Skip to contentSkip to conversation

ASoC updates

PATCH In review Mark Brown 5 participants 15 messages Nov 5, 2008 – Nov 7, 2008

Subsystems (6)
lore
Share & tools
lore

Mark Brown

author
#0 lore raw

cover ASoC updates

The following changes since commit 0ee4663617fb0f78cec4cc6558a096ccbd8c3ffc:
Takashi Iwai (1):
ALSA: ASoC - Remove unnecessary inclusion of linux/version.h

are available in the git repository at:

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

David Anders (1):
ASoC: Add new parameter to s3c24xx_pcm_enqueue

Marek Vasut (1):
ASoC: Add Palm/PXA27x unified ASoC audio driver

Mark Brown (1):
ASoC: Remove core version number

Troy Kisky (1):
ASoC: TLV320AIC23B Support more sample rates

 arch/arm/mach-pxa/include/mach/palmasoc.h |   13 ++
 include/sound/soc.h                       |    2 -
 sound/soc/codecs/tlv320aic23.c            |  227 +++++++++++++++++++-----
 sound/soc/pxa/Kconfig                     |    9 +
 sound/soc/pxa/Makefile                    |    2 +
 sound/soc/pxa/palm27x.c                   |  269 +++++++++++++++++++++++++++++
 sound/soc/s3c24xx/s3c24xx-pcm.c           |   12 +-
 sound/soc/soc-core.c                      |    1 -
 8 files changed, 483 insertions(+), 52 deletions(-)
 create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h
 create mode 100644 sound/soc/pxa/palm27x.c
author
#1 lore raw

PATCH 1/4 ASoC: TLV320AIC23B Support more sample rates

From: Troy Kisky <[email protected]>

Add support for more sample rates, different crystals and split playback/capture rates.

---
 sound/soc/codecs/tlv320aic23.c |  227 ++++++++++++++++++++++++++++++++--------
 1 files changed, 182 insertions(+), 45 deletions(-)
sound/soc/codecs/tlv320aic23.c +182 -45
index 44308da..c1b2060 100644
@@ -37,12 +37,6 @@
 
 #define AIC23_VERSION "0.1"
 
-struct tlv320aic23_srate_reg_info {
-	u32 sample_rate;
-	u8 control;		/* SR3, SR2, SR1, SR0 and BOSR */
-	u8 divider;		/* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
-};
-
 /*
  * AIC23 register cache
  */
@@ -261,19 +255,155 @@ static const struct snd_soc_dapm_route intercon[] = {
 
 };
 
