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Re: [alsa-devel] [PATCH] ASoC: max98090: add shutdown callback for max98090
by Jianqun 10 Apr '15

10 Apr '15

Hi Anish Kumar 在 04/10/2015 03:38 AM, Anish Kumar 写道: >> To fix pop noise when shutdown,the pop noise during shutdown >> is the pmic cutoff power of codec without any notice. > Is the power turned off without any intimation to the codec driver? > Or is it physically removed? Can you describe the probelm > better? When device is shutdown with playing music, the origin driver does nothing for shutdown operation, as a result, pmic will cut off the power of codec IC, thus cause a pop noise from speaker and headphone. So the patch added a shutdown callback for driver, enable volume smoothing and disable zero cross, this is referenced from mic_input which does the samething before mic switch. I'm very happy if some suggestion from maximintegrate . :-) > > +static void max98090_i2c_shutdown(struct i2c_client *i2c) > +{ > + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); > + > + dev_info(&i2c->dev, "shut down device\n"); > + > + /* Enable volume smoothing, disable zero cross. This will cause > + * a quick 40ms ramp to mute on shutdown. > + */ > + regmap_write(max98090->regmap, > + M98090_REG_LEVEL_CONTROL, M98090_VSENN_MASK); > + regmap_write(max98090->regmap, > + M98090_REG_DEVICE_SHUTDOWN, 0x00); > + msleep(40); > +} > + > >

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[alsa-devel] ASoC: Failed to create DAPM debugfs
by Fabio Estevam 10 Apr '15

10 Apr '15

Hi, On a imx6q-hummingboard the following warning is seen when running linux-next 20150408: [ 2.270006] imx-spdif sound-spdif: ASoC: Failed to create DAPM debugfs directory Looking at sound/soc/fsl/imx-spdif.c and sound/soc/fsl/fsl_spdif.c I am not sure how to avoid such warning. Any ideas? Thanks, Fabio Estevam

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[alsa-devel] [PATCH v2] ASoC: fsl: Add the audio interface acronyms in Kconfig text
by Fabio Estevam 09 Apr '15

09 Apr '15

To keep consistency with the other Kconfig entries, use the audio interface acronyms (SSI and SPDIF) in the Kconfig menu text. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Fix typo in 'interfacce' sound/soc/fsl/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 081e406..fa44fc6 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -24,7 +24,7 @@ config SND_SOC_FSL_SAI in-tree drivers select it automatically. config SND_SOC_FSL_SSI - tristate "Synchronous Serial Interface module support" + tristate "Synchronous Serial Interface module (SSI) support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) select REGMAP_MMIO @@ -35,7 +35,7 @@ config SND_SOC_FSL_SSI in-tree drivers select it automatically. config SND_SOC_FSL_SPDIF - tristate "Sony/Philips Digital Interface module support" + tristate "Sony/Philips Digital Interface (SPDIF) module support" select REGMAP_MMIO select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) -- 1.9.1

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Re: [alsa-devel] [Intel-gfx] [PATCH 15/19] drm/i915: HSW cdclk support
by Daniel Vetter 09 Apr '15

09 Apr '15

On Thu, Apr 09, 2015 at 04:41:26PM +0300, Mika Kahola wrote: > On Thu, 2015-04-09 at 11:32 +0200, Daniel Vetter wrote: > > On Thu, Apr 09, 2015 at 10:24:24AM +0300, Mika Kahola wrote: > > > I did some testing on audio part with HDMI-HDMI and DP-HDMI cables > > > connected to my Haswell box. Before applying the patch I tested as a > > > reference with the latest -nightly (04-08-2015), 4.0-rc7. Unfortunately, > > > I failed to get any audio over HDMI cable. For a reference I tested with > > > the very same setup the vanillla kernel from Linus tree 4.0-rc7 and with > > > that kernel the audio worked ok. Then I did some GIT bisecting and it > > > turned out that the first commit that I failed to get audio working was > > > aa2fee4286e43b4784982b17669b02cc99c1ae55. > > > I rerun the bisecting and this time the result was > > commit 0a599838737a2527c35e4d94f794aefe59df1781 > Merge: 2d846c7 a59d719 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Wed Apr 8 11:29:56 2015 +0200 > > Merge branch 'for-linus' into for-next > > Back merge HD-audio quirks to for-next branch, so that we can apply > a couple of more quirks. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Adding Takashi and intel audio folks. -Daniel > > > I don't have this sha1 anywhere. Can you please double-check? > > > > Also to avoid such issues please always add at least the commit subject > > and author, so that I can find it if it's somehow rebased. > > -Daniel > > > > > > > > To test this patch audio functionality I checkout the -nightly version > > > that works for me and apply the patch and test it. I'll come back with > > > the results later on. > > > > > > I had the module option i915.disable_power_well=0 > > > > > > The test routine that I used for audio testing was > > > speaker-test -c 2 -r 48000 -f S16_LE -t pink --device=plughw:0,3 > > > > > > On Tue, 2015-04-07 at 15:52 +0200, Daniel Vetter wrote: > > > > On Tue, Apr 07, 2015 at 12:29:25PM +0300, Mika Kahola wrote: > > > > > Definitely a good idea to check the audio part as well if there is > > > > > a doubt that by changing CD clock the audio would fail. I can check > > > > > this and I'll get back once I have the results. > > > > > > > > We force a full modeset, which should result in an interrupt on the audio > > > > side, which should result in the audio driver re-reading the current > > > > cdclk. If that's no the case it's buggy already. > > > > -Daniel > > > > > > -- > > > Mika Kahola, Intel OTC > > > > > > > > > > > > > -- > Mika Kahola, Intel OTC > > > -- Daniel Vetter Software Engineer, Intel Corporation http://blog.ffwll.ch

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[alsa-devel] [PATCH 1/3] ASoC: Create card debugfs directory earlier
by Lars-Peter Clausen 09 Apr '15

09 Apr '15

Create the card debugfs directory at the begining of the initilization rather then the end as various steps in the initilization sequence will try to register files and sub-directories in the card directory. Fixes: 4e2576bd36a1 ("ASoC: soc-core: initialize debugfs in snd_soc_instantiate_card()") Reported-by: Fabio Estevam <festevam(a)gmail.com> Reported-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2801578..7bd1599 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1555,6 +1555,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) goto base_error; } + soc_init_card_debugfs(card); + card->dapm.bias_level = SND_SOC_BIAS_OFF; card->dapm.dev = card->dev; card->dapm.card = card; @@ -1684,8 +1686,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); - soc_init_card_debugfs(card); - return 0; probe_aux_dev_err: @@ -1699,6 +1699,7 @@ card_probe_error: if (card->remove) card->remove(card); + soc_cleanup_card_debugfs(card); snd_card_free(card->snd_card); base_error: -- 1.8.0

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Re: [alsa-devel] [PATCH 0/5] HD-audio widget power-save fixes
by Takashi Iwai 09 Apr '15

09 Apr '15

At Thu, 09 Apr 2015 17:12:40 +0800, hwang4 wrote: > > > > On 2015年04月09日 16:40, Takashi Iwai wrote: > > Hi, > > > > this is a series of small fixes for the recently added HD-audio widget > > power-save feature. Especially the last patch should fix the Realtek > > codec breakage. > > > > The patches are found in topic/hda branch of sound git tree, too. > I tested the topic/hda branch on a machine with alc255 codec, the > internal speaker, headphone, internal mic and external mic all worked > very well. Great, I merged now. Takashi

1 0

[alsa-devel] [PATCH] ASoC: rt286: Restore default in probe
by Bard Liao 09 Apr '15

09 Apr '15

RT286 can't do register reset. If the hardware power is still existing in power off, rt286 will keep the register settings. So, we need to restore the default register value in probe to make sure the cache value is the same as the real register value. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 842cfb9..87af81b 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1250,6 +1250,14 @@ static int rt286_i2c_probe(struct i2c_client *i2c, rt286->i2c = i2c; i2c_set_clientdata(i2c, rt286); + /* restore codec default */ + for (i = 0; i < INDEX_CACHE_SIZE; i++) + regmap_write(rt286->regmap, rt286->index_cache[i].reg, + rt286->index_cache[i].def); + for (i = 0; i < ARRAY_SIZE(rt286_reg); i++) + regmap_write(rt286->regmap, rt286_reg[i].reg, + rt286_reg[i].def); + if (pdata) rt286->pdata = *pdata; -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [patch] ASoC: Intel: do cast earlier in sst_cdev_tstamp()
by Dan Carpenter 09 Apr '15

09 Apr '15

My static checker complains about these because it looks like the multiply can overflow and then we cast to a larger data type. I don't think this is a problem, but it's also harmless to do the cast earlier so let's silence the static checker warning. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 718838b..7b50a9d 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -381,7 +381,7 @@ static int sst_cdev_tstamp(struct device *dev, unsigned int str_id, tstamp->copied_total = fw_tstamp.ring_buffer_counter; tstamp->pcm_frames = fw_tstamp.frames_decoded; tstamp->pcm_io_frames = div_u64(fw_tstamp.hardware_counter, - (u64)((stream->num_ch) * SST_GET_BYTES_PER_SAMPLE(24))); + (u64)stream->num_ch * SST_GET_BYTES_PER_SAMPLE(24)); tstamp->sampling_rate = fw_tstamp.sampling_frequency; dev_dbg(dev, "PCM = %u\n", tstamp->pcm_io_frames);

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Re: [alsa-devel] [PATCH 0/4] More aggressive PM for HD-audio
by Takashi Iwai 09 Apr '15

09 Apr '15

At Thu, 26 Mar 2015 20:24:43 +0800, Hui Wang wrote: > > [1 <text/plain; utf-8 (8bit)>] > On 03/21/2015 02:38 PM, Hui Wang wrote: > > On 03/21/2015 12:20 AM, David Henningsson wrote: > >> > >> On 2015-03-18 09:50, Takashi Iwai wrote: > >>> Hi, > >>> > >>> here is a patchset for supporting more aggressive PM for HD-audio. > >>> This allows to change the power state of each widget more dynamically > >>> with jack and stream states. It's activated only when the codec > >>> driver (or via sysfs or f/w patch) sets codec->power_mgmt flag. > >>> > >>> In theory, this should work for the recent Realtek codecs, but > >>> currently I have no machine for test. > >>> > >>> David, could you or your team check whether this works for ALC282 or > >>> such? Just add like: > >>> > >>> --- a/sound/pci/hda/patch_realtek.c > >>> +++ b/sound/pci/hda/patch_realtek.c > >>> @@ -5415,6 +5415,7 @@ static int patch_alc269(struct hda_codec *codec) > >>> > >>> spec = codec->spec; > >>> spec->gen.shared_mic_vref_pin = 0x18; > >>> + codec->power_mgmt = 1; > >>> > >>> snd_hda_pick_fixup(codec, alc269_fixup_models, > >>> alc269_fixup_tbl, alc269_fixups); > >>> > >>> > >>> The patchset is for for-next branch of sound git tree, but they might > >>> be applicable to 4.0-rc (or even older), too. The current patches are > >>> found in topic/hda-power branch. > >> > >> So I hoped to be able to look at this today, but it turns out the > >> machine I was thinking of using for testing has an ALC262 codec, > >> which hardly counts as "new". > >> > >> Hui, is this something you feel like taking on? Otherwise I'll try to > >> talk to someone in Taipei. > >> > > OK， I will look for the machine to do the test next week. > > > > Regards, > > Hui. > > > Sorry for late response, today is my first day in the office back from > vacation, I checked all machines in the Beijing office, none of them has > the ALC282 codec, I will continue to look for the machine from other office. > > And I did the test on the machines with the alc283, alc255, alc292 and > alc269, the testing result were same, there were no sound output from > internal speaker or headphone, and the internal mic or external mic > can't record any sound. The test steps as below: > > 1. power_save_node = 0 > checkout the hda-power branch, build the kernel based on this branch. > Install the kernel to the above machines and boot into the desktop > test internal speaker and internal mic, works very well, plug a headset, > test headphone and external mic, works very well. > run pm_suspend, wait 5 seconds, wakeup the system, redo the above test, > everything works very well. OK, this is expected. The patch shouldn't touch this case. > 2. power_save_node = 1 > enable the power_save_node as below: > @@ -5426,6 +5426,8 @@ static int patch_alc269(struct hda_codec *codec) > > alc_auto_parse_customize_define(codec); > > + codec->power_save_node = 1; > + > if (has_cdefine_beep(codec)) > spec->gen.beep_nid = 0x01; > rebuild the kernel, install the kernel to the above machines and boot > into the desktop > test internal speaker and internal mic, we can play sound to internal > speaker without any errors, but I can't hear any sound from the speaker; > I can use the internal mic to record without errors, but recorded file > did not include any sound pcm (maybe all 0x00 or 0xff) > I plug a headset into the headset jack, the detection works very well, > but I can't hear sound from headphone when play a sound, and I can't use > headset mic to record any sound as well. > > > And I attached 2 alsa-info.txt, one is the power_save_node=0, the other > is the power_save_node=1 Thanks. The alsa-info.sh outputs show no difference but the power state, so the widget attributes seem kept with the power state change, as it seems. Could you give alsa-info.sh output *during* playing with power_save_node=1? Takashi

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[alsa-devel] [PATCH v2] ASoC: fsl_ssi: Do not create debugfs directory
by Fabio Estevam 09 Apr '15

09 Apr '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit 81c7cfd1b22a0 ("ASoC: Move debugfs registration to the component level") ASoC core code deals with registering debugfs, so we should not be creating the debugfs directory locally in order to avoid the following warning: fsl-ssi-dai 2028000.ssi: ASoC: Failed to create component debugfs directory Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Only remove dir creation/remove as suggested by Mark Brown sound/soc/fsl/fsl_ssi.h | 1 - sound/soc/fsl/fsl_ssi_dbg.c | 13 +++---------- 2 files changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 5065105..68aef90 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -211,7 +211,6 @@ struct device; #if IS_ENABLED(CONFIG_DEBUG_FS) struct fsl_ssi_dbg { - struct dentry *dbg_dir; struct dentry *dbg_stats; struct { diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 5469ffb..46cea84 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -142,22 +142,15 @@ static const struct file_operations fsl_ssi_stats_ops = { int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) { - ssi_dbg->dbg_dir = debugfs_create_dir(dev_name(dev), NULL); - if (!ssi_dbg->dbg_dir) + ssi_dbg->dbg_stats = debugfs_create_file("stats", S_IRUGO, NULL, + ssi_dbg, &fsl_ssi_stats_ops); + if (!ssi_dbg->dbg_stats) return -ENOMEM; - ssi_dbg->dbg_stats = debugfs_create_file("stats", S_IRUGO, - ssi_dbg->dbg_dir, ssi_dbg, &fsl_ssi_stats_ops); - if (!ssi_dbg->dbg_stats) { - debugfs_remove(ssi_dbg->dbg_dir); - return -ENOMEM; - } - return 0; } void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg) { debugfs_remove(ssi_dbg->dbg_stats); - debugfs_remove(ssi_dbg->dbg_dir); } -- 1.9.1

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Re: [alsa-devel] [PATCH] ALSA: hda - make power_save threshold per-codec
by Takashi Iwai 09 Apr '15

09 Apr '15

At Wed, 8 Apr 2015 23:24:52 -0700, Matthew Garrett wrote: > > On Wed, Apr 8, 2015 at 10:20 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Wed, 8 Apr 2015 18:53:48 -0700, > > Matthew Garrett wrote: > > > > > > Modern hardware will often have multiple HDA devices, and the desired > > > power saving configuration may vary depending on the codecs attached to > > > each of them. Push the power_save value down to the individual codec > > > structures, keeping the module parameter as a global control mechanism > > > for compatibility purposes. > > > > This patch doesn't apply any longer at all to the recent code, since > > the power saving stuff has been already translated to the standard > > runtime PM. Take a look at linux-next tree. > > Ah! Wonderful! > > > Also we can't break the current behavior controlling the power save > > via option. Many programs rely on this mechanism, thus changing this > > would break its behavior. > > Yeah, I kept that behaviour - anything that touches > /sys/module/snd_hda_intel/parameters/power_save just changes the > settings for all codecs. > > > In the latest code, the common control via power_save module option is > > still there for compatibility, but you can also adjust the value for > > each codec in runtime attribute of each codec sysfs in addition. > > That's ideal - I'll pull down next and rewrite my udev rules. Thanks! Let me know if you find anything broken with the new code. It's pretty new, so any intensive testing is appreciated. thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: hda - Always allow access for POWER_STATE verbs
by Takashi Iwai 09 Apr '15

09 Apr '15

The hdac regmap code checks whether the codec is powered on while accessing, but there must be an exception -- the verbs to control the power state. Currently HD-audio driver doesn't access them via regmap, so this patch doesn't fix any current behavior, but it's just for future. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_regmap.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 64876fa357c9..51f1b5c8a91c 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -239,7 +239,7 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) int verb = get_verb(reg); int err; - if (!codec_is_running(codec)) + if (!codec_is_running(codec) && verb != AC_VERB_GET_POWER_STATE) return -EAGAIN; reg |= (codec->addr << 28); if (is_stereo_amp_verb(reg)) @@ -265,16 +265,16 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) unsigned int verb; int i, bytes, err; - if (!codec_is_running(codec)) - return codec->lazy_cache ? 0 : -EAGAIN; - reg &= ~0x00080000U; /* drop GET bit */ reg |= (codec->addr << 28); + verb = get_verb(reg); + + if (!codec_is_running(codec) && verb != AC_VERB_SET_POWER_STATE) + return codec->lazy_cache ? 0 : -EAGAIN; if (is_stereo_amp_verb(reg)) return hda_reg_write_stereo_amp(codec, reg, val); - verb = get_verb(reg); if (verb == AC_VERB_SET_PROC_COEF) return hda_reg_write_coef(codec, reg, val); -- 2.3.5

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[alsa-devel] [PATCH 0/5 alsa-lib] amixer fixes for enumerated elements
by Takashi Sakamoto 09 Apr '15

09 Apr '15

The amixer still has issues according to sset operation to enumerated items. The issues are: - Fail to handle omitted value of each channel. - The name of enumerated element is truncated. - Unclear separators for values of channels. This patchset fixes these issues, including a small code refactoring for better reading. Takashi Sakamoto (5): amixer: gather local variables in the beginning of functions amixer: fix a bug to handle omitted value in parameter for enumerated element amixer: expand local storage for item name according to kernel code amixer: arrange validation logic amixer: use the same characters for separator. amixer/amixer.c | 60 ++++++++++++++++++++++++++++++++++++++++----------------- 1 file changed, 42 insertions(+), 18 deletions(-) -- 2.1.0

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Re: [alsa-devel] Audio Jack Out does not work
by Dan Carpenter 09 Apr '15

09 Apr '15

So it's 03ad6a8c93b6df2 ('ALSA: hda - Fix "PCM" name being used on one DAC when there are two DACs') which causes the problem? Have you tried to just revert that patch? git show 03ad6a8c93b6df2d65c305b5b5f9474068b45bfb | patch -p1 -R regards, dan carpenter

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[alsa-devel] [PATCH] ALSA: hda - make power_save threshold per-codec
by Matthew Garrett 09 Apr '15

09 Apr '15

Modern hardware will often have multiple HDA devices, and the desired power saving configuration may vary depending on the codecs attached to each of them. Push the power_save value down to the individual codec structures, keeping the module parameter as a global control mechanism for compatibility purposes. Signed-off-by: Matthew Garrett <mjg59(a)coreos.com> --- Documentation/sound/alsa/HD-Audio.txt | 5 +++-- sound/pci/hda/hda_codec.c | 28 ++++++++++++++++++++-------- sound/pci/hda/hda_codec.h | 19 ++++--------------- sound/pci/hda/hda_controller.c | 6 +++--- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 7 +++---- sound/pci/hda/hda_sysfs.c | 30 ++++++++++++++++++++++++++++++ sound/pci/hda/hda_tegra.c | 4 ++-- 8 files changed, 66 insertions(+), 35 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 42a0a39..99f2fb8 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -569,8 +569,9 @@ Power-Saving The power-saving is a kind of auto-suspend of the device. When the device is inactive for a certain time, the device is automatically turned off to save the power. The time to go down is specified via -`power_save` module option, and this option can be changed dynamically -via sysfs. +`power_save` module option. This option can be changed dynamically +via sysfs, either globally (via /sys/module/snd_hda_*/parameters/power_save) +or per-codec (via the power_save sysfs parameter in the codec directory) The power-saving won't work when the analog loopback is enabled on some codecs. Make sure that you mute all unneeded signal routes when diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2fe86d2..5e8394a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -895,7 +895,6 @@ int snd_hda_bus_new(struct snd_card *card, bus->private_data = temp->private_data; bus->pci = temp->pci; bus->modelname = temp->modelname; - bus->power_save = temp->power_save; bus->ops = temp->ops; mutex_init(&bus->cmd_mutex); @@ -1436,8 +1435,9 @@ static void snd_hda_codec_dev_release(struct device *dev) * Returns 0 if successful, or a negative error code. */ int snd_hda_codec_new(struct hda_bus *bus, - unsigned int codec_addr, - struct hda_codec **codecp) + unsigned int codec_addr, + struct hda_codec **codecp, + int power_save) { struct hda_codec *codec; char component[31]; @@ -1495,6 +1495,7 @@ int snd_hda_codec_new(struct hda_bus *bus, INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); codec->depop_delay = -1; + codec->power_save = power_save; codec->fixup_id = HDA_FIXUP_ID_NOT_SET; #ifdef CONFIG_PM @@ -5112,23 +5113,34 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) codec->power_transition = 0; } -#define power_save(codec) \ - ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) - /* Transition to powered down */ static void __snd_hda_power_down(struct hda_codec *codec) { if (!codec->power_on || codec->power_count || codec->power_transition) return; - if (power_save(codec)) { + if (codec->power_save >= 0) { codec->power_transition = -1; /* avoid reentrance */ queue_delayed_work(codec->bus->workq, &codec->power_work, - msecs_to_jiffies(power_save(codec) * 1000)); + msecs_to_jiffies(codec->power_save * 1000)); } } /** + * + * snd_hda_power_save_set - set the power-down time for the codec + * + * @codec: HD-audio codec + * @value: seconds to wait before powering down the codec + */ +void snd_hda_power_save_set(struct hda_codec *codec, int value) +{ + codec->power_save = value; + snd_hda_power_save(codec, 0, false); +} +EXPORT_SYMBOL_GPL(snd_hda_power_save_set); + +/** * snd_hda_power_save - Power-up/down/sync the codec * @codec: HD-audio codec * @delta: the counter delta to change diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 9c8820f..faaab58 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -106,7 +106,6 @@ struct hda_bus_template { void *private_data; struct pci_dev *pci; const char *modelname; - int *power_save; struct hda_bus_ops ops; }; @@ -123,7 +122,6 @@ struct hda_bus { void *private_data; struct pci_dev *pci; const char *modelname; - int *power_save; struct hda_bus_ops ops; /* codec linked list */ @@ -398,6 +396,7 @@ struct hda_codec { struct snd_array jacks; #endif + int power_save; int depop_delay; /* depop delay in ms, -1 for default delay time */ /* fix-up list */ @@ -423,7 +422,7 @@ enum { int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, struct hda_bus **busp); int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - struct hda_codec **codecp); + struct hda_codec **codecp, int power_save); int snd_hda_codec_configure(struct hda_codec *codec); int snd_hda_codec_update_widgets(struct hda_codec *codec); @@ -588,9 +587,11 @@ const char *snd_hda_get_jack_location(u32 cfg); * power saving */ #ifdef CONFIG_PM +void snd_hda_power_save_set(struct hda_codec *codec, int value); void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait); void snd_hda_update_power_acct(struct hda_codec *codec); #else +static inline void snd_hda_power_save_set(struct hda_codec *codec, int value) {} static inline void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait) {} #endif @@ -634,18 +635,6 @@ static inline void snd_hda_power_down(struct hda_codec *codec) snd_hda_power_save(codec, -1, false); } -/** - * snd_hda_power_sync - Synchronize the power-save status - * @codec: HD-audio codec - * - * Synchronize the actual power state with the power account; - * called when power_save parameter is changed - */ -static inline void snd_hda_power_sync(struct hda_codec *codec) -{ - snd_hda_power_save(codec, 0, false); -} - #ifdef CONFIG_SND_HDA_PATCH_LOADER /* * patch firmware diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 17c2637..4bb13ae 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1813,7 +1813,7 @@ static int get_jackpoll_interval(struct azx *chip) /* Codec initialization */ int azx_codec_create(struct azx *chip, const char *model, unsigned int max_slots, - int *power_save_to) + int power_save) { struct hda_bus_template bus_temp; int c, codecs, err; @@ -1827,7 +1827,6 @@ int azx_codec_create(struct azx *chip, const char *model, bus_temp.ops.attach_pcm = azx_attach_pcm_stream; bus_temp.ops.bus_reset = azx_bus_reset; #ifdef CONFIG_PM - bus_temp.power_save = power_save_to; bus_temp.ops.pm_notify = azx_power_notify; #endif #ifdef CONFIG_SND_HDA_DSP_LOADER @@ -1886,7 +1885,8 @@ int azx_codec_create(struct azx *chip, const char *model, for (c = 0; c < max_slots; c++) { if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { struct hda_codec *codec; - err = snd_hda_codec_new(chip->bus, c, &codec); + err = snd_hda_codec_new(chip->bus, c, &codec, + power_save); if (err < 0) continue; codec->jackpoll_interval = get_jackpoll_interval(chip); diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index c90d10f..d187121 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -45,7 +45,7 @@ irqreturn_t azx_interrupt(int irq, void *dev_id); /* Codec interface */ int azx_codec_create(struct azx *chip, const char *model, unsigned int max_slots, - int *power_save_to); + int power_save); int azx_codec_configure(struct azx *chip); int azx_mixer_create(struct azx *chip); int azx_init_stream(struct azx *chip); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a8a1e14..d51779a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -174,7 +174,6 @@ static struct kernel_param_ops param_ops_xint = { #define param_check_xint param_check_int static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; -static int *power_save_addr = &power_save; module_param(power_save, xint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " "(in second, 0 = disable)."); @@ -187,7 +186,7 @@ static bool power_save_controller = 1; module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #else -static int *power_save_addr; +static int power_save = -1; #endif /* CONFIG_PM */ static int align_buffer_size = -1; @@ -754,7 +753,7 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) if (!chip->bus || chip->disabled) continue; list_for_each_entry(c, &chip->bus->codec_list, list) - snd_hda_power_sync(c); + snd_hda_power_save_set(c, power_save); } mutex_unlock(&card_list_lock); return 0; @@ -1898,7 +1897,7 @@ static int azx_probe_continue(struct azx *chip) /* create codec instances */ err = azx_codec_create(chip, model[dev], azx_max_codecs[chip->driver_type], - power_save_addr); + power_save); if (err < 0) goto out_free; diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index ccc962a..69bbb0b 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -44,8 +44,37 @@ static ssize_t power_off_acct_show(struct device *dev, return sprintf(buf, "%u\n", jiffies_to_msecs(codec->power_off_acct)); } +static ssize_t power_save_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct hda_codec *codec = dev_get_drvdata(dev); + + return sprintf(buf, "%d\n", jiffies_to_msecs(codec->power_save)); +} + +static ssize_t power_save_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct hda_codec *codec = dev_get_drvdata(dev); + unsigned long val; + int err = kstrtol(buf, 0, &val); + + if (err < 0) + return err; + + if (val < -1) + return -EINVAL; + + snd_hda_power_save_set(codec, val); + return count; +} + static DEVICE_ATTR_RO(power_on_acct); static DEVICE_ATTR_RO(power_off_acct); +static DEVICE_ATTR_RW(power_save); + #endif /* CONFIG_PM */ #define CODEC_INFO_SHOW(type) \ @@ -759,6 +788,7 @@ static struct attribute *hda_dev_attrs[] = { #ifdef CONFIG_PM &dev_attr_power_on_acct.attr, &dev_attr_power_off_acct.attr, + &dev_attr_power_save.attr, #endif #ifdef CONFIG_SND_HDA_RECONFIG &dev_attr_init_verbs.attr, diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 375e94f..69b76fc 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -82,7 +82,7 @@ module_param(power_save, bint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout (in seconds, 0 = disable)."); #else -static int power_save = 0; +static int power_save = -1; #endif /* @@ -503,7 +503,7 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; /* create codec instances */ - err = azx_codec_create(chip, NULL, 0, &power_save); + err = azx_codec_create(chip, NULL, 0, power_save); if (err < 0) goto out_free; -- 2.3.4

