Merge branch 'for-linus' into for-next
For taking back the recent change of HDA HDMI fixes for i915 HSW/BDW.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index e7193aa..d4510eb 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -655,17 +655,6 @@
and next kernels are found in for-linus and for-next branches,
respectively.
-If you are using the latest Linus tree, it'd be better to pull the
-above GIT tree onto it. If you are using the older kernels, an easy
-way to try the latest ALSA code is to build from the snapshot
-tarball. There are daily tarballs and the latest snapshot tarball.
-All can be built just like normal alsa-driver release packages, that
-is, installed via the usual spells: configure, make and make
-install(-modules). See INSTALL in the package. The snapshot tarballs
-are found at:
-
-- ftp://ftp.suse.com/pub/people/tiwai/snapshot/
-
Sending a Bug Report
~~~~~~~~~~~~~~~~~~~~
@@ -699,7 +688,12 @@
alsa-info
~~~~~~~~~
The script `alsa-info.sh` is a very useful tool to gather the audio
-device information. You can fetch the latest version from:
+device information. It's included in alsa-utils package. The latest
+version can be found on git repository:
+
+- git://git.alsa-project.org/alsa-utils.git
+
+The script can be fetched directly from the following URL, too:
- http://www.alsa-project.org/alsa-info.sh
@@ -836,15 +830,11 @@
(mixer) elements, set/get the control element value, simulate the PCM
operation, the jack plugging simulation, etc.
-The package is found in:
-
-- ftp://ftp.suse.com/pub/people/tiwai/misc/
-
-A git repository is available:
+The program is found in the git repository below:
- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git
-See README file in the tarball for more details about hda-emu
+See README file in the repository for more details about hda-emu
program.
diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt
index 630c492..81476b4 100644
--- a/Documentation/sound/alsa/compress_offload.txt
+++ b/Documentation/sound/alsa/compress_offload.txt
@@ -149,7 +149,7 @@
================
When playing thru an album, the decoders have the ability to skip the encoder
delay and padding and directly move from one track content to another. The end
-user can perceive this as gapless playback as we dont have silence while
+user can perceive this as gapless playback as we don't have silence while
switching from one track to another
Also, there might be low-intensity noises due to encoding. Perfect gapless is
diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
index 6faab48..c45bd79 100644
--- a/Documentation/sound/alsa/soc/dapm.txt
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -132,7 +132,7 @@
SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls,
ARRAY_SIZE(wm8731_output_mixer_controls)),
-If you dont want the mixer elements prefixed with the name of the mixer widget,
+If you don't want the mixer elements prefixed with the name of the mixer widget,
you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same
as for SND_SOC_DAPM_MIXER.
diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt
index ff88f52..f3f28b7 100644
--- a/Documentation/sound/alsa/soc/overview.txt
+++ b/Documentation/sound/alsa/soc/overview.txt
@@ -63,7 +63,7 @@
and any audio DSP drivers for that platform.
* Machine class driver: The machine driver class acts as the glue that
- decribes and binds the other component drivers together to form an ALSA
+ describes and binds the other component drivers together to form an ALSA
"sound card device". It handles any machine specific controls and
machine level audio events (e.g. turning on an amp at start of playback).
diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt
index 0b191a2..1b6473f 100644
--- a/Documentation/sound/alsa/timestamping.txt
+++ b/Documentation/sound/alsa/timestamping.txt
@@ -129,7 +129,7 @@
interpolation of the results
In some hardware-specific configuration, the system timestamp is
-latched by a low-level audio subsytem, and the information provided
+latched by a low-level audio subsystem, and the information provided
back to the driver. Due to potential delays in the communication with
the hardware, there is a risk of misalignment with the avail and delay
information. To make sure applications are not confused, a
diff --git a/include/sound/hda_chmap.h b/include/sound/hda_chmap.h
index e20d219..babd445 100644
--- a/include/sound/hda_chmap.h
+++ b/include/sound/hda_chmap.h
@@ -36,6 +36,8 @@
int (*chmap_validate)(struct hdac_chmap *hchmap, int ca,
int channels, unsigned char *chmap);
+ int (*get_spk_alloc)(struct hdac_device *hdac, int pcm_idx);
+
void (*get_chmap)(struct hdac_device *hdac, int pcm_idx,
unsigned char *chmap);
void (*set_chmap)(struct hdac_device *hdac, int pcm_idx,
diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h
index f5842bc..796cabf 100644
--- a/include/sound/hda_i915.h
+++ b/include/sound/hda_i915.h
@@ -10,8 +10,8 @@
int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable);
int snd_hdac_display_power(struct hdac_bus *bus, bool enable);
void snd_hdac_i915_set_bclk(struct hdac_bus *bus);
-int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate);
-int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid,
+int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate);
+int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid,
bool *audio_enabled, char *buffer, int max_bytes);
int snd_hdac_i915_init(struct hdac_bus *bus);
int snd_hdac_i915_exit(struct hdac_bus *bus);
@@ -28,12 +28,12 @@
static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus)
{
}
-static inline int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid,
- int rate)
+static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec,
+ hda_nid_t nid, int rate)
{
return 0;
}
-static inline int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid,
+static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid,
bool *audio_enabled, char *buffer,
int max_bytes)
{
diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index 67bf49d..609cadb 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -672,7 +672,7 @@
/* global timers (device member) */
#define SNDRV_TIMER_GLOBAL_SYSTEM 0
-#define SNDRV_TIMER_GLOBAL_RTC 1
+#define SNDRV_TIMER_GLOBAL_RTC 1 /* unused */
#define SNDRV_TIMER_GLOBAL_HPET 2
#define SNDRV_TIMER_GLOBAL_HRTIMER 3
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index 6d12ca9..9749f9e 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -141,35 +141,6 @@
Say Y here to use the HR-timer backend as the default sequencer
timer.
-config SND_RTCTIMER
- tristate "RTC Timer support"
- depends on RTC
- select SND_TIMER
- help
- Say Y here to enable RTC timer support for ALSA. ALSA uses
- the RTC timer as a precise timing source and maps the RTC
- timer to ALSA's timer interface. The ALSA sequencer code also
- can use this timing source.
-
- To compile this driver as a module, choose M here: the module
- will be called snd-rtctimer.
-
- Note that this option is exclusive with the new RTC drivers
- (CONFIG_RTC_CLASS) since this requires the old API.
-
-config SND_SEQ_RTCTIMER_DEFAULT
- bool "Use RTC as default sequencer timer"
- depends on SND_RTCTIMER && SND_SEQUENCER
- depends on !SND_SEQ_HRTIMER_DEFAULT
- default y
- help
- Say Y here to use the RTC timer as the default sequencer
- timer. This is strongly recommended because it ensures
- precise MIDI timing even when the system timer runs at less
- than 1000 Hz.
-
- If in doubt, say Y.
-
config SND_DYNAMIC_MINORS
bool "Dynamic device file minor numbers"
help
diff --git a/sound/core/Makefile b/sound/core/Makefile
index 48ab4b8f8..e85d9dd 100644
--- a/sound/core/Makefile
+++ b/sound/core/Makefile
@@ -37,7 +37,6 @@
obj-$(CONFIG_SND_HWDEP) += snd-hwdep.o
obj-$(CONFIG_SND_TIMER) += snd-timer.o
obj-$(CONFIG_SND_HRTIMER) += snd-hrtimer.o
-obj-$(CONFIG_SND_RTCTIMER) += snd-rtctimer.o
obj-$(CONFIG_SND_PCM) += snd-pcm.o
obj-$(CONFIG_SND_DMAENGINE_PCM) += snd-pcm-dmaengine.o
obj-$(CONFIG_SND_RAWMIDI) += snd-rawmidi.o
diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c
index 656d9a9..e2f2702 100644
--- a/sound/core/hrtimer.c
+++ b/sound/core/hrtimer.c
@@ -38,37 +38,53 @@
struct snd_hrtimer {
struct snd_timer *timer;
struct hrtimer hrt;
- atomic_t running;
+ bool in_callback;
};
static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
{
struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt);
struct snd_timer *t = stime->timer;
- unsigned long oruns;
+ ktime_t delta;
+ unsigned long ticks;
+ enum hrtimer_restart ret = HRTIMER_NORESTART;
- if (!atomic_read(&stime->running))
- return HRTIMER_NORESTART;
+ spin_lock(&t->lock);
+ if (!t->running)
+ goto out; /* fast path */
+ stime->in_callback = true;
+ ticks = t->sticks;
+ spin_unlock(&t->lock);
- oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
- snd_timer_interrupt(stime->timer, t->sticks * oruns);
+ /* calculate the drift */
+ delta = ktime_sub(hrt->base->get_time(), hrtimer_get_expires(hrt));
+ if (delta.tv64 > 0)
+ ticks += ktime_divns(delta, ticks * resolution);
- if (!atomic_read(&stime->running))
- return HRTIMER_NORESTART;
- return HRTIMER_RESTART;
+ snd_timer_interrupt(stime->timer, ticks);
+
+ spin_lock(&t->lock);
+ if (t->running) {
+ hrtimer_add_expires_ns(hrt, t->sticks * resolution);
+ ret = HRTIMER_RESTART;
+ }
+
+ stime->in_callback = false;
+ out:
+ spin_unlock(&t->lock);
+ return ret;
}
static int snd_hrtimer_open(struct snd_timer *t)
{
struct snd_hrtimer *stime;
- stime = kmalloc(sizeof(*stime), GFP_KERNEL);
+ stime = kzalloc(sizeof(*stime), GFP_KERNEL);
if (!stime)
return -ENOMEM;
hrtimer_init(&stime->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
stime->timer = t;
stime->hrt.function = snd_hrtimer_callback;
- atomic_set(&stime->running, 0);
t->private_data = stime;
return 0;
}
@@ -78,6 +94,11 @@
struct snd_hrtimer *stime = t->private_data;
if (stime) {
+ spin_lock_irq(&t->lock);
+ t->running = 0; /* just to be sure */
+ stime->in_callback = 1; /* skip start/stop */
+ spin_unlock_irq(&t->lock);
+
hrtimer_cancel(&stime->hrt);
kfree(stime);
t->private_data = NULL;
@@ -89,18 +110,19 @@
{
struct snd_hrtimer *stime = t->private_data;
- atomic_set(&stime->running, 0);
- hrtimer_try_to_cancel(&stime->hrt);
+ if (stime->in_callback)
+ return 0;
hrtimer_start(&stime->hrt, ns_to_ktime(t->sticks * resolution),
HRTIMER_MODE_REL);
- atomic_set(&stime->running, 1);
return 0;
}
static int snd_hrtimer_stop(struct snd_timer *t)
{
struct snd_hrtimer *stime = t->private_data;
- atomic_set(&stime->running, 0);
+
+ if (stime->in_callback)
+ return 0;
hrtimer_try_to_cancel(&stime->hrt);
return 0;
}
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 3a9b66c..0aca397 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1886,8 +1886,8 @@
snd_timer_interrupt(substream->timer, 1);
#endif
_end:
- snd_pcm_stream_unlock_irqrestore(substream, flags);
kill_fasync(&runtime->fasync, SIGIO, POLL_IN);
+ snd_pcm_stream_unlock_irqrestore(substream, flags);
}
EXPORT_SYMBOL(snd_pcm_period_elapsed);
diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c
deleted file mode 100644
index f3420d1..0000000
--- a/sound/core/rtctimer.c
+++ /dev/null
@@ -1,187 +0,0 @@
-/*
- * RTC based high-frequency timer
- *
- * Copyright (C) 2000 Takashi Iwai
- * based on rtctimer.c by Steve Ratcliffe
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
-
-#include <linux/init.h>
-#include <linux/interrupt.h>
-#include <linux/module.h>
-#include <linux/log2.h>
-#include <sound/core.h>
-#include <sound/timer.h>
-
-#if IS_ENABLED(CONFIG_RTC)
-
-#include <linux/mc146818rtc.h>
-
-#define RTC_FREQ 1024 /* default frequency */
-#define NANO_SEC 1000000000L /* 10^9 in sec */
-
-/*
- * prototypes
- */
-static int rtctimer_open(struct snd_timer *t);
-static int rtctimer_close(struct snd_timer *t);
-static int rtctimer_start(struct snd_timer *t);
-static int rtctimer_stop(struct snd_timer *t);
-
-
-/*
- * The hardware dependent description for this timer.
