Merge branch 'topic/asoc' into for-linus
diff --git a/Documentation/sound/alsa/soc/jack.txt b/Documentation/sound/alsa/soc/jack.txt
new file mode 100644
index 0000000..fcf82a4
--- /dev/null
+++ b/Documentation/sound/alsa/soc/jack.txt
@@ -0,0 +1,71 @@
+ASoC jack detection
+===================
+
+ALSA has a standard API for representing physical jacks to user space,
+the kernel side of which can be seen in include/sound/jack.h. ASoC
+provides a version of this API adding two additional features:
+
+ - It allows more than one jack detection method to work together on one
+ user visible jack. In embedded systems it is common for multiple
+ to be present on a single jack but handled by separate bits of
+ hardware.
+
+ - Integration with DAPM, allowing DAPM endpoints to be updated
+ automatically based on the detected jack status (eg, turning off the
+ headphone outputs if no headphones are present).
+
+This is done by splitting the jacks up into three things working
+together: the jack itself represented by a struct snd_soc_jack, sets of
+snd_soc_jack_pins representing DAPM endpoints to update and blocks of
+code providing jack reporting mechanisms.
+
+For example, a system may have a stereo headset jack with two reporting
+mechanisms, one for the headphone and one for the microphone. Some
+systems won't be able to use their speaker output while a headphone is
+connected and so will want to make sure to update both speaker and
+headphone when the headphone jack status changes.
+
+The jack - struct snd_soc_jack
+==============================
+
+This represents a physical jack on the system and is what is visible to
+user space. The jack itself is completely passive, it is set up by the
+machine driver and updated by jack detection methods.
+
+Jacks are created by the machine driver calling snd_soc_jack_new().
+
+snd_soc_jack_pin
+================
+
+These represent a DAPM pin to update depending on some of the status
+bits supported by the jack. Each snd_soc_jack has zero or more of these
+which are updated automatically. They are created by the machine driver
+and associated with the jack using snd_soc_jack_add_pins(). The status
+of the endpoint may configured to be the opposite of the jack status if
+required (eg, enabling a built in microphone if a microphone is not
+connected via a jack).
+
+Jack detection methods
+======================
+
+Actual jack detection is done by code which is able to monitor some
+input to the system and update a jack by calling snd_soc_jack_report(),
+specifying a subset of bits to update. The jack detection code should
+be set up by the machine driver, taking configuration for the jack to
+update and the set of things to report when the jack is connected.
+
+Often this is done based on the status of a GPIO - a handler for this is
+provided by the snd_soc_jack_add_gpio() function. Other methods are
+also available, for example integrated into CODECs. One example of
+CODEC integrated jack detection can be see in the WM8350 driver.
+
+Each jack may have multiple reporting mechanisms, though it will need at
+least one to be useful.
+
+Machine drivers
+===============
+
+These are all hooked together by the machine driver depending on the
+system hardware. The machine driver will set up the snd_soc_jack and
+the list of pins to update then set up one or more jack detection
+mechanisms to update that jack based on their current status.
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index 7793d2a..0afd1a8 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -238,6 +238,8 @@
bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97)
{
+ unsigned long gsr;
+
#ifdef CONFIG_PXA25x
if (cpu_is_pxa25x())
pxa_ac97_warm_pxa25x();
@@ -254,10 +256,10 @@
else
#endif
BUG();
-
- if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) {
+ gsr = GSR | gsr_bits;
+ if (!(gsr & (GSR_PCR | GSR_SCR))) {
printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n",
- __func__, gsr_bits);
+ __func__, gsr);
return false;
}
@@ -268,6 +270,8 @@
bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97)
{
+ unsigned long gsr;
+
#ifdef CONFIG_PXA25x
if (cpu_is_pxa25x())
pxa_ac97_cold_pxa25x();
@@ -285,9 +289,10 @@
#endif
BUG();
- if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) {
+ gsr = GSR | gsr_bits;
+ if (!(gsr & (GSR_PCR | GSR_SCR))) {
printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n",
- __func__, gsr_bits);
+ __func__, gsr);
return false;
}
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 97738e2..bfda7a8 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -122,6 +122,9 @@
unsigned int bypass_state;
unsigned int codec_powered;
unsigned int codec_muted;
+
+ struct snd_pcm_substream *master_substream;
+ struct snd_pcm_substream *slave_substream;
};
/*
@@ -1217,6 +1220,50 @@
return 0;
}
+static int twl4030_startup(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 twl4030_priv *twl4030 = codec->private_data;
+
+ /* If we already have a playback or capture going then constrain
+ * this substream to match it.
