January 2014 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH] ASoC: simple-card: fix one bug to writing to the platform data
by Jean-Francois Moine 13 Jan '14

13 Jan '14

Xiubo Li wrote: > It's a bug that writing to the platform data directly, for it should > be constant. So just copy it before writing. > > Signed-off-by: Xiubo Li <Li.Xiubo at freescale.com> > --- > sound/soc/generic/simple-card.c | 40 +++++++++++++++++++++------------------- > 1 file changed, 21 insertions(+), 19 deletions(-) > > diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c > index 406e4ea..5b65324 100644 > --- a/sound/soc/generic/simple-card.c > +++ b/sound/soc/generic/simple-card.c [snip] > @@ -204,36 +205,37 @@ static int asoc_simple_card_probe(struct platform_device *pdev) > struct device_node *np = pdev->dev.of_node; > struct device_node *of_cpu, *of_codec, *of_platform; > struct device *dev = &pdev->dev; > + int ret; > > cinfo = NULL; > of_cpu = NULL; > of_codec = NULL; > of_platform = NULL; > + > + cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); > + if (!cinfo) > + return -ENOMEM; > + > if (np && of_device_is_available(np)) { > - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); > - if (cinfo) { > - int ret; > - cinfo->snd_card.dev = &pdev->dev; > - ret = asoc_simple_card_parse_of(np, cinfo, dev, > - &of_cpu, > - &of_codec, > - &of_platform); > - if (ret < 0) { > - if (ret != -EPROBE_DEFER) > - dev_err(dev, "parse error %d\n", ret); > - return ret; > - } > - } else { > - return -ENOMEM; > + cinfo->snd_card.dev = dev; > + > + ret = asoc_simple_card_parse_of(np, cinfo, dev, > + &of_cpu, > + &of_codec, > + &of_platform); > + if (ret < 0) { > + if (ret != -EPROBE_DEFER) > + dev_err(dev, "parse error %d\n", ret); > + return ret; > } > } else { > - cinfo = pdev->dev.platform_data; > - if (!cinfo) { > + if (!pdev->dev.platform_data) { > dev_err(dev, "no info for asoc-simple-card\n"); > return -EINVAL; > } > > - cinfo->snd_card.dev = &pdev->dev; > + memcpy(cinfo, pdev->dev.platform_data, sizeof(cinfo)); > + cinfo->snd_card.dev = dev; > } > > if (!cinfo->name || If the original cinfo is not used anymore, the use of its structure to handle the card information is not a good idea: - almost all cinfo information are in the struct snd_soc_card, - this cinfo structure cannot be extended to handle many DAI links, - it contains simple-card information which are of no use for the platform caller. So, I'd rather have seen: - the removal of 'snd_link' and 'snd_card' from the platform interface (struct asoc_simple_card_info), - the definition of a local struct simple_card_data containing the struct snd_soc_card and a pointer to an array of fmt/sysclk values (one per DAI link). -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

3 3

[alsa-devel] [PATCH] ASoC: wm8804: Allow control of master clock divider in PLL generation
by Daniel Matuschek 13 Jan '14

