Merge tag 'sound-6.11-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A small collection of fixes:

   - Revert of FireWire changes that caused a long-time regression

   - Another long-time regression fix for AMD HDMI

   - MIDI2 UMP fixes

   - HD-audio Conexant codec fixes and a quirk"

* tag 'sound-6.11-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda: Conditionally use snooping for AMD HDMI
  ALSA: usb-audio: Correct surround channels in UAC1 channel map
  ALSA: seq: ump: Explicitly reset RPN with Null RPN
  ALSA: seq: ump: Transmit RPN/NRPN message at each MSB/LSB data reception
  ALSA: seq: ump: Use the common RPN/bank conversion context
  ALSA: ump: Explicitly reset RPN with Null RPN
  ALSA: ump: Transmit RPN/NRPN message at each MSB/LSB data reception
  Revert "ALSA: firewire-lib: operate for period elapse event in process context"
  Revert "ALSA: firewire-lib: obsolete workqueue for period update"
  ALSA: hda/realtek: Add quirk for Acer Aspire E5-574G
  ALSA: seq: ump: Optimize conversions from SysEx to UMP
  ALSA: hda/conexant: Mute speakers at suspend / shutdown
  ALSA: hda/generic: Add a helper to mute speakers at suspend/shutdown
  ALSA: hda: conexant: Fix headset auto detect fail in the polling mode

include/sound/ump_convert.h

@@ -13,6 +13,7 @@ struct ump_cvt_to_ump_bank {
 	unsigned char cc_nrpn_msb, cc_nrpn_lsb;
 	unsigned char cc_data_msb, cc_data_lsb;
 	unsigned char cc_bank_msb, cc_bank_lsb;
+	bool cc_data_msb_set, cc_data_lsb_set;
 };
 
 /* context for converting from MIDI1 byte stream to UMP packet */

sound/core/seq/seq_ports.h

@@ -7,6 +7,7 @@
 #define __SND_SEQ_PORTS_H
 
 #include <sound/seq_kernel.h>
+#include <sound/ump_convert.h>
 #include "seq_lock.h"
 
 /* list of 'exported' ports */
@@ -42,17 +43,6 @@ struct snd_seq_port_subs_info {
 	int (*close)(void *private_data, struct snd_seq_port_subscribe *info);
 };
 
-/* context for converting from legacy control event to UMP packet */
-struct snd_seq_ump_midi2_bank {
-	bool rpn_set;
-	bool nrpn_set;
-	bool bank_set;
-	unsigned char cc_rpn_msb, cc_rpn_lsb;
-	unsigned char cc_nrpn_msb, cc_nrpn_lsb;
-	unsigned char cc_data_msb, cc_data_lsb;
-	unsigned char cc_bank_msb, cc_bank_lsb;
-};
-
 struct snd_seq_client_port {
 
 	struct snd_seq_addr addr;	/* client/port number */
@@ -88,7 +78,7 @@ struct snd_seq_client_port {
 	unsigned char ump_group;
 
 #if IS_ENABLED(CONFIG_SND_SEQ_UMP)
-	struct snd_seq_ump_midi2_bank midi2_bank[16]; /* per channel */
+	struct ump_cvt_to_ump_bank midi2_bank[16]; /* per channel */
 #endif
 };
 

sound/core/seq/seq_ump_convert.c

@@ -368,7 +368,7 @@ static int cvt_ump_midi1_to_midi2(struct snd_seq_client *dest,
 	struct snd_seq_ump_event ev_cvt;
 	const union snd_ump_midi1_msg *midi1 = (const union snd_ump_midi1_msg *)event->ump;
 	union snd_ump_midi2_msg *midi2 = (union snd_ump_midi2_msg *)ev_cvt.ump;
-	struct snd_seq_ump_midi2_bank *cc;
+	struct ump_cvt_to_ump_bank *cc;
 
 	ev_cvt = *event;
 	memset(&ev_cvt.ump, 0, sizeof(ev_cvt.ump));
@@ -789,28 +789,45 @@ static int paf_ev_to_ump_midi2(const struct snd_seq_event *event,
 	return 1;
 }
 
