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authorTakashi Iwai <tiwai@suse.de>2020-03-30 10:46:51 +0300
committerTakashi Iwai <tiwai@suse.de>2020-03-30 10:46:51 +0300
commitaa21c3d4b941739651e77747d2f7a20a6c1d87bc (patch)
treeae46596c4fe5803fbd4746459a09a09cff5487e8 /sound
parent476c02e0b4fd9071d158f6a1a1dfea1d36ee0ffd (diff)
parent652bb5d8df4b3a79ed350db35cda12637e63efa7 (diff)
downloadlinux-aa21c3d4b941739651e77747d2f7a20a6c1d87bc.tar.xz
Merge branch 'for-next' into for-linus
Diffstat (limited to 'sound')
-rw-r--r--sound/core/device.c21
-rw-r--r--sound/core/info.c2
-rw-r--r--sound/core/oss/pcm_oss.c23
-rw-r--r--sound/core/oss/pcm_plugin.c120
-rw-r--r--sound/core/oss/rate.c2
-rw-r--r--sound/core/pcm.c2
-rw-r--r--sound/core/pcm_dmaengine.c2
-rw-r--r--sound/core/pcm_misc.c17
-rw-r--r--sound/core/pcm_native.c47
-rw-r--r--sound/drivers/aloop.c6
-rw-r--r--sound/drivers/dummy.c6
-rw-r--r--sound/firewire/bebob/bebob.c2
-rw-r--r--sound/firewire/digi00x/digi00x.c2
-rw-r--r--sound/firewire/fireface/ff.c2
-rw-r--r--sound/firewire/fireworks/fireworks.c2
-rw-r--r--sound/firewire/tascam/tascam-hwdep.c2
-rw-r--r--sound/firewire/tascam/tascam.c2
-rw-r--r--sound/hda/hdac_device.c2
-rw-r--r--sound/isa/sb/emu8000_pcm.c4
-rw-r--r--sound/pci/ali5451/ali5451.c6
-rw-r--r--sound/pci/emu10k1/emu10k1_main.c4
-rw-r--r--sound/pci/hda/Kconfig1
-rw-r--r--sound/pci/hda/hda_codec.c2
-rw-r--r--sound/pci/hda/hda_controller.c2
-rw-r--r--sound/pci/hda/patch_ca0132.c2
-rw-r--r--sound/pci/hda/patch_hdmi.c313
-rw-r--r--sound/pci/korg1212/korg1212.c2
-rw-r--r--sound/pci/rme9652/hdsp.c3
-rw-r--r--sound/pci/via82xx.c6
-rw-r--r--sound/pci/via82xx_modem.c6
-rw-r--r--sound/ppc/keywest.c9
-rw-r--r--sound/usb/Makefile1
-rw-r--r--sound/usb/card.c38
-rw-r--r--sound/usb/clock.c59
-rw-r--r--sound/usb/format.c37
-rw-r--r--sound/usb/midi.c2
-rw-r--r--sound/usb/mixer.c33
-rw-r--r--sound/usb/mixer_quirks.c5
-rw-r--r--sound/usb/mixer_s1810c.c595
-rw-r--r--sound/usb/mixer_s1810c.h7
-rw-r--r--sound/usb/pcm.c7
-rw-r--r--sound/usb/proc.c2
-rw-r--r--sound/usb/quirks-table.h2
-rw-r--r--sound/usb/quirks.c88
-rw-r--r--sound/usb/quirks.h2
-rw-r--r--sound/usb/stream.c3
-rw-r--r--sound/usb/usbaudio.h1
47 files changed, 1118 insertions, 386 deletions
diff --git a/sound/core/device.c b/sound/core/device.c
index cdc5af526739..bf0b04a7ee79 100644
--- a/sound/core/device.c
+++ b/sound/core/device.c
@@ -237,3 +237,24 @@ void snd_device_free_all(struct snd_card *card)
list_for_each_entry_safe_reverse(dev, next, &card->devices, list)
__snd_device_free(dev);
}
+
+/**
+ * snd_device_get_state - Get the current state of the given device
+ * @card: the card instance
+ * @device_data: the data pointer to release
+ *
+ * Returns the current state of the given device object. For the valid
+ * device, either @SNDRV_DEV_BUILD, @SNDRV_DEV_REGISTERED or
+ * @SNDRV_DEV_DISCONNECTED is returned.
+ * Or for a non-existing device, -1 is returned as an error.
+ */
+int snd_device_get_state(struct snd_card *card, void *device_data)
+{
+ struct snd_device *dev;
+
+ dev = look_for_dev(card, device_data);
+ if (dev)
+ return dev->state;
+ return -1;
+}
+EXPORT_SYMBOL_GPL(snd_device_get_state);
diff --git a/sound/core/info.c b/sound/core/info.c
index ca87ae4c30ba..8c6bc5241df5 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -604,7 +604,7 @@ int snd_info_card_free(struct snd_card *card)
*/
int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len)
{
- int c = -1;
+ int c;
if (snd_BUG_ON(!buffer || !buffer->buffer))
return 1;
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 13db77771f0f..930def8201f4 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -884,20 +884,17 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream)
sformat = snd_pcm_plug_slave_format(format, sformat_mask);
if ((__force int)sformat < 0 ||
- !snd_mask_test(sformat_mask, (__force int)sformat)) {
- for (sformat = (__force snd_pcm_format_t)0;
- (__force int)sformat <= (__force int)SNDRV_PCM_FORMAT_LAST;
- sformat = (__force snd_pcm_format_t)((__force int)sformat + 1)) {
- if (snd_mask_test(sformat_mask, (__force int)sformat) &&
+ !snd_mask_test_format(sformat_mask, sformat)) {
+ pcm_for_each_format(sformat) {
+ if (snd_mask_test_format(sformat_mask, sformat) &&
snd_pcm_oss_format_to(sformat) >= 0)
- break;
- }
- if ((__force int)sformat > (__force int)SNDRV_PCM_FORMAT_LAST) {
- pcm_dbg(substream->pcm, "Cannot find a format!!!\n");
- err = -EINVAL;
- goto failure;
+ goto format_found;
}
+ pcm_dbg(substream->pcm, "Cannot find a format!!!\n");
+ err = -EINVAL;
+ goto failure;
}
+ format_found:
err = _snd_pcm_hw_param_set(sparams, SNDRV_PCM_HW_PARAM_FORMAT, (__force int)sformat, 0);
if (err < 0)
goto failure;
@@ -1220,8 +1217,10 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const
if (ret < 0)
break;
}
+ mutex_unlock(&runtime->oss.params_lock);
ret = __snd_pcm_lib_xfer(substream, (void *)ptr, true,
frames, in_kernel);
+ mutex_lock(&runtime->oss.params_lock);
if (ret != -EPIPE && ret != -ESTRPIPE)
break;
/* test, if we can't store new data, because the stream */
@@ -1257,8 +1256,10 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p
ret = snd_pcm_oss_capture_position_fixup(substream, &delay);
if (ret < 0)
break;
+ mutex_unlock(&runtime->oss.params_lock);
ret = __snd_pcm_lib_xfer(substream, (void *)ptr, true,
frames, in_kernel);
+ mutex_lock(&runtime->oss.params_lock);
if (ret == -EPIPE) {
if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) {
ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index 752d078908e9..fbda4ebf38b3 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -196,82 +196,74 @@ int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin)
return 0;
}
-snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames)
+static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug,
+ snd_pcm_sframes_t frames)
{
- struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next;
- int stream;
+ struct snd_pcm_plugin *plugin, *plugin_next;
- if (snd_BUG_ON(!plug))
- return -ENXIO;
- if (drv_frames == 0)
- return 0;
- stream = snd_pcm_plug_stream(plug);
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- plugin = snd_pcm_plug_last(plug);
- while (plugin && drv_frames > 0) {
- if (drv_frames > plugin->buf_frames)
- drv_frames = plugin->buf_frames;
- plugin_prev = plugin->prev;
- if (plugin->src_frames)
- drv_frames = plugin->src_frames(plugin, drv_frames);
- plugin = plugin_prev;
+ plugin = snd_pcm_plug_first(plug);
+ while (plugin && frames > 0) {
+ plugin_next = plugin->next;
+ if (plugin->dst_frames) {
+ frames = plugin->dst_frames(plugin, frames);
+ if (frames < 0)
+ return frames;
}
- } else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
- plugin = snd_pcm_plug_first(plug);
- while (plugin && drv_frames > 0) {
- plugin_next = plugin->next;
- if (plugin->dst_frames)
- drv_frames = plugin->dst_frames(plugin, drv_frames);
- if (drv_frames > plugin->buf_frames)
- drv_frames = plugin->buf_frames;
- plugin = plugin_next;
+ if (frames > plugin->buf_frames)
+ frames = plugin->buf_frames;
+ plugin = plugin_next;
+ }
+ return frames;
+}
+
+static snd_pcm_sframes_t calc_src_frames(struct snd_pcm_substream *plug,
+ snd_pcm_sframes_t frames)
+{
+ struct snd_pcm_plugin *plugin, *plugin_prev;
+
+ plugin = snd_pcm_plug_last(plug);
+ while (plugin && frames > 0) {
+ if (frames > plugin->buf_frames)
+ frames = plugin->buf_frames;
+ plugin_prev = plugin->prev;
+ if (plugin->src_frames) {
+ frames = plugin->src_frames(plugin, frames);
+ if (frames < 0)
+ return frames;
}
- } else
+ plugin = plugin_prev;
+ }
+ return frames;
+}
+
+snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames)
+{
+ if (snd_BUG_ON(!plug))
+ return -ENXIO;
+ switch (snd_pcm_plug_stream(plug)) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ return calc_src_frames(plug, drv_frames);
+ case SNDRV_PCM_STREAM_CAPTURE:
+ return calc_dst_frames(plug, drv_frames);
+ default:
snd_BUG();
- return drv_frames;
+ return -EINVAL;
+ }
}
snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t clt_frames)
{
- struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next;
- snd_pcm_sframes_t frames;
- int stream;
-
if (snd_BUG_ON(!plug))
return -ENXIO;
- if (clt_frames == 0)
- return 0;
- frames = clt_frames;
- stream = snd_pcm_plug_stream(plug);
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- plugin = snd_pcm_plug_first(plug);
- while (plugin && frames > 0) {
- plugin_next = plugin->next;
- if (plugin->dst_frames) {
- frames = plugin->dst_frames(plugin, frames);
- if (frames < 0)
- return frames;
- }
- if (frames > plugin->buf_frames)
- frames = plugin->buf_frames;
- plugin = plugin_next;
- }
- } else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
- plugin = snd_pcm_plug_last(plug);
- while (plugin) {
- if (frames > plugin->buf_frames)
- frames = plugin->buf_frames;
- plugin_prev = plugin->prev;
- if (plugin->src_frames) {
- frames = plugin->src_frames(plugin, frames);
- if (frames < 0)
- return frames;
- }
- plugin = plugin_prev;
- }
- } else
+ switch (snd_pcm_plug_stream(plug)) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ return calc_dst_frames(plug, clt_frames);
+ case SNDRV_PCM_STREAM_CAPTURE:
+ return calc_src_frames(plug, clt_frames);
+ default:
snd_BUG();
- return frames;
+ return -EINVAL;
+ }
}
static int snd_pcm_plug_formats(const struct snd_mask *mask,
diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c
index 7cd09cef6961..d381f4c967c9 100644
--- a/sound/core/oss/rate.c
+++ b/sound/core/oss/rate.c
@@ -47,7 +47,7 @@ struct rate_priv {
unsigned int pos;
rate_f func;
snd_pcm_sframes_t old_src_frames, old_dst_frames;
- struct rate_channel channels[0];
+ struct rate_channel channels[];
};
static void rate_init(struct snd_pcm_plugin *plugin)
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index a141a301369f..b6d2331a82f7 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -1019,7 +1019,7 @@ static ssize_t show_pcm_class(struct device *dev,
str = "none";
else
str = strs[pcm->dev_class];
- return snprintf(buf, PAGE_SIZE, "%s\n", str);
+ return sprintf(buf, "%s\n", str);
}
static DEVICE_ATTR(pcm_class, 0444, show_pcm_class, NULL);
diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c
index 5749a8a49784..b37c70864b57 100644
--- a/sound/core/pcm_dmaengine.c
+++ b/sound/core/pcm_dmaengine.c
@@ -426,7 +426,7 @@ int snd_dmaengine_pcm_refine_runtime_hwparams(
* default assumption is that it supports 1, 2 and 4 bytes
* widths.
