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

[alsa-devel] [PATCH] ASoC: dapm: Fix typos in comment
by Peng Donglin 27 Sep '17

27 Sep '17

Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcef67a..0867eaa 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4167,7 +4167,7 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, * @dapm: DAPM context * @pin: pin name * - * Enables input/output pin and its parents or children widgets iff there is + * Enables input/output pin and its parents or children widgets if there is * a valid audio route and active audio stream. * * Requires external locking. @@ -4187,7 +4187,7 @@ int snd_soc_dapm_enable_pin_unlocked(struct snd_soc_dapm_context *dapm, * @dapm: DAPM context * @pin: pin name * - * Enables input/output pin and its parents or children widgets iff there is + * Enables input/output pin and its parents or children widgets if there is * a valid audio route and active audio stream. * * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to -- 1.9.1

3 2

[alsa-devel] usb/sound/caiaq: warning in init_card/usb_submit_urb
by Andrey Konovalov 27 Sep '17

27 Sep '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 on the actual type of the endpoint. usb 1-1: BOGUS urb xfer, pipe 3 != type 1 ------------[ cut here ]------------ WARNING: CPU: 1 PID: 1150 at drivers/usb/core/urb.c:449 usb_submit_urb+0xf8a/0x11d0 Modules linked in: CPU: 1 PID: 1150 Comm: kworker/1:1 Not tainted 4.14.0-rc2-42660-g24b7bd59eec0 #277 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Workqueue: usb_hub_wq hub_event task: ffff880069428000 task.stack: ffff880069440000 RIP: 0010:usb_submit_urb+0xf8a/0x11d0 drivers/usb/core/urb.c:448 RSP: 0018:ffff8800694462e8 EFLAGS: 00010282 RAX: 0000000000000029 RBX: ffff88006a570ef0 RCX: 0000000000000000 RDX: 0000000000000029 RSI: ffffffff85876dc0 RDI: ffffed000d288c4f RBP: ffff8800694463e8 R08: 1ffff1000d288af5 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 1ffff1000d288c64 R13: 0000000000000001 R14: 0000000000000003 R15: ffff8800697c4288 FS: 0000000000000000(0000) GS:ffff88006c900000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007facf21b4000 CR3: 000000006a6d4000 CR4: 00000000000006e0 Call Trace: init_card sound/usb/caiaq/device.c:467 snd_probe+0x81c/0x1150 sound/usb/caiaq/device.c:525 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 4e 92 8b ff 45 89 e8 44 89 f1 4c 89 fa 48 89 c6 48 c7 c7 c0 5a c8 85 e8 f0 4a dd fd <0f> ff e9 9b f7 ff ff e8 da cb 26 fe e9 80 f7 ff ff e8 70 a1 f4 ---[ end trace 1d323f93391df1ca ]---

1 0

[alsa-devel] [PATCH 0/3] ALSA: BeBoB: Fine-tuning for six function implementations
by SF Markus Elfring 27 Sep '17

27 Sep '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 6 Sep 2017 12:16:54 +0200 Three update suggestions were taken into account from static source code analysis. Markus Elfring (3): Use common error handling code in snd_bebob_stream_start_duplex() Adjust six checks for null pointers Improve a size determination in parse_stream_formation() sound/firewire/bebob/bebob_stream.c | 35 +++++++++++++++++------------------ 1 file changed, 17 insertions(+), 18 deletions(-) -- 2.14.1

2 8

[alsa-devel] [PATCH v2] ASoC: Intel: Kbl: Add Playback DAI for fixup
by Harsha Priya 27 Sep '17

27 Sep '17

'Kbl Audio Headset Playback' FE DAI also needs SSP hw param fix. Add this dai also to be handled in kabylake_ssp_fixup() call. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> --- v2: change patch title and log sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 88ff542..e767283 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -302,6 +302,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || + !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; channels->min = channels->max = 2; -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: add Component level items
by Kuninori Morimoto 27 Sep '17

27 Sep '17

Hi Mark These are resend of one of "replace component" patch series. I posted it on below, but few weeks past, and its base branch was updated now. I need/want to resend these. Subject: Prepare to covert Codec/Platform into Component Date: Fri, 01 Sep 2017 14:18:11 +0900 Because it is big patch-set, to avoid sending patch bomb, I will repost it as small amount of patch set. To review from big picture, please check above patch-set. I will continue to resend remainings of it if this small patch-set were accepted Kuninori Morimoto (2): ASoC: add Component level pcm_new/pcm_free v2 ASoC: add Component level set_bias_level include/sound/soc.h | 14 +++++++++++++ sound/soc/soc-core.c | 55 +++++++++++++++++++++++++++++++++++++++++++++++++++- sound/soc/soc-pcm.c | 29 +++++++++++++++++++-------- 3 files changed, 89 insertions(+), 9 deletions(-) -- 1.9.1 Best regards --- Kuninori Morimoto

