Merge remote-tracking branches 'asoc/topic/blackfin', 'asoc/topic/davinci', 'asoc/topic/fsl', 'asoc/topic/hdmi' and 'asoc/topic/intel' into asoc-next
diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c
index 5bf1501..864df26 100644
--- a/sound/soc/blackfin/bf5xx-ad1836.c
+++ b/sound/soc/blackfin/bf5xx-ad1836.c
@@ -87,27 +87,18 @@
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "Failed to register card\n");
return ret;
}
-static int bf5xx_ad1836_driver_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver bf5xx_ad1836_driver = {
.driver = {
.name = "bfin-snd-ad1836",
.pm = &snd_soc_pm_ops,
},
.probe = bf5xx_ad1836_driver_probe,
- .remove = bf5xx_ad1836_driver_remove,
};
module_platform_driver(bf5xx_ad1836_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c
index 523baf58..72ac789 100644
--- a/sound/soc/blackfin/bfin-eval-adau1373.c
+++ b/sound/soc/blackfin/bfin-eval-adau1373.c
@@ -154,16 +154,7 @@
card->dev = &pdev->dev;
- return snd_soc_register_card(&bfin_eval_adau1373);
-}
-
-static int bfin_eval_adau1373_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1373);
}
static struct platform_driver bfin_eval_adau1373_driver = {
@@ -172,7 +163,6 @@
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adau1373_probe,
- .remove = bfin_eval_adau1373_remove,
};
module_platform_driver(bfin_eval_adau1373_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c
index f9e926d..5c67f72 100644
--- a/sound/soc/blackfin/bfin-eval-adau1701.c
+++ b/sound/soc/blackfin/bfin-eval-adau1701.c
@@ -94,16 +94,7 @@
card->dev = &pdev->dev;
- return snd_soc_register_card(&bfin_eval_adau1701);
-}
-
-static int bfin_eval_adau1701_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1701);
}
static struct platform_driver bfin_eval_adau1701_driver = {
@@ -112,7 +103,6 @@
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adau1701_probe,
- .remove = bfin_eval_adau1701_remove,
};
module_platform_driver(bfin_eval_adau1701_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c
index 27eee66..1037477 100644
--- a/sound/soc/blackfin/bfin-eval-adav80x.c
+++ b/sound/soc/blackfin/bfin-eval-adav80x.c
@@ -119,16 +119,7 @@
card->dev = &pdev->dev;
- return snd_soc_register_card(&bfin_eval_adav80x);
-}
-
-static int bfin_eval_adav80x_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adav80x);
}
static const struct platform_device_id bfin_eval_adav80x_ids[] = {
@@ -144,7 +135,6 @@
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adav80x_probe,
- .remove = bfin_eval_adav80x_remove,
.id_table = bfin_eval_adav80x_ids,
};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index a92e4d4..70e5a75 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -80,7 +80,6 @@
select SND_SOC_MAX9877 if I2C
select SND_SOC_MC13783 if MFD_MC13XXX
select SND_SOC_ML26124 if I2C
- select SND_SOC_HDMI_CODEC
select SND_SOC_PCM1681 if I2C
select SND_SOC_PCM1792A if SPI_MASTER
select SND_SOC_PCM3008
@@ -447,9 +446,6 @@
config SND_SOC_DMIC
tristate
-config SND_SOC_HDMI_CODEC
- tristate "HDMI stub CODEC"
-
config SND_SOC_ES8328
tristate "Everest Semi ES8328 CODEC"
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 5b6c8af..be1491a 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -73,7 +73,6 @@
snd-soc-max9850-objs := max9850.o
snd-soc-mc13783-objs := mc13783.o
snd-soc-ml26124-objs := ml26124.o
-snd-soc-hdmi-codec-objs := hdmi.o
snd-soc-pcm1681-objs := pcm1681.o
snd-soc-pcm1792a-codec-objs := pcm1792a.o
snd-soc-pcm3008-objs := pcm3008.o
@@ -266,7 +265,6 @@
obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o
obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o
obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o
-obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o
obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o
obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c
deleted file mode 100644
index bd42ad3..0000000
--- a/sound/soc/codecs/hdmi.c
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- * ALSA SoC codec driver for HDMI audio codecs.
- * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/
- * Author: Ricardo Neri <ricardo.neri@ti.com>
- *
- * 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 program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-#include <linux/module.h>
-#include <sound/soc.h>
-#include <linux/of.h>
-#include <linux/of_device.h>
-
-#define DRV_NAME "hdmi-audio-codec"
-
-static const struct snd_soc_dapm_widget hdmi_widgets[] = {
- SND_SOC_DAPM_INPUT("RX"),
- SND_SOC_DAPM_OUTPUT("TX"),
-};
-
-static const struct snd_soc_dapm_route hdmi_routes[] = {
- { "Capture", NULL, "RX" },
- { "TX", NULL, "Playback" },
-};
-
-static struct snd_soc_dai_driver hdmi_codec_dai = {
- .name = "hdmi-hifi",
- .playback = {
- .stream_name = "Playback",
- .channels_min = 2,
- .channels_max = 8,
- .rates = SNDRV_PCM_RATE_32000 |
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
- SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE,
- .sig_bits = 24,
- },
- .capture = {
- .stream_name = "Capture",
- .channels_min = 2,
- .channels_max = 2,
- .rates = SNDRV_PCM_RATE_32000 |
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
- SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S24_LE,
- },
-
-};
-
-#ifdef CONFIG_OF
-static const struct of_device_id hdmi_audio_codec_ids[] = {
- { .compatible = "linux,hdmi-audio", },
- { }
-};
-MODULE_DEVICE_TABLE(of, hdmi_audio_codec_ids);
-#endif
-
-static struct snd_soc_codec_driver hdmi_codec = {
- .dapm_widgets = hdmi_widgets,
- .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets),
- .dapm_routes = hdmi_routes,
- .num_dapm_routes = ARRAY_SIZE(hdmi_routes),
- .ignore_pmdown_time = true,
-};
-
-static int hdmi_codec_probe(struct platform_device *pdev)
-{
- return snd_soc_register_codec(&pdev->dev, &hdmi_codec,
- &hdmi_codec_dai, 1);
-}
-
-static int hdmi_codec_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_codec(&pdev->dev);
- return 0;
-}
-
-static struct platform_driver hdmi_codec_driver = {
- .driver = {
- .name = DRV_NAME,
- .of_match_table = of_match_ptr(hdmi_audio_codec_ids),
- },
-
- .probe = hdmi_codec_probe,
- .remove = hdmi_codec_remove,
-};
-
-module_platform_driver(hdmi_codec_driver);
-
-MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
-MODULE_DESCRIPTION("ASoC generic HDMI codec driver");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 7d45d98..4495a40 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -80,12 +80,13 @@
/* McASP specific data */
int tdm_slots;
+ u32 tdm_mask[2];
+ int slot_width;
u8 op_mode;
u8 num_serializer;
u8 *serial_dir;
u8 version;
u8 bclk_div;
- u16 bclk_lrclk_ratio;
int streams;
u32 irq_request[2];
int dma_request[2];
@@ -556,8 +557,21 @@
mcasp->bclk_div = div;
break;
- case 2: /* BCLK/LRCLK ratio */
- mcasp->bclk_lrclk_ratio = div;
+ case 2: /*
+ * BCLK/LRCLK ratio descries how many bit-clock cycles
+ * fit into one frame. The clock ratio is given for a
+ * full period of data (for I2S format both left and
+ * right channels), so it has to be divided by number
+ * of tdm-slots (for I2S - divided by 2).
+ * Instead of storing this ratio, we calculate a new
+ * tdm_slot width by dividing the the ratio by the
+ * number of configured tdm slots.