*/
- for (i = SNDRV_PCM_FORMAT_FIRST; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+ pcm_for_each_format(i) {
int bits = snd_pcm_format_physical_width(i);
/*
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index a6a541511534..cf3e8c26e00a 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -42,6 +42,11 @@ struct pcm_format_data {
/* we do lots of calculations on snd_pcm_format_t; shut up sparse */
#define INT __force int
+static bool valid_format(snd_pcm_format_t format)
+{
+ return (INT)format >= 0 && (INT)format <= (INT)SNDRV_PCM_FORMAT_LAST;
+}
+
static const struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = {
[SNDRV_PCM_FORMAT_S8] = {
.width = 8, .phys = 8, .le = -1, .signd = 1,
@@ -259,7 +264,7 @@ static const struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] =
int snd_pcm_format_signed(snd_pcm_format_t format)
{
int val;
- if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+ if (!valid_format(format))
return -EINVAL;
if ((val = pcm_formats[(INT)format].signd) < 0)
return -EINVAL;
@@ -307,7 +312,7 @@ EXPORT_SYMBOL(snd_pcm_format_linear);
int snd_pcm_format_little_endian(snd_pcm_format_t format)
{
int val;
- if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+ if (!valid_format(format))
return -EINVAL;
if ((val = pcm_formats[(INT)format].le) < 0)
return -EINVAL;
@@ -343,7 +348,7 @@ EXPORT_SYMBOL(snd_pcm_format_big_endian);
int snd_pcm_format_width(snd_pcm_format_t format)
{
int val;
- if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+ if (!valid_format(format))
return -EINVAL;
if ((val = pcm_formats[(INT)format].width) == 0)
return -EINVAL;
@@ -361,7 +366,7 @@ EXPORT_SYMBOL(snd_pcm_format_width);
int snd_pcm_format_physical_width(snd_pcm_format_t format)
{
int val;
- if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+ if (!valid_format(format))
return -EINVAL;
if ((val = pcm_formats[(INT)format].phys) == 0)
return -EINVAL;
@@ -394,7 +399,7 @@ EXPORT_SYMBOL(snd_pcm_format_size);
*/
const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format)
{
- if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+ if (!valid_format(format))
return NULL;
if (! pcm_formats[(INT)format].phys)
return NULL;
@@ -418,7 +423,7 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int
unsigned char *dst;
const unsigned char *pat;
- if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+ if (!valid_format(format))
return -EINVAL;
if (samples == 0)
return 0;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index d5443eeb8b63..8e1060b084f1 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -228,6 +228,9 @@ int snd_pcm_info_user(struct snd_pcm_substream *substream,
return err;
}
+/* macro for simplified cast */
+#define PARAM_MASK_BIT(b) (1U << (__force int)(b))
+
static bool hw_support_mmap(struct snd_pcm_substream *substream)
{
if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP))
@@ -257,7 +260,7 @@ static int constrain_mask_params(struct snd_pcm_substream *substream,
return -EINVAL;
/* This parameter is not requested to change by a caller. */
- if (!(params->rmask & (1 << k)))
+ if (!(params->rmask & PARAM_MASK_BIT(k)))
continue;
if (trace_hw_mask_param_enabled())
@@ -271,7 +274,7 @@ static int constrain_mask_params(struct snd_pcm_substream *substream,
/* Set corresponding flag so that the caller gets it. */
trace_hw_mask_param(substream, k, 0, &old_mask, m);
- params->cmask |= 1 << k;
+ params->cmask |= PARAM_MASK_BIT(k);
}
return 0;
@@ -293,7 +296,7 @@ static int constrain_interval_params(struct snd_pcm_substream *substream,
return -EINVAL;
/* This parameter is not requested to change by a caller. */
- if (!(params->rmask & (1 << k)))
+ if (!(params->rmask & PARAM_MASK_BIT(k)))
continue;
if (trace_hw_interval_param_enabled())
@@ -307,7 +310,7 @@ static int constrain_interval_params(struct snd_pcm_substream *substream,
/* Set corresponding flag so that the caller gets it. */
trace_hw_interval_param(substream, k, 0, &old_interval, i);
- params->cmask |= 1 << k;
+ params->cmask |= PARAM_MASK_BIT(k);
}
return 0;
@@ -349,7 +352,7 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream,
* have 0 so that the parameters are never changed anymore.
*/
for (k = 0; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++)
- vstamps[k] = (params->rmask & (1 << k)) ? 1 : 0;
+ vstamps[k] = (params->rmask & PARAM_MASK_BIT(k)) ? 1 : 0;
/* Due to the above design, actual sequence number starts at 2. */
stamp = 2;
@@ -417,7 +420,7 @@ retry:
hw_param_interval(params, r->var));
}
- params->cmask |= (1 << r->var);
+ params->cmask |= PARAM_MASK_BIT(r->var);
vstamps[r->var] = stamp;
again = true;
}
@@ -486,9 +489,9 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream,
params->info = 0;
params->fifo_size = 0;
- if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_SAMPLE_BITS))
+ if (params->rmask & PARAM_MASK_BIT(SNDRV_PCM_HW_PARAM_SAMPLE_BITS))
params->msbits = 0;
- if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_RATE)) {
+ if (params->rmask & PARAM_MASK_BIT(SNDRV_PCM_HW_PARAM_RATE)) {
params->rate_num = 0;
params->rate_den = 0;
}
@@ -2293,21 +2296,21 @@ static int snd_pcm_hw_rule_mulkdiv(struct snd_pcm_hw_params *params,
static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
- unsigned int k;
+ snd_pcm_format_t k;
const struct snd_interval *i =
hw_param_interval_c(params, rule->deps[0]);
struct snd_mask m;
struct snd_mask *mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
snd_mask_any(&m);
- for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) {
+ pcm_for_each_format(k) {
int bits;
- if (! snd_mask_test(mask, k))
+ if (!snd_mask_test_format(mask, k))
continue;
bits = snd_pcm_format_physical_width(k);
if (bits <= 0)
continue; /* ignore invalid formats */
if ((unsigned)bits < i->min || (unsigned)bits > i->max)
- snd_mask_reset(&m, k);
+ snd_mask_reset(&m, (__force unsigned)k);
}
return snd_mask_refine(mask, &m);
}
@@ -2316,14 +2319,15 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_interval t;
- unsigned int k;
+ snd_pcm_format_t k;
+
t.min = UINT_MAX;
t.max = 0;
t.openmin = 0;
t.openmax = 0;
- for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) {
+ pcm_for_each_format(k) {
int bits;
- if (! snd_mask_test(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k))
+ if (!snd_mask_test_format(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k))
continue;
bits = snd_pcm_format_physical_width(k);
if (bits <= 0)
@@ -2505,16 +2509,16 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
unsigned int mask = 0;
if (hw->info & SNDRV_PCM_INFO_INTERLEAVED)
- mask |= 1 << SNDRV_PCM_ACCESS_RW_INTERLEAVED;
+ mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_RW_INTERLEAVED);
if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED)
- mask |= 1 << SNDRV_PCM_ACCESS_RW_NONINTERLEAVED;
+ mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_RW_NONINTERLEAVED);
if (hw_support_mmap(substream)) {
if (hw->info & SNDRV_PCM_INFO_INTERLEAVED)
- mask |= 1 << SNDRV_PCM_ACCESS_MMAP_INTERLEAVED;
+ mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_INTERLEAVED);
if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED)
- mask |= 1 << SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED;
+ mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED);
if (hw->info & SNDRV_PCM_INFO_COMPLEX)
- mask |= 1 << SNDRV_PCM_ACCESS_MMAP_COMPLEX;
+ mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_COMPLEX);
}
err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_ACCESS, mask);
if (err < 0)
@@ -2524,7 +2528,8 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
if (err < 0)
return err;
- err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT, 1 << SNDRV_PCM_SUBFORMAT_STD);
+ err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT,
+ PARAM_MASK_BIT(SNDRV_PCM_SUBFORMAT_STD));
if (err < 0)
return err;
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index d78a27271d6d..251eaf1152e2 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -118,7 +118,7 @@ struct loopback_cable {
struct loopback_setup {
unsigned int notify: 1;
unsigned int rate_shift;
- unsigned int format;
+ snd_pcm_format_t format;
unsigned int rate;
unsigned int channels;
struct snd_ctl_elem_id active_id;
@@ -1432,7 +1432,7 @@ static int loopback_format_info(struct snd_kcontrol *kcontrol,
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 1;
uinfo->value.integer.min = 0;
- uinfo->value.integer.max = SNDRV_PCM_FORMAT_LAST;
+ uinfo->value.integer.max = (__force int)SNDRV_PCM_FORMAT_LAST;
uinfo->value.integer.step = 1;
return 0;
}
@@ -1443,7 +1443,7 @@ static int loopback_format_get(struct snd_kcontrol *kcontrol,
struct loopback *loopback = snd_kcontrol_chip(kcontrol);
ucontrol->value.integer.value[0] =
- loopback->setup[kcontrol->id.subdevice]
+ (__force int)loopback->setup[kcontrol->id.subdevice]
[kcontrol->id.device].format;
return 0;
}
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 02ac3f4e0c02..b5486de08b97 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -901,10 +901,10 @@ static int snd_card_dummy_new_mixer(struct snd_dummy *dummy)
static void print_formats(struct snd_dummy *dummy,
struct snd_info_buffer *buffer)
{
- int i;
+ snd_pcm_format_t i;
- for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
- if (dummy->pcm_hw.formats & (1ULL << i))
+ pcm_for_each_format(i) {
+ if (dummy->pcm_hw.formats & pcm_format_to_bits(i))
snd_iprintf(buffer, " %s", snd_pcm_format_name(i));
}
}
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 976d8cb9a34f..2c8e3392a490 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -509,7 +509,7 @@ MODULE_DEVICE_TABLE(ieee1394, bebob_id_table);
static struct fw_driver bebob_driver = {
.driver = {
.owner = THIS_MODULE,
- .name = "snd-bebob",
+ .name = KBUILD_MODNAME,
.bus = &fw_bus_type,
},
.probe = bebob_probe,
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
index 1f5fc0e7c024..c84b913a9fe0 100644
--- a/sound/firewire/digi00x/digi00x.c
+++ b/sound/firewire/digi00x/digi00x.c
@@ -192,7 +192,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_dg00x_id_table);
static struct fw_driver dg00x_driver = {
.driver = {
.owner = THIS_MODULE,
- .name = "snd-firewire-digi00x",
