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authorLinus Torvalds <torvalds@linux-foundation.org>2016-05-19 13:41:32 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2016-05-19 13:41:32 -0700
commitf4c80d5a16eb4b08a0d9ade154af1ebdc63f5752 (patch)
tree5334acabf48210285333bc80d4a3e326efb36750 /sound
parent7afd16f882887c9adc69cd1794f5e57777723217 (diff)
parent17e1717c11a34f9b0956e33e0c4a4e4ae8c51a57 (diff)
downloadlinux-stericsson-f4c80d5a16eb4b08a0d9ade154af1ebdc63f5752.tar.gz
Merge tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "This time was again a relatively calm development cycle; most of updates are about drivers, and no radical changes are seen in any core code. Here are some highlights: ALSA core: - Continued hardening of ALSA hrtimer - A few leak fixes in timer interface - Fix poll error handling in PCM and compress - Add error propagation in compress API - Removal of dead rtctimer driver HD-audio: - Native ELD notify support for i915 HDMI - Realtek ALC234 & co support - Code refactoring to standardize chmap support - Continued development for SKL HDMI core support Firewire: - Apply delayed card registration to all drivers - Improved / stabilized the handling of PCM stream start / stop - Add tracepoints to dump a part of isochronous packet data - Fixed incoming/outgoing packet parameter usages - Add support for M-Audio profire series USB-audio: - Fixes for UAC2 clock source - SS+ support - Workaround for oft-seen repeated sample rate read errors ASoC: - Further slow progress on the topology code - Substantial updates and improvements for the da7219, es8328, fsl-ssi, Intel and rcar drivers. - Compress error handling in WM ADSP driver" * tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (177 commits) ALSA: firewire-lib: change a member of event structure to suppress sparse wanings to bool type sound: oss: Use setup_timer and mod_timer. ASoC: hdac_hdmi: Remove the unused 'timeout' variable ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix samples being dropped at Playback startup ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: pcm5102a: Add support for PCM5102A codec ASoC: hdac_hdmi: add link management ASoC: Intel: Skylake: add link management ALSA: hdac: add link pm and ref counting ALSA: au88x0: Fix zero clear of stream->resources ASoC: rt298: Add DMI match for Broxton-P reference platform ASoC: rt298: fix null deref on acpi driver data ASoC: dapm: deprecate MICBIAS widget type ALSA: firewire-lib: drop skip argument from helper functions to queue a packet ALSA: firewire-lib: add context information to tracepoints ALSA: firewire-lib: permit to flush queued packets only in process context for better PCM period granularity ...
Diffstat (limited to 'sound')
-rw-r--r--sound/core/Kconfig29
-rw-r--r--sound/core/Makefile1
-rw-r--r--sound/core/compress_offload.c25
-rw-r--r--sound/core/hrtimer.c56
-rw-r--r--sound/core/pcm_dmaengine.c11
-rw-r--r--sound/core/pcm_iec958.c65
-rw-r--r--sound/core/pcm_lib.c4
-rw-r--r--sound/core/pcm_native.c4
-rw-r--r--sound/core/rtctimer.c187
-rw-r--r--sound/core/seq/seq.c2
-rw-r--r--sound/core/timer.c5
-rw-r--r--sound/firewire/Kconfig1
-rw-r--r--sound/firewire/Makefile3
-rw-r--r--sound/firewire/amdtp-stream-trace.h110
-rw-r--r--sound/firewire/amdtp-stream.c210
-rw-r--r--sound/firewire/amdtp-stream.h37
-rw-r--r--sound/firewire/bebob/bebob.c217
-rw-r--r--sound/firewire/bebob/bebob.h6
-rw-r--r--sound/firewire/bebob/bebob_stream.c101
-rw-r--r--sound/firewire/dice/dice.c41
-rw-r--r--sound/firewire/digi00x/amdtp-dot.c2
-rw-r--r--sound/firewire/digi00x/digi00x-transaction.c7
-rw-r--r--sound/firewire/digi00x/digi00x.c107
-rw-r--r--sound/firewire/digi00x/digi00x.h3
-rw-r--r--sound/firewire/fireworks/fireworks.c168
-rw-r--r--sound/firewire/fireworks/fireworks.h4
-rw-r--r--sound/firewire/fireworks/fireworks_stream.c84
-rw-r--r--sound/firewire/lib.c32
-rw-r--r--sound/firewire/lib.h3
-rw-r--r--sound/firewire/oxfw/oxfw-stream.c3
-rw-r--r--sound/firewire/oxfw/oxfw.c151
-rw-r--r--sound/firewire/oxfw/oxfw.h4
-rw-r--r--sound/firewire/tascam/tascam-stream.c26
-rw-r--r--sound/firewire/tascam/tascam.c118
-rw-r--r--sound/firewire/tascam/tascam.h2
-rw-r--r--sound/hda/ext/hdac_ext_bus.c4
-rw-r--r--sound/hda/ext/hdac_ext_controller.c66
-rw-r--r--sound/hda/hdac_controller.c17
-rw-r--r--sound/hda/hdac_i915.c47
-rw-r--r--sound/hda/hdmi_chmap.c44
-rw-r--r--sound/hda/local.h10
-rw-r--r--sound/isa/wavefront/wavefront_synth.c9
-rw-r--r--sound/oss/waveartist.c8
-rw-r--r--sound/pci/au88x0/au88x0_core.c14
-rw-r--r--sound/pci/au88x0/au88x0_pcm.c5
-rw-r--r--sound/pci/ctxfi/cttimer.c6
-rw-r--r--sound/pci/ens1370.c2
-rw-r--r--sound/pci/hda/Kconfig10
-rw-r--r--sound/pci/hda/hda_generic.c2
-rw-r--r--sound/pci/hda/patch_hdmi.c386
-rw-r--r--sound/pci/hda/patch_realtek.c15
-rw-r--r--sound/pci/intel8x0.c20
-rw-r--r--sound/pci/lx6464es/lx_core.c2
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c4
-rw-r--r--sound/soc/au1x/dbdma2.c4
-rw-r--r--sound/soc/bcm/bcm2835-i2s.c27
-rw-r--r--sound/soc/codecs/Kconfig10
-rw-r--r--sound/soc/codecs/Makefile4
-rw-r--r--sound/soc/codecs/ak4642.c4
-rw-r--r--sound/soc/codecs/arizona.c32
-rw-r--r--sound/soc/codecs/cs42l56.c2
-rw-r--r--sound/soc/codecs/cs47l24.c46
-rw-r--r--sound/soc/codecs/da7213.c99
-rw-r--r--sound/soc/codecs/da7213.h35
-rw-r--r--sound/soc/codecs/da7218.c32
-rw-r--r--sound/soc/codecs/da7218.h21
-rw-r--r--sound/soc/codecs/da7219.c38
-rw-r--r--sound/soc/codecs/da7219.h20
-rw-r--r--sound/soc/codecs/es8328.c194
-rw-r--r--sound/soc/codecs/es8328.h23
-rw-r--r--sound/soc/codecs/hdac_hdmi.c194
-rw-r--r--sound/soc/codecs/hdmi-codec.c432
-rw-r--r--sound/soc/codecs/pcm5102a.c69
-rw-r--r--sound/soc/codecs/rt298.c19
-rw-r--r--sound/soc/codecs/rt5645.c16
-rw-r--r--sound/soc/codecs/wm5102.c4
-rw-r--r--sound/soc/codecs/wm5110.c6
-rw-r--r--sound/soc/codecs/wm_adsp.c292
-rw-r--r--sound/soc/codecs/wm_adsp.h1
-rw-r--r--sound/soc/davinci/Kconfig6
-rw-r--r--sound/soc/davinci/davinci-i2s.c80
-rw-r--r--sound/soc/davinci/davinci-mcasp.c92
-rw-r--r--sound/soc/davinci/davinci-mcasp.h5
-rw-r--r--sound/soc/dwc/designware_i2s.c14
-rw-r--r--sound/soc/fsl/fsl_sai.c24
-rw-r--r--sound/soc/fsl/fsl_ssi.c81
-rw-r--r--sound/soc/fsl/imx-pcm-fiq.c2
-rw-r--r--sound/soc/intel/Kconfig16
-rw-r--r--sound/soc/intel/atom/sst-atom-controls.c2
-rw-r--r--sound/soc/intel/boards/Makefile2
-rw-r--r--sound/soc/intel/boards/broadwell.c2
-rw-r--r--sound/soc/intel/boards/bxt_rt298.c353
-rw-r--r--sound/soc/intel/boards/bytcr_rt5640.c2
-rw-r--r--sound/soc/intel/boards/bytcr_rt5651.c2
-rw-r--r--sound/soc/intel/boards/cht_bsw_max98090_ti.c2
-rw-r--r--sound/soc/intel/boards/cht_bsw_rt5645.c2
-rw-r--r--sound/soc/intel/boards/cht_bsw_rt5672.c2
-rw-r--r--sound/soc/intel/boards/haswell.c2
-rw-r--r--sound/soc/intel/boards/skl_nau88l25_max98357a.c87
-rw-r--r--sound/soc/intel/boards/skl_nau88l25_ssm4567.c86
-rw-r--r--sound/soc/intel/boards/skl_rt286.c59
-rw-r--r--sound/soc/intel/common/sst-acpi.h9
-rw-r--r--sound/soc/intel/haswell/sst-haswell-pcm.c2
-rw-r--r--sound/soc/intel/skylake/Makefile2
-rw-r--r--sound/soc/intel/skylake/bxt-sst.c328
-rw-r--r--sound/soc/intel/skylake/skl-messages.c122
-rw-r--r--sound/soc/intel/skylake/skl-nhlt.c15
-rw-r--r--sound/soc/intel/skylake/skl-pcm.c119
-rw-r--r--sound/soc/intel/skylake/skl-sst-dsp.c2
-rw-r--r--sound/soc/intel/skylake/skl-sst-dsp.h15
-rw-r--r--sound/soc/intel/skylake/skl-sst.c13
-rw-r--r--sound/soc/intel/skylake/skl-topology.c27
-rw-r--r--sound/soc/intel/skylake/skl-topology.h2
-rw-r--r--sound/soc/intel/skylake/skl-tplg-interface.h2
-rw-r--r--sound/soc/intel/skylake/skl.c39
-rw-r--r--sound/soc/intel/skylake/skl.h6
-rw-r--r--sound/soc/rockchip/rockchip_i2s.c87
-rw-r--r--sound/soc/soc-core.c14
-rw-r--r--sound/soc/soc-generic-dmaengine-pcm.c57
-rw-r--r--sound/usb/card.c4
-rw-r--r--sound/usb/clock.c4
-rw-r--r--sound/usb/helper.c1
-rw-r--r--sound/usb/midi.c1
-rw-r--r--sound/usb/mixer.c78
-rw-r--r--sound/usb/usbaudio.h1
125 files changed, 4505 insertions, 1492 deletions
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_dmaengine.c b/sound/core/pcm_dmaengine.c
index 697c166acf05..8eb58c709b14 100644
--- a/sound/core/pcm_dmaengine.c
+++ b/sound/core/pcm_dmaengine.c
@@ -106,8 +106,9 @@ EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config);
* direction of the substream. If the substream is a playback stream the dst
* fields will be initialized, if it is a capture stream the src fields will be
* initialized. The {dst,src}_addr_width field will only be initialized if the
- * addr_width field of the DAI DMA data struct is not equal to
- * DMA_SLAVE_BUSWIDTH_UNDEFINED.
+ * SND_DMAENGINE_PCM_DAI_FLAG_PACK flag is set or if the addr_width field of
+ * the DAI DMA data struct is not equal to DMA_SLAVE_BUSWIDTH_UNDEFINED. If
+ * both conditions are met the latter takes priority.
*/
void snd_dmaengine_pcm_set_config_from_dai_data(
const struct snd_pcm_substream *substream,
@@ -117,11 +118,17 @@ void snd_dmaengine_pcm_set_config_from_dai_data(
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
slave_config->dst_addr = dma_data->addr;
slave_config->dst_maxburst = dma_data->maxburst;
+ if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)
+ slave_config->dst_addr_width =
+ DMA_SLAVE_BUSWIDTH_UNDEFINED;
if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED)
slave_config->dst_addr_width = dma_data->addr_width;
} else {
slave_config->src_addr = dma_data->addr;
slave_config->src_maxburst = dma_data->maxburst;
+ if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)
+ slave_config->src_addr_width =
+ DMA_SLAVE_BUSWIDTH_UNDEFINED;
if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED)
slave_config->src_addr_width = dma_data->addr_width;
}
diff --git a/sound/core/pcm_iec958.c b/sound/core/pcm_iec958.c
index 36b2d7aca1bd..5e6aed64f451 100644
--- a/sound/core/pcm_iec958.c
+++ b/sound/core/pcm_iec958.c
@@ -9,30 +9,18 @@
#include <linux/types.h>
#include <sound/asoundef.h>
#include <sound/pcm.h>
+#include <sound/pcm_params.h>
#include <sound/pcm_iec958.h>
-/**
- * snd_pcm_create_iec958_consumer - create consumer format IEC958 channel status
- * @runtime: pcm runtime structure with ->rate filled in
- * @cs: channel status buffer, at least four bytes
- * @len: length of channel status buffer
- *
- * Create the consumer format channel status data in @cs of maximum size
- * @len corresponding to the parameters of the PCM runtime @runtime.
- *
- * Drivers may wish to tweak the contents of the buffer after creation.
- *
- * Returns: length of buffer, or negative error code if something failed.
- */
-int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
- size_t len)
+static int create_iec958_consumer(uint rate, uint sample_width,
+ u8 *cs, size_t len)
{
unsigned int fs, ws;
if (len < 4)
return -EINVAL;
- switch (runtime->rate) {
+ switch (rate) {
case 32000:
fs = IEC958_AES3_CON_FS_32000;
break;
@@ -59,7 +47,7 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
}
if (len > 4) {
- switch (snd_pcm_format_width(runtime->format)) {
+ switch (sample_width) {
case 16:
ws = IEC958_AES4_CON_WORDLEN_20_16;
break;
@@ -71,6 +59,7 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
IEC958_AES4_CON_MAX_WORDLEN_24;
break;
case 24:
+ case 32: /* Assume 24-bit width for 32-bit samples. */
ws = IEC958_AES4_CON_WORDLEN_24_20 |
IEC958_AES4_CON_MAX_WORDLEN_24;
break;
@@ -92,4 +81,46 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
return len;
}
+
+/**
+ * snd_pcm_create_iec958_consumer - create consumer format IEC958 channel status
+ * @runtime: pcm runtime structure with ->rate filled in
+ * @cs: channel status buffer, at least four bytes
+ * @len: length of channel status buffer
+ *
+ * Create the consumer format channel status data in @cs of maximum size
+ * @len corresponding to the parameters of the PCM runtime @runtime.
+ *
+ * Drivers may wish to tweak the contents of the buffer after creation.
+ *
+ * Returns: length of buffer, or negative error code if something failed.
+ */
+int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
+ size_t len)
+{
+ return create_iec958_consumer(runtime->rate,
+ snd_pcm_format_width(runtime->format),
+ cs, len);
+}
EXPORT_SYMBOL(snd_pcm_create_iec958_consumer);
+
+/**
+ * snd_pcm_create_iec958_consumer_hw_params - create IEC958 channel status
+ * @hw_params: the hw_params instance for extracting rate and sample format
+ * @cs: channel status buffer, at least four bytes
+ * @len: length of channel status buffer
+ *
+ * Create the consumer format channel status data in @cs of maximum size
+ * @len corresponding to the parameters of the PCM runtime @runtime.
+ *
+ * Drivers may wish to tweak the contents of the buffer after creation.
+ *
+ * Returns: length of buffer, or negative error code if something failed.
+ */
+int snd_pcm_create_iec958_consumer_hw_params(struct snd_pcm_hw_params *params,
+ u8 *cs, size_t len)
+{
+ return create_iec958_consumer(params_rate(params), params_width(params),
+ cs, len);
+}
+EXPORT_SYMBOL(snd_pcm_create_iec958_consumer_hw_params);
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..9c04faf206b2
--- /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(unsigned int, 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(unsigned int, 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..31b510c5ca0b 100644
--- a/sound/hda/ext/hdac_ext_bus.c
+++ b/sound/hda/ext/hdac_ext_bus.c
@@ -105,6 +105,9 @@ int snd_hdac_ext_bus_init(struct hdac_ext_bus *ebus, struct device *dev,
INIT_LIST_HEAD(&ebus->hlink_list);
ebus->idx = idx++;
+ mutex_init(&ebus->lock);
+ ebus->cmd_dma_state = true;
+
return 0;
}
EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_init);
@@ -144,6 +147,7 @@ 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/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c
index 548cc1e4114b..860f8cad6602 100644
--- a/sound/hda/ext/hdac_ext_controller.c
+++ b/sound/hda/ext/hdac_ext_controller.c
@@ -186,6 +186,9 @@ int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus)
hlink->lcaps = readl(hlink->ml_addr + AZX_REG_ML_LCAP);
hlink->lsdiid = readw(hlink->ml_addr + AZX_REG_ML_LSDIID);
+ /* since link in On, update the ref */
+ hlink->ref_count = 1;
+
list_add_tail(&hlink->list, &ebus->hlink_list);
}
@@ -327,3 +330,66 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus)
return 0;
}
EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all);
+
+int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus,
+ struct hdac_ext_link *link)
+{
+ int ret = 0;
+
+ mutex_lock(&ebus->lock);
+
+ /*
+ * if we move from 0 to 1, count will be 1 so power up this link
+ * as well, also check the dma status and trigger that
+ */
+ if (++link->ref_count == 1) {
+ if (!ebus->cmd_dma_state) {
+ snd_hdac_bus_init_cmd_io(&ebus->bus);
+ ebus->cmd_dma_state = true;
+ }
+
+ ret = snd_hdac_ext_bus_link_power_up(link);
+ }
+
+ mutex_unlock(&ebus->lock);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_get);
+
+int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus,
+ struct hdac_ext_link *link)
+{
+ int ret = 0;
+ struct hdac_ext_link *hlink;
+ bool link_up = false;
+
+ mutex_lock(&ebus->lock);
+
+ /*
+ * if we move from 1 to 0, count will be 0
+ * so power down this link as well
+ */
+ if (--link->ref_count == 0) {
+ ret = snd_hdac_ext_bus_link_power_down(link);
+
+ /*
+ * now check if all links are off, if so turn off
+ * cmd dma as well
+ */
+ list_for_each_entry(hlink, &ebus->hlink_list, list) {
+ if (hlink->ref_count) {
+ link_up = true;
+ break;
+ }
+ }
+
+ if (!link_up) {
+ snd_hdac_bus_stop_cmd_io(&ebus->bus);
+ ebus->cmd_dma_state = false;
+ }
+ }
+
+ mutex_unlock(&ebus->lock);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_put);
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/hda/local.h b/sound/hda/local.h
index d692f417ddc0..0d5bb159d538 100644
--- a/sound/hda/local.h
+++ b/sound/hda/local.h
@@ -16,6 +16,16 @@ static inline int get_wcaps_type(unsigned int wcaps)
return (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
}
+static inline unsigned int get_wcaps_channels(u32 wcaps)
+{
+ unsigned int chans;
+
+ chans = (wcaps & AC_WCAP_CHAN_CNT_EXT) >> 13;
+ chans = (chans + 1) * 2;
+
+ return chans;
+}
+
extern const struct attribute_group *hdac_dev_attr_groups[];
int hda_widget_sysfs_init(struct hdac_device *codec);
void hda_widget_sysfs_exit(struct hdac_device *codec);
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/oss/waveartist.c b/sound/oss/waveartist.c
index b36ea47527e8..0b8d0de87273 100644
--- a/sound/oss/waveartist.c
+++ b/sound/oss/waveartist.c
@@ -1414,11 +1414,9 @@ attach_waveartist(struct address_info *hw, const struct waveartist_mixer_info *m
else {
#ifdef CONFIG_ARCH_NETWINDER
if (machine_is_netwinder()) {
- init_timer(&vnc_timer);
- vnc_timer.function = vnc_slider_tick;
- vnc_timer.expires = jiffies;
- vnc_timer.data = nr_waveartist_devs;
- add_timer(&vnc_timer);
+ setup_timer(&vnc_timer, vnc_slider_tick,
+ nr_waveartist_devs);
+ mod_timer(&vnc_timer, jiffies);
vnc_configure_mixer(devc, 0);
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/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 276897033639..1267e1af0fae 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -652,7 +652,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period)
| SSC_BF(RCMR_STTDLY, 1)
| SSC_BF(RCMR_START, SSC_START_RISING_RF)
- | SSC_BF(RCMR_CKI, SSC_CKI_FALLING)
+ | SSC_BF(RCMR_CKI, SSC_CKI_RISING)
| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
| SSC_BF(RCMR_CKS, SSC_CKS_DIV);
@@ -692,7 +692,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
rcmr = SSC_BF(RCMR_PERIOD, 0)
| SSC_BF(RCMR_STTDLY, START_DELAY)
| SSC_BF(RCMR_START, SSC_START_RISING_RF)
- | SSC_BF(RCMR_CKI, SSC_CKI_FALLING)
+ | SSC_BF(RCMR_CKI, SSC_CKI_RISING)
| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
| SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
SSC_CKS_PIN : SSC_CKS_CLOCK);
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 5741c0aa6c03..b5d1caa04d8e 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -206,8 +206,8 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
stype = substream->stream;
pcd = to_dmadata(substream);
- DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d "
- "runtime->min_align %d\n",
+ DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %zu "
+ "runtime->min_align %lu\n",
(unsigned long)runtime->dma_area,
(unsigned long)runtime->dma_addr, runtime->dma_bytes,
runtime->min_align);
diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c
index 1c1f2210387b..6ba20498202e 100644
--- a/sound/soc/bcm/bcm2835-i2s.c
+++ b/sound/soc/bcm/bcm2835-i2s.c
@@ -259,6 +259,9 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream,
case SNDRV_PCM_FORMAT_S16_LE:
data_length = 16;
break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ data_length = 24;
+ break;
case SNDRV_PCM_FORMAT_S32_LE:
data_length = 32;
break;
@@ -273,13 +276,20 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream,
/* otherwise calculate a fitting block ratio */
bclk_ratio = 2 * data_length;
- /* set target clock rate*/
- clk_set_rate(dev->clk, sampling_rate * bclk_ratio);
+ /* Clock should only be set up here if CPU is clock master */
+ switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ case SND_SOC_DAIFMT_CBS_CFM:
+ clk_set_rate(dev->clk, sampling_rate * bclk_ratio);
+ break;
+ default:
+ break;
+ }
/* Setup the frame format */
format = BCM2835_I2S_CHEN;
- if (data_length > 24)
+ if (data_length >= 24)
format |= BCM2835_I2S_CHWEX;
format |= BCM2835_I2S_CHWID((data_length-8)&0xf);
@@ -570,6 +580,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = {
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_192000,
.formats = SNDRV_PCM_FMTBIT_S16_LE
+ | SNDRV_PCM_FMTBIT_S24_LE
| SNDRV_PCM_FMTBIT_S32_LE
},
.capture = {
@@ -577,6 +588,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = {
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_192000,
.formats = SNDRV_PCM_FMTBIT_S16_LE
+ | SNDRV_PCM_FMTBIT_S24_LE
| SNDRV_PCM_FMTBIT_S32_LE
},
.ops = &bcm2835_i2s_dai_ops,
@@ -678,6 +690,15 @@ static int bcm2835_i2s_probe(struct platform_device *pdev)
dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].maxburst = 2;
dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].maxburst = 2;
+ /*
+ * Set the PACK flag to enable S16_LE support (2 S16_LE values
+ * packed into 32-bit transfers).
