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

[alsa-devel] SEGFault when optininal snd_ctl_ext_callback::read_event() not set
by Wischer, Timo (ADITG/ESB) 05 Oct '17

05 Oct '17

Hi all, snd_ctl_ext_callback::read_event() callback is mentioned as optional (see http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=include/control_extern…) but there is no NULL check and the NULL pointer will be called if the read_event function callback pointer is not set. (see http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/control/control_ex…) I think a default function has to be provided which will be called when the callback is not set or the read_event() callback should not be marked as optional. What is your opinion? Best regards Timo Wischer Advanced Driver Information Technology GmbH Engineering Software Base (ADITG/ESB) Robert-Bosch-Str. 200 31139 Hildesheim Germany Tel. +49 5121 49 6938 Fax +49 5121 49 6999 twischer(a)de.adit-jv.com ADIT is a joint venture company of Robert Bosch GmbH/Robert Bosch Car Multimedia GmbH and DENSO Corporation Sitz: Hildesheim, Registergericht: Amtsgericht Hildesheim HRB 3438 Geschäftsführung: Wilhelm Grabow, Ken Yaguchi

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[alsa-devel] notifying userspace of hwparams change
by Tim Harvey 05 Oct '17

05 Oct '17

Greetings, I've got an HDMI receiver chip I'm working on a driver for which provides the audio signal from the HDMI source. An HDMI source can be a variety of sample-rates, channels, widths, and formats. If I understand correctly I need to register the device as supporting all the possible rates/channels/formats then constrain these to what is actually available at the time in the startup callback with snd_pcm_hw_contstraint_minmax and snd_pcm_hw_constraint_mask. How do I notify userspace when the chip tells me that the HDMI audio source rate/channels/format has changed? I know on the Video side the driver sends a V4L2_EVENT_SOURCE_CHANGE event to userspace when input format/resolution has changed so that userspace can stop streaming, call VIDIOC_QUERY_DV_TIMINGS and then VIDIOC_S_DV_TIMINGS with new timings. Regards, Tim

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[alsa-devel] [PATCH] ALSA: core: Add new ACCESS_TLV flag for coefficient TLV controls
by Charles Keepax 05 Oct '17

