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

[alsa-devel] [PATCH] ALSA: hda/realtek - Add quirks for Dell Optiplex 5060, 7060 for audio out while using headphones
by Girija Kumar Kasinadhuni 12 Dec '18

12 Dec '18

There are two audio ports on these machines - headset and mic. There' s no sound when hp is inserted into the headset port, though the hp is detected. Without pulseaudio, this issue is seen on both the Optiplex machines. The issue goes away with the "options snd-hda-intel model=dell-headset-dock" quirk in the /etc/modprobe.d/alsa-base.conf or /etc/modprobe.d/alsa.conf Getting this fix into the kernel instead of fixing in an application layer above, or in a .conf file. This is also helpful for audio clients using the ALSA api like cras from ChromiumOS. Signed-off-by: Girija Kumar Kasinadhuni <gkumar(a)neverware.com> --- alsa-info: Optiplex 5060 - - without fix: http://www.alsa-project.org/db/?f=2cf3c4354e82bde4aa480612a914535982fa3f91 - with fix: http://www.alsa-project.org/db/?f=7a928960400ec828b82632411284ad8a63727b92 Optiplex 7060 - without fix: http://www.alsa-project.org/db/?f=69fa554faa712e7cad1f188be4a5192353d64f2c - with fix: http://www.alsa-project.org/db/?f=80ce1fb178b377b27b0795ac1c55e8cbee1b5b96 sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8d75597028ee..581f6a288abf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6444,6 +6444,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", ALC288_FIXUP_DELL_XPS_13), SND_PCI_QUIRK(0x1028, 0x0669, "Dell Optiplex 9020m", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x085b, "Dell Optiplex 5060", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x085a, "Dell Optiplex 7060", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x069a, "Dell Vostro 5480", ALC290_FIXUP_SUBWOOFER_HSJACK), SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- 2.17.1

2 1

[alsa-devel] RME Fireface alsa driver
by thet 11 Dec '18

11 Dec '18

I posted this a couple of weeks ago in the alsa users list but there was no response. I hope someone here can help. I have a FF400 and I have been using ffado with it. It came to my attention that there exists a ALSA driver for the FF400 since the midi was working over alsa and midi is not supported in the ffado driver. Also my system can see the FF400 in the sound control panel without jack or ffado running. I am using Mint 19 cinnamon and the liquorix kernel. (though so far I have not noticed any difference between the liquorix kernel and the generic one except pre-empt performance). I would like to try using just the alsa driver and not ffado. Since I'm using the alsa midi support anyway it would be less total software to use the alsa driver for everything - assuming it supports everything I need for serious music recording. However I can't get any sound out of the FF400 using the ALSA driver - it works ok with FFADO though. Also the midi works - presumably using the alsa driver. I have tried using alsa directly in the sound control panel, using the alsa output from audacious, and the alsa driver from jack. I get an error about setting the frames/period/buffer size. The red light on the FF400 that shows no connection to a host stays on - even though the system can see it is present. This light goes off if I start ffado and jack, or if I start ffado mixer. Is there a guide somewhere on how to set up the ALSA drivers for the FF400? thanks! Thet

2 1

[alsa-devel] [PATCH 0/5 v2] ASoC: merge simple-scu-card into simple-card - step1
by Kuninori Morimoto 11 Dec '18

11 Dec '18

Hi Mark These are based on latest "for-4.21" branch. Kuninori Morimoto (5): ASoC: simple-scu-card: add dai-link support ASoC: simple-scu-card: care link / dai count ASoC: simple-scu-card: tidyup "convert-rate/channels" parsing ASoC: simple-scu-card: use cpu/codec pointer on graph_dai_props ASoC: simple-scu-card: care multi DPCM codec_conf sound/soc/generic/simple-scu-card.c | 240 +++++++++++++++++++++++++++--------- 1 file changed, 185 insertions(+), 55 deletions(-) -- 2.7.4 Best regards --- Kuninori Morimoto

