Add Freescale ASRC driver

The Asynchronous Sample Rate Converter (ASRC) converts the sampling rate of a signal associated with an input clock into a signal associated with a different output clock. The driver currently works as a Front End of DPCM with other Back Ends DAI links such as ESAI<->CS42888 and SSI<->WM8962 and SAI. It converts the original sample rate to a common rate supported by Back Ends for playback while converts the common rate of Back Ends to a desired rate for capture. It has 3 pairs to support three different substreams within totally 10 channels.

Documentation/devicetree/bindings/sound/fsl,asrc.txt new file +20 -0

+Required properties:
+
+  - compatible : Compatible list, must contain "fsl,imx35-asrc" or
+    "fsl,imx53-asrc".
+
+  - reg : Offset and length of the register set for the device.
+
+  - interrupts : Contains the spdif interrupt.
+
+  - dmas : Generic dma devicetree binding as described in
+	   Documentation/devicetree/bindings/dma/dma.txt.
+
+  - dma-names : Six dmas have to be defined: "rxa", "rxb", "rxc", "txa", "txb",
+		and "txc".
+
+  - clocks : Contains an entry for each entry in clock-names.
+
+  - clock-names : Includes the following entries:
+	"mem"		Peripheral access clock to access registers.
+	"ipg"		Peripheral clock to driver module.

sound/soc/fsl/fsl_asrc.c new file +17 -0

+
+static const struct platform_device_id fsl_asrc_devtype[] = {
+	{
+		.name = "imx35-asrc",
+		.driver_data = IMX35_ASRC,
+	}, {
+		.name = "imx53-asrc",
+		.driver_data = IMX53_ASRC,
+	}, {
+		/* sentinel */
+	}
+};
+MODULE_DEVICE_TABLE(platform, fsl_asrc_devtype);
+
+static const struct of_device_id fsl_asrc_ids[] = {
+	{
+		.compatible = "fsl,imx35-asrc",

sound/soc/fsl/fsl_asrc.c new file +14 -0

+static irqreturn_t fsl_asrc_isr(int irq, void *dev_id)
+{
+	struct fsl_asrc *asrc_priv = (struct fsl_asrc *)dev_id;
+	struct device *dev = &asrc_priv->pdev->dev;
+	enum asrc_pair_index index;
+	u32 status;
+
+	regmap_read(asrc_priv->regmap, REG_ASRSTR, &status);
+
+	/* Clean overload error */
+	regmap_write(asrc_priv->regmap, REG_ASRSTR, ASRSTR_AOLE);
+
+	/*
+	 * We here use dev_dbg() for all exceptions because ASRC itself does

sound/soc/fsl/fsl_asrc.c new file +6 -0

+		}
+	}
+
+	return IRQ_HANDLED;
+}
+

sound/soc/fsl/fsl_asrc.c new file +13 -0

+	struct fsl_asrc *asrc_priv = pair->asrc_priv;
+	enum asrc_pair_index index = pair->index;
+
+	regmap_update_bits(asrc_priv->regmap, REG_ASRMCR(index),
+			   ASRMCRi_EXTTHRSHi_MASK |
+			   ASRMCRi_INFIFO_THRESHOLD_MASK |
+			   ASRMCRi_OUTFIFO_THRESHOLD_MASK,
+			   ASRMCRi_EXTTHRSHi |
+			   ASRMCRi_INFIFO_THRESHOLD(in) |
+			   ASRMCRi_OUTFIFO_THRESHOLD(out));
+
+	return 0;
+}

Quoted 8 lines · Nicolin Chen

The Asynchronous Sample Rate Converter (ASRC) converts the sampling rate of a signal associated with an input clock into a signal associated with a different output clock. The driver currently works as a Front End of DPCM with other Back Ends DAI links such as ESAI<->CS42888 and SSI<->WM8962 and SAI. It converts the original sample rate to a common rate supported by Back Ends for playback while converts the common rate of Back Ends to a desired rate for capture. It has 3 pairs to support three different substreams within totally 10 channels.

(...)

Quoted 20 lines

Documentation/devicetree/bindings/sound/fsl,asrc.txt new file +20 -0

+Required properties:
+
+  - compatible : Compatible list, must contain "fsl,imx35-asrc" or
+    "fsl,imx53-asrc".
+
+  - reg : Offset and length of the register set for the device.
+
+  - interrupts : Contains the spdif interrupt.
+
+  - dmas : Generic dma devicetree binding as described in
+	   Documentation/devicetree/bindings/dma/dma.txt.
+
+  - dma-names : Six dmas have to be defined: "rxa", "rxb", "rxc", "txa", "txb",
+		and "txc".
+
+  - clocks : Contains an entry for each entry in clock-names.
+
+  - clock-names : Includes the following entries:
+	"mem"		Peripheral access clock to access registers.
+	"ipg"		Peripheral clock to driver module.

Documentation/devicetree/bindings/sound/fsl,asrc.txt line 38

Quoted 10 lines

Documentation/devicetree/bindings/sound/fsl,asrc.txt new file +10 -0

+	"asrck_<0-f>"	Clock sources for input and output clock.
+
+   - big-endian : If this property is absent, the native endian mode will
+		  be in use as default, or the big endian mode will be in use
+		  for all the device registers.
+
+   - fsl,asrc-rate : Defines a mutual sample rate used by Back End DAI link.
+
+   - fsl,asrc-width : Defines a mutual sample width used by Back End DAI link.
+

indentation for the properties....

- compatible : Should be "fsl,imx35-asrc" or "fsl,imx53-asrc".
- reg : Offset and length of the register set for the device.
....