05 Oct '17

ASoC can use the TLV mechanism to pass coefficient data for controls that exceed the normal 512 byte limit. However, there is no mechanism to inform user-space the control expects data through the TLV mechanism. Currently amixer uses a combination of the absense of non-TLV read/write permissions and the type being BYTES. This is not ideal as it is not a clear signal and prevents controls that might support both access mechanisms. Add a specific flag SNDRV_CTL_ELEM_ACCESS_TLV_COEFF to let user-space know when it is dealing with a TLV control that is being used to pass actual coefficient data. Then add this flag to the controls generated by the current users of these types of controls. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Hi, This builds on work done by Sakamoto-san [1] and the changes mentioned to amixer in the commit message are found in [2]. I think something like this should probably be added, what I am less clear on is where we go from here? Sakamoto-san's original patches also added a tag type SNDRV_CTL_TVLT_COEFF, to keep the data matching the normal format found in TLV controls. The trouble is neither of the two users really follow this convention at the moment. The ADSP code just takes the data as is with no additional headers and the Skylake code does have a tag and a length as part of the data passed through the TLV however the tag part looks like it is used for some internal purpose, so I am far from clear it would support a fixed value to identify the data without some changes as well. Changing the ADSP code is probably acceptable since we have had very few users in the field, and since this work has been done improvements in the firmware tool chains mean most firmwares are created with much smaller dedicated purpose controls rather than the large block ones that were initially the reason for using these controls on our side. Finally, since [3] tinymix has been updated to add some TLV information into control writes/reads which will break the ADSP code once it trickles down to users. So I would be keen to discuss this at this years conference and decide how we are going to handle these controls? As Mark says [4] maybe we should see if there is a better approach we should all migrate across to? Thanks, Charles [1] https://patchwork.kernel.org/patch/9304649/ [2] https://patchwork.kernel.org/patch/8147001/ [3] https://github.com/tinyalsa/tinyalsa/commit/3f813e47784674a3909fe1277bc10b7… [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-September/125743.… include/sound/soc.h | 3 ++- include/uapi/sound/asound.h | 1 + sound/core/control.c | 1 + sound/soc/codecs/wm_adsp.c | 3 ++- sound/soc/intel/skylake/skl-topology.c | 1 + sound/soc/soc-ops.c | 3 +++ 6 files changed, 10 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d776cdee30d7..d82297a95caf 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -323,7 +323,8 @@ #define SND_SOC_BYTES_TLV(xname, xcount, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | \ - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK | \ + SNDRV_CTL_ELEM_ACCESS_TLV_COEFF, \ .tlv.c = (snd_soc_bytes_tlv_callback), \ .info = snd_soc_bytes_info_ext, \ .private_value = (unsigned long)&(struct soc_bytes_ext) \ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1949923a40bf..4bf0b2f835e9 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -853,6 +853,7 @@ typedef int __bitwise snd_ctl_elem_iface_t; #define SNDRV_CTL_ELEM_ACCESS_TLV_WRITE (1<<5) /* TLV write is possible */ #define SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE (SNDRV_CTL_ELEM_ACCESS_TLV_READ|SNDRV_CTL_ELEM_ACCESS_TLV_WRITE) #define SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND (1<<6) /* TLV command is possible */ +#define SNDRV_CTL_ELEM_ACCESS_TLV_COEFF (1<<7) /* Coefficient data is accepted */ #define SNDRV_CTL_ELEM_ACCESS_INACTIVE (1<<8) /* control does actually nothing, but may be updated */ #define SNDRV_CTL_ELEM_ACCESS_LOCK (1<<9) /* write lock */ #define SNDRV_CTL_ELEM_ACCESS_OWNER (1<<10) /* write lock owner */ diff --git a/sound/core/control.c b/sound/core/control.c index 56b3e2d49c82..e7a74d48633b 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -266,6 +266,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, SNDRV_CTL_ELEM_ACCESS_INACTIVE | SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND | + SNDRV_CTL_ELEM_ACCESS_TLV_COEFF | SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK); err = snd_ctl_new(&kctl, count, access, NULL); diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 65c059b5ffd7..1d7fe0d8a18a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1156,7 +1156,8 @@ static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) wr = SNDRV_CTL_ELEM_ACCESS_TLV_WRITE; vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE; - out = SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + out = SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK | + SNDRV_CTL_ELEM_ACCESS_TLV_COEFF; } else { rd = SNDRV_CTL_ELEM_ACCESS_READ; wr = SNDRV_CTL_ELEM_ACCESS_WRITE; diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 27bcb62568fb..01b01c6a2bf3 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2870,6 +2870,7 @@ static int skl_tplg_control_load(struct snd_soc_component *cmpnt, tplg_bc = container_of(hdr, struct snd_soc_tplg_bytes_control, hdr); if (kctl->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + kctl->access |= SNDRV_CTL_ELEM_ACCESS_TLV_COEFF; sb = (struct soc_bytes_ext *)kctl->private_value; if (tplg_bc->priv.size) return skl_init_algo_data( diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 500f98c730b9..5ac2afe12b5d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -776,6 +776,9 @@ int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, unsigned int count = size < params->max ? size : params->max; int ret = -ENXIO; + if (!(kcontrol->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_COEFF)) + return -ENXIO; + switch (op_flag) { case SNDRV_CTL_TLV_OP_READ: if (params->get) -- 2.11.0

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[alsa-devel] [PATCH] sound/pci/asihpi: Convert timers to use timer_setup()
by Kees Cook 05 Oct '17

