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authorLinus Torvalds <torvalds@linux-foundation.org>2013-07-03 19:52:22 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2013-07-03 19:52:22 -0700
commit1286da8bc009cb2aee7f285e94623fc974c0c983 (patch)
tree51ec0a79c3de63fa809b831ae0cbb5b85e44482f /sound/soc/fsl
parent9e220385c4eb8b7e66174a60ea0e15b6b296f228 (diff)
parent1ba65ae4bdbd43265c51ee4c30ff21a48124b6d8 (diff)
Merge tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "A relative calm release at this time with a flat diffstat. The only significant change in the ALSA core side is the support for more than 32 card instances, configurable via kconfig. Other than that, in both ASoC and other parts, mostly some improvements and fixes on the driver side. - hda: More quirks for ALC269-variants on Dell & co, VIA codec fixes - hda: Haswell HDMI audio fixes, runtime PM improvements - hda: Intel BayTrail support, ALC5505 DSP support - es1968: MediaForte M56VAP support - usb-audio: Improved support for Yamaha/Roland devices - usb-audio: M2Tech hiFace, Audio Advantage Micro II support - hdspm: wordclock fixes - ASoC: Pending fixes for WM8962 - ASoC: Cleanups and fixes for Blackfin, SGTL5000 and UX500 - ASoC: Generalisation of the Bluetooth and HDMI stub drivers - ASoC: SSM2518 and RT5640 codec drivers. - ASoC: Tegra CPUs with RT5640 machine driver - ASoC: AC'97 refactoring bug fixes - ASoC: ADAU1701 driver fixes - Clean up of *_set_drvdata() in a wide range of drivers" * tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (284 commits) ALSA: vmaster: Fix the regression of missing vmaster hook call ALSA: hda - Add Dell SSID to support Headset Mic recording ASoC: adau1701: remove control_data assignment ASoC: adau1701: more direct regmap usage ASoC: ac97: fixup multi-platform AC'97 module build failure ASoC: pxa2xx: fixup multi-platform AC'97 build failures ASoC: tegra20-ac97: Remove unused variable ASoC: tegra20-ac97: Remove duplicate error message ALSA: usb-audio: Add Audio Advantage Micro II ASoC: tas5086: fix Mid-Z implementation ASoC: tas5086: fix TAS5086_CLOCK_CONTROL register size ALSA: Replace the magic number 44 with const ALSA: hda - Fix the max length of control name in generic parser ALSA: hda - Guess what, it's two more Dell headset mic quirks ALSA: hda - Yet another Dell headset mic quirk ALSA: hda - Add support for ALC5505 DSP power-save mode ASoC: mfld: Remove unused variable ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE ALSA: usb-audio: claim autodetected PCM interfaces all at once ALSA: usb-audio: remove superfluous Roland quirks ...
Diffstat (limited to 'sound/soc/fsl')
-rw-r--r--sound/soc/fsl/Kconfig17
-rw-r--r--sound/soc/fsl/Makefile13
-rw-r--r--sound/soc/fsl/eukrea-tlv320.c2
-rw-r--r--sound/soc/fsl/fsl_ssi.c13
-rw-r--r--sound/soc/fsl/imx-audmux.c8
-rw-r--r--sound/soc/fsl/imx-mc13783.c2
-rw-r--r--sound/soc/fsl/imx-pcm-dma.c2
-rw-r--r--sound/soc/fsl/imx-pcm-fiq.c92
-rw-r--r--sound/soc/fsl/imx-pcm.c145
-rw-r--r--sound/soc/fsl/imx-pcm.h10
-rw-r--r--sound/soc/fsl/imx-sgtl5000.c39
-rw-r--r--sound/soc/fsl/imx-ssi.c55
-rw-r--r--sound/soc/fsl/imx-ssi.h3
-rw-r--r--sound/soc/fsl/imx-wm8962.c323
-rw-r--r--sound/soc/fsl/mpc5200_psc_ac97.c10
-rw-r--r--sound/soc/fsl/mx27vis-aic32x4.c2
-rw-r--r--sound/soc/fsl/phycore-ac97.c2
-rw-r--r--sound/soc/fsl/wm1133-ev1.c2
18 files changed, 489 insertions, 251 deletions
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 3843a18d4e56..aa438546c912 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -108,18 +108,13 @@ if SND_IMX_SOC
config SND_SOC_IMX_SSI
tristate
-config SND_SOC_IMX_PCM
- tristate
-
config SND_SOC_IMX_PCM_FIQ
bool
select FIQ
- select SND_SOC_IMX_PCM
config SND_SOC_IMX_PCM_DMA
bool
select SND_SOC_GENERIC_DMAENGINE_PCM
- select SND_SOC_IMX_PCM
config SND_SOC_IMX_AUDMUX
tristate
@@ -173,6 +168,18 @@ config SND_SOC_EUKREA_TLV320
Enable I2S based access to the TLV320AIC23B codec attached
to the SSI interface
+config SND_SOC_IMX_WM8962
+ tristate "SoC Audio support for i.MX boards with wm8962"
+ depends on OF && I2C
+ select SND_SOC_WM8962
+ select SND_SOC_IMX_PCM_DMA
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_FSL_SSI
+ select SND_SOC_FSL_UTILS
+ help
+ Say Y if you want to add support for SoC audio on an i.MX board with
+ a wm8962 codec.