(...)

Quoted 17 lines

sound/soc/fsl/fsl_asrc.c new file +17 -0

+
+static const struct platform_device_id fsl_asrc_devtype[] = {
+	{
+		.name = "imx35-asrc",
+		.driver_data = IMX35_ASRC,
+	}, {
+		.name = "imx53-asrc",
+		.driver_data = IMX53_ASRC,
+	}, {
+		/* sentinel */
+	}
+};
+MODULE_DEVICE_TABLE(platform, fsl_asrc_devtype);
+
+static const struct of_device_id fsl_asrc_ids[] = {
+	{
+		.compatible = "fsl,imx35-asrc",

sound/soc/fsl/fsl_asrc.c line 116

Quoted 10 lines

sound/soc/fsl/fsl_asrc.c new file +10 -0

+		.data = &fsl_asrc_devtype[IMX35_ASRC],
+	}, {
+		.compatible = "fsl,imx53-asrc",
+		.data = &fsl_asrc_devtype[IMX53_ASRC],
+	}, {
+		/* sentinel */
+	}
+};
+MODULE_DEVICE_TABLE(of, fsl_asrc_ids);
+

move these ids after probe/remove... every driver follows same thing...

Quoted 14 lines

sound/soc/fsl/fsl_asrc.c new file +14 -0

+static irqreturn_t fsl_asrc_isr(int irq, void *dev_id)
+{
+	struct fsl_asrc *asrc_priv = (struct fsl_asrc *)dev_id;
+	struct device *dev = &asrc_priv->pdev->dev;
+	enum asrc_pair_index index;
+	u32 status;
+
+	regmap_read(asrc_priv->regmap, REG_ASRSTR, &status);
+
+	/* Clean overload error */
+	regmap_write(asrc_priv->regmap, REG_ASRSTR, ASRSTR_AOLE);
+
+	/*
+	 * We here use dev_dbg() for all exceptions because ASRC itself does

sound/soc/fsl/fsl_asrc.c line 140

Quoted 10 lines

sound/soc/fsl/fsl_asrc.c new file +10 -0

+	 * not care if FIFO overflowed or underrun while a warning in the
+	 * interrupt would result a ridged conversion.
+	 */
+	for (index = ASRC_PAIR_A; index < ASRC_PAIR_MAX_NUM; index++) {
+		if (!asrc_priv->pair[index])
+			continue;
+
+		if (status & ASRSTR_ATQOL) {
+			asrc_priv->pair[index]->error |= ASRC_TASK_Q_OVERLOAD;
+			dev_dbg(dev, "ASRC Task Queue FIFO overload");

missed terminating new line (\n)...

sound/soc/fsl/fsl_asrc.c line 145

Quoted 5 lines

sound/soc/fsl/fsl_asrc.c new file +5 -0

+		}
+
+		if (status & ASRSTR_AOOL(index)) {
+			asrc_priv->pair[index]->error |= ASRC_OUTPUT_TASK_OVERLOAD;
+			pair_dbg("Output Task Overload");

same

sound/soc/fsl/fsl_asrc.c line 150

Quoted 5 lines

sound/soc/fsl/fsl_asrc.c new file +5 -0

+		}
+
+		if (status & ASRSTR_AIOL(index)) {
+			asrc_priv->pair[index]->error |= ASRC_INPUT_TASK_OVERLOAD;
+			pair_dbg("Input Task Overload");

same

sound/soc/fsl/fsl_asrc.c line 155

Quoted 5 lines

sound/soc/fsl/fsl_asrc.c new file +5 -0

+		}
+
+		if (status & ASRSTR_AODO(index)) {
+			asrc_priv->pair[index]->error |= ASRC_OUTPUT_BUFFER_OVERFLOW;
+			pair_dbg("Output Data Buffer has overflowed");

same

sound/soc/fsl/fsl_asrc.c line 160

Quoted 5 lines

sound/soc/fsl/fsl_asrc.c new file +5 -0

+		}
+
+		if (status & ASRSTR_AIDU(index)) {
+			asrc_priv->pair[index]->error |= ASRC_INPUT_BUFFER_UNDERRUN;
+			pair_dbg("Input Data Buffer has underflowed");

same..

Quoted 6 lines

sound/soc/fsl/fsl_asrc.c new file +6 -0

+		}
+	}
+
+	return IRQ_HANDLED;
+}
+

(...)

sound/soc/fsl/fsl_asrc.c line 240

Quoted 5 lines

sound/soc/fsl/fsl_asrc.c new file +5 -0

+/**
+ * Configure input and output thresholds
+ */
+static int fsl_asrc_set_watermarks(struct fsl_asrc_pair *pair, u32 in, u32 out)
+{

static void for function..?

Quoted 13 lines

sound/soc/fsl/fsl_asrc.c new file +13 -0

+	struct fsl_asrc *asrc_priv = pair->asrc_priv;
+	enum asrc_pair_index index = pair->index;
+
+	regmap_update_bits(asrc_priv->regmap, REG_ASRMCR(index),
+			   ASRMCRi_EXTTHRSHi_MASK |
+			   ASRMCRi_INFIFO_THRESHOLD_MASK |
+			   ASRMCRi_OUTFIFO_THRESHOLD_MASK,
+			   ASRMCRi_EXTTHRSHi |
+			   ASRMCRi_INFIFO_THRESHOLD(in) |
+			   ASRMCRi_OUTFIFO_THRESHOLD(out));
+
+	return 0;
+}

(...)