05 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: Thomas Gleixner <tglx(a)linutronix.de> Signed-off-by: Kees Cook <keescook(a)chromium.org> --- This requires commit 686fef928bba ("timer: Prepare to change timer callback argument type") in v4.14-rc3, but should be otherwise stand-alone. --- sound/pci/asihpi/asihpi.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 70d023a85bf5..614322171c79 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -749,9 +749,9 @@ static inline unsigned int modulo_min(unsigned int a, unsigned int b, /** Timer function, equivalent to interrupt service routine for cards */ -static void snd_card_asihpi_timer_function(unsigned long data) +static void snd_card_asihpi_timer_function(struct timer_list *t) { - struct snd_card_asihpi_pcm *dpcm = (struct snd_card_asihpi_pcm *)data; + struct snd_card_asihpi_pcm *dpcm = from_timer(dpcm, t, timer); struct snd_pcm_substream *substream = dpcm->substream; struct snd_card_asihpi *card = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime; @@ -948,7 +948,7 @@ static void snd_card_asihpi_int_task(unsigned long data) asihpi = (struct snd_card_asihpi *)a->snd_card->private_data; if (asihpi->llmode_streampriv) snd_card_asihpi_timer_function( - (unsigned long)asihpi->llmode_streampriv); + &asihpi->llmode_streampriv->timer); } static void snd_card_asihpi_isr(struct hpi_adapter *a) @@ -1059,8 +1059,7 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) If internal and other stream playing, can't switch */ - setup_timer(&dpcm->timer, snd_card_asihpi_timer_function, - (unsigned long) dpcm); + timer_setup(&dpcm->timer, snd_card_asihpi_timer_function, 0); dpcm->substream = substream; runtime->private_data = dpcm; runtime->private_free = snd_card_asihpi_runtime_free; @@ -1240,8 +1239,7 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream) if (err) return -EIO; - setup_timer(&dpcm->timer, snd_card_asihpi_timer_function, - (unsigned long) dpcm); + timer_setup(&dpcm->timer, snd_card_asihpi_timer_function, 0); dpcm->substream = substream; runtime->private_data = dpcm; runtime->private_free = snd_card_asihpi_runtime_free; -- 2.7.4 -- Kees Cook Pixel Security

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[alsa-devel] [PATCH] sound/core: Convert timers to use timer_setup()
by Kees Cook 05 Oct '17

05 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. This adds a pointer back to struct snd_timer. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Ingo Molnar <mingo(a)kernel.org> Cc: Dan Carpenter <dan.carpenter(a)oracle.com> Cc: alsa-devel(a)alsa-project.org Cc: Thomas Gleixner <tglx(a)linutronix.de> Signed-off-by: Kees Cook <keescook(a)chromium.org> --- This requires commit 686fef928bba ("timer: Prepare to change timer callback argument type") in v4.14-rc3, but should be otherwise stand-alone. --- sound/core/timer.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 6cdd04a45962..09acaf2b2e57 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1028,15 +1028,17 @@ EXPORT_SYMBOL(snd_timer_global_register); struct snd_timer_system_private { struct timer_list tlist; + struct snd_timer *snd_timer; unsigned long last_expires; unsigned long last_jiffies; unsigned long correction; }; -static void snd_timer_s_function(unsigned long data) +static void snd_timer_s_function(struct timer_list *t) { - struct snd_timer *timer = (struct snd_timer *)data; - struct snd_timer_system_private *priv = timer->private_data; + struct snd_timer_system_private *priv = from_timer(priv, t, + tlist); + struct snd_timer *timer = priv->snd_timer; unsigned long jiff = jiffies; if (time_after(jiff, priv->last_expires)) priv->correction += (long)jiff - (long)priv->last_expires; @@ -1118,7 +1120,8 @@ static int snd_timer_register_system(void) snd_timer_free(timer); return -ENOMEM; } - setup_timer(&priv->tlist, snd_timer_s_function, (unsigned long) timer); + priv->snd_timer = timer; + timer_setup(&priv->tlist, snd_timer_s_function, 0); timer->private_data = priv; timer->private_free = snd_timer_free_system; return snd_timer_global_register(timer); -- 2.7.4 -- Kees Cook Pixel Security

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[alsa-devel] [PATCH] ASoC: fsi: Use of_device_get_match_data() helper
by Geert Uytterhoeven 04 Oct '17