13 Jan '14

Signed-off-by: Daniel Matuschek <daniel(a)matuschek.net> After some discussions of the patch last week, here is a new version. Simply reducing the post_table did not work, as for some frequencies both settings (MCLKDIV=0 and MCLKDIV=1) are needed (e.g. 96 and 192kHz) WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using an additional mclk_div divider, is is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. The behavior has to be controlled by the sound card driver, because some sample frequency share the same setting. e.g. 192kHz and 96kHz use 24.576MHz master clock. The only difference is the MCLK divider. --- sound/soc/codecs/wm8804.c | 17 ++++++++++++++--- sound/soc/codecs/wm8804.h | 4 ++++ 2 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 1704b1e..4619bf8 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -63,6 +63,7 @@ struct wm8804_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; + int mclk_div; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -318,7 +319,7 @@ static struct { #define FIXED_PLL_SIZE ((1ULL << 22) * 10) static int pll_factors(struct pll_div *pll_div, unsigned int target, - unsigned int source) + unsigned int source, unsigned int mclk_div) { u64 Kpart; unsigned long int K, Ndiv, Nmod, tmp; @@ -330,7 +331,8 @@ static int pll_factors(struct pll_div *pll_div, unsigned int target, */ for (i = 0; i < ARRAY_SIZE(post_table); i++) { tmp = target * post_table[i].div; - if (tmp >= 90000000 && tmp <= 100000000) { + if ((tmp >= 90000000 && tmp <= 100000000) && + (mclk_div == post_table[i].mclkdiv)) { pll_div->freqmode = post_table[i].freqmode; pll_div->mclkdiv = post_table[i].mclkdiv; target *= post_table[i].div; @@ -387,8 +389,12 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, } else { int ret; struct pll_div pll_div; + struct wm8804_priv *wm8804; - ret = pll_factors(&pll_div, freq_out, freq_in); + wm8804 = snd_soc_codec_get_drvdata(codec); + + ret = pll_factors(&pll_div, freq_out, freq_in, + wm8804->mclk_div); if (ret) return ret; @@ -452,6 +458,7 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { struct snd_soc_codec *codec; + struct wm8804_priv *wm8804; codec = dai->codec; switch (div_id) { @@ -459,6 +466,10 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, snd_soc_update_bits(codec, WM8804_PLL5, 0x30, (div & 0x3) << 4); break; + case WM8804_MCLK_DIV: + wm8804 = snd_soc_codec_get_drvdata(codec); + wm8804->mclk_div = div; + break; default: dev_err(dai->dev, "Unknown clock divider: %d\n", div_id); return -EINVAL; diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index 8ec14f5..e72d4f4 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -57,5 +57,9 @@ #define WM8804_CLKOUT_SRC_OSCCLK 4 #define WM8804_CLKOUT_DIV 1 +#define WM8804_MCLK_DIV 2 + +#define WM8804_MCLKDIV_256FS 0 +#define WM8804_MCLKDIV_128FS 1 #endif /* _WM8804_H */ -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: dpcm: don't do hw_param when BE has done hw_param
by Nenghua Cao 13 Jan '14

13 Jan '14

From: Nenghua Cao <nhcao(a)marvell.com> It fixes the following case: Two FEs connects the same BE; FE1 & BE path has been opened and hw_paramed. At this momment, FE2 & BE path is being opened and hw_paramed. The BE dai will do hw_param again even if it has done hw_param. It is not reasonable. FE1------------>BE FE2-------------^ Signed-off-by: Nenghua Cao <nhcao(a)marvell.com> --- sound/soc/soc-pcm.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 891b9a9..ec07e37 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1339,7 +1339,6 @@ static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) continue; if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE)) continue; -- 1.7.0.4

4 15

[alsa-devel] [PATCH] alsa-restore.rules: refer to correct attr
by Dave Reisner 13 Jan '14

13 Jan '14

$attr{number} in the RUN rule is an empty expansion. This makes sense, because the path doesn't exist -- i.e., it refers to the path: /sys/devices/pci0000:00/foo/bar/sound/card0/controlC0/number Instead, refer to $attr{device/number}, which does exist. Signed-off-by: Dave Reisner <dreisner(a)archlinux.org> --- This seems to have been broken when it was first introduced a little over three years ago (de7c3eff0e). Please CC me on replies as I am not subscribed to the list. alsactl/90-alsa-restore.rules.in | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/alsactl/90-alsa-restore.rules.in b/alsactl/90-alsa-restore.rules.in index 88e12e0..c68119d 100644 --- a/alsactl/90-alsa-restore.rules.in +++ b/alsactl/90-alsa-restore.rules.in @@ -2,7 +2,7 @@ ACTION=="add", SUBSYSTEM=="sound", KERNEL=="controlC*", KERNELS!="card*", GOTO=" GOTO="alsa_restore_end" LABEL="alsa_restore_go" -TEST!="@daemonswitch@", RUN+="@sbindir@/alsactl restore $attr{number}" -TEST=="@daemonswitch@", RUN+="@sbindir@/alsactl nrestore $attr{number}" +TEST!="@daemonswitch@", RUN+="@sbindir@/alsactl restore $attr{device/number}" +TEST=="@daemonswitch@", RUN+="@sbindir@/alsactl nrestore $attr{device/number}" LABEL="alsa_restore_end" -- 1.8.5.2

3 2

[alsa-devel] [PATCH] ASoC: simple-card: remove snd_link and snd_card from struct asoc_xx_info
by Xiubo Li 13 Jan '14

