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

Re: [alsa-devel] [RFC 42/42] sound/soc/kirkwood: don't check resource with devm_ioremap_resource
by Mark Brown 10 May '13

10 May '13

On Fri, May 10, 2013 at 10:17:27AM +0200, Wolfram Sang wrote: > devm_ioremap_resource does sanity checks on the given resource. No need to > duplicate this in the driver. Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com>

1 0

Re: [alsa-devel] [RFC 41/42] sound/soc/fsl: don't check resource with devm_ioremap_resource
by Mark Brown 10 May '13

10 May '13

On Fri, May 10, 2013 at 10:17:26AM +0200, Wolfram Sang wrote: > devm_ioremap_resource does sanity checks on the given resource. No need to > duplicate this in the driver. Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com>

1 0

[alsa-devel] [PATCH v3] ASoC: sgtl5000: Fix driver probe after reset
by Fabio Estevam 10 May '13

10 May '13

sgtl5000 codec does not have a reset line, nor a reset command in software. After a 'reboot' command in Linux or after pressing the system's reset button the sgtl5000 driver fails to probe: sgtl5000 0-000a: Device with ID register ffff is not a sgtl5000 sgtl5000 0-000a: ASoC: failed to probe CODEC -19 imx-sgtl5000 sound.12: ASoC: failed to instantiate card -19 imx-sgtl5000 sound.12: snd_soc_register_card failed (-19) As the sgtl5000 codec did not go through a real reset, we cannot rely on the sgtl5000_reg_defaults table, since these are only valid after a clean power-on reset. Fix this issue by explicitly reading all the sgtl5000 registers and filling sgtl5000_reg_defaults with such values. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v2: - Do not use reg_defaults_raw as it is not the correct purpose - Manually build sgtl5000_reg_default - Improve commitlog Changes since v1: - Remove sgtl5000_reg_defaults array - Do not use num_reg_defaults_raw sound/soc/codecs/sgtl5000.c | 58 ++++++++++++++++++++++++------------------- 1 file changed, 33 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 327b443..311dfb5 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -35,31 +35,7 @@ #define SGTL5000_MAX_REG_OFFSET 0x013A /* default value of sgtl5000 registers */ -static const struct reg_default sgtl5000_reg_defaults[] = { - { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, - { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, - { SGTL5000_CHIP_SSS_CTRL, 0x0008 }, - { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, - { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, - { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, - { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, - { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, - { SGTL5000_CHIP_ANA_POWER, 0x7060 }, - { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, - { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, - { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, - { SGTL5000_DAP_SURROUND, 0x0040 }, - { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, - { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, - { SGTL5000_DAP_EQ_BASS_BAND2, 0x002f }, - { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, - { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, - { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, - { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, - { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, - { SGTL5000_DAP_AVC_DECAY, 0x0050 }, -}; +static struct reg_default sgtl5000_reg_defaults[SGTL5000_MAX_REG_OFFSET + 1]; /* regulator supplies for sgtl5000, VDDD is an optional external supply */ enum sgtl5000_regulator_supplies { @@ -1355,6 +1331,33 @@ err_regulator_free: } +/* + * Read all the sgtl5000 registers to fill the cache, as + * we cannot rely on a pre-defined table containing the + * power-on reset values of the registers as done in most + * of the other codec drivers. + * + * We follow this approach here because sgtl5000 does not have + * a reset line, nor a reset command in software, and this way + * we can guarantee that we always have sane register values + * stored in the reg_defaults table after a reset + */ +static int sgtl5000_fill_cache(struct snd_soc_codec *codec) +{ + int i, reg, ret; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + for (i = 0; i < SGTL5000_MAX_REG_OFFSET; i += 2) { + ret = regmap_read(sgtl5000->regmap, i, &reg); + if (ret < 0) + return ret; + sgtl5000_reg_defaults[i].reg = i; + sgtl5000_reg_defaults[i].def = reg; + } + + return 0; +} + static int sgtl5000_probe(struct snd_soc_codec *codec) { int ret; @@ -1377,6 +1380,11 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) if (ret) goto err; + /* Build the reg_defaults manually by reading the registers */ + ret = sgtl5000_fill_cache(codec); + if (ret) + goto err; + /* enable small pop, introduce 400ms delay in turning off */ snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, SGTL5000_SMALL_POP, -- 1.7.9.5

4 4

Re: [alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix driver probe after reset
by Mark Brown 10 May '13

10 May '13

On Thu, May 09, 2013 at 12:08:38PM -0500, Matt Sealey wrote: > match a reset chip powered on for the first time. What we should > actually be doing is dumping these into the chip as the first thing we > do at probe, and then they DO become the reasonable defaults. Yes, this is eactly what I keep saying :/ > The fix is keep using cache/regmaps or whatever, but it definitely > would be an absolute requirement to reset the codec manually by > putting in all the right register values before using the cache, which You don't need to disable the cache at any point.

1 0

[alsa-devel] TLV question - ranges not in volume order
by David Henningsson 10 May '13

10 May '13

Hi, I recently came across a strange ASoC volume control. In this case, the values are not in volume order, i e, the control is declared something like below. I believe the numbers correctly match the hardware. However alsamixer seems not to handle this "reordering" correctly. My question is if this is something that should be fixed in alsamixer (or possibly alsa-lib?), or does one need to write some kind of mapping table in the kernel, just to make something come in volume order to userspace? Any TLV documentation on what TLVs are valid and invalid is also welcome :-) static const unsigned int tlv[] = { TLV_DB_RANGE_HEAD(8), 0, 0, TLV_DB_SCALE_ITEM(-100, 0, 1), 1, 1, TLV_DB_SCALE_ITEM(400, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(200, 0, 0), 3, 3, TLV_DB_SCALE_ITEM(1000, 0, 0), 4, 4, TLV_DB_SCALE_ITEM(0, 0, 0), 5, 5, TLV_DB_SCALE_ITEM(800, 0, 0), 6, 6, TLV_DB_SCALE_ITEM(600, 0, 0), 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0), }; -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

4 6

Re: [alsa-devel] [RFC PATCH 1/3] drm/915: Add private api for power well usage
by Daniel Vetter 10 May '13

10 May '13

Hi all, Three things: 1. Any reason public mailing lists (intel-gfx, alsa-devel) aren't cc'ed? Afaics no secret stuff is being discussed here ... Please resend the patches with mailings lists cc'ed. 2. On a quick read of the hda driver stuff I don't think the power well enable/disable places are at the right spot: We force the power well on whenever the hda codec is used, but we must only do that when we want to output audio to external screens. And since that state transition can only really happen due to a modeset change on the gfx side I don't think we need any autosuspend delays either. The use-case I'm thinking of is media playback with just the panel enabled: In that case we can switch off the power well on the display side, and I don't think the power well is required for audio play back on the integrated speakerrs/headphone-jack either. 3. The moduel depencies seem to still not work out dynamically, at least I think we still have a hard chain between hda-intel and i915.ko (just with one thing in between now). Cheers, Daniel On Thu, May 9, 2013 at 9:23 AM, Wang, Xingchao <xingchao.wang(a)intel.com> wrote: > Hi All, > > This is a newer version patchset for power well feature implementation. > > I will send out a document later to describe the design, especially for David and Takashi as they could not see Bspec > so it maybe a bit confuse. Meanwhile Liam will send out a meeting request for discussion, welcome Daniel, Paulo, Jesse to join us. > > Basically, here's the latest working status on my side: > > I tested the power well feature on Haswell ULT board, there's only one eDP port used. > Without this patch and we enabled power well feature, gfx driver will shutdown power well, audio driver will crash. > Applied this patch, display audio driver will request power well on before initialize the card. In gfx side, it will enable power well. > > Also power_save feature in audio side should be enabled, I set "5" second to let codec enter suspend when free for 5s long. > Audio controller driver detects all codecs are suspended it will release power well and suspend too. Gfx driver will receive the request and shut down power well. > > If user trigger some audio operations(cat codec# info), audio controller driver will request power well again... > > If gfx side decided to disable power well now, while audio is in use, power well would be kept on. > > Generally audio drvier will request power well on need and release it immediately after suspend. Gfx should make decision at his side. > > The new introduced module "hda-i915" has dependency on i915, only built in when i915 enabled. So it's safe for call. > This module was based on David's original patch, which added pm_suspend/resume callback for hdmi codec. As the codecs and controller are both depending on powerwell, > and codec *must* already on power so I moved the power well request/release to controller side and changed the module name to "hda-i915". > David, would you like me to add you as author for the second patch? :) > > For non-Haswell platform, power well is no need atm. If the module is built in and gfx will reject power well request at its side, so it's safe too. > > thanks > --xingchao > > >> -----Original Message----- >> From: Wang Xingchao [mailto:xingchao.wang@linux.intel.com] >> Sent: Thursday, May 09, 2013 3:01 PM >> To: david.henningsson(a)canonical.com; Girdwood, Liam R; tiwai(a)suse.de; >> daniel(a)ffwll.ch >> Cc: Barnes, Jesse; Li, Jocelyn; Lin, Mengdong; Wang, Xingchao; Zanoni, Paulo R; >> Wang Xingchao >> Subject: [RFC PATCH 1/3] drm/915: Add private api for power well usage >> >> Haswell Display audio depends on power well in graphic side, it should request >> power well before use it and release power well after use. >> I915 will not shutdown power well if it detects audio is using. >> This patch protects display audio crash for Intel Haswell mobile >> C3 stepping board. >> >> Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> >> --- >> drivers/gpu/drm/i915/intel_pm.c | 127 >> ++++++++++++++++++++++++++++++++++++--- >> include/drm/audio_drm.h | 39 ++++++++++++ >> 2 files changed, 159 insertions(+), 7 deletions(-) create mode 100644 >> include/drm/audio_drm.h >> >> diff --git a/drivers/gpu/drm/i915/intel_pm.c b/drivers/gpu/drm/i915/intel_pm.c >> index 2557926..cba753e 100644 >> --- a/drivers/gpu/drm/i915/intel_pm.c >> +++ b/drivers/gpu/drm/i915/intel_pm.c >> @@ -4298,18 +4298,12 @@ bool intel_using_power_well(struct drm_device >> *dev) >> return true; >> } >> >> -void intel_set_power_well(struct drm_device *dev, bool enable) >> +void set_power_well(struct drm_device *dev, bool enable) >> { >> struct drm_i915_private *dev_priv = dev->dev_private; >> bool is_enabled, enable_requested; >> uint32_t tmp; >> >> - if (!HAS_POWER_WELL(dev)) >> - return; >> - >> - if (!i915_disable_power_well && !enable) >> - return; >> - >> tmp = I915_READ(HSW_PWR_WELL_DRIVER); >> is_enabled = tmp & HSW_PWR_WELL_STATE; >> enable_requested = tmp & HSW_PWR_WELL_ENABLE; @@ -4332,6 >> +4326,125 @@ void intel_set_power_well(struct drm_device *dev, bool >> enable) >> } >> } >> >> +/* Global drm_device for display audio drvier usage */ static struct >> +drm_device *power_well_device; >> +/* Lock protecting power well setting */ >> +DEFINE_SPINLOCK(powerwell_lock); static bool i915_power_well_using; >> + >> +struct power_well { >> + int user_cnt; >> + struct list_head power_well_list; >> +}; >> + >> +static struct power_well hsw_power = { >> + .user_cnt = 0, >> + .power_well_list = LIST_HEAD_INIT(hsw_power.power_well_list), >> +}; >> + >> +struct power_well_user { >> + char name[16]; >> + struct list_head list; >> +}; >> + >> +void i915_request_power_well(const char *name) { >> + struct power_well_user *user; >> + >> + DRM_DEBUG_KMS("request power well for %s\n", name); >> + spin_lock_irq(&powerwell_lock); >> + if (!power_well_device) >> + goto out; >> + >> + if (!IS_HASWELL(power_well_device)) >> + goto out; >> + >> + list_for_each_entry(user, &hsw_power.power_well_list, list) { >> + if (!strcmp(user->name, name)) { >> + goto out; >> + } >> + } >> + >> + user = kzalloc(sizeof(*user), GFP_KERNEL); >> + if (user == NULL) { >> + goto out; >> + } >> + strcpy(user->name, name); >> + >> + list_add(&user->list, &hsw_power.power_well_list); >> + hsw_power.user_cnt++; >> + >> + if (hsw_power.user_cnt == 1) { >> + set_power_well(power_well_device, true); >> + DRM_DEBUG_KMS("%s set power well on\n", user->name); >> + } >> +out: >> + spin_unlock_irq(&powerwell_lock); >> + return; >> +} >> +EXPORT_SYMBOL_GPL(i915_request_power_well); >> + >> +void i915_release_power_well(const char *name) { >> + struct power_well_user *user; >> + >> + DRM_DEBUG_KMS("release power well from %s\n", name); >> + spin_lock_irq(&powerwell_lock); >> + if (!power_well_device) >> + goto out; >> + >> + if (!IS_HASWELL(power_well_device)) >> + goto out; >> + >> + list_for_each_entry(user, &hsw_power.power_well_list, list) { >> + if (!strcmp(user->name, name)) { >> + list_del(&user->list); >> + hsw_power.user_cnt--; >> + DRM_DEBUG_KMS("Releaseing power well:%s, user_cnt:%d, >> i915 using:%d\n", >> + user->name, hsw_power.user_cnt, >> i915_power_well_using); >> + if (!hsw_power.user_cnt && !i915_power_well_using) >> + set_power_well(power_well_device, false); >> + goto out; >> + } >> + } >> + >> + DRM_DEBUG_KMS("power well %s doesnot exist\n", name); >> +out: >> + spin_unlock_irq(&powerwell_lock); >> + return; >> +} >> +EXPORT_SYMBOL_GPL(i915_release_power_well); >> + >> +/* TODO: Call this when i915 module unload */ void >> +remove_power_well(void) { >> + spin_lock_irq(&powerwell_lock); >> + i915_power_well_using = false; >> + power_well_device = NULL; >> + spin_unlock_irq(&powerwell_lock); >> +} >> + >> +void intel_set_power_well(struct drm_device *dev, bool enable) { >> + if (!HAS_POWER_WELL(dev)) >> + return; >> + >> + spin_lock_irq(&powerwell_lock); >> + power_well_device = dev; >> + i915_power_well_using = enable; >> + if (!enable && hsw_power.user_cnt) { >> + DRM_DEBUG_KMS("Display audio power well busy using now\n"); >> + goto out; >> + } >> + >> + if (!i915_disable_power_well && !enable) >> + goto out; >> + set_power_well(dev, enable); >> +out: >> + spin_unlock_irq(&powerwell_lock); >> + return; >> +} >> + >> /* >> * Starting with Haswell, we have a "Power Down Well" that can be turned off >> * when not needed anymore. We have 4 registers that can request the >> power well diff --git a/include/drm/audio_drm.h b/include/drm/audio_drm.h >> new file mode 100644 index 0000000..215295f >> --- /dev/null >> +++ b/include/drm/audio_drm.h >> @@ -0,0 +1,39 @@ >> +/************************************************************** >> ******** >> +**** >> + * >> + * Copyright 2013 Intel Inc. >> + * All Rights Reserved. >> + * >> + * Permission is hereby granted, free of charge, to any person >> +obtaining a >> + * copy of this software and associated documentation files (the >> + * "Software"), to deal in the Software without restriction, including >> + * without limitation the rights to use, copy, modify, merge, publish, >> + * distribute, sub license, and/or sell copies of the Software, and to >> + * permit persons to whom the Software is furnished to do so, subject >> +to >> + * the following conditions: >> + * >> + * The above copyright notice and this permission notice (including the >> + * next paragraph) shall be included in all copies or substantial >> +portions >> + * of the Software. >> + * >> + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, >> +EXPRESS OR >> + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF >> +MERCHANTABILITY, >> + * FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT. IN NO >> EVENT >> +SHALL >> + * THE COPYRIGHT HOLDERS, AUTHORS AND/OR ITS SUPPLIERS BE LIABLE >> FOR >> +ANY CLAIM, >> + * DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, >> TORT >> +OR >> + * OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE >> SOFTWARE >> +OR THE >> + * USE OR OTHER DEALINGS IN THE SOFTWARE. >> + * >> + * >> + >> +*************************************************************** >> ******** >> +***/ >> +/* >> + * Authors: >> + * Wang Xingchao <xingchao.wang(a)intel.com> */ >> + >> +#ifndef _AUDIO_DRM_H_ >> +#define _AUDIO_DRM_H_ >> + >> +extern void i915_request_power_well(const char *name); extern void >> +i915_release_power_well(const char *name); >> + >> +#endif /* _AUDIO_DRM_H_ */ >> -- >> 1.7.9.5 > -- Daniel Vetter Software Engineer, Intel Corporation +41 (0) 79 365 57 48 - http://blog.ffwll.ch

4 5

[alsa-devel] [PATCH v4] ASoC: sgtl5000: Fix driver probe after reset
by Fabio Estevam 10 May '13

10 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> After a 'reboot' command in Linux or after pressing the system's reset button the sgtl5000 driver fails to probe: sgtl5000 0-000a: Device with ID register ffff is not a sgtl5000 sgtl5000 0-000a: ASoC: failed to probe CODEC -19 imx-sgtl5000 sound.12: ASoC: failed to instantiate card -19 imx-sgtl5000 sound.12: snd_soc_register_card failed (-19) sgtl5000 codec does not have a reset line, nor a reset command in software, so after a system reset the codec does not contain the default register values from sgtl5000_reg_defaults[] anymore, as these are only valid after a power-on-reset cycle. Fix this issue by explicitly reading all the reset register values from sgtl5000_reg_defaults[] and writing them back into sgtl5000 to ensure a sane state. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v3: - Read sgtl5000_reg_defaults and write these values into sgtl5000 Changes since v2: - Do not use reg_defaults_raw as it is not the correct purpose - Manually build sgtl5000_reg_default - Improve commitlog Changes since v1: - Remove sgtl5000_reg_defaults array - Do not use num_reg_defaults_raw sound/soc/codecs/sgtl5000.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 327b443..be23fce 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1474,6 +1474,31 @@ static const struct regmap_config sgtl5000_regmap = { .num_reg_defaults = ARRAY_SIZE(sgtl5000_reg_defaults), }; +/* + * Write all the default values from sgtl5000_reg_defaults[] array into the + * sgtl5000 registers, to make sure we always start with the sane registers + * values as stated in the datasheet. + * + * Since sgtl5000 does not have a reset line, nor a reset command in software, + * we follow this approach to guarantee we always start from the default values + * and avoid problems like, not being able to probe after an audio playback + * followed by a system reset or a 'reboot' command in Linux + */ +static int sgtl5000_fill_defaults(struct sgtl5000_priv *sgtl5000) +{ + int i, ret, val, index; + + for (i = 0; i < ARRAY_SIZE(sgtl5000_reg_defaults); i++) { + val = sgtl5000_reg_defaults[i].def; + index = sgtl5000_reg_defaults[i].reg; + ret = regmap_write(sgtl5000->regmap, index, val); + if (ret) + return ret; + } + + return 0; +} + static int sgtl5000_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -1494,6 +1519,11 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, sgtl5000); + /* Ensure sgtl5000 will start with sane register values */ + ret = sgtl5000_fill_defaults(sgtl5000); + if (ret) + return ret; + ret = snd_soc_register_codec(&client->dev, &sgtl5000_driver, &sgtl5000_dai, 1); return ret; -- 1.7.9.5

2 1

Re: [alsa-devel] BUG: USB audio discontinuities with 'UHCI: implement new semantics for URB_ISO_ASAP'
by Alan Stern 09 May '13

09 May '13

On Wed, 8 May 2013, Joe Rayhawk wrote: > For what it's worth, the following -stable patch > > "ALSA: USB: adjust for changed 3.8 USB API" > c75c5ab575af7db707689cdbb5a5c458e9a034bb > > fixes the discontinuous playback on period sizes between 139 and 192, > but the discontinuous playback on period sizes between 48 and 138 has > been replaced by rather worse kernel blocking and kernel errors: > > jrayhawk@richardiv:~$ sudo tail -F -n 0 /var/log/kern.log & > [1] 4490 > jrayhawk@richardiv:~$ time perl -e 'print pack "H*", "00FF" x 2048' | aplay --period-size=192 -r 48000 -f S16_LE -c2 -D hw:0,0 > Playing raw data 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo > > real 0m0.053s > user 0m0.008s > sys 0m0.000s > jrayhawk@richardiv:~$ time perl -e 'print pack "H*", "00FF" x 2048' | aplay --period-size=48 -r 48000 -f S16_LE -c2 -D hw:0,0 > Playing raw data 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo > May 8 17:01:56 richardiv kernel: [ 1680.923474] cannot submit urb (err = -18) > May 8 17:01:56 richardiv kernel: [ 1680.924472] cannot submit urb (err = -18) > > real 0m10.023s > user 0m0.008s > sys 0m0.004s I don't know if you want to investigate this in any detail. If you do, please post a usbmon trace of a failed playback. Alan Stern -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

[alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix driver probe after reset
by Fabio Estevam 09 May '13

09 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 codec does not have a reset line, nor a reset command in software. After a 'reboot' command in Linux or after pressing the system's reset button the sgtl5000 driver fails to probe: sgtl5000 0-000a: Device with ID register ffff is not a sgtl5000 sgtl5000 0-000a: ASoC: failed to probe CODEC -19 imx-sgtl5000 sound.12: ASoC: failed to instantiate card -19 imx-sgtl5000 sound.12: snd_soc_register_card failed (-19) As the sgtl5000 codec did not go through a real reset, we cannot rely on the sgtl5000_reg_defaults array, since these are only valid after a clean power-on reset. To fix this issue, use 'num_reg_defaults_raw' which tells the regcache driver to actually read the codec registers and then fill the cache with these values. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Remove sgtl5000_reg_defaults array - Do not use num_reg_defaults_raw sound/soc/codecs/sgtl5000.c | 30 +----------------------------- 1 file changed, 1 insertion(+), 29 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 327b443..fa342f7 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -34,33 +34,6 @@ #define SGTL5000_DAP_REG_OFFSET 0x0100 #define SGTL5000_MAX_REG_OFFSET 0x013A -/* default value of sgtl5000 registers */ -static const struct reg_default sgtl5000_reg_defaults[] = { - { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, - { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, - { SGTL5000_CHIP_SSS_CTRL, 0x0008 }, - { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, - { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, - { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, - { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, - { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, - { SGTL5000_CHIP_ANA_POWER, 0x7060 }, - { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, - { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, - { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, - { SGTL5000_DAP_SURROUND, 0x0040 }, - { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, - { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, - { SGTL5000_DAP_EQ_BASS_BAND2, 0x002f }, - { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, - { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, - { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, - { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, - { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, - { SGTL5000_DAP_AVC_DECAY, 0x0050 }, -}; - /* regulator supplies for sgtl5000, VDDD is an optional external supply */ enum sgtl5000_regulator_supplies { VDDA, @@ -1470,8 +1443,7 @@ static const struct regmap_config sgtl5000_regmap = { .readable_reg = sgtl5000_readable, .cache_type = REGCACHE_RBTREE, - .reg_defaults = sgtl5000_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(sgtl5000_reg_defaults), + .num_reg_defaults_raw = SGTL5000_MAX_REG_OFFSET, }; static int sgtl5000_i2c_probe(struct i2c_client *client, -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: Fix driver probe after reset
by Fabio Estevam 09 May '13

09 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 codec does not have a reset line, nor a reset command in software. After a 'reboot' command in Linux or after pressing the system's reset button the sgtl5000 driver fails to probe: sgtl5000 0-000a: Device with ID register ffff is not a sgtl5000 sgtl5000 0-000a: ASoC: failed to probe CODEC -19 imx-sgtl5000 sound.12: ASoC: failed to instantiate card -19 imx-sgtl5000 sound.12: snd_soc_register_card failed (-19) As the sgtl5000 codec did not go through a real reset, we cannot rely on the reg_defaults array, since these are only valid after a clean power-on reset. To fix this issue, use 'reg_defaults_raw' which tells the regcache driver to actually read the codec registers and then fill the cache. 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 327b443..c59fae8 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1470,8 +1470,8 @@ static const struct regmap_config sgtl5000_regmap = { .readable_reg = sgtl5000_readable, .cache_type = REGCACHE_RBTREE, - .reg_defaults = sgtl5000_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(sgtl5000_reg_defaults), + .reg_defaults_raw = sgtl5000_reg_defaults, + .num_reg_defaults_raw = ARRAY_SIZE(sgtl5000_reg_defaults), }; static int sgtl5000_i2c_probe(struct i2c_client *client, -- 1.7.9.5

3 2

Re: [alsa-devel] [RFC] ASoC: sgtl5000: Remove cache support
by Mark Brown 08 May '13

08 May '13

On Tue, May 07, 2013 at 03:06:39PM -0500, Matt Sealey wrote: > On Mon, May 6, 2013 at 5:06 PM, Mark Brown <broonie(a)kernel.org> wrote: > Just for reference the supplies on the Efika MX are VDDD 1.2V, VDDDA > 1.65V and VDDIO 2.775V respectively. For the sake of a couple 0.1uF > capacitors you're not really saving much money in the lowest cost > configuration, and you're doubling to tripling power consumption. If > 15mW makes a difference to you (especially if you're supplying the > above from rails connected to other devices and you're concerned about > current limiting etc.) it makes sense to go for the lower power one. Well. some applications just don't care about power consumption. > It is hard to imagine an i.MX board without 1.2V, 1.6V and 3.3V from a > PMIC to supply the codec so letting VDDD float and using the internal > LDO seems like an exercise in selling the codec to people who aren't > using other Freescale parts.. Even if there's physically a supply at the appropriate voltage people can have concerns about running noisy digital stuff near their analogue, the on-chip regulators aren't purely about component cost. Not sure that this is a consideration here but... > I think most of the weird problems is the LDO configuration itself - > Linux assumes that it's not set up, when if VDDD isn't there, it is > already. Weird sequencing of supplies by boards (either not bringing > any supplies down or not bringing them down correctly in a reasonable > order) on warm reboot and enabling it *regardless* at revision 0x11 or > above which is just turning (on our boards) a 1.2V supply into a > better filtered 1.2V supply and making the chip run a little hotter. Is it actually having a postive impact on performance? > I am curious about the version check more than anything. Either the > internal LDO is required to filter poor supplies on some boards (which > ones?) or revisions equal or above 0x11 had some DIFFERENT digital > supply requirement - I seem to remember the reference manual stating > 1.2V was a suitable input supply, but the minimum is 1.1V in the > latest RM. Maybe if you run it at 1.2V there's some problem. It's not > documented why that decision would be made.. no errata.. That sounds like tolerances doesn't it? Spec of 1.2V. > >> So maybe we don't disable the cache, or it gets ported to regmap, but > >> the order should be enable regulators in a non-cached way then fill > >> the cache AFTERWARDS - the code at ToT mainline doesn't fix the cache > > I'm sorry, this makes no sense to me. If we know the physical defaults > > for the device and we can get the device into that state then we should > > have no need to read from the cache. If we don't know what they are > > then they shouldn't be listed. > In the case VDDD isn't supplied when the other two supplies are > brought up, then the LDO is enabled, so if the register controlling > the LDO configuration is in the cache.. that might explain a lot... I still don't understand this at all. Is the power on value for the register variable? > > I think the point that you lost me was your assumption that the > > regulator enable needs to be done without the cache. What makes you > > believe that this is the case? > Because that's what Freescale did in their code as per their comments > - if only because they needed to do this kind of setup before the > cache/defaults were populated. I assume they populate the cache from > the chip at probe because the defaults change per revision and the > presence of VDDD or not may change some of the registers in the cache. We just have no idea why they're doing what they're doing; this is all guesswork. I've not seen anything that indicates that anyone who's sent any code has any understanding of what they're changing. > It may also be on boards like i.MX6 Sabrelite that the "defaults" are > not the "defaults" over a warm reboot, since the chip never powers off > (just "optionally" loses clock depending on whether CLKO is > reconfigured or not at some point) and probably retains everything you > ever set. This is why most drivers do a hardware reset of the device on gaining control of it, in order to ensure that the hardware is in a known state. I see that the driver doesn't do that except through the regulators. I see this driver doesn't do that which isn't terribly clever, though it's not clear that the chip has suitable support. > We want to do two possible things... > * At probe, dump some known good defaults into the chip and use those as a cache > * At probe, read out all the registers assuming they're already known > good defaults into a cache Clearly the first option is more sane since otherwise you've no idea if the chip is configured sensibly. > The intent being in the first case to make for damn sure the codec is > in a well known state, and in the second case JUST to reduce i2c > writes further down the line. Freescale seem to like the second one, This isn't the sole function of the register cache, it's also used to implement most of suspend and resume and to allow the device to be powered off when idle but still present controls. > Mainline is doing the first set, although it's definitely possible > those defaults are not the true defaults if they were derived from a > board like the Sabrelite where a warm reboot might retain all the > state from previous boots.. the precedence of that list of defaults is > basically unknown.. and I recall Freescale updating them in the BSP a > lot. I'd *really* hope that the defaults come from the chip datasheet. If they don't that's extremely disappointing. > So the problem here might be some quirky board designs, a random > implementation of an erratum workaround with no documentation, a bunch My impression based on what I'm seeing here, including things like your comments about lots of register default changes, is that people working on the driver haven't had much understanding of what they're doing and have got confused as a result. Sounds like a good first step is getting the register defaults sane...

3 5

[alsa-devel] [PATCH v2] ASoC: wm_adsp: Expose coefficient blocks as ALSA binary controls
by Dimitris Papastamos 08 May '13

08 May '13

Add initial support for runtime tuning for the ADSP cores. This is achieved by exposing the coefficient configuration blocks as ALSA binary controls. The current code assumes that no controls on the DSP are volatile. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 458 ++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 3 + 2 files changed, 454 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 9af1bdd..a9b9ebb 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -21,6 +21,7 @@ #include <linux/regmap.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> +#include <linux/workqueue.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -206,6 +207,36 @@ static struct { { .file = "rx-anc" }, }; +struct wm_coeff_ctl_ops { + int (*xget)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*xput)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*xinfo)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +}; + +struct wm_coeff { + struct device *dev; + struct list_head ctl_list; + struct regmap *regmap; +}; + +struct wm_coeff_ctl { + const char *name; + struct snd_card *card; + struct wm_adsp_alg_region region; + struct wm_coeff_ctl_ops ops; + struct wm_adsp *adsp; + void *private; + unsigned int enabled:1; + struct list_head list; + void *cache; + size_t len; + unsigned int dirty:1; + struct snd_kcontrol *kcontrol; +}; + static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -290,6 +321,181 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, } } +static int wm_coeff_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = ctl->len; + return 0; +} + +static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, + const void *buf, size_t len) +{ + struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_adsp_alg_region *region = &ctl->region; + const struct wm_adsp_region *mem; + struct wm_adsp *adsp = ctl->adsp; + void *scratch; + int ret; + unsigned int reg; + + mem = wm_adsp_find_region(adsp, region->type); + if (!mem) { + adsp_err(adsp, "No base for region %x\n", + region->type); + return -EINVAL; + } + + reg = ctl->region.base; + reg = wm_adsp_region_to_reg(mem, reg); + + scratch = kmemdup(buf, ctl->len, GFP_KERNEL | GFP_DMA); + if (!scratch) + return -ENOMEM; + + ret = regmap_raw_write(wm_coeff->regmap, reg, scratch, + ctl->len); + if (ret) { + adsp_err(adsp, "Failed to write %zu bytes to %x\n", + ctl->len, reg); + kfree(scratch); + return ret; + } + + kfree(scratch); + + return 0; +} + +static int wm_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + char *p = ucontrol->value.bytes.data; + + memcpy(ctl->cache, p, ctl->len); + + if (!ctl->enabled) { + ctl->dirty = 1; + return 0; + } + + return wm_coeff_write_control(kcontrol, p, ctl->len); +} + +static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, + void *buf, size_t len) +{ + struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_adsp_alg_region *region = &ctl->region; + const struct wm_adsp_region *mem; + struct wm_adsp *adsp = ctl->adsp; + void *scratch; + int ret; + unsigned int reg; + + mem = wm_adsp_find_region(adsp, region->type); + if (!mem) { + adsp_err(adsp, "No base for region %x\n", + region->type); + return -EINVAL; + } + + reg = ctl->region.base; + reg = wm_adsp_region_to_reg(mem, reg); + + scratch = kmalloc(ctl->len, GFP_KERNEL | GFP_DMA); + if (!scratch) + return -ENOMEM; + + ret = regmap_raw_read(wm_coeff->regmap, reg, scratch, ctl->len); + if (ret) { + adsp_err(adsp, "Failed to read %zu bytes from %x\n", + ctl->len, reg); + kfree(scratch); + return ret; + } + + memcpy(buf, scratch, ctl->len); + kfree(scratch); + + return 0; +} + +static int wm_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + char *p = ucontrol->value.bytes.data; + + memcpy(p, ctl->cache, ctl->len); + return 0; +} + +static int wm_coeff_add_kcontrol(struct wm_coeff *wm_coeff, + struct wm_coeff_ctl *ctl, + const struct snd_kcontrol_new *kctl) +{ + int ret; + struct snd_kcontrol *kcontrol; + + kcontrol = snd_ctl_new1(kctl, wm_coeff); + ret = snd_ctl_add(ctl->card, kcontrol); + if (ret < 0) { + dev_err(wm_coeff->dev, "Failed to add %s: %d\n", + kctl->name, ret); + return ret; + } + ctl->kcontrol = kcontrol; + return 0; +} + +struct wmfw_ctl_work { + struct wm_coeff *wm_coeff; + struct wm_coeff_ctl *ctl; + struct work_struct work; +}; + +static int wmfw_add_ctl(struct wm_coeff *wm_coeff, + struct wm_coeff_ctl *ctl) +{ + struct snd_kcontrol_new *kcontrol; + int ret; + + if (!wm_coeff || !ctl || !ctl->name || !ctl->card) + return -EINVAL; + + kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); + if (!kcontrol) + return -ENOMEM; + kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + + kcontrol->name = ctl->name; + kcontrol->info = wm_coeff_info; + kcontrol->get = wm_coeff_get; + kcontrol->put = wm_coeff_put; + kcontrol->private_value = (unsigned long)ctl; + + ret = wm_coeff_add_kcontrol(wm_coeff, + ctl, kcontrol); + if (ret < 0) + goto err_kcontrol; + + kfree(kcontrol); + + list_add(&ctl->list, &wm_coeff->ctl_list); + return 0; + +err_kcontrol: + kfree(kcontrol); + return ret; +} + static int wm_adsp_load(struct wm_adsp *dsp) { LIST_HEAD(buf_list); @@ -503,7 +709,156 @@ out: return ret; } -static int wm_adsp_setup_algs(struct wm_adsp *dsp) +static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff) +{ + struct wm_coeff_ctl *ctl; + int ret; + + list_for_each_entry(ctl, &wm_coeff->ctl_list, + list) { + if (!ctl->enabled || ctl->dirty) + continue; + ret = wm_coeff_read_control(ctl->kcontrol, + ctl->cache, + ctl->len); + if (ret < 0) + return ret; + } + + return 0; +} + +static int wm_coeff_sync_controls(struct wm_coeff *wm_coeff) +{ + struct wm_coeff_ctl *ctl; + int ret; + + list_for_each_entry(ctl, &wm_coeff->ctl_list, + list) { + if (!ctl->enabled) + continue; + if (ctl->dirty) { + ret = wm_coeff_write_control(ctl->kcontrol, + ctl->cache, + ctl->len); + if (ret < 0) + return ret; + ctl->dirty = 0; + } + } + + return 0; +} + +static void wm_adsp_ctl_work(struct work_struct *work) +{ + struct wmfw_ctl_work *ctl_work = container_of(work, + struct wmfw_ctl_work, + work); + + wmfw_add_ctl(ctl_work->wm_coeff, ctl_work->ctl); + kfree(ctl_work); +} + +static int wm_adsp_create_control(struct snd_soc_codec *codec, + const struct wm_adsp_alg_region *region) + +{ + struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); + struct wm_coeff_ctl *ctl; + struct wmfw_ctl_work *ctl_work; + char *name; + char *region_name; + int ret; + + name = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!name) + return -ENOMEM; + + switch (region->type) { + case WMFW_ADSP1_PM: + region_name = "PM"; + break; + case WMFW_ADSP1_DM: + region_name = "DM"; + break; + case WMFW_ADSP2_XM: + region_name = "XM"; + break; + case WMFW_ADSP2_YM: + region_name = "YM"; + break; + case WMFW_ADSP1_ZM: + region_name = "ZM"; + break; + default: + return -EINVAL; + } + + snprintf(name, PAGE_SIZE, "DSP%d %s %x", + dsp->num, region_name, region->alg); + + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + if (!strcmp(ctl->name, name)) { + if (!ctl->enabled) + ctl->enabled = 1; + return 0; + } + } + + ctl = kzalloc(sizeof(*ctl), GFP_KERNEL); + if (!ctl) { + ret = -ENOMEM; + goto err_name; + } + ctl->region = *region; + ctl->name = kmemdup(name, strlen(name) + 1, GFP_KERNEL); + if (!ctl->name) { + ret = -ENOMEM; + goto err_ctl; + } + ctl->enabled = 1; + ctl->dirty = 0; + ctl->ops.xget = wm_coeff_get; + ctl->ops.xput = wm_coeff_put; + ctl->card = codec->card->snd_card; + ctl->adsp = dsp; + + ctl->len = region->len; + ctl->cache = kzalloc(ctl->len, GFP_KERNEL); + if (!ctl->cache) { + ret = -ENOMEM; + goto err_ctl_name; + } + + ctl_work = kzalloc(sizeof(*ctl_work), GFP_KERNEL); + if (!ctl_work) { + ret = -ENOMEM; + goto err_ctl_cache; + } + + ctl_work->wm_coeff = dsp->wm_coeff; + ctl_work->ctl = ctl; + INIT_WORK(&ctl_work->work, wm_adsp_ctl_work); + schedule_work(&ctl_work->work); + + kfree(name); + + return 0; + +err_ctl_cache: + kfree(ctl->cache); +err_ctl_name: + kfree(ctl->name); +err_ctl: + kfree(ctl); +err_name: + kfree(name); + return ret; +} + +static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) { struct regmap *regmap = dsp->regmap; struct wmfw_adsp1_id_hdr adsp1_id; @@ -644,7 +999,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP1_DM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].dm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp1_alg[i + 1].dm); + region->len -= be32_to_cpu(adsp1_alg[i].dm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region DM with ID %x\n", + be32_to_cpu(adsp1_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -652,7 +1016,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP1_ZM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].zm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp1_alg[i + 1].zm); + region->len -= be32_to_cpu(adsp1_alg[i].zm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", + be32_to_cpu(adsp1_alg[i].alg.id)); + } break; case WMFW_ADSP2: @@ -672,7 +1045,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_XM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].xm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].xm); + region->len -= be32_to_cpu(adsp2_alg[i].xm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region XM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -680,7 +1062,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_YM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].ym); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].ym); + region->len -= be32_to_cpu(adsp2_alg[i].ym); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region YM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -688,7 +1079,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_ZM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].zm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].zm); + region->len -= be32_to_cpu(adsp2_alg[i].zm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } break; } } @@ -884,6 +1284,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = w->codec; struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; + struct wm_coeff_ctl *ctl; int ret; int val; @@ -921,7 +1322,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp); + ret = wm_adsp_setup_algs(dsp, codec); if (ret != 0) goto err; @@ -929,6 +1330,16 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + /* Initialize caches for enabled and non-dirty controls */ + ret = wm_coeff_init_control_caches(dsp->wm_coeff); + if (ret != 0) + goto err; + + /* Sync dirty controls */ + ret = wm_coeff_sync_controls(dsp->wm_coeff); + if (ret != 0) + goto err; + /* Start the core running */ regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_CORE_ENA | ADSP1_START, @@ -945,6 +1356,11 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, 0); + + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + ctl->enabled = 0; + } break; default: @@ -997,6 +1413,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; struct wm_adsp_alg_region *alg_region; + struct wm_coeff_ctl *ctl; unsigned int val; int ret; @@ -1062,7 +1479,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp); + ret = wm_adsp_setup_algs(dsp, codec); if (ret != 0) goto err; @@ -1070,6 +1487,16 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + /* Initialize caches for enabled and non-dirty controls */ + ret = wm_coeff_init_control_caches(dsp->wm_coeff); + if (ret != 0) + goto err; + + /* Sync dirty controls */ + ret = wm_coeff_sync_controls(dsp->wm_coeff); + if (ret != 0) + goto err; + ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, ADSP2_CORE_ENA | ADSP2_START, @@ -1107,6 +1534,11 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ret); } + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + ctl->enabled = 0; + } + while (!list_empty(&dsp->alg_regions)) { alg_region = list_first_entry(&dsp->alg_regions, struct wm_adsp_alg_region, @@ -1145,36 +1577,48 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) INIT_LIST_HEAD(&adsp->alg_regions); + adsp->wm_coeff = kzalloc(sizeof(*adsp->wm_coeff), + GFP_KERNEL); + if (!adsp->wm_coeff) + return -ENOMEM; + adsp->wm_coeff->regmap = adsp->regmap; + adsp->wm_coeff->dev = adsp->dev; + INIT_LIST_HEAD(&adsp->wm_coeff->ctl_list); + if (dvfs) { adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); if (IS_ERR(adsp->dvfs)) { ret = PTR_ERR(adsp->dvfs); dev_err(adsp->dev, "Failed to get DCVDD: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_enable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to enable DCVDD: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_set_voltage(adsp->dvfs, 1200000, 1800000); if (ret != 0) { dev_err(adsp->dev, "Failed to initialise DVFS: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_disable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to disable DCVDD: %d\n", ret); - return ret; + goto out_coeff; } } return 0; + +out_coeff: + kfree(adsp->wm_coeff); + return ret; } EXPORT_SYMBOL_GPL(wm_adsp2_init); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index cb8871a..e6e705a 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -30,6 +30,7 @@ struct wm_adsp_alg_region { unsigned int alg; int type; unsigned int base; + size_t len; }; struct wm_adsp { @@ -53,6 +54,8 @@ struct wm_adsp { bool running; struct regulator *dvfs; + + struct wm_coeff *wm_coeff; }; #define WM_ADSP1(wname, num) \ -- 1.8.2.2

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: Mark CHIP_ANA_POWER as volatile
by Fabio Estevam 08 May '13

08 May '13

Register CHIP_ANA_POWER (address: 0x0030) contains all of the power down controls for the analog blocks. Mark it as a volatile register, so that sgtl5000 can be functional after a reset. Without this patch the following error happens after a audio playback followed by a reset: sgtl5000 0-000a: Device with ID register ffff is not a sgtl5000 sgtl5000 0-000a: ASoC: failed to probe CODEC -19 imx-sgtl5000 sound.12: ASoC: failed to instantiate card -19 imx-sgtl5000 sound.12: snd_soc_register_card failed (-19) Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 327b443..8b704ff 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -965,6 +965,7 @@ static bool sgtl5000_volatile(struct device *dev, unsigned int reg) switch (reg) { case SGTL5000_CHIP_ID: case SGTL5000_CHIP_ADCDAC_CTRL: + case SGTL5000_CHIP_ANA_POWER: case SGTL5000_CHIP_ANA_STATUS: return true; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Apply pin-enablement workaround to all Haswell HDMI codecs
by Takashi Iwai 08 May '13

08 May '13

This is a revised patch based on Mengdong Lin's fix patch, which is a supplement to a previous patch [1611a9c9: ALSA: hda - Add fixup for Haswell to enable all pin and convertor widgets]. Some Haswell BIOS will disable the 2nd and 3rd pin/covertor widgets when the HD-A controller changes state from D3 to D0. So when the controller resumes after a system or runtime suspend, these widgets are disabled and programming these widgets to D0 will cause H/W error and codec will not respond. In addition, we found out that some BIOS disables the pins at S3 although it shows up at boot. This confuses the driver utterly, and the hardware falls into the fatal communication error like the above. So in this patch, we apply intel_haswell_enable_all_pins() not only as a fixup to a certain device (with 8086:2010) but to all Haswell machines. The codec driver basically assumes that all pins are exposed, so it's anyway better to see them from the beginning. Even if all pins and converters are shown by this call, there should be no regression in practice: the pin default configurations are still kept, thus the disabled pins are handled as disabled by the driver properly. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 54 +++++++++++++++++++--------------------------- 1 file changed, 22 insertions(+), 32 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 32930e6..e12f7a0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1832,12 +1832,10 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, #define INTEL_EN_ALL_PIN_CVTS 0x01 /* enable 2nd & 3rd pins and convertors */ static void intel_haswell_enable_all_pins(struct hda_codec *codec, - const struct hda_fixup *fix, int action) + bool update_tree) { unsigned int vendor_param; - if (action != HDA_FIXUP_ACT_PRE_PROBE) - return; vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, INTEL_GET_VENDOR_VERB, 0); if (vendor_param == -1 || vendor_param & INTEL_EN_ALL_PIN_CVTS) @@ -1849,8 +1847,8 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, if (vendor_param == -1) return; - snd_hda_codec_update_widgets(codec); - return; + if (update_tree) + snd_hda_codec_update_widgets(codec); } static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) @@ -1868,30 +1866,20 @@ static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) INTEL_SET_VENDOR_VERB, vendor_param); } +/* Haswell needs to re-issue the vendor-specific verbs before turning to D0. + * Otherwise you may get severe h/w communication errors. + */ +static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (power_state == AC_PWRST_D0) { + intel_haswell_enable_all_pins(codec, false); + intel_haswell_fixup_enable_dp12(codec); + } - -/* 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_enable_all_pins, - }, -}; - + snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, power_state); + snd_hda_codec_set_power_to_all(codec, fg, power_state); +} static int patch_generic_hdmi(struct hda_codec *codec) { @@ -1904,11 +1892,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->spec = spec; hdmi_array_init(spec, 4); - snd_hda_pick_fixup(codec, hdmi_models, hdmi_fixup_tbl, hdmi_fixups); - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - - if (codec->vendor_id == 0x80862807) + if (codec->vendor_id == 0x80862807) { + intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); + } if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; @@ -1916,6 +1903,9 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + if (codec->vendor_id == 0x80862807) + codec->patch_ops.set_power_state = haswell_set_power_state; + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.8.2.1

1 0

[alsa-devel] [PATCH] ALSA: hda - Apply pin-enablement workaround to all Haswell HDMI codecs
by Takashi Iwai 08 May '13

08 May '13

This is a revised patch based on Mengdong Lin's fix patch, which is a supplement to a previous patch [1611a9c9: ALSA: hda - Add fixup for Haswell to enable all pin and convertor widgets]. Some Haswell BIOS will disable the 2nd and 3rd pin/covertor widgets when the HD-A controller changes state from D3 to D0. So when the controller resumes after a system or runtime suspend, these widgets are disabled and programming these widgets to D0 will cause H/W error and codec will not respond. In addition, we found out that some BIOS disables the pins at S3 although it shows up at boot. This confuses the driver utterly, and the hardware falls into the fatal communication error like the above. So in this patch, we apply intel_haswell_enable_all_pins() not only as a fixup to a certain device (with 8086:2010) but to all Haswell machines. The codec driver basically assumes that all pins are exposed, so it's anyway better to see them from the beginning. Even if all pins and converters are shown by this call, there should be no regression in practice: the pin default configurations are still kept, thus the disabled pins are handled as disabled by the driver properly. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 54 +++++++++++++++++++--------------------------- 1 file changed, 22 insertions(+), 32 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 32930e6..e12f7a0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1832,12 +1832,10 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, #define INTEL_EN_ALL_PIN_CVTS 0x01 /* enable 2nd & 3rd pins and convertors */ static void intel_haswell_enable_all_pins(struct hda_codec *codec, - const struct hda_fixup *fix, int action) + bool update_tree) { unsigned int vendor_param; - if (action != HDA_FIXUP_ACT_PRE_PROBE) - return; vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, INTEL_GET_VENDOR_VERB, 0); if (vendor_param == -1 || vendor_param & INTEL_EN_ALL_PIN_CVTS) @@ -1849,8 +1847,8 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, if (vendor_param == -1) return; - snd_hda_codec_update_widgets(codec); - return; + if (update_tree) + snd_hda_codec_update_widgets(codec); } static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) @@ -1868,30 +1866,20 @@ static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) INTEL_SET_VENDOR_VERB, vendor_param); } +/* Haswell needs to re-issue the vendor-specific verbs before turning to D0. + * Otherwise you may get severe h/w communication errors. + */ +static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (power_state == AC_PWRST_D0) { + intel_haswell_enable_all_pins(codec, false); + intel_haswell_fixup_enable_dp12(codec); + } - -/* 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_enable_all_pins, - }, -}; - + snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, power_state); + snd_hda_codec_set_power_to_all(codec, fg, power_state); +} static int patch_generic_hdmi(struct hda_codec *codec) { @@ -1904,11 +1892,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->spec = spec; hdmi_array_init(spec, 4); - snd_hda_pick_fixup(codec, hdmi_models, hdmi_fixup_tbl, hdmi_fixups); - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - - if (codec->vendor_id == 0x80862807) + if (codec->vendor_id == 0x80862807) { + intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); + } if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; @@ -1916,6 +1903,9 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + if (codec->vendor_id == 0x80862807) + codec->patch_ops.set_power_state = haswell_set_power_state; + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.8.2.1

1 0

Re: [alsa-devel] [RFC] ASoC: sgtl5000: Remove cache support
by Fabio Estevam 08 May '13

08 May '13

On Tue, May 7, 2013 at 5:06 PM, Matt Sealey <matt(a)genesi-usa.com> wrote: > Right, and in theory - I can't find an example of anyone implementing > VDDD as unconnected - if you float VDDD then the SGTL5000 will start Actually I can't find see anyone that implemented VDDD connected, except Efika. All these boards have sgtl5000 VDDD unconnected: mx51evk, mx28evk, mx53smd, mx35pdk, mx25pdk. mx6qsabrelite connects VDD to GND.

2 1

[alsa-devel] [PATCH v2] ALSA: hda - re-apply fixup when resuming haswell hdmi codec
by mengdong.lin＠intel.com 07 May '13

07 May '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch is a supplement to a previous patch: "ALSA: hda - Add fixup for Haswell to enable all pin and convertor widgets". Some Haswell BIOS will disable the 2nd and 3rd pin/covertor widgets when the HD-A controller changes state from D3 to D0. So when the controller resumes after a system or runtime suspend, these widgets are disabled and programming these widgets to D0 will cause H/W error and codec will not respond. So this patch adds a "set_power_state" ops for Haswell codec. Before setting the codec to D0, this function will apply a BIOS-specific fixup to enable the 2nd & 3rd pins/cvts if need. And since BIOS will also disable DP1.2, so this function will check and enable DP1.2 mode if need. 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 adb5dce..080398e 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1842,7 +1842,7 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, { unsigned int vendor_param; - if (action != HDA_FIXUP_ACT_PRE_PROBE) + if (action != HDA_FIXUP_ACT_PRE_PROBE && action != HDA_FIXUP_ACT_INIT) return; vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, INTEL_GET_VENDOR_VERB, 0); @@ -1855,7 +1855,8 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, if (vendor_param == -1) return; - snd_hda_codec_update_widgets(codec); + if (action == HDA_FIXUP_ACT_PRE_PROBE) + snd_hda_codec_update_widgets(codec); return; } @@ -1874,7 +1875,24 @@ static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) INTEL_SET_VENDOR_VERB, vendor_param); } +static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (AC_PWRST_D0 == power_state) { + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); + + intel_haswell_fixup_enable_dp12(codec); + + } + + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + snd_hda_codec_set_power_to_all(codec, fg, power_state); + + return; +} /* available models for fixup */ enum { @@ -1922,6 +1940,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + + if (codec->vendor_id == 0x80862807) + codec->patch_ops.set_power_state = haswell_set_power_state; + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.7.10.4

3 5

Re: [alsa-devel] [PATCH] ASoC: wm0010: fix error return code in wm0010_boot()
by Mark Brown 07 May '13

07 May '13

On Tue, May 07, 2013 at 07:38:52PM +0800, Wei Yongjun wrote: > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Fix to return -ENOMEM in the memory malloc of 'out' and 'img_swap' error > handling case instead of 0, as done elsewhere in this function. Applied, thanks. Please note my new e-mail address.

1 0

[alsa-devel] [PATCH] ASoC: sam9g20ek: Let device core handle pinctrl
by Mark Brown 07 May '13

07 May '13

Since commit ab78029 (drivers/pinctrl: grab default handles from device core) we can rely on device core for handling pinctrl so remove devm_pinctrl_get_select_default() from the driver. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/atmel/sam9g20_wm8731.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 2d6fbd0..802717e 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -38,8 +38,6 @@ #include <linux/platform_device.h> #include <linux/i2c.h> -#include <linux/pinctrl/consumer.h> - #include <linux/atmel-ssc.h> #include <sound/core.h> @@ -203,15 +201,8 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) struct device_node *codec_np, *cpu_np; struct clk *pllb; struct snd_soc_card *card = &snd_soc_at91sam9g20ek; - struct pinctrl *pinctrl; int ret; - pinctrl = devm_pinctrl_get_select_default(&pdev->dev); - if (IS_ERR(pinctrl)) { - dev_err(&pdev->dev, "Failed to request pinctrl for mck\n"); - return PTR_ERR(pinctrl); - } - if (!np) { if (!(machine_is_at91sam9g20ek() || machine_is_at91sam9g20ek_2mmc())) -- 1.7.10.4

3 3

[alsa-devel] [PATCH] ALSA: hda - re-apply fixup when resuming haswell hdmi codec
by mengdong.lin＠intel.com 07 May '13

07 May '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch is a supplement to a previous patch: "ALSA: hda - Add fixup for Haswell to enable all pin and convertor widgets". Some Haswell BIOS will disable the 2nd and 3rd pin/covertor widgets when the HD-A controller changes state from D3 to D0. So when the controller resumes after a system or runtime suspend, these widgets are disabled and programming these widgets to D0 will cause H/W error and codec will not respond. So this patch adds a "set_power_state" ops for Haswell codec. Before setting the codec to D0, this function will apply a BIOS-specific fixup to enable the 2nd & 3rd pins/cvts if need. And since BIOS will also disable DP1.2, so this function will check and enable DP1.2 mode if need. 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 adb5dce..9ecb55c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1842,7 +1842,7 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, { unsigned int vendor_param; - if (action != HDA_FIXUP_ACT_PRE_PROBE) + if (action != HDA_FIXUP_ACT_PRE_PROBE && action != HDA_FIXUP_ACT_INIT) return; vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, INTEL_GET_VENDOR_VERB, 0); @@ -1855,7 +1855,8 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, if (vendor_param == -1) return; - snd_hda_codec_update_widgets(codec); + if (action == HDA_FIXUP_ACT_PRE_PROBE) + snd_hda_codec_update_widgets(codec); return; } @@ -1874,7 +1875,27 @@ static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) INTEL_SET_VENDOR_VERB, vendor_param); } +static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + unsigned int state; + unsigned int vendor_param; + + if (AC_PWRST_D0 == power_state) { + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); + + intel_haswell_fixup_enable_dp12(codec); + + } + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + + snd_hda_codec_set_power_to_all(codec, fg, power_state); + + return; +} /* available models for fixup */ enum { @@ -1922,6 +1943,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + + if (codec->vendor_id == 0x80862807) + codec->patch_ops.set_power_state = haswell_set_power_state; + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.7.10.4

2 1

[alsa-devel] [PATCH]ALSA: HDA: Fix Oops caused by dereference NULL pointer
by Wang YanQing 07 May '13

07 May '13

The interrupt handler azx_interrupt will call azx_update_rirb, which may call snd_hda_queue_unsol_event, snd_hda_queue_unsol_event will dereference chip->bus pointer. The problem is we alloc chip->bus in azx_codec_create which will be called after we enable IRQ and enable unsolicited event in azx_probe. This will cause Oops due dereference NULL pointer. I meet it, good luck:) Signed-off-by: Wang YanQing <udknight(a)gmail.com> --- sound/pci/hda/hda_codec.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 622f726..9c76752 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -618,6 +618,9 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) unsigned int wp; trace_hda_unsol_event(bus, res, res_ex); + if (!bus) + return 0; + unsol = bus->unsol; if (!unsol) return 0; -- 1.7.12.4.dirty

2 1

[alsa-devel] [PATCH 1/6] ASoC: Samsung: Remove redundant platform_set_drvdata()
by Sachin Kamat 07 May '13

07 May '13

Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> Cc: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/soc/samsung/smdk_wm8580pcm.c | 1 - sound/soc/samsung/smdk_wm8994pcm.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index e43bd42..23a9204 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -176,7 +176,6 @@ static int snd_smdk_probe(struct platform_device *pdev) static int snd_smdk_remove(struct platform_device *pdev) { snd_soc_unregister_card(&smdk_pcm); - platform_set_drvdata(pdev, NULL); return 0; } diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 3688a32..0c84ca0 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -146,7 +146,6 @@ static int snd_smdk_probe(struct platform_device *pdev) static int snd_smdk_remove(struct platform_device *pdev) { snd_soc_unregister_card(&smdk_pcm); - platform_set_drvdata(pdev, NULL); return 0; } -- 1.7.9.5

4 12

[alsa-devel] [PATCH v2 12/13] ASoC: fsl: imx-audmux: Let device core handle pinctrl
by Fabio Estevam 07 May '13

07 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit ab78029 (drivers/pinctrl: grab default handles from device core), we can rely on device core for handling pinctrl. So remove devm_pinctrl_get_select_default() from the driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Rebased against topic/fsl branch sound/soc/fsl/imx-audmux.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 47f046a..e260f1f 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -26,7 +26,6 @@ #include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/slab.h> -#include <linux/pinctrl/consumer.h> #include "imx-audmux.h" @@ -247,7 +246,6 @@ EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); static int imx_audmux_probe(struct platform_device *pdev) { struct resource *res; - struct pinctrl *pinctrl; const struct of_device_id *of_id = of_match_device(imx_audmux_dt_ids, &pdev->dev); @@ -256,12 +254,6 @@ static int imx_audmux_probe(struct platform_device *pdev) if (IS_ERR(audmux_base)) return PTR_ERR(audmux_base); - pinctrl = devm_pinctrl_get_select_default(&pdev->dev); - if (IS_ERR(pinctrl)) { - dev_err(&pdev->dev, "setup pinctrl failed!"); - return PTR_ERR(pinctrl); - } - audmux_clk = devm_clk_get(&pdev->dev, "audmux"); if (IS_ERR(audmux_clk)) { dev_dbg(&pdev->dev, "cannot get clock: %ld\n", -- 1.7.9.5

2 1

Re: [alsa-devel] [RFC] ASoC: sgtl5000: Remove cache support
by Mark Brown 07 May '13

07 May '13

On Mon, May 06, 2013 at 03:19:27PM -0500, Matt Sealey wrote: > I have asked before and I will ask again; does anyone have any insight Not on the list you haven't as far as I can tell... this is the first I've heard of any concern about this code, I'd not be surprised if others were similarly unaware of any concerns. > The patch came from out of nowhere, and the comments seem to say that > the LDO gets enabled on every chip higher than a specific version. > Now, this is totally against the design documentation for the SGTL5000 > which specifically states that using the internal LDO is quite > possibly the *worst* power consuming case and NOT best practise for > putting the codec on a design, with the sole benefit being reduced > external component count. Are you sure it's purely about power consumption? A common application for LDOs in this situation is to clean up noise on the incoming power supply, improving rejection of noise. This is mostly irrelevant for digital supplies but can be desirable for analogue ones. Of course this looks like it's being used as a digital supply so... I have to say your explanation is a little curious, though - normally an LDO would require an external capacitor and so would either increase the component count (adding the capactior) or have no effect (if one is already present for decoupling). What is the supply for this LDO and where does the output go? > What happens if we just take out the internal LDO setup conditional on > version, and only initialize it if VDDD is missing? Do boards that If the LDO is enabled and connected to a supply which is also driven externally that's not a clever idea, the two regulators will both be trying to regulate the same supply. > I am not entirely sure how well the commit linked by Fabio would work > on mainline since if it uses hw_read/write it should also be updating > the cache with those values to be sure they can later be modified... > the BSP code does that then fills the cache with the existing values > afterwards. That commit is just bogus in the way it interacts with the cache, half of it looks awfully like it's trying to work around some breakage in the way the cache was being set up. > So maybe we don't disable the cache, or it gets ported to regmap, but > the order should be enable regulators in a non-cached way then fill > the cache AFTERWARDS - the code at ToT mainline doesn't fix the cache I'm sorry, this makes no sense to me. If we know the physical defaults for the device and we can get the device into that state then we should have no need to read from the cache. If we don't know what they are then they shouldn't be listed. > after it does regulator/power regs setup, I assume because cache > filling is now somehow automated in ASoC (and if this is the case, There is no cache filling to automate, if (as is the case here) register defaults are provided they're used. > then the cache io setup needs to be done after regulator/power setup, > the defaults list/cache needs to be updated for those registers before > any register caching io is done..). And maybe setting up the regulator > is kind of pointless anyway (meaning that whole caching/not-caching > thing just falls away?) Again this makes no sense, it seems like you're very confused here. The whole purpose of the register defaults is to reflect the default values of the registers. The infrastructure relies on this being the case, otherwise it will get confused for example during cache sync when it decides if a write to the device is required. I think the point that you lost me was your assumption that the regulator enable needs to be done without the cache. What makes you believe that this is the case? Note that as I said in another post the driver is currently totally broken with regard to anything that actually powers off the device, there's never any attempt made to restore the register cache.

1 0

[alsa-devel] [PATCH 13/13] ASoC: mxs: mxs-saif: Let device core handle pinctrl
by Fabio Estevam 06 May '13

06 May '13

Since commit ab78029 (drivers/pinctrl: grab default handles from device core), we can rely on device core for handling pinctrl. So remove devm_pinctrl_get_select_default() from the driver. Cc: Mark Brown <broonie(a)kernel.org> Cc: <alsa-devel(a)alsa-project.org> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 3a2aa1d..12723b0 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -27,7 +27,6 @@ #include <linux/delay.h> #include <linux/time.h> #include <linux/fsl/mxs-dma.h> -#include <linux/pinctrl/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -661,7 +660,6 @@ static int mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; - struct pinctrl *pinctrl; int ret = 0; struct device_node *master; @@ -701,12 +699,6 @@ static int mxs_saif_probe(struct platform_device *pdev) mxs_saif[saif->id] = saif; - pinctrl = devm_pinctrl_get_select_default(&pdev->dev); - if (IS_ERR(pinctrl)) { - ret = PTR_ERR(pinctrl); - return ret; - } - saif->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(saif->clk)) { ret = PTR_ERR(saif->clk); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 12/13] ASoC: fsl: imx-audmux: Let device core handle pinctrl
by Fabio Estevam 06 May '13

06 May '13

Since commit ab78029 (drivers/pinctrl: grab default handles from device core), we can rely on device core for handling pinctrl. So remove devm_pinctrl_get_select_default() from the driver. Cc: Mark Brown <broonie(a)kernel.org> Cc: <alsa-devel(a)alsa-project.org> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 3f333e5..22967c6 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -26,7 +26,6 @@ #include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/slab.h> -#include <linux/pinctrl/consumer.h> #include "imx-audmux.h" @@ -247,7 +246,6 @@ EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); static int imx_audmux_probe(struct platform_device *pdev) { struct resource *res; - struct pinctrl *pinctrl; const struct of_device_id *of_id = of_match_device(imx_audmux_dt_ids, &pdev->dev); @@ -256,12 +254,6 @@ static int imx_audmux_probe(struct platform_device *pdev) if (IS_ERR(audmux_base)) return PTR_ERR(audmux_base); - pinctrl = devm_pinctrl_get_select_default(&pdev->dev); - if (IS_ERR(pinctrl)) { - dev_err(&pdev->dev, "setup pinctrl failed!"); - return PTR_ERR(pinctrl); - } - audmux_clk = clk_get(&pdev->dev, "audmux"); if (IS_ERR(audmux_clk)) { dev_dbg(&pdev->dev, "cannot get clock: %ld\n", -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: mxs-sgtl5000: Remove unneeded 'ret' variable
by Fabio Estevam 06 May '13

06 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Variable 'ret' is not needed here, so just remove it. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index b1d9b5e..4f74b05 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -116,7 +116,7 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct device_node *saif_np[2], *codec_np; - int i, ret = 0; + int i; if (!np) return 1; /* no device tree */ @@ -142,7 +142,7 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) of_node_put(saif_np[0]); of_node_put(saif_np[1]); - return ret; + return 0; } static int mxs_sgtl5000_probe(struct platform_device *pdev) -- 1.7.9.5

3 5

[alsa-devel] [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
by mengdong.lin＠intel.com 06 May '13

06 May '13

From: Mengdong Lin <mengdong.lin(a)intel.com> In system resume, Haswell codec cannot be programmed to D0 before Gfx driver initializes the display pipeline and audio, which will trigger an unsol event on the pin with HDMI/DP cable connected. Otherwise, the connected pin will stay in D3 with right channel muted and thus no sound can be heard. This patch - adds a codec flag to delay resuming a codec. System resume will skip the codecs if this flag is set, and these codecs will be resumed on later codec access. - adds a set_power_state ops for Haswell HDMI codec. In a delayed resume, this ops will enable and wait for the unsol event, and then resume the codec. A 300ms timeout is set in case unsol event is lost. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 04b5738..bcb7205 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5508,11 +5508,15 @@ int snd_hda_resume(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { - hda_call_codec_resume(codec); + if (codec->support_delay_resume) + codec->resume_delayed = 1; + else + hda_call_codec_resume(codec); } return 0; } EXPORT_SYMBOL_HDA(snd_hda_resume); + #endif /* CONFIG_PM */ /* diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 23ca172..5b5e5f4 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -886,6 +886,8 @@ struct hda_codec { unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ unsigned int pm_down_notified:1; /* PM notified to controller */ unsigned int in_pm:1; /* suspend/resume being performed */ + unsigned int support_delay_resume:1; /* codec support delay resume */ + unsigned int resume_delayed:1; /* resume delayed by PM */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 78e1827..d116908 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -98,6 +98,14 @@ struct hdmi_spec { */ struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; + +#ifdef CONFIG_PM + /* + * Non-generic Intel Haswell specific + */ + unsigned int ready_to_resume:1; + wait_queue_head_t resume_wq; +#endif }; @@ -977,6 +985,13 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; +#ifdef CONFIG_PM + if (codec->resume_delayed) { + spec->ready_to_resume = 1; + wake_up(&spec->resume_wq); + } +#endif + hdmi_present_sense(&spec->pins[pin_idx], 1); snd_hda_jack_report_sync(codec); } @@ -1846,6 +1861,63 @@ static const struct hda_fixup hdmi_fixups[] = { }; +#ifdef CONFIG_PM +static void intel_haswell_wait_ready_to_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + spec->ready_to_resume = 0; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin; + hda_nid_t pin_nid; + struct hda_jack_tbl *jack; + + per_pin = &spec->pins[pin_idx]; + pin_nid = per_pin->pin_nid; + jack = snd_hda_jack_tbl_get(codec, pin_nid); + if (jack) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | jack->tag); + } + + wait_event_timeout(spec->resume_wq, + spec->ready_to_resume, msecs_to_jiffies(300)); + if (!spec->ready_to_resume) + snd_printd(KERN_WARNING "HDMI: Haswell not ready to resume\n"); +} + +static void intel_haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (codec->resume_delayed && power_state == AC_PWRST_D0) { + intel_haswell_wait_ready_to_resume(codec); + codec->resume_delayed = 0; + } + + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + + snd_hda_codec_set_power_to_all(codec, fg, power_state); +} + +static inline void intel_haswell_allow_delay_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + + init_waitqueue_head(&spec->resume_wq); + codec->support_delay_resume = 1; + + codec->patch_ops.set_power_state = + intel_haswell_set_power_state; +} +#else +define intel_haswell_allow_delay_resume NULL +#endif + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1868,6 +1940,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + + if (codec->vendor_id == 0x80862807) + intel_haswell_allow_delay_resume(codec); + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.7.10.4

4 51

[alsa-devel] [PATCH] alsactl: sprintf to snprintf prevent buffer overflow
by Doug Goldstein 06 May '13

06 May '13

sprintf() is a bit dangerous unless you explicitly know your type size and want to keep it in sync always. Its safer to just use snprintf() and ensure your string doesn't overflow and is NULL terminated. Signed-off-by: Doug Goldstein <cardoe(a)cardoe.com> --- This was written to fix the buffer overflow crash in alsactl with the 1.0.27 release, which is documented in Gentoo bug #468160 [1]. The issue is actually already fixed upstream by 95788fea25c1a59985828d4b91af0772d077600b however the fix just changes the size and continues to use sprintf() which IMHO is still dangerous. The attached patch will fix the issue in a safer manner for good. [1] https://bugs.gentoo.org/show_bug.cgi?id=468160 --- alsactl/lock.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/alsactl/lock.c b/alsactl/lock.c index fce208b..587a109 100644 --- a/alsactl/lock.c +++ b/alsactl/lock.c @@ -53,9 +53,9 @@ static int state_lock_(const char *file, int lock, int timeout) lck.l_len = 11; lck.l_pid = 0; if (lock) { - sprintf(lcktxt, "%10li\n", (long)getpid()); + snprintf(lcktxt, sizeof(lcktxt), "%10li\n", (long)getpid()); } else { - sprintf(lcktxt, "%10s\n", ""); + snprintf(lcktxt, sizeof(lcktxt), "%10s\n", ""); } while (fd < 0 && timeout-- > 0) { fd = open(nfile, O_RDWR); -- 1.8.2.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: mips/hal2: Remove redundant platform_set_drvdata()
by Sachin Kamat 06 May '13

06 May '13

Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> Cc: Thomas Bogendoerfer <tsbogend(a)alpha.fanken.de> --- sound/mips/hal2.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index 7420c59..2b7f6e8 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -922,7 +922,6 @@ static int hal2_remove(struct platform_device *pdev) struct snd_card *card = platform_get_drvdata(pdev); snd_card_free(card); - platform_set_drvdata(pdev, NULL); return 0; } -- 1.7.9.5

2 3

[alsa-devel] [PATCH resend 2/2] ALSA: mips/sgio2audio: Remove redundant platform_set_drvdata()
by Sachin Kamat 06 May '13

06 May '13

Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> Cc: Vivien Chappelier <vivien.chappelier(a)linux-mips.org> --- Resending becasue of a typo in patch title --- sound/mips/sgio2audio.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 01a03ef..cfe99ae 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -963,7 +963,6 @@ static int snd_sgio2audio_remove(struct platform_device *pdev) struct snd_card *card = platform_get_drvdata(pdev); snd_card_free(card); - platform_set_drvdata(pdev, NULL); return 0; } -- 1.7.9.5

1 0

[alsa-devel] [RFC 26/26] HID/ALSA: wiimote: add speaker support
by David Herrmann 05 May '13