+
config SND_SOC_IMX_SGTL5000
tristate "SoC Audio support for i.MX boards with sgtl5000"
depends on OF && I2C
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index afd34794db53..d4b4aa8b5649 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -30,18 +30,11 @@ obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o
# i.MX Platform Support
snd-soc-imx-ssi-objs := imx-ssi.o
snd-soc-imx-audmux-objs := imx-audmux.o
-snd-soc-imx-pcm-objs := imx-pcm.o
-ifneq ($(CONFIG_SND_SOC_IMX_PCM_FIQ),)
- snd-soc-imx-pcm-objs += imx-pcm-fiq.o
-endif
-ifneq ($(CONFIG_SND_SOC_IMX_PCM_DMA),)
- snd-soc-imx-pcm-objs += imx-pcm-dma.o
-endif
-
obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o
obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o
-obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o
+obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o
+obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm-dma.o
# i.MX Machine Support
snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
@@ -49,6 +42,7 @@ snd-soc-phycore-ac97-objs := phycore-ac97.o
snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
snd-soc-wm1133-ev1-objs := wm1133-ev1.o
snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
+snd-soc-imx-wm8962-objs := imx-wm8962.o
snd-soc-imx-mc13783-objs := imx-mc13783.o
obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
@@ -56,4 +50,5 @@ obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
+obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o
obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 75ffdf0e2aad..9a4a0ca2c1de 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -80,7 +80,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = {
.name = "tlv320aic23",
.stream_name = "TLV320AIC23",
.codec_dai_name = "tlv320aic23-hifi",
- .platform_name = "imx-fiq-pcm-audio.0",
+ .platform_name = "imx-ssi.0",
.codec_name = "tlv320aic23-codec.0-001a",
.cpu_dai_name = "imx-ssi.0",
.ops = &eukrea_tlv320_snd_ops,
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 0f0bed6def9e..2f2d837df07f 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -122,7 +122,6 @@ struct fsl_ssi_private {
bool new_binding;
bool ssi_on_imx;
struct clk *clk;
- struct platform_device *imx_pcm_pdev;
struct snd_dmaengine_dai_dma_data dma_params_tx;
struct snd_dmaengine_dai_dma_data dma_params_rx;
struct imx_dma_data filter_data_tx;
@@ -809,13 +808,9 @@ static int fsl_ssi_probe(struct platform_device *pdev)
}
if (ssi_private->ssi_on_imx) {
- ssi_private->imx_pcm_pdev =
- platform_device_register_simple("imx-pcm-audio",
- -1, NULL, 0);
- if (IS_ERR(ssi_private->imx_pcm_pdev)) {
- ret = PTR_ERR(ssi_private->imx_pcm_pdev);
+ ret = imx_pcm_dma_init(pdev);
+ if (ret)
goto error_dev;
- }
}
/*
@@ -854,7 +849,7 @@ done:
error_dai:
if (ssi_private->ssi_on_imx)
- platform_device_unregister(ssi_private->imx_pcm_pdev);
+ imx_pcm_dma_exit(pdev);
snd_soc_unregister_component(&pdev->dev);
error_dev:
@@ -889,7 +884,7 @@ static int fsl_ssi_remove(struct platform_device *pdev)
if (!ssi_private->new_binding)
platform_device_unregister(ssi_private->pdev);
if (ssi_private->ssi_on_imx) {
- platform_device_unregister(ssi_private->imx_pcm_pdev);
+ imx_pcm_dma_exit(pdev);
clk_disable_unprepare(ssi_private->clk);
clk_put(ssi_private->clk);
}
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index 47f046a8fdab..e260f1f899db 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -26,7 +26,6 @@
#include <linux/of_device.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
-#include <linux/pinctrl/consumer.h>
#include "imx-audmux.h"
@@ -247,7 +246,6 @@ EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port);
static int imx_audmux_probe(struct platform_device *pdev)
{
struct resource *res;
- struct pinctrl *pinctrl;
const struct of_device_id *of_id =
