September 2018 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH] ASoC: max98373: usleep_range() needs include/delay.h
by Benson Leung 07 Sep '18

07 Sep '18

Hi Grant, On Thu, Sep 06, 2018 at 05:27:28PM -0700, Grant Grundler wrote: > Commit ca917f9fe1a0fab added use of usleep_range() but not > the corresponding "include <linux/delay.h>". The result is > with Chrome OS won't build because warnings are forced > to be errors: > mnt/host/source/src/third_party/kernel/v4.4/sound/soc/codecs/max98373.c:734:2: error: implicit declaration of function 'usleep_range' [-Werror,-Wimplicit-function-declaration] > usleep_range(10000, 11000); > ^ > > Including delay.h "fixes" this. > > Signed-off-by: Grant Grundler <grundler(a)chromium.org> Reviewed-by: Benson Leung <bleung(a)chromium.org> Thanks! -- Benson Leung Staff Software Engineer Chrome OS Kernel Google Inc. bleung(a)google.com Chromium OS Project bleung(a)chromium.org

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[alsa-devel] [asoc:for-4.20 399/404] sound/soc/soc-dapm.c:3768:9: warning: 'ret' may be used uninitialized in this function
by kbuild test robot 06 Sep '18

06 Sep '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 9ab708aef61f5620113269a9d1bdb1543d1207d0 commit: 4a75aae17b2a802a7267206414050408392c374c [399/404] ASoC: dapm: Add support for multi-CODEC CODEC to CODEC links config: x86_64-randconfig-x006-201835 (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout 4a75aae17b2a802a7267206414050408392c374c # save the attached .config to linux build tree make ARCH=x86_64 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 >>): Cyclomatic Complexity 3 sound/soc/soc-dapm.c:dapm_power_one_widget Cyclomatic Complexity 6 sound/soc/soc-dapm.c:snd_soc_dapm_check_dynamic_path Cyclomatic Complexity 2 sound/soc/soc-dapm.c:soc_dapm_update_bits Cyclomatic Complexity 15 sound/soc/soc-dapm.c:dapm_widget_update Cyclomatic Complexity 3 sound/soc/soc-dapm.c:pop_dbg Cyclomatic Complexity 12 sound/soc/soc-dapm.c:dapm_seq_check_event Cyclomatic Complexity 1 sound/soc/soc-dapm.c:dapm_free_path Cyclomatic Complexity 1 sound/soc/soc-dapm.c:dapm_kcontrol_free Cyclomatic Complexity 2 sound/soc/soc-dapm.c:pop_wait Cyclomatic Complexity 13 sound/soc/soc-dapm.c:dapm_seq_run_coalesced Cyclomatic Complexity 42 sound/soc/soc-dapm.c:dapm_seq_run Cyclomatic Complexity 13 sound/soc/soc-dapm.c:dapm_widget_show_component Cyclomatic Complexity 3 sound/soc/soc-dapm.c:dapm_widget_show Cyclomatic Complexity 6 sound/soc/soc-dapm.c:dapm_wcache_lookup Cyclomatic Complexity 2 sound/soc/soc-dapm.c:soc_dapm_read Cyclomatic Complexity 3 sound/soc/soc-dapm.c:dapm_connect_mux Cyclomatic Complexity 6 sound/soc/soc-dapm.c:dapm_set_mixer_path_status Cyclomatic Complexity 3 sound/soc/soc-dapm.c:dapm_connect_mixer Cyclomatic Complexity 17 sound/soc/soc-dapm.c:dapm_update_widget_flags Cyclomatic Complexity 22 sound/soc/soc-dapm.c:snd_soc_dapm_add_path Cyclomatic Complexity 19 sound/soc/soc-dapm.c:snd_soc_dapm_add_route Cyclomatic Complexity 10 sound/soc/soc-dapm.c:snd_soc_dapm_del_route Cyclomatic Complexity 7 sound/soc/soc-dapm.c:dapm_find_widget Cyclomatic Complexity 11 sound/soc/soc-dapm.c:snd_soc_dapm_weak_route Cyclomatic Complexity 4 sound/soc/soc-dapm.c:snd_soc_dapm_set_pin Cyclomatic Complexity 4 sound/soc/soc-dapm.c:dapm_new_dai_link Cyclomatic Complexity 9 sound/soc/soc-dapm.c:dapm_is_shared_kcontrol Cyclomatic Complexity 3 sound/soc/soc-dapm.c:dapm_kcontrol_add_widget Cyclomatic Complexity 17 sound/soc/soc-dapm.c:dapm_widget_power_read_file Cyclomatic Complexity 2 sound/soc/soc-dapm.c:soc_dapm_test_bits Cyclomatic Complexity 8 sound/soc/soc-dapm.c:dapm_supply_check_power Cyclomatic Complexity 2 sound/soc/soc-dapm.c:snd_soc_dapm_free_kcontrol Cyclomatic Complexity 42 sound/soc/soc-dapm.c:snd_soc_dai_link_event Cyclomatic Complexity 2 include/linux/device.h:devm_kmalloc_array Cyclomatic Complexity 1 include/linux/device.h:devm_kcalloc Cyclomatic Complexity 6 sound/soc/soc-dapm.c:snd_soc_dapm_alloc_kcontrol Cyclomatic Complexity 1 sound/soc/soc-dapm.c:dapm_debugfs_cleanup Cyclomatic Complexity 7 sound/soc/soc-dapm.c:dapm_mark_endpoints_dirty Cyclomatic Complexity 1 sound/soc/soc-dapm.c:dapm_kcontrol_get_value Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_kcontrol_widget Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_kcontrol_dapm Cyclomatic Complexity 3 sound/soc/soc-dapm.c:snd_soc_dapm_force_bias_level Cyclomatic Complexity 9 sound/soc/soc-dapm.c:snd_soc_dapm_set_bias_level Cyclomatic Complexity 10 sound/soc/soc-dapm.c:dapm_pre_sequence_async Cyclomatic Complexity 11 sound/soc/soc-dapm.c:dapm_post_sequence_async Cyclomatic Complexity 47 sound/soc/soc-dapm.c:dapm_power_widgets Cyclomatic Complexity 9 sound/soc/soc-dapm.c:soc_dapm_mux_update_power Cyclomatic Complexity 9 sound/soc/soc-dapm.c:soc_dapm_mixer_update_power Cyclomatic Complexity 3 sound/soc/soc-dapm.c:soc_dapm_stream_event Cyclomatic Complexity 9 sound/soc/soc-dapm.c:soc_dapm_shutdown_dapm Cyclomatic Complexity 6 sound/soc/soc-dapm.c:snd_soc_dapm_dai_get_connected_widgets Cyclomatic Complexity 6 sound/soc/soc-dapm.c:dapm_regulator_event Cyclomatic Complexity 4 sound/soc/soc-dapm.c:dapm_pinctrl_event Cyclomatic Complexity 3 sound/soc/soc-dapm.c:dapm_clock_event Cyclomatic Complexity 4 sound/soc/soc-dapm.c:snd_soc_dapm_debugfs_init Cyclomatic Complexity 2 sound/soc/soc-dapm.c:snd_soc_dapm_mux_update_power Cyclomatic Complexity 2 sound/soc/soc-dapm.c:snd_soc_dapm_mixer_update_power Cyclomatic Complexity 6 sound/soc/soc-dapm.c:snd_soc_dapm_free_widget Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_reset_cache Cyclomatic Complexity 6 sound/soc/soc-dapm.c:dapm_free_widgets Cyclomatic Complexity 3 sound/soc/soc-dapm.c:snd_soc_dapm_sync_unlocked Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_sync Cyclomatic Complexity 4 sound/soc/soc-dapm.c:snd_soc_dapm_add_routes Cyclomatic Complexity 2 sound/soc/soc-dapm.c:snd_soc_dapm_del_routes Cyclomatic Complexity 3 sound/soc/soc-dapm.c:snd_soc_dapm_weak_routes Cyclomatic Complexity 11 sound/soc/soc-dapm.c:snd_soc_dapm_get_volsw Cyclomatic Complexity 11 sound/soc/soc-dapm.c:snd_soc_dapm_put_volsw Cyclomatic Complexity 5 sound/soc/soc-dapm.c:snd_soc_dapm_get_enum_double Cyclomatic Complexity 8 sound/soc/soc-dapm.c:snd_soc_dapm_put_enum_double Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_info_pin_switch Cyclomatic Complexity 24 sound/soc/soc-dapm.c:snd_soc_dapm_new_control_unlocked Cyclomatic Complexity 13 sound/soc/soc-dapm.c:dapm_kcontrol_data_alloc Cyclomatic Complexity 14 sound/soc/soc-dapm.c:dapm_create_or_share_kcontrol Cyclomatic Complexity 9 sound/soc/soc-dapm.c:dapm_new_mixer Cyclomatic Complexity 10 sound/soc/soc-dapm.c:dapm_new_mux Cyclomatic Complexity 3 sound/soc/soc-dapm.c:dapm_new_pga Cyclomatic Complexity 13 sound/soc/soc-dapm.c:snd_soc_dapm_new_widgets Cyclomatic Complexity 6 sound/soc/soc-dapm.c:snd_soc_dapm_new_dai Cyclomatic Complexity 8 sound/soc/soc-dapm.c:dapm_connect_dai_link_widgets Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_new_control Cyclomatic Complexity 3 sound/soc/soc-dapm.c:snd_soc_dapm_new_controls Cyclomatic Complexity 6 sound/soc/soc-dapm.c:snd_soc_dapm_new_dai_widgets Cyclomatic Complexity 14 sound/soc/soc-dapm.c:snd_soc_dapm_link_dai_widgets Cyclomatic Complexity 5 sound/soc/soc-dapm.c:snd_soc_dapm_connect_dai_link_widgets Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_stream_event Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_enable_pin_unlocked Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_enable_pin Cyclomatic Complexity 3 sound/soc/soc-dapm.c:snd_soc_dapm_force_enable_pin_unlocked Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_force_enable_pin Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_disable_pin_unlocked Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_disable_pin Cyclomatic Complexity 2 sound/soc/soc-dapm.c:snd_soc_dapm_put_pin_switch Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_nc_pin_unlocked Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_nc_pin Cyclomatic Complexity 2 sound/soc/soc-dapm.c:snd_soc_dapm_get_pin_status Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_get_pin_switch Cyclomatic Complexity 2 sound/soc/soc-dapm.c:snd_soc_dapm_ignore_suspend Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_free Cyclomatic Complexity 7 sound/soc/soc-dapm.c:snd_soc_dapm_shutdown sound/soc/soc-dapm.c: In function 'snd_soc_dai_link_event': >> sound/soc/soc-dapm.c:3768:9: warning: 'ret' may be used uninitialized in this function [-Wmaybe-uninitialized] return ret; ^~~ vim +/ret +3768 sound/soc/soc-dapm.c 4ba1327a Mark Brown 2008-05-13 3613 c74184ed Mark Brown 2012-04-04 3614 static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, c74184ed Mark Brown 2012-04-04 3615 struct snd_kcontrol *kcontrol, int event) c74184ed Mark Brown 2012-04-04 3616 { 4a75aae1 Charles Keepax 2018-09-05 3617 struct snd_soc_dapm_path *path; c74184ed Mark Brown 2012-04-04 3618 struct snd_soc_dai *source, *sink; 249dc495 Charles Keepax 2018-08-15 3619 struct snd_soc_pcm_runtime *rtd = w->priv; c6615082 Nikesh Oswal 2015-02-02 3620 const struct snd_soc_pcm_stream *config = w->params + w->params_select; c74184ed Mark Brown 2012-04-04 3621 struct snd_pcm_substream substream; 9747cec2 Mark Brown 2012-04-26 3622 struct snd_pcm_hw_params *params = NULL; 8053f216 Nicolin Chen 2016-07-26 3623 struct snd_pcm_runtime *runtime = NULL; 3ba66feb Takashi Iwai 2018-07-25 3624 unsigned int fmt; c74184ed Mark Brown 2012-04-04 3625 int ret; c74184ed Mark Brown 2012-04-04 3626 bf4edea8 Takashi Iwai 2013-11-07 3627 if (WARN_ON(!config) || a3423b02 Lars-Peter Clausen 2015-08-11 3628 WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) || a3423b02 Lars-Peter Clausen 2015-08-11 3629 list_empty(&w->edges[SND_SOC_DAPM_DIR_IN]))) bf4edea8 Takashi Iwai 2013-11-07 3630 return -EINVAL; c74184ed Mark Brown 2012-04-04 3631 c74184ed Mark Brown 2012-04-04 3632 /* Be a little careful as we don't want to overflow the mask array */ c74184ed Mark Brown 2012-04-04 3633 if (config->formats) { c74184ed Mark Brown 2012-04-04 3634 fmt = ffs(config->formats) - 1; c74184ed Mark Brown 2012-04-04 3635 } else { 30a6a1a4 Liam Girdwood 2012-11-19 3636 dev_warn(w->dapm->dev, "ASoC: Invalid format %llx specified\n", c74184ed Mark Brown 2012-04-04 3637 config->formats); c74184ed Mark Brown 2012-04-04 3638 fmt = 0; c74184ed Mark Brown 2012-04-04 3639 } c74184ed Mark Brown 2012-04-04 3640 c74184ed Mark Brown 2012-04-04 3641 /* Currently very limited parameter selection */ 9747cec2 Mark Brown 2012-04-26 3642 params = kzalloc(sizeof(*params), GFP_KERNEL); 9747cec2 Mark Brown 2012-04-26 3643 if (!params) { 9747cec2 Mark Brown 2012-04-26 3644 ret = -ENOMEM; 9747cec2 Mark Brown 2012-04-26 3645 goto out; 9747cec2 Mark Brown 2012-04-26 3646 } 9747cec2 Mark Brown 2012-04-26 3647 snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt); c74184ed Mark Brown 2012-04-04 3648 9747cec2 Mark Brown 2012-04-26 3649 hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min = c74184ed Mark Brown 2012-04-04 3650 config->rate_min; 9747cec2 Mark Brown 2012-04-26 3651 hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max = c74184ed Mark Brown 2012-04-04 3652 config->rate_max; c74184ed Mark Brown 2012-04-04 3653 9747cec2 Mark Brown 2012-04-26 3654 hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->min c74184ed Mark Brown 2012-04-04 3655 = config->channels_min; 9747cec2 Mark Brown 2012-04-26 3656 hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max c74184ed Mark Brown 2012-04-04 3657 = config->channels_max; c74184ed Mark Brown 2012-04-04 3658 c74184ed Mark Brown 2012-04-04 3659 memset(&substream, 0, sizeof(substream)); c74184ed Mark Brown 2012-04-04 3660 8053f216 Nicolin Chen 2016-07-26 3661 /* Allocate a dummy snd_pcm_runtime for startup() and other ops() */ 8053f216 Nicolin Chen 2016-07-26 3662 runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); 8053f216 Nicolin Chen 2016-07-26 3663 if (!runtime) { 8053f216 Nicolin Chen 2016-07-26 3664 ret = -ENOMEM; 8053f216 Nicolin Chen 2016-07-26 3665 goto out; 8053f216 Nicolin Chen 2016-07-26 3666 } 8053f216 Nicolin Chen 2016-07-26 3667 substream.runtime = runtime; 249dc495 Charles Keepax 2018-08-15 3668 substream.private_data = rtd; 8053f216 Nicolin Chen 2016-07-26 3669 c74184ed Mark Brown 2012-04-04 3670 switch (event) { c74184ed Mark Brown 2012-04-04 3671 case SND_SOC_DAPM_PRE_PMU: c74184ed Mark Brown 2012-04-04 3672 substream.stream = SNDRV_PCM_STREAM_CAPTURE; 4a75aae1 Charles Keepax 2018-09-05 3673 snd_soc_dapm_widget_for_each_source_path(w, path) { 4a75aae1 Charles Keepax 2018-09-05 3674 source = path->source->priv; 4a75aae1 Charles Keepax 2018-09-05 3675 9900a422 Kuninori Morimoto 2017-09-25 3676 if (source->driver->ops->startup) { 4a75aae1 Charles Keepax 2018-09-05 3677 ret = source->driver->ops->startup(&substream, 4a75aae1 Charles Keepax 2018-09-05 3678 source); 9b8ef9f6 Jeeja KP 2015-10-20 3679 if (ret < 0) { 9b8ef9f6 Jeeja KP 2015-10-20 3680 dev_err(source->dev, 4a75aae1 Charles Keepax 2018-09-05 3681 "ASoC: startup() failed: %d\n", 4a75aae1 Charles Keepax 2018-09-05 3682 ret); 9b8ef9f6 Jeeja KP 2015-10-20 3683 goto out; 9b8ef9f6 Jeeja KP 2015-10-20 3684 } 9b8ef9f6 Jeeja KP 2015-10-20 3685 source->active++; 9b8ef9f6 Jeeja KP 2015-10-20 3686 } 93e6958a Benoit Cousson 2014-07-08 3687 ret = soc_dai_hw_params(&substream, params, source); 93e6958a Benoit Cousson 2014-07-08 3688 if (ret < 0) 9747cec2 Mark Brown 2012-04-26 3689 goto out; 4a75aae1 Charles Keepax 2018-09-05 3690 } c74184ed Mark Brown 2012-04-04 3691 c74184ed Mark Brown 2012-04-04 3692 substream.stream = SNDRV_PCM_STREAM_PLAYBACK; 4a75aae1 Charles Keepax 2018-09-05 3693 snd_soc_dapm_widget_for_each_sink_path(w, path) { 4a75aae1 Charles Keepax 2018-09-05 3694 sink = path->sink->priv; 4a75aae1 Charles Keepax 2018-09-05 3695 9900a422 Kuninori Morimoto 2017-09-25 3696 if (sink->driver->ops->startup) { 4a75aae1 Charles Keepax 2018-09-05 3697 ret = sink->driver->ops->startup(&substream, 4a75aae1 Charles Keepax 2018-09-05 3698 sink); 9b8ef9f6 Jeeja KP 2015-10-20 3699 if (ret < 0) { 9b8ef9f6 Jeeja KP 2015-10-20 3700 dev_err(sink->dev, 4a75aae1 Charles Keepax 2018-09-05 3701 "ASoC: startup() failed: %d\n", 4a75aae1 Charles Keepax 2018-09-05 3702 ret); 9b8ef9f6 Jeeja KP 2015-10-20 3703 goto out; 9b8ef9f6 Jeeja KP 2015-10-20 3704 } 9b8ef9f6 Jeeja KP 2015-10-20 3705 sink->active++; 9b8ef9f6 Jeeja KP 2015-10-20 3706 } 93e6958a Benoit Cousson 2014-07-08 3707 ret = soc_dai_hw_params(&substream, params, sink); 93e6958a Benoit Cousson 2014-07-08 3708 if (ret < 0) 9747cec2 Mark Brown 2012-04-26 3709 goto out; 4a75aae1 Charles Keepax 2018-09-05 3710 } c74184ed Mark Brown 2012-04-04 3711 break; c74184ed Mark Brown 2012-04-04 3712 c74184ed Mark Brown 2012-04-04 3713 case SND_SOC_DAPM_POST_PMU: 4a75aae1 Charles Keepax 2018-09-05 3714 snd_soc_dapm_widget_for_each_sink_path(w, path) { 4a75aae1 Charles Keepax 2018-09-05 3715 sink = path->sink->priv; 4a75aae1 Charles Keepax 2018-09-05 3716 da18396f Mark Brown 2013-02-06 3717 ret = snd_soc_dai_digital_mute(sink, 0, da18396f Mark Brown 2013-02-06 3718 SNDRV_PCM_STREAM_PLAYBACK); c74184ed Mark Brown 2012-04-04 3719 if (ret != 0 && ret != -ENOTSUPP) 4a75aae1 Charles Keepax 2018-09-05 3720 dev_warn(sink->dev, 4a75aae1 Charles Keepax 2018-09-05 3721 "ASoC: Failed to unmute: %d\n", ret); 9747cec2 Mark Brown 2012-04-26 3722 ret = 0; 4a75aae1 Charles Keepax 2018-09-05 3723 } c74184ed Mark Brown 2012-04-04 3724 break; c74184ed Mark Brown 2012-04-04 3725 c74184ed Mark Brown 2012-04-04 3726 case SND_SOC_DAPM_PRE_PMD: 4a75aae1 Charles Keepax 2018-09-05 3727 snd_soc_dapm_widget_for_each_sink_path(w, path) { 4a75aae1 Charles Keepax 2018-09-05 3728 sink = path->sink->priv; 4a75aae1 Charles Keepax 2018-09-05 3729 da18396f Mark Brown 2013-02-06 3730 ret = snd_soc_dai_digital_mute(sink, 1, da18396f Mark Brown 2013-02-06 3731 SNDRV_PCM_STREAM_PLAYBACK); c74184ed Mark Brown 2012-04-04 3732 if (ret != 0 && ret != -ENOTSUPP) 4a75aae1 Charles Keepax 2018-09-05 3733 dev_warn(sink->dev, 4a75aae1 Charles Keepax 2018-09-05 3734 "ASoC: Failed to mute: %d\n", ret); 9747cec2 Mark Brown 2012-04-26 3735 ret = 0; 4a75aae1 Charles Keepax 2018-09-05 3736 } 4a75aae1 Charles Keepax 2018-09-05 3737 4a75aae1 Charles Keepax 2018-09-05 3738 snd_soc_dapm_widget_for_each_source_path(w, path) { 4a75aae1 Charles Keepax 2018-09-05 3739 source = path->source->priv; 9b8ef9f6 Jeeja KP 2015-10-20 3740 9b8ef9f6 Jeeja KP 2015-10-20 3741 source->active--; 9900a422 Kuninori Morimoto 2017-09-25 3742 if (source->driver->ops->shutdown) { 9b8ef9f6 Jeeja KP 2015-10-20 3743 substream.stream = SNDRV_PCM_STREAM_CAPTURE; 4a75aae1 Charles Keepax 2018-09-05 3744 source->driver->ops->shutdown(&substream, 4a75aae1 Charles Keepax 2018-09-05 3745 source); 4a75aae1 Charles Keepax 2018-09-05 3746 } 9b8ef9f6 Jeeja KP 2015-10-20 3747 } 9b8ef9f6 Jeeja KP 2015-10-20 3748 4a75aae1 Charles Keepax 2018-09-05 3749 snd_soc_dapm_widget_for_each_sink_path(w, path) { 4a75aae1 Charles Keepax 2018-09-05 3750 sink = path->sink->priv; 4a75aae1 Charles Keepax 2018-09-05 3751 9b8ef9f6 Jeeja KP 2015-10-20 3752 sink->active--; 9900a422 Kuninori Morimoto 2017-09-25 3753 if (sink->driver->ops->shutdown) { 9b8ef9f6 Jeeja KP 2015-10-20 3754 substream.stream = SNDRV_PCM_STREAM_PLAYBACK; 9b8ef9f6 Jeeja KP 2015-10-20 3755 sink->driver->ops->shutdown(&substream, sink); 9b8ef9f6 Jeeja KP 2015-10-20 3756 } 4a75aae1 Charles Keepax 2018-09-05 3757 } c74184ed Mark Brown 2012-04-04 3758 break; c74184ed Mark Brown 2012-04-04 3759 c74184ed Mark Brown 2012-04-04 3760 default: a6ed0608 Takashi Iwai 2013-11-06 3761 WARN(1, "Unknown event %d\n", event); 75881df3 Sudip Mukherjee 2015-09-10 3762 ret = -EINVAL; c74184ed Mark Brown 2012-04-04 3763 } c74184ed Mark Brown 2012-04-04 3764 9747cec2 Mark Brown 2012-04-26 3765 out: 8053f216 Nicolin Chen 2016-07-26 3766 kfree(runtime); 9747cec2 Mark Brown 2012-04-26 3767 kfree(params); 9747cec2 Mark Brown 2012-04-26 @3768 return ret; c74184ed Mark Brown 2012-04-04 3769 } c74184ed Mark Brown 2012-04-04 3770 :::::: The code at line 3768 was first introduced by commit :::::: 9747cec21e2d55ee996249b01ec30dd32389c0ba ASoC: dapm: Move CODEC<->CODEC params off stack :::::: TO: Mark Brown <broonie(a)opensource.wolfsonmicro.com> :::::: CC: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- 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] ASoC: rsnd: Add device tree binding for r8a77990
by Yoshihiro Kaneko 06 Sep '18