+ */
+ dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].flags =
+ SND_DMAENGINE_PCM_DAI_FLAG_PACK;
+ dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].flags =
+ SND_DMAENGINE_PCM_DAI_FLAG_PACK;
+
/* BCLK ratio - use default */
dev->bclk_ratio = 0;
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 7ef3a0c16478..b3afae990e39 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -88,12 +88,14 @@ config SND_SOC_ALL_CODECS
select SND_SOC_MC13783 if MFD_MC13XXX
select SND_SOC_ML26124 if I2C
select SND_SOC_NAU8825 if I2C
+ select SND_SOC_HDMI_CODEC
select SND_SOC_PCM1681 if I2C
select SND_SOC_PCM179X_I2C if I2C
select SND_SOC_PCM179X_SPI if SPI_MASTER
select SND_SOC_PCM3008
select SND_SOC_PCM3168A_I2C if I2C
select SND_SOC_PCM3168A_SPI if SPI_MASTER
+ select SND_SOC_PCM5102A
select SND_SOC_PCM512x_I2C if I2C
select SND_SOC_PCM512x_SPI if SPI_MASTER
select SND_SOC_RT286 if I2C
@@ -477,6 +479,11 @@ config SND_SOC_BT_SCO
config SND_SOC_DMIC
tristate
+config SND_SOC_HDMI_CODEC
+ tristate
+ select SND_PCM_ELD
+ select SND_PCM_IEC958
+
config SND_SOC_ES8328
tristate "Everest Semi ES8328 CODEC"
@@ -575,6 +582,9 @@ config SND_SOC_PCM3168A_SPI
select SND_SOC_PCM3168A
select REGMAP_SPI
+config SND_SOC_PCM5102A
+ tristate
+
config SND_SOC_PCM512x
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 185a712a7fe7..b7b99416537f 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -81,6 +81,7 @@ snd-soc-max9850-objs := max9850.o
snd-soc-mc13783-objs := mc13783.o
snd-soc-ml26124-objs := ml26124.o
snd-soc-nau8825-objs := nau8825.o
+snd-soc-hdmi-codec-objs := hdmi-codec.o
snd-soc-pcm1681-objs := pcm1681.o
snd-soc-pcm179x-codec-objs := pcm179x.o
snd-soc-pcm179x-i2c-objs := pcm179x-i2c.o
@@ -89,6 +90,7 @@ snd-soc-pcm3008-objs := pcm3008.o
snd-soc-pcm3168a-objs := pcm3168a.o
snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o
snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o
+snd-soc-pcm5102a-objs := pcm5102a.o
snd-soc-pcm512x-objs := pcm512x.o
snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o
snd-soc-pcm512x-spi-objs := pcm512x-spi.o
@@ -290,6 +292,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o
obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o
obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o
obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o
+obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o
obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o
obj-$(CONFIG_SND_SOC_PCM179X) += snd-soc-pcm179x-codec.o
obj-$(CONFIG_SND_SOC_PCM179X_I2C) += snd-soc-pcm179x-i2c.o
@@ -298,6 +301,7 @@ obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
obj-$(CONFIG_SND_SOC_PCM3168A) += snd-soc-pcm3168a.o
obj-$(CONFIG_SND_SOC_PCM3168A_I2C) += snd-soc-pcm3168a-i2c.o
obj-$(CONFIG_SND_SOC_PCM3168A_SPI) += snd-soc-pcm3168a-spi.o
+obj-$(CONFIG_SND_SOC_PCM5102A) += snd-soc-pcm5102a.o
obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o
obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o
obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index cda27c22812a..1ee8506c06c7 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -608,9 +608,7 @@ static struct clk *ak4642_of_parse_mcko(struct device *dev)
of_property_read_string(np, "clock-output-names", &clk_name);
- clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name,
- (parent_clk_name) ? 0 : CLK_IS_ROOT,
- rate);
+ clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, 0, rate);
if (!IS_ERR(clk))
of_clk_add_provider(np, of_clk_src_simple_get, clk);
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 83959312f7a0..664a8c044ffb 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -221,6 +221,8 @@ int arizona_init_spk(struct snd_soc_codec *codec)
switch (arizona->type) {
case WM8997:
+ case CS47L24:
+ case WM1831:
break;
default:
ret = snd_soc_dapm_new_controls(dapm, &arizona_spkr, 1);
@@ -1134,7 +1136,6 @@ int arizona_anc_ev(struct snd_soc_dapm_widget *w,
int event)
{
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
- unsigned int mask = 0x3 << w->shift;
unsigned int val;
switch (event) {
@@ -1148,7 +1149,7 @@ int arizona_anc_ev(struct snd_soc_dapm_widget *w,
return 0;
}
- snd_soc_update_bits(codec, ARIZONA_CLOCK_CONTROL, mask, val);
+ snd_soc_write(codec, ARIZONA_CLOCK_CONTROL, val);
return 0;
}
@@ -2047,7 +2048,21 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
init_ratio, Fref, refdiv);
while (div <= ARIZONA_FLL_MAX_REFDIV) {
- for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO;
+ /* start from init_ratio because this may already give a
+ * fractional N.K
+ */
+ for (ratio = init_ratio; ratio > 0; ratio--) {
+ if (target % (ratio * Fref)) {
+ cfg->refdiv = refdiv;
+ cfg->fratio = ratio - 1;
+ arizona_fll_dbg(fll,
+ "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n",
+ Fref, refdiv, div, ratio);
+ return ratio;
+ }
+ }
+
+ for (ratio = init_ratio + 1; ratio <= ARIZONA_FLL_MAX_FRATIO;
ratio++) {
if ((ARIZONA_FLL_VCO_CORNER / 2) /
(fll->vco_mult * ratio) < Fref) {
@@ -2073,17 +2088,6 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
}
}
- for (ratio = init_ratio - 1; ratio > 0; ratio--) {
- if (target % (ratio * Fref)) {
- cfg->refdiv = refdiv;
- cfg->fratio = ratio - 1;
- arizona_fll_dbg(fll,
- "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n",
- Fref, refdiv, div, ratio);
- return ratio;
- }
- }
-
div *= 2;
Fref /= 2;
refdiv++;
diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index 7cd5f769bb61..eec1ff853b98 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/sound/soc/codecs/cs42l56.c
@@ -56,7 +56,7 @@ struct cs42l56_private {
u8 iface;
u8 iface_fmt;
u8 iface_inv;
-#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE)
+#if IS_ENABLED(CONFIG_INPUT)
struct input_dev *beep;
struct work_struct beep_work;
int beep_rate;
diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index 00e9b6fc1b5c..5ec5a682d186 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -807,6 +807,9 @@ static const struct snd_soc_dapm_route cs47l24_dapm_routes[] = {
{ "IN2L PGA", NULL, "IN2L" },
{ "IN2R PGA", NULL, "IN2R" },
+ { "Audio Trace DSP", NULL, "DSP2" },
+ { "Audio Trace DSP", NULL, "SYSCLK" },
+
ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
@@ -1016,6 +1019,27 @@ static struct snd_soc_dai_driver cs47l24_dai[] = {
.formats = CS47L24_FORMATS,
},
},
+ {
+ .name = "cs47l24-cpu-trace",
+ .capture = {
+ .stream_name = "Audio Trace CPU",
+ .channels_min = 1,
+ .channels_max = 6,
+ .rates = CS47L24_RATES,
+ .formats = CS47L24_FORMATS,
+ },
+ .compress_new = snd_soc_new_compress,
+ },
+ {
+ .name = "cs47l24-dsp-trace",
+ .capture = {
+ .stream_name = "Audio Trace DSP",
+ .channels_min = 1,
+ .channels_max = 6,
+ .rates = CS47L24_RATES,
+ .formats = CS47L24_FORMATS,
+ },
+ },
};
static int cs47l24_open(struct snd_compr_stream *stream)
@@ -1027,6 +1051,8 @@ static int cs47l24_open(struct snd_compr_stream *stream)
if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-voicectrl") == 0) {
n_adsp = 2;
+ } else if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-trace") == 0) {
+ n_adsp = 1;
} else {
dev_err(arizona->dev,
"No suitable compressed stream for DAI '%s'\n",
@@ -1041,10 +1067,16 @@ static irqreturn_t cs47l24_adsp2_irq(int irq, void *data)
{
struct cs47l24_priv *priv = data;
struct arizona *arizona = priv->core.arizona;
- int ret;
+ int serviced = 0;
+ int i, ret;
- ret = wm_adsp_compr_handle_irq(&priv->core.adsp[2]);
- if (ret == -ENODEV) {
+ for (i = 1; i <= 2; ++i) {
+ ret = wm_adsp_compr_handle_irq(&priv->core.adsp[i]);
+ if (ret != -ENODEV)
+ serviced++;
+ }
+
+ if (!serviced) {
dev_err(arizona->dev, "Spurious compressed data IRQ\n");
return IRQ_NONE;
}
@@ -1160,6 +1192,7 @@ static struct snd_compr_ops cs47l24_compr_ops = {
static struct snd_soc_platform_driver cs47l24_compr_platform = {
.compr_ops = &cs47l24_compr_ops,
};
+
static int cs47l24_probe(struct platform_device *pdev)
{
struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
@@ -1228,9 +1261,9 @@ static int cs47l24_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "Failed to register platform: %d\n", ret);
return ret;
}
+
ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cs47l24,
cs47l24_dai, ARRAY_SIZE(cs47l24_dai));
-
if (ret < 0) {
dev_err(&pdev->dev, "Failed to register codec: %d\n", ret);
snd_soc_unregister_platform(&pdev->dev);
@@ -1241,10 +1274,15 @@ static int cs47l24_probe(struct platform_device *pdev)
static int cs47l24_remove(struct platform_device *pdev)
{
+ struct cs47l24_priv *cs47l24 = platform_get_drvdata(pdev);
+
snd_soc_unregister_platform(&pdev->dev);
snd_soc_unregister_codec(&pdev->dev);
pm_runtime_disable(&pdev->dev);
+ wm_adsp2_remove(&cs47l24->core.adsp[1]);
+ wm_adsp2_remove(&cs47l24->core.adsp[2]);
+
return 0;
}
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 7278f93460c1..e5527bc570ae 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -726,6 +726,68 @@ static const struct snd_kcontrol_new da7213_dapm_mixoutr_controls[] = {
/*
+ * DAPM Events
+ */
+
+static int da7213_dai_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec);
+ u8 pll_ctrl, pll_status;
+ int i = 0;
+ bool srm_lock = false;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ /* Enable DAI clks for master mode */
+ if (da7213->master)
+ snd_soc_update_bits(codec, DA7213_DAI_CLK_MODE,
+ DA7213_DAI_CLK_EN_MASK,
+ DA7213_DAI_CLK_EN_MASK);
+
+ /* PC synchronised to DAI */
+ snd_soc_update_bits(codec, DA7213_PC_COUNT,
+ DA7213_PC_FREERUN_MASK, 0);
+
+ /* Slave mode, if SRM not enabled no need for status checks */
+ pll_ctrl = snd_soc_read(codec, DA7213_PLL_CTRL);
+ if (!(pll_ctrl & DA7213_PLL_SRM_EN))
+ return 0;
+
+ /* Check SRM has locked */
+ do {
+ pll_status = snd_soc_read(codec, DA7213_PLL_STATUS);
+ if (pll_status & DA7219_PLL_SRM_LOCK) {
+ srm_lock = true;
+ } else {
+ ++i;
+ msleep(50);
+ }
+ } while ((i < DA7213_SRM_CHECK_RETRIES) & (!srm_lock));
+
+ if (!srm_lock)
+ dev_warn(codec->dev, "SRM failed to lock\n");
+
+ return 0;
+ case SND_SOC_DAPM_POST_PMD:
+ /* PC free-running */
+ snd_soc_update_bits(codec, DA7213_PC_COUNT,
+ DA7213_PC_FREERUN_MASK,
+ DA7213_PC_FREERUN_MASK);
+
+ /* Disable DAI clks if in master mode */
+ if (da7213->master)
+ snd_soc_update_bits(codec, DA7213_DAI_CLK_MODE,
+ DA7213_DAI_CLK_EN_MASK, 0);
+ return 0;
+ default:
+ return -EINVAL;
+ }
+}
+
+
+/*
* DAPM widgets
*/
@@ -736,7 +798,8 @@ static const struct snd_soc_dapm_widget da7213_dapm_widgets[] = {
/* Use a supply here as this controls both input & output DAIs */
SND_SOC_DAPM_SUPPLY("DAI", DA7213_DAI_CTRL, DA7213_DAI_EN_SHIFT,
- DA7213_NO_INVERT, NULL, 0),
+ DA7213_NO_INVERT, da7213_dai_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
/*
* Input
@@ -1143,11 +1206,9 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
/* Set master/slave mode */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
- dai_clk_mode |= DA7213_DAI_CLK_EN_MASTER_MODE;
da7213->master = true;
break;
case SND_SOC_DAIFMT_CBS_CFS:
- dai_clk_mode |= DA7213_DAI_CLK_EN_SLAVE_MODE;
da7213->master = false;
break;
default:
@@ -1281,28 +1342,28 @@ static int da7213_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
pll_ctrl = 0;
/* Workout input divider based on MCLK rate */
- if ((da7213->mclk_rate == 32768) && (source == DA7213_SYSCLK_PLL)) {
+ if (da7213->mclk_rate == 32768) {
/* 32KHz PLL Mode */
- indiv_bits = DA7213_PLL_INDIV_10_20_MHZ;
- indiv = DA7213_PLL_INDIV_10_20_MHZ_VAL;
+ indiv_bits = DA7213_PLL_INDIV_9_TO_18_MHZ;
+ indiv = DA7213_PLL_INDIV_9_TO_18_MHZ_VAL;
freq_ref = 3750000;
pll_ctrl |= DA7213_PLL_32K_MODE;
} else {
/* 5 - 54MHz MCLK */
if (da7213->mclk_rate < 5000000) {
goto pll_err;
- } else if (da7213->mclk_rate <= 10000000) {
- indiv_bits = DA7213_PLL_INDIV_5_10_MHZ;
- indiv = DA7213_PLL_INDIV_5_10_MHZ_VAL;
- } else if (da7213->mclk_rate <= 20000000) {
- indiv_bits = DA7213_PLL_INDIV_10_20_MHZ;
- indiv = DA7213_PLL_INDIV_10_20_MHZ_VAL;
- } else if (da7213->mclk_rate <= 40000000) {
- indiv_bits = DA7213_PLL_INDIV_20_40_MHZ;
- indiv = DA7213_PLL_INDIV_20_40_MHZ_VAL;
+ } else if (da7213->mclk_rate <= 9000000) {
+ indiv_bits = DA7213_PLL_INDIV_5_TO_9_MHZ;
+ indiv = DA7213_PLL_INDIV_5_TO_9_MHZ_VAL;
+ } else if (da7213->mclk_rate <= 18000000) {
+ indiv_bits = DA7213_PLL_INDIV_9_TO_18_MHZ;
+ indiv = DA7213_PLL_INDIV_9_TO_18_MHZ_VAL;
+ } else if (da7213->mclk_rate <= 36000000) {
+ indiv_bits = DA7213_PLL_INDIV_18_TO_36_MHZ;
+ indiv = DA7213_PLL_INDIV_18_TO_36_MHZ_VAL;
} else if (da7213->mclk_rate <= 54000000) {
- indiv_bits = DA7213_PLL_INDIV_40_54_MHZ;
- indiv = DA7213_PLL_INDIV_40_54_MHZ_VAL;
+ indiv_bits = DA7213_PLL_INDIV_36_TO_54_MHZ;
+ indiv = DA7213_PLL_INDIV_36_TO_54_MHZ_VAL;
} else {
goto pll_err;
}
@@ -1547,6 +1608,10 @@ static int da7213_probe(struct snd_soc_codec *codec)
/* Default to using SRM for slave mode */
da7213->srm_en = true;
+ /* Default PC counter to free-running */
+ snd_soc_update_bits(codec, DA7213_PC_COUNT, DA7213_PC_FREERUN_MASK,
+ DA7213_PC_FREERUN_MASK);
+
/* Enable all Gain Ramps */
snd_soc_update_bits(codec, DA7213_AUX_L_CTRL,
DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN);
diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h
index 030fd691b076..fbb7a356a501 100644
--- a/sound/soc/codecs/da7213.h
+++ b/sound/soc/codecs/da7213.h
@@ -142,6 +142,9 @@
* Bit fields
*/
+/* DA7213_PLL_STATUS = 0x03 */
+#define DA7219_PLL_SRM_LOCK (0x1 << 1)
+
/* DA7213_SR = 0x22 */
#define DA7213_SR_8000 (0x1 << 0)
#define DA7213_SR_11025 (0x2 << 0)
@@ -160,10 +163,10 @@
#define DA7213_VMID_EN (0x1 << 7)
/* DA7213_PLL_CTRL = 0x27 */
-#define DA7213_PLL_INDIV_5_10_MHZ (0x0 << 2)
-#define DA7213_PLL_INDIV_10_20_MHZ (0x1 << 2)
-#define DA7213_PLL_INDIV_20_40_MHZ (0x2 << 2)
-#define DA7213_PLL_INDIV_40_54_MHZ (0x3 << 2)
+#define DA7213_PLL_INDIV_5_TO_9_MHZ (0x0 << 2)
+#define DA7213_PLL_INDIV_9_TO_18_MHZ (0x1 << 2)
+#define DA7213_PLL_INDIV_18_TO_36_MHZ (0x2 << 2)
+#define DA7213_PLL_INDIV_36_TO_54_MHZ (0x3 << 2)
#define DA7213_PLL_INDIV_MASK (0x3 << 2)
#define DA7213_PLL_MCLK_SQR_EN (0x1 << 4)
#define DA7213_PLL_32K_MODE (0x1 << 5)
@@ -178,8 +181,6 @@
#define DA7213_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0)
#define DA7213_DAI_CLK_POL_INV (0x1 << 2)
#define DA7213_DAI_WCLK_POL_INV (0x1 << 3)
-#define DA7213_DAI_CLK_EN_SLAVE_MODE (0x0 << 7)
-#define DA7213_DAI_CLK_EN_MASTER_MODE (0x1 << 7)
#define DA7213_DAI_CLK_EN_MASK (0x1 << 7)
/* DA7213_DAI_CTRL = 0x29 */
@@ -412,6 +413,9 @@
#define DA7213_DMIC_CLK_RATE_SHIFT 2
#define DA7213_DMIC_CLK_RATE_MASK (0x1 << 2)
+/* DA7213_PC_COUNT = 0x94 */
+#define DA7213_PC_FREERUN_MASK (0x1 << 0)
+
/* DA7213_DIG_CTRL = 0x99 */
#define DA7213_DAC_L_INV_SHIFT 3
#define DA7213_DAC_R_INV_SHIFT 7
@@ -495,15 +499,16 @@
#define DA7213_ALC_AVG_ITERATIONS 5
/* PLL related */
-#define DA7213_SYSCLK_MCLK 0
-#define DA7213_SYSCLK_PLL 1
-#define DA7213_PLL_FREQ_OUT_90316800 90316800
-#define DA7213_PLL_FREQ_OUT_98304000 98304000
-#define DA7213_PLL_FREQ_OUT_94310400 94310400
-#define DA7213_PLL_INDIV_5_10_MHZ_VAL 2
-#define DA7213_PLL_INDIV_10_20_MHZ_VAL 4
-#define DA7213_PLL_INDIV_20_40_MHZ_VAL 8
-#define DA7213_PLL_INDIV_40_54_MHZ_VAL 16
+#define DA7213_SYSCLK_MCLK 0
+#define DA7213_SYSCLK_PLL 1
+#define DA7213_PLL_FREQ_OUT_90316800 90316800
+#define DA7213_PLL_FREQ_OUT_98304000 98304000
+#define DA7213_PLL_FREQ_OUT_94310400 94310400
+#define DA7213_PLL_INDIV_5_TO_9_MHZ_VAL 2
+#define DA7213_PLL_INDIV_9_TO_18_MHZ_VAL 4
+#define DA7213_PLL_INDIV_18_TO_36_MHZ_VAL 8
+#define DA7213_PLL_INDIV_36_TO_54_MHZ_VAL 16
+#define DA7213_SRM_CHECK_RETRIES 8
enum da7213_clk_src {
DA7213_CLKSRC_MCLK = 0,
diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c
index 93575f251866..99ce23e113bf 100644
--- a/sound/soc/codecs/da7218.c
+++ b/sound/soc/codecs/da7218.c
@@ -1868,27 +1868,27 @@ static int da7218_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
/* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */
if (da7218->mclk_rate == 32768) {
- indiv_bits = DA7218_PLL_INDIV_2_5_MHZ;
- indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL;
+ indiv_bits = DA7218_PLL_INDIV_9_TO_18_MHZ;
+ indiv = DA7218_PLL_INDIV_9_TO_18_MHZ_VAL;
} else if (da7218->mclk_rate < 2000000) {
dev_err(codec->dev, "PLL input clock %d below valid range\n",
da7218->mclk_rate);
return -EINVAL;
- } else if (da7218->mclk_rate <= 5000000) {
- indiv_bits = DA7218_PLL_INDIV_2_5_MHZ;
- indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL;
- } else if (da7218->mclk_rate <= 10000000) {
- indiv_bits = DA7218_PLL_INDIV_5_10_MHZ;
- indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL;
- } else if (da7218->mclk_rate <= 20000000) {
- indiv_bits = DA7218_PLL_INDIV_10_20_MHZ;
- indiv = DA7218_PLL_INDIV_10_20_MHZ_VAL;
- } else if (da7218->mclk_rate <= 40000000) {
- indiv_bits = DA7218_PLL_INDIV_20_40_MHZ;
- indiv = DA7218_PLL_INDIV_20_40_MHZ_VAL;
+ } else if (da7218->mclk_rate <= 4500000) {
+ indiv_bits = DA7218_PLL_INDIV_2_TO_4_5_MHZ;
+ indiv = DA7218_PLL_INDIV_2_TO_4_5_MHZ_VAL;
+ } else if (da7218->mclk_rate <= 9000000) {
+ indiv_bits = DA7218_PLL_INDIV_4_5_TO_9_MHZ;
+ indiv = DA7218_PLL_INDIV_4_5_TO_9_MHZ_VAL;
+ } else if (da7218->mclk_rate <= 18000000) {
+ indiv_bits = DA7218_PLL_INDIV_9_TO_18_MHZ;
+ indiv = DA7218_PLL_INDIV_9_TO_18_MHZ_VAL;
+ } else if (da7218->mclk_rate <= 36000000) {
+ indiv_bits = DA7218_PLL_INDIV_18_TO_36_MHZ;
+ indiv = DA7218_PLL_INDIV_18_TO_36_MHZ_VAL;
} else if (da7218->mclk_rate <= 54000000) {
- indiv_bits = DA7218_PLL_INDIV_40_54_MHZ;
- indiv = DA7218_PLL_INDIV_40_54_MHZ_VAL;
+ indiv_bits = DA7218_PLL_INDIV_36_TO_54_MHZ;
+ indiv = DA7218_PLL_INDIV_36_TO_54_MHZ_VAL;
} else {
dev_err(codec->dev, "PLL input clock %d above valid range\n",
da7218->mclk_rate);
diff --git a/sound/soc/codecs/da7218.h b/sound/soc/codecs/da7218.h
index c2c59049a2ad..477cd37723cf 100644
--- a/sound/soc/codecs/da7218.h
+++ b/sound/soc/codecs/da7218.h
@@ -876,15 +876,11 @@
/* DA7218_PLL_CTRL = 0x91 */
#define DA7218_PLL_INDIV_SHIFT 0
#define DA7218_PLL_INDIV_MASK (0x7 << 0)
-#define DA7218_PLL_INDIV_2_5_MHZ (0x0 << 0)
-#define DA7218_PLL_INDIV_5_10_MHZ (0x1 << 0)
-#define DA7218_PLL_INDIV_10_20_MHZ (0x2 << 0)
-#define DA7218_PLL_INDIV_20_40_MHZ (0x3 << 0)
-#define DA7218_PLL_INDIV_40_54_MHZ (0x4 << 0)
-#define DA7218_PLL_INDIV_2_10_MHZ_VAL 2
-#define DA7218_PLL_INDIV_10_20_MHZ_VAL 4
-#define DA7218_PLL_INDIV_20_40_MHZ_VAL 8
-#define DA7218_PLL_INDIV_40_54_MHZ_VAL 16
+#define DA7218_PLL_INDIV_2_TO_4_5_MHZ (0x0 << 0)
+#define DA7218_PLL_INDIV_4_5_TO_9_MHZ (0x1 << 0)
+#define DA7218_PLL_INDIV_9_TO_18_MHZ (0x2 << 0)
+#define DA7218_PLL_INDIV_18_TO_36_MHZ (0x3 << 0)
+#define DA7218_PLL_INDIV_36_TO_54_MHZ (0x4 << 0)
#define DA7218_PLL_MCLK_SQR_EN_SHIFT 4
#define DA7218_PLL_MCLK_SQR_EN_MASK (0x1 << 4)
#define DA7218_PLL_MODE_SHIFT 6
@@ -1336,6 +1332,13 @@
#define DA7218_PLL_FREQ_OUT_90316 90316800
#define DA7218_PLL_FREQ_OUT_98304 98304000
+/* PLL Frequency Dividers */
+#define DA7218_PLL_INDIV_2_TO_4_5_MHZ_VAL 1
+#define DA7218_PLL_INDIV_4_5_TO_9_MHZ_VAL 2
+#define DA7218_PLL_INDIV_9_TO_18_MHZ_VAL 4
+#define DA7218_PLL_INDIV_18_TO_36_MHZ_VAL 8
+#define DA7218_PLL_INDIV_36_TO_54_MHZ_VAL 16
+
/* ALC Calibration */
#define DA7218_ALC_CALIB_DELAY_MIN 2500
#define DA7218_ALC_CALIB_DELAY_MAX 5000
diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c
index 81c0708b85c1..5c93899f1f0e 100644
--- a/sound/soc/codecs/da7219.c
+++ b/sound/soc/codecs/da7219.c
@@ -11,6 +11,7 @@
* option) any later version.
