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

[alsa-devel] [PATCH v3 1/1] ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pins
by mengdong.lin＠intel.com 17 Jan '13

17 Jan '13

From: Mengdong Lin <mengdong.lin(a)intel.com> Haswell HDMI codec pins may report invalid connection list entries, which will cause failure to play audio via HDMI or Display Port. So this patch adds fixup for Haswell to workaround this hardware issue: enable DP1.2 mode and override the pins' connection list entries with proper value. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Xingchao Wang <xingchao.wang(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 71555cc..59abe73 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1687,6 +1687,30 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .unsol_event = hdmi_unsol_event, }; +static void intel_haswell_fixup_connect_list(struct hda_codec *codec) +{ + unsigned int vendor_param; + hda_nid_t list[3] = {0x2, 0x3, 0x4}; + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || vendor_param & 0x02) + return; + + /* enable DP1.2 mode */ + vendor_param |= 0x02; + snd_hda_codec_read(codec, 0x08, 0, 0x781, vendor_param); + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || !(vendor_param & 0x02)) + return; + + /* override 3 pins connection list */ + snd_hda_override_conn_list(codec, 0x05, 3, list); + snd_hda_override_conn_list(codec, 0x06, 3, list); + snd_hda_override_conn_list(codec, 0x07, 3, list); +} + + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1696,6 +1720,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + + if (codec->vendor_id == 0x80862807) + intel_haswell_fixup_connect_list(codec); + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); -- 1.7.9.5

4 5

[alsa-devel] [PATCH 0/3] ...and three more fixes for the new generic parser
by David Henningsson 16 Jan '13

16 Jan '13

These regressions were found when running the test tool over even more alsa-infos. Btw, would you mind updating the hda-migrate branch too? It's useful to have your latest fixes from the hda-gen-parser branch when I'm running pre-release QA testing. David Henningsson (3): ALSA: hda - do not add non-existing Mic boost controls ALSA: hda - force different capture controls if amp caps differ ALSA: hda - Make sure fill_all_dac_nids is called for digital only codecs sound/pci/hda/hda_generic.c | 31 ++++++++++++++++++++++++++----- 1 file changed, 26 insertions(+), 5 deletions(-) -- 1.7.9.5

2 4

[alsa-devel] alsa-util fix (Re: [PATCH] alsa-lib: Fix build with automake-1.13)
by Marko Lindqvist 16 Jan '13

16 Jan '13

On 4 January 2013 01:40, Marko Lindqvist <cazfi74(a)gmail.com> wrote: > Automake-1.13 removed long obsolete AM_CONFIG_HEADER completely ( > http://lists.gnu.org/archive/html/automake/2012-12/msg00038.html ) and > errors out upon seeing it. > > Attached patch replaces it with proper AC_CONFIG_HEADERS. > > > I'm not subscribed to the list. I'm mass-submitting automake-1.13 fixes. Please see what you can do so I don't need to do the same when automake-1.14 comes out: http://cazfi.livejournal.com/195108.html - ML

2 1

[alsa-devel] [PATCH] ASoC: fsi: Remove __devinitconst
by Thierry Reding 16 Jan '13

16 Jan '13

__devinitconst and friends have recently been removed and must not be used anymore. Signed-off-by: Thierry Reding <thierry.reding(a)avionic-design.de> --- sound/soc/sh/fsi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 9157612..c724026 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -2160,7 +2160,7 @@ static struct fsi_core fsi2_core = { .b_mclk = B_MST_CTLR, }; -static struct of_device_id fsi_of_match[] __devinitconst = { +static struct of_device_id fsi_of_match[] = { { .compatible = "renesas,sh_fsi", .data = &fsi1_core}, { .compatible = "renesas,sh_fsi2", .data = &fsi2_core}, {}, -- 1.8.1

2 1

[alsa-devel] [PATCH] ASoC: wm8978: SND_SOC_DAIFMT_NB_NF become default clock inversion
by Kuninori Morimoto 16 Jan '13

16 Jan '13

Current wm8978 driver return -EINVAL if it doesn't have SND_SOC_DAIFMT_INV_MASK flags on snd_soc_dai_ops :: set_fmt. But default settings should be treated as SND_SOC_DAIFMT_NB_NF (= no clock inversion). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/wm8978.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index f347af3..ba01981 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -692,8 +692,6 @@ static int wm8978_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* clock inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; case SND_SOC_DAIFMT_IB_IF: iface |= 0x180; break; @@ -703,8 +701,9 @@ static int wm8978_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) case SND_SOC_DAIFMT_NB_IF: iface |= 0x80; break; + case SND_SOC_DAIFMT_NB_NF: default: - return -EINVAL; + break; } snd_soc_write(codec, WM8978_AUDIO_INTERFACE, iface); -- 1.7.9.5

2 3

[alsa-devel] [PATCH 0/3] Fixes for the new generic parser
by David Henningsson 16 Jan '13

16 Jan '13

Hi Takashi, I've run my hda-emu summary test tool to check for errors in the 110 codecs we have of certified systems (those in codecs/canonical) against the test/hda-migrate branch. After these fixes and the one I sent yesterday, there are no remaining regressions (as seen by this test tool) compared to the current sound tree. David Henningsson (3): ALSA: hda - Check array bounds in get_input_path ALSA: hda - fix wrong adc_idx in generic parser ALSA: hda - initialize channel counts correctly sound/pci/hda/hda_generic.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) -- 1.7.9.5

2 4

[alsa-devel] [PATCH] ASoC: arizona: Don't request FLL lock IRQ
by Mark Brown 16 Jan '13

16 Jan '13

We only log the result and since the interrupt triggers on loss of lock during shutdown this may lead to spurious interrupts during shutdown delaying the process. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 19 ------------------- sound/soc/codecs/arizona.h | 1 - 2 files changed, 20 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 9da5eee..b3621d6 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1008,17 +1008,6 @@ int arizona_init_dai(struct arizona_priv *priv, int id) } EXPORT_SYMBOL_GPL(arizona_init_dai); -static irqreturn_t arizona_fll_lock(int irq, void *data) -{ - struct arizona_fll *fll = data; - - arizona_fll_dbg(fll, "Lock status changed\n"); - - complete(&fll->lock); - - return IRQ_HANDLED; -} - static irqreturn_t arizona_fll_clock_ok(int irq, void *data) { struct arizona_fll *fll = data; @@ -1258,7 +1247,6 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, { int ret; - init_completion(&fll->lock); init_completion(&fll->ok); fll->id = id; @@ -1269,13 +1257,6 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name), "FLL%d clock OK", id); - ret = arizona_request_irq(arizona, lock_irq, fll->lock_name, - arizona_fll_lock, fll); - if (ret != 0) { - dev_err(arizona->dev, "Failed to get FLL%d lock IRQ: %d\n", - id, ret); - } - ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name, arizona_fll_clock_ok, fll); if (ret != 0) { diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index b3b97f56..67fd49b 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -200,7 +200,6 @@ struct arizona_fll { int id; unsigned int base; unsigned int vco_mult; - struct completion lock; struct completion ok; unsigned int fref; unsigned int fout; -- 1.7.10.4

1 0

[alsa-devel] Fujitsu Lifebook s7110 (ALC262) no sound
by Raymond Yau 16 Jan '13

16 Jan '13

> after patching the clean 12.04 install with teh last alsa version (see comment #13) and with external speakers plugged (3.5 mm jack), I have got the follwoing Alsa log: http://www.alsa-project.org/db/?f=778415ad65d437d9ff5e5dd341e0e556ff848bc1 http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commitdiff;h=3d… -/* - * fujitsu model - * 0x14 = headphone/spdif-out, 0x15 = internal speaker, - * 0x1b = port replicator headphone out - */ Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Surround Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x032110f0: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=03, enabled=1 Connection: 2 0x0c 0x0d* Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Front Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0xb7231110: [Fixed] HP Out at Oth Mobile-In Conn = ATAPI, Color = Black DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0d Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Dock Headphone Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x2321101f: [Jack] HP Out at Sep Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Connection: 2 0x0c* 0x0d If seem that BIOS did not setup the internal speaker node and need pin fixup Pulseaudio expect headphone Jack and speaker phantom port https://launchpadlibrarian.net/128561206/pulseverbose2.log

2 1

[alsa-devel] [PATCH v3 0/4] RME HDSP fixes
by Adrian Knoth 16 Jan '13

16 Jan '13

Hi! Changes compared to v2: - use snd_printd to print iobox detection time - use KERN_ERR in case no iobox was found - define HDSP_S300 as a dummy for (0x100 | HDSP_S200) Adrian Knoth (4): ALSA: hdsp - Fix detection for RME RPM/Multiface/Digiface ioboxes ALSA: hdsp - Implement generic function to toggle settings ALSA: hdsp - Use HDSP_TOGGLE_SETTING to alter settings ALSA: hdsp - Remove obsolete settings functions sound/pci/rme9652/hdsp.c | 462 ++++++++++++---------------------------------- 1 file changed, 117 insertions(+), 345 deletions(-) -- 1.7.10.4

2 5

[alsa-devel] [PATCH] ASoC: maintainers: Update email address.
by Liam Girdwood 16 Jan '13

16 Jan '13

From: Liam Girdwood <liam.r.girdwood(a)intel.com> I'm no longer at TI. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)intel.com> --- MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 915564e..3c2e275 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7082,7 +7082,7 @@ F: include/uapi/sound/ F: sound/ SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) -M: Liam Girdwood <lrg(a)ti.com> +M: Liam Girdwood <lgirdwood(a)gmail.com> M: Mark Brown <broonie(a)opensource.wolfsonmicro.com> T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -- 1.7.10.4

2 1

[alsa-devel] [PATCH v2 0/4] RME HDSP fixes
by Adrian Knoth 15 Jan '13

15 Jan '13

Hi, here's v2 of my previous series of patches for the RME HDSP. Changes to the previous version: - correct time value in verbose output ("iobox found after %d ms") - define and use constants for iobox detection (no more 0x10a0) - hdsp_toggle_setting: only change register if newval != oldval Adrian Knoth (4): ALSA: hdsp - Fix detection for RME RPM/Multiface/Digiface ioboxes ALSA: hdsp - Implement generic function to toggle settings ALSA: hdsp - Use HDSP_TOGGLE_SETTING to alter settings ALSA: hdsp - Remove obsolete settings functions sound/pci/rme9652/hdsp.c | 461 ++++++++++++---------------------------------- 1 file changed, 116 insertions(+), 345 deletions(-) -- 1.7.10.4

2 5

[alsa-devel] [PATCH (sound-unstable)] ALSA: hda - fix OOPS in hda_mark_cmd_cache_dirty
by David Henningsson 15 Jan '13

15 Jan '13

Obvious copy-paste error. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b28e403..e6cdad7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3794,7 +3794,7 @@ static void hda_mark_cmd_cache_dirty(struct hda_codec *codec) } for (i = 0; i < codec->amp_cache.buf.used; i++) { struct hda_amp_info *amp; - amp = snd_array_elem(&codec->cmd_cache.buf, i); + amp = snd_array_elem(&codec->amp_cache.buf, i); amp->head.dirty = 1; } } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] hda-emu: Make it work with sound-unstable.git
by David Henningsson 15 Jan '13

15 Jan '13

To be able to build the new code (so we can run aggregate tests on it) the following patch needs to be applied. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- configure.ac | 8 +++++++ hda-ctlsh.c | 5 ++++ include/linux/ctype.h | 64 +++++++++++++++++++++++++++++++++++++++++++++++++ snd-wrapper.c | 32 +++++++++++++++++++++++++ 4 files changed, 109 insertions(+) create mode 100644 include/linux/ctype.h diff --git a/configure.ac b/configure.ac index 3103eb5..80b68a6 100644 --- a/configure.ac +++ b/configure.ac @@ -182,6 +182,14 @@ if test "$OLD_HDA_CMD" = "1"; then AC_DEFINE(OLD_HDA_CMD) fi +AC_MSG_CHECKING(for user_mutex) +if grep -q 'struct mutex user_mutex' hda/hda_codec.h; then + AC_DEFINE(HAVE_USER_MUTEX) + AC_MSG_RESULT(yes) +else + AC_MSG_RESULT(no) +fi + AC_MSG_CHECKING(for new pm_notify callback) ac_save_CFLAGS="$CFLAGS" CFLAGS="$CFLAGS -Iinclude" diff --git a/hda-ctlsh.c b/hda-ctlsh.c index 5ad95fc..04fc318 100644 --- a/hda-ctlsh.c +++ b/hda-ctlsh.c @@ -537,6 +537,11 @@ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key) return -ENOENT; } +int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp) +{ + return -ENOENT; +} + #ifdef HAVE_USER_PINCFGS static void get_drv_pincfgs(void) { diff --git a/include/linux/ctype.h b/include/linux/ctype.h new file mode 100644 index 0000000..8acfe31 --- /dev/null +++ b/include/linux/ctype.h @@ -0,0 +1,64 @@ +#ifndef _LINUX_CTYPE_H +#define _LINUX_CTYPE_H + +/* + * NOTE! This ctype does not handle EOF like the standard C + * library is required to. + */ + +#define _U 0x01 /* upper */ +#define _L 0x02 /* lower */ +#define _D 0x04 /* digit */ +#define _C 0x08 /* cntrl */ +#define _P 0x10 /* punct */ +#define _S 0x20 /* white space (space/lf/tab) */ +#define _X 0x40 /* hex digit */ +#define _SP 0x80 /* hard space (0x20) */ + +extern const unsigned char _ctype[]; + +#define __ismask(x) (_ctype[(int)(unsigned char)(x)]) + +#define isalnum(c) ((__ismask(c)&(_U|_L|_D)) != 0) +#define isalpha(c) ((__ismask(c)&(_U|_L)) != 0) +#define iscntrl(c) ((__ismask(c)&(_C)) != 0) +#define isdigit(c) ((__ismask(c)&(_D)) != 0) +#define isgraph(c) ((__ismask(c)&(_P|_U|_L|_D)) != 0) +#define islower(c) ((__ismask(c)&(_L)) != 0) +#define isprint(c) ((__ismask(c)&(_P|_U|_L|_D|_SP)) != 0) +#define ispunct(c) ((__ismask(c)&(_P)) != 0) +/* Note: isspace() must return false for %NUL-terminator */ +#define isspace(c) ((__ismask(c)&(_S)) != 0) +#define isupper(c) ((__ismask(c)&(_U)) != 0) +#define isxdigit(c) ((__ismask(c)&(_D|_X)) != 0) + +#define isascii(c) (((unsigned char)(c))<=0x7f) +#define toascii(c) (((unsigned char)(c))&0x7f) + +static inline unsigned char __tolower(unsigned char c) +{ + if (isupper(c)) + c -= 'A'-'a'; + return c; +} + +static inline unsigned char __toupper(unsigned char c) +{ + if (islower(c)) + c -= 'a'-'A'; + return c; +} + +#define tolower(c) __tolower(c) +#define toupper(c) __toupper(c) + +/* + * Fast implementation of tolower() for internal usage. Do not use in your + * code. + */ +static inline char _tolower(const char c) +{ + return c | 0x20; +} + +#endif diff --git a/snd-wrapper.c b/snd-wrapper.c index 69cd19c..c2649d9 100644 --- a/snd-wrapper.c +++ b/snd-wrapper.c @@ -27,6 +27,35 @@ #include "hda/hda_codec.h" #include "hda-types.h" #include "hda-log.h" +#include <linux/ctype.h> + +/* Copied from lib/ctype.c */ +const unsigned char _ctype[] = { +_C,_C,_C,_C,_C,_C,_C,_C, /* 0-7 */ +_C,_C|_S,_C|_S,_C|_S,_C|_S,_C|_S,_C,_C, /* 8-15 */ +_C,_C,_C,_C,_C,_C,_C,_C, /* 16-23 */ +_C,_C,_C,_C,_C,_C,_C,_C, /* 24-31 */ +_S|_SP,_P,_P,_P,_P,_P,_P,_P, /* 32-39 */ +_P,_P,_P,_P,_P,_P,_P,_P, /* 40-47 */ +_D,_D,_D,_D,_D,_D,_D,_D, /* 48-55 */ +_D,_D,_P,_P,_P,_P,_P,_P, /* 56-63 */ +_P,_U|_X,_U|_X,_U|_X,_U|_X,_U|_X,_U|_X,_U, /* 64-71 */ +_U,_U,_U,_U,_U,_U,_U,_U, /* 72-79 */ +_U,_U,_U,_U,_U,_U,_U,_U, /* 80-87 */ +_U,_U,_U,_P,_P,_P,_P,_P, /* 88-95 */ +_P,_L|_X,_L|_X,_L|_X,_L|_X,_L|_X,_L|_X,_L, /* 96-103 */ +_L,_L,_L,_L,_L,_L,_L,_L, /* 104-111 */ +_L,_L,_L,_L,_L,_L,_L,_L, /* 112-119 */ +_L,_L,_L,_P,_P,_P,_P,_C, /* 120-127 */ +0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, /* 128-143 */ +0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, /* 144-159 */ +_S|_SP,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P, /* 160-175 */ +_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P, /* 176-191 */ +_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U, /* 192-207 */ +_U,_U,_U,_U,_U,_U,_U,_P,_U,_U,_U,_U,_U,_U,_U,_L, /* 208-223 */ +_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L, /* 224-239 */ +_L,_L,_L,_L,_L,_L,_L,_P,_L,_L,_L,_L,_L,_L,_L,_L}; /* 240-255 */ + int snd_pcm_format_width(int format) { @@ -46,6 +75,9 @@ int snd_pcm_format_width(int format) int snd_hda_create_hwdep(struct hda_codec *codec) { +#ifdef HAVE_USER_MUTEX + mutex_init(&codec->user_mutex); +#endif return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/2] A couple of pending HD-audio patches
by Takashi Iwai 15 Jan '13

15 Jan '13

Hi, these are yet another couple of HD-audio patches, one for merging to 3.8, and another an enhancement for 3.9. Takashi

1 2

[alsa-devel] [PATCH 00/12] Use standard fixup table for STAC/IDT codecs
by Takashi Iwai 15 Jan '13

15 Jan '13

Hi, this is a series of patches to convert the STAC/IDT codec parser to use the standard fixup table. Obviously it's a step forward for the migration of patch_sigmatel.c to the generic parser. Takashi

1 13

[alsa-devel] [PATCH RFC 0/2] Yet more patches for HD-audio generic parser
by Takashi Iwai 15 Jan '13

15 Jan '13

Hi, these two patches have slipped from the previous patch series. Takashi

1 2

[alsa-devel] sound/pci/hda/patch_sigmatel.c:2219:24: warning: unused variable 'spec'
by Fengguang Wu 15 Jan '13

15 Jan '13

Hi Takashi, FYI, there are new compile warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 629965377895ebe6a546184911d588fd29596dc8 commit: 8018ee4e722f6c408d475262b54dc8a73fdd349f ALSA: hda - Use standard fixup table for IDT92HD71Bxx date: 2 hours ago config: make ARCH=i386 defconfig All warnings: sound/pci/hda/patch_sigmatel.c: In function 'stac92hd71bxx_fixup_hp': >> sound/pci/hda/patch_sigmatel.c:2219:24: warning: unused variable 'spec' [-Wunused-variable] vim +/spec +2219 sound/pci/hda/patch_sigmatel.c 8018ee4e Takashi Iwai 2013-01-14 2216 static void stac92hd71bxx_fixup_hp(struct hda_codec *codec, 8018ee4e Takashi Iwai 2013-01-14 2217 const struct hda_fixup *fix, int action) 8018ee4e Takashi Iwai 2013-01-14 2218 { 8018ee4e Takashi Iwai 2013-01-14 @2219 struct sigmatel_spec *spec = codec->spec; 8018ee4e Takashi Iwai 2013-01-14 2220 8018ee4e Takashi Iwai 2013-01-14 2221 if (action != HDA_FIXUP_ACT_PRE_PROBE) 8018ee4e Takashi Iwai 2013-01-14 2222 return; 8018ee4e Takashi Iwai 2013-01-14 2223 8018ee4e Takashi Iwai 2013-01-14 2224 if (hp_blike_system(codec->subsystem_id)) { 8018ee4e Takashi Iwai 2013-01-14 2225 unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f); 8018ee4e Takashi Iwai 2013-01-14 2226 if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT || 8018ee4e Takashi Iwai 2013-01-14 2227 get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER || --- 0-DAY kernel build testing backend Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

2 1

[alsa-devel] sound/pci/hda/patch_sigmatel.c:2260:26: sparse: obsolete array initializer, use C99 syntax
by Fengguang Wu 15 Jan '13

15 Jan '13

Hi Takashi, FYI, there are new sparse warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 629965377895ebe6a546184911d588fd29596dc8 commit: 8018ee4e722f6c408d475262b54dc8a73fdd349f ALSA: hda - Use standard fixup table for IDT92HD71Bxx date: 2 hours ago >> sound/pci/hda/patch_sigmatel.c:2260:26: sparse: obsolete array initializer, use C99 syntax sound/pci/hda/patch_sigmatel.c:2266:26: sparse: obsolete array initializer, use C99 syntax vim +2260 sound/pci/hda/patch_sigmatel.c 8018ee4e Takashi Iwai 2013-01-14 2249 static const struct hda_fixup stac92hd71bxx_fixups[] = { 8018ee4e Takashi Iwai 2013-01-14 2250 [STAC_92HD71BXX_REF] = { 8018ee4e Takashi Iwai 2013-01-14 2251 .type = HDA_FIXUP_FUNC, 8018ee4e Takashi Iwai 2013-01-14 2252 .v.func = stac92hd71bxx_fixup_ref, 8018ee4e Takashi Iwai 2013-01-14 2253 }, 8018ee4e Takashi Iwai 2013-01-14 2254 [STAC_DELL_M4_1] = { 8018ee4e Takashi Iwai 2013-01-14 2255 .type = HDA_FIXUP_PINS, 8018ee4e Takashi Iwai 2013-01-14 2256 .v.pins = dell_m4_1_pin_configs, 8018ee4e Takashi Iwai 2013-01-14 2257 .chained = true, 8018ee4e Takashi Iwai 2013-01-14 2258 .chain_id = STAC_92HD71BXX_NO_SMUX, 8018ee4e Takashi Iwai 2013-01-14 2259 }, 8018ee4e Takashi Iwai 2013-01-14 @2260 [STAC_DELL_M4_2] { 8018ee4e Takashi Iwai 2013-01-14 2261 .type = HDA_FIXUP_PINS, 8018ee4e Takashi Iwai 2013-01-14 2262 .v.pins = dell_m4_2_pin_configs, 8018ee4e Takashi Iwai 2013-01-14 2263 .chained = true, 8018ee4e Takashi Iwai 2013-01-14 2264 .chain_id = STAC_92HD71BXX_NO_DMIC, 8018ee4e Takashi Iwai 2013-01-14 2265 }, 8018ee4e Takashi Iwai 2013-01-14 2266 [STAC_DELL_M4_3] { 8018ee4e Takashi Iwai 2013-01-14 2267 .type = HDA_FIXUP_PINS, 8018ee4e Takashi Iwai 2013-01-14 2268 .v.pins = dell_m4_3_pin_configs, --- 0-DAY kernel build testing backend Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

2 1

[alsa-devel] [PATCH 1/4] ALSA: hdsp - Fix detection for RME RPM/Multiface/Digiface ioboxes
by Adrian Knoth 15 Jan '13

15 Jan '13

The current iobox detection code reportedly fails for various users, so simply do what the Win32 driver does instead. Patch originally by Karl Grill <kgrill(a)chello.at> and then modified to comply with kernel coding guidelines + current HEAD. Signed-off-by: Karl Grill <kgrill(a)chello.at> Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 4fae81f..bcad9ba 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -154,10 +154,11 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_BIGENDIAN_MODE 0x200 #define HDSP_RD_MULTIPLE 0x400 #define HDSP_9652_ENABLE_MIXER 0x800 +#define HDSP_DUNNO 0x1000 #define HDSP_TDO 0x10000000 -#define HDSP_S_PROGRAM (HDSP_PROGRAM|HDSP_CONFIG_MODE_0) -#define HDSP_S_LOAD (HDSP_PROGRAM|HDSP_CONFIG_MODE_1) +#define HDSP_S_PROGRAM (HDSP_DUNNO|HDSP_PROGRAM|HDSP_CONFIG_MODE_0) +#define HDSP_S_LOAD (HDSP_DUNNO|HDSP_PROGRAM|HDSP_CONFIG_MODE_1) /* Control Register bits */ @@ -671,13 +672,23 @@ static unsigned int hdsp_read(struct hdsp *hdsp, int reg) static int hdsp_check_for_iobox (struct hdsp *hdsp) { + int i; + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; - if (hdsp_read (hdsp, HDSP_statusRegister) & HDSP_ConfigError) { - snd_printk("Hammerfall-DSP: no IO box connected!\n"); - hdsp->state &= ~HDSP_FirmwareLoaded; - return -EIO; + for (i = 0; i < 500; i++) { + if (0 == (hdsp_read(hdsp, HDSP_statusRegister) & + HDSP_ConfigError)) { + if (i) { + snd_printk("Hammerfall-DSP: IO box found after %d ms\n", + i); + } + return 0; + } + msleep(20); } - return 0; + snd_printk("Hammerfall-DSP: no IO box connected!\n"); + hdsp->state &= ~HDSP_FirmwareLoaded; + return -EIO; } static int hdsp_wait_for_iobox(struct hdsp *hdsp, unsigned int loops, @@ -728,6 +739,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout waiting for download preparation\n"); + hdsp_write(hdsp, HDSP_control2Reg, 0x800); return -EIO; } @@ -737,17 +749,15 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { hdsp_write(hdsp, HDSP_fifoData, cache[i]); if (hdsp_fifo_wait (hdsp, 127, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout during firmware loading\n"); + hdsp_write(hdsp, HDSP_control2Reg, 0x800); return -EIO; } } - ssleep(3); - - if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { - snd_printk ("Hammerfall-DSP: timeout at end of firmware loading\n"); - return -EIO; - } + hdsp_fifo_wait(hdsp, 3, HDSP_LONG_WAIT); + hdsp_write(hdsp, HDSP_control2Reg, 0x800); + ssleep(3); #ifdef SNDRV_BIG_ENDIAN hdsp->control2_register = HDSP_BIGENDIAN_MODE; #else @@ -773,24 +783,51 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp) { if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - hdsp_write (hdsp, HDSP_control2Reg, HDSP_PROGRAM); - hdsp_write (hdsp, HDSP_fifoData, 0); - if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) < 0) - return -EIO; + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + hdsp_write(hdsp, HDSP_fifoData, 0); - hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT) < 0) { + hdsp_write(hdsp, HDSP_control2Reg, 0x900); + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + } + + hdsp_write(hdsp, HDSP_control2Reg, 0x820); hdsp_write (hdsp, HDSP_fifoData, 0); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT) < 0) { + hdsp->io_type = Multiface; + snd_printk("Hammerfall-DSP: Multiface found\n"); + return 0; + } - if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) { - hdsp_write(hdsp, HDSP_control2Reg, HDSP_VERSION_BIT); - hdsp_write(hdsp, HDSP_control2Reg, HDSP_S_LOAD); - if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) - hdsp->io_type = RPM; - else - hdsp->io_type = Multiface; - } else { + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + hdsp_write(hdsp, HDSP_fifoData, 0); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT) == 0) { hdsp->io_type = Digiface; + snd_printk("Hammerfall-DSP: Digiface found\n"); + return 0; } + + hdsp_write(hdsp, HDSP_control2Reg, 0x900); + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + hdsp_write(hdsp, HDSP_fifoData, 0); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT) == 0) { + hdsp->io_type = Multiface; + snd_printk("Hammerfall-DSP: Multiface found\n"); + return 0; + } + + hdsp_write(hdsp, HDSP_control2Reg, 0x900); + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + hdsp_write(hdsp, HDSP_fifoData, 0); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT) < 0) { + hdsp->io_type = Multiface; + snd_printk("Hammerfall-DSP: Multiface found\n"); + return 0; + } + + hdsp->io_type = RPM; + snd_printk("Hammerfall-DSP: RPM found\n"); + return 0; } else { /* firmware was already loaded, get iobox type */ if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version2) -- 1.7.10.4

2 6

[alsa-devel] ASoC fixes for atmel
by Mark Brown 15 Jan '13

15 Jan '13

The following changes since commit 9931faca02c604c22335f5a935a501bb2ace6e20: Linux 3.8-rc3 (2013-01-09 18:59:55 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-atmel-pinctrl for you to fetch changes up to ea03c81521bde526570e1dec96eaa21fe5ac84a2: ASoC: atmel-ssc: add pinctrl selection to driver (2013-01-12 00:00:33 +0000) ---------------------------------------------------------------- ASoC: atmel: Fixes for pinctrl Due to a series of problems with the handling of Atmel, a combination of making changes that make other branches instantly buggy and a general failure to deal with the resulting issues effectively, v3.8 Atmel audio currently won't work at all for DT boards without adding pinctrl definitions and a request for those. ---------------------------------------------------------------- Bo Shen (2): ARM: at91/dts: add pinctrl support for SSC peripheral ASoC: atmel-ssc: add pinctrl selection to driver arch/arm/boot/dts/at91sam9260.dtsi | 18 ++++++++++++++++++ arch/arm/boot/dts/at91sam9263.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9g45.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9n12.dtsi | 26 ++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9x5.dtsi | 32 +++++++++++++++++++++++++------- drivers/misc/atmel-ssc.c | 8 ++++++++ 6 files changed, 149 insertions(+), 7 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: hda - mute LEDs on HP: also check PCI SSID
by David Henningsson 14 Jan '13

14 Jan '13

If the codec subvendor is not set to HP, maybe the PCI SSID is set to HP. If so, there might still be a valid mute LED string in BIOS/UEFI. Cc: stable(a)vger.kernel.org Buglink: https://bugs.launchpad.net/bugs/1099372 Reported-and-tested-by: Tammy Yang <tammy.yang(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a86547c..40c74a2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4948,7 +4948,8 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) &spec->gpio_led_polarity); return 1; } - if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) { + if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP || + codec->bus->pci->subsystem_vendor == PCI_VENDOR_ID_HP) { while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { if (sscanf(dev->name, "HP_Mute_LED_%d_%x", -- 1.7.9.5

2 3

[alsa-devel] [RFC PATCH] ALSA: usb - fix race in creation of M-Audio Fast track pro driver
by David Henningsson 14 Jan '13

14 Jan '13

A patch in the 3.2 kernel caused regression with hotplugging the M-Audio Fast track pro, or sound after suspend. I don't have the device so I haven't done a full analysis, but it seems userspace (both udev and pulseaudio) got confused when a card was created, immediately destroyed, and then created again. However, at least one person in the bug report (martin djfun) reports that this patch resolves the issue for him. It also leaves a message in the log: "snd-usb-audio: probe of 1-1.1:1.1 failed with error -5" which is a bit misleading. It is better than non-working audio, but maybe there's a more elegant solution? BugLink: https://bugs.launchpad.net/bugs/1095315 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/usb/quirks.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) ...oh, and if you end up taking this patch as it is, you should probably add cc to stable (3.2+). diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index acc12f0..e71fc8b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -387,11 +387,13 @@ static int snd_usb_fasttrackpro_boot_quirk(struct usb_device *dev) * rules */ err = usb_driver_set_configuration(dev, 2); - if (err < 0) { + if (err < 0) snd_printdd("error usb_driver_set_configuration: %d\n", err); - return -ENODEV; - } + /* Always return an error, so that we stop creating a device + that will just be destroyed and recreated with a new + configuration */ + return -ENODEV; } else snd_printk(KERN_INFO "usb-audio: Fast Track Pro config OK\n"); -- 1.7.9.5

3 4

[alsa-devel] [PATCH 0/3] M-Audio C400 corrections
by Eldad Zack 14 Jan '13

14 Jan '13

Hi Takashi, The following are specific fixes for the M-Audio FT C400. * Use the clock selector instead of the internal clock and add names for the clocks instead of Unit 129 and Unit 130. * Fix an issue where the latency would change every time the stream is started. Uses skip_packets. I think they could go in current mainline, since it is only for this device, but I also checked that they apply cleanly against for-next. Cheers, Eldad Eldad Zack (3): ALSA: usb-audio: correct M-Audio C400 clock source quirk ALSA: usb-audio: M-Audio FT C400 skip packet quirk ALSA: usb-audio: selector map for M-Audio FT C400 sound/usb/mixer_maps.c | 13 +++++++++++++ sound/usb/quirks-table.h | 4 ++-- sound/usb/quirks.c | 11 +++++++++++ 3 files changed, 26 insertions(+), 2 deletions(-) -- 1.7.12.4

2 4

Re: [alsa-devel] [RFC PATCH 4/4] alsa/soc: add hdmi audio codec based on cdf
by Sachin Kamat 14 Jan '13

14 Jan '13

+CC: ALSA mailing list, Mark Brown On 13 January 2013 18:22, Rahul Sharma <rahul.sharma(a)samsung.com> wrote: > This patch registers hdmi-audio codec to the ALSA framework. This is the second > client to the hdmi panel. Once notified by the CDF Core it proceeds towards > audio setting and audio control. It also subscribes for hpd notification to > implement hpd related audio requirements. > > Signed-off-by: Rahul Sharma <rahul.sharma(a)samsung.com> > --- > sound/soc/codecs/Kconfig | 4 + > sound/soc/codecs/Makefile | 2 + > sound/soc/codecs/exynos_hdmi_audio.c | 307 +++++++++++++++++++++++++++++++++++ > 3 files changed, 313 insertions(+) > create mode 100644 sound/soc/codecs/exynos_hdmi_audio.c > > diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig > index b92759a..93f3f6b 100644 > --- a/sound/soc/codecs/Kconfig > +++ b/sound/soc/codecs/Kconfig > @@ -496,3 +496,7 @@ config SND_SOC_ML26124 > > config SND_SOC_TPA6130A2 > tristate > + > +config SND_SOC_EXYNOS_HDMI_AUDIO > + tristate > + default y Do you want to enable this by default? Shouldn't this be depending on HDMI support? > diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile > index 9bd4d95..bfe93e6 100644 > --- a/sound/soc/codecs/Makefile > +++ b/sound/soc/codecs/Makefile > @@ -112,6 +112,7 @@ snd-soc-wm9705-objs := wm9705.o > snd-soc-wm9712-objs := wm9712.o > snd-soc-wm9713-objs := wm9713.o > snd-soc-wm-hubs-objs := wm_hubs.o > +snd-soc-exynos-hdmi-audio-objs := exynos_hdmi_audio.o > > # Amp > snd-soc-max9877-objs := max9877.o > @@ -230,6 +231,7 @@ obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o > obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o > obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o > obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o > +obj-$(CONFIG_SND_SOC_EXYNOS_HDMI_AUDIO) += snd-soc-exynos-hdmi-audio.o > > # Amp > obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o > diff --git a/sound/soc/codecs/exynos_hdmi_audio.c b/sound/soc/codecs/exynos_hdmi_audio.c > new file mode 100644 > index 0000000..50e8564 > --- /dev/null > +++ b/sound/soc/codecs/exynos_hdmi_audio.c > @@ -0,0 +1,307 @@ > +/* > + * ALSA SoC codec driver for HDMI audio on Samsung Exynos processors. > + * Copyright (C) 2012 Samsung corp. Copyright (c) 2012 (-13?) Samsung Electronics Co., Ltd. > + * Author: Rahul Sharma <rahul.sharma(a)samsung.com> > + * > + * This program is free software; you can redistribute it and/or > + * modify it under the terms of the GNU General Public License > + * version 2 as published by the Free Software Foundation. > + * > + * This program is distributed in the hope that it will be useful, but > + * WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * General Public License for more details. > + * > + */ > +#include <linux/module.h> > +#include <linux/delay.h> > +#include <sound/soc.h> > +#include <video/display.h> > +#include <video/exynos_hdmi.h> > +#include <sound/pcm.h> > +#include <sound/pcm_params.h> > +#include <linux/of_platform.h> > +#include <linux/platform_device.h> > + > +#undef dev_info > + > +#define dev_info(dev, format, arg...) \ > + dev_printk(KERN_CRIT, dev, format, ##arg) You may directly use dev_crit instead of dev_printk. > + > +static struct snd_soc_codec_driver hdmi_codec; > + > +/* platform device pointer for eynos hdmi audio device. */ > +static struct platform_device *exynos_hdmi_audio_pdev; > + > +struct hdmi_audio_context { > + struct platform_device *pdev; > + atomic_t plugged; > + struct display_entity_audio_params audio_params; > + struct display_entity *entity; > + struct display_entity_notifier notf; > + struct display_event_subscriber subscriber; > +}; > + > +static int exynos_hdmi_audio_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, > + struct snd_soc_dai *dai) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct hdmi_audio_context *ctx = snd_soc_codec_get_drvdata(codec); > + int ret; > + > + dev_info(codec->dev, "[%d] %s\n", __LINE__, __func__); How about making this a debug message as it does not convey anything useful? > + > + ctx->audio_params.type = DISPLAY_ENTITY_AUDIO_I2S; > + > + switch (params_channels(params)) { > + case 6: > + case 4: > + case 2: > + case 1: > + ctx->audio_params.channels = params_channels(params); > + break; > + default: > + dev_err(codec->dev, "%d channels not supported\n", > + params_channels(params)); > + return -EINVAL; > + } > + > + switch (params_format(params)) { > + case SNDRV_PCM_FORMAT_S8: > + ctx->audio_params.bits_per_sample = 8; > + break; > + case SNDRV_PCM_FORMAT_S16_LE: > + ctx->audio_params.bits_per_sample = 12; > + break; > + case SNDRV_PCM_FORMAT_S24_LE: > + ctx->audio_params.bits_per_sample = 16; > + break; > + default: > + dev_err(codec->dev, "Format(%d) not supported\n", > + params_format(params)); > + return -EINVAL; > + } > + > + switch (params_rate(params)) { > + case 32000: > + case 44100: > + case 88200: > + case 176400: > + case 48000: > + case 96000: > + case 192000: > + ctx->audio_params.sf = params_rate(params); > + break; > + default: > + dev_err(codec->dev, "%d Rate supported\n", > + params_rate(params)); > + return -EINVAL; > + } > + > + /* checking here to cache audioparms for hpd plug handling */ > + if (!atomic_read(&ctx->plugged)) > + return -EINVAL; > + > + ret = > + display_entity_hdmi_init_audio(ctx->entity, &ctx->audio_params); > + return ret; > +} > + > +static int exynos_hdmi_audio_trigger(struct snd_pcm_substream *substream, > + int cmd, struct snd_soc_dai *dai) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct hdmi_audio_context *ctx = snd_soc_codec_get_drvdata(codec); > + int ret; > + > + dev_info(codec->dev, "[%d] %s\n", __LINE__, __func__); ditto > + > + /* checking here to cache audioparms for hpd plug handling */ > + if (!atomic_read(&ctx->plugged)) > + return -EINVAL; > + > + switch (cmd) { > + case SNDRV_PCM_TRIGGER_START: > + ret = display_entity_hdmi_set_audiostate(ctx->entity, > + DISPLAY_ENTITY_AUDIOSTATE_ON); > + if (ret) { > + dev_err(codec->dev, "audio enable failed.\n"); > + return -EINVAL; > + } > + break; > + case SNDRV_PCM_TRIGGER_STOP: > + ret = display_entity_hdmi_set_audiostate(ctx->entity, > + DISPLAY_ENTITY_AUDIOSTATE_OFF); > + break; > + default: > + ret = -EINVAL; > + break; > + } > + > + return ret; > +} > + > +static const struct snd_soc_dai_ops exynos_hdmi_audio_dai_ops = { > + .hw_params = exynos_hdmi_audio_hw_params, > + .trigger = exynos_hdmi_audio_trigger, > +}; > + > +static struct snd_soc_dai_driver hdmi_codec_dai = { > + .name = "exynos-hdmi-audio", > + .playback = { > + .channels_min = 2, > + .channels_max = 8, > + .rates = SNDRV_PCM_RATE_32000 | > + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | > + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | > + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, > + .formats = SNDRV_PCM_FMTBIT_S16_LE | > + SNDRV_PCM_FMTBIT_S24_LE, > + }, > + .ops = &exynos_hdmi_audio_dai_ops, > +}; > + > +void hdmi_audio_event_notify(struct display_entity *entity, > + enum display_entity_event_type type, > + unsigned int value, void *context) > +{ > + struct hdmi_audio_context *ctx = (struct hdmi_audio_context *)context; > + > + if (type == DISPLAY_ENTITY_HDMI_HOTPLUG) { > + dev_info(&ctx->pdev->dev, "[%d][%s] hpd(%d)\n", __LINE__, > + __func__, value); > + atomic_set(&ctx->plugged, !!value); > + } > +} > + > +int exynos_hdmi_audio_notification(struct display_entity_notifier *notf, > + struct display_entity *entity, int status) > +{ > + struct hdmi_audio_context *ctx = container_of(notf, > + struct hdmi_audio_context, notf); > + struct exynos_hdmi_control_ops *exynos_ops = > + (struct exynos_hdmi_control_ops *)entity->private; > + int hpd; > + > + if (status != DISPLAY_ENTITY_NOTIFIER_CONNECT && entity) > + return -EINVAL; > + > + dev_info(&ctx->pdev->dev, "[%d][%s]\n", __LINE__, __func__); > + > + ctx->entity = entity; > + > + ctx->subscriber.context = ctx; > + ctx->subscriber.notify = hdmi_audio_event_notify; > + > + display_entity_subscribe_event(entity, &ctx->subscriber); > + > + exynos_ops->get_hpdstate(entity, &hpd); > + atomic_set(&ctx->plugged, !!hpd); > + > + return 0; > +} > + > +static __devinit int hdmi_codec_probe(struct platform_device *pdev) __devinit is not necessary. +{ > + int ret; > + struct hdmi_audio_context *ctx; > + struct device_node *dev_node; > + struct platform_device *disp_pdev; > + > + dev_info(&pdev->dev, "[%d][%s]\n", __LINE__, __func__); > + > + ret = snd_soc_register_codec(&pdev->dev, &hdmi_codec, > + &hdmi_codec_dai, 1); > + > + if (ret) { > + dev_err(&pdev->dev, "register_codec failed (%d)\n", ret); > + return ret; > + } > + > + ctx = devm_kzalloc(&pdev->dev, sizeof(struct hdmi_audio_context), > + GFP_KERNEL); > + if (ctx == NULL) > + return -ENOMEM; > + > + ctx->pdev = pdev; > + atomic_set(&ctx->plugged, 0); > + platform_set_drvdata(pdev, ctx); > + > + dev_node = of_find_compatible_node(NULL, NULL, > + "samsung,exynos5-hdmi"); > + if (!dev_node) { > + dev_err(&pdev->dev, "[%d][%s] dt node not found.\n", > + __LINE__, __func__); > + return -EINVAL; > + } > + > + disp_pdev = of_find_device_by_node(dev_node); > + if (!disp_pdev) { > + dev_err(&pdev->dev, "[ERROR][%d][%s] No pdev\n", > + __LINE__, __func__); > + return -EINVAL; > + } > + > + ctx->notf.dev = &disp_pdev->dev; > + ctx->notf.notify = exynos_hdmi_audio_notification; > + > + ret = display_entity_register_notifier(&ctx->notf); > + if (ret) { > + dev_err(&pdev->dev, "[%d][%s] entity registe failed.\n", > + __LINE__, __func__); > + return -EINVAL; > + } > + return ret; > +} > + > +static __devexit int hdmi_codec_remove(struct platform_device *pdev) __devexit is not necessary. > +{ > + dev_info(&pdev->dev, " %s:%s:%d", __FILE__, __func__, __LINE__); > + mdelay(1000); > + > + snd_soc_unregister_codec(&pdev->dev); > + return 0; > +} > + > +static struct platform_driver hdmi_codec_driver = { > + .driver = { > + .name = "exynos-hdmi-audio-codec", > + .owner = THIS_MODULE, > + }, > + > + .probe = hdmi_codec_probe, > + .remove = __devexit_p(hdmi_codec_remove), > +}; > + > +static int __init hdmi_codec_init(void) > +{ > + int ret; > + > + ret = platform_driver_register(&hdmi_codec_driver); > + if (ret < 0) > + return -EINVAL; > + > + exynos_hdmi_audio_pdev = platform_device_register_simple > + ("exynos-hdmi-audio-codec", -1, NULL, 0); > + if (IS_ERR_OR_NULL(exynos_hdmi_audio_pdev)) { > + ret = PTR_ERR(exynos_hdmi_audio_pdev); > + platform_driver_unregister(&hdmi_codec_driver); > + return ret; > + } > + > + return 0; > +} > +static void __exit hdmi_codec_exit(void) > +{ > + platform_driver_unregister(&hdmi_codec_driver); > + platform_device_unregister(exynos_hdmi_audio_pdev); > +} > + > +module_init(hdmi_codec_init); > +module_exit(hdmi_codec_exit); > + > +MODULE_AUTHOR("Rahul Sharma <rahul.sharma(a)samsung.com>"); > +MODULE_DESCRIPTION("ASoC EXYNOS HDMI codec driver"); > +MODULE_LICENSE("GPL"); > +MODULE_ALIAS("platform:" DRV_NAME); > -- > 1.8.0 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-media" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html -- With warm regards, Sachin

2 1

[alsa-devel] [PATCH 0/2] ASoC: fsi: add DT support
by Kuninori Morimoto 14 Jan '13

14 Jan '13

Hi Mark These patches adds device tree support on FSI driver These are based on mark/topic/fsi branch Kuninori Morimoto (2): ASoC: fsi: don't use platform info pointer on probe() ASoC: fsi: add device tree support .../devicetree/bindings/sound/renesas,fsi.txt | 23 +++++ sound/soc/sh/fsi.c | 98 ++++++++++++++++---- 2 files changed, 104 insertions(+), 17 deletions(-) Best regards --- Kuninori Morimoto

2 7

Re: [alsa-devel] [PATCH v2 3/3] ASoC: tegra: add tegra machine driver using wm9712 codec
by Mark Brown 14 Jan '13

14 Jan '13

On Tue, Jan 08, 2013 at 11:52:33PM +0100, Lucas Stach wrote: > Am Dienstag, den 08.01.2013, 15:17 -0700 schrieb Stephen Warren: > > Hmmm. I thought that could be auto-instantiated based on probing the > > AC'97 bus for whatever was there? > AC97 is in principle probeable by reading the first regs of the codec, > but all the infrastructure to do proper hotplugging on the AC97 bus is > missing in the kernel. As I didn't want to be the one to add all this > infrastructure to a dying audio platform, I opted to do the same thing > as all the other platform using AC97 and instantiate the codec from the > machine driver. That's about the size of it, yes. We should write the device tree as though we'd got better infrastructure here since this isn't something that ought to be required but right now this is how you do AC'97 in ASoC. It'd be really nice if someone had the time to work on the AC'97 infrastructure here but I'd not hold my breath given that it's rarely used in new designs.

