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

[alsa-devel] [PATCH v3] ASoC: dapm: Avoid creating kcontrol for params
by yesanishhere＠gmail.com 09 Oct '17

09 Oct '17

From: anish kumar <yesanishhere(a)gmail.com> Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- Changes in this version: - Fixed the coding guideline comment from charles. Changes in v2: - Review comments from charles to move common function of error handling in seperate function. Changes in v1: - Included the review comments from charles regarding releasing memory in error path. sound/soc/soc-dapm.c | 140 +++++++++++++++++++++++++++++++-------------------- 1 file changed, 86 insertions(+), 54 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d55cac6..dbf3838 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3778,18 +3778,27 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, return 0; } -int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - const struct snd_soc_pcm_stream *params, - unsigned int num_params, - struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) +void +snd_soc_dapm_free_kcontrol(struct snd_soc_card *card, + unsigned long *private_value, + int num_params, + const char **w_param_text) +{ + int count; + + devm_kfree(card->dev, (void *)*private_value); + for (count = 0 ; count < num_params; count++) + devm_kfree(card->dev, (void *)w_param_text[count]); + devm_kfree(card->dev, w_param_text); +} + +static struct snd_kcontrol_new * +snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card, + char *link_name, + const struct snd_soc_pcm_stream *params, + int num_params, const char **w_param_text, + unsigned long *private_value) { - struct snd_soc_dapm_widget template; - struct snd_soc_dapm_widget *w; - char *link_name; - int ret, count; - unsigned long private_value; - const char **w_param_text; struct soc_enum w_param_enum[] = { SOC_ENUM_SINGLE(0, 0, 0, NULL), }; @@ -3798,19 +3807,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, snd_soc_dapm_dai_link_get, snd_soc_dapm_dai_link_put), }; + struct snd_kcontrol_new *kcontrol_news; const struct snd_soc_pcm_stream *config = params; - - w_param_text = devm_kcalloc(card->dev, num_params, - sizeof(char *), GFP_KERNEL); - if (!w_param_text) - return -ENOMEM; - - link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", - source->name, sink->name); - if (!link_name) { - ret = -ENOMEM; - goto outfree_w_param; - } + int count; for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { @@ -3827,47 +3826,86 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, strlen(config->stream_name) + 1, GFP_KERNEL); } - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } + if (!w_param_text[count]) + goto outfree_w_param; config++; } + w_param_enum[0].items = num_params; w_param_enum[0].texts = w_param_text; - memset(&template, 0, sizeof(template)); - template.reg = SND_SOC_NOPM; - template.id = snd_soc_dapm_dai_link; - template.name = link_name; - template.event = snd_soc_dai_link_event; - template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD; - template.num_kcontrols = 1; - /* duplicate w_param_enum on heap so that memory persists */ - private_value = + *private_value = (unsigned long) devm_kmemdup(card->dev, (void *)(kcontrol_dai_link[0].private_value), sizeof(struct soc_enum), GFP_KERNEL); - if (!private_value) { + if (!*private_value) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_link_name; + goto outfree_w_param; } - kcontrol_dai_link[0].private_value = private_value; + kcontrol_dai_link[0].private_value = *private_value; /* duplicate kcontrol_dai_link on heap so that memory persists */ - template.kcontrol_news = - devm_kmemdup(card->dev, &kcontrol_dai_link[0], + kcontrol_news = devm_kmemdup(card->dev, &kcontrol_dai_link[0], sizeof(struct snd_kcontrol_new), GFP_KERNEL); - if (!template.kcontrol_news) { + if (!kcontrol_news) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_private_value; + goto outfree_w_param; } + return kcontrol_news; +outfree_w_param: + snd_soc_dapm_free_kcontrol(card, private_value, num_params, w_param_text); + return NULL; +} + +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + unsigned int num_params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget template; + struct snd_soc_dapm_widget *w; + const char **w_param_text; + unsigned long private_value; + char *link_name; + int ret; + + link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", + source->name, sink->name); + if (!link_name) + return -ENOMEM; + + memset(&template, 0, sizeof(template)); + template.reg = SND_SOC_NOPM; + template.id = snd_soc_dapm_dai_link; + template.name = link_name; + template.event = snd_soc_dai_link_event; + template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD; + template.kcontrol_news = NULL; + + /* allocate memory for control, only in case of multiple configs */ + if (num_params > 1) { + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) { + ret = -ENOMEM; + goto param_fail; + } + + template.num_kcontrols = 1; + template.kcontrol_news = + snd_soc_dapm_alloc_kcontrol(card, + link_name, params, num_params, + w_param_text, &private_value); + if (!template.kcontrol_news) { + ret = -ENOMEM; + goto param_fail; + } + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); @@ -3899,15 +3937,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kfree(card->dev, w); outfree_kcontrol_news: devm_kfree(card->dev, (void *)template.kcontrol_news); -outfree_private_value: - devm_kfree(card->dev, (void *)private_value); -outfree_link_name: + snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text); +param_fail: devm_kfree(card->dev, link_name); - for (count = 0 ; count < num_params; count++) - devm_kfree(card->dev, (void *)w_param_text[count]); -outfree_w_param: - devm_kfree(card->dev, w_param_text); - return ret; } -- 2.5.4 (Apple Git-61)