2 1

[alsa-devel] [PATCH 1/4] ASoC: tegra_alc5632: Use card DAPM context to access widgets
by Lars-Peter Clausen 08 Apr '15

08 Apr '15

The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/tegra/tegra_alc5632.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 6dcd06a..ba272e2 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -101,9 +101,6 @@ static const struct snd_kcontrol_new tegra_alc5632_controls[] = { static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card); snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, @@ -118,7 +115,7 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) &tegra_alc5632_hp_jack_gpio); } - snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin(&rtd->card->dapm, "MICBIAS1"); return 0; } -- 1.8.0

2 4

[alsa-devel] [PATCH v4 0/5] ALSA: jack: Refactoring for jack kctls
by Jie Yang 08 Apr '15

08 Apr '15

Currently only hda will create kctls for hda jacks, for asoc, people may need create jack kctls in specific driver. Here we are introducing kctls for each jack, by creating kctls and add them to jack controls list (considering exist of combo jack). At the same time, we will report events for each control in the list. With this new implementation, both HDA and soc jack kctls works fine, they are compatible for old HDA jack kctls, too. For soc, a. snd_jack_new() with NULL jack_kctl; b. for each pin, call snd_jack_add_kctls() to add one kctl(it will call snd_ctl_add() to add kctl to card); For hda, a. call snd_jack_kctl_new()(in __snd_hda_jack_add_kctl()) to new a jack_kctl(also new kctl); and use snd_hda_ctl_add to add kctl to card; b. new a hda_jack_tbl, appoint jack_tbl->jack_kctl = jack_kctl; c. for non-phantom jack, call snd_jack_new() with this jack_kctl, then the jack_kctl will be added to the new created jack; Changes in v4: 1. use snd_ctl_find_id() to get avaliable index; 2. add initial kctl for snd_jack_new(compatible for HDA); 3. add struct snd_jack_kctl * field to struct hda_jack_tbl; 4. new kctls for soc jack during jack pins creating. 5. add a patch to remove exporting snd_kctl_jack_new(). Changes in v3: 1. replace bit index with bit mask in jack_kctl; 2. add exception for SND_JACK_HEADSET and SND_JACK_AVOUT, only create one jack kctl for these two combo jacks, respectively. 3. add NULL check, mem kfree, and fix some potential risk. Change in v2: 1. define jack_kctl struct, and put jack kctl related stuff there; 2. add a patch to remove the existing controls for HDA jack. Jie Yang (5): ALSA: jack: create jack kcontrols for every jack input ALSA: jack: add a parameter to pass kctl for snd_jack_new ALSA: hda - Update to use the new jack kctls method ASoC: jack: create kctls according to jack pins info ALSA: jack: remove export snd_kctl_jack_new() include/sound/jack.h | 25 ++++++- sound/core/ctljack.c | 1 - sound/core/jack.c | 152 +++++++++++++++++++++++++++++++++++++++- sound/pci/hda/hda_jack.c | 52 +++++++------- sound/pci/hda/hda_jack.h | 4 +- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/soc-jack.c | 3 +- 7 files changed, 200 insertions(+), 39 deletions(-) -- 1.9.1

6 20

[alsa-devel] [PATCH] ASoC: rt5645: Adds push button support for rt5650
by Bard Liao 08 Apr '15

08 Apr '15

This patch exports rt5650_set_button_detect function for machine driver to support rt5650 headset button detection function. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 147 +++++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/rt5645.h | 7 ++- 2 files changed, 146 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 69528ae..e547a994 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2368,6 +2368,8 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, static int rt5645_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_PREPARE: if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { @@ -2398,8 +2400,9 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: snd_soc_write(codec, RT5645_DEPOP_M2, 0x1100); - snd_soc_update_bits(codec, RT5645_GEN_CTRL1, - RT5645_DIG_GATE_CTRL, 0); + if (!rt5645->en_button_func) + snd_soc_update_bits(codec, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, 0); snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_VREF1 | RT5645_PWR_MB | RT5645_PWR_BG | RT5645_PWR_VREF2 | @@ -2414,6 +2417,37 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, return 0; } +static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, + bool enable) +{ + if (enable) { + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "ADC L power"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "ADC R power"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "LDO2"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync_unlocked(&codec->dapm); + snd_soc_update_bits(codec, RT5645_INT_IRQ_ST, 0x8, 0x8); + snd_soc_update_bits(codec, RT5650_4BTN_IL_CMD2, 0x8000, 0x8000); + snd_soc_read(codec, RT5650_4BTN_IL_CMD1); + } else { + snd_soc_update_bits(codec, RT5650_4BTN_IL_CMD2, 0x8000, 0x0); + snd_soc_update_bits(codec, RT5645_INT_IRQ_ST, 0x8, 0x0); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "ADC L power"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "ADC R power"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "LDO2"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync_unlocked(&codec->dapm); + } +} + static int rt5645_jack_detect(struct snd_soc_codec *codec) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); @@ -2449,17 +2483,24 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec) val = snd_soc_read(codec, RT5645_IN1_CTRL3) & 0x7; dev_dbg(codec->dev, "val = %d\n", val); - if (val == 1 || val == 2) - jack_type = SND_JACK_HEADSET; - else - jack_type = SND_JACK_HEADPHONE; - snd_soc_dapm_disable_pin(&codec->dapm, "micbias1"); snd_soc_dapm_disable_pin(&codec->dapm, "micbias2"); if (rt5645->pdata.jd_mode == 0) snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); snd_soc_dapm_sync(&codec->dapm); + + if (val == 1 || val == 2) { + jack_type = SND_JACK_HEADSET; + if (rt5645->en_button_func) + rt5645_enable_push_button_irq(codec, true); + } else { + jack_type = SND_JACK_HEADPHONE; + } + + } else { /* jack out */ + if (rt5645->en_button_func) + rt5645_enable_push_button_irq(codec, false); } snd_soc_jack_report(rt5645->hp_jack, jack_type, SND_JACK_HEADPHONE); @@ -2480,6 +2521,42 @@ int rt5645_set_jack_detect(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(rt5645_set_jack_detect); +int rt5650_set_button_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + int ret; + + rt5645->jack = jack; + rt5645->codec_irq.gpiod_dev = codec->dev; + rt5645->codec_irq.name = "rt5650 irq"; + rt5645->codec_irq.report = SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3; + rt5645->codec_irq.debounce_time = 150; + rt5645->codec_irq.wake = true; + rt5645->codec_irq.data = (struct rt5645_priv *)rt5645; + rt5645->codec_irq.jack_status_check = rt5645_irq_detection; + + ret = snd_soc_jack_add_gpios(rt5645->jack, 1, + &rt5645->codec_irq); + if (ret) { + dev_err(codec->dev, "Adding jack GPIO failed\n"); + return ret; + } + + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); + regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1, + RT5645_HP_CB_MASK, RT5645_HP_CB_PU); + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); + rt5645->en_button_func = true; + + return 0; +} +EXPORT_SYMBOL_GPL(rt5650_set_button_detect); + static void rt5645_jack_detect_work(struct work_struct *work) { struct rt5645_priv *rt5645 = @@ -2498,6 +2575,62 @@ static irqreturn_t rt5645_irq(int irq, void *data) return IRQ_HANDLED; } +static int rt5645_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5650_4BTN_IL_CMD1); + pr_debug("val=0x%x\n", val); + btn_type = val & 0xfff0; + snd_soc_write(codec, RT5650_4BTN_IL_CMD1, val); + + return btn_type; +} + +static int rt5645_irq_detection(void *data) +{ + struct rt5645_priv *rt5645 = (struct rt5645_priv *)data; + struct snd_soc_jack *jack = &rt5645->jack; + int btn_type, report = SND_JACK_HEADSET; + + btn_type = rt5645_button_detect(rt5645->codec); + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + report |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + report |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + report |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + report |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + dev_err(rt5645->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + /* keep the jack status */ + report = jack->status; + break; + } + + pr_debug("report=%x\n", report); + + return report; +} + static int rt5645_probe(struct snd_soc_codec *codec) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index db78e94..bf29d8b 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2184,7 +2184,9 @@ struct rt5645_priv { struct i2c_client *i2c; struct snd_soc_jack *hp_jack; struct snd_soc_jack *mic_jack; + struct snd_soc_jack *jack; struct delayed_work jack_detect_work; + struct snd_soc_jack_gpio codec_irq; int codec_type; int sysclk; @@ -2196,9 +2198,12 @@ struct rt5645_priv { int pll_src; int pll_in; int pll_out; + + bool en_button_func; }; int rt5645_set_jack_detect(struct snd_soc_codec *codec, struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); - +int rt5650_set_button_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); #endif /* __RT5645_H__ */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v3] ASoC: max98090: add shutdown callback for max98090
by Caesar Wang 08 Apr '15

08 Apr '15

To fix pop noise when shutdown,the pop noise during shutdown is the pmic cutoff power of codec without any notice. Signed-off-by: jay.xu <xjq(a)rock-chips.com> Signed-off-by: zhengxing <zhengxing(a)rock-chips.com> Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> --- Changes in v3: - modify the shutdown function before remove(..) - fix the `Serien-cc` Changes in v2: - remove the dev_info(..) - fix the comment style - add the max98090_i2c_shutdown() in remove() fuction sound/soc/codecs/max98090.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index b112b1c..3e33ef2 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2605,8 +2605,24 @@ err_enable: return ret; } +static void max98090_i2c_shutdown(struct i2c_client *i2c) +{ + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); + + /* + * Enable volume smoothing, disable zero cross. This will cause + * a quick 40ms ramp to mute on shutdown. + */ + regmap_write(max98090->regmap, + M98090_REG_LEVEL_CONTROL, M98090_VSENN_MASK); + regmap_write(max98090->regmap, + M98090_REG_DEVICE_SHUTDOWN, 0x00); + msleep(40); +} + static int max98090_i2c_remove(struct i2c_client *client) { + max98090_i2c_shutdown(client); snd_soc_unregister_codec(&client->dev); return 0; } @@ -2696,6 +2712,7 @@ static struct i2c_driver max98090_i2c_driver = { .acpi_match_table = ACPI_PTR(max98090_acpi_match), }, .probe = max98090_i2c_probe, + .shutdown = max98090_i2c_shutdown, .remove = max98090_i2c_remove, .id_table = max98090_i2c_id, }; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: Remove custom fsl_ssi_debugfs_create/remove()
by Fabio Estevam 08 Apr '15

08 Apr '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit 81c7cfd1b22a0 ("ASoC: Move debugfs registration to the component level") ASoC core code deals with registering debugfs, so we should remove the custom fsl_ssi_debugfs_create/remove() to avoid the following warning: fsl-ssi-dai 2028000.ssi: ASoC: Failed to create component debugfs directory Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 11 +---------- sound/soc/fsl/fsl_ssi.h | 13 ------------- sound/soc/fsl/fsl_ssi_dbg.c | 22 ---------------------- 3 files changed, 1 insertion(+), 45 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6b0c8f7..b8cdb8f 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1411,10 +1411,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) } } - ret = fsl_ssi_debugfs_create(&ssi_private->dbg_stats, &pdev->dev); - if (ret) - goto error_irq; - /* * If codec-handle property is missing from SSI node, we assume * that the machine driver uses new binding which does not require @@ -1441,7 +1437,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (IS_ERR(ssi_private->pdev)) { ret = PTR_ERR(ssi_private->pdev); dev_err(&pdev->dev, "failed to register platform: %d\n", ret); - goto error_sound_card; + goto error_irq; } done: @@ -1451,9 +1447,6 @@ done: return 0; -error_sound_card: - fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); - error_irq: snd_soc_unregister_component(&pdev->dev); @@ -1468,8 +1461,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) { struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev); - fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); - if (ssi_private->pdev) platform_device_unregister(ssi_private->pdev); snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 5065105..4767df1 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -241,10 +241,6 @@ struct fsl_ssi_dbg { void fsl_ssi_dbg_isr(struct fsl_ssi_dbg *ssi_dbg, u32 sisr); -int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev); - -void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg); - #else struct fsl_ssi_dbg { @@ -254,15 +250,6 @@ static inline void fsl_ssi_dbg_isr(struct fsl_ssi_dbg *stats, u32 sisr) { } -static inline int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, - struct device *dev) -{ - return 0; -} - -static inline void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg) -{ -} #endif /* ! IS_ENABLED(CONFIG_DEBUG_FS) */ #endif diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 5469ffb..118064c 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -139,25 +139,3 @@ static const struct file_operations fsl_ssi_stats_ops = { .llseek = seq_lseek, .release = single_release, }; - -int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) -{ - ssi_dbg->dbg_dir = debugfs_create_dir(dev_name(dev), NULL); - if (!ssi_dbg->dbg_dir) - return -ENOMEM; - - ssi_dbg->dbg_stats = debugfs_create_file("stats", S_IRUGO, - ssi_dbg->dbg_dir, ssi_dbg, &fsl_ssi_stats_ops); - if (!ssi_dbg->dbg_stats) { - debugfs_remove(ssi_dbg->dbg_dir); - return -ENOMEM; - } - - return 0; -} - -void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg) -{ - debugfs_remove(ssi_dbg->dbg_stats); - debugfs_remove(ssi_dbg->dbg_dir); -} -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: remove unused functions
by Sudip Mukherjee 08 Apr '15

08 Apr '15

these functions were never called by anyone. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/intel/atom/sst/sst_pvt.c | 24 ------------------------ 1 file changed, 24 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index 2d74249..adb32fe 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -111,30 +111,6 @@ int sst_wait_interruptible(struct intel_sst_drv *sst_drv_ctx, } -static unsigned long long read_shim_data(struct intel_sst_drv *sst, int addr) -{ - unsigned long long val = 0; - - switch (sst->dev_id) { - case SST_MRFLD_PCI_ID: - case SST_BYT_ACPI_ID: - val = sst_shim_read64(sst->shim, addr); - break; - } - return val; -} - -static void write_shim_data(struct intel_sst_drv *sst, int addr, - unsigned long long data) -{ - switch (sst->dev_id) { - case SST_MRFLD_PCI_ID: - case SST_BYT_ACPI_ID: - sst_shim_write64(sst->shim, addr, (u64) data); - break; - } -} - /* * sst_wait_timeout - wait on event for timeout * -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ALSA: bebob: fix to processing in big-endian machine for sending cue
by Takashi Sakamoto 08 Apr '15

08 Apr '15

Some M-Audio devices require to receive bootup command just after powering on, while codes in BeBoB driver doesn't work properly in big-endian machine because the command should be aligned by little-endian. This commit fixes this bug. This fix should go to stable kernel. Cc: Takayuki Shiroma <t.shiroma.oki(a)gmail.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_maudio.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index a422aaa..9ee25a6 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -96,10 +96,10 @@ int snd_bebob_maudio_load_firmware(struct fw_unit *unit) struct fw_device *device = fw_parent_device(unit); int err, rcode; u64 date; - __be32 cues[3] = { - MAUDIO_BOOTLOADER_CUE1, - MAUDIO_BOOTLOADER_CUE2, - MAUDIO_BOOTLOADER_CUE3 + __le32 cues[3] = { + cpu_to_le32(MAUDIO_BOOTLOADER_CUE1), + cpu_to_le32(MAUDIO_BOOTLOADER_CUE2), + cpu_to_le32(MAUDIO_BOOTLOADER_CUE3) }; /* check date of software used to build */ -- 2.1.0

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Create AFG sysfs node at last
by Takashi Iwai 08 Apr '15

08 Apr '15

... so that user-space can know that the whole nodes have been created. Unfortunately, this can't be implemented easily in race-free way, so it's a kind of compromise. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_sysfs.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index b358d5157802..18aea43d230d 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -313,12 +313,12 @@ static void widget_tree_free(struct hdac_device *codec) if (!tree) return; + free_widget_node(tree->afg, &widget_afg_group); if (tree->nodes) { for (p = tree->nodes; *p; p++) free_widget_node(*p, &widget_node_group); kfree(tree->nodes); } - free_widget_node(tree->afg, &widget_afg_group); if (tree->root) kobject_put(tree->root); kfree(tree); @@ -362,13 +362,6 @@ static int widget_tree_create(struct hdac_device *codec) if (!tree->root) return -ENOMEM; - if (codec->afg) { - err = add_widget_node(tree->root, codec->afg, - &widget_afg_group, &tree->afg); - if (err < 0) - return err; - } - tree->nodes = kcalloc(codec->num_nodes + 1, sizeof(*tree->nodes), GFP_KERNEL); if (!tree->nodes) @@ -381,6 +374,13 @@ static int widget_tree_create(struct hdac_device *codec) return err; } + if (codec->afg) { + err = add_widget_node(tree->root, codec->afg, + &widget_afg_group, &tree->afg); + if (err < 0) + return err; + } + kobject_uevent(tree->root, KOBJ_CHANGE); return 0; } -- 2.3.4

1 1

[alsa-devel] [PATCH v2] ASoC: max98090: add shutdown callback for max98090
by Caesar Wang 08 Apr '15

08 Apr '15

To fix pop noise when shutdown,the pop noise during shutdown is the pmic cutoff power of codec without any notice. Signed-off-by: jay.xu <xjq(a)rock-chips.com> Signed-off-by: zhengxing <zhengxing(a)rock-chips.com> Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> Serien-cc: linux-kernel(a)vger.kernel.org Serien-cc: devicetree(a)vger.kernel.org Serien-cc: dianders(a)chromium.org --- Changes in v2: - remove the dev_info(..) - fix the comment style - add the max98090_i2c_shutdown() in remove() fuction sound/soc/codecs/max98090.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index b112b1c..e59013a 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2607,10 +2607,26 @@ err_enable: static int max98090_i2c_remove(struct i2c_client *client) { + max98090_i2c_shutdown(client); snd_soc_unregister_codec(&client->dev); return 0; } +static void max98090_i2c_shutdown(struct i2c_client *i2c) +{ + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); + + /* + * Enable volume smoothing, disable zero cross. This will cause + * a quick 40ms ramp to mute on shutdown. + */ + regmap_write(max98090->regmap, + M98090_REG_LEVEL_CONTROL, M98090_VSENN_MASK); + regmap_write(max98090->regmap, + M98090_REG_DEVICE_SHUTDOWN, 0x00); + msleep(40); +} + #ifdef CONFIG_PM static int max98090_runtime_resume(struct device *dev) { @@ -2697,6 +2713,7 @@ static struct i2c_driver max98090_i2c_driver = { }, .probe = max98090_i2c_probe, .remove = max98090_i2c_remove, + .shutdown = max98090_i2c_shutdown, .id_table = max98090_i2c_id, }; -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: Intel: Fix a buffer overflow issue
by Jie Yang 08 Apr '15

08 Apr '15

0day robot reported a buffer overflow issue: ... sound/soc/intel/haswell/sst-haswell-pcm.c:1107 hsw_pcm_probe() error: buffer\ overflow 'hsw_dais' 4 <= 4 sound/soc/intel/haswell/sst-haswell-pcm.c:1109 hsw_pcm_probe() error: buffer\ overflow 'hsw_dais' 4 <= 4 ... Fix it by initializing the index(i) to correct value. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/haswell/sst-haswell-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 157b3a6..23ae040 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -1103,7 +1103,7 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) return 0; err: - for (;i >= 0; i--) { + for (--i; i >= 0; i--) { if (hsw_dais[i].playback.channels_min) snd_dma_free_pages(&priv_data->dmab[i][0]); if (hsw_dais[i].capture.channels_min) -- 1.9.1

2 1

[alsa-devel] [PATCH 0/1] ASoC: max98090: add shutdown callback for max98090
by Caesar Wang 08 Apr '15

08 Apr '15

Tested on veyron devices,play music then shutdown device,here carefully to speaker or headphone. Caesar Wang (1): ASoC: max98090: add shutdown callback for max98090 sound/soc/codecs/max98090.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) -- 1.9.1

3 6

Re: [alsa-devel] Add delay in alc283_init
by Takashi Iwai 08 Apr '15

08 Apr '15

At Wed, 8 Apr 2015 08:56:48 +0000, Kailang wrote: > > Hi Takashi, > > This patch will make more stable for getting pin sense during resume state. > This is tested for Chrome Box machine. > Many Thanks. Thanks, applied now with Cc to stable. Takashi > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0000-add-delay-before-get-pin-sense.patch <application/octet-stream (base64)>] >

1 0

[alsa-devel] 1st draft: ALSA Json Gateway and HTML5 UI
by Fulup Ar Foll 07 Apr '15

07 Apr '15

Hi, I would need some help to finish a json version of amixer. Porting cget is done, but for cset I have an issue with "snd_hctl_find_elem (handle, id)". I'm not sure what's the right method to pass it. I've tried every options I could imagine. But if it works on Mint/32bit, it miserably fail on OpenSuse/64. Furthermore amixer.c as this strange comment /* FIXME: Remove it when hctl find works ok !!! */ and they do not use snd_hctl_find_elem. I posted a small C code to highlight the issue at [1]https://gist.github.com/fulup-bzh/891e5627be325150337f My question is simple, did I miss something ? should I move to lower level API ? Note: While the painting for my json-gateway is not dry, I nevertheless pushed a initial running version with cget at [2]https://github.com/fulup-bzh/AlsaJsonGateway I also have a small HTML5/UI at [3]http://breizhme.net/alsajson/mixers/ajg#/scarlett?card=2 While UI code is very dependant on each soundcard [in my case Scarlett/Focurite] the Json-Gateway is 100% independent of the board. If anything one have an idea, please let me know. Fulup References 1. https://gist.github.com/fulup-bzh/891e5627be325150337f 2. https://github.com/fulup-bzh/AlsaJsonGateway 3. http://breizhme.net/alsajson/mixers/ajg#/scarlett?card=2

2 1

[alsa-devel] [PATCH][ASoC]Add ability to remove rate constraints from generic ASoC AC'97 CODEC driver
by Maciej S. Szmigiero 07 Apr '15

07 Apr '15

Add ability to remove rate constraints from generic ASoC AC'97 CODEC driver via passed platform data, make it selectable in config. This way this driver can be used for platforms which don't need specialized AC'97 CODEC drivers while at the same avoiding code duplication from implementing equivalent functionality in a controller driver. There is no change in behavior when no platform data is passed. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> diff --git a/include/sound/soc-ac97.h b/include/sound/soc-ac97.h new file mode 100644 index 0000000..ceb4e2f --- /dev/null +++ b/include/sound/soc-ac97.h @@ -0,0 +1,17 @@ +/* + * Platform data for generic ASoC AC97 CODEC driver. + * + * 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. + */ +#ifndef __LINUX_SND__SOC_AC97_H +#define __LINUX_SND__SOC_AC97_H + +struct snd_soc_ac97_codec_platform_data { + bool playback_rate_constrained; + bool capture_rate_constrained; +}; + +#endif /* __LINUX_SND__SOC_AC97_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0bddd92..92d6d39 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,7 +16,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AB8500_CODEC if ABX500_CORE - select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS + select SND_SOC_AC97_CODEC select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD193X_SPI if SPI_MASTER select SND_SOC_AD193X_I2C if I2C @@ -210,8 +210,9 @@ config SND_SOC_AB8500_CODEC tristate config SND_SOC_AC97_CODEC - tristate + tristate "Build generic ASoC AC97 CODEC driver" select SND_AC97_CODEC + select SND_SOC_AC97_BUS config SND_SOC_AD1836 tristate diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index d0ac723..c3fb845 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -20,9 +20,16 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> +#include <sound/soc-ac97.h> #include <sound/initval.h> #include <sound/soc.h> +struct ac97_private { + struct snd_ac97 *ac97; + bool playback_rate_constrained; + bool capture_rate_constrained; +}; + static const struct snd_soc_dapm_widget ac97_widgets[] = { SND_SOC_DAPM_INPUT("RX"), SND_SOC_DAPM_OUTPUT("TX"), @@ -33,22 +40,53 @@ static const struct snd_soc_dapm_route ac97_routes[] = { { "TX", NULL, "AC97 Playback" }, }; +static const unsigned int default_rates[] = { + 8000, 11025, 22050, 44100, 48000 +}; + +static const struct snd_pcm_hw_constraint_list default_rate_constraints = { + .count = ARRAY_SIZE(default_rates), + .list = default_rates, +}; + +static int ac97_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ac97_private *ac97 = snd_soc_codec_get_drvdata(codec); + bool constrained; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + constrained = ac97->playback_rate_constrained; + else + constrained = ac97->capture_rate_constrained; + + if (!constrained) + return 0; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &default_rate_constraints); + else + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &default_rate_constraints); + + return 0; +} + static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + struct ac97_private *ac97 = snd_soc_codec_get_drvdata(codec); int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; - return snd_ac97_set_rate(ac97, reg, substream->runtime->rate); + return snd_ac97_set_rate(ac97->ac97, reg, substream->runtime->rate); } -#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000) - static const struct snd_soc_dai_ops ac97_dai_ops = { + .startup = ac97_startup, .prepare = ac97_prepare, }; @@ -58,20 +96,21 @@ static struct snd_soc_dai_driver ac97_dai = { .stream_name = "AC97 Playback", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .capture = { .stream_name = "AC97 Capture", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &ac97_dai_ops, }; static int ac97_soc_probe(struct snd_soc_codec *codec) { - struct snd_ac97 *ac97; + struct ac97_private *ac97 = snd_soc_codec_get_drvdata(codec); + struct snd_ac97_bus *ac97_bus; struct snd_ac97_template ac97_template; int ret; @@ -83,31 +122,28 @@ static int ac97_soc_probe(struct snd_soc_codec *codec) return ret; memset(&ac97_template, 0, sizeof(struct snd_ac97_template)); - ret = snd_ac97_mixer(ac97_bus, &ac97_template, &ac97); + ret = snd_ac97_mixer(ac97_bus, &ac97_template, &ac97->ac97); if (ret < 0) return ret; - snd_soc_codec_set_drvdata(codec, ac97); - return 0; } #ifdef CONFIG_PM static int ac97_soc_suspend(struct snd_soc_codec *codec) { - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + struct ac97_private *ac97 = snd_soc_codec_get_drvdata(codec); - snd_ac97_suspend(ac97); + snd_ac97_suspend(ac97->ac97); return 0; } static int ac97_soc_resume(struct snd_soc_codec *codec) { + struct ac97_private *ac97 = snd_soc_codec_get_drvdata(codec); - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - - snd_ac97_resume(ac97); + snd_ac97_resume(ac97->ac97); return 0; } @@ -129,6 +165,27 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = { static int ac97_probe(struct platform_device *pdev) { + struct ac97_private *ac97 = + devm_kzalloc(&pdev->dev, sizeof(struct ac97_private), + GFP_KERNEL); + struct snd_soc_ac97_codec_platform_data *pdata = + pdev->dev.platform_data; + + if (!ac97) + return -ENOMEM; + + if (pdata) { + ac97->playback_rate_constrained = + pdata->playback_rate_constrained; + ac97->capture_rate_constrained = + pdata->capture_rate_constrained; + } else { + ac97->playback_rate_constrained = 1; + ac97->capture_rate_constrained = 1; + } + + platform_set_drvdata(pdev, ac97); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_ac97, &ac97_dai, 1); }

