October 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: codecs: msm8916-wcd-analog: fix micbias level
by Jean-François Têtu 04 Oct '17

04 Oct '17

The macro used to set the microphone bias level causes the snd_soc_write() call to overwrite other fields in the CDC_A_MICB_1_VAL register. The macro also does not return the proper level value to use. This fixes this by preserving all bits from the register that are not the level while setting the level. Signed-off-by: Jean-François Têtu <jean-francois.tetu(a)savoirfairelinux.com> --- sound/soc/codecs/msm8916-wcd-analog.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index f562f2d..e1534fe 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -104,7 +104,7 @@ #define CDC_A_MICB_1_VAL (0xf141) #define MICB_MIN_VAL 1600 #define MICB_STEP_SIZE 50 -#define MICB_VOLTAGE_REGVAL(v) ((v - MICB_MIN_VAL)/MICB_STEP_SIZE) +#define MICB_VOLTAGE_REGVAL(v) (((v - MICB_MIN_VAL)/MICB_STEP_SIZE) << 3) #define MICB_1_VAL_MICB_OUT_VAL_MASK GENMASK(7, 3) #define MICB_1_VAL_MICB_OUT_VAL_V2P70V ((0x16) << 3) #define MICB_1_VAL_MICB_OUT_VAL_V1P80V ((0x4) << 3) @@ -349,8 +349,9 @@ static void pm8916_wcd_analog_micbias_enable(struct snd_soc_codec *codec) | MICB_1_CTL_EXT_PRECHARG_EN_ENABLE); if (wcd->micbias_mv) { - snd_soc_write(codec, CDC_A_MICB_1_VAL, - MICB_VOLTAGE_REGVAL(wcd->micbias_mv)); + snd_soc_update_bits(codec, CDC_A_MICB_1_VAL, + MICB_1_VAL_MICB_OUT_VAL_MASK, + MICB_VOLTAGE_REGVAL(wcd->micbias_mv)); /* * Special headset needs MICBIAS as 2.7V so wait for * 50 msec for the MICBIAS to reach 2.7 volts. -- 2.7.4

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[alsa-devel] [PATCH 0/6 v3] ASoC: rsnd: add MIX Volume Ramp support
by Kuninori Morimoto 04 Oct '17

04 Oct '17

Hi Mark These are v3 of MIX Volume Ramp support on Renesas R-Car sound. 0001 - 0005 are prepare/cleanup patch for it. v2 -> v3 - tidyup typo: INITIALIZEED -> INITIALIZED - flags macro Kuninori Morimoto (6): ASoC: rsnd: add generic rsnd_flags_xxx() macro ASoC: rsnd: use generic rsnd_flags_xxx() macro on ADG ASoC: rsnd: DVC kctrl sets once ASoC: rsnd: CTU kctrl sets once ASoC: rsnd: makes volume ramp rate list generic ASoC: rsnd: add MIX Volume Ramp support sound/soc/sh/rcar/adg.c | 13 ++-- sound/soc/sh/rcar/core.c | 27 ++++++++ sound/soc/sh/rcar/ctu.c | 8 +++ sound/soc/sh/rcar/dvc.c | 41 ++++-------- sound/soc/sh/rcar/mix.c | 158 +++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/sh/rcar/rsnd.h | 12 +++- sound/soc/sh/rcar/ssi.c | 27 ++++---- 7 files changed, 227 insertions(+), 59 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/6] ASoC: rsnd: add MIX Volume Ramp support
by Kuninori Morimoto 04 Oct '17

04 Oct '17

Hi Mark These adds MIX Volume Ramp support on Renesas R-Car sound. 0001 - 0005 are prepare/cleanup patch for it. Kuninori Morimoto (6): ASoC: rsnd: add generic rsnd_flags_xxx() macro ASoC: rsnd: use generic rsnd_flags_xxx() macro on ADG ASoC: rsnd: DVC kctrl sets once ASoC: rsnd: CTU kctrl sets once ASoC: rsnd: makes volume ramp rate list generic ASoC: rsnd: add MIX Volume Ramp support sound/soc/sh/rcar/adg.c | 13 ++-- sound/soc/sh/rcar/core.c | 27 ++++++++ sound/soc/sh/rcar/ctu.c | 8 +++ sound/soc/sh/rcar/dvc.c | 41 ++++-------- sound/soc/sh/rcar/mix.c | 158 +++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/sh/rcar/rsnd.h | 12 +++- sound/soc/sh/rcar/ssi.c | 27 ++++---- 7 files changed, 227 insertions(+), 59 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/6 v2] ASoC: rsnd: add MIX Volume Ramp support
by Kuninori Morimoto 04 Oct '17