- */
-static struct snd_timer_hardware rtc_hw = {
- .flags = SNDRV_TIMER_HW_AUTO |
- SNDRV_TIMER_HW_FIRST |
- SNDRV_TIMER_HW_TASKLET,
- .ticks = 100000000L, /* FIXME: XXX */
- .open = rtctimer_open,
- .close = rtctimer_close,
- .start = rtctimer_start,
- .stop = rtctimer_stop,
-};
-
-static int rtctimer_freq = RTC_FREQ; /* frequency */
-static struct snd_timer *rtctimer;
-static struct tasklet_struct rtc_tasklet;
-static rtc_task_t rtc_task;
-
-
-static int
-rtctimer_open(struct snd_timer *t)
-{
- int err;
-
- err = rtc_register(&rtc_task);
- if (err < 0)
- return err;
- t->private_data = &rtc_task;
- return 0;
-}
-
-static int
-rtctimer_close(struct snd_timer *t)
-{
- rtc_task_t *rtc = t->private_data;
- if (rtc) {
- rtc_unregister(rtc);
- tasklet_kill(&rtc_tasklet);
- t->private_data = NULL;
- }
- return 0;
-}
-
-static int
-rtctimer_start(struct snd_timer *timer)
-{
- rtc_task_t *rtc = timer->private_data;
- if (snd_BUG_ON(!rtc))
- return -EINVAL;
- rtc_control(rtc, RTC_IRQP_SET, rtctimer_freq);
- rtc_control(rtc, RTC_PIE_ON, 0);
- return 0;
-}
-
-static int
-rtctimer_stop(struct snd_timer *timer)
-{
- rtc_task_t *rtc = timer->private_data;
- if (snd_BUG_ON(!rtc))
- return -EINVAL;
- rtc_control(rtc, RTC_PIE_OFF, 0);
- return 0;
-}
-
-static void rtctimer_tasklet(unsigned long data)
-{
- snd_timer_interrupt((struct snd_timer *)data, 1);
-}
-
-/*
- * interrupt
- */
-static void rtctimer_interrupt(void *private_data)
-{
- tasklet_schedule(private_data);
-}
-
-
-/*
- * ENTRY functions
- */
-static int __init rtctimer_init(void)
-{
- int err;
- struct snd_timer *timer;
-
- if (rtctimer_freq < 2 || rtctimer_freq > 8192 ||
- !is_power_of_2(rtctimer_freq)) {
- pr_err("ALSA: rtctimer: invalid frequency %d\n", rtctimer_freq);
- return -EINVAL;
- }
-
- /* Create a new timer and set up the fields */
- err = snd_timer_global_new("rtc", SNDRV_TIMER_GLOBAL_RTC, &timer);
- if (err < 0)
- return err;
-
- timer->module = THIS_MODULE;
- strcpy(timer->name, "RTC timer");
- timer->hw = rtc_hw;
- timer->hw.resolution = NANO_SEC / rtctimer_freq;
-
- tasklet_init(&rtc_tasklet, rtctimer_tasklet, (unsigned long)timer);
-
- /* set up RTC callback */
- rtc_task.func = rtctimer_interrupt;
- rtc_task.private_data = &rtc_tasklet;
-
- err = snd_timer_global_register(timer);
- if (err < 0) {
- snd_timer_global_free(timer);
- return err;
- }
- rtctimer = timer; /* remember this */
-
- return 0;
-}
-
-static void __exit rtctimer_exit(void)
-{
- if (rtctimer) {
- snd_timer_global_free(rtctimer);
- rtctimer = NULL;
- }
-}
-
-
-/*
- * exported stuff
- */
-module_init(rtctimer_init)
-module_exit(rtctimer_exit)
-
-module_param(rtctimer_freq, int, 0444);
-MODULE_PARM_DESC(rtctimer_freq, "timer frequency in Hz");
-
-MODULE_LICENSE("GPL");
-
-MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_RTC));
-
-#endif /* IS_ENABLED(CONFIG_RTC) */
diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c
index 7e0aabb..639544b 100644
--- a/sound/core/seq/seq.c
+++ b/sound/core/seq/seq.c
@@ -47,8 +47,6 @@
int seq_default_timer_device =
#ifdef CONFIG_SND_SEQ_HRTIMER_DEFAULT
SNDRV_TIMER_GLOBAL_HRTIMER
-#elif defined(CONFIG_SND_SEQ_RTCTIMER_DEFAULT)
- SNDRV_TIMER_GLOBAL_RTC
#else
SNDRV_TIMER_GLOBAL_SYSTEM
#endif
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 6469bed..0cfc028 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -37,8 +37,6 @@
#if IS_ENABLED(CONFIG_SND_HRTIMER)
#define DEFAULT_TIMER_LIMIT 4
-#elif IS_ENABLED(CONFIG_SND_RTCTIMER)
-#define DEFAULT_TIMER_LIMIT 2
#else
#define DEFAULT_TIMER_LIMIT 1
#endif
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index 2a779c2..ab894ed 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -134,6 +134,7 @@
Say Y here to include support for TASCAM.
* FW-1884
* FW-1082
+ * FW-1804
To compile this driver as a module, choose M here: the module
will be called snd-firewire-tascam.
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index ed29026..4484242 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -102,6 +102,10 @@
*/
void amdtp_stream_destroy(struct amdtp_stream *s)
{
+ /* Not initialized. */
+ if (s->protocol == NULL)
+ return;
+
WARN_ON(amdtp_stream_running(s));
kfree(s->protocol);
mutex_destroy(&s->mutex);
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 3e4e075..f7e2cbd 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -67,7 +67,7 @@
#define MODEL_MAUDIO_PROJECTMIX 0x00010091
static int
-name_device(struct snd_bebob *bebob, unsigned int vendor_id)
+name_device(struct snd_bebob *bebob)
{
struct fw_device *fw_dev = fw_parent_device(bebob->unit);
char vendor[24] = {0};
@@ -126,6 +126,17 @@
return err;
}
+static void bebob_free(struct snd_bebob *bebob)
+{
+ snd_bebob_stream_destroy_duplex(bebob);
+ fw_unit_put(bebob->unit);
+
+ kfree(bebob->maudio_special_quirk);
+
+ mutex_destroy(&bebob->mutex);
+ kfree(bebob);
+}
+
/*
* This module releases the FireWire unit data after all ALSA character devices
* are released by applications. This is for releasing stream data or finishing
@@ -137,18 +148,11 @@
{
struct snd_bebob *bebob = card->private_data;
- snd_bebob_stream_destroy_duplex(bebob);
- fw_unit_put(bebob->unit);
+ mutex_lock(&devices_mutex);
+ clear_bit(bebob->card_index, devices_used);
+ mutex_unlock(&devices_mutex);
- kfree(bebob->maudio_special_quirk);
-
- if (bebob->card_index >= 0) {
- mutex_lock(&devices_mutex);
- clear_bit(bebob->card_index, devices_used);
- mutex_unlock(&devices_mutex);
- }
-
- mutex_destroy(&bebob->mutex);
+ bebob_free(card->private_data);
}
static const struct snd_bebob_spec *
@@ -176,16 +180,17 @@
return strncmp(name, "FW Audiophile Bootloader", 15) != 0;
}
-static int
-bebob_probe(struct fw_unit *unit,
- const struct ieee1394_device_id *entry)
+static void
+do_registration(struct work_struct *work)
{
- struct snd_card *card;
- struct snd_bebob *bebob;
- const struct snd_bebob_spec *spec;
+ struct snd_bebob *bebob =
+ container_of(work, struct snd_bebob, dwork.work);
unsigned int card_index;
int err;
+ if (bebob->registered)
+ return;
+
mutex_lock(&devices_mutex);
for (card_index = 0; card_index < SNDRV_CARDS; card_index++) {
@@ -193,64 +198,39 @@
break;
}
if (card_index >= SNDRV_CARDS) {
- err = -ENOENT;
- goto end;
+ mutex_unlock(&devices_mutex);
+ return;
}
- if ((entry->vendor_id == VEN_FOCUSRITE) &&
- (entry->model_id == MODEL_FOCUSRITE_SAFFIRE_BOTH))
- spec = get_saffire_spec(unit);
- else if ((entry->vendor_id == VEN_MAUDIO1) &&
- (entry->model_id == MODEL_MAUDIO_AUDIOPHILE_BOTH) &&
- !check_audiophile_booted(unit))
- spec = NULL;
- else
- spec = (const struct snd_bebob_spec *)entry->driver_data;
-
- if (spec == NULL) {
- if ((entry->vendor_id == VEN_MAUDIO1) ||
- (entry->vendor_id == VEN_MAUDIO2))
- err = snd_bebob_maudio_load_firmware(unit);
- else
- err = -ENOSYS;
- goto end;
+ err = snd_card_new(&bebob->unit->device, index[card_index],
+ id[card_index], THIS_MODULE, 0, &bebob->card);
+ if (err < 0) {
+ mutex_unlock(&devices_mutex);
+ return;
}
- err = snd_card_new(&unit->device, index[card_index], id[card_index],
- THIS_MODULE, sizeof(struct snd_bebob), &card);
- if (err < 0)
- goto end;
- bebob = card->private_data;
- bebob->card_index = card_index;
- set_bit(card_index, devices_used);
- card->private_free = bebob_card_free;
-
- bebob->card = card;
- bebob->unit = fw_unit_get(unit);
- bebob->spec = spec;
- mutex_init(&bebob->mutex);
- spin_lock_init(&bebob->lock);
- init_waitqueue_head(&bebob->hwdep_wait);
-
- err = name_device(bebob, entry->vendor_id);
+ err = name_device(bebob);
if (err < 0)
goto error;
- if ((entry->vendor_id == VEN_MAUDIO1) &&
- (entry->model_id == MODEL_MAUDIO_FW1814))
- err = snd_bebob_maudio_special_discover(bebob, true);
- else if ((entry->vendor_id == VEN_MAUDIO1) &&
- (entry->model_id == MODEL_MAUDIO_PROJECTMIX))
- err = snd_bebob_maudio_special_discover(bebob, false);
- else
+ if (bebob->spec == &maudio_special_spec) {
+ if (bebob->entry->model_id == MODEL_MAUDIO_FW1814)
+ err = snd_bebob_maudio_special_discover(bebob, true);
+ else
+ err = snd_bebob_maudio_special_discover(bebob, false);
+ } else {
err = snd_bebob_stream_discover(bebob);
+ }
+ if (err < 0)
+ goto error;
+
+ err = snd_bebob_stream_init_duplex(bebob);
if (err < 0)
goto error;
snd_bebob_proc_init(bebob);
- if ((bebob->midi_input_ports > 0) ||
- (bebob->midi_output_ports > 0)) {
+ if (bebob->midi_input_ports > 0 || bebob->midi_output_ports > 0) {
err = snd_bebob_create_midi_devices(bebob);
if (err < 0)
goto error;
@@ -264,16 +244,75 @@
if (err < 0)
goto error;
- err = snd_bebob_stream_init_duplex(bebob);
+ err = snd_card_register(bebob->card);
if (err < 0)
goto error;
- if (!bebob->maudio_special_quirk) {
- err = snd_card_register(card);
- if (err < 0) {
- snd_bebob_stream_destroy_duplex(bebob);
- goto error;
- }
+ set_bit(card_index, devices_used);
+ mutex_unlock(&devices_mutex);
+
+ /*
+ * After registered, bebob instance can be released corresponding to
+ * releasing the sound card instance.