-/* tlv320aic23 related */
-static const struct tlv320aic23_srate_reg_info srate_reg_info[] = {
-	{4000, 0x06, 1},	/*  4000 */
-	{8000, 0x06, 0},	/*  8000 */
-	{16000, 0x0C, 1},	/* 16000 */
-	{22050, 0x11, 1},	/* 22050 */
-	{24000, 0x00, 1},	/* 24000 */
-	{32000, 0x0C, 0},	/* 32000 */
-	{44100, 0x11, 0},	/* 44100 */
-	{48000, 0x00, 0},	/* 48000 */
-	{88200, 0x1F, 0},	/* 88200 */
-	{96000, 0x0E, 0},	/* 96000 */
+/* AIC23 driver data */
+struct aic23 {
+	struct snd_soc_codec codec;
+	int mclk;
+	int requested_adc;
+	int requested_dac;
+};
+
+/*
+ * Common Crystals used
+ * 11.2896 Mhz /128 = *88.2k  /192 = 58.8k
+ * 12.0000 Mhz /125 = *96k    /136 = 88.235K
+ * 12.2880 Mhz /128 = *96k    /192 = 64k
+ * 16.9344 Mhz /128 = 132.3k /192 = *88.2k
+ * 18.4320 Mhz /128 = 144k   /192 = *96k
+ */
+
+/*
+ * Normal BOSR 0-256/2 = 128, 1-384/2 = 192
+ * USB BOSR 0-250/2 = 125, 1-272/2 = 136
+ */
+static const int bosr_usb_divisor_table[] = {
+	128, 125, 192, 136
+};
+#define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7))
+#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11)        | (1<<15))
+static const unsigned short sr_valid_mask[] = {
+	LOWER_GROUP|UPPER_GROUP,	/* Normal, bosr - 0*/
+	LOWER_GROUP|UPPER_GROUP,	/* Normal, bosr - 1*/
+	LOWER_GROUP,			/* Usb, bosr - 0*/
+	UPPER_GROUP,			/* Usb, bosr - 1*/
+};
+/*
+ * Every divisor is a factor of 11*12
+ */
+#define SR_MULT (11*12)
+#define A(x) (x) ? (SR_MULT/x) : 0
+static const unsigned char sr_adc_mult_table[] = {
+	A(2), A(2), A(12), A(12),  A(0), A(0), A(3), A(1),
+	A(2), A(2), A(11), A(11),  A(0), A(0), A(0), A(1)
 };
+static const unsigned char sr_dac_mult_table[] = {
+	A(2), A(12), A(2), A(12),  A(0), A(0), A(3), A(1),
+	A(2), A(11), A(2), A(11),  A(0), A(0), A(0), A(1)
+};
+
+unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
+		int dac, int dac_l, int dac_h, int need_dac)
+{
+	if ((adc >= adc_l) && (adc <= adc_h) &&
+			(dac >= dac_l) && (dac <= dac_h)) {
+		unsigned diff_adc = need_adc - adc;
+		unsigned diff_dac = need_dac - dac;
+		unsigned score;
+		if (((int)diff_adc) < 0)
+			diff_adc = -diff_adc;
+		if (((int)diff_dac) < 0)
+			diff_dac = -diff_dac;
+		score = diff_adc + diff_dac;
+		return score;
+	}
+	return 0xffffffff;
+}
+int find_rate(int mclk, u32 need_adc, u32 need_dac)
+{
+	int i, j;
+	int best_i = -1;
+	int best_j = -1;
+	int best_div = 0;
+	unsigned best_score = 0xffffffff;
+	int adc_l, adc_h, dac_l, dac_h;
+
+	need_adc *= SR_MULT;
+	need_dac *= SR_MULT;
+	/*
+	 * rates given are +/- 1/32
+	 */
+	adc_l = need_adc - (need_adc >> 5);
+	adc_h = need_adc + (need_adc >> 5);
+	dac_l = need_dac - (need_dac >> 5);
+	dac_h = need_dac + (need_dac >> 5);
+	for (i = 0; i < 4; i++) {
+		int base = mclk / bosr_usb_divisor_table[i];
+		int mask = sr_valid_mask[i];
+		for (j = 0; j < 16; j++, mask >>= 1) {
+			int adc;
+			int dac;
+			int score;
+			if ((mask & 1) == 0)
+				continue;
+			adc = base * sr_adc_mult_table[j];
+			dac = base * sr_dac_mult_table[j];
+			score = get_score(adc, adc_l, adc_h, need_adc,
+					dac, dac_l, dac_h, need_dac);
+			if (best_score > score) {
+				best_score = score;
+				best_i = i;
+				best_j = j;
+				best_div = 0;
+			}
+			score = get_score((adc >> 1), adc_l, adc_h, need_adc,
+					(dac >> 1), dac_l, dac_h, need_dac);
+			/* prefer to have a /2 */
+			if ((score != 0xffffffff) && (best_score >= score)) {
+				best_score = score;
+				best_i = i;
+				best_j = j;
+				best_div = 1;
+			}
+		}
+	}
+	return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT);
+}
+
+static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk,
+		u32 *sample_rate_adc, u32 *sample_rate_dac)
+{
+	int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE);
+	int sr = (src >> 2) & 0x0f;
+	int val = (mclk / bosr_usb_divisor_table[src & 3]);
+	int adc = (val * sr_adc_mult_table[sr]) / SR_MULT;
+	int dac = (val * sr_dac_mult_table[sr]) / SR_MULT;
+	if (src & TLV320AIC23_CLKIN_HALF) {
+		adc >>= 1;
+		dac >>= 1;
+	}
+	*sample_rate_adc = adc;
+	*sample_rate_dac = dac;
+}
+
+static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk,
+		u32 sample_rate_adc, u32 sample_rate_dac)
+{
+	/* Search for the right sample rate */
+	int data = find_rate(mclk, sample_rate_adc, sample_rate_dac);
+	if (data < 0) {
+		printk(KERN_ERR "%s:Invalid rate %u,%u requested\n",
+				__func__, sample_rate_adc, sample_rate_dac);
+		return -EINVAL;
+	}
+	tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
+	if (1) {
+		int adc, dac;
+		get_current_sample_rates(codec, mclk, &adc, &dac);
+		printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n",
+			adc, dac, data);
+	}
+	return 0;
+}
 
 static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
 {
@@ -293,27 +423,30 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
-	u16 iface_reg, data;
-	u8 count = 0;
+	u16 iface_reg;
+	int ret;
+	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
+	u32 sample_rate_adc = aic23->requested_adc;
+	u32 sample_rate_dac = aic23->requested_dac;
+	u32 sample_rate = params_rate(params);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		aic23->requested_dac = sample_rate_dac = sample_rate;
+		if (!sample_rate_adc)
+			sample_rate_adc = sample_rate;
+	} else {
+		aic23->requested_adc = sample_rate_adc = sample_rate;
+		if (!sample_rate_dac)
+			sample_rate_dac = sample_rate;
+	}
+	ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc,
+			sample_rate_dac);
+	if (ret < 0)
+		return ret;
 
 	iface_reg =
 	    tlv320aic23_read_reg_cache(codec,
 				       TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
-
-	/* Search for the right sample rate */
-	/* Verify what happens if the rate is not supported
-	 * now it goes to 96Khz */
-	while ((srate_reg_info[count].sample_rate != params_rate(params)) &&
-	       (count < ARRAY_SIZE(srate_reg_info))) {
-		count++;
-	}
-
-	data =  (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) |
-		(srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) |
-		TLV320AIC23_USB_CLK_ON;
-
-	tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
-
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		break;
@@ -349,12 +482,17 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
+	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
 
