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* [PATCH 1/1] [ASOC]: AD1980 audio codec ASOC driver
@ 2008-05-12 10:45 Bryan Wu
  2008-05-12 11:54 ` Mark Brown
  2008-05-13 10:59 ` Takashi Iwai
  0 siblings, 2 replies; 8+ messages in thread
From: Bryan Wu @ 2008-05-12 10:45 UTC (permalink / raw)
  To: liam.girdwood, linux-kernel; +Cc: Cliff Cai, Bryan Wu

From: Cliff Cai <cliff.cai@analog.com>

Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
---
 sound/soc/codecs/Kconfig  |    4 +
 sound/soc/codecs/Makefile |    2 +
 sound/soc/codecs/ad1980.c |  322 +++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/ad1980.h |   23 ++++
 4 files changed, 351 insertions(+), 0 deletions(-)
 create mode 100644 sound/soc/codecs/ad1980.c
 create mode 100644 sound/soc/codecs/ad1980.h

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 3903ab7..62ef183 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -2,6 +2,10 @@ config SND_SOC_AC97_CODEC
 	tristate
 	depends on SND_SOC
 
+config SND_SOC_AD1980
+	tristate
+	depends on SND_SOC
+
 config SND_SOC_WM8731
 	tristate
 	depends on SND_SOC
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 4e1314c..0568d32 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,4 +1,5 @@
 snd-soc-ac97-objs := ac97.o
+snd-soc-ad1980-objs := ad1980.o
 snd-soc-wm8731-objs := wm8731.o
 snd-soc-wm8750-objs := wm8750.o
 snd-soc-wm8753-objs := wm8753.o
@@ -8,6 +9,7 @@ snd-soc-cs4270-objs := cs4270.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
 
