December 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH trivial] ASoC: xlnx: Grammar s/the the/the/
by Geert Uytterhoeven 19 Dec '18

19 Dec '18

Fixes: 33f8db9a89200c18 ("ASoC: xlnx: enable i2s driver build") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/xilinx/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 25e287feb58c33e8..723a583a8d570f62 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -1,5 +1,5 @@ config SND_SOC_XILINX_I2S - tristate "Audio support for the the Xilinx I2S" + tristate "Audio support for the Xilinx I2S" help Select this option to enable Xilinx I2S Audio. This enables I2S playback and capture using xilinx soft IP. In transmitter -- 2.17.1

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[alsa-devel] [PATCH] ASoC: Intel: Haswell/Broadwell: fix setting for .dynamic field
by Rander Wang 19 Dec '18

19 Dec '18

For some reason this field was set to zero when all other drivers use .dynamic = 1 for front-ends. This change was tested on Dell XPS13 and has no impact with the existing legacy driver. The SOF driver also works with this change which enables it to override the fixed topology. Signed-off-by: Rander Wang <rander.wang(a)linux.intel.com> --- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 68e6543e6cb0..99f2a0156ae8 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -192,7 +192,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index eab1f439dd3f..a4022983a7ce 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -146,7 +146,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, -- 2.14.1

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[alsa-devel] Loading coefficients for Audio IP
by Daniel Baluta 19 Dec '18

19 Dec '18

Hello Pierre, Liam, Mark, What is the standard ALSA way of loading tables of coefficients for various audio IPs? We have IPs like PDM or ASRC which need large table of coefficients depending on configuration. This tables can reach up to 120K in size. Creating this kind of tables as arrays of integers in kernel doesn't seem to be a common used solution and we will end up with large files that contains only numbers. Another approach that we are thinking of is to create and compile this tables in userspace as raw binary files and then loading them in kernel using the firmware interface. Would this be an acceptable approach? thanks, Daniel.

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[alsa-devel] [PATCH] ALSA: hda/realtek: Enable audio jacks of ASUS UX391UA with, ALC294
by Wandrille RONCE 19 Dec '18

19 Dec '18

From c87ce2119f010db78933c424b8ba6738c3056dff Mon Sep 17 00:00:00 2001 From: Wandrille RONCE <w(a)ndrille.fr> Date: Mon, 17 Dec 2018 19:00:00 +0100 Subject: [PATCH] ALSA: hda/realtek: Enable audio jacks of ASUS UX391UA with ALC294 --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 15021c839372..90f30e769a42 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6584,6 +6584,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x14a1, "ASUS UX533FD", ALC294_FIXUP_ASUS_SPK), + SND_PCI_QUIRK(0x1043, 0x10a1, "ASUS UX391UA", ALC294_FIXUP_ASUS_SPK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), -- 2.19.1

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[alsa-devel] 4.21 merge window
by Takashi Iwai 19 Dec '18

19 Dec '18

Hi, we're approaching to 4.20 final release, and it means that the merge window of 4.21 will be closed soon, too. Actually, I'll be traveling from the next Saturday (more exactly, already leaving on Friday afternoon), so I'd like to close the merges of major changes in today, and prepare the pull request for tomorrow. If you still have anything really important that must be in 4.21, please let me know. Thanks! Takashi

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[alsa-devel] [PATCH] ALSA: bebob: fix model-id of unit for Apogee Ensemble
by Takashi Sakamoto 19 Dec '18

