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-rw-r--r--sound/soc/codecs/ak4642.c2
-rw-r--r--sound/soc/codecs/mxc_hdmi.c22
-rw-r--r--sound/soc/codecs/mxc_spdif.c6
-rw-r--r--sound/soc/codecs/wm8962.c6
-rw-r--r--sound/soc/codecs/wm_hubs.c18
-rw-r--r--sound/soc/imx/Makefile1
-rw-r--r--sound/soc/imx/imx-cs42888.c2
-rw-r--r--sound/soc/imx/imx-esai.c4
-rw-r--r--sound/soc/imx/imx-ssi.c45
-rw-r--r--sound/soc/imx/imx-ssi.h5
-rwxr-xr-xsound/soc/imx/imx-wm8958.c89
-rw-r--r--sound/soc/imx/imx-wm8962.c7
-rw-r--r--sound/soc/pxa/pxa-ssp.c61
-rw-r--r--sound/soc/samsung/neo1973_wm8753.c4
-rw-r--r--sound/soc/soc-core.c18
-rw-r--r--sound/soc/soc-dapm.c14
16 files changed, 125 insertions, 179 deletions
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 79c1b3d79f1e..7d45197f5cfd 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -143,7 +143,7 @@
* min : 0xFE : -115.0 dB
* mute: 0xFF
*/
-static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1);
static const struct snd_kcontrol_new ak4642_snd_controls[] = {
diff --git a/sound/soc/codecs/mxc_hdmi.c b/sound/soc/codecs/mxc_hdmi.c
index 834d785b2c27..9590181d4647 100644
--- a/sound/soc/codecs/mxc_hdmi.c
+++ b/sound/soc/codecs/mxc_hdmi.c
@@ -153,30 +153,26 @@ static struct cea_channel_speaker_allocation channel_allocations[] = {
{ .ca_index = 0x01, .speakers = { 0, 0, 0, 0, 0, LFE, FR, FL } },
/* Dolby Surround */
{ .ca_index = 0x02, .speakers = { 0, 0, 0, 0, FC, 0, FR, FL } },
- /* surround40 */
-{ .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } },
- /* surround41 */
-{ .ca_index = 0x09, .speakers = { 0, 0, RR, RL, 0, LFE, FR, FL } },
- /* surround50 */
-{ .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } },
- /* surround51 */
-{ .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } },
- /* 6.1 */
-{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } },
- /* surround71 */
-{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } },
-
{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } },
{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } },
{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } },
{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } },
{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } },
+{ .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } },
+{ .ca_index = 0x09, .speakers = { 0, 0, RR, RL, 0, LFE, FR, FL } },
+{ .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } },
+ /* surround51 */
+{ .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } },
{ .ca_index = 0x0c, .speakers = { 0, RC, RR, RL, 0, 0, FR, FL } },
{ .ca_index = 0x0d, .speakers = { 0, RC, RR, RL, 0, LFE, FR, FL } },
{ .ca_index = 0x0e, .speakers = { 0, RC, RR, RL, FC, 0, FR, FL } },
+ /* 6.1 */
+{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } },
{ .ca_index = 0x10, .speakers = { RRC, RLC, RR, RL, 0, 0, FR, FL } },
{ .ca_index = 0x11, .speakers = { RRC, RLC, RR, RL, 0, LFE, FR, FL } },
{ .ca_index = 0x12, .speakers = { RRC, RLC, RR, RL, FC, 0, FR, FL } },
+ /* surround71 */
+{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } },
{ .ca_index = 0x14, .speakers = { FRC, FLC, 0, 0, 0, 0, FR, FL } },
{ .ca_index = 0x15, .speakers = { FRC, FLC, 0, 0, 0, LFE, FR, FL } },
{ .ca_index = 0x16, .speakers = { FRC, FLC, 0, 0, FC, 0, FR, FL } },
diff --git a/sound/soc/codecs/mxc_spdif.c b/sound/soc/codecs/mxc_spdif.c
index d22f84fa214b..883aa4d26687 100644
--- a/sound/soc/codecs/mxc_spdif.c
+++ b/sound/soc/codecs/mxc_spdif.c
@@ -1138,12 +1138,6 @@ static int mxc_spdif_startup(struct snd_pcm_substream *substream,
/* disable all the interrupts */
spdif_intr_enable(0xffffff, 0);
- /* enable spdif_xtal_clk */
- clk_enable(plat_data->spdif_core_clk);