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
+obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
 obj-$(CONFIG_SND_SOC_WM8731)	+= snd-soc-wm8731.o
 obj-$(CONFIG_SND_SOC_WM8750)	+= snd-soc-wm8750.o
 obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
new file mode 100644
index 0000000..dab1904
--- /dev/null
+++ b/sound/soc/codecs/ad1980.c
@@ -0,0 +1,322 @@
+/*
+ * ad1980.c  --  ALSA Soc AD1980 codec support
+ *
+ * Copyright:	Analog Device Inc.
+ * Author:	Roy Huang <roy.huang@analog.com>
+ * 		Cliff Cai <cliff.cai@analog.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ *  Revision history
+ *    1st July 2007   Initial version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "ad1980.h"
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+	unsigned int reg);
+static int ac97_write(struct snd_soc_codec *codec,
+	unsigned int reg, unsigned int val);
+
+/*
+ * AD1980 register cache
+ */
+static const u16 ad1980_reg[] = {
+	0x0090, 0x8000, 0x8000, 0x8000, /* 0 - 6  */
+	0x0000, 0x0000, 0x8008, 0x8008, /* 8 - e  */
+	0x8808, 0x8808, 0x0000, 0x8808, /* 10 - 16 */
+	0x8808, 0x0000, 0x8000, 0x0000, /* 18 - 1e */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 20 - 26 */
+	0x03c7, 0x0000, 0xbb80, 0xbb80, /* 28 - 2e */
+	0xbb80, 0xbb80, 0x0000, 0x8080, /* 30 - 36 */
+	0x8080, 0x2000, 0x0000, 0x0000, /* 38 - 3e */
+	0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
+	0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
+	0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
+	0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
+	0x8080, 0x0000, 0x0000, 0x0000, /* 60 - 66 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
+	0x0000, 0x0000, 0x1001, 0x0000, /* 70 - 76 */
+	0x0000, 0x0000, 0x4144, 0x5370  /* 78 - 7e */
+};
+
+static const char *ad1980_rec_sel[] = {"Mic", "CD", "NC", "AUX", "Line",
+		"Stereo Mix", "Mono Mix", "Phone"};
+
+static const struct soc_enum ad1980_cap_src =
+	SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 7, ad1980_rec_sel);
+
+static const struct snd_kcontrol_new ad1980_snd_ac97_controls[] = {
+SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1),
+SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1),
+
+SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
+SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
+
+SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1),
+SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1),
+
+SOC_DOUBLE("PCM Capture Volume", AC97_REC_GAIN, 8, 0, 31, 0),
+SOC_SINGLE("PCM Capture Switch", AC97_REC_GAIN, 15, 1, 1),
+
+SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1),
+SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
+
+SOC_SINGLE("Phone Capture Volume", AC97_PHONE, 0, 31, 1),
+SOC_SINGLE("Phone Capture Switch", AC97_PHONE, 15, 1, 1),
+
+SOC_SINGLE("Mic Volume", AC97_MIC, 0, 31, 1),
+SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
+
+SOC_SINGLE("Stereo Mic Switch", AC97_AD_MISC, 6, 1, 0),
+SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0),
+
+SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1),
+SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1),
+
+SOC_ENUM("Capture Source", ad1980_cap_src),
+
+SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0),
+};
+
+/* add non dapm controls */
+static int ad1980_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(ad1980_snd_ac97_controls); i++) {
+		err = snd_ctl_add(codec->card, snd_soc_cnew( \
+				&ad1980_snd_ac97_controls[i], codec, NULL));
+		if (err < 0)
+			return err;
+	}
+	return 0;
+}
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+
+	if (reg == AC97_RESET || reg == AC97_INT_PAGING || \
+			reg == AC97_POWERDOWN || reg == AC97_EXTENDED_STATUS  \
+			|| reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2)
+		return soc_ac97_ops.read(codec->ac97, reg);
+	else {
+		reg = reg >> 1;
+
+		if (reg > (ARRAY_SIZE(ad1980_reg)))
+			return -EINVAL;
+
+		return cache[reg];
+	}
+}
+
+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int val)
+{
+	u16 *cache = codec->reg_cache;
+
+	soc_ac97_ops.write(codec->ac97, reg, val);
+	reg = reg >> 1;
+	if (reg <= (ARRAY_SIZE(ad1980_reg)))
+		cache[reg] = val;
+
+	return 0;
+}
+
+struct snd_soc_codec_dai ad1980_dai = {
+	.name = "AC97",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
+};
+EXPORT_SYMBOL_GPL(ad1980_dai);
+
+static int ad1980_reset(struct snd_soc_codec *codec, int try_warm)
+{
+	u16 retry_cnt = 0;
+
+retry:
+	if (try_warm && soc_ac97_ops.warm_reset) {
+		soc_ac97_ops.warm_reset(codec->ac97);
+		if (ac97_read(codec, AC97_RESET) == 0x0090)
+			return 1;
+	}
+
+	soc_ac97_ops.reset(codec->ac97);
+	/* Set bit 16slot in register 74h, then every slot will has only 16
+	 * bits. This command is sent out in 20bit mode, in which case the
+	 * first nibble of data is eaten by the addr. (Tag is always 16 bit)*/
+	ac97_write(codec, AC97_AD_SERIAL_CFG, 0x9900);
+
+	if (ac97_read(codec, AC97_RESET)  != 0x0090)
+		goto err;
+	return 0;
+
+err:
+	while (retry_cnt++ < 10)
+		goto retry;
+
+	printk(KERN_ERR "AD1980 AC97 reset failed\n");
+	return -EIO;
+}
+
+static int ad1980_soc_suspend(struct platform_device *pdev,
+	pm_message_t state)
+{
+	return 0;
+}
+
+static int ad1980_soc_resume(struct platform_device *pdev)
+{
+	return 0;
+}
+
+static int ad1980_soc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+	u16 vendor_id2;
+
+	printk(KERN_INFO "AD1980 SoC Audio Codec\n");
+
+	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (socdev->codec == NULL)
+		return -ENOMEM;
+	codec = socdev->codec;
+	mutex_init(&codec->mutex);
+
+	codec->reg_cache =
+		kzalloc(sizeof(u16) * ARRAY_SIZE(ad1980_reg), GFP_KERNEL);
+	if (codec->reg_cache == NULL) {
+		ret = -ENOMEM;
+		goto cache_err;
+	}
+	memcpy(codec->reg_cache, ad1980_reg, sizeof(u16) * \
+			ARRAY_SIZE(ad1980_reg));
+	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(ad1980_reg);
+	codec->reg_cache_step = 2;
+	codec->name = "AD1980";
+	codec->owner = THIS_MODULE;
+	codec->dai = &ad1980_dai;
+	codec->num_dai = 1;
+	codec->write = ac97_write;
+	codec->read = ac97_read;
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+	if (ret < 0) {
+		printk(KERN_ERR "ad1980: failed to register AC97 codec\n");
+		goto codec_err;
+	}
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0)
+		goto pcm_err;
+
+
+	ret = ad1980_reset(codec, 0);
+	if (ret < 0) {
+		printk(KERN_ERR "AC97 link error\n");
+		goto reset_err;
+	}
+
+	/* Read out vendor ID to make sure it is ad1980 */
+	if (ac97_read(codec, AC97_VENDOR_ID1) != 0x4144)
+		goto reset_err;
+
+	vendor_id2 = ac97_read(codec, AC97_VENDOR_ID2);
+
+	if (vendor_id2 != 0x5370) {
+		if (vendor_id2 != 0x5374)
+			goto reset_err;
+		else
+			printk(KERN_WARNING "ad1980: "
+				"Found AD1981 - only 2/2 IN/OUT Channels "
+				"supported\n");
+	}
+
+	ac97_write(codec, AC97_MASTER, 0x0000); /* unmute line out volume */
+	ac97_write(codec, AC97_PCM, 0x0000);	/* unmute PCM out volume */
+	ac97_write(codec, AC97_REC_GAIN, 0x0000);/* unmute record volume */
+
+	ad1980_add_controls(codec);
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "ad1980: failed to register card\n");
+		goto reset_err;
+	}
+
+	return 0;
+
+reset_err:
+	snd_soc_free_pcms(socdev);
+
+pcm_err:
+	snd_soc_free_ac97_codec(codec);
+
+codec_err:
+	kfree(codec->reg_cache);
+
+cache_err:
+	kfree(socdev->codec);
+	socdev->codec = NULL;
+	return ret;
+}
+
+static int ad1980_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	if (codec == NULL)
+		return 0;
+
+	snd_soc_dapm_free(socdev);
+	snd_soc_free_pcms(socdev);
+	snd_soc_free_ac97_codec(codec);
+	kfree(codec->reg_cache);
+	kfree(codec);
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ad1980 = {
+	.probe = 	ad1980_soc_probe,
+	.remove = 	ad1980_soc_remove,
+	.suspend =	ad1980_soc_suspend,
+	.resume =	ad1980_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ad1980);
+
+MODULE_DESCRIPTION("ASoC ad1980 driver");
+MODULE_AUTHOR("Roy Huang, Cliff Cai");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad1980.h b/sound/soc/codecs/ad1980.h
new file mode 100644
index 0000000..5d4710d
--- /dev/null
+++ b/sound/soc/codecs/ad1980.h
@@ -0,0 +1,23 @@
+/*
+ * ad1980.h  --  ad1980 Soc Audio driver
+ */
+
+#ifndef _AD1980_H
+#define _AD1980_H
+/* Bit definition of Power-Down Control/Status Register */
+#define ADC		0x0001
+#define DAC		0x0002
+#define ANL		0x0004
+#define REF		0x0008
+#define PR0		0x0100
+#define PR1		0x0200
+#define PR2		0x0400
+#define PR3		0x0800
+#define PR4		0x1000
+#define PR5		0x2000
+#define PR6		0x4000
+
+extern struct snd_soc_codec_dai ad1980_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ad1980;
+
+#endif
-- 
1.5.5