2 6

[alsa-devel] Applied "ASoC: simple-scu-card: care link / dai count" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: simple-scu-card: care link / dai count 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 b8d8f9a58fd971ec215fb8bfec5a4fb6e539a3ab Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 11 Dec 2018 03:25:08 +0000 Subject: [PATCH] ASoC: simple-scu-card: care link / dai count In DPCM case, it uses CPU-dummy / dummy-Codec dai links. If sound card is caring only DPCM, link count = dai count, but, if non DPCM case, link count != dai count. Now, we want to merge simple-card and simple-scu-card, then, we need to care both link / dai count more carefly This patch cares it, and prepare for merging simple card Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 90 ++++++++++++++++++++++++++--- 1 file changed, 83 insertions(+), 7 deletions(-) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 666ace6175be..1c6382fa54a0 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -256,6 +256,80 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) return 0; } +static void asoc_simple_card_get_dais_count(struct device *dev, + int *link_num, + int *dais_num, + int *ccnf_num) +{ + struct device_node *top = dev->of_node; + struct device_node *node; + int loop; + int num; + + /* + * link_num : number of links. + * CPU-Codec / CPU-dummy / dummy-Codec + * dais_num : number of DAIs + * ccnf_num : number of codec_conf + * same number for "dummy-Codec" + * + * ex1) + * CPU0 --- Codec0 link : 5 + * CPU1 --- Codec1 dais : 7 + * CPU2 -/ ccnf : 1 + * CPU3 --- Codec2 + * + * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec + * => 7 DAIs = 4xCPU + 3xCodec + * => 1 ccnf = 1xdummy-Codec + * + * ex2) + * CPU0 --- Codec0 link : 5 + * CPU1 --- Codec1 dais : 6 + * CPU2 -/ ccnf : 1 + * CPU3 -/ + * + * => 5 links = 1xCPU-Codec + 3xCPU-dummy + 1xdummy-Codec + * => 6 DAIs = 4xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec + * + * ex3) + * CPU0 --- Codec0 link : 6 + * CPU1 -/ dais : 6 + * CPU2 --- Codec1 ccnf : 2 + * CPU3 -/ + * + * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec + * => 6 DAIs = 4xCPU + 2xCodec + * => 2 ccnf = 2xdummy-Codec + */ + if (!top) { + (*link_num) = 1; + (*dais_num) = 2; + (*ccnf_num) = 0; + return; + } + + loop = 1; + node = of_get_child_by_name(top, PREFIX "dai-link"); + if (!node) { + node = top; + loop = 0; + } + + do { + num = of_get_child_count(node); + (*dais_num) += num; + if (num > 2) { + (*link_num) += num; + (*ccnf_num)++; + } else { + (*link_num)++; + } + node = of_get_next_child(top, node); + } while (loop && node); +} + static int asoc_simple_card_probe(struct platform_device *pdev) { struct simple_card_data *priv; @@ -263,18 +337,20 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct simple_dai_props *dai_props; struct snd_soc_card *card; struct device *dev = &pdev->dev; - struct device_node *np = dev->of_node; - int num, ret, i; + int ret, i; + int lnum = 0, dnum = 0, cnum = 0; /* Allocate the private data */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; - num = of_get_child_count(np); + asoc_simple_card_get_dais_count(dev, &lnum, &dnum, &cnum); + if (!lnum || !dnum) + return -EINVAL; - dai_props = devm_kcalloc(dev, num, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, num, sizeof(*dai_link), GFP_KERNEL); + dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); + dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); if (!dai_props || !dai_link) return -ENOMEM; @@ -284,7 +360,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) * see * soc-core.c :: snd_soc_init_multicodec() */ - for (i = 0; i < num; i++) { + for (i = 0; i < lnum; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; dai_link[i].platform = &dai_props[i].platform; @@ -298,7 +374,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) card->owner = THIS_MODULE; card->dev = dev; card->dai_link = priv->dai_link; - card->num_links = num; + card->num_links = lnum; card->codec_conf = &priv->codec_conf; card->num_configs = 1; -- 2.19.0.rc2