+ */
+ if (twl4030->master_substream) {
+ struct snd_pcm_runtime *master_runtime;
+ master_runtime = twl4030->master_substream->runtime;
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ master_runtime->rate,
+ master_runtime->rate);
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ master_runtime->sample_bits,
+ master_runtime->sample_bits);
+
+ twl4030->slave_substream = substream;
+ } else
+ twl4030->master_substream = substream;
+
+ return 0;
+}
+
+static void twl4030_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 twl4030_priv *twl4030 = codec->private_data;
+
+ if (twl4030->master_substream == substream)
+ twl4030->master_substream = twl4030->slave_substream;
+
+ twl4030->slave_substream = NULL;
+}
+
static int twl4030_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -1224,8 +1271,13 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
+ struct twl4030_priv *twl4030 = codec->private_data;
u8 mode, old_mode, format, old_format;
+ if (substream == twl4030->slave_substream)
+ /* Ignoring hw_params for slave substream */
+ return 0;
+
/* bit rate */
old_mode = twl4030_read_reg_cache(codec,
TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
@@ -1259,6 +1311,9 @@
case 48000:
mode |= TWL4030_APLL_RATE_48000;
break;
+ case 96000:
+ mode |= TWL4030_APLL_RATE_96000;
+ break;
default:
printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n",
params_rate(params));
@@ -1384,6 +1439,8 @@
#define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
static struct snd_soc_dai_ops twl4030_dai_ops = {
+ .startup = twl4030_startup,
+ .shutdown = twl4030_shutdown,
.hw_params = twl4030_hw_params,
.set_sysclk = twl4030_set_dai_sysclk,
.set_fmt = twl4030_set_dai_fmt,
@@ -1395,7 +1452,7 @@
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
- .rates = TWL4030_RATES,
+ .rates = TWL4030_RATES | SNDRV_PCM_RATE_96000,
.formats = TWL4030_FORMATS,},
.capture = {
.stream_name = "Capture",
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
index 33dbb14..cb63765 100644
--- a/sound/soc/codecs/twl4030.h
+++ b/sound/soc/codecs/twl4030.h
@@ -109,6 +109,7 @@
#define TWL4030_APLL_RATE_32000 0x80
#define TWL4030_APLL_RATE_44100 0x90
#define TWL4030_APLL_RATE_48000 0xA0
+#define TWL4030_APLL_RATE_96000 0xE0
#define TWL4030_SEL_16K 0x04
#define TWL4030_CODECPDZ 0x02
#define TWL4030_OPT_MODE 0x01
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index 3265817..6e23a81 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -317,6 +317,41 @@
return -EIO;
}
+#ifdef CONFIG_PM
+static int wm9705_soc_suspend(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff);
+
+ return 0;
+}
+
+static int wm9705_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ int i, ret;
+ u16 *cache = codec->reg_cache;
+
+ ret = wm9705_reset(codec);
+ if (ret < 0) {
+ printk(KERN_ERR "could not reset AC97 codec\n");
+ return ret;
+ }
+
+ for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) {
+ soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
+ }
+
+ return 0;
+}
+#else
+#define wm9705_soc_suspend NULL
+#define wm9705_soc_resume NULL
+#endif
+
static int wm9705_soc_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -407,6 +442,8 @@
struct snd_soc_codec_device soc_codec_dev_wm9705 = {
.probe = wm9705_soc_probe,
.remove = wm9705_soc_remove,
+ .suspend = wm9705_soc_suspend,
+ .resume = wm9705_soc_resume,
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705);
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index b3eb857..2c4892c 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -697,6 +697,23 @@
else
position = in_be32(&dma_channel->dar);
+ /*
+ * When capture is started, the SSI immediately starts to fill its FIFO.