*/
+#include <linux/acpi.h>
#include <linux/clk.h>
#include <linux/i2c.h>
#include <linux/of_device.h>
@@ -1025,7 +1026,7 @@ static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai,
if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq))
return 0;
- if (((freq < 2000000) && (freq != 32768)) || (freq > 54000000)) {
+ if ((freq < 2000000) || (freq > 54000000)) {
dev_err(codec_dai->dev, "Unsupported MCLK value %d\n",
freq);
return -EINVAL;
@@ -1079,21 +1080,21 @@ static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
dev_err(codec->dev, "PLL input clock %d below valid range\n",
da7219->mclk_rate);
return -EINVAL;
- } else if (da7219->mclk_rate <= 5000000) {
- indiv_bits = DA7219_PLL_INDIV_2_5_MHZ;
- indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL;
- } else if (da7219->mclk_rate <= 10000000) {
- indiv_bits = DA7219_PLL_INDIV_5_10_MHZ;
- indiv = DA7219_PLL_INDIV_5_10_MHZ_VAL;
- } else if (da7219->mclk_rate <= 20000000) {
- indiv_bits = DA7219_PLL_INDIV_10_20_MHZ;
- indiv = DA7219_PLL_INDIV_10_20_MHZ_VAL;
- } else if (da7219->mclk_rate <= 40000000) {
- indiv_bits = DA7219_PLL_INDIV_20_40_MHZ;
- indiv = DA7219_PLL_INDIV_20_40_MHZ_VAL;
+ } else if (da7219->mclk_rate <= 4500000) {
+ indiv_bits = DA7219_PLL_INDIV_2_TO_4_5_MHZ;
+ indiv = DA7219_PLL_INDIV_2_TO_4_5_MHZ_VAL;
+ } else if (da7219->mclk_rate <= 9000000) {
+ indiv_bits = DA7219_PLL_INDIV_4_5_TO_9_MHZ;
+ indiv = DA7219_PLL_INDIV_4_5_TO_9_MHZ_VAL;
+ } else if (da7219->mclk_rate <= 18000000) {
+ indiv_bits = DA7219_PLL_INDIV_9_TO_18_MHZ;
+ indiv = DA7219_PLL_INDIV_9_TO_18_MHZ_VAL;
+ } else if (da7219->mclk_rate <= 36000000) {
+ indiv_bits = DA7219_PLL_INDIV_18_TO_36_MHZ;
+ indiv = DA7219_PLL_INDIV_18_TO_36_MHZ_VAL;
} else if (da7219->mclk_rate <= 54000000) {
- indiv_bits = DA7219_PLL_INDIV_40_54_MHZ;
- indiv = DA7219_PLL_INDIV_40_54_MHZ_VAL;
+ indiv_bits = DA7219_PLL_INDIV_36_TO_54_MHZ;
+ indiv = DA7219_PLL_INDIV_36_TO_54_MHZ_VAL;
} else {
dev_err(codec->dev, "PLL input clock %d above valid range\n",
da7219->mclk_rate);
@@ -1426,6 +1427,12 @@ static const struct of_device_id da7219_of_match[] = {
};
MODULE_DEVICE_TABLE(of, da7219_of_match);
+static const struct acpi_device_id da7219_acpi_match[] = {
+ { .id = "DLGS7219", },
+ { }
+};
+MODULE_DEVICE_TABLE(acpi, da7219_acpi_match);
+
static enum da7219_micbias_voltage
da7219_of_micbias_lvl(struct snd_soc_codec *codec, u32 val)
{
@@ -1955,6 +1962,7 @@ static struct i2c_driver da7219_i2c_driver = {
.driver = {
.name = "da7219",
.of_match_table = of_match_ptr(da7219_of_match),
+ .acpi_match_table = ACPI_PTR(da7219_acpi_match),
},
.probe = da7219_i2c_probe,
.remove = da7219_i2c_remove,
diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h
index 5a787e738084..ff2a2f02ce40 100644
--- a/sound/soc/codecs/da7219.h
+++ b/sound/soc/codecs/da7219.h
@@ -194,11 +194,11 @@
/* DA7219_PLL_CTRL = 0x20 */
#define DA7219_PLL_INDIV_SHIFT 2
#define DA7219_PLL_INDIV_MASK (0x7 << 2)
-#define DA7219_PLL_INDIV_2_5_MHZ (0x0 << 2)
-#define DA7219_PLL_INDIV_5_10_MHZ (0x1 << 2)
-#define DA7219_PLL_INDIV_10_20_MHZ (0x2 << 2)
-#define DA7219_PLL_INDIV_20_40_MHZ (0x3 << 2)
-#define DA7219_PLL_INDIV_40_54_MHZ (0x4 << 2)
+#define DA7219_PLL_INDIV_2_TO_4_5_MHZ (0x0 << 2)
+#define DA7219_PLL_INDIV_4_5_TO_9_MHZ (0x1 << 2)
+#define DA7219_PLL_INDIV_9_TO_18_MHZ (0x2 << 2)
+#define DA7219_PLL_INDIV_18_TO_36_MHZ (0x3 << 2)
+#define DA7219_PLL_INDIV_36_TO_54_MHZ (0x4 << 2)
#define DA7219_PLL_MCLK_SQR_EN_SHIFT 5
#define DA7219_PLL_MCLK_SQR_EN_MASK (0x1 << 5)
#define DA7219_PLL_MODE_SHIFT 6
@@ -761,11 +761,11 @@
#define DA7219_PLL_FREQ_OUT_98304 98304000
/* PLL Frequency Dividers */
-#define DA7219_PLL_INDIV_2_5_MHZ_VAL 1
-#define DA7219_PLL_INDIV_5_10_MHZ_VAL 2
-#define DA7219_PLL_INDIV_10_20_MHZ_VAL 4
-#define DA7219_PLL_INDIV_20_40_MHZ_VAL 8
-#define DA7219_PLL_INDIV_40_54_MHZ_VAL 16
+#define DA7219_PLL_INDIV_2_TO_4_5_MHZ_VAL 1
+#define DA7219_PLL_INDIV_4_5_TO_9_MHZ_VAL 2
+#define DA7219_PLL_INDIV_9_TO_18_MHZ_VAL 4
+#define DA7219_PLL_INDIV_18_TO_36_MHZ_VAL 8
+#define DA7219_PLL_INDIV_36_TO_54_MHZ_VAL 16
/* SRM */
#define DA7219_SRM_CHECK_RETRIES 8
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index afa6c5db9dcc..2086d7107622 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -26,18 +26,30 @@
#include <sound/tlv.h>
#include "es8328.h"
-#define ES8328_SYSCLK_RATE_1X 11289600
-#define ES8328_SYSCLK_RATE_2X 22579200
+static const unsigned int rates_12288[] = {
+ 8000, 12000, 16000, 24000, 32000, 48000, 96000,
+};
-/* Run the codec at 22.5792 or 11.2896 MHz to support these rates */
-static struct {
- int rate;
- u8 ratio;
-} mclk_ratios[] = {
- { 8000, 9 },
- {11025, 7 },
- {22050, 4 },
- {44100, 2 },
+static const int ratios_12288[] = {
+ 10, 7, 6, 4, 3, 2, 0,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_12288 = {
+ .count = ARRAY_SIZE(rates_12288),
+ .list = rates_12288,
+};
+
+static const unsigned int rates_11289[] = {
+ 8018, 11025, 22050, 44100, 88200,
+};
+
+static const int ratios_11289[] = {
+ 9, 7, 4, 2, 0,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_11289 = {
+ .count = ARRAY_SIZE(rates_11289),
+ .list = rates_11289,
};
/* regulator supplies for sgtl5000, VDDD is an optional external supply */
@@ -57,16 +69,28 @@ static const char * const supply_names[ES8328_SUPPLY_NUM] = {
"HPVDD",
};
-#define ES8328_RATES (SNDRV_PCM_RATE_44100 | \
+#define ES8328_RATES (SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_32000 | \
SNDRV_PCM_RATE_22050 | \
- SNDRV_PCM_RATE_11025)
-#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
+ SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_11025 | \
+ SNDRV_PCM_RATE_8000)
+#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
struct es8328_priv {
struct regmap *regmap;
struct clk *clk;
int playback_fs;
bool deemph;
+ int mclkdiv2;
+ const struct snd_pcm_hw_constraint_list *sysclk_constraints;
+ const int *mclk_ratios;
struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM];
};
@@ -439,54 +463,131 @@ static int es8328_mute(struct snd_soc_dai *dai, int mute)
mute ? ES8328_DACCONTROL3_DACMUTE : 0);
}
+static int es8328_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+
+ if (es8328->sysclk_constraints)
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ es8328->sysclk_constraints);
+
+ return 0;
+}
+
static int es8328_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
- int clk_rate;
int i;
int reg;
- u8 ratio;
+ int wl;
+ int ratio;
+
+ if (!es8328->sysclk_constraints) {
+ dev_err(codec->dev, "No MCLK configured\n");
+ return -EINVAL;
+ }
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
reg = ES8328_DACCONTROL2;
else
reg = ES8328_ADCCONTROL5;
- clk_rate = clk_get_rate(es8328->clk);
+ for (i = 0; i < es8328->sysclk_constraints->count; i++)
+ if (es8328->sysclk_constraints->list[i] == params_rate(params))
+ break;
- if ((clk_rate != ES8328_SYSCLK_RATE_1X) &&
- (clk_rate != ES8328_SYSCLK_RATE_2X)) {
- dev_err(codec->dev,
- "%s: clock is running at %d Hz, not %d or %d Hz\n",
- __func__, clk_rate,
- ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X);
+ if (i == es8328->sysclk_constraints->count) {
+ dev_err(codec->dev, "LRCLK %d unsupported with current clock\n",
+ params_rate(params));
return -EINVAL;
}
- /* find master mode MCLK to sampling frequency ratio */
- ratio = mclk_ratios[0].rate;
- for (i = 1; i < ARRAY_SIZE(mclk_ratios); i++)
- if (params_rate(params) <= mclk_ratios[i].rate)
- ratio = mclk_ratios[i].ratio;
+ ratio = es8328->mclk_ratios[i];
+ snd_soc_update_bits(codec, ES8328_MASTERMODE,
+ ES8328_MASTERMODE_MCLKDIV2,
+ es8328->mclkdiv2 ? ES8328_MASTERMODE_MCLKDIV2 : 0);
+
+ switch (params_width(params)) {
+ case 16:
+ wl = 3;
+ break;
+ case 18:
+ wl = 2;
+ break;
+ case 20:
+ wl = 1;
+ break;
+ case 24:
+ wl = 0;
+ break;
+ case 32:
+ wl = 4;
+ break;
+ default:
+ return -EINVAL;
+ }
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ snd_soc_update_bits(codec, ES8328_DACCONTROL1,
+ ES8328_DACCONTROL1_DACWL_MASK,
+ wl << ES8328_DACCONTROL1_DACWL_SHIFT);
+
es8328->playback_fs = params_rate(params);
es8328_set_deemph(codec);
- }
+ } else
+ snd_soc_update_bits(codec, ES8328_ADCCONTROL4,
+ ES8328_ADCCONTROL4_ADCWL_MASK,
+ wl << ES8328_ADCCONTROL4_ADCWL_SHIFT);
return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio);
}
+static int es8328_set_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+ int mclkdiv2 = 0;
+
+ switch (freq) {
+ case 0:
+ es8328->sysclk_constraints = NULL;
+ es8328->mclk_ratios = NULL;
+ break;
+ case 22579200:
+ mclkdiv2 = 1;
+ /* fallthru */
+ case 11289600:
+ es8328->sysclk_constraints = &constraints_11289;
+ es8328->mclk_ratios = ratios_11289;
+ break;
+ case 24576000:
+ mclkdiv2 = 1;
+ /* fallthru */
+ case 12288000:
+ es8328->sysclk_constraints = &constraints_12288;
+ es8328->mclk_ratios = ratios_12288;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ es8328->mclkdiv2 = mclkdiv2;
+ return 0;
+}
+
static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
- struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
- int clk_rate;
- u8 mode = ES8328_DACCONTROL1_DACWL_16;
+ u8 dac_mode = 0;
+ u8 adc_mode = 0;
/* set master/slave audio interface */
if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM)
@@ -495,13 +596,16 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
- mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+ dac_mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+ adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_I2S;
break;
case SND_SOC_DAIFMT_RIGHT_J:
- mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+ dac_mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+ adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_RJUST;
break;
case SND_SOC_DAIFMT_LEFT_J:
- mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+ dac_mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+ adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_LJUST;
break;
default:
return -EINVAL;
@@ -511,18 +615,14 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF)
return -EINVAL;
- snd_soc_write(codec, ES8328_DACCONTROL1, mode);
- snd_soc_write(codec, ES8328_ADCCONTROL4, mode);
+ snd_soc_update_bits(codec, ES8328_DACCONTROL1,
+ ES8328_DACCONTROL1_DACFORMAT_MASK, dac_mode);
+ snd_soc_update_bits(codec, ES8328_ADCCONTROL4,
+ ES8328_ADCCONTROL4_ADCFORMAT_MASK, adc_mode);
/* Master serial port mode, with BCLK generated automatically */
- clk_rate = clk_get_rate(es8328->clk);
- if (clk_rate == ES8328_SYSCLK_RATE_1X)
- snd_soc_write(codec, ES8328_MASTERMODE,
- ES8328_MASTERMODE_MSC);
- else
- snd_soc_write(codec, ES8328_MASTERMODE,
- ES8328_MASTERMODE_MCLKDIV2 |
- ES8328_MASTERMODE_MSC);
+ snd_soc_update_bits(codec, ES8328_MASTERMODE,
+ ES8328_MASTERMODE_MSC, ES8328_MASTERMODE_MSC);
return 0;
}
@@ -579,8 +679,10 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec,
}
static const struct snd_soc_dai_ops es8328_dai_ops = {
+ .startup = es8328_startup,
.hw_params = es8328_hw_params,
.digital_mute = es8328_mute,
+ .set_sysclk = es8328_set_sysclk,
.set_fmt = es8328_set_dai_fmt,
};
@@ -601,6 +703,7 @@ static struct snd_soc_dai_driver es8328_dai = {
.formats = ES8328_FORMATS,
},
.ops = &es8328_dai_ops,
+ .symmetric_rates = 1,
};
static int es8328_suspend(struct snd_soc_codec *codec)
@@ -708,6 +811,7 @@ const struct regmap_config es8328_regmap_config = {
.val_bits = 8,
.max_register = ES8328_REG_MAX,
.cache_type = REGCACHE_RBTREE,
+ .use_single_rw = true,
};
EXPORT_SYMBOL_GPL(es8328_regmap_config);
diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h
index 156c748c89c7..1a736e72a929 100644
--- a/sound/soc/codecs/es8328.h
+++ b/sound/soc/codecs/es8328.h
@@ -22,7 +22,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
#define ES8328_CONTROL1_VMIDSEL_50k (1 << 0)
#define ES8328_CONTROL1_VMIDSEL_500k (2 << 0)
#define ES8328_CONTROL1_VMIDSEL_5k (3 << 0)
-#define ES8328_CONTROL1_VMIDSEL_MASK (7 << 0)
+#define ES8328_CONTROL1_VMIDSEL_MASK (3 << 0)
#define ES8328_CONTROL1_ENREF (1 << 2)
#define ES8328_CONTROL1_SEQEN (1 << 3)
#define ES8328_CONTROL1_SAMEFS (1 << 4)
@@ -84,7 +84,20 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
#define ES8328_ADCCONTROL1 0x09
#define ES8328_ADCCONTROL2 0x0a
#define ES8328_ADCCONTROL3 0x0b
+
#define ES8328_ADCCONTROL4 0x0c
+#define ES8328_ADCCONTROL4_ADCFORMAT_MASK (3 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_I2S (0 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_LJUST (1 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_RJUST (2 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_PCM (3 << 0)
+#define ES8328_ADCCONTROL4_ADCWL_SHIFT 2
+#define ES8328_ADCCONTROL4_ADCWL_MASK (7 << 2)
+#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_NORMAL (0 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_INV (1 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK2 (0 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK1 (1 << 5)
+
#define ES8328_ADCCONTROL5 0x0d
#define ES8328_ADCCONTROL5_RATEMASK (0x1f << 0)
@@ -109,15 +122,13 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
#define ES8328_ADCCONTROL14 0x16
#define ES8328_DACCONTROL1 0x17
+#define ES8328_DACCONTROL1_DACFORMAT_MASK (3 << 1)
#define ES8328_DACCONTROL1_DACFORMAT_I2S (0 << 1)
#define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1)
#define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1)
#define ES8328_DACCONTROL1_DACFORMAT_PCM (3 << 1)
-#define ES8328_DACCONTROL1_DACWL_24 (0 << 3)
-#define ES8328_DACCONTROL1_DACWL_20 (1 << 3)
-#define ES8328_DACCONTROL1_DACWL_18 (2 << 3)
-#define ES8328_DACCONTROL1_DACWL_16 (3 << 3)
-#define ES8328_DACCONTROL1_DACWL_32 (4 << 3)
+#define ES8328_DACCONTROL1_DACWL_SHIFT 3
+#define ES8328_DACCONTROL1_DACWL_MASK (7 << 3)
#define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6)
#define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6)
#define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK2 (0 << 6)
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index aaa038ffc8a5..181cd3bf0b92 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -29,6 +29,7 @@
#include <sound/hdaudio_ext.h>
#include <sound/hda_i915.h>
#include <sound/pcm_drm_eld.h>
+#include <sound/hda_chmap.h>
#include "../../hda/local.h"
#include "hdac_hdmi.h"
@@ -60,11 +61,17 @@ struct hdac_hdmi_cvt {
struct hdac_hdmi_cvt_params params;
};
+/* Currently only spk_alloc, more to be added */
+struct hdac_hdmi_parsed_eld {
+ u8 spk_alloc;
+};
+
struct hdac_hdmi_eld {
bool monitor_present;
bool eld_valid;
int eld_size;
char eld_buffer[ELD_MAX_SIZE];
+ struct hdac_hdmi_parsed_eld info;
};
struct hdac_hdmi_pin {
@@ -76,6 +83,10 @@ struct hdac_hdmi_pin {
struct hdac_ext_device *edev;
int repoll_count;
struct delayed_work work;
+ struct mutex lock;
+ bool chmap_set;
+ unsigned char chmap[8]; /* ALSA API channel-map */
+ int channels; /* current number of channels */
};
struct hdac_hdmi_pcm {
@@ -100,8 +111,22 @@ struct hdac_hdmi_priv {
int num_pin;
int num_cvt;
struct mutex pin_mutex;
+ struct hdac_chmap chmap;
};
+static struct hdac_hdmi_pcm *get_hdmi_pcm_from_id(struct hdac_hdmi_priv *hdmi,
+ int pcm_idx)
+{
+ struct hdac_hdmi_pcm *pcm;
+
+ list_for_each_entry(pcm, &hdmi->pcm_list, head) {
+ if (pcm->pcm_id == pcm_idx)
+ return pcm;
+ }
+
+ return NULL;
+}
+
static inline struct hdac_ext_device *to_hda_ext_device(struct device *dev)
{
struct hdac_device *hdac = dev_to_hdac_dev(dev);
@@ -278,26 +303,31 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac,
int i;
const u8 *eld_buf;
u8 conn_type;
- int channels = 2;
+ int channels, ca;
list_for_each_entry(pin, &hdmi->pin_list, head) {
if (pin->nid == pin_nid)
break;
}
+ ca = snd_hdac_channel_allocation(&hdac->hdac, pin->eld.info.spk_alloc,
+ pin->channels, pin->chmap_set, true, pin->chmap);
+
+ channels = snd_hdac_get_active_channels(ca);
+ hdmi->chmap.ops.set_channel_count(&hdac->hdac, cvt_nid, channels);
+
+ snd_hdac_setup_channel_mapping(&hdmi->chmap, pin->nid, false, ca,
+ pin->channels, pin->chmap, pin->chmap_set);
+
eld_buf = pin->eld.eld_buffer;
conn_type = drm_eld_get_conn_type(eld_buf);
- /* setup channel count */
- snd_hdac_codec_write(&hdac->hdac, cvt_nid, 0,
- AC_VERB_SET_CVT_CHAN_COUNT, channels - 1);
-
switch (conn_type) {
case DRM_ELD_CONN_TYPE_HDMI:
hdmi_audio_infoframe_init(&frame);
- /* Default stereo for now */
frame.channels = channels;
+ frame.channel_allocation = ca;
ret = hdmi_audio_infoframe_pack(&frame, buffer, sizeof(buffer));
if (ret < 0)
@@ -311,7 +341,7 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac,
dp_ai.len = 0x1b;
dp_ai.ver = 0x11 << 2;
dp_ai.CC02_CT47 = channels - 1;
- dp_ai.CA = 0;
+ dp_ai.CA = ca;
dip = (u8 *)&dp_ai;
break;
@@ -370,17 +400,23 @@ static int hdac_hdmi_playback_prepare(struct snd_pcm_substream *substream,
struct hdac_ext_device *hdac = snd_soc_dai_get_drvdata(dai);
struct hdac_hdmi_priv *hdmi = hdac->private_data;
struct hdac_hdmi_dai_pin_map *dai_map;
+ struct hdac_hdmi_pin *pin;
struct hdac_ext_dma_params *dd;
int ret;
dai_map = &hdmi->dai_map[dai->id];
+ pin = dai_map->pin;
dd = (struct hdac_ext_dma_params *)snd_soc_dai_get_dma_data(dai, substream);
dev_dbg(&hdac->hdac.dev, "stream tag from cpu dai %d format in cvt 0x%x\n",
dd->stream_tag, dd->format);
+ mutex_lock(&pin->lock);
+ pin->channels = substream->runtime->channels;
+
ret = hdac_hdmi_setup_audio_infoframe(hdac, dai_map->cvt->nid,
dai_map->pin->nid);
+ mutex_unlock(&pin->lock);
if (ret < 0)
return ret;
@@ -640,6 +676,12 @@ static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream,
snd_hdac_codec_write(&hdac->hdac, dai_map->pin->nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+ mutex_lock(&dai_map->pin->lock);
+ dai_map->pin->chmap_set = false;
+ memset(dai_map->pin->chmap, 0, sizeof(dai_map->pin->chmap));
+ dai_map->pin->channels = 0;
+ mutex_unlock(&dai_map->pin->lock);
+
dai_map->pin = NULL;
}
}
@@ -647,10 +689,19 @@ static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream,
static int
hdac_hdmi_query_cvt_params(struct hdac_device *hdac, struct hdac_hdmi_cvt *cvt)
{
+ unsigned int chans;
+ struct hdac_ext_device *edev = to_ehdac_device(hdac);
+ struct hdac_hdmi_priv *hdmi = edev->private_data;
int err;
- /* Only stereo supported as of now */
- cvt->params.channels_min = cvt->params.channels_max = 2;
+ chans = get_wcaps(hdac, cvt->nid);
+ chans = get_wcaps_channels(chans);
+
+ cvt->params.channels_min = 2;
+
+ cvt->params.channels_max = chans;
+ if (chans > hdmi->chmap.channels_max)
+ hdmi->chmap.channels_max = chans;
err = snd_hdac_query_supported_pcm(hdac, cvt->nid,
&cvt->params.rates,
@@ -1008,6 +1059,12 @@ static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid)
return hdac_hdmi_query_cvt_params(&edev->hdac, cvt);
}
+static void hdac_hdmi_parse_eld(struct hdac_ext_device *edev,
+ struct hdac_hdmi_pin *pin)
+{
+ pin->eld.info.spk_alloc = pin->eld.eld_buffer[DRM_ELD_SPEAKER];
+}
+
static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll)
{
struct hdac_ext_device *edev = pin->edev;
@@ -1065,6 +1122,7 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll)
snd_jack_report(pcm->jack, SND_JACK_AVOUT);
}
+ hdac_hdmi_parse_eld(edev, pin);
print_hex_dump_bytes("ELD: ", DUMP_PREFIX_OFFSET,
pin->eld.eld_buffer, pin->eld.eld_size);
@@ -1123,6 +1181,7 @@ static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid)
hdmi->num_pin++;
pin->edev = edev;
+ mutex_init(&pin->lock);
INIT_DELAYED_WORK(&pin->work, hdac_hdmi_repoll_eld);
return 0;
@@ -1342,6 +1401,19 @@ static struct i915_audio_component_audio_ops aops = {
.pin_eld_notify = hdac_hdmi_eld_notify_cb,
};
+static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card,
+ int device)
+{
+ struct snd_soc_pcm_runtime *rtd;
+
+ list_for_each_entry(rtd, &card->rtd_list, list) {
+ if (rtd->pcm && (rtd->pcm->device == device))
+ return rtd->pcm;
+ }
+
+ return NULL;
+}
+
int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device)
{
char jack_name[NAME_SIZE];
@@ -1351,6 +1423,8 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device)
snd_soc_component_get_dapm(&codec->component);
struct hdac_hdmi_priv *hdmi = edev->private_data;
struct hdac_hdmi_pcm *pcm;
+ struct snd_pcm *snd_pcm;
+ int err;
/*
* this is a new PCM device, create new pcm and
@@ -1362,6 +1436,18 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device)
pcm->pcm_id = device;
pcm->cvt = hdmi->dai_map[dai->id].cvt;
+ snd_pcm = hdac_hdmi_get_pcm_from_id(dai->component->card, device);
+ if (snd_pcm) {
+ err = snd_hdac_add_chmap_ctls(snd_pcm, device, &hdmi->chmap);
+ if (err < 0) {
+ dev_err(&edev->hdac.dev,
+ "chmap control add failed with err: %d for pcm: %d\n",
+ err, device);
+ kfree(pcm);
+ return err;
+ }
+ }
+
list_add_tail(&pcm->head, &hdmi->pcm_list);
sprintf(jack_name, "HDMI/DP, pcm=%d Jack", device);
@@ -1378,10 +1464,18 @@ static int hdmi_codec_probe(struct snd_soc_codec *codec)
struct snd_soc_dapm_context *dapm =
snd_soc_component_get_dapm(&codec->component);
struct hdac_hdmi_pin *pin;
+ struct hdac_ext_link *hlink = NULL;
int ret;
edev->scodec = codec;
+ /*
+ * hold the ref while we probe, also no need to drop the ref on
+ * exit, we call pm_runtime_suspend() so that will do for us
+ */
+ hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev));
+ snd_hdac_ext_bus_link_get(edev->ebus, hlink);
+
ret = create_fill_widget_route_map(dapm);
if (ret < 0)
return ret;
@@ -1475,19 +1569,83 @@ static struct snd_soc_codec_driver hdmi_hda_codec = {
.idle_bias_off = true,
};
+static void hdac_hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx,
+ unsigned char *chmap)
+{
+ struct hdac_ext_device *edev = to_ehdac_device(hdac);
+ struct hdac_hdmi_priv *hdmi = edev->private_data;
+ struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx);
+ struct hdac_hdmi_pin *pin = pcm->pin;
+
+ /* chmap is already set to 0 in caller */
+ if (!pin)
+ return;
+
+ memcpy(chmap, pin->chmap, ARRAY_SIZE(pin->chmap));
+}
+
+static void hdac_hdmi_set_chmap(struct hdac_device *hdac, int pcm_idx,
+ unsigned char *chmap, int prepared)
+{
+ struct hdac_ext_device *edev = to_ehdac_device(hdac);
+ struct hdac_hdmi_priv *hdmi = edev->private_data;
+ struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx);
+ struct hdac_hdmi_pin *pin = pcm->pin;
+
+ mutex_lock(&pin->lock);
+ pin->chmap_set = true;
+ memcpy(pin->chmap, chmap, ARRAY_SIZE(pin->chmap));
+ if (prepared)
+ hdac_hdmi_setup_audio_infoframe(edev, pcm->cvt->nid, pin->nid);
+ mutex_unlock(&pin->lock);
+}
+
+static bool is_hdac_hdmi_pcm_attached(struct hdac_device *hdac, int pcm_idx)
+{
+ struct hdac_ext_device *edev = to_ehdac_device(hdac);
+ struct hdac_hdmi_priv *hdmi = edev->private_data;
+ struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx);
+ struct hdac_hdmi_pin *pin = pcm->pin;
+
+ return pin ? true:false;
+}
+
+static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx)
+{
+ struct hdac_ext_device *edev = to_ehdac_device(hdac);
+ struct hdac_hdmi_priv *hdmi = edev->private_data;
+ struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx);
+ struct hdac_hdmi_pin *pin = pcm->pin;
+
+ if (!pin || !pin->eld.eld_valid)
+ return 0;
+
+ return pin->eld.info.spk_alloc;
+}
