From b1132b3d874dcef296a63b46680a41b02de66bb4 Mon Sep 17 00:00:00 2001
From: Haavard Skinnemoen <haavard.skinnemoen@atmel.com>
Date: Fri, 19 Sep 2008 18:50:45 +0200
Subject: ALSA: ASoC: Fix at32-pcm build breakage with PM enabled

s/PDC_PTCR/ATMEL_PDC_PTCR/

Signed-off-by: Haavard Skinnemoen <haavard.skinnemoen@atmel.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/at32/at32-pcm.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c
index 435f1daf177..c83584f989a 100644
--- a/sound/soc/at32/at32-pcm.c
+++ b/sound/soc/at32/at32-pcm.c
@@ -434,7 +434,8 @@ static int at32_pcm_suspend(struct platform_device *pdev,
 	params = prtd->params;
 
 	/* Disable the PDC and save the PDC registers */
-	ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable);
+	ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+		   params->mask->pdc_disable);
 
 	prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr);
 	prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr);
@@ -464,7 +465,7 @@ static int at32_pcm_resume(struct platform_device *pdev,
 	ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save);
 	ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save);
 
-	ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable);
+	ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, params->mask->pdc_enable);
 	return 0;
 }
 #else /* CONFIG_PM */
-- 
cgit v1.2.3-18-g5258


From 399ccdc1cd4e92e541d4dacbbf18c52bd693418b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 25 Sep 2008 14:51:03 +0200
Subject: ALSA: fix locking in snd_pcm_open*() and snd_rawmidi_open*()

The PCM and rawmidi open callbacks have a lock against card->controls_list
but it takes a wrong one, card->controls_rwsem, instead of a right one
card->ctl_files_rwlock.  This patch fixes them.

This change also fixes automatically the potential deadlocks due to
mm->mmap_sem in munmap and copy_from/to_user, reported by Sitsofe
Wheeler:

  A: snd_ctl_elem_user_tlv(): card->controls_rwsem => mm->mmap_sem
  B: snd_pcm_open(): card->open_mutex => card->controls_rwsem
  C: munmap: mm->mmap_sem => snd_pcm_release(): card->open_mutex

The patch breaks the chain.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/pcm.c     | 4 ++--
 sound/core/rawmidi.c | 4 ++--
 2 files changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 9dd9bc73fe1..ece25c718e9 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -781,7 +781,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream,
 		return -ENODEV;
 
 	card = pcm->card;
-	down_read(&card->controls_rwsem);
+	read_lock(&card->ctl_files_rwlock);
 	list_for_each_entry(kctl, &card->ctl_files, list) {
 		if (kctl->pid == current->pid) {
 			prefer_subdevice = kctl->prefer_pcm_subdevice;
@@ -789,7 +789,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream,
 				break;
 		}
 	}
-	up_read(&card->controls_rwsem);
+	read_unlock(&card->ctl_files_rwlock);
 
 	switch (stream) {
 	case SNDRV_PCM_STREAM_PLAYBACK:
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index f7ea7287c59..b917a9f981c 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -418,7 +418,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file)
 	mutex_lock(&rmidi->open_mutex);
 	while (1) {
 		subdevice = -1;
-		down_read(&card->controls_rwsem);
+		read_lock(&card->ctl_files_rwlock);
 		list_for_each_entry(kctl, &card->ctl_files, list) {
 			if (kctl->pid == current->pid) {
 				subdevice = kctl->prefer_rawmidi_subdevice;
@@ -426,7 +426,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file)
 					break;
 			}
 		}
-		up_read(&card->controls_rwsem);
+		read_unlock(&card->ctl_files_rwlock);
 		err = snd_rawmidi_kernel_open(rmidi->card, rmidi->device,
 					      subdevice, fflags, rawmidi_file);
 		if (err >= 0)
-- 
cgit v1.2.3-18-g5258


From 24e8fc498e9618338854bfbcf8d1d737e0bf1775 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 25 Sep 2008 17:51:11 +0200
Subject: ALSA: remove unneeded power_mutex lock in snd_pcm_drop

The power_mutex lock in snd_pcm_drop may cause a possible deadlock
chain, and above all, it's unneeded.  Let's get rid of it.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/pcm_native.c | 13 +++----------
 1 file changed, 3 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index c49b9d9e303..c487025d345 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1546,16 +1546,10 @@ static int snd_pcm_drop(struct snd_pcm_substream *substream)
 	card = substream->pcm->card;
 
 	if (runtime->status->state == SNDRV_PCM_STATE_OPEN ||
-	    runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED)
+	    runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED ||
+	    runtime->status->state == SNDRV_PCM_STATE_SUSPENDED)
 		return -EBADFD;
 
-	snd_power_lock(card);
-	if (runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) {
-		result = snd_power_wait(card, SNDRV_CTL_POWER_D0);
-		if (result < 0)
-			goto _unlock;
-	}
-
 	snd_pcm_stream_lock_irq(substream);
 	/* resume pause */
 	if (runtime->status->state == SNDRV_PCM_STATE_PAUSED)
@@ -1564,8 +1558,7 @@ static int snd_pcm_drop(struct snd_pcm_substream *substream)
 	snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP);
 	/* runtime->control->appl_ptr = runtime->status->hw_ptr; */
 	snd_pcm_stream_unlock_irq(substream);
- _unlock:
-	snd_power_unlock(card);
+
 	return result;
 }
 
-- 
cgit v1.2.3-18-g5258


From ec2cd95f340fb07b905839ee219b3846ecf58396 Mon Sep 17 00:00:00 2001
From: Timur Tabi <timur@freescale.com>
Date: Tue, 29 Jul 2008 16:35:52 -0500
Subject: ALSA: make the CS4270 driver a new-style I2C driver

Update the CS4270 ALSA device driver to use the new-style I2C interface.
Starting with the 2.6.27 PowerPC kernel, I2C devices that have entries in the
device trees can no longer be probed by old-style I2C drivers.  The device
tree for Freescale MPC8610 HPCD has included an entry for the CS4270 since
2.6.25, but that entry was previously ignored by the PowerPC I2C subsystem.
Since that's no longer the case, the best solution is to update the CS4270
driver to a new-style interface, rather than try to revert the behavior of
new PowerPC I2C subsystem.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/codecs/cs4270.c | 71 ++++++++++-------------------------------------
 1 file changed, 15 insertions(+), 56 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 9deb8c74fdf..82d94f00aa4 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -490,34 +490,7 @@ static int cs4270_mute(struct snd_soc_dai *dai, int mute)
 
 #endif
 
-static int cs4270_i2c_probe(struct i2c_adapter *adap, int addr, int kind);
-
-/*
- * Notify the driver that a new I2C bus has been found.
- *
- * This function is called for each I2C bus in the system.  The function
- * then asks the I2C subsystem to probe that bus at the addresses on which
- * our device (the CS4270) could exist.  If a device is found at one of
- * those addresses, then our probe function (cs4270_i2c_probe) is called.
- */
-static int cs4270_i2c_attach(struct i2c_adapter *adapter)
-{
-	return i2c_probe(adapter, &addr_data, cs4270_i2c_probe);
-}
-
-static int cs4270_i2c_detach(struct i2c_client *client)
-{
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-
-	i2c_detach_client(client);
-	codec->control_data = NULL;
-
-	kfree(codec->reg_cache);
-	codec->reg_cache = NULL;
-
-	kfree(client);
-	return 0;
-}
+static int cs4270_i2c_probe(struct i2c_client *, const struct i2c_device_id *);
 
 /* A list of non-DAPM controls that the CS4270 supports */
 static const struct snd_kcontrol_new cs4270_snd_controls[] = {
@@ -525,14 +498,19 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = {
 		CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1)
 };
 
+static const struct i2c_device_id cs4270_id[] = {
+	{"cs4270", 0},
+	{}
+};
+MODULE_DEVICE_TABLE(i2c, cs4270_id);
+
 static struct i2c_driver cs4270_i2c_driver = {
 	.driver = {
 		.name = "CS4270 I2C",
 		.owner = THIS_MODULE,
 	},
-	.id =             I2C_DRIVERID_CS4270,
-	.attach_adapter = cs4270_i2c_attach,
-	.detach_client =  cs4270_i2c_detach,
+	.id_table = cs4270_id,
+	.probe = cs4270_i2c_probe,
 };
 
