January 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: MAINTAINERS: fsl: Change Fabio's email address
by Fabio Estevam 21 Jan '19

21 Jan '19

I prefer to use my personal email address for kernel related work. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- MAINTAINERS | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index a3d1222b81ad..f3de979b8e58 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6178,7 +6178,7 @@ FREESCALE SOC SOUND DRIVERS M: Timur Tabi <timur(a)kernel.org> M: Nicolin Chen <nicoleotsuka(a)gmail.com> M: Xiubo Li <Xiubo.Lee(a)gmail.com> -R: Fabio Estevam <fabio.estevam(a)nxp.com> +R: Fabio Estevam <festevam(a)gmail.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) L: linuxppc-dev(a)lists.ozlabs.org S: Maintained @@ -10954,7 +10954,7 @@ F: include/linux/nvmem-consumer.h F: include/linux/nvmem-provider.h NXP SGTL5000 DRIVER -M: Fabio Estevam <fabio.estevam(a)nxp.com> +M: Fabio Estevam <festevam(a)gmail.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained F: Documentation/devicetree/bindings/sound/sgtl5000.txt -- 2.17.1

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[alsa-devel] Applied "ASoC: soc-core: add new snd_soc_flush_all_delayed_work()" to the asoc tree
by Mark Brown 21 Jan '19

21 Jan '19

The patch ASoC: soc-core: add new snd_soc_flush_all_delayed_work() 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 65462e445f78cb2f9378443be7ba8b7e07300694 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 21 Jan 2019 09:32:37 +0900 Subject: [PATCH] ASoC: soc-core: add new snd_soc_flush_all_delayed_work() soc-core is calling flush_delayed_work() many times for same purpose. Same code in many places makes code un-understandable. This patch adds new snd_soc_flush_all_delayed_work() for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2c63921675d5..eeb794d8a262 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -425,6 +425,14 @@ struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime); +static void snd_soc_flush_all_delayed_work(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + + for_each_card_rtds(card, rtd) + flush_delayed_work(&rtd->delayed_work); +} + static void codec2codec_close_delayed_work(struct work_struct *work) { /* @@ -494,8 +502,7 @@ int snd_soc_suspend(struct device *dev) } /* close any waiting streams */ - for_each_card_rtds(card, rtd) - flush_delayed_work(&rtd->delayed_work); + snd_soc_flush_all_delayed_work(card); for_each_card_rtds(card, rtd) { @@ -2228,11 +2235,8 @@ static int soc_probe(struct platform_device *pdev) static int soc_cleanup_card_resources(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd; - /* make sure any delayed work runs */ - for_each_card_rtds(card, rtd) - flush_delayed_work(&rtd->delayed_work); + snd_soc_flush_all_delayed_work(card); /* free the ALSA card at first; this syncs with pending operations */ snd_card_free(card->snd_card); @@ -2275,8 +2279,7 @@ int snd_soc_poweroff(struct device *dev) * Flush out pmdown_time work - we actually do want to run it * now, we're shutting down so no imminent restart. */ - for_each_card_rtds(card, rtd) - flush_delayed_work(&rtd->delayed_work); + snd_soc_flush_all_delayed_work(card); snd_soc_dapm_shutdown(card); -- 2.20.1

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[alsa-devel] Applied "ASoC: soc-core: merge card resources cleanup method" to the asoc tree
by Mark Brown 21 Jan '19

