September 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v6 1/2] ASoC: samsung: Add DT bindings documentation for TM2 sound subsystem
by Sylwester Nawrocki 02 Sep '16

02 Sep '16

This patch adds DT binding documentation for Exynos5433 based TM2 and TM2E boards sound subsystem. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Rob Herring <robh(a)kernel.org> --- Changes since v5: - none. Changes since v4: - indentation changes. Changes since v2: - none. Changes since initial version: - dropped clocks, clock-names properties, instead properties from the CODEC node will be used, - property renames: 'samsung,model' -> 'model', 'samsung,i2s-controller' -> 'i2s-controller', 'samsung,speaker-amplifier' -> 'audio-amplifier', - added 'audio-codec' property. --- .../bindings/sound/samsung,tm2-audio.txt | 38 ++++++++++++++++++++++ 1 file changed, 38 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt new file mode 100644 index 0000000..94442e5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt @@ -0,0 +1,38 @@ +Samsung Exynos5433 TM2(E) audio complex with WM5110 codec + +Required properties: + + - compatible : "samsung,tm2-audio" + - model : the user-visible name of this sound complex + - audio-codec : the phandle of the wm5110 audio codec node, + as described in ../mfd/arizona.txt + - i2s-controller : the phandle of the I2S controller + - audio-amplifier : the phandle of the MAX98504 amplifier + - samsung,audio-routing : a list of the connections between audio components; + each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source; valid names for sources and sinks are the + WM5110's and MAX98504's pins and the jacks on the + board: HP, SPK, Main Mic, Sub Mic, Third Mic, + Headset Mic + - mic-bias-gpios : GPIO pin that enables the Main Mic bias regulator + + +Example: + +sound { + compatible = "samsung,tm2-audio"; + audio-codec = <&wm5110>; + i2s-controller = <&i2s0>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3 2 0>; + model = "wm5110"; + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "Main Mic", "MICBIAS2", + "IN1R", "Main Mic"; +}; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: ux500: fix spelling mistake "Unsopported" -> "Unsupported"
by Colin King 02 Sep '16

02 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistakes in dev_err messages Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/ux500/ux500_msp_dai.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 6d5698b..b343efd 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -187,7 +187,7 @@ static int setup_clocking(struct snd_soc_dai *dai, default: dev_err(dai->dev, - "%s: Error: Unsopported inversion (fmt = 0x%x)!\n", + "%s: Error: Unsupported inversion (fmt = 0x%x)!\n", __func__, fmt); return -EINVAL; @@ -218,7 +218,7 @@ static int setup_clocking(struct snd_soc_dai *dai, break; default: - dev_err(dai->dev, "%s: Error: Unsopported master (fmt = 0x%x)!\n", + dev_err(dai->dev, "%s: Error: Unsupported master (fmt = 0x%x)!\n", __func__, fmt); return -EINVAL; @@ -374,7 +374,7 @@ static int setup_msp_config(struct snd_pcm_substream *substream, break; default: - dev_err(dai->dev, "%s: Error: Unsopported format (%d)!\n", + dev_err(dai->dev, "%s: Error: Unsupported format (%d)!\n", __func__, fmt); return -EINVAL; } -- 2.9.3

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[alsa-devel] [PATCH 1/3] ALSA: timer: fix NULL pointer dereference in read()/ioctl() race
by Vegard Nossum 02 Sep '16