^ permalink raw reply related	[flat|nested] 8+ messages in thread

* Re: [PATCH 1/1] [ASOC]: AD1980 audio codec ASOC driver
  2008-05-12 10:45 [PATCH 1/1] [ASOC]: AD1980 audio codec ASOC driver Bryan Wu
@ 2008-05-12 11:54 ` Mark Brown
  2008-05-12 11:56   ` Mark Brown
  2008-05-13  3:00   ` Cai, Cliff
  2008-05-13 10:59 ` Takashi Iwai
  1 sibling, 2 replies; 8+ messages in thread
From: Mark Brown @ 2008-05-12 11:54 UTC (permalink / raw)
  To: Bryan Wu; +Cc: liam.girdwood, linux-kernel, Cliff Cai, alsa-devel

On Mon, May 12, 2008 at 06:45:12PM +0800, Bryan Wu wrote:
> From: Cliff Cai <cliff.cai@analog.com>
> 
> Signed-off-by: Cliff Cai <cliff.cai@analog.com>
> Signed-off-by: Bryan Wu <cooloney@kernel.org>

Thanks, I've applied this to the ASoC git tree.  CCing in
alsa-devel@alsa-project.org - ALSA patches should go via there.

> +static int ad1980_soc_suspend(struct platform_device *pdev,
> +	pm_message_t state)
> +{
> +	return 0;
> +}
> +
> +static int ad1980_soc_resume(struct platform_device *pdev)
> +{
> +	return 0;
> +}

Are you sure about these?  I would expect the suspend and resume
functions to either do some register writes or be omitted if they don't
do anything.  Standard AC97 codecs would have some power management via
register 0x26 if they were doing anything.