13 Jan '14

Whether the dt is used or not, almost all the simple card information for the DAI link and sound card are initialized in the simple card driver. And for the platform caller, the snd_link and snd_card are of no use, so remove it from struct asoc_simple_card_info, and let them to be the simple card driver's global data. This is also fix one bug about writing to the platform data directly, for it should be constant. And now only one DAI link is supported for simple card. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- This patch maybe not very perfect, such as supporting many DAI links, etc. And there still need followed patches to fulfil these. include/sound/simple_card.h | 4 --- sound/soc/generic/simple-card.c | 54 +++++++++++++++++++++++------------------ 2 files changed, 30 insertions(+), 28 deletions(-) diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 6c74527..e1ac996 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -29,10 +29,6 @@ struct asoc_simple_card_info { unsigned int daifmt; struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; - - /* used in simple-card.c */ - struct snd_soc_dai_link snd_link; - struct snd_soc_card snd_card; }; #endif /* __SIMPLE_CARD_H */ diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index dbd93cc..94dc743 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -14,9 +14,6 @@ #include <linux/module.h> #include <sound/simple_card.h> -#define asoc_simple_get_card_info(p) \ - container_of(p->dai_link, struct asoc_simple_card_info, snd_link) - static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set, unsigned int daifmt) @@ -41,7 +38,8 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *info = asoc_simple_get_card_info(rtd); + struct asoc_simple_card_info *info = + snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; unsigned int daifmt = info->daifmt; @@ -58,6 +56,16 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static struct snd_soc_dai_link simple_dai_link = { + .init = asoc_simple_card_dai_init, +}; + +static struct snd_soc_card simple_snd_card = { + .owner = THIS_MODULE, + .dai_link = &simple_dai_link, + .num_links = 1, +}; + static int asoc_simple_card_sub_parse_of(struct device_node *np, struct asoc_simple_dai *dai, @@ -144,7 +152,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, /* DAPM routes */ if (of_property_read_bool(node, "simple-audio-card,routing")) { - ret = snd_soc_of_parse_audio_routing(&info->snd_card, + ret = snd_soc_of_parse_audio_routing(&simple_snd_card, "simple-audio-card,routing"); if (ret) return ret; @@ -213,7 +221,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); if (cinfo) { int ret; - cinfo->snd_card.dev = &pdev->dev; + simple_snd_card.dev = dev; ret = asoc_simple_card_parse_of(np, cinfo, dev, &of_cpu, &of_codec, @@ -233,7 +241,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return -EINVAL; } - cinfo->snd_card.dev = &pdev->dev; + simple_snd_card.dev = dev; } if (!cinfo->name || @@ -247,28 +255,26 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } /* - * init snd_soc_dai_link + * init simple_dai_link */ - cinfo->snd_link.name = cinfo->name; - cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; - cinfo->snd_link.platform_name = cinfo->platform; - cinfo->snd_link.codec_name = cinfo->codec; - cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; - cinfo->snd_link.cpu_of_node = of_cpu; - cinfo->snd_link.codec_of_node = of_codec; - cinfo->snd_link.platform_of_node = of_platform; - cinfo->snd_link.init = asoc_simple_card_dai_init; + simple_dai_link.name = cinfo->name; + simple_dai_link.stream_name = cinfo->name; + simple_dai_link.cpu_dai_name = cinfo->cpu_dai.name; + simple_dai_link.platform_name = cinfo->platform; + simple_dai_link.codec_name = cinfo->codec; + simple_dai_link.codec_dai_name = cinfo->codec_dai.name; + simple_dai_link.cpu_of_node = of_cpu; + simple_dai_link.codec_of_node = of_codec; + simple_dai_link.platform_of_node = of_platform; /* - * init snd_soc_card + * init simple_snd_card */ - cinfo->snd_card.name = cinfo->card; - cinfo->snd_card.owner = THIS_MODULE; - cinfo->snd_card.dai_link = &cinfo->snd_link; - cinfo->snd_card.num_links = 1; + simple_snd_card.name = cinfo->card; + + snd_soc_card_set_drvdata(&simple_snd_card, cinfo); - return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); + return devm_snd_soc_register_card(dev, &simple_snd_card); } static const struct of_device_id asoc_simple_of_match[] = { -- 1.8.4

