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

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix to fail safely if module not available in path
by Subhransu S. Prusty 31 Dec '16

31 Dec '16

From: G Kranthi <gudishax.kranthikumar(a)intel.com> If a module is not available in a pipeline, fail safely rather than causing oops. Signed-off-by: G Kranthi <gudishax.kranthikumar(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- sound/soc/intel/skylake/skl-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 10fa10d..7e97ae6 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -254,6 +254,9 @@ static int skl_pcm_open(struct snd_pcm_substream *substream, snd_pcm_set_sync(substream); mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream); + if (!mconfig) + return -EINVAL; + skl_tplg_d0i3_get(skl, mconfig->d0i3_caps); return 0; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: nau8825: fix invalid configuration in Pre-Scalar of FLL
by John Hsu 31 Dec '16

31 Dec '16

The clk_ref_div is not configured in the correct position of the register. The patch fixes that clk_ref_div, Pre-Scalar, is assigned the wrong value. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 3 ++- sound/soc/codecs/nau8825.h | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index f69f738..e4fb017 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2078,7 +2078,8 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, NAU8825_FLL_INTEGER_MASK, fll_param->fll_int); /* FLL pre-scaler */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL4, - NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); + NAU8825_FLL_REF_DIV_MASK, + fll_param->clk_ref_div << NAU8825_FLL_REF_DIV_SFT); /* select divided VCO input */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, NAU8825_FLL_CLK_SW_MASK, NAU8825_FLL_CLK_SW_REF); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index fa571ec..5732ce1 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -137,7 +137,8 @@ #define NAU8825_FLL_CLK_SRC_FS (0x3 << NAU8825_FLL_CLK_SRC_SFT) /* FLL4 (0x07) */ -#define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) +#define NAU8825_FLL_REF_DIV_SFT 10 +#define NAU8825_FLL_REF_DIV_MASK (0x3 << NAU8825_FLL_REF_DIV_SFT) /* FLL5 (0x08) */ #define NAU8825_FLL_PDB_DAC_EN (0x1 << 15) -- 2.6.4

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Re: [alsa-devel] [PATCH] ALSA: snd-usb: fix IRQ triggered NULL pointer dereference
by Takashi Iwai 30 Dec '16

30 Dec '16

On Wed, 21 Dec 2016 11:38:54 +0100, Ioan-Adrian Ratiu wrote: > > >> > Please take the time to fully analyze my patch and let's have a > >> > discussion based on it, not reject it outright and replace it with > >> > a quick and dirty delay hack. > >> > >> OK. I'll deliberately check it again so that I have no overlooks. I > >> with this thread also catch the other developers enough helpful to > >> you. (I just eventually caught your patch in LKML and not developer > >> for this category of devices.) > > > > Sorry for the late reply, as I've been (still) off and had bad net > > connections. > > > > About your fix: Sakamoto-san is right, it's no good way to fix this > > kind or problem. The easiest option right now is just to revert my > > previous fix, as it obviously introduces another regression. The > > correct fix will be taken after that. > > > > I'm going to prepare a revert patch and ask Linus to take it before > > rc1. > > I agree with reverting the initial commit decision because my problem > disappears with it. > > But can you please state a reason for why my patch is "no good way to > fix"? Being too intrusive is not a good reason. "Being too intrusive" is the exact reason why it's not good as a "regression fix" like this case. The logic you've implemented in the patch itself looks good (although the code introduces a bug, the unbalance of snd_usb_*lock_shutdown()). The only point I couldn't take it is that it's rather a fundamental change, not a quick fix for a regression. What's the worst case scenario in a regression fix? It's when a fix introduces yet another regression. It'd be much worse if the new regression is deeper. The complicated logical change has a potential risk of such. Thus an intrusive change is not always good as a "regression fix". In general, if the change were trivial, it's obviously OK to take as a regression fix -- where the logic is also trivial in most cases, too. But when the fix becomes complex, we need to rethink. Especially when the original buggy commit is small, reverting it is often better as a clear cut. Think in that way: you're addressing a deeper problem that was revealed accidentally by my previous bad fix. Writing the change as if it were merely a regression fix gives the wrong understanding to readers :) That said, I'd appreciate if you respin the fix again with a combination of my previous fix and brush up the code a bit as a whole, and document it not as a regression fix but as a complete fix of the existing races. I can write it in my side quickly, but it'd be almost in the same form as you posted, I suppose. thanks, Takashi

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[alsa-devel] [PATCH] alsactl: Check existence of `alsactl` in udev rule
by Paul Menzel 30 Dec '16