+ .name = KBUILD_MODNAME,
.bus = &fw_bus_type,
},
.probe = snd_dg00x_probe,
diff --git a/sound/firewire/fireface/ff.c b/sound/firewire/fireface/ff.c
index f5a016560eb8..b62a4fd22407 100644
--- a/sound/firewire/fireface/ff.c
+++ b/sound/firewire/fireface/ff.c
@@ -224,7 +224,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_ff_id_table);
static struct fw_driver ff_driver = {
.driver = {
.owner = THIS_MODULE,
- .name = "snd-fireface",
+ .name = KBUILD_MODNAME,
.bus = &fw_bus_type,
},
.probe = snd_ff_probe,
diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c
index 134fc9ee26b9..b1cc013a3540 100644
--- a/sound/firewire/fireworks/fireworks.c
+++ b/sound/firewire/fireworks/fireworks.c
@@ -362,7 +362,7 @@ MODULE_DEVICE_TABLE(ieee1394, efw_id_table);
static struct fw_driver efw_driver = {
.driver = {
.owner = THIS_MODULE,
- .name = "snd-fireworks",
+ .name = KBUILD_MODNAME,
.bus = &fw_bus_type,
},
.probe = efw_probe,
diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c
index c29a97f6f638..6f38335fe10b 100644
--- a/sound/firewire/tascam/tascam-hwdep.c
+++ b/sound/firewire/tascam/tascam-hwdep.c
@@ -17,6 +17,7 @@
static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf,
long count, loff_t *offset)
+ __releases(&tscm->lock)
{
struct snd_firewire_event_lock_status event = {
.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS,
@@ -36,6 +37,7 @@ static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf,
static long tscm_hwdep_read_queue(struct snd_tscm *tscm, char __user *buf,
long remained, loff_t *offset)
+ __releases(&tscm->lock)
{
char __user *pos = buf;
unsigned int type = SNDRV_FIREWIRE_EVENT_TASCAM_CONTROL;
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
index addc464503bc..5dac0d9fc58e 100644
--- a/sound/firewire/tascam/tascam.c
+++ b/sound/firewire/tascam/tascam.c
@@ -224,7 +224,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_tscm_id_table);
static struct fw_driver tscm_driver = {
.driver = {
.owner = THIS_MODULE,
- .name = "snd-firewire-tascam",
+ .name = KBUILD_MODNAME,
.bus = &fw_bus_type,
},
.probe = snd_tscm_probe,
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index 9a526aeef8da..e3119f5cb0d5 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -204,7 +204,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_set_chip_name);
*/
int snd_hdac_codec_modalias(struct hdac_device *codec, char *buf, size_t size)
{
- return snprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n",
+ return scnprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n",
codec->vendor_id, codec->revision_id, codec->type);
}
EXPORT_SYMBOL_GPL(snd_hdac_codec_modalias);
diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c
index e377ac93f37f..8e8257c574b0 100644
--- a/sound/isa/sb/emu8000_pcm.c
+++ b/sound/isa/sb/emu8000_pcm.c
@@ -435,7 +435,7 @@ enum {
#define LOOP_WRITE(rec, offset, _buf, count, mode) \
do { \
struct snd_emu8000 *emu = (rec)->emu; \
- unsigned short *buf = (unsigned short *)(_buf); \
+ unsigned short *buf = (__force unsigned short *)(_buf); \
snd_emu8000_write_wait(emu, 1); \
EMU8000_SMALW_WRITE(emu, offset); \
while (count > 0) { \
@@ -492,7 +492,7 @@ static int emu8k_pcm_silence(struct snd_pcm_substream *subs,
#define LOOP_WRITE(rec, pos, _buf, count, mode) \
do { \
struct snd_emu8000 *emu = rec->emu; \
- unsigned short *buf = (unsigned short *)(_buf); \
+ unsigned short *buf = (__force unsigned short *)(_buf); \
snd_emu8000_write_wait(emu, 1); \
EMU8000_SMALW_WRITE(emu, pos + rec->loop_start[0]); \
if (rec->voices > 1) \
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 4f524a9dbbca..4462375d2d82 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -1070,7 +1070,7 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream,
{
struct snd_ali *codec = snd_pcm_substream_chip(substream);
struct snd_pcm_substream *s;
- unsigned int what, whati, capture_flag;
+ unsigned int what, whati;
struct snd_ali_voice *pvoice, *evoice;
unsigned int val;
int do_start;
@@ -1088,7 +1088,7 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream,
return -EINVAL;
}
- what = whati = capture_flag = 0;
+ what = whati = 0;
snd_pcm_group_for_each_entry(s, substream) {
if ((struct snd_ali *) snd_pcm_substream_chip(s) == codec) {
pvoice = s->runtime->private_data;
@@ -1110,8 +1110,6 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream,
evoice->running = 0;
}
snd_pcm_trigger_done(s, substream);
- if (pvoice->mode)
- capture_flag = 1;
}
}
spin_lock(&codec->reg_lock);
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index a89a7e603ca8..6ff581733a19 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1789,6 +1789,7 @@ int snd_emu10k1_create(struct snd_card *card,
int idx, err;
int is_audigy;
size_t page_table_size;
+ __le32 *pgtbl;
unsigned int silent_page;
const struct snd_emu_chip_details *c;
static const struct snd_device_ops ops = {
@@ -2009,8 +2010,9 @@ int snd_emu10k1_create(struct snd_card *card,
/* Clear silent pages and set up pointers */
memset(emu->silent_page.area, 0, emu->silent_page.bytes);
silent_page = emu->silent_page.addr << emu->address_mode;
+ pgtbl = (__le32 *)emu->ptb_pages.area;
for (idx = 0; idx < (emu->address_mode ? MAXPAGES1 : MAXPAGES0); idx++)
- ((u32 *)emu->ptb_pages.area)[idx] = cpu_to_le32(silent_page | idx);
+ pgtbl[idx] = cpu_to_le32(silent_page | idx);
/* set up voice indices */
for (idx = 0; idx < NUM_G; idx++) {
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index bd48335d09d7..e1d3082a4fe9 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -184,6 +184,7 @@ comment "Set to Y if you want auto-loading the codec driver"
config SND_HDA_CODEC_CA0132_DSP
bool "Support new DSP code for CA0132 codec"
depends on SND_HDA_CODEC_CA0132
+ default y
select SND_HDA_DSP_LOADER
select FW_LOADER
help
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 53e7732ef752..a34a2c9f4bcf 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -88,7 +88,7 @@ struct hda_conn_list {
struct list_head list;
int len;
hda_nid_t nid;
- hda_nid_t conns[0];
+ hda_nid_t conns[];
};
/* look up the cached results */
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 2609e391ce54..9765652a73d7 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -373,7 +373,7 @@ static int azx_get_sync_time(ktime_t *device,
u32 wallclk_ctr, wallclk_cycles;
bool direction;
u32 dma_select;
- u32 timeout = 200;
+ u32 timeout;
u32 retry_count = 0;
runtime = substream->runtime;
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 10223e080d59..34fe753a46fb 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -2699,7 +2699,7 @@ struct dsp_image_seg {
u32 magic;
u32 chip_addr;
u32 count;
- u32 data[0];
+ u32 data[];
};
static const u32 g_magic_value = 0x4c46584d;
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 5119a9ae3d8a..bb287a916dae 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -154,7 +154,6 @@ struct hdmi_spec {
struct hda_multi_out multiout;
struct hda_pcm_stream pcm_playback;
- bool use_jack_detect; /* jack detection enabled */
bool use_acomp_notifier; /* use eld_notify callback for hotplug */
bool acomp_registered; /* audio component registered in this driver */
struct drm_audio_component_audio_ops drm_audio_ops;
@@ -753,7 +752,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
* Unsolicited events
*/
-static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll);
+static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll);
static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid,
int dev_id)
@@ -764,8 +763,7 @@ static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid,
if (pin_idx < 0)
return;
mutex_lock(&spec->pcm_lock);
- if (hdmi_present_sense(get_pin(spec, pin_idx), 1))
- snd_hda_jack_report_sync(codec);
+ hdmi_present_sense(get_pin(spec, pin_idx), 1);
mutex_unlock(&spec->pcm_lock);
}
@@ -779,21 +777,9 @@ static void jack_callback(struct hda_codec *codec,
check_presence_and_report(codec, jack->nid, jack->dev_id);
}
-static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
+static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res,
+ struct hda_jack_tbl *jack)
{
- int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
- struct hda_jack_tbl *jack;
-
- if (codec->dp_mst) {
- int dev_entry =
- (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT;
-
- jack = snd_hda_jack_tbl_get_from_tag(codec, tag, dev_entry);
- } else {
- jack = snd_hda_jack_tbl_get_from_tag(codec, tag, 0);
- }
- if (!jack)
- return;
jack->jack_dirty = 1;
codec_dbg(codec,
@@ -853,7 +839,7 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res)
}
if (subtag == 0)
- hdmi_intrinsic_event(codec, res);
+ hdmi_intrinsic_event(codec, res, jack);
else
hdmi_non_intrinsic_event(codec, res);
}
@@ -1480,21 +1466,60 @@ static void hdmi_pcm_reset_pin(struct hdmi_spec *spec,
per_pin->channels = 0;
}
+static struct snd_jack *pin_idx_to_pcm_jack(struct hda_codec *codec,
+ struct hdmi_spec_per_pin *per_pin)
+{
+ struct hdmi_spec *spec = codec->spec;
+
+ if (per_pin->pcm_idx >= 0)
+ return spec->pcm_rec[per_pin->pcm_idx].jack;
+ else
+ return NULL;
+}
+
/* update per_pin ELD from the given new ELD;
* setup info frame and notification accordingly
+ * also notify ELD kctl and report jack status changes
*/
-static bool update_eld(struct hda_codec *codec,
+static void update_eld(struct hda_codec *codec,
struct hdmi_spec_per_pin *per_pin,
- struct hdmi_eld *eld)
+ struct hdmi_eld *eld,
+ int repoll)
{
struct hdmi_eld *pin_eld = &per_pin->sink_eld;
struct hdmi_spec *spec = codec->spec;
+ struct snd_jack *pcm_jack;
bool old_eld_valid = pin_eld->eld_valid;
bool eld_changed;
int pcm_idx;
+ if (eld->eld_valid) {
+ if (eld->eld_size <= 0 ||
+ snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer,
+ eld->eld_size) < 0) {
+ eld->eld_valid = false;
+ if (repoll) {
+ schedule_delayed_work(&per_pin->work,
+ msecs_to_jiffies(300));
+ return;
+ }
+ }
+ }
+
+ if (!eld->eld_valid || eld->eld_size <= 0) {
+ eld->eld_valid = false;
+ eld->eld_size = 0;
+ }
+
/* for monitor disconnection, save pcm_idx firstly */
pcm_idx = per_pin->pcm_idx;
+
+ /*
+ * pcm_idx >=0 before update_eld() means it is in monitor
+ * disconnected event. Jack must be fetched before update_eld().