of_match_device(imx_audmux_dt_ids, &pdev->dev);
@@ -256,12 +254,6 @@ static int imx_audmux_probe(struct platform_device *pdev)
if (IS_ERR(audmux_base))
return PTR_ERR(audmux_base);
- pinctrl = devm_pinctrl_get_select_default(&pdev->dev);
- if (IS_ERR(pinctrl)) {
- dev_err(&pdev->dev, "setup pinctrl failed!");
- return PTR_ERR(pinctrl);
- }
-
audmux_clk = devm_clk_get(&pdev->dev, "audmux");
if (IS_ERR(audmux_clk)) {
dev_dbg(&pdev->dev, "cannot get clock: %ld\n",
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index 4ae30f21fdb5..9df173c091a6 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -64,7 +64,7 @@ static struct snd_soc_dai_link imx_mc13783_dai_mc13783[] = {
.codec_dai_name = "mc13783-hifi",
.codec_name = "mc13783-codec",
.cpu_dai_name = "imx-ssi.0",
- .platform_name = "imx-pcm-audio.0",
+ .platform_name = "imx-ssi.0",
.ops = &imx_mc13783_hifi_ops,
.symmetric_rates = 1,
.dai_fmt = FMT_SSI,
diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c
index c246fb514930..fde4d2ea68c8 100644
--- a/sound/soc/fsl/imx-pcm-dma.c
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -67,8 +67,10 @@ int imx_pcm_dma_init(struct platform_device *pdev)
SND_DMAENGINE_PCM_FLAG_NO_DT |
SND_DMAENGINE_PCM_FLAG_COMPAT);
}
+EXPORT_SYMBOL_GPL(imx_pcm_dma_init);
void imx_pcm_dma_exit(struct platform_device *pdev)
{
snd_dmaengine_pcm_unregister(&pdev->dev);
}
+EXPORT_SYMBOL_GPL(imx_pcm_dma_exit);
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index 670b96b0ce2f..310d90290320 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -225,6 +225,22 @@ static int snd_imx_close(struct snd_pcm_substream *substream)
return 0;
}
+static int snd_imx_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int ret;
+
+ ret = dma_mmap_writecombine(substream->pcm->card->dev, vma,
+ runtime->dma_area, runtime->dma_addr, runtime->dma_bytes);
+
+ pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+ return ret;
+}
+
static struct snd_pcm_ops imx_pcm_ops = {
.open = snd_imx_open,
.close = snd_imx_close,
@@ -236,6 +252,54 @@ static struct snd_pcm_ops imx_pcm_ops = {
.mmap = snd_imx_pcm_mmap,
};
+static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = IMX_SSI_DMABUF_SIZE;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+ if (!buf->area)
+ return -ENOMEM;
+ buf->bytes = size;
+
+ return 0;
+}
+
+static u64 imx_pcm_dmamask = DMA_BIT_MASK(32);
+
+static int imx_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret = 0;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &imx_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+ ret = imx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+ }
+
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ ret = imx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ }
+
+out:
+ return ret;
+}
+
static int ssi_irq = 0;
static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd)
@@ -268,6 +332,27 @@ static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd)
return 0;
}
+static void imx_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+
+ dma_free_writecombine(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
static void imx_pcm_fiq_free(struct snd_pcm *pcm)
{
mxc_set_irq_fiq(ssi_irq, 0);
@@ -314,3 +399,10 @@ failed_register:
return ret;
}
+EXPORT_SYMBOL_GPL(imx_pcm_fiq_init);
+
+void imx_pcm_fiq_exit(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+}
+EXPORT_SYMBOL_GPL(imx_pcm_fiq_exit);
diff --git a/sound/soc/fsl/imx-pcm.c b/sound/soc/fsl/imx-pcm.c
deleted file mode 100644
index c49896442d8e..000000000000
--- a/sound/soc/fsl/imx-pcm.c
+++ /dev/null
@@ -1,145 +0,0 @@
-/*
- * Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de>
- *
- * This code is based on code copyrighted by Freescale,
- * Liam Girdwood, Javier Martin and probably others.