30 Dec '16

From: Jordi Mallach <jordi(a)debian.org> Date: Thu, 10 Jan 2013 00:17:58 +0000 Include the line `TEST=="/usr/sbin/alsactl"` in the udev rule, to properly fix the state restoring for users with split `/usr` filesystems [1]. Upstream the patch from the Debian package [2]. [1] https://bugs.debian.org/670490 "Debian Bug report logs - #670490 alsa-utils: Restore sound volume in udev" [2] https://sources.debian.net/src/alsa-utils/1.1.2-1/debian/patches/udev_test_… Signed-off-by: Paul Menzel <paulepanter(a)users.sourceforge.net> CC: Jordi Mallach <jordi(a)debian.org> --- Please apply with `git am saved-messages.mbox`. alsactl/90-alsa-restore.rules.in | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/alsactl/90-alsa-restore.rules.in b/alsactl/90-alsa-restore.rules.in index c0c1b23..f190b85 100644 --- a/alsactl/90-alsa-restore.rules.in +++ b/alsactl/90-alsa-restore.rules.in @@ -2,7 +2,7 @@ ACTION=="add", SUBSYSTEM=="sound", KERNEL=="controlC*", KERNELS!="card*", TEST== GOTO="alsa_restore_end" LABEL="alsa_restore_go" -TEST!="@daemonswitch@", RUN+="@sbindir@/alsactl restore $attr{device/number}" -TEST=="@daemonswitch@", RUN+="@sbindir@/alsactl nrestore $attr{device/number}" +TEST!="@daemonswitch@", TEST=="@sbindir@/alsactl", RUN+="@sbindir@/alsactl restore $attr{device/number}" +TEST=="@daemonswitch@", TEST=="@sbindir@/alsactl", RUN+="@sbindir@/alsactl nrestore $attr{device/number}" LABEL="alsa_restore_end" -- 2.11.0

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[alsa-devel] [PATCH 6/6] alsa-lib:pcm: don't return bogus buffer levels in xrun state
by sutar.mounesh＠gmail.com 30 Dec '16

30 Dec '16

From: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 320c948..127b06f 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -884,6 +884,9 @@ static snd_pcm_sframes_t snd_pcm_dmix_avail_update(snd_pcm_t *pcm) if (( err = snd_pcm_dmix_sync_ptr(pcm)) < 0) return err; } + if (dmix->state == SND_PCM_STATE_XRUN) + return -EPIPE; + return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index f4fa32b..5b897f5 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -570,6 +570,9 @@ static snd_pcm_sframes_t snd_pcm_dshare_avail_update(snd_pcm_t *pcm) if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) return err; } + if (dshare->state == SND_PCM_STATE_XRUN) + return -EPIPE; + return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 0729a5a..61579e8 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -454,6 +454,9 @@ static snd_pcm_sframes_t snd_pcm_dsnoop_avail_update(snd_pcm_t *pcm) if (err < 0) return err; } + if (dsnoop->state == SND_PCM_STATE_XRUN) + return -EPIPE; + return snd_pcm_mmap_capture_avail(pcm); } -- 1.7.9.5

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[alsa-devel] [PATCH 5/6] alsa-lib:pcm: check state before enter poll on timer.
by sutar.mounesh＠gmail.com 30 Dec '16