+ * This means that the DMA controller is not started until the FIFO is
+ * full. However, ALSA calls this function before that happens, when
+ * MR.DAR is still zero. In this case, just return zero to indicate
+ * that nothing has been received yet.
+ */
+ if (!position)
+ return 0;
+
+ if ((position < dma_private->dma_buf_phys) ||
+ (position > dma_private->dma_buf_end)) {
+ dev_err(substream->pcm->card->dev,
+ "dma pointer is out of range, halting stream\n");
+ return SNDRV_PCM_POS_XRUN;
+ }
+
frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys);
/*
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 169bca2..3711d84 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -60,6 +60,13 @@
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE)
#endif
+/* SIER bitflag of interrupts to enable */
+#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \
+ CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \
+ CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \
+ CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \
+ CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN)
+
/**
* fsl_ssi_private: per-SSI private data
*
@@ -140,7 +147,7 @@
were interrupted for. We mask it with the Interrupt Enable register
so that we only check for events that we're interested in.
*/
- sisr = in_be32(&ssi->sisr) & in_be32(&ssi->sier);
+ sisr = in_be32(&ssi->sisr) & SIER_FLAGS;
if (sisr & CCSR_SSI_SISR_RFRC) {
ssi_private->stats.rfrc++;
@@ -324,12 +331,7 @@
*/
/* 4. Enable the interrupts and DMA requests */
- out_be32(&ssi->sier,
- CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE |
- CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN |
- CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN |
- CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE |
- CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN);
+ out_be32(&ssi->sier, SIER_FLAGS);
/*
* Set the watermark for transmit FIFI 0 and receive FIFO 0. We
@@ -466,28 +468,12 @@
case SNDRV_PCM_TRIGGER_START:
clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
setbits32(&ssi->scr,
CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
- } else {
- long timeout = jiffies + 10;
-
+ else
setbits32(&ssi->scr,
CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
-
- /* Wait until the SSI has filled its FIFO. Without this
- * delay, ALSA complains about overruns. When the FIFO
- * is full, the DMA controller initiates its first
- * transfer. Until then, however, the DMA's DAR
- * register is zero, which translates to an
- * out-of-bounds pointer. This makes ALSA think an
- * overrun has occurred.
- */
- while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) &&
- (jiffies < timeout));
- if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0))
- return -EIO;
- }
break;
case SNDRV_PCM_TRIGGER_STOP:
@@ -606,39 +592,52 @@
.ops = &fsl_ssi_dai_ops,
};
+/* Show the statistics of a flag only if its interrupt is enabled. The
+ * compiler will optimze this code to a no-op if the interrupt is not
+ * enabled.
+ */
+#define SIER_SHOW(flag, name) \
+ do { \
+ if (SIER_FLAGS & CCSR_SSI_SIER_##flag) \
+ length += sprintf(buf + length, #name "=%u\n", \
+ ssi_private->stats.name); \
+ } while (0)
+
+
/**
* fsl_sysfs_ssi_show: display SSI statistics
*
- * Display the statistics for the current SSI device.
+ * Display the statistics for the current SSI device. To avoid confusion,
+ * we only show those counts that are enabled.