04 Oct '17

Use the of_device_get_match_data() helper instead of open coding. Note that when used with DT, there's always a valid match. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/sh/fsi.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 6d3c7706d93fbc10..c3aaf4788557c4e1 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1932,14 +1932,9 @@ static int fsi_probe(struct platform_device *pdev) core = NULL; if (np) { - const struct of_device_id *of_id; - - of_id = of_match_device(fsi_of_match, &pdev->dev); - if (of_id) { - core = of_id->data; - fsi_of_parse("fsia", np, &info.port_a, &pdev->dev); - fsi_of_parse("fsib", np, &info.port_b, &pdev->dev); - } + core = of_device_get_match_data(&pdev->dev); + fsi_of_parse("fsia", np, &info.port_a, &pdev->dev); + fsi_of_parse("fsib", np, &info.port_b, &pdev->dev); } else { const struct platform_device_id *id_entry = pdev->id_entry; if (id_entry) -- 2.7.4

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[alsa-devel] usb/sound/bcd2000: warning in bcd2000_init_device
by Andrey Konovalov 04 Oct '17

04 Oct '17

Hi! I've got the following report while fuzzing the kernel with syzkaller. On commit e19b205be43d11bff638cad4487008c48d21c103 (4.14-rc2). It seems that there's no check of the endpoint type. usb 1-1: BOGUS urb xfer, pipe 1 != type 3 ------------[ cut here ]------------ WARNING: CPU: 0 PID: 1846 at drivers/usb/core/urb.c:449 usb_submit_urb+0xf8a/0x11d0 Modules linked in: CPU: 0 PID: 1846 Comm: kworker/0:2 Not tainted 4.14.0-rc2-42613-g1488251d1a98 #238 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Workqueue: usb_hub_wq hub_event task: ffff880064296300 task.stack: ffff8800643b0000 RIP: 0010:usb_submit_urb+0xf8a/0x11d0 drivers/usb/core/urb.c:448 RSP: 0018:ffff8800643b6140 EFLAGS: 00010286 RAX: 0000000000000029 RBX: ffff880063842400 RCX: 0000000000000000 RDX: 0000000000000029 RSI: ffffffff85a58800 RDI: ffffed000c876c1a RBP: ffff8800643b6240 R08: 1ffff1000c876ac0 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 1ffff1000c876c2f R13: 0000000000000003 R14: 0000000000000001 R15: ffff88006325c768 FS: 0000000000000000(0000) GS:ffff88006c800000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007fa2a51c0000 CR3: 000000006acc8000 CR4: 00000000000006f0 Call Trace: bcd2000_init_device sound/usb/bcd2000/bcd2000.c:289 bcd2000_init_midi sound/usb/bcd2000/bcd2000.c:345 bcd2000_probe+0xe64/0x19e0 sound/usb/bcd2000/bcd2000.c:406 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: 48 8b 85 30 ff ff ff 48 8d b8 98 00 00 00 e8 6e 91 8b ff 45 89 e8 44 89 f1 4c 89 fa 48 89 c6 48 c7 c7 c0 5a c8 85 e8 10 50 dd fd <0f> ff e9 9b f7 ff ff e8 ba cf 26 fe e9 80 f7 ff ff e8 a0 a5 f4 ---[ end trace bad127706d5fe2d6 ]---

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[alsa-devel] [PATCH v2] ASoC: codecs: msm8916-wcd-analog: configure micbias in mbhc setup
by Damien Riegel 04 Oct '17