1 0

[alsa-devel] [PATCH 1/8] regmap: Add support for 24 bit wide register addresses
by Lars-Peter Clausen 14 Jan '13

14 Jan '13

Since regmap already has support for formatting 24 bit wide values, so adding support for 24 bit wide registers is pretty much straight forward. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- drivers/base/regmap/regmap.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 42d5cb0..26af93a 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -500,6 +500,12 @@ struct regmap *regmap_init(struct device *dev, } break; + case 24: + if (reg_endian != REGMAP_ENDIAN_BIG) + goto err_map; + map->format.format_reg = regmap_format_24; + break; + case 32: switch (reg_endian) { case REGMAP_ENDIAN_BIG: -- 1.8.0

2 17

[alsa-devel] [PATCH] ASoC: simple-card: add asoc_simple_dai for initializing
by Kuninori Morimoto 13 Jan '13

13 Jan '13

Current simple-card driver calls asoc_simple_card_dai_init() if platform had a asoc_simple_card_dai_init pointer. And, this initialization function works only when platform has an applicable initial value for each dai settings. And basically, almost all sound card requires certain initialization. This means that almost all platform has initialization settings, and driver do nothing if it doesn't have settings. And additionally, current simple-card supports sysclk settings but it was only for codec. In order to abolish deviation between cpu and codec, and in order to simplify processing, this patch adds asoc_simple_dai, and removed pointless struct asoc_simple_dai_init_info which was trigger of calling asoc_simple_card_dai_init(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- >> Mark This patch is based on both mark/topic/simple-card and mark/topic/fsi. I used "git merge". Please let me know if I should rebase it. >> Simon Please give this patch your Acked-by, since it modifies mach-shmobile. arch/arm/mach-shmobile/board-ap4evb.c | 35 +++++++------- arch/arm/mach-shmobile/board-armadillo800eva.c | 34 +++++++------- arch/arm/mach-shmobile/board-kzm9g.c | 20 ++++---- arch/arm/mach-shmobile/board-mackerel.c | 35 +++++++------- arch/sh/boards/mach-ecovec24/setup.c | 19 ++++---- arch/sh/boards/mach-se/7724/setup.c | 21 ++++----- include/sound/simple_card.h | 12 ++--- sound/soc/generic/simple-card.c | 58 ++++++++++++------------ 8 files changed, 116 insertions(+), 118 deletions(-) diff --git a/arch/arm/mach-shmobile/board-ap4evb.c b/arch/arm/mach-shmobile/board-ap4evb.c index 4c97903..08294fa 100644 --- a/arch/arm/mach-shmobile/board-ap4evb.c +++ b/arch/arm/mach-shmobile/board-ap4evb.c @@ -686,21 +686,21 @@ static struct platform_device fsi_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_ak4643_init_info = { - .fmt = SND_SOC_DAIFMT_LEFT_J, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, - .sysclk = 11289600, -}; - static struct asoc_simple_card_info fsi2_ak4643_info = { .name = "AK4643", .card = "FSI2A-AK4643", - .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0013", .platform = "sh_fsi2", - .codec_dai = "ak4642-hifi", - .init = &fsi2_ak4643_init_info, + .daifmt = SND_SOC_DAIFMT_LEFT_J, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS, + }, + .codec_dai = { + .name = "ak4642-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + .sysclk = 11289600, + }, }; static struct platform_device fsi_ak4643_device = { @@ -809,19 +809,18 @@ static struct platform_device lcdc1_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = { - .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_NF, -}; - static struct asoc_simple_card_info fsi2_hdmi_info = { .name = "HDMI", .card = "FSI2B-HDMI", - .cpu_dai = "fsib-dai", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", - .codec_dai = "sh_mobile_hdmi-hifi", - .init = &fsi2_hdmi_init_info, + .cpu_dai = { + .name = "fsib-dai", + .fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "sh_mobile_hdmi-hifi", + }, }; static struct platform_device fsi_hdmi_device = { diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c index 042a21e..22a1cc5 100644 --- a/arch/arm/mach-shmobile/board-armadillo800eva.c +++ b/arch/arm/mach-shmobile/board-armadillo800eva.c @@ -807,21 +807,21 @@ static struct platform_device fsi_device = { }; /* FSI-WM8978 */ -static struct asoc_simple_dai_init_info fsi_wm8978_init_info = { - .fmt = SND_SOC_DAIFMT_I2S, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_NB_NF, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, - .sysclk = 12288000, -}; - static struct asoc_simple_card_info fsi_wm8978_info = { .name = "wm8978", .card = "FSI2A-WM8978", - .cpu_dai = "fsia-dai", .codec = "wm8978.0-001a", .platform = "sh_fsi2", - .codec_dai = "wm8978-hifi", - .init = &fsi_wm8978_init_info, + .daifmt = SND_SOC_DAIFMT_I2S, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "wm8978-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_NB_NF, + .sysclk = 12288000, + }, }; static struct platform_device fsi_wm8978_device = { @@ -833,18 +833,18 @@ static struct platform_device fsi_wm8978_device = { }; /* FSI-HDMI */ -static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = { - .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM, -}; - static struct asoc_simple_card_info fsi2_hdmi_info = { .name = "HDMI", .card = "FSI2B-HDMI", - .cpu_dai = "fsib-dai", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", - .codec_dai = "sh_mobile_hdmi-hifi", - .init = &fsi2_hdmi_init_info, + .cpu_dai = { + .name = "fsib-dai", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + }, + .codec_dai = { + .name = "sh_mobile_hdmi-hifi", + }, }; static struct platform_device fsi_hdmi_device = { diff --git a/arch/arm/mach-shmobile/board-kzm9g.c b/arch/arm/mach-shmobile/board-kzm9g.c index 9ca7ce5..f170aed 100644 --- a/arch/arm/mach-shmobile/board-kzm9g.c +++ b/arch/arm/mach-shmobile/board-kzm9g.c @@ -526,21 +526,21 @@ static struct platform_device fsi_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_ak4648_init_info = { - .fmt = SND_SOC_DAIFMT_LEFT_J, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, - .sysclk = 11289600, -}; - static struct asoc_simple_card_info fsi2_ak4648_info = { .name = "AK4648", .card = "FSI2A-AK4648", - .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0012", .platform = "sh_fsi2", - .codec_dai = "ak4642-hifi", - .init = &fsi2_ak4648_init_info, + .daifmt = SND_SOC_DAIFMT_LEFT_J, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS, + }, + .codec_dai = { + .name = "ak4642-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + .sysclk = 11289600, + }, }; static struct platform_device fsi_ak4648_device = { diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c index b5d210b..3fd716d 100644 --- a/arch/arm/mach-shmobile/board-mackerel.c +++ b/arch/arm/mach-shmobile/board-mackerel.c @@ -502,19 +502,18 @@ static struct platform_device hdmi_lcdc_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = { - .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_NF, -}; - static struct asoc_simple_card_info fsi2_hdmi_info = { .name = "HDMI", .card = "FSI2B-HDMI", - .cpu_dai = "fsib-dai", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", - .codec_dai = "sh_mobile_hdmi-hifi", - .init = &fsi2_hdmi_init_info, + .cpu_dai = { + .name = "fsib-dai", + .fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "sh_mobile_hdmi-hifi", + }, }; static struct platform_device fsi_hdmi_device = { @@ -893,21 +892,21 @@ static struct platform_device fsi_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_ak4643_init_info = { - .fmt = SND_SOC_DAIFMT_LEFT_J, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, - .sysclk = 11289600, -}; - static struct asoc_simple_card_info fsi2_ak4643_info = { .name = "AK4643", .card = "FSI2A-AK4643", - .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0013", .platform = "sh_fsi2", - .codec_dai = "ak4642-hifi", - .init = &fsi2_ak4643_init_info, + .daifmt = SND_SOC_DAIFMT_LEFT_J, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS, + }, + .codec_dai = { + .name = "ak4642-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + .sysclk = 11289600, + }, }; static struct platform_device fsi_ak4643_device = { diff --git a/arch/sh/boards/mach-ecovec24/setup.c b/arch/sh/boards/mach-ecovec24/setup.c index 8ebe4c7..065e9600 100644 --- a/arch/sh/boards/mach-ecovec24/setup.c +++ b/arch/sh/boards/mach-ecovec24/setup.c @@ -897,21 +897,20 @@ static struct platform_device fsi_device = { .resource = fsi_resources, }; -static struct asoc_simple_dai_init_info fsi_da7210_init_info = { - .fmt = SND_SOC_DAIFMT_I2S, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS | - SND_SOC_DAIFMT_IB_NF, -}; - static struct asoc_simple_card_info fsi_da7210_info = { .name = "DA7210", .card = "FSIB-DA7210", - .cpu_dai = "fsib-dai", .codec = "da7210.0-001a", .platform = "sh_fsi.0", - .codec_dai = "da7210-hifi", - .init = &fsi_da7210_init_info, + .daifmt = SND_SOC_DAIFMT_I2S, + .cpu_dai = { + .name = "fsib-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "da7210-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + }, }; static struct platform_device fsi_da7210_device = { diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index 975608f..4010e63 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -299,22 +299,21 @@ static struct platform_device fsi_device = { .resource = fsi_resources, }; -static struct asoc_simple_dai_init_info fsi2_ak4642_init_info = { - .fmt = SND_SOC_DAIFMT_LEFT_J, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS | - SND_SOC_DAIFMT_IB_NF, - .sysclk = 11289600, -}; - static struct asoc_simple_card_info fsi_ak4642_info = { .name = "AK4642", .card = "FSIA-AK4642", - .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0012", .platform = "sh_fsi.0", - .codec_dai = "ak4642-hifi", - .init = &fsi2_ak4642_init_info, + .daifmt = SND_SOC_DAIFMT_LEFT_J, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "ak4642-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + .sysclk = 11289600, + }, }; static struct platform_device fsi_ak4642_device = { diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 4b62b8d..6c74527 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -14,21 +14,21 @@ #include <sound/soc.h> -struct asoc_simple_dai_init_info { +struct asoc_simple_dai { + const char *name; unsigned int fmt; - unsigned int cpu_daifmt; - unsigned int codec_daifmt; unsigned int sysclk; }; struct asoc_simple_card_info { const char *name; const char *card; - const char *cpu_dai; const char *codec; const char *platform; - const char *codec_dai; - struct asoc_simple_dai_init_info *init; /* for snd_link.init */ + + 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; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index bc050ec..6cf8355 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -16,33 +16,38 @@ #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) +{ + int ret = 0; + + daifmt |= set->fmt; + + if (!ret && daifmt) + ret = snd_soc_dai_set_fmt(dai, daifmt); + + if (!ret && set->sysclk) + ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); + + return ret; +} + static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *cinfo = asoc_simple_get_card_info(rtd); - struct asoc_simple_dai_init_info *iinfo = cinfo->init; + struct asoc_simple_card_info *info = asoc_simple_get_card_info(rtd); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; - unsigned int cpu_daifmt = iinfo->fmt | iinfo->cpu_daifmt; - unsigned int codec_daifmt = iinfo->fmt | iinfo->codec_daifmt; + unsigned int daifmt = info->daifmt; int ret; - if (codec_daifmt) { - ret = snd_soc_dai_set_fmt(codec, codec_daifmt); - if (ret < 0) - return ret; - } - - if (iinfo->sysclk) { - ret = snd_soc_dai_set_sysclk(codec, 0, iinfo->sysclk, 0); - if (ret < 0) - return ret; - } + ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt); + if (ret < 0) + return ret; - if (cpu_daifmt) { - ret = snd_soc_dai_set_fmt(cpu, cpu_daifmt); - if (ret < 0) - return ret; - } + ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt); + if (ret < 0) + return ret; return 0; } @@ -59,10 +64,10 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (!cinfo->name || !cinfo->card || - !cinfo->cpu_dai || !cinfo->codec || !cinfo->platform || - !cinfo->codec_dai) { + !cinfo->cpu_dai.name || + !cinfo->codec_dai.name) { dev_err(dev, "insufficient asoc_simple_card_info settings\n"); return -EINVAL; } @@ -72,14 +77,11 @@ static int asoc_simple_card_probe(struct platform_device *pdev) */ cinfo->snd_link.name = cinfo->name; cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai; + 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; - - /* enable snd_link.init if cinfo has settings */ - if (cinfo->init) - cinfo->snd_link.init = asoc_simple_card_dai_init; + cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; + cinfo->snd_link.init = asoc_simple_card_dai_init; /* * init snd_soc_card -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: tlv320dac33: Remove suspend/resume soc driver operations
by Peter Ujfalusi 13 Jan '13

13 Jan '13

With idle_bias_off these are no longer needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320dac33.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 782b0cd..4f35839 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1452,20 +1452,6 @@ static int dac33_soc_remove(struct snd_soc_codec *codec) return 0; } -static int dac33_soc_suspend(struct snd_soc_codec *codec) -{ - dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int dac33_soc_resume(struct snd_soc_codec *codec) -{ - dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { .read = dac33_read_reg_cache, .write = dac33_write_locked, @@ -1476,8 +1462,6 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { .reg_cache_default = dac33_reg, .probe = dac33_soc_probe, .remove = dac33_soc_remove, - .suspend = dac33_soc_suspend, - .resume = dac33_soc_resume, .controls = dac33_snd_controls, .num_controls = ARRAY_SIZE(dac33_snd_controls), -- 1.8.1

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Provide explicit trace of coefficient writes
by Mark Brown 12 Jan '13

12 Jan '13

Helpful for debugging. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4a233b8..14af8e2 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -775,6 +775,9 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) return -ENOMEM; } + adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n", + file, blocks, le32_to_cpu(blk->len), + reg); ret = regmap_raw_write(regmap, reg, blk->data, le32_to_cpu(blk->len)); if (ret != 0) { -- 1.7.10.4

1 0

[alsa-devel] [PATCH 0/2 v2] at91/ssc: fixes on ASoC tree for 3.8
by Nicolas Ferre 12 Jan '13

12 Jan '13

Hi Mark, Olof, Arnd, This series goes on top of Linus' v3.8-rc3 and fixes an error that we have while compiling DTBs for AT91: ERROR (phandle_references): Reference to non-existent node or label "pinctrl_ssc0_tx" ERROR: Input tree has errors, aborting (use -f to force output) make[3]: *** [arch/arm/boot/dts/at91sam9g20ek.dtb] Error 2 I have included pending material that is dealing with SSC and pinctrl. The pinctrl part can be merged more easily now that the big pinctrl update for AT91 has been merged upstream. This material was designed to enter Mark's fixes queue, but as discussed with Olof, we can imagine merging everything through arm-soc or split the series (of 2 patches) and let them progress upstream separated (option that I do not like even if I know that the consequences are not so dramatic). So please, Olof, if you feel confortable with this series, tell us what you prefer and we will make our best to make this material go forward... Thanks for your help, best regards, v2: - remove the fix for typo in SSC status property: already merged upstream Bo Shen (2): ARM: at91/dts: add pinctrl support for SSC peripheral ASoC: atmel-ssc: add pinctrl selection to driver arch/arm/boot/dts/at91sam9260.dtsi | 18 ++++++++++++++++++ arch/arm/boot/dts/at91sam9263.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9g45.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9n12.dtsi | 26 ++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9x5.dtsi | 32 +++++++++++++++++++++++++------- drivers/misc/atmel-ssc.c | 8 ++++++++ 6 files changed, 149 insertions(+), 7 deletions(-) -- 1.8.0

4 7

[alsa-devel] sound/pci/hda/hda_hwdep.c:487 parse_hints() warn: inconsistent returns mutex:&codec->user_mutex: locked (478) unlocked (447, 449, 452, 457, 487)
by Fengguang Wu 12 Jan '13

12 Jan '13

Hi Takashi, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 02e5aa7b459f69538d4ec88658d064816a7692d6 commit: bea06a147981e88465f566acd6c6e98b297f7712 ALSA: hda - Protect user-defined arrays via mutex date: 3 hours ago sound/pci/hda/hda_hwdep.c:487 parse_hints() warn: inconsistent returns mutex:&codec->user_mutex: locked (478) unlocked (447,449,452,457,487) vim +487 sound/pci/hda/hda_hwdep.c 43b62713 Takashi Iwai 2009-03-02 441 char *key, *val; 43b62713 Takashi Iwai 2009-03-02 442 struct hda_hint *hint; bea06a14 Takashi Iwai 2013-01-10 443 int err = 0; 1e1be432 Takashi Iwai 2008-07-30 444 e7d2860b André Goddard Rosa 2009-12-14 445 buf = skip_spaces(buf); 43b62713 Takashi Iwai 2009-03-02 446 if (!*buf || *buf == '#' || *buf == '\n') 4ea6fbc8 Takashi Iwai 2009-06-17 @447 return 0; 43b62713 Takashi Iwai 2009-03-02 448 if (*buf == '=') 43b62713 Takashi Iwai 2009-03-02 @449 return -EINVAL; 43b62713 Takashi Iwai 2009-03-02 450 key = kstrndup_noeol(buf, 1024); 43b62713 Takashi Iwai 2009-03-02 451 if (!key) 1e1be432 Takashi Iwai 2008-07-30 @452 return -ENOMEM; 43b62713 Takashi Iwai 2009-03-02 453 /* extract key and val */ 43b62713 Takashi Iwai 2009-03-02 454 val = strchr(key, '='); 43b62713 Takashi Iwai 2009-03-02 455 if (!val) { 43b62713 Takashi Iwai 2009-03-02 456 kfree(key); 43b62713 Takashi Iwai 2009-03-02 @457 return -EINVAL; 43b62713 Takashi Iwai 2009-03-02 458 } 43b62713 Takashi Iwai 2009-03-02 459 *val++ = 0; e7d2860b André Goddard Rosa 2009-12-14 460 val = skip_spaces(val); 43b62713 Takashi Iwai 2009-03-02 461 remove_trail_spaces(key); 43b62713 Takashi Iwai 2009-03-02 462 remove_trail_spaces(val); bea06a14 Takashi Iwai 2013-01-10 463 mutex_lock(&codec->user_mutex); 43b62713 Takashi Iwai 2009-03-02 464 hint = get_hint(codec, key); 43b62713 Takashi Iwai 2009-03-02 465 if (hint) { 43b62713 Takashi Iwai 2009-03-02 466 /* replace */ 43b62713 Takashi Iwai 2009-03-02 467 kfree(hint->key); 43b62713 Takashi Iwai 2009-03-02 468 hint->key = key; 43b62713 Takashi Iwai 2009-03-02 469 hint->val = val; bea06a14 Takashi Iwai 2013-01-10 470 goto unlock; 43b62713 Takashi Iwai 2009-03-02 471 } 43b62713 Takashi Iwai 2009-03-02 472 /* allocate a new hint entry */ 43b62713 Takashi Iwai 2009-03-02 473 if (codec->hints.used >= MAX_HINTS) 43b62713 Takashi Iwai 2009-03-02 474 hint = NULL; 43b62713 Takashi Iwai 2009-03-02 475 else 43b62713 Takashi Iwai 2009-03-02 476 hint = snd_array_new(&codec->hints); 1e1be432 Takashi Iwai 2008-07-30 477 if (!hint) { 1e1be432 Takashi Iwai 2008-07-30 @478 return -ENOMEM; bea06a14 Takashi Iwai 2013-01-10 479 goto unlock; 1e1be432 Takashi Iwai 2008-07-30 480 } 43b62713 Takashi Iwai 2009-03-02 481 hint->key = key; 43b62713 Takashi Iwai 2009-03-02 482 hint->val = val; bea06a14 Takashi Iwai 2013-01-10 483 unlock: bea06a14 Takashi Iwai 2013-01-10 484 mutex_unlock(&codec->user_mutex); bea06a14 Takashi Iwai 2013-01-10 485 if (err) bea06a14 Takashi Iwai 2013-01-10 486 kfree(key); bea06a14 Takashi Iwai 2013-01-10 @487 return err; 4ea6fbc8 Takashi Iwai 2009-06-17 488 } 4ea6fbc8 Takashi Iwai 2009-06-17 489 4ea6fbc8 Takashi Iwai 2009-06-17 490 static ssize_t hints_store(struct device *dev, --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

2 1

[alsa-devel] sound/pci/hda/hda_generic.c:1529 indep_hp_put() warn: inconsistent returns mutex:&spec->pcm_mutex: locked (1526) unlocked (1517, 1529)
by Fengguang Wu 12 Jan '13

12 Jan '13

Hi Takashi, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 02e5aa7b459f69538d4ec88658d064816a7692d6 commit: a5de6f4d021608c50ff5ff51d208b5030da92957 ALSA: hda - Implement independent HP control date: 33 hours ago sound/pci/hda/hda_generic.c:1529 indep_hp_put() warn: inconsistent returns mutex:&spec->pcm_mutex: locked (1526) unlocked (1517,1529) vim +1529 sound/pci/hda/hda_generic.c a5de6f4d Takashi Iwai 2012-12-21 1511 struct hda_gen_spec *spec = codec->spec; a5de6f4d Takashi Iwai 2012-12-21 1512 unsigned int select = ucontrol->value.enumerated.item[0]; a5de6f4d Takashi Iwai 2012-12-21 1513 a5de6f4d Takashi Iwai 2012-12-21 1514 mutex_lock(&spec->pcm_mutex); a5de6f4d Takashi Iwai 2012-12-21 1515 if (spec->active_streams) { a5de6f4d Takashi Iwai 2012-12-21 1516 mutex_unlock(&spec->pcm_mutex); a5de6f4d Takashi Iwai 2012-12-21 @1517 return -EBUSY; a5de6f4d Takashi Iwai 2012-12-21 1518 } a5de6f4d Takashi Iwai 2012-12-21 1519 a5de6f4d Takashi Iwai 2012-12-21 1520 if (spec->indep_hp_enabled != select) { a5de6f4d Takashi Iwai 2012-12-21 1521 spec->indep_hp_enabled = select; a5de6f4d Takashi Iwai 2012-12-21 1522 if (spec->indep_hp_enabled) a5de6f4d Takashi Iwai 2012-12-21 1523 spec->multiout.hp_out_nid[0] = 0; a5de6f4d Takashi Iwai 2012-12-21 1524 else a5de6f4d Takashi Iwai 2012-12-21 1525 spec->multiout.hp_out_nid[0] = spec->alt_dac_nid; a5de6f4d Takashi Iwai 2012-12-21 @1526 return 1; a5de6f4d Takashi Iwai 2012-12-21 1527 } a5de6f4d Takashi Iwai 2012-12-21 1528 mutex_unlock(&spec->pcm_mutex); a5de6f4d Takashi Iwai 2012-12-21 @1529 return 0; a5de6f4d Takashi Iwai 2012-12-21 1530 } a5de6f4d Takashi Iwai 2012-12-21 1531 a5de6f4d Takashi Iwai 2012-12-21 1532 static const struct snd_kcontrol_new indep_hp_ctl = { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] sound/pci/hda/hda_codec.c:3433 snd_hda_codec_write_cache() error: we previously assumed 'c' could be null (see line 3430)
by Fengguang Wu 12 Jan '13

12 Jan '13

Hi Takashi, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 02e5aa7b459f69538d4ec88658d064816a7692d6 commit: 95836563e7f636fc57732f5e687ca2f170eb674f ALSA: hda - Introduce cache & flush cmd / amp writes date: 33 hours ago sound/pci/hda/hda_codec.c:3433 snd_hda_codec_write_cache() error: we previously assumed 'c' could be null (see line 3430) vim +/c +3433 sound/pci/hda/hda_codec.c aa2936f5 Takashi Iwai 2009-05-26 3424 /* parm may contain the verb stuff for get/set amp */ aa2936f5 Takashi Iwai 2009-05-26 3425 verb = verb | (parm >> 8); aa2936f5 Takashi Iwai 2009-05-26 3426 parm &= 0xff; aa2936f5 Takashi Iwai 2009-05-26 3427 key = build_cmd_cache_key(nid, verb); aa2936f5 Takashi Iwai 2009-05-26 3428 mutex_lock(&codec->bus->cmd_mutex); aa2936f5 Takashi Iwai 2009-05-26 3429 c = get_alloc_hash(&codec->cmd_cache, key); aa2936f5 Takashi Iwai 2009-05-26 @3430 if (c) aa2936f5 Takashi Iwai 2009-05-26 3431 c->val = parm; 95836563 Takashi Iwai 2012-12-13 3432 if (codec->cached_write) 95836563 Takashi Iwai 2012-12-13 @3433 c->dirty = 1; aa2936f5 Takashi Iwai 2009-05-26 3434 mutex_unlock(&codec->bus->cmd_mutex); aa2936f5 Takashi Iwai 2009-05-26 3435 return 0; b3ac5636 Takashi Iwai 2007-08-10 3436 } --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] [RESEND PATCH] ASoC: OMAP: HDMI: Initialize IEC-60958 channel status word
by Ricardo Neri 12 Jan '13

12 Jan '13

As the IEC-60958 channel status word is set by ANDing and ORing with the appropriate definitions, the word bytes need to be initialized to zero to avoid misconfiguration due to previous hw_params calls. Signed-off-by: Ricardo Neri <rneri(a)dextratech.com> --- sound/soc/omap/omap-hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c index f59c69f..75ad378 100644 --- a/sound/soc/omap/omap-hdmi.c +++ b/sound/soc/omap/omap-hdmi.c @@ -110,6 +110,8 @@ static int omap_hdmi_dai_hw_params(struct snd_pcm_substream *substream, /* * fill the IEC-60958 channel status word */ + /* initialize the word bytes */ + memset(iec->status, 0, sizeof(iec->status)); /* specify IEC-60958-3 (commercial use) */ iec->status[0] &= ~IEC958_AES0_PROFESSIONAL; -- 1.7.10.4

2 2

[alsa-devel] [patch] ASoC: pcm: delete some dead code
by Dan Carpenter 12 Jan '13

12 Jan '13

I've removed several unreachable returns. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index cf191e6..1d6af4d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1729,20 +1729,16 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) /* startup must always be called for new BEs */ ret = dpcm_be_dai_startup(fe, stream); - if (ret < 0) { + if (ret < 0) goto disconnect; - return ret; - } /* keep going if FE state is > open */ if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_OPEN) return 0; ret = dpcm_be_dai_hw_params(fe, stream); - if (ret < 0) { + if (ret < 0) goto close; - return ret; - } /* keep going if FE state is > hw_params */ if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_HW_PARAMS) @@ -1750,10 +1746,8 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) ret = dpcm_be_dai_prepare(fe, stream); - if (ret < 0) { + if (ret < 0) goto hw_free; - return ret; - } /* run the stream event for each BE */ dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_NOP);

2 1

[alsa-devel] [PATCH] ASoC: fsl: fix multiple definition of init_module
by Shawn Guo 12 Jan '13

12 Jan '13

With commit f2818d0 (ASoC: fsl: fix miscompilation of snd-soc-imx-pcm), we will see the following build error when building modules with CONFIG_SND_IMX_SOC=m in imx_v6_v7_defconfig. CC [M] sound/soc/fsl/phycore-ac97.o LD [M] sound/soc/fsl/snd-soc-fsl-ssi.o LD [M] sound/soc/fsl/snd-soc-fsl-utils.o LD [M] sound/soc/fsl/snd-soc-imx-ssi.o LD [M] sound/soc/fsl/snd-soc-imx-audmux.o LD [M] sound/soc/fsl/snd-soc-imx-pcm.o sound/soc/fsl/imx-pcm-dma.o: In function `init_module': imx-pcm-dma.c:(.init.text+0x0): multiple definition of `init_module' sound/soc/fsl/imx-pcm-fiq.o:imx-pcm-fiq.c:(.init.text+0x0): first defined here sound/soc/fsl/imx-pcm-dma.o: In function `cleanup_module': imx-pcm-dma.c:(.exit.text+0x0): multiple definition of `cleanup_module' sound/soc/fsl/imx-pcm-fiq.o:imx-pcm-fiq.c:(.exit.text+0x0): first defined here make[4]: *** [sound/soc/fsl/snd-soc-imx-pcm.o] Error 1 Instead of using bool for SND_SOC_IMX_PCM_FIQ and SND_SOC_IMX_PCM_DMA to fix the original issue, we should completely remove SND_SOC_IMX_PCM and have imx-pcm.o statically linked with imx-pcm-fiq.o or imx-pcm-dma.o. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- Mark, To be clear, I think it's a v3.8 material. Shawn sound/soc/fsl/Kconfig | 9 ++------- sound/soc/fsl/Makefile | 5 ++++- sound/soc/fsl/imx-pcm.c | 3 --- 3 files changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3b98159..a210c8d 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -108,18 +108,13 @@ if SND_IMX_SOC config SND_SOC_IMX_SSI tristate -config SND_SOC_IMX_PCM - tristate - config SND_SOC_IMX_PCM_FIQ - bool + tristate select FIQ - select SND_SOC_IMX_PCM config SND_SOC_IMX_PCM_DMA - bool + tristate select SND_SOC_DMAENGINE_PCM - select SND_SOC_IMX_PCM config SND_SOC_IMX_AUDMUX tristate diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index afd3479..ec14579 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -41,7 +41,10 @@ endif obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o -obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o +obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += snd-soc-imx-pcm-fiq.o +snd-soc-imx-pcm-fiq-y := imx-pcm-fiq.o imx-pcm.o +obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += snd-soc-imx-pcm-dma.o +snd-soc-imx-pcm-dma-y := imx-pcm-dma.o imx-pcm.o # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o diff --git a/sound/soc/fsl/imx-pcm.c b/sound/soc/fsl/imx-pcm.c index d5cd9ef..0c9f188 100644 --- a/sound/soc/fsl/imx-pcm.c +++ b/sound/soc/fsl/imx-pcm.c @@ -31,7 +31,6 @@ int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); return ret; } -EXPORT_SYMBOL_GPL(snd_imx_pcm_mmap); static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { @@ -80,7 +79,6 @@ int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) out: return ret; } -EXPORT_SYMBOL_GPL(imx_pcm_new); void imx_pcm_free(struct snd_pcm *pcm) { @@ -102,7 +100,6 @@ void imx_pcm_free(struct snd_pcm *pcm) buf->area = NULL; } } -EXPORT_SYMBOL_GPL(imx_pcm_free); MODULE_DESCRIPTION("Freescale i.MX PCM driver"); MODULE_AUTHOR("Sascha Hauer <s.hauer(a)pengutronix.de>"); -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 0/5] ASoC: twl6040: Updates for 3.9
by Peter Ujfalusi 12 Jan '13

12 Jan '13

Hi Mark, Liam, Changes since v1: - patch 2 for twl6040_resume() has been replaced with a simpler one Cover letter from v1: Small fixe from Misael for twl6040 codec driver and devm_* conversion, switch to system workqueue and dead code removal. Regards, Peter --- Misael Lopez Cruz (1): ASoC: twl6040: Convert PLUGINT to no-suspend irq Peter Ujfalusi (4): ASoC: twl6040: Only set the bias_level once in twl6040_resume() ASoC: twl6040: Convert to use devm_* when possible ASoC: twl6040: Switch to use system workqueue for jack reporting ASoC: twl6040: Remove leftover code from hs/hf ramp implementation sound/soc/codecs/twl6040.c | 62 +++++++--------------------------------------- 1 file changed, 9 insertions(+), 53 deletions(-) -- 1.8.1

2 6

[alsa-devel] [PATCH] ASoC: twl4030: Remove suspend/resume soc driver operations
by Peter Ujfalusi 12 Jan '13

12 Jan '13

With idle_bias_off these are no longer needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl4030.c | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 63b280b..5611b7b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2267,18 +2267,6 @@ static struct snd_soc_dai_driver twl4030_dai[] = { }, }; -static int twl4030_soc_suspend(struct snd_soc_codec *codec) -{ - twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int twl4030_soc_resume(struct snd_soc_codec *codec) -{ - twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int twl4030_soc_probe(struct snd_soc_codec *codec) { struct twl4030_priv *twl4030; @@ -2316,8 +2304,6 @@ static int twl4030_soc_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { .probe = twl4030_soc_probe, .remove = twl4030_soc_remove, - .suspend = twl4030_soc_suspend, - .resume = twl4030_soc_resume, .read = twl4030_read_reg_cache, .write = twl4030_write, .set_bias_level = twl4030_set_bias_level, -- 1.8.1

2 1

Re: [alsa-devel] [PATCH v2 06/10] ASoC: tegra: remove auxdata
by Stephen Warren 11 Jan '13

11 Jan '13

On 01/11/2013 01:01 AM, Prashant Gaikwad wrote: > Configlink clock information is added to device tree. Get the clocks > using device node. Remove AUXDATA. Cc'ing in the ASoC maintainers as an FYI and for Acks; this patch needs to go through the Tegra tree due to dependencies. > > Signed-off-by: Prashant Gaikwad <pgaikwad(a)nvidia.com> > --- > sound/soc/tegra/tegra30_ahub.c | 14 ++------------ > 1 files changed, 2 insertions(+), 12 deletions(-) > > diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c > index bb31c41..2355630 100644 > --- a/sound/soc/tegra/tegra30_ahub.c > +++ b/sound/soc/tegra/tegra30_ahub.c > @@ -299,15 +299,6 @@ static const char * const configlink_clocks[] = { > "spdif_in", > }; > > -struct of_dev_auxdata ahub_auxdata[] = { > - OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080300, "tegra30-i2s.0", NULL), > - OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080400, "tegra30-i2s.1", NULL), > - OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080500, "tegra30-i2s.2", NULL), > - OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080600, "tegra30-i2s.3", NULL), > - OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080700, "tegra30-i2s.4", NULL), > - {} > -}; > - > #define LAST_REG(name) \ > (TEGRA30_AHUB_##name + \ > (TEGRA30_AHUB_##name##_STRIDE * TEGRA30_AHUB_##name##_COUNT) - 4) > @@ -451,7 +442,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) > * Ensure that here. > */ > for (i = 0; i < ARRAY_SIZE(configlink_clocks); i++) { > - clk = clk_get_sys(NULL, configlink_clocks[i]); > + clk = clk_get(&pdev->dev, configlink_clocks[i]); > if (IS_ERR(clk)) { > dev_err(&pdev->dev, "Can't get clock %s\n", > configlink_clocks[i]); > @@ -569,8 +560,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) > goto err_pm_disable; > } > > - of_platform_populate(pdev->dev.of_node, NULL, ahub_auxdata, > - &pdev->dev); > + of_platform_populate(pdev->dev.of_node, NULL, NULL, &pdev->dev); > > return 0; > >

2 1

[alsa-devel] [PATCH 0/5] ASoC: twl6040: Updates for 3.9
by Peter Ujfalusi 11 Jan '13

11 Jan '13

Hi Mark, Liam, Small fixes from Misael for twl6040 codec driver and devm_* conversion, switch to system workqueue and dead code removal. Regards, Peter --- Misael Lopez Cruz (2): ASoC: twl6040: Convert PLUGINT to no-suspend irq ASoC: twl6040: Prevent extra power transitions during resume Peter Ujfalusi (3): ASoC: twl6040: Convert to use devm_* when possible ASoC: twl6040: Switch to use system workqueue for jack reporting ASoC: twl6040: Remove leftover code from hs/hf ramp implementation sound/soc/codecs/twl6040.c | 67 +++++++++------------------------------------- 1 file changed, 13 insertions(+), 54 deletions(-) -- 1.8.1

2 12

Re: [alsa-devel] Oops in sound/usb/pcm.c:match_endpoint_audioformats() in current -git
by Takashi Iwai 11 Jan '13

11 Jan '13

At Fri, 11 Jan 2013 08:47:25 +0100, Jens Axboe wrote: > > On 2013-01-10 21:19, Takashi Iwai wrote: > > From: Takashi Iwai <tiwai(a)suse.de> > > Subject: [PATCH v2] ALSA: usb-audio: Fix NULL dereference by access to > > non-existing substream > > > > The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for > > audioformat mismatch] introduced the correction of parameters to be > > set for sync EP. But since the new code assumes that the sync EP is > > always paired with the data EP of another direction, it triggers Oops > > when a device only with a single direction is used. > > > > This patch adds a proper check of sync EP type and the presence of the > > paired substream for avoiding the crash. > > > > Reported-by: Jens Axboe <axboe(a)kernel.dk> > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > Confirmed, it works. You can add my tested-by too. Thanks Takashi! Thanks for quick testing. Applied it now. Takashi

1 0

Re: [alsa-devel] Oops in sound/usb/pcm.c:match_endpoint_audioformats() in current -git
by Takashi Iwai 10 Jan '13

10 Jan '13

At Thu, 10 Jan 2013 13:49:22 +0100, Jens Axboe wrote: > > On 2013-01-10 11:21, Takashi Iwai wrote: > > At Thu, 10 Jan 2013 09:23:48 +0100, > > Jens Axboe wrote: > >> > >> Hi, > >> > >> I have a USB DAC that I use for music. Just upgraded my workstation from > >> 3.7-rc7 to 3.8-rc3 this morning, and when the DAC is switched on, I get > >> an immediate oops. I have attached the picture. It's crashing here: > >> > >> (gdb) l *snd_usb_pcm_prepare+0x153 > >> 0x161c is in snd_usb_pcm_prepare (sound/usb/pcm.c:470). > >> 465 snd_pcm_format_t pcm_format) > >> 466 { > >> 467 int i; > >> 468 int score = 0; > >> 469 > >> 470 if (fp->channels < 1) { > >> 471 snd_printdd("%s: (fmt @%p) no channels\n", __func__, fp); > >> 472 return 0; > >> 473 } > >> 474 > >> > >> 'fp' is clearly NULL. > > > > Hmm... it's a function called only from a single place in > > configure_sync_endpoint() and it's in list_for_each_entry(): > > > > list_for_each_entry(fp, &sync_subs->fmt_list, list) { > > int score = match_endpoint_audioformats(fp, subs->cur_audiofmt, > > subs->cur_rate, subs->pcm_format); > > > > so it must be hardly NULL in normal situations. > > The only scenario I can imagine right now is that the whole structure > > is still uninitialized while it's called. If so, it's a race problem, > > and shouldn't be new to 3.8. A patch like below might paper over the > > problem although it's far from perfect. > > > > > >> Let me know if you want a bisect. > > > > Yeah, that'll be really helpful. > > Here it is, it's from the one introducing the audioformat lookup. > Confirmed that 3.8-rc3 with this backed out works fine, too. So should > be fairly confident in that result. Thanks! I'm slowly understanding what's happening there. Could you try the patch below? Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: usb-audio: Fix NULL dereference by access to non-existing substream The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for audioformat mismatch] introduced the correction of parameters to be set for sync EP. But since the new code assumes that the sync EP is always paired with the data EP of another direction, it triggers Oops when a device only with a single direction is used. This patch adds a proper check of sync EP type and the presence of the paired substream for avoiding the crash. Reported-by: Jens Axboe <axboe(a)kernel.dk> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..21c0001 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -511,6 +511,17 @@ static int configure_sync_endpoint(struct snd_usb_substream *subs) struct snd_usb_substream *sync_subs = &subs->stream->substream[subs->direction ^ 1]; + if (subs->sync_endpoint->type != SND_USB_ENDPOINT_TYPE_DATA || + !subs->stream) { + ret = snd_usb_endpoint_set_params(subs->sync_endpoint, + subs->pcm_format, + subs->channels, + subs->period_bytes, + subs->cur_rate, + subs->cur_audiofmt, + NULL); + } + /* Try to find the best matching audioformat. */ list_for_each_entry(fp, &sync_subs->fmt_list, list) { int score = match_endpoint_audioformats(fp, subs->cur_audiofmt, -- 1.8.1

2 2

[alsa-devel] [PATCH] ASoC: dapm: Fix sense of regulator bypass mode
by Mark Brown 10 Jan '13

10 Jan '13

Enable bypass when the regulator is idle, not when it is in use. This is consistent with what the few existing users actually want. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dc18d78..17dba18 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1023,7 +1023,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, if (SND_SOC_DAPM_EVENT_ON(event)) { if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { - ret = regulator_allow_bypass(w->regulator, true); + ret = regulator_allow_bypass(w->regulator, false); if (ret != 0) dev_warn(w->dapm->dev, "ASoC: Failed to bypass %s: %d\n", @@ -1033,7 +1033,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, return regulator_enable(w->regulator); } else { if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { - ret = regulator_allow_bypass(w->regulator, false); + ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, "ASoC: Failed to unbypass %s: %d\n", @@ -3039,6 +3039,14 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, w->name, ret); return NULL; } + + if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { + ret = regulator_allow_bypass(w->regulator, true); + if (ret != 0) + dev_warn(w->dapm->dev, + "ASoC: Failed to unbypass %s: %d\n", + w->name, ret); + } break; case snd_soc_dapm_clock_supply: #ifdef CONFIG_CLKDEV_LOOKUP -- 1.7.10.4

1 0

[alsa-devel] [GIT PULL] ASoC updates for v3.8
by Mark Brown 10 Jan '13

10 Jan '13

The following changes since commit 9931faca02c604c22335f5a935a501bb2ace6e20: Linux 3.8-rc3 (2013-01-09 18:59:55 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-3.8-rc2 for you to fetch changes up to 49a170bcf28af5ef372cdbed7750ba8d4162bb0e: Merge remote-tracking branch 'asoc/fix/wm5100' into tmp (2013-01-10 12:22:30 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.8 Nothing terribly exciting here except for the DOUBLE_RANGE fix which just hadn't worked before, nobody noticed due to lack of use. ---------------------------------------------------------------- Axel Lin (5): ASoC: sta529: Fix update register bits in sta529_set_dai_fmt ASoC: wm2200: Fix setting dai format in wm2200_set_fmt ASoC: lm49453: Fix mask for setting mode bit in lm49453_set_dai_fmt() ASoC: arizona: Do proper shift for setting AIF rate ASoC: cs42l52: Catch no-match case in cs42l52_get_clk Chuansheng Liu (2): ASoC: core: fix the memory leak in case of device_add() failure ASoC: core: fix the memory leak in case of remove_aux_dev() Daniel Mack (2): ASoC: cs4271: fix sparse warning ASoC: cs4271: fix property check Fabio Estevam (2): ASoC: soc-core: Remove unused 'ret' variable ASoC: sgtl5000: Fix maximum value for microphone gain MR.Swami.Reddy(a)ti.com (2): ASoC: lm49453: Fix adc, mic and sidetone volume ranges ASoC: lm49453: Update lm49453_reg_defs values as per LM49453 HW revision-B Mark Brown (20): ASoC: core: Fix SOC_DOUBLE_RANGE() macros ASoC: arizona: Correct FLL source definitions ASoC: wm5102: Improve speaker enable performance ASoC: wm2200: Remove DSP B and left justified AIF modes ASoC: arizona: Remove DSP B and left justified AIF modes ASoC: wm5100: Remove DSP B and left justified formats ASoC: wm2000: Fix sense of speech clarity enable ASoC: wm_adsp: Ensure that block writes are from DMA aligned addresses Merge remote-tracking branch 'asoc/fix/arizona' into tmp Merge remote-tracking branch 'asoc/fix/core' into tmp Merge remote-tracking branch 'asoc/fix/cs4271' into tmp Merge remote-tracking branch 'asoc/fix/cs42l52' into tmp Merge remote-tracking branch 'asoc/fix/lm49453' into tmp Merge remote-tracking branch 'asoc/fix/pxa' into tmp Merge remote-tracking branch 'asoc/fix/sgtl5000' into tmp Merge remote-tracking branch 'asoc/fix/sta529' into tmp Merge remote-tracking branch 'asoc/fix/wm-adsp' into tmp Merge remote-tracking branch 'asoc/fix/wm2000' into tmp Merge remote-tracking branch 'asoc/fix/wm2200' into tmp Merge remote-tracking branch 'asoc/fix/wm5100' into tmp Mike Dunn (2): ALSA: pxa27x: fix ac97 cold reset ALSA: pxa27x: fix ac97 warm reset Patrick Lai (1): ASoC: pcm: allow backend hardware to be freed in pause state arch/arm/mach-pxa/include/mach/mfp-pxa27x.h | 3 + arch/arm/mach-pxa/pxa27x.c | 4 +- include/sound/cs4271.h | 2 +- include/sound/soc.h | 10 ++- sound/arm/pxa2xx-ac97-lib.c | 26 ++++++- sound/soc/codecs/arizona.c | 9 +-- sound/soc/codecs/arizona.h | 18 ++--- sound/soc/codecs/cs4271.c | 6 +- sound/soc/codecs/cs42l52.c | 4 +- sound/soc/codecs/lm49453.c | 106 ++++++++++----------------- sound/soc/codecs/sgtl5000.c | 4 +- sound/soc/codecs/sta529.c | 9 ++- sound/soc/codecs/wm2000.c | 4 +- sound/soc/codecs/wm2200.c | 8 +- sound/soc/codecs/wm5100.c | 6 -- sound/soc/codecs/wm5102.c | 48 +++++++++++- sound/soc/codecs/wm_adsp.c | 23 +++++- sound/soc/soc-core.c | 35 ++++++++- sound/soc/soc-pcm.c | 1 + 19 files changed, 201 insertions(+), 125 deletions(-)

2 1

[alsa-devel] [PATCH RFC 0/8] Use generic parser for HD-audio codec drivers
by Takashi Iwai 10 Jan '13

10 Jan '13

Hi, this is the next series of patches to convert various codec drivers to use the new generic parser code. Most of changes are obvious, but some places it gets tricky. As mentioned in another mail, the patches are found in sound-unstable git tree test/hda-migrate branch git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git Takashi

2 10

[alsa-devel] [PATCH] ASoC: ak4642: add Device Tree support
by Kuninori Morimoto 10 Jan '13

10 Jan '13

Support for loading the ak4642 codec module via devicetree. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/ak4642.txt | 17 ++++++++++ sound/soc/codecs/ak4642.c | 33 ++++++++++++++++++-- 2 files changed, 48 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak4642.txt diff --git a/Documentation/devicetree/bindings/sound/ak4642.txt b/Documentation/devicetree/bindings/sound/ak4642.txt new file mode 100644 index 0000000..623d4e7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4642.txt @@ -0,0 +1,17 @@ +AK4642 I2C transmitter + +This device supports I2C mode only. + +Required properties: + + - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648" + - reg : The chip select number on the I2C bus + +Example: + +&i2c { + ak4648: ak4648@0x12 { + compatible = "asahi-kasei,ak4642"; + reg = <0x12>; + }; +}; diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 1f0cdab..c78794d 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -26,6 +26,7 @@ #include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <linux/module.h> #include <sound/soc.h> #include <sound/initval.h> @@ -513,12 +514,31 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static struct of_device_id ak4642_of_match[]; static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct device_node *np = i2c->dev.of_node; + const struct snd_soc_codec_driver *driver; + + driver = NULL; + if (np) { + const struct of_device_id *of_id; + + of_id = of_match_device(ak4642_of_match, &i2c->dev); + if (of_id) + driver = of_id->data; + } else { + driver = (struct snd_soc_codec_driver *)id->driver_data; + } + + if (!driver) { + dev_err(&i2c->dev, "no driver\n"); + return -EINVAL; + } + return snd_soc_register_codec(&i2c->dev, - (struct snd_soc_codec_driver *)id->driver_data, - &ak4642_dai, 1); + driver, &ak4642_dai, 1); } static int ak4642_i2c_remove(struct i2c_client *client) @@ -527,6 +547,14 @@ static int ak4642_i2c_remove(struct i2c_client *client) return 0; } +static struct of_device_id ak4642_of_match[] __devinitconst = { + { .compatible = "asahi-kasei,ak4642", .data = &soc_codec_dev_ak4642}, + { .compatible = "asahi-kasei,ak4643", .data = &soc_codec_dev_ak4642}, + { .compatible = "asahi-kasei,ak4648", .data = &soc_codec_dev_ak4648}, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4642_of_match); + static const struct i2c_device_id ak4642_i2c_id[] = { { "ak4642", (kernel_ulong_t)&soc_codec_dev_ak4642 }, { "ak4643", (kernel_ulong_t)&soc_codec_dev_ak4642 }, @@ -539,6 +567,7 @@ static struct i2c_driver ak4642_i2c_driver = { .driver = { .name = "ak4642-codec", .owner = THIS_MODULE, + .of_match_table = ak4642_of_match, }, .probe = ak4642_i2c_probe, .remove = ak4642_i2c_remove, -- 1.7.9.5

2 1

Re: [alsa-devel] Oops in sound/usb/pcm.c:match_endpoint_audioformats() in current -git
by Takashi Iwai 10 Jan '13

10 Jan '13

At Thu, 10 Jan 2013 09:23:48 +0100, Jens Axboe wrote: > > Hi, > > I have a USB DAC that I use for music. Just upgraded my workstation from > 3.7-rc7 to 3.8-rc3 this morning, and when the DAC is switched on, I get > an immediate oops. I have attached the picture. It's crashing here: > > (gdb) l *snd_usb_pcm_prepare+0x153 > 0x161c is in snd_usb_pcm_prepare (sound/usb/pcm.c:470). > 465 snd_pcm_format_t pcm_format) > 466 { > 467 int i; > 468 int score = 0; > 469 > 470 if (fp->channels < 1) { > 471 snd_printdd("%s: (fmt @%p) no channels\n", __func__, fp); > 472 return 0; > 473 } > 474 > > 'fp' is clearly NULL. Hmm... it's a function called only from a single place in configure_sync_endpoint() and it's in list_for_each_entry(): list_for_each_entry(fp, &sync_subs->fmt_list, list) { int score = match_endpoint_audioformats(fp, subs->cur_audiofmt, subs->cur_rate, subs->pcm_format); so it must be hardly NULL in normal situations. The only scenario I can imagine right now is that the whole structure is still uninitialized while it's called. If so, it's a race problem, and shouldn't be new to 3.8. A patch like below might paper over the problem although it's far from perfect. > Let me know if you want a bisect. Yeah, that'll be really helpful. Or, at least, could you try to copy sound/usb/* from 3.7 to 3.8 and see whether it's really a regression there? thanks, Takashi --- diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ad181d5..3bfc564 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -81,8 +81,7 @@ static void snd_usb_audio_pcm_free(struct snd_pcm *pcm) */ static void snd_usb_init_substream(struct snd_usb_stream *as, - int stream, - struct audioformat *fp) + int stream) { struct snd_usb_substream *subs = &as->substream[stream]; @@ -96,6 +95,13 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, subs->speed = snd_usb_get_speed(subs->dev); snd_usb_set_pcm_ops(as->pcm, stream); +} + +static void snd_usb_add_substream_format(struct snd_usb_stream *as, + int stream, + struct audioformat *fp) +{ + struct snd_usb_substream *subs = &as->substream[stream]; list_add_tail(&fp->list, &subs->fmt_list); subs->formats |= fp->formats; @@ -342,7 +348,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) return err; - snd_usb_init_substream(as, stream, fp); + snd_usb_add_substream_format(as, stream, fp); return add_chmap(as->pcm, stream, subs); } @@ -370,7 +376,9 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, else strcpy(pcm->name, "USB Audio"); - snd_usb_init_substream(as, stream, fp); + snd_usb_init_substream(as, SNDRV_PCM_STREAM_PLAYBACK); + snd_usb_init_substream(as, SNDRV_PCM_STREAM_CAPTURE); + snd_usb_add_substream_format(as, stream, fp); list_add(&as->list, &chip->pcm_list); chip->pcm_devs++;

1 0

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: Make ebox44_table static
by Sachin Kamat 10 Jan '13

10 Jan '13

Fixes the following sparse warning: sound/usb/mixer_quirks.c:1209:23: warning: symbol 'ebox44_table' was not declared. Should it be static? Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/usb/mixer_quirks.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 0422b13..15520de 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1206,7 +1206,7 @@ static int snd_c400_create_mixer(struct usb_mixer_interface *mixer) * are valid they presents mono controls as L and R channels of * stereo. So we provide a good mixer here. */ -struct std_mono_table ebox44_table[] = { +static struct std_mono_table ebox44_table[] = { { .unitid = 4, .control = 1, -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ALSA: hdspm: Fix wordclock status on AES32
by Andre Schramm 09 Jan '13

09 Jan '13

Use correct bitmask for AES32 cards to determine wordclock lock state, add missing bitmask for sync check and make output of the corresponding control and /proc coherent. Signed-off-by: Andre Schramm <andre.schramm(a)iosono-sound.com> diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 6e02e06..36e247f 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -441,6 +441,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); */ /* status */ #define HDSPM_AES32_wcLock 0x0200000 +#define HDSPM_AES32_wcSync 0x0100000 #define HDSPM_AES32_wcFreq_bit 22 /* (status >> HDSPM_AES32_wcFreq_bit) & 0xF gives WC frequency (cf function HDSPM_bit2freq */ @@ -3467,10 +3468,12 @@ static int hdspm_wc_sync_check(struct hdspm *hdspm) switch (hdspm->io_type) { case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister); - if (status & HDSPM_wcSync) - return 2; - else if (status & HDSPM_wcLock) - return 1; + if (status & HDSPM_AES32_wcLock) { + if (status & HDSPM_AES32_wcSync) + return 2; + else + return 1; + } return 0; break; @@ -4658,6 +4661,7 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry, unsigned int status; unsigned int status2; unsigned int timecode; + unsigned int wcLock, wcSync; int pref_syncref; char *autosync_ref; int x; @@ -4751,8 +4755,11 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry, snd_iprintf(buffer, "--- Status:\n"); + wcLock = status & HDSPM_AES32_wcLock; + wcSync = wcLock && (status & HDSPM_AES32_wcSync); + snd_iprintf(buffer, "Word: %s Frequency: %d\n", - (status & HDSPM_AES32_wcLock) ? "Sync " : "No Lock", + (wcLock) ? (wcSync ? "Sync " : "Lock") : "No Lock", HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF)); for (x = 0; x < 8; x++) { -- 1.7.1

3 4

[alsa-devel] [PATCH] Revert "ALSA: hda - Shut up pins at power-saving mode with Conexnat codecs"
by David Henningsson 09 Jan '13

09 Jan '13

This reverts commit 697c373e34613609cb5450f98b91fefb6e910588. The original patch was meant to remove clicking, but in fact caused even more clicking instead. Thanks to c4pp4 for doing most of the work with this bug. BugLink: https://bugs.launchpad.net/bugs/886975 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- @c4pp4: Is this consistent with what you have done to reduce the clicking? Will you recommend upstream Linux to take the patch below? sound/pci/hda/patch_conexant.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 60890bf..dd798c3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -558,24 +558,12 @@ static int conexant_build_controls(struct hda_codec *codec) return 0; } -#ifdef CONFIG_PM -static int conexant_suspend(struct hda_codec *codec) -{ - snd_hda_shutup_pins(codec); - return 0; -} -#endif - static const struct hda_codec_ops conexant_patch_ops = { .build_controls = conexant_build_controls, .build_pcms = conexant_build_pcms, .init = conexant_init, .free = conexant_free, .set_power_state = conexant_set_power, -#ifdef CONFIG_PM - .suspend = conexant_suspend, -#endif - .reboot_notify = snd_hda_shutup_pins, }; #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -4405,10 +4393,6 @@ static const struct hda_codec_ops cx_auto_patch_ops = { .init = cx_auto_init, .free = conexant_free, .unsol_event = snd_hda_jack_unsol_event, -#ifdef CONFIG_PM - .suspend = conexant_suspend, -#endif - .reboot_notify = snd_hda_shutup_pins, }; /* -- 1.7.9.5

2 7

Re: [alsa-devel] [3.6.9 -> 3.7.1 regression] sound: snd_hda_intel codec probing issue?
by Takashi Iwai 09 Jan '13