+static void reset_rpn(struct ump_cvt_to_ump_bank *cc)
+{
+	cc->rpn_set = 0;
+	cc->nrpn_set = 0;
+	cc->cc_rpn_msb = cc->cc_rpn_lsb = 0;
+	cc->cc_data_msb = cc->cc_data_lsb = 0;
+	cc->cc_data_msb_set = cc->cc_data_lsb_set = 0;
+}
+
 /* set up the MIDI2 RPN/NRPN packet data from the parsed info */
-static void fill_rpn(struct snd_seq_ump_midi2_bank *cc,
-		     union snd_ump_midi2_msg *data,
-		     unsigned char channel)
+static int fill_rpn(struct ump_cvt_to_ump_bank *cc,
+		    union snd_ump_midi2_msg *data,
+		    unsigned char channel,
+		    bool flush)
 {
+	if (!(cc->cc_data_lsb_set || cc->cc_data_msb_set))
+		return 0; // skip
+	/* when not flushing, wait for complete data set */
+	if (!flush && (!cc->cc_data_lsb_set || !cc->cc_data_msb_set))
+		return 0; // skip
+
 	if (cc->rpn_set) {
 		data->rpn.status = UMP_MSG_STATUS_RPN;
 		data->rpn.bank = cc->cc_rpn_msb;
 		data->rpn.index = cc->cc_rpn_lsb;
-		cc->rpn_set = 0;
-		cc->cc_rpn_msb = cc->cc_rpn_lsb = 0;
-	} else {
+	} else if (cc->nrpn_set) {
 		data->rpn.status = UMP_MSG_STATUS_NRPN;
 		data->rpn.bank = cc->cc_nrpn_msb;
 		data->rpn.index = cc->cc_nrpn_lsb;
-		cc->nrpn_set = 0;
-		cc->cc_nrpn_msb = cc->cc_nrpn_lsb = 0;
+	} else {
+		return 0; // skip
 	}
+
 	data->rpn.data = upscale_14_to_32bit((cc->cc_data_msb << 7) |
 					     cc->cc_data_lsb);
 	data->rpn.channel = channel;
-	cc->cc_data_msb = cc->cc_data_lsb = 0;
+
+	reset_rpn(cc);
+	return 1;
 }
 
 /* convert CC event to MIDI 2.0 UMP */
@@ -822,29 +839,39 @@ static int cc_ev_to_ump_midi2(const struct snd_seq_event *event,
 	unsigned char channel = event->data.control.channel & 0x0f;
 	unsigned char index = event->data.control.param & 0x7f;
 	unsigned char val = event->data.control.value & 0x7f;
-	struct snd_seq_ump_midi2_bank *cc = &dest_port->midi2_bank[channel];
+	struct ump_cvt_to_ump_bank *cc = &dest_port->midi2_bank[channel];
+	int ret;
 
 	/* process special CC's (bank/rpn/nrpn) */
 	switch (index) {
 	case UMP_CC_RPN_MSB:
+		ret = fill_rpn(cc, data, channel, true);
 		cc->rpn_set = 1;
 		cc->cc_rpn_msb = val;
-		return 0; // skip
+		if (cc->cc_rpn_msb == 0x7f && cc->cc_rpn_lsb == 0x7f)
+			reset_rpn(cc);
+		return ret;
 	case UMP_CC_RPN_LSB:
+		ret = fill_rpn(cc, data, channel, true);
 		cc->rpn_set = 1;
 		cc->cc_rpn_lsb = val;
-		return 0; // skip
+		if (cc->cc_rpn_msb == 0x7f && cc->cc_rpn_lsb == 0x7f)
+			reset_rpn(cc);
+		return ret;
 	case UMP_CC_NRPN_MSB:
+		ret = fill_rpn(cc, data, channel, true);
 		cc->nrpn_set = 1;
 		cc->cc_nrpn_msb = val;
-		return 0; // skip
+		return ret;
 	case UMP_CC_NRPN_LSB:
+		ret = fill_rpn(cc, data, channel, true);
 		cc->nrpn_set = 1;
 		cc->cc_nrpn_lsb = val;
-		return 0; // skip
+		return ret;
 	case UMP_CC_DATA:
+		cc->cc_data_msb_set = 1;
 		cc->cc_data_msb = val;
-		return 0; // skip
+		return fill_rpn(cc, data, channel, false);
 	case UMP_CC_BANK_SELECT:
 		cc->bank_set = 1;
 		cc->cc_bank_msb = val;
@@ -854,11 +881,9 @@ static int cc_ev_to_ump_midi2(const struct snd_seq_event *event,
 		cc->cc_bank_lsb = val;
 		return 0; // skip
 	case UMP_CC_DATA_LSB:
+		cc->cc_data_lsb_set = 1;
 		cc->cc_data_lsb = val;
-		if (!(cc->rpn_set || cc->nrpn_set))
-			return 0; // skip
-		fill_rpn(cc, data, channel);
-		return 1;
+		return fill_rpn(cc, data, channel, false);
 	}
 