06 Sep '18

From: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> This patch adds the device tree binding of the r8a77990 SoC. Signed-off-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Yoshihiro Kaneko <ykaneko0929(a)gmail.com> --- This patch is based on the devel branch of Simon Horman's renesas tree. Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index b86d790..07aba8d 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -352,6 +352,7 @@ Required properties: - "renesas,rcar_sound-r8a7794" (R-Car E2) - "renesas,rcar_sound-r8a7795" (R-Car H3) - "renesas,rcar_sound-r8a7796" (R-Car M3-W) + - "renesas,rcar_sound-r8a77990" (R-Car E3) - reg : Should contain the register physical address. required register is SRU/ADG/SSI if generation1 -- 1.9.1

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[alsa-devel] [PATCH] ASoC: sgtl5000: avoid division by zero if lo_vag is zero
by Colin King 06 Sep '18

06 Sep '18

From: Colin Ian King <colin.king(a)canonical.com> In the case where lo_vag <= SGTL5000_LINE_OUT_GND_BASE, lo_vag is set to zero and later vol_quot is computed by dividing by lo_vag causing a division by zero error. Fix this by avoiding a zero division and set vol_quot to zero in this specific case so that the lowest setting for i is correctly set. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 60764f6201b1..add18d6d77da 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1218,7 +1218,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_component *component) * Searching for a suitable index solving this formula: * idx = 40 * log10(vag_val / lo_cagcntrl) + 15 */ - vol_quot = (vag * 100) / lo_vag; + vol_quot = lo_vag ? (vag * 100) / lo_vag : 0; lo_vol = 0; for (i = 0; i < ARRAY_SIZE(vol_quot_table); i++) { if (vol_quot >= vol_quot_table[i]) -- 2.17.1

4 3

[alsa-devel] Applied "ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked" to the asoc tree
by Mark Brown 06 Sep '18

06 Sep '18

The patch ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked has been applied to the asoc tree at https://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 3bbf5d34fd4a0c41246290b70338095ae291851b Mon Sep 17 00:00:00 2001 From: Charles Keepax <ckeepax(a)opensource.cirrus.com> Date: Wed, 5 Sep 2018 15:20:58 +0100 Subject: [PATCH] ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked Currently DAPM has a lot of similar code to handle errors from snd_soc_dapm_new_control_unlocked, and much of this code does not really accurately reflect what the function returns. Firstly, most places will check for a return value of -EPROBE_DEFER and silence any error messages in that case. The one notable exception here being dapm_kcontrol_data_alloc which does currently print any error messages in the case of snd_soc_dapm_new_control_unlocked returning NULL or an error. Additionally the error prints being silenced in these case are redundant as snd_soc_dapm_new_control_unlocked can only return -EPROBE_DEFER or NULL when failing. Secondly, most places will treat a return value of NULL as an -ENOMEM. This is not correct either since any error except EPROBE_DEFER will cause a return value of NULL from snd_soc_dapm_new_control_unlocked. Centralise this handling and the error messages within snd_soc_dapm_new_control_unlocked and update the callers to simply check IS_ERR and return. Note that this update is slightly simpler in the case of dapm_kcontrol_data_alloc where that is fairly close to the handling that was already in place. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 118 +++++++-------------------------------- sound/soc/soc-topology.c | 11 ---- 2 files changed, 19 insertions(+), 110 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0a738cb439be..d13a25ce1275 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -364,10 +364,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, ret = PTR_ERR(data->widget); goto err_data; } - if (!data->widget) { - ret = -ENOMEM; - goto err_data; - } } break; case snd_soc_dapm_demux: @@ -402,10 +398,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, ret = PTR_ERR(data->widget); goto err_data; } - if (!data->widget) { - ret = -ENOMEM; - goto err_data; - } snd_soc_dapm_add_path(widget->dapm, data->widget, widget, NULL, NULL); @@ -3433,23 +3425,8 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); w = snd_soc_dapm_new_control_unlocked(dapm, widget); - /* Do not nag about probe deferrals */ - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s (%d)\n", - widget->name, ret); - goto out_unlock; - } - if (!w) - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s\n", - widget->name); - -out_unlock: mutex_unlock(&dapm->card->dapm_mutex); + return w; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); @@ -3464,24 +3441,20 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, int ret; if ((w = dapm_cnew_widget(widget)) == NULL) - return NULL; + return ERR_PTR(-ENOMEM); switch (w->id) { case snd_soc_dapm_regulator_supply: w->regulator = devm_regulator_get(dapm->dev, w->name); if (IS_ERR(w->regulator)) { ret = PTR_ERR(w->regulator); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) { ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) - dev_warn(w->dapm->dev, + dev_warn(dapm->dev, "ASoC: Failed to bypass %s: %d\n", w->name, ret); } @@ -3490,22 +3463,14 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->pinctrl = devm_pinctrl_get(dapm->dev); if (IS_ERR(w->pinctrl)) { ret = PTR_ERR(w->pinctrl); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } break; case snd_soc_dapm_clock_supply: w->clk = devm_clk_get(dapm->dev, w->name); if (IS_ERR(w->clk)) { ret = PTR_ERR(w->clk); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } break; default: @@ -3519,7 +3484,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->name = kstrdup_const(widget->name, GFP_KERNEL); if (w->name == NULL) { kfree(w); - return NULL; + return ERR_PTR(-ENOMEM); } switch (w->id) { @@ -3596,6 +3561,13 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, /* machine layer sets up unconnected pins and insertions */ w->connected = 1; return w; + +request_failed: + if (ret != -EPROBE_DEFER) + dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", + w->name, ret); + + return ERR_PTR(ret); } /** @@ -3621,19 +3593,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, w = snd_soc_dapm_new_control_unlocked(dapm, widget); if (IS_ERR(w)) { ret = PTR_ERR(w); - /* Do not nag about probe deferrals */ - if (ret == -EPROBE_DEFER) - break; - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s (%d)\n", - widget->name, ret); - break; - } - if (!w) { - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s\n", - widget->name); - ret = -ENOMEM; break; } widget++; @@ -3944,21 +3903,8 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); - if (IS_ERR(w)) { - ret = PTR_ERR(w); - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(card->dev, - "ASoC: Failed to create %s widget (%d)\n", - link_name, ret); - goto outfree_kcontrol_news; - } - if (!w) { - dev_err(card->dev, "ASoC: Failed to create %s widget\n", - link_name); - ret = -ENOMEM; + if (IS_ERR(w)) goto outfree_kcontrol_news; - } w->params = params; w->num_params = num_params; @@ -3999,21 +3945,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.name); w = snd_soc_dapm_new_control_unlocked(dapm, &template); - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create %s widget (%d)\n", - dai->driver->playback.stream_name, ret); - return ret; - } - if (!w) { - dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", - dai->driver->playback.stream_name); - return -ENOMEM; - } + if (IS_ERR(w)) + return PTR_ERR(w); w->priv = dai; dai->playback_widget = w; @@ -4028,21 +3961,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.name); w = snd_soc_dapm_new_control_unlocked(dapm, &template); - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create %s widget (%d)\n", - dai->driver->playback.stream_name, ret); - return ret; - } - if (!w) { - dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", - dai->driver->capture.stream_name); - return -ENOMEM; - } + if (IS_ERR(w)) + return PTR_ERR(w); w->priv = dai; dai->capture_widget = w; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 66e77e020745..17f81b9a5754 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1565,17 +1565,6 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, widget = snd_soc_dapm_new_control_unlocked(dapm, &template); if (IS_ERR(widget)) { ret = PTR_ERR(widget); - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(tplg->dev, - "ASoC: failed to create widget %s controls (%d)\n", - w->name, ret); - goto hdr_err; - } - if (widget == NULL) { - dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n", - w->name); - ret = -ENOMEM; goto hdr_err; } -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: dapm: Cosmetic tidy up of snd_soc_dapm_new_control" to the asoc tree
by Mark Brown 06 Sep '18