09 Jan '13

At Tue, 08 Jan 2013 21:25:21 +0100, Vincent Blut wrote: > > Le mardi 08 janvier 2013 à 13:56 +0100, Takashi Iwai a écrit : > > At Tue, 08 Jan 2013 13:28:55 +0100, > > From: Takashi Iwai <tiwai(a)suse.de> > > Subject: [PATCH] ALSA: hda - Disable runtime D3 for Intel CPT & co > > > > We've got a few bug reports that the runtime D3 results in the dead > > HD-audio controller. It seems that the problem is in a deeper level > > than the sound driver itself, so as a temporal solution, disable the > > feature for these controllers again. > > > > Reported-by: Vincent Blut <vincent.debian(a)free.fr> > > Reported-by: Maurizio Avogadro <mavoga(a)gmail.com> > > Cc: <stable(a)vger.kernel.org> [v3.7] > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- > > sound/pci/hda/hda_intel.c | 13 ++++++++----- > > 1 file changed, 8 insertions(+), 5 deletions(-) > > > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > > index cca8727..0b6aeba 100644 > > --- a/sound/pci/hda/hda_intel.c > > +++ b/sound/pci/hda/hda_intel.c > > @@ -573,9 +573,12 @@ enum { > > #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ > > > > /* quirks for Intel PCH */ > > -#define AZX_DCAPS_INTEL_PCH \ > > +#define AZX_DCAPS_INTEL_PCH_NOPM \ > > (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | \ > > - AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_PM_RUNTIME) > > + AZX_DCAPS_COUNT_LPIB_DELAY) > > + > > +#define AZX_DCAPS_INTEL_PCH \ > > + (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME) > > > > /* quirks for ATI SB / AMD Hudson */ > > #define AZX_DCAPS_PRESET_ATI_SB \ > > @@ -3586,13 +3589,13 @@ static void azx_remove(struct pci_dev *pci) > > static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > > /* CPT */ > > { PCI_DEVICE(0x8086, 0x1c20), > > - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > > /* PBG */ > > { PCI_DEVICE(0x8086, 0x1d20), > > - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > > /* Panther Point */ > > { PCI_DEVICE(0x8086, 0x1e20), > > - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > > /* Lynx Point */ > > { PCI_DEVICE(0x8086, 0x8c20), > > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > Well, my sound controller behaves nicely with the above quirk. > > Tested-by: Vincent Blut <vincent.debian(a)free.fr> OK, I queued the patch now. thanks, Takashi

1 0

Re: [alsa-devel] [PATCH v2 1/3] ASoC: tegra: add ac97 host driver
by Stephen Warren 09 Jan '13

09 Jan '13

On 01/08/2013 03:22 PM, Lucas Stach wrote: > Am Dienstag, den 08.01.2013, 15:10 -0700 schrieb Stephen Warren: >> On 01/04/2013 06:18 PM, Lucas Stach wrote: >>> This adds the driver for the Tegra 2x AC97 host controller. >>> diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c >>> +static inline void tegra20_ac97_start_playback(struct tegra20_ac97 *ac97) >>> +{ >>> + regmap_update_bits(ac97->regmap, TEGRA20_AC97_FIFO1_SCR, >>> + TEGRA20_AC97_FIFO_SCR_PB_QRT_MT_EN, >>> + TEGRA20_AC97_FIFO_SCR_PB_QRT_MT_EN); >>> + >>> + regmap_update_bits(ac97->regmap, TEGRA20_AC97_CTRL, >>> + TEGRA20_AC97_CTRL_PCM_DAC_EN | >>> + TEGRA20_AC97_CTRL_STM_EN, >>> + TEGRA20_AC97_CTRL_PCM_DAC_EN | >>> + TEGRA20_AC97_CTRL_STM_EN); >>> +} >> >> That sets both PCM_DAC_EN and STM_EN, but ... >> >>> +static inline void tegra20_ac97_stop_playback(struct tegra20_ac97 *ac97) >>> +{ >>> + regmap_update_bits(ac97->regmap, TEGRA20_AC97_FIFO1_SCR, >>> + TEGRA20_AC97_FIFO_SCR_PB_QRT_MT_EN, 0); >>> + >>> + regmap_update_bits(ac97->regmap, TEGRA20_AC97_CTRL, >>> + TEGRA20_AC97_CTRL_PCM_DAC_EN, 0); >>> +} >> >> ... that only clears DAC_EN. Should it clear STM_EN too? >> > To be honest I have no idea what STM really is, seems to be some form of > packed format selection. If not set playback outputs only more or less > random noise. Only PCM_EN controls FIFO and DMA operations though, so > it's ok to just disable this to stop playback. I'd be tempted to just hard-code those bits on in probe() then, but it's not a big deal, so not worth spinning the patch for that. I guess I meant to say, Reviewed-by: Stephen Warren <swarren(a)nvidia.com>

1 0

[alsa-devel] [PATCH 0/3] ALSA: pxa27x: ac97 reset fixes
by Mike Dunn 08 Jan '13

08 Jan '13

AC97 reset is broken on the pxa27x. These patches fix it by addressing multiple problems: - cold reset fails because the GCR register is not set correctly - warm reset fails because some code used to work around a hardware bug is broken - the intent of the work-around code is obfuscated by unfortunate function naming This patch set replaces some earlier individual patches that had some problems. Thanks Robert and Igor. Changelog: v2: - don't specify the LPM flag DRIVE_HIGH in #define for the mfp config - patches 2 and 3 are combined into one patch - mfp level and direction are set only once, during driver initialization Mike Dunn (3): ALSA: pxa2xx: fix ac97 cold reset ALSA: pxa27x: fix ac97 warm reset ALSA: pxa27x: rename pxa27x_assert_ac97reset() arch/arm/mach-pxa/include/mach/mfp-pxa27x.h | 3 ++ arch/arm/mach-pxa/pxa27x.c | 24 ++++++++++++------ sound/arm/pxa2xx-ac97-lib.c | 34 +++++++++++++++++++++----- 3 files changed, 46 insertions(+), 15 deletions(-) -- 1.7.8.6

4 9

[alsa-devel] [asoc:topic/adsp 4/4] sound/soc/codecs/wm_adsp.c:664:29: sparse: incorrect type in assignment (different modifiers)
by kbuild test robot 08 Jan '13

08 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/adsp head: 25326bb263894a528288716901bd434372927844 commit: 25326bb263894a528288716901bd434372927844 [4/4] ASoC: wm_adsp: Implement support for algorithm-specific coefficient blocks sparse warnings: + sound/soc/codecs/wm_adsp.c:664:29: sparse: incorrect type in assignment (different modifiers) sound/soc/codecs/wm_adsp.c:664:29: expected struct wm_adsp_region *mem sound/soc/codecs/wm_adsp.c:664:29: got struct wm_adsp_region const * sound/soc/codecs/wm_adsp.c: In function 'wm_adsp_load_coeff': sound/soc/codecs/wm_adsp.c:664:8: warning: assignment discards 'const' qualifier from pointer target type [enabled by default] vim +664 sound/soc/codecs/wm_adsp.c 2159ad93 Mark Brown 2012-10-11 648 case WMFW_NAME_TEXT: 2159ad93 Mark Brown 2012-10-11 649 case WMFW_INFO_TEXT: 2159ad93 Mark Brown 2012-10-11 650 break; 2159ad93 Mark Brown 2012-10-11 651 case WMFW_ABSOLUTE: 2159ad93 Mark Brown 2012-10-11 652 region_name = "register"; 2159ad93 Mark Brown 2012-10-11 653 reg = offset; 2159ad93 Mark Brown 2012-10-11 654 break; 25326bb2 Mark Brown 2013-01-08 655 25326bb2 Mark Brown 2013-01-08 656 case WMFW_ADSP1_DM: 25326bb2 Mark Brown 2013-01-08 657 case WMFW_ADSP1_ZM: 25326bb2 Mark Brown 2013-01-08 658 case WMFW_ADSP2_XM: 25326bb2 Mark Brown 2013-01-08 659 case WMFW_ADSP2_YM: 25326bb2 Mark Brown 2013-01-08 660 adsp_dbg(dsp, "%s.%d: %d bytes in %x for %x\n", 25326bb2 Mark Brown 2013-01-08 661 file, blocks, le32_to_cpu(blk->len), 25326bb2 Mark Brown 2013-01-08 662 type, le32_to_cpu(blk->id)); 25326bb2 Mark Brown 2013-01-08 663 25326bb2 Mark Brown 2013-01-08 @664 mem = wm_adsp_find_region(dsp, type); 25326bb2 Mark Brown 2013-01-08 665 if (!mem) { 25326bb2 Mark Brown 2013-01-08 666 adsp_err(dsp, "No base for region %x\n", type); 25326bb2 Mark Brown 2013-01-08 667 break; 25326bb2 Mark Brown 2013-01-08 668 } 25326bb2 Mark Brown 2013-01-08 669 25326bb2 Mark Brown 2013-01-08 670 reg = 0; 25326bb2 Mark Brown 2013-01-08 671 list_for_each_entry(alg_region, 25326bb2 Mark Brown 2013-01-08 672 &dsp->alg_regions, list) { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] [asoc:topic/adsp 4/4] sound/soc/codecs/wm_adsp.c:664:8: warning: assignment discards 'const' qualifier from pointer target type
by kbuild test robot 08 Jan '13

08 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/adsp head: 25326bb263894a528288716901bd434372927844 commit: 25326bb263894a528288716901bd434372927844 [4/4] ASoC: wm_adsp: Implement support for algorithm-specific coefficient blocks config: make ARCH=x86_64 allmodconfig All warnings: sound/soc/codecs/wm_adsp.c: In function 'wm_adsp_setup_algs': sound/soc/codecs/wm_adsp.c:465:3: warning: format '%d' expects argument of type 'int', but argument 4 has type 'size_t' [-Wformat] sound/soc/codecs/wm_adsp.c:480:3: warning: format '%zx' expects argument of type 'size_t', but argument 4 has type 'unsigned int' [-Wformat] sound/soc/codecs/wm_adsp.c: In function 'wm_adsp_load_coeff': sound/soc/codecs/wm_adsp.c:664:8: warning: assignment discards 'const' qualifier from pointer target type [enabled by default] vim +/const +664 sound/soc/codecs/wm_adsp.c 2159ad93 Mark Brown 2012-10-11 648 case WMFW_NAME_TEXT: 2159ad93 Mark Brown 2012-10-11 649 case WMFW_INFO_TEXT: 2159ad93 Mark Brown 2012-10-11 650 break; 2159ad93 Mark Brown 2012-10-11 651 case WMFW_ABSOLUTE: 2159ad93 Mark Brown 2012-10-11 652 region_name = "register"; 2159ad93 Mark Brown 2012-10-11 653 reg = offset; 2159ad93 Mark Brown 2012-10-11 654 break; 25326bb2 Mark Brown 2013-01-08 655 25326bb2 Mark Brown 2013-01-08 656 case WMFW_ADSP1_DM: 25326bb2 Mark Brown 2013-01-08 657 case WMFW_ADSP1_ZM: 25326bb2 Mark Brown 2013-01-08 658 case WMFW_ADSP2_XM: 25326bb2 Mark Brown 2013-01-08 659 case WMFW_ADSP2_YM: 25326bb2 Mark Brown 2013-01-08 660 adsp_dbg(dsp, "%s.%d: %d bytes in %x for %x\n", 25326bb2 Mark Brown 2013-01-08 661 file, blocks, le32_to_cpu(blk->len), 25326bb2 Mark Brown 2013-01-08 662 type, le32_to_cpu(blk->id)); 25326bb2 Mark Brown 2013-01-08 663 25326bb2 Mark Brown 2013-01-08 @664 mem = wm_adsp_find_region(dsp, type); 25326bb2 Mark Brown 2013-01-08 665 if (!mem) { 25326bb2 Mark Brown 2013-01-08 666 adsp_err(dsp, "No base for region %x\n", type); 25326bb2 Mark Brown 2013-01-08 667 break; 25326bb2 Mark Brown 2013-01-08 668 } 25326bb2 Mark Brown 2013-01-08 669 25326bb2 Mark Brown 2013-01-08 670 reg = 0; 25326bb2 Mark Brown 2013-01-08 671 list_for_each_entry(alg_region, 25326bb2 Mark Brown 2013-01-08 672 &dsp->alg_regions, list) { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] [asoc:topic/adsp 3/4] sound/soc/codecs/wm_adsp.c:464:3: warning: format '%d' expects argument of type 'int', but argument 4 has type 'size_t'
by kbuild test robot 08 Jan '13

08 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/adsp head: 25326bb263894a528288716901bd434372927844 commit: 7265971f1bda27fdf97a19db3608ae15e2cdc7d2 [3/4] ASoC: wm_asdp: Validate sanity of algorithm count config: make ARCH=x86_64 allmodconfig All warnings: sound/soc/codecs/wm_adsp.c: In function 'wm_adsp_setup_algs': sound/soc/codecs/wm_adsp.c:464:3: warning: format '%d' expects argument of type 'int', but argument 4 has type 'size_t' [-Wformat] sound/soc/codecs/wm_adsp.c:479:3: warning: format '%zx' expects argument of type 'size_t', but argument 4 has type 'unsigned int' [-Wformat] vim +464 sound/soc/codecs/wm_adsp.c aab034fe Mark Brown 2012-10-26 448 aab034fe Mark Brown 2012-10-26 449 pos = sizeof(adsp2_id) / 2; aab034fe Mark Brown 2012-10-26 450 term = pos + ((sizeof(*adsp2_alg) * algs) / 2); aab034fe Mark Brown 2012-10-26 451 break; aab034fe Mark Brown 2012-10-26 452 aab034fe Mark Brown 2012-10-26 453 default: aab034fe Mark Brown 2012-10-26 454 BUG_ON(NULL == "Unknown DSP type"); aab034fe Mark Brown 2012-10-26 455 return -EINVAL; aab034fe Mark Brown 2012-10-26 456 } aab034fe Mark Brown 2012-10-26 457 aab034fe Mark Brown 2012-10-26 458 if (algs == 0) { aab034fe Mark Brown 2012-10-26 459 adsp_err(dsp, "No algorithms\n"); aab034fe Mark Brown 2012-10-26 460 return -EINVAL; aab034fe Mark Brown 2012-10-26 461 } aab034fe Mark Brown 2012-10-26 462 7265971f Mark Brown 2012-12-19 463 if (algs > 1024) { 7265971f Mark Brown 2012-12-19 @464 adsp_err(dsp, "Algorithm count %d excessive\n", algs); 7265971f Mark Brown 2012-12-19 465 print_hex_dump_bytes(dev_name(dsp->dev), DUMP_PREFIX_OFFSET, 7265971f Mark Brown 2012-12-19 466 buf, buf_size); 7265971f Mark Brown 2012-12-19 467 return -EINVAL; 7265971f Mark Brown 2012-12-19 468 } 7265971f Mark Brown 2012-12-19 469 aab034fe Mark Brown 2012-10-26 470 /* Read the terminator first to validate the length */ aab034fe Mark Brown 2012-10-26 471 ret = regmap_raw_read(regmap, mem->base + term, &val, sizeof(val)); aab034fe Mark Brown 2012-10-26 472 if (ret != 0) { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] [asoc:topic/adsp 1/4] sound/soc/codecs/wm_adsp.c:446:3: warning: format '%zx' expects argument of type 'size_t', but argument 4 has type 'unsigned int'
by kbuild test robot 08 Jan '13

08 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/adsp head: 25326bb263894a528288716901bd434372927844 commit: aab034fe60460c0e352a64e94f2323edc12ff296 [1/4] ASoC: wm_adsp: Add support for parsing algorithms config: make ARCH=x86_64 allmodconfig All warnings: sound/soc/codecs/wm_adsp.c: In function 'wm_adsp_setup_algs': sound/soc/codecs/wm_adsp.c:446:3: warning: format '%zx' expects argument of type 'size_t', but argument 4 has type 'unsigned int' [-Wformat] vim +446 sound/soc/codecs/wm_adsp.c aab034fe Mark Brown 2012-10-26 430 } aab034fe Mark Brown 2012-10-26 431 aab034fe Mark Brown 2012-10-26 432 if (algs == 0) { aab034fe Mark Brown 2012-10-26 433 adsp_err(dsp, "No algorithms\n"); aab034fe Mark Brown 2012-10-26 434 return -EINVAL; aab034fe Mark Brown 2012-10-26 435 } aab034fe Mark Brown 2012-10-26 436 aab034fe Mark Brown 2012-10-26 437 /* Read the terminator first to validate the length */ aab034fe Mark Brown 2012-10-26 438 ret = regmap_raw_read(regmap, mem->base + term, &val, sizeof(val)); aab034fe Mark Brown 2012-10-26 439 if (ret != 0) { aab034fe Mark Brown 2012-10-26 440 adsp_err(dsp, "Failed to read algorithm list end: %d\n", aab034fe Mark Brown 2012-10-26 441 ret); aab034fe Mark Brown 2012-10-26 442 return ret; aab034fe Mark Brown 2012-10-26 443 } aab034fe Mark Brown 2012-10-26 444 aab034fe Mark Brown 2012-10-26 445 if (be32_to_cpu(val) != 0xbedead) aab034fe Mark Brown 2012-10-26 @446 adsp_warn(dsp, "Algorithm list end %zx 0x%x != 0xbeadead\n", aab034fe Mark Brown 2012-10-26 447 term, be32_to_cpu(val)); aab034fe Mark Brown 2012-10-26 448 aab034fe Mark Brown 2012-10-26 449 alg = kzalloc((term - pos) * 2, GFP_KERNEL); aab034fe Mark Brown 2012-10-26 450 if (!alg) aab034fe Mark Brown 2012-10-26 451 return -ENOMEM; aab034fe Mark Brown 2012-10-26 452 aab034fe Mark Brown 2012-10-26 453 ret = regmap_raw_read(regmap, mem->base + pos, alg, (term - pos) * 2); aab034fe Mark Brown 2012-10-26 454 if (ret != 0) { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] [PATCH v2] ASoC: mxs-saif: Use a signed integer for error value
by Fabio Estevam 08 Jan '13

08 Jan '13

saif->id and saif->master_id are unsigned, so they can not be negative. Fix the following warning when building with W=1 option: sound/soc/mxs/mxs-saif.c: In function 'mxs_saif_probe': sound/soc/mxs/mxs-saif.c:676:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] sound/soc/mxs/mxs-saif.c:688:3: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] sound/soc/mxs/mxs-saif.c:692:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] Use a signed variable 'ret' to handle the error values. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Keep the attributions in the case of of_alias_get_id is not negative. sound/soc/mxs/mxs-saif.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 752675d..8f17094 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -672,9 +672,12 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!saif) return -ENOMEM; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; + ret = of_alias_get_id(np, "saif"); + if (ret < 0) + return ret; + else + saif->id = ret; + /* * If there is no "fsl,saif-master" phandle, it's a saif * master. Otherwise, it's a slave and its phandle points @@ -684,12 +687,14 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!master) { saif->master_id = saif->id; } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; + ret = of_alias_get_id(master, "saif"); + if (ret < 0) + return ret; + else + saif->master_id = ret; } - if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { + if (saif->master_id >= ARRAY_SIZE(mxs_saif)) { dev_err(&pdev->dev, "get wrong master id\n"); return -EINVAL; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Factor out calculation of memory base addresses
by Mark Brown 08 Jan '13

08 Jan '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 30 +++++++++++++++++++++++++----- 1 file changed, 25 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6bff303..ed9a6c4 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -156,6 +156,26 @@ static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, return NULL; } +static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, + unsigned int offset) +{ + switch (region->type) { + case WMFW_ADSP1_PM: + return region->base + (offset * 3); + case WMFW_ADSP1_DM: + return region->base + (offset * 2); + case WMFW_ADSP2_XM: + return region->base + (offset * 2); + case WMFW_ADSP2_YM: + return region->base + (offset * 2); + case WMFW_ADSP1_ZM: + return region->base + (offset * 2); + default: + WARN_ON(NULL != "Unknown memory region type"); + return offset; + } +} + static int wm_adsp_load(struct wm_adsp *dsp) { const struct firmware *firmware; @@ -283,27 +303,27 @@ static int wm_adsp_load(struct wm_adsp *dsp) case WMFW_ADSP1_PM: BUG_ON(!mem); region_name = "PM"; - reg = mem->base + (offset * 3); + reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP1_DM: BUG_ON(!mem); region_name = "DM"; - reg = mem->base + (offset * 2); + reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP2_XM: BUG_ON(!mem); region_name = "XM"; - reg = mem->base + (offset * 2); + reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP2_YM: BUG_ON(!mem); region_name = "YM"; - reg = mem->base + (offset * 2); + reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP1_ZM: BUG_ON(!mem); region_name = "ZM"; - reg = mem->base + (offset * 2); + reg = wm_adsp_region_to_reg(mem, offset); break; default: adsp_warn(dsp, -- 1.7.10.4

1 1

Re: [alsa-devel] [3.6.9 -> 3.7.1 regression] sound: snd_hda_intel codec probing issue?
by Takashi Iwai 08 Jan '13

08 Jan '13

At Fri, 28 Dec 2012 15:25:40 +0100, Vincent Blut wrote: > > Hi, > > Since I updated to Linux 3.7.1, listening to some audio/video bits > frequently cause the following: > > [ 7896.166946] hda-intel: azx_get_response timeout, switching to polling > mode: last cmd=0x020c0000 > [ 7897.173444] hda-intel: No response from codec, disabling MSI: last > cmd=0x020c0000 > [ 7898.179932] hda_intel: azx_get_response timeout, switching to > single_cmd mode: last cmd=0x020c0000 > [ 7898.179983] hda-codec: out of range cmd 0:0:20:400:fffff7ff > [ 9445.034371] plugin-containe[5873]: segfault at 7f44bb95e639 ip > 00007f44e454bca0 sp 00007f44c91165f8 error 4 in > libc-2.13.so[7f44e442c000+180000] > > It seems to be a codec probing failure (?). This is really fatal because > the sound become very choppy and can't recover until I reboot. > I'll try to play with 'probe_mask' kernel parameter to see if I can > narrow the correct codec slots! > > By the way I can't reproduce this on 3.6.9, so is there something that > changed in this area in 3.7.1? If it's new in 3.7, this could be a regression by runtime D3. Try to pass power_save_controller=0 option to snd-hda-intel module (or change it via sysfs dynamically). thanks, Takashi

4 10

[alsa-devel] [PATCH] ALSA: hda - print power state for AFG node in proc file
by David Henningsson 08 Jan '13

08 Jan '13

It seems useful, and power states are required for AFG nodes, so I see no reason not to print it. As a bonus, also print the AFG nid. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_proc.c | 2 ++ 1 file changed, 2 insertions(+) (Feel free to rephrase "State of AFG node 0x%02x:\n" if you like.) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 045e5d3..d6c8416 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -603,6 +603,8 @@ static void print_codec_info(struct snd_info_entry *entry, print_amp_caps(buffer, codec, codec->afg, HDA_INPUT); snd_iprintf(buffer, "Default Amp-Out caps: "); print_amp_caps(buffer, codec, codec->afg, HDA_OUTPUT); + snd_iprintf(buffer, "State of AFG node 0x%02x:\n", codec->afg); + print_power_state(buffer, codec, codec->afg); nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); if (! nid || nodes < 0) { -- 1.7.9.5

2 3

[alsa-devel] [PATCH] ASoC: mxs-saif: Use a signed integer for error value
by Fabio Estevam 08 Jan '13

08 Jan '13

saif->id and saif->master_id are unsigned, so they can not be negative. Fix the following warning when building with W=1 option: sound/soc/mxs/mxs-saif.c: In function 'mxs_saif_probe': sound/soc/mxs/mxs-saif.c:676:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] sound/soc/mxs/mxs-saif.c:688:3: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] sound/soc/mxs/mxs-saif.c:692:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] Use a signed variable 'ret' to handle the error values. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 752675d..a920206 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -672,9 +672,9 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!saif) return -ENOMEM; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; + ret = of_alias_get_id(np, "saif"); + if (ret < 0) + return ret; /* * If there is no "fsl,saif-master" phandle, it's a saif * master. Otherwise, it's a slave and its phandle points @@ -684,12 +684,12 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!master) { saif->master_id = saif->id; } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; + ret = of_alias_get_id(master, "saif"); + if (ret < 0) + return ret; } - if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { + if (saif->master_id >= ARRAY_SIZE(mxs_saif)) { dev_err(&pdev->dev, "get wrong master id\n"); return -EINVAL; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH RESEND v5 2/2] ASoC: mxs-saif: Remove platform data
by Fabio Estevam 08 Jan '13

08 Jan '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> All MXS users have been converted to device tree and the board files have been removed. No need to keep platform data in the driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> Acked-by: Dong Aisheng <dong.aisheng(a)linaro.org> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> --- Changes since v4: - Rebased against fix/mxs branch and fix file removal Changes since v3: - Remove pdev->id check Changes since v2: - Also remove 'if (np)' as dt is the only probing mechanism. Changes since v1: - Also remove mxs_saif_platform_data and sound/saif.h, which was used only to include mxs_saif_platform_data definition. include/sound/saif.h | 16 ---------------- sound/soc/mxs/mxs-saif.c | 44 ++++++++++++++++---------------------------- 2 files changed, 16 insertions(+), 44 deletions(-) delete mode 100644 include/sound/saif.h diff --git a/include/sound/saif.h b/include/sound/saif.h deleted file mode 100644 index f22f3e1..0000000 --- a/include/sound/saif.h +++ /dev/null @@ -1,16 +0,0 @@ -/* - * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SAIF_H__ -#define __SOUND_SAIF_H__ - -struct mxs_saif_platform_data { - bool master_mode; /* if true use master mode */ - int master_id; /* id of the master if in slave mode */ -}; -#endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index c294fbb..05dbadb 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -32,7 +32,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <sound/saif.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/mxs.h> @@ -660,43 +659,32 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; - struct mxs_saif_platform_data *pdata; struct pinctrl *pinctrl; int ret = 0; + struct device_node *master; - - if (!np && pdev->id >= ARRAY_SIZE(mxs_saif)) + if (!np) return -EINVAL; saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; - if (np) { - struct device_node *master; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; - /* - * If there is no "fsl,saif-master" phandle, it's a saif - * master. Otherwise, it's a slave and its phandle points - * to the master. - */ - master = of_parse_phandle(np, "fsl,saif-master", 0); - if (!master) { - saif->master_id = saif->id; - } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; - } + saif->id = of_alias_get_id(np, "saif"); + if (saif->id < 0) + return saif->id; + /* + * If there is no "fsl,saif-master" phandle, it's a saif + * master. Otherwise, it's a slave and its phandle points + * to the master. + */ + master = of_parse_phandle(np, "fsl,saif-master", 0); + if (!master) { + saif->master_id = saif->id; } else { - saif->id = pdev->id; - pdata = pdev->dev.platform_data; - if (pdata && !pdata->master_mode) - saif->master_id = pdata->master_id; - else - saif->master_id = saif->id; + saif->master_id = of_alias_get_id(master, "saif"); + if (saif->master_id < 0) + return saif->master_id; } if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { -- 1.7.9.5

2 3

[alsa-devel] [PATCH v5 2/2] ASoC: mxs-saif: Remove platform data
by Fabio Estevam 08 Jan '13

08 Jan '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> All MXS users have been converted to device tree and the board files have been removed. No need to keep platform data in the driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> Acked-by: Dong Aisheng <dong.aisheng(a)linaro.org> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> --- Changes since v4: - Rebased against fix/mxs branch and fix file removal Changes since v3: - Remove pdev->id check Changes since v2: - Also remove 'if (np)' as dt is the only probing mechanism. Changes since v1: - Also remove mxs_saif_platform_data and sound/saif.h, which was used only to include mxs_saif_platform_data definition. include/sound/saif.h | 16 ---------------- sound/soc/mxs/mxs-saif.c | 44 ++++++++++++++++---------------------------- 2 files changed, 16 insertions(+), 44 deletions(-) delete mode 100644 include/sound/saif.h diff --git a/include/sound/saif.h b/include/sound/saif.h deleted file mode 100644 index f22f3e1..0000000 --- a/include/sound/saif.h +++ /dev/null @@ -1,16 +0,0 @@ -/* - * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SAIF_H__ -#define __SOUND_SAIF_H__ - -struct mxs_saif_platform_data { - bool master_mode; /* if true use master mode */ - int master_id; /* id of the master if in slave mode */ -}; -#endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index c294fbb..05dbadb 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -32,7 +32,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <sound/saif.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/mxs.h> @@ -660,43 +659,32 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; - struct mxs_saif_platform_data *pdata; struct pinctrl *pinctrl; int ret = 0; + struct device_node *master; - - if (!np && pdev->id >= ARRAY_SIZE(mxs_saif)) + if (!np) return -EINVAL; saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; - if (np) { - struct device_node *master; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; - /* - * If there is no "fsl,saif-master" phandle, it's a saif - * master. Otherwise, it's a slave and its phandle points - * to the master. - */ - master = of_parse_phandle(np, "fsl,saif-master", 0); - if (!master) { - saif->master_id = saif->id; - } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; - } + saif->id = of_alias_get_id(np, "saif"); + if (saif->id < 0) + return saif->id; + /* + * If there is no "fsl,saif-master" phandle, it's a saif + * master. Otherwise, it's a slave and its phandle points + * to the master. + */ + master = of_parse_phandle(np, "fsl,saif-master", 0); + if (!master) { + saif->master_id = saif->id; } else { - saif->id = pdev->id; - pdata = pdev->dev.platform_data; - if (pdata && !pdata->master_mode) - saif->master_id = pdata->master_id; - else - saif->master_id = saif->id; + saif->master_id = of_alias_get_id(master, "saif"); + if (saif->master_id < 0) + return saif->master_id; } if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { -- 1.7.9.5

2 5

Re: [alsa-devel] About Alsa arecord
by Raymond Yau 08 Jan '13

08 Jan '13

> Good morning! > Are you saying I should modify the source code of alsa-driver-1.0.25? >> >> > Hello! >> > There have 2 stereo ADCs support one stereo microphone and one legacy mixer recording simultaneously, said in the document of ALC888 Datasheet .I inputed 1KHz sine wave into every pins (Line1-R,Line1-L,MIC1-R,MIC1_L,MIC2-R,MIC2-L,LINE2-R,LINE2-L) of ALC888 by signal genrator,and used command 'arecord -c 4 -f S16_LE -r 44100 >test.wav' recording in fedora14.but ALL audio data are recorded in channel 1,2.No any data in channel 3,4.Why?How can I record every pins audio data in each channel? >> >> In alc_auto_create_input_ctls , add the connection from node 0x0b of mixer ADC as input source similar to those pin complex >> >> Those Mic/line/CD playback volume and switches of node 0x0b affect the signal of the new input source ”mixer” > > Yes, also need a switch which mute the input from 0x0b for all output jacks to prevent feedback if using mic the current hda driver only support stereo capture and you have to implement multi channel capturing similar to multi channel playback in hda_codec.c snd_hda_multi_in_analog_prepare for (i = 1; i < num_adcs; i++) { snd_hda_codec_setup_stream(codec, adc_nids[i], stream_tag, i * 2, format); or try multi plugin of two stereo capture devices

1 0

[alsa-devel] [PATCH] ASoC: wm_adsp: Ensure that block writes are from DMA aligned addresses
by Mark Brown 07 Jan '13

07 Jan '13

Otherwise we won't run correctly on systems that require this for larger data transfers. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5d8bb1d..a765458 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -169,6 +169,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) const struct wm_adsp_region *mem; const char *region_name; char *file, *text; + void *buf; unsigned int reg; int regions = 0; int ret, offset, type, sizes; @@ -322,8 +323,18 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - ret = regmap_raw_write(regmap, reg, region->data, + buf = kmemdup(region->data, le32_to_cpu(region->len), + GFP_KERNEL); + if (!buf) { + adsp_err(dsp, "Out of memory\n"); + return -ENOMEM; + } + + ret = regmap_raw_write(regmap, reg, buf, le32_to_cpu(region->len)); + + kfree(buf); + if (ret != 0) { adsp_err(dsp, "%s.%d: Failed to write %d bytes at %d in %s: %d\n", -- 1.7.10.4

1 0

[alsa-devel] [PATCH 0/4] ALSA: pxa27x: ac97 reset fixes
by Mike Dunn 07 Jan '13

07 Jan '13

AC97 reset is broken on the pxa27x. These patches fix it by addressing multiple problems: - cold reset fails because the GCR register is not set correctly - warm reset fails because some code used to work around a hardware bug is broken - the intent of the work-around code is obfuscated by unfortunate function naming This patch set replaces some earlier individual patches that had some problems. Thanks Robert and Igor. Mike Dunn (4): ALSA: pxa27x: fix ac97 cold reset ALSA: pxa27x: fix ac97 warm reset bug work-around code ALSA: pxa27x: ac97 controller driver requests gpio ALSA: pxa27x: rename pxa27x_assert_ac97() arch/arm/mach-pxa/include/mach/mfp-pxa27x.h | 3 ++ arch/arm/mach-pxa/pxa27x.c | 38 +++++++++++++++++++------- sound/arm/pxa2xx-ac97-lib.c | 26 +++++++++++++----- 3 files changed, 50 insertions(+), 17 deletions(-) -- 1.7.8.6

4 21

[alsa-devel] [PATCH v2] ALSA: hda - add mute LED for HP Pavilion 17 (Realtek codec)
by David Henningsson 07 Jan '13

07 Jan '13

The mute LED is in this case connected to the Mic1 VREF. The machine also exposes the following string in BIOS: "HP_Mute_LED_0_A", so if more machines are coming, it probably makes sense to try to do something more generic, like for the IDT codec. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1096789 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6ee3459..71ae23d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5992,6 +5992,30 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec, spec->automute_hook = alc269_quanta_automute; } +/* update mute-LED according to the speaker mute state via mic1 VREF pin */ +static void alc269_fixup_mic1_mute_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + unsigned int pinval = AC_PINCTL_IN_EN + (enabled ? + AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80); + snd_hda_set_pin_ctl_cache(codec, 0x18, pinval); +} + +static void alc269_fixup_mic1_mute(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + switch (action) { + case ALC_FIXUP_ACT_BUILD: + spec->vmaster_mute.hook = alc269_fixup_mic1_mute_hook; + snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true); + /* fallthru */ + case ALC_FIXUP_ACT_INIT: + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + break; + } +} + /* update mute-LED according to the speaker mute state via mic2 VREF pin */ static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled) { @@ -6043,6 +6067,7 @@ enum { ALC269_FIXUP_DMIC, ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, + ALC269_FIXUP_MIC1_MUTE_LED, ALC269_FIXUP_MIC2_MUTE_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, @@ -6171,6 +6196,10 @@ static const struct alc_fixup alc269_fixups[] = { { } }, }, + [ALC269_FIXUP_MIC1_MUTE_LED] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_mic1_mute, + }, [ALC269_FIXUP_MIC2_MUTE_LED] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_mic2_mute, @@ -6215,6 +6244,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x1972, "HP Pavilion 17", ALC269_FIXUP_MIC1_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), -- 1.7.9.5

2 1

[alsa-devel] Alsa snapshot directory layout changed
by David Henningsson 07 Jan '13

07 Jan '13

I have a script that picks daily snapshots from ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-snapshot.tar.gz ...and I just noticed that the directory layout has changed, so what was "alsa-driver" is now "alsa-driver/alsa". Is this expected, i e, should I update my script to follow this new layout, or was it a mistake? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

3 3

[alsa-devel] [PATCH] ALSA: pxa2xx: fix ac97 cold reset for pxa27x
by Mike Dunn 07 Jan '13

07 Jan '13

Currently, ac97 reset is broken on PXA27x. Through trial-and-error (the pxa270 developer's manual is mostly incoherent on the topic of ac97 reset), I fixed it by setting the WARM_RST bit in the GCR register at the end of the cold reset, and then skipping the warm reset if the link is already up and running. It appears that setting the WARM_RST bit is a necessary final step during cold reset. I think that the PXA25x and PXA3xx may currently be working correctly because WARM_RST is set within the warm reset routine, and the codec drivers always follow a cold reset with a warm reset during their initialization. so this combination effectively completes a cold reset. This doesn't work on the PXA27x because its warm reset routine contains additional code for working around a hardware bug in the warm reset sequence, which causes the reset sequence to fail. I only have a PXA27x platform for testing, so this patch only affects the PXA27x. Signed-off-by: Mike Dunn <mikedunn(a)newsguy.com> --- As an aside... I don't understand how always following a cold reset with a warm reset (as the codec drivers do) makes sense. Unless I'm mistaken, a warm reset can only occur on a link that's currently down. All the timing diagrams I've seen for reset (warm and cold) show the link down (BITCLK stopped) at the start of the reset sequence. The SYNC line is used to effect the warm reset, and SYNC is already pulsing every frame on a link that's up and running, so it seems physically impossible to do a warm reset on a link that's currently up. But I'm no ac97 expert, so I may just be embarassing myself. Any insight gratefully received. Thanks! sound/arm/pxa2xx-ac97-lib.c | 15 +++++++++++++++ 1 files changed, 15 insertions(+), 0 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 6fc0ae9..aa242ff 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -138,6 +138,10 @@ static inline void pxa_ac97_warm_pxa27x(void) { gsr_bits = 0; + /* don't attempt warm reset if link is up */ + if (GSR & (GSR_SCR | GSR_PCR)) + return; + /* warm reset broken on Bulverde, so manually keep AC97 reset high */ pxa27x_assert_ac97reset(reset_gpio, 1); udelay(10); @@ -148,6 +152,8 @@ static inline void pxa_ac97_warm_pxa27x(void) static inline void pxa_ac97_cold_pxa27x(void) { + unsigned int timeout; + GCR &= GCR_COLD_RST; /* clear everything but nCRST */ GCR &= ~GCR_COLD_RST; /* then assert nCRST */ @@ -159,6 +165,15 @@ static inline void pxa_ac97_cold_pxa27x(void) clk_disable(ac97conf_clk); GCR = GCR_COLD_RST; udelay(50); + + /* + * Setting the WARM_RST bit is a necessary part of a cold reset, at + * least on the buggy and poorly documented PXA27x ac97 controller. + */ + GCR |= GCR_WARM_RST; + timeout = 100; /* wait for the codec-ready bit to be set */ + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); } #endif -- 1.7.8.6

3 6

[alsa-devel] [PATCH 1/2] ALSA: hda - add mute LED for HP Pavilion 17 (Realtek codec)
by David Henningsson 07 Jan '13

07 Jan '13

The mute LED is in this case connected to the Mic1 VREF. The machine also exposes the following string in BIOS: "HP_Mute_LED_0_A", so if more machines are coming, it probably makes sense to try to do something more generic, like for the IDT codec. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1096789 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 31 ++++++++++++++++++++++++++++++- 1 file changed, 30 insertions(+), 1 deletion(-) I know this is a copy-paste, but I didn't really find a better and more condensed way to do it. Perhaps if we get a third variation it would make sense to write a more generic version. Also, Alsa-info is available at https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1096789/+attachment/34… diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6ee3459..2e6803e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5992,7 +5992,30 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec, spec->automute_hook = alc269_quanta_automute; } -/* update mute-LED according to the speaker mute state via mic2 VREF pin */ +/* update mute-LED according to the speaker mute state via mic VREF pin */ +static void alc269_fixup_mic1_mute_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + unsigned int pinval = AC_PINCTL_IN_EN + (enabled ? + AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80); + snd_hda_set_pin_ctl_cache(codec, 0x18, pinval); +} + +static void alc269_fixup_mic1_mute(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + switch (action) { + case ALC_FIXUP_ACT_BUILD: + spec->vmaster_mute.hook = alc269_fixup_mic1_mute_hook; + snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true); + /* fallthru */ + case ALC_FIXUP_ACT_INIT: + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + break; + } +} + static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled) { struct hda_codec *codec = private_data; @@ -6043,6 +6066,7 @@ enum { ALC269_FIXUP_DMIC, ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, + ALC269_FIXUP_MIC1_MUTE_LED, ALC269_FIXUP_MIC2_MUTE_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, @@ -6171,6 +6195,10 @@ static const struct alc_fixup alc269_fixups[] = { { } }, }, + [ALC269_FIXUP_MIC1_MUTE_LED] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_mic1_mute, + }, [ALC269_FIXUP_MIC2_MUTE_LED] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_mic2_mute, @@ -6215,6 +6243,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x1972, "HP Pavilion 17", ALC269_FIXUP_MIC1_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), -- 1.7.9.5

2 4

[alsa-devel] About Alsa arecord
by FF 07 Jan '13

07 Jan '13

Hello! There have 2 stereo ADCs support one stereo microphone and one legacy mixer recording simultaneously, said in the document of ALC888 Datasheet .I inputed 1KHz sine wave into every pins (Line1-R,Line1-L,MIC1-R,MIC1_L,MIC2-R,MIC2-L,LINE2-R,LINE2-L) of ALC888 by signal genrator,and used command 'arecord -c 4 -f S16_LE -r 44100 >test.wav' recording in fedora14.but ALL audio data are recorded in channel 1,2.No any data in channel 3,4.Why?How can I record every pins audio data in each channel? Thank you!Best regards.

2 1

[alsa-devel] Roland-Boss RC-300 support
by Jean-Samuel Reynaud 06 Jan '13

06 Jan '13

Hi All, I have a Boss RC-300 loop station with an USB connector. The windows driver add an audio device but it was not working under linux. So please find attached, the patch for supporting audio in this device. Note that midi is not working yet but I'm working on it... Regards,

2 1

Re: [alsa-devel] Missing exactly 3 of 8 audio packets?
by Daniel Griscom 06 Jan '13

06 Jan '13

Dear Daniel (M) and Clemens, I'd like to conclude this thread if I could. It seems clear that configuring our USB audio card's descriptor with a bInterval value of 1 (a packet every microframe) was too aggressive, and was part of what caused the packet stuttering issue. (Right?) We almost certainly should increase bInterval; I'll try 2, and if that seems stable will probably go to 3 (4 microframes) for some margin. However, I'm still concerned that the symptoms don't sound like a simple starvation issue, where I'd expect to have a missing frame every once in a while. Instead, we were seeing reliable operation interspersed with periods where lots of packets were being consistently omitted. To me, this suggests that there's some other issue going on, either in the USB stack or in the USB hardware itself. Given that, I'm worried that increasing bInterval will just move the symptoms around, making it harder to duplicate but not truly becoming robust. Any thoughts on this? Should I try chasing the issue into the Linux USB development mailing list, or will they just say "set bInterval to a larger value!"? Thanks, Dan -- Daniel T. Griscom griscom(a)suitable.com Suitable Systems http://www.suitable.com/ 1 Centre Street, Suite 204 (781) 665-0053 Wakefield, MA 01880-2400

2 1

Re: [alsa-devel] [PATCH v3 0/5] ARM: OMAP4+: HDMI: Update platform devices for audio
by Ricardo Neri 05 Jan '13

05 Jan '13

Hi Mark, Tomi, Just wondering if you had a chance to take a look at these patches. Thanks! Ricardo On 12/17/2012 07:30 PM, Ricardo Neri wrote: > Hi Mark, Tomi, Liam, Tony, > > This set aims to be the version 3 of my previous submission[1] and aims to > address the comments that Mark and Tomi kindly provided on such submission. > > The creation of the platform device for the HDMI audio interface from within the > OMAPDSS HDMI driver that was previously submitted[2] is resubmitted to be > complemented with code to relocate to arch/arm/mach-omap2/display.c the > creation of the platform devices for the HDMI ASoC codec and card drivers. This > series does not break the HDMI audio functionality in any patch. > > Also, the names of the platform devices are changed to give them more logical > and more descriptive names. > > As the commit > > commit 14840b9a83c6a56629db2ba0ec247503e975f143 > Author: Ricardo Neri <ricardo.n...(a)ti.com> > Date: Tue Nov 6 00:19:17 2012 -0600 > > OMAPDSS: HDMI: Create platform device for audio support > > is reverted in Tomi's git://gitorious.org/linux-omap-dss2/linux.git master > branch, this series applies cleanly. > > Changes from v1: > *Put in a single series all the patches related to platform device updates. > *Now HDMI audio works correctly in every patch. > *Remove reference to the TPD12S015 HDMI companion chip as the ASoC drivers > are not aware of this and other chips could be used in the future. > > Changes from v2: > *Split in two patches the renaming and the relocation of the platform devices. > *Create a separate patch to pass only the address offset of the DMA data port > to the audio interface platform device. > *Keep the name hdmi-audio-codec to not refer to explicitly to OMAP. The codec > can be made generic in a different patch series submitted to alsa-devel. > > BR, > > Ricardo > > > [1]. http://www.mail-archive.com/linux-omap@vger.kernel.org/msg80785.html > [2]. http://www.mail-archive.com/linux-omap@vger.kernel.org/msg79795.html > > Ricardo Neri (5): > ARM: OMAP4+: HDMI: Rename platform devices for ASoC drivers > ARM: OMAP4: Assign IDs to DSS HDMI devices > ARM4: OMAP4+: HDMI: Relocate devices for audio codec and card > ARM: OMAP4+: HDMI: Relocate the device for audio interface > ARM: OMAP4+: HDMI: Refine the DMA port resource for audio > > arch/arm/mach-omap2/board-4430sdp.c | 9 ++--- > arch/arm/mach-omap2/board-omap4panda.c | 9 ++--- > arch/arm/mach-omap2/devices.c | 31 ---------------- > arch/arm/mach-omap2/display.c | 31 ++++++++++++++++ > drivers/video/omap2/dss/hdmi.c | 62 ++++++++++++++++++++++++++++++++ > sound/soc/omap/omap-hdmi-card.c | 4 +-- > sound/soc/omap/omap-hdmi.c | 5 ++- > sound/soc/omap/omap-hdmi.h | 2 -- > 8 files changed, 103 insertions(+), 50 deletions(-) > > -- > 1.7.10.4 >

2 2

[alsa-devel] [PATCH] ASoC: wm2000: Fix sense of speech clarity enable
by Mark Brown 04 Jan '13

04 Jan '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 1cbe88f..12bcae6 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -209,9 +209,9 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY); if (wm2000->speech_clarity) - ret &= ~WM2000_SPEECH_CLARITY; - else ret |= WM2000_SPEECH_CLARITY; + else + ret &= ~WM2000_SPEECH_CLARITY; wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret); wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: arizona: Allow runtime reconfiguration of the output mode
by Mark Brown 04 Jan '13

04 Jan '13

Some systems use external analogue switches to connect more analogue devices to the CODEC than are supported by the device. In some systems this requires changing the switched output from single ended to differential mode dynamically at runtime. Add a new function arizona_set_output_mode() to support this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 34 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 3 +++ 2 files changed, 37 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 7724b4b..6d98cf4 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1176,6 +1176,40 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, } EXPORT_SYMBOL_GPL(arizona_init_fll); +/** + * arizona_set_output_mode - Set the mode of the specified output + * + * @codec: Device to configure + * @output: Output number + * @diff: True to set the output to differential mode + * + * Some systems use external analogue switches to connect more + * analogue devices to the CODEC than are supported by the device. In + * some systems this requires changing the switched output from single + * ended to differential mode dynamically at runtime, an operation + * supported using this function. + * + * Most systems have a single static configuration and should use + * platform data instead. + */ +int arizona_set_output_mode(struct snd_soc_codec *codec, int output, bool diff) +{ + unsigned int reg, val; + + if (output < 1 || output > 6) + return -EINVAL; + + reg = ARIZONA_OUTPUT_PATH_CONFIG_1L + (output - 1) * 8; + + if (diff) + val = ARIZONA_OUT1_MONO; + else + val = 0; + + return snd_soc_update_bits(codec, reg, ARIZONA_OUT1_MONO, val); +} +EXPORT_SYMBOL_GPL(arizona_set_output_mode); + MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 68dfae5..697ff6f 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -213,4 +213,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source, extern int arizona_init_dai(struct arizona_priv *priv, int dai); +int arizona_set_output_mode(struct snd_soc_codec *codec, int output, + bool diff); + #endif -- 1.7.10.4

1 0

Re: [alsa-devel] draft -- ALSA firewire + digi003
by Robin Gareus 04 Jan '13

04 Jan '13

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Clemens, Hi ALSA-devs, I'm working with Damien Zammit on the Digidesign-003 driver which is based on your alsa-kprivate dice code. We were able to work out most of the issues so far but are now stuck with timing problems. More specifically: the jitter of async message timing is quite large and the AMDTP ISO stream has regular drop-outs. Comparing 1394 data dumps, OSX sends async messages at regular 125 (+-5us) intervals. On GNU/Linux we've seen a delta -40us, + 120us. For async messages this is not a big deal. But.. Just playing audio via aplay: there are short dropouts every couple of seconds but otherwise the devices plays fine indefinitely. Interestingly, when running cyclictest while playing sound: there are no dropouts but the stream drops after ~10-15 seconds. Any hints why that may be or how to debug the issue further? In order to rule out kernel related issues (3.0.0 is quite old), we've rebased the work (and alsa-kprivate's firewire-kernel-streaming branch) onto alsa-driver.git. The issue remains the same with 3.2.0-4-rt Alas, neither of us has detailed knowledge nor specs for AMDTP, not to mention in-depth knowledge about Linux' firewire stack timings. Any help or leads are appreciated. As we believe we are close to a fully functional driver for the 003, we are very keen to resolve the following two separate issues: 1) The timing problem we have observed with AMDTP over firewire. 2) A framework for the capture stream in ALSA firewire. The rebased driver code is at git://gareus.org/alsa-driver.git http://gareus.org/gitweb/?p=alsa-driver.git The broken-out digi003 driver is available from: https://github.com/zamaudio/snd-firewire-003 https://github.com/x42/003amdtp 1394 bus data dumps of both OSX as well as Linux can be found at http://www.zamaudio.com/?p=715 Cheers! robin -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) iQIcBAEBAgAGBQJQ5t9CAAoJEKCQvOAs9X8ELbsP/Re0wxZKU6v2rbBVhV/FC6zr JYmx/jAsmoC7vtVZMKE4gGcCreszX58oU7RG323qR9iKSEgXeqtuThZSTT6l3N39 KpHW4T094FyQxmvxRYrZnRlHgWexvLfJT/YiuTUX50L/CKgS75ajRfLqADZaHSy5 Oc+CEy50Jmoi/9/dTuM0pVPVvt5APFqLyAisw1/5wzhYMTHF0mBByn5AHXBp7bIl SVg7IoJuTNd4dYgo3PtKw9OqhjMGZ1g5DNHA9OaE7qY12r3TMMKm7Sq2iWS4HQoq RMQLOVtyMOVUKyddK8ubWi0Cm2lYjHUXUNSLyZLcdWYTWbk5eWo6Ze83Tbl33h4T ngiWt+DWpqKpm+U0UQDOU6p+DSAosKHsVcPPPkYVkwXKAkHctXKlPUst5NHEhhK3 5mbVQECHE9+BBy9qY2eimDZ/p3+5C2kIMofDga6jiuFWfHMqKp30uacUHwP4xY38 jHLaTmpX7VNwNYGnFLvGve709ejNjL8ByFKrUKpV1nAMTO4QP4SipHuApBF3BWS7 hAGBD6mKQAPl6AhvuJaxtCdTQc/tkAF0Ep1LClthsHnyhqPVCX+L5sb/z7vVmCkp cFxGXJq6TujcW4kkKPfZCNsnHqea8lgY6tEonc8aRmcIZvF+L4Yp+p4/r9OxL6ma xIIgIVDp1X9HxVayBDWL =lVsY -----END PGP SIGNATURE-----

1 0

[alsa-devel] [PATCH 1/2] ASoC: arizona: Remove DSP B and left justified AIF modes
by Mark Brown 04 Jan '13