21 Jan '19

The patch ASoC: soc-core: merge card resources cleanup method 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 53e947a0e1f770b9707febb7054b856878945d50 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 21 Jan 2019 09:32:42 +0900 Subject: [PATCH] ASoC: soc-core: merge card resources cleanup method We need to cleanup card resources when snd_soc_instantiate_card() was failed, or when snd_soc_unbind_card() was called. But they are cleanuping card resources on each way. Same code in many places makes code un-understandable. This patch reuses soc_cleanup_card_resources() for cleanuping code resource. Then, it makes avoiding cleanup order. It will be called from snd_soc_instantiate_card() and snd_soc_unbind_card(). Then, original soc_cleanup_card_resources() included snd_soc_flush_all_delayed_work(), but it is now separated. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 103 ++++++++++++++++++------------------------- 1 file changed, 43 insertions(+), 60 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index eeb794d8a262..d59b5ea9fa25 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2008,6 +2008,29 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) } } +static int soc_cleanup_card_resources(struct snd_soc_card *card) +{ + /* free the ALSA card at first; this syncs with pending operations */ + if (card->snd_card) + snd_card_free(card->snd_card); + + /* remove and free each DAI */ + soc_remove_dai_links(card); + soc_remove_pcm_runtimes(card); + + /* remove auxiliary devices */ + soc_remove_aux_devices(card); + + snd_soc_dapm_free(&card->dapm); + soc_cleanup_card_debugfs(card); + + /* remove the card */ + if (card->remove) + card->remove(card); + + return 0; +} + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; @@ -2017,6 +2040,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); + card->dapm.bias_level = SND_SOC_BIAS_OFF; + card->dapm.dev = card->dev; + card->dapm.card = card; + list_add(&card->dapm.list, &card->dapm_list); + /* check whether any platform is ignore machine FE and using topology */ soc_check_tplg_fes(card); @@ -2024,14 +2052,14 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) for_each_card_prelinks(card, i, dai_link) { ret = soc_bind_dai_link(card, dai_link); if (ret != 0) - goto base_error; + goto probe_end; } /* bind aux_devs too */ for (i = 0; i < card->num_aux_devs; i++) { ret = soc_bind_aux_dev(card, i); if (ret != 0) - goto base_error; + goto probe_end; } /* add predefined DAI links to the list */ @@ -2045,16 +2073,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) dev_err(card->dev, "ASoC: can't create sound card for card %s: %d\n", card->name, ret); - goto base_error; + goto probe_end; } soc_init_card_debugfs(card); - card->dapm.bias_level = SND_SOC_BIAS_OFF; - card->dapm.dev = card->dev; - card->dapm.card = card; - list_add(&card->dapm.list, &card->dapm_list); - #ifdef CONFIG_DEBUG_FS snd_soc_dapm_debugfs_init(&card->dapm, card->debugfs_card_root); #endif @@ -2076,7 +2099,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) if (card->probe) { ret = card->probe(card); if (ret < 0) - goto card_probe_error; + goto probe_end; } /* probe all components used by DAI links on this card */ @@ -2087,7 +2110,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) dev_err(card->dev, "ASoC: failed to instantiate card %d\n", ret); - goto probe_dai_err; + goto probe_end; } } } @@ -2095,7 +2118,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) /* probe auxiliary components */ ret = soc_probe_aux_devices(card); if (ret < 0) - goto probe_dai_err; + goto probe_end; /* * Find new DAI links added during probing components and bind them. @@ -2107,10 +2130,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) ret = soc_init_dai_link(card, dai_link); if (ret) - goto probe_dai_err; + goto probe_end; ret = soc_bind_dai_link(card, dai_link); if (ret) - goto probe_dai_err; + goto probe_end; } /* probe all DAI links on this card */ @@ -2121,7 +2144,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) dev_err(card->dev, "ASoC: failed to instantiate card %d\n", ret); - goto probe_dai_err; + goto probe_end; } } } @@ -2168,7 +2191,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) { dev_err(card->dev, "ASoC: %s late_probe() failed: %d\n", card->name, ret); - goto probe_aux_dev_err; + goto probe_end; } } @@ -2178,33 +2201,17 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) { dev_err(card->dev, "ASoC: failed to register soundcard %d\n", ret); - goto probe_aux_dev_err; + goto probe_end; } card->instantiated = 1; dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); - mutex_unlock(&card->mutex); - mutex_unlock(&client_mutex); - - return 0; -probe_aux_dev_err: - soc_remove_aux_devices(card); - -probe_dai_err: - soc_remove_dai_links(card); - -card_probe_error: - if (card->remove) - card->remove(card); - - snd_soc_dapm_free(&card->dapm); - soc_cleanup_card_debugfs(card); - snd_card_free(card->snd_card); +probe_end: + if (ret < 0) + soc_cleanup_card_resources(card); -base_error: - soc_remove_pcm_runtimes(card); mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); @@ -2233,31 +2240,6 @@ static int soc_probe(struct platform_device *pdev) return snd_soc_register_card(card); } -static int soc_cleanup_card_resources(struct snd_soc_card *card) -{ - /* make sure any delayed work runs */ - snd_soc_flush_all_delayed_work(card); - - /* free the ALSA card at first; this syncs with pending operations */ - snd_card_free(card->snd_card); - - /* remove and free each DAI */ - soc_remove_dai_links(card); - soc_remove_pcm_runtimes(card); - - /* remove auxiliary devices */ - soc_remove_aux_devices(card); - - snd_soc_dapm_free(&card->dapm); - soc_cleanup_card_debugfs(card); - - /* remove the card */ - if (card->remove) - card->remove(card); - - return 0; -} - /* removes a socdev */ static int soc_remove(struct platform_device *pdev) { @@ -2823,6 +2805,7 @@ static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) if (card->instantiated) { card->instantiated = false; snd_soc_dapm_shutdown(card); + snd_soc_flush_all_delayed_work(card); soc_cleanup_card_resources(card); if (!unregister) list_add(&card->list, &unbind_card_list); -- 2.20.1