 	data->cc.status = status;
@@ -887,7 +912,7 @@ static int pgm_ev_to_ump_midi2(const struct snd_seq_event *event,
 			       unsigned char status)
 {
 	unsigned char channel = event->data.control.channel & 0x0f;
-	struct snd_seq_ump_midi2_bank *cc = &dest_port->midi2_bank[channel];
+	struct ump_cvt_to_ump_bank *cc = &dest_port->midi2_bank[channel];
 
 	data->pg.status = status;
 	data->pg.channel = channel;
@@ -924,8 +949,9 @@ static int ctrl14_ev_to_ump_midi2(const struct snd_seq_event *event,
 {
 	unsigned char channel = event->data.control.channel & 0x0f;
 	unsigned char index = event->data.control.param & 0x7f;
-	struct snd_seq_ump_midi2_bank *cc = &dest_port->midi2_bank[channel];
+	struct ump_cvt_to_ump_bank *cc = &dest_port->midi2_bank[channel];
 	unsigned char msb, lsb;
+	int ret;
 
 	msb = (event->data.control.value >> 7) & 0x7f;
 	lsb = event->data.control.value & 0x7f;
@@ -939,28 +965,27 @@ static int ctrl14_ev_to_ump_midi2(const struct snd_seq_event *event,
 		cc->cc_bank_lsb = lsb;
 		return 0; // skip
 	case UMP_CC_RPN_MSB:
-		cc->cc_rpn_msb = msb;
-		fallthrough;
 	case UMP_CC_RPN_LSB:
-		cc->rpn_set = 1;
+		ret = fill_rpn(cc, data, channel, true);
+		cc->cc_rpn_msb = msb;
 		cc->cc_rpn_lsb = lsb;
-		return 0; // skip
+		cc->rpn_set = 1;
+		if (cc->cc_rpn_msb == 0x7f && cc->cc_rpn_lsb == 0x7f)
+			reset_rpn(cc);
+		return ret;
 	case UMP_CC_NRPN_MSB:
-		cc->cc_nrpn_msb = msb;
-		fallthrough;
 	case UMP_CC_NRPN_LSB:
+		ret = fill_rpn(cc, data, channel, true);
+		cc->cc_nrpn_msb = msb;
 		cc->nrpn_set = 1;
 		cc->cc_nrpn_lsb = lsb;
-		return 0; // skip
+		return ret;
 	case UMP_CC_DATA:
-		cc->cc_data_msb = msb;
-		fallthrough;
 	case UMP_CC_DATA_LSB:
+		cc->cc_data_msb_set = cc->cc_data_lsb_set = 1;
+		cc->cc_data_msb = msb;
 		cc->cc_data_lsb = lsb;
-		if (!(cc->rpn_set || cc->nrpn_set))
-			return 0; // skip
-		fill_rpn(cc, data, channel);
-		return 1;
+		return fill_rpn(cc, data, channel, false);
 	}
 
 	data->cc.status = UMP_MSG_STATUS_CC;
@@ -1192,44 +1217,53 @@ static int cvt_sysex_to_ump(struct snd_seq_client *dest,
 {
 	struct snd_seq_ump_event ev_cvt;
 	unsigned char status;
-	u8 buf[6], *xbuf;
+	u8 buf[8], *xbuf;
 	int offset = 0;
 	int len, err;
+	bool finished = false;
 
 	if (!snd_seq_ev_is_variable(event))
 		return 0;
 
