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

[alsa-devel] [PATCH] ALSA: isa/wavefront: Fix potential Spectre v1 vulnerabilities
by Gustavo A. R. Silva 20 Dec '18

20 Dec '18

header->number 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/isa/wavefront/wavefront_synth.c:792 wavefront_send_patch() warn: potential spectre issue 'dev->patch_status' [w] (local cap) sound/isa/wavefront/wavefront_synth.c:819 wavefront_send_program() warn: potential spectre issue 'dev->prog_status' [w] (local cap) sound/isa/wavefront/wavefront_synth.c:1197 wavefront_send_alias() warn: potential spectre issue 'dev->sample_status' [w] sound/isa/wavefront/wavefront_synth.c:1248 wavefront_send_multisample() warn: potential spectre issue 'dev->sample_status' [w] sound/isa/wavefront/wavefront_synth.c:1548 wavefront_synth_control() warn: potential spectre issue 'dev->sample_status' [r] (local cap) Fix this by sanitizing header->number before using it to index dev->patch_status, dev->prog_status and dev->sample_status. 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/isa/wavefront/wavefront_synth.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 0b1e4b34b299..80196a533a36 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -31,6 +31,7 @@ #include <linux/moduleparam.h> #include <linux/slab.h> #include <linux/module.h> +#include <linux/nospec.h> #include <sound/core.h> #include <sound/snd_wavefront.h> #include <sound/initval.h> @@ -788,7 +789,8 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header) if (header->number >= ARRAY_SIZE(dev->patch_status)) return -EINVAL; - + header->number = array_index_nospec(header->number, + ARRAY_SIZE(dev->patch_status)); dev->patch_status[header->number] |= WF_SLOT_FILLED; bptr = buf; @@ -815,7 +817,8 @@ wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header) if (header->number >= ARRAY_SIZE(dev->prog_status)) return -EINVAL; - + header->number = array_index_nospec(header->number, + ARRAY_SIZE(dev->prog_status)); dev->prog_status[header->number] = WF_SLOT_USED; /* XXX need to zero existing SLOT_USED bit for program_status[i] @@ -1194,6 +1197,10 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) return -EIO; } + if (header->number >= ARRAY_SIZE(dev->sample_status)) + return -EINVAL; + header->number = array_index_nospec(header->number, + ARRAY_SIZE(dev->sample_status)); dev->sample_status[header->number] = (WF_SLOT_FILLED|WF_ST_ALIAS); return (0); @@ -1245,6 +1252,10 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) return -EIO; } + if (header->number >= ARRAY_SIZE(dev->sample_status)) + return -EINVAL; + header->number = array_index_nospec(header->number, + ARRAY_SIZE(dev->sample_status)); dev->sample_status[header->number] = (WF_SLOT_FILLED|WF_ST_MULTISAMPLE); kfree(msample_hdr); @@ -1539,12 +1550,13 @@ wavefront_synth_control (snd_wavefront_card_t *acard, case WFC_IDENTIFY_SLOT_TYPE: i = wc->wbuf[0] | (wc->wbuf[1] << 7); - if (i <0 || i >= WF_MAX_SAMPLE) { + if (i < 0 || i >= WF_MAX_SAMPLE) { snd_printk ("invalid slot ID %d\n", i); wc->status = EINVAL; return -EINVAL; } + i = array_index_nospec(i, WF_MAX_SAMPLE); wc->rbuf[0] = dev->sample_status[i]; wc->status = 0; return 0; -- 2.20.1

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[alsa-devel] Console downloaders give *The certificate of ‘www.alsa-project.org’ hasn't got a known issuer.*
by Paul Menzel 20 Dec '18

20 Dec '18

[Please CC, as I am not subscribed.] Dear ALSA folks, Despite working in the browser (Mozilla Firefox), GNU Wget and curl give the error below trying to download the script `alsa-info.sh`. $ wget https://www.alsa-project.org/alsa-info.sh --2018-12-18 17:27:57-- https://www.alsa-project.org/alsa-info.sh Resolving www.alsa-project.org (www.alsa-project.org)... 77.48.224.243 Connecting to www.alsa-project.org (www.alsa-project.org)|77.48.224.243|:443... connected. ERROR: The certificate of ‘www.alsa-project.org’ is not trusted. ERROR: The certificate of ‘www.alsa-project.org’ hasn't got a known issuer. Kind regards, Paul

