Merge tag 'sound-4.16-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A series of small fixes in ASoC, HD-audio and core stuff:

   - a UAF fix in ALSA PCM core

   - yet more hardening for ALSA sequencer

   - a regression fix for the previous HD-audio power_save option change

   - various ASoC codec fixes (sgtl5000, rt5651, hdmi-codec, wm_adsp)

   - minor ASoC platform fixes (AMD ACP, sun4i)"

* tag 'sound-4.16-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - Revert power_save option default value
  ALSA: pcm: Fix UAF in snd_pcm_oss_get_formats()
  ALSA: seq: Clear client entry before deleting else at closing
  ALSA: seq: Fix possible UAF in snd_seq_check_queue()
  ASoC: amd: 16bit resolution support for i2s sp instance
  ASoC: wm_adsp: For TLV controls only register TLV get/set
  ASoC: sun4i-i2s: Fix RX slot number of SUN8I
  ASoC: hdmi-codec: Fix module unloading caused kernel crash
  ASoC: rt5651: Fix regcache sync errors on resume
  ASoC: sgtl5000: Fix suspend/resume
  MAINTAINERS: Add myself as sgtl5000 maintainer
  ASoC: samsung: Add the DT binding files entry to MAINTAINERS
  sgtl5000: change digital_mute policy

MAINTAINERS

@@ -9925,6 +9925,13 @@ F:	Documentation/ABI/stable/sysfs-bus-nvmem
 F:	include/linux/nvmem-consumer.h
 F:	include/linux/nvmem-provider.h
 
+NXP SGTL5000 DRIVER
+M:	Fabio Estevam <fabio.estevam@nxp.com>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Maintained
+F:	Documentation/devicetree/bindings/sound/sgtl5000.txt
+F:	sound/soc/codecs/sgtl5000*
+
 NXP TDA998X DRM DRIVER
 M:	Russell King <linux@armlinux.org.uk>
 S:	Supported
@@ -12107,6 +12114,7 @@ M:	Sylwester Nawrocki <s.nawrocki@samsung.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:	Supported
 F:	sound/soc/samsung/
+F:	Documentation/devicetree/bindings/sound/samsung*
 
 SAMSUNG EXYNOS PSEUDO RANDOM NUMBER GENERATOR (RNG) DRIVER
 M:	Krzysztof Kozlowski <krzk@kernel.org>

sound/core/oss/pcm_oss.c

@@ -1762,10 +1762,9 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file)
 		return -ENOMEM;
 	_snd_pcm_hw_params_any(params);
 	err = snd_pcm_hw_refine(substream, params);
-	format_mask = hw_param_mask_c(params, SNDRV_PCM_HW_PARAM_FORMAT);
-	kfree(params);
 	if (err < 0)
-		return err;
+		goto error;
+	format_mask = hw_param_mask_c(params, SNDRV_PCM_HW_PARAM_FORMAT);
 	for (fmt = 0; fmt < 32; ++fmt) {
 		if (snd_mask_test(format_mask, fmt)) {
 			int f = snd_pcm_oss_format_to(fmt);
@@ -1773,7 +1772,10 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file)
 				formats |= f;
 		}
 	}
-	return formats;
+
+ error:
+	kfree(params);
+	return err < 0 ? err : formats;
 }
 
 static int snd_pcm_oss_set_format(struct snd_pcm_oss_file *pcm_oss_file, int format)

sound/core/seq/seq_clientmgr.c

@@ -255,12 +255,12 @@ static int seq_free_client1(struct snd_seq_client *client)
 
 	if (!client)
 		return 0;
-	snd_seq_delete_all_ports(client);
-	snd_seq_queue_client_leave(client->number);
 	spin_lock_irqsave(&clients_lock, flags);
 	clienttablock[client->number] = 1;
 	clienttab[client->number] = NULL;
 	spin_unlock_irqrestore(&clients_lock, flags);
+	snd_seq_delete_all_ports(client);
+	snd_seq_queue_client_leave(client->number);
 	snd_use_lock_sync(&client->use_lock);
 	snd_seq_queue_client_termination(client->number);
 	if (client->pool)