+ */
+ bebob->card->private_free = bebob_card_free;
+ bebob->card->private_data = bebob;
+ bebob->registered = true;
+
+ return;
+error:
+ mutex_unlock(&devices_mutex);
+ snd_bebob_stream_destroy_duplex(bebob);
+ snd_card_free(bebob->card);
+ dev_info(&bebob->unit->device,
+ "Sound card registration failed: %d\n", err);
+}
+
+static int
+bebob_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry)
+{
+ struct snd_bebob *bebob;
+ const struct snd_bebob_spec *spec;
+
+ if (entry->vendor_id == VEN_FOCUSRITE &&
+ entry->model_id == MODEL_FOCUSRITE_SAFFIRE_BOTH)
+ spec = get_saffire_spec(unit);
+ else if (entry->vendor_id == VEN_MAUDIO1 &&
+ entry->model_id == MODEL_MAUDIO_AUDIOPHILE_BOTH &&
+ !check_audiophile_booted(unit))
+ spec = NULL;
+ else
+ spec = (const struct snd_bebob_spec *)entry->driver_data;
+
+ if (spec == NULL) {
+ if (entry->vendor_id == VEN_MAUDIO1 ||
+ entry->vendor_id == VEN_MAUDIO2)
+ return snd_bebob_maudio_load_firmware(unit);
+ else
+ return -ENODEV;
+ }
+
+ /* Allocate this independent of sound card instance. */
+ bebob = kzalloc(sizeof(struct snd_bebob), GFP_KERNEL);
+ if (bebob == NULL)
+ return -ENOMEM;
+
+ bebob->unit = fw_unit_get(unit);
+ bebob->entry = entry;
+ bebob->spec = spec;
+ dev_set_drvdata(&unit->device, bebob);
+
+ mutex_init(&bebob->mutex);
+ spin_lock_init(&bebob->lock);
+ init_waitqueue_head(&bebob->hwdep_wait);
+
+ /* Allocate and register this sound card later. */
+ INIT_DEFERRABLE_WORK(&bebob->dwork, do_registration);
+
+ if (entry->vendor_id != VEN_MAUDIO1 ||
+ (entry->model_id != MODEL_MAUDIO_FW1814 &&
+ entry->model_id != MODEL_MAUDIO_PROJECTMIX)) {
+ snd_fw_schedule_registration(unit, &bebob->dwork);
} else {
/*
* This is a workaround. This bus reset seems to have an effect
@@ -285,19 +324,11 @@
* signals from dbus and starts I/Os. To avoid I/Os till the
* future bus reset, registration is done in next update().
*/
- bebob->deferred_registration = true;
fw_schedule_bus_reset(fw_parent_device(bebob->unit)->card,
false, true);
}
- dev_set_drvdata(&unit->device, bebob);
-end:
- mutex_unlock(&devices_mutex);
- return err;
-error:
- mutex_unlock(&devices_mutex);
- snd_card_free(card);
- return err;
+ return 0;
}
/*
@@ -324,15 +355,11 @@
if (bebob == NULL)
return;
- fcp_bus_reset(bebob->unit);
-
- if (bebob->deferred_registration) {
- if (snd_card_register(bebob->card) < 0) {
- snd_bebob_stream_destroy_duplex(bebob);
- snd_card_free(bebob->card);
- }
- bebob->deferred_registration = false;
- }
+ /* Postpone a workqueue for deferred registration. */
+ if (!bebob->registered)
+ snd_fw_schedule_registration(unit, &bebob->dwork);
+ else
+ fcp_bus_reset(bebob->unit);
}
static void bebob_remove(struct fw_unit *unit)
@@ -342,8 +369,20 @@
if (bebob == NULL)
return;
- /* No need to wait for releasing card object in this context. */
- snd_card_free_when_closed(bebob->card);
+ /*
+ * Confirm to stop the work for registration before the sound card is
+ * going to be released. The work is not scheduled again because bus
+ * reset handler is not called anymore.
+ */
+ cancel_delayed_work_sync(&bebob->dwork);
+
+ if (bebob->registered) {
+ /* No need to wait for releasing card object in this context. */
+ snd_card_free_when_closed(bebob->card);
+ } else {
+ /* Don't forget this case. */
+ bebob_free(bebob);
+ }
}
static const struct snd_bebob_rate_spec normal_rate_spec = {
diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h
index b50bb33d..2a442a7 100644
--- a/sound/firewire/bebob/bebob.h
+++ b/sound/firewire/bebob/bebob.h
@@ -83,6 +83,10 @@
struct mutex mutex;
spinlock_t lock;
+ bool registered;
+ struct delayed_work dwork;
+
+ const struct ieee1394_device_id *entry;
const struct snd_bebob_spec *spec;
unsigned int midi_input_ports;
@@ -111,7 +115,6 @@
/* for M-Audio special devices */
void *maudio_special_quirk;
- bool deferred_registration;
/* For BeBoB version quirk. */
unsigned int version;
diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c
index 8b64aef..96fe68f4 100644
--- a/sound/firewire/dice/dice.c
+++ b/sound/firewire/dice/dice.c
@@ -20,8 +20,6 @@
#define WEISS_CATEGORY_ID 0x00
#define LOUD_CATEGORY_ID 0x10
-#define PROBE_DELAY_MS (2 * MSEC_PER_SEC)
-
/*
* Some models support several isochronous channels, while these streams are not
* always available. In this case, add the model name to this list.
@@ -201,6 +199,10 @@
dice_card_strings(dice);
+ err = snd_dice_stream_init_duplex(dice);
+ if (err < 0)
+ goto error;
+
snd_dice_create_proc(dice);
err = snd_dice_create_pcm(dice);
@@ -229,28 +231,14 @@
return;
error:
+ snd_dice_stream_destroy_duplex(dice);
snd_dice_transaction_destroy(dice);
+ snd_dice_stream_destroy_duplex(dice);
snd_card_free(dice->card);
dev_info(&dice->unit->device,
"Sound card registration failed: %d\n", err);
}
-static void schedule_registration(struct snd_dice *dice)
-{
- struct fw_card *fw_card = fw_parent_device(dice->unit)->card;
- u64 now, delay;
-
- now = get_jiffies_64();
- delay = fw_card->reset_jiffies + msecs_to_jiffies(PROBE_DELAY_MS);
-
- if (time_after64(delay, now))
- delay -= now;
- else
- delay = 0;
-
- mod_delayed_work(system_wq, &dice->dwork, delay);
-}
-
static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id)
{
struct snd_dice *dice;
@@ -273,15 +261,9 @@
init_completion(&dice->clock_accepted);
init_waitqueue_head(&dice->hwdep_wait);
- err = snd_dice_stream_init_duplex(dice);
- if (err < 0) {
- dice_free(dice);
- return err;
- }
-
/* Allocate and register this sound card later. */
INIT_DEFERRABLE_WORK(&dice->dwork, do_registration);
- schedule_registration(dice);
+ snd_fw_schedule_registration(unit, &dice->dwork);
return 0;
}
@@ -312,7 +294,7 @@
/* Postpone a workqueue for deferred registration. */
if (!dice->registered)
- schedule_registration(dice);
+ snd_fw_schedule_registration(unit, &dice->dwork);
/* The handler address register becomes initialized. */
snd_dice_transaction_reinit(dice);
diff --git a/sound/firewire/digi00x/digi00x-transaction.c b/sound/firewire/digi00x/digi00x-transaction.c
index 554324d..735d356 100644
--- a/sound/firewire/digi00x/digi00x-transaction.c
+++ b/sound/firewire/digi00x/digi00x-transaction.c
@@ -126,12 +126,17 @@
return err;
error:
fw_core_remove_address_handler(&dg00x->async_handler);
- dg00x->async_handler.address_callback = NULL;
+ dg00x->async_handler.callback_data = NULL;
return err;
}
void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x)
{
+ if (dg00x->async_handler.callback_data == NULL)
+ return;
+
snd_fw_async_midi_port_destroy(&dg00x->out_control);
fw_core_remove_address_handler(&dg00x->async_handler);
+
+ dg00x->async_handler.callback_data = NULL;
}
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
index 1f33b7a..cc4776c 100644
--- a/sound/firewire/digi00x/digi00x.c
+++ b/sound/firewire/digi00x/digi00x.c
@@ -40,10 +40,8 @@
return 0;
}
-static void dg00x_card_free(struct snd_card *card)
+static void dg00x_free(struct snd_dg00x *dg00x)
{
- struct snd_dg00x *dg00x = card->private_data;
-
snd_dg00x_stream_destroy_duplex(dg00x);
snd_dg00x_transaction_unregister(dg00x);
@@ -52,28 +50,24 @@
mutex_destroy(&dg00x->mutex);
}
-static int snd_dg00x_probe(struct fw_unit *unit,
- const struct ieee1394_device_id *entry)
+static void dg00x_card_free(struct snd_card *card)
{
- struct snd_card *card;
- struct snd_dg00x *dg00x;
+ dg00x_free(card->private_data);
+}
+
+static void do_registration(struct work_struct *work)
+{
+ struct snd_dg00x *dg00x =
+ container_of(work, struct snd_dg00x, dwork.work);
int err;
- /* create card */
- err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE,
- sizeof(struct snd_dg00x), &card);
+ if (dg00x->registered)
+ return;
+
+ err = snd_card_new(&dg00x->unit->device, -1, NULL, THIS_MODULE, 0,
+ &dg00x->card);
if (err < 0)
- return err;
- card->private_free = dg00x_card_free;
-
- /* initialize myself */
- dg00x = card->private_data;
- dg00x->card = card;
- dg00x->unit = fw_unit_get(unit);
-
- mutex_init(&dg00x->mutex);
- spin_lock_init(&dg00x->lock);
- init_waitqueue_head(&dg00x->hwdep_wait);
+ return;
err = name_card(dg00x);
if (err < 0)
@@ -101,35 +95,86 @@
if (err < 0)
goto error;
- err = snd_card_register(card);
+ err = snd_card_register(dg00x->card);
if (err < 0)
goto error;
+ dg00x->card->private_free = dg00x_card_free;
+ dg00x->card->private_data = dg00x;
+ dg00x->registered = true;
+
+ return;
+error:
+ snd_dg00x_transaction_unregister(dg00x);
+ snd_dg00x_stream_destroy_duplex(dg00x);
+ snd_card_free(dg00x->card);
+ dev_info(&dg00x->unit->device,
+ "Sound card registration failed: %d\n", err);
+}
+
+static int snd_dg00x_probe(struct fw_unit *unit,
+ const struct ieee1394_device_id *entry)
+{
+ struct snd_dg00x *dg00x;
+
+ /* Allocate this independent of sound card instance. */
+ dg00x = kzalloc(sizeof(struct snd_dg00x), GFP_KERNEL);
+ if (dg00x == NULL)
+ return -ENOMEM;
+
+ dg00x->unit = fw_unit_get(unit);
dev_set_drvdata(&unit->device, dg00x);
- return err;
-error:
- snd_card_free(card);
- return err;
+ mutex_init(&dg00x->mutex);
+ spin_lock_init(&dg00x->lock);
+ init_waitqueue_head(&dg00x->hwdep_wait);
+
+ /* Allocate and register this sound card later. */
+ INIT_DEFERRABLE_WORK(&dg00x->dwork, do_registration);
+ snd_fw_schedule_registration(unit, &dg00x->dwork);
+
+ return 0;
}
static void snd_dg00x_update(struct fw_unit *unit)
{
struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device);
+ /* Postpone a workqueue for deferred registration. */
+ if (!dg00x->registered)
+ snd_fw_schedule_registration(unit, &dg00x->dwork);
+
snd_dg00x_transaction_reregister(dg00x);
- mutex_lock(&dg00x->mutex);
- snd_dg00x_stream_update_duplex(dg00x);
- mutex_unlock(&dg00x->mutex);
+ /*
+ * After registration, userspace can start packet streaming, then this
+ * code block works fine.