*/
static ssize_t fsl_sysfs_ssi_show(struct device *dev,
struct device_attribute *attr, char *buf)
{
struct fsl_ssi_private *ssi_private =
- container_of(attr, struct fsl_ssi_private, dev_attr);
- ssize_t length;
+ container_of(attr, struct fsl_ssi_private, dev_attr);
+ ssize_t length = 0;
- length = sprintf(buf, "rfrc=%u", ssi_private->stats.rfrc);
- length += sprintf(buf + length, "\ttfrc=%u", ssi_private->stats.tfrc);
- length += sprintf(buf + length, "\tcmdau=%u", ssi_private->stats.cmdau);
- length += sprintf(buf + length, "\tcmddu=%u", ssi_private->stats.cmddu);
- length += sprintf(buf + length, "\trxt=%u", ssi_private->stats.rxt);
- length += sprintf(buf + length, "\trdr1=%u", ssi_private->stats.rdr1);
- length += sprintf(buf + length, "\trdr0=%u", ssi_private->stats.rdr0);
- length += sprintf(buf + length, "\ttde1=%u", ssi_private->stats.tde1);
- length += sprintf(buf + length, "\ttde0=%u", ssi_private->stats.tde0);
- length += sprintf(buf + length, "\troe1=%u", ssi_private->stats.roe1);
- length += sprintf(buf + length, "\troe0=%u", ssi_private->stats.roe0);
- length += sprintf(buf + length, "\ttue1=%u", ssi_private->stats.tue1);
- length += sprintf(buf + length, "\ttue0=%u", ssi_private->stats.tue0);
- length += sprintf(buf + length, "\ttfs=%u", ssi_private->stats.tfs);
- length += sprintf(buf + length, "\trfs=%u", ssi_private->stats.rfs);
- length += sprintf(buf + length, "\ttls=%u", ssi_private->stats.tls);
- length += sprintf(buf + length, "\trls=%u", ssi_private->stats.rls);
- length += sprintf(buf + length, "\trff1=%u", ssi_private->stats.rff1);
- length += sprintf(buf + length, "\trff0=%u", ssi_private->stats.rff0);
- length += sprintf(buf + length, "\ttfe1=%u", ssi_private->stats.tfe1);
- length += sprintf(buf + length, "\ttfe0=%u\n", ssi_private->stats.tfe0);
+ SIER_SHOW(RFRC_EN, rfrc);
+ SIER_SHOW(TFRC_EN, tfrc);
+ SIER_SHOW(CMDAU_EN, cmdau);
+ SIER_SHOW(CMDDU_EN, cmddu);
+ SIER_SHOW(RXT_EN, rxt);
+ SIER_SHOW(RDR1_EN, rdr1);
+ SIER_SHOW(RDR0_EN, rdr0);
+ SIER_SHOW(TDE1_EN, tde1);
+ SIER_SHOW(TDE0_EN, tde0);
+ SIER_SHOW(ROE1_EN, roe1);
+ SIER_SHOW(ROE0_EN, roe0);
+ SIER_SHOW(TUE1_EN, tue1);
+ SIER_SHOW(TUE0_EN, tue0);
+ SIER_SHOW(TFS_EN, tfs);
+ SIER_SHOW(RFS_EN, rfs);
+ SIER_SHOW(TLS_EN, tls);
+ SIER_SHOW(RLS_EN, rls);
+ SIER_SHOW(RFF1_EN, rff1);
+ SIER_SHOW(RFF0_EN, rff0);
+ SIER_SHOW(TFE1_EN, tfe1);
+ SIER_SHOW(TFE0_EN, tfe0);
return length;
}
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index d6882be..9c09b94 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -146,6 +146,17 @@
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
int err = 0;
+ if (cpu_is_omap343x() && mcbsp_data->bus_id == 1) {
+ /*
+ * McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer.
+ * Set constraint for minimum buffer size to the same than FIFO
+ * size in order to avoid underruns in playback startup because
+ * HW is keeping the DMA request active until FIFO is filled.