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[alsa-devel] [PATCH] sound/pci/riptide or drivers/base/firmware: Fix a possible sleep-in-atomic bug
by Jia-Ju Bai 09 Oct '17

09 Oct '17

The riptide driver may sleep under a spinlock, and the function call path is: snd_riptide_prepare (acquire the spinlock) setsampleformat sendcmd riptide_reset try_to_load_firmware request_firmware _request_firmware (drivers/base/firmware_class.c) _request_firmware_prepare kzalloc(GFP_KERNEL) --> may sleep To fix it, GFP_KERNEL is replaced with GFP_ATOMIC in _request_firmware_prepare. This bug is found by my static analysis tool and my code review. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)163.com> --- drivers/base/firmware_class.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/base/firmware_class.c b/drivers/base/firmware_class.c index 4b57cf5..d89f676 100644 --- a/drivers/base/firmware_class.c +++ b/drivers/base/firmware_class.c @@ -1138,7 +1138,7 @@ static inline void kill_pending_fw_fallback_reqs(bool only_kill_custom) { } struct firmware_buf *buf; int ret; - *firmware_p = firmware = kzalloc(sizeof(*firmware), GFP_KERNEL); + *firmware_p = firmware = kzalloc(sizeof(*firmware), GFP_ATOMIC); if (!firmware) { dev_err(device, "%s: kmalloc(struct firmware) failed\n", __func__); -- 1.7.9.5

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[alsa-devel] [PATCH] opl3: Fix a possible sleep-in-atomic bug in snd_opl3_find_patch
by Jia-Ju Bai 09 Oct '17

09 Oct '17

The driver may sleep under a spinlock, and the function call path is: snd_opl3_note_on (acquire the spinlock) snd_opl3_find_patch kzalloc(GFP_KERNEL) --> may sleep To fix it, GFP_KERNEL is replaced with GFP_ATOMIC. This bug is found by my static analysis tool and my code review. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)163.com> --- sound/drivers/opl3/opl3_synth.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index ddcc1a3..2e1cb2b 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -325,7 +325,7 @@ struct fm_patch *snd_opl3_find_patch(struct snd_opl3 *opl3, int prog, int bank, if (!create_patch) return NULL; - patch = kzalloc(sizeof(*patch), GFP_KERNEL); + patch = kzalloc(sizeof(*patch), GFP_ATOMIC); if (!patch) return NULL; patch->prog = prog; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: cygnus: Remove unnecessary active_slots check
by Christos Gkekas 08 Oct '17