05 May '13

Classic Wii Remotes provide a speaker on the device. We can stream PCM data, mute and change volume via special protocol requests and remote audio registers. The device supports several different formats, but fully understood are only signed 8-bit PCM and something that looks like 4-bit Yamaha ADPCM. Theoretically, we can set data-rates from 183Hz up to 12MHz, but a realistic range for Bluetooth l2cap is 1000-4000 Hz. Data is streamed as 20bytes blocks and must be sent in a constant rate. There is no way to read the current position. It would be way too slow, anyway. However, if we stream data too fast, older devices will report overrun errors. It is nearly impossible to avoid them as we have no fine-grained control over bluetooth radio. But that's probably also the reason why newer devices no longer send these errors. Note that 8bit PCM at such low rates has quite bad sound quality. You also need to use user-space programs like "sox/play" as other applications require rates >4000/8000Hz. The 4bit Yamaha ADPCM should improve sound-quality a lot, however, the kernel currently doesn't support this. So we either need to add ALSA support or add an internal encoder. But this will be implemented in follow-up patches so we first get the infrastructure ready. Thanks to David Henningsson for very helpful advice on the driver layout. The driver is also very loosely based on the sgio2audio driver by Vivien Chappelier <vivien.chappelier(a)linux-mips.org> and the dummy alsa sound driver. Cc: alsa-devel(a)alsa-project.org Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: David Herrmann <dh.herrmann(a)gmail.com> --- Hi This is my second revision of the Wii Remote sound-driver. It is based on the hotplugging rework in this series. It's the only patch that is marked as RFC but I thought I'd just include it in this series for completenes. If someone wants to give it a try, I used the "sox" tools for testing as most other tools require rates >4000/8000Hz. Sample created via: sox input.wav -b 8 -e signed -c 1 -r 2000 output.raw Play via: (prefix with: SOX_OPTS="--buffer 128" for better responsiveness) AUDIODEV=hw:1 play -e signed -b 8 -c 1 -r 2000 output.raw Comments welcome! I tested this with 4 different devices and all worked well. I also tested hotplugging while playing audio and I never got any deadlocks or oopses. It's the first time I worked with the sound-layer, but I'm now pretty sure I got everything right. Maybe I could even relax the heavy-locking but better be safe. Cheers David drivers/hid/Kconfig | 4 + drivers/hid/Makefile | 3 + drivers/hid/hid-wiimote-core.c | 66 +++- drivers/hid/hid-wiimote-modules.c | 5 + drivers/hid/hid-wiimote-speaker.c | 662 ++++++++++++++++++++++++++++++++++++++ drivers/hid/hid-wiimote.h | 17 + 6 files changed, 755 insertions(+), 2 deletions(-) create mode 100644 drivers/hid/hid-wiimote-speaker.c diff --git a/drivers/hid/Kconfig b/drivers/hid/Kconfig index e8ef86c..3bc81c8 100644 --- a/drivers/hid/Kconfig +++ b/drivers/hid/Kconfig @@ -699,6 +699,7 @@ config HID_WIIMOTE tristate "Nintendo Wii / Wii U peripherals" depends on HID depends on LEDS_CLASS + depends on !SND || HIGH_RES_TIMERS select POWER_SUPPLY select INPUT_FF_MEMLESS ---help--- @@ -715,6 +716,9 @@ config HID_WIIMOTE the Wii U Gamepad) might be supported in the future. But currently support is limited to Bluetooth based devices. + If Alsa sound support (CONFIG_SND) is enabled, this driver also provides + a sound card interface for Wii Remote speakers. + If unsure, say N. To compile this driver as a module, choose M here: the diff --git a/drivers/hid/Makefile b/drivers/hid/Makefile index 8b7106b..d149e10 100644 --- a/drivers/hid/Makefile +++ b/drivers/hid/Makefile @@ -32,6 +32,9 @@ hid-wiimote-y := hid-wiimote-core.o hid-wiimote-modules.o ifdef CONFIG_DEBUG_FS hid-wiimote-y += hid-wiimote-debug.o endif +ifdef CONFIG_SND + hid-wiimote-y += hid-wiimote-speaker.o +endif obj-$(CONFIG_HID_A4TECH) += hid-a4tech.o obj-$(CONFIG_HID_ACRUX) += hid-axff.o diff --git a/drivers/hid/hid-wiimote-core.c b/drivers/hid/hid-wiimote-core.c index 279a07d..ea07fc3 100644 --- a/drivers/hid/hid-wiimote-core.c +++ b/drivers/hid/hid-wiimote-core.c @@ -309,8 +309,8 @@ void wiiproto_req_ir2(struct wiimote_data *wdata, __u8 flags) #define wiiproto_req_weeprom(wdata, os, buf, sz) \ wiiproto_req_wmem((wdata), true, (os), (buf), (sz)) -static void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, - __u32 offset, const __u8 *buf, __u8 size) +void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, + __u32 offset, const __u8 *buf, __u8 size) { __u8 cmd[22]; @@ -359,6 +359,58 @@ void wiiproto_req_rmem(struct wiimote_data *wdata, bool eeprom, __u32 offset, wiimote_queue(wdata, cmd, sizeof(cmd)); } +void wiiproto_req_speaker(struct wiimote_data *wdata, bool on) +{ + __u8 cmd[2]; + + cmd[0] = WIIPROTO_REQ_SPEAKER; + cmd[1] = on ? 0x04 : 0x00; + + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); +} + +void wiiproto_req_mute(struct wiimote_data *wdata, bool on) +{ + __u8 cmd[2]; + + cmd[0] = WIIPROTO_REQ_MUTE; + cmd[1] = on ? 0x04 : 0x00; + + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); +} + +/* send audio data, possibly fragmented into two blocks */ +size_t wiiproto_req_audio(struct wiimote_data *wdata, const __u8 *b1, + size_t len1, const __u8 *b2, size_t len2) +{ + __u8 cmd[22]; + size_t pos, len; + + pos = 0; + len = min_t(size_t, len1, 20ULL); + memcpy(&cmd[2 + pos], b1, len); + pos += len; + + if (pos < 20 && len2) { + len = min_t(size_t, len2, 20ULL - pos); + memcpy(&cmd[2 + pos], b2, len); + pos += len; + } + + cmd[0] = WIIPROTO_REQ_AUDIO; + cmd[1] = (pos & 0xff) << 3; + + for ( ; pos < 20; ++pos) + cmd[pos] = 0; + + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); + + return pos; +} + /* requries the cmd-mutex to be held */ int wiimote_cmd_write(struct wiimote_data *wdata, __u32 offset, const __u8 *wmem, __u8 size) @@ -570,6 +622,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_LED4, WIIMOD_ACCEL, WIIMOD_IR, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_GEN10] = (const __u8[]){ @@ -582,6 +635,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_LED4, WIIMOD_ACCEL, WIIMOD_IR, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_GEN20] = (const __u8[]){ @@ -595,6 +649,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_ACCEL, WIIMOD_IR, WIIMOD_BUILTIN_MP, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_BALANCE_BOARD] = (const __u8[]) { @@ -1469,6 +1524,13 @@ static void handler_return(struct wiimote_data *wdata, const __u8 *payload) if (wiimote_cmd_pending(wdata, cmd, 0)) { wdata->state.cmd_err = err; wiimote_cmd_complete(wdata); + } else if (cmd == WIIPROTO_REQ_AUDIO && err == 4) { + /* If we stream audio data too fast, we get overrun errors + * from the device. However, on audio-period bounds, + * we have a very hard time sending audio at the exact rate + * as we have no control over the bluetooth l2cap IRQs. + * Hence, drop any overrun errors. */ + hid_dbg(wdata->hdev, "audio overrun error\n"); } else if (err) { hid_warn(wdata->hdev, "Remote error %hhu on req %hhu\n", err, cmd); diff --git a/drivers/hid/hid-wiimote-modules.c b/drivers/hid/hid-wiimote-modules.c index ab25a64..25483aa 100644 --- a/drivers/hid/hid-wiimote-modules.c +++ b/drivers/hid/hid-wiimote-modules.c @@ -2074,6 +2074,11 @@ const struct wiimod_ops *wiimod_table[WIIMOD_NUM] = { [WIIMOD_IR] = &wiimod_ir, [WIIMOD_BUILTIN_MP] = &wiimod_builtin_mp, [WIIMOD_NO_MP] = &wiimod_no_mp, +#ifdef CONFIG_SND + [WIIMOD_SPEAKER] = &wiimod_speaker, +#else + [WIIMOD_SPEAKER] = &wiimod_dummy, +#endif }; const struct wiimod_ops *wiimod_ext_table[WIIMOTE_EXT_NUM] = { diff --git a/drivers/hid/hid-wiimote-speaker.c b/drivers/hid/hid-wiimote-speaker.c new file mode 100644 index 0000000..89b7f9f --- /dev/null +++ b/drivers/hid/hid-wiimote-speaker.c @@ -0,0 +1,662 @@ +/* + * Driver for audio speakers of Nintendo Wii / Wii U peripherals + * Copyright (c) 2012-2013 David Herrmann <dh.herrmann(a)gmail.com> + */ + +/* + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + */ + +/* + * Audio Speakers + * Some Wii peripherals provide an audio speaker that supports 8bit PCM, 4bit + * Yamaha ADPCM and some other mostly unknown formats. Not all setup options + * are known, but we know how to setup an 8bit PCM or 4bit ADPCM stream and + * adjust volume. Data is sent as 20bytes chunks and needs to be streamed at + * a constant rate. + */ + +#include <linux/device.h> +#include <linux/hid.h> +#include <linux/hrtimer.h> +#include <linux/interrupt.h> +#include <linux/mutex.h> +#include <linux/spinlock.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include "hid-wiimote.h" + +struct wiimote_speaker { + spinlock_t lock; + struct snd_card *card; + struct wiimote_data *wdata; + unsigned int online : 1; + unsigned int mute : 1; + __u8 volume; + + unsigned long pos; + struct snd_pcm_substream *subs; + struct mutex runlock; + + struct hrtimer timer; + struct tasklet_struct tasklet; +}; + +enum wiimod_speaker_mode { + WIIMOD_SPEAKER_MODE_PCM8, + /* TODO: 4bit Yamaha ADPCM is currently not implemented */ + WIIMOD_SPEAKER_MODE_ADPCM4, +}; + +static int wiimod_speaker_enable(struct wiimote_speaker *speaker) +{ + struct wiimote_data *wdata = speaker->wdata; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (!speaker->online) { + speaker->online = 1; + wiiproto_req_speaker(wdata, true); + wiiproto_req_mute(wdata, speaker->mute); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + +static void wiimod_speaker_disable(struct wiimote_speaker *speaker) +{ + struct wiimote_data *wdata = speaker->wdata; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->online) { + speaker->online = 0; + wiiproto_req_speaker(wdata, false); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +static void wiimod_speaker_set_mute(struct wiimote_speaker *speaker, bool mute) +{ + struct wiimote_data *wdata = speaker->wdata; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->mute != mute) { + speaker->mute = mute; + if (speaker->online) + wiiproto_req_mute(wdata, mute); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +static void wiimod_speaker_set_volume(struct wiimote_speaker *speaker, + __u8 volume) +{ + struct wiimote_data *wdata = speaker->wdata; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->volume != volume) { + speaker->volume = volume; + if (speaker->online) + wiiproto_req_wmem(wdata, false, 0xa20005, &volume, + sizeof(volume)); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +/* Change speaker configuration. \mode can be one of WIIMOD_SPEAKER_MODE_*, + * \rate is the PCM sample rate and \volume is the requested volume. */ +static int wiimod_speaker_setup(struct wiimote_speaker *speaker, + __u8 mode, __u16 rate, __s16 volume) +{ + struct wiimote_data *wdata = speaker->wdata; + unsigned long flags; + __u8 config[7], m, wmem; + __u16 r; + int ret; + + if (!rate) + return -EINVAL; + + switch (mode) { + case WIIMOD_SPEAKER_MODE_PCM8: + r = 12000000ULL / rate; + m = 0x40; + break; + case WIIMOD_SPEAKER_MODE_ADPCM4: + r = 6000000ULL / rate; + m = 0x00; + break; + default: + return -EINVAL; + } + + config[0] = 0x00; + config[1] = m; + config[2] = r & 0x00ff; + config[3] = (r & 0xff00) >> 8; + config[4] = volume & 0xff; + config[5] = 0x00; + config[6] = 0x00; + + wiimote_cmd_acquire_noint(wdata); + + /* mute speaker during setup and read/write volume field */ + spin_lock_irqsave(&wdata->state.lock, flags); + + wiiproto_req_mute(wdata, true); + if (volume < 0) + config[4] = speaker->volume; + else + speaker->volume = volume; + + spin_unlock_irqrestore(&wdata->state.lock, flags); + + /* power speaker */ + wmem = 0x01; + ret = wiimote_cmd_write(wdata, 0xa20009, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + + /* prepare setup */ + wmem = 0x08; + ret = wiimote_cmd_write(wdata, 0xa20001, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + + /* write configuration */ + ret = wiimote_cmd_write(wdata, 0xa20001, config, sizeof(config)); + if (ret) + goto out_unlock; + + /* enable speaker */ + wmem = 0x01; + ret = wiimote_cmd_write(wdata, 0xa20008, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + +out_unlock: + wiimote_cmd_release(wdata); + return ret; +} + +/* returns true if a period has elapsed */ +static bool wiimod_speaker_push(struct wiimote_speaker *speaker) +{ + struct wiimote_data *wdata = speaker->wdata; + struct snd_pcm_runtime *runtime = speaker->subs->runtime; + unsigned long flags; + bool elapsed; + unsigned long buflen, plen, len; + __u8 *src; + + spin_lock_irqsave(&wdata->state.lock, flags); + + buflen = frames_to_bytes(runtime, runtime->buffer_size); + plen = frames_to_bytes(runtime, runtime->period_size); + src = runtime->dma_area; + + len = buflen - speaker->pos; + + /* no need to send data while muted or offline */ + if (speaker->online && !speaker->mute) { + if (len < 20) + wiiproto_req_audio(wdata, &src[speaker->pos], + len, src, 20 - len); + else + wiiproto_req_audio(wdata, &src[speaker->pos], + 20, NULL, 0); + } + + speaker->pos += 20; + speaker->pos %= buflen; + elapsed = (speaker->pos % plen) < 20; + + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return elapsed; +} + +/* timer handling */ + +static void wiimod_speaker_task(unsigned long data) +{ + struct wiimote_speaker *speaker = (void*)data; + unsigned long flags; + bool elapsed = false; + + spin_lock_irqsave(&speaker->lock, flags); + if (speaker->wdata) + elapsed = wiimod_speaker_push(speaker); + spin_unlock_irqrestore(&speaker->lock, flags); + + if (elapsed) + snd_pcm_period_elapsed(speaker->subs); +} + +static enum hrtimer_restart wiimod_speaker_tick(struct hrtimer *timer) +{ + struct wiimote_speaker *speaker = container_of(timer, + struct wiimote_speaker, + timer); + unsigned long missed; + __u64 ival; + + tasklet_schedule(&speaker->tasklet); + + ival = 1000000000ULL * 20U; + ival /= speaker->subs->runtime->rate; + + missed = hrtimer_forward_now(timer, ns_to_ktime(ival)); + if (missed > 1) + snd_printdd("wiimote: speaker: missed %lu timer interrupts\n", + missed - 1); + + return HRTIMER_RESTART; +} + +/* PCM layer */ + +static const struct snd_pcm_hardware wiimod_speaker_playback_hw = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S8, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 500, + .rate_max = 2000, /* TODO: >2000 only supported by 2nd gen */ + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = 32768, + .period_bytes_min = 128, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, +}; + +static int wiimod_speaker_playback_open(struct snd_pcm_substream *substream) +{ + struct wiimote_speaker *speaker = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long flags; + int ret; + + runtime->hw = wiimod_speaker_playback_hw; + + spin_lock_irqsave(&speaker->lock, flags); + ret = -ENODEV; + if (speaker->wdata) { + runtime->private_data = speaker->wdata; + ret = wiimod_speaker_enable(speaker); + } + spin_unlock_irqrestore(&speaker->lock, flags); + + return ret; +} + +static int wiimod_speaker_playback_close(struct snd_pcm_substream *substream) +{ + struct wiimote_speaker *speaker = snd_pcm_substream_chip(substream); + unsigned long flags; + + spin_lock_irqsave(&speaker->lock, flags); + if (speaker->wdata) + wiimod_speaker_disable(speaker); + spin_unlock_irqrestore(&speaker->lock, flags); + + hrtimer_cancel(&speaker->timer); + tasklet_kill(&speaker->tasklet); + + return 0; +} + +static int wiimod_speaker_playback_hw_params(struct snd_pcm_substream *subs, + struct snd_pcm_hw_params *hw) +{ + return snd_pcm_lib_alloc_vmalloc_buffer(subs, params_buffer_bytes(hw)); +} + +static int wiimod_speaker_playback_hw_free(struct snd_pcm_substream *subs) +{ + return snd_pcm_lib_free_vmalloc_buffer(subs); +} + +static int wiimod_speaker_playback_prepare(struct snd_pcm_substream *subs) +{ + struct wiimote_speaker *speaker = snd_pcm_substream_chip(subs); + struct wiimote_data *wdata; + struct snd_pcm_runtime *runtime = subs->runtime; + int ret, online; + unsigned long flags; + + /* runlock synchronizes with device hotplugging */ + mutex_lock(&speaker->runlock); + + spin_lock_irqsave(&speaker->lock, flags); + wdata = speaker->wdata; + online = speaker->online; + spin_unlock_irqrestore(&speaker->lock, flags); + + if (!wdata || !online) { + ret = -ENODEV; + goto unlock; + } + + spin_lock_irqsave(&wdata->state.lock, flags); + speaker->pos = 0; + speaker->subs = subs; + spin_unlock_irqrestore(&wdata->state.lock, flags); + + /* Perform speaker setup. This may take a few milliseconds and the + * handlers perform synchronous network operations so this + * may sleep. */ + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S8: + ret = wiimod_speaker_setup(speaker, WIIMOD_SPEAKER_MODE_PCM8, + runtime->rate, -1); + break; + default: + ret = -EINVAL; + } + +unlock: + mutex_unlock(&speaker->runlock); + return ret; +} + +static int wiimod_speaker_playback_trigger(struct snd_pcm_substream *subs, + int cmd) +{ + struct wiimote_speaker *speaker = snd_pcm_substream_chip(subs); + struct snd_pcm_runtime *runtime = subs->runtime; + struct wiimote_data *wdata; + unsigned long flags; + __u64 ival; + + spin_lock_irqsave(&speaker->lock, flags); + + wdata = speaker->wdata; + if (!wdata || !speaker->online) + goto unlock; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* unmute device on start if not muted by user-space */ + spin_lock(&wdata->state.lock); + if (!speaker->mute) + wiiproto_req_mute(wdata, false); + spin_unlock(&wdata->state.lock); + + ival = 20U * 1000000000ULL / runtime->rate; + hrtimer_start(&speaker->timer, ktime_set(0, ival), + HRTIMER_MODE_REL); + break; + case SNDRV_PCM_TRIGGER_STOP: + /* mute device when stopping transmission */ + spin_lock(&wdata->state.lock); + wiiproto_req_mute(wdata, true); + spin_unlock(&wdata->state.lock); + + hrtimer_cancel(&speaker->timer); + break; + case SNDRV_PCM_TRIGGER_RESUME: + /* unmute device on start if not muted by user-space */ + spin_lock(&wdata->state.lock); + if (!speaker->mute) + wiiproto_req_mute(wdata, false); + spin_unlock(&wdata->state.lock); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + /* mute device when stopping transmission */ + spin_lock(&wdata->state.lock); + wiiproto_req_mute(wdata, true); + spin_unlock(&wdata->state.lock); + break; + } + +unlock: + spin_unlock_irqrestore(&speaker->lock, flags); + return 0; +} + +static snd_pcm_uframes_t wiimod_speaker_playback_pointer(struct snd_pcm_substream *subs) +{ + struct wiimote_speaker *speaker = snd_pcm_substream_chip(subs); + + return bytes_to_frames(subs->runtime, speaker->pos); +} + +static struct snd_pcm_ops wiimod_speaker_playback_ops = { + .open = wiimod_speaker_playback_open, + .close = wiimod_speaker_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = wiimod_speaker_playback_hw_params, + .hw_free = wiimod_speaker_playback_hw_free, + .prepare = wiimod_speaker_playback_prepare, + .trigger = wiimod_speaker_playback_trigger, + .pointer = wiimod_speaker_playback_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, +}; + +/* volume control */ + +static int wiimod_speaker_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 1; + info->value.integer.min = 0; + info->value.integer.max = 0xff; + + return 0; +} + +static int wiimod_speaker_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_speaker *speaker = snd_kcontrol_chip(kcontrol); + + val->value.integer.value[0] = speaker->volume; + + return 0; +} + +static int wiimod_speaker_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_speaker *speaker = snd_kcontrol_chip(kcontrol); + unsigned long value, flags; + + value = val->value.integer.value[0]; + if (value > 0xff) + value = 0xff; + + spin_lock_irqsave(&speaker->lock, flags); + if (speaker->wdata) + wiimod_speaker_set_volume(speaker, value); + spin_unlock_irqrestore(&speaker->lock, flags); + + return 0; +} + +static const struct snd_kcontrol_new wiimod_speaker_volume = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = wiimod_speaker_volume_info, + .get = wiimod_speaker_volume_get, + .put = wiimod_speaker_volume_put, +}; + +/* mute control */ + +static int wiimod_speaker_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_speaker *speaker = snd_kcontrol_chip(kcontrol); + + val->value.integer.value[0] = !speaker->mute; + + return 0; +} + +static int wiimod_speaker_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_speaker *speaker = snd_kcontrol_chip(kcontrol); + unsigned long flags; + + spin_lock_irqsave(&speaker->lock, flags); + if (speaker->wdata) + wiimod_speaker_set_mute(speaker, + !val->value.integer.value[0]); + spin_unlock_irqrestore(&speaker->lock, flags); + + return 0; +} + +static const struct snd_kcontrol_new wiimod_speaker_mute = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_mono_info, + .get = wiimod_speaker_mute_get, + .put = wiimod_speaker_mute_put, +}; + +/* initialization and setup */ + +static int wiimod_speaker_probe(const struct wiimod_ops *ops, + struct wiimote_data *wdata) +{ + int ret; + struct wiimote_speaker *speaker; + struct snd_card *card; + struct snd_kcontrol *kcontrol; + struct snd_pcm *pcm; + + /* create sound card device */ + ret = snd_card_create(-1, NULL, THIS_MODULE, + sizeof(struct wiimote_speaker), &card); + if (ret) + return ret; + speaker = card->private_data; + + wdata->speaker = speaker; + speaker->wdata = wdata; + speaker->card = card; + speaker->mute = 1; + speaker->volume = 0xff; + strcpy(card->driver, "hid-wiimote"); + strcpy(card->shortname, "wiimote"); + strcpy(card->longname, "Nintendo Wii Remote speaker"); + + spin_lock_init(&speaker->lock); + mutex_init(&speaker->runlock); + hrtimer_init(&speaker->timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + speaker->timer.function = wiimod_speaker_tick; + tasklet_init(&speaker->tasklet, wiimod_speaker_task, + (unsigned long)speaker); + + /* create volume control */ + kcontrol = snd_ctl_new1(&wiimod_speaker_volume, speaker); + if (!kcontrol) { + ret = -ENOMEM; + goto err_free; + } + + ret = snd_ctl_add(card, kcontrol); + if (ret) { + snd_ctl_free_one(kcontrol); + goto err_free; + } + + /* create mute control */ + kcontrol = snd_ctl_new1(&wiimod_speaker_mute, speaker); + if (!kcontrol) { + ret = -ENOMEM; + goto err_free; + } + + ret = snd_ctl_add(card, kcontrol); + if (ret) { + snd_ctl_free_one(kcontrol); + goto err_free; + } + + /* create PCM sub-device for playback */ + ret = snd_pcm_new(card, "Speaker", 0, 1, 0, &pcm); + if (ret) + goto err_free; + + pcm->private_data = speaker; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &wiimod_speaker_playback_ops); + + /* register sound card */ + snd_card_set_dev(card, &wdata->hdev->dev); + ret = snd_card_register(card); + if (ret) + goto err_free; + + return 0; + +err_free: + snd_card_free(card); + wdata->speaker = NULL; + return ret; +} + +static void wiimod_speaker_remove(const struct wiimod_ops *ops, + struct wiimote_data *wdata) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + if (!speaker) + return; + + mutex_lock(&speaker->runlock); + spin_lock_irqsave(&speaker->lock, flags); + + wdata->speaker = NULL; + speaker->online = 0; + speaker->wdata = NULL; + + spin_lock(&wdata->state.lock); + wiiproto_req_mute(wdata, true); + wiiproto_req_speaker(wdata, false); + spin_unlock(&wdata->state.lock); + + spin_unlock_irqrestore(&speaker->lock, flags); + mutex_unlock(&speaker->runlock); + + hrtimer_cancel(&speaker->timer); + tasklet_kill(&speaker->tasklet); + snd_card_free_when_closed(speaker->card); +} + +const struct wiimod_ops wiimod_speaker = { + .flags = 0, + .arg = 0, + .probe = wiimod_speaker_probe, + .remove = wiimod_speaker_remove, +}; diff --git a/drivers/hid/hid-wiimote.h b/drivers/hid/hid-wiimote.h index f1474f3..63a9244 100644 --- a/drivers/hid/hid-wiimote.h +++ b/drivers/hid/hid-wiimote.h @@ -147,6 +147,7 @@ struct wiimote_data { struct input_dev *mp; struct timer_list timer; struct wiimote_debug *debug; + struct wiimote_speaker *speaker; union { struct input_dev *input; @@ -171,6 +172,7 @@ enum wiimod_module { WIIMOD_IR, WIIMOD_BUILTIN_MP, WIIMOD_NO_MP, + WIIMOD_SPEAKER, WIIMOD_NUM, WIIMOD_NULL = WIIMOD_NUM, }; @@ -207,9 +209,12 @@ enum wiiproto_reqs { WIIPROTO_REQ_LED = 0x11, WIIPROTO_REQ_DRM = 0x12, WIIPROTO_REQ_IR1 = 0x13, + WIIPROTO_REQ_SPEAKER = 0x14, WIIPROTO_REQ_SREQ = 0x15, WIIPROTO_REQ_WMEM = 0x16, WIIPROTO_REQ_RMEM = 0x17, + WIIPROTO_REQ_AUDIO = 0x18, + WIIPROTO_REQ_MUTE = 0x19, WIIPROTO_REQ_IR2 = 0x1a, WIIPROTO_REQ_STATUS = 0x20, WIIPROTO_REQ_DATA = 0x21, @@ -263,6 +268,12 @@ extern void wiiproto_req_status(struct wiimote_data *wdata); extern void wiiproto_req_accel(struct wiimote_data *wdata, __u8 accel); extern void wiiproto_req_ir1(struct wiimote_data *wdata, __u8 flags); extern void wiiproto_req_ir2(struct wiimote_data *wdata, __u8 flags); +extern void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, + __u32 offset, const __u8 *buf, __u8 size); +extern void wiiproto_req_speaker(struct wiimote_data *wdata, bool on); +extern void wiiproto_req_mute(struct wiimote_data *wdata, bool on); +extern size_t wiiproto_req_audio(struct wiimote_data *wdata, const __u8 *b1, + size_t len1, const __u8 *b2, size_t len2); extern int wiimote_cmd_write(struct wiimote_data *wdata, __u32 offset, const __u8 *wmem, __u8 size); extern ssize_t wiimote_cmd_read(struct wiimote_data *wdata, __u32 offset, @@ -287,6 +298,12 @@ static inline void wiidebug_deinit(void *u) { } #endif +#ifdef CONFIG_SND + +extern const struct wiimod_ops wiimod_speaker; + +#endif + /* requires the state.lock spinlock to be held */ static inline bool wiimote_cmd_pending(struct wiimote_data *wdata, int cmd, __u32 opt) -- 1.8.2.2

1 0

[alsa-devel] [RFC] Sound/HID: wiimote: add speaker support
by David Herrmann 05 May '13

05 May '13

Classic Wii Remotes provide a speaker on the device. We can stream PCM data, mute and change volume via special protocol requests and remote audio registers. The device supports several different formats, but fully understood are only signed 8-bit PCM and something that looks like 4-bit Yamaha ADPCM. Theoretically, we can set data-rates from 183Hz up to 12MHz, but a realistic range for Bluetooth l2cap is 1500-4000 Hz. Data is streamed as 20bytes blocks and must be sent in a constant rate. There is no way to read the current position. It would be way too slow, anyway. Signed-off-by: David Herrmann <dh.herrmann(a)gmail.com> --- Hi I'm reworking large parts of the HID Nintendo Wii Remote driver and wanted to add speaker support. I have never worked with the sound subsystem until now and need some help. The Wii Remote is a Bluetooth gamepad that features a small speaker on the remote. All information was gathered via reverse-engineering so I cannot tell you any specifics about the hardware, sorry. What I figured out so far is how to set up the speaker, stream data, set volume and mute the speaker. Supported formats are: signed 8-bit PCM 4-bit Yamaha ADPCM I implemented only the 8-bit PCM as I couldn't figure out whether the kernel supports the other format? The wiiproto_* helpers in hid-wiimote-core.c are used to send the requests. Data is streamed as 20-bytes packets and must be sent at a constant rate. I would really appreciate if someone can have a look at hid-wiimote-speaker.c and help me fill the gaps. I figured out how to write the basic snd_card management and PCM setup, but I am stuck with buffer management now. I have a function called wiiproto_req_audio() which sends up to 20bytes of PCM data to the device. However, I cannot figure out how to integrate that into the snd_pcm object. The problem is that all other drivers I found have large DMA buffers which are filled whenever an interrupt signals more data. However, I don't have this setup but instead I need a timer (does the sound-core provide that?) which causes the "copy" callback to be called with 20bytes of data at a constant rate. Any help welcome! And if this whole idea is stupid, please tell me! ;) Thanks David drivers/hid/Kconfig | 3 + drivers/hid/Makefile | 3 + drivers/hid/hid-wiimote-core.c | 54 +++- drivers/hid/hid-wiimote-modules.c | 5 + drivers/hid/hid-wiimote-speaker.c | 539 ++++++++++++++++++++++++++++++++++++++ drivers/hid/hid-wiimote.h | 17 ++ 6 files changed, 619 insertions(+), 2 deletions(-) create mode 100644 drivers/hid/hid-wiimote-speaker.c diff --git a/drivers/hid/Kconfig b/drivers/hid/Kconfig index e8ef86c..0b4d7e2 100644 --- a/drivers/hid/Kconfig +++ b/drivers/hid/Kconfig @@ -715,6 +715,9 @@ config HID_WIIMOTE the Wii U Gamepad) might be supported in the future. But currently support is limited to Bluetooth based devices. + If Alsa sound support (CONFIG_SND) is enabled, this driver also provides + a sound card interface for Wii Remote speakers. + If unsure, say N. To compile this driver as a module, choose M here: the diff --git a/drivers/hid/Makefile b/drivers/hid/Makefile index 8b7106b..d149e10 100644 --- a/drivers/hid/Makefile +++ b/drivers/hid/Makefile @@ -32,6 +32,9 @@ hid-wiimote-y := hid-wiimote-core.o hid-wiimote-modules.o ifdef CONFIG_DEBUG_FS hid-wiimote-y += hid-wiimote-debug.o endif +ifdef CONFIG_SND + hid-wiimote-y += hid-wiimote-speaker.o +endif obj-$(CONFIG_HID_A4TECH) += hid-a4tech.o obj-$(CONFIG_HID_ACRUX) += hid-axff.o diff --git a/drivers/hid/hid-wiimote-core.c b/drivers/hid/hid-wiimote-core.c index 906c146..53f1e88 100644 --- a/drivers/hid/hid-wiimote-core.c +++ b/drivers/hid/hid-wiimote-core.c @@ -17,6 +17,7 @@ #include <linux/module.h> #include <linux/mutex.h> #include <linux/spinlock.h> +#include <linux/uaccess.h> #include "hid-ids.h" #include "hid-wiimote.h" @@ -309,8 +310,8 @@ void wiiproto_req_ir2(struct wiimote_data *wdata, __u8 flags) #define wiiproto_req_weeprom(wdata, os, buf, sz) \ wiiproto_req_wmem((wdata), true, (os), (buf), (sz)) -static void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, - __u32 offset, const __u8 *buf, __u8 size) +void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, + __u32 offset, const __u8 *buf, __u8 size) { __u8 cmd[22]; @@ -359,6 +360,52 @@ void wiiproto_req_rmem(struct wiimote_data *wdata, bool eeprom, __u32 offset, wiimote_queue(wdata, cmd, sizeof(cmd)); } +void wiiproto_req_speaker(struct wiimote_data *wdata, bool on) +{ + __u8 cmd[2]; + + cmd[0] = WIIPROTO_REQ_SPEAKER; + cmd[1] = on ? 0x04 : 0x00; + + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); +} + +void wiiproto_req_mute(struct wiimote_data *wdata, bool on) +{ + __u8 cmd[2]; + + cmd[0] = WIIPROTO_REQ_MUTE; + cmd[1] = on ? 0x04 : 0x00; + + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); +} + +/* must not be called from atomic contexts */ +int wiiproto_req_audio_user(struct wiimote_data *wdata, + void __user *buf, size_t len) +{ + unsigned long flags; + __u8 cmd[22]; + + if (len > 20) + len = 20; + + cmd[0] = WIIPROTO_REQ_AUDIO; + cmd[1] = (len & 0xff) << 3; + + if (copy_from_user(&cmd[2], buf, len)) + return -EFAULT; + + spin_lock_irqsave(&wdata->state.lock, flags); + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + /* requries the cmd-mutex to be held */ int wiimote_cmd_write(struct wiimote_data *wdata, __u32 offset, const __u8 *wmem, __u8 size) @@ -570,6 +617,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_LED4, WIIMOD_ACCEL, WIIMOD_IR, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_GEN10] = (const __u8[]){ @@ -582,6 +630,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_LED4, WIIMOD_ACCEL, WIIMOD_IR, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_GEN20] = (const __u8[]){ @@ -595,6 +644,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_ACCEL, WIIMOD_IR, WIIMOD_BUILTIN_MP, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_BALANCE_BOARD] = (const __u8[]) { diff --git a/drivers/hid/hid-wiimote-modules.c b/drivers/hid/hid-wiimote-modules.c index 4ce0e02..7acf721 100644 --- a/drivers/hid/hid-wiimote-modules.c +++ b/drivers/hid/hid-wiimote-modules.c @@ -2072,6 +2072,11 @@ const struct wiimod_ops *wiimod_table[WIIMOD_NUM] = { [WIIMOD_IR] = &wiimod_ir, [WIIMOD_BUILTIN_MP] = &wiimod_builtin_mp, [WIIMOD_NO_MP] = &wiimod_no_mp, +#ifdef CONFIG_SND + [WIIMOD_SPEAKER] = &wiimod_speaker, +#else + [WIIMOD_SPEAKER] = &wiimod_dummy, +#endif }; const struct wiimod_ops *wiimod_ext_table[WIIMOTE_EXT_NUM] = { diff --git a/drivers/hid/hid-wiimote-speaker.c b/drivers/hid/hid-wiimote-speaker.c new file mode 100644 index 0000000..ecec5d9 --- /dev/null +++ b/drivers/hid/hid-wiimote-speaker.c @@ -0,0 +1,539 @@ +/* + * Driver for audio speakers of Nintendo Wii / Wii U peripherals + * Copyright (c) 2012-2013 David Herrmann <dh.herrmann(a)gmail.com> + */ + +/* + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + */ + +/* + * Audio Speakers + * Some Wii peripherals provide an audio speaker that supports 8bit PCM and + * some other mostly unknown formats. Not all setup options are known, but we + * know how to setup an 8bit PCM or 4bit ADPCM stream and adjust volume. Data + * is sent as 20bytes chunks and needs to be streamed at a constant rate. + */ + +#include <linux/device.h> +#include <linux/hid.h> +#include <linux/mutex.h> +#include <linux/spinlock.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include "hid-wiimote.h" + +struct wiimote_speaker { + struct snd_card *card; + unsigned int online : 1; + unsigned int mute : 1; + __u8 volume; +}; + +enum wiimod_speaker_mode { + WIIMOD_SPEAKER_MODE_PCM8, + WIIMOD_SPEAKER_MODE_ADPCM4, +}; + +static int wiimod_speaker_enable(struct wiimote_data *wdata) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (!speaker->online) { + speaker->online = 1; + wiiproto_req_speaker(wdata, true); + wiiproto_req_mute(wdata, speaker->mute); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + +static void wiimod_speaker_disable(struct wiimote_data *wdata) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->online) { + speaker->online = 0; + wiiproto_req_speaker(wdata, false); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +static void wiimod_speaker_set_mute(struct wiimote_data *wdata, bool mute) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->mute != mute) { + speaker->mute = mute; + if (speaker->online) + wiiproto_req_mute(wdata, mute); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +static void wiimod_speaker_set_volume(struct wiimote_data *wdata, __u8 volume) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->volume != volume) { + speaker->volume = volume; + if (speaker->online) + wiiproto_req_wmem(wdata, false, 0xa20005, &volume, + sizeof(volume)); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +/* Change speaker configuration. \mode can be one of WIIMOD_SPEAKER_MODE_*, + * \rate is the PCM sample rate and \volume is the requested volume. */ +static int wiimod_speaker_setup(struct wiimote_data *wdata, + __u8 mode, __u16 rate, __s16 volume) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + __u8 config[7], m, wmem; + __u16 r; + int ret; + + if (!rate) + return -EINVAL; + + switch (mode) { + case WIIMOD_SPEAKER_MODE_PCM8: + r = 12000000ULL / rate; + m = 0x40; + break; + case WIIMOD_SPEAKER_MODE_ADPCM4: + r = 6000000ULL / rate; + m = 0x00; + break; + default: + return -EINVAL; + } + + config[0] = 0x00; + config[1] = m; + config[2] = r & 0x00ff; + config[3] = (r & 0xff00) >> 8; + config[4] = volume & 0xff; + config[5] = 0x00; + config[6] = 0x00; + + wiimote_cmd_acquire_noint(wdata); + + /* mute speaker during setup and read/write volume field */ + spin_lock_irqsave(&wdata->state.lock, flags); + + wiiproto_req_mute(wdata, true); + if (volume < 0) + config[4] = speaker->volume; + else + speaker->volume = volume; + + spin_unlock_irqrestore(&wdata->state.lock, flags); + + /* power speaker */ + wmem = 0x01; + ret = wiimote_cmd_write(wdata, 0xa20009, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + + /* prepare setup */ + wmem = 0x08; + ret = wiimote_cmd_write(wdata, 0xa20001, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + + /* write configuration */ + ret = wiimote_cmd_write(wdata, 0xa20001, config, sizeof(config)); + if (ret) + goto out_unlock; + + /* enable speaker */ + wmem = 0x01; + ret = wiimote_cmd_write(wdata, 0xa20008, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + + /* unmute speaker after setup if not muted */ + spin_lock_irqsave(&wdata->state.lock, flags); + if (!speaker->mute) + wiiproto_req_mute(wdata, false); + spin_unlock_irqrestore(&wdata->state.lock, flags); + +out_unlock: + wiimote_cmd_release(wdata); + return ret; +} + +/* PCM layer */ + +static const struct snd_pcm_hardware wiimod_speaker_playback_hw = { + .info = SNDRV_PCM_INFO_NONINTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S8, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 1500, + .rate_max = 4000, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = 20, + .period_bytes_min = 20, + .period_bytes_max = 20, + .periods_min = 1, + .periods_max = 1, +}; + +static int wiimod_speaker_playback_open(struct snd_pcm_substream *substream) +{ + struct wiimote_data *wdata = snd_pcm_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = wiimod_speaker_playback_hw; + runtime->private_data = wdata; + + return wiimod_speaker_enable(wdata); +} + +static int wiimod_speaker_playback_close(struct snd_pcm_substream *substream) +{ + struct wiimote_data *wdata = snd_pcm_chip(substream); + + wiimod_speaker_disable(wdata); + + return 0; +} + +static int wiimod_speaker_playback_hw_params(struct snd_pcm_substream *subs, + struct snd_pcm_hw_params *hw) +{ + /* TODO: anything to do here? */ + + return 0; +} + +static int wiimod_speaker_playback_hw_free(struct snd_pcm_substream *subs) +{ + return 0; +} + +static int wiimod_speaker_playback_prepare(struct snd_pcm_substream *subs) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + struct snd_pcm_runtime *runtime = subs->runtime; + size_t buf_size, period_size, periods; + int ret; + + /* TODO: If I have set the hw-params to a fixed 20 bytes buffer, do + * I actually need to do any computations here? */ + buf_size = frames_to_bytes(runtime, runtime->buffer_size); + period_size = frames_to_bytes(runtime, runtime->period_size); + periods = runtime->periods; + + switch (runtime->format) { + case SNDRV_PCM_FMTBIT_S8: + ret = wiimod_speaker_setup(wdata, WIIMOD_SPEAKER_MODE_PCM8, + runtime->rate, -1); + break; + default: + return -EINVAL; + } + + return ret; +} + +static int wiimod_speaker_playback_trigger(struct snd_pcm_substream *subs, + int cmd) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + /* unmute device on start if not muted by user-space */ + spin_lock_irqsave(&wdata->state.lock, flags); + if (!speaker->mute) + wiiproto_req_mute(wdata, false); + spin_unlock_irqrestore(&wdata->state.lock, flags); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + /* mute device when stopping transmission */ + spin_lock_irqsave(&wdata->state.lock, flags); + wiiproto_req_mute(wdata, true); + spin_unlock_irqrestore(&wdata->state.lock, flags); + break; + } + + return 0; +} + +static snd_pcm_uframes_t wiimod_speaker_playback_pointer(struct snd_pcm_substream *subs) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + unsigned long pos; + + /* There is no way to read the current position. Even if there was a + * way to do that, it would be so slow that it would be useless. + * TODO: Should we interpolate via a timer? Is this function actually + * needed? When is it called? */ + pos = 0; + + pos %= 20; + + return bytes_to_frames(subs->runtime, pos); +} + +static int wiimod_speaker_playback_copy(struct snd_pcm_substream *subs, + int channel, snd_pcm_uframes_t pos, + void __user *buf, + snd_pcm_uframes_t count) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + struct snd_pcm_runtime *runtime = subs->runtime; + + count = frames_to_bytes(runtime, count); + pos = frames_to_bytes(runtime, pos); + + /* TODO: copy from "buf" "count" bytes to "protobuf + pos" + * With the given hw-params, can pos be non-zero? Can count != 20? + * If not, should I simply send a 20byte frame to the device here? */ + + return 0; +} + +static int wiimod_speaker_playback_silence(struct snd_pcm_substream *subs, + int channel, snd_pcm_uframes_t pos, + snd_pcm_uframes_t count) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + struct snd_pcm_runtime *runtime = subs->runtime; + + count = frames_to_bytes(runtime, count); + pos = frames_to_bytes(runtime, pos); + + /* TODO: set "count" bytes of "protobuf + pos" to zero + * Can "pos" actually be non-zero with the given hw-params? If not, + * can I simply send a 20byte zeroed frame to the remote device? */ + + return 0; +} + +/* TODO: is there a reason this cannot be "const"? */ +static struct snd_pcm_ops wiimod_speaker_playback_ops = { + .open = wiimod_speaker_playback_open, + .close = wiimod_speaker_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = wiimod_speaker_playback_hw_params, + .hw_free = wiimod_speaker_playback_hw_free, + .prepare = wiimod_speaker_playback_prepare, + .trigger = wiimod_speaker_playback_trigger, + .pointer = wiimod_speaker_playback_pointer, + .copy = wiimod_speaker_playback_copy, + .silence = wiimod_speaker_playback_silence, +}; + +/* volume control */ + +static int wiimod_speaker_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 1; + info->value.integer.min = 0; + info->value.integer.max = 0xff; + + return 0; +} + +static int wiimod_speaker_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_data *wdata = snd_kcontrol_chip(kcontrol); + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + val->value.integer.value[0] = speaker->volume; + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + +static int wiimod_speaker_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_data *wdata = snd_kcontrol_chip(kcontrol); + unsigned long value; + + value = val->value.integer.value[0]; + if (value > 0xff) + value = 0xff; + + wiimod_speaker_set_volume(wdata, value); + + return 0; +} + +static const struct snd_kcontrol_new wiimod_speaker_volume = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = wiimod_speaker_volume_info, + .get = wiimod_speaker_volume_get, + .put = wiimod_speaker_volume_put, +}; + +/* mute control */ + +static int wiimod_speaker_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_data *wdata = snd_kcontrol_chip(kcontrol); + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + val->value.integer.value[0] = !!speaker->mute; + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + +static int wiimod_speaker_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_data *wdata = snd_kcontrol_chip(kcontrol); + + wiimod_speaker_set_mute(wdata, val->value.integer.value[0]); + + return 0; +} + +/* TODO: Is *_IFACE_CARD the right interface? */ +static const struct snd_kcontrol_new wiimod_speaker_mute = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "PCM Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_mono_info, + .get = wiimod_speaker_mute_get, + .put = wiimod_speaker_mute_put, +}; + +/* initialization and setup */ + +static int wiimod_speaker_probe(const struct wiimod_ops *ops, + struct wiimote_data *wdata) +{ + int ret; + struct wiimote_speaker *speaker; + struct snd_card *card; + struct snd_kcontrol *kcontrol; + struct snd_pcm *pcm; + + /* create sound card device */ + ret = snd_card_create(-1, NULL, THIS_MODULE, + sizeof(struct wiimote_speaker), &card); + if (ret) + return ret; + speaker = card->private_data; + + wdata->speaker = speaker; + speaker->card = card; + speaker->mute = 1; + speaker->volume = 0xff; + strcpy(card->driver, "hid-wiimote"); + strcpy(card->shortname, "wiimote"); + strcpy(card->longname, "Nintendo Wii Remote speaker"); + + /* create volume control */ + kcontrol = snd_ctl_new1(&wiimod_speaker_volume, wdata); + if (!kcontrol) { + ret = -ENOMEM; + goto err_free; + } + + ret = snd_ctl_add(card, kcontrol); + if (ret) { + snd_ctl_free_one(kcontrol); + goto err_free; + } + + /* create mute control */ + kcontrol = snd_ctl_new1(&wiimod_speaker_mute, wdata); + if (!kcontrol) { + ret = -ENOMEM; + goto err_free; + } + + ret = snd_ctl_add(card, kcontrol); + if (ret) { + snd_ctl_free_one(kcontrol); + goto err_free; + } + + /* create PCM sub-device for playback */ + ret = snd_pcm_new(card, "Speaker", 0, 1, 0, &pcm); + if (ret) + goto err_free; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &wiimod_speaker_playback_ops); + pcm->private_data = wdata; + + /* register sound card */ + snd_card_set_dev(card, &wdata->hdev->dev); + ret = snd_card_register(card); + if (ret) + goto err_free; + + return 0; + +err_free: + snd_card_free(card); + wdata->speaker = NULL; + return ret; +} + +static void wiimod_speaker_remove(const struct wiimod_ops *ops, + struct wiimote_data *wdata) +{ + struct wiimote_speaker *speaker = wdata->speaker; + + if (!speaker) + return; + + snd_card_free_when_closed(speaker->card); + wdata->speaker = NULL; +} + +const struct wiimod_ops wiimod_speaker = { + .flags = 0, + .arg = 0, + .probe = wiimod_speaker_probe, + .remove = wiimod_speaker_remove, +}; diff --git a/drivers/hid/hid-wiimote.h b/drivers/hid/hid-wiimote.h index 30caef5..4907bc7 100644 --- a/drivers/hid/hid-wiimote.h +++ b/drivers/hid/hid-wiimote.h @@ -148,6 +148,7 @@ struct wiimote_data { struct timer_list timer; struct wiimote_ext *ext; struct wiimote_debug *debug; + struct wiimote_speaker *speaker; union { struct input_dev *input; @@ -172,6 +173,7 @@ enum wiimod_module { WIIMOD_IR, WIIMOD_BUILTIN_MP, WIIMOD_NO_MP, + WIIMOD_SPEAKER, WIIMOD_NUM, WIIMOD_NULL = WIIMOD_NUM, }; @@ -208,9 +210,12 @@ enum wiiproto_reqs { WIIPROTO_REQ_LED = 0x11, WIIPROTO_REQ_DRM = 0x12, WIIPROTO_REQ_IR1 = 0x13, + WIIPROTO_REQ_SPEAKER = 0x14, WIIPROTO_REQ_SREQ = 0x15, WIIPROTO_REQ_WMEM = 0x16, WIIPROTO_REQ_RMEM = 0x17, + WIIPROTO_REQ_AUDIO = 0x18, + WIIPROTO_REQ_MUTE = 0x19, WIIPROTO_REQ_IR2 = 0x1a, WIIPROTO_REQ_STATUS = 0x20, WIIPROTO_REQ_DATA = 0x21, @@ -241,6 +246,12 @@ extern void wiiproto_req_status(struct wiimote_data *wdata); extern void wiiproto_req_accel(struct wiimote_data *wdata, __u8 accel); extern void wiiproto_req_ir1(struct wiimote_data *wdata, __u8 flags); extern void wiiproto_req_ir2(struct wiimote_data *wdata, __u8 flags); +extern void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, + __u32 offset, const __u8 *buf, __u8 size); +extern void wiiproto_req_speaker(struct wiimote_data *wdata, bool on); +extern void wiiproto_req_mute(struct wiimote_data *wdata, bool on); +extern int wiiproto_req_audio_user(struct wiimote_data *wdata, + void __user *buf, size_t len); extern int wiimote_cmd_write(struct wiimote_data *wdata, __u32 offset, const __u8 *wmem, __u8 size); extern ssize_t wiimote_cmd_read(struct wiimote_data *wdata, __u32 offset, @@ -283,6 +294,12 @@ static inline void wiidebug_deinit(void *u) { } #endif +#ifdef CONFIG_SND + +extern const struct wiimod_ops wiimod_speaker; + +#endif + /* requires the state.lock spinlock to be held */ static inline bool wiimote_cmd_pending(struct wiimote_data *wdata, int cmd, __u32 opt) -- 1.8.2.1

2 6

Re: [alsa-devel] [PATCH] ASoC: fix kernel message grepability
by Mark Brown 05 May '13

05 May '13

On Sat, May 04, 2013 at 10:21:38PM +0200, Michał Mirosław wrote: > Signed-off-by: Michał Mirosław <mirq-linux(a)rere.qmqm.pl> Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: sgtl5000: Convert to use regmap directly
by Fabio Estevam 05 May '13

05 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 145 +++++++++++++++++++++++++++++++++---------- 1 file changed, 111 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 92bbfec..96b65b1 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -16,6 +16,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/clk.h> +#include <linux/regmap.h> #include <linux/regulator/driver.h> #include <linux/regulator/machine.h> #include <linux/regulator/consumer.h> @@ -34,30 +35,30 @@ #define SGTL5000_MAX_REG_OFFSET 0x013A /* default value of sgtl5000 registers */ -static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET] = { - [SGTL5000_CHIP_CLK_CTRL] = 0x0008, - [SGTL5000_CHIP_I2S_CTRL] = 0x0010, - [SGTL5000_CHIP_SSS_CTRL] = 0x0008, - [SGTL5000_CHIP_DAC_VOL] = 0x3c3c, - [SGTL5000_CHIP_PAD_STRENGTH] = 0x015f, - [SGTL5000_CHIP_ANA_HP_CTRL] = 0x1818, - [SGTL5000_CHIP_ANA_CTRL] = 0x0111, - [SGTL5000_CHIP_LINE_OUT_VOL] = 0x0404, - [SGTL5000_CHIP_ANA_POWER] = 0x7060, - [SGTL5000_CHIP_PLL_CTRL] = 0x5000, - [SGTL5000_DAP_BASS_ENHANCE] = 0x0040, - [SGTL5000_DAP_BASS_ENHANCE_CTRL] = 0x051f, - [SGTL5000_DAP_SURROUND] = 0x0040, - [SGTL5000_DAP_EQ_BASS_BAND0] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND1] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND2] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND3] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND4] = 0x002f, - [SGTL5000_DAP_MAIN_CHAN] = 0x8000, - [SGTL5000_DAP_AVC_CTRL] = 0x0510, - [SGTL5000_DAP_AVC_THRESHOLD] = 0x1473, - [SGTL5000_DAP_AVC_ATTACK] = 0x0028, - [SGTL5000_DAP_AVC_DECAY] = 0x0050, +static const struct reg_default sgtl5000_reg_defaults[] = { + { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, + { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, + { SGTL5000_CHIP_SSS_CTRL, 0x0008 }, + { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, + { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, + { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, + { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, + { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, + { SGTL5000_CHIP_ANA_POWER, 0x7060 }, + { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, + { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, + { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, + { SGTL5000_DAP_SURROUND, 0x0040 }, + { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, + { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, + { SGTL5000_DAP_EQ_BASS_BAND2, 0x002f }, + { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, + { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, + { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, + { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, + { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, + { SGTL5000_DAP_AVC_DECAY, 0x0050 }, }; /* regulator supplies for sgtl5000, VDDD is an optional external supply */ @@ -112,6 +113,7 @@ struct sgtl5000_priv { int fmt; /* i2s data format */ struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM]; struct ldo_regulator *ldo; + struct regmap *regmap; }; /* @@ -958,17 +960,76 @@ static struct snd_soc_dai_driver sgtl5000_dai = { .symmetric_rates = 1, }; -static int sgtl5000_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool sgtl5000_volatile(struct device *dev, unsigned int reg) { switch (reg) { case SGTL5000_CHIP_ID: case SGTL5000_CHIP_ADCDAC_CTRL: case SGTL5000_CHIP_ANA_STATUS: - return 1; + return true; } - return 0; + return false; +} + +static bool sgtl5000_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SGTL5000_CHIP_ID: + case SGTL5000_CHIP_DIG_POWER: + case SGTL5000_CHIP_CLK_CTRL: + case SGTL5000_CHIP_I2S_CTRL: + case SGTL5000_CHIP_SSS_CTRL: + case SGTL5000_CHIP_ADCDAC_CTRL: + case SGTL5000_CHIP_DAC_VOL: + case SGTL5000_CHIP_PAD_STRENGTH: + case SGTL5000_CHIP_ANA_ADC_CTRL: + case SGTL5000_CHIP_ANA_HP_CTRL: + case SGTL5000_CHIP_ANA_CTRL: + case SGTL5000_CHIP_LINREG_CTRL: + case SGTL5000_CHIP_REF_CTRL: + case SGTL5000_CHIP_MIC_CTRL: + case SGTL5000_CHIP_LINE_OUT_CTRL: + case SGTL5000_CHIP_LINE_OUT_VOL: + case SGTL5000_CHIP_ANA_POWER: + case SGTL5000_CHIP_PLL_CTRL: + case SGTL5000_CHIP_CLK_TOP_CTRL: + case SGTL5000_CHIP_ANA_STATUS: + case SGTL5000_CHIP_SHORT_CTRL: + case SGTL5000_CHIP_ANA_TEST2: + case SGTL5000_DAP_CTRL: + case SGTL5000_DAP_PEQ: + case SGTL5000_DAP_BASS_ENHANCE: + case SGTL5000_DAP_BASS_ENHANCE_CTRL: + case SGTL5000_DAP_AUDIO_EQ: + case SGTL5000_DAP_SURROUND: + case SGTL5000_DAP_FLT_COEF_ACCESS: + case SGTL5000_DAP_COEF_WR_B0_MSB: + case SGTL5000_DAP_COEF_WR_B0_LSB: + case SGTL5000_DAP_EQ_BASS_BAND0: + case SGTL5000_DAP_EQ_BASS_BAND1: + case SGTL5000_DAP_EQ_BASS_BAND2: + case SGTL5000_DAP_EQ_BASS_BAND3: + case SGTL5000_DAP_EQ_BASS_BAND4: + case SGTL5000_DAP_MAIN_CHAN: + case SGTL5000_DAP_MIX_CHAN: + case SGTL5000_DAP_AVC_CTRL: + case SGTL5000_DAP_AVC_THRESHOLD: + case SGTL5000_DAP_AVC_ATTACK: + case SGTL5000_DAP_AVC_DECAY: + case SGTL5000_DAP_COEF_WR_B1_MSB: + case SGTL5000_DAP_COEF_WR_B1_LSB: + case SGTL5000_DAP_COEF_WR_B2_MSB: + case SGTL5000_DAP_COEF_WR_B2_LSB: + case SGTL5000_DAP_COEF_WR_A1_MSB: + case SGTL5000_DAP_COEF_WR_A1_LSB: + case SGTL5000_DAP_COEF_WR_A2_MSB: + case SGTL5000_DAP_COEF_WR_A2_LSB: + return true; + + default: + return false; + } } #ifdef CONFIG_SUSPEND @@ -1300,7 +1361,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); /* setup i2c data ops */ - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + codec->control_data = sgtl5000->regmap; + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -1391,11 +1453,6 @@ static struct snd_soc_codec_driver sgtl5000_driver = { .suspend = sgtl5000_suspend, .resume = sgtl5000_resume, .set_bias_level = sgtl5000_set_bias_level, - .reg_cache_size = ARRAY_SIZE(sgtl5000_regs), - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, - .reg_cache_default = sgtl5000_regs, - .volatile_register = sgtl5000_volatile_register, .controls = sgtl5000_snd_controls, .num_controls = ARRAY_SIZE(sgtl5000_snd_controls), .dapm_widgets = sgtl5000_dapm_widgets, @@ -1404,6 +1461,19 @@ static struct snd_soc_codec_driver sgtl5000_driver = { .num_dapm_routes = ARRAY_SIZE(sgtl5000_dapm_routes), }; +static const struct regmap_config sgtl5000_regmap = { + .reg_bits = 16, + .val_bits = 16, + + .max_register = SGTL5000_MAX_REG_OFFSET, + .volatile_reg = sgtl5000_volatile, + .readable_reg = sgtl5000_readable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = sgtl5000_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(sgtl5000_reg_defaults), +}; + static int sgtl5000_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -1415,6 +1485,13 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, if (!sgtl5000) return -ENOMEM; + sgtl5000->regmap = devm_regmap_init_i2c(client, &sgtl5000_regmap); + if (IS_ERR(sgtl5000->regmap)) { + ret = PTR_ERR(sgtl5000->regmap); + dev_err(&client->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + i2c_set_clientdata(client, sgtl5000); ret = snd_soc_register_codec(&client->dev, -- 1.7.9.5

2 1

Re: [alsa-devel] [RFC] ASoC: sgtl5000: Remove cache support
by Fabio Estevam 05 May '13

05 May '13

Hi Mark, On Fri, May 3, 2013 at 7:36 PM, Mark Brown <broonie(a)sirena.org.uk> wrote: > On Fri, May 03, 2013 at 04:49:09PM -0300, Fabio Estevam wrote: > >> .set_bias_level = sgtl5000_set_bias_level, >> - .reg_cache_size = ARRAY_SIZE(sgtl5000_regs), >> - .reg_word_size = sizeof(u16), >> - .reg_cache_step = 2, >> - .reg_cache_default = sgtl5000_regs, > > So, the most obvious thing here is to convert the driver to regmap. It > has a much more baked in idea of sparse register maps which should help > if nothing else, and will allow the device initialisation to be moved to > the normal device probe which is better practice. Thanks for your suggestion. I have just sent a patch converting sgtl5000 to use regmap. Unfortunately the result is the same: the codec cannot probe after a reset. > Otherwise the normal issue with this sort of thing is a failure to power > down the device properly, the LDO workaround looks suspicous here as > does the fact that the driver doesn't restore the register cache when > powering on the device in resume (nor does it makr the device as cache > only). I will try to investigate the power down and LDO workaround as suggested. Thanks, Fabio Estevam

2 2

Re: [alsa-devel] Conexant CX20588 breakage in Kernel 3.9
by Takashi Iwai 04 May '13

04 May '13

At Sat, 4 May 2013 00:56:10 +0200, Christopher K. wrote: > > Works great, thanks for the help! OK, now applied. Thanks for a quick test. Takashi > > > On Fri, May 3, 2013 at 4:58 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > At Fri, 3 May 2013 15:51:20 +0200, > > Christopher K. wrote: > > > > > > Hi, > > > > > > Kernel 3.9 breaks audio on internal speakers for the Conexant CX20588. > > > EAPD is disabled for Node 0x1d. Enabling in hda_analyzer works fine. > > > > > > Someone already postet on > > > https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1173618 > > > > > > Is this the right place to report this bug? > > > > The kernel bugzilla or in alsa-devel ML, at best with Cc to > > maintainer. > > > > Does the patch below fix your problem? If so, I'll queue it to the > > upstream and Cc to stable kernel. > > > > > > thanks, > > > > Takashi > > > > --- > > diff --git a/sound/pci/hda/patch_conexant.c > > b/sound/pci/hda/patch_conexant.c > > index 84b81c8..e08cd33 100644 > > --- a/sound/pci/hda/patch_conexant.c > > +++ b/sound/pci/hda/patch_conexant.c > > @@ -63,6 +63,7 @@ struct conexant_spec { > > > > /* extra EAPD pins */ > > unsigned int num_eapds; > > + bool dynamic_eapd; > > hda_nid_t eapds[4]; > > > > #ifdef ENABLE_CXT_STATIC_QUIRKS > > @@ -3155,7 +3156,7 @@ static void cx_auto_parse_eapd(struct hda_codec > > *codec) > > * thus it might control over all pins. > > */ > > if (spec->num_eapds > 2) > > - spec->gen.own_eapd_ctl = 1; > > + spec->dynamic_eapd = 1; > > } > > > > static void cx_auto_turn_eapd(struct hda_codec *codec, int num_pins, > > @@ -3194,10 +3195,19 @@ static int cx_auto_build_controls(struct hda_codec > > *codec) > > return 0; > > } > > > > +static int cx_auto_init(struct hda_codec *codec) > > +{ > > + struct conexant_spec *spec = codec->spec; > > + snd_hda_gen_init(codec); > > + if (!spec->dynamic_eapd) > > + cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, > > true); > > + return 0; > > +} > > + > > static const struct hda_codec_ops cx_auto_patch_ops = { > > .build_controls = cx_auto_build_controls, > > .build_pcms = snd_hda_gen_build_pcms, > > - .init = snd_hda_gen_init, > > + .init = cx_auto_init, > > .free = snd_hda_gen_free, > > .unsol_event = snd_hda_jack_unsol_event, > > #ifdef CONFIG_PM > > @@ -3348,7 +3358,8 @@ static int patch_conexant_auto(struct hda_codec > > *codec) > > > > cx_auto_parse_beep(codec); > > cx_auto_parse_eapd(codec); > > - if (spec->gen.own_eapd_ctl) > > + spec->gen.own_eapd_ctl = 1; > > + if (spec->dynamic_eapd) > > spec->gen.vmaster_mute.hook = cx_auto_vmaster_hook; > > > > switch (codec->vendor_id) { > > > [2 <text/html; UTF-8 (quoted-printable)>] >

1 0

[alsa-devel] [PATCH] ASoC: sgtl5000: Fix comment about register addresses
by Fabio Estevam 04 May '13

04 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> The list below the comment relates to sgtl5000 registers addresses, so change the comment to improve the description. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 8a9f435..4b69229 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -12,7 +12,7 @@ #define _SGTL5000_H /* - * Register values. + * Registers addresses */ #define SGTL5000_CHIP_ID 0x0000 #define SGTL5000_CHIP_DIG_POWER 0x0002 -- 1.7.9.5

2 1

[alsa-devel] [RFC] ASoC: sgtl5000: Remove cache support
by Fabio Estevam 03 May '13

03 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Currently sgtl5000 driver is probed correctly after a power-on reset, but after audio is played and the board is reset, we are no longer able to probe again: sgtl5000 0-000a: Device with ID register ffff is not a sgtl5000 sgtl5000 0-000a: ASoC: failed to probe CODEC -19 imx-sgtl5000 sound.12: ASoC: failed to instantiate card -19 imx-sgtl5000 sound.12: snd_soc_register_card failed (-19) Removing the cache support mechanism allows us to probe sgtl5000 driver successfully in subsequent board resets. Tested on a mx6qsabrelite board. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- I would like to fix it properly and looking for some ideas/feedbacks as to what a proper fix would be. Should I complete the sgtl5000_regs array with all the sgtl5000 registers? Thanks! sound/soc/codecs/sgtl5000.c | 31 ------------------------------- 1 file changed, 31 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 92bbfec..3884162 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -33,33 +33,6 @@ #define SGTL5000_DAP_REG_OFFSET 0x0100 #define SGTL5000_MAX_REG_OFFSET 0x013A -/* default value of sgtl5000 registers */ -static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET] = { - [SGTL5000_CHIP_CLK_CTRL] = 0x0008, - [SGTL5000_CHIP_I2S_CTRL] = 0x0010, - [SGTL5000_CHIP_SSS_CTRL] = 0x0008, - [SGTL5000_CHIP_DAC_VOL] = 0x3c3c, - [SGTL5000_CHIP_PAD_STRENGTH] = 0x015f, - [SGTL5000_CHIP_ANA_HP_CTRL] = 0x1818, - [SGTL5000_CHIP_ANA_CTRL] = 0x0111, - [SGTL5000_CHIP_LINE_OUT_VOL] = 0x0404, - [SGTL5000_CHIP_ANA_POWER] = 0x7060, - [SGTL5000_CHIP_PLL_CTRL] = 0x5000, - [SGTL5000_DAP_BASS_ENHANCE] = 0x0040, - [SGTL5000_DAP_BASS_ENHANCE_CTRL] = 0x051f, - [SGTL5000_DAP_SURROUND] = 0x0040, - [SGTL5000_DAP_EQ_BASS_BAND0] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND1] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND2] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND3] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND4] = 0x002f, - [SGTL5000_DAP_MAIN_CHAN] = 0x8000, - [SGTL5000_DAP_AVC_CTRL] = 0x0510, - [SGTL5000_DAP_AVC_THRESHOLD] = 0x1473, - [SGTL5000_DAP_AVC_ATTACK] = 0x0028, - [SGTL5000_DAP_AVC_DECAY] = 0x0050, -}; - /* regulator supplies for sgtl5000, VDDD is an optional external supply */ enum sgtl5000_regulator_supplies { VDDA, @@ -1391,10 +1364,6 @@ static struct snd_soc_codec_driver sgtl5000_driver = { .suspend = sgtl5000_suspend, .resume = sgtl5000_resume, .set_bias_level = sgtl5000_set_bias_level, - .reg_cache_size = ARRAY_SIZE(sgtl5000_regs), - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, - .reg_cache_default = sgtl5000_regs, .volatile_register = sgtl5000_volatile_register, .controls = sgtl5000_snd_controls, .num_controls = ARRAY_SIZE(sgtl5000_snd_controls), -- 1.7.9.5

1 0

[alsa-devel] [PATCH] drm/i915: Add private api for power well usage -- alignment between graphic team and audio team
by Li, Jocelyn 03 May '13

03 May '13

Hi Guys, I apologize that I may have misunderstood on the agreement we have made on power well work. I assumed that audio team is expected to create private gfx driver API to control the power well between gfx and audio for internal releases. That's why I asked Xingchao work on that. So the correct implementation approach between gfx and audio should be, #1, Daniel will work on the existing private API in gfx driver for power well control that can be used by the audio driver to enable/disable the power well. #2, Once #1 is done, Audio team will integrate Daniel's patch into audio driver and add/improve on anything that is missing Please correct me if my understanding is not correct. Thanks, Jocelyn -----Original Message----- From: Daniel Vetter [mailto:daniel.vetter@ffwll.ch] Sent: Thursday, April 25, 2013 3:52 PM To: Wang, Xingchao Cc: Zanoni, Paulo R; ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; Li, Jocelyn; intel-gfx(a)lists.freedesktop.org; alsa-devel(a)alsa-project.org; Wang Xingchao; Takashi Iwai; Barnes, Jesse; Wysocki, Rafael J Subject: Re: [PATCH] drm/i915: Add private api for power well usage On Thu, Apr 25, 2013 at 4:36 AM, Wang, Xingchao <xingchao.wang(a)intel.com> wrote: > Hi Daniel/Paulo, > > Any comments on this? > Add Jesse and Rafael in loop. Well I've figured we want to get the power well integration working before we waste time reviewing an interface which might or might not quite work. But if you want, I can drop two quick comments: - I think the functions needs some prefix like i915_ - Imo the power well should just be reference counted (the current interface name suggests it's generally useable, but actually can only cope with one request from the audio driver). - Is the spin_lock (irq-disabling even) really required for the audio driver to work? power well enabling can take up to 20ms, so this won't work as-is. Can't we just use a mutex? - I'd need to see the audio driver side of this to check how this works for coping with arbitrary module load ordering. Cheers, Daniel > > thanks > --xingchao > > >> -----Original Message----- >> From: Wang, Xingchao >> Sent: Wednesday, April 24, 2013 3:29 PM >> To: daniel.vetter(a)ffwll.ch; Zanoni, Paulo R; 'Takashi Iwai' >> Cc: ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; >> Li, Jocelyn; intel-gfx(a)lists.freedesktop.org; alsa-devel(a)alsa-project.org; 'Wang Xingchao' >> Subject: RE: [PATCH] drm/i915: Add private api for power well usage >> >> Hi Guys, >> >> Sorry I should Add Takashi and alsa maillist in loop. >> >> I tested this patch against Daniel's "drm-intel-next-queued" Branch, >> on Haswell Mobile C3 stepping machines. >> >> Test steps: >> 1. only connect one monitor through HDMI cable 2. echo 1 > >> /sys/module/i915/parameters/disable_power_well ==> I think this could >> be set as 1 by default if applied my patch. >> 3. plug out HDMI cable >> 4. cat /proc/asound/card0/codec#0 >> >> At step 3, i915 will try to disable power well but rejected as >> display audio is using power well. >> Without the patch, display audio would crash at step 4. >> >> I did not list suspend/resume test here, as during suspend sequence, >> i915 will _ENABLE_ power well, that's a BUG? >> >> For me above test case is the only one to reproduce the issue, if you >> have more ideas and need more test, please let me know. >> >> I'm afraid there's some risk because the dependency between alsa >> driver and drm driver. For example, if i915 module was not loaded, >> display audio Will call >> request_power_well() and meet symbol error. Even if i915 module >> loaded after snd-hda-intel module, there's such risk too. Any comment on this? >> >> Another risk is the pointer "power_well_device", it should be set >> NULL when >> i915 module unloaded, I added the caller but have no idea where to >> call it, if anyone has suggestion, that would be fine. >> >> thanks >> --xingchao >> >> >> > -----Original Message----- >> > From: Wang Xingchao [mailto:xingchao.wang@linux.intel.com] >> > Sent: Thursday, April 24, 2014 10:20 PM >> > To: daniel.vetter(a)ffwll.ch; Zanoni, Paulo R >> > Cc: ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; >> > Li, Jocelyn; intel-gfx(a)lists.freedesktop.org; Wang, Xingchao; Wang >> > Xingchao >> > Subject: [PATCH] drm/i915: Add private api for power well usage >> > >> > Display audio depends on power well in graphic side, it should >> > request power well before use it and release power well after use. >> > I915 will not shutdown power well if it detects audio is using. >> > This patch protects display audio crash for Intel Haswell mobile >> > C3 stepping board. >> > >> > Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> >> > --- >> > drivers/gpu/drm/i915/intel_pm.c | 73 >> > +++++++++++++++++++++++++++++++++++---- >> > include/drm/audio_drm.h | 39 +++++++++++++++++++++ >> > sound/pci/hda/hda_intel.c | 8 +++++ >> > 3 files changed, 113 insertions(+), 7 deletions(-) create mode >> > 100644 include/drm/audio_drm.h >> > >> > diff --git a/drivers/gpu/drm/i915/intel_pm.c >> > b/drivers/gpu/drm/i915/intel_pm.c index 2557926..9983f56 100644 >> > --- a/drivers/gpu/drm/i915/intel_pm.c >> > +++ b/drivers/gpu/drm/i915/intel_pm.c >> > @@ -4298,18 +4298,12 @@ bool intel_using_power_well(struct >> > drm_device >> > *dev) >> > return true; >> > } >> > >> > -void intel_set_power_well(struct drm_device *dev, bool enable) >> > +void set_power_well(struct drm_device *dev, bool enable) >> > { >> > struct drm_i915_private *dev_priv = dev->dev_private; >> > bool is_enabled, enable_requested; >> > uint32_t tmp; >> > >> > - if (!HAS_POWER_WELL(dev)) >> > - return; >> > - >> > - if (!i915_disable_power_well && !enable) >> > - return; >> > - >> > tmp = I915_READ(HSW_PWR_WELL_DRIVER); >> > is_enabled = tmp & HSW_PWR_WELL_STATE; >> > enable_requested = tmp & HSW_PWR_WELL_ENABLE; @@ -4332,6 >> > +4326,71 @@ void intel_set_power_well(struct drm_device *dev, bool >> > +enable) >> > } >> > } >> > >> > +/* Global drm_device for display audio drvier usage */ static >> > +struct drm_device *power_well_device; static bool >> > +audio_power_well_using, i915_power_well_using; >> > +/* Lock protecting power well setting */ >> > +DEFINE_SPINLOCK(powerwell_lock); >> > + >> > +void request_power_well(void) >> > +{ >> > + DRM_DEBUG_KMS("Display audio requesting power well\n"); >> > + spin_lock_irq(&powerwell_lock); >> > + if (!power_well_device) >> > + goto out; >> > + if (i915_power_well_using == false) >> > + set_power_well(power_well_device, true); >> > + audio_power_well_using = true; >> > +out: >> > + spin_unlock_irq(&powerwell_lock); >> > + return; >> > +} >> > +EXPORT_SYMBOL_GPL(request_power_well); >> > + >> > +void release_power_well(void) >> > +{ >> > + DRM_DEBUG_KMS("Display audio releasing power well\n"); >> > + spin_lock_irq(&powerwell_lock); >> > + if (!power_well_device) >> > + goto out; >> > + if (i915_power_well_using == false) >> > + set_power_well(power_well_device, false); >> > + audio_power_well_using = false; >> > +out: >> > + spin_unlock_irq(&powerwell_lock); >> > + return; >> > +} >> > +EXPORT_SYMBOL_GPL(release_power_well); >> > + >> > +/* TODO: Call this when i915 module unload */ void >> > +remove_power_well(void) { >> > + spin_lock_irq(&powerwell_lock); >> > + audio_power_well_using = false; >> > + i915_power_well_using = false; >> > + power_well_device = NULL; >> > + spin_unlock_irq(&powerwell_lock); } >> > + >> > +void intel_set_power_well(struct drm_device *dev, bool enable) { >> > + if (!HAS_POWER_WELL(dev)) >> > + return; >> > + >> > + spin_lock_irq(&powerwell_lock); >> > + power_well_device = dev; >> > + i915_power_well_using = enable; >> > + if (!enable && audio_power_well_using) >> > + goto out; >> > + >> > + if (!i915_disable_power_well && !enable) >> > + goto out; >> > + set_power_well(dev, enable); >> > +out: >> > + spin_unlock_irq(&powerwell_lock); >> > + return; >> > +} >> > + >> > /* >> > * Starting with Haswell, we have a "Power Down Well" that can be >> > turned >> off >> > * when not needed anymore. We have 4 registers that can request >> > the power well diff --git a/include/drm/audio_drm.h >> > b/include/drm/audio_drm.h new file mode 100644 index >> > 0000000..4412b36 >> > --- /dev/null >> > +++ b/include/drm/audio_drm.h >> > @@ -0,0 +1,39 @@ >> > >> +/************************************************************** >> > ******** >> > +**** >> > + * >> > + * Copyright 2013 Intel Inc. >> > + * All Rights Reserved. >> > + * >> > + * Permission is hereby granted, free of charge, to any person >> > +obtaining a >> > + * copy of this software and associated documentation files (the >> > + * "Software"), to deal in the Software without restriction, >> > +including >> > + * without limitation the rights to use, copy, modify, merge, >> > +publish, >> > + * distribute, sub license, and/or sell copies of the Software, >> > +and to >> > + * permit persons to whom the Software is furnished to do so, >> > +subject to >> > + * the following conditions: >> > + * >> > + * The above copyright notice and this permission notice >> > +(including the >> > + * next paragraph) shall be included in all copies or substantial >> > +portions >> > + * of the Software. >> > + * >> > + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY >> KIND, >> > +EXPRESS OR >> > + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF >> > +MERCHANTABILITY, >> > + * FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT. IN NO >> > EVENT >> > +SHALL >> > + * THE COPYRIGHT HOLDERS, AUTHORS AND/OR ITS SUPPLIERS BE LIABLE >> > FOR >> > +ANY CLAIM, >> > + * DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, >> > TORT >> > +OR >> > + * OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE >> > SOFTWARE >> > +OR THE >> > + * USE OR OTHER DEALINGS IN THE SOFTWARE. >> > + * >> > + * >> > + >> > >> +*************************************************************** >> > ******** >> > +***/ >> > +/* >> > + * Authors: >> > + * Wang Xingchao <xingchao.wang(a)intel.com> */ >> > + >> > +#ifndef _AUDIO_DRM_H_ >> > +#define _AUDIO_DRM_H_ >> > + >> > +extern void request_power_well(void); extern void >> > +release_power_well(void); >> > + >> > +#endif /* _AUDIO_DRM_H_ */ >> > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c >> > index 418bfc0..59a85de 100644 >> > --- a/sound/pci/hda/hda_intel.c >> > +++ b/sound/pci/hda/hda_intel.c >> > @@ -50,6 +50,7 @@ >> > #include <linux/clocksource.h> >> > #include <linux/time.h> >> > #include <linux/completion.h> >> > +#include <drm/audio_drm.h> >> > >> > #ifdef CONFIG_X86 >> > /* for snoop control */ >> > @@ -2869,6 +2870,7 @@ static int azx_suspend(struct device *dev) >> > pci_disable_device(pci); >> > pci_save_state(pci); >> > pci_set_power_state(pci, PCI_D3hot); >> > + release_power_well(); >> > return 0; >> > } >> > >> > @@ -2881,6 +2883,7 @@ static int azx_resume(struct device *dev) >> > if (chip->disabled) >> > return 0; >> > >> > + request_power_well(); >> > pci_set_power_state(pci, PCI_D0); >> > pci_restore_state(pci); >> > if (pci_enable_device(pci) < 0) { @@ -2913,6 +2916,7 @@ static >> > int azx_runtime_suspend(struct device >> > *dev) >> > >> > azx_stop_chip(chip); >> > azx_clear_irq_pending(chip); >> > + release_power_well(); >> > return 0; >> > } >> > >> > @@ -2921,6 +2925,7 @@ static int azx_runtime_resume(struct device *dev) >> > struct snd_card *card = dev_get_drvdata(dev); >> > struct azx *chip = card->private_data; >> > >> > + request_power_well(); >> > azx_init_pci(chip); >> > azx_init_chip(chip, 1); >> > return 0; >> > @@ -3671,6 +3676,8 @@ static int azx_probe(struct pci_dev *pci, >> > return -ENOENT; >> > } >> > >> > + request_power_well(); >> > + >> > err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); >> > if (err < 0) { >> > snd_printk(KERN_ERR "hda-intel: Error creating >> > card!\n"); @@ >> > -3806,6 +3813,7 @@ static void azx_remove(struct pci_dev *pci) >> > if (card) >> > snd_card_free(card); >> > pci_set_drvdata(pci, NULL); >> > + release_power_well(); >> > } >> > >> > /* PCI IDs */ >> > -- >> > 1.7.9.5 > -- Daniel Vetter Software Engineer, Intel Corporation +41 (0) 79 365 57 48 - http://blog.ffwll.ch

9 23

[alsa-devel] Conexant CX20588 breakage in Kernel 3.9
by Christopher K. 03 May '13

03 May '13

Hi, Kernel 3.9 breaks audio on internal speakers for the Conexant CX20588. EAPD is disabled for Node 0x1d. Enabling in hda_analyzer works fine. Someone already postet on https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1173618 Is this the right place to report this bug? Regards, Chris

2 1

[alsa-devel] Audio miniconference (probably) moved to Edinburgh
by David Henningsson 03 May '13

03 May '13

Just to keep the wider crowd informed: We're currently discussing that we won't have any Audio miniconference on LPC (Linux Plumber's Conference) this year. Instead, we'll have an audio miniconference day in Edinburgh in October, colocated with LinuxCon Europe, Kernel Summit, ELCE and GStreamer conferences. The exact day for this miniconference has not been decided yet. The main reason for this move is proximity, as several participants are located in northern Europe. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

1 0

[alsa-devel] [PATCH]: x86_64, audio: Fix system panic when DMA > 40 bits
by Mike Travis 03 May '13

03 May '13

The audio driver mistakenly allows 64 bit addresses to be created for the audio driver on Nvidia GPUs. Unfortunately, the hardware normally only supports up to 40 bits of DMA. This can cause system panics as well as misdirected data when the address is > 40 bits as the upper part the address is truncated. Signed-off-by: Mike Travis <travis(a)sgi.com> Reviewed-by: Mike Habeck <habeck(a)sgi.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bcd40ee..45eb165 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -615,7 +615,7 @@ enum { /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ (AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\ - AZX_DCAPS_ALIGN_BUFSIZE) + AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_NO_64BIT) #define AZX_DCAPS_PRESET_CTHDA \ (AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_4K_BDLE_BOUNDARY) -- 1.8.2.1

4 7

[alsa-devel] ASoC updates for v3.10
by Mark Brown 03 May '13

03 May '13

The following changes since commit c1be5a5b1b355d40e6cf79cc979eb66dafa24ad1: Linux 3.9 (2013-04-28 17:36:01 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.10-4 for you to fetch changes up to 9631c02e240236ec206394fb52380647d0e70e4f: Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus (2013-05-03 09:30:29 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.10 A few more bug fixes, the DAPM clock fix is actually a driver specific one since currently there's only one user of the clock support due to the problems relying on the clock API. ---------------------------------------------------------------- Dan Carpenter (1): ASoC: wm8994: missing break in wm8994_aif3_hw_params() Fabio Baltieri (1): ASoC: dapm: use clk_prepare_enable and clk_disable_unprepare Marek Belisko (2): ASoC: McASP: Add pins output direction for rx clocks when configured in CBS_CFS format ASoC: McASP: Fix receive clock polarity in DAIFMT_NB_NF mode. Mark Brown (2): Merge remote-tracking branch 'asoc/fix/davinci' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus sound/soc/codecs/wm8994.c | 1 + sound/soc/davinci/davinci-mcasp.c | 7 +++++-- sound/soc/soc-dapm.c | 4 ++-- 3 files changed, 8 insertions(+), 4 deletions(-)

2 1

[alsa-devel] [PATCH -next 0/3] mop500_ab8500 fixes
by Fabio Baltieri 03 May '13

03 May '13

Hi, these are small fixes to the mop500_ab8500 asoc driver, developed on top of today's ASoC-next. The first patch is actually non ab8500 specific, but as ab8500 seems to be the only codec to use the SND_SOC_DAPM_CLOCK_SUPPLY widget I'm posting it here anyway. The other two patches sort out a mixup of card and codec controls in mop500_ab8500. Thanks, Fabio Fabio Baltieri (3): ASoC: dapm: use clk_prepare_enable and clk_disable_unprepare ASoC: ux500: move clock controls to ab8500-codec ASoC: ux500: register controls to card instead of codec sound/soc/codecs/ab8500-codec.c | 10 ++++++++++ sound/soc/soc-dapm.c | 4 ++-- sound/soc/ux500/mop500_ab8500.c | 20 +++++--------------- 3 files changed, 17 insertions(+), 17 deletions(-) -- 1.8.2

3 11

[alsa-devel] [PATCH] ASoC: McASP: Fix receive clock polarity in DAIFMT_NB_NF mode.
by Marek Belisko 03 May '13

03 May '13

According documentation bit ACLKRPOL is set to 0 (receiver samples data on falling edge) and when set to 1 (receiver samples data on rising edge). I2S data are always sampled on falling edge and valid during rising edge of bit clock. So in case of capture data transmitter sample data on falling edge and macsp must read then on rising edge. Signed-off-by: Marek Belisko <marek.belisko(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 849a187..c46112e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -568,7 +568,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); - mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); break; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ALSA: atmel: Remove redundant platform_set_drvdata()
by Sachin Kamat 03 May '13

03 May '13

Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/atmel/abdac.c | 2 -- sound/atmel/ac97c.c | 2 -- 2 files changed, 4 deletions(-) diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index 071ce1b..872d59e 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -583,8 +583,6 @@ static int atmel_abdac_remove(struct platform_device *pdev) free_irq(dac->irq, dac); snd_card_free(card); - platform_set_drvdata(pdev, NULL); - return 0; } diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 6b7e2b5..ae63d22 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -1199,8 +1199,6 @@ static int atmel_ac97c_remove(struct platform_device *pdev) snd_card_set_dev(card, NULL); snd_card_free(card); - platform_set_drvdata(pdev, NULL); - return 0; } -- 1.7.9.5

4 3

[alsa-devel] [PATCH] ASoC: Added the condition for the channels_min of cpu_dai
by JongHo Kim 02 May '13

02 May '13

Only playback device, the pcm substream for capture does not need to be generated. If codec dai driver has both playback.channels_min and capture.channels_min, and cpu dai driver has only playback.channels_min, soc_new_pcm() will make the pcm substream for capture although DataIN is not connected. So added the condition for the channels_min of cpu_dai in soc_new_pcm(). Signed-off-by: JongHo Kim <furmuwon(a)gmail.com> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index cf191e6..c405e39 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2018,9 +2018,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (cpu_dai->driver->capture.channels_min) capture = 1; } else { - if (codec_dai->driver->playback.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (codec_dai->driver->capture.channels_min && + cpu_dai->driver->capture.channels_min) capture = 1; } -- 1.8.1.1

2 4

[alsa-devel] [PATCH] ASoC: max98090: Fix build warning when CONFIG_PM_RUNTIME is not set
by Axel Lin 02 May '13

02 May '13

Fix below build warning when CONFIG_PM_RUNTIME is not set. CC [M] sound/soc/codecs/max98090.o sound/soc/codecs/max98090.c:2339:12: warning: 'max98090_runtime_resume' defined but not used [-Wunused-function] sound/soc/codecs/max98090.c:2350:12: warning: 'max98090_runtime_suspend' defined but not used [-Wunused-function] Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98090.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index ce0d364..4f39a50 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2336,6 +2336,7 @@ static int max98090_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME static int max98090_runtime_resume(struct device *dev) { struct max98090_priv *max98090 = dev_get_drvdata(dev); @@ -2355,6 +2356,7 @@ static int max98090_runtime_suspend(struct device *dev) return 0; } +#endif static const struct dev_pm_ops max98090_pm = { SET_RUNTIME_PM_OPS(max98090_runtime_suspend, -- 1.8.1.2

2 1

[alsa-devel] [PATCH v2 1/4] ASoC: imx-sgtl5000: Do not enter the error path on success
by Fabio Estevam 02 May '13

02 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Return on success instead of entering the error path. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Keep of_node_put in the success path. sound/soc/fsl/imx-sgtl5000.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 9584e78..5a6aaa3 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -174,6 +174,11 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) } platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + clk_fail: clk_put(data->codec_clk); fail: -- 1.7.9.5

3 12

[alsa-devel] Mono output on emu10k1 with alsa broken
by Thomas Richter 01 May '13

01 May '13

Trying to configure mono (one channel) output on emu10k1 hardware using the alsa version that comes with kernel 3.9 is broken. Instead of expecting mono samples, the alsa callback expects twice as many stereo samples. Not getting the required number of samples then causes a buffer underrun, and sound gets interrupted or messed up. In fact, what the audio hardware does is that it plays not only the buffer, but also the data "behind the buffer", causing wierd noises. The channel count is setup as channels = 1; snd_pcm_hw_params_malloc(&HWParms); snd_pcm_open(&SoundStream,CardName,SND_PCM_STREAM_PLAYBACK,SND_PCM_NONBLOCK | SND_PCM_ASYNC); snd_pcm_hw_params_set_channels_min(SoundStream,HWParms,&channels); snd_pcm_hw_params_set_channels_max(SoundStream,HWParms,&channels); snd_pcm_hw_params_set_channels_first(SoundStream,HWParms,&channels); plus error checking. Then, a bit later, the number of configured channels is verified: snd_pcm_hw_params_get_channels(HWParms,&channels); channels is at this point still 1, i.e. the driver accepted the mono output, but does then not play correctly. Note that the same call sequence works for stereo, and it also works for mono for other underlying hardware drivers, i.e. mono output works on snd-hda-intel, for example. It does not on emu10k1.

