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authorShengjiu Wang <shengjiu.wang@freescale.com>2015-07-17 14:18:21 +0800
committerNitin Garg <nitin.garg@freescale.com>2015-09-17 09:24:01 -0500
commit6d07037d44cb28b931b2b4541cfda8de95bbe9e0 (patch)
treeb76c4bdc5d7e48d970a9df87f8c0f71e81935d12 /sound
parent8554c92f2db690ac5940fe3b0e663481c904eb31 (diff)
MLK-11259-2: ASoC: fsl: enable ASRC p2p for ssi->wm8962
Enable ASRC p2p for ssi->wm8962, base on the new p2p script, which support to select dualfifo for source/destination device. Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Diffstat (limited to 'sound')
-rw-r--r--sound/soc/fsl/fsl_asrc_dma.c11
-rw-r--r--sound/soc/fsl/imx-wm8962.c105
2 files changed, 103 insertions, 13 deletions
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 28bc57f897a4..f0bf0b6657f5 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -1,7 +1,7 @@
/*
* Freescale ASRC ALSA SoC Platform (DMA) driver
*
- * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ * Copyright (C) 2014-2015 Freescale Semiconductor, Inc.
*
* Author: Nicolin Chen <nicoleotsuka@gmail.com>
*
@@ -157,6 +157,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream,
struct device *dev_be;
u8 dir = tx ? OUT : IN;
dma_cap_mask_t mask;
+ enum sdma_peripheral_type be_peripheral_type;
int ret;
/* Fetch the Back-End dma_data from DPCM */
@@ -212,6 +213,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream,
tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx");
tmp_data = tmp_chan->private;
pair->dma_data.dma_request = tmp_data->dma_request;
+ be_peripheral_type = tmp_data->peripheral_type;
dma_release_channel(tmp_chan);
/* Get DMA request of Front-End */
@@ -222,6 +224,11 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream,
pair->dma_data.priority = tmp_data->priority;
dma_release_channel(tmp_chan);
+ if (tx && be_peripheral_type == IMX_DMATYPE_SSI_DUAL)
+ pair->dma_data.dst_dualfifo = true;
+ if (!tx && be_peripheral_type == IMX_DMATYPE_SSI_DUAL)
+ pair->dma_data.src_dualfifo = true;
+
pair->dma_chan[dir] = dma_request_channel(mask, filter, &pair->dma_data);
if (!pair->dma_chan[dir]) {
dev_err(dev, "failed to request DMA channel for Back-End\n");
@@ -233,6 +240,8 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream,
else
buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ memset(&config_be, 0, sizeof(config_be));
+
config_be.direction = DMA_DEV_TO_DEV;
config_be.src_addr_width = buswidth;
config_be.src_maxburst = dma_params_be->maxburst;
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 4b28a76eed93..7b1cf6b1552e 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2013-2014 Freescale Semiconductor, Inc.
+ * Copyright (C) 2013-2015 Freescale Semiconductor, Inc.
*
* Based on imx-sgtl5000.c
* Copyright (C) 2012 Freescale Semiconductor, Inc.