+ */
+ pcm_jack = pin_idx_to_pcm_jack(codec, per_pin);
+
if (spec->dyn_pcm_assign) {
if (eld->eld_valid) {
hdmi_attach_hda_pcm(spec, per_pin);
@@ -1509,6 +1534,8 @@ static bool update_eld(struct hda_codec *codec,
*/
if (pcm_idx == -1)
pcm_idx = per_pin->pcm_idx;
+ if (!pcm_jack)
+ pcm_jack = pin_idx_to_pcm_jack(codec, per_pin);
if (eld->eld_valid)
snd_hdmi_show_eld(codec, &eld->info);
@@ -1547,42 +1574,17 @@ static bool update_eld(struct hda_codec *codec,
SNDRV_CTL_EVENT_MASK_VALUE |
SNDRV_CTL_EVENT_MASK_INFO,
&get_hdmi_pcm(spec, pcm_idx)->eld_ctl->id);
- return eld_changed;
-}
-static struct snd_jack *pin_idx_to_pcm_jack(struct hda_codec *codec,
- struct hdmi_spec_per_pin *per_pin)
-{
- struct hdmi_spec *spec = codec->spec;
- struct snd_jack *jack = NULL;
- struct hda_jack_tbl *jack_tbl;
-
- /* if !dyn_pcm_assign, get jack from hda_jack_tbl
- * in !dyn_pcm_assign case, spec->pcm_rec[].jack is not
- * NULL even after snd_hda_jack_tbl_clear() is called to
- * free snd_jack. This may cause access invalid memory
- * when calling snd_jack_report
- */
- if (per_pin->pcm_idx >= 0 && spec->dyn_pcm_assign) {
- jack = spec->pcm_rec[per_pin->pcm_idx].jack;
- } else if (!spec->dyn_pcm_assign) {
- /*
- * jack tbl doesn't support DP MST
- * DP MST will use dyn_pcm_assign,
- * so DP MST will never come here
- */
- jack_tbl = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid,
- per_pin->dev_id);
- if (jack_tbl)
- jack = jack_tbl->jack;
- }
- return jack;
+ if (eld_changed && pcm_jack)
+ snd_jack_report(pcm_jack,
+ (eld->monitor_present && eld->eld_valid) ?
+ SND_JACK_AVOUT : 0);
}
+
/* update ELD and jack state via HD-audio verbs */
-static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
+static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
int repoll)
{
- struct hda_jack_tbl *jack;
struct hda_codec *codec = per_pin->codec;
struct hdmi_spec *spec = codec->spec;
struct hdmi_eld *eld = &spec->temp_eld;
@@ -1597,9 +1599,11 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
* the unsolicited response to avoid custom WARs.
*/
int present;
- bool ret;
- bool do_repoll = false;
- struct snd_jack *pcm_jack = NULL;
+ int ret;
+
+ ret = snd_hda_power_up_pm(codec);
+ if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec)))
+ goto out;
present = snd_hda_jack_pin_sense(codec, pin_nid, dev_id);
@@ -1618,62 +1622,12 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
if (spec->ops.pin_get_eld(codec, pin_nid, dev_id,
eld->eld_buffer, &eld->eld_size) < 0)
eld->eld_valid = false;
- else {
- if (snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer,
- eld->eld_size) < 0)
- eld->eld_valid = false;
- }
- if (!eld->eld_valid && repoll)
- do_repoll = true;
}
- if (do_repoll) {
- schedule_delayed_work(&per_pin->work, msecs_to_jiffies(300));
- } else {
- /*
- * pcm_idx >=0 before update_eld() means it is in monitor
- * disconnected event. Jack must be fetched before
- * update_eld().
- */
- pcm_jack = pin_idx_to_pcm_jack(codec, per_pin);
- update_eld(codec, per_pin, eld);
- if (!pcm_jack)
- pcm_jack = pin_idx_to_pcm_jack(codec, per_pin);
- }
-
- ret = !repoll || !eld->monitor_present || eld->eld_valid;
-
- jack = snd_hda_jack_tbl_get_mst(codec, pin_nid, per_pin->dev_id);
- if (jack) {
- jack->block_report = !ret;
- jack->pin_sense = (eld->monitor_present && eld->eld_valid) ?
- AC_PINSENSE_PRESENCE : 0;
-
- if (spec->dyn_pcm_assign && pcm_jack && !do_repoll) {
- int state = 0;
-
- if (jack->pin_sense & AC_PINSENSE_PRESENCE)
- state = SND_JACK_AVOUT;
- snd_jack_report(pcm_jack, state);
- }
-
- /*
- * snd_hda_jack_pin_sense() call at the beginning of this
- * function, updates jack->pins_sense and clears
- * jack->jack_dirty, therefore snd_hda_jack_report_sync() will
- * not override the jack->pin_sense.
- *
- * snd_hda_jack_report_sync() is superfluous for dyn_pcm_assign
- * case. The jack->pin_sense update was already performed, and
- * hda_jack->jack is NULL for dyn_pcm_assign.
- *
- * Don't call snd_hda_jack_report_sync() for
- * dyn_pcm_assign.
- */
- ret = ret && !spec->dyn_pcm_assign;
- }
+ update_eld(codec, per_pin, eld, repoll);
mutex_unlock(&per_pin->lock);
- return ret;
+ out:
+ snd_hda_power_down_pm(codec);
}
/* update ELD and jack state via audio component */
@@ -1682,64 +1636,25 @@ static void sync_eld_via_acomp(struct hda_codec *codec,
{
struct hdmi_spec *spec = codec->spec;
struct hdmi_eld *eld = &spec->temp_eld;
- struct snd_jack *jack = NULL;
- bool changed;
- int size;
mutex_lock(&per_pin->lock);
eld->monitor_present = false;
- size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid,
+ eld->eld_size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid,
per_pin->dev_id, &eld->monitor_present,
eld->eld_buffer, ELD_MAX_SIZE);
- if (size > 0) {
- size = min(size, ELD_MAX_SIZE);
- if (snd_hdmi_parse_eld(codec, &eld->info,
- eld->eld_buffer, size) < 0)
- size = -EINVAL;
- }
-
- if (size > 0) {
- eld->eld_valid = true;
- eld->eld_size = size;
- } else {
- eld->eld_valid = false;
- eld->eld_size = 0;
- }
-
- /* pcm_idx >=0 before update_eld() means it is in monitor
- * disconnected event. Jack must be fetched before update_eld()
- */
- jack = pin_idx_to_pcm_jack(codec, per_pin);
- changed = update_eld(codec, per_pin, eld);
- if (jack == NULL)
- jack = pin_idx_to_pcm_jack(codec, per_pin);
- if (changed && jack)
- snd_jack_report(jack,
- (eld->monitor_present && eld->eld_valid) ?
- SND_JACK_AVOUT : 0);
+ eld->eld_valid = (eld->eld_size > 0);
+ update_eld(codec, per_pin, eld, 0);
mutex_unlock(&per_pin->lock);
}
-static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
+static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
{
struct hda_codec *codec = per_pin->codec;
- int ret;
- /* no temporary power up/down needed for component notifier */
- if (!codec_has_acomp(codec)) {
- ret = snd_hda_power_up_pm(codec);
- if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) {
- snd_hda_power_down_pm(codec);
- return false;
- }
- ret = hdmi_present_sense_via_verbs(per_pin, repoll);
- snd_hda_power_down_pm(codec);
- } else {
+ if (!codec_has_acomp(codec))
+ hdmi_present_sense_via_verbs(per_pin, repoll);
+ else
sync_eld_via_acomp(codec, per_pin);
- ret = false; /* don't call snd_hda_jack_report_sync() */
- }
-
- return ret;
}
static void hdmi_repoll_eld(struct work_struct *work)
@@ -1759,8 +1674,7 @@ static void hdmi_repoll_eld(struct work_struct *work)
per_pin->repoll_count = 0;
mutex_lock(&spec->pcm_lock);
- if (hdmi_present_sense(per_pin, per_pin->repoll_count))
- snd_hda_jack_report_sync(per_pin->codec);
+ hdmi_present_sense(per_pin, per_pin->repoll_count);
mutex_unlock(&spec->pcm_lock);
}
@@ -2206,15 +2120,23 @@ static void free_hdmi_jack_priv(struct snd_jack *jack)
pcm->jack = NULL;
}
-static int add_hdmi_jack_kctl(struct hda_codec *codec,
- struct hdmi_spec *spec,
- int pcm_idx,
- const char *name)
+static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx)
{
+ char hdmi_str[32] = "HDMI/DP";
+ struct hdmi_spec *spec = codec->spec;
+ struct hdmi_spec_per_pin *per_pin = get_pin(spec, pcm_idx);
struct snd_jack *jack;
+ int pcmdev = get_pcm_rec(spec, pcm_idx)->device;
int err;
- err = snd_jack_new(codec->card, name, SND_JACK_AVOUT, &jack,
+ if (pcmdev > 0)
+ sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev);
+ if (!spec->dyn_pcm_assign &&
+ !is_jack_detectable(codec, per_pin->pin_nid))
+ strncat(hdmi_str, " Phantom",
+ sizeof(hdmi_str) - strlen(hdmi_str) - 1);
+
+ err = snd_jack_new(codec->card, hdmi_str, SND_JACK_AVOUT, &jack,
true, false);
if (err < 0)
return err;
@@ -2225,48 +2147,6 @@ static int add_hdmi_jack_kctl(struct hda_codec *codec,
return 0;
}
-static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx)
-{
- char hdmi_str[32] = "HDMI/DP";
- struct hdmi_spec *spec = codec->spec;
- struct hdmi_spec_per_pin *per_pin;
- struct hda_jack_tbl *jack;
- int pcmdev = get_pcm_rec(spec, pcm_idx)->device;
- bool phantom_jack;
- int ret;
-
- if (pcmdev > 0)
- sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev);
-
- if (spec->dyn_pcm_assign)
- return add_hdmi_jack_kctl(codec, spec, pcm_idx, hdmi_str);
-
- /* for !dyn_pcm_assign, we still use hda_jack for compatibility */
- /* if !dyn_pcm_assign, it must be non-MST mode.
- * This means pcms and pins are statically mapped.
- * And pcm_idx is pin_idx.