04 Jan '13

These are not supported. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a4d7fb9..7724b4b 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -503,15 +503,9 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_A: mode = 0; break; - case SND_SOC_DAIFMT_DSP_B: - mode = 1; - break; case SND_SOC_DAIFMT_I2S: mode = 2; break; - case SND_SOC_DAIFMT_LEFT_J: - mode = 3; - break; default: arizona_aif_err(dai, "Unsupported DAI format %d\n", fmt & SND_SOC_DAIFMT_FORMAT_MASK); -- 1.7.10.4

1 1

Re: [alsa-devel] sound: usb-audio: [PATCH]x2 full capture/playback/spdif support for Digidesign Mbox 2
by Damien Zammit 04 Jan '13

04 Jan '13

On 23 December 2012 14:36, Damien Zammit <damien.zammit(a)gmail.com> wrote: > I have fixed both the kernel panic and the broken pipe issue with midi. > I believe this completes the patch, I just don't know how to submit it to you. Please find attached the patch you accepted previously: (Version 5) Plus an incremental patch (Version 5-2) ***Please ignore patch (Version 6)*** I hope this is the easiest way for you to review it. It compiles, runs, and works on the hardware now without errors. Thanks, Damien

2 6

Re: [alsa-devel] [PATCH] sound: Add support for Creative BT-D1 via usb sound quirks
by Takashi Iwai 04 Jan '13

04 Jan '13

At Thu, 03 Jan 2013 12:59:07 +0100, Alexander Schremmer wrote: > > On 03.01.2013 10:58, Takashi Iwai wrote: > > > Thanks for the patch. The changes look OK, but could you fix trivial > > coding issues reported by scripts/checkpatch.pl and resend? > > Oops, my sending workflow ate the tabs, here it is again (hopefully now > working correctly): > > > From: Alexander Schremmer <alex(a)alexanderweb.de> > > Support the Creative BT-D1 Bluetooth USB audio device. Before this > patch, Linux had trouble finding the correct USB descriptors and bailed > out with these messages: > > no or invalid class specific endpoint descriptor > > Now it still prints these messages on hotplug: > > snd-usb-audio: probe of ...:1.0 failed with error -5 > snd-usb-audio: probe of ...:1.2 failed with error -5 > snd-usb-audio: probe of ...:1.3 failed with error -5 > > But the device works correctly, including the HID support. > > The patch is diff'ed against 3.8-rc1 but should apply to older kernels > as well. > > Signed-off-by: Alexander Schremmer <alex(a)alexanderweb.de> Thanks, applied. Takashi > > --- > diff -uprN linux-3.8-rc1/sound/usb/quirks-table.h > linux-3.8-rc1-mod/sound/usb/quirks-table.h > --- linux-3.8-rc1/sound/usb/quirks-table.h 2012-12-22 02:19:00.000000000 > +0100 > +++ linux-3.8-rc1-mod/sound/usb/quirks-table.h 2012-12-26 > 11:41:28.000000000 +0100 > @@ -50,6 +50,28 @@ > } > }, > > +{ > + /* Creative BT-D1 */ > + USB_DEVICE(0x041e, 0x0005), > + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { > + .ifnum = 1, > + .type = QUIRK_AUDIO_FIXED_ENDPOINT, > + .data = &(const struct audioformat) { > + .formats = SNDRV_PCM_FMTBIT_S16_LE, > + .channels = 2, > + .iface = 1, > + .altsetting = 1, > + .altset_idx = 1, > + .endpoint = 0x03, > + .ep_attr = USB_ENDPOINT_XFER_ISOC, > + .attributes = 0, > + .rates = SNDRV_PCM_RATE_CONTINUOUS, > + .rate_min = 48000, > + .rate_max = 48000, > + } > + } > +}, > + > /* Creative/Toshiba Multimedia Center SB-0500 */ > { > USB_DEVICE(0x041e, 0x3048), >

1 0

[alsa-devel] [sound:for-linus 4/4] sound/oss/pas2_card.c:338:5: warning: too many arguments for format
by kbuild test robot 04 Jan '13

04 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: b56458b36c8922d1d57c2320151194059dfb9a31 commit: b56458b36c8922d1d57c2320151194059dfb9a31 [4/4] sound: oss/pas2: Fix possible access out of array config: make ARCH=x86_64 allyesconfig All warnings: sound/oss/pas2_card.c: In function 'attach_pas_card': sound/oss/pas2_card.c:338:5: warning: too many arguments for format [-Wformat-extra-args] vim +338 sound/oss/pas2_card.c ^1da177e Linus Torvalds 2005-04-16 322 return pas_model; ^1da177e Linus Torvalds 2005-04-16 323 } ^1da177e Linus Torvalds 2005-04-16 324 ^1da177e Linus Torvalds 2005-04-16 325 static void __init attach_pas_card(struct address_info *hw_config) ^1da177e Linus Torvalds 2005-04-16 326 { ^1da177e Linus Torvalds 2005-04-16 327 pas_irq = hw_config->irq; ^1da177e Linus Torvalds 2005-04-16 328 ^1da177e Linus Torvalds 2005-04-16 329 if (detect_pas_hw(hw_config)) ^1da177e Linus Torvalds 2005-04-16 330 { ^1da177e Linus Torvalds 2005-04-16 331 ^1da177e Linus Torvalds 2005-04-16 332 if ((pas_model = pas_read(0xFF88))) ^1da177e Linus Torvalds 2005-04-16 333 { ^1da177e Linus Torvalds 2005-04-16 334 char temp[100]; ^1da177e Linus Torvalds 2005-04-16 335 b56458b3 Asim Kadav 2013-01-03 336 if (pas_model < 0 || b56458b3 Asim Kadav 2013-01-03 337 pas_model >= ARRAY_SIZE(pas_model_names)) { b56458b3 Asim Kadav 2013-01-03 @338 printk(KERN_ERR, "pas2 unrecognized model.\n"); b56458b3 Asim Kadav 2013-01-03 339 return; b56458b3 Asim Kadav 2013-01-03 340 } ^1da177e Linus Torvalds 2005-04-16 341 sprintf(temp, ^1da177e Linus Torvalds 2005-04-16 342 "%s rev %d", pas_model_names[(int) pas_model], ^1da177e Linus Torvalds 2005-04-16 343 pas_read(0x2789)); ^1da177e Linus Torvalds 2005-04-16 344 conf_printf(temp, hw_config); ^1da177e Linus Torvalds 2005-04-16 345 } ^1da177e Linus Torvalds 2005-04-16 346 if (config_pas_hw(hw_config)) --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

3 2

[alsa-devel] [Tiny6410/Mini6410] Sound support
by Alexander Nestorov 03 Jan '13

03 Jan '13

I'm working on a Tiny6410 device (S3C6410) on top of a 3.7 kernel and sound isn't working with the default settings. I also wasn't able to make it work enablig the related stuff to AC97 in menuconfig so I'm wondering if there is actually any support for this device? Looking at the code in sound/soc, there is something for the SMDK6410 but I'm not really sure if that's compatible with my device or even if that code is fully functional. Can I get some help with this, please? I'll appreciate any kind of help. Regards PS: I'm including Kukjin Kim and Sangbeom Kim as of recomendation of <broonie> from #alsa-soc at Freenode -- alexandernst

1 0

[alsa-devel] [PATCH v2] ASoC: pxa2xx-ac97: fix oops when codec_driver probe runs
by Mike Dunn 03 Jan '13

03 Jan '13

Change log: v2: pxa2xx_ac97_hw_probe() is called *before* snd_soc_register_dais(), and likewise pxa2xx_ac97_hw_remove() is called after snd_soc_unregister_dais(). Thanks much Mark. This patch fixes a kernel oops that occurs when the snd_soc_codec_driver probe function runs. The exception occurs when the codec calls soc_ac97_ops.reset(). The pxa2xx-ac97's clocks have not been initialized at this point, but the cpu's reset function (pxa_ac97_cold_pxa27x()) expects them to be, resulting in a NULL pointer dereference in the pxa's clk_enable(). The pxa2xx-ac97's clocks are initialized in pxa2xx_ac97_hw_probe(). To ensure that the clocks are initialized when the codec_driver probe runs, the calls to pxa2xx_ac97_hw_{probe,remove}() are moved from the cpu's dai_driver probe/remove functions to those of its platform_driver. This is necessary because snd_soc_instantiate_card() always calls the codec_driver probe before any of the dai_drivers' probes. This change leaves nothing in the cpu's dai_driver probe/remove functions, so those functions are eliminated. Tested on the pxa27x; palmtreo680 machine using the wm9712 codec. Signed-off-by: Mike Dunn <mikedunn(a)newsguy.com> --- sound/soc/pxa/pxa2xx-ac97.c | 27 ++++++++++----------------- 1 files changed, 10 insertions(+), 17 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 4b0a009..ba04f49 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -104,17 +104,6 @@ static int pxa2xx_ac97_resume(struct snd_soc_dai *dai) #define pxa2xx_ac97_resume NULL #endif -static int pxa2xx_ac97_probe(struct snd_soc_dai *dai) -{ - return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev)); -} - -static int pxa2xx_ac97_remove(struct snd_soc_dai *dai) -{ - pxa2xx_ac97_hw_remove(to_platform_device(dai->dev)); - return 0; -} - static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -184,8 +173,6 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { { .name = "pxa2xx-ac97", .ac97_control = 1, - .probe = pxa2xx_ac97_probe, - .remove = pxa2xx_ac97_remove, .suspend = pxa2xx_ac97_suspend, .resume = pxa2xx_ac97_resume, .playback = { @@ -236,15 +223,20 @@ EXPORT_SYMBOL_GPL(soc_ac97_ops); static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { + int ret; + if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); return -ENXIO; } - /* Punt most of the init to the SoC probe; we may need the machine - * driver to do interesting things with the clocking to get us up - * and running. - */ + ret = pxa2xx_ac97_hw_probe(pdev); + if (ret < 0) { + dev_err(&pdev->dev, "pxa2xx_ac97_hw_probe failed with %d\n", + ret); + return ret; + } + return snd_soc_register_dais(&pdev->dev, pxa_ac97_dai_driver, ARRAY_SIZE(pxa_ac97_dai_driver)); } @@ -252,6 +244,7 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(pxa_ac97_dai_driver)); + pxa2xx_ac97_hw_remove(pdev); return 0; } -- 1.7.8.6

1 0

[alsa-devel] [PATCH] ALSA: hda - Switch "On" and "Off" for "Mute-LED Mode" kcontrol
by David Henningsson 03 Jan '13

03 Jan '13

The vmaster hook sends 1 for enabled/unmuted and 0 for disabled/muted, but "Mute-LED Mode" being "On" refers to the LED being on, not the volume being on. Therefore "On" and "Off" should be switched. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8353c77..b8fb0a5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2531,7 +2531,7 @@ static int vmaster_mute_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static const char * const texts[] = { - "Off", "On", "Follow Master" + "On", "Off", "Follow Master" }; unsigned int index; -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH] sound: Add support for Creative BT-D1 via usb sound quirks
by Takashi Iwai 03 Jan '13

03 Jan '13

At Sat, 29 Dec 2012 14:26:25 +0100, Alexander Schremmer wrote: > > From: Alexander Schremmer <alex(a)alexanderweb.de> > > Support the Creative BT-D1 Bluetooth USB audio device. Before this > patch, Linux had trouble finding the correct USB descriptors and bailed > out with these messages: > > no or invalid class specific endpoint descriptor > > Now it still prints these messages on hotplug: > > snd-usb-audio: probe of ...:1.0 failed with error -5 > snd-usb-audio: probe of ...:1.2 failed with error -5 > snd-usb-audio: probe of ...:1.3 failed with error -5 > > But the device works correctly, including the HID support. > > The patch is diff'ed against 3.8-rc1 but should apply to older kernels > as well. > > Signed-off-by: Alexander Schremmer <alex(a)alexanderweb.de> Thanks for the patch. The changes look OK, but could you fix trivial coding issues reported by scripts/checkpatch.pl and resend? Takashi > > --- > diff -uprN linux-3.8-rc1/sound/usb/quirks-table.h > linux-3.8-rc1-mod/sound/usb/quirks-table.h > --- linux-3.8-rc1/sound/usb/quirks-table.h 2012-12-22 > 02:19:00.000000000 +0100 > +++ linux-3.8-rc1-mod/sound/usb/quirks-table.h 2012-12-26 > 11:41:28.000000000 +0100 > @@ -50,6 +50,28 @@ > } > }, > > +{ > + /* Creative BT-D1 */ > + USB_DEVICE(0x041e, 0x0005), > + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { > + .ifnum = 1, > + .type = QUIRK_AUDIO_FIXED_ENDPOINT, > + .data = &(const struct audioformat) { > + .formats = SNDRV_PCM_FMTBIT_S16_LE, > + .channels = 2, > + .iface = 1, > + .altsetting = 1, > + .altset_idx = 1, > + .endpoint = 0x03, > + .ep_attr = USB_ENDPOINT_XFER_ISOC, > + .attributes = 0, > + .rates = SNDRV_PCM_RATE_CONTINUOUS, > + .rate_min = 48000, > + .rate_max = 48000, > + } > + } > +}, > + > /* Creative/Toshiba Multimedia Center SB-0500 */ > { > USB_DEVICE(0x041e, 0x3048), >

1 0

Re: [alsa-devel] [PATCH] pas2: Added a fix where a sound card failure should not result in reading beyond array memory index.
by Takashi Iwai 03 Jan '13

03 Jan '13

At Fri, 28 Dec 2012 12:28:57 -0600, Asim Kadav wrote: > > Signed-off-by: Asim Kadav <kadav(a)cs.wisc.edu> > --- > sound/oss/pas2_card.c | 4 ++++ > 1 file changed, 4 insertions(+) > > diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c > index dabf8a8..0b8cf64 100644 > --- a/sound/oss/pas2_card.c > +++ b/sound/oss/pas2_card.c > @@ -332,6 +332,10 @@ static void __init attach_pas_card(struct address_info *hw_config) > if ((pas_model = pas_read(0xFF88))) > { > char temp[100]; > + if ((int )pas_model < 0 || (int )pas_model >= 5) { Try to run scripts/checkpatch.pl. You'll get a few errors there. But, you can simply remove the cast to int instead of fixing the coding style issues since the cast is superfluous. Also, use ARRAY_SIZE() instead of a magic number 5. > + printk ("pas2card unrecognized model.\n"); Put a KERN_ERR prefix. thanks, Takashi > + return; > + } > > sprintf(temp, > "%s rev %d", pas_model_names[(int) pas_model], > -- > 1.7.11.3

1 0

[alsa-devel] ASoC: EA3250 V2 machine driver
by Bryan Kaminski 02 Jan '13

02 Jan '13

Hello, I have an Embedded Artists LPC3250 V2 development board, which has the NXP UDA1380 audio codec on-board. I'm wondering if someone could help me get started on making an ASoC driver for it. There already exists a machine driver for a similar Phytec 3250 development board, with the same audio codec. I think it should be possible to modify the existing Phytec driver, but I'm unable to accomplish that. I modified some source and everything compiled, but I can't get the driver to load correctly. After loading the module with 'modprobe', I'm unable to get ALSA to recognize any sound cards. Can anyone give me a jump-start on this? Sorry if this is the wrong list to post on. I would post at the LPCLinux official site but it has been down for multiple weeks now. :| Bryan

1 0

[alsa-devel] [PATCH 0/2] ASoC: palm27x: machine driver fixes and updates
by Mike Dunn 02 Jan '13

02 Jan '13

Hi, A fix and an update to the palm27x machine driver. Happy New Year! Mike Dunn (2): ASoC: palm27x: fix widgets and routes in dai_link init ASoC: palm27x: register card in platform_driver probe sound/soc/pxa/palm27x.c | 36 +++++++++--------------------------- 1 files changed, 9 insertions(+), 27 deletions(-) -- 1.7.8.6

2 3

[alsa-devel] [PATCH 0/8] ASoC: OMAP-twl4030: Add support for more3 boards
by Peter Ujfalusi 02 Jan '13

02 Jan '13

Hello, This series will update the omap-twl4030 machine driver and the twl4030 codec driver to enable support for more boards. With this change we can remove two more machine drivers: zoom2 and sdp3430 since they will be supported by the common omap-twl4030 machine driver. The arch/arm/mach-omap2/ patches will go as a separate series to avoid cross tree issues. Regards, Peter --- Peter Ujfalusi (8): ASoC: twl4030: Correct the support for Voice port ASoC: zoom2: No need to configure the Voice port anymore ASoC: sdp3430: No need to configure the Voice port anymore ASoC: twl4030: Convert MICBIAS to SUPPLY widget ASoC: twl4030: Configure extmute pinmux when the dedicated pin is in use ASoC: sdp3430: No need to configure pin mux for extmute ASoC: omap-twl4030: Add support for routing, voice port and jack detect ASoC: OMAP: Remove obsolete machine drivers for Zoom2 and SDP3430 .../devicetree/bindings/sound/omap-twl4030.txt | 46 ++++ sound/soc/codecs/twl4030.c | 71 ++++-- sound/soc/omap/Kconfig | 19 +- sound/soc/omap/Makefile | 4 - sound/soc/omap/omap-twl4030.c | 204 ++++++++++++++- sound/soc/omap/omap3pandora.c | 8 +- sound/soc/omap/sdp3430.c | 278 --------------------- sound/soc/omap/zoom2.c | 207 --------------- 8 files changed, 304 insertions(+), 533 deletions(-) delete mode 100644 sound/soc/omap/sdp3430.c delete mode 100644 sound/soc/omap/zoom2.c -- 1.8.0.2

2 9

[alsa-devel] [PATCH] ASoC: pxa2xx-ac97: fix oops when codec_driver probe runs
by Mike Dunn 02 Jan '13

02 Jan '13

This patch fixes a kernel oops that occurs when the snd_soc_codec_driver probe function runs. The exception occurs when the codec calls soc_ac97_ops.reset(). The pxa2xx-ac97's clocks have not been initialized at this point, but the cpu's reset function (pxa_ac97_cold_pxa27x()) expects them to be, resulting in a NULL pointer dereference in the pxa's clk_enable(). The pxa2xx-ac97's clocks are initialized in pxa2xx_ac97_hw_probe(). To ensure that the clocks are initialized when the codec_driver probe runs, the calls to pxa2xx_ac97_hw_{probe,remove}() are moved from the cpu's dai_driver probe/remove functions to those of its platform_driver. This is necessary because snd_soc_instantiate_card() always calls the codec_driver probe before any of the dai_drivers' probes. This change leaves nothing in the cpu's dai_driver probe/remove functions, so those functions are eliminated. Tested on the pxa27x; palmtreo680 machine using the wm9712 codec. Signed-off-by: Mike Dunn <mikedunn(a)newsguy.com> --- I'm new to the ASoC code. Experts, please have a look. Thanks! sound/soc/pxa/pxa2xx-ac97.c | 38 +++++++++++++++++++------------------- 1 files changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 4b0a009..940e9f4 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -104,17 +104,6 @@ static int pxa2xx_ac97_resume(struct snd_soc_dai *dai) #define pxa2xx_ac97_resume NULL #endif -static int pxa2xx_ac97_probe(struct snd_soc_dai *dai) -{ - return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev)); -} - -static int pxa2xx_ac97_remove(struct snd_soc_dai *dai) -{ - pxa2xx_ac97_hw_remove(to_platform_device(dai->dev)); - return 0; -} - static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -184,8 +173,6 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { { .name = "pxa2xx-ac97", .ac97_control = 1, - .probe = pxa2xx_ac97_probe, - .remove = pxa2xx_ac97_remove, .suspend = pxa2xx_ac97_suspend, .resume = pxa2xx_ac97_resume, .playback = { @@ -236,21 +223,34 @@ EXPORT_SYMBOL_GPL(soc_ac97_ops); static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { + int ret; + if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); return -ENXIO; } - /* Punt most of the init to the SoC probe; we may need the machine - * driver to do interesting things with the clocking to get us up - * and running. - */ - return snd_soc_register_dais(&pdev->dev, pxa_ac97_dai_driver, - ARRAY_SIZE(pxa_ac97_dai_driver)); + ret = snd_soc_register_dais(&pdev->dev, pxa_ac97_dai_driver, + ARRAY_SIZE(pxa_ac97_dai_driver)); + if (ret < 0) { + dev_err(&pdev->dev, "snd_soc_register_dais failed with %d\n", + ret); + return ret; + } + + ret = pxa2xx_ac97_hw_probe(pdev); + if (ret < 0) { + dev_err(&pdev->dev, "pxa2xx_ac97_hw_probe failed with %d\n", + ret); + snd_soc_unregister_dais(&pdev->dev, + ARRAY_SIZE(pxa_ac97_dai_driver)); + } + return ret; } static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { + pxa2xx_ac97_hw_remove(pdev); snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(pxa_ac97_dai_driver)); return 0; } -- 1.7.8.6

2 1

[alsa-devel] Looking for ALSA developer
by Eben Upton 01 Jan '13

01 Jan '13

We're looking for someone to do some contract development on the ALSA driver for the Raspberry Pi. Would anyone here be interested in taking this on? A little background: Our driver is architecturally fairly similar to a USB sound card driver, sitting on top of a stream-like connection to a remote entity implemented in software on an internal DSP. Due to our limited understanding of ALSA internals, there are a number of bugs which lead to underrun conditions in common audio applications, and we'd like to pay someone on and hourly basis to go through the failing cases, root cause them, and propose patches. Eben Upton Executive Director, Raspberry Pi Foundation

1 0

[alsa-devel] [PATCH 0/6] ARM: OMAP3: Audio support update for omap-twl4030
by Peter Ujfalusi 31 Dec '12

31 Dec '12

Hello, omap-twl4030 ASoC machine driver can support more boards with the following series: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-December/058119.h… Rework the omap-twl4030 audio support code in arch/arm/mach-omap2/ to take advantages of this. The same machine driver can support the zoom2 and sdp3430 boards as well. Regards, Peter --- Peter Ujfalusi (6): ARM: OMAP: 3430sdp: Enable extmute functionality for audio ARM: OMAP: zoom: Zoom2 does not have extmute functionality ARM: OMAP2+: twl-common: Add default twl4030 audio configuration ARM: OMAP2+: twl-common: Allow boards to customize the twl4030 audio setup ARM: OMAP: zoom: Audio support via the common omap-twl4030 machine driver ARM: OMAP: sdp3430: Audio support via the common omap-twl4030 machine driver arch/arm/mach-omap2/board-3430sdp.c | 20 +++++++++++++++++++ arch/arm/mach-omap2/board-cm-t35.c | 2 +- arch/arm/mach-omap2/board-devkit8000.c | 2 +- arch/arm/mach-omap2/board-igep0020.c | 2 +- arch/arm/mach-omap2/board-omap3beagle.c | 2 +- arch/arm/mach-omap2/board-omap3evm.c | 2 +- arch/arm/mach-omap2/board-overo.c | 2 +- arch/arm/mach-omap2/board-zoom-peripherals.c | 29 ++++++++++++++++++++-------- arch/arm/mach-omap2/twl-common.c | 17 ++++++++++------ arch/arm/mach-omap2/twl-common.h | 3 ++- 10 files changed, 60 insertions(+), 21 deletions(-) -- 1.8.0.2

1 6

Re: [alsa-devel] Help with adding mic bias to omap-twl4030 audio driver
by Peter Ujfalusi 31 Dec '12

31 Dec '12

Hi Neil, On 12/18/2012 12:18 AM, NeilBrown wrote: > Hi Peter, > I recently discovered your omap-twl4030 driver and thought to use it to > replace the hacked-around driver I'm using for the GTA04 mobile phone > platform. > > It works fine for playing sound out the speaker, but the microphone doesn't > work - I think because there is no mechanism to enable the microphone bias. > > I copied some bits out of the old driver - just enough to make it work. > However this probably makes the code specific to my platform. > > So I'm wondering if you could give me some guidelines on how this "should" > be done. Should these tables be passed in with the platform_data (I haven't > graduated to device-tree yet)? I'm going to send a series soon to update the omap-twl4030 machine driver. This update going to add support for microphones and all other routings as well. With the upcoming series the omap-twl4030 will support zoom2 and sdp3430. I'll CC you with the series so you can take a look if it is going to help with your board. > > My 'hacked-up' driver has a few more entries in this table than just those > few, but I wanted to strip it down to what is needed for the main microphone > to work. > > Thanks for any help you can provide, > NeilBrown > > P.S is there some good reason to use the name > struct omap_tw4030_pdata > rather than > struct omap_twl4030_pdata > > (i.e. with the 'l')?? > > diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c > index 3b97b879..aa4d866 100644 > --- a/sound/soc/omap/omap-twl4030.c > +++ b/sound/soc/omap/omap-twl4030.c > @@ -87,6 +87,32 @@ static struct snd_soc_ops omap_twl4030_ops = { > .hw_params = omap_twl4030_hw_params, > }; > > +static const struct snd_soc_dapm_widget gta04_dapm_widgets[] = { > + SND_SOC_DAPM_MIC("Internal Mic", NULL), > +}; > + > +static const struct snd_soc_dapm_route audio_map[] = { > + {"MAINMIC", NULL, "Mic Bias 1"}, > + {"Mic Bias 1", NULL, "Internal Mic"}, > +}; > + > +static int omap3gta04_init(struct snd_soc_pcm_runtime *runtime) > +{ > + int ret; > + struct snd_soc_codec *codec = runtime->codec; > + struct snd_soc_dapm_context *dapm = &codec->dapm; > + > + ret = snd_soc_dapm_new_controls(dapm, gta04_dapm_widgets, > + ARRAY_SIZE(gta04_dapm_widgets)); > + if (ret < 0) > + return ret; > + > + snd_soc_dapm_add_routes(dapm, audio_map, > + ARRAY_SIZE(audio_map)); > + > + return snd_soc_dapm_sync(dapm); > +} > + > /* Digital audio interface glue - connects codec <--> CPU */ > static struct snd_soc_dai_link omap_twl4030_dai_links[] = { > { > @@ -97,6 +123,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = { > .platform_name = "omap-pcm-audio", > .codec_name = "twl4030-codec", > .ops = &omap_twl4030_ops, > + .init = &omap3gta04_init, > }, > }; > > -- Péter

1 0

[alsa-devel] [PATCH v2 0/2] ASoC: palm27x: machine driver fixes and updates
by Mike Dunn 29 Dec '12

29 Dec '12

Hi, A fix and an update to the palm27x machine driver. Change log: v2: Patch #2: in palm27x_asoc_remove(), the call to platform_device_unregister() is replaced with snd_soc_unregister_card(). This version v2 is unsolicited; I noticed the omission today. Patch preparation error; tripping over my own patches. No new ones for a while! Thanks for your patience. Mike Dunn (2): ASoC: palm27x: fix widgets and routes in dai_link init ASoC: palm27x: register card in platform_driver probe sound/soc/pxa/palm27x.c | 38 ++++++++++---------------------------- 1 files changed, 10 insertions(+), 28 deletions(-) -- 1.7.8.6

1 2

[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for Focusrite Scarlett 18i6
by Martin Schwenke 29 Dec '12

29 Dec '12

Probing this device currently fails in snd_usb_audio_probe() because the call to snd_usb_create_mixer() fails. This is due to unknown or non-standard interface descriptor subtypes in parse_audio_unit(): usbaudio: unit 51: unexpected type 0x09 snd-usb-audio: probe of 1-8:1.0 failed with error -5 Some people are working around this by recompiling usb-audio with the call to snd_usb_create_mixer() commented out. It would be nice to avoid that. While the best idea would be to look into the mixer creation failure, a reasonable short-term solution is to use quirks to only probe the trouble-free interfaces. This allows audio and MIDI interfaces to be used without any obvious issues. Interface 0 is the main one to ignore. It contains lots of control-fu, including the unexpected interface descriptor subtypes. Interface 5 is for firmware updates and I'm not sure how to get support for this. Interface 3 is some sort of control interface that I don't understand: Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 9 bInCollection 1 baInterfaceNr( 0) 1 Signed-off-by: Martin Schwenke <martin(a)meltin.net> --- sound/usb/quirks-table.h | 52 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 52 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 88d8ceb..c9dde5f 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2882,6 +2882,58 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* + * Focusrite Scarlett 18i6 + * + * Avoid mixer creation, which otherwise fails because some of + * the interface descriptor subtypes for interface 0 are + * unknown. That should be fixed or worked-around but this at + * least allows the device to be used successfully with a DAW + * and an external mixer. See comments below about other + * ignored interfaces. + */ + USB_DEVICE(0x1235, 0x8004), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Scarlett 18i6", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = & (const struct snd_usb_audio_quirk[]) { + { + /* InterfaceSubClass 1 (Control Device) */ + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + /* InterfaceSubClass 1 (Control Device) */ + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + /* InterfaceSubClass 1 (Device Firmware Update) */ + .ifnum = 5, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, + +{ + /* * Some USB MIDI devices don't have an audio control interface, * so we have to grab MIDI streaming interfaces here. */ -- 1.7.10.4

4 5

[alsa-devel] EMU-1212m "no Audiodock - no firmware" patch
by oeai 29 Dec '12

29 Dec '12

Aloha! i've been posting about my problem earlier, but i decided to DIY the problem was that when going up from suspend the firmware for additional card (that i don't have) was trying to load, -but there's no card - sorry, i'm gonna check few more thousand times that caused my cpu to load insanely so in the patch i've changed few more strings for debugging but actually the one (but not the only) bug is here *** 728,739 **** ! if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) { --- 729,740 ---- ! if (reg & !EMU_HANA_OPTION_DOCK_OFFLINE) { the thing is in the "!"EMU_HANA_OPTION_DOCK_OFFLINE -> then the firmware should be loaded. and to debug some errors if something i've put the one more else state so if no audiodock - it could continue to load config *** 777,784 **** /* Unmute all. Default is muted after a firmware load */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); } } ! snd_printk(KERN_INFO "emu1010: firmware thread stopping\n"); return 0; } --- 778,790 ---- /* Unmute all. Default is muted after a firmware load */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); } + else if ((reg & 0x3f) == 0x15) + { + snd_printk(KERN_INFO "emu1212m: trying to load config emu_hana_fpga_config"); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG,0x02); + } } ! snd_printk(KERN_INFO "emu1010: emu dock firmware thread stopping\n"); return 0; } *************** then i've commented the power_down for audiodock - it should be in other place, but as i'm 1212m user - it works for me - you know. *** 841,847 **** outl(0x0005a000, emu->port + HCFG); /* Disable 48Volt power to Audio Dock */ ! snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg); --- 847,853 ---- outl(0x0005a000, emu->port + HCFG); /* Disable 48Volt power to Audio Dock */ ! /*snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0);*/ /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg); *************** and here i've changed that the emu_model was firstly enabled and then ecard - since i don't know what's the ecard i thought that it is a second module of my emu card. *** 2079,2088 **** { if (emu->card_capabilities->ca_cardbus_chip) snd_emu10k1_cardbus_init(emu); ! if (emu->card_capabilities->ecard) ! snd_emu10k1_ecard_init(emu); ! else if (emu->card_capabilities->emu_model) snd_emu10k1_emu1010_init(emu); else snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE); snd_emu10k1_init(emu, emu->enable_ir, 1); --- 2098,2111 ---- { if (emu->card_capabilities->ca_cardbus_chip) snd_emu10k1_cardbus_init(emu); ! ! if (emu->card_capabilities->emu_model) ! { snd_emu10k1_emu1010_init(emu); + + } + else if (emu->card_capabilities->ecard) + snd_emu10k1_ecard_init(emu); else snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE); snd_emu10k1_init(emu, emu->enable_ir, 1); Now i still have the problem of waking up i cannot request_firmware() properly on resume and it is trying to load nothing and other changes in patch - that was trying to debug the situation so i can at least alsa-reload at anytime. also i'd like to ask if i can use the ctxfi modules - CA0110-IBG i got xfi card (it is second of 5) that actually uses as i believe the same logic - maybe a lil modified, but not so much imho. -- RA OEAI

3 8

[alsa-devel] Fwd: Request for HW auto-recognition: HP Pavilion dv7 with STAC92xx
by Nicolas Maître 27 Dec '12

27 Dec '12

Hi, At Wed, 12 Dec 2012 18:08:25 +0100, Tomas Pospisek wrote: > > Hello, > > I have a HP Pavilion dv7 laptop (LS066EA#UUZ), with: > > # cat /proc/asound/card0/codec* | grep Codec > Codec: IDT 92HD81B1X5 > Codec: Intel CougarPoint HDMI > > when play sound without any tweaks, I get a very weak sound output. If I > want to have "beats audio", that is subwoofer and all speakers, I need to > set: > > # cat /etc/modprobe.d/alsa-base.conf > options snd-hda-intel model=hp-dv7-4000 > > Could the snd_hda_intel alsa driver be updated to recognize and setup the > driver accordingly? > > alsa-info.sh output here: > > > http://www.alsa-project.org/db/?f=54ea874f43f12c1572aff3ee3113dc4b0ad0b398 I'm not quite sure that I have the exact same hardware, but I think so. I have 5 speakers (rear left and right, front left and right + subwoofer) Here is my alsa-info: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1056669/+attachm… In my case, model=hp-dv7-4000 doesn't fully solve the problem. The subwoofer works but it is not mapped as such. After some tests, it seems that the alsa's speakers rear left and rear right are mapped to the subwoofer. The front ones are alright. With the model=hp-dv5 that I'm using, the subwoofer isnt recognized at all, but the rear speakers are correctly mapped. Also, with hp-dv7-4000, the internal microphone is still used when a micro jack is plugged in. That's not the case with hp-dv5. If you think there's a configuration issue on my side, please let me know, but I think hp-dv5 is actually a better default choice if I have to choose between both. Kind regards, -- nimai

4 7

Re: [alsa-devel] [PATCH 001/001] ASoC: Replace max98090 Device Driver
by Mark Brown 27 Dec '12

27 Dec '12

On Thu, Dec 13, 2012 at 11:19:29AM -0800, Jerry Wong wrote: For legibility please submit future revisions as a two stage patch series, one removing the old driver and another adding the new one. > +/* codec platform data */ > +struct max98090_pdata { > + > + int irq; This should be passed in using the normal mechanism provided by the bus, not custom implemented. > + int power_over_performance; This doesn't look like platform data to me, it looks like something that should be configured dynamically at runtime. > + /* Equalizers for DAC */ > + struct max98090_eq_cfg *eq_cfg; > + unsigned int eq_cfgcnt; Coefficients should be configured using SND_SOC_BYTES(). Some older drivers use this method but it's deprecated. > + { 0xE0, 0x00 }, /* E0 */ > + { 0xE1, 0x00 }, /* E1 */ > + { 0xE2, 0x00 }, /* E2 */ Do these number-only registers actually exist in the published register map or are they undefined? If they're undefined and not used by software there's no need to include them here. > + switch (reg) { > + case M98090_REG_01_IRQ_STATUS: > + case M98090_REG_02_JACK_STATUS: > + case M98090_REG_FF_REV_ID: Including the register number in the constants seems to rather defeat the point of having symbolic names for the registers; it means people still have to remember the register numbers. > +static int max98090_get_enab_tlv(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) What's the relevance of _tlv here? It looks like these are normal get and sets. > +static const char * max98090_dsts_text[] = > + { "None", "Left", "Right", "Left and Right" }; > + > +static const struct soc_enum max98090_dsts_enum = > + SOC_ENUM_SINGLE(M98090_REG_1A_LVL_SIDETONE, M98090_DSTS_SHIFT, > + ARRAY_SIZE(max98090_dsts_text), max98090_dsts_text); > +static const struct snd_kcontrol_new max98090_snd_controls[] = { > + SOC_SINGLE("DMIC MIC Left Enable", M98090_REG_13_MIC_CFG1, > + M98090_DIGMICL_SHIFT, M98090_DIGMICL_NUM - 1, 0), > + SOC_SINGLE("DMIC MIC Right Enable", M98090_REG_13_MIC_CFG1, > + M98090_DIGMICR_SHIFT, M98090_DIGMICR_NUM - 1, 0), This looks like it should be DAPM. > + SOC_ENUM_EXT("External MIC Mux", max98090_extmic_mux_enum, > + max98090_extmic_mux_get, max98090_extmic_mux_set), Likewise. > + SOC_ENUM("Digital Sidetone Source", max98090_dsts_enum), DAPM. > + SOC_SINGLE_EXT_TLV("Digital Sidetone Level Volume", > + M98090_REG_1A_LVL_SIDETONE, M98090_DVST_SHIFT, > + M98090_DVST_NUM - 1, 1, max98090_get_enab_tlv, > + max98090_put_enab_tlv, max98090_micboost_tlv), Level Volume? > + SOC_SINGLE_TLV("Digital Gain Volume", M98090_REG_27_DAI_LVL, > + M98090_DAI_LVL_DVG_SHIFT, M98090_DAI_LVL_DVG_NUM - 1, 0, > + max98090_dvg_tlv), Gain Volume? > + SOC_SINGLE("Digital EQ 3 Band Enable", M98090_REG_41_DSP_EQ_EN, > + M98090_EQ3BANDEN_SHIFT, M98090_EQ3BANDEN_NUM - 1, 0), Switch. > + SOC_SINGLE("DAC HP Playback Performance Mode", M98090_REG_43_DAC_CFG, > + M98090_DAC_PERFMODE_SHIFT, M98090_DAC_PERFMODE_NUM - 1, 0), > + SOC_SINGLE("DAC High Performance Mode", M98090_REG_43_DAC_CFG, > + M98090_DACHP_SHIFT, M98090_DACHP_NUM - 1, 0), These should either be enumerations with real names of Switches if it's just an on/off (looks like the former). > + SOC_SINGLE_TLV("Headphone Left Volume", M98090_REG_2C_LVL_HP_LEFT, > + M98090_HPVOLL_SHIFT, M98090_HPVOLL_NUM - 1, 0, > + max98090_hp_tlv), > + SOC_SINGLE_TLV("Headphone Right Volume", M98090_REG_2D_LVL_HP_RIGHT, > + M98090_HPVOLR_SHIFT, M98090_HPVOLR_NUM - 1, 0, > + max98090_hp_tlv), Should be a single stereo control (and similarly for the other output controls). > + SOC_SINGLE("Quick Setup System Clock", M98090_REG_04_QCFG_SYS_CLK, > + M98090_CLK_ALL_SHIFT, M98090_CLK_ALL_NUM - 1, 0), What do these controls do? > + if (val >= 1) { > + if (w->reg == M98090_REG_10_LVL_MIC1) > + max98090->mic1pre = val - 1; /* Update for volatile */ > + else > + max98090->mic2pre = val - 1; /* Update for volatile */ > + } Use braces consistently within an if statemeent - if you use them for the outer one use them for the inner one too. > +static const struct snd_soc_dapm_route max98090_dapm_routes[] = { > + > + {"MIC1 Input", NULL, "MIC1"}, > + {"MIC2 Input", NULL, "MIC2"}, > + {"MIC1 Input", NULL, "MICBIAS"}, > + {"MIC2 Input", NULL, "MICBIAS"}, Unless this chip is *very* unusual I expect the bias is something that's connected externally and therefore these connections should be being made by the machine driver. > + > + /* Check for user calculated MI and NI ratios */ > + for (i = 0; i < ARRAY_SIZE(user_pclk_rates); i++) { > + if ((user_pclk_rates[i] == max98090->sysclk) && > + (user_lrclk_rates[i] == max98090->lrclk)) { > + dev_info(codec->dev, > + "Found user supported PCLK to LRCLK rates\n"); > + dev_info(codec->dev, "i %d ni %lld mi %lld\n", > + i, ni_value[i], mi_value[i]); dev_dbg() would be fine but dev_info() is too loud. > +/* > + * This accommodates an inverted logic in the MAX98090 chip > + * for Bit Clock Invert (BCI). The inverted logic is only seen > + * for the case of TDM mode. The remaining cases have normal logic. > + */ > + if (max98090->tdm_slots > 1) { > + regval ^= M98090_DAI_BCI_MASK; > + } Coding style, the indentation is inconsistent here and no braces for single line if () statements. > + if (max98090->jack_state == M98090_JACK_STATE_HEADSET) { > + /* > + * Set to normal bias level. > + */ > + snd_soc_update_bits(codec, M98090_REG_12_MIC_BIAS, > + M98090_MBVSEL_MASK, M98090_MBVSEL_2V4); > + > + snd_soc_update_bits(codec, M98090_REG_3E_PWR_EN_IN, > + M98090_PWR_MBEN_MASK, M98090_PWR_MBEN_MASK); > + } This looks suspicous, why does the headset state have any bearing here? > +static int max98090_dai_digital_mute(struct snd_soc_dai *codec_dai, int mute) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + int regval; > + > + dev_info(codec->dev, "max98090_dai_digital_mute\n"); Drop this, it's not conveying any useful information. > +static void max98090_dmic_switch(struct snd_soc_codec *codec, int state) > +{ I have no idea what this function is supposed to do... > +static void max98090_headset_button_event(struct snd_soc_codec *codec) > +{ > + dev_info(codec->dev, "max98090_headset_button_event\n"); > +} Delete this, it doesn't do anything except log. > + switch (reg & (M98090_LSNS_MASK | M98090_JKSNS_MASK)) { > + case M98090_LSNS_MASK | M98090_JKSNS_MASK: > + { > + dev_info(codec->dev, "No Headset Detected\n"); > + > + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; > + > + max98090_set_bias_level(codec, SND_SOC_BIAS_STANDBY); > + > + snd_soc_dapm_disable_pin(&codec->dapm, "HPL"); > + snd_soc_dapm_disable_pin(&codec->dapm, "HPR"); > + snd_soc_dapm_force_enable_pin(&codec->dapm, "SPKL"); > + snd_soc_dapm_force_enable_pin(&codec->dapm, "SPKR"); > + snd_soc_dapm_disable_pin(&codec->dapm, "MIC1"); > + snd_soc_dapm_disable_pin(&codec->dapm, "MIC2"); > + snd_soc_dapm_force_enable_pin(&codec->dapm, "DMIC1"); > + snd_soc_dapm_force_enable_pin(&codec->dapm, "DMIC2"); > + max98090_dmic_switch(codec, 1); > + > + break; > + } This doesn't look obviously sensible... calling set_bias_level() outside of DAPM is going to get overridden in pretty short order, none of the DAPM pin setting looks like a good idea (especially not forcing on the speakers whenever there's no headset) and there's no DAPM sync to make any of this take effect. I'm a bit confused about the intended effect. I also can't see any interaction with soc-jack... What is this code actually intended to do? > + dev_info(codec->dev, "***** max98090_interrupt *****\n"); > + There's lots of these noisy, uninformative prints at info level throughout the driver. > + /* Send work to be scheduled */ > + if (active & M98090_IRQ_CLD_MASK) { > + dev_info(codec->dev, "M98090_IRQ_CLD_MASK\n"); > + } This doesn't actually schedule anything... > + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); > + if (ret != 0) { > + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); > + return ret; > + } SND_SOC_REGMAP. > + if ( (request_threaded_irq(pdata->irq, NULL, > + max98090_interrupt, IRQF_TRIGGER_FALLING, > + "max98090_interrupt", codec)) < 0) { > + dev_info(codec->dev, "request_irq failed\n"); > + } > + /* > + * Clear any old interrupts. Coding style and it's not terribly helpful to hide the return code from users. > + /* Turn on VCM bandgap reference */ > + snd_soc_write(codec, M98090_REG_42_BIAS_CNTL, > + M98090_VCM_MODE_MASK); This shouldn't be managed at runtime? > +/* > + * Driver revision > + */ > +#define MAX98090_REVISION "0.01.00" The kernel already has perfectly good versioning.

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm8985: Refactor set_pll code to avoid gcc warnings
by Fabio Estevam 27 Dec '12

27 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Refactor set_pll code to avoid the following warnings: sound/soc/codecs/wm8985.c:852:50: warning: 'pll_div.k' may be used uninitialized in this function sound/soc/codecs/wm8985.c:849:9: warning: 'pll_div.n' may be used uninitialized in this function sound/soc/codecs/wm8985.c:848:23: warning: 'pll_div.div2' may be used uninitialized in this function Do the same as in commit 86ce6c9a (ASoC: WM8804: Refactor set_pll code to avoid GCC warnings). Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8985.c | 43 ++++++++++++++++++++----------------------- 1 file changed, 20 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index ab37826..dd6ce3b 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -830,33 +830,30 @@ static int wm8985_set_pll(struct snd_soc_dai *dai, int pll_id, struct pll_div pll_div; codec = dai->codec; - if (freq_in && freq_out) { + if (!freq_in || !freq_out) { + /* disable the PLL */ + snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, + WM8985_PLLEN_MASK, 0); + } else { ret = pll_factors(&pll_div, freq_out * 4 * 2, freq_in); if (ret) return ret; - } - /* disable the PLL before reprogramming it */ - snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, - WM8985_PLLEN_MASK, 0); - - if (!freq_in || !freq_out) - return 0; - - /* set PLLN and PRESCALE */ - snd_soc_write(codec, WM8985_PLL_N, - (pll_div.div2 << WM8985_PLL_PRESCALE_SHIFT) - | pll_div.n); - /* set PLLK */ - snd_soc_write(codec, WM8985_PLL_K_3, pll_div.k & 0x1ff); - snd_soc_write(codec, WM8985_PLL_K_2, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8985_PLL_K_1, (pll_div.k >> 18)); - /* set the source of the clock to be the PLL */ - snd_soc_update_bits(codec, WM8985_CLOCK_GEN_CONTROL, - WM8985_CLKSEL_MASK, WM8985_CLKSEL); - /* enable the PLL */ - snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, - WM8985_PLLEN_MASK, WM8985_PLLEN); + /* set PLLN and PRESCALE */ + snd_soc_write(codec, WM8985_PLL_N, + (pll_div.div2 << WM8985_PLL_PRESCALE_SHIFT) + | pll_div.n); + /* set PLLK */ + snd_soc_write(codec, WM8985_PLL_K_3, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8985_PLL_K_2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8985_PLL_K_1, (pll_div.k >> 18)); + /* set the source of the clock to be the PLL */ + snd_soc_update_bits(codec, WM8985_CLOCK_GEN_CONTROL, + WM8985_CLKSEL_MASK, WM8985_CLKSEL); + /* enable the PLL */ + snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, + WM8985_PLLEN_MASK, WM8985_PLLEN); + } return 0; } -- 1.7.9.5

2 3

Re: [alsa-devel] [PATCH V5 2/2] ASoC: SAMSUNG: Add DT support for i2s
by Padma Venkat 27 Dec '12

27 Dec '12

Hi, On Sat, Dec 22, 2012 at 12:32 AM, Kukjin Kim <kgene.kim(a)samsung.com> wrote: > Padma Venkat wrote: >> >> Hi, >> >> On Wed, Dec 19, 2012 at 10:39 PM, Mark Brown >> <broonie(a)opensource.wolfsonmicro.com> wrote: >> > On Wed, Dec 19, 2012 at 01:24:14PM +0000, Grant Likely wrote: >> >> On Thu, 13 Dec 2012 16:12:53 +0530, Padmavathi Venna >> <padma.v(a)samsung.com> wrote: >> > >> >> > +- compatible : "samsung,samsung-i2s" >> > >> >> Isn't that kind of redundant? :-) >> > >> >> The format of the compatible strings should be "<vendor>,<part- >> number>-i2s". >> >> Please be specific about the part number that you're doing the binding >> >> for. For example; use "samsung,exynos4210-i2s" instead of >> "samsung,exynos-i2s". >> > >> > There are actually versioned IPs here (where the versions are used >> > publically in a few places) but it's not clearly documented which is >> > which. It would be reasonable to use the IP versions here I think. >> >> Samsung has three i2s drivers one for s3c24xx, one for s3c2412 and one >> for rest of the platforms. The above mentioned other platforms has >> Version 3/4/5 of i2s controllers. This dt binding is for for the i2s > > Where is the version defined such as 3, 4, 5? So, what is the > "sound/soc/Samsung/s3c-i2s-v2.[ch]"? Versions 3, 4, 5 are defined in dev-audio.c file of corresponding platforms. s3c-i2s-v2 is used in s3c2412 platform. > >> driver that has support for Version 3/4/5 of i2s controller. So >> "samsung,i2s-v5" is okay as compatible name? Please suggest me. >> > I agree with using version here but we need some consensus about that. > > - Kukjin > > -- > To unsubscribe from this list: send the line "unsubscribe linux-samsung-soc" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html Thanks Padma

2 2

[alsa-devel] [PATCH] ASoC: cs42l52: Convert to devm_input_allocate_device()
by Axel Lin 27 Dec '12

27 Dec '12

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 99bb1c6..73a8049 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1040,7 +1040,7 @@ static void cs42l52_init_beep(struct snd_soc_codec *codec) struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); int ret; - cs42l52->beep = input_allocate_device(); + cs42l52->beep = devm_input_allocate_device(codec->dev); if (!cs42l52->beep) { dev_err(codec->dev, "Failed to allocate beep device\n"); return; @@ -1061,7 +1061,6 @@ static void cs42l52_init_beep(struct snd_soc_codec *codec) ret = input_register_device(cs42l52->beep); if (ret != 0) { - input_free_device(cs42l52->beep); cs42l52->beep = NULL; dev_err(codec->dev, "Failed to register beep device\n"); } @@ -1078,7 +1077,6 @@ static void cs42l52_free_beep(struct snd_soc_codec *codec) struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); device_remove_file(codec->dev, &dev_attr_beep); - input_unregister_device(cs42l52->beep); cancel_work_sync(&cs42l52->beep_work); cs42l52->beep = NULL; -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH 1/2] ASoC: core: fix the memory leak in case of device_add() failure
by Mark Brown 27 Dec '12

27 Dec '12

On Wed, Dec 26, 2012 at 12:56:05AM +0800, Chuansheng Liu wrote: > > After called device_initialize(), even device_add() returns > error, we still need use the put_device() to release the reference > to call rtd_release(), which will do the free() action. Applied both, thanks.

1 0

[alsa-devel] [PATCH] ASoC: cs42l52: Catch no-match case in cs42l52_get_clk
by Axel Lin 24 Dec '12

24 Dec '12

In the case of no-match, return -EINVAL instead of 0. Since we assign i to ret in the for loop, ret always less than ARRAY_SIZE(clk_map_table). Thus remove the boundary checking for ret. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 99bb1c6..9811a54 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -737,7 +737,7 @@ static const struct cs42l52_clk_para clk_map_table[] = { static int cs42l52_get_clk(int mclk, int rate) { - int i, ret = 0; + int i, ret = -EINVAL; u_int mclk1, mclk2 = 0; for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { @@ -749,8 +749,6 @@ static int cs42l52_get_clk(int mclk, int rate) } } } - if (ret > ARRAY_SIZE(clk_map_table)) - return -EINVAL; return ret; } -- 1.7.9.5

3 2

Re: [alsa-devel] [PATCH 3/7] ASoC: tegra: setup DAP3<->DAC3 connection by default
by Mark Brown 24 Dec '12

24 Dec '12

On Thu, Dec 20, 2012 at 12:17:32AM +0100, Lucas Stach wrote: > This connection is used by the AC97 controller. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 4/7] ASoC: tegra: add function to set ac97 rate
by Mark Brown 24 Dec '12

24 Dec '12

On Thu, Dec 20, 2012 at 12:17:33AM +0100, Lucas Stach wrote: > AC97 uses a fixed rate, unrelated to the sample rate. Add a function to > make the setup more trivial. Applied, though the above isn't strictly true - a number of later AC'97 CODECs had PLLs which allowed them to use non-standard AC'97 MCLKs.