 	setup_ump_event(&ev_cvt, event);
-	for (;;) {
+	while (!finished) {
 		len = snd_seq_expand_var_event_at(event, sizeof(buf), buf, offset);
 		if (len <= 0)
 			break;
-		if (WARN_ON(len > 6))
+		if (WARN_ON(len > sizeof(buf)))
 			break;
-		offset += len;
+
 		xbuf = buf;
+		status = UMP_SYSEX_STATUS_CONTINUE;
+		/* truncate the sysex start-marker */
 		if (*xbuf == UMP_MIDI1_MSG_SYSEX_START) {
 			status = UMP_SYSEX_STATUS_START;
-			xbuf++;
 			len--;
-			if (len > 0 && xbuf[len - 1] == UMP_MIDI1_MSG_SYSEX_END) {
+			offset++;
+			xbuf++;
+		}
+
+		/* if the last of this packet or the 1st byte of the next packet
+		 * is the end-marker, finish the transfer with this packet
+		 */
+		if (len > 0 && len < 8 &&
+		    xbuf[len - 1] == UMP_MIDI1_MSG_SYSEX_END) {
+			if (status == UMP_SYSEX_STATUS_START)
 				status = UMP_SYSEX_STATUS_SINGLE;
-				len--;
-			}
-		} else {
-			if (xbuf[len - 1] == UMP_MIDI1_MSG_SYSEX_END) {
+			else
 				status = UMP_SYSEX_STATUS_END;
-				len--;
-			} else {
-				status = UMP_SYSEX_STATUS_CONTINUE;
-			}
+			len--;
+			finished = true;
 		}
+
+		len = min(len, 6);
 		fill_sysex7_ump(dest_port, ev_cvt.ump, status, xbuf, len);
 		err = __snd_seq_deliver_single_event(dest, dest_port,
 						     (struct snd_seq_event *)&ev_cvt,
 						     atomic, hop);
 		if (err < 0)
 			return err;
+		offset += len;
 	}
 	return 0;
 }

sound/core/ump_convert.c

@@ -287,25 +287,42 @@ static int cvt_legacy_system_to_ump(struct ump_cvt_to_ump *cvt,
 	return 4;
 }
 
-static void fill_rpn(struct ump_cvt_to_ump_bank *cc,
-		     union snd_ump_midi2_msg *midi2)
+static void reset_rpn(struct ump_cvt_to_ump_bank *cc)
 {
+	cc->rpn_set = 0;
+	cc->nrpn_set = 0;
+	cc->cc_rpn_msb = cc->cc_rpn_lsb = 0;
+	cc->cc_data_msb = cc->cc_data_lsb = 0;
+	cc->cc_data_msb_set = cc->cc_data_lsb_set = 0;
+}
+
+static int fill_rpn(struct ump_cvt_to_ump_bank *cc,
+		    union snd_ump_midi2_msg *midi2,
+		    bool flush)
+{
+	if (!(cc->cc_data_lsb_set || cc->cc_data_msb_set))
+		return 0; // skip
+	/* when not flushing, wait for complete data set */
+	if (!flush && (!cc->cc_data_lsb_set || !cc->cc_data_msb_set))
+		return 0; // skip
+
 	if (cc->rpn_set) {
 		midi2->rpn.status = UMP_MSG_STATUS_RPN;
 		midi2->rpn.bank = cc->cc_rpn_msb;
 		midi2->rpn.index = cc->cc_rpn_lsb;
-		cc->rpn_set = 0;
-		cc->cc_rpn_msb = cc->cc_rpn_lsb = 0;
-	} else {
+	} else if (cc->nrpn_set) {
 		midi2->rpn.status = UMP_MSG_STATUS_NRPN;
 		midi2->rpn.bank = cc->cc_nrpn_msb;
 		midi2->rpn.index = cc->cc_nrpn_lsb;
-		cc->nrpn_set = 0;
-		cc->cc_nrpn_msb = cc->cc_nrpn_lsb = 0;
+	} else {
+		return 0; // skip
 	}
+
 	midi2->rpn.data = upscale_14_to_32bit((cc->cc_data_msb << 7) |
 					      cc->cc_data_lsb);
-	cc->cc_data_msb = cc->cc_data_lsb = 0;
+
+	reset_rpn(cc);
+	return 1;
 }
 
 /* convert to a MIDI 1.0 Channel Voice message */
@@ -318,6 +335,7 @@ static int cvt_legacy_cmd_to_ump(struct ump_cvt_to_ump *cvt,
 	struct ump_cvt_to_ump_bank *cc;
 	union snd_ump_midi2_msg *midi2 = (union snd_ump_midi2_msg *)data;
 	unsigned char status, channel;
+	int ret;
 
