Linux kernel mirror (for testing)
git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git
kernel
os
linux
Merge tag 'sound-4.3-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"This ended up with a larger set of fixes than wished, unfortunately.
As diffstat shows, the majority of changes are for various ASoC
drivers (Realtek, Wolfson codec drivers, etc), in addition to a couple
of HD-audio regression fixes. All these are reasonably small and
nothing to scare much"
* tag 'sound-4.3-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (29 commits)
ALSA: hda - Disable power_save_node for Thinkpads
ALSA: hda/tegra - async probe for avoiding module loading deadlock
ASoC: rt5645: Prevent the pop sound in case of playback and the jack is plugging
ASoC: rt5645: Increase the delay time to remove the pop sound
ASoC: rt5645: Use the type SOC_DAPM_SINGLE_AUTODISABLE to prevent the weird sound in runtime of power up
ASoC: pxa: pxa2xx-ac97: fix dma requestor lines
MAINTAINERS: Update website and git repo for Wolfson Microelectronics
ASoC: fsl_ssi: Fix checking of dai format for AC97 mode
ASoC: wm0010: fix error path
ASoC: wm0010: fix memory leak
ASoC: wm8960: correct the max register value of mic boost pga
ASoC: wm8962: remove 64k sample rate support
ASoC: davinci-mcasp: Fix devm_kasprintf format string
ASoC: fix broken pxa SoC support
ASoC: davinci-mcasp: Set .symmetric_rates = 1 in snd_soc_dai_driver
ASoC: au1x: psc-i2s: Fix unused variable 'ret' warning
ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM
ASoC: mediatek: Increase periods_min in capture
ASoC: davinci-mcasp: Revise the FIFO threshold calculation
ASoC: wm8960: correct gain value for input PGA and add microphone PGA
...
+166
-92
+3
-6
MAINTAINERS
+3
-6
MAINTAINERS
···
11249
11249
M: Liam Girdwood <lgirdwood@gmail.com>
11250
11250
M: Mark Brown <broonie@kernel.org>
11251
11251
L: linux-kernel@vger.kernel.org
11252
-
W: http://opensource.wolfsonmicro.com/node/15
11253
11252
W: http://www.slimlogic.co.uk/?p=48
11254
11253
T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regulator.git
11255
11254
S: Supported
···
11377
11378
M: Mark Brown <broonie@kernel.org>
11378
11379
M: Liam Girdwood <lrg@slimlogic.co.uk>
11379
11380
L: linux-input@vger.kernel.org
11380
-
T: git git://opensource.wolfsonmicro.com/linux-2.6-touch
11381
-
W: http://opensource.wolfsonmicro.com/node/7
11381
+
W: https://github.com/CirrusLogic/linux-drivers/wiki
11382
11382
S: Supported
11383
11383
F: drivers/input/touchscreen/*wm97*
11384
11384
F: include/linux/wm97xx.h
11385
11385
11386
11386
WOLFSON MICROELECTRONICS DRIVERS
11387
11387
L: patches@opensource.wolfsonmicro.com
11388
-
T: git git://opensource.wolfsonmicro.com/linux-2.6-asoc
11389
-
T: git git://opensource.wolfsonmicro.com/linux-2.6-audioplus
11390
-
W: http://opensource.wolfsonmicro.com/content/linux-drivers-wolfson-devices
11388
+
T: git https://github.com/CirrusLogic/linux-drivers.git
11389
+
W: https://github.com/CirrusLogic/linux-drivers/wiki
11391
11390
S: Supported
11392
11391
F: Documentation/hwmon/wm83??
11393
11392
F: arch/arm/mach-s3c64xx/mach-crag6410*
+8
-7
sound/arm/Kconfig
+8
-7
sound/arm/Kconfig
···
9
9
Drivers that are implemented on ASoC can be found in
10
10
"ALSA for SoC audio support" section.
