diff options
62 files changed, 1437 insertions, 1035 deletions
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index e7193aac669c..d4510ebf2e8c 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -655,17 +655,6 @@ development branches in general while the development for the current 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 @@ problems. 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 @@ can get a proc-file dump at the current state, get a list of control (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 630c492c3dc2..8ba556a131c3 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -149,7 +149,7 @@ Gapless Playback ================ 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 @@ -184,7 +184,7 @@ Sequence flow for gapless would be: - Fill data of the first track - Trigger start - User-space finished sending all, -- Indicaite next track data by sending set_next_track +- Indicate next track data by sending set_next_track - Set metadata of the next track - then call partial_drain to flush most of buffer in DSP - Fill data of the next track diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index 6faab4880006..c45bd79f291e 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -132,7 +132,7 @@ SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0), 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 ff88f52eec98..f3f28b7ae242 100644 --- a/Documentation/sound/alsa/soc/overview.txt +++ b/Documentation/sound/alsa/soc/overview.txt @@ -63,7 +63,7 @@ multiple re-usable component drivers :- 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 0b191a23f534..1b6473f393a8 100644 --- a/Documentation/sound/alsa/timestamping.txt +++ b/Documentation/sound/alsa/timestamping.txt @@ -129,7 +129,7 @@ will be required to issue multiple queries and perform an 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 e20d219a0304..babd445c7505 100644 --- a/include/sound/hda_chmap.h +++ b/include/sound/hda_chmap.h @@ -36,6 +36,8 @@ struct hdac_chmap_ops { 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 f5842bcd9c94..796cabf6be5e 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 int snd_hdac_display_power(struct hdac_bus *bus, bool enable) 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 67bf49d8c944..609cadb8739d 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -672,7 +672,7 @@ enum { /* 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 6d12ca9bcb80..9749f9e8b45c 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -141,35 +141,6 @@ config SND_SEQ_HRTIMER_DEFAULT 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 48ab4b8f8279..e85d9dd12c2d 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -37,7 +37,6 @@ obj-$(CONFIG_SND) += snd.o obj-$(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/compress_offload.c b/sound/core/compress_offload.c index a9933c07a6bf..9b3334be9df2 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -288,9 +288,12 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, stream = &data->stream; mutex_lock(&stream->device->lock); /* write is allowed when stream is running or has been steup */ - if (stream->runtime->state != SNDRV_PCM_STATE_SETUP && - stream->runtime->state != SNDRV_PCM_STATE_PREPARED && - stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_RUNNING: + break; + default: mutex_unlock(&stream->device->lock); return -EBADFD; } @@ -391,14 +394,13 @@ static unsigned int snd_compr_poll(struct file *f, poll_table *wait) int retval = 0; if (snd_BUG_ON(!data)) - return -EFAULT; + return POLLERR; + stream = &data->stream; - if (snd_BUG_ON(!stream)) - return -EFAULT; mutex_lock(&stream->device->lock); if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) { - retval = -EBADFD; + retval = snd_compr_get_poll(stream) | POLLERR; goto out; } poll_wait(f, &stream->runtime->sleep, wait); @@ -421,10 +423,7 @@ static unsigned int snd_compr_poll(struct file *f, poll_table *wait) retval = snd_compr_get_poll(stream); break; default: - if (stream->direction == SND_COMPRESS_PLAYBACK) - retval = POLLOUT | POLLWRNORM | POLLERR; - else - retval = POLLIN | POLLRDNORM | POLLERR; + retval = snd_compr_get_poll(stream) | POLLERR; break; } out: @@ -802,9 +801,9 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) if (snd_BUG_ON(!data)) return -EFAULT; + stream = &data->stream; - if (snd_BUG_ON(!stream)) - return -EFAULT; + mutex_lock(&stream->device->lock); switch (_IOC_NR(cmd)) { case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index 656d9a9032dc..e2f27022b363 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -38,37 +38,53 @@ static unsigned int resolution; 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; - - if (!atomic_read(&stime->running)) - return HRTIMER_NORESTART; - - oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution)); - snd_timer_interrupt(stime->timer, t->sticks * oruns); + ktime_t delta; + unsigned long ticks; + enum hrtimer_restart ret = HRTIMER_NORESTART; + + spin_lock(&t->lock); + if (!t->running) + goto out; /* fast path */ + stime->in_callback = true; + ticks = t->sticks; + spin_unlock(&t->lock); + + /* calculate the drift */ + delta = ktime_sub(hrt->base->get_time(), hrtimer_get_expires(hrt)); + if (delta.tv64 > 0) + ticks += ktime_divns(delta, ticks * resolution); + + snd_timer_interrupt(stime->timer, ticks); + + spin_lock(&t->lock); + if (t->running) { + hrtimer_add_expires_ns(hrt, t->sticks * resolution); + ret = HRTIMER_RESTART; + } - if (!atomic_read(&stime->running)) - return HRTIMER_NORESTART; - return 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 @@ static int snd_hrtimer_close(struct snd_timer *t) 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 @@ static int snd_hrtimer_start(struct snd_timer *t) { 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 3a9b66c6e09c..bb1261591a1f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1886,8 +1886,8 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) 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); @@ -2595,6 +2595,8 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, }; int err; + if (WARN_ON(pcm->streams[stream].chmap_kctl)) + return -EBUSY; info = kzalloc(sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 9106d8e2300e..c61fd50f771f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3161,7 +3161,7 @@ static unsigned int snd_pcm_playback_poll(struct file *file, poll_table * wait) substream = pcm_file->substream; if (PCM_RUNTIME_CHECK(substream)) - return -ENXIO; + return POLLOUT | POLLWRNORM | POLLERR; runtime = substream->runtime; poll_wait(file, &runtime->sleep, wait); @@ -3200,7 +3200,7 @@ static unsigned int snd_pcm_capture_poll(struct file *file, poll_table * wait) substream = pcm_file->substream; if (PCM_RUNTIME_CHECK(substream)) - return -ENXIO; + return POLLIN | POLLRDNORM | POLLERR; runtime = substream->runtime; poll_wait(file, &runtime->sleep, wait); diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c deleted file mode 100644 index f3420d11a12f..000000000000 --- 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 7e0aabb808a6..639544b4fb04 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -47,8 +47,6 @@ int seq_default_timer_card = -1; 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 6469bedda2f3..e722022d325d 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 @@ -1225,6 +1223,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, tu->tstamp = *tstamp; if ((tu->filter & (1 << event)) == 0 || !tu->tread) return; + memset(&r1, 0, sizeof(r1)); r1.event = event; r1.tstamp = *tstamp; r1.val = resolution; @@ -1267,6 +1266,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, } if ((tu->filter & (1 << SNDRV_TIMER_EVENT_RESOLUTION)) && tu->last_resolution != resolution) { + memset(&r1, 0, sizeof(r1)); r1.event = SNDRV_TIMER_EVENT_RESOLUTION; r1.tstamp = tstamp; r1.val = resolution; @@ -1739,6 +1739,7 @@ static int snd_timer_user_params(struct file *file, if (tu->timeri->flags & SNDRV_TIMER_IFLG_EARLY_EVENT) { if (tu->tread) { struct snd_timer_tread tread; + memset(&tread, 0, sizeof(tread)); tread.event = SNDRV_TIMER_EVENT_EARLY; tread.tstamp.tv_sec = 0; tread.tstamp.tv_nsec = 0; diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 2a779c2f63ab..ab894ed1ff67 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -134,6 +134,7 @@ config SND_FIREWIRE_TASCAM 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/Makefile b/sound/firewire/Makefile index 003c09029786..0ee1fb115d88 100644 --- a/sound/firewire/Makefile +++ b/sound/firewire/Makefile @@ -1,3 +1,6 @@ +# To find a header included by define_trace.h. +CFLAGS_amdtp-stream.o := -I$(src) + snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \ fcp.o cmp.o amdtp-stream.o amdtp-am824.o snd-isight-objs := isight.o diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h new file mode 100644 index 000000000000..16225792b722 --- /dev/null +++ b/sound/firewire/amdtp-stream-trace.h @@ -0,0 +1,110 @@ +/* + * amdtp-stream-trace.h - tracepoint definitions to dump a part of packet data + * + * Copyright (c) 2016 Takashi Sakamoto + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#undef TRACE_SYSTEM +#define TRACE_SYSTEM snd_firewire_lib + +#if !defined(_AMDTP_STREAM_TRACE_H) || defined(TRACE_HEADER_MULTI_READ) +#define _AMDTP_STREAM_TRACE_H + +#include <linux/tracepoint.h> + +TRACE_EVENT(in_packet, + TP_PROTO(const struct amdtp_stream *s, u32 cycles, u32 cip_header[2], unsigned int payload_quadlets, unsigned int index), + TP_ARGS(s, cycles, cip_header, payload_quadlets, index), + TP_STRUCT__entry( + __field(unsigned int, second) + __field(unsigned int, cycle) + __field(int, channel) + __field(int, src) + __field(int, dest) + __field(u32, cip_header0) + __field(u32, cip_header1) + __field(unsigned int, payload_quadlets) + __field(unsigned int, packet_index) + __field(bool, irq) + __field(unsigned int, index) + ), + TP_fast_assign( + __entry->second = cycles / CYCLES_PER_SECOND; + __entry->cycle = cycles % CYCLES_PER_SECOND; + __entry->channel = s->context->channel; + __entry->src = fw_parent_device(s->unit)->node_id; + __entry->dest = fw_parent_device(s->unit)->card->node_id; + __entry->cip_header0 = cip_header[0]; + __entry->cip_header1 = cip_header[1]; + __entry->payload_quadlets = payload_quadlets; + __entry->packet_index = s->packet_index; + __entry->irq = in_interrupt(); + __entry->index = index; + ), + TP_printk( + "%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u", + __entry->second, + __entry->cycle, + __entry->src, + __entry->dest, + __entry->channel, + __entry->cip_header0, + __entry->cip_header1, + __entry->payload_quadlets, + __entry->packet_index, + __entry->irq, + __entry->index) +); + +TRACE_EVENT(out_packet, + TP_PROTO(const struct amdtp_stream *s, u32 cycles, __be32 *cip_header, unsigned int payload_length, unsigned int index), + TP_ARGS(s, cycles, cip_header, payload_length, index), + TP_STRUCT__entry( + __field(unsigned int, second) + __field(unsigned int, cycle) + __field(int, channel) + __field(int, src) + __field(int, dest) + __field(u32, cip_header0) + __field(u32, cip_header1) + __field(unsigned int, payload_quadlets) + __field(unsigned int, packet_index) + __field(bool, irq) + __field(unsigned int, index) + ), + TP_fast_assign( + __entry->second = cycles / CYCLES_PER_SECOND; + __entry->cycle = cycles % CYCLES_PER_SECOND; + __entry->channel = s->context->channel; + __entry->src = fw_parent_device(s->unit)->card->node_id; + __entry->dest = fw_parent_device(s->unit)->node_id; + __entry->cip_header0 = be32_to_cpu(cip_header[0]); + __entry->cip_header1 = be32_to_cpu(cip_header[1]); + __entry->payload_quadlets = payload_length / 4; + __entry->packet_index = s->packet_index; + __entry->irq = in_interrupt(); + __entry->index = index; + ), + TP_printk( + "%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u", + __entry->second, + __entry->cycle, + __entry->src, + __entry->dest, + __entry->channel, + __entry->cip_header0, + __entry->cip_header1, + __entry->payload_quadlets, + __entry->packet_index, + __entry->irq, + __entry->index) +); + +#endif + +#undef TRACE_INCLUDE_PATH +#define TRACE_INCLUDE_PATH . +#undef TRACE_INCLUDE_FILE +#define TRACE_INCLUDE_FILE amdtp-stream-trace +#include <trace/define_trace.h> diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index ed2902609a4c..00060c4a9deb 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -19,6 +19,10 @@ #define CYCLES_PER_SECOND 8000 #define TICKS_PER_SECOND (TICKS_PER_CYCLE * CYCLES_PER_SECOND) +/* Always support Linux tracing subsystem. */ +#define CREATE_TRACE_POINTS +#include "amdtp-stream-trace.h" + #define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 microseconds */ /* isochronous header parameters */ @@ -87,7 +91,6 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, init_waitqueue_head(&s->callback_wait); s->callbacked = false; - s->sync_slave = NULL; s->fmt = fmt; s->process_data_blocks = process_data_blocks; @@ -102,6 +105,10 @@ EXPORT_SYMBOL(amdtp_stream_init); */ 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); @@ -244,7 +251,6 @@ void amdtp_stream_pcm_prepare(struct amdtp_stream *s) tasklet_kill(&s->period_tasklet); s->pcm_buffer_pointer = 0; s->pcm_period_pointer = 0; - s->pointer_flush = true; } EXPORT_SYMBOL(amdtp_stream_pcm_prepare); @@ -349,7 +355,6 @@ static void update_pcm_pointers(struct amdtp_stream *s, s->pcm_period_pointer += frames; if (s->pcm_period_pointer >= pcm->runtime->period_size) { s->pcm_period_pointer -= pcm->runtime->period_size; - s->pointer_flush = false; tasklet_hi_schedule(&s->period_tasklet); } } @@ -363,9 +368,8 @@ static void pcm_period_tasklet(unsigned long data) snd_pcm_period_elapsed(pcm); } -static int queue_packet(struct amdtp_stream *s, - unsigned int header_length, - unsigned int payload_length, bool skip) +static int queue_packet(struct amdtp_stream *s, unsigned int header_length, + unsigned int payload_length) { struct fw_iso_packet p = {0}; int err = 0; @@ -376,8 +380,10 @@ static int queue_packet(struct amdtp_stream *s, p.interrupt = IS_ALIGNED(s->packet_index + 1, INTERRUPT_INTERVAL); p.tag = TAG_CIP; p.header_length = header_length; - p.payload_length = (!skip) ? payload_length : 0; - p.skip = skip; + if (payload_length > 0) + p.payload_length = payload_length; + else + p.skip = true; err = fw_iso_context_queue(s->context, &p, &s->buffer.iso_buffer, s->buffer.packets[s->packet_index].offset); if (err < 0) { @@ -392,27 +398,30 @@ end: } static inline int queue_out_packet(struct amdtp_stream *s, - unsigned int payload_length, bool skip) + unsigned int payload_length) { - return queue_packet(s, OUT_PACKET_HEADER_SIZE, - payload_length, skip); + return queue_packet(s, OUT_PACKET_HEADER_SIZE, payload_length); } static inline int queue_in_packet(struct amdtp_stream *s) { return queue_packet(s, IN_PACKET_HEADER_SIZE, - amdtp_stream_get_max_payload(s), false); + amdtp_stream_get_max_payload(s)); } -static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, - unsigned int syt) +static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle, + unsigned int index) { __be32 *buffer; + unsigned int syt; + unsigned int data_blocks; unsigned int payload_length; unsigned int pcm_frames; struct snd_pcm_substream *pcm; buffer = s->buffer.packets[s->packet_index].buffer; + syt = calculate_syt(s, cycle); + data_blocks = calculate_data_blocks(s, syt); pcm_frames = s->process_data_blocks(s, buffer + 2, data_blocks, &syt); buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) | @@ -424,9 +433,11 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, (syt & CIP_SYT_MASK)); s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff; - payload_length = 8 + data_blocks * 4 * s->data_block_quadlets; - if (queue_out_packet(s, payload_length, false) < 0) + + trace_out_packet(s, cycle, buffer, payload_length, index); + + if (queue_out_packet(s, payload_length) < 0) return -EIO; pcm = ACCESS_ONCE(s->pcm); @@ -438,19 +449,24 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, } static int handle_in_packet(struct amdtp_stream *s, - unsigned int payload_quadlets, __be32 *buffer, - unsigned int *data_blocks, unsigned int syt) + unsigned int payload_quadlets, unsigned int cycle, + unsigned int index) { + __be32 *buffer; u32 cip_header[2]; - unsigned int fmt, fdf; + unsigned int fmt, fdf, syt; unsigned int data_block_quadlets, data_block_counter, dbc_interval; + unsigned int data_blocks; struct snd_pcm_substream *pcm; unsigned int pcm_frames; bool lost; + buffer = s->buffer.packets[s->packet_index].buffer; cip_header[0] = be32_to_cpu(buffer[0]); cip_header[1] = be32_to_cpu(buffer[1]); + trace_in_packet(s, cycle, cip_header, payload_quadlets, index); + /* * This module supports 'Two-quadlet CIP header with SYT field'. * For convenience, also check FMT field is AM824 or not. @@ -460,7 +476,7 @@ static int handle_in_packet(struct amdtp_stream *s, dev_info_ratelimited(&s->unit->device, "Invalid CIP header for AMDTP: %08X:%08X\n", cip_header[0], cip_header[1]); - *data_blocks = 0; + data_blocks = 0; pcm_frames = 0; goto end; } @@ -471,7 +487,7 @@ static int handle_in_packet(struct amdtp_stream *s, dev_info_ratelimited(&s->unit->device, "Detect unexpected protocol: %08x %08x\n", cip_header[0], cip_header[1]); - *data_blocks = 0; + data_blocks = 0; pcm_frames = 0; goto end; } @@ -480,7 +496,7 @@ static int handle_in_packet(struct amdtp_stream *s, fdf = (cip_header[1] & CIP_FDF_MASK) >> CIP_FDF_SHIFT; if (payload_quadlets < 3 || (fmt == CIP_FMT_AM && fdf == AMDTP_FDF_NO_DATA)) { - *data_blocks = 0; + data_blocks = 0; } else { data_block_quadlets = (cip_header[0] & CIP_DBS_MASK) >> CIP_DBS_SHIFT; @@ -494,12 +510,12 @@ static int handle_in_packet(struct amdtp_stream *s, if (s->flags & CIP_WRONG_DBS) data_block_quadlets = s->data_block_quadlets; - *data_blocks = (payload_quadlets - 2) / data_block_quadlets; + data_blocks = (payload_quadlets - 2) / data_block_quadlets; } /* Check data block counter continuity */ data_block_counter = cip_header[0] & CIP_DBC_MASK; - if (*data_blocks == 0 && (s->flags & CIP_EMPTY_HAS_WRONG_DBC) && + if (data_blocks == 0 && (s->flags & CIP_EMPTY_HAS_WRONG_DBC) && s->data_block_counter != UINT_MAX) data_block_counter = s->data_block_counter; @@ -510,10 +526,10 @@ static int handle_in_packet(struct amdtp_stream *s, } else if (!