sound/soc/fsl/fsl_asrc.c line 1016

Quoted 5 lines

sound/soc/fsl/fsl_asrc.c new file +5 -0

+static const struct dev_pm_ops fsl_asrc_pm = {
+	SET_RUNTIME_PM_OPS(fsl_asrc_runtime_suspend, fsl_asrc_runtime_resume, NULL)
+	SET_SYSTEM_SLEEP_PM_OPS(fsl_asrc_suspend, fsl_asrc_resume)
+};
+

move device ids to here...

sound/soc/fsl/fsl_asrc.c line 1025

Quoted 10 lines

sound/soc/fsl/fsl_asrc.c new file +9 -0

+static struct platform_driver fsl_asrc_driver = {
+	.probe = fsl_asrc_probe,
+	.driver = {
+		.name = "fsl-asrc",
+		.of_match_table = fsl_asrc_ids,
+		.pm = &fsl_asrc_pm,
+	},
+};
+module_platform_driver(fsl_asrc_driver);

Thanks...

Signature · 2 lines

Regards,
Varka Bhadram.

On 07/25/2014 09:33 AM, Nicolin Chen wrote:
(...)

Quoted 21 lines

sound/soc/fsl/fsl_asrc.c new file +21 -0

+
+static const struct platform_device_id fsl_asrc_devtype[] = {
+	{
+		.name = "imx35-asrc",
+		.driver_data = IMX35_ASRC,
+	}, {
+		.name = "imx53-asrc",
+		.driver_data = IMX53_ASRC,
+	}, {
+		/* sentinel */
+	}
+};
+MODULE_DEVICE_TABLE(platform, fsl_asrc_devtype);
+
+static const struct of_device_id fsl_asrc_ids[] = {
+	{
+		.compatible = "fsl,imx35-asrc",
+		.data = &fsl_asrc_devtype[IMX35_ASRC],
+	}, {
+		.compatible = "fsl,imx53-asrc",
+		.data = &fsl_asrc_devtype[IMX53_ASRC],

sound/soc/fsl/fsl_asrc.c line 116

Quoted 6 lines

sound/soc/fsl/fsl_asrc.c new file +6 -0

+	}, {
+		/* sentinel */
+	}
+};
+MODULE_DEVICE_TABLE(of, fsl_asrc_ids);
+

move these ids after probe/remove... every driver follows same thing...

Hi Nicolin,

Quoted 36 lines · Nicolin Chen

Hi Varka,

Regarding a point you suggested.

Quoted 23 lines · Varka Bhadram

On 07/25/2014 09:33 AM, Nicolin Chen wrote:
(...)

Quoted 20 lines

sound/soc/fsl/fsl_asrc.c new file +20 -0

+
+static const struct platform_device_id fsl_asrc_devtype[] = {
+	{
+		.name = "imx35-asrc",
+		.driver_data = IMX35_ASRC,
+	}, {
+		.name = "imx53-asrc",
+		.driver_data = IMX53_ASRC,
+	}, {
+		/* sentinel */
+	}
+};
+MODULE_DEVICE_TABLE(platform, fsl_asrc_devtype);
+
+static const struct of_device_id fsl_asrc_ids[] = {
+	{
+		.compatible = "fsl,imx35-asrc",
+		.data = &fsl_asrc_devtype[IMX35_ASRC],
+	}, {
+		.compatible = "fsl,imx53-asrc",

Quoted 8 lines

sound/soc/fsl/fsl_asrc.c line 116

Quoted 7 lines

sound/soc/fsl/fsl_asrc.c new file +7 -0

+		.data = &fsl_asrc_devtype[IMX53_ASRC],
+	}, {
+		/* sentinel */
+	}
+};
+MODULE_DEVICE_TABLE(of, fsl_asrc_ids);
+

move these ids after probe/remove... every driver follows same thing...

Quoted 5 lines

Hmm.. fsl_asrc_ids is called in probe(), so it's probably not a good choice to put them after probe/remove. And actually not every driver does so. For example drivers/i2c/busses/i2c-s3c2410.c

I think it should be okay to put here if it contains data.

You are using these ids in the probe .Tt should be above the probe() so we can see of_match_device() directly...
and also one more thing that the interrupt handler also would be immediately above the probe()

ex:
static irqreturn_t fsl_asrc_isr(int irq, void *dev_id) {
...
}

device ids..

your_probe()
{
...
}

your_remove()
{
...
}

Now every driver that is coming following this style...

If your not using device ids in probe then it would be like

static irqreturn_t fsl_asrc_isr(int irq, void *dev_id) {
...
}

your_probe()
{
...
}

your_remove()
{
...
}

device ids..

Thanks...

Signature · 2 lines

Regards,
Varka Bhadram.

Hi Varka,

Regarding a point you suggested.

Quoted 23 lines · Varka Bhadram

On 07/25/2014 09:33 AM, Nicolin Chen wrote:
(...)

Quoted 20 lines

sound/soc/fsl/fsl_asrc.c new file +20 -0

+
+static const struct platform_device_id fsl_asrc_devtype[] = {
+	{
+		.name = "imx35-asrc",
+		.driver_data = IMX35_ASRC,
+	}, {
+		.name = "imx53-asrc",
+		.driver_data = IMX53_ASRC,
+	}, {
+		/* sentinel */
+	}
+};
+MODULE_DEVICE_TABLE(platform, fsl_asrc_devtype);
+
+static const struct of_device_id fsl_asrc_ids[] = {
+	{
+		.compatible = "fsl,imx35-asrc",
+		.data = &fsl_asrc_devtype[IMX35_ASRC],
+	}, {
+		.compatible = "fsl,imx53-asrc",

Quoted 8 lines

sound/soc/fsl/fsl_asrc.c line 116

Quoted 7 lines

sound/soc/fsl/fsl_asrc.c new file +7 -0

+		.data = &fsl_asrc_devtype[IMX53_ASRC],
+	}, {
+		/* sentinel */
+	}
+};
+MODULE_DEVICE_TABLE(of, fsl_asrc_ids);
+

move these ids after probe/remove... every driver follows same thing...