30 Dec '16

From: Andreas Pape <apape(a)de.adit-jv.com> To avoid the chances of timeout, we need to check the enter poll in state xrun. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 779845c..2928512 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -549,6 +549,42 @@ int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix) snd_timer_stop(dmix->timer); return 0; } +/* + * This is the only operation guaranteed to be called before entering poll(). + * Direct plugins use fd of snd_timer to poll on, these timers do NOT check state of substream in kernel by intention. + * Only the enter to xrun might be notified once (SND_TIMER_EVENT_MSTOP). + * If xrun event was not correctly handled or was ignored it will never be evaluated again afterwards. + * This will result in snd_pcm_wait() always returning timeout. + * In contrast poll() on pcm hardware checks ALSA state and will immediately return POLLERR on XRUN. + * + * To prevent timeout and applications endlessly spinning without xrun detected we add a state check + * here which may trigger the xrun sequence. + * + * return count of filled descriptors or negative error code + */ +int snd_pcm_direct_poll_descriptors(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int space) +{ + if (pcm->poll_fd < 0) { + SNDMSG("poll_fd < 0"); + return -EIO; + } + if (space >= 1 && pfds) { + pfds->fd = pcm->poll_fd; + pfds->events = pcm->poll_events | POLLERR | POLLNVAL; + } else { + return 0; + } + + /*this will also evaluate slave state and enter xrun if necessary*/ + switch (snd_pcm_state(pcm)) { + case SND_PCM_STATE_XRUN: + return -EPIPE; + default: + break; + } + return 1; +} + int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 569d2be..3a1c1c0 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -301,6 +301,7 @@ int snd_pcm_direct_parse_bindings(snd_pcm_direct_t *dmix, snd_config_t *cfg); int snd_pcm_direct_nonblock(snd_pcm_t *pcm, int nonblock); int snd_pcm_direct_async(snd_pcm_t *pcm, int sig, pid_t pid); +int snd_pcm_direct_poll_descriptors(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int space); int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents); int snd_pcm_direct_info(snd_pcm_t *pcm, snd_pcm_info_t * info); int snd_pcm_direct_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 5c3eb2d..320c948 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -463,17 +463,23 @@ static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) static snd_pcm_state_t snd_pcm_dmix_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; snd_pcm_state_t state; state = snd_pcm_state(dmix->spcm); switch (state) { - case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dmix->state = state; return state; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) + return err; + break; default: break; } + snd_pcm_direct_client_chk_xrun(dmix, pcm); + if (dmix->state == STATE_RUN_PENDING) return SNDRV_PCM_STATE_RUNNING; return dmix->state; @@ -969,7 +975,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dmix_fast_ops = { .avail_update = snd_pcm_dmix_avail_update, .mmap_commit = snd_pcm_dmix_mmap_commit, .htimestamp = snd_pcm_dmix_htimestamp, - .poll_descriptors = NULL, + .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_dmix_poll_revents, }; diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 707e2c2..f4fa32b 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -255,17 +255,22 @@ static int snd_pcm_dshare_status(snd_pcm_t *pcm, snd_pcm_status_t * status) static snd_pcm_state_t snd_pcm_dshare_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; snd_pcm_state_t state; state = snd_pcm_state(dshare->spcm); switch (state) { - case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dshare->state = state; return state; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) + return err; + break; default: break; } + snd_pcm_direct_client_chk_xrun(dshare, pcm); if (dshare->state == STATE_RUN_PENDING) return SNDRV_PCM_STATE_RUNNING; return dshare->state; @@ -646,7 +651,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dshare_fast_ops = { .avail_update = snd_pcm_dshare_avail_update, .mmap_commit = snd_pcm_dshare_mmap_commit, .htimestamp = snd_pcm_dshare_htimestamp, - .poll_descriptors = NULL, + .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_direct_poll_revents, }; diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 512e957..0729a5a 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -208,17 +208,22 @@ static int snd_pcm_dsnoop_status(snd_pcm_t *pcm, snd_pcm_status_t * status) static snd_pcm_state_t snd_pcm_dsnoop_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dsnoop = pcm->private_data; + int err; snd_pcm_state_t state; state = snd_pcm_state(dsnoop->spcm); switch (state) { - case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dsnoop->state = state; return state; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dsnoop)) < 0) + return err; + break; default: break; } + snd_pcm_direct_client_chk_xrun(dsnoop, pcm); return dsnoop->state; } @@ -531,7 +536,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dsnoop_fast_ops = { .avail_update = snd_pcm_dsnoop_avail_update, .mmap_commit = snd_pcm_dsnoop_mmap_commit, .htimestamp = snd_pcm_dsnoop_htimestamp, - .poll_descriptors = NULL, + .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_direct_poll_revents, }; -- 1.7.9.5

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[alsa-devel] [PATCH] Revert "ASoC: core: replace aux_comp_list to component_dev_list"
by Nikita Yushchenko 29 Dec '16

29 Dec '16

This reverts commit 1a653aa44725668590b36bbe2d7fe4736a69f055 ("ASoC: core: replace aux_comp_list to component_dev_list"). That commit tries to remove card->aux_comp_list, using flagged entries in card->component_dev_list instead. However, components are added to card->component_dev_list in soc_probe_component(). Before that commit, it traversed card->aux_comp_list and found aux devices added by soc_bind_aux_dev(). After that commit, it traverses card->component_dev_list and finds no aux devices, because soc_probe_component() was not yet called for them. Thus all aux devices are lost and any setup that needs them no longer works. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> --- include/sound/soc.h | 4 +++- sound/soc/soc-core.c | 17 +++++------------ 2 files changed, 8 insertions(+), 13 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2b502f6cc6d0..8172a512632a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -809,10 +809,10 @@ struct snd_soc_component { unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ unsigned int registered_as_component:1; - unsigned int auxiliary:1; /* for auxiliary component of the card */ unsigned int suspended:1; /* is in suspend PM state */ struct list_head list; + struct list_head list_aux; /* for auxiliary component of the card */ struct list_head card_list; struct snd_soc_dai_driver *dai_drv; @@ -1152,6 +1152,7 @@ struct snd_soc_card { */ struct snd_soc_aux_dev *aux_dev; int num_aux_devs; + struct list_head aux_comp_list; const struct snd_kcontrol_new *controls; int num_controls; @@ -1547,6 +1548,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); + INIT_LIST_HEAD(&card->aux_comp_list); INIT_LIST_HEAD(&card->component_dev_list); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f1901bb1466e..15657a65e4ca 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1747,8 +1747,7 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) } component->init = aux_dev->init; - component->auxiliary = 1; - + list_add(&component->list_aux, &card->aux_comp_list); return 0; err_defer: @@ -1764,10 +1763,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - list_for_each_entry(comp, &card->component_dev_list, card_list) { - if (!comp->auxiliary) - continue; - + list_for_each_entry(comp, &card->aux_comp_list, list_aux) { if (comp->driver->probe_order == order) { ret = soc_probe_component(card, comp); if (ret < 0) { @@ -1791,14 +1787,11 @@ static void soc_remove_aux_devices(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { list_for_each_entry_safe(comp, _comp, - &card->component_dev_list, card_list) { - - if (!comp->auxiliary) - continue; - + &card->aux_comp_list, list_aux) { if (comp->driver->remove_order == order) { soc_remove_component(comp); - comp->auxiliary = 0; + /* remove it from the card's aux_comp_list */ + list_del(&comp->list_aux); } } } -- 2.1.4