04 Oct '17

Hi Mark These are v2 of MIX Volume Ramp support on Renesas R-Car sound. 0001 - 0005 are prepare/cleanup patch for it. Kuninori Morimoto (6): ASoC: rsnd: add generic rsnd_flags_xxx() macro ASoC: rsnd: use generic rsnd_flags_xxx() macro on ADG ASoC: rsnd: DVC kctrl sets once ASoC: rsnd: CTU kctrl sets once ASoC: rsnd: makes volume ramp rate list generic ASoC: rsnd: add MIX Volume Ramp support sound/soc/sh/rcar/adg.c | 13 ++-- sound/soc/sh/rcar/core.c | 27 ++++++++ sound/soc/sh/rcar/ctu.c | 8 +++ sound/soc/sh/rcar/dvc.c | 41 ++++-------- sound/soc/sh/rcar/mix.c | 158 +++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/sh/rcar/rsnd.h | 12 +++- sound/soc/sh/rcar/ssi.c | 27 ++++---- 7 files changed, 227 insertions(+), 59 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/9] ASoC: cygnus: Various improvements and fixes
by Lori Hikichi 04 Oct '17

04 Oct '17

This patch series contains an number of improvements and refinements to the driver. There is also a fix for a problem when transferring four or more channels in TDM mode. Lori Hikichi (9): ASoC: cygnus: Add support for 384kHz frame rates ASoC: cygnus: Update bindings for audio clock changes ASoC: cygnus: Allow each port to select its clock source ASoC: cygnus: Only enable MCLK pins when in use ASoC: cygnus: Remove support for 8 bit audio and for mono ASoc: cygnus: Fix problems with multichannel transfers ASoC: cygnus: Remove set_fmt from SPDIF dai ops ASoC: cygnus: Add EXPORT_SYMBOL for helper function ASoC: cygnus: Tidy up of structure access .../bindings/sound/brcm,cygnus-audio.txt | 70 +- sound/soc/bcm/cygnus-pcm.c | 50 +- sound/soc/bcm/cygnus-ssp.c | 1336 ++++++++++---------- sound/soc/bcm/cygnus-ssp.h | 51 +- 4 files changed, 739 insertions(+), 768 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v1 0/3] Minor updates to driver
by Lori Hikichi 04 Oct '17

04 Oct '17

This patchset contains a few small updates to the DAI driver. Lori Hikichi (3): ASoC: cygnus: Add EXPORT_SYMBOL for helper function ASoC: cygnus: Remove set_fmt from SPDIF dai ops ASoC: cygnus: Remove support for 8 bit audio and for mono sound/soc/bcm/cygnus-ssp.c | 32 +++++++++++++------------------- 1 file changed, 13 insertions(+), 19 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/5] ASoC: soc-core: component functions cleanup
by Kuninori Morimoto 04 Oct '17

04 Oct '17

Hi Mark These are component functions customize. And is preparation for platform replace to component. I'm posting platform replace patch-set [RFC] to ML. These are preparation but independent from it. Kuninori Morimoto (5): ASoC: soc-core: rename "cmpnt" to "component" ASoC: soc-core: add component lookup functions ASoC: soc-core: add component /unregister_exp/remove functions ASoC: soc-core: add snd_soc_add_component() ASoC: soc-devres: use expanded unregister on devm register component include/sound/soc.h | 16 ++++++-- sound/soc/soc-core.c | 102 ++++++++++++++++++++++++++++++++++--------------- sound/soc/soc-devres.c | 6 +-- 3 files changed, 87 insertions(+), 37 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: codecs: msm8916-wcd-analog: fix module autoload
by Nicolas Dechesne 03 Oct '17

03 Oct '17

If the driver is built as a module, autoload won't work because the module alias information is not filled. So user-space can't match the registered device with the corresponding module. Export the module alias information using the MODULE_DEVICE_TABLE() macro. Before this patch: $ modinfo snd_soc_msm8916_analog | grep alias $ After this patch: $ modinfo snd_soc_msm8916_analog | grep alias alias: of:N*T*Cqcom,pm8916-wcd-analog-codecC* alias: of:N*T*Cqcom,pm8916-wcd-analog-codec Signed-off-by: Nicolas Dechesne <nicolas.dechesne(a)linaro.org> --- sound/soc/codecs/msm8916-wcd-analog.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 549c269acc7d..d79e5b35e003 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -1241,6 +1241,8 @@ static const struct of_device_id pm8916_wcd_analog_spmi_match_table[] = { { } }; +MODULE_DEVICE_TABLE(of, pm8916_wcd_analog_spmi_match_table); + static struct platform_driver pm8916_wcd_analog_spmi_driver = { .driver = { .name = "qcom,pm8916-wcd-spmi-codec", -- 2.11.0