02 Sep '16

I got this with syzkaller: ================================================================== BUG: KASAN: null-ptr-deref on address 0000000000000020 Read of size 32 by task syz-executor/22519 CPU: 1 PID: 22519 Comm: syz-executor Not tainted 4.8.0-rc2+ #169 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS rel-1.9.3-0-ge2fc41e-prebuilt.qemu-project.org 04/01/2 014 0000000000000001 ffff880111a17a00 ffffffff81f9f141 ffff880111a17a90 ffff880111a17c50 ffff880114584a58 ffff880114584a10 ffff880111a17a80 ffffffff8161fe3f ffff880100000000 ffff880118d74a48 ffff880118d74a68 Call Trace: [<ffffffff81f9f141>] dump_stack+0x83/0xb2 [<ffffffff8161fe3f>] kasan_report_error+0x41f/0x4c0 [<ffffffff8161ff74>] kasan_report+0x34/0x40 [<ffffffff82c84b54>] ? snd_timer_user_read+0x554/0x790 [<ffffffff8161e79e>] check_memory_region+0x13e/0x1a0 [<ffffffff8161e9c1>] kasan_check_read+0x11/0x20 [<ffffffff82c84b54>] snd_timer_user_read+0x554/0x790 [<ffffffff82c84600>] ? snd_timer_user_info_compat.isra.5+0x2b0/0x2b0 [<ffffffff817d0831>] ? proc_fault_inject_write+0x1c1/0x250 [<ffffffff817d0670>] ? next_tgid+0x2a0/0x2a0 [<ffffffff8127c278>] ? do_group_exit+0x108/0x330 [<ffffffff8174653a>] ? fsnotify+0x72a/0xca0 [<ffffffff81674dfe>] __vfs_read+0x10e/0x550 [<ffffffff82c84600>] ? snd_timer_user_info_compat.isra.5+0x2b0/0x2b0 [<ffffffff81674cf0>] ? do_sendfile+0xc50/0xc50 [<ffffffff81745e10>] ? __fsnotify_update_child_dentry_flags+0x60/0x60 [<ffffffff8143fec6>] ? kcov_ioctl+0x56/0x190 [<ffffffff81e5ada2>] ? common_file_perm+0x2e2/0x380 [<ffffffff81746b0e>] ? __fsnotify_parent+0x5e/0x2b0 [<ffffffff81d93536>] ? security_file_permission+0x86/0x1e0 [<ffffffff816728f5>] ? rw_verify_area+0xe5/0x2b0 [<ffffffff81675355>] vfs_read+0x115/0x330 [<ffffffff81676371>] SyS_read+0xd1/0x1a0 [<ffffffff816762a0>] ? vfs_write+0x4b0/0x4b0 [<ffffffff82001c2c>] ? __this_cpu_preempt_check+0x1c/0x20 [<ffffffff8150455a>] ? __context_tracking_exit.part.4+0x3a/0x1e0 [<ffffffff816762a0>] ? vfs_write+0x4b0/0x4b0 [<ffffffff81005524>] do_syscall_64+0x1c4/0x4e0 [<ffffffff810052fc>] ? syscall_return_slowpath+0x16c/0x1d0 [<ffffffff83c3276a>] entry_SYSCALL64_slow_path+0x25/0x25 ================================================================== There are a couple of problems that I can see: - ioctl(SNDRV_TIMER_IOCTL_SELECT), which potentially sets tu->queue/tu->tqueue to NULL on memory allocation failure, so read() would get a NULL pointer dereference like the above splat - the same ioctl() can free tu->queue/to->tqueue which means read() could potentially see (and dereference) the freed pointer We can fix the NULL pointer dereference by not touching the pointers until we have allocated the new memory (similar to what is done in ioctl(SNDRV_TIMER_IOCTL_PARAMS)) and we can fix the use-after-free by taking the ioctl_lock mutex when dereferencing ->queue/->tqueue, since that's always held over all the ioctl() code. Just looking at the code I find it likely that there are more problems here such as tu->qhead pointing outside the buffer if the size is changed concurrently using SNDRV_TIMER_IOCTL_PARAMS. Signed-off-by: Vegard Nossum <vegard.nossum(a)oracle.com> --- sound/core/timer.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 9a6157e..7899e37 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1602,15 +1602,25 @@ static int snd_timer_user_tselect(struct file *file, kfree(tu->tqueue); tu->tqueue = NULL; if (tu->tread) { - tu->tqueue = kmalloc(tu->queue_size * sizeof(struct snd_timer_tread), + struct snd_timer_tread *ttr; + ttr = kmalloc(tu->queue_size * sizeof(struct snd_timer_tread), GFP_KERNEL); - if (tu->tqueue == NULL) + if (ttr) { + kfree(tu->tqueue); + tu->tqueue = ttr; + } else { err = -ENOMEM; + } } else { - tu->queue = kmalloc(tu->queue_size * sizeof(struct snd_timer_read), + struct snd_timer_read *tr; + tr = kmalloc(tu->queue_size * sizeof(struct snd_timer_read), GFP_KERNEL); - if (tu->queue == NULL) + if (tr) { + kfree(tu->queue); + tu->queue = tr; + } else { err = -ENOMEM; + } } if (err < 0) { @@ -1958,6 +1968,7 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, tu->qused--; spin_unlock_irq(&tu->qlock); + mutex_lock(&tu->ioctl_lock); if (tu->tread) { if (copy_to_user(buffer, &tu->tqueue[qhead], sizeof(struct snd_timer_tread))) @@ -1967,6 +1978,7 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, sizeof(struct snd_timer_read))) err = -EFAULT; } + mutex_unlock(&tu->ioctl_lock); spin_lock_irq(&tu->qlock); if (err < 0) -- 2.10.0.rc0.1.g07c9292