2 4

[alsa-devel] [RFC PATCH 0/7] Fix year 2038 issue for sound subsystem
by Baolin Wang 27 Sep '17

27 Sep '17

Since many structures will use timespec type variables to record time stamp in uapi/asound.h, which are not year 2038 safe on 32bit system. This patchset tries to introduce new structures removing timespec type to compatible native mode and compat mode. Moreover this patchset also converts the internal structrures to use timespec64 type and related APIs. Baolin Wang (7): sound: Replace timespec with timespec64 sound: core: Avoid using timespec for struct snd_pcm_status sound: core: Avoid using timespec for struct snd_pcm_sync_ptr sound: core: Avoid using timespec for struct snd_rawmidi_status sound: core: Avoid using timespec for struct snd_timer_status uapi: sound: Avoid using timespec for struct snd_ctl_elem_value sound: core: Avoid using timespec for struct snd_timer_tread include/sound/pcm.h | 113 ++++++++- include/sound/timer.h | 4 +- include/uapi/sound/asound.h | 15 +- sound/core/pcm.c | 14 +- sound/core/pcm_compat.c | 466 +++++++++++++++++++++++++++++-------- sound/core/pcm_lib.c | 33 +-- sound/core/pcm_native.c | 227 ++++++++++++++---- sound/core/rawmidi.c | 74 +++++- sound/core/rawmidi_compat.c | 90 +++++-- sound/core/timer.c | 247 ++++++++++++++++---- sound/core/timer_compat.c | 25 +- sound/pci/hda/hda_controller.c | 10 +- sound/soc/intel/skylake/skl-pcm.c | 4 +- 13 files changed, 1046 insertions(+), 276 deletions(-) -- 1.7.9.5

5 31

[alsa-devel] [PATCH v2] ASoC: add null_snd_soc_ops and reduce NULL ops check
by Kuninori Morimoto 26 Sep '17

26 Sep '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Double NULL pointer check for ops and ops->func is difficult to read and might be forget to check it if new func was add. This patch adds new null_snd_soc_ops and use it if rtd->dai_link didn't have it to avoid NULL ops, and reduces ops NULL check. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - rebased on mark/topic/core sound/soc/soc-core.c | 5 +++++ sound/soc/soc-pcm.c | 16 ++++++++-------- 2 files changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e162206..1b5b519 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -614,6 +614,8 @@ struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_get_dai_substream); +static const struct snd_soc_ops null_snd_soc_ops; + static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -626,6 +628,9 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( INIT_LIST_HEAD(&rtd->component_list); rtd->card = card; rtd->dai_link = dai_link; + if (!rtd->dai_link->ops) + rtd->dai_link->ops = &null_snd_soc_ops; + rtd->codec_dais = kzalloc(sizeof(struct snd_soc_dai *) * dai_link->num_codecs, GFP_KERNEL); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 975965d..09fed701 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -511,7 +511,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->rx_mask = 0; } - if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { + if (rtd->dai_link->ops->startup) { ret = rtd->dai_link->ops->startup(substream); if (ret < 0) { pr_err("ASoC: %s startup failed: %d\n", @@ -585,7 +585,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) return 0; config_err: - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + if (rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); machine_err: @@ -692,7 +692,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->driver->ops->shutdown(substream, codec_dai); } - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + if (rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); if (platform->driver->ops && platform->driver->ops->close) @@ -751,7 +751,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); - if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { + if (rtd->dai_link->ops->prepare) { ret = rtd->dai_link->ops->prepare(substream); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: machine prepare error:" @@ -855,7 +855,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { + if (rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: machine hw_params" @@ -948,7 +948,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, codec_dai->rate = 0; } - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + if (rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream); mutex_unlock(&rtd->pcm_mutex); @@ -994,7 +994,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) } /* free any machine hw params */ - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + if (rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream); /* free any DMA resources */ @@ -1045,7 +1045,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - if (rtd->dai_link->ops && rtd->dai_link->ops->trigger) { + if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) return ret; -- 1.9.1

2 1

[alsa-devel] [PATCH] Revert "ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members"
by Takashi Sakamoto 26 Sep '17

26 Sep '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

1 0

[alsa-devel] [Resend] [PATCH v1 3/4] ASoC: dapm: Avoid creating kcontrol for params
by yesanishhere＠gmail.com 25 Sep '17