 	BUILD_BUG_ON(sizeof(union snd_ump_midi1_msg) != 4);
 	BUILD_BUG_ON(sizeof(union snd_ump_midi2_msg) != 8);
@@ -358,24 +376,33 @@ static int cvt_legacy_cmd_to_ump(struct ump_cvt_to_ump *cvt,
 	case UMP_MSG_STATUS_CC:
 		switch (buf[1]) {
 		case UMP_CC_RPN_MSB:
+			ret = fill_rpn(cc, midi2, true);
 			cc->rpn_set = 1;
 			cc->cc_rpn_msb = buf[2];
-			return 0; // skip
+			if (cc->cc_rpn_msb == 0x7f && cc->cc_rpn_lsb == 0x7f)
+				reset_rpn(cc);
+			return ret;
 		case UMP_CC_RPN_LSB:
+			ret = fill_rpn(cc, midi2, true);
 			cc->rpn_set = 1;
 			cc->cc_rpn_lsb = buf[2];
-			return 0; // skip
+			if (cc->cc_rpn_msb == 0x7f && cc->cc_rpn_lsb == 0x7f)
+				reset_rpn(cc);
+			return ret;
 		case UMP_CC_NRPN_MSB:
+			ret = fill_rpn(cc, midi2, true);
 			cc->nrpn_set = 1;
 			cc->cc_nrpn_msb = buf[2];
-			return 0; // skip
+			return ret;
 		case UMP_CC_NRPN_LSB:
+			ret = fill_rpn(cc, midi2, true);
 			cc->nrpn_set = 1;
 			cc->cc_nrpn_lsb = buf[2];
-			return 0; // skip
+			return ret;
 		case UMP_CC_DATA:
+			cc->cc_data_msb_set = 1;
 			cc->cc_data_msb = buf[2];
-			return 0; // skip
+			return fill_rpn(cc, midi2, false);
 		case UMP_CC_BANK_SELECT:
 			cc->bank_set = 1;
 			cc->cc_bank_msb = buf[2];
@@ -385,12 +412,9 @@ static int cvt_legacy_cmd_to_ump(struct ump_cvt_to_ump *cvt,
 			cc->cc_bank_lsb = buf[2];
 			return 0; // skip
 		case UMP_CC_DATA_LSB:
+			cc->cc_data_lsb_set = 1;
 			cc->cc_data_lsb = buf[2];
-			if (cc->rpn_set || cc->nrpn_set)
-				fill_rpn(cc, midi2);
-			else
-				return 0; // skip
-			break;
+			return fill_rpn(cc, midi2, false);
 		default:
 			midi2->cc.index = buf[1];
 			midi2->cc.data = upscale_7_to_32bit(buf[2]);

sound/firewire/amdtp-stream.c

@@ -77,6 +77,8 @@
 // overrun. Actual device can skip more, then this module stops the packet streaming.
 #define IR_JUMBO_PAYLOAD_MAX_SKIP_CYCLES	5
 
+static void pcm_period_work(struct work_struct *work);
+
 /**
  * amdtp_stream_init - initialize an AMDTP stream structure
  * @s: the AMDTP stream to initialize
@@ -105,6 +107,7 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
 	s->flags = flags;
 	s->context = ERR_PTR(-1);
 	mutex_init(&s->mutex);
+	INIT_WORK(&s->period_work, pcm_period_work);
 	s->packet_index = 0;
 
 	init_waitqueue_head(&s->ready_wait);
@@ -347,6 +350,7 @@ EXPORT_SYMBOL(amdtp_stream_get_max_payload);
  */
 void amdtp_stream_pcm_prepare(struct amdtp_stream *s)
 {
+	cancel_work_sync(&s->period_work);
 	s->pcm_buffer_pointer = 0;
 	s->pcm_period_pointer = 0;
 }
@@ -611,19 +615,21 @@ static void update_pcm_pointers(struct amdtp_stream *s,
 		// The program in user process should periodically check the status of intermediate
 		// buffer associated to PCM substream to process PCM frames in the buffer, instead
 		// of receiving notification of period elapsed by poll wait.
-		if (!pcm->runtime->no_period_wakeup) {
-			if (in_softirq()) {
-				// In software IRQ context for 1394 OHCI.
-				snd_pcm_period_elapsed(pcm);
-			} else {
-				// In process context of ALSA PCM application under acquired lock of
-				// PCM substream.
-				snd_pcm_period_elapsed_under_stream_lock(pcm);
-			}
-		}
+		if (!pcm->runtime->no_period_wakeup)
+			queue_work(system_highpri_wq, &s->period_work);
 	}
 }
 