 	/* deactivate */
 	if (!codec->active) {
 		udelay(50);
 		tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
 	}
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		aic23->requested_dac = 0;
+	else
+		aic23->requested_adc = 0;
 }
 
 static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
@@ -422,12 +560,9 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				      int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-
-	switch (freq) {
-	case 12000000:
-		return 0;
-	}
-	return -EINVAL;
+	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
+	aic23->mclk = freq;
+	return 0;
 }
 
 static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
@@ -659,14 +794,15 @@ static int tlv320aic23_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec;
+	struct aic23 *aic23;
 	int ret = 0;
 
 	printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION);
 
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
+	aic23 = kzalloc(sizeof(struct aic23), GFP_KERNEL);
+	if (aic23 == NULL)
 		return -ENOMEM;
-
+	codec = &aic23->codec;
 	socdev->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
@@ -687,6 +823,7 @@ static int tlv320aic23_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->codec;
+	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
 
 	if (codec->control_data)
 		tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -697,7 +834,7 @@ static int tlv320aic23_remove(struct platform_device *pdev)
 	i2c_del_driver(&tlv320aic23_i2c_driver);
 #endif
 	kfree(codec->reg_cache);
-	kfree(codec);
+	kfree(aic23);
 
 	return 0;
 }
Signature · 1 line
1.5.6.5
author
#2 lore raw

PATCH 2/4 ASoC: Add Palm/PXA27x unified ASoC audio driver

From: Marek Vasut <[email protected]>

this patch adds asoc audio driver for pxa27x based Palm PDAs. I tested it for palmtx, t5 and ld, it should work with palmz72 as well (slapin, please test). I sent it here some time ago, but now I got to fixing bugs in it. It should be somehow mostly ok and ready for applying.

[Converted to use snd_soc_dapm_nc_pin() -- broonie]

---
 arch/arm/mach-pxa/include/mach/palmasoc.h |   13 ++
 sound/soc/pxa/Kconfig                     |    9 +
 sound/soc/pxa/Makefile                    |    2 +
 sound/soc/pxa/palm27x.c                   |  269 +++++++++++++++++++++++++++++
 4 files changed, 293 insertions(+), 0 deletions(-)
 create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h
 create mode 100644 sound/soc/pxa/palm27x.c
arch/arm/mach-pxa/include/mach/palmasoc.h new file +13 -0
new file mode 100644
index 0000000..6c4b1f7
@@ -0,0 +1,13 @@
+#ifndef _INCLUDE_PALMASOC_H_
+#define _INCLUDE_PALMASOC_H_
+struct palm27x_asoc_info {
+	int	jack_gpio;
+};
+
+#ifdef CONFIG_SND_PXA2XX_SOC_PALM27X
+void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data);
+#else
+static inline void palm27x_asoc_set_pdata(struct palm27x_asoc_info *data) {}
+#endif
+
+#endif
sound/soc/pxa/Kconfig +9 -0
index 4235524..1187603 100644
@@ -79,6 +79,15 @@ config SND_PXA2XX_SOC_EM_X270
 	  Say Y if you want to add support for SoC audio on
 	  CompuLab EM-x270.
 
+config SND_PXA2XX_SOC_PALM27X
+	tristate "SoC Audio support for Palm T|X, T5 and LifeDrive"
+	depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || MACH_PALMT5)
+	select SND_PXA2XX_SOC_AC97
+	select SND_SOC_WM9712
+	help
+	  Say Y if you want to add support for SoC audio on
+	  Palm T|X, T5 or LifeDrive handheld computer.
+
 config SND_SOC_ZYLONITE
 	tristate "SoC Audio support for Marvell Zylonite"
 	depends on SND_PXA2XX_SOC && MACH_ZYLONITE
sound/soc/pxa/Makefile +2 -0
index 00258ab..1a3a36e 100644
@@ -16,6 +16,7 @@ snd-soc-tosa-objs := tosa.o
 snd-soc-e800-objs := e800_wm9712.o
 snd-soc-spitz-objs := spitz.o
 snd-soc-em-x270-objs := em-x270.o
+snd-soc-palm27x-objs := palm27x.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -23,3 +24,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
 obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
 obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
 obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
+obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
sound/soc/pxa/palm27x.c new file +269 -0

269 lines not on this page · raw message

Signature · 1 line
1.5.6.5
author
#3 lore raw

PATCH 3/4 ASoC: Remove core version number

Rather than try to remember to keep the core version number updated (which hasn't been happening) just remove it. It was much more useful when ASoC was out of tree.

---
 include/sound/soc.h  |    2 --
 sound/soc/soc-core.c |    1 -
 2 files changed, 0 insertions(+), 3 deletions(-)
include/sound/soc.h +0 -2
index da0040b..fa1b99b 100644
@@ -21,8 +21,6 @@
 #include <sound/control.h>
 #include <sound/ac97_codec.h>
 
-#define SND_SOC_VERSION "0.13.2"
-
 /*
  * Convenience kcontrol builders
  */
sound/soc/soc-core.c +0 -1
index 1a17368..9feaa7b 100644
@@ -1965,7 +1965,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
 
 static int __devinit snd_soc_init(void)
 {
-	printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION);
 	return platform_driver_register(&soc_driver);
 }
 
Signature · 1 line
1.5.6.5
author
#4 lore raw

PATCH 4/4 ASoC: Add new parameter to s3c24xx_pcm_enqueue

From: David Anders <[email protected]>

The S3C24xx dma does not allow more than one buffer to be enqueue prior to the dma transfers starting. This patch adds an additional parameter to s3c24xx_pcm_enqueue() to allow for passing an initial dma maximum load value.