 /*
@@ -561,11 +539,11 @@ static struct snd_soc_device *cs4270_socdev;
  * Note: snd_soc_new_pcms() must be called before this function can be called,
  * because of snd_ctl_add().
  */
-static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind)
+static int cs4270_i2c_probe(struct i2c_client *i2c_client,
+	const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = cs4270_socdev;
 	struct snd_soc_codec *codec = socdev->codec;
-	struct i2c_client *i2c_client = NULL;
 	int i;
 	int ret = 0;
 
@@ -578,12 +556,6 @@ static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind)
 
 	/* Note: codec_dai->codec is NULL here */
 
-	i2c_client = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
-	if (!i2c_client) {
-		printk(KERN_ERR "cs4270: could not allocate I2C client\n");
-		return -ENOMEM;
-	}
-
 	codec->reg_cache = kzalloc(CS4270_NUMREGS, GFP_KERNEL);
 	if (!codec->reg_cache) {
 		printk(KERN_ERR "cs4270: could not allocate register cache\n");
@@ -591,13 +563,6 @@ static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind)
 		goto error;
 	}
 
-	i2c_set_clientdata(i2c_client, codec);
-	strcpy(i2c_client->name, "CS4270");
-
-	i2c_client->driver = &cs4270_i2c_driver;
-	i2c_client->adapter = adapter;
-	i2c_client->addr = addr;
-
 	/* Verify that we have a CS4270 */
 
 	ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID);
@@ -612,18 +577,10 @@ static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind)
 		goto error;
 	}
 
-	printk(KERN_INFO "cs4270: found device at I2C address %X\n", addr);
+	printk(KERN_INFO "cs4270: found device at I2C address %X\n",
+		i2c_client->addr);
 	printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF);
 
-	/* Tell the I2C layer a new client has arrived */
-
-	ret = i2c_attach_client(i2c_client);
-	if (ret) {
-		printk(KERN_ERR "cs4270: could not attach codec, "
-			"I2C address %x, error code %i\n", addr, ret);
-		goto error;
-	}
-
 	codec->control_data = i2c_client;
 	codec->read = cs4270_read_reg_cache;
 	codec->write = cs4270_i2c_write;
@@ -648,6 +605,8 @@ static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind)
 			goto error;
 	}
 
+	i2c_set_clientdata(i2c_client, codec);
+
 	return 0;
 
 error:
-- 
cgit v1.2.3-18-g5258


From 9778e9a0eafe796c2affcd1fa1fa8a3765e026e6 Mon Sep 17 00:00:00 2001
From: Jean Delvare <khali@linux-fr.org>
Date: Sat, 27 Sep 2008 20:30:52 +0200
Subject: ALSA: ASoC: Fix another cs4270 error path

Conversion to new-style i2c driver missed the error path of the
probe function. Fix it.

Signed-off-by: Jean Delvare <khali@linux-fr.org>
Cc: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/codecs/cs4270.c | 7 +------
 1 file changed, 1 insertion(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 82d94f00aa4..d68650de39b 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -610,17 +610,12 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 	return 0;
 
 error:
-	if (codec->control_data) {
-		i2c_detach_client(i2c_client);
-		codec->control_data = NULL;
-	}
+	codec->control_data = NULL;
 
 	kfree(codec->reg_cache);
 	codec->reg_cache = NULL;
 	codec->reg_cache_size = 0;
 
-	kfree(i2c_client);
-
 	return ret;
 }
 
-- 
cgit v1.2.3-18-g5258


From e3145dfb7b4262fa55907006b75da799de8c1be3 Mon Sep 17 00:00:00 2001
From: Jean Delvare <khali@linux-fr.org>
Date: Tue, 30 Sep 2008 11:40:37 +0200
Subject: ALSA: ASoC: Fix cs4270 error path

The error path in cs4270_probe/cs4270_remove is pretty broken:
* If cs4270_probe fails, codec is leaked.
* If snd_soc_register_card fails, cs4270_i2c_driver stays registered.
* If I2C support is enabled but no I2C device is found, i2c_del_driver
  is never called (neither in cs4270_probe nor in cs4270_remove.

Fix all 3 problems by implementing a clean error path in cs4270_probe
and jumping to its labels as needed.

Signed-off-by: Jean Delvare <khali@linux-fr.org>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/codecs/cs4270.c | 25 ++++++++++++++++++-------
 1 file changed, 18 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index d68650de39b..0bbd94501d7 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -681,7 +681,7 @@ static int cs4270_probe(struct platform_device *pdev)
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
 		printk(KERN_ERR "cs4270: failed to create PCMs\n");
-		return ret;
+		goto error_free_codec;
 	}
 
 #ifdef USE_I2C
@@ -690,8 +690,7 @@ static int cs4270_probe(struct platform_device *pdev)
 	ret = i2c_add_driver(&cs4270_i2c_driver);
 	if (ret) {
 		printk(KERN_ERR "cs4270: failed to attach driver");
-		snd_soc_free_pcms(socdev);
-		return ret;
+		goto error_free_pcms;
 	}
 
 	/* Did we find a CS4270 on the I2C bus? */
@@ -713,10 +712,23 @@ static int cs4270_probe(struct platform_device *pdev)
 	ret = snd_soc_register_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "cs4270: failed to register card\n");
-		snd_soc_free_pcms(socdev);
-		return ret;
+		goto error_del_driver;
 	}
 
+	return 0;
+
+error_del_driver:
+#ifdef USE_I2C
+	i2c_del_driver(&cs4270_i2c_driver);
+
+error_free_pcms:
+#endif
+	snd_soc_free_pcms(socdev);
+
+error_free_codec:
+	kfree(socdev->codec);
+	socdev->codec = NULL;
+
 	return ret;
 }
 
@@ -727,8 +739,7 @@ static int cs4270_remove(struct platform_device *pdev)
 	snd_soc_free_pcms(socdev);
 
 #ifdef USE_I2C
-	if (socdev->codec->control_data)
-		i2c_del_driver(&cs4270_i2c_driver);
+	i2c_del_driver(&cs4270_i2c_driver);
 #endif
 
 	kfree(socdev->codec);
-- 
cgit v1.2.3-18-g5258


From 24918b61b55c21e09a3e07cd82e1b3a8154782dc Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 30 Sep 2008 12:58:54 +0200
Subject: ALSA: hda - Fix model for Dell Inspiron 1525

Dell Inspiron 1525 seems to have a buggy BIOS setup and screws up
the recent codec parser, as reported by Oleksandr Natalenko:
    http://lkml.org/lkml/2008/9/12/203

This patch adds the working model, dell-3stack, statically.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
---
 sound/pci/hda/patch_sigmatel.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index ad994fcab72..f3da621f25c 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1683,8 +1683,8 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
 	/* Dell 3 stack systems with verb table in BIOS */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f3, "Dell Inspiron 1420", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x0227, "Dell Vostro 1400  ", STAC_DELL_BIOS),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x022f, "Dell     ", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x022e, "Dell     ", STAC_DELL_BIOS),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x022f, "Dell Inspiron 1525", STAC_DELL_3ST),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x0242, "Dell     ", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x0243, "Dell     ", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x02ff, "Dell     ", STAC_DELL_BIOS),
-- 
cgit v1.2.3-18-g5258


From 4037314afc6eea3eab7e0447884c12b57a081f63 Mon Sep 17 00:00:00 2001
From: Rob Sims <openmoko-z@robsims.com>
Date: Wed, 1 Oct 2008 21:47:31 +0200
Subject: ASoC: Set correct name for WM8753 rec mixer output

Rob Sims wrote:

"I can't seem to turn on register 0x17, bit 3 in the sound chip, except
by codec_reg_write; the mixer lacks direct or indirect control.  It
seems there are two names for the output of the rec mixer:
Capture ST Mixer
Playback Mixer

Would the following do the trick?"

I confirm that this solves the audio problems I was having.