1 0

[alsa-devel] [PATCH v2] ASoC: wm8993: Refactor set_pll code to avoid GCC warnings
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Refactor set_pll code to avoid the following warnings: sound/soc/codecs/wm8983.c:873:40: warning: 'pll_div.k' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm8983.c:870:9: warning: 'pll_div.n' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm8983.c:869:23: warning: 'pll_div.div2' may be used uninitialized in this function [-Wuninitialized] Do the same as in commit 86ce6c9a (ASoC: WM8804: Refactor set_pll code to avoid GCC warnings). Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Fix the comment sound/soc/codecs/wm8983.c | 41 ++++++++++++++++++++++------------------- 1 file changed, 22 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 9fe1e04..e7c73ff 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -851,30 +851,33 @@ static int wm8983_set_pll(struct snd_soc_dai *dai, int pll_id, struct pll_div pll_div; codec = dai->codec; - if (freq_in && freq_out) { + if (!freq_in || !freq_out) { + /* disable the PLL */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, 0); + return 0; + } else { ret = pll_factors(&pll_div, freq_out * 4 * 2, freq_in); if (ret) return ret; - } - - /* disable the PLL before re-programming it */ - snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, - WM8983_PLLEN_MASK, 0); - if (!freq_in || !freq_out) - return 0; + /* disable the PLL before re-programming it */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, 0); + + /* set PLLN and PRESCALE */ + snd_soc_write(codec, WM8983_PLL_N, + (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT) + | pll_div.n); + /* set PLLK */ + snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18)); + /* enable the PLL */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, WM8983_PLLEN); + } - /* set PLLN and PRESCALE */ - snd_soc_write(codec, WM8983_PLL_N, - (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT) - | pll_div.n); - /* set PLLK */ - snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff); - snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18)); - /* enable the PLL */ - snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, - WM8983_PLLEN_MASK, WM8983_PLLEN); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: wm8804: Remove redundant check
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> The condition "if (!freq_in || !freq_out)" has already been tested previously, so no need to do it again. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8804.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index d321a87..1704b1e 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -395,9 +395,6 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, /* power down the PLL before reprogramming it */ snd_soc_update_bits(codec, WM8804_PWRDN, 0x1, 0x1); - if (!freq_in || !freq_out) - return 0; - /* set PLLN and PRESCALE */ snd_soc_update_bits(codec, WM8804_PLL4, 0xf | 0x10, pll_div.n | (pll_div.prescale << 4)); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: ASoC: cs4271: add optional soft reset workaround
by Daniel Mack 24 Dec '12

24 Dec '12

The CS4271 requires its LRCLK and MCLK to be stable before its RESET line is de-asserted. That also means that clocks cannot be changed without putting the chip back into hardware reset, which also requires a complete re-initialization of all registers. One (undocumented) workaround is to assert and de-assert the PDN bit in the MODE2 register. This patch adds a new flag to both the DT bindings as well as to the platform data to enable that workaround. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- Documentation/devicetree/bindings/sound/cs4271.txt | 12 ++++++++ include/sound/cs4271.h | 15 ++++++++++ sound/soc/codecs/cs4271.c | 34 ++++++++++++++++++++++ 3 files changed, 61 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt index a850fb9..e2cd1d7 100644 --- a/Documentation/devicetree/bindings/sound/cs4271.txt +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -20,6 +20,18 @@ Optional properties: !RESET pin - cirrus,amuteb-eq-bmutec: When given, the Codec's AMUTEB=BMUTEC flag is enabled. + - cirrus,enable-soft-reset: + The CS4271 requires its LRCLK and MCLK to be stable before its RESET + line is de-asserted. That also means that clocks cannot be changed + without putting the chip back into hardware reset, which also requires + a complete re-initialization of all registers. + + One (undocumented) workaround is to assert and de-assert the PDN bit + in the MODE2 register. This workaround can be enabled with this DT + property. + + Note that this is not needed in case the clocks are stable + throughout the entire runtime of the codec. Examples: diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index dd8c48d..70f4535 100644 --- a/include/sound/cs4271.h +++ b/include/sound/cs4271.h @@ -20,6 +20,21 @@ struct cs4271_platform_data { int gpio_nreset; /* GPIO driving Reset pin, if any */ bool amutec_eq_bmutec; /* flag to enable AMUTEC=BMUTEC */ + + /* + * The CS4271 requires its LRCLK and MCLK to be stable before its RESET + * line is de-asserted. That also means that clocks cannot be changed + * without putting the chip back into hardware reset, which also requires + * a complete re-initialization of all registers. + * + * One (undocumented) workaround is to assert and de-assert the PDN bit + * in the MODE2 register. This workaround can be enabled with the + * following flag. + * + * Note that this is not needed in case the clocks are stable + * throughout the entire runtime of the codec. + */ + bool enable_soft_reset; }; #endif /* __CS4271_H */ diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index ac99238..ffa72dd 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -167,6 +167,8 @@ struct cs4271_private { int gpio_nreset; /* GPIO that disable serial bus, if any */ int gpio_disable; + /* enable soft reset workaround */ + bool enable_soft_reset; }; /* @@ -325,6 +327,33 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, int i, ret; unsigned int ratio, val; + if (cs4271->enable_soft_reset) { + /* + * Put the codec in soft reset and back again in case it's not + * currently streaming data. This way of bringing the codec in + * sync to the current clocks is not explicitly documented in + * the data sheet, but it seems to work fine, and in contrast + * to a read hardware reset, we don't have to sync back all + * registers every time. + */ + + if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + !dai->capture_active) || + (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + !dai->playback_active)) { + ret = snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_PDN, + CS4271_MODE2_PDN); + if (ret < 0) + return ret; + + ret = snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; + } + } + cs4271->rate = params_rate(params); /* Configure DAC */ @@ -484,6 +513,10 @@ static int cs4271_probe(struct snd_soc_codec *codec) if (of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) amutec_eq_bmutec = true; + + if (of_get_property(codec->dev->of_node, + "cirrus,enable-soft-reset", NULL)) + cs4271->enable_soft_reset = true; } #endif @@ -492,6 +525,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) gpio_nreset = cs4271plat->gpio_nreset; amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; + cs4271->enable_soft_reset = cs4271plat->enable_soft_reset; } if (gpio_nreset >= 0) -- 1.7.11.7

3 2

[alsa-devel] [PATCH] ASoC: soc-core: Remove unused 'ret' variable
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> commit 9bde4f0b1c (ASoC: core: Fix SOC_DOUBLE_RANGE() macros) introduced the following build warning: sound/soc/soc-core.c:2999:6: warning: unused variable 'ret' [-Wunused-variable] Remove the unused 'ret' variable. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/soc-core.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e0d4630..29046e7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2996,7 +2996,6 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - int ret; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: atmel-soc: make it buildable on other architectures
by Joachim Eastwood 24 Dec '12

24 Dec '12

Not very useful on non AT91/AVR32 platforms but it provides more build coverage and prepares for ARM multiplatform. Also fixes a couple of format type warnings. Signed-off-by: Joachim Eastwood <manabian(a)gmail.com> --- Hi, Build tested on RM9200 and x86_64. ARCH_AT91 depend is pushed down to the machine drivers since they still relies on mach includes. regards Joachim Eastwood sound/soc/atmel/Kconfig | 6 +++--- sound/soc/atmel/atmel-pcm-pdc.c | 4 ++-- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 2 -- 4 files changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index d1b691b..3fdd87f 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -1,6 +1,6 @@ config SND_ATMEL_SOC tristate "SoC Audio for the Atmel System-on-Chip" - depends on ARCH_AT91 + depends on HAS_IOMEM help Say Y or M if you want to add support for codecs attached to the ATMEL SSC interface. You will also need @@ -24,7 +24,7 @@ config SND_ATMEL_SOC_SSC config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" - depends on ATMEL_SSC && SND_ATMEL_SOC && AT91_PROGRAMMABLE_CLOCKS + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && AT91_PROGRAMMABLE_CLOCKS select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_WM8731 @@ -34,7 +34,7 @@ config SND_AT91_SOC_SAM9G20_WM8731 config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" - depends on ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC + depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_TLV320AIC23 diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c index 6a293c7..054ea4d 100644 --- a/sound/soc/atmel/atmel-pcm-pdc.c +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -159,7 +159,7 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, pr_debug("atmel-pcm: " "hw_params: DMA for %s initialized " - "(dma_bytes=%u, period_size=%u)\n", + "(dma_bytes=%zu, period_size=%zu)\n", prtd->params->name, runtime->dma_bytes, prtd->period_size); @@ -201,7 +201,7 @@ static int atmel_pcm_trigger(struct snd_pcm_substream *substream, int ret = 0; pr_debug("atmel-pcm:buffer_size = %ld," - "dma_area = %p, dma_bytes = %u\n", + "dma_area = %p, dma_bytes = %zu\n", rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); switch (cmd) { diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index e99f181..3109db7 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -49,7 +49,7 @@ static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, buf->private_data = NULL; buf->area = dma_alloc_coherent(pcm->card->dev, size, &buf->addr, GFP_KERNEL); - pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%d\n", + pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n", (void *)buf->area, (void *)buf->addr, size); if (!buf->area) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 1c76634..2755750 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -42,8 +42,6 @@ #include <sound/initval.h> #include <sound/soc.h> -#include <mach/hardware.h> - #include "atmel-pcm.h" #include "atmel_ssc_dai.h" -- 1.8.0

4 4

[alsa-devel] [PATCH] ASoC: atmel-ssc: make it buildable on other architectures
by Joachim Eastwood 24 Dec '12

24 Dec '12

Not very useful on non AT91/AVR32 platforms but it provides more build coverage and prepares for ARM multiplatform. Also fixes a truncated warning that would come when building on a 64-bit arch. Signed-off-by: Joachim Eastwood <manabian(a)gmail.com> --- Hi Mark, Sending it to you since seem like you're the last one who commited changes to the driver and the driver is mostly used on ASoC. Tested on RM9200 and build tested on x86_64. regards Joachim Eastwood drivers/misc/Kconfig | 2 +- drivers/misc/atmel-ssc.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/misc/Kconfig b/drivers/misc/Kconfig index b151b7c..8f59d88 100644 --- a/drivers/misc/Kconfig +++ b/drivers/misc/Kconfig @@ -192,7 +192,7 @@ config ICS932S401 config ATMEL_SSC tristate "Device driver for Atmel SSC peripheral" - depends on AVR32 || ARCH_AT91 + depends on HAS_IOMEM ---help--- This option enables device driver support for Atmel Synchronized Serial Communication peripheral (SSC). diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c index 158da5a..0cee274 100644 --- a/drivers/misc/atmel-ssc.c +++ b/drivers/misc/atmel-ssc.c @@ -167,7 +167,7 @@ static int ssc_probe(struct platform_device *pdev) /* disable all interrupts */ clk_enable(ssc->clk); - ssc_writel(ssc->regs, IDR, ~0UL); + ssc_writel(ssc->regs, IDR, -1); ssc_readl(ssc->regs, SR); clk_disable(ssc->clk); -- 1.8.0

4 3

[alsa-devel] [PATCH v2] ASoC: lm49453: Update lm49453_reg_defs values as per LM49453 HW revision-B
by MR.Swami.Reddy＠ti.com 24 Dec '12

24 Dec '12

Update lm49453_reg_defs values as per LM49453 HW revision-B Signed-off-by: M R Swami Reddy <mr.swami.reddy(a)ti.com> --- sound/soc/codecs/lm49453.c | 61 +++++++++++--------------------------------- 1 files changed, 15 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 99b0a9d..808e50a 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -111,9 +111,9 @@ static struct reg_default lm49453_reg_defs[] = { { 101, 0x00 }, { 102, 0x00 }, { 103, 0x01 }, - { 105, 0x01 }, - { 106, 0x00 }, - { 107, 0x01 }, + { 104, 0x01 }, + { 105, 0x00 }, + { 106, 0x01 }, { 107, 0x00 }, { 108, 0x00 }, { 109, 0x00 }, @@ -163,56 +163,25 @@ static struct reg_default lm49453_reg_defs[] = { { 184, 0x00 }, { 185, 0x00 }, { 186, 0x00 }, - { 189, 0x00 }, + { 187, 0x00 }, { 188, 0x00 }, - { 194, 0x00 }, - { 195, 0x00 }, - { 196, 0x00 }, - { 197, 0x00 }, - { 200, 0x00 }, - { 201, 0x00 }, - { 202, 0x00 }, - { 203, 0x00 }, - { 204, 0x00 }, - { 205, 0x00 }, - { 208, 0x00 }, + { 189, 0x00 }, + { 208, 0x06 }, { 209, 0x00 }, - { 210, 0x00 }, - { 211, 0x00 }, - { 213, 0x00 }, - { 214, 0x00 }, - { 215, 0x00 }, - { 216, 0x00 }, - { 217, 0x00 }, - { 218, 0x00 }, - { 219, 0x00 }, + { 210, 0x08 }, + { 211, 0x54 }, + { 212, 0x14 }, + { 213, 0x0d }, + { 214, 0x0d }, + { 215, 0x14 }, + { 216, 0x60 }, { 221, 0x00 }, { 222, 0x00 }, + { 223, 0x00 }, { 224, 0x00 }, - { 225, 0x00 }, - { 226, 0x00 }, - { 227, 0x00 }, - { 228, 0x00 }, - { 229, 0x00 }, - { 230, 0x13 }, - { 231, 0x00 }, - { 232, 0x80 }, - { 233, 0x0C }, - { 234, 0xDD }, - { 235, 0x00 }, - { 236, 0x04 }, - { 237, 0x00 }, - { 238, 0x00 }, - { 239, 0x00 }, - { 240, 0x00 }, - { 241, 0x00 }, - { 242, 0x00 }, - { 243, 0x00 }, - { 244, 0x00 }, - { 245, 0x00 }, { 248, 0x00 }, { 249, 0x00 }, - { 254, 0x00 }, + { 250, 0x00 }, { 255, 0x00 }, };

2 1

[alsa-devel] [PATCH v2] ASoC: lm49453: Fix adc, mic and sidetone volume ranges
by MR.Swami.Reddy＠ti.com 24 Dec '12

24 Dec '12

Add adc, mic, sidetone volume ranges and appropriately added the controls. Fix the DAC HP/EP/LS/LO/HA maximum gain values. Signed-off-by: MR Swami Reddy <mr.swami.reddy(a)ti.com> -- sound/soc/codecs/lm49453.c | 43 ++++++++++++++++++++++++------------------- 1 files changed, 24 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 808e50a..379fcdf 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -494,36 +494,41 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), }; /* TLV Declarations */ -static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); -static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_dac_tlv, -7650, 150, 1); +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 200, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0); +static const DECLARE_TLV_DB_SCALE(stn_tlv, -7200, 150, 0); static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { /* Sidetone supports mono only */ SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), }; static const struct snd_kcontrol_new lm49453_snd_controls[] = { /* mic1 and mic2 supports mono only */ - SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, - 0, digital_tlv), - SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, - 0, digital_tlv), + SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_MICL_REG, 0, 15, 0, mic_tlv), + SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_MICR_REG, 0, 15, 0, mic_tlv), + + SOC_SINGLE_TLV("ADCL Volume", LM49453_P0_ADC_LEVELL_REG, 0, 63, + 0, adc_dac_tlv), + SOC_SINGLE_TLV("ADCR Volume", LM49453_P0_ADC_LEVELR_REG, 0, 63, + 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, - LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC1_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG, - LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC2_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum), SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum), @@ -538,16 +543,16 @@ static const struct snd_kcontrol_new lm49453_snd_controls[] = { 2, 1, 0), SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG, - LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_HP_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG, - LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_LO_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG, - LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_LS_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG, - LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_HA_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG, - 0, 6, 0, digital_tlv), + 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, 0, 3, 0, port_tlv),

4 4

[alsa-devel] [PATCH] ASoC: wm8962: Add device tree support
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Add device tree support. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Documentation/devicetree/bindings/sound/wm8962.txt | 16 ++++++++++++++++ sound/soc/codecs/wm8962.c | 7 +++++++ 2 files changed, 23 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8962.txt diff --git a/Documentation/devicetree/bindings/sound/wm8962.txt b/Documentation/devicetree/bindings/sound/wm8962.txt new file mode 100644 index 0000000..dceb3b1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8962.txt @@ -0,0 +1,16 @@ +WM8962 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "wlf,wm8962" + + - reg : the I2C address of the device. + +Example: + +codec: wm8962@1a { + compatible = "wlf,wm8962"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8fd38cb..1ebeff4 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3758,10 +3758,17 @@ static const struct i2c_device_id wm8962_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8962_i2c_id); +static const struct of_device_id wm8962_of_match[] = { + { .compatible = "wlf,wm8962", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8962_of_match); + static struct i2c_driver wm8962_i2c_driver = { .driver = { .name = "wm8962", .owner = THIS_MODULE, + .of_match_table = wm8962_of_match, .pm = &wm8962_pm, }, .probe = wm8962_i2c_probe, -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH] ASoC: core: giving WARN when device starting from non-off bias with idle_bias_off
by Mark Brown 24 Dec '12

24 Dec '12

On Fri, Dec 21, 2012 at 06:17:12PM +0800, Chuansheng Liu wrote: > > Just found some cases that some codec drivers set the bias to _STANDBY and > set idle_bias_off to 1 during probing. > It will cause unpaired runtime_get_sync/put() issue. Also as Mark suggested, > there is no reason to start from _STANDBY bias with idle_bias_off == 1. Applied, thanks.

1 0

[alsa-devel] [PATCH RFT] ASoC: lm49453: Fix mask for setting mode bit in lm49453_set_dai_fmt()
by Axel Lin 24 Dec '12

24 Dec '12

The mode variable is either 0 or 1. To update mode setting, the mask should be BIT(0) rather than BIT(1). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi M R Swami, I don't have the datasheet. Can you check if this patch is correct? Thanks, Axel sound/soc/codecs/lm49453.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index d75257d..c0d203b 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1218,7 +1218,7 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) } snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG, - LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5), + LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(0)|BIT(5), (aif_val | mode | clk_phase)); snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift); -- 1.7.9.5

3 2

[alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix maximum value for microphone gain
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3. >From Eric Nelson: "We also found that for the microphones we have here (commodity PC boom mics) a default value of 2 for the gain gives the best results." So change the default microphone gain as well. Signed-off-by: Eric Nelson <eric.nelson(a)boundarydevices.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Add sgtl5000 into the Subject sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index cb1675c..92bbfec 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 5, 1, 0), SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, - 0, 4, 0, mic_gain_tlv), + 0, 3, 0, mic_gain_tlv), }; /* mute the codec used by alsa core */ @@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN); - snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2); /* * disable DAP -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: Fix maximum value for microphone gain
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3. >From Eric Nelson: "We also found that for the microphones we have here (commodity PC boom mics) a default value of 2 for the gain gives the best results." So change the default microphone gain as well. Signed-off-by: Eric Nelson <eric.nelson(a)boundarydevices.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index cb1675c..92bbfec 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 5, 1, 0), SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, - 0, 4, 0, mic_gain_tlv), + 0, 3, 0, mic_gain_tlv), }; /* mute the codec used by alsa core */ @@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN); - snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2); /* * disable DAP -- 1.7.9.5

2 1

[alsa-devel] Does it will effect the precision of DMA position?
by Zhang wei 24 Dec '12

24 Dec '12

Hello guys. In the function snd_pcm_update_hw_ptr0(): pos -= pos % runtime->min_align; after running it, the 'pos' maybe smaller than before.because there is need to aligned.so that means lost or drop some Audio data for ever ,is that true? The data playbacked is a bit of different from we send to? Best Regards zw

1 0

[alsa-devel] [alsa-lib, dmix] Support for S24_3BE
by Vincent Vinel 24 Dec '12

24 Dec '12

Hello, The bugtracker seems to be down, so I'm posting here instead. I was trying to use software mixing with my Native Instruments USB soundcard (snd-usb-caiaq), but all attemps to use the dmix pcm ended up with the rather unhelpful error : ALSA lib pcm_dmix.c:1311:(_snd_pcm_dmix_open) Unsupported format When using the hw pcm, however, aplay was kind enough to display the list of available formats, which in my case was limited to S24_3BE. A quick check confirmed that dmix doesn't support that format, which happens to be the only format available for NI USB soundcards (as defined in snd-usb-caiaq). Is it worth writing a patch for 24bit big-endian support in dmix ? External soundcards like these ones are usually used for realtime effects processing and low-latency audio, things for which dmix proably isn't suited anyway. -- Vincent Vinel

2 1

Re: [alsa-devel] [PATCH V5 2/2] ASoC: SAMSUNG: Add DT support for i2s
by Grant Likely 24 Dec '12

24 Dec '12

On Thu, 13 Dec 2012 16:12:53 +0530, Padmavathi Venna <padma.v(a)samsung.com> wrote: > Add support for device based discovery. > > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> > --- > .../devicetree/bindings/sound/samsung-i2s.txt | 75 +++++++ > sound/soc/samsung/dma.c | 1 + > sound/soc/samsung/dma.h | 1 + > sound/soc/samsung/i2s.c | 233 ++++++++++++++++---- > 4 files changed, 269 insertions(+), 41 deletions(-) > create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt > > diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt > new file mode 100644 > index 0000000..bff47b4 > --- /dev/null > +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt > @@ -0,0 +1,75 @@ > +* Samsung I2S controller > + > +Required SoC Specific Properties: > + > +- compatible : "samsung,samsung-i2s" Isn't that kind of redundant? :-) The format of the compatible strings should be "<vendor>,<part-number>-i2s". Please be specific about the part number that you're doing the binding for. For example; use "samsung,exynos4210-i2s" instead of "samsung,exynos-i2s". > +- reg: physical base address of the controller and length of memory mapped > + region. > + > +[PRELIMINARY: the dma channel allocation will change once there are > +official DMA bindings] > + > +- tx-dma-channel-secondary: The dma channel specifier for secondary tx > + operations. The format of the dma specifier depends on the dma > + controller. > +- tx-dma-channel: The dma channel specifier for tx operations. The format of > + the dma specifier depends on the dma controller. > +- rx-dma-channel: The dma channel specifier for rx operations. The format of > + the dma specifier depends on the dma controller. > + > +Optional SoC Specific Properties: > + > +- samsung,supports-6ch: If the I2S Primary sound source has 5.1 Channel > + support, this flag is enabled. > +- samsung,supports-rstclr: This flag should be set if I2S software reset bit > + control is required. When this flag is set I2S software reset bit will be > + enabled or disabled based on need. > +- samsung,supports-secdai:If I2S block has a secondary FIFO and internal DMA, > + then this flag is enabled. > +- samsung,idma-addr: Internal DMA register base address of the audio > + sub system(used in secondary sound source). > + > +Required Board Specific Properties: > + > +- gpios: The gpio specifier for data out,data in, LRCLK, CDCLK and SCLK > + interface lines. The format of the gpio specifier depends on the gpio > + controller. > + The syntax of samsung gpio specifier is > + <[phandle of the gpio controller node] > + [pin number within the gpio controller] > + [mux function] > + [flags and pull up/down] > + [drive strength]> > + > +Aliases: > + > +- All the I2S controller nodes are represented in the aliases node using > + the following format 'i2s{n}' where n is a unique number for the alias. Are the aliases actually being used now? If not, then drop this section. I don't see any references in this patch. Otherwise the binding looks good to me. g.

4 3

[alsa-devel] sound: usb-audio: full capture/playback/spdif support for Digidesign Mbox 2
by Damien Zammit 23 Dec '12

23 Dec '12

Hi folks, I have put together full support for Digidesign Mbox 2, but it needs someone to help me merge it into alsa. I haven't seen the latest snd-usb-audio code since last year, it seems to have branched into multiple files. I have worked out how the Digidesign Mbox 2 needs to be configured, it supports SPDIF i/o simultaneously with analogue but can be toggled on and off by using different altsettings. Also the sample clock rate must be configured via a special method. Therefore I have provided for this in a device_setup parameter with 4 settings: 48Khz Spdif+analog = 0 44.1Khz Spdif+analog = 1 48Khz Analog only = 2 44.1Khz Analog only = 3 ###### audio /* Digidesign Mbox 2 workaround: * supports S24_3BE BIG ENDIAN in an unusual way */ if (chip->usb_id == USB_ID(0x0dba, 0x3000)) pcm_format = SNDRV_PCM_FORMAT_S24_3BE; static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno); static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) { ..... ..... /* Digidesign Mbox 2: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0dba, 0x3000) && mbox2_skip_setting_quirk(chip, iface_no, altno)) continue; ..... } #define MBOX2_SET_48K_SPDIF 0 #define MBOX2_SET_44K_SPDIF 1 #define MBOX2_SET_48K_ANALOG 2 #define MBOX2_SET_44K_ANALOG 3 static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno) { u8 srate[3]; /* Reset ifaces 2-5 to 0 altsetting. */ usb_set_interface(chip->dev, iface, 0); if (device_setup[chip->index] == MBOX2_SET_48K_SPDIF || device_setup[chip->index] == MBOX2_SET_48K_ANALOG) { //setup 48k srate[0]=0x80; srate[1]=0xbb; srate[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x01, 0x22, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003, 1000); snd_printk(KERN_INFO "Digidesign Mbox 2: 24 Bit 48000Hz"); if (device_setup[chip->index] == MBOX2_SET_48K_SPDIF) { //setup spdif snd_printk(KERN_INFO ", SPDIF\n"); if (altno != 2) return 1; } else { //setup analog snd_printk(KERN_INFO ", ANALOGUE\n"); if (altno != 3) return 1; } } else { //setup 44k srate[0]=0x44; srate[1]=0xac; srate[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x01, 0x22, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003, 1000); snd_printk(KERN_INFO "Digidesign Mbox 2: 24 Bit 44100Hz"); if (device_setup[chip->index] == MBOX2_SET_44K_SPDIF || device_setup[chip->index] == MBOX2_SET_44K_ANALOG) { //setup spdif snd_printk(KERN_INFO ", SPDIF\n"); if (altno != 2) return 1; } else { //setup analog snd_printk(KERN_INFO ", ANALOGUE\n"); if (altno != 3) return 1; } } return 0; } #define MBOX2_FIRMWARE_SIZE 646 #define MBOX2_BOOT_LOADING 0x01 /* Hard coded into the device */ #define MBOX2_BOOT_READY 0x02 /* Hard coded into the device */ static int snd_usb_mbox2_boot_quirk(struct usb_device *dev) { struct usb_host_config *config = dev->actconfig; int err; u8 enablemagic[3]; u8 bootresponse; int fwsize; if ((fwsize = le16_to_cpu(get_cfg_desc(config)->wTotalLength)) == MBOX2_FIRMWARE_SIZE) { snd_printdd("Sending Digidesign Mbox 2 boot sequence...\n"); /* From USB Snoop, * * SetupPacket = RT RQ VHVL INDX SIZE * RT = 0xRT Request Type * RQ = 0xRQ Request * VHVL = 0xVLVH Value in reverse byte order * INDX = 0xDXIN Index in reverse byte order * SIZE = 0xZESI Size in reverse byte order * * Magic boot code setup packet: c0 85 01 00 00 00 12 00 * RQ RT VLVH DXIN ZESI * becomes 0x85, 0xc0, 0x0001, 0x0000, &RETURNDATA, 0x0012, TIMEOUT * for snd_usb_ctl_msg() */ mbox2_reboot: snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012, 1000); if (bootresponse == MBOX2_BOOT_LOADING) { snd_printdd("device not ready, resending boot sequence...\n"); goto mbox2_reboot; } if (bootresponse == MBOX2_BOOT_READY) { snd_printdd("device initialised!\n"); err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, &dev->descriptor, sizeof(dev->descriptor)); config = dev->actconfig; if (err < 0) snd_printdd("error usb_get_descriptor: %d\n", err); err = usb_reset_configuration(dev); if (err < 0) snd_printdd("error usb_reset_configuration: %d\n", err); snd_printdd("mbox2_boot: new boot length = %d\n", le16_to_cpu(get_cfg_desc(config)->wTotalLength)); /* Successful boot, (should reset all altsettings to 0 like windows does, but works without it) * but now we will try sending capture/playback enable magic * * 80 bb 00 = 24bit mode - S24_3BE * 44 ac 00 = 16bit mode? * */ /* We want 48khz mode for now */ enablemagic[0]=0x80; enablemagic[1]=0xbb; enablemagic[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x01, 0x22, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &enablemagic, 0x0003, 1000); return 0; /* Succesful boot */ } snd_printdd("unknown bootresponse, ignoring device: %d\n",bootresponse); return -ENODEV; } snd_printk("Invalid firmware size: %d\n",fwsize); return -ENODEV; } ####### quirk: { USB_DEVICE(0x0dba, 0x3000), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Digidesign", .product_name = "Mbox 2", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 1, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 2, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 3, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 4, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 5, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 6, .type = QUIRK_MIDI_MBOX2, .data = & (const struct snd_usb_midi_endpoint_info) { .out_ep = 0x02, .out_cables = 0x0001, .in_ep = 0x81, .in_interval = 0x01, .in_cables = 0x0001 } }, { .ifnum = -1 } } } }, ##### midi case QUIRK_MIDI_MBOX2: /* Digidesign Mbox 2 uses MIDIMAN MIDI protocol */ umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops; /* * We have to make sure that the USB core looks * again at interface 6 by calling usb_set_interface() on it. */ usb_set_interface(umidi->chip->dev, 6, 0); memcpy(&endpoints[0], quirk->data, sizeof(struct snd_usb_midi_endpoint_info)); err = 0; break; Damien

4 26

[alsa-devel] [PATCH] ALSA: usb-audio: support delay calculation on capture streams
by Pierre-Louis Bossart 21 Dec '12

21 Dec '12

Enable delay report on capture path. The delay is reset when an URB is retired and increment at each call to .pointer based on frame counter changes. The precision of the delay information is limited to 1ms as in the playback case. This reverts commit 3f94fad09538ec988919ec3f371841182df71d04. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/pcm.c | 23 ++++++++++++++++++++--- 1 file changed, 20 insertions(+), 3 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..3a38447 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -59,7 +59,12 @@ snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, /* Approximation based on number of samples per USB frame (ms), some truncation for 44.1 but the estimate is good enough */ - est_delay = subs->last_delay - (frame_diff * rate / 1000); + est_delay = frame_diff * rate / 1000; + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) + est_delay = subs->last_delay - est_delay; + else + est_delay = subs->last_delay + est_delay; + if (est_delay < 0) est_delay = 0; return est_delay; @@ -78,8 +83,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->delay = snd_usb_pcm_delay(subs, substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); @@ -1147,6 +1151,10 @@ static void retire_capture_urb(struct snd_usb_substream *subs, int i, period_elapsed = 0; unsigned long flags; unsigned char *cp; + int current_frame_number; + + /* read frame number here, update pointer in critical section */ + current_frame_number = usb_get_current_frame_number(subs->dev); stride = runtime->frame_bits >> 3; @@ -1180,6 +1188,15 @@ static void retire_capture_urb(struct snd_usb_substream *subs, subs->transfer_done -= runtime->period_size; period_elapsed = 1; } + /* capture delay is by construction limited to one URB, + * reset delays here + */ + runtime->delay = subs->last_delay = 0; + + /* realign last_frame_number */ + subs->last_frame_number = current_frame_number; + subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ + spin_unlock_irqrestore(&subs->lock, flags); /* copy a data chunk */ if (oldptr + bytes > runtime->buffer_size * stride) { -- 1.7.9.5

2 1

[alsa-devel] [PATCH RFT] ASoC: wm2200: Fix setting dai format in wm2200_set_fmt
by Axel Lin 21 Dec '12

21 Dec '12

According to the defines in wm2200.h: /* * R1284 (0x504) - Audio IF 1_5 */ #define WM2200_AIF1_FMT_MASK 0x0007 /* AIF1_FMT - [2:0] */ #define WM2200_AIF1_FMT_SHIFT 0 /* AIF1_FMT - [2:0] */ #define WM2200_AIF1_FMT_WIDTH 3 /* AIF1_FMT - [2:0] */ We should not left shift 1 bit for fmt_val when setting dai format. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Mark, wm2200.h defines WM2200_AIF1_FMT_SHIFT to be 0, so current code looks wrong to me. I don't have the datasheet, so can you check if this patch is correct? Axel sound/soc/codecs/wm2200.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index afcf31d..a12fc2f 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1626,7 +1626,7 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV, lrclk); snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_5, - WM2200_AIF1_FMT_MASK << 1, fmt_val << 1); + WM2200_AIF1_FMT_MASK, fmt_val); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/3] ASoC: fsi: cleanup FSI driver
by Kuninori Morimoto 21 Dec '12

21 Dec '12

Hi Mark Cc Simon These patches changed platform setting to use FSI new style, and cleanup FSI driver. Kuninori Morimoto (3): ASoC: fsi: remove platform depended .set_rate() callback support ASoC: fsi: remove SH_FSI_xxx_INV flags ASoC: fsi: cleanup sh_fsi.h arch/arm/mach-shmobile/board-ap4evb.c | 11 +-- arch/arm/mach-shmobile/board-mackerel.c | 11 +-- arch/sh/boards/mach-ecovec24/setup.c | 12 +-- arch/sh/boards/mach-se/7724/setup.c | 12 +-- include/sound/sh_fsi.h | 70 +------------- sound/soc/sh/fsi.c | 156 +++---------------------------- 6 files changed, 27 insertions(+), 245 deletions(-) >> Simon #2 patch modifies both FSI driver and platform. So, it needs your Acked-by Best regards --- Kuninori Morimoto

1 5

[alsa-devel] [PATCHv2 0/9] ALSA: CA0132: DSP Update
by Ian Minett 21 Dec '12

21 Dec '12

From: Ian Minett <ian_minett(a)creativelabs.com> Thanks for the comments. This resubmit should help to correct many of the points raised. For example, use of C99-style init for structures, the previous large patch is now split into several subpatches, and there are more descriptions to the code and individual patchfiles. Patch Summary: 1: Add new definitions and structures that will be used by the DSP routines 2: Init chip registers, DSP parameters and mixer settings 3: Add the new DSP mixer kcontrols and helpers. 4: Update and add PCM callbacks and enhancements. 5: Add unsol handler for DSP response and jack detection reporting 6: Add tuning controls 7/8: Code shuffle and tidy, split into two for a cleaner diff. 9: Updates to the mic sample rate and module description, and fix potential errors on init. Thanks, - Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com>

1 9

Re: [alsa-devel] [PATCH 5/7] ASoC: tegra: add ac97 host driver
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > This adds the driver for the Tegra 2x AC97 host controller. The AC'97 DT binding file should really be added as part of this patch. I'm not at all familiar with AC'97, but this mostly looks fine. I have a bunch of comments below though, mostly for my own understanding. > diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig > +config SND_SOC_TEGRA20_AC97 > + tristate > + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC This should also select SND_SOC_TEGRA20_DAS, just like the I2S driver. > diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c > +static struct tegra20_ac97 *workdata; Why not put that into the device's drvdata? Oh, is there no way to stash driver-private data into the struct snd_ac97? > +static void tegra20_ac97_codec_reset(struct snd_ac97 *ac97) > +{ > + u32 readback; > + unsigned long timeout; > + > + /* reset line is not driven by DAC pad group, have to toggle GPIO */ > + gpio_set_value(workdata->reset_gpio, 0); > + udelay(2); I'm not quite sure what that implies. Might some (Tegra) HW designs use a dedicated signal from the AC'97 HW to reset the CODEC and others not? That would make the GPIO optional. Or, does the comment mean that Tegra doesn't ever have a dedicated CODEC reset signal, so we always have to use a GPIO? If so, this comment might be more appropriate inside probe() where I asked my related questions. > +static void tegra20_ac97_codec_warm_reset(struct snd_ac97 *ac97) > +{ > + u32 readback; > + unsigned long timeout; > + > + /* > + * although sync line is driven by the DAC pad group warm reset using > + * the controller cmd is not working, have to toggle sync line > + * manually. > + */ > + gpio_request(workdata->sync_gpio, "codec-sync"); Hmm. There's an AC'97 command to reset the CODEC and we don't implement it? Uggh. > +static const struct snd_soc_dai_ops tegra20_ac97_dai_ops = { > + .trigger = tegra20_ac97_trigger, > +}; No .set_fmt or .hw_params? Does the AC'97 ASoC core code handle that somehow? I guess that's what soc_ac97_ops is for? > +static struct snd_soc_dai_driver tegra20_ac97_dai = { > + .name = "tegra-ac97-pcm", > + .ac97_control = 1, > + .probe = tegra20_ac97_probe, > + .playback = { > + .stream_name = "PCM Playback", Out of curiosity, why "PCM Playback" not just "Playback"? > +static bool tegra20_ac97_volatile_reg(struct device *dev, unsigned int reg) > +{ > + switch (reg) { > + case TEGRA20_AC97_CMD: ... > + return true; CMD is volatile; did we put status bits in there? > +static int tegra20_ac97_platform_probe(struct platform_device *pdev) > + ac97->reset_gpio = of_get_named_gpio(pdev->dev.of_node, > + "nvidia,codec-reset-gpio", 0); > + if (gpio_is_valid(ac97->reset_gpio)) { > + ret = devm_gpio_request_one(&pdev->dev, ac97->reset_gpio, > + GPIOF_OUT_INIT_HIGH, "codec-reset"); Shouldn't this get the flags from the GPIO specifier, and deal with active-high/active-low resets, or is the polarity set by the AC'97 spec? > + if (ret) { > + dev_err(&pdev->dev, "could not get codec-reset GPIO\n"); > + goto err_clk_put; > + } > + } else { > + dev_err(&pdev->dev, "no codec-reset GPIO supplied\n"); > + goto err_clk_put; > + } Is a reset GPIO necessarily required? What if a board arranged to reset the CODEC during power-on-reset, and there was no software control over the reset? > + ac97->sync_gpio = of_get_named_gpio(pdev->dev.of_node, > + "nvidia,codec-sync-gpio", 0); > + if (!gpio_is_valid(ac97->sync_gpio)) { > + dev_err(&pdev->dev, "no codec-sync GPIO supplied\n"); > + goto err_clk_put; > + } I don't know what this is, so I'll ask if it's strictly required too. > + ret = snd_soc_register_dais(&pdev->dev, &tegra20_ac97_dai, 1); snd_soc_register_dai() would be marginally simpler. > + ret = clk_prepare_enable(ac97->clk_ac97); > + if (ret) { > + dev_err(&pdev->dev, "clk_enable failed: %d\n", ret); > + goto err_asoc_utils_fini; > + } Can you do runtime PM instead? See the I2S driver for an example. > +MODULE_LICENSE("GPL"); "GPL v2"

2 1

Re: [alsa-devel] [PATCH 5/7] ASoC: tegra: add ac97 host driver
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/20/2012 01:10 PM, Lucas Stach wrote: > Am Donnerstag, den 20.12.2012, 12:44 -0700 schrieb Stephen Warren: >> On 12/19/2012 04:17 PM, Lucas Stach wrote: >>> This adds the driver for the Tegra 2x AC97 host controller. >>> +static void tegra20_ac97_codec_warm_reset(struct snd_ac97 *ac97) >>> +{ >>> + u32 readback; >>> + unsigned long timeout; >>> + >>> + /* >>> + * although sync line is driven by the DAC pad group warm reset using >>> + * the controller cmd is not working, have to toggle sync line >>> + * manually. >>> + */ >>> + gpio_request(workdata->sync_gpio, "codec-sync"); >> >> Hmm. There's an AC'97 command to reset the CODEC and we don't implement >> it? Uggh. > > As far as I could figure from the downstream register doc there is a > command implemented in the host controller that's supposed to do the > warm reset, but in my testing it didn't work. Sadly I have not received > any documentation about the host controller from NVIDIA up until now, so > I don't now if we could make this work somehow. Bug Submission ID#: > 196075 What system was that bug submitted into? It's not a regular NVIDIA internal bug number. I'll see who I can ask about this feature... >>> + ac97->sync_gpio = of_get_named_gpio(pdev->dev.of_node, >>> + "nvidia,codec-sync-gpio", 0); >>> + if (!gpio_is_valid(ac97->sync_gpio)) { >>> + dev_err(&pdev->dev, "no codec-sync GPIO supplied\n"); >>> + goto err_clk_put; >>> + } >> >> I don't know what this is, so I'll ask if it's strictly required too. > > Until we manage to get the host controller warm reset command working > this is fixed to be the _FS line of the DAP connected to the codec, so > strictly required. I'd like to investigate whether the warm reset command can be made to work before actually committing to a DT binding that requires this GPIO to be specified; the DT binding is supposed to be basically static.

1 0

Re: [alsa-devel] [PATCH 7/7] ASoC: tegra: add machine driver using wm9712 codec
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > This adds a simple machine driver using the Wolfson wm9712 AC97 > codec. If the AC'97 code is going to be reworked to probe the CODEC rather than manually specify it in DT, then will this machine driver become generic for any AC'97 system, and hence you could rename the DT binding to nvidia,tegra-audio-ac97? > diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt > +- nvidia,audio-codec : The phandle of the WM9712 audio codec I wonder how you'll get that if you dynamically probe the AC'97 bus... > diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig > +config SND_SOC_TEGRA_WM9712 > + tristate "SoC Audio support for Tegra boards using a WM9712 codec" > + depends on SND_SOC_TEGRA I'd suggest making this depend on the Tegra AC97 module, or ARCH_TEGRA_2x_SOC instead, since it can only run on Tegra20. The AC'97 module has been removed on Tegra30 and replaced with an HD Audio controller instead. > + select SND_SOC_TEGRA20_AC97 if ARCH_TEGRA_2x_SOC That select (or a dependency as mentioned above) is always required. > + select SND_SOC_AC97_BUS Shouldn'y the AC'97 driver select that? > diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c > +static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd) > + snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); Hmm. I guess that potentially ties this driver to whichever CODECs actually expose a widget with that exact name... > +static int tegra_wm9712_remove(struct snd_soc_card *card) > +{ > + return 0; > +} Do you need to implement this at all if it does nothing? > +MODULE_LICENSE("GPL"); "GPL v2"

2 1

Re: [alsa-devel] [PATCH 6/7] ASoC: tegra: add ac97 host controller to device tree
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > This adds the DT bindings and a default entry for the Tegra 2x > AC97 host controller. Everything here except the .dtsi file should be part of the previous patch. > diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt > +- nvidia,codec-sync-gpio : The Tegra GPIO controller's phandle and the number > + of the GPIO corresponding with the AC97 DAP _FS line Hmm, for I2S at least, FS (frame sync I assume) definitely is something that the audio controller itself drives. Are you absolutely sure that this should be represented as a GPIO here? > diff --git a/arch/arm/boot/dts/tegra20.dtsi b/arch/arm/boot/dts/tegra20.dtsi > + tegra_ac97: ac97@70002000 { There's only 1 node named ac97, so you can leave out the unit address since it's not required to generate a unique node name. In other words just "ac97".

1 0

Re: [alsa-devel] [PATCH 1/7] ARM: tegra: add ac97 clock
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > Add clock used by the AC97 host controller. This is going to conflict with Prashant's clock driver rework. Can you rebase it on top of that once it's checked in. That might be a couple weeks out though.

1 0

Re: [alsa-devel] [PATCH 4/7] ASoC: tegra: add function to set ac97 rate
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > AC97 uses a fixed rate, unrelated to the sample rate. Add a function to > make the setup more trivial. Acked-by: Stephen Warren <swarren(a)nvidia.com>

1 0

Re: [alsa-devel] [PATCH 3/7] ASoC: tegra: setup DAP3<->DAC3 connection by default
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > This connection is used by the AC97 controller. Acked-by: Stephen Warren <swarren(a)nvidia.com>

1 0

[alsa-devel] Fwd: Low volume with Realtek ALC 883.
by prasannatsmkumar 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 11:07 PM, prasannatsmkumar <prasannatsmkumar(a)gmail.com> wrote: > Hi ALSA people, > > I am having a Realtek ALC 883 chip. The sound volume is very low > compared to what I hear in windows - I maintain same speaker volume > and play same audio. If I keep the volume as 10% in windows I have to > keep a volume of at least 25+% in Linux. Is there any way to improve > it? I have been seeing such a situation for past 4 or 5 years (from > Fedora 9 period till today). I have tried several distributions > (Fedora, Vector Linux, PC Linux OS, Ubuntu, Mint) and all have the > same behaviour. Currently I am running Ubuntu 11.10 with 3.5 kernel. > > $cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.25. > > Should I file a bug? If so in which component (kernel or pulseaudio)? > > Thanks in advance, > PrasannaKumar Resending this as this mail was bounced by alsa devel list. Please feel free to ask for information necessary to solve this issue. I have copied output of some commands that may be useful for understanding or digging this issue. dmesg | grep hda output: [ 30.648663] snd_hda_intel 0000:00:1b.0: >irq 43 for MSI/MSI-X [ 30.716741] hda_codec: ALC883: SKU not ready 0x411111f0 lspci -vvv output (only info related audio device is in mail all others are stripped): 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 01) Subsystem: Intel Corporation Device d605 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 43 Region 0: Memory at 501c0000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0300c Data: 4189 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed- WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=01 Status: NegoPending- InProgress- VC1: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=1 ArbSelect=Fixed TC/VC=80 Status: NegoPending- InProgress- Capabilities: [130 v1] Root Complex Link Desc: PortNumber=0f ComponentID=02 EltType=Config Link0: Desc: TargetPort=00 TargetComponent=02 AssocRCRB- LinkType=MemMapped LinkValid+ Addr: 00000000fed1c000 Kernel driver in use: snd_hda_intel Kernel modules: snd-hda-intel amixer output: Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 13 [42%] [-27.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [off] Front Right: Playback 31 [100%] [12.00dB] [off] Simple mixer control 'Front Mic Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 3 [100%] [30.00dB] Front Right: 3 [100%] [30.00dB] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [off] Front Right: Playback 31 [100%] [12.00dB] [off] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 8 [26%] [0.00dB] [on] Front Right: Capture 8 [26%] [0.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-12.00dB] [off] Front Right: Capture 0 [0%] [-12.00dB] [off] Simple mixer control '2ch-Upmix Master',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 255 Front Left: 255 [100%] Front Right: 255 [100%] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '4ch' '6ch' Item0: '2ch' Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Rear Mic' 'Front Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Rear Mic' 'Front Mic' 'Line' Item0: 'Rear Mic' Simple mixer control 'Rear Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Rear Mic Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 2 [67%] [20.00dB] Front Right: 2 [67%] [20.00dB] Thanks and Regards, PrasannaKumar

1 1

[alsa-devel] Fwd: Low volume with Realtek ALC 883.
by prasannatsmkumar 20 Dec '12

20 Dec '12

Hi ALSA / pulseaudio people, Resending this as this mail was bounced by alsa devel list. I am having a Realtek ALC 883 chip. The sound volume is very low compared to what I hear in windows - I maintain same speaker volume and play same audio. If I keep the volume as 10% in windows I have to keep a volume of at least 25+% in Linux. Is there any way to improve it? I have been seeing such a situation for past 4 or 5 years (from Fedora 9 period till today). I have tried several distributions (Fedora, Vector Linux, PC Linux OS, Ubuntu, Mint) and all have the same behaviour. Currently I am running Ubuntu 11.10 with 3.5 kernel. $cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.25. Should I file a bug? If so in which component (kernel or pulseaudio)? Thanks in advance, PrasannaKumar

1 0

[alsa-devel] [PATCH RFT] ASoC: sta529: Fix update register bits in sta529_set_dai_fmt
by Axel Lin 20 Dec '12

20 Dec '12

Both the mask and mode settings are wrong in current code. According to the datasheet: S2PCFG0 (0x0A) BIT[3:1] DATA_FORMAT serial interface protocol format: 000: left Justified 001: I2S (default) 010: right justified 100: PCM no delay 101: PCM delay 111: DSP Thus fixes the defines for LEFT_J_DATA_FORMAT, I2S_DATA_FORMAT, and RIGHT_J_DATA_FORMAT. Also adds define for DATA_FORMAT_MSK. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Rajeev, I don't have this hardware, can you test this patch? Thanks, Axel sound/soc/codecs/sta529.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index ab355c4..40c07be 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -74,9 +74,10 @@ SNDRV_PCM_FMTBIT_S32_LE) #define S2PC_VALUE 0x98 #define CLOCK_OUT 0x60 -#define LEFT_J_DATA_FORMAT 0x10 -#define I2S_DATA_FORMAT 0x12 -#define RIGHT_J_DATA_FORMAT 0x14 +#define DATA_FORMAT_MSK 0x0E +#define LEFT_J_DATA_FORMAT 0x00 +#define I2S_DATA_FORMAT 0x02 +#define RIGHT_J_DATA_FORMAT 0x04 #define CODEC_MUTE_VAL 0x80 #define POWER_CNTLMSAK 0x40 @@ -289,7 +290,7 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return -EINVAL; } - snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode); + snd_soc_update_bits(codec, STA529_S2PCFG0, DATA_FORMAT_MSK, mode); return 0; } -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: pcm: allow backend hardware to be freed in pause state
by Patrick Lai 20 Dec '12

20 Dec '12

When front-end PCM session is in paused state, back-end PCM session will be put in paused state as well if given front-end PCM session is the only client of given back-end. Then, application closes front-end PCM session, DPCM framework will not allow back-end enters HW_FREE state so back-end will never get shutdown completely. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ef22d0b..d340644 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1240,6 +1240,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; -- 1.7.8.3 Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

3 2

[alsa-devel] [PATCH RFT] ASoC: arizona: Do proper shift for setting aif1 rate
by Axel Lin 20 Dec '12

20 Dec '12

ARIZONA_AIF1_RATE_MASK is 0x7800 /* AIF1_RATE - [14:11] */ Thus we need left shift ARIZONA_AIF1_RATE_SHIFT when setting aif1 rate. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/arizona.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index adf397b..38248a7 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -714,7 +714,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, 8); + ARIZONA_AIF1_RATE_MASK, + 8 << ARIZONA_AIF1_RATE_SHIFT); break; default: arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH] ASoC: dapm: Fix the unpaired runtime_get/put cases
by Mark Brown 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 06:37:26AM +0000, Liu, Chuansheng wrote: > Meanwhile, is it useful to add one warning there for that case? > After all, in probing, set the bias to _STANDBY even idle_bias_off == 1, and calling get_runtime_sync(), it > will let the code more obscure. So giving a warning there to indicate the driver: > it is not suggested that in probing, set the bias to _STANDBY even idle_bias_off == 1. Probably, send a patch please. Like I say it is possible to start off in _STANDBY providing the driver grabs the runtime PM reference too but I can't think of any reason for doing that so the warning seems sensible.

1 0

Re: [alsa-devel] [PATCH 2/7] ASoC: allow wolfson wm9712 codec to be instantiated using device tree
by Mark Brown 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 12:17:31AM +0100, Lucas Stach wrote: > static struct platform_driver wm9712_codec_driver = { > .driver = { > .name = "wm9712-codec", > .owner = THIS_MODULE, > + .of_match_table = wm9712_of_match, > }, No, this is really not a sane way of handling AC'97 at all. They're not platform devices, there is no sane reason why they should appear in the device tree as such. Really we ought to be just hotplugging them, AC'97 is perfectly enumerable.