Signed-off-by David Anders <danders at amltd.com>
---
 sound/soc/s3c24xx/s3c24xx-pcm.c |   12 ++++++++----
 1 files changed, 8 insertions(+), 4 deletions(-)
sound/soc/s3c24xx/s3c24xx-pcm.c +8 -4
index e13e614..341198b 100644
@@ -78,7 +78,8 @@ struct s3c24xx_runtime_data {
  * place a dma buffer onto the queue for the dma system
  * to handle.
 */
-static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
+static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream,
+				int dma_max)
 {
 	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
 	dma_addr_t pos = prtd->dma_pos;
@@ -86,7 +87,10 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
 
 	DBG("Entered %s\n", __func__);
 
-	while (prtd->dma_loaded < prtd->dma_limit) {
+	if (!dma_max)
+		dma_max = prtd->dma_limit;
+
+	while (prtd->dma_loaded < dma_max) {
 		unsigned long len = prtd->dma_period;
 
 		DBG("dma_loaded: %d\n", prtd->dma_loaded);
@@ -132,7 +136,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
 	spin_lock(&prtd->lock);
 	if (prtd->state & ST_RUNNING) {
 		prtd->dma_loaded--;
-		s3c24xx_pcm_enqueue(substream);
+		s3c24xx_pcm_enqueue(substream, 0);
 	}
 
 	spin_unlock(&prtd->lock);
@@ -249,7 +253,7 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream)
 	prtd->dma_pos = prtd->dma_start;
 
 	/* enqueue dma buffers */
-	s3c24xx_pcm_enqueue(substream);
+	s3c24xx_pcm_enqueue(substream, 1);
 
 	return ret;
 }
Signature · 1 line
1.5.6.5
#5 lore raw

Re: [PATCH 2/4] ASoC: Add Palm/PXA27x unified ASoC audio driver

In reply to Mark Brown
Quoted 8 lines · On Wednesday 05 of November 2008 19:53:29 Mark Brown wrote:

From: Marek Vasut <[email protected]>

this patch adds asoc audio driver for pxa27x based Palm PDAs. I tested it for palmtx, t5 and ld, it should work with palmz72 as well (slapin, please test). I sent it here some time ago, but now I got to fixing bugs in it. It should be somehow mostly ok and ready for applying.

[Converted to use snd_soc_dapm_nc_pin() -- broonie]

Mark, thanks a lot for fixing it, I'm really overloaded with school recently and I dont see it ending any soon.

Quoted 344 lines
---
 arch/arm/mach-pxa/include/mach/palmasoc.h |   13 ++
 sound/soc/pxa/Kconfig                     |    9 +
 sound/soc/pxa/Makefile                    |    2 +
 sound/soc/pxa/palm27x.c                   |  269
+++++++++++++++++++++++++++++ 4 files changed, 293 insertions(+), 0
deletions(-)
 create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h
 create mode 100644 sound/soc/pxa/palm27x.c
(quoted diff) +0 -0

b/arch/arm/mach-pxa/include/mach/palmasoc.h new file mode 100644 index 0000000..6c4b1f7

arch/arm/mach-pxa/include/mach/palmasoc.h new file +10 -0
@@ -0,0 +1,13 @@
+#ifndef _INCLUDE_PALMASOC_H_
+#define _INCLUDE_PALMASOC_H_
+struct palm27x_asoc_info {
+	int	jack_gpio;
+};
+
+#ifdef CONFIG_SND_PXA2XX_SOC_PALM27X
+void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data);
+#else
+static inline void palm27x_asoc_set_pdata(struct palm27x_asoc_info *data)

{} +#endif

arch/arm/mach-pxa/include/mach/palmasoc.h new file +2 -0
+
+#endif
sound/soc/pxa/Kconfig +9 -0
index 4235524..1187603 100644
@@ -79,6 +79,15 @@ config SND_PXA2XX_SOC_EM_X270
 	  Say Y if you want to add support for SoC audio on
 	  CompuLab EM-x270.
 