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[alsa-devel] Applied "ASoC: soc-core: reduce if/else nest on soc_probe_link_dais" to the asoc tree
by Mark Brown 21 Jan '19

21 Jan '19

The patch ASoC: soc-core: reduce if/else nest on soc_probe_link_dais 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 52293596f5afff10d14e033aa3edfc801a31b3a1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 21 Jan 2019 09:32:50 +0900 Subject: [PATCH] ASoC: soc-core: reduce if/else nest on soc_probe_link_dais Deep nested codec is not readable. Let's reduce if/else nest. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 37 +++++++++++++++++-------------------- 1 file changed, 17 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d59b5ea9fa25..7fc10a41e0e9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1612,27 +1612,24 @@ static int soc_probe_link_dais(struct snd_soc_card *card, dai_link->stream_name); return ret; } - } else { - - if (!dai_link->params) { - /* create the pcm */ - ret = soc_new_pcm(rtd, num); - if (ret < 0) { - dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", - dai_link->stream_name, ret); - return ret; - } - ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd); - if (ret < 0) - return ret; - ret = soc_link_dai_pcm_new(rtd->codec_dais, - rtd->num_codecs, rtd); - if (ret < 0) - return ret; - } else { - INIT_DELAYED_WORK(&rtd->delayed_work, - codec2codec_close_delayed_work); + } else if (!dai_link->params) { + /* create the pcm */ + ret = soc_new_pcm(rtd, num); + if (ret < 0) { + dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", + dai_link->stream_name, ret); + return ret; } + ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd); + if (ret < 0) + return ret; + ret = soc_link_dai_pcm_new(rtd->codec_dais, + rtd->num_codecs, rtd); + if (ret < 0) + return ret; + } else { + INIT_DELAYED_WORK(&rtd->delayed_work, + codec2codec_close_delayed_work); } return 0; -- 2.20.1

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[alsa-devel] [PATCH] ASoC: simple-card: rename to asoc_simple_card_canonicalize_platform()
by Kuninori Morimoto 21 Jan '19

21 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current simple-card is using asoc_simple_card_canonicalize_dailink(). Its naming is "dailink", but is for "platform". We already have asoc_simple_card_canonicalize_cpu() for "cpu", let's follow same naming rule. It never return error, so, void function is better idea. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/simple_card_utils.h | 2 +- sound/soc/generic/audio-graph-card.c | 11 +++-------- sound/soc/generic/simple-card-utils.c | 7 ++----- sound/soc/generic/simple-card.c | 11 +++-------- 4 files changed, 9 insertions(+), 22 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index ab5a2ba..7afe453 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -108,7 +108,7 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep, int asoc_simple_card_init_dai(struct snd_soc_dai *dai, struct asoc_simple_dai *simple_dai); -int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link); +void asoc_simple_card_canonicalize_platform(struct snd_soc_dai_link *dai_link); void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, int is_single_links); diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 42b077c..bb12351 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -307,14 +307,12 @@ static int graph_dai_link_of_dpcm(struct graph_priv *priv, "prefix"); } + asoc_simple_card_canonicalize_platform(dai_link); + ret = asoc_simple_card_of_parse_tdm(ep, dai); if (ret) return ret; - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - return ret; - ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, NULL, &dai_link->dai_fmt); if (ret < 0) @@ -405,10 +403,6 @@ static int graph_dai_link_of(struct graph_priv *priv, if (ret < 0) return ret; - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - return ret; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, "%s-%s", dai_link->cpu_dai_name, @@ -419,6 +413,7 @@ static int graph_dai_link_of(struct graph_priv *priv, dai_link->ops = &graph_ops; dai_link->init = graph_dai_init; + asoc_simple_card_canonicalize_platform(dai_link); asoc_simple_card_canonicalize_cpu(dai_link, of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 3c0901d..5c1424f 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -394,16 +394,13 @@ int asoc_simple_card_init_dai(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai); -int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link) +void asoc_simple_card_canonicalize_platform(struct snd_soc_dai_link *dai_link) { /* Assumes platform == cpu */ if (!dai_link->platforms->of_node) dai_link->platforms->of_node = dai_link->cpu_of_node; - - return 0; - } -EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_dailink); +EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_platform); void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, int is_single_links) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d8a0d1e..08df261 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -297,14 +297,12 @@ static int simple_dai_link_of_dpcm(struct simple_priv *priv, simple_get_conversion(dev, np, &dai_props->adata); + asoc_simple_card_canonicalize_platform(dai_link); + ret = asoc_simple_card_of_parse_tdm(np, dai); if (ret) return ret; - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - return ret; - snprintf(prop, sizeof(prop), "%smclk-fs", prefix); of_property_read_u32(top, PREFIX "mclk-fs", &dai_props->mclk_fs); of_property_read_u32(node, prop, &dai_props->mclk_fs); @@ -409,10 +407,6 @@ static int simple_dai_link_of(struct simple_priv *priv, if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - goto dai_link_of_err; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, "%s-%s", dai_link->cpu_dai_name, @@ -424,6 +418,7 @@ static int simple_dai_link_of(struct simple_priv *priv, dai_link->init = simple_dai_init; asoc_simple_card_canonicalize_cpu(dai_link, single_cpu); + asoc_simple_card_canonicalize_platform(dai_link); dai_link_of_err: of_node_put(node); -- 2.7.4