3 2

[alsa-devel] How to make Realtek HD-A codec alc282 create "Input Source" control to select a input for recording?
by Lin, Mengdong 13 Jan '14

13 Jan '14

Hi, How to make the driver create a "input source" control for ALC282 with 3 Mic pins? My platform uses a Realtek HD-A codec ALC282 card, it has 3 Mic pins: Mic = 0x18 (MIC1) Internal Mic = 0x19 (MIC2) Internal Mic = 0x12 (DMIC) Recording on Mic1 (0x18) cannot work because the mixer 0x23 always mutes the input from 0x18. And there is no "input source" control to let the mixer select and unmute 0x18. I've tried to disable pin sense on pin 0x18 as a board fix-up like below, but "input source" control is still not created: caps = snd_hda_query_pin_caps(codec, 0x18); caps &= ~AC_PINCAP_PRES_DETECT; snd_hda_override_pin_caps(codec, 0x18, caps); snd_hda_codec_set_pincfg(codec, 0x18, 0x03a19120); ... set 'Jack Detect Override' Thanks Mengdong Here is the alsa-info: upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.61 !!################################ !!Script ran on: Thu Jan 9 06:09:23 UTC 2014 !!Linux Distribution !!------------------ Ubuntu Saucy Salamander (development branch) \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu Saucy Salamander (development branch)" NAME="Ubuntu" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu Saucy Salamander (development branch)" HOME_URL="http://www.ubuntu.com/" SUPPORT_URL="http://help.ubuntu.com/" BUG_REPORT_URL="http://bugs.launchpad.net/ubuntu/" !!DMI Information !!--------------- Manufacturer: Intel Corporation Product Name: Broadwell Client platform Product Version: 0.1 Firmware Version: BDW-E1R1.86C.0052.R01.1311260206 !!Kernel Information !!------------------ Kernel release: 3.13.0-rc4-drm-bdw+ Operating System: GNU/Linux Architecture: x86_64 Processor: x86_64 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k3.13.0-rc4-drm-bdw+ Library version: 1.0.27.2 Utilities version: 1.0.27.1 !!Loaded ALSA modules !!------------------- snd_hda_intel snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xd1030000 irq 61 1 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xd1034000 irq 59 !!PCI Soundcards installed in the system !!-------------------------------------- 00:03.0 Audio device: Intel Corporation Device 160c (rev 04) 00:1b.0 Audio device: Intel Corporation Device 9ca0 !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:03.0 0403: 8086:160c (rev 04) Subsystem: 8086:2010 -- 00:1b.0 0403: 8086:9ca0 Subsystem: 8086:7270 !!Modprobe options (Sound related) !!-------------------------------- snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 snd_usb_audio: index=-2 snd_usb_caiaq: index=-2 snd_usb_ua101: index=-2 snd_usb_us122l: index=-2 snd_usb_usx2y: index=-2 snd_cmipci: mpu_port=0x330 fm_port=0x388 snd_pcsp: index=-2 snd_usb_audio: index=-2 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 32,1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 32,1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Intel Broadwell HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862808 Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D3, actual=D3 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D3, actual=D3 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=3 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="ELD", index=0, device=3 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=01, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=7 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=1, device=0 Control: name="IEC958 Playback Pro Mask", index=1, device=0 Control: name="IEC958 Playback Default", index=1, device=0 Control: name="IEC958 Playback Switch", index=1, device=0 Control: name="ELD", index=0, device=7 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=02, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 3 0x02* 0x03 0x04 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=8 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=2, device=0 Control: name="IEC958 Playback Pro Mask", index=2, device=0 Control: name="IEC958 Playback Default", index=2, device=0 Control: name="IEC958 Playback Switch", index=2, device=0 Control: name="ELD", index=0, device=8 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=03, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono Codec: Realtek ALC282 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0282 Subsystem Id: 0x10ec0282 Revision Id: 0x100002 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=3, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC282 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x54 0x54] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC282 Analog", type="Audio", device=0 Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 Amp-In vals: [0x29 0x29] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 Amp-In vals: [0x97 0x97] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x22 Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Internal Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 5 0x18 0x19 0x1a 0x1b 0x1d Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x0d Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 Amp-In vals: [0x29 0x29] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Phantom Jack", index=1, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x90a60930: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00010014: OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x0c Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x17 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x0f Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x03 0x03] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x03a19820: [Jack] Mic at Ext Left Conn = 1/8, Color = Pink DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x19 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x90a7092f: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x2, Sequence = 0xf Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1a [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1d [Pin Complex] wcaps 0x400400: Mono Pincap 0x00000020: IN Pin Default 0x40189d2d: [N/A] Speaker at Ext N/A Conn = DIN, Color = Pink DefAssociation = 0x2, Sequence = 0xd Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1e [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=117 Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000001c: OUT HP Detect Pin Default 0x0321401f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c 0x0d* Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 6 0x18 0x19 0x1a 0x1b 0x1d 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 6 0x18 0x19 0x1a 0x1b 0x1d 0x0b --endcollapse--