sound/core/seq/seq_prioq.c

@@ -87,7 +87,7 @@ void snd_seq_prioq_delete(struct snd_seq_prioq **fifo)
 	if (f->cells > 0) {
 		/* drain prioQ */
 		while (f->cells > 0)
-			snd_seq_cell_free(snd_seq_prioq_cell_out(f));
+			snd_seq_cell_free(snd_seq_prioq_cell_out(f, NULL));
 	}
 	
 	kfree(f);
@@ -214,8 +214,18 @@ int snd_seq_prioq_cell_in(struct snd_seq_prioq * f,
 	return 0;
 }
 
+/* return 1 if the current time >= event timestamp */
+static int event_is_ready(struct snd_seq_event *ev, void *current_time)
+{
+	if ((ev->flags & SNDRV_SEQ_TIME_STAMP_MASK) == SNDRV_SEQ_TIME_STAMP_TICK)
+		return snd_seq_compare_tick_time(current_time, &ev->time.tick);
+	else
+		return snd_seq_compare_real_time(current_time, &ev->time.time);
+}
+
 /* dequeue cell from prioq */
-struct snd_seq_event_cell *snd_seq_prioq_cell_out(struct snd_seq_prioq *f)
+struct snd_seq_event_cell *snd_seq_prioq_cell_out(struct snd_seq_prioq *f,
+						  void *current_time)
 {
 	struct snd_seq_event_cell *cell;
 	unsigned long flags;
@@ -227,6 +237,8 @@ struct snd_seq_event_cell *snd_seq_prioq_cell_out(struct snd_seq_prioq *f)
 	spin_lock_irqsave(&f->lock, flags);
 
 	cell = f->head;
+	if (cell && current_time && !event_is_ready(&cell->event, current_time))
+		cell = NULL;
 	if (cell) {
 		f->head = cell->next;
 
@@ -252,18 +264,6 @@ int snd_seq_prioq_avail(struct snd_seq_prioq * f)
 	return f->cells;
 }
 
-
-/* peek at cell at the head of the prioq */
-struct snd_seq_event_cell *snd_seq_prioq_cell_peek(struct snd_seq_prioq * f)
-{
-	if (f == NULL) {
-		pr_debug("ALSA: seq: snd_seq_prioq_cell_in() called with NULL prioq\n");
-		return NULL;
-	}
-	return f->head;
-}
-
-
 static inline int prioq_match(struct snd_seq_event_cell *cell,
 			      int client, int timestamp)
 {

sound/core/seq/seq_prioq.h

@@ -44,14 +44,12 @@ void snd_seq_prioq_delete(struct snd_seq_prioq **fifo);
 int snd_seq_prioq_cell_in(struct snd_seq_prioq *f, struct snd_seq_event_cell *cell);
 
 /* dequeue cell from prioq */ 
-struct snd_seq_event_cell *snd_seq_prioq_cell_out(struct snd_seq_prioq *f);
+struct snd_seq_event_cell *snd_seq_prioq_cell_out(struct snd_seq_prioq *f,
+						  void *current_time);
 
 /* return number of events available in prioq */
 int snd_seq_prioq_avail(struct snd_seq_prioq *f);
 
-/* peek at cell at the head of the prioq */
-struct snd_seq_event_cell *snd_seq_prioq_cell_peek(struct snd_seq_prioq *f);
-
 /* client left queue */
 void snd_seq_prioq_leave(struct snd_seq_prioq *f, int client, int timestamp);        
 

sound/core/seq/seq_queue.c

@@ -277,30 +277,20 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop)
 
       __again:
 	/* Process tick queue... */
-	while ((cell = snd_seq_prioq_cell_peek(q->tickq)) != NULL) {
-		if (snd_seq_compare_tick_time(&q->timer->tick.cur_tick,
-					      &cell->event.time.tick)) {
-			cell = snd_seq_prioq_cell_out(q->tickq);
-			if (cell)
-				snd_seq_dispatch_event(cell, atomic, hop);
-		} else {
-			/* event remains in the queue */
+	for (;;) {
+		cell = snd_seq_prioq_cell_out(q->tickq,
+					      &q->timer->tick.cur_tick);
+		if (!cell)
 			break;
-		}
+		snd_seq_dispatch_event(cell, atomic, hop);
 	}
 