11
11
12
+
config SND_PXA2XX_LIB
13
+
tristate
14
+
select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
15
+
select SND_DMAENGINE_PCM
16
+
17
+
config SND_PXA2XX_LIB_AC97
18
+
bool
19
+
12
20
if SND_ARM
13
21
14
22
config SND_ARMAACI
···
28
20
config SND_PXA2XX_PCM
29
21
tristate
30
22
select SND_PCM
31
-
32
-
config SND_PXA2XX_LIB
33
-
tristate
34
-
select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
35
-
36
-
config SND_PXA2XX_LIB_AC97
37
-
bool
38
23
39
24
config SND_PXA2XX_AC97
40
25
tristate "AC97 driver for the Intel PXA2xx chip"
+25
-5
sound/pci/hda/hda_tegra.c
+25
-5
sound/pci/hda/hda_tegra.c
···
73
73
struct clk *hda2codec_2x_clk;
74
74
struct clk *hda2hdmi_clk;
75
75
void __iomem *regs;
76
+
struct work_struct probe_work;
76
77
};
77
78
78
79
#ifdef CONFIG_PM
···
295
294
static int hda_tegra_dev_free(struct snd_device *device)
296
295
{
297
296
struct azx *chip = device->device_data;
297
+
struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip);
298
298
299
+
cancel_work_sync(&hda->probe_work);
299
300
if (azx_bus(chip)->chip_init) {
300
301
azx_stop_all_streams(chip);
301
302
azx_stop_chip(chip);
···
429
426
/*
430
427
* constructor
431
428
*/
429
+
430
+
static void hda_tegra_probe_work(struct work_struct *work);
431
+
432
432
static int hda_tegra_create(struct snd_card *card,
433
433
unsigned int driver_caps,
434
434
struct hda_tegra *hda)
···
457
451
458
452
chip->single_cmd = false;
459
453
chip->snoop = true;
454
+
455
+
INIT_WORK(&hda->probe_work, hda_tegra_probe_work);
460
456
461
457
err = azx_bus_init(chip, NULL, &hda_tegra_io_ops);
462
458
if (err < 0)
···
507
499
card->private_data = chip;
508
500
509
501
dev_set_drvdata(&pdev->dev, card);
502
+
schedule_work(&hda->probe_work);
503
+
504
+
return 0;
505
+
506
+
out_free:
507
+
snd_card_free(card);
508
+
return err;
509
+
}
510
+
511
+
static void hda_tegra_probe_work(struct work_struct *work)
512
+
{
513
+
struct hda_tegra *hda = container_of(work, struct hda_tegra, probe_work);
514
+
struct azx *chip = &hda->chip;
515
+
struct platform_device *pdev = to_platform_device(hda->dev);
516
+
int err;
510
517
511
518
err = hda_tegra_first_init(chip, pdev);
512
519
if (err < 0)
···
543
520
chip->running = 1;
544
521
snd_hda_set_power_save(&chip->bus, power_save * 1000);
545
522
546
-
return 0;
547
-
548
-
out_free:
549
-
snd_card_free(card);
550
-
return err;
523
+
out_free:
524
+
return; /* no error return from async probe */
551
525
}
552
526
553
527
static int hda_tegra_remove(struct platform_device *pdev)
+19
-12
sound/pci/hda/patch_realtek.c
+19
-12
sound/pci/hda/patch_realtek.c
···
4188
4188
}
4189
4189
}
4190
4190
4191
+
/* fixup for Thinkpad docks: add dock pins, avoid HP parser fixup */
4192
+
static void alc_fixup_tpt440_dock(struct hda_codec *codec,
4193
+
const struct hda_fixup *fix, int action)
4194
+
{
4195
+
static const struct hda_pintbl pincfgs[] = {
4196
+
{ 0x16, 0x21211010 }, /* dock headphone */
4197
+
{ 0x19, 0x21a11010 }, /* dock mic */
4198
+
{ }
4199
+
};
4200
+
struct alc_spec *spec = codec->spec;
4201
+
4202
+
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
4203
+
spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
4204
+
codec->power_save_node = 0; /* avoid click noises */
4205
+