+ */
+ mcasp->slot_width = div / mcasp->tdm_slots;
+ if (div % mcasp->tdm_slots)
+ dev_warn(mcasp->dev,
+ "%s(): BCLK/LRCLK %d is not divisible by %d tdm slots",
+ __func__, div, mcasp->tdm_slots);
break;
default:
@@ -596,12 +610,92 @@
return 0;
}
+/* All serializers must have equal number of channels */
+static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp, int stream,
+ int serializers)
+{
+ struct snd_pcm_hw_constraint_list *cl = &mcasp->chconstr[stream];
+ unsigned int *list = (unsigned int *) cl->list;
+ int slots = mcasp->tdm_slots;
+ int i, count = 0;
+
+ if (mcasp->tdm_mask[stream])
+ slots = hweight32(mcasp->tdm_mask[stream]);
+
+ for (i = 2; i <= slots; i++)
+ list[count++] = i;
+
+ for (i = 2; i <= serializers; i++)
+ list[count++] = i*slots;
+
+ cl->count = count;
+
+ return 0;
+}
+
+static int davinci_mcasp_set_ch_constraints(struct davinci_mcasp *mcasp)
+{
+ int rx_serializers = 0, tx_serializers = 0, ret, i;
+
+ for (i = 0; i < mcasp->num_serializer; i++)
+ if (mcasp->serial_dir[i] == TX_MODE)
+ tx_serializers++;
+ else if (mcasp->serial_dir[i] == RX_MODE)
+ rx_serializers++;
+
+ ret = davinci_mcasp_ch_constraint(mcasp, SNDRV_PCM_STREAM_PLAYBACK,
+ tx_serializers);
+ if (ret)
+ return ret;
+
+ ret = davinci_mcasp_ch_constraint(mcasp, SNDRV_PCM_STREAM_CAPTURE,
+ rx_serializers);
+
+ return ret;
+}
+
+
+static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai,
+ unsigned int tx_mask,
+ unsigned int rx_mask,
+ int slots, int slot_width)
+{
+ struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
+
+ dev_dbg(mcasp->dev,
+ "%s() tx_mask 0x%08x rx_mask 0x%08x slots %d width %d\n",
+ __func__, tx_mask, rx_mask, slots, slot_width);
+
+ if (tx_mask >= (1<<slots) || rx_mask >= (1<<slots)) {
+ dev_err(mcasp->dev,
+ "Bad tdm mask tx: 0x%08x rx: 0x%08x slots %d\n",
+ tx_mask, rx_mask, slots);
+ return -EINVAL;
+ }
+
+ if (slot_width &&
+ (slot_width < 8 || slot_width > 32 || slot_width % 4 != 0)) {
+ dev_err(mcasp->dev, "%s: Unsupported slot_width %d\n",
+ __func__, slot_width);
+ return -EINVAL;
+ }
+
+ mcasp->tdm_slots = slots;
+ mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = rx_mask;
+ mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = tx_mask;
+ mcasp->slot_width = slot_width;
+
+ return davinci_mcasp_set_ch_constraints(mcasp);
+}
+
static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
- int word_length)
+ int sample_width)
{
u32 fmt;
- u32 tx_rotate = (word_length / 4) & 0x7;
- u32 mask = (1ULL << word_length) - 1;
+ u32 tx_rotate = (sample_width / 4) & 0x7;
+ u32 mask = (1ULL << sample_width) - 1;
+ u32 slot_width = sample_width;
+
/*
* For captured data we should not rotate, inversion and masking is
* enoguh to get the data to the right position:
@@ -614,28 +708,23 @@
u32 rx_rotate = 0;
/*
- * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv()
- * callback, take it into account here. That allows us to for example
- * send 32 bits per channel to the codec, while only 16 of them carry
- * audio payload.
- * The clock ratio is given for a full period of data (for I2S format
- * both left and right channels), so it has to be divided by number of
- * tdm-slots (for I2S - divided by 2).
+ * Setting the tdm slot width either with set_clkdiv() or
+ * set_tdm_slot() allows us to for example send 32 bits per
+ * channel to the codec, while only 16 of them carry audio
+ * payload.
*/
- if (mcasp->bclk_lrclk_ratio) {
- u32 slot_length = mcasp->bclk_lrclk_ratio / mcasp->tdm_slots;
-
+ if (mcasp->slot_width) {
/*
- * When we have more bclk then it is needed for the data, we
- * need to use the rotation to move the received samples to have
- * correct alignment.
+ * When we have more bclk then it is needed for the
+ * data, we need to use the rotation to move the
+ * received samples to have correct alignment.
*/
- rx_rotate = (slot_length - word_length) / 4;
- word_length = slot_length;
+ slot_width = mcasp->slot_width;
+ rx_rotate = (slot_width - sample_width) / 4;
}
/* mapping of the XSSZ bit-field as described in the datasheet */
- fmt = (word_length >> 1) - 1;
+ fmt = (slot_width >> 1) - 1;
if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) {
mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXSSZ(fmt),
@@ -776,33 +865,50 @@
/*
* If more than one serializer is needed, then use them with
- * their specified tdm_slots count. Otherwise, one serializer
- * can cope with the transaction using as many slots as channels
- * in the stream, requires channels symmetry
+ * all the specified tdm_slots. Otherwise, one serializer can
+ * cope with the transaction using just as many slots as there
+ * are channels in the stream.
