1xpsc_ac97_probe,
-	.remove			= au1xpsc_ac97_remove,
-	.suspend		= au1xpsc_ac97_suspend,
-	.resume			= au1xpsc_ac97_resume,
-	.playback = {
-		.rates		= AC97_RATES,
-		.formats	= AC97_FMTS,
-		.channels_min	= 2,
-		.channels_max	= 2,
+static struct platform_driver au1xpsc_ac97_driver = {
+	.driver	= {
+		.name	= "au1xpsc_ac97",
+		.owner	= THIS_MODULE,
 	},
-	.capture = {
-		.rates		= AC97_RATES,
-		.formats	= AC97_FMTS,
-		.channels_min	= 2,
-		.channels_max	= 2,
-	},
-	.ops = &au1xpsc_ac97_dai_ops,
+	.probe		= au1xpsc_ac97_probe,
+	.remove		= __devexit_p(au1xpsc_ac97_remove),
+	.suspend	= au1xpsc_ac97_suspend,
+	.resume		= au1xpsc_ac97_resume,
 };
-EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai);

-static int __init au1xpsc_ac97_init(void)
+static int __init au1xpsc_ac97_load(void)
 {
 	au1xpsc_ac97_workdata = NULL;
-	return snd_soc_register_dai(&au1xpsc_ac97_dai);
+	return platform_driver_register(&au1xpsc_ac97_driver);
 }

-static void __exit au1xpsc_ac97_exit(void)
+static void __exit au1xpsc_ac97_unload(void)
 {
-	snd_soc_unregister_dai(&au1xpsc_ac97_dai);
+	platform_driver_unregister(&au1xpsc_ac97_driver);
 }

-module_init(au1xpsc_ac97_init);
-module_exit(au1xpsc_ac97_exit);
+module_init(au1xpsc_ac97_load);
+module_exit(au1xpsc_ac97_unload);

 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver");
-MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index bb58932..f830f5a 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -1,8 +1,8 @@
 /*
  * Au12x0/Au1550 PSC ALSA ASoC audio support.
  *
- * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
- *	Manuel Lauss <mano@roarinelk.homelinux.net>
+ * (c) 2007-2009 MSC Vertriebsges.m.b.H.,
+ *	Manuel Lauss <manuel.lauss@gmail.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
@@ -262,8 +262,31 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 	return ret;
 }

-static int au1xpsc_i2s_probe(struct platform_device *pdev,
-			     struct snd_soc_dai *dai)
+static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
+	.trigger	= au1xpsc_i2s_trigger,
+	.hw_params	= au1xpsc_i2s_hw_params,
+	.set_fmt	= au1xpsc_i2s_set_fmt,
+};
+
+struct snd_soc_dai au1xpsc_i2s_dai = {
+	.name			= "au1xpsc_i2s",
+	.playback = {
+		.rates		= AU1XPSC_I2S_RATES,
+		.formats	= AU1XPSC_I2S_FMTS,
+		.channels_min	= 2,
+		.channels_max	= 8,	/* 2 without external help */
+	},
+	.capture = {
+		.rates		= AU1XPSC_I2S_RATES,
+		.formats	= AU1XPSC_I2S_FMTS,
+		.channels_min	= 2,
+		.channels_max	= 8,	/* 2 without external help */
+	},
+	.ops = &au1xpsc_i2s_dai_ops,
+};
+EXPORT_SYMBOL(au1xpsc_i2s_dai);
+
+static int au1xpsc_i2s_probe(struct platform_device *pdev)
 {
 	struct resource *r;
 	unsigned long sel;
@@ -294,9 +317,7 @@ static int au1xpsc_i2s_probe(struct platform_device *pdev,
 	if (!au1xpsc_i2s_workdata->mmio)
 		goto out1;

-	/* preserve PSC clock source set up by platform (dev.platform_data
-	 * is already occupied by soc layer)
-	 */
+	/* preserve PSC clock source set up by platform  */
 	sel = au_readl(PSC_SEL(au1xpsc_i2s_workdata)) & PSC_SEL_CLK_MASK;
 	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
 	au_sync();
@@ -312,9 +333,16 @@ static int au1xpsc_i2s_probe(struct platform_device *pdev,
 	 * be running yet; depending on clock input for PSC a wait might
 	 * time out.
 	 */
+	ret = au1xpsc_asoc_dbdma_init(pdev);
+	if (ret)
+		goto out2;

-	return 0;
+	ret = snd_soc_register_dai(&au1xpsc_i2s_dai);
+	if (!ret)
+		return 0;

+out2:
+	iounmap(au1xpsc_i2s_workdata->mmio);
 out1:
 	release_resource(au1xpsc_i2s_workdata->ioarea);
 	kfree(au1xpsc_i2s_workdata->ioarea);
@@ -324,9 +352,11 @@ out0:
 	return ret;
 }