snd_hda_apply_pincfgs(codec, pincfgs);
4206
+
}
4207
+
}
4208
+
4191
4209
static void alc_shutup_dell_xps13(struct hda_codec *codec)
4192
4210
{
4193
4211
struct alc_spec *spec = codec->spec;
···
4580
4562
ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC,
4581
4563
ALC293_FIXUP_DELL1_MIC_NO_PRESENCE,
4582
4564
ALC292_FIXUP_TPT440_DOCK,
4583
-
ALC292_FIXUP_TPT440_DOCK2,
4584
4565
ALC283_FIXUP_BXBT2807_MIC,
4585
4566
ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED,
4586
4567
ALC282_FIXUP_ASPIRE_V5_PINS,
···
5046
5029
},
5047
5030
[ALC292_FIXUP_TPT440_DOCK] = {
5048
5031
.type = HDA_FIXUP_FUNC,
5049
-
.v.func = alc269_fixup_pincfg_no_hp_to_lineout,
5050
-
.chained = true,
5051
-
.chain_id = ALC292_FIXUP_TPT440_DOCK2
5052
-
},
5053
-
[ALC292_FIXUP_TPT440_DOCK2] = {
5054
-
.type = HDA_FIXUP_PINS,
5055
-
.v.pins = (const struct hda_pintbl[]) {
5056
-
{ 0x16, 0x21211010 }, /* dock headphone */
5057
-
{ 0x19, 0x21a11010 }, /* dock mic */
5058
-
{ }
5059
-
},
5032
+
.v.func = alc_fixup_tpt440_dock,
5060
5033
.chained = true,
5061
5034
.chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST
5062
5035
},
-1
sound/soc/au1x/psc-i2s.c
-1
sound/soc/au1x/psc-i2s.c
+13
-9
sound/soc/codecs/rt5645.c
+13
-9
sound/soc/codecs/rt5645.c
···
732
732
static const struct snd_kcontrol_new rt5645_dac_l_mix[] = {
733
733
SOC_DAPM_SINGLE("Stereo ADC Switch", RT5645_AD_DA_MIXER,
734
734
RT5645_M_ADCMIX_L_SFT, 1, 1),
735
-
SOC_DAPM_SINGLE("DAC1 Switch", RT5645_AD_DA_MIXER,
735
+
SOC_DAPM_SINGLE_AUTODISABLE("DAC1 Switch", RT5645_AD_DA_MIXER,
736
736
RT5645_M_DAC1_L_SFT, 1, 1),
737
737
};
738
738
739
739
static const struct snd_kcontrol_new rt5645_dac_r_mix[] = {
740
740
SOC_DAPM_SINGLE("Stereo ADC Switch", RT5645_AD_DA_MIXER,
741
741
RT5645_M_ADCMIX_R_SFT, 1, 1),
742
-
SOC_DAPM_SINGLE("DAC1 Switch", RT5645_AD_DA_MIXER,
742
+
SOC_DAPM_SINGLE_AUTODISABLE("DAC1 Switch", RT5645_AD_DA_MIXER,
743
743
RT5645_M_DAC1_R_SFT, 1, 1),
744
744
};
745
745
···
1381
1381
regmap_write(rt5645->regmap, RT5645_PR_BASE +
1382
1382
RT5645_MAMP_INT_REG2, 0xfc00);
1383
1383
snd_soc_write(codec, RT5645_DEPOP_M2, 0x1140);
1384
-
mdelay(5);
1384
+
msleep(40);
1385
1385
rt5645->hp_on = true;
1386
1386
} else {
1387
1387
/* depop parameters */
···
2829
2829
snd_soc_dapm_sync(dapm);
2830
2830
rt5645->jack_type = SND_JACK_HEADPHONE;
2831
2831
}
2832
-
2833
-
snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200);
2834
-
snd_soc_write(codec, RT5645_DEPOP_M1, 0x001d);
2835
-
snd_soc_write(codec, RT5645_DEPOP_M1, 0x0001);
2836
2832
} else { /* jack out */
2837
2833
rt5645->jack_type = 0;
2838
2834
2835
+
regmap_update_bits(rt5645->regmap, RT5645_HP_VOL,
2836
+
RT5645_L_MUTE | RT5645_R_MUTE,
2837
+
RT5645_L_MUTE | RT5645_R_MUTE);
2839
2838
regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2,
2840
2839
RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD);
2841
2840
regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1,
···
2879
2880
rt5645->en_button_func = true;
2880
2881
regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1,
2881
2882
RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ);
2882
-
regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1,
2883
-
RT5645_HP_CB_MASK, RT5645_HP_CB_PU);
2884
2883
regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1,