2 6

[alsa-devel] [PATCH 00/14] Fix fallout from per-user struct clk patches
by Russell King - ARM Linux 07 Apr '15

07 Apr '15

Sorry for posting this soo close to the 4.1 merge window, I had completely forgotten about this chunk of work I did earlier this month. The per-user struct clk patches rather badly broke clkdev and various other places. This was reported, but was forgotten about. Really, the per-user clk stuff should've been reverted, but we've lived with it far too long for that. So, our only other option is to now rush these patches into 4.1 and hope for the best. The series cleans up quite a number of places too... arch/arm/mach-davinci/da850.c | 1 + arch/arm/mach-lpc32xx/clock.c | 5 +- arch/arm/mach-omap1/board-nokia770.c | 2 +- arch/arm/mach-omap2/clkt2xxx_virt_prcm_set.c | 12 +--- arch/arm/mach-omap2/omap_device.c | 24 +++----- arch/arm/mach-pxa/eseries.c | 1 + arch/arm/mach-pxa/lubbock.c | 1 + arch/arm/mach-pxa/tosa.c | 1 + arch/arm/plat-orion/common.c | 6 +- arch/sh/kernel/cpu/sh4a/clock-sh7734.c | 3 +- arch/sh/kernel/cpu/sh4a/clock-sh7757.c | 4 +- arch/sh/kernel/cpu/sh4a/clock-sh7785.c | 4 +- arch/sh/kernel/cpu/sh4a/clock-sh7786.c | 4 +- arch/sh/kernel/cpu/sh4a/clock-shx3.c | 4 +- drivers/clk/clk-s2mps11.c | 4 +- drivers/clk/clkdev.c | 83 ++++++++++++++++++++-------- drivers/media/platform/omap3isp/isp.c | 18 ------ drivers/media/platform/omap3isp/isp.h | 1 - include/linux/clk.h | 27 ++++----- include/linux/clkdev.h | 6 +- include/media/omap3isp.h | 6 -- sound/soc/sh/migor.c | 3 +- 22 files changed, 108 insertions(+), 112 deletions(-) -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

10 39

[alsa-devel] [PATCH] ALSA: hda_intel: add AZX_DCAPS_I915_POWERWELL for SKL and BSW
by libin.yang＠intel.com 07 Apr '15

07 Apr '15

From: Libin Yang <libin.yang(a)intel.com> HDMI/DP codec on SKL/BSW is in the power well. The power well must be turned on before probing the HDMI/DP codec. This is a temporary patch, which will power on the powerwell by adding AZX_DCAPS_I915_POWERWELL for SKL and BSW. After restructuring and new flag is added, this patch will be reverted. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/hda_intel.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 15a8299..e1c2105 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -297,8 +297,12 @@ enum { AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ AZX_DCAPS_SNOOP_TYPE(SCH)) +#define AZX_DCAPS_INTEL_BRASWELL \ + (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_I915_POWERWELL) + #define AZX_DCAPS_INTEL_SKYLAKE \ - (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG) + (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\ + AZX_DCAPS_I915_POWERWELL) /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -1991,7 +1995,7 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Braswell */ { PCI_DEVICE(0x8086, 0x2284), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BRASWELL }, /* ICH6 */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, -- 1.9.1

2 1

[alsa-devel] [PATCH 1/4] ASoC: wm8804: Add support for hardware reset line
by Charles Keepax 07 Apr '15

07 Apr '15

It is best to use the physical reset if it is available. This patch adds support for a GPIO controlled physical reset for the chip. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8804.c | 26 +++++++++++++++++++++----- 1 files changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index f44da83..cc168c4 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -13,6 +13,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> +#include <linux/gpio/consumer.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/of_device.h> @@ -61,6 +62,8 @@ struct wm8804_priv { struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; int mclk_div; + + struct gpio_desc *reset; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -182,7 +185,7 @@ static bool wm8804_volatile(struct device *dev, unsigned int reg) } } -static int wm8804_reset(struct wm8804_priv *wm8804) +static int wm8804_soft_reset(struct wm8804_priv *wm8804) { return regmap_write(wm8804->regmap, WM8804_RST_DEVID1, 0x0); } @@ -586,6 +589,14 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) wm8804->regmap = regmap; + wm8804->reset = devm_gpiod_get_optional(dev, "wlf,reset", + GPIOD_OUT_LOW); + if (IS_ERR(wm8804->reset)) { + ret = PTR_ERR(wm8804->reset); + dev_err(dev, "Failed to get reset line: %d\n", ret); + return ret; + } + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) wm8804->supplies[i].supply = wm8804_supply_names[i]; @@ -620,6 +631,9 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) return ret; } + if (wm8804->reset) + gpiod_set_value_cansleep(wm8804->reset, 1); + ret = regmap_read(regmap, WM8804_RST_DEVID1, &id1); if (ret < 0) { dev_err(dev, "Failed to read device ID: %d\n", ret); @@ -648,10 +662,12 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) } dev_info(dev, "revision %c\n", id1 + 'A'); - ret = wm8804_reset(wm8804); - if (ret < 0) { - dev_err(dev, "Failed to issue reset: %d\n", ret); - goto err_reg_enable; + if (!wm8804->reset) { + ret = wm8804_soft_reset(wm8804); + if (ret < 0) { + dev_err(dev, "Failed to issue reset: %d\n", ret); + goto err_reg_enable; + } } ret = snd_soc_register_codec(dev, &soc_codec_dev_wm8804, -- 1.7.2.5

2 5

[alsa-devel] [PATCH v2 1/4] ASoC: wm8804: Add support for hardware reset line
by Charles Keepax 07 Apr '15

07 Apr '15

It is best to use the physical reset if it is available. This patch adds support for a GPIO controlled physical reset for the chip. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Corrected email address on two of the patches Thanks, Charles sound/soc/codecs/wm8804.c | 26 +++++++++++++++++++++----- 1 files changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index f44da83..cc168c4 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -13,6 +13,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> +#include <linux/gpio/consumer.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/of_device.h> @@ -61,6 +62,8 @@ struct wm8804_priv { struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; int mclk_div; + + struct gpio_desc *reset; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -182,7 +185,7 @@ static bool wm8804_volatile(struct device *dev, unsigned int reg) } } -static int wm8804_reset(struct wm8804_priv *wm8804) +static int wm8804_soft_reset(struct wm8804_priv *wm8804) { return regmap_write(wm8804->regmap, WM8804_RST_DEVID1, 0x0); } @@ -586,6 +589,14 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) wm8804->regmap = regmap; + wm8804->reset = devm_gpiod_get_optional(dev, "wlf,reset", + GPIOD_OUT_LOW); + if (IS_ERR(wm8804->reset)) { + ret = PTR_ERR(wm8804->reset); + dev_err(dev, "Failed to get reset line: %d\n", ret); + return ret; + } + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) wm8804->supplies[i].supply = wm8804_supply_names[i]; @@ -620,6 +631,9 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) return ret; } + if (wm8804->reset) + gpiod_set_value_cansleep(wm8804->reset, 1); + ret = regmap_read(regmap, WM8804_RST_DEVID1, &id1); if (ret < 0) { dev_err(dev, "Failed to read device ID: %d\n", ret); @@ -648,10 +662,12 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) } dev_info(dev, "revision %c\n", id1 + 'A'); - ret = wm8804_reset(wm8804); - if (ret < 0) { - dev_err(dev, "Failed to issue reset: %d\n", ret); - goto err_reg_enable; + if (!wm8804->reset) { + ret = wm8804_soft_reset(wm8804); + if (ret < 0) { + dev_err(dev, "Failed to issue reset: %d\n", ret); + goto err_reg_enable; + } } ret = snd_soc_register_codec(dev, &soc_codec_dev_wm8804, -- 1.7.2.5

1 3

[alsa-devel] bug
by Nikita N. 07 Apr '15

07 Apr '15

is any programmer listening here? -- http://www.fastmail.com - A no graphics, no pop-ups email service

10 35

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Fix ruledata setup in davinci_mcasp_startup
by Peter Ujfalusi 07 Apr '15

07 Apr '15

Passing &mcasp->ruledata[dir] to snd_pcm_hw_rule_add() is not correct since commit: 7b3d165a2821 ASoC: davinci-mcasp: Index ruledata in drvdata with substream->stream now sets up the struct based on the substream->stream (0 or 1) while we pass a pointer which we take with dir (1 or 2). This will lead kernel crash. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0b6b1b286201..bb4b78eada58 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1128,6 +1128,8 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + struct davinci_mcasp_ruledata *ruledata = + &mcasp->ruledata[substream->stream]; u32 max_channels = 0; int i, dir; @@ -1149,7 +1151,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->serial_dir[i] == dir) max_channels++; } - mcasp->ruledata[substream->stream].serializers = max_channels; + ruledata->serializers = max_channels; max_channels *= mcasp->tdm_slots; /* * If the already active stream has less channels than the calculated @@ -1172,12 +1174,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { int ret; - mcasp->ruledata[substream->stream].mcasp = mcasp; + ruledata->mcasp = mcasp; ret = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, davinci_mcasp_hw_rule_rate, - &mcasp->ruledata[dir], + ruledata, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (ret) @@ -1185,7 +1187,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, ret = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, davinci_mcasp_hw_rule_format, - &mcasp->ruledata[dir], + ruledata, SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (ret) @@ -1193,7 +1195,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, ret = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, davinci_mcasp_hw_rule_channels, - &mcasp->ruledata[dir], + ruledata, SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_FORMAT, -1); if (ret) -- 2.3.5

2 1

[alsa-devel] [PATCH 1/4] ASoC: wm8804: Add support for hardware reset line
by Charles Keepax 07 Apr '15

07 Apr '15

It is best to use the physical reset if it is available. This patch adds support for a GPIO controlled physical reset for the chip. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8804.c | 26 +++++++++++++++++++++----- 1 files changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index f44da83..cc168c4 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -13,6 +13,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> +#include <linux/gpio/consumer.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/of_device.h> @@ -61,6 +62,8 @@ struct wm8804_priv { struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; int mclk_div; + + struct gpio_desc *reset; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -182,7 +185,7 @@ static bool wm8804_volatile(struct device *dev, unsigned int reg) } } -static int wm8804_reset(struct wm8804_priv *wm8804) +static int wm8804_soft_reset(struct wm8804_priv *wm8804) { return regmap_write(wm8804->regmap, WM8804_RST_DEVID1, 0x0); } @@ -586,6 +589,14 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) wm8804->regmap = regmap; + wm8804->reset = devm_gpiod_get_optional(dev, "wlf,reset", + GPIOD_OUT_LOW); + if (IS_ERR(wm8804->reset)) { + ret = PTR_ERR(wm8804->reset); + dev_err(dev, "Failed to get reset line: %d\n", ret); + return ret; + } + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) wm8804->supplies[i].supply = wm8804_supply_names[i]; @@ -620,6 +631,9 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) return ret; } + if (wm8804->reset) + gpiod_set_value_cansleep(wm8804->reset, 1); + ret = regmap_read(regmap, WM8804_RST_DEVID1, &id1); if (ret < 0) { dev_err(dev, "Failed to read device ID: %d\n", ret); @@ -648,10 +662,12 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) } dev_info(dev, "revision %c\n", id1 + 'A'); - ret = wm8804_reset(wm8804); - if (ret < 0) { - dev_err(dev, "Failed to issue reset: %d\n", ret); - goto err_reg_enable; + if (!wm8804->reset) { + ret = wm8804_soft_reset(wm8804); + if (ret < 0) { + dev_err(dev, "Failed to issue reset: %d\n", ret); + goto err_reg_enable; + } } ret = snd_soc_register_codec(dev, &soc_codec_dev_wm8804, -- 1.7.2.5

2 5

[alsa-devel] [PATCH] ALSA: hda_intel: add AZX_DCAPS_I915_POWERWELL for skl
by libin.yang＠intel.com 07 Apr '15

07 Apr '15

From: Libin Yang <libin.yang(a)intel.com> HDMI/DP codec on SKL is in the power well. The power well must be turned on before probing the HDMI/DP codec. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/hda_intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 15a8299..4744a96 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -298,7 +298,8 @@ enum { AZX_DCAPS_SNOOP_TYPE(SCH)) #define AZX_DCAPS_INTEL_SKYLAKE \ - (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG) + (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\ + AZX_DCAPS_I915_POWERWELL) /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ -- 1.9.1

4 19

[alsa-devel] steinberg ur44 - soundcard not connecting, acl problem (systemd?)
by James Miller 06 Apr '15

06 Apr '15

Hi, I've just bought a steinberg ur44. It is reported as having worked in an earlier kernel to the 3.18 rt custom/3.19 low latency generic that I am using. The low latency kernel is generic but the realtime is compiled with the alsa quirks as described here: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074120.html On plugging it in, the card is recognised by udev but is disconnected after seemingly being set up. The same error seems to be occurring when I start the machine with the card plugged in. There seems to be an ACL error reported in journalctl. I include the appropriate lines from the udevadm and from journalctl udevadm: sudo udevadm monitor monitor will print the received events for: UDEV - the event which udev sends out after rule processing KERNEL - the kernel uevent KERNEL[2741.779546] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5 (usb) KERNEL[2741.779705] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.0 (usb) KERNEL[2741.781720] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1 (usb) KERNEL[2741.787500] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3 (sound) KERNEL[2741.787611] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/controlC3 (sound) KERNEL[2741.787813] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/pcmC3D0p (sound) KERNEL[2741.787933] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/pcmC3D0c (sound) KERNEL[2741.788073] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/midiC3D0 (sound) KERNEL[2741.788189] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/midi3 (sound) KERNEL[2741.789228] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/dmmidi3 (sound) KERNEL[2741.789294] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/midi3 (sound) KERNEL[2741.789315] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/dmmidi3 (sound) KERNEL[2741.789365] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/midiC3D0 (sound) KERNEL[2741.789418] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/pcmC3D0p (sound) KERNEL[2741.789436] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/pcmC3D0c (sound) KERNEL[2741.789487] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/controlC3 (sound) KERNEL[2741.789503] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3 (sound) KERNEL[2741.789556] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.2 (usb) KERNEL[2741.789577] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.3 (usb) KERNEL[2741.790043] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.4 (usb) UDEV [2741.810128] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5 (usb) UDEV [2741.810172] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.3 (usb) UDEV [2741.810195] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.2 (usb) UDEV [2741.810217] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1 (usb) UDEV [2741.810237] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3 (sound) UDEV [2741.810260] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/pcmC3D0c (sound) UDEV [2741.810279] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/pcmC3D0c (sound) UDEV [2741.810301] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.4 (usb) UDEV [2741.815664] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/pcmC3D0p (sound) UDEV [2741.815903] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/pcmC3D0p (sound) UDEV [2741.820639] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/midi3 (sound) UDEV [2741.821530] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/midi3 (sound) UDEV [2741.823891] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/dmmidi3 (sound) UDEV [2741.823931] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/dmmidi3 (sound) UDEV [2741.826596] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/midiC3D0 (sound) UDEV [2741.827398] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/midiC3D0 (sound) UDEV [2741.837736] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/controlC3 (sound) UDEV [2741.839478] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/controlC3 (sound) UDEV [2741.840171] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3 (sound) UDEV [2742.842156] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.0 (usb) journactl (on hotplug add): Apr 06 19:26:09 super-R720 kernel: usb 1-5: USB disconnect, device number 4 Apr 06 19:26:09 super-R720 systemd-udevd[309]: error opening USB device 'descriptors' file Apr 06 19:26:11 super-R720 kernel: usb 1-5: new high-speed USB device number 5 using ehci-pci Apr 06 19:26:12 super-R720 kernel: usb 1-5: New USB device found, idVendor=0499, idProduct=150e Apr 06 19:26:12 super-R720 kernel: usb 1-5: New USB device strings: Mfr=1, Product=2, SerialNumber=0 Apr 06 19:26:12 super-R720 kernel: usb 1-5: Product: Steinberg UR44 Apr 06 19:26:12 super-R720 kernel: usb 1-5: Manufacturer: Yamaha Corporation Apr 06 19:26:12 super-R720 kernel: snd-usb-audio: probe of 1-5:1.1 failed with error -16 Apr 06 19:26:12 super-R720 mtp-probe[3550]: checking bus 1, device 5: "/sys/devices/pci0000:00/0000:00:1a.7/usb1/1-5" Apr 06 19:26:12 super-R720 mtp-probe[3550]: bus: 1, device: 5 was not an MTP device Apr 06 19:26:12 super-R720 systemd-udevd[3547]: error opening ATTR{/sys/devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/controlC3/ Apr 06 19:26:12 super-R720 systemd-udevd[3560]: Failed to apply ACL on /dev/dmmidi3: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3560]: Failed to apply ACL on /dev/dmmidi3: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3559]: Failed to apply ACL on /dev/midi3: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3559]: Failed to apply ACL on /dev/midi3: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3558]: Failed to apply ACL on /dev/snd/midiC3D0: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3558]: Failed to apply ACL on /dev/snd/midiC3D0: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3549]: Failed to apply ACL on /dev/snd/pcmC3D0c: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3549]: Failed to apply ACL on /dev/snd/pcmC3D0c: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3548]: Failed to apply ACL on /dev/snd/pcmC3D0p: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3548]: Failed to apply ACL on /dev/snd/pcmC3D0p: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3547]: Failed to apply ACL on /dev/snd/controlC3: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3547]: Failed to apply ACL on /dev/snd/controlC3: No such file or directory journalctl -b: systemd-udevd[416]: error opening ATTR{/sys/devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/controlC3/. I can compile a kernel reasonably well, and can understand and program in c, c++, but am not well versed in lower level usb hardware coding. I'd be grateful for any help, many thanks! Many thanks -- James Stewart Miller Bsc(hons) Psych.

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[alsa-devel] [asoc:topic/intel 37/37] sound/soc/intel/atom/sst/sst_drv_interface.c:277:19: sparse: cast removes address space of expression
by kbuild test robot 06 Apr '15

06 Apr '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: b97169da06992ef04081e66ed22bbdb23dbf6610 commit: b97169da06992ef04081e66ed22bbdb23dbf6610 [37/37] ASoC: Intel: create atom folder and move atom platform files in reproduce: # apt-get install sparse git checkout b97169da06992ef04081e66ed22bbdb23dbf6610 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/intel/atom/sst/sst_drv_interface.c:277:19: sparse: cast removes address space of expression >> sound/soc/intel/atom/sst/sst_drv_interface.c:278:17: sparse: incorrect type in argument 2 (different address spaces) sound/soc/intel/atom/sst/sst_drv_interface.c:278:17: expected void const volatile [noderef] <asn:2>*src sound/soc/intel/atom/sst/sst_drv_interface.c:278:17: got void * >> sound/soc/intel/atom/sst/sst_drv_interface.c:285:19: sparse: cast removes address space of expression >> sound/soc/intel/atom/sst/sst_drv_interface.c:285:14: sparse: incorrect type in assignment (different address spaces) sound/soc/intel/atom/sst/sst_drv_interface.c:285:14: expected void [noderef] <asn:2>*addr sound/soc/intel/atom/sst/sst_drv_interface.c:285:14: got void * >> sound/soc/intel/atom/sst/sst_drv_interface.c:372:19: sparse: cast removes address space of expression >> sound/soc/intel/atom/sst/sst_drv_interface.c:373:17: sparse: incorrect type in argument 2 (different address spaces) sound/soc/intel/atom/sst/sst_drv_interface.c:373:17: expected void const volatile [noderef] <asn:2>*src sound/soc/intel/atom/sst/sst_drv_interface.c:373:17: got void * >> sound/soc/intel/atom/sst/sst_drv_interface.c:560:19: sparse: cast removes address space of expression >> sound/soc/intel/atom/sst/sst_drv_interface.c:561:25: sparse: incorrect type in argument 2 (different address spaces) sound/soc/intel/atom/sst/sst_drv_interface.c:561:25: expected void const volatile [noderef] <asn:2>*src sound/soc/intel/atom/sst/sst_drv_interface.c:561:25: got void * -- >> sound/soc/intel/atom/sst/sst_loader.c:55:26: sparse: incorrect type in argument 1 (different address spaces) sound/soc/intel/atom/sst/sst_loader.c:55:26: expected void [noderef] <asn:2>*to sound/soc/intel/atom/sst/sst_loader.c:55:26: got void *dst >> sound/soc/intel/atom/sst/sst_loader.c:55:31: sparse: incorrect type in argument 2 (different address spaces) sound/soc/intel/atom/sst/sst_loader.c:55:31: expected void const *from sound/soc/intel/atom/sst/sst_loader.c:55:31: got void const [noderef] <asn:2>*src >> sound/soc/intel/atom/sst/sst_loader.c:220:43: sparse: incorrect type in argument 2 (different address spaces) sound/soc/intel/atom/sst/sst_loader.c:220:43: expected void *destn sound/soc/intel/atom/sst/sst_loader.c:220:43: got void [noderef] <asn:2>* -- >> sound/soc/intel/atom/sst/sst_pvt.c:114:20: sparse: symbol 'read_shim_data' was not declared. Should it be static? >> sound/soc/intel/atom/sst/sst_pvt.c:127:6: sparse: symbol 'write_shim_data' was not declared. Should it be static? Please review and possibly fold the followup patch. vim +277 sound/soc/intel/atom/sst/sst_drv_interface.c 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 271 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 272 /* update bytes sent */ 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 273 stream->cumm_bytes += bytes; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 274 dev_dbg(dev, "bytes copied %d inc by %ld\n", stream->cumm_bytes, bytes); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 275 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 276 memcpy_fromio(&fw_tstamp, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 @277 ((void *)(ctx->mailbox + ctx->tstamp) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 @278 +(str_id * sizeof(fw_tstamp))), 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 279 sizeof(fw_tstamp)); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 280 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 281 fw_tstamp.bytes_copied = stream->cumm_bytes; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 282 dev_dbg(dev, "bytes sent to fw %llu inc by %ld\n", 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 283 fw_tstamp.bytes_copied, bytes); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 284 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 @285 addr = ((void *)(ctx->mailbox + ctx->tstamp)) + 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 286 (str_id * sizeof(fw_tstamp)); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 287 offset = offsetof(struct snd_sst_tstamp, bytes_copied); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 288 sst_shim_write(addr, offset, fw_tstamp.bytes_copied); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 289 return 0; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 290 } 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 291 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 292 static int sst_cdev_set_metadata(struct device *dev, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 293 unsigned int str_id, struct snd_compr_metadata *metadata) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 294 { 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 295 int retval = 0; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 296 struct stream_info *str_info; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 297 struct intel_sst_drv *ctx = dev_get_drvdata(dev); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 298 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 299 dev_dbg(dev, "set metadata for stream %d\n", str_id); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 300 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 301 str_info = get_stream_info(ctx, str_id); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 302 if (!str_info) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 303 return -EINVAL; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 304 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 305 dev_dbg(dev, "pipe id = %d\n", str_info->pipe_id); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 306 retval = sst_prepare_and_post_msg(ctx, str_info->task_id, IPC_CMD, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 307 IPC_IA_SET_STREAM_PARAMS_MRFLD, str_info->pipe_id, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 308 sizeof(*metadata), metadata, NULL, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 309 true, true, true, false); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 310 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 311 return retval; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 312 } 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 313 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 314 static int sst_cdev_stream_pause(struct device *dev, unsigned int str_id) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 315 { 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 316 struct intel_sst_drv *ctx = dev_get_drvdata(dev); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 317 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 318 return sst_pause_stream(ctx, str_id); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 319 } 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 320 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 321 static int sst_cdev_stream_pause_release(struct device *dev, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 322 unsigned int str_id) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 323 { 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 324 struct intel_sst_drv *ctx = dev_get_drvdata(dev); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 325 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 326 return sst_resume_stream(ctx, str_id); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 327 } 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 328 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 329 static int sst_cdev_stream_start(struct device *dev, unsigned int str_id) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 330 { 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 331 struct stream_info *str_info; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 332 struct intel_sst_drv *ctx = dev_get_drvdata(dev); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 333 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 334 str_info = get_stream_info(ctx, str_id); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 335 if (!str_info) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 336 return -EINVAL; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 337 str_info->prev = str_info->status; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 338 str_info->status = STREAM_RUNNING; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 339 return sst_start_stream(ctx, str_id); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 340 } 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 341 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 342 static int sst_cdev_stream_drop(struct device *dev, unsigned int str_id) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 343 { 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 344 struct intel_sst_drv *ctx = dev_get_drvdata(dev); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 345 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 346 return sst_drop_stream(ctx, str_id); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 347 } 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 348 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 349 static int sst_cdev_stream_drain(struct device *dev, unsigned int str_id) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 350 { 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 351 struct intel_sst_drv *ctx = dev_get_drvdata(dev); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 352 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 353 return sst_drain_stream(ctx, str_id, false); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 354 } 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 355 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 356 static int sst_cdev_stream_partial_drain(struct device *dev, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 357 unsigned int str_id) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 358 { 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 359 struct intel_sst_drv *ctx = dev_get_drvdata(dev); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 360 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 361 return sst_drain_stream(ctx, str_id, true); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 362 } 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 363 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 364 static int sst_cdev_tstamp(struct device *dev, unsigned int str_id, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 365 struct snd_compr_tstamp *tstamp) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 366 { 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 367 struct snd_sst_tstamp fw_tstamp = {0,}; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 368 struct stream_info *stream; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 369 struct intel_sst_drv *ctx = dev_get_drvdata(dev); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 370 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 371 memcpy_fromio(&fw_tstamp, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 @372 ((void *)(ctx->mailbox + ctx->tstamp) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 @373 +(str_id * sizeof(fw_tstamp))), 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 374 sizeof(fw_tstamp)); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 375 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 376 stream = get_stream_info(ctx, str_id); :::::: The code at line 277 was first introduced by commit :::::: 7adab122a57c5ade8efc2e4de67c72b084c31cda ASoC: Intel: sst - add compressed ops handling :::::: TO: Vinod Koul <vinod.koul(a)intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

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[alsa-devel] [PATCH v3 0/5] Intel SST driver restructure
by Jie Yang 06 Apr '15

