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

[alsa-devel] [PATCH] ALSA: pcm: Update hardware pointer before start capture
by Ricardo Biehl Pasquali 10 Sep '18

10 Sep '18

This ensures the transfer loop won't waste a run to read the few frames (if any) between start and hw_ptr update. It will wait for the next interrupt with wait_for_avail(). Signed-off-by: Ricardo Biehl Pasquali <pasqualirb(a)gmail.com> --- sound/core/pcm_lib.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7f71c2449..40013b26f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2172,6 +2172,10 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, if (err < 0) goto _end_unlock; + runtime->twake = runtime->control->avail_min ? : 1; + if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) + snd_pcm_update_hw_ptr(substream); + if (!is_playback && runtime->status->state == SNDRV_PCM_STATE_PREPARED) { if (size >= runtime->start_threshold) { @@ -2185,10 +2189,8 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, } } - runtime->twake = runtime->control->avail_min ? : 1; - if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) - snd_pcm_update_hw_ptr(substream); avail = snd_pcm_avail(substream); + while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t cont; -- 2.17.1

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[alsa-devel] [PATCH] ASoC: core: add support to card rebind
by Srinivas Kandagatla 10 Sep '18

10 Sep '18

Current behaviour of ASoC core w.r.t to component removal is that it unregisters dependent sound card totally. There is no support to rebind the card if the component comes back. Typical use case is DSP restart or kernel modules itself. With this patch, core now maintains list of cards that are unbind due to any of its depended components are removed and card not unregistered yet. This list is cleared when the card is rebind successfully or when the card is unregistered from machine driver. This list of unbind cards are tried to bind once again after every new component is successfully added, giving a fair chance for card bind to be successful. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Hi Mark, This time I tried doing this without component framework, by maintaining list of ubind card in ASoC core that are still not unregistered. This list is tried to bind for every successful add of component. It looks much better than the component framework one! thanks, srini include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 86 ++++++++++++++++++++++++++++++-------------- 2 files changed, 61 insertions(+), 27 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a4915148f739..d9a8aea213cf 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1092,6 +1092,7 @@ struct snd_soc_card { /* lists of probed devices belonging to this card */ struct list_head component_dev_list; + struct list_head list; struct list_head widgets; struct list_head paths; @@ -1359,6 +1360,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->dapm_list); INIT_LIST_HEAD(&card->aux_comp_list); INIT_LIST_HEAD(&card->component_dev_list); + INIT_LIST_HEAD(&card->list); } static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6d33634b934b..795796978fc8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -52,6 +52,7 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); static DEFINE_MUTEX(client_mutex); static LIST_HEAD(component_list); +static LIST_HEAD(unbind_card_list); /* * This is a timeout to do a DAPM powerdown after a stream is closed(). @@ -2678,6 +2679,34 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); +static int snd_soc_bind_card(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + int ret; + + ret = snd_soc_instantiate_card(card); + if (ret != 0) + return ret; + + /* deactivate pins to sleep state */ + list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int j; + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + } + + if (!cpu_dai->active) + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + + return ret; +} + /** * snd_soc_register_card - Register a card with the ASoC core * @@ -2687,7 +2716,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); int snd_soc_register_card(struct snd_soc_card *card) { int i, ret; - struct snd_soc_pcm_runtime *rtd; if (!card->name || !card->dev) return -EINVAL; @@ -2719,28 +2747,23 @@ int snd_soc_register_card(struct snd_soc_card *card) mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); - ret = snd_soc_instantiate_card(card); - if (ret != 0) - return ret; - - /* deactivate pins to sleep state */ - list_for_each_entry(rtd, &card->rtd_list, list) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int j; - - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); - } + return snd_soc_bind_card(card); +} +EXPORT_SYMBOL_GPL(snd_soc_register_card); - if (!cpu_dai->active) - pinctrl_pm_select_sleep_state(cpu_dai->dev); +static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) +{ + if (card->instantiated) { + card->instantiated = false; + snd_soc_dapm_shutdown(card); + soc_cleanup_card_resources(card); + if (!unregister) + list_add(&card->list, &unbind_card_list); + } else { + if (unregister) + list_del(&card->list); } - - return ret; } -EXPORT_SYMBOL_GPL(snd_soc_register_card); /** * snd_soc_unregister_card - Unregister a card with the ASoC core @@ -2750,12 +2773,8 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card); */ int snd_soc_unregister_card(struct snd_soc_card *card) { - if (card->instantiated) { - card->instantiated = false; - snd_soc_dapm_shutdown(card); - soc_cleanup_card_resources(card); - dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); - } + snd_soc_unbind_card(card, true); + dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); return 0; } @@ -3101,7 +3120,7 @@ static void snd_soc_component_del_unlocked(struct snd_soc_component *component) struct snd_soc_card *card = component->card; if (card) - snd_soc_unregister_card(card); + snd_soc_unbind_card(card, false); list_del(&component->list); } @@ -3141,6 +3160,18 @@ static void convert_endianness_formats(struct snd_soc_pcm_stream *stream) stream->formats |= endianness_format_map[i]; } +static void snd_soc_try_rebind_card(void) +{ + struct snd_soc_card *card, *c; + + if (!list_empty(&unbind_card_list)) { + list_for_each_entry_safe(card, c, &unbind_card_list, list) { + if (!snd_soc_bind_card(card)) + list_del(&card->list); + } + } +} + int snd_soc_add_component(struct device *dev, struct snd_soc_component *component, const struct snd_soc_component_driver *component_driver, @@ -3168,6 +3199,7 @@ int snd_soc_add_component(struct device *dev, } snd_soc_component_add(component); + snd_soc_try_rebind_card(); return 0; -- 2.18.0

