Commit c95d947f authored by Takashi Iwai's avatar Takashi Iwai

Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v3.7

Nothing too exciting except for the ams-delta change which is relatively
lerge due to the fact that the driver loading had been totally broken as
the driver needed a newer API to function.
parents 71aa5ebe 9233ef6b
......@@ -444,16 +444,28 @@ static struct omap1_cam_platform_data ams_delta_camera_platform_data = {
.lclk_khz_max = 1334, /* results in 5fps CIF, 10fps QCIF */
};
static struct platform_device ams_delta_audio_device = {
.name = "ams-delta-audio",
.id = -1,
};
static struct platform_device cx20442_codec_device = {
.name = "cx20442-codec",
.id = -1,
};
static struct platform_device *ams_delta_devices[] __initdata = {
&latch1_gpio_device,
&latch2_gpio_device,
&ams_delta_kp_device,
&ams_delta_camera_device,
&ams_delta_audio_device,
};
static struct platform_device *late_devices[] __initdata = {
&ams_delta_nand_device,
&ams_delta_lcd_device,
&cx20442_codec_device,
};
static void __init ams_delta_init(void)
......
snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o
snd-soc-dmaengine-pcm-objs := soc-dmaengine-pcm.o
obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o
ifneq ($(CONFIG_SND_SOC_DMAENGINE_PCM),)
snd-soc-core-objs += soc-dmaengine-pcm.o
endif
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
obj-$(CONFIG_SND_SOC) += codecs/
......
......@@ -178,6 +178,12 @@
#define DA9055_AIF_WORD_S24_LE (2 << 2)
#define DA9055_AIF_WORD_S32_LE (3 << 2)
/* MIC_L_CTRL bit fields */
#define DA9055_MIC_L_MUTE_EN (1 << 6)
/* MIC_R_CTRL bit fields */
#define DA9055_MIC_R_MUTE_EN (1 << 6)
/* MIXIN_L_CTRL bit fields */
#define DA9055_MIXIN_L_MIX_EN (1 << 3)
......@@ -476,7 +482,7 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
u8 reg_val, adc_left, adc_right;
u8 reg_val, adc_left, adc_right, mic_left, mic_right;
int avg_left_data, avg_right_data, offset_l, offset_r;
if (ucontrol->value.integer.value[0]) {
......@@ -485,6 +491,16 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol,
* offsets must be done first
*/
/* Save current values from Mic control registers */
mic_left = snd_soc_read(codec, DA9055_MIC_L_CTRL);
mic_right = snd_soc_read(codec, DA9055_MIC_R_CTRL);
/* Mute Mic PGA Left and Right */
snd_soc_update_bits(codec, DA9055_MIC_L_CTRL,
DA9055_MIC_L_MUTE_EN, DA9055_MIC_L_MUTE_EN);
snd_soc_update_bits(codec, DA9055_MIC_R_CTRL,
DA9055_MIC_R_MUTE_EN, DA9055_MIC_R_MUTE_EN);
/* Save current values from ADC control registers */
adc_left = snd_soc_read(codec, DA9055_ADC_L_CTRL);
adc_right = snd_soc_read(codec, DA9055_ADC_R_CTRL);
......@@ -520,6 +536,10 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol,
/* Restore original values of ADC control registers */
snd_soc_write(codec, DA9055_ADC_L_CTRL, adc_left);
snd_soc_write(codec, DA9055_ADC_R_CTRL, adc_right);
/* Restore original values of Mic control registers */
snd_soc_write(codec, DA9055_MIC_L_CTRL, mic_left);
snd_soc_write(codec, DA9055_MIC_R_CTRL, mic_right);
}
return snd_soc_put_volsw(kcontrol, ucontrol);
......