06 Apr '15

In order to prepare for the upstreaming of Skylake drivers and to allow reuse of code between different Intel audio DSP products, this patch series restructures the Intel directory as follows: 1. Create "common" directory. Move common files into this directory. 2. Create "boards" directory. Move all machine drivers into this directory. 3. Create "haswell" directory. Move all remaining hsw/bdw code into this directory. 4. Create "baytrail" directory. Move remaining baytrail driver to this directory. 5. Create "atom" directory. Move all MCG platform drivers(including sst/ folder) to this directory. Changes in v3: Fix ATT43256.config 0day build issue. Changes in v2: Add "Reviewed-by" and "Acked-by" items from each platforms' maintainer. Todo next step: 1. Change and move atom related codes(in atom/sst/) to corresponding and try share as many as possible codes(e.g. sst_acpi.c). 2. Share more common codes about IPC for different platforms, e.g. The methods for queueing and sending/reciving Tx/Rx messages, The methods for mailbox IO, The IPC error return codes, etc. Jie Yang (5): ASoC: Intel: create common folder and move common files in ASoC: Intel: create haswell folder and move haswell platform files in ASoC: Intel: create boards folder and move sst boards files in ASoC: Intel: create baytrail folder and move baytrail platform files in ASoC: Intel: create atom folder and move atom platform files in sound/soc/intel/Makefile | 42 +++------------------- sound/soc/intel/atom/Makefile | 7 ++++ sound/soc/intel/{ => atom}/sst-atom-controls.c | 0 sound/soc/intel/{ => atom}/sst-atom-controls.h | 0 sound/soc/intel/{ => atom}/sst-mfld-dsp.h | 0 .../intel/{ => atom}/sst-mfld-platform-compress.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform.h | 0 sound/soc/intel/{ => atom}/sst/Makefile | 0 sound/soc/intel/{ => atom}/sst/sst.c | 2 +- sound/soc/intel/{ => atom}/sst/sst.h | 0 sound/soc/intel/{ => atom}/sst/sst_acpi.c | 2 +- sound/soc/intel/{ => atom}/sst/sst_drv_interface.c | 2 +- sound/soc/intel/{ => atom}/sst/sst_ipc.c | 2 +- sound/soc/intel/{ => atom}/sst/sst_loader.c | 2 +- sound/soc/intel/{ => atom}/sst/sst_pci.c | 0 sound/soc/intel/{ => atom}/sst/sst_pvt.c | 2 +- sound/soc/intel/{ => atom}/sst/sst_stream.c | 2 +- sound/soc/intel/baytrail/Makefile | 4 +++ sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c | 4 +-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c | 4 +-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h | 0 sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c | 4 +-- sound/soc/intel/boards/Makefile | 15 ++++++++ sound/soc/intel/{ => boards}/broadwell.c | 6 ++-- sound/soc/intel/{ => boards}/byt-max98090.c | 2 +- sound/soc/intel/{ => boards}/byt-rt5640.c | 4 +-- .../{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} | 4 +-- sound/soc/intel/{ => boards}/cht_bsw_rt5645.c | 4 +-- sound/soc/intel/{ => boards}/cht_bsw_rt5672.c | 4 +-- sound/soc/intel/{ => boards}/haswell.c | 6 ++-- sound/soc/intel/{ => boards}/mfld_machine.c | 0 sound/soc/intel/common/Makefile | 6 ++++ sound/soc/intel/{ => common}/sst-acpi.c | 0 sound/soc/intel/{ => common}/sst-dsp-priv.h | 0 sound/soc/intel/{ => common}/sst-dsp.c | 0 sound/soc/intel/{ => common}/sst-dsp.h | 0 sound/soc/intel/{ => common}/sst-firmware.c | 0 sound/soc/intel/haswell/Makefile | 4 +++ sound/soc/intel/{ => haswell}/sst-haswell-dsp.c | 6 ++-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.c | 4 +-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.h | 0 sound/soc/intel/{ => haswell}/sst-haswell-pcm.c | 6 ++-- 43 files changed, 77 insertions(+), 73 deletions(-) create mode 100644 sound/soc/intel/atom/Makefile rename sound/soc/intel/{ => atom}/sst-atom-controls.c (100%) rename sound/soc/intel/{ => atom}/sst-atom-controls.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-dsp.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-compress.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform.h (100%) rename sound/soc/intel/{ => atom}/sst/Makefile (100%) rename sound/soc/intel/{ => atom}/sst/sst.c (99%) rename sound/soc/intel/{ => atom}/sst/sst.h (100%) rename sound/soc/intel/{ => atom}/sst/sst_acpi.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_drv_interface.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_ipc.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_loader.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_pci.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_pvt.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_stream.c (99%) create mode 100644 sound/soc/intel/baytrail/Makefile rename sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h (100%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c (99%) create mode 100644 sound/soc/intel/boards/Makefile rename sound/soc/intel/{ => boards}/broadwell.c (98%) rename sound/soc/intel/{ => boards}/byt-max98090.c (99%) rename sound/soc/intel/{ => boards}/byt-rt5640.c (98%) rename sound/soc/intel/{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} (98%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5645.c (99%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5672.c (99%) rename sound/soc/intel/{ => boards}/haswell.c (98%) rename sound/soc/intel/{ => boards}/mfld_machine.c (100%) create mode 100644 sound/soc/intel/common/Makefile rename sound/soc/intel/{ => common}/sst-acpi.c (100%) rename sound/soc/intel/{ => common}/sst-dsp-priv.h (100%) rename sound/soc/intel/{ => common}/sst-dsp.c (100%) rename sound/soc/intel/{ => common}/sst-dsp.h (100%) rename sound/soc/intel/{ => common}/sst-firmware.c (100%) create mode 100644 sound/soc/intel/haswell/Makefile rename sound/soc/intel/{ => haswell}/sst-haswell-dsp.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.h (100%) rename sound/soc/intel/{ => haswell}/sst-haswell-pcm.c (99%) -- 1.9.1

2 6

[alsa-devel] [PATCH] ASoC: wm5102: Remove set of volume update bits for output 3R
by Charles Keepax 06 Apr '15

06 Apr '15

The earpiece on wm5102 is mono, thus there is no output 3R. Don't toggle the volume update bits for this output, although worth noting that doing so had no negative effects it is just redundant. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 6d0fe0a..0c6d1bc 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1861,7 +1861,6 @@ static unsigned int wm5102_digital_vu[] = { ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_DAC_DIGITAL_VOLUME_3L, - ARIZONA_DAC_DIGITAL_VOLUME_3R, ARIZONA_DAC_DIGITAL_VOLUME_4L, ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_DAC_DIGITAL_VOLUME_5L, -- 1.7.2.5

2 1

[alsa-devel] [PATCH/alsa-lib] amixer: fix 'cset' operation to set several channels
by Takashi Sakamoto 06 Apr '15

06 Apr '15

For an enumerated element with several channels, when given comma-separated string values, 'cset' operation sets these values just to the first channel, instead of corresponding channels. This commit adds missing-increment to fix this bug. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- amixer/amixer.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index 36c92eb..e9fe735 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -1290,8 +1290,10 @@ static int sset_enum(snd_mixer_elem_t *elem, unsigned int argc, char **argv) if (snd_mixer_selem_set_enum_item(elem, item++, ival) >= 0) check_flag = 1; /* skip separators */ - while (*ptr == ',' || isspace(*ptr)) + while (*ptr == ',' || isspace(*ptr)) { ptr++; + chn++; + } } } return check_flag; -- 2.1.0

2 3

[alsa-devel] [PATCH] ALSA: emu10k1: add Audigy 5/Rx
by Michael Gernoth 05 Apr '15

05 Apr '15

The Audigy 5/Rx is essentially an Audigy 4 behind a PLX PCIe- bridge with an additional TOSLINK output. Signed-off-by: Michael Gernoth <michael(a)gernoth.net> --- sound/pci/emu10k1/emu10k1_main.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index b4458a6..8e510e9 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1325,6 +1325,22 @@ static int snd_emu10k1_dev_free(struct snd_device *device) } static struct snd_emu_chip_details emu_chip_details[] = { + /* Audigy 5/Rx SB1550 */ + /* Tested by michael(a)gernoth.net 28 Mar 2015 */ + /* DSP: CA10300-IAT LF + * DAC: Cirrus Logic CS4382-KQZ + * ADC: Philips 1361T + * AC97: Sigmatel STAC9750 + * CA0151: None + */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10241102, + .driver = "Audigy2", .name = "SB Audigy 5/Rx [SB1550]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .adc_1361t = 1, /* 24 bit capture instead of 16bit */ + .ac97_chip = 1} , /* Audigy4 (Not PRO) SB0610 */ /* Tested by James(a)superbug.co.uk 4th April 2006 */ /* A_IOCFG bits -- 2.1.4

2 1

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

05 Apr '15

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

2 8

[alsa-devel] [PATCH 0/16] fix error return code
by Julia Lawall 05 Apr '15

05 Apr '15

The complate semantic patch that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @ok exists@ identifier f,ret,i; expression e; constant c; @@ // identify a function that returns a negative return value at least once. f(...) { ... when any ( return -c@i; | ret = -c@i; ... when != ret = e return ret; | if (ret < 0) { ... return ret; } ) ... when any } @r exists@ identifier ret,ok.f,fn; expression e1,e2,e3,e4,e5,e6,x; statement S,S1; position p1,p2,p3; @@ // identify a case where the return variable is set to a non-negative value // and then returned in error-handling code f(...) { ... when any ( if@p1 ($ret < 0\|ret != 0$) { ... return ret; } | ret@p1 = 0 ) ... when != $ret = e1\|ret++\|ret--\|ret+=e1\|ret-=e1$ when != &ret when any ( if (<+... ret = e5 ...+>) S1 | if (<+... &ret ...+>) S1 | if@p2(<+...x = fn(...)...+>) { ... when != ret = e6 when forall return@p3 ret; } | break; | x = fn(...) ... when != $ret = e4\|ret++\|ret--\|ret+=e4\|ret-=e4$ when != &ret ( if (<+... ret = e3 ...+>) S | if (<+... &ret ...+>) S | if@p2(<+...$x != 0\|x < 0\|x == NULL\|IS_ERR(x)$...+>) { ... when != ret = e2 when forall return@p3 ret; } ) ) ... when any } @printer depends on r@ position p; identifier ok.f,pr; constant char [] c; @@ f(...) { <...pr@p(...,c,...)...> } @bad0 exists@ identifier r.ret,ok.f,g != {ERR_PTR,IS_ERR}; position p != printer.p; @@ f(...) { ... when any g@p(...,ret,...) ... when any } @bad depends on !bad0 exists@ position r.p1,r.p2; statement S1,S2; identifier r.ret; expression e1; @@ // ignore the above if there is some path where the variable is set to // something else ( if@p1 ($ret < 0\|ret != 0$) S1 | ret@p1 = 0 ) ... when any $ret = e1\|ret++\|ret--\|ret+=e1\|ret-=e1\|&ret$ ... when any if@p2(...) S2 @bad1 depends on !bad0 && !bad exists@ position r.p2; statement S2; identifier r.ret; expression e1; constant c; @@ ret = -c ... when != $ret = e1\|ret++\|ret--\|ret+=e1\|ret-=e1$ when != &ret when any if@p2(...) S2 @bad2 depends on !bad0 && !bad && !bad1 exists@ position r.p1,r.p2; identifier r.ret; expression e1; statement S2; constant c; @@ // likewise ignore it if there has been an intervening return ret@p1 = 0 ... when != if (...) { ... ret = e1 ... return ret; } when != if (...) { ... return -c; } when any if@p2(...) S2 @script:python depends on !bad0 && !bad && !bad1 && !bad2@ p1 << r.p1; p2 << r.p2; p3 << r.p3; @@ cocci.print_main("",p1) cocci.print_secs("",p2) cocci.print_secs("",p3) // </smpl>

2 2

Re: [alsa-devel] [PATCH] ALSA: hda/realtek - Support for microphone in Dell's XPS 13 9343 with ALC288.
by Takashi Iwai 05 Apr '15

05 Apr '15

At Thu, 2 Apr 2015 12:33:40 +0100, me(a)jdiez.me wrote: > > From: Jose Diez <me(a)jdiez.me> > > Original patch by Kailang Yang did not compile on 4.0-rc5. It was > referencing a member field `core` in `struct hda_codec` in order to > access the `vendor_id`. This has always been in the codec structure > itself. > > This patch fixes the build and is verified to work properly on 4.0-rc5, > running actual hardware. > > Signed-off-by: Jose Diez <me(a)jdiez.me> I prefer splitting the Dell-specific part as its own quirk, as mentioned in the review of Kailang's patch. Then it'll be applicable to both 4.1 and older kernels. thanks, Takashi > --- > sound/pci/hda/patch_realtek.c | 122 ++++++++++++++++++++++++++++++++++++++++++ > 1 file changed, 122 insertions(+) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 7438213..a8fc2c8 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -3583,6 +3583,14 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) > WRITE_COEF(0x32, 0x42a3), > {} > }; > + static struct coef_fw coef0286[] = { > + UPDATE_COEF(0x4f, 0xfcc0, 0xc400), > + UPDATE_COEF(0x50, 0x2000, 0x2000), > + UPDATE_COEF(0x56, 0x0006, 0x0006), > + UPDATE_COEF(0x66, 0x0008, 0), > + UPDATE_COEF(0x67, 0x2000, 0), > + {} > + }; > static struct coef_fw coef0292[] = { > WRITE_COEF(0x76, 0x000e), > WRITE_COEF(0x6c, 0x2400), > @@ -3613,6 +3621,10 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) > case 0x10ec0283: > alc_process_coef_fw(codec, coef0233); > break; > + case 0x10ec0286: > + case 0x10ec0288: > + alc_process_coef_fw(codec, coef0286); > + break; > case 0x10ec0292: > alc_process_coef_fw(codec, coef0292); > break; > @@ -3642,6 +3654,14 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, > WRITE_COEF(0x26, 0x008c), > {} > }; > + static struct coef_fw coef0286[] = { > + UPDATE_COEF(0x50, 0x2000, 0), > + UPDATE_COEF(0x56, 0x0006, 0), > + UPDATE_COEF(0x4f, 0xfcc0, 0xc400), > + UPDATE_COEF(0x66, 0x0008, 0x0008), > + UPDATE_COEF(0x67, 0x2000, 0x2000), > + {} > + }; > static struct coef_fw coef0292[] = { > WRITE_COEF(0x19, 0xa208), > WRITE_COEF(0x2e, 0xacf0), > @@ -3674,6 +3694,13 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, > alc_process_coef_fw(codec, coef0233); > snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); > break; > + case 0x10ec0286: > + case 0x10ec0288: > + alc_update_coef_idx(codec, 0x4f, 0x000c, 0); > + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); > + alc_process_coef_fw(codec, coef0286); > + 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_process_coef_fw(codec, coef0292); > @@ -3709,6 +3736,14 @@ static void alc_headset_mode_default(struct hda_codec *codec) > WRITE_COEF(0x32, 0x4ea3), > {} > }; > + static struct coef_fw coef0286[] = { > + UPDATE_COEF(0x4f, 0xfcc0, 0xc400), /* Set to TRS type */ > + UPDATE_COEF(0x50, 0x2000, 0x2000), > + UPDATE_COEF(0x56, 0x0006, 0x0006), > + UPDATE_COEF(0x66, 0x0008, 0), > + UPDATE_COEF(0x67, 0x2000, 0), > + {} > + }; > static struct coef_fw coef0292[] = { > WRITE_COEF(0x76, 0x000e), > WRITE_COEF(0x6c, 0x2400), > @@ -3737,6 +3772,11 @@ static void alc_headset_mode_default(struct hda_codec *codec) > case 0x10ec0283: > alc_process_coef_fw(codec, coef0233); > break; > + case 0x10ec0286: > + case 0x10ec0288: > + alc_process_coef_fw(codec, coef0286); > + break; > + break; > case 0x10ec0292: > alc_process_coef_fw(codec, coef0292); > break; > @@ -3765,6 +3805,13 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) > WRITE_COEF(0x32, 0x4ea3), > {} > }; > + static struct coef_fw coef0286[] = { > + UPDATE_COEF(0x50, 0x2000, 0x2000), > + UPDATE_COEF(0x56, 0x0006, 0x0006), > + UPDATE_COEF(0x66, 0x0008, 0), > + UPDATE_COEF(0x67, 0x2000, 0), > + {} > + }; > static struct coef_fw coef0292[] = { > WRITE_COEF(0x6b, 0xd429), > WRITE_COEF(0x76, 0x0008), > @@ -3791,6 +3838,12 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) > case 0x10ec0283: > alc_process_coef_fw(codec, coef0233); > break; > + case 0x10ec0286: > + case 0x10ec0288: > + alc_update_coef_idx(codec, 0x4f, 0xfcc0, 0xd400); > + msleep(300); > + alc_process_coef_fw(codec, coef0286); > + break; > case 0x10ec0292: > alc_process_coef_fw(codec, coef0292); > break; > @@ -3819,6 +3872,13 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) > WRITE_COEF(0x32, 0x4ea3), > {} > }; > + static struct coef_fw coef0286[] = { > + UPDATE_COEF(0x50, 0x2000, 0x2000), > + UPDATE_COEF(0x56, 0x0006, 0x0006), > + UPDATE_COEF(0x66, 0x0008, 0), > + UPDATE_COEF(0x67, 0x2000, 0), > + {} > + }; > static struct coef_fw coef0292[] = { > WRITE_COEF(0x6b, 0xe429), > WRITE_COEF(0x76, 0x0008), > @@ -3845,6 +3905,12 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) > case 0x10ec0283: > alc_process_coef_fw(codec, coef0233); > break; > + case 0x10ec0286: > + case 0x10ec0288: > + alc_update_coef_idx(codec, 0x4f, 0xfcc0, 0xe400); > + msleep(300); > + alc_process_coef_fw(codec, coef0286); > + break; > case 0x10ec0292: > alc_process_coef_fw(codec, coef0292); > break; > @@ -3869,6 +3935,11 @@ static void alc_determine_headset_type(struct hda_codec *codec) > conteol) */ > {} > }; > + static struct coef_fw coef0286[] = { > + //UPDATE_COEF(0x4f, 0xfcc0, 0x5400), /* Combo Jack auto detect */ > + UPDATE_COEF(0x4f, 0xfcc0, 0xd400), > + {} > + }; > static struct coef_fw coef0293[] = { > UPDATE_COEF(0x4a, 0x000f, 0x0008), /* Combo Jack auto detect */ > WRITE_COEF(0x45, 0xD429), /* Set to ctia type */ > @@ -3896,6 +3967,13 @@ static void alc_determine_headset_type(struct hda_codec *codec) > val = alc_read_coef_idx(codec, 0x46); > is_ctia = (val & 0x0070) == 0x0070; > break; > + case 0x10ec0286: > + case 0x10ec0288: > + alc_process_coef_fw(codec, coef0286); > + msleep(350); > + val = alc_read_coef_idx(codec, 0x50); > + is_ctia = (val & 0x0070) == 0x0070; > + break; > case 0x10ec0292: > alc_write_coef_idx(codec, 0x6b, 0xd429); > msleep(300); > @@ -3990,8 +4068,15 @@ static void alc_update_headset_jack_cb(struct hda_codec *codec, > struct hda_jack_callback *jack) > { > struct alc_spec *spec = codec->spec; > + int present; > spec->current_headset_type = ALC_HEADSET_TYPE_UNKNOWN; > snd_hda_gen_hp_automute(codec, jack); > + present = spec->gen.hp_jack_present ? 0x40 : 0; > + if (codec->bus->pci->subsystem_vendor == 0x1028 && > + codec->vendor_id == 0x10ec0288) > + /* Headset Mic enable and disable, ony for Dell Dino */ > + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, > + present); > } > > static void alc_probe_headset_mode(struct hda_codec *codec) > @@ -4418,6 +4503,8 @@ enum { > ALC286_FIXUP_HP_GPIO_LED, > ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, > ALC280_FIXUP_HP_DOCK_PINS, > + ALC286_FIXUP_DELL1_MIC_NO_PRESENCE, > + ALC286_FIXUP_DELL_XPS_13_GPIO6, > }; > > static const struct hda_fixup alc269_fixups[] = { > @@ -4906,6 +4993,27 @@ static const struct hda_fixup alc269_fixups[] = { > .chained = true, > .chain_id = ALC280_FIXUP_HP_GPIO4 > }, > + [ALC286_FIXUP_DELL1_MIC_NO_PRESENCE] = { > + .type = HDA_FIXUP_PINS, > + .v.pins = (const struct hda_pintbl[]) { > + { 0x18, 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 > + }, > + [ALC286_FIXUP_DELL_XPS_13_GPIO6] = { > + .type = HDA_FIXUP_VERBS, > + .v.verbs = (const struct hda_verb[]) { > + {0x01, AC_VERB_SET_GPIO_MASK, 0x40}, > + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x40}, > + {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, > + { } > + }, > + .chained = true, > + .chain_id = ALC286_FIXUP_DELL1_MIC_NO_PRESENCE > + }, > }; > > static const struct snd_pci_quirk alc269_fixup_tbl[] = { > @@ -5132,6 +5240,13 @@ static const struct hda_model_fixup alc269_fixup_models[] = { > {0x1b, 0x411111f0}, \ > {0x1e, 0x411111f0} > > +#define ALC286_STANDARD_PINS \ > + {0x17, 0x411111f0}, \ > + {0x18, 0x411111f0}, \ > + {0x19, 0x411111f0}, \ > + {0x1a, 0x411111f0}, \ > + {0x1e, 0x411111f0} > + > #define ALC290_STANDARD_PINS \ > {0x12, 0x99a30130}, \ > {0x13, 0x40000000}, \ > @@ -5317,6 +5432,13 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { > {0x19, 0x03a11020}, > {0x1d, 0x40e00001}, > {0x21, 0x0321101f}), > + SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC286_FIXUP_DELL_XPS_13_GPIO6, > + ALC286_STANDARD_PINS, > + {0x12, 0x90a60120}, > + {0x13, 0x40000000}, > + {0x14, 0x90170110}, > + {0x1d, 0x4076832d}, > + {0x21, 0x0321101f}), > SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, > ALC290_STANDARD_PINS, > {0x14, 0x411111f0}, > -- > 2.3.5 >

1 0

[alsa-devel] [PATCH 0/10] fix error return code
by Julia Lawall 05 Apr '15

05 Apr '15

These patches fix cases where the return code appears to be unintentially nonnegative. The complete semantic match that finds the problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @ok exists@ identifier f,ret,i; expression e; constant c; @@ f(...) { <+... ( return -c@i; | ret = -c@i; ... when != ret = e return ret; | if (ret < 0) { ... return ret; } ) ...+> } @r exists@ identifier ret,l,ok.f; expression e1,e2,e3; statement S; position p1,p2,p3; @@ f(...) { ... when any ( if@p1 ($ret < 0\|ret != 0$) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret ( if (<+... ret = e3 ...+>) S | if (<+... &ret ...+>) S | if@p2(...) { ... when != ret = e2 when forall return@p3 ret; } ) ... when any } @bad exists@ position r.p1,r.p2; statement S1,S2; identifier r.ret; expression e1; @@ ( if@p1 ($ret < 0\|ret != 0$) S1 | ret@p1 = 0 ) ... when any ret = e1 ... when any if@p2(...) S2 @bad2@ position r.p1,r.p2; identifier r.ret; expression e1; statement S2; @@ ret@p1 = 0 ... when != if (...) { ... ret = e1 ... return ret; } when any if@p2(...) S2 @script:python depends on !bad && !bad2@ p1 << r.p1; p2 << r.p2; p3 << r.p3; @@ cocci.print_main("",p1) cocci.print_secs("",p2) cocci.print_secs("",p3) // </smpl>

2 20

[alsa-devel] [PATCH] sound: Use const struct ac97_quirk
by Joe Perches 05 Apr '15

05 Apr '15

Use const to reduce data by ~3Kb. Signed-off-by: Joe Perches <joe(a)perches.com> --- include/sound/ac97_codec.h | 4 +++- sound/pci/ac97/ac97_codec.c | 3 ++- sound/pci/ad1889.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/cs5535audio/cs5535audio.c | 2 +- sound/pci/intel8x0.c | 2 +- sound/pci/via82xx.c | 2 +- 7 files changed, 10 insertions(+), 7 deletions(-) diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index d315a08..0e9d75b 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -608,7 +608,9 @@ struct ac97_quirk { int type; /* quirk type above */ }; -int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, const char *override); +int snd_ac97_tune_hardware(struct snd_ac97 *ac97, + const struct ac97_quirk *quirk, + const char *override); int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate); /* diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 5bca1a3..82259ca 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2902,7 +2902,8 @@ static int apply_quirk_str(struct snd_ac97 *ac97, const char *typestr) * Return: Zero if successful, or a negative error code on failure. */ -int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, const char *override) +int snd_ac97_tune_hardware(struct snd_ac97 *ac97, + const struct ac97_quirk *quirk, const char *override) { int result; diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 850a8c9..66ddd98 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -747,7 +747,7 @@ snd_ad1889_proc_init(struct snd_ad1889 *chip) snd_info_set_text_ops(entry, chip, snd_ad1889_proc_read); } -static struct ac97_quirk ac97_quirks[] = { +static const struct ac97_quirk ac97_quirks[] = { { .subvendor = 0x11d4, /* AD */ .subdevice = 0x1889, /* AD1889 */ diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index d5f15c9..42a20c8 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1390,7 +1390,7 @@ static irqreturn_t snd_atiixp_interrupt(int irq, void *dev_id) * ac97 mixer section */ -static struct ac97_quirk ac97_quirks[] = { +static const struct ac97_quirk ac97_quirks[] = { { .subvendor = 0x103c, .subdevice = 0x006b, diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 802c33f..963b912 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -43,7 +43,7 @@ static char *ac97_quirk; module_param(ac97_quirk, charp, 0444); MODULE_PARM_DESC(ac97_quirk, "AC'97 board specific workarounds."); -static struct ac97_quirk ac97_quirks[] = { +static const struct ac97_quirk ac97_quirks[] = { #if 0 /* Not yet confirmed if all 5536 boards are HP only */ { .subvendor = PCI_VENDOR_ID_AMD, diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 2c5484e..749069a 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1795,7 +1795,7 @@ static struct ac97_pcm ac97_pcm_defs[] = { }, }; -static struct ac97_quirk ac97_quirks[] = { +static const struct ac97_quirk ac97_quirks[] = { { .subvendor = 0x0e11, .subdevice = 0x000e, diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 8622283..3dd038b 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1812,7 +1812,7 @@ static void snd_via82xx_mixer_free_ac97(struct snd_ac97 *ac97) chip->ac97 = NULL; } -static struct ac97_quirk ac97_quirks[] = { +static const struct ac97_quirk ac97_quirks[] = { { .subvendor = 0x1106, .subdevice = 0x4161,

2 1

[alsa-devel] linux-next - strange audio bug
by Valdis Kletnieks 04 Apr '15

04 Apr '15

Next-20150310 works OK, next-20150324 has the bug. Dell Latitude E6530 laptop, lspci tells me: 01:00.1 Audio device: NVIDIA Corporation GF108 High Definition Audio Controller (rev a1) Subsystem: Dell Device 0535 ... Kernel driver in use: snd_hda_intel Symptoms: Audio works perfectly at boot, and for some amount of time after. But somewhere around 6-8 hours of uptime, it gets into a strange state where I'll get no audio... and then it will cut in for all of 2 seconds or so (fairly consistent, that part), and then cut back out for anywhere from 10 seconds to a minute, then another 2 seconds of sound.. lather rinse repeat. Before I start the thankless task of bisecting this (which means I'll be able to do only 1, *maybe* 2 steps a day), does this ring any bells?