+
static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev)
{
struct hdac_device *codec = &edev->hdac;
struct hdac_hdmi_priv *hdmi_priv;
struct snd_soc_dai_driver *hdmi_dais = NULL;
+ struct hdac_ext_link *hlink = NULL;
int num_dais = 0;
int ret = 0;
+ /* hold the ref while we probe */
+ hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev));
+ snd_hdac_ext_bus_link_get(edev->ebus, hlink);
+
hdmi_priv = devm_kzalloc(&codec->dev, sizeof(*hdmi_priv), GFP_KERNEL);
if (hdmi_priv == NULL)
return -ENOMEM;
edev->private_data = hdmi_priv;
+ snd_hdac_register_chmap_ops(codec, &hdmi_priv->chmap);
+ hdmi_priv->chmap.ops.get_chmap = hdac_hdmi_get_chmap;
+ hdmi_priv->chmap.ops.set_chmap = hdac_hdmi_set_chmap;
+ hdmi_priv->chmap.ops.is_pcm_attached = is_hdac_hdmi_pcm_attached;
+ hdmi_priv->chmap.ops.get_spk_alloc = hdac_hdmi_get_spk_alloc;
dev_set_drvdata(&codec->dev, edev);
@@ -1516,8 +1674,12 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev)
}
/* ASoC specific initialization */
- return snd_soc_register_codec(&codec->dev, &hdmi_hda_codec,
- hdmi_dais, num_dais);
+ ret = snd_soc_register_codec(&codec->dev, &hdmi_hda_codec,
+ hdmi_dais, num_dais);
+
+ snd_hdac_ext_bus_link_put(edev->ebus, hlink);
+
+ return ret;
}
static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev)
@@ -1556,6 +1718,8 @@ static int hdac_hdmi_runtime_suspend(struct device *dev)
struct hdac_ext_device *edev = to_hda_ext_device(dev);
struct hdac_device *hdac = &edev->hdac;
struct hdac_bus *bus = hdac->bus;
+ struct hdac_ext_bus *ebus = hbus_to_ebus(bus);
+ struct hdac_ext_link *hlink = NULL;
int err;
dev_dbg(dev, "Enter: %s\n", __func__);
@@ -1579,6 +1743,9 @@ static int hdac_hdmi_runtime_suspend(struct device *dev)
return err;
}
+ hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev));
+ snd_hdac_ext_bus_link_put(ebus, hlink);
+
return 0;
}
@@ -1587,6 +1754,8 @@ static int hdac_hdmi_runtime_resume(struct device *dev)
struct hdac_ext_device *edev = to_hda_ext_device(dev);
struct hdac_device *hdac = &edev->hdac;
struct hdac_bus *bus = hdac->bus;
+ struct hdac_ext_bus *ebus = hbus_to_ebus(bus);
+ struct hdac_ext_link *hlink = NULL;
int err;
dev_dbg(dev, "Enter: %s\n", __func__);
@@ -1595,6 +1764,9 @@ static int hdac_hdmi_runtime_resume(struct device *dev)
if (!bus)
return 0;
+ hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev));
+ snd_hdac_ext_bus_link_get(ebus, hlink);
+
err = snd_hdac_display_power(bus, true);
if (err < 0) {
dev_err(bus->dev, "Cannot turn on display power on i915\n");
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
new file mode 100644
index 000000000000..8e36e883e453
--- /dev/null
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -0,0 +1,432 @@
+/*
+ * ALSA SoC codec for HDMI encoder drivers
+ * Copyright (C) 2015 Texas Instruments Incorporated - http://www.ti.com/
+ * Author: Jyri Sarha <jsarha@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ */
+#include <linux/module.h>
+#include <linux/string.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pcm_drm_eld.h>
+#include <sound/hdmi-codec.h>
+#include <sound/pcm_iec958.h>
+
+#include <drm/drm_crtc.h> /* This is only to get MAX_ELD_BYTES */
+
+struct hdmi_codec_priv {
+ struct hdmi_codec_pdata hcd;
+ struct snd_soc_dai_driver *daidrv;
+ struct hdmi_codec_daifmt daifmt[2];
+ struct mutex current_stream_lock;
+ struct snd_pcm_substream *current_stream;
+ struct snd_pcm_hw_constraint_list ratec;
+ uint8_t eld[MAX_ELD_BYTES];
+};
+
+static const struct snd_soc_dapm_widget hdmi_widgets[] = {
+ SND_SOC_DAPM_OUTPUT("TX"),
+};
+
+static const struct snd_soc_dapm_route hdmi_routes[] = {
+ { "TX", NULL, "Playback" },
+};
+
+enum {
+ DAI_ID_I2S = 0,
+ DAI_ID_SPDIF,
+};
+
+static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+ struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+ uinfo->count = sizeof(hcp->eld);
+
+ return 0;
+}
+
+static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+ struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component);
+
+ memcpy(ucontrol->value.bytes.data, hcp->eld, sizeof(hcp->eld));
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new hdmi_controls[] = {
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "ELD",
+ .info = hdmi_eld_ctl_info,
+ .get = hdmi_eld_ctl_get,
+ },
+};
+
+static int hdmi_codec_new_stream(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+ int ret = 0;
+
+ mutex_lock(&hcp->current_stream_lock);
+ if (!hcp->current_stream) {
+ hcp->current_stream = substream;
+ } else if (hcp->current_stream != substream) {
+ dev_err(dai->dev, "Only one simultaneous stream supported!\n");
+ ret = -EINVAL;
+ }
+ mutex_unlock(&hcp->current_stream_lock);
+
+ return ret;
+}
+
+static int hdmi_codec_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+ int ret = 0;
+
+ dev_dbg(dai->dev, "%s()\n", __func__);
+
+ ret = hdmi_codec_new_stream(substream, dai);
+ if (ret)
+ return ret;
+
+ if (hcp->hcd.ops->audio_startup) {
+ ret = hcp->hcd.ops->audio_startup(dai->dev->parent);
+ if (ret) {
+ mutex_lock(&hcp->current_stream_lock);
+ hcp->current_stream = NULL;
+ mutex_unlock(&hcp->current_stream_lock);
+ return ret;
+ }
+ }
+
+ if (hcp->hcd.ops->get_eld) {
+ ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->eld,
+ sizeof(hcp->eld));
+
+ if (!ret) {
+ ret = snd_pcm_hw_constraint_eld(substream->runtime,
+ hcp->eld);
+ if (ret)
+ return ret;
+ }
+ }
+ return 0;
+}
+
+static void hdmi_codec_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+
+ dev_dbg(dai->dev, "%s()\n", __func__);
+
+ WARN_ON(hcp->current_stream != substream);
+
+ hcp->hcd.ops->audio_shutdown(dai->dev->parent);
+
+ mutex_lock(&hcp->current_stream_lock);
+ hcp->current_stream = NULL;
+ mutex_unlock(&hcp->current_stream_lock);
+}
+
+static int hdmi_codec_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+ struct hdmi_codec_params hp = {
+ .iec = {
+ .status = { 0 },
+ .subcode = { 0 },
+ .pad = 0,
+ .dig_subframe = { 0 },
+ }
+ };
+ int ret;
+
+ dev_dbg(dai->dev, "%s() width %d rate %d channels %d\n", __func__,
+ params_width(params), params_rate(params),
+ params_channels(params));
+
+ if (params_width(params) > 24)
+ params->msbits = 24;
+
+ ret = snd_pcm_create_iec958_consumer_hw_params(params, hp.iec.status,
+ sizeof(hp.iec.status));
+ if (ret < 0) {
+ dev_err(dai->dev, "Creating IEC958 channel status failed %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = hdmi_codec_new_stream(substream, dai);
+ if (ret)
+ return ret;
+
+ hdmi_audio_infoframe_init(&hp.cea);
+ hp.cea.channels = params_channels(params);
+ hp.cea.coding_type = HDMI_AUDIO_CODING_TYPE_STREAM;
+ hp.cea.sample_size = HDMI_AUDIO_SAMPLE_SIZE_STREAM;
+ hp.cea.sample_frequency = HDMI_AUDIO_SAMPLE_FREQUENCY_STREAM;
+
+ hp.sample_width = params_width(params);
+ hp.sample_rate = params_rate(params);
+ hp.channels = params_channels(params);
+
+ return hcp->hcd.ops->hw_params(dai->dev->parent, &hcp->daifmt[dai->id],
+ &hp);
+}
+
+static int hdmi_codec_set_fmt(struct snd_soc_dai *dai,
+ unsigned int fmt)
+{
+ struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+ struct hdmi_codec_daifmt cf = { 0 };
+ int ret = 0;
+
+ dev_dbg(dai->dev, "%s()\n", __func__);
+
+ if (dai->id == DAI_ID_SPDIF) {
+ cf.fmt = HDMI_SPDIF;
+ } else {
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ cf.bit_clk_master = 1;
+ cf.frame_clk_master = 1;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ cf.frame_clk_master = 1;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ cf.bit_clk_master = 1;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ cf.frame_clk_inv = 1;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ cf.bit_clk_inv = 1;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ cf.frame_clk_inv = 1;
+ cf.bit_clk_inv = 1;
+ break;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ cf.fmt = HDMI_I2S;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ cf.fmt = HDMI_DSP_A;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ cf.fmt = HDMI_DSP_B;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ cf.fmt = HDMI_RIGHT_J;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ cf.fmt = HDMI_LEFT_J;
+ break;
+ case SND_SOC_DAIFMT_AC97:
+ cf.fmt = HDMI_AC97;
+ break;
+ default:
+ dev_err(dai->dev, "Invalid DAI interface format\n");
+ return -EINVAL;
+ }
+ }
+
+ hcp->daifmt[dai->id] = cf;
+
+ return ret;
+}
+
+static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+
+ dev_dbg(dai->dev, "%s()\n", __func__);
+
+ if (hcp->hcd.ops->digital_mute)
+ return hcp->hcd.ops->digital_mute(dai->dev->parent, mute);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops hdmi_dai_ops = {
+ .startup = hdmi_codec_startup,
+ .shutdown = hdmi_codec_shutdown,
+ .hw_params = hdmi_codec_hw_params,
+ .set_fmt = hdmi_codec_set_fmt,
+ .digital_mute = hdmi_codec_digital_mute,
+};
+
+
+#define HDMI_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
+ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\
+ SNDRV_PCM_RATE_192000)
+
+#define SPDIF_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |\
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE)
+
+/*
+ * This list is only for formats allowed on the I2S bus. So there is
+ * some formats listed that are not supported by HDMI interface. For
+ * instance allowing the 32-bit formats enables 24-precision with CPU
+ * DAIs that do not support 24-bit formats. If the extra formats cause
+ * problems, we should add the video side driver an option to disable
+ * them.
+ */
+#define I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |\
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE |\
+ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE)
+
+static struct snd_soc_dai_driver hdmi_i2s_dai = {
+ .name = "i2s-hifi",
+ .id = DAI_ID_I2S,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 8,
+ .rates = HDMI_RATES,
+ .formats = I2S_FORMATS,
+ .sig_bits = 24,
+ },
+ .ops = &hdmi_dai_ops,
+};
+
+static const struct snd_soc_dai_driver hdmi_spdif_dai = {
+ .name = "spdif-hifi",
+ .id = DAI_ID_SPDIF,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = HDMI_RATES,
+ .formats = SPDIF_FORMATS,
+ },
+ .ops = &hdmi_dai_ops,
+};
+
+static struct snd_soc_codec_driver hdmi_codec = {
+ .controls = hdmi_controls,
+ .num_controls = ARRAY_SIZE(hdmi_controls),
+ .dapm_widgets = hdmi_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets),
+ .dapm_routes = hdmi_routes,
+ .num_dapm_routes = ARRAY_SIZE(hdmi_routes),
+};
+
+static int hdmi_codec_probe(struct platform_device *pdev)
+{
+ struct hdmi_codec_pdata *hcd = pdev->dev.platform_data;
+ struct device *dev = &pdev->dev;
+ struct hdmi_codec_priv *hcp;
+ int dai_count, i = 0;
+ int ret;
+
+ dev_dbg(dev, "%s()\n", __func__);
+
+ if (!hcd) {
+ dev_err(dev, "%s: No plalform data\n", __func__);
+ return -EINVAL;
+ }
+
+ dai_count = hcd->i2s + hcd->spdif;
+ if (dai_count < 1 || !hcd->ops || !hcd->ops->hw_params ||
+ !hcd->ops->audio_shutdown) {
+ dev_err(dev, "%s: Invalid parameters\n", __func__);
+ return -EINVAL;
+ }
+
+ hcp = devm_kzalloc(dev, sizeof(*hcp), GFP_KERNEL);
+ if (!hcp)
+ return -ENOMEM;
+
+ hcp->hcd = *hcd;
+ mutex_init(&hcp->current_stream_lock);
+
+ hcp->daidrv = devm_kzalloc(dev, dai_count * sizeof(*hcp->daidrv),
+ GFP_KERNEL);
+ if (!hcp->daidrv)
+ return -ENOMEM;
+
+ if (hcd->i2s) {
+ hcp->daidrv[i] = hdmi_i2s_dai;
+ hcp->daidrv[i].playback.channels_max =
+ hcd->max_i2s_channels;
+ i++;
+ }
+
+ if (hcd->spdif)
+ hcp->daidrv[i] = hdmi_spdif_dai;
+
+ ret = snd_soc_register_codec(dev, &hdmi_codec, hcp->daidrv,
+ dai_count);
+ if (ret) {
+ dev_err(dev, "%s: snd_soc_register_codec() failed (%d)\n",
+ __func__, ret);
+ return ret;
+ }
+
+ dev_set_drvdata(dev, hcp);
+ return 0;
+}
+
+static int hdmi_codec_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver hdmi_codec_driver = {
+ .driver = {
+ .name = HDMI_CODEC_DRV_NAME,
+ },
+ .probe = hdmi_codec_probe,
+ .remove = hdmi_codec_remove,
+};
+
+module_platform_driver(hdmi_codec_driver);
+
+MODULE_AUTHOR("Jyri Sarha <jsarha@ti.com>");
+MODULE_DESCRIPTION("HDMI Audio Codec Driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" HDMI_CODEC_DRV_NAME);
diff --git a/sound/soc/codecs/pcm5102a.c b/sound/soc/codecs/pcm5102a.c
new file mode 100644
index 000000000000..ed515677409b
--- /dev/null
+++ b/sound/soc/codecs/pcm5102a.c
@@ -0,0 +1,69 @@
+/*
+ * Driver for the PCM5102A codec
+ *
+ * Author: Florian Meier <florian.meier@koalo.de>
+ * Copyright 2013
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+
+#include <sound/soc.h>
+
+static struct snd_soc_dai_driver pcm5102a_dai = {
+ .name = "pcm5102a-hifi",
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE
+ },
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_pcm5102a;
+
+static int pcm5102a_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pcm5102a,
+ &pcm5102a_dai, 1);
+}
+
+static int pcm5102a_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+static const struct of_device_id pcm5102a_of_match[] = {
+ { .compatible = "ti,pcm5102a", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, pcm5102a_of_match);
+
+static struct platform_driver pcm5102a_codec_driver = {
+ .probe = pcm5102a_probe,
+ .remove = pcm5102a_remove,
+ .driver = {
+ .name = "pcm5102a-codec",
+ .owner = THIS_MODULE,
+ .of_match_table = pcm5102a_of_match,
+ },
+};
+
+module_platform_driver(pcm5102a_codec_driver);
+
+MODULE_DESCRIPTION("ASoC PCM5102A codec driver");
+MODULE_AUTHOR("Florian Meier <florian.meier@koalo.de>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c
index f0e6c06e89ac..a1aaffc20862 100644
--- a/sound/soc/codecs/rt298.c
+++ b/sound/soc/codecs/rt298.c
@@ -17,6 +17,7 @@
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/spi/spi.h>
+#include <linux/dmi.h>
#include <linux/acpi.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -1132,6 +1133,17 @@ static const struct acpi_device_id rt298_acpi_match[] = {
};
MODULE_DEVICE_TABLE(acpi, rt298_acpi_match);
+static const struct dmi_system_id force_combo_jack_table[] = {
+ {
+ .ident = "Intel Broxton P",
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Intel Corp"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "Broxton P")
+ }
+ },
+ { }
+};
+
static int rt298_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -1184,11 +1196,16 @@ static int rt298_i2c_probe(struct i2c_client *i2c,
/* enable jack combo mode on supported devices */
acpiid = acpi_match_device(dev->driver->acpi_match_table, dev);
- if (acpiid) {
+ if (acpiid && acpiid->driver_data) {
rt298->pdata = *(struct rt298_platform_data *)
acpiid->driver_data;
}
+ if (dmi_check_system(force_combo_jack_table)) {
+ rt298->pdata.cbj_en = true;
+ rt298->pdata.gpio2_en = false;
+ }
+
/* VREF Charging */
regmap_update_bits(rt298->regmap, 0x04, 0x80, 0x80);
regmap_update_bits(rt298->regmap, 0x1b, 0x860, 0x860);
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 7af5e7380d61..3c6594da6c9c 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3286,10 +3286,8 @@ static void rt5645_jack_detect_work(struct work_struct *work)
if (btn_type == 0)/* button release */
report = rt5645->jack_type;
else {
- if (rt5645->pdata.jd_invert) {
- mod_timer(&rt5645->btn_check_timer,
- msecs_to_jiffies(100));
- }
+ mod_timer(&rt5645->btn_check_timer,
+ msecs_to_jiffies(100));
}
break;
@@ -3557,6 +3555,12 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = {
DMI_MATCH(DMI_SYS_VENDOR, "GOOGLE"),
},
},
+ {
+ .ident = "Google Setzer",
+ .matches = {
+ DMI_MATCH(DMI_PRODUCT_NAME, "Setzer"),
+ },
+ },
{ }
};
@@ -3810,9 +3814,9 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
if (rt5645->pdata.jd_invert) {
regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV);
- setup_timer(&rt5645->btn_check_timer,
- rt5645_btn_check_callback, (unsigned long)rt5645);
}
+ setup_timer(&rt5645->btn_check_timer,
+ rt5645_btn_check_callback, (unsigned long)rt5645);
INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work);
INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work);
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 1bae17ee8817..da60e3fe5ee7 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -2098,10 +2098,14 @@ static int wm5102_probe(struct platform_device *pdev)
static int wm5102_remove(struct platform_device *pdev)
{
+ struct wm5102_priv *wm5102 = platform_get_drvdata(pdev);
+
snd_soc_unregister_platform(&pdev->dev);
snd_soc_unregister_codec(&pdev->dev);
pm_runtime_disable(&pdev->dev);
+ wm_adsp2_remove(&wm5102->core.adsp[0]);
+
return 0;
}
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 2728ac545ffe..b5820e4d5471 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -2437,10 +2437,16 @@ static int wm5110_probe(struct platform_device *pdev)
static int wm5110_remove(struct platform_device *pdev)
{
+ struct wm5110_priv *wm5110 = platform_get_drvdata(pdev);
+ int i;
+
snd_soc_unregister_platform(&pdev->dev);
snd_soc_unregister_codec(&pdev->dev);
pm_runtime_disable(&pdev->dev);
+ for (i = 0; i < WM5110_NUM_ADSP; i++)
+ wm_adsp2_remove(&wm5110->core.adsp[i]);
+
return 0;
}
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index d3b1cb15e7f0..a07bd7c2c587 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -160,6 +160,8 @@
#define ADSP2_RAM_RDY_SHIFT 0
#define ADSP2_RAM_RDY_WIDTH 1
+#define ADSP_MAX_STD_CTRL_SIZE 512
+
struct wm_adsp_buf {
struct list_head list;
void *buf;
@@ -271,8 +273,11 @@ struct wm_adsp_buffer {
__be32 words_written[2]; /* total words written (64 bit) */
};
+struct wm_adsp_compr;
+
struct wm_adsp_compr_buf {
struct wm_adsp *dsp;
+ struct wm_adsp_compr *compr;
struct wm_adsp_buffer_region *regions;
u32 host_buf_ptr;
@@ -435,6 +440,7 @@ struct wm_coeff_ctl {
size_t len;
unsigned int set:1;
struct snd_kcontrol *kcontrol;
+ struct soc_bytes_ext bytes_ext;
unsigned int flags;
};
@@ -711,10 +717,17 @@ static void wm_adsp2_show_fw_status(struct wm_adsp *dsp)
be16_to_cpu(scratch[3]));
}
+static inline struct wm_coeff_ctl *bytes_ext_to_ctl(struct soc_bytes_ext *ext)
+{
+ return container_of(ext, struct wm_coeff_ctl, bytes_ext);
+}
+
static int wm_coeff_info(struct snd_kcontrol *kctl,
struct snd_ctl_elem_info *uinfo)
{
- struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value;
+ struct soc_bytes_ext *bytes_ext =
+ (struct soc_bytes_ext *)kctl->private_value;
+ struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
uinfo->count = ctl->len;
@@ -763,7 +776,9 @@ static int wm_coeff_write_control(struct wm_coeff_ctl *ctl,
static int wm_coeff_put(struct snd_kcontrol *kctl,
struct snd_ctl_elem_value *ucontrol)
{
- struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value;
+ struct soc_bytes_ext *bytes_ext =
+ (struct soc_bytes_ext *)kctl->private_value;
+ struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
char *p = ucontrol->value.bytes.data;
int ret = 0;
@@ -780,6 +795,29 @@ static int wm_coeff_put(struct snd_kcontrol *kctl,
return ret;
}
+static int wm_coeff_tlv_put(struct snd_kcontrol *kctl,
+ const unsigned int __user *bytes, unsigned int size)
+{
+ struct soc_bytes_ext *bytes_ext =
+ (struct soc_bytes_ext *)kctl->private_value;
+ struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
+ int ret = 0;
+
+ mutex_lock(&ctl->dsp->pwr_lock);
+
+ if (copy_from_user(ctl->cache, bytes, size)) {
+ ret = -EFAULT;
+ } else {
+ ctl->set = 1;
+ if (ctl->enabled)
+ ret = wm_coeff_write_control(ctl, ctl->cache, size);
+ }
+
+ mutex_unlock(&ctl->dsp->pwr_lock);
+
+ return ret;
+}
+
static int wm_coeff_read_control(struct wm_coeff_ctl *ctl,
void *buf, size_t len)
{
@@ -822,7 +860,9 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl,
static int wm_coeff_get(struct snd_kcontrol *kctl,
struct snd_ctl_elem_value *ucontrol)
{
- struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value;
+ struct soc_bytes_ext *bytes_ext =
+ (struct soc_bytes_ext *)kctl->private_value;
+ struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
char *p = ucontrol->value.bytes.data;
int ret = 0;
@@ -845,12 +885,72 @@ static int wm_coeff_get(struct snd_kcontrol *kctl,
return ret;
}
+static int wm_coeff_tlv_get(struct snd_kcontrol *kctl,
+ unsigned int __user *bytes, unsigned int size)
+{
+ struct soc_bytes_ext *bytes_ext =
+ (struct soc_bytes_ext *)kctl->private_value;
+ struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
+ int ret = 0;
+
+ mutex_lock(&ctl->dsp->pwr_lock);
+
+ if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) {
+ if (ctl->enabled)
+ ret = wm_coeff_read_control(ctl, ctl->cache, size);
+ else
+ ret = -EPERM;
+ } else {
+ if (!ctl->flags && ctl->enabled)
+ ret = wm_coeff_read_control(ctl, ctl->cache, size);
+ }
+
+ if (!ret && copy_to_user(bytes, ctl->cache, size))
+ ret = -EFAULT;
+
+ mutex_unlock(&ctl->dsp->pwr_lock);
+
+ return ret;
+}
+
struct wmfw_ctl_work {
struct wm_adsp *dsp;
struct wm_coeff_ctl *ctl;
struct work_struct work;
};
+static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len)
+{
+ unsigned int out, rd, wr, vol;
+
+ if (len > ADSP_MAX_STD_CTRL_SIZE) {
+ rd = SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+ wr = SNDRV_CTL_ELEM_ACCESS_TLV_WRITE;
+ vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE;
+
+ out = SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK;
+ } else {
+ rd = SNDRV_CTL_ELEM_ACCESS_READ;
+ wr = SNDRV_CTL_ELEM_ACCESS_WRITE;
+ vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE;
+
+ out = 0;
+ }
+
+ if (in) {
+ if (in & WMFW_CTL_FLAG_READABLE)
+ out |= rd;
+ if (in & WMFW_CTL_FLAG_WRITEABLE)
+ out |= wr;
+ if (in & WMFW_CTL_FLAG_VOLATILE)
+ out |= vol;
+ } else {
+ out |= rd | wr | vol;
+ }
+
+ return out;
+}
+
static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl)
{
struct snd_kcontrol_new *kcontrol;
@@ -868,19 +968,15 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl)
kcontrol->info = wm_coeff_info;
kcontrol->get = wm_coeff_get;
kcontrol->put = wm_coeff_put;
- kcontrol->private_value = (unsigned long)ctl;
+ kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ kcontrol->tlv.c = snd_soc_bytes_tlv_callback;
+ kcontrol->private_value = (unsigned long)&ctl->bytes_ext;
- if (ctl->flags) {
- if (ctl->flags & WMFW_CTL_FLAG_WRITEABLE)
- kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_WRITE;
- if (ctl->flags & WMFW_CTL_FLAG_READABLE)
- kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_READ;
- if (ctl->flags & WMFW_CTL_FLAG_VOLATILE)
- kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE;
- } else {
- kcontrol->access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
- kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE;
- }
+ ctl->bytes_ext.max = ctl->len;
+ ctl->bytes_ext.get = wm_coeff_tlv_get;
+ ctl->bytes_ext.put = wm_coeff_tlv_put;
+
+ kcontrol->access = wmfw_convert_flags(ctl->flags, ctl->len);
ret = snd_soc_add_card_controls(dsp->card, kcontrol, 1);
if (ret < 0)
@@ -944,6 +1040,13 @@ static void wm_adsp_ctl_work(struct work_struct *work)
kfree(ctl_work);
}
+static void wm_adsp_free_ctl_blk(struct wm_coeff_ctl *ctl)
+{
+ kfree(ctl->cache);
+ kfree(ctl->name);
+ kfree(ctl);
+}
+
static int wm_adsp_create_control(struct wm_adsp *dsp,
const struct wm_adsp_alg_region *alg_region,
unsigned int offset, unsigned int len,