1 0

[alsa-devel] [PATCH] MAINTAINERS: Add linux-samsung-soc list to all related entries
by Tomasz Figa 01 May '13

01 May '13

Several entries in MAINTAINERS file related to Samsung SoCs do not point to linux-samsung-soc mailing list, which is supposed to collect all Samsung SoC-related threads, to ease following of Samsung SoC-related work. This leads to a problem with people skipping this mailing list in their posts, even though they are related to Samsung SoCs. This patch adds pointers to linux-samsung-soc mailing list to affected entries of MAINTAINERS file. Signed-off-by: Tomasz Figa <tomasz.figa(a)gmail.com> --- MAINTAINERS | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 6c0d68b..07cb8da 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1139,6 +1139,7 @@ F: arch/arm/mach-exynos*/ ARM/SAMSUNG MOBILE MACHINE SUPPORT M: Kyungmin Park <kyungmin.park(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: arch/arm/mach-s5pv210/mach-aquila.c F: arch/arm/mach-s5pv210/mach-goni.c @@ -1150,6 +1151,7 @@ M: Kyungmin Park <kyungmin.park(a)samsung.com> M: Kamil Debski <k.debski(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org L: linux-media(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/media/platform/s5p-g2d/ @@ -1158,6 +1160,7 @@ M: Kyungmin Park <kyungmin.park(a)samsung.com> M: Sylwester Nawrocki <s.nawrocki(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org L: linux-media(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: arch/arm/plat-samsung/include/plat/*fimc* F: drivers/media/platform/s5p-fimc/ @@ -1168,6 +1171,7 @@ M: Kamil Debski <k.debski(a)samsung.com> M: Jeongtae Park <jtp.park(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org L: linux-media(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: arch/arm/plat-samsung/s5p-dev-mfc.c F: drivers/media/platform/s5p-mfc/ @@ -1177,6 +1181,7 @@ M: Kyungmin Park <kyungmin.park(a)samsung.com> M: Tomasz Stanislawski <t.stanislaws(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org L: linux-media(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/media/platform/s5p-tv/ @@ -2679,6 +2684,7 @@ M: Joonyoung Shim <jy0922.shim(a)samsung.com> M: Seung-Woo Kim <sw0312.kim(a)samsung.com> M: Kyungmin Park <kyungmin.park(a)samsung.com> L: dri-devel(a)lists.freedesktop.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) T: git git://git.kernel.org/pub/scm/linux/kernel/git/daeinki/drm-exynos.git S: Supported F: drivers/gpu/drm/exynos @@ -3142,6 +3148,7 @@ F: Documentation/extcon/ EXYNOS DP DRIVER M: Jingoo Han <jg1.han(a)samsung.com> L: linux-fbdev(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/video/exynos/exynos_dp* F: include/video/exynos_dp* @@ -3151,6 +3158,7 @@ M: Inki Dae <inki.dae(a)samsung.com> M: Donghwa Lee <dh09.lee(a)samsung.com> M: Kyungmin Park <kyungmin.park(a)samsung.com> L: linux-fbdev(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/video/exynos/exynos_mipi* F: include/video/exynos_mipi* @@ -6892,12 +6900,14 @@ F: drivers/platform/x86/samsung-laptop.c SAMSUNG AUDIO (ASoC) DRIVERS M: Sangbeom Kim <sbkim73(a)samsung.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Supported F: sound/soc/samsung SAMSUNG FRAMEBUFFER DRIVER M: Jingoo Han <jg1.han(a)samsung.com> L: linux-fbdev(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/video/s3c-fb.c @@ -7087,6 +7097,7 @@ F: drivers/mmc/host/sdhci-pltfm.[ch] SECURE DIGITAL HOST CONTROLLER INTERFACE (SDHCI) SAMSUNG DRIVER M: Ben Dooks <ben-linux(a)fluff.org> L: linux-mmc(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/mmc/host/sdhci-s3c.c -- 1.8.2.1

4 3

[alsa-devel] [PATCH] ASoC: McASP: Add pins output direction for rx clocks when configured in CBS_CFS format.
by Marek Belisko 30 Apr '13

30 Apr '13

When McASP is bit clock and frame clock master enable pin output for rx clocks. Signed-off-by: Marek Belisko <marek.belisko(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d54e0d9..2953c55 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -505,7 +505,10 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, ACLKX | AFSX); + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, + ACLKX | ACLKR); + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, + AFSX | AFSR); break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 0/3] ASoC: Add devicetree support for spdif dummy codecs.
by Marek Belisko 30 Apr '13

30 Apr '13

changes from v1 (for patch 2 and 3): - add prefix linux for compatible property (for both tx and rx) - rename spdif_transceiver.c to spdif_transmitter.c - add bindings documentation Marek Belisko (3): ASoC: spdif_transceiver: Change driver filename to spdif_transmitter.c. ASoC: spdif_transmitter: Add DT support. ASoC: spdif_receiver: Add DT support. .../devicetree/bindings/sound/spdif-receiver.txt | 10 ++++++++++ .../bindings/sound/spdif-transmitter.txt | 10 ++++++++++ sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/spdif_receiver.c | 10 ++++++++++ .../{spdif_transciever.c => spdif_transmitter.c} | 10 ++++++++++ 5 files changed, 41 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/spdif-receiver.txt create mode 100644 Documentation/devicetree/bindings/sound/spdif-transmitter.txt rename sound/soc/codecs/{spdif_transciever.c => spdif_transmitter.c} (88%) -- 1.7.9.5

4 7

Re: [alsa-devel] linux-next: Tree for Apr 30 (sound)
by Randy Dunlap 30 Apr '13

30 Apr '13

On 04/29/13 23:57, Stephen Rothwell wrote: > Hi all, > > Please do not add any v3.11 destined work to your linux-next included > branches until after v3.10-rc1 is released. > > Changes since 20130429: > when CONFIG_PROC_FS is not enabled: sound/core/memalloc.c: In function 'snd_mem_exit': sound/core/memalloc.c:525:20: error: 'SND_MEM_PROC_FILE' undeclared (first use in this function) -- ~Randy

1 0

[alsa-devel] [patch] ASoC: wm8994: missing break in wm8994_aif3_hw_params()
by Dan Carpenter 30 Apr '13

30 Apr '13

The missing break here means that we always return early and the function is a no-op. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 14094f5..1eb152c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2882,6 +2882,7 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, default: return 0; } + break; default: return 0; }

3 3

Re: [alsa-devel] what is the difference between SNDRV_PCM_FMTBIT_S24_LE and SNDRV_PCM_FMTBIT_S24_3LE
by Clemens Ladisch 30 Apr '13

30 Apr '13

(please do not drop the mailing list from replies.) Zhang wei wrote: > I do not really understand. For example,an codec is wm8985 ... The SNDRV_PCM_* symbols specify the format in which samples are stored in memory. I do not know how ASoC maps these symbols to on-the-wire formats. >>Most devices that support 24 bits use SNDRV_PCM_FORMAT_S32_LE, which >>stores the sample's bits in the upper bits of a 32-bit word. > > But how to distinguish the SNDRV_PCM_FORMAT_S24_LE and the REAL > SNDRV_PCM_FORMAT_S32_LE? If a 24-bit sample with the value 0x123456 is stored in memory, it looks like this: SNDRV_PCM_FORMAT_S24_LE: 56 34 12 00 SNDRV_PCM_FORMAT_S32_LE: 00 56 34 12 Regards, Clemens

3 2

[alsa-devel] [PATCH, for-next] ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
by Eldad Zack 30 Apr '13

30 Apr '13

Current code does this: be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]) Which is effectively (neglecting the index): be16_to_cpu(be16_to_cpu(*((u16 *) buf))) This means the int16 in the buffer is not converted at all. Daniel Mack confirmed that the driver works on little endian CPUs, leading to the conclusion that the device-side structure is actually little endian. This changes the code to use le16_to_cpu(). Caught by sparse. Acked-by: Daniel Mack <zonque(a)gmail.com> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/caiaq/input.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index efc70ae..4b3fb91 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -488,13 +488,13 @@ static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *cdev, unsigned int len) { unsigned int i, pad_id; - uint16_t pressure; + __le16 *pressure = (__le16 *) buf; for (i = 0; i < MASCHINE_PADS; i++) { - pressure = be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]); - pad_id = pressure >> 12; - - input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), pressure & 0xfff); + pad_id = le16_to_cpu(*pressure) >> 12; + input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), + le16_to_cpu(*pressure) & 0xfff); + pressure++; } input_sync(cdev->input_dev); -- 1.8.1.5

3 2

[alsa-devel] ASoC : playback and capture counts in soc_new_pcm
by JongHo Kim 29 Apr '13

29 Apr '13

Hi I am developing the ALSA ASoC part. I has some problem. Now my situation must only use the cpu_dai(AP I2S IP) to half duplex. So, I want to allocate either a capture substream or a playback substream. But ASoC(soc-pcm.c) referred to the playback(capture)_channel_min of codec_dai at the calling snd_pcm_new in soc_new_pcm. I do not want to modify the codec_dai codes. As a result, hope to see below /dev/snd/pcmC0D0p /dev/snd/pcmC0D1c attach the ASCII schematic diagram. Slaves Master +--------------+ +-----------------+ |I2S.0 data_out|---------------|data_in(AIF1) | |I2S.0 bclk |-------+-------|bclk | |I2S.0 lrclk |-------|--+----|lrclk | +--------------+ | | | | +--------------+ | | | AUDIO CODEC | |I2S.1 bclk |-------+ | | | |I2S.1 lrclk |----------+ | | |I2S.1 data_in |---------------|data_out(AIF1) | +--------------+ +-----------------+ attach the i2s and dai_link codes. ----------------------------------------------------------------------------- static struct snd_soc_dai_driver board_i2s_dai_hf_playback_template = { .probe = board_i2s_probe, .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &board_i2s_dai_ops, }; static struct snd_soc_dai_driver board_i2s_dai_hf_capture_template = { .probe = board_i2s_probe, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &board_i2s_dai_ops, }; static struct snd_soc_dai_link board_dai[] = { { .name = "audcodec_l0", .stream_name = "audcodec HiFi", .cpu_dai_name = "board-i2s.0", .codec_dai_name = "audcodec-aif1", .platform_name = "board-pcm-audio", .codec_name = "audcodec-codec", .ops = &board_ops, }, { .name = "audcodec_l1", .stream_name = "audcodec HiFi", .cpu_dai_name = "board-i2s.1", .codec_dai_name = "audcodec-aif1", .platform_name = "board-pcm-audio", .codec_name = "audcodec-codec", .ops = &board_ops, }, }; ----------------------------------------------------------------------------- So, I suggest the next diff. --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2014,9 +2014,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (cpu_dai->driver->capture.channels_min) capture = 1; } else { - if (codec_dai->driver->playback.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) capture = 1; } Thanks.

2 2

[alsa-devel] ALSA SOC Driver (mxs-wm8753)
by John Graham 29 Apr '13

29 Apr '13

Hi there, I'm completely new to both kernel/ALSA driver programming (at least at a serious level) and was looking for some help on a project I'm working on - I'm part way there but I've kind of hit a brick wall and don't know where to turn. I'm trying to get a codec working on a new platform (new for the codec - the platform itself is not new). Here are the details: The audio codec is the Wolfson WM8753, which already has a codec driver in the kernel mainline tree. We also have a vendor-supplied (but GPL) patch to make the wm8753 work on a CC9M2443 (based on the Samsung S3C2443). This includes the glue code ("platform" code?) in sound/soc/s3c24xx/cc9m2443js_wm8753.c. We have a product with this combination fully working. The platform is the Karo TX28 (based on the Freescale i.MX28), and there are already drivers in the kernel tree to make the TX28 work with an SGTL5000 codec in sound/soc/mxs/mxs-sgtl5000.c. We also have a product with this combination fully working. Now we want to move the WM8753 over from the CC9M2443 to work on the TX28. To simplify, what I *have* is: CC9M2443 <----> WM8753 TX28 <----> SGTL5000 and what I *want* is: /-> WM8753 TX28 <-/ What I was hoping I could do was just "mash" the existing drivers together - i.e. copy sound/soc/mxs/mxs-sgtl5000.c into sound/soc/mxs/mxs-wm8753.c, change the parts in it that were specific for the SGTL5000 to the parts that drive the wm8753 in sound/soc/s3c24xx/cc9m2443js_wm8753.c, tell the TX28 to probe for it and that it would "magically work" (hey, crazier things have happened...). Obviously, it does not work, but I've got it a little way down the line. The WM8752 is probed for by the TX28 and recognized - the I2C control works, and you can change the mixer settings with amixer. When I try to record (using arecord) I can see with an oscilloscope the I2S clock lines look correct (in that they look the same as when the CC9M2443 talks to the WM8753 at the same audio rate). The problem now is that the I2S data lines don't look right - the line from codec to TX28 looks like a (more-or-less) square wave with the same period as the I2S clock, but only going from ~3.1 - ~3.5 V, as opposed to the 0 - ~3.3 V I expect. Furthermore, it's a pretty regular square wave - it doesn't have all the breaks you see on a working I2S data line. I know I probably haven't provided the right detail to get to the bottom of this problem, but part of my problem is that I'm so new to this that I'm not entirely sure where the problem might lie, or what extra detail to include to ask for more help. Is anyone able to point me in the right direction/suggest some ideas as to what might be wrong/anything else at all? Any help would be much appreciated - many thanks, John Graham

2 1

[alsa-devel] Configure a CODEC which has I2C connected to CPU, but no PCM bus connection
by Craig McQueen 29 Apr '13

29 Apr '13

I have an embedded device with somewhat unusual configuration: there is a second audio CODEC (Wolfson WM8940) which is connected to the PCM interface of another device (a radio module, which is acting as PCM bus master), so that radio device can do analog audio I/O. So there is no CPU<-->CODEC PCM link, but the CODEC _is_ connected to the CPU's I2C, and the CPU needs to configure the CODEC registers to interface to the radio module. See the attached diagram which shows a simplified view of the configuration. This configuration came about because both the CPU and the radio module can _only_ operate as PCM bus masters. I am not sure how I could fit this into the ALSA ASoC architecture. Is the ALSA ASoC driver architecture able to accommodate this scenario? Can I use the WM8940 driver to configure the second CODEC through I2C? Could I simply set up a machine driver that defines a DAI including a pointer to the WM8940 CODEC DAI, but NULL pointer for CPU DAI? Can I set it up so the Wolfson CODEC gains could appear as ALSA mixer widgets? Regards, Craig McQueen

2 1

[alsa-devel] [PATCH] ASoC: wm8962: Convert to devm_input_allocate_device()
by Mark Brown 29 Apr '13

29 Apr '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 50b0983..1cee9b3 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3189,7 +3189,7 @@ static void wm8962_init_beep(struct snd_soc_codec *codec) struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int ret; - wm8962->beep = input_allocate_device(); + wm8962->beep = devm_input_allocate_device(codec->dev); if (!wm8962->beep) { dev_err(codec->dev, "Failed to allocate beep device\n"); return; @@ -3210,7 +3210,6 @@ static void wm8962_init_beep(struct snd_soc_codec *codec) ret = input_register_device(wm8962->beep); if (ret != 0) { - input_free_device(wm8962->beep); wm8962->beep = NULL; dev_err(codec->dev, "Failed to register beep device\n"); } @@ -3227,7 +3226,6 @@ static void wm8962_free_beep(struct snd_soc_codec *codec) struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); device_remove_file(codec->dev, &dev_attr_beep); - input_unregister_device(wm8962->beep); cancel_work_sync(&wm8962->beep_work); wm8962->beep = NULL; -- 1.7.10.4

4 11

[alsa-devel] ALSA: usb-audio: caiaq: endianness bug
by Eldad Zack 29 Apr '13

29 Apr '13

Hi Daniel, I think I found a endian usage issue in the caiaq driver. I have a patch to fix it (below), but it assumes that the driver works currently on little endian CPUs - I don't have a device to test it with. Cheers, Eldad -- >8 -- Subject: [PATCH] ALSA: usb-audio: caiaq: fix endianness bug Current code does this: be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]) Which is effectively: be16_to_cpu(be16_to_cpu(*((u16 *) buf))) This means the int16 in the buffer is not converted at all. Assuming the device-side structure is actually little endian, change the code to use le16_to_cpu(). Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/caiaq/input.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index efc70ae..2f4ae4d 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -488,13 +488,12 @@ static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *cdev, unsigned int len) { unsigned int i, pad_id; - uint16_t pressure; + __le16 *pressure = (__le16 *) buf; for (i = 0; i < MASCHINE_PADS; i++) { - pressure = be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]); - pad_id = pressure >> 12; - - input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), pressure & 0xfff); + pad_id = le16_to_cpu(*pressure) >> 12; + input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), le16_to_cpu(*pressure) & 0xfff); + pressure++; } input_sync(cdev->input_dev); -- 1.8.1.5

2 2

Re: [alsa-devel] linux-next: Tree for Apr 29 (sound/soc)
by Randy Dunlap 29 Apr '13

29 Apr '13

On 04/29/13 02:17, Stephen Rothwell wrote: > Hi all, > > Changes since 20130426: > on i386: CONFIG_I2C=m sound/built-in.o: In function `snd_soc_codec_set_cache_io': (.text+0x1ebe4): undefined reference to `regmap_init_i2c' make[1]: *** [vmlinux] Error 1 -- ~Randy

1 0

[alsa-devel] [PATCH 0/3] snd-firewire-lib: add handling CMP output connection
by Takashi Sakamoto 29 Apr '13

29 Apr '13

Current implementation of snd-firewire-lib can handle only CMP input connection. This series of patch enable it to handle both CMP output and input connection according to IEC 61883-1. Note that this series of patch still supports only a point-to-point connection without overlaying. The overlaying and any broadcast connection are not supported. Each connections gain its own isochronous resource. Takashi Sakamoto (3): rename macros, variables and functions related to CMP plug Add "direction" member to cmp_connection structure Add handling CMP output connection sound/firewire/cmp.c | 192 +++++++++++++++++++++++++++++++++------------ sound/firewire/cmp.h | 13 ++- sound/firewire/speakers.c | 2 +- 3 files changed, 152 insertions(+), 55 deletions(-) -- 1.7.10.4

2 12

[alsa-devel] Missing patches for 3.10 merge?
by Takashi Iwai 29 Apr '13

29 Apr '13

Hi, if anyone posted fix patches that should be applied to 3.10, but still unapplied, please let me know ASAP. I got a complete home (and system) FS crash on Friday (due to too much experiment with btrfs) and might have lost some incoming mails. thanks, Takashi

2 2

[alsa-devel] [PATCH] ALSA: compress: fix the states to check for allowing read
by Vinod Koul 29 Apr '13

29 Apr '13

for reading compressed data, we need to allow when we are paused, draining or stopped. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Cc: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Cc: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 10 ++++++++-- 1 files changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a0bc47f..5389b9a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -311,8 +311,14 @@ static ssize_t snd_compr_read(struct file *f, char __user *buf, stream = &data->stream; mutex_lock(&stream->device->lock); - /* read is allowed when stream is running */ - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + /* read is allowed when stream is running, paused, draining and setup + * (yes setup is state which we transistion to after stop, so if user + * wants to read data after stop we allow that + */ + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING || + stream->runtime->state != SNDRV_PCM_STATE_DRAINING || + stream->runtime->state != SNDRV_PCM_STATE_PAUSED || + stream->runtime->state != SNDRV_PCM_STATE_SETUP) { retval = -EBADFD; goto out; } -- 1.7.0.4

4 8

[alsa-devel] [PATCH] compress: fix the states to check for allowing read
by Vinod Koul 29 Apr '13

29 Apr '13

for reading compressed data, we need to allow when we are paused, draining or stopped. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Cc: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Cc: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 12 ++++++++++-- 1 files changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a0bc47f..7b774b7 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -311,8 +311,16 @@ static ssize_t snd_compr_read(struct file *f, char __user *buf, stream = &data->stream; mutex_lock(&stream->device->lock); - /* read is allowed when stream is running */ - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + /* read is allowed when stream is running, paused, draining and setup + * (yes setup is state which we transition to after stop, so if user + * wants to read data after stop we allow that + */ + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_XRUN: + case SNDRV_PCM_STATE_SUSPENDED: + case SNDRV_PCM_STATE_DISCONNECTED: retval = -EBADFD; goto out; } -- 1.7.0.4

2 1

[alsa-devel] [PATCH 0/4] snd-firewire-lib: add handling AMDTP receive stream
by Takashi Sakamoto 29 Apr '13

29 Apr '13

Current implementation of snd-firewire-lib can handle only AMDTP transmit stream with PCM. This series of patch enable it to handle both AMDTP receive and transmit stream with PCM according to IEC 61883-6. I develop with Echo Audio's Fireworks (AudioFirePre8). Unfortunately it's not fully compliant to IEC 61883-6. The last patch includes some work arounds for Fireworks. The snd-fireworks is still under development and not prepared for commit. Takashi Sakamoto (4): Rename functions, structure and member name for AMDTP Add macros for fixed value related to AMDTP Add "direction" member to amdtp_stream structure for AMDTP Add handling AMDTP receive stream sound/firewire/amdtp.c | 422 ++++++++++++++++++++++++++++++++++++--------- sound/firewire/amdtp.h | 64 +++---- sound/firewire/speakers.c | 33 ++-- 3 files changed, 393 insertions(+), 126 deletions(-) -- 1.7.10.4

2 13

[alsa-devel] [PATCH 1/1] ALSA: compress_core: Ensure device is stopped when stream closed
by Richard Fitzgerald 29 Apr '13

29 Apr '13

If the stream state is running or paused when device file is closed snd_compr_free() will send a SNDRV_PCM_TRIGGER_STOP to the stream. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 91e6bcb..a81c5e4 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -132,6 +132,11 @@ static int snd_compr_open(struct inode *inode, struct file *f) static int snd_compr_free(struct inode *inode, struct file *f) { struct snd_compr_file *data = f->private_data; + + if ((data->stream.runtime->state == SNDRV_PCM_STATE_RUNNING) || + (data->stream.runtime->state == SNDRV_PCM_STATE_PAUSED)) + data->stream.ops->trigger(&data->stream, SNDRV_PCM_TRIGGER_STOP); + data->stream.ops->free(&data->stream); kfree(data->stream.runtime->buffer); kfree(data->stream.runtime); -- 1.7.2.5

3 5

[alsa-devel] [PATCH] ALSA: hda - Move Thinkpad X220 to use auto parser
by David Henningsson 29 Apr '13

29 Apr '13

This enables better volume controls than the current model parser. Also, because the original quirk for X220 was added to fix docking station support, add the TP410 fixup instead. Reported-by: Willian Jon McCann <william.jon.mccann(a)gmail.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5499643..84b81c8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2946,7 +2946,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), @@ -3308,6 +3307,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: USB: adjust for changed 3.8 USB API
by Clemens Ladisch 29 Apr '13

29 Apr '13

The recent changes in the USB API ("implement new semantics for URB_ISO_ASAP") made the former meaning of the URB_ISO_ASAP flag the default, and changed this flag to mean that URBs can be delayed. This is not the behaviour wanted by any of the audio drivers because it leads to discontinuous playback with very small period sizes. Therefore, our URBs need to be submitted without this flag. Reported-by: Joe Rayhawk <jrayhawk(a)fairlystable.org> Cc: <stable(a)vger.kernel.org> # 3.8 only Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/6fire/pcm.c | 1 - sound/usb/caiaq/audio.c | 3 --- sound/usb/endpoint.c | 5 ++--- sound/usb/misc/ua101.c | 3 +-- sound/usb/usx2y/usb_stream.c | 1 - sound/usb/usx2y/usbusx2yaudio.c | 1 - sound/usb/usx2y/usx2yhwdeppcm.c | 1 - 7 files changed, 3 insertions(+), 12 deletions(-) diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index e2ca12f..40dd50a 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -575,7 +575,6 @@ static void usb6fire_pcm_init_urb(struct pcm_urb *urb, urb->instance.pipe = in ? usb_rcvisocpipe(chip->dev, ep) : usb_sndisocpipe(chip->dev, ep); urb->instance.interval = 1; - urb->instance.transfer_flags = URB_ISO_ASAP; urb->instance.complete = handler; urb->instance.context = urb; urb->instance.number_of_packets = PCM_N_PACKETS_PER_URB; diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index fde9a7a..b45e29b 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -670,7 +670,6 @@ static void read_completed(struct urb *urb) if (send_it) { out->number_of_packets = outframe; - out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); } else { struct snd_usb_caiaq_cb_info *oinfo = out->context; @@ -686,7 +685,6 @@ requeue: } urb->number_of_packets = FRAMES_PER_URB; - urb->transfer_flags = URB_ISO_ASAP; usb_submit_urb(urb, GFP_ATOMIC); } @@ -751,7 +749,6 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) * BYTES_PER_FRAME; urbs[i]->context = &dev->data_cb_info[i]; urbs[i]->interval = 1; - urbs[i]->transfer_flags = URB_ISO_ASAP; urbs[i]->number_of_packets = FRAMES_PER_URB; urbs[i]->complete = (dir == SNDRV_PCM_STREAM_CAPTURE) ? read_completed : write_completed; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 21049b8..63cca3a 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -677,7 +677,7 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, if (!u->urb->transfer_buffer) goto out_of_memory; u->urb->pipe = ep->pipe; - u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP; + u->urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP; u->urb->interval = 1 << ep->datainterval; u->urb->context = u; u->urb->complete = snd_complete_urb; @@ -716,8 +716,7 @@ static int sync_ep_set_params(struct snd_usb_endpoint *ep, u->urb->transfer_dma = ep->sync_dma + i * 4; u->urb->transfer_buffer_length = 4; u->urb->pipe = ep->pipe; - u->urb->transfer_flags = URB_ISO_ASAP | - URB_NO_TRANSFER_DMA_MAP; + u->urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP; u->urb->number_of_packets = 1; u->urb->interval = 1 << ep->syncinterval; u->urb->context = u; diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 8b81cb5..6ad617b 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -1120,8 +1120,7 @@ static int alloc_stream_urbs(struct ua101 *ua, struct ua101_stream *stream, usb_init_urb(&urb->urb); urb->urb.dev = ua->dev; urb->urb.pipe = stream->usb_pipe; - urb->urb.transfer_flags = URB_ISO_ASAP | - URB_NO_TRANSFER_DMA_MAP; + urb->urb.transfer_flags = URB_NO_TRANSFER_DMA_MAP; urb->urb.transfer_buffer = addr; urb->urb.transfer_dma = dma; urb->urb.transfer_buffer_length = max_packet_size; diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index 1e7a47a..bf618e1 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -69,7 +69,6 @@ static void init_pipe_urbs(struct usb_stream_kernel *sk, unsigned use_packsize, ++u, transfer += transfer_length) { struct urb *urb = urbs[u]; struct usb_iso_packet_descriptor *desc; - urb->transfer_flags = URB_ISO_ASAP; urb->transfer_buffer = transfer; urb->dev = dev; urb->pipe = pipe; diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 520ef96..b376532 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -503,7 +503,6 @@ static int usX2Y_urbs_start(struct snd_usX2Y_substream *subs) if (0 == i) atomic_set(&subs->state, state_STARTING3); urb->dev = usX2Y->dev; - urb->transfer_flags = URB_ISO_ASAP; for (pack = 0; pack < nr_of_packs(); pack++) { urb->iso_frame_desc[pack].offset = subs->maxpacksize * pack; urb->iso_frame_desc[pack].length = subs->maxpacksize; diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index cc56007..f2a1acd 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -443,7 +443,6 @@ static int usX2Y_usbpcm_urbs_start(struct snd_usX2Y_substream *subs) if (0 == u) atomic_set(&subs->state, state_STARTING3); urb->dev = usX2Y->dev; - urb->transfer_flags = URB_ISO_ASAP; for (pack = 0; pack < nr_of_packs(); pack++) { urb->iso_frame_desc[pack].offset = subs->maxpacksize * (pack + u * nr_of_packs()); urb->iso_frame_desc[pack].length = subs->maxpacksize;

3 2

[alsa-devel] [TINYCOMPRESS][PATCH] compress: check for config first
by Vinod Koul 26 Apr '13

26 Apr '13

if invalid config is passed, then we return error for memory Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- compress.c | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-) diff --git a/compress.c b/compress.c index e80e598..ee792c3 100644 --- a/compress.c +++ b/compress.c @@ -218,8 +218,13 @@ struct compress *compress_open(unsigned int card, unsigned int device, struct snd_compr_caps caps; char fn[256]; + if (!config) { + oops(&bad_compress, -EINVAL, "passed bad config"); + return &bad_compress; + } + compress = calloc(1, sizeof(struct compress)); - if (!compress || !config) { + if (!compress) { oops(&bad_compress, errno, "cannot allocate compress object"); return &bad_compress; } -- 1.7.0.4

1 0

[alsa-devel] [TINYCOMPRESS][PATCH 1/1] compress: Add support for capture streams
by Richard Fitzgerald 26 Apr '13

26 Apr '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- compress.c | 64 +++++++++++++++++++++++++++++++--- include/tinycompress/tinycompress.h | 2 +- 2 files changed, 59 insertions(+), 7 deletions(-) diff --git a/compress.c b/compress.c index 86cca88..e80e598 100644 --- a/compress.c +++ b/compress.c @@ -240,13 +240,12 @@ struct compress *compress_open(unsigned int card, unsigned int device, oops(&bad_compress, -EINVAL, "can't deduce device direction from given flags"); goto config_fail; } + if (flags & COMPRESS_OUT) { - /* this should be removed once we have capture tested */ - oops(&bad_compress, -EINVAL, "this version doesnt support capture"); - goto config_fail; + compress->fd = open(fn, O_RDONLY); + } else { + compress->fd = open(fn, O_WRONLY); } - - compress->fd = open(fn, O_WRONLY); if (compress->fd < 0) { oops(&bad_compress, errno, "cannot open device '%s'", fn); goto config_fail; @@ -403,7 +402,60 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size int compress_read(struct compress *compress, void *buf, unsigned int size) { - return oops(compress, -ENOTTY, "Not supported yet in lib"); + struct snd_compr_avail avail; + struct pollfd fds; + int to_read = 0; + int num_read, total = 0, ret; + char* cbuf = buf; + const unsigned int frag_size = compress->config->fragment_size; + + if (!(compress->flags & COMPRESS_OUT)) + return oops(compress, -EINVAL, "Invalid flag set"); + if (!is_compress_ready(compress)) + return oops(compress, -ENODEV, "device not ready"); + fds.fd = compress->fd; + fds.events = POLLIN; + + while (size) { + if (ioctl(compress->fd, SNDRV_COMPRESS_AVAIL, &avail)) + return oops(compress, errno, "cannot get avail"); + + if ( (avail.avail < frag_size) && (avail.avail < size) ) { + /* Less than one fragment available and not at the + * end of the read, so poll + */ + ret = poll(&fds, 1, compress->max_poll_wait_ms); + /* A pause will cause -EBADFD or zero. + * This is not an error, just stop reading */ + if ((ret == 0) || (ret == -EBADFD)) + break; + if (ret < 0) + return oops(compress, errno, "poll error"); + if (fds.revents & POLLIN) { + continue; + } + if (fds.revents & POLLERR) { + return oops(compress, -EIO, "poll returned error!"); + } + } + /* read avail bytes */ + if (size > avail.avail) + to_read = avail.avail; + else + to_read = size; + num_read = read(compress->fd, cbuf, to_read); + /* If play was paused the read returns -EBADFD */ + if (num_read == -EBADFD) + break; + if (num_read < 0) + return oops(compress, errno, "read failed!"); + + size -= num_read; + cbuf += num_read; + total += num_read; + } + + return total; } int compress_start(struct compress *compress) diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index ba9f3b6..9867afc 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -76,7 +76,7 @@ struct compr_gapless_mdata { __u32 encoder_padding; }; -#define COMPRESS_OUT 0x00000000 +#define COMPRESS_OUT 0x20000000 #define COMPRESS_IN 0x10000000 struct compress; -- 1.7.2.5

2 1

[alsa-devel] [PATCH 00/21] more arm build fixes
by Arnd Bergmann 26 Apr '13

26 Apr '13

Hi subsystem maintainers, Here is another set of patches that resulted from build testing on linux-next. Please apply directly into your trees if you agree, or let me know if I made a mistake. I can take whatever remains through the arm-soc tree if you prefer that or I don't hear back. Arnd Arnd Bergmann (21): ARM: topology: export cpu_topology ARM: default machine descriptor for multiplatform ARM: shmobile: don't call irqchip_init unconditionally ARM: orion5x: include linux/cpu.h atm: he: use mdelay instead of large udelay constants ALSA: ali5451: use mdelay instead of large udelay constants oss/dmabuf: use dma_map_single drm/nouveau: use mdelay instead of large udelay constants drm: export drm_vm_open_locked [SCSI] nsp32: use mdelay instead of large udelay constants irqdomain: export irq_domain_add_simple irqchip: s3c24xx: add missing __init annotations iommu: tegra: print dma_addr_t using %lld cpufreq: pxa2xx: initialize variables thermal: cooling: avoid uninitialied used gcc warning OF: remove #ifdef from linux/of_platform.h X.509: do not emit any informational output USB: ehci-msm: USB_MSM_OTG needs USB_PHY USB: lpc32xx: ISP1301 needs USB_PHY USB: OMAP: ISP1301 needs USB_PHY USB: OHCI: avoid conflicting platform drivers arch/arm/Kconfig | 1 - arch/arm/configs/lpc32xx_defconfig | 1 + arch/arm/configs/msm_defconfig | 1 + arch/arm/configs/omap1_defconfig | 1 + arch/arm/kernel/devtree.c | 7 ++ arch/arm/kernel/setup.c | 11 +- arch/arm/kernel/topology.c | 1 + arch/arm/mach-orion5x/common.c | 1 + arch/arm/mach-shmobile/intc-r8a7740.c | 13 +- drivers/atm/he.c | 2 +- drivers/cpufreq/pxa2xx-cpufreq.c | 5 +- drivers/gpu/drm/drm_vm.c | 1 + drivers/gpu/drm/nouveau/core/engine/disp/dacnv50.c | 3 +- drivers/iommu/tegra-gart.c | 3 +- drivers/iommu/tegra-smmu.c | 2 +- drivers/irqchip/irq-s3c24xx.c | 4 +- drivers/scsi/nsp32.c | 2 +- drivers/thermal/cpu_cooling.c | 17 +-- drivers/usb/gadget/Kconfig | 2 + drivers/usb/host/Kconfig | 1 + drivers/usb/host/ohci-hcd.c | 136 ++++++++++++++++++--- drivers/usb/phy/Makefile | 2 +- include/linux/of_platform.h | 13 +- kernel/irq/irqdomain.c | 1 + lib/build_OID_registry | 2 - sound/oss/dmabuf.c | 3 +- sound/pci/ali5451/ali5451.c | 8 +- 27 files changed, 187 insertions(+), 57 deletions(-) -- 1.8.1.2 Cc: "James E.J. Bottomley" <JBottomley(a)parallels.com> Cc: Benjamin Herrenschmidt <benh(a)kernel.crashing.org> Cc: David Airlie <airlied(a)linux.ie> Cc: Felipe Balbi <balbi(a)ti.com> Cc: Grant Likely <grant.likely(a)linaro.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Inki Dae <inki.dae(a)samsung.com> Cc: Jason Cooper <jason(a)lakedaemon.net> Cc: Joerg Roedel <joro(a)8bytes.org> Cc: Nicolas Pitre <nico(a)linaro.org> Cc: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> Cc: Rob Herring <robherring2(a)gmail.com> Cc: Russell King <rmk+kernel(a)arm.linux.org.uk> Cc: Simon Horman <horms+renesas(a)verge.net.au> Cc: Thomas Gleixner <tglx(a)linutronix.de> Cc: Viresh Kumar <viresh.kumar(a)linaro.org> Cc: Will Deacon <will.deacon(a)arm.com> Cc: alsa-devel(a)alsa-project.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-atm-general(a)lists.sourceforge.net Cc: linux-scsi(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: netdev(a)vger.kernel.org

2 4

[alsa-devel] [PATCH] alsa-lib: conf - Add Scarlett 2i2 USB to S/PDIF blacklist
by David Henningsson 26 Apr '13

26 Apr '13

The Scarlett 2i2 USB interface does not have any S/PDIF connections. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/conf/cards/USB-Audio.conf | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 71c2971..0d93175 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -47,6 +47,7 @@ USB-Audio.pcm.iec958_device { "Plantronics USB Headset" 999 "Plantronics Wireless Audio" 999 "SB WoW Headset" 999 + "Scarlett 2i2 USB" 999 "Sennheiser USB headset" 999 "USB Device 0x46d:0x992" 999 } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
by David Henningsson 26 Apr '13

26 Apr '13

The Scarlett 2i2 seems to take almost 500 ms to set the sample rate, even if the clock is currently set to that value. This patch speeds up prepare of the device, by avoiding setting the clock to something it already is. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/usb/clock.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b0ec364..3a2ce39 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -356,6 +356,8 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return clock; prev_rate = get_sample_rate_v2(chip, iface, fmt->altsetting, clock); + if (prev_rate == rate) + return 0; cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, clock); writeable = uac2_control_is_writeable(cs_desc->bmControls, UAC2_CS_CONTROL_SAM_FREQ - 1); -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH 04/28] proc: Supply PDE attribute setting accessor functions [RFC]
by Vasant Hegde 25 Apr '13

25 Apr '13

On 04/16/2013 11:56 PM, David Howells wrote: > Supply accessor functions to set attributes in proc_dir_entry structs. > > The following are supplied: proc_set_size() and proc_set_user(). > > Signed-off-by: David Howells<dhowells(a)redhat.com> > cc: linuxppc-dev(a)lists.ozlabs.org > cc: linux-media(a)vger.kernel.org > cc: netdev(a)vger.kernel.org > cc: linux-wireless(a)vger.kernel.org > cc: linux-pci(a)vger.kernel.org > cc: netfilter-devel(a)vger.kernel.org > cc: alsa-devel(a)alsa-project.org > --- > > arch/powerpc/kernel/proc_powerpc.c | 2 +- > arch/powerpc/platforms/pseries/reconfig.c | 2 +- arch/powerpc side changes looks good. -Vasant > drivers/media/pci/ttpci/av7110_ir.c | 2 +- > drivers/net/irda/vlsi_ir.c | 2 +- > drivers/net/wireless/airo.c | 34 +++++++++-------------------- > drivers/pci/proc.c | 2 +- > fs/proc/generic.c | 13 +++++++++++ > include/linux/proc_fs.h | 5 ++++ > kernel/configs.c | 2 +- > kernel/profile.c | 2 +- > net/netfilter/xt_recent.c | 3 +-- > sound/core/info.c | 2 +- > 12 files changed, 38 insertions(+), 33 deletions(-) > > diff --git a/arch/powerpc/kernel/proc_powerpc.c b/arch/powerpc/kernel/proc_powerpc.c > index 41d8ee9..feb8580 100644 > --- a/arch/powerpc/kernel/proc_powerpc.c > +++ b/arch/powerpc/kernel/proc_powerpc.c > @@ -83,7 +83,7 @@ static int __init proc_ppc64_init(void) > &page_map_fops, vdso_data); > if (!pde) > return 1; > - pde->size = PAGE_SIZE; > + proc_set_size(pde, PAGE_SIZE); > > return 0; > } > diff --git a/arch/powerpc/platforms/pseries/reconfig.c b/arch/powerpc/platforms/pseries/reconfig.c > index d6491bd..f93cdf5 100644 > --- a/arch/powerpc/platforms/pseries/reconfig.c > +++ b/arch/powerpc/platforms/pseries/reconfig.c > @@ -452,7 +452,7 @@ static int proc_ppc64_create_ofdt(void) > > ent = proc_create("powerpc/ofdt", S_IWUSR, NULL,&ofdt_fops); > if (ent) > - ent->size = 0; > + proc_set_size(ent, 0); > > return 0; > } > diff --git a/drivers/media/pci/ttpci/av7110_ir.c b/drivers/media/pci/ttpci/av7110_ir.c > index eb82286..0e763a7 100644 > --- a/drivers/media/pci/ttpci/av7110_ir.c > +++ b/drivers/media/pci/ttpci/av7110_ir.c > @@ -375,7 +375,7 @@ int av7110_ir_init(struct av7110 *av7110) > if (av_cnt == 1) { > e = proc_create("av7110_ir", S_IWUSR, NULL,&av7110_ir_proc_fops); > if (e) > - e->size = 4 + 256 * sizeof(u16); > + proc_set_size(e, 4 + 256 * sizeof(u16)); > } > > tasklet_init(&av7110->ir.ir_tasklet, av7110_emit_key, (unsigned long)&av7110->ir); > diff --git a/drivers/net/irda/vlsi_ir.c b/drivers/net/irda/vlsi_ir.c > index e22cd4e..5f47584 100644 > --- a/drivers/net/irda/vlsi_ir.c > +++ b/drivers/net/irda/vlsi_ir.c > @@ -1678,7 +1678,7 @@ vlsi_irda_probe(struct pci_dev *pdev, const struct pci_device_id *id) > IRDA_WARNING("%s: failed to create proc entry\n", > __func__); > } else { > - ent->size = 0; > + proc_set_size(ent, 0); > } > idev->proc_entry = ent; > } > diff --git a/drivers/net/wireless/airo.c b/drivers/net/wireless/airo.c > index 66e398d..21d0233 100644 > --- a/drivers/net/wireless/airo.c > +++ b/drivers/net/wireless/airo.c > @@ -4507,73 +4507,63 @@ static int setup_proc_entry( struct net_device *dev, > airo_entry); > if (!apriv->proc_entry) > goto fail; > - apriv->proc_entry->uid = proc_kuid; > - apriv->proc_entry->gid = proc_kgid; > + proc_set_user(apriv->proc_entry, proc_kuid, proc_kgid); > > /* Setup the StatsDelta */ > entry = proc_create_data("StatsDelta", S_IRUGO& proc_perm, > apriv->proc_entry,&proc_statsdelta_ops, dev); > if (!entry) > goto fail_stats_delta; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Stats */ > entry = proc_create_data("Stats", S_IRUGO& proc_perm, > apriv->proc_entry,&proc_stats_ops, dev); > if (!entry) > goto fail_stats; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Status */ > entry = proc_create_data("Status", S_IRUGO& proc_perm, > apriv->proc_entry,&proc_status_ops, dev); > if (!entry) > goto fail_status; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Config */ > entry = proc_create_data("Config", proc_perm, > apriv->proc_entry,&proc_config_ops, dev); > if (!entry) > goto fail_config; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the SSID */ > entry = proc_create_data("SSID", proc_perm, > apriv->proc_entry,&proc_SSID_ops, dev); > if (!entry) > goto fail_ssid; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the APList */ > entry = proc_create_data("APList", proc_perm, > apriv->proc_entry,&proc_APList_ops, dev); > if (!entry) > goto fail_aplist; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the BSSList */ > entry = proc_create_data("BSSList", proc_perm, > apriv->proc_entry,&proc_BSSList_ops, dev); > if (!entry) > goto fail_bsslist; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the WepKey */ > entry = proc_create_data("WepKey", proc_perm, > apriv->proc_entry,&proc_wepkey_ops, dev); > if (!entry) > goto fail_wepkey; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > - > + proc_set_user(entry, proc_kuid, proc_kgid); > return 0; > > fail_wepkey: > @@ -5695,10 +5685,8 @@ static int __init airo_init_module( void ) > > airo_entry = proc_mkdir_mode("driver/aironet", airo_perm, NULL); > > - if (airo_entry) { > - airo_entry->uid = proc_kuid; > - airo_entry->gid = proc_kgid; > - } > + if (airo_entry) > + proc_set_user(airo_entry, proc_kuid, proc_kgid); > > for (i = 0; i< 4&& io[i]&& irq[i]; i++) { > airo_print_info("", "Trying to configure ISA adapter at irq=%d " > diff --git a/drivers/pci/proc.c b/drivers/pci/proc.c > index 12e4fb5..7cde7c1 100644 > --- a/drivers/pci/proc.c > +++ b/drivers/pci/proc.c > @@ -419,7 +419,7 @@ int pci_proc_attach_device(struct pci_dev *dev) > &proc_bus_pci_operations, dev); > if (!e) > return -ENOMEM; > - e->size = dev->cfg_size; > + proc_set_size(e, dev->cfg_size); > dev->procent = e; > > return 0; > diff --git a/fs/proc/generic.c b/fs/proc/generic.c > index 1c07cad..5f6f6c3 100644 > --- a/fs/proc/generic.c > +++ b/fs/proc/generic.c > @@ -498,6 +498,19 @@ out: > return NULL; > } > EXPORT_SYMBOL(proc_create_data); > + > +void proc_set_size(struct proc_dir_entry *de, loff_t size) > +{ > + de->size = size; > +} > +EXPORT_SYMBOL(proc_set_size); > + > +void proc_set_user(struct proc_dir_entry *de, kuid_t uid, kgid_t gid) > +{ > + de->uid = uid; > + de->gid = gid; > +} > +EXPORT_SYMBOL(proc_set_user); > > static void free_proc_entry(struct proc_dir_entry *de) > { > diff --git a/include/linux/proc_fs.h b/include/linux/proc_fs.h > index 805edac..28a4d7e 100644 > --- a/include/linux/proc_fs.h > +++ b/include/linux/proc_fs.h > @@ -130,6 +130,9 @@ static inline struct proc_dir_entry *proc_create(const char *name, umode_t mode, > extern struct proc_dir_entry *proc_net_mkdir(struct net *net, const char *name, > struct proc_dir_entry *parent); > > +extern void proc_set_size(struct proc_dir_entry *, loff_t); > +extern void proc_set_user(struct proc_dir_entry *, kuid_t, kgid_t); > + > extern struct file *proc_ns_fget(int fd); > extern bool proc_ns_inode(struct inode *inode); > > @@ -158,6 +161,8 @@ static inline struct proc_dir_entry *proc_mkdir(const char *name, > struct proc_dir_entry *parent) {return NULL;} > static inline struct proc_dir_entry *proc_mkdir_mode(const char *name, > umode_t mode, struct proc_dir_entry *parent) { return NULL; } > +static inline void proc_set_size(struct proc_dir_entry *de, loff_t size) {} > +static inline void proc_set_user(struct proc_dir_entry *de, kuid_t uid, kgid_t gid) {} > > struct tty_driver; > static inline void proc_tty_register_driver(struct tty_driver *driver) {}; > diff --git a/kernel/configs.c b/kernel/configs.c > index 42e8fa0..c18b1f1 100644 > --- a/kernel/configs.c > +++ b/kernel/configs.c > @@ -79,7 +79,7 @@ static int __init ikconfig_init(void) > if (!entry) > return -ENOMEM; > > - entry->size = kernel_config_data_size; > + proc_set_size(entry, kernel_config_data_size); > > return 0; > } > diff --git a/kernel/profile.c b/kernel/profile.c > index 524ce5e..0bf4007 100644 > --- a/kernel/profile.c > +++ b/kernel/profile.c > @@ -600,7 +600,7 @@ int __ref create_proc_profile(void) /* false positive from hotcpu_notifier */ > NULL,&proc_profile_operations); > if (!entry) > return 0; > - entry->size = (1+prof_len) * sizeof(atomic_t); > + proc_set_size(entry, (1 + prof_len) * sizeof(atomic_t)); > hotcpu_notifier(profile_cpu_callback, 0); > return 0; > } > diff --git a/net/netfilter/xt_recent.c b/net/netfilter/xt_recent.c > index 3db2d38..1e657cf 100644 > --- a/net/netfilter/xt_recent.c > +++ b/net/netfilter/xt_recent.c > @@ -401,8 +401,7 @@ static int recent_mt_check(const struct xt_mtchk_param *par, > ret = -ENOMEM; > goto out; > } > - pde->uid = uid; > - pde->gid = gid; > + proc_set_user(pde, uid, gid); > #endif > spin_lock_bh(&recent_lock); > list_add_tail(&t->list,&recent_net->tables); > diff --git a/sound/core/info.c b/sound/core/info.c > index 3aa8864..c7f41c3 100644 > --- a/sound/core/info.c > +++ b/sound/core/info.c > @@ -970,7 +970,7 @@ int snd_info_register(struct snd_info_entry * entry) > mutex_unlock(&info_mutex); > return -ENOMEM; > } > - p->size = entry->size; > + proc_set_size(p, entry->size); > } > entry->p = p; > if (entry->parent) > > _______________________________________________ > Linuxppc-dev mailing list > Linuxppc-dev(a)lists.ozlabs.org > https://lists.ozlabs.org/listinfo/linuxppc-dev >

1 0

[alsa-devel] [PATCH v5 00/10] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 25 Apr '13

25 Apr '13

Hi, This series adds DT support for phycore-ac97. It is now based on the pcm dma cleanups from Lars. Beside ac97 support, the series adds fsl-ssi imx-pcm-fiq and generic DMA binding handling. Regards, Markus Changes in v5: - Pass of_node via parent device to imx-pcm-dma. Changes in v4: - phycore-ac97 uses the iomux functions based on audmux phandle now. - fsl-ssi imx-pcm-fiq got some new comments about problems with imx-ssi and incompatible codecs Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (10): ASoC: phycore-ac97: Add DT support ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ARM: imx: Export ac97 reset functions ASoC: fsl-ssi: imx ac97 support ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97 ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 10 + .../bindings/sound/phytec,phycore-ac97.txt | 14 + arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl_ssi.c | 510 +++++++++++++++++---- sound/soc/fsl/imx-pcm-dma.c | 4 +- sound/soc/fsl/imx-pcm-fiq.c | 16 +- sound/soc/fsl/imx-pcm.h | 9 + sound/soc/fsl/imx-ssi.c | 7 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/phycore-ac97.c | 166 ++++++- 12 files changed, 631 insertions(+), 122 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (88%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt

3 15

[alsa-devel] ASoC updates for v3.10
by Mark Brown 25 Apr '13

25 Apr '13

The following changes since commit 22f38f792ec53e2a93be13ecb609bbe911ed8ff9: ASoC: ux500: Use generic dmaengine PCM (2013-04-18 15:04:44 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.10-3 for you to fetch changes up to 5cc50fc858a5ab37dc2744d72d7ffed96f23afd8: Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next (2013-04-23 19:26:15 +0100) Diffstat and log are inflated by Linus' tree again, plus git has apparently got itself a bit confused about what to diff against and picked the wrong side of a merge. ---------------------------------------------------------------- ASoC: More updates for v3.10 A few more fixes, nothing too major though the DMA changes fix modular builds. ---------------------------------------------------------------- Aaro Koskinen (1): ARM: 7692/1: iop3xx: move IOP3XX_PERIPHERAL_VIRT_BASE Abhijeet Dharmapurikar (1): irqchip: gic: fix irq_trigger return Akinobu Mita (2): sparc/srmmu: clear trailing edge of bitmap properly sparc/iommu: fix typo s/265KB/256KB/ Alban Bedel (1): ASoC: wm8903: Add the DAC boost control Alex Elder (1): rbd: do a safe list traversal in rbd_img_request_submit() Alexander Duyck (1): igb: Revert support for build_skb in igb Alistair Popple (1): powerpc: Fix audit crash due to save/restore PPR changes Andreas Schwab (1): kbuild: generate generic headers before recursing into scripts Andrei Epure (1): drivers/video/mmp/core.c: fix use-after-free bug Andrey Smirnov (2): ASoC: si476x: Convert SI476X codec to use regmap ASoC: si476x: Cosmetic changes to SI476X codec driver Andy Lutomirski (2): userns: Check uid_map's opener's fsuid, not the current fsuid userns: Changing any namespace id mappings should require privileges Antonio Quartulli (1): batman-adv: make is_my_mac() check for the current mesh only Arend van Spriel (2): brcmfmac: remove advertising P2P device support brcmsmac: request firmware in .start() callback Ariel Elior (1): bnx2x: Fix status blocks configuration Arnd Bergmann (6): ASoC: samsung: use irq resource for idma ASoC: samsung: fix module_device_table ASoC: samsung: export idma_reg_addr_init ASoC: samsung: move plat/ headers to local directory net: ax88796: avoid 64 bit arithmetic ASoC: ux500: forward declare msp_i2s_platform_data Bastian Hecht (1): ARM: 7697/1: hw_breakpoint: do not use __cpuinitdata for dbg_cpu_pm_nb Ben Greear (1): net: rate-limit warn-bad-offload splats. Bjørn Mork (4): net: cdc_mbim: remove bogus sizeof() net: qmi_wwan: fixup missing ethernet header (firmware bug workaround) net: qmi_wwan: fixup destination address (firmware bug workaround) net: qmi_wwan: prevent duplicate mac address on link (firmware bug workaround) Charles Keepax (12): ASoC: arizona: Move selection of FLL REFCLK into init ASoC: arizona: Tidy up SYNCCLK selection and cache values ASoC: arizona: Factor out check for enabled FLL ASoC: arizona: Factor out FLL disable ASoC: arizona: Factor out FLL enable ASoC: arizona: Improve suppression of noop FLL updates ASoC: arizona: Add support for directly setting the FLL REFCLK ASoC: arizona: Add convience define for clearing SYNCCLK ASoC: arizona: Don't enable FLL on REFCLK configuration ASoC: wm5102: Correctly use SOC_VALUE_ENUM for ISRC FSL controls ASoC: wm0010: Constify usage of firmware filenames ASoC: wm0010: Report filename when we fail to load firmware Christian Ruppert (1): stmmac: prevent interrupt loop with MMC RX IPC Counter Christoph Fritz (1): can: sja1000: fix handling on dt properties on little endian systems Cong Wang (1): ipv6: statically link register_inet6addr_notifier() Dan Carpenter (2): ASoC: tas5086: signedness bug in tas5086_hw_params() irda: small read past the end of array in debug code Daniel Mack (5): ASoC: ak4104: convert to direct regmap API usage ASoC: ak4104: correct tranceiver enable handling ASoC: Add codec driver for AK5386 ALSA: ASoC: add codec driver for TI TAS5086 ASoC: davinci-mcasp: clean up davinci_hw_common_param() David S. Miller (10): Merge branch 'master' of git://git.kernel.org/.../pablo/nf into netfilter Merge branch 'fixes-for-3.9' of git://gitorious.org/linux-can/linux-can Merge branch 'fixes' of git://git.kernel.org/.../jesse/openvswitch Merge tag 'batman-adv-fix-for-davem' of git://git.open-mesh.org/linux-merge Merge branch 'qlogic' Merge branch 'for-davem' of git://git.kernel.org/.../linville/wireless Merge branch 'master' of git://git.kernel.org/.../pablo/nf sparc64: Fix race in TLB batch processing. Merge branch 'bonding' Merge branch 'qmi_wwan' David Ward (1): net/802/mrp: fix possible race condition when calling mrp_pdu_queue() Dmitry Kravkov (1): bnx2x: Prevent UNDI FW illegal host access Dmitry Popov (1): tcp: incoming connections might use wrong route under synflood Emese Revfy (1): kernel/signal.c: stop info leak via the tkill and the tgkill syscalls Eric Dumazet (3): net: drop dst before queueing fragments bonding: fix l23 and l34 load balancing in forwarding path tcp: call tcp_replace_ts_recent() from tcp_ack() Eric W. Biederman (1): userns: Don't let unprivileged users trick privileged users into setting the id_map Felix Fietkau (2): ath9k_htc: accept 1.x firmware newer than 1.3 ath9k_hw: change AR9580 initvals to fix a stability issue Florian Westphal (2): netfilter: nf_nat: fix race when unloading protocol modules netfilter: xt_rpfilter: skip locally generated broadcast/multicast, too Greg Rose (1): ixgbe: Fix a bug in setting VF VLAN via PF H. Peter Anvin (2): x86, microcode: Verify the family before dispatching microcode patching Merge remote-tracking branch 'efi/urgent' into x86/urgent Heiko Stübner (1): ASoC: samsung: fix neo1973-wm8753 compilation Himanshu Madhani (1): qlcnic: fix beaconing test for 82xx adapter Hong Zhiguo (1): openvswitch: correct an invalid BUG_ON Illia Ragozin (1): ARM: 7696/1: Fix kexec by setting outer_cache.inv_all for Feroceon Ivan Vecera (1): be2net: take care of __vlan_put_tag return value Jason Gerecke (2): Input: wacom - fix "can not retrieve extra class descriptor" for 24HDT Input: wacom - correct reported resolution for Intuos4 Wireless Jeeja KP (1): ASoC: compress - add support for metadata apis Jesse Gross (1): openvswitch: Preallocate reply skb in ovs_vport_cmd_set(). Jitendra Kalsaria (3): qlcnic: Stop traffic before performing loopback test qlge: Fix ethtool autoneg advertising. qlge: Update version to 1.00.00.32. Joe Perches (1): checkpatch: fix stringification macro defect Johannes Berg (2): mac80211: fix LED in idle handling mac80211: fix cfg80211 interaction on auth/assoc request John W. Linville (3): Merge branch 'for-john' of git://git.kernel.org/.../jberg/mac80211 Merge branch 'for-john' of git://x-git.kernel.org/.../jberg/mac80211 Merge branch 'master' of git://git.kernel.org/.../linville/wireless into for-davem Joonsoo Kim (1): ARM: 7694/1: ARM, TCM: initialize TCM in paging_init(), instead of setup_arch() Josh Wu (1): avr32: fix build error in atstk1006_defconfig Jozsef Kadlecsik (3): netfilter: ipset: list:set: fix reference counter update netfilter: ipset: hash:*net*: nomatch flag not excluded on set resize netfilter: ipset: bitmap:ip,mac: fix listing with timeout K. Y. Srinivasan (1): x86, hyperv: Handle Xen emulation of Hyper-V more gracefully Kefeng Wang (2): sparc:remove unused declaration smp_boot_cpus() sparc:cleanup unused code in smp_32.h Kevin Hao (1): powerpc: add a missing label in resume_kernel Kuninori Morimoto (58): ASoC: fsi: remove unused irq ASoC: add snd_soc_register_component() ASoC: add .name for snd_soc_component_driver ASoC: fsi: use snd_soc_register_component() instead of snd_soc_register_dais() ASoC: pxa2xx-ac97: move EXPORT_SYMBOL_GPL() next to definition ASoC: snd_soc_register_component() uses properly snd_soc_register_dai[s]() ASoC: switch over to use snd_soc_register_component() on davinci i2s ASoC: switch over to use snd_soc_register_component() on davinci vcif ASoC: switch over to use snd_soc_register_component() on davinci mcasp ASoC: switch over to use snd_soc_register_component() on dw i2s ASoC: switch over to use snd_soc_register_component() on mpc5200 ac97 ASoC: switch over to use snd_soc_register_component() on atmel ssc ASoC: switch over to use snd_soc_register_component() on au1x i2sc ASoC: switch over to use snd_soc_register_component() on au1x psc-ac97 ASoC: switch over to use snd_soc_register_component() on au1x psc-i2s ASoC: switch over to use snd_soc_register_component() on au1x ac97c ASoC: switch over to use snd_soc_register_component() on bf6xx i2s ASoC: switch over to use snd_soc_register_component() on bf5xx ac97 ASoC: switch over to use snd_soc_register_component() on bf5xx i2s ASoC: switch over to use snd_soc_register_component() on bf5xx tdm ASoC: switch over to use snd_soc_register_component() on ep93xx ac97 ASoC: switch over to use snd_soc_register_component() on ep93xx i2s ASoC: switch over to use snd_soc_register_component() on mpc5200 i2s ASoC: switch over to use snd_soc_register_component() on fsl ssi ASoC: switch over to use snd_soc_register_component() on imx ssi ASoC: switch over to use snd_soc_register_component() on jz4740 i2s ASoC: switch over to use snd_soc_register_component() on kirkwood i2s ASoC: switch over to use snd_soc_register_component() on sst ASoC: switch over to use snd_soc_register_component() on mxs saif ASoC: switch over to use snd_soc_register_component() on nuc900 ac97 ASoC: switch over to use snd_soc_register_component() on omap mcbsp ASoC: switch over to use snd_soc_register_component() on omap mcpdm ASoC: switch over to use snd_soc_register_component() on omap dmic ASoC: switch over to use snd_soc_register_component() on omap hdmi ASoC: switch over to use snd_soc_register_component() on pxa ssp ASoC: switch over to use snd_soc_register_component() on pxa2xx ac97 ASoC: switch over to use snd_soc_register_component() on pxa2xx i2s ASoC: switch over to use snd_soc_register_component() on mmp sspa ASoC: switch over to use snd_soc_register_component() on s6000 i2s ASoC: switch over to use snd_soc_register_component() on s3c24xx i2s ASoC: switch over to use snd_soc_register_component() on s3c i2s ASoC: switch over to use snd_soc_register_component() on s3c ac97 ASoC: switch over to use snd_soc_register_component() on samsung spdif ASoC: switch over to use snd_soc_register_component() on samsung i2s ASoC: switch over to use snd_soc_register_component() on samsung pcm ASoC: switch over to use snd_soc_register_component() on goni_wm8994 ASoC: switch over to use snd_soc_register_component() on sh4 ssi ASoC: switch over to use snd_soc_register_component() on sh4 hac ASoC: switch over to use snd_soc_register_component() on spear spdif out ASoC: switch over to use snd_soc_register_component() on spear spdif in ASoC: switch over to use snd_soc_register_component() on tegra30 i2s ASoC: switch over to use snd_soc_register_component() on tegra20 spdif ASoC: switch over to use snd_soc_register_component() on tegra20 ac97 ASoC: switch over to use snd_soc_register_component() on tegra20 i2s ASoC: switch over to use snd_soc_register_component() on txx9aclc ac97 ASoC: switch over to use snd_soc_register_component() on ux500 msp ASoC: snd_soc_[un]register_dai[s]() become non global function ASoC: switch over to use snd_soc_register_component() on sh4 siu Lars-Peter Clausen (5): ASoC: spear_pcm: Staticize non-exported structs ASoC: generic-dmaengine-pcm: Add support for half-duplex ASoC: mxs-pcm: Set SNDRV_PCM_INFO_HALF_DUPLEX ASoC: mxs: Setup dma data in DAI probe ASoC: mxs: Use generic dmaengine PCM Len Brown (2): intel_idle: additional Haswell CPU-id tools/power turbostat: additional Haswell CPU-id Linus Torvalds (32): Merge branches 'timers-urgent-for-linus', 'irq-urgent-for-linus' and 'core-urgent-for-linus' of git://git.kernel.org/.../tip/tip vm: add vm_iomap_memory() helper function Merge branch 'for-linus' of git://git.kernel.org/.../dtor/input Merge branch 'rc-fixes' of git://git.kernel.org/.../mmarek/kbuild Merge tag 'for-linus' of git://git.kernel.org/.../sfr/next-fixes Merge git://git.kernel.org/pub/scm/virt/kvm/kvm Merge branch 'fixes' of git://git.linaro.org/people/rmk/linux-arm s390: move dummy io_remap_pfn_range() to asm/pgtable.h Merge git://git.kernel.org/.../davem/net Merge branch 'for-linus' of git://git.kernel.org/.../sage/ceph-client Merge branch 'akpm' (fixes from Andrew) Merge branch 'release' of git://git.kernel.org/.../lenb/linux Revert "block: add missing block_bio_complete() tracepoint" Merge branch 'for_linus' of git://cavan.codon.org.uk/platform-drivers-x86 Merge branch 'userns-fixes' of git://git.kernel.org/.../luto/linux Merge branch 'for-linus' of git://git.kernel.org/.../mszeredi/fuse Merge branch 'for-linus' of git://git.kernel.org/.../dtor/input vm: convert HPET mmap to vm_iomap_memory() helper vm: convert mtdchar mmap to vm_iomap_memory() helper vm: convert fb_mmap to vm_iomap_memory() helper vm: convert snd_pcm_lib_mmap_iomem() to vm_iomap_memory() helper mtdchar: remove no-longer-used vma helpers Merge tag 'fixes-for-linus' of git://git.kernel.org/.../arm/arm-soc net: fix incorrect credentials passing Merge git://git.kernel.org/.../davem/net Merge git://git.kernel.org/.../davem/sparc Merge branch 'fixes' of git://git.linaro.org/people/rmk/linux-arm Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'x86-kdump-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'vm_ioremap_memory-examples' Merge branch 'perf-urgent-for-linus' of git://git.kernel.org/.../tip/tip Linux 3.9-rc8 Lucas Stach (1): net: fec: fix regression in link change accounting Ludovic Desroches (1): dmaengine: at_hdmac: fix race condition in atc_advance_work() Marc Kleine-Budde (1): can: mcp251x: add missing IRQF_ONESHOT to request_threaded_irq Marc Zyngier (3): ARM: KVM: fix KVM_CAP_ARM_SET_DEVICE_ADDR reporting ARM: KVM: fix L_PTE_S2_RDWR to actually be Read/Write ARM: KVM: fix unbalanced get_cpu() in access_dcsw Marcelo Tosatti (1): Merge branch 'kvm-arm-fixes-3.9' of git://github.com/columbia/linux-kvm-arm Mark Brown (76): ASoC: arizona: Only allow input volume updates when inputs are enabled ASoC: wm8960: Add input boost volume control ASoC: arizona: If we only have a clock to synchronise with make it REFCLK mfd: arizona: Define additional FLL control registers mfd: wm5102: Map in additional FLL control registers ASoC: arizona: Increase FLL synchroniser bandwidth for high frequencies ASoC: arizona: Optimise FLL loop gains ASoC: arizona: Ensure synchroniser is disabled when not needed ASoC: arizona: Suppress reference calculations when setting REFCLK to 0 ASoC: arizona: Provide defines for the clock rates ASoC: wm_adsp: Handle old .bin files ASoC: arizona: Ensure we clock two channels for I2S mode ASoC: wm8994: Support constraining the maximum number of channels clocked mfd: wm5102: Add additional speaker control registers ASoC: arizona: Factor out speaker widgets from CODEC drivers ASoC: arizona: Log thermal events ASoC: arizona: Suppress speaker enable if thermal shutdown is flagged ASoC: arizona: Basic support for ISRC rate selection ASoC: wm8994: Restore AIFnCLK after reducing it for low clock rates extcon: arizona: Factor out magic application ASoC: arizona: Fix interaction between headphone outputs and identification extcon: arizona: Fix interaction between headphone outputs and identification Merge tag 'arizona-extcon-asoc' of git://git.kernel.org/.../broonie/misc into asoc-arizona ASoC: arizona: Add delay after powering up line level outputs Merge branch 'topic/pxa' of git://git.kernel.org/.../broonie/sound into asoc-component ASoC: wm5102: Implement OSR support ASoC: wm2000: Expose some more registers for diagnostics ASoC: wm_adsp: Provide defines for firmwares ASoC: wm_adsp: Split ADSP1 and ADSP2 firmware controls Merge branch 'fix/samsung' of git://git.kernel.org/.../broonie/sound into asoc-samsung ASoC: core: Merge Samsung fixes to avoid trivial context conflict ASoC: wm8994: Remove duplicate revision cache ASoC: wm_adsp: Add support for firmware wide coefficient blocks Merge remote-tracking branch 'asoc/fix/compress' into asoc-next Merge remote-tracking branch 'asoc/fix/core' into asoc-next Merge remote-tracking branch 'asoc/fix/samsung' into asoc-next Merge remote-tracking branch 'asoc/fix/tegra' into asoc-next Merge remote-tracking branch 'asoc/fix/wm8903' into asoc-next Merge remote-tracking branch 'asoc/topic/adau1373' into asoc-next Merge remote-tracking branch 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4104' into asoc-next Merge remote-tracking branch 'asoc/topic/ak5386' into asoc-next Merge remote-tracking branch 'asoc/topic/arizona' into asoc-next Merge remote-tracking branch 'asoc/topic/atmel' into asoc-next Merge remote-tracking branch 'asoc/topic/component' into asoc-next Merge remote-tracking branch 'asoc/topic/compress' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-next Merge remote-tracking branch 'asoc/topic/cs42l73' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/fsi' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/maintainers' into asoc-next Merge remote-tracking branch 'asoc/topic/max98088' into asoc-next Merge remote-tracking branch 'asoc/topic/max98090' into asoc-next Merge remote-tracking branch 'asoc/topic/samsung' into asoc-next Merge remote-tracking branch 'asoc/topic/si476x' into asoc-next Merge remote-tracking branch 'asoc/topic/spear' into asoc-next Merge remote-tracking branch 'asoc/topic/tas5086' into asoc-next Merge remote-tracking branch 'asoc/topic/tegra' into asoc-next Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next Merge remote-tracking branch 'asoc/topic/wm-hubs' into asoc-next Merge remote-tracking branch 'asoc/topic/wm0010' into asoc-next Merge remote-tracking branch 'asoc/topic/wm2000' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8903' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8960' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/max98088' into asoc-next Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next Masami Hiramatsu (1): kprobes: Fix a double lock bug of kprobe_mutex Matt Fleming (1): x86, efivars: firmware bug workarounds should be in platform code Matthew Garrett (4): Move utf16 functions to kernel core and rename efi: Pass boot services variable info to runtime code efi: Distinguish between "remaining space" and actually used space Revert "hp-wmi: Add support for SMBus hotkeys" Michael Bohan (1): hrtimer: Don't reinitialize a cpu_base lock on CPU_UP Michael Neuling (1): powerpc: fix compiling CONFIG_PPC_TRANSACTIONAL_MEM when CONFIG_ALTIVEC=n Michal Bachraty (3): ASoC: davinci-mcasp: Add support for multichannel playback ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes Miklos Szeredi (1): fuse: fix type definitions in uapi header Mugunthan V N (1): drivers: net: ethernet: cpsw: get slave VLAN id from slave node instead of cpsw node Naoya Horiguchi (3): hugetlbfs: stop setting VM_DONTDUMP in initializing vma(VM_HUGETLB) fs/binfmt_elf.c: fix hugetlb memory check in vma_dump_size() hugetlbfs: add swap entry check in follow_hugetlb_page() Oleg Nesterov (1): thinkpad-acpi: kill hotkey_thread_mutex Patrick McHardy (1): netfilter: nf_ct_sip: don't drop packets with offsets pointing outside the packet Paul Bolle (2): ASoC: samsung: remove last traces of neo1973-gta01 ASoC: Ux500: remove test for undefined Kconfig macro Paul Handrigan (1): ASoC: cs42l73: If Internal MCLK is >= 6.4MHz, then set SCLK to 64*Fs. Ping Cheng (2): Input: wacom - DTH2242 Grip Pen id was off by one bit Input: wacom - fix "can not retrieve extra class descriptor" for DTH2242 Rafał Miłecki (1): ssb: implement spurious tone avoidance Richard Weinberger (2): x86,efi: Check max_size only if it is non-zero. x86,efi: Implement efi_no_storage_paranoia parameter Rob Herring (1): ARM: highbank: fix cache flush ordering for cpu hotplug Russell King (2): ARM: Do 15e0d9e37c (ARM: pm: let platforms select cpu_suspend support) properly Merge branch 'for-rmk/740t' of git://git.kernel.org/.../will/linux into fixes Ryo Tsutsui (2): ASoC: dapm: Remove redundant clear_walk() for supply widgets ASoC: dapm: Only clear paths we've walked Sachin Kamat (5): ASoC: max98090: Remove unneeded version.h header include ASoC: max98090: Convert to devm_regmap_init_i2c() ASoC: max98090: Make struct dev_pm_ops const ASoC: max98090: Fix checkpatch errors related to spacing ASoC: max98090: Remove unnecessary braces Sam Ravnborg (3): sparc: use generic headers bbc_i2c: fix section mismatch warning sparc: use asm-generic version of types.h Scott Wood (4): kvm/powerpc/e500mc: fix tlb invalidation on cpu migration kvm/ppc/e500: h2g_tlb1_rmap: esel 0 is valid kvm/ppc/e500: g2h_tlb1_map: clear old bit before setting new bit kvm/ppc/e500: eliminate tlb_refs Sergey Vlasov (2): x86/Kconfig: Make EFI select UCS2_STRING efi: Export efi_query_variable_store() for efivars.ko Shahed Shaikh (1): qlcnic: Fix typo in logs Shawn Guo (2): ASoC: generic-dmaengine-pcm: use a more common dma name ASoC: generic-dmaengine-pcm: call dma_request_slave_channel() Sritej Velaga (2): qlcnic: fix TSO race condition qlge: Fix receive path to drop error frames Stephane Eranian (1): perf/x86: Fix offcore_rsp valid mask for SNB/IVB Stephen Boyd (1): ARM: 7699/1: sched_clock: Add more notrace to prevent recursion Stephen Warren (10): ASoC: tegra: assume CONFIG_OF in tegra_asoc_utils_init ASoC: tegra_wm8753: minor cleanup ASoC: tegra_wm9712: assume CONFIG_OF ASoC: tegra_alc5632: assume CONFIG_OF, and other cleanup ASoC: tegra trimslice: assume CONFIG_OF, and other cleanup ASoC: tegra_wm8903: assume CONFIG_OF, remove platform data ASoC: tegra: add Tegra114 support to the AHUB driver ASoC: tegra: add Tegra114 support to tegra_asoc_utils.c ASoC: export snd_soc_register_component ASoC: dapm: Implement mixer control sharing Thomas Gleixner (1): kthread: Prevent unpark race which puts threads on the wrong cpu Thomas Graf (1): tcp: Reallocate headroom if it would overflow csum_start Thomas Petazzoni (1): net: mvmdio: add select PHYLIB Tim Gardner (2): ASoC: adau1373: adau1373_hw_params: Silence overflow warning ASoC: wm_hubs: Silence reg_r and reg_l 'may be used uninitialized' warnings Tommi Rantala (1): perf: Treat attr.config as u64 in perf_swevent_init() Vinod Koul (1): ASoC: mid-x86 - add support for meaadata apis Vyacheslav Dubeyko (1): hfsplus: fix potential overflow in hfsplus_file_truncate() Wei Yongjun (5): ASoC: tas5086: use module_i2c_driver to simplify the code ASoC: arizona: remove duplicated include from arizona.c tuntap: fix error return code in tun_set_iff() esp4: fix error return code in esp_output() pkt_sched: fix error return code in fw_change_attrs() Will Deacon (6): ARM: tlbflush: remove ARMv3 support ARM: cache: remove ARMv3 support code ARM: mm: fix numerous hideous errors in proc-arm740.S ARM: mm: remove broken condition check for v4 flushing ARM: modules: don't export cpu_set_pte_ext when !MMU ARM: 7698/1: perf: fix group validation when using enable_on_exec Willy Tarreau (1): net: mvneta: fix improper tx queue usage in mvneta_tx() Xishi Qiu (1): mm/vmscan: fix error return in kswapd_run() Yegor Yefremov (2): ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode ASoC: davinci-mcasp: don't overwrite DIT settings Yinghai Lu (4): x86, kdump: Set crashkernel_low automatically x86, kdump: Retore crashkernel= to allocate under 896M x86, kdump: Change crashkernel_high/low= to crashkernel=,high/low kexec: use Crash kernel for Crash kernel low Zefir Kurtisi (1): ath9k: fix DFS detector called in softirq context Zoltan Puskas (1): ASoC: atmel: Add slave mode support to SSC in DSP Mode A nikolay(a)redhat.com (7): bonding: fix netdev event NULL pointer dereference bonding: IFF_BONDING is not stripped on enslave failure bonding: mc addresses don't get deleted on enslave failure bonding: vlans don't get deleted on enslave failure bonding: primary_slave & curr_active_slave are not cleaned on enslave failure bonding: disable netpoll on enslave failure bonding: in bond_mc_swap() bond's mc addr list is walked without lock stephen hemminger (1): bridge: make user modified path cost sticky Documentation/devicetree/bindings/sound/ak5386.txt | 19 + .../bindings/sound/nvidia,tegra30-ahub.txt | 26 +- .../devicetree/bindings/sound/ti,tas5086.txt | 32 ++ Documentation/kernel-parameters.txt | 29 +- Makefile | 5 +- arch/arm/include/asm/glue-cache.h | 8 - arch/arm/include/asm/hardware/iop3xx.h | 2 +- arch/arm/include/asm/pgtable-3level.h | 2 +- arch/arm/include/asm/tlbflush.h | 11 +- arch/arm/kernel/hw_breakpoint.c | 2 +- arch/arm/kernel/perf_event.c | 5 +- arch/arm/kernel/sched_clock.c | 4 +- arch/arm/kernel/setup.c | 3 - arch/arm/kernel/tcm.c | 1 - arch/arm/kvm/arm.c | 1 + arch/arm/kvm/coproc.c | 4 +- arch/arm/mach-highbank/hotplug.c | 10 +- arch/arm/mach-s3c24xx/dma-s3c2410.c | 2 - arch/arm/mach-s3c24xx/dma-s3c2412.c | 2 - arch/arm/mach-s3c24xx/dma-s3c2440.c | 2 - arch/arm/mach-s3c24xx/dma-s3c2443.c | 2 - arch/arm/mm/Kconfig | 5 +- arch/arm/mm/Makefile | 1 - arch/arm/mm/cache-feroceon-l2.c | 1 + arch/arm/mm/cache-v3.S | 137 ----- arch/arm/mm/cache-v4.S | 2 +- arch/arm/mm/mmu.c | 2 + arch/arm/mm/proc-arm740.S | 30 +- arch/arm/mm/proc-arm920.S | 2 +- arch/arm/mm/proc-arm926.S | 2 +- arch/arm/mm/proc-mohawk.S | 2 +- arch/arm/mm/proc-sa1100.S | 2 +- arch/arm/mm/proc-syms.c | 2 + arch/arm/mm/proc-v6.S | 2 +- arch/arm/mm/proc-xsc3.S | 2 +- arch/arm/mm/proc-xscale.S | 2 +- arch/arm/{kernel => mm}/tcm.h | 0 arch/arm/plat-samsung/devs.c | 6 + arch/avr32/include/asm/io.h | 4 + arch/powerpc/kernel/entry_64.S | 4 +- arch/powerpc/kernel/process.c | 2 + arch/powerpc/kernel/signal_32.c | 2 + arch/powerpc/kernel/signal_64.c | 2 + arch/powerpc/kernel/tm.S | 2 + arch/powerpc/kvm/e500.h | 24 +- arch/powerpc/kvm/e500_mmu_host.c | 84 +-- arch/powerpc/kvm/e500mc.c | 7 +- arch/s390/include/asm/io.h | 4 - arch/s390/include/asm/pgtable.h | 4 + arch/sparc/include/asm/Kbuild | 5 + arch/sparc/include/asm/cputime.h | 6 - arch/sparc/include/asm/emergency-restart.h | 6 - arch/sparc/include/asm/mutex.h | 9 - arch/sparc/include/asm/pgtable_64.h | 1 + arch/sparc/include/asm/serial.h | 6 - arch/sparc/include/asm/smp_32.h | 5 - arch/sparc/include/asm/switch_to_64.h | 3 +- arch/sparc/include/asm/tlbflush_64.h | 37 +- arch/sparc/include/uapi/asm/Kbuild | 1 - arch/sparc/include/uapi/asm/types.h | 17 - arch/sparc/kernel/smp_64.c | 41 +- arch/sparc/lib/bitext.c | 6 +- arch/sparc/mm/iommu.c | 2 +- arch/sparc/mm/srmmu.c | 4 +- arch/sparc/mm/tlb.c | 38 +- arch/sparc/mm/tsb.c | 57 +- arch/sparc/mm/ultra.S | 119 +++- arch/x86/Kconfig | 1 + arch/x86/boot/compressed/eboot.c | 47 ++ arch/x86/include/asm/efi.h | 7 + arch/x86/include/uapi/asm/bootparam.h | 1 + arch/x86/kernel/cpu/mshyperv.c | 18 +- arch/x86/kernel/cpu/perf_event_intel.c | 20 +- arch/x86/kernel/microcode_core_early.c | 38 +- arch/x86/kernel/setup.c | 45 +- arch/x86/platform/efi/efi.c | 168 +++++- block/blk-core.c | 1 + drivers/block/rbd.c | 3 +- drivers/char/hpet.c | 14 +- drivers/dma/at_hdmac.c | 9 +- drivers/extcon/extcon-arizona.c | 107 ++-- drivers/firmware/Kconfig | 1 + drivers/firmware/efivars.c | 98 +--- drivers/idle/intel_idle.c | 1 + drivers/input/tablet/wacom_wac.c | 8 +- drivers/irqchip/irq-gic.c | 3 +- drivers/md/dm.c | 1 + drivers/md/raid5.c | 11 +- drivers/mfd/wm5102-tables.c | 10 + drivers/mtd/mtdchar.c | 59 +- drivers/net/bonding/bond_main.c | 90 ++- drivers/net/can/mcp251x.c | 10 +- drivers/net/can/sja1000/sja1000_of_platform.c | 31 +- drivers/net/ethernet/8390/ax88796.c | 2 +- drivers/net/ethernet/broadcom/bnx2x/bnx2x_cmn.c | 7 +- drivers/net/ethernet/broadcom/bnx2x/bnx2x_main.c | 4 + drivers/net/ethernet/emulex/benet/be_main.c | 5 +- drivers/net/ethernet/freescale/fec.c | 1 + drivers/net/ethernet/intel/igb/igb.h | 8 - drivers/net/ethernet/intel/igb/igb_main.c | 110 +--- drivers/net/ethernet/intel/ixgbe/ixgbe_sriov.c | 6 + drivers/net/ethernet/marvell/Kconfig | 2 +- drivers/net/ethernet/marvell/mvneta.c | 9 +- .../net/ethernet/qlogic/qlcnic/qlcnic_83xx_hw.c | 15 +- drivers/net/ethernet/qlogic/qlcnic/qlcnic_io.c | 3 +- drivers/net/ethernet/qlogic/qlcnic/qlcnic_sysfs.c | 4 +- drivers/net/ethernet/qlogic/qlge/qlge.h | 2 +- drivers/net/ethernet/qlogic/qlge/qlge_ethtool.c | 2 +- drivers/net/ethernet/qlogic/qlge/qlge_main.c | 36 +- drivers/net/ethernet/stmicro/stmmac/mmc_core.c | 1 + drivers/net/ethernet/ti/cpsw.c | 2 +- drivers/net/tun.c | 2 +- drivers/net/usb/cdc_mbim.c | 2 +- drivers/net/usb/qmi_wwan.c | 104 ++++ .../net/wireless/ath/ath9k/ar9580_1p0_initvals.h | 2 +- .../net/wireless/ath/ath9k/dfs_pattern_detector.c | 4 +- drivers/net/wireless/ath/ath9k/dfs_pri_detector.c | 4 +- drivers/net/wireless/ath/ath9k/htc_drv_init.c | 2 +- drivers/net/wireless/b43/phy_n.c | 3 +- .../net/wireless/brcm80211/brcmfmac/wl_cfg80211.c | 7 +- .../net/wireless/brcm80211/brcmsmac/mac80211_if.c | 264 ++++----- drivers/platform/x86/hp-wmi.c | 4 - drivers/platform/x86/thinkpad_acpi.c | 10 - drivers/sbus/char/bbc_i2c.c | 4 +- drivers/ssb/driver_chipcommon_pmu.c | 29 + drivers/video/fbmem.c | 39 +- drivers/video/mmp/core.c | 2 - fs/binfmt_elf.c | 1 + fs/bio.c | 2 - fs/hfsplus/extents.c | 2 +- fs/hugetlbfs/inode.c | 2 +- fs/proc/array.c | 1 + include/linux/blktrace_api.h | 1 - include/linux/efi.h | 9 +- include/linux/kexec.h | 2 + include/linux/mfd/arizona/core.h | 3 + include/linux/mfd/arizona/registers.h | 42 ++ include/linux/mfd/wm8994/pdata.h | 8 + include/linux/mm.h | 2 + include/linux/netfilter/ipset/ip_set_ahash.h | 30 +- include/linux/sched.h | 5 +- include/linux/ssb/ssb_driver_chipcommon.h | 2 + include/linux/swiotlb.h | 1 + include/linux/ucs2_string.h | 14 + include/net/addrconf.h | 1 + include/net/irda/irlmp.h | 3 +- include/net/scm.h | 4 +- include/sound/dmaengine_pcm.h | 5 + include/sound/soc-dai.h | 8 - include/sound/soc.h | 20 + include/sound/tas5086.h | 7 + include/sound/tegra_wm8903.h | 26 - include/trace/events/block.h | 8 +- include/trace/events/sched.h | 2 +- include/uapi/linux/fuse.h | 436 +++++++-------- kernel/events/core.c | 2 +- kernel/hrtimer.c | 3 +- kernel/kexec.c | 118 +++- kernel/kprobes.c | 19 +- kernel/kthread.c | 52 +- kernel/signal.c | 2 +- kernel/smpboot.c | 14 +- kernel/trace/blktrace.c | 26 +- kernel/user_namespace.c | 22 +- lib/Kconfig | 3 + lib/Makefile | 2 + lib/swiotlb.c | 19 +- lib/ucs2_string.c | 51 ++ mm/hugetlb.c | 12 +- mm/memory.c | 47 ++ mm/vmscan.c | 2 +- net/802/mrp.c | 4 + net/batman-adv/main.c | 5 +- net/batman-adv/main.h | 2 +- net/batman-adv/routing.c | 38 +- net/batman-adv/translation-table.c | 2 +- net/batman-adv/vis.c | 4 +- net/bridge/br_if.c | 3 +- net/bridge/br_private.h | 1 + net/bridge/br_stp_if.c | 1 + net/core/dev.c | 3 + net/ipv4/esp4.c | 6 +- net/ipv4/ip_fragment.c | 14 +- net/ipv4/netfilter/ipt_rpfilter.c | 8 +- net/ipv4/syncookies.c | 4 +- net/ipv4/tcp_input.c | 64 +-- net/ipv4/tcp_output.c | 8 +- net/ipv6/addrconf.c | 24 +- net/ipv6/addrconf_core.c | 19 + net/ipv6/netfilter/ip6t_rpfilter.c | 8 +- net/ipv6/reassembly.c | 12 +- net/irda/iriap.c | 3 +- net/irda/irlmp.c | 10 +- net/mac80211/iface.c | 27 +- net/mac80211/mlme.c | 24 +- net/netfilter/ipset/ip_set_bitmap_ipmac.c | 6 +- net/netfilter/ipset/ip_set_hash_ipportnet.c | 18 + net/netfilter/ipset/ip_set_hash_net.c | 22 +- net/netfilter/ipset/ip_set_hash_netiface.c | 22 +- net/netfilter/ipset/ip_set_hash_netport.c | 18 + net/netfilter/ipset/ip_set_list_set.c | 10 +- net/netfilter/nf_conntrack_sip.c | 6 +- net/netfilter/nf_nat_core.c | 40 +- net/openvswitch/datapath.c | 30 +- net/openvswitch/flow.c | 2 +- net/sched/cls_fw.c | 2 +- scripts/checkpatch.pl | 1 + sound/core/pcm_native.c | 12 +- sound/soc/atmel/atmel_ssc_dai.c | 54 +- sound/soc/au1x/ac97c.c | 9 +- sound/soc/au1x/i2sc.c | 9 +- sound/soc/au1x/psc-ac97.c | 9 +- sound/soc/au1x/psc-i2s.c | 9 +- sound/soc/blackfin/bf5xx-ac97.c | 9 +- sound/soc/blackfin/bf5xx-i2s.c | 9 +- sound/soc/blackfin/bf5xx-tdm.c | 9 +- sound/soc/blackfin/bf6xx-i2s.c | 9 +- sound/soc/cirrus/ep93xx-ac97.c | 9 +- sound/soc/cirrus/ep93xx-i2s.c | 9 +- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/adau1373.c | 5 +- sound/soc/codecs/ak4104.c | 55 +- sound/soc/codecs/ak5386.c | 152 +++++ sound/soc/codecs/arizona.c | 530 +++++++++++++++--- sound/soc/codecs/arizona.h | 34 +- sound/soc/codecs/cs4271.c | 166 +++--- sound/soc/codecs/cs42l73.c | 6 +- sound/soc/codecs/max98088.c | 32 +- sound/soc/codecs/max98090.c | 45 +- sound/soc/codecs/si476x.c | 47 +- sound/soc/codecs/tas5086.c | 591 ++++++++++++++++++++ sound/soc/codecs/wm0010.c | 6 +- sound/soc/codecs/wm2000.c | 4 +- sound/soc/codecs/wm2000.h | 2 + sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5102.c | 129 ++--- sound/soc/codecs/wm5102.h | 6 +- sound/soc/codecs/wm5110.c | 54 +- sound/soc/codecs/wm5110.h | 6 +- sound/soc/codecs/wm8903.c | 4 + sound/soc/codecs/wm8960.c | 10 + sound/soc/codecs/wm8994.c | 68 ++- sound/soc/codecs/wm8994.h | 3 +- sound/soc/codecs/wm_adsp.c | 124 +++- sound/soc/codecs/wm_adsp.h | 5 +- sound/soc/codecs/wm_hubs.c | 9 +- sound/soc/davinci/davinci-i2s.c | 15 +- sound/soc/davinci/davinci-mcasp.c | 110 ++-- sound/soc/davinci/davinci-mcasp.h | 2 +- sound/soc/davinci/davinci-pcm.c | 9 +- sound/soc/davinci/davinci-vcif.c | 11 +- sound/soc/dwc/designware_i2s.c | 9 +- sound/soc/fsl/fsl_ssi.c | 11 +- sound/soc/fsl/fsl_ssi.h | 8 + sound/soc/fsl/imx-audmux.c | 3 +- sound/soc/fsl/imx-pcm-fiq.c | 2 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/imx-ssi.c | 13 +- sound/soc/fsl/mpc5200_psc_ac97.c | 8 +- sound/soc/fsl/mpc5200_psc_i2s.c | 9 +- sound/soc/jz4740/jz4740-i2s.c | 9 +- sound/soc/kirkwood/kirkwood-i2s.c | 11 +- sound/soc/mid-x86/sst_platform.c | 20 +- sound/soc/mid-x86/sst_platform.h | 2 + sound/soc/mxs/Kconfig | 2 +- sound/soc/mxs/mxs-pcm.c | 139 +---- sound/soc/mxs/mxs-saif.c | 14 +- sound/soc/nuc900/nuc900-ac97.c | 9 +- sound/soc/omap/omap-dmic.c | 9 +- sound/soc/omap/omap-hdmi.c | 9 +- sound/soc/omap/omap-mcbsp.c | 9 +- sound/soc/omap/omap-mcpdm.c | 9 +- sound/soc/pxa/mmp-sspa.c | 9 +- sound/soc/pxa/pxa-ssp.c | 9 +- sound/soc/pxa/pxa2xx-ac97.c | 11 +- sound/soc/pxa/pxa2xx-i2s.c | 9 +- sound/soc/s6000/s6000-i2s.c | 9 +- sound/soc/samsung/Kconfig | 5 +- sound/soc/samsung/ac97.c | 14 +- sound/soc/samsung/goni_wm8994.c | 11 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 16 +- sound/soc/samsung/idma.c | 11 +- sound/soc/samsung/neo1973_wm8753.c | 3 +- sound/soc/samsung/pcm.c | 11 +- .../include/plat => sound/soc/samsung}/regs-ac97.h | 0 .../include/plat => sound/soc/samsung}/regs-iis.h | 0 sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 9 +- sound/soc/samsung/s3c-i2s-v2.h | 7 +- sound/soc/samsung/s3c2412-i2s.c | 12 +- sound/soc/samsung/s3c24xx-i2s.c | 13 +- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- sound/soc/samsung/spdif.c | 11 +- sound/soc/sh/fsi.c | 14 +- sound/soc/sh/hac.c | 10 +- sound/soc/sh/migor.c | 2 +- sound/soc/sh/siu_dai.c | 11 +- sound/soc/sh/ssi.c | 10 +- sound/soc/soc-compress.c | 27 + sound/soc/soc-core.c | 99 +++- sound/soc/soc-dapm.c | 319 ++++++----- sound/soc/soc-generic-dmaengine-pcm.c | 44 +- sound/soc/soc-utils.c | 25 + sound/soc/spear/spdif_in.c | 9 +- sound/soc/spear/spdif_out.c | 9 +- sound/soc/spear/spear_pcm.c | 4 +- sound/soc/tegra/tegra20_ac97.c | 15 +- sound/soc/tegra/tegra20_i2s.c | 15 +- sound/soc/tegra/tegra20_spdif.c | 15 +- sound/soc/tegra/tegra30_ahub.c | 64 ++- sound/soc/tegra/tegra30_ahub.h | 16 + sound/soc/tegra/tegra30_i2s.c | 15 +- sound/soc/tegra/tegra_alc5632.c | 15 +- sound/soc/tegra/tegra_asoc_utils.c | 34 +- sound/soc/tegra/tegra_asoc_utils.h | 1 + sound/soc/tegra/tegra_wm8753.c | 15 +- sound/soc/tegra/tegra_wm8903.c | 179 +++--- sound/soc/tegra/tegra_wm9712.c | 7 +- sound/soc/tegra/trimslice.c | 56 +- sound/soc/txx9/txx9aclc-ac97.c | 9 +- sound/soc/ux500/ux500_msp_dai.c | 13 +- sound/soc/ux500/ux500_msp_dai.h | 5 - sound/soc/ux500/ux500_msp_i2s.h | 1 + tools/power/x86/turbostat/turbostat.c | 5 +- 326 files changed, 5225 insertions(+), 2620 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak5386.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,tas5086.txt delete mode 100644 arch/arm/mm/cache-v3.S rename arch/arm/{kernel => mm}/tcm.h (100%) delete mode 100644 arch/sparc/include/asm/cputime.h delete mode 100644 arch/sparc/include/asm/emergency-restart.h delete mode 100644 arch/sparc/include/asm/mutex.h delete mode 100644 arch/sparc/include/asm/serial.h delete mode 100644 arch/sparc/include/uapi/asm/types.h create mode 100644 include/linux/ucs2_string.h create mode 100644 include/sound/tas5086.h delete mode 100644 include/sound/tegra_wm8903.h create mode 100644 lib/ucs2_string.c create mode 100644 sound/soc/codecs/ak5386.c create mode 100644 sound/soc/codecs/tas5086.c rename {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-ac97.h (100%) rename {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-iis.h (100%)

2 1

Re: [alsa-devel] [PATCH] drm/i915: Add private api for power well usage
by Wang, Xingchao 25 Apr '13

25 Apr '13

Hi Daniel/Paulo, Any comments on this? Add Jesse and Rafael in loop. thanks --xingchao > -----Original Message----- > From: Wang, Xingchao > Sent: Wednesday, April 24, 2013 3:29 PM > To: daniel.vetter(a)ffwll.ch; Zanoni, Paulo R; 'Takashi Iwai' > Cc: ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; Li, Jocelyn; > intel-gfx(a)lists.freedesktop.org; alsa-devel(a)alsa-project.org; 'Wang Xingchao' > Subject: RE: [PATCH] drm/i915: Add private api for power well usage > > Hi Guys, > > Sorry I should Add Takashi and alsa maillist in loop. > > I tested this patch against Daniel's "drm-intel-next-queued" Branch, on > Haswell Mobile C3 stepping machines. > > Test steps: > 1. only connect one monitor through HDMI cable 2. echo 1 > > /sys/module/i915/parameters/disable_power_well ==> I think this could be set > as 1 by default if applied my patch. > 3. plug out HDMI cable > 4. cat /proc/asound/card0/codec#0 > > At step 3, i915 will try to disable power well but rejected as display audio is > using power well. > Without the patch, display audio would crash at step 4. > > I did not list suspend/resume test here, as during suspend sequence, i915 will > _ENABLE_ power well, that's a BUG? > > For me above test case is the only one to reproduce the issue, if you have more > ideas and need more test, please let me know. > > I'm afraid there's some risk because the dependency between alsa driver and > drm driver. For example, if i915 module was not loaded, display audio Will call > request_power_well() and meet symbol error. Even if i915 module loaded after > snd-hda-intel module, there's such risk too. Any comment on this? > > Another risk is the pointer "power_well_device", it should be set NULL when > i915 module unloaded, I added the caller but have no idea where to call it, if > anyone has suggestion, that would be fine. > > thanks > --xingchao > > > > -----Original Message----- > > From: Wang Xingchao [mailto:xingchao.wang@linux.intel.com] > > Sent: Thursday, April 24, 2014 10:20 PM > > To: daniel.vetter(a)ffwll.ch; Zanoni, Paulo R > > Cc: ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; > > Li, Jocelyn; intel-gfx(a)lists.freedesktop.org; Wang, Xingchao; Wang > > Xingchao > > Subject: [PATCH] drm/i915: Add private api for power well usage > > > > Display audio depends on power well in graphic side, it should request > > power well before use it and release power well after use. > > I915 will not shutdown power well if it detects audio is using. > > This patch protects display audio crash for Intel Haswell mobile > > C3 stepping board. > > > > Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> > > --- > > drivers/gpu/drm/i915/intel_pm.c | 73 > > +++++++++++++++++++++++++++++++++++---- > > include/drm/audio_drm.h | 39 +++++++++++++++++++++ > > sound/pci/hda/hda_intel.c | 8 +++++ > > 3 files changed, 113 insertions(+), 7 deletions(-) create mode > > 100644 include/drm/audio_drm.h > > > > diff --git a/drivers/gpu/drm/i915/intel_pm.c > > b/drivers/gpu/drm/i915/intel_pm.c index 2557926..9983f56 100644 > > --- a/drivers/gpu/drm/i915/intel_pm.c > > +++ b/drivers/gpu/drm/i915/intel_pm.c > > @@ -4298,18 +4298,12 @@ bool intel_using_power_well(struct drm_device > > *dev) > > return true; > > } > > > > -void intel_set_power_well(struct drm_device *dev, bool enable) > > +void set_power_well(struct drm_device *dev, bool enable) > > { > > struct drm_i915_private *dev_priv = dev->dev_private; > > bool is_enabled, enable_requested; > > uint32_t tmp; > > > > - if (!HAS_POWER_WELL(dev)) > > - return; > > - > > - if (!i915_disable_power_well && !enable) > > - return; > > - > > tmp = I915_READ(HSW_PWR_WELL_DRIVER); > > is_enabled = tmp & HSW_PWR_WELL_STATE; > > enable_requested = tmp & HSW_PWR_WELL_ENABLE; @@ -4332,6 > > +4326,71 @@ void intel_set_power_well(struct drm_device *dev, bool > > +enable) > > } > > } > > > > +/* Global drm_device for display audio drvier usage */ static struct > > +drm_device *power_well_device; static bool audio_power_well_using, > > +i915_power_well_using; > > +/* Lock protecting power well setting */ > > +DEFINE_SPINLOCK(powerwell_lock); > > + > > +void request_power_well(void) > > +{ > > + DRM_DEBUG_KMS("Display audio requesting power well\n"); > > + spin_lock_irq(&powerwell_lock); > > + if (!power_well_device) > > + goto out; > > + if (i915_power_well_using == false) > > + set_power_well(power_well_device, true); > > + audio_power_well_using = true; > > +out: > > + spin_unlock_irq(&powerwell_lock); > > + return; > > +} > > +EXPORT_SYMBOL_GPL(request_power_well); > > + > > +void release_power_well(void) > > +{ > > + DRM_DEBUG_KMS("Display audio releasing power well\n"); > > + spin_lock_irq(&powerwell_lock); > > + if (!power_well_device) > > + goto out; > > + if (i915_power_well_using == false) > > + set_power_well(power_well_device, false); > > + audio_power_well_using = false; > > +out: > > + spin_unlock_irq(&powerwell_lock); > > + return; > > +} > > +EXPORT_SYMBOL_GPL(release_power_well); > > + > > +/* TODO: Call this when i915 module unload */ void > > +remove_power_well(void) { > > + spin_lock_irq(&powerwell_lock); > > + audio_power_well_using = false; > > + i915_power_well_using = false; > > + power_well_device = NULL; > > + spin_unlock_irq(&powerwell_lock); > > +} > > + > > +void intel_set_power_well(struct drm_device *dev, bool enable) { > > + if (!HAS_POWER_WELL(dev)) > > + return; > > + > > + spin_lock_irq(&powerwell_lock); > > + power_well_device = dev; > > + i915_power_well_using = enable; > > + if (!enable && audio_power_well_using) > > + goto out; > > + > > + if (!i915_disable_power_well && !enable) > > + goto out; > > + set_power_well(dev, enable); > > +out: > > + spin_unlock_irq(&powerwell_lock); > > + return; > > +} > > + > > /* > > * Starting with Haswell, we have a "Power Down Well" that can be turned > off > > * when not needed anymore. We have 4 registers that can request the > > power well diff --git a/include/drm/audio_drm.h > > b/include/drm/audio_drm.h new file mode 100644 index 0000000..4412b36 > > --- /dev/null > > +++ b/include/drm/audio_drm.h > > @@ -0,0 +1,39 @@ > > > +/************************************************************** > > ******** > > +**** > > + * > > + * Copyright 2013 Intel Inc. > > + * All Rights Reserved. > > + * > > + * Permission is hereby granted, free of charge, to any person > > +obtaining a > > + * copy of this software and associated documentation files (the > > + * "Software"), to deal in the Software without restriction, > > +including > > + * without limitation the rights to use, copy, modify, merge, > > +publish, > > + * distribute, sub license, and/or sell copies of the Software, and > > +to > > + * permit persons to whom the Software is furnished to do so, subject > > +to > > + * the following conditions: > > + * > > + * The above copyright notice and this permission notice (including > > +the > > + * next paragraph) shall be included in all copies or substantial > > +portions > > + * of the Software. > > + * > > + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY > KIND, > > +EXPRESS OR > > + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF > > +MERCHANTABILITY, > > + * FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT. IN NO > > EVENT > > +SHALL > > + * THE COPYRIGHT HOLDERS, AUTHORS AND/OR ITS SUPPLIERS BE LIABLE > > FOR > > +ANY CLAIM, > > + * DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, > > TORT > > +OR > > + * OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE > > SOFTWARE > > +OR THE > > + * USE OR OTHER DEALINGS IN THE SOFTWARE. > > + * > > + * > > + > > > +*************************************************************** > > ******** > > +***/ > > +/* > > + * Authors: > > + * Wang Xingchao <xingchao.wang(a)intel.com> */ > > + > > +#ifndef _AUDIO_DRM_H_ > > +#define _AUDIO_DRM_H_ > > + > > +extern void request_power_well(void); extern void > > +release_power_well(void); > > + > > +#endif /* _AUDIO_DRM_H_ */ > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > > index 418bfc0..59a85de 100644 > > --- a/sound/pci/hda/hda_intel.c > > +++ b/sound/pci/hda/hda_intel.c > > @@ -50,6 +50,7 @@ > > #include <linux/clocksource.h> > > #include <linux/time.h> > > #include <linux/completion.h> > > +#include <drm/audio_drm.h> > > > > #ifdef CONFIG_X86 > > /* for snoop control */ > > @@ -2869,6 +2870,7 @@ static int azx_suspend(struct device *dev) > > pci_disable_device(pci); > > pci_save_state(pci); > > pci_set_power_state(pci, PCI_D3hot); > > + release_power_well(); > > return 0; > > } > > > > @@ -2881,6 +2883,7 @@ static int azx_resume(struct device *dev) > > if (chip->disabled) > > return 0; > > > > + request_power_well(); > > pci_set_power_state(pci, PCI_D0); > > pci_restore_state(pci); > > if (pci_enable_device(pci) < 0) { > > @@ -2913,6 +2916,7 @@ static int azx_runtime_suspend(struct device > > *dev) > > > > azx_stop_chip(chip); > > azx_clear_irq_pending(chip); > > + release_power_well(); > > return 0; > > } > > > > @@ -2921,6 +2925,7 @@ static int azx_runtime_resume(struct device *dev) > > struct snd_card *card = dev_get_drvdata(dev); > > struct azx *chip = card->private_data; > > > > + request_power_well(); > > azx_init_pci(chip); > > azx_init_chip(chip, 1); > > return 0; > > @@ -3671,6 +3676,8 @@ static int azx_probe(struct pci_dev *pci, > > return -ENOENT; > > } > > > > + request_power_well(); > > + > > err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); > > if (err < 0) { > > snd_printk(KERN_ERR "hda-intel: Error creating card!\n"); @@ > > -3806,6 +3813,7 @@ static void azx_remove(struct pci_dev *pci) > > if (card) > > snd_card_free(card); > > pci_set_drvdata(pci, NULL); > > + release_power_well(); > > } > > > > /* PCI IDs */ > > -- > > 1.7.9.5

2 2

[alsa-devel] code improvements in /sound/usb/mixer.c
by Daniel Schürmann 25 Apr '13

25 Apr '13

Hi Takashi, Hi alsa-devel. in my last mail I had supposed scaling issues with mixer.c, but finally it turns out that was just a debugging problem of myself. I was struggled how types are handled in the code. I have some ideas for improvement but I am not sure if it its worth the work since nothing is broken. 1. get rid of convert_bytes_value(cval, value_set): This function return any type in a integer container, which is confusing. 2. use type casts in convert_signed_value(): gcc knows perfecly how to convert types. 3. merge convert_signed_value() and snd_usb_combine_bytes() to get rid of the intermediate integer variant. --- I am currently working on a patch that introduces a get_ctl_value_range_v2. I want to get rid of the three single queries for min max and rate. Also here, nothing broken. Is it worth to do the work? Kind regards Daniel

1 0

[alsa-devel] [PATCH v2] ALSA: snd-usb-audio: fixed mixer range reading during probing, when compiled with CONFIG_PM defined
by Daniel Schürmann 25 Apr '13

25 Apr '13

During probing, snd_usb_autoresume() fails with -ENODEV. This brakes get_ctl_value() in mixer.c when reading the range values. In the new version snd_usb_autoresume() returns 0 in this case which brakes not the following query. The following snd_usb_suspend() fails silently during probing anyway. The patch also makes the error messages more significant. The original error messages where identified as misleading during debug. Signed-off-by: Daniel Schürmann <daschuer(a)mixxx.org> --- sound/usb/card.c | 7 +++++-- sound/usb/mixer.c | 11 +++++++---- 2 files changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 5254b18..2f55d00 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -629,9 +629,12 @@ static void usb_audio_disconnect(struct usb_interface *intf) int snd_usb_autoresume(struct snd_usb_audio *chip) { int err = -ENODEV; - + + if (chip->probing) { + return 0; + } down_read(&chip->shutdown_rwsem); - if (!chip->shutdown && !chip->probing) + if (!chip->shutdown) err = usb_autopm_get_interface(chip->pm_intf); up_read(&chip->shutdown_rwsem); diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ca4739c..5d4f3c6 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -333,9 +333,12 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v memset(buf, 0, sizeof(buf)); - ret = snd_usb_autoresume(chip) ? -EIO : 0; - if (ret) + ret = snd_usb_autoresume(chip); + if (ret) { + snd_printk(KERN_ERR "snd_usb_autoresume() failed with %d\n", ret); + ret = -EIO; goto error; + } down_read(&chip->shutdown_rwsem); if (chip->shutdown) @@ -351,8 +354,8 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v if (ret < 0) { error: - snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", - request, validx, idx, cval->val_type); + snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d, ret = %d\n", + bRequest, validx, idx, cval->val_type, ret); return ret; } -- 1.7.9.5

2 2

Re: [alsa-devel] [PATCH v2 0/3] USB quirks for Yamaha THR series amps
by Takashi Iwai 25 Apr '13

25 Apr '13

At Wed, 24 Apr 2013 21:19:18 -0400, Trulan Martin wrote: > > The following USB quirks enable MIDI control of the Yamaha THR10, THR5A, and THR10C amps. Each device has its own USB ID. Thanks, applied now. Takashi > > Please cc me as I am not subscribed to the list. > > Thanks, > Trulan >

1 0

[alsa-devel] Asynchronous audio USB chips: choppy playback since 3.8-rc7
by David Henningsson 24 Apr '13

24 Apr '13

Hi ALSA developers, Just to get your attention here on what seems to be an USB audio regression. The bug is described in detail here: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1136110?comments=all Quoting the bug: " This bug seems to affect only a certain kind of hardware, which is called "Asynchronous USB Digital Audio Codec (DAC)". It's said that such a DAC hosts the clock itself (USB Device Host). An ordinary DAC, so called "Synchronous USB DAC", uses the clock hosted by the mother board, which is not affected by this bug. When this bug affects an asynchronous USB DAC, the audio played by the DAC is constantly interrupted. The playback itself does not stop, but the output becomes discontinous, filling with constant crackling noises, destroying everything the DAC plays. " According to the bug reporter, which seems to have done quite a bit of research, this started between 3.8-rc6 and 3.8-rc7 as well as stable kernels and the bug also lists a few commits which could be the cause, none under sound/usb though. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

5 11

[alsa-devel] [PATCH] ARM: allmodconfig regressions in linux-next
by Arnd Bergmann 24 Apr '13

24 Apr '13

Hi everyone, These are a couple of simple patches to fix bugs that break allyesconfig and that are only present in linux-next at the moment. Please apply to the appropriate trees or provide feedback! Arnd Bergmann (5): mfd: ab8500: remove broken ab8500_debug_register_interrupt logic usb: phy: phy core cannot yet be a module ASoC: don't call of_dma_request_slave_channel directly hwrng: bcm2835: fix MODULE_LICENSE tag clocksource: kona: adapt to CLOCKSOURCE_OF_DECLARE change drivers/char/hw_random/bcm2835-rng.c | 2 +- drivers/clocksource/bcm_kona_timer.c | 15 ++++----------- drivers/mfd/ab8500-debugfs.c | 10 ---------- drivers/usb/phy/Kconfig | 2 +- sound/soc/soc-generic-dmaengine-pcm.c | 9 +++++---- 5 files changed, 11 insertions(+), 27 deletions(-) Cc: Christian Daudt <csd(a)broadcom.com> Cc: Dom Cobley <popcornmix(a)gmail.com> Cc: Felipe Balbi <balbi(a)ti.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Herbert Xu <herbert(a)gondor.apana.org.au> Cc: John Stultz <john.stultz(a)linaro.org> Cc: Jonas ABERG <jonas.aberg(a)stericsson.com> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Lee Jones <lee.jones(a)linaro.org> Cc: Linus Walleij <linus.walleij(a)stericsson.com> Cc: Lubomir Rintel <lkundrak(a)v3.sk> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Matt Mackall <mpm(a)selenic.com> Cc: Mattias WALLIN <mattias.wallin(a)stericsson.com> Cc: Per FORLIN <per.forlin(a)stericsson.com> Cc: Samuel Ortiz <sameo(a)linux.intel.com> Cc: Shawn Guo <shawn.guo(a)linaro.org> Cc: Stephen Warren <swarren(a)wwwdotorg.org> Cc: alsa-devel(a)alsa-project.org Cc: linux-rpi-kernel(a)lists.infradead.org Cc: linux-usb(a)vger.kernel.org Cc: srinidhi kasagar <srinidhi.kasagar(a)stericsson.com> -- 1.8.1.2

3 4

[alsa-devel] [PATCH] fixed mixer range reading during probing when compiled with CONFIG_PM defined
by Daniel Schürmann 24 Apr '13

24 Apr '13

During probing, snd_usb_autoresume() failes with -ENODEV. This brakes get_ctl_value() in mixer.c when rading the range values. The new code does not try to call snd_usb_autoresume() in this case. The following snd_usb_suspend() fails silently during probing anyway. The patch also makes the error messages more significant. The original error messages where identified as missleading during debug. Signed-off-by: Daniel Schürmann <daschuer(a)mixxx.org> --- sound/usb/mixer.c | 23 +++++++++++++++-------- 1 file changed, 15 insertions(+), 8 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ca4739c..c2daa58 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -289,9 +289,11 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v int timeout = 10; int idx = 0, err; - err = snd_usb_autoresume(cval->mixer->chip); - if (err < 0) - return -EIO; + if (!chip->probing) { + err = snd_usb_autoresume(cval->mixer->chip); + if (err < 0) + return -EIO; + } down_read(&chip->shutdown_rwsem); while (timeout-- > 0) { if (chip->shutdown) @@ -333,9 +335,14 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v memset(buf, 0, sizeof(buf)); - ret = snd_usb_autoresume(chip) ? -EIO : 0; - if (ret) - goto error; + if (!chip->probing) { + ret = snd_usb_autoresume(chip); + if (ret) { + snd_printk(KERN_ERR "snd_usb_autoresume() failed with %d\n", ret); + ret = -EIO; + goto error; + } + } down_read(&chip->shutdown_rwsem); if (chip->shutdown) @@ -351,8 +358,8 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v if (ret < 0) { error: - snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", - request, validx, idx, cval->val_type); + snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d, ret = %d\n", + bRequest, validx, idx, cval->val_type, ret); return ret; } -- 1.7.9.5

2 3

Re: [alsa-devel] [PATCH] drm/i915: Add private api for power well usage
by Wang, Xingchao 24 Apr '13

24 Apr '13

Hi Guys, Sorry I should Add Takashi and alsa maillist in loop. I tested this patch against Daniel's "drm-intel-next-queued" Branch, on Haswell Mobile C3 stepping machines. Test steps: 1. only connect one monitor through HDMI cable 2. echo 1 > /sys/module/i915/parameters/disable_power_well ==> I think this could be set as 1 by default if applied my patch. 3. plug out HDMI cable 4. cat /proc/asound/card0/codec#0 At step 3, i915 will try to disable power well but rejected as display audio is using power well. Without the patch, display audio would crash at step 4. I did not list suspend/resume test here, as during suspend sequence, i915 will _ENABLE_ power well, that's a BUG? For me above test case is the only one to reproduce the issue, if you have more ideas and need more test, please let me know. I'm afraid there's some risk because the dependency between alsa driver and drm driver. For example, if i915 module was not loaded, display audio Will call request_power_well() and meet symbol error. Even if i915 module loaded after snd-hda-intel module, there's such risk too. Any comment on this? Another risk is the pointer "power_well_device", it should be set NULL when i915 module unloaded, I added the caller but have no idea where to call it, if anyone has suggestion, that would be fine. thanks --xingchao > -----Original Message----- > From: Wang Xingchao [mailto:xingchao.wang@linux.intel.com] > Sent: Thursday, April 24, 2014 10:20 PM > To: daniel.vetter(a)ffwll.ch; Zanoni, Paulo R > Cc: ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; Li, Jocelyn; > intel-gfx(a)lists.freedesktop.org; Wang, Xingchao; Wang Xingchao > Subject: [PATCH] drm/i915: Add private api for power well usage > > Display audio depends on power well in graphic side, it should request power > well before use it and release power well after use. > I915 will not shutdown power well if it detects audio is using. > This patch protects display audio crash for Intel Haswell mobile > C3 stepping board. > > Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> > --- > drivers/gpu/drm/i915/intel_pm.c | 73 > +++++++++++++++++++++++++++++++++++---- > include/drm/audio_drm.h | 39 +++++++++++++++++++++ > sound/pci/hda/hda_intel.c | 8 +++++ > 3 files changed, 113 insertions(+), 7 deletions(-) create mode 100644 > include/drm/audio_drm.h > > diff --git a/drivers/gpu/drm/i915/intel_pm.c b/drivers/gpu/drm/i915/intel_pm.c > index 2557926..9983f56 100644 > --- a/drivers/gpu/drm/i915/intel_pm.c > +++ b/drivers/gpu/drm/i915/intel_pm.c > @@ -4298,18 +4298,12 @@ bool intel_using_power_well(struct drm_device > *dev) > return true; > } > > -void intel_set_power_well(struct drm_device *dev, bool enable) > +void set_power_well(struct drm_device *dev, bool enable) > { > struct drm_i915_private *dev_priv = dev->dev_private; > bool is_enabled, enable_requested; > uint32_t tmp; > > - if (!HAS_POWER_WELL(dev)) > - return; > - > - if (!i915_disable_power_well && !enable) > - return; > - > tmp = I915_READ(HSW_PWR_WELL_DRIVER); > is_enabled = tmp & HSW_PWR_WELL_STATE; > enable_requested = tmp & HSW_PWR_WELL_ENABLE; @@ -4332,6 > +4326,71 @@ void intel_set_power_well(struct drm_device *dev, bool enable) > } > } > > +/* Global drm_device for display audio drvier usage */ static struct > +drm_device *power_well_device; static bool audio_power_well_using, > +i915_power_well_using; > +/* Lock protecting power well setting */ > +DEFINE_SPINLOCK(powerwell_lock); > + > +void request_power_well(void) > +{ > + DRM_DEBUG_KMS("Display audio requesting power well\n"); > + spin_lock_irq(&powerwell_lock); > + if (!power_well_device) > + goto out; > + if (i915_power_well_using == false) > + set_power_well(power_well_device, true); > + audio_power_well_using = true; > +out: > + spin_unlock_irq(&powerwell_lock); > + return; > +} > +EXPORT_SYMBOL_GPL(request_power_well); > + > +void release_power_well(void) > +{ > + DRM_DEBUG_KMS("Display audio releasing power well\n"); > + spin_lock_irq(&powerwell_lock); > + if (!power_well_device) > + goto out; > + if (i915_power_well_using == false) > + set_power_well(power_well_device, false); > + audio_power_well_using = false; > +out: > + spin_unlock_irq(&powerwell_lock); > + return; > +} > +EXPORT_SYMBOL_GPL(release_power_well); > + > +/* TODO: Call this when i915 module unload */ void > +remove_power_well(void) { > + spin_lock_irq(&powerwell_lock); > + audio_power_well_using = false; > + i915_power_well_using = false; > + power_well_device = NULL; > + spin_unlock_irq(&powerwell_lock); > +} > + > +void intel_set_power_well(struct drm_device *dev, bool enable) { > + if (!HAS_POWER_WELL(dev)) > + return; > + > + spin_lock_irq(&powerwell_lock); > + power_well_device = dev; > + i915_power_well_using = enable; > + if (!enable && audio_power_well_using) > + goto out; > + > + if (!i915_disable_power_well && !enable) > + goto out; > + set_power_well(dev, enable); > +out: > + spin_unlock_irq(&powerwell_lock); > + return; > +} > + > /* > * Starting with Haswell, we have a "Power Down Well" that can be turned off > * when not needed anymore. We have 4 registers that can request the > power well diff --git a/include/drm/audio_drm.h b/include/drm/audio_drm.h > new file mode 100644 index 0000000..4412b36 > --- /dev/null > +++ b/include/drm/audio_drm.h > @@ -0,0 +1,39 @@ > +/************************************************************** > ******** > +**** > + * > + * Copyright 2013 Intel Inc. > + * All Rights Reserved. > + * > + * Permission is hereby granted, free of charge, to any person > +obtaining a > + * copy of this software and associated documentation files (the > + * "Software"), to deal in the Software without restriction, including > + * without limitation the rights to use, copy, modify, merge, publish, > + * distribute, sub license, and/or sell copies of the Software, and to > + * permit persons to whom the Software is furnished to do so, subject > +to > + * the following conditions: > + * > + * The above copyright notice and this permission notice (including the > + * next paragraph) shall be included in all copies or substantial > +portions > + * of the Software. > + * > + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, > +EXPRESS OR > + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF > +MERCHANTABILITY, > + * FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT. IN NO > EVENT > +SHALL > + * THE COPYRIGHT HOLDERS, AUTHORS AND/OR ITS SUPPLIERS BE LIABLE > FOR > +ANY CLAIM, > + * DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, > TORT > +OR > + * OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE > SOFTWARE > +OR THE > + * USE OR OTHER DEALINGS IN THE SOFTWARE. > + * > + * > + > +*************************************************************** > ******** > +***/ > +/* > + * Authors: > + * Wang Xingchao <xingchao.wang(a)intel.com> */ > + > +#ifndef _AUDIO_DRM_H_ > +#define _AUDIO_DRM_H_ > + > +extern void request_power_well(void); > +extern void release_power_well(void); > + > +#endif /* _AUDIO_DRM_H_ */ > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index > 418bfc0..59a85de 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -50,6 +50,7 @@ > #include <linux/clocksource.h> > #include <linux/time.h> > #include <linux/completion.h> > +#include <drm/audio_drm.h> > > #ifdef CONFIG_X86 > /* for snoop control */ > @@ -2869,6 +2870,7 @@ static int azx_suspend(struct device *dev) > pci_disable_device(pci); > pci_save_state(pci); > pci_set_power_state(pci, PCI_D3hot); > + release_power_well(); > return 0; > } > > @@ -2881,6 +2883,7 @@ static int azx_resume(struct device *dev) > if (chip->disabled) > return 0; > > + request_power_well(); > pci_set_power_state(pci, PCI_D0); > pci_restore_state(pci); > if (pci_enable_device(pci) < 0) { > @@ -2913,6 +2916,7 @@ static int azx_runtime_suspend(struct device *dev) > > azx_stop_chip(chip); > azx_clear_irq_pending(chip); > + release_power_well(); > return 0; > } > > @@ -2921,6 +2925,7 @@ static int azx_runtime_resume(struct device *dev) > struct snd_card *card = dev_get_drvdata(dev); > struct azx *chip = card->private_data; > > + request_power_well(); > azx_init_pci(chip); > azx_init_chip(chip, 1); > return 0; > @@ -3671,6 +3676,8 @@ static int azx_probe(struct pci_dev *pci, > return -ENOENT; > } > > + request_power_well(); > + > err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); > if (err < 0) { > snd_printk(KERN_ERR "hda-intel: Error creating card!\n"); @@ > -3806,6 +3813,7 @@ static void azx_remove(struct pci_dev *pci) > if (card) > snd_card_free(card); > pci_set_drvdata(pci, NULL); > + release_power_well(); > } > > /* PCI IDs */ > -- > 1.7.9.5

1 0

[alsa-devel] [PATCH 0/3] Yamaha THR10 and related USB quirks
by Trulan Martin 24 Apr '13

24 Apr '13

The following patches add USB quirks for several Yamaha amps. Please CC me as I am not subscribed to the mailing list.

2 4

[alsa-devel] Per-host .asoundrc on shared $HOME
by Mark Hills 23 Apr '13

23 Apr '13

There is a very old thread [1] on alsa-user on this topic which seems to go through some yucky solutions. I have the same use case, and post some findings here. Since pre-2003 alsa-lib calls wordexp(3) on the filenames. I am using the patch below in /usr/share/alsa/alsa.conf It follows a precedent set by other files such as .Xdefaults-hostname. I'd submit a patch, except there seems to be no guarantee that HOSTNAME is in the environment (on Slackware it is explicitly set in /etc/profile.) Without it I can only see a bash local variable. >From the old thread I imagine this function (or at least this mail) would be useful to others. Would it be better to patch the C code to provide a token (eg. '\h') that is substituted by the hostname? Or is it enough to assume that HOSTNAME would be set? I didn't consider short vs. long hostnames, I suppose $HOSTNAME gives the user/admin some control over this. This: ~/.asoundrc-$(hostname -s) may work, but I don't see that forking a process is an option. [1] http://thread.gmane.org/gmane.linux.alsa.user/27532/focus=27542 -- Mark --- alsa.conf.orig 2013-04-23 21:22:45.105454493 +0100 +++ alsa.conf 2013-04-23 21:16:41.707053308 +0100 @@ -17,6 +17,7 @@ } "/etc/asound.conf" "~/.asoundrc" + "~/.asoundrc-$HOSTNAME" ] errors false }

1 0

[alsa-devel] [PATCH 00/21] ARM build fixes for 3.10
by Arnd Bergmann 23 Apr '13

23 Apr '13

Hi everyone, This is a set of bug fixes for build errors and warnings I am seeing in the various defconfigs and/or allmodconfig builds on linux-next. The majority of these showed up because we now include a lot more platforms in allmodconfig than we used to. Please apply to appropriate maintainer trees or provide feedback, I can take care of the rest sending those through arm-soc that I get no feedback on. Arnd Arnd Bergmann (21): Turn off -Wmaybe-uninitialized when building with -Os ARM: compressed/head.S: work around new binutils warning ARM: use optimized do_div only for EABI ARM: kvm: define KVM_ARM_MAX_VCPUS unconditionally ARM: OMAP2+: add dependencies on ARCH_MULTI_V6/V7 ARM: tegra: call cpu_do_idle from C code ARM: tegra: unify tegra_idle_device definitions ASoC: ux500: forward declare msp_i2s_platform_data spi/sirf: fix MODULE_DEVICE_TABLE ssb: fix alignment of struct bcma_device_id drm/tilcdc: use only a single module device table mmc: sdhci-tegra: fix MODULE_DEVICE_TABLE misc: mark spear13xx-pcie-gadget as broken staging: nvec: allow modular build staging: comedi/dawboard2000: use mdelay for large delays staging/zsmalloc: don't use pgtable-mapping from modules tty/serial/sirf: fix MODULE_DEVICE_TABLE usb: phy: tegra: don't call into tegra-ehci directly usb: phy: don't mark usb_bind_phy as __init MFD: ab8500: export ab8500_gpadc_sw_hw_convert properly mtd: omap2: allow bulding as a module Makefile | 2 +- arch/arm/boot/compressed/Makefile | 2 +- arch/arm/boot/compressed/head.S | 12 ++++++++++++ arch/arm/include/asm/div64.h | 2 +- arch/arm/kvm/Kconfig | 6 +++--- arch/arm/mach-omap2/Kconfig | 5 +++++ arch/arm/mach-tegra/cpuidle-tegra114.c | 5 ++--- arch/arm/mach-tegra/cpuidle-tegra20.c | 4 +--- arch/arm/mach-tegra/cpuidle-tegra30.c | 4 +--- arch/arm/mach-tegra/cpuidle.c | 2 ++ arch/arm/mach-tegra/cpuidle.h | 4 ++++ arch/arm/mach-tegra/pm.c | 5 +++++ arch/arm/mach-tegra/sleep-tegra20.S | 2 +- drivers/gpu/drm/tilcdc/tilcdc_panel.c | 1 - drivers/gpu/drm/tilcdc/tilcdc_slave.c | 1 - drivers/gpu/drm/tilcdc/tilcdc_tfp410.c | 1 - drivers/mfd/ab8500-gpadc.c | 2 +- drivers/misc/Kconfig | 2 +- drivers/mmc/host/sdhci-tegra.c | 2 +- drivers/mtd/nand/Kconfig | 2 +- drivers/spi/spi-sirf.c | 2 +- drivers/staging/comedi/drivers/daqboard2000.c | 14 +++++++------- drivers/staging/nvec/Kconfig | 10 +++++----- drivers/staging/zsmalloc/zsmalloc-main.c | 2 +- drivers/tty/serial/sirfsoc_uart.c | 2 +- drivers/usb/host/ehci-tegra.c | 10 +++++----- drivers/usb/otg/otg.c | 2 +- drivers/usb/phy/tegra_usb_phy.c | 13 +++++++++---- include/linux/mod_devicetable.h | 5 +++-- include/linux/usb/tegra_usb_phy.h | 10 +++++----- sound/soc/ux500/ux500_msp_i2s.h | 1 + 31 files changed, 82 insertions(+), 55 deletions(-) -- 1.8.1.2 Cc: Alan Stern <stern(a)rowland.harvard.edu> Cc: Artem Bityutskiy <artem.bityutskiy(a)linux.intel.com> Cc: Chris Ball <cjb(a)laptop.org> Cc: Dave Airlie <airlied(a)linux.ie> Cc: David Woodhouse <dwmw2(a)infradead.org> Cc: Felipe Balbi <balbi(a)ti.com> Cc: Grant Likely <grant.likely(a)secretlab.ca> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Russell King <linux(a)arm.linux.org.uk> Cc: Samuel Ortiz <sameo(a)linux.intel.com> Cc: Stephen Warren <swarren(a)nvidia.com> Cc: Tony Lindgren <tony(a)atomide.com> Cc: Viresh Kumar <viresh.kumar(a)st.com> Cc: alsa-devel(a)alsa-project.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-mtd(a)lists.infradead.org Cc: linux-usb(a)vger.kernel.org Cc: spi-devel-general(a)lists.sourceforge.net

2 2

Re: [alsa-devel] BUG: USB audio discontinuities with 'UHCI: implement new semantics for URB_ISO_ASAP'
by Alan Stern 23 Apr '13

23 Apr '13

On Wed, 17 Apr 2013, Joe Rayhawk wrote: > Small buffer/period sizes on usb audio playback though UHCI works fine on v3.7 but causes audio discontinuities/delays on v3.8 and v3.9-rc7. > > I've bisected the behavior down to > > c44b225077bb1fb25ed5cd5c4f226897b91bedd4 > 'UHCI: implement new semantics for URB_ISO_ASAP' > > Tested on two UHCI hosts and three usb-audio devices, all from different vendors. Can you provide a usbmon trace showing the problems? And maybe also a similar trace made under a 3.7 kernel, where the problem doesn't occur? Alan Stern -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

3 10

[alsa-devel] [PATCH 2/2] ALSA: ASoc: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
by Michal Bachraty 23 Apr '13

23 Apr '13

For TDM mode, BCLK-to-LCLK ratio is computed as (tdm_slots) x (word_length). I2S mode is only subset of TDM mode with specific tdm_slots = 2 channels. Also bclk_lrclk_ratio can be greater than 255, therefore u16 need to be used. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 7 ++++--- sound/soc/davinci/davinci-mcasp.h | 2 +- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 63633e9..d54e0d9 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -636,11 +636,12 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, * callback, take it into account here. That allows us to for example * send 32 bits per channel to the codec, while only 16 of them carry * audio payload. - * The clock ratio is given for a full period of data (both left and - * right channels), so it has to be divided by 2. + * The clock ratio is given for a full period of data (for I2S format + * both left and right channels), so it has to be divided by number of + * tdm-slots (for I2S - divided by 2). */ if (dev->bclk_lrclk_ratio) - word_length = dev->bclk_lrclk_ratio / 2; + word_length = dev->bclk_lrclk_ratio / dev->tdm_slots; /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 0edd3b5..a9ac0c1 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -38,7 +38,7 @@ struct davinci_audio_dev { u8 num_serializer; u8 *serial_dir; u8 version; - u8 bclk_lrclk_ratio; + u16 bclk_lrclk_ratio; /* McASP FIFO related */ u8 txnumevt; -- 1.7.9.5

3 2

[alsa-devel] [PATCH 1/2] ALSA: ASoC davinci-pcm, davinci-mcasp Clean up active_serializers
by Michal Bachraty 23 Apr '13

23 Apr '13

As pointed of by Vaibhav, commit message: "ASoC: davinci-mcasp: Add support for multichannel playback" number of active serializers can be hidden into fifo_level variable, which is set in davimci-mcasp. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 8 +++++--- sound/soc/davinci/davinci-pcm.c | 11 +++++------ sound/soc/davinci/davinci-pcm.h | 1 - 3 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 8a9c0ee..63633e9 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -834,17 +834,20 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int word_length; u8 fifo_level; u8 slots = dev->tdm_slots; + u8 active_serializers; int channels; struct snd_interval *pcm_channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); channels = pcm_channels->min; + active_serializers = (channels + slots - 1) / slots; + if (davinci_hw_common_param(dev, substream->stream, channels) == -EINVAL) return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - fifo_level = dev->txnumevt; + fifo_level = dev->txnumevt * active_serializers; else - fifo_level = dev->rxnumevt; + fifo_level = dev->rxnumevt * active_serializers; if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) davinci_hw_dit_param(dev); @@ -889,7 +892,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; - dma_params->active_serializers = (channels + slots - 1) / slots; davinci_config_channel_size(dev, word_length); return 0; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 0161579..8460edc 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -182,7 +182,6 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) unsigned short acnt; unsigned int count; unsigned int fifo_level; - unsigned char serializers = prtd->params->active_serializers; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; @@ -196,14 +195,14 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) data_type = prtd->params->data_type; count = period_size / data_type; if (fifo_level) - count /= fifo_level * serializers; + count /= fifo_level; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; dst_bidx = 4; - src_cidx = data_type * fifo_level * serializers; + src_cidx = data_type * fifo_level; dst_cidx = 0; } else { src = prtd->params->dma_addr; @@ -211,7 +210,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src_bidx = 0; dst_bidx = data_type; src_cidx = 0; - dst_cidx = data_type * fifo_level * serializers; + dst_cidx = data_type * fifo_level; } acnt = prtd->params->acnt; @@ -226,8 +225,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) ASYNC); else edma_set_transfer_params(prtd->asp_link[0], acnt, - fifo_level * serializers, - count, fifo_level * serializers, + fifo_level, + count, fifo_level, ABSYNC); } diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 0d84d32..fbb710c 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -27,7 +27,6 @@ struct davinci_pcm_dma_params { unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; - unsigned char active_serializers; /* num. of active audio serializers */ }; int davinci_soc_platform_register(struct device *dev); -- 1.7.9.5

3 2

[alsa-devel] [PATCH 1/2] ALSA: ASoc: spdif_transciever: Add DT support
by Michal Bachraty 23 Apr '13

23 Apr '13

Add devicetree support for this dummy audio soc driver. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/codecs/spdif_transciever.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index 112a49d..4c8c295 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -20,6 +20,7 @@ #include <sound/soc.h> #include <sound/pcm.h> #include <sound/initval.h> +#include <linux/of.h> #define DRV_NAME "spdif-dit" @@ -52,12 +53,21 @@ static int spdif_dit_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id spdif_dit_dt_ids[] = { + { .compatible = DRV_NAME, }, + { } +}; +MODULE_DEVICE_TABLE(of, spdif_dits_dt_ids); +#endif + static struct platform_driver spdif_dit_driver = { .probe = spdif_dit_probe, .remove = spdif_dit_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(spdif_dit_dt_ids), }, }; -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: generic-dmaengine-pcm: call dma_request_slave_channel()
by Shawn Guo 23 Apr '13

23 Apr '13

dma_request_slave_channel() is a more appropriate API for dmaengine clients that adopt generic DMA bindings to call. Let's use it instead of of_dma_request_slave_channel() to save <linux/of_dma.h> include. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/soc-generic-dmaengine-pcm.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 98e1310..e29ec3c 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -21,7 +21,6 @@ #include <sound/soc.h> #include <linux/dma-mapping.h> #include <linux/of.h> -#include <linux/of_dma.h> #include <sound/dmaengine_pcm.h> @@ -219,19 +218,19 @@ static const char * const dmaengine_pcm_dma_channel_names[] = { }; static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, - struct device_node *of_node) + struct device *dev) { unsigned int i; - if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !of_node) + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !dev->of_node) return; if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { - pcm->chan[0] = of_dma_request_slave_channel(of_node, "rx-tx"); + pcm->chan[0] = dma_request_slave_channel(dev, "rx-tx"); pcm->chan[1] = pcm->chan[0]; } else { for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - pcm->chan[i] = of_dma_request_slave_channel(of_node, + pcm->chan[i] = dma_request_slave_channel(dev, dmaengine_pcm_dma_channel_names[i]); } } @@ -255,7 +254,7 @@ int snd_dmaengine_pcm_register(struct device *dev, pcm->config = config; pcm->flags = flags; - dmaengine_pcm_request_chan_of(pcm, dev->of_node); + dmaengine_pcm_request_chan_of(pcm, dev); if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) return snd_soc_add_platform(dev, &pcm->platform, -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: generic-dmaengine-pcm: use a more common dma name
by Shawn Guo 23 Apr '13

23 Apr '13

The examples in Documentation/devicetree/bindings/dma/dma.txt recommends the name for dma channel doing both RX and TX to be "rx-tx". This becomes a common pattern that has been adopted by platforms that converts to generic DMA bindings. Let's follow this common pattern in generic-dmaengine-pcm. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 5fd5ed4..98e1310 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -227,7 +227,7 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, return; if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { - pcm->chan[0] = of_dma_request_slave_channel(of_node, "tx_rx"); + pcm->chan[0] = of_dma_request_slave_channel(of_node, "rx-tx"); pcm->chan[1] = pcm->chan[0]; } else { for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { -- 1.7.9.5

3 2

[alsa-devel] [PATCH, for-next] ALSA: usb-audio: use FMTBITs in parse_audio_format_i
by Eldad Zack 23 Apr '13

23 Apr '13

Replace the usage of SNDRV_PCM_FORMAT_* macros with the equivalent SNDRV_PCM_FMTBIT_* macros, and the result be can assigned directly to the formats field. Modify the variable name to reflect the change. The following sparse messages are silenced: sound/usb/format.c:377:44: warning: incorrect type in assignment (different base types) sound/usb/format.c:377:44: expected int [signed] pcm_format sound/usb/format.c:377:44: got restricted snd_pcm_format_t [usertype] <noident> sound/usb/format.c:379:44: warning: incorrect type in assignment (different base types) sound/usb/format.c:379:44: expected int [signed] pcm_format sound/usb/format.c:379:44: got restricted snd_pcm_format_t [usertype] <noident> sound/usb/format.c:382:36: warning: incorrect type in assignment (different base types) sound/usb/format.c:382:36: expected int [signed] pcm_format sound/usb/format.c:382:36: got restricted snd_pcm_format_t [usertype] <noident> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/format.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 020ede0..e025e28 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -365,7 +365,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, { struct usb_interface_descriptor *altsd = get_iface_desc(iface); int protocol = altsd->bInterfaceProtocol; - int pcm_format, ret; + int pcm_formats, ret; if (fmt->bFormatType == UAC_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx @@ -377,14 +377,14 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ if (chip->setup == 0x00 && fp->altsetting == 6) - pcm_format = SNDRV_PCM_FORMAT_S16_BE; + pcm_formats = SNDRV_PCM_FMTBIT_S16_BE; else - pcm_format = SNDRV_PCM_FORMAT_S16_LE; + pcm_formats = SNDRV_PCM_FMTBIT_S16_LE; break; default: - pcm_format = SNDRV_PCM_FORMAT_S16_LE; + pcm_formats = SNDRV_PCM_FMTBIT_S16_LE; } - fp->formats = 1uLL << pcm_format; + fp->formats = pcm_formats; } else { fp->formats = parse_audio_format_i_type(chip, fp, format, fmt, protocol); -- 1.8.1.5

2 7

[alsa-devel] [RFC] ASoC: Current state of Dynamic PCM API
by Sebastian Hesselbarth 22 Apr '13

22 Apr '13

Hi, I am rewriting Marvell Kirkwood ASoC driver to also support Dove and reflect the audio controller's capabilities. The audio controller can have one DMA stream for playback and capture respectively. While capture source can be either I2S or SPDIF, playback can be on both interfaces simultaneously. The setup I have in mind for a generic Kirkwood/Dove soundcard is: (DMA)-->(CPU_DAI)--+<-->(I2S Codec) | +--->(SPDIF TX Codec) | +<---(SPDIF RX Codec) I sent a mail on how to handle multiple codecs on a single cpu_dai earlier, and Mark pointed me to Dynamic PCM API. Unfortunately, current mainline ASoC has no driver using DPCM as an example and integration of DPCM patches look kind-of incomplete to me. Can any of you point me to patches, repositories, or documentation about DPCM where I can start with? Regards, Sebastian

2 1

Re: [alsa-devel] [PATCH] ASoC: add RT5640 CODEC driver
by Bard 22 Apr '13

22 Apr '13

Dear Mark, Please see my reply below. Thanks. -----Original Message----- From: Mark Brown [mailto:broonie@opensource.wolfsonmicro.com] Sent: Tuesday, April 16, 2013 10:38 PM To: Bard Cc: Flove; Oder Chiou; swarren(a)nvidia.com; bard Subject: Re: [PATCH] ASoC: add RT5640 CODEC driver On Tue, Apr 16, 2013 at 10:10:37PM +0800, bardliao(a)realtek.com wrote: Please remember to post things to the list... just a quick scan through here: > +static int rt5640_reg_init(struct snd_soc_codec *codec) { > + int i; > + for (i = 0; i < RT5640_INIT_REG_LEN; i++) > + snd_soc_write(codec, init_list[i].reg, init_list[i].val); > + return 0; > +} This should be converted over to regmap. There's support for paging in the core, look at regmap_range. Bard: I will use regmap_register_patch to do that. > +/* IN1/IN2 Input Type */ > +static const char * const rt5640_input_mode[] = { > + "Single ended", "Differential"}; > + > +static const SOC_ENUM_SINGLE_DECL( > + rt5640_in1_mode_enum, RT5640_IN1_IN2, > + RT5640_IN_SFT1, rt5640_input_mode); > + > +static const SOC_ENUM_SINGLE_DECL( > + rt5640_in2_mode_enum, RT5640_IN3_IN4, > + RT5640_IN_SFT2, rt5640_input_mode); Platform data... Bard: Can I export a function that machine can configure it by this function? > +static int rt5640_vol_rescale_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) { > + struct soc_mixer_control *mc = > + (struct soc_mixer_control *)kcontrol->private_value; > + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); > + unsigned int val, val2; > + > + val = VOL_RESCALE_MAX_VOL - ucontrol->value.integer.value[0]; > + val2 = VOL_RESCALE_MAX_VOL - ucontrol->value.integer.value[1]; > + return snd_soc_update_bits_locked(codec, mc->reg, RT5640_L_VOL_MASK | > + RT5640_R_VOL_MASK, val << mc->shift | val2); } This looks like a variant on the _RANGE controls? Bard: Yes, we want to limit the max value of volume. If it is not good, I will change it. > +static int hp_event(struct snd_soc_dapm_widget *w, > + struct snd_kcontrol *kcontrol, int event) { > + switch (event) { > + case SND_SOC_DAPM_POST_PMU: > + break; > + > + case SND_SOC_DAPM_PRE_PMD: > + break; > + > + default: > + return 0; > + } > + return 0; > +} Remove this, it doesn't actually do anything any more. Bard: I will remove it. ------Please consider the environment before printing this e-mail.