4 3

[alsa-devel] [PATCH] ALSA: hda: fix possible null dereference
by Sudip Mukherjee 04 Apr '15

04 Apr '15

we are dereferencing pcm first then checking pcm. instead now lets put them in same if condition so that pcm is checked first. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/hda/hda_codec.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 145cae7..7df18e2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3717,9 +3717,8 @@ static int add_std_chmaps(struct hda_codec *codec) struct snd_pcm_chmap *chmap; const struct snd_pcm_chmap_elem *elem; - if (pcm->own_chmap) - continue; - if (!pcm || !hinfo->substreams) + if (!pcm || pcm->own_chmap || + !hinfo->substreams) continue; elem = hinfo->chmap ? hinfo->chmap : snd_pcm_std_chmaps; err = snd_pcm_add_chmap_ctls(pcm->pcm, str, elem, -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix subsystem ID read regression
by David Henningsson 04 Apr '15

04 Apr '15

A regression was introduced in 7639a06c23c7d4cda3: if AC_PAR_SUBSYSTEM_ID reads as zero, one should retry using AC_VERB_GET_SUBSYSTEM_ID. This seems to hit many codecs (my own laptop included), and causes quirks for some machines not to apply correctly. Reported-by: TienFu Chen <tienfu.chen(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/hda/hdac_device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 6e8ee1d..53b6b95 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -91,7 +91,7 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, codec->power_caps = snd_hdac_read_parm(codec, fg, AC_PAR_POWER_STATE); /* reread ssid if not set by parameter */ - if (codec->subsystem_id == -1) + if (codec->subsystem_id == -1 || codec->subsystem_id == 0) snd_hdac_read(codec, fg, AC_VERB_GET_SUBSYSTEM_ID, 0, &codec->subsystem_id); -- 1.9.1

2 1

Re: [alsa-devel] [RFC] ALSA: usb-audio: reduce "cannot get freq at ep" spew
by Takashi Iwai 04 Apr '15

04 Apr '15

At Tue, 31 Mar 2015 07:36:28 +0000, Eric Wong wrote: > > Eric Wong <normalperson(a)yhbt.net> wrote: > > I also had another generic patch prepared before I noticed Joe's > > quirk addition for the MS Lifecam HD-5000. > > This is the generic patch I prepared before I noticed Joe's quirk > addition (applies cleanly to 3.19 and 4.0-rc6): > > ------------------------------8<--------------------------- > Subject: [PATCH] ALSA: usb-audio: reduce "cannot get freq at ep" spew > > If a device fails to support reading the sample rate, it will likely > fail again and there is no point in logging the message every time the > sample rate is set. > > This reduces dmesg noise when using the Benchmark DAC1 PRE as a USB > sound card. Hm, this shuts up all the rest for the device, so it also hides if the message comes from multiple EPs. Takashi > > Signed-off-by: Eric Wong <normalperson(a)yhbt.net> > --- > sound/usb/clock.c | 7 +++++-- > sound/usb/usbaudio.h | 3 ++- > 2 files changed, 7 insertions(+), 3 deletions(-) > > diff --git a/sound/usb/clock.c b/sound/usb/clock.c > index 03fed66..512fe12 100644 > --- a/sound/usb/clock.c > +++ b/sound/usb/clock.c > @@ -307,8 +307,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, > USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, > UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, > data, sizeof(data))) < 0) { > - dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n", > - iface, fmt->altsetting, ep); > + if (!chip->no_get_freq) { > + dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n", > + iface, fmt->altsetting, ep); > + chip->no_get_freq = 1; > + } > return 0; /* some devices don't support reading */ > } > > diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h > index 91d0380..be65671 100644 > --- a/sound/usb/usbaudio.h > +++ b/sound/usb/usbaudio.h > @@ -43,7 +43,8 @@ struct snd_usb_audio { > unsigned int in_pm:1; > unsigned int autosuspended:1; > unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ > - > + unsigned int no_get_freq:1; > + > int num_interfaces; > int num_suspended_intf; > > -- > EW >

1 0

Re: [alsa-devel] [PATCH] ALSA: usb-audio: don't try to get Benchmark DAC1 sample rate
by Takashi Iwai 04 Apr '15

04 Apr '15

At Tue, 31 Mar 2015 07:34:05 +0000, Eric Wong wrote: > > Adding this quirk allows us to avoid the noisy > "cannot get freq at ep 0x1" message in dmesg output every time > playback starts. > > This ought to affect other Benchmark DAC1 variations using the same > "Microchip Technology, Inc." chip as well, but I have only tested > with the "Pre" variant. > > Signed-off-by: Eric Wong <normalperson(a)yhbt.net> > Cc: Joe Turner <joe(a)oampo.co.uk> > Cc: Takashi Iwai <tiwai(a)suse.de> Applied, thanks. Takashi > --- > I also had another generic patch prepared before I noticed Joe's > quirk addition for the MS Lifecam HD-5000. > > sound/usb/quirks.c | 9 +++++++-- > 1 file changed, 7 insertions(+), 2 deletions(-) > > diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c > index 753a47d..9a28365 100644 > --- a/sound/usb/quirks.c > +++ b/sound/usb/quirks.c > @@ -1113,8 +1113,13 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, > > bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) > { > - /* MS Lifecam HD-5000 doesn't support reading the sample rate. */ > - return chip->usb_id == USB_ID(0x045E, 0x076D); > + /* devices which do not support reading the sample rate. */ > + switch (chip->usb_id) { > + case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */ > + case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ > + return true; > + } > + return false; > } > > /* Marantz/Denon USB DACs need a vendor cmd to switch > -- > EW >

1 0

[alsa-devel] Support Dell headset mode for ALC256
by Kailang 04 Apr '15

04 Apr '15

Hi Takashi, Dell has new platform with ALC256. Attach patch is support headset mode for ALC256. Thanks. BR, Kailang

2 1

Re: [alsa-devel] [PATCH 76/86] sound/pci: use uapi/linux/pci_ids.h directly
by Takashi Iwai 04 Apr '15

04 Apr '15

At Sun, 29 Mar 2015 15:42:59 +0200, Michael S. Tsirkin wrote: > > Header moved from linux/pci_ids.h to uapi/linux/pci_ids.h, > use the new header directly so we can drop > the wrapper in include/linux/pci_ids.h. > > Signed-off-by: Michael S. Tsirkin <mst(a)redhat.com> Do we need this change in the driver code at all...? I thought include/uapi is already included in the include path. Takashi > --- > sound/pci/ctxfi/xfi.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c > index f2f3277..7e26876 100644 > --- a/sound/pci/ctxfi/xfi.c > +++ b/sound/pci/ctxfi/xfi.c > @@ -11,7 +11,7 @@ > #include <linux/init.h> > #include <linux/pci.h> > #include <linux/moduleparam.h> > -#include <linux/pci_ids.h> > +#include <uapi/linux/pci_ids.h> > #include <linux/module.h> > #include <sound/core.h> > #include <sound/initval.h> > -- > MST >

1 0

[alsa-devel] [PATCH v1.1 07/14] media: omap3isp: remove unused clkdev
by Laurent Pinchart 03 Apr '15

03 Apr '15

From: Russell King <rmk+kernel(a)arm.linux.org.uk> No merged platform supplies xclks via platform data. As we want to slightly change the clkdev interface, rather than fixing this unused code, remove it instead. Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> Signed-off-by: Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> --- drivers/media/platform/omap3isp/isp.c | 24 ------------------------ drivers/media/platform/omap3isp/isp.h | 1 - include/media/omap3isp.h | 6 ------ 3 files changed, 31 deletions(-) In case this would be helpful, I've rebased the original patch on top of the linuxtv master branch scheduled for merge in v4.1. diff --git a/drivers/media/platform/omap3isp/isp.c b/drivers/media/platform/omap3isp/isp.c index ff51c4f..18d0a87 100644 --- a/drivers/media/platform/omap3isp/isp.c +++ b/drivers/media/platform/omap3isp/isp.c @@ -304,7 +304,6 @@ static struct clk *isp_xclk_src_get(struct of_phandle_args *clkspec, void *data) static int isp_xclk_init(struct isp_device *isp) { - struct isp_platform_data *pdata = isp->pdata; struct device_node *np = isp->dev->of_node; struct clk_init_data init; unsigned int i; @@ -335,26 +334,6 @@ static int isp_xclk_init(struct isp_device *isp) xclk->clk = clk_register(NULL, &xclk->hw); if (IS_ERR(xclk->clk)) return PTR_ERR(xclk->clk); - - /* When instantiated from DT we don't need to register clock - * aliases. - */ - if (np) - continue; - - if (!pdata || (pdata->xclks[i].con_id == NULL && - pdata->xclks[i].dev_id == NULL)) - continue; - - xclk->lookup = kzalloc(sizeof(*xclk->lookup), GFP_KERNEL); - if (xclk->lookup == NULL) - return -ENOMEM; - - xclk->lookup->con_id = pdata->xclks[i].con_id; - xclk->lookup->dev_id = pdata->xclks[i].dev_id; - xclk->lookup->clk = xclk->clk; - - clkdev_add(xclk->lookup); } if (np) @@ -376,9 +355,6 @@ static void isp_xclk_cleanup(struct isp_device *isp) if (!IS_ERR(xclk->clk)) clk_unregister(xclk->clk); - - if (xclk->lookup) - clkdev_drop(xclk->lookup); } } diff --git a/drivers/media/platform/omap3isp/isp.h b/drivers/media/platform/omap3isp/isp.h index 431224e..e579943 100644 --- a/drivers/media/platform/omap3isp/isp.h +++ b/drivers/media/platform/omap3isp/isp.h @@ -132,7 +132,6 @@ enum isp_xclk_id { struct isp_xclk { struct isp_device *isp; struct clk_hw hw; - struct clk_lookup *lookup; struct clk *clk; enum isp_xclk_id id; diff --git a/include/media/omap3isp.h b/include/media/omap3isp.h index 0f0c08b..048f8f9 100644 --- a/include/media/omap3isp.h +++ b/include/media/omap3isp.h @@ -150,13 +150,7 @@ struct isp_platform_subdev { struct isp_bus_cfg *bus; }; -struct isp_platform_xclk { - const char *dev_id; - const char *con_id; -}; - struct isp_platform_data { - struct isp_platform_xclk xclks[2]; struct isp_platform_subdev *subdevs; void (*set_constraints)(struct isp_device *isp, bool enable); }; -- Regards, Laurent Pinchart

2 1

Re: [alsa-devel] [PATCH 1/6] ASoC: jz4740: Remove Makefile entry for removed file
by Greg Kroah-Hartman 03 Apr '15

03 Apr '15

On Fri, Mar 27, 2015 at 10:51:43AM +0000, Markos Chandras wrote: > On 04/22/2014 09:46 PM, Lars-Peter Clausen wrote: > > Commit 0406a40a0 ("ASoC: jz4740: Use the generic dmaengine PCM driver") > > jz4740-pcm.c file, but neglected to remove the Makefile entries. > > > > Fixes: 0406a40a0 ("ASoC: jz4740: Use the generic dmaengine PCM driver") > > Reported-by: kbuild test robot <fengguang.wu(a)intel.com> > > Reported-by: Ralf Baechle <ralf(a)linux-mips.org> > > Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> > > --- > > sound/soc/jz4740/Makefile | 2 -- > > 1 file changed, 2 deletions(-) > > > > diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile > > index be873c1..d32c540 100644 > > --- a/sound/soc/jz4740/Makefile > > +++ b/sound/soc/jz4740/Makefile > > @@ -1,10 +1,8 @@ > > # > > # Jz4740 Platform Support > > # > > -snd-soc-jz4740-objs := jz4740-pcm.o > > snd-soc-jz4740-i2s-objs := jz4740-i2s.o > > > > -obj-$(CONFIG_SND_JZ4740_SOC) += snd-soc-jz4740.o > > obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o > > > > # Jz4740 Machine Support > > > Hi, > > This patch (eebdec044e82) is present in 3.15-rc1 but the build failure > that was introduced by 0406a40a0 ("ASoC: jz4740: Use the generic > dmaengine PCM driver") is present in 3.14-rc1 so 3.14 is still broken. > > Greg, would it be possible to cherry pick eebdec044e82 ("ASoC: jz4740: > Remove Makefile entry for removed file") to 3.14 stable branch? It seems > to apply without conflicts. Now applied, thanks. greg k-h

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[alsa-devel] dummy codec + simple card combination
by Belisko Marek 03 Apr '15

03 Apr '15

Hi, I'm trying to add audio support for gta04 (for gsm + bt and fm audio). All audio sources are clock masters so basically connection is dummy codecs -> cpu dai (mcbsp ports). For implementing such thing I can see 2 options: 1. simple card driver which can easily describe cpu dai interface but dummy codec cannot because there are no DT bindings (there was sever attempts to add DT bindings for dummy-codec but AFAIK none was accepted) This turns it to only solution to create gta04 soundcard which will get from DT cpu dai links and use in card driver dummy codecs directly. Is my assumption correct? I would like to get some feedback before wasting my time with implementing something which cannot be pushed mainline. Thanks for all suggestions. BR, marek -- as simple and primitive as possible ------------------------------------------------- Marek Belisko - OPEN-NANDRA Freelance Developer Ruska Nova Ves 219 | Presov, 08005 Slovak Republic Tel: +421 915 052 184 skype: marekwhite twitter: #opennandra web: http://open-nandra.com

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[alsa-devel] Advanced Gtk+ Sequencer aka GSequencer now on GitHub
by Joël Krähemann 03 Apr '15

03 Apr '15

Hi all, me just migrated gsequencer to GitHub. http://www.gsequencer.org Further me did renamed binary from `ags` to `gsequencer`. About the state of the sequencer: Version 0.4.2-44 is believed to be stable. Since me don't have the time to test it well, me need people to test it. Please, let me know your experiences. MIDI support is intended to be added for release Version 0.4.3 and many other new exciting features. If someone can help with ALSA rawmidi, please let me know. I don't want to do JACK, since it is believed to be somehow the Qt thing. What me can say so far about MIDI me read years ago about it in "The computer music tutorial - The MIT Press, ISBN 0-262-68082-3". And MIDI consist of messages which are routeable. Someone giving specification how the MIDI message struct is passed to ALSA and how it looks like would be great. Thank you in advance Joël

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[alsa-devel] [PATCH] ASoC: cs4271: Increase delay time after reset
by pascal.huerst＠gmail.com 02 Apr '15

02 Apr '15

From: Pascal Huerst <pascal.huerst(a)gmail.com> The delay time after a reset in the codec probe callback was too short, and did not work on certain hw because the codec needs more time to power on. This increases the delay time from 1us to 1ms. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- sound/soc/codecs/cs4271.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 7d3a6ac..e770ee6 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -561,10 +561,10 @@ static int cs4271_codec_probe(struct snd_soc_codec *codec) if (gpio_is_valid(cs4271->gpio_nreset)) { /* Reset codec */ gpio_direction_output(cs4271->gpio_nreset, 0); - udelay(1); + mdelay(1); gpio_set_value(cs4271->gpio_nreset, 1); /* Give the codec time to wake up */ - udelay(1); + mdelay(1); } ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, -- 1.9.3

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[alsa-devel] [PATCH] ASoC: atmel: ac97c: Remove unused variable
by Fabio Estevam 02 Apr '15

02 Apr '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Remove the unused 'match' variable to fix the following build warning: sound/atmel/ac97c.c:919:29: warning: unused variable 'match' [-Wunused-variable] Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/atmel/ac97c.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index cf4cedf..6dad042 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -916,7 +916,6 @@ static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) { struct ac97c_platform_data *pdata; struct device_node *node = dev->of_node; - const struct of_device_id *match; if (!node) { dev_err(dev, "Device does not have associated DT data\n"); -- 1.9.1

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[alsa-devel] [RFC 0/11] dw_hdmi cleanups, audio preparation, helpers and ahb audio support
by Russell King - ARM Linux 02 Apr '15

02 Apr '15

I'm sending this series for comments, and to allow people to update their code bases (for those who are using my AHB audio driver on iMX6). This is really several series, and I'm not expecting this to be ready for the upcoming merge window. That said, if anyone wants to say that they're happy with some of the initial four patches, I'm happy to queue those for David, if he's willing to take them. This series applies on top of my previous series, which finished with the patch "drm: bridge/dw_hdmi: adjust n/cts setting order". The first four patches are a few cleanups to the dw_hdmi driver. The next two patches introduce new interfaces to the dw_hdmi driver to support AHB audio, including the errata found on iMX6 which requires N to be programmed to zero when the audio FIFO is not full. I'm expecting some discussion with these last two as we try to work out how to deal with the two variants of audio support for this part (AHB audio vs I2S audio). The following three add various helpers to the DRM and ALSA subsystems for audio support which is non-specific to the AHB audio driver. For DRM, this is a helper macro to obtain the pointer into the short audio descriptors. For ALSA, it's a set of helpers to restrict an audio PCM device's capabilities according to the ELD, and to generate the IEC958 channel status data. If these helpers are acceptable, we can convert a number of drivers to them. The ELD helper isn't quite finished, but should be sufficient for initial testing. The last two patches add support for the dw_hdmi audio. This is an ALSA driver rather than an ASoC driver as it's a pretty simple affair. I couldn't find a suitable location in the sound/ subtree to place it, so I've placed it next to the bridge - yes, we probably need to find it a better home in the sound/ subtree, but it currently makes use of dw_hdmi.h in the directory it lives. drivers/gpu/drm/bridge/Kconfig | 11 + drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.c | 566 +++++++++++++++++++++++++++++ drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.h | 14 + drivers/gpu/drm/bridge/dw_hdmi.c | 250 ++++++++----- drivers/gpu/drm/bridge/dw_hdmi.h | 3 + include/drm/bridge/dw_hdmi.h | 7 + include/drm/drm_edid.h | 19 + include/sound/pcm_drm_eld.h | 6 + include/sound/pcm_iec958.h | 9 + sound/core/Kconfig | 6 + sound/core/Makefile | 3 + sound/core/pcm_drm_eld.c | 91 +++++ sound/core/pcm_iec958.c | 70 ++++ 14 files changed, 964 insertions(+), 92 deletions(-) -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

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[alsa-devel] [PATCH] ASoC: atmel-pcm-dma: increase buffer_bytes_max
by Alexandre Belloni 02 Apr '15

02 Apr '15

atmel-pcm-dma is not limited to a buffer size of 64kB like atmel-pcm-pdc. Increase buffer_bytes_max to 512kB to allow for higher bit rates (i.e. 32bps at 192kHz) to work correctly. By default, keep the prealloc at 64kB. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- sound/soc/atmel/atmel-pcm-dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index b8e7bad05eb1..b6625c8c411b 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -54,7 +54,7 @@ static const struct snd_pcm_hardware atmel_pcm_dma_hardware = { .period_bytes_max = 2 * 0xffff, /* if 2 bytes format */ .periods_min = 8, .periods_max = 1024, /* no limit */ - .buffer_bytes_max = ATMEL_SSC_DMABUF_SIZE, + .buffer_bytes_max = 512 * 1024, }; /** @@ -119,7 +119,7 @@ static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream, static const struct snd_dmaengine_pcm_config atmel_dmaengine_pcm_config = { .prepare_slave_config = atmel_pcm_configure_dma, .pcm_hardware = &atmel_pcm_dma_hardware, - .prealloc_buffer_size = ATMEL_SSC_DMABUF_SIZE, + .prealloc_buffer_size = 64 * 1024, }; int atmel_pcm_dma_platform_register(struct device *dev) -- 2.1.0

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[alsa-devel] ASoC: cs4271: init/timing problem
by Pascal Huerst 02 Apr '15

02 Apr '15

Hey all, We have a custom built hw, based on am335x and from time to time we need to rmmod/modprobe the ASoC machine driver. Depending on the hw, the first call to regmap_update_bits(..) in cs4271_codec_probe(..) fails with -EREMOTEIO. The error is originated in: drivers/i2c/busses/i2c-omap.c and happens if no i2c package acknowledge is received by the host. I think this is a timing issue and on some devices, the codec is just not ready yet, for communication. What is the right way to fix that? My attempt would be something like this: const int retries = 5; while (retries) { ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN | CS4271_MODE2_CPEN, CS4271_MODE2_PDN | CS4271_MODE2_CPEN); if (ret == -EREMOTEIO) { retries--; udelay(1); continue; } else if (ret) return ret; break; } But I might also just add a proper delay before the first call to regmap_update_bits(..) So what is the proper way to handle this? Thanks Pascal

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Re: [alsa-devel] Realtek ALC662 issues - Intel DG41CN
by Raymond Yau 02 Apr '15

02 Apr '15

> > >> You don't need any model since the pin defaults of your Intel DG41CN are correct unless you want to use CD audio in connector > > Originally, after the install there was no "model" there. It did not work even then. This entry that you see is because of my trials to make it work. You have to post output of alsa-info.sh without specify any model > > >>You should check the front audio panel of your computer chassis if jack state of headphone is alway plugged > Don't think that is the case. Because it works just fine on the Windows OS. In fact, as I have mentioned, there is some choppiness/stutter sound that I hear occasionally when I play in right now on Fedora/Ubuntu (while most of the time there is no sound). This behaviour goes away when I put the headphone (and then there is no sound at all from the main speaker) Do the front panel headphone still work when you use rear panel jacks for 5.1 sound in windows ? > > >>You have to provide pulseaudio verbose log > Pulseaudio is not a problem because I tested it by killing Pulseaudio too. The same problem persists. Though, if you insist I can surely get you the output. Killing pulseaudio won't work since it can auto-spwan http://lists.freedesktop.org/archives/pulseaudio-discuss/2015-March/023568.… Check the pulseaudio verbose log , as your line out and headphone share the same playback volume control, dB min of line out path and headphone path should be same value http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/m… <http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix…> http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… If Front is shared by HP and Line out, Why set volume to zero dB ? volume should remain unchanged , only need to restore save values after reboot > > >>You can disable multi channels by using early patching and hint multi_io=0 to get back the headphone playback volume contro > How do I do that? Appreciate your help here! > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… Codec: Realtek ALC662 rev1 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0662 Subsystem Id: 0x8086434e Revision Id: 0x100101 Early Patching ~~~~~~~~~~~~~~ When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a firmware file for modifying the HD-audio setup before initializing the codec. This can work basically like the reconfiguration via sysfs in the above, but it does it before the first codec configuration. A patch file is a plain text file which looks like below: ------------------------------------------------------------------------ [codec] 0x12345678 0xabcd1234 2 [hint] multi_io = no ------------------------------------------------------------------------ The file needs to have a line `[codec]`. The next line should contain three numbers indicating the codec vendor-id (0x12345678 in the example), the codec subsystem-id (0xabcd1234) and the address (2) of the codec. The rest patch entries are applied to this specified codec until another codec entry is given. Passing 0 or a negative number to the first or the second value will make the check of the corresponding field be skipped. It'll be useful for really broken devices that don't initialize SSID properly. The hd-audio driver reads the file via request_firmware(). Thus, a patch file has to be located on the appropriate firmware path, typically, /lib/firmware. For example, when you pass the option `patch=hda-init.fw`, the file /lib/firmware/hda-init.fw must be present Post output of alsa-info.sh after you reboot with early patching You will get headphone playback volume control but loss those surround , center and lfe playback volume controls http://www.intel.com/support/motherboards/desktop/sb/CS-034206.htm Seem your motherboard does not support multistreaming as it is not in the supported list Do this mean that indep_hp should be explicitly disabled for alc662 ?