+config SND_PXA2XX_SOC_PALM27X
+	tristate "SoC Audio support for Palm T|X, T5 and LifeDrive"
+	depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || MACH_PALMT5)
+	select SND_PXA2XX_SOC_AC97
+	select SND_SOC_WM9712
+	help
+	  Say Y if you want to add support for SoC audio on
+	  Palm T|X, T5 or LifeDrive handheld computer.
+
 config SND_SOC_ZYLONITE
 	tristate "SoC Audio support for Marvell Zylonite"
 	depends on SND_PXA2XX_SOC && MACH_ZYLONITE
sound/soc/pxa/Makefile +2 -0
index 00258ab..1a3a36e 100644
@@ -16,6 +16,7 @@ snd-soc-tosa-objs := tosa.o
 snd-soc-e800-objs := e800_wm9712.o
 snd-soc-spitz-objs := spitz.o
 snd-soc-em-x270-objs := em-x270.o
+snd-soc-palm27x-objs := palm27x.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -23,3 +24,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
 obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
 obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
 obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
+obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
sound/soc/pxa/palm27x.c new file +269 -0
new file mode 100644
index 0000000..e364abc
@@ -0,0 +1,269 @@
+/*
+ * linux/sound/soc/pxa/palm27x.c
+ *
+ * SoC Audio driver for Palm T|X, T5 and LifeDrive
+ *
+ * based on tosa.c
+ *
+ * Copyright (C) 2008 Marek Vasut <[email protected]>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/gpio.h>
+#include <linux/interrupt.h>
+#include <linux/irq.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/audio.h>
+#include <mach/palmasoc.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static int palm27x_jack_func = 1;
+static int palm27x_spk_func = 1;
+static int palm27x_ep_gpio = -1;
+
+static void palm27x_ext_control(struct snd_soc_codec *codec)
+{
+	if (!palm27x_spk_func)
+		snd_soc_dapm_enable_pin(codec, "Speaker");
+	else
+		snd_soc_dapm_disable_pin(codec, "Speaker");
+
+	if (!palm27x_jack_func)
+		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+	else
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+
+	snd_soc_dapm_sync(codec);
+}
+
+static int palm27x_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->socdev->codec;
+
+	/* check the jack status at stream startup */
+	palm27x_ext_control(codec);
+	return 0;
+}
+
+static struct snd_soc_ops palm27x_ops = {
+	.startup = palm27x_startup,
+};
+
+static irqreturn_t palm27x_interrupt(int irq, void *v)
+{
+	palm27x_spk_func = gpio_get_value(palm27x_ep_gpio);
+	palm27x_jack_func = !palm27x_spk_func;
+	return IRQ_HANDLED;
+}
+
+static int palm27x_get_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = palm27x_jack_func;
+	return 0;
+}
+
+static int palm27x_set_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (palm27x_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	palm27x_jack_func = ucontrol->value.integer.value[0];
+	palm27x_ext_control(codec);
+	return 1;
+}
+
+static int palm27x_get_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = palm27x_spk_func;
+	return 0;
+}
+
+static int palm27x_set_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (palm27x_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	palm27x_spk_func = ucontrol->value.integer.value[0];
+	palm27x_ext_control(codec);
+	return 1;
+}
+
+/* PalmTX machine dapm widgets */
+static const struct snd_soc_dapm_widget palm27x_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/* PalmTX audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* headphone connected to HPOUTL, HPOUTR */
+	{"Headphone Jack", NULL, "HPOUTL"},
+	{"Headphone Jack", NULL, "HPOUTR"},
+
+	/* ext speaker connected to ROUT2, LOUT2 */
+	{"Speaker", NULL, "LOUT2"},
+	{"Speaker", NULL, "ROUT2"},
+};
+
+static const char *jack_function[] = {"Headphone", "Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum palm27x_enum[] = {
+	SOC_ENUM_SINGLE_EXT(2, jack_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new palm27x_controls[] = {
+	SOC_ENUM_EXT("Jack Function", palm27x_enum[0], palm27x_get_jack,
+		palm27x_set_jack),
+	SOC_ENUM_EXT("Speaker Function", palm27x_enum[1], palm27x_get_spk,
+		palm27x_set_spk),
+};
+
+static int palm27x_ac97_init(struct snd_soc_codec *codec)
+{
+	int i, err;
+
+	snd_soc_dapm_nc_pin(codec, "OUT3");
+	snd_soc_dapm_nc_pin(codec, "MONOOUT");
+
+	/* add palm27x specific controls */
+	for (i = 0; i < ARRAY_SIZE(palm27x_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				snd_soc_cnew(&palm27x_controls[i],
+						codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	/* add palm27x specific widgets */
+	snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets,
+				ARRAY_SIZE(palm27x_dapm_widgets));
+
+	/* set up palm27x specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_sync(codec);
+	return 0;
+}
+
+static struct snd_soc_dai_link palm27x_dai[] = {
+{
+	.name = "AC97 HiFi",
+	.stream_name = "AC97 HiFi",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+	.init = palm27x_ac97_init,
+	.ops = &palm27x_ops,
+},
+{
+	.name = "AC97 Aux",
+	.stream_name = "AC97 Aux",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+	.ops = &palm27x_ops,
+},
+};
+
+static struct snd_soc_machine palm27x_asoc = {
+	.name = "Palm/PXA27x",
+	.dai_link = palm27x_dai,
+	.num_links = ARRAY_SIZE(palm27x_dai),
+};
+
+static struct snd_soc_device palm27x_snd_devdata = {
+	.machine = &palm27x_asoc,
+	.platform = &pxa2xx_soc_platform,
+	.codec_dev = &soc_codec_dev_wm9712,
+};
+
+static struct platform_device *palm27x_snd_device;
+
+static int __init palm27x_asoc_init(void)
+{
+	int ret;
+
+	if (!(machine_is_palmtx() || machine_is_palmt5() ||
+		machine_is_palmld()))
+		return -ENODEV;
+
+	ret = gpio_request(palm27x_ep_gpio, "Headphone Jack");
+	if (ret)
+		return ret;
+	ret = gpio_direction_input(palm27x_ep_gpio);
+	if (ret)
+		goto err_alloc;
+
+	if (request_irq(gpio_to_irq(palm27x_ep_gpio), palm27x_interrupt,
+			IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING,
+			"Headphone jack", NULL))
+		goto err_alloc;
+
+	palm27x_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!palm27x_snd_device) {
+		ret = -ENOMEM;
+		goto err_dev;
+	}
+
+	platform_set_drvdata(palm27x_snd_device, &palm27x_snd_devdata);
+	palm27x_snd_devdata.dev = &palm27x_snd_device->dev;
+	ret = platform_device_add(palm27x_snd_device);
+
+	if (ret != 0)
+		goto put_device;
+
+	return 0;
+
+put_device:
+	platform_device_put(palm27x_snd_device);
+err_dev:
+	free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
+err_alloc:
+	gpio_free(palm27x_ep_gpio);
+
+	return ret;
+}
+
+static void __exit palm27x_asoc_exit(void)
+{
+	free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
+	gpio_free(palm27x_ep_gpio);
+	platform_device_unregister(palm27x_snd_device);
+}
+
+void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data)
+{
+	palm27x_ep_gpio = data->jack_gpio;
+}
+
+module_init(palm27x_asoc_init);
+module_exit(palm27x_asoc_exit);
+
+/* Module information */
+MODULE_AUTHOR("Marek Vasut <[email protected]>");
+MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive");
+MODULE_LICENSE("GPL");
#6 lore raw

