Linus,

please pull sound fixes for v4.5-rc2 from:

  git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git 
tags/sound-4.5-rc2

The topmost commit is 6639484ddaf6707b41082c9fa9ca9af342df6402

----------------------------------------------------------------

sound fixes for 4.5-rc2

There are a few fixes in ALSA core for bugs that have been spotted by
fuzzer.  Also a temporary workaround for PowerPC (and possibly other)
builds with incompatible ioctls was applied to compress API.

Other than that, a few trivial fixes and quirks for FireWire BeBoB,
USB-audio and HD-audio are found, too.

----------------------------------------------------------------

Aaron Plattner (1):
      ALSA: hda - Add new GPU codec ID 0x10de0083 to snd-hda

Guillaume Fougnies (1):
      ALSA: usb-audio: Fix TEAC UD-501/UD-503/NT-503 usb delay

Libin Yang (1):
      ALSA: hda - disable dynamic clock gating on Broxton before reset

Lucas Tanure (1):
      ALSA: bebob: Use a signed return type for get_formation_index

Randy Dunlap (1):
      ALSA: timer: fix SND_PCM_TIMER Kconfig text

Takashi Iwai (5):
      ALSA: seq: Fix incorrect sanity check at snd_seq_oss_synth_cleanup()
      ALSA: seq: Degrade the error message for too many opens
      ALSA: compress: Disable GET_CODEC_CAPS ioctl for some architectures
      ALSA: Add missing dependency on CONFIG_SND_TIMER
      ALSA: dummy: Disable switching timer backend via sysfs

---
 sound/core/Kconfig                  |  6 +++---
 sound/core/compress_offload.c       | 11 +++++++++++
 sound/core/seq/oss/seq_oss_init.c   |  2 +-
 sound/core/seq/oss/seq_oss_synth.c  |  2 +-
 sound/drivers/dummy.c               |  2 +-
 sound/firewire/bebob/bebob_stream.c | 14 +++++++++-----
 sound/isa/Kconfig                   |  4 ++++
 sound/pci/Kconfig                   |  3 +++
 sound/pci/hda/hda_intel.c           | 13 +++++++++++++
 sound/pci/hda/patch_hdmi.c          |  1 +
 sound/sparc/Kconfig                 |  1 +
 sound/usb/quirks.c                  | 14 +++++++++++++-
 12 files changed, 61 insertions(+), 12 deletions(-)

diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index e3e949126a56..a2a1e24becc6 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -97,11 +97,11 @@ config SND_PCM_TIMER
        bool "PCM timer interface" if EXPERT
        default y
        help
-         If you disable this option, pcm timer will be inavailable, so
-         those stubs used pcm timer (e.g. dmix, dsnoop & co) may work
+         If you disable this option, pcm timer will be unavailable, so
+         those stubs that use pcm timer (e.g. dmix, dsnoop & co) may work
          incorrectlly.
 
-         For some embedded device, we may disable it to reduce memory
+         For some embedded devices, we may disable it to reduce memory
          footprint, about 20KB on x86_64 platform.
 
 config SND_SEQUENCER_OSS
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 18b8dc45bb8f..7fac3cae8abd 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -46,6 +46,13 @@
 #include <sound/compress_offload.h>
 #include <sound/compress_driver.h>
 
+/* struct snd_compr_codec_caps overflows the ioctl bit size for some
+ * architectures, so we need to disable the relevant ioctls.
+ */
+#if _IOC_SIZEBITS < 14
+#define COMPR_CODEC_CAPS_OVERFLOW
+#endif
+
 /* TODO:
  * - add substream support for multiple devices in case of
  *     SND_DYNAMIC_MINORS is not used
@@ -440,6 +447,7 @@ out:
        return retval;
 }
 
+#ifndef COMPR_CODEC_CAPS_OVERFLOW
 static int
 snd_compr_get_codec_caps(struct snd_compr_stream *stream, unsigned long arg)
 {
@@ -463,6 +471,7 @@ out:
        kfree(caps);
        return retval;
 }
+#endif /* !COMPR_CODEC_CAPS_OVERFLOW */
 