- spdif_softreset();
- /* disable all the interrupts */
- spdif_intr_enable(0xffffff, 0);
-
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
ret = mxc_spdif_playback_startup(substream, dai);
else
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 2f5ce28d70a0..190720ec029d 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3159,9 +3159,6 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream,
int aif0 = 0;
int adctl3 = 0;
- if (codec->dapm.bias_level != SND_SOC_BIAS_OFF)
- return 0;
-
wm8962->bclk = snd_soc_params_to_bclk(params);
if (params_channels(params) == 1)
wm8962->bclk *= 2;
@@ -3463,9 +3460,6 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
if (fll_div.theta || fll_div.lambda)
fll1 |= WM8962_FLL_FRAC;
- /* Stop the FLL while we reconfigure */
- snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_ENA, 0);
-
snd_soc_update_bits(codec, WM8962_FLL_CONTROL_2,
WM8962_FLL_OUTDIV_MASK |
WM8962_FLL_REFCLK_DIV_MASK,
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 9e370d14ad88..8712a9f6d6ec 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -562,14 +562,14 @@ SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0),
};
static const struct snd_kcontrol_new line2_mix[] = {
-SOC_DAPM_SINGLE("IN2R Switch", WM8993_LINE_MIXER2, 2, 1, 0),
-SOC_DAPM_SINGLE("IN2L Switch", WM8993_LINE_MIXER2, 1, 1, 0),
+SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 2, 1, 0),
+SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 1, 1, 0),
SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0),
};
static const struct snd_kcontrol_new line2n_mix[] = {
-SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER2, 6, 1, 0),
-SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 5, 1, 0),
+SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER2, 5, 1, 0),
+SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 6, 1, 0),
};
static const struct snd_kcontrol_new line2p_mix[] = {
@@ -589,6 +589,8 @@ SND_SOC_DAPM_INPUT("IN2RP:VXRP"),
SND_SOC_DAPM_MICBIAS("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0),
SND_SOC_DAPM_MICBIAS("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0),
+SND_SOC_DAPM_SUPPLY("LINEOUT_VMID_BUF", WM8993_ANTIPOP1, 7, 0, NULL, 0),
+
SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0,
in1l_pga, ARRAY_SIZE(in1l_pga)),
SND_SOC_DAPM_MIXER("IN1R PGA", WM8993_POWER_MANAGEMENT_2, 4, 0,
@@ -794,9 +796,11 @@ static const struct snd_soc_dapm_route lineout1_diff_routes[] = {
};
static const struct snd_soc_dapm_route lineout1_se_routes[] = {
+ { "LINEOUT1N Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" },
+ { "LINEOUT1P Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" },
@@ -804,8 +808,8 @@ static const struct snd_soc_dapm_route lineout1_se_routes[] = {
};
static const struct snd_soc_dapm_route lineout2_diff_routes[] = {
- { "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" },
- { "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" },
+ { "LINEOUT2 Mixer", "IN1L Switch", "IN1L PGA" },
+ { "LINEOUT2 Mixer", "IN1R Switch", "IN1R PGA" },
{ "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" },
{ "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" },
@@ -813,9 +817,11 @@ static const struct snd_soc_dapm_route lineout2_diff_routes[] = {
};
static const struct snd_soc_dapm_route lineout2_se_routes[] = {
+ { "LINEOUT2N Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" },
+ { "LINEOUT2P Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" },
{ "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" },
diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile
index 3c851809e26d..adeef06f3c7f 100644
--- a/sound/soc/imx/Makefile
+++ b/sound/soc/imx/Makefile