Hmm.. fsl_asrc_ids is called in probe(), so it's probably not a good choice to put them after probe/remove. And actually not every driver does so. For example drivers/i2c/busses/i2c-s3c2410.c

I think it should be okay to put here if it contains data.

What is the best way to test/debug the alc668 driver these days?

I am running kernel 3.14.1 (debian testing/sid).

The laptop is an ASUS N750JK.

Sound works for front left/right speakers but the back speakers and external lfe channel is not working on this laptop.

- I have found a couple of kernel bug reports for other ASUS models with this chipset but they are both marked resolved. Should I open a new ticket ?

https://bugzilla.kernel.org/show_bug.cgi?id=65091
https://bugzilla.kernel.org/show_bug.cgi?id=66271

- I can hear static through the front speakers (with pulseaudio test) if I set pins 0x19 and 0x1a to Line Out (Center/LFE) and Internal (Center/LFE) or vice versa with hdajackretask.

Pin 0x16 does not appear to work at all.

I see this output when I apply the config in hdajackretask.

0x12 0x90a60140
0x14 0x99130110
0x15 0x04211020
0x16 0x99134111
0x18 0x01a19840
0x19 0x01014011
0x1a 0x411111f0
0x1b 0x411111f0
0x1d 0x40c6852d
0x1e 0x014b1180
0x1f 0x411111f0
1

--
Patrick Shirkey
Boost Hardware Ltd
_______________________________________________
Alsa-devel mailing list
[email protected]
http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

Quoted 9 lines

What is the best way to test/debug the alc668 driver these days?

I am running kernel 3.14.1 (debian testing/sid).

The laptop is an ASUS N750JK.

Sound works for front left/right speakers but the back speakers and external lfe channel is not working on this laptop.

You have to post output of alsa-info.sh

Do the BIOS setup all the pin default ?

How many audio jacks ?

Do it support headset or headphone ?

Do you mean there are four internal speakers ?

Do it support surround40 in windows ?

The external sonic master and headphone need two volume controls but alc668 only have three

The best way to test is using latest git version and hda jack retask to

1) four internal speaker and headphone
2) two internal speaker, headphone and external subwoofer

Once you confirm the nodes of the speakers, subwooder
You need to decide whether headphone share the volume controls with the speaker/subwoofer

Quoted 1 line

Quoted 18 lines

- I have found a couple of kernel bug reports for other ASUS models with
this chipset but they are both marked resolved. Should I open a new ticket
?

https://bugzilla.kernel.org/show_bug.cgi?id=65091
https://bugzilla.kernel.org/show_bug.cgi?id=66271

- I can hear static through the front speakers (with pulseaudio test) if
I set pins 0x19 and 0x1a to Line Out (Center/LFE) and Internal (Center/LFE) or vice versa with hdajackretask.

Pin 0x16 does not appear to work at all.

I see this output when I apply the config in hdajackretask.

Quoted 4 lines

0x12 0x90a60140
0x14 0x99130110
0x15 0x04211020
0x16 0x99134111

Quoted 10 lines

0x18 0x01a19840
0x19 0x01014011
0x1a 0x411111f0
0x1b 0x411111f0
0x1d 0x40c6852d
0x1e 0x014b1180
0x1f 0x411111f0
1

Quoted 7 lines

Quoted 7 lines

What is the best way to test/debug the alc668 driver these days?

I am running kernel 3.14.1 (debian testing/sid).

The laptop is an ASUS N750JK.

Quoted 5 lines

Quoted 2 lines

Sound works for front left/right speakers but the back speakers and external lfe channel is not working on this laptop.

You have to post output of alsa-info.sh

http://boosthardware.com/alsa-info-alc668-n750jk.txt

Quoted 3 lines

How many audio jacks ?

Standard headphone/mic + yellow external lfe (sonic master). There is also hdmi but I am not testing that at the moment.

Quoted 5 lines

Do it support headset or headphone ?

Do you mean there are four internal speakers ?

There are two front speakers which produce sound through the front bottom of the case. There are also two back speakers which do not produce sound in my tests so far.

Quoted 2 lines

Do it support surround40 in windows ?

I am not running windows OS on this machine but I suspect it does.

Quoted 5 lines

The external sonic master and headphone need two volume controls but alc668 only have three

The best way to test is using latest git version and hda jack retask to

The process has changed somewhat since I last did this.

Quoted 8 lines

1) four internal speaker and headphone
2) two internal speaker, headphone and external subwoofer

Once you confirm the nodes of the speakers, subwooder
You need to decide whether headphone share the volume controls with the speaker/subwoofer

Quoted 1 line

I have spent some time on this step already with 3.14.1

Is there a specific reason this process is not already scripted other than no one has taken the time to do it?

Can we not create a test script that will run through all the logical combinations and provide all the relevant info?

Quoted 32 lines

Quoted 32 lines

- I have found a couple of kernel bug reports for other ASUS models with
this chipset but they are both marked resolved. Should I open a new ticket
?

https://bugzilla.kernel.org/show_bug.cgi?id=65091
https://bugzilla.kernel.org/show_bug.cgi?id=66271

- I can hear static through the front speakers (with pulseaudio test) if
I set pins 0x19 and 0x1a to Line Out (Center/LFE) and Internal (Center/LFE) or vice versa with hdajackretask.