1 0

[alsa-devel] Adding wm8962 support to mx6qsabresd
by Fabio Estevam 20 Dec '12

20 Dec '12

Hi, I am working on adding audio support to mx6qsabresd board, which contains a wm8962 codec. Here is the log with debug information turned on that I am getting so far: wm8962 0-001a: customer id 0 revision D wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 3a => 0 wm8962 0-001a: read 3a => 0 wm8962 0-001a: read 39 => 0 wm8962 0-001a: read 39 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 200 => ffffffff wm8962 0-001a: read 201 => 0 wm8962 0-001a: read 202 => 0 wm8962 0-001a: read 203 => ffffffff wm8962 0-001a: read 204 => 8100 wm8962 0-001a: read 205 => 8100 wm8962 0-001a: DMIC not in use, disabling input: WM8962 Beep Generator as /devices/soc.0/2100000.aips-bus/21a0000.i2c/i2c-0/0-001a/input/input0 wm8962 0-001a: read 31 => ffffffff wm8962 0-001a: read 31 => ffffffff wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 30 => ffffffff wm8962 0-001a: read 30 => ffffffff wm8962 0-001a: read 300 => 1c00 wm8962 0-001a: read 17 => 160 wm8962 0-001a: read 48 => 0 wm8962 0-001a: read 8 => ffffffff wm8962 0-001a: read 52 => 4 wm8962 0-001a: read 19 => 0 wm8962 0-001a: write a0 = 149 wm8962 0-001a: write a1 = 271 wm8962 0-001a: write a2 = 7 wm8962 0-001a: FLL configured for 24000000Hz->11289600Hz wm8962 0-001a: No audio clocks configured ... imx-wm8962 sound.14: wm8962 <-> 2028000.ssi mapping ok root@freescale /home$ ./aplay Front_Left.wav wm8962 0-001a: hw_params set BCLK 1411200Hz LRCLK 44100Hz wm8962 0-001a: wm8962->sysclk_rate = 11289600 wm8962 0-001a: wm8962->lrclk = 44100 wm8962 0-001a: Selected sysclk ratio 256 wm8962 0-001a: read 4 => ffffffff wm8962 0-001a: Failed to read DSPCLK: -1 wm8962 0-001a: read 3 => 100 wm8962 0-001a: write 3 = 100 wm8962 0-001a: ASoC: POST_PMU: HPOUTR PGA event failed: -5 wm8962 0-001a: read 29 => 100 wm8962 0-001a: write 29 = 100 wm8962 0-001a: ASoC: POST_PMU: SPKOUTR PGA event failed: -5 wm8962 0-001a: read 2 => 100 wm8962 0-001a: write 2 = 100 wm8962 0-001a: ASoC: POST_PMU: HPOUTL PGA event failed: -5 wm8962 0-001a: read 28 => 100 wm8962 0-001a: write 28 = 100 wm8962 0-001a: ASoC: POST_PMU: SPKOUTL PGA event failed: -5 wm8962 0-001a: read 42 => ffffffff wm8962 0-001a: Failed to read DCS status: -1 wm8962 0-001a: DC servo complete after 1ms Playing WAVE 'Front_Left.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono Any idea of why register 4 and 42 read fails? wm8962 0-001a: read 4 => ffffffff wm8962 0-001a: Failed to read DSPCLK: -1 wm8962 0-001a: read 42 => ffffffff wm8962 0-001a: Failed to read DCS status: -1 Any suggestion is appreciated. Thanks, Fabio Estevam

3 2

[alsa-devel] how to check the CODEC registers vaule
by Zhang wei 20 Dec '12

20 Dec '12

Hello everyone,I am confused again : ( I read the wm8985.pdf and found the description: If the address is not recognised or the R/W bit is '1' when operating in write only mode,the WM8985 returns to the idle conditon and waits for a new start condition and valid address. Does it mean the wm8985 registers write only, can not be read? I found other kinds of codec datasheet same as it. Is it necessary to read back the value from the register ? Thanks a lot. Best Regards zw

1 0

[alsa-devel] [PATCH 1/2] Revert "ALSA: usb-audio: ignore delay calculation for capture stream"
by Pierre-Louis Bossart 19 Dec '12

19 Dec '12

This reverts commit 3f94fad09538ec988919ec3f371841182df71d04. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..b09b895 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -78,8 +78,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->delay = snd_usb_pcm_delay(subs, substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ASoC: core: Fix SOC_DOUBLE_RANGE() macros
by Mark Brown 19 Dec '12

19 Dec '12

Although we've had macros defining double _RANGE controls for a while now they've not actually been backed up properly by the implementation, it's treated everything as mono. Fix that by implementing the handling in the stereo controls, ensuring that the mono controls don't mistakenly get treated as stereo. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 10 ++++++---- sound/soc/soc-core.c | 32 ++++++++++++++++++++++++++++++-- 2 files changed, 36 insertions(+), 6 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4a27822..1e4ea78 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -58,8 +58,9 @@ .info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \ .put = snd_soc_put_volsw_range, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = xshift, .min = xmin,\ - .max = xmax, .platform_max = xmax, .invert = xinvert} } + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .platform_max = xmax, .invert = xinvert} } #define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -88,8 +89,9 @@ .info = snd_soc_info_volsw_range, \ .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = xshift, .min = xmin,\ - .max = xmax, .platform_max = xmax, .invert = xinvert} } + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b1738d5..d4732ac 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2922,7 +2922,7 @@ int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, platform_max = mc->platform_max; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; uinfo->value.integer.max = platform_max - min; @@ -2946,12 +2946,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; unsigned int shift = mc->shift; int min = mc->min; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; + int ret; val = ((ucontrol->value.integer.value[0] + min) & mask); if (invert) @@ -2959,7 +2961,21 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val_mask = mask << shift; val = val << shift; - return snd_soc_update_bits_locked(codec, reg, val_mask, val); + ret = snd_soc_update_bits_locked(codec, reg, val_mask, val); + if (ret != 0) + return ret; + + if (snd_soc_volsw_is_stereo(mc)) { + val = ((ucontrol->value.integer.value[1] + min) & mask); + if (invert) + val = max - val; + val_mask = mask << shift; + val = val << shift; + + ret = snd_soc_update_bits_locked(codec, rreg, val_mask, val); + } + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); @@ -2979,11 +2995,13 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; unsigned int shift = mc->shift; int min = mc->min; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; + int ret; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; @@ -2993,6 +3011,16 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = ucontrol->value.integer.value[0] - min; + if (snd_soc_volsw_is_stereo(mc)) { + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, rreg) >> shift) & mask; + if (invert) + ucontrol->value.integer.value[1] = + max - ucontrol->value.integer.value[1]; + ucontrol->value.integer.value[1] = + ucontrol->value.integer.value[1] - min; + } + return 0; } EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); -- 1.7.10.4

2 1

[alsa-devel] Newbie help with WM8962 and tinyalsa
by Randi Eckstein 19 Dec '12

19 Dec '12

If anyone has an explanation on how to use tinymix to play on a wm8962, I'd really appreciate it. Thanks, Randi Eckstein

2 1

Re: [alsa-devel] [PATCH] ASoC: dapm: Fix the unpaired runtime_get/put cases
by Mark Brown 19 Dec '12

19 Dec '12

On Wed, Dec 19, 2012 at 06:36:37PM +0800, Chuansheng Liu wrote: > But some devices has been set to STANDY bias directly during device probing, > such as cs42l73_probe(): > cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); > Then it will cause runtime_get() not be called but laterly runtime_put() will > be called. Also found some other uppaired cases. This is just a bug in the driver, if it's idle_bias_off then it really should be starting in _OFF or at the very least starting actually in _STANDBY (including taking the runtime reference) rather than partially in _STANDBY. > So here add new flag get_runtime, and the logic will be: > 1/ when device is from off to non-off bias, runtime_get() will be called if not yet; > 2/ When device is off bias, runtime_put() will be called if runtime_get() has > been called; This is really not a good idea at all, it's just adding new special cases and making the code more obscure.

1 0

[alsa-devel] Speaker not working; muted?
by Martin Michlmayr 19 Dec '12

19 Dec '12

I've a problem with audio on my HP 2560p laptop (using snd_hda_intel): the speaker no longer works. However, I successfully get sound on a headphone. I've attached the output of alsa-info. It would be great if someone could take a look at let me know if this is a bug in the driver or wrong settings. Background: I used ALSA successfully in the past to play audio with the in-built speaker. At some point, I used aumix (which requires OSS modules) to turn the volume to zero. I believe I turned the volume back up again. I also rebooted my laptop at some point. I don't know if it's related to using aumix or something else, but at some point the in-built speaker stopped working... I can still listen to audio using a headphone. Another observation is that my laptop has a speaker mute button. When I reboot my laptop, and the BIOS or grub is running, I see that the mute button is activated. However, pressing the button doesn't change anything... when the Linux kernel loads, the mute button is still on, but when ALSA starts the button changes colour: it seems to me that ALSA thinks that mute is off while in reality it is on. -- Martin Michlmayr http://www.cyrius.com/

1 0

[alsa-devel] [PATCH 1/2] Revert "ALSA: usb-audio: ignore delay calculation for capture stream"
by Pierre-Louis Bossart 19 Dec '12

19 Dec '12

This reverts commit 3f94fad09538ec988919ec3f371841182df71d04. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..b09b895 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -78,8 +78,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->delay = snd_usb_pcm_delay(subs, substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); -- 1.7.9.5

1 1

[alsa-devel] [PATCH 1/3] ASoC: arizona: Add noise gate hold time enumeration
by Mark Brown 18 Dec '12

18 Dec '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 9 +++++++++ sound/soc/codecs/arizona.h | 2 ++ 2 files changed, 11 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a85a68f..89c0433 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -323,6 +323,15 @@ const struct soc_enum arizona_lhpf4_mode = arizona_lhpf_mode_text); EXPORT_SYMBOL_GPL(arizona_lhpf4_mode); +static const char *arizona_ng_hold_text[] = { + "30ms", "120ms", "250ms", "500ms", +}; + +const struct soc_enum arizona_ng_hold = + SOC_ENUM_SINGLE(ARIZONA_NOISE_GATE_CONTROL, ARIZONA_NGATE_HOLD_SHIFT, + 4, arizona_ng_hold_text); +EXPORT_SYMBOL_GPL(arizona_ng_hold); + int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index db0c8dd..68dfae5 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -176,6 +176,8 @@ extern const struct soc_enum arizona_lhpf2_mode; extern const struct soc_enum arizona_lhpf3_mode; extern const struct soc_enum arizona_lhpf4_mode; +extern const struct soc_enum arizona_ng_hold; + extern int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); -- 1.7.10.4

1 2

[alsa-devel] [PATCH] ASoC: arizona: Correct FLL source definitions
by Mark Brown 18 Dec '12

18 Dec '12

The FLL source constants were numbered as a simple enumeration but were being used in the code as direct values to be written to the registers. Renumber the constants to reflect the usage. Reported-by: Ryo Tsutsui <Ryo.Tsutsui(a)wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.h | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 7f22b4f..68dfae5 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -34,15 +34,15 @@ #define ARIZONA_FLL_SRC_MCLK1 0 #define ARIZONA_FLL_SRC_MCLK2 1 -#define ARIZONA_FLL_SRC_SLIMCLK 2 -#define ARIZONA_FLL_SRC_FLL1 3 -#define ARIZONA_FLL_SRC_FLL2 4 -#define ARIZONA_FLL_SRC_AIF1BCLK 5 -#define ARIZONA_FLL_SRC_AIF2BCLK 6 -#define ARIZONA_FLL_SRC_AIF3BCLK 7 -#define ARIZONA_FLL_SRC_AIF1LRCLK 8 -#define ARIZONA_FLL_SRC_AIF2LRCLK 9 -#define ARIZONA_FLL_SRC_AIF3LRCLK 10 +#define ARIZONA_FLL_SRC_SLIMCLK 3 +#define ARIZONA_FLL_SRC_FLL1 4 +#define ARIZONA_FLL_SRC_FLL2 5 +#define ARIZONA_FLL_SRC_AIF1BCLK 8 +#define ARIZONA_FLL_SRC_AIF2BCLK 9 +#define ARIZONA_FLL_SRC_AIF3BCLK 10 +#define ARIZONA_FLL_SRC_AIF1LRCLK 12 +#define ARIZONA_FLL_SRC_AIF2LRCLK 13 +#define ARIZONA_FLL_SRC_AIF3LRCLK 14 #define ARIZONA_MIXER_VOL_MASK 0x00FE #define ARIZONA_MIXER_VOL_SHIFT 1 -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/1] ALSA: hda - bug fix for invalid connection list of more Haswell HDMI codec pins
by mengdong.lin＠intel.com 18 Dec '12

18 Dec '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Haswell HDMI codec may report invalid connection list entries on all three pins. But our previous patch only fixes the HDMI pin, not the DP pin, so audio playbak via DP can still fail. This patch adds fixup for Haswell to set proper connection list entries for the other 2 pins including the DP pin. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 42fdf14..73d298a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1710,8 +1710,14 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, if (vendor_param == -1 || !(vendor_param & 0x02)) return; + snd_hda_override_conn_list(codec, 0x05, 3, list); + hdmi_read_pin_conn(codec, 1); + snd_hda_override_conn_list(codec, 0x06, 3, list); hdmi_read_pin_conn(codec, 1); + + snd_hda_override_conn_list(codec, 0x07, 3, list); + hdmi_read_pin_conn(codec, 1); } -- 1.7.9.5

3 5

Re: [alsa-devel] [pulseaudio-discuss] Front mic not working properly
by Raymond Yau 18 Dec '12

18 Dec '12

>> > >> > I'm totally stuck with a sound problem I fail to debug on my own. >> > >> > My setup: >> > Asus P5Q Motherboard with Realtec ALC1200 soundchip. Stereo speakers plugged in to the rear output and headphone speaker/mic combo plugged in to the front jacks. System: Ubuntu 12.10 >> > Desired behaviour: Sound is played back on headphones AND rear speaker. Front mic should "just-work" Asus P5Q specification Realtek ALC1200, 8-channel High Definition Audio CODEC - Support Jack-Detection, Multi-streaming, and Front Panel Jack-Retasking Alt_playback is still not enabled to support multi streaming playback on realtec codecs >> > Problem 2: >> > This is the biggest problem. >> > As soon as I plug in the front mic, Inside gnome-alsamixer the volume for master, surround,center,lfe start to jump between on and off. Why is the Front mic jack controlling these values? Do you mean BIOS did not setup pin default correctly after asus-p5q model was removed? http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commitdiff;h=6f… > How can I debug this behaviour? Post the output of alsa-info.sh > I just want them to stay the same, if the mic is plugged in or not. >> > Inside the gnome volume settings, the output port starts to jump between "Analog Output" and "Analog Headphones" >> >> This look like problem in Jack detection of the headphone and Mic Jack of front audio panel >> >> alsamixer -c0 >> >> Does the playback switch change rapidly on and off in alsamixer ? >> >> Do your computer chassis have a hda front audio panel and you connect them to the front audio connector of the motherboard according to the user manual ? >> >> Does it help if you disabled the automute switch ? >> > Thanks a lot. Your input helped me to at least work around the problem. My chassis has a HDA and an AC97 connector. The HDA connector was plugged in and correctly configured in BIOS. > I now use the AC97 connector (and changed the BIOS settings) and everything is working properly. Checking the front audio panel specification of your computer chassis Setting front audio panel to AC97 in BIOS of Asus motherboard change the miscellaneous bit of pin default of front HP and Mic Jack （i.e. misc = NO_PRESENCE) The alsa driver won't enable the unsolicited event on these nodes > > Regarding your questions: Disabling automute didn't help. And yes, the playback switch toggled rapidly. http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commitdi… Try disable trigger sense by codec->no_trigger_sense = 1; > > One thing I didn't test: I mistakenly plugged the rear speaker into the front channel jack (plugging the green connector into a green jack seemed so obvious). I don't know is this is the same channel as the headphone connector on the front panel... > > In AC97 mode it doesn't matter if I use the front channel jack or the rear channel jack (both at the rear of the computer) to connect my main speakers.

1 0

[alsa-devel] [PATCH 0/2] A few fixes for Reatek codecs
by Takashi Iwai 17 Dec '12

17 Dec '12

Hi, here is a patch series for fixing some obvious errors I've overseen for Realtek codecs. It'll be queued for 3.8-rc1. Takashi

1 2

[alsa-devel] [sound:topic/hda-ca0132-dsp 4/11] sound/pci/hda/patch_ca0132.c:714 dspio_send() warn: always true condition '(res >= 0) => (0-u32max >= 0)'
by Dan Carpenter 17 Dec '12

17 Dec '12

Hi Ian, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/hda-ca0132-dsp head: cdc83c59e59d57b250be46c4b9d31e3b2b5ae589 commit: 01ef7dbffb411d9d78d1150b268d9c757f9f2f93 [4/11] ALSA: hda - Update CA0132 codec to load DSP firmware binary + sound/pci/hda/patch_ca0132.c:714 dspio_send() warn: always true condition '(res >= 0) => (0-u32max >= 0)' sound/pci/hda/patch_ca0132.c:1781 dspxfr_one_seg() info: why not propagate 'status' from dsp_dma_stop() instead of (-5)? sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: potential null dereference 'dma_engine'. (kzalloc returns null) sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: we previously assumed 'dma_engine' could be null (see line 1857) git remote add sound git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git remote update sound git checkout 01ef7dbffb411d9d78d1150b268d9c757f9f2f93 vim +714 sound/pci/hda/patch_ca0132.c 01ef7dbf Ian Minett 2012-09-20 698 VENDOR_CHIPIO_PLL_PMU_WRITE, 0xff); 01ef7dbf Ian Minett 2012-09-20 699 mutex_unlock(&spec->chipio_mutex); 01ef7dbf Ian Minett 2012-09-20 700 } 01ef7dbf Ian Minett 2012-09-20 701 01ef7dbf Ian Minett 2012-09-20 702 /* 01ef7dbf Ian Minett 2012-09-20 703 * CA0132 DSP IO stuffs 01ef7dbf Ian Minett 2012-09-20 704 */ 01ef7dbf Ian Minett 2012-09-20 705 static int dspio_send(struct hda_codec *codec, unsigned int reg, 01ef7dbf Ian Minett 2012-09-20 706 unsigned int data) 01ef7dbf Ian Minett 2012-09-20 707 { 01ef7dbf Ian Minett 2012-09-20 708 unsigned int res; 01ef7dbf Ian Minett 2012-09-20 709 int retry = 50; 01ef7dbf Ian Minett 2012-09-20 710 01ef7dbf Ian Minett 2012-09-20 711 /* send bits of data specified by reg to dsp */ 01ef7dbf Ian Minett 2012-09-20 712 do { 01ef7dbf Ian Minett 2012-09-20 713 res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, data); 01ef7dbf Ian Minett 2012-09-20 @714 if ((res >= 0) && (res != VENDOR_STATUS_DSPIO_BUSY)) 01ef7dbf Ian Minett 2012-09-20 715 return res; 01ef7dbf Ian Minett 2012-09-20 716 } while (--retry); 01ef7dbf Ian Minett 2012-09-20 717 01ef7dbf Ian Minett 2012-09-20 718 return -EIO; 01ef7dbf Ian Minett 2012-09-20 719 } 01ef7dbf Ian Minett 2012-09-20 720 01ef7dbf Ian Minett 2012-09-20 721 static void dspio_write_wait(struct hda_codec *codec) 01ef7dbf Ian Minett 2012-09-20 722 { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] The new alsa-driver repository
by Jaroslav Kysela 17 Dec '12

17 Dec '12

Hello all, the ALSA server has new alsa-driver repository which contains several branches for now: master - just empty one with README build - the previous master branch (but all was moved to the alsa/ subtree) mirror - mirrored code from upstream (ALSA parts only) release - merging branch for mirror/build trees to create a relationship between them All branches are ophan. It means that they have no common parent. I put some handy configuration hints to the ALSA WIKI: http://www.alsa-project.org/main/index.php/GIT_Server#Syncing_the_branch_re… The reason to move the old master branch to the alsa/ subtree is quite simple - you can merge/pull the branch to the full linux-2.6 kernel tree now and do your development there. Also, cherry picks between mirror and the linux-2.6 tree should work. Another handy extension - in the current repo - might be to create the linux-2.6 branch with the whole upstream and create another branch for merging the build branch with the whole tree like 'release26' or so. In this way, we can manage another releases of the out-of-kernel build code for the ALSA tree in stable linux kernel versions. Comments, other ideas are welcome. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix clicking noise on Thinkpad T61/R61i
by David Henningsson 17 Dec '12

17 Dec '12

The bug reporter reports that setting the speaker pin to D3 before turning off its pinctl fixes the clicking noise on powersave for Thinkpad T61. Thanks to c4pp4 for doing most of the work with this bug, including reporting and testing, and writing preliminary patches. BugLink: https://bugs.launchpad.net/bugs/886975 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 36 ++++++++++++++++++++++++++++++++---- 1 file changed, 32 insertions(+), 4 deletions(-) Note: c4pp4 pointed out that in his case, only the speaker needed to go to D3 before pinctl, but noticed no regressions from the below implementation, so I made the fixup more generic (easier to reuse later). diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a3a2263..27e6ca7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -144,6 +144,7 @@ struct conexant_spec { unsigned int asus:1; unsigned int pin_eapd_ctrls:1; unsigned int fixup_stereo_dmic:1; + unsigned int d3_before_shutup:1; /* Needs speaker to go to D3 before pinctl */ unsigned int adc_switching:1; @@ -558,13 +559,22 @@ static int conexant_build_controls(struct hda_codec *codec) return 0; } -#ifdef CONFIG_PM static int conexant_suspend(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; + + if (spec->d3_before_shutup) + snd_hda_codec_set_power_to_all(codec, codec->afg, AC_PWRST_D3, + false); snd_hda_shutup_pins(codec); return 0; } -#endif + +static void conexant_reboot_notify(struct hda_codec *codec) +{ + conexant_suspend(codec); +} + static const struct hda_codec_ops conexant_patch_ops = { .build_controls = conexant_build_controls, @@ -575,7 +585,7 @@ static const struct hda_codec_ops conexant_patch_ops = { #ifdef CONFIG_PM .suspend = conexant_suspend, #endif - .reboot_notify = snd_hda_shutup_pins, + .reboot_notify = conexant_reboot_notify, }; #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -4408,7 +4418,7 @@ static const struct hda_codec_ops cx_auto_patch_ops = { #ifdef CONFIG_PM .suspend = conexant_suspend, #endif - .reboot_notify = snd_hda_shutup_pins, + .reboot_notify = conexant_reboot_notify, }; /* @@ -4421,6 +4431,7 @@ enum { CXT_PINCFG_LEMOTE_A1205, CXT_FIXUP_STEREO_DMIC, CXT_FIXUP_INC_MIC_BOOST, + CXT_FIXUP_D3_BEFORE_SHUTUP, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -4430,6 +4441,13 @@ static void cxt_fixup_stereo_dmic(struct hda_codec *codec, spec->fixup_stereo_dmic = 1; } +static void cxt_fixup_d3_before_shutup(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + spec->d3_before_shutup = 1; +} + static void cxt5066_increase_mic_boost(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4499,6 +4517,15 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt5066_increase_mic_boost, }, + [CXT_FIXUP_D3_BEFORE_SHUTUP] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_d3_before_shutup, + }, +}; + +static const struct snd_pci_quirk cxt5045_fixups[] = { + SND_PCI_QUIRK(0x17aa, 0x20ac, "Lenovo T61/R61i", CXT_FIXUP_D3_BEFORE_SHUTUP), + {} }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -4553,6 +4580,7 @@ static int patch_conexant_auto(struct hda_codec *codec) switch (codec->vendor_id) { case 0x14f15045: codec->single_adc_amp = 1; + snd_hda_pick_fixup(codec, NULL, cxt5045_fixups, cxt_fixups); break; case 0x14f15051: add_cx5051_fake_mutes(codec); -- 1.7.9.5

2 3

[alsa-devel] ASoC updates for v3.8
by Mark Brown 17 Dec '12

17 Dec '12

2 3

[alsa-devel] [PATCH 0/3] at91/ssc: fixes on ASoC tree for 3.8
by Nicolas Ferre 17 Dec '12

17 Dec '12

Hi Mark, This series goes on top of current Linus' git tree and fixes an error that we have while compiling DTBs for AT91: ERROR (phandle_references): Reference to non-existent node or label "pinctrl_ssc0_tx" ERROR: Input tree has errors, aborting (use -f to force output) make[3]: *** [arch/arm/boot/dts/at91sam9g20ek.dtb] Error 2 I have included pending material that is dealing with SSC and pinctrl. The pinctrl part can be merged more easily now that the big pinctrl update for AT91 has been merged upstream. I have also included a typo patch that you have already taken as 315656b (ASoC: atmel-ssc: change disable to disable in dts node) in your topic/atmel branch but that was not described properly. If you find it easier, I can provide you a git tree for this material: just tell me... Thanks for your help, best regards, Bo Shen (3): ARM: at91/dts: fix typo in SSC status property ARM: at91/dts: add pinctrl support for SSC peripheral ASoC: atmel-ssc: add pinctrl selection to driver arch/arm/boot/dts/at91sam9260.dtsi | 18 +++++++++++++++++ arch/arm/boot/dts/at91sam9263.dtsi | 40 ++++++++++++++++++++++++++++++++++++-- arch/arm/boot/dts/at91sam9g45.dtsi | 40 ++++++++++++++++++++++++++++++++++++-- arch/arm/boot/dts/at91sam9n12.dtsi | 26 +++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9x5.dtsi | 32 +++++++++++++++++++++++------- drivers/misc/atmel-ssc.c | 8 ++++++++ 6 files changed, 153 insertions(+), 11 deletions(-) -- 1.8.0

2 4

Re: [alsa-devel] [Resend PATCH v6 1/4] pinctrl: atmel-ssc: add pinctrl support
by Nicolas Ferre 17 Dec '12

17 Dec '12

On 12/17/2012 10:39 AM, Jean-Christophe PLAGNIOL-VILLARD : > On 11:24 Wed 21 Nov , Mark Brown wrote: >> On Wed, Nov 21, 2012 at 10:17:09AM +0800, Bo Shen wrote: >>> Add pinctrl support for atmel ssc peripheral >>> >>> Signed-off-by: Bo Shen <voice.shen(a)atmel.com> >> >> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > you handle this one with scc Hold on Jean-Christophe, I made a little experiment this morning, and this patch is not applying correctly on: - linus' tree - linux-next - neither Mark's for-next branch So, *I* will manage this patch and try to make sure that Mark is able to seamlessly include it in his tree on top of a known source for a future ASoC "fixes" pull-request. Mark, what about rebasing this one on top of current Linus' tree (toguether with the one that you have just applied on the topic/atmel branch: (ASoC: atmel-ssc: change disable to disable in dts node)? I hope that once reworked, they will be able to reach Linus' tree soon because the lack of this patch is causing DTB compilation errors on all AT91... Best regards, -- Nicolas Ferre

2 1

[alsa-devel] [PATCH] ASoC: atmel-ssc: change disable to disable in dts node
by Bo Shen 17 Dec '12

17 Dec '12

Change the value of status to disabled to keep the consistent Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- arch/arm/boot/dts/at91sam9260.dtsi | 2 +- arch/arm/boot/dts/at91sam9263.dtsi | 4 ++-- arch/arm/boot/dts/at91sam9g45.dtsi | 4 ++-- arch/arm/boot/dts/at91sam9x5.dtsi | 2 +- 4 files changed, 6 insertions(+), 6 deletions(-) diff --git a/arch/arm/boot/dts/at91sam9260.dtsi b/arch/arm/boot/dts/at91sam9260.dtsi index aaa42d8..03c56cb 100644 --- a/arch/arm/boot/dts/at91sam9260.dtsi +++ b/arch/arm/boot/dts/at91sam9260.dtsi @@ -217,7 +217,7 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfffbc000 0x4000>; interrupts = <14 4 5>; - status = "disable"; + status = "disabled"; }; adc0: adc@fffe0000 { diff --git a/arch/arm/boot/dts/at91sam9263.dtsi b/arch/arm/boot/dts/at91sam9263.dtsi index 3b721ee..15f12dd 100644 --- a/arch/arm/boot/dts/at91sam9263.dtsi +++ b/arch/arm/boot/dts/at91sam9263.dtsi @@ -179,14 +179,14 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfff98000 0x4000>; interrupts = <16 4 5>; - status = "disable"; + status = "disabled"; }; ssc1: ssc@fff9c000 { compatible = "atmel,at91rm9200-ssc"; reg = <0xfff9c000 0x4000>; interrupts = <17 4 5>; - status = "disable"; + status = "disabled"; }; macb0: ethernet@fffbc000 { diff --git a/arch/arm/boot/dts/at91sam9g45.dtsi b/arch/arm/boot/dts/at91sam9g45.dtsi index acfa207..44a38d0 100644 --- a/arch/arm/boot/dts/at91sam9g45.dtsi +++ b/arch/arm/boot/dts/at91sam9g45.dtsi @@ -232,14 +232,14 @@ compatible = "atmel,at91sam9g45-ssc"; reg = <0xfff9c000 0x4000>; interrupts = <16 4 5>; - status = "disable"; + status = "disabled"; }; ssc1: ssc@fffa0000 { compatible = "atmel,at91sam9g45-ssc"; reg = <0xfffa0000 0x4000>; interrupts = <17 4 5>; - status = "disable"; + status = "disabled"; }; adc0: adc@fffb0000 { diff --git a/arch/arm/boot/dts/at91sam9x5.dtsi b/arch/arm/boot/dts/at91sam9x5.dtsi index 69667d0..0beff72 100644 --- a/arch/arm/boot/dts/at91sam9x5.dtsi +++ b/arch/arm/boot/dts/at91sam9x5.dtsi @@ -92,7 +92,7 @@ compatible = "atmel,at91sam9g45-ssc"; reg = <0xf0010000 0x4000>; interrupts = <28 4 5>; - status = "disable"; + status = "disabled"; }; tcb0: timer@f8008000 { -- 1.7.9.5

3 2

[alsa-devel] "hda-intel: azx_get_response timeout"
by Mantas M. 17 Dec '12

17 Dec '12

After upgrading to Linux 3.7 (on Arch Linux, x86_64), I've been getting the following messages in dmesg: > [Dec11 21:52] hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x001f0500 > [ +1.006709] hda-intel: No response from codec, disabling MSI: last cmd=0x001f0500 > [ +1.006773] hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x001f0500 This seems to happen when resuming from suspend-to-memory, and is new to v3.7 – I haven't seen any such messages on any kernels I've used earlier (between v3.0 and v3.6.9). The device is: > # lspci -v > 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06) > Subsystem: ASUSTeK Computer Inc. Device 1643 > Flags: bus master, fast devsel, latency 0, IRQ 22 > Memory at d5200000 (64-bit, non-prefetchable) [size=16K] > Capabilities: [50] Power Management version 2 > Capabilities: [60] MSI: Enable- Count=1/1 Maskable- 64bit+ > Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 > Capabilities: [100] Virtual Channel > Capabilities: [130] Root Complex Link > Kernel driver in use: snd_hda_intel Is there anything I can/should do about this? -- Mantas Mikulėnas <grawity(a)gmail.com>

2 1

Re: [alsa-devel] Could you help me about the ALSA?
by Zhang wei 17 Dec '12

17 Dec '12

> Yes,when buffer_size wraped the 'pos' is smallar than last one.If we > want to confirm the hw_base need to adjust there must be consider > another confidition: > > hdelta = jiffies - runtime->hw_ptr_jiffies; > > if (hdelta > runtime->hw_ptr_buffer_jiffies/2) > > how to connect this code with checking unnormal interrupt(lost or delayed)? This condition is for the double interrupt acknowledge problems (see comment). It makes sure, that the DMA buffer position was wrapped using another external timing source. Indeed, it's mostly a lowlevel driver fault to call this pointer update routine with the in_interrupt flag when the period was not really elapsed. -------------------------------------- I am sorry. I forget signed my info and this mail sent by me. Yes,I saw the comment and known the target of the pices of code and known alsa use jiffies as another timing source to detect the wrap situation,but I do not understand how to detect or check that case.In other word,why does alsa use the value runtime->hw_ptr_buffer_jiffies/2 not runtime->hw_ptr_buffer_jiffies/3 or runtime->hw_ptr_buffer_jiffies/4 ...? Best Regards zw

1 0

Re: [alsa-devel] Could you help me about the ALSA?
by Jaroslav Kysela 17 Dec '12

17 Dec '12

Date 14.12.2012 09:07, Zhang wei wrote: > Good afternoon . > > Excuse me if I disturb you, I found your contact information from the > alsa-devel > http://mailman.alsa-project.org/pipermail/alsa-devel/2010-August/030605.html > . > > Recently I am studying the alsa-driver and the code blocked me which > from the function snd_pcm_update_hw_ptr0inthe pcm_lib.c. > > =====================start==============================è > > if (in_interrupt) { > > /* we know that one period was processed */ > > /* delta = "expected next hw_ptr" for in_interrupt != > 0 */ > > delta = runtime->hw_ptr_interrupt + runtime->period_size; > > if (delta > new_hw_ptr) { > > /* check for double acknowledged interrupts */ > > hdelta = jiffies - runtime->hw_ptr_jiffies; > > if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { > > hw_base += runtime->buffer_size; > > if (hw_base >= runtime->boundary) > > hw_base = 0; > > new_hw_ptr = hw_base + pos; > > goto __delta; > > } > > } > > } > > ç================end================================= > > I’ve already read your mail but I still do not understand your intend. I > know the target of the pices of code is prevent the variable hw_ptr and > hw_base delayed update from some kind of case .so my question is > > (1) how could happen on delta>new_hw_ptr? And why? > > (2) Why we can conclue the hw_ptr crossed the boundary of > buffer_size when hdelta>runitme->hw_ptr_jiffies/2? What principle we > based on? The interrupts can be lost or delayed in some cases (small period sizes etc.). So you need to check for these situations. The driver returns only position in the ring buffer without count of "cross" points, so we need to trace these situations using another clock source (like system clock). Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

3 6

[alsa-devel] [PATCH] ASoC: cs42l52: Fix off-by-one boundary checking for cs42l52_get_clk()
by Axel Lin 16 Dec '12

16 Dec '12

The caller uses the return value of cs42l52_get_clk as array index, the valid array index is 0 .. ARRAY_SIZE(clk_map_table) - 1. Thus we need to ensure ret is less than ARRAY_SIZE(clk_map_table). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 99bb1c6..66817f6 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -749,7 +749,7 @@ static int cs42l52_get_clk(int mclk, int rate) } } } - if (ret > ARRAY_SIZE(clk_map_table)) + if (ret >= ARRAY_SIZE(clk_map_table)) return -EINVAL; return ret; } -- 1.7.9.5

1 1

[alsa-devel] [PATCH] ASoC: pcm: allow backend hardware to be freed in pause state
by Patrick Lai 15 Dec '12

15 Dec '12

When front-end PCM session is in paused state, back-end PCM session will be put in paused state as well if given front-end PCM session is the only client of given back-end. Then, application closes front-end PCM session, DPCM framework will not allow back-end enters HW_FREE state so back-end will never get shutdown completely. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ef22d0b..d340644 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1240,7 +1240,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; -- The Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation 1.7.8.3

2 1

[alsa-devel] [PATCH] ASoC: Prevent pop_wait overwrite
by Misael Lopez Cruz 15 Dec '12

15 Dec '12

pop_wait is used to determine if a deferred playback close needs to be cancelled when the a PCM is open or if after the power-down delay expires it needs to run. pop_wait is associated with the CODEC DAI, so the CODEC DAI must be unique. This holds true for most CODECs, except for the dummy CODEC and its DAI. In DAI links with non-unique dummy CODECs (e.g. front-ends), pop_wait can be overwritten by another DAI link using also a dummy CODEC. Failure to cancel a deferred close can cause mute due to the DAPM STOP event sent in the deferred work. One scenario where pop_wait is overwritten and causing mute is below (where hw:0,0 and hw:0,1 are two front-ends with default pmdown_time = 5 secs): aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1 sleep 1 aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 & aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE Since CODECs may not be unique, pop_wait is moved to the PCM runtime structure. Creating separate dummy CODECs for each DAI link can also solve the problem, but at this point it's only pop_wait variable in the CODEC DAI that has negative effects by not being unique. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> --- include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/soc-compress.c | 2 +- sound/soc/soc-pcm.c | 12 ++++++------ 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 628db7b..3953cea 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -242,7 +242,6 @@ struct snd_soc_dai { unsigned int symmetric_rates:1; struct snd_pcm_runtime *runtime; unsigned int active; - unsigned char pop_wait:1; unsigned char probed:1; struct snd_soc_dapm_widget *playback_widget; diff --git a/include/sound/soc.h b/include/sound/soc.h index 91244a0..769e27c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1039,6 +1039,7 @@ struct snd_soc_pcm_runtime { struct snd_soc_dpcm_runtime dpcm[2]; long pmdown_time; + unsigned char pop_wait:1; /* runtime devices */ struct snd_pcm *pcm; diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 967d0e1..5fbfb06 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -113,7 +113,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); } else - codec_dai->pop_wait = 1; + rtd->pop_wait = 1; schedule_delayed_work(&rtd->delayed_work, msecs_to_jiffies(rtd->pmdown_time)); } else { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5c3ca2a..d7711fc 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -334,11 +334,11 @@ static void close_delayed_work(struct work_struct *work) dev_dbg(rtd->dev, "ASoC: pop wq checking: %s status: %s waiting: %s\n", codec_dai->driver->playback.stream_name, codec_dai->playback_active ? "active" : "inactive", - codec_dai->pop_wait ? "yes" : "no"); + rtd->pop_wait ? "yes" : "no"); /* are we waiting on this codec DAI stream */ - if (codec_dai->pop_wait == 1) { - codec_dai->pop_wait = 0; + if (rtd->pop_wait == 1) { + rtd->pop_wait = 0; snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); } @@ -408,7 +408,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); } else { /* start delayed pop wq here for playback streams */ - codec_dai->pop_wait = 1; + rtd->pop_wait = 1; schedule_delayed_work(&rtd->delayed_work, msecs_to_jiffies(rtd->pmdown_time)); } @@ -480,8 +480,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) /* cancel any delayed stream shutdown that is pending */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - codec_dai->pop_wait) { - codec_dai->pop_wait = 0; + rtd->pop_wait) { + rtd->pop_wait = 0; cancel_delayed_work(&rtd->delayed_work); } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] configure: do not detect incorrect cross-compiler
by Olivier Blin 15 Dec '12

15 Dec '12

On Ubuntu 11.04, configuring with --build=x86_64-unknown-linux-gnu --host=x86_64-linux-gnu finds a wrong cross-compiler: checking for cross-compiler... x86_64-x86_64-pc-linux-gnu-gcc This happens because of a dash vs underscore inconsistency in configure.in: host=x86_64-pc-linux-gnu host_cpu=x86_64 host_os=linux-gnu which ${host_cpu}-${host_os}-gcc >/dev/null 2>&1 && echo ${host_cpu}-${host-os}-gcc This bug has been introduced in the initial --with-host support from 2002, commit eb267ade29c9a49c07b1c33dc9bf7a6790217400 This configure command is about "cross-compiling for i586", where the system compiler is used, which just -m32 additional options. The --build value comes from config.guess. --- configure.in | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.in b/configure.in index e397dad..4af917b 100644 --- a/configure.in +++ b/configure.in @@ -31,7 +31,7 @@ then which ${program_prefix}gcc >/dev/null 2>&1 && CC=${program_prefix}gcc which ${host_cpu}-${host_os}-gcc >/dev/null 2>&1 \ - && CC=${host_cpu}-${host-os}-gcc + && CC=${host_cpu}-${host_os}-gcc which ${host_cpu}-${host_vendor}-${host_os}-gcc >/dev/null 2>&1 \ && CC=${host_cpu}-${host_vendor}-${host_os}-gcc -- 1.8.0.2

2 1

[alsa-devel] [PATCH] usb-audio: ignore-quirk for HP Wireless Audio
by Eldad Zack 15 Dec '12

15 Dec '12

As Joe Cooper <swelljoe(a)gmail.com> reported, "On most HP Envy laptops the snd-usb-audio module causes the system to become unresponsive and Gnome Shell 3 to crash.". See also: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-December/057729.h… Add a quirk to ignore this device (for now) to solve the instability issue and allow other USB audio devices to be used. Reported-by: Joe Cooper <swelljoe(a)gmail.com> Tested-by: Isaac Smith <hunternet93(a)gmail.com> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/quirks-table.h | 36 ++++++++++++++++++++++++++++++++++++ 1 files changed, 36 insertions(+), 0 deletions(-) Applies against mainline 3.7.0 and sound-3.8. diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 49f9af9..e3c59f0 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -99,6 +99,42 @@ }, /* + * HP Wireless Audio + * When not ignored, causes instability issues for some users, forcing them to + * blacklist the entire module. + */ +{ + USB_DEVICE(0x0424, 0xb832), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Standard Microsystems Corp.", + .product_name = "HP Wireless Audio", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + /* Mixer */ + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* Playback */ + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* Capture */ + { + .ifnum = 2, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* HID Device, .ifnum = 3 */ + { + .ifnum = -1, + } + } + } +}, + +/* * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface * class matches do not take effect without an explicit ID match. */ -- 1.7.8.6

2 1

[alsa-devel] [PATCH v2 1/1] ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pin
by mengdong.lin＠intel.com 14 Dec '12

14 Dec '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Haswell HDMI codec pins may report invalid connection list entries, which will cause failure to play audio via HDMI. So this patch - allow generic hdmi patch to pick up and apply fixups for different codecs. - add fixup for Haswell to workaround this hardware issue: enable DP1.2 mode and override the connection list entries with proper value. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Xingchao Wang <xingchao.wang(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 71555cc..42fdf14 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -38,6 +38,7 @@ #include <sound/tlv.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_jack.h" static bool static_hdmi_pcm; @@ -79,6 +80,7 @@ struct hdmi_spec_per_pin { }; struct hdmi_spec { + struct hda_gen_spec gen; int num_cvts; struct hdmi_spec_per_cvt cvts[MAX_HDMI_CVTS]; @@ -1687,6 +1689,54 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .unsol_event = hdmi_unsol_event, }; +static void intel_haswell_fixup_connect_list(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + unsigned int vendor_param; + hda_nid_t list[3] = {0x2, 0x3, 0x4}; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || vendor_param & 0x02) + return; + + /* enable DP1.2 mode */ + vendor_param |= 0x02; + snd_hda_codec_read(codec, 0x08, 0, 0x781, vendor_param); + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || !(vendor_param & 0x02)) + return; + + snd_hda_override_conn_list(codec, 0x06, 3, list); + hdmi_read_pin_conn(codec, 1); +} + + +/* available models for fixup */ +enum { + INTEL_HASWELL, +}; + +static const struct hda_model_fixup hdmi_models[] = { + {.id = INTEL_HASWELL, .name = "Haswell"}, + {} +}; + +static const struct snd_pci_quirk hdmi_fixup_tbl[] = { + SND_PCI_QUIRK(0x8086, 0x2010, "Haswell", INTEL_HASWELL), + {} /* terminator */ +}; + +static const struct hda_fixup hdmi_fixups[] = { + [INTEL_HASWELL] = { + .type = HDA_FIXUP_FUNC, + .v.func = intel_haswell_fixup_connect_list, + }, +}; + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1696,6 +1746,11 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + snd_hda_gen_init(&spec->gen); + + snd_hda_pick_fixup(codec, hdmi_models, hdmi_fixup_tbl, hdmi_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); -- 1.7.9.5

2 2

[alsa-devel] [PATCH 1/1] ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pin
by mengdong.lin＠intel.com 14 Dec '12

14 Dec '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Haswell HDMI codec pins may report invalid connection list entries, which will cause failure to play audio via HDMI. So this patch - allow generic hdmi patch to pick up and apply fixups for different codecs. - add fixup for Haswell to workaround this hardware issue: enable DP1.2 mode and override invalid connection list entries reported by hardware. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Xingchao Wang <xingchao.wang(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 71555cc..2bf1758 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -38,6 +38,7 @@ #include <sound/tlv.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_jack.h" static bool static_hdmi_pcm; @@ -79,6 +80,7 @@ struct hdmi_spec_per_pin { }; struct hdmi_spec { + struct hda_gen_spec gen; int num_cvts; struct hdmi_spec_per_cvt cvts[MAX_HDMI_CVTS]; @@ -1687,6 +1689,69 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .unsol_event = hdmi_unsol_event, }; +static void intel_haswell_fixup_enable_dp12_mode(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + unsigned int vendor_param; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || vendor_param & 0x02) + return; + + vendor_param |= 0x02; + snd_hda_codec_read(codec, 0x08, 0, 0x781, vendor_param); +} + +static void intel_haswell_fixup_connect_list(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + hda_nid_t list[3] = {0x2, 0x3, 0x4}; + unsigned int vendor_param; + + if (action != HDA_FIXUP_ACT_PROBE) + return; + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || !(vendor_param & 0x02)) + return; + + snd_hda_override_conn_list(codec, 0x06, 3, list); + hdmi_read_pin_conn(codec, 1); +} + + +/* available models for fixup */ +enum { + INTEL_HASWELL, + INTEL_HASWELL_CONNECT_LIST, +}; + +static const struct hda_model_fixup hdmi_models[] = { + {.id = INTEL_HASWELL, .name = "Haswell"}, + {} +}; + +static const struct snd_pci_quirk hdmi_fixup_tbl[] = { + SND_PCI_QUIRK(0x8086, 0x2010, "Haswell", INTEL_HASWELL), + {} /* terminator */ +}; + +static const struct hda_fixup hdmi_fixups[] = { + [INTEL_HASWELL] = { + .type = HDA_FIXUP_FUNC, + .v.func = intel_haswell_fixup_enable_dp12_mode, + .chained = true, + .chain_id = INTEL_HASWELL_CONNECT_LIST, + }, + [INTEL_HASWELL_CONNECT_LIST] = { + .type = HDA_FIXUP_FUNC, + .v.func = intel_haswell_fixup_connect_list, + }, +}; + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1696,11 +1761,19 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + snd_hda_gen_init(&spec->gen); + + snd_hda_pick_fixup(codec, hdmi_models, hdmi_fixup_tbl, hdmi_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); return -EINVAL; } + + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + codec->patch_ops = generic_hdmi_patch_ops; generic_hdmi_init_per_pins(codec); -- 1.7.9.5

3 4

Re: [alsa-devel] ALSA:aloop -- inconsistent {in-softirq-W} -> {softirq-on-W} usage
by Takashi Iwai 14 Dec '12

14 Dec '12

At Thu, 13 Dec 2012 21:36:05 +0100, Andreas Fenkart wrote: > > > Hi, > > Seen this error when stopping playback to loopback. > > # aplay -Dhw:0 -fcd /dev/urandom > Playing raw data '/dev/urandom' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo > ^CAborted by signa[ 65.562992] > [ 65.565493] ================================= > [ 65.570044] [ INFO: inconsistent lock state ] > [ 65.574601] 3.7.0-rc8+ #303 Not tainted > [ 65.578605] --------------------------------- > [ 65.583156] inconsistent {IN-SOFTIRQ-W} -> {SOFTIRQ-ON-W} usage. > [ 65.589436] aplay/156 [HC0[0]:SC0[0]:HE1:SE1] takes: > [ 65.594620] (&(&dpcm->timer_lock)->rlock){+.?...}, at: [<c02e90cd>] loopback_close+0x29/0x98 > [ 65.603604] {IN-SOFTIRQ-W} state was registered at: > [ 65.608699] [<c005d8ef>] mark_irqflags+0xa3/0x11c > [ 65.613820] [<c005dca1>] __lock_acquire+0x339/0x77c > [ 65.619112] [<c005e575>] lock_acquire+0x65/0xac > [ 65.624038] [<c03e0c17>] _raw_spin_lock+0x1f/0x2c > [ 65.629155] [<c02e8cdd>] loopback_timer_start+0x3d/0x108 > [ 65.634899] [<c02e9455>] loopback_timer_function+0x21/0x3c > [ 65.640821] [<c0035f3f>] call_timer_fn+0x63/0xfc > [ 65.645849] [<c0036127>] run_timer_softirq+0x14f/0x190 > [ 65.651410] [<c00311d1>] __do_softirq+0xb1/0x174 > [ 65.656426] [<c0031529>] irq_exit+0x6d/0x70 > [ 65.660989] [<c000dde9>] handle_IRQ+0x2d/0x60 > [ 65.665743] [<c000848d>] omap3_intc_handle_irq+0x59/0x60 > [ 65.671484] [<c000cfff>] __irq_svc+0x3f/0x64 > [ 65.676138] [<c000de86>] default_idle+0x22/0x24 > [ 65.681063] [<c000df89>] cpu_idle+0x89/0x94 > [ 65.685626] [<c05cf5bb>] start_kernel+0x25b/0x264 > [ 65.690743] irq event stamp: 74155 > [ 65.694296] hardirqs last enabled at (74155): [<c03df9b5>] __mutex_unlock_slowpath+0x85/0xec > [ 65.703215] hardirqs last disabled at (74154): [<c03df96b>] __mutex_unlock_slowpath+0x3b/0xec > [ 65.712132] softirqs last enabled at (72724): [<c0031215>] __do_softirq+0xf5/0x174 > [ 65.720141] softirqs last disabled at (72717): [<c0031529>] irq_exit+0x6d/0x70 > [ 65.727697] > [ 65.727697] other info that might help us debug this: > [ 65.734516] Possible unsafe locking scenario: > [ 65.734516] > [ 65.740697] CPU0 > [ 65.743250] ---- > [ 65.745802] lock(&(&dpcm->timer_lock)->rlock); > [ 65.750639] <Interrupt> > [ 65.753373] lock(&(&dpcm->timer_lock)->rlock); > [ 65.758392] > [ 65.758392] *** DEADLOCK *** > [ 65.758392] > [ 65.764581] 1 lock held by aplay/156: > [ 65.768404] #0: (&pcm->open_mutex){+.+.+.}, at: [<c02e0c49>] snd_pcm_release+0x1d/0x7c > [ 65.776907] > [ 65.776907] stack backtrace: > [ 65.781487] [<c0011a15>] (unwind_backtrace+0x1/0xa0) from [<c005d2c9>] (print_usage_bug+0x149/0x184) > [ 65.791039] [<c005d2c9>] (print_usage_bug+0x149/0x184) from [<c005d4c5>] (mark_lock_irq+0x1c1/0x1e0) > [ 65.800590] [<c005d4c5>] (mark_lock_irq+0x1c1/0x1e0) from [<c005d5f7>] (mark_lock+0x113/0x368) > [ 65.809597] [<c005d5f7>] (mark_lock+0x113/0x368) from [<c005d915>] (mark_irqflags+0xc9/0x11c) > [ 65.818514] [<c005d915>] (mark_irqflags+0xc9/0x11c) from [<c005dca1>] (__lock_acquire+0x339/0x77c) > [ 65.827884] [<c005dca1>] (__lock_acquire+0x339/0x77c) from [<c005e575>] (lock_acquire+0x65/0xac) > [ 65.846083] [<c03e0c17>] (_raw_spin_lock+0x1f/0x2c) from [<c02e90cd>] (loopback_close+0x29/0x98) > [ 65.855273] [<c02e90cd>] (loopback_close+0x29/0x98) from [<c02e0bf1>] (snd_pcm_release_substream+0x39/0x74) > [ 65.865458] [<c02e0bf1>] (snd_pcm_release_substream+0x39/0x74) from [<c02e0c4f>] (snd_pcm_release+0x23/0x7c) > [ 65.875748] [<c02e0c4f>] (snd_pcm_release+0x23/0x7c) from [<c00b4dad>] (__fput+0x61/0x170) > [ 65.884409] [<c00b4dad>] (__fput+0x61/0x170) from [<c0040485>] (task_work_run+0x5d/0x8c) > [ 65.892879] [<c0040485>] (task_work_run+0x5d/0x8c) from [<c001017f>] (do_work_pending+0x67/0x7c) > [ 65.902069] [<c001017f>] (do_work_pending+0x67/0x7c) from [<c000d569>] (work_pending+0x9/0x18) > > This seems to be a real deadlock, since loopback_timer_stop is > triggered from ioctl, while loopback_timer_start can be called > from softirq. Could you check the latest Linus tree? The spinlocks of aloop have been already fixed there. Takashi