@@ -19,6 +19,7 @@ snd-soc-imx-wm8962-objs := imx-wm8962.o
snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
snd-soc-imx-cs42888-objs := imx-cs42888.o
snd-soc-imx-spdif-objs := imx-spdif.o
+snd-soc-imx-si4763-objs := imx-si4763.o
snd-soc-imx-hdmi-objs := imx-hdmi.o imx-hdmi-dai.o imx-hdmi-dma.o hdmi_pcm.o
obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
diff --git a/sound/soc/imx/imx-cs42888.c b/sound/soc/imx/imx-cs42888.c
index 5a22939b8cc0..128c9959a25f 100644
--- a/sound/soc/imx/imx-cs42888.c
+++ b/sound/soc/imx/imx-cs42888.c
@@ -138,7 +138,7 @@ static int config_asrc(struct snd_pcm_substream *substream,
|| !asrc_func)
return -EINVAL;
- if (channel != 2)
+ if ((channel != 2) && (channel != 4) && (channel != 6))
return -EINVAL;
ret = asrc_req_pair(channel, &asrc_esai_data.asrc_index);
diff --git a/sound/soc/imx/imx-esai.c b/sound/soc/imx/imx-esai.c
index daa3a8ae7a11..f11757b0e7cc 100644
--- a/sound/soc/imx/imx-esai.c
+++ b/sound/soc/imx/imx-esai.c
@@ -320,11 +320,11 @@ static int imx_esai_hw_tx_params(struct snd_pcm_substream *substream,
if (iprtd->asrc_enable) {
switch (iprtd->output_bit) {
- case 16:
+ case ASRC_WIDTH_16_BIT:
tfcr |= ESAI_WORD_LEN_16;
tcr |= ESAI_TCR_TSHFD_MSB | ESAI_TCR_TSWS_STL32_WDL16;
break;
- case 24:
+ case ASRC_WIDTH_24_BIT:
tfcr |= ESAI_WORD_LEN_24;
tcr |= ESAI_TCR_TSHFD_MSB | ESAI_TCR_TSWS_STL32_WDL24;
break;
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index a37c98b925ac..e9b102db7f01 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -1,8 +1,13 @@
/*
* imx-ssi.c -- ALSA Soc Audio Layer
*
+ * Copyright 2010-2012 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
* Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de>
*
+ * 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.
* This code is based on code copyrighted by Freescale,
* Liam Girdwood, Javier Martin and probably others.
*
@@ -110,7 +115,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
break;
case SND_SOC_DAIFMT_DSP_A:
/* data on rising edge of bclk, frame high 1clk before data */
- strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS;
+ strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS;
break;
}
@@ -213,28 +218,40 @@ static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
switch (div_id) {
case IMX_SSI_TX_DIV_2:
+ if (div & (~SSI_STCCR_DIV2))
+ return EINVAL;
stccr &= ~SSI_STCCR_DIV2;
stccr |= div;
break;
case IMX_SSI_TX_DIV_PSR:
+ if (div & (~SSI_STCCR_PSR))
+ return EINVAL;
stccr &= ~SSI_STCCR_PSR;
stccr |= div;
break;
case IMX_SSI_TX_DIV_PM:
- stccr &= ~0xff;
+ if (div & (~SSI_STCCR_PM_MASK))
+ return EINVAL;
+ stccr &= ~SSI_STCCR_PM_MASK;
stccr |= SSI_STCCR_PM(div);
break;
case IMX_SSI_RX_DIV_2:
- stccr &= ~SSI_STCCR_DIV2;
- stccr |= div;
+ if (div & (~SSI_SRCCR_DIV2))
+ return EINVAL;
+ srccr &= ~SSI_SRCCR_DIV2;
+ srccr |= div;
break;
case IMX_SSI_RX_DIV_PSR:
- stccr &= ~SSI_STCCR_PSR;
- stccr |= div;
+ if (div & (~SSI_SRCCR_PSR))
+ return EINVAL;
+ srccr &= ~SSI_SRCCR_PSR;
+ srccr |= div;
break;
case IMX_SSI_RX_DIV_PM:
- stccr &= ~0xff;
- stccr |= SSI_STCCR_PM(div);
+ if (div & (~SSI_SRCCR_PM_MASK))
+ return EINVAL;
+ srccr &= ~SSI_SRCCR_PM_MASK;
+ srccr |= SSI_STCCR_PM(div);
break;
default:
return -EINVAL;
@@ -497,7 +514,9 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai)
snd_soc_dai_set_drvdata(dai, ssi);
val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) |
- SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize);
+ SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize) |
+ SSI_SFCSR_TFWM1(ssi->dma_params_tx.burstsize) |
+ SSI_SFCSR_RFWM1(ssi->dma_params_rx.burstsize) ;
writel(val, ssi->base + SSI_SFCSR);
return 0;
@@ -567,10 +586,10 @@ static void setup_channel_to_ac97(struct imx_ssi *imx_ssi)