*/
- active_serializers = (channels + total_slots - 1) / total_slots;
- if (active_serializers == 1)
- active_slots = channels;
- else
- active_slots = total_slots;
+ if (mcasp->tdm_mask[stream]) {
+ active_slots = hweight32(mcasp->tdm_mask[stream]);
+ active_serializers = (channels + active_slots - 1) /
+ active_slots;
+ if (active_serializers == 1) {
+ active_slots = channels;
+ for (i = 0; i < total_slots; i++) {
+ if ((1 << i) & mcasp->tdm_mask[stream]) {
+ mask |= (1 << i);
+ if (--active_slots <= 0)
+ break;
+ }
+ }
+ }
+ } else {
+ active_serializers = (channels + total_slots - 1) / total_slots;
+ if (active_serializers == 1)
+ active_slots = channels;
+ else
+ active_slots = total_slots;
- for (i = 0; i < active_slots; i++)
- mask |= (1 << i);
-
+ for (i = 0; i < active_slots; i++)
+ mask |= (1 << i);
+ }
mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC);
if (!mcasp->dat_port)
busel = TXSEL;
- mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
- mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
- mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
- FSXMOD(total_slots), FSXMOD(0x1FF));
-
- mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
- mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
- mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
- FSRMOD(total_slots), FSRMOD(0x1FF));
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
+ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
+ mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
+ FSXMOD(total_slots), FSXMOD(0x1FF));
+ } else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
+ mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
+ mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
+ FSRMOD(total_slots), FSRMOD(0x1FF));
+ }
return 0;
}
@@ -922,6 +1028,9 @@
int sbits = params_width(params);
int ppm, div;
+ if (mcasp->slot_width)
+ sbits = mcasp->slot_width;
+
div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*slots,
&ppm);
if (ppm)
@@ -1027,6 +1136,9 @@
struct snd_interval range;
int i;
+ if (rd->mcasp->slot_width)
+ sbits = rd->mcasp->slot_width;
+
snd_interval_any(&range);
range.empty = 1;
@@ -1069,10 +1181,14 @@
for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) {
if (snd_mask_test(fmt, i)) {
- uint bclk_freq = snd_pcm_format_width(i)*slots*rate;
+ uint sbits = snd_pcm_format_width(i);
int ppm;
- davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm);
+ if (rd->mcasp->slot_width)
+ sbits = rd->mcasp->slot_width;
+
+ davinci_mcasp_calc_clk_div(rd->mcasp, sbits*slots*rate,
+ &ppm);
if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) {
snd_mask_set(&nfmt, i);
count++;
@@ -1094,6 +1210,10 @@
&mcasp->ruledata[substream->stream];
u32 max_channels = 0;
int i, dir;
+ int tdm_slots = mcasp->tdm_slots;
+
+ if (mcasp->tdm_mask[substream->stream])
+ tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]);
mcasp->substreams[substream->stream] = substream;
@@ -1114,7 +1234,7 @@
max_channels++;
}
ruledata->serializers = max_channels;
- max_channels *= mcasp->tdm_slots;
+ max_channels *= tdm_slots;
/*
* If the already active stream has less channels than the calculated
* limnit based on the seirializers * tdm_slots, we need to use that as
@@ -1124,15 +1244,25 @@
*/
if (mcasp->channels && mcasp->channels < max_channels)
max_channels = mcasp->channels;
+ /*
+ * But we can always allow channels upto the amount of
+ * the available tdm_slots.
+ */
+ if (max_channels < tdm_slots)
+ max_channels = tdm_slots;
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_CHANNELS,
2, max_channels);
- if (mcasp->chconstr[substream->stream].count)
- snd_pcm_hw_constraint_list(substream->runtime,
- 0, SNDRV_PCM_HW_PARAM_CHANNELS,
- &mcasp->chconstr[substream->stream]);
+ snd_pcm_hw_constraint_list(substream->runtime,
+ 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &mcasp->chconstr[substream->stream]);
+
+ if (mcasp->slot_width)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ 8, mcasp->slot_width);
/*
* If we rely on implicit BCLK divider setting we should
@@ -1184,6 +1314,7 @@
.set_fmt = davinci_mcasp_set_dai_fmt,
.set_clkdiv = davinci_mcasp_set_clkdiv,
.set_sysclk = davinci_mcasp_set_sysclk,
+ .set_tdm_slot = davinci_mcasp_set_tdm_slot,
};
static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai)
@@ -1514,59 +1645,6 @@
return pdata;
}
-/* All serializers must have equal number of channels */
-static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp,
- struct snd_pcm_hw_constraint_list *cl,
- int serializers)
-{
- unsigned int *list;
- int i, count = 0;
-
- if (serializers <= 1)
- return 0;
-
- list = devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
- (mcasp->tdm_slots + serializers - 2),
- GFP_KERNEL);
- if (!list)
- return -ENOMEM;
-
- for (i = 2; i <= mcasp->tdm_slots; i++)