2885
2884
RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL);
2886
2885
}
···
3200
3203
.callback = strago_quirk_cb,
3201
3204
.matches = {
3202
3205
DMI_MATCH(DMI_PRODUCT_NAME, "Celes"),
3206
+
},
3207
+
},
3208
+
{
3209
+
.ident = "Google Ultima",
3210
+
.callback = strago_quirk_cb,
3211
+
.matches = {
3212
+
DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"),
3203
3213
},
3204
3214
},
3205
3215
{ }
+9
-14
sound/soc/codecs/wm0010.c
+9
-14
sound/soc/codecs/wm0010.c
···
577
577
struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec);
578
578
unsigned long flags;
579
579
int ret;
580
-
const struct firmware *fw;
581
580
struct spi_message m;
582
581
struct spi_transfer t;
583
582
struct dfw_pllrec pll_rec;
···
622
623
wm0010->state = WM0010_OUT_OF_RESET;
623
624
spin_unlock_irqrestore(&wm0010->irq_lock, flags);
624
625
625
-
/* First the bootloader */
626
-
ret = request_firmware(&fw, "wm0010_stage2.bin", codec->dev);
627
-
if (ret != 0) {
628
-
dev_err(codec->dev, "Failed to request stage2 loader: %d\n",
629
-
ret);
630
-
goto abort;
631
-
}
632
-
633
626
if (!wait_for_completion_timeout(&wm0010->boot_completion,
634
627
msecs_to_jiffies(20)))
635
628
dev_err(codec->dev, "Failed to get interrupt from DSP\n");
···
664
673
665
674
img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA);
666
675
if (!img_swap)
667
-
goto abort;
676
+
goto abort_out;
668
677
669
678
/* We need to re-order for 0010 */
670
679
byte_swap_64((u64 *)&pll_rec, img_swap, len);
···
679
688
spi_message_add_tail(&t, &m);
680
689
681
690
ret = spi_sync(spi, &m);
682
-
if (ret != 0) {
691
+
if (ret) {
683
692
dev_err(codec->dev, "First PLL write failed: %d\n", ret);
684
-
goto abort;
693
+
goto abort_swap;
685
694
}
686
695
687
696
/* Use a second send of the message to get the return status */
688
697
ret = spi_sync(spi, &m);
689
-
if (ret != 0) {
698
+
if (ret) {
690
699
dev_err(codec->dev, "Second PLL write failed: %d\n", ret);
691
-
goto abort;
700
+
goto abort_swap;
692
701
}
693
702
694
703
p = (u32 *)out;
···
721
730
722
731
return 0;
723
732
733
+
abort_swap:
734
+
kfree(img_swap);
735
+
abort_out:
736
+
kfree(out);
724
737
abort:
725
738
/* Put the chip back into reset */
726
739
wm0010_halt(codec);
+18
-8
sound/soc/codecs/wm8960.c
+18
-8
sound/soc/codecs/wm8960.c
···
211
211
return wm8960_set_deemph(codec);
212
212
}
213
213
214
-
static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0);
215
-
static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1);
214
+
static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
215
+
static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1725, 75, 0);
216
+
static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
216
217
static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
217
218
static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
218
-
static const DECLARE_TLV_DB_SCALE(boost_tlv, -1200, 300, 1);
219
+
static const DECLARE_TLV_DB_SCALE(lineinboost_tlv, -1500, 300, 1);
220
+
static const unsigned int micboost_tlv[] = {
221
+
TLV_DB_RANGE_HEAD(2),
222
+
0, 1, TLV_DB_SCALE_ITEM(0, 1300, 0),
223
+
2, 3, TLV_DB_SCALE_ITEM(2000, 900, 0),
224
+
};
219
225
220
226
static const struct snd_kcontrol_new wm8960_snd_controls[] = {