Re: [PATCH 2/4] ASoC: Add Palm/PXA27x unified ASoC audio driver

In reply to Mark Brown

At Wed, 5 Nov 2008 18:53:29 +0000,

Quoted 27 lines · Mark Brown

From: Marek Vasut <[email protected]>

this patch adds asoc audio driver for pxa27x based Palm PDAs. I tested it for palmtx, t5 and ld, it should work with palmz72 as well (slapin, please test). I sent it here some time ago, but now I got to fixing bugs in it. It should be somehow mostly ok and ready for applying.

[Converted to use snd_soc_dapm_nc_pin() -- broonie]

---
 arch/arm/mach-pxa/include/mach/palmasoc.h |   13 ++
 sound/soc/pxa/Kconfig                     |    9 +
 sound/soc/pxa/Makefile                    |    2 +
 sound/soc/pxa/palm27x.c                   |  269 +++++++++++++++++++++++++++++
 4 files changed, 293 insertions(+), 0 deletions(-)
 create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h
 create mode 100644 sound/soc/pxa/palm27x.c
arch/arm/mach-pxa/include/mach/palmasoc.h new file +0 -0
new file mode 100644
index 0000000..6c4b1f7

arch/arm/mach-pxa/include/mach/palmasoc.h line 8

Quoted 8 lines
arch/arm/mach-pxa/include/mach/palmasoc.h new file +8 -0
@@ -0,0 +1,13 @@
+#ifndef _INCLUDE_PALMASOC_H_
+#define _INCLUDE_PALMASOC_H_
+struct palm27x_asoc_info {
+	int	jack_gpio;
+};
+
+#ifdef CONFIG_SND_PXA2XX_SOC_PALM27X
+void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data);

I wonder whether it's really __init.
Note that the driver can be a module in theory.

Takashi

author
#7 lore raw

Re: [PATCH 2/4] ASoC: Add Palm/PXA27x unified ASoC audio driver

Quoted 4 lines · Takashi Iwai

arch/arm/mach-pxa/include/mach/palmasoc.h line 8

Quoted 2 lines
arch/arm/mach-pxa/include/mach/palmasoc.h new file +2 -0
+#ifdef CONFIG_SND_PXA2XX_SOC_PALM27X
+void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data);

I wonder whether it's really __init.

Yes. This will be set up by the machine init code which has to be built in. It should be platform data but we can't support that yet.

Quoted 1 line

Note that the driver can be a module in theory.

Yeah, that practice won't actually build so should probably be masked out in Kconfig.

#8 lore raw

Re: [PATCH 1/4] ASoC: TLV320AIC23B Support more sample rates

In reply to Mark Brown

At Wed, 5 Nov 2008 18:53:28 +0000,

Quoted 17 lines · Mark Brown

From: Troy Kisky <[email protected]>

Add support for more sample rates, different crystals and split playback/capture rates.

---
 sound/soc/codecs/tlv320aic23.c |  227 ++++++++++++++++++++++++++++++++--------
 1 files changed, 182 insertions(+), 45 deletions(-)
sound/soc/codecs/tlv320aic23.c +0 -0
index 44308da..c1b2060 100644

sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 305

Quoted 2 lines
sound/soc/codecs/tlv320aic23.c +2 -0
+unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
+		int dac, int dac_l, int dac_h, int need_dac)

Should be static.