writel(SSI_SCR_SYN | SSI_SCR_NET, base + SSI_SCR);
- writel(SSI_SFCSR_RFWM0(8) |
- SSI_SFCSR_TFWM0(8) |
- SSI_SFCSR_RFWM1(8) |
- SSI_SFCSR_TFWM1(8), base + SSI_SFCSR);
+ writel(SSI_SFCSR_RFWM0(imx_ssi->dma_params_rx.burstsize) |
+ SSI_SFCSR_TFWM0(imx_ssi->dma_params_tx.burstsize) |
+ SSI_SFCSR_RFWM1(imx_ssi->dma_params_rx.burstsize) |
+ SSI_SFCSR_TFWM1(imx_ssi->dma_params_tx.burstsize), base + SSI_SFCSR);
writel(SSI_STCCR_WL(16) | SSI_STCCR_DC(12), base + SSI_STCCR);
writel(SSI_STCCR_WL(16) | SSI_STCCR_DC(12), base + SSI_SRCCR);
diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h
index 1b43c620c735..c57f9fe4edaa 100644
--- a/sound/soc/imx/imx-ssi.h
+++ b/sound/soc/imx/imx-ssi.h
@@ -223,9 +223,6 @@ int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
struct snd_pcm *pcm);
void imx_pcm_free(struct snd_pcm *pcm);
-/*
- * Do not change this as the FIQ handler depends on this size
- */
-#define IMX_SSI_DMABUF_SIZE (64 * 1024)
+#define IMX_SSI_DMABUF_SIZE (256 * 1024)
#endif /* _IMX_SSI_H */
diff --git a/sound/soc/imx/imx-wm8958.c b/sound/soc/imx/imx-wm8958.c
index 328bb0b6ca10..dc7a1f59b37c 100755
--- a/sound/soc/imx/imx-wm8958.c
+++ b/sound/soc/imx/imx-wm8958.c
@@ -26,7 +26,6 @@
#include <linux/io.h>
#include <linux/fsl_devices.h>
#include <linux/slab.h>
-#include <linux/gpio.h>
#include <linux/clk.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -47,14 +46,11 @@ struct imx_priv {
int sysclk; /*mclk from the outside*/
int codec_sysclk;
int dai_hifi;
- int hp_status;
- int hp_irq;
struct platform_device *pdev;
struct wm8994 *wm8958;
};
static struct imx_priv card_priv;
static struct snd_soc_card snd_soc_card_imx;
-
struct clk *codec_mclk;
static struct snd_soc_jack hs_jack;
@@ -109,11 +105,6 @@ static int imx_hifi_hw_params(struct snd_pcm_substream *substream,
int ret = 0;
unsigned int pll_out;
u32 dai_format;
- /* only need to do this once as capture and playback are sync */
-
- if (priv->dai_hifi)
- return 0;
- priv->dai_hifi = 1;
dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM;
@@ -187,71 +178,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
};
-static void headphone_detect_handler(struct work_struct *wor)
-{
-
- struct imx_priv *priv = &card_priv;
- struct platform_device *pdev = priv->pdev;
- struct mxc_audio_platform_data *plat = pdev->dev.platform_data;
- char *envp[3];
- char *buf;
-
- /*sysfs_notify(&pdev->dev.kobj, NULL, "headphone");*/
- priv->hp_status = gpio_get_value(plat->hp_gpio);
-
- /* setup a message for userspace headphone in */
- buf = kmalloc(32, GFP_ATOMIC);
- if (!buf) {
- pr_err("%s kmalloc failed\n", __func__);
- return;
- }
-
- if (priv->hp_status == 0)
- snprintf(buf, 32, "STATE=%d", 2);
- else
- snprintf(buf, 32, "STATE=%d", 0);
-
- envp[0] = "NAME=headphone";
- envp[1] = buf;
- envp[2] = NULL;
- kobject_uevent_env(&pdev->dev.kobj, KOBJ_CHANGE, envp);
- kfree(buf);
-
- enable_irq(priv->hp_irq);
-
- return;
-}
-
-static DECLARE_DELAYED_WORK(hp_event, headphone_detect_handler);
-
-static irqreturn_t imx_headphone_detect_handler(int irq, void *data)
-{
- disable_irq_nosync(irq);
- schedule_delayed_work(&hp_event, msecs_to_jiffies(200));
- return IRQ_HANDLED;
-}
-
-static ssize_t show_headphone(struct device_driver *dev, char *buf)
-{
- struct imx_priv *priv = &card_priv;
- struct platform_device *pdev = priv->pdev;
- struct mxc_audio_platform_data *plat = pdev->dev.platform_data;
-
- /* determine whether hp is plugged in */
- priv->hp_status = gpio_get_value(plat->hp_gpio);
-
- if (priv->hp_status)
- strcpy(buf, "speaker\n");
- else
- strcpy(buf, "headphone\n");
-
- /*pr_info("show_headphone hp_status = %d \r",priv->hp_status);*/
-
- return strlen(buf);