2 1

[alsa-devel] I connected my digital piano and it just worked :-)
by David Henningsson 22 Apr '13

22 Apr '13

On a slightly different note; I needed to connect my digital piano to my laptop the other day, but the laptop had no MIDI interface. So I tried connecting it through USB, because the piano had a USB port too. I knew it required a 3rd party driver on Windows, so I didn't expect much. It just worked. :-) From someone who knows that fixing hardware often takes a lot of effort and hard work, I just wanted to shout a "thank you" and a "keep up the good work" to all contributors! :-) -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

1 0

[alsa-devel] [PATCH] ALSA: hda - Limit internal mic boost for a few Asus machines
by David Henningsson 22 Apr '13

22 Apr '13

These are being reported as being so noisy at high mic boost levels, so they are unusable in practice. Therefore artificially limit the boosts. BugLink: https://bugs.launchpad.net/bugs/1089795 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 42 ++++++++++++++++++++++++++++++++++++++++- 1 file changed, 41 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 28b3f5a..6e5d383 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3156,6 +3156,38 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, } } +static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->gen.autocfg; + int i; + + /* The mic boosts on level 2 and 3 are too noisy + on the internal mic input. + Therefore limit the boost to 0 or 1. */ + + if (action != HDA_FIXUP_ACT_PROBE) + return; + + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; + unsigned int defcfg; + if (cfg->inputs[i].type != AUTO_PIN_MIC) + continue; + defcfg = snd_hda_codec_get_pincfg(codec, nid); + if (snd_hda_get_input_pin_attr(defcfg) != INPUT_PIN_ATTR_INT) + continue; + + snd_hda_override_amp_caps(codec, nid, HDA_INPUT, + (0x00 << AC_AMPCAP_OFFSET_SHIFT) | + (0x01 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x2f << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); + } +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -3190,6 +3222,7 @@ enum { ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, + ALC269_FIXUP_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -3428,6 +3461,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC271_FIXUP_DMIC, }, + [ALC269_FIXUP_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3458,11 +3495,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 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, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), - SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/7] Add capture support to compress API
by Richard Fitzgerald 22 Apr '13

22 Apr '13

This chain of patches does some clearing up and add capture support to the compressed API. Missing is support for capture streams that don't use the copy callback. Also I have included two patches at the end of the chain that remove the buffer pointers and just use the cumulative values however as our setup uses the copy callback these are untested past build so should be taken with caution. Thanks, Charles Thess patches apply to: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git branch "for-next" based on top of commit 126825e ALSA: snd-usb: add quirks handler for DSD streams - RichF Charles Keepax (7): ALSA: compress_core: Update calc_avail to use cumulative values ALSA: compress_core: Calculate avail correctly for capture streams ALSA: compress_core: Deconstify copy callback buffer ALSA: compress_core: Add support for capture streams ASoC: soc-compress: Deduce stream direction ALSA: compress_core: Remove unused hw_pointer ALSA: compress_core: Rework writes to use cumulative values include/sound/compress_driver.h | 4 +- sound/core/compress_offload.c | 117 +++++++++++++++++++++++++------------- sound/soc/soc-compress.c | 11 +++- 3 files changed, 87 insertions(+), 45 deletions(-) -- 1.7.2.5

6 15

[alsa-devel] [PATCH v3] ALSA: snd-usb-audio: set the timeout for usb control set messages to 5000 ms
by Daniel Schürmann 22 Apr '13

22 Apr '13

This is an other version of the patch send via git send-email. I still distinguish between set and get but as long both are 5000 ms GCC will remove it anyway. IMHO this is more easy read and there is no need to explain why we use a get timeout for set messages. Commit message: Set the timeout for USB control set messages according to the USB 2 spec, using the macros from include/linux/usb.h. The get timout becomes 5000 ms even though it is 500 ms in the spec. This patch is required to run the Hercules RMX2 which needs a timeout of 1240 ms. Signed-off-by: Daniel Schürmann <daschuer(a)mixxx.org> --- sound/usb/helper.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/usb/helper.c b/sound/usb/helper.c index c1db28f..6209024 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -86,14 +86,22 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, { int err; void *buf = NULL; + int timeout; if (size > 0) { buf = kmemdup(data, size, GFP_KERNEL); if (!buf) return -ENOMEM; } + + if (requesttype & USB_DIR_IN) + timeout = USB_CTRL_GET_TIMEOUT; + else + timeout = USB_CTRL_SET_TIMEOUT; + err = usb_control_msg(dev, pipe, request, requesttype, - value, index, buf, size, 1000); + value, index, buf, size, timeout); + if (size > 0) { memcpy(data, buf, size); kfree(buf); -- 1.7.9.5

3 2

[alsa-devel] [patch] ALSA: compress: info leak in snd_compr_get_caps()
by Dan Carpenter 22 Apr '13

22 Apr '13

If the ->get_caps() function doesn't clear the buffer then there would stack information leaked to userspace. For example, soc_compr_get_caps() can return success without clearing the buffer. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Perhaps the soc_compr_get_caps() function should return an error code if the platform->driver->compr_ops is NULL. I'm not sure about that, and it's a separate issue anyway. diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index c84abc8..8d3190a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -375,6 +375,7 @@ snd_compr_get_caps(struct snd_compr_stream *stream, unsigned long arg) if (!stream->ops->get_caps) return -ENXIO; + memset(&caps, 0, sizeof(caps)); retval = stream->ops->get_caps(stream, &caps); if (retval) goto out;

4 4

Re: [alsa-devel] [PATCH v2] Add M2Tech hiFace USB-SPDIF driver
by Pavel Hofman 22 Apr '13

22 Apr '13

On 22.4.2013 10:31, Pietro Cipriano wrote: > Il 22/04/2013 09:40, Pavel Hofman ha scritto: >> >> >> On 20.4.2013 22:15, Daniel Mack wrote: >>> On 13.02.2013 18:11, Antonio Ospite wrote: >>>> Add driver for M2Tech hiFace USB-SPDIF interface and compatible >>>> devices. >>>> >>>> M2Tech hiFace and compatible devices offer a Hi-End S/PDIF Output >>>> Interface, see http://www.m2tech.biz/hiface.html >>> >>> Just curious: is there a new version of this patch set coming? >> >> I have read users report large latencies (unable to watch movies) of >> hiFace1 in OSX while reportedly they have not identified any major >> latency in windows. Perhaps the fixed PCM_PACKET_SIZE > The issue is related to a no controllable buffer dimension, which > depends by Audio Core thus present only on OSx. A easy workaround is of > to set up a higher output sampling frequency, for instance at 192KHz the > delay is no more perceptible, obviously the DAC must be compliant with > this setting. > > Pietro Thanks for very fast reponse. So is it possible to reduce the period size in the linux driver? Right now if I calculate correctly it is 4096 bytes / 32bits / 2 channels / 48000 Hz = 10 ms. Thanks, Pavel.

1 0

Re: [alsa-devel] [PATCH 26/28] proc: Supply a function to remove a proc entry by PDE [RFC]
by Rafael J. Wysocki 22 Apr '13

22 Apr '13

On Tuesday, April 16, 2013 07:27:34 PM David Howells wrote: > Supply a function (proc_remove()) to remove a proc entry (and any subtree > rooted there) by proc_dir_entry pointer rather than by name and (optionally) > root dir entry pointer. This allows us to eliminate all remaining pde->name > accesses outside of procfs. > > Signed-off-by: David Howells <dhowells(a)redhat.com> > cc: linux-acpi(a)vger.kernel.org > cc: openipmi-developer(a)lists.sourceforge.net > cc: devicetree-discuss(a)lists.ozlabs.org > cc: linux-pci(a)vger.kernel.org > cc: netdev(a)vger.kernel.org > cc: netfilter-devel(a)vger.kernel.org > cc: alsa-devel(a)alsa-project.org The ACPI changes look good to me. Acked-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> > --- > > drivers/acpi/sbs.c | 21 ++++----------------- > drivers/char/ipmi/ipmi_msghandler.c | 2 +- > drivers/misc/sgi-gru/gruprocfs.c | 2 +- > drivers/of/base.c | 11 +---------- > drivers/pci/proc.c | 12 +++--------- > fs/proc/generic.c | 7 +++++++ > fs/proc/vmcore.c | 2 +- > include/linux/proc_fs.h | 2 ++ > kernel/irq/proc.c | 6 +----- > net/8021q/vlanproc.c | 9 ++------- > net/core/pktgen.c | 6 ++---- > net/ipv4/netfilter/ipt_CLUSTERIP.c | 4 ++-- > net/ipv6/proc.c | 3 +-- > net/netfilter/xt_hashlimit.c | 4 ++-- > sound/core/info.c | 19 +++++-------------- > 15 files changed, 35 insertions(+), 75 deletions(-) > > diff --git a/drivers/acpi/sbs.c b/drivers/acpi/sbs.c > index a296e08..b6241ee 100644 > --- a/drivers/acpi/sbs.c > +++ b/drivers/acpi/sbs.c > @@ -521,19 +521,6 @@ acpi_sbs_add_fs(struct proc_dir_entry **dir, > return 0; > } > > -static void > -acpi_sbs_remove_fs(struct proc_dir_entry **dir, > - struct proc_dir_entry *parent_dir) > -{ > - if (*dir) { > - remove_proc_entry(ACPI_SBS_FILE_INFO, *dir); > - remove_proc_entry(ACPI_SBS_FILE_STATE, *dir); > - remove_proc_entry(ACPI_SBS_FILE_ALARM, *dir); > - remove_proc_entry((*dir)->name, parent_dir); > - *dir = NULL; > - } > -} > - > /* Smart Battery Interface */ > static struct proc_dir_entry *acpi_battery_dir = NULL; > > @@ -836,8 +823,8 @@ static void acpi_battery_remove(struct acpi_sbs *sbs, int id) > power_supply_unregister(&battery->bat); > } > #ifdef CONFIG_ACPI_PROCFS_POWER > - if (battery->proc_entry) > - acpi_sbs_remove_fs(&battery->proc_entry, acpi_battery_dir); > + proc_remove(battery->proc_entry); > + battery->proc_entry = NULL; > #endif > } > > @@ -873,8 +860,8 @@ static void acpi_charger_remove(struct acpi_sbs *sbs) > if (sbs->charger.dev) > power_supply_unregister(&sbs->charger); > #ifdef CONFIG_ACPI_PROCFS_POWER > - if (sbs->charger_entry) > - acpi_sbs_remove_fs(&sbs->charger_entry, acpi_ac_dir); > + proc_remove(sbs->charger_entry); > + sbs->charger_entry = NULL; > #endif > } > -- I speak only for myself. Rafael J. Wysocki, Intel Open Source Technology Center.

1 0

[alsa-devel] [PATCH v2] ALSA: snd-usb-audio: set the timeout for usb control set messages to 5000 ms
by Daniel Schürmann 20 Apr '13

20 Apr '13

Thank you Daniel for review. Here is an updated version of the patch against https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/ 581cbef46ae60073252208fc1f26dd1044f6e215 Set the timeout for USB control set messages according to the USB 2 spec §9.2.6.4 to 5000 ms. To avoid new issues, the get timeout is unchanged at 1000 ms, though it is 500 ms in the spec. This patch is required to run the Hercules RMX2 which needs a timeout > 1240 ms Signed-off-by: Daniel Schürmann <daschuer(a)mixxx.org> --- diff --git a/sound/usb/helper.c b/sound/usb/helper.c index c1db28f..c1caae4 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -23,6 +23,11 @@ #include "helper.h" #include "quirks.h" +/* Value from 9.2.6.4 USB 2 spec */ +#define USB_MSG_SET_TIMEOUT 5000 +/* Value from spec is 500 but we pick 1000 for legacy reasons */ +#define USB_MSG_GET_TIMEOUT 1000 + /* * combine bytes and get an integer value */ @@ -86,14 +91,19 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, { int err; void *buf = NULL; + int timeout; if (size > 0) { buf = kmemdup(data, size, GFP_KERNEL); if (!buf) return -ENOMEM; } + if (requesttype & USB_DIR_IN) + timeout = USB_MSG_GET_TIMEOUT; + else + timeout = USB_MSG_SET_TIMEOUT; err = usb_control_msg(dev, pipe, request, requesttype, - value, index, buf, size, 1000); + value, index, buf, size, timeout); if (size > 0) { memcpy(data, buf, size); kfree(buf);

3 4

[alsa-devel] DAPM widget power off
by yuvaraj r 20 Apr '13

20 Apr '13

Hi All, I have two different dais 1. Hifi Multimedia 2. Voice call dai Before opening voice call dai. I closed Hifi stream. After 5 sec Voice call stream dapm widget got powered off automatically. I am assuming that stream close event of Multimedia causing problem. What will issue in my cause. Thanks Yuvaraj

2 1

[alsa-devel] snd_pcm_avail() and snd_pcm_avail_update() always returns buffer_size
by Nate Case 20 Apr '13

20 Apr '13

Hello, I'm debugging an issue where both snd_pcm_avail_update() and snd_pcm_avail() are always returning buffer_size, even after successfully calling snd_pcm_writei() repeatedly. This is causing problems when I first start playback. My goal was to fill up the playback buffer until it's almost full when first starting, and then enter my regular playback loop where I write period_size at a time. However, I'm not able to tell when the playback buffer is full from snd_pcm_avail() or snd_pcm_avail_update() as expected. In case it's useful, the full snd_pcm_dump() output of the playback device is available here: http://pastebin.com/ai3e0cer To summarize the configuration, I'm using a chain of PCM devices that looks like this: Soft volume -> Rate conversion -> Route conversion -> HW PCM (USB audio) (notes: buffer_size for soft volume PCM is 640 [80ms], period size is 160 [20ms]. buffer_size for HW PCM is 3846 and period_size is 960. I'm opening the playback device in blocking mode, although the same thing appears to happen in non-blocking mode) Here is the sequence of events I'm seeing: 1) Call snd_pcm_writei() with 160 frames successfully. Function returns immediately in less than 1ms because the buffer is empty. 2) snd_pcm_avail() returns 640 (expected 640-160 = 480) snd_pcm_dump() shows the following for the HW PCM device: appl_ptr : 960 hw_ptr : 0 3) Call snd_pcm_writei() again with 160 frames successfully. Function returns immediately again. 4) snd_pcm_avail() returns 640 (expected 480-160 = 320) snd_pcm_dump() shows the following for the HW PCM device: appl_ptr : 1920 hw_ptr : 0 5) Cycle repeats two more times, with these values for apply_ptr/hw_ptr reported after each subsequent snd_pcm_writei() call: appl_ptr : 2880 hw_ptr : 288 appl_ptr : 3840 hw_ptr : 288 All along, snd_pcm_avail() returns 640. 6) Call snd_pcm_writei() with 160 frames again. This time, the buffer is actually full, so it blocks for 20 ms as expected. 7) snd_pcm_avail() returns 640 snd_pcm_dump() shows the following for the HW PCM device: appl_ptr : 4800 hw_ptr : 960 8) Continue this loop, with each subsequent snd_pcm_writei() call blocking for 20 ms, while snd_pcm_avail() always returns 640. Any ideas on why this could be happening? Or am I misunderstanding something here? Thanks, Nate Case <ncase(a)xes-inc.com>

2 2

Re: [alsa-devel] PCM buffering ussues ( or probably the lack of understanding )
by Lars-Peter Clausen 20 Apr '13

20 Apr '13

On 04/20/2013 12:59 PM, marcelg wrote: > Lars Peter, > > Thanks for your reply. > > Some things are more clear now, however one issue I still wonder. > > > Quote > > " >> Suppose I want to have some buffering inside my pcm driver, how will I be >> able to get some samples in advance before starting playback ? Can I just >> call pcm_period_elapsed() a few times in a row to get the buffer filled >> before starting the actual playcback, or does this cause side effects ? >> > Why'd you want to do that? > " > > > If Alsa is not having an internal buffer as you pointed out. An my driver > for example advertises > min periods = 1 and max_periods = 10. Then it means my drivers is capable > of buffering up to 10 periods. > > However what I notice is that the Alsa layer pcm_copy() always supplies > "period_bytes" after each call to period_elapsed() > > This would imply that even my driver is able to buffer 10 periods, it never > receives more data as one period, because I call period_elapsed() > as soon as the data is consumed. The ALSA core keeps track of how many bytes are available in your buffer. It will supply your driver with as many bytes as there is space in the buffer. So if you start playback you should see it supplying you more than one period worth of bytes. Once the buffer is full you'll naturally only get one new period of each period that has elapsed. But as I said before if you don't do anything other then copying data to your buffer in the copy callback, you should probably get rid of the copy callback altogether and let ALSA take care of writing the data to your buffer. This will also allow you to use for example mmap. - Lars

1 0

[alsa-devel] Invitation to connect on LinkedIn
by Barry Song via LinkedIn 20 Apr '13

20 Apr '13

LinkedIn ------------ Barry Song requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, I'd like to add you to my professional network on LinkedIn. - Barry Accept invitation from Barry Song http://www.linkedin.com/e/jpxkk3-hfqntbbx-71/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View profile of Barry Song http://www.linkedin.com/e/jpxkk3-hfqntbbx-71/rso/196461153/a2mu/name/138036… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-hfqntbbx-71/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

1 0

[alsa-devel] PCM buffering ussues ( or probably the lack of understanding )
by marcelg 20 Apr '13

20 Apr '13

I am working on a PCM driver, have some issues and thing which are unclear to me, hopefully somebody here can clarify something for me. My PCM driver is straigh forward: - Copy callback copies the frames received in the callback into a small ringbuffer and starts a DMA transfer to the hardware. - DMA interrupt each times take the next chunk from the ringbuffer. - In the DMA interrupt the number of frames is counted, if te number >= period_size reported in runtime_struct pcm_period_elapsed() is called. - the pointer which is read by the pcm_pointer() callback is updated in each DMA interrupt. When using mplayer this works without problems, however with other applications like mpg123 its causes a lot of problems, for some reaosns I do not understand yet. It has something to do with buffer handling. basically in the cm driver I can influence a few parameters: periods_min/periods_max period_bytes_min, period_bytes_max buffer_size ( should be sufficient to hold periods_max with period_bytes_max I assume ) As far as I understand periods_min/periods_max is the size (number of buffer entries) of the internal ringbuffer Alsa uses. - Why is there both a min and a max ? How is chosen what will be used ? - Currently my period_min is set to 1 and my period_max set to 4, If I increase the period_max value it does not start properly, it seems that the samples are not deliverered in time. Is there any explanation for this, does the Alsa layer not any pre-buffering before the first copy() is called ? - Is there any advice / guidline for period min/max values ? I checked sourecs of severl drivers, some use 1 and some use values like 32 or more for period_max. The same applies to buffer_bytes min/max, currently my min is set to 128bytes , max to 2kB, larger values of max seem to cause more problems. From the documentation I understand that it does not matter if pcm_period_elapsed() is not always called, How dows the ALSA layer then know how many samples are played. the pointer value read the pcm_pointer() only counts from [0..period_max] so what if 2 or more calls to period_elapsed() are skipped ? Suppose I want to have some buffering inside my pcm driver, how will I be able to get some samples in advance before starting playback ? Can I just call pcm_period_elapsed() a few times in a row to get the buffer filled before starting the actual playcback, or does this cause side effects ? I noticed that the number of frames passed to pcm_copy is often less than the periods_size, is this determined by the application ? Lots of questions, but I try to understand the odd behaviour with certain applications ( and the best settings for a pcm driver which will work with most applications ) Any help would be appreciated. Marcel

3 2

Re: [alsa-devel] [PATCH 4/7] ALSA: compress_core: Add support for capture streams
by Vinod Koul 20 Apr '13

20 Apr '13

On Thu, Apr 18, 2013 at 11:02:08AM +0100, Charles Keepax wrote: > Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> > Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> > --- > sound/core/compress_offload.c | 43 +++++++++++++++++++++++++++++++++++++++- > 1 files changed, 41 insertions(+), 2 deletions(-) > > diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c > index 52ca4cc..52a2765 100644 > --- a/sound/core/compress_offload.c > +++ b/sound/core/compress_offload.c > @@ -296,7 +296,41 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, > static ssize_t snd_compr_read(struct file *f, char __user *buf, > size_t count, loff_t *offset) > { > - return -ENXIO; > + struct snd_compr_file *data = f->private_data; > + struct snd_compr_stream *stream; > + size_t avail; > + int retval; > + > + if (snd_BUG_ON(!data)) > + return -EFAULT; > + > + stream = &data->stream; > + mutex_lock(&stream->device->lock); > + > + /* read is allowed when stream is running */ > + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { > + retval = -EBADFD; > + goto out; minor one but shouldnt we allow read even when we have stoped or paused the stream. Stop is important as you might want to get all buffers after you have stopped. > + } > + > + avail = snd_compr_get_avail(stream); > + pr_debug("avail returned %ld\n", (unsigned long)avail); > + /* calculate how much we can read from buffer */ > + if (avail > count) > + avail = count; > + > + if (stream->ops->copy) { > + retval = stream->ops->copy(stream, buf, avail); > + } else { > + retval = -ENXIO; > + goto out; > + } > + if (retval > 0) > + stream->runtime->total_bytes_transferred += retval; > + > +out: > + mutex_unlock(&stream->device->lock); > + return retval; > } > > static int snd_compr_mmap(struct file *f, struct vm_area_struct *vma) > @@ -481,9 +515,14 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) > retval = stream->ops->set_params(stream, params); > if (retval) > goto out; > - stream->runtime->state = SNDRV_PCM_STATE_SETUP; > + > stream->metadata_set = false; > stream->next_track = false; > + > + if (stream->direction == SND_COMPRESS_PLAYBACK) > + stream->runtime->state = SNDRV_PCM_STATE_SETUP; > + else > + stream->runtime->state = SNDRV_PCM_STATE_PREPARED; > } else { > return -EPERM; > } > -- > 1.7.2.5 >

3 2

[alsa-devel] Solved Hercules RMX2
by Daniel Schürmann 20 Apr '13

20 Apr '13

Hallo Alsa Developers, I have fixed Support for Hercules RMX2. You can find the Bug and the patch here: https://bugs.launchpad.net/mixxx/+bug/1096687 The issue was caused by the first time setting the rate. I have changed the usb_msg_timeout to 1500 ms to make it work. The measures time is a ~1240 ms. All following rate settings are finished in not time < 4 ms Sound is now working fine but there are still IO errors just after connecting the device errno 5 - input/output error Here the extended Log: Apr 14 23:50:37 KellerPc kernel: [19213.746929] snd_usb_autoresume() -19 ID:6f8b113 chip->shutdown 0 chip->probing 1. Apr 14 23:50:37 KellerPc kernel: [19213.746934] -5 1 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0x0, type = 4 Apr 14 23:50:37 KellerPc kernel: [19213.747053] snd_usb_autoresume() -19 ID:6f8b113 chip->shutdown 0 chip->probing 1. Apr 14 23:50:37 KellerPc kernel: [19213.747058] -5 1 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0x0, type = 4 Apr 14 23:50:37 KellerPc kernel: [19213.747430] snd_usb_autoresume() -19 ID:6f8b113 chip->shutdown 0 chip->probing 1. Apr 14 23:50:37 KellerPc kernel: [19213.747435] -5 1 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0x0, type = 4 Apr 14 23:50:37 KellerPc kernel: [19213.747554] snd_usb_autoresume() -19 ID:6f8b113 chip->shutdown 0 chip->probing 1. Apr 14 23:50:37 KellerPc kernel: [19213.747559] -5 1 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0x0, type = 4 Apr 14 23:50:37 KellerPc kernel: [19213.752599] usbcore: registered new interface driver snd-usb-audio Apr 14 23:50:37 KellerPc kernel: [19213.775204] snd_usb_ctl_msg() dev:ed5eb000 pipe:80000d80 request:02, requesttype:a1, value:0201, index:0a00, data_size:8. Apr 14 23:50:37 KellerPc kernel: [19213.776264] snd_usb_ctl_msg() dev:ed5eb000 pipe:80000d80 request:02, requesttype:a1, value:0201, index:0a00, data_size:8. Apr 14 23:50:37 KellerPc kernel: [19213.776436] snd_usb_ctl_msg() dev:ed5eb000 pipe:80000d80 request:02, requesttype:a1, value:0201, index:0a00, data_size:8. The current solution is fine for me, but maybe there is an other place for tweaking the startup timing. Any ideas? Is it possible to include the fix in trunk? Thank you. Kind regards, Daniel Schürmann

3 9

[alsa-devel] [PATCH 2/2] ALSA: ASoC: spdif_receiver: Add DT support
by Michal Bachraty 19 Apr '13

19 Apr '13

Add devicetree support for this dummy audio soc driver. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/codecs/spdif_receiver.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c index dd8d856..5d469c7 100644 --- a/sound/soc/codecs/spdif_receiver.c +++ b/sound/soc/codecs/spdif_receiver.c @@ -21,6 +21,9 @@ #include <sound/soc.h> #include <sound/pcm.h> #include <sound/initval.h> +#include <linux/of.h> + +#define DRV_NAME "spdif-dir" #define STUB_RATES SNDRV_PCM_RATE_8000_192000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -51,12 +54,21 @@ static int spdif_dir_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id spdif_dit_dt_ids[] = { + { .compatible = DRV_NAME, }, + { } +}; +MODULE_DEVICE_TABLE(of, spdif_dits_dt_ids); +#endif + static struct platform_driver spdif_dir_driver = { .probe = spdif_dir_probe, .remove = spdif_dir_remove, .driver = { .name = "spdif-dir", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(spdif_dit_dt_ids), }, }; -- 1.7.9.5

1 0

Re: [alsa-devel] [PATCH 5/7] ASoC: soc-compress: Deduce stream direction
by Vinod Koul 19 Apr '13

19 Apr '13

On Thu, Apr 18, 2013 at 11:02:38AM +0100, Charles Keepax wrote: > Previously we just hard coded all streams as playback streams, this > patch checks the DAI to see if it is a capture or playback stream. It is > worth noting that at this time only unidirectional streams are > supported. well the intention was the compress device would be either playback or record. you should have two devices if you support both. > > Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> > Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> > --- > sound/soc/soc-compress.c | 9 ++++++++- > 1 files changed, 8 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c > index da83e56..3853f7e 100644 > --- a/sound/soc/soc-compress.c > +++ b/sound/soc/soc-compress.c > @@ -384,7 +384,14 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) > /* check client and interface hw capabilities */ > snprintf(new_name, sizeof(new_name), "%s %s-%d", > rtd->dai_link->stream_name, codec_dai->name, num); > - direction = SND_COMPRESS_PLAYBACK; > + > + if (codec_dai->driver->playback.channels_min) > + direction = SND_COMPRESS_PLAYBACK; > + else if (codec_dai->driver->capture.channels_min) > + direction = SND_COMPRESS_CAPTURE; > + else > + return -EINVAL; > + > compr = kzalloc(sizeof(*compr), GFP_KERNEL); > if (compr == NULL) { > snd_printk(KERN_ERR "Cannot allocate compr\n"); > -- > 1.7.2.5 >

1 0

[alsa-devel] ALSA framework problem: FE disconnected, BE still connected!
by lkasam＠codeaurora.org 19 Apr '13

19 Apr '13

Hi Liam, We have a problem encountered in ALSA framework wrt FE/BE disconnection and fix to it is mentioned in below code change. please check it and provide your valuable comments. ---------------------- From: Laxminath Kasam <lkasam(a)codeaurora.org> Date: Wed, 3 Apr 2013 14:03:17 +0530 Subject: [PATCH] ASoC: audio: Fix BE not disconnected even codec teardown Scenario: When FE1(LowLatency) comes in which opens BE1 and puts it in state SND_SOC_DPCM_STATE_START. Similarly FE2(LPA) comes in with connecting to same BE. Now FE2 is in SND_SOC_DPCM_STATE_PREPARE state and waits for TRIGGER_START. Meanwhile FE1 is closed which doesn't affect BE1 and FE2 which are connected. Also, for FE2 pcm_close is called before TRIGGER_START itself. In such a case, BE1 is not getting disconnected while Codec is teared down resulting in failures. Fix: Whenever BE1 is found to be in SND_SOC_DPCM_STATE_START state and FE1 being connected is not started yet, and close sequence happens for this FE, then allow hw_free to happen for BE1 if this is the only FE connected to this BE1. This allows BE to be disconnected which calls afe_close then followed by Codec teardown not resulting in any failures. Signed-off-by: Laxminath Kasam <lkasam(a)codeaurora.org> --- sound/soc/soc-pcm.c | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 915c3c2..ed0ea86 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1765,7 +1765,12 @@ static int soc_dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) (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)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + !((be->dpcm[stream].state == SND_SOC_DPCM_STATE_START)&& + ((fe->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (fe->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && + (fe->dpcm[stream].state != + SND_SOC_DPCM_STATE_SUSPEND)))) continue; dev_dbg(be->dev, "dpcm: hw_free BE %s\n", -- ----------------------------------------- Please let me know if you are looking for more details. Thanks Kasam

3 3

[alsa-devel] [PATCH 1/4] ASoC: imx-sgtl5000: Do not enter the error path on success
by Fabio Estevam 19 Apr '13

19 Apr '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Return on success instead of entering the error path. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-sgtl5000.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 9584e78..88fc02c 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -174,6 +174,9 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) } platform_set_drvdata(pdev, data); + + return 0; + clk_fail: clk_put(data->codec_clk); fail: -- 1.7.9.5

2 10

Re: [alsa-devel] [PATCH 26/28] proc: Supply a function to remove a proc entry by PDE [RFC]
by Bjorn Helgaas 18 Apr '13

18 Apr '13

On Tue, Apr 16, 2013 at 12:27 PM, David Howells <dhowells(a)redhat.com> wrote: > Supply a function (proc_remove()) to remove a proc entry (and any subtree > rooted there) by proc_dir_entry pointer rather than by name and (optionally) > root dir entry pointer. This allows us to eliminate all remaining pde->name > accesses outside of procfs. > > Signed-off-by: David Howells <dhowells(a)redhat.com> > cc: linux-acpi(a)vger.kernel.org > cc: openipmi-developer(a)lists.sourceforge.net > cc: devicetree-discuss(a)lists.ozlabs.org > cc: linux-pci(a)vger.kernel.org > cc: netdev(a)vger.kernel.org > cc: netfilter-devel(a)vger.kernel.org > cc: alsa-devel(a)alsa-project.org > --- > > drivers/acpi/sbs.c | 21 ++++----------------- > drivers/char/ipmi/ipmi_msghandler.c | 2 +- > drivers/misc/sgi-gru/gruprocfs.c | 2 +- > drivers/of/base.c | 11 +---------- > drivers/pci/proc.c | 12 +++--------- For the drivers/pci part: Acked-by: Bjorn Helgaas <bhelgaas(a)google.com> I assume you'll merge this via something other than my tree. > fs/proc/generic.c | 7 +++++++ > fs/proc/vmcore.c | 2 +- > include/linux/proc_fs.h | 2 ++ > kernel/irq/proc.c | 6 +----- > net/8021q/vlanproc.c | 9 ++------- > net/core/pktgen.c | 6 ++---- > net/ipv4/netfilter/ipt_CLUSTERIP.c | 4 ++-- > net/ipv6/proc.c | 3 +-- > net/netfilter/xt_hashlimit.c | 4 ++-- > sound/core/info.c | 19 +++++-------------- > 15 files changed, 35 insertions(+), 75 deletions(-) > > diff --git a/drivers/acpi/sbs.c b/drivers/acpi/sbs.c > index a296e08..b6241ee 100644 > --- a/drivers/acpi/sbs.c > +++ b/drivers/acpi/sbs.c > @@ -521,19 +521,6 @@ acpi_sbs_add_fs(struct proc_dir_entry **dir, > return 0; > } > > -static void > -acpi_sbs_remove_fs(struct proc_dir_entry **dir, > - struct proc_dir_entry *parent_dir) > -{ > - if (*dir) { > - remove_proc_entry(ACPI_SBS_FILE_INFO, *dir); > - remove_proc_entry(ACPI_SBS_FILE_STATE, *dir); > - remove_proc_entry(ACPI_SBS_FILE_ALARM, *dir); > - remove_proc_entry((*dir)->name, parent_dir); > - *dir = NULL; > - } > -} > - > /* Smart Battery Interface */ > static struct proc_dir_entry *acpi_battery_dir = NULL; > > @@ -836,8 +823,8 @@ static void acpi_battery_remove(struct acpi_sbs *sbs, int id) > power_supply_unregister(&battery->bat); > } > #ifdef CONFIG_ACPI_PROCFS_POWER > - if (battery->proc_entry) > - acpi_sbs_remove_fs(&battery->proc_entry, acpi_battery_dir); > + proc_remove(battery->proc_entry); > + battery->proc_entry = NULL; > #endif > } > > @@ -873,8 +860,8 @@ static void acpi_charger_remove(struct acpi_sbs *sbs) > if (sbs->charger.dev) > power_supply_unregister(&sbs->charger); > #ifdef CONFIG_ACPI_PROCFS_POWER > - if (sbs->charger_entry) > - acpi_sbs_remove_fs(&sbs->charger_entry, acpi_ac_dir); > + proc_remove(sbs->charger_entry); > + sbs->charger_entry = NULL; > #endif > } > > diff --git a/drivers/char/ipmi/ipmi_msghandler.c b/drivers/char/ipmi/ipmi_msghandler.c > index 1420bbb..4d439d2 100644 > --- a/drivers/char/ipmi/ipmi_msghandler.c > +++ b/drivers/char/ipmi/ipmi_msghandler.c > @@ -4541,7 +4541,7 @@ static void __exit cleanup_ipmi(void) > del_timer_sync(&ipmi_timer); > > #ifdef CONFIG_PROC_FS > - remove_proc_entry(proc_ipmi_root->name, NULL); > + proc_remove(proc_ipmi_root); > #endif /* CONFIG_PROC_FS */ > > driver_unregister(&ipmidriver.driver); > diff --git a/drivers/misc/sgi-gru/gruprocfs.c b/drivers/misc/sgi-gru/gruprocfs.c > index 950dbe9..797d796 100644 > --- a/drivers/misc/sgi-gru/gruprocfs.c > +++ b/drivers/misc/sgi-gru/gruprocfs.c > @@ -355,7 +355,7 @@ static void delete_proc_files(void) > for (p = proc_files; p->name; p++) > if (p->entry) > remove_proc_entry(p->name, proc_gru); > - remove_proc_entry("gru", proc_gru->parent); > + proc_remove(proc_gru); > } > } > > diff --git a/drivers/of/base.c b/drivers/of/base.c > index 321d3ef..9c70436 100644 > --- a/drivers/of/base.c > +++ b/drivers/of/base.c > @@ -1452,16 +1452,7 @@ int of_attach_node(struct device_node *np) > #ifdef CONFIG_PROC_DEVICETREE > static void of_remove_proc_dt_entry(struct device_node *dn) > { > - struct device_node *parent = dn->parent; > - struct property *prop = dn->properties; > - > - while (prop) { > - remove_proc_entry(prop->name, dn->pde); > - prop = prop->next; > - } > - > - if (dn->pde) > - remove_proc_entry(dn->pde->name, parent->pde); > + proc_remove(dn->pde); > } > #else > static void of_remove_proc_dt_entry(struct device_node *dn) > diff --git a/drivers/pci/proc.c b/drivers/pci/proc.c > index 7cde7c1..0812608 100644 > --- a/drivers/pci/proc.c > +++ b/drivers/pci/proc.c > @@ -427,20 +427,14 @@ int pci_proc_attach_device(struct pci_dev *dev) > > int pci_proc_detach_device(struct pci_dev *dev) > { > - struct proc_dir_entry *e; > - > - if ((e = dev->procent)) { > - remove_proc_entry(e->name, dev->bus->procdir); > - dev->procent = NULL; > - } > + proc_remove(dev->procent); > + dev->procent = NULL; > return 0; > } > > int pci_proc_detach_bus(struct pci_bus* bus) > { > - struct proc_dir_entry *de = bus->procdir; > - if (de) > - remove_proc_entry(de->name, proc_bus_pci_dir); > + proc_remove(bus->procdir); > return 0; > } > > diff --git a/fs/proc/generic.c b/fs/proc/generic.c > index 2c6d6be..4b48033 100644 > --- a/fs/proc/generic.c > +++ b/fs/proc/generic.c > @@ -630,3 +630,10 @@ const char *get_proc_name(const struct proc_dir_entry *de) > return de->name; > } > EXPORT_SYMBOL_GPL(get_proc_name); > + > +void proc_remove(struct proc_dir_entry *de) > +{ > + if (de) > + remove_proc_subtree(de->name, de->parent); > +} > +EXPORT_SYMBOL(proc_remove); > diff --git a/fs/proc/vmcore.c b/fs/proc/vmcore.c > index 38edddc..17f7e08 100644 > --- a/fs/proc/vmcore.c > +++ b/fs/proc/vmcore.c > @@ -699,7 +699,7 @@ void vmcore_cleanup(void) > struct list_head *pos, *next; > > if (proc_vmcore) { > - remove_proc_entry(proc_vmcore->name, proc_vmcore->parent); > + proc_remove(proc_vmcore); > proc_vmcore = NULL; > } > > diff --git a/include/linux/proc_fs.h b/include/linux/proc_fs.h > index 7abc72a..b8949b1 100644 > --- a/include/linux/proc_fs.h > +++ b/include/linux/proc_fs.h > @@ -67,6 +67,7 @@ struct proc_dir_entry *proc_create_data(const char *name, umode_t mode, > struct proc_dir_entry *parent, > const struct file_operations *proc_fops, > void *data); > +extern void proc_remove(struct proc_dir_entry *); > extern void remove_proc_entry(const char *name, struct proc_dir_entry *parent); > extern int remove_proc_subtree(const char *name, struct proc_dir_entry *parent); > > @@ -103,6 +104,7 @@ static inline struct proc_dir_entry *proc_create_data(const char *name, > { > return NULL; > } > +static inline void proc_remove(struct proc_dir_entry *de) {} > #define remove_proc_entry(name, parent) do {} while (0) > #define remove_proc_subtree(name, parent) do {} while (0) > > diff --git a/kernel/irq/proc.c b/kernel/irq/proc.c > index d59ae37..19ed5c4 100644 > --- a/kernel/irq/proc.c > +++ b/kernel/irq/proc.c > @@ -366,11 +366,7 @@ void unregister_irq_proc(unsigned int irq, struct irq_desc *desc) > > void unregister_handler_proc(unsigned int irq, struct irqaction *action) > { > - if (action->dir) { > - struct irq_desc *desc = irq_to_desc(irq); > - > - remove_proc_entry(action->dir->name, desc->dir); > - } > + proc_remove(action->dir); > } > > static void register_default_affinity_proc(void) > diff --git a/net/8021q/vlanproc.c b/net/8021q/vlanproc.c > index 959ddbb..1d0e8921 100644 > --- a/net/8021q/vlanproc.c > +++ b/net/8021q/vlanproc.c > @@ -184,14 +184,9 @@ int vlan_proc_add_dev(struct net_device *vlandev) > */ > int vlan_proc_rem_dev(struct net_device *vlandev) > { > - struct vlan_net *vn = net_generic(dev_net(vlandev), vlan_net_id); > - > /** NOTE: This will consume the memory pointed to by dent, it seems. */ > - if (vlan_dev_priv(vlandev)->dent) { > - remove_proc_entry(vlan_dev_priv(vlandev)->dent->name, > - vn->proc_vlan_dir); > - vlan_dev_priv(vlandev)->dent = NULL; > - } > + proc_remove(vlan_dev_priv(vlandev)->dent); > + vlan_dev_priv(vlandev)->dent = NULL; > return 0; > } > > diff --git a/net/core/pktgen.c b/net/core/pktgen.c > index f6af4fe..6c41e97 100644 > --- a/net/core/pktgen.c > +++ b/net/core/pktgen.c > @@ -1904,7 +1904,7 @@ static void pktgen_change_name(const struct pktgen_net *pn, struct net_device *d > if (pkt_dev->odev != dev) > continue; > > - remove_proc_entry(pkt_dev->entry->name, pn->proc_dir); > + proc_remove(pkt_dev->entry); > > pkt_dev->entry = proc_create_data(dev->name, 0600, > pn->proc_dir, > @@ -3576,8 +3576,6 @@ static void _rem_dev_from_if_list(struct pktgen_thread *t, > static int pktgen_remove_device(struct pktgen_thread *t, > struct pktgen_dev *pkt_dev) > { > - struct pktgen_net *pn = t->net; > - > pr_debug("remove_device pkt_dev=%p\n", pkt_dev); > > if (pkt_dev->running) { > @@ -3597,7 +3595,7 @@ static int pktgen_remove_device(struct pktgen_thread *t, > _rem_dev_from_if_list(t, pkt_dev); > > if (pkt_dev->entry) > - remove_proc_entry(pkt_dev->entry->name, pn->proc_dir); > + proc_remove(pkt_dev->entry); > > #ifdef CONFIG_XFRM > free_SAs(pkt_dev); > diff --git a/net/ipv4/netfilter/ipt_CLUSTERIP.c b/net/ipv4/netfilter/ipt_CLUSTERIP.c > index e4738fe..0b732ef 100644 > --- a/net/ipv4/netfilter/ipt_CLUSTERIP.c > +++ b/net/ipv4/netfilter/ipt_CLUSTERIP.c > @@ -105,7 +105,7 @@ clusterip_config_entry_put(struct clusterip_config *c) > * functions are also incrementing the refcount on their own, > * so it's safe to remove the entry even if it's in use. */ > #ifdef CONFIG_PROC_FS > - remove_proc_entry(c->pde->name, c->pde->parent); > + proc_remove(c->pde); > #endif > return; > } > @@ -736,7 +736,7 @@ static void __exit clusterip_tg_exit(void) > { > pr_info("ClusterIP Version %s unloading\n", CLUSTERIP_VERSION); > #ifdef CONFIG_PROC_FS > - remove_proc_entry(clusterip_procdir->name, clusterip_procdir->parent); > + proc_remove(clusterip_procdir); > #endif > nf_unregister_hook(&cip_arp_ops); > xt_unregister_target(&clusterip_tg_reg); > diff --git a/net/ipv6/proc.c b/net/ipv6/proc.c > index 7ea6e18..537d9ee 100644 > --- a/net/ipv6/proc.c > +++ b/net/ipv6/proc.c > @@ -287,8 +287,7 @@ int snmp6_unregister_dev(struct inet6_dev *idev) > return -ENOENT; > if (!idev->stats.proc_dir_entry) > return -EINVAL; > - remove_proc_entry(idev->stats.proc_dir_entry->name, > - net->mib.proc_net_devsnmp6); > + proc_remove(idev->stats.proc_dir_entry); > idev->stats.proc_dir_entry = NULL; > return 0; > } > diff --git a/net/netfilter/xt_hashlimit.c b/net/netfilter/xt_hashlimit.c > index d2fd53b..40002ef 100644 > --- a/net/netfilter/xt_hashlimit.c > +++ b/net/netfilter/xt_hashlimit.c > @@ -330,7 +330,7 @@ static void htable_destroy(struct xt_hashlimit_htable *hinfo) > parent = hashlimit_net->ip6t_hashlimit; > > if(parent != NULL) > - remove_proc_entry(hinfo->pde->name, parent); > + proc_remove(hinfo->pde); > > htable_selective_cleanup(hinfo, select_all); > vfree(hinfo); > @@ -887,7 +887,7 @@ static void __net_exit hashlimit_proc_net_exit(struct net *net) > pde = hashlimit_net->ip6t_hashlimit; > > hlist_for_each_entry(hinfo, &hashlimit_net->htables, node) > - remove_proc_entry(hinfo->pde->name, pde); > + proc_remove(hinfo->pde); > > hashlimit_net->ipt_hashlimit = NULL; > hashlimit_net->ip6t_hashlimit = NULL; > diff --git a/sound/core/info.c b/sound/core/info.c > index c7f41c3..3c9bd6b 100644 > --- a/sound/core/info.c > +++ b/sound/core/info.c > @@ -153,13 +153,6 @@ EXPORT_SYMBOL(snd_seq_root); > struct snd_info_entry *snd_oss_root; > #endif > > -static void snd_remove_proc_entry(struct proc_dir_entry *parent, > - struct proc_dir_entry *de) > -{ > - if (de) > - remove_proc_entry(de->name, parent); > -} > - > static loff_t snd_info_entry_llseek(struct file *file, loff_t offset, int orig) > { > struct snd_info_private_data *data; > @@ -580,7 +573,7 @@ int __exit snd_info_done(void) > #ifdef CONFIG_SND_OSSEMUL > snd_info_free_entry(snd_oss_root); > #endif > - snd_remove_proc_entry(NULL, snd_proc_root); > + proc_remove(snd_proc_root); > } > return 0; > } > @@ -642,7 +635,7 @@ void snd_info_card_id_change(struct snd_card *card) > { > mutex_lock(&info_mutex); > if (card->proc_root_link) { > - snd_remove_proc_entry(snd_proc_root, card->proc_root_link); > + proc_remove(card->proc_root_link); > card->proc_root_link = NULL; > } > if (strcmp(card->id, card->proc_root->name)) > @@ -661,10 +654,8 @@ void snd_info_card_disconnect(struct snd_card *card) > if (!card) > return; > mutex_lock(&info_mutex); > - if (card->proc_root_link) { > - snd_remove_proc_entry(snd_proc_root, card->proc_root_link); > - card->proc_root_link = NULL; > - } > + proc_remove(card->proc_root_link); > + card->proc_root_link = NULL; > if (card->proc_root) > snd_info_disconnect(card->proc_root); > mutex_unlock(&info_mutex); > @@ -856,7 +847,7 @@ static void snd_info_disconnect(struct snd_info_entry *entry) > list_del_init(&entry->list); > root = entry->parent == NULL ? snd_proc_root : entry->parent->p; > snd_BUG_ON(!root); > - snd_remove_proc_entry(root, entry->p); > + proc_remove(entry->p); > entry->p = NULL; > } > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-pci" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html

2 1

[alsa-devel] [PATCH 00/17] ASoC: Towards a generic dmaengine PCM driver
by Lars-Peter Clausen 18 Apr '13

18 Apr '13

Hi, This series adds a (almost) generic dmaengine PCM driver for ASoC. The tegra, spear, omap, imx, mxs, ep93xx and ux500 platforms are converted to use the generic driver instead of their own custom implementation. The driver builds on top of the dmaengine PCM library and adds the missing bits. The driver was primarily built for handling devicetree based platforms, but the series also adds support for compat platforms which don't use devicetree yet or have to support non-devicetree instantiation for backwards compatibility. Since non of the converted platforms have devicetree support yet (at least not using the generic DMA bindings) this series doesn't enable devicetree support for them either. But enabling support for them is very easy and can be done by removing the SND_DMAENGINE_PCM_NO_DT flag for these platforms. For platforms using devicetree the DMA channels are requested early on before the ASoC platform driver is registered. This allows us, at least in theory, to use probe deferral if the DMA driver hasn't been probed yet. Unfortunately of_dma_request_slave_channel either returns a valid channel or NULL, so we can't distinguish between the case where there is no DMA channel assigned and the case where the DMA driver hasn't been probed yet. I plan to submit a patch which lets of_dma_request_slave_channel return a ERR_PTR, but for now the generic PCM dmaengine driver assumes that the DMA driver will always be probed before it. For the compat case the DMA channel gets requested from within the PCM driver's pcm_new callback. This requires some minor changes to some of the DAI drivers moving the initialization of the DAI DMA data pointer from the DAI startup callback to the DAI probe callback. Unfortunately the driver is not fully generic yet and we still need some platform specific data to make it work, but the plan is to eventually make the driver fully generic in the future. The platform specific bits currently are: * The snd_pcm_hardware struct which contains device specific parameters like the maximum period size and maximum number of periods as well as whether the driver supports pause and resume. The idea is to get rid of the per platform snd_pcm_hardware struct by extending the dmaengine API to a point where we can query all necessary information from the dmaengine driver to fill in the snd_pcm_hardware struct at runtime. * The size of the pre-allocated audio buffer. All platforms seem use different sizes and since some of them might be sensitive to changes to the buffer size this series keeps the original size for now. But we should probably try to decide on a sane default for this and maybe allow ASoC board drivers to overwrite it somehow. * A callback to fill in the dma_slave_config. Currently this is either NULL (for platforms which don't need to configure the DMA channel at runtime) or uses the generic snd_dmaengine_pcm_prepare_slave_config(), which uses snd_hwparams_to_dma_slave_config() and snd_dmaengine_pcm_set_config_from_dai_data(). The later requires the platform to use the generic DMA DAI data struct though, hopefully we'll be able to convert all platforms to use this generic struct, and once that's done we can always use snd_dmaengine_pcm_prepare_slave_config() unconditionally. * For platforms not using devicetree platforms a callback to request the dma channel. This can either be a completely custom function which returns the channel or a filter function which will be passed to dma_request_channel. The route forward here probably is to let the DAI driver provide the filter function or implement a table based DMA channel lookup like we have for other subsystems (like clk or regulator) where the board file provides a mapping based on the device name. Platforms which use the dmaengine pcm library but are not converted to the generic driver yet are atmel and mmp. mmp use a custom genalloc based memory allocator, we may add support for this to the generic driver at a later point. The atmel PCM driver is very interlocked with atmel SSC DAI driver, it'll probably require some serious refactoring before it can use the generic dmaengine PCM driver. The driver uses snd_pcm_lib_preallocate_pages to preallocate the memory buffer. snd_pcm_lib_preallocate_pages internally uses dma_alloc_coherent, while most of the platforms converted in this series use dma_alloc_writecombine. Laxman Dewangan submitted a patch[1] a couple of months ago to add support for dma_alloc_writecombine to snd_pcm_lib_preallocate_pages, I always thought the patch had been merged, but it looks like I was mistaken. So if there are concerns about performance regression due to using coherent instead of writecombine mappings we'll have to postpone converting these platforms until Laxman's patch has been merged. The compat path of this series hasn't been runtime tested yet and neither have the platforms which are converted in this series. I probably broke one or two of them by accident, so please test the patches. Thanks, - Lars [1] https://lkml.org/lkml/2012/7/2/94 Lars-Peter Clausen (17): ASoC: dmaengine-pcm: Make requesting the DMA channel at PCM open optional ASoC: Add snd_soc_{add,remove}_platform ASoC: Add a generic dmaengine_pcm driver ASoC: dmaengine_pcm: Add support for compat platforms ASoC: tegra: Use generic dmaengine PCM ASoC: spear: Setup dma data in DAI probe ASoC: spear: Use generic dmaengine PCM ASoC: dmaengine-pcm: Add support for platforms which can't report residue ASoC: mxs: Setup dma data in DAI probe ASoC: mxs: Use generic dmaengine PCM ASoC: imx: Setup dma data in DAI probe ASoC: imx: Use generic dmaengine PCM ASoC: omap: Setup dma data in DAI probe ASoC: omap: Use generic dmaengine PCM ASoC: ep93xx: Setup dma data in DAI probe ASoC: ep93xx: Use generic dmaengine PCM ASoC: ux500: Use generic dmaengine PCM include/sound/dmaengine_pcm.h | 65 +++++++- include/sound/soc.h | 4 + sound/soc/Kconfig | 4 + sound/soc/Makefile | 4 + sound/soc/atmel/atmel-pcm-dma.c | 6 +- sound/soc/cirrus/Kconfig | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 14 +- sound/soc/cirrus/ep93xx-i2s.c | 14 +- sound/soc/cirrus/ep93xx-pcm.c | 137 ++-------------- sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl_ssi.c | 19 ++- sound/soc/fsl/imx-pcm-dma.c | 76 ++------- sound/soc/fsl/imx-pcm.c | 6 +- sound/soc/fsl/imx-pcm.h | 5 + sound/soc/fsl/imx-ssi.c | 22 +-- sound/soc/mxs/Kconfig | 2 +- sound/soc/mxs/mxs-pcm.c | 135 ++-------------- sound/soc/mxs/mxs-saif.c | 3 +- sound/soc/omap/Kconfig | 2 +- sound/soc/omap/omap-dmic.c | 4 +- sound/soc/omap/omap-hdmi.c | 12 +- sound/soc/omap/omap-mcbsp.c | 6 +- sound/soc/omap/omap-mcpdm.c | 7 +- sound/soc/omap/omap-pcm.c | 186 ++-------------------- sound/soc/pxa/mmp-pcm.c | 5 +- sound/soc/soc-core.c | 85 +++++++--- sound/soc/soc-dmaengine-pcm.c | 82 +++++++--- sound/soc/soc-generic-dmaengine-pcm.c | 284 ++++++++++++++++++++++++++++++++++ sound/soc/spear/spdif_in.c | 12 +- sound/soc/spear/spdif_out.c | 7 +- sound/soc/spear/spear_pcm.c | 151 ++---------------- sound/soc/tegra/Kconfig | 2 +- sound/soc/tegra/tegra_pcm.c | 171 ++------------------ sound/soc/ux500/Kconfig | 2 +- sound/soc/ux500/ux500_pcm.c | 159 ++----------------- 35 files changed, 651 insertions(+), 1046 deletions(-) create mode 100644 sound/soc/soc-generic-dmaengine-pcm.c -- 1.8.0

7 48

Re: [alsa-devel] [PATCH 04/28] proc: Supply PDE attribute setting accessor functions [RFC]
by Bjorn Helgaas 18 Apr '13

18 Apr '13

On Tue, Apr 16, 2013 at 12:26 PM, David Howells <dhowells(a)redhat.com> wrote: > Supply accessor functions to set attributes in proc_dir_entry structs. > > The following are supplied: proc_set_size() and proc_set_user(). > > Signed-off-by: David Howells <dhowells(a)redhat.com> > cc: linuxppc-dev(a)lists.ozlabs.org > cc: linux-media(a)vger.kernel.org > cc: netdev(a)vger.kernel.org > cc: linux-wireless(a)vger.kernel.org > cc: linux-pci(a)vger.kernel.org > cc: netfilter-devel(a)vger.kernel.org > cc: alsa-devel(a)alsa-project.org > --- > > arch/powerpc/kernel/proc_powerpc.c | 2 +- > arch/powerpc/platforms/pseries/reconfig.c | 2 +- > drivers/media/pci/ttpci/av7110_ir.c | 2 +- > drivers/net/irda/vlsi_ir.c | 2 +- > drivers/net/wireless/airo.c | 34 +++++++++-------------------- > drivers/pci/proc.c | 2 +- For the drivers/pci part: Acked-by: Bjorn Helgaas <bhelgaas(a)google.com> > fs/proc/generic.c | 13 +++++++++++ > include/linux/proc_fs.h | 5 ++++ > kernel/configs.c | 2 +- > kernel/profile.c | 2 +- > net/netfilter/xt_recent.c | 3 +-- > sound/core/info.c | 2 +- > 12 files changed, 38 insertions(+), 33 deletions(-) > > diff --git a/arch/powerpc/kernel/proc_powerpc.c b/arch/powerpc/kernel/proc_powerpc.c > index 41d8ee9..feb8580 100644 > --- a/arch/powerpc/kernel/proc_powerpc.c > +++ b/arch/powerpc/kernel/proc_powerpc.c > @@ -83,7 +83,7 @@ static int __init proc_ppc64_init(void) > &page_map_fops, vdso_data); > if (!pde) > return 1; > - pde->size = PAGE_SIZE; > + proc_set_size(pde, PAGE_SIZE); > > return 0; > } > diff --git a/arch/powerpc/platforms/pseries/reconfig.c b/arch/powerpc/platforms/pseries/reconfig.c > index d6491bd..f93cdf5 100644 > --- a/arch/powerpc/platforms/pseries/reconfig.c > +++ b/arch/powerpc/platforms/pseries/reconfig.c > @@ -452,7 +452,7 @@ static int proc_ppc64_create_ofdt(void) > > ent = proc_create("powerpc/ofdt", S_IWUSR, NULL, &ofdt_fops); > if (ent) > - ent->size = 0; > + proc_set_size(ent, 0); > > return 0; > } > diff --git a/drivers/media/pci/ttpci/av7110_ir.c b/drivers/media/pci/ttpci/av7110_ir.c > index eb82286..0e763a7 100644 > --- a/drivers/media/pci/ttpci/av7110_ir.c > +++ b/drivers/media/pci/ttpci/av7110_ir.c > @@ -375,7 +375,7 @@ int av7110_ir_init(struct av7110 *av7110) > if (av_cnt == 1) { > e = proc_create("av7110_ir", S_IWUSR, NULL, &av7110_ir_proc_fops); > if (e) > - e->size = 4 + 256 * sizeof(u16); > + proc_set_size(e, 4 + 256 * sizeof(u16)); > } > > tasklet_init(&av7110->ir.ir_tasklet, av7110_emit_key, (unsigned long) &av7110->ir); > diff --git a/drivers/net/irda/vlsi_ir.c b/drivers/net/irda/vlsi_ir.c > index e22cd4e..5f47584 100644 > --- a/drivers/net/irda/vlsi_ir.c > +++ b/drivers/net/irda/vlsi_ir.c > @@ -1678,7 +1678,7 @@ vlsi_irda_probe(struct pci_dev *pdev, const struct pci_device_id *id) > IRDA_WARNING("%s: failed to create proc entry\n", > __func__); > } else { > - ent->size = 0; > + proc_set_size(ent, 0); > } > idev->proc_entry = ent; > } > diff --git a/drivers/net/wireless/airo.c b/drivers/net/wireless/airo.c > index 66e398d..21d0233 100644 > --- a/drivers/net/wireless/airo.c > +++ b/drivers/net/wireless/airo.c > @@ -4507,73 +4507,63 @@ static int setup_proc_entry( struct net_device *dev, > airo_entry); > if (!apriv->proc_entry) > goto fail; > - apriv->proc_entry->uid = proc_kuid; > - apriv->proc_entry->gid = proc_kgid; > + proc_set_user(apriv->proc_entry, proc_kuid, proc_kgid); > > /* Setup the StatsDelta */ > entry = proc_create_data("StatsDelta", S_IRUGO & proc_perm, > apriv->proc_entry, &proc_statsdelta_ops, dev); > if (!entry) > goto fail_stats_delta; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Stats */ > entry = proc_create_data("Stats", S_IRUGO & proc_perm, > apriv->proc_entry, &proc_stats_ops, dev); > if (!entry) > goto fail_stats; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Status */ > entry = proc_create_data("Status", S_IRUGO & proc_perm, > apriv->proc_entry, &proc_status_ops, dev); > if (!entry) > goto fail_status; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Config */ > entry = proc_create_data("Config", proc_perm, > apriv->proc_entry, &proc_config_ops, dev); > if (!entry) > goto fail_config; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the SSID */ > entry = proc_create_data("SSID", proc_perm, > apriv->proc_entry, &proc_SSID_ops, dev); > if (!entry) > goto fail_ssid; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the APList */ > entry = proc_create_data("APList", proc_perm, > apriv->proc_entry, &proc_APList_ops, dev); > if (!entry) > goto fail_aplist; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the BSSList */ > entry = proc_create_data("BSSList", proc_perm, > apriv->proc_entry, &proc_BSSList_ops, dev); > if (!entry) > goto fail_bsslist; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the WepKey */ > entry = proc_create_data("WepKey", proc_perm, > apriv->proc_entry, &proc_wepkey_ops, dev); > if (!entry) > goto fail_wepkey; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > - > + proc_set_user(entry, proc_kuid, proc_kgid); > return 0; > > fail_wepkey: > @@ -5695,10 +5685,8 @@ static int __init airo_init_module( void ) > > airo_entry = proc_mkdir_mode("driver/aironet", airo_perm, NULL); > > - if (airo_entry) { > - airo_entry->uid = proc_kuid; > - airo_entry->gid = proc_kgid; > - } > + if (airo_entry) > + proc_set_user(airo_entry, proc_kuid, proc_kgid); > > for (i = 0; i < 4 && io[i] && irq[i]; i++) { > airo_print_info("", "Trying to configure ISA adapter at irq=%d " > diff --git a/drivers/pci/proc.c b/drivers/pci/proc.c > index 12e4fb5..7cde7c1 100644 > --- a/drivers/pci/proc.c > +++ b/drivers/pci/proc.c > @@ -419,7 +419,7 @@ int pci_proc_attach_device(struct pci_dev *dev) > &proc_bus_pci_operations, dev); > if (!e) > return -ENOMEM; > - e->size = dev->cfg_size; > + proc_set_size(e, dev->cfg_size); > dev->procent = e; > > return 0; > diff --git a/fs/proc/generic.c b/fs/proc/generic.c > index 1c07cad..5f6f6c3 100644 > --- a/fs/proc/generic.c > +++ b/fs/proc/generic.c > @@ -498,6 +498,19 @@ out: > return NULL; > } > EXPORT_SYMBOL(proc_create_data); > + > +void proc_set_size(struct proc_dir_entry *de, loff_t size) > +{ > + de->size = size; > +} > +EXPORT_SYMBOL(proc_set_size); > + > +void proc_set_user(struct proc_dir_entry *de, kuid_t uid, kgid_t gid) > +{ > + de->uid = uid; > + de->gid = gid; > +} > +EXPORT_SYMBOL(proc_set_user); > > static void free_proc_entry(struct proc_dir_entry *de) > { > diff --git a/include/linux/proc_fs.h b/include/linux/proc_fs.h > index 805edac..28a4d7e 100644 > --- a/include/linux/proc_fs.h > +++ b/include/linux/proc_fs.h > @@ -130,6 +130,9 @@ static inline struct proc_dir_entry *proc_create(const char *name, umode_t mode, > extern struct proc_dir_entry *proc_net_mkdir(struct net *net, const char *name, > struct proc_dir_entry *parent); > > +extern void proc_set_size(struct proc_dir_entry *, loff_t); > +extern void proc_set_user(struct proc_dir_entry *, kuid_t, kgid_t); > + > extern struct file *proc_ns_fget(int fd); > extern bool proc_ns_inode(struct inode *inode); > > @@ -158,6 +161,8 @@ static inline struct proc_dir_entry *proc_mkdir(const char *name, > struct proc_dir_entry *parent) {return NULL;} > static inline struct proc_dir_entry *proc_mkdir_mode(const char *name, > umode_t mode, struct proc_dir_entry *parent) { return NULL; } > +static inline void proc_set_size(struct proc_dir_entry *de, loff_t size) {} > +static inline void proc_set_user(struct proc_dir_entry *de, kuid_t uid, kgid_t gid) {} > > struct tty_driver; > static inline void proc_tty_register_driver(struct tty_driver *driver) {}; > diff --git a/kernel/configs.c b/kernel/configs.c > index 42e8fa0..c18b1f1 100644 > --- a/kernel/configs.c > +++ b/kernel/configs.c > @@ -79,7 +79,7 @@ static int __init ikconfig_init(void) > if (!entry) > return -ENOMEM; > > - entry->size = kernel_config_data_size; > + proc_set_size(entry, kernel_config_data_size); > > return 0; > } > diff --git a/kernel/profile.c b/kernel/profile.c > index 524ce5e..0bf4007 100644 > --- a/kernel/profile.c > +++ b/kernel/profile.c > @@ -600,7 +600,7 @@ int __ref create_proc_profile(void) /* false positive from hotcpu_notifier */ > NULL, &proc_profile_operations); > if (!entry) > return 0; > - entry->size = (1+prof_len) * sizeof(atomic_t); > + proc_set_size(entry, (1 + prof_len) * sizeof(atomic_t)); > hotcpu_notifier(profile_cpu_callback, 0); > return 0; > } > diff --git a/net/netfilter/xt_recent.c b/net/netfilter/xt_recent.c > index 3db2d38..1e657cf 100644 > --- a/net/netfilter/xt_recent.c > +++ b/net/netfilter/xt_recent.c > @@ -401,8 +401,7 @@ static int recent_mt_check(const struct xt_mtchk_param *par, > ret = -ENOMEM; > goto out; > } > - pde->uid = uid; > - pde->gid = gid; > + proc_set_user(pde, uid, gid); > #endif > spin_lock_bh(&recent_lock); > list_add_tail(&t->list, &recent_net->tables); > diff --git a/sound/core/info.c b/sound/core/info.c > index 3aa8864..c7f41c3 100644 > --- a/sound/core/info.c > +++ b/sound/core/info.c > @@ -970,7 +970,7 @@ int snd_info_register(struct snd_info_entry * entry) > mutex_unlock(&info_mutex); > return -ENOMEM; > } > - p->size = entry->size; > + proc_set_size(p, entry->size); > } > entry->p = p; > if (entry->parent) > > -- > To unsubscribe from this list: send the line "unsubscribe linux-pci" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

[alsa-devel] [PATCH v4 00/10] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 18 Apr '13

18 Apr '13

Hi, This series adds DT support for phycore-ac97. It is now based on the pcm dma cleanups from Lars. Beside ac97 support, the series adds fsl-ssi imx-pcm-fiq and generic DMA binding handling. Regards, Markus Changes in v4: - phycore-ac97 uses the iomux functions based on audmux phandle now. - fsl-ssi imx-pcm-fiq got some new comments about problems with imx-ssi and incompatible codecs Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (10): ASoC: phycore-ac97: Add DT support ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ARM: imx: Export ac97 reset functions ASoC: fsl-ssi: imx ac97 support ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97 ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 10 ++ Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt | 14 ++ arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl_ssi.c | 512 ++++++++++++++++++++++++++++++++++++++++++++++++---------- sound/soc/fsl/imx-pcm-dma.c | 4 +- sound/soc/fsl/imx-pcm-fiq.c | 16 +- sound/soc/fsl/imx-pcm.h | 9 ++ sound/soc/fsl/imx-ssi.c | 7 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/phycore-ac97.c | 166 ++++++++++++++++--- 12 files changed, 633 insertions(+), 122 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (88%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt

3 13

[alsa-devel] More ASoC updates for v3.10
by Mark Brown 18 Apr '13

18 Apr '13

The following changes since commit b02e48f2f8755e51fe8ce246988a29bd96f6b85e: MAINTAINERS: Update e-mail address (2013-04-12 11:53:41 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next for you to fetch changes up to 24568ea4bef5ab8106206eddf5512434421c00ed: Merge remote-tracking branch 'asoc/topic/max98088' into asoc-next (2013-04-18 15:05:35 +0100) There's a bunch of extra stuff in the diffstat/log since your for-next is further behind Linus' tree than one of my branches. ---------------------------------------------------------------- ASoC: More updates for v3.10 The main additional change here is Lars-Peter's DMA work plus the platform conversions which have been tested - getting this in mainline will make life easier for development after the merge window. These factor a large chunk of code out of the drivers for the platforms using dmaengine, greatly simplifying development. ---------------------------------------------------------------- Al Viro (3): ecryptfs: close rmmod race procfs: add proc_remove_subtree() palinfo fixes Alex Williamson (1): vfio-pci: Fix possible integer overflow Alexandre Belloni (1): gpio: pca953x: fix irq_domain_add_simple usage Alexey Khoroshilov (1): tty: mxser: fix cycle termination condition in mxser_probe() and mxser_module_init() Alexey Pelykh (1): OMAP/serial: Revert bad fix of Rx FIFO threshold granularity Andrea Arcangeli (2): x86/mm/cpa: Convert noop to functional fix x86/mm/cpa/selftest: Fix false positive in CPA self test Andrey Vagin (1): mnt: release locks on error path in do_loopback Arnd Bergmann (5): block: avoid using uninitialized value in from queue_var_store Merge tag 'mxs-fixes-3.9-4' of git://git.linaro.org/people/shawnguo/linux-2.6 into fixes Merge tag 'imx-fixes-3.9-5' of git://git.linaro.org/people/shawnguo/linux-2.6 into fixes Merge tag 'samsung-fixes-for-v3.9' of git://git.kernel.org/.../kgene/linux-samsung into fixes Merge tag 'mvebu_fixes_for_v3.9_round3' of git://git.infradead.org/users/jcooper/linux into fixes Artem Savkov (1): cfg80211: sched_scan_mtx lock in cfg80211_conn_work() Arun Easi (1): [SCSI] qla2xxx: Fix crash during firmware dump procedure. Asai Thambi S P (3): mtip32xx: recovery from command timeout mtip32xx: return 0 from pci probe in case of rebuild mtip32xx: Add debugfs entry device_status Asias He (7): tcm_vhost: Use ACCESS_ONCE for vs->vs_tpg[target] access tcm_vhost: Use vq->private_data to indicate if the endpoint is setup tcm_vhost: Initialize vq->last_used_idx when set endpoint tcm_vhost: Remove double check of response tcm_vhost: Fix tv_cmd leak in vhost_scsi_handle_vq tcm_vhost: Add vhost_scsi_send_bad_target() helper tcm_vhost: Send bad target to guest when cmd fails Bing Zhao (1): mwifiex: complete last internal scan Boris Ostrovsky (2): x86, mm: Patch out arch_flush_lazy_mmu_mode() when running on bare metal x86/mm: Flush lazy MMU when DEBUG_PAGEALLOC is set Brian King (1): [SCSI] ibmvscsi: Fix slave_configure deadlock Calvin Owens (1): drm/nouveau: fix unconditional return waiting on memory Chen Gang (3): perf: Fix strncpy() use, always make sure it's NUL terminated perf: Fix strncpy() use, use strlcpy() instead of strncpy() ftrace: Fix strncpy() use, use strlcpy() instead of strncpy() Chris Metcalf (1): tile: comment assumption about __insn_mtspr for <asm/irqflags.h> Christian Ruppert (1): ARC: Add implicit compiler barrier to raw_local_irq* functions Christoph Paasch (1): ipv6/tcp: Stop processing ICMPv6 redirect messages Christopher Harvey (1): drm/mgag200: Index 24 in extended CRTC registers is 24 in hex, not decimal. Chuck Lever (1): SUNRPC: Remove extra xprt_put() Daniel Vetter (1): drm/fb-helper: Fix locking in drm_fb_helper_hotplug_event Dave Airlie (2): Merge branch 'drm-nouveau-fixes-3.9' of git://anongit.freedesktop.org/git/nouveau/linux-2.6 into drm-fixes udl: handle EDID failure properly. Dave Hansen (1): x86-32: Fix possible incomplete TLB invalidate with PAE pagetables David S. Miller (3): Merge branch 'infoleaks' Merge branch 'wireless' Merge branch 'fixes-for-3.9' of git://gitorious.org/linux-can/linux-can David Woodhouse (1): libata: fix DMA to stack in reading devslp_timing parameters Dirk Behme (1): ARM i.MX6: Fix ldb_di clock selection Dirk Brandewie (1): cpufreq / intel_pstate: Set timer timeout correctly Dmitry Tarnyagin (1): remoteproc/ste: fix memory leak on shutdown Dylan Reid (1): ASoC: max98088: Fix logging of hardware revision. Eldad Zack (1): ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_* Eric Dumazet (1): selinux: add a skb_owned_by() hook Franky Lin (1): brcmfmac: do not proceed if fail to download nvram to dongle Gabor Juhos (1): rt2x00: rt2x00pci: fix build error on Ralink RT3x5x SoCs Greg Ungerer (1): m68k: define a local gpio_request_one() function Gregory CLEMENT (1): arm: mvebu: Fix the irq map function in SMP mode Guennadi Liakhovetski (1): tty: Documentation: fix a path in a DocBook template Guenter Roeck (1): misc/vmw_vmci: Add dependency on CONFIG_NET Haiyang Zhang (2): hyperv: Fix a kernel warning from netvsc_linkstatus_callback() hyperv: Fix RNDIS send_completion code path Hante Meuleman (3): brcmfmac: fix tkip mic tx/rx ap swap bug. brcmfmac: fix stopping AP. brcmfmac: fix returning cipher_suite for get_key operation. Huacai Chen (1): PM / reboot: call syscore_shutdown() after disable_nonboot_cpus() Jan Kiszka (1): ftrace: Consistently restore trace function on sysctl enabling Jens Axboe (2): mtip32xx: fix two smatch warnings Revert "loop: cleanup partitions when detaching loop device" Jiri Pirko (2): net: ipv4: reset check_lifetime_work after changing lifetime net: ipv4: fix schedule while atomic bug in check_lifetime() Joe Carnuccio (1): [SCSI] Revert "qla2xxx: Add setting of driver version string for vendor application." Joe Lawrence (1): [SCSI] st: Take additional queue ref in st_probe John Gong (1): [SCSI] libsas: use right function to alloc smp response John W. Linville (2): Merge branch 'for-john' of git://git.kernel.org/.../jberg/mac80211 Merge branch 'master' of git://git.kernel.org/.../linville/wireless into for-davem Josef Bacik (1): Btrfs: make sure nbytes are right after log replay Jussi Kivilinna (1): crypto: gcm - fix assumption that assoc has one segment Keir Fraser (1): xen/events: Handle VIRQ_TIMER before any other hardirq in event loop. Kleber Sacilotto de Souza (1): [SCSI] ipr: fix addition of abort command to HRRQ free queue Konrad Rzeszutek Wilk (1): xen/mmu: On early bootup, flush the TLB when changing RO->RW bits Xen provided pagetables. Krzysztof Mazur (1): [libata] Fix HDIO_DRIVE_* ioctl() Linux 3.9 regression Lars-Peter Clausen (9): ASoC: dmaengine-pcm: Make requesting the DMA channel at PCM open optional ASoC: Add snd_soc_{add, remove}_platform ASoC: Add a generic dmaengine_pcm driver ASoC: dmaengine_pcm: Add support for compat platforms ASoC: tegra: Use generic dmaengine PCM ASoC: dmaengine-pcm: Add support for platforms which can't report residue ASoC: imx: Setup dma data in DAI probe ASoC: imx: Use generic dmaengine PCM ASoC: ux500: Use generic dmaengine PCM Li Fei (1): hwspinlock: fix __hwspin_lock_request error path Linus Torvalds (39): Merge tag 'trace-fixes-3.9-rc6' of git://git.kernel.org/.../rostedt/linux-trace Merge tag 'upstream-linus' of git://git.kernel.org/.../jgarzik/libata-dev Merge tag 'for-linus' of git://git.kernel.org/.../sfr/next-fixes Merge tag 'remoteproc-3.9-fixes' of git://git.kernel.org/.../ohad/remoteproc Merge tag 'hwspinlock-3.9-fix' of git://git.kernel.org/.../ohad/hwspinlock spinlocks and preemption points need to be at least compiler barriers Merge tag 'for-linus-20130409' of git://git.kernel.dk/linux-block Merge tag 'vfio-v3.9-rc7' of git://github.com/awilliam/linux-vfio Merge branch 'for-linus' of git://git.kernel.org/.../viro/vfs Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge git://git.kernel.org/.../herbert/crypto-2.6 Merge tag 'nfs-for-3.9-4' of git://git.linux-nfs.org/projects/trondmy/linux-nfs Merge tag 'nfs-for-3.9-5' of git://git.linux-nfs.org/projects/trondmy/linux-nfs Merge tag 'for-linus' of git://linux-c6x.org/git/projects/linux-c6x-upstreaming Merge git://git.kernel.org/.../davem/net Merge tag 'trace-fixes-3.9-rc-v2' of git://git.kernel.org/.../rostedt/linux-trace Merge tag 'stable/for-linus-3.9-rc6-tag' of git://git.kernel.org/.../konrad/xen Merge tag 'tty-3.9-rc6' of git://git.kernel.org/.../gregkh/tty Merge tag 'char-misc-3.9-rc6' of git://git.kernel.org/.../gregkh/char-misc Merge tag 'arm-soc-fixes' of git://git.kernel.org/.../arm/arm-soc Merge tag 'gpio-fixes-v3.9-rc6' of git://git.kernel.org/.../linusw/linux-gpio Merge tag 'regmap-v3.9-rc6' of git://git.kernel.org/.../broonie/regmap Merge tag 'pm-3.9-rc7' of git://git.kernel.org/.../rafael/linux-pm Merge branch 'fixes' of git://git.infradead.org/users/vkoul/slave-dma Merge tag 'sound-3.9' of git://git.kernel.org/.../tiwai/sound Merge branch 'for-next' of git://git.samba.org/sfrench/cifs-2.6 Merge tag 'scsi-fixes' of git://git.kernel.org/.../jejb/scsi Merge git://git.kernel.org/.../nab/target-pending kobject: fix kset_find_obj() race with concurrent last kobject_put() Add file_ns_capable() helper function for open-time capability checking Merge tag 'trace-fixes-v3.9-rc-v3' of git://git.kernel.org/.../rostedt/linux-trace Merge branch 'for-linus' of git://git.kernel.org/.../mason/linux-btrfs Merge git://www.linux-watchdog.org/linux-watchdog Merge branch 'for-linus' of git://git.kernel.org/.../gerg/m68knommu Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge branch 'perf-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'sched-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Linux 3.9-rc7 Linus Walleij (1): crypto: ux500 - add missing comma Lukasz Dorau (1): [SCSI] libsas: fix handling vacant phy in sas_set_ex_phy() Marek Vasut (1): ARM: mxs: Slow down the I2C clock speed Mark Brown (13): regmap: Back out work buffer fix Merge remote-tracking branch 'asoc/fix/compress' into tmp Merge remote-tracking branch 'asoc/fix/core' into tmp Merge remote-tracking branch 'asoc/fix/samsung' into tmp Merge remote-tracking branch 'asoc/fix/tegra' into tmp Merge remote-tracking branch 'asoc/fix/wm8903' into tmp Merge branch 'topic/core' of git://git.kernel.org/.../broonie/sound into asoc-dma Merge tag 'v3.9-rc7' into asoc-dma Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/max98088' into asoc-next Mark Salter (1): add memory barrier to arch_local_irq_restore Markus Pargmann (3): ARM: clk-imx35: Bugfix iomux clock ARM: imx35 Bugfix admux clock ASoC: fsl-ssi: Add SACNT definitions Mathias Krause (16): atm: update msg_namelen in vcc_recvmsg() ax25: fix info leak via msg_name in ax25_recvmsg() Bluetooth: fix possible info leak in bt_sock_recvmsg() Bluetooth: RFCOMM - Fix missing msg_namelen update in rfcomm_sock_recvmsg() Bluetooth: SCO - Fix missing msg_namelen update in sco_sock_recvmsg() caif: Fix missing msg_namelen update in caif_seqpkt_recvmsg() irda: Fix missing msg_namelen update in irda_recvmsg_dgram() iucv: Fix missing msg_namelen update in iucv_sock_recvmsg() l2tp: fix info leak in l2tp_ip6_recvmsg() llc: Fix missing msg_namelen update in llc_ui_recvmsg() netrom: fix info leak via msg_name in nr_recvmsg() NFC: llcp: fix info leaks via msg_name in llcp_sock_recvmsg() rose: fix info leak via msg_name in rose_recvmsg() tipc: fix info leaks via msg_name in recv_msg/recv_stream VSOCK: vmci - fix possible info leak in vmci_transport_dgram_dequeue() VSOCK: Fix missing msg_namelen update in vsock_stream_recvmsg() Michael Riesch (1): rtnetlink: Call nlmsg_parse() with correct header length Michael Wolf (1): powerpc: pSeries_lpar_hpte_remove fails from Adjunct partition being performed before the ANDCOND test Namhyung Kim (2): tracing: Fix double free when function profile init failed tracing: Fix possible NULL pointer dereferences Neil Horman (1): e100: Add dma mapping error check Nicholas Bellinger (1): target: Fix incorrect fallthrough of ALUA Standby/Offline/Transition CDBs Nicolas Ferre (1): watchdog: Revert the AT91RM9200_WATCHDOG dependency Nigel Roberts (2): ARM: Kirkwood: Fix typo in the definition of ix2-200 rebuild LED Fix GE0/GE1 init on ix2-200 as GE0 has no PHY Padmavathi Venna (1): DMA: PL330: Add check if device tree compatible Paul Moore (1): lsm: add the missing documentation for the security_skb_owned_by() hook Peter Ujfalusi (1): dmaengine: omap-dma: Start DMA without delay for cyclic channels Robert Shade (1): ath9k: Re-enable interrupts after a channel change failure Robert Tivy (1): remoteproc: fix FW_CONFIG typo Roland Stigge (1): MAINTAINERS: Add maintainer for LPC32xx Sachin Prabhu (1): cifs: Allow passwords which begin with a delimitor Samu Kallio (1): x86, mm, paravirt: Fix vmalloc_fault oops during lazy MMU updates Saurav Kashyap (3): [SCSI] qla2xxx: Update copyright dates information in LICENSE.qla2xxx file. [SCSI] qla2xxx: Remove debug code that msleeps for random duration. [SCSI] qla2xxx: Update the driver version to 8.04.00.13-k. Sean Young (1): Revert "tty/8250_pnp: serial port detection regression since v3.7" Shan Hai (2): libata: Use integer return value for atapi_command_packet_set libata: Set max sector to 65535 for Slimtype DVD A DS8A8SH drive Shawn Guo (1): ARM: imx: provide twd clock lookup from device tree Sjur Brændeland (1): remoteproc: fix error path of handle_vdev Sridhar Samudrala (1): sctp: remove 'sridhar' from maintainers list Stanislaw Gruszka (1): sched/cputime: Fix accounting on multi-threaded processes Stas Sergeev (1): ASoC: define playback and capture streams in dummy codec Stefan Raspl (1): qeth: fix qeth_wait_for_threads() deadlock for OSN devices Stephane Eranian (2): perf/x86: Fix uninitialized pt_regs in intel_pmu_drain_bts_buffer() perf: Fix ring_buffer perf_output_space() boundary calculation Steven Rostedt (Red Hat) (3): tracing: Fix race with update_max_tr_single and changing tracers ftrace: Do not call stub functions in control loop ftrace: Move ftrace_filter_lseek out of CONFIG_DYNAMIC_FTRACE section Suleiman Souhlal (1): vfs: Revert spurious fix to spinning prevention in prune_icache_sb Suman Anna (1): remoteproc: fix the error check for idr_alloc Sylwester Nawrocki (2): ARM: S3C24XX: Correct NR_IRQS definition for s3c2440 ARM: S3C24XX: Fix interrupt pending register offset of the EINT controller Takashi Iwai (2): Merge tag 'asoc-v3.9-rc6' of git://git.kernel.org/.../broonie/sound into for-linus Merge tag 'asoc-maintainers-v3.9-rc6' of git://git.kernel.org/.../broonie/sound into for-linus Tejun Heo (1): sched: Convert BUG_ON()s in try_to_wake_up_local() to WARN_ON_ONCE()s Thomas Gleixner (1): sched_clock: Prevent 64bit inatomicity on 32bit systems Tim Gardner (1): rt2x00: rt2x00pci_regbusy_read() - only print register access failure once Trond Myklebust (5): NFSv4: Fix a memory leak in nfs4_discover_server_trunking NFSv4/4.1: Fix bugs in nfs4[01]_walk_client_list SUNRPC: Fix a potential memory leak in rpc_new_client NFSv4: Fix another potential state manager deadlock NFSv4: Doh! Typo in the fix to nfs41_walk_client_list Ursula Braun (1): af_iucv: fix recvmsg by replacing skb_pull() function Wei Yongjun (4): netrom: fix invalid use of sizeof in nr_recvmsg() can: gw: use kmem_cache_free() instead of kfree() [SCSI] lpfc: fix potential NULL pointer dereference in lpfc_sli4_rq_put() perf: Fix error return code Yaniv Rosner (1): bnx2x: Fix KR2 rapid link flap Youquan Song (1): ata_piix: Fix DVD not dectected at some Haswell platforms Yuval Mintz (1): bnx2x: Prevent null pointer dereference in AFEX mode libin (1): sched/debug: Fix sd->*_idx limit range avoiding overflow nikolay(a)redhat.com (2): Revert "bonding: remove sysfs before removing devices" bonding: fix bonding_masters race condition in bond unloading wenxiong(a)linux.vnet.ibm.com (2): [SCSI] ipr: ipr_test_msi() fails when running with msi-x enabled adapter [SCSI] ipr: dlpar failed when adding an adapter back willy tarreau (1): net: mvneta: enable features before registering the driver Documentation/DocBook/device-drivers.tmpl | 2 +- Documentation/scsi/LICENSE.qla2xxx | 2 +- MAINTAINERS | 7 +- Makefile | 2 +- arch/arc/include/asm/irqflags.h | 12 +- arch/arm/boot/dts/imx28-m28evk.dts | 1 - arch/arm/boot/dts/imx28-sps1.dts | 1 - arch/arm/boot/dts/imx6qdl.dtsi | 1 + arch/arm/boot/dts/kirkwood-iomega_ix2_200.dts | 14 +- arch/arm/mach-imx/clk-imx35.c | 2 + arch/arm/mach-imx/clk-imx6q.c | 3 +- arch/arm/mach-kirkwood/board-iomega_ix2_200.c | 7 +- arch/arm/mach-mvebu/irq-armada-370-xp.c | 16 +- arch/arm/mach-s3c24xx/include/mach/irqs.h | 4 +- arch/arm/mach-s3c24xx/irq.c | 2 +- arch/c6x/include/asm/irqflags.h | 2 +- arch/ia64/kernel/palinfo.c | 77 +---- arch/m68k/include/asm/gpio.h | 20 ++ arch/powerpc/platforms/pseries/lpar.c | 8 +- arch/tile/include/asm/irqflags.h | 10 +- arch/x86/include/asm/paravirt.h | 5 +- arch/x86/include/asm/paravirt_types.h | 2 + arch/x86/include/asm/tlb.h | 2 +- arch/x86/kernel/cpu/perf_event_intel_ds.c | 3 +- arch/x86/kernel/paravirt.c | 25 +- arch/x86/lguest/boot.c | 1 + arch/x86/mm/fault.c | 6 +- arch/x86/mm/pageattr-test.c | 2 +- arch/x86/mm/pageattr.c | 12 +- arch/x86/mm/pgtable.c | 7 + arch/x86/xen/mmu.c | 13 +- block/blk-sysfs.c | 2 + block/partition-generic.c | 1 - crypto/gcm.c | 17 +- drivers/ata/ata_piix.c | 14 +- drivers/ata/libata-core.c | 6 +- drivers/ata/libata-scsi.c | 8 +- drivers/base/regmap/regmap.c | 3 +- drivers/block/loop.c | 21 +- drivers/block/mtip32xx/mtip32xx.c | 327 ++++++++++++++------ drivers/block/mtip32xx/mtip32xx.h | 18 +- drivers/cpufreq/intel_pstate.c | 1 - drivers/crypto/ux500/cryp/cryp_core.c | 2 +- drivers/dma/omap-dma.c | 20 +- drivers/dma/pl330.c | 38 ++- drivers/gpio/gpio-pca953x.c | 2 +- drivers/gpu/drm/drm_fb_helper.c | 8 +- drivers/gpu/drm/mgag200/mgag200_mode.c | 13 +- drivers/gpu/drm/nouveau/nv50_display.c | 2 +- drivers/gpu/drm/udl/udl_connector.c | 4 + drivers/hwspinlock/hwspinlock_core.c | 2 + drivers/misc/vmw_vmci/Kconfig | 2 +- drivers/net/bonding/bond_main.c | 11 +- drivers/net/ethernet/broadcom/bnx2x/bnx2x_link.c | 18 +- drivers/net/ethernet/broadcom/bnx2x/bnx2x_main.c | 3 +- drivers/net/ethernet/intel/e100.c | 36 ++- drivers/net/ethernet/marvell/mvneta.c | 9 +- drivers/net/hyperv/netvsc.c | 17 +- drivers/net/hyperv/netvsc_drv.c | 2 - drivers/net/hyperv/rndis_filter.c | 14 +- drivers/net/wireless/ath/ath9k/main.c | 4 + drivers/net/wireless/brcm80211/brcmfmac/dhd_sdio.c | 6 +- .../net/wireless/brcm80211/brcmfmac/wl_cfg80211.c | 46 +-- drivers/net/wireless/mwifiex/scan.c | 11 +- drivers/net/wireless/rt2x00/Kconfig | 7 + drivers/net/wireless/rt2x00/Makefile | 1 + drivers/net/wireless/rt2x00/rt2400pci.c | 1 + drivers/net/wireless/rt2x00/rt2500pci.c | 1 + drivers/net/wireless/rt2x00/rt2800pci.c | 1 + drivers/net/wireless/rt2x00/rt2x00mmio.c | 216 +++++++++++++ drivers/net/wireless/rt2x00/rt2x00mmio.h | 119 +++++++ drivers/net/wireless/rt2x00/rt2x00pci.c | 176 ----------- drivers/net/wireless/rt2x00/rt2x00pci.h | 88 ------ drivers/net/wireless/rt2x00/rt61pci.c | 1 + drivers/remoteproc/Kconfig | 2 +- drivers/remoteproc/remoteproc_core.c | 6 +- drivers/remoteproc/ste_modem_rproc.c | 7 +- drivers/s390/net/qeth_core.h | 3 + drivers/s390/net/qeth_core_main.c | 19 ++ drivers/s390/net/qeth_l2_main.c | 2 + drivers/s390/net/qeth_l3_main.c | 2 + drivers/scsi/ibmvscsi/ibmvscsi.c | 2 +- drivers/scsi/ipr.c | 13 +- drivers/scsi/libsas/sas_expander.c | 14 +- drivers/scsi/lpfc/lpfc_sli.c | 3 +- drivers/scsi/qla2xxx/qla_attr.c | 5 - drivers/scsi/qla2xxx/qla_dbg.c | 3 +- drivers/scsi/qla2xxx/qla_def.h | 1 - drivers/scsi/qla2xxx/qla_gbl.h | 3 - drivers/scsi/qla2xxx/qla_init.c | 4 +- drivers/scsi/qla2xxx/qla_mbx.c | 58 ---- drivers/scsi/qla2xxx/qla_version.h | 2 +- drivers/scsi/st.c | 8 +- drivers/target/target_core_alua.c | 3 + drivers/tty/mxser.c | 8 +- drivers/tty/serial/8250/8250_pnp.c | 12 +- drivers/tty/serial/omap-serial.c | 11 + drivers/vfio/pci/vfio_pci.c | 3 +- drivers/vhost/tcm_vhost.c | 198 ++++++++---- drivers/watchdog/Kconfig | 2 +- drivers/xen/events.c | 19 +- fs/btrfs/tree-log.c | 48 ++- fs/cifs/connect.c | 16 +- fs/ecryptfs/miscdev.c | 14 +- fs/inode.c | 2 +- fs/namespace.c | 2 +- fs/nfs/nfs4client.c | 45 ++- fs/nfs/nfs4proc.c | 1 + fs/nfs/nfs4state.c | 8 +- fs/proc/generic.c | 119 +++++-- include/asm-generic/tlb.h | 7 +- include/linux/ata.h | 2 +- include/linux/capability.h | 2 + include/linux/ftrace.h | 5 +- include/linux/libata.h | 1 + include/linux/preempt.h | 22 +- include/linux/proc_fs.h | 2 + include/linux/security.h | 12 + include/linux/spinlock_up.h | 29 +- include/net/iucv/af_iucv.h | 8 + include/sound/dmaengine_pcm.h | 92 +++++- include/sound/soc-dapm.h | 1 - include/sound/soc.h | 13 +- kernel/capability.c | 24 ++ kernel/events/core.c | 4 +- kernel/events/internal.h | 2 +- kernel/events/ring_buffer.c | 22 +- kernel/sched/clock.c | 26 ++ kernel/sched/core.c | 8 +- kernel/sched/cputime.c | 2 +- kernel/sys.c | 3 +- kernel/trace/ftrace.c | 54 ++-- kernel/trace/trace.c | 9 +- kernel/trace/trace_stack.c | 2 +- lib/kobject.c | 9 +- mm/memory.c | 1 + net/atm/common.c | 2 + net/ax25/af_ax25.c | 1 + net/bluetooth/af_bluetooth.c | 4 +- net/bluetooth/rfcomm/sock.c | 1 + net/bluetooth/sco.c | 1 + net/caif/caif_socket.c | 2 + net/can/gw.c | 6 +- net/core/rtnetlink.c | 4 +- net/ipv4/devinet.c | 60 +++- net/ipv4/tcp_output.c | 1 + net/ipv6/tcp_ipv6.c | 1 + net/irda/af_irda.c | 2 + net/iucv/af_iucv.c | 36 +-- net/l2tp/l2tp_ip6.c | 1 + net/llc/af_llc.c | 2 + net/netrom/af_netrom.c | 1 + net/nfc/llcp/sock.c | 3 + net/rose/af_rose.c | 1 + net/sunrpc/clnt.c | 11 +- net/tipc/socket.c | 7 + net/vmw_vsock/af_vsock.c | 2 + net/vmw_vsock/vmci_transport.c | 3 +- net/wireless/sme.c | 2 + security/capability.c | 6 + security/security.c | 5 + security/selinux/hooks.c | 7 + sound/soc/Kconfig | 4 + sound/soc/Makefile | 4 + sound/soc/atmel/atmel-pcm-dma.c | 29 +- sound/soc/cirrus/edb93xx.c | 1 - sound/soc/cirrus/ep93xx-ac97.c | 9 +- sound/soc/cirrus/ep93xx-i2s.c | 16 +- sound/soc/cirrus/ep93xx-pcm.c | 38 +-- sound/soc/cirrus/ep93xx-pcm.h | 20 -- sound/soc/cirrus/simone.c | 2 - sound/soc/cirrus/snappercl15.c | 1 - sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/cs4271.c | 166 +++++----- sound/soc/codecs/max98088.c | 32 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm8903.c | 2 + sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl_ssi.c | 55 ++-- sound/soc/fsl/fsl_ssi.h | 8 + sound/soc/fsl/imx-audmux.c | 3 +- sound/soc/fsl/imx-pcm-dma.c | 120 +------ sound/soc/fsl/imx-pcm-fiq.c | 6 +- sound/soc/fsl/imx-pcm.c | 6 +- sound/soc/fsl/imx-pcm.h | 24 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/imx-ssi.c | 51 ++- sound/soc/fsl/imx-ssi.h | 7 +- sound/soc/mxs/mxs-pcm.c | 43 +-- sound/soc/mxs/mxs-pcm.h | 4 +- sound/soc/mxs/mxs-saif.c | 6 +- sound/soc/omap/am3517evm.c | 1 - sound/soc/omap/ams-delta.c | 1 - sound/soc/omap/mcbsp.c | 14 +- sound/soc/omap/mcbsp.h | 7 +- sound/soc/omap/n810.c | 1 - sound/soc/omap/omap-abe-twl6040.c | 1 - sound/soc/omap/omap-dmic.c | 38 +-- sound/soc/omap/omap-hdmi.c | 24 +- sound/soc/omap/omap-mcbsp.c | 18 +- sound/soc/omap/omap-mcpdm.c | 109 ++++--- sound/soc/omap/omap-pcm.c | 86 +---- sound/soc/omap/omap-pcm.h | 40 --- sound/soc/omap/omap-twl4030.c | 1 - sound/soc/omap/omap3pandora.c | 9 +- sound/soc/omap/osk5912.c | 1 - sound/soc/omap/rx51.c | 1 - sound/soc/pxa/mmp-pcm.c | 34 +- sound/soc/samsung/i2s.c | 17 +- sound/soc/soc-compress.c | 14 +- sound/soc/soc-core.c | 116 ++++--- sound/soc/soc-dapm.c | 1 - sound/soc/soc-dmaengine-pcm.c | 150 +++++---- sound/soc/soc-generic-dmaengine-pcm.c | 284 +++++++++++++++++ sound/soc/soc-io.c | 5 +- sound/soc/soc-utils.c | 25 ++ sound/soc/spear/spear_pcm.c | 19 +- sound/soc/tegra/Kconfig | 2 +- sound/soc/tegra/tegra20_ac97.c | 13 +- sound/soc/tegra/tegra20_ac97.h | 4 +- sound/soc/tegra/tegra20_i2s.c | 13 +- sound/soc/tegra/tegra20_i2s.h | 4 +- sound/soc/tegra/tegra20_spdif.c | 7 +- sound/soc/tegra/tegra20_spdif.h | 4 +- sound/soc/tegra/tegra30_ahub.c | 8 +- sound/soc/tegra/tegra30_ahub.h | 8 +- sound/soc/tegra/tegra30_i2s.c | 13 +- sound/soc/tegra/tegra30_i2s.h | 4 +- sound/soc/tegra/tegra_pcm.c | 209 +------------ sound/soc/tegra/tegra_pcm.h | 7 - sound/soc/ux500/Kconfig | 2 +- sound/soc/ux500/ux500_pcm.c | 229 ++------------ sound/soc/ux500/ux500_pcm.h | 14 - sound/usb/mixer_quirks.c | 4 +- sound/usb/quirks.c | 2 +- 235 files changed, 2790 insertions(+), 2211 deletions(-) create mode 100644 drivers/net/wireless/rt2x00/rt2x00mmio.c create mode 100644 drivers/net/wireless/rt2x00/rt2x00mmio.h delete mode 100644 sound/soc/cirrus/ep93xx-pcm.h delete mode 100644 sound/soc/omap/omap-pcm.h create mode 100644 sound/soc/soc-generic-dmaengine-pcm.c

3 4

[alsa-devel] [PATCH] sgtl5000.txt: Include supplies as required properties
by Fabio Estevam 18 Apr '13

18 Apr '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 driver cannot probe succesfully if VDDA and VDDIO regulators are not provided. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- .../devicetree/bindings/sound/sgtl5000.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 9cc4444..520eb98 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -5,9 +5,15 @@ Required properties: - reg : the I2C address of the device +- VDDA-supply: regulator that provides VDDA voltage + +- VDDIO-supply: regulator that provides VDDIO voltage + Example: codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; + VDDA-supply = <&reg_2p5v>; + VDDIO-supply = <&reg_3p3v>; }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] alsa-lib: pcm.c: fix indentation
by Daniel Mack 18 Apr '13

18 Apr '13

Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- src/pcm/pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 05737d9..2963711 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1634,11 +1634,11 @@ static const char *const snd_pcm_type_names[] = { PCMTYPE(LADSPA), PCMTYPE(DMIX), PCMTYPE(JACK), - PCMTYPE(DSNOOP), - PCMTYPE(IEC958), + PCMTYPE(DSNOOP), + PCMTYPE(IEC958), PCMTYPE(SOFTVOL), - PCMTYPE(IOPLUG), - PCMTYPE(EXTPLUG), + PCMTYPE(IOPLUG), + PCMTYPE(EXTPLUG), PCMTYPE(MMAP_EMUL), }; -- 1.8.1.4

2 2

[alsa-devel] [PATCH v3 0/5] ALSA: snd-usb: add support for DSD
by Daniel Mack 18 Apr '13

18 Apr '13

In contrast to the first two approaches, this one keeps the userspace unaware of the actual transport on the bus. More specifically, the DSD_U16_LE format may be implemented to actually send DOP samples to the device, depending on the quirks entries. Also, as there is quite some confusion in both data file formats as well as hardware sample formats about whether to stuff the oldest bit in the MSB or the LSB of a byte, ALSA defines that now to be in the MSB, as far as the ALSA userspace layer is concerned. The driver learned functions to re-encode the bytes according to what the hardware supports. I also wrote a very basic test tool which I published here: https://github.com/zonque/alsa-dsd-player Thanks, Daniel Daniel Mack (5): ALSA: add DSD formats ALSA: snd-usb: use ep->stride from urb callbacks ALSA: snd-usb: add support for DSD DOP stream transport ALSA: snd-usb: add support for bit-reversed byte formats ALSA: snd-usb: add quirks handler for DSD streams include/sound/pcm.h | 2 + include/uapi/sound/asound.h | 4 +- sound/core/pcm.c | 2 + sound/core/pcm_misc.c | 8 +++ sound/usb/card.h | 8 +++ sound/usb/endpoint.c | 9 ++++ sound/usb/format.c | 3 ++ sound/usb/pcm.c | 118 ++++++++++++++++++++++++++++++++++++------- sound/usb/quirks.c | 28 ++++++++++ sound/usb/quirks.h | 4 ++ 10 files changed, 167 insertions(+), 19 deletions(-) -- 1.7.10.4

3 12

Re: [alsa-devel] [PATCH -next] ALSA: hda - fix error return code in patch_alc662()
by Takashi Iwai 18 Apr '13

18 Apr '13

At Thu, 18 Apr 2013 11:12:59 +0800, Wei Yongjun wrote: > > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Fix to return a negative error code from the error handling > case instead of 0, as returned elsewhere in this function. > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Thanks, applied. Takashi > --- > sound/pci/hda/patch_realtek.c | 3 ++- > 1 file changed, 2 insertions(+), 1 deletion(-) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index f15c36b..a3965b0 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -3787,7 +3787,8 @@ static int patch_alc662(struct hda_codec *codec) > if ((alc_get_coef0(codec) & (1 << 14)) && > codec->bus->pci->subsystem_vendor == 0x1025 && > spec->cdefine.platform_type == 1) { > - if (alc_codec_rename(codec, "ALC272X") < 0) > + err = alc_codec_rename(codec, "ALC272X"); > + if (err < 0) > goto error; > } > >

1 0

Re: [alsa-devel] [PATCH] ASoC: add RT5640 CODEC driver
by Mark Brown 17 Apr '13

17 Apr '13

On Tue, Apr 16, 2013 at 04:29:40PM -0600, Stephen Warren wrote: > a) There are many different VDD inputs to the chip. I imagine that some > of these only need to be enabled/active under certain conditions. For > example, perhaps the SPKVDD or MICVDD inputs are only required if > actively using the speakers or microphones, and could be disabled at > other times? If so, all of these "optional" or "part-time" VDD inputs > should be represented as regulators, and retrieved/manipulated using > Linux APIs such as regulator_get(), regulator_enable(), etc. There's framework support for regulator supplies in ASoC so you just need to specify the name and the core will do the rest. > b) Some VDD inputs are optional; for example, either I believe that a > board should either provide DCVDD, or provide LDO1_IN and assert the > LDO1_EN input signal. Similarly, I think that either MICVDD should be > provided, or SPKVDDL be provided, coupled with LDO2_EN register bit > being set. This may require the RT5640 driver to provide two regulator > objects (LDO1, LDO2), which the board file or device tree can connect > back to the DCVDD and MICVDD inputs if appropriate for the board's HW > configuration. In the case where LDO1 is used, a separate fixed > regulator with GPIO should be used to control the LDO1_EN GPIO input. That said if boards generally don't use external supplies and use the built in regulators then it's probably best to at least have the driver assume that by default. > > Bard: You are right. The mic type should be defined according to circuit. > > And it should be change in run time. > > So, should we just remove this control in codec driver? > Mark's point is that the driver should either/both accept some platform > data structure, or parse device tree, to configure the input pins. I > don't think completely removing all mechanisms to configure this would > be a good idea, since then how would the board/device-tree author > configure the CODEC? Indeed, there definitely should be configuration - however what I was commenting on there was a user-visible ALSA control which should definitely go.

3 6

Re: [alsa-devel] [PATCH 26/28] proc: Supply a function to remove a proc entry by PDE [RFC]
by Grant Likely 17 Apr '13

17 Apr '13

On Tue, 16 Apr 2013 19:27:34 +0100, David Howells <dhowells(a)redhat.com> wrote: > Supply a function (proc_remove()) to remove a proc entry (and any subtree > rooted there) by proc_dir_entry pointer rather than by name and (optionally) > root dir entry pointer. This allows us to eliminate all remaining pde->name > accesses outside of procfs. > > Signed-off-by: David Howells <dhowells(a)redhat.com> > cc: linux-acpi(a)vger.kernel.org > cc: openipmi-developer(a)lists.sourceforge.net > cc: devicetree-discuss(a)lists.ozlabs.org > cc: linux-pci(a)vger.kernel.org > cc: netdev(a)vger.kernel.org > cc: netfilter-devel(a)vger.kernel.org > cc: alsa-devel(a)alsa-project.org Looks okay to me. For the OF portion: Acked-by: Grant Likely <grant.likely(a)linaro.or> > --- > > drivers/acpi/sbs.c | 21 ++++----------------- > drivers/char/ipmi/ipmi_msghandler.c | 2 +- > drivers/misc/sgi-gru/gruprocfs.c | 2 +- > drivers/of/base.c | 11 +---------- > drivers/pci/proc.c | 12 +++--------- > fs/proc/generic.c | 7 +++++++ > fs/proc/vmcore.c | 2 +- > include/linux/proc_fs.h | 2 ++ > kernel/irq/proc.c | 6 +----- > net/8021q/vlanproc.c | 9 ++------- > net/core/pktgen.c | 6 ++---- > net/ipv4/netfilter/ipt_CLUSTERIP.c | 4 ++-- > net/ipv6/proc.c | 3 +-- > net/netfilter/xt_hashlimit.c | 4 ++-- > sound/core/info.c | 19 +++++-------------- > 15 files changed, 35 insertions(+), 75 deletions(-) > > diff --git a/drivers/acpi/sbs.c b/drivers/acpi/sbs.c > index a296e08..b6241ee 100644 > --- a/drivers/acpi/sbs.c > +++ b/drivers/acpi/sbs.c > @@ -521,19 +521,6 @@ acpi_sbs_add_fs(struct proc_dir_entry **dir, > return 0; > } > > -static void > -acpi_sbs_remove_fs(struct proc_dir_entry **dir, > - struct proc_dir_entry *parent_dir) > -{ > - if (*dir) { > - remove_proc_entry(ACPI_SBS_FILE_INFO, *dir); > - remove_proc_entry(ACPI_SBS_FILE_STATE, *dir); > - remove_proc_entry(ACPI_SBS_FILE_ALARM, *dir); > - remove_proc_entry((*dir)->name, parent_dir); > - *dir = NULL; > - } > -} > - > /* Smart Battery Interface */ > static struct proc_dir_entry *acpi_battery_dir = NULL; > > @@ -836,8 +823,8 @@ static void acpi_battery_remove(struct acpi_sbs *sbs, int id) > power_supply_unregister(&battery->bat); > } > #ifdef CONFIG_ACPI_PROCFS_POWER > - if (battery->proc_entry) > - acpi_sbs_remove_fs(&battery->proc_entry, acpi_battery_dir); > + proc_remove(battery->proc_entry); > + battery->proc_entry = NULL; > #endif > } > > @@ -873,8 +860,8 @@ static void acpi_charger_remove(struct acpi_sbs *sbs) > if (sbs->charger.dev) > power_supply_unregister(&sbs->charger); > #ifdef CONFIG_ACPI_PROCFS_POWER > - if (sbs->charger_entry) > - acpi_sbs_remove_fs(&sbs->charger_entry, acpi_ac_dir); > + proc_remove(sbs->charger_entry); > + sbs->charger_entry = NULL; > #endif > } > > diff --git a/drivers/char/ipmi/ipmi_msghandler.c b/drivers/char/ipmi/ipmi_msghandler.c > index 1420bbb..4d439d2 100644 > --- a/drivers/char/ipmi/ipmi_msghandler.c > +++ b/drivers/char/ipmi/ipmi_msghandler.c > @@ -4541,7 +4541,7 @@ static void __exit cleanup_ipmi(void) > del_timer_sync(&ipmi_timer); > > #ifdef CONFIG_PROC_FS > - remove_proc_entry(proc_ipmi_root->name, NULL); > + proc_remove(proc_ipmi_root); > #endif /* CONFIG_PROC_FS */ > > driver_unregister(&ipmidriver.driver); > diff --git a/drivers/misc/sgi-gru/gruprocfs.c b/drivers/misc/sgi-gru/gruprocfs.c > index 950dbe9..797d796 100644 > --- a/drivers/misc/sgi-gru/gruprocfs.c > +++ b/drivers/misc/sgi-gru/gruprocfs.c > @@ -355,7 +355,7 @@ static void delete_proc_files(void) > for (p = proc_files; p->name; p++) > if (p->entry) > remove_proc_entry(p->name, proc_gru); > - remove_proc_entry("gru", proc_gru->parent); > + proc_remove(proc_gru); > } > } > > diff --git a/drivers/of/base.c b/drivers/of/base.c > index 321d3ef..9c70436 100644 > --- a/drivers/of/base.c > +++ b/drivers/of/base.c > @@ -1452,16 +1452,7 @@ int of_attach_node(struct device_node *np) > #ifdef CONFIG_PROC_DEVICETREE > static void of_remove_proc_dt_entry(struct device_node *dn) > { > - struct device_node *parent = dn->parent; > - struct property *prop = dn->properties; > - > - while (prop) { > - remove_proc_entry(prop->name, dn->pde); > - prop = prop->next; > - } > - > - if (dn->pde) > - remove_proc_entry(dn->pde->name, parent->pde); > + proc_remove(dn->pde); > } > #else > static void of_remove_proc_dt_entry(struct device_node *dn) > diff --git a/drivers/pci/proc.c b/drivers/pci/proc.c > index 7cde7c1..0812608 100644 > --- a/drivers/pci/proc.c > +++ b/drivers/pci/proc.c > @@ -427,20 +427,14 @@ int pci_proc_attach_device(struct pci_dev *dev) > > int pci_proc_detach_device(struct pci_dev *dev) > { > - struct proc_dir_entry *e; > - > - if ((e = dev->procent)) { > - remove_proc_entry(e->name, dev->bus->procdir); > - dev->procent = NULL; > - } > + proc_remove(dev->procent); > + dev->procent = NULL; > return 0; > } > > int pci_proc_detach_bus(struct pci_bus* bus) > { > - struct proc_dir_entry *de = bus->procdir; > - if (de) > - remove_proc_entry(de->name, proc_bus_pci_dir); > + proc_remove(bus->procdir); > return 0; > } > > diff --git a/fs/proc/generic.c b/fs/proc/generic.c > index 2c6d6be..4b48033 100644 > --- a/fs/proc/generic.c > +++ b/fs/proc/generic.c > @@ -630,3 +630,10 @@ const char *get_proc_name(const struct proc_dir_entry *de) > return de->name; > } > EXPORT_SYMBOL_GPL(get_proc_name); > + > +void proc_remove(struct proc_dir_entry *de) > +{ > + if (de) > + remove_proc_subtree(de->name, de->parent); > +} > +EXPORT_SYMBOL(proc_remove); > diff --git a/fs/proc/vmcore.c b/fs/proc/vmcore.c > index 38edddc..17f7e08 100644 > --- a/fs/proc/vmcore.c > +++ b/fs/proc/vmcore.c > @@ -699,7 +699,7 @@ void vmcore_cleanup(void) > struct list_head *pos, *next; > > if (proc_vmcore) { > - remove_proc_entry(proc_vmcore->name, proc_vmcore->parent); > + proc_remove(proc_vmcore); > proc_vmcore = NULL; > } > > diff --git a/include/linux/proc_fs.h b/include/linux/proc_fs.h > index 7abc72a..b8949b1 100644 > --- a/include/linux/proc_fs.h > +++ b/include/linux/proc_fs.h > @@ -67,6 +67,7 @@ struct proc_dir_entry *proc_create_data(const char *name, umode_t mode, > struct proc_dir_entry *parent, > const struct file_operations *proc_fops, > void *data); > +extern void proc_remove(struct proc_dir_entry *); > extern void remove_proc_entry(const char *name, struct proc_dir_entry *parent); > extern int remove_proc_subtree(const char *name, struct proc_dir_entry *parent); > > @@ -103,6 +104,7 @@ static inline struct proc_dir_entry *proc_create_data(const char *name, > { > return NULL; > } > +static inline void proc_remove(struct proc_dir_entry *de) {} > #define remove_proc_entry(name, parent) do {} while (0) > #define remove_proc_subtree(name, parent) do {} while (0) > > diff --git a/kernel/irq/proc.c b/kernel/irq/proc.c > index d59ae37..19ed5c4 100644 > --- a/kernel/irq/proc.c > +++ b/kernel/irq/proc.c > @@ -366,11 +366,7 @@ void unregister_irq_proc(unsigned int irq, struct irq_desc *desc) > > void unregister_handler_proc(unsigned int irq, struct irqaction *action) > { > - if (action->dir) { > - struct irq_desc *desc = irq_to_desc(irq); > - > - remove_proc_entry(action->dir->name, desc->dir); > - } > + proc_remove(action->dir); > } > > static void register_default_affinity_proc(void) > diff --git a/net/8021q/vlanproc.c b/net/8021q/vlanproc.c > index 959ddbb..1d0e8921 100644 > --- a/net/8021q/vlanproc.c > +++ b/net/8021q/vlanproc.c > @@ -184,14 +184,9 @@ int vlan_proc_add_dev(struct net_device *vlandev) > */ > int vlan_proc_rem_dev(struct net_device *vlandev) > { > - struct vlan_net *vn = net_generic(dev_net(vlandev), vlan_net_id); > - > /** NOTE: This will consume the memory pointed to by dent, it seems. */ > - if (vlan_dev_priv(vlandev)->dent) { > - remove_proc_entry(vlan_dev_priv(vlandev)->dent->name, > - vn->proc_vlan_dir); > - vlan_dev_priv(vlandev)->dent = NULL; > - } > + proc_remove(vlan_dev_priv(vlandev)->dent); > + vlan_dev_priv(vlandev)->dent = NULL; > return 0; > } > > diff --git a/net/core/pktgen.c b/net/core/pktgen.c > index f6af4fe..6c41e97 100644 > --- a/net/core/pktgen.c > +++ b/net/core/pktgen.c > @@ -1904,7 +1904,7 @@ static void pktgen_change_name(const struct pktgen_net *pn, struct net_device *d > if (pkt_dev->odev != dev) > continue; > > - remove_proc_entry(pkt_dev->entry->name, pn->proc_dir); > + proc_remove(pkt_dev->entry); > > pkt_dev->entry = proc_create_data(dev->name, 0600, > pn->proc_dir, > @@ -3576,8 +3576,6 @@ static void _rem_dev_from_if_list(struct pktgen_thread *t, > static int pktgen_remove_device(struct pktgen_thread *t, > struct pktgen_dev *pkt_dev) > { > - struct pktgen_net *pn = t->net; > - > pr_debug("remove_device pkt_dev=%p\n", pkt_dev); > > if (pkt_dev->running) { > @@ -3597,7 +3595,7 @@ static int pktgen_remove_device(struct pktgen_thread *t, > _rem_dev_from_if_list(t, pkt_dev); > > if (pkt_dev->entry) > - remove_proc_entry(pkt_dev->entry->name, pn->proc_dir); > + proc_remove(pkt_dev->entry); > > #ifdef CONFIG_XFRM > free_SAs(pkt_dev); > diff --git a/net/ipv4/netfilter/ipt_CLUSTERIP.c b/net/ipv4/netfilter/ipt_CLUSTERIP.c > index e4738fe..0b732ef 100644 > --- a/net/ipv4/netfilter/ipt_CLUSTERIP.c > +++ b/net/ipv4/netfilter/ipt_CLUSTERIP.c > @@ -105,7 +105,7 @@ clusterip_config_entry_put(struct clusterip_config *c) > * functions are also incrementing the refcount on their own, > * so it's safe to remove the entry even if it's in use. */ > #ifdef CONFIG_PROC_FS > - remove_proc_entry(c->pde->name, c->pde->parent); > + proc_remove(c->pde); > #endif > return; > } > @@ -736,7 +736,7 @@ static void __exit clusterip_tg_exit(void) > { > pr_info("ClusterIP Version %s unloading\n", CLUSTERIP_VERSION); > #ifdef CONFIG_PROC_FS > - remove_proc_entry(clusterip_procdir->name, clusterip_procdir->parent); > + proc_remove(clusterip_procdir); > #endif > nf_unregister_hook(&cip_arp_ops); > xt_unregister_target(&clusterip_tg_reg); > diff --git a/net/ipv6/proc.c b/net/ipv6/proc.c > index 7ea6e18..537d9ee 100644 > --- a/net/ipv6/proc.c > +++ b/net/ipv6/proc.c > @@ -287,8 +287,7 @@ int snmp6_unregister_dev(struct inet6_dev *idev) > return -ENOENT; > if (!idev->stats.proc_dir_entry) > return -EINVAL; > - remove_proc_entry(idev->stats.proc_dir_entry->name, > - net->mib.proc_net_devsnmp6); > + proc_remove(idev->stats.proc_dir_entry); > idev->stats.proc_dir_entry = NULL; > return 0; > } > diff --git a/net/netfilter/xt_hashlimit.c b/net/netfilter/xt_hashlimit.c > index d2fd53b..40002ef 100644 > --- a/net/netfilter/xt_hashlimit.c > +++ b/net/netfilter/xt_hashlimit.c > @@ -330,7 +330,7 @@ static void htable_destroy(struct xt_hashlimit_htable *hinfo) > parent = hashlimit_net->ip6t_hashlimit; > > if(parent != NULL) > - remove_proc_entry(hinfo->pde->name, parent); > + proc_remove(hinfo->pde); > > htable_selective_cleanup(hinfo, select_all); > vfree(hinfo); > @@ -887,7 +887,7 @@ static void __net_exit hashlimit_proc_net_exit(struct net *net) > pde = hashlimit_net->ip6t_hashlimit; > > hlist_for_each_entry(hinfo, &hashlimit_net->htables, node) > - remove_proc_entry(hinfo->pde->name, pde); > + proc_remove(hinfo->pde); > > hashlimit_net->ipt_hashlimit = NULL; > hashlimit_net->ip6t_hashlimit = NULL; > diff --git a/sound/core/info.c b/sound/core/info.c > index c7f41c3..3c9bd6b 100644 > --- a/sound/core/info.c > +++ b/sound/core/info.c > @@ -153,13 +153,6 @@ EXPORT_SYMBOL(snd_seq_root); > struct snd_info_entry *snd_oss_root; > #endif > > -static void snd_remove_proc_entry(struct proc_dir_entry *parent, > - struct proc_dir_entry *de) > -{ > - if (de) > - remove_proc_entry(de->name, parent); > -} > - > static loff_t snd_info_entry_llseek(struct file *file, loff_t offset, int orig) > { > struct snd_info_private_data *data; > @@ -580,7 +573,7 @@ int __exit snd_info_done(void) > #ifdef CONFIG_SND_OSSEMUL > snd_info_free_entry(snd_oss_root); > #endif > - snd_remove_proc_entry(NULL, snd_proc_root); > + proc_remove(snd_proc_root); > } > return 0; > } > @@ -642,7 +635,7 @@ void snd_info_card_id_change(struct snd_card *card) > { > mutex_lock(&info_mutex); > if (card->proc_root_link) { > - snd_remove_proc_entry(snd_proc_root, card->proc_root_link); > + proc_remove(card->proc_root_link); > card->proc_root_link = NULL; > } > if (strcmp(card->id, card->proc_root->name)) > @@ -661,10 +654,8 @@ void snd_info_card_disconnect(struct snd_card *card) > if (!card) > return; > mutex_lock(&info_mutex); > - if (card->proc_root_link) { > - snd_remove_proc_entry(snd_proc_root, card->proc_root_link); > - card->proc_root_link = NULL; > - } > + proc_remove(card->proc_root_link); > + card->proc_root_link = NULL; > if (card->proc_root) > snd_info_disconnect(card->proc_root); > mutex_unlock(&info_mutex); > @@ -856,7 +847,7 @@ static void snd_info_disconnect(struct snd_info_entry *entry) > list_del_init(&entry->list); > root = entry->parent == NULL ? snd_proc_root : entry->parent->p; > snd_BUG_ON(!root); > - snd_remove_proc_entry(root, entry->p); > + proc_remove(entry->p); > entry->p = NULL; > } > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ -- Grant Likely, B.Sc, P.Eng. Secret Lab Technologies, Ltd.

1 0

[alsa-devel] [PATCH v2] ASoC: wm_adsp: Expose coefficient blocks as ALSA binary controls
by Dimitris Papastamos 17 Apr '13

17 Apr '13

Add initial support for runtime tuning for the ADSP cores. This is achieved by exposing the coefficient configuration blocks as ALSA binary controls. The current code assumes that no controls on the DSP are volatile. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 435 ++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 3 + 2 files changed, 431 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3470b64..080d05f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -215,6 +215,36 @@ static struct { [WM_ADSP_FW_RX_ANC] = { .file = "rx-anc" }, }; +struct wm_coeff_ctl_ops { + int (*xget)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*xput)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*xinfo)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +}; + +struct wm_coeff { + struct device *dev; + struct list_head ctl_list; + struct regmap *regmap; +}; + +struct wm_coeff_ctl { + const char *name; + struct snd_card *card; + struct wm_adsp_alg_region region; + struct wm_coeff_ctl_ops ops; + struct wm_adsp *adsp; + void *private; + unsigned int enabled:1; + struct list_head list; + void *cache; + size_t len; + unsigned int dirty:1; + struct snd_kcontrol *kcontrol; +}; + static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -334,6 +364,175 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, } } +static int wm_coeff_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = ctl->len; + return 0; +} + +static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, + const void *buf, size_t len) +{ + struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_adsp_alg_region *region = &ctl->region; + const struct wm_adsp_region *mem; + struct wm_adsp *adsp = ctl->adsp; + void *scratch; + int ret; + unsigned int reg; + + mem = wm_adsp_find_region(adsp, region->type); + if (!mem) { + adsp_err(adsp, "No base for region %x\n", + region->type); + return -EINVAL; + } + + reg = ctl->region.base; + reg = wm_adsp_region_to_reg(mem, reg); + + scratch = kmemdup(buf, ctl->len, GFP_KERNEL | GFP_DMA); + if (!scratch) + return -ENOMEM; + + ret = regmap_raw_write(wm_coeff->regmap, reg, scratch, + ctl->len); + if (ret) { + adsp_err(adsp, "Failed to write %zu bytes to %x\n", + ctl->len, reg); + kfree(scratch); + return ret; + } + + kfree(scratch); + + return 0; +} + +static int wm_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + char *p = ucontrol->value.bytes.data; + + memcpy(ctl->cache, p, ctl->len); + + if (!ctl->enabled) { + ctl->dirty = 1; + return 0; + } + + return wm_coeff_write_control(kcontrol, p, ctl->len); +} + +static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, + void *buf, size_t len) +{ + struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_adsp_alg_region *region = &ctl->region; + const struct wm_adsp_region *mem; + struct wm_adsp *adsp = ctl->adsp; + void *scratch; + int ret; + unsigned int reg; + + mem = wm_adsp_find_region(adsp, region->type); + if (!mem) { + adsp_err(adsp, "No base for region %x\n", + region->type); + return -EINVAL; + } + + reg = ctl->region.base; + reg = wm_adsp_region_to_reg(mem, reg); + + scratch = kmalloc(ctl->len, GFP_KERNEL | GFP_DMA); + if (!scratch) + return -ENOMEM; + + ret = regmap_raw_read(wm_coeff->regmap, reg, scratch, ctl->len); + if (ret) { + adsp_err(adsp, "Failed to read %zu bytes from %x\n", + ctl->len, reg); + kfree(scratch); + return ret; + } + + memcpy(buf, scratch, ctl->len); + kfree(scratch); + + return 0; +} + +static int wm_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + char *p = ucontrol->value.bytes.data; + + memcpy(p, ctl->cache, ctl->len); + return 0; +} + +static int wm_coeff_add_kcontrol(struct wm_coeff *wm_coeff, + struct wm_coeff_ctl *ctl, + const struct snd_kcontrol_new *kctl) +{ + int ret; + struct snd_kcontrol *kcontrol; + + kcontrol = snd_ctl_new1(kctl, wm_coeff); + ret = snd_ctl_add(ctl->card, kcontrol); + if (ret < 0) { + dev_err(wm_coeff->dev, "Failed to add %s: %d\n", + kctl->name, ret); + return ret; + } + ctl->kcontrol = kcontrol; + return 0; +} + +static int wmfw_add_ctl(struct wm_coeff *wm_coeff, + struct wm_coeff_ctl *ctl) +{ + struct snd_kcontrol_new *kcontrol; + int ret; + + if (!wm_coeff || !ctl || !ctl->name || !ctl->card) + return -EINVAL; + + kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); + if (!kcontrol) + return -ENOMEM; + kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + + kcontrol->name = ctl->name; + kcontrol->info = wm_coeff_info; + kcontrol->get = wm_coeff_get; + kcontrol->put = wm_coeff_put; + kcontrol->private_value = (unsigned long)ctl; + + ret = wm_coeff_add_kcontrol(wm_coeff, + ctl, kcontrol); + if (ret < 0) + goto err_kcontrol; + + kfree(kcontrol); + + list_add(&ctl->list, &wm_coeff->ctl_list); + return 0; + +err_kcontrol: + kfree(kcontrol); + return ret; +} + static int wm_adsp_load(struct wm_adsp *dsp) { LIST_HEAD(buf_list); @@ -547,7 +746,140 @@ out: return ret; } -static int wm_adsp_setup_algs(struct wm_adsp *dsp) +static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff) +{ + struct wm_coeff_ctl *ctl; + int ret; + + list_for_each_entry(ctl, &wm_coeff->ctl_list, + list) { + if (!ctl->enabled || ctl->dirty) + continue; + ret = wm_coeff_read_control(ctl->kcontrol, + ctl->cache, + ctl->len); + if (ret < 0) + return ret; + } + + return 0; +} + +static int wm_coeff_sync_controls(struct wm_coeff *wm_coeff) +{ + struct wm_coeff_ctl *ctl; + int ret; + + list_for_each_entry(ctl, &wm_coeff->ctl_list, + list) { + if (!ctl->enabled) + continue; + if (ctl->dirty) { + ret = wm_coeff_write_control(ctl->kcontrol, + ctl->cache, + ctl->len); + if (ret < 0) + return ret; + ctl->dirty = 0; + } + } + + return 0; +} + +static int wm_adsp_create_control(struct snd_soc_codec *codec, + const struct wm_adsp_alg_region *region) + +{ + struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); + struct wm_coeff_ctl *ctl; + char *name; + char *region_name; + int ret; + + name = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!name) + return -ENOMEM; + + switch (region->type) { + case WMFW_ADSP1_PM: + region_name = "PM"; + break; + case WMFW_ADSP1_DM: + region_name = "DM"; + break; + case WMFW_ADSP2_XM: + region_name = "XM"; + break; + case WMFW_ADSP2_YM: + region_name = "YM"; + break; + case WMFW_ADSP1_ZM: + region_name = "ZM"; + break; + default: + return -EINVAL; + } + + snprintf(name, PAGE_SIZE, "DSP%d %s %x", + dsp->num, region_name, region->alg); + + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + if (!strcmp(ctl->name, name)) { + if (!ctl->enabled) + ctl->enabled = 1; + return 0; + } + } + + ctl = kzalloc(sizeof(*ctl), GFP_KERNEL); + if (!ctl) { + ret = -ENOMEM; + goto err_name; + } + ctl->region = *region; + ctl->name = kmemdup(name, strlen(name) + 1, GFP_KERNEL); + if (!ctl->name) { + ret = -ENOMEM; + goto err_ctl; + } + ctl->enabled = 1; + ctl->dirty = 0; + ctl->ops.xget = wm_coeff_get; + ctl->ops.xput = wm_coeff_put; + ctl->card = codec->card->snd_card; + ctl->adsp = dsp; + + ctl->len = region->len; + ctl->cache = kzalloc(ctl->len, GFP_KERNEL); + if (!ctl->cache) { + ret = -ENOMEM; + goto err_ctl_name; + } + + ret = wmfw_add_ctl(dsp->wm_coeff, ctl); + if (ret < 0) { + ret = -ENOMEM; + goto err_ctl_cache; + } + + kfree(name); + + return 0; + +err_ctl_cache: + kfree(ctl->cache); +err_ctl_name: + kfree(ctl->name); +err_ctl: + kfree(ctl); +err_name: + kfree(name); + return ret; +} + +static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) { struct regmap *regmap = dsp->regmap; struct wmfw_adsp1_id_hdr adsp1_id; @@ -730,7 +1062,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP1_DM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].dm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp1_alg[i + 1].dm); + region->len -= be32_to_cpu(adsp1_alg[i].dm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region DM with ID %x\n", + be32_to_cpu(adsp1_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -738,7 +1079,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP1_ZM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].zm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp1_alg[i + 1].zm); + region->len -= be32_to_cpu(adsp1_alg[i].zm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", + be32_to_cpu(adsp1_alg[i].alg.id)); + } break; case WMFW_ADSP2: @@ -758,7 +1108,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_XM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].xm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].xm); + region->len -= be32_to_cpu(adsp2_alg[i].xm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region XM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -766,7 +1125,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_YM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].ym); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].ym); + region->len -= be32_to_cpu(adsp2_alg[i].ym); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region YM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -774,7 +1142,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_ZM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].zm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].zm); + region->len -= be32_to_cpu(adsp2_alg[i].zm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } break; } } @@ -986,6 +1363,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = w->codec; struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; + struct wm_coeff_ctl *ctl; int ret; int val; @@ -1023,7 +1401,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp); + ret = wm_adsp_setup_algs(dsp, codec); if (ret != 0) goto err; @@ -1031,6 +1409,16 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + /* Initialize caches for enabled and non-dirty controls */ + ret = wm_coeff_init_control_caches(dsp->wm_coeff); + if (ret != 0) + goto err; + + /* Sync dirty controls */ + ret = wm_coeff_sync_controls(dsp->wm_coeff); + if (ret != 0) + goto err; + /* Start the core running */ regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_CORE_ENA | ADSP1_START, @@ -1047,6 +1435,11 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, 0); + + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + ctl->enabled = 0; + } break; default: @@ -1099,6 +1492,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; struct wm_adsp_alg_region *alg_region; + struct wm_coeff_ctl *ctl; unsigned int val; int ret; @@ -1164,7 +1558,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp); + ret = wm_adsp_setup_algs(dsp, codec); if (ret != 0) goto err; @@ -1172,6 +1566,16 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + /* Initialize caches for enabled and non-dirty controls */ + ret = wm_coeff_init_control_caches(dsp->wm_coeff); + if (ret != 0) + goto err; + + /* Sync dirty controls */ + ret = wm_coeff_sync_controls(dsp->wm_coeff); + if (ret != 0) + goto err; + ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, ADSP2_CORE_ENA | ADSP2_START, @@ -1209,6 +1613,11 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ret); } + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + ctl->enabled = 0; + } + while (!list_empty(&dsp->alg_regions)) { alg_region = list_first_entry(&dsp->alg_regions, struct wm_adsp_alg_region, @@ -1247,36 +1656,48 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) INIT_LIST_HEAD(&adsp->alg_regions); + adsp->wm_coeff = kzalloc(sizeof(*adsp->wm_coeff), + GFP_KERNEL); + if (!adsp->wm_coeff) + return -ENOMEM; + adsp->wm_coeff->regmap = adsp->regmap; + adsp->wm_coeff->dev = adsp->dev; + INIT_LIST_HEAD(&adsp->wm_coeff->ctl_list); + if (dvfs) { adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); if (IS_ERR(adsp->dvfs)) { ret = PTR_ERR(adsp->dvfs); dev_err(adsp->dev, "Failed to get DCVDD: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_enable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to enable DCVDD: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_set_voltage(adsp->dvfs, 1200000, 1800000); if (ret != 0) { dev_err(adsp->dev, "Failed to initialise DVFS: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_disable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to disable DCVDD: %d\n", ret); - return ret; + goto out_coeff; } } return 0; + +out_coeff: + kfree(adsp->wm_coeff); + return ret; } EXPORT_SYMBOL_GPL(wm_adsp2_init); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index fea5146..6e890b9 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -30,6 +30,7 @@ struct wm_adsp_alg_region { unsigned int alg; int type; unsigned int base; + size_t len; }; struct wm_adsp { @@ -55,6 +56,8 @@ struct wm_adsp { bool running; struct regulator *dvfs; + + struct wm_coeff *wm_coeff; }; #define WM_ADSP1(wname, num) \ -- 1.8.2.1

2 1

[alsa-devel] [PATCH 0/7] Add capture support to compress API
by Charles Keepax 17 Apr '13

17 Apr '13

This chain of patches does some clearing up and add capture support to the compressed API. Missing is support for capture streams that don't use the copy callback. Also I have included two patches at the end of the chain that remove the buffer pointers and just use the cumulative values however as our setup uses the copy callback these are untested past build so should be taken with caution. Thanks, Charles Charles Keepax (7): ALSA: compress_core: Update calc_avail to use cumulative values ALSA: compress_core: Calculate avail correctly for capture streams ALSA: compress_core: Deconstify copy callback buffer ALSA: compress_core: Add support for capture streams ASoC: soc-compress: Deduce stream direction ALSA: compress_core: Remove unused hw_pointer ALSA: compress_core: Rework writes to use cumulative values include/sound/compress_driver.h | 4 +- sound/core/compress_offload.c | 117 +++++++++++++++++++++++++------------- sound/soc/soc-compress.c | 11 +++- 3 files changed, 87 insertions(+), 45 deletions(-) -- 1.7.2.5

4 10

Re: [alsa-devel] [RFC] sound: ASoC: multiple DAIs/codecs on single audio stream
by Mark Brown 17 Apr '13

17 Apr '13

On Tue, Apr 16, 2013 at 07:45:08PM +0200, Sebastian Hesselbarth wrote: > The controller has a dma controller for one playback buffer. It can > play the stream to both I2S and SPDIF interfaces with independent mute. > Now, for example, I have a board with i2s codec and spdif tx diode > connected to the above controller: > DMA (platform) -+-> I2S (dai) ----> I2S_Codec (codec) > | > +-> SPDIF (dai) --> SPDIF_TX (codec) > Is there any way in the current ASoC API to have the above setup? As discussed on IRC this is probably best represented using soc-pcm at the minute.

1 0

[alsa-devel] [PATCH v3 00/15] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 17 Apr '13

17 Apr '13

Hi, This series adds DT support for phycore-ac97. It is now based on the pcm dma cleanups from Lars. Beside ac97 support, the series adds fsl-ssi imx-pcm-fiq and generic DMA binding handling. @Lars: There are 4 new patches at the beginning of the series that fix some problems of your cleanups. I think they were in the changes you posted. If not, could you please add them to your branch? Regards, Markus Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (15): ASoC: imx-ssi: Fix includes ASoC: dmaengine-pcm: Replace of specific helper function ASoC: dmaengine-pcm: Fix function name ASoC: dmaengine-pcm: Fix pcm_request_chan function name ASoC: phycore-ac97: Add DT support ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl-ssi: Add SACNT definitions ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ARM: imx: Export ac97 reset functions ASoC: fsl-ssi: imx ac97 support ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97 ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 7 + Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt | 12 ++ arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- include/sound/dmaengine_pcm.h | 2 +- sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl_ssi.c | 484 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++------------- sound/soc/fsl/fsl_ssi.h | 8 ++ sound/soc/fsl/imx-pcm-dma.c | 4 +- sound/soc/fsl/imx-pcm-fiq.c | 16 ++- sound/soc/fsl/imx-pcm.h | 9 ++ sound/soc/fsl/imx-ssi.c | 7 +- sound/soc/fsl/imx-ssi.h | 2 + sound/soc/fsl/phycore-ac97.c | 184 +++++++++++++++++++++++---- sound/soc/soc-generic-dmaengine-pcm.c | 9 +- sound/soc/spear/spear_pcm.c | 2 +- 16 files changed, 634 insertions(+), 128 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (92%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt

4 22

[alsa-devel] Fwd: Solved Hercules RMX2
by Daniel Schürmann 17 Apr '13

17 Apr '13

Hi Daniel, thank your advice! I am not a Hercules developer, I am developer of the Mixxx DJ Software and I am just trying to make the thing work! I have just read within the USB 2.0 speck to find out if there are specific timeouts to deal with. I have found: § 9.2.6.4 Standard Device Requests If I understand it correct, there is a 5-second limit for storing data into the device. I think changing the sample rate is such a operation. So what about changing the Limit to 5000 ms for all "Set requests" bmRequest = 21. See Audio20 § 5.2.2 All other timeouts might be reduced to 500 ms but I would leave them untouched at 1000 ms , just in case. I would prefer this general solution over a device specific one. What do you think? Kind regards, Daniel 2013/4/16 Daniel Mack <zonque(a)gmail.com> > On 16.04.2013 21:23, Daniel Mack wrote: > > On 16.04.2013 21:14, Daniel Schürmann wrote: > > >> Against which Kernel should I patch? > > > > > https://git.kernel.org/cgit/linux/kernel/git/broonie/sound.git/log/?h=for-n… > > Eh, sorry. This one: > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/ for-next > >

2 1

[alsa-devel] [PATCH] ASoC: max98088: Fix logging of hardware revision.
by Dylan Reid 17 Apr '13

17 Apr '13

The hardware revision of the codec is based at 0x40. Subtract that before convering to ASCII. The same as it is done for 98095. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/max98088.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 3a7b7fd..3eeada5 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -2024,7 +2024,7 @@ static int max98088_probe(struct snd_soc_codec *codec) ret); goto err_access; } - dev_info(codec->dev, "revision %c\n", ret + 'A'); + dev_info(codec->dev, "revision %c\n", ret - 0x40 + 'A'); snd_soc_write(codec, M98088_REG_51_PWR_SYS, M98088_PWRSV); -- 1.8.1.3.605.g02339dd

2 1

[alsa-devel] [PATCH] ASoC: define playback and capture streams in dummy codec
by Stas Sergeev 17 Apr '13

17 Apr '13

Hi. This patch adds a playback and capture streams to the dummy codec DAI configuration. Most permissive set of sampling rates and formats is used. Signed-off-by: Stas Sergeev <stsp(a)users.sourceforge.net>

5 12

Re: [alsa-devel] hda: sink info in case of AMD Radeon GPU card (no ELD?)
by Rafał Miłecki 17 Apr '13

17 Apr '13

2013/1/19 Rafał Miłecki <zajec5(a)gmail.com>: > Hello, > > I'm interested in polishing HDMI audio support for Radeon cards and I > hope to get some help here. One of the current problems is that hda > has no information about TV audio capabilities. It results in a well > known: > ALSA sound/pci/hda/hda_eld.c:337 HDMI: ELD buf size is 0, force 128 > ALSA sound/pci/hda/hda_eld.c:356 HDMI: invalid ELD data byte 0 > It's expected, as radeon (GPU driver) doesn't pass ELD struct anywhere > to the GPU and it's not done automatically neither. > > Unfortunately it doesn't seem to be trivial to solve. You may be > familiar with other GPUs like Nvidia or Intel, where it's done in a > pretty simple way, for example: >> for (i = 0; i < len; i++) >> I915_WRITE(hdmiw_hdmiedid, *((uint32_t *)eld + i)); > but it can't be handled that way on Radeon cards. > > If you take a look at evergreend.h [0] you can see we can't simply > write ELD to the GPU buffer, but we have to fill specific registers > which look more like SADs (Short Audio Descriptors). > > My question is how to handle this on the alsa side. We can't just read > ELD, we probably have to read some registers from the audio card. Is > something like that implemented anywhere else? Did anyone started > working on it? Anyone interested in it? > > [0] http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob;f=drive… I tried hacking this sink info problem again. I've installed "fglrx" on my AMD Radeon HD 6970M (BARTS == DCE5) and verified it sets SAD registers: 5f84 7f077f07 5f88 00500705 5f8c 00000001 5f90 00000001 5f94 00000001 5f98 00000001 5f9c 00c01e05 5fa0 00000001 5fa4 00000001 5fa8 00000607 5fac 00017e07 5fb0 00007e07 5fb4 00000001 5fb8 00000001 Reading ELD of course doesn't work (that's why we get warning in dmesg) and I did some manual confirmation with hda-verb (reading ELD length): $ sudo ./hda-verb /dev/snd/hwC1D0 3 0xf2e 0x8 nid = 0x3, verb = 0xf2e, param = 0x8 value = 0x0 Plus (just for sure) reading first byte of ELD (in case length is always zero, but data is available): $ sudo ./hda-verb /dev/snd/hwC1D0 3 0xf2f 0 | grep val value = 0x0 $ sudo ./hda-verb /dev/snd/hwC1D0 3 0xf2f 1 | grep val value = 0x0 $ sudo ./hda-verb /dev/snd/hwC1D0 3 0xf2f 2 | grep val value = 0x0 So it seems ELD is really not available in this configuration. I thought about vendor specific verbs. Maybe values from the GPU registers are just copied to the audio? I've tried reading every possible vendor specific verb with the attached script (verbs.php). Unfortunately any verb didn't return anything interesting :( > grep value verbs.txt | sort | uniq -c 9336832 value = 0x0 18432 value = 0x1 6144 value = 0x40 2048 value = 0x415f 55296 value = 0x86 2048 value = 0xa6 16384 value = 0xb6 Does anyone have any idea how we can/should read that values from GPU in the audio driver? :| I guess the must be copied somewhere, but can't find the correct place. -- Rafał

1 0

Re: [alsa-devel] Enhance support for SigmaDSP chips
by Lars-Peter Clausen 17 Apr '13

17 Apr '13

On 04/11/2013 11:56 AM, Daniel Mack wrote: > Hi Lars, > Hi Cliff, > > I'm looking into the SigmaDSP AD1701 support, as a new prospective > project will use it. What's going to be used on this chip is not the > DAC/ADC features though, but the internal, programmable DSP things, and > designers will use the SigmaStudio IDE to generate the firmware. > > So I'm wondering how to support this kind of chip properly in ALSA, > which is not straight-forward, as the controls exported by it are > specific to the loaded firmware of course. Also, the IDE is free to > re-allocate every sub-addresses of its controls when something changes > in the layout of the building blocks. > > One idea I have in mind is to have a little parser script that reads the > generated sources of the IDE and dump a device-tree node snippet which > can then be put into the final DT. The driver would need to learn about > how to interpret that DT nodes, which will most likely just contain a > name, a sub-address, along with some mask and shift values. > > I wanted to ask for your opinion on that before I start. Have you ever > used any of the DSP features from ALSA/ASoC? Are there any pitfalls I > should be aware of? Is anyone actively working on improvements on the > driver? > > An alternative approach is to write a userspace library of course, but > that's going to be problematic once anybody wants to use the DSP > features in parallel to the DAC/ADC. Hi, I just yesterday wrote a small post how you can program the DSP registers. http://ez.analog.com/message/83094#83094 I'd like to be able to put the information about the different algorithms into the DSP firmware file itself, so we can auto instantiate the controls from the driver. The problem is that it is not so easy to generalize the export of the algorithms from SigmaStudio. Another problem is that you don't know how to calculate the algorithm parameters at runtime for some of the more complex algorithms. I don't think it's a good idea to put the control info into the devicetree and would rather prefer to see them go into the firmware. But as I said auto-generating this from exported SigmaStudio files is not so easy. So maybe manually creating the control info table might be an option. Btw. in case you haven't seen it the SigmaTcp tools is quite useful during the development of the firmware since it allows you to connect SigmaStudio directly to the DSP on the board. http://wiki.analog.com/resources/tools-software/linux-software/sigmatcp - Lars

3 4

[alsa-devel] [PATCH] ASoC: wm_adsp: Expose coefficient blocks as ALSA binary controls
by Dimitris Papastamos 17 Apr '13

17 Apr '13

Add initial support for runtime tuning for the ADSP cores. This is achieved by exposing the coefficient configuration blocks as ALSA binary controls. The current code assumes that no controls on the DSP are volatile. Change-Id: I9cd57efabe6c195c731f5a17f53617a99d2f09da Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 435 ++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 3 + 2 files changed, 431 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3470b64..080d05f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -215,6 +215,36 @@ static struct { [WM_ADSP_FW_RX_ANC] = { .file = "rx-anc" }, }; +struct wm_coeff_ctl_ops { + int (*xget)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*xput)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*xinfo)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +}; + +struct wm_coeff { + struct device *dev; + struct list_head ctl_list; + struct regmap *regmap; +}; + +struct wm_coeff_ctl { + const char *name; + struct snd_card *card; + struct wm_adsp_alg_region region; + struct wm_coeff_ctl_ops ops; + struct wm_adsp *adsp; + void *private; + unsigned int enabled:1; + struct list_head list; + void *cache; + size_t len; + unsigned int dirty:1; + struct snd_kcontrol *kcontrol; +}; + static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -334,6 +364,175 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, } } +static int wm_coeff_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = ctl->len; + return 0; +} + +static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, + const void *buf, size_t len) +{ + struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_adsp_alg_region *region = &ctl->region; + const struct wm_adsp_region *mem; + struct wm_adsp *adsp = ctl->adsp; + void *scratch; + int ret; + unsigned int reg; + + mem = wm_adsp_find_region(adsp, region->type); + if (!mem) { + adsp_err(adsp, "No base for region %x\n", + region->type); + return -EINVAL; + } + + reg = ctl->region.base; + reg = wm_adsp_region_to_reg(mem, reg); + + scratch = kmemdup(buf, ctl->len, GFP_KERNEL | GFP_DMA); + if (!scratch) + return -ENOMEM; + + ret = regmap_raw_write(wm_coeff->regmap, reg, scratch, + ctl->len); + if (ret) { + adsp_err(adsp, "Failed to write %zu bytes to %x\n", + ctl->len, reg); + kfree(scratch); + return ret; + } + + kfree(scratch); + + return 0; +} + +static int wm_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + char *p = ucontrol->value.bytes.data; + + memcpy(ctl->cache, p, ctl->len); + + if (!ctl->enabled) { + ctl->dirty = 1; + return 0; + } + + return wm_coeff_write_control(kcontrol, p, ctl->len); +} + +static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, + void *buf, size_t len) +{ + struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_adsp_alg_region *region = &ctl->region; + const struct wm_adsp_region *mem; + struct wm_adsp *adsp = ctl->adsp; + void *scratch; + int ret; + unsigned int reg; + + mem = wm_adsp_find_region(adsp, region->type); + if (!mem) { + adsp_err(adsp, "No base for region %x\n", + region->type); + return -EINVAL; + } + + reg = ctl->region.base; + reg = wm_adsp_region_to_reg(mem, reg); + + scratch = kmalloc(ctl->len, GFP_KERNEL | GFP_DMA); + if (!scratch) + return -ENOMEM; + + ret = regmap_raw_read(wm_coeff->regmap, reg, scratch, ctl->len); + if (ret) { + adsp_err(adsp, "Failed to read %zu bytes from %x\n", + ctl->len, reg); + kfree(scratch); + return ret; + } + + memcpy(buf, scratch, ctl->len); + kfree(scratch); + + return 0; +} + +static int wm_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + char *p = ucontrol->value.bytes.data; + + memcpy(p, ctl->cache, ctl->len); + return 0; +} + +static int wm_coeff_add_kcontrol(struct wm_coeff *wm_coeff, + struct wm_coeff_ctl *ctl, + const struct snd_kcontrol_new *kctl) +{ + int ret; + struct snd_kcontrol *kcontrol; + + kcontrol = snd_ctl_new1(kctl, wm_coeff); + ret = snd_ctl_add(ctl->card, kcontrol); + if (ret < 0) { + dev_err(wm_coeff->dev, "Failed to add %s: %d\n", + kctl->name, ret); + return ret; + } + ctl->kcontrol = kcontrol; + return 0; +} + +static int wmfw_add_ctl(struct wm_coeff *wm_coeff, + struct wm_coeff_ctl *ctl) +{ + struct snd_kcontrol_new *kcontrol; + int ret; + + if (!wm_coeff || !ctl || !ctl->name || !ctl->card) + return -EINVAL; + + kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); + if (!kcontrol) + return -ENOMEM; + kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + + kcontrol->name = ctl->name; + kcontrol->info = wm_coeff_info; + kcontrol->get = wm_coeff_get; + kcontrol->put = wm_coeff_put; + kcontrol->private_value = (unsigned long)ctl; + + ret = wm_coeff_add_kcontrol(wm_coeff, + ctl, kcontrol); + if (ret < 0) + goto err_kcontrol; + + kfree(kcontrol); + + list_add(&ctl->list, &wm_coeff->ctl_list); + return 0; + +err_kcontrol: + kfree(kcontrol); + return ret; +} + static int wm_adsp_load(struct wm_adsp *dsp) { LIST_HEAD(buf_list); @@ -547,7 +746,140 @@ out: return ret; } -static int wm_adsp_setup_algs(struct wm_adsp *dsp) +static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff) +{ + struct wm_coeff_ctl *ctl; + int ret; + + list_for_each_entry(ctl, &wm_coeff->ctl_list, + list) { + if (!ctl->enabled || ctl->dirty) + continue; + ret = wm_coeff_read_control(ctl->kcontrol, + ctl->cache, + ctl->len); + if (ret < 0) + return ret; + } + + return 0; +} + +static int wm_coeff_sync_controls(struct wm_coeff *wm_coeff) +{ + struct wm_coeff_ctl *ctl; + int ret; + + list_for_each_entry(ctl, &wm_coeff->ctl_list, + list) { + if (!ctl->enabled) + continue; + if (ctl->dirty) { + ret = wm_coeff_write_control(ctl->kcontrol, + ctl->cache, + ctl->len); + if (ret < 0) + return ret; + ctl->dirty = 0; + } + } + + return 0; +} + +static int wm_adsp_create_control(struct snd_soc_codec *codec, + const struct wm_adsp_alg_region *region) + +{ + struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); + struct wm_coeff_ctl *ctl; + char *name; + char *region_name; + int ret; + + name = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!name) + return -ENOMEM; + + switch (region->type) { + case WMFW_ADSP1_PM: + region_name = "PM"; + break; + case WMFW_ADSP1_DM: + region_name = "DM"; + break; + case WMFW_ADSP2_XM: + region_name = "XM"; + break; + case WMFW_ADSP2_YM: + region_name = "YM"; + break; + case WMFW_ADSP1_ZM: + region_name = "ZM"; + break; + default: + return -EINVAL; + } + + snprintf(name, PAGE_SIZE, "DSP%d %s %x", + dsp->num, region_name, region->alg); + + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + if (!strcmp(ctl->name, name)) { + if (!ctl->enabled) + ctl->enabled = 1; + return 0; + } + } + + ctl = kzalloc(sizeof(*ctl), GFP_KERNEL); + if (!ctl) { + ret = -ENOMEM; + goto err_name; + } + ctl->region = *region; + ctl->name = kmemdup(name, strlen(name) + 1, GFP_KERNEL); + if (!ctl->name) { + ret = -ENOMEM; + goto err_ctl; + } + ctl->enabled = 1; + ctl->dirty = 0; + ctl->ops.xget = wm_coeff_get; + ctl->ops.xput = wm_coeff_put; + ctl->card = codec->card->snd_card; + ctl->adsp = dsp; + + ctl->len = region->len; + ctl->cache = kzalloc(ctl->len, GFP_KERNEL); + if (!ctl->cache) { + ret = -ENOMEM; + goto err_ctl_name; + } + + ret = wmfw_add_ctl(dsp->wm_coeff, ctl); + if (ret < 0) { + ret = -ENOMEM; + goto err_ctl_cache; + } + + kfree(name); + + return 0; + +err_ctl_cache: + kfree(ctl->cache); +err_ctl_name: + kfree(ctl->name); +err_ctl: + kfree(ctl); +err_name: + kfree(name); + return ret; +} + +static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) { struct regmap *regmap = dsp->regmap; struct wmfw_adsp1_id_hdr adsp1_id; @@ -730,7 +1062,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP1_DM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].dm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp1_alg[i + 1].dm); + region->len -= be32_to_cpu(adsp1_alg[i].dm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region DM with ID %x\n", + be32_to_cpu(adsp1_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -738,7 +1079,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP1_ZM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].zm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp1_alg[i + 1].zm); + region->len -= be32_to_cpu(adsp1_alg[i].zm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", + be32_to_cpu(adsp1_alg[i].alg.id)); + } break; case WMFW_ADSP2: @@ -758,7 +1108,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_XM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].xm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].xm); + region->len -= be32_to_cpu(adsp2_alg[i].xm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region XM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -766,7 +1125,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_YM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].ym); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].ym); + region->len -= be32_to_cpu(adsp2_alg[i].ym); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region YM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -774,7 +1142,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_ZM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].zm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].zm); + region->len -= be32_to_cpu(adsp2_alg[i].zm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } break; } } @@ -986,6 +1363,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = w->codec; struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; + struct wm_coeff_ctl *ctl; int ret; int val; @@ -1023,7 +1401,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp); + ret = wm_adsp_setup_algs(dsp, codec); if (ret != 0) goto err; @@ -1031,6 +1409,16 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + /* Initialize caches for enabled and non-dirty controls */ + ret = wm_coeff_init_control_caches(dsp->wm_coeff); + if (ret != 0) + goto err; + + /* Sync dirty controls */ + ret = wm_coeff_sync_controls(dsp->wm_coeff); + if (ret != 0) + goto err; + /* Start the core running */ regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_CORE_ENA | ADSP1_START, @@ -1047,6 +1435,11 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, 0); + + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + ctl->enabled = 0; + } break; default: @@ -1099,6 +1492,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; struct wm_adsp_alg_region *alg_region; + struct wm_coeff_ctl *ctl; unsigned int val; int ret; @@ -1164,7 +1558,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp); + ret = wm_adsp_setup_algs(dsp, codec); if (ret != 0) goto err; @@ -1172,6 +1566,16 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + /* Initialize caches for enabled and non-dirty controls */ + ret = wm_coeff_init_control_caches(dsp->wm_coeff); + if (ret != 0) + goto err; + + /* Sync dirty controls */ + ret = wm_coeff_sync_controls(dsp->wm_coeff); + if (ret != 0) + goto err; + ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, ADSP2_CORE_ENA | ADSP2_START, @@ -1209,6 +1613,11 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ret); } + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + ctl->enabled = 0; + } + while (!list_empty(&dsp->alg_regions)) { alg_region = list_first_entry(&dsp->alg_regions, struct wm_adsp_alg_region, @@ -1247,36 +1656,48 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) INIT_LIST_HEAD(&adsp->alg_regions); + adsp->wm_coeff = kzalloc(sizeof(*adsp->wm_coeff), + GFP_KERNEL); + if (!adsp->wm_coeff) + return -ENOMEM; + adsp->wm_coeff->regmap = adsp->regmap; + adsp->wm_coeff->dev = adsp->dev; + INIT_LIST_HEAD(&adsp->wm_coeff->ctl_list); + if (dvfs) { adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); if (IS_ERR(adsp->dvfs)) { ret = PTR_ERR(adsp->dvfs); dev_err(adsp->dev, "Failed to get DCVDD: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_enable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to enable DCVDD: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_set_voltage(adsp->dvfs, 1200000, 1800000); if (ret != 0) { dev_err(adsp->dev, "Failed to initialise DVFS: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_disable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to disable DCVDD: %d\n", ret); - return ret; + goto out_coeff; } } return 0; + +out_coeff: + kfree(adsp->wm_coeff); + return ret; } EXPORT_SYMBOL_GPL(wm_adsp2_init); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index fea5146..6e890b9 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -30,6 +30,7 @@ struct wm_adsp_alg_region { unsigned int alg; int type; unsigned int base; + size_t len; }; struct wm_adsp { @@ -55,6 +56,8 @@ struct wm_adsp { bool running; struct regulator *dvfs; + + struct wm_coeff *wm_coeff; }; #define WM_ADSP1(wname, num) \ -- 1.8.2.1

