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authorMark Brown <broonie@kernel.org>2015-06-22 10:24:19 +0100
committerMark Brown <broonie@kernel.org>2015-06-22 10:24:19 +0100
commit208a128f6b19eedd1819cb1b19f29dc99ca1d27e (patch)
tree5d95c61efeb06ed6ec827267bc05e89498836bec /sound/soc/fsl
parentd21504d4c993838b31d970d392b1b78eb33cfd61 (diff)
parent11e688862c4c8162119a4ca51c3326555966c8bb (diff)
Merge tag 'asoc-v4.2' into asoc-next
ASoC: Updates for v4.2 The big thing this release has been Liam's addition of topology support to the core. We've also seen quite a bit of driver work and the continuation of Lars' refactoring for component support. - Support for loading ASoC topology maps from firmware, intended to be used to allow self-describing DSP firmware images to be built which can map controls added by the DSP to userspace without the kernel needing to know about individual DSP firmwares. - Lots of refactoring to avoid direct access to snd_soc_codec where it's not needed supporting future refactoring. - Big refactoring and cleanup serieses for the Wolfson ADSP and TI TAS2552 drivers. - Support for TI TAS571x power amplifiers. - Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs. - Support for x86 systems with RT5650 and Qualcomm Storm. # gpg: Signature made Mon 08 Jun 2015 18:48:37 BST using RSA key ID 5D5487D0 # gpg: Oops: keyid_from_fingerprint: no pubkey # gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>" # gpg: aka "Mark Brown <broonie@debian.org>" # gpg: aka "Mark Brown <broonie@kernel.org>" # gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>" # gpg: aka "Mark Brown <broonie@linaro.org>" # gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
Diffstat (limited to 'sound/soc/fsl')
-rw-r--r--sound/soc/fsl/fsl_dma.c4
-rw-r--r--sound/soc/fsl/fsl_sai.c144
-rw-r--r--sound/soc/fsl/fsl_sai.h9
-rw-r--r--sound/soc/fsl/fsl_spdif.c10
-rw-r--r--sound/soc/fsl/fsl_ssi.c7
-rw-r--r--sound/soc/fsl/imx-audmux.c2
-rw-r--r--sound/soc/fsl/imx-mc13783.c6
7 files changed, 151 insertions, 31 deletions
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 93d7e56c6066..ccadefceeff2 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -445,7 +445,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
return ret;
}
- dma->assigned = 1;
+ dma->assigned = true;
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware);
@@ -814,7 +814,7 @@ static int fsl_dma_close(struct snd_pcm_substream *substream)
substream->runtime->private_data = NULL;
}
- dma->assigned = 0;
+ dma->assigned = false;
return 0;
}
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index ec79c3d5e65e..5c73bea7b11e 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -1,7 +1,7 @@
/*
* Freescale ALSA SoC Digital Audio Interface (SAI) driver.
*
- * Copyright 2012-2013 Freescale Semiconductor, Inc.
+ * Copyright 2012-2015 Freescale Semiconductor, Inc.
*
* 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
@@ -27,6 +27,17 @@
#define FSL_SAI_FLAGS (FSL_SAI_CSR_SEIE |\
FSL_SAI_CSR_FEIE)
+static u32 fsl_sai_rates[] = {
+ 8000, 11025, 12000, 16000, 22050,
+ 24000, 32000, 44100, 48000, 64000,
+ 88200, 96000, 176400, 192000
+};
+
+static struct snd_pcm_hw_constraint_list fsl_sai_rate_constraints = {
+ .count = ARRAY_SIZE(fsl_sai_rates),
+ .list = fsl_sai_rates,
+};
+
static irqreturn_t fsl_sai_isr(int irq, void *devid)
{
struct fsl_sai *sai = (struct fsl_sai *)devid;
@@ -251,12 +262,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
break;
case SND_SOC_DAIFMT_CBM_CFM:
+ sai->is_slave_mode = true;
break;
case SND_SOC_DAIFMT_CBS_CFM:
val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
break;
case SND_SOC_DAIFMT_CBM_CFS:
val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
+ sai->is_slave_mode = true;
break;
default:
return -EINVAL;
@@ -288,6 +301,79 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
return ret;
}
+static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq)
+{
+ struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai);
+ unsigned long clk_rate;
+ u32 savediv = 0, ratio, savesub = freq;
+ u32 id;
+ int ret = 0;
+
+ /* Don't apply to slave mode */
+ if (sai->is_slave_mode)
+ return 0;
+
+ for (id = 0; id < FSL_SAI_MCLK_MAX; id++) {
+ clk_rate = clk_get_rate(sai->mclk_clk[id]);
+ if (!clk_rate)
+ continue;
+
+ ratio = clk_rate / freq;
+
+ ret = clk_rate - ratio * freq;
+
+ /*
+ * Drop the source that can not be
+ * divided into the required rate.