-
 	/* Process time queue... */
-	while ((cell = snd_seq_prioq_cell_peek(q->timeq)) != NULL) {
-		if (snd_seq_compare_real_time(&q->timer->cur_time,
-					      &cell->event.time.time)) {
-			cell = snd_seq_prioq_cell_out(q->timeq);
-			if (cell)
-				snd_seq_dispatch_event(cell, atomic, hop);
-		} else {
-			/* event remains in the queue */
+	for (;;) {
+		cell = snd_seq_prioq_cell_out(q->timeq, &q->timer->cur_time);
+		if (!cell)
 			break;
-		}
+		snd_seq_dispatch_event(cell, atomic, hop);
 	}
 
 	/* free lock */

sound/pci/hda/hda_intel.c

@@ -181,11 +181,15 @@ static const struct kernel_param_ops param_ops_xint = {
 };
 #define param_check_xint param_check_int
 
-static int power_save = -1;
+static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;
 module_param(power_save, xint, 0644);
 MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
 		 "(in second, 0 = disable).");
 
+static bool pm_blacklist = true;
+module_param(pm_blacklist, bool, 0644);
+MODULE_PARM_DESC(pm_blacklist, "Enable power-management blacklist");
+
 /* reset the HD-audio controller in power save mode.
  * this may give more power-saving, but will take longer time to
  * wake up.
@@ -2300,10 +2304,9 @@ static int azx_probe_continue(struct azx *chip)
 
 	val = power_save;
 #ifdef CONFIG_PM
-	if (val == -1) {
+	if (pm_blacklist) {
 		const struct snd_pci_quirk *q;
 
-		val = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;
 		q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist);
 		if (q && val) {
 			dev_info(chip->card->dev, "device %04x:%04x is on the power_save blacklist, forcing power_save to 0\n",

sound/soc/amd/acp-pcm-dma.c

@@ -579,13 +579,6 @@ static int acp_init(void __iomem *acp_mmio, u32 asic_type)
 		for (bank = 1; bank < 48; bank++)
 			acp_set_sram_bank_state(acp_mmio, bank, false);
 	}
-
-	/* Stoney supports 16bit resolution */
-	if (asic_type == CHIP_STONEY) {
-		val = acp_reg_read(acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN);
-		val |= 0x03;
-		acp_reg_write(val, acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN);
-	}
 	return 0;
 }
 
@@ -774,6 +767,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
 {
 	int status;
 	uint64_t size;
+	u32 val = 0;
 	struct page *pg;
 	struct snd_pcm_runtime *runtime;
 	struct audio_substream_data *rtd;
@@ -786,6 +780,14 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
 	if (WARN_ON(!rtd))
 		return -EINVAL;
 
+	if (adata->asic_type == CHIP_STONEY) {
+		val = acp_reg_read(adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val |= ACP_I2S_SP_16BIT_RESOLUTION_EN;
+		else
+			val |= ACP_I2S_MIC_16BIT_RESOLUTION_EN;
+		acp_reg_write(val, adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN);
+	}
 	size = params_buffer_bytes(params);
 	status = snd_pcm_lib_malloc_pages(substream, size);
 	if (status < 0)

sound/soc/amd/acp.h

@@ -70,6 +70,8 @@
 #define CAPTURE_END_DMA_DESCR_CH15 7
 
 #define mmACP_I2S_16BIT_RESOLUTION_EN       0x5209
+#define ACP_I2S_MIC_16BIT_RESOLUTION_EN 0x01
+#define ACP_I2S_SP_16BIT_RESOLUTION_EN	0x02
 enum acp_dma_priority_level {
 	/* 0x0 Specifies the DMA channel is given normal priority */
 	ACP_DMA_PRIORITY_LEVEL_NORMAL = 0x0,

sound/soc/codecs/hdmi-codec.c

@@ -798,12 +798,7 @@ static int hdmi_codec_probe(struct platform_device *pdev)
 