06 Sep '18

The patch ASoC: dapm: Cosmetic tidy up of snd_soc_dapm_new_control has been applied to the asoc tree at https://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 94e630a35d3383b42f12a873a5404bdf61e38e42 Mon Sep 17 00:00:00 2001 From: Charles Keepax <ckeepax(a)opensource.cirrus.com> Date: Wed, 5 Sep 2018 15:20:59 +0100 Subject: [PATCH] ASoC: dapm: Cosmetic tidy up of snd_soc_dapm_new_control Move the function snd_soc_dapm_new_control to be next to snd_soc_dapm_new_controls and add some kernel doc for it. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 37 +++++++++++++++++++++++-------------- 1 file changed, 23 insertions(+), 14 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d13a25ce1275..c111e69b9a09 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3417,20 +3417,6 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); -struct snd_soc_dapm_widget * -snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_widget *widget) -{ - struct snd_soc_dapm_widget *w; - - mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - w = snd_soc_dapm_new_control_unlocked(dapm, widget); - mutex_unlock(&dapm->card->dapm_mutex); - - return w; -} -EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); - struct snd_soc_dapm_widget * snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget) @@ -3570,6 +3556,29 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, return ERR_PTR(ret); } +/** + * snd_soc_dapm_new_control - create new dapm control + * @dapm: DAPM context + * @widget: widget template + * + * Creates new DAPM control based upon a template. + * + * Returns a widget pointer on success or an error pointer on failure + */ +struct snd_soc_dapm_widget * +snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_widget *widget) +{ + struct snd_soc_dapm_widget *w; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + w = snd_soc_dapm_new_control_unlocked(dapm, widget); + mutex_unlock(&dapm->card->dapm_mutex); + + return w; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); + /** * snd_soc_dapm_new_controls - create new dapm controls * @dapm: DAPM context -- 2.19.0.rc1

1 0

[alsa-devel] [PATCH v2 1/5] ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked
by Charles Keepax 06 Sep '18

06 Sep '18

Currently DAPM has a lot of similar code to handle errors from snd_soc_dapm_new_control_unlocked, and much of this code does not really accurately reflect what the function returns. Firstly, most places will check for a return value of -EPROBE_DEFER and silence any error messages in that case. The one notable exception here being dapm_kcontrol_data_alloc which does currently print any error messages in the case of snd_soc_dapm_new_control_unlocked returning NULL or an error. Additionally the error prints being silenced in these case are redundant as snd_soc_dapm_new_control_unlocked can only return -EPROBE_DEFER or NULL when failing. Secondly, most places will treat a return value of NULL as an -ENOMEM. This is not correct either since any error except EPROBE_DEFER will cause a return value of NULL from snd_soc_dapm_new_control_unlocked. Centralise this handling and the error messages within snd_soc_dapm_new_control_unlocked and update the callers to simply check IS_ERR and return. Note that this update is slightly simpler in the case of dapm_kcontrol_data_alloc where that is fairly close to the handling that was already in place. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Changes since v1: - Improved the commit message to be a more descriptive of the change Thanks, Charles sound/soc/soc-dapm.c | 118 ++++++++--------------------------------------- sound/soc/soc-topology.c | 11 ----- 2 files changed, 19 insertions(+), 110 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0a738cb439be..d13a25ce1275 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -364,10 +364,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, ret = PTR_ERR(data->widget); goto err_data; } - if (!data->widget) { - ret = -ENOMEM; - goto err_data; - } } break; case snd_soc_dapm_demux: @@ -402,10 +398,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, ret = PTR_ERR(data->widget); goto err_data; } - if (!data->widget) { - ret = -ENOMEM; - goto err_data; - } snd_soc_dapm_add_path(widget->dapm, data->widget, widget, NULL, NULL); @@ -3433,23 +3425,8 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); w = snd_soc_dapm_new_control_unlocked(dapm, widget); - /* Do not nag about probe deferrals */ - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s (%d)\n", - widget->name, ret); - goto out_unlock; - } - if (!w) - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s\n", - widget->name); - -out_unlock: mutex_unlock(&dapm->card->dapm_mutex); + return w; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); @@ -3464,24 +3441,20 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, int ret; if ((w = dapm_cnew_widget(widget)) == NULL) - return NULL; + return ERR_PTR(-ENOMEM); switch (w->id) { case snd_soc_dapm_regulator_supply: w->regulator = devm_regulator_get(dapm->dev, w->name); if (IS_ERR(w->regulator)) { ret = PTR_ERR(w->regulator); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) { ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) - dev_warn(w->dapm->dev, + dev_warn(dapm->dev, "ASoC: Failed to bypass %s: %d\n", w->name, ret); } @@ -3490,22 +3463,14 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->pinctrl = devm_pinctrl_get(dapm->dev); if (IS_ERR(w->pinctrl)) { ret = PTR_ERR(w->pinctrl); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } break; case snd_soc_dapm_clock_supply: w->clk = devm_clk_get(dapm->dev, w->name); if (IS_ERR(w->clk)) { ret = PTR_ERR(w->clk); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } break; default: @@ -3519,7 +3484,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->name = kstrdup_const(widget->name, GFP_KERNEL); if (w->name == NULL) { kfree(w); - return NULL; + return ERR_PTR(-ENOMEM); } switch (w->id) { @@ -3596,6 +3561,13 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, /* machine layer sets up unconnected pins and insertions */ w->connected = 1; return w; + +request_failed: + if (ret != -EPROBE_DEFER) + dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", + w->name, ret); + + return ERR_PTR(ret); } /** @@ -3621,19 +3593,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, w = snd_soc_dapm_new_control_unlocked(dapm, widget); if (IS_ERR(w)) { ret = PTR_ERR(w); - /* Do not nag about probe deferrals */ - if (ret == -EPROBE_DEFER) - break; - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s (%d)\n", - widget->name, ret); - break; - } - if (!w) { - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s\n", - widget->name); - ret = -ENOMEM; break; } widget++; @@ -3944,21 +3903,8 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); - if (IS_ERR(w)) { - ret = PTR_ERR(w); - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(card->dev, - "ASoC: Failed to create %s widget (%d)\n", - link_name, ret); - goto outfree_kcontrol_news; - } - if (!w) { - dev_err(card->dev, "ASoC: Failed to create %s widget\n", - link_name); - ret = -ENOMEM; + if (IS_ERR(w)) goto outfree_kcontrol_news; - } w->params = params; w->num_params = num_params; @@ -3999,21 +3945,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.name); w = snd_soc_dapm_new_control_unlocked(dapm, &template); - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create %s widget (%d)\n", - dai->driver->playback.stream_name, ret); - return ret; - } - if (!w) { - dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", - dai->driver->playback.stream_name); - return -ENOMEM; - } + if (IS_ERR(w)) + return PTR_ERR(w); w->priv = dai; dai->playback_widget = w; @@ -4028,21 +3961,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.name); w = snd_soc_dapm_new_control_unlocked(dapm, &template); - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create %s widget (%d)\n", - dai->driver->playback.stream_name, ret); - return ret; - } - if (!w) { - dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", - dai->driver->capture.stream_name); - return -ENOMEM; - } + if (IS_ERR(w)) + return PTR_ERR(w); w->priv = dai; dai->capture_widget = w; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 66e77e020745..17f81b9a5754 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1565,17 +1565,6 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, widget = snd_soc_dapm_new_control_unlocked(dapm, &template); if (IS_ERR(widget)) { ret = PTR_ERR(widget); - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(tplg->dev, - "ASoC: failed to create widget %s controls (%d)\n", - w->name, ret); - goto hdr_err; - } - if (widget == NULL) { - dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n", - w->name); - ret = -ENOMEM; goto hdr_err; } -- 2.11.0

2 5

[alsa-devel] Applied "ASoC: dapm: Add support for multi-CODEC CODEC to CODEC links" to the asoc tree
by Mark Brown 06 Sep '18

06 Sep '18

The patch ASoC: dapm: Add support for multi-CODEC CODEC to CODEC links has been applied to the asoc tree at https://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 4a75aae17b2a802a7267206414050408392c374c Mon Sep 17 00:00:00 2001 From: Charles Keepax <ckeepax(a)opensource.cirrus.com> Date: Wed, 5 Sep 2018 15:21:01 +0100 Subject: [PATCH] ASoC: dapm: Add support for multi-CODEC CODEC to CODEC links Currently multi-CODEC is not supported on CODEC to CODEC links. There are common applications where this would be useful, such as connecting two mono amplifiers to an audio CODEC. Adding support simply requires an update of snd_soc_dai_link_event to loop over the attached CODEC DAIs. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 123 ++++++++++++++++++++++++------------------- 1 file changed, 70 insertions(+), 53 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index bbfcb7fc05cc..40f27c95da61 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3614,7 +3614,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls); static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_dapm_path *source_p, *sink_p; + struct snd_soc_dapm_path *path; struct snd_soc_dai *source, *sink; struct snd_soc_pcm_runtime *rtd = w->priv; const struct snd_soc_pcm_stream *config = w->params + w->params_select; @@ -3629,17 +3629,6 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, list_empty(&w->edges[SND_SOC_DAPM_DIR_IN]))) return -EINVAL; - /* We only support a single source and sink, pick the first */ - source_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_OUT], - struct snd_soc_dapm_path, - list_node[SND_SOC_DAPM_DIR_OUT]); - sink_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_IN], - struct snd_soc_dapm_path, - list_node[SND_SOC_DAPM_DIR_IN]); - - source = source_p->source->priv; - sink = sink_p->sink->priv; - /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3681,59 +3670,90 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: substream.stream = SNDRV_PCM_STREAM_CAPTURE; - if (source->driver->ops->startup) { - ret = source->driver->ops->startup(&substream, source); - if (ret < 0) { - dev_err(source->dev, - "ASoC: startup() failed: %d\n", ret); - goto out; + snd_soc_dapm_widget_for_each_source_path(w, path) { + source = path->source->priv; + + if (source->driver->ops->startup) { + ret = source->driver->ops->startup(&substream, + source); + if (ret < 0) { + dev_err(source->dev, + "ASoC: startup() failed: %d\n", + ret); + goto out; + } + source->active++; } - source->active++; + ret = soc_dai_hw_params(&substream, params, source); + if (ret < 0) + goto out; } - ret = soc_dai_hw_params(&substream, params, source); - if (ret < 0) - goto out; substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - if (sink->driver->ops->startup) { - ret = sink->driver->ops->startup(&substream, sink); - if (ret < 0) { - dev_err(sink->dev, - "ASoC: startup() failed: %d\n", ret); - goto out; + snd_soc_dapm_widget_for_each_sink_path(w, path) { + sink = path->sink->priv; + + if (sink->driver->ops->startup) { + ret = sink->driver->ops->startup(&substream, + sink); + if (ret < 0) { + dev_err(sink->dev, + "ASoC: startup() failed: %d\n", + ret); + goto out; + } + sink->active++; } - sink->active++; + ret = soc_dai_hw_params(&substream, params, sink); + if (ret < 0) + goto out; } - ret = soc_dai_hw_params(&substream, params, sink); - if (ret < 0) - goto out; break; case SND_SOC_DAPM_POST_PMU: - ret = snd_soc_dai_digital_mute(sink, 0, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(sink->dev, "ASoC: Failed to unmute: %d\n", ret); - ret = 0; + snd_soc_dapm_widget_for_each_sink_path(w, path) { + sink = path->sink->priv; + + ret = snd_soc_dai_digital_mute(sink, 0, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(sink->dev, + "ASoC: Failed to unmute: %d\n", ret); + ret = 0; + } break; case SND_SOC_DAPM_PRE_PMD: - ret = snd_soc_dai_digital_mute(sink, 1, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(sink->dev, "ASoC: Failed to mute: %d\n", ret); - ret = 0; + snd_soc_dapm_widget_for_each_sink_path(w, path) { + sink = path->sink->priv; + + ret = snd_soc_dai_digital_mute(sink, 1, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(sink->dev, + "ASoC: Failed to mute: %d\n", ret); + ret = 0; + } + + snd_soc_dapm_widget_for_each_source_path(w, path) { + source = path->source->priv; - source->active--; - if (source->driver->ops->shutdown) { - substream.stream = SNDRV_PCM_STREAM_CAPTURE; - source->driver->ops->shutdown(&substream, source); + source->active--; + if (source->driver->ops->shutdown) { + substream.stream = SNDRV_PCM_STREAM_CAPTURE; + source->driver->ops->shutdown(&substream, + source); + } } - sink->active--; - if (sink->driver->ops->shutdown) { - substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - sink->driver->ops->shutdown(&substream, sink); + snd_soc_dapm_widget_for_each_sink_path(w, path) { + sink = path->sink->priv; + + sink->active--; + if (sink->driver->ops->shutdown) { + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + sink->driver->ops->shutdown(&substream, sink); + } } break; @@ -4043,9 +4063,6 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, int i; if (rtd->dai_link->params) { - if (rtd->num_codecs > 1) - dev_warn(card->dev, "ASoC: Multiple codecs not supported yet\n"); - playback_cpu = cpu_dai->capture_widget; capture_cpu = cpu_dai->playback_widget; } else { -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: dapm: Move CODEC to CODEC params from the widget to the runtime" to the asoc tree
by Mark Brown 06 Sep '18