+ */
+ if (ret != 0 && clk_rate / ret < 1000)
+ continue;
+
+ dev_dbg(dai->dev,
+ "ratio %d for freq %dHz based on clock %ldHz\n",
+ ratio, freq, clk_rate);
+
+ if (ratio % 2 == 0 && ratio >= 2 && ratio <= 512)
+ ratio /= 2;
+ else
+ continue;
+
+ if (ret < savesub) {
+ savediv = ratio;
+ sai->mclk_id[tx] = id;
+ savesub = ret;
+ }
+
+ if (ret == 0)
+ break;
+ }
+
+ if (savediv == 0) {
+ dev_err(dai->dev, "failed to derive required %cx rate: %d\n",
+ tx ? 'T' : 'R', freq);
+ return -EINVAL;
+ }
+
+ if ((tx && sai->synchronous[TX]) || (!tx && !sai->synchronous[RX])) {
+ regmap_update_bits(sai->regmap, FSL_SAI_RCR2,
+ FSL_SAI_CR2_MSEL_MASK,
+ FSL_SAI_CR2_MSEL(sai->mclk_id[tx]));
+ regmap_update_bits(sai->regmap, FSL_SAI_RCR2,
+ FSL_SAI_CR2_DIV_MASK, savediv - 1);
+ } else {
+ regmap_update_bits(sai->regmap, FSL_SAI_TCR2,
+ FSL_SAI_CR2_MSEL_MASK,
+ FSL_SAI_CR2_MSEL(sai->mclk_id[tx]));
+ regmap_update_bits(sai->regmap, FSL_SAI_TCR2,
+ FSL_SAI_CR2_DIV_MASK, savediv - 1);
+ }
+
+ dev_dbg(dai->dev, "best fit: clock id=%d, div=%d, deviation =%d\n",
+ sai->mclk_id[tx], savediv, savesub);
+
+ return 0;
+}
+
static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *cpu_dai)
@@ -297,6 +383,24 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
unsigned int channels = params_channels(params);
u32 word_width = snd_pcm_format_width(params_format(params));
u32 val_cr4 = 0, val_cr5 = 0;
+ int ret;
+
+ if (!sai->is_slave_mode) {
+ ret = fsl_sai_set_bclk(cpu_dai, tx,
+ 2 * word_width * params_rate(params));
+ if (ret)
+ return ret;
+
+ /* Do not enable the clock if it is already enabled */
+ if (!(sai->mclk_streams & BIT(substream->stream))) {
+ ret = clk_prepare_enable(sai->mclk_clk[sai->mclk_id[tx]]);
+ if (ret)
+ return ret;
+
+ sai->mclk_streams |= BIT(substream->stream);
+ }
+
+ }
if (!sai->is_dsp_mode)
val_cr4 |= FSL_SAI_CR4_SYWD(word_width);
@@ -322,6 +426,22 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
return 0;
}
+static int fsl_sai_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+
+ if (!sai->is_slave_mode &&
+ sai->mclk_streams & BIT(substream->stream)) {
+ clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[tx]]);
+ sai->mclk_streams &= ~BIT(substream->stream);
+ }
+
+ return 0;
+}
+
+
static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *cpu_dai)
{
@@ -410,7 +530,10 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream,
regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx), FSL_SAI_CR3_TRCE,
FSL_SAI_CR3_TRCE);
- return 0;
+ ret = snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &fsl_sai_rate_constraints);
+
+ return ret;
}
static void fsl_sai_shutdown(struct snd_pcm_substream *substream,
@@ -428,6 +551,7 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
.set_sysclk = fsl_sai_set_dai_sysclk,
.set_fmt = fsl_sai_set_dai_fmt,
.hw_params = fsl_sai_hw_params,
+ .hw_free = fsl_sai_hw_free,
.trigger = fsl_sai_trigger,
.startup = fsl_sai_startup,
.shutdown = fsl_sai_shutdown,
@@ -463,14 +587,18 @@ static struct snd_soc_dai_driver fsl_sai_dai = {
.stream_name = "CPU-Playback",
.channels_min = 1,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_96000,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .rates = SNDRV_PCM_RATE_KNOT,
.formats = FSL_SAI_FORMATS,
},
.capture = {
.stream_name = "CPU-Capture",
.channels_min = 1,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_96000,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .rates = SNDRV_PCM_RATE_KNOT,
.formats = FSL_SAI_FORMATS,
},
.ops = &fsl_sai_pcm_dai_ops,
@@ -600,8 +728,9 @@ static int fsl_sai_probe(struct platform_device *pdev)
sai->bus_clk = NULL;
}
- for (i = 0; i < FSL_SAI_MCLK_MAX; i++) {
- sprintf(tmp, "mclk%d", i + 1);
+ sai->mclk_clk[0] = sai->bus_clk;
+ for (i = 1; i < FSL_SAI_MCLK_MAX; i++) {
+ sprintf(tmp, "mclk%d", i);
sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp);
if (IS_ERR(sai->mclk_clk[i])) {
dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n",
