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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/usb/mixer.c
parent7afd16f882887c9adc69cd1794f5e57777723217 (diff)
parent17e1717c11a34f9b0956e33e0c4a4e4ae8c51a57 (diff)
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/usb/mixer.c')
-rw-r--r--sound/usb/mixer.c78
1 files changed, 76 insertions, 2 deletions
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) {