- *
- * 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/dma-mapping.h>
-#include <linux/module.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include "imx-pcm.h"
-
-int snd_imx_pcm_mmap(struct snd_pcm_substream *substream,
- struct vm_area_struct *vma)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- int ret;
-
- ret = dma_mmap_writecombine(substream->pcm->card->dev, vma,
- runtime->dma_area, runtime->dma_addr, runtime->dma_bytes);
-
- pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret,
- runtime->dma_area,
- runtime->dma_addr,
- runtime->dma_bytes);
- return ret;
-}
-EXPORT_SYMBOL_GPL(snd_imx_pcm_mmap);
-
-static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
-{
- struct snd_pcm_substream *substream = pcm->streams[stream].substream;
- struct snd_dma_buffer *buf = &substream->dma_buffer;
- size_t size = IMX_SSI_DMABUF_SIZE;
-
- buf->dev.type = SNDRV_DMA_TYPE_DEV;
- buf->dev.dev = pcm->card->dev;
- buf->private_data = NULL;
- buf->area = dma_alloc_writecombine(pcm->card->dev, size,
- &buf->addr, GFP_KERNEL);
- if (!buf->area)
- return -ENOMEM;
- buf->bytes = size;
-
- return 0;
-}
-
-static u64 imx_pcm_dmamask = DMA_BIT_MASK(32);
-
-int imx_pcm_new(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_card *card = rtd->card->snd_card;
- struct snd_pcm *pcm = rtd->pcm;
- int ret = 0;
-
- if (!card->dev->dma_mask)
- card->dev->dma_mask = &imx_pcm_dmamask;
- if (!card->dev->coherent_dma_mask)
- card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
- if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
- ret = imx_pcm_preallocate_dma_buffer(pcm,
- SNDRV_PCM_STREAM_PLAYBACK);
- if (ret)
- goto out;
- }
-
- if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
- ret = imx_pcm_preallocate_dma_buffer(pcm,
- SNDRV_PCM_STREAM_CAPTURE);
- if (ret)
- goto out;
- }
-
-out:
- return ret;
-}
-EXPORT_SYMBOL_GPL(imx_pcm_new);
-
-void imx_pcm_free(struct snd_pcm *pcm)
-{
- struct snd_pcm_substream *substream;
- struct snd_dma_buffer *buf;
- int stream;
-
- for (stream = 0; stream < 2; stream++) {
- substream = pcm->streams[stream].substream;
- if (!substream)
- continue;
-
- buf = &substream->dma_buffer;
- if (!buf->area)
- continue;
-
- dma_free_writecombine(pcm->card->dev, buf->bytes,
- buf->area, buf->addr);
- buf->area = NULL;
- }
-}
-EXPORT_SYMBOL_GPL(imx_pcm_free);
-
-static int imx_pcm_probe(struct platform_device *pdev)
-{
- if (strcmp(pdev->id_entry->name, "imx-fiq-pcm-audio") == 0)
- return imx_pcm_fiq_init(pdev);
-
- return imx_pcm_dma_init(pdev);
-}
-
-static int imx_pcm_remove(struct platform_device *pdev)
-{
- if (strcmp(pdev->id_entry->name, "imx-fiq-pcm-audio") == 0)
- snd_soc_unregister_platform(&pdev->dev);
- else
- imx_pcm_dma_exit(pdev);
-
- return 0;
-}
-
-static struct platform_device_id imx_pcm_devtype[] = {
- { .name = "imx-pcm-audio", },
- { .name = "imx-fiq-pcm-audio", },
- { /* sentinel */ }
-};
-MODULE_DEVICE_TABLE(platform, imx_pcm_devtype);
-
-static struct platform_driver imx_pcm_driver = {
- .driver = {
- .name = "imx-pcm",
- .owner = THIS_MODULE,
- },
- .id_table = imx_pcm_devtype,
- .probe = imx_pcm_probe,
- .remove = imx_pcm_remove,
-};
-module_platform_driver(imx_pcm_driver);
-
-MODULE_DESCRIPTION("Freescale i.MX PCM driver");
-MODULE_AUTHOR("Sascha Hauer <s.hauer@pengutronix.de>");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h
index b7fa0d75c687..67f656c7c320 100644
--- a/sound/soc/fsl/imx-pcm.h
+++ b/sound/soc/fsl/imx-pcm.h
@@ -32,11 +32,6 @@ imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data,
dma_data->peripheral_type = IMX_DMATYPE_SSI;
}
-int snd_imx_pcm_mmap(struct snd_pcm_substream *substream,
- struct vm_area_struct *vma);
-int imx_pcm_new(struct snd_soc_pcm_runtime *rtd);
-void imx_pcm_free(struct snd_pcm *pcm);
-
#ifdef CONFIG_SND_SOC_IMX_PCM_DMA
int imx_pcm_dma_init(struct platform_device *pdev);
void imx_pcm_dma_exit(struct platform_device *pdev);
@@ -53,11 +48,16 @@ static inline void imx_pcm_dma_exit(struct platform_device *pdev)
#ifdef CONFIG_SND_SOC_IMX_PCM_FIQ