^ permalink raw reply	[flat|nested] 8+ messages in thread

* Re: [PATCH 1/1] [ASOC]: AD1980 audio codec ASOC driver
  2008-05-12 11:54 ` Mark Brown
@ 2008-05-12 11:56   ` Mark Brown
  2008-05-13  3:00   ` Cai, Cliff
  1 sibling, 0 replies; 8+ messages in thread
From: Mark Brown @ 2008-05-12 11:56 UTC (permalink / raw)
  To: Bryan Wu, liam.girdwood, linux-kernel, Cliff Cai, alsa-devel

On Mon, May 12, 2008 at 12:54:16PM +0100, Mark Brown wrote:

> Thanks, I've applied this to the ASoC git tree.  CCing in

...actually, it's already there so I've not applied it - sorry for the
noise.

^ permalink raw reply	[flat|nested] 8+ messages in thread

* RE: [PATCH 1/1] [ASOC]: AD1980 audio codec ASOC driver
  2008-05-12 11:54 ` Mark Brown
  2008-05-12 11:56   ` Mark Brown
@ 2008-05-13  3:00   ` Cai, Cliff
  2008-05-13 14:01     ` Mark Brown
  1 sibling, 1 reply; 8+ messages in thread
From: Cai, Cliff @ 2008-05-13  3:00 UTC (permalink / raw)
  To: Mark Brown, Bryan Wu; +Cc: liam.girdwood, linux-kernel, alsa-devel


 ok,we will implement these two functions later.

Best Regards
Cliff Cai

-----Original Message-----
From: Mark Brown [mailto:broonie@opensource.wolfsonmicro.com] 
Sent: Monday, May 12, 2008 7:54 PM
To: Bryan Wu
Cc: liam.girdwood@wolfsonmicro.com; linux-kernel@vger.kernel.org; Cliff
Cai; alsa-devel@alsa-project.org
Subject: Re: [PATCH 1/1] [ASOC]: AD1980 audio codec ASOC driver

On Mon, May 12, 2008 at 06:45:12PM +0800, Bryan Wu wrote:
> From: Cliff Cai <cliff.cai@analog.com>
> 
> Signed-off-by: Cliff Cai <cliff.cai@analog.com>
> Signed-off-by: Bryan Wu <cooloney@kernel.org>

Thanks, I've applied this to the ASoC git tree.  CCing in
alsa-devel@alsa-project.org - ALSA patches should go via there.

> +static int ad1980_soc_suspend(struct platform_device *pdev,
> +	pm_message_t state)
> +{
> +	return 0;
> +}
> +
> +static int ad1980_soc_resume(struct platform_device *pdev) {
> +	return 0;
> +}

Are you sure about these?  I would expect the suspend and resume
functions to either do some register writes or be omitted if they don't
do anything.  Standard AC97 codecs would have some power management via
register 0x26 if they were doing anything.

^ permalink raw reply	[flat|nested] 8+ messages in thread

* Re: [PATCH 1/1] [ASOC]: AD1980 audio codec ASOC driver
  2008-05-12 10:45 [PATCH 1/1] [ASOC]: AD1980 audio codec ASOC driver Bryan Wu
  2008-05-12 11:54 ` Mark Brown
@ 2008-05-13 10:59 ` Takashi Iwai
  1 sibling, 0 replies; 8+ messages in thread
From: Takashi Iwai @ 2008-05-13 10:59 UTC (permalink / raw)
  To: Bryan Wu; +Cc: liam.girdwood, Cliff Cai, Mark Brown, linux-kernel, alsa-devel

At Mon, 12 May 2008 18:45:12 +0800,
Bryan Wu wrote:
> diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
(snip)
> +static int ad1980_add_controls(struct snd_soc_codec *codec)
> +{
> +	int err, i;
> +
> +	for (i = 0; i < ARRAY_SIZE(ad1980_snd_ac97_controls); i++) {
> +		err = snd_ctl_add(codec->card, snd_soc_cnew( \

The backslash isn't needed.

> +static unsigned int ac97_read(struct snd_soc_codec *codec,
> +	unsigned int reg)
> +{
> +	u16 *cache = codec->reg_cache;
> +
> +	if (reg == AC97_RESET || reg == AC97_INT_PAGING || \
> +			reg == AC97_POWERDOWN || reg == AC97_EXTENDED_STATUS  \
> +			|| reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2)

Ditto.  Maybe a switch is a better choice here.

> +		return soc_ac97_ops.read(codec->ac97, reg);
> +	else {
> +		reg = reg >> 1;
> +
> +		if (reg > (ARRAY_SIZE(ad1980_reg)))

Isn't it reg >= ARRAY_SIZE(ad1980_reg) ??