+ */
+ if (dg00x->registered) {
+ mutex_lock(&dg00x->mutex);
+ snd_dg00x_stream_update_duplex(dg00x);
+ mutex_unlock(&dg00x->mutex);
+ }
}
static void snd_dg00x_remove(struct fw_unit *unit)
{
struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device);
- /* No need to wait for releasing card object in this context. */
- snd_card_free_when_closed(dg00x->card);
+ /*
+ * Confirm to stop the work for registration before the sound card is
+ * going to be released. The work is not scheduled again because bus
+ * reset handler is not called anymore.
+ */
+ cancel_delayed_work_sync(&dg00x->dwork);
+
+ if (dg00x->registered) {
+ /* No need to wait for releasing card object in this context. */
+ snd_card_free_when_closed(dg00x->card);
+ } else {
+ /* Don't forget this case. */
+ dg00x_free(dg00x);
+ }
}
static const struct ieee1394_device_id snd_dg00x_id_table[] = {
diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h
index 907e739..2cd465c0 100644
--- a/sound/firewire/digi00x/digi00x.h
+++ b/sound/firewire/digi00x/digi00x.h
@@ -37,6 +37,9 @@
struct mutex mutex;
spinlock_t lock;
+ bool registered;
+ struct delayed_work dwork;
+
struct amdtp_stream tx_stream;
struct fw_iso_resources tx_resources;
diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c
index 8f27b67..71a0613 100644
--- a/sound/firewire/fireworks/fireworks.c
+++ b/sound/firewire/fireworks/fireworks.c
@@ -168,11 +168,34 @@
sizeof(struct snd_efw_phys_grp) * hwinfo->phys_in_grp_count);
memcpy(&efw->phys_out_grps, hwinfo->phys_out_grps,
sizeof(struct snd_efw_phys_grp) * hwinfo->phys_out_grp_count);
+
+ /* AudioFire8 (since 2009) and AudioFirePre8 */
+ if (hwinfo->type == MODEL_ECHO_AUDIOFIRE_9)
+ efw->is_af9 = true;
+ /* These models uses the same firmware. */
+ if (hwinfo->type == MODEL_ECHO_AUDIOFIRE_2 ||
+ hwinfo->type == MODEL_ECHO_AUDIOFIRE_4 ||
+ hwinfo->type == MODEL_ECHO_AUDIOFIRE_9 ||
+ hwinfo->type == MODEL_GIBSON_RIP ||
+ hwinfo->type == MODEL_GIBSON_GOLDTOP)
+ efw->is_fireworks3 = true;
end:
kfree(hwinfo);
return err;
}
+static void efw_free(struct snd_efw *efw)
+{
+ snd_efw_stream_destroy_duplex(efw);
+ snd_efw_transaction_remove_instance(efw);
+ fw_unit_put(efw->unit);
+
+ kfree(efw->resp_buf);
+
+ mutex_destroy(&efw->mutex);
+ kfree(efw);
+}
+
/*
* This module releases the FireWire unit data after all ALSA character devices
* are released by applications. This is for releasing stream data or finishing
@@ -184,28 +207,24 @@
{
struct snd_efw *efw = card->private_data;
- snd_efw_stream_destroy_duplex(efw);
- snd_efw_transaction_remove_instance(efw);
- fw_unit_put(efw->unit);
-
- kfree(efw->resp_buf);
-
if (efw->card_index >= 0) {
mutex_lock(&devices_mutex);
clear_bit(efw->card_index, devices_used);
mutex_unlock(&devices_mutex);
}
- mutex_destroy(&efw->mutex);
+ efw_free(card->private_data);
}
-static int
-efw_probe(struct fw_unit *unit,
- const struct ieee1394_device_id *entry)
+static void
+do_registration(struct work_struct *work)
{
- struct snd_card *card;
- struct snd_efw *efw;
- int card_index, err;
+ struct snd_efw *efw = container_of(work, struct snd_efw, dwork.work);
+ unsigned int card_index;
+ int err;
+
+ if (efw->registered)
+ return;
mutex_lock(&devices_mutex);
@@ -215,24 +234,16 @@
break;
}
if (card_index >= SNDRV_CARDS) {
- err = -ENOENT;
- goto end;
+ mutex_unlock(&devices_mutex);
+ return;
}
- err = snd_card_new(&unit->device, index[card_index], id[card_index],
- THIS_MODULE, sizeof(struct snd_efw), &card);
- if (err < 0)
- goto end;
- efw = card->private_data;
- efw->card_index = card_index;
- set_bit(card_index, devices_used);
- card->private_free = efw_card_free;
-
- efw->card = card;
- efw->unit = fw_unit_get(unit);
- mutex_init(&efw->mutex);
- spin_lock_init(&efw->lock);
- init_waitqueue_head(&efw->hwdep_wait);
+ err = snd_card_new(&efw->unit->device, index[card_index],
+ id[card_index], THIS_MODULE, 0, &efw->card);
+ if (err < 0) {
+ mutex_unlock(&devices_mutex);
+ return;
+ }
/* prepare response buffer */
snd_efw_resp_buf_size = clamp(snd_efw_resp_buf_size,
@@ -248,16 +259,10 @@
err = get_hardware_info(efw);
if (err < 0)
goto error;
- /* AudioFire8 (since 2009) and AudioFirePre8 */
- if (entry->model_id == MODEL_ECHO_AUDIOFIRE_9)
- efw->is_af9 = true;
- /* These models uses the same firmware. */
- if (entry->model_id == MODEL_ECHO_AUDIOFIRE_2 ||
- entry->model_id == MODEL_ECHO_AUDIOFIRE_4 ||
- entry->model_id == MODEL_ECHO_AUDIOFIRE_9 ||
- entry->model_id == MODEL_GIBSON_RIP ||
- entry->model_id == MODEL_GIBSON_GOLDTOP)
- efw->is_fireworks3 = true;
+
+ err = snd_efw_stream_init_duplex(efw);
+ if (err < 0)
+ goto error;
snd_efw_proc_init(efw);
@@ -275,44 +280,93 @@
if (err < 0)
goto error;
- err = snd_efw_stream_init_duplex(efw);
+ err = snd_card_register(efw->card);
if (err < 0)
goto error;
- err = snd_card_register(card);
- if (err < 0) {
- snd_efw_stream_destroy_duplex(efw);
- goto error;
- }
+ set_bit(card_index, devices_used);
+ mutex_unlock(&devices_mutex);
- dev_set_drvdata(&unit->device, efw);
-end:
- mutex_unlock(&devices_mutex);
- return err;
+ /*
+ * After registered, efw instance can be released corresponding to
+ * releasing the sound card instance.
+ */
+ efw->card->private_free = efw_card_free;
+ efw->card->private_data = efw;
+ efw->registered = true;
+
+ return;
error:
- snd_efw_transaction_remove_instance(efw);
mutex_unlock(&devices_mutex);
- snd_card_free(card);
- return err;
+ snd_efw_transaction_remove_instance(efw);
+ snd_efw_stream_destroy_duplex(efw);
+ snd_card_free(efw->card);
+ dev_info(&efw->unit->device,
+ "Sound card registration failed: %d\n", err);
+}
+
+static int
+efw_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry)
+{
+ struct snd_efw *efw;
+
+ efw = kzalloc(sizeof(struct snd_efw), GFP_KERNEL);
+ if (efw == NULL)
+ return -ENOMEM;
+
+ efw->unit = fw_unit_get(unit);
+ dev_set_drvdata(&unit->device, efw);
+
+ mutex_init(&efw->mutex);
+ spin_lock_init(&efw->lock);
+ init_waitqueue_head(&efw->hwdep_wait);
+
+ /* Allocate and register this sound card later. */
+ INIT_DEFERRABLE_WORK(&efw->dwork, do_registration);
+ snd_fw_schedule_registration(unit, &efw->dwork);
+
+ return 0;
}
static void efw_update(struct fw_unit *unit)
{
struct snd_efw *efw = dev_get_drvdata(&unit->device);
+ /* Postpone a workqueue for deferred registration. */
+ if (!efw->registered)
+ snd_fw_schedule_registration(unit, &efw->dwork);
+
snd_efw_transaction_bus_reset(efw->unit);
- mutex_lock(&efw->mutex);
- snd_efw_stream_update_duplex(efw);
- mutex_unlock(&efw->mutex);
+ /*
+ * After registration, userspace can start packet streaming, then this
+ * code block works fine.
+ */
+ if (efw->registered) {
+ mutex_lock(&efw->mutex);
+ snd_efw_stream_update_duplex(efw);
+ mutex_unlock(&efw->mutex);
+ }
}
static void efw_remove(struct fw_unit *unit)
{
struct snd_efw *efw = dev_get_drvdata(&unit->device);
- /* No need to wait for releasing card object in this context. */
- snd_card_free_when_closed(efw->card);
+ /*
+ * Confirm to stop the work for registration before the sound card is
+ * going to be released. The work is not scheduled again because bus
+ * reset handler is not called anymore.
+ */
+ cancel_delayed_work_sync(&efw->dwork);
+
+ if (efw->registered) {
+ /* No need to wait for releasing card object in this context. */
+ snd_card_free_when_closed(efw->card);
+ } else {
+ /* Don't forget this case. */
+ efw_free(efw);
+ }
}
static const struct ieee1394_device_id efw_id_table[] = {
diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h
index 96c4e0c..471c772 100644
--- a/sound/firewire/fireworks/fireworks.h
+++ b/sound/firewire/fireworks/fireworks.h
@@ -65,6 +65,9 @@
struct mutex mutex;
spinlock_t lock;
+ bool registered;
+ struct delayed_work dwork;
+
/* for transaction */
u32 seqnum;
bool resp_addr_changable;
diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c
index f80aafa..ca4dfcf 100644
--- a/sound/firewire/lib.c
+++ b/sound/firewire/lib.c
@@ -67,6 +67,38 @@
}
EXPORT_SYMBOL(snd_fw_transaction);
+#define PROBE_DELAY_MS (2 * MSEC_PER_SEC)
+
+/**
+ * snd_fw_schedule_registration - schedule work for sound card registration
+ * @unit: an instance for unit on IEEE 1394 bus
+ * @dwork: delayed work with callback function
+ *
+ * This function is not designed for general purposes. When new unit is
+ * connected to IEEE 1394 bus, the bus is under bus-reset state because of
+ * topological change. In this state, units tend to fail both of asynchronous
+ * and isochronous communication. To avoid this problem, this function is used
+ * to postpone sound card registration after the state. The callers must
+ * set up instance of delayed work in advance.
+ */
+void snd_fw_schedule_registration(struct fw_unit *unit,
+ struct delayed_work *dwork)
+{
+ u64 now, delay;
+
+ now = get_jiffies_64();
+ delay = fw_parent_device(unit)->card->reset_jiffies
+ + msecs_to_jiffies(PROBE_DELAY_MS);
+
+ if (time_after64(delay, now))
+ delay -= now;
+ else
+ delay = 0;
+
+ mod_delayed_work(system_wq, dwork, delay);
+}
+EXPORT_SYMBOL(snd_fw_schedule_registration);
+
static void async_midi_port_callback(struct fw_card *card, int rcode,
void *data, size_t length,
void *callback_data)
diff --git a/sound/firewire/lib.h b/sound/firewire/lib.h
index f3f6f84..f676931 100644
--- a/sound/firewire/lib.h
+++ b/sound/firewire/lib.h
@@ -22,6 +22,9 @@
return rcode == RCODE_TYPE_ERROR || rcode == RCODE_ADDRESS_ERROR;
}
+void snd_fw_schedule_registration(struct fw_unit *unit,
+ struct delayed_work *dwork);
+
struct snd_fw_async_midi_port;
typedef int (*snd_fw_async_midi_port_fill)(
struct snd_rawmidi_substream *substream,
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index abedc22..e629b88 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -118,15 +118,8 @@
return err;
}
-/*
- * This module releases the FireWire unit data after all ALSA character devices
- * are released by applications. This is for releasing stream data or finishing
- * transactions safely. Thus at returning from .remove(), this module still keep
- * references for the unit.