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[alsa-devel] "alsactl store" fails to save Echo Gina3G mixer
by S. Christian Collins 03 Oct '17

03 Oct '17

Greetings, I have two PCI sound cards in my PC: 1. Sound Blaster Audigy 2 ZS Gold (primary card) 2. Echo Gina3G If I save the mixer for the Audigy 2 via: alsactl -f ~/asound0.state store 0 ...the Audigy 2's mixer is properly saved. However, when I try to save the Gina3G mixer: alsactl -f ~/asound1.state store 1 ...I get the following error: alsactl: control.c:2513: snd_ctl_elem_value_get_integer: Assertion `idx < sizeof(obj->value.integer.value) / sizeof(obj->value.integer.value[0])' failed. Aborted I have tried to find any info on this error online, but to no avail. Does anybody know what is going on here? ** My System ** OS: KDE Neon 5.10.5 64-bit (Plasma Desktop 5.10.5, KDE Frameworks 5.37.0, Qt 5.9.1) Linux Kernel: 4.10.0.32-generic Motherboard: ASRock X58 Extreme3 (Intel X58 chipset) CPU: Intel Core i7 930 (2.8 GHz quad-core) RAM: 12GB DDR3 Video: NVIDIA GeForce GTX 970 w/ 4GB RAM (PCI Express) Sound Card #1: Sound Blaster Audigy 2 ZS Gold Sound Card #2: Echo Gina3G NVIDIA video driver: 375.66

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[alsa-devel] Applied "ASoC: codecs: msm8916-wcd-analog: fix module autoload" to the asoc tree
by Mark Brown 03 Oct '17

03 Oct '17

The patch ASoC: codecs: msm8916-wcd-analog: fix module autoload has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 46d69e141d479585c105a4d5b2337cd2ce6967e5 Mon Sep 17 00:00:00 2001 From: Nicolas Dechesne <nicolas.dechesne(a)linaro.org> Date: Tue, 3 Oct 2017 11:49:51 +0200 Subject: [PATCH] ASoC: codecs: msm8916-wcd-analog: fix module autoload If the driver is built as a module, autoload won't work because the module alias information is not filled. So user-space can't match the registered device with the corresponding module. Export the module alias information using the MODULE_DEVICE_TABLE() macro. Before this patch: $ modinfo snd_soc_msm8916_analog | grep alias $ After this patch: $ modinfo snd_soc_msm8916_analog | grep alias alias: of:N*T*Cqcom,pm8916-wcd-analog-codecC* alias: of:N*T*Cqcom,pm8916-wcd-analog-codec Signed-off-by: Nicolas Dechesne <nicolas.dechesne(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/msm8916-wcd-analog.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index f562f2d86907..3593c578e3e7 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -1239,6 +1239,8 @@ static const struct of_device_id pm8916_wcd_analog_spmi_match_table[] = { { } }; +MODULE_DEVICE_TABLE(of, pm8916_wcd_analog_spmi_match_table); + static struct platform_driver pm8916_wcd_analog_spmi_driver = { .driver = { .name = "qcom,pm8916-wcd-spmi-codec", -- 2.14.1

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[alsa-devel] [PATCH] ASoC: rcar: skip disabled-SSI nodes
by Kuninori Morimoto 03 Oct '17

03 Oct '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> The current device tree representation of the R-Car SSI assumes that they are numbered consecutively, starting from 0. Alas, this is not the case with the R8A77995 (D3) SoC which SSI1/SSI2 aren't present. In order to keep the existing device trees working, I'm suggesting to use a disabled node for SSI0/SSI1. Teach the SSI probe to just skip disabled nodes. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index fffc07e..03c2a1f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1112,6 +1112,9 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) i = 0; for_each_child_of_node(node, np) { + if (!of_device_is_available(np)) + goto skip; + ssi = rsnd_ssi_get(priv, i); snprintf(name, RSND_SSI_NAME_SIZE, "%s.%d", @@ -1148,7 +1151,7 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) of_node_put(np); goto rsnd_ssi_probe_done; } - +skip: i++; } -- 1.9.1

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[alsa-devel] [alsa-lib][PATCH] test/seq-decoder: enable timestamping for external subscribers
by Antonio Ospite 03 Oct '17