08 Oct '17

Variable active_slots is unsigned so checking whether it is less than zero is not necessary. Signed-off-by: Christos Gkekas <chris.gekas(a)gmail.com> --- sound/soc/bcm/cygnus-ssp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index 15c438f..cb46128 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -998,7 +998,7 @@ static int cygnus_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, active_slots = hweight32(tx_mask); - if ((active_slots < 0) || (active_slots > 16)) + if (active_slots > 16) return -EINVAL; /* Slot value must be even */ -- 2.7.4

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[alsa-devel] [BUG] haswell: A possible sleep-in-atomic bug in hsw_irq_thread
by Jia-Ju Bai 08 Oct '17

08 Oct '17

According to sst-haswell-ipc.c, the driver may sleep under a spinlock, and the function call path is: hsw_irq_thread (acquire the spinlock) hsw_process_notification hsw_log_message mutex_lock --> may sleep This bug is found by my static analysis tool and my code review. Thanks, Jia-Ju Bai

1 0

[alsa-devel] [PATCH] ALSA: seq: resize buffer for overflow
by Mark Salyzyn 06 Oct '17

06 Oct '17

Can not replicate, issue discovered in fuzzing. Stack trace below. No functional or performance testing done regarding the fix. Trap at (reformatted): snd_seq_oss_readq_puts(struct seq_oss_readq *q, int dev, unsigned char *data, int len) { union evrec rec; int result; memset(&rec, 0, sizeof(rec)); rec.c[0] = SEQ_MIDIPUTC; rec.c[2] = dev; while (len-- > 0) { rec.c[1] = *data++; // data is RBX HERE 'data' pointer just passed a page boundary, so the buffer supplied was short. Caller must have been handed an ev->type equal to SNDDRV_SEQ_EVENT_SYSEX, which resulted in handing off ev->data.ext.ptr[ev->data.ext.len] buffer. Intuited that the source of the event and buffer was referenced in snd_midi_event_encode_byte() passing a larger length than the allocated buffer. BUG: unable to handle kernel paging request at ffffc900008ab000 IP: [<ffffffff82e2fc05>] snd_seq_oss_readq_puts+0xd5/0x170 sound/core/seq/oss/seq_oss_readq.c:112 PGD 1da091067 PUD 1da092067 Oops: 0000 [#1] PREEMPT SMP KASAN Dumping ftrace buffer: (ftrace buffer empty) Modules linked in: CPU: 1 PID: 3264 Comm: XXXXXXXX Hardware name: XXXXXXXXXX task: ffff8801cdd9e000 task.stack: ffff8801ce648000 RIP: 0010:[<ffffffff82e2fc05>] [<ffffffff82e2fc05>] snd_seq_oss_readq_puts+0xd5/0x170 sound/core/seq/oss/seq_oss_readq.c:112 RSP: 0018:ffff8801ce64f1c0 EFLAGS: 00010246 RAX: 0000000000000000 RBX: ffffc900008ab000 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 0000000000000001 RDI: ffffffff858e5780 RBP: ffff8801ce64f260 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 1ffff10039cc9df2 R12: 000000003fffffa4 R13: dffffc0000000000 R14: ffff8801ce64f238 R15: ffffc900008ab001 FS: 00007fe3d3d9e700(0000) GS:ffff8801db300000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: ffffc900008ab000 CR3: 00000001d19b7000 CR4: 00000000001406e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Stack: 1ffff10039cc9e3b ffff8801ce64f1f8 ffff8801d0b9aa00 0000000041b58ab3 ffffffff841daf3c ffffffff82e2fb30 0000000000000286 0000000000000005 ffffffff838aa5d5 ffffffff861962c0 dffffc0000000000 ffff8801ce64f260 Call Trace: [<ffffffff82e2ebbe>] send_midi_event sound/core/seq/oss/seq_oss_midi.c:616 [inline] [<ffffffff82e2ebbe>] snd_seq_oss_midi_input+0x8ce/0xa70 sound/core/seq/oss/seq_oss_midi.c:535 [<ffffffff82e2758d>] snd_seq_oss_event_input+0x15d/0x220 