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

[alsa-devel] SoundBlaster X-fi HD SPDIF fix
by Richard Körber 05 Apr '15

05 Apr '15

Hi! I own a Creative USB SoundBlaster X-fi HD and tried to use its SPDIF optical output. However it didn't work with ALSA out of the box. The reason is that for SPDIF output device #1 must be used. The fix is: In /usr/share/alsa/cards/USB-Audio.conf, change the iec958_device block like this: USB-Audio.pcm.iec958_device { "USB Sound Blaster HD" 1 } Digital output works fine for me after that. BTW, I tried this line in /etc/asound.conf cards.USB-Audio.pcm.iec958_device."USB Sound Blaster HD" 1 but it didn't have any effect. Why was that? Thank you! Regards Richard Körber

2 8

[alsa-devel] [PATCH] add Steinberg UR44 USB Audio Interface support to quirks-table
by Cyrus Harmon 03 Feb '15

03 Feb '15

Signed-off-by: Cyrus Harmon <cyrus(a)bobobeach.com> --- sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f652b10..987ca01 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -436,6 +436,36 @@ YAMAHA_DEVICE(0x105d, NULL), } } }, +{ + USB_DEVICE(0x0499, 0x150e), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Yamaha", */ + /* .product_name = "Steinberg UR44", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_YAMAHA + }, + { + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), -- 1.8.3.2

5 32

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Correct rx format unit configuration
by Peter Ujfalusi 14 Jan '15

14 Jan '15

In case of capture we should not use rotation. The reverse and mask is enough to get the data align correctly from the bus to MCU: Format data from bus after reverse (XRBUF) S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| With this patch all supported formats will work for playback and capture. Reported-by: Jyri Sarha <jsarha(a)ti.com> (broken S24_3LE capture) Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6a6b2ff7d7d7..68347b55f6e1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -467,8 +467,17 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, { u32 fmt; u32 tx_rotate = (word_length / 4) & 0x7; - u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; + /* + * For captured data we should not rotate, inversion and masking is + * enoguh to get the data to the right position: + * Format data from bus after reverse (XRBUF) + * S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| + * S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| + */ + u32 rx_rotate = 0; /* * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() -- 2.1.0

3 6

[alsa-devel] "ak" vs "asahi-kasei" (was: Re: [PATCH 3/3 v3] ARM: shmobile: lager: add sound support on DT)
by Geert Uytterhoeven 13 Jan '15

13 Jan '15

Hi Morimoto-san, CC devicetree, alsa-devel On Wed, Jun 11, 2014 at 9:40 AM, Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> wrote: >> On Wed, Jun 11, 2014 at 8:54 AM, Kuninori Morimoto >> <kuninori.morimoto.gx(a)renesas.com> wrote: >> > + compatible = "asahi-kasei,ak4643"; >> >> As Sergei already commented, >> Documentation/devicetree/bindings/vendor-prefixes.txt >> says >> >> ak Asahi Kasei Corp. > > I know, but, > > grep asahi-kasei sound/soc/codecs/* > sound/soc/codecs/ak4104.c: { .compatible = "asahi-kasei,ak4104", }, > sound/soc/codecs/ak4554.c: { .compatible = "asahi-kasei,ak4554" }, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4642", .data = &ak4642_drvdata}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4643", .data = &ak4643_drvdata}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4648", .data = &ak4648_drvdata}, > sound/soc/codecs/ak5386.c: { .compatible = "asahi-kasei,ak5386", }, > > ak4642 driver can't probe by "ak"; That's bad. And all but one of these match entries have been added _after_ the addition of "ak" to vendor-prefixes.txt... Outside sound, there's also "asahi-kasei,ak8975" vs. "ak,ak8975". > My code is based on "code", not "document" Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

14 44

Re: [alsa-devel] [FFADO-devel] [RFC] libhinawa: a light-weight I/O library for status/transactions to ALSA FireWire devices
by Jonathan Woithe 12 Jan '15

