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Merge remote-tracking branches 'asoc/topic/blackfin', 'asoc/topic/davinci', 'asoc/topic/fsl', 'asoc/topic/hdmi' and 'asoc/topic/intel' into asoc-next

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This commit is contained in:
Mark Brown 2015-09-23 11:01:19 -07:00
commit f7b93159ab
13 changed files with 209 additions and 319 deletions
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11
sound/soc/blackfin/bf5xx-ad1836.c
View File

@ -87,27 +87,18 @@ static int bf5xx_ad1836_driver_probe(struct platform_device *pdev)
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);

12
sound/soc/blackfin/bfin-eval-adau1373.c
View File

@ -154,16 +154,7 @@ static int bfin_eval_adau1373_probe(struct platform_device *pdev)
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 @@ static struct platform_driver bfin_eval_adau1373_driver = {
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adau1373_probe,
.remove = bfin_eval_adau1373_remove,
};
module_platform_driver(bfin_eval_adau1373_driver);

12
sound/soc/blackfin/bfin-eval-adau1701.c
View File

@ -94,16 +94,7 @@ static int bfin_eval_adau1701_probe(struct platform_device *pdev)
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 @@ static struct platform_driver bfin_eval_adau1701_driver = {
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adau1701_probe,
.remove = bfin_eval_adau1701_remove,
};
module_platform_driver(bfin_eval_adau1701_driver);

12
sound/soc/blackfin/bfin-eval-adav80x.c
View File

@ -119,16 +119,7 @@ static int bfin_eval_adav80x_probe(struct platform_device *pdev)
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 @@ static struct platform_driver bfin_eval_adav80x_driver = {
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adav80x_probe,
.remove = bfin_eval_adav80x_remove,
.id_table = bfin_eval_adav80x_ids,
};

4
sound/soc/codecs/Kconfig
View File

@ -80,7 +80,6 @@ config SND_SOC_ALL_CODECS
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_BT_SCO
config SND_SOC_DMIC
tristate
config SND_SOC_HDMI_CODEC
tristate "HDMI stub CODEC"
config SND_SOC_ES8328
tristate "Everest Semi ES8328 CODEC"

2
sound/soc/codecs/Makefile
View File

@ -73,7 +73,6 @@ snd-soc-max98925-objs := max98925.o
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-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o
obj-$(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

109
sound/soc/codecs/hdmi.c
View File

@ -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);

305
sound/soc/davinci/davinci-mcasp.c
View File

@ -80,12 +80,13 @@ struct davinci_mcasp {
/* 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 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
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 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
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 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
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 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream,
/*
* 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;
for (i = 0; i < active_slots; i++)
mask |= (1 << i);
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);
}
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 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
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 @@ static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params,
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 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params,
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 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
&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 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
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 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
*/
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 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
.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 @@ nodata:
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 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
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;

1
sound/soc/fsl/fsl-asoc-card.c
View File

@ -593,6 +593,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = {
{ .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,

1
sound/soc/fsl/fsl_sai.c
View File

@ -801,6 +801,7 @@ static const struct of_device_id fsl_sai_ids[] = {
{ .compatible = "fsl,imx6sx-sai", },
{ /* sentinel */ }
};
MODULE_DEVICE_TABLE(of, fsl_sai_ids);
static struct platform_driver fsl_sai_driver = {
.probe = fsl_sai_probe,

17
sound/soc/intel/atom/sst-mfld-platform-pcm.c
View File

@ -368,23 +368,6 @@ static void sst_media_close(struct snd_pcm_substream *substream,
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)
{

9
sound/soc/intel/boards/broadwell.c
View File

@ -266,18 +266,11 @@ static int broadwell_audio_probe(struct platform_device *pdev)
{
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",
},

33
sound/soc/intel/skylake/skl-pcm.c
View File

@ -509,17 +509,6 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
},
},
{
.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,
@ -538,28 +527,6 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
.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)