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 ...

Linus Torvalds committed 11 years ago

Commit ddff42e2 parents 966966a 7f57d8021 files · +166 −92Browse files

Unified · Split

21 files

MAINTAINERS +3 −6
sound/arm/Kconfig +8 −7
sound/pci/hda/hda_tegra.c +25 −5
sound/pci/hda/patch_realtek.c +19 −12
sound/soc/au1x/psc-i2s.c +0 −1
sound/soc/codecs/rt5645.c +13 −9
sound/soc/codecs/wm0010.c +9 −14
sound/soc/codecs/wm8960.c +18 −8
sound/soc/codecs/wm8962.c +2 −1
sound/soc/davinci/davinci-mcasp.c +7 −7
sound/soc/fsl/fsl-asoc-card.c +2 −1
sound/soc/fsl/fsl_ssi.c +3 −2
sound/soc/intel/haswell/sst-haswell-ipc.c +12 −8
sound/soc/mediatek/mtk-afe-pcm.c +17 −0
sound/soc/pxa/Kconfig +0 −2
sound/soc/pxa/pxa2xx-ac97.c +2 −2
sound/soc/soc-dapm.c +1 −1
sound/soc/soc-utils.c +9 −0
sound/soc/spear/Kconfig +1 −1
sound/soc/sti/uniperif_player.c +10 −4
sound/soc/sti/uniperif_reader.c +5 −1

MAINTAINERS+3−6

@@ -11249,7 +11249,6 @@ VOLTAGE AND CURRENT REGULATOR FRAMEWORK · 11248 unmodified lines
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
@@ -11378,17 +11377,15 @@ WM97XX TOUCHSCREEN DRIVERS · 122 unmodified lines
11378	11377	M: Mark Brown <broonie@kernel.org>
11379	11378	M: Liam Girdwood <lrg@slimlogic.co.uk>
11380	11379	L: linux-input@vger.kernel.org
11381		−T: git git://opensource.wolfsonmicro.com/linux-2.6-touch
11382		−W: http://opensource.wolfsonmicro.com/node/7
	11380	+W: https://github.com/CirrusLogic/linux-drivers/wiki
11383	11381	S: Supported
11384	11382	F: drivers/input/touchscreen/wm97
11385	11383	F: include/linux/wm97xx.h
11386	11384
11387	11385	WOLFSON MICROELECTRONICS DRIVERS
11388	11386	L: patches@opensource.wolfsonmicro.com
11389		−T: git git://opensource.wolfsonmicro.com/linux-2.6-asoc
11390		−T: git git://opensource.wolfsonmicro.com/linux-2.6-audioplus
11391		−W: http://opensource.wolfsonmicro.com/content/linux-drivers-wolfson-devices
	11387	+T: git https://github.com/CirrusLogic/linux-drivers.git
	11388	+W: https://github.com/CirrusLogic/linux-drivers/wiki
11392	11389	S: Supported
11393	11390	F: Documentation/hwmon/wm83??
11394	11391	F: arch/arm/mach-s3c64xx/mach-crag6410*

sound/arm/Kconfig+8−7

@@ -9,6 +9,14 @@ menuconfig SND_ARM · 8 unmodified lines
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
@@ -21,13 +29,6 @@ config SND_PXA2XX_PCM · 6 unmodified lines
21	29	tristate
22	30	select SND_PCM
23	31
24		−config SND_PXA2XX_LIB
25		− tristate
26		− select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
27		−
28		−config SND_PXA2XX_LIB_AC97
29		− bool
30		−
31	32	config SND_PXA2XX_AC97
32	33	tristate "AC97 driver for the Intel PXA2xx chip"
33	34	depends on ARCH_PXA

