Merge tag 'sound-4.5-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "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"

* tag 'sound-4.5-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - disable dynamic clock gating on Broxton before reset
  ALSA: hda - Add new GPU codec ID 0x10de0083 to snd-hda
  ALSA: dummy: Disable switching timer backend via sysfs
  ALSA: timer: fix SND_PCM_TIMER Kconfig text
  ALSA: Add missing dependency on CONFIG_SND_TIMER
  ALSA: bebob: Use a signed return type for get_formation_index
  ALSA: usb-audio: Fix TEAC UD-501/UD-503/NT-503 usb delay
  ALSA: compress: Disable GET_CODEC_CAPS ioctl for some architectures
  ALSA: seq: Degrade the error message for too many opens
  ALSA: seq: Fix incorrect sanity check at snd_seq_oss_synth_cleanup()

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

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 @@ snd_compr_get_caps(struct snd_compr_stream *stream, unsigned long arg)
 	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 @@ snd_compr_get_codec_caps(struct snd_compr_stream *stream, unsigned long arg)
 	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;

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;
 	}

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];

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
 

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,

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.

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.

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);
 

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),

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.
 

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
 	 */