Signed-off-by: Jonas Bonn <jonas.bonn@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/codecs/wm8753.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 5761164fe16..e873414840c 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -583,7 +583,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 
 	/* out 4 */
 	{"Out4 Mux", "VREF", "VREF"},
-	{"Out4 Mux", "Capture ST", "Capture ST Mixer"},
+	{"Out4 Mux", "Capture ST", "Playback Mixer"},
 	{"Out4 Mux", "LOUT2", "LOUT2"},
 	{"Out 4", NULL, "Out4 Mux"},
 	{"OUT4", NULL, "Out 4"},
@@ -607,7 +607,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	/* Capture Right Mux */
 	{"Capture Right Mux", "PGA", "Right Capture Volume"},
 	{"Capture Right Mux", "Line or RXP-RXN", "Line Right Mux"},
-	{"Capture Right Mux", "Sidetone", "Capture ST Mixer"},
+	{"Capture Right Mux", "Sidetone", "Playback Mixer"},
 
 	/* Mono Capture mixer-mux */
 	{"Capture Right Mixer", "Stereo", "Capture Right Mux"},
-- 
cgit v1.2.3-18-g5258


From 4dbf95ba6c344186ec6d38ff514dc675da464bec Mon Sep 17 00:00:00 2001
From: Risto Suominen <Risto.Suominen@gmail.com>
Date: Mon, 25 Aug 2008 08:02:12 +0200
Subject: ALSA: snd-powermac: mixers for PowerMac G4 AGP

Add mixer controls for PowerMac G4 AGP (Screamer).

This patch fixes the regression in the recent snd-powermac which
doesn't support some G3/G4 PowerMacs:
    http://lkml.org/lkml/2008/10/1/220

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Tested-by: Mariusz Kozlowski <m.kozlowski@tuxland.pl>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/ppc/awacs.c | 19 ++++++++++++++++---
 1 file changed, 16 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c
index 566a6d0daf4..543d4f1784a 100644
--- a/sound/ppc/awacs.c
+++ b/sound/ppc/awacs.c
@@ -621,6 +621,13 @@ static struct snd_kcontrol_new snd_pmac_screamer_mixers_imac[] __initdata = {
 	AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0),
 };
 
+static struct snd_kcontrol_new snd_pmac_screamer_mixers_g4agp[] __initdata = {
+	AWACS_VOLUME("Line out Playback Volume", 2, 6, 1),
+	AWACS_VOLUME("Master Playback Volume", 5, 6, 1),
+	AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0),
+	AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0),
+};
+
 static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac7500[] __initdata = {
 	AWACS_VOLUME("Line out Playback Volume", 2, 6, 1),
 	AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0),
@@ -768,6 +775,7 @@ static void snd_pmac_awacs_resume(struct snd_pmac *chip)
 #define IS_IMAC (machine_is_compatible("PowerMac2,1") \
 		|| machine_is_compatible("PowerMac2,2") \
 		|| machine_is_compatible("PowerMac4,1"))
+#define IS_G4AGP (machine_is_compatible("PowerMac3,1"))
 
 static int imac;
 
@@ -850,6 +858,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
 {
 	int pm7500 = IS_PM7500;
 	int beige = IS_BEIGE;
+	int g4agp = IS_G4AGP;
 	int err, vol;
 
 	imac = IS_IMAC;
@@ -939,7 +948,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
 				snd_pmac_awacs_mixers);
 	if (err < 0)
 		return err;
-	if (beige)
+	if (beige || g4agp)
 		;
 	else if (chip->model == PMAC_SCREAMER)
 		err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers2),
@@ -961,13 +970,17 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
 		err = build_mixers(chip,
 				   ARRAY_SIZE(snd_pmac_screamer_mixers_imac),
 				   snd_pmac_screamer_mixers_imac);
+	else if (g4agp)
+		err = build_mixers(chip,
+				   ARRAY_SIZE(snd_pmac_screamer_mixers_g4agp),
+				   snd_pmac_screamer_mixers_g4agp);
 	else
 		err = build_mixers(chip,
 				   ARRAY_SIZE(snd_pmac_awacs_mixers_pmac),
 				   snd_pmac_awacs_mixers_pmac);
 	if (err < 0)
 		return err;
-	chip->master_sw_ctl = snd_ctl_new1((pm7500 || imac)
+	chip->master_sw_ctl = snd_ctl_new1((pm7500 || imac || g4agp)
 			? &snd_pmac_awacs_master_sw_imac
 			: &snd_pmac_awacs_master_sw, chip);
 	err = snd_ctl_add(chip->card, chip->master_sw_ctl);
@@ -1012,7 +1025,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
 			return err;
 	}
 
-	if (beige)
+	if (beige || g4agp)
 		err = build_mixers(chip,
 				ARRAY_SIZE(snd_pmac_screamer_mic_boost_beige),
 				snd_pmac_screamer_mic_boost_beige);
-- 
cgit v1.2.3-18-g5258


From 030b655b062fe5190fc490e0091ea50307d7a86f Mon Sep 17 00:00:00 2001
From: Risto Suominen <Risto.Suominen@gmail.com>
Date: Mon, 25 Aug 2008 08:04:23 +0200
Subject: ALSA: snd-powermac: HP detection for 1st iMac G3 SL

Correct headphone detection for 1st generation iMac G3 Slot-loading (Screamer).

This patch fixes the regression in the recent snd-powermac which
doesn't support some G3/G4 PowerMacs:
    http://lkml.org/lkml/2008/10/1/220

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Tested-by: Mariusz Kozlowski <m.kozlowski@tuxland.pl>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/ppc/awacs.c | 31 ++++++++++++++++++++++---------
 1 file changed, 22 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c
index 543d4f1784a..106c48225bb 100644
--- a/sound/ppc/awacs.c
+++ b/sound/ppc/awacs.c
@@ -695,7 +695,10 @@ static struct snd_kcontrol_new snd_pmac_awacs_speaker_vol[] __initdata = {
 static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw __initdata =
 AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_SPKMUTE, 1);
 
-static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac __initdata =
+static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac1 __initdata =
+AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 1);
+
+static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac2 __initdata =
 AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 0);
 
 
@@ -772,12 +775,12 @@ static void snd_pmac_awacs_resume(struct snd_pmac *chip)
 
 #define IS_PM7500 (machine_is_compatible("AAPL,7500"))
 #define IS_BEIGE (machine_is_compatible("AAPL,Gossamer"))
-#define IS_IMAC (machine_is_compatible("PowerMac2,1") \
-		|| machine_is_compatible("PowerMac2,2") \
+#define IS_IMAC1 (machine_is_compatible("PowerMac2,1"))
+#define IS_IMAC2 (machine_is_compatible("PowerMac2,2") \
 		|| machine_is_compatible("PowerMac4,1"))
 #define IS_G4AGP (machine_is_compatible("PowerMac3,1"))
 
-static int imac;
+static int imac1, imac2;
 
 #ifdef PMAC_SUPPORT_AUTOMUTE
 /*
@@ -823,13 +826,18 @@ static void snd_pmac_awacs_update_automute(struct snd_pmac *chip, int do_notify)
 		{
 			int reg = chip->awacs_reg[1]
 				| (MASK_HDMUTE | MASK_SPKMUTE);
-			if (imac) {
+			if (imac1) {
+				reg &= ~MASK_SPKMUTE;
+				reg |= MASK_PAROUT1;
+			} else if (imac2) {
 				reg &= ~MASK_SPKMUTE;
 				reg &= ~MASK_PAROUT1;
 			}
 			if (snd_pmac_awacs_detect_headphone(chip))
 				reg &= ~MASK_HDMUTE;
-			else if (imac)
+			else if (imac1)
+				reg &= ~MASK_PAROUT1;
+			else if (imac2)
 				reg |= MASK_PAROUT1;
 			else
 				reg &= ~MASK_SPKMUTE;
@@ -859,9 +867,12 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
 	int pm7500 = IS_PM7500;
 	int beige = IS_BEIGE;
 	int g4agp = IS_G4AGP;
+	int imac;
 	int err, vol;
 
-	imac = IS_IMAC;
+	imac1 = IS_IMAC1;
+	imac2 = IS_IMAC2;
+	imac = imac1 || imac2;
 	/* looks like MASK_GAINLINE triggers something, so we set here
 	 * as start-up
 	 */
@@ -1017,8 +1028,10 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
 					snd_pmac_awacs_speaker_vol);
 		if (err < 0)
 			return err;
-		chip->speaker_sw_ctl = snd_ctl_new1(imac
-				? &snd_pmac_awacs_speaker_sw_imac
+		chip->speaker_sw_ctl = snd_ctl_new1(imac1
+				? &snd_pmac_awacs_speaker_sw_imac1
+				: imac2
+				? &snd_pmac_awacs_speaker_sw_imac2
 				: &snd_pmac_awacs_speaker_sw, chip);
 		err = snd_ctl_add(chip->card, chip->speaker_sw_ctl);
 		if (err < 0)
-- 
cgit v1.2.3-18-g5258


From 68c072388d2339af504c033a51886ea7c6b8d806 Mon Sep 17 00:00:00 2001
From: Rene Herman <rene.herman@gmail.com>
Date: Tue, 15 Jul 2008 16:24:50 +0200
Subject: ALSA: re-order AC97 codec ID table.