int imx_pcm_fiq_init(struct platform_device *pdev);
+void imx_pcm_fiq_exit(struct platform_device *pdev);
#else
static inline int imx_pcm_fiq_init(struct platform_device *pdev)
{
return -ENODEV;
}
+
+static inline void imx_pcm_fiq_exit(struct platform_device *pdev)
+{
+}
#endif
#endif /* _IMX_PCM_H */
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index 9584e78858df..7a8bc1220b2e 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -128,28 +128,18 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
goto fail;
}
- data->codec_clk = clk_get(&codec_dev->dev, NULL);
- if (IS_ERR(data->codec_clk)) {
- /* assuming clock enabled by default */
- data->codec_clk = NULL;
- ret = of_property_read_u32(codec_np, "clock-frequency",
- &data->clk_frequency);
- if (ret) {
- dev_err(&codec_dev->dev,
- "clock-frequency missing or invalid\n");
- goto fail;
- }
- } else {
- data->clk_frequency = clk_get_rate(data->codec_clk);
- clk_prepare_enable(data->codec_clk);
- }
+ data->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
+ if (IS_ERR(data->codec_clk))
+ goto fail;
+
+ data->clk_frequency = clk_get_rate(data->codec_clk);
data->dai.name = "HiFi";
data->dai.stream_name = "HiFi";
data->dai.codec_dai_name = "sgtl5000";
data->dai.codec_of_node = codec_np;
data->dai.cpu_of_node = ssi_np;
- data->dai.platform_name = "imx-pcm-audio";
+ data->dai.platform_of_node = ssi_np;
data->dai.init = &imx_sgtl5000_dai_init;
data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM;
@@ -157,10 +147,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
data->card.dev = &pdev->dev;
ret = snd_soc_of_parse_card_name(&data->card, "model");
if (ret)
- goto clk_fail;
+ goto fail;
ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
if (ret)
- goto clk_fail;
+ goto fail;
data->card.num_links = 1;
data->card.owner = THIS_MODULE;
data->card.dai_link = &data->dai;
@@ -170,12 +160,15 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
ret = snd_soc_register_card(&data->card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
- goto clk_fail;
+ goto fail;
}
platform_set_drvdata(pdev, data);
-clk_fail:
- clk_put(data->codec_clk);
+ of_node_put(ssi_np);
+ of_node_put(codec_np);
+
+ return 0;
+
fail:
if (ssi_np)
of_node_put(ssi_np);
@@ -189,10 +182,6 @@ static int imx_sgtl5000_remove(struct platform_device *pdev)
{
struct imx_sgtl5000_data *data = platform_get_drvdata(pdev);
- if (data->codec_clk) {
- clk_disable_unprepare(data->codec_clk);
- clk_put(data->codec_clk);
- }
snd_soc_unregister_card(&data->card);
return 0;
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index c6fa03e2114a..51be3772cba9 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -501,13 +501,12 @@ static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97)
imx_ssi_ac97_read(ac97, 0);
}
-struct snd_ac97_bus_ops soc_ac97_ops = {
+static struct snd_ac97_bus_ops imx_ssi_ac97_ops = {
.read = imx_ssi_ac97_read,
.write = imx_ssi_ac97_write,
.reset = imx_ssi_ac97_reset,
.warm_reset = imx_ssi_ac97_warm_reset
};
-EXPORT_SYMBOL_GPL(soc_ac97_ops);
static int imx_ssi_probe(struct platform_device *pdev)
{
@@ -583,6 +582,12 @@ static int imx_ssi_probe(struct platform_device *pdev)
platform_set_drvdata(pdev, ssi);
+ ret = snd_soc_set_ac97_ops(&imx_ssi_ac97_ops);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret);
+ goto failed_register;
+ }
+
ret = snd_soc_register_component(&pdev->dev, &imx_component,
dai, 1);
if (ret) {
@@ -590,46 +595,25 @@ static int imx_ssi_probe(struct platform_device *pdev)
goto failed_register;
}
- ssi->soc_platform_pdev_fiq = platform_device_alloc("imx-fiq-pcm-audio", pdev->id);
- if (!ssi->soc_platform_pdev_fiq) {
- ret = -ENOMEM;
- goto failed_pdev_fiq_alloc;
- }
+ ret = imx_pcm_fiq_init(pdev);
+ if (ret)
+ goto failed_pcm_fiq;
- platform_set_drvdata(ssi->soc_platform_pdev_fiq, ssi);
- ret = platform_device_add(ssi->soc_platform_pdev_fiq);
- if (ret) {
- dev_err(&pdev->dev, "failed to add platform device\n");
- goto failed_pdev_fiq_add;
- }
-
- ssi->soc_platform_pdev = platform_device_alloc("imx-pcm-audio", pdev->id);