- */
- per_pin = get_pin(spec, pcm_idx);
- phantom_jack = !is_jack_detectable(codec, per_pin->pin_nid);
- if (phantom_jack)
- strncat(hdmi_str, " Phantom",
- sizeof(hdmi_str) - strlen(hdmi_str) - 1);
- ret = snd_hda_jack_add_kctl_mst(codec, per_pin->pin_nid,
- per_pin->dev_id, hdmi_str, phantom_jack,
- 0, NULL);
- if (ret < 0)
- return ret;
- jack = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid,
- per_pin->dev_id);
- if (jack == NULL)
- return 0;
- /* assign jack->jack to pcm_rec[].jack to
- * align with dyn_pcm_assign mode
- */
- spec->pcm_rec[pcm_idx].jack = jack->jack;
- return 0;
-}
-
static int generic_hdmi_build_controls(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
@@ -2355,7 +2235,6 @@ static int generic_hdmi_init(struct hda_codec *codec)
int pin_idx;
mutex_lock(&spec->bind_lock);
- spec->use_jack_detect = !codec->jackpoll_interval;
for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
hda_nid_t pin_nid = per_pin->pin_nid;
@@ -2365,12 +2244,8 @@ static int generic_hdmi_init(struct hda_codec *codec)
hdmi_init_pin(codec, pin_nid);
if (codec_has_acomp(codec))
continue;
- if (spec->use_jack_detect)
- snd_hda_jack_detect_enable(codec, pin_nid, dev_id);
- else
- snd_hda_jack_detect_enable_callback_mst(codec, pin_nid,
- dev_id,
- jack_callback);
+ snd_hda_jack_detect_enable_callback_mst(codec, pin_nid, dev_id,
+ jack_callback);
}
mutex_unlock(&spec->bind_lock);
return 0;
@@ -2532,12 +2407,6 @@ static void reprogram_jack_detect(struct hda_codec *codec, hda_nid_t nid,
unsigned int val = use_acomp ? 0 : (AC_USRSP_EN | tbl->tag);
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_UNSOLICITED_ENABLE, val);
- } else {
- /* if no jack entry was defined beforehand, create a new one
- * at need (i.e. only when notifier is cleared)
- */
- if (!use_acomp)
- snd_hda_jack_detect_enable(codec, nid, dev_id);
}
}
@@ -2553,13 +2422,11 @@ static void generic_acomp_notifier_set(struct drm_audio_component *acomp,
spec->use_acomp_notifier = use_acomp;
spec->codec->relaxed_resume = use_acomp;
/* reprogram each jack detection logic depending on the notifier */
- if (spec->use_jack_detect) {
- for (i = 0; i < spec->num_pins; i++)
- reprogram_jack_detect(spec->codec,
- get_pin(spec, i)->pin_nid,
- get_pin(spec, i)->dev_id,
- use_acomp);
- }
+ for (i = 0; i < spec->num_pins; i++)
+ reprogram_jack_detect(spec->codec,
+ get_pin(spec, i)->pin_nid,
+ get_pin(spec, i)->dev_id,
+ use_acomp);
mutex_unlock(&spec->bind_lock);
}
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 21ab9cc50c71..65a887b217ee 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -30,7 +30,7 @@
#if K1212_DEBUG_LEVEL > 0
#define K1212_DEBUG_PRINTK(fmt,args...) printk(KERN_DEBUG fmt,##args)
#else
-#define K1212_DEBUG_PRINTK(fmt,...)
+#define K1212_DEBUG_PRINTK(fmt,...) do { } while (0)
#endif
#if K1212_DEBUG_LEVEL > 1
#define K1212_DEBUG_PRINTK_VERBOSE(fmt,args...) printk(KERN_DEBUG fmt,##args)
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index cc06f0a1a7e4..227aece17e39 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -3353,7 +3353,8 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
return;
}
} else {
- int err = -EINVAL;
+ int err;
+
err = hdsp_request_fw_loader(hdsp);
if (err < 0) {
snd_iprintf(buffer,
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 799789c8eea9..8b03e2dc503f 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -414,6 +414,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
{
unsigned int i, idx, ofs, rest;
struct via82xx *chip = snd_pcm_substream_chip(substream);
+ __le32 *pgtbl;
if (dev->table.area == NULL) {
/* the start of each lists must be aligned to 8 bytes,
@@ -435,6 +436,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
/* fill the entries */
idx = 0;
ofs = 0;
+ pgtbl = (__le32 *)dev->table.area;
for (i = 0; i < periods; i++) {
rest = fragsize;
/* fill descriptors for a period.
@@ -451,7 +453,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
return -EINVAL;
}
addr = snd_pcm_sgbuf_get_addr(substream, ofs);
- ((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr);
+ pgtbl[idx << 1] = cpu_to_le32(addr);
r = snd_pcm_sgbuf_get_chunk_size(substream, ofs, rest);
rest -= r;
if (! rest) {
@@ -466,7 +468,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
"tbl %d: at %d size %d (rest %d)\n",
idx, ofs, r, rest);
*/
- ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag);
+ pgtbl[(idx<<1) + 1] = cpu_to_le32(r | flag);
dev->idx_table[idx].offset = ofs;
dev->idx_table[idx].size = r;
ofs += r;
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index 84e589803e2e..607b7100db1c 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -267,6 +267,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
{
unsigned int i, idx, ofs, rest;
struct via82xx_modem *chip = snd_pcm_substream_chip(substream);
+ __le32 *pgtbl;
if (dev->table.area == NULL) {
/* the start of each lists must be aligned to 8 bytes,
@@ -288,6 +289,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
/* fill the entries */
idx = 0;
ofs = 0;
+ pgtbl = (__le32 *)dev->table.area;
for (i = 0; i < periods; i++) {
rest = fragsize;
/* fill descriptors for a period.
@@ -304,7 +306,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
return -EINVAL;
}
addr = snd_pcm_sgbuf_get_addr(substream, ofs);
- ((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr);
+ pgtbl[idx << 1] = cpu_to_le32(addr);
r = PAGE_SIZE - (ofs % PAGE_SIZE);
if (rest < r)
r = rest;
@@ -321,7 +323,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
"tbl %d: at %d size %d (rest %d)\n",
idx, ofs, r, rest);
*/
- ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag);
+ pgtbl[(idx<<1) + 1] = cpu_to_le32(r | flag);
dev->idx_table[idx].offset = ofs;
dev->idx_table[idx].size = r;
ofs += r;
diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c
index 093806d735c6..9554a0c506af 100644
--- a/sound/ppc/keywest.c
+++ b/sound/ppc/keywest.c
@@ -40,6 +40,7 @@ static int keywest_probe(struct i2c_client *client,
static int keywest_attach_adapter(struct i2c_adapter *adapter)
{
struct i2c_board_info info;
+ struct i2c_client *client;
if (! keywest_ctx)
return -EINVAL;
@@ -50,9 +51,11 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter)
memset(&info, 0, sizeof(struct i2c_board_info));
strlcpy(info.type, "keywest", I2C_NAME_SIZE);
info.addr = keywest_ctx->addr;
- keywest_ctx->client = i2c_new_device(adapter, &info);
- if (!keywest_ctx->client)
- return -ENODEV;
+ client = i2c_new_client_device(adapter, &info);
+ if (IS_ERR(client))
+ return PTR_ERR(client);
+ keywest_ctx->client = client;
+
/*
* We know the driver is already loaded, so the device should be
* already bound. If not it means binding failed, and then there
diff --git a/sound/usb/Makefile b/sound/usb/Makefile
index 78edd7d2f418..56031026b113 100644
--- a/sound/usb/Makefile
+++ b/sound/usb/Makefile
@@ -13,6 +13,7 @@ snd-usb-audio-objs := card.o \
mixer_scarlett.o \
mixer_scarlett_gen2.o \
mixer_us16x08.o \
+ mixer_s1810c.o \
pcm.o \
power.o \
proc.o \
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 827fb0bc8b56..fd6fd1726ea0 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -72,6 +72,7 @@ static int device_setup[SNDRV_CARDS]; /* device parameter for this card */
static bool ignore_ctl_error;
static bool autoclock = true;
static char *quirk_alias[SNDRV_CARDS];
+static char *delayed_register[SNDRV_CARDS];
bool snd_usb_use_vmalloc = true;
bool snd_usb_skip_validation;
@@ -95,6 +96,8 @@ module_param(autoclock, bool, 0444);
MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes).");
module_param_array(quirk_alias, charp, NULL, 0444);
MODULE_PARM_DESC(quirk_alias, "Quirk aliases, e.g. 0123abcd:5678beef.");
+module_param_array(delayed_register, charp, NULL, 0444);
+MODULE_PARM_DESC(delayed_register, "Quirk for delayed registration, given by id:iface, e.g. 0123abcd:4.");
module_param_named(use_vmalloc, snd_usb_use_vmalloc, bool, 0444);
MODULE_PARM_DESC(use_vmalloc, "Use vmalloc for PCM intermediate buffers (default: yes).");
module_param_named(skip_validation, snd_usb_skip_validation, bool, 0444);
@@ -525,6 +528,21 @@ static bool get_alias_id(struct usb_device *dev, unsigned int *id)
return false;
}
+static bool check_delayed_register_option(struct snd_usb_audio *chip, int iface)
+{
+ int i;
+ unsigned int id, inum;
+
+ for (i = 0; i < ARRAY_SIZE(delayed_register); i++) {
+ if (delayed_register[i] &&
+ sscanf(delayed_register[i], "%x:%x", &id, &inum) == 2 &&
+ id == chip->usb_id)
+ return inum != iface;
+ }
+
+ return false;
+}
+
static const struct usb_device_id usb_audio_ids[]; /* defined below */
/* look for the corresponding quirk */
@@ -662,10 +680,22 @@ static int usb_audio_probe(struct usb_interface *intf,
goto __error;
}
- /* we are allowed to call snd_card_register() many times */
- err = snd_card_register(chip->card);
- if (err < 0)
- goto __error;
+ if (chip->need_delayed_register) {
+ dev_info(&dev->dev,
+ "Found post-registration device assignment: %08x:%02x\n",
+ chip->usb_id, ifnum);
+ chip->need_delayed_register = false; /* clear again */
+ }
+
+ /* we are allowed to call snd_card_register() many times, but first
+ * check to see if a device needs to skip it or do anything special
+ */
+ if (!snd_usb_registration_quirk(chip, ifnum) &&
+ !check_delayed_register_option(chip, ifnum)) {
+ err = snd_card_register(chip->card);
+ if (err < 0)
+ goto __error;
+ }
if (quirk && quirk->shares_media_device) {
/* don't want to fail when snd_media_device_create() fails */
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index a48313dfa967..b118cf97607f 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -151,16 +151,15 @@ static int uac_clock_selector_set_val(struct snd_usb_audio *chip, int selector_i
return ret;
}
-/*
- * Assume the clock is valid if clock source supports only one single sample
- * rate, the terminal is connected directly to it (there is no clock selector)
- * and clock type is internal. This is to deal with some Denon DJ controllers
- * that always reports that clock is invalid.