(s->flags & CIP_DBC_IS_END_EVENT)) { lost = data_block_counter != s->data_block_counter; } else { - if ((*data_blocks > 0) && (s->tx_dbc_interval > 0)) + if (data_blocks > 0 && s->tx_dbc_interval > 0) dbc_interval = s->tx_dbc_interval; else - dbc_interval = *data_blocks; + dbc_interval = data_blocks; lost = data_block_counter != ((s->data_block_counter + dbc_interval) & 0xff); @@ -526,13 +542,14 @@ static int handle_in_packet(struct amdtp_stream *s, return -EIO; } - pcm_frames = s->process_data_blocks(s, buffer + 2, *data_blocks, &syt); + syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK; + pcm_frames = s->process_data_blocks(s, buffer + 2, data_blocks, &syt); if (s->flags & CIP_DBC_IS_END_EVENT) s->data_block_counter = data_block_counter; else s->data_block_counter = - (data_block_counter + *data_blocks) & 0xff; + (data_block_counter + data_blocks) & 0xff; end: if (queue_in_packet(s) < 0) return -EIO; @@ -544,29 +561,50 @@ end: return 0; } -static void out_stream_callback(struct fw_iso_context *context, u32 cycle, +/* + * In CYCLE_TIMER register of IEEE 1394, 7 bits are used to represent second. On + * the other hand, in DMA descriptors of 1394 OHCI, 3 bits are used to represent + * it. Thus, via Linux firewire subsystem, we can get the 3 bits for second. + */ +static inline u32 compute_cycle_count(u32 tstamp) +{ + return (((tstamp >> 13) & 0x07) * 8000) + (tstamp & 0x1fff); +} + +static inline u32 increment_cycle_count(u32 cycle, unsigned int addend) +{ + cycle += addend; + if (cycle >= 8 * CYCLES_PER_SECOND) + cycle -= 8 * CYCLES_PER_SECOND; + return cycle; +} + +static inline u32 decrement_cycle_count(u32 cycle, unsigned int subtrahend) +{ + if (cycle < subtrahend) + cycle += 8 * CYCLES_PER_SECOND; + return cycle - subtrahend; +} + +static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, size_t header_length, void *header, void *private_data) { struct amdtp_stream *s = private_data; - unsigned int i, syt, packets = header_length / 4; - unsigned int data_blocks; + unsigned int i, packets = header_length / 4; + u32 cycle; if (s->packet_index < 0) return; - /* - * Compute the cycle of the last queued packet. - * (We need only the four lowest bits for the SYT, so we can ignore - * that bits 0-11 must wrap around at 3072.) - */ - cycle += QUEUE_LENGTH - packets; + cycle = compute_cycle_count(tstamp); - for (i = 0; i < packets; ++i) { - syt = calculate_syt(s, ++cycle); - data_blocks = calculate_data_blocks(s, syt); + /* Align to actual cycle count for the last packet. */ + cycle = increment_cycle_count(cycle, QUEUE_LENGTH - packets); - if (handle_out_packet(s, data_blocks, syt) < 0) { + for (i = 0; i < packets; ++i) { + cycle = increment_cycle_count(cycle, 1); + if (handle_out_packet(s, cycle, i) < 0) { s->packet_index = -1; amdtp_stream_pcm_abort(s); return; @@ -576,15 +614,15 @@ static void out_stream_callback(struct fw_iso_context *context, u32 cycle, fw_iso_context_queue_flush(s->context); } -static void in_stream_callback(struct fw_iso_context *context, u32 cycle, +static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, size_t header_length, void *header, void *private_data) { struct amdtp_stream *s = private_data; - unsigned int p, syt, packets; + unsigned int i, packets; unsigned int payload_quadlets, max_payload_quadlets; - unsigned int data_blocks; - __be32 *buffer, *headers = header; + __be32 *headers = header; + u32 cycle; if (s->packet_index < 0) return; @@ -592,70 +630,44 @@ static void in_stream_callback(struct fw_iso_context *context, u32 cycle, /* The number of packets in buffer */ packets = header_length / IN_PACKET_HEADER_SIZE; + cycle = compute_cycle_count(tstamp); + + /* Align to actual cycle count for the last packet. */ + cycle = decrement_cycle_count(cycle, packets); + /* For buffer-over-run prevention. */ max_payload_quadlets = amdtp_stream_get_max_payload(s) / 4; - for (p = 0; p < packets; p++) { - buffer = s->buffer.packets[s->packet_index].buffer; + for (i = 0; i < packets; i++) { + cycle = increment_cycle_count(cycle, 1); /* The number of quadlets in this packet */ payload_quadlets = - (be32_to_cpu(headers[p]) >> ISO_DATA_LENGTH_SHIFT) / 4; + (be32_to_cpu(headers[i]) >> ISO_DATA_LENGTH_SHIFT) / 4; if (payload_quadlets > max_payload_quadlets) { dev_err(&s->unit->device, "Detect jumbo payload: %02x %02x\n", payload_quadlets, max_payload_quadlets); - s->packet_index = -1; break; } - syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK; - if (handle_in_packet(s, payload_quadlets, buffer, - &data_blocks, syt) < 0) { - s->packet_index = -1; + if (handle_in_packet(s, payload_quadlets, cycle, i) < 0) break; - } - - /* Process sync slave stream */ - if (s->sync_slave && s->sync_slave->callbacked) { - if (handle_out_packet(s->sync_slave, - data_blocks, syt) < 0) { - s->packet_index = -1; - break; - } - } } - /* Queueing error or detecting discontinuity */ - if (s->packet_index < 0) { + /* Queueing error or detecting invalid payload. */ + if (i < packets) { + s->packet_index = -1; amdtp_stream_pcm_abort(s); - - /* Abort sync slave. */ - if (s->sync_slave) { - s->sync_slave->packet_index = -1; - amdtp_stream_pcm_abort(s->sync_slave); - } return; } - /* when sync to device, flush the packets for slave stream */ - if (s->sync_slave && s->sync_slave->callbacked) - fw_iso_context_queue_flush(s->sync_slave->context); - fw_iso_context_queue_flush(s->context); } -/* processing is done by master callback */ -static void slave_stream_callback(struct fw_iso_context *context, u32 cycle, - size_t header_length, void *header, - void *private_data) -{ - return; -} - /* this is executed one time */ static void amdtp_stream_first_callback(struct fw_iso_context *context, - u32 cycle, size_t header_length, + u32 tstamp, size_t header_length, void *header, void *private_data) { struct amdtp_stream *s = private_data; @@ -669,12 +681,10 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context, if (s->direction == AMDTP_IN_STREAM) context->callback.sc = in_stream_callback; - else if (s->flags & CIP_SYNC_TO_DEVICE) - context->callback.sc = slave_stream_callback; else context->callback.sc = out_stream_callback; - context->callback.sc(context, cycle, header_length, header, s); + context->callback.sc(context, tstamp, header_length, header, s); } /** @@ -713,8 +723,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) goto err_unlock; } - if (s->direction == AMDTP_IN_STREAM && - s->flags & CIP_SKIP_INIT_DBC_CHECK) + if (s->direction == AMDTP_IN_STREAM) s->data_block_counter = UINT_MAX; else s->data_block_counter = 0; @@ -755,7 +764,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) if (s->direction == AMDTP_IN_STREAM) err = queue_in_packet(s); else - err = queue_out_packet(s, 0, true); + err = queue_out_packet(s, 0); if (err < 0) goto err_context; } while (s->packet_index > 0); @@ -794,11 +803,24 @@ EXPORT_SYMBOL(amdtp_stream_start); */ unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s) { - /* this optimization is allowed to be racy */ - if (s->pointer_flush && amdtp_stream_running(s)) + /* + * This function is called in software IRQ context of period_tasklet or + * process context. + * + * When the software IRQ context was scheduled by software IRQ context + * of IR/IT contexts, queued packets were already handled. Therefore, + * no need to flush the queue in buffer anymore. + * + * When the process context reach here, some packets will be already + * queued in the buffer. These packets should be handled immediately + * to keep better granularity of PCM pointer. + * + * Later, the process