- */
-static void oxfw_card_free(struct snd_card *card)
+static void oxfw_free(struct snd_oxfw *oxfw)
{
- struct snd_oxfw *oxfw = card->private_data;
unsigned int i;
snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream);
@@ -144,6 +137,17 @@
mutex_destroy(&oxfw->mutex);
}
+/*
+ * This module releases the FireWire unit data after all ALSA character devices
+ * are released by applications. This is for releasing stream data or finishing
+ * transactions safely. Thus at returning from .remove(), this module still keep
+ * references for the unit.
+ */
+static void oxfw_card_free(struct snd_card *card)
+{
+ oxfw_free(card->private_data);
+}
+
static int detect_quirks(struct snd_oxfw *oxfw)
{
struct fw_device *fw_dev = fw_parent_device(oxfw->unit);
@@ -205,33 +209,18 @@
return 0;
}
-static int oxfw_probe(struct fw_unit *unit,
- const struct ieee1394_device_id *entry)
+static void do_registration(struct work_struct *work)
{
- struct snd_card *card;
- struct snd_oxfw *oxfw;
+ struct snd_oxfw *oxfw = container_of(work, struct snd_oxfw, dwork.work);
int err;
- if (entry->vendor_id == VENDOR_LOUD && !detect_loud_models(unit))
- return -ENODEV;
+ if (oxfw->registered)
+ return;
- err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE,
- sizeof(*oxfw), &card);
+ err = snd_card_new(&oxfw->unit->device, -1, NULL, THIS_MODULE, 0,
+ &oxfw->card);
if (err < 0)
- return err;
-
- card->private_free = oxfw_card_free;
- oxfw = card->private_data;
- oxfw->card = card;
- mutex_init(&oxfw->mutex);
- oxfw->unit = fw_unit_get(unit);
- oxfw->entry = entry;
- spin_lock_init(&oxfw->lock);
- init_waitqueue_head(&oxfw->hwdep_wait);
-
- err = snd_oxfw_stream_discover(oxfw);
- if (err < 0)
- goto error;
+ return;
err = name_card(oxfw);
if (err < 0)
@@ -241,6 +230,19 @@
if (err < 0)
goto error;
+ err = snd_oxfw_stream_discover(oxfw);
+ if (err < 0)
+ goto error;
+
+ err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->rx_stream);
+ if (err < 0)
+ goto error;
+ if (oxfw->has_output) {
+ err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->tx_stream);
+ if (err < 0)
+ goto error;
+ }
+
err = snd_oxfw_create_pcm(oxfw);
if (err < 0)
goto error;
@@ -255,54 +257,97 @@
if (err < 0)
goto error;
- err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->rx_stream);
+ err = snd_card_register(oxfw->card);
if (err < 0)
goto error;
- if (oxfw->has_output) {
- err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->tx_stream);
- if (err < 0)
- goto error;
- }
- err = snd_card_register(card);
- if (err < 0) {
- snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream);
- if (oxfw->has_output)
- snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream);
- goto error;
- }
+ /*
+ * After registered, oxfw instance can be released corresponding to
+ * releasing the sound card instance.
+ */
+ oxfw->card->private_free = oxfw_card_free;
+ oxfw->card->private_data = oxfw;
+ oxfw->registered = true;
+
+ return;
+error:
+ snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream);
+ if (oxfw->has_output)
+ snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream);
+ snd_card_free(oxfw->card);
+ dev_info(&oxfw->unit->device,
+ "Sound card registration failed: %d\n", err);
+}
+
+static int oxfw_probe(struct fw_unit *unit,
+ const struct ieee1394_device_id *entry)
+{
+ struct snd_oxfw *oxfw;
+
+ if (entry->vendor_id == VENDOR_LOUD && !detect_loud_models(unit))
+ return -ENODEV;
+
+ /* Allocate this independent of sound card instance. */
+ oxfw = kzalloc(sizeof(struct snd_oxfw), GFP_KERNEL);
+ if (oxfw == NULL)
+ return -ENOMEM;
+
+ oxfw->entry = entry;
+ oxfw->unit = fw_unit_get(unit);
dev_set_drvdata(&unit->device, oxfw);
+ mutex_init(&oxfw->mutex);
+ spin_lock_init(&oxfw->lock);
+ init_waitqueue_head(&oxfw->hwdep_wait);
+
+ /* Allocate and register this sound card later. */
+ INIT_DEFERRABLE_WORK(&oxfw->dwork, do_registration);
+ snd_fw_schedule_registration(unit, &oxfw->dwork);
+
return 0;
-error:
- snd_card_free(card);
- return err;
}
static void oxfw_bus_reset(struct fw_unit *unit)
{
struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device);
+ if (!oxfw->registered)
+ snd_fw_schedule_registration(unit, &oxfw->dwork);
+
fcp_bus_reset(oxfw->unit);
- mutex_lock(&oxfw->mutex);
+ if (oxfw->registered) {
+ mutex_lock(&oxfw->mutex);
- snd_oxfw_stream_update_simplex(oxfw, &oxfw->rx_stream);
- if (oxfw->has_output)
- snd_oxfw_stream_update_simplex(oxfw, &oxfw->tx_stream);
+ snd_oxfw_stream_update_simplex(oxfw, &oxfw->rx_stream);
+ if (oxfw->has_output)
+ snd_oxfw_stream_update_simplex(oxfw, &oxfw->tx_stream);
- mutex_unlock(&oxfw->mutex);
+ mutex_unlock(&oxfw->mutex);
- if (oxfw->entry->vendor_id == OUI_STANTON)
- snd_oxfw_scs1x_update(oxfw);
+ if (oxfw->entry->vendor_id == OUI_STANTON)
+ snd_oxfw_scs1x_update(oxfw);
+ }
}
static void oxfw_remove(struct fw_unit *unit)
{
struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device);
- /* No need to wait for releasing card object in this context. */
- snd_card_free_when_closed(oxfw->card);
+ /*
+ * Confirm to stop the work for registration before the sound card is
+ * going to be released. The work is not scheduled again because bus
+ * reset handler is not called anymore.
+ */
+ cancel_delayed_work_sync(&oxfw->dwork);
+
+ if (oxfw->registered) {
+ /* No need to wait for releasing card object in this context. */
+ snd_card_free_when_closed(oxfw->card);
+ } else {
+ /* Don't forget this case. */
+ oxfw_free(oxfw);
+ }
}
static const struct compat_info griffin_firewave = {
diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h
index 9beecc2..2047dcb 100644
--- a/sound/firewire/oxfw/oxfw.h
+++ b/sound/firewire/oxfw/oxfw.h
@@ -36,10 +36,12 @@
struct snd_oxfw {
struct snd_card *card;
struct fw_unit *unit;
- const struct device_info *device_info;
struct mutex mutex;
spinlock_t lock;
+ bool registered;
+ struct delayed_work dwork;
+
bool wrong_dbs;
bool has_output;
u8 *tx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES];
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
index e281c33..9dc93a7 100644
--- a/sound/firewire/tascam/tascam.c
+++ b/sound/firewire/tascam/tascam.c
@@ -85,10 +85,8 @@
return 0;
}
-static void tscm_card_free(struct snd_card *card)
+static void tscm_free(struct snd_tscm *tscm)
{
- struct snd_tscm *tscm = card->private_data;
-
snd_tscm_transaction_unregister(tscm);
snd_tscm_stream_destroy_duplex(tscm);
@@ -97,44 +95,36 @@
mutex_destroy(&tscm->mutex);
}
-static int snd_tscm_probe(struct fw_unit *unit,
- const struct ieee1394_device_id *entry)
+static void tscm_card_free(struct snd_card *card)
{
- struct snd_card *card;
- struct snd_tscm *tscm;
+ tscm_free(card->private_data);
+}
+
+static void do_registration(struct work_struct *work)
+{
+ struct snd_tscm *tscm = container_of(work, struct snd_tscm, dwork.work);
int err;
- /* create card */
- err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE,
- sizeof(struct snd_tscm), &card);
+ err = snd_card_new(&tscm->unit->device, -1, NULL, THIS_MODULE, 0,
+ &tscm->card);
if (err < 0)
- return err;
- card->private_free = tscm_card_free;
-
- /* initialize myself */
- tscm = card->private_data;
- tscm->card = card;
- tscm->unit = fw_unit_get(unit);
-
- mutex_init(&tscm->mutex);
- spin_lock_init(&tscm->lock);
- init_waitqueue_head(&tscm->hwdep_wait);
+ return;
err = identify_model(tscm);
if (err < 0)
goto error;
- snd_tscm_proc_init(tscm);
+ err = snd_tscm_transaction_register(tscm);
+ if (err < 0)
+ goto error;
err = snd_tscm_stream_init_duplex(tscm);
if (err < 0)
goto error;
- err = snd_tscm_create_pcm_devices(tscm);
- if (err < 0)
- goto error;
+ snd_tscm_proc_init(tscm);
- err = snd_tscm_transaction_register(tscm);
+ err = snd_tscm_create_pcm_devices(tscm);
if (err < 0)
goto error;
@@ -146,35 +136,91 @@
if (err < 0)
goto error;
- err = snd_card_register(card);
+ err = snd_card_register(tscm->card);
if (err < 0)
goto error;
+ /*
+ * After registered, tscm instance can be released corresponding to
+ * releasing the sound card instance.
+ */
+ tscm->card->private_free = tscm_card_free;
+ tscm->card->private_data = tscm;
+ tscm->registered = true;
+
+ return;
+error:
+ snd_tscm_transaction_unregister(tscm);
+ snd_tscm_stream_destroy_duplex(tscm);
+ snd_card_free(tscm->card);
+ dev_info(&tscm->unit->device,
+ "Sound card registration failed: %d\n", err);
+}
+
+static int snd_tscm_probe(struct fw_unit *unit,
+ const struct ieee1394_device_id *entry)
+{
+ struct snd_tscm *tscm;
+
+ /* Allocate this independent of sound card instance. */
+ tscm = kzalloc(sizeof(struct snd_tscm), GFP_KERNEL);
+ if (tscm == NULL)
+ return -ENOMEM;
+
+ /* initialize myself */
+ tscm->unit = fw_unit_get(unit);
dev_set_drvdata(&unit->device, tscm);
- return err;
-error:
- snd_card_free(card);
- return err;
+ mutex_init(&tscm->mutex);
+ spin_lock_init(&tscm->lock);
+ init_waitqueue_head(&tscm->hwdep_wait);
+
+ /* Allocate and register this sound card later. */
+ INIT_DEFERRABLE_WORK(&tscm->dwork, do_registration);
+ snd_fw_schedule_registration(unit, &tscm->dwork);
+
+ return 0;
}
static void snd_tscm_update(struct fw_unit *unit)
{
struct snd_tscm *tscm = dev_get_drvdata(&unit->device);
+ /* Postpone a workqueue for deferred registration. */
+ if (!tscm->registered)
+ snd_fw_schedule_registration(unit, &tscm->dwork);
+
snd_tscm_transaction_reregister(tscm);
- mutex_lock(&tscm->mutex);
- snd_tscm_stream_update_duplex(tscm);
- mutex_unlock(&tscm->mutex);
+ /*
+ * After registration, userspace can start packet streaming, then this
+ * code block works fine.
+ */
+ if (tscm->registered) {
+ mutex_lock(&tscm->mutex);
+ snd_tscm_stream_update_duplex(tscm);
+ mutex_unlock(&tscm->mutex);
+ }
}
static void snd_tscm_remove(struct fw_unit *unit)
{
struct snd_tscm *tscm = dev_get_drvdata(&unit->device);
- /* No need to wait for releasing card object in this context. */
- snd_card_free_when_closed(tscm->card);
+ /*
+ * Confirm to stop the work for registration before the sound card is
+ * going to be released. The work is not scheduled again because bus
+ * reset handler is not called anymore.