sound/core/seq/oss/seq_oss_event.c:439 [<ffffffff82e0b650>] snd_seq_deliver_single_event.constprop.11+0x310/0x7c0 sound/core/seq/seq_clientmgr.c:621 [<ffffffff82e0be16>] deliver_to_subscribers sound/core/seq/seq_clientmgr.c:676 [inline] [<ffffffff82e0be16>] snd_seq_deliver_event+0x316/0x740 sound/core/seq/seq_clientmgr.c:807 [<ffffffff82e0cf9e>] snd_seq_kernel_client_dispatch+0x11e/0x150 sound/core/seq/seq_clientmgr.c:2314 [<ffffffff82e31775>] dummy_input+0x235/0x320 sound/core/seq/seq_dummy.c:104 [<ffffffff82e0b650>] snd_seq_deliver_single_event.constprop.11+0x310/0x7c0 sound/core/seq/seq_clientmgr.c:621 [<ffffffff82e0bc2d>] snd_seq_deliver_event+0x12d/0x740 sound/core/seq/seq_clientmgr.c:818 [<ffffffff82e0d0ed>] snd_seq_dispatch_event+0x11d/0x520 sound/core/seq/seq_clientmgr.c:892 [<ffffffff82e1117e>] snd_seq_check_queue.part.3+0x38e/0x510 sound/core/seq/seq_queue.c:285 [<ffffffff82e11d8d>] snd_seq_check_queue sound/core/seq/seq_queue.c:357 [inline] [<ffffffff82e11d8d>] snd_seq_enqueue_event+0x32d/0x3d0 sound/core/seq/seq_queue.c:363 [<ffffffff82e0c444>] snd_seq_client_enqueue_event+0x204/0x3e0 sound/core/seq/seq_clientmgr.c:951 [<ffffffff82e0cc55>] kernel_client_enqueue.part.10+0xb5/0xd0 sound/core/seq/seq_clientmgr.c:2251 [<ffffffff82e0cd3f>] kernel_client_enqueue sound/core/seq/seq_clientmgr.c:2241 [inline] [<ffffffff82e0cd3f>] snd_seq_kernel_client_enqueue_blocking+0xcf/0x110 sound/core/seq/seq_clientmgr.c:2279 [<ffffffff82e27f68>] insert_queue sound/core/seq/oss/seq_oss_rw.c:189 [inline] [<ffffffff82e27f68>] snd_seq_oss_write+0x538/0x850 sound/core/seq/oss/seq_oss_rw.c:148 [<ffffffff82e1fab4>] odev_write+0x64/0x90 sound/core/seq/oss/seq_oss.c:177 [<ffffffff8156ab43>] __vfs_write+0x103/0x680 fs/read_write.c:510 [<ffffffff8156ec70>] vfs_write+0x170/0x4e0 fs/read_write.c:560 [<ffffffff81572669>] SYSC_write fs/read_write.c:607 [inline] [<ffffffff81572669>] SyS_write+0xd9/0x1b0 fs/read_write.c:599 [<ffffffff838aad85>] entry_SYSCALL_64_fastpath+0x23/0xc6 Code: 4d 9a eb 4e e8 2d aa 53 fe 4c 8d 7b 01 48 89 d8 48 89 d9 48 c1 e8 03 83 e1 07 42 0f b6 04 28 38 c8 7f 08 84 c0 0f 85 80 00 00 00 <41> 0f b6 47 ff 41 83 ec 01 48 8b b5 68 ff ff ff 48 8b bd 70 ff RIP [<ffffffff82e2fc05>] snd_seq_oss_readq_puts+0xd5/0x170 sound/core/seq/oss/seq_oss_readq.c:112 RSP <ffff8801ce64f1c0> CR2: ffffc900008ab000 Signed-off-by: Mark Salyzyn <salyzyn(a)android.com> --- sound/core/seq/seq_midi_event.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c index 90bbbdbeba03..cbc501fd47c2 100644 --- a/sound/core/seq/seq_midi_event.c +++ b/sound/core/seq/seq_midi_event.c @@ -192,8 +192,7 @@ EXPORT_SYMBOL(snd_midi_event_no_status); /* * resize buffer */ -#if 0 -int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) +static int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) { unsigned char *new_buf, *old_buf; unsigned long flags; @@ -212,7 +211,6 @@ int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) kfree(old_buf); return 0; } -#endif /* 0 */ /* * read bytes and encode to sequencer event if finished @@ -297,6 +295,8 @@ int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, } else if (dev->type == ST_SYSEX) { if (c == MIDI_CMD_COMMON_SYSEX_END || dev->read >= dev->bufsize) { + if (dev->read > dev->bufsize) + snd_midi_event_resize_buffer(dev, dev->read); ev->flags &= ~SNDRV_SEQ_EVENT_LENGTH_MASK; ev->flags |= SNDRV_SEQ_EVENT_LENGTH_VARIABLE; ev->type = SNDRV_SEQ_EVENT_SYSEX; -- 2.14.2.920.gcf0c67979c-goog