2 5

[alsa-devel] [PATCH] ASoC: wm8804: Cleanup of post_table
by Daniel Matuschek 12 Jan '14

12 Jan '14

Signed-off-by: Daniel Matuschek <inof(a)crazy-audio.com> Existing post_table has duplicate settings for some PLL settings. Only the first setting was used with mclk_div=1 With the updated post_table, the driver uses mclk_div=0 whenever it is possible. This allows the WM8804 to work at 256xfs internal clock and only use 128xfs internal clock for high sample rates where 256xfs is not supported anymore. --- sound/soc/codecs/wm8804.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 1704b1e..f605b8b 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -307,9 +307,7 @@ static struct { unsigned int mclkdiv; } post_table[] = { { 2, 0, 0 }, - { 4, 0, 1 }, { 4, 1, 0 }, - { 8, 1, 1 }, { 8, 2, 0 }, { 16, 2, 1 }, { 12, 3, 0 }, -- 1.7.9.5

2 1

[alsa-devel] searching volunteer to test Hoontech DSP24 to check if the driver is realy working well
by YvanM 12 Jan '14

12 Jan '14

Hi everybody ! First please apologize if I'm in the wrong place. Secondly thanks for all the work you are all doing ! I have a Hoontech DSP24 (http://fr.audiofanzine.com/carte-son-rack/hoontech/DSP24-+-ADDA2000-DSP2000…), but I had problem to make it working. I don't use it so, if someone with knowledge wants to check if this soundcard is really recognized, I can send it to him. After that, feel free to play with it, improve the driver, or even sell it ! The problem I had : - spent 3 months to find on the Internet the following lines to had to /etc/modprobe.d/alsa-base to make the card recognized : stall snd-ice1712 /sbin/modprobe --ignore-install snd-ice1712 && { /sbin/modprobe -Qb snd-ice1712 ; } options snd-ice1712 model=dsp24 - impossible to have "normal" stereo : it seems that each logical canal (right and left) is redirected to both right and left speaker. I someone wants to have a try with it, do not hesitate to mail me. It is a pity that a so good soundcard can't be used anymore… And as I said, feel then free to do what you want with it. Thanks, Yvan

3 2

[alsa-devel] [PATCH] ALSA: Enable CONFIG_ZONE_DMA for smaller PCI DMA masks
by Takashi Iwai 11 Jan '14