Pin 0x16 does not appear to work at all.

I see this output when I apply the config in hdajackretask.

0x12 0x90a60140
0x14 0x99130110
0x15 0x04211020
0x16 0x99134111
0x18 0x01a19840
0x19 0x01014011
0x1a 0x411111f0
0x1b 0x411111f0
0x1d 0x40c6852d
0x1e 0x014b1180
0x1f 0x411111f0
1

Quoted 10 lines

Quoted 10 lines

Quoted 9 lines

What is the best way to test/debug the alc668 driver these days?

I am running kernel 3.14.1 (debian testing/sid).

The laptop is an ASUS N750JK.

Sound works for front left/right speakers but the back speakers and external lfe channel is not working on this laptop.

Quoted 4 lines

Quoted 2 lines

You have to post output of alsa-info.sh

http://boosthardware.com/alsa-info-alc668-n750jk.txt

Seem BIOS only set up the front speakers

Quoted 4 lines

Quoted 3 lines

How many audio jacks ?

Quoted 3 lines

Standard headphone/mic + yellow external lfe (sonic master). There is also
hdmi but I am not testing that at the moment.

http://www.asus.com/Notebooks_Ultrabooks/N750JK/specifications/

Do you mean headset ?

Quoted 19 lines

Quoted 5 lines

Do it support headset or headphone ?

Do you mean there are four internal speakers ?

There are two front speakers which produce sound through the front bottom
of the case. There are also two back speakers which do not produce sound in my tests so far.

Quoted 2 lines

Do it support surround40 in windows ?

I am not running windows OS on this machine but I suspect it does.

Quoted 3 lines

The external sonic master and headphone need two volume controls but alc668 only have three

Quoted 6 lines

Quoted 1 line

The best way to test is using latest git version and hda jack retask

to

Quoted 1 line

The process has changed somewhat since I last did this.

https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/hda?qt=grep&q=alc668

Quoted 6 lines

Quoted 6 lines

1) four internal speaker and headphone
2) two internal speaker, headphone and external subwoofer

Once you confirm the nodes of the speakers, subwooder You need to decide whether headphone share the volume controls with

Quoted 5 lines

the

Quoted 2 lines

speaker/subwoofer

Quoted 1 line

I have spent some time on this step already with 3.14.1

Did you retask the node of subwoofer as speaker or line out ?

For multi channel, the sequence number is ascending and the driver expect they have same type i.e, all speakers or all line out jacks

Did you find the node of the rear speakers in (1) ?

Did surround 4.0 work when you fix the rear speaker ?

Did you find the node of the external subwoofer in (2) ?

Did the sonic master subwoofer detected ?
hda jack sense test -a

Do the subwoofer only use right channel ?

Do surround 2.1 works ?

Quoted 4 lines

Is there a specific reason this process is not already scripted other than
no one has taken the time to do it?

Do surround 4.1 or 5.1 work when you fix both speaker and subwoofer ?

Do you prefer driver create headphone volume control or subwoofer volume control ?

Quoted 4 lines

Can we not create a test script that will run through all the logical combinations and provide all the relevant info?

The driver hardcore the name of volume control to bass when there are two speaker nodes for those 2.1

Quoted 19 lines

Quoted 19 lines

Quoted 19 lines

- I have found a couple of kernel bug reports for other ASUS models with
this chipset but they are both marked resolved. Should I open a new ticket
?

https://bugzilla.kernel.org/show_bug.cgi?id=65091
https://bugzilla.kernel.org/show_bug.cgi?id=66271

- I can hear static through the front speakers (with pulseaudio test) if
I set pins 0x19 and 0x1a to Line Out (Center/LFE) and Internal (Center/LFE) or vice versa with hdajackretask.

Pin 0x16 does not appear to work at all.

I see this output when I apply the config in hdajackretask.

Quoted 5 lines

Quoted 5 lines

Quoted 5 lines

0x12 0x90a60140
0x14 0x99130110
0x15 0x04211020
0x16 0x99134111
0x18 0x01a19840

Quoted 8 lines

Quoted 8 lines

Quoted 8 lines

0x19 0x01014011
0x1a 0x411111f0
0x1b 0x411111f0
0x1d 0x40c6852d
0x1e 0x014b1180
0x1f 0x411111f0
1

Documentation/devicetree/bindings/sound/fsl,asrc.txt line 33

Quoted 3 lines

Documentation/devicetree/bindings/sound/fsl,asrc.txt new file +3 -0

+   - big-endian : If this property is absent, the native endian mode will
+                 be in use as default, or the big endian mode will be in use
+                 for all the device registers.

Native endian is meaningless. If a CPU supports both BE and LE, there is no native endianness. The endianness of the kernel is dynamic while the endianness of registers in HW is fixed.

Just choose an endianness to assume by default (presumably little). That way this describes the HW and always works with a kernel of arbitrary endianness.

Quoted 10 lines

Quoted 10 lines

Quoted 10 lines

Quoted 9 lines

What is the best way to test/debug the alc668 driver these days?

I am running kernel 3.14.1 (debian testing/sid).

The laptop is an ASUS N750JK.

Sound works for front left/right speakers but the back speakers and external lfe channel is not working on this laptop.

Quoted 7 lines

Quoted 4 lines

Quoted 2 lines

You have to post output of alsa-info.sh

http://boosthardware.com/alsa-info-alc668-n750jk.txt

Seem BIOS only set up the front speakers

There is nothing in the bios to configure the audio device

Quoted 8 lines

Quoted 7 lines

Quoted 3 lines

How many audio jacks ?