-}
-
-static DRIVER_ATTR(headphone, S_IRUGO | S_IWUSR, show_headphone, NULL);
-
static int imx_wm8958_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
@@ -356,7 +282,6 @@ static int __devinit imx_wm8958_probe(struct platform_device *pdev)
priv->pdev = pdev;
priv->wm8958 = wm8958;
- priv->hp_irq = gpio_to_irq(plat->hp_gpio);
imx_audmux_config(plat->src_port, plat->ext_port);
@@ -366,23 +291,9 @@ static int __devinit imx_wm8958_probe(struct platform_device *pdev)
}
priv->sysclk = plat->sysclk;
-
- ret = driver_create_file(pdev->dev.driver, &driver_attr_headphone);
- if (ret < 0) {
- ret = -EINVAL;
- return ret;
- }
-
hs_jack_gpios[0].gpio = plat->hp_gpio;
hs_jack_gpios[0].invert = plat->hp_active_low;
- if (plat->hp_gpio != -1) {
- ret = request_irq(priv->hp_irq,
- imx_headphone_detect_handler,
- IRQ_TYPE_EDGE_BOTH, pdev->name, priv);
- }
-
-
return ret;
}
diff --git a/sound/soc/imx/imx-wm8962.c b/sound/soc/imx/imx-wm8962.c
index aa3122f6d765..76299a659b41 100644
--- a/sound/soc/imx/imx-wm8962.c
+++ b/sound/soc/imx/imx-wm8962.c
@@ -98,9 +98,6 @@ static int imx_hifi_hw_params(struct snd_pcm_substream *substream,
u32 dai_format;
unsigned int pll_out;
- if (codec->dapm.bias_level != SND_SOC_BIAS_OFF)
- return 0;
-
dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM;
@@ -128,8 +125,8 @@ static int imx_hifi_hw_params(struct snd_pcm_substream *substream,
else
pll_out = sample_rate * 256;
- ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL_MCLK,
- WM8962_FLL_MCLK, priv->sysclk,
+ ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL_OSC,
+ WM8962_FLL_OSC, priv->sysclk,
pll_out);
if (ret < 0)
pr_err("Failed to start FLL: %d\n", ret);
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 8ad93ee2e92b..b583e604a1ea 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -668,6 +668,38 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
return 0;
}
+static void pxa_ssp_set_running_bit(struct snd_pcm_substream *substream,
+ struct ssp_device *ssp, int value)
+{
+ uint32_t sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
+ uint32_t sscr1 = pxa_ssp_read_reg(ssp, SSCR1);
+ uint32_t sspsp = pxa_ssp_read_reg(ssp, SSPSP);
+ uint32_t sssr = pxa_ssp_read_reg(ssp, SSSR);
+
+ if (value && (sscr0 & SSCR0_SSE))
+ pxa_ssp_write_reg(ssp, SSCR0, sscr0 & ~SSCR0_SSE);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (value)
+ sscr1 |= SSCR1_TSRE;
+ else
+ sscr1 &= ~SSCR1_TSRE;
+ } else {
+ if (value)
+ sscr1 |= SSCR1_RSRE;
+ else
+ sscr1 &= ~SSCR1_RSRE;
+ }
+
+ pxa_ssp_write_reg(ssp, SSCR1, sscr1);
+
+ if (value) {
+ pxa_ssp_write_reg(ssp, SSSR, sssr);
+ pxa_ssp_write_reg(ssp, SSPSP, sspsp);
+ pxa_ssp_write_reg(ssp, SSCR0, sscr0 | SSCR0_SSE);
+ }
+}
+
static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *cpu_dai)
{
@@ -681,42 +713,21 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
pxa_ssp_enable(ssp);
break;
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- val = pxa_ssp_read_reg(ssp, SSCR1);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- val |= SSCR1_TSRE;
- else
- val |= SSCR1_RSRE;
- pxa_ssp_write_reg(ssp, SSCR1, val);
+ pxa_ssp_set_running_bit(substream, ssp, 1);
val = pxa_ssp_read_reg(ssp, SSSR);
pxa_ssp_write_reg(ssp, SSSR, val);
break;
case SNDRV_PCM_TRIGGER_START:
- val = pxa_ssp_read_reg(ssp, SSCR1);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- val |= SSCR1_TSRE;
- else
- val |= SSCR1_RSRE;
- pxa_ssp_write_reg(ssp, SSCR1, val);
- pxa_ssp_enable(ssp);
+ pxa_ssp_set_running_bit(substream, ssp, 1);
break;
case SNDRV_PCM_TRIGGER_STOP:
- val = pxa_ssp_read_reg(ssp, SSCR1);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- val &= ~SSCR1_TSRE;
- else
- val &= ~SSCR1_RSRE;