03 Oct '17

Events sent by external clients subscribed to the input port are not timestamped. This inconsistent behavior may surprise newbies who look at seq-decoder as a reference example. See the example below using "vkeybd --addr 128:0" to connect to seq-decoder, the events sent by vkeybd are on a different queue with no timestamps: ... EVENT>>> Type = 66, flags = 0x0, time = 0 ticks Source = 0.1, dest = 128.0, queue = 253 Event = Port Subscribed; 129:0 -> 128:0 EVENT>>> Type = 66, flags = 0x1, time = 4.829712627 Source = 0.1, dest = 128.0, queue = 0 Event = Port Subscribed; 129:0 -> 128:0 EVENT>>> Type = 10, flags = 0x0, time = 0 ticks Source = 129.0, dest = 128.0, queue = 253 Event = Controller; ch=0, param=0, value=0 EVENT>>> Type = 11, flags = 0x0, time = 0 ticks Source = 129.0, dest = 128.0, queue = 253 Event = Program Change; ch=0, program=0 ... After the change events are on the main queue and are timestamped: ... EVENT>>> Type = 66, flags = 0x1, time = 4.280907223 Source = 0.1, dest = 128.0, queue = 0 Event = Port Subscribed; 129:0 -> 128:0 EVENT>>> Type = 66, flags = 0x1, time = 4.280912063 Source = 0.1, dest = 128.0, queue = 0 Event = Port Subscribed; 129:0 -> 128:0 EVENT>>> Type = 10, flags = 0x1, time = 4.280990702 Source = 129.0, dest = 128.0, queue = 0 Event = Controller; ch=0, param=0, value=0 EVENT>>> Type = 11, flags = 0x1, time = 4.280994862 Source = 129.0, dest = 128.0, queue = 0 Event = Program Change; ch=0, program=0 ... Signed-off-by: Antonio Ospite <ao2(a)ao2.it> --- Hi, while looking into improving the alsamidisrc[1,2] GStreamer element, I took a better look at what the sequencer API could do and started using test/seq-decoder.c as a reference example. I noticed the issue discussed above and I tried to fix it in order to have a more consistent behavior of event timestamping. I think the new behavior is less surprising for newbies but I may still be missing something, please let me know if I am. Thanks, Antonio [1] https://cgit.freedesktop.org/gstreamer/gst-plugins-base/tree/ext/alsa/gstal… [2] https://bugzilla.gnome.org/show_bug.cgi?id=787683 test/seq-decoder.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/test/seq-decoder.c b/test/seq-decoder.c index b110e98d..38755553 100644 --- a/test/seq-decoder.c +++ b/test/seq-decoder.c @@ -283,6 +283,12 @@ void event_decoder(snd_seq_t *handle, int argc, char *argv[]) snd_seq_port_info_set_name(pinfo, "Input"); snd_seq_port_info_set_type(pinfo, SND_SEQ_PORT_TYPE_MIDI_GENERIC); snd_seq_port_info_set_capability(pinfo, SND_SEQ_PORT_CAP_WRITE | SND_SEQ_PORT_CAP_READ | SND_SEQ_PORT_CAP_SUBS_WRITE); + + /* Enable timestamping for events sent by external subscribers. */ + snd_seq_port_info_set_timestamping(pinfo, 1); + snd_seq_port_info_set_timestamp_real(pinfo, 1); + snd_seq_port_info_set_timestamp_queue(pinfo, queue); + if ((err = snd_seq_create_port(handle, pinfo)) < 0) { fprintf(stderr, "Cannot create input port: %s\n", snd_strerror(err)); return; -- 2.14.1

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[alsa-devel] [alsa-lib][PATCH] seq: fix snd_seq_set_queue_tempo() usage example in the documentation
by Antonio Ospite 03 Oct '17

03 Oct '17

snd_seq_set_queue_tempo() requires a queue id as the second argument, fix the example in documentation to reflect that. Also add the queue id as an argument of the set_tempo() function, just to keep the whole example compilable. Signed-off-by: Antonio Ospite <ao2(a)ao2.it> --- src/seq/seq.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/seq/seq.c b/src/seq/seq.c index d5ed1c6a..808c7915 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -452,13 +452,13 @@ For setting these tempo parameters, use #snd_seq_queue_tempo_t record. For example, to set the tempo of the queue <code>q</code> to 48 PPQ, 60 BPM, \code -void set_tempo(snd_seq_t *handle) +void set_tempo(snd_seq_t *handle, int queue) { snd_seq_queue_tempo_t *tempo; snd_seq_queue_tempo_alloca(&tempo); snd_seq_queue_tempo_set_tempo(tempo, 1000000); // 60 BPM snd_seq_queue_tempo_set_ppq(tempo, 48); // 48 PPQ - snd_seq_set_queue_tempo(handle, tempo); + snd_seq_set_queue_tempo(handle, queue, tempo); } \endcode -- 2.14.2