+ */
+ cancel_delayed_work_sync(&tscm->dwork);
+
+ if (tscm->registered) {
+ /* No need to wait for releasing card object in this context. */
+ snd_card_free_when_closed(tscm->card);
+ } else {
+ /* Don't forget this case. */
+ tscm_free(tscm);
+ }
}
static const struct ieee1394_device_id snd_tscm_id_table[] = {
diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h
index 30ab77e..1f61011 100644
--- a/sound/firewire/tascam/tascam.h
+++ b/sound/firewire/tascam/tascam.h
@@ -51,6 +51,8 @@
struct mutex mutex;
spinlock_t lock;
+ bool registered;
+ struct delayed_work dwork;
const struct snd_tscm_spec *spec;
struct fw_iso_resources tx_resources;
diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index 607bbea..c9af022 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -158,22 +158,40 @@
}
EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk);
-/* There is a fixed mapping between audio pin node and display port
- * on current Intel platforms:
+/* There is a fixed mapping between audio pin node and display port.
+ * on SNB, IVY, HSW, BSW, SKL, BXT, KBL:
* Pin Widget 5 - PORT B (port = 1 in i915 driver)
* Pin Widget 6 - PORT C (port = 2 in i915 driver)
* Pin Widget 7 - PORT D (port = 3 in i915 driver)
+ *
+ * on VLV, ILK:
+ * Pin Widget 4 - PORT B (port = 1 in i915 driver)
+ * Pin Widget 5 - PORT C (port = 2 in i915 driver)
+ * Pin Widget 6 - PORT D (port = 3 in i915 driver)
*/
-static int pin2port(hda_nid_t pin_nid)
+static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid)
{
- if (WARN_ON(pin_nid < 5 || pin_nid > 7))
+ int base_nid;
+
+ switch (codec->vendor_id) {
+ case 0x80860054: /* ILK */
+ case 0x80862804: /* ILK */
+ case 0x80862882: /* VLV */
+ base_nid = 3;
+ break;
+ default:
+ base_nid = 4;
+ break;
+ }
+
+ if (WARN_ON(pin_nid <= base_nid || pin_nid > base_nid + 3))
return -1;
- return pin_nid - 4;
+ return pin_nid - base_nid;
}
/**
* snd_hdac_sync_audio_rate - Set N/CTS based on the sample rate
- * @bus: HDA core bus
+ * @codec: HDA codec
* @nid: the pin widget NID
* @rate: the sample rate to set
*
@@ -183,14 +201,15 @@
* This function sets N/CTS value based on the given sample rate.
* Returns zero for success, or a negative error code.
*/
-int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate)
+int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate)
{
+ struct hdac_bus *bus = codec->bus;
struct i915_audio_component *acomp = bus->audio_component;
int port;
if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate)
return -ENODEV;
- port = pin2port(nid);
+ port = pin2port(codec, nid);
if (port < 0)
return -EINVAL;
return acomp->ops->sync_audio_rate(acomp->dev, port, rate);
@@ -199,7 +218,7 @@
/**
* snd_hdac_acomp_get_eld - Get the audio state and ELD via component
- * @bus: HDA core bus
+ * @codec: HDA codec
* @nid: the pin widget NID
* @audio_enabled: the pointer to store the current audio state
* @buffer: the buffer pointer to store ELD bytes
@@ -217,16 +236,17 @@
* thus it may be over @max_bytes. If it's over @max_bytes, it implies
* that only a part of ELD bytes have been fetched.
*/
-int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid,
+int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid,
bool *audio_enabled, char *buffer, int max_bytes)
{
+ struct hdac_bus *bus = codec->bus;
struct i915_audio_component *acomp = bus->audio_component;
int port;
if (!acomp || !acomp->ops || !acomp->ops->get_eld)
return -ENODEV;
- port = pin2port(nid);
+ port = pin2port(codec, nid);
if (port < 0)
return -EINVAL;
return acomp->ops->get_eld(acomp->dev, port, audio_enabled,
@@ -338,6 +358,9 @@
struct i915_audio_component *acomp;
int ret;
+ if (WARN_ON(hdac_acomp))
+ return -EBUSY;
+
if (!i915_gfx_present())
return -ENODEV;
@@ -371,6 +394,7 @@
out_err:
kfree(acomp);
bus->audio_component = NULL;
+ hdac_acomp = NULL;
dev_info(dev, "failed to add i915 component master (%d)\n", ret);
return ret;
@@ -404,6 +428,7 @@
kfree(acomp);
bus->audio_component = NULL;
+ hdac_acomp = NULL;
return 0;
}
diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c
index d7ec862..c6c75e7 100644
--- a/sound/hda/hdmi_chmap.c
+++ b/sound/hda/hdmi_chmap.c
@@ -625,13 +625,30 @@
WARN_ON(count != channels);
}
+static int spk_mask_from_spk_alloc(int spk_alloc)
+{
+ int i;
+ int spk_mask = eld_speaker_allocation_bits[0];
+
+ for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) {
+ if (spk_alloc & (1 << i))
+ spk_mask |= eld_speaker_allocation_bits[i];
+ }
+
+ return spk_mask;
+}
+
static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag,
unsigned int size, unsigned int __user *tlv)
{
struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol);
struct hdac_chmap *chmap = info->private_data;
+ int pcm_idx = kcontrol->private_value;
unsigned int __user *dst;
int chs, count = 0;
+ unsigned long max_chs;
+ int type;
+ int spk_alloc, spk_mask;
if (size < 8)
return -ENOMEM;
@@ -639,40 +656,59 @@
return -EFAULT;
size -= 8;
dst = tlv + 2;
- for (chs = 2; chs <= chmap->channels_max; chs++) {
+
+ spk_alloc = chmap->ops.get_spk_alloc(chmap->hdac, pcm_idx);
+ spk_mask = spk_mask_from_spk_alloc(spk_alloc);
+
+ max_chs = hweight_long(spk_mask);
+
+ for (chs = 2; chs <= max_chs; chs++) {
int i;
struct hdac_cea_channel_speaker_allocation *cap;
cap = channel_allocations;
for (i = 0; i < ARRAY_SIZE(channel_allocations); i++, cap++) {
int chs_bytes = chs * 4;
- int type = chmap->ops.chmap_cea_alloc_validate_get_type(
- chmap, cap, chs);
unsigned int tlv_chmap[8];
- if (type < 0)
+ if (cap->channels != chs)
continue;
+
+ if (!(cap->spk_mask == (spk_mask & cap->spk_mask)))
+ continue;
+
+ type = chmap->ops.chmap_cea_alloc_validate_get_type(
+ chmap, cap, chs);
+ if (type < 0)
+ return -ENODEV;
if (size < 8)
return -ENOMEM;
+
if (put_user(type, dst) ||
put_user(chs_bytes, dst + 1))
return -EFAULT;
+
dst += 2;
size -= 8;
count += 8;
+
if (size < chs_bytes)
return -ENOMEM;
+
size -= chs_bytes;
count += chs_bytes;
chmap->ops.cea_alloc_to_tlv_chmap(chmap, cap,
tlv_chmap, chs);
+
if (copy_to_user(dst, tlv_chmap, chs_bytes))
return -EFAULT;
dst += chs;
}
}
+
if (put_user(count, tlv + 1))
return -EFAULT;
+
return 0;
}
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index a6d6d8d..df5741a 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -432,7 +432,10 @@
#endif
//printk(KERN_INFO "vortex: pointer = 0x%x\n", current_ptr);
spin_unlock(&chip->lock);
- return (bytes_to_frames(substream->runtime, current_ptr));
+ current_ptr = bytes_to_frames(substream->runtime, current_ptr);
+ if (current_ptr >= substream->runtime->buffer_size)
+ current_ptr = 0;
+ return current_ptr;
}
/* operators */
diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c
index a5d4604..8f94534 100644
--- a/sound/pci/ctxfi/cttimer.c
+++ b/sound/pci/ctxfi/cttimer.c
@@ -49,7 +49,7 @@
spinlock_t lock; /* global timer lock (for xfitimer) */
spinlock_t list_lock; /* lock for instance list */
struct ct_atc *atc;
- struct ct_timer_ops *ops;
+ const struct ct_timer_ops *ops;
struct list_head instance_head;
struct list_head running_head;
unsigned int wc; /* current wallclock */
@@ -128,7 +128,7 @@
#define ct_systimer_free ct_systimer_prepare
-static struct ct_timer_ops ct_systimer_ops = {
+static const struct ct_timer_ops ct_systimer_ops = {
.init = ct_systimer_init,
.free_instance = ct_systimer_free,
.prepare = ct_systimer_prepare,
@@ -322,7 +322,7 @@
ct_xfitimer_irq_stop(atimer);
}
-static struct ct_timer_ops ct_xfitimer_ops = {
+static const struct ct_timer_ops ct_xfitimer_ops = {
.prepare = ct_xfitimer_prepare,
.start = ct_xfitimer_start,
.stop = ct_xfitimer_stop,
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 0dc44eb..626cd21 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -1548,7 +1548,7 @@
int val = 0;
spin_lock_irq(&ensoniq->reg_lock);
- if ((ensoniq->ctrl & ES_1371_GPIO_OUTM) >= 4)
+ if (ensoniq->ctrl & ES_1371_GPIO_OUT(4))
val = 1;
ucontrol->value.integer.value[0] = val;
spin_unlock_irq(&ensoniq->reg_lock);
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index dfaf1a9..320445f 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -5434,6 +5434,7 @@
spec->cur_adc_stream_tag = stream_tag;
spec->cur_adc_format = format;
snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format);
+ call_pcm_capture_hook(hinfo, codec, substream, HDA_GEN_PCM_ACT_PREPARE);
return 0;
}
@@ -5444,6 +5445,7 @@
struct hda_gen_spec *spec = codec->spec;
snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
spec->cur_adc = 0;
+ call_pcm_capture_hook(hinfo, codec, substream, HDA_GEN_PCM_ACT_CLEANUP);
return 0;
}
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 1483f85..d2e57c72 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -114,6 +114,9 @@
int (*setup_stream)(struct hda_codec *codec, hda_nid_t cvt_nid,
hda_nid_t pin_nid, u32 stream_tag, int format);
+ void (*pin_cvt_fixup)(struct hda_codec *codec,
+ struct hdmi_spec_per_pin *per_pin,
+ hda_nid_t cvt_nid);
};
struct hdmi_pcm {
@@ -684,7 +687,8 @@
if (!channels)
return;
- if (is_haswell_plus(codec))
+ /* some HW (e.g. HSW+) needs reprogramming the amp at each time */
+ if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP)
snd_hda_codec_write(codec, pin_nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_UNMUTE);
@@ -864,9 +868,6 @@
struct hdmi_spec *spec = codec->spec;
int err;
- if (is_haswell_plus(codec))
- haswell_verify_D0(codec, cvt_nid, pin_nid);
-
err = spec->ops.pin_hbr_setup(codec, pin_nid, is_hbr_format(format));
if (err) {
@@ -884,7 +885,7 @@
* of the pin.
*/
static int hdmi_choose_cvt(struct hda_codec *codec,
- int pin_idx, int *cvt_id, int *mux_id)
+ int pin_idx, int *cvt_id)
{
struct hdmi_spec *spec = codec->spec;
struct hdmi_spec_per_pin *per_pin;
@@ -925,8 +926,6 @@
if (cvt_id)
*cvt_id = cvt_idx;
- if (mux_id)
- *mux_id = mux_idx;
return 0;
}
@@ -1019,9 +1018,6 @@
int mux_idx;
struct hdmi_spec *spec = codec->spec;
- if (!is_haswell_plus(codec) && !is_valleyview_plus(codec))
- return;
-
/* On Intel platform, the mapping of converter nid to
* mux index of the pins are always the same.