04 Oct '17

The very first time a headset is plugged in, detection is unreliable because bias hasn't been configured yet, it's done once a mechanical insertion interrupt has been triggered, so following insertions (and thus detections) are not affected. To fix the very first detection, the bias must also be configured in the function that setup the MBHC. Move pm8916_wcd_setup_mbhc after pm8916_mbhc_configure_bias to avoid a forward declaration. Signed-off-by: Damien Riegel <damien.riegel(a)savoirfairelinux.com> --- Changes in v2: - squash the two patches in one patch only sound/soc/codecs/msm8916-wcd-analog.c | 94 +++++++++++++++++++---------------- 1 file changed, 50 insertions(+), 44 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index f562f2d86907..71494e9dbdcb 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -443,50 +443,6 @@ static int pm8916_wcd_analog_enable_micbias_int1(struct wcd->micbias1_cap_mode); } -static void pm8916_wcd_setup_mbhc(struct pm8916_wcd_analog_priv *wcd) -{ - struct snd_soc_codec *codec = wcd->codec; - u32 plug_type = 0; - u32 int_en_mask; - - snd_soc_write(codec, CDC_A_MBHC_DET_CTL_1, - CDC_A_MBHC_DET_CTL_L_DET_EN | - CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_INSERTION | - CDC_A_MBHC_DET_CTL_MIC_CLAMP_CTL_AUTO | - CDC_A_MBHC_DET_CTL_MBHC_BIAS_EN); - - if (wcd->hphl_jack_type_normally_open) - plug_type |= CDC_A_HPHL_PLUG_TYPE_NO; - - if (wcd->gnd_jack_type_normally_open) - plug_type |= CDC_A_GND_PLUG_TYPE_NO; - - snd_soc_write(codec, CDC_A_MBHC_DET_CTL_2, - CDC_A_MBHC_DET_CTL_HS_L_DET_PULL_UP_CTRL_I_3P0 | - CDC_A_MBHC_DET_CTL_HS_L_DET_COMPA_CTRL_V0P9_VDD | - plug_type | - CDC_A_MBHC_DET_CTL_HPHL_100K_TO_GND_EN); - - - snd_soc_write(codec, CDC_A_MBHC_DBNC_TIMER, - CDC_A_MBHC_DBNC_TIMER_INSREM_DBNC_T_256_MS | - CDC_A_MBHC_DBNC_TIMER_BTN_DBNC_T_16MS); - - /* enable MBHC clock */ - snd_soc_update_bits(codec, CDC_D_CDC_DIG_CLK_CTL, - DIG_CLK_CTL_D_MBHC_CLK_EN_MASK, - DIG_CLK_CTL_D_MBHC_CLK_EN); - - int_en_mask = MBHC_SWITCH_INT; - if (wcd->mbhc_btn_enabled) - int_en_mask |= MBHC_BUTTON_PRESS_DET | MBHC_BUTTON_RELEASE_DET; - - snd_soc_update_bits(codec, CDC_D_INT_EN_CLR, int_en_mask, 0); - snd_soc_update_bits(codec, CDC_D_INT_EN_SET, int_en_mask, int_en_mask); - wcd->mbhc_btn0_released = false; - wcd->detect_accessory_type = true; -} - static int pm8916_mbhc_configure_bias(struct pm8916_wcd_analog_priv *priv, bool micbias2_enabled) { @@ -534,6 +490,56 @@ static int pm8916_mbhc_configure_bias(struct pm8916_wcd_analog_priv *priv, return 0; } +static void pm8916_wcd_setup_mbhc(struct pm8916_wcd_analog_priv *wcd) +{ + struct snd_soc_codec *codec = wcd->codec; + bool micbias_enabled = false; + u32 plug_type = 0; + u32 int_en_mask; + + snd_soc_write(codec, CDC_A_MBHC_DET_CTL_1, + CDC_A_MBHC_DET_CTL_L_DET_EN | + CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_INSERTION | + CDC_A_MBHC_DET_CTL_MIC_CLAMP_CTL_AUTO | + CDC_A_MBHC_DET_CTL_MBHC_BIAS_EN); + + if (wcd->hphl_jack_type_normally_open) + plug_type |= CDC_A_HPHL_PLUG_TYPE_NO; + + if (wcd->gnd_jack_type_normally_open) + plug_type |= CDC_A_GND_PLUG_TYPE_NO; + + snd_soc_write(codec, CDC_A_MBHC_DET_CTL_2, + CDC_A_MBHC_DET_CTL_HS_L_DET_PULL_UP_CTRL_I_3P0 | + CDC_A_MBHC_DET_CTL_HS_L_DET_COMPA_CTRL_V0P9_VDD | + plug_type | + CDC_A_MBHC_DET_CTL_HPHL_100K_TO_GND_EN); + + + snd_soc_write(codec, CDC_A_MBHC_DBNC_TIMER, + CDC_A_MBHC_DBNC_TIMER_INSREM_DBNC_T_256_MS | + CDC_A_MBHC_DBNC_TIMER_BTN_DBNC_T_16MS); + + /* enable MBHC clock */ + snd_soc_update_bits(codec, CDC_D_CDC_DIG_CLK_CTL, + DIG_CLK_CTL_D_MBHC_CLK_EN_MASK, + DIG_CLK_CTL_D_MBHC_CLK_EN); + + if (snd_soc_read(codec, CDC_A_MICB_2_EN) & CDC_A_MICB_2_EN_ENABLE) + micbias_enabled = true; + + pm8916_mbhc_configure_bias(wcd, micbias_enabled); + + int_en_mask = MBHC_SWITCH_INT; + if (wcd->mbhc_btn_enabled) + int_en_mask |= MBHC_BUTTON_PRESS_DET | MBHC_BUTTON_RELEASE_DET; + + snd_soc_update_bits(codec, CDC_D_INT_EN_CLR, int_en_mask, 0); + snd_soc_update_bits(codec, CDC_D_INT_EN_SET, int_en_mask, int_en_mask); + wcd->mbhc_btn0_released = false; + wcd->detect_accessory_type = true; +} + static int pm8916_wcd_analog_enable_micbias_int2(struct snd_soc_dapm_widget *w, struct snd_kcontrol -- 2.14.1