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[alsa-devel] ASoC Sound Driver I2S over LPE, Intel Baytrail SoC. MinnowBoard Max
by Michael Pogrebinsky 02 Oct '17

02 Oct '17

Hi all, I have the MinnowBoard Max with Baytrail Atom (ValleyView) SoC. I'm attempting to write an Alsa (ASoC) driver for my sound card to work with this board. PCM should work through I2S for this sound card I have. >From my understanding Baytrail's I2S goes over LPE. I read this thread:https://www.mail-archive.com/elinux-minnowboard@lists.elinux.org/msg… So I understand that I need the 3 components for my ASoC driver. However I am having trouble at understanding how to communicate with the LPE to get the I2S working. Here is what I have done so far: I am running Ubuntu 14.04 LTS, kernel : 4.3.0-040300rc4 First of all I downloaded the firmware fw_sst_0f28.bin-48kHz_i2s_master fromhttps://kernel.googlesource.com/pub/scm/linux/kernel/git/firmware/linux… And put it /lib/firmware/intel/. Later I went to the Bios and enabled LPE in PCI mode. I do see the LPE controller now, but it shows as UNCLAIMED. Later I looked at the sound/soc/inte/baytrail/ and from my understanding in order for me to get the I2S working I need to load this module so I did modprobe snd-soc-sst-baytrail-pcm. Later I wrote a simple codec and machine driver (looking at the example of sound/soc/intel/boards/byt_max98090) however everytime I attempt to register the sound card with snd_soc_register_card, I get the following error message in dmesg : "ASoC: CPU DAI baytrail-pcm-audio not registered". >From this I deduct that although I loaded the snd-soc-sst-baytrail-pcm module, the driver has not actually been loaded. I'm clueless on how to proceed from here. I would highly appreciate your help.

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[alsa-devel] usb/sound/usx2y: warning in usb_stream_new/__alloc_pages_slowpath
by Andrey Konovalov 02 Oct '17

02 Oct '17

Hi! I've got the following report while fuzzing the kernel with syzkaller. On commit ebb2c2437d8008d46796902ff390653822af6cc4 (Sep 18). ------------[ cut here ]------------ WARNING: CPU: 1 PID: 1846 at mm/page_alloc.c:3883 __alloc_pages_slowpath+0x1ef2/0x2d70 Modules linked in: CPU: 1 PID: 1846 Comm: kworker/1:2 Not tainted 4.14.0-rc1-42251-gebb2c2437d80 #215 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Workqueue: usb_hub_wq hub_event task: ffff8800643c18c0 task.stack: ffff88006b658000 RIP: 0010:trace_reclaim_retry_zone ./include/trace/events/oom.h:31 RIP: 0010:should_reclaim_retry mm/page_alloc.c:3783 RIP: 0010:__alloc_pages_slowpath+0x1ef2/0x2d70 mm/page_alloc.c:4039 RSP: 0018:ffff88006b65d938 EFLAGS: 00010246 RAX: 00000000ffffa666 RBX: 00000000014000c0 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 0000000000000034 RDI: 000000000140c0c0 RBP: ffff88006b65e088 R08: 0000000000000000 R09: fffffffffff00f88 R10: 0000000000000000 R11: 0000000000000085 R12: ffff88006b65e130 R13: ffff88006b65e270 R14: ffff88006b65e0f0 R15: 000000000140c0c0 FS: 0000000000000000(0000) GS:ffff88006c900000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000020829000 CR3: 0000000063cce000 CR4: 00000000000006e0 Call Trace: __alloc_pages_nodemask+0x921/0xf70 mm/page_alloc.c:4217 alloc_pages_current+0xbb/0x1f0 mm/mempolicy.c:2035 alloc_pages ./include/linux/gfp.h:505 __get_free_pages+0x14/0x50 mm/page_alloc.c:4248 usb_stream_new+0x50f/0x9f0 sound/usb/usx2y/usb_stream.c:214 us122l_start+0xb0/0x250 sound/usb/usx2y/us122l.c:355 us122l_create_card sound/usb/usx2y/us122l.c:502 us122l_usb_probe sound/usb/usx2y/us122l.c:588 snd_us122l_probe+0xa62/0x15d0 sound/usb/usx2y/us122l.c:623 usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_set_configuration+0x104e/0x1870 drivers/usb/core/message.c:1932 generic_probe+0x73/0xe0 drivers/usb/core/generic.c:174 usb_probe_device+0xaf/0xe0 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_new_device+0x7b8/0x1020 drivers/usb/core/hub.c:2457 hub_port_connect drivers/usb/core/hub.c:4903 hub_port_connect_change drivers/usb/core/hub.c:5009 port_event drivers/usb/core/hub.c:5115 hub_event+0x194d/0x3740 drivers/usb/core/hub.c:5195 process_one_work+0xc7f/0x1db0 kernel/workqueue.c:2119 worker_thread+0x221/0x1850 kernel/workqueue.c:2253 kthread+0x3a1/0x470 kernel/kthread.c:231 ret_from_fork+0x2a/0x40 arch/x86/entry/entry_64.S:431 Code: be 47 00 00 00 48 c7 c7 20 42 8b 85 c6 05 5e 35 99 05 01 e8 91 1f cb ff 48 8b 8d d8 f8 ff ff 44 8b 9d 50 f9 ff ff e9 57 fc ff ff <0f> ff e9 79 e3 ff ff 0f ff c7 85 e8 f8 ff ff 00 00 00 00 89 d8 ---[ end trace 78800971cbd5f94f ]---