context will sometimes schedules software IRQ + * context of the period_tasklet. Then, no need to flush the queue by + * the same reason as described for IR/IT contexts. + */ + if (!in_interrupt() && amdtp_stream_running(s)) fw_iso_context_flush_completions(s->context); - else - s->pointer_flush = true; return ACCESS_ONCE(s->pcm_buffer_pointer); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 8775704a3665..c1bc7fad056e 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -17,8 +17,6 @@ * @CIP_BLOCKING: In blocking mode, each packet contains either zero or * SYT_INTERVAL samples, with these two types alternating so that * the overall sample rate comes out right. - * @CIP_SYNC_TO_DEVICE: In sync to device mode, time stamp in out packets is - * generated by in packets. Defaultly this driver generates timestamp. * @CIP_EMPTY_WITH_TAG0: Only for in-stream. Empty in-packets have TAG0. * @CIP_DBC_IS_END_EVENT: Only for in-stream. The value of dbc in an in-packet * corresponds to the end of event in the packet. Out of IEC 61883. @@ -26,8 +24,6 @@ * The value of data_block_quadlets is used instead of reported value. * @CIP_SKIP_DBC_ZERO_CHECK: Only for in-stream. Packets with zero in dbc is * skipped for detecting discontinuity. - * @CIP_SKIP_INIT_DBC_CHECK: Only for in-stream. The value of dbc in first - * packet is not continuous from an initial value. * @CIP_EMPTY_HAS_WRONG_DBC: Only for in-stream. The value of dbc in empty * packet is wrong but the others are correct. * @CIP_JUMBO_PAYLOAD: Only for in-stream. The number of data blocks in an @@ -37,14 +33,12 @@ enum cip_flags { CIP_NONBLOCKING = 0x00, CIP_BLOCKING = 0x01, - CIP_SYNC_TO_DEVICE = 0x02, - CIP_EMPTY_WITH_TAG0 = 0x04, - CIP_DBC_IS_END_EVENT = 0x08, - CIP_WRONG_DBS = 0x10, - CIP_SKIP_DBC_ZERO_CHECK = 0x20, - CIP_SKIP_INIT_DBC_CHECK = 0x40, - CIP_EMPTY_HAS_WRONG_DBC = 0x80, - CIP_JUMBO_PAYLOAD = 0x100, + CIP_EMPTY_WITH_TAG0 = 0x02, + CIP_DBC_IS_END_EVENT = 0x04, + CIP_WRONG_DBS = 0x08, + CIP_SKIP_DBC_ZERO_CHECK = 0x10, + CIP_EMPTY_HAS_WRONG_DBC = 0x20, + CIP_JUMBO_PAYLOAD = 0x40, }; /** @@ -132,12 +126,10 @@ struct amdtp_stream { struct tasklet_struct period_tasklet; unsigned int pcm_buffer_pointer; unsigned int pcm_period_pointer; - bool pointer_flush; /* To wait for first packet. */ bool callbacked; wait_queue_head_t callback_wait; - struct amdtp_stream *sync_slave; /* For backends to process data blocks. */ void *protocol; @@ -223,23 +215,6 @@ static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc) return sfc & 1; } -static inline void amdtp_stream_set_sync(enum cip_flags sync_mode, - struct amdtp_stream *master, - struct amdtp_stream *slave) -{ - if (sync_mode == CIP_SYNC_TO_DEVICE) { - master->flags |= CIP_SYNC_TO_DEVICE; - slave->flags |= CIP_SYNC_TO_DEVICE; - master->sync_slave = slave; - } else { - master->flags &= ~CIP_SYNC_TO_DEVICE; - slave->flags &= ~CIP_SYNC_TO_DEVICE; - master->sync_slave = NULL; - } - - slave->sync_slave = NULL; -} - /** * amdtp_stream_wait_callback - sleep till callbacked or timeout * @s: the AMDTP stream diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 3e4e0756e3fe..f7e2cbd2a313 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -67,7 +67,7 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS); #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 @@ end: 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 @@ bebob_card_free(struct snd_card *card) { struct snd_bebob *bebob = card->private_data; - snd_bebob_stream_destroy_duplex(bebob); - fw_unit_put(bebob->unit); - - 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_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 @@ check_audiophile_booted(struct fw_unit *unit) 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 @@ bebob_probe(struct fw_unit *unit, 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); + err = name_device(bebob); 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); + goto error; - err = name_device(bebob, entry->vendor_id); + 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; - 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 - err = snd_bebob_stream_discover(bebob); + 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 @@ bebob_probe(struct fw_unit *unit, 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 @@ bebob_probe(struct fw_unit *unit, * 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 @@ bebob_update(struct fw_unit *unit) 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 @@ static void bebob_remove(struct fw_unit *unit) 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 b50bb33d9d46..e7f1bb925b12 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -83,6 +83,10 @@ struct snd_bebob { 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; @@ -90,7 +94,6 @@ struct snd_bebob { bool connected; - struct amdtp_stream *master; struct amdtp_stream tx_stream; struct amdtp_stream rx_stream; struct cmp_connection out_conn; @@ -111,7 +114,6 @@ struct snd_bebob { /* for M-Audio special devices */ void *maudio_special_quirk; - bool deferred_registration; /* For BeBoB version quirk. */ unsigned int version; diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 77cbb02bff34..4d3034a68bdf 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -484,30 +484,6 @@ destroy_both_connections(struct snd_bebob *bebob) } static int -get_sync_mode(struct snd_bebob *bebob, enum cip_flags *sync_mode) -{ - enum snd_bebob_clock_type src; - int err; - - err = snd_bebob_stream_get_clock_src(bebob, &src); - if (err < 0) - return err; - - switch (src) { - case SND_BEBOB_CLOCK_TYPE_INTERNAL: - case SND_BEBOB_CLOCK_TYPE_EXTERNAL: - *sync_mode = CIP_SYNC_TO_DEVICE; - break; - default: - case SND_BEBOB_CLOCK_TYPE_SYT: - *sync_mode = 0; - break; - } - - return 0; -} - -static int start_stream(struct snd_bebob *bebob, struct amdtp_stream *stream, unsigned int rate) { @@ -550,8 +526,6 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) goto end; } - bebob->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK; - /* * BeBoB v3 transfers packets with these qurks: * - In the beginning of streaming, the value of dbc is incremented @@ -584,8 +558,6 @@ end: int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) { const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate; - struct amdtp_stream *master, *slave; - enum cip_flags sync_mode; unsigned int curr_rate; int err = 0; @@ -593,22 +565,11 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) if (bebob->substreams_counter == 0) goto end; - err = get_sync_mode(bebob, &sync_mode); - if (err < 0) - goto end; - if (sync_mode == CIP_SYNC_TO_DEVICE) { - master = &bebob->tx_stream; - slave = &bebob->rx_stream; - } else { - master = &bebob->rx_stream; - slave = &bebob->tx_stream; - } - /* * Considering JACK/FFADO streaming: * TODO: This can be removed hwdep functionality becomes popular. */ - err = check_connection_used_by_others(bebob, master); + err = check_connection_used_by_others(bebob, &bebob->rx_stream); if (err < 0) goto end; @@ -618,11 +579,12 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) * At bus reset, connections should not be broken here. So streams need * to be re-started. This is a reason to use SKIP_INIT_DBC_CHECK flag. */ - if (amdtp_streaming_error(master)) - amdtp_stream_stop(master); - if (amdtp_streaming_error(slave)) - amdtp_stream_stop(slave); - if (!amdtp_stream_running(master) && !amdtp_stream_running(slave)) + if (amdtp_streaming_error(&bebob->rx_stream)) + amdtp_stream_stop(&bebob->rx_stream); + if (amdtp_streaming_error(&bebob->tx_stream)) + amdtp_stream_stop(&bebob->tx_stream); + if (!amdtp_stream_running(&bebob->rx_stream) && + !amdtp_stream_running(&bebob->tx_stream)) break_both_connections(bebob); /* stop streams if rate is different */ @@ -635,16 +597,13 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) if (rate == 0) rate = curr_rate; if (rate != curr_rate) { - amdtp_stream_stop(master); - amdtp_stream_stop(slave); + amdtp_stream_stop(&bebob->rx_stream); + amdtp_stream_stop(&bebob->tx_stream); break_both_connections(bebob); } /* master should be always running */ - if (!amdtp_stream_running(master)) { - amdtp_stream_set_sync(sync_mode, master, slave); - bebob->master = master; - + if (!amdtp_stream_running(&bebob->rx_stream)) { /* * NOTE: * If establishing connections at first, Yamaha GO46 @@ -666,7 +625,7 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) if (err < 0) goto end; - err = start_stream(bebob, master, rate); + err = start_stream(bebob, &bebob->rx_stream, rate); if (err < 0) { dev_err(&bebob->unit->device, "fail to run AMDTP master stream:%d\n", err); @@ -685,15 +644,16 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) dev_err(&bebob->unit->device, "fail to ensure sampling rate: %d\n", err); - amdtp_stream_stop(master); + amdtp_stream_stop(&bebob->rx_stream); break_both_connections(bebob); goto end; } } /* wait first callback */ - if (!amdtp_stream_wait_callback(master, CALLBACK_TIMEOUT)) { - amdtp_stream_stop(master); + if (!amdtp_stream_wait_callback(&bebob->rx_stream, + CALLBACK_TIMEOUT)) { + amdtp_stream_stop(&bebob->rx_stream); break_both_connections(bebob); err = -ETIMEDOUT; goto end; @@ -701,20 +661,21 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) } /* start slave if needed */ - if (!amdtp_stream_running(slave)) { - err = start_stream(bebob, slave, rate); + if (!amdtp_stream_running(&bebob->tx_stream)) { + err = start_stream(bebob, &bebob->tx_stream, rate); if (err < 0) { dev_err(&bebob->unit->device, "fail to run AMDTP slave stream:%d\n", err); - amdtp_stream_stop(master); + amdtp_stream_stop(&bebob->rx_stream); break_both_connections(bebob); goto end; } /* wait first callback */ - if (!amdtp_stream_wait_callback(slave, CALLBACK_TIMEOUT)) { - amdtp_stream_stop(slave); - amdtp_stream_stop(master); + if (!amdtp_stream_wait_callback(&bebob->tx_stream, + CALLBACK_TIMEOUT)) { + amdtp_stream_stop(&bebob->tx_stream); + amdtp_stream_stop(&bebob->rx_stream); break_both_connections(bebob); err = -ETIMEDOUT; } @@ -725,22 +686,12 @@ end: void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob) { - struct amdtp_stream *master, *slave; - - if (bebob->master == &bebob->rx_stream) { - slave = &bebob->tx_stream; - master = &bebob->rx_stream; - } else { - slave = &bebob->rx_stream; - master = &bebob->tx_stream; - } - if (bebob->substreams_counter == 0) { - amdtp_stream_pcm_abort(master); - amdtp_stream_stop(master); + amdtp_stream_pcm_abort(&bebob->rx_stream); + amdtp_stream_stop(&bebob->rx_stream); - amdtp_stream_pcm_abort(slave); - amdtp_stream_stop(slave); + amdtp_stream_pcm_abort(&bebob->tx_stream); + amdtp_stream_stop(&bebob->tx_stream); break_both_connections(bebob); } diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 8b64aef31a86..25e9f77275c4 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -20,8 +20,6 @@ MODULE_LICENSE("GPL v2"); #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 @@ static void do_registration(struct work_struct *work) 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 @@ static void do_registration(struct work_struct *work) 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 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) 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 @@ static void dice_bus_reset(struct fw_unit *unit) /* 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); @@ -335,6 +317,13 @@ static const struct ieee1394_device_id dice_id_table[] = { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, }, + /* M-Audio Profire 610/2626 has a different value in version field. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID, + .vendor_id = 0x000d6c, + .specifier_id = 0x000d6c, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, dice_id_table); diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c index 0ac92aba5bc1..b3cffd01a19f 100644 --- a/sound/firewire/digi00x/amdtp-dot.c +++ b/sound/firewire/digi00x/amdtp-dot.c @@ -421,7 +421,7 @@ int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit, /* Use different mode between incoming/outgoing. */ if (dir == AMDTP_IN_STREAM) { - flags = CIP_NONBLOCKING | CIP_SKIP_INIT_DBC_CHECK; + flags = CIP_NONBLOCKING; process_data_blocks = process_tx_data_blocks; } else { flags = CIP_BLOCKING; diff --git a/sound/firewire/digi00x/digi00x-transaction.c b/sound/firewire/digi00x/digi00x-transaction.c index 554324d8c602..735d35640807 100644 --- a/sound/firewire/digi00x/digi00x-transaction.c +++ b/sound/firewire/digi00x/digi00x-transaction.c @@ -126,12 +126,17 @@ int snd_dg00x_transaction_register(struct snd_dg00x *dg00x) 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 1f33b7a1fca4..cc4776c6ded3 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -40,10 +40,8 @@ static int name_card(struct snd_dg00x *dg00x) 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 @@ static void dg00x_card_free(struct snd_card *card) 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; - int err; + dg00x_free(card->private_data); +} - /* create card */ - err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, - sizeof(struct snd_dg00x), &card); - if (err < 0) - return err; - card->private_free = dg00x_card_free; +static void do_registration(struct work_struct *work) +{ + struct snd_dg00x *dg00x = + container_of(work, struct snd_dg00x, dwork.work); + int err; - /* initialize myself */ - dg00x = card->private_data; - dg00x->card = card; - dg00x->unit = fw_unit_get(unit); + if (dg00x->registered) + return; - mutex_init(&dg00x->mutex); - spin_lock_init(&dg00x->lock); - init_waitqueue_head(&dg00x->hwdep_wait); + err = snd_card_new(&dg00x->unit->device, -1, NULL, THIS_MODULE, 0, + &dg00x->card); + if (err < 0) + return; err = name_card(dg00x); if (err < 0) @@ -101,35 +95,86 @@ static int snd_dg00x_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_card_register(card); + err = snd_card_register(dg00x->card); if (err < 0) goto error; - dev_set_drvdata(&unit->device, dg00x); + dg00x->card->private_free = dg00x_card_free; + dg00x->card->private_data = dg00x; + dg00x->registered = true; - return err; + return; error: - snd_card_free(card); - return err; + 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); + + 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 907e73993677..2cd465c0caae 100644 --- a/sound/firewire/digi00x/digi00x.h +++ b/sound/firewire/digi00x/digi00x.h @@ -37,6 +37,9 @@ struct snd_dg00x { 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 8f27b67503c8..71a0613d3da0 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -168,11 +168,34 @@ get_hardware_info(struct snd_efw *efw) 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 @@ efw_card_free(struct snd_card *card) { 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 @@ efw_probe(struct fw_unit *unit, 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 @@ efw_probe(struct fw_unit *unit, 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 @@ efw_probe(struct fw_unit *unit, 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; - } - - dev_set_drvdata(&unit->device, efw); -end: + set_bit(card_index, devices_used); 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 96c4e0c6a9bd..03ed35237e2b 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -65,6 +65,9 @@ struct snd_efw { struct mutex mutex; spinlock_t lock; + bool registered; + struct delayed_work dwork; + /* for transaction */ u32 seqnum; bool resp_addr_changable; @@ -81,7 +84,6 @@ struct snd_efw { unsigned int pcm_capture_channels[SND_EFW_MULTIPLIER_MODES]; unsigned int pcm_playback_channels[SND_EFW_MULTIPLIER_MODES]; - struct amdtp_stream *master; struct amdtp_stream tx_stream; struct amdtp_stream rx_stream; struct cmp_connection out_conn; diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index 425db8d88235..ee47924aef0d 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -121,23 +121,6 @@ destroy_stream(struct snd_efw *efw, struct amdtp_stream *stream) } static int -get_sync_mode(struct snd_efw *efw, enum cip_flags *sync_mode) -{ - enum snd_efw_clock_source clock_source; - int err; - - err = snd_efw_command_get_clock_source(efw, &clock_source); - if (err < 0) - return err; - - if (clock_source == SND_EFW_CLOCK_SOURCE_SYTMATCH) - return -ENOSYS; - - *sync_mode = CIP_SYNC_TO_DEVICE; - return 0; -} - -static int check_connection_used_by_others(struct snd_efw *efw, struct amdtp_stream *s) { struct cmp_connection *conn; @@ -208,9 +191,6 @@ end: int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate) { - struct amdtp_stream *master, *slave; - unsigned int slave_substreams; - enum cip_flags sync_mode; unsigned int curr_rate; int err = 0; @@ -218,32 +198,19 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate) if (efw->playback_substreams == 0 && efw->capture_substreams == 0) goto end; - err = get_sync_mode(efw, &sync_mode); - if (err < 0) - goto end; - if (sync_mode == CIP_SYNC_TO_DEVICE) { - master = &efw->tx_stream; - slave = &efw->rx_stream; - slave_substreams = efw->playback_substreams; - } else { - master = &efw->rx_stream; - slave = &efw->tx_stream; - slave_substreams = efw->capture_substreams; - } - /* * Considering JACK/FFADO streaming: * TODO: This can be removed hwdep functionality becomes popular. */ - err = check_connection_used_by_others(efw, master); + err = check_connection_used_by_others(efw, &efw->rx_stream); if (err < 0) goto end; /* packet queueing error */ - if (amdtp_streaming_error(slave)) - stop_stream(efw, slave); - if (amdtp_streaming_error(master)) - stop_stream(efw, master); + if (amdtp_streaming_error(&efw->tx_stream)) + stop_stream(efw, &efw->tx_stream); + if (amdtp_streaming_error(&efw->rx_stream)) + stop_stream(efw, &efw->rx_stream); /* stop streams if rate is different */ err = snd_efw_command_get_sampling_rate(efw, &curr_rate); @@ -252,20 +219,17 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate) if (rate == 0) rate = curr_rate; if (rate != curr_rate) { - stop_stream(efw, slave); - stop_stream(efw, master); + stop_stream(efw, &efw->tx_stream); + stop_stream(efw, &efw->rx_stream); } /* master should be always running */ - if (!amdtp_stream_running(master)) { - amdtp_stream_set_sync(sync_mode, master, slave); - efw->master = master; - + if (!amdtp_stream_running(&efw->rx_stream)) { err = snd_efw_command_set_sampling_rate(efw, rate); if (err < 0) goto end; - err = start_stream(efw, master, rate); + err = start_stream(efw, &efw->rx_stream, rate); if (err < 0) { dev_err(&efw->unit->device, "fail to start AMDTP master stream:%d\n", err); @@ -274,12 +238,13 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate) } /* start slave if needed */ - if (slave_substreams > 0 && !amdtp_stream_running(slave)) { - err = start_stream(efw, slave, rate); + if (efw->capture_substreams > 0 && + !amdtp_stream_running(&efw->tx_stream)) { + err = start_stream(efw, &efw->tx_stream, rate); if (err < 0) { dev_err(&efw->unit->device, "fail to start AMDTP slave stream:%d\n", err); - stop_stream(efw, master); + stop_stream(efw, &efw->rx_stream); } } end: @@ -288,26 +253,11 @@ end: void snd_efw_stream_stop_duplex(struct snd_efw *efw) { - struct amdtp_stream *master, *slave; - unsigned int master_substreams, slave_substreams; - - if (efw->master == &efw->rx_stream) { - slave = &efw->tx_stream; - master = &efw->rx_stream; - slave_substreams = efw->capture_substreams; - master_substreams = efw->playback_substreams; - } else { - slave = &efw->rx_stream; - master = &efw->tx_stream; - slave_substreams = efw->playback_substreams; - master_substreams = efw->capture_substreams; - } - - if (slave_substreams == 0) { - stop_stream(efw, slave); + if (efw->capture_substreams == 0) { + stop_stream(efw, &efw->tx_stream); - if (master_substreams == 0) - stop_stream(efw, master); + if (efw->playback_substreams == 0) + stop_stream(efw, &efw->rx_stream); } } diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c index f80aafa44c89..ca4dfcf43175 100644 --- a/sound/firewire/lib.c +++ b/sound/firewire/lib.c @@ -67,6 +67,38 @@ int snd_fw_transaction(struct fw_unit *unit, int tcode, } 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 f3f6f84c48d6..f6769312ebfc 100644 --- a/sound/firewire/lib.h +++ b/sound/firewire/lib.h @@ -22,6 +22,9 @@ static inline bool rcode_is_permanent_error(int rcode) 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-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 7cb5743c073b..d9361f352133 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -242,8 +242,7 @@ int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw, * blocks than IEC 61883-6 defines. */ if (stream == &oxfw->tx_stream) { - oxfw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK | - CIP_JUMBO_PAYLOAD; + oxfw->tx_stream.flags |= CIP_JUMBO_PAYLOAD; if (oxfw->wrong_dbs) oxfw->tx_stream.flags |= CIP_WRONG_DBS; } diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index abedc2207261..e629b88f7d93 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -118,15 +118,8 @@ end: 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 @@ static void oxfw_card_free(struct snd_card *card) 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); @@ -205,41 +209,39 @@ static int detect_quirks(struct snd_oxfw *oxfw) 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; + return; - 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 = name_card(oxfw); + if (err < 0) + goto error; - err = snd_oxfw_stream_discover(oxfw); + err = detect_quirks(oxfw); if (err < 0) goto error; - err = name_card(oxfw); + err = snd_oxfw_stream_discover(oxfw); if (err < 0) goto error; - err = detect_quirks(oxfw); + 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) @@ -255,54 +257,97 @@ static int oxfw_probe(struct fw_unit *unit, 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 9beecc214767..2047dcb27625 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-stream.c b/sound/firewire/tascam/tascam-stream.c index 0e6dd5c61f53..4ad3bd7fd445 100644 --- a/sound/firewire/tascam/tascam-stream.c +++ b/sound/firewire/tascam/tascam-stream.c @@ -381,19 +381,17 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) if (err < 0) return err; if (curr_rate != rate || - amdtp_streaming_error(&tscm->tx_stream) || - amdtp_streaming_error(&tscm->rx_stream)) { + amdtp_streaming_error(&tscm->rx_stream) || + amdtp_streaming_error(&tscm->tx_stream)) { finish_session(tscm); - amdtp_stream_stop(&tscm->tx_stream); amdtp_stream_stop(&tscm->rx_stream); + amdtp_stream_stop(&tscm->tx_stream); release_resources(tscm); } - if (!amdtp_stream_running(&tscm->tx_stream)) { - amdtp_stream_set_sync(CIP_SYNC_TO_DEVICE, - &tscm->tx_stream, &tscm->rx_stream); + if (!amdtp_stream_running(&tscm->rx_stream)) { err = keep_resources(tscm, rate); if (err < 0) goto error; @@ -406,27 +404,27 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) if (err < 0) goto error; - err = amdtp_stream_start(&tscm->tx_stream, - tscm->tx_resources.channel, + err = amdtp_stream_start(&tscm->rx_stream, + tscm->rx_resources.channel, fw_parent_device(tscm->unit)->max_speed); if (err < 0) goto error; - if (!amdtp_stream_wait_callback(&tscm->tx_stream, + if (!amdtp_stream_wait_callback(&tscm->rx_stream, CALLBACK_TIMEOUT)) { err = -ETIMEDOUT; goto error; } } - if (!amdtp_stream_running(&tscm->rx_stream)) { - err = amdtp_stream_start(&tscm->rx_stream, - tscm->rx_resources.channel, + if (!amdtp_stream_running(&tscm->tx_stream)) { + err = amdtp_stream_start(&tscm->tx_stream, + tscm->tx_resources.channel, fw_parent_device(tscm->unit)->max_speed); if (err < 0) goto error; - if (!amdtp_stream_wait_callback(&tscm->rx_stream, + if (!amdtp_stream_wait_callback(&tscm->tx_stream, CALLBACK_TIMEOUT)) { err = -ETIMEDOUT; goto error; @@ -435,8 +433,8 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) return 0; error: - amdtp_stream_stop(&tscm->tx_stream); amdtp_stream_stop(&tscm->rx_stream); + amdtp_stream_stop(&tscm->tx_stream); finish_session(tscm); release_resources(tscm); diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index e281c338e562..9dc93a7eb9da 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -85,10 +85,8 @@ static int identify_model(struct snd_tscm *tscm) 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 @@ static void tscm_card_free(struct snd_card *card) 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_stream_init_duplex(tscm); + err = snd_tscm_transaction_register(tscm); if (err < 0) goto error; - err = snd_tscm_create_pcm_devices(tscm); + err = snd_tscm_stream_init_duplex(tscm); if (err < 0) goto error; - err = snd_tscm_transaction_register(tscm); + snd_tscm_proc_init(tscm); + + err = snd_tscm_create_pcm_devices(tscm); if (err < 0) goto error; @@ -146,35 +136,91 @@ static int snd_tscm_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_card_register(card); + err = snd_card_register(tscm->card); if (err < 0) goto error; - dev_set_drvdata(&unit->device, tscm); + /* + * 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 err; + return; error: - snd_card_free(card); - return err; + 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); + + 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 30ab77e924f7..1f61011579a7 100644 --- a/sound/firewire/tascam/tascam.h +++ b/sound/firewire/tascam/tascam.h @@ -51,6 +51,8 @@ struct snd_tscm { 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/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 2433f7c81472..64de0a3d6d93 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -144,6 +144,7 @@ int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) if (!edev) return -ENOMEM; hdev = &edev->hdac; + edev->ebus = ebus; snprintf(name, sizeof(name), "ehdaudio%dD%d", ebus->idx, addr); diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 8c486235c905..9fee464e5d49 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -80,6 +80,22 @@ void snd_hdac_bus_init_cmd_io(struct hdac_bus *bus) } EXPORT_SYMBOL_GPL(snd_hdac_bus_init_cmd_io); +/* wait