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[alsa-devel] Applied "ASoC: simple-scu-card: tidyup "convert-rate/channels" parsing" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: simple-scu-card: tidyup "convert-rate/channels" parsing 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 09ee833679761f4995daaa6fd6200c77b8fcdaef Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 11 Dec 2018 03:25:18 +0000 Subject: [PATCH] ASoC: simple-scu-card: tidyup "convert-rate/channels" parsing simple-scu-card.c is supporting "convert-rate/channels" which is used for DPCM. But, sound card might have multi codecs, and each codec might need each convert-rate/channels. This patch supports each codec's convert-rate/channles support. top node convert-rate/channels will overwrite settings if exist. It can't support each codec's convert-rate/channels if sound card had multi codecs without this patch. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 1c6382fa54a0..3c4e28995872 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -26,6 +26,7 @@ struct simple_card_data { struct asoc_simple_dai dai; struct snd_soc_dai_link_component codecs; struct snd_soc_dai_link_component platform; + struct asoc_simple_card_data adata; } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_card_data adata; @@ -86,7 +87,11 @@ static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + asoc_simple_card_convert_fixup(&dai_props->adata, params); + + /* overwrite by top level adata if exist */ asoc_simple_card_convert_fixup(&priv->adata, params); return 0; @@ -177,6 +182,8 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, "prefix"); } + asoc_simple_card_parse_convert(dev, link, prefix, &dai_props->adata); + ret = asoc_simple_card_of_parse_tdm(np, &dai_props->dai); if (ret) return ret; -- 2.19.0.rc2