@@ -1032,11 +1135,6 @@ static int wm_adsp_create_control(struct wm_adsp *dsp,
ctl->flags = flags;
ctl->offset = offset;
- if (len > 512) {
- adsp_warn(dsp, "Truncating control %s from %d\n",
- ctl->name, len);
- len = 512;
- }
ctl->len = len;
ctl->cache = kzalloc(ctl->len, GFP_KERNEL);
if (!ctl->cache) {
@@ -1564,6 +1662,19 @@ static struct wm_adsp_alg_region *wm_adsp_create_region(struct wm_adsp *dsp,
return alg_region;
}
+static void wm_adsp_free_alg_regions(struct wm_adsp *dsp)
+{
+ struct wm_adsp_alg_region *alg_region;
+
+ while (!list_empty(&dsp->alg_regions)) {
+ alg_region = list_first_entry(&dsp->alg_regions,
+ struct wm_adsp_alg_region,
+ list);
+ list_del(&alg_region->list);
+ kfree(alg_region);
+ }
+}
+
static int wm_adsp1_setup_algs(struct wm_adsp *dsp)
{
struct wmfw_adsp1_id_hdr adsp1_id;
@@ -1994,7 +2105,6 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w,
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec);
struct wm_adsp *dsp = &dsps[w->shift];
- struct wm_adsp_alg_region *alg_region;
struct wm_coeff_ctl *ctl;
int ret;
unsigned int val;
@@ -2074,13 +2184,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w,
list_for_each_entry(ctl, &dsp->ctl_list, list)
ctl->enabled = 0;
- while (!list_empty(&dsp->alg_regions)) {
- alg_region = list_first_entry(&dsp->alg_regions,
- struct wm_adsp_alg_region,
- list);
- list_del(&alg_region->list);
- kfree(alg_region);
- }
+
+ wm_adsp_free_alg_regions(dsp);
break;
default:
@@ -2222,7 +2327,6 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec);
struct wm_adsp *dsp = &dsps[w->shift];
- struct wm_adsp_alg_region *alg_region;
struct wm_coeff_ctl *ctl;
int ret;
@@ -2240,9 +2344,13 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
if (ret != 0)
goto err;
+ mutex_lock(&dsp->pwr_lock);
+
if (wm_adsp_fw[dsp->fw].num_caps != 0)
ret = wm_adsp_buffer_init(dsp);
+ mutex_unlock(&dsp->pwr_lock);
+
break;
case SND_SOC_DAPM_PRE_PMD:
@@ -2269,13 +2377,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
list_for_each_entry(ctl, &dsp->ctl_list, list)
ctl->enabled = 0;
- while (!list_empty(&dsp->alg_regions)) {
- alg_region = list_first_entry(&dsp->alg_regions,
- struct wm_adsp_alg_region,
- list);
- list_del(&alg_region->list);
- kfree(alg_region);
- }
+ wm_adsp_free_alg_regions(dsp);
if (wm_adsp_fw[dsp->fw].num_caps != 0)
wm_adsp_buffer_free(dsp);
@@ -2340,6 +2442,54 @@ int wm_adsp2_init(struct wm_adsp *dsp)
}
EXPORT_SYMBOL_GPL(wm_adsp2_init);
+void wm_adsp2_remove(struct wm_adsp *dsp)
+{
+ struct wm_coeff_ctl *ctl;
+
+ while (!list_empty(&dsp->ctl_list)) {
+ ctl = list_first_entry(&dsp->ctl_list, struct wm_coeff_ctl,
+ list);
+ list_del(&ctl->list);
+ wm_adsp_free_ctl_blk(ctl);
+ }
+}
+EXPORT_SYMBOL_GPL(wm_adsp2_remove);
+
+static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr)
+{
+ return compr->buf != NULL;
+}
+
+static int wm_adsp_compr_attach(struct wm_adsp_compr *compr)
+{
+ /*
+ * Note this will be more complex once each DSP can support multiple
+ * streams
+ */
+ if (!compr->dsp->buffer)
+ return -EINVAL;
+
+ compr->buf = compr->dsp->buffer;
+ compr->buf->compr = compr;
+
+ return 0;
+}
+
+static void wm_adsp_compr_detach(struct wm_adsp_compr *compr)
+{
+ if (!compr)
+ return;
+
+ /* Wake the poll so it can see buffer is no longer attached */
+ if (compr->stream)
+ snd_compr_fragment_elapsed(compr->stream);
+
+ if (wm_adsp_compr_attached(compr)) {
+ compr->buf->compr = NULL;
+ compr->buf = NULL;
+ }
+}
+
int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream)
{
struct wm_adsp_compr *compr;
@@ -2393,6 +2543,7 @@ int wm_adsp_compr_free(struct snd_compr_stream *stream)
mutex_lock(&dsp->pwr_lock);
+ wm_adsp_compr_detach(compr);
dsp->compr = NULL;
kfree(compr->raw_buf);
@@ -2689,6 +2840,8 @@ err_buffer:
static int wm_adsp_buffer_free(struct wm_adsp *dsp)
{
if (dsp->buffer) {
+ wm_adsp_compr_detach(dsp->buffer->compr);
+
kfree(dsp->buffer->regions);
kfree(dsp->buffer);
@@ -2698,25 +2851,6 @@ static int wm_adsp_buffer_free(struct wm_adsp *dsp)
return 0;
}
-static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr)
-{
- return compr->buf != NULL;
-}
-
-static int wm_adsp_compr_attach(struct wm_adsp_compr *compr)
-{
- /*
- * Note this will be more complex once each DSP can support multiple
- * streams
- */
- if (!compr->dsp->buffer)
- return -EINVAL;
-
- compr->buf = compr->dsp->buffer;
-
- return 0;
-}
-
int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd)
{
struct wm_adsp_compr *compr = stream->runtime->private_data;
@@ -2805,21 +2939,41 @@ static int wm_adsp_buffer_update_avail(struct wm_adsp_compr_buf *buf)
avail += wm_adsp_buffer_size(buf);
adsp_dbg(buf->dsp, "readindex=0x%x, writeindex=0x%x, avail=%d\n",
- buf->read_index, write_index, avail);
+ buf->read_index, write_index, avail * WM_ADSP_DATA_WORD_SIZE);
buf->avail = avail;
return 0;
}
+static int wm_adsp_buffer_get_error(struct wm_adsp_compr_buf *buf)
+{
+ int ret;
+
+ ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(error), &buf->error);
+ if (ret < 0) {
+ adsp_err(buf->dsp, "Failed to check buffer error: %d\n", ret);
+ return ret;
+ }
+ if (buf->error != 0) {
+ adsp_err(buf->dsp, "Buffer error occurred: %d\n", buf->error);
+ return -EIO;
+ }
+
+ return 0;
+}
+
int wm_adsp_compr_handle_irq(struct wm_adsp *dsp)
{
- struct wm_adsp_compr_buf *buf = dsp->buffer;
- struct wm_adsp_compr *compr = dsp->compr;
+ struct wm_adsp_compr_buf *buf;
+ struct wm_adsp_compr *compr;
int ret = 0;
mutex_lock(&dsp->pwr_lock);
+ buf = dsp->buffer;
+ compr = dsp->compr;
+
if (!buf) {
ret = -ENODEV;
goto out;
@@ -2827,16 +2981,9 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp)
adsp_dbg(dsp, "Handling buffer IRQ\n");
- ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(error), &buf->error);
- if (ret < 0) {
- adsp_err(dsp, "Failed to check buffer error: %d\n", ret);
- goto out;
- }
- if (buf->error != 0) {
- adsp_err(dsp, "Buffer error occurred: %d\n", buf->error);
- ret = -EIO;
- goto out;
- }
+ ret = wm_adsp_buffer_get_error(buf);
+ if (ret < 0)
+ goto out_notify; /* Wake poll to report error */
ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(irq_count),
&buf->irq_count);
@@ -2851,6 +2998,7 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp)
goto out;
}
+out_notify:
if (compr && compr->stream)
snd_compr_fragment_elapsed(compr->stream);
@@ -2879,14 +3027,16 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream,
struct snd_compr_tstamp *tstamp)
{
struct wm_adsp_compr *compr = stream->runtime->private_data;
- struct wm_adsp_compr_buf *buf = compr->buf;
struct wm_adsp *dsp = compr->dsp;
+ struct wm_adsp_compr_buf *buf;
int ret = 0;
adsp_dbg(dsp, "Pointer request\n");
mutex_lock(&dsp->pwr_lock);
+ buf = compr->buf;
+
if (!compr->buf) {
ret = -ENXIO;
goto out;
@@ -2909,6 +3059,10 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream,
* DSP to inform us once a whole fragment is available.
*/
if (buf->avail < wm_adsp_compr_frag_words(compr)) {
+ ret = wm_adsp_buffer_get_error(buf);
+ if (ret < 0)
+ goto out;
+
ret = wm_adsp_buffer_reenable_irq(buf);
if (ret < 0) {
adsp_err(dsp,
diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h
index b61cb57e600f..feb61e2c4bb4 100644
--- a/sound/soc/codecs/wm_adsp.h
+++ b/sound/soc/codecs/wm_adsp.h
@@ -92,6 +92,7 @@ extern const struct snd_kcontrol_new wm_adsp_fw_controls[];
int wm_adsp1_init(struct wm_adsp *dsp);
int wm_adsp2_init(struct wm_adsp *dsp);
+void wm_adsp2_remove(struct wm_adsp *dsp);
int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec);
int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec);
int wm_adsp1_event(struct snd_soc_dapm_widget *w,
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 50ca291cc225..6b732d8e5896 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -16,7 +16,11 @@ config SND_EDMA_SOC
- DRA7xx family
config SND_DAVINCI_SOC_I2S
- tristate
+ tristate "DaVinci Multichannel Buffered Serial Port (McBSP) support"
+ depends on SND_EDMA_SOC
+ help
+ Say Y or M here if you want to have support for McBSP IP found in
+ Texas Instruments DaVinci DA850 SoCs.
config SND_DAVINCI_SOC_MCASP
tristate "Multichannel Audio Serial Port (McASP) support"
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index ec98548a5fc9..384961651904 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -4,9 +4,15 @@
* Author: Vladimir Barinov, <vbarinov@embeddedalley.com>
* Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
*
+ * DT support (c) 2016 Petr Kulhavy, Barix AG <petr@barix.com>
+ * based on davinci-mcasp.c DT support
+ *
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
+ *
+ * TODO:
+ * on DA850 implement HW FIFOs instead of DMA into DXR and DRR registers
*/
#include <linux/init.h>
@@ -650,13 +656,24 @@ static const struct snd_soc_component_driver davinci_i2s_component = {
static int davinci_i2s_probe(struct platform_device *pdev)
{
+ struct snd_dmaengine_dai_dma_data *dma_data;
struct davinci_mcbsp_dev *dev;
struct resource *mem, *res;
void __iomem *io_base;
int *dma;
int ret;
- mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
+ if (!mem) {
+ dev_warn(&pdev->dev,
+ "\"mpu\" mem resource not found, using index 0\n");
+ mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!mem) {
+ dev_err(&pdev->dev, "no mem resource?\n");
+ return -ENODEV;
+ }
+ }
+
io_base = devm_ioremap_resource(&pdev->dev, mem);
if (IS_ERR(io_base))
return PTR_ERR(io_base);
@@ -666,39 +683,43 @@ static int davinci_i2s_probe(struct platform_device *pdev)
if (!dev)
return -ENOMEM;
- dev->clk = clk_get(&pdev->dev, NULL);
- if (IS_ERR(dev->clk))
- return -ENODEV;
- clk_enable(dev->clk);
-
dev->base = io_base;
- dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr =
- (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG);
+ /* setup DMA, first TX, then RX */
+ dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
+ dma_data->addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG);
- dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr =
- (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG);
-
- /* first TX, then RX */
res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
- if (!res) {
- dev_err(&pdev->dev, "no DMA resource\n");
- ret = -ENXIO;
- goto err_release_clk;
+ if (res) {
+ dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK];
+ *dma = res->start;
+ dma_data->filter_data = dma;
+ } else if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) {
+ dma_data->filter_data = "tx";
+ } else {
+ dev_err(&pdev->dev, "Missing DMA tx resource\n");
+ return -ENODEV;
}
- dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK];
- *dma = res->start;
- dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = dma;
+
+ dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE];
+ dma_data->addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG);
res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
- if (!res) {
- dev_err(&pdev->dev, "no DMA resource\n");
- ret = -ENXIO;
- goto err_release_clk;
+ if (res) {
+ dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE];
+ *dma = res->start;
+ dma_data->filter_data = dma;
+ } else if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) {
+ dma_data->filter_data = "rx";
+ } else {
+ dev_err(&pdev->dev, "Missing DMA rx resource\n");
+ return -ENODEV;
}
- dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE];
- *dma = res->start;
- dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = dma;
+
+ dev->clk = clk_get(&pdev->dev, NULL);
+ if (IS_ERR(dev->clk))
+ return -ENODEV;
+ clk_enable(dev->clk);
dev->dev = &pdev->dev;
dev_set_drvdata(&pdev->dev, dev);
@@ -737,11 +758,18 @@ static int davinci_i2s_remove(struct platform_device *pdev)
return 0;
}
+static const struct of_device_id davinci_i2s_match[] = {
+ { .compatible = "ti,da850-mcbsp" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, davinci_i2s_match);
+
static struct platform_driver davinci_mcbsp_driver = {
.probe = davinci_i2s_probe,
.remove = davinci_i2s_remove,
.driver = {
.name = "davinci-mcbsp",
+ .of_match_table = of_match_ptr(davinci_i2s_match),
},
};
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index e1324989bd6b..0f66fda2c772 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -489,7 +489,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG,
- ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR);
+ ACLKX | AFSX | ACLKR | AHCLKR | AFSR);
mcasp->bclk_master = 0;
break;
default:
@@ -540,21 +540,19 @@ out:
return ret;
}
-static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
+static int __davinci_mcasp_set_clkdiv(struct davinci_mcasp *mcasp, int div_id,
int div, bool explicit)
{
- struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
-
pm_runtime_get_sync(mcasp->dev);
switch (div_id) {
- case 0: /* MCLK divider */
+ case MCASP_CLKDIV_AUXCLK: /* MCLK divider */
mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG,
AHCLKXDIV(div - 1), AHCLKXDIV_MASK);
mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG,
AHCLKRDIV(div - 1), AHCLKRDIV_MASK);
break;
- case 1: /* BCLK divider */
+ case MCASP_CLKDIV_BCLK: /* BCLK divider */
mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG,
ACLKXDIV(div - 1), ACLKXDIV_MASK);
mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG,
@@ -563,7 +561,8 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
mcasp->bclk_div = div;
break;
- case 2: /*
+ case MCASP_CLKDIV_BCLK_FS_RATIO:
+ /*
* BCLK/LRCLK ratio descries how many bit-clock cycles
* fit into one frame. The clock ratio is given for a
* full period of data (for I2S format both left and
@@ -591,7 +590,9 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
int div)
{
- return __davinci_mcasp_set_clkdiv(dai, div_id, div, 1);
+ struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
+
+ return __davinci_mcasp_set_clkdiv(mcasp, div_id, div, 1);
}
static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
@@ -999,27 +1000,53 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp,
}
static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp,
- unsigned int bclk_freq,
- int *error_ppm)
+ unsigned int bclk_freq, bool set)
{
- int div = mcasp->sysclk_freq / bclk_freq;
- int rem = mcasp->sysclk_freq % bclk_freq;
+ int error_ppm;
+ unsigned int sysclk_freq = mcasp->sysclk_freq;
+ u32 reg = mcasp_get_reg(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG);
+ int div = sysclk_freq / bclk_freq;
+ int rem = sysclk_freq % bclk_freq;
+ int aux_div = 1;
+
+ if (div > (ACLKXDIV_MASK + 1)) {
+ if (reg & AHCLKXE) {
+ aux_div = div / (ACLKXDIV_MASK + 1);
+ if (div % (ACLKXDIV_MASK + 1))
+ aux_div++;
+
+ sysclk_freq /= aux_div;
+ div = sysclk_freq / bclk_freq;
+ rem = sysclk_freq % bclk_freq;
+ } else if (set) {
+ dev_warn(mcasp->dev, "Too fast reference clock (%u)\n",
+ sysclk_freq);
+ }
+ }
if (rem != 0) {
if (div == 0 ||
- ((mcasp->sysclk_freq / div) - bclk_freq) >
- (bclk_freq - (mcasp->sysclk_freq / (div+1)))) {
+ ((sysclk_freq / div) - bclk_freq) >
+ (bclk_freq - (sysclk_freq / (div+1)))) {
div++;
rem = rem - bclk_freq;
}
}
- if (error_ppm)
- *error_ppm =
- (div*1000000 + (int)div64_long(1000000LL*rem,
- (int)bclk_freq))
- /div - 1000000;
+ error_ppm = (div*1000000 + (int)div64_long(1000000LL*rem,
+ (int)bclk_freq)) / div - 1000000;
+
+ if (set) {
+ if (error_ppm)
+ dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n",
+ error_ppm);
+
+ __davinci_mcasp_set_clkdiv(mcasp, MCASP_CLKDIV_BCLK, div, 0);
+ if (reg & AHCLKXE)
+ __davinci_mcasp_set_clkdiv(mcasp, MCASP_CLKDIV_AUXCLK,
+ aux_div, 0);
+ }
- return div;
+ return error_ppm;
}
static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
@@ -1044,18 +1071,11 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
int slots = mcasp->tdm_slots;
int rate = params_rate(params);
int sbits = params_width(params);
- int ppm, div;
if (mcasp->slot_width)
sbits = mcasp->slot_width;
- div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*slots,
- &ppm);
- if (ppm)
- dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n",
- ppm);
-
- __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0);
+ davinci_mcasp_calc_clk_div(mcasp, rate * sbits * slots, true);
}
ret = mcasp_common_hw_param(mcasp, substream->stream,
@@ -1166,7 +1186,8 @@ static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params,
davinci_mcasp_dai_rates[i];
int ppm;
- davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm);
+ ppm = davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq,
+ false);
if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) {
if (range.empty) {
range.min = davinci_mcasp_dai_rates[i];
@@ -1205,8 +1226,9 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params,
if (rd->mcasp->slot_width)
sbits = rd->mcasp->slot_width;
- davinci_mcasp_calc_clk_div(rd->mcasp, sbits*slots*rate,
- &ppm);
+ ppm = davinci_mcasp_calc_clk_div(rd->mcasp,
+ sbits * slots * rate,
+ false);
if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) {
snd_mask_set(&nfmt, i);
count++;
@@ -1230,11 +1252,15 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
int i, dir;
int tdm_slots = mcasp->tdm_slots;
- if (mcasp->tdm_mask[substream->stream])
- tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]);
+ /* Do not allow more then one stream per direction */
+ if (mcasp->substreams[substream->stream])
+ return -EBUSY;
mcasp->substreams[substream->stream] = substream;
+ if (mcasp->tdm_mask[substream->stream])
+ tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]);
+
if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
return 0;
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index a3be108a8c17..1e8787fb3fb7 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -306,4 +306,9 @@
#define NUMEVT(x) (((x) & 0xFF) << 8)
#define NUMDMA_MASK (0xFF)
+/* clock divider IDs */
+#define MCASP_CLKDIV_AUXCLK 0 /* HCLK divider from AUXCLK */
+#define MCASP_CLKDIV_BCLK 1 /* BCLK divider from HCLK */
+#define MCASP_CLKDIV_BCLK_FS_RATIO 2 /* to set BCLK FS ration */
+
#endif /* DAVINCI_MCASP_H */
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index bff258d7bcea..0db69b7e9617 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -100,6 +100,7 @@ struct dw_i2s_dev {
struct device *dev;
u32 ccr;
u32 xfer_resolution;
+ u32 fifo_th;
/* data related to DMA transfers b/w i2s and DMAC */
union dw_i2s_snd_dma_data play_dma_data;
@@ -147,17 +148,18 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream)
static void i2s_start(struct dw_i2s_dev *dev,
struct snd_pcm_substream *substream)
{
+ struct i2s_clk_config_data *config = &dev->config;
u32 i, irq;
i2s_write_reg(dev->i2s_base, IER, 1);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- for (i = 0; i < 4; i++) {
+ for (i = 0; i < (config->chan_nr / 2); i++) {
irq = i2s_read_reg(dev->i2s_base, IMR(i));
i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30);
}
i2s_write_reg(dev->i2s_base, ITER, 1);
} else {
- for (i = 0; i < 4; i++) {
+ for (i = 0; i < (config->chan_nr / 2); i++) {
irq = i2s_read_reg(dev->i2s_base, IMR(i));
i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03);
}
@@ -231,14 +233,16 @@ static void dw_i2s_config(struct dw_i2s_dev *dev, int stream)
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
i2s_write_reg(dev->i2s_base, TCR(ch_reg),
dev->xfer_resolution);
- i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02);
+ i2s_write_reg(dev->i2s_base, TFCR(ch_reg),
+ dev->fifo_th - 1);
irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg));
i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30);
i2s_write_reg(dev->i2s_base, TER(ch_reg), 1);
} else {
i2s_write_reg(dev->i2s_base, RCR(ch_reg),
dev->xfer_resolution);
- i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07);
+ i2s_write_reg(dev->i2s_base, RFCR(ch_reg),
+ dev->fifo_th - 1);
irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg));
i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03);
i2s_write_reg(dev->i2s_base, RER(ch_reg), 1);
@@ -498,6 +502,7 @@ static int dw_configure_dai(struct dw_i2s_dev *dev,
*/
u32 comp1 = i2s_read_reg(dev->i2s_base, dev->i2s_reg_comp1);
u32 comp2 = i2s_read_reg(dev->i2s_base, dev->i2s_reg_comp2);
+ u32 fifo_depth = 1 << (1 + COMP1_FIFO_DEPTH_GLOBAL(comp1));
u32 idx;
if (dev->capability & DWC_I2S_RECORD &&
@@ -536,6 +541,7 @@ static int dw_configure_dai(struct dw_i2s_dev *dev,
dev->capability |= DW_I2S_SLAVE;
}
+ dev->fifo_th = fifo_depth / 2;
return 0;
}
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 0754df771e3b..2147994ab46f 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -21,6 +21,8 @@
#include <sound/core.h>
#include <sound/dmaengine_pcm.h>
#include <sound/pcm_params.h>
+#include <linux/mfd/syscon.h>
+#include <linux/mfd/syscon/imx6q-iomuxc-gpr.h>
#include "fsl_sai.h"
#include "imx-pcm.h"
@@ -786,10 +788,12 @@ static int fsl_sai_probe(struct platform_device *pdev)
{
struct device_node *np = pdev->dev.of_node;
struct fsl_sai *sai;
+ struct regmap *gpr;
struct resource *res;
void __iomem *base;
char tmp[8];
int irq, ret, i;
+ int index;
sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL);
if (!sai)
@@ -797,7 +801,8 @@ static int fsl_sai_probe(struct platform_device *pdev)
sai->pdev = pdev;
- if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai"))
+ if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") ||
+ of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai"))
sai->sai_on_imx = true;
sai->is_lsb_first = of_property_read_bool(np, "lsb-first");
@@ -877,6 +882,22 @@ static int fsl_sai_probe(struct platform_device *pdev)
fsl_sai_dai.symmetric_samplebits = 0;
}
+ if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) &&
+ of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) {
+ gpr = syscon_regmap_lookup_by_compatible("fsl,imx6ul-iomuxc-gpr");
+ if (IS_ERR(gpr)) {
+ dev_err(&pdev->dev, "cannot find iomuxc registers\n");
+ return PTR_ERR(gpr);
+ }
+
+ index = of_alias_get_id(np, "sai");
+ if (index < 0)
+ return index;
+
+ regmap_update_bits(gpr, IOMUXC_GPR1, MCLK_DIR(index),
+ MCLK_DIR(index));
+ }
+
sai->dma_params_rx.addr = res->start + FSL_SAI_RDR;
sai->dma_params_tx.addr = res->start + FSL_SAI_TDR;
sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX;
@@ -898,6 +919,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
static const struct of_device_id fsl_sai_ids[] = {
{ .compatible = "fsl,vf610-sai", },
{ .compatible = "fsl,imx6sx-sai", },
+ { .compatible = "fsl,imx6ul-sai", },
{ /* sentinel */ }
};
MODULE_DEVICE_TABLE(of, fsl_sai_ids);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index ed8de1035cda..632ecc0e3956 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -137,6 +137,7 @@ static bool fsl_ssi_volatile_reg(struct device *dev, unsigned int reg)
case CCSR_SSI_SACDAT:
case CCSR_SSI_SATAG:
case CCSR_SSI_SACCST:
+ case CCSR_SSI_SOR:
return true;
default:
return false;
@@ -261,6 +262,7 @@ struct fsl_ssi_private {
struct fsl_ssi_dbg dbg_stats;
const struct fsl_ssi_soc_data *soc;
+ struct device *dev;
};
/*
@@ -400,6 +402,26 @@ static void fsl_ssi_rxtx_config(struct fsl_ssi_private *ssi_private,
}
/*
+ * Clear RX or TX FIFO to remove samples from the previous
+ * stream session which may be still present in the FIFO and
+ * may introduce bad samples and/or channel slipping.