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[alsa-devel] [PATCH] ASoC: rockchip: Fix wrong allocation size of dapm routes
by Jeffy Chen 02 Oct '17

02 Oct '17

The allocation size of dapm routes is wrong, correct it. Fixes: d9f9c167edae ("ASoC: rockchip: Init dapm routes dynamically") Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> --- sound/soc/rockchip/rk3399_gru_sound.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 30eed83e8a13..008452e55ef8 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -493,14 +493,18 @@ static int rockchip_sound_of_parse_dais(struct device *dev, struct device_node *np_codec; struct snd_soc_dai_link *dai; struct snd_soc_dapm_route *routes; - int i, index; + int i, index, max_num_routes; card->dai_link = devm_kzalloc(dev, sizeof(rockchip_dais), GFP_KERNEL); if (!card->dai_link) return -ENOMEM; - routes = devm_kzalloc(dev, sizeof(rockchip_routes), + max_num_routes = 0; + for (i = 0; i < ARRAY_SIZE(rockchip_dais); i++) + max_num_routes += rockchip_routes[i].num_routes; + + routes = devm_kzalloc(dev, max_num_routes * sizeof(*routes), GFP_KERNEL); if (!routes) return -ENOMEM; -- 2.11.0

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[alsa-devel] [PATCH] ALSA: line6: make snd_pcm_ops const
by Bhumika Goyal 02 Oct '17

02 Oct '17

Make these const as they are only passed to a const argument of the function snd_pcm_set_ops in the file referencing them. Also, add const to the declaration in the headers. Structures found using Coccinelle and changes done by hand. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/usb/line6/capture.c | 2 +- sound/usb/line6/capture.h | 2 +- sound/usb/line6/playback.c | 2 +- sound/usb/line6/playback.h | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/line6/capture.c b/sound/usb/line6/capture.c index 7c81256..947d616 100644 --- a/sound/usb/line6/capture.c +++ b/sound/usb/line6/capture.c @@ -248,7 +248,7 @@ static int snd_line6_capture_close(struct snd_pcm_substream *substream) } /* capture operators */ -struct snd_pcm_ops snd_line6_capture_ops = { +const struct snd_pcm_ops snd_line6_capture_ops = { .open = snd_line6_capture_open, .close = snd_line6_capture_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/usb/line6/capture.h b/sound/usb/line6/capture.h index 890b21b..b67ccc3 100644 --- a/sound/usb/line6/capture.h +++ b/sound/usb/line6/capture.h @@ -17,7 +17,7 @@ #include "driver.h" #include "pcm.h" -extern struct snd_pcm_ops snd_line6_capture_ops; +extern const struct snd_pcm_ops snd_line6_capture_ops; extern void line6_capture_copy(struct snd_line6_pcm *line6pcm, char *fbuf, int fsize); diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c index 812d181..819e9b2 100644 --- a/sound/usb/line6/playback.c +++ b/sound/usb/line6/playback.c @@ -393,7 +393,7 @@ static int snd_line6_playback_close(struct snd_pcm_substream *substream) } /* playback operators */ -struct snd_pcm_ops snd_line6_playback_ops = { +const struct snd_pcm_ops snd_line6_playback_ops = { .open = snd_line6_playback_open, .close = snd_line6_playback_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/usb/line6/playback.h b/sound/usb/line6/playback.h index 51fce29..d8d3b8a 100644 --- a/sound/usb/line6/playback.h +++ b/sound/usb/line6/playback.h @@ -27,7 +27,7 @@ */ #define USE_CLEAR_BUFFER_WORKAROUND 1 -extern struct snd_pcm_ops snd_line6_playback_ops; +extern const struct snd_pcm_ops snd_line6_playback_ops; extern int line6_create_audio_out_urbs(struct snd_line6_pcm *line6pcm); extern int line6_submit_audio_out_all_urbs(struct snd_line6_pcm *line6pcm); -- 1.9.1