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

07 Sep '18

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); > ^ Note that this isn't causing a warning upstream, presumably due to an implicit inclusion that isn't present in the v4.4 kernel that you appear to be using, or gets missed due to config differences.

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[alsa-devel] [bug report] ASoC: dapm: Move connection of CODEC to CODEC DAIs
by Dan Carpenter 07 Sep '18

07 Sep '18

Hello Charles Keepax, The patch 778ff5bb8689: "ASoC: dapm: Move connection of CODEC to CODEC DAIs" from Sep 5, 2018, leads to the following static checker warning: sound/soc/soc-dapm.c:4089 dapm_connect_dai_link_widgets() error: 'playback' dereferencing possible ERR_PTR() sound/soc/soc-dapm.c:4110 dapm_connect_dai_link_widgets() error: 'capture' dereferencing possible ERR_PTR() sound/soc/soc-dapm.c 4077 4078 for_each_rtd_codec_dai(rtd, i, codec_dai) { 4079 4080 /* connect BE DAI playback if widgets are valid */ 4081 codec = codec_dai->playback_widget; 4082 4083 if (playback_cpu && codec) { 4084 if (!playback) { 4085 playback = snd_soc_dapm_new_dai(card, rtd, 4086 playback_cpu, 4087 codec); 4088 4089 snd_soc_dapm_add_path(&card->dapm, playback_cpu, 4090 playback, NULL, NULL); ^^^^^^^^ snd_soc_dapm_new_dai() returns error pointers. 4091 } 4092 4093 dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", 4094 cpu_dai->component->name, playback_cpu->name, 4095 codec_dai->component->name, codec->name); 4096 4097 snd_soc_dapm_add_path(&card->dapm, playback, codec, 4098 NULL, NULL); 4099 } 4100 4101 /* connect BE DAI capture if widgets are valid */ 4102 codec = codec_dai->capture_widget; 4103 4104 if (codec && capture_cpu) { 4105 if (!capture) { 4106 capture = snd_soc_dapm_new_dai(card, rtd, 4107 codec, 4108 capture_cpu); 4109 4110 snd_soc_dapm_add_path(&card->dapm, capture, 4111 capture_cpu, NULL, NULL); ^^^^^^^^^^^ 4112 } 4113 4114 dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", 4115 codec_dai->component->name, codec->name, 4116 cpu_dai->component->name, capture_cpu->name); 4117 4118 snd_soc_dapm_add_path(&card->dapm, codec, capture, 4119 NULL, NULL); regards, dan carpenter