11 Jan '14

The PCI devices with DMA masks smaller than 32bit should enable CONFIG_ZONE_DMA. Since the recent change of page allocator, page allocations via dma_alloc_coherent() with the limited DMA mask bits may fail more frequently, ended up with no available buffers, when CONFIG_ZONE_DMA isn't enabled. With CONFIG_ZONE_DMA, the system has much more chance to obtain such pages. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=68221 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/Kconfig | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 3a0504dfeb8a..8756c8e32922 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -25,6 +25,7 @@ config SND_ALS300 select SND_PCM select SND_AC97_CODEC select SND_OPL3_LIB + select ZONE_DMA help Say 'Y' or 'M' to include support for Avance Logic ALS300/ALS300+ @@ -49,6 +50,7 @@ config SND_ALI5451 tristate "ALi M5451 PCI Audio Controller" select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for the integrated AC97 sound device on motherboards using the ALi M5451 Audio Controller @@ -153,6 +155,7 @@ config SND_AZT3328 select SND_PCM select SND_RAWMIDI select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for Aztech AZF3328 (PCI168) soundcards. @@ -459,6 +462,7 @@ config SND_EMU10K1 select SND_HWDEP select SND_RAWMIDI select SND_AC97_CODEC + select ZONE_DMA help Say Y to include support for Sound Blaster PCI 512, Live!, Audigy and E-mu APS (partially supported) soundcards. @@ -474,6 +478,7 @@ config SND_EMU10K1X tristate "Emu10k1X (Dell OEM Version)" select SND_AC97_CODEC select SND_RAWMIDI + select ZONE_DMA help Say Y here to include support for the Dell OEM version of the Sound Blaster Live!. @@ -507,6 +512,7 @@ config SND_ES1938 select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on ESS Solo-1 (ES1938, ES1946, ES1969) chips. @@ -518,6 +524,7 @@ config SND_ES1968 tristate "ESS ES1968/1978 (Maestro-1/2/2E)" select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on ESS Maestro 1/2/2E chips. @@ -606,6 +613,7 @@ config SND_ICE1712 select SND_MPU401_UART select SND_AC97_CODEC select BITREVERSE + select ZONE_DMA help Say Y here to include support for soundcards based on the ICE1712 (Envy24) chip. @@ -693,6 +701,7 @@ config SND_LX6464ES config SND_MAESTRO3 tristate "ESS Allegro/Maestro3" select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on ESS Maestro 3 (Allegro) chips. @@ -789,6 +798,7 @@ config SND_SIS7019 tristate "SiS 7019 Audio Accelerator" depends on X86 && !X86_64 select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for the SiS 7019 Audio Accelerator. @@ -800,6 +810,7 @@ config SND_SONICVIBES select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on the S3 SonicVibes chip. @@ -811,6 +822,7 @@ config SND_TRIDENT tristate "Trident 4D-Wave DX/NX; SiS 7018" select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on Trident 4D-Wave DX/NX or SiS 7018 chips. -- 1.8.5.2

1 0

Re: [alsa-devel] [PATCH v2] ASoC: fsl_esai: Add ESAI CPU DAI driver
by Nicolin Chen 10 Jan '14

10 Jan '14

Resent this because of losing attached file. On Fri, Jan 10, 2014 at 10:32:52AM +0800, Nicolin Chen wrote: > On Thu, Jan 09, 2014 at 06:44:53PM +0000, Mark Brown wrote: > > On Thu, Jan 09, 2014 at 06:57:58PM +0800, Nicolin Chen wrote: > > > > > +/** > > > + * This function configures the ratio between MCLK (HCK) and BCLK (SCK) > > > + * (For DAI Master Mode only) > > > + * > > > + * Note: Machine driver should calculate the ratio to call this function. > > > + * Only effective after calling set_dai_sysclk() to set HCK direction. > > > + */ > > > +static int fsl_esai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) > > > > Why does the machine driver have to do this by hand, being able to > > override is fine but having sensible defaults is easier? Or does it > > actually do that and the comment just needs updating? > > The divider part of ESAI is pretty complicated due to caring about three > configure bits - ETO, ETI, HCKD. (I've attached a diagram to this mail.) > So setting sysclk() alone is not enough to take care all the configurations. > That's why I designed these two interfaces at the first place. And it should > be hard to find a default situation. > > But there is one approach to omit this calling for machine driver is to do > the set_bclk_ratio() at this CPU DAI driver. And this might be a good idea > because we can then separate the settings between PLAYBACK and CAPTURE, even > though we might then need to check the Master or Slave state to apply the > settings accordingly. > > I'll try this idea in v3 if you feel it's plausible. > > > > + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component, > > > + &fsl_esai_dai, 1); > > > + if (ret) { > > > + dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); > > > + return ret; > > > + } > > > + > > > + ret = imx_pcm_dma_init(pdev); > > > + if (ret) > > > + dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); > > > + > > > + /* Reset ESAI unit */ > > > + regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ERST); > > > + > > > + /* > > > + * We need to enable ESAI so as to access some of its registers. > > > + * Otherwise, we would fail to dump regmap from user space. > > > + */ > > > + regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ESAIEN); > > > > I would expect to see the hardware initialisation before we start > > registering with the core otherwise the core might start trying to run > > prior to the hardware being initialised. > > Will switch this initial code in v3 as your suggestion. > > Thank you, > Nicolin Chen

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