221
227
SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
222
-
0, 63, 0, adc_tlv),
228
+
0, 63, 0, inpga_tlv),
223
229
SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
224
230
6, 1, 0),
225
231
SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
226
232
7, 1, 0),
227
233
228
234
SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
229
-
WM8960_INBMIX1, 4, 7, 0, boost_tlv),
235
+
WM8960_INBMIX1, 4, 7, 0, lineinboost_tlv),
230
236
SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume",
231
-
WM8960_INBMIX1, 1, 7, 0, boost_tlv),
237
+
WM8960_INBMIX1, 1, 7, 0, lineinboost_tlv),
232
238
SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume",
233
-
WM8960_INBMIX2, 4, 7, 0, boost_tlv),
239
+
WM8960_INBMIX2, 4, 7, 0, lineinboost_tlv),
234
240
SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume",
235
-
WM8960_INBMIX2, 1, 7, 0, boost_tlv),
241
+
WM8960_INBMIX2, 1, 7, 0, lineinboost_tlv),
242
+
SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT1 Volume",
243
+
WM8960_RINPATH, 4, 3, 0, micboost_tlv),
244
+
SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT1 Volume",
245
+
WM8960_LINPATH, 4, 3, 0, micboost_tlv),
236
246
237
247
SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC,
238
248
0, 255, 0, dac_tlv),
+2
-1
sound/soc/codecs/wm8962.c
+2
-1
sound/soc/codecs/wm8962.c
···
2944
2944
WM8962_DAC_MUTE, val);
2945
2945
}
2946
2946
2947
-
#define WM8962_RATES SNDRV_PCM_RATE_8000_96000
2947
+
#define WM8962_RATES (SNDRV_PCM_RATE_8000_48000 |\
2948
+
SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
2948
2949
2949
2950
#define WM8962_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
2950
2951
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+7
-7
sound/soc/davinci/davinci-mcasp.c
+7
-7
sound/soc/davinci/davinci-mcasp.c
···
663
663
u8 rx_ser = 0;
664
664
u8 slots = mcasp->tdm_slots;
665
665
u8 max_active_serializers = (channels + slots - 1) / slots;
666
-
int active_serializers, numevt, n;
666
+
int active_serializers, numevt;
667
667
u32 reg;
668
668
/* Default configuration */
669
669
if (mcasp->version < MCASP_VERSION_3)
···
745
745
* The number of words for numevt need to be in steps of active
746
746
* serializers.
747
747
*/
748
-
n = numevt % active_serializers;
749
-
if (n)
750
-
numevt += (active_serializers - n);
748
+
numevt = (numevt / active_serializers) * active_serializers;
749
+
751
750
while (period_words % numevt && numevt > 0)
752
751
numevt -= active_serializers;
753
752
if (numevt <= 0)
···
1298
1299
.ops = &davinci_mcasp_dai_ops,
1299
1300
1300
1301
.symmetric_samplebits = 1,
1302
+
.symmetric_rates = 1,
1301
1303
},
1302
1304
{
1303
1305
.name = "davinci-mcasp.1",
···
1685
1685
1686
1686
irq = platform_get_irq_byname(pdev, "common");
1687
1687
if (irq >= 0) {
1688
-
irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common\n",
1688
+
irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common",
1689
1689
dev_name(&pdev->dev));
1690
1690
ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
1691
1691
davinci_mcasp_common_irq_handler,
···
1702
1702
1703
1703
irq = platform_get_irq_byname(pdev, "rx");
1704
1704
if (irq >= 0) {
1705
-
irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n",
1705
+
irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx",