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[alsa-devel] [PATCH v2] ASoC: Intel: hdac_hdmi: Limit sampling rates at dai creation
by Yong Zhi 07 Sep '18

07 Sep '18

Playback of 44.1Khz contents with HDMI plugged returns "Invalid pipe config" because HDMI paths in the FW topology are configured to operate at 48Khz. This patch filters out sampling rates not supported at hdac_hdmi_create_dais() to let user space SRC to do the converting. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> --- v2: - Add Reviewed-by tag. - Use (and tested) Takashi's sample code in his v1 review. sound/soc/codecs/hdac_hdmi.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 7b8533abf637..590dc1e33a60 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1410,6 +1410,12 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdev, if (ret) return ret; + /* Filter out 44.1, 88.2 and 176.4Khz */ + rates &= ~(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_176400); + if (!rates) + return -EINVAL; + sprintf(dai_name, "intel-hdmi-hifi%d", i+1); hdmi_dais[i].name = devm_kstrdup(&hdev->dev, dai_name, GFP_KERNEL); -- 2.7.4

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[alsa-devel] Applied "ASoC: max98373: usleep_range() needs include/delay.h" to the asoc tree
by Mark Brown 07 Sep '18

07 Sep '18

The patch ASoC: max98373: usleep_range() needs include/delay.h 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 3004136b90bedc9e254ff659adb7a60299e9495e Mon Sep 17 00:00:00 2001 From: Grant Grundler <grundler(a)chromium.org> Date: Thu, 6 Sep 2018 17:27:28 -0700 Subject: [PATCH] ASoC: max98373: usleep_range() needs include/delay.h 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> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98373.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 1093f766d0d2..d6868c9a9ce6 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -2,6 +2,7 @@ // Copyright (c) 2017, Maxim Integrated #include <linux/acpi.h> +#include <linux/delay.h> #include <linux/i2c.h> #include <linux/module.h> #include <linux/regmap.h> -- 2.19.0.rc1

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[alsa-devel] [PATCH] ASoC: cs4265: Add a MIC pre. switch
by Matt Flax 07 Sep '18

07 Sep '18

This patch adds a MIC preamp enable switch as a SOC_SINGLE. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/codecs/cs4265.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index d9eebf6a..03476f93 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -161,6 +161,7 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { 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), + SOC_SINGLE("MIC Preamp", CS4265_PWRCTL, 3, 1, 1), }; static const struct snd_soc_dapm_widget cs4265_dapm_widgets[] = { -- 2.17.1

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[alsa-devel] [PATCH v2] ASoC: rsnd: Add device tree binding for r8a77990
by Yoshihiro Kaneko 07 Sep '18

07 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. v2 [Yoshihiro Kaneko] - Rebased 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 9e764270..58971a6 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -353,6 +353,7 @@ Required properties: - "renesas,rcar_sound-r8a7795" (R-Car H3) - "renesas,rcar_sound-r8a7796" (R-Car M3-W) - "renesas,rcar_sound-r8a77965" (R-Car M3-N) + - "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: dapm: Avoid uninitialised variable warning
by Charles Keepax 07 Sep '18

07 Sep '18

Commit 4a75aae17b2a ("ASoC: dapm: Add support for multi-CODEC CODEC to CODEC links") adds loops that iterate over multiple CODECs in snd_soc_dai_link_event. This also introduced a compiler warning for a potentially uninitialised variable in the case no CODECs are present. This should never be the case as the DAI link must by definition contain at least 1 CODEC however probably best to avoid the compiler warning by initialising ret to zero. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9f4edcd19e022..e496bc642c990 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3623,7 +3623,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_pcm_hw_params *params = NULL; struct snd_pcm_runtime *runtime = NULL; unsigned int fmt; - int ret; + int ret = 0; config = rtd->dai_link->params + rtd->params_select; -- 2.11.0

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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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