......@@ -820,10 +820,10 @@ static const struct snd_soc_dapm_route intercon[] = {
{"VIBRA DAC", NULL, "Vibra Playback"},
/* ADC -> Stream mapping */
{"ADC Left", NULL, "Legacy Capture"},
{"ADC Left", NULL, "Capture"},
{"ADC Right", NULL, "Legacy Capture"},
{"ADC Right", NULL, "Capture"},
{"Legacy Capture" , NULL, "ADC Left"},
{"Capture", NULL, "ADC Left"},
{"Legacy Capture", NULL, "ADC Right"},
{"Capture" , NULL, "ADC Right"},
/* Capture path */
{"Analog Left Capture Route", "Headset Mic", "HSMIC"},
......
......@@ -1028,7 +1028,7 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L,
WM2200_DAC_DIGITAL_VOLUME_2R, WM2200_OUT2L_VOL_SHIFT, 0x9f, 0,
digital_tlv),
SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT,
WM2200_SPK1R_MUTE_SHIFT, 1, 0),
WM2200_SPK1R_MUTE_SHIFT, 1, 1),
};
WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE);
......@@ -2091,6 +2091,7 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c,
switch (wm2200->rev) {
case 0:
case 1:
ret = regmap_register_patch(wm2200->regmap, wm2200_reva_patch,
ARRAY_SIZE(wm2200_reva_patch));
if (ret != 0) {
......
......@@ -575,56 +575,53 @@ static struct snd_soc_card ams_delta_audio_card = {
};
/* Module init/exit */
static struct platform_device *ams_delta_audio_platform_device;
static struct platform_device *cx20442_platform_device;
static int __init ams_delta_module_init(void)
static __devinit int ams_delta_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &ams_delta_audio_card;
int ret;
if (!(machine_is_ams_delta()))
return -ENODEV;
ams_delta_audio_platform_device =
platform_device_alloc("soc-audio", -1);
if (!ams_delta_audio_platform_device)
return -ENOMEM;
card->dev = &pdev->dev;
platform_set_drvdata(ams_delta_audio_platform_device,
&ams_delta_audio_card);
ret = platform_device_add(ams_delta_audio_platform_device);
if (ret)
goto err;
/*
* Codec platform device could be registered from elsewhere (board?),
* but I do it here as it makes sense only if used with the card.
*/
cx20442_platform_device =
platform_device_register_simple("cx20442-codec", -1, NULL, 0);
ret = snd_soc_register_card(card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
card->dev = NULL;
return ret;
}
return 0;
err:
platform_device_put(ams_delta_audio_platform_device);
return ret;
}
late_initcall(ams_delta_module_init);
static void __exit ams_delta_module_exit(void)
static int __devexit ams_delta_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
if (tty_unregister_ldisc(N_V253) != 0)
dev_warn(&ams_delta_audio_platform_device->dev,
dev_warn(&pdev->dev,
"failed to unregister V253 line discipline\n");
snd_soc_jack_free_gpios(&ams_delta_hook_switch,
ARRAY_SIZE(ams_delta_hook_switch_gpios),
ams_delta_hook_switch_gpios);
platform_device_unregister(cx20442_platform_device);
platform_device_unregister(ams_delta_audio_platform_device);
snd_soc_unregister_card(card);
card->dev = NULL;
return 0;
}
module_exit(ams_delta_module_exit);
#define DRV_NAME "ams-delta-audio"
static struct platform_driver ams_delta_driver = {
.driver = {
.name = DRV_NAME,
.owner = THIS_MODULE,
},
.probe = ams_delta_probe,
.remove = __devexit_p(ams_delta_remove),
};
module_platform_driver(ams_delta_driver);
MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>");
MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:" DRV_NAME);
......@@ -216,7 +216,7 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk");
twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk");
twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out");
twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator");
twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator");
twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic");
twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic");
......
......@@ -40,7 +40,6 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <plat/omap_hwmod.h>
#include "omap-mcpdm.h"
#include "omap-pcm.h"
......@@ -260,13 +259,9 @@ static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream,
mutex_lock(&mcpdm->mutex);
if (!dai->active) {
/* Enable watch dog for ES above ES 1.0 to avoid saturation */
if (omap_rev() != OMAP4430_REV_ES1_0) {
u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL,
ctrl | MCPDM_WD_EN);
}
omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl | MCPDM_WD_EN);
omap_mcpdm_open_streams(mcpdm);
}
mutex_unlock(&mcpdm->mutex);
......