Standard headphone/mic + yellow external lfe (sonic master). There is also
hdmi but I am not testing that at the moment.

Quoted 4 lines

http://www.asus.com/Notebooks_Ultrabooks/N750JK/specifications/

Do you mean headset ?

There are the following jacks:

1 x headphone

1 x mic(line in)

1 x sonic master (yellow)
1 x hdmi (intel Haswell hdmi)

Quoted 9 lines

Quoted 9 lines

Quoted 5 lines

Do it support headset or headphone ?

Do you mean there are four internal speakers ?

There are two front speakers which produce sound through the front bottom
of the case. There are also two back speakers which do not produce

Quoted 20 lines

Quoted 20 lines

in my tests so far.

Quoted 2 lines

Do it support surround40 in windows ?

I am not running windows OS on this machine but I suspect it does.

Quoted 4 lines

The external sonic master and headphone need two volume controls but alc668 only have three

The best way to test is using latest git version and hda jack retask

to

Quoted 8 lines

1) four internal speaker and headphone
2) two internal speaker, headphone and external subwoofer

Once you confirm the nodes of the speakers, subwooder You need to decide whether headphone share the volume controls with

Quoted 8 lines

Quoted 5 lines

the

Quoted 2 lines

speaker/subwoofer

Quoted 1 line

I have spent some time on this step already with 3.14.1

Did you retask the node of subwoofer as speaker or line out ?

I have tried both but I have not been able to identify the subwoofer.

I have tried playing with: 16, 19, 1a

Quoted 2 lines

Did you find the node of the rear speakers in (1) ?

I have not found them yet.

Quoted 2 lines

Did surround 4.0 work when you fix the rear speaker ?

Only the front speakers play audio but they don't work with 4.0 enabled

Quoted 2 lines

Did you find the node of the external subwoofer in (2) ?

Not yet.

Quoted 3 lines

Did the sonic master subwoofer detected ?
hda jack sense test -a

I have grabbed the scripts but I get this error when running hda-analyzer:

Quoted 3 lines

Do the subwoofer only use right channel ?

I cannot hear the subwoofer at this point

Quoted 2 lines

Do surround 2.1 works ?

It says it is working but I do not get sound out of the sub woofer. I can hear static(whitenoise) through the front speakers (with pulseaudio test) if I set pins 0x19 and 0x1a to Line Out (Center/LFE) and Internal (Center/LFE) or vice versa with hdajackretask.

Quoted 5 lines

Quoted 4 lines

Is there a specific reason this process is not already scripted other than
no one has taken the time to do it?

Quoted 2 lines

Do surround 4.1 or 5.1 work when you fix both speaker and subwoofer ?

Surround 4.1/5/1 do not produce any sound from my tests so far (using PA Test)

Quoted 1 line

Do you prefer driver create headphone volume control or subwoofer

Quoted 2 lines

control ?

I would like the lfe to be sent through the yellow sonic master jack. I am not concerned with the headset jack at the moment. A cursory test shows just the right channel is working on the headset jack.

Quoted 6 lines

Quoted 4 lines

Can we not create a test script that will run through all the logical combinations and provide all the relevant info?

The driver hardcore the name of volume control to bass when there are

Quoted 2 lines

speaker nodes for those 2.1

I do not see any controls for the bass volume in alsamixer

Quoted 11 lines

Quoted 11 lines

Quoted 11 lines

Quoted 11 lines

- I have found a couple of kernel bug reports for other ASUS models with
this chipset but they are both marked resolved. Should I open a new ticket
?

https://bugzilla.kernel.org/show_bug.cgi?id=65091
https://bugzilla.kernel.org/show_bug.cgi?id=66271

- I can hear static through the front speakers (with pulseaudio

Quoted 6 lines

Quoted 4 lines

Quoted 4 lines

Quoted 4 lines

if
I set pins 0x19 and 0x1a to Line Out (Center/LFE) and Internal (Center/LFE) or vice versa with hdajackretask.

For multi channel, the sequence number is ascending and the driver

Quoted 2 lines

they have same type i.e, all speakers or all line out jacks

You can use hda-verb GET_PIN_SENSE to test all the pin complex nodes which pincap support DETECT

Quoted 10 lines

Quoted 2 lines

Do surround 2.1 works ?

It says it is working but I do not get sound out of the sub woofer. I can
hear static(whitenoise) through the front speakers (with pulseaudio test)
if I set pins 0x19 and 0x1a to Line Out (Center/LFE) and Internal (Center/LFE) or vice versa with hdajackretask.

http://git.alsa-project.org/?p=alsa-tools.git;a=blob;f=hdajackretask/README;hb=HEAD

Did you use apply button(dyanmic reconfiguartion) or installment boot override (early patching) ?

Quoted 4 lines

I do not see any controls for the bass volume in alsamixer

https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda?id=f37bc7a88d374448a1f4bba9267d308606d78bf2

Quoted 13 lines

Quoted 2 lines

Quoted 2 lines

Quoted 2 lines

Quoted 2 lines

- I have found a couple of kernel bug reports for other ASUS

models

Quoted 2 lines

Quoted 2 lines

Quoted 2 lines

Quoted 2 lines

with
this chipset but they are both marked resolved. Should I open a

new

Quoted 7 lines

Quoted 7 lines

Quoted 7 lines

Quoted 7 lines

ticket
?

https://bugzilla.kernel.org/show_bug.cgi?id=65091
https://bugzilla.kernel.org/show_bug.cgi?id=66271

- I can hear static through the front speakers (with pulseaudio

test)

Quoted 6 lines

Quoted 6 lines

Quoted 4 lines

Quoted 4 lines

Quoted 4 lines

if
I set pins 0x19 and 0x1a to Line Out (Center/LFE) and Internal (Center/LFE) or vice versa with hdajackretask.