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Re: [alsa-devel] [PATCH] alsactl: Move systemd unit start-up from basic.target to sound.target
by Takashi Iwai 06 Oct '17

06 Oct '17

On Tue, 26 Sep 2017 20:36:12 +0200, Chris Mayo wrote: > > Ensures soundcard is ready before restoring state. > > sound.target added to systemd in v18: > https://cgit.freedesktop.org/systemd/systemd/commit/?id=88dfa2938af > > Simplify dependencies: > - After=alsa-state.service is not needed because both units test for > @daemonswitch@ with opposite outcomes. > > - After=sysinit.target is automatically added by systemd. > > First proposed by Tom Yan. > > Signed-off-by: Chris Mayo <aklhfex(a)gmail.com> > --- > > I need this because my Sound Blaster Titanium is not ready by the time > basic.target is reached. > > It was proposed last year by Tom Yan but looks like it didn't get in > because requested config checks and documentation didn't materialise. > https://www.spinics.net/lists/alsa-devel/msg45537.html > > I've added those (PKG_CHECK_MODULES outputs a message at configure time). > > I've left ConditionPathExistsGlob=/dev/snd/control* in > alsa-restore.service because although it would normally be started by the > udev rule and sound.target there's no guarantee that is how it was started. Thanks, this looks like a sensitive change. Applied now. Takashi

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Add sample rate quirk for Plantronics P610
by Kai-Heng Feng 06 Oct '17

06 Oct '17

Like other Plantronics devices, P610 does not support sample rate reading. Apply sample rate quirk to it. BugLink: https://bugs.launchpad.net/bugs/1719853 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index b8cb57aeec77..9ddaae3784f5 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1138,6 +1138,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */ case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x047F, 0xC022): /* Plantronics C310 */ + case USB_ID(0x047F, 0xC02F): /* Plantronics P610 */ case USB_ID(0x047F, 0xC036): /* Plantronics C520-M */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */ -- 2.14.1

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[alsa-devel] [asoc:topic/dapm 4/4] sound//soc/soc-dapm.c:3940:2: warning: 'w_param_text' may be used uninitialized in this function
by kbuild test robot 05 Oct '17