19 Dec '18

This commit fixes hard-coded model-id for an unit of Apogee Ensemble with a correct value. This unit uses DM1500 ASIC produced ArchWave AG (formerly known as BridgeCo AG). I note that this model supports three modes in the number of data channels in tx/rx streams; 8 ch pairs, 10 ch pairs, 18 ch pairs. The mode is switched by Vendor-dependent AV/C command, like: $ cd linux-firewire-utils $ ./firewire-request /dev/fw1 fcp 0x00ff000003dbeb0600000000 (8ch pairs) $ ./firewire-request /dev/fw1 fcp 0x00ff000003dbeb0601000000 (10ch pairs) $ ./firewire-request /dev/fw1 fcp 0x00ff000003dbeb0602000000 (18ch pairs) When switching between different mode, the unit disappears from IEEE 1394 bus, then appears on the bus with different combination of stream formats. In a mode of 18 ch pairs, available sampling rate is up to 96.0 kHz, else up to 192.0 kHz. $ ./hinawa-config-rom-printer /dev/fw1 { 'bus-info': { 'adj': False, 'bmc': True, 'chip_ID': 21474898341, 'cmc': True, 'cyc_clk_acc': 100, 'generation': 2, 'imc': True, 'isc': True, 'link_spd': 2, 'max_ROM': 1, 'max_rec': 512, 'name': '1394', 'node_vendor_ID': 987, 'pmc': False}, 'root-directory': [ ['HARDWARE_VERSION', 19], [ 'NODE_CAPABILITIES', { 'addressing': {'64': True, 'fix': True, 'prv': False}, 'misc': {'int': False, 'ms': False, 'spt': True}, 'state': { 'atn': False, 'ded': False, 'drq': True, 'elo': False, 'init': False, 'lst': True, 'off': False}, 'testing': {'bas': False, 'ext': False}}], ['VENDOR', 987], ['DESCRIPTOR', 'Apogee Electronics'], ['MODEL', 126702], ['DESCRIPTOR', 'Ensemble'], ['VERSION', 5297], [ 'UNIT', [ ['SPECIFIER_ID', 41005], ['VERSION', 65537], ['MODEL', 126702], ['DESCRIPTOR', 'Ensemble']]], [ 'DEPENDENT_INFO', [ ['SPECIFIER_ID', 2037], ['VERSION', 1], [(58, 'IMMEDIATE'), 16777159], [(59, 'IMMEDIATE'), 1048576], [(60, 'IMMEDIATE'), 16777159], [(61, 'IMMEDIATE'), 6291456]]]]} Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 672d13488454..d91874275d2c 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -408,7 +408,7 @@ static const struct ieee1394_device_id bebob_id_table[] = { /* Apogee Electronics, DA/AD/DD-16X (X-FireWire card) */ SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x00010048, &spec_normal), /* Apogee Electronics, Ensemble */ - SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x00001eee, &spec_normal), + SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x01eeee, &spec_normal), /* ESI, Quatafire610 */ SND_BEBOB_DEV_ENTRY(VEN_ESI, 0x00010064, &spec_normal), /* AcousticReality, eARMasterOne */ -- 2.19.1

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[alsa-devel] [PATCH] ALSA: emu10k1: Fix potential Spectre v1 vulnerabilities
by Gustavo A. R. Silva 19 Dec '18

19 Dec '18

ipcm->substream is indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. This issue was detected with the help of Smatch: sound/pci/emu10k1/emufx.c:1031 snd_emu10k1_ipcm_poke() warn: potential spectre issue 'emu->fx8010.pcm' [r] (local cap) sound/pci/emu10k1/emufx.c:1075 snd_emu10k1_ipcm_peek() warn: potential spectre issue 'emu->fx8010.pcm' [r] (local cap) Fix this by sanitizing ipcm->substream before using it to index emu->fx8010.pcm Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. [1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2 Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/emu10k1/emufx.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 6ebe817801ea..1f25e6d029d8 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -36,6 +36,7 @@ #include <linux/init.h> #include <linux/mutex.h> #include <linux/moduleparam.h> +#include <linux/nospec.h> #include <sound/core.h> #include <sound/tlv.h> @@ -1026,6 +1027,8 @@ static int snd_emu10k1_ipcm_poke(struct snd_emu10k1 *emu, if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT) return -EINVAL; + ipcm->substream = array_index_nospec(ipcm->substream, + EMU10K1_FX8010_PCM_COUNT); if (ipcm->channels > 32) return -EINVAL; pcm = &emu->fx8010.pcm[ipcm->substream]; @@ -1072,6 +1075,8 @@ static int snd_emu10k1_ipcm_peek(struct snd_emu10k1 *emu, if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT) return -EINVAL; + ipcm->substream = array_index_nospec(ipcm->substream, + EMU10K1_FX8010_PCM_COUNT); pcm = &emu->fx8010.pcm[ipcm->substream]; mutex_lock(&emu->fx8010.lock); spin_lock_irq(&emu->reg_lock); -- 2.19.2