Signed-off-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/pci/ac97/ac97_codec.c | 20 ++++++++++----------
 1 file changed, 10 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 8c49a00a5e3..f6a7d721649 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -67,8 +67,8 @@ struct ac97_codec_id {
 };
 
 static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = {
-{ 0x414b4d00, 0xffffff00, "Asahi Kasei",	NULL,	NULL },
 { 0x41445300, 0xffffff00, "Analog Devices",	NULL,	NULL },
+{ 0x414b4d00, 0xffffff00, "Asahi Kasei",	NULL,	NULL },
 { 0x414c4300, 0xffffff00, "Realtek",		NULL,	NULL },
 { 0x414c4700, 0xffffff00, "Realtek",		NULL,	NULL },
 { 0x434d4900, 0xffffff00, "C-Media Electronics", NULL,	NULL },
@@ -94,11 +94,6 @@ static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = {
 };
 
 static const struct ac97_codec_id snd_ac97_codec_ids[] = {
-{ 0x414b4d00, 0xffffffff, "AK4540",		NULL,		NULL },
-{ 0x414b4d01, 0xffffffff, "AK4542",		NULL,		NULL },
-{ 0x414b4d02, 0xffffffff, "AK4543",		NULL,		NULL },
-{ 0x414b4d06, 0xffffffff, "AK4544A",		NULL,		NULL },
-{ 0x414b4d07, 0xffffffff, "AK4545",		NULL,		NULL },
 { 0x41445303, 0xffffffff, "AD1819",		patch_ad1819,	NULL },
 { 0x41445340, 0xffffffff, "AD1881",		patch_ad1881,	NULL },
 { 0x41445348, 0xffffffff, "AD1881A",		patch_ad1881,	NULL },
@@ -112,20 +107,25 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = {
 { 0x41445374, 0xffffffff, "AD1981B",		patch_ad1981b,	NULL },
 { 0x41445375, 0xffffffff, "AD1985",		patch_ad1985,	NULL },
 { 0x41445378, 0xffffffff, "AD1986",		patch_ad1986,	NULL },
+{ 0x414b4d00, 0xffffffff, "AK4540",		NULL,		NULL },
+{ 0x414b4d01, 0xffffffff, "AK4542",		NULL,		NULL },
+{ 0x414b4d02, 0xffffffff, "AK4543",		NULL,		NULL },
+{ 0x414b4d06, 0xffffffff, "AK4544A",		NULL,		NULL },
+{ 0x414b4d07, 0xffffffff, "AK4545",		NULL,		NULL },
 { 0x414c4300, 0xffffff00, "ALC100,100P", 	NULL,		NULL },
 { 0x414c4710, 0xfffffff0, "ALC200,200P",	NULL,		NULL },
 { 0x414c4721, 0xffffffff, "ALC650D",		NULL,	NULL }, /* already patched */
 { 0x414c4722, 0xffffffff, "ALC650E",		NULL,	NULL }, /* already patched */
 { 0x414c4723, 0xffffffff, "ALC650F",		NULL,	NULL }, /* already patched */
 { 0x414c4720, 0xfffffff0, "ALC650",		patch_alc650,	NULL },
-{ 0x414c4760, 0xfffffff0, "ALC655",		patch_alc655,	NULL },
-{ 0x414c4781, 0xffffffff, "ALC658D",		NULL,	NULL }, /* already patched */
-{ 0x414c4780, 0xfffffff0, "ALC658",		patch_alc655,	NULL },
-{ 0x414c4790, 0xfffffff0, "ALC850",		patch_alc850,	NULL },
 { 0x414c4730, 0xffffffff, "ALC101",		NULL,		NULL },
 { 0x414c4740, 0xfffffff0, "ALC202",		NULL,		NULL },
 { 0x414c4750, 0xfffffff0, "ALC250",		NULL,		NULL },
+{ 0x414c4760, 0xfffffff0, "ALC655",		patch_alc655,	NULL },
 { 0x414c4770, 0xfffffff0, "ALC203",		NULL,		NULL },
+{ 0x414c4781, 0xffffffff, "ALC658D",		NULL,	NULL }, /* already patched */
+{ 0x414c4780, 0xfffffff0, "ALC658",		patch_alc655,	NULL },
+{ 0x414c4790, 0xfffffff0, "ALC850",		patch_alc850,	NULL },
 { 0x434d4941, 0xffffffff, "CMI9738",		patch_cm9738,	NULL },
 { 0x434d4961, 0xffffffff, "CMI9739",		patch_cm9739,	NULL },
 { 0x434d4969, 0xffffffff, "CMI9780",		patch_cm9780,	NULL },
-- 
cgit v1.2.3-18-g5258


From 1cd2224cd01898a13138f4ab476932cfb689839e Mon Sep 17 00:00:00 2001
From: Matthew Ranostay <mranostay@embeddedalley.com>
Date: Fri, 18 Jul 2008 18:20:52 +0200
Subject: ALSA: hda: digital pc-beep support hd-audio codecs

Added digital pc-beep support using linear tone generation for hd-codecs along
with initial support for several IDT codecs.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/pci/Kconfig              |   8 +++
 sound/pci/hda/Makefile         |   1 +
 sound/pci/hda/hda_beep.c       | 134 +++++++++++++++++++++++++++++++++++++++++
 sound/pci/hda/hda_beep.h       |  44 ++++++++++++++
 sound/pci/hda/hda_codec.h      |   4 ++
 sound/pci/hda/patch_sigmatel.c |  63 +++++++++++++++++++
 6 files changed, 254 insertions(+)
 create mode 100644 sound/pci/hda/hda_beep.c
 create mode 100644 sound/pci/hda/hda_beep.h

(limited to 'sound')

diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 31f52d3fc21..db9e31fd061 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -517,6 +517,14 @@ config SND_HDA_HWDEP
 	  This interface can be used for out-of-band communication
 	  with codecs for debugging purposes.
 