25 Sep '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 v1: - Included the review comments from charles regarding releasing memory in error path. sound/soc/soc-dapm.c | 130 ++++++++++++++++++++++++++++++--------------------- 1 file changed, 78 insertions(+), 52 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d55cac6..e659c74 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3778,18 +3778,13 @@ 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) +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 +3793,9 @@ 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 +3812,90 @@ 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], - sizeof(struct snd_kcontrol_new), - GFP_KERNEL); - if (!template.kcontrol_news) { + kcontrol_news = + devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + if (!kcontrol_news) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; goto outfree_private_value; } + return kcontrol_news; + +outfree_private_value: + devm_kfree(card->dev, (void *)*private_value); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + devm_kfree(card->dev, (void *)w_param_text[count]); + 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, count; + + 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; + + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) { + ret = -ENOMEM; + goto param_fail; + } + + /* allocate memory for control, only in case of multiple configs */ + if (num_params > 1) { + 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 outfree_link_name; + } + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); @@ -3899,15 +3927,13 @@ 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: - 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: +outfree_link_name: devm_kfree(card->dev, w_param_text); - +param_fail: + devm_kfree(card->dev, link_name); return ret; } -- 2.5.4 (Apple Git-61)

4 4

[alsa-devel] Applied "ASoC: max98927: Added support for DSP_A and DSP_B format" to the asoc tree
by Mark Brown 25 Sep '17

25 Sep '17

The patch ASoC: max98927: Added support for DSP_A and DSP_B format 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 4eee20246c0fa77a7a7b189ac806d5f78d675546 Mon Sep 17 00:00:00 2001 From: Ryan Lee <ryans.lee(a)maximintegrated.com> Date: Thu, 14 Sep 2017 17:30:38 -0700 Subject: [PATCH] ASoC: max98927: Added support for DSP_A and DSP_B format Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98927.c | 35 ++++++++++++++++++++--------------- sound/soc/codecs/max98927.h | 6 ++++-- 2 files changed, 24 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index d9dbbe72f8ad..6f7b3ef48e25 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -1,7 +1,7 @@ /* * max98927.c -- MAX98927 ALSA Soc Audio driver * - * Copyright (C) 2016 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -146,6 +146,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); unsigned int mode = 0; unsigned int format = 0; + bool use_pdm = false; unsigned int invert = 0; dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); @@ -187,22 +188,27 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - max98927->iface |= SND_SOC_DAIFMT_I2S; format = MAX98927_PCM_FORMAT_I2S; break; case SND_SOC_DAIFMT_LEFT_J: - max98927->iface |= SND_SOC_DAIFMT_LEFT_J; format = MAX98927_PCM_FORMAT_LJ; break; + case SND_SOC_DAIFMT_DSP_A: + format = MAX98927_PCM_FORMAT_TDM_MODE1; + break; + case SND_SOC_DAIFMT_DSP_B: + format = MAX98927_PCM_FORMAT_TDM_MODE0; + break; case SND_SOC_DAIFMT_PDM: - max98927->iface |= SND_SOC_DAIFMT_PDM; + use_pdm = true; break; default: return -EINVAL; } + max98927->iface = fmt & SND_SOC_DAIFMT_FORMAT_MASK; - /* pcm channel configuration */ - if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) { + if (!use_pdm) { + /* pcm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0018_PCM_RX_EN_A, MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, @@ -217,13 +223,11 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 0); - } else regmap_update_bits(max98927->regmap, - MAX98927_R0018_PCM_RX_EN_A, - MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); - - /* pdm channel configuration */ - if (max98927->iface & SND_SOC_DAIFMT_PDM) { + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 0); + } else { + /* pdm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0035_PDM_RX_CTRL, MAX98927_PDM_RX_EN_MASK, 1); @@ -231,10 +235,11 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(max98927->regmap, MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 3); - } else + regmap_update_bits(max98927->regmap, - MAX98927_R0035_PDM_RX_CTRL, - MAX98927_PDM_RX_EN_MASK, 0); + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + } return 0; } diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index ece6a608cbe1..bf7a6f92562a 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -1,7 +1,7 @@ /* * max98927.h -- MAX98927 ALSA Soc Audio driver * - * Copyright 2013-15 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -161,7 +161,9 @@ #define MAX98927_PCM_MODE_CFG_FORMAT_SHIFT (3) #define MAX98927_PCM_FORMAT_I2S (0x0 << 0) #define MAX98927_PCM_FORMAT_LJ (0x1 << 0) - +#define MAX98927_PCM_FORMAT_TDM_MODE0 (0x3 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE1 (0x4 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE2 (0x5 << 0) #define MAX98927_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) -- 2.14.1

1 0