- */
static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip,
struct audioformat *fmt,
int source_id)
{
+ bool ret = false;
+ int count;
+ unsigned char data;
+ struct usb_device *dev = chip->dev;
+
if (fmt->protocol == UAC_VERSION_2) {
struct uac_clock_source_descriptor *cs_desc =
snd_usb_find_clock_source(chip->ctrl_intf, source_id);
@@ -168,13 +167,51 @@ static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip,
if (!cs_desc)
return false;
- return (fmt->nr_rates == 1 &&
- (fmt->clock & 0xff) == cs_desc->bClockID &&
- (cs_desc->bmAttributes & 0x3) !=
- UAC_CLOCK_SOURCE_TYPE_EXT);
+ /*
+ * Assume the clock is valid if clock source supports only one
+ * single sample rate, the terminal is connected directly to it
+ * (there is no clock selector) and clock type is internal.
+ * This is to deal with some Denon DJ controllers that always
+ * reports that clock is invalid.
+ */
+ if (fmt->nr_rates == 1 &&
+ (fmt->clock & 0xff) == cs_desc->bClockID &&
+ (cs_desc->bmAttributes & 0x3) !=
+ UAC_CLOCK_SOURCE_TYPE_EXT)
+ return true;
+ }
+
+ /*
+ * MOTU MicroBook IIc
+ * Sample rate changes takes more than 2 seconds for this device. Clock
+ * validity request returns false during that period.
+ */
+ if (chip->usb_id == USB_ID(0x07fd, 0x0004)) {
+ count = 0;
+
+ while ((!ret) && (count < 50)) {
+ int err;
+
+ msleep(100);
+
+ err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+ USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+ UAC2_CS_CONTROL_CLOCK_VALID << 8,
+ snd_usb_ctrl_intf(chip) | (source_id << 8),
+ &data, sizeof(data));
+ if (err < 0) {
+ dev_warn(&dev->dev,
+ "%s(): cannot get clock validity for id %d\n",
+ __func__, source_id);
+ return false;
+ }
+
+ ret = !!data;
+ count++;
+ }
}
- return false;
+ return ret;
}
static bool uac_clock_source_is_valid(struct snd_usb_audio *chip,
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 9f5cb4ed3a0c..50e1874c847c 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -247,6 +247,36 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
return 0;
}
+
+/*
+ * Presonus Studio 1810c supports a limited set of sampling
+ * rates per altsetting but reports the full set each time.
+ * If we don't filter out the unsupported rates and attempt
+ * to configure the card, it will hang refusing to do any
+ * further audio I/O until a hard reset is performed.
+ *
+ * The list of supported rates per altsetting (set of available
+ * I/O channels) is described in the owner's manual, section 2.2.
+ */
+static bool s1810c_valid_sample_rate(struct audioformat *fp,
+ unsigned int rate)
+{
+ switch (fp->altsetting) {
+ case 1:
+ /* All ADAT ports available */
+ return rate <= 48000;
+ case 2:
+ /* Half of ADAT ports available */
+ return (rate == 88200 || rate == 96000);
+ case 3:
+ /* Analog I/O only (no S/PDIF nor ADAT) */
+ return rate >= 176400;
+ default:
+ return false;
+ }
+ return false;
+}
+
/*
* Helper function to walk the array of sample rate triplets reported by
* the device. The problem is that we need to parse whole array first to
@@ -283,6 +313,12 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip,
}
for (rate = min; rate <= max; rate += res) {
+
+ /* Filter out invalid rates on Presonus Studio 1810c */
+ if (chip->usb_id == USB_ID(0x0194f, 0x010c) &&
+ !s1810c_valid_sample_rate(fp, rate))
+ goto skip_rate;
+
if (fp->rate_table)
fp->rate_table[nr_rates] = rate;
if (!fp->rate_min || rate < fp->rate_min)
@@ -297,6 +333,7 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip,
break;
}
+skip_rate:
/* avoid endless loop */
if (res == 0)
break;
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 392e5fda680c..be5c597ed40c 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -91,7 +91,7 @@ struct usb_ms_endpoint_descriptor {
__u8 bDescriptorType;
__u8 bDescriptorSubtype;
__u8 bNumEmbMIDIJack;
- __u8 baAssocJackID[0];
+ __u8 baAssocJackID[];
} __attribute__ ((packed));
struct snd_usb_midi_in_endpoint;
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 81b2db0edd5f..721d12130d0c 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -292,6 +292,11 @@ static int uac2_ctl_value_size(int val_type)
* retrieve a mixer value
*/
+static inline int mixer_ctrl_intf(struct usb_mixer_interface *mixer)
+{
+ return get_iface_desc(mixer->hostif)->bInterfaceNumber;
+}
+
static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request,
int validx, int *value_ret)
{
@@ -306,7 +311,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request,
return -EIO;
while (timeout-- > 0) {
- idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8);
+ idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8);
err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request,
USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
validx, idx, buf, val_len);
@@ -354,7 +359,7 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request,
if (ret)
goto error;
- idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8);
+ idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8);
ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest,
USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
validx, idx, buf, size);
@@ -479,7 +484,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval,
return -EIO;
while (timeout-- > 0) {
- idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8);
+ idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8);
err = snd_usb_ctl_msg(chip->dev,
usb_sndctrlpipe(chip->dev, 0), request,
USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
@@ -901,6 +906,12 @@ static int parse_term_effect_unit(struct mixer_build *state,
struct usb_audio_term *term,
void *p1, int id)
{
+ struct uac2_effect_unit_descriptor *d = p1;
+ int err;
+
+ err = __check_input_term(state, d->bSourceID, term);
+ if (err < 0)
+ return err;
term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */
term->id = id;
return 0;
@@ -1203,7 +1214,7 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval,
get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) {
usb_audio_err(cval->head.mixer->chip,
"%d:%d: cannot get min/max values for control %d (id %d)\n",
- cval->head.id, snd_usb_ctrl_intf(cval->head.mixer->chip),
+ cval->head.id, mixer_ctrl_intf(cval->head.mixer),
cval->control, cval->head.id);
return -EINVAL;
}
@@ -1422,7 +1433,7 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol,
if (ret)
goto error;
- idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8);
+ idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8);
if (cval->head.mixer->protocol == UAC_VERSION_2) {
struct uac2_connectors_ctl_blk uac2_conn;
@@ -1674,6 +1685,16 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer,
/* get min/max values */
get_min_max_with_quirks(cval, 0, kctl);
+ /* skip a bogus volume range */
+ if (cval->max <= cval->min) {
+ usb_audio_dbg(mixer->chip,
+ "[%d] FU [%s] skipped due to invalid volume\n",
+ cval->head.id, kctl->id.name);
+ snd_ctl_free_one(kctl);
+ return;
+ }
+
+
if (control == UAC_FU_VOLUME) {
check_mapped_dB(map, cval);
if (cval->dBmin < cval->dBmax || !cval->initialized) {
@@ -3203,7 +3224,7 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry,
list_for_each_entry(mixer, &chip->mixer_list, list) {
snd_iprintf(buffer,
"USB Mixer: usb_id=0x%08x, ctrlif=%i, ctlerr=%i\n",
- chip->usb_id, snd_usb_ctrl_intf(chip),
+ chip->usb_id, mixer_ctrl_intf(mixer),
mixer->ignore_ctl_error);
snd_iprintf(buffer, "Card: %s\n", chip->card->longname);
for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) {
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index c237e24f08d9..02b036b2aefb 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -34,6 +34,7 @@
#include "mixer_scarlett.h"
#include "mixer_scarlett_gen2.h"
#include "mixer_us16x08.h"
+#include "mixer_s1810c.h"
#include "helper.h"
struct std_mono_table {
@@ -2277,6 +2278,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
case USB_ID(0x2a39, 0x3fd4): /* RME */
err = snd_rme_controls_create(mixer);
break;
+
+ case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */
+ err = snd_sc1810_init_mixer(mixer);
+ break;
}
return err;
diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c
new file mode 100644
index 000000000000..6483e47bafd0
--- /dev/null
+++ b/sound/usb/mixer_s1810c.c
@@ -0,0 +1,595 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * Presonus Studio 1810c driver for ALSA
+ * Copyright (C) 2019 Nick Kossifidis <mickflemm@gmail.com>
+ *
+ * Based on reverse engineering of the communication protocol
+ * between the windows driver / Univeral Control (UC) program
+ * and the device, through usbmon.
+ *
+ * For now this bypasses the mixer, with all channels split,
+ * so that the software can mix with greater flexibility.
+ * It also adds controls for the 4 buttons on the front of
+ * the device.
+ */
+
+#include <linux/usb.h>
+#include <linux/usb/audio-v2.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/control.h>
+
+#include "usbaudio.h"
+#include "mixer.h"
+#include "mixer_quirks.h"
+#include "helper.h"
+#include "mixer_s1810c.h"
+
+#define SC1810C_CMD_REQ 160
+#define SC1810C_CMD_REQTYPE \
+ (USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT)
+#define SC1810C_CMD_F1 0x50617269
+#define SC1810C_CMD_F2 0x14
+
+/*
+ * DISCLAIMER: These are just guesses based on the
+ * dumps I got.
+ *
+ * It seems like a selects between
+ * device (0), mixer (0x64) and output (0x65)
+ *
+ * For mixer (0x64):
+ * * b selects an input channel (see below).
+ * * c selects an output channel pair (see below).
+ * * d selects left (0) or right (1) of that pair.
+ * * e 0-> disconnect, 0x01000000-> connect,
+ * 0x0109-> used for stereo-linking channels,
+ * e is also used for setting volume levels
+ * in which case b is also set so I guess
+ * this way it is possible to set the volume
+ * level from the specified input to the
+ * specified output.
+ *
+ * IN Channels:
+ * 0 - 7 Mic/Inst/Line (Analog inputs)
+ * 8 - 9 S/PDIF
+ * 10 - 17 ADAT
+ * 18 - 35 DAW (Inputs from the host)
+ *
+ * OUT Channels (pairs):
+ * 0 -> Main out
+ * 1 -> Line1/2
+ * 2 -> Line3/4
+ * 3 -> S/PDIF
+ * 4 -> ADAT?
+ *
+ * For device (0):
+ * * b and c are not used, at least not on the
+ * dumps I got.
+ * * d sets the control id to be modified
+ * (see below).
+ * * e sets the setting for that control.
+ * (so for the switches I was interested
+ * in it's 0/1)
+ *
+ * For output (0x65):
+ * * b is the output channel (see above).
+ * * c is zero.
+ * * e I guess the same as with mixer except 0x0109
+ * which I didn't see in my dumps.
+ *
+ * The two fixed fields have the same values for
+ * mixer and output but a different set for device.
+ */
+struct s1810c_ctl_packet {
+ u32 a;
+ u32 b;
+ u32 fixed1;
+ u32 fixed2;
+ u32 c;
+ u32 d;
+ u32 e;
+};
+
+#define SC1810C_CTL_LINE_SW 0
+#define SC1810C_CTL_MUTE_SW 1
+#define SC1810C_CTL_AB_SW 3
+#define SC1810C_CTL_48V_SW 4
+
+#define SC1810C_SET_STATE_REQ 161
+#define SC1810C_SET_STATE_REQTYPE SC1810C_CMD_REQTYPE
+#define SC1810C_SET_STATE_F1 0x64656D73
+#define SC1810C_SET_STATE_F2 0xF4
+
+#define SC1810C_GET_STATE_REQ 162
+#define SC1810C_GET_STATE_REQTYPE \
+ (USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN)
+#define SC1810C_GET_STATE_F1 SC1810C_SET_STATE_F1
+#define SC1810C_GET_STATE_F2 SC1810C_SET_STATE_F2
+
+#define SC1810C_STATE_F1_IDX 2
+#define SC1810C_STATE_F2_IDX 3
+
+/*
+ * This packet includes mixer volumes and
+ * various other fields, it's an extended
+ * version of ctl_packet, with a and b
+ * being zero and different f1/f2.
