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sound fixes for 4.5-rc2

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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.
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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()
This commit is contained in:
Linus Torvalds 2016-01-29 12:34:39 -08:00
commit b54544322e
12 changed files with 61 additions and 12 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

6
sound/core/Kconfig
View file

@ -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

11
sound/core/compress_offload.c
View file

@ -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;

2
sound/core/seq/oss/seq_oss_init.c
View file

@ -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;
}

2
sound/core/seq/oss/seq_oss_synth.c
View file

@ -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];

2
sound/drivers/dummy.c
View file

@ -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

14
sound/firewire/bebob/bebob_stream.c
View file

@ -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,

4
sound/isa/Kconfig
View file

@ -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.

3
sound/pci/Kconfig
View file

@ -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.

13
sound/pci/hda/hda_intel.c
View file

@ -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);

1
sound/pci/hda/patch_hdmi.c
View file

@ -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),

1
sound/sparc/Kconfig
View file

@ -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.

14
sound/usb/quirks.c
View file

@ -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
*/