-static void au1xpsc_i2s_remove(struct platform_device *pdev,
-			       struct snd_soc_dai *dai)
+static int __devexit au1xpsc_i2s_remove(struct platform_device *pdev)
 {
+	snd_soc_unregister_dai(&au1xpsc_i2s_dai);
+	au1xpsc_asoc_dbdma_exit();
+
 	au_writel(0, I2S_CFG(au1xpsc_i2s_workdata));
 	au_sync();
 	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
@@ -337,9 +367,11 @@ static void au1xpsc_i2s_remove(struct platform_device *pdev,
 	kfree(au1xpsc_i2s_workdata->ioarea);
 	kfree(au1xpsc_i2s_workdata);
 	au1xpsc_i2s_workdata = NULL;
+
+	return 0;
 }

-static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai)
+static int au1xpsc_i2s_suspend(struct platform_device *pdev, pm_message_t m)
 {
 	/* save interesting register and disable PSC */
 	au1xpsc_i2s_workdata->pm[0] =
@@ -353,7 +385,7 @@ static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai)
 	return 0;
 }

-static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai)
+static int au1xpsc_i2s_resume(struct platform_device *pdev)
 {
 	/* select I2S mode and PSC clock */
 	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
@@ -367,48 +399,31 @@ static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai)
 	return 0;
 }

-static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
-	.trigger	= au1xpsc_i2s_trigger,
-	.hw_params	= au1xpsc_i2s_hw_params,
-	.set_fmt	= au1xpsc_i2s_set_fmt,
-};
-
-struct snd_soc_dai au1xpsc_i2s_dai = {
-	.name			= "au1xpsc_i2s",
-	.probe			= au1xpsc_i2s_probe,
-	.remove			= au1xpsc_i2s_remove,
-	.suspend		= au1xpsc_i2s_suspend,
-	.resume			= au1xpsc_i2s_resume,
-	.playback = {
-		.rates		= AU1XPSC_I2S_RATES,
-		.formats	= AU1XPSC_I2S_FMTS,
-		.channels_min	= 2,
-		.channels_max	= 8,	/* 2 without external help */
+static struct platform_driver au1xpsc_i2s_driver = {
+	.driver	= {
+		.name	= "au1xpsc_i2s",
+		.owner	= THIS_MODULE,
 	},
-	.capture = {
-		.rates		= AU1XPSC_I2S_RATES,
-		.formats	= AU1XPSC_I2S_FMTS,
-		.channels_min	= 2,
-		.channels_max	= 8,	/* 2 without external help */
-	},
-	.ops = &au1xpsc_i2s_dai_ops,
+	.probe		= au1xpsc_i2s_probe,
+	.remove		= __devexit_p(au1xpsc_i2s_remove),
+	.suspend	= au1xpsc_i2s_suspend,
+	.resume		= au1xpsc_i2s_resume,
 };
-EXPORT_SYMBOL(au1xpsc_i2s_dai);

-static int __init au1xpsc_i2s_init(void)
+static int __init au1xpsc_i2s_load(void)
 {
 	au1xpsc_i2s_workdata = NULL;
-	return snd_soc_register_dai(&au1xpsc_i2s_dai);
+	return platform_driver_register(&au1xpsc_i2s_driver);
 }

-static void __exit au1xpsc_i2s_exit(void)
+static void __exit au1xpsc_i2s_unload(void)
 {
-	snd_soc_unregister_dai(&au1xpsc_i2s_dai);
+	platform_driver_unregister(&au1xpsc_i2s_driver);
 }

-module_init(au1xpsc_i2s_init);
-module_exit(au1xpsc_i2s_exit);
+module_init(au1xpsc_i2s_load);
+module_exit(au1xpsc_i2s_unload);

 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver");
-MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
index 8fdb1a0..9ee4f59 100644
--- a/sound/soc/au1x/psc.h
+++ b/sound/soc/au1x/psc.h
@@ -1,8 +1,8 @@
 /*
  * Au12x0/Au1550 PSC ALSA ASoC audio support.
  *
- * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
- *	Manuel Lauss <mano@roarinelk.homelinux.net>
+ * (c) 2007-2009 MSC Vertriebsges.m.b.H.,
+ *	Manuel Lauss <manuel.lauss@gmail.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
@@ -21,6 +21,9 @@ extern struct snd_soc_dai au1xpsc_i2s_dai;
 extern struct snd_soc_platform au1xpsc_soc_platform;
 extern struct snd_ac97_bus_ops soc_ac97_ops;