+ */
+struct s1810c_state_packet {
+ u32 fields[63];
+};
+
+#define SC1810C_STATE_48V_SW 58
+#define SC1810C_STATE_LINE_SW 59
+#define SC1810C_STATE_MUTE_SW 60
+#define SC1810C_STATE_AB_SW 62
+
+struct s1810_mixer_state {
+ uint16_t seqnum;
+ struct mutex usb_mutex;
+ struct mutex data_mutex;
+};
+
+static int
+snd_s1810c_send_ctl_packet(struct usb_device *dev, u32 a,
+ u32 b, u32 c, u32 d, u32 e)
+{
+ struct s1810c_ctl_packet pkt = { 0 };
+ int ret = 0;
+
+ pkt.fixed1 = SC1810C_CMD_F1;
+ pkt.fixed2 = SC1810C_CMD_F2;
+
+ pkt.a = a;
+ pkt.b = b;
+ pkt.c = c;
+ pkt.d = d;
+ /*
+ * Value for settings 0/1 for this
+ * output channel is always 0 (probably because
+ * there is no ADAT output on 1810c)
+ */
+ pkt.e = (c == 4) ? 0 : e;
+
+ ret = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0),
+ SC1810C_CMD_REQ,
+ SC1810C_CMD_REQTYPE, 0, 0, &pkt, sizeof(pkt));
+ if (ret < 0) {
+ dev_warn(&dev->dev, "could not send ctl packet\n");
+ return ret;
+ }
+ return 0;
+}
+
+/*
+ * When opening Universal Control the program periodicaly
+ * sends and receives state packets for syncinc state between
+ * the device and the host.
+ *
+ * Note that if we send only the request to get data back we'll
+ * get an error, we need to first send an empty state packet and
+ * then ask to receive a filled. Their seqnumbers must also match.
+ */
+static int
+snd_sc1810c_get_status_field(struct usb_device *dev,
+ u32 *field, int field_idx, uint16_t *seqnum)
+{
+ struct s1810c_state_packet pkt_out = { { 0 } };
+ struct s1810c_state_packet pkt_in = { { 0 } };
+ int ret = 0;
+
+ pkt_out.fields[SC1810C_STATE_F1_IDX] = SC1810C_SET_STATE_F1;
+ pkt_out.fields[SC1810C_STATE_F2_IDX] = SC1810C_SET_STATE_F2;
+ ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
+ SC1810C_SET_STATE_REQ,
+ SC1810C_SET_STATE_REQTYPE,
+ (*seqnum), 0, &pkt_out, sizeof(pkt_out));
+ if (ret < 0) {
+ dev_warn(&dev->dev, "could not send state packet (%d)\n", ret);
+ return ret;
+ }
+
+ ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
+ SC1810C_GET_STATE_REQ,
+ SC1810C_GET_STATE_REQTYPE,
+ (*seqnum), 0, &pkt_in, sizeof(pkt_in));
+ if (ret < 0) {
+ dev_warn(&dev->dev, "could not get state field %u (%d)\n",
+ field_idx, ret);
+ return ret;
+ }
+
+ (*field) = pkt_in.fields[field_idx];
+ (*seqnum)++;
+ return 0;
+}
+
+/*
+ * This is what I got when bypassing the mixer with
+ * all channels split. I'm not 100% sure of what's going
+ * on, I could probably clean this up based on my observations
+ * but I prefer to keep the same behavior as the windows driver.
+ */
+static int snd_s1810c_init_mixer_maps(struct snd_usb_audio *chip)
+{
+ u32 a, b, c, e, n, off;
+ struct usb_device *dev = chip->dev;
+
+ /* Set initial volume levels ? */
+ a = 0x64;
+ e = 0xbc;
+ for (n = 0; n < 2; n++) {
+ off = n * 18;
+ for (b = off, c = 0; b < 18 + off; b++) {
+ /* This channel to all outputs ? */
+ for (c = 0; c <= 8; c++) {
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e);
+ }
+ /* This channel to main output (again) */
+ snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e);
+ }
+ /*
+ * I noticed on UC that DAW channels have different
+ * initial volumes, so this makes sense.
+ */
+ e = 0xb53bf0;
+ }
+
+ /* Connect analog outputs ? */
+ a = 0x65;
+ e = 0x01000000;
+ for (b = 1; b < 3; b++) {
+ snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e);
+ }
+ snd_s1810c_send_ctl_packet(dev, a, 0, 0, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, 0, 0, 1, e);
+
+ /* Set initial volume levels for S/PDIF mappings ? */
+ a = 0x64;
+ e = 0xbc;
+ c = 3;
+ for (n = 0; n < 2; n++) {
+ off = n * 18;
+ for (b = off; b < 18 + off; b++) {
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e);
+ }
+ e = 0xb53bf0;
+ }
+
+ /* Connect S/PDIF output ? */
+ a = 0x65;
+ e = 0x01000000;
+ snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e);
+
+ /* Connect all outputs (again) ? */
+ a = 0x65;
+ e = 0x01000000;
+ for (b = 0; b < 4; b++) {
+ snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e);
+ }
+
+ /* Basic routing to get sound out of the device */
+ a = 0x64;
+ e = 0x01000000;
+ for (c = 0; c < 4; c++) {
+ for (b = 0; b < 36; b++) {
+ if ((c == 0 && b == 18) || /* DAW1/2 -> Main */
+ (c == 1 && b == 20) || /* DAW3/4 -> Line3/4 */
+ (c == 2 && b == 22) || /* DAW4/5 -> Line5/6 */
+ (c == 3 && b == 24)) { /* DAW5/6 -> S/PDIF */
+ /* Left */
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0);
+ b++;
+ /* Right */
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0);
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e);
+ } else {
+ /* Leave the rest disconnected */
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0);
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0);
+ }
+ }
+ }
+
+ /* Set initial volume levels for S/PDIF (again) ? */
+ a = 0x64;
+ e = 0xbc;
+ c = 3;
+ for (n = 0; n < 2; n++) {
+ off = n * 18;
+ for (b = off; b < 18 + off; b++) {
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e);
+ }
+ e = 0xb53bf0;
+ }
+
+ /* Connect S/PDIF outputs (again) ? */
+ a = 0x65;
+ e = 0x01000000;
+ snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e);
+
+ /* Again ? */
+ snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e);
+
+ return 0;
+}
+
+/*
+ * Sync state with the device and retrieve the requested field,
+ * whose index is specified in (kctl->private_value & 0xFF),
+ * from the received fields array.
+ */
+static int
+snd_s1810c_get_switch_state(struct usb_mixer_interface *mixer,
+ struct snd_kcontrol *kctl, u32 *state)
+{
+ struct snd_usb_audio *chip = mixer->chip;
+ struct s1810_mixer_state *private = mixer->private_data;
+ u32 field = 0;
+ u32 ctl_idx = (u32) (kctl->private_value & 0xFF);
+ int ret = 0;
+
+ mutex_lock(&private->usb_mutex);
+ ret = snd_sc1810c_get_status_field(chip->dev, &field,
+ ctl_idx, &private->seqnum);
+ if (ret < 0)
+ goto unlock;
+
+ *state = field;
+ unlock:
+ mutex_unlock(&private->usb_mutex);
+ return ret ? ret : 0;
+}
+
+/*
+ * Send a control packet to the device for the control id
+ * specified in (kctl->private_value >> 8) with value
+ * specified in (kctl->private_value >> 16).