- if (!ssi->soc_platform_pdev) {
- ret = -ENOMEM;
- goto failed_pdev_alloc;
- }
-
- platform_set_drvdata(ssi->soc_platform_pdev, ssi);
- ret = platform_device_add(ssi->soc_platform_pdev);
- if (ret) {
- dev_err(&pdev->dev, "failed to add platform device\n");
- goto failed_pdev_add;
- }
+ ret = imx_pcm_dma_init(pdev);
+ if (ret)
+ goto failed_pcm_dma;
return 0;
-failed_pdev_add:
- platform_device_put(ssi->soc_platform_pdev);
-failed_pdev_alloc:
- platform_device_del(ssi->soc_platform_pdev_fiq);
-failed_pdev_fiq_add:
- platform_device_put(ssi->soc_platform_pdev_fiq);
-failed_pdev_fiq_alloc:
+failed_pcm_dma:
+ imx_pcm_fiq_exit(pdev);
+failed_pcm_fiq:
snd_soc_unregister_component(&pdev->dev);
failed_register:
release_mem_region(res->start, resource_size(res));
clk_disable_unprepare(ssi->clk);
failed_clk:
+ snd_soc_set_ac97_ops(NULL);
return ret;
}
@@ -639,8 +623,8 @@ static int imx_ssi_remove(struct platform_device *pdev)
struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
struct imx_ssi *ssi = platform_get_drvdata(pdev);
- platform_device_unregister(ssi->soc_platform_pdev);
- platform_device_unregister(ssi->soc_platform_pdev_fiq);
+ imx_pcm_dma_exit(pdev);
+ imx_pcm_fiq_exit(pdev);
snd_soc_unregister_component(&pdev->dev);
@@ -649,6 +633,7 @@ static int imx_ssi_remove(struct platform_device *pdev)
release_mem_region(res->start, resource_size(res));
clk_disable_unprepare(ssi->clk);
+ snd_soc_set_ac97_ops(NULL);
return 0;
}
diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h
index bb6b3dbb13fd..d5003cefca8d 100644
--- a/sound/soc/fsl/imx-ssi.h
+++ b/sound/soc/fsl/imx-ssi.h
@@ -211,9 +211,6 @@ struct imx_ssi {
struct imx_dma_data filter_data_rx;
int enabled;
-
- struct platform_device *soc_platform_pdev;
- struct platform_device *soc_platform_pdev_fiq;
};
#endif /* _IMX_SSI_H */
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
new file mode 100644
index 000000000000..52a36a90f4f4
--- /dev/null
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -0,0 +1,323 @@
+/*
+ * Copyright 2013 Freescale Semiconductor, Inc.
+ *
+ * Based on imx-sgtl5000.c
+ * Copyright 2012 Freescale Semiconductor, Inc.
+ * Copyright 2012 Linaro Ltd.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/of_i2c.h>
+#include <linux/slab.h>
+#include <linux/clk.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include <sound/soc-dapm.h>
+#include <linux/pinctrl/consumer.h>
+
+#include "../codecs/wm8962.h"
+#include "imx-audmux.h"
+
+#define DAI_NAME_SIZE 32
+
+struct imx_wm8962_data {
+ struct snd_soc_dai_link dai;
+ struct snd_soc_card card;
+ char codec_dai_name[DAI_NAME_SIZE];
+ char platform_name[DAI_NAME_SIZE];
+ struct clk *codec_clk;
+ unsigned int clk_frequency;
+};
+
+struct imx_priv {
+ struct platform_device *pdev;
+};
+static struct imx_priv card_priv;
+
+static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_MIC("AMIC", NULL),
+ SND_SOC_DAPM_MIC("DMIC", NULL),
+};
+
+static int sample_rate = 44100;
+static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE;
+
+static int imx_hifi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ sample_rate = params_rate(params);
+ sample_format = params_format(params);
+
+ return 0;
+}
+
+static struct snd_soc_ops imx_hifi_ops = {
+ .hw_params = imx_hifi_hw_params,
+};
+
+static int imx_wm8962_set_bias_level(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ struct imx_priv *priv = &card_priv;
+ struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev);
+ struct device *dev = &priv->pdev->dev;
+ unsigned int pll_out;
+ int ret;
+
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_PREPARE:
+ if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
+ if (sample_format == SNDRV_PCM_FORMAT_S24_LE)
+ pll_out = sample_rate * 384;
+ else
+ pll_out = sample_rate * 256;
+
+ ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+ WM8962_FLL_MCLK, data->clk_frequency,
+ pll_out);
+ if (ret < 0) {
+ dev_err(dev, "failed to start FLL: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(codec_dai,