12 Jan '15

Hi Takashi S On Tue, Sep 23, 2014 at 08:39:34PM +0900, Takashi Sakamoto wrote: > For some reasons (I describe later), I start to develop a light-weight > I/O library for status/transactions to ALSA FireWire devices. > https://github.com/takaswie/libhinawa > > This library gives ways to use ALSA FireWire driver functionality such as > lock/unlock kernel streaming, EFW transaction. This is one of my > solutions to some issues between ALSA/FFADO. As I understand it, this library will be a thin wrapper around whatever kernel interface is used to communicate these details out to userspace (I think byte-type mixer controls are the preferred avenue at present). In principle I have no problem with this approach so long as it remains lightweight. The situation to avoid is the one we had with libraw1394; when the new stack came about the preferred way to interact with it switched back to the native kernel interface since (as I understand the situation) it was felt that libraw1394 didn't really provide a worthwhile API abstraction. Out of interest, is there any signficance in the library's name "hinawa"? > I hope to merge this library into alsa-tools package in future, and this > is a first RFC for it. It's under development and I require your > comments about its APIs and structure ... I will try to find some time to look at the current proposal in the next week or so. > Dice notification is an actual example. Dice based devices transfer > notification to a specific address on host controller. The address space > is exclusive resources on an system. For Dice notification, ALSA Dice > driver gives a way to utilize it for applications As well as Fireworks > situation, some userspace stuffs are needed to use this functionality. For sure. One thing to keep in mind is that while this is an issue for DICE devices it may not apply to others. The library API should be structured to keep in mind that not all features will be needed across the board. We don't want to burden all subsequent firewire-audio streaming drivers with baggage which is only required for a small subset. Regards jonathan

2 2

[alsa-devel] [PATCH - ALSA: snd-virtuoso 1/1] Xonar DSX support
by sgiurgiu11＠gmail.com 04 Jan '15

04 Jan '15

From: Sergiu Giurgiu <sgiurgiu11(a)gmail.com> This patch adds support for ASUS - Xonar DSX sound cards. Tested on openSUSE 12.2 with kernel: Linux 3.4.6-2.10-desktop #1 SMP PREEMPT Thu Jul 26 09:36:26 UTC 2012 (641c197) x86_64 x86_64 x86_64 GNU/Linux Works: - play sounds - adjust volume on master channel. - mute . Since Xonar DS uses the same chip, everything that works for DS should work for DSX as well. Thank you, Sergiu Giurgiu. Signed-off-by: Sergiu Giurgiu <sgiurgiu11(a)gmail.com> diff --git a/mirror/sound/pci/oxygen/virtuoso.c b/mirror/sound/pci/oxygen/virtuoso.c index 3d71423..8104eab 100644 --- a/mirror/sound/pci/oxygen/virtuoso.c +++ b/mirror/sound/pci/oxygen/virtuoso.c @@ -52,6 +52,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8522) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; diff --git a/mirror/sound/pci/oxygen/xonar_wm87x6.c b/mirror/sound/pci/oxygen/xonar_wm87x6.c index 63cff90..b555b62 100644 --- a/mirror/sound/pci/oxygen/xonar_wm87x6.c +++ b/mirror/sound/pci/oxygen/xonar_wm87x6.c @@ -1255,7 +1255,6 @@ static void dump_wm87x6_registers(struct oxygen *chip, } static const struct oxygen_model model_xonar_ds = { - .shortname = "Xonar DS", .longname = "Asus Virtuoso 66", .chip = "AV200", .init = xonar_ds_init, @@ -1327,6 +1326,11 @@ int __devinit get_xonar_wm87x6_model(struct oxygen *chip, switch (id->subdevice) { case 0x838e: chip->model = model_xonar_ds; + chip->model.shortname = "Xonar DS"; + break; + case 0x8522: + chip->model = model_xonar_ds; + chip->model.shortname = "Xonar DSX"; break; case 0x835e: chip->model = model_xonar_hdav_slim; -- 1.7.10.4

4 5

[alsa-devel] [PATCH] pcm: add new 32-bit DSD sample format
by Jurgen Kramer 21 Nov '14