05 Oct '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/dapm head: 19ad683abc8534cd755403381c73c0ddce7b9f6d commit: 19ad683abc8534cd755403381c73c0ddce7b9f6d [4/4] ASoC: dapm: Avoid creating kcontrol for params config: i386-randconfig-x079-201740 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout 19ad683abc8534cd755403381c73c0ddce7b9f6d # save the attached .config to linux build tree make ARCH=i386 Note: it may well be a FALSE warning. FWIW you are at least aware of it now. http://gcc.gnu.org/wiki/Better_Uninitialized_Warnings All warnings (new ones prefixed by >>): sound//soc/soc-dapm.c: In function 'snd_soc_dapm_new_pcm': >> sound//soc/soc-dapm.c:3940:2: warning: 'w_param_text' may be used uninitialized in this function [-Wmaybe-uninitialized] snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vim +/w_param_text +3940 sound//soc/soc-dapm.c 3862 3863 int snd_soc_dapm_new_pcm(struct snd_soc_card *card, 3864 const struct snd_soc_pcm_stream *params, 3865 unsigned int num_params, 3866 struct snd_soc_dapm_widget *source, 3867 struct snd_soc_dapm_widget *sink) 3868 { 3869 struct snd_soc_dapm_widget template; 3870 struct snd_soc_dapm_widget *w; 3871 const char **w_param_text; 3872 unsigned long private_value; 3873 char *link_name; 3874 int ret; 3875 3876 link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", 3877 source->name, sink->name); 3878 if (!link_name) 3879 return -ENOMEM; 3880 3881 memset(&template, 0, sizeof(template)); 3882 template.reg = SND_SOC_NOPM; 3883 template.id = snd_soc_dapm_dai_link; 3884 template.name = link_name; 3885 template.event = snd_soc_dai_link_event; 3886 template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | 3887 SND_SOC_DAPM_PRE_PMD; 3888 template.kcontrol_news = NULL; 3889 3890 /* allocate memory for control, only in case of multiple configs */ 3891 if (num_params > 1) { 3892 w_param_text = devm_kcalloc(card->dev, num_params, 3893 sizeof(char *), GFP_KERNEL); 3894 if (!w_param_text) { 3895 ret = -ENOMEM; 3896 goto param_fail; 3897 } 3898 3899 template.num_kcontrols = 1; 3900 template.kcontrol_news = 3901 snd_soc_dapm_alloc_kcontrol(card, 3902 link_name, params, num_params, 3903 w_param_text, &private_value); 3904 if (!template.kcontrol_news) { 3905 ret = -ENOMEM; 3906 goto param_fail; 3907 } 3908 } 3909 dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); 3910 3911 w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); 3912 if (IS_ERR(w)) { 3913 ret = PTR_ERR(w); 3914 /* Do not nag about probe deferrals */ 3915 if (ret != -EPROBE_DEFER) 3916 dev_err(card->dev, 3917 "ASoC: Failed to create %s widget (%d)\n", 3918 link_name, ret); 3919 goto outfree_kcontrol_news; 3920 } 3921 if (!w) { 3922 dev_err(card->dev, "ASoC: Failed to create %s widget\n", 3923 link_name); 3924 ret = -ENOMEM; 3925 goto outfree_kcontrol_news; 3926 } 3927 3928 w->params = params; 3929 w->num_params = num_params; 3930 3931 ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); 3932 if (ret) 3933 goto outfree_w; 3934 return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); 3935 3936 outfree_w: 3937 devm_kfree(card->dev, w); 3938 outfree_kcontrol_news: 3939 devm_kfree(card->dev, (void *)template.kcontrol_news); > 3940 snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text); 3941 param_fail: 3942 devm_kfree(card->dev, link_name); 3943 return ret; 3944 } 3945 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 00/18] use ARRAY_SIZE macro
by Jérémy Lefaure 05 Oct '17

05 Oct '17

Hi everyone, Using ARRAY_SIZE improves the code readability. I used coccinelle (I made a change to the array_size.cocci file [1]) to find several places where ARRAY_SIZE could be used instead of other macros or sizeof division. I tried to divide the changes into a patch per subsystem (excepted for staging). If one of the patch should be split into several patches, let me know. In order to reduce the size of the To: and Cc: lines, each patch of the series is sent only to the maintainers and lists concerned by the patch. This cover letter is sent to every list concerned by this series. This series is based on linux-next next-20170929. Each patch has been tested by building the relevant files with W=1. This series contains the following patches: [PATCH 01/18] sound: use ARRAY_SIZE [PATCH 02/18] tracing/filter: use ARRAY_SIZE [PATCH 03/18] media: use ARRAY_SIZE [PATCH 04/18] IB/mlx5: Use ARRAY_SIZE [PATCH 05/18] net: use ARRAY_SIZE [PATCH 06/18] drm: use ARRAY_SIZE [PATCH 07/18] scsi: bfa: use ARRAY_SIZE [PATCH 08/18] ecryptfs: use ARRAY_SIZE [PATCH 09/18] nfsd: use ARRAY_SIZE [PATCH 10/18] orangefs: use ARRAY_SIZE [PATCH 11/18] dm space map metadata: use ARRAY_SIZE [PATCH 12/18] x86: use ARRAY_SIZE [PATCH 13/18] tpm: use ARRAY_SIZE [PATCH 14/18] ipmi: use ARRAY_SIZE [PATCH 15/18] acpi: use ARRAY_SIZE [PATCH 16/18] media: staging: atomisp: use ARRAY_SIZE [PATCH 17/18] staging: rtl8723bs: use ARRAY_SIZE [PATCH 18/18] staging: rtlwifi: use ARRAY_SIZE [1]: https://lkml.org/lkml/2017/9/13/689