+ */
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096, UINT_MAX);
+ }
+
if (!cpu_dai->active)
err = omap_mcbsp_request(mcbsp_data->bus_id);
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 5998ab3..ad8a10f 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -116,6 +116,16 @@
Say Y if you want to add support for SoC audio on the
Marvell Zylonite reference platform.
+config SND_PXA2XX_SOC_MAGICIAN
+ tristate "SoC Audio support for HTC Magician"
+ depends on SND_PXA2XX_SOC && MACH_MAGICIAN
+ select SND_PXA2XX_SOC_I2S
+ select SND_PXA_SOC_SSP
+ select SND_SOC_UDA1380
+ help
+ Say Y if you want to add support for SoC audio on the
+ HTC Magician.
+
config SND_PXA2XX_SOC_MIOA701
tristate "SoC Audio support for MIO A701"
depends on SND_PXA2XX_SOC && MACH_MIOA701
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 8ed881c..4b90c3c 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -20,6 +20,7 @@
snd-soc-em-x270-objs := em-x270.o
snd-soc-palm27x-objs := palm27x.o
snd-soc-zylonite-objs := zylonite.o
+snd-soc-magician-objs := magician.o
snd-soc-mioa701-objs := mioa701_wm9713.o
obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
@@ -31,5 +32,6 @@
obj-$(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
+obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
new file mode 100644
index 0000000..f7c4544
--- /dev/null
+++ b/sound/soc/pxa/magician.c
@@ -0,0 +1,560 @@
+/*
+ * SoC audio for HTC Magician
+ *
+ * Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com>
+ *
+ * based on spitz.c,
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Richard Purdie <richard@openedhand.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/magician.h>
+#include <asm/mach-types.h>
+#include "../codecs/uda1380.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+#include "pxa-ssp.h"
+
+#define MAGICIAN_MIC 0
+#define MAGICIAN_MIC_EXT 1
+
+static int magician_hp_switch;
+static int magician_spk_switch = 1;
+static int magician_in_sel = MAGICIAN_MIC;
+
+static void magician_ext_control(struct snd_soc_codec *codec)
+{
+ if (magician_spk_switch)
+ snd_soc_dapm_enable_pin(codec, "Speaker");
+ else
+ snd_soc_dapm_disable_pin(codec, "Speaker");
+ if (magician_hp_switch)
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ else
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+
+ switch (magician_in_sel) {
+ case MAGICIAN_MIC:
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_enable_pin(codec, "Call Mic");
+ break;
+ case MAGICIAN_MIC_EXT:
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
+ snd_soc_dapm_enable_pin(codec, "Headset Mic");
+ break;
+ }
+
+ snd_soc_dapm_sync(codec);
+}
+
+static int magician_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->card->codec;
+
+ /* check the jack status at stream startup */
+ magician_ext_control(codec);
+
+ return 0;
+}
+
+/*
+ * Magician uses SSP port for playback.