1706
1706
dev_name(&pdev->dev));
1707
1707
ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
1708
1708
davinci_mcasp_rx_irq_handler,
···
1717
1717
1718
1718
irq = platform_get_irq_byname(pdev, "tx");
1719
1719
if (irq >= 0) {
1720
-
irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx\n",
1720
+
irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx",
1721
1721
dev_name(&pdev->dev));
1722
1722
ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
1723
1723
davinci_mcasp_tx_irq_handler,
+2
-1
sound/soc/fsl/fsl-asoc-card.c
+2
-1
sound/soc/fsl/fsl-asoc-card.c
+3
-2
sound/soc/fsl/fsl_ssi.c
+3
-2
sound/soc/fsl/fsl_ssi.c
···
249
249
250
250
static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private)
251
251
{
252
-
return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97);
252
+
return (ssi_private->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) ==
253
+
SND_SOC_DAIFMT_AC97;
253
254
}
254
255
255
256
static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private)
···
948
947
CCSR_SSI_SCR_TCH_EN);
949
948
}
950
949
951
-
if (fmt & SND_SOC_DAIFMT_AC97)
950
+
if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97)
952
951
fsl_ssi_setup_ac97(ssi_private);
953
952
954
953
return 0;
+12
-8
sound/soc/intel/haswell/sst-haswell-ipc.c
+12
-8
sound/soc/intel/haswell/sst-haswell-ipc.c
···
302
302
struct sst_hsw_ipc_dx_reply dx;
303
303
void *dx_context;
304
304
dma_addr_t dx_context_paddr;
305
+
enum sst_hsw_device_id dx_dev;
306
+
enum sst_hsw_device_mclk dx_mclk;
307
+
enum sst_hsw_device_mode dx_mode;
308
+
u32 dx_clock_divider;
305
309
306
310
/* boot */
307
311
wait_queue_head_t boot_wait;
···
1404
1400
1405
1401
trace_ipc_request("set device config", dev);
1406
1402
1407
-
config.ssp_interface = dev;
1408
-
config.clock_frequency = mclk;
1409
-
config.mode = mode;
1410
-
config.clock_divider = clock_divider;
1403
+
hsw->dx_dev = config.ssp_interface = dev;
1404
+
hsw->dx_mclk = config.clock_frequency = mclk;
1405
+
hsw->dx_mode = config.mode = mode;
1406
+
hsw->dx_clock_divider = config.clock_divider = clock_divider;
1411
1407
if (mode == SST_HSW_DEVICE_TDM_CLOCK_MASTER)
1412
1408
config.channels = 4;
1413
1409
else
···
1708
1704
return -EIO;
1709
1705
}
1710
1706
1711
-
/* Set ADSP SSP port settings */
1712
-
ret = sst_hsw_device_set_config(hsw, SST_HSW_DEVICE_SSP_0,
1713
-
SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
1714
-
SST_HSW_DEVICE_CLOCK_MASTER, 9);
1707
+
/* Set ADSP SSP port settings - sadly the FW does not store SSP port
1708
+
settings as part of the PM context. */
1709
+
ret = sst_hsw_device_set_config(hsw, hsw->dx_dev, hsw->dx_mclk,
1710
+
hsw->dx_mode, hsw->dx_clock_divider);
1715
1711
if (ret < 0)
1716
1712
dev_err(dev, "error: SSP re-initialization failed\n");
1717
1713
+17
sound/soc/mediatek/mtk-afe-pcm.c
+17
sound/soc/mediatek/mtk-afe-pcm.c
···
549
549
memif->substream = substream;
550
550
551
551
snd_soc_set_runtime_hwparams(substream, &mtk_afe_hardware);
552
+
553
+
/*
554
+
* Capture cannot use ping-pong buffer since hw_ptr at IRQ may be
555
+
* smaller than period_size due to AFE's internal buffer.
556
+
* This easily leads to overrun when avail_min is period_size.
557
+
* One more period can hold the possible unread buffer.