sound/pci/hda/hda_tegra.c+25−5

@@ -73,6 +73,7 @@ struct hda_tegra { · 72 unmodified lines
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
@@ -294,7 +295,9 @@ static int hda_tegra_dev_disconnect(struct snd_device *device) · 215 unmodified lines
294	295	static int hda_tegra_dev_free(struct snd_device *device)
295	296	{
296	297	struct azx *chip = device->device_data;
	298	+ struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip);
297	299
	300	+ cancel_work_sync(&hda->probe_work);
298	301	if (azx_bus(chip)->chip_init) {
299	302	azx_stop_all_streams(chip);
300	303	azx_stop_chip(chip);
@@ -426,6 +429,9 @@ static int hda_tegra_first_init(struct azx chip, struct platform_device pdev) · 125 unmodified lines
426	429	/*
427	430	* constructor
428	431	*/
	432	+
	433	+static void hda_tegra_probe_work(struct work_struct *work);
	434	+
429	435	static int hda_tegra_create(struct snd_card *card,
430	436	unsigned int driver_caps,
431	437	struct hda_tegra *hda)
@@ -452,6 +458,8 @@ static int hda_tegra_create(struct snd_card *card, · 20 unmodified lines
452	458	chip->single_cmd = false;
453	459	chip->snoop = true;
454	460
	461	+ INIT_WORK(&hda->probe_work, hda_tegra_probe_work);
	462	+
455	463	err = azx_bus_init(chip, NULL, &hda_tegra_io_ops);
456	464	if (err < 0)
457	465	return err;
@@ -499,6 +507,21 @@ static int hda_tegra_probe(struct platform_device *pdev) · 41 unmodified lines
499	507	card->private_data = chip;
500	508
501	509	dev_set_drvdata(&pdev->dev, card);
	510	+ schedule_work(&hda->probe_work);
	511	+
	512	+ return 0;
	513	+
	514	+out_free:
	515	+ snd_card_free(card);
	516	+ return err;
	517	+}
	518	+
	519	+static void hda_tegra_probe_work(struct work_struct *work)
	520	+{
	521	+ struct hda_tegra *hda = container_of(work, struct hda_tegra, probe_work);
	522	+ struct azx *chip = &hda->chip;
	523	+ struct platform_device *pdev = to_platform_device(hda->dev);
	524	+ int err;
502	525
503	526	err = hda_tegra_first_init(chip, pdev);
504	527	if (err < 0)
@@ -520,11 +543,8 @@ static int hda_tegra_probe(struct platform_device *pdev) · 15 unmodified lines
520	543	chip->running = 1;
521	544	snd_hda_set_power_save(&chip->bus, power_save * 1000);
522	545
523		− return 0;
524		−
525		−out_free:
526		− snd_card_free(card);
527		− return err;
	546	+ out_free:
	547	+ return; /* no error return from async probe */
528	548	}
529	549
530	550	static int hda_tegra_remove(struct platform_device *pdev)

sound/pci/hda/patch_realtek.c+19−12

@@ -4188,6 +4188,24 @@ static void alc_fixup_disable_aamix(struct hda_codec *codec, · 4187 unmodified lines
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;
@@ -4562,7 +4580,6 @@ enum { · 368 unmodified lines
4562	4580	ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC,
4563	4581	ALC293_FIXUP_DELL1_MIC_NO_PRESENCE,
4564	4582	ALC292_FIXUP_TPT440_DOCK,
4565		− ALC292_FIXUP_TPT440_DOCK2,
4566	4583	ALC283_FIXUP_BXBT2807_MIC,
4567	4584	ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED,
4568	4585	ALC282_FIXUP_ASPIRE_V5_PINS,
@@ -5029,17 +5046,7 @@ static const struct hda_fixup alc269_fixups[] = { · 460 unmodified lines
5029	5046	},
5030	5047	[ALC292_FIXUP_TPT440_DOCK] = {
5031	5048	.type = HDA_FIXUP_FUNC,
5032		− .v.func = alc269_fixup_pincfg_no_hp_to_lineout,
5033		− .chained = true,
5034		− .chain_id = ALC292_FIXUP_TPT440_DOCK2
5035		− },
5036		− [ALC292_FIXUP_TPT440_DOCK2] = {
5037		− .type = HDA_FIXUP_PINS,
5038		− .v.pins = (const struct hda_pintbl[]) {
5039		− { 0x16, 0x21211010 }, /* dock headphone */
5040		− { 0x19, 0x21a11010 }, /* dock mic */
5041		− { }
5042		− },
	5049	+ .v.func = alc_fixup_tpt440_dock,
5043	5050	.chained = true,
5044	5051	.chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST
5045	5052	},

sound/soc/au1x/psc-i2s.c+0−1

@@ -296,7 +296,6 @@ static int au1xpsc_i2s_drvprobe(struct platform_device *pdev) · 295 unmodified lines
296	296	{
297	297	struct resource iores, dmares;
298	298	unsigned long sel;
299		− int ret;
300	299	struct au1xpsc_audio_data *wd;
301	300
302	301	wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data),

sound/soc/codecs/rt5645.c+13−9

@@ -732,14 +732,14 @@ static const struct snd_kcontrol_new rt5645_mono_adc_r_mix[] = { · 731 unmodified lines
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
@@ -1381,7 +1381,7 @@ static void hp_amp_power(struct snd_soc_codec *codec, int on) · 635 unmodified lines
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 */
@@ -2829,13 +2829,12 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) · 1441 unmodified lines
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,
@@ -2880,8 +2879,6 @@ int rt5645_set_jack_detect(struct snd_soc_codec *codec, · 38 unmodified lines
2880	2879	rt5645->en_button_func = true;
2881	2880	regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1,
2882	2881	RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ);
2883		− regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1,
2884		− RT5645_HP_CB_MASK, RT5645_HP_CB_PU);
2885	2882	regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1,
2886	2883	RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL);
2887	2884	}
@@ -3205,6 +3202,13 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { · 317 unmodified lines
3205	3202	DMI_MATCH(DMI_PRODUCT_NAME, "Celes"),
3206	3203	},
3207	3204	},
	3205	+ {
	3206	+ .ident = "Google Ultima",
	3207	+ .callback = strago_quirk_cb,
	3208	+ .matches = {
	3209	+ DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"),
	3210	+ },
	3211	+ },
3208	3212	{ }
3209	3213	};
3210	3214