06 Sep '18

The patch ASoC: dapm: Move CODEC to CODEC params from the widget to the runtime has been applied to the asoc tree at https://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 243bcfafcd9a23a20867fd488dc3a35264918d87 Mon Sep 17 00:00:00 2001 From: Charles Keepax <ckeepax(a)opensource.cirrus.com> Date: Wed, 5 Sep 2018 15:21:02 +0100 Subject: [PATCH] ASoC: dapm: Move CODEC to CODEC params from the widget to the runtime Larger CODECs may contain many several hundred widgets and which set of parameters is selected only needs to be recorded on a per DAI basis. As such move the selected CODEC to CODEC link params to be stored in the runtime rather than the DAPM widget, to save some memory. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-dapm.h | 3 --- include/sound/soc.h | 2 ++ sound/soc/soc-dapm.c | 19 +++++++++++-------- 3 files changed, 13 insertions(+), 11 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index cb177fa21ce7..bd8163f151cb 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -584,9 +584,6 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ struct pinctrl *pinctrl; /* attached pinctrl */ - const struct snd_soc_pcm_stream *params; /* params for dai links */ - unsigned int num_params; /* number of params for dai links */ - unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 6b68b31e3140..821bf99992b8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1130,6 +1130,8 @@ struct snd_soc_pcm_runtime { enum snd_soc_pcm_subclass pcm_subclass; struct snd_pcm_ops ops; + unsigned int params_select; /* currently selected param for dai link */ + /* Dynamic PCM BE runtime data */ struct snd_soc_dpcm_runtime dpcm[2]; int fe_compr; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 40f27c95da61..9f4edcd19e02 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1018,9 +1018,10 @@ static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) struct snd_kcontrol *kcontrol; struct snd_soc_dapm_context *dapm = w->dapm; struct snd_card *card = dapm->card->snd_card; + struct snd_soc_pcm_runtime *rtd = w->priv; /* create control for links with > 1 config */ - if (w->num_params <= 1) + if (rtd->dai_link->num_params <= 1) return 0; /* add kcontrol */ @@ -3617,13 +3618,15 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_path *path; struct snd_soc_dai *source, *sink; struct snd_soc_pcm_runtime *rtd = w->priv; - const struct snd_soc_pcm_stream *config = w->params + w->params_select; + const struct snd_soc_pcm_stream *config; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; struct snd_pcm_runtime *runtime = NULL; unsigned int fmt; int ret; + config = rtd->dai_link->params + rtd->params_select; + if (WARN_ON(!config) || WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) || list_empty(&w->edges[SND_SOC_DAPM_DIR_IN]))) @@ -3772,8 +3775,9 @@ static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_pcm_runtime *rtd = w->priv; - ucontrol->value.enumerated.item[0] = w->params_select; + ucontrol->value.enumerated.item[0] = rtd->params_select; return 0; } @@ -3782,18 +3786,19 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_pcm_runtime *rtd = w->priv; /* Can't change the config when widget is already powered */ if (w->power) return -EBUSY; - if (ucontrol->value.enumerated.item[0] == w->params_select) + if (ucontrol->value.enumerated.item[0] == rtd->params_select) return 0; - if (ucontrol->value.enumerated.item[0] >= w->num_params) + if (ucontrol->value.enumerated.item[0] >= rtd->dai_link->num_params) return -EINVAL; - w->params_select = ucontrol->value.enumerated.item[0]; + rtd->params_select = ucontrol->value.enumerated.item[0]; return 0; } @@ -3936,8 +3941,6 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, if (IS_ERR(w)) goto outfree_kcontrol_news; - w->params = rtd->dai_link->params; - w->num_params = rtd->dai_link->num_params; w->priv = rtd; return w; -- 2.19.0.rc1

1 0

[alsa-devel] [PATCH] ASoC: hdac_hdmi: remove redundant check for !port condition
by Colin King 06 Sep '18

06 Sep '18

From: Colin Ian King <colin.king(a)canonical.com> The !port check is redundant as it being performed in the following check. Remove it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/hdac_hdmi.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 7b8533abf637..dc6a0dfea050 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1961,9 +1961,6 @@ static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdev, int pcm_idx) port = list_first_entry(&pcm->port_list, struct hdac_hdmi_port, head); - if (!port) - return 0; - if (!port || !port->eld.eld_valid) return 0; -- 2.17.1

2 1

[alsa-devel] [PATCH 0/4] ASoC: rsnd: bugfix patches for linus/master, and v4.20
by Kuninori Morimoto 06 Sep '18

06 Sep '18

Hi Mark These are bug fix patches for Renesas Sound driver. 1) : for mark/for-4.20 2) - 4) : for linus/master Kuninori Morimoto (4): 1) ASoC: rsnd: gen: use tab instead of white-space 2) ASoC: rsnd: adg: care clock-frequency size 3) ASoC: rsnd: don't fallback to PIO mode when -EPROBE_DEFER 4) ASoC: rsnd: don't use %p for dev_dbg() sound/soc/sh/rcar/adg.c | 9 +++++++-- sound/soc/sh/rcar/core.c | 10 +++++++++- sound/soc/sh/rcar/dma.c | 6 +++++- sound/soc/sh/rcar/gen.c | 48 ++++++++++++++++++++++++------------------------ 4 files changed, 45 insertions(+), 28 deletions(-) -- 2.7.4 Best regards --- Kuninori Morimoto

2 8

[alsa-devel] [PATCH 1/1] ASoC: cs4265: fix MMTLR Data switch control
by Sébastien Szymanski 06 Sep '18

06 Sep '18

The MMTLR bit is in the CS4265_SPDIF_CTL2 register at address 0x12 bit 0 and not at address 0x0 bit 1. Fix this. Signed-off-by: Sébastien Szymanski <sebastien.szymanski(a)armadeus.com> --- sound/soc/codecs/cs4265.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 275677de669f..407554175282 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -157,8 +157,8 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, 3, 1, 0), SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), - SOC_SINGLE("MMTLR Data Switch", 0, - 1, 1, 0), + SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, + 0, 1, 0), SOC_ENUM("Mono Channel Select", spdif_mono_select_enum), SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24), }; -- 2.16.4

1 0

[alsa-devel] [PATCH 0/1] Fix ADI axi-i2s + Xilinx AXI-DMA capture
by Luca Ceresoli 06 Sep '18

06 Sep '18

Hi, here is a fix for a nasty audio capture problem when the axi-i2s output stream is fed to a Xilinx AXI-DMA. The commit is simple and hopefully well described, but I am not 100% sure the solution is the correct one. If it is not, I'll be glad to know which one is the best. It solved my problem anyway. Luca Luca Ceresoli (1): axi-i2s: set period size register sound/soc/adi/axi-i2s.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) -- 2.17.1

4 8

[alsa-devel] [RFC] UAPI: Check headers by compiling all together as C++
by David Howells 06 Sep '18

06 Sep '18

Here's a set of patches that inserts a step into the build process to make sure that the UAPI headers can all be built together with C++ (if the compiler being used supports C++). All but the final patch perform fixups, including: (1) Fix member names that conflict with C++ reserved words by providing alternates that can be used anywhere. An anonymous union is used so that that the conflicting name is still available outside of C++. (2) Fix the use of flexible arrays in structs that get embedded (which is illegal in C++). (3) Remove the use of internal kernel structs in UAPI structures. (4) Fix symbol collisions. (5) Replace usage of u32 and co. with __u32 and co. (6) Fix use of sparsely initialised arrays (which g++ doesn't implement). (7) Remove some use of PAGE_SIZE since this isn't valid outside of the kernel. And lastly: (8) Compile all of the UAPI headers (with a few exceptions) together as C++ to catch new errors occurring as part of the regular build process. The patches can also be found here: http://git.kernel.org/cgit/linux/kernel/git/dhowells/linux-fs.git/log/?h=ua… Thanks, David --- David Howells (11): UAPI: drm: Fix use of C++ keywords as structural members UAPI: keys: Fix use of C++ keywords as structural members UAPI: virtio_net: Fix use of C++ keywords as structural members UAPI: bcache: Fix use of embedded flexible array UAPI: coda: Don't use internal kernel structs in UAPI UAPI: netfilter: Fix symbol collision issues UAPI: nilfs2: Fix use of undefined byteswapping functions UAPI: sound: Fix use of u32 and co. in UAPI headers UAPI: ndctl: Fix g++-unsupported initialisation in headers UAPI: ndctl: Remove use of PAGE_SIZE UAPI: Check headers build for C++ Makefile | 1 include/linux/ndctl.h | 22 ++++ include/uapi/drm/i810_drm.h | 7 + include/uapi/drm/msm_drm.h | 7 + include/uapi/linux/bcache.h | 2 include/uapi/linux/coda_psdev.h | 4 + include/uapi/linux/keyctl.h | 7 + include/uapi/linux/ndctl.h | 20 ++- include/uapi/linux/netfilter/nfnetlink_cthelper.h | 2 include/uapi/linux/netfilter_ipv4/ipt_ECN.h | 9 -- include/uapi/linux/nilfs2_ondisk.h | 21 ++-- include/uapi/linux/virtio_net.h | 7 + include/uapi/sound/skl-tplg-interface.h | 106 +++++++++--------- scripts/headers-c++.sh | 124 +++++++++++++++++++++ 14 files changed, 255 insertions(+), 84 deletions(-) create mode 100644 include/linux/ndctl.h create mode 100755 scripts/headers-c++.sh

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[alsa-devel] [PATCH] ASoC: uniphier: change status to orphan
by Katsuhiro Suzuki 05 Sep '18

05 Sep '18

Since I'm leaving from Socionext, I'll unable to access specification documents of this hardware (these are not public). So change the state to orphan until someone will maintain this driver. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- MAINTAINERS | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index a8683e549777..4788a5335d35 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -13450,9 +13450,8 @@ F: drivers/i2c/busses/i2c-synquacer.c F: Documentation/devicetree/bindings/i2c/i2c-synquacer.txt SOCIONEXT UNIPHIER SOUND DRIVER -M: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -S: Maintained +S: Orphan F: sound/soc/uniphier/ SOEKRIS NET48XX LED SUPPORT -- 2.18.0

2 2

[alsa-devel] [PATCH v2 1/3] ASoC: tlv320aic31xx: Add overflow detection support
by Andrew F. Davis 04 Sep '18

04 Sep '18

Similar to short circuit detection, when the ADC/DAC is saturated and overflows poor audio quality can result and should be reported to the user. This device support Automatic Dynamic Range Compression (DRC) to reduce this but it is not enabled currently in this driver. Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- Changes from v1: - Change errors to warning for non-fatal events - Rebased on broonie/sound.git for-next sound/soc/codecs/tlv320aic31xx.c | 34 ++++++++++++++++++++++++++++++-- sound/soc/codecs/tlv320aic31xx.h | 7 +++++++ 2 files changed, 39 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index d8783af7d31c..6e8f79ec9629 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1356,7 +1356,7 @@ static irqreturn_t aic31xx_irq(int irq, void *data) if (value) handled = true; else - goto exit; + goto read_overflow; if (value & AIC31XX_HPLSCDETECT) dev_err(dev, "Short circuit on Left output is detected\n"); @@ -1366,6 +1366,35 @@ static irqreturn_t aic31xx_irq(int irq, void *data) AIC31XX_HPRSCDETECT)) dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); +read_overflow: + ret = regmap_read(aic31xx->regmap, AIC31XX_OFFLAG, &value); + if (ret) { + dev_err(dev, "Failed to read overflow flag: %d\n", ret); + goto exit; + } + + if (value) + handled = true; + else + goto exit; + + if (value & AIC31XX_DAC_OF_LEFT) + dev_warn(dev, "Left-channel DAC overflow has occurred\n"); + if (value & AIC31XX_DAC_OF_RIGHT) + dev_warn(dev, "Right-channel DAC overflow has occurred\n"); + if (value & AIC31XX_DAC_OF_SHIFTER) + dev_warn(dev, "DAC barrel shifter overflow has occurred\n"); + if (value & AIC31XX_ADC_OF) + dev_warn(dev, "ADC overflow has occurred\n"); + if (value & AIC31XX_ADC_OF_SHIFTER) + dev_warn(dev, "ADC barrel shifter overflow has occurred\n"); + if (value & ~(AIC31XX_DAC_OF_LEFT | + AIC31XX_DAC_OF_RIGHT | + AIC31XX_DAC_OF_SHIFTER | + AIC31XX_ADC_OF | + AIC31XX_ADC_OF_SHIFTER)) + dev_warn(dev, "Unknown overflow interrupt flags: 0x%08x\n", value); + exit: if (handled) return IRQ_HANDLED; @@ -1447,7 +1476,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, AIC31XX_GPIO1_FUNC_SHIFT); regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, - AIC31XX_SC); + AIC31XX_SC | + AIC31XX_ENGINE); ret = devm_request_threaded_irq(aic31xx->dev, aic31xx->irq, NULL, aic31xx_irq, diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 52e171988906..2636f2c6bc79 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -173,6 +173,13 @@ struct aic31xx_pdata { #define AIC31XX_HPRDRVPWRSTATUS_MASK BIT(1) #define AIC31XX_SPRDRVPWRSTATUS_MASK BIT(0) +/* AIC31XX_OFFLAG */ +#define AIC31XX_DAC_OF_LEFT BIT(7) +#define AIC31XX_DAC_OF_RIGHT BIT(6) +#define AIC31XX_DAC_OF_SHIFTER BIT(5) +#define AIC31XX_ADC_OF BIT(3) +#define AIC31XX_ADC_OF_SHIFTER BIT(1) + /* AIC31XX_INTRDACFLAG */ #define AIC31XX_HPLSCDETECT BIT(7) #define AIC31XX_HPRSCDETECT BIT(6) -- 2.18.0