+ */
+static int magician_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int acps, acds, width, rate;
+ unsigned int div4 = PXA_SSP_CLK_SCDB_4;
+ int ret = 0;
+
+ rate = params_rate(params);
+ width = snd_pcm_format_physical_width(params_format(params));
+
+ /*
+ * rate = SSPSCLK / (2 * width(16 or 32))
+ * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
+ */
+ switch (params_rate(params)) {
+ case 8000:
+ /* off by a factor of 2: bug in the PXA27x audio clock? */
+ acps = 32842000;
+ switch (width) {
+ case 16:
+ /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_16;
+ break;
+ case 32:
+ /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_8;
+ }
+ break;
+ case 11025:
+ acps = 5622000;
+ switch (width) {
+ case 16:
+ /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_4;
+ break;
+ case 32:
+ /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ }
+ break;
+ case 22050:
+ acps = 5622000;
+ switch (width) {
+ case 16:
+ /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ break;
+ case 32:
+ /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_1;
+ }
+ break;
+ case 44100:
+ acps = 5622000;
+ switch (width) {
+ case 16:
+ /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ break;
+ case 32:
+ /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_1;
+ }
+ break;
+ case 48000:
+ acps = 12235000;
+ switch (width) {
+ case 16:
+ /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ break;
+ case 32:
+ /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_1;
+ }
+ break;
+ case 96000:
+ acps = 12235000;
+ switch (width) {
+ case 16:
+ /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_1;
+ break;
+ case 32:
+ /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ div4 = PXA_SSP_CLK_SCDB_1;
+ break;
+ }
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1);
+ if (ret < 0)
+ return ret;
+
+ /* set audio clock as clock source */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ /* set the SSP audio system clock ACDS divider */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai,
+ PXA_SSP_AUDIO_DIV_ACDS, acds);
+ if (ret < 0)
+ return ret;
+
+ /* set the SSP audio system clock SCDB divider4 */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai,
+ PXA_SSP_AUDIO_DIV_SCDB, div4);
+ if (ret < 0)
+ return ret;
+
+ /* set SSP audio pll clock */
+ ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+/*
+ * Magician uses I2S for capture.
+ */
+static int magician_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret = 0;
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as output */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops magician_capture_ops = {
+ .startup = magician_startup,
+ .hw_params = magician_capture_hw_params,
+};
+
+static struct snd_soc_ops magician_playback_ops = {
+ .startup = magician_startup,
+ .hw_params = magician_playback_hw_params,
+};
+
+static int magician_get_hp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = magician_hp_switch;
+ return 0;
+}
+
+static int magician_set_hp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (magician_hp_switch == ucontrol->value.integer.value[0])
+ return 0;
+
+ magician_hp_switch = ucontrol->value.integer.value[0];
+ magician_ext_control(codec);
+ return 1;
+}
+
+static int magician_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = magician_spk_switch;
+ return 0;
+}
+
+static int magician_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (magician_spk_switch == ucontrol->value.integer.value[0])
+ return 0;
+
+ magician_spk_switch = ucontrol->value.integer.value[0];
+ magician_ext_control(codec);
+ return 1;
+}
+
+static int magician_get_input(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = magician_in_sel;
+ return 0;
+}
+
+static int magician_set_input(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ if (magician_in_sel == ucontrol->value.integer.value[0])
+ return 0;
+
+ magician_in_sel = ucontrol->value.integer.value[0];
+
+ switch (magician_in_sel) {
+ case MAGICIAN_MIC:
+ gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1);
+ break;
+ case MAGICIAN_MIC_EXT:
+ gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0);
+ }
+
+ return 1;
+}
+