* The pin nid may be 0, this means all pins will not
@@ -1032,6 +1028,17 @@
intel_not_share_assigned_cvt(codec, pin_nid, mux_idx);
}
+/* skeleton caller of pin_cvt_fixup ops */
+static void pin_cvt_fixup(struct hda_codec *codec,
+ struct hdmi_spec_per_pin *per_pin,
+ hda_nid_t cvt_nid)
+{
+ struct hdmi_spec *spec = codec->spec;
+
+ if (spec->ops.pin_cvt_fixup)
+ spec->ops.pin_cvt_fixup(codec, per_pin, cvt_nid);
+}
+
/* called in hdmi_pcm_open when no pin is assigned to the PCM
* in dyn_pcm_assign mode.
*/
@@ -1049,7 +1056,7 @@
if (pcm_idx < 0)
return -EINVAL;
- err = hdmi_choose_cvt(codec, -1, &cvt_idx, NULL);
+ err = hdmi_choose_cvt(codec, -1, &cvt_idx);
if (err)
return err;
@@ -1057,7 +1064,7 @@
per_cvt->assigned = 1;
hinfo->nid = per_cvt->cvt_nid;
- intel_not_share_assigned_cvt_nid(codec, 0, per_cvt->cvt_nid);
+ pin_cvt_fixup(codec, NULL, per_cvt->cvt_nid);
set_bit(pcm_idx, &spec->pcm_in_use);
/* todo: setup spdif ctls assign */
@@ -1089,7 +1096,7 @@
{
struct hdmi_spec *spec = codec->spec;
struct snd_pcm_runtime *runtime = substream->runtime;
- int pin_idx, cvt_idx, pcm_idx, mux_idx = 0;
+ int pin_idx, cvt_idx, pcm_idx;
struct hdmi_spec_per_pin *per_pin;
struct hdmi_eld *eld;
struct hdmi_spec_per_cvt *per_cvt = NULL;
@@ -1118,7 +1125,7 @@
}
}
- err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx, &mux_idx);
+ err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx);
if (err < 0) {
mutex_unlock(&spec->pcm_lock);
return err;
@@ -1135,11 +1142,10 @@
snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0,
AC_VERB_SET_CONNECT_SEL,
- mux_idx);
+ per_pin->mux_idx);
/* configure unused pins to choose other converters */
- if (is_haswell_plus(codec) || is_valleyview_plus(codec))
- intel_not_share_assigned_cvt(codec, per_pin->pin_nid, mux_idx);
+ pin_cvt_fixup(codec, per_pin, 0);
snd_hda_spdif_ctls_assign(codec, pcm_idx, per_cvt->cvt_nid);
@@ -1372,12 +1378,7 @@
* and this can make HW reset converter selection on a pin.
*/
if (eld->eld_valid && !old_eld_valid && per_pin->setup) {
- if (is_haswell_plus(codec) || is_valleyview_plus(codec)) {
- intel_verify_pin_cvt_connect(codec, per_pin);
- intel_not_share_assigned_cvt(codec, per_pin->pin_nid,
- per_pin->mux_idx);
- }
-
+ pin_cvt_fixup(codec, per_pin, 0);
hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm);
}
@@ -1484,7 +1485,7 @@
mutex_lock(&per_pin->lock);
eld->monitor_present = false;
- size = snd_hdac_acomp_get_eld(&codec->bus->core, per_pin->pin_nid,
+ size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid,
&eld->monitor_present, eld->eld_buffer,
ELD_MAX_SIZE);
if (size > 0) {
@@ -1711,7 +1712,7 @@
* skip pin setup and return 0 to make audio playback
* be ongoing
*/
- intel_not_share_assigned_cvt_nid(codec, 0, cvt_nid);
+ pin_cvt_fixup(codec, NULL, cvt_nid);
snd_hda_codec_setup_stream(codec, cvt_nid,
stream_tag, 0, format);
mutex_unlock(&spec->pcm_lock);
@@ -1724,23 +1725,21 @@
}
per_pin = get_pin(spec, pin_idx);
pin_nid = per_pin->pin_nid;
- if (is_haswell_plus(codec) || is_valleyview_plus(codec)) {
- /* Verify pin:cvt selections to avoid silent audio after S3.
- * After S3, the audio driver restores pin:cvt selections
- * but this can happen before gfx is ready and such selection
- * is overlooked by HW. Thus multiple pins can share a same
- * default convertor and mute control will affect each other,
- * which can cause a resumed audio playback become silent
- * after S3.
- */
- intel_verify_pin_cvt_connect(codec, per_pin);
- intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx);
- }
+
+ /* Verify pin:cvt selections to avoid silent audio after S3.
+ * After S3, the audio driver restores pin:cvt selections
+ * but this can happen before gfx is ready and such selection
+ * is overlooked by HW. Thus multiple pins can share a same
+ * default convertor and mute control will affect each other,
+ * which can cause a resumed audio playback become silent
+ * after S3.
+ */
+ pin_cvt_fixup(codec, per_pin, 0);
/* Call sync_audio_rate to set the N/CTS/M manually if necessary */
/* Todo: add DP1.2 MST audio support later */
if (codec_has_acomp(codec))
- snd_hdac_sync_audio_rate(&codec->bus->core, pin_nid, runtime->rate);
+ snd_hdac_sync_audio_rate(&codec->core, pin_nid, runtime->rate);
non_pcm = check_non_pcm_per_cvt(codec, cvt_nid);
mutex_lock(&per_pin->lock);
@@ -1837,6 +1836,18 @@
.cleanup = generic_hdmi_playback_pcm_cleanup,
};
+static int hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx)
+{
+ struct hda_codec *codec = container_of(hdac, struct hda_codec, core);
+ struct hdmi_spec *spec = codec->spec;
+ struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx);
+
+ if (!per_pin)
+ return 0;
+
+ return per_pin->sink_eld.info.spk_alloc;
+}
+
static void hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx,
unsigned char *chmap)
{
@@ -2075,6 +2086,20 @@
snd_array_free(&spec->cvts);
}
+static void generic_spec_free(struct hda_codec *codec)
+{
+ struct hdmi_spec *spec = codec->spec;
+
+ if (spec) {
+ if (spec->i915_bound)
+ snd_hdac_i915_exit(&codec->bus->core);
+ hdmi_array_free(spec);
+ kfree(spec);
+ codec->spec = NULL;
+ }
+ codec->dp_mst = false;
+}
+
static void generic_hdmi_free(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
@@ -2099,10 +2124,7 @@
spec->pcm_rec[pcm_idx].jack = NULL;
}
- if (spec->i915_bound)
- snd_hdac_i915_exit(&codec->bus->core);
- hdmi_array_free(spec);
- kfree(spec);
+ generic_spec_free(codec);
}
#ifdef CONFIG_PM
@@ -2140,6 +2162,55 @@
.setup_stream = hdmi_setup_stream,
};
+/* allocate codec->spec and assign/initialize generic parser ops */
+static int alloc_generic_hdmi(struct hda_codec *codec)
+{
+ struct hdmi_spec *spec;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+
+ spec->ops = generic_standard_hdmi_ops;
+ mutex_init(&spec->pcm_lock);
+ snd_hdac_register_chmap_ops(&codec->core, &spec->chmap);
+
+ spec->chmap.ops.get_chmap = hdmi_get_chmap;
+ spec->chmap.ops.set_chmap = hdmi_set_chmap;
+ spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached;
+ spec->chmap.ops.get_spk_alloc = hdmi_get_spk_alloc,
+
+ codec->spec = spec;
+ hdmi_array_init(spec, 4);
+
+ codec->patch_ops = generic_hdmi_patch_ops;
+
+ return 0;
+}
+
+/* generic HDMI parser */
+static int patch_generic_hdmi(struct hda_codec *codec)
+{
+ int err;
+
+ err = alloc_generic_hdmi(codec);
+ if (err < 0)
+ return err;
+
+ err = hdmi_parse_codec(codec);
+ if (err < 0) {
+ generic_spec_free(codec);
+ return err;
+ }
+
+ generic_hdmi_init_per_pins(codec);
+ return 0;
+}
+
+/*
+ * Intel codec parsers and helpers
+ */
+
static void intel_haswell_fixup_connect_list(struct hda_codec *codec,
hda_nid_t nid)
{
@@ -2217,12 +2288,23 @@
static void intel_pin_eld_notify(void *audio_ptr, int port)
{
struct hda_codec *codec = audio_ptr;
- int pin_nid = port + 0x04;
+ int pin_nid;
/* we assume only from port-B to port-D */
if (port < 1 || port > 3)
return;
+ switch (codec->core.vendor_id) {
+ case 0x80860054: /* ILK */
+ case 0x80862804: /* ILK */
+ case 0x80862882: /* VLV */
+ pin_nid = port + 0x03;
+ break;
+ default:
+ pin_nid = port + 0x04;
+ break;
+ }
+
/* skip notification during system suspend (but not in runtime PM);
* the state will be updated at resume
*/
@@ -2236,95 +2318,161 @@
check_presence_and_report(codec, pin_nid);
}
-static int patch_generic_hdmi(struct hda_codec *codec)
+/* register i915 component pin_eld_notify callback */
+static void register_i915_notifier(struct hda_codec *codec)
+{
+ struct hdmi_spec *spec = codec->spec;
+
+ spec->use_acomp_notifier = true;
+ spec->i915_audio_ops.audio_ptr = codec;
+ /* intel_audio_codec_enable() or intel_audio_codec_disable()
+ * will call pin_eld_notify with using audio_ptr pointer
+ * We need make sure audio_ptr is really setup
+ */
+ wmb();
+ spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify;
+ snd_hdac_i915_register_notifier(&spec->i915_audio_ops);
+}
+
+/* setup_stream ops override for HSW+ */
+static int i915_hsw_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid,
+ hda_nid_t pin_nid, u32 stream_tag, int format)
+{
+ haswell_verify_D0(codec, cvt_nid, pin_nid);
+ return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format);
+}
+
+/* pin_cvt_fixup ops override for HSW+ and VLV+ */
+static void i915_pin_cvt_fixup(struct hda_codec *codec,
+ struct hdmi_spec_per_pin *per_pin,
+ hda_nid_t cvt_nid)
+{
+ if (per_pin) {
+ intel_verify_pin_cvt_connect(codec, per_pin);
+ intel_not_share_assigned_cvt(codec, per_pin->pin_nid,
+ per_pin->mux_idx);
+ } else {
+ intel_not_share_assigned_cvt_nid(codec, 0, cvt_nid);
+ }
+}
+
+/* Intel Haswell and onwards; audio component with eld notifier */
+static int patch_i915_hsw_hdmi(struct hda_codec *codec)
{
struct hdmi_spec *spec;
+ int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- spec->ops = generic_standard_hdmi_ops;
- mutex_init(&spec->pcm_lock);
- snd_hdac_register_chmap_ops(&codec->core, &spec->chmap);
-
- spec->chmap.ops.get_chmap = hdmi_get_chmap;
- spec->chmap.ops.set_chmap = hdmi_set_chmap;
- spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached;
-
- codec->spec = spec;
- hdmi_array_init(spec, 4);
-
-#ifdef CONFIG_SND_HDA_I915
- /* Try to bind with i915 for Intel HSW+ codecs (if not done yet) */
- if ((codec->core.vendor_id >> 16) == 0x8086 &&
- is_haswell_plus(codec)) {
-#if 0
- /* on-demand binding leads to an unbalanced refcount when
- * both i915 and hda drivers are probed concurrently;
- * disabled temporarily for now
- */
- if (!codec->bus->core.audio_component)
- if (!snd_hdac_i915_init(&codec->bus->core))
- spec->i915_bound = true;
-#endif
- /* use i915 audio component notifier for hotplug */
- if (codec->bus->core.audio_component)
- spec->use_acomp_notifier = true;
+ /* HSW+ requires i915 binding */
+ if (!codec->bus->core.audio_component) {
+ codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n");
+ return -ENODEV;
}
-#endif
- if (is_haswell_plus(codec)) {
- intel_haswell_enable_all_pins(codec, true);
- intel_haswell_fixup_enable_dp12(codec);
+ err = alloc_generic_hdmi(codec);
+ if (err < 0)
+ return err;
+ spec = codec->spec;
+
+ intel_haswell_enable_all_pins(codec, true);
+ intel_haswell_fixup_enable_dp12(codec);
+
+ /* For Haswell/Broadwell, the controller is also in the power well and
+ * can cover the codec power request, and so need not set this flag.