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[alsa-devel] [PATCH] Revert "ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members"
by Takashi Sakamoto 02 Oct '17

02 Oct '17

This reverts commit 275353bb684e to fix a regression which can abort 'alsactl' program in alsa-utils due to assertion in alsa-lib. alsactl: control.c:2513: snd_ctl_elem_value_get_integer: Assertion `idx < sizeof(obj->value.integer.value) / sizeof(obj->value.integer.value[0])' failed. alsactl: control.c:2976: snd_ctl_elem_value_get_integer: Assertion `idx < ARRAY_SIZE(obj->value.integer.value)' failed. This commit is a band-aid. In a point of usage of ALSA control interface, the drivers still bring an issue that they prevent userspace applications to have a consistent way to parse each levels of the dimension information via ALSA control interface. Let me investigate this issue. Current implementation of the drivers have three control element sets with dimension information: * 'Monitor Mixer Volume' (type: integer) * 'VMixer Volume' (type: integer) * 'VU-meters' (type: boolean) Although the number of elements named as 'Monitor Mixer Volume' differs depending on drivers in this group, it can be calculated by macros defined by each driver (= (BX_NUM - BX_ANALOG_IN) * BX_ANALOG_IN). Each of the elements has one member for value and has dimension information with 2 levels (= BX_ANALOG_IN * (BX_NUM - BX_ANALOG_IN)). For these elements, userspace applications are expected to handle the dimension information so that all of the elements construct a matrix where the number of rows and columns are represented by the dimension information. The same way is applied to elements named as 'VMixer Volume'. The number of these elements can also be calculated by macros defined by each drivers (= PX_ANALOG_IN * BX_ANALOG_IN). Each of the element has one member for value and has dimension information with 2 levels (= BX_ANALOG_IN * PX_ANALOG_IN). All of the elements construct a matrix with the dimension information. An element named as 'VU-meters' gets a different way in a point of dimension information. The element includes 96 members for value. The element has dimension information with 3 levels (= 3 or 2 * 16 * 2). For this element, userspace applications are expected to handle the dimension information so that all of the members for value construct a matrix where the number of rows and columns are represented by the dimension information. This is different from the way for the former. As a summary, the drivers were not designed to produce a consistent way to parse the dimension information. This makes it hard for general userspace applications such as amixer to parse the information by a consistent way, and actually no userspace applications except for 'echomixer' utilize the dimension information. Additionally, no drivers excluding this group use the information. The reverted commit was written based on the latter way. A commit 860c1994a70a ('ALSA: control: add dimension validator for userspace elements') is written based on the latter way, too. The patch should be reconsider too in the same time to re-define a consistent way to parse the dimension information. Reported-by: Mark Hills <mark(a)xwax.org> Reported-by: S. Christian Collins <s.chriscollins(a)gmail.com> Fixes: 275353bb684e ('ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members') Cc: <stable(a)vger.kernel.org> # v4.8+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/pci/echoaudio/echoaudio.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 7326695bca33..d68f99e076a8 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1272,11 +1272,11 @@ static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol, chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = ECHOGAIN_MAXOUT; uinfo->dimen.d[0] = num_busses_out(chip); uinfo->dimen.d[1] = num_busses_in(chip); - uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1]; return 0; } @@ -1344,11 +1344,11 @@ static int snd_echo_vmixer_info(struct snd_kcontrol *kcontrol, chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = ECHOGAIN_MAXOUT; uinfo->dimen.d[0] = num_busses_out(chip); uinfo->dimen.d[1] = num_pipes_out(chip); - uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1]; return 0; } @@ -1728,6 +1728,7 @@ static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 96; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = 0; #ifdef ECHOCARD_HAS_VMIXER @@ -1737,7 +1738,6 @@ static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol, #endif uinfo->dimen.d[1] = 16; /* 16 channels */ uinfo->dimen.d[2] = 2; /* 0=level, 1=peak */ - uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1] * uinfo->dimen.d[2]; return 0; } -- 2.11.0

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[alsa-devel] [PATCH 0/2] ASoC: img-spdif-in: Add power management
by Ed Blake 02 Oct '17