@@ -664,8 +793,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
if (sai->sai_on_imx)
return imx_pcm_dma_init(pdev);
else
- return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL,
- SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
+ return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
}
static const struct of_device_id fsl_sai_ids[] = {
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 34667209b607..066280953c85 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -72,13 +72,15 @@
/* SAI Transmit and Recieve Configuration 2 Register */
#define FSL_SAI_CR2_SYNC BIT(30)
-#define FSL_SAI_CR2_MSEL_MASK (0xff << 26)
+#define FSL_SAI_CR2_MSEL_MASK (0x3 << 26)
#define FSL_SAI_CR2_MSEL_BUS 0
#define FSL_SAI_CR2_MSEL_MCLK1 BIT(26)
#define FSL_SAI_CR2_MSEL_MCLK2 BIT(27)
#define FSL_SAI_CR2_MSEL_MCLK3 (BIT(26) | BIT(27))
+#define FSL_SAI_CR2_MSEL(ID) ((ID) << 26)
#define FSL_SAI_CR2_BCP BIT(25)
#define FSL_SAI_CR2_BCD_MSTR BIT(24)
+#define FSL_SAI_CR2_DIV_MASK 0xff
/* SAI Transmit and Recieve Configuration 3 Register */
#define FSL_SAI_CR3_TRCE BIT(16)
@@ -120,7 +122,7 @@
#define FSL_SAI_CLK_MAST2 2
#define FSL_SAI_CLK_MAST3 3
-#define FSL_SAI_MCLK_MAX 3
+#define FSL_SAI_MCLK_MAX 4
/* SAI data transfer numbers per DMA request */
#define FSL_SAI_MAXBURST_TX 6
@@ -132,11 +134,14 @@ struct fsl_sai {
struct clk *bus_clk;
struct clk *mclk_clk[FSL_SAI_MCLK_MAX];
+ bool is_slave_mode;
bool is_lsb_first;
bool is_dsp_mode;
bool sai_on_imx;
bool synchronous[2];
+ unsigned int mclk_id[2];
+ unsigned int mclk_streams;
struct snd_dmaengine_dai_dma_data dma_params_rx;
struct snd_dmaengine_dai_dma_data dma_params_tx;
};
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 91eb3aef7f02..8e932219cb3a 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -417,11 +417,9 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
if (clk != STC_TXCLK_SPDIF_ROOT)
goto clk_set_bypass;
- /*
- * The S/PDIF block needs a clock of 64 * fs * txclk_df.
- * So request 64 * fs * (txclk_df + 1) to get rounded.
- */
- ret = clk_set_rate(spdif_priv->txclk[rate], 64 * sample_rate * (txclk_df + 1));
+ /* The S/PDIF block needs a clock of 64 * fs * txclk_df */
+ ret = clk_set_rate(spdif_priv->txclk[rate],
+ 64 * sample_rate * txclk_df);
if (ret) {
dev_err(&pdev->dev, "failed to set tx clock rate\n");
return ret;
@@ -1060,7 +1058,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) {
for (txclk_df = 1; txclk_df <= 128; txclk_df++) {
- rate_ideal = rate[index] * (txclk_df + 1) * 64;
+ rate_ideal = rate[index] * txclk_df * 64;
if (round)
rate_actual = clk_round_rate(clk, rate_ideal);
else
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 0d48804218b1..c7647e066cfd 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1292,13 +1292,6 @@ static int fsl_ssi_probe(struct platform_device *pdev)
void __iomem *iomem;
char name[64];
- /* SSIs that are not connected on the board should have a
- * status = "disabled"
- * property in their device tree nodes.
- */
- if (!of_device_is_available(np))
- return -ENODEV;
-
of_id = of_match_device(fsl_ssi_ids, &pdev->dev);
if (!of_id || !of_id->data)
return -EINVAL;
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index d9050d946ae7..fc57da341d61 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -184,7 +184,7 @@ static enum imx_audmux_type {
IMX31_AUDMUX,
} audmux_type;
-static struct platform_device_id imx_audmux_ids[] = {
+static const struct platform_device_id imx_audmux_ids[] = {
{
.name = "imx21-audmux",
.driver_data = IMX21_AUDMUX,
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index 9e6493d4e7ff..bb0459018b45 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -45,11 +45,7 @@ static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream,
if (ret)
return ret;
- ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 16);
- if (ret)
- return ret;
-
- return 0;
+ return snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 16);
}
static struct snd_soc_ops imx_mc13783_hifi_ops = {