+ *
+ * Note: The SOR is not documented in recent IMX datasheet, but
+ * is described in IMX51 reference manual at section 56.3.3.15.
+ */
+static void fsl_ssi_fifo_clear(struct fsl_ssi_private *ssi_private,
+ bool is_rx)
+{
+ if (is_rx) {
+ regmap_update_bits(ssi_private->regs, CCSR_SSI_SOR,
+ CCSR_SSI_SOR_RX_CLR, CCSR_SSI_SOR_RX_CLR);
+ } else {
+ regmap_update_bits(ssi_private->regs, CCSR_SSI_SOR,
+ CCSR_SSI_SOR_TX_CLR, CCSR_SSI_SOR_TX_CLR);
+ }
+}
+
+/*
* Calculate the bits that have to be disabled for the current stream that is
* getting disabled. This keeps the bits enabled that are necessary for the
* second stream to work if 'stream_active' is true.
@@ -474,9 +496,11 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable,
* (online configuration)
*/
if (enable) {
- regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier);
+ fsl_ssi_fifo_clear(ssi_private, vals->scr & CCSR_SSI_SCR_RE);
+
regmap_update_bits(regs, CCSR_SSI_SRCR, vals->srcr, vals->srcr);
regmap_update_bits(regs, CCSR_SSI_STCR, vals->stcr, vals->stcr);
+ regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier);
} else {
u32 sier;
u32 srcr;
@@ -506,8 +530,40 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable,
config_done:
/* Enabling of subunits is done after configuration */
- if (enable)
+ if (enable) {
+ if (ssi_private->use_dma && (vals->scr & CCSR_SSI_SCR_TE)) {
+ /*
+ * Be sure the Tx FIFO is filled when TE is set.
+ * Otherwise, there are some chances to start the
+ * playback with some void samples inserted first,
+ * generating a channel slip.
+ *
+ * First, SSIEN must be set, to let the FIFO be filled.
+ *
+ * Notes:
+ * - Limit this fix to the DMA case until FIQ cases can
+ * be tested.
+ * - Limit the length of the busy loop to not lock the
+ * system too long, even if 1-2 loops are sufficient
+ * in general.
+ */
+ int i;
+ int max_loop = 100;
+ regmap_update_bits(regs, CCSR_SSI_SCR,
+ CCSR_SSI_SCR_SSIEN, CCSR_SSI_SCR_SSIEN);
+ for (i = 0; i < max_loop; i++) {
+ u32 sfcsr;
+ regmap_read(regs, CCSR_SSI_SFCSR, &sfcsr);
+ if (CCSR_SSI_SFCSR_TFCNT0(sfcsr))
+ break;
+ }
+ if (i == max_loop) {
+ dev_err(ssi_private->dev,
+ "Timeout waiting TX FIFO filling\n");
+ }
+ }
regmap_update_bits(regs, CCSR_SSI_SCR, vals->scr, vals->scr);
+ }
}
@@ -670,6 +726,15 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
if (IS_ERR(ssi_private->baudclk))
return -EINVAL;
+ /*
+ * Hardware limitation: The bclk rate must be
+ * never greater than 1/5 IPG clock rate
+ */
+ if (freq * 5 > clk_get_rate(ssi_private->clk)) {
+ dev_err(cpu_dai->dev, "bitclk > ipgclk/5\n");
+ return -EINVAL;
+ }
+
baudclk_is_used = ssi_private->baudclk_streams & ~(BIT(substream->stream));
/* It should be already enough to divide clock by setting pm alone */
@@ -686,13 +751,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
else
clkrate = clk_round_rate(ssi_private->baudclk, tmprate);
- /*
- * Hardware limitation: The bclk rate must be
- * never greater than 1/5 IPG clock rate
- */
- if (clkrate * 5 > clk_get_rate(ssi_private->clk))
- continue;
-
clkrate /= factor;
afreq = clkrate / (i + 1);
@@ -1158,14 +1216,14 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = {
.playback = {
.stream_name = "CPU-Playback",
.channels_min = 1,
- .channels_max = 2,
+ .channels_max = 32,
.rates = FSLSSI_I2S_RATES,
.formats = FSLSSI_I2S_FORMATS,
},
.capture = {
.stream_name = "CPU-Capture",
.channels_min = 1,
- .channels_max = 2,
+ .channels_max = 32,
.rates = FSLSSI_I2S_RATES,
.formats = FSLSSI_I2S_FORMATS,
},
@@ -1402,6 +1460,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
}
ssi_private->soc = of_id->data;
+ ssi_private->dev = &pdev->dev;
sprop = of_get_property(np, "fsl,mode", NULL);
if (sprop) {
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index e63cd5ecfd8f..dac6688540dc 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -220,7 +220,7 @@ static int snd_imx_pcm_mmap(struct snd_pcm_substream *substream,
ret = dma_mmap_wc(substream->pcm->card->dev, vma, runtime->dma_area,
runtime->dma_addr, runtime->dma_bytes);
- pr_debug("%s: ret: %d %p %pad 0x%08x\n", __func__, ret,
+ pr_debug("%s: ret: %d %p %pad 0x%08zx\n", __func__, ret,
runtime->dma_area,
&runtime->dma_addr,
runtime->dma_bytes);
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 1120f4f4d011..91c15abb625e 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -58,6 +58,21 @@ config SND_SOC_INTEL_HASWELL_MACH
Say Y if you have such a device
If unsure select "N".
+config SND_SOC_INTEL_BXT_RT298_MACH
+ tristate "ASoC Audio driver for Broxton with RT298 I2S mode"
+ depends on X86 && ACPI && I2C
+ select SND_SOC_INTEL_SST
+ select SND_SOC_INTEL_SKYLAKE
+ select SND_SOC_RT298
+ select SND_SOC_DMIC
+ select SND_SOC_HDAC_HDMI
+ select SND_HDA_DSP_LOADER
+ help
+ This adds support for ASoC machine driver for Broxton platforms
+ with RT286 I2S audio codec.
+ Say Y if you have such a device
+ If unsure select "N".
+
config SND_SOC_INTEL_BYT_RT5640_MACH
tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec"
depends on X86_INTEL_LPSS && I2C
@@ -162,6 +177,7 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH
config SND_SOC_INTEL_SKYLAKE
tristate
select SND_HDA_EXT_CORE
+ select SND_HDA_DSP_LOADER
select SND_SOC_TOPOLOGY
select SND_SOC_INTEL_SST
diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c
index b97e6adcf1b2..98720a93de8a 100644
--- a/sound/soc/intel/atom/sst-atom-controls.c
+++ b/sound/soc/intel/atom/sst-atom-controls.c
@@ -195,7 +195,7 @@ static int sst_check_and_send_slot_map(struct sst_data *drv, struct snd_kcontrol
if (e->w && e->w->power)
ret = sst_send_slot_map(drv);
- else
+ else if (!e->w)
dev_err(&drv->pdev->dev, "Slot control: %s doesn't have DAPM widget!!!\n",
kcontrol->id.name);
return ret;
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index 3310c0f9c356..a8506774f510 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -2,6 +2,7 @@ snd-soc-sst-haswell-objs := haswell.o
snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
snd-soc-sst-broadwell-objs := broadwell.o
+snd-soc-sst-bxt-rt298-objs := bxt_rt298.o
snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o
snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o
snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
@@ -14,6 +15,7 @@ snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o
obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
+obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o
obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o
obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 3f8a1e10bed0..7486a0022fde 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -201,7 +201,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
{
/* SSP0 - Codec */
.name = "Codec",
- .be_id = 0,
+ .id = 0,
.cpu_dai_name = "snd-soc-dummy-dai",
.platform_name = "snd-soc-dummy",
.no_pcm = 1,
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
new file mode 100644
index 000000000000..f4787515c0ed
--- /dev/null
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -0,0 +1,353 @@
+/*
+ * Intel Broxton-P I2S Machine Driver
+ *
+ * Copyright (C) 2014-2016, Intel Corporation. All rights reserved.
+ *
+ * Modified from:
+ * Intel Skylake I2S Machine driver
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License version
+ * 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/pcm_params.h>
+#include "../../codecs/hdac_hdmi.h"
+#include "../../codecs/rt298.h"
+
+static struct snd_soc_jack broxton_headset;
+/* Headset jack detection DAPM pins */
+
+enum {
+ BXT_DPCM_AUDIO_PB = 0,
+ BXT_DPCM_AUDIO_CP,
+ BXT_DPCM_AUDIO_REF_CP,
+ BXT_DPCM_AUDIO_HDMI1_PB,
+ BXT_DPCM_AUDIO_HDMI2_PB,
+ BXT_DPCM_AUDIO_HDMI3_PB,
+};
+
+static struct snd_soc_jack_pin broxton_headset_pins[] = {
+ {
+ .pin = "Mic Jack",
+ .mask = SND_JACK_MICROPHONE,
+ },
+ {
+ .pin = "Headphone Jack",
+ .mask = SND_JACK_HEADPHONE,
+ },
+};
+
+static const struct snd_kcontrol_new broxton_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Speaker"),
+ SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+ SOC_DAPM_PIN_SWITCH("Mic Jack"),
+};
+
+static const struct snd_soc_dapm_widget broxton_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_MIC("DMIC2", NULL),
+ SND_SOC_DAPM_MIC("SoC DMIC", NULL),
+ SND_SOC_DAPM_SPK("HDMI1", NULL),
+ SND_SOC_DAPM_SPK("HDMI2", NULL),
+ SND_SOC_DAPM_SPK("HDMI3", NULL),
+};
+
+static const struct snd_soc_dapm_route broxton_rt298_map[] = {
+ /* speaker */
+ {"Speaker", NULL, "SPOR"},
+ {"Speaker", NULL, "SPOL"},
+
+ /* HP jack connectors - unknown if we have jack detect */
+ {"Headphone Jack", NULL, "HPO Pin"},
+
+ /* other jacks */
+ {"MIC1", NULL, "Mic Jack"},
+
+ /* digital mics */
+ {"DMIC1 Pin", NULL, "DMIC2"},
+ {"DMic", NULL, "SoC DMIC"},
+
+ {"HDMI1", NULL, "hif5 Output"},
+ {"HDMI2", NULL, "hif6 Output"},
+ {"HDMI3", NULL, "hif7 Output"},
+
+ /* CODEC BE connections */
+ { "AIF1 Playback", NULL, "ssp5 Tx"},
+ { "ssp5 Tx", NULL, "codec0_out"},
+
+ { "codec0_in", NULL, "ssp5 Rx" },
+ { "ssp5 Rx", NULL, "AIF1 Capture" },
+
+ { "dmic01_hifi", NULL, "DMIC01 Rx" },
+ { "DMIC01 Rx", NULL, "Capture" },
+
+ { "hifi3", NULL, "iDisp3 Tx"},
+ { "iDisp3 Tx", NULL, "iDisp3_out"},
+ { "hifi2", NULL, "iDisp2 Tx"},
+ { "iDisp2 Tx", NULL, "iDisp2_out"},
+ { "hifi1", NULL, "iDisp1 Tx"},
+ { "iDisp1 Tx", NULL, "iDisp1_out"},
+
+};
+
+static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ int ret = 0;
+
+ ret = snd_soc_card_jack_new(rtd->card, "Headset",
+ SND_JACK_HEADSET | SND_JACK_BTN_0,
+ &broxton_headset,
+ broxton_headset_pins, ARRAY_SIZE(broxton_headset_pins));
+
+ if (ret)
+ return ret;
+
+ rt298_mic_detect(codec, &broxton_headset);
+ return 0;
+}
+
+static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dai *dai = rtd->codec_dai;
+
+ return hdac_hdmi_jack_init(dai, BXT_DPCM_AUDIO_HDMI1_PB + dai->id);
+}
+
+static int broxton_ssp5_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+ /* The ADSP will covert the FE rate to 48k, stereo */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ /* set SSP5 to 24 bit */
+ snd_mask_none(fmt);
+ snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+
+ return 0;
+}
+
+static int broxton_rt298_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT298_SCLK_S_PLL,
+ 19200000, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+ return ret;
+ }
+
+ return ret;
+}
+
+static struct snd_soc_ops broxton_rt298_ops = {
+ .hw_params = broxton_rt298_hw_params,
+};
+
+/* broxton digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link broxton_rt298_dais[] = {
+ /* Front End DAI links */
+ [BXT_DPCM_AUDIO_PB]
+ {
+ .name = "Bxt Audio Port",
+ .stream_name = "Audio",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "0000:00:0e.0",
+ .nonatomic = 1,
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ [BXT_DPCM_AUDIO_CP]
+ {
+ .name = "Bxt Audio Capture Port",
+ .stream_name = "Audio Record",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "0000:00:0e.0",
+ .nonatomic = 1,
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_capture = 1,
+ },
+ [BXT_DPCM_AUDIO_REF_CP]
+ {
+ .name = "Bxt Audio Reference cap",
+ .stream_name = "refcap",
+ .cpu_dai_name = "Reference Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .init = NULL,
+ .dpcm_capture = 1,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [BXT_DPCM_AUDIO_HDMI1_PB]
+ {
+ .name = "Bxt HDMI Port1",
+ .stream_name = "Hdmi1",
+ .cpu_dai_name = "HDMI1 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .dpcm_playback = 1,
+ .init = NULL,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [BXT_DPCM_AUDIO_HDMI2_PB]
+ {
+ .name = "Bxt HDMI Port2",
+ .stream_name = "Hdmi2",
+ .cpu_dai_name = "HDMI2 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .dpcm_playback = 1,
+ .init = NULL,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [BXT_DPCM_AUDIO_HDMI3_PB]
+ {
+ .name = "Bxt HDMI Port3",
+ .stream_name = "Hdmi3",
+ .cpu_dai_name = "HDMI3 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .dpcm_playback = 1,
+ .init = NULL,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ /* Back End DAI links */
+ {
+ /* SSP5 - Codec */
+ .name = "SSP5-Codec",
+ .id = 0,
+ .cpu_dai_name = "SSP5 Pin",
+ .platform_name = "0000:00:0e.0",
+ .no_pcm = 1,
+ .codec_name = "i2c-INT343A:00",
+ .codec_dai_name = "rt298-aif1",
+ .init = broxton_rt298_codec_init,
+ .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ignore_pmdown_time = 1,
+ .be_hw_params_fixup = broxton_ssp5_fixup,
+ .ops = &broxton_rt298_ops,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ },
+ {
+ .name = "dmic01",
+ .id = 1,
+ .cpu_dai_name = "DMIC01 Pin",
+ .codec_name = "dmic-codec",
+ .codec_dai_name = "dmic-hifi",
+ .platform_name = "0000:00:0e.0",
+ .ignore_suspend = 1,
+ .dpcm_capture = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp1",
+ .id = 3,
+ .cpu_dai_name = "iDisp1 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi1",
+ .platform_name = "0000:00:0e.0",
+ .init = broxton_hdmi_init,
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp2",
+ .id = 4,
+ .cpu_dai_name = "iDisp2 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi2",
+ .platform_name = "0000:00:0e.0",
+ .init = broxton_hdmi_init,
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp3",
+ .id = 5,
+ .cpu_dai_name = "iDisp3 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi3",
+ .platform_name = "0000:00:0e.0",
+ .init = broxton_hdmi_init,
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+};
+
+/* broxton audio machine driver for SPT + RT298S */
+static struct snd_soc_card broxton_rt298 = {
+ .name = "broxton-rt298",
+ .owner = THIS_MODULE,
+ .dai_link = broxton_rt298_dais,
+ .num_links = ARRAY_SIZE(broxton_rt298_dais),
+ .controls = broxton_controls,
+ .num_controls = ARRAY_SIZE(broxton_controls),
+ .dapm_widgets = broxton_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(broxton_widgets),
+ .dapm_routes = broxton_rt298_map,
+ .num_dapm_routes = ARRAY_SIZE(broxton_rt298_map),
+ .fully_routed = true,
+};
+
+static int broxton_audio_probe(struct platform_device *pdev)
+{
+ broxton_rt298.dev = &pdev->dev;
+
+ return devm_snd_soc_register_card(&pdev->dev, &broxton_rt298);
+}
+
+static struct platform_driver broxton_audio = {
+ .probe = broxton_audio_probe,
+ .driver = {
+ .name = "bxt_alc298s_i2s",
+ },
+};
+module_platform_driver(broxton_audio)
+
+/* Module information */
+MODULE_AUTHOR("Ramesh Babu <Ramesh.Babu@intel.com>");
+MODULE_AUTHOR("Senthilnathan Veppur <senthilnathanx.veppur@intel.com>");
+MODULE_DESCRIPTION("Intel SST Audio for Broxton");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:bxt_alc298s_i2s");
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 032a2e753f0b..88efb62439ba 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -304,7 +304,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = {
/* back ends */
{
.name = "SSP2-Codec",
- .be_id = 1,
+ .id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 1c95ccc886c4..35f591eab3c9 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -267,7 +267,7 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = {
/* back ends */
{
.name = "SSP2-Codec",
- .be_id = 1,
+ .id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
index ac60b04540fd..cdcced9f32b6 100644
--- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c
+++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
@@ -255,7 +255,7 @@ static struct snd_soc_dai_link cht_dailink[] = {
/* back ends */
{
.name = "SSP2-Codec",
- .be_id = 1,
+ .id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c
index 3f2c1ea3a83e..d7ef292c402d 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5645.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5645.c
@@ -295,7 +295,7 @@ static struct snd_soc_dai_link cht_dailink[] = {
/* back ends */
{
.name = "SSP2-Codec",
- .be_id = 1,
+ .id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c
index 2e5347f8f96c..df9d254baa18 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5672.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5672.c
@@ -273,7 +273,7 @@ static struct snd_soc_dai_link cht_dailink[] = {
{
/* SSP2 - Codec */
.name = "SSP2-Codec",
- .be_id = 1,
+ .id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
index 22558572cb9c..863f1d5e2a2c 100644
--- a/sound/soc/intel/boards/haswell.c
+++ b/sound/soc/intel/boards/haswell.c
@@ -156,7 +156,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
{
/* SSP0 - Codec */
.name = "Codec",
- .be_id = 0,
+ .id = 0,
.cpu_dai_name = "snd-soc-dummy-dai",
.platform_name = "snd-soc-dummy",
.no_pcm = 1,
diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
index 72176b79a18d..d2808652b974 100644
--- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c
+++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
@@ -30,6 +30,16 @@
static struct snd_soc_jack skylake_headset;
static struct snd_soc_card skylake_audio_card;
+struct skl_hdmi_pcm {
+ struct list_head head;
+ struct snd_soc_dai *codec_dai;
+ int device;
+};
+
+struct skl_nau8825_private {
+ struct list_head hdmi_pcm_list;
+};
+
enum {
SKL_DPCM_AUDIO_PB = 0,
SKL_DPCM_AUDIO_CP,
@@ -192,23 +202,56 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai = rtd->codec_dai;
+ struct skl_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
+
+ pcm->device = SKL_DPCM_AUDIO_HDMI1_PB;
+ pcm->codec_dai = dai;
- return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB);
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
}
static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai = rtd->codec_dai;
+ struct skl_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
+
+ pcm->device = SKL_DPCM_AUDIO_HDMI2_PB;
+ pcm->codec_dai = dai;
- return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI2_PB);
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
}
static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai = rtd->codec_dai;
+ struct skl_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
- return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI3_PB);
+ pcm->device = SKL_DPCM_AUDIO_HDMI3_PB;
+ pcm->codec_dai = dai;
+
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
}
static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd)
@@ -391,7 +434,6 @@ static struct snd_soc_dai_link skylake_dais[] = {
.platform_name = "0000:00:1f.3",
.init = NULL,
.dpcm_capture = 1,
- .ignore_suspend = 1,
.nonatomic = 1,
.dynamic = 1,
.ops = &skylaye_refcap_ops,
@@ -456,7 +498,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
{
/* SSP0 - Codec */
.name = "SSP0-Codec",
- .be_id = 0,
+ .id = 0,
.cpu_dai_name = "SSP0 Pin",
.platform_name = "0000:00:1f.3",
.no_pcm = 1,
@@ -472,7 +514,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
{
/* SSP1 - Codec */
.name = "SSP1-Codec",
- .be_id = 1,
+ .id = 1,
.cpu_dai_name = "SSP1 Pin",
.platform_name = "0000:00:1f.3",
.no_pcm = 1,
@@ -489,7 +531,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
{
.name = "dmic01",
- .be_id = 2,
+ .id = 2,
.cpu_dai_name = "DMIC01 Pin",
.codec_name = "dmic-codec",
.codec_dai_name = "dmic-hifi",
@@ -501,7 +543,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
{
.name = "iDisp1",
- .be_id = 3,
+ .id = 3,
.cpu_dai_name = "iDisp1 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi1",
@@ -512,7 +554,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
{
.name = "iDisp2",
- .be_id = 4,
+ .id = 4,
.cpu_dai_name = "iDisp2 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi2",
@@ -523,7 +565,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
{
.name = "iDisp3",
- .be_id = 5,
+ .id = 5,
.cpu_dai_name = "iDisp3 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi3",
@@ -534,6 +576,21 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
};
+static int skylake_card_late_probe(struct snd_soc_card *card)
+{
+ struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(card);
+ struct skl_hdmi_pcm *pcm;
+ int err;
+
+ list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+ err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
/* skylake audio machine driver for SPT + NAU88L25 */
static struct snd_soc_card skylake_audio_card = {
.name = "sklnau8825max",
@@ -547,11 +604,21 @@ static struct snd_soc_card skylake_audio_card = {
.dapm_routes = skylake_map,
.num_dapm_routes = ARRAY_SIZE(skylake_map),
.fully_routed = true,
+ .late_probe = skylake_card_late_probe,
};
static int skylake_audio_probe(struct platform_device *pdev)
{
+ struct skl_nau8825_private *ctx;
+
+ ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+ if (!ctx)
+ return -ENOMEM;
+
+ INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
skylake_audio_card.dev = &pdev->dev;
+ snd_soc_card_set_drvdata(&skylake_audio_card, ctx);
return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card);
}
diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
index 5f1ca99ae9b0..e19aa99c4f72 100644
--- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
+++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
@@ -34,6 +34,15 @@
static struct snd_soc_jack skylake_headset;
static struct snd_soc_card skylake_audio_card;
+struct skl_hdmi_pcm {
+ struct list_head head;
+ struct snd_soc_dai *codec_dai;
+ int device;
+};
+
+struct skl_nau88125_private {
+ struct list_head hdmi_pcm_list;
+};
enum {
SKL_DPCM_AUDIO_PB = 0,
SKL_DPCM_AUDIO_CP,
@@ -222,24 +231,57 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai = rtd->codec_dai;
+ struct skl_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
+
+ pcm->device = SKL_DPCM_AUDIO_HDMI1_PB;
+ pcm->codec_dai = dai;
+
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
- return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB);
+ return 0;