sound/soc/codecs/wm0010.c+9−14

@@ -577,7 +577,6 @@ static int wm0010_boot(struct snd_soc_codec *codec) · 576 unmodified lines
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;
@@ -623,14 +622,6 @@ static int wm0010_boot(struct snd_soc_codec *codec) · 39 unmodified lines
623	622	wm0010->state = WM0010_OUT_OF_RESET;
624	623	spin_unlock_irqrestore(&wm0010->irq_lock, flags);
625	624
626		− /* First the bootloader */
627		− ret = request_firmware(&fw, "wm0010_stage2.bin", codec->dev);
628		− if (ret != 0) {
629		− dev_err(codec->dev, "Failed to request stage2 loader: %d\n",
630		− ret);
631		− goto abort;
632		− }
633		−
634	625	if (!wait_for_completion_timeout(&wm0010->boot_completion,
635	626	msecs_to_jiffies(20)))
636	627	dev_err(codec->dev, "Failed to get interrupt from DSP\n");
@@ -673,7 +664,7 @@ static int wm0010_boot(struct snd_soc_codec *codec) · 36 unmodified lines
673	664
674	665	img_swap = kzalloc(len, GFP_KERNEL \| GFP_DMA);
675	666	if (!img_swap)
676		− goto abort;
	667	+ goto abort_out;
677	668
678	669	/* We need to re-order for 0010 */
679	670	byte_swap_64((u64 *)&pll_rec, img_swap, len);
@@ -688,16 +679,16 @@ static int wm0010_boot(struct snd_soc_codec *codec) · 8 unmodified lines
688	679	spi_message_add_tail(&t, &m);
689	680
690	681	ret = spi_sync(spi, &m);
691		− if (ret != 0) {
	682	+ if (ret) {
692	683	dev_err(codec->dev, "First PLL write failed: %d\n", ret);
693		− goto abort;
	684	+ goto abort_swap;
694	685	}
695	686
696	687	/* Use a second send of the message to get the return status */
697	688	ret = spi_sync(spi, &m);
698		− if (ret != 0) {
	689	+ if (ret) {
699	690	dev_err(codec->dev, "Second PLL write failed: %d\n", ret);
700		− goto abort;
	691	+ goto abort_swap;
701	692	}
702	693
703	694	p = (u32 *)out;
@@ -730,6 +721,10 @@ static int wm0010_boot(struct snd_soc_codec *codec) · 26 unmodified lines
730	721
731	722	return 0;
732	723
	724	+abort_swap:
	725	+ kfree(img_swap);
	726	+abort_out:
	727	+ kfree(out);
733	728	abort:
734	729	/* Put the chip back into reset */
735	730	wm0010_halt(codec);

sound/soc/codecs/wm8960.c+18−8

@@ -211,28 +211,38 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol, · 210 unmodified lines
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),

sound/soc/codecs/wm8962.c+2−1

@@ -2944,7 +2944,8 @@ static int wm8962_mute(struct snd_soc_dai *dai, int mute) · 2943 unmodified lines
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)

sound/soc/davinci/davinci-mcasp.c+7−7

@@ -663,7 +663,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, · 662 unmodified lines
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)
@@ -745,9 +745,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, · 75 unmodified lines
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)
@@ -1299,6 +1298,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { · 545 unmodified lines
1299	1298	.ops = &davinci_mcasp_dai_ops,
1300	1299
1301	1300	.symmetric_samplebits = 1,
	1301	+ .symmetric_rates = 1,
1302	1302	},
1303	1303	{
1304	1304	.name = "davinci-mcasp.1",
@@ -1685,7 +1685,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) · 380 unmodified lines
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,
@@ -1702,7 +1702,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) · 10 unmodified lines
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,
@@ -1717,7 +1717,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) · 8 unmodified lines
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,

sound/soc/fsl/fsl-asoc-card.c+2−1

@@ -488,7 +488,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) · 487 unmodified lines
488	488	priv->dai_fmt \|= SND_SOC_DAIFMT_CBM_CFM;
489	489	} else {
490	490	dev_err(&pdev->dev, "unknown Device Tree compatible\n");
491		− return -EINVAL;
	491	+ ret = -EINVAL;
	492	+ goto asrc_fail;
492	493	}
493	494
494	495	/* Common settings for corresponding Freescale CPU DAI driver */