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[alsa-devel] [PATCH] ALSA: sh: aica: 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 requires adding a pointer to hold the timer's target substream, as there won't be a way to pass this in the future. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Bhumika Goyal <bhumirks(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> 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/sh/aica.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index fdc680ae8aa0..2b26311405a4 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -299,14 +299,14 @@ static void run_spu_dma(struct work_struct *work) } } -static void aica_period_elapsed(unsigned long timer_var) +static void aica_period_elapsed(struct timer_list *t) { + struct snd_card_aica *dreamcastcard = from_timer(dreamcastcard, + t, timer); + struct snd_pcm_substream *substream = dreamcastcard->timer_substream; /*timer function - so cannot sleep */ int play_period; struct snd_pcm_runtime *runtime; - struct snd_pcm_substream *substream; - struct snd_card_aica *dreamcastcard; - substream = (struct snd_pcm_substream *) timer_var; runtime = substream->runtime; dreamcastcard = substream->pcm->private_data; /* Have we played out an additional period? */ @@ -336,12 +336,12 @@ static void spu_begin_dma(struct snd_pcm_substream *substream) /*get the queue to do the work */ schedule_work(&(dreamcastcard->spu_dma_work)); /* Timer may already be running */ - if (unlikely(dreamcastcard->timer.data)) { + if (unlikely(dreamcastcard->timer_substream)) { mod_timer(&dreamcastcard->timer, jiffies + 4); return; } - setup_timer(&dreamcastcard->timer, aica_period_elapsed, - (unsigned long) substream); + timer_setup(&dreamcastcard->timer, aica_period_elapsed, 0); + dreamcastcard->timer_substream = substream; mod_timer(&dreamcastcard->timer, jiffies + 4); } @@ -379,7 +379,7 @@ static int snd_aicapcm_pcm_close(struct snd_pcm_substream { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; flush_work(&(dreamcastcard->spu_dma_work)); - if (dreamcastcard->timer.data) + if (dreamcastcard->timer_substream) del_timer(&dreamcastcard->timer); kfree(dreamcastcard->channel); spu_disable(); @@ -600,7 +600,7 @@ static int snd_aica_probe(struct platform_device *devptr) { int err; struct snd_card_aica *dreamcastcard; - dreamcastcard = kmalloc(sizeof(struct snd_card_aica), GFP_KERNEL); + dreamcastcard = kzalloc(sizeof(struct snd_card_aica), GFP_KERNEL); if (unlikely(!dreamcastcard)) return -ENOMEM; err = snd_card_new(&devptr->dev, index, SND_AICA_DRIVER, @@ -619,8 +619,6 @@ static int snd_aica_probe(struct platform_device *devptr) err = snd_aicapcmchip(dreamcastcard, 0); if (unlikely(err < 0)) goto freedreamcast; - dreamcastcard->timer.data = 0; - dreamcastcard->channel = NULL; /* Add basic controls */ err = add_aicamixer_controls(dreamcastcard); if (unlikely(err < 0)) -- 2.7.4 -- Kees Cook Pixel Security

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[alsa-devel] [PATCH 0/2] ASoC: img-i2s-in: Add power management
by Ed Blake 05 Oct '17