+config SND_HDA_INPUT_BEEP
+	bool "Support digital beep via input layer"
+	depends on SND_HDA_INTEL
+	depends on INPUT=y || INPUT=SND_HDA_INTEL
+	help
+	  Say Y here to build a digital beep interface for HD-audio
+	  driver. This interface is used to generate digital beeps.
+
 config SND_HDA_CODEC_REALTEK
 	bool "Build Realtek HD-audio codec support"
 	depends on SND_HDA_INTEL
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index ab0c726d648..6db92fd954d 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -5,6 +5,7 @@ snd-hda-intel-y := hda_intel.o
 snd-hda-intel-y += hda_codec.o
 snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o
 snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o
+snd-hda-intel-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o
 snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o
 snd-hda-intel-$(CONFIG_SND_HDA_CODEC_REALTEK) += patch_realtek.o
 snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CMEDIA) += patch_cmedia.o
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
new file mode 100644
index 00000000000..5a764c48139
--- /dev/null
+++ b/sound/pci/hda/hda_beep.c
@@ -0,0 +1,134 @@
+/*
+ * Digital Beep Input Interface for HD-audio codec
+ *
+ * Author: Matthew Ranostay <mranostay@embeddedalley.com>
+ * Copyright (c) 2008 Embedded Alley Solutions Inc
+ *
+ *  This driver 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.
+ *
+ *  This driver 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ */
+
+#include <linux/input.h>
+#include <linux/pci.h>
+#include <linux/workqueue.h>
+#include <sound/core.h>
+#include "hda_beep.h"
+
+enum {
+	DIGBEEP_HZ_STEP = 46875,	/* 46.875 Hz */
+	DIGBEEP_HZ_MIN = 93750,		/* 93.750 Hz */
+	DIGBEEP_HZ_MAX = 12000000,	/* 12 KHz */
+};
+
+static void snd_hda_generate_beep(struct work_struct *work)
+{
+	struct hda_beep *beep =
+		container_of(work, struct hda_beep, beep_work);
+	struct hda_codec *codec = beep->codec;
+
+	/* generate tone */
+	snd_hda_codec_write_cache(codec, beep->nid, 0,
+			AC_VERB_SET_BEEP_CONTROL, beep->tone);
+}
+
+static int snd_hda_beep_event(struct input_dev *dev, unsigned int type,
+				unsigned int code, int hz)
+{
+	struct hda_beep *beep = input_get_drvdata(dev);
+
+	switch (code) {
+	case SND_BELL:
+		if (hz)
+			hz = 1000;
+	case SND_TONE:
+		hz *= 1000; /* fixed point */
+		hz = hz - DIGBEEP_HZ_MIN;
+		if (hz < 0)
+			hz = 0; /* turn off PC beep*/
+		else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN))
+			hz = 0xff;
+		else {
+			hz /= DIGBEEP_HZ_STEP;
+			hz++;
+		}
+		break;
+	default:
+		return -1;
+	}
+	beep->tone = hz;
+
+	/* schedule beep event */
+	schedule_work(&beep->beep_work);
+	return 0;
+}
+
+int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
+{
+	struct input_dev *input_dev;
+	struct hda_beep *beep;
+	int err;
+
+	beep = kzalloc(sizeof(*beep), GFP_KERNEL);
+	if (beep == NULL)
+		return -ENOMEM;
+	snprintf(beep->phys, sizeof(beep->phys),
+		"card%d/codec#%d/beep0", codec->bus->card->number, codec->addr);
+	input_dev = input_allocate_device();
+
+	/* setup digital beep device */
+	input_dev->name = "HDA Digital PCBeep";
+	input_dev->phys = beep->phys;
+	input_dev->id.bustype = BUS_PCI;
+
+	input_dev->id.vendor = codec->vendor_id >> 16;
+	input_dev->id.product = codec->vendor_id & 0xffff;
+	input_dev->id.version = 0x01;
+
+	input_dev->evbit[0] = BIT_MASK(EV_SND);
+	input_dev->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE);
+	input_dev->event = snd_hda_beep_event;
+	input_dev->dev.parent = &codec->bus->pci->dev;
+	input_set_drvdata(input_dev, beep);
+
+	err = input_register_device(input_dev);
+	if (err < 0) {
+		kfree(input_dev);
+		kfree(beep);
+		return err;
+	}
+
+	/* enable linear scale */
+	snd_hda_codec_write(codec, nid, 0,
+		AC_VERB_SET_DIGI_CONVERT_2, 0x01);
+
+	beep->nid = nid;
+	beep->dev = input_dev;
+	beep->codec = codec;
+	codec->beep = beep;
+
+	INIT_WORK(&beep->beep_work, &snd_hda_generate_beep);
+	return 0;
+}
+
+void snd_hda_detach_beep_device(struct hda_codec *codec)
+{
+	struct hda_beep *beep = codec->beep;
+	if (beep) {
+		cancel_work_sync(&beep->beep_work);
+		flush_scheduled_work();
+
+		input_unregister_device(beep->dev);
+		kfree(beep);
+	}
+}
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
new file mode 100644
index 00000000000..de4036e6e71
--- /dev/null
+++ b/sound/pci/hda/hda_beep.h
@@ -0,0 +1,44 @@
+/*
+ * Digital Beep Input Interface for HD-audio codec
+ *
+ * Author: Matthew Ranostay <mranostay@embeddedalley.com>
+ * Copyright (c) 2008 Embedded Alley Solutions Inc
+ *
+ *  This driver 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.
+ *
+ *  This driver 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ */
+
+#ifndef __SOUND_HDA_BEEP_H
+#define __SOUND_HDA_BEEP_H
+
+#include "hda_codec.h"
+
+/* beep information */
+struct hda_beep {
+	struct input_dev *dev;
+	struct hda_codec *codec;
+	char phys[32];
+	int tone;
+	int nid;
+	struct work_struct beep_work; /* scheduled task for beep event */
+};
+
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+int snd_hda_attach_beep_device(struct hda_codec *codec, int nid);
+void snd_hda_detach_beep_device(struct hda_codec *codec);
+#else
+#define snd_hda_attach_beep_device(...)
+#define snd_hda_detach_beep_device(...)
+#endif
+#endif
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index efc682888b3..3a63c445d36 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -449,6 +449,7 @@ enum {
  */
 
 struct hda_bus;
+struct hda_beep;
 struct hda_codec;
 struct hda_pcm;
 struct hda_pcm_stream;
@@ -634,6 +635,9 @@ struct hda_codec {
 	/* codec specific info */
 	void *spec;
 
+	/* beep device */
+	struct hda_beep *beep;
+
 	/* widget capabilities cache */
 	unsigned int num_nodes;
 	hda_nid_t start_nid;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index f3da621f25c..6ee73ed23dd 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -33,6 +33,7 @@
 #include "hda_codec.h"
 #include "hda_local.h"
 #include "hda_patch.h"
+#include "hda_beep.h"
 
 #define NUM_CONTROL_ALLOC	32
 #define STAC_PWR_EVENT		0x20
@@ -164,6 +165,8 @@ struct sigmatel_spec {
 	unsigned int num_dmuxes;
 	hda_nid_t dig_in_nid;
 	hda_nid_t mono_nid;
+	hda_nid_t anabeep_nid;
+	hda_nid_t digbeep_nid;
 
 	/* pin widgets */
 	hda_nid_t *pin_nids;
@@ -690,6 +693,8 @@ static struct hda_verb d965_core_init[] = {
 static struct hda_verb stac927x_core_init[] = {
 	/* set master volume and direct control */	
 	{ 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+	/* enable analog pc beep path */
+	{ 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5},
 	{}
 };
 
@@ -829,8 +834,11 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT),
 
+	/* analog pc-beep replaced with digital beep support */
+	/*
 	HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT),
 	HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT),
+	*/
 
 	HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT),
 	HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT),
@@ -2609,6 +2617,34 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec)
 				"Mono Mux", spec->mono_nid);
 }
 
+/* create PC beep volume controls */
+static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec,
+						hda_nid_t nid)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	u32 caps = query_amp_caps(codec, nid, HDA_OUTPUT);
+	int err;
+
+	/* check for mute support for the the amp */
+	if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) {
+		err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE,
+			"PC Beep Playback Switch",
+			HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT));
+			if (err < 0)
+				return err;
+	}
+
+	/* check to see if there is volume support for the amp */
+	if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) {
+		err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL,
+			"PC Beep Playback Volume",
+			HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT));
+			if (err < 0)
+				return err;
+	}
+	return 0;
+}
+
 /* labels for dmic mux inputs */
 static const char *stac92xx_dmic_labels[5] = {
 	"Analog Inputs", "Digital Mic 1", "Digital Mic 2",
@@ -2844,6 +2880,28 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 	if (err < 0)
 		return err;
 
+	/* setup analog beep controls */
+	if (spec->anabeep_nid > 0) {
+		err = stac92xx_auto_create_beep_ctls(codec,
+			spec->anabeep_nid);
+		if (err < 0)
+			return err;
+	}
+
+	/* setup digital beep controls and input device */
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+	if (spec->digbeep_nid > 0) {
+		hda_nid_t nid = spec->digbeep_nid;
+
+		err = stac92xx_auto_create_beep_ctls(codec, nid);
+		if (err < 0)
+			return err;
+		err = snd_hda_attach_beep_device(codec, nid);
+		if (err < 0)
+			return err;
+	}
+#endif
+
 	if (hp_speaker_swap == 1) {
 		/* Restore the hp_outs and line_outs */
 		memcpy(spec->autocfg.hp_pins, spec->autocfg.line_out_pins,
@@ -3158,6 +3216,7 @@ static void stac92xx_free(struct hda_codec *codec)
 		kfree(spec->bios_pin_configs);
 
 	kfree(spec);
+	snd_hda_detach_beep_device(codec);
 }
 
 static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid,
@@ -3546,6 +3605,7 @@ again:
 	spec->aloopback_mask = 0x01;
 	spec->aloopback_shift = 8;
 