2 4

[alsa-devel] [PATCH 0/2][RFC] ASoC: add for_each_xxx_codec macro
by Kuninori Morimoto 04 Sep '18

04 Sep '18

Hi Mark These are not so important patches. But current code is using for loop for dai_link codecs and rtd codec_dai, but I think using for_each_xxx macro for these can be more readable. These patches do it. Kuninori Morimoto (2): ASoC: add for_each_link_codecs() macro ASoC: add for_each_rtd_codec_dai() macro include/sound/soc.h | 11 ++ sound/soc/intel/boards/kbl_rt5663_max98927.c | 5 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 5 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 5 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 5 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 5 +- sound/soc/meson/axg-card.c | 11 +- sound/soc/soc-core.c | 55 ++++---- sound/soc/soc-dapm.c | 14 +- sound/soc/soc-pcm.c | 154 ++++++++++----------- 10 files changed, 133 insertions(+), 137 deletions(-) -- 2.7.4

2 6

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Add MICBIAS off setting
by Andrew F. Davis 04 Sep '18

04 Sep '18

Leaving microphone bias off is a valid setting and even used in the DT binding document example. Add this setting here and document the same. Although it may not make much sense to enable a microphone here without any bias, it is a valid setting that can be chosen by DT and may be needed for some boards. Signed-off-by: Andrew F. Davis <afd(a)ti.com> Acked-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/sound/tlv320aic31xx.txt | 1 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 1 + sound/soc/codecs/tlv320aic31xx.c | 1 + 3 files changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt index 5b3c33bb99e5..411cc46a2c58 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt @@ -24,6 +24,7 @@ Optional properties: - reset-gpios - GPIO specification for the active low RESET input. - ai31xx-micbias-vg - MicBias Voltage setting + 0 or MICBIAS_OFF - MICBIAS output is powered off 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD diff --git a/include/dt-bindings/sound/tlv320aic31xx-micbias.h b/include/dt-bindings/sound/tlv320aic31xx-micbias.h index c6895a18a455..069484070fcf 100644 --- a/include/dt-bindings/sound/tlv320aic31xx-micbias.h +++ b/include/dt-bindings/sound/tlv320aic31xx-micbias.h @@ -2,6 +2,7 @@ #ifndef __DT_TLV320AIC31XX_MICBIAS_H #define __DT_TLV320AIC31XX_MICBIAS_H +#define MICBIAS_OFF 0 #define MICBIAS_2_0V 1 #define MICBIAS_2_5V 2 #define MICBIAS_AVDDV 3 diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index bf92d36b8f8a..7d87df518fed 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1421,6 +1421,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, fwnode_property_read_u32(aic31xx->dev->fwnode, "ai31xx-micbias-vg", &micbias_value); switch (micbias_value) { + case MICBIAS_OFF: case MICBIAS_2_0V: case MICBIAS_2_5V: case MICBIAS_AVDDV: -- 2.18.0

2 9

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Remove regulator notification handling
by Andrew F. Davis 04 Sep '18

04 Sep '18

The notification handler sets the reset line but never re-enables it as no one is listening for an enable event, this is certainly broken and was most likely just copied from other CODECs, remove this code. Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.c | 55 +------------------------------- 1 file changed, 1 insertion(+), 54 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 7d87df518fed..f8d0f3ebed9c 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -147,11 +147,6 @@ static const char * const aic31xx_supply_names[] = { #define AIC31XX_NUM_SUPPLIES ARRAY_SIZE(aic31xx_supply_names) -struct aic31xx_disable_nb { - struct notifier_block nb; - struct aic31xx_priv *aic31xx; -}; - struct aic31xx_priv { struct snd_soc_component *component; u8 i2c_regs_status; @@ -162,7 +157,6 @@ struct aic31xx_priv { int micbias_vg; struct aic31xx_pdata pdata; struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; - struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES]; unsigned int sysclk; u8 p_div; int rate_div_line; @@ -1116,28 +1110,6 @@ static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } -static int aic31xx_regulator_event(struct notifier_block *nb, - unsigned long event, void *data) -{ - struct aic31xx_disable_nb *disable_nb = - container_of(nb, struct aic31xx_disable_nb, nb); - struct aic31xx_priv *aic31xx = disable_nb->aic31xx; - - if (event & REGULATOR_EVENT_DISABLE) { - /* - * Put codec to reset and as at least one of the - * supplies was disabled. - */ - if (aic31xx->gpio_reset) - gpiod_set_value(aic31xx->gpio_reset, 1); - - regcache_mark_dirty(aic31xx->regmap); - dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__); - } - - return 0; -} - static int aic31xx_reset(struct aic31xx_priv *aic31xx) { int ret = 0; @@ -1263,26 +1235,12 @@ static int aic31xx_set_bias_level(struct snd_soc_component *component, static int aic31xx_codec_probe(struct snd_soc_component *component) { struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); - int i, ret; + int ret; dev_dbg(aic31xx->dev, "## %s\n", __func__); aic31xx->component = component; - for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) { - aic31xx->disable_nb[i].nb.notifier_call = - aic31xx_regulator_event; - aic31xx->disable_nb[i].aic31xx = aic31xx; - ret = regulator_register_notifier(aic31xx->supplies[i].consumer, - &aic31xx->disable_nb[i].nb); - if (ret) { - dev_err(component->dev, - "Failed to request regulator notifier: %d\n", - ret); - return ret; - } - } - regcache_cache_only(aic31xx->regmap, true); regcache_mark_dirty(aic31xx->regmap); @@ -1297,19 +1255,8 @@ static int aic31xx_codec_probe(struct snd_soc_component *component) return 0; } -static void aic31xx_codec_remove(struct snd_soc_component *component) -{ - struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); - int i; - - for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) - regulator_unregister_notifier(aic31xx->supplies[i].consumer, - &aic31xx->disable_nb[i].nb); -} - static const struct snd_soc_component_driver soc_codec_driver_aic31xx = { .probe = aic31xx_codec_probe, - .remove = aic31xx_codec_remove, .set_bias_level = aic31xx_set_bias_level, .controls = common31xx_snd_controls, .num_controls = ARRAY_SIZE(common31xx_snd_controls), -- 2.18.0

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[alsa-devel] [PATCH 1/4] ASoC: tlv320aic31xx: Add short circuit detection support
by Andrew F. Davis 04 Sep '18

04 Sep '18

These devices support detecting and reporting short circuits across the output stages. Add support for reporting these issue. Do this by registering an interrupt if available and enabling this error to trigger that interrupt in the device. Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.c | 55 ++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 16 ++++++++++ 2 files changed, 71 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index f8d0f3ebed9c..d8783af7d31c 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -161,6 +161,7 @@ struct aic31xx_priv { u8 p_div; int rate_div_line; bool master_dapm_route_applied; + int irq; }; struct aic31xx_rate_divs { @@ -1338,6 +1339,40 @@ static const struct acpi_device_id aic31xx_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match); #endif +static irqreturn_t aic31xx_irq(int irq, void *data) +{ + struct aic31xx_priv *aic31xx = data; + struct device *dev = aic31xx->dev; + unsigned int value; + bool handled = false; + int ret; + + ret = regmap_read(aic31xx->regmap, AIC31XX_INTRDACFLAG, &value); + if (ret) { + dev_err(dev, "Failed to read interrupt mask: %d\n", ret); + goto exit; + } + + if (value) + handled = true; + else + goto exit; + + if (value & AIC31XX_HPLSCDETECT) + dev_err(dev, "Short circuit on Left output is detected\n"); + if (value & AIC31XX_HPRSCDETECT) + dev_err(dev, "Short circuit on Right output is detected\n"); + if (value & ~(AIC31XX_HPLSCDETECT | + AIC31XX_HPRSCDETECT)) + dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); + +exit: + if (handled) + return IRQ_HANDLED; + else + return IRQ_NONE; +} + static int aic31xx_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1360,6 +1395,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return ret; } aic31xx->dev = &i2c->dev; + aic31xx->irq = i2c->irq; aic31xx->codec_type = id->driver_data; @@ -1404,6 +1440,25 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return ret; } + if (aic31xx->irq > 0) { + regmap_update_bits(aic31xx->regmap, AIC31XX_GPIO1, + AIC31XX_GPIO1_FUNC_MASK, + AIC31XX_GPIO1_INT1 << + AIC31XX_GPIO1_FUNC_SHIFT); + + regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, + AIC31XX_SC); + + ret = devm_request_threaded_irq(aic31xx->dev, aic31xx->irq, + NULL, aic31xx_irq, + IRQF_ONESHOT, "aic31xx-irq", + aic31xx); + if (ret) { + dev_err(aic31xx->dev, "Unable to request IRQ\n"); + return ret; + } + } + if (aic31xx->codec_type & DAC31XX_BIT) return devm_snd_soc_register_component(&i2c->dev, &soc_codec_driver_aic31xx, diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 0b587585b38b..52e171988906 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -191,6 +191,22 @@ struct aic31xx_pdata { #define AIC31XX_SC BIT(3) #define AIC31XX_ENGINE BIT(2) +/* AIC31XX_GPIO1 */ +#define AIC31XX_GPIO1_FUNC_MASK GENMASK(5, 2) +#define AIC31XX_GPIO1_FUNC_SHIFT 2 +#define AIC31XX_GPIO1_DISABLED 0x00 +#define AIC31XX_GPIO1_INPUT 0x01 +#define AIC31XX_GPIO1_GPI 0x02 +#define AIC31XX_GPIO1_GPO 0x03 +#define AIC31XX_GPIO1_CLKOUT 0x04 +#define AIC31XX_GPIO1_INT1 0x05 +#define AIC31XX_GPIO1_INT2 0x06 +#define AIC31XX_GPIO1_ADC_WCLK 0x07 +#define AIC31XX_GPIO1_SBCLK 0x08 +#define AIC31XX_GPIO1_SWCLK 0x09 +#define AIC31XX_GPIO1_ADC_MOD_CLK 0x10 +#define AIC31XX_GPIO1_SDOUT 0x11 + /* AIC31XX_DACSETUP */ #define AIC31XX_SOFTSTEP_MASK GENMASK(1, 0) -- 2.18.0

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[alsa-devel] [PATCH 0/5] Set SNDRV_PCM_INFO_SYNC_APPLPTR to indirect PCM usages
by Takashi Iwai 04 Sep '18

04 Sep '18

Hi, this is a patch set to pass the missing SNDRV_PCM_INFO_SYNC_APPLPTR flag to the drivers using the indirect PCM helpers. Also, one cleanup of the indirect PCM helper usage in mips/hal2 driver is included. Takashi === Takashi Iwai (5): ALSA: rme32: Use SNDRV_PCM_INFO_SYNC_APPLPTR info flag ALSA: mips: Use SNDRV_PCM_INFO_SYNC_APPLPTR info flag ALSA: emu10k1: Use SNDRV_PCM_INFO_SYNC_APPLPTR info flag ALSA: cs46xx: Use SNDRV_PCM_INFO_SYNC_APPLPTR info flag ALSA: mips: Cleanup indirect PCM helper usages sound/mips/hal2.c | 13 +++++-------- sound/pci/cs46xx/cs46xx_lib.c | 6 ++++-- sound/pci/emu10k1/emupcm.c | 3 ++- sound/pci/rme32.c | 22 ++++++++-------------- 4 files changed, 19 insertions(+), 25 deletions(-) -- 2.18.0

2 6

[alsa-devel] Possibility of routing at the kernel level
by Mark Farrugia 04 Sep '18

04 Sep '18

Hi all, I am currently using the ALSA loopback driver to create a "virtual" sound card to abstract underlying real sound cards. I capture from the loopback in user-space, and then subsequently playback to a real sound card. The issue with this is that there is a significant latency introduced (~100ms) due to the number of transitions of the stream from user to kernel space. I am seeking a way to avoid this. My question is this: is there any way of routing PCM substreams at the kernel level, so that we can avoid the need to pipe audio back up to user-space with the loopback capturer? If there is no current solution that would achieve this, I would proposition to design and implement a system that uses a top-level ALSA driver, which then routes dynamically to one or many subscribed lower-level "audio path" drivers. The system would also have the option to loopback upon itself. Does anyone see any reason that I couldn't (or shouldn't) attempt such a solution? All feedback is welcome. Regards, Mark

2 3

[alsa-devel] [PATCH 00/13] ASoC: add full BUSIF support
by Kuninori Morimoto 04 Sep '18

04 Sep '18

Hi Mark Cc: Jiada, Timo, Dragos Dragos, Jiada, Timo added full BUSIF support, expand TDM mode, TDM split mode and its related features to Renesas sound driver on v4.14 base kernel. Thank you Dragos, Jiada, Timo. Unfortunately, I can't confirm expand TDM mode, TDM split mode on my environment. But I could confirm/test full BUSIF support. These patches are for it. Dragos Tarcatu (3): ASoC: rsnd: add support for 16/24 bit slot widths ASoC: rsnd: add support for the DSP_A/DSP_B formats ASoC: rsnd: add support for 8 bit S8 format Jiada Wang (9): ASoC: rsnd: add warning message to rsnd_kctrl_accept_runtime() ASoC: rsnd: remove is_play parameter from hw_rule function ASoC: rsnd: ssi: Fix issue in dma data address assignment ASoC: rsnd: ssi: Check runtime channel number rather than hw_params ASoc: rsnd: dma: Calculate dma address with consider of BUSIF ASoc: rsnd: dma: Calculate PDMACHCRE with consider of BUSIF ASoC: rsnd: ssiu: Support BUSIF other than BUSIF0 ASoC: rsnd: ssiu: Support to init different BUSIF instance ASoC: rsnd: fixup not to call clk_get/set under non-atomic Kuninori Morimoto (1): ASoC: rsnd: merge .nolock_start and .prepare sound/soc/sh/rcar/core.c | 135 +++++++++++++++++++++++++---------------------- sound/soc/sh/rcar/dma.c | 100 ++++++++++++++++++++++------------- sound/soc/sh/rcar/gen.c | 27 ++++++++-- sound/soc/sh/rcar/rsnd.h | 64 ++++++++++++++++------ sound/soc/sh/rcar/ssi.c | 105 ++++++++++++++++++++++-------------- sound/soc/sh/rcar/ssiu.c | 92 +++++++++++++++++++++++++++----- 6 files changed, 349 insertions(+), 174 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ALSA: HDA: Fix several mismatch for register mask and value
by Keyon Jie 04 Sep '18