05 Oct '17

This patch set adds power managment to the img-i2s-in driver. The first patch adds system PM suspend / resume handling and the second patch adds runtime PM. Ed Blake (2): ASoC: img-i2s-in: Add suspend / resume handling ASoC: img-i2s-in: Add runtime PM sound/soc/img/img-i2s-in.c | 126 ++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 114 insertions(+), 12 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] conf: HdmiLpeAudio: remove the "front" pcm definition
by Tanu Kaskinen 05 Oct '17

05 Oct '17

PulseAudio assumes that the "front" pcm device always refers to an analog device, not HDMI. While that assumption is not really valid, the reality is that without that assumption PulseAudio can't know whether "front" and "hdmi" refer to a different or the same device. The HDMI LPE driver doesn't allow audio streaming while the HDMI cable is unplugged, so PulseAudio has to know when it's plugged in and when it's not. If both "front" and "hdmi" devices exist, PulseAudio will notice that HDMI is unplugged, but it doesn't know that "front" refers to the same device, and PulseAudio will try to use the "front" device with bad consequences. The kernel driver's refusal to stream any audio makes PulseAudio enter an infinite loop and then the kernel kills PulseAudio, because it consumes too much CPU time in a realtime thread. While the looping in PulseAudio could probably be fixed, that wouldn't change the fact that PulseAudio thinks that there is an analog device. I believe it's best to avoid having the same device as both "front" and "hdmi" in alsa-lib. I removed also the surround configuration includes. I don't think they had any effect anyway, so I wonder why they were there in the first place. BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=100488 --- src/conf/cards/HdmiLpeAudio.conf | 24 ------------------------ 1 file changed, 24 deletions(-) diff --git a/src/conf/cards/HdmiLpeAudio.conf b/src/conf/cards/HdmiLpeAudio.conf index 9fa30da0..a1e493da 100644 --- a/src/conf/cards/HdmiLpeAudio.conf +++ b/src/conf/cards/HdmiLpeAudio.conf @@ -2,30 +2,6 @@ # Configuration for the Intel HDMI/DP LPE audio # -<confdir:pcm/front.conf> - -HdmiLpeAudio.pcm.front.0 { - @args [ CARD ] - @args.CARD { - type string - } - type softvol - slave.pcm { - type hw - card $CARD - } - control { - name "PCM Playback Volume" - card $CARD - } -} - -<confdir:pcm/surround40.conf> -<confdir:pcm/surround21.conf> -<confdir:pcm/surround41.conf> -<confdir:pcm/surround50.conf> -<confdir:pcm/surround51.conf> - <confdir:pcm/hdmi.conf> HdmiLpeAudio.pcm.hdmi.0 { -- 2.14.2

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[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

1 0

[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 ]---

5 15

[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

3 4

[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

2 1

[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

2 4

[alsa-devel] [PATCH] ASoC: img-parallel-out: Add pm_runtime_get/put to set_fmt callback
by Ed Blake 04 Oct '17

04 Oct '17

Add pm_runtime_get_sync and pm_runtime_put calls to set_fmt callback function. This fixes a bus error during boot when CONFIG_SUSPEND is defined when this function gets called while the device is runtime disabled and device registers are accessed while the clock is disabled. Signed-off-by: Ed Blake <ed.blake(a)sondrel.com> --- sound/soc/img/img-parallel-out.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/img/img-parallel-out.c b/sound/soc/img/img-parallel-out.c index 23b0f0f..2fc8a63 100644 --- a/sound/soc/img/img-parallel-out.c +++ b/sound/soc/img/img-parallel-out.c @@ -164,9 +164,11 @@ static int img_prl_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } + pm_runtime_get_sync(prl->dev); reg = img_prl_out_readl(prl, IMG_PRL_OUT_CTL); reg = (reg & ~IMG_PRL_OUT_CTL_EDGE_MASK) | control_set; img_prl_out_writel(prl, reg, IMG_PRL_OUT_CTL); + pm_runtime_put(prl->dev); return 0; } -- 1.9.1

2 2

[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

3 2