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[alsa-devel] [PATCH] ASoC: dapm: Skip suspending widgets with ignore flag
by Zhiwei Jiang 21 Jan '19

21 Jan '19

For endpoint wigdets which have set the suspend ignore flag, asoc shouldn't mark them as dirty when ASoC suspend function is called. Signed-off-by: Zhiwei Jiang <qq282012236(a)gmail.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 a5178845065b..68c49090f561 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -278,7 +278,7 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card) mutex_lock(&card->dapm_mutex); list_for_each_entry(w, &card->widgets, list) { - if (w->is_ep) { + if (w->is_ep && !w->ignore_suspend) { dapm_mark_dirty(w, "Rechecking endpoints"); if (w->is_ep & SND_SOC_DAPM_EP_SINK) dapm_widget_invalidate_output_paths(w); -- 2.19.0

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[alsa-devel] testing hardware with alsaloop.
by tom＠hitechsys.com 21 Jan '19

21 Jan '19

Hello Im running alsaloop test to test the audio on our hardware. Im specifically running this command - alsaloop -C hw:0,0 -P hw:1,0 -S 1 -l 480 -c 6 -v When I measure the delay through my hardware I get 11mSec - which is fine, and the audio quality is good as well. However after several hours of running this loop test the delay has increased to 90mSec. Can someone advice what is going wrong here please. Best regards Tom Favell | Hi Tech Systems Ltd http://www.hitechsys.com For support go to https://support.hitechsys.com/ T: +44 (0) 1256 780880 M: +44 (0) 79797 55359 http://www.hitechsys.com © 2019 Hi Tech Systems Ltd. Holbrook House, Oakley Lane, Basingstoke, RG23 7JY, UK Registered in England No. 2130346 | VAT No. GB 450390370 The information contained in this e-mail is confidential and is intended solely for the use of the named addressee. Access, copying or re-use of the information in it by any other person is not authorized. The Internet is not a secure transmission medium and therefore the content of this e-mail may have been changed without the consent of the originator. If you are not the intended recipient, please destroy this email and contact the sender.http://www.hitechsys.com

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[alsa-devel] [PATCH] ALSA: hda - Add mute LED support for HP ProBook 470 G5
by Anthony Wong 21 Jan '19

21 Jan '19

From: Anthony Wong <anthony.wong(a)canonical.com> Support speaker and mic mute LEDs on HP ProBook 470 G5. BugLink: https://bugs.launchpad.net/bugs/1811254 Signed-off-by: Anthony Wong <anthony.wong(a)canonical.com> Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 88ce2f1..8b5071b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -966,6 +966,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), -- 2.7.4

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[alsa-devel] [PATCH 0/9] ALSA: fireface: add support for Fireface UCX
by Takashi Sakamoto 21 Jan '19