1 0

Re: [alsa-devel] [PATCH] fix dig out type to SPDIF when there are two digital pins with different type - VT1708S lost SPDIF device after upgrade
by Raymond Yau 02 Apr '15

02 Apr '15

> > > > Thank you for your reply, > > > > Please see below the output of alsa-info.sh before and after the headphone plugged in > > > > Before: http://www.alsa-project.org/db/?f=a316eebf7b150b5897838be37bce4e8c9d95eaae > > After: http://www.alsa-project.org/db/?f=0f163003e9005972db5f97f21a45be67816e3801 > > > > I have 6 Jacks at the rear panel and two at the top (one output one input) > > I am afraid I do not know how to switch channel mode to retask. > > > > Also, the computer is of a certain age, and the issue has started several month ago but I never really looked into it, so I am not sure that change in power management will be relevant has it seems quite recent. > > What is the model of your asus motherbroad ? > > Seem lost the digial device 1 for SPDIF out > > Pin Default 0x985601f0: [Fixed] Digital Out at Int HDMI > > Is this internal spdif connector ? > > Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 > Device: name="VT1708S Digital", type="HDMI", device=3 > > APLAY > > **** List of PLAYBACK Hardware Devices **** > card 0: Intel [HDA Intel], device 0: VT1708S Analog [VT1708S Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: Intel [HDA Intel], device 3: VT1708S Digital [VT1708S Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > > Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital > Control: name="IEC958 Playback Con Mask", index=0, device=0 > Control: name="IEC958 Playback Pro Mask", index=0, device=0 > Control: name="IEC958 Playback Default", index=0, device=0 > Control: name="IEC958 Playback Switch", index=0, device=0 > Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 > Device: name="VT1708S Digital", type="HDMI", device=3 > Converter: stream=5, channel=0 > Digital: > Digital category: 0x0 > IEC Coding Type: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > > Power states: D0 D1 D2 D3 > Power: setting=D0, actual=D0 > > Node 0x15 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=5, channel=0 > Digital: > Digital category: 0x0 > IEC Coding Type: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > > Power states: D0 D1 D2 D3 > Power: setting=D0, actual=D0 > > Node 0x20 [Pin Complex] wcaps 0x400701: Stereo Digital > Control: name="HDMI Phantom Jack", index=0, device=0 > Pincap 0x00000010: OUT > Pin Default 0x985601f0: [Fixed] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > > Power states: D0 D1 D2 D3 > Power: setting=D0, actual=D0 > Connection: 1 > 0x12 > Node 0x21 [Pin Complex] wcaps 0x400701: Stereo Digital > Control: name="SPDIF Phantom Jack", index=0, device=0 > Pincap 0x00000010: OUT > Pin Default 0x074511f0: [Jack] SPDIF Out at Ext Rear Panel > Conn = Optical, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > > Power states: D0 D1 D2 D3 > Power: setting=D0, actual=D0 > Connection: 1 > 0x15 Do the driver need a pin fixup for the digital pin or more check when assign dig out type ? If codec has two digital pin complex with diffferent type, check whether the primary digital pin complex support DETECT case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins)) { codec_info(codec, "ignore pin 0x%x, too many assigned pins\n", nid); continue; } cfg->dig_out_pins[cfg->dig_outs] = nid; cfg->dig_out_type[cfg->dig_outs] = (loc == AC_JACK_LOC_HDMI) ? HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF; cfg->dig_outs++; + if (cfg->dig_outs == 2) + if (cfg->dig_out_type[0] != cfg->dig_out_type[1]) + if (! snd_hda_query_pin_caps(codec, cfg->dig_out_pins[0]) & AC_PINCAP_PRES_DETECT)) + cfg-> dig_out_type[0] = HDA_PCM_TYPE_SPDIF; break;

1 0

[alsa-devel] [PATCH v2 0/5] Intel SST driver restructure
by Jie Yang 02 Apr '15

02 Apr '15

In order to prepare for the upstreaming of Skylake drivers and to allow reuse of code between different Intel audio DSP products, this patch series restructures the Intel directory as follows: 1. Create "common" directory. Move common files into this directory. 2. Create "boards" directory. Move all machine drivers into this directory. 3. Create "haswell" directory. Move all remaining hsw/bdw code into this directory. 4. Create "baytrail" directory. Move remaining baytrail driver to this directory. 5. Create "atom" directory. Move all MCG platform drivers(including sst/ folder) to this directory. Changes in v2: add "Reviewed-by" and "Acked-by" items from each platforms' maintainer. Todo next step: 1. Change and move atom related codes(in atom/sst/) to corresponding and try share as many as possible codes(e.g. sst_acpi.c). 2. Share more common codes about IPC for different platforms, e.g. The methods for queueing and sending/reciving Tx/Rx messages, The methods for mailbox IO, The IPC error return codes, etc. Jie Yang (5): ASoC: Intel: create common folder and move common files in ASoC: Intel: create haswell folder and move haswell platform files in ASoC: Intel: create boards folder and move sst boards files in ASoC: Intel: create baytrail folder and move baytrail platform files in ASoC: Intel: create atom folder and move atom platform files in sound/soc/intel/Makefile | 42 +++------------------- sound/soc/intel/atom/Makefile | 7 ++++ sound/soc/intel/{ => atom}/sst-atom-controls.c | 0 sound/soc/intel/{ => atom}/sst-atom-controls.h | 0 sound/soc/intel/{ => atom}/sst-mfld-dsp.h | 0 .../intel/{ => atom}/sst-mfld-platform-compress.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform.h | 0 sound/soc/intel/{ => atom}/sst/Makefile | 0 sound/soc/intel/{ => atom}/sst/sst.c | 0 sound/soc/intel/{ => atom}/sst/sst.h | 0 sound/soc/intel/{ => atom}/sst/sst_acpi.c | 0 sound/soc/intel/{ => atom}/sst/sst_drv_interface.c | 0 sound/soc/intel/{ => atom}/sst/sst_ipc.c | 0 sound/soc/intel/{ => atom}/sst/sst_loader.c | 0 sound/soc/intel/{ => atom}/sst/sst_pci.c | 0 sound/soc/intel/{ => atom}/sst/sst_pvt.c | 0 sound/soc/intel/{ => atom}/sst/sst_stream.c | 0 sound/soc/intel/baytrail/Makefile | 4 +++ sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c | 4 +-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c | 4 +-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h | 0 sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c | 4 +-- sound/soc/intel/boards/Makefile | 15 ++++++++ sound/soc/intel/{ => boards}/broadwell.c | 6 ++-- sound/soc/intel/{ => boards}/byt-max98090.c | 2 +- sound/soc/intel/{ => boards}/byt-rt5640.c | 4 +-- .../{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} | 0 sound/soc/intel/{ => boards}/cht_bsw_rt5645.c | 0 sound/soc/intel/{ => boards}/cht_bsw_rt5672.c | 0 sound/soc/intel/{ => boards}/haswell.c | 6 ++-- sound/soc/intel/{ => boards}/mfld_machine.c | 0 sound/soc/intel/common/Makefile | 6 ++++ sound/soc/intel/{ => common}/sst-acpi.c | 0 sound/soc/intel/{ => common}/sst-dsp-priv.h | 0 sound/soc/intel/{ => common}/sst-dsp.c | 0 sound/soc/intel/{ => common}/sst-dsp.h | 0 sound/soc/intel/{ => common}/sst-firmware.c | 0 sound/soc/intel/haswell/Makefile | 4 +++ sound/soc/intel/{ => haswell}/sst-haswell-dsp.c | 6 ++-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.c | 4 +-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.h | 0 sound/soc/intel/{ => haswell}/sst-haswell-pcm.c | 6 ++-- 43 files changed, 64 insertions(+), 60 deletions(-) create mode 100644 sound/soc/intel/atom/Makefile rename sound/soc/intel/{ => atom}/sst-atom-controls.c (100%) rename sound/soc/intel/{ => atom}/sst-atom-controls.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-dsp.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-compress.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform.h (100%) rename sound/soc/intel/{ => atom}/sst/Makefile (100%) rename sound/soc/intel/{ => atom}/sst/sst.c (100%) rename sound/soc/intel/{ => atom}/sst/sst.h (100%) rename sound/soc/intel/{ => atom}/sst/sst_acpi.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_drv_interface.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_ipc.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_loader.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_pci.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_pvt.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_stream.c (100%) create mode 100644 sound/soc/intel/baytrail/Makefile rename sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h (100%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c (99%) create mode 100644 sound/soc/intel/boards/Makefile rename sound/soc/intel/{ => boards}/broadwell.c (98%) rename sound/soc/intel/{ => boards}/byt-max98090.c (99%) rename sound/soc/intel/{ => boards}/byt-rt5640.c (98%) rename sound/soc/intel/{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} (100%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5645.c (100%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5672.c (100%) rename sound/soc/intel/{ => boards}/haswell.c (98%) rename sound/soc/intel/{ => boards}/mfld_machine.c (100%) create mode 100644 sound/soc/intel/common/Makefile rename sound/soc/intel/{ => common}/sst-acpi.c (100%) rename sound/soc/intel/{ => common}/sst-dsp-priv.h (100%) rename sound/soc/intel/{ => common}/sst-dsp.c (100%) rename sound/soc/intel/{ => common}/sst-dsp.h (100%) rename sound/soc/intel/{ => common}/sst-firmware.c (100%) create mode 100644 sound/soc/intel/haswell/Makefile rename sound/soc/intel/{ => haswell}/sst-haswell-dsp.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.h (100%) rename sound/soc/intel/{ => haswell}/sst-haswell-pcm.c (99%) -- 1.9.1

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Re: [alsa-devel] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used
by Dan Carpenter 01 Apr '15

01 Apr '15

Hello Jyri Sarha, The patch a75a053f1eef: "ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used" from Mar 20, 2015, leads to the following Smatch warning: sound/soc/davinci/davinci-mcasp.c:1152 davinci_mcasp_startup() error: buffer overflow 'mcasp->ruledata' 2 <= 2 sound/soc/davinci/davinci-mcasp.c 1138 1139 /* 1140 * Limit the maximum allowed channels for the first stream: 1141 * number of serializers for the direction * tdm slots per serializer 1142 */ 1143 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) 1144 dir = TX_MODE; 1145 else 1146 dir = RX_MODE; dir is either 1 or 2. 1147 1148 for (i = 0; i < mcasp->num_serializer; i++) { 1149 if (mcasp->serial_dir[i] == dir) 1150 max_channels++; 1151 } 1152 mcasp->ruledata[dir].serializers = max_channels; ^^^^^^^^^^^^^ ->ruledata[] has only two elements so ->ruledata[2] is beyond the end of the array. 1153 max_channels *= mcasp->tdm_slots; regards, dan carpenter

3 2

[alsa-devel] [PATCH] ASoC: omap-hdmi-audio: No not use IEC958_AES1_PRO_MODE_NOTID
by Jyri Sarha 01 Apr '15

01 Apr '15

No IEC958_AES?_PRO_* macros should be used in HDMI consumer audio mode and IEC958_AES1_PRO_MODE_NOTID should be applied to byte 1 when applicable. However IEC958_AES1_PRO_MODE_NOTID is defined as 0 so this fix does not affect the functionality in any way. Reported-by: Russell King <linux(a)arm.linux.org.uk> Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- When I refactored the OMAP HDMI audio functionality I intentionally wanted to keep all the HW bits identical to the earlier implementation so that I could compare the register values of the old and the new implementations one to one. One of these days I should go trough and check all the bits now that new implementation has settled in. Best regards, Jyri sound/soc/omap/omap-hdmi-audio.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index f7eb42a..4775da4 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -142,8 +142,6 @@ static int hdmi_dai_hw_params(struct snd_pcm_substream *substream, iec->status[0] |= IEC958_AES0_CON_EMPHASIS_NONE; - iec->status[0] |= IEC958_AES1_PRO_MODE_NOTID; - iec->status[1] = IEC958_AES1_CON_GENERAL; iec->status[2] |= IEC958_AES2_CON_SOURCE_UNSPEC; -- 1.9.1

2 1

[alsa-devel] [PATCH 1/8] ASoC: wm8971: Use system_power_efficient_wq instead of custom workqueue
by Lars-Peter Clausen 01 Apr '15

01 Apr '15

The delayed work used by the wm8971 driver to manage the caps charging doesn't have any special requirements that would justify using a custom workqueue, just use the generic system_power_efficient_wq instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8971.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 39ddb9b..44baacd 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -31,8 +31,6 @@ #define WM8971_REG_COUNT 43 -static struct workqueue_struct *wm8971_workq = NULL; - /* codec private data */ struct wm8971_priv { unsigned int sysclk; @@ -636,7 +634,8 @@ static int wm8971_resume(struct snd_soc_codec *codec) reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_ON; - queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, + queue_delayed_work(system_power_efficient_wq, + &codec->dapm.delayed_work, msecs_to_jiffies(1000)); } @@ -649,9 +648,6 @@ static int wm8971_probe(struct snd_soc_codec *codec) u16 reg; INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8971_work); - wm8971_workq = create_workqueue("wm8971"); - if (wm8971_workq == NULL) - return -ENOMEM; wm8971_reset(codec); @@ -659,7 +655,8 @@ static int wm8971_probe(struct snd_soc_codec *codec) reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; - queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, + queue_delayed_work(system_power_efficient_wq, + &codec->dapm.delayed_work, msecs_to_jiffies(1000)); /* set the update bits */ @@ -681,8 +678,6 @@ static int wm8971_remove(struct snd_soc_codec *codec) { wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (wm8971_workq) - destroy_workqueue(wm8971_workq); return 0; } -- 1.8.0

3 9

[alsa-devel] [PATCH] ASoC: atmel-pcm-pdc: merge atmel-pcm back in
by Alexandre Belloni 01 Apr '15

01 Apr '15

atmel-pcm.c was split into two files to create a generic framework for both PDC and DMA. atmel-pcm-dma.c is using the generic dmaengine framework since 95e0e07e710e (ASoC: atmel-pcm: use generic dmaengine framework). Merge atmel-pcm.c in atmel-pcm-pdc.c as this is now the only user. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- sound/soc/atmel/Makefile | 2 - sound/soc/atmel/atmel-pcm-pdc.c | 79 ++++++++++++++++++++++++++ sound/soc/atmel/atmel-pcm.c | 121 ---------------------------------------- sound/soc/atmel/atmel-pcm.h | 5 -- 4 files changed, 79 insertions(+), 128 deletions(-) delete mode 100644 sound/soc/atmel/atmel-pcm.c diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 466a821da98c..b327e5cc8de3 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -1,10 +1,8 @@ # AT91 Platform Support -snd-soc-atmel-pcm-objs := atmel-pcm.o snd-soc-atmel-pcm-pdc-objs := atmel-pcm-pdc.o snd-soc-atmel-pcm-dma-objs := atmel-pcm-dma.o snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o -obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o obj-$(CONFIG_SND_ATMEL_SOC_PDC) += snd-soc-atmel-pcm-pdc.o obj-$(CONFIG_SND_ATMEL_SOC_DMA) += snd-soc-atmel-pcm-dma.o obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c index a366b3503c28..da861b44413f 100644 --- a/sound/soc/atmel/atmel-pcm-pdc.c +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -47,6 +47,85 @@ #include "atmel-pcm.h" +static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = ATMEL_SSC_DMABUF_SIZE; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n", + (void *)buf->area, (void *)(long)buf->addr, size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} + +static int atmel_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + pr_debug("atmel-pcm: allocating PCM playback DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + pr_debug("atmel-pcm: allocating PCM capture DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +static void atmel_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + /*--------------------------------------------------------------------------*\ * Hardware definition \*--------------------------------------------------------------------------*/ diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c deleted file mode 100644 index 8ae3fa5ac60a..000000000000 --- a/sound/soc/atmel/atmel-pcm.c +++ /dev/null @@ -1,121 +0,0 @@ -/* - * atmel-pcm.c -- ALSA PCM interface for the Atmel atmel SoC. - * - * Copyright (C) 2005 SAN People - * Copyright (C) 2008 Atmel - * - * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> - * - * Based on at91-pcm. by: - * Frank Mandarino <fmandarino(a)endrelia.com> - * Copyright 2006 Endrelia Technologies Inc. - * - * Based on pxa2xx-pcm.c by: - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, Inc. - * - * 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. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - */ - -#include <linux/module.h> -#include <linux/dma-mapping.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include "atmel-pcm.h" - -static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, - int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = ATMEL_SSC_DMABUF_SIZE; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_coherent(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n", - (void *)buf->area, (void *)(long)buf->addr, size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -int atmel_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, - vma->vm_end - vma->vm_start, vma->vm_page_prot); -} -EXPORT_SYMBOL_GPL(atmel_pcm_mmap); - -int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - pr_debug("atmel-pcm: allocating PCM playback DMA buffer\n"); - ret = atmel_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - pr_debug("atmel-pcm: allocating PCM capture DMA buffer\n"); - ret = atmel_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} -EXPORT_SYMBOL_GPL(atmel_pcm_new); - -void atmel_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - dma_free_coherent(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} -EXPORT_SYMBOL_GPL(atmel_pcm_free); - diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h index 12ae814eff21..6eaf081cad50 100644 --- a/sound/soc/atmel/atmel-pcm.h +++ b/sound/soc/atmel/atmel-pcm.h @@ -83,11 +83,6 @@ struct atmel_pcm_dma_params { #define ssc_readx(base, reg) (__raw_readl((base) + (reg))) #define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) -int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd); -void atmel_pcm_free(struct snd_pcm *pcm); -int atmel_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma); - #if defined(CONFIG_SND_ATMEL_SOC_PDC) || \ defined(CONFIG_SND_ATMEL_SOC_PDC_MODULE) int atmel_pcm_pdc_platform_register(struct device *dev); -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: rsnd: add Synchronous SRC mode
by Kuninori Morimoto 01 Apr '15

01 Apr '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Renesas R-Car sound SRC (= Sampling Rate Converter) has Asynchronous/Synchronous SRC mode. Asynchronous mode is already supported via DPCM. This patch adds Synchronous mode on it. The condition of enabling Synchronous mode are - SoC is clock master - Sound uses SRC - Sound doesn't use DVC - Sound card uses DPCM (= rsrc-card card) amixer set "SRC Out Rate" on aplay xxx.wav & amixer set "SRC Out Rate" 48000 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/src.c | 126 ++++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 121 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index a0a2bda..3beb32e 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -22,13 +22,15 @@ struct rsnd_src { struct rsnd_src_platform_info *info; /* rcar_snd.h */ struct rsnd_mod mod; + struct rsnd_kctrl_cfg_s sen; /* sync convert enable */ + struct rsnd_kctrl_cfg_s sync; /* sync convert */ u32 convert_rate; /* sampling rate convert */ int err; }; #define RSND_SRC_NAME_SIZE 16 -#define rsnd_src_convert_rate(s) ((s)->convert_rate) +#define rsnd_enable_sync_convert(src) ((src)->sen.val) #define rsnd_src_of_node(priv) \ of_get_child_by_name(rsnd_priv_to_dev(priv)->of_node, "rcar_sound,src") @@ -233,6 +235,30 @@ int rsnd_src_ssi_irq_disable(struct rsnd_mod *ssi_mod) return 0; } +static u32 rsnd_src_convert_rate(struct rsnd_src *src) +{ + struct rsnd_mod *mod = &src->mod; + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + u32 convert_rate; + + if (!runtime) + return 0; + + if (!rsnd_enable_sync_convert(src)) + return src->convert_rate; + + convert_rate = src->sync.val; + + if (!convert_rate) + convert_rate = src->convert_rate; + + if (!convert_rate) + convert_rate = runtime->rate; + + return convert_rate; +} + unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv, struct rsnd_dai_stream *io, struct snd_pcm_runtime *runtime) @@ -333,6 +359,9 @@ static int rsnd_src_init(struct rsnd_mod *mod, src->err = 0; + /* reset sync convert_rate */ + src->sync.val = 0; + /* * Initialize the operation of the SRC internal circuits * see rsnd_src_start() @@ -356,6 +385,9 @@ static int rsnd_src_quit(struct rsnd_mod *mod, src->convert_rate = 0; + /* reset sync convert_rate */ + src->sync.val = 0; + return 0; } @@ -672,6 +704,7 @@ static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod) struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct rsnd_src *src = rsnd_mod_to_src(mod); u32 convert_rate = rsnd_src_convert_rate(src); + u32 cr, route; uint ratio; int ret; @@ -692,13 +725,21 @@ static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod) if (ret < 0) return ret; - rsnd_mod_write(mod, SRC_SRCCR, 0x00011110); - + cr = 0x00011110; + route = 0x0; if (convert_rate) { - /* Gen1/Gen2 are not compatible */ - rsnd_mod_write(mod, SRC_ROUTE_MODE0, 1); + route = 0x1; + + if (rsnd_enable_sync_convert(src)) { + cr |= 0x1; + route |= rsnd_io_is_play(io) ? + (0x1 << 24) : (0x1 << 25); + } } + rsnd_mod_write(mod, SRC_SRCCR, cr); + rsnd_mod_write(mod, SRC_ROUTE_MODE0, route); + switch (rsnd_mod_id(mod)) { case 5: case 6: @@ -811,6 +852,80 @@ static int rsnd_src_stop_gen2(struct rsnd_mod *mod, return ret; } +static void rsnd_src_reconvert_update(struct rsnd_mod *mod) +{ + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct rsnd_src *src = rsnd_mod_to_src(mod); + u32 convert_rate = rsnd_src_convert_rate(src); + u32 fsrate; + + if (!runtime) + return; + + if (!convert_rate) + convert_rate = runtime->rate; + + fsrate = 0x0400000 / convert_rate * runtime->rate; + + /* update IFS */ + rsnd_mod_write(mod, SRC_IFSVR, fsrate); +} + +static int rsnd_src_pcm_new(struct rsnd_mod *mod, + struct snd_soc_pcm_runtime *rtd) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct rsnd_dai *rdai = rsnd_io_to_rdai(io); + struct rsnd_src *src = rsnd_mod_to_src(mod); + int ret; + + /* + * enable SRC sync convert if possible + */ + + /* + * Gen1 is not supported + */ + if (rsnd_is_gen1(priv)) + return 0; + + /* + * SRC sync convert needs clock master + */ + if (!rsnd_rdai_is_clk_master(rdai)) + return 0; + + /* + * We can't use SRC sync convert + * if it has DVC + */ + if (rsnd_io_to_mod_dvc(io)) + return 0; + + /* + * enable sync convert + */ + ret = rsnd_kctrl_new_s(mod, rtd, + rsnd_io_is_play(io) ? + "SRC Out Rate Switch" : + "SRC In Rate Switch", + rsnd_src_reconvert_update, + &src->sen, 1); + if (ret < 0) + return ret; + + ret = rsnd_kctrl_new_s(mod, rtd, + rsnd_io_is_play(io) ? + "SRC Out Rate" : + "SRC In Rate", + rsnd_src_reconvert_update, + &src->sync, 192000); + + return ret; +} + static struct rsnd_mod_ops rsnd_src_gen2_ops = { .name = SRC_NAME, .dma_req = rsnd_src_dma_req, @@ -821,6 +936,7 @@ static struct rsnd_mod_ops rsnd_src_gen2_ops = { .start = rsnd_src_start_gen2, .stop = rsnd_src_stop_gen2, .hw_params = rsnd_src_hw_params, + .pcm_new = rsnd_src_pcm_new, }; struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id) -- 1.9.1

2 1

[alsa-devel] [PATCH 00/25] treewide: Use bool function return values of true/false not 1/0
by Joe Perches 01 Apr '15

01 Apr '15

Joe Perches (25): arm: Use bool function return values of true/false not 1/0 arm64: Use bool function return values of true/false not 1/0 hexagon: Use bool function return values of true/false not 1/0 ia64: Use bool function return values of true/false not 1/0 mips: Use bool function return values of true/false not 1/0 powerpc: Use bool function return values of true/false not 1/0 s390: Use bool function return values of true/false not 1/0 sparc: Use bool function return values of true/false not 1/0 tile: Use bool function return values of true/false not 1/0 unicore32: Use bool function return values of true/false not 1/0 x86: Use bool function return values of true/false not 1/0 virtio_console: Use bool function return values of true/false not 1/0 csiostor: Use bool function return values of true/false not 1/0 dcache: Use bool function return values of true/false not 1/0 nfsd: nfs4state: Use bool function return values of true/false not 1/0 include/linux: Use bool function return values of true/false not 1/0 sound: Use bool function return values of true/false not 1/0 rcu: tree_plugin: Use bool function return values of true/false not 1/0 sched: Use bool function return values of true/false not 1/0 ftrace: Use bool function return values of true/false not 1/0 slub: Use bool function return values of true/false not 1/0 bridge: Use bool function return values of true/false not 1/0 netfilter: Use bool function return values of true/false not 1/0 security: Use bool function return values of true/false not 1/0 sound: wm5100-tables: Use bool function return values of true/false not 1/0 arch/arm/include/asm/dma-mapping.h | 8 ++-- arch/arm/include/asm/kvm_emulate.h | 2 +- arch/arm/mach-omap2/powerdomain.c | 14 +++--- arch/arm64/include/asm/dma-mapping.h | 2 +- arch/hexagon/include/asm/dma-mapping.h | 2 +- arch/ia64/include/asm/dma-mapping.h | 2 +- arch/mips/include/asm/dma-mapping.h | 2 +- arch/powerpc/include/asm/dcr-native.h | 2 +- arch/powerpc/include/asm/dma-mapping.h | 4 +- arch/powerpc/include/asm/kvm_book3s_64.h | 4 +- arch/powerpc/sysdev/dcr.c | 2 +- arch/s390/include/asm/dma-mapping.h | 2 +- arch/sparc/mm/init_64.c | 8 ++-- arch/tile/include/asm/dma-mapping.h | 2 +- arch/unicore32/include/asm/dma-mapping.h | 2 +- arch/x86/include/asm/archrandom.h | 2 +- arch/x86/include/asm/dma-mapping.h | 2 +- arch/x86/include/asm/kvm_para.h | 2 +- arch/x86/kvm/cpuid.h | 2 +- arch/x86/kvm/vmx.c | 72 ++++++++++++++-------------- drivers/char/virtio_console.c | 2 +- drivers/scsi/csiostor/csio_scsi.c | 4 +- fs/dcache.c | 12 ++--- fs/nfsd/nfs4state.c | 2 +- include/linux/blkdev.h | 2 +- include/linux/ide.h | 2 +- include/linux/kgdb.h | 2 +- include/linux/mfd/db8500-prcmu.h | 2 +- include/linux/mm.h | 2 +- include/linux/power_supply.h | 8 ++-- include/linux/ssb/ssb_driver_extif.h | 2 +- include/linux/ssb/ssb_driver_gige.h | 16 +++---- include/sound/soc.h | 4 +- kernel/rcu/tree_plugin.h | 4 +- kernel/sched/auto_group.h | 2 +- kernel/sched/completion.c | 16 ++++--- kernel/trace/ftrace.c | 10 ++-- mm/slub.c | 12 ++--- net/bridge/br_private.h | 2 +- net/ipv4/netfilter/ipt_ah.c | 2 +- net/netfilter/ipset/ip_set_hash_ip.c | 8 ++-- net/netfilter/ipset/ip_set_hash_ipmark.c | 8 ++-- net/netfilter/ipset/ip_set_hash_ipport.c | 8 ++-- net/netfilter/ipset/ip_set_hash_ipportip.c | 8 ++-- net/netfilter/ipset/ip_set_hash_ipportnet.c | 8 ++-- net/netfilter/ipset/ip_set_hash_net.c | 8 ++-- net/netfilter/ipset/ip_set_hash_netiface.c | 8 ++-- net/netfilter/ipset/ip_set_hash_netport.c | 8 ++-- net/netfilter/ipset/ip_set_hash_netportnet.c | 8 ++-- net/netfilter/xt_connlimit.c | 2 +- net/netfilter/xt_hashlimit.c | 2 +- net/netfilter/xt_ipcomp.c | 2 +- security/apparmor/file.c | 8 ++-- security/apparmor/policy.c | 10 ++-- sound/soc/codecs/wm5100-tables.c | 12 ++--- 55 files changed, 178 insertions(+), 176 deletions(-) -- 2.1.2

3 4

[alsa-devel] [PATCH v3 0/5] ALSA: hda - add hda controller to hda core
by Vinod Koul 01 Apr '15

01 Apr '15

This incorporates the comments from Takashi on 1st patch and uses the change from Takashi's unstable tree for controller and then adds the DSP loader code on top. This is based on sound/topic/hda-core Comments welcome Jeeja KP (1): ALSA: hda: move common hda controller register defines up Vinod Koul (4): ALSA: hda: add HDA_MAX_CODECS ALSA: hda - Add the controller helper codes to hda-core module ALSA: hda - move dsp lock to hdac ALSA: hda - add the DSP loader code include/sound/hda_registers.h | 177 +++++++++++++ include/sound/hdaudio.h | 185 ++++++++++++- sound/hda/Makefile | 3 +- sound/hda/hdac_bus.c | 13 +- sound/hda/hdac_controller.c | 401 ++++++++++++++++++++++++++++ sound/hda/hdac_stream.c | 565 ++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_controller.c | 17 +- sound/pci/hda/hda_controller.h | 171 +----------- 8 files changed, 1347 insertions(+), 185 deletions(-) create mode 100644 include/sound/hda_registers.h create mode 100644 sound/hda/hdac_controller.c create mode 100644 sound/hda/hdac_stream.c -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: rcar: mark device data as constant
by Uwe Kleine-König 01 Apr '15

01 Apr '15

A driver's device data should and can be const. This is a follow-up on commit 33187fb4a203 (ASoC: rsnd: constify of_device_id array) which marked the of_device_id as const. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- This is also patch 4 of my series[1] which degraded to this hunk when rebasing to next because all the other changes were already addressed by Fabian. Best regards Uwe [1] http://thread.gmane.org/gmane.linux.alsa.devel/136241/ sound/soc/sh/rcar/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 3d8e4d60c83d..9f48d75fa992 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -99,11 +99,11 @@ #define RSND_RATES SNDRV_PCM_RATE_8000_96000 #define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) -static struct rsnd_of_data rsnd_of_data_gen1 = { +static const struct rsnd_of_data rsnd_of_data_gen1 = { .flags = RSND_GEN1, }; -static struct rsnd_of_data rsnd_of_data_gen2 = { +static const struct rsnd_of_data rsnd_of_data_gen2 = { .flags = RSND_GEN2, }; -- 2.1.4 -- Pengutronix e.K. | Uwe Kleine-König | Industrial Linux Solutions | http://www.pengutronix.de/ |

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[alsa-devel] [PATCH v2 0/4] ASoC: constification and random cleanups
by Uwe Kleine-König 01 Apr '15