}
static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai = rtd->codec_dai;
+ struct skl_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
+
+ pcm->device = SKL_DPCM_AUDIO_HDMI2_PB;
+ pcm->codec_dai = dai;
- return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI2_PB);
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
}
static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai = rtd->codec_dai;
+ struct skl_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
- return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI3_PB);
+ pcm->device = SKL_DPCM_AUDIO_HDMI3_PB;
+ pcm->codec_dai = dai;
+
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
}
static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd)
@@ -440,7 +482,6 @@ static struct snd_soc_dai_link skylake_dais[] = {
.platform_name = "0000:00:1f.3",
.init = NULL,
.dpcm_capture = 1,
- .ignore_suspend = 1,
.nonatomic = 1,
.dynamic = 1,
.ops = &skylaye_refcap_ops,
@@ -505,7 +546,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
{
/* SSP0 - Codec */
.name = "SSP0-Codec",
- .be_id = 0,
+ .id = 0,
.cpu_dai_name = "SSP0 Pin",
.platform_name = "0000:00:1f.3",
.no_pcm = 1,
@@ -523,7 +564,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
{
/* SSP1 - Codec */
.name = "SSP1-Codec",
- .be_id = 1,
+ .id = 1,
.cpu_dai_name = "SSP1 Pin",
.platform_name = "0000:00:1f.3",
.no_pcm = 1,
@@ -540,7 +581,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
{
.name = "dmic01",
- .be_id = 2,
+ .id = 2,
.cpu_dai_name = "DMIC01 Pin",
.codec_name = "dmic-codec",
.codec_dai_name = "dmic-hifi",
@@ -552,7 +593,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
{
.name = "iDisp1",
- .be_id = 3,
+ .id = 3,
.cpu_dai_name = "iDisp1 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi1",
@@ -563,7 +604,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
{
.name = "iDisp2",
- .be_id = 4,
+ .id = 4,
.cpu_dai_name = "iDisp2 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi2",
@@ -574,7 +615,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
{
.name = "iDisp3",
- .be_id = 5,
+ .id = 5,
.cpu_dai_name = "iDisp3 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi3",
@@ -585,6 +626,21 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
};
+static int skylake_card_late_probe(struct snd_soc_card *card)
+{
+ struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(card);
+ struct skl_hdmi_pcm *pcm;
+ int err;
+
+ list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+ err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
/* skylake audio machine driver for SPT + NAU88L25 */
static struct snd_soc_card skylake_audio_card = {
.name = "sklnau8825adi",
@@ -600,11 +656,21 @@ static struct snd_soc_card skylake_audio_card = {
.codec_conf = ssm4567_codec_conf,
.num_configs = ARRAY_SIZE(ssm4567_codec_conf),
.fully_routed = true,
+ .late_probe = skylake_card_late_probe,
};
static int skylake_audio_probe(struct platform_device *pdev)
{
+ struct skl_nau88125_private *ctx;
+
+ ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+ if (!ctx)
+ return -ENOMEM;
+
+ INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
skylake_audio_card.dev = &pdev->dev;
+ snd_soc_card_set_drvdata(&skylake_audio_card, ctx);
return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card);
}
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
index 2016397a8e75..426b48233fdb 100644
--- a/sound/soc/intel/boards/skl_rt286.c
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -30,6 +30,16 @@
static struct snd_soc_jack skylake_headset;
+struct skl_hdmi_pcm {
+ struct list_head head;
+ struct snd_soc_dai *codec_dai;
+ int device;
+};
+
+struct skl_rt286_private {
+ struct list_head hdmi_pcm_list;
+};
+
enum {
SKL_DPCM_AUDIO_PB = 0,
SKL_DPCM_AUDIO_CP,
@@ -142,9 +152,20 @@ static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_hdmi_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai = rtd->codec_dai;
+ struct skl_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
- return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB + dai->id);
+ pcm->device = SKL_DPCM_AUDIO_HDMI1_PB + dai->id;
+ pcm->codec_dai = dai;
+
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
}
static unsigned int rates[] = {
@@ -317,7 +338,6 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
.platform_name = "0000:00:1f.3",
.init = NULL,
.dpcm_capture = 1,
- .ignore_suspend = 1,
.nonatomic = 1,
.dynamic = 1,
},
@@ -375,7 +395,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
{
/* SSP0 - Codec */
.name = "SSP0-Codec",
- .be_id = 0,
+ .id = 0,
.cpu_dai_name = "SSP0 Pin",
.platform_name = "0000:00:1f.3",
.no_pcm = 1,
@@ -393,7 +413,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
},
{
.name = "dmic01",
- .be_id = 1,
+ .id = 1,
.cpu_dai_name = "DMIC01 Pin",
.codec_name = "dmic-codec",
.codec_dai_name = "dmic-hifi",
@@ -405,7 +425,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
},
{
.name = "iDisp1",
- .be_id = 2,
+ .id = 2,
.cpu_dai_name = "iDisp1 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi1",
@@ -416,7 +436,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
},
{
.name = "iDisp2",
- .be_id = 3,
+ .id = 3,
.cpu_dai_name = "iDisp2 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi2",
@@ -427,7 +447,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
},
{
.name = "iDisp3",
- .be_id = 4,
+ .id = 4,
.cpu_dai_name = "iDisp3 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi3",
@@ -438,6 +458,21 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
},
};
+static int skylake_card_late_probe(struct snd_soc_card *card)
+{
+ struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(card);
+ struct skl_hdmi_pcm *pcm;
+ int err;
+
+ list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+ err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
/* skylake audio machine driver for SPT + RT286S */
static struct snd_soc_card skylake_rt286 = {
.name = "skylake-rt286",
@@ -451,11 +486,21 @@ static struct snd_soc_card skylake_rt286 = {
.dapm_routes = skylake_rt286_map,
.num_dapm_routes = ARRAY_SIZE(skylake_rt286_map),
.fully_routed = true,
+ .late_probe = skylake_card_late_probe,
};
static int skylake_audio_probe(struct platform_device *pdev)
{
+ struct skl_rt286_private *ctx;
+
+ ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+ if (!ctx)
+ return -ENOMEM;
+
+ INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
skylake_rt286.dev = &pdev->dev;
+ snd_soc_card_set_drvdata(&skylake_rt286, ctx);
return devm_snd_soc_register_card(&pdev->dev, &skylake_rt286);
}
diff --git a/sound/soc/intel/common/sst-acpi.h b/sound/soc/intel/common/sst-acpi.h
index 4dcfb7e5ed70..8398cb227ba9 100644
--- a/sound/soc/intel/common/sst-acpi.h
+++ b/sound/soc/intel/common/sst-acpi.h
@@ -12,10 +12,19 @@
*
*/
+#include <linux/kconfig.h>
+#include <linux/stddef.h>
#include <linux/acpi.h>
/* translation fron HID to I2C name, needed for DAI codec_name */
+#if IS_ENABLED(CONFIG_ACPI)
const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]);
+#else
+inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN])
+{
+ return NULL;
+}
+#endif
/* acpi match */
struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines);
diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c
index 1aa819c7e09b..994256b39b9c 100644
--- a/sound/soc/intel/haswell/sst-haswell-pcm.c
+++ b/sound/soc/intel/haswell/sst-haswell-pcm.c
@@ -445,7 +445,7 @@ static int create_adsp_page_table(struct snd_pcm_substream *substream,
pages = snd_sgbuf_aligned_pages(size);
- dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n",
+ dev_dbg(rtd->dev, "generating page table for %p size 0x%zx pages %d\n",
dma_area, size, pages);
for (i = 0; i < pages; i++) {
diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile
index 914b6dab9bea..c28f5d0e1d99 100644
--- a/sound/soc/intel/skylake/Makefile
+++ b/sound/soc/intel/skylake/Makefile
@@ -5,6 +5,6 @@ obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o
# Skylake IPC Support
snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o skl-sst-cldma.o \
- skl-sst.o
+ skl-sst.o bxt-sst.o
obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o
diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c
new file mode 100644
index 000000000000..965ce40ce752
--- /dev/null
+++ b/sound/soc/intel/skylake/bxt-sst.c
@@ -0,0 +1,328 @@
+/*
+ * bxt-sst.c - DSP library functions for BXT platform
+ *
+ * Copyright (C) 2015-16 Intel Corp
+ * Author:Rafal Redzimski <rafal.f.redzimski@intel.com>
+ * Jeeja KP <jeeja.kp@intel.com>
+ *
+ * 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; version 2 of the License.
+ *
+ * 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.
+ */
+
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/firmware.h>
+#include <linux/device.h>
+
+#include "../common/sst-dsp.h"
+#include "../common/sst-dsp-priv.h"
+#include "skl-sst-ipc.h"
+
+#define BXT_BASEFW_TIMEOUT 3000
+#define BXT_INIT_TIMEOUT 500
+#define BXT_IPC_PURGE_FW 0x01004000
+
+#define BXT_ROM_INIT 0x5
+#define BXT_ADSP_SRAM0_BASE 0x80000
+
+/* Firmware status window */
+#define BXT_ADSP_FW_STATUS BXT_ADSP_SRAM0_BASE
+#define BXT_ADSP_ERROR_CODE (BXT_ADSP_FW_STATUS + 0x4)
+
+#define BXT_ADSP_SRAM1_BASE 0xA0000
+
+static unsigned int bxt_get_errorcode(struct sst_dsp *ctx)
+{
+ return sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE);
+}
+
+static int sst_bxt_prepare_fw(struct sst_dsp *ctx,
+ const void *fwdata, u32 fwsize)
+{
+ int stream_tag, ret, i;
+ u32 reg;
+
+ stream_tag = ctx->dsp_ops.prepare(ctx->dev, 0x40, fwsize, &ctx->dmab);
+ if (stream_tag < 0) {
+ dev_err(ctx->dev, "Failed to prepare DMA FW loading err: %x\n",
+ stream_tag);
+ return stream_tag;
+ }
+
+ ctx->dsp_ops.stream_tag = stream_tag;
+ memcpy(ctx->dmab.area, fwdata, fwsize);
+
+ /* Purge FW request */
+ sst_dsp_shim_write(ctx, SKL_ADSP_REG_HIPCI, SKL_ADSP_REG_HIPCI_BUSY |
+ BXT_IPC_PURGE_FW | (stream_tag - 1));
+
+ ret = skl_dsp_enable_core(ctx);
+ if (ret < 0) {
+ dev_err(ctx->dev, "Boot dsp core failed ret: %d\n", ret);
+ ret = -EIO;
+ goto base_fw_load_failed;
+ }
+
+ for (i = BXT_INIT_TIMEOUT; i > 0; --i) {
+ reg = sst_dsp_shim_read(ctx, SKL_ADSP_REG_HIPCIE);
+
+ if (reg & SKL_ADSP_REG_HIPCIE_DONE) {
+ sst_dsp_shim_update_bits_forced(ctx,
+ SKL_ADSP_REG_HIPCIE,
+ SKL_ADSP_REG_HIPCIE_DONE,
+ SKL_ADSP_REG_HIPCIE_DONE);
+ break;
+ }
+ mdelay(1);
+ }
+ if (!i) {
+ dev_info(ctx->dev, "Waiting for HIPCIE done, reg: 0x%x\n", reg);
+ sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_HIPCIE,
+ SKL_ADSP_REG_HIPCIE_DONE,
+ SKL_ADSP_REG_HIPCIE_DONE);
+ }
+
+ /* enable Interrupt */
+ skl_ipc_int_enable(ctx);
+ skl_ipc_op_int_enable(ctx);
+
+ for (i = BXT_INIT_TIMEOUT; i > 0; --i) {
+ if (SKL_FW_INIT ==
+ (sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS) &
+ SKL_FW_STS_MASK)) {
+
+ dev_info(ctx->dev, "ROM loaded, continue FW loading\n");
+ break;
+ }
+ mdelay(1);
+ }
+ if (!i) {
+ dev_err(ctx->dev, "Timeout for ROM init, HIPCIE: 0x%x\n", reg);
+ ret = -EIO;
+ goto base_fw_load_failed;
+ }
+
+ return ret;
+
+base_fw_load_failed:
+ ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, stream_tag);
+ skl_dsp_disable_core(ctx);
+ return ret;
+}
+
+static int sst_transfer_fw_host_dma(struct sst_dsp *ctx)
+{
+ int ret;
+
+ ctx->dsp_ops.trigger(ctx->dev, true, ctx->dsp_ops.stream_tag);
+ ret = sst_dsp_register_poll(ctx, BXT_ADSP_FW_STATUS, SKL_FW_STS_MASK,
+ BXT_ROM_INIT, BXT_BASEFW_TIMEOUT, "Firmware boot");
+
+ ctx->dsp_ops.trigger(ctx->dev, false, ctx->dsp_ops.stream_tag);
+ ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, ctx->dsp_ops.stream_tag);
+
+ return ret;
+}
+
+static int bxt_load_base_firmware(struct sst_dsp *ctx)
+{
+ const struct firmware *fw = NULL;
+ struct skl_sst *skl = ctx->thread_context;
+ int ret;
+
+ ret = request_firmware(&fw, ctx->fw_name, ctx->dev);
+ if (ret < 0) {
+ dev_err(ctx->dev, "Request firmware failed %d\n", ret);
+ goto sst_load_base_firmware_failed;
+ }
+
+ ret = sst_bxt_prepare_fw(ctx, fw->data, fw->size);
+ /* Retry Enabling core and ROM load. Retry seemed to help */
+ if (ret < 0) {
+ ret = sst_bxt_prepare_fw(ctx, fw->data, fw->size);
+ if (ret < 0) {
+ dev_err(ctx->dev, "Core En/ROM load fail:%d\n", ret);
+ goto sst_load_base_firmware_failed;
+ }
+ }
+
+ ret = sst_transfer_fw_host_dma(ctx);
+ if (ret < 0) {
+ dev_err(ctx->dev, "Transfer firmware failed %d\n", ret);
+ dev_info(ctx->dev, "Error code=0x%x: FW status=0x%x\n",
+ sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE),
+ sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS));
+
+ skl_dsp_disable_core(ctx);
+ } else {
+ dev_dbg(ctx->dev, "Firmware download successful\n");
+ ret = wait_event_timeout(skl->boot_wait, skl->boot_complete,
+ msecs_to_jiffies(SKL_IPC_BOOT_MSECS));
+ if (ret == 0) {
+ dev_err(ctx->dev, "DSP boot fail, FW Ready timeout\n");
+ skl_dsp_disable_core(ctx);
+ ret = -EIO;
+ } else {
+ skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING);
+ ret = 0;
+ }
+ }
+
+sst_load_base_firmware_failed:
+ release_firmware(fw);
+ return ret;
+}
+
+static int bxt_set_dsp_D0(struct sst_dsp *ctx)
+{
+ struct skl_sst *skl = ctx->thread_context;
+ int ret;
+
+ skl->boot_complete = false;
+
+ ret = skl_dsp_enable_core(ctx);
+ if (ret < 0) {
+ dev_err(ctx->dev, "enable dsp core failed ret: %d\n", ret);
+ return ret;
+ }
+
+ /* enable interrupt */
+ skl_ipc_int_enable(ctx);
+ skl_ipc_op_int_enable(ctx);
+
+ ret = wait_event_timeout(skl->boot_wait, skl->boot_complete,
+ msecs_to_jiffies(SKL_IPC_BOOT_MSECS));
+ if (ret == 0) {
+ dev_err(ctx->dev, "ipc: error DSP boot timeout\n");
+ dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n",
+ sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE),
+ sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS));
+ return -EIO;
+ }
+
+ skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING);
+ return 0;
+}
+
+static int bxt_set_dsp_D3(struct sst_dsp *ctx)
+{
+ struct skl_ipc_dxstate_info dx;
+ struct skl_sst *skl = ctx->thread_context;
+ int ret = 0;
+
+ if (!is_skl_dsp_running(ctx))
+ return ret;
+
+ dx.core_mask = SKL_DSP_CORE0_MASK;
+ dx.dx_mask = SKL_IPC_D3_MASK;
+
+ ret = skl_ipc_set_dx(&skl->ipc, SKL_INSTANCE_ID,
+ SKL_BASE_FW_MODULE_ID, &dx);
+ if (ret < 0) {
+ dev_err(ctx->dev, "Failed to set DSP to D3 state: %d\n", ret);
+ return ret;
+ }
+
+ ret = skl_dsp_disable_core(ctx);
+ if (ret < 0) {
+ dev_err(ctx->dev, "disbale dsp core failed: %d\n", ret);
+ ret = -EIO;
+ }
+
+ skl_dsp_set_state_locked(ctx, SKL_DSP_RESET);
+ return 0;
+}
+
+static struct skl_dsp_fw_ops bxt_fw_ops = {
+ .set_state_D0 = bxt_set_dsp_D0,
+ .set_state_D3 = bxt_set_dsp_D3,
+ .load_fw = bxt_load_base_firmware,
+ .get_fw_errcode = bxt_get_errorcode,
+};
+
+static struct sst_ops skl_ops = {
+ .irq_handler = skl_dsp_sst_interrupt,
+ .write = sst_shim32_write,
+ .read = sst_shim32_read,
+ .ram_read = sst_memcpy_fromio_32,
+ .ram_write = sst_memcpy_toio_32,
+ .free = skl_dsp_free,
+};
+
+static struct sst_dsp_device skl_dev = {
+ .thread = skl_dsp_irq_thread_handler,
+ .ops = &skl_ops,
+};
+
+int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
+ const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
+ struct skl_sst **dsp)
+{
+ struct skl_sst *skl;
+ struct sst_dsp *sst;
+ int ret;
+
+ skl = devm_kzalloc(dev, sizeof(*skl), GFP_KERNEL);
+ if (skl == NULL)
+ return -ENOMEM;
+
+ skl->dev = dev;
+ skl_dev.thread_context = skl;
+
+ skl->dsp = skl_dsp_ctx_init(dev, &skl_dev, irq);
+ if (!skl->dsp) {
+ dev_err(skl->dev, "skl_dsp_ctx_init failed\n");
+ return -ENODEV;
+ }
+
+ sst = skl->dsp;
+ sst->fw_name = fw_name;
+ sst->dsp_ops = dsp_ops;
+ sst->fw_ops = bxt_fw_ops;
+ sst->addr.lpe = mmio_base;
+ sst->addr.shim = mmio_base;
+
+ sst_dsp_mailbox_init(sst, (BXT_ADSP_SRAM0_BASE + SKL_ADSP_W0_STAT_SZ),
+ SKL_ADSP_W0_UP_SZ, BXT_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ);
+
+ ret = skl_ipc_init(dev, skl);
+ if (ret)
+ return ret;
+
+ skl->boot_complete = false;
+ init_waitqueue_head(&skl->boot_wait);
+
+ ret = sst->fw_ops.load_fw(sst);
+ if (ret < 0) {
+ dev_err(dev, "Load base fw failed: %x", ret);
+ return ret;
+ }
+
+ if (dsp)
+ *dsp = skl;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(bxt_sst_dsp_init);
+
+
+void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx)
+{
+ skl_ipc_free(&ctx->ipc);
+ ctx->dsp->cl_dev.ops.cl_cleanup_controller(ctx->dsp);
+
+ if (ctx->dsp->addr.lpe)
+ iounmap(ctx->dsp->addr.lpe);
+
+ ctx->dsp->ops->free(ctx->dsp);
+}
+EXPORT_SYMBOL_GPL(bxt_sst_dsp_cleanup);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Broxton IPC driver");
diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index 79c5089b85d6..226db84ba20f 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -72,6 +72,105 @@ static void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable)
skl_ipc_set_large_config(&ctx->ipc, &msg, (u32 *)&mask);
}
+static int skl_dsp_setup_spib(struct device *dev, unsigned int size,
+ int stream_tag, int enable)
+{
+ struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+ struct hdac_bus *bus = ebus_to_hbus(ebus);
+ struct hdac_stream *stream = snd_hdac_get_stream(bus,
+ SNDRV_PCM_STREAM_PLAYBACK, stream_tag);
+ struct hdac_ext_stream *estream;
+
+ if (!stream)
+ return -EINVAL;
+
+ estream = stream_to_hdac_ext_stream(stream);
+ /* enable/disable SPIB for this hdac stream */
+ snd_hdac_ext_stream_spbcap_enable(ebus, enable, stream->index);
+
+ /* set the spib value */
+ snd_hdac_ext_stream_set_spib(ebus, estream, size);
+
+ return 0;
+}
+
+static int skl_dsp_prepare(struct device *dev, unsigned int format,
+ unsigned int size, struct snd_dma_buffer *dmab)
+{
+ struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+ struct hdac_bus *bus = ebus_to_hbus(ebus);
+ struct hdac_ext_stream *estream;
+ struct hdac_stream *stream;
+ struct snd_pcm_substream substream;
+ int ret;
+
+ if (!bus)
+ return -ENODEV;
+
+ memset(&substream, 0, sizeof(substream));
+ substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+
+ estream = snd_hdac_ext_stream_assign(ebus, &substream,
+ HDAC_EXT_STREAM_TYPE_HOST);
+ if (!estream)
+ return -ENODEV;
+
+ stream = hdac_stream(estream);
+
+ /* assign decouple host dma channel */
+ ret = snd_hdac_dsp_prepare(stream, format, size, dmab);
+ if (ret < 0)
+ return ret;
+
+ skl_dsp_setup_spib(dev, size, stream->stream_tag, true);
+
+ return stream->stream_tag;
+}
+
+static int skl_dsp_trigger(struct device *dev, bool start, int stream_tag)
+{
+ struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+ struct hdac_stream *stream;
+ struct hdac_bus *bus = ebus_to_hbus(ebus);
+
+ if (!bus)
+ return -ENODEV;
+
+ stream = snd_hdac_get_stream(bus,
+ SNDRV_PCM_STREAM_PLAYBACK, stream_tag);
+ if (!stream)
+ return -EINVAL;
+
+ snd_hdac_dsp_trigger(stream, start);
+
+ return 0;
+}
+
+static int skl_dsp_cleanup(struct device *dev,
+ struct snd_dma_buffer *dmab, int stream_tag)
+{
+ struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+ struct hdac_stream *stream;
+ struct hdac_ext_stream *estream;
+ struct hdac_bus *bus = ebus_to_hbus(ebus);
+
+ if (!bus)
+ return -ENODEV;
+
+ stream = snd_hdac_get_stream(bus,
+ SNDRV_PCM_STREAM_PLAYBACK, stream_tag);
+ if (!stream)
+ return -EINVAL;
+
+ estream = stream_to_hdac_ext_stream(stream);
+ skl_dsp_setup_spib(dev, 0, stream_tag, false);
+ snd_hdac_ext_stream_release(estream, HDAC_EXT_STREAM_TYPE_HOST);
+
+ snd_hdac_dsp_cleanup(stream, dmab);
+
+ return 0;
+}
+
static struct skl_dsp_loader_ops skl_get_loader_ops(void)
{
struct skl_dsp_loader_ops loader_ops;
@@ -84,6 +183,21 @@ static struct skl_dsp_loader_ops skl_get_loader_ops(void)
return loader_ops;
};
+static struct skl_dsp_loader_ops bxt_get_loader_ops(void)
+{
+ struct skl_dsp_loader_ops loader_ops;
+
+ memset(&loader_ops, 0, sizeof(loader_ops));
+
+ loader_ops.alloc_dma_buf = skl_alloc_dma_buf;
+ loader_ops.free_dma_buf = skl_free_dma_buf;
+ loader_ops.prepare = skl_dsp_prepare;
+ loader_ops.trigger = skl_dsp_trigger;
+ loader_ops.cleanup = skl_dsp_cleanup;
+
+ return loader_ops;
+};
+
static const struct skl_dsp_ops dsp_ops[] = {
{
.id = 0x9d70,
@@ -91,6 +205,12 @@ static const struct skl_dsp_ops dsp_ops[] = {
.init = skl_sst_dsp_init,
.cleanup = skl_sst_dsp_cleanup
},
+ {
+ .id = 0x5a98,
+ .loader_ops = bxt_get_loader_ops,
+ .init = bxt_sst_dsp_init,
+ .cleanup = bxt_sst_dsp_cleanup
+ },
};
static int skl_get_dsp_ops(int pci_id)
@@ -744,7 +864,7 @@ int skl_init_module(struct skl_sst *ctx,
return ret;
}
mconfig->m_state = SKL_MODULE_INIT_DONE;
-
+ kfree(param_data);
return ret;
}
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 14d1916ea9f8..7d73648e5f9a 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -25,11 +25,12 @@ static u8 OSC_UUID[16] = {0x6E, 0x88, 0x9F, 0xA6, 0xEB, 0x6C, 0x94, 0x45,
#define DSDT_NHLT_PATH "\\_SB.PCI0.HDAS"
-void *skl_nhlt_init(struct device *dev)
+struct nhlt_acpi_table *skl_nhlt_init(struct device *dev)
{
acpi_handle handle;
union acpi_object *obj;
struct nhlt_resource_desc *nhlt_ptr = NULL;
+ struct nhlt_acpi_table *nhlt_table = NULL;
if (ACPI_FAILURE(acpi_get_handle(NULL, DSDT_NHLT_PATH, &handle))) {
dev_err(dev, "Requested NHLT device not found\n");
@@ -39,18 +40,20 @@ void *skl_nhlt_init(struct device *dev)
obj = acpi_evaluate_dsm(handle, OSC_UUID, 1, 1, NULL);
if (obj && obj->type == ACPI_TYPE_BUFFER) {
nhlt_ptr = (struct nhlt_resource_desc *)obj->buffer.pointer;
-
- return memremap(nhlt_ptr->min_addr, nhlt_ptr->length,
+ nhlt_table = (struct nhlt_acpi_table *)
+ memremap(nhlt_ptr->min_addr, nhlt_ptr->length,
MEMREMAP_WB);
+ ACPI_FREE(obj);
+ return nhlt_table;
}
dev_err(dev, "device specific method to extract NHLT blob failed\n");