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[alsa-devel] Applied "ASoC: simple-scu-card: use cpu/codec pointer on graph_dai_props" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: simple-scu-card: use cpu/codec pointer on graph_dai_props 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 e6a3ff2ed4c9c4df217c64e0fc0a280a24055790 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 11 Dec 2018 03:25:27 +0000 Subject: [PATCH] ASoC: simple-scu-card: use cpu/codec pointer on graph_dai_props In DPCM case, it uses CPU-dummy / dummy-Codec dai links, and non DPCM case, it uses CPU-Codec dai links. Now, we want to merge simple-card and simple-scu-card. These sound cards are using silimar but not same logic on each functions. Then, of course we want to share same logic. To compromise, this patch uses cpu/codec pointer on simple-scu-card. It is same logic with simple-card, thus easy merging. This is prepare for merging simple card Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 77 +++++++++++++++++++---------- 1 file changed, 52 insertions(+), 25 deletions(-) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 3c4e28995872..944a6f435f88 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -23,12 +23,14 @@ struct simple_card_data { struct snd_soc_card snd_card; struct snd_soc_codec_conf codec_conf; struct simple_dai_props { - struct asoc_simple_dai dai; + struct asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; struct snd_soc_dai_link_component platform; struct asoc_simple_card_data adata; } *dai_props; struct snd_soc_dai_link *dai_link; + struct asoc_simple_dai *dais; struct asoc_simple_card_data adata; }; @@ -47,8 +49,17 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream) struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + int ret; + + ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); + if (ret) + return ret; + + ret = asoc_simple_card_clk_enable(dai_props->codec_dai); + if (ret) + asoc_simple_card_clk_disable(dai_props->cpu_dai); - return asoc_simple_card_clk_enable(&dai_props->dai); + return ret; } static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) @@ -58,7 +69,9 @@ static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - asoc_simple_card_clk_disable(&dai_props->dai); + asoc_simple_card_clk_disable(dai_props->cpu_dai); + + asoc_simple_card_clk_disable(dai_props->codec_dai); } static const struct snd_soc_ops asoc_simple_card_ops = { @@ -68,19 +81,21 @@ static const struct snd_soc_ops asoc_simple_card_ops = { static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai; - struct snd_soc_dai_link *dai_link; - struct simple_dai_props *dai_props; - int num = rtd->num; + struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + int ret; - dai_link = simple_priv_to_link(priv, num); - dai_props = simple_priv_to_props(priv, num); - dai = dai_link->dynamic ? - rtd->cpu_dai : - rtd->codec_dai; + ret = asoc_simple_card_init_dai(rtd->codec_dai, + dai_props->codec_dai); + if (ret < 0) + return ret; - return asoc_simple_card_init_dai(dai, &dai_props->dai); + ret = asoc_simple_card_init_dai(rtd->cpu_dai, + dai_props->cpu_dai); + if (ret < 0) + return ret; + + return 0; } static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -101,13 +116,14 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, struct device_node *np, struct device_node *codec, struct simple_card_data *priv, - int idx, bool is_fe, + int *dai_idx, int link_idx, int is_fe, bool is_top_level_node) { struct device *dev = simple_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx); + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, link_idx); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, link_idx); struct snd_soc_card *card = simple_priv_to_card(priv); + struct asoc_simple_dai *dai; char *prefix = ""; int ret; @@ -129,12 +145,15 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; + dai = + dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + ret = asoc_simple_card_parse_cpu(np, dai_link, DAI, CELL, &is_single_links); if (ret) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, &dai_props->dai); + ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, dai); if (ret < 0) return ret; @@ -155,11 +174,14 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_simple_card_be_hw_params_fixup; + dai = + dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + ret = asoc_simple_card_parse_codec(np, dai_link, DAI, CELL); if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, &dai_props->dai); + ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, dai); if (ret < 0) return ret; @@ -184,7 +206,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, asoc_simple_card_parse_convert(dev, link, prefix, &dai_props->adata); - ret = asoc_simple_card_of_parse_tdm(np, &dai_props->dai); + ret = asoc_simple_card_of_parse_tdm(np, dai); if (ret) return ret; @@ -215,7 +237,8 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) struct device_node *codec; struct snd_soc_card *card = simple_priv_to_card(priv); bool is_fe; - int ret, i, loop; + int ret, loop; + int dai_idx, link_idx; if (!top) return -EINVAL; @@ -231,7 +254,8 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) asoc_simple_card_parse_convert(dev, top, PREFIX, &priv->adata); loop = 1; - i = 0; + link_idx = 0; + dai_idx = 0; node = of_get_child_by_name(top, PREFIX "dai-link"); if (!node) { node = dev->of_node; @@ -248,10 +272,10 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) is_fe = (np != codec); ret = asoc_simple_card_dai_link_of(node, np, codec, priv, - i, is_fe, !loop); + &dai_idx, link_idx++, + is_fe, !loop); if (ret < 0) return ret; - i++; } node = of_get_next_child(top, node); } while (loop && node); @@ -342,6 +366,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct simple_card_data *priv; struct snd_soc_dai_link *dai_link; struct simple_dai_props *dai_props; + struct asoc_simple_dai *dais; struct snd_soc_card *card; struct device *dev = &pdev->dev; int ret, i; @@ -358,7 +383,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); - if (!dai_props || !dai_link) + dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); + if (!dai_props || !dai_link || !dais) return -ENOMEM; /* @@ -375,6 +401,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) priv->dai_props = dai_props; priv->dai_link = dai_link; + priv->dais = dais; /* Init snd_soc_card */ card = simple_priv_to_card(priv); -- 2.19.0.rc2