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[alsa-devel] [PATCH] ASoC: Fix binding and probing of auxiliary components
by Sylwester Nawrocki 29 Dec '16

29 Dec '16

Currently binding of auxiliary devices doesn't work as in soc_bind_aux_dev() function a bound component is not being added to any list and in soc_probe_aux_devices() we are trying to walk the component_dev_list list to probe auxiliary components but at that time this list doesn't contain any auxiliary components since they are being added to the card only in soc_probe_component(). This patch adds a list to the card where are stored bound but not probed auxiliary devices, so that all aux devices can be probed. Fixes: 1a653aa44725 "ASoC: core: replace aux_comp_list to component_dev_list" Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- I'm not sure if this is a correct approach, without this patch the amplifier component doesn't get bound to the card on the TM2 board. --- include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 10 +++++----- 2 files changed, 8 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2b502f6..b86168a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -813,6 +813,7 @@ struct snd_soc_component { unsigned int suspended:1; /* is in suspend PM state */ struct list_head list; + struct list_head card_aux_list; /* for auxiliary bound components */ struct list_head card_list; struct snd_soc_dai_driver *dai_drv; @@ -1152,6 +1153,7 @@ struct snd_soc_card { */ struct snd_soc_aux_dev *aux_dev; int num_aux_devs; + struct list_head aux_comp_list; const struct snd_kcontrol_new *controls; int num_controls; @@ -1547,6 +1549,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); + INIT_LIST_HEAD(&card->aux_comp_list); INIT_LIST_HEAD(&card->component_dev_list); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f1901bb..baa1afa 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1748,6 +1748,7 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) component->init = aux_dev->init; component->auxiliary = 1; + list_add(&component->card_aux_list, &card->aux_comp_list); return 0; @@ -1758,16 +1759,14 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) static int soc_probe_aux_devices(struct snd_soc_card *card) { - struct snd_soc_component *comp; + struct snd_soc_component *comp, *tmp; int order; int ret; for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - list_for_each_entry(comp, &card->component_dev_list, card_list) { - if (!comp->auxiliary) - continue; - + list_for_each_entry_safe(comp, tmp, &card->aux_comp_list, + card_aux_list) { if (comp->driver->probe_order == order) { ret = soc_probe_component(card, comp); if (ret < 0) { @@ -1776,6 +1775,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card) comp->name, ret); return ret; } + list_del(&comp->card_aux_list); } } } -- 1.9.1

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[alsa-devel] Stream format conversion in SoC driver
by Giedrius Trainavičius 28 Dec '16

28 Dec '16

Hi, We are developing a soundcard for Raspberry Pi, you may find the current driver source code here: https://github.com/raspberrypi/linux/blob/rpi-4.4.y/sound/soc/bcm/pisound.c The card natively supports 24 and 32 bit PCM streams at 48kHz, 96kHz and 192kHz, symmetric capture and playback. Is it possible to have ALSA do the necessary format conversions for the PCM streams in natively unsupported formats to the closest natively supported format? What would be required to enable it? Thank you, Giedrius.

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Re: [alsa-devel] [PATCH] ASoC : fsl_ssi : Correct the condition to check AC97 mode
by Andreas Schwab 28 Dec '16

28 Dec '16

On Dez 28 2016, Timur Tabi <timur(a)tabi.org> wrote: > Andreas Schwab wrote: >>> > + return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97) == >>> > + SND_SOC_DAIFMT_AC97; >> This is never true. > > I think the right parenthesis should be at the end of the expression. That would still test the wrong bitfield. And all that is obsolete anyway since commit 5b64c173cd (of Sep 2015). Andreas. -- Andreas Schwab, schwab(a)linux-m68k.org GPG Key fingerprint = 58CA 54C7 6D53 942B 1756 01D3 44D5 214B 8276 4ED5 "And now for something completely different."

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