 static int hdmi_codec_remove(struct platform_device *pdev)
 {
-	struct device *dev = &pdev->dev;
-	struct hdmi_codec_priv *hcp;
-
-	hcp = dev_get_drvdata(dev);
-	kfree(hcp->chmap_info);
-	snd_soc_unregister_codec(dev);
+	snd_soc_unregister_codec(&pdev->dev);
 
 	return 0;
 }

sound/soc/codecs/rt5651.c

@@ -1722,6 +1722,7 @@ static const struct regmap_config rt5651_regmap = {
 	.num_reg_defaults = ARRAY_SIZE(rt5651_reg),
 	.ranges = rt5651_ranges,
 	.num_ranges = ARRAY_SIZE(rt5651_ranges),
+	.use_single_rw = true,
 };
 
 #if defined(CONFIG_OF)

sound/soc/codecs/sgtl5000.c

@@ -529,10 +529,15 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
 static int sgtl5000_digital_mute(struct snd_soc_dai *codec_dai, int mute)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	u16 adcdac_ctrl = SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT;
+	u16 i2s_pwr = SGTL5000_I2S_IN_POWERUP;
 
-	snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL,
-			adcdac_ctrl, mute ? adcdac_ctrl : 0);
+	/*
+	 * During 'digital mute' do not mute DAC
+	 * because LINE_IN would be muted aswell. We want to mute
+	 * only I2S block - this can be done by powering it off
+	 */
+	snd_soc_update_bits(codec, SGTL5000_CHIP_DIG_POWER,
+			i2s_pwr, mute ? 0 : i2s_pwr);
 
 	return 0;
 }
@@ -871,15 +876,26 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream,
 static int sgtl5000_set_bias_level(struct snd_soc_codec *codec,
 				   enum snd_soc_bias_level level)
 {
+	struct sgtl5000_priv *sgtl = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 	case SND_SOC_BIAS_PREPARE:
 	case SND_SOC_BIAS_STANDBY:
+		regcache_cache_only(sgtl->regmap, false);
+		ret = regcache_sync(sgtl->regmap);
+		if (ret) {
+			regcache_cache_only(sgtl->regmap, true);
+			return ret;
+		}
+
 		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
 				    SGTL5000_REFTOP_POWERUP,
 				    SGTL5000_REFTOP_POWERUP);
 		break;
 	case SND_SOC_BIAS_OFF:
+		regcache_cache_only(sgtl->regmap, true);
 		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
 				    SGTL5000_REFTOP_POWERUP, 0);
 		break;
@@ -1237,6 +1253,10 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 	 */
 	snd_soc_write(codec, SGTL5000_DAP_CTRL, 0);
 
+	/* Unmute DAC after start */
+	snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL,
+		SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT, 0);
+
 	return 0;
 
 err:

sound/soc/codecs/wm_adsp.c

@@ -1204,12 +1204,14 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl)
 		kcontrol->put = wm_coeff_put_acked;
 		break;
 	default:
-		kcontrol->get = wm_coeff_get;
-		kcontrol->put = wm_coeff_put;
-
-		ctl->bytes_ext.max = ctl->len;
-		ctl->bytes_ext.get = wm_coeff_tlv_get;
-		ctl->bytes_ext.put = wm_coeff_tlv_put;
+		if (kcontrol->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) {
+			ctl->bytes_ext.max = ctl->len;
+			ctl->bytes_ext.get = wm_coeff_tlv_get;
+			ctl->bytes_ext.put = wm_coeff_tlv_put;
+		} else {
+			kcontrol->get = wm_coeff_get;
+			kcontrol->put = wm_coeff_put;
+		}
 		break;
 	}
 

sound/soc/sunxi/sun4i-i2s.c

@@ -104,7 +104,7 @@
 
 #define SUN8I_I2S_CHAN_CFG_REG		0x30
 #define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM_MASK	GENMASK(6, 4)
-#define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(chan)	(chan - 1)
+#define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(chan)	((chan - 1) << 4)
 #define SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM_MASK	GENMASK(2, 0)
 #define SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM(chan)	(chan - 1)