02 Oct '17

This patch set adds power managment to the img-spdif-in driver. The first patch adds system PM suspend / resume handling and the second patch adds runtime PM. Ed Blake (2): ASoC: img-spdif-in: Add suspend / resume handling ASoC: img-spdif-in: Add runtime PM sound/soc/img/img-spdif-in.c | 111 ++++++++++++++++++++++++++++++++++++++----- 1 file changed, 99 insertions(+), 12 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v1 0/2] Fix first detection issue with headset
by Damien Riegel 02 Oct '17

02 Oct '17

This series fix an issue with headset being misidentified on the very first time a device is plugged in by calling the function that configures the bias when the MBHC is setup. I've split this series into two patches because pm8916_wcd_setup_mbhc needs to be moved a few lines below to avoid forward declaring it. It is overkill but that was not part of the code snippet that Srinivas acked, and I wanted to be extra sure not to misattribute an Acked-by. Damien Riegel (2): ASoC: codecs: msm8916-wcd-analog: move pm8916_wcd_setup_mbhc ASoC: codecs: msm8916-wcd-analog: configure micbias in mbhc setup sound/soc/codecs/msm8916-wcd-analog.c | 94 +++++++++++++++++++---------------- 1 file changed, 50 insertions(+), 44 deletions(-) -- 2.14.1

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[alsa-devel] [PATCH v2] ALSA: hda - program ICT bits to support HBR audio
by Pierre-Louis Bossart 02 Oct '17

02 Oct '17

From: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> On recent Intel platforms (Haswell, Broadwell, Skylake, ApolloLake, KabyLake, ...), the IEC Coding Type (ICT) bitfield in the Digital Converter Control #3 needs to be set explicitly for HDMI/DisplayPort High Bit Rate (HBR) audio playback to work. This was not required in earlier platforms when HBR was first introduced. The ICT bits are defined in Section 7.3.3.9 of the HDaudio 1.0a specification. Since the ICT bitfield was not specified for HDAudio 1.0 devices (before 2009), we only program it on machines more recent than Haswell. We tested that this fix is not needed on Baytrail-I (MinnowBoard Turbot) and believe by extension it also does not apply to Braswell. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=98797 Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- v2: move digi3 verb in Set section, only set ICT for haswell and newer include/sound/hda_verbs.h | 1 + sound/pci/hda/patch_hdmi.c | 21 +++++++++++++++++++++ 2 files changed, 22 insertions(+) diff --git a/include/sound/hda_verbs.h b/include/sound/hda_verbs.h index d0509db6d0ec..d2004d7feb2b 100644 --- a/include/sound/hda_verbs.h +++ b/include/sound/hda_verbs.h @@ -54,6 +54,7 @@ enum { #define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c #define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d #define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e /* unused */ +#define AC_VERB_SET_DIGI_CONVERT_3 0x73e #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f /* f10-f1a: GPIO */ #define AC_VERB_GET_GPIO_DATA 0x0f15 diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 53f9311370de..8566de298ea3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -906,6 +906,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, hda_nid_t pin_nid, u32 stream_tag, int format) { struct hdmi_spec *spec = codec->spec; + unsigned int param; int err; err = spec->ops.pin_hbr_setup(codec, pin_nid, is_hbr_format(format)); @@ -915,6 +916,26 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, return err; } + if (is_haswell_plus(codec)) { + + /* + * on recent platforms IEC Coding Type is required for HBR + * support, read current Digital Converter settings and set + * ICT bitfield if needed. + */ + param = snd_hda_codec_read(codec, cvt_nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); + + param = (param >> 16) & ~(AC_DIG3_ICT); + + /* on recent platforms ICT mode is required for HBR support */ + if (is_hbr_format(format)) + param |= 0x1; + + snd_hda_codec_write(codec, cvt_nid, 0, + AC_VERB_SET_DIGI_CONVERT_3, param); + } + snd_hda_codec_setup_stream(codec, cvt_nid, stream_tag, 0, format); return 0; } -- 2.11.0

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[alsa-devel] [PATCH 0/2] ASoC: soc-core: add missing component functions
by Kuninori Morimoto 02 Oct '17

02 Oct '17

Hi Mark These are resend of one of "replace to component" patch set. These add missing functions which exist in "platforms" but not exist in "component". Kuninori Morimoto (2): ASoC: soc-core: add component lookup functions ASoC: soc-core: add snd_soc_add_component() include/sound/soc.h | 7 ++++++ sound/soc/soc-core.c | 60 ++++++++++++++++++++++++++++++++++++++++++---------- 2 files changed, 56 insertions(+), 11 deletions(-) -- 1.9.1

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