For multi channel, the sequence number is ascending and the driver

expect

Quoted 2 lines

Quoted 2 lines

they have same type i.e, all speakers or all line out jacks

Pin Default 0x90170110: [Fixed] Speaker at Int N/A
Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE

The rear speaker need same default association but sequence number higher than that of the front speaker

The external subwoofer may use [jack] speaker with same default association and higher sequence number if it can be detected

The driver won't use all of them for multi channel if you retasked it as line out

Quoted 3 lines

You can use hda-verb GET_PIN_SENSE to test all the pin complex nodes

Quoted 2 lines

pincap support DETECT

I have tried a few attempts but it's pretty confusing.

These ones don't do anything:

# hda-verb /dev/snd/hwC1D0  GET_PIN_SENSE
# hda-verb /dev/snd/hwC1D0  0x0 GET_PIN_SENSE

I don't know what to make of this output:

# hda-verb /dev/snd/hwC1D0  0x0 GET_PIN_SENSE PIN_CAP
nid = 0x0, verb = 0xf09, param = 0xc
value = 0x0
# hda-verb /dev/snd/hwC0D0  0x0 GET_PIN_SENSE PIN_CAP
nid = 0x0, verb = 0xf09, param = 0xc
value = 0x0
# hda-verb /dev/snd/hwC0D0  0x16 GET_PIN_SENSE PIN_CAP
nid = 0x16, verb = 0xf09, param = 0xc
value = 0xffffffff
# hda-verb /dev/snd/hwC0D0  0x19 GET_PIN_SENSE PIN_CAP

nid = 0x19, verb = 0xf09, param = 0xc
value = 0xffffffff
# hda-verb /dev/snd/hwC0D0  0x1a GET_PIN_SENSE PIN_CAP
nid = 0x1a, verb = 0xf09, param = 0xc
value = 0xffffffff
# hda-verb /dev/snd/hwC0D0  0x1b GET_PIN_SENSE PIN_CAP
nid = 0x1b, verb = 0xf09, param = 0xc
value = 0xffffffff

Quoted 5 lines

Quoted 5 lines

Quoted 4 lines

Do surround 2.1 works ?

Quoted 9 lines

Quoted 7 lines

It says it is working but I do not get sound out of the sub woofer. I can
hear static(whitenoise) through the front speakers (with pulseaudio test)
if I set pins 0x19 and 0x1a to Line Out (Center/LFE) and Internal (Center/LFE) or vice versa with hdajackretask.

Quoted 4 lines

Did you use apply button(dyanmic reconfiguartion) or installment boot override (early patching) ?

I have tried both options.

After rebooting with the apply option set I don't hear any static or audio with the PA test app.

Quoted 6 lines

Quoted 4 lines

I do not see any controls for the bass volume in alsamixer

Quoted 1 line

I have checked alsamixer again after the last round of tinkering with hdajackretask and" Speaker CLFE" is now showing up but it does not have volume control just mute on/off.

I see the following new controls

Speaker CFLE (mute only)
Speaker Front (mute + vol)
Speaker Side (mute only)
Speaker Surround (mute + vol)

Quoted 31 lines

Quoted 15 lines

Quoted 4 lines

Quoted 4 lines

Quoted 4 lines

Quoted 4 lines

- I have found a couple of kernel bug reports for other ASUS

models

Quoted 2 lines

Quoted 2 lines

Quoted 2 lines

Quoted 2 lines

with
this chipset but they are both marked resolved. Should I open a

new

Quoted 7 lines

Quoted 7 lines

Quoted 7 lines

Quoted 7 lines

ticket
?

https://bugzilla.kernel.org/show_bug.cgi?id=65091
https://bugzilla.kernel.org/show_bug.cgi?id=66271

- I can hear static through the front speakers (with pulseaudio

test)

Quoted 6 lines

Quoted 6 lines

Quoted 4 lines

Quoted 4 lines

Quoted 4 lines

if
I set pins 0x19 and 0x1a to Line Out (Center/LFE) and Internal (Center/LFE) or vice versa with hdajackretask.

For multi channel, the sequence number is ascending and the driver

expect

Quoted 2 lines

Quoted 2 lines

they have same type i.e, all speakers or all line out jacks

Pin Default 0x90170110: [Fixed] Speaker at Int N/A
Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE

Quoted 9 lines

The rear speaker need same default association but sequence number higher than that of the front speaker

The external subwoofer may use [jack] speaker with same default association and higher sequence number if it can be detected

The driver won't use all of them for multi channel if you retasked it as line out

I have tried several combinations with hdajackretask

I am not able to get audio out of the physical rear speakers or sonic master output.
:

So what do I need to do to write a script that will test all the combinations and play audio through them? Or am I missing something that someone else has already written for this purpose?

Currently I am using this process:

1: hdajackretask - modify params
2: pulseaudio -k
3: hdajackretask : apply settings
4: pulseaudio -D
5: open PA audio config
6: select audio output option and press test speakers button 7: manually press every button

http://git.alsa-project.org/?p=alsa.git;a=commitdiff;h=48ed9474fc6724dc1486d30e9a274ab27ec001da;hp=56c640a98975b03a6a78a44b2c01e49dfd7b5263

Have you try the fix ?

Quoted 1 line

Quoted 2 lines

Quoted 2 lines

You can use hda-verb GET_PIN_SENSE to test all the pin complex nodes

which

Quoted 5 lines

Quoted 2 lines

pincap support DETECT

I have tried a few attempts but it's pretty confusing.