+static int magician_spk_power(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+static int magician_hp_power(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+static int magician_mic_bias(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+/* magician machine dapm widgets */
+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
+ SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
+ SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias),
+ SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias),
+};
+
+/* magician machine audio_map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* Headphone connected to VOUTL, VOUTR */
+ {"Headphone Jack", NULL, "VOUTL"},
+ {"Headphone Jack", NULL, "VOUTR"},
+
+ /* Speaker connected to VOUTL, VOUTR */
+ {"Speaker", NULL, "VOUTL"},
+ {"Speaker", NULL, "VOUTR"},
+
+ /* Mics are connected to VINM */
+ {"VINM", NULL, "Headset Mic"},
+ {"VINM", NULL, "Call Mic"},
+};
+
+static const char *input_select[] = {"Call Mic", "Headset Mic"};
+static const struct soc_enum magician_in_sel_enum =
+ SOC_ENUM_SINGLE_EXT(2, input_select);
+
+static const struct snd_kcontrol_new uda1380_magician_controls[] = {
+ SOC_SINGLE_BOOL_EXT("Headphone Switch",
+ (unsigned long)&magician_hp_switch,
+ magician_get_hp, magician_set_hp),
+ SOC_SINGLE_BOOL_EXT("Speaker Switch",
+ (unsigned long)&magician_spk_switch,
+ magician_get_spk, magician_set_spk),
+ SOC_ENUM_EXT("Input Select", magician_in_sel_enum,
+ magician_get_input, magician_set_input),
+};
+
+/*
+ * Logic for a uda1380 as connected on a HTC Magician
+ */
+static int magician_uda1380_init(struct snd_soc_codec *codec)
+{
+ int err;
+
+ /* NC codec pins */
+ snd_soc_dapm_nc_pin(codec, "VOUTLHP");
+ snd_soc_dapm_nc_pin(codec, "VOUTRHP");
+
+ /* FIXME: is anything connected here? */
+ snd_soc_dapm_nc_pin(codec, "VINL");
+ snd_soc_dapm_nc_pin(codec, "VINR");
+
+ /* Add magician specific controls */
+ err = snd_soc_add_controls(codec, uda1380_magician_controls,
+ ARRAY_SIZE(uda1380_magician_controls));
+ if (err < 0)
+ return err;
+
+ /* Add magician specific widgets */
+ snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
+ ARRAY_SIZE(uda1380_dapm_widgets));
+
+ /* Set up magician specific audio path interconnects */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_sync(codec);
+ return 0;
+}
+
+/* magician digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link magician_dai[] = {
+{
+ .name = "uda1380",
+ .stream_name = "UDA1380 Playback",
+ .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1],
+ .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK],
+ .init = magician_uda1380_init,
+ .ops = &magician_playback_ops,
+},
+{
+ .name = "uda1380",
+ .stream_name = "UDA1380 Capture",
+ .cpu_dai = &pxa_i2s_dai,
+ .codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE],
+ .ops = &magician_capture_ops,
+}
+};
+
+/* magician audio machine driver */
+static struct snd_soc_card snd_soc_card_magician = {
+ .name = "Magician",
+ .dai_link = magician_dai,
+ .num_links = ARRAY_SIZE(magician_dai),
+ .platform = &pxa2xx_soc_platform,
+};
+
+/* magician audio private data */
+static struct uda1380_setup_data magician_uda1380_setup = {
+ .i2c_address = 0x18,
+ .dac_clk = UDA1380_DAC_CLK_WSPLL,
+};
+
+/* magician audio subsystem */
+static struct snd_soc_device magician_snd_devdata = {
+ .card = &snd_soc_card_magician,
+ .codec_dev = &soc_codec_dev_uda1380,
+ .codec_data = &magician_uda1380_setup,
+};
+
+static struct platform_device *magician_snd_device;
+
+static int __init magician_init(void)
+{
+ int ret;
+
+ if (!machine_is_magician())
+ return -ENODEV;
+
+ ret = gpio_request(EGPIO_MAGICIAN_CODEC_POWER, "CODEC_POWER");
+ if (ret)
+ goto err_request_power;
+ ret = gpio_request(EGPIO_MAGICIAN_CODEC_RESET, "CODEC_RESET");
+ if (ret)
+ goto err_request_reset;
+ ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER");
+ if (ret)
+ goto err_request_spk;
+ ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER");
+ if (ret)
+ goto err_request_ep;
+ ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER");
+ if (ret)
+ goto err_request_mic;
+ ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0");
+ if (ret)
+ goto err_request_in_sel0;
+ ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1");
+ if (ret)
+ goto err_request_in_sel1;
+
+ gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 1);
+ gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0);