04 Sep '18

E.g. for snd_hdac_ext_link_clear_stream_id(), we should set (1 << stream) as mask, and 0 as value, here correct it and several similar mismatches. And, here also remove unreadable register_mask usage for those mask value updating. Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> --- sound/hda/ext/hdac_ext_stream.c | 22 +++++++--------------- 1 file changed, 7 insertions(+), 15 deletions(-) diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index 1bd27576db98..a835558ddbc9 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -146,7 +146,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_decouple); */ void snd_hdac_ext_link_stream_start(struct hdac_ext_stream *stream) { - snd_hdac_updatel(stream->pplc_addr, AZX_REG_PPLCCTL, 0, AZX_PPLCCTL_RUN); + snd_hdac_updatel(stream->pplc_addr, AZX_REG_PPLCCTL, + AZX_PPLCCTL_RUN, AZX_PPLCCTL_RUN); } EXPORT_SYMBOL_GPL(snd_hdac_ext_link_stream_start); @@ -171,7 +172,8 @@ void snd_hdac_ext_link_stream_reset(struct hdac_ext_stream *stream) snd_hdac_ext_link_stream_clear(stream); - snd_hdac_updatel(stream->pplc_addr, AZX_REG_PPLCCTL, 0, AZX_PPLCCTL_STRST); + snd_hdac_updatel(stream->pplc_addr, AZX_REG_PPLCCTL, + AZX_PPLCCTL_STRST, AZX_PPLCCTL_STRST); udelay(3); timeout = 50; do { @@ -242,7 +244,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_link_set_stream_id); void snd_hdac_ext_link_clear_stream_id(struct hdac_ext_link *link, int stream) { - snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, 0, (1 << stream)); + snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, (1 << stream), 0); } EXPORT_SYMBOL_GPL(snd_hdac_ext_link_clear_stream_id); @@ -415,7 +417,6 @@ void snd_hdac_ext_stream_spbcap_enable(struct hdac_bus *bus, bool enable, int index) { u32 mask = 0; - u32 register_mask = 0; if (!bus->spbcap) { dev_err(bus->dev, "Address of SPB capability is NULL\n"); @@ -424,12 +425,8 @@ void snd_hdac_ext_stream_spbcap_enable(struct hdac_bus *bus, mask |= (1 << index); - register_mask = readl(bus->spbcap + AZX_REG_SPB_SPBFCCTL); - - mask |= register_mask; - if (enable) - snd_hdac_updatel(bus->spbcap, AZX_REG_SPB_SPBFCCTL, 0, mask); + snd_hdac_updatel(bus->spbcap, AZX_REG_SPB_SPBFCCTL, mask, mask); else snd_hdac_updatel(bus->spbcap, AZX_REG_SPB_SPBFCCTL, mask, 0); } @@ -503,7 +500,6 @@ void snd_hdac_ext_stream_drsm_enable(struct hdac_bus *bus, bool enable, int index) { u32 mask = 0; - u32 register_mask = 0; if (!bus->drsmcap) { dev_err(bus->dev, "Address of DRSM capability is NULL\n"); @@ -512,12 +508,8 @@ void snd_hdac_ext_stream_drsm_enable(struct hdac_bus *bus, mask |= (1 << index); - register_mask = readl(bus->drsmcap + AZX_REG_SPB_SPBFCCTL); - - mask |= register_mask; - if (enable) - snd_hdac_updatel(bus->drsmcap, AZX_REG_DRSM_CTL, 0, mask); + snd_hdac_updatel(bus->drsmcap, AZX_REG_DRSM_CTL, mask, mask); else snd_hdac_updatel(bus->drsmcap, AZX_REG_DRSM_CTL, mask, 0); } -- 2.14.1

2 1

[alsa-devel] [PATCH v2] sound: Convert to using %pOFn instead of device_node.name
by Rob Herring 04 Sep '18

04 Sep '18

In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- v2: - Fix error in strncmp sound/aoa/soundbus/i2sbus/core.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 000b58522106..bd7c5029fc59 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -157,18 +157,19 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct device_node *child = NULL, *sound = NULL; struct resource *r; int i, layout = 0, rlen, ok = force; - static const char *rnames[] = { "i2sbus: %s (control)", - "i2sbus: %s (tx)", - "i2sbus: %s (rx)" }; + char node_name[6]; + static const char *rnames[] = { "i2sbus: %pOFn (control)", + "i2sbus: %pOFn (tx)", + "i2sbus: %pOFn (rx)" }; static irq_handler_t ints[] = { i2sbus_bus_intr, i2sbus_tx_intr, i2sbus_rx_intr }; - if (strlen(np->name) != 5) + if (snprintf(node_name, sizeof(node_name), "%pOFn", np) != 5) return 0; - if (strncmp(np->name, "i2s-", 4)) + if (strncmp(node_name, "i2s-", 4)) return 0; dev = kzalloc(sizeof(struct i2sbus_dev), GFP_KERNEL); @@ -228,13 +229,13 @@ static int i2sbus_add_dev(struct macio_dev *macio, dev->sound.pcmid = -1; dev->macio = macio; dev->control = control; - dev->bus_number = np->name[4] - 'a'; + dev->bus_number = node_name[4] - 'a'; INIT_LIST_HEAD(&dev->sound.codec_list); for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) { dev->interrupts[i] = -1; snprintf(dev->rnames[i], sizeof(dev->rnames[i]), - rnames[i], np->name); + rnames[i], np); } for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) { int irq = irq_of_parse_and_map(np, i); -- 2.17.1

2 1

[alsa-devel] KMSAN: uninit-value in snd_midi_event_encode_byte
by syzbot 04 Sep '18

04 Sep '18

Hello, syzbot found the following crash on: HEAD commit: 2dca2cbde67a kmsan: fix build warnings with CONFIG_KMSAN=n git tree: https://github.com/google/kmsan.git/master console output: https://syzkaller.appspot.com/x/log.txt?x=1660ef6a400000 kernel config: https://syzkaller.appspot.com/x/.config?x=820d6393634b55e3 dashboard link: https://syzkaller.appspot.com/bug?extid=194dffdb8b22fc5d207a compiler: clang version 8.0.0 (trunk 339414) Unfortunately, I don't have any reproducer for this crash yet. IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+194dffdb8b22fc5d207a(a)syzkaller.appspotmail.com ================================================================== BUG: KMSAN: uninit-value in snd_midi_event_encode_byte+0x569/0xff0 sound/core/seq/seq_midi_event.c:195 CPU: 0 PID: 1623 Comm: kworker/0:1H Not tainted 4.19.0-rc1+ #36 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: events_highpri snd_vmidi_output_work Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x14b/0x190 lib/dump_stack.c:113 kmsan_report+0x183/0x2b0 mm/kmsan/kmsan.c:956 __msan_warning+0x70/0xc0 mm/kmsan/kmsan_instr.c:645 snd_midi_event_encode_byte+0x569/0xff0 sound/core/seq/seq_midi_event.c:195 snd_vmidi_output_work+0x34e/0x5b0 sound/core/seq/seq_virmidi.c:161 process_one_work+0x1605/0x1f40 kernel/workqueue.c:2153 worker_thread+0x11a2/0x2590 kernel/workqueue.c:2296 kthread+0x465/0x4a0 kernel/kthread.c:247 ret_from_fork+0x35/0x40 arch/x86/entry/entry_64.S:416 Uninit was stored to memory at: kmsan_save_stack_with_flags mm/kmsan/kmsan.c:256 [inline] kmsan_save_stack mm/kmsan/kmsan.c:271 [inline] kmsan_internal_chain_origin+0x128/0x210 mm/kmsan/kmsan.c:573 __msan_chain_origin+0x69/0xc0 mm/kmsan/kmsan_instr.c:482 __snd_rawmidi_transmit_peek sound/core/rawmidi.c:1103 [inline] snd_rawmidi_transmit+0xa75/0xbf0 sound/core/rawmidi.c:1228 snd_vmidi_output_work+0x2ac/0x5b0 sound/core/seq/seq_virmidi.c:159 process_one_work+0x1605/0x1f40 kernel/workqueue.c:2153 worker_thread+0x11a2/0x2590 kernel/workqueue.c:2296 kthread+0x465/0x4a0 kernel/kthread.c:247 ret_from_fork+0x35/0x40 arch/x86/entry/entry_64.S:416 Uninit was created at: kmsan_save_stack_with_flags mm/kmsan/kmsan.c:256 [inline] kmsan_internal_poison_shadow+0xb8/0x1b0 mm/kmsan/kmsan.c:181 kmsan_kmalloc+0x98/0x100 mm/kmsan/kmsan_hooks.c:91 __kmalloc_node+0x7bf/0x11c0 mm/slub.c:3828 kmalloc_node include/linux/slab.h:555 [inline] kvmalloc_node+0x19d/0x3e0 mm/util.c:423 kvmalloc include/linux/mm.h:577 [inline] snd_rawmidi_runtime_create sound/core/rawmidi.c:132 [inline] open_substream+0x3c8/0xaa0 sound/core/rawmidi.c:276 rawmidi_open_priv+0x347/0x1000 sound/core/rawmidi.c:327 snd_rawmidi_open+0x7d4/0x1120 sound/core/rawmidi.c:424 soundcore_open+0x9be/0xa60 sound/sound_core.c:597 chrdev_open+0xc26/0xdb0 fs/char_dev.c:417 do_dentry_open+0xc9d/0x1660 fs/open.c:771 vfs_open+0xaf/0xe0 fs/open.c:880 do_last fs/namei.c:3418 [inline] path_openat+0x17a6/0x67c0 fs/namei.c:3534 do_filp_open+0x259/0x610 fs/namei.c:3564 do_sys_open+0x630/0x940 fs/open.c:1063 __do_sys_open fs/open.c:1081 [inline] __se_sys_open fs/open.c:1076 [inline] __x64_sys_open+0xf2/0x130 fs/open.c:1076 do_syscall_64+0x15b/0x220 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x63/0xe7 ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#bug-status-tracking for how to communicate with syzbot.

3 16

[alsa-devel] Microphone detected, but no output for all ASUS G751xx with ALC668 chipset
by Håvard 03 Sep '18

03 Sep '18

Have tried enabling all GPIO settings as well as tried every possible pin combination I could think of. This problem has persisted for almost 4-years with no fix or patch to speak of. Would be great if someone finally had a look at it! -Håvard

1 0

[alsa-devel] [PATCH] ASoC: core: Don't schedule DAPM work if already in target state
by Jon Hunter 03 Sep '18

03 Sep '18

When dapm_power_widgets() is called, the dapm_pre_sequence_async() and dapm_post_sequence_async() functions are scheduled for all DAPM contexts (apart from the card DAPM context) regardless of whether the DAPM context is already in the desired state. The overhead of this is not insignificant and the more DAPM contexts there are the more overhead there is. For example, on the Tegra124 Jetson TK1, when profiling the time taken to execute the dapm_power_widgets() the following times were observed. Times for function dapm_power_widgets() are (us): Min 23, Ave 190, Max 434, Count 39 Here 'Count' is the number of times that dapm_power_widgets() has been called. Please note that the above time were measured using ktime_get() to log the time on entry and exit from dapm_power_widgets(). So it should be noted that these times may not be purely the time take to execute this function if it is preempted. However, after applying this patch and measuring the time taken to execute dapm_power_widgets() again a significant improvement is seen as shown below. Times for function dapm_power_widgets() are (us): Min 4, Ave 16, Max 82, Count 39 Therefore, optimise the dapm_power_widgets() function by only scheduling the dapm_pre/post_sequence_async() work if the DAPM context is not in the desired state. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.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 461d951917c0..2e9231aeabb9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1953,7 +1953,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) dapm_pre_sequence_async(&card->dapm, 0); /* Run other bias changes in parallel */ list_for_each_entry(d, &card->dapm_list, list) { - if (d != &card->dapm) + if (d != &card->dapm && d->bias_level != d->target_bias_level) async_schedule_domain(dapm_pre_sequence_async, d, &async_domain); } @@ -1977,7 +1977,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) /* Run all the bias changes in parallel */ list_for_each_entry(d, &card->dapm_list, list) { - if (d != &card->dapm) + if (d != &card->dapm && d->bias_level != d->target_bias_level) async_schedule_domain(dapm_post_sequence_async, d, &async_domain); } -- 2.7.4

3 3

[alsa-devel] [PATCH] ASoC: q6routing: initialize data correctly
by Srinivas Kandagatla 03 Sep '18

03 Sep '18

Some of the router data fields are left as default zeros which are valid dai ids, so initialize these to invalid value of -1. Without intializing these correctly get_session_from_id() can return incorrect session resulting in not closing the opened copp and messing up with the copp ref count. Fixes: e3a33673e845 ("ASoC: qdsp6: q6routing: Add q6routing driver") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6routing.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index dc94c5c53788..c6b51571be94 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -960,8 +960,10 @@ static int msm_routing_probe(struct snd_soc_component *c) { int i; - for (i = 0; i < MAX_SESSIONS; i++) + for (i = 0; i < MAX_SESSIONS; i++) { routing_data->sessions[i].port_id = -1; + routing_data->sessions[i].fedai_id = -1; + } return 0; } -- 2.18.0

2 1

[alsa-devel] [PATCH] sound: soc: fix soc-core.c kernel-doc warning
by Randy Dunlap 03 Sep '18

03 Sep '18

From: Randy Dunlap <rdunlap(a)infradead.org> Fix kernel-doc warning: ../sound/soc/soc-core.c:2918: warning: Excess function parameter 'legacy_dai_naming' description in 'snd_soc_register_dais' Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 2 -- 1 file changed, 2 deletions(-) --- lnx-419-rc2.orig/sound/soc/soc-core.c +++ lnx-419-rc2/sound/soc/soc-core.c @@ -2910,8 +2910,6 @@ static struct snd_soc_dai *soc_add_dai(s * @component: The component the DAIs are registered for * @dai_drv: DAI driver to use for the DAIs * @count: Number of DAIs - * @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the - * parent's name. */ static int snd_soc_register_dais(struct snd_soc_component *component, struct snd_soc_dai_driver *dai_drv, size_t count)

2 2

[alsa-devel] [PATCH v2 0/2] Add support for MikroElektronika PROTO codec
by Codrin Ciubotariu 03 Sep '18

03 Sep '18

This patchset is based on Florian's initial PROTO driver: https://lkml.org/lkml/2013/5/22/342 My changes made the driver more generic, so that we could connect other I2S CPU DAIs, by using device-tree. Tested with sama5d2 I2S driver. Changes in v2: - removed unnecessary code that added constraints for the fixed SYSLK; - fixed errors reported by kbuild; - adjusted "mikroe" in vendor prefixes; - updated description in Kconfig; Codrin Ciubotariu (2): ASoC: Add driver for PROTO Audio CODEC (with a WM8731) ASoC: mikroe-proto: dt-bindings: add DT bindings for PROTO board .../bindings/sound/mikroe,mikroe-proto.txt | 23 +++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/atmel/Kconfig | 11 ++ sound/soc/atmel/Makefile | 2 + sound/soc/atmel/mikroe-proto.c | 165 ++++++++++++++++++ 5 files changed, 202 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt create mode 100644 sound/soc/atmel/mikroe-proto.c -- 2.17.1