 /* revisit this with snd_pcm_preallocate_xxx */
 static int snd_compr_allocate_buffer(struct snd_compr_stream *stream,
@@ -801,9 +810,11 @@ static long snd_compr_ioctl(struct file *f, unsigned int 
cmd, unsigned long arg)
        case _IOC_NR(SNDRV_COMPRESS_GET_CAPS):
                retval = snd_compr_get_caps(stream, arg);
                break;
+#ifndef COMPR_CODEC_CAPS_OVERFLOW
        case _IOC_NR(SNDRV_COMPRESS_GET_CODEC_CAPS):
                retval = snd_compr_get_codec_caps(stream, arg);
                break;
+#endif
        case _IOC_NR(SNDRV_COMPRESS_SET_PARAMS):
                retval = snd_compr_set_params(stream, arg);
                break;
diff --git a/sound/core/seq/oss/seq_oss_init.c 
b/sound/core/seq/oss/seq_oss_init.c
index b1221b29728e..6779e82b46dd 100644
--- a/sound/core/seq/oss/seq_oss_init.c
+++ b/sound/core/seq/oss/seq_oss_init.c
@@ -202,7 +202,7 @@ snd_seq_oss_open(struct file *file, int level)
 
        dp->index = i;
        if (i >= SNDRV_SEQ_OSS_MAX_CLIENTS) {
-               pr_err("ALSA: seq_oss: too many applications\n");
+               pr_debug("ALSA: seq_oss: too many applications\n");
                rc = -ENOMEM;
                goto _error;
        }
diff --git a/sound/core/seq/oss/seq_oss_synth.c 
b/sound/core/seq/oss/seq_oss_synth.c
index 0f3b38184fe5..b16dbef04174 100644
--- a/sound/core/seq/oss/seq_oss_synth.c
+++ b/sound/core/seq/oss/seq_oss_synth.c
@@ -308,7 +308,7 @@ snd_seq_oss_synth_cleanup(struct seq_oss_devinfo *dp)
        struct seq_oss_synth *rec;
        struct seq_oss_synthinfo *info;
 
-       if (snd_BUG_ON(dp->max_synthdev >= SNDRV_SEQ_OSS_MAX_SYNTH_DEVS))
+       if (snd_BUG_ON(dp->max_synthdev > SNDRV_SEQ_OSS_MAX_SYNTH_DEVS))
                return;
        for (i = 0; i < dp->max_synthdev; i++) {
                info = &dp->synths[i];
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 75b74850c005..bde33308f0d6 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -87,7 +87,7 @@ MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-128) 
for dummy driver.");
 module_param(fake_buffer, bool, 0444);
 MODULE_PARM_DESC(fake_buffer, "Fake buffer allocations.");
 #ifdef CONFIG_HIGH_RES_TIMERS
-module_param(hrtimer, bool, 0644);
+module_param(hrtimer, bool, 0444);
 MODULE_PARM_DESC(hrtimer, "Use hrtimer as the timer source.");
 #endif
 
diff --git a/sound/firewire/bebob/bebob_stream.c 
b/sound/firewire/bebob/bebob_stream.c
index 926e5dcbb66a..5022c9b97ddf 100644
--- a/sound/firewire/bebob/bebob_stream.c
+++ b/sound/firewire/bebob/bebob_stream.c
@@ -47,14 +47,16 @@ static const unsigned int bridgeco_freq_table[] = {
        [6] = 0x07,
 };
 
-static unsigned int
-get_formation_index(unsigned int rate)
+static int
+get_formation_index(unsigned int rate, unsigned int *index)
 {
        unsigned int i;
 
        for (i = 0; i < ARRAY_SIZE(snd_bebob_rate_table); i++) {
-               if (snd_bebob_rate_table[i] == rate)
-                       return i;
+               if (snd_bebob_rate_table[i] == rate) {
+                       *index = i;
+                       return 0;
+               }
        }
        return -EINVAL;
 }
@@ -425,7 +427,9 @@ make_both_connections(struct snd_bebob *bebob, unsigned int 
rate)
                goto end;
 
        /* confirm params for both streams */
-       index = get_formation_index(rate);
+       err = get_formation_index(rate, &index);
+       if (err < 0)
+               goto end;
        pcm_channels = bebob->tx_stream_formations[index].pcm;
        midi_channels = bebob->tx_stream_formations[index].midi;
        err = amdtp_am824_set_parameters(&bebob->tx_stream, rate,
diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig
index 0216475fc759..37adcc6cbe6b 100644
--- a/sound/isa/Kconfig
+++ b/sound/isa/Kconfig
@@ -3,6 +3,7 @@
 config SND_WSS_LIB
         tristate
         select SND_PCM
+       select SND_TIMER
 