for cmd dmas till they are stopped */ +static void hdac_wait_for_cmd_dmas(struct hdac_bus *bus) +{ + unsigned long timeout; + + timeout = jiffies + msecs_to_jiffies(100); + while ((snd_hdac_chip_readb(bus, RIRBCTL) & AZX_RBCTL_DMA_EN) + && time_before(jiffies, timeout)) + udelay(10); + + timeout = jiffies + msecs_to_jiffies(100); + while ((snd_hdac_chip_readb(bus, CORBCTL) & AZX_CORBCTL_RUN) + && time_before(jiffies, timeout)) + udelay(10); +} + /** * snd_hdac_bus_stop_cmd_io - clean up CORB/RIRB buffers * @bus: HD-audio core bus @@ -90,6 +106,7 @@ void snd_hdac_bus_stop_cmd_io(struct hdac_bus *bus) /* disable ringbuffer DMAs */ snd_hdac_chip_writeb(bus, RIRBCTL, 0); snd_hdac_chip_writeb(bus, CORBCTL, 0); + hdac_wait_for_cmd_dmas(bus); /* disable unsolicited responses */ snd_hdac_chip_updatel(bus, GCTL, AZX_GCTL_UNSOL, 0); spin_unlock_irq(&bus->reg_lock); diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 607bbeaebddf..c9af022676c2 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -158,22 +158,40 @@ void snd_hdac_i915_set_bclk(struct hdac_bus *bus) } 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 @@ static int pin2port(hda_nid_t pin_nid) * 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 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); /** * 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 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); * 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 @@ int snd_hdac_i915_init(struct hdac_bus *bus) 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_master_del: 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 @@ int snd_hdac_i915_exit(struct hdac_bus *bus) 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 d7ec86263828..c6c75e7e0981 100644 --- a/sound/hda/hdmi_chmap.c +++ b/sound/hda/hdmi_chmap.c @@ -625,13 +625,30 @@ static void hdmi_cea_alloc_to_tlv_chmap(struct hdac_chmap *hchmap, 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 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, 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/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 69f76ff5693d..718d5e3b7806 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -785,6 +785,9 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header) DPRINT (WF_DEBUG_LOAD_PATCH, "downloading patch %d\n", header->number); + if (header->number >= ARRAY_SIZE(dev->patch_status)) + return -EINVAL; + dev->patch_status[header->number] |= WF_SLOT_FILLED; bptr = buf; @@ -809,6 +812,9 @@ wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header) DPRINT (WF_DEBUG_LOAD_PATCH, "downloading program %d\n", header->number); + if (header->number >= ARRAY_SIZE(dev->prog_status)) + return -EINVAL; + dev->prog_status[header->number] = WF_SLOT_USED; /* XXX need to zero existing SLOT_USED bit for program_status[i] @@ -898,6 +904,9 @@ wavefront_send_sample (snd_wavefront_t *dev, header->number = x; } + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + if (header->size) { /* XXX it's a debatable point whether or not RDONLY semantics diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 4667c3232b7f..4a054d720112 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2151,8 +2151,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_SRC)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } if (stream->type != VORTEX_PCM_A3D) { @@ -2162,7 +2161,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, VORTEX_RESOURCE_MIXIN)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } } @@ -2175,8 +2174,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_A3D)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); dev_err(vortex->card->dev, "out of A3D sources. Sorry\n"); return -EBUSY; @@ -2290,8 +2288,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, VORTEX_RESOURCE_MIXOUT)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } if ((src[i] = @@ -2299,8 +2296,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_SRC)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } } diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index a6d6d8d0867a..df5741a78fd2 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -432,7 +432,10 @@ static snd_pcm_uframes_t snd_vortex_pcm_pointer(struct snd_pcm_substream *substr #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 a5d460453d7b..8f945341720b 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -49,7 +49,7 @@ struct ct_timer { 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 @@ static void ct_systimer_prepare(struct ct_timer_instance *ti) #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 @@ static void ct_xfitimer_free_global(struct ct_timer *atimer) 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 0dc44ebb0032..626cd2167d29 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1548,7 +1548,7 @@ static int snd_es1373_line_get(struct snd_kcontrol *kcontrol, 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/Kconfig b/sound/pci/hda/Kconfig index bb02c2d48fd5..7f3b5ed81995 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -50,9 +50,13 @@ config SND_HDA_RECONFIG bool "Allow dynamic codec reconfiguration" help Say Y here to enable the HD-audio codec re-configuration feature. - This adds the sysfs interfaces to allow user to clear the whole - codec configuration, change the codec setup, add extra verbs, - and re-configure the codec dynamically. + It allows user to clear the whole codec configuration, change the + codec setup, add extra verbs, and re-configure the codec dynamically. + + Note that this item alone doesn't provide the sysfs interface, but + enables the feature just for the patch loader below. + If you need the traditional sysfs entries for the manual interaction, + turn on CONFIG_SND_HDA_HWDEP as well. config SND_HDA_INPUT_BEEP bool "Support digital beep via input layer" diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index dfaf1a93fb8a..320445f3bf73 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5434,6 +5434,7 @@ static int dyn_adc_capture_pcm_prepare(struct hda_pcm_stream *hinfo, 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 @@ static int dyn_adc_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, 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 a010d704e0e2..d0d5ad8beac5 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -114,6 +114,9 @@ struct hdmi_ops { 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 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, 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 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, 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 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, * 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 @@ static int hdmi_choose_cvt(struct hda_codec *codec, if (cvt_id) *cvt_id = cvt_idx; - if (mux_id) - *mux_id = mux_idx; return 0; } @@ -1019,9 +1018,6 @@ static void intel_not_share_assigned_cvt_nid(struct hda_codec *codec, 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 @@ static void intel_not_share_assigned_cvt_nid(struct hda_codec *codec, 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 @@ static int hdmi_pcm_open_no_pin(struct hda_pcm_stream *hinfo, 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 @@ static int hdmi_pcm_open_no_pin(struct hda_pcm_stream *hinfo, 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 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, { 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 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, } } - 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 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, 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 @@ static void update_eld(struct hda_codec *codec, * 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 @@ static void sync_eld_via_acomp(struct hda_codec *codec, 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 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, * 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 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, } 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 @@ static const struct hda_pcm_ops generic_ops = { .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 @@ static void hdmi_array_free(struct hdmi_spec *spec) 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 @@ static void generic_hdmi_free(struct hda_codec *codec) 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 @@ static const struct hdmi_ops generic_standard_hdmi_ops = { .