+	spec->digbeep_nid = 0x1c;
 	spec->mux_nids = stac92hd73xx_mux_nids;
 	spec->adc_nids = stac92hd73xx_adc_nids;
 	spec->dmic_nids = stac92hd73xx_dmic_nids;
@@ -3680,6 +3740,7 @@ again:
 	spec->gpio_dir = 0x01;
 	spec->gpio_data = 0x01;
 
+	spec->digbeep_nid = 0x26;
 	spec->mux_nids = stac92hd71bxx_mux_nids;
 	spec->adc_nids = stac92hd71bxx_adc_nids;
 	spec->dmic_nids = stac92hd71bxx_dmic_nids;
@@ -3854,6 +3915,7 @@ static int patch_stac927x(struct hda_codec *codec)
 		stac92xx_set_config_regs(codec);
 	}
 
+	spec->digbeep_nid = 0x23;
 	spec->adc_nids = stac927x_adc_nids;
 	spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids);
 	spec->mux_nids = stac927x_mux_nids;
@@ -3974,6 +4036,7 @@ static int patch_stac9205(struct hda_codec *codec)
 		stac92xx_set_config_regs(codec);
 	}
 
+	spec->digbeep_nid = 0x23;
 	spec->adc_nids = stac9205_adc_nids;
 	spec->num_adcs = ARRAY_SIZE(stac9205_adc_nids);
 	spec->mux_nids = stac9205_mux_nids;
-- 
cgit v1.2.3-18-g5258


From b38addb2da26c0eeab5b538cfbd9d306c50a4726 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Mon, 28 Jul 2008 10:19:39 +0200
Subject: ALSA: usb-audio: add BOSS GT-10 support

Add a quirk entry for the BOSS GT-10.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
---
 sound/usb/usbquirks.h | 31 +++++++++++++++++++++++++++++--
 1 file changed, 29 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h
index 9ea726c049c..3f68359d494 100644
--- a/sound/usb/usbquirks.h
+++ b/sound/usb/usbquirks.h
@@ -1383,7 +1383,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		}
 	}
 },
-
 {
 	/* Roland SonicCell */
 	USB_DEVICE(0x0582, 0x00c2),
@@ -1415,7 +1414,35 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		}
 	}
 },
-
+{
+	/* BOSS GT-10 */
+	USB_DEVICE(0x0582, 0x00da),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_MIDI_FIXED_ENDPOINT,
+				.data = & (const struct snd_usb_midi_endpoint_info) {
+					.out_cables = 0x0001,
+					.in_cables  = 0x0001
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 
 /* Guillemot devices */
 {
-- 
cgit v1.2.3-18-g5258


From d0513fc6c37b009004cf5c7a8e90af0adb3755bc Mon Sep 17 00:00:00 2001
From: Matthew Ranostay <mranostay@embeddedalley.com>
Date: Sun, 27 Jul 2008 10:30:30 +0200
Subject: ALSA: hda: added 92HD81/83 support

Added support for 92HD81/83 family of codecs.
This also includes a pwr_mapping array for pins that have more than
one amp to power down.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/pci/hda/hda_codec.h      |   1 +
 sound/pci/hda/patch_sigmatel.c | 248 +++++++++++++++++++++++++++++++++++++----
 2 files changed, 227 insertions(+), 22 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 3a63c445d36..2f112626f24 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -121,6 +121,7 @@ enum {
 #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1	0x71d
 #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2	0x71e
 #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_3	0x71f
+#define AC_VERB_SET_EAPD				0x788
 #define AC_VERB_SET_CODEC_RESET			0x7ff
 
 /*
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6ee73ed23dd..23a7b2228e3 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -71,6 +71,11 @@ enum {
 	STAC_92HD73XX_MODELS
 };
 
+enum {
+	STAC_92HD83XXX_REF,
+	STAC_92HD83XXX_MODELS
+};
+
 enum {
 	STAC_92HD71BXX_REF,
 	STAC_DELL_M4_1,
@@ -145,6 +150,7 @@ struct sigmatel_spec {
 
 	/* power management */
 	unsigned int num_pwrs;
+	unsigned int *pwr_mapping;
 	hda_nid_t *pwr_nids;
 	hda_nid_t *dac_list;
 
@@ -240,6 +246,33 @@ static hda_nid_t stac92hd73xx_dmux_nids[2] = {
 	0x20, 0x21,
 };
 
+#define STAC92HD83XXX_NUM_DMICS	2
+static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = {
+	0x11, 0x12, 0
+};
+
+#define STAC92HD81_DAC_COUNT 2
+#define STAC92HD83_DAC_COUNT 3
+static hda_nid_t stac92hd83xxx_dac_nids[STAC92HD73_DAC_COUNT] = {
+	0x13, 0x14, 0x22,
+};
+
+static hda_nid_t stac92hd83xxx_dmux_nids[2] = {
+	0x17, 0x18,
+};
+
+static hda_nid_t stac92hd83xxx_adc_nids[2] = {
+	0x15, 0x16,
+};
+
+static hda_nid_t stac92hd83xxx_pwr_nids[4] = {
+	0xa, 0xb, 0xd, 0xe,
+};
+
+static unsigned int stac92hd83xxx_pwr_mapping[4] = {
+	0x03, 0x0c, 0x10, 0x40,
+};
+
 static hda_nid_t stac92hd71bxx_pwr_nids[3] = {
 	0x0a, 0x0d, 0x0f
 };
@@ -353,6 +386,11 @@ static hda_nid_t stac92hd73xx_pin_nids[13] = {
 	0x14, 0x1e, 0x22
 };
 
+static hda_nid_t stac92hd83xxx_pin_nids[14] = {
+	0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
+	0x0f, 0x10, 0x11, 0x12, 0x13,
+	0x1d, 0x1e, 0x1f, 0x20
+};
 static hda_nid_t stac92hd71bxx_pin_nids[10] = {
 	0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
 	0x0f, 0x14, 0x18, 0x19, 0x1e,
@@ -631,6 +669,19 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = {
 	{}
 };
 
+static struct hda_verb stac92hd83xxx_core_init[] = {
+	/* start of config #1 */
+	{ 0xe, AC_VERB_SET_CONNECT_SEL, 0x3},
+
+	/* start of config #2 */
+	{ 0xa, AC_VERB_SET_CONNECT_SEL, 0x0},
+	{ 0xb, AC_VERB_SET_CONNECT_SEL, 0x0},
+	{ 0xd, AC_VERB_SET_CONNECT_SEL, 0x1},
+
+	/* power state controls amps */
+	{ 0x01, AC_VERB_SET_EAPD, 1 << 2},
+};
+
 static struct hda_verb stac92hd71bxx_core_init[] = {
 	/* set master volume and direct control */
 	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
@@ -701,6 +752,8 @@ static struct hda_verb stac927x_core_init[] = {
 static struct hda_verb stac9205_core_init[] = {
 	/* set master volume and direct control */	
 	{ 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+	/* enable analog pc beep path */
+	{ 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5},
 	{}
 };
 
@@ -823,6 +876,33 @@ static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = {
 	{ } /* end */
 };
 
+
+static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_OUTPUT),
+
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_OUTPUT),
+
+	HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x1b, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x1b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("DAC1 Capture Switch", 0x1b, 0x1, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Front Mic Capture Volume", 0x1b, 0x2, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Capture Switch", 0x1b, 0x2, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Line In Capture Volume", 0x1b, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("Line In Capture Switch", 0x1b, 0x3, HDA_INPUT),
+
+	/*
+	HDA_CODEC_VOLUME("Mic Capture Volume", 0x1b, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Capture Switch", 0x1b 0x4, HDA_INPUT),
+	*/
+	{ } /* end */
+};
+
 static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
 	STAC_INPUT_SOURCE(2),
 
@@ -1333,6 +1413,27 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 	{} /* terminator */
 };
 
+static unsigned int ref92hd83xxx_pin_configs[14] = {
+	0x02214030, 0x02211010, 0x02a19020, 0x02170130,
+	0x01014050, 0x01819040, 0x01014020, 0x90a3014e,
+	0x40f000f0, 0x40f000f0, 0x40f000f0, 0x40f000f0,
+	0x01451160, 0x98560170,
+};
+
+static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
+	[STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs,
+};
+
+static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
+	[STAC_92HD83XXX_REF] = "ref",
+};
+
+static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
+	/* SigmaTel reference board */
+	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
+		      "DFI LanParty", STAC_92HD71BXX_REF),
+};
+
 static unsigned int ref92hd71bxx_pin_configs[10] = {
 	0x02214030, 0x02a19040, 0x01a19020, 0x01014010,
 	0x0181302e, 0x01114010, 0x01019020, 0x90a000f0,
@@ -2587,8 +2688,8 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec,
 }
 