@@ -33,7 +33,7 @@
#define DAI_NAME_SIZE 32
struct imx_wm8962_data {
- struct snd_soc_dai_link dai;
+ struct snd_soc_dai_link dai[3];
struct snd_soc_card card;
char codec_dai_name[DAI_NAME_SIZE];
char platform_name[DAI_NAME_SIZE];
@@ -53,6 +53,9 @@ struct imx_priv {
struct snd_pcm_substream *second_stream;
struct snd_kcontrol *headphone_kctl;
struct snd_card *snd_card;
+ struct platform_device *asrc_pdev;
+ u32 asrc_rate;
+ u32 asrc_format;
};
static struct imx_priv card_priv;
@@ -490,6 +493,25 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card)
return ret;
}
+static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params) {
+ struct imx_priv *priv = &card_priv;
+ struct snd_interval *rate;
+ struct snd_mask *mask;
+
+ if (!priv->asrc_pdev)
+ return -EINVAL;
+
+ rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ rate->max = rate->min = priv->asrc_rate;
+
+ mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ snd_mask_none(mask);
+ snd_mask_set(mask, priv->asrc_format);
+
+ return 0;
+}
+
static int imx_wm8962_probe(struct platform_device *pdev)
{
struct device_node *np = pdev->dev.of_node;
@@ -501,8 +523,12 @@ static int imx_wm8962_probe(struct platform_device *pdev)
struct clk *codec_clk = NULL;
int int_port, ext_port;
int ret;
+ struct platform_device *asrc_pdev = NULL;
+ struct device_node *asrc_np;
+ u32 width;
priv->pdev = pdev;
+ priv->asrc_pdev = NULL;
cpu_np = of_parse_phandle(pdev->dev.of_node, "cpu-dai", 0);
if (!cpu_np) {
@@ -571,6 +597,12 @@ audmux_bypass:
goto fail;
}
+ asrc_np = of_parse_phandle(pdev->dev.of_node, "asrc-controller", 0);
+ if (asrc_np) {
+ asrc_pdev = of_find_device_by_node(asrc_np);
+ priv->asrc_pdev = asrc_pdev;
+ }
+
priv->first_stream = NULL;
priv->second_stream = NULL;
@@ -597,16 +629,66 @@ audmux_bypass:
priv->mic_gpio = of_get_named_gpio_flags(np, "mic-det-gpios", 0,
(enum of_gpio_flags *)&priv->mic_active_low);
- 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(&cpu_pdev->dev);
- data->dai.platform_of_node = cpu_np;
- data->dai.ops = &imx_hifi_ops;
- data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ data->dai[0].name = "HiFi";
+ data->dai[0].stream_name = "HiFi";
+ data->dai[0].codec_dai_name = "wm8962";
+ data->dai[0].codec_of_node = codec_np;
+ data->dai[0].cpu_dai_name = dev_name(&cpu_pdev->dev);
+ data->dai[0].platform_of_node = cpu_np;
+ data->dai[0].ops = &imx_hifi_ops;
+ data->dai[0].dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM;
+ data->card.num_links = 1;
+
+ if (asrc_pdev) {
+ data->dai[0].ignore_pmdown_time = 1;
+ data->dai[1].name = "HiFi-ASRC-FE";
+ data->dai[1].stream_name = "HiFi-ASRC-FE";
+ data->dai[1].codec_name = "snd-soc-dummy";
+ data->dai[1].codec_dai_name = "snd-soc-dummy-dai";
+ data->dai[1].cpu_of_node = asrc_np;
+ data->dai[1].platform_of_node = asrc_np;
+ data->dai[1].dynamic = 1;
+ data->dai[1].ignore_pmdown_time = 1;
+ data->dai[1].dpcm_playback = 1;
+ data->dai[1].dpcm_capture = 1;
+
+ data->dai[2].name = "HiFi-ASRC-BE";
+ data->dai[2].stream_name = "HiFi-ASRC-BE";
+ data->dai[2].codec_dai_name = "wm8962";
+ data->dai[2].codec_of_node = codec_np;
+ data->dai[2].cpu_dai_name = dev_name(&cpu_pdev->dev);
+ data->dai[2].platform_name = "snd-soc-dummy";
+ data->dai[2].ops = &imx_hifi_ops;
+ data->dai[2].be_hw_params_fixup = be_hw_params_fixup;
+ data->dai[2].no_pcm = 1;
+ data->dai[2].ignore_pmdown_time = 1;
+ data->dai[2].dpcm_playback = 1;
+ data->dai[2].dpcm_capture = 1;
+ data->card.num_links = 3;
+
+ ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
+ &priv->asrc_rate);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to get output rate\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to get output rate\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ if (width == 24)
+ priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
+ else
+ priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
+ }
+
data->card.dev = &pdev->dev;
ret = snd_soc_of_parse_card_name(&data->card, "model");
if (ret)
@@ -614,8 +696,7 @@ audmux_bypass:
ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
if (ret)
goto fail;
- data->card.num_links = 1;
- data->card.dai_link = &data->dai;
+ 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);