558
+
*/
559
+
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
560
+
ret = snd_pcm_hw_constraint_minmax(runtime,
561
+
SNDRV_PCM_HW_PARAM_PERIODS,
562
+
3,
563
+
mtk_afe_hardware.periods_max);
564
+
if (ret < 0) {
565
+
dev_err(afe->dev, "hw_constraint_minmax failed\n");
566
+
return ret;
567
+
}
568
+
}
552
569
ret = snd_pcm_hw_constraint_integer(runtime,
553
570
SNDRV_PCM_HW_PARAM_PERIODS);
554
571
if (ret < 0)
-2
sound/soc/pxa/Kconfig
-2
sound/soc/pxa/Kconfig
···
1
1
config SND_PXA2XX_SOC
2
2
tristate "SoC Audio for the Intel PXA2xx chip"
3
3
depends on ARCH_PXA
4
-
select SND_ARM
5
4
select SND_PXA2XX_LIB
6
5
help
7
6
Say Y or M if you want to add support for codecs attached to
···
24
25
config SND_PXA2XX_SOC_AC97
25
26
tristate
26
27
select AC97_BUS
27
-
select SND_ARM
28
28
select SND_PXA2XX_LIB_AC97
29
29
select SND_SOC_AC97_BUS
30
30
+2
-2
sound/soc/pxa/pxa2xx-ac97.c
+2
-2
sound/soc/pxa/pxa2xx-ac97.c
···
49
49
.reset = pxa2xx_ac97_cold_reset,
50
50
};
51
51
52
-
static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 12;
52
+
static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 11;
53
53
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
54
54
.addr = __PREG(PCDR),
55
55
.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
···
57
57
.filter_data = &pxa2xx_ac97_pcm_stereo_in_req,
58
58
};
59
59
60
-
static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 11;
60
+
static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 12;
61
61
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
62
62
.addr = __PREG(PCDR),
63
63
.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+1
-1
sound/soc/soc-dapm.c
+1
-1
sound/soc/soc-dapm.c
+9
sound/soc/soc-utils.c
+9
sound/soc/soc-utils.c
···
101
101
SNDRV_PCM_FMTBIT_S32_LE | \
102
102
SNDRV_PCM_FMTBIT_U32_LE | \
103
103
SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
104
+
/*
105
+
* The dummy CODEC is only meant to be used in situations where there is no
106
+
* actual hardware.
107
+
*
108
+
* If there is actual hardware even if it does not have a control bus
109
+
* the hardware will still have constraints like supported samplerates, etc.
110
+
* which should be modelled. And the data flow graph also should be modelled
111
+
* using DAPM.
112
+
*/
104
113
static struct snd_soc_dai_driver dummy_dai = {
105
114
.name = "snd-soc-dummy-dai",
106
115
.playback = {
+1
-1
sound/soc/spear/Kconfig
+1
-1
sound/soc/spear/Kconfig
+10
-4
sound/soc/sti/uniperif_player.c
+10
-4
sound/soc/sti/uniperif_player.c
···
989
989
if (!info)
990
990
return -ENOMEM;
991
991
992
-
of_property_read_u32(pnode, "version", &player->ver);
993
-
if (player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
992
+
if (of_property_read_u32(pnode, "version", &player->ver) ||
993
+
player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
994
994
dev_err(dev, "Unknown uniperipheral version ");
995
995
return -EINVAL;
996
996
}
···
998
998
if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
999
999
info->underflow_enabled = 1;
1000
1000
1001
-
of_property_read_u32(pnode, "uniperiph-id", &info->id);
1001
+
if (of_property_read_u32(pnode, "uniperiph-id", &info->id)) {
1002
+
dev_err(dev, "uniperipheral id not defined");
1003
+
return -EINVAL;
1004
+
}
1002
1005
1003
1006
/* Read the device mode property */
1004
-
of_property_read_string(pnode, "mode", &mode);
1007
+
if (of_property_read_string(pnode, "mode", &mode)) {
1008
+
dev_err(dev, "uniperipheral mode not defined");
1009
+
return -EINVAL;
1010
+
}
1005
1011
1006
1012
if (strcasecmp(mode, "hdmi") == 0)
1007
1013
info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_HDMI;
+5
-1
sound/soc/sti/uniperif_reader.c
+5
-1
sound/soc/sti/uniperif_reader.c
···
316
316
if (!info)
317
317
return -ENOMEM;
318
318
319
-
of_property_read_u32(node, "version", &reader->ver);
319
+
if (of_property_read_u32(node, "version", &reader->ver) ||
320
+
reader->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
321
+
dev_err(&pdev->dev, "Unknown uniperipheral version ");
322
+
return -EINVAL;
323
+
}
320
324
321
325
/* Save the info structure */
322
326
reader->info = info;