+ WM8962_SYSCLK_FLL, pll_out,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(dev, "failed to set SYSCLK: %d\n", ret);
+ return ret;
+ }
+ }
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (dapm->bias_level == SND_SOC_BIAS_PREPARE) {
+ ret = snd_soc_dai_set_sysclk(codec_dai,
+ WM8962_SYSCLK_MCLK, data->clk_frequency,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(dev,
+ "failed to switch away from FLL: %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+ 0, 0, 0);
+ if (ret < 0) {
+ dev_err(dev, "failed to stop FLL: %d\n", ret);
+ return ret;
+ }
+ }
+ break;
+
+ default:
+ break;
+ }
+
+ dapm->bias_level = level;
+
+ return 0;
+}
+
+static int imx_wm8962_late_probe(struct snd_soc_card *card)
+{
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ struct imx_priv *priv = &card_priv;
+ struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev);
+ struct device *dev = &priv->pdev->dev;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
+ data->clk_frequency, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ dev_err(dev, "failed to set sysclk in %s\n", __func__);
+
+ return ret;
+}
+
+static int imx_wm8962_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct device_node *ssi_np, *codec_np;
+ struct platform_device *ssi_pdev;
+ struct imx_priv *priv = &card_priv;
+ struct i2c_client *codec_dev;
+ struct imx_wm8962_data *data;
+ int int_port, ext_port;
+ int ret;
+
+ priv->pdev = pdev;
+
+ ret = of_property_read_u32(np, "mux-int-port", &int_port);
+ if (ret) {
+ dev_err(&pdev->dev, "mux-int-port missing or invalid\n");
+ return ret;
+ }
+ ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+ if (ret) {
+ dev_err(&pdev->dev, "mux-ext-port missing or invalid\n");
+ return ret;
+ }
+
+ /*
+ * The port numbering in the hardware manual starts at 1, while
+ * the audmux API expects it starts at 0.
+ */
+ int_port--;
+ ext_port--;
+ ret = imx_audmux_v2_configure_port(int_port,
+ IMX_AUDMUX_V2_PTCR_SYN |
+ IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_TFSDIR |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+ if (ret) {
+ dev_err(&pdev->dev, "audmux internal port setup failed\n");
+ return ret;
+ }
+ imx_audmux_v2_configure_port(ext_port,
+ IMX_AUDMUX_V2_PTCR_SYN,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+ if (ret) {
+ dev_err(&pdev->dev, "audmux external port setup failed\n");
+ return ret;
+ }
+
+ ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0);
+ codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
+ if (!ssi_np || !codec_np) {
+ dev_err(&pdev->dev, "phandle missing or invalid\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ ssi_pdev = of_find_device_by_node(ssi_np);
+ if (!ssi_pdev) {
+ dev_err(&pdev->dev, "failed to find SSI platform device\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+ codec_dev = of_find_i2c_device_by_node(codec_np);
+ if (!codec_dev || !codec_dev->driver) {
+ dev_err(&pdev->dev, "failed to find codec platform device\n");
+ return -EINVAL;
+ }
+
+ data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+ if (!data) {
+ ret = -ENOMEM;
+ goto fail;
+ }
+
+ data->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
+ if (IS_ERR(data->codec_clk)) {
+ ret = PTR_ERR(data->codec_clk);
+ dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret);
+ goto fail;
+ }
+
+ data->clk_frequency = clk_get_rate(data->codec_clk);
+ ret = clk_prepare_enable(data->codec_clk);
+ if (ret) {
+ dev_err(&codec_dev->dev, "failed to enable codec clk: %d\n", ret);
+ goto fail;
+ }
+
+ data->dai.name = "HiFi";
+ data->dai.stream_name = "HiFi";
+ data->dai.codec_dai_name = "wm8962";
+ data->dai.codec_of_node = codec_np;
+ data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev);
+ data->dai.platform_of_node = ssi_np;
+ data->dai.ops = &imx_hifi_ops;
+ data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+
+ data->card.dev = &pdev->dev;
+ ret = snd_soc_of_parse_card_name(&data->card, "model");
+ if (ret)
+ goto clk_fail;
+ ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
+ if (ret)
+ goto clk_fail;
+ data->card.num_links = 1;