+
+ /* we may need to have the clock running here - pH5 */
+ gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 1);
+ udelay(5);
+ gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 0);
+
+ magician_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!magician_snd_device) {
+ ret = -ENOMEM;
+ goto err_pdev;
+ }
+
+ platform_set_drvdata(magician_snd_device, &magician_snd_devdata);
+ magician_snd_devdata.dev = &magician_snd_device->dev;
+ ret = platform_device_add(magician_snd_device);
+ if (ret) {
+ platform_device_put(magician_snd_device);
+ goto err_pdev;
+ }
+
+ return 0;
+
+err_pdev:
+ gpio_free(EGPIO_MAGICIAN_IN_SEL1);
+err_request_in_sel1:
+ gpio_free(EGPIO_MAGICIAN_IN_SEL0);
+err_request_in_sel0:
+ gpio_free(EGPIO_MAGICIAN_MIC_POWER);
+err_request_mic:
+ gpio_free(EGPIO_MAGICIAN_EP_POWER);
+err_request_ep:
+ gpio_free(EGPIO_MAGICIAN_SPK_POWER);
+err_request_spk:
+ gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
+err_request_reset:
+ gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
+err_request_power:
+ return ret;
+}
+
+static void __exit magician_exit(void)
+{
+ platform_device_unregister(magician_snd_device);
+
+ gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0);
+ gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0);
+ gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0);
+ gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 0);
+
+ gpio_free(EGPIO_MAGICIAN_IN_SEL1);
+ gpio_free(EGPIO_MAGICIAN_IN_SEL0);
+ gpio_free(EGPIO_MAGICIAN_MIC_POWER);
+ gpio_free(EGPIO_MAGICIAN_EP_POWER);
+ gpio_free(EGPIO_MAGICIAN_SPK_POWER);
+ gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
+ gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
+}
+
+module_init(magician_init);
+module_exit(magician_exit);
+
+MODULE_AUTHOR("Philipp Zabel");
+MODULE_DESCRIPTION("ALSA SoC Magician");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 7acd3fe..308a657 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -627,12 +627,18 @@
u32 sscr0;
u32 sspsp;
int width = snd_pcm_format_physical_width(params_format(params));
+ int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf;
/* select correct DMA params */
if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
dma = 1; /* capture DMA offset is 1,3 */
- if (chn == 2)
- dma += 2; /* stereo DMA offset is 2, mono is 0 */
+ /* Network mode with one active slot (ttsa == 1) can be used
+ * to force 16-bit frame width on the wire (for S16_LE), even
+ * with two channels. Use 16-bit DMA transfers for this case.
+ */
+ if (((chn == 2) && (ttsa != 1)) || (width == 32))
+ dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */
+
cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma);
@@ -712,7 +718,7 @@
/* When we use a network mode, we always require TDM slots
* - complain loudly and fail if they've not been set up yet.
*/
- if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) {
+ if ((sscr0 & SSCR0_MOD) && !ttsa) {
dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
return -EINVAL;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6e710f7..99712f6 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -98,7 +98,7 @@
int err;
codec->ac97->dev.bus = &ac97_bus_type;
- codec->ac97->dev.parent = NULL;
+ codec->ac97->dev.parent = codec->card->dev;
codec->ac97->dev.release = soc_ac97_device_release;
dev_set_name(&codec->ac97->dev, "%d-%d:%s",
@@ -767,11 +767,21 @@
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_card *card = socdev->card;
+ struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai;
- dev_dbg(socdev->dev, "scheduling resume work\n");
-
- if (!schedule_work(&card->deferred_resume_work))
- dev_err(socdev->dev, "resume work item may be lost\n");
+ /* AC97 devices might have other drivers hanging off them so
+ * need to resume immediately. Other drivers don't have that
+ * problem and may take a substantial amount of time to resume
+ * due to I/O costs and anti-pop so handle them out of line.
+ */
+ if (cpu_dai->ac97_control) {
+ dev_dbg(socdev->dev, "Resuming AC97 immediately\n");
+ soc_resume_deferred(&card->deferred_resume_work);
+ } else {
+ dev_dbg(socdev->dev, "Scheduling resume work\n");
+ if (!schedule_work(&card->deferred_resume_work))
+ dev_err(socdev->dev, "resume work item may be lost\n");
+ }
return 0;
}