- pxa_ssp_write_reg(ssp, SSCR1, val);
+ pxa_ssp_set_running_bit(substream, ssp, 0);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
pxa_ssp_disable(ssp);
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- val = pxa_ssp_read_reg(ssp, SSCR1);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- val &= ~SSCR1_TSRE;
- else
- val &= ~SSCR1_RSRE;
- pxa_ssp_write_reg(ssp, SSCR1, val);
+ pxa_ssp_set_running_bit(substream, ssp, 0);
break;
default:
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 16152ed08648..c1290da71c62 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -425,7 +425,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.platform_name = "samsung-audio",
.cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "wm8753-hifi",
- .codec_name = "wm8753-codec.0-001a",
+ .codec_name = "wm8753.0-001a",
.init = neo1973_wm8753_init,
.ops = &neo1973_hifi_ops,
},
@@ -434,7 +434,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.stream_name = "Voice",
.cpu_dai_name = "dfbmcs320-pcm",
.codec_dai_name = "wm8753-voice",
- .codec_name = "wm8753-codec.0-001a",
+ .codec_name = "wm8753.0-001a",
.ops = &neo1973_voice_ops,
},
};
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 5f36274336ae..ad12a4c17b72 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -30,6 +30,7 @@
#include <linux/bitops.h>
#include <linux/debugfs.h>
#include <linux/platform_device.h>
+#include <linux/ctype.h>
#include <linux/slab.h>
#include <sound/ac97_codec.h>
#include <sound/core.h>
@@ -1956,9 +1957,20 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
"%s", card->name);
snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
"%s", card->long_name ? card->long_name : card->name);
- if (card->driver_name)
- strlcpy(card->snd_card->driver, card->driver_name,
- sizeof(card->snd_card->driver));
+ snprintf(card->snd_card->driver, sizeof(card->snd_card->driver),
+ "%s", card->driver_name ? card->driver_name : card->name);
+ for (i = 0; i < ARRAY_SIZE(card->snd_card->driver); i++) {
+ switch (card->snd_card->driver[i]) {
+ case '_':
+ case '-':
+ case '\0':
+ break;
+ default:
+ if (!isalnum(card->snd_card->driver[i]))
+ card->snd_card->driver[i] = '_';
+ break;
+ }
+ }
if (card->late_probe) {
ret = card->late_probe(card);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 32ab7fc4579a..d5ec2060d121 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -67,6 +67,7 @@ static int dapm_up_seq[] = {
[snd_soc_dapm_out_drv] = 10,
[snd_soc_dapm_hp] = 10,
[snd_soc_dapm_spk] = 10,
+ [snd_soc_dapm_line] = 10,
[snd_soc_dapm_post] = 11,
};
@@ -75,6 +76,7 @@ static int dapm_down_seq[] = {
[snd_soc_dapm_adc] = 1,
[snd_soc_dapm_hp] = 2,
[snd_soc_dapm_spk] = 2,
+ [snd_soc_dapm_line] = 2,
[snd_soc_dapm_out_drv] = 2,
[snd_soc_dapm_pga] = 4,
[snd_soc_dapm_mixer_named_ctl] = 5,
@@ -2615,9 +2617,13 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
* standby.
*/
if (powerdown) {
- snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE);
+ if (dapm->bias_level == SND_SOC_BIAS_ON)
+ snd_soc_dapm_set_bias_level(dapm,
+ SND_SOC_BIAS_PREPARE);
dapm_seq_run(dapm, &down_list, 0, false);
- snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY);
+ if (dapm->bias_level == SND_SOC_BIAS_PREPARE)
+ snd_soc_dapm_set_bias_level(dapm,
+ SND_SOC_BIAS_STANDBY);
}
}
@@ -2630,7 +2636,9 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card)
list_for_each_entry(codec, &card->codec_dev_list, list) {
soc_dapm_shutdown_codec(&codec->dapm);
- snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF);
+ if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+ snd_soc_dapm_set_bias_level(&codec->dapm,
+ SND_SOC_BIAS_OFF);
}
}
generated by cgit v1.2.3 (git 2.0.4) at 2024-07-04 07:27:38 +0000