......@@ -15,13 +15,13 @@
#include <linux/slab.h>
#include <linux/dma-mapping.h>
#include <linux/dmaengine.h>
#include <linux/platform_data/dma-mmp_tdma.h>
#include <linux/platform_data/mmp_audio.h>
#include <sound/pxa2xx-lib.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <mach/sram.h>
#include <sound/dmaengine_pcm.h>
struct mmp_dma_data {
......
......@@ -212,7 +212,7 @@ static struct snd_soc_dai_link bells_dai_wm5102[] = {
{
.name = "Sub",
.stream_name = "Sub",
.cpu_dai_name = "wm5102-aif3",
.cpu_dai_name = "wm5110-aif3",
.codec_dai_name = "wm9081-hifi",
.codec_name = "wm9081.1-006c",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
......@@ -247,7 +247,7 @@ static struct snd_soc_dai_link bells_dai_wm5110[] = {
{
.name = "Sub",
.stream_name = "Sub",
.cpu_dai_name = "wm5102-aif3",
.cpu_dai_name = "wm5110-aif3",
.codec_dai_name = "wm9081-hifi",
.codec_name = "wm9081.1-006c",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
......
......@@ -20,6 +20,7 @@
#include <linux/sh_dma.h>
#include <linux/slab.h>
#include <linux/module.h>
#include <linux/workqueue.h>
#include <sound/soc.h>
#include <sound/sh_fsi.h>
......@@ -223,7 +224,7 @@ struct fsi_stream {
*/
struct dma_chan *chan;
struct sh_dmae_slave slave; /* see fsi_handler_init() */
struct tasklet_struct tasklet;
struct work_struct work;
dma_addr_t dma;
};
......@@ -1085,9 +1086,9 @@ static void fsi_dma_complete(void *data)
snd_pcm_period_elapsed(io->substream);
}
static void fsi_dma_do_tasklet(unsigned long data)
static void fsi_dma_do_work(struct work_struct *work)
{
struct fsi_stream *io = (struct fsi_stream *)data;
struct fsi_stream *io = container_of(work, struct fsi_stream, work);
struct fsi_priv *fsi = fsi_stream_to_priv(io);
struct snd_soc_dai *dai;
struct dma_async_tx_descriptor *desc;
......@@ -1129,7 +1130,7 @@ static void fsi_dma_do_tasklet(unsigned long data)
* FIXME
*
* In DMAEngine case, codec and FSI cannot be started simultaneously
* since FSI is using tasklet.
* since FSI is using the scheduler work queue.
* Therefore, in capture case, probably FSI FIFO will have got
* overflow error in this point.
* in that case, DMA cannot start transfer until error was cleared.
......@@ -1153,7 +1154,7 @@ static bool fsi_dma_filter(struct dma_chan *chan, void *param)
static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io)
{
tasklet_schedule(&io->tasklet);
schedule_work(&io->work);
return 0;
}
......@@ -1195,14 +1196,14 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct dev
return fsi_stream_probe(fsi, dev);
}
tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io);
INIT_WORK(&io->work, fsi_dma_do_work);
return 0;
}
static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io)
{
tasklet_kill(&io->tasklet);
cancel_work_sync(&io->work);
fsi_stream_stop(fsi, io);
......
......@@ -22,7 +22,7 @@
/**
* snd_soc_jack_new - Create a new jack
* @card: ASoC card
* @codec: ASoC codec
* @id: an identifying string for this jack
* @type: a bitmask of enum snd_jack_type values that can be detected by
* this jack
......@@ -133,12 +133,13 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_add_zones);
/**
* snd_soc_jack_get_type - Based on the mic bias value, this function returns
* the type of jack from the zones delcared in the jack type
* the type of jack from the zones declared in the jack type
*
* @jack: ASoC jack
* @micbias_voltage: mic bias voltage at adc channel when jack is plugged in
*
* Based on the mic bias value passed, this function helps identify
* the type of jack from the already delcared jack zones
* the type of jack from the already declared jack zones
*/
int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage)
{
......
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