21 Nov '14

Add the new DSD_U32_LE sample format to alsa-lib. NB include/pcm.h and include/sound/asound.h are updated so a new sync with the kernel headers is not needed Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- include/pcm.h | 4 +++- include/sound/asound.h | 3 ++- src/pcm/pcm.c | 2 ++ src/pcm/pcm_misc.c | 6 ++++++ 4 files changed, 13 insertions(+), 2 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 11e9f0d..db88ad5 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -211,7 +211,9 @@ typedef enum _snd_pcm_format { SND_PCM_FORMAT_DSD_U8, /* Direct Stream Digital (DSD) in 2-byte samples (x16) */ SND_PCM_FORMAT_DSD_U16_LE, - SND_PCM_FORMAT_LAST = SND_PCM_FORMAT_DSD_U16_LE, + /* Direct Stream Digital (DSD) in 4-byte samples (x32) */ + SND_PCM_FORMAT_DSD_U32_LE, + SND_PCM_FORMAT_LAST = SND_PCM_FORMAT_DSD_U32_LE, #if __BYTE_ORDER == __LITTLE_ENDIAN /** Signed 16 bit CPU endian */ diff --git a/include/sound/asound.h b/include/sound/asound.h index 32168f7..6ee5867 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -219,7 +219,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ #define SNDRV_PCM_FORMAT_DSD_U8 ((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD (x8) */ #define SNDRV_PCM_FORMAT_DSD_U16_LE ((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD (x16), little endian */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U16_LE +#define SNDRV_PCM_FORMAT_DSD_U32_LE ((__force snd_pcm_format_t) 50) /* DSD, 4-byte samples DSD (x32), little endian */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U32_LE #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 1399a5b..2e24338 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1565,6 +1565,7 @@ static const char *const snd_pcm_format_names[] = { FORMAT(G723_40_1B), FORMAT(DSD_U8), FORMAT(DSD_U16_LE), + FORMAT(DSD_U32_LE), }; static const char *const snd_pcm_format_aliases[SND_PCM_FORMAT_LAST+1] = { @@ -1624,6 +1625,7 @@ static const char *const snd_pcm_format_descriptions[] = { FORMATD(G723_40_1B, "G.723 (ADPCM) 40 kbit/s, 1 sample in 1 byte"), FORMATD(DSD_U8, "Direct Stream Digital, 1-byte (x8), oldest bit in MSB"), FORMATD(DSD_U16_LE, "Direct Stream Digital, 2-byte (x16), little endian, oldest bits in MSB"), + FORMATD(DSD_U32_LE, "Direct Stream Digital, 4-byte (x32), little endian, oldest bits in MSB"), }; static const char *const snd_pcm_type_names[] = { diff --git a/src/pcm/pcm_misc.c b/src/pcm/pcm_misc.c index 46fc771..9272179 100644 --- a/src/pcm/pcm_misc.c +++ b/src/pcm/pcm_misc.c @@ -64,6 +64,7 @@ int snd_pcm_format_signed(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U18_3BE: case SNDRV_PCM_FORMAT_DSD_U8: case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 0; default: return -EINVAL; @@ -154,6 +155,7 @@ int snd_pcm_format_little_endian(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U18_3BE: case SNDRV_PCM_FORMAT_DSD_U8: case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 0; default: return -EINVAL; @@ -232,6 +234,7 @@ int snd_pcm_format_width(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U32_BE: case SNDRV_PCM_FORMAT_FLOAT_LE: case SNDRV_PCM_FORMAT_FLOAT_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 32; case SNDRV_PCM_FORMAT_FLOAT64_LE: case SNDRV_PCM_FORMAT_FLOAT64_BE: @@ -292,6 +295,7 @@ int snd_pcm_format_physical_width(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_FLOAT_BE: case SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE: case SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 32; case SNDRV_PCM_FORMAT_FLOAT64_LE: case SNDRV_PCM_FORMAT_FLOAT64_BE: @@ -348,6 +352,7 @@ ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples) case SNDRV_PCM_FORMAT_U32_BE: case SNDRV_PCM_FORMAT_FLOAT_LE: case SNDRV_PCM_FORMAT_FLOAT_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return samples * 4; case SNDRV_PCM_FORMAT_FLOAT64_LE: case SNDRV_PCM_FORMAT_FLOAT64_BE: @@ -394,6 +399,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) return 0x8080808080808080ULL; case SNDRV_PCM_FORMAT_DSD_U8: case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 0x6969696969696969ULL; #ifdef SNDRV_LITTLE_ENDIAN case SNDRV_PCM_FORMAT_U16_LE: -- 1.9.3

3 7

[alsa-devel] AKAI EIE PRO
by Martin Svensson 20 Nov '14

20 Nov '14

Dont know if its ok to ask a question here so please let me know if its not ok then i wont ask again ;) i just bought an external usb AKAI EIE PRO sound card, is there any future plans to implement it in Alsa? Sincerally Martin aka "ztealmax"

4 7

[alsa-devel] [PATCH v6] ASoC: dapm: add code to configure dai link parameters
by Nikesh Oswal 04 Nov '14