- list[count++] = i;
-
- for (i = 2; i <= serializers; i++)
- list[count++] = i*mcasp->tdm_slots;
-
- cl->count = count;
- cl->list = list;
-
- return 0;
-}
-
-
-static int davinci_mcasp_init_ch_constraints(struct davinci_mcasp *mcasp)
-{
- int rx_serializers = 0, tx_serializers = 0, ret, i;
-
- for (i = 0; i < mcasp->num_serializer; i++)
- if (mcasp->serial_dir[i] == TX_MODE)
- tx_serializers++;
- else if (mcasp->serial_dir[i] == RX_MODE)
- rx_serializers++;
-
- ret = davinci_mcasp_ch_constraint(mcasp, &mcasp->chconstr[
- SNDRV_PCM_STREAM_PLAYBACK],
- tx_serializers);
- if (ret)
- return ret;
-
- ret = davinci_mcasp_ch_constraint(mcasp, &mcasp->chconstr[
- SNDRV_PCM_STREAM_CAPTURE],
- rx_serializers);
-
- return ret;
-}
-
enum {
PCM_EDMA,
PCM_SDMA,
@@ -1783,7 +1861,28 @@
mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE;
}
- ret = davinci_mcasp_init_ch_constraints(mcasp);
+ /* Allocate memory for long enough list for all possible
+ * scenarios. Maximum number tdm slots is 32 and there cannot
+ * be more serializers than given in the configuration. The
+ * serializer directions could be taken into account, but it
+ * would make code much more complex and save only couple of
+ * bytes.
+ */
+ mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list =
+ devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
+ (32 + mcasp->num_serializer - 2),
+ GFP_KERNEL);
+
+ mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list =
+ devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
+ (32 + mcasp->num_serializer - 2),
+ GFP_KERNEL);
+
+ if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list ||
+ !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list)
+ return -ENOMEM;
+
+ ret = davinci_mcasp_set_ch_constraints(mcasp);
if (ret)
goto err;
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 96f55ae..0901d5e 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -593,6 +593,7 @@
{ .compatible = "fsl,imx-audio-wm8960", },
{}
};
+MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
static struct platform_driver fsl_asoc_card_driver = {
.probe = fsl_asoc_card_probe,
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index a18fd92..9366b5a 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -801,6 +801,7 @@
{ .compatible = "fsl,imx6sx-sai", },
{ /* sentinel */ }
};
+MODULE_DEVICE_TABLE(of, fsl_sai_ids);
static struct platform_driver fsl_sai_driver = {
.probe = fsl_sai_probe,
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 683e501..5e9c316 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -368,23 +368,6 @@
kfree(stream);
}
-static inline unsigned int get_current_pipe_id(struct snd_soc_dai *dai,
- struct snd_pcm_substream *substream)
-{
- struct sst_data *sst = snd_soc_dai_get_drvdata(dai);
- struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map;
- struct sst_runtime_stream *stream =
- substream->runtime->private_data;
- u32 str_id = stream->stream_info.str_id;
- unsigned int pipe_id;
-
- pipe_id = map[str_id].device_id;
-
- dev_dbg(dai->dev, "got pipe_id = %#x for str_id = %d\n",
- pipe_id, str_id);
- return pipe_id;
-}
-
static int sst_media_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 8bafaf6..3f8a1e1 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -266,18 +266,11 @@
{
broadwell_rt286.dev = &pdev->dev;
- return snd_soc_register_card(&broadwell_rt286);
-}
-
-static int broadwell_audio_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_card(&broadwell_rt286);
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &broadwell_rt286);
}
static struct platform_driver broadwell_audio = {
.probe = broadwell_audio_probe,
- .remove = broadwell_audio_remove,
.driver = {
.name = "broadwell-audio",
},
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 7d617bf..bea2673 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -510,17 +510,6 @@
},
},
{
- .name = "DMIC23 Pin",
- .ops = &skl_dmic_dai_ops,
- .capture = {
- .stream_name = "DMIC23 Rx",
- .channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
- .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
- },
-},
-{
.name = "HD-Codec Pin",
.ops = &skl_link_dai_ops,
.playback = {
@@ -538,28 +527,6 @@
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
-{
- .name = "HD-Codec-SPK Pin",
- .ops = &skl_link_dai_ops,
- .playback = {
- .stream_name = "HD-Codec-SPK Tx",
- .channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
- .rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
- },
-},
-{
- .name = "HD-Codec-AMIC Pin",
- .ops = &skl_link_dai_ops,
- .capture = {
- .stream_name = "HD-Codec-AMIC Rx",
- .channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
- .rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
- },
-},
};
static int skl_platform_open(struct snd_pcm_substream *substream)