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[alsa-devel] [PATCH] ASoC: xlnx: change license header format style
by Maruthi Srinivas Bayyavarapu 20 Dec '18

20 Dec '18

Changed License header from C to C++ style comment block. Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> --- sound/soc/xilinx/xlnx_i2s.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/xilinx/xlnx_i2s.c b/sound/soc/xilinx/xlnx_i2s.c index d4ae9ef..8b35316 100644 --- a/sound/soc/xilinx/xlnx_i2s.c +++ b/sound/soc/xilinx/xlnx_i2s.c @@ -1,12 +1,11 @@ // SPDX-License-Identifier: GPL-2.0 -/* - * Xilinx ASoC I2S audio support - * - * Copyright (C) 2018 Xilinx, Inc. - * - * Author: Praveen Vuppala <praveenv(a)xilinx.com> - * Author: Maruthi Srinivas Bayyavarapu <maruthis(a)xilinx.com> - */ +// +// Xilinx ASoC I2S audio support +// +// Copyright (C) 2018 Xilinx, Inc. +// +// Author: Praveen Vuppala <praveenv(a)xilinx.com> +// Author: Maruthi Srinivas Bayyavarapu <maruthis(a)xilinx.com> #include <linux/io.h> #include <linux/module.h> -- 2.7.4

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[alsa-devel] [PATCH 0/3] add Xilinx audio formatter driver
by Maruthi Srinivas Bayyavarapu 20 Dec '18

20 Dec '18

Audio formatter IP supports two streaming interfaces - MM2S for playback and S2MM for capture. The driver enables DMA functionality for both the interfaces. Patchset includes devicetree bindings documentation, driver and build enablement. Maruthi Srinivas Bayyavarapu (3): dt-bindings: ASoC: xlnx,audio-formatter: Document audio formatter bindings ASoC: xlnx: add pcm formatter platform driver ASoC: xlnx: enable audio formatter driver build .../bindings/sound/xlnx,audio-formatter.txt | 29 ++ sound/soc/xilinx/Kconfig | 7 + sound/soc/xilinx/Makefile | 2 + sound/soc/xilinx/xlnx_formatter_pcm.c | 561 +++++++++++++++++++++ 4 files changed, 599 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt create mode 100644 sound/soc/xilinx/xlnx_formatter_pcm.c -- 2.7.4

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[alsa-devel] [PATCH 0/9] ASoC/ARM: Merge the davinci and omap audio directories
by Peter Ujfalusi 20 Dec '18