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[alsa-devel] [PATCH] ALSA: rme9652: Fix potential Spectre v1 vulnerability
by Gustavo A. R. Silva 19 Dec '18

19 Dec '18

info->channel is indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. This issue was detected with the help of Smatch: sound/pci/rme9652/hdsp.c:4100 snd_hdsp_channel_info() warn: potential spectre issue 'hdsp->channel_map' [r] (local cap) Fix this by sanitizing info->channel before using it to index hdsp->channel_map Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. Also, notice that I refactored the code a bit in order to get rid of the following checkpatch warning: ERROR: do not use assignment in if condition FILE: sound/pci/rme9652/hdsp.c:4103: if ((mapped_channel = hdsp->channel_map[info->channel]) < 0) [1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2 Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/rme9652/hdsp.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 1bff4b1b39cd..ba99ff0e93e0 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -30,6 +30,7 @@ #include <linux/math64.h> #include <linux/vmalloc.h> #include <linux/io.h> +#include <linux/nospec.h> #include <sound/core.h> #include <sound/control.h> @@ -4092,15 +4093,16 @@ static int snd_hdsp_channel_info(struct snd_pcm_substream *substream, struct snd_pcm_channel_info *info) { struct hdsp *hdsp = snd_pcm_substream_chip(substream); - int mapped_channel; + unsigned int channel = info->channel; - if (snd_BUG_ON(info->channel >= hdsp->max_channels)) + if (snd_BUG_ON(channel >= hdsp->max_channels)) return -EINVAL; + channel = array_index_nospec(channel, hdsp->max_channels); - if ((mapped_channel = hdsp->channel_map[info->channel]) < 0) + if (hdsp->channel_map[channel] < 0) return -EINVAL; - info->offset = mapped_channel * HDSP_CHANNEL_BUFFER_BYTES; + info->offset = hdsp->channel_map[channel] * HDSP_CHANNEL_BUFFER_BYTES; info->first = 0; info->step = 32; return 0; -- 2.19.2

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[alsa-devel] [PATCH v3] ASoC: AMD: Fix race condition between register access
by Agrawal, Akshu 19 Dec '18