Quoted 4 lines
sound/soc/codecs/tlv320aic23.c +4 -0
+{
+	if ((adc >= adc_l) && (adc <= adc_h) &&
+			(dac >= dac_l) && (dac <= dac_h)) {
+		unsigned diff_adc = need_adc - adc;

sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 319

Quoted 10 lines
sound/soc/codecs/tlv320aic23.c +10 -0
+		unsigned diff_dac = need_dac - dac;
+		unsigned score;
+		if (((int)diff_adc) < 0)
+			diff_adc = -diff_adc;
+		if (((int)diff_dac) < 0)
+			diff_dac = -diff_dac;
+		score = diff_adc + diff_dac;
+		return score;
+	}
+	return 0xffffffff;

-1 would be simpler and safer (for possible missing f's).

sound/soc/codecs/tlv320aic23.c

Quoted 2 lines
sound/soc/codecs/tlv320aic23.c +2 -0
+}
+int find_rate(int mclk, u32 need_adc, u32 need_dac)

Use static.

sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 327

Quoted 6 lines
sound/soc/codecs/tlv320aic23.c +6 -0
+{
+	int i, j;
+	int best_i = -1;
+	int best_j = -1;
+	int best_div = 0;
+	unsigned best_score = 0xffffffff;

-1 would be safer.

Quoted 62 lines
sound/soc/codecs/tlv320aic23.c +62 -0
+	int adc_l, adc_h, dac_l, dac_h;
+
+	need_adc *= SR_MULT;
+	need_dac *= SR_MULT;
+	/*
+	 * rates given are +/- 1/32
+	 */
+	adc_l = need_adc - (need_adc >> 5);
+	adc_h = need_adc + (need_adc >> 5);
+	dac_l = need_dac - (need_dac >> 5);
+	dac_h = need_dac + (need_dac >> 5);
+	for (i = 0; i < 4; i++) {
+		int base = mclk / bosr_usb_divisor_table[i];
+		int mask = sr_valid_mask[i];
+		for (j = 0; j < 16; j++, mask >>= 1) {
+			int adc;
+			int dac;
+			int score;
+			if ((mask & 1) == 0)
+				continue;
+			adc = base * sr_adc_mult_table[j];
+			dac = base * sr_dac_mult_table[j];
+			score = get_score(adc, adc_l, adc_h, need_adc,
+					dac, dac_l, dac_h, need_dac);
+			if (best_score > score) {
+				best_score = score;
+				best_i = i;
+				best_j = j;
+				best_div = 0;
+			}
+			score = get_score((adc >> 1), adc_l, adc_h, need_adc,
+					(dac >> 1), dac_l, dac_h, need_dac);
+			/* prefer to have a /2 */
+			if ((score != 0xffffffff) && (best_score >= score)) {
+				best_score = score;
+				best_i = i;
+				best_j = j;
+				best_div = 1;
+			}
+		}
+	}
+	return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT);
+}
+
+static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk,
+		u32 *sample_rate_adc, u32 *sample_rate_dac)
+{
+	int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE);
+	int sr = (src >> 2) & 0x0f;
+	int val = (mclk / bosr_usb_divisor_table[src & 3]);
+	int adc = (val * sr_adc_mult_table[sr]) / SR_MULT;
+	int dac = (val * sr_dac_mult_table[sr]) / SR_MULT;
+	if (src & TLV320AIC23_CLKIN_HALF) {
+		adc >>= 1;
+		dac >>= 1;
+	}
+	*sample_rate_adc = adc;
+	*sample_rate_dac = dac;
+}
+
+static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk,
+		u32 sample_rate_adc, u32 sample_rate_dac)

sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 399

Quoted 10 lines
sound/soc/codecs/tlv320aic23.c +10 -0
+{
+	/* Search for the right sample rate */
+	int data = find_rate(mclk, sample_rate_adc, sample_rate_dac);
+	if (data < 0) {
+		printk(KERN_ERR "%s:Invalid rate %u,%u requested\n",
+				__func__, sample_rate_adc, sample_rate_dac);
+		return -EINVAL;
+	}
+	tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
+	if (1) {

For what?

Takashi

#9 lore raw

Re: [PATCH 1/4] ASoC: TLV320AIC23B Support more sample rates

Quoted 5 lines · Takashi Iwai

sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 399

Quoted 1 line · Mark Brown
sound/soc/codecs/tlv320aic23.c +1 -0
+	if (1) {

For what?

That should be an ifdef DEBUG - sorry, looks like I've sent an old version of the patch. My bad, sorry. Troy, when addressing Takashi's other comments please send an incremental patch.

#10 lore raw

PATCH 1/1 ASoC: TLV320AIC23B Support more sample rates

Add support for more sample rates, different crystals and split playback/capture rates.

Mark, this is an incremental patch to the #ifdef DEBUG version

sound/soc/codecs/tlv320aic23.c +9 -14
index 0096842..8a9ef21 100644
@@ -301,30 +301,25 @@ static const unsigned char sr_dac_mult_table[] = {
 	A(2), A(11), A(2), A(11),  A(0), A(0), A(0), A(1)
 };
 
-unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
+static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
 		int dac, int dac_l, int dac_h, int need_dac)
 {
 	if ((adc >= adc_l) && (adc <= adc_h) &&
 			(dac >= dac_l) && (dac <= dac_h)) {
-		unsigned diff_adc = need_adc - adc;
-		unsigned diff_dac = need_dac - dac;
-		unsigned score;
-		if (((int)diff_adc) < 0)
-			diff_adc = -diff_adc;
-		if (((int)diff_dac) < 0)
-			diff_dac = -diff_dac;
-		score = diff_adc + diff_dac;
-		return score;
+		int diff_adc = need_adc - adc;
+		int diff_dac = need_dac - dac;
+		return abs(diff_adc) + abs(diff_dac);
 	}
-	return 0xffffffff;
+	return UINT_MAX;
 }
-int find_rate(int mclk, u32 need_adc, u32 need_dac)
+
+static int find_rate(int mclk, u32 need_adc, u32 need_dac)
 {
 	int i, j;
 	int best_i = -1;
 	int best_j = -1;
 	int best_div = 0;
-	unsigned best_score = 0xffffffff;
+	unsigned best_score = UINT_MAX;
 	int adc_l, adc_h, dac_l, dac_h;
 
 	need_adc *= SR_MULT;
@@ -359,7 +354,7 @@ int find_rate(int mclk, u32 need_adc, u32 need_dac)
 			score = get_score((adc >> 1), adc_l, adc_h, need_adc,
 					(dac >> 1), dac_l, dac_h, need_dac);
 			/* prefer to have a /2 */
-			if ((score != 0xffffffff) && (best_score >= score)) {
+			if ((score != UINT_MAX) && (best_score >= score)) {
 				best_score = score;
 				best_i = i;
 				best_j = j;
Signature · 1 line
1.5.4.3
Acked-by on v1 1/1
#11 lore raw

Re: [PATCH 1/1] [ALSA] ASoC: TLV320AIC23B Support more sample rates

Quoted 6 lines · Troy Kisky

Add support for more sample rates, different crystals and split playback/capture rates.

Mark, this is an incremental patch to the #ifdef DEBUG version

I'll squash this down and push it today or Monday.

#12 lore raw

Re: [PATCH 1/4] ASoC: TLV320AIC23B Support more sample rates

In reply to Takashi Iwai
Quoted 7 lines · Takashi Iwai

sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 319

Quoted 3 lines · Mark Brown
sound/soc/codecs/tlv320aic23.c +2 -0
+		return score;
 
+	return 0xffffffff;

-1 would be simpler and safer (for possible missing f's).

I'm not sure that'd add to the clarity - the goal isn't to have all bits set, the goal is to have a much larger score than could ever otherwise be achieved so putting in a very large constant expresses that much more directly.

#13 lore raw

Re: [PATCH 1/4] ASoC: TLV320AIC23B Support more sample rates

At Thu, 6 Nov 2008 10:49:06 +0000,

Quoted 6 lines · Mark Brown
Quoted 3 lines · Takashi Iwai
Quoted 1 line · Mark Brown
sound/soc/codecs/tlv320aic23.c +1 -0
+		return score;
Quoted 8 lines
Quoted 3 lines

sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 319

Quoted 1 line
sound/soc/codecs/tlv320aic23.c +1 -0
+	return 0xffffffff;

-1 would be simpler and safer (for possible missing f's).

I'm not sure that'd add to the clarity - the goal isn't to have all bits set, the goal is to have a much larger score than could ever otherwise be achieved so putting in a very large constant expresses that much more directly.

In that case, the best form is UINT_MAX.

Takashi

#14 lore raw

Re: [PATCH 1/4] ASoC: TLV320AIC23B Support more sample rates

In reply to Mark Brown

At Thu, 6 Nov 2008 10:49:06 +0000,

Quoted 6 lines · Mark Brown
Quoted 3 lines · Takashi Iwai
Quoted 1 line · Mark Brown
sound/soc/codecs/tlv320aic23.c +1 -0
+		return score;
Quoted 8 lines
Quoted 3 lines

sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 319

Quoted 1 line
sound/soc/codecs/tlv320aic23.c +1 -0
+	return 0xffffffff;

-1 would be simpler and safer (for possible missing f's).

I'm not sure that'd add to the clarity - the goal isn't to have all bits set, the goal is to have a much larger score than could ever otherwise be achieved so putting in a very large constant expresses that much more directly.

Also, regarding unsigned integer:

Quoted 4 lines
sound/soc/codecs/tlv320aic23.c +4 -0
+unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
+		int dac, int dac_l, int dac_h, int need_dac)
+{
+	if ((adc >= adc_l) && (adc <= adc_h) &&

sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 317

Quoted 10 lines
sound/soc/codecs/tlv320aic23.c +10 -0
+			(dac >= dac_l) && (dac <= dac_h)) {
+		unsigned diff_adc = need_adc - adc;
+		unsigned diff_dac = need_dac - dac;
+		unsigned score;
+		if (((int)diff_adc) < 0)
+			diff_adc = -diff_adc;
+		if (((int)diff_dac) < 0)
+			diff_dac = -diff_dac;
+		score = diff_adc + diff_dac;
+		return score;

It's more readable when diff_adc, diff_dac are int so that you have no ugly cast:

		int diff_adc = need_adc - adc;
		int diff_dac = need_dac - dac;
		unsigned score;
		if (diff_adc < 0)
			diff_adc = -diff_adc;
		if (diff_dac < 0)
			diff_dac = -diff_dac;
		score = diff_adc + diff_dac;
		return score;

And, even more readable with abs() (not surprisingly defined in linux/kernel.h):

		int diff_adc = need_adc - adc;
		int diff_dac = need_dac - dac;
		return abs(diff_adc) + abs(diff_dac);

Takashi