+ data->card.dai_link = &data->dai;
+ data->card.dapm_widgets = imx_wm8962_dapm_widgets;
+ data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets);
+
+ data->card.late_probe = imx_wm8962_late_probe;
+ data->card.set_bias_level = imx_wm8962_set_bias_level;
+
+ ret = snd_soc_register_card(&data->card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+ goto clk_fail;
+ }
+
+ platform_set_drvdata(pdev, data);
+ of_node_put(ssi_np);
+ of_node_put(codec_np);
+
+ return 0;
+
+clk_fail:
+ if (!IS_ERR(data->codec_clk))
+ clk_disable_unprepare(data->codec_clk);
+fail:
+ if (ssi_np)
+ of_node_put(ssi_np);
+ if (codec_np)
+ of_node_put(codec_np);
+
+ return ret;
+}
+
+static int imx_wm8962_remove(struct platform_device *pdev)
+{
+ struct imx_wm8962_data *data = platform_get_drvdata(pdev);
+
+ if (!IS_ERR(data->codec_clk))
+ clk_disable_unprepare(data->codec_clk);
+ snd_soc_unregister_card(&data->card);
+
+ return 0;
+}
+
+static const struct of_device_id imx_wm8962_dt_ids[] = {
+ { .compatible = "fsl,imx-audio-wm8962", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids);
+
+static struct platform_driver imx_wm8962_driver = {
+ .driver = {
+ .name = "imx-wm8962",
+ .owner = THIS_MODULE,
+ .of_match_table = imx_wm8962_dt_ids,
+ },
+ .probe = imx_wm8962_probe,
+ .remove = imx_wm8962_remove,
+};
+module_platform_driver(imx_wm8962_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-wm8962");
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
index 4141b35ef0bb..3ef7a0c92efa 100644
--- a/sound/soc/fsl/mpc5200_psc_ac97.c
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -131,13 +131,12 @@ static void psc_ac97_cold_reset(struct snd_ac97 *ac97)
psc_ac97_warm_reset(ac97);
}
-struct snd_ac97_bus_ops soc_ac97_ops = {
+static struct snd_ac97_bus_ops psc_ac97_ops = {
.read = psc_ac97_read,
.write = psc_ac97_write,
.reset = psc_ac97_cold_reset,
.warm_reset = psc_ac97_warm_reset,
};
-EXPORT_SYMBOL_GPL(soc_ac97_ops);
static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
@@ -290,6 +289,12 @@ static int psc_ac97_of_probe(struct platform_device *op)
if (rc != 0)
return rc;
+ rc = snd_soc_set_ac97_ops(&psc_ac97_ops);
+ if (rc != 0) {
+ dev_err(&op->dev, "Failed to set AC'97 ops: %d\n", ret);
+ return rc;
+ }
+
rc = snd_soc_register_component(&op->dev, &psc_ac97_component,
psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai));
if (rc != 0) {
@@ -318,6 +323,7 @@ static int psc_ac97_of_remove(struct platform_device *op)
{
mpc5200_audio_dma_destroy(op);
snd_soc_unregister_component(&op->dev);
+ snd_soc_set_ac97_ops(NULL);
return 0;
}
diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c
index 3d1074179057..f4c3bda5e69e 100644
--- a/sound/soc/fsl/mx27vis-aic32x4.c
+++ b/sound/soc/fsl/mx27vis-aic32x4.c
@@ -161,7 +161,7 @@ static struct snd_soc_dai_link mx27vis_aic32x4_dai = {
.name = "tlv320aic32x4",
.stream_name = "TLV320AIC32X4",
.codec_dai_name = "tlv320aic32x4-hifi",
- .platform_name = "imx-pcm-audio.0",
+ .platform_name = "imx-ssi.0",
.codec_name = "tlv320aic32x4.0-0018",
.cpu_dai_name = "imx-ssi.0",
.ops = &mx27vis_aic32x4_snd_ops,
diff --git a/sound/soc/fsl/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c
index f8da6dd115ed..ae403c29688f 100644
--- a/sound/soc/fsl/phycore-ac97.c
+++ b/sound/soc/fsl/phycore-ac97.c
@@ -33,7 +33,7 @@ static struct snd_soc_dai_link imx_phycore_dai_ac97[] = {
.codec_dai_name = "wm9712-hifi",
.codec_name = "wm9712-codec",
.cpu_dai_name = "imx-ssi.0",
- .platform_name = "imx-fiq-pcm-audio.0",
+ .platform_name = "imx-ssi.0",
.ops = &imx_phycore_hifi_ops,
},
};
diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
index fe54a69073e5..fce63252bdbb 100644
--- a/sound/soc/fsl/wm1133-ev1.c
+++ b/sound/soc/fsl/wm1133-ev1.c
@@ -245,7 +245,7 @@ static struct snd_soc_dai_link wm1133_ev1_dai = {
.stream_name = "Audio",
.cpu_dai_name = "imx-ssi.0",
.codec_dai_name = "wm8350-hifi",
- .platform_name = "imx-fiq-pcm-audio.0",
+ .platform_name = "imx-ssi.0",
.codec_name = "wm8350-codec.0-0x1a",
.init = wm1133_ev1_init,
.ops = &wm1133_ev1_ops,