Quoted 7 lines

These ones don't do anything:

# hda-verb /dev/snd/hwC1D0  GET_PIN_SENSE
# hda-verb /dev/snd/hwC1D0  0x0 GET_PIN_SENSE

I don't know what to make of this output:

Bit 31 is PD

VALUE should return 0 when Jack is UNplugged and 0x80000000 when Jack is plugged for those pin which support DETECT

But your alc668 is on card 1 , the second hda controller

Quoted 6 lines

I see the following new controls

Speaker CFLE (mute only)
Speaker Front (mute + vol)
Speaker Side (mute only)
Speaker Surround (mute + vol)

hda-jack-retask won't check whether your codec have enough audio output if you changed all the nodes to speaker

But the driver expect multiout support up to 8 channels when parsing the pin

As alc668 only support 6 channels, hda_generic.c just assign audio output and create volume control but Jack detection controls and mute switch are created according to the number of parsed pins which support Jack detect and mute switch

Do the headphone still work with pulseaudio since there is no headphone playback volume and headphone is share SPEAKER FRONT PLAYBAACK VOLUME control with front speaker ?

if the connected retasked pin is not the correct pin
,you can use hda_analyzer to change the pin ctls of the remaining unconnected pin to OUT, unmute and coonect the pin to second/third dac through audio mixers

Quoted 1 line

Quoted 3 lines

Have you try the fix ?

Quoted 1 line

I did not see that update.

run.py works now without any modifications on this machine.

I have generated a graph.

http://boosthardware.com/hda_graph-1.png

Quoted 19 lines

Quoted 2 lines

Quoted 2 lines

You can use hda-verb GET_PIN_SENSE to test all the pin complex nodes

which

Quoted 5 lines

Quoted 2 lines

pincap support DETECT

I have tried a few attempts but it's pretty confusing.

Quoted 7 lines

These ones don't do anything:

# hda-verb /dev/snd/hwC1D0  GET_PIN_SENSE
# hda-verb /dev/snd/hwC1D0  0x0 GET_PIN_SENSE

I don't know what to make of this output:

Bit 31 is PD

VALUE should return 0 when Jack is UNplugged and 0x80000000 when Jack

Quoted 5 lines

plugged for those pin which support DETECT

But your alc668 is on card 1 , the second hda controller

Sorry. Missed that one.

# hda-verb /dev/snd/hwC1D0  0x16 GET_PIN_SENSE PIN_CAP
nid = 0x16, verb = 0xf09, param = 0xc
value = 0x0
# hda-verb /dev/snd/hwC1D0  0x19 GET_PIN_SENSE PIN_CAP
nid = 0x19, verb = 0xf09, param = 0xc
value = 0x0
# hda-verb /dev/snd/hwC1D0  0x11a GET_PIN_SENSE PIN_CAP
invalid nid 0x11a
# hda-verb /dev/snd/hwC1D0  0x1a GET_PIN_SENSE PIN_CAP
nid = 0x1a, verb = 0xf09, param = 0xc
value = 0x0
# hda-verb /dev/snd/hwC1D0  0x1b GET_PIN_SENSE PIN_CAP
nid = 0x1b, verb = 0xf09, param = 0xc
value = 0x0

Quoted 8 lines

Quoted 6 lines

I see the following new controls

Speaker CFLE (mute only)
Speaker Front (mute + vol)
Speaker Side (mute only)
Speaker Surround (mute + vol)

hda-jack-retask won't check whether your codec have enough audio output

Quoted 6 lines

you changed all the nodes to speaker

But the driver expect multiout support up to 8 channels when parsing the pin

As alc668 only support 6 channels, hda_generic.c just assign audio

Quoted 1 line

and create volume control but Jack detection controls and mute switch

Quoted 7 lines

created according to the number of parsed pins which support Jack detect and mute switch

Do the headphone still work with pulseaudio since there is no headphone playback volume and headphone is share SPEAKER FRONT PLAYBAACK VOLUME control with front speaker ?

With speaker-test I can hear audio on the right channel

speaker-test -c2 -t wav -D hw:1,0

audio comes out of front left/right

speaker-test -c4 -t wav -D hw:1,0

front left/right has no audio
rear left/right audio comes out of front left/right speakers

# speaker-test -c6 -t wav -D hw:1,0

speaker-test 1.0.28

Playback device is hw:1,0
Stream parameters are 48000Hz, S16_LE, 6 channels
WAV file(s)
Channels count (6) not available for playbacks: Invalid argument Setting of hwparams failed: Invalid argument

Do you mean the plugging of subwoofer cannot be detected

hda-jack-sense-test -c 1 -a

The graph don't contain enough into for debugg

Post output of alsa-info.sh after retasked

Seem headphone is more important than multichannel speaker when evaluating the badness of config

You may need to configure alsa driver --with-debug=verbose to produce more information in system log about how it evaluate the badness

alsa-info:

retask and apply
http://boosthardware.com/alsa-info1.txt

store and reboot
http://boosthardware.com/alsa-info2.txt

Quoted 6 lines

alsa-info:

retask and apply
http://boosthardware.com/alsa-info1.txt

Quoted 4 lines

store and reboot
http://boosthardware.com/alsa-info2.txt

Any reason to use asus-mode4 and hda-jack-retask at the same time ?

Try model=auto when you use patch=,hda-jack-retask.fw

Your alsa-info missing many important info , need all info from alsactl, amixer, sysfs about the driver init pins and user init pins, allay/arecord -l

No information about which Jack have been retasked