 config SND_SB_COMMON
         tristate
@@ -42,6 +43,7 @@ config SND_AD1816A
        select SND_OPL3_LIB
        select SND_MPU401_UART
        select SND_PCM
+       select SND_TIMER
        help
          Say Y here to include support for Analog Devices SoundPort
          AD1816A or compatible sound chips.
@@ -209,6 +211,7 @@ config SND_GUSCLASSIC
        tristate "Gravis UltraSound Classic"
        select SND_RAWMIDI
        select SND_PCM
+       select SND_TIMER
        help
          Say Y here to include support for Gravis UltraSound Classic
          soundcards.
@@ -221,6 +224,7 @@ config SND_GUSEXTREME
        select SND_OPL3_LIB
        select SND_MPU401_UART
        select SND_PCM
+       select SND_TIMER
        help
          Say Y here to include support for Gravis UltraSound Extreme
          soundcards.
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 656ce39bddbc..8f6594a7d37f 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -155,6 +155,7 @@ config SND_AZT3328
        select SND_PCM
        select SND_RAWMIDI
        select SND_AC97_CODEC
+       select SND_TIMER
        depends on ZONE_DMA
        help
          Say Y here to include support for Aztech AZF3328 (PCI168)
@@ -463,6 +464,7 @@ config SND_EMU10K1
        select SND_HWDEP
        select SND_RAWMIDI
        select SND_AC97_CODEC
+       select SND_TIMER
        depends on ZONE_DMA
        help
          Say Y to include support for Sound Blaster PCI 512, Live!,
@@ -889,6 +891,7 @@ config SND_YMFPCI
        select SND_OPL3_LIB
        select SND_MPU401_UART
        select SND_AC97_CODEC
+       select SND_TIMER
        help
          Say Y here to include support for Yamaha PCI audio chips -
          YMF724, YMF724F, YMF740, YMF740C, YMF744, YMF754.
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 256e6cda218f..4045dca3d699 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -90,6 +90,8 @@ enum {
 #define NVIDIA_HDA_ENABLE_COHBIT      0x01
 
 /* Defines for Intel SCH HDA snoop control */
+#define INTEL_HDA_CGCTL         0x48
+#define INTEL_HDA_CGCTL_MISCBDCGE        (0x1 << 6)
 #define INTEL_SCH_HDA_DEVC      0x78
 #define INTEL_SCH_HDA_DEVC_NOSNOOP       (0x1<<11)
 
@@ -534,10 +536,21 @@ static void hda_intel_init_chip(struct azx *chip, bool 
full_reset)
 {
        struct hdac_bus *bus = azx_bus(chip);
        struct pci_dev *pci = chip->pci;
+       u32 val;
 
        if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
                snd_hdac_set_codec_wakeup(bus, true);
+       if (IS_BROXTON(pci)) {
+               pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val);
+               val = val & ~INTEL_HDA_CGCTL_MISCBDCGE;
+               pci_write_config_dword(pci, INTEL_HDA_CGCTL, val);
+       }
        azx_init_chip(chip, full_reset);
+       if (IS_BROXTON(pci)) {
+               pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val);
+               val = val | INTEL_HDA_CGCTL_MISCBDCGE;
+               pci_write_config_dword(pci, INTEL_HDA_CGCTL, val);
+       }
        if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
                snd_hdac_set_codec_wakeup(bus, false);
 
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 426a29a1c19b..1f52b55d77c9 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -3653,6 +3653,7 @@ HDA_CODEC_ENTRY(0x10de0070, "GPU 70 HDMI/DP",     
patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0071, "GPU 71 HDMI/DP",  patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0072, "GPU 72 HDMI/DP",  patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de007d, "GPU 7d HDMI/DP",  patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0083, "GPU 83 HDMI/DP",  patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI",      patch_nvhdmi_2ch),
 HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP",   patch_via_hdmi),
 HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP",   patch_via_hdmi),
diff --git a/sound/sparc/Kconfig b/sound/sparc/Kconfig
index d75deba5617d..dfcd38647606 100644
--- a/sound/sparc/Kconfig
+++ b/sound/sparc/Kconfig
@@ -22,6 +22,7 @@ config SND_SUN_AMD7930
 config SND_SUN_CS4231
        tristate "Sun CS4231"
        select SND_PCM
+       select SND_TIMER
        help
          Say Y here to include support for CS4231 sound device on Sun.
 
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 23ea6d800c4c..a75d9ce7d77a 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1205,8 +1205,12 @@ void snd_usb_set_interface_quirk(struct usb_device *dev)
         * "Playback Design" products need a 50ms delay after setting the
         * USB interface.
         */
-       if (le16_to_cpu(dev->descriptor.idVendor) == 0x23ba)
+       switch (le16_to_cpu(dev->descriptor.idVendor)) {
+       case 0x23ba: /* Playback Design */
+       case 0x0644: /* TEAC Corp. */
                mdelay(50);
+               break;
+       }
 }
 
 void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
@@ -1221,6 +1225,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, 
unsigned int pipe,
            (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
                mdelay(20);
 
+       /*
+        * "TEAC Corp." products need a 20ms delay after each
+        * class compliant request
+        */
+       if ((le16_to_cpu(dev->descriptor.idVendor) == 0x0644) &&
+           (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
+               mdelay(20);
+
        /* Marantz/Denon devices with USB DAC functionality need a delay
         * after each class compliant request
         */

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