+ */
+static int
+snd_s1810c_set_switch_state(struct usb_mixer_interface *mixer,
+ struct snd_kcontrol *kctl)
+{
+ struct snd_usb_audio *chip = mixer->chip;
+ struct s1810_mixer_state *private = mixer->private_data;
+ u32 pval = (u32) kctl->private_value;
+ u32 ctl_id = (pval >> 8) & 0xFF;
+ u32 ctl_val = (pval >> 16) & 0x1;
+ int ret = 0;
+
+ mutex_lock(&private->usb_mutex);
+ ret = snd_s1810c_send_ctl_packet(chip->dev, 0, 0, 0, ctl_id, ctl_val);
+ mutex_unlock(&private->usb_mutex);
+ return ret;
+}
+
+/* Generic get/set/init functions for switch controls */
+
+static int
+snd_s1810c_switch_get(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ctl_elem)
+{
+ struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl);
+ struct usb_mixer_interface *mixer = list->mixer;
+ struct s1810_mixer_state *private = mixer->private_data;
+ u32 pval = (u32) kctl->private_value;
+ u32 ctl_idx = pval & 0xFF;
+ u32 state = 0;
+ int ret = 0;
+
+ mutex_lock(&private->data_mutex);
+ ret = snd_s1810c_get_switch_state(mixer, kctl, &state);
+ if (ret < 0)
+ goto unlock;
+
+ switch (ctl_idx) {
+ case SC1810C_STATE_LINE_SW:
+ case SC1810C_STATE_AB_SW:
+ ctl_elem->value.enumerated.item[0] = (int)state;
+ break;
+ default:
+ ctl_elem->value.integer.value[0] = (long)state;
+ }
+
+ unlock:
+ mutex_unlock(&private->data_mutex);
+ return (ret < 0) ? ret : 0;
+}
+
+static int
+snd_s1810c_switch_set(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ctl_elem)
+{
+ struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl);
+ struct usb_mixer_interface *mixer = list->mixer;
+ struct s1810_mixer_state *private = mixer->private_data;
+ u32 pval = (u32) kctl->private_value;
+ u32 ctl_idx = pval & 0xFF;
+ u32 curval = 0;
+ u32 newval = 0;
+ int ret = 0;
+
+ mutex_lock(&private->data_mutex);
+ ret = snd_s1810c_get_switch_state(mixer, kctl, &curval);
+ if (ret < 0)
+ goto unlock;
+
+ switch (ctl_idx) {
+ case SC1810C_STATE_LINE_SW:
+ case SC1810C_STATE_AB_SW:
+ newval = (u32) ctl_elem->value.enumerated.item[0];
+ break;
+ default:
+ newval = (u32) ctl_elem->value.integer.value[0];
+ }
+
+ if (curval == newval)
+ goto unlock;
+
+ kctl->private_value &= ~(0x1 << 16);
+ kctl->private_value |= (unsigned int)(newval & 0x1) << 16;
+ ret = snd_s1810c_set_switch_state(mixer, kctl);
+
+ unlock:
+ mutex_unlock(&private->data_mutex);
+ return (ret < 0) ? 0 : 1;
+}
+
+static int
+snd_s1810c_switch_init(struct usb_mixer_interface *mixer,
+ const struct snd_kcontrol_new *new_kctl)
+{
+ struct snd_kcontrol *kctl;
+ struct usb_mixer_elem_info *elem;
+
+ elem = kzalloc(sizeof(struct usb_mixer_elem_info), GFP_KERNEL);
+ if (!elem)
+ return -ENOMEM;
+
+ elem->head.mixer = mixer;
+ elem->control = 0;
+ elem->head.id = 0;
+ elem->channels = 1;
+
+ kctl = snd_ctl_new1(new_kctl, elem);
+ if (!kctl) {
+ kfree(elem);
+ return -ENOMEM;
+ }
+ kctl->private_free = snd_usb_mixer_elem_free;
+
+ return snd_usb_mixer_add_control(&elem->head, kctl);
+}
+
+static int
+snd_s1810c_line_sw_info(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char *const texts[2] = {
+ "Preamp On (Mic/Inst)",
+ "Preamp Off (Line in)"
+ };
+
+ return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts);
+}
+
+static const struct snd_kcontrol_new snd_s1810c_line_sw = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Line 1/2 Source Type",
+ .info = snd_s1810c_line_sw_info,
+ .get = snd_s1810c_switch_get,
+ .put = snd_s1810c_switch_set,
+ .private_value = (SC1810C_STATE_LINE_SW | SC1810C_CTL_LINE_SW << 8)
+};
+
+static const struct snd_kcontrol_new snd_s1810c_mute_sw = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Mute Main Out Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = snd_s1810c_switch_get,
+ .put = snd_s1810c_switch_set,
+ .private_value = (SC1810C_STATE_MUTE_SW | SC1810C_CTL_MUTE_SW << 8)
+};
+
+static const struct snd_kcontrol_new snd_s1810c_48v_sw = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "48V Phantom Power On Mic Inputs Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = snd_s1810c_switch_get,
+ .put = snd_s1810c_switch_set,
+ .private_value = (SC1810C_STATE_48V_SW | SC1810C_CTL_48V_SW << 8)
+};
+
+static int
+snd_s1810c_ab_sw_info(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char *const texts[2] = {
+ "1/2",
+ "3/4"
+ };
+
+ return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts);
+}
+
+static const struct snd_kcontrol_new snd_s1810c_ab_sw = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Headphone 1 Source Route",
+ .info = snd_s1810c_ab_sw_info,
+ .get = snd_s1810c_switch_get,
+ .put = snd_s1810c_switch_set,
+ .private_value = (SC1810C_STATE_AB_SW | SC1810C_CTL_AB_SW << 8)
+};
+
+static void snd_sc1810_mixer_state_free(struct usb_mixer_interface *mixer)
+{
+ struct s1810_mixer_state *private = mixer->private_data;
+ kfree(private);
+ mixer->private_data = NULL;
+}
+
+/* Entry point, called from mixer_quirks.c */
+int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer)
+{
+ struct s1810_mixer_state *private = NULL;
+ struct snd_usb_audio *chip = mixer->chip;
+ struct usb_device *dev = chip->dev;
+ int ret = 0;
+
+ /* Run this only once */
+ if (!list_empty(&chip->mixer_list))
+ return 0;
+
+ dev_info(&dev->dev,
+ "Presonus Studio 1810c, device_setup: %u\n", chip->setup);
+ if (chip->setup == 1)
+ dev_info(&dev->dev, "(8out/18in @ 48KHz)\n");
+ else if (chip->setup == 2)
+ dev_info(&dev->dev, "(6out/8in @ 192KHz)\n");
+ else
+ dev_info(&dev->dev, "(8out/14in @ 96KHz)\n");
+
+ ret = snd_s1810c_init_mixer_maps(chip);
+ if (ret < 0)
+ return ret;
+
+ private = kzalloc(sizeof(struct s1810_mixer_state), GFP_KERNEL);
+ if (!private)
+ return -ENOMEM;
+
+ mutex_init(&private->usb_mutex);
+ mutex_init(&private->data_mutex);
+
+ mixer->private_data = private;
+ mixer->private_free = snd_sc1810_mixer_state_free;
+
+ private->seqnum = 1;
+
+ ret = snd_s1810c_switch_init(mixer, &snd_s1810c_line_sw);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_s1810c_switch_init(mixer, &snd_s1810c_mute_sw);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_s1810c_switch_init(mixer, &snd_s1810c_48v_sw);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw);
+ if (ret < 0)
+ return ret;
+ return ret;
+}
diff --git a/sound/usb/mixer_s1810c.h b/sound/usb/mixer_s1810c.h
new file mode 100644
index 000000000000..a79a3743cff3
--- /dev/null
+++ b/sound/usb/mixer_s1810c.h
@@ -0,0 +1,7 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Presonus Studio 1810c driver for ALSA
+ * Copyright (C) 2019 Nick Kossifidis <mickflemm@gmail.com>
+ */
+
+int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer);
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index bd258f1ec2dd..a4e4064f9aee 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -357,7 +357,12 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
ep = 0x81;
ifnum = 1;
goto add_sync_ep_from_ifnum;
- case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */
+ case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II/IIc */
+ /* MicroBook IIc */
+ if (altsd->bInterfaceClass == USB_CLASS_AUDIO)
+ return 0;
+
+ /* MicroBook II */
ep = 0x84;
ifnum = 0;
goto add_sync_ep_from_ifnum;
diff --git a/sound/usb/proc.c b/sound/usb/proc.c
index ffbf4bd9208c..4174ad11fca6 100644
--- a/sound/usb/proc.c
+++ b/sound/usb/proc.c
@@ -70,7 +70,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s
snd_iprintf(buffer, " Interface %d\n", fp->iface);
snd_iprintf(buffer, " Altset %d\n", fp->altsetting);
snd_iprintf(buffer, " Format:");
- for (fmt = 0; fmt <= SNDRV_PCM_FORMAT_LAST; ++fmt)
+ pcm_for_each_format(fmt)
if (fp->formats & pcm_format_to_bits(fmt))
snd_iprintf(buffer, " %s",
snd_pcm_format_name(fmt));
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index d187aa6d50db..1c8719292eee 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3472,7 +3472,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
},
/* MOTU Microbook II */
{
- USB_DEVICE(0x07fd, 0x0004),
+ USB_DEVICE_VENDOR_SPEC(0x07fd, 0x0004),
.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
.vendor_name = "MOTU",
.product_name = "MicroBookII",
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 7f558f4b4520..86f192a3043d 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1252,6 +1252,38 @@ static int fasttrackpro_skip_setting_quirk(struct snd_usb_audio *chip,
return 0; /* keep this altsetting */
}
+static int s1810c_skip_setting_quirk(struct snd_usb_audio *chip,
+ int iface, int altno)
+{
+ /*
+ * Altno settings:
+ *
+ * Playback (Interface 1):
+ * 1: 6 Analog + 2 S/PDIF
+ * 2: 6 Analog + 2 S/PDIF
+ * 3: 6 Analog
+ *
+ * Capture (Interface 2):
+ * 1: 8 Analog + 2 S/PDIF + 8 ADAT
+ * 2: 8 Analog + 2 S/PDIF + 4 ADAT
+ * 3: 8 Analog
+ */
+
+ /*
+ * I'll leave 2 as the default one and
+ * use device_setup to switch to the
+ * other two.
+ */
+ if ((chip->setup == 0 || chip->setup > 2) && altno != 2)
+ return 1;
+ else if (chip->setup == 1 && altno != 1)
+ return 1;
+ else if (chip->setup == 2 && altno != 3)
+ return 1;
+
+ return 0;
+}
+
int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip,
int iface,
int altno)
@@ -1265,6 +1297,10 @@ int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip,
/* fasttrackpro usb: skip altsets incompatible with device_setup */
if (chip->usb_id == USB_ID(0x0763, 0x2012))
return fasttrackpro_skip_setting_quirk(chip, iface, altno);
+ /* presonus studio 1810c: skip altsets incompatible with device_setup */
+ if (chip->usb_id == USB_ID(0x0194f, 0x010c))
+ return s1810c_skip_setting_quirk(chip, iface, altno);
+
return 0;
}
@@ -1316,7 +1352,15 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev,
case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */
return snd_usb_axefx3_boot_quirk(dev);
case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */
- return snd_usb_motu_microbookii_boot_quirk(dev);
+ /*
+ * For some reason interface 3 with vendor-spec class is
+ * detected on MicroBook IIc.
+ */
+ if (get_iface_desc(intf->altsetting)->bInterfaceClass ==
+ USB_CLASS_VENDOR_SPEC &&
+ get_iface_desc(intf->altsetting)->bInterfaceNumber < 3)
+ return snd_usb_motu_microbookii_boot_quirk(dev);
+ break;
}
return 0;
@@ -1754,5 +1798,47 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip,
else
fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC;
break;
+ case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook IIc */
+ /*
+ * MaxPacketsOnly attribute is erroneously set in endpoint
+ * descriptors. As a result this card produces noise with
+ * all sample rates other than 96 KHz.
+ */
+ fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX;
+ break;
}
}
+
+/*
+ * registration quirk:
+ * the registration is skipped if a device matches with the given ID,
+ * unless the interface reaches to the defined one. This is for delaying
+ * the registration until the last known interface, so that the card and
+ * devices appear at the same time.
+ */
+
+struct registration_quirk {
+ unsigned int usb_id; /* composed via USB_ID() */
+ unsigned int interface; /* the interface to trigger register */
+};
+
+#define REG_QUIRK_ENTRY(vendor, product, iface) \
+ { .usb_id = USB_ID(vendor, product), .interface = (iface) }
+
+static const struct registration_quirk registration_quirks[] = {
+ REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */
+ { 0 } /* terminator */
+};
+
+/* return true if skipping registration */
+bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface)
+{
+ const struct registration_quirk *q;
+
+ for (q = registration_quirks; q->usb_id; q++)
+ if (chip->usb_id == q->usb_id)
+ return iface != q->interface;
+
+ /* Register as normal */
+ return false;
+}
diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h
index df0355843a4c..c76cf24a640a 100644
--- a/sound/usb/quirks.h
+++ b/sound/usb/quirks.h
@@ -51,4 +51,6 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip,
struct audioformat *fp,
int stream);
+bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface);
+
#endif /* __USBAUDIO_QUIRKS_H */
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index afd5aa574611..15296f2c902c 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -502,6 +502,9 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip,
subs = &as->substream[stream];
if (subs->ep_num)
continue;
+ if (snd_device_get_state(chip->card, as->pcm) !=
+ SNDRV_DEV_BUILD)
+ chip->need_delayed_register = true;
err = snd_pcm_new_stream(as->pcm, stream, 1);
if (err < 0)
return err;
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 6fe3ab582ec6..1c892c7f14d7 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -34,6 +34,7 @@ struct snd_usb_audio {
unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */
unsigned int tx_length_quirk:1; /* Put length specifier in transfers */
unsigned int setup_fmt_after_resume_quirk:1; /* setup the format to interface after resume */
+ unsigned int need_delayed_register:1; /* warn for delayed registration */
int num_interfaces;
int num_suspended_intf;
int sample_rate_read_error;