> +static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
> +	unsigned int val)
> +{
> +	u16 *cache = codec->reg_cache;
> +
> +	soc_ac97_ops.write(codec->ac97, reg, val);
> +	reg = reg >> 1;
> +	if (reg <= (ARRAY_SIZE(ad1980_reg)))

And reg < ARRAY_SIZE(ad1980_reg)


thanks,

Takashi

^ permalink raw reply	[flat|nested] 8+ messages in thread

* Re: [PATCH 1/1] [ASOC]: AD1980 audio codec ASOC driver
  2008-05-13  3:00   ` Cai, Cliff
@ 2008-05-13 14:01     ` Mark Brown
  2008-05-13 15:07       ` Bryan Wu
  0 siblings, 1 reply; 8+ messages in thread
From: Mark Brown @ 2008-05-13 14:01 UTC (permalink / raw)
  To: Cai, Cliff; +Cc: Bryan Wu, liam.girdwood, linux-kernel, alsa-devel

On Tue, May 13, 2008 at 11:00:58AM +0800, Cai, Cliff wrote:

>  ok,we will implement these two functions later.

So they can be removed for now?

What's the current status of merging the Blackfin ASoC support?  We've
had patches in the ASoC git tree for some time (along with the AD1980
driver) - it'd be good to get everything merged into ALSA.

^ permalink raw reply	[flat|nested] 8+ messages in thread

* Re: [PATCH 1/1] [ASOC]: AD1980 audio codec ASOC driver
  2008-05-13 14:01     ` Mark Brown
@ 2008-05-13 15:07       ` Bryan Wu
  2008-05-13 15:24         ` Mark Brown
  0 siblings, 1 reply; 8+ messages in thread
From: Bryan Wu @ 2008-05-13 15:07 UTC (permalink / raw)
  To: Cai, Cliff, Bryan Wu, liam.girdwood, linux-kernel, alsa-devel

On Tue, May 13, 2008 at 10:01 PM, Mark Brown
<broonie@opensource.wolfsonmicro.com> wrote:
> On Tue, May 13, 2008 at 11:00:58AM +0800, Cai, Cliff wrote:
>
>  >  ok,we will implement these two functions later.
>
>  So they can be removed for now?
>
>  What's the current status of merging the Blackfin ASoC support?

We plan to cleanup the Blackfin ASoC code, after that we will send out
the code for merging.

> We've had patches in the ASoC git tree for some time (along with the AD1980
>  driver) - it'd be good to get everything merged into ALSA.
>

Do you mean there is another version AD1980 in ASoC git tree?

Thanks
-Bryan

^ permalink raw reply	[flat|nested] 8+ messages in thread

* Re: [PATCH 1/1] [ASOC]: AD1980 audio codec ASOC driver
  2008-05-13 15:07       ` Bryan Wu
@ 2008-05-13 15:24         ` Mark Brown
  0 siblings, 0 replies; 8+ messages in thread
From: Mark Brown @ 2008-05-13 15:24 UTC (permalink / raw)
  To: Bryan Wu; +Cc: Cai, Cliff, liam.girdwood, linux-kernel, alsa-devel

On Tue, May 13, 2008 at 11:07:16PM +0800, Bryan Wu wrote:
> On Tue, May 13, 2008 at 10:01 PM, Mark Brown

> > We've had patches in the ASoC git tree for some time (along with the AD1980
> >  driver) - it'd be good to get everything merged into ALSA.

> Do you mean there is another version AD1980 in ASoC git tree?

We're carrying both AD1980 and Blackfin platform code.  The AD1980
driver is currently identical to the one you just sent.

Everything is in the dev branch of:

	git://opensource.wolfsonmicro.com/linux-2.6-asoc

^ permalink raw reply	[flat|nested] 8+ messages in thread

end of thread, other threads:[~2008-05-13 15:24 UTC | newest]

Thread overview: 8+ messages (download: mbox.gz / follow: Atom feed)
-- links below jump to the message on this page --
2008-05-12 10:45 [PATCH 1/1] [ASOC]: AD1980 audio codec ASOC driver Bryan Wu
2008-05-12 11:54 ` Mark Brown
2008-05-12 11:56   ` Mark Brown
2008-05-13  3:00   ` Cai, Cliff
2008-05-13 14:01     ` Mark Brown
2008-05-13 15:07       ` Bryan Wu
2008-05-13 15:24         ` Mark Brown
2008-05-13 10:59 ` Takashi Iwai

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