2 4

[alsa-devel] ASoC: pxa: remove pxa2xx-pcm driver, which caused regression
by petrcvekcz＠gmail.com 02 Sep '18

02 Sep '18

From: Petr Cvek <petrcvekcz(a)gmail.com> The switch of PXA to dmaengine caused a regression for devices which are using SSP/I2S and probably AC97. The problem is in multiple allocation of DMA by function pxa2xx_soc_pcm_new() called by pxa-pcm-audio and (SSP/I2S/AC97). The regression was found on HTC Magician and after playback kernel generates an oops: WARNING: CPU: 0 PID: 892 at drivers/dma/dmaengine.c:788 dma_release_channel+0x44/0xa0 chan reference count 0 != 1 Modules linked in: ... snd_soc_magician snd_soc_uda1380 snd_soc_pxa2xx_i2s spi_pxa2xx_platform snd_soc_pxa2xx snd_soc_pxa_ssp snd_pxa2xx_lib snd_pcm_dmaengine i2c_pxa snd_soc_core snd_pcm snd_timer i2c_core snd ssp soundcore [last unloaded: pwm_bl] CPU: 0 PID: 892 Comm: aplay Tainted: G C 4.18.0-rc6-next-20180726-magician+ #16 Hardware name: HTC Magician [<c0107d90>] (unwind_backtrace) from [<c010594c>] (show_stack+0x10/0x14) [<c010594c>] (show_stack) from [<c0111680>] (__warn+0xd4/0xec) [<c0111680>] (__warn) from [<c01112d0>] (warn_slowpath_fmt+0x44/0x6c) [<c01112d0>] (warn_slowpath_fmt) from [<c034908c>] (dma_release_channel+0x44/0xa0) [<c034908c>] (dma_release_channel) from [<bf330528>] (snd_dmaengine_pcm_close_release_chan+0x44/0xb1c [snd_pcm_dmaengine]) [<bf330528>] (snd_dmaengine_pcm_close_release_chan [snd_pcm_dmaengine]) from [<bf26c29c>] (soc_pcm_components_close+0x8c/0xac [snd_soc_core]) [<bf26c29c>] (soc_pcm_components_close [snd_soc_core]) from [<bf26c3a0>] (soc_pcm_close+0xe4/0xd44 [snd_soc_core]) [<bf26c3a0>] (soc_pcm_close [snd_soc_core]) from [<bf167254>] (snd_pcm_release_substream+0x88/0x108 [snd_pcm]) [<bf167254>] (snd_pcm_release_substream [snd_pcm]) from [<bf167330>] (snd_pcm_release+0x5c/0xd2c [snd_pcm]) [<bf167330>] (snd_pcm_release [snd_pcm]) from [<c01c39e4>] (__fput+0xdc/0x1e8) [<c01c39e4>] (__fput) from [<c0129e7c>] (task_work_run+0xc4/0xd4) [<c0129e7c>] (task_work_run) from [<c01053ac>] (do_work_pending+0xc0/0xc8) [<c01053ac>] (do_work_pending) from [<c0101068>] (slow_work_pending+0xc/0x20) Exception stack(0xc2fdffb0 to 0xc2fdfff8) ffa0: 00000000 0154d1a0 01542a3c 00000000 ffc0: 0154d150 00000000 0009283c 00000006 000003f0 0154d4c8 00000000 0154d4c8 ffe0: b6ee7d70 bef13960 b6e547b8 b6cfe018 60000010 00000004 Removing bindings for snd_soc_magician fixes the problem (tested). Other machines should have the same problem if they use at least one of pxa-ssp-dai or pxa2xx-i2s. Considering pxa2xx-ac97 uses the function pxa2xx_soc_pcm_new() too it should produce the same problem (untested). After removal from I2S/SSP/AC97 machines there seems to be no user of pxa2xx-pcm, so remove the driver completely, change platform device in arch/arm/mach-pxa/devices.c to snd-soc-dummy and an example in the documentation. Signed-off-by: Petr Cvek <petrcvekcz(a)gmail.com> --- Documentation/sound/soc/machine.rst | 2 +- arch/arm/mach-pxa/devices.c | 2 +- sound/soc/pxa/Kconfig | 43 ++++++++++---------------- sound/soc/pxa/Makefile | 2 -- sound/soc/pxa/corgi.c | 1 - sound/soc/pxa/e740_wm9705.c | 2 -- sound/soc/pxa/e750_wm9705.c | 2 -- sound/soc/pxa/e800_wm9712.c | 2 -- sound/soc/pxa/em-x270.c | 2 -- sound/soc/pxa/hx4700.c | 1 - sound/soc/pxa/imote2.c | 1 - sound/soc/pxa/magician.c | 2 -- sound/soc/pxa/mioa701_wm9713.c | 2 -- sound/soc/pxa/palm27x.c | 2 -- sound/soc/pxa/poodle.c | 1 - sound/soc/pxa/pxa2xx-pcm.c | 48 ----------------------------- sound/soc/pxa/raumfeld.c | 2 -- sound/soc/pxa/spitz.c | 1 - sound/soc/pxa/tosa.c | 2 -- sound/soc/pxa/z2.c | 1 - sound/soc/pxa/zylonite.c | 3 -- 21 files changed, 18 insertions(+), 106 deletions(-) delete mode 100644 sound/soc/pxa/pxa2xx-pcm.c diff --git a/Documentation/sound/soc/machine.rst b/Documentation/sound/soc/machine.rst index 515c9444deaf..a65f41ade2ca 100644 --- a/Documentation/sound/soc/machine.rst +++ b/Documentation/sound/soc/machine.rst @@ -65,7 +65,7 @@ struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. .stream_name = "WM8731", .cpu_dai_name = "pxa-is2-dai", .codec_dai_name = "wm8731-hifi", - .platform_name = "pxa-pcm-audio", + .platform_name = "snd-soc-dummy", .codec_name = "wm8713-codec.0-001a", .init = corgi_wm8731_init, .ops = &corgi_ops, diff --git a/arch/arm/mach-pxa/devices.c b/arch/arm/mach-pxa/devices.c index a24783a03827..82ee7f6c1dcf 100644 --- a/arch/arm/mach-pxa/devices.c +++ b/arch/arm/mach-pxa/devices.c @@ -371,7 +371,7 @@ struct platform_device pxa_device_asoc_ssp4 = { }; struct platform_device pxa_device_asoc_platform = { - .name = "pxa-pcm-audio", + .name = "snd-soc-dummy", .id = -1, }; diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 776e148b0aa2..d865c1041c6b 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,12 +1,3 @@ -config SND_PXA2XX_SOC - tristate "SoC Audio for the Intel PXA2xx chip" - depends on ARCH_PXA || COMPILE_TEST - select SND_PXA2XX_LIB - help - Say Y or M if you want to add support for codecs attached to - the PXA2xx AC97, I2S or SSP interface. You will also need - to select the audio interfaces to support below. - config SND_MMP_SOC bool "Soc Audio for Marvell MMP chips" depends on ARCH_MMP @@ -43,7 +34,7 @@ config SND_MMP_SOC_SSPA config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" - depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx && I2C + depends on PXA_SHARP_C7xx && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8731 help @@ -52,7 +43,7 @@ config SND_PXA2XX_SOC_CORGI config SND_PXA2XX_SOC_SPITZ tristate "SoC Audio support for Sharp Zaurus SL-Cxx00" - depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00 && I2C + depends on PXA_SHARP_Cxx00 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8750 help @@ -61,7 +52,7 @@ config SND_PXA2XX_SOC_SPITZ config SND_PXA2XX_SOC_Z2 tristate "SoC Audio support for Zipit Z2" - depends on SND_PXA2XX_SOC && MACH_ZIPIT2 && I2C + depends on MACH_ZIPIT2 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8750 help @@ -69,7 +60,7 @@ config SND_PXA2XX_SOC_Z2 config SND_PXA2XX_SOC_POODLE tristate "SoC Audio support for Poodle" - depends on SND_PXA2XX_SOC && MACH_POODLE && I2C + depends on MACH_POODLE && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8731 help @@ -78,7 +69,7 @@ config SND_PXA2XX_SOC_POODLE config SND_PXA2XX_SOC_TOSA tristate "SoC AC97 Audio support for Tosa" - depends on SND_PXA2XX_SOC && MACH_TOSA + depends on MACH_TOSA depends on MFD_TC6393XB select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 @@ -88,7 +79,7 @@ config SND_PXA2XX_SOC_TOSA config SND_PXA2XX_SOC_E740 tristate "SoC AC97 Audio support for e740" - depends on SND_PXA2XX_SOC && MACH_E740 + depends on MACH_E740 select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -97,7 +88,7 @@ config SND_PXA2XX_SOC_E740 config SND_PXA2XX_SOC_E750 tristate "SoC AC97 Audio support for e750" - depends on SND_PXA2XX_SOC && MACH_E750 + depends on MACH_E750 select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -106,7 +97,7 @@ config SND_PXA2XX_SOC_E750 config SND_PXA2XX_SOC_E800 tristate "SoC AC97 Audio support for e800" - depends on SND_PXA2XX_SOC && MACH_E800 + depends on MACH_E800 select SND_SOC_WM9712 select SND_PXA2XX_SOC_AC97 help @@ -115,8 +106,7 @@ config SND_PXA2XX_SOC_E800 config SND_PXA2XX_SOC_EM_X270 tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" - depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ - MACH_CM_X300) + depends on MACH_EM_X270 || MACH_EXEDA || MACH_CM_X300 select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -125,8 +115,7 @@ config SND_PXA2XX_SOC_EM_X270 config SND_PXA2XX_SOC_PALM27X bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive" - depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \ - MACH_PALMT5 || MACH_PALMTE2) + depends on MACH_PALMLD || MACH_PALMTX || MACH_PALMT5 || MACH_PALMTE2 select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -155,7 +144,7 @@ config SND_SOC_TTC_DKB config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" - depends on SND_PXA2XX_SOC && MACH_ZYLONITE + depends on MACH_ZYLONITE select SND_PXA2XX_SOC_AC97 select SND_PXA_SOC_SSP select SND_SOC_WM9713 @@ -165,7 +154,7 @@ config SND_SOC_ZYLONITE config SND_SOC_RAUMFELD tristate "SoC Audio support Raumfeld audio adapter" - depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) + depends on MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR depends on I2C && SPI_MASTER select SND_PXA_SOC_SSP select SND_SOC_CS4270 @@ -175,7 +164,7 @@ config SND_SOC_RAUMFELD config SND_PXA2XX_SOC_HX4700 tristate "SoC Audio support for HP iPAQ hx4700" - depends on SND_PXA2XX_SOC && MACH_H4700 && I2C + depends on MACH_H4700 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_AK4641 help @@ -184,7 +173,7 @@ config SND_PXA2XX_SOC_HX4700 config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" - depends on SND_PXA2XX_SOC && MACH_MAGICIAN && I2C + depends on MACH_MAGICIAN && I2C select SND_PXA2XX_SOC_I2S select SND_PXA_SOC_SSP select SND_SOC_UDA1380 @@ -194,7 +183,7 @@ config SND_PXA2XX_SOC_MAGICIAN config SND_PXA2XX_SOC_MIOA701 tristate "SoC Audio support for MIO A701" - depends on SND_PXA2XX_SOC && MACH_MIOA701 + depends on MACH_MIOA701 select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9713 help @@ -203,7 +192,7 @@ config SND_PXA2XX_SOC_MIOA701 config SND_PXA2XX_SOC_IMOTE2 tristate "SoC Audio support for IMote 2" - depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C + depends on MACH_INTELMOTE2 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8940 help diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 5b265662f04f..93c0fdd041ed 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -1,13 +1,11 @@ # SPDX-License-Identifier: GPL-2.0 # PXA Platform Support -snd-soc-pxa2xx-objs := pxa2xx-pcm.o snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o snd-soc-pxa-ssp-objs := pxa-ssp.o snd-soc-mmp-objs := mmp-pcm.o snd-soc-mmp-sspa-objs := mmp-sspa.o -obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 054e0d65db9d..d36193e45be9 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -265,7 +265,6 @@ static struct snd_soc_dai_link corgi_dai = { .stream_name = "WM8731", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm8731.0-001b", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 8ab7032631b7..cfae28571ec7 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -90,7 +90,6 @@ static struct snd_soc_dai_link e740_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9705-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", }, { @@ -98,7 +97,6 @@ static struct snd_soc_dai_link e740_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9705-aux", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", }, }; diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 82bcbbb1841b..21c1f504a141 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -73,7 +73,6 @@ static struct snd_soc_dai_link e750_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9705-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", /* use ops to check startup state */ }, @@ -82,7 +81,6 @@ static struct snd_soc_dai_link e750_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9705-aux", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", }, }; diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 1ed8aa2348f1..2475366f0e0b 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -74,7 +74,6 @@ static struct snd_soc_dai_link e800_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", }, { @@ -82,7 +81,6 @@ static struct snd_soc_dai_link e800_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", }, }; diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index e046770ce70e..76b466b3eaee 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -36,7 +36,6 @@ static struct snd_soc_dai_link em_x270_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", }, { @@ -44,7 +43,6 @@ static struct snd_soc_dai_link em_x270_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", }, }; diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 6cdef5d4954e..79ffeab616f7 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -144,7 +144,6 @@ static struct snd_soc_dai_link hx4700_dai = { .stream_name = "AK4641", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "ak4641-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "ak4641.0-0012", .init = hx4700_ak4641_init, .dai_fmt = SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index 78475376f971..84dff8d34472 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -51,7 +51,6 @@ static struct snd_soc_dai_link imote2_dai = { .stream_name = "WM8940", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8940-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm8940-codec.0-0034", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 935a248e5bf6..a7e7fa08e0a4 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -296,7 +296,6 @@ static struct snd_soc_dai_link magician_dai[] = { .stream_name = "UDA1380 Playback", .cpu_dai_name = "pxa-ssp-dai.0", .codec_dai_name = "uda1380-hifi-playback", - .platform_name = "pxa-pcm-audio", .codec_name = "uda1380-codec.0-0018", .ops = &magician_playback_ops, }, @@ -305,7 +304,6 @@ static struct snd_soc_dai_link magician_dai[] = { .stream_name = "UDA1380 Capture", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "uda1380-hifi-capture", - .platform_name = "pxa-pcm-audio", .codec_name = "uda1380-codec.0-0018", .ops = &magician_capture_ops, } diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 47052fe3f5f4..121dcb1def3c 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -150,7 +150,6 @@ static struct snd_soc_dai_link mioa701_dai[] = { .codec_dai_name = "wm9713-hifi", .codec_name = "wm9713-codec", .init = mioa701_wm9713_init, - .platform_name = "pxa-pcm-audio", .ops = &mioa701_ops, }, { @@ -159,7 +158,6 @@ static struct snd_soc_dai_link mioa701_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9713-aux", .codec_name = "wm9713-codec", - .platform_name = "pxa-pcm-audio", .ops = &mioa701_ops, }, }; diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 97167048572d..490717c7a5ce 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -94,7 +94,6 @@ static struct snd_soc_dai_link palm27x_dai[] = { .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .codec_name = "wm9712-codec", - .platform_name = "pxa-pcm-audio", .init = palm27x_ac97_init, }, { @@ -103,7 +102,6 @@ static struct snd_soc_dai_link palm27x_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .codec_name = "wm9712-codec", - .platform_name = "pxa-pcm-audio", }, }; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index b6693f32fc02..728904266930 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -229,7 +229,6 @@ static struct snd_soc_dai_link poodle_dai = { .stream_name = "WM8731", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm8731.0-001b", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c deleted file mode 100644 index 72eaaef1b426..000000000000 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ /dev/null @@ -1,48 +0,0 @@ -/* - * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/dma-mapping.h> -#include <linux/module.h> -#include <linux/dmaengine.h> -#include <linux/of.h> - -#include <sound/core.h> -#include <sound/soc.h> -#include <sound/pxa2xx-lib.h> -#include <sound/dmaengine_pcm.h> - -static const struct snd_soc_component_driver pxa2xx_soc_platform = { - .ops = &pxa2xx_pcm_ops, - .pcm_new = pxa2xx_soc_pcm_new, - .pcm_free = pxa2xx_pcm_free_dma_buffers, -}; - -static int pxa2xx_soc_platform_probe(struct platform_device *pdev) -{ - return devm_snd_soc_register_component(&pdev->dev, &pxa2xx_soc_platform, - NULL, 0); -} - -static struct platform_driver pxa_pcm_driver = { - .driver = { - .name = "pxa-pcm-audio", - }, - - .probe = pxa2xx_soc_platform_probe, -}; - -module_platform_driver(pxa_pcm_driver); - -MODULE_AUTHOR("Nicolas Pitre"); -MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:pxa-pcm-audio"); diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 111a907c4eb9..b2ae5aec0b83 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -205,7 +205,6 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .name = "CS4270", \ .stream_name = "CS4270", \ .cpu_dai_name = "pxa-ssp-dai.0", \ - .platform_name = "pxa-pcm-audio", \ .codec_dai_name = "cs4270-hifi", \ .codec_name = "cs4270.0-0048", \ .dai_fmt = SND_SOC_DAIFMT_I2S | \ @@ -220,7 +219,6 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .stream_name = "Playback", \ .cpu_dai_name = "pxa-ssp-dai.1", \ .codec_dai_name = "ak4104-hifi", \ - .platform_name = "pxa-pcm-audio", \ .dai_fmt = SND_SOC_DAIFMT_I2S | \ SND_SOC_DAIFMT_NB_NF | \ SND_SOC_DAIFMT_CBS_CFS, \ diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 1671da648e95..57e4ecb9cbe2 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -262,7 +262,6 @@ static struct snd_soc_dai_link spitz_dai = { .stream_name = "WM8750", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm8750.0-001b", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index ae9c12e1ea2a..b2ddfeea6b85 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -188,7 +188,6 @@ static struct snd_soc_dai_link tosa_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", .ops = &tosa_ops, }, @@ -197,7 +196,6 @@ static struct snd_soc_dai_link tosa_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", .ops = &tosa_ops, }, diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 5b0eccd2b4dd..5e2307959b7b 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -162,7 +162,6 @@ static struct snd_soc_dai_link z2_dai = { .stream_name = "WM8750", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm8750.0-001b", .init = z2_wm8750_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 230eee450f45..753c828ade37 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -132,7 +132,6 @@ static struct snd_soc_dai_link zylonite_dai[] = { .name = "AC97", .stream_name = "AC97 HiFi", .codec_name = "wm9713-codec", - .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9713-hifi", .init = zylonite_wm9713_init, @@ -141,7 +140,6 @@ static struct snd_soc_dai_link zylonite_dai[] = { .name = "AC97 Aux", .stream_name = "AC97 Aux", .codec_name = "wm9713-codec", - .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9713-aux", }, @@ -149,7 +147,6 @@ static struct snd_soc_dai_link zylonite_dai[] = { .name = "WM9713 Voice", .stream_name = "WM9713 Voice", .codec_name = "wm9713-codec", - .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa-ssp-dai.2", .codec_dai_name = "wm9713-voice", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | -- 2.18.0