1 0

[alsa-devel] Can Realtek HD-A codec ALC282 support D3cold?
by Lin, Mengdong 17 Apr '13

17 Apr '13

Hi, (Sorry to post this message again) We want to put an idle HD-A codec into D3cold for more power saving and preparation for being power gated. The Realtek ALC282 is used in our test. Could anyone help to confirm whether ALC282 can support D3cold or not? I'm told that ALC282 can support D3cold. But my test shows it cannot. At first, ALC282 HW does not report that it can support D3cold. Its parameter "Supported Power States" returns 0xc000000f, means it only supports: EPSS, CLKSTOP, D3hot, D2, D1 and D0. Is this a just mistake in reporting the bit, or D3cold is not supported all? Then when I explicitly set the audio function group to D3cold by verb 0x705, and then get its power state after 200ms by verb 0xf05, The codec response is 0x00000000: No PS-Error, both PS-Act and PS-set is D0. So it seems codec HW does not process the set power state command at all. Is there someone from Realtek or familiar with ALC282 here can share some information about this issue? Many thanks Mengdong

2 1

Re: [alsa-devel] [PATCH 04/28] proc: Supply PDE attribute setting accessor functions [RFC]
by Mauro Carvalho Chehab 16 Apr '13

16 Apr '13

Em 16-04-2013 15:26, David Howells escreveu: > Supply accessor functions to set attributes in proc_dir_entry structs. > > The following are supplied: proc_set_size() and proc_set_user(). > > Signed-off-by: David Howells <dhowells(a)redhat.com> > cc: linuxppc-dev(a)lists.ozlabs.org > cc: linux-media(a)vger.kernel.org > cc: netdev(a)vger.kernel.org > cc: linux-wireless(a)vger.kernel.org > cc: linux-pci(a)vger.kernel.org > cc: netfilter-devel(a)vger.kernel.org > cc: alsa-devel(a)alsa-project.org > --- > > arch/powerpc/kernel/proc_powerpc.c | 2 +- > arch/powerpc/platforms/pseries/reconfig.c | 2 +- > drivers/media/pci/ttpci/av7110_ir.c | 2 +- Weird that av7110 IR uses /proc... Well, this is an old, obsolete driver for hardware that is not sold anymore for a long time... So, be it. Acked-by: Mauro Carvalho Chehab <mchehab(a)redhat.com> > drivers/net/irda/vlsi_ir.c | 2 +- > drivers/net/wireless/airo.c | 34 +++++++++-------------------- > drivers/pci/proc.c | 2 +- > fs/proc/generic.c | 13 +++++++++++ > include/linux/proc_fs.h | 5 ++++ > kernel/configs.c | 2 +- > kernel/profile.c | 2 +- > net/netfilter/xt_recent.c | 3 +-- > sound/core/info.c | 2 +- > 12 files changed, 38 insertions(+), 33 deletions(-) > > diff --git a/arch/powerpc/kernel/proc_powerpc.c b/arch/powerpc/kernel/proc_powerpc.c > index 41d8ee9..feb8580 100644 > --- a/arch/powerpc/kernel/proc_powerpc.c > +++ b/arch/powerpc/kernel/proc_powerpc.c > @@ -83,7 +83,7 @@ static int __init proc_ppc64_init(void) > &page_map_fops, vdso_data); > if (!pde) > return 1; > - pde->size = PAGE_SIZE; > + proc_set_size(pde, PAGE_SIZE); > > return 0; > } > diff --git a/arch/powerpc/platforms/pseries/reconfig.c b/arch/powerpc/platforms/pseries/reconfig.c > index d6491bd..f93cdf5 100644 > --- a/arch/powerpc/platforms/pseries/reconfig.c > +++ b/arch/powerpc/platforms/pseries/reconfig.c > @@ -452,7 +452,7 @@ static int proc_ppc64_create_ofdt(void) > > ent = proc_create("powerpc/ofdt", S_IWUSR, NULL, &ofdt_fops); > if (ent) > - ent->size = 0; > + proc_set_size(ent, 0); > > return 0; > } > diff --git a/drivers/media/pci/ttpci/av7110_ir.c b/drivers/media/pci/ttpci/av7110_ir.c > index eb82286..0e763a7 100644 > --- a/drivers/media/pci/ttpci/av7110_ir.c > +++ b/drivers/media/pci/ttpci/av7110_ir.c > @@ -375,7 +375,7 @@ int av7110_ir_init(struct av7110 *av7110) > if (av_cnt == 1) { > e = proc_create("av7110_ir", S_IWUSR, NULL, &av7110_ir_proc_fops); > if (e) > - e->size = 4 + 256 * sizeof(u16); > + proc_set_size(e, 4 + 256 * sizeof(u16)); > } > > tasklet_init(&av7110->ir.ir_tasklet, av7110_emit_key, (unsigned long) &av7110->ir); > diff --git a/drivers/net/irda/vlsi_ir.c b/drivers/net/irda/vlsi_ir.c > index e22cd4e..5f47584 100644 > --- a/drivers/net/irda/vlsi_ir.c > +++ b/drivers/net/irda/vlsi_ir.c > @@ -1678,7 +1678,7 @@ vlsi_irda_probe(struct pci_dev *pdev, const struct pci_device_id *id) > IRDA_WARNING("%s: failed to create proc entry\n", > __func__); > } else { > - ent->size = 0; > + proc_set_size(ent, 0); > } > idev->proc_entry = ent; > } > diff --git a/drivers/net/wireless/airo.c b/drivers/net/wireless/airo.c > index 66e398d..21d0233 100644 > --- a/drivers/net/wireless/airo.c > +++ b/drivers/net/wireless/airo.c > @@ -4507,73 +4507,63 @@ static int setup_proc_entry( struct net_device *dev, > airo_entry); > if (!apriv->proc_entry) > goto fail; > - apriv->proc_entry->uid = proc_kuid; > - apriv->proc_entry->gid = proc_kgid; > + proc_set_user(apriv->proc_entry, proc_kuid, proc_kgid); > > /* Setup the StatsDelta */ > entry = proc_create_data("StatsDelta", S_IRUGO & proc_perm, > apriv->proc_entry, &proc_statsdelta_ops, dev); > if (!entry) > goto fail_stats_delta; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Stats */ > entry = proc_create_data("Stats", S_IRUGO & proc_perm, > apriv->proc_entry, &proc_stats_ops, dev); > if (!entry) > goto fail_stats; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Status */ > entry = proc_create_data("Status", S_IRUGO & proc_perm, > apriv->proc_entry, &proc_status_ops, dev); > if (!entry) > goto fail_status; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Config */ > entry = proc_create_data("Config", proc_perm, > apriv->proc_entry, &proc_config_ops, dev); > if (!entry) > goto fail_config; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the SSID */ > entry = proc_create_data("SSID", proc_perm, > apriv->proc_entry, &proc_SSID_ops, dev); > if (!entry) > goto fail_ssid; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the APList */ > entry = proc_create_data("APList", proc_perm, > apriv->proc_entry, &proc_APList_ops, dev); > if (!entry) > goto fail_aplist; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the BSSList */ > entry = proc_create_data("BSSList", proc_perm, > apriv->proc_entry, &proc_BSSList_ops, dev); > if (!entry) > goto fail_bsslist; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the WepKey */ > entry = proc_create_data("WepKey", proc_perm, > apriv->proc_entry, &proc_wepkey_ops, dev); > if (!entry) > goto fail_wepkey; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > - > + proc_set_user(entry, proc_kuid, proc_kgid); > return 0; > > fail_wepkey: > @@ -5695,10 +5685,8 @@ static int __init airo_init_module( void ) > > airo_entry = proc_mkdir_mode("driver/aironet", airo_perm, NULL); > > - if (airo_entry) { > - airo_entry->uid = proc_kuid; > - airo_entry->gid = proc_kgid; > - } > + if (airo_entry) > + proc_set_user(airo_entry, proc_kuid, proc_kgid); > > for (i = 0; i < 4 && io[i] && irq[i]; i++) { > airo_print_info("", "Trying to configure ISA adapter at irq=%d " > diff --git a/drivers/pci/proc.c b/drivers/pci/proc.c > index 12e4fb5..7cde7c1 100644 > --- a/drivers/pci/proc.c > +++ b/drivers/pci/proc.c > @@ -419,7 +419,7 @@ int pci_proc_attach_device(struct pci_dev *dev) > &proc_bus_pci_operations, dev); > if (!e) > return -ENOMEM; > - e->size = dev->cfg_size; > + proc_set_size(e, dev->cfg_size); > dev->procent = e; > > return 0; > diff --git a/fs/proc/generic.c b/fs/proc/generic.c > index 1c07cad..5f6f6c3 100644 > --- a/fs/proc/generic.c > +++ b/fs/proc/generic.c > @@ -498,6 +498,19 @@ out: > return NULL; > } > EXPORT_SYMBOL(proc_create_data); > + > +void proc_set_size(struct proc_dir_entry *de, loff_t size) > +{ > + de->size = size; > +} > +EXPORT_SYMBOL(proc_set_size); > + > +void proc_set_user(struct proc_dir_entry *de, kuid_t uid, kgid_t gid) > +{ > + de->uid = uid; > + de->gid = gid; > +} > +EXPORT_SYMBOL(proc_set_user); > > static void free_proc_entry(struct proc_dir_entry *de) > { > diff --git a/include/linux/proc_fs.h b/include/linux/proc_fs.h > index 805edac..28a4d7e 100644 > --- a/include/linux/proc_fs.h > +++ b/include/linux/proc_fs.h > @@ -130,6 +130,9 @@ static inline struct proc_dir_entry *proc_create(const char *name, umode_t mode, > extern struct proc_dir_entry *proc_net_mkdir(struct net *net, const char *name, > struct proc_dir_entry *parent); > > +extern void proc_set_size(struct proc_dir_entry *, loff_t); > +extern void proc_set_user(struct proc_dir_entry *, kuid_t, kgid_t); > + > extern struct file *proc_ns_fget(int fd); > extern bool proc_ns_inode(struct inode *inode); > > @@ -158,6 +161,8 @@ static inline struct proc_dir_entry *proc_mkdir(const char *name, > struct proc_dir_entry *parent) {return NULL;} > static inline struct proc_dir_entry *proc_mkdir_mode(const char *name, > umode_t mode, struct proc_dir_entry *parent) { return NULL; } > +static inline void proc_set_size(struct proc_dir_entry *de, loff_t size) {} > +static inline void proc_set_user(struct proc_dir_entry *de, kuid_t uid, kgid_t gid) {} > > struct tty_driver; > static inline void proc_tty_register_driver(struct tty_driver *driver) {}; > diff --git a/kernel/configs.c b/kernel/configs.c > index 42e8fa0..c18b1f1 100644 > --- a/kernel/configs.c > +++ b/kernel/configs.c > @@ -79,7 +79,7 @@ static int __init ikconfig_init(void) > if (!entry) > return -ENOMEM; > > - entry->size = kernel_config_data_size; > + proc_set_size(entry, kernel_config_data_size); > > return 0; > } > diff --git a/kernel/profile.c b/kernel/profile.c > index 524ce5e..0bf4007 100644 > --- a/kernel/profile.c > +++ b/kernel/profile.c > @@ -600,7 +600,7 @@ int __ref create_proc_profile(void) /* false positive from hotcpu_notifier */ > NULL, &proc_profile_operations); > if (!entry) > return 0; > - entry->size = (1+prof_len) * sizeof(atomic_t); > + proc_set_size(entry, (1 + prof_len) * sizeof(atomic_t)); > hotcpu_notifier(profile_cpu_callback, 0); > return 0; > } > diff --git a/net/netfilter/xt_recent.c b/net/netfilter/xt_recent.c > index 3db2d38..1e657cf 100644 > --- a/net/netfilter/xt_recent.c > +++ b/net/netfilter/xt_recent.c > @@ -401,8 +401,7 @@ static int recent_mt_check(const struct xt_mtchk_param *par, > ret = -ENOMEM; > goto out; > } > - pde->uid = uid; > - pde->gid = gid; > + proc_set_user(pde, uid, gid); > #endif > spin_lock_bh(&recent_lock); > list_add_tail(&t->list, &recent_net->tables); > diff --git a/sound/core/info.c b/sound/core/info.c > index 3aa8864..c7f41c3 100644 > --- a/sound/core/info.c > +++ b/sound/core/info.c > @@ -970,7 +970,7 @@ int snd_info_register(struct snd_info_entry * entry) > mutex_unlock(&info_mutex); > return -ENOMEM; > } > - p->size = entry->size; > + proc_set_size(p, entry->size); > } > entry->p = p; > if (entry->parent) > > -- > 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 >

1 0

[alsa-devel] [PATCH] tinycompress: cplay: Correct a couple of small message typos
by Richard Fitzgerald 16 Apr '13

16 Apr '13

Change-Id: I2b25d006328f1a9458f232f77c61e509c9b54d3c Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- cplay.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/cplay.c b/cplay.c index 60995a4..d9364f4 100644 --- a/cplay.c +++ b/cplay.c @@ -337,7 +337,7 @@ void play_samples(char *name, unsigned int card, unsigned int device, goto BUF_EXIT; } if (wrote != num_read) { - /* TODO: Buufer pointer needs to be set here */ + /* TODO: Buffer pointer needs to be set here */ fprintf(stderr, "We wrote %d, DSP accepted %d\n", num_read, wrote); } if (verbose) { @@ -348,7 +348,7 @@ void play_samples(char *name, unsigned int card, unsigned int device, } while (num_read > 0); if (verbose) - printf("%s: exit sucess\n", __func__); + printf("%s: exit success\n", __func__); /* issue drain if it supports */ compress_drain(compress); free(buffer); -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: fsi: fixup sparse errors
by Kuninori Morimoto 16 Apr '13

16 Apr '13

This patch fixup below sparse errors ${LINUX}/sound/soc/sh/fsi.c:1459:9: \ error: incompatible types in conditional expression (different base types) ${LINUX}/sound/soc/sh/fsi.c:1634:25: \ error: incompatible types in conditional expression (different base types) ${LINUX}/sound/soc/sh/fsi.c:1639:17: \ error: incompatible types in conditional expression (different base types) ${LINUX}/sound/soc/sh/fsi.c:2093:9: \ error: incompatible types in conditional expression (different base types) ${LINUX}/sound/soc/sh/fsi.c:2105:9: \ error: incompatible types in conditional expression (different base types) Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index c724026a..7bc767b 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -276,7 +276,7 @@ struct fsi_stream_handler { int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev); int (*transfer)(struct fsi_priv *fsi, struct fsi_stream *io); int (*remove)(struct fsi_priv *fsi, struct fsi_stream *io); - void (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io, + int (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io, int enable); }; #define fsi_stream_handler_call(io, func, args...) \ @@ -1189,7 +1189,7 @@ static int fsi_pio_push(struct fsi_priv *fsi, struct fsi_stream *io) samples); } -static void fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, +static int fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, int enable) { struct fsi_master *master = fsi_get_master(fsi); @@ -1202,6 +1202,8 @@ static void fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, if (fsi_is_clk_master(fsi)) fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); + + return 0; } static int fsi_pio_push_init(struct fsi_priv *fsi, struct fsi_stream *io) @@ -1410,7 +1412,7 @@ static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) return 0; } -static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, +static int fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, int start) { struct fsi_master *master = fsi_get_master(fsi); @@ -1423,6 +1425,8 @@ static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, if (fsi_is_clk_master(fsi)) fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); + + return 0; } static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev) -- 1.7.9.5

1 0

[alsa-devel] about snd_pcm_writei() parameter
by Zhang wei 16 Apr '13

16 Apr '13

hi all. Is there have some limitation on the parameter size of the interface: snd_pcm_sframes_t snd_pcm_readi(snd_pcm_t *pcm, void *buffer, snd_pcm_uframes_t size)? should be the SIZE as any value ? For example, set the size < period_size, ok? does the parameter depend on the period_size? Thanks a lot. ---- BR zw

1 0

Re: [alsa-devel] [PATCH 01/17] ASoC: dmaengine-pcm: Make requesting the DMA channel at PCM open optional
by Lars-Peter Clausen 16 Apr '13

16 Apr '13

On 04/16/2013 01:28 AM, Ryan Mallon wrote: > On 16/04/13 03:19, Lars-Peter Clausen wrote: >> Refactor the dmaengine PCM library to allow the DMA channel to be requested >> before opening a PCM substream. snd_dmaengine_pcm_open() now expects a DMA >> channel instead of a filter function and filter parameter as its parameters. >> snd_dmaengine_pcm_close() is updated to not release the DMA channel. This allows >> a dmaengine based PCM driver to request its channels before the substream is >> opened. >> >> The patch also introduces two new functions, snd_dmaengine_pcm_open_request_chan() >> and snd_dmaengine_pcm_close_release_chan(), which have the same signature and >> behaviour of the old snd_dmaengine_pcm_{open,close}() and internally use the new >> variants of these functions. All users of snd_dmaengine_pcm_{open,close}() are >> updated to use snd_dmaengine_pcm_open_request_chan() and >> snd_dmaengine_pcm_close_release_chan(). >> >> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> > > <snip> > >> --- a/sound/soc/cirrus/ep93xx-pcm.c >> +++ b/sound/soc/cirrus/ep93xx-pcm.c >> @@ -69,7 +69,8 @@ static int ep93xx_pcm_open(struct snd_pcm_substream *substream) >> >> snd_soc_set_runtime_hwparams(substream, &ep93xx_pcm_hardware); >> >> - return snd_dmaengine_pcm_open(substream, ep93xx_pcm_dma_filter, >> + return snd_dmaengine_pcm_open_request_chan(substream, >> + ep93xx_pcm_dma_filter, >> snd_soc_dai_get_dma_data(rtd->cpu_dai, substream)); > > ep93xx_pcm_dma_filter is now unused and can be removed? It's still used in the same way as before. - Lars

1 0

Re: [alsa-devel] [PATCH 16/17] ASoC: ep93xx: Use generic dmaengine PCM
by Lars-Peter Clausen 16 Apr '13

16 Apr '13

On 04/16/2013 01:48 AM, Ryan Mallon wrote: > On 16/04/13 03:20, Lars-Peter Clausen wrote: >> Use the generic dmaengine PCM driver instead of a custom implementation. >> >> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> >> --- >> sound/soc/cirrus/Kconfig | 2 +- >> sound/soc/cirrus/ep93xx-i2s.c | 4 +- >> sound/soc/cirrus/ep93xx-pcm.c | 138 ++++-------------------------------------- >> 3 files changed, 14 insertions(+), 130 deletions(-) >> >> diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig >> index 88143db..2c20f01 100644 >> --- a/sound/soc/cirrus/Kconfig >> +++ b/sound/soc/cirrus/Kconfig >> @@ -1,7 +1,7 @@ >> config SND_EP93XX_SOC >> tristate "SoC Audio support for the Cirrus Logic EP93xx series" >> depends on ARCH_EP93XX && SND_SOC >> - select SND_SOC_DMAENGINE_PCM >> + select SND_SOC_GENERIC_DMAENGINE_PCM >> help >> Say Y or M if you want to add support for codecs attached to >> the EP93xx I2S or AC97 interfaces. >> diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c >> index 453a89c..ef731e6 100644 >> --- a/sound/soc/cirrus/ep93xx-i2s.c >> +++ b/sound/soc/cirrus/ep93xx-i2s.c >> @@ -140,8 +140,8 @@ static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream) >> >> static int ep93xx_i2s_dai_probe(struct snd_soc_dai *dai) >> { >> - dai->playback_dma_data = ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_PLAYBACK]; >> - dai->capute_dma_data = ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_CAPTURE] >> + dai->playback_dma_data = &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_PLAYBACK]; >> + dai->capture_dma_data = &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_CAPTURE]; > > Why did this change? I can't see a change to ep93xx_i2s_dma_data. Was > this just incorrect in the previous patch? All patches in a series > should be buildable/runnable to assist things like git bisect. > Yes, that part belongs to the previous patch and only ended up here by accident. >> >> return 0; >> } >> diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c >> index 4880326..0e9f56e 100644 >> --- a/sound/soc/cirrus/ep93xx-pcm.c >> +++ b/sound/soc/cirrus/ep93xx-pcm.c >> @@ -14,20 +14,14 @@ >> >> #include <linux/module.h> >> #include <linux/init.h> >> -#include <linux/device.h> >> -#include <linux/slab.h> >> +#include <linux/platform_device.h> >> #include <linux/dmaengine.h> >> -#include <linux/dma-mapping.h> >> >> -#include <sound/core.h> >> #include <sound/pcm.h> >> -#include <sound/pcm_params.h> >> #include <sound/soc.h> >> #include <sound/dmaengine_pcm.h> >> >> #include <linux/platform_data/dma-ep93xx.h> >> -#include <mach/hardware.h> >> -#include <mach/ep93xx-regs.h> > > Awesome. It looks like this header can also be removed from ep93xx-i2s.c > also, which gets us a step closer to making this header private to the > ep93xx soc code in arch/arm. > > ~Ryan >

1 0

[alsa-devel] [PATCH] ALSA: hda - Add codec delay to the capture time stamp.
by Dylan Reid 16 Apr '13

16 Apr '13

For capture, the delay through the codec contributes to the time stamp of the sample recorded at the A to D. Rename the codec time stamp function appropriately. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_intel.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 63734b5..7b213d5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1889,7 +1889,7 @@ static void azx_timecounter_init(struct snd_pcm_substream *substream, tc->cycle_last = last; } -static u64 azx_subtract_codec_delay(struct snd_pcm_substream *substream, +static u64 azx_adjust_codec_delay(struct snd_pcm_substream *substream, u64 nsec) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); @@ -1903,6 +1903,9 @@ static u64 azx_subtract_codec_delay(struct snd_pcm_substream *substream, codec_nsecs = div_u64(codec_frames * 1000000000LL, substream->runtime->rate); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + return nsec + codec_nsecs; + return (nsec > codec_nsecs) ? nsec - codec_nsecs : 0; } @@ -1914,7 +1917,7 @@ static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, nsec = timecounter_read(&azx_dev->azx_tc); nsec = div_u64(nsec, 3); /* can be optimized */ - nsec = azx_subtract_codec_delay(substream, nsec); + nsec = azx_adjust_codec_delay(substream, nsec); *ts = ns_to_timespec(nsec); -- 1.8.1.3.605.g02339dd

2 1

[alsa-devel] ASoC updates for v3.10
by Mark Brown 15 Apr '13

15 Apr '13

The following changes since commit b02e48f2f8755e51fe8ce246988a29bd96f6b85e: MAINTAINERS: Update e-mail address (2013-04-12 11:53:41 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.10 for you to fetch changes up to 5cbad7d39ad229c68a724e5e139fd845b93766b2: Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next (2013-04-12 13:57:31 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.10 A bunch of changes here, the most interesting one subsystem wise being Morimoto-san's work to create snd_soc_component which doesn't do much for now but will be pretty important going forwards: - Add a new component object type which will form the basis of moving to a more generic handling of SoC and off-SoC components, contributed by Kuninori Morimoto. - A fairly large set of cleanups for the dmaengine integration from Lars-Peter Clausen, starting to move towards being able to have a generic driver based on the library. - Performance optimisations to DAPM from Ryo Tsutsui. - Support for mixer control sharing in DAPM from Stephen Warren. - Multiplatform ARM cleanups from Arnd Bergmann. - New CODEC drivers for AK5385 and TAS5086 from Daniel Mack. ---------------------------------------------------------------- Alban Bedel (1): ASoC: wm8903: Add the DAC boost control Alexandru Gheorghiu (1): ASoC: core: Use PTR_RET function Andrey Smirnov (2): ASoC: si476x: Convert SI476X codec to use regmap ASoC: si476x: Cosmetic changes to SI476X codec driver Arnd Bergmann (4): ASoC: samsung: use irq resource for idma ASoC: samsung: fix module_device_table ASoC: samsung: export idma_reg_addr_init ASoC: samsung: move plat/ headers to local directory Charles Keepax (12): ASoC: arizona: Move selection of FLL REFCLK into init ASoC: arizona: Tidy up SYNCCLK selection and cache values ASoC: arizona: Factor out check for enabled FLL ASoC: arizona: Factor out FLL disable ASoC: arizona: Factor out FLL enable ASoC: arizona: Improve suppression of noop FLL updates ASoC: arizona: Add support for directly setting the FLL REFCLK ASoC: arizona: Add convience define for clearing SYNCCLK ASoC: arizona: Don't enable FLL on REFCLK configuration ASoC: wm5102: Correctly use SOC_VALUE_ENUM for ISRC FSL controls ASoC: wm0010: Constify usage of firmware filenames ASoC: wm0010: Report filename when we fail to load firmware Dan Carpenter (1): ASoC: tas5086: signedness bug in tas5086_hw_params() Daniel Mack (7): ASoC: ak4104: convert to direct regmap API usage ASoC: ak4104: correct tranceiver enable handling ASoC: Add codec driver for AK5386 ALSA: ASoC: add codec driver for TI TAS5086 ASoC: davinci-mcasp: clean up davinci_hw_common_param() ASoC: cs4271: convert to direct regmap API usage ASoC: cs4271: switch to mute_stream Dylan Reid (1): ASoC: max98088: Add TLV data for volume controls. Fabio Estevam (2): ASoC: fsl: imx-audmux: Use devm_clk_get() ASoC: fsl: imx-pcm-fiq: Use 'unsigned int' for period Heiko Stübner (1): ASoC: samsung: fix neo1973-wm8753 compilation Jeeja KP (1): ASoC: compress - add support for metadata apis Kuninori Morimoto (61): ASoC: core: tidyup snd_soc_register_codec() fail case ASoC: core: use snd_soc_register_dais() on codec ASoC: core: remove codec from list if registration failed ASoC: fsi: remove unused irq ASoC: add snd_soc_register_component() ASoC: add .name for snd_soc_component_driver ASoC: fsi: use snd_soc_register_component() instead of snd_soc_register_dais() ASoC: pxa2xx-ac97: move EXPORT_SYMBOL_GPL() next to definition ASoC: snd_soc_register_component() uses properly snd_soc_register_dai[s]() ASoC: switch over to use snd_soc_register_component() on davinci i2s ASoC: switch over to use snd_soc_register_component() on davinci vcif ASoC: switch over to use snd_soc_register_component() on davinci mcasp ASoC: switch over to use snd_soc_register_component() on dw i2s ASoC: switch over to use snd_soc_register_component() on mpc5200 ac97 ASoC: switch over to use snd_soc_register_component() on atmel ssc ASoC: switch over to use snd_soc_register_component() on au1x i2sc ASoC: switch over to use snd_soc_register_component() on au1x psc-ac97 ASoC: switch over to use snd_soc_register_component() on au1x psc-i2s ASoC: switch over to use snd_soc_register_component() on au1x ac97c ASoC: switch over to use snd_soc_register_component() on bf6xx i2s ASoC: switch over to use snd_soc_register_component() on bf5xx ac97 ASoC: switch over to use snd_soc_register_component() on bf5xx i2s ASoC: switch over to use snd_soc_register_component() on bf5xx tdm ASoC: switch over to use snd_soc_register_component() on ep93xx ac97 ASoC: switch over to use snd_soc_register_component() on ep93xx i2s ASoC: switch over to use snd_soc_register_component() on mpc5200 i2s ASoC: switch over to use snd_soc_register_component() on fsl ssi ASoC: switch over to use snd_soc_register_component() on imx ssi ASoC: switch over to use snd_soc_register_component() on jz4740 i2s ASoC: switch over to use snd_soc_register_component() on kirkwood i2s ASoC: switch over to use snd_soc_register_component() on sst ASoC: switch over to use snd_soc_register_component() on mxs saif ASoC: switch over to use snd_soc_register_component() on nuc900 ac97 ASoC: switch over to use snd_soc_register_component() on omap mcbsp ASoC: switch over to use snd_soc_register_component() on omap mcpdm ASoC: switch over to use snd_soc_register_component() on omap dmic ASoC: switch over to use snd_soc_register_component() on omap hdmi ASoC: switch over to use snd_soc_register_component() on pxa ssp ASoC: switch over to use snd_soc_register_component() on pxa2xx ac97 ASoC: switch over to use snd_soc_register_component() on pxa2xx i2s ASoC: switch over to use snd_soc_register_component() on mmp sspa ASoC: switch over to use snd_soc_register_component() on s6000 i2s ASoC: switch over to use snd_soc_register_component() on s3c24xx i2s ASoC: switch over to use snd_soc_register_component() on s3c i2s ASoC: switch over to use snd_soc_register_component() on s3c ac97 ASoC: switch over to use snd_soc_register_component() on samsung spdif ASoC: switch over to use snd_soc_register_component() on samsung i2s ASoC: switch over to use snd_soc_register_component() on samsung pcm ASoC: switch over to use snd_soc_register_component() on goni_wm8994 ASoC: switch over to use snd_soc_register_component() on sh4 ssi ASoC: switch over to use snd_soc_register_component() on sh4 hac ASoC: switch over to use snd_soc_register_component() on spear spdif out ASoC: switch over to use snd_soc_register_component() on spear spdif in ASoC: switch over to use snd_soc_register_component() on tegra30 i2s ASoC: switch over to use snd_soc_register_component() on tegra20 spdif ASoC: switch over to use snd_soc_register_component() on tegra20 ac97 ASoC: switch over to use snd_soc_register_component() on tegra20 i2s ASoC: switch over to use snd_soc_register_component() on txx9aclc ac97 ASoC: switch over to use snd_soc_register_component() on ux500 msp ASoC: snd_soc_[un]register_dai[s]() become non global function ASoC: switch over to use snd_soc_register_component() on sh4 siu Lars-Peter Clausen (28): ASoC: core: Drop unused "dapm" field form soc_enum struct ASoC: core: Remove unused "n_widgets" field from snd_soc_dapm struct ASoC: spear_pcm: Staticize non-exported structs ASoC: omap: Call omap_mcbsp_set_threshold() from mcbsp hw_params ASoC: ux500_pcm: Remove duplicated SNDRV_PCM_HW_PARAM_PERIODS constraint ASoC: spear_pcm: No need to wrap snd_dmaengine_pcm_close() ASoC: omap-pcm: No need to wrap snd_dmaengine_pcm_close() ASoC: tegra_pcm: No need to wrap snd_dmaengine_pcm_close() ASoC: ux500_pcm: No need to wrap snd_dmaengine_pcm_close() ASoC: atmel-pcm-dma: No need to wrap snd_dmaengine_pcm_close() ASoC: ux500_pcm: No need to use snd_dmaengine_pcm_set_data() ASoC: speaer_pcm: No need to use snd_dmaengine_pcm_set_data() ASoC: atmel-pcm-dma: Do not use snd_dmaengine_pcm_{set,get}_data() ASoC: ep93xx: Use ep93xx_dma_params instead of ep93xx_pcm_dma_params ASoC: mmp-pcm: Allocate dma filter parameters on the stack ASoC: imx-pcm: Embed the imx_dma_data struct in the dma_params struct ASoC: mxs: Embed the mxs_dma_data struct in the mxs_pcm_dma_params struct ASoC: dmaengine-pcm: Remove snd_dmaengine_pcm_{set,get}_data ASoC: Constify the 'driver' field of snd_soc_platform ASoC: Constify the 'ops' field of snd_soc_platform_driver ASoC: Constify the 'compr_ops' field of snd_soc_platform_driver ASoC: dmaengine_pcm: Setup device_fc in snd_hwparams_to_dma_slave_config ASoC: dmaengine-pcm: Add a common DAI DMA data struct ASoC: omap: Use common DAI DMA data ASoC: tegra: Use common DAI DMA data struct ASoC: fsl: Use common DAI DMA data struct ASoC: ep93xx_pcm: Fix compile error ASoC: ux500_pcm: Use the same snd_pcm_hardware for playback and capture Mark Brown (72): ASoC: arizona: Only allow input volume updates when inputs are enabled ASoC: wm8960: Add input boost volume control ASoC: omap: Check regulator enable for DAC on Pandora ASoC: arizona: If we only have a clock to synchronise with make it REFCLK mfd: arizona: Define additional FLL control registers mfd: wm5102: Map in additional FLL control registers ASoC: arizona: Increase FLL synchroniser bandwidth for high frequencies ASoC: arizona: Optimise FLL loop gains ASoC: arizona: Ensure synchroniser is disabled when not needed ASoC: arizona: Suppress reference calculations when setting REFCLK to 0 Merge tag 'v3.9-rc2' into asoc-core ASoC: arizona: Provide defines for the clock rates ASoC: wm_adsp: Handle old .bin files ASoC: arizona: Ensure we clock two channels for I2S mode ASoC: wm8994: Support constraining the maximum number of channels clocked mfd: wm5102: Add additional speaker control registers ASoC: arizona: Factor out speaker widgets from CODEC drivers ASoC: arizona: Log thermal events ASoC: arizona: Suppress speaker enable if thermal shutdown is flagged ASoC: arizona: Basic support for ISRC rate selection ASoC: wm8994: Restore AIFnCLK after reducing it for low clock rates extcon: arizona: Factor out magic application ASoC: arizona: Fix interaction between headphone outputs and identification extcon: arizona: Fix interaction between headphone outputs and identification Merge tag 'arizona-extcon-asoc' of git://git.kernel.org/.../broonie/misc into asoc-arizona ASoC: arizona: Add delay after powering up line level outputs Merge branch 'topic/pxa' of git://git.kernel.org/.../broonie/sound into asoc-component ASoC: wm5102: Implement OSR support ASoC: wm2000: Expose some more registers for diagnostics ASoC: wm_adsp: Provide defines for firmwares ASoC: wm_adsp: Split ADSP1 and ADSP2 firmware controls Merge branch 'topic/omap' of git://git.kernel.org/.../broonie/sound into HEAD Merge branch 'fix/samsung' of git://git.kernel.org/.../broonie/sound into asoc-samsung ASoC: core: Merge Samsung fixes to avoid trivial context conflict ASoC: wm8994: Remove duplicate revision cache ASoC: wm_adsp: Add support for firmware wide coefficient blocks Merge remote-tracking branch 'asoc/fix/compress' into asoc-next Merge remote-tracking branch 'asoc/fix/core' into asoc-next Merge remote-tracking branch 'asoc/fix/samsung' into asoc-next Merge remote-tracking branch 'asoc/fix/tegra' into asoc-next Merge remote-tracking branch 'asoc/fix/wm8903' into asoc-next Merge remote-tracking branch 'asoc/topic/adau1373' into asoc-next Merge remote-tracking branch 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4104' into asoc-next Merge remote-tracking branch 'asoc/topic/ak5386' into asoc-next Merge remote-tracking branch 'asoc/topic/arizona' into asoc-next Merge remote-tracking branch 'asoc/topic/atmel' into asoc-next Merge remote-tracking branch 'asoc/topic/component' into asoc-next Merge remote-tracking branch 'asoc/topic/compress' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-next Merge remote-tracking branch 'asoc/topic/cs42l73' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/fsi' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/maintainers' into asoc-next Merge remote-tracking branch 'asoc/topic/max98088' into asoc-next Merge remote-tracking branch 'asoc/topic/max98090' into asoc-next Merge remote-tracking branch 'asoc/topic/samsung' into asoc-next Merge remote-tracking branch 'asoc/topic/si476x' into asoc-next Merge remote-tracking branch 'asoc/topic/spear' into asoc-next Merge remote-tracking branch 'asoc/topic/tas5086' into asoc-next Merge remote-tracking branch 'asoc/topic/tegra' into asoc-next Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next Merge remote-tracking branch 'asoc/topic/wm-hubs' into asoc-next Merge remote-tracking branch 'asoc/topic/wm0010' into asoc-next Merge remote-tracking branch 'asoc/topic/wm2000' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8903' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8960' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next Michal Bachraty (1): ASoC: davinci-mcasp: Add support for multichannel playback Paul Bolle (3): ASoC: samsung: remove last traces of neo1973-gta01 ASoC: codecs: remove hidden prompt ASoC: Ux500: remove test for undefined Kconfig macro Paul Handrigan (1): ASoC: cs42l73: If Internal MCLK is >= 6.4MHz, then set SCLK to 64*Fs. Peter Ujfalusi (4): ASoC: omap3pandora: Fix compilation error ASoC: omap-mcpdm: Collect link direction configuration under a struct ASoC: omap-mcpdm: Fix for full duplex audio use case ASoC: omap-mcpdm: Remove leftower define for IO address Ryo Tsutsui (2): ASoC: dapm: Remove redundant clear_walk() for supply widgets ASoC: dapm: Only clear paths we've walked Sachin Kamat (5): ASoC: max98090: Remove unneeded version.h header include ASoC: max98090: Convert to devm_regmap_init_i2c() ASoC: max98090: Make struct dev_pm_ops const ASoC: max98090: Fix checkpatch errors related to spacing ASoC: max98090: Remove unnecessary braces Sascha Hauer (1): ASoC: imx-ssi: Fix AC97 rates Sebastien Guiriec (2): ASoC: omap-mcpdm: Clean up with devm_* function ASoC: omap-dmic: Clean up with devm_request_and_ioremap Shawn Guo (1): ASoC: imx-sgtl5000: use of_node to match cpu dai Silviu-Mihai Popescu (1): ASoC: omap: convert to devm_ioremap_resource() Stephen Warren (10): ASoC: tegra: assume CONFIG_OF in tegra_asoc_utils_init ASoC: tegra_wm8753: minor cleanup ASoC: tegra_wm9712: assume CONFIG_OF ASoC: tegra_alc5632: assume CONFIG_OF, and other cleanup ASoC: tegra trimslice: assume CONFIG_OF, and other cleanup ASoC: tegra_wm8903: assume CONFIG_OF, remove platform data ASoC: tegra: add Tegra114 support to the AHUB driver ASoC: tegra: add Tegra114 support to tegra_asoc_utils.c ASoC: export snd_soc_register_component ASoC: dapm: Implement mixer control sharing Tim Gardner (2): ASoC: adau1373: adau1373_hw_params: Silence overflow warning ASoC: wm_hubs: Silence reg_r and reg_l 'may be used uninitialized' warnings Vinod Koul (1): ASoC: mid-x86 - add support for meaadata apis Wei Yongjun (2): ASoC: tas5086: use module_i2c_driver to simplify the code ASoC: arizona: remove duplicated include from arizona.c Yegor Yefremov (2): ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode ASoC: davinci-mcasp: don't overwrite DIT settings Zoltan Puskas (1): ASoC: atmel: Add slave mode support to SSC in DSP Mode A Documentation/devicetree/bindings/sound/ak5386.txt | 19 + .../bindings/sound/nvidia,tegra30-ahub.txt | 26 +- .../devicetree/bindings/sound/ti,tas5086.txt | 32 ++ arch/arm/mach-s3c24xx/dma-s3c2410.c | 2 - arch/arm/mach-s3c24xx/dma-s3c2412.c | 2 - arch/arm/mach-s3c24xx/dma-s3c2440.c | 2 - arch/arm/mach-s3c24xx/dma-s3c2443.c | 2 - arch/arm/plat-samsung/devs.c | 6 + drivers/extcon/extcon-arizona.c | 107 ++-- drivers/mfd/wm5102-tables.c | 10 + include/linux/mfd/arizona/core.h | 3 + include/linux/mfd/arizona/registers.h | 42 ++ include/linux/mfd/wm8994/pdata.h | 8 + include/sound/dmaengine_pcm.h | 27 +- include/sound/soc-dai.h | 8 - include/sound/soc-dapm.h | 1 - include/sound/soc.h | 29 +- include/sound/tas5086.h | 7 + include/sound/tegra_wm8903.h | 26 - sound/soc/atmel/atmel-pcm-dma.c | 25 +- sound/soc/atmel/atmel_ssc_dai.c | 54 +- sound/soc/au1x/ac97c.c | 9 +- sound/soc/au1x/i2sc.c | 9 +- sound/soc/au1x/psc-ac97.c | 9 +- sound/soc/au1x/psc-i2s.c | 9 +- sound/soc/blackfin/bf5xx-ac97.c | 9 +- sound/soc/blackfin/bf5xx-i2s.c | 9 +- sound/soc/blackfin/bf5xx-tdm.c | 9 +- sound/soc/blackfin/bf6xx-i2s.c | 9 +- sound/soc/cirrus/edb93xx.c | 1 - sound/soc/cirrus/ep93xx-ac97.c | 18 +- sound/soc/cirrus/ep93xx-i2s.c | 25 +- sound/soc/cirrus/ep93xx-pcm.c | 37 +- sound/soc/cirrus/ep93xx-pcm.h | 20 - sound/soc/cirrus/simone.c | 2 - sound/soc/cirrus/snappercl15.c | 1 - sound/soc/codecs/Kconfig | 10 +- sound/soc/codecs/Makefile | 4 + sound/soc/codecs/adau1373.c | 5 +- sound/soc/codecs/ak4104.c | 55 +- sound/soc/codecs/ak5386.c | 152 +++++ sound/soc/codecs/arizona.c | 530 +++++++++++++++--- sound/soc/codecs/arizona.h | 34 +- sound/soc/codecs/cs4271.c | 166 +++--- sound/soc/codecs/cs42l73.c | 6 +- sound/soc/codecs/max98088.c | 30 +- sound/soc/codecs/max98090.c | 45 +- sound/soc/codecs/si476x.c | 47 +- sound/soc/codecs/tas5086.c | 591 ++++++++++++++++++++ sound/soc/codecs/wm0010.c | 6 +- sound/soc/codecs/wm2000.c | 4 +- sound/soc/codecs/wm2000.h | 2 + sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5102.c | 131 ++--- sound/soc/codecs/wm5102.h | 6 +- sound/soc/codecs/wm5110.c | 54 +- sound/soc/codecs/wm5110.h | 6 +- sound/soc/codecs/wm8903.c | 6 + sound/soc/codecs/wm8960.c | 10 + sound/soc/codecs/wm8994.c | 68 ++- sound/soc/codecs/wm8994.h | 3 +- sound/soc/codecs/wm_adsp.c | 124 +++- sound/soc/codecs/wm_adsp.h | 5 +- sound/soc/codecs/wm_hubs.c | 9 +- sound/soc/davinci/davinci-i2s.c | 15 +- sound/soc/davinci/davinci-mcasp.c | 97 +++- sound/soc/davinci/davinci-pcm.c | 16 +- sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/davinci/davinci-vcif.c | 11 +- sound/soc/dwc/designware_i2s.c | 9 +- sound/soc/fsl/fsl_ssi.c | 45 +- sound/soc/fsl/imx-audmux.c | 3 +- sound/soc/fsl/imx-pcm-dma.c | 58 +- sound/soc/fsl/imx-pcm-fiq.c | 6 +- sound/soc/fsl/imx-pcm.h | 19 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/imx-ssi.c | 42 +- sound/soc/fsl/imx-ssi.h | 7 +- sound/soc/fsl/mpc5200_psc_ac97.c | 8 +- sound/soc/fsl/mpc5200_psc_i2s.c | 9 +- sound/soc/jz4740/jz4740-i2s.c | 9 +- sound/soc/kirkwood/kirkwood-i2s.c | 11 +- sound/soc/mid-x86/sst_platform.c | 20 +- sound/soc/mid-x86/sst_platform.h | 2 + sound/soc/mxs/mxs-pcm.c | 43 +- sound/soc/mxs/mxs-pcm.h | 4 +- sound/soc/mxs/mxs-saif.c | 17 +- sound/soc/nuc900/nuc900-ac97.c | 9 +- sound/soc/omap/am3517evm.c | 1 - sound/soc/omap/ams-delta.c | 1 - sound/soc/omap/mcbsp.c | 14 +- sound/soc/omap/mcbsp.h | 7 +- sound/soc/omap/n810.c | 1 - sound/soc/omap/omap-abe-twl6040.c | 1 - sound/soc/omap/omap-dmic.c | 47 +- sound/soc/omap/omap-hdmi.c | 33 +- sound/soc/omap/omap-mcbsp.c | 27 +- sound/soc/omap/omap-mcpdm.c | 118 ++-- sound/soc/omap/omap-pcm.c | 83 +-- sound/soc/omap/omap-pcm.h | 40 -- sound/soc/omap/omap-twl4030.c | 1 - sound/soc/omap/omap3pandora.c | 9 +- sound/soc/omap/osk5912.c | 1 - sound/soc/omap/rx51.c | 1 - sound/soc/pxa/mmp-pcm.c | 33 +- sound/soc/pxa/mmp-sspa.c | 9 +- sound/soc/pxa/pxa-ssp.c | 9 +- sound/soc/pxa/pxa2xx-ac97.c | 11 +- sound/soc/pxa/pxa2xx-i2s.c | 9 +- sound/soc/s6000/s6000-i2s.c | 9 +- sound/soc/samsung/Kconfig | 5 +- sound/soc/samsung/ac97.c | 14 +- sound/soc/samsung/goni_wm8994.c | 11 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 33 +- sound/soc/samsung/idma.c | 11 +- sound/soc/samsung/neo1973_wm8753.c | 3 +- sound/soc/samsung/pcm.c | 11 +- .../include/plat => sound/soc/samsung}/regs-ac97.h | 0 .../include/plat => sound/soc/samsung}/regs-iis.h | 0 sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 9 +- sound/soc/samsung/s3c-i2s-v2.h | 7 +- sound/soc/samsung/s3c2412-i2s.c | 12 +- sound/soc/samsung/s3c24xx-i2s.c | 13 +- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- sound/soc/samsung/spdif.c | 11 +- sound/soc/sh/fsi.c | 14 +- sound/soc/sh/hac.c | 10 +- sound/soc/sh/migor.c | 2 +- sound/soc/sh/siu_dai.c | 11 +- sound/soc/sh/ssi.c | 10 +- sound/soc/soc-compress.c | 41 +- sound/soc/soc-core.c | 130 ++++- sound/soc/soc-dapm.c | 320 ++++++----- sound/soc/soc-dmaengine-pcm.c | 68 ++- sound/soc/soc-io.c | 5 +- sound/soc/spear/spdif_in.c | 9 +- sound/soc/spear/spdif_out.c | 9 +- sound/soc/spear/spear_pcm.c | 22 +- sound/soc/tegra/tegra20_ac97.c | 28 +- sound/soc/tegra/tegra20_ac97.h | 4 +- sound/soc/tegra/tegra20_i2s.c | 28 +- sound/soc/tegra/tegra20_i2s.h | 4 +- sound/soc/tegra/tegra20_spdif.c | 22 +- sound/soc/tegra/tegra20_spdif.h | 4 +- sound/soc/tegra/tegra30_ahub.c | 72 ++- sound/soc/tegra/tegra30_ahub.h | 24 +- sound/soc/tegra/tegra30_i2s.c | 28 +- sound/soc/tegra/tegra30_i2s.h | 4 +- sound/soc/tegra/tegra_alc5632.c | 15 +- sound/soc/tegra/tegra_asoc_utils.c | 34 +- sound/soc/tegra/tegra_asoc_utils.h | 1 + sound/soc/tegra/tegra_pcm.c | 48 +- sound/soc/tegra/tegra_pcm.h | 7 - sound/soc/tegra/tegra_wm8753.c | 15 +- sound/soc/tegra/tegra_wm8903.c | 179 +++--- sound/soc/tegra/tegra_wm9712.c | 7 +- sound/soc/tegra/trimslice.c | 56 +- sound/soc/txx9/txx9aclc-ac97.c | 9 +- sound/soc/ux500/ux500_msp_dai.c | 13 +- sound/soc/ux500/ux500_msp_dai.h | 5 - sound/soc/ux500/ux500_pcm.c | 76 +-- sound/soc/ux500/ux500_pcm.h | 14 - 164 files changed, 3437 insertions(+), 1710 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak5386.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,tas5086.txt create mode 100644 include/sound/tas5086.h delete mode 100644 include/sound/tegra_wm8903.h delete mode 100644 sound/soc/cirrus/ep93xx-pcm.h create mode 100644 sound/soc/codecs/ak5386.c create mode 100644 sound/soc/codecs/tas5086.c delete mode 100644 sound/soc/omap/omap-pcm.h rename {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-ac97.h (100%) rename {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-iis.h (100%)

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix headset mic support for Asus X101CH
by David Henningsson 15 Apr '13

15 Apr '13

With this patch, a TRRS headset mic cannot be successfully detected on the Asus X101CH, and we can also distinguish between headphone and headset automatically. Buglink: https://bugs.launchpad.net/bugs/1169138 Co-authored-by: Kailang <kailang(a)realtek.com> Tested-by: Luis Henriques <luis.henriques(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Now that we have better headset mic support infrastructure, I remembered I forgot to upstream this patch. It was originally provided by Realtek (therefore, I cannot answer for the msleeps and exactly why this is needed). I just rewrote it for the current kernel version. sound/pci/hda/patch_realtek.c | 56 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 56 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6cf27f5..1b7aed5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2680,6 +2680,34 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec, spec->gen.automute_hook = alc269_quanta_automute; } +static void alc269_x101_hp_automute_hook(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + struct alc_spec *spec = codec->spec; + int vref; + msleep(200); + snd_hda_gen_hp_automute(codec, jack); + + vref = spec->gen.hp_jack_present ? PIN_VREF80 : 0; + msleep(100); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + vref); + msleep(500); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + vref); +} + +static void alc269_fixup_x101_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + spec->gen.hp_automute_hook = alc269_x101_hp_automute_hook; + } +} + + /* update mute-LED according to the speaker mute state via mic VREF pin */ static void alc269_fixup_mic_mute_hook(void *private_data, int enabled) { @@ -2837,6 +2865,9 @@ enum { ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, + ALC269_FIXUP_ASUS_X101_FUNC, + ALC269_FIXUP_ASUS_X101_VERB, + ALC269_FIXUP_ASUS_X101, ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, @@ -2996,6 +3027,30 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_no_hp_to_lineout, }, + [ALC269_FIXUP_ASUS_X101_FUNC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_x101_headset_mic, + }, + [ALC269_FIXUP_ASUS_X101_VERB] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x20, AC_VERB_SET_COEF_INDEX, 0x08}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0310}, + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_ASUS_X101_FUNC + }, + [ALC269_FIXUP_ASUS_X101] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x04a1182c }, /* Headset mic */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_ASUS_X101_VERB + }, [ALC271_FIXUP_AMIC_MIC2] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -3044,6 +3099,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101), SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), -- 1.7.9.5

2 2

[alsa-devel] [PATCH v2 00/11] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 15 Apr '13

15 Apr '13

Hi, This series adds DT support for phycore-ac97 using the fsl-ssi driver. In version 2 I discarded most of the imx-ssi work and integrated the ac97-slave support in fsl-ssi, including support for imx-pcm-fiq as alternative to dma. There are some other changes to get phycore-ac97 working. The first two patches have notes about the detailed changes since version 1. The rest of the patches is new and necessary for the ac97-slave support. Regards, Markus Markus Pargmann (11): ASoC: phycore-ac97: Add DT support ASoC: imx-pcm-dma: Add support for DMA init by ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl-ssi: Add SACNT definitions ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Setup generic imx dma params ARM: imx: Export ac97 reset functions ASoC: fsl-ssi: imx ac97 support ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97 ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc Documentation/devicetree/bindings/powerpc/fsl/ssi.txt | 73 - Documentation/devicetree/bindings/sound/fsl,ssi.txt | 13 b/Documentation/devicetree/bindings/sound/fsl,ssi.txt | 73 + b/Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt | 12 b/arch/arm/mach-imx/mach-pca100.c | 7 b/arch/arm/mach-imx/mach-pcm043.c | 7 b/sound/soc/fsl/Kconfig | 2 b/sound/soc/fsl/fsl_ssi.c | 70 + b/sound/soc/fsl/fsl_ssi.h | 8 b/sound/soc/fsl/imx-pcm-dma.c | 31 b/sound/soc/fsl/imx-pcm-fiq.c | 14 b/sound/soc/fsl/imx-pcm.h | 6 b/sound/soc/fsl/imx-ssi.c | 7 b/sound/soc/fsl/imx-ssi.h | 1 b/sound/soc/fsl/phycore-ac97.c | 185 ++++ sound/soc/fsl/fsl_ssi.c | 379 +++++++--- sound/soc/fsl/imx-pcm.h | 9 17 files changed, 664 insertions(+), 233 deletions(-)

4 34

[alsa-devel] Stupid question #1 of many -- unit test harness?
by TJ Olaes 15 Apr '13

15 Apr '13

Hi. So, I'm attempting to write the driver for the eClaro card for a couple of reasons, one of which is that I currently don't have sound on mine. So... is there some kind of test harness framework for new drivers? I have no prior idea how to move forward to build this thing, so I'm snooping around in the oxygen source files and trying to copy things that look like they make sense. For instance, I noted that there's mention of an "AK4396" related to output. I assume this is some kind of chip, and I saw in the Claro code that something called "ak4396_init(...)" is called, so I just copied that line and pasted it into the eclaro_init(...) function I'm writing. I have no idea how I'm going to test this short of just compiling the kernel and booting into it. I feel like that's the brute-force, non-smart way to do it, and that you guys who've been doing this for a while might have ideas and recommendations on how to better test my code so I'm not rebooting my machine each time I change a line and recompile. I'd appreciate your guidance, community. Thanks. -TJ

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: disable autopm for MIDI devices
by Clemens Ladisch 15 Apr '13

15 Apr '13

Commit 88a8516a2128 (ALSA: usbaudio: implement USB autosuspend) introduced autopm for all USB audio/MIDI devices. However, many MIDI devices, such as synthesizers, do not merely transmit MIDI messages but use their MIDI inputs to control other functions. With autopm, these devices would get powered down as soon as the last MIDI port device is closed on the host. Even some plain MIDI interfaces could get broken: they automatically send Active Sensing messages while powered up, but as soon as these messages cease, the receiving device would interpret this as an accidental disconnection. Commit f5f165418cab (ALSA: usb-audio: Fix missing autopm for MIDI input) introduced another regression: some devices (e.g. the Roland GAIA SH-01) are self-powered but do a reset whenever the USB interface's power state changes. To work around all this, just disable autopm for all USB MIDI devices. Reported-by: Laurens Holst Cc: <stable(a)vger.kernel.org> Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/midi.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 34b9bb7..e5fee18 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -126,7 +126,6 @@ struct snd_usb_midi { struct snd_usb_midi_in_endpoint *in; } endpoints[MIDI_MAX_ENDPOINTS]; unsigned long input_triggered; - bool autopm_reference; unsigned int opened[2]; unsigned char disconnected; unsigned char input_running; @@ -1040,7 +1039,6 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir, { struct snd_usb_midi* umidi = substream->rmidi->private_data; struct snd_kcontrol *ctl; - int err; down_read(&umidi->disc_rwsem); if (umidi->disconnected) { @@ -1051,13 +1049,6 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir, mutex_lock(&umidi->mutex); if (open) { if (!umidi->opened[0] && !umidi->opened[1]) { - err = usb_autopm_get_interface(umidi->iface); - umidi->autopm_reference = err >= 0; - if (err < 0 && err != -EACCES) { - mutex_unlock(&umidi->mutex); - up_read(&umidi->disc_rwsem); - return -EIO; - } if (umidi->roland_load_ctl) { ctl = umidi->roland_load_ctl; ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; @@ -1080,8 +1071,6 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir, snd_ctl_notify(umidi->card, SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); } - if (umidi->autopm_reference) - usb_autopm_put_interface(umidi->iface); } } mutex_unlock(&umidi->mutex); @@ -2256,6 +2245,8 @@ int snd_usbmidi_create(struct snd_card *card, return err; } + usb_autopm_get_interface_no_resume(umidi->iface); + list_add_tail(&umidi->list, midi_list); return 0; }

2 1

[alsa-devel] [PATCH] ALSA: hda - Implement headset jack functionality for some Dell hw
by David Henningsson 15 Apr '13

15 Apr '13

On some machines, there is a headset jack that can support both headphone, headsets (of both CTIA and OMTP type) and mic-in. On other machines, the headset jack supports headphone, headsets (both CTIA and OMTP), but not mic-in. This patch implements that functionality as different capture sources. Buglink: https://bugs.launchpad.net/bugs/1169143 Tested-by: David Chen <david.chen(a)canonical.com> Co-authored-by: Kailang <kailang(a)realtek.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 386 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 386 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6cf27f5..f3307eb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -52,6 +52,20 @@ enum { ALC_INIT_GPIO3, }; +enum { + ALC_HEADSET_MODE_UNKNOWN, + ALC_HEADSET_MODE_UNPLUGGED, + ALC_HEADSET_MODE_HEADSET, + ALC_HEADSET_MODE_MIC, + ALC_HEADSET_MODE_HEADPHONE, +}; + +enum { + ALC_HEADSET_TYPE_UNKNOWN, + ALC_HEADSET_TYPE_CTIA, + ALC_HEADSET_TYPE_OMTP, +}; + struct alc_customize_define { unsigned int sku_cfg; unsigned char port_connectivity; @@ -87,6 +101,11 @@ struct alc_spec { unsigned int gpio_led; /* used for alc269_fixup_hp_gpio_led() */ + hda_nid_t headset_mic_pin; + hda_nid_t headphone_mic_pin; + int current_headset_mode; + int current_headset_type; + /* hooks */ void (*init_hook)(struct hda_codec *codec); #ifdef CONFIG_PM @@ -2798,6 +2817,302 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, } } +static void alc_headset_mode_unplugged(struct hda_codec *codec) +{ + int val; + + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x1b, 0x0c0b); + alc_write_coef_idx(codec, 0x45, 0xc429); + val = alc_read_coef_idx(codec, 0x35); + alc_write_coef_idx(codec, 0x35, val & 0xbfff); + alc_write_coef_idx(codec, 0x06, 0x2104); + alc_write_coef_idx(codec, 0x1a, 0x0001); + alc_write_coef_idx(codec, 0x26, 0x0004); + alc_write_coef_idx(codec, 0x32, 0x42a3); + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x76, 0x000e); + alc_write_coef_idx(codec, 0x6c, 0x2400); + alc_write_coef_idx(codec, 0x18, 0x7308); + alc_write_coef_idx(codec, 0x6b, 0xc429); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x15, 0x0d40); + alc_write_coef_idx(codec, 0xb7, 0x802b); + break; + } + snd_printdd("Headset jack set to unplugged mode.\n"); +} + + +static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, + hda_nid_t mic_pin) +{ + int val; + + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x45, 0xc429); + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + val = alc_read_coef_idx(codec, 0x35); + alc_write_coef_idx(codec, 0x35, val | 1<<14); + alc_write_coef_idx(codec, 0x06, 0x2100); + alc_write_coef_idx(codec, 0x1a, 0x0021); + alc_write_coef_idx(codec, 0x26, 0x008c); + snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); + break; + case 0x10ec0292: + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + alc_write_coef_idx(codec, 0x19, 0xa208); + alc_write_coef_idx(codec, 0x2e, 0xacf0); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + alc_write_coef_idx(codec, 0xb7, 0x802b); + alc_write_coef_idx(codec, 0xb5, 0x1040); + val = alc_read_coef_idx(codec, 0xc3); + alc_write_coef_idx(codec, 0xc3, val | 1<<12); + snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); + break; + } + snd_printdd("Headset jack set to mic-in mode.\n"); +} + +static void alc_headset_mode_default(struct hda_codec *codec) +{ + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x06, 0x2100); + alc_write_coef_idx(codec, 0x32, 0x4ea3); + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x76, 0x000e); + alc_write_coef_idx(codec, 0x6c, 0x2400); + alc_write_coef_idx(codec, 0x6b, 0xc429); + alc_write_coef_idx(codec, 0x18, 0x7308); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0041); + alc_write_coef_idx(codec, 0x15, 0x0d40); + alc_write_coef_idx(codec, 0xb7, 0x802b); + break; + } + snd_printdd("Headset jack set to headphone (default) mode.\n"); +} + +/* Iphone type */ +static void alc_headset_mode_ctia(struct hda_codec *codec) +{ + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x45, 0xd429); + alc_write_coef_idx(codec, 0x1b, 0x0c2b); + alc_write_coef_idx(codec, 0x32, 0x4ea3); + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x6b, 0xd429); + alc_write_coef_idx(codec, 0x76, 0x0008); + alc_write_coef_idx(codec, 0x18, 0x7388); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x15, 0x0d60); + alc_write_coef_idx(codec, 0xc3, 0x0000); + break; + } + snd_printdd("Headset jack set to iPhone-style headset mode.\n"); +} + +/* Nokia type */ +static void alc_headset_mode_omtp(struct hda_codec *codec) +{ + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x45, 0xe429); + alc_write_coef_idx(codec, 0x1b, 0x0c2b); + alc_write_coef_idx(codec, 0x32, 0x4ea3); + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x6b, 0xe429); + alc_write_coef_idx(codec, 0x76, 0x0008); + alc_write_coef_idx(codec, 0x18, 0x7388); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x15, 0x0d50); + alc_write_coef_idx(codec, 0xc3, 0x0000); + break; + } + snd_printdd("Headset jack set to Nokia-style headset mode.\n"); +} + +static void alc_determine_headset_type(struct hda_codec *codec) +{ + int val; + bool is_ctia = false; + struct alc_spec *spec = codec->spec; + + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x45, 0xd029); + msleep(300); + val = alc_read_coef_idx(codec, 0x46); + is_ctia = (val & 0x0070) == 0x0070; + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x6b, 0xd429); + msleep(300); + val = alc_read_coef_idx(codec, 0x6c); + is_ctia = (val & 0x001c) == 0x001c; + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); + alc_write_coef_idx(codec, 0xb7, 0x802b); + alc_write_coef_idx(codec, 0x15, 0x0d60); + alc_write_coef_idx(codec, 0xc3, 0x0c00); + msleep(300); + val = alc_read_coef_idx(codec, 0xbe); + is_ctia = (val & 0x1c02) == 0x1c02; + break; + } + + snd_printdd("Headset jack detected iPhone-style headset: %s\n", + is_ctia ? "yes" : "no"); + spec->current_headset_type = is_ctia ? ALC_HEADSET_TYPE_CTIA : ALC_HEADSET_TYPE_OMTP; +} + +static void alc_update_headset_mode(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]]; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + + int new_headset_mode; + + if (!snd_hda_jack_detect(codec, hp_pin)) + new_headset_mode = ALC_HEADSET_MODE_UNPLUGGED; + else if (mux_pin == spec->headset_mic_pin) + new_headset_mode = ALC_HEADSET_MODE_HEADSET; + else if (mux_pin == spec->headphone_mic_pin) + new_headset_mode = ALC_HEADSET_MODE_MIC; + else + new_headset_mode = ALC_HEADSET_MODE_HEADPHONE; + + if (new_headset_mode == spec->current_headset_mode) + return; + + switch (new_headset_mode) { + case ALC_HEADSET_MODE_UNPLUGGED: + alc_headset_mode_unplugged(codec); + spec->gen.hp_jack_present = false; + break; + case ALC_HEADSET_MODE_HEADSET: + if (spec->current_headset_type == ALC_HEADSET_TYPE_UNKNOWN) + alc_determine_headset_type(codec); + if (spec->current_headset_type == ALC_HEADSET_TYPE_CTIA) + alc_headset_mode_ctia(codec); + else if (spec->current_headset_type == ALC_HEADSET_TYPE_OMTP) + alc_headset_mode_omtp(codec); + spec->gen.hp_jack_present = true; + break; + case ALC_HEADSET_MODE_MIC: + alc_headset_mode_mic_in(codec, hp_pin, spec->headphone_mic_pin); + spec->gen.hp_jack_present = false; + break; + case ALC_HEADSET_MODE_HEADPHONE: + alc_headset_mode_default(codec); + spec->gen.hp_jack_present = true; + break; + } + if (new_headset_mode != ALC_HEADSET_MODE_MIC) { + snd_hda_set_pin_ctl_cache(codec, hp_pin, + AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); + if (spec->headphone_mic_pin) + snd_hda_set_pin_ctl_cache(codec, spec->headphone_mic_pin, + PIN_VREFHIZ); + } + spec->current_headset_mode = new_headset_mode; + + snd_hda_gen_update_outputs(codec); +} + +static void alc_update_headset_mode_hook(struct hda_codec *codec, + struct snd_ctl_elem_value *ucontrol) +{ + alc_update_headset_mode(codec); +} + +static void alc_update_headset_jack_cb(struct hda_codec *codec, struct hda_jack_tbl *jack) +{ + struct alc_spec *spec = codec->spec; + spec->current_headset_type = ALC_HEADSET_MODE_UNKNOWN; + snd_hda_gen_hp_automute(codec, jack); +} + +static void alc_probe_headset_mode(struct hda_codec *codec) +{ + int i; + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->gen.autocfg; + + /* Find mic pins */ + for (i = 0; i < cfg->num_inputs; i++) { + if (cfg->inputs[i].is_headset_mic && !spec->headset_mic_pin) + spec->headset_mic_pin = cfg->inputs[i].pin; + if (cfg->inputs[i].is_headphone_mic && !spec->headphone_mic_pin) + spec->headphone_mic_pin = cfg->inputs[i].pin; + } + + spec->gen.cap_sync_hook = alc_update_headset_mode_hook; + spec->gen.automute_hook = alc_update_headset_mode; + spec->gen.hp_automute_hook = alc_update_headset_jack_cb; +} + +static void alc_fixup_headset_mode(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC | HDA_PINCFG_HEADPHONE_MIC; + break; + case HDA_FIXUP_ACT_PROBE: + alc_probe_headset_mode(codec); + break; + case HDA_FIXUP_ACT_INIT: + spec->current_headset_mode = 0; + alc_update_headset_mode(codec); + break; + } +} + +static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct alc_spec *spec = codec->spec; + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + } + else + alc_fixup_headset_mode(codec, fix, action); +} + +static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + int val; + alc_write_coef_idx(codec, 0xc4, 0x8000); + val = alc_read_coef_idx(codec, 0xc2); + alc_write_coef_idx(codec, 0xc2, val & 0xfe); + snd_hda_set_pin_ctl_cache(codec, 0x18, 0); + } + alc_fixup_headset_mode(codec, fix, action); +} + static void alc271_hp_gate_mic_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -2837,6 +3152,10 @@ enum { ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, + ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, + ALC269_FIXUP_HEADSET_MODE, + ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, @@ -2996,6 +3315,35 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_no_hp_to_lineout, }, + [ALC269_FIXUP_DELL1_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x01a1913d }, /* use as headphone mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, + [ALC269_FIXUP_DELL2_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x16, 0x21014020 }, /* dock line out */ + { 0x19, 0x21a19030 }, /* dock mic */ + { 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, + [ALC269_FIXUP_HEADSET_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode, + }, + [ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_no_hp_mic, + }, [ALC271_FIXUP_AMIC_MIC2] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -3030,6 +3378,26 @@ static const struct hda_fixup alc269_fixups[] = { 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(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c7, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ec, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ed, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ee, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f3, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), @@ -3536,6 +3904,8 @@ enum { ALC662_FIXUP_NO_JACK_DETECT, ALC662_FIXUP_ZOTAC_Z68, ALC662_FIXUP_INV_DMIC, + ALC668_FIXUP_DELL_MIC_NO_PRESENCE, + ALC668_FIXUP_HEADSET_MODE, }; static const struct hda_fixup alc662_fixups[] = { @@ -3696,6 +4066,20 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC668_FIXUP_DELL_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a1913d }, /* use as headphone mic, without its own jack detect */ + { 0x1b, 0x03a1113c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_MODE + }, + [ALC668_FIXUP_HEADSET_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_alc668, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -3704,6 +4088,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), + SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), -- 1.7.9.5

2 1

[alsa-devel] PROBLEM: iMX6 SSI - set up as a master of external audio codec
by Andrey Gelman 15 Apr '13

15 Apr '13

Hi, everyone ! I'm bringing up a wm8731 audio codec on an iMX6 based board. I use the slave mode, i.e. SSI outputs all the clocks towards the codec IC. I work with 3.0.35 Android kernel provided by freescale. While, generally, the playback is Ok, I face 2 issues: - cannot play back mono streams (too high rate, one speaker produces only noise) - cannot play back streams with sampling rate as low as 8kHz (the sound is accompanied by high distortion) In the mainline kernel (commits to v3.6), there is a workaround to the 1-st issue by simply removing mono support from imx-ssi.c. Similar approach would fix the 2-nd issue as well - SSI would report it does not support 8kHz sampling rate. In a later commit, this workaround has been reverted, enabling mono support back. My question is this: Is this a real limitation of SSI ? Does it support mono playback or not ? Does SSI support 8kHz sampling rate ? If yes, what could be possible reason to the distortions I get, while higher rates are played back well ? Is there a real fix to these issues, or only workarounds are available ? Thanks, Andrey Gelman

1 0

[alsa-devel] [PATCH 20/30] ASoC: samsung: convert to dmaengine API
by Arnd Bergmann 15 Apr '13