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 haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, 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 */ @@ -2236,93 +2318,159 @@ static void intel_pin_eld_notify(void *audio_ptr, int port) 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; + struct hdmi_spec *spec = codec->spec; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; + 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); +} - spec->ops = generic_standard_hdmi_ops; - mutex_init(&spec->pcm_lock); - snd_hdac_register_chmap_ops(&codec->core, &spec->chmap); +/* 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); +} - 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; +/* 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); + } +} - codec->spec = spec; - hdmi_array_init(spec, 4); +/* 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; -#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; - /* 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 + 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. - * For previous platforms, there is no such power well feature. */ - if (is_valleyview_plus(codec) || is_skylake(codec) || - is_broxton(codec)) + if (!is_haswell(codec) && !is_broadwell(codec)) 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.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; } - 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; + + 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; } - /* 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; + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + spec = codec->spec; - generic_hdmi_init_per_pins(codec); + /* For Valleyview/Cherryview, only the display codec is in the display + * power well and can use link_power ops to request/release the power. + */ + codec->core.link_power_control = 1; + codec->depop_delay = 0; + codec->auto_runtime_pm = 1; - 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); + spec->ops.pin_cvt_fixup = i915_pin_cvt_fixup; + + err = hdmi_parse_codec(codec); + if (err < 0) { + generic_spec_free(codec); + return err; } - WARN_ON(spec->dyn_pcm_assign && !codec_has_acomp(codec)); + generic_hdmi_init_per_pins(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; } /* @@ -3492,21 +3640,21 @@ HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), 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/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4918ffa5ba68..002f153bc659 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -342,6 +342,11 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0293: alc_update_coef_idx(codec, 0xa, 1<<13, 0); break; + case 0x10ec0234: + case 0x10ec0274: + case 0x10ec0294: + alc_update_coef_idx(codec, 0x10, 1<<15, 0); + break; case 0x10ec0662: if ((coef & 0x00f0) == 0x0030) alc_update_coef_idx(codec, 0x4, 1<<10, 0); /* EAPD Ctrl */ @@ -2647,6 +2652,7 @@ enum { ALC269_TYPE_ALC255, ALC269_TYPE_ALC256, ALC269_TYPE_ALC225, + ALC269_TYPE_ALC294, }; /* @@ -2677,6 +2683,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC255: case ALC269_TYPE_ALC256: case ALC269_TYPE_ALC225: + case ALC269_TYPE_ALC294: ssids = alc269_ssids; break; default: @@ -6028,6 +6035,11 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0225: spec->codec_variant = ALC269_TYPE_ALC225; break; + case 0x10ec0234: + case 0x10ec0274: + case 0x10ec0294: + spec->codec_variant = ALC269_TYPE_ALC294; + break; } if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { @@ -6942,6 +6954,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0225, "ALC225", patch_alc269), HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269), HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0234, "ALC234", patch_alc269), HDA_CODEC_ENTRY(0x10ec0235, "ALC233", patch_alc269), HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269), HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269), @@ -6952,6 +6965,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0269, "ALC269", patch_alc269), HDA_CODEC_ENTRY(0x10ec0270, "ALC270", patch_alc269), HDA_CODEC_ENTRY(0x10ec0272, "ALC272", patch_alc662), + HDA_CODEC_ENTRY(0x10ec0274, "ALC274", patch_alc269), HDA_CODEC_ENTRY(0x10ec0275, "ALC275", patch_alc269), HDA_CODEC_ENTRY(0x10ec0276, "ALC276", patch_alc269), HDA_CODEC_ENTRY(0x10ec0280, "ALC280", patch_alc269), @@ -6964,6 +6978,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0290, "ALC290", patch_alc269), HDA_CODEC_ENTRY(0x10ec0292, "ALC292", patch_alc269), HDA_CODEC_ENTRY(0x10ec0293, "ALC293", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0294, "ALC294", patch_alc269), HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269), HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861), HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd), diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 8151318a69a2..9720a30dbfff 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 @@ static int snd_intel8x0_inside_vm(struct pci_dev *pci) 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 f3d62020ef66..a80684bdc30d 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -644,7 +644,7 @@ static int lx_pipe_wait_for_state(struct lx6464es *chip, u32 pipe, if (err < 0) return err; - if (current_state == state) + if (!err && current_state == state) return 0; mdelay(1); diff --git a/sound/usb/card.c b/sound/usb/card.c index 3fc63583a537..69860da473ea 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -350,6 +350,7 @@ static int snd_usb_audio_create(struct usb_interface *intf, case USB_SPEED_HIGH: case USB_SPEED_WIRELESS: case USB_SPEED_SUPER: + case USB_SPEED_SUPER_PLUS: break; default: dev_err(&dev->dev, "unknown device speed %d\n", snd_usb_get_speed(dev)); @@ -450,6 +451,9 @@ static int snd_usb_audio_create(struct usb_interface *intf, case USB_SPEED_SUPER: strlcat(card->longname, ", super speed", sizeof(card->longname)); break; + case USB_SPEED_SUPER_PLUS: + strlcat(card->longname, ", super speed plus", sizeof(card->longname)); + break; default: break; } diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 7ccbcaf6a147..26dd5f20f149 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -309,6 +309,9 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, * support reading */ if (snd_usb_get_sample_rate_quirk(chip)) return 0; + /* the firmware is likely buggy, don't repeat to fail too many times */ + if (chip->sample_rate_read_error > 2) + return 0; if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, @@ -316,6 +319,7 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, data, sizeof(data))) < 0) { dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n", iface, fmt->altsetting, ep); + chip->sample_rate_read_error++; return 0; /* some devices don't support reading */ } diff --git a/sound/usb/helper.c b/sound/usb/helper.c index 51ed1ac825fd..7712e2b84183 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -120,6 +120,7 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, case USB_SPEED_HIGH: case USB_SPEED_WIRELESS: case USB_SPEED_SUPER: + case USB_SPEED_SUPER_PLUS: if (get_endpoint(alts, 0)->bInterval >= 1 && get_endpoint(alts, 0)->bInterval <= 4) return get_endpoint(alts, 0)->bInterval - 1; diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 47de8af42f16..7ba92921bf28 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -911,6 +911,7 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, switch (snd_usb_get_speed(ep->umidi->dev)) { case USB_SPEED_HIGH: case USB_SPEED_SUPER: + case USB_SPEED_SUPER_PLUS: count = 1; break; default: diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4f85757009b3..2f8c388ef84f 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 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, 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 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) 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 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, __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 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, 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 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, continue; info = (struct usb_mixer_elem_info *)list; - if (info->control != control) + if (count > 1 && info->control != control) continue; switch (attribute) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b665d85555cb..4d5c89a7ba2b 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -47,6 +47,7 @@ struct snd_usb_audio { int num_interfaces; int num_suspended_intf; + int sample_rate_read_error; struct list_head pcm_list; /* list of pcm streams */ struct list_head ep_list; /* list of audio-related endpoints */ |