 /* labels for mono mux outputs */
-static const char *stac92xx_mono_labels[3] = {
-	"DAC0", "DAC1", "Mixer"
+static const char *stac92xx_mono_labels[4] = {
+	"DAC0", "DAC1", "Mixer", "DAC2"
 };
 
 /* create mono mux for mono out on capable codecs */
@@ -2692,16 +2793,19 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 			}
 		continue;
 found:
-		wcaps = get_wcaps(codec, nid);
+		wcaps = get_wcaps(codec, nid) &
+			(AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP);
 
-		if (wcaps & AC_WCAP_OUT_AMP) {
+		if (wcaps) {
 			sprintf(name, "%s Capture Volume",
 				stac92xx_dmic_labels[dimux->num_items]);
 
 			err = stac92xx_add_control(spec,
 				STAC_CTL_WIDGET_VOL,
 				name,
-				HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
+				HDA_COMPOSE_AMP_VAL(nid, 3, 0,
+				(wcaps & AC_WCAP_OUT_AMP) ?
+				HDA_OUTPUT : HDA_INPUT));
 			if (err < 0)
 				return err;
 		}
@@ -2825,8 +2929,8 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 		hp_speaker_swap = 1;
 	}
 	if (spec->autocfg.mono_out_pin) {
-		int dir = (get_wcaps(codec, spec->autocfg.mono_out_pin)
-				& AC_WCAP_OUT_AMP) ? HDA_OUTPUT : HDA_INPUT;
+		int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) &
+			(AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP);
 		u32 caps = query_amp_caps(codec,
 				spec->autocfg.mono_out_pin, dir);
 		hda_nid_t conn_list[1];
@@ -2848,21 +2952,26 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 						!(wcaps & AC_WCAP_LR_SWAP))
 					spec->mono_nid = conn_list[0];
 		}
-		/* all mono outs have a least a mute/unmute switch */
-		err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE,
-			"Mono Playback Switch",
-			HDA_COMPOSE_AMP_VAL(spec->autocfg.mono_out_pin,
-					1, 0, dir));
-		if (err < 0)
-			return err;
-		/* check to see if there is volume support for the amp */
-		if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) {
-			err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL,
-				"Mono Playback Volume",
-				HDA_COMPOSE_AMP_VAL(spec->autocfg.mono_out_pin,
-					1, 0, dir));
+		if (dir) {
+			hda_nid_t nid = spec->autocfg.mono_out_pin;
+
+			/* most mono outs have a least a mute/unmute switch */
+			dir = (dir & AC_WCAP_OUT_AMP) ? HDA_OUTPUT : HDA_INPUT;
+			err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE,
+				"Mono Playback Switch",
+				HDA_COMPOSE_AMP_VAL(nid, 1, 0, dir));
 			if (err < 0)
 				return err;
+			/* check for volume support for the amp */
+			if ((caps & AC_AMPCAP_NUM_STEPS)
+					>> AC_AMPCAP_NUM_STEPS_SHIFT) {
+				err = stac92xx_add_control(spec,
+					STAC_CTL_WIDGET_VOL,
+					"Mono Playback Volume",
+				HDA_COMPOSE_AMP_VAL(nid, 1, 0, dir));
+				if (err < 0)
+					return err;
+			}
 		}
 
 		stac92xx_auto_set_pinctl(codec, spec->autocfg.mono_out_pin,
@@ -2942,7 +3051,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 
 	if (spec->autocfg.dig_out_pin)
 		spec->multiout.dig_out_nid = dig_out;
-	if (spec->autocfg.dig_in_pin)
+	if (dig_in && spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = dig_in;
 
 	if (spec->kctl_alloc)
@@ -3338,7 +3447,12 @@ static void stac92xx_pin_sense(struct hda_codec *codec, int idx)
 	val = snd_hda_codec_read(codec, codec->afg, 0, 0x0fec, 0x0)
 							& 0x000000ff;
 	presence = get_hp_pin_presence(codec, nid);
-	idx = 1 << idx;
+
+	/* several codecs have two power down bits */
+	if (spec->pwr_mapping)
+		idx = spec->pwr_mapping[idx];
+	else
+		idx = 1 << idx;
 
 	if (presence)
 		val &= ~idx;
@@ -3674,6 +3788,94 @@ again:
 	return 0;
 }
 
+static struct hda_input_mux stac92hd83xxx_dmux = {
+	.num_items = 3,
+	.items = {
+		{ "Analog Inputs", 0x03 },
+		{ "Digital Mic 1", 0x04 },
+		{ "Digital Mic 2", 0x05 },
+	}
+};
+
+static int patch_stac92hd83xxx(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec;
+	int err;
+
+	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (spec == NULL)
+		return -ENOMEM;
+
+	codec->spec = spec;
+	spec->mono_nid = 0x19;
+	spec->digbeep_nid = 0x21;
+	spec->dmic_nids = stac92hd83xxx_dmic_nids;
+	spec->dmux_nids = stac92hd83xxx_dmux_nids;
+	spec->adc_nids = stac92hd83xxx_adc_nids;
+	spec->pwr_nids = stac92hd83xxx_pwr_nids;
+	spec->pwr_mapping = stac92hd83xxx_pwr_mapping;
+	spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids);
+	spec->multiout.dac_nids = stac92hd83xxx_dac_nids;
+
+	spec->init = stac92hd83xxx_core_init;
+	switch (codec->vendor_id) {
+	case 0x111d7605:
+		spec->multiout.num_dacs = STAC92HD81_DAC_COUNT;
+		break;
+	default:
+		spec->num_pwrs--;
+		spec->init++; /* switch to config #2 */
+		spec->multiout.num_dacs = STAC92HD83_DAC_COUNT;
+	}
+
+	spec->mixer = stac92hd83xxx_mixer;
+	spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids);
+	spec->num_dmuxes = ARRAY_SIZE(stac92hd83xxx_dmux_nids);
+	spec->num_adcs = ARRAY_SIZE(stac92hd83xxx_adc_nids);
+	spec->num_dmics = STAC92HD83XXX_NUM_DMICS;
+	spec->dinput_mux = &stac92hd83xxx_dmux;
+	spec->pin_nids = stac92hd83xxx_pin_nids;
+	spec->board_config = snd_hda_check_board_config(codec,
+							STAC_92HD83XXX_MODELS,
+							stac92hd83xxx_models,
+							stac92hd83xxx_cfg_tbl);
+again:
+	if (spec->board_config < 0) {
+		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
+			" STAC92HD83XXX, using BIOS defaults\n");
+		err = stac92xx_save_bios_config_regs(codec);
+		if (err < 0) {
+			stac92xx_free(codec);
+			return err;
+		}
+		spec->pin_configs = spec->bios_pin_configs;
+	} else {
+		spec->pin_configs = stac92hd83xxx_brd_tbl[spec->board_config];
+		stac92xx_set_config_regs(codec);
+	}
+
+	err = stac92xx_parse_auto_config(codec, 0x1d, 0);
+	if (!err) {
+		if (spec->board_config < 0) {
+			printk(KERN_WARNING "hda_codec: No auto-config is "
+			       "available, default to model=ref\n");
+			spec->board_config = STAC_92HD83XXX_REF;
+			goto again;
+		}
+		err = -EINVAL;
+	}
+
+	if (err < 0) {
+		stac92xx_free(codec);
+		return err;
+	}
+
+	codec->patch_ops = stac92xx_patch_ops;
+
+	return 0;
+}
+
+
 static int patch_stac92hd71bxx(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
@@ -4395,6 +4597,8 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = {
  	{ .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 },
  	{ .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 },
 	{ .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx},
+	{ .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx},
+	{ .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx},
 	{ .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx },
-- 
cgit v1.2.3-18-g5258


From 0b18cb1854152a62492aae088cb80cbeb5c0288d Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Mon, 28 Jul 2008 17:07:07 +0200
Subject: ALSA: Fix commit: Add automatic model setting for the Acer Aspire
 5920G laptop

There is a whitespace at the end of added line. Remove it.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 66025161bd6..38017a129ba 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7942,7 +7942,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE),
-	SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE), 
+	SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */
 	SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
 	SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
-- 
cgit v1.2.3-18-g5258


From e76d8ceaaff9d7fc1ba2b1963a9f34151832223b Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 28 Jul 2008 19:05:35 +0100
Subject: ALSA: Add jack reporting API

Currently very few systems provide information about jack status to user
space, even though many have hardware facilities to do detection. Those
systems that do use an input device with the existing SW_HEADPHONE_INSERT
switch type to do so, often independently of ALSA.