21 Jan '19

Hi, In 2012, RME GmbH shipped Fireface UFX. This model is one of latter models of Fireface series and support both of IEEE 1394 bus and USB. Furthermore, it supports two types of remote control unit (Basic Remote and Advanced Remote Control) with by 9pin mini-din connector. This patchset adds support for this model, as a part of devices by ALSA firewire stack (therefore it's not a device of USB). At present, maintain of isochronous communication and synchronization status are cleared but asynchronous communication for MIDI messages are not yet, therefore this commit adds support for isochronous communication for PCM frames. This work is based on my packet analysis from rent device for a short term (2 weeks). I'm happy to get your responses till next weekend. Driver code in a remote branch is available for backport till Linux kernel v4.17 or later for your test. https://github.com/takaswie/snd-firewire-improve/tree/topic/ff800-midi Regards Takashi Sakamoto (9): ALSA: fireface: rename protocol layer for former models ALSA: fireface: unify protocol layer for FF400/FF800 ALSA: fireface: obsolete proc node to leave one node ALSA: fireface: add protocol-dependent operation to dump status ALSA: fireface: add protocol-dependent operation to switch mode to fetch PCM frame ALSA: fireface: add protocol-dependent operation to get clock status ALSA: fireface: code refactoring for dump of sync status ALSA: fireface: code refactoring to parse of clock configuration ALSA: fireface: add support for Fireface UCX sound/firewire/Kconfig | 1 + sound/firewire/fireface/Makefile | 4 +- sound/firewire/fireface/ff-pcm.c | 2 +- sound/firewire/fireface/ff-proc.c | 217 +------- sound/firewire/fireface/ff-protocol-ff400.c | 161 ------ sound/firewire/fireface/ff-protocol-ff800.c | 143 ----- sound/firewire/fireface/ff-protocol-former.c | 549 +++++++++++++++++++ sound/firewire/fireface/ff-protocol-latter.c | 273 +++++++++ sound/firewire/fireface/ff-stream.c | 40 +- sound/firewire/fireface/ff-transaction.c | 63 --- sound/firewire/fireface/ff.c | 41 +- sound/firewire/fireface/ff.h | 13 +- 12 files changed, 886 insertions(+), 621 deletions(-) delete mode 100644 sound/firewire/fireface/ff-protocol-ff400.c delete mode 100644 sound/firewire/fireface/ff-protocol-ff800.c create mode 100644 sound/firewire/fireface/ff-protocol-former.c create mode 100644 sound/firewire/fireface/ff-protocol-latter.c -- 2.19.1

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[alsa-devel] [PATCH v1] ASoC: Intel: Skylake: Switch to modern UUID API
by Andy Shevchenko 21 Jan '19