3 10

[alsa-devel] token to prevent spam?
by Lee, Keerock 01 Sep '18

01 Sep '18

Hi all, Could anyone please let me know how to acquire Token to prevent spam when you create an account for the alsa-project.org? I am having trouble in creating an account. Thank you, Keerock

1 0

[alsa-devel] [PATCH] ALSA: control: Make sure that id->index does not overflow in function snd_ctl_replace
by Xiao Yang 01 Sep '18

01 Sep '18

From: Young_X <YangX92(a)hotmail.com> The ALSA control code expects that the range of assigned indices to a control is continuous and does not overflow. Currently there are no checks to enforce this. If a control with a overflowing index range is created that control becomes effectively inaccessible and unremovable since snd_ctl_find_id() will not be able to find it. This patch adds a check that makes sure that controls with a overflowing index range can not be created. (same issue as CVE-2014-4656) Signed-off-by: Young_X <YangX92(a)hotmail.com> --- sound/core/control.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/core/control.c b/sound/core/control.c index 9aa15bf..6435772 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -441,6 +441,11 @@ int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, goto error; } id = kcontrol->id; + if (id.index > UINT_MAX - kcontrol->count) { + ret = -EINVAL; + goto error; + } + down_write(&card->controls_rwsem); old = snd_ctl_find_id(card, &id); if (!old) { -- 2.7.4

2 1

[alsa-devel] [asoc:for-4.20 333/374] sound/pci/hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'?
by kbuild test robot 01 Sep '18

01 Sep '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 5fb6589acc3860304436aa436e7ea33712de6fc2 commit: 6bae5ea9498926440ffc883f3dbceb0adc65e492 [333/374] ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers config: sh-allmodconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 6bae5ea9498926440ffc883f3dbceb0adc65e492 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sound/pci/hda/patch_ca0132.c: In function 'patch_ca0132': >> sound/pci/hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'? [-Werror=implicit-function-declaration] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^~~~~~~~~ pcim_iomap sound/pci/hda/patch_ca0132.c:7650:18: warning: assignment makes pointer from integer without a cast [-Wint-conversion] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^ cc1: some warnings being treated as errors vim +7650 sound/pci/hda/patch_ca0132.c d5c016b56 Gabriele Martino 2015-05-18 7581 95c6e9cb7 Ian Minett 2011-06-15 7582 static int patch_ca0132(struct hda_codec *codec) 95c6e9cb7 Ian Minett 2011-06-15 7583 { 95c6e9cb7 Ian Minett 2011-06-15 7584 struct ca0132_spec *spec; a73d511c4 Ian Minett 2012-12-20 7585 int err; d5c016b56 Gabriele Martino 2015-05-18 7586 const struct snd_pci_quirk *quirk; 95c6e9cb7 Ian Minett 2011-06-15 7587 4e76a8833 Takashi Iwai 2014-02-25 7588 codec_dbg(codec, "patch_ca0132\n"); 95c6e9cb7 Ian Minett 2011-06-15 7589 95c6e9cb7 Ian Minett 2011-06-15 7590 spec = kzalloc(sizeof(*spec), GFP_KERNEL); 95c6e9cb7 Ian Minett 2011-06-15 7591 if (!spec) 95c6e9cb7 Ian Minett 2011-06-15 7592 return -ENOMEM; 95c6e9cb7 Ian Minett 2011-06-15 7593 codec->spec = spec; 993884f6a Chih-Chung Chang 2013-03-25 7594 spec->codec = codec; 95c6e9cb7 Ian Minett 2011-06-15 7595 225068ab2 Takashi Iwai 2015-05-29 7596 codec->patch_ops = ca0132_patch_ops; 225068ab2 Takashi Iwai 2015-05-29 7597 codec->pcm_format_first = 1; 225068ab2 Takashi Iwai 2015-05-29 7598 codec->no_sticky_stream = 1; 225068ab2 Takashi Iwai 2015-05-29 7599 d5c016b56 Gabriele Martino 2015-05-18 7600 /* Detect codec quirk */ d5c016b56 Gabriele Martino 2015-05-18 7601 quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); d5c016b56 Gabriele Martino 2015-05-18 7602 if (quirk) d5c016b56 Gabriele Martino 2015-05-18 7603 spec->quirk = quirk->value; d5c016b56 Gabriele Martino 2015-05-18 7604 else d5c016b56 Gabriele Martino 2015-05-18 7605 spec->quirk = QUIRK_NONE; d5c016b56 Gabriele Martino 2015-05-18 7606 e24aa0a4c Takashi Iwai 2014-08-10 7607 spec->dsp_state = DSP_DOWNLOAD_INIT; a7e76271b Ian Minett 2012-12-20 7608 spec->num_mixers = 1; 017310fbe Connor McAdams 2018-05-08 7609 017310fbe Connor McAdams 2018-05-08 7610 /* Set which mixers each quirk uses. */ 017310fbe Connor McAdams 2018-05-08 7611 switch (spec->quirk) { 017310fbe Connor McAdams 2018-05-08 7612 case QUIRK_SBZ: e25e34450 Connor McAdams 2018-08-08 7613 spec->mixers[0] = desktop_mixer; 017310fbe Connor McAdams 2018-05-08 7614 snd_hda_codec_set_name(codec, "Sound Blaster Z"); 017310fbe Connor McAdams 2018-05-08 7615 break; e25e34450 Connor McAdams 2018-08-08 7616 case QUIRK_R3D: e25e34450 Connor McAdams 2018-08-08 7617 spec->mixers[0] = desktop_mixer; e25e34450 Connor McAdams 2018-08-08 7618 snd_hda_codec_set_name(codec, "Recon3D"); e25e34450 Connor McAdams 2018-08-08 7619 break; 017310fbe Connor McAdams 2018-05-08 7620 case QUIRK_R3DI: 017310fbe Connor McAdams 2018-05-08 7621 spec->mixers[0] = r3di_mixer; 017310fbe Connor McAdams 2018-05-08 7622 snd_hda_codec_set_name(codec, "Recon3Di"); 017310fbe Connor McAdams 2018-05-08 7623 break; 017310fbe Connor McAdams 2018-05-08 7624 default: a7e76271b Ian Minett 2012-12-20 7625 spec->mixers[0] = ca0132_mixer; 017310fbe Connor McAdams 2018-05-08 7626 break; 017310fbe Connor McAdams 2018-05-08 7627 } a7e76271b Ian Minett 2012-12-20 7628 08eca6b1f Connor McAdams 2018-08-08 7629 /* Setup whether or not to use alt functions/controls/pci_mmio */ 009b8f979 Connor McAdams 2018-05-08 7630 switch (spec->quirk) { 009b8f979 Connor McAdams 2018-05-08 7631 case QUIRK_SBZ: e42c7c731 Connor McAdams 2018-08-08 7632 case QUIRK_R3D: 08eca6b1f Connor McAdams 2018-08-08 7633 spec->use_alt_controls = true; 08eca6b1f Connor McAdams 2018-08-08 7634 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7635 spec->use_pci_mmio = true; 08eca6b1f Connor McAdams 2018-08-08 7636 break; 009b8f979 Connor McAdams 2018-05-08 7637 case QUIRK_R3DI: 47cdf76e4 Connor McAdams 2018-05-08 7638 spec->use_alt_controls = true; 009b8f979 Connor McAdams 2018-05-08 7639 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7640 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7641 break; 009b8f979 Connor McAdams 2018-05-08 7642 default: 47cdf76e4 Connor McAdams 2018-05-08 7643 spec->use_alt_controls = false; 009b8f979 Connor McAdams 2018-05-08 7644 spec->use_alt_functions = false; 08eca6b1f Connor McAdams 2018-08-08 7645 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7646 break; 009b8f979 Connor McAdams 2018-05-08 7647 } 009b8f979 Connor McAdams 2018-05-08 7648 08eca6b1f Connor McAdams 2018-08-08 7649 if (spec->use_pci_mmio) { 08eca6b1f Connor McAdams 2018-08-08 @7650 spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); 08eca6b1f Connor McAdams 2018-08-08 7651 if (spec->mem_base == NULL) { 08eca6b1f Connor McAdams 2018-08-08 7652 codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE."); 08eca6b1f Connor McAdams 2018-08-08 7653 spec->quirk = QUIRK_NONE; 08eca6b1f Connor McAdams 2018-08-08 7654 } 08eca6b1f Connor McAdams 2018-08-08 7655 } 08eca6b1f Connor McAdams 2018-08-08 7656 5aaca44d8 Ian Minett 2012-12-20 7657 spec->base_init_verbs = ca0132_base_init_verbs; 5aaca44d8 Ian Minett 2012-12-20 7658 spec->base_exit_verbs = ca0132_base_exit_verbs; 5aaca44d8 Ian Minett 2012-12-20 7659 993884f6a Chih-Chung Chang 2013-03-25 7660 INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed); 993884f6a Chih-Chung Chang 2013-03-25 7661 95c6e9cb7 Ian Minett 2011-06-15 7662 ca0132_init_chip(codec); 95c6e9cb7 Ian Minett 2011-06-15 7663 95c6e9cb7 Ian Minett 2011-06-15 7664 ca0132_config(codec); 95c6e9cb7 Ian Minett 2011-06-15 7665 d5c016b56 Gabriele Martino 2015-05-18 7666 err = ca0132_prepare_verbs(codec); d5c016b56 Gabriele Martino 2015-05-18 7667 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7668 goto error; d5c016b56 Gabriele Martino 2015-05-18 7669 a73d511c4 Ian Minett 2012-12-20 7670 err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); a73d511c4 Ian Minett 2012-12-20 7671 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7672 goto error; a73d511c4 Ian Minett 2012-12-20 7673 95c6e9cb7 Ian Minett 2011-06-15 7674 return 0; cc91ceaf3 Takashi Iwai 2017-09-04 7675 cc91ceaf3 Takashi Iwai 2017-09-04 7676 error: cc91ceaf3 Takashi Iwai 2017-09-04 7677 ca0132_free(codec); cc91ceaf3 Takashi Iwai 2017-09-04 7678 return err; 95c6e9cb7 Ian Minett 2011-06-15 7679 } 95c6e9cb7 Ian Minett 2011-06-15 7680 :::::: The code at line 7650 was first introduced by commit :::::: 08eca6b1f1468a4021bac7b3929fd3eb491e2629 ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio :::::: TO: Connor McAdams <conmanx360(a)gmail.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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