This patch introduces a standard method for representing jacks to user
space into ALSA. It allows drivers to register jacks for a sound card with
the input subsystem, binding the input device to the card to help user
space associate the input devices with their sound cards. The created
input devices are named in the form "card longname jack" where jack is
provided by the driver when allocating a jack. By default the parent for
the input device is the sound card but this can be overridden by the
card driver.

The existing user space API with SW_HEADPHONE_INSERT is preserved.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/core/jack.c | 163 ++++++++++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 163 insertions(+)
 create mode 100644 sound/core/jack.c

(limited to 'sound')

diff --git a/sound/core/jack.c b/sound/core/jack.c
new file mode 100644
index 00000000000..8133a2b173a
--- /dev/null
+++ b/sound/core/jack.c
@@ -0,0 +1,163 @@
+/*
+ *  Jack abstraction layer
+ *
+ *  Copyright 2008 Wolfson Microelectronics
+ *
+ *   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.
+ *
+ *   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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#include <linux/input.h>
+#include <sound/jack.h>
+#include <sound/core.h>
+
+static int snd_jack_dev_free(struct snd_device *device)
+{
+	struct snd_jack *jack = device->device_data;
+
+	/* If the input device is registered with the input subsystem
+	 * then we need to use a different deallocator. */
+	if (jack->registered)
+		input_unregister_device(jack->input_dev);
+	else
+		input_free_device(jack->input_dev);
+
+	kfree(jack);
+
+	return 0;
+}
+
+static int snd_jack_dev_register(struct snd_device *device)
+{
+	struct snd_jack *jack = device->device_data;
+	struct snd_card *card = device->card;
+	int err;
+
+	snprintf(jack->name, sizeof(jack->name), "%s %s",
+		 card->longname, jack->id);
+	jack->input_dev->name = jack->name;
+
+	/* Default to the sound card device. */
+	if (!jack->input_dev->dev.parent)
+		jack->input_dev->dev.parent = card->dev;
+
+	err = input_register_device(jack->input_dev);
+	if (err == 0)
+		jack->registered = 1;
+
+	return err;
+}
+
+/**
+ * snd_jack_new - Create a new jack
+ * @card:  the card instance
+ * @id:    an identifying string for this jack
+ * @type:  a bitmask of enum snd_jack_type values that can be detected by
+ *         this jack
+ * @jjack: Used to provide the allocated jack object to the caller.
+ *
+ * Creates a new jack object.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ * On success jjack will be initialised.
+ */
+int snd_jack_new(struct snd_card *card, const char *id, int type,
+		 struct snd_jack **jjack)
+{
+	struct snd_jack *jack;
+	int err;
+	static struct snd_device_ops ops = {
+		.dev_free = snd_jack_dev_free,
+		.dev_register = snd_jack_dev_register,
+	};
+
+	jack = kzalloc(sizeof(struct snd_jack), GFP_KERNEL);
+	if (jack == NULL)
+		return -ENOMEM;
+
+	jack->id = id;
+
+	jack->input_dev = input_allocate_device();
+	if (jack->input_dev == NULL) {
+		err = -ENOMEM;
+		goto fail_input;
+	}
+
+	jack->input_dev->phys = "ALSA";
+
+	jack->type = type;
+
+	if (type & SND_JACK_HEADPHONE)
+		input_set_capability(jack->input_dev, EV_SW,
+				     SW_HEADPHONE_INSERT);
+	if (type & SND_JACK_MICROPHONE)
+		input_set_capability(jack->input_dev, EV_SW,
+				     SW_MICROPHONE_INSERT);
+
+	err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops);
+	if (err < 0)
+		goto fail_input;
+
+	*jjack = jack;
+
+	return 0;
+
+fail_input:
+	input_free_device(jack->input_dev);
+	kfree(jack);
+	return err;
+}
+EXPORT_SYMBOL(snd_jack_new);
+
+/**
+ * snd_jack_set_parent - Set the parent device for a jack
+ *
+ * @jack:   The jack to configure
+ * @parent: The device to set as parent for the jack.
+ *
+ * Set the parent for the jack input device in the device tree.  This
+ * function is only valid prior to registration of the jack.  If no
+ * parent is configured then the parent device will be the sound card.
+ */
+void snd_jack_set_parent(struct snd_jack *jack, struct device *parent)
+{
+	WARN_ON(jack->registered);
+
+	jack->input_dev->dev.parent = parent;
+}
+EXPORT_SYMBOL(snd_jack_set_parent);
+
+/**
+ * snd_jack_report - Report the current status of a jack
+ *
+ * @jack:   The jack to report status for
+ * @status: The current status of the jack
+ */
+void snd_jack_report(struct snd_jack *jack, int status)
+{
+	if (jack->type & SND_JACK_HEADPHONE)
+		input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT,
+				    status & SND_JACK_HEADPHONE);
+	if (jack->type & SND_JACK_MICROPHONE)
+		input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT,
+				    status & SND_JACK_MICROPHONE);
+
+	input_sync(jack->input_dev);
+}
+EXPORT_SYMBOL(snd_jack_report);
+
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("Jack detection support for ALSA");
+MODULE_LICENSE("GPL");
-- 
cgit v1.2.3-18-g5258


From 0d94e41abe271c86df06bcf72d24f9ca7ce771f0 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 28 Jul 2008 19:05:36 +0100
Subject: ALSA: Build jack detection

Since jack detection requires the input subsystem which may not be
desired on small systems it is not built unless required by a driver
that is being built. Drivers using jack detection should use a pattern
like this:

config SND_FOO
        tristate "..."
        ...
        select SND_JACK if INPUT=y || INPUT=SND

to ensure that the jack detection API is enabled if the input subsystem
is.  If the input subsystem is not enabled then a stub version of the
API is provided.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/core/Kconfig  | 6 ++++++
 sound/core/Makefile | 1 +
 2 files changed, 7 insertions(+)

(limited to 'sound')

diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index 335d45ecde6..9c4da1cd4a6 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -12,6 +12,12 @@ config SND_HWDEP
 config SND_RAWMIDI
 	tristate
 
+# To be effective this also requires INPUT - users should say:
+#    select SND_JACK if INPUT=y || INPUT=SND
+# to avoid having to force INPUT on.
+config SND_JACK
+	bool
+
 config SND_SEQUENCER
 	tristate "Sequencer support"
 	select SND_TIMER
diff --git a/sound/core/Makefile b/sound/core/Makefile
index da8e685eef9..d57125a5687 100644
--- a/sound/core/Makefile
+++ b/sound/core/Makefile
@@ -7,6 +7,7 @@ snd-y     := sound.o init.o memory.o info.o control.o misc.o device.o
 snd-$(CONFIG_ISA_DMA_API) += isadma.o
 snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o info_oss.o
 snd-$(CONFIG_SND_VMASTER) += vmaster.o
+snd-$(CONFIG_SND_JACK)	  += jack.o
 
 snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \
 		pcm_memory.o
-- 
cgit v1.2.3-18-g5258


From f6154d6d007c69a330acc9021ec77cca5da9dc4c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 29 Jul 2008 12:08:16 +0200
Subject: ALSA: hda - use input_free_device()

Use input_free_devce() correctly instead of kfree() at error path.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/pci/hda/hda_beep.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 5a764c48139..9b77b3e0fa9 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -103,7 +103,7 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
 
 	err = input_register_device(input_dev);
 	if (err < 0) {
-		kfree(input_dev);
+		input_free_device(input_dev);
 		kfree(beep);
 		return err;
 	}
-- 
cgit v1.2.3-18-g5258