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[alsa-devel] Applied "ASoC: simple-scu-card: care multi DPCM codec_conf" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: simple-scu-card: care multi DPCM codec_conf 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 0a26d6df4fd8aa8f4fb444446d162e01c3e349c0 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 11 Dec 2018 03:25:37 +0000 Subject: [PATCH] ASoC: simple-scu-card: care multi DPCM codec_conf Current simple-scu-card didn't care about codec_conf for multi DPCM case. This patch cares it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 30 +++++++++++++++++++---------- 1 file changed, 20 insertions(+), 10 deletions(-) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 944a6f435f88..9d7299d536a8 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -21,17 +21,18 @@ struct simple_card_data { struct snd_soc_card snd_card; - struct snd_soc_codec_conf codec_conf; struct simple_dai_props { struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; struct snd_soc_dai_link_component platform; struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_dai *dais; struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; }; #define simple_priv_to_card(priv) (&(priv)->snd_card) @@ -116,7 +117,8 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, struct device_node *np, struct device_node *codec, struct simple_card_data *priv, - int *dai_idx, int link_idx, int is_fe, + int *dai_idx, int link_idx, + int *conf_idx, int is_fe, bool is_top_level_node) { struct device *dev = simple_priv_to_dev(priv); @@ -165,6 +167,8 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, asoc_simple_card_canonicalize_cpu(dai_link, is_single_links); } else { + struct snd_soc_codec_conf *cconf; + /* FE is dummy */ dai_link->cpu_of_node = NULL; dai_link->cpu_dai_name = "snd-soc-dummy-dai"; @@ -177,6 +181,9 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, dai = dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + cconf = + dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; + ret = asoc_simple_card_parse_codec(np, dai_link, DAI, CELL); if (ret < 0) return ret; @@ -192,14 +199,12 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, return ret; /* check "prefix" from top node */ - snd_soc_of_parse_audio_prefix(card, - &priv->codec_conf, + snd_soc_of_parse_audio_prefix(card, cconf, dai_link->codecs->of_node, PREFIX "prefix"); /* check "prefix" from each node if top doesn't have */ - if (!priv->codec_conf.of_node) - snd_soc_of_parse_node_prefix(np, - &priv->codec_conf, + if (!cconf->of_node) + snd_soc_of_parse_node_prefix(np, cconf, dai_link->codecs->of_node, "prefix"); } @@ -238,7 +243,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) struct snd_soc_card *card = simple_priv_to_card(priv); bool is_fe; int ret, loop; - int dai_idx, link_idx; + int dai_idx, link_idx, conf_idx; if (!top) return -EINVAL; @@ -256,6 +261,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) loop = 1; link_idx = 0; dai_idx = 0; + conf_idx = 0; node = of_get_child_by_name(top, PREFIX "dai-link"); if (!node) { node = dev->of_node; @@ -273,6 +279,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) ret = asoc_simple_card_dai_link_of(node, np, codec, priv, &dai_idx, link_idx++, + &conf_idx, is_fe, !loop); if (ret < 0) return ret; @@ -368,6 +375,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct simple_dai_props *dai_props; struct asoc_simple_dai *dais; struct snd_soc_card *card; + struct snd_soc_codec_conf *cconf; struct device *dev = &pdev->dev; int ret, i; int lnum = 0, dnum = 0, cnum = 0; @@ -384,6 +392,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); + cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); if (!dai_props || !dai_link || !dais) return -ENOMEM; @@ -402,6 +411,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) priv->dai_props = dai_props; priv->dai_link = dai_link; priv->dais = dais; + priv->codec_conf = cconf; /* Init snd_soc_card */ card = simple_priv_to_card(priv); @@ -409,8 +419,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) card->dev = dev; card->dai_link = priv->dai_link; card->num_links = lnum; - card->codec_conf = &priv->codec_conf; - card->num_configs = 1; + card->codec_conf = cconf; + card->num_configs = cnum; ret = asoc_simple_card_parse_of(priv); if (ret < 0) { -- 2.19.0.rc2

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[alsa-devel] [PATCH 0/7] ALSA: HD-audio display power fixes
by Takashi Iwai 11 Dec '18

