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-rw-r--r--include/sound/ump_convert.h1
-rw-r--r--sound/core/seq/seq_ports.h14
-rw-r--r--sound/core/seq/seq_ump_convert.c132
-rw-r--r--sound/core/ump_convert.c60
-rw-r--r--sound/firewire/amdtp-stream.c38
-rw-r--r--sound/firewire/amdtp-stream.h1
-rw-r--r--sound/pci/hda/hda_controller.h2
-rw-r--r--sound/pci/hda/hda_generic.c63
-rw-r--r--sound/pci/hda/hda_generic.h1
-rw-r--r--sound/pci/hda/hda_intel.c10
-rw-r--r--sound/pci/hda/patch_conexant.c56
-rw-r--r--sound/pci/hda/patch_realtek.c1
-rw-r--r--sound/usb/stream.c4
13 files changed, 241 insertions, 142 deletions
diff --git a/include/sound/ump_convert.h b/include/sound/ump_convert.h
index 28c364c63245..d099ae27f849 100644
--- a/include/sound/ump_convert.h
+++ b/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 */
diff --git a/sound/core/seq/seq_ports.h b/sound/core/seq/seq_ports.h
index b111382f697a..9e36738c0dd0 100644
--- a/sound/core/seq/seq_ports.h
+++ b/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
};
diff --git a/sound/core/seq/seq_ump_convert.c b/sound/core/seq/seq_ump_convert.c
index e90b27a135e6..4dd540cbb1cb 100644
--- a/sound/core/seq/seq_ump_convert.c
+++ b/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;
}
diff --git a/sound/core/ump_convert.c b/sound/core/ump_convert.c
index f67c44c83fde..0fe13d031656 100644
--- a/sound/core/ump_convert.c
+++ b/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]);
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 1a163bbcabd7..c827d7d8d800 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/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);
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index a1ed2e80f91a..775db3fc4959 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/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;
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index c2d0109866e6..68c883f202ca 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/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 */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index f64d9dc197a3..9cff87dfbecb 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/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
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 8f5ecf740c49..08544601b4ce 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/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 */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index b33602e64d17..97d33a48ff17 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/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");
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 17389a3801bd..f030669243f9 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/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;
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ba0ce8750ca4..1645d21d422f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/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),
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index d5409f387945..e14c725acebf 100644
--- a/sound/usb/stream.c
+++ b/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 */
generated by cgit v1.2.3-58-ga151 (git 2.39.0) at 2024-11-21 23:20:25 +0000