21 Jan '19

Switch the driver to use modern UUID API, i.e. guid_t type and accompanying functions, such as guid_equal(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/skylake/skl-pcm.c | 12 ++++++------ sound/soc/intel/skylake/skl-sst-dsp.h | 6 +++--- sound/soc/intel/skylake/skl-sst-utils.c | 23 +++++++---------------- sound/soc/intel/skylake/skl-sst.c | 4 ++-- sound/soc/intel/skylake/skl-topology.c | 24 ++++++++++++------------ sound/soc/intel/skylake/skl-topology.h | 6 +++--- 6 files changed, 33 insertions(+), 42 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 823e39103edd..86f8ebc77a32 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1268,12 +1268,12 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) { struct skl_sst *ctx = skl->skl_sst; struct skl_module_inst_id *pin_id; - uuid_le *uuid_mod, *uuid_tplg; + guid_t *uuid_mod, *uuid_tplg; struct skl_module *skl_module; struct uuid_module *module; int i, ret = -EIO; - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; if (list_empty(&ctx->uuid_list)) { dev_err(ctx->dev, "Module list is empty\n"); @@ -1281,7 +1281,7 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) } list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { mconfig->id.module_id = module->id; if (mconfig->module) mconfig->module->loadable = module->is_loadable; @@ -1298,7 +1298,7 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) for (i = 0; i < skl->nr_modules; i++) { skl_module = skl->modules[i]; uuid_tplg = &skl_module->uuid; - if (!uuid_le_cmp(*uuid_mod, *uuid_tplg)) { + if (guid_equal(uuid_mod, uuid_tplg)) { mconfig->module = skl_module; ret = 0; break; @@ -1310,13 +1310,13 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) list_for_each_entry(module, &ctx->uuid_list, list) { for (i = 0; i < MAX_IN_QUEUE; i++) { pin_id = &mconfig->m_in_pin[i].id; - if (!uuid_le_cmp(pin_id->mod_uuid, module->uuid)) + if (guid_equal(&pin_id->mod_uuid, &module->uuid)) pin_id->module_id = module->id; } for (i = 0; i < MAX_OUT_QUEUE; i++) { pin_id = &mconfig->m_out_pin[i].id; - if (!uuid_le_cmp(pin_id->mod_uuid, module->uuid)) + if (guid_equal(&pin_id->mod_uuid, &module->uuid)) pin_id->module_id = module->id; } } diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h index e1d6f6719f7e..cbc7a93d56c2 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -177,7 +177,7 @@ struct skl_dsp_loader_ops { #define MAX_INSTANCE_BUFF 2 struct uuid_module { - uuid_le uuid; + guid_t uuid; int id; int is_loadable; int max_instance; @@ -241,8 +241,8 @@ void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx); int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, unsigned int offset, int index); -int skl_get_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int instance_id); -int skl_put_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int *pvt_id); +int skl_get_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int instance_id); +int skl_put_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int *pvt_id); int skl_get_pvt_instance_id_map(struct skl_sst *ctx, int module_id, int instance_id); void skl_freeup_uuid_list(struct skl_sst *ctx); diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c index 2ae405617876..85551321c35b 100644 --- a/sound/soc/intel/skylake/skl-sst-utils.c +++ b/sound/soc/intel/skylake/skl-sst-utils.c @@ -21,17 +21,11 @@ #include "../common/sst-dsp-priv.h" #include "skl-sst-ipc.h" - -#define UUID_STR_SIZE 37 #define DEFAULT_HASH_SHA256_LEN 32 /* FW Extended Manifest Header id = $AE1 */ #define SKL_EXT_MANIFEST_HEADER_MAGIC 0x31454124 -struct UUID { - u8 id[16]; -}; - union seg_flags { u32 ul; struct { @@ -65,7 +59,7 @@ struct module_type { struct adsp_module_entry { u32 struct_id; u8 name[8]; - struct UUID uuid; + u8 uuid[16]; struct module_type type; u8 hash1[DEFAULT_HASH_SHA256_LEN]; u32 entry_point; @@ -184,13 +178,13 @@ static inline int skl_pvtid_128(struct uuid_module *module) * This generates a 128 bit private unique id for a module TYPE so that * module instance is unique */ -int skl_get_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int instance_id) +int skl_get_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int instance_id) { struct uuid_module *module; int pvt_id; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { pvt_id = skl_pvtid_128(module); if (pvt_id >= 0) { @@ -214,13 +208,13 @@ EXPORT_SYMBOL_GPL(skl_get_pvt_id); * * This frees a 128 bit private unique id previously generated */ -int skl_put_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int *pvt_id) +int skl_put_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int *pvt_id) { int i; struct uuid_module *module; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { if (*pvt_id != 0) i = (*pvt_id) / 64; @@ -247,7 +241,6 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, struct adsp_fw_hdr *adsp_hdr; struct adsp_module_entry *mod_entry; int