01 Apr '15

Hello, this is the series sent on Sunday rebased on next. There was a conflict with commit c660c0a80586 (ASoC: fsi: constify of_device_id array) which did a part of my 2nd patch but already conflicted with the first one. Mark said he already took patch 3, but that should not be a problem. Patch 4 applied just fine to next, so this is just a resend. Best regards Uwe Uwe Kleine-König (4): ASoC: fsi: reorder code to make a forward declaration superfluous ASoC: fsi: mark several data structures as const ASoC: fsi: fix license specification ASoC: mark of_device_id and device data as constant sound/soc/sh/fsi.c | 71 ++++++++++++++++++++++++------------------------ sound/soc/sh/rcar/core.c | 4 +-- 2 files changed, 37 insertions(+), 38 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH 0/5] Intel SST driver restructure
by Jie Yang 01 Apr '15

01 Apr '15

In order to prepare for the upstreaming of Skylake drivers and to allow reuse of code between different Intel audio DSP products, this patch series restructures the Intel directory as follows: 1. Create "common" directory. Move common files into this directory. 2. Create "boards" directory. Move all machine drivers into this directory. 3. Create "haswell" directory. Move all remaining hsw/bdw code into this directory. 4. Create "baytrail" directory. Move remaining baytrail driver to this directory. 5. Create "atom" directory. Move all MCG platform drivers(including sst/ folder) to this directory. Todo next step: 1. Change and move atom related codes(in atom/sst/) to corresponding and try share as many as possible codes(e.g. sst_acpi.c). 2. Share more common codes about IPC for different platforms, e.g. The methods for queueing and sending/reciving Tx/Rx messages, The methods for mailbox IO, The IPC error return codes, etc. Jie Yang (5): ASoC: Intel: create common folder and move common files in ASoC: Intel: create haswell folder and move haswell platform files in ASoC: Intel: create boards folder and move sst boards files in ASoC: Intel: create baytrail folder and move baytrail platform files in ASoC: Intel: create atom folder and move atom platform files in sound/soc/intel/Makefile | 42 +++------------------- sound/soc/intel/atom/Makefile | 7 ++++ sound/soc/intel/{ => atom}/sst-atom-controls.c | 0 sound/soc/intel/{ => atom}/sst-atom-controls.h | 0 sound/soc/intel/{ => atom}/sst-mfld-dsp.h | 0 .../intel/{ => atom}/sst-mfld-platform-compress.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform.h | 0 sound/soc/intel/{ => atom}/sst/Makefile | 0 sound/soc/intel/{ => atom}/sst/sst.c | 0 sound/soc/intel/{ => atom}/sst/sst.h | 0 sound/soc/intel/{ => atom}/sst/sst_acpi.c | 0 sound/soc/intel/{ => atom}/sst/sst_drv_interface.c | 0 sound/soc/intel/{ => atom}/sst/sst_ipc.c | 0 sound/soc/intel/{ => atom}/sst/sst_loader.c | 0 sound/soc/intel/{ => atom}/sst/sst_pci.c | 0 sound/soc/intel/{ => atom}/sst/sst_pvt.c | 0 sound/soc/intel/{ => atom}/sst/sst_stream.c | 0 sound/soc/intel/baytrail/Makefile | 4 +++ sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c | 4 +-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c | 4 +-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h | 0 sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c | 4 +-- sound/soc/intel/boards/Makefile | 15 ++++++++ sound/soc/intel/{ => boards}/broadwell.c | 6 ++-- sound/soc/intel/{ => boards}/byt-max98090.c | 2 +- sound/soc/intel/{ => boards}/byt-rt5640.c | 4 +-- .../{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} | 0 sound/soc/intel/{ => boards}/cht_bsw_rt5645.c | 0 sound/soc/intel/{ => boards}/cht_bsw_rt5672.c | 0 sound/soc/intel/{ => boards}/haswell.c | 6 ++-- sound/soc/intel/{ => boards}/mfld_machine.c | 0 sound/soc/intel/common/Makefile | 6 ++++ sound/soc/intel/{ => common}/sst-acpi.c | 0 sound/soc/intel/{ => common}/sst-dsp-priv.h | 0 sound/soc/intel/{ => common}/sst-dsp.c | 0 sound/soc/intel/{ => common}/sst-dsp.h | 0 sound/soc/intel/{ => common}/sst-firmware.c | 0 sound/soc/intel/haswell/Makefile | 4 +++ sound/soc/intel/{ => haswell}/sst-haswell-dsp.c | 6 ++-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.c | 4 +-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.h | 0 sound/soc/intel/{ => haswell}/sst-haswell-pcm.c | 6 ++-- 43 files changed, 64 insertions(+), 60 deletions(-) create mode 100644 sound/soc/intel/atom/Makefile rename sound/soc/intel/{ => atom}/sst-atom-controls.c (100%) rename sound/soc/intel/{ => atom}/sst-atom-controls.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-dsp.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-compress.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform.h (100%) rename sound/soc/intel/{ => atom}/sst/Makefile (100%) rename sound/soc/intel/{ => atom}/sst/sst.c (100%) rename sound/soc/intel/{ => atom}/sst/sst.h (100%) rename sound/soc/intel/{ => atom}/sst/sst_acpi.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_drv_interface.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_ipc.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_loader.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_pci.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_pvt.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_stream.c (100%) create mode 100644 sound/soc/intel/baytrail/Makefile rename sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h (100%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c (99%) create mode 100644 sound/soc/intel/boards/Makefile rename sound/soc/intel/{ => boards}/broadwell.c (98%) rename sound/soc/intel/{ => boards}/byt-max98090.c (99%) rename sound/soc/intel/{ => boards}/byt-rt5640.c (98%) rename sound/soc/intel/{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} (100%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5645.c (100%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5672.c (100%) rename sound/soc/intel/{ => boards}/haswell.c (98%) rename sound/soc/intel/{ => boards}/mfld_machine.c (100%) create mode 100644 sound/soc/intel/common/Makefile rename sound/soc/intel/{ => common}/sst-acpi.c (100%) rename sound/soc/intel/{ => common}/sst-dsp-priv.h (100%) rename sound/soc/intel/{ => common}/sst-dsp.c (100%) rename sound/soc/intel/{ => common}/sst-dsp.h (100%) rename sound/soc/intel/{ => common}/sst-firmware.c (100%) create mode 100644 sound/soc/intel/haswell/Makefile rename sound/soc/intel/{ => haswell}/sst-haswell-dsp.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.h (100%) rename sound/soc/intel/{ => haswell}/sst-haswell-pcm.c (99%) -- 1.9.1

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[alsa-devel] Rane SL2/3/4 driver?
by Be 01 Apr '15

01 Apr '15

Hi, I recently published the first version of Crossfade GNU/Linux, a distro for DJing from USB drives on any PC (see http://mixxx.org/forums/viewtopic.php?f=1&t=7031 for details). I just tried it on my friend's Macbook which he uses with a Rane SL4 USB sound card. I realized there is no Linux driver for the current line of sound cards sold for digital vinyl emulation with Serato DJ. These are the Rane SL2, SL3, and SL4 (http://serato.com/hardware/all/mixers-interfaces ; the Denon DS1 was also recently announced but not available yet: http://serato.com/latest/blog/17973/announcing-denon-ds1 ). They range from 4 in/4 out to 10 in/10 out and all have phono preamps built in. There appears to be support for the old and discontinued Rane SL1, which used USB 1.1, but the SL2/3/4 use USB 2.0. Has anyone thought about writing a driver for these?

1 1

[alsa-devel] [RESEND PATCH] MAINTAINERS: change the Atmel audio alsa driver entry
by Nicolas Ferre 31 Mar '15

31 Mar '15

I take over the the maintainship of Atmel alsa drivers from Voice. Thanks for your work! Signed-off-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> Acked-by: Bo Shen <voice.shen(a)atmel.com> Cc: Mark Brown <broonie(a)kernel.org> --- Hi Mark, Here is the MAINTAINERS' change that I sent to the wrong address. Can you please take it with your tree? Thanks, bye. MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 6cb40f8736ee..a915bb124fe7 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1762,7 +1762,7 @@ S: Supported F: drivers/tty/serial/atmel_serial.c ATMEL Audio ALSA driver -M: Bo Shen <voice.shen(a)atmel.com> +M: Nicolas Ferre <nicolas.ferre(a)atmel.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/atmel -- 2.1.3

2 1

[alsa-devel] [PATCH 1/4] ASoC: fsi: reorder code to make a forward declaration superfluous
by Uwe Kleine-König 31 Mar '15

31 Mar '15

Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/sh/fsi.c | 69 +++++++++++++++++++++++++++--------------------------- 1 file changed, 34 insertions(+), 35 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index b87b22e88e43..dab6aa661a22 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1876,7 +1876,40 @@ static void fsi_handler_init(struct fsi_priv *fsi, } } -static struct of_device_id fsi_of_match[]; +static struct fsi_core fsi1_core = { + .ver = 1, + + /* Interrupt */ + .int_st = INT_ST, + .iemsk = IEMSK, + .imsk = IMSK, +}; + +static struct fsi_core fsi2_core = { + .ver = 2, + + /* Interrupt */ + .int_st = CPU_INT_ST, + .iemsk = CPU_IEMSK, + .imsk = CPU_IMSK, + .a_mclk = A_MST_CTLR, + .b_mclk = B_MST_CTLR, +}; + +static struct of_device_id fsi_of_match[] = { + { .compatible = "renesas,sh_fsi", .data = &fsi1_core}, + { .compatible = "renesas,sh_fsi2", .data = &fsi2_core}, + {}, +}; +MODULE_DEVICE_TABLE(of, fsi_of_match); + +static struct platform_device_id fsi_id_table[] = { + { "sh_fsi", (kernel_ulong_t)&fsi1_core }, + { "sh_fsi2", (kernel_ulong_t)&fsi2_core }, + {}, +}; +MODULE_DEVICE_TABLE(platform, fsi_id_table); + static int fsi_probe(struct platform_device *pdev) { struct fsi_master *master; @@ -2072,40 +2105,6 @@ static struct dev_pm_ops fsi_pm_ops = { .resume = fsi_resume, }; -static struct fsi_core fsi1_core = { - .ver = 1, - - /* Interrupt */ - .int_st = INT_ST, - .iemsk = IEMSK, - .imsk = IMSK, -}; - -static struct fsi_core fsi2_core = { - .ver = 2, - - /* Interrupt */ - .int_st = CPU_INT_ST, - .iemsk = CPU_IEMSK, - .imsk = CPU_IMSK, - .a_mclk = A_MST_CTLR, - .b_mclk = B_MST_CTLR, -}; - -static struct of_device_id fsi_of_match[] = { - { .compatible = "renesas,sh_fsi", .data = &fsi1_core}, - { .compatible = "renesas,sh_fsi2", .data = &fsi2_core}, - {}, -}; -MODULE_DEVICE_TABLE(of, fsi_of_match); - -static struct platform_device_id fsi_id_table[] = { - { "sh_fsi", (kernel_ulong_t)&fsi1_core }, - { "sh_fsi2", (kernel_ulong_t)&fsi2_core }, - {}, -}; -MODULE_DEVICE_TABLE(platform, fsi_id_table); - static struct platform_driver fsi_driver = { .driver = { .name = "fsi-pcm-audio", -- 2.1.4

2 6

[alsa-devel] [PATCH] ASoC: pcm512x: Do not redefine DIV_ROUND_CLOSEST_ULL
by Fabio Estevam 31 Mar '15

31 Mar '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> DIV_ROUND_CLOSEST_ULL is already defined at <linux/kernel.h>, so use the definition from this header file rather than re-defining it locally. This fixes the following build warning: sound/soc/codecs/pcm512x.c:34:0: warning: "DIV_ROUND_CLOSEST_ULL" redefined [enabled by default] In file included from include/linux/list.h:8:0, from include/linux/module.h:9, from sound/soc/codecs/pcm512x.c:19: include/linux/kernel.h:110:0: note: this is the location of the previous definition Reported-by: Olof's autobuilder <build(a)lixom.net> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/pcm512x.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 5a30fdd..e12764d 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -18,6 +18,7 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/clk.h> +#include <linux/kernel.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> @@ -31,8 +32,6 @@ #define DIV_ROUND_DOWN_ULL(ll, d) \ ({ unsigned long long _tmp = (ll); do_div(_tmp, d); _tmp; }) -#define DIV_ROUND_CLOSEST_ULL(ll, d) \ - ({ unsigned long long _tmp = (ll)+(d)/2; do_div(_tmp, d); _tmp; }) #define PCM512x_NUM_SUPPLIES 3 static const char * const pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { -- 1.9.1

2 1

Re: [alsa-devel] [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC.
by Mark Brown 31 Mar '15

31 Mar '15

On Mon, Mar 30, 2015 at 08:16:24PM -0700, Scott Branden wrote: The CC list for this patch is pretty wide - please look at who you're sending this to and try to send to only relevant people (for example I'm not sure the Raspberry Pi people need to review this). People working upstream get a lot of mail so it's helpful to avoid filling their inboxes with random irrelevant stuff. > sound/soc/bcm/Kconfig | 11 + > sound/soc/bcm/Makefile | 5 +- > sound/soc/bcm/cygnus-pcm.c | 918 +++++++++++++++++++++++++ > sound/soc/bcm/cygnus-pcm.h | 45 ++ > sound/soc/bcm/cygnus-ssp.c | 1613 ++++++++++++++++++++++++++++++++++++++++++++ > sound/soc/bcm/cygnus-ssp.h | 84 +++ > 6 files changed, 2675 insertions(+), 1 deletion(-) Send the DMA and DAI drivers as separate patches please, it makes review easier. > +config SND_SOC_CYGNUS > + tristate "SoC platform audio for Broadcom Cygnus chips" > + depends on ARCH_BCM_CYGNUS || COMPILE_TEST > + default ARCH_BCM_CYGNUS Remove the default setting here - we don't do this for other drivers, we shouldn't do it here. > +/* > + * PERIOD_BYTES_MIN is the number of bytes to at which the interrupt will tick. > + * This number should be a multiple of 256 > + */ > +#define PERIOD_BYTES_MIN 0x100 This sounds like it's a setting rather than actually the minimum? > +static const struct snd_pcm_hardware cygnus_pcm_hw = { > + .info = SNDRV_PCM_INFO_MMAP | > + SNDRV_PCM_INFO_MMAP_VALID | > + SNDRV_PCM_INFO_INTERLEAVED, The DMA controller is integrated into the IP? > +static int enable_count; This looks bogus - why is this a global variable not part of the device struct and if it does need to be global why does it need no locking? > + if (aio->portnum == 0) > + *p_rbuf = RINGBUF_REG_PLAYBACK(0); > + else if (aio->portnum == 1) > + *p_rbuf = RINGBUF_REG_PLAYBACK(2); > + else if (aio->portnum == 2) > + *p_rbuf = RINGBUF_REG_PLAYBACK(4); > + else if (aio->portnum == 3) > + *p_rbuf = RINGBUF_REG_PLAYBACK(6); /*SPDIF */ > + else > + status = -EINVAL; This looks like a switch statement, there are many places in the code where you're writing switch statements as chains of ifs. > +static void ringbuf_inc(void __iomem *audio_io, bool is_playback, > + const struct ringbuf_regs *p_rbuf) > +{ > + u32 regval, endval, active_ptr; > + > + if (is_playback) > + active_ptr = p_rbuf->wraddr; > + else > + active_ptr = p_rbuf->rdaddr; > + > + endval = readl(audio_io + p_rbuf->endaddr); > + regval = readl(audio_io + active_ptr); > + regval = regval + p_rbuf->period_bytes; > + if (regval > endval) > + regval -= p_rbuf->buf_size; > + > + writel(regval, audio_io + active_ptr); > +} So it looks like we're getting an interrupt per period and we have to manually advance to the next one? > +static irqreturn_t cygnus_dma_irq(int irq, void *data) > +{ > + u32 r5_status; > + struct cygnus_audio *cygaud; > + > + cygaud = (struct cygnus_audio *)data; If you need to cast away from void something is very wrong. > + /* > + * R5 status bits Description > + * 0 ESR0 (playback FIFO interrupt) > + * 1 ESR1 (playback rbuf interrupt) > + * 2 ESR2 (capture rbuf interrupt) > + * 3 ESR3 (Freemark play. interrupt) > + * 4 ESR4 (Fullmark capt. interrupt) > + */ > + r5_status = readl(cygaud->audio + INTH_R5F_STATUS_OFFSET); > + > + /* If playback interrupt happened */ > + if (ANY_PLAYBACK_IRQ & r5_status) > + handle_playback_irq(cygaud); > + > + /* If capture interrupt happened */ > + if (ANY_CAPTURE_IRQ & r5_status) > + handle_capture_irq(cygaud); > + > + /* > + * clear r5 interrupts after servicing them to avoid overwriting > + * esr_status > + */ > + writel(r5_status, cygaud->audio + INTH_R5F_CLEAR_OFFSET); This feels racy especially given that we seem to need every interrupt delivering. What if another period happens during the servicing? I don't understand what "overwriting esr_status" means here. > + return IRQ_HANDLED; If neither playback nor capture interrupts were flagged we should return IRQ_NONE. > +/* > + * This code is identical to what is done by the framework, when we do not > + * supply a 'copy' function. Having our own copy hook in place allows for > + * us to easily add some diagnotics when needed. > + */ > +int cygnus_pcm_copy(struct snd_pcm_substream *substream, int channel, > + snd_pcm_uframes_t pos, > + void __user *buf, snd_pcm_uframes_t count) This is obviously not suitable for mainline - "let's just cut'n'paste the shared code in case we want to add diagnostics in future" does not scale, you can always add local diagnostics in the core code. > +}; > +/* Blank line between these two please. Missing blanks between bits of code seem to be a common issue in this driver. > +static int audio_ssp_in_enable(struct cygnus_aio_port *aio, bool enable) > +{ > + u32 value; > + > + if (enable) { > + } else { Why not just write two functions? > +static int audio_ssp_source_bitres(struct cygnus_aio_port *aio, unsigned bits) > +{ > + u32 mask = 0x1f; > + u32 value = 0; > + > + if ((bits == 8) || (bits == 16) || (bits == 32)) { > + value = readl(aio->audio + aio->regs.bf_sourcech_cfg); > + value &= ~(mask << BF_SRC_CFGX_BIT_RES); > + > + /* 32 bit mode is coded as 0 */ > + if ((bits == 8) || (bits == 16)) > + value |= (bits << BF_SRC_CFGX_BIT_RES); > + > + writel(value, aio->audio + aio->regs.bf_sourcech_cfg); > + return 0; > + } > + > + pr_err("Bit resolution not supported %d\n", bits); > + return -EINVAL; dev_err() (this applies throughout the driver). It's not clear why this is a function and not just written as a single statement in the one place that it's used (there are multiple calls but they're all together and could just be inlined). > + if (!aio->bitrate) { > + pr_err("%s Use .set_clkdiv() to set bitrate\n", __func__); > + return -EINVAL; > + } This seems bogus - why are we enforcing the use of set_clkdiv()? Can't we figure out something esneible? > + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { > + /* Set the SSP up as slave */ > + case SND_SOC_DAIFMT_CBM_CFM: The comments here look odd due to the indentation and really aren't adding much anyway. > + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { > + case SND_SOC_DAIFMT_RIGHT_J: > + case SND_SOC_DAIFMT_LEFT_J: > + return -EINVAL; These are just eaten by the default case. > + case SND_SOC_DAIFMT_DSP_A: > + case SND_SOC_DAIFMT_DSP_B: > + ssp_newcfg |= BIT(I2S_OUT_CFGX_TDM_MODE); > + > + /* DSP_A = data after FS, DSP_B = data during FS */ > + if (SND_SOC_DAIFMT_DSP_A) > + ssp_newcfg |= BIT(I2S_OUT_CFGX_DATA_ALIGNMENT); That if statement isn't doing what was intended... > + } else { > + > + switch (cmd) { > + case SNDRV_PCM_TRIGGER_START: > + audio_ssp_in_enable(aio, 1); > + break; > + > + case SNDRV_PCM_TRIGGER_STOP: > + audio_ssp_in_enable(aio, 0); > + break; > + } We can just ignore other triggers? It's not clear why this is different to the playback case. > +int cygnus_ssp_get_mode(struct snd_soc_dai *cpu_dai) > +{ > + struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai); > + > + return aio->mode; > +} > +EXPORT_SYMBOL(cygnus_ssp_get_mode); What is this for, it's setting off alarm bells? Note also that ASoC is _GPL() only. > +static const struct snd_kcontrol_new pll_tweak_controls[] = { > + SOC_SINGLE_EXT("PLL Tweak", 0, 0, PLL_NDIV_FRACT_MAX, 0, > + pll_tweak_get, pll_tweak_put), > +}; Whatever this control is doing it should be a separate patch (as I think we discussed in person a ELC?), it's clearly something that's unusual and is likely to block the rest of the code as a result. At a high level this needs at least some documentation. > +int cygnus_ssp_add_pll_tweak_controls(struct snd_soc_pcm_runtime *rtd) > +{ > + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; > + > + return snd_soc_add_dai_controls(cpu_dai, > + pll_tweak_controls, > + ARRAY_SIZE(pll_tweak_controls)); > +} > +EXPORT_SYMBOL(cygnus_ssp_add_pll_tweak_controls); Again, why is this being exported and why is it not _GPL? If the driver is adding controls I'd expect it to just add its controls itself. > +static int cygnus_ssp_probe(struct platform_device *pdev) > +{ > + struct device *dev = &pdev->dev; > + struct device_node *child_node; > + struct resource *res = pdev->resource; > + struct cygnus_audio *cygaud; > + int err = -EINVAL; > + int node_count; > + int active_port_count; > + > + if (!of_match_device(cygnus_ssp_of_match, dev)) { > + dev_err(dev, "Failed to find ssp controller\n"); > + return -ENODEV; > + } This error message is misleading - you mean to say that the driver got loaded for a device it doesn't understand. > + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > + cygaud->audio = devm_ioremap_resource(dev, res); > + if (IS_ERR(cygaud->audio)) { > + dev_err(dev, "audio_io ioremap failed\n"); > + return PTR_ERR(cygaud->audio); devm_ioremap_resource() will complain for you, and in general you should be printing error codes. > + node_count = 0; This doesn't seem to be needed? > + node_count = of_get_child_count(pdev->dev.of_node); > + if ((node_count < 1) || (node_count > CYGNUS_MAX_PORTS)) { > + dev_err(dev, "Incorrct number of child nodes\n"); > + return -EINVAL; Spelling mistake there and it would be helpful to the user to tell them what we parsed.

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Re: [alsa-devel] [pulseaudio-discuss] Can't get Pulseadio to playback stream on Raspbian with HifiBerry DAC+
by Raymond Yau 31 Mar '15

31 Mar '15

2015-3-30 上午12:22 於 "David Martinez" <david(a)sojournmusic.net> 寫道： > > Label says PiFi DAC+. v2.0 > > I'll need to look up how to apply the patch. I'm not familiar with how to do it. This look like driver bug since capabilities should be pvolume instead of volume if your sound card are half duplex Simple mixer control 'PCM',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 215 Front Left: 172 [80%] Front Right: 172 [80%] Simple mixer control 'Playback',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 1 Front Left: 0 [0%] [-6.00dB] Front Right: 0 [0%] [-6.00dB] Simple mixer control 'Playback Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 1 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] > >> >> > >> > I'll check HiFiBerry's site and see if I can get it. >> > >> >> https://www.hifiberry.com/dacplus/ >> >> What is the label on the black DAC chip ? >> >> Aplay/Arecord output >> !!-------------------- >> >> APLAY >> >> **** List of PLAYBACK Hardware Devices **** >> card 0: sndrpihifiberry [snd_rpi_hifiberry_dacplus], device 0: HiFiBerry DAC HiFi pcm512x-hifi-0 [] >> Subdevices: 1/1 >> Subdevice #0: subdevice #0 >> >> ARECORD >> >> **** List of CAPTURE Hardware Devices **** >> >> Seem half duplex >> >>

1 1

[alsa-devel] [RFC v2][PATCH 00/11] digi00x: new driver for Digidesign 002/003 family
by Takashi Sakamoto 31 Mar '15

31 Mar '15

This patchset is revised version of my previous set. [alsa-devel] [RFC][PATCH 00/11] digi00x: new driver for Digidesign 002/003 family http://mailman.alsa-project.org/pipermail/alsa-devel/2015-March/089141.html Improvements from the previous set: * improve double-oh-three (DOT) application * extend hwdep interface for asynchronous messaging * limit sampling rate for PCM substream when clock source is external or any PCM substreams are running * support s16 samples for PCM playback In this time, I test PCM functionality with my test programs. As a result, I can confirm that S/PDIF and ADAT output from my Digi 002 Rack includes expected PCM samples as my programs generates. I think we can judge the implementation of DOT is proper for Digi 002 rack. Currently, several issues remains: * The port for MIDI machine control message is not tested yet, because 002/003 console model are required. * When allocates 2 or more channel numbers for the device, after 15 to 20 seconds from playbacking, any PCM samples causes noisy sound. Then, all of LED on the front panel light. The streaming still continues correctly. * The actual effects of external clock source is not clear. When set the clock source is somewhat external, even if stopping the clock source, the device continues to sound PCM samples against my expectation. * The meaning of asynchronous messages is unknown. This patchset adds a functionality to receive it in userspace. You can test it with updated libhinawa sample script. https://github.com/takaswie/libhinawa Well, I'll stop my work for this driver in this developing period (for Linux 4.0), because this work is not in my original plan. I hope any users test this driver and report the result till Linux 4.2 merge window. If not, this driver will not be merged into upstream, naturally. I prepare for DKMS package to test this patchset. Please use it if needed. https://github.com/takaswie/snd-firewire-improve Takashi Sakamoto (11): ALSA: digi00x: add skeleton for Digi 002/003 device driver ALSA: digi00x: add streaming functionality ALSA: digi00x: add proc node for clock status ALSA: digi00x: add PCM functionality ALSA: digi00x: add MIDI functionality ALSA: digi00x: add hwdep interface ALSA: digi00x: support unknown asynchronous message ALSA: digi00x: support MIDI ports for device control ALSA: firewire-lib: allows to implement external MIDI callback function ALSA: digi00x: improve MIDI capture/playback ALSA: digi00x: apply double-oh-three algorithm to multiplex PCM samples include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 8 + sound/firewire/Kconfig | 10 + sound/firewire/Makefile | 1 + sound/firewire/amdtp.c | 30 ++- sound/firewire/amdtp.h | 5 + sound/firewire/digi00x/Makefile | 3 + sound/firewire/digi00x/digi00x-hwdep.c | 200 +++++++++++++++ sound/firewire/digi00x/digi00x-midi.c | 199 +++++++++++++++ sound/firewire/digi00x/digi00x-pcm.c | 344 +++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-proc.c | 86 +++++++ sound/firewire/digi00x/digi00x-protocol.c | 406 ++++++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-stream.c | 393 +++++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x.c | 179 +++++++++++++ sound/firewire/digi00x/digi00x.h | 167 ++++++++++++ 15 files changed, 2026 insertions(+), 8 deletions(-) create mode 100644 sound/firewire/digi00x/Makefile create mode 100644 sound/firewire/digi00x/digi00x-hwdep.c create mode 100644 sound/firewire/digi00x/digi00x-midi.c create mode 100644 sound/firewire/digi00x/digi00x-pcm.c create mode 100644 sound/firewire/digi00x/digi00x-proc.c create mode 100644 sound/firewire/digi00x/digi00x-protocol.c create mode 100644 sound/firewire/digi00x/digi00x-stream.c create mode 100644 sound/firewire/digi00x/digi00x.c create mode 100644 sound/firewire/digi00x/digi00x.h -- 2.1.0

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Re: [alsa-devel] [PATCH 00/25] treewide: Use bool function return values of true/false not 1/0
by Joe Perches 31 Mar '15

31 Mar '15

On Mon, 2015-03-30 at 17:07 -0700, Casey Schaufler wrote: > On 3/30/2015 4:45 PM, Joe Perches wrote: > > Joe Perches (25): > > arm: Use bool function return values of true/false not 1/0 [etc...] > Why, and why these in particular? bool functions are probably better returning bool values instead of 1 and 0. Especially when the functions intermix returning returning 1/0 and true/false. (there are only a couple of those though) These are all the remaining instances in the kernel tree.