+extern int au1xpsc_asoc_dbdma_init(struct platform_device *pdev);
+extern void au1xpsc_asoc_dbdma_exit(void);
+
 struct au1xpsc_audio_data {
 	void __iomem *mmio;

diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
index 27683eb..86380f6 100644
--- a/sound/soc/au1x/sample-ac97.c
+++ b/sound/soc/au1x/sample-ac97.c
@@ -1,7 +1,7 @@
 /*
  * Sample Au12x0/Au1550 PSC AC97 sound machine.
  *
- * Copyright (c) 2007-2008 Manuel Lauss <mano@roarinelk.homelinux.net>
+ * Copyright (c) 2007-2009 Manuel Lauss <manuel.lauss@gmail.com>
  *
  *  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
@@ -44,13 +44,13 @@ static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = {

 static struct snd_soc_card au1xpsc_sample_ac97_machine = {
 	.name		= "Au1xxx PSC AC97 Audio",
+	.platform	= &au1xpsc_soc_platform, /* see dbdma2.c */
 	.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,
 };

@@ -82,7 +82,14 @@ static struct resource au1xpsc_psc1_res[] = {
 	},
 };

-static struct platform_device *au1xpsc_sample_ac97_dev;
+static struct platform_device psc_ac97_dev = {
+	.name		= "au1xpsc_ac97",
+	.id		= 0,
+	.resource	= au1xpsc_psc1_res,
+	.num_resources	= ARRAY_SIZE(au1xpsc_psc1_res),
+};
+
+static struct platform_device *au1xpsc_sample_asoc_dev;

 static int __init au1xpsc_sample_ac97_load(void)
 {
@@ -99,41 +106,42 @@ static int __init au1xpsc_sample_ac97_load(void)
 	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;
+	/* register the PSC-AC97 platform device */
+	ret = platform_device_register(&psc_ac97_dev);
+	if (ret)
+		return -ENODEV;

-	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;
+	/* register machine fabric with ASoC core */
+	ret = -ENOMEM;
+	au1xpsc_sample_asoc_dev = platform_device_alloc("soc-audio", -1);
+	if (!au1xpsc_sample_asoc_dev)
+		goto out;

-	platform_set_drvdata(au1xpsc_sample_ac97_dev,
+	platform_set_drvdata(au1xpsc_sample_asoc_dev,
 			     &au1xpsc_sample_ac97_devdata);
-	au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev;
-	ret = platform_device_add(au1xpsc_sample_ac97_dev);
+	au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_asoc_dev->dev;
+	ret = platform_device_add(au1xpsc_sample_asoc_dev);

 	if (ret) {
-		platform_device_put(au1xpsc_sample_ac97_dev);
-		au1xpsc_sample_ac97_dev = NULL;
+		platform_device_put(au1xpsc_sample_asoc_dev);
+		au1xpsc_sample_asoc_dev = NULL;
 	}

 out:
+	platform_device_unregister(&psc_ac97_dev);
 	return ret;
 }

 static void __exit au1xpsc_sample_ac97_exit(void)
 {
-	platform_device_unregister(au1xpsc_sample_ac97_dev);
+	platform_device_unregister(au1xpsc_sample_asoc_dev);
+	platform_device_unregister(&psc_ac97_dev);
 }

 module_init(au1xpsc_sample_ac97_load);
@@ -141,4 +149,4 @@ module_exit(au1xpsc_sample_ac97_exit);

 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine");
-MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
+MODULE_AUTHOR("Manuel Lauss");
-- 
1.6.3.1

Re: [RFC PATCH] ASoC: au1x: platform driver restructuring

[Mark Brown]

On Sat, Jun 13, 2009 at 05:00:00PM +0200, Manuel Lauss wrote:

> -static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai)
> +static int au1xpsc_ac97_suspend(struct platform_device *pdev, pm_message_t m)
>  {
>  	/* save interesting registers and disable PSC */
>  	au1xpsc_ac97_workdata->pm[0] =
> @@ -328,7 +366,7 @@ static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai)
>  	return 0;
>  }

I'm don't think this is a good idea - it will remove any sequencing
between the suspend of the various ASoC device components, meaning that
the AC97 controller could suspend before the rest of ASoC.  This would
break if ASoC then comes along and tries to suspend the CODEC over an
AC97 link which has been disabled.  With I2S it's probably less serious
but there's a risk of audio issues if the hardware does something dodgy
when suspending or if the SoC is the clock master and the CODEC gets
upset about having clocks stopping when it's using them.

In order to do this properly I have a plan to add a calls into the ASoC
core that the individual drivers can call when they get suspend and
resume calls.  The core will suspend the entire ASoC device when it gets
the first suspend call and will wait for all the suspended devices to
resume before it resumes the ASoC device.  I've not actually added those
calls yet, though.  I can add stubs quickly enough, though - I should
probably do that just now so that we don't need a separate driver
conversion step.