sound/soc/fsl/fsl_ssi.c+3−2

@@ -249,7 +249,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids); · 248 unmodified lines
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)
@@ -947,7 +948,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, · 691 unmodified lines
947	948	CCSR_SSI_SCR_TCH_EN);
948	949	}
949	950
950		− if (fmt & SND_SOC_DAIFMT_AC97)
	951	+ if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97)
951	952	fsl_ssi_setup_ac97(ssi_private);
952	953
953	954	return 0;

sound/soc/intel/haswell/sst-haswell-ipc.c+12−8

@@ -302,6 +302,10 @@ struct sst_hsw { · 301 unmodified lines
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;
@@ -1400,10 +1404,10 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, · 1092 unmodified lines
1400	1404
1401	1405	trace_ipc_request("set device config", dev);
1402	1406
1403		− config.ssp_interface = dev;
1404		− config.clock_frequency = mclk;
1405		− config.mode = mode;
1406		− config.clock_divider = clock_divider;
	1407	+ hsw->dx_dev = config.ssp_interface = dev;
	1408	+ hsw->dx_mclk = config.clock_frequency = mclk;
	1409	+ hsw->dx_mode = config.mode = mode;
	1410	+ hsw->dx_clock_divider = config.clock_divider = clock_divider;
1407	1411	if (mode == SST_HSW_DEVICE_TDM_CLOCK_MASTER)
1408	1412	config.channels = 4;
1409	1413	else
@@ -1704,10 +1708,10 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw) · 294 unmodified lines
1704	1708	return -EIO;
1705	1709	}
1706	1710
1707		− /* Set ADSP SSP port settings */
1708		− ret = sst_hsw_device_set_config(hsw, SST_HSW_DEVICE_SSP_0,
1709		− SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
1710		− SST_HSW_DEVICE_CLOCK_MASTER, 9);
	1711	+ /* Set ADSP SSP port settings - sadly the FW does not store SSP port
	1712	+ settings as part of the PM context. */
	1713	+ ret = sst_hsw_device_set_config(hsw, hsw->dx_dev, hsw->dx_mclk,
	1714	+ hsw->dx_mode, hsw->dx_clock_divider);
1711	1715	if (ret < 0)
1712	1716	dev_err(dev, "error: SSP re-initialization failed\n");
1713	1717

sound/soc/mediatek/mtk-afe-pcm.c+17−0

@@ -549,6 +549,23 @@ static int mtk_afe_dais_startup(struct snd_pcm_substream *substream, · 548 unmodified lines
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)

sound/soc/pxa/Kconfig+0−2

@@ -1,7 +1,6 @@
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
@@ -25,7 +24,6 @@ config SND_PXA2XX_AC97 · 17 unmodified lines
25	24	config SND_PXA2XX_SOC_AC97
26	25	tristate
27	26	select AC97_BUS
28		− select SND_ARM
29	27	select SND_PXA2XX_LIB_AC97
30	28	select SND_SOC_AC97_BUS
31	29

sound/soc/pxa/pxa2xx-ac97.c+2−2

@@ -49,7 +49,7 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { · 48 unmodified lines
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,
@@ -57,7 +57,7 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { · 1 unmodified line
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,

sound/soc/soc-dapm.c+1−1

@@ -3501,7 +3501,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, · 3500 unmodified lines
3501	3501
3502	3502	default:
3503	3503	WARN(1, "Unknown event %d\n", event);
3504		− return -EINVAL;
	3504	+ ret = -EINVAL;
3505	3505	}
3506	3506
3507	3507	out:

sound/soc/soc-utils.c+9−0

@@ -101,6 +101,15 @@ static struct snd_soc_codec_driver dummy_codec; · 100 unmodified lines
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 = {

sound/soc/spear/Kconfig+1−1

@@ -1,6 +1,6 @@
1	1	config SND_SPEAR_SOC
2	2	tristate
3		− select SND_DMAENGINE_PCM
	3	+ select SND_SOC_GENERIC_DMAENGINE_PCM
4	4
5	5	config SND_SPEAR_SPDIF_OUT
6	6	tristate

sound/soc/sti/uniperif_player.c+10−4

@@ -989,8 +989,8 @@ static int uni_player_parse_dt(struct platform_device *pdev, · 988 unmodified lines
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	}
@@ -998,10 +998,16 @@ static int uni_player_parse_dt(struct platform_device *pdev, · 1 unmodified line
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;

sound/soc/sti/uniperif_reader.c+5−1

@@ -316,7 +316,11 @@ static int uni_reader_parse_dt(struct platform_device *pdev, · 315 unmodified lines
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;