From 599151336638d57b98d92338aa59c048e3a3e97d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 25 Jan 2016 11:01:47 +0100 Subject: [PATCH 01/10] ALSA: seq: Fix incorrect sanity check at snd_seq_oss_synth_cleanup() ALSA sequencer OSS emulation code has a sanity check for currently opened devices, but there is a thinko there, eventually it spews warnings and skips the operation wrongly like: WARNING: CPU: 1 PID: 7573 at sound/core/seq/oss/seq_oss_synth.c:311 Fix this off-by-one error. Reported-by: Dmitry Vyukov Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_synth.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) 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]; From da10816e3d923565b470fec78a674baba794ed33 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 25 Jan 2016 11:24:56 +0100 Subject: [PATCH 02/10] ALSA: seq: Degrade the error message for too many opens ALSA OSS sequencer spews a kernel error message ("ALSA: seq_oss: too many applications") when user-space tries to open more than the limit. This means that it can easily fill the log buffer. Since it's merely a normal error, it's safe to suppress it via pr_debug() instead. Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_init.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) 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; } From 462b3f161beb62eeb290f4ec52f5ead29a2f8ac7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 25 Jan 2016 13:59:21 +0100 Subject: [PATCH 03/10] ALSA: compress: Disable GET_CODEC_CAPS ioctl for some architectures Some architectures like PowerPC can handle the maximum struct size in an ioctl only up to 13 bits, and struct snd_compr_codec_caps used by SNDRV_COMPRESS_GET_CODEC_CAPS ioctl overflows this limit. This problem was revealed recently by a powerpc change, as it's now treated as a fatal build error. This patch is a stop-gap for that: for architectures with less than 14 bit ioctl struct size, get rid of the handling of the relevant ioctl. We should provide an alternative equivalent ioctl code later, but for now just paper over it. Luckily, the compress API hasn't been used on such architectures, so the impact must be effectively zero. Reviewed-by: Mark Brown Acked-by: Sudip Mukherjee Cc: Signed-off-by: Takashi Iwai --- sound/core/compress_offload.c | 11 +++++++++++ 1 file changed, 11 insertions(+) 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 #include +/* 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; From 5a4ff9ec8d6edd2ab1cfe8ce6a080d6e57cbea9a Mon Sep 17 00:00:00 2001 From: Guillaume Fougnies Date: Tue, 26 Jan 2016 00:28:27 +0100 Subject: [PATCH 04/10] ALSA: usb-audio: Fix TEAC UD-501/UD-503/NT-503 usb delay TEAC UD-501/UD-503/NT-503 fail to switch properly between different rate/format. Similar to 'Playback Design', this patch corrects the invalid clock source error for TEAC products and avoids complete freeze of the usb interface of 503 series. Signed-off-by: Guillaume Fougnies Cc: Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) 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 */ From 07905298e4d5777eb58516cdc242f7ac1ca387a2 Mon Sep 17 00:00:00 2001 From: Lucas Tanure Date: Mon, 25 Jan 2016 19:30:23 -0200 Subject: [PATCH 05/10] ALSA: bebob: Use a signed return type for get_formation_index The return type "unsigned int" was used by the get_formation_index function despite of the aspect that it will eventually return a negative error code. So, change to signed int and get index by reference in the parameters. Done with the help of Coccinelle. [Fix the missing braces suggested by Julia Lawall -- tiwai] Signed-off-by: Lucas Tanure Reviewed-by: Takashi Sakamoto Tested-by: Takashi Sakamoto Cc: Signed-off-by: Takashi Iwai --- sound/firewire/bebob/bebob_stream.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) 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, From 61595dca742a9ba9a4c998b9af1f468adc816275 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Jan 2016 07:05:56 +0100 Subject: [PATCH 06/10] ALSA: Add missing dependency on CONFIG_SND_TIMER Since the build of PCM timer may be disabled via Kconfig now, each driver that provides a timer interface needs to set CONFIG_SND_TIMER explicitly. Otherwise it may get a build error due to missing symbol. Fixes: 90bbaf66ee7b ('ALSA: timer: add config item to export PCM timer disabling for expert') Reported-by: kbuild test robot Cc: # v4.4+ Signed-off-by: Takashi Iwai --- sound/isa/Kconfig | 4 ++++ sound/pci/Kconfig | 3 +++ sound/sparc/Kconfig | 1 + 3 files changed, 8 insertions(+) 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/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. From ac1efcfb35f73657ec1e28b1e8a3c895529c1108 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Wed, 27 Jan 2016 18:04:10 -0800 Subject: [PATCH 07/10] ALSA: timer: fix SND_PCM_TIMER Kconfig text Fix spelling and typos for SND_PCM_TIMER. Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/core/Kconfig | 6 +++--- 1 file changed, 3 insertions(+), 3 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 From 7ee96216c31aabe1eb42fb91ff50dae9fcd014b2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Jan 2016 07:54:16 +0100 Subject: [PATCH 08/10] ALSA: dummy: Disable switching timer backend via sysfs ALSA dummy driver can switch the timer backend between system timer and hrtimer via its hrtimer module option. This can be also switched dynamically via sysfs, but it may lead to a memory corruption when switching is done while a PCM stream is running; the stream instance for the newly switched timer method tries to access the memory that was allocated by another timer method although the sizes differ. As the simplest fix, this patch just disables the switch via sysfs by dropping the writable bit. BugLink: http://lkml.kernel.org/r/CACT4Y+ZGEeEBntHW5WHn2GoeE0G_kRrCmUh6=dWyy-wfzvuJLg@mail.gmail.com Reported-by: Dmitry Vyukov Cc: Signed-off-by: Takashi Iwai --- sound/drivers/dummy.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) 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 From 3ec622f40913ae036f218e5e7e92df9c1f1753d9 Mon Sep 17 00:00:00 2001 From: Aaron Plattner Date: Thu, 28 Jan 2016 14:07:38 -0800 Subject: [PATCH 09/10] ALSA: hda - Add new GPU codec ID 0x10de0083 to snd-hda Vendor ID 0x10de0083 is used by a yet-to-be-named GPU chip. This chip also has the 2-ch audio swapping bug, so patch_nvhdmi is appropriate here. Signed-off-by: Aaron Plattner Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) 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), From 6639484ddaf6707b41082c9fa9ca9af342df6402 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Fri, 29 Jan 2016 20:39:09 +0800 Subject: [PATCH 10/10] ALSA: hda - disable dynamic clock gating on Broxton before reset On Broxton, to make sure the reset controller works properly, MISCBDCGE bit (bit 6) in CGCTL (0x48) of PCI configuration space need be cleared before reset and set back to 1 after reset. Otherwise, it may prevent the CORB/RIRB logic from being reset. Signed-off-by: Libin Yang Cc: # v4.4+ Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) 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);