+static void pcm_period_work(struct work_struct *work)
+{
+	struct amdtp_stream *s = container_of(work, struct amdtp_stream,
+					      period_work);
+	struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
+
+	if (pcm)
+		snd_pcm_period_elapsed(pcm);
+}
+
 static int queue_packet(struct amdtp_stream *s, struct fw_iso_packet *params,
 			bool sched_irq)
 {
@@ -1849,11 +1855,14 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d,
 {
 	struct amdtp_stream *irq_target = d->irq_target;
 
-	// Process isochronous packets queued till recent isochronous cycle to handle PCM frames.
 	if (irq_target && amdtp_stream_running(irq_target)) {
-		// In software IRQ context, the call causes dead-lock to disable the tasklet
-		// synchronously.
-		if (!in_softirq())
+		// use wq to prevent AB/BA deadlock competition for
+		// substream lock:
+		// fw_iso_context_flush_completions() acquires
+		// lock by ohci_flush_iso_completions(),
+		// amdtp-stream process_rx_packets() attempts to
+		// acquire same lock by snd_pcm_elapsed()
+		if (current_work() != &s->period_work)
 			fw_iso_context_flush_completions(irq_target->context);
 	}
 
@@ -1909,6 +1918,7 @@ static void amdtp_stream_stop(struct amdtp_stream *s)
 		return;
 	}
 
+	cancel_work_sync(&s->period_work);
 	fw_iso_context_stop(s->context);
 	fw_iso_context_destroy(s->context);
 	s->context = ERR_PTR(-1);

sound/firewire/amdtp-stream.h

@@ -191,6 +191,7 @@ struct amdtp_stream {
 
 	/* For a PCM substream processing. */
 	struct snd_pcm_substream *pcm;
+	struct work_struct period_work;
 	snd_pcm_uframes_t pcm_buffer_pointer;
 	unsigned int pcm_period_pointer;
 	unsigned int pcm_frame_multiplier;

sound/pci/hda/hda_controller.h

@@ -28,7 +28,7 @@
 #else
 #define AZX_DCAPS_I915_COMPONENT 0		/* NOP */
 #endif
-/* 14 unused */
+#define AZX_DCAPS_AMD_ALLOC_FIX	(1 << 14)	/* AMD allocation workaround */
 #define AZX_DCAPS_CTX_WORKAROUND (1 << 15)	/* X-Fi workaround */
 #define AZX_DCAPS_POSFIX_LPIB	(1 << 16)	/* Use LPIB as default */
 #define AZX_DCAPS_AMD_WORKAROUND (1 << 17)	/* AMD-specific workaround */

sound/pci/hda/hda_generic.c

@@ -4955,6 +4955,69 @@ void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on)
 }
 EXPORT_SYMBOL_GPL(snd_hda_gen_stream_pm);
 
+/* forcibly mute the speaker output without caching; return true if updated */
+static bool force_mute_output_path(struct hda_codec *codec, hda_nid_t nid)
+{
+	if (!nid)
+		return false;
+	if (!nid_has_mute(codec, nid, HDA_OUTPUT))
+		return false; /* no mute, skip */
+	if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) &
+	    snd_hda_codec_amp_read(codec, nid, 1, HDA_OUTPUT, 0) &
+	    HDA_AMP_MUTE)
+		return false; /* both channels already muted, skip */
+
+	/* direct amp update without caching */
+	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+			    AC_AMP_SET_OUTPUT | AC_AMP_SET_LEFT |
+			    AC_AMP_SET_RIGHT | HDA_AMP_MUTE);
+	return true;
+}
+
+/**
+ * snd_hda_gen_shutup_speakers - Forcibly mute the speaker outputs
+ * @codec: the HDA codec
+ *
+ * Forcibly mute the speaker outputs, to be called at suspend or shutdown.
+ *
+ * The mute state done by this function isn't cached, hence the original state
+ * will be restored at resume.
+ *
+ * Return true if the mute state has been changed.
+ */
+bool snd_hda_gen_shutup_speakers(struct hda_codec *codec)
+{
+	struct hda_gen_spec *spec = codec->spec;
+	const int *paths;
+	const struct nid_path *path;
+	int i, p, num_paths;
+	bool updated = false;
+
+	/* if already powered off, do nothing */
+	if (!snd_hdac_is_power_on(&codec->core))
+		return false;
+
+	if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) {
+		paths = spec->out_paths;
+		num_paths = spec->autocfg.line_outs;
+	} else {
+		paths = spec->speaker_paths;
+		num_paths = spec->autocfg.speaker_outs;
+	}
+
+	for (i = 0; i < num_paths; i++) {
+		path = snd_hda_get_path_from_idx(codec, paths[i]);
+		if (!path)
+			continue;
+		for (p = 0; p < path->depth; p++)
+			if (force_mute_output_path(codec, path->path[p]))
+				updated = true;
+	}
+
+	return updated;
+}
+EXPORT_SYMBOL_GPL(snd_hda_gen_shutup_speakers);
+
 /**
  * snd_hda_gen_parse_auto_config - Parse the given BIOS configuration and
  * set up the hda_gen_spec