1 0

[alsa-devel] Polling issues with the PCM multi plugin
by Devin Anderson 14 Dec '12

14 Dec '12

Hi all, On the jack-devel list, we've been having a discussion about the ALSA PCM multi plugin: http://thread.gmane.org/gmane.comp.audio.jackit/26571/focus=26571 I've been having some issues with setting up my 2 Echo Layla 3Gs as a PCM multi. In particular, I've found that there are times when polling the file descriptors of the multi when poll() returns, but there isn't actually a period of data available for capture/playback, even though the poll descriptors indicate that the data should be available. After some debugging and learning about the PCM multi driver: 1.) The PCM multi driver only returns poll descriptors for one of the devices that makes up the multi 2.) When this happens, the device that's actually being polled *does* have data available 3.) When this happens, the device that's not being polled does not yet have a period's worth of data available After making sure that I had word clock enabled and that my word clock cable was working properly, I started delving into the problem some more. From what I can gather, the PCM multi device uses the function `snd_pcm_link()` to sync its devices, and, for start, stop, drain, pause, suspend, resume, reset, and prepare operations, this appears to work fine; however, I can't find any evidence that a PCM device checks to make sure that all of its linked slave devices have their poll descriptors ready with events before indicating that events are available. Questions: 1.) Should `snd_pcm_link` make I/O availability indication dependent on I/O availability of its linked slave devices? 2.) If not, how should this operation be handled? I'm sure it *can* be handled in userspace by bypassing the multi and polling the file descriptors of all sound cards directly, but duplicating this code for each application that wants to poll multiple sound cards seems wrong. Thanks, -- Devin Anderson surfacepatterns (at) gmail (dot) com blog - http://surfacepatterns.blogspot.com/ psinsights - http://psinsights.googlecode.com/ synthclone - http://synthclone.googlecode.com/

2 5

[alsa-devel] [PATCH] ASoC: wm5110: Split input PGA controls
by Mark Brown 14 Dec '12

14 Dec '12

Though the controls are named as stereo controls in the part the main use case for the analogue inputs to the WM5102 is mono. Reflect this in the controls exposed to userspace, providing a series of mono controls rather than stereo ones. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 77 +++++++++++++++++++++++++++------------------ 1 file changed, 46 insertions(+), 31 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 794f050..55a1348 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -55,37 +55,52 @@ SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL, ARIZONA_IN4_OSR_SHIFT, 1, 0), -SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1R_CONTROL, - ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2R_CONTROL, - ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3R_CONTROL, - ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), - -SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN4 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L, - ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_MUTE_SHIFT, 1, 1), - -SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN4 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4L, - ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), +SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, + ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_SINGLE("IN1L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN1R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN4L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_IN4L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN4R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4R, + ARIZONA_IN4R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN4L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_IN4L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN4R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4R, + ARIZONA_IN4R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm2000: Use clock API integration to configure MCLK divisor
by Mark Brown 14 Dec '12

14 Dec '12

Since we are now using the clock API integration to manage MCLK we can now use clk_get_rate() to determine if we need to divide MCLK without relying on platform data. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/wm2000.h | 3 --- sound/soc/codecs/wm2000.c | 6 +++--- 2 files changed, 3 insertions(+), 6 deletions(-) diff --git a/include/sound/wm2000.h b/include/sound/wm2000.h index aa388ca..4de81f4 100644 --- a/include/sound/wm2000.h +++ b/include/sound/wm2000.h @@ -15,9 +15,6 @@ struct wm2000_platform_data { /** Filename for system-specific image to download to device. */ const char *download_file; - /** Divide MCLK by 2 for system clock? */ - unsigned int mclkdiv2:1; - /** Disable speech clarity enhancement, for use when an * external algorithm is used. */ unsigned int speech_enh_disable:1; diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 0aba8ce..85550dc 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -73,7 +73,6 @@ struct wm2000_priv { unsigned int anc_eng_ena:1; unsigned int spk_ena:1; - unsigned int mclk_div:1; unsigned int speech_clarity:1; int anc_download_size; @@ -133,6 +132,7 @@ static int wm2000_poll_bit(struct i2c_client *i2c, static int wm2000_power_up(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + unsigned long rate; int ret; BUG_ON(wm2000->anc_mode != ANC_OFF); @@ -145,7 +145,8 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) return ret; } - if (!wm2000->mclk_div) { + rate = clk_get_rate(wm2000->mclk); + if (rate <= 13500000) { dev_dbg(&i2c->dev, "Disabling MCLK divider\n"); wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_MCLK_DIV2_ENA_CLR); @@ -847,7 +848,6 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, filename = "wm2000_anc.bin"; pdata = dev_get_platdata(&i2c->dev); if (pdata) { - wm2000->mclk_div = pdata->mclkdiv2; wm2000->speech_clarity = !pdata->speech_enh_disable; if (pdata->download_file) -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm5102: Split input PGA controls
by Mark Brown 14 Dec '12

14 Dec '12

Though the controls are named as stereo controls in the part the main use case for the analogue inputs to the WM5102 is mono. Reflect this in the controls exposed to userspace, providing a series of mono controls rather than stereo ones. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 64 +++++++++++++++++++++++++++------------------ 1 file changed, 38 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index b20feb2..3831645 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -611,32 +611,44 @@ SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, ARIZONA_IN3_OSR_SHIFT, 1, 0), -SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1R_CONTROL, - ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2R_CONTROL, - ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3R_CONTROL, - ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), - -SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), - -SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), +SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, + ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_SINGLE("IN1L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN1R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), -- 1.7.10.4

1 0

[alsa-devel] Presonus 44VSL AudioBox drivers
by Nicholas Hadaway 14 Dec '12

14 Dec '12

I own a Presonus AudioBox 44VSL and it does not work with alsa currently. I am an active linux user that runs git kernels and keeps an active development environment. I would love to get this hardware supported. what do I need to do to help you guys with this? Thanks! -Nick

1 0

[alsa-devel] Request for HW auto-recognition: HP Pavilion dv7 with STAC92xx
by Tomas Pospisek 13 Dec '12

13 Dec '12

Hello, I have a HP Pavilion dv7 laptop (LS066EA#UUZ), with: # cat /proc/asound/card0/codec* | grep Codec Codec: IDT 92HD81B1X5 Codec: Intel CougarPoint HDMI when play sound without any tweaks, I get a very weak sound output. If I want to have "beats audio", that is subwoofer and all speakers, I need to set: # cat /etc/modprobe.d/alsa-base.conf options snd-hda-intel model=hp-dv7-4000 Could the snd_hda_intel alsa driver be updated to recognize and setup the driver accordingly? alsa-info.sh output here: http://www.alsa-project.org/db/?f=54ea874f43f12c1572aff3ee3113dc4b0ad0b398 Thanks, *t

2 3

[alsa-devel] [PATCH v2 1/1] ALSA: hda - use usleep_range in link reset and change timeout check
by mengdong.lin＠intel.com 13 Dec '12

13 Dec '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Reducing the time on HDA link reset can help to reduce the driver loading time. So we replace msleep with usleep_range to get more accurate time control and change the value to a smaller one. And a 100ms timeout is set for both entering and exiting the link reset. Signed-off-by: Xingchao Wang <xingchao.wang(a)intel.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 72b085a..ed3a290 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1041,7 +1041,7 @@ static void azx_power_notify(struct hda_bus *bus, bool power_up); /* reset codec link */ static int azx_reset(struct azx *chip, int full_reset) { - int count; + unsigned long timeout; if (!full_reset) goto __skip; @@ -1052,24 +1052,26 @@ static int azx_reset(struct azx *chip, int full_reset) /* reset controller */ azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET); - count = 50; - while (azx_readb(chip, GCTL) && --count) - msleep(1); + timeout = jiffies + msecs_to_jiffies(100); + while (azx_readb(chip, GCTL) && + time_before(jiffies, timeout)) + usleep_range(500, 1000); /* delay for >= 100us for codec PLL to settle per spec * Rev 0.9 section 5.5.1 */ - msleep(1); + usleep_range(500, 1000); /* Bring controller out of reset */ azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); - count = 50; - while (!azx_readb(chip, GCTL) && --count) - msleep(1); + timeout = jiffies + msecs_to_jiffies(100); + while (!azx_readb(chip, GCTL) && + time_before(jiffies, timeout)) + usleep_range(500, 1000); /* Brent Chartrand said to wait >= 540us for codecs to initialize */ - msleep(1); + usleep_range(1000, 1200); __skip: /* check to see if controller is ready */ -- 1.7.9.5

5 12

[alsa-devel] [PATCH 0/2] Make ASUS Xonar U3 S/PDIF channel accessible
by Denis Washington 13 Dec '12

13 Dec '12

Hello, This is my first ALSA patch ever (in fact, also my first kernel patch and my first e-mail to this mailing list); as such, I'm sorry when I did something wrong and/or didn't follow the procedure properly. I hope you can forgive me. I recently bought an ASUS Xonar U3 USB audio card. It general it works well with Linux, but unfortunately, it's PCM output channel is unusable for headphones because it allows virtually no volume control (maximum loudness is already reach at a volume level of 2 or 3). The card does also have a digital (S/PDIF) output channel, which however isn't switched on by the Linux USB audio driver: http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg28970.html When I boot into Windows (where the ASUS driver seemingly enables the channel) and then reboot into Linux, I can use this channel and control volume as expected. It's just this turning on of S/PDIF which is missing, as suggested by the alsa-user thread linked to above. The patchset that follows enables S/PDIF on the Xonar U3 by using the (seemingly compatible) mixer quirks code as for the U1, just as done and tested by the initiator of the alsa-user thread, together with a very minor cleanup (renaming internal xonar_u1_* identifiers to just xonar_u_*). The change seems to work well (tested with the 3.7.0 Ubuntu kernel as found in today' daily version). Regard, Denis Washington

2 4

Re: [alsa-devel] [PATCH 1/2] ASoC: tlv320aic326x: Support for TI TLV320AIC3262 audio driver
by Mark Brown 13 Dec '12

13 Dec '12

On Wed, Dec 12, 2012 at 04:23:48PM +0530, Mehar Bajwa wrote: > TI TLV320AIC3262 is a flexible, low-power, low-voltage stereo audio > codec with digital microphone inputs and programmable outputs. It includes > powertune capabilities, enhanced fully-programmable miniDSP, fixed predefined > and parameterizable signal processing blocks, integrated PLL and flexible digital > interfaces. Is this in some way different to the patch set you sent yesterday?

2 1

Re: [alsa-devel] [PATCH] ASoC: max98090: Resubmit device driver
by Mark Brown 13 Dec '12

13 Dec '12

On Wed, Dec 12, 2012 at 04:36:10PM -0800, Jerry Wong wrote: > I have made the changes you suggested below, except: I've not read the patch yet but I see that once more I need to ask you to look at and follow SubmittingPatches - you're getting closer but the at least the changelog is nothing like it should be. Please look at both SubmittingPatches and other people's patches and try to follow the standard form.

1 0

[alsa-devel] [PATCH 1/1] ASoC: tpa6130a2: Use devm_* APIs
by Sachin Kamat 13 Dec '12

13 Dec '12

Converted to use devm_gpio_request and devm_regulator_get APIs. These are device managed and make error handling and cleanup a bit simpler. Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- Compile tested on for-next branch of following tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git --- sound/soc/codecs/tpa6130a2.c | 23 ++++++----------------- 1 files changed, 6 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 565ff39..ec78073 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -398,7 +398,8 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, TPA6130A2_MUTE_L; if (data->power_gpio >= 0) { - ret = gpio_request(data->power_gpio, "tpa6130a2 enable"); + ret = devm_gpio_request(dev, data->power_gpio, + "tpa6130a2 enable"); if (ret < 0) { dev_err(dev, "Failed to request power GPIO (%d)\n", data->power_gpio); @@ -419,16 +420,16 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, break; } - data->supply = regulator_get(dev, regulator); + data->supply = devm_regulator_get(dev, regulator); if (IS_ERR(data->supply)) { ret = PTR_ERR(data->supply); dev_err(dev, "Failed to request supply: %d\n", ret); - goto err_regulator; + goto err_gpio; } ret = tpa6130a2_power(1); if (ret != 0) - goto err_power; + goto err_gpio; /* Read version */ @@ -440,15 +441,10 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, /* Disable the chip */ ret = tpa6130a2_power(0); if (ret != 0) - goto err_power; + goto err_gpio; return 0; -err_power: - regulator_put(data->supply); -err_regulator: - if (data->power_gpio >= 0) - gpio_free(data->power_gpio); err_gpio: tpa6130a2_client = NULL; @@ -457,14 +453,7 @@ err_gpio: static int __devexit tpa6130a2_remove(struct i2c_client *client) { - struct tpa6130a2_data *data = i2c_get_clientdata(client); - tpa6130a2_power(0); - - if (data->power_gpio >= 0) - gpio_free(data->power_gpio); - - regulator_put(data->supply); tpa6130a2_client = NULL; return 0; -- 1.7.4.1

2 3

[alsa-devel] [Resend PATCH v6 1/4] pinctrl: atmel-ssc: add pinctrl support
by Bo Shen 13 Dec '12

13 Dec '12

Add pinctrl support for atmel ssc peripheral Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Resend - based on linux-next-20121115 - commit id: 3805711eb31d3b980c4a55a6994bbc8b22a9ec65 Change since v5 - correct the pull setting, without pull up Change since v4 - This is initial version --- arch/arm/boot/dts/at91sam9260.dtsi | 18 ++++++++++++++++++ arch/arm/boot/dts/at91sam9263.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9g45.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9n12.dtsi | 26 ++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9x5.dtsi | 32 +++++++++++++++++++++++++------- drivers/misc/atmel-ssc.c | 8 ++++++++ 6 files changed, 149 insertions(+), 7 deletions(-) diff --git a/arch/arm/boot/dts/at91sam9260.dtsi b/arch/arm/boot/dts/at91sam9260.dtsi index 3164dcd..f9dc7db 100644 --- a/arch/arm/boot/dts/at91sam9260.dtsi +++ b/arch/arm/boot/dts/at91sam9260.dtsi @@ -217,6 +217,22 @@ }; }; + ssc0 { + pinctrl_ssc0_tx: ssc0_tx-0 { + atmel,pins = + <1 16 0x1 0x0 /* PB16 periph A */ + 1 17 0x1 0x0 /* PB17 periph A */ + 1 18 0x1 0x0>; /* PB18 periph A */ + }; + + pinctrl_ssc0_rx: ssc0_rx-0 { + atmel,pins = + <1 19 0x1 0x0 /* PB19 periph A */ + 1 20 0x1 0x0 /* PB20 periph A */ + 1 21 0x1 0x0>; /* PB21 periph A */ + }; + }; + pioA: gpio@fffff400 { compatible = "atmel,at91rm9200-gpio"; reg = <0xfffff400 0x200>; @@ -350,6 +366,8 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfffbc000 0x4000>; interrupts = <14 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>; status = "disable"; }; diff --git a/arch/arm/boot/dts/at91sam9263.dtsi b/arch/arm/boot/dts/at91sam9263.dtsi index 96e0bc9..6e9a4ff 100644 --- a/arch/arm/boot/dts/at91sam9263.dtsi +++ b/arch/arm/boot/dts/at91sam9263.dtsi @@ -165,6 +165,38 @@ }; }; + ssc0 { + pinctrl_ssc0_tx: ssc0_tx-0 { + atmel,pins = + <1 0 0x2 0x0 /* PB0 periph B */ + 1 1 0x2 0x0 /* PB1 periph B */ + 1 2 0x2 0x0>; /* PB2 periph B */ + }; + + pinctrl_ssc0_rx: ssc0_rx-0 { + atmel,pins = + <1 3 0x2 0x0 /* PB3 periph B */ + 1 4 0x2 0x0 /* PB4 periph B */ + 1 5 0x2 0x0>; /* PB5 periph B */ + }; + }; + + ssc1 { + pinctrl_ssc1_tx: ssc1_tx-0 { + atmel,pins = + <1 6 0x1 0x0 /* PB6 periph A */ + 1 7 0x1 0x0 /* PB7 periph A */ + 1 8 0x1 0x0>; /* PB8 periph A */ + }; + + pinctrl_ssc1_rx: ssc1_rx-0 { + atmel,pins = + <1 9 0x1 0x0 /* PB9 periph A */ + 1 10 0x1 0x0 /* PB10 periph A */ + 1 11 0x1 0x0>; /* PB11 periph A */ + }; + }; + pioA: gpio@fffff200 { compatible = "atmel,at91rm9200-gpio"; reg = <0xfffff200 0x200>; @@ -262,6 +294,8 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfff98000 0x4000>; interrupts = <16 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>; status = "disable"; }; @@ -269,6 +303,8 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfff9c000 0x4000>; interrupts = <17 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc1_tx &pinctrl_ssc1_rx>; status = "disable"; }; diff --git a/arch/arm/boot/dts/at91sam9g45.dtsi b/arch/arm/boot/dts/at91sam9g45.dtsi index c40c3eb..63cf206 100644 --- a/arch/arm/boot/dts/at91sam9g45.dtsi +++ b/arch/arm/boot/dts/at91sam9g45.dtsi @@ -198,6 +198,38 @@ }; }; + ssc0 { + pinctrl_ssc0_tx: ssc0_tx-0 { + atmel,pins = + <3 0 0x1 0x0 /* PD0 periph A */ + 3 1 0x1 0x0 /* PD1 periph A */ + 3 2 0x1 0x0>; /* PD2 periph A */ + }; + + pinctrl_ssc0_rx: ssc0_rx-0 { + atmel,pins = + <3 3 0x1 0x0 /* PD3 periph A */ + 3 4 0x1 0x0 /* PD4 periph A */ + 3 5 0x1 0x0>; /* PD5 periph A */ + }; + }; + + ssc1 { + pinctrl_ssc1_tx: ssc1_tx-0 { + atmel,pins = + <3 10 0x1 0x0 /* PD10 periph A */ + 3 11 0x1 0x0 /* PD11 periph A */ + 3 12 0x1 0x0>; /* PD12 periph A */ + }; + + pinctrl_ssc1_rx: ssc1_rx-0 { + atmel,pins = + <3 13 0x1 0x0 /* PD13 periph A */ + 3 14 0x1 0x0 /* PD14 periph A */ + 3 15 0x1 0x0>; /* PD15 periph A */ + }; + }; + pioA: gpio@fffff200 { compatible = "atmel,at91rm9200-gpio"; reg = <0xfffff200 0x200>; @@ -331,6 +363,8 @@ compatible = "atmel,at91sam9g45-ssc"; reg = <0xfff9c000 0x4000>; interrupts = <16 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>; status = "disable"; }; @@ -338,6 +372,8 @@ compatible = "atmel,at91sam9g45-ssc"; reg = <0xfffa0000 0x4000>; interrupts = <17 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc1_tx &pinctrl_ssc1_rx>; status = "disable"; }; diff --git a/arch/arm/boot/dts/at91sam9n12.dtsi b/arch/arm/boot/dts/at91sam9n12.dtsi index 7b644c5..b211da6 100644 --- a/arch/arm/boot/dts/at91sam9n12.dtsi +++ b/arch/arm/boot/dts/at91sam9n12.dtsi @@ -28,6 +28,7 @@ tcb1 = &tcb1; i2c0 = &i2c0; i2c1 = &i2c1; + ssc0 = &ssc0; }; cpus { cpu@0 { @@ -199,6 +200,22 @@ }; }; + ssc0 { + pinctrl_ssc0_tx: ssc0_tx-0 { + atmel,pins = + <0 24 0x2 0x0 /* PA24 periph B */ + 0 25 0x2 0x0 /* PA25 periph B */ + 0 26 0x2 0x0>; /* PA26 periph B */ + }; + + pinctrl_ssc0_rx: ssc0_rx-0 { + atmel,pins = + <0 27 0x2 0x0 /* PA27 periph B */ + 0 28 0x2 0x0 /* PA28 periph B */ + 0 29 0x2 0x0>; /* PA29 periph B */ + }; + }; + pioA: gpio@fffff400 { compatible = "atmel,at91sam9x5-gpio", "atmel,at91rm9200-gpio"; reg = <0xfffff400 0x200>; @@ -249,6 +266,15 @@ status = "disabled"; }; + ssc0: ssc@f0010000 { + compatible = "atmel,at91sam9g45-ssc"; + reg = <0xf0010000 0x4000>; + interrupts = <28 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>; + status = "disable"; + }; + usart0: serial@f801c000 { compatible = "atmel,at91sam9260-usart"; reg = <0xf801c000 0x4000>; diff --git a/arch/arm/boot/dts/at91sam9x5.dtsi b/arch/arm/boot/dts/at91sam9x5.dtsi index 1cb26c28..df84f39 100644 --- a/arch/arm/boot/dts/at91sam9x5.dtsi +++ b/arch/arm/boot/dts/at91sam9x5.dtsi @@ -88,13 +88,6 @@ interrupts = <1 4 7>; }; - ssc0: ssc@f0010000 { - compatible = "atmel,at91sam9g45-ssc"; - reg = <0xf0010000 0x4000>; - interrupts = <28 4 5>; - status = "disable"; - }; - tcb0: timer@f8008000 { compatible = "atmel,at91sam9x5-tcb"; reg = <0xf8008000 0x100>; @@ -214,6 +207,22 @@ }; }; + ssc0 { + pinctrl_ssc0_tx: ssc0_tx-0 { + atmel,pins = + <0 24 0x2 0x0 /* PA24 periph B */ + 0 25 0x2 0x0 /* PA25 periph B */ + 0 26 0x2 0x0>; /* PA26 periph B */ + }; + + pinctrl_ssc0_rx: ssc0_rx-0 { + atmel,pins = + <0 27 0x2 0x0 /* PA27 periph B */ + 0 28 0x2 0x0 /* PA28 periph B */ + 0 29 0x2 0x0>; /* PA29 periph B */ + }; + }; + pioA: gpio@fffff400 { compatible = "atmel,at91sam9x5-gpio", "atmel,at91rm9200-gpio"; reg = <0xfffff400 0x200>; @@ -266,6 +275,15 @@ status = "disabled"; }; + ssc0: ssc@f0010000 { + compatible = "atmel,at91sam9g45-ssc"; + reg = <0xf0010000 0x4000>; + interrupts = <28 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>; + status = "disable"; + }; + usart0: serial@f801c000 { compatible = "atmel,at91sam9260-usart"; reg = <0xf801c000 0x200>; diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c index a769719..3b3d0e0 100644 --- a/drivers/misc/atmel-ssc.c +++ b/drivers/misc/atmel-ssc.c @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/of.h> +#include <linux/pinctrl/consumer.h> /* Serialize access to ssc_list and user count */ static DEFINE_SPINLOCK(user_lock); @@ -131,6 +132,13 @@ static int ssc_probe(struct platform_device *pdev) struct resource *regs; struct ssc_device *ssc; const struct atmel_ssc_platform_data *plat_dat; + struct pinctrl *pinctrl; + + pinctrl = devm_pinctrl_get_select_default(&pdev->dev); + if (IS_ERR(pinctrl)) { + dev_err(&pdev->dev, "Failed to request pinctrl\n"); + return PTR_ERR(pinctrl); + } ssc = devm_kzalloc(&pdev->dev, sizeof(struct ssc_device), GFP_KERNEL); if (!ssc) { -- 1.7.9.5

2 3

[alsa-devel] "disable" or "disabled" in DT?
by Takashi Iwai 13 Dec '12

13 Dec '12

Hi, in the recent commit for atmel-ssc to add DT support, you added the lines status = "disable"; Now looking through all commits for merging 3.8-rc1, I noticed that other codes have status = "disabled"; I think this should be kept consistent. Takashi

2 1

[alsa-devel] ALSA sound/usb/mixer.c:415 cannot get current value for control 2 ch 1: err = -22
by isaac smith 12 Dec '12

12 Dec '12

Success! I applied the patch against kernel 3.6.9 (AMD's Catalyst driver doesn't support 3.7 yet) and I can now use the snd-usb-audio driver without it hanging up on the wireless audio. I tested a USB sound card, it worked perfectly. There's no sign of the HP Wireless card in ALSA. Thanks a lot! Now I can finally use my slightly decent speakers and my USB MIDI controller. On Tue, Dec 11, 2012 at 12:59 PM, Eldad Zack <eldad(a)fogrefinery.com> wrote: > > On Mon, 10 Dec 2012, isaac smith wrote: > >> I have the same issue with my HP Envy 15. The sound card in question >> is an HP Wireless Audio card that only works with HP-branded wireless >> speakers, so playing to the card produces no output whatsoever, but >> does hang whatever process tries to access it. I doubt that many >> people ever use this feature, so unless someone invests the time to >> get it working I'd like a way to just ignore it completely. I've >> blacklisted the snd-usb-audio module for now, but I have a USB sound >> card and a USB MIDI controller that I'd like to be able to use. > > Hi Isaac, > > Try this patch - applies against 3.7. > > Let me know if that works for you and you can successfully use other USB > audio devices. > > Cheers, > Eldad > > diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h > index 88d8ceb..be68c66 100644 > --- a/sound/usb/quirks-table.h > +++ b/sound/usb/quirks-table.h > @@ -99,6 +99,42 @@ > }, > > /* > + * Standard Microsystems Corp. > + * HP Wireless Audio > + */ > +{ > + USB_DEVICE(0x0424, 0xb832), > + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { > + .vendor_name = "Standard Microsystems Corp.", > + .product_name = "HP Wireless Audio", > + .ifnum = QUIRK_ANY_INTERFACE, > + .type = QUIRK_COMPOSITE, > + .data = (const struct snd_usb_audio_quirk[]) { > + /* Mixer */ > + { > + .ifnum = 0, > + .type = QUIRK_IGNORE_INTERFACE, > + }, > + /* Playback */ > + { > + .ifnum = 1, > + .type = QUIRK_IGNORE_INTERFACE, > + }, > + /* Capture */ > + { > + .ifnum = 2, > + .type = QUIRK_IGNORE_INTERFACE, > + }, > + /* HID Device, maybe standard compliant? */ > + { > + /* .ifnum = 3 */ > + .ifnum = -1, > + } > + } > + } > +}, > + > +/* > * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface > * class matches do not take effect without an explicit ID match. > */ > >

1 0

[alsa-devel] Bug in aplay?
by Damien Zammit 12 Dec '12

12 Dec '12

Hi, I think i found a small bug in alsa utility "aplay". If you have a sound device that supports the S24_3BE format natively, and you have a wav file of the same format, aplay will not play it because it detects the wrong format and assumes it is default format. (ie 8000Hz Mono 1 channel). I compiled my own version of aplay and it works because I hacked the default format to match this format, but it is not the ideal solution. pi@raspberrypi ~ $ aplay --version aplay: version 1.0.25 by Jaroslav Kysela <perex(a)perex.cz> pi@raspberrypi ~ $ aplay -D hw:1 48000-S24_3BE.wav Playing raw data '48000-S24_3BE.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono aplay: set_params:1081: Sample format non available Available formats: - S24_3BE My version: pi@raspberrypi ~ $ ./aplay -D hw:1 48000-S24_3BE.wav Playing raw data '48000-S24_3BE.wav' : Signed 24 bit Big Endian in 3bytes, Rate 48000 Hz, Stereo aplay: set_params:1145: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S24_3BE SUBFORMAT: STD SAMPLE_BITS: 24 FRAME_BITS: 48 CHANNELS: 2 RATE: 48000 PERIOD_TIME: 125000 PERIOD_SIZE: 6000 PERIOD_BYTES: 36000 PERIODS: 4 BUFFER_TIME: 500000 BUFFER_SIZE: 24000 BUFFER_BYTES: 144000 TICK_TIME: 0 pi@raspberrypi ~ $ This is making it tricky for me to use my sound card. For some reason it works better with jackd. The error above is also a mystery to me, but it seems to be detecting the format better. Regards, Damien

3 6

[alsa-devel] [PATCH 0/2] Fix usb-audio delay calculation
by Takashi Iwai 12 Dec '12

12 Dec '12

Hi, the following two patches are for fixing the wrong PCM delay reported from usb-audio driver. The first one is trivial, just to fix / optimize for the capture direction. The second one is a deeper fix for corner cases like playback pause. Takashi

2 7

[alsa-devel] [PATCH 2/2] ASoC: aic: Support for AIC family DSPs
by Mehar Bajwa 12 Dec '12