19 Dec '18

During simultaneous running of playback and capture, we got hit by incorrect value write on common register. This was due to race condition between 2 streams. Fixing this by locking the common register access. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Added 2 helper functions, removed locking in ch enable/disable as they are separate registers. v3: Modified helper acp_reg_read_mod_write, moved pte locks to helper sound/soc/amd/acp-pcm-dma.c | 66 ++++++++++++++++++++++++++++++--------------- 1 file changed, 44 insertions(+), 22 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 0ac4b5b..0d4c2eb 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -121,6 +121,9 @@ .periods_max = CAPTURE_MAX_NUM_PERIODS, }; +/* Lock to protect access to registers */ +static DEFINE_SPINLOCK(lock); + static u32 acp_reg_read(void __iomem *acp_mmio, u32 reg) { return readl(acp_mmio + (reg * 4)); @@ -131,6 +134,31 @@ static void acp_reg_write(u32 val, void __iomem *acp_mmio, u32 reg) writel(val, acp_mmio + (reg * 4)); } +static void acp_reg_write_srbm_targ(void __iomem *acp_mmio, + u32 addr, u32 data) +{ + unsigned long flags; + + spin_lock_irqsave(&lock, flags); + acp_reg_write(addr, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); + acp_reg_write(data, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + spin_unlock_irqrestore(&lock, flags); +} + +static void acp_reg_read_mod_write(void __iomem *acp_mmio, u32 addr, + u32 mask, u32 value) +{ + u32 val; + unsigned long flags; + + spin_lock_irqsave(&lock, flags); + val = acp_reg_read(acp_mmio, addr); + val &= ~mask; + val |= (mask & value); + acp_reg_write(val, acp_mmio, addr); + spin_unlock_irqrestore(&lock, flags); +} + /* * Configure a given dma channel parameters - enable/disable, * number of descriptors, priority @@ -172,15 +200,12 @@ static void config_dma_descriptor_in_sram(void __iomem *acp_mmio, sram_offset = (descr_idx * sizeof(acp_dma_dscr_transfer_t)); /* program the source base address. */ - acp_reg_write(sram_offset, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - acp_reg_write(descr_info->src, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + acp_reg_write_srbm_targ(acp_mmio, sram_offset, descr_info->src); /* program the destination base address. */ - acp_reg_write(sram_offset + 4, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - acp_reg_write(descr_info->dest, acp_mmio, mmACP_SRBM_Targ_Idx_Data); - + acp_reg_write_srbm_targ(acp_mmio, sram_offset + 4, descr_info->dest); /* program the number of bytes to be transferred for this descriptor. */ - acp_reg_write(sram_offset + 8, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - acp_reg_write(descr_info->xfer_val, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + acp_reg_write_srbm_targ(acp_mmio, sram_offset + 8, + descr_info->xfer_val); } static void pre_config_reset(void __iomem *acp_mmio, u16 ch_num) @@ -311,24 +336,22 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, u32 offset; offset = ACP_DAGB_GRP_SRBM_SRAM_BASE_OFFSET + (pte_offset * 8); + for (page_idx = 0; page_idx < (num_of_pages); page_idx++) { /* Load the low address of page int ACP SRAM through SRBM */ - acp_reg_write((offset + (page_idx * 8)), - acp_mmio, mmACP_SRBM_Targ_Idx_Addr); addr = page_to_phys(pg); low = lower_32_bits(addr); high = upper_32_bits(addr); - acp_reg_write(low, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + acp_reg_write_srbm_targ(acp_mmio, (offset + (page_idx * 8)), + low); /* Load the High address of page int ACP SRAM through SRBM */ - acp_reg_write((offset + (page_idx * 8) + 4), - acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - /* page enable in ACP */ high |= BIT(31); - acp_reg_write(high, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + acp_reg_write_srbm_targ(acp_mmio, (offset + (page_idx * 8) + 4), + high); /* Move to next physically contiguos page */ pg++; @@ -870,29 +893,28 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, } } if (adata->asic_type == CHIP_STONEY) { - val = acp_reg_read(adata->acp_mmio, - mmACP_I2S_16BIT_RESOLUTION_EN); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: - val |= ACP_I2S_BT_16BIT_RESOLUTION_EN; + val = ACP_I2S_BT_16BIT_RESOLUTION_EN; break; case I2S_SP_INSTANCE: default: - val |= ACP_I2S_SP_16BIT_RESOLUTION_EN; + val = ACP_I2S_SP_16BIT_RESOLUTION_EN; } } else { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: - val |= ACP_I2S_BT_16BIT_RESOLUTION_EN; + val = ACP_I2S_BT_16BIT_RESOLUTION_EN; break; case I2S_SP_INSTANCE: default: - val |= ACP_I2S_MIC_16BIT_RESOLUTION_EN; + val = ACP_I2S_MIC_16BIT_RESOLUTION_EN; } } - acp_reg_write(val, adata->acp_mmio, - mmACP_I2S_16BIT_RESOLUTION_EN); + acp_reg_read_mod_write(adata->acp_mmio, + mmACP_I2S_16BIT_RESOLUTION_EN, + val, val); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -- 1.9.1

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[alsa-devel] [PATCH] ASoC: eukrea-tlv320: Do not enable on ARM64
by Andrey Smirnov 19 Dec '18

19 Dec '18

ARCH_MXC is also used by i.MX8/AArch64 SoC, building the driver for that platform results in: sound/soc/fsl/eukrea-tlv320.c:28:10: fatal error: asm/mach-types.h: No such file or directory #include <asm/mach-types.h> ^~~~~~~~~~~~~~~~~~ compilation terminated. Since the code in question doesn't seem to have any use on i.MX8, disable it by adding a !ARM64 as a dependecy. Signed-off-by: Andrey Smirnov <andrew.smirnov(a)gmail.com> --- Sorry for the noise if this is fixed already. Thanks, Andrey Smirnov sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 6ec19fb4a934..2e75b5bc5f1d 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -221,7 +221,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on ARCH_MXC && I2C + depends on ARCH_MXC && !ARM64 && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI -- 2.19.1

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