sound/pci/hda/hda_generic.h

@@ -353,5 +353,6 @@ int snd_hda_gen_add_mute_led_cdev(struct hda_codec *codec,
 int snd_hda_gen_add_micmute_led_cdev(struct hda_codec *codec,
 				     int (*callback)(struct led_classdev *,
 						     enum led_brightness));
+bool snd_hda_gen_shutup_speakers(struct hda_codec *codec);
 
 #endif /* __SOUND_HDA_GENERIC_H */

sound/pci/hda/hda_intel.c

@@ -40,6 +40,7 @@
 
 #ifdef CONFIG_X86
 /* for snoop control */
+#include <linux/dma-map-ops.h>
 #include <asm/set_memory.h>
 #include <asm/cpufeature.h>
 #endif
@@ -306,7 +307,7 @@ enum {
 
 /* quirks for ATI HDMI with snoop off */
 #define AZX_DCAPS_PRESET_ATI_HDMI_NS \
-	(AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_SNOOP_OFF)
+	(AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_AMD_ALLOC_FIX)
 
 /* quirks for AMD SB */
 #define AZX_DCAPS_PRESET_AMD_SB \
@@ -1702,6 +1703,13 @@ static void azx_check_snoop_available(struct azx *chip)
 	if (chip->driver_caps & AZX_DCAPS_SNOOP_OFF)
 		snoop = false;
 
+#ifdef CONFIG_X86
+	/* check the presence of DMA ops (i.e. IOMMU), disable snoop conditionally */
+	if ((chip->driver_caps & AZX_DCAPS_AMD_ALLOC_FIX) &&
+	    !get_dma_ops(chip->card->dev))
+		snoop = false;
+#endif
+
 	chip->snoop = snoop;
 	if (!snoop) {
 		dev_info(chip->card->dev, "Force to non-snoop mode\n");

sound/pci/hda/patch_conexant.c

@@ -21,12 +21,6 @@
 #include "hda_jack.h"
 #include "hda_generic.h"
 
-enum {
-	CX_HEADSET_NOPRESENT = 0,
-	CX_HEADSET_PARTPRESENT,
-	CX_HEADSET_ALLPRESENT,
-};
-
 struct conexant_spec {
 	struct hda_gen_spec gen;
 
@@ -48,7 +42,6 @@ struct conexant_spec {
 	unsigned int gpio_led;
 	unsigned int gpio_mute_led_mask;
 	unsigned int gpio_mic_led_mask;
-	unsigned int headset_present_flag;
 	bool is_cx8070_sn6140;
 };
 
@@ -212,6 +205,8 @@ static void cx_auto_shutdown(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
 
+	snd_hda_gen_shutup_speakers(codec);
+
 	/* Turn the problematic codec into D3 to avoid spurious noises
 	   from the internal speaker during (and after) reboot */
 	cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false);
@@ -250,48 +245,19 @@ static void cx_process_headset_plugin(struct hda_codec *codec)
 	}
 }
 
-static void cx_update_headset_mic_vref(struct hda_codec *codec, unsigned int res)
+static void cx_update_headset_mic_vref(struct hda_codec *codec, struct hda_jack_callback *event)
 {
-	unsigned int phone_present, mic_persent, phone_tag, mic_tag;
-	struct conexant_spec *spec = codec->spec;
+	unsigned int mic_present;
 