20 Dec '18

Hi, Changes since RFC: - Added Jarkko's Acked-by/Tested-by for the {omap->ti} changes - Rebased on top of next-20181210 (20181214+ is not booting on any of my boards) For Texas Instruments audio support we have two directories at the moment: davinci omap In reality TI have the following SoC lines supported by Linux: daVinci OMAP Sitara Keystone We already needed to have cross directory 'links' for DRA7x/AM5 since they use McASP (davinci directory) and sDMA (omap directory). Keystone II line of SoCs are using drivers from davinci directory (McASP and EDMA). Which is kind of fine as they are in the same place. With the new AM654 (http://www.ti.com/lit/pdf/spruid7) we will have completely new DMA and there is no place to put that as it does not belong under davinci or omap, but it makes no sense to create yet another TI related directory just for the small udma-pcm. Another issue with the current support is how the Kconfigs are written. The DAI drivers are depending on the ASoC platform drivers and it is a bit of a mess at the moment. For simple-sound-card one need to know what DMA is used and then select the DAI driver (if the Kconfig is right). I have also cleaned up the Kconfig prompts and comments a bit to be more precise. As can be seen in the diffstat I tried to do the merger with minimal code change, resisting the urge to rename the davinci-i2c driver to davinci-asp. With the merger I have created a new Kconfig structure: - Submenu for TI related settings for clarity - Only the DAI drivers are selectable and the platform drivers are selected based on the available (enabled) DMA engines - The remaining ASoC machine drivers are selecting the DAIs they need - The davinci-evm is now a single entry, combining all the various daVinci EVMS it supports and selects the needed DAIs based on the enabled boards The series also includes patches to updated the arch/arm/ files (defconfigs, source files using audio Kconfig symbols). I think I have catched all (I hope) issues regarding to Kconfig thanks to the various autobuild machines out there kindly testing my linux-next-wip branch. Mark, Tony, Sekhar: This series - as I mentioned - is based on next-20181210. Not sure what tag, branch, commit I should base this, but I can resend v2 if needed. Regards, Peter --- Peter Ujfalusi (9): ASoC: ti: Merge davinci and omap directories MAINTAINERS: Add entry for sound/soc/ti and update the OMAP audio support ARM: OMAP1: Makefile: Update for new MCBSP Kconfig option ARM: OMAP2: Update for new MCBSP Kconfig option ARM: davinci: dm365-evm: Update for the new ASoC Kcofnig options ARM: omap2plus_defconfig: Update the audio options ARM: omap1_defconfig: Do not select ASoC by default ARM: davinci_all_defconfig: Update the audio options ASoC: ti: Kconfig: Remove the deprecated options MAINTAINERS | 11 +- arch/arm/configs/davinci_all_defconfig | 5 +- arch/arm/configs/omap1_defconfig | 2 - arch/arm/configs/omap2plus_defconfig | 14 +- arch/arm/mach-davinci/board-dm365-evm.c | 4 +- arch/arm/mach-omap1/Makefile | 2 +- arch/arm/mach-omap2/Makefile | 2 +- arch/arm/mach-omap2/pdata-quirks.c | 4 +- sound/soc/Kconfig | 3 +- sound/soc/Makefile | 3 +- sound/soc/davinci/Kconfig | 106 --------- sound/soc/davinci/Makefile | 16 -- sound/soc/omap/Kconfig | 120 ---------- sound/soc/omap/Makefile | 30 --- sound/soc/ti/Kconfig | 209 ++++++++++++++++++ sound/soc/ti/Makefile | 44 ++++ sound/soc/{omap => ti}/ams-delta.c | 0 sound/soc/{davinci => ti}/davinci-evm.c | 4 +- sound/soc/{davinci => ti}/davinci-i2s.c | 0 sound/soc/{davinci => ti}/davinci-i2s.h | 0 sound/soc/{davinci => ti}/davinci-mcasp.c | 14 +- sound/soc/{davinci => ti}/davinci-mcasp.h | 0 sound/soc/{davinci => ti}/davinci-vcif.c | 0 sound/soc/{davinci => ti}/edma-pcm.c | 0 sound/soc/{davinci => ti}/edma-pcm.h | 4 +- sound/soc/{omap => ti}/n810.c | 0 sound/soc/{omap => ti}/omap-abe-twl6040.c | 0 sound/soc/{omap => ti}/omap-dmic.c | 0 sound/soc/{omap => ti}/omap-dmic.h | 0 .../omap-hdmi-audio.c => ti/omap-hdmi.c} | 0 sound/soc/{omap => ti}/omap-mcbsp-priv.h | 0 sound/soc/{omap => ti}/omap-mcbsp-st.c | 0 sound/soc/{omap => ti}/omap-mcbsp.c | 0 sound/soc/{omap => ti}/omap-mcbsp.h | 0 sound/soc/{omap => ti}/omap-mcpdm.c | 0 sound/soc/{omap => ti}/omap-mcpdm.h | 0 sound/soc/{omap => ti}/omap-twl4030.c | 0 sound/soc/{omap => ti}/omap3pandora.c | 0 sound/soc/{omap => ti}/osk5912.c | 0 sound/soc/{omap => ti}/rx51.c | 0 sound/soc/{omap => ti}/sdma-pcm.c | 0 sound/soc/{omap => ti}/sdma-pcm.h | 4 +- 42 files changed, 294 insertions(+), 307 deletions(-) delete mode 100644 sound/soc/davinci/Kconfig delete mode 100644 sound/soc/davinci/Makefile delete mode 100644 sound/soc/omap/Kconfig delete mode 100644 sound/soc/omap/Makefile create mode 100644 sound/soc/ti/Kconfig create mode 100644 sound/soc/ti/Makefile rename sound/soc/{omap => ti}/ams-delta.c (100%) rename sound/soc/{davinci => ti}/davinci-evm.c (99%) rename sound/soc/{davinci => ti}/davinci-i2s.c (100%) rename sound/soc/{davinci => ti}/davinci-i2s.h (100%) rename sound/soc/{davinci => ti}/davinci-mcasp.c (99%) rename sound/soc/{davinci => ti}/davinci-mcasp.h (100%) rename sound/soc/{davinci => ti}/davinci-vcif.c (100%) rename sound/soc/{davinci => ti}/edma-pcm.c (100%) rename sound/soc/{davinci => ti}/edma-pcm.h (91%) rename sound/soc/{omap => ti}/n810.c (100%) rename sound/soc/{omap => ti}/omap-abe-twl6040.c (100%) rename sound/soc/{omap => ti}/omap-dmic.c (100%) rename sound/soc/{omap => ti}/omap-dmic.h (100%) rename sound/soc/{omap/omap-hdmi-audio.c => ti/omap-hdmi.c} (100%) rename sound/soc/{omap => ti}/omap-mcbsp-priv.h (100%) rename sound/soc/{omap => ti}/omap-mcbsp-st.c (100%) rename sound/soc/{omap => ti}/omap-mcbsp.c (100%) rename sound/soc/{omap => ti}/omap-mcbsp.h (100%) rename sound/soc/{omap => ti}/omap-mcpdm.c (100%) rename sound/soc/{omap => ti}/omap-mcpdm.h (100%) rename sound/soc/{omap => ti}/omap-twl4030.c (100%) rename sound/soc/{omap => ti}/omap3pandora.c (100%) rename sound/soc/{omap => ti}/osk5912.c (100%) rename sound/soc/{omap => ti}/rx51.c (100%) rename sound/soc/{omap => ti}/sdma-pcm.c (100%) rename sound/soc/{omap => ti}/sdma-pcm.h (85%) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] alsa-lib: commit ce2095c41f2891c51f5dbd28e0317200314c5a75 breaks speaker-test/pulseaudio
by Diego Viola 19 Dec '18