One other thing I'd say is that before registering the DAIs you should
assign the dev field in the DAI structure to be the struct device that
was used to probe the DAI.

Re: [RFC PATCH] ASoC: au1x: platform driver restructuring

[Manuel Lauss]

On Sat, Jun 13, 2009 at 7:44 PM, Mark
Brown<broonie@opensource.wolfsonmicro.com> wrote:
> On Sat, Jun 13, 2009 at 05:00:00PM +0200, Manuel Lauss wrote:
>
>> -static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai)
>> +static int au1xpsc_ac97_suspend(struct platform_device *pdev, pm_message_t m)
>>  {
>>       /* save interesting registers and disable PSC */
>>       au1xpsc_ac97_workdata->pm[0] =
>> @@ -328,7 +366,7 @@ static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai)
>>       return 0;
>>  }
>
> I'm don't think this is a good idea - it will remove any sequencing
> between the suspend of the various ASoC device components, meaning that
> the AC97 controller could suspend before the rest of ASoC.  This would

Reverted back,

> One other thing I'd say is that before registering the DAIs you should
> assign the dev field in the DAI structure to be the struct device that
> was used to probe the DAI.

Done.

This patch unfortunately breaks PM (no sound after suspend-to-ram), so
may take a while until I resend.

Thank you!
        Manuel Lauss

Re: [RFC PATCH] ASoC: au1x: platform driver restructuring

[Mark Brown]

On Sat, Jun 13, 2009 at 10:17:15PM +0200, Manuel Lauss wrote:
> On Sat, Jun 13, 2009 at 7:44 PM, Mark

> > I'm don't think this is a good idea - it will remove any sequencing
> > between the suspend of the various ASoC device components, meaning that
> > the AC97 controller could suspend before the rest of ASoC.  This would

> Reverted back,

OK, when you do resubmit you might want to check for the patch below
should make it in shortly:

>From 831dc0f10f7b2a4856094ff160c018bf19f77527 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sat, 13 Jun 2009 19:55:02 +0100
Subject: [PATCH] ASoC: Add stub suspend and resume calls for ASoC subdevices

Now that ASoC subdevices can be regular devices they can have normal
suspend and resume calls from their buses.  However, suspending them
individually is not desirable since this can lead to problems such as
pops and clicks from devices being suspended with their signals being
amplified or clocks being stopped suddenly.

This will be resolved by having the normal device model suspend and
resume calls call into ASoC which will suspend the entire card while any
of its components are suspended.  At present this is not yet implemented
but in order to aid the transition of drivers to the standard device
model this patch adds API calls for the notifications.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h  |    5 +++++
 sound/soc/soc-core.c |   39 +++++++++++++++++++++++++++++++++++++++
 2 files changed, 44 insertions(+), 0 deletions(-)

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 5297ba7..e6704c0 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -192,6 +192,11 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform);
 int snd_soc_register_codec(struct snd_soc_codec *codec);
 void snd_soc_unregister_codec(struct snd_soc_codec *codec);
 
+#ifdef CONFIG_PM
+int snd_soc_suspend_device(struct device *dev);
+int snd_soc_resume_device(struct device *dev);
+#endif
+
 /* pcm <-> DAI connect */
 void snd_soc_free_pcms(struct snd_soc_device *socdev);
 int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index e1a920c..4414117 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -788,6 +788,45 @@ static int soc_resume(struct platform_device *pdev)
 	return 0;
 }
 
+/**
+ * snd_soc_suspend_device: Notify core of device suspend
+ *
+ * @dev: Device being suspended.
+ *
+ * In order to ensure that the entire audio subsystem is suspended in a
+ * coordinated fashion ASoC devices should suspend themselves when
+ * called by ASoC.  When the standard kernel suspend process asks the
+ * device to suspend it should call this function to initiate a suspend
+ * of the entire ASoC card.
+ *
+ * \note Currently this function is stubbed out.
+ */
+int snd_soc_suspend_device(struct device *dev)
+{
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_suspend_device);
+
+/**
+ * snd_soc_resume_device: Notify core of device resume
+ *
+ * @dev: Device being resumed.
+ *
+ * In order to ensure that the entire audio subsystem is resumed in a
+ * coordinated fashion ASoC devices should resume themselves when called
+ * by ASoC.  When the standard kernel resume process asks the device
+ * to resume it should call this function.  Once all the components of
+ * the card have notified that they are ready to be resumed the card
+ * will be resumed.
+ *
+ * \note Currently this function is stubbed out.
+ */
+int snd_soc_resume_device(struct device *dev)
+{
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_resume_device);
+
 #else
 #define soc_suspend	NULL
 #define soc_resume	NULL
-- 
1.6.3.1