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[alsa-devel] [PATCH] ASoC: rockchip: Allocate enough memory so we don't overflow routes
by Douglas Anderson 04 Oct '17

04 Oct '17

In the recent commit d9f9c167edae ("ASoC: rockchip: Init dapm routes dynamically") we improperly allocated memory for the card->dapm_routes causing us to overflow the allocation on every boot. Oops. Let's allocate the correct amount of memory. We'll also add a check to make sure that we don't overrun memory even if we encounter some sort of weird device tree. Fixes: d9f9c167edae ("ASoC: rockchip: Init dapm routes dynamically") Signed-off-by: Douglas Anderson <dianders(a)chromium.org> --- sound/soc/rockchip/rk3399_gru_sound.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 30eed83e8a13..d64fbbd50544 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -494,13 +494,17 @@ static int rockchip_sound_of_parse_dais(struct device *dev, struct snd_soc_dai_link *dai; struct snd_soc_dapm_route *routes; int i, index; + int 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), + num_routes = 0; + for (i = 0; i < ARRAY_SIZE(rockchip_routes); i++) + num_routes += rockchip_routes[i].num_routes; + routes = devm_kzalloc(dev, num_routes * sizeof(*routes), GFP_KERNEL); if (!routes) return -ENOMEM; @@ -538,6 +542,12 @@ static int rockchip_sound_of_parse_dais(struct device *dev, dai->platform_of_node = np_cpu; dai->cpu_of_node = np_cpu; + if (card->num_dapm_routes + rockchip_routes[index].num_routes > + num_routes) { + dev_err(dev, "Too many routes\n"); + return -EINVAL; + } + memcpy(routes + card->num_dapm_routes, rockchip_routes[index].routes, rockchip_routes[index].num_routes * sizeof(*routes)); -- 2.14.2.822.g60be5d43e6-goog

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[alsa-devel] [PATCH 0/3] ASoC: img-i2s-out: Add power management
by Ed Blake 04 Oct '17

04 Oct '17

This patch set adds power managment to the img-i2s-out driver. The first patch renames existing 'suspend' and 'resume' functions to 'runtime_suspend' and 'runtime_resume', which is what they actually are, the second patch adds system PM suspend / resume handling and the third patch adds control of the sys clock to runtime PM. Ed Blake (3): ASoC: img-i2s-out: Rename suspend / resume funcs ASoC: img-i2s-out: Add suspend / resume handling ASoC: img-i2s-out: Add control of sys clock to runtime PM sound/soc/img/img-i2s-out.c | 116 +++++++++++++++++++++++++++++++++----------- 1 file changed, 87 insertions(+), 29 deletions(-) -- 1.9.1

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