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[alsa-devel] [PATCH v2 00/11] ASoC: hdac: Add hdac generic driver
by Subhransu S. Prusty 02 Sep '16

02 Sep '16

HDA devices generically can be modelled with DAPM in ASoC. We would like to create a framework for the HDA devices in ASoC. This will follow HDA standard. This will be registered as generic "virtual" class device (Not implemented in this series). If there is no match found for vendor id and device id, then the class driver will be loaded. For vendor specific devices, vendor quirks will be loaded based on the match. Additional widgets, controls and route will be created in a vendor specific way through vendor ops. This series adds a framework in ASoC to model HDA codecs. HDA widgets are enumerated and one or multiple DAPM widget(s) are created. Connection list is queried for each widget to identify the connection between two endpoints and modelled using DAPM graph. Set of event handlers are defined for each widget type. Based on DAPM events required verbs are sent to program codec. Finally a function is exported to query for the device endpoint configuration to create machine controls. Features yet to be added: - Optional features of widgets will be added using additional dapm widgets or controls - Pin retasking will be done by adding all possible dapm route map - Class driver to load the generic driver if no match found - Jack detection changes in v2: - Addressed review comment from Takashi - Moved the widget_list to hdac_ext_device - helper to fill the widget list is now called from hdac_ext_bus_device_init instead of driver probe - Fixed a NULL pointer dereferencing issue - Reorder patch to fix 'auto build test error' - Playback with hs/spk and amic capture now working for rt286 - And some more cleanup and code refactoring to avoid long function definition Subhransu S. Prusty (11): ALSA: hdac: Add codec helper library ALSA: hda - Add macro to test pin widget's input capability ASoC: hdac: Add a generic hdac driver framework ALSA: hdac: Move 'to_hda_ext_device' helper to core ASoC: hdac: Create DAPM model for HDA widgets ASoC: hdac: Build DAPM graph by querying through widget connection list ASoC: hdac: Register widget event handlers ALSA: hda - macro to get default config device of pin widgets ASoC: hdac: Export API to create machine controls ALSA: hdac: Implement a match function ASoC: hdac: Add coefficient init callback for RT286 include/sound/hdaudio_ext.h | 40 + sound/hda/ext/Makefile | 3 +- sound/hda/ext/hdac_ext_bus.c | 24 + sound/hda/ext/hdac_ext_codec.c | 156 ++++ sound/hda/ext/hdac_ext_codec.h | 26 + sound/hda/local.h | 13 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_generic.c | 1743 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hdac_generic.h | 36 + sound/soc/codecs/hdac_hdmi.c | 8 +- 11 files changed, 2048 insertions(+), 8 deletions(-) create mode 100644 sound/hda/ext/hdac_ext_codec.c create mode 100644 sound/hda/ext/hdac_ext_codec.h create mode 100644 sound/soc/codecs/hdac_generic.c create mode 100644 sound/soc/codecs/hdac_generic.h -- 1.9.1

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[alsa-devel] [PATCH 0/6] constify snd_pcm_ops structures
by Julia Lawall 02 Sep '16

02 Sep '16

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[alsa-devel] [PATCH] pcm_hw: Respect LIBASOUND_THREAD_SAFE env var
by David Henningsson 02 Sep '16

02 Sep '16

Without this patch, there no way to revert to the old behaviour with pcm_hw, so currently there will always be some extra locking. This seems to be an oversight? Reported-by: Samuel Thibault <samuel.thibault(a)ens-lyon.org> Signed-off-by: David Henningsson <diwic(a)ubuntu.com> --- src/pcm/pcm_hw.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 3a5634c..e6a8a4f 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -1514,7 +1514,8 @@ int snd_pcm_hw_open_fd(snd_pcm_t **pcmp, const char *name, pcm->poll_events = info.stream == SND_PCM_STREAM_PLAYBACK ? POLLOUT : POLLIN; pcm->tstamp_type = tstamp_type; #ifdef THREAD_SAFE_API - pcm->thread_safe = 1; + if (pcm->thread_safe != -1) + pcm->thread_safe = 1; #endif ret = snd_pcm_hw_mmap_status(pcm); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: arizona: Wait for resume before enabling FLL
by Richard Fitzgerald 01 Sep '16