04 Nov '14

dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 143 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 146 insertions(+), 7 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 6b59471..3ee031e 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -531,6 +532,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index ed9e2d7..51c6c4f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -906,6 +906,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b87d7d8..1db2168 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1461,7 +1461,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1473,7 +1474,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index cdc837e..68f6057 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -729,6 +729,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* skip control creation for links with 1 config */ + if (w->num_params == 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* reset 'walked' bit for each dapm path */ static void dapm_clear_walk_output(struct snd_soc_dapm_context *dapm, struct list_head *sink) @@ -2664,6 +2694,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3142,6 +3175,9 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + /* Select the configuration set by alsa control */ + config += w->params_select; + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3222,8 +3258,35 @@ out: return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3231,14 +3294,48 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = kcalloc(num_params, sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + if (!link_name) { + ret = -ENOMEM; + goto outfree_w_param; + } + snprintf(link_name, len, "%s-%s", source->name, sink->name); + for (count = 0 ; count < num_params; count++) { + if (!config->stream_name) + dev_warn(card->dapm.dev, + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); + w_param_text[count] = kmemdup((void *)(config->stream_name), + strlen(config->stream_name) + 1, GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3246,6 +3343,25 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + private_value = + (unsigned long) kmemdup((void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + template.kcontrol_news = kmemdup(&kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3253,15 +3369,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + kfree(w); +outfree_kcontrol_news: + kfree(template.kcontrol_news); +outfree_private_value: + kfree((void *)private_value); +outfree_link_name: + kfree(link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + kfree(w_param_text[count]); + kfree(w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, -- 1.7.9.5

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[alsa-devel] [PATCHv4] ASoC: simple-card: binding: update binding to support the new style.
by Xiubo Li 04 Nov '14