12 Dec '12

AIC family of audio CODECs from TI features a programmable miniDSP for performing signal processing operations. Due to commonality of functions across the CODECs a common library will be used to provide support for them. Signed-off-by: Mehar Bajwa <mehar.bajwa(a)ti.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/aic3xxx_cfw.h | 529 +++++++++++++++++++ sound/soc/codecs/aic3xxx_cfw_ops.c | 989 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/aic3xxx_cfw_ops.h | 98 ++++ 5 files changed, 1623 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/aic3xxx_cfw.h create mode 100644 sound/soc/codecs/aic3xxx_cfw_ops.c create mode 100644 sound/soc/codecs/aic3xxx_cfw_ops.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 73d8cea..52b3601 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -188,6 +188,11 @@ config SND_SOC_ADAV80X config SND_SOC_ADS117X tristate +config SND_SOC_AIC_CFW + tristate + default y if SND_SOC_TLV320AIC3262=y + default m if SND_SOC_TLV320AIC3262=m + config SND_SOC_AK4104 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a3f8f4f..ffda11b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -9,6 +9,7 @@ snd-soc-adau1701-objs := adau1701.o snd-soc-adau1373-objs := adau1373.o snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o +snd-soc-aic3xxx-cfw-ops-objs := aic3xxx_cfw_ops.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4641-objs := ak4641.o @@ -132,6 +133,7 @@ obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o +obj-$(CONFIG_SND_SOC_AIC_CFW) += snd-soc-aic3xxx-cfw-ops.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o diff --git a/sound/soc/codecs/aic3xxx_cfw.h b/sound/soc/codecs/aic3xxx_cfw.h new file mode 100644 index 0000000..b88485d --- /dev/null +++ b/sound/soc/codecs/aic3xxx_cfw.h @@ -0,0 +1,529 @@ +/* + * aic3xxx_cfw.h -- SoC audio for TI OMAP44XX SDP + * Codec Firmware Declarations + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef AIC_FIRMWARE_H_ +#define AIC_FIRMWARE_H_ + + +#define AIC_FW_MAGIC 0xC0D1F1ED + + +/** \defgroup pd Arbitrary Limitations */ +/* @{ */ +#ifndef AIC_MAX_ID +# define AIC_MAX_ID (64) /**<Max length of string identifies*/ +# define AIC_MAX_VARS (256) /**<Number of "variables" alive at the*/ + /**<same time in an acx file*/ +#endif + +/* @} */ + + + +/** \defgroup st Enums, Flags, Macros and Supporting Types */ +/* @{ */ + + +/** + * Device Family Identifier + * + */ +enum __attribute__ ((__packed__)) aic_dfamily { + AIC_DFM_TYPE_A, + AIC_DFM_TYPE_B, + AIC_DFM_TYPE_C +}; + +/** + * Device Identifier + * + */ +enum __attribute__ ((__packed__)) aic_device { + AIC_DEV_DAC3120, + AIC_DEV_DAC3100, + AIC_DEV_AIC3120, + AIC_DEV_AIC3100, + AIC_DEV_AIC3110, + AIC_DEV_AIC3111, + AIC_DEV_AIC36, + AIC_DEV_AIC3206, + AIC_DEV_AIC3204, + AIC_DEV_AIC3254, + AIC_DEV_AIC3256, + AIC_DEV_AIC3253, + AIC_DEV_AIC3212, + AIC_DEV_AIC3262, + AIC_DEV_AIC3017, + AIC_DEV_AIC3008, + + AIC_DEV_AIC3266, + AIC_DEV_AIC3285, +}; + +/** + * Transition Sequence Identifier + * + */ +enum aic_transition_t { + AIC_TRN_INIT, + AIC_TRN_RESUME, + AIC_TRN_NEUTRAL, + AIC_TRN_A_MUTE, + AIC_TRN_D_MUTE, + AIC_TRN_AD_MUTE, + AIC_TRN_A_UNMUTE, + AIC_TRN_D_UNMUTE, + AIC_TRN_AD_UNMUTE, + AIC_TRN_SUSPEND, + AIC_TRN_EXIT, + AIC_TRN_N +}; + +#ifndef __cplusplus +static const char *const aic_transition_id[] = { + [AIC_TRN_INIT] "INIT", + [AIC_TRN_RESUME] "RESUME", + [AIC_TRN_NEUTRAL] "NEUTRAL", + [AIC_TRN_A_MUTE] "A_MUTE", + [AIC_TRN_D_MUTE] "D_MUTE", + [AIC_TRN_AD_MUTE] "AD_MUTE", + [AIC_TRN_A_UNMUTE] "A_UNMUTE", + [AIC_TRN_D_UNMUTE] "D_UNMUTE", + [AIC_TRN_AD_UNMUTE]"AD_UNMUTE", + [AIC_TRN_SUSPEND] "SUSPEND", + [AIC_TRN_EXIT] "EXIT", +}; +#endif + +/* @} */ + +/** \defgroup ds Data Structures */ +/* @{ */ + + +/** +* AIC Command +* These commands do not appear in the register +* set of the device. +*/ +enum __attribute__ ((__packed__)) aic_cmd_id { + AIC_CMD_NOP = 0x80, + AIC_CMD_DELAY, + AIC_CMD_UPDTBITS, + AIC_CMD_WAITBITS, + AIC_CMD_LOCK, + AIC_CMD_BURST, + AIC_CMD_RBURST, + AIC_CMD_LOAD_VAR_IM, + AIC_CMD_LOAD_VAR_ID, + AIC_CMD_STORE_VAR, + AIC_CMD_COND, + AIC_CMD_BRANCH, + AIC_CMD_BRANCH_IM, + AIC_CMD_BRANCH_ID, + AIC_CMD_PRINT, + AIC_CMD_OP_ADD = 0xC0, + AIC_CMD_OP_SUB, + AIC_CMD_OP_MUL, + AIC_CMD_OP_DIV, + AIC_CMD_OP_AND, + AIC_CMD_OP_OR, + AIC_CMD_OP_SHL, + AIC_CMD_OP_SHR, + AIC_CMD_OP_RR, + AIC_CMD_OP_XOR, + AIC_CMD_OP_NOT, + AIC_CMD_OP_LNOT, +}; + +/** +* AIC Delay +* Used for the cmd command delay +* Has one parameter of delay time in ms +*/ +struct aic_cmd_delay { + u16 delay; + enum aic_cmd_id cid; + u8 delay_fine; +}; + +/** +* AIC Lock +* Take codec mutex to avoid clashing with DAPM operations +*/ +struct aic_cmd_lock { + u16 lock; + enum aic_cmd_id cid; + u8 unused; +}; + + +/** + * AIC UPDTBITS, WAITBITS, CHKBITS + * Both these cmd commands have same arguments + * cid will be used to specify which command it is + * has parameters of book, page, offset and mask + */ +struct aic_cmd_bitop { + u16 unused1; + enum aic_cmd_id cid; + u8 mask; +}; + +/** + * AIC CMD Burst header + * Burst writes inside command array + * Followed by burst address, first byte + */ +struct aic_cmd_bhdr { + u16 len; + enum aic_cmd_id cid; + u8 unused; +}; + +/** + * AIC CMD Burst + * Burst writes inside command array + * Followed by data to the extent indicated in previous len + * Can be safely cast to aic_burst + */ +struct aic_cmd_burst { + u8 book; + u8 page; + u8 offset; + u8 data[1]; +}; +#define AIC_CMD_BURST_LEN(n) (2 + ((n) - 1 + 3)/4) + +/** + * AIC CMD Scratch register + * For load + * if (svar != dvar) + * dvar = setbits(svar, mask) // Ignore reg + * else + * dvar = setbits(reg, mask) + * For store + * if (svar != dvar) + * reg = setbits(svar, dvar) + * else + * reg = setbits(svar, mask) + * + */ +struct aic_cmd_ldst { + u8 dvar; + u8 svar; + enum aic_cmd_id cid; + u8 mask; +}; + +/** + * AIC CMD Conditional + * May only precede branch. Followed by nmatch+1 jump + * instructions + * cond = svar&mask + * At each of the following nmatch+1 branch command + * if (cond == match) + * take the branch + */ +struct aic_cmd_cond { + u8 svar; + u8 nmatch; + enum aic_cmd_id cid; + u8 mask; +}; +#define AIC_CMD_COND_LEN(nm) (1 + ((nm)+1)) + +/** + * AIC CMD Goto + * For branch, break, continue and stop + */ +struct aic_cmd_branch { + u16 address; + enum aic_cmd_id cid; + u8 match; +}; + +/** + * AIC Debug print + * For diagnostics + */ +struct aic_cmd_print { + u8 fmtlen; + u8 nargs; + enum aic_cmd_id cid; + char fmt[1]; +}; + +#define AIC_CMD_PRINT_LEN(p) (1 + ((p).fmtlen/4) + (((p).nargs + 3)/4)) +#define AIC_CMD_PRINT_ARG(p) (1 + ((p).fmtlen/4)) + +/** + * AIC Arithmetic and logical operations + * Bit 5 indicates if op1 is indirect + * Bit 6 indicates if op2 is indirect + */ +struct aic_cmd_op { + u8 op1; + u8 op2; + enum aic_cmd_id cid; + u8 dst; +}; +#define AIC_CMD_OP1_ID (1u<<5) +#define AIC_CMD_OP2_ID (1u<<4) + +#define AIC_CMD_OP_START AIC_CMD_OP_ADD +#define AIC_CMD_OP_END (AIC_CMD_OP_LNOT|AIC_CMD_OP1_ID|AIC_CMD_OP2_ID) +#define AIC_CMD_OP_IS_UNARY(x) \ + (((x) == AIC_CMD_OP_NOT) || ((x) == AIC_CMD_OP_LNOT)) + + +/** + * AIC Register + * + * A single reg write + * + */ +union aic_register { + struct { + u8 book; + u8 page; + u8 offset; + u8 data; + }; + u32 bpod; +}; + + + +/** + * AIC Command + * + * Can be a either a + * -# single register write, or + * -# command + * + */ +union aic_cmd { + struct { + u16 unused1; + enum aic_cmd_id cid; + u8 unused2; + }; + union aic_register reg; + struct aic_cmd_delay delay; + struct aic_cmd_lock lock; + struct aic_cmd_bitop bitop; + struct aic_cmd_bhdr bhdr; + struct aic_cmd_burst burst; + struct aic_cmd_ldst ldst; + struct aic_cmd_cond cond; + struct aic_cmd_branch branch; + struct aic_cmd_print print; + u8 print_arg[4]; + struct aic_cmd_op op; +}; + +#define AIC_REG_IS_CMD(x) ((x).cid >= AIC_CMD_DELAY) + +/** + * AIC Block Type + * + * Block identifier + * + */ +enum __attribute__ ((__packed__)) aic_block_t { + AIC_BLOCK_SYSTEM_PRE, + AIC_BLOCK_A_INST, + AIC_BLOCK_A_A_COEF, + AIC_BLOCK_A_B_COEF, + AIC_BLOCK_A_F_COEF, + AIC_BLOCK_D_INST, + AIC_BLOCK_D_A1_COEF, + AIC_BLOCK_D_B1_COEF, + AIC_BLOCK_D_A2_COEF, + AIC_BLOCK_D_B2_COEF, + AIC_BLOCK_D_F_COEF, + AIC_BLOCK_SYSTEM_POST, + AIC_BLOCK_N, + AIC_BLOCK_INVALID, +}; +#define AIC_BLOCK_D_A_COEF AIC_BLOCK_D_A1_COEF +#define AIC_BLOCK_D_B_COEF AIC_BLOCK_D_B1_COEF + +/** + * AIC Block + * + * A block of logically grouped sequences/commands/cmd-commands + * + */ +struct aic_block { + enum aic_block_t type; + int ncmds; + union aic_cmd cmd[]; +}; +#define AIC_BLOCK_SIZE(ncmds) (sizeof(struct aic_block) + \ + ((ncmds)*sizeof(union aic_cmd))) + +/** + * AIC Image + * + * A downloadable image + */ +struct aic_image { + char name[AIC_MAX_ID]; /**< Name of the pfw/overlay/configuration*/ + char *desc; /**< User string*/ + int mute_flags; + struct aic_block *block[AIC_BLOCK_N]; +}; + + + +/** + * AIC PLL + * + * PLL configuration sequence and match critirea + */ +struct aic_pll { + char name[AIC_MAX_ID]; /**< Name of the PLL sequence*/ + char *desc; /**< User string*/ + struct aic_block *seq; +}; + +/** + * AIC Control + * + * Run-time control for a process flow + */ +struct aic_control { + char name[AIC_MAX_ID]; /**< Control identifier*/ + char *desc; /**< User string*/ + int mute_flags; + + int min; /**< Min value of control (*100)*/ + int max; /**< Max value of control (*100)*/ + int step; /**< Control step size (*100)*/ + + int imax; /**< Max index into controls array*/ + int ireset; /**< Reset control to defaults*/ + int icur; /**< Last value set*/ + struct aic_block **output; /**< Array of sequences to send*/ +}; + +/** + * Process flow + * + * Complete description of a process flow + */ +struct aic_pfw { + char name[AIC_MAX_ID]; /**< Name of the process flow*/ + char *desc; /**< User string*/ + u32 version; + u8 prb_a; + u8 prb_d; + int novly; /**< Number of overlays (1 or more)*/ + int ncfg; /**< Number of configurations (0 or more)*/ + int nctrl; /**< Number of run-time controls*/ + struct aic_image *base; /**< Base sequence*/ + struct aic_image **ovly_cfg; /**< Overlay and cfg*/ + /**< patches (if any)*/ + struct aic_control **ctrl; /**< Array of run-time controls*/ +}; + +#define AIC_OCFG_NDX(p, o, c) (((o)*(p)->ncfg)+(c)) +/** + * Process transition + * + * Sequence for specific state transisitions within the driver + * + */ +struct aic_transition { + char name[AIC_MAX_ID]; /**< Name of the transition*/ + char *desc; /**< User string*/ + struct aic_block *block; +}; + +/** + * Device audio mode + * + * Link operating modes to process flows, + * configurations and sequences + * + */ +struct aic_mode { + char name[AIC_MAX_ID]; + char *desc; /**< User string*/ + u32 flags; + u8 pfw; + u8 ovly; + u8 cfg; + u8 pll; + struct aic_block *entry; + struct aic_block *exit; +}; + +struct aic_asoc_toc_entry { + char etext[AIC_MAX_ID]; + int mode; + int cfg; +}; + +struct aic_asoc_toc { + int nentries; + struct aic_asoc_toc_entry entry[]; +}; + +/** + * AIC Project + * + * Top level structure describing the AIC project + */ +struct aic_project { + u32 magic; /**< magic number for identifying F/W file*/ + u32 if_id; /**< Interface match code */ + u32 size; /**< Total size of the firmware (including this header)*/ + u32 cksum; /**< CRC32 of the pickled firmware */ + u32 version; /**< Firmware version (from CFD file)*/ + u32 tstamp; /**< Time stamp of firmware build (epoch seconds)*/ + char name[AIC_MAX_ID]; /**< Project name*/ + char *desc; /**< User string*/ + enum aic_dfamily dfamily; /**< Device family*/ + enum aic_device device; /**< Device identifier*/ + u32 flags; /**< AIC flags*/ + + struct aic_transition **transition; /**< Transition sequences*/ + + u16 npll; /**< Number of PLL settings*/ + struct aic_pll **pll; /**< PLL settings*/ + + u16 npfw; /**< Number of process flows*/ + struct aic_pfw **pfw; /**< Process flows*/ + + u16 nmode; /**< Number of operating modes*/ + struct aic_mode **mode; /**< Modes*/ + + struct aic_asoc_toc *asoc_toc; /**< list of amixer controls*/ +}; + + +/* @} */ + +/* **AIC_INTERFACE_ID=0x3FA6D547** */ + +#endif /* AIC_FIRMWARE_H_ */ diff --git a/sound/soc/codecs/aic3xxx_cfw_ops.c b/sound/soc/codecs/aic3xxx_cfw_ops.c new file mode 100644 index 0000000..79b975b --- /dev/null +++ b/sound/soc/codecs/aic3xxx_cfw_ops.c @@ -0,0 +1,989 @@ +/* + * linux/sound/soc/codecs/aic3xxx/aic3xxx_cfw_ops.c + * + * Copyright (C) 2011 Texas Instruments Inc., + * + * This package is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * THIS PACKAGE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + * + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <linux/slab.h> +#include <sound/tlv.h> + +#include "aic3xxx_cfw.h" +#include "aic3xxx_cfw_ops.h" + +#define aic_warn(_cfw, fmt, ...)\ + dev_warn(_cfw->dev, "AIC: " fmt, ##__VA_ARGS__) +#define aic_error(_cfw, fmt, ...)\ + dev_err(_cfw->dev, "AIC: " fmt, ##__VA_ARGS__) +#define aic_dbg(_cfw, fmt, ...)\ + dev_dbg(_cfw->dev, "AIC: " fmt, ##__VA_ARGS__) + +#if defined(CONFIG_MFD_AIC3111) || defined(CONFIG_AIC3111_CORE) + +# define AIC3XXX_AIC_DEVICE "aic3111_cfw" +#elif defined(CONFIG_MFD_AIC3256) || defined(CONFIG_AIC3256_CORE) + +# define AIC3XXX_AIC_DEVICE "aic3256_cfw" +#elif defined(CONFIG_MFD_AIC262) || defined(CONFIG_AIC3262_CORE) +# define AIC3XXX_AIC_DEVICE "aic3262_cfw" +#else +# define AIC3XXX_AIC_DEVICE "aic3xxx_cfw" +#endif + +/* **Code beyond this point is compilable on host** */ + +/* + * Firmware version numbers are used to make sure that the + * host and target code stay in sync. It is _not_ recommended + * to provide this number from the outside (E.g., from a makefile) + * Instead, a set of automated tools are relied upon to keep the numbers + * in sync at the time of host testing. + */ +#undef AIC_FW_IF_ID +#define AIC_FW_IF_ID 0x3FA6D547 +static int aic_dlimage(struct aic_state *ps, struct aic_image *pim); +static int aic_dlcfg(struct aic_state *ps, struct aic_image *pim); +static int aic_dlctl(struct aic_state *ps, struct aic_block *pb, + u32 mute_flags); + +static void aic_dlcmds(struct aic_state *ps, struct aic_block *pb); +static int aic_set_mode_id(struct aic_state *ps); +static int aic_mute(struct aic_state *ps, int mute, u32 flags); +static int aic_setmode_cfg_u(struct aic_state *ps, int mode, int cfg); +static int aic_setcfg_u(struct aic_state *ps, int cfg); +static int aic_transition_u(struct aic_state *ps, char *ttype); +static int aic_set_pll_u(struct aic_state *ps, int asi); +static int aic_control_u(struct aic_state *ps, char *cname, int param); +static struct aic_project *aic_unpickle(void *pcfw, int n); + +static void aic_wait(struct aic_state *ps, unsigned int reg, u8 mask, + u8 data); +static void aic_set_bits(u8 *data, u8 mask, u8 val); + +int aic_init(struct aic_state *ps, const struct aic3xxx_codec_ops *ops, + struct snd_soc_codec *codec) +{ + ps->ops = ops; + ps->codec = codec; + ps->pjt = NULL; + mutex_init(&ps->mutex); + + return 0; +} +EXPORT_SYMBOL_GPL(aic_init); + +int aic_lock(struct aic_state *ps, int lock) +{ + if (lock) + mutex_lock(&ps->mutex); + else + mutex_unlock(&ps->mutex); + return 0; +} +EXPORT_SYMBOL_GPL(aic_lock); + +int aic_reload(struct aic_state *ps, void *pcfw, int n) +{ + ps->pjt = aic_unpickle(pcfw, n); + ps->cur_mode_id = + ps->cur_mode = ps->cur_pll = ps->cur_pfw = + ps->cur_ovly = ps->cur_cfg = -1; + if (ps->pjt == NULL) { + aic_error(ps, "version mismatch\n"); + return -EINVAL; + } + return 0; +} +EXPORT_SYMBOL_GPL(aic_reload); + +int aic_setmode(struct aic_state *ps, int mode) +{ + struct aic_project *pjt; + int ret; + + aic_lock(ps, 1); + pjt = ps->pjt; + if (pjt == NULL) { + aic_lock(ps, 0); + return -EINVAL; + } + ret = aic_setmode_cfg_u(ps, mode, pjt->mode[mode]->cfg); + aic_lock(ps, 0); + return ret; +} + +int aic_setcfg(struct aic_state *ps, int cfg) +{ + int ret; + + aic_lock(ps, 1); + ret = aic_setcfg_u(ps, cfg); + aic_lock(ps, 0); + return ret; +} + +static int aic_setcfg_u(struct aic_state *ps, int cfg) +{ + struct aic_project *pjt = ps->pjt; + struct aic_pfw *pfw; + struct aic_image *patch; + + if (pjt == NULL) + return -EINVAL; + if (ps->cur_pfw < 0 || ps->cur_pfw >= pjt->npfw) + return -EINVAL; /* Non miniDSP */ + if (ps->cur_cfg == cfg) + return 0; + pfw = pjt->pfw[ps->cur_pfw]; + if (pfw->ncfg == 0 && cfg != 0) + return -EINVAL; + if (cfg > 0 && cfg >= pfw->ncfg) + return -EINVAL; + ps->cur_cfg = cfg; + aic_set_mode_id(ps); + patch = + pfw->ovly_cfg[AIC_OCFG_NDX(pfw, ps->cur_ovly, ps->cur_cfg)]; + if (pfw->ncfg != 0) + return aic_dlcfg(ps, patch); + return 0; +} + +int aic_setmode_cfg(struct aic_state *ps, int mode, int cfg) +{ + int ret; + + aic_lock(ps, 1); + ret = aic_setmode_cfg_u(ps, mode, cfg); + aic_lock(ps, 0); + return ret; +} +EXPORT_SYMBOL_GPL(aic_setmode_cfg); + +static int aic_setmode_cfg_u(struct aic_state *ps, int mode, int cfg) +{ + struct aic_project *pjt = ps->pjt; + struct aic_mode *pmode; + int which = 0, ocndx; + + if (pjt == NULL) + goto err; + if ((mode < 0) || (mode >= pjt->nmode)) + goto err; + if (cfg < 0) + goto err; + if (mode == ps->cur_mode) + return aic_setcfg_u(ps, cfg); + + /* Apply exit sequence for previous mode if present */ + if (ps->cur_mode >= 0) + aic_dlcmds(ps, pjt->mode[ps->cur_mode]->exit); + pmode = pjt->mode[mode]; + if (pjt->mode[mode]->pfw < pjt->npfw) { /* New mode uses miniDSP */ + struct aic_image *im; + struct aic_pfw *pfw = pjt->pfw[pmode->pfw]; + + /* Make sure cfg is valid and supported in this mode */ + if (pfw->ncfg == 0 && cfg != 0) + goto err; + if (cfg > 0 && cfg >= pfw->ncfg) + goto err; + + /* + * Decisions about which miniDSP to stop/restart are taken + * on the basis of sections present in the _base_ image + * This allows for correct sync mode operation even in cases + * where the base PFW uses both miniDSPs where a particular + * overlay applies only to one + */ + im = pfw->base; + if (im->block[AIC_BLOCK_A_INST]) + which |= AIC3XXX_COPS_MDSP_A; + if (im->block[AIC_BLOCK_D_INST]) + which |= AIC3XXX_COPS_MDSP_D; + + if (pmode->pfw != ps->cur_pfw) { + + /* New mode requires different PFW */ + ps->cur_pfw = pmode->pfw; + ps->cur_ovly = 0; + ps->cur_cfg = 0; + + which = ps->ops->stop(ps->codec, which); + aic_dlimage(ps, im); + if (pmode->ovly && pmode->ovly < pfw->novly) { + + /* New mode uses ovly */ + ocndx = AIC_OCFG_NDX(pfw, pmode->ovly, cfg); + aic_dlimage(ps, + pfw->ovly_cfg[ocndx]); + } else if (pfw->ncfg > 0) { + + /* new mode needs only a cfg change */ + ocndx = AIC_OCFG_NDX(pfw, 0, cfg); + aic_dlimage(ps, + pfw->ovly_cfg[ocndx]); + } + ps->ops->restore(ps->codec, which); + + } else if (pmode->ovly != ps->cur_ovly) { + + /* New mode requires only an ovly change */ + ocndx = AIC_OCFG_NDX(pfw, pmode->ovly, cfg); + which = ps->ops->stop(ps->codec, which); + aic_dlimage(ps, pfw->ovly_cfg[ocndx]); + ps->ops->restore(ps->codec, which); + } else if (pfw->ncfg > 0 && cfg != ps->cur_cfg) { + + /* New mode requires only a cfg change */ + ocndx = AIC_OCFG_NDX(pfw, pmode->ovly, cfg); + aic_dlcfg(ps, pfw->ovly_cfg[ocndx]); + } + ps->cur_ovly = pmode->ovly; + ps->cur_cfg = cfg; + + ps->cur_mode = mode; + aic_set_pll_u(ps, 0); + + } else if (pjt->mode[mode]->pfw != 0xFF) { + + /* Not bypass mode */ + aic_warn(ps, "Bad pfw setting detected (%d). Max pfw=%d", + pmode->pfw, pjt->npfw); + } + ps->cur_mode = mode; + aic_set_mode_id(ps); + + /* Transition to netural mode */ + aic_transition_u(ps, "NEUTRAL"); + + /* Apply entry sequence if present */ + aic_dlcmds(ps, pmode->entry); + + aic_dbg(ps, "setmode_cfg: DONE (mode=%d pfw=%d ovly=%d cfg=%d)", + ps->cur_mode, ps->cur_pfw, ps->cur_ovly, ps->cur_cfg); + return 0; + +err: + aic_dbg(ps, "Failed to set firmware mode"); + return -EINVAL; +} + +int aic_transition(struct aic_state *ps, char *ttype) +{ + int ret; + + aic_lock(ps, 1); + ret = aic_transition_u(ps, ttype); + aic_lock(ps, 0); + return ret; +} +EXPORT_SYMBOL_GPL(aic_transition); + +static int aic_transition_u(struct aic_state *ps, char *ttype) +{ + int i; + + if (ps->pjt == NULL) + return -EINVAL; + for (i = 0; i < AIC_TRN_N; ++i) { + if (!strcasecmp(ttype, aic_transition_id[i])) { + struct aic_transition *pt = ps->pjt->transition[i]; + aic_dbg(ps, "Sending transition %s[%d]", ttype, i); + if (pt) + aic_dlcmds(ps, pt->block); + return 0; + } + } + aic_warn(ps, "Transition %s not present or invalid", ttype); + return 0; +} + +int aic_set_pll(struct aic_state *ps, int asi) +{ + int ret; + + aic_lock(ps, 1); + ret = aic_set_pll_u(ps, asi); + aic_lock(ps, 0); + return ret; +} +EXPORT_SYMBOL_GPL(aic_set_pll); + +static int aic_set_pll_u(struct aic_state *ps, int asi) +{ + struct aic_project *pjt = ps->pjt; + int pll_id; + + if (pjt == NULL) + return -EINVAL; + if (ps->cur_mode < 0) + return -EINVAL; + pll_id = pjt->mode[ps->cur_mode]->pll; + if (ps->cur_pll != pll_id) { + aic_dbg(ps, "Re-configuring PLL: %s==>%d", + pjt->pll[pll_id]->name, + pll_id); + aic_dlcmds(ps, pjt->pll[pll_id]->seq); + ps->cur_pll = pll_id; + } + return 0; +} + +int aic_control(struct aic_state *ps, char *cname, int param) +{ + int ret; + + aic_lock(ps, 1); + ret = aic_control_u(ps, cname, param); + aic_lock(ps, 0); + return ret; +} + +static int aic_control_u(struct aic_state *ps, char *cname, int param) +{ + struct aic_pfw *pfw; + int i; + + if (ps->cur_pfw < 0 || ps->cur_pfw >= ps->pjt->npfw) { + aic_warn(ps, "Not in MiniDSP mode"); + return 0; + } + pfw = ps->pjt->pfw[ps->cur_pfw]; + for (i = 0; i < pfw->nctrl; ++i) { + struct aic_control *pc = pfw->ctrl[i]; + if (strcasecmp(cname, pfw->ctrl[i]->name)) + continue; + if (param < 0 || param > pc->imax) { + aic_warn(ps, "Parameter out of range\n"); + return -EINVAL; + } + aic_dbg(ps, "Sending control %s[%d]", cname, param); + pc->icur = param; + aic_dlctl(ps, pc->output[param], pc->mute_flags); + return 0; + } + aic_warn(ps, "Control named %s not found in pfw %s", cname, pfw->name); + + return -EINVAL; +} + +static void aic_op(struct aic_state *ps, unsigned char *var, + struct aic_cmd_op cmd) +{ + u32 op1, op2; + u32 cid = cmd.cid; + + op1 = cmd.op1; + op2 = cmd.op2; + if (cid & AIC_CMD_OP1_ID) + op1 = var[op1]; + if (cid & AIC_CMD_OP2_ID) + op2 = var[op2]; + cid &= ~(AIC_CMD_OP1_ID | AIC_CMD_OP2_ID); + + switch (cid) { + case AIC_CMD_OP_ADD: + var[cmd.dst] = op1 + op2; + break; + case AIC_CMD_OP_SUB: + var[cmd.dst] = op1 - op2; + break; + case AIC_CMD_OP_MUL: + var[cmd.dst] = op1 * op2; + break; + case AIC_CMD_OP_DIV: + var[cmd.dst] = op1 / op2; + break; + case AIC_CMD_OP_AND: + var[cmd.dst] = op1 & op2; + break; + case AIC_CMD_OP_OR: + var[cmd.dst] = op1 | op2; + break; + case AIC_CMD_OP_SHL: + var[cmd.dst] = (op1 << op2); + break; + case AIC_CMD_OP_SHR: + var[cmd.dst] = (op1 >> op2); + break; + case AIC_CMD_OP_RR: + while (op2--) + var[cmd.dst] = (op1 >> 1) | ((op1 & 1) << 7); + break; + case AIC_CMD_OP_XOR: + var[cmd.dst] = op1 ^ op2; + break; + case AIC_CMD_OP_NOT: + var[cmd.dst] = ~op1; + break; + case AIC_CMD_OP_LNOT: + var[cmd.dst] = !op1; + break; + default: + break; + } +} + +static void aic_dlcmds(struct aic_state *ps, struct aic_block *pb) +{ + int pc = 0, cond = 0; + unsigned char var[256]; + + if (!pb) + return; + while (pc < pb->ncmds) { + union aic_cmd *c = &(pb->cmd[pc]); + if (c->cid != AIC_CMD_BRANCH_IM && + c->cid != AIC_CMD_BRANCH_ID && c->cid != AIC_CMD_NOP) + cond = 0; + switch (c->cid) { + case 0 ... (AIC_CMD_NOP - 1): + ps->ops->reg_write(ps->codec, c->reg.bpod, + c->reg.data); + pc += 1; + break; + case AIC_CMD_NOP: + pc += 1; + break; + case AIC_CMD_DELAY: + mdelay(c->delay.delay); + pc += 1; + break; + case AIC_CMD_UPDTBITS: + ps->ops->set_bits(ps->codec, c[1].reg.bpod, + c->bitop.mask, c[1].reg.data); + pc += 2; + break; + case AIC_CMD_WAITBITS: + aic_wait(ps, c[1].reg.bpod, c->bitop.mask, + c[1].reg.data); + pc += 2; + break; + case AIC_CMD_LOCK: + if (c->delay.delay) + ps->ops->lock(ps->codec); + else + ps->ops->unlock(ps->codec); + pc += 1; + break; + case AIC_CMD_BURST: + ps->ops->bulk_write(ps->codec, c[1].reg.bpod, + c->bhdr.len, c[1].burst.data); + pc += AIC_CMD_BURST_LEN(c->bhdr.len); + break; + case AIC_CMD_RBURST: + ps->ops->bulk_read(ps->codec, c[1].reg.bpod, + c->bhdr.len, c[1].burst.data); + pc += AIC_CMD_BURST_LEN(c->bhdr.len); + break; + case AIC_CMD_LOAD_VAR_IM: + aic_set_bits(&var[c->ldst.dvar], + c->ldst.mask, c->ldst.svar); + pc += 1; + break; + case AIC_CMD_LOAD_VAR_ID: + if (c->ldst.svar != c->ldst.dvar) { + aic_set_bits(&var[c->ldst.dvar], + c->ldst.mask, + var[c->ldst.svar]); + pc += 1; + } else { + u8 data; + data = ps->ops->reg_read(ps->codec, + c[1].reg.bpod); + aic_set_bits(&var[c->ldst.dvar], + c->ldst.mask, data); + pc += 2; + } + break; + case AIC_CMD_STORE_VAR: + if (c->ldst.svar != c->ldst.dvar) + ps->ops->set_bits(ps->codec, + c[1].reg.bpod, + var[c->ldst.dvar], + var[c->ldst.svar]); + else + ps->ops->set_bits(ps->codec, + c[1].reg.bpod, + c->ldst.mask, + var[c->ldst.svar]); + pc += 2; + break; + case AIC_CMD_COND: + cond = var[c->cond.svar] & c->cond.mask; + pc += 1; + break; + case AIC_CMD_BRANCH: + pc = c->branch.address; + break; + case AIC_CMD_BRANCH_IM: + if (c->branch.match == cond) + pc = c->branch.address; + else + pc += 1; + break; + case AIC_CMD_BRANCH_ID: + if (var[c->branch.match] == cond) + pc = c->branch.address; + else + pc += 1; + break; + case AIC_CMD_PRINT: + { + union aic_cmd *parglist = + c + AIC_CMD_PRINT_ARG(c->print); + printk(c->print.fmt, + var[parglist->print_arg[0]], + var[parglist->print_arg[1]], + var[parglist->print_arg[2]], + var[parglist->print_arg[3]]); + pc += AIC_CMD_PRINT_LEN(c->print); + } + break; + case AIC_CMD_OP_START ... AIC_CMD_OP_END: + aic_op(ps, var, c->op); + pc += 1; + break; + default: + aic_warn(ps, "Unknown cmd command %x. Skipped", c->cid); + pc += 1; + break; + } + } +} + +static void aic_wait(struct aic_state *ps, unsigned int reg, u8 mask, + u8 data) +{ + while ((ps->ops->reg_read(ps->codec, reg) & mask) != data) + usleep_range(500, 1000); +} + +static inline void aic_set_bits(u8 *data, u8 mask, u8 val) +{ + *data = (*data & (~mask)) | (val & mask); +} + +static const struct { + u32 mdsp; + int buf_a, buf_b; + u32 swap; +} csecs[] = { + { + .mdsp = AIC3XXX_COPS_MDSP_A, + .swap = AIC3XXX_ABUF_MDSP_A, + .buf_a = AIC_BLOCK_A_A_COEF, + .buf_b = AIC_BLOCK_A_B_COEF + }, + { + .mdsp = AIC3XXX_COPS_MDSP_D, + .swap = AIC3XXX_ABUF_MDSP_D1, + .buf_a = AIC_BLOCK_D_A1_COEF, + .buf_b = AIC_BLOCK_D_B1_COEF + }, + { + .mdsp = AIC3XXX_COPS_MDSP_D, + .swap = AIC3XXX_ABUF_MDSP_D2, + .buf_a = AIC_BLOCK_D_A2_COEF, + .buf_b = AIC_BLOCK_D_B2_COEF + }, +}; +static int aic_dlctl(struct aic_state *ps, struct aic_block *pb, + u32 mute_flags) +{ + int i, btype = pb->type; + int run_state = ps->ops->lock(ps->codec); + + aic_dbg(ps, "Download CTL"); + for (i = 0; i < sizeof(csecs) / sizeof(csecs[0]); ++i) { + if (csecs[i].buf_a != btype && csecs[i].buf_b != btype) + continue; + aic_dbg(ps, "\tDownload once to %d\n", btype); + aic_dlcmds(ps, pb); + if (run_state & csecs[i].mdsp) { + aic_dbg(ps, "Download again to make sure it reaches B"); + aic_mute(ps, 1, run_state & mute_flags); + ps->ops->bswap(ps->codec, csecs[i].swap); + aic_mute(ps, 0, run_state & mute_flags); + aic_dlcmds(ps, pb); + } + break; + } + ps->ops->unlock(ps->codec); + return 0; +} + +static int aic_dlcfg(struct aic_state *ps, struct aic_image *pim) +{ + int i, run_state, swap; + + aic_dbg(ps, "Download CFG %s", pim->name); + run_state = ps->ops->lock(ps->codec); + swap = 0; + for (i = 0; i < sizeof(csecs) / sizeof(csecs[0]); ++i) { + if (!pim->block[csecs[i].buf_a]) + continue; + aic_dlcmds(ps, pim->block[csecs[i].buf_a]); + aic_dlcmds(ps, pim->block[csecs[i].buf_b]); + if (run_state & csecs[i].mdsp) + swap |= csecs[i].swap; + } + if (swap) { + aic_mute(ps, 1, run_state & pim->mute_flags); + ps->ops->bswap(ps->codec, swap); + aic_mute(ps, 0, run_state & pim->mute_flags); + for (i = 0; i < sizeof(csecs) / sizeof(csecs[0]); ++i) { + if (!pim->block[csecs[i].buf_a]) + continue; + if (!(run_state & csecs[i].mdsp)) + continue; + aic_dlcmds(ps, pim->block[csecs[i].buf_a]); + aic_dlcmds(ps, pim->block[csecs[i].buf_b]); + } + } + ps->ops->unlock(ps->codec); + return 0; +} + +static int aic_dlimage(struct aic_state *ps, struct aic_image *pim) +{ + int i; + + if (!pim) + return 0; + aic_dbg(ps, "Download IMAGE %s", pim->name); + for (i = 0; i < AIC_BLOCK_N; ++i) + aic_dlcmds(ps, pim->block[i]); + return 0; +} + +static int aic_mute(struct aic_state *ps, int mute, u32 flags) +{ + if ((flags & AIC3XXX_COPS_MDSP_D) && (flags & AIC3XXX_COPS_MDSP_A)) + aic_transition_u(ps, + mute ? "AD_MUTE" : "AD_UNMUTE"); + else if (flags & AIC3XXX_COPS_MDSP_D) + aic_transition_u(ps, mute ? "D_MUTE" : "D_UNMUTE"); + else if (flags & AIC3XXX_COPS_MDSP_A) + aic_transition_u(ps, mute ? "A_MUTE" : "A_UNMUTE"); + return 0; +} + +static inline void *aic_ndx2ptr(void *p, u8 *base) +{ + return &base[(int)p]; +} +static inline char *aic_desc(void *p, u8 *base) +{ + if (p) + return aic_ndx2ptr(p, base); + return NULL; +} + +static void aic_unpickle_image(struct aic_image *im, void *p) +{ + int i; + + im->desc = aic_desc(im->desc, p); + for (i = 0; i < AIC_BLOCK_N; ++i) + if (im->block[i]) + im->block[i] = aic_ndx2ptr(im->block[i], p); +} + +static void aic_unpickle_control(struct aic_control *ct, void *p) +{ + int i; + + ct->output = aic_ndx2ptr(ct->output, p); + ct->desc = aic_desc(ct->desc, p); + for (i = 0; i <= ct->imax; ++i) + ct->output[i] = aic_ndx2ptr(ct->output[i], p); +} + +static struct aic_project *aic_unpickle(void *p, int n) +{ + struct aic_project *pjt = p; + int i, j; + + if (pjt->magic != AIC_FW_MAGIC || pjt->size != n || + pjt->if_id != AIC_FW_IF_ID) { + return NULL; + } + + pjt->desc = aic_desc(pjt->desc, p); + pjt->transition = aic_ndx2ptr(pjt->transition, p); + for (i = 0; i < AIC_TRN_N; i++) { + if (!pjt->transition[i]) + continue; + pjt->transition[i] = aic_ndx2ptr(pjt->transition[i], p); + pjt->transition[i]->desc = aic_desc( + pjt->transition[i]->desc, p); + pjt->transition[i]->block = aic_ndx2ptr( + pjt->transition[i]->block, p); + } + pjt->pll = aic_ndx2ptr(pjt->pll, p); + for (i = 0; i < pjt->npll; i++) { + pjt->pll[i] = aic_ndx2ptr(pjt->pll[i], p); + pjt->pll[i]->desc = aic_desc(pjt->pll[i]->desc, p); + pjt->pll[i]->seq = aic_ndx2ptr(pjt->pll[i]->seq, p); + } + + pjt->pfw = aic_ndx2ptr(pjt->pfw, p); + for (i = 0; i < pjt->npfw; i++) { + pjt->pfw[i] = aic_ndx2ptr(pjt->pfw[i], p); + pjt->pfw[i]->desc = aic_desc(pjt->pfw[i]->desc, p); + if (pjt->pfw[i]->base) { + pjt->pfw[i]->base = aic_ndx2ptr( + pjt->pfw[i]->base, p); + aic_unpickle_image(pjt->pfw[i]->base, p); + } + pjt->pfw[i]->ovly_cfg = aic_ndx2ptr( + pjt->pfw[i]->ovly_cfg, p); + for (j = 0; j < pjt->pfw[i]->novly * pjt->pfw[i]->ncfg; ++j) { + pjt->pfw[i]->ovly_cfg[j] = aic_ndx2ptr( + pjt->pfw[i]->ovly_cfg[j], p); + aic_unpickle_image(pjt->pfw[i]->ovly_cfg[j], p); + } + if (pjt->pfw[i]->nctrl) + pjt->pfw[i]->ctrl = aic_ndx2ptr( + pjt->pfw[i]->ctrl, p); + for (j = 0; j < pjt->pfw[i]->nctrl; ++j) { + pjt->pfw[i]->ctrl[j] = aic_ndx2ptr( + pjt->pfw[i]->ctrl[j], p); + aic_unpickle_control(pjt->pfw[i]->ctrl[j], p); + } + } + + pjt->mode = aic_ndx2ptr(pjt->mode, p); + for (i = 0; i < pjt->nmode; i++) { + pjt->mode[i] = aic_ndx2ptr(pjt->mode[i], p); + pjt->mode[i]->desc = aic_desc(pjt->mode[i]->desc, p); + if (pjt->mode[i]->entry) + pjt->mode[i]->entry = aic_ndx2ptr( + pjt->mode[i]->entry, p); + if (pjt->mode[i]->exit) + pjt->mode[i]->exit = aic_ndx2ptr( + pjt->mode[i]->exit, p); + } + if (pjt->asoc_toc) + pjt->asoc_toc = aic_ndx2ptr(pjt->asoc_toc, p); + else + return NULL; + + return pjt; +} +static int aic_set_mode_id(struct aic_state *ps) +{ + struct aic_asoc_toc *toc = ps->pjt->asoc_toc; + int i; + + for (i = 0; i < toc->nentries; ++i) { + if (toc->entry[i].cfg == ps->cur_cfg && + toc->entry[i].mode == ps->cur_mode) { + ps->cur_mode_id = i; + return 0; + } + } + aic_dbg(ps, "Unknown mode,cfg combination [%d,%d]", ps->cur_mode, + ps->cur_cfg); + return -EINVAL; +} + +/* **Code beyond this point is not compilable on host** */ + +static int aic3xxx_get_control(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct aic_state *ps = (struct aic_state *)kcontrol->private_value; + struct aic_pfw *pfw; + int i; + + if (ps->cur_pfw >= ps->pjt->npfw) { + aic_dbg(ps, "Not in MiniDSP mode"); + return 0; + } + pfw = ps->pjt->pfw[ps->cur_pfw]; + for (i = 0; i < pfw->nctrl; ++i) { + if (!strcasecmp(kcontrol->id.name, pfw->ctrl[i]->name)) { + struct aic_control *pc = pfw->ctrl[i]; + ucontrol->value.integer.value[0] = pc->icur; + return 0; + } + } + return 0; +} + +static int aic3xxx_put_control(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct aic_state *ps = (struct aic_state *)kcontrol->private_value; + + aic_control(ps, kcontrol->id.name, + ucontrol->value.integer.value[0]); + return 0; +} + +static int aic3xxx_info_control(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *ucontrol) +{ + struct aic_state *ps = (struct aic_state *)kcontrol->private_value; + struct aic_pfw *pfw; + int i; + + if (ps->cur_pfw >= ps->pjt->npfw) { + aic_dbg(ps, "Not in MiniDSP mode"); + return 0; + } + pfw = ps->pjt->pfw[ps->cur_pfw]; + for (i = 0; i < pfw->nctrl; ++i) { + if (!strcasecmp(kcontrol->id.name, pfw->ctrl[i]->name)) { + struct aic_control *pc = pfw->ctrl[i]; + ucontrol->value.integer.min = 0; + ucontrol->value.integer.max = pc->imax; + if (pc->imax == 1) + ucontrol->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + ucontrol->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + } + } + + ucontrol->count = 1; + return 0; +} +int aic_add_controls(struct snd_soc_codec *codec, struct aic_state *ps) +{ + int i, j; + struct aic_pfw *pfw; + + for (j = 0; j < ps->pjt->npfw; ++j) { + pfw = ps->pjt->pfw[j]; + + for (i = 0; i < pfw->nctrl; ++i) { + struct aic_control *pc = pfw->ctrl[i]; + struct snd_kcontrol_new *generic_control = + devm_kzalloc(codec->dev, + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + unsigned int *tlv_array = + devm_kzalloc(codec->dev, + 4 * sizeof(unsigned int), GFP_KERNEL); + + if (generic_control == NULL) + return -ENOMEM; + generic_control->access = + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_READWRITE; + tlv_array[0] = SNDRV_CTL_TLVT_DB_SCALE; + tlv_array[1] = 2 * sizeof(unsigned int); + tlv_array[2] = pc->min; + tlv_array[3] = ((pc->step) & TLV_DB_SCALE_MASK); + if (pc->step > 0) + generic_control->tlv.p = tlv_array; + generic_control->name = pc->name; + generic_control->private_value = (unsigned long) ps; + generic_control->get = aic3xxx_get_control; + generic_control->put = aic3xxx_put_control; + generic_control->info = aic3xxx_info_control; + generic_control->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + snd_soc_add_codec_controls(codec, generic_control, 1); + aic_dbg(ps, "Added control %s", pc->name); + } + } + return 0; + +} +EXPORT_SYMBOL_GPL(aic_add_controls); + +static int aic3xxx_get_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_enum *e = (struct soc_enum *) kcontrol->private_value; + struct aic_state *ps = (struct aic_state *) e->mask; + + ucontrol->value.enumerated.item[0] = ps->cur_mode_id; + + return 0; +} + +static int aic3xxx_put_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_enum *e = (struct soc_enum *) kcontrol->private_value; + struct aic_state *ps = (struct aic_state *) e->mask; + struct aic_asoc_toc *toc; + int index, ret; + + aic_lock(ps, 1); + toc = ps->pjt->asoc_toc; + + index = ucontrol->value.enumerated.item[0]; + if (index < 0 || index >= toc->nentries) { + aic_lock(ps, 0); + return -EINVAL; + } + ret = aic_setmode_cfg_u(ps, toc->entry[index].mode, + toc->entry[index].cfg); + aic_lock(ps, 0); + return ret; +} +EXPORT_SYMBOL_GPL(aic3xxx_put_mode); + +int aic_add_modes(struct snd_soc_codec *codec, struct aic_state *ps) +{ + int j; + struct aic_asoc_toc *toc = ps->pjt->asoc_toc; + struct soc_enum *mode_cfg_enum = + devm_kzalloc(codec->dev, sizeof(struct soc_enum), GFP_KERNEL); + struct snd_kcontrol_new *mode_cfg_control = + devm_kzalloc(codec->dev, sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + + if (mode_cfg_enum == NULL) + goto mem_err; + if (mode_cfg_control == NULL) + goto mem_err; + + mode_cfg_enum->dtexts = devm_kzalloc(codec->dev, + toc->nentries * sizeof(char *), + GFP_KERNEL); + if (mode_cfg_enum->dtexts == NULL) + goto mem_err; + + for (j = 0; j < toc->nentries; j++) + mode_cfg_enum->dtexts[j] = toc->entry[j].etext; + + mode_cfg_enum->reg = j; + mode_cfg_enum->max = toc->nentries; + mode_cfg_enum->mask = (unsigned int) ps; + mode_cfg_control->name = "Codec Firmware Setmode"; + mode_cfg_control->get = aic3xxx_get_mode; + mode_cfg_control->put = aic3xxx_put_mode; + mode_cfg_control->info = snd_soc_info_enum_ext; + mode_cfg_control->private_value = (unsigned long) mode_cfg_enum; + mode_cfg_control->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + snd_soc_add_codec_controls(codec, mode_cfg_control, 1); + return 0; +mem_err: + kfree(mode_cfg_control); + kfree(mode_cfg_enum); + kfree(mode_cfg_enum->texts); + return -ENOMEM; + +} +EXPORT_SYMBOL_GPL(aic_add_modes); + +MODULE_AUTHOR("Hari Rajagopala <harik(a)ti.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/aic3xxx_cfw_ops.h b/sound/soc/codecs/aic3xxx_cfw_ops.h new file mode 100644 index 0000000..6a600fb --- /dev/null +++ b/sound/soc/codecs/aic3xxx_cfw_ops.h @@ -0,0 +1,98 @@ +/* + * aic3xxx_cfw_ops.h -- SoC audio for TI OMAP44XX SDP + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef AIC3XXX_CFW_OPS_H_ +#define AIC3XXX_CFW_OPS_H_ + +#include <linux/mutex.h> +#include <sound/soc.h> +#include <linux/cdev.h> + +struct aic_project; +struct aic3xxx_codec_ops; + +struct aic_state { + const char *dev_name; + int version; + struct aic_project *pjt; + const struct aic3xxx_codec_ops *ops; + struct snd_soc_codec *codec; + struct device *dev; + struct mutex mutex; + int cur_mode_id; + int cur_pll; + int cur_mode; + int cur_pfw; + int cur_ovly; + int cur_cfg; + struct cdev cdev; + int is_open; +}; + +int aic_init(struct aic_state *ps, const struct aic3xxx_codec_ops *ops, + struct snd_soc_codec *codec); +int aic_lock(struct aic_state *ps, int lock); +int aic_reload(struct aic_state *ps, void *pcfw, int n); +int aic_setmode(struct aic_state *ps, int mode); +int aic_setmode_cfg(struct aic_state *ps, int mode, int cfg); +int aic_setcfg(struct aic_state *ps, int cfg); +int aic_transition(struct aic_state *ps, char *ttype); +int aic_set_pll(struct aic_state *ps, int asi); +int aic_control(struct aic_state *ps, char *cname, int param); +int aic_add_controls(struct snd_soc_codec *codec, struct aic_state *ps); +int aic_add_modes(struct snd_soc_codec *codec, struct aic_state *ps); + + +#define AIC3XXX_COPS_MDSP_D_L (0x00000002u) +#define AIC3XXX_COPS_MDSP_D_R (0x00000001u) +#define AIC3XXX_COPS_MDSP_D (AIC3XXX_COPS_MDSP_D_L|AIC3XXX_COPS_MDSP_D_R) + +#define AIC3XXX_COPS_MDSP_A_L (0x00000020u) +#define AIC3XXX_COPS_MDSP_A_R (0x00000010u) +#define AIC3XXX_COPS_MDSP_A (AIC3XXX_COPS_MDSP_A_L|AIC3XXX_COPS_MDSP_A_R) + +#define AIC3XXX_COPS_MDSP_ALL (AIC3XXX_COPS_MDSP_D|AIC3XXX_COPS_MDSP_A) + +#define AIC3XXX_ABUF_MDSP_D1 (0x00000001u) +#define AIC3XXX_ABUF_MDSP_D2 (0x00000002u) +#define AIC3XXX_ABUF_MDSP_A (0x00000010u) +#define AIC3XXX_ABUF_MDSP_ALL \ + (AIC3XXX_ABUF_MDSP_D1|AIC3XXX_ABUF_MDSP_D2|AIC3XXX_ABUF_MDSP_A) + + +struct aic3xxx_codec_ops { + int (*reg_read) (struct snd_soc_codec *codec, unsigned int reg); + int (*reg_write) (struct snd_soc_codec *codec, unsigned int reg, + unsigned char val); + int (*set_bits) (struct snd_soc_codec *codec, unsigned int reg, + unsigned char mask, unsigned char val); + int (*bulk_read) (struct snd_soc_codec *codec, unsigned int reg, + int count, u8 *buf); + int (*bulk_write) (struct snd_soc_codec *codec, unsigned int reg, + int count, const u8 *buf); + int (*lock) (struct snd_soc_codec *codec); + int (*unlock) (struct snd_soc_codec *codec); + int (*stop) (struct snd_soc_codec *codec, int mask); + int (*restore) (struct snd_soc_codec *codec, int runstate); + int (*bswap) (struct snd_soc_codec *codec, int mask); +}; + +MODULE_LICENSE("GPL"); + +#endif -- 1.7.0.4

1 0

[alsa-devel] how to update the hw_ptr in the interrupt?
by Zhang wei 12 Dec '12

12 Dec '12

Hello every one.i am in a trouble and could anyone help me figure out ? I have spent couple days trying to understand ,but I do not know the pricinple clearly which decides how to adjust the hw_ptr in the interrupt routine. Below is the reference sourcecode in the snd_pcm_update_hw_ptr0(). When the DMA interrupt ocurrs the process will enter the snd_pcm_update_hw_ptr0() with the in_interrupt=1. ========================================================== if (in_interrupt) { /* we know that one period was processed */ /* delta = "expected next hw_ptr" for in_interrupt != 0 */ delta = runtime->hw_ptr_interrupt + runtime->period_size; if (delta > new_hw_ptr) { /* check for double acknowledged interrupts */ hdelta = jiffies - runtime->hw_ptr_jiffies; if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) hw_base = 0; new_hw_ptr = hw_base + pos; goto __delta; } } } ========================================================== The question is: what can we get from the result of the compare between delta and new_hw_ptr.? Why we can conclue it? The delta represent the expected location in the range of *boundary* , and it producted based the hw_ptr_interrupt(*not fixed*) and perod_size(fixed)? The new_hw_ptr is producted by runtime->hw_ptr_base plus pos. What Is the relationship between delta and new_hw_ptr? Why we can compare them with each other ? Any tips will be ok. Thanks a lot. Best Regards zw

1 0

[alsa-devel] [PATCH] ASoC: arizona: Implement tristate support
by Mark Brown 12 Dec '12

12 Dec '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 231fae0..a85a68f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -859,11 +859,27 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, return snd_soc_dapm_sync(&codec->dapm); } +static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + int base = dai->driver->base; + unsigned int reg; + + if (tristate) + reg = ARIZONA_AIF1_TRI; + else + reg = 0; + + return snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_TRI, reg); +} + const struct snd_soc_dai_ops arizona_dai_ops = { .startup = arizona_startup, .set_fmt = arizona_set_fmt, .hw_params = arizona_hw_params, .set_sysclk = arizona_dai_set_sysclk, + .set_tristate = arizona_set_tristate, }; EXPORT_SYMBOL_GPL(arizona_dai_ops); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm8993: Refactor set_pll code to avoid GCC warnings
by Fabio Estevam 12 Dec '12

12 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Refactor set_pll code to avoid the following warnings: sound/soc/codecs/wm8983.c:873:40: warning: 'pll_div.k' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm8983.c:870:9: warning: 'pll_div.n' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm8983.c:869:23: warning: 'pll_div.div2' may be used uninitialized in this function [-Wuninitialized] Do the same as in commit 86ce6c9a (ASoC: WM8804: Refactor set_pll code to avoid GCC warnings). Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8983.c | 41 ++++++++++++++++++++++------------------- 1 file changed, 22 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 9fe1e04..e7c73ff 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -851,30 +851,33 @@ static int wm8983_set_pll(struct snd_soc_dai *dai, int pll_id, struct pll_div pll_div; codec = dai->codec; - if (freq_in && freq_out) { + if (!freq_in || !freq_out) { + /* disable the PLL before re-programming it */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, 0); + return 0; + } else { ret = pll_factors(&pll_div, freq_out * 4 * 2, freq_in); if (ret) return ret; - } - - /* disable the PLL before re-programming it */ - snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, - WM8983_PLLEN_MASK, 0); - if (!freq_in || !freq_out) - return 0; + /* disable the PLL before re-programming it */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, 0); + + /* set PLLN and PRESCALE */ + snd_soc_write(codec, WM8983_PLL_N, + (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT) + | pll_div.n); + /* set PLLK */ + snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18)); + /* enable the PLL */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, WM8983_PLLEN); + } - /* set PLLN and PRESCALE */ - snd_soc_write(codec, WM8983_PLL_N, - (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT) - | pll_div.n); - /* set PLLK */ - snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff); - snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18)); - /* enable the PLL */ - snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, - WM8983_PLLEN_MASK, WM8983_PLLEN); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: wm8962: Unconditionally wait for the FLL to lock
by Mark Brown 12 Dec '12

12 Dec '12

If the FLL is being shut down we will exit early so there is no need to check here and in fact we're checking the wrong thing anyway. Reported-by: Graeme Gregory <gg(a)slimlogic.co.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 26 ++++++++++++-------------- 1 file changed, 12 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 09816e8..50b0983 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2875,22 +2875,20 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, ret = 0; - if (fll1 & WM8962_FLL_ENA) { - /* This should be a massive overestimate but go even - * higher if we'll error out - */ - if (wm8962->irq) - timeout = msecs_to_jiffies(5); - else - timeout = msecs_to_jiffies(1); + /* This should be a massive overestimate but go even + * higher if we'll error out + */ + if (wm8962->irq) + timeout = msecs_to_jiffies(5); + else + timeout = msecs_to_jiffies(1); - timeout = wait_for_completion_timeout(&wm8962->fll_lock, - timeout); + timeout = wait_for_completion_timeout(&wm8962->fll_lock, + timeout); - if (timeout == 0 && wm8962->irq) { - dev_err(codec->dev, "FLL lock timed out"); - ret = -ETIMEDOUT; - } + if (timeout == 0 && wm8962->irq) { + dev_err(codec->dev, "FLL lock timed out"); + ret = -ETIMEDOUT; } wm8962->fll_fref = Fref; -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ALSA - Replace msleep() with usleep_range() to save module loading time
by Wang Xingchao 12 Dec '12

12 Dec '12

msleep(1) is not so accurate and may cause almost 15ms delay in azx_reset(). usleep_range() can reduce ~30ms loading time for Intel Haswell which has two HDA controllers. Signed-off-by: Wang Xingchao <wangxingchao2011(a)gmail.com> --- sound/pci/hda/hda_intel.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index eb48109..865df90 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1075,22 +1075,22 @@ static int azx_reset(struct azx *chip, int full_reset) count = 50; while (azx_readb(chip, GCTL) && --count) - msleep(1); + usleep_range(1000, 1500); /* delay for >= 100us for codec PLL to settle per spec * Rev 0.9 section 5.5.1 */ - msleep(1); + usleep_range(1000, 1500); /* Bring controller out of reset */ azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); count = 50; while (!azx_readb(chip, GCTL) && --count) - msleep(1); + usleep_range(1000, 1500); /* Brent Chartrand said to wait >= 540us for codecs to initialize */ - msleep(1); + usleep_range(1000, 1500); __skip: /* check to see if controller is ready */ -- 1.7.9.5

3 2

[alsa-devel] ALSA sound/usb/mixer.c:415 cannot get current value for control 2 ch 1: err = -22
by isaac smith 11 Dec '12

11 Dec '12

Hello, I have the same issue with my HP Envy 15. The sound card in question is an HP Wireless Audio card that only works with HP-branded wireless speakers, so playing to the card produces no output whatsoever, but does hang whatever process tries to access it. I doubt that many people ever use this feature, so unless someone invests the time to get it working I'd like a way to just ignore it completely. I've blacklisted the snd-usb-audio module for now, but I have a USB sound card and a USB MIDI controller that I'd like to be able to use. Here's the output of "lsusb -vd 0424:b832" (apologies for length): Bus 004 Device 004: ID 0424:b832 Standard Microsystems Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x0424 Standard Microsystems Corp. idProduct 0xb832 bcdDevice 75.28 iManufacturer 1 iProduct 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 225 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 3 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 72 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0402 Headset bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 11 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 12 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 1 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 12 bSourceID 2 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 0 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 0 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 48000 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 1 Milliseconds wLockDelay 0 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.10 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 74 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x8a EP 10 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 10 Device Status: 0x060a (Bus Powered) Remote Wakeup Enabled > > On Thu, 6 Dec 2012, Joe Cooper wrote: > > On most HP Envy laptops the snd-usb-audio module causes the system to > > become unresponsive and Gnome Shell 3 to crash. This effects every kernel > > I've tried from 3.3.4 on up to 3.7rc8. It also produces these errors on the > > console: > > > > ALSA sound/usb/mixer.c:415 cannot get current value for control 2 ch 1: err > > = -22 > > > > ALSA sound/usb/clock.c:228 3:1:1: cannot set freq 44100 to ep 0x1 > > > > Blacklisting this module allows the system to function, but with most of > > its speakers not working. It also prevents use of other USB audio devices, > > making multitrack recording impossible. > > > > I have a ticket about it in the Fedora bug tracker, but was referred to > > alsa-devel. That ticket links to other discussions of this bug, as well. > > Here's that ticket: https://bugzilla.redhat.com/show_bug.cgi?id=884368 > > > > alsa-info is here: https://bugzilla.redhat.com/attachment.cgi?id=658989 > > > > Please let me know if you need further details, or if there's anything else > > I can do to help troubleshoot this. > > Hi Joe, > > I can try to help. > > Do you get any sound output at all (even if it stutters or sounds > wrong)? > > Can you please send the output of "lsusb -v -d 0424:b832"? > I think I can work out a patch for you to skip this device, so > you can use other audio usb devices until this gets proper support. > > Cheers, > Eldad

2 1

[alsa-devel] Denon DN-X 1600 Support
by Florian Beginn 11 Dec '12

11 Dec '12

Hello, i just made my first steps fiddling with the alsa-driver and got my DN-X 1600 working. It's a dj-mixer connected via usb, offering 8 channel input/output and midi. Samplerate can be set to either 44.1, 48 or 96khz. It seems to not answer properly on samplerate requests from snd_usb_ctl_msg() so i hardcoded it to 96khz. Regrettably, this is a pretty ugly workaround and somehow broke midi capability, although audio is running fine. I'd be happy if one of you could help me work out how to do this right/better. Find attached the patches and all information i could gather. P.S. I'm not used to git (well, any VCS) and hope the diff format is ok for you. --- a/sound/usb/clock.c 2012-01-25 08:56:49.000000000 +0100 +++ b/sound/usb/clock.c 2012-12-11 17:33:47.685196544 +0100 @@ -255,6 +255,14 @@ int err, crate; int clock = snd_usb_clock_find_source(chip, fmt->clock); + /* + * Denon DN-X 1600 won't respond when asked for samplerate. + * Currently hardcoded to only work with 96khz + */ + + if (chip->usb_id == USB_ID(0x154e, 0x500e) && rate == 96000) + return 0; + if (clock < 0) return clock; --- a/sound/usb/format.c 2012-12-07 17:11:00.000000000 +0100 +++ b/sound/usb/format.c 2012-12-11 17:42:14.069181583 +0100 @@ -281,6 +281,21 @@ int nr_triplets, data_size, ret = 0; int clock = snd_usb_clock_find_source(chip, fp->clock); + /* + * Denon DN-X 1600 won't respond when asked for samplerate. + * Currently hardcoded to only work with 96khz + */ + if (chip->usb_id == USB_ID(0x154e, 0x500e)) + { + fp->nr_rates = 1; + fp->rate_min = 96000; + fp->rate_max = 96000; + fp->rates = SNDRV_PCM_RATE_96000; + fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); + fp->rate_table[0] = 96000; + return 0; + } + if (clock < 0) { snd_printk(KERN_ERR "%s(): unable to find clock source (clock %d)\n", __func__, clock); --- a/sound/usb/pcm.c 2012-01-25 08:56:49.000000000 +0100 +++ b/sound/usb/pcm.c 2012-12-11 17:57:11.579301294 +0100 @@ -198,6 +198,13 @@ if (!(fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL)) return 0; + /* + * Denon DN-X 1600 doesn't properly handle this -> bail out, too + */ + + if (chip->usb_id == USB_ID(0x154e, 0x500e)) + return 0; + switch (altsd->bInterfaceProtocol) { case UAC_VERSION_1: default: cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.25. lsusb (...) Bus 002 Device 003: ID 154e:500e D&M Holdings, Inc. (Denon/Marantz) lsusb -t (...) /: Bus 02.Port 1: Dev 1, Class=root_hub, Driver=ehci_hcd/2p, 480M |__ Port 1: Dev 2, If 0, Class=hub, Driver=hub/6p, 480M |__ Port 2: Dev 3, If 0, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 3, If 1, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 3, If 2, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 3, If 3, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 3, If 4, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 3, If 5, Class=HID, Driver=usbhid, 480M (...) lsusb -v (...) Bus 002 Device 003: ID 154e:500e D&M Holdings, Inc. (Denon/Marantz) Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 ? bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x154e D&M Holdings, Inc. (Denon/Marantz) idProduct 0x500e bcdDevice 0.01 iManufacturer 1 iProduct 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 385 bNumInterfaces 6 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 0 bInterfaceCount 3 bFunctionClass 255 Vendor Specific Class bFunctionSubClass 0 bFunctionProtocol 32 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 32 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 2.00 bCategory 0 wTotalLength 92 bmControl 0x01 Latency control Control (read-only) AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bCSourceID 7 bNrChannels 8 bmChannelConfig 0x000000ff Front Left (FL) Front Right (FR) Front Center (FC) Low Frequency Effects (LFE) Back Left (BL) Back Right (BR) Front Left of Center (FLC) Front Right of Center (FRC) bmControls 0x0555 Copy Protect Control (read-only) Connector Control (read-only) Overload Control (read-only) Cluster Control (read-only) Underflow Control (read-only) Overflow Control (read-only) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 1 bCSourceID 7 bmControls 0x0155 Copy Protect Control (read-only) Connector Control (read-only) Overload Control (read-only) Underflow Control (read-only) Overflow Control (read-only) iTerminal 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0201 Microphone bAssocTerminal 0 bCSourceID 7 bNrChannels 8 bmChannelConfig 0x000000ff Front Left (FL) Front Right (FR) Front Center (FC) Low Frequency Effects (LFE) Back Left (BL) Back Right (BR) Front Left of Center (FLC) Front Right of Center (FRC) bmControls 0x0555 Copy Protect Control (read-only) Connector Control (read-only) Overload Control (read-only) Cluster Control (read-only) Underflow Control (read-only) Overflow Control (read-only) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 3 bCSourceID 7 bmControls 0x0155 Copy Protect Control (read-only) Connector Control (read-only) Overload Control (read-only) Underflow Control (read-only) Overflow Control (read-only) iTerminal 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 5 bmAttributes 0x02 Internal variable Clock bmControls 0x03 Clock Frequency Control (read/write) bAssocTerminal 0 iClockSource 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 6 bmAttributes 0x02 Internal variable Clock bmControls 0x03 Clock Frequency Control (read/write) bAssocTerminal 0 iClockSource 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 11 (CLOCK_SELECTOR) bUnitID 7 bNrInPins 2 baCSourceID( 0) 5 baCSourceID( 1) 6 bmControls 0x01 Clock Selector Control (read-only) iClockSelector 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bmControls 0x05 Active Alternate Setting Control (read-only) Valid Alternate Setting Control (read-only) bFormatType 1 bmFormats 1 PCM bNrChannels 8 bmChannelConfig 0x000000ff Front Left (FL) Front Right (FR) Front Center (FC) Low Frequency Effects (LFE) Back Left (BL) Back Right (BR) Front Left of Center (FLC) Front Right of Center (FRC) iChannelNames 0 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 3 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0138 1x 312 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x15 Pitch Control (read-only) Data Overrun Control (read-only) Data Underrun Control (read-only) bLockDelayUnits 1 Milliseconds wLockDelay 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bmControls 0x05 Active Alternate Setting Control (read-only) Valid Alternate Setting Control (read-only) bFormatType 1 bmFormats 1 PCM bNrChannels 8 bmChannelConfig 0x000000ff Front Left (FL) Front Right (FR) Front Center (FC) Low Frequency Effects (LFE) Back Left (BL) Back Right (BR) Front Left of Center (FLC) Front Right of Center (FRC) iChannelNames 0 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 3 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0138 1x 312 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x15 Pitch Control (read-only) Data Overrun Control (read-only) Data Underrun Control (read-only) bLockDelayUnits 1 Milliseconds wLockDelay 0 Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 3 bInterfaceCount 2 bFunctionClass 1 Audio bFunctionSubClass 0 bFunctionProtocol 0 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 9 bInCollection 1 baInterfaceNr( 0) 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 0 MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 97 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 17 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 2 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 18 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 3 bNrInputPins 1 baSourceID( 0) 2 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 19 bNrInputPins 1 baSourceID( 0) 18 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 20 bNrInputPins 1 baSourceID( 0) 17 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 6 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 2 baAssocJackID( 0) 1 baAssocJackID( 1) 17 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 6 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 2 baAssocJackID( 0) 3 baAssocJackID( 1) 19 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 5 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.11 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 34 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1

1 0

[alsa-devel] [PATCH 2/2] Rename internal xonar_u1_* names to just xonar_u_*
by Denis Washington 11 Dec '12

11 Dec '12

Now that the code applies both to the U1 and the U3, xonar_u1 is not a good name anymore. Signed-off-by: Denis Washington <denisw(a)online.de> --- sound/usb/mixer.h | 2 +- sound/usb/mixer_quirks.c | 24 ++++++++++++------------ 2 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index a7f3d45..9e7ec0e 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -22,7 +22,7 @@ struct usb_mixer_interface { u8 rc_buffer[6]; u8 audigy2nx_leds[3]; - u8 xonar_u1_status; + u8 xonar_u_status; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 2223089..de6db70 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -416,23 +416,23 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, } } -static int snd_xonar_u1_switch_get(struct snd_kcontrol *kcontrol, +static int snd_xonar_u_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = !!(mixer->xonar_u1_status & 0x02); + ucontrol->value.integer.value[0] = !!(mixer->xonar_u_status & 0x02); return 0; } -static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, +static int snd_xonar_u_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); u8 old_status, new_status; int err, changed; - old_status = mixer->xonar_u1_status; + old_status = mixer->xonar_u_status; if (ucontrol->value.integer.value[0]) new_status = old_status | 0x02; else @@ -449,27 +449,27 @@ static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, up_read(&mixer->chip->shutdown_rwsem); if (err < 0) return err; - mixer->xonar_u1_status = new_status; + mixer->xonar_u_status = new_status; return changed; } -static struct snd_kcontrol_new snd_xonar_u1_output_switch = { +static struct snd_kcontrol_new snd_xonar_u_output_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Digital Playback Switch", .info = snd_ctl_boolean_mono_info, - .get = snd_xonar_u1_switch_get, - .put = snd_xonar_u1_switch_put, + .get = snd_xonar_u_switch_get, + .put = snd_xonar_u_switch_put, }; -static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) +static int snd_xonar_u_controls_create(struct usb_mixer_interface *mixer) { int err; err = snd_ctl_add(mixer->chip->card, - snd_ctl_new1(&snd_xonar_u1_output_switch, mixer)); + snd_ctl_new1(&snd_xonar_u_output_switch, mixer)); if (err < 0) return err; - mixer->xonar_u1_status = 0x05; + mixer->xonar_u_status = 0x05; return 0; } @@ -1110,7 +1110,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0b05, 0x1739): /* ASUS Xonar U1 */ case USB_ID(0x0b05, 0x1743): /* ASUS Xonar U1 (2) */ case USB_ID(0x0b05, 0x17a0): /* ASUS Xonar U3 */ - err = snd_xonar_u1_controls_create(mixer); + err = snd_xonar_u_controls_create(mixer); break; case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ -- 1.8.0

1 0

[alsa-devel] [PATCH 1/2] Enable S/PDIF on the ASUS Xonar U3
by Denis Washington 11 Dec '12

11 Dec '12

The only required change is to extend the existing Xonar U1 mixer quirks to the U3, which seems to be controlled the same way. Signed-off-by: Denis Washington <denisw(a)online.de> --- sound/usb/mixer_quirks.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index ae2b714..2223089 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1107,8 +1107,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_ftu_create_mixer(mixer); break; - case USB_ID(0x0b05, 0x1739): - case USB_ID(0x0b05, 0x1743): + case USB_ID(0x0b05, 0x1739): /* ASUS Xonar U1 */ + case USB_ID(0x0b05, 0x1743): /* ASUS Xonar U1 (2) */ + case USB_ID(0x0b05, 0x17a0): /* ASUS Xonar U3 */ err = snd_xonar_u1_controls_create(mixer); break; -- 1.8.0

1 0

Re: [alsa-devel] is it a memory bug?
by l9jj＠163.com 11 Dec '12

11 Dec '12

thanks a lot. 在 2012-12-10 19:41:58, Clemens Ladisch <clemens(a)ladisch.de>写道: l9jj(a)163.com wrote: > just call alloca() without free()? <http://linux.die.net/man/3/alloca> Regards, Clemens _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel l9jj(a)163.com ——网易邮箱手机客户端——

1 0