 	/* In cx8070 and sn6140, the node 16 can only be config to headphone or disabled,
 	 * the node 19 can only be config to microphone or disabled.
 	 * Check hp&mic tag to process headset pulgin&plugout.
 	 */
-	phone_tag = snd_hda_codec_read(codec, 0x16, 0, AC_VERB_GET_UNSOLICITED_RESPONSE, 0x0);
-	mic_tag = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_UNSOLICITED_RESPONSE, 0x0);
-	if ((phone_tag & (res >> AC_UNSOL_RES_TAG_SHIFT)) ||
-	    (mic_tag & (res >> AC_UNSOL_RES_TAG_SHIFT))) {
-		phone_present = snd_hda_codec_read(codec, 0x16, 0, AC_VERB_GET_PIN_SENSE, 0x0);
-		if (!(phone_present & AC_PINSENSE_PRESENCE)) {/* headphone plugout */
-			spec->headset_present_flag = CX_HEADSET_NOPRESENT;
-			snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20);
-			return;
-		}
-		if (spec->headset_present_flag == CX_HEADSET_NOPRESENT) {
-			spec->headset_present_flag = CX_HEADSET_PARTPRESENT;
-		} else if (spec->headset_present_flag == CX_HEADSET_PARTPRESENT) {
-			mic_persent = snd_hda_codec_read(codec, 0x19, 0,
-							 AC_VERB_GET_PIN_SENSE, 0x0);
-			/* headset is present */
-			if ((phone_present & AC_PINSENSE_PRESENCE) &&
-			    (mic_persent & AC_PINSENSE_PRESENCE)) {
-				cx_process_headset_plugin(codec);
-				spec->headset_present_flag = CX_HEADSET_ALLPRESENT;
-			}
-		}
-	}
-}
-
-static void cx_jack_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	struct conexant_spec *spec = codec->spec;
-
-	if (spec->is_cx8070_sn6140)
-		cx_update_headset_mic_vref(codec, res);
-
-	snd_hda_jack_unsol_event(codec, res);
+	mic_present = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0);
+	if (!(mic_present & AC_PINSENSE_PRESENCE)) /* mic plugout */
+		snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20);
+	else
+		cx_process_headset_plugin(codec);
 }
 
 static int cx_auto_suspend(struct hda_codec *codec)
@@ -305,7 +271,7 @@ static const struct hda_codec_ops cx_auto_patch_ops = {
 	.build_pcms = snd_hda_gen_build_pcms,
 	.init = cx_auto_init,
 	.free = cx_auto_free,
-	.unsol_event = cx_jack_unsol_event,
+	.unsol_event = snd_hda_jack_unsol_event,
 	.suspend = cx_auto_suspend,
 	.check_power_status = snd_hda_gen_check_power_status,
 };
@@ -1163,7 +1129,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
 	case 0x14f11f86:
 	case 0x14f11f87:
 		spec->is_cx8070_sn6140 = true;
-		spec->headset_present_flag = CX_HEADSET_NOPRESENT;
+		snd_hda_jack_detect_enable_callback(codec, 0x19, cx_update_headset_mic_vref);
 		break;
 	}
 

sound/pci/hda/patch_realtek.c

@@ -9872,6 +9872,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
 	SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x0840, "Acer Aspire E1", ALC269VB_FIXUP_ASPIRE_E1_COEF),
+	SND_PCI_QUIRK(0x1025, 0x100c, "Acer Aspire E5-574G", ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1025, 0x101c, "Acer Veriton N2510G", ALC269_FIXUP_LIFEBOOK),
 	SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC),

sound/usb/stream.c

@@ -244,8 +244,8 @@ static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits,
 		SNDRV_CHMAP_FR,		/* right front */
 		SNDRV_CHMAP_FC,		/* center front */
 		SNDRV_CHMAP_LFE,	/* LFE */
-		SNDRV_CHMAP_SL,		/* left surround */
-		SNDRV_CHMAP_SR,		/* right surround */
+		SNDRV_CHMAP_RL,		/* left surround */
+		SNDRV_CHMAP_RR,		/* right surround */
 		SNDRV_CHMAP_FLC,	/* left of center */
 		SNDRV_CHMAP_FRC,	/* right of center */
 		SNDRV_CHMAP_RC,		/* surround */