i, num_entry, size; - uuid_le *uuid_bin; const char *buf; struct skl_sst *skl = ctx->thread_context; struct uuid_module *module; @@ -279,8 +272,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, return -EINVAL; } - mod_entry = (struct adsp_module_entry *) - (buf + offset + adsp_hdr->len); + mod_entry = (struct adsp_module_entry *)(buf + offset + adsp_hdr->len); num_entry = adsp_hdr->num_modules; @@ -307,8 +299,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, goto free_uuid_list; } - uuid_bin = (uuid_le *)mod_entry->uuid.id; - memcpy(&module->uuid, uuid_bin, sizeof(module->uuid)); + guid_copy(&module->uuid, (guid_t *)&mod_entry->uuid); module->id = (i | (index << 12)); module->is_loadable = mod_entry->type.load_type; diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 5951bbdf1f1a..92806119a755 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -420,9 +420,9 @@ static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid) struct skl_module_table *module_entry = NULL; int ret = 0; char mod_name[64]; /* guid str = 32 chars + 4 hyphens */ - uuid_le *uuid_mod; + guid_t *uuid_mod; - uuid_mod = (uuid_le *)guid; + uuid_mod = (guid_t *)guid; snprintf(mod_name, sizeof(mod_name), "%s%pUL%s", "intel/dsp_fw_", uuid_mod, ".bin"); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 76dde12cc2bb..160a5c864b3f 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -577,7 +577,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) int ret = 0; list_for_each_entry(w_module, &pipe->w_list, node) { - uuid_le *uuid_mod; + guid_t *uuid_mod; w = w_module->w; mconfig = w->priv; @@ -585,7 +585,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) if (mconfig->id.module_id < 0) { dev_err(skl->skl_sst->dev, "module %pUL id not populated\n", - (uuid_le *)mconfig->guid); + (guid_t *)mconfig->guid); return -EIO; } @@ -619,7 +619,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) * FE/BE params */ skl_tplg_update_module_params(w, ctx); - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; mconfig->id.pvt_id = skl_get_pvt_id(ctx, uuid_mod, mconfig->id.instance_id); if (mconfig->id.pvt_id < 0) @@ -658,9 +658,9 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx, struct skl_module_cfg *mconfig = NULL; list_for_each_entry(w_module, &pipe->w_list, node) { - uuid_le *uuid_mod; + guid_t *uuid_mod; mconfig = w_module->w->priv; - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; if (mconfig->module->loadable && ctx->dsp->fw_ops.unload_mod && mconfig->m_state > SKL_MODULE_UNINIT) { @@ -916,12 +916,12 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w, return 0; } -static int skl_get_module_id(struct skl_sst *ctx, uuid_le *uuid) +static int skl_get_module_id(struct skl_sst *ctx, guid_t *uuid) { struct uuid_module *module; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid, module->uuid) == 0) + if (guid_equal(uuid, &module->uuid)) return module->id; } @@ -2121,11 +2121,11 @@ static int skl_tplg_add_pipe(struct device *dev, return 0; } -static int skl_tplg_get_uuid(struct device *dev, u8 *guid, +static int skl_tplg_get_uuid(struct device *dev, guid_t *guid, struct snd_soc_tplg_vendor_uuid_elem *uuid_tkn) { if (uuid_tkn->token == SKL_TKN_UUID) { - memcpy(guid, &uuid_tkn->uuid, 16); + guid_copy(guid, (guid_t *)&uuid_tkn->uuid); return 0; } @@ -2151,7 +2151,7 @@ static int skl_tplg_fill_pin(struct device *dev, break; case SKL_TKN_UUID: - ret = skl_tplg_get_uuid(dev, m_pin[pin_index].id.mod_uuid.b, + ret = skl_tplg_get_uuid(dev, &m_pin[pin_index].id.mod_uuid, (struct snd_soc_tplg_vendor_uuid_elem *)tkn_elem); if (ret < 0) return ret; @@ -2666,7 +2666,7 @@ static int skl_tplg_get_tokens(struct device *dev, case SND_SOC_TPLG_TUPLE_TYPE_UUID: if (is_module_guid) { - ret = skl_tplg_get_uuid(dev, mconfig->guid, + ret = skl_tplg_get_uuid(dev, (guid_t *)mconfig->guid, array->uuid); is_module_guid = false; } else { @@ -3484,7 +3484,7 @@ static int skl_tplg_get_manifest_uuid(struct device *dev, if (uuid_tkn->token == SKL_TKN_UUID) { mod = skl->modules[ref_count]; - memcpy(&mod->uuid, &uuid_tkn->uuid, sizeof(uuid_tkn->uuid)); + guid_copy(&mod->uuid, (guid_t *)&uuid_tkn->uuid); ref_count++; } else { dev_err(dev, "Not an UUID token tkn %d\n", uuid_tkn->token); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 82282cac9751..9311e248b6f9 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -224,7 +224,7 @@ struct skl_mod_inst_map { struct skl_uuid_inst_map { u16 inst_id; u16 reserved; - uuid_le mod_uuid; + guid_t mod_uuid; } __packed; struct skl_kpb_params { @@ -236,7 +236,7 @@ struct skl_kpb_params { }; struct skl_module_inst_id { - uuid_le mod_uuid; + guid_t mod_uuid; int module_id; u32 instance_id; int pvt_id; @@ -369,7 +369,7 @@ struct skl_module_res { }; struct skl_module { - uuid_le uuid; + guid_t uuid; u8 loadable; u8 input_pin_type; u8 output_pin_type; -- 2.18.0

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