15 Apr '13

In order to build the exynos kernel with CONFIG_ARCH_MULTIPLATFORM, we must convert all users of the Samsung private DMA interface to the generic dmaengine API. This version of the patch adds the generic dmaengine API as an alternative to the existing samsung specific one. Once all the older platforms provide support for the common dmaengine interfaces, we can remove the old code. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> --- sound/soc/samsung/dma.c | 219 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/samsung/dma.h | 15 +++- sound/soc/samsung/i2s.c | 2 - sound/soc/samsung/pcm.c | 1 - sound/soc/samsung/spdif.c | 1 - 5 files changed, 232 insertions(+), 6 deletions(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 21b7926..9fd53df 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -22,8 +22,14 @@ #include <sound/pcm_params.h> #include <asm/dma.h> + +#ifdef CONFIG_SAMSUNG_DMADEV #include <mach/hardware.h> #include <mach/dma.h> +#else +#include <linux/dmaengine.h> +#include <linux/amba/pl330.h> +#endif #include "dma.h" @@ -62,11 +68,13 @@ struct runtime_data { static void audio_buffdone(void *data); +#ifdef CONFIG_SAMSUNG_DMADEV /* dma_enqueue * * place a dma buffer onto the queue for the dma system * to handle. */ + static void dma_enqueue(struct snd_pcm_substream *substream) { struct runtime_data *prtd = substream->runtime->private_data; @@ -265,6 +273,217 @@ static int dma_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } +#else +/* dma_enqueue + * + * place a dma buffer onto the queue for the dma system + * to handle. + */ + +static void dma_enqueue(struct snd_pcm_substream *substream) +{ + struct runtime_data *prtd = substream->runtime->private_data; + dma_addr_t pos = prtd->dma_pos; + unsigned long period = prtd->dma_period; + unsigned int limit; + enum dma_transfer_direction direction; + struct dma_chan *chan = prtd->params->ch; + struct dma_async_tx_descriptor *desc; + + pr_debug("Entered %s\n", __func__); + + limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; + + pr_debug("%s: loaded %d, limit %d\n", + __func__, prtd->dma_loaded, limit); + + direction = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); + + while (prtd->dma_loaded < limit) { + pr_debug("dma_loaded: %d\n", prtd->dma_loaded); + + if ((pos + period) > prtd->dma_end) { + period = prtd->dma_end - pos; + pr_debug("%s: corrected dma len %ld\n", + __func__, period); + } + + desc = dmaengine_prep_dma_cyclic(chan, pos, + prtd->dma_period*limit, period, direction, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + + if (desc) { + desc->callback = audio_buffdone; + desc->callback_param = substream; + dmaengine_submit(desc); + } + + prtd->dma_loaded++; + pos += period; + if (pos >= prtd->dma_end) + pos = prtd->dma_start; + } + + prtd->dma_pos = pos; +} + +static void audio_buffdone(void *data) +{ + struct snd_pcm_substream *substream = data; + struct runtime_data *prtd = substream->runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + if (prtd->state & ST_RUNNING) { + prtd->dma_pos += prtd->dma_period; + if (prtd->dma_pos >= prtd->dma_end) + prtd->dma_pos = prtd->dma_start; + + if (substream) + snd_pcm_period_elapsed(substream); + } +} + +static int dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned long totbytes = params_buffer_bytes(params); + struct s3c_dma_params *dma = + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!dma) + return 0; + + /* this may get called several times by oss emulation + * with different params -HW */ + if (prtd->params == NULL) { + struct dma_slave_config config; + dma_cap_mask_t mask; + + /* prepare DMA */ + prtd->params = dma; + + pr_debug("params %p, channel %d\n", prtd->params, + prtd->params->channel); + + dma_cap_zero(mask); + dma_cap_set(DMA_CYCLIC, mask); + + prtd->params->ch = dma_request_slave_channel_compat(mask, + pl330_filter, (void *)prtd->params->channel, + rtd->cpu_dai->dev, prtd->params->ch_name); + + memset(&config, 0, sizeof(struct dma_slave_config)); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + config.direction = DMA_MEM_TO_DEV; + config.dst_addr = prtd->params->dma_addr; + config.dst_addr_width = prtd->params->dma_size; + config.dst_maxburst = 1; + dmaengine_slave_config(prtd->params->ch, &config); + } else { + config.direction = DMA_DEV_TO_MEM; + config.src_addr = prtd->params->dma_addr; + config.src_addr_width = prtd->params->dma_size; + config.src_maxburst = 1; + dmaengine_slave_config(prtd->params->ch, &config); + } + } + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + runtime->dma_bytes = totbytes; + + spin_lock_irq(&prtd->lock); + prtd->dma_loaded = 0; + prtd->dma_period = params_period_bytes(params); + prtd->dma_start = runtime->dma_addr; + prtd->dma_pos = prtd->dma_start; + prtd->dma_end = prtd->dma_start + totbytes; + spin_unlock_irq(&prtd->lock); + + return 0; +} + +static int dma_hw_free(struct snd_pcm_substream *substream) +{ + struct runtime_data *prtd = substream->runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + snd_pcm_set_runtime_buffer(substream, NULL); + + if (prtd->params) { + dmaengine_terminate_all(prtd->params->ch); + dma_release_channel(prtd->params->ch); + prtd->params = NULL; + } + + return 0; +} + +static int dma_prepare(struct snd_pcm_substream *substream) +{ + struct runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!prtd->params) + return 0; + + /* flush the DMA channel */ + dmaengine_terminate_all(prtd->params->ch); + + prtd->dma_loaded = 0; + prtd->dma_pos = prtd->dma_start; + + /* enqueue dma buffers */ + dma_enqueue(substream); + + return ret; +} + +static int dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->state |= ST_RUNNING; + dma_async_issue_pending(prtd->params->ch); + break; + + case SNDRV_PCM_TRIGGER_STOP: + prtd->state &= ~ST_RUNNING; + dmaengine_terminate_all(prtd->params->ch); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} +#endif static snd_pcm_uframes_t dma_pointer(struct snd_pcm_substream *substream) diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 189a7a6..6bd1857 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -12,13 +12,24 @@ #ifndef _S3C_AUDIO_H #define _S3C_AUDIO_H +#ifdef CONFIG_SAMSUNG_DMADEV +#include <mach/dma.h> +#endif + struct s3c_dma_params { +#ifdef CONFIG_SAMSUNG_DMADEV struct s3c2410_dma_client *client; /* stream identifier */ + unsigned ch; + struct samsung_dma_ops *ops; +#else + struct s3c2410_dma_client { + char *name; /* unused */ + } *client; + struct dma_chan *ch; +#endif int channel; /* Channel ID */ dma_addr_t dma_addr; int dma_size; /* Size of the DMA transfer */ - unsigned ch; - struct samsung_dma_ops *ops; char *ch_name; }; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index d7231e3..61f2622 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -22,8 +22,6 @@ #include <sound/soc.h> #include <sound/pcm_params.h> -#include <mach/dma.h> - #include <linux/platform_data/asoc-s3c.h> #include "dma.h" diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 13bab79..b5f267c 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -20,7 +20,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "pcm.h" diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 5008e5b..ee792aa 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -18,7 +18,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "spdif.h" -- 1.8.1.2

4 5

[alsa-devel] [PATCH v2] ALSA: hda - Apply codec delay to wallclock.
by Dylan Reid 15 Apr '13

15 Apr '13

For playback add the codec-side delay to the timestamp, for capture subtract it. This brings the timestamps in line with the time that was recently added to the delay reporting. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Changes since v1: Calculate nsec from frames, subtract delay for playback as well as capture. sound/pci/hda/hda_intel.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735567e..3e6f2c6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1889,6 +1889,23 @@ static void azx_timecounter_init(struct snd_pcm_substream *substream, tc->cycle_last = last; } +static u64 azx_subtract_codec_delay(struct snd_pcm_substream *substream, + u64 nsec) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + u64 codec_frames, codec_nsecs; + + if (!hinfo->ops.get_delay) + return nsec; + + codec_frames = hinfo->ops.get_delay(hinfo, apcm->codec, substream); + codec_nsecs = div_u64(codec_frames * 1000000000LL, + substream->runtime->rate); + + return (nsec > codec_nsecs) ? nsec - codec_nsecs : 0; +} + static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, struct timespec *ts) { @@ -1897,6 +1914,7 @@ static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, nsec = timecounter_read(&azx_dev->azx_tc); nsec = div_u64(nsec, 3); /* can be optimized */ + nsec = azx_subtract_codec_delay(substream, nsec); *ts = ns_to_timespec(nsec); -- 1.8.1.3.605.g02339dd

3 9

[alsa-devel] Can Realtek HD-A codec ALC282 support D3cold?
by Lin, Mengdong 15 Apr '13

15 Apr '13

Hi, We want to put an idle HD-A codec into D3cold for more power saving and preparation for being power gated. The Realtek ALC282 is used in our test. Could anyone help to confirm whether ALC282 can support D3cold or not? I'm told that ALC282 can support D3cold. But my test shows it cannot. At first, ALC282 HW does not report that it can support D3cold. Its parameter "Supported Power States" returns 0xc000000f, means it only supports: EPSS, CLKSTOP, D3hot, D2, D1 and D0. Is this a just mistake in reporting the bit, or D3cold is not supported all? Then when I explicitly set the audio function group to D3cold by verb 0x705, and then get its power state after 200ms by verb 0xf05, The codec response is 0x00000000: No PS-Error, both PS-Act and PS-set is D0. So it seems the command to D3cold is not processed by codec HW at all. Is there someone from Realtek or with experience on ALC282 here can share some information about this issue? Many thanks Mengdong

1 0

[alsa-devel] M-Audio Delta 1010LT: S/PDIF clock source not working
by Jonas Petersen 15 Apr '13

15 Apr '13

Hi, I am trying to capture from S/PDIF. I get a signal but the clocking is broken somehow. I have 'Multi Track Internal Clock' on 'ICE958 Input'. 'Word Clock Status' and 'Word Clock Sync' are 'Off'. What happens is the following: It captures what comes in on S/PDIF but it seems it will always use the internal clock with the setting from 'Multi Track Internal Clock Default'. So even though 'Multi Track Internal Clock' is on 'ICE958 Input', the setting of 'Multi Track Internal Clock Default' will clock the signal. A 1 kHz sine wave at 48 kHz samplerate captured with 'Multi Track Internal Clock Default' on 96 kHz will result in a 2 kHz sine. When the sample frequency rates are matched, it will capture at correct rate but with hiccups due to the rates not perfectly aligned. What can I do narrow down the cause? Reards Jonas

4 8

[alsa-devel] USB Quirks for Boss JS-8
by Keith A. Milner 15 Apr '13

15 Apr '13

Hi, I'm looking at developing a quirk for the Boss JS-8. The lsusb is as follows: Bus 001 Device 003: ID 0582:0109 Roland Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x0582 Roland Corp. idProduct 0x0109 bcdDevice 0.01 iManufacturer 1 BOSS iProduct 2 JS-8 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 167 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x05 EP 5 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 02 02 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 8 Mass Storage bDeviceSubClass 6 SCSI bDeviceProtocol 80 Bulk (Zip) bMaxPacketSize0 64 bNumConfigurations 0 Device Status: 0x0001 Self Powered I wasn't entirely sure about the quirk syntax as the two interfaces are at EP 4 and EP5. Does this mean I need to ignore the (non-existant) lower interfaces? Here's what I came up with: { /* Boss JS-8 Jam Station */ USB_DEVICE(0x0582, 0x0109), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "BOSS", */ /* .product_name = "JS-8", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 1, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 2, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 3, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 4, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 5, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = -1 } } } }, regards, -- Keith A. Milner

5 9

[alsa-devel] alsa-project.org access, I want to tackle building an eClaro driver
by TJ Olaes 13 Apr '13

13 Apr '13

Hello. I hate to post to this mailing list regarding access to the alsa-project.org wiki, but I've gone through the site and can't, for the life of me, seem to find an admin email or something like it on the wiki. I'm trying to sign up to the wiki and am having trouble with the Captcha. I put in different version numbers of linux kernels but all my attempts to register seem to be thwarted because I'm not putting in the correct info. I have no idea what I'm doing wrong. What I'm really trying to do is get involved in alsa driver development... I think this would be an appropriate place for me to attempt to address my issue, which is the lack of an HT Omega eClaro card driver. I really, really want to contribute but it seems I'm being thwarted by the developer website. If I could get contact info for an admin or something, I would go bug them, but it seems that info isn't posted on the wiki. Sorry to bug y'all. Thanks. -TJ

3 3

Re: [alsa-devel] [PATCH] alsa/usb: Add quirk for 192KHz recording on E-Mu devices
by Takashi Iwai 13 Apr '13

13 Apr '13

At Fri, 12 Apr 2013 22:33:59 -0500, Calvin Owens wrote: > > When recording at 176.2KHz or 192Khz, the device adds a 32-bit length > header to the capture packets, which obviously needs to be ignored for > recording to work properly. > > Userspace expected: L0 L1 L2 R0 R1 R2 > ...but actually got: R2 L0 L1 L2 R0 R1 > > Also, the last byte of the length header being interpreted as L0 of > the first sample caused spikes every 0.5ms, resulting in a loud 16KHz > tone (about the highest 'B' on a piano) being present throughout > captures. > > Tested at all sample rates on an E-Mu 0404USB, and tested for > regressions on a generic USB headset. > > Signed-off-by: Calvin Owens <jcalvinowens(a)gmail.com> Thanks, applied for 3.10 kernel (but with Cc to stable) now. Takashi > --- > sound/usb/card.h | 1 + > sound/usb/pcm.c | 2 +- > sound/usb/quirks.c | 1 + > sound/usb/stream.c | 1 + > 4 files changed, 4 insertions(+), 1 deletion(-) > > diff --git a/sound/usb/card.h b/sound/usb/card.h > index 8a751b4..d32ea41 100644 > --- a/sound/usb/card.h > +++ b/sound/usb/card.h > @@ -116,6 +116,7 @@ struct snd_usb_substream { > unsigned int altset_idx; /* USB data format: index of alternate setting */ > unsigned int txfr_quirk:1; /* allow sub-frame alignment */ > unsigned int fmt_type; /* USB audio format type (1-3) */ > + unsigned int pkt_offset_adj; /* Bytes to drop from beginning of packets (for non-compliant devices) */ > > unsigned int running: 1; /* running status */ > > diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c > index f94397b..a481fea 100644 > --- a/sound/usb/pcm.c > +++ b/sound/usb/pcm.c > @@ -1170,7 +1170,7 @@ static void retire_capture_urb(struct snd_usb_substream *subs, > stride = runtime->frame_bits >> 3; > > for (i = 0; i < urb->number_of_packets; i++) { > - cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset; > + cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset + subs->pkt_offset_adj; > if (urb->iso_frame_desc[i].status && printk_ratelimit()) { > snd_printdd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); > // continue; > diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c > index 5325a38..7e292b9 100644 > --- a/sound/usb/quirks.c > +++ b/sound/usb/quirks.c > @@ -837,6 +837,7 @@ static void set_format_emu_quirk(struct snd_usb_substream *subs, > break; > } > snd_emuusb_set_samplerate(subs->stream->chip, emu_samplerate_id); > + subs->pkt_offset_adj = (emu_samplerate_id >= EMU_QUIRK_SR_176400HZ) ? 4 : 0; > } > > void snd_usb_set_format_quirk(struct snd_usb_substream *subs, > diff --git a/sound/usb/stream.c b/sound/usb/stream.c > index ad181d5..0927cc6 100644 > --- a/sound/usb/stream.c > +++ b/sound/usb/stream.c > @@ -94,6 +94,7 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, > subs->dev = as->chip->dev; > subs->txfr_quirk = as->chip->txfr_quirk; > subs->speed = snd_usb_get_speed(subs->dev); > + subs->pkt_offset_adj = 0; > > snd_usb_set_pcm_ops(as->pcm, stream); > > -- > 1.8.1.5 >

1 0

Re: [alsa-devel] [PATCH] ASoC: samsung: fix neo1973-wm8753 compilation
by Mark Brown 12 Apr '13

12 Apr '13

On Thu, Apr 11, 2013 at 09:02:43PM +0200, Heiko Stübner wrote: > Commit b2ca78717cea (ARM: S3C24XX: make gta02.h local) already replaced > the GTA02_GPIO_* constants in neo1973-wm8753.c but forgot to remove the > inclusion of mach/gta02.h before moving the file out of mach/. Applied, thanks.

1 0

[alsa-devel] ASoC update for v3.9-rc6
by Mark Brown 12 Apr '13

12 Apr '13

The following changes since commit 31880c37c11e28cb81c70757e38392b42e695dc6: Linux 3.9-rc6 (2013-04-07 20:49:54 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-maintainers-v3.9-rc6 for you to fetch changes up to b02e48f2f8755e51fe8ce246988a29bd96f6b85e: MAINTAINERS: Update e-mail address (2013-04-12 11:53:41 +0100) May as well go along with the other sound updates. ---------------------------------------------------------------- MAINTAINERS: Update e-mail address Update the e-mail address I use for subsystems. ---------------------------------------------------------------- Mark Brown (1): MAINTAINERS: Update e-mail address MAINTAINERS | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-)

2 1

[alsa-devel] [PATCH 18/30] ASoC: samsung: move plat/ headers to local directory
by Arnd Bergmann 12 Apr '13

12 Apr '13

The plat/iis.h and plat/ac97.h files in the samsung platform are only needed by the ASoC drivers, so they can be moved into the same directory, as one more step towards a multiplatform build. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> --- arch/arm/mach-s3c24xx/dma-s3c2410.c | 2 -- arch/arm/mach-s3c24xx/dma-s3c2412.c | 2 -- arch/arm/mach-s3c24xx/dma-s3c2440.c | 2 -- arch/arm/mach-s3c24xx/dma-s3c2443.c | 2 -- sound/soc/samsung/ac97.c | 2 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-ac97.h | 0 {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-iis.h | 0 sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c24xx-i2s.c | 2 +- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- 12 files changed, 6 insertions(+), 14 deletions(-) rename {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-ac97.h (100%) rename {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-iis.h (100%) diff --git a/arch/arm/mach-s3c24xx/dma-s3c2410.c b/arch/arm/mach-s3c24xx/dma-s3c2410.c index a6c94b8..30aa53f 100644 --- a/arch/arm/mach-s3c24xx/dma-s3c2410.c +++ b/arch/arm/mach-s3c24xx/dma-s3c2410.c @@ -25,10 +25,8 @@ #include <plat/regs-serial.h> #include <mach/regs-gpio.h> -#include <plat/regs-ac97.h> #include <plat/regs-dma.h> #include <mach/regs-lcd.h> -#include <plat/regs-iis.h> #include <plat/regs-spi.h> static struct s3c24xx_dma_map __initdata s3c2410_dma_mappings[] = { diff --git a/arch/arm/mach-s3c24xx/dma-s3c2412.c b/arch/arm/mach-s3c24xx/dma-s3c2412.c index c0e8c3f..ab1700e 100644 --- a/arch/arm/mach-s3c24xx/dma-s3c2412.c +++ b/arch/arm/mach-s3c24xx/dma-s3c2412.c @@ -25,10 +25,8 @@ #include <plat/regs-serial.h> #include <mach/regs-gpio.h> -#include <plat/regs-ac97.h> #include <plat/regs-dma.h> #include <mach/regs-lcd.h> -#include <plat/regs-iis.h> #include <plat/regs-spi.h> #define MAP(x) { (x)| DMA_CH_VALID, (x)| DMA_CH_VALID, (x)| DMA_CH_VALID, (x)| DMA_CH_VALID } diff --git a/arch/arm/mach-s3c24xx/dma-s3c2440.c b/arch/arm/mach-s3c24xx/dma-s3c2440.c index 1c08eccd..cd25de2 100644 --- a/arch/arm/mach-s3c24xx/dma-s3c2440.c +++ b/arch/arm/mach-s3c24xx/dma-s3c2440.c @@ -25,10 +25,8 @@ #include <plat/regs-serial.h> #include <mach/regs-gpio.h> -#include <plat/regs-ac97.h> #include <plat/regs-dma.h> #include <mach/regs-lcd.h> -#include <plat/regs-iis.h> #include <plat/regs-spi.h> static struct s3c24xx_dma_map __initdata s3c2440_dma_mappings[] = { diff --git a/arch/arm/mach-s3c24xx/dma-s3c2443.c b/arch/arm/mach-s3c24xx/dma-s3c2443.c index 000e4c6..5fe3539 100644 --- a/arch/arm/mach-s3c24xx/dma-s3c2443.c +++ b/arch/arm/mach-s3c24xx/dma-s3c2443.c @@ -25,10 +25,8 @@ #include <plat/regs-serial.h> #include <mach/regs-gpio.h> -#include <plat/regs-ac97.h> #include <plat/regs-dma.h> #include <mach/regs-lcd.h> -#include <plat/regs-iis.h> #include <plat/regs-spi.h> #define MAP(x) { \ diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 0df3c56..c76abdf 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -20,7 +20,7 @@ #include <sound/soc.h> #include <mach/dma.h> -#include <plat/regs-ac97.h> +#include "regs-ac97.h" #include <linux/platform_data/asoc-s3c.h> #include "dma.h" diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 15a3817..fa91376 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -20,7 +20,7 @@ #include <sound/soc.h> #include <sound/jack.h> -#include <plat/regs-iis.h> +#include "regs-iis.h" #include <asm/mach-types.h> #include "s3c24xx-i2s.h" diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index a301d8c..ccc601d 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -21,7 +21,7 @@ #include <sound/soc.h> #include <asm/mach-types.h> -#include <plat/regs-iis.h> +#include "regs-iis.h" #include <mach/gta02.h> #include "../codecs/wm8753.h" diff --git a/arch/arm/plat-samsung/include/plat/regs-ac97.h b/sound/soc/samsung/regs-ac97.h similarity index 100% rename from arch/arm/plat-samsung/include/plat/regs-ac97.h rename to sound/soc/samsung/regs-ac97.h diff --git a/arch/arm/plat-samsung/include/plat/regs-iis.h b/sound/soc/samsung/regs-iis.h similarity index 100% rename from arch/arm/plat-samsung/include/plat/regs-iis.h rename to sound/soc/samsung/regs-iis.h diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index a5826ea..704460a 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -24,7 +24,7 @@ #include <sound/soc.h> #include <sound/jack.h> -#include <plat/regs-iis.h> +#include "regs-iis.h" #include <asm/mach-types.h> #include "s3c24xx-i2s.h" diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 13f6dd1..a7b17c1 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -24,7 +24,7 @@ #include <sound/pcm_params.h> #include <mach/dma.h> -#include <plat/regs-iis.h> +#include "regs-iis.h" #include "dma.h" #include "s3c24xx-i2s.h" diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 333e1b7..1b7b52b 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -18,7 +18,7 @@ #include <sound/soc.h> #include <sound/s3c24xx_uda134x.h> -#include <plat/regs-iis.h> +#include "regs-iis.h" #include "s3c24xx-i2s.h" -- 1.8.1.2

3 5

[alsa-devel] [PATCH] MAINTAINERS: Update e-mail address
by Mark Brown 12 Apr '13

12 Apr '13

Update the e-mail address I use for subsystems. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- MAINTAINERS | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 836a618..c6108c3 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6625,7 +6625,7 @@ S: Supported F: fs/reiserfs/ REGISTER MAP ABSTRACTION -M: Mark Brown <broonie(a)opensource.wolfsonmicro.com> +M: Mark Brown <broonie(a)kernel.org> T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regmap.git S: Supported F: drivers/base/regmap/ @@ -7374,7 +7374,7 @@ F: sound/ SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) M: Liam Girdwood <lgirdwood(a)gmail.com> -M: Mark Brown <broonie(a)opensource.wolfsonmicro.com> +M: Mark Brown <broonie(a)kernel.org> 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) W: http://alsa-project.org/main/index.php/ASoC @@ -7463,7 +7463,7 @@ F: drivers/clk/spear/ SPI SUBSYSTEM M: Grant Likely <grant.likely(a)secretlab.ca> -M: Mark Brown <broonie(a)opensource.wolfsonmicro.com> +M: Mark Brown <broonie(a)kernel.org> L: spi-devel-general(a)lists.sourceforge.net Q: http://patchwork.kernel.org/project/spi-devel-general/list/ T: git git://git.secretlab.ca/git/linux-2.6.git @@ -8708,7 +8708,7 @@ F: drivers/scsi/vmw_pvscsi.h VOLTAGE AND CURRENT REGULATOR FRAMEWORK M: Liam Girdwood <lrg(a)ti.com> -M: Mark Brown <broonie(a)opensource.wolfsonmicro.com> +M: Mark Brown <broonie(a)kernel.org> W: http://opensource.wolfsonmicro.com/node/15 W: http://www.slimlogic.co.uk/?p=48 T: git git://git.kernel.org/pub/scm/linux/kernel/git/lrg/regulator.git -- 1.7.10.4

1 0

[alsa-devel] alsa 1.0.27 release
by Jaroslav Kysela 12 Apr '13

12 Apr '13

Hello all, the alsa 1.0.27 packages were released. You may download these packages from the main ALSA page: http://www.alsa-project.org . There are no updates for pyalsa and alsa-oss. The full changelog is provided here: http://www.alsa-project.org/main/index.php/Changes_v1.0.26_v1.0.27 Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

1 0

[alsa-devel] [PATCH] asoc: add dummy codec
by Stas Sergeev 12 Apr '13

12 Apr '13

Hi. The attached patch adds a dummy codec for the asoc subsystem. I use it with the driver that programs the soc to drive I2S from an internal PLLs, and the audio then goes to HDMI, so no external codec is used. I've found spdif_transciever.c, which exactly is such a dummy driver, but it is bound to the particular plat device, so I made the device name a config option, and if not set, the plat device is created with the default name. It may be possible to avoid the usage of the codec driver at all, but I haven't succeeded with this, and the existance of spdif_transciever.c suggests there is no such way. Thoughts? Signed-off-by: Stas Sergeev <stsp(a)users.sourceforge.net>

3 14

Re: [alsa-devel] [PATCH 2/4] mfd: wm5102: Map in additional FLL control registers
by Mark Brown 11 Apr '13

11 Apr '13

On Mon, Apr 08, 2013 at 06:22:23PM +0200, Samuel Ortiz wrote: > Hi Mark, > > On Wed, Mar 06, 2013 at 02:28:14PM +0800, Mark Brown wrote: > > Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > --- > > drivers/mfd/wm5102-tables.c | 8 ++++++++ > > 1 file changed, 8 insertions(+) > Did you merge this patchset through the ASoC tree ? Yes, let me know if that's a problem and I can back stuff out but it seemed sensible.

1 0

[alsa-devel] [PATCH 22/30] ASoC: samsung/idma: export idma_reg_addr_init
by Arnd Bergmann 11 Apr '13

11 Apr '13

The idma_reg_addr_init function is used by the samsung i2s driver, which can be a loadable module, so we have to export this function. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> --- sound/soc/samsung/idma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index f36a541..6e5fed3 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -411,6 +411,7 @@ void idma_reg_addr_init(void __iomem *regs, dma_addr_t addr) idma.regs = regs; idma.lp_tx_addr = addr; } +EXPORT_SYMBOL_GPL(idma_reg_addr_init); static struct snd_soc_platform_driver asoc_idma_platform = { .ops = &idma_ops, -- 1.8.1.2

2 1

[alsa-devel] [PATCH 21/30] ASoC: samsung/i2s: fix module_device_table
by Arnd Bergmann 11 Apr '13

11 Apr '13

The second argument to the module_device_table macro must be the name of the device id array. In the samsung i2s driver, there was a small typo, resulting in a build error when building it as a loadable module. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> --- sound/soc/samsung/i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 61f2622..f7c6816 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1289,7 +1289,7 @@ static struct platform_device_id samsung_i2s_driver_ids[] = { }, {}, }; -MODULE_DEVICE_TABLE(platform, samsung-i2s-driver-ids); +MODULE_DEVICE_TABLE(platform, samsung_i2s_driver_ids); #ifdef CONFIG_OF static struct samsung_i2s_dai_data samsung_i2s_dai_data_array[] = { -- 1.8.1.2

2 1

[alsa-devel] [PATCH 19/30] ASoC: samsung: use irq resource for idma
by Arnd Bergmann 11 Apr '13

11 Apr '13

With multiplatform kernels, we cannot use hardwired IRQ numbers in device drivers. This changes the idma driver to use a proper resource, like all other drivers do. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> --- arch/arm/plat-samsung/devs.c | 6 ++++++ sound/soc/samsung/idma.c | 10 ++++++++-- 2 files changed, 14 insertions(+), 2 deletions(-) diff --git a/arch/arm/plat-samsung/devs.c b/arch/arm/plat-samsung/devs.c index 78be9c0..b441908 100644 --- a/arch/arm/plat-samsung/devs.c +++ b/arch/arm/plat-samsung/devs.c @@ -145,14 +145,20 @@ struct platform_device s3c_device_camif = { /* ASOC DMA */ +#ifdef CONFIG_PLAT_S5P +static struct resource samsung_asoc_idma_resource = DEFINE_RES_IRQ(IRQ_I2S0); + struct platform_device samsung_asoc_idma = { .name = "samsung-idma", .id = -1, + .num_resources = 1, + .resource = &samsung_asoc_idma_resource, .dev = { .dma_mask = &samsung_device_dma_mask, .coherent_dma_mask = DMA_BIT_MASK(32), } }; +#endif /* FB */ diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index a07950b..f36a541 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -68,6 +68,8 @@ static struct idma_info { dma_addr_t lp_tx_addr; } idma; +static int idma_irq; + static void idma_getpos(dma_addr_t *src) { *src = idma.lp_tx_addr + @@ -305,7 +307,7 @@ static int idma_open(struct snd_pcm_substream *substream) if (prtd == NULL) return -ENOMEM; - ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd); + ret = request_irq(idma_irq, iis_irq, 0, "i2s", prtd); if (ret < 0) { pr_err("fail to claim i2s irq , ret = %d\n", ret); kfree(prtd); @@ -324,7 +326,7 @@ static int idma_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct idma_ctrl *prtd = runtime->private_data; - free_irq(IRQ_I2S0, prtd); + free_irq(idma_irq, prtd); if (!prtd) pr_err("idma_close called with prtd == NULL\n"); @@ -418,6 +420,10 @@ static struct snd_soc_platform_driver asoc_idma_platform = { static int asoc_idma_platform_probe(struct platform_device *pdev) { + idma_irq = platform_get_irq(pdev, 0); + if (idma_irq < 0) + return idma_irq; + return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform); } -- 1.8.1.2

2 1

[alsa-devel] [PATCH resend] sound: at73c213: Use dev_pm_ops
by Lars-Peter Clausen 11 Apr '13

11 Apr '13

Use dev_pm_ops instead of the deprecated legacy suspend/resume callbacks. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/spi/at73c213.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 4dd60d8..a1a24b9 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -1075,10 +1075,11 @@ out: return 0; } -#ifdef CONFIG_PM -static int snd_at73c213_suspend(struct spi_device *spi, pm_message_t msg) +#ifdef CONFIG_PM_SLEEP + +static int snd_at73c213_suspend(struct device *dev) { - struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_at73c213 *chip = card->private_data; ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); @@ -1087,9 +1088,9 @@ static int snd_at73c213_suspend(struct spi_device *spi, pm_message_t msg) return 0; } -static int snd_at73c213_resume(struct spi_device *spi) +static int snd_at73c213_resume(struct device *dev) { - struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_at73c213 *chip = card->private_data; clk_enable(chip->board->dac_clk); @@ -1097,18 +1098,21 @@ static int snd_at73c213_resume(struct spi_device *spi) return 0; } + +static SIMPLE_DEV_PM_OPS(at73c213_pm_ops, snd_at73c213_suspend, + snd_at73c213_resume); +#define AT73C213_PM_OPS (&at73c213_pm_ops) + #else -#define snd_at73c213_suspend NULL -#define snd_at73c213_resume NULL +#define AT73C213_PM_OPS NULL #endif static struct spi_driver at73c213_driver = { .driver = { .name = "at73c213", + .pm = AT73C213_PM_OPS, }, .probe = snd_at73c213_probe, - .suspend = snd_at73c213_suspend, - .resume = snd_at73c213_resume, .remove = snd_at73c213_remove, }; -- 1.8.0

2 1

[alsa-devel] [PATCH 0/3] Headphone Mic patches
by David Henningsson 11 Apr '13

11 Apr '13

I'm preparing upstreaming of some new hardware, which has headphone mic support, i e, you can use the headphone jack to plug in a TRS/TS mic instead of plugging in a headphone in that jack. For this hardware, the existing hp_mic functionality won't work as the headphone and mic are on different pins. David Henningsson (3): ALSA: hda - allow "Headphone Mic" parser flag ALSA: hda - add some small convenience functions to auto parser ALSA: hda - Handle Headphone Mic jack more generic sound/pci/hda/hda_auto_parser.c | 68 ++++++++++++++++++++++++++++++++------- sound/pci/hda/hda_auto_parser.h | 29 +++++++++++++++-- sound/pci/hda/hda_generic.c | 12 +------ sound/pci/hda/hda_jack.c | 43 +++++++++++++++++-------- 4 files changed, 113 insertions(+), 39 deletions(-) -- 1.7.9.5

2 4

[alsa-devel] what is the difference between SNDRV_PCM_FMTBIT_S24_LE and SNDRV_PCM_FMTBIT_S24_3LE
by Zhang wei 11 Apr '13

11 Apr '13

There are two definitions In the file include/sound/asound.h SNDRV_PCM_FORMAT_S24_3LE and SNDRV_PCM_FORMAT_S24_LE. What is the difference between them? Thanks a lot. zw

2 1

[alsa-devel] VIA VT1731
by ALEKSANDR BOIKO 11 Apr '13

11 Apr '13

Hello! I'm using DAC Audio-GD Master 7 with Voyage MPD. There is a problem with playback audio files different frequency. I've read there is a path - solution. How can I get it? Help please.

2 1

[alsa-devel] ASOC snd_soc_dai_ops for clock mux control
by Gellner, Christoph (ADITG/SW1) 10 Apr '13

10 Apr '13

Hi all, I'm working on the driver of an audio CPU DAI IP providing - Internal clocks which can be used for master clock generation by TX as well as RX section - One oversampling clock available to RX section - One oversampling clock available to TX section. Kernel version is 3.5.7 I want to add support to the CPU DAI driver to - Specify the rate of the input clocks - Select which of the clock inputs shall be used by RX and which shall be used by TX section I'm currently wondering how to map this to existing snd_soc_dai_ops functions. Currently I see only set_sysclk, set_pll, set_clkdiv for clock control. I plan to use set_sysclk to specify the rate as well as the direction of each clock. What is currently missing for me is a way to specify the clock to be used by RX or TX section. I can't really add this in set_sysclk. In case I switch to clock X when set_sysclk is called with clock ID of X, I need a way to distinguish if RX or TX section or both need to switch to this clock. But I can't really double the internal clock X in virtual clocks X_TX, X_RX (one time for TX and one time for RX section) as the internal clock X has only one rate for both sections and this would not represent the hardware correctly. Do you have any recommendation how to implement ? What do you think about adding a new function pointer to snd_soc_dai_ops ? int (*set_clkmux)(struct snd_soc_dai *dai, int mux_id, int source_id); I think this would be the cleanest solution as it models the features of the hardware (the clock multiplexers for RX or TX sections) It might be possible to use set_pll or set_clkdiv to realize. I could use set_pll and ignore the freq_in and freq_out I could use set_clkdiv and interpret div (divider integer value) as a source_id But this is not a 100% clean solution in my opinion. Best regards Christoph

3 3

[alsa-devel] [PATCH 1/3] ALSA: compress_core: Update calc_avail to use cumulative values
by Charles Keepax 10 Apr '13

10 Apr '13

The app_pointer is managed locally by the compress core for memory mapped DSPs but for DSPs that are not memory mapped this would have to be manually updated from within the DSP driver itself, which is hardly very idiomatic. This patch switches to using the cumulative values to calculate the available buffer space because these are already gracefully passed out of the DSP driver to the compress core and otherwise should be functionally equivalent. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 23 +++++------------------ 1 files changed, 5 insertions(+), 18 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index c84abc8..27bd81a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -160,8 +160,6 @@ static int snd_compr_update_tstamp(struct snd_compr_stream *stream, static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, struct snd_compr_avail *avail) { - long avail_calc; /*this needs to be signed variable */ - memset(avail, 0, sizeof(*avail)); snd_compr_update_tstamp(stream, &avail->tstamp); /* Still need to return avail even if tstamp can't be filled in */ @@ -184,22 +182,11 @@ static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, return stream->runtime->buffer_size; } - /* FIXME: this routine isn't consistent, in one test we use - * cumulative values and in the other byte offsets. Do we - * really need the byte offsets if the cumulative values have - * been updated? In the PCM interface app_ptr and hw_ptr are - * already cumulative */ - - avail_calc = stream->runtime->buffer_size - - (stream->runtime->app_pointer - stream->runtime->hw_pointer); - pr_debug("calc avail as %ld, app_ptr %lld, hw+ptr %lld\n", avail_calc, - stream->runtime->app_pointer, - stream->runtime->hw_pointer); - if (avail_calc >= stream->runtime->buffer_size) - avail_calc -= stream->runtime->buffer_size; - pr_debug("ret avail as %ld\n", avail_calc); - avail->avail = avail_calc; - return avail_calc; + avail->avail = stream->runtime->buffer_size - + (stream->runtime->total_bytes_available - + stream->runtime->total_bytes_transferred); + pr_debug("ret avail as %lld\n", avail->avail); + return avail->avail; } static inline size_t snd_compr_get_avail(struct snd_compr_stream *stream) -- 1.7.2.5

5 19

[alsa-devel] [PATCH v2] davinci-mcasp: Add support for multichannel playback
by Michal Bachraty 10 Apr '13

10 Apr '13

Davinci McASP has support for I2S multichannel playback. For I2S playback/receive, each serializer is capable to play 2 channels (L/R) audio data.Serializer function (Playback-receive-none) is configured in DT, depending on hardware specification. It is possible to play less channels than configured in DT. For that purpose,only specific number of active serializers are enabled. McASP FIFO need to have DMA transfer Bcnt set to number of enabled serializers, otherwise no data are transfered to McASP and Alsa generates "DMA/IRQ playback write error (DMA or IRQ trouble?)" error. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 56 ++++++++++++++++++++++++++++++++----- sound/soc/davinci/davinci-pcm.c | 16 ++++++----- sound/soc/davinci/davinci-pcm.h | 1 + 3 files changed, 59 insertions(+), 14 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index afef3fb..b84bb73 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -235,6 +235,10 @@ #define DISMOD (val)(val<<2) #define TXSTATE BIT(4) #define RXSTATE BIT(5) +#define SRMOD_MASK 3 +#define SRMOD_INACTIVE 0 +#define SRMOD_TX 1 +#define SRMOD_RX 2 /* * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits @@ -657,12 +661,15 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, return 0; } -static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, + int channels) { int i; u8 tx_ser = 0; u8 rx_ser = 0; - + u8 ser; + u8 slots = dev->tdm_slots; + u8 max_active_serializers = (channels + slots - 1) / slots; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -680,16 +687,42 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) } for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) + tx_ser++; + if (dev->serial_dir[i] == RX_MODE) + rx_ser++; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ser = tx_ser; + else + ser = rx_ser; + + if (ser < max_active_serializers) { + dev_warn(dev->dev, "stream has more channels (%d) than are " + "enabled in mcasp (%d)\n", channels, ser * slots); + return -EINVAL; + } + + tx_ser = 0; + rx_ser = 0; + + for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) { + if (dev->serial_dir[i] == TX_MODE && + tx_ser < max_active_serializers) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; - } else if (dev->serial_dir[i] == RX_MODE) { + } else if (dev->serial_dir[i] == RX_MODE && + rx_ser < max_active_serializers) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; + } else { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + SRMOD_INACTIVE, SRMOD_MASK); } } @@ -729,6 +762,8 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); } } + + return 0; } static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) @@ -812,8 +847,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, &dev->dma_params[substream->stream]; int word_length; u8 fifo_level; + u8 slots = dev->tdm_slots; + int channels; + struct snd_interval *pcm_channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels = pcm_channels->min; - davinci_hw_common_param(dev, substream->stream); + if (davinci_hw_common_param(dev, substream->stream, channels)) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = dev->txnumevt; else @@ -862,6 +903,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; + dma_params->active_serializers = (channels + slots - 1) / slots; davinci_config_channel_size(dev, word_length); return 0; @@ -936,13 +978,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .name = "davinci-mcasp.0", .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index bb57552..3af8b50 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -182,6 +182,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) unsigned short acnt; unsigned int count; unsigned int fifo_level; + unsigned char serializers = prtd->params->active_serializers; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; @@ -195,14 +196,14 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) data_type = prtd->params->data_type; count = period_size / data_type; if (fifo_level) - count /= fifo_level; + count /= fifo_level * serializers; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; - dst_bidx = 0; - src_cidx = data_type * fifo_level; + dst_bidx = 4; + src_cidx = data_type * fifo_level * serializers; dst_cidx = 0; } else { src = prtd->params->dma_addr; @@ -210,7 +211,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src_bidx = 0; dst_bidx = data_type; src_cidx = 0; - dst_cidx = data_type * fifo_level; + dst_cidx = data_type * fifo_level * serializers; } acnt = prtd->params->acnt; @@ -224,9 +225,10 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, ASYNC); else - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, - ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, + fifo_level * serializers, + count, fifo_level * serializers, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index fbb710c..0d84d32 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -27,6 +27,7 @@ struct davinci_pcm_dma_params { unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; + unsigned char active_serializers; /* num. of active audio serializers */ }; int davinci_soc_platform_register(struct device *dev); -- 1.7.9.5

5 23

Re: [alsa-devel] [alsa-cvslog] [ALSA GIT]ALSA utilities repository branch master updated. v1.0.26-26-ge05b903
by Takashi Iwai 10 Apr '13

10 Apr '13

At Mon, 8 Apr 2013 14:51:54 +0200 (CEST), GIT server wrote: > > This is an automated email from the git hooks/post-receive script. It was > generated because a ref change was pushed to the repository containing > the project "ALSA utilities repository". > > The branch, master has been updated > via e05b903b1fb16e967d838edac408304cd4470fee (commit) > from 46cb355d4ff41c8065b02644b0534ab42d16cb72 (commit) > > Those revisions listed above that are new to this repository have > not appeared on any other notification email; so we list those > revisions in full, below. > > - Log ----------------------------------------------------------------- > commit e05b903b1fb16e967d838edac408304cd4470fee > Author: Jaroslav Kysela <perex(a)perex.cz> > Date: Mon Apr 8 14:49:31 2013 +0200 > > alsactl: move systemd config to the daemon mode > > Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> I'm not thrilled by the silent default behavior change like this. This will affect all systems using systemd from now on. I understand the reason behind it, but I wonder whether it's an overkill. Yet another daemon, unconditionally no matter whether hotplug or not? Hmm... Takashi

4 13

[alsa-devel] [PATCH] ALSA: snd-usb: Playback Design: use usb_set_inferface quirk from more locations
by Daniel Mack 10 Apr '13

10 Apr '13

It turns out the devices from Playback Design need the delay quirk after usb_set_interface from clocks.c as well. Make it a proper quirks function and factor out the code to quirks.c. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/clock.c | 3 +++ sound/usb/pcm.c | 7 +------ sound/usb/quirks.c | 10 ++++++++++ sound/usb/quirks.h | 1 + 4 files changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index ae35e7d..b0ec364 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -32,6 +32,7 @@ #include "card.h" #include "helper.h" #include "clock.h" +#include "quirks.h" static struct uac_clock_source_descriptor * snd_usb_find_clock_source(struct usb_host_interface *ctrl_iface, @@ -392,7 +393,9 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, * interface is active. */ if (rate != prev_rate) { usb_set_interface(dev, iface, 0); + snd_usb_set_interface_quirk(dev); usb_set_interface(dev, iface, fmt->altsetting); + snd_usb_set_interface_quirk(dev); } return 0; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 754cb5b..815a37d 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -349,12 +349,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; - /* - * "Playback Design" products need a 50ms delay after setting the - * USB interface. - */ - if (le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) - mdelay(50); + snd_usb_set_interface_quirk(dev); } subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7882a82..551ae1a 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -892,6 +892,16 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) ep->skip_packets = 16; } +void snd_usb_set_interface_quirk(struct usb_device *dev) +{ + /* + * "Playback Design" products need a 50ms delay after setting the + * USB interface. + */ + if (le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) + mdelay(50); +} + void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size) diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index f10d00e..665e972 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -26,6 +26,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep); +void snd_usb_set_interface_quirk(struct usb_device *dev); void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size); -- 1.7.10.4

2 1

Re: [alsa-devel] [PATCH] ASoC: wm8903: Fix the bypass to HP/LINEOUT when no DAC or ADC is running
by Mark Brown 09 Apr '13

09 Apr '13

On Tue, Apr 09, 2013 at 05:13:59PM +0200, Alban Bedel wrote: > The Charge Pump needs the DSP clock to work properly, without it the > bypass to HP/LINEOUT is not working properly. This requirement is not > mentioned in the datasheet but has been confirmed by Mark Brown from > Wolfson. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 2/2] ASoC: wm8903: Fix the bypass to HP/LINEOUT when no DAC or ADC is running
by Mark Brown 09 Apr '13

09 Apr '13

On Wed, Mar 20, 2013 at 05:37:33PM +0100, Alban Bedel wrote: > output enable sequence requiere the DSP clock. Some input from Wolfson > will be needed to find out if this is just an undocumented requirement > or a chip bug. It seems it's an undocumented requirement. > + { "Left Output Mixer", "Left Bypass Switch", "CLK_DSP" }, > + { "Left Output Mixer", "Right Bypass Switch", "CLK_DSP" }, > + { "Right Output Mixer", "Left Bypass Switch", "CLK_DSP" }, > + { "Right Output Mixer", "Right Bypass Switch", "CLK_DSP" }, The actual change here should be to add a route: { "Charge Pump", NULL, "CLK_DSP" }, Can you submit an update for this or should I do one?

1 0

[alsa-devel] [PATCH v3 3/3] ALSA: hda - delay resuming Haswell HDMI codec in system resume
by mengdong.lin＠intel.com 09 Apr '13

09 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> In system resume, Haswell codec cannot be programmed to D0 before Gfx driver initializes the display pipeline and audio, which will trigger an unsol event on the pin with HDMI/DP cable connected. Otherwise, the connected pin will stay in D3 with right channel muted and thus no sound can be heard. In this patch, Haswell codec will claim it needs "delaying resume" by setting flag "support_delay_resume", so system resume will skip it and mark it as "delay_resumed". On later codec access, Haswell codec will be resumed and hda_call_codec_resume() will call its set_power_states ops intel_haswell_set_ power_state(). For a delayed resume, this ops will enable and wait for the unsol event, and then resume the codec. A 300ms timeout is set in case unsol event is lost. 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 ede8215..22f276c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -89,6 +89,14 @@ struct hdmi_spec { */ struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; + +#ifdef CONFIG_PM + /* + * Non-generic Intel Haswell specific + */ + unsigned int ready_to_resume:1; + wait_queue_head_t resume_wq; +#endif }; @@ -975,6 +983,13 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; +#ifdef CONFIG_PM + if (codec->resume_delayed) { + spec->ready_to_resume = 1; + wake_up(&spec->resume_wq); + } +#endif + hdmi_present_sense(get_pin(spec, pin_idx), 1); snd_hda_jack_report_sync(codec); } @@ -1725,6 +1740,11 @@ static int generic_hdmi_init(struct hda_codec *codec) hdmi_init_pin(codec, pin_nid); snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); } + +#ifdef CONFIG_PM + spec->ready_to_resume = 0; +#endif + return 0; } @@ -1855,6 +1875,60 @@ static const struct hda_fixup hdmi_fixups[] = { }; +#ifdef CONFIG_PM +static void intel_haswell_wait_ready_to_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin; + hda_nid_t pin_nid; + struct hda_jack_tbl *jack; + + per_pin = get_pin(spec, pin_idx); + pin_nid = per_pin->pin_nid; + jack = snd_hda_jack_tbl_get(codec, pin_nid); + if (jack) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | jack->tag); + } + + wait_event_timeout(spec->resume_wq, + spec->ready_to_resume, msecs_to_jiffies(300)); + if (!spec->ready_to_resume) + snd_printk(KERN_WARNING "HDMI: Haswell not ready to resume\n"); +} + +static void intel_haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (codec->resume_delayed && power_state == AC_PWRST_D0) { + intel_haswell_wait_ready_to_resume(codec); + codec->resume_delayed = 0; + } + + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + + snd_hda_codec_set_power_to_all(codec, fg, power_state); +} + +static void intel_haswell_allow_delay_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + + init_waitqueue_head(&spec->resume_wq); + codec->support_delay_resume = 1; + + codec->patch_ops.set_power_state = intel_haswell_set_power_state; +} +#else +define intel_haswell_allow_delay_resume do { } while (0) +#endif + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1878,6 +1952,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + + if (codec->vendor_id == 0x80862807) + intel_haswell_allow_delay_resume(codec); + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.7.10.4

1 0

[alsa-devel] [PATCH v2 2/3] ALSA: hda - system resume may delay resuming a codec
by mengdong.lin＠intel.com 09 Apr '13

09 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> On system resume, a codec may not be ready to resume for some external dependency. e.g. Intel Haswell HDMI codec depends on Gfx to initialize the display pipeline before the codec can resume. To avoid the external dependency blocking system resume, this patch allow system resume ops "snd_hda_resume" skip a not ready codec and mark it as "resume delayed". So the codec resume is delayed until later codec access, assuming the external dependency may be resolved and waiting is acceptable at that time. A codec that needs delaying resume shall explicitly set "support_delay_resume" flag on initialization in its patch ops. And it's up to the codec when it's ready and how to finish the delayed resume. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ba147d7..81e91b8 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5518,7 +5518,10 @@ int snd_hda_resume(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { - hda_call_codec_resume(codec); + if (codec->support_delay_resume) + codec->resume_delayed = 1; + else + hda_call_codec_resume(codec); } return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6061360..5fe7362 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -889,6 +889,8 @@ struct hda_codec { unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ unsigned int pm_down_notified:1; /* PM notified to controller */ unsigned int in_pm:1; /* suspend/resume being performed */ + unsigned int support_delay_resume:1; /* codec support delay resume */ + unsigned int resume_delayed:1; /* resume delayed by PM */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ -- 1.7.10.4

1 0

[alsa-devel] [PATCH v2 1/3] ALSA: hda - add new function snd_hda_bus_reset() for codec communitation error
by mengdong.lin＠intel.com 09 Apr '13

09 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> A new function snd_hda_bus_reset() is defined to reset all codecs on the bus. This is implemented as temporary suspend-and-resume calls to codecs as an ad hoc solution. It will be called by bus reset ops azx_bus_reset() as a rescue of codec communitation error, replacing calls to snd_hda_suspend and snd_hda_resume. It's because snd_hda_resume need change for system resume: It will delay resuming a codec that is not ready to resume for some external dependency, to avoid blocking system resume. But this change of "delay resume" is not suitable for recovering from codec communication error, which need resume the codec immediately after suspend to simulate a bus reset. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 17286b3..ba147d7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5523,6 +5523,29 @@ int snd_hda_resume(struct hda_bus *bus) return 0; } EXPORT_SYMBOL_HDA(snd_hda_resume); + +/** + * snd_hda_bus_reset - reset the codecs on the bus. + * @bus: the HDA bus + * + * This is implemented as temporary suspend-and-resume calls + * to codecs as an ad hoc solution. + * + * Returns 0 if successful. + */ +int snd_hda_bus_reset(struct hda_bus *bus) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &bus->codec_list, list) { + cancel_delayed_work_sync(&codec->jackpoll_work); + if (hda_codec_is_power_on(codec)) + hda_call_codec_suspend(codec, false); + hda_call_codec_resume(codec); + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_bus_reset); #endif /* CONFIG_PM */ /* diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c93f902..6061360 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1076,6 +1076,7 @@ void snd_hda_unlock_devices(struct hda_bus *bus); #ifdef CONFIG_PM int snd_hda_suspend(struct hda_bus *bus); int snd_hda_resume(struct hda_bus *bus); +int snd_hda_bus_reset(struct hda_bus *bus); #endif static inline diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735567e..6f11c7a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1659,8 +1659,7 @@ static void azx_bus_reset(struct hda_bus *bus) struct azx_pcm *p; list_for_each_entry(p, &chip->pcm_list, list) snd_pcm_suspend_all(p->pcm); - snd_hda_suspend(chip->bus); - snd_hda_resume(chip->bus); + snd_hda_bus_reset(chip->bus); } #endif bus->in_reset = 0; -- 1.7.10.4

1 0

[alsa-devel] [PATCH 3/3] ALSA: hda - delay resuming Haswell HDMI codec in system resume
by mengdong.lin＠intel.com 09 Apr '13

09 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> In system resume, Haswell codec cannot be programmed to D0 before Gfx driver initializes the display pipeline and audio, which will trigger an unsol event on the pin with HDMI/DP cable connected. Otherwise, the connected pin will stay in D3 with right channel muted and thus no sound can be heard. In this patch, Haswell codec will claim it needs "delaying resume" by setting flag "support_delay_resume", so system resume will skip it and mark it as "delay_resumed". On later codec access, Haswell codec will be resumed and hda_call_codec_resume() will call its set_power_states ops intel_haswell_set_ power_state(). For a delayed resume, this ops will enable and wait for the unsol event, and then resume the codec. A 300ms timeout is set in case unsol event is lost. 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 ede8215..f39e374 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -89,6 +89,14 @@ struct hdmi_spec { */ struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; + +#ifdef CONFIG_PM + /* + * Non-generic Intel Haswell specific + */ + unsigned int ready_to_resume:1; + wait_queue_head_t resume_wq; +#endif }; @@ -975,6 +983,13 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; +#ifdef CONFIG_PM + if (codec->resume_delayed) { + spec->ready_to_resume = 1; + wake_up(&spec->resume_wq); + } +#endif + hdmi_present_sense(get_pin(spec, pin_idx), 1); snd_hda_jack_report_sync(codec); } @@ -1725,6 +1740,11 @@ static int generic_hdmi_init(struct hda_codec *codec) hdmi_init_pin(codec, pin_nid); snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); } + +#ifdef CONFIG_PM + spec->ready_to_resume = 0; +#endif + return 0; } @@ -1855,6 +1875,61 @@ static const struct hda_fixup hdmi_fixups[] = { }; +#ifdef CONFIG_PM +static void intel_haswell_wait_ready_to_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin; + hda_nid_t pin_nid; + struct hda_jack_tbl *jack; + + per_pin = get_pin(spec, pin_idx); + pin_nid = per_pin->pin_nid; + jack = snd_hda_jack_tbl_get(codec, pin_nid); + if (jack) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | jack->tag); + } + + wait_event_timeout(spec->resume_wq, + spec->ready_to_resume, msecs_to_jiffies(300)); + if (!spec->ready_to_resume) + snd_printd(KERN_WARNING "HDMI: Haswell not ready to resume\n"); +} + +static void intel_haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (codec->resume_delayed && power_state == AC_PWRST_D0) { + intel_haswell_wait_ready_to_resume(codec); + codec->resume_delayed = 0; + } + + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + + snd_hda_codec_set_power_to_all(codec, fg, power_state); +} + +static inline void intel_haswell_allow_delay_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + + init_waitqueue_head(&spec->resume_wq); + codec->support_delay_resume = 1; + + codec->patch_ops.set_power_state = + intel_haswell_set_power_state; +} +#else +define intel_haswell_allow_delay_resume NULL +#endif + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1878,6 +1953,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + + if (codec->vendor_id == 0x80862807) + intel_haswell_allow_delay_resume(codec); + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ALSA: hda - Apply codec delay to wallclock.
by Dylan Reid 09 Apr '13

09 Apr '13

For playback add the codec-side delay to the timestamp, for capture subtract it. This brings the timestamps in line with the time that was recently added to the delay reporting. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_intel.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735567e..ec8ac71 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1889,6 +1889,27 @@ static void azx_timecounter_init(struct snd_pcm_substream *substream, tc->cycle_last = last; } +static u64 azx_add_codec_delay_nsec(struct snd_pcm_substream *substream, + u64 nsec) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + int stream = substream->stream; + struct hda_pcm_stream *hinfo = apcm->hinfo[stream]; + struct hda_codec *codec = apcm->codec; + u64 codec_nsec; + + if (hinfo->ops.get_delay) { + codec_nsec = + hinfo->ops.get_delay(hinfo, codec, substream) * 1000000; + if (stream == SNDRV_PCM_STREAM_CAPTURE) + nsec = (nsec > codec_nsec) ? nsec - codec_nsec : 0; + else if (stream == SNDRV_PCM_STREAM_PLAYBACK) + nsec += codec_nsec; + } + + return nsec; +} + static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, struct timespec *ts) { @@ -1897,6 +1918,7 @@ static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, nsec = timecounter_read(&azx_dev->azx_tc); nsec = div_u64(nsec, 3); /* can be optimized */ + nsec = azx_add_codec_delay_nsec(substream, nsec); *ts = ns_to_timespec(nsec); -- 1.8.1.3.605.g02339dd

3 7

[alsa-devel] [PATCH 2/3] ALSA: hda - system resume may delay resuming a codec
by mengdong.lin＠intel.com 08 Apr '13

08 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> On system resume, a codec may not be ready to resume for some external dependency. e.g. Intel Haswell HDMI codec depends on Gfx to initialize the display pipeline before the codec can resume. To avoid the external dependency blocking system resume, this patch allow system resume ops "snd_hda_resume" skip a not ready codec and mark it as "resume delayed". So the codec resume is delayed until later codec access, assuming the external dependency may be resolved and waiting is acceptable at that time. A codec that needs delaying resume shall explicitly set "support_delay_resume" flag on initialization in its patch ops. And it's up to the codec when it's ready and how to finish the delayed resume. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a05af8b..53b1a1c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5518,7 +5518,10 @@ int snd_hda_resume(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { - hda_call_codec_resume(codec); + if (codec->support_delay_resume) + codec->resume_delayed = 1; + else + hda_call_codec_resume(codec); } return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6061360..5fe7362 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -889,6 +889,8 @@ struct hda_codec { unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ unsigned int pm_down_notified:1; /* PM notified to controller */ unsigned int in_pm:1; /* suspend/resume being performed */ + unsigned int support_delay_resume:1; /* codec support delay resume */ + unsigned int resume_delayed:1; /* resume delayed by PM */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/3] ALSA: hda - add new function snd_hda_bus_reset() for codec communitation error
by mengdong.lin＠intel.com 08 Apr '13

08 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> A new function snd_hda_bus_reset() is defined to suspend and resume all codecs on the hda bus. It will be called by bus reset ops azx_bus_reset() as a rescue of codec communitation error. snd_hda_suspend & snd_hda_resume will no longer be called by azx_bus_reset. So snd_hda_resume will only be used in system resume and may delay resuming a codec. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 17286b3..a05af8b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5523,6 +5523,26 @@ int snd_hda_resume(struct hda_bus *bus) return 0; } EXPORT_SYMBOL_HDA(snd_hda_resume); + +/** + * snd_hda_bus_reset - reset the codecs, i.e. suspend and resume the codecs + * @bus: the HDA bus + * + * Returns 0 if successful. + */ +int snd_hda_bus_reset(struct hda_bus *bus) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &bus->codec_list, list) { + cancel_delayed_work_sync(&codec->jackpoll_work); + if (hda_codec_is_power_on(codec)) + hda_call_codec_suspend(codec, false); + hda_call_codec_resume(codec); + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_bus_reset); #endif /* CONFIG_PM */ /* diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c93f902..6061360 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1076,6 +1076,7 @@ void snd_hda_unlock_devices(struct hda_bus *bus); #ifdef CONFIG_PM int snd_hda_suspend(struct hda_bus *bus); int snd_hda_resume(struct hda_bus *bus); +int snd_hda_bus_reset(struct hda_bus *bus); #endif static inline diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735567e..6f11c7a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1659,8 +1659,7 @@ static void azx_bus_reset(struct hda_bus *bus) struct azx_pcm *p; list_for_each_entry(p, &chip->pcm_list, list) snd_pcm_suspend_all(p->pcm); - snd_hda_suspend(chip->bus); - snd_hda_resume(chip->bus); + snd_hda_bus_reset(chip->bus); } #endif bus->in_reset = 0; -- 1.7.10.4

2 1

Re: [alsa-devel] [RFC] revoke(2) and generic handling of things like remove_proc_entry()
by Clemens Ladisch 08 Apr '13

08 Apr '13

Al Viro wrote: > BTW, snd_card_disconnect() doesn't do anything to existing mappings; smells > like a bug, and there we do have ones with non-trivial ->mmap(). Could > ALSA folks comment? I don't know of any hotplug sound driver that maps memory from a device. All hotplug buses (PCIe, USB, FireWire) involve DMAing from/to main memory. When a sound device is unplugged, sample transfers just stop, but the buffer itself stays as long as the device node is kept opened. (In the case of a capture device, that actually makes sense.) Regards, Clemens

1 0

Re: [alsa-devel] [PATCH] ASoC: Ux500: remove test for undefined Kconfig macro
by Mark Brown 08 Apr '13

08 Apr '13

On Fri, Mar 29, 2013 at 01:28:21PM +0100, Paul Bolle wrote: > A test for CONFIG_SND_SOC_UX500_AB5500 was added in v3.5. But there > never was a corresponding Kconfig symbol so this test has always > evaluated to true. And since AB5500 support was removed in v3.5 it > appears safe to remove this test and a few lines of code. Applied, thanks.

1 0

[alsa-devel] [PATCH] ALSA: hda: fix mute led on HP Envy 15
by Dan Savilonis 08 Apr '13

08 Apr '13

This laptop reports the string HP_Mute_LED_P_G in DMI, but the guess made in hp_set_mute_led() is wrong. So, hard-code the GPIO id. Signed-off-by: Dan Savilonis <djs(a)n-cube.org> --- sound/pci/hda/patch_sigmatel.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1d9d642..3a6b111 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -100,6 +100,7 @@ enum { STAC_92HD83XXX_HEADSET_JACK, STAC_92HD83XXX_HP, STAC_HP_ENVY_BASS, + STAC_HP_ENVY_15, STAC_92HD83XXX_MODELS }; @@ -2092,6 +2093,15 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ } +static void stac92hd83xxx_fixup_hp_envy_15(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->gpio_led = 0x01; /* GPIO0 */ +} + static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -2101,6 +2111,7 @@ static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, spec->headset_jack = 1; } + static const struct hda_fixup stac92hd83xxx_fixups[] = { [STAC_92HD83XXX_REF] = { .type = HDA_FIXUP_PINS, @@ -2169,6 +2180,12 @@ static const struct hda_fixup stac92hd83xxx_fixups[] = { {} }, }, + [STAC_HP_ENVY_15] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd83xxx_fixup_hp_envy_15, + .chained = true, + .chain_id = STAC_92HD83XXX_HP, + }, }; static const struct hda_model_fixup stac92hd83xxx_models[] = { @@ -2184,6 +2201,7 @@ static const struct hda_model_fixup stac92hd83xxx_models[] = { { .id = STAC_92HD83XXX_HP_MIC_LED, .name = "hp-mic-led" }, { .id = STAC_92HD83XXX_HEADSET_JACK, .name = "headset-jack" }, { .id = STAC_HP_ENVY_BASS, .name = "hp-envy-bass" }, + { .id = STAC_HP_ENVY_15, .name = "hp-envy-15" }, {} }; @@ -2269,6 +2287,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP Pavilion dv5", STAC_92HD83XXX_HP_INV_LED), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x148a, "HP Mini", STAC_92HD83XXX_HP_LED), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1688, + "HP Envy 15", STAC_HP_ENVY_15), SND_PCI_QUIRK_VENDOR(PCI_VENDOR_ID_HP, "HP", STAC_92HD83XXX_HP), {} /* terminator */ }; @@ -3968,6 +3988,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) snd_hda_add_verbs(codec, stac92hd83xxx_core_init); + snd_hda_pick_fixup(codec, stac92hd83xxx_models, stac92hd83xxx_fixup_tbl, stac92hd83xxx_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); -- 1.8.1.2

1 0

[alsa-devel] [PATCH 0/8] More dmaengine PCM cleanups
by Lars-Peter Clausen 07 Apr '13

07 Apr '13

Hi, The first few patches in this series are a few unrelated cleanups to different dmaengine PCM drivers and one fix for a small compile error I introduced in the last dmaengine PCM cleanup series. The last four patches introduce a common DAI DMA data struct, which will be used by tegra, omap and fsl/imx. Having a common DAI DMA data struct allows us to factor out some redundant code from these drivers into a common helper function in the dmaengine PCM library. This should hopefully be the last set of cleanups before the generic dmaengine PCM driver. - Lars Lars-Peter Clausen (8): ASoC: ep93xx_pcm: Fix compile error ASoC: tegra: Don't claim to support PCM pause and resume ASoC: ux500_pcm: Use the same snd_pcm_hardware for playback and capture ASoC: dmaengine_pcm: Setup device_fc in snd_hwparams_to_dma_slave_config ASoC: dmaengine-pcm: Add a common DAI DMA data struct ASoC: tegra: Use common DAI DMA data struct ASoC: fsl: Use common DAI DMA data struct ASoC: omap: Use common DAI DMA data include/sound/dmaengine_pcm.h | 24 ++++++++++++++++++ sound/soc/atmel/atmel-pcm-dma.c | 2 -- sound/soc/cirrus/ep93xx-i2s.c | 4 +-- sound/soc/fsl/fsl_ssi.c | 23 ++++++++++++------ sound/soc/fsl/imx-pcm-dma.c | 25 ++++++------------- sound/soc/fsl/imx-pcm-fiq.c | 4 +-- sound/soc/fsl/imx-pcm.h | 16 ++++-------- sound/soc/fsl/imx-ssi.c | 21 +++++++++------- sound/soc/fsl/imx-ssi.h | 7 ++++-- sound/soc/omap/am3517evm.c | 1 - sound/soc/omap/ams-delta.c | 1 - sound/soc/omap/mcbsp.c | 14 ++++++----- sound/soc/omap/mcbsp.h | 7 +++--- sound/soc/omap/n810.c | 1 - sound/soc/omap/omap-abe-twl6040.c | 1 - sound/soc/omap/omap-dmic.c | 22 ++++++++--------- sound/soc/omap/omap-hdmi.c | 24 ++++++++---------- sound/soc/omap/omap-mcbsp.c | 6 ++--- sound/soc/omap/omap-mcpdm.c | 31 +++++++++++------------- sound/soc/omap/omap-pcm.c | 42 ++++---------------------------- sound/soc/omap/omap-pcm.h | 39 ------------------------------ sound/soc/omap/omap-twl4030.c | 1 - sound/soc/omap/omap3pandora.c | 1 - sound/soc/omap/osk5912.c | 1 - sound/soc/omap/rx51.c | 1 - sound/soc/soc-dmaengine-pcm.c | 39 ++++++++++++++++++++++++++++++ sound/soc/tegra/tegra20_ac97.c | 13 +++++----- sound/soc/tegra/tegra20_ac97.h | 4 +-- sound/soc/tegra/tegra20_i2s.c | 13 +++++----- sound/soc/tegra/tegra20_i2s.h | 4 +-- sound/soc/tegra/tegra20_spdif.c | 7 +++--- sound/soc/tegra/tegra20_spdif.h | 4 +-- sound/soc/tegra/tegra30_ahub.c | 8 +++--- sound/soc/tegra/tegra30_ahub.h | 8 +++--- sound/soc/tegra/tegra30_i2s.c | 13 +++++----- sound/soc/tegra/tegra30_i2s.h | 4 +-- sound/soc/tegra/tegra_pcm.c | 40 +++--------------------------- sound/soc/tegra/tegra_pcm.h | 7 ------ sound/soc/ux500/ux500_pcm.c | 51 ++++++++++++--------------------------- sound/soc/ux500/ux500_pcm.h | 14 ----------- 40 files changed, 226 insertions(+), 322 deletions(-) delete mode 100644 sound/soc/omap/omap-pcm.h -- 1.8.0

7 25

[alsa-devel] Samsung S5PV210 I2S DMA
by Mike Dyer 07 Apr '13

07 Apr '13

Hi All, I'm attempting to play audio into a WM8960 codec attached to I2S 0 on a Samsung S5PV210. The DMA engine emits a lot of messages complaining of (I think) a read fault: dma-pl330 dma-pl330.0: Reset Channel-0 CS-20000f FTC-40000 dma-pl330 dma-pl330.0: Reset Channel-0 CS-20000f FTC-40000 ... I've had audio playing using the I2S idma driver, so I know clocks + mixer settings are OK. I'd like to get the external DMA working because I also need to do capture. Any suggestions on how to debug? Cheers, Mike

1 0

[alsa-devel] [PATCH] ALSA: hda/realtek - Add a quirk for AC700 Chromebook.
by Dylan Reid 07 Apr '13

07 Apr '13

Correct pin configs for the Acer AC700. Most importantly indicate that SPDIF is connected, it routes to HDMI out. Similar to Aspire models, chain in the DMIC fixup and allow it to be applied to this codec (ALC269VB) as well. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_realtek.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1bd2d49..145d078 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2617,7 +2617,8 @@ static void alc271_fixup_dmic(struct hda_codec *codec, }; unsigned int cfg; - if (strcmp(codec->chip_name, "ALC271X")) + if (strcmp(codec->chip_name, "ALC271X") && + strcmp(codec->chip_name, "ALC269VB")) return; cfg = snd_hda_codec_get_pincfg(codec, 0x12); if (get_defcfg_connect(cfg) == AC_JACK_PORT_FIXED) @@ -2838,6 +2839,7 @@ enum { ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, + ALC269_FIXUP_ACER_AC700, }; static const struct hda_fixup alc269_fixups[] = { @@ -3010,6 +3012,19 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC271_FIXUP_AMIC_MIC2, }, + [ALC269_FIXUP_ACER_AC700] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x99a3092f }, /* int-mic */ + { 0x14, 0x99130110 }, /* speaker */ + { 0x18, 0x03a11c20 }, /* mic */ + { 0x1e, 0x0346101e }, /* SPDIF1 */ + { 0x21, 0x0321101f }, /* HP out */ + { } + }, + .chained = true, + .chain_id = ALC271_FIXUP_DMIC, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3034,6 +3049,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), + SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), -- 1.8.1.3.605.g02339dd

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_*
by Eldad Zack 07 Apr '13

07 Apr '13

The usb_control_msg() function expects __u16 types and performs the endianness conversions by itself. However, in three places, a conversion is performed before it is handed over to usb_control_msg(), which leads to a double conversion (= no conversion): * snd_usb_nativeinstruments_boot_quirk() * snd_nativeinstruments_control_get() * snd_nativeinstruments_control_put() Caught by sparse: sound/usb/mixer_quirks.c:512:38: warning: incorrect type in argument 6 (different base types) sound/usb/mixer_quirks.c:512:38: expected unsigned short [unsigned] [usertype] index sound/usb/mixer_quirks.c:512:38: got restricted __le16 [usertype] <noident> sound/usb/mixer_quirks.c:543:35: warning: incorrect type in argument 5 (different base types) sound/usb/mixer_quirks.c:543:35: expected unsigned short [unsigned] [usertype] value sound/usb/mixer_quirks.c:543:35: got restricted __le16 [usertype] <noident> sound/usb/mixer_quirks.c:543:56: warning: incorrect type in argument 6 (different base types) sound/usb/mixer_quirks.c:543:56: expected unsigned short [unsigned] [usertype] index sound/usb/mixer_quirks.c:543:56: got restricted __le16 [usertype] <noident> sound/usb/quirks.c:502:35: warning: incorrect type in argument 5 (different base types) sound/usb/quirks.c:502:35: expected unsigned short [unsigned] [usertype] value sound/usb/quirks.c:502:35: got restricted __le16 [usertype] <noident> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/mixer_quirks.c | 4 ++-- sound/usb/quirks.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 497d274..ebe9144 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -509,7 +509,7 @@ static int snd_nativeinstruments_control_get(struct snd_kcontrol *kcontrol, else ret = usb_control_msg(dev, usb_rcvctrlpipe(dev, 0), bRequest, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, - 0, cpu_to_le16(wIndex), + 0, wIndex, &tmp, sizeof(tmp), 1000); up_read(&mixer->chip->shutdown_rwsem); @@ -540,7 +540,7 @@ static int snd_nativeinstruments_control_put(struct snd_kcontrol *kcontrol, else ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), bRequest, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, - cpu_to_le16(wValue), cpu_to_le16(wIndex), + wValue, wIndex, NULL, 0, 1000); up_read(&mixer->chip->shutdown_rwsem); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a2ac004..68e9220 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -499,7 +499,7 @@ static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev) { int ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), 0xaf, USB_TYPE_VENDOR | USB_RECIP_DEVICE, - cpu_to_le16(1), 0, NULL, 0, 1000); + 1, 0, NULL, 0, 1000); if (ret < 0) return ret; -- 1.8.1.5

3 2

[alsa-devel] [PATCH 1/2] ALSA: add DSD formats
by Daniel Mack 07 Apr '13

07 Apr '13

This patch adds two formats for Direct Stream Digital (DSD), a pulse-density encoding format which is described here: https://en.wikipedia.org/wiki/Direct_Stream_Digital DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit stream. In order to provide a compatibility way for pushing DSD samples through ordinary PCM channels, the "DoP open Standard" was invented. See http://www.dsd-guide.com for a copy of the documentation. The two new types added by this patch describe streams that are capable of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8 or x16 data rate, respectively). DSD itself specifies samples in *bit*, while DOP and ALSA handle them as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample rare configuration, according to the following table: configured hardware 352.8kHz 705.6KHz 1411.2KHz <---- sample rate 8-bit 2.8MHz 5.6MHz 11.2MHz 16-bit 5.6MHz 11.2MHz `-----------------------------------' actual DSD sample rates Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- include/sound/pcm.h | 2 ++ include/uapi/sound/asound.h | 4 +++- sound/core/pcm.c | 2 ++ 3 files changed, 7 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index aa7b0a8..d957046 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -181,6 +181,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_G723_24_1B _SNDRV_PCM_FMTBIT(G723_24_1B) #define SNDRV_PCM_FMTBIT_G723_40 _SNDRV_PCM_FMTBIT(G723_40) #define SNDRV_PCM_FMTBIT_G723_40_1B _SNDRV_PCM_FMTBIT(G723_40_1B) +#define SNDRV_PCM_FMTBIT_DSD_DOP_U8 _SNDRV_PCM_FMTBIT(DSD_DOP_U8) +#define SNDRV_PCM_FMTBIT_DSD_DOP_U16 _SNDRV_PCM_FMTBIT(DSD_DOP_U16) #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1774a5c..eb9eda8 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -214,7 +214,9 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_G723_24_1B ((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */ #define SNDRV_PCM_FORMAT_G723_40 ((__force snd_pcm_format_t) 46) /* 8 Samples in 5 bytes */ #define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_G723_40_1B +#define SNDRV_PCM_FORMAT_DSD_DOP_U8 ((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD DOP format (x8) */ +#define SNDRV_PCM_FORMAT_DSD_DOP_U16 ((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD DOP format (x16) */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_DOP_U16 #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 578327e..578a761 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -209,6 +209,8 @@ static char *snd_pcm_format_names[] = { FORMAT(G723_24_1B), FORMAT(G723_40), FORMAT(G723_40_1B), + FORMAT(DSD_DOP_U8), + FORMAT(DSD_DOP_U16), }; const char *snd_pcm_format_name(snd_pcm_format_t format) -- 1.8.1.4

3 8

Re: [alsa-devel] [PATCH 2/2] alsa-lib: bring pcm.h and pcm.c in sync with the kernel list
by Jussi Laako 06 Apr '13

06 Apr '13

On 04/05/2013 10:59 PM, Andreas Koch wrote: > However, they are interpreting the ASIO spec for DSD differently. They > express buffer size in samples, which means that for DSD it is measured > in bits. That's how pretty much everybody, including Sony who co-authored the spec, has been interpreting it so far... Otherwise all the latency calculations become inconsistent and need special exceptions for DSD. I agree the ASIO spec is not very clear, however it always talks about number of samples and then specifically mentions that DSD samples are _packed_ into bytes and thus all buffer sizes have to multiples of eight. I have a hidden configuration flag to flip that either way as necessary, but default is that sizes are in _bits_. - Jussi

2 5

[alsa-devel] [PATCH v2 1/2] alsa-lib: pcm.c: fix indentation
by Daniel Mack 06 Apr '13

06 Apr '13

Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- src/pcm/pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 05737d9..2963711 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1634,11 +1634,11 @@ static const char *const snd_pcm_type_names[] = { PCMTYPE(LADSPA), PCMTYPE(DMIX), PCMTYPE(JACK), - PCMTYPE(DSNOOP), - PCMTYPE(IEC958), + PCMTYPE(DSNOOP), + PCMTYPE(IEC958), PCMTYPE(SOFTVOL), - PCMTYPE(IOPLUG), - PCMTYPE(EXTPLUG), + PCMTYPE(IOPLUG), + PCMTYPE(EXTPLUG), PCMTYPE(MMAP_EMUL), }; -- 1.8.1.4

1 1