return NULL;
}
-void skl_nhlt_free(void *addr)
+void skl_nhlt_free(struct nhlt_acpi_table *nhlt)
{
- memunmap(addr);
+ memunmap((void *) nhlt);
}
static struct nhlt_specific_cfg *skl_get_specific_cfg(
@@ -120,7 +123,7 @@ struct nhlt_specific_cfg
struct hdac_bus *bus = ebus_to_hbus(&skl->ebus);
struct device *dev = bus->dev;
struct nhlt_specific_cfg *sp_config;
- struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt;
+ struct nhlt_acpi_table *nhlt = skl->nhlt;
u16 bps = (s_fmt == 16) ? 16 : 32;
u8 j;
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index dab0900eef26..7c81b31748ff 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -51,7 +51,7 @@ static struct snd_pcm_hardware azx_pcm_hw = {
.rate_min = 8000,
.rate_max = 48000,
.channels_min = 1,
- .channels_max = HDA_QUAD,
+ .channels_max = 8,
.buffer_bytes_max = AZX_MAX_BUF_SIZE,
.period_bytes_min = 128,
.period_bytes_max = AZX_MAX_BUF_SIZE / 2,
@@ -213,7 +213,7 @@ static int skl_be_prepare(struct snd_pcm_substream *substream,
struct skl_sst *ctx = skl->skl_sst;
struct skl_module_cfg *mconfig;
- if ((dai->playback_active > 1) || (dai->capture_active > 1))
+ if (dai->playback_widget->power || dai->capture_widget->power)
return 0;
mconfig = skl_tplg_be_get_cpr_module(dai, substream->stream);
@@ -402,23 +402,33 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
struct skl_module_cfg *mconfig;
struct hdac_ext_bus *ebus = get_bus_ctx(substream);
struct hdac_ext_stream *stream = get_hdac_ext_stream(substream);
+ struct snd_soc_dapm_widget *w;
int ret;
mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream);
if (!mconfig)
return -EIO;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ w = dai->playback_widget;
+ else
+ w = dai->capture_widget;
+
switch (cmd) {
case SNDRV_PCM_TRIGGER_RESUME:
- skl_pcm_prepare(substream, dai);
- /*
- * enable DMA Resume enable bit for the stream, set the dpib
- * & lpib position to resune before starting the DMA
- */
- snd_hdac_ext_stream_drsm_enable(ebus, true,
- hdac_stream(stream)->index);
- snd_hdac_ext_stream_set_dpibr(ebus, stream, stream->dpib);
- snd_hdac_ext_stream_set_lpib(stream, stream->lpib);
+ if (!w->ignore_suspend) {
+ skl_pcm_prepare(substream, dai);
+ /*
+ * enable DMA Resume enable bit for the stream, set the
+ * dpib & lpib position to resume before starting the
+ * DMA
+ */
+ snd_hdac_ext_stream_drsm_enable(ebus, true,
+ hdac_stream(stream)->index);
+ snd_hdac_ext_stream_set_dpibr(ebus, stream,
+ stream->dpib);
+ snd_hdac_ext_stream_set_lpib(stream, stream->lpib);
+ }
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
@@ -448,7 +458,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
return ret;
ret = skl_decoupled_trigger(substream, cmd);
- if (cmd == SNDRV_PCM_TRIGGER_SUSPEND) {
+ if ((cmd == SNDRV_PCM_TRIGGER_SUSPEND) && !w->ignore_suspend) {
/* save the dpib and lpib positions */
stream->dpib = readl(ebus->bus.remap_addr +
AZX_REG_VS_SDXDPIB_XBASE +
@@ -523,7 +533,6 @@ static int skl_link_pcm_prepare(struct snd_pcm_substream *substream,
if (!link)
return -EINVAL;
- snd_hdac_ext_bus_link_power_up(link);
snd_hdac_ext_link_stream_reset(link_dev);
snd_hdac_ext_link_stream_setup(link_dev, format_val);
@@ -682,7 +691,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
.playback = {
.stream_name = "HDMI1 Playback",
.channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
+ .channels_max = 8,
.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |
SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |
@@ -697,7 +706,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
.playback = {
.stream_name = "HDMI2 Playback",
.channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
+ .channels_max = 8,
.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |
SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |
@@ -712,7 +721,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
.playback = {
.stream_name = "HDMI3 Playback",
.channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
+ .channels_max = 8,
.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |
SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |
@@ -760,12 +769,84 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
},
},
{
+ .name = "SSP2 Pin",
+ .ops = &skl_be_ssp_dai_ops,
+ .playback = {
+ .stream_name = "ssp2 Tx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "ssp2 Rx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
+{
+ .name = "SSP3 Pin",
+ .ops = &skl_be_ssp_dai_ops,
+ .playback = {
+ .stream_name = "ssp3 Tx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "ssp3 Rx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
+{
+ .name = "SSP4 Pin",
+ .ops = &skl_be_ssp_dai_ops,
+ .playback = {
+ .stream_name = "ssp4 Tx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "ssp4 Rx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
+{
+ .name = "SSP5 Pin",
+ .ops = &skl_be_ssp_dai_ops,
+ .playback = {
+ .stream_name = "ssp5 Tx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "ssp5 Rx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
+{
.name = "iDisp1 Pin",
.ops = &skl_link_dai_ops,
.playback = {
.stream_name = "iDisp1 Tx",
.channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
+ .channels_max = 8,
.rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE |
SNDRV_PCM_FMTBIT_S24_LE,
@@ -777,7 +858,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
.playback = {
.stream_name = "iDisp2 Tx",
.channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
+ .channels_max = 8,
.rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|
SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE |
@@ -790,7 +871,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
.playback = {
.stream_name = "iDisp3 Tx",
.channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
+ .channels_max = 8,
.rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|
SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE |
diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c
index 2962ef22fc84..13c19855ee1a 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.c
+++ b/sound/soc/intel/skylake/skl-sst-dsp.c
@@ -336,8 +336,6 @@ void skl_dsp_free(struct sst_dsp *dsp)
skl_ipc_int_disable(dsp);
free_irq(dsp->irq, dsp);
- dsp->cl_dev.ops.cl_cleanup_controller(dsp);
- skl_cldma_int_disable(dsp);
skl_ipc_op_int_disable(dsp);
skl_ipc_int_disable(dsp);
diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h
index b6e310d49dd6..deabe7308d3b 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.h
+++ b/sound/soc/intel/skylake/skl-sst-dsp.h
@@ -118,16 +118,25 @@ struct skl_dsp_fw_ops {
int (*set_state_D0)(struct sst_dsp *ctx);
int (*set_state_D3)(struct sst_dsp *ctx);
unsigned int (*get_fw_errcode)(struct sst_dsp *ctx);
- int (*load_mod)(struct sst_dsp *ctx, u16 mod_id, char *mod_name);
+ int (*load_mod)(struct sst_dsp *ctx, u16 mod_id, u8 *mod_name);
int (*unload_mod)(struct sst_dsp *ctx, u16 mod_id);
};
struct skl_dsp_loader_ops {
+ int stream_tag;
+
int (*alloc_dma_buf)(struct device *dev,
struct snd_dma_buffer *dmab, size_t size);
int (*free_dma_buf)(struct device *dev,
struct snd_dma_buffer *dmab);
+ int (*prepare)(struct device *dev, unsigned int format,
+ unsigned int byte_size,
+ struct snd_dma_buffer *bufp);
+ int (*trigger)(struct device *dev, bool start, int stream_tag);
+
+ int (*cleanup)(struct device *dev, struct snd_dma_buffer *dmab,
+ int stream_tag);
};
struct skl_load_module_info {
@@ -160,6 +169,10 @@ int skl_dsp_boot(struct sst_dsp *ctx);
int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
struct skl_sst **dsp);
+int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
+ const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
+ struct skl_sst **dsp);
void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx);
+void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx);
#endif /*__SKL_SST_DSP_H__*/
diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c
index 348a734f8e24..13ec8d53b526 100644
--- a/sound/soc/intel/skylake/skl-sst.c
+++ b/sound/soc/intel/skylake/skl-sst.c
@@ -20,6 +20,7 @@
#include <linux/delay.h>
#include <linux/device.h>
#include <linux/err.h>
+#include <linux/uuid.h>
#include "../common/sst-dsp.h"
#include "../common/sst-dsp-priv.h"
#include "../common/sst-ipc.h"
@@ -304,14 +305,16 @@ static int skl_transfer_module(struct sst_dsp *ctx,
return ret;
}
-static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, char *guid)
+static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid)
{
struct skl_module_table *module_entry = NULL;
int ret = 0;
char mod_name[64]; /* guid str = 32 chars + 4 hyphens */
+ uuid_le *uuid_mod;
- snprintf(mod_name, sizeof(mod_name), "%s%s%s",
- "intel/dsp_fw_", guid, ".bin");
+ uuid_mod = (uuid_le *)guid;
+ snprintf(mod_name, sizeof(mod_name), "%s%pUL%s",
+ "intel/dsp_fw_", uuid_mod, ".bin");
module_entry = skl_module_get_from_id(ctx, mod_id);
if (module_entry == NULL) {
@@ -451,6 +454,10 @@ void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx)
skl_clear_module_table(ctx->dsp);
skl_ipc_free(&ctx->ipc);
ctx->dsp->ops->free(ctx->dsp);
+ if (ctx->boot_complete) {
+ ctx->dsp->cl_dev.ops.cl_cleanup_controller(ctx->dsp);
+ skl_cldma_int_disable(ctx->dsp);
+ }
}
EXPORT_SYMBOL_GPL(skl_sst_dsp_cleanup);
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index cdb78b7e5a14..3e036b0349b9 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -154,13 +154,32 @@ static void skl_dump_mconfig(struct skl_sst *ctx,
dev_dbg(ctx->dev, "ch_cfg = %d\n", mcfg->out_fmt[0].ch_cfg);
}
+static void skl_tplg_update_chmap(struct skl_module_fmt *fmt, int chs)
+{
+ int slot_map = 0xFFFFFFFF;
+ int start_slot = 0;
+ int i;
+
+ for (i = 0; i < chs; i++) {
+ /*
+ * For 2 channels with starting slot as 0, slot map will
+ * look like 0xFFFFFF10.
+ */
+ slot_map &= (~(0xF << (4 * i)) | (start_slot << (4 * i)));
+ start_slot++;
+ }
+ fmt->ch_map = slot_map;
+}
+
static void skl_tplg_update_params(struct skl_module_fmt *fmt,
struct skl_pipe_params *params, int fixup)
{
if (fixup & SKL_RATE_FIXUP_MASK)
fmt->s_freq = params->s_freq;
- if (fixup & SKL_CH_FIXUP_MASK)
+ if (fixup & SKL_CH_FIXUP_MASK) {
fmt->channels = params->ch;
+ skl_tplg_update_chmap(fmt, fmt->channels);
+ }
if (fixup & SKL_FMT_FIXUP_MASK) {
fmt->valid_bit_depth = skl_get_bit_depth(params->s_fmt);
@@ -1564,6 +1583,8 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt,
return -ENOMEM;
w->priv = mconfig;
+ memcpy(&mconfig->guid, &dfw_config->uuid, 16);
+
mconfig->id.module_id = dfw_config->module_id;
mconfig->id.instance_id = dfw_config->instance_id;
mconfig->mcps = dfw_config->max_mcps;
@@ -1593,10 +1614,6 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt,
mconfig->time_slot = dfw_config->time_slot;
mconfig->formats_config.caps_size = dfw_config->caps.caps_size;
- if (dfw_config->is_loadable)
- memcpy(mconfig->guid, dfw_config->uuid,
- ARRAY_SIZE(dfw_config->uuid));
-
mconfig->m_in_pin = devm_kzalloc(bus->dev, (mconfig->max_in_queue) *
sizeof(*mconfig->m_in_pin),
GFP_KERNEL);
diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index d2d923002d5c..e4b399cd7868 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -281,7 +281,7 @@ enum skl_module_state {
};
struct skl_module_cfg {
- char guid[SKL_UUID_STR_SZ];
+ u8 guid[16];
struct skl_module_inst_id id;
u8 domain;
bool homogenous_inputs;
diff --git a/sound/soc/intel/skylake/skl-tplg-interface.h b/sound/soc/intel/skylake/skl-tplg-interface.h
index 1db88a63ac17..a32e5e9cc530 100644
--- a/sound/soc/intel/skylake/skl-tplg-interface.h
+++ b/sound/soc/intel/skylake/skl-tplg-interface.h
@@ -181,7 +181,7 @@ struct skl_dfw_pipe {
} __packed;
struct skl_dfw_module {
- char uuid[SKL_UUID_STR_SZ];
+ u8 uuid[16];
u16 module_id;
u16 instance_id;
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 3982f5536f2d..06d8c263c68f 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -229,7 +229,12 @@ static int skl_suspend(struct device *dev)
* running, we need to save the state for these and continue
*/
if (skl->supend_active) {
+ /* turn off the links and stop the CORB/RIRB DMA if it is On */
snd_hdac_ext_bus_link_power_down_all(ebus);
+
+ if (ebus->cmd_dma_state)
+ snd_hdac_bus_stop_cmd_io(&ebus->bus);
+
enable_irq_wake(bus->irq);
pci_save_state(pci);
pci_disable_device(pci);
@@ -255,6 +260,7 @@ static int skl_resume(struct device *dev)
struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
struct skl *skl = ebus_to_skl(ebus);
struct hdac_bus *bus = ebus_to_hbus(ebus);
+ struct hdac_ext_link *hlink = NULL;
int ret;
/* Turned OFF in HDMI codec driver after codec reconfiguration */
@@ -276,8 +282,29 @@ static int skl_resume(struct device *dev)
ret = pci_enable_device(pci);
snd_hdac_ext_bus_link_power_up_all(ebus);
disable_irq_wake(bus->irq);
+ /*
+ * turn On the links which are On before active suspend
+ * and start the CORB/RIRB DMA if On before
+ * active suspend.
+ */
+ list_for_each_entry(hlink, &ebus->hlink_list, list) {
+ if (hlink->ref_count)
+ snd_hdac_ext_bus_link_power_up(hlink);
+ }
+
+ if (ebus->cmd_dma_state)
+ snd_hdac_bus_init_cmd_io(&ebus->bus);
} else {
ret = _skl_resume(ebus);
+
+ /* turn off the links which are off before suspend */
+ list_for_each_entry(hlink, &ebus->hlink_list, list) {
+ if (!hlink->ref_count)
+ snd_hdac_ext_bus_link_power_down(hlink);
+ }
+
+ if (!ebus->cmd_dma_state)
+ snd_hdac_bus_stop_cmd_io(&ebus->bus);
}
return ret;
@@ -613,6 +640,7 @@ static int skl_probe(struct pci_dev *pci,
struct skl *skl;
struct hdac_ext_bus *ebus = NULL;
struct hdac_bus *bus = NULL;
+ struct hdac_ext_link *hlink = NULL;
int err;
/* we use ext core ops, so provide NULL for ops here */
@@ -643,7 +671,7 @@ static int skl_probe(struct pci_dev *pci,
err = skl_machine_device_register(skl,
(void *)pci_id->driver_data);
if (err < 0)
- goto out_free;
+ goto out_nhlt_free;
err = skl_init_dsp(skl);
if (err < 0) {
@@ -679,6 +707,12 @@ static int skl_probe(struct pci_dev *pci,
}
}
+ /*
+ * we are done probling so decrement link counts
+ */
+ list_for_each_entry(hlink, &ebus->hlink_list, list)
+ snd_hdac_ext_bus_link_put(ebus, hlink);
+
/*configure PM */
pm_runtime_put_noidle(bus->dev);
pm_runtime_allow(bus->dev);
@@ -693,6 +727,8 @@ out_dsp_free:
skl_free_dsp(skl);
out_mach_free:
skl_machine_device_unregister(skl);
+out_nhlt_free:
+ skl_nhlt_free(skl->nhlt);
out_free:
skl->init_failed = 1;
skl_free(ebus);
@@ -743,6 +779,7 @@ static void skl_remove(struct pci_dev *pci)
skl_free_dsp(skl);
skl_machine_device_unregister(skl);
skl_dmic_device_unregister(skl);
+ skl_nhlt_free(skl->nhlt);
skl_free(ebus);
dev_set_drvdata(&pci->dev, NULL);
}
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index 39e16fa7a92b..4b4b3876aea9 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -66,7 +66,7 @@ struct skl {
struct platform_device *dmic_dev;
struct platform_device *i2s_dev;
- void *nhlt; /* nhlt ptr */
+ struct nhlt_acpi_table *nhlt; /* nhlt ptr */
struct skl_sst *skl_sst; /* sst skl ctx */
struct skl_dsp_resource resource;
@@ -103,8 +103,8 @@ struct skl_dsp_ops {
int skl_platform_unregister(struct device *dev);
int skl_platform_register(struct device *dev);
-void *skl_nhlt_init(struct device *dev);
-void skl_nhlt_free(void *addr);
+struct nhlt_acpi_table *skl_nhlt_init(struct device *dev);
+void skl_nhlt_free(struct nhlt_acpi_table *addr);
struct nhlt_specific_cfg *skl_get_ep_blob(struct skl *skl, u32 instance,
u8 link_type, u8 s_fmt, u8 no_ch, u32 s_rate, u8 dirn);
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 2f8e20416bd3..574c6af28c06 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -34,6 +34,13 @@ struct rk_i2s_dev {
struct regmap *regmap;
+/*
+ * Used to indicate the tx/rx status.
+ * I2S controller hopes to start the tx and rx together,
+ * also to stop them when they are both try to stop.
+*/
+ bool tx_start;
+ bool rx_start;
bool is_master_mode;
};
@@ -75,29 +82,37 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on)
I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE);
regmap_update_bits(i2s->regmap, I2S_XFER,
- I2S_XFER_TXS_START,
- I2S_XFER_TXS_START);
+ I2S_XFER_TXS_START | I2S_XFER_RXS_START,
+ I2S_XFER_TXS_START | I2S_XFER_RXS_START);
+
+ i2s->tx_start = true;
} else {
+ i2s->tx_start = false;
+
regmap_update_bits(i2s->regmap, I2S_DMACR,
I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE);
- regmap_update_bits(i2s->regmap, I2S_XFER,
- I2S_XFER_TXS_START,
- I2S_XFER_TXS_STOP);
+ if (!i2s->rx_start) {
+ regmap_update_bits(i2s->regmap, I2S_XFER,
+ I2S_XFER_TXS_START |
+ I2S_XFER_RXS_START,
+ I2S_XFER_TXS_STOP |
+ I2S_XFER_RXS_STOP);
- regmap_update_bits(i2s->regmap, I2S_CLR,
- I2S_CLR_TXC,
- I2S_CLR_TXC);
+ regmap_update_bits(i2s->regmap, I2S_CLR,
+ I2S_CLR_TXC | I2S_CLR_RXC,
+ I2S_CLR_TXC | I2S_CLR_RXC);
- regmap_read(i2s->regmap, I2S_CLR, &val);
-
- /* Should wait for clear operation to finish */
- while (val & I2S_CLR_TXC) {
regmap_read(i2s->regmap, I2S_CLR, &val);
- retry--;
- if (!retry) {
- dev_warn(i2s->dev, "fail to clear\n");
- break;
+
+ /* Should wait for clear operation to finish */
+ while (val) {
+ regmap_read(i2s->regmap, I2S_CLR, &val);
+ retry--;
+ if (!retry) {
+ dev_warn(i2s->dev, "fail to clear\n");
+ break;
+ }
}
}
}
@@ -113,29 +128,37 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on)
I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE);
regmap_update_bits(i2s->regmap, I2S_XFER,
- I2S_XFER_RXS_START,
- I2S_XFER_RXS_START);
+ I2S_XFER_TXS_START | I2S_XFER_RXS_START,
+ I2S_XFER_TXS_START | I2S_XFER_RXS_START);
+
+ i2s->rx_start = true;
} else {
+ i2s->rx_start = false;
+
regmap_update_bits(i2s->regmap, I2S_DMACR,
I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE);
- regmap_update_bits(i2s->regmap, I2S_XFER,
- I2S_XFER_RXS_START,
- I2S_XFER_RXS_STOP);
+ if (!i2s->tx_start) {
+ regmap_update_bits(i2s->regmap, I2S_XFER,
+ I2S_XFER_TXS_START |
+ I2S_XFER_RXS_START,
+ I2S_XFER_TXS_STOP |
+ I2S_XFER_RXS_STOP);
- regmap_update_bits(i2s->regmap, I2S_CLR,
- I2S_CLR_RXC,
- I2S_CLR_RXC);
+ regmap_update_bits(i2s->regmap, I2S_CLR,
+ I2S_CLR_TXC | I2S_CLR_RXC,
+ I2S_CLR_TXC | I2S_CLR_RXC);
- regmap_read(i2s->regmap, I2S_CLR, &val);
-
- /* Should wait for clear operation to finish */
- while (val & I2S_CLR_RXC) {
regmap_read(i2s->regmap, I2S_CLR, &val);
- retry--;
- if (!retry) {
- dev_warn(i2s->dev, "fail to clear\n");
- break;
+
+ /* Should wait for clear operation to finish */
+ while (val) {
+ regmap_read(i2s->regmap, I2S_CLR, &val);
+ retry--;
+ if (!retry) {
+ dev_warn(i2s->dev, "fail to clear\n");
+ break;
+ }
}
}
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d2e62b159610..16369cad4803 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -930,7 +930,18 @@ static struct snd_soc_component *soc_find_component(
return NULL;
}
-static struct snd_soc_dai *snd_soc_find_dai(
+/**
+ * snd_soc_find_dai - Find a registered DAI
+ *
+ * @dlc: name of the DAI and optional component info to match
+ *
+ * This function will search all regsitered components and their DAIs to
+ * find the DAI of the same name. The component's of_node and name
+ * should also match if being specified.
+ *
+ * Return: pointer of DAI, or NULL if not found.
+ */
+struct snd_soc_dai *snd_soc_find_dai(
const struct snd_soc_dai_link_component *dlc)
{
struct snd_soc_component *component;
@@ -959,6 +970,7 @@ static struct snd_soc_dai *snd_soc_find_dai(
return NULL;
}
+EXPORT_SYMBOL_GPL(snd_soc_find_dai);
static bool soc_is_dai_link_bound(struct snd_soc_card *card,
struct snd_soc_dai_link *dai_link)
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 6fd1906af387..6cef3977507a 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -163,31 +163,42 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea
}
/*
- * Prepare formats mask for valid/allowed sample types. If the dma does
- * not have support for the given physical word size, it needs to be
- * masked out so user space can not use the format which produces
- * corrupted audio.
- * In case the dma driver does not implement the slave_caps the default
- * assumption is that it supports 1, 2 and 4 bytes widths.
+ * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep
+ * hw.formats set to 0, meaning no restrictions are in place.
+ * In this case it's the responsibility of the DAI driver to
+ * provide the supported format information.
*/
- for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
- int bits = snd_pcm_format_physical_width(i);
-
- /* Enable only samples with DMA supported physical widths */
- switch (bits) {
- case 8:
- case 16:
- case 24:
- case 32:
- case 64:
- if (addr_widths & (1 << (bits / 8)))
- hw.formats |= (1LL << i);
- break;
- default:
- /* Unsupported types */
- break;
+ if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK))
+ /*
+ * Prepare formats mask for valid/allowed sample types. If the
+ * dma does not have support for the given physical word size,
+ * it needs to be masked out so user space can not use the
+ * format which produces corrupted audio.
+ * In case the dma driver does not implement the slave_caps the
+ * default assumption is that it supports 1, 2 and 4 bytes
+ * widths.
+ */
+ for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+ int bits = snd_pcm_format_physical_width(i);
+
+ /*
+ * Enable only samples with DMA supported physical
+ * widths
+ */
+ switch (bits) {
+ case 8:
+ case 16:
+ case 24:
+ case 32:
+ case 64:
+ if (addr_widths & (1 << (bits / 8)))
+ hw.formats |= (1LL << i);
+ break;
+ default:
+ /* Unsupported types */
+ break;
+ }
}
- }
return snd_soc_set_runtime_hwparams(substream, &hw);
}
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 */