19 Dec '18

Hello Takashi, I currently have an issue with speaker-test and pulseaudio, and the root of the problem seems to be a regression in alsa-lib. The issue occurs when I run speaker-test, it creates endless output with no sound, and the output does not stop until I hit ctrl-c. I've been investigating about this issue and concluded (through a git bisect) that this is a regression in alsa-lib, and this is the bad commit that introduces the bug: commit ce2095c41f2891c51f5dbd28e0317200314c5a75 Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Mar 29 09:51:46 2018 +0200 pcm: ioplug: Implement proper drain behavior This patch fixes the draining behavior of ioplug in the following ways: - When no draining ioplug callback is defined, implement the draining loop using snd_pcm_wait*() and sync with the drain finishes. This is equivalent with the implementation in the kernel write(). Similarly as in kernel code, for non-blocking mode, it returns immediately after setting DRAINING state. - The hw_ptr update function checks the PCM state and stops the stream if the draining finishes. - When draining ioplug callback is defined, leave the whole draining operation to it. The callback is supposed to return -EAGAIN for non-blocking case, too. - When an error happens during draining, it drops the stream, for a safety reason. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> I've sanity checked and can confirm that going back to the previous commit fixes the issue. My original bug report on the pulseaudio bug tracker: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/issues/593 Thanks, and please let me know if you need more information. Regards, Diego

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[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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