+ */
+ if (!is_haswell(codec) && !is_broadwell(codec))
+ codec->core.link_power_control = 1;
+
+ codec->patch_ops.set_power_state = haswell_set_power_state;
+ codec->dp_mst = true;
+ codec->depop_delay = 0;
+ codec->auto_runtime_pm = 1;
+
+ spec->ops.setup_stream = i915_hsw_setup_stream;
+ spec->ops.pin_cvt_fixup = i915_pin_cvt_fixup;
+
+ err = hdmi_parse_codec(codec);
+ if (err < 0) {
+ generic_spec_free(codec);
+ return err;
}
+ generic_hdmi_init_per_pins(codec);
+ register_i915_notifier(codec);
+ return 0;
+}
+
+/* Intel Baytrail and Braswell; with eld notifier */
+static int patch_i915_byt_hdmi(struct hda_codec *codec)
+{
+ struct hdmi_spec *spec;
+ int err;
+
+ /* requires i915 binding */
+ if (!codec->bus->core.audio_component) {
+ codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n");
+ return -ENODEV;
+ }
+
+ err = alloc_generic_hdmi(codec);
+ if (err < 0)
+ return err;
+ spec = codec->spec;
+
/* For Valleyview/Cherryview, only the display codec is in the display
* power well and can use link_power ops to request/release the power.
- * For Haswell/Broadwell, the controller is also in the power well and
- * can cover the codec power request, and so need not set this flag.
- * For previous platforms, there is no such power well feature.
*/
- if (is_valleyview_plus(codec) || is_skylake(codec) ||
- is_broxton(codec))
- codec->core.link_power_control = 1;
+ codec->core.link_power_control = 1;
- if (hdmi_parse_codec(codec) < 0) {
- if (spec->i915_bound)
- snd_hdac_i915_exit(&codec->bus->core);
- codec->spec = NULL;
- kfree(spec);
- return -EINVAL;
- }
- codec->patch_ops = generic_hdmi_patch_ops;
- if (is_haswell_plus(codec)) {
- codec->patch_ops.set_power_state = haswell_set_power_state;
- codec->dp_mst = true;
- }
+ codec->depop_delay = 0;
+ codec->auto_runtime_pm = 1;
- /* Enable runtime pm for HDMI audio codec of HSW/BDW/SKL/BYT/BSW */
- if (is_haswell_plus(codec) || is_valleyview_plus(codec))
- codec->auto_runtime_pm = 1;
+ spec->ops.pin_cvt_fixup = i915_pin_cvt_fixup;
+
+ err = hdmi_parse_codec(codec);
+ if (err < 0) {
+ generic_spec_free(codec);
+ return err;
+ }
generic_hdmi_init_per_pins(codec);
-
-
- if (codec_has_acomp(codec)) {
- codec->depop_delay = 0;
- spec->i915_audio_ops.audio_ptr = codec;
- /* intel_audio_codec_enable() or intel_audio_codec_disable()
- * will call pin_eld_notify with using audio_ptr pointer
- * We need make sure audio_ptr is really setup
- */
- wmb();
- spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify;
- snd_hdac_i915_register_notifier(&spec->i915_audio_ops);
- }
-
- WARN_ON(spec->dyn_pcm_assign && !codec_has_acomp(codec));
+ register_i915_notifier(codec);
return 0;
}
+/* Intel IronLake, SandyBridge and IvyBridge; with eld notifier */
+static int patch_i915_cpt_hdmi(struct hda_codec *codec)
+{
+ struct hdmi_spec *spec;
+ int err;
+
+ /* no i915 component should have been bound before this */
+ if (WARN_ON(codec->bus->core.audio_component))
+ return -EBUSY;
+
+ err = alloc_generic_hdmi(codec);
+ if (err < 0)
+ return err;
+ spec = codec->spec;
+
+ /* Try to bind with i915 now */
+ err = snd_hdac_i915_init(&codec->bus->core);
+ if (err < 0)
+ goto error;
+ spec->i915_bound = true;
+
+ err = hdmi_parse_codec(codec);
+ if (err < 0)
+ goto error;
+
+ generic_hdmi_init_per_pins(codec);
+ register_i915_notifier(codec);
+ return 0;
+
+ error:
+ generic_spec_free(codec);
+ return err;
+}
+
/*
* Shared non-generic implementations
*/
@@ -3493,21 +3641,21 @@
HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi),
HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi),
HDA_CODEC_ENTRY(0x11069f85, "VX11 HDMI/DP", patch_generic_hdmi),
-HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI", patch_i915_cpt_hdmi),
HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI", patch_generic_hdmi),
HDA_CODEC_ENTRY(0x80862802, "Cantiga HDMI", patch_generic_hdmi),
HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI", patch_generic_hdmi),
-HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI", patch_generic_hdmi),
-HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI", patch_generic_hdmi),
-HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_generic_hdmi),
-HDA_CODEC_ENTRY(0x80862807, "Haswell HDMI", patch_generic_hdmi),
-HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_generic_hdmi),
-HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_generic_hdmi),
-HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_generic_hdmi),
-HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI", patch_i915_cpt_hdmi),
+HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI", patch_i915_cpt_hdmi),
+HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_i915_cpt_hdmi),
+HDA_CODEC_ENTRY(0x80862807, "Haswell HDMI", patch_i915_hsw_hdmi),
+HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_i915_hsw_hdmi),
+HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_i915_hsw_hdmi),
+HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi),
+HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi),
HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi),
-HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_generic_hdmi),
-HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi),
+HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi),
HDA_CODEC_ENTRY(0x808629fb, "Crestline HDMI", patch_generic_hdmi),
/* special ID for generic HDMI */
HDA_CODEC_ENTRY(HDA_CODEC_ID_GENERIC_HDMI, "Generic HDMI", patch_generic_hdmi),
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 8151318..9720a30 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -42,12 +42,6 @@
#include <asm/pgtable.h>
#include <asm/cacheflush.h>
-#ifdef CONFIG_KVM_GUEST
-#include <linux/kvm_para.h>
-#else
-#define kvm_para_available() (0)
-#endif
-
MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455");
MODULE_LICENSE("GPL");
@@ -2972,25 +2966,17 @@
goto fini;
}
- /* detect KVM and Parallels virtual environments */
- result = kvm_para_available();
-#ifdef X86_FEATURE_HYPERVISOR
- result = result || boot_cpu_has(X86_FEATURE_HYPERVISOR);
-#endif
- if (!result)
- goto fini;
-
/* check for known (emulated) devices */
+ result = 0;
if (pci->subsystem_vendor == PCI_SUBVENDOR_ID_REDHAT_QUMRANET &&
pci->subsystem_device == PCI_SUBDEVICE_ID_QEMU) {
/* KVM emulated sound, PCI SSID: 1af4:1100 */
msg = "enable KVM";
+ result = 1;
} else if (pci->subsystem_vendor == 0x1ab8) {
/* Parallels VM emulated sound, PCI SSID: 1ab8:xxxx */
msg = "enable Parallels VM";
- } else {
- msg = "disable (unknown or VT-d) VM";
- result = 0;
+ result = 1;
}
fini:
diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c
index f3d6202..a80684b 100644
--- a/sound/pci/lx6464es/lx_core.c
+++ b/sound/pci/lx6464es/lx_core.c
@@ -644,7 +644,7 @@
if (err < 0)
return err;
- if (current_state == state)
+ if (!err && current_state == state)
return 0;
mdelay(1);
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 4f85757..2f8c388 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -45,6 +45,7 @@
#include <linux/bitops.h>
#include <linux/init.h>
#include <linux/list.h>
+#include <linux/log2.h>
#include <linux/slab.h>
#include <linux/string.h>
#include <linux/usb.h>
@@ -1378,6 +1379,71 @@
snd_usb_mixer_add_control(&cval->head, kctl);
}
+static int parse_clock_source_unit(struct mixer_build *state, int unitid,
+ void *_ftr)
+{
+ struct uac_clock_source_descriptor *hdr = _ftr;
+ struct usb_mixer_elem_info *cval;
+ struct snd_kcontrol *kctl;
+ char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+ int ret;
+
+ if (state->mixer->protocol != UAC_VERSION_2)
+ return -EINVAL;
+
+ if (hdr->bLength != sizeof(*hdr)) {
+ usb_audio_dbg(state->chip,
+ "Bogus clock source descriptor length of %d, ignoring.\n",
+ hdr->bLength);
+ return 0;
+ }
+
+ /*
+ * The only property of this unit we are interested in is the
+ * clock source validity. If that isn't readable, just bail out.
+ */
+ if (!uac2_control_is_readable(hdr->bmControls,
+ ilog2(UAC2_CS_CONTROL_CLOCK_VALID)))
+ return 0;
+
+ cval = kzalloc(sizeof(*cval), GFP_KERNEL);
+ if (!cval)
+ return -ENOMEM;
+
+ snd_usb_mixer_elem_init_std(&cval->head, state->mixer, hdr->bClockID);
+
+ cval->min = 0;
+ cval->max = 1;
+ cval->channels = 1;
+ cval->val_type = USB_MIXER_BOOLEAN;
+ cval->control = UAC2_CS_CONTROL_CLOCK_VALID;
+
+ if (uac2_control_is_writeable(hdr->bmControls,
+ ilog2(UAC2_CS_CONTROL_CLOCK_VALID)))
+ kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval);
+ else {
+ cval->master_readonly = 1;
+ kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval);
+ }
+
+ if (!kctl) {
+ kfree(cval);
+ return -ENOMEM;
+ }
+
+ kctl->private_free = snd_usb_mixer_elem_free;
+ ret = snd_usb_copy_string_desc(state, hdr->iClockSource,
+ name, sizeof(name));
+ if (ret > 0)
+ snprintf(kctl->id.name, sizeof(kctl->id.name),
+ "%s Validity", name);
+ else
+ snprintf(kctl->id.name, sizeof(kctl->id.name),
+ "Clock Source %d Validity", hdr->bClockID);
+
+ return snd_usb_mixer_add_control(&cval->head, kctl);
+}
+
/*
* parse a feature unit
*
@@ -2126,10 +2192,11 @@
switch (p1[2]) {
case UAC_INPUT_TERMINAL:
- case UAC2_CLOCK_SOURCE:
return 0; /* NOP */
case UAC_MIXER_UNIT:
return parse_audio_mixer_unit(state, unitid, p1);
+ case UAC2_CLOCK_SOURCE:
+ return parse_clock_source_unit(state, unitid, p1);
case UAC_SELECTOR_UNIT:
case UAC2_CLOCK_SELECTOR:
return parse_audio_selector_unit(state, unitid, p1);
@@ -2307,6 +2374,7 @@
__u8 unitid = (index >> 8) & 0xff;
__u8 control = (value >> 8) & 0xff;
__u8 channel = value & 0xff;
+ unsigned int count = 0;
if (channel >= MAX_CHANNELS) {
usb_audio_dbg(mixer->chip,
@@ -2315,6 +2383,12 @@
return;
}
+ for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem)
+ count++;
+
+ if (count == 0)
+ return;
+
for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) {
struct usb_mixer_elem_info *info;
@@ -2322,7 +2396,7 @@
continue;
info = (struct usb_mixer_elem_info *)list;
- if (info->control != control)
+ if (count > 1 && info->control != control)
continue;
switch (attribute) {