01 Sep '16

When enabling an FLL use pm_runtime_get_sync() instead of pm_runtime_get() to ensure that all the register settings have been written out and the codec is powered-up before we write the enable bit. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ecfdbfc..8b8f2d2 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2268,7 +2268,7 @@ static int arizona_enable_fll(struct arizona_fll *fll) ARIZONA_FLL1_SYNC_BW); if (!already_enabled) - pm_runtime_get(arizona->dev); + pm_runtime_get_sync(arizona->dev); regmap_update_bits_async(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rockchip: use SPI dependency for rt5514
by Arnd Bergmann 01 Sep '16

01 Sep '16

The rk3399 scans the spi_bus_type to find the rt5514 driver, but does not actually have a Kconfig dependency on SPI, so we can end up with a link failure: sound/soc/codecs/snd-soc-rt5514-spi.o: In function `rt5514_spi_driver_init': rt5514-spi.c:(.init.text+0x14): undefined reference to `__spi_register_driver' sound/soc/codecs/snd-soc-rt5514-spi.o: In function `rt5514_spi_burst_read': rt5514-spi.c:(.text.rt5514_spi_burst_read+0x18c): undefined reference to `spi_sync' sound/soc/codecs/snd-soc-rt5514-spi.o: In function `rt5514_spi_burst_write': rt5514-spi.c:(.text.rt5514_spi_burst_write+0x1b4): undefined reference to `spi_sync' sound/soc/rockchip/snd-soc-rk3399-gru-sound.o: In function `rockchip_sound_probe': rk3399_gru_sound.c:(.text.rockchip_sound_probe+0x128): undefined reference to `spi_bus_type' This adds the missing dependency. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: c6eac8a36a84 ("ASoC: rockchip: Add machine driver for RK3399 GRU Boards") --- sound/soc/rockchip/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index d82d763d4854..c783f9a22595 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -44,7 +44,7 @@ config SND_SOC_ROCKCHIP_RT5645 config SND_SOC_RK3399_GRU_SOUND tristate "ASoC support multiple codecs for Rockchip RK3399 GRU boards" - depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP && SPI select SND_SOC_ROCKCHIP_I2S select SND_SOC_MAX98357A select SND_SOC_RT5514 -- 2.9.0

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[alsa-devel] [PATCH v9 0/2] sun4i-codec: Add Line-In, FM-In, Mic 2, Capture Source, Differential Line-In
by Danny Milosavljevic 01 Sep '16

01 Sep '16

This patch adds support for some mixer controls: - Line-In - FM-In - Mic 2 - Capture Source - Differential Line-In v9 changes compared to v8 are: - added Line Differential Capture Switch. - split Capture Source into Left Capture Select, Right Capture Select. - added Line Capture Volume. - rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field it's used in. - rename "Line-In" to "Line". - rename "Power Amplifier Playback Volume" to "Headphone Playback Volume". v8 changes compared to v7 are: - fixed the routes for line and mic capturing. v7 changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top. - sun7i-specific things (A20-specific things) are now less grouped-together. - rename "Power Amplifier Volume" to "Power Amplifier Playback Volume". v6 changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. v5 changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. v4 changes compared to v3 are: - names of the input are not uppercase anymore. - bit index constants are now named as in the A20 user manual v1.4. - added Mic1-In, Mac2-In. - added Mic1 and Mic2 Pre-Amplifiers. v3 changes compared to v2 are: - added DAPM routes. v2 changes compared to v1 are: - moved Line-In and FM-In playback switches to their respective sun4i_codec_*_mixer_controls. v1 changes: - added linein, fmin output volumes and switches. Danny Milosavljevic (2): ASoC: sun4i-codec: Distinguish sun4i from sun7i Add mixer controls: Line-In, FM-In, Mic 2, Capture Source, Differential Line-In. sound/soc/sunxi/sun4i-codec.c | 278 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 259 insertions(+), 19 deletions(-)

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[alsa-devel] CIP Discontinuity with snd-firewire-digi00x
by lin-alsa＠noblejury.com 01 Sep '16

01 Sep '16

All, I've been working to set up a Digi002 Rack mixer with Jack using the Alsa backend. Unfortunately, the mixer seems to desync or something after a few minutes - hours. dmesg gives me `snd-firewire-digi00x fw1.0: Detect discontinuity of CIP: XX XX` with XX XX replaced with various hex pairs. I'm not sure how to debug this, and any help would be much appreciated. Also, cat /proc/asound/card1/firewire/clock gives `Optical mode: adat Sampling Rate: 88200 Clock Source: internal` so I don't think it's an external clock issue. Lin

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