11 Dec '18

Hi, this patch is about fixing possible HD-audio display power unbalance. Basically the "fix" is done by refactoring the whole relevant code. It starts from the Intel HD-audio runtime PM refactoring, followed by the display PM API change, and lots of code cleanups. Since it changes the display power API function, it hits both legacy and ASoC drivers. The patches are kept in topic/hda-pm-refactor branch of my sound git tree. This will be an immutable branch once when merged to for-next, so that it can be merged to ASoC tree if any conflicting change needs to be applied. Takashi === Takashi Iwai (7): ALSA: hda/intel: Refactoring PM code ALSA: hda: Refactor display power management ALSA: hda/intel: Drop superfluous AZX_DCAPS_I915_POWERWELL checks ALSA: hda/intel: Properly free the display power at error path ALSA: hda: Make snd_hdac_display_power() void function ASoC: hdac_hdmi: Add missing display power-off at driver removal ALSA: hda/hdmi: Always set display_power_control for Intel HSW+ codecs include/sound/hda_codec.h | 1 + include/sound/hda_component.h | 9 +- include/sound/hdaudio.h | 7 +- sound/hda/hdac_component.c | 39 ++++--- sound/hda/hdac_device.c | 17 --- sound/pci/hda/hda_codec.c | 16 ++- sound/pci/hda/hda_controller.c | 11 -- sound/pci/hda/hda_intel.c | 192 +++++++++++++-------------------- sound/pci/hda/patch_hdmi.c | 8 +- sound/soc/codecs/hdac_hdmi.c | 22 ++-- sound/soc/intel/skylake/skl.c | 40 ++----- 11 files changed, 134 insertions(+), 228 deletions(-) -- 2.19.2

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[alsa-devel] Realtek ALC294 doesn't work in 4.20-rc6
by Anthony Ettinger 11 Dec '18

11 Dec '18

Does anyone know when a fix will be applied? This is affecting Asus Zenbook UX433F -- Anthony Ettinger http://linkedin.com/in/chovy

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[alsa-devel] [PATCH 0/8] ALSA: fireface: add support for Fireface 800 with MIDI functionality only
by Takashi Sakamoto 11 Dec '18

11 Dec '18

Hi, In 2004, RME GmbH shipped Fireface 800 as a flagship model of its Fireface series. This model consists of some chips: - TI TSB81BA3D for physical layer on cable environment of EEE 1394 bus - TI TSB82AA2 for link layer for 1394 OHCI bus bridge to PCI bus - Xilinx Spartan-3 FPGA XC3S400 - Xilinx High-Performance CPLD XC9572XL A remarkable point of this model is that its physical/link layer support S800 (800Mbps) mode of IEEE 1394b. Within devices of audio and music unit on IEEE 1394 bus, this is an unique model to support it, as long as I know. This commit adds support for the model with its MIDI functionality only. The functionality is quite similar to the one of Fireface 400. In patch 01-06, some codes specific to Fireface 400 are arranged to be shared by both models. On the other hand, there're some differences. The largest difference is that the model has no register to switch enabling/disabling tx asynchronous transactions for MIDI messages. Therefore the functionality work well without any assist by userspace. For this work, I use v2.77 firmware and v3.1.24 driver shipped by RME GmbH. If finding some differences between older firmware, please report it with enough information for debug. To help users for test, I prepare a remote branch including these patches. This can be backport to Linux v4.17 or later. https://github.com/takaswie/snd-firewire-improve/tree/topic/ff800-midi Takashi Sakamoto (8): ALSA: fireface: share some registers for status of clock synchronization ALSA: fireface: share status and configuration dump ALSA: fireface: share helper function to get current sampling rate and clock source ALSA: fireface: add support for second optical interface for ADAT stream ALSA: fireface: share register for async transaction of MIDI messages ALSA: fireface: add driver data for register for MIDI high address ALSA: fireface: localize a handler for MIDI messages on tx transaction ALSA: fireface: add support for Fireface 800 with MIDI functionality only sound/firewire/Kconfig | 1 + sound/firewire/fireface/Makefile | 3 +- sound/firewire/fireface/ff-pcm.c | 2 +- sound/firewire/fireface/ff-proc.c | 193 +++++++++++++- sound/firewire/fireface/ff-protocol-ff400.c | 282 ++------------------ sound/firewire/fireface/ff-protocol-ff800.c | 27 ++ sound/firewire/fireface/ff-stream.c | 2 +- sound/firewire/fireface/ff-transaction.c | 157 ++++++----- sound/firewire/fireface/ff.c | 54 +++- sound/firewire/fireface/ff.h | 31 ++- 10 files changed, 406 insertions(+), 346 deletions(-) create mode 100644 sound/firewire/fireface/ff-protocol-ff800.c -- 2.19.1

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