04 Nov '14

This patch will split the DT binding into old style and new style, the new style will be easier to add many DAI links from old single DAI link DTs. This patch will maintian compatibility with the old DTs. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Changes in v4: - Follow Jyri's advice. - Fix some small words or sentences errors. - Fix discription of the commit commnet. - Apply against the current code. - Add Notes. .../devicetree/bindings/sound/simple-card.txt | 215 ++++++++++++++------- 1 file changed, 148 insertions(+), 67 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index c2e9841..5fe0796 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -1,15 +1,19 @@ -Simple-Card: +Device-Tree bindings for Simple Card Simple-Card specifies audio DAI connections of SoC <-> codec. -Required properties: +=== Top level's properties and subnodes === +*** Required properties *** - compatible : "simple-audio-card" -Optional properties: - +*** Optional properties *** - simple-audio-card,name : User specified audio sound card name, one string property. +- simple-audio-card,format : CPU/CODEC common audio format. + "i2s", "right_j", "left_j" , "dsp_a" + "dsp_b", "ac97", "pdm", "msb", "lsb" + (This is used for single DAI link & old style.) - simple-audio-card,widgets : Please refer to widgets.txt. - simple-audio-card,routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the @@ -17,63 +21,99 @@ Optional properties: source. - simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec mclk. - -Optional subnodes: - -- simple-audio-card,dai-link : Container for dai-link level - properties and the CPU and CODEC - sub-nodes. This container may be - omitted when the card has only one - DAI link. See the examples and the - section bellow. - -Dai-link subnode properties and subnodes: - -If dai-link subnode is omitted and the subnode properties are directly -under "sound"-node the subnode property and subnode names have to be -prefixed with "simple-audio-card,"-prefix. - -Required dai-link subnodes: - -- cpu : CPU sub-node -- codec : CODEC sub-node - -Optional dai-link subnode properties: - +- simple-audio-card,frame-master : Indicates DAI link frame master. One phandle to a + CODEC DAI node. + (This is used for single DAI link & old style.) +- simple-audio-card,bitclock-master : Indicates DAI link bit clock master. One phandle to + a CODEC DAI node. + (This is used for single DAI link & old style.) + +*** Optional subnodes *** +- simple-audio-card,dai-link : Container for DAI link level properties and the CPU + and CODEC nodes. This container may be omitted + when the card has only one DAI link and using the old + style. See the examples and the section bellow. +- simple-audio-card,cpu : CPU DAI node. + (This is used for single DAI link & old style.) +- simple-audio-card,codec : CODEC DAI node. + (This is used for single DAI link & old style.) + +=== DAI link node's properties and its subnodes === + +*** Required subnodes *** +- cpu : CPU DAI node. +- codec : CODEC DAI node. + +*** Optional properties *** - format : CPU/CODEC common audio format. "i2s", "right_j", "left_j" , "dsp_a" "dsp_b", "ac97", "pdm", "msb", "lsb" -- frame-master : Indicates dai-link frame master. - phandle to a cpu or codec subnode. -- bitclock-master : Indicates dai-link bit clock master. - phandle to a cpu or codec subnode. -- bitclock-inversion : bool property. Add this if the - dai-link uses bit clock inversion. -- frame-inversion : bool property. Add this if the - dai-link uses frame clock inversion. - -For backward compatibility the frame-master and bitclock-master -properties can be used as booleans in codec subnode to indicate if the -codec is the dai-link frame or bit clock master. In this case there -should be no dai-link node, the same properties should not be present -at sound-node level, and the bitclock-inversion and frame-inversion -properties should also be placed in the codec node if needed. - -Required CPU/CODEC subnodes properties: - -- sound-dai : phandle and port of CPU/CODEC - -Optional CPU/CODEC subnodes properties: - +- frame-master : Indicates DAI link frame master. One phandle to a + CODEC DAI node. + (This is One boolean property for old style.) +- bitclock-master : Indicates DAI link bit clock master. One phandle to + a CODEC DAI node. + (This is one boolean property for old style.) + +For backward compatibility, the frame-master and bitclock-master +can be used as boolean properties in CPU/CODEC DAI node indicating +that the CODEC DAI is the DAI link frame master and bit clock master. +In this case there should be no DAI link level nodes or boolean +properties of them. + +For all the properties in DAI link level will be applied to both +CPU and CODEC DAIs. + +=== CPU/CODEC DAI node's properties and its subnodes === + +*** Required properties *** +- sound-dai : One phandle and port of CPU/CODEC DAI device node. + +*** Optional properties *** +- bitclock-inversion : Boolean property. Add this if the DAI device uses bit + clock inversion. +- frame-inversion : Boolean property. Add this if the DAI device uses frame + clock inversion. +- frame-master : Boolean property. Add this if the CODEC DAI device is + frame master. + (This is used for single DAI link & old style.) +- bitclock-master : Boolean property. Add this if the CODEC DAI device is + bit clock master. + (This is used for single DAI link & old style.) - dai-tdm-slot-num : Please refer to tdm-slot.txt. - dai-tdm-slot-width : Please refer to tdm-slot.txt. -- clocks / system-clock-frequency : specify subnode's clock if needed. - it can be specified via "clocks" if system has - clock node (= common clock), or "system-clock-frequency" - (if system doens't support common clock) +- clocks / system-clock-frequency : Specify CPU/CODEC DAI's clock if needed. It can + be specified via "clocks" if system has clock node + (= common clock), or "system-clock-frequency"(if system + doens't support common clock) + +=== Notes === +*** DAI hardware clock masters *** +For [prefix]frame-master and [prefix]bitclock-master, which prefix +is "simple-audio-card,", are both wrt the CODEC DAI. If the CODEC DAI +is bit clock and frame master then the CPU DAI will be bit clock and +frame slave, and vice versa. + +=== Examples === +*** CPU & CODEC DAI DT nodes *** +&i2c0 { + ak4648: ak4648@12 { + #sound-dai-cells = <0>; + compatible = "asahi-kasei,ak4648"; + reg = <0x12>; + }; +}; -Example 1 - single DAI link: +sh_fsi2: sh_fsi2@ec230000 { + #sound-dai-cells = <1>; + compatible = "renesas,sh_fsi2"; + reg = <0xec230000 0x400>; + interrupt-parent = <&gic>; + interrupts = <0 146 0x4>; +}; +Example 1 - single DAI link & old style: +bitclock-master and frame-master as phandles. sound { compatible = "simple-audio-card"; simple-audio-card,name = "VF610-Tower-Sound-Card"; @@ -91,6 +131,7 @@ sound { simple-audio-card,cpu { sound-dai = <&sh_fsi2 0>; + bitclock-inversion; }; dailink0_master: simple-audio-card,codec { @@ -99,24 +140,64 @@ sound { }; }; -&i2c0 { - ak4648: ak4648@12 { - #sound-dai-cells = <0>; - compatible = "asahi-kasei,ak4648"; - reg = <0x12>; +Example 2 - single DAI link & old style: +bitclock-master and frame-master as boolean properties. +sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,format = "left_j"; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; }; -}; -sh_fsi2: sh_fsi2@ec230000 { - #sound-dai-cells = <1>; - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupt-parent = <&gic>; - interrupts = <0 146 0x4>; + simple-audio-card,codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + bitclock-master; + frame-master; + bitclock-inversion; + }; }; -Example 2 - many DAI links: +Example 3 - single DAI link & new style: +sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,dai-link { + format = "i2s"; + bitclock-master = <&dailink0_master>; + frame-master = <&dailink0_master>; + cpu { + sound-dai = <&sh_fsi2 0>; + frame-inversion; + }; + dailink0_master: codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + frame-inversion; + }; + }; +}; +Example 4 - many DAI links: sound { compatible = "simple-audio-card"; simple-audio-card,name = "Cubox Audio"; -- 2.1.0.27.g96db324

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