1 0

Re: [alsa-devel] Realtek ALC662 issues - Intel DG41CN
by Raymond Yau 31 Mar '15

31 Mar '15

> > > Well, I was just trying different modes. The last one was ASUS-MODE1. Before this I tried the "laptop" mode and the "generic" one, both of which did not work. > > What do you suggest? You don't need any model since the pin defaults of your Intel DG41CN are correct unless you want to use CD audio in connector !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x14 0x01014010 0x15 0x411111f0 0x16 0x411111f0 0x18 0x01a19840 0x19 0x02a19c50 0x1a 0x0181304f 0x1b 0x02214c20 0x1c 0x593301f0 0x1d 0x4004c601 0x1e 0x01441130 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN http://www.intel.com/support/motherboards/desktop/dg41cn/sb/CS-031343.htm You should check the front audio panel of your computer chassis if jack state of headphone is alway plugged http://www.intel.com/support/motherboards/desktop/sb/cs-015851.htm http://fedoraproject.org/wiki/How_to_debug_PulseAudio_problems You have to provide pulseaudio verbose log Default setting support multi channels but NO headphone playback volume control http://www.intel.com/support/motherboards/desktop/sb/CS-034198.htm You can disable multi channels by using early patching and hint multi_io=0 to get back the headphone playback volume control - multi_io (bool): try probing multi-I/O config (e.g. shared line-in/surround, mic/clfe jacks) https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… >> >> > I have recently installed Fedora 21 Mate on my desktop which runs on >> Intel DG41CN motherboard with Realtek ALC662 Audio. Prior to this I >> was running Linux Mint. It was working all fine then, I may have fixed >> it when I had installed Mint two years ago but I do not remember what >> was done then. I also run Windows 7 on the same machine and there is >> no problem with the audio. >> >> > So the problem I have is that there is no sound from my speakers. A >> few times, there is a little "choppy" sound but thats it. When I put >> the headphones on, the sound is there. >> >> > I must also mention that I also installed Ubuntu Mate on it and faced >> the same problem. >> >> > I have run the Alsa diagnostic tool and the output is uploaded here: >> http://www.alsa-project.org/db/?f=1f6dd98e26f68e890258fa2fe5eb08ee343c5c13 >> >> >I will really appreciate any help. I am stuck bad here.. >> >> Any reason to use asus-mode1 ? >> >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >> >> Seem asus-mode1 is a broken pin fixup since node 0x21 is not any pin complex >> >> Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono >> >> + [ALC662_FIXUP_ASUS_MODE1] = { >> + .type = ALC_FIXUP_PINS, >> + .v.pins = (const struct alc_pincfg[]) { >> + { 0x14, 0x99130110 }, /* speaker */ >> + { 0x18, 0x01a19c20 }, /* mic */ >> + { 0x19, 0x99a3092f }, /* int-mic */ >> + { 0x21, 0x0121401f }, /* HP out */ >> + { } >> + }, >> + .chained = true, >> + .chain_id = ALC662_FIXUP_SKU_IGNORE >> + }, >> >> Line out is muted when headphone jack is plugged >> >> control.30 { >> iface CARD >> name 'Front Headphone Jack' >> value true >> comment { >> access read >> type BOOLEAN >> count 1 >> } >> } >> >> Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Control: name="Speaker Playback Switch", index=0, device=0 >> ControlAmp: chs=3, dir=Out, idx=0, ofs=0 >> Control: name="Speaker Phantom Jack", index=0, device=0 >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0001003c: IN OUT HP EAPD Detect >> EAPD 0x2: EAPD >> Pin Default 0x01014010: [Jack] Line Out at Ext Rear >> Conn = 1/8, Color = Green >> DefAssociation = 0x1, Sequence = 0x0 >> Pin-ctls: 0x00: >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x0c > >

1 0

Re: [alsa-devel] Realtek ALC662 issues - Intel DG41CN
by Raymond Yau 30 Mar '15

30 Mar '15

> I have recently installed Fedora 21 Mate on my desktop which runs on Intel DG41CN motherboard with Realtek ALC662 Audio. Prior to this I was running Linux Mint. It was working all fine then, I may have fixed it when I had installed Mint two years ago but I do not remember what was done then. I also run Windows 7 on the same machine and there is no problem with the audio. > So the problem I have is that there is no sound from my speakers. A few times, there is a little "choppy" sound but thats it. When I put the headphones on, the sound is there. > I must also mention that I also installed Ubuntu Mate on it and faced the same problem. > I have run the Alsa diagnostic tool and the output is uploaded here: http://www.alsa-project.org/db/?f=1f6dd98e26f68e890258fa2fe5eb08ee343c5c13 >I will really appreciate any help. I am stuck bad here.. Any reason to use asus-mode1 ? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Seem asus-mode1 is a broken pin fixup since node 0x21 is not any pin complex Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono + [ALC662_FIXUP_ASUS_MODE1] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x99130110 }, /* speaker */ + { 0x18, 0x01a19c20 }, /* mic */ + { 0x19, 0x99a3092f }, /* int-mic */ + { 0x21, 0x0121401f }, /* HP out */ + { } + }, + .chained = true, + .chain_id = ALC662_FIXUP_SKU_IGNORE + }, Line out is muted when headphone jack is plugged control.30 { iface CARD name 'Front Headphone Jack' value true comment { access read type BOOLEAN count 1 } } Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x0c

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Re: [alsa-devel] baytrail-pcm-audio: audio broken on Lenovo Thinkpad Tablet 10
by Sébastien Bourdeauducq 30 Mar '15

30 Mar '15

Hi, On 02/23/2015 08:58 PM, Jarkko Nikula wrote: >> Is there a way I can try sound/soc/codecs/rt5670.c instead? I edited it >> so that it attaches to my ACPI ID (and rt5640 doesn't), but that did not >> work. I noticed that the kernel has some Baytrail-specific modules for >> RT564x, and I guess those would need to be changed as well - but it's >> not straightforward. >> > Basically three changes are needed: > > 1. Hack add "10EC5640" to rt5670.c as you have done > 2. Add a new machine driver sound/soc/intel/sst-acpi.c/byt-rt5670 > I guess cht_bsw_rt5672.c can be used as a starting point for audio > routes and byt-rt5640.c for dai link definitions.mess > 3. Hack change sound/soc/intel/sst-acpi.c so that "10EC5640" loads > the new machine driver instead of byt-rt5640. Thanks. I did that and I get the following in the kernel log: [ 6.831422] byt-rt5670 byt-rt5670: ASoC: CPU DAI baytrail-pcm-audio not regis tered [ 6.831443] byt-rt5670 byt-rt5670: snd_soc_register_card failed -517 [ 6.831453] platform byt-rt5670: Driver byt-rt5670 requests probe deferral [ 6.884013] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [ 6.884023] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed -22 Note: it seems to me that all Baytrail machines will have this "error: invalid DMA engine" message. AFAICT it is emitted from sst-firmware.c because sst->pdata->dma_engine is 0 and in sst-acpi.c the function sst_acpi_probe does not set it for Baytrail because resindex_dma_base is -1 in sst_acpi_baytrail_desc. [ 6.924094] baytrail-pcm-audio baytrail-pcm-audio: FW version: 04.05.13.a0 [ 6.924106] baytrail-pcm-audio baytrail-pcm-audio: Build type: a0 [ 6.924109] baytrail-pcm-audio baytrail-pcm-audio: Build date: Apr 2 2014 14:14:39 [ 6.931503] byt-rt5670 byt-rt5670: rt5670-aif1 <-> baytrail-pcm-audio mapping ok [ 6.931624] byt-rt5670 byt-rt5670: ASoC: no source widget found for ssp2 Tx [ 6.931629] byt-rt5670 byt-rt5670: ASoC: Failed to add route ssp2 Tx -> direct -> AIF1 Playback [ 6.931636] byt-rt5670 byt-rt5670: ASoC: no source widget found for codec_out0 [ 6.931640] byt-rt5670 byt-rt5670: ASoC: Failed to add route codec_out0 -> direct -> ssp2 Tx [ 6.931647] byt-rt5670 byt-rt5670: ASoC: no source widget found for codec_out1 [ 6.931650] byt-rt5670 byt-rt5670: ASoC: Failed to add route codec_out1 -> direct -> ssp2 Tx [ 6.931656] byt-rt5670 byt-rt5670: ASoC: no source widget found for ssp2 Rx [ 6.931660] byt-rt5670 byt-rt5670: ASoC: Failed to add route ssp2 Rx -> direct -> codec_in0 [ 6.931666] byt-rt5670 byt-rt5670: ASoC: no source widget found for ssp2 Rx [ 6.931670] byt-rt5670 byt-rt5670: ASoC: Failed to add route ssp2 Rx -> direct -> codec_in1 [ 6.931677] byt-rt5670 byt-rt5670: ASoC: no sink widget found for ssp2 Rx [ 6.931680] byt-rt5670 byt-rt5670: ASoC: Failed to add route AIF1 Capture -> direct -> ssp2 Rx My byt-rt5670 is there: https://gist.github.com/sbourdeauducq/8c5da8ac0da13e21ee31 I have to say that I am unfamiliar with Intel SST and I'm don't quite know what I am doing there. I wish sound devices were straightforward DMA-to-DAC devices instead of this complicated DSP architecture which AFAICT boils down to a glorified memcpy/resampler under Linux (this seems to be a pattern with this SoC though; I had a look at how to get the MIPI cameras to work on this machine and it's an even worse mess). > However, above changes may not make audio working since there has been > couple other new Baytrail based machines with correct codec but audio > seems to not move between the DSP and codec chip. Despite the error messages, the audio device is created and userspace applications can send data into it, but no sound is coming out. I also have to say that the settings in alsamixer are a terrible mess which seems not to originate from my hacks. > One assumption is if they use another SSP port than SSP2. Unfortunately > it's difficult to test since Linux DSP FW is hard-coded for SSP2. If only there were no DSP in the first place... Sébastien

1 0

[alsa-devel] [PATCH 0/5 v2] ASoC: rsnd: DPCM base sampling rate convert support
by Kuninori Morimoto 30 Mar '15

30 Mar '15

Hi Mark These are v2 of remains of my previous patch set. Basically, these doesn't need soc-core's new feature, but it is assuming DPCM base sampling rate convert has both playback/capture in same time. 1 - 2: new Renesas sound card for DPCM 3 : bug fix patch. it can be applied in linus/master branch 4 - 5: new feature, but based on 3) patch Kuninori Morimoto (5): 1) ASoC: rsrc-card: add Renesas sampling rate convert sound card support 2) ASoC: rsrc-card: add .be_hw_params_fixup support for convert rate 3) ASoC: rsnd: call clk_prepare/unprepare() in probe/remove 4) ASoC: rsnd: remove useless debug message 5) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 67 +++++++++++++ sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 - sound/soc/sh/rcar/core.c | 35 ++++++- sound/soc/sh/rcar/dma.c | 4 - sound/soc/sh/rcar/dvc.c | 31 +++--- sound/soc/sh/rcar/gen.c | 6 -- sound/soc/sh/rcar/rsnd.h | 19 +++- sound/soc/sh/rcar/rsrc-card.c | 512 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/src.c | 93 +++++++++++------- sound/soc/sh/rcar/ssi.c | 36 ++++--- 12 files changed, 729 insertions(+), 86 deletions(-) _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

3 7

[alsa-devel] Atomic operations not multi-thread aware
by Joël Krähemann 30 Mar '15

30 Mar '15

Hi, please take a a look at gsequencer.org especially at: libags-audio * http://gsequencer.org/api/ags/ags-AgsPort.html gsequencer * http://gsequencer.org/api/ags/ags-AgsLineMember.html I can crash my application while modifying values of the LADSPA plugin assigned dial widget what uses atomic operations and a task engine ... Where can me read about state of Atomic Operations of linux or glibc? Me doing further investigation. bests Joël

2 2

[alsa-devel] [PATCH v2 0/2] ALSA: jack: Refactoring for jack kctls
by Jie Yang 30 Mar '15

30 Mar '15

Currently only hda will create kctls for hda jacks, for asoc, people may need create jack kctls in specific driver. Here we are introducing kctls for each jack, by creating kctls and add them to jack controls list (considering exist of combo jack). At the same time, we will report events for each control in the list. With this implement, we are no longer need hda jack specific kctls, so here also remove jack kctls for hda. Any comments are welcome. Jie Yang (2): ALSA: jack: create jack kcontrols for every jack input device ALSA: hda - Remove jack kctls include/sound/jack.h | 8 ++++ sound/core/Kconfig | 3 -- sound/core/Makefile | 3 +- sound/core/jack.c | 99 +++++++++++++++++++++++++++++++++++++++++++++++- sound/pci/hda/Kconfig | 10 +---- sound/pci/hda/hda_jack.c | 45 ++-------------------- sound/pci/hda/hda_jack.h | 3 -- 7 files changed, 110 insertions(+), 61 deletions(-) -- 1.9.1

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Re: [alsa-devel] getting the sound to work on my asus m6s00v
by Raymond Yau 28 Mar '15

28 Mar '15

2015-3-28 下午5:52 於 "Christian Gleerup" <cromozon(a)swn.nu> 寫道： > > Trying again with my previous kernel since the new kernel seems broken regarding loading the sound module > > nothin plugged in > ../hdajacksensetest/hdajacksensetest -a; for i in 0x14 0x15 0x15; do ./hda-verb /dev/snd/hwC0D0 $i SET_PIN_SENSE 0 nid ;./hda-verb /dev/snd/hwC0D0 $i GET_PIN_SENSE 0 nid; done > Pin 0x14 (Green Headphone, Rear side): present = No > Pin 0x15 (Not connected): present = Yes > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Pink Mic, Rear side): present = No > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = No > Pin 0x1b (Not connected): present = No > Pin 0x1c (Internal CD, Mobile-In): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Green SPDIF Out, Rear side): present = No > Pin 0x1f (Not connected): present = No > nid = 0x14, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x14, verb = 0xf09, param = 0x0 > value = 0x7fffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0xffffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0xffffffff > > Headphone plugged in > ../hdajacksensetest/hdajacksensetest -a; for i in 0x14 0x15 0x15; do ./hda-verb /dev/snd/hwC0D0 $i SET_PIN_SENSE 0 nid ;./hda-verb /dev/snd/hwC0D0 $i GET_PIN_SENSE 0 nid; done > Pin 0x14 (Green Headphone, Rear side): present = No > Pin 0x15 (Not connected): present = No > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Pink Mic, Rear side): present = No > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = No > Pin 0x1b (Not connected): present = No > Pin 0x1c (Internal CD, Mobile-In): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Green SPDIF Out, Rear side): present = No > Pin 0x1f (Not connected): present = No > nid = 0x14, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x14, verb = 0xf09, param = 0x0 > value = 0x7fffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0xffffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0xffffffff 0xffffffff = 0x80000000 + 0x7ffffff Try the modified version of hda_jack_sense_test.py which need hda_analyzer 's codec parser and alsa python https://bugzilla.kernel.org/show_bug.cgi?id=66621#c39 it need time for the impedance measurement to complete and become stable, You may need to increase time.sleep(0.02) if you still get 0xffffffff https://bugzilla.kernel.org/show_bug.cgi?id=66621#c41 # the hda_codec.py file comes from Jaroslav Kysela's hda_analyzer program. from hda_codec import * import time def parseoptions(): from optparse import OptionParser parser = OptionParser() parser.add_option("-c", "--card", dest="cardindex", default=0, metavar="CARD", help="card index (as can be seen in /proc/asound/cards)") parser.add_option("-i", "--codec", dest="codecindex", default=0, metavar="CODEC", help="codec index (as can be seen in /proc/asound/cardX/codecY)") (options, args) = parser.parse_args() return int(options.cardindex), int(options.codecindex) cardindex, codecindex = parseoptions() codec = HDACodec(cardindex, codecindex) codec.analyze() for nid in codec.nodes: node = codec.get_node(nid) if WIDGET_TYPE_IDS[node.wtype] == 'PIN': pd = " " impedance = " " if ((node.pincaps & 4) == 4): if (node.pincaps & 2): codec.rw(nid, VERBS['SET_PIN_SENSE'], 0) time.sleep(0.02) sense = codec.rw(nid, VERBS['GET_PIN_SENSE'], 0) if (node.pincaps & 1): impedance = "Imp=%d" % (sense & 0x7fffffff) if (sense & 0x80000000): pd = "PD=1" else: pd = "PD=0" print "Pin 0x%.2x [%s] %s %s at %s (%s) %s : %s %s" % (nid, node.jack_conn_name, node.jack_type_name, node.jack_location_name, node.jack_location2_name, node.jack_color_name, node.defcfg_misc , pd, impedance) > > Mic plugged in > ../hdajacksensetest/hdajacksensetest -a; for i in 0x14 0x15 0x15; do ./hda-verb /dev/snd/hwC0D0 $i SET_PIN_SENSE 0 nid ;./hda-verb /dev/snd/hwC0D0 $i GET_PIN_SENSE 0 nid; done > Pin 0x14 (Green Headphone, Rear side): present = Yes > Pin 0x15 (Not connected): present = No > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Pink Mic, Rear side): present = No > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = Yes > Pin 0x1b (Not connected): present = No > Pin 0x1c (Internal CD, Mobile-In): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Green SPDIF Out, Rear side): present = No > Pin 0x1f (Not connected): present = No > nid = 0x14, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x14, verb = 0xf09, param = 0x0 > value = 0xffffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0x7fffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0x7fffffff > > > > > >

1 0

[alsa-devel] getting the sound to work on my asus m6s00v
by Christian Gleerup 28 Mar '15

28 Mar '15

I had some problems getting the sound to work on my asus m6s00v (I think that is the model, it is a bit worn so I can't really read it) the symptoms was * on the integrated speakers, no sound at all * on the headphone speakers, sound for a very very short period that faded into nothing. with help from debianuser on freenode#alsa we managed to get it working. the option that i had to load the module with is the following: sudo modprobe snd-hda-intel model=3stack-digout the status now (perfect)... * sound on integrated speaker. * sound in headphones (disables integrated speaker output which is fine) * mic works, both for direct playback and recording * when recording it is the 'digital' channel that controls the volume of the recording level, all others Notice: I don't have OSS or pulseaudio installed. I am thinking it would be really nice if the driver automatically could detect that this option is required. this is the output from the alsa-info script http://www.alsa-project.org/db/?f=28b98a3bd1efc6a80dbd5c54b52c4472d8ea5496 let me know if I can do anything to help with this. Regards.

3 18

[alsa-devel] [PATCH RFT] ASoC: max98925: Fix bit-width 24 settings in max98925_dai_hw_params
by Axel Lin 27 Mar '15

27 Mar '15

Trivial typo fix. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Anish, This looks like a typo, can you confirm if this patch is correct. Thanks. sound/soc/codecs/max98925.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 0796dfa..9b5a17d 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -442,8 +442,8 @@ static int max98925_dai_hw_params(struct snd_pcm_substream *substream, case 24: regmap_update_bits(max98925->regmap, MAX98925_FORMAT, - M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); - max98925->ch_size = 32; + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_24); + max98925->ch_size = 24; break; case 32: regmap_update_bits(max98925->regmap, -- 1.9.1

2 1

Re: [alsa-devel] snd_usb_audio - siberia elite prism soundcard issues
by Raymond Yau 27 Mar '15

27 Mar '15

>> >> > >> >> > >> >> numid=11,iface=MIXER,name='Speaker Playback Volume' ; type=INTEGER,access=rw---R--,values=8,min=0,max=74,step=0 : values=54,54,54,54,54,54,54,54 | dBminmax-min=-74.00dB,max=0.00dB >> >> >> >> You need workaround since pulseaudio don't like volume control with more than two channels >> >> >> >> https://bugs.freedesktop.org/show_bug.cgi?id=84983#c35 >> >> >> >> Can you post output of your usb audio device ? >> >> >> >> lsusb -vvvv >> > >> > >> > >> > Certainly! >> > >> > Here it is: >> > >> > Bus 001 Device 003: ID 1038:1225 Ideazon, Inc. >> > Device Descriptor: >> > bLength 18 >> > bDescriptorType 1 >> > bcdUSB 2.00 >> > bDeviceClass 0 (Defined at Interface level) >> > bDeviceSubClass 0 >> > bDeviceProtocol 0 >> > bMaxPacketSize0 64 >> > idVendor 0x1038 Ideazon, Inc. >> > idProduct 0x1225 >> > bcdDevice 0.03 >> > iManufacturer 1 SteelSeries >> > iProduct 2 SteelSeries Siberia Elite >> > iSerial 3 000000000000 >> > bNumConfigurations 1 >> >> > >> > AudioControl Interface Descriptor: >> > bLength 13 >> > bDescriptorType 36 >> > bDescriptorSubtype 4 (MIXER_UNIT) >> > Warning: Descriptor too short >> >> https://github.com/gregkh/usbutils/blob/master/lsusb.c >> >> Seem no warning for descriptor too short for uac2 and can dump those uac2 specific variables >> >> > bUnitID 10 >> > bNrInPins 2 >> > baSourceID( 0) 9 >> > baSourceID( 1) 1 >> > bNrChannels 8 >> > wChannelConfig 0x063f >> > Left Front (L) >> > Right Front (R) >> > Center Front (C) >> > Low Freqency Enhancement (LFE) >> > Left Surround (LS) >> > Right Surround (RS) >> > Side Left (SL) >> > Side Right (SR) >> > iChannelNames 0 >> > bmControls 0x00 >> > bmControls 0x00 >> > iMixer 33 (error) >> > > > > Maybe it's the language barrier, but I am not sure if you wanted me to do something. I, out of curiosity, tried to compile that lsusb but I'm missing headers such as config.h. > > Should you need any more feedback from me about this device, please do not hesitate asking through mailing list or directly, just consider I'm not dev! :) > Do you mean there is no sound using alsa hw device since pulseaudio does not support the multichannel speaker playback volume comtrol ? speaker-test -c8 -t wav -D hw:CARD=Elite speaker-test -c2 -t wav -D hw:CARD=Elite You need to know how to compile alsa driver for you running kernel if you want to fix the problem >> >> > > > amixer: Control hw:2 element read error: Invalid argument If alsamixer or amixer cannot set the speaker playback volume Try skip the problematic PCM Volume controls by adding if (state->chip->usb_id == USB_ID(0x1038,0x1225)) if(unitid == 10) ich_has_control = 0; If the driver really need this mixer unit for the playback path, you need to fix those controls bUnitID 10 bNrInPins 2 baSourceID( 0) 9 microphone baSourceID( 1) 1 usb streaming This look like the first four problematic controls related to microphone and the last control related to streaming numid=5,iface=MIXER,name='PCM Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=6,iface=MIXER,name='PCM Volume',index=1 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=7,iface=MIXER,name='PCM Volume',index=2 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=8,iface=MIXER,name='PCM Volume',index=3 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=9,iface=MIXER,name='PCM Volume',index=4 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument

2 1

[alsa-devel] [PATCH 1/4] ASoC: rt5645: Redefine format config for rt5650
by Bard Liao 27 Mar '15

27 Mar '15

rt5650 and rt5645 use different register bits for format configuration. This patch modifies rt5645_hw_params and rt5645_set_dai_fmt to support both codecs. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Mark, I am think if we should use different codec driver for rt5645 and rt5650. In fact, they are different Die. What do you think? --- --- sound/soc/codecs/rt5645.c | 39 +++++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4c384a1..0133c8c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2049,7 +2049,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int val_len = 0, val_clk, mask_clk; + unsigned int val_len = 0, val_clk, mask_clk, dl_sft; int pre_div, bclk_ms, frame_size; rt5645->lrck[dai->id] = params_rate(params); @@ -2063,6 +2063,16 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); return -EINVAL; } + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + dl_sft = 4; + break; + default: + dl_sft = 2; + break; + } + bclk_ms = frame_size > 32; rt5645->bclk[dai->id] = rt5645->lrck[dai->id] * (32 << bclk_ms); @@ -2075,13 +2085,13 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, case 16: break; case 20: - val_len |= RT5645_I2S_DL_20; + val_len = 0x1; break; case 24: - val_len |= RT5645_I2S_DL_24; + val_len = 0x2; break; case 8: - val_len |= RT5645_I2S_DL_8; + val_len = 0x3; break; default: return -EINVAL; @@ -2093,7 +2103,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS1_SFT | pre_div << RT5645_I2S_PD1_SFT; snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; case RT5645_AIF2: @@ -2101,7 +2111,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS2_SFT | pre_div << RT5645_I2S_PD2_SFT; snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; default: @@ -2116,7 +2126,16 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int reg_val = 0; + unsigned int reg_val = 0, pol_sft; + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + pol_sft = 8; + break; + default: + pol_sft = 7; + break; + } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -2134,7 +2153,7 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_IB_NF: - reg_val |= RT5645_I2S_BP_INV; + reg_val |= (1 << pol_sft); break; default: return -EINVAL; @@ -2158,12 +2177,12 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (dai->id) { case RT5645_AIF1: snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; case RT5645_AIF2: snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; default: -- 1.8.1.1.439.g50a6b54

2 4

[alsa-devel] [PATCH 1/4] ASoC: rt5645: Redefine format config for rt5650
by Bard Liao 27 Mar '15

27 Mar '15

rt5650 and rt5645 use different register bits for format configuration. This patch modifies rt5645_hw_params and rt5645_set_dai_fmt to support both codecs. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Mark, I am think if we should use different codec driver for rt5645 and rt5650. In fact, they are different Die. What do you think? --- --- sound/soc/codecs/rt5645.c | 39 +++++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4c384a1..0133c8c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2049,7 +2049,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int val_len = 0, val_clk, mask_clk; + unsigned int val_len = 0, val_clk, mask_clk, dl_sft; int pre_div, bclk_ms, frame_size; rt5645->lrck[dai->id] = params_rate(params); @@ -2063,6 +2063,16 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); return -EINVAL; } + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + dl_sft = 4; + break; + default: + dl_sft = 2; + break; + } + bclk_ms = frame_size > 32; rt5645->bclk[dai->id] = rt5645->lrck[dai->id] * (32 << bclk_ms); @@ -2075,13 +2085,13 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, case 16: break; case 20: - val_len |= RT5645_I2S_DL_20; + val_len = 0x1; break; case 24: - val_len |= RT5645_I2S_DL_24; + val_len = 0x2; break; case 8: - val_len |= RT5645_I2S_DL_8; + val_len = 0x3; break; default: return -EINVAL; @@ -2093,7 +2103,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS1_SFT | pre_div << RT5645_I2S_PD1_SFT; snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; case RT5645_AIF2: @@ -2101,7 +2111,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS2_SFT | pre_div << RT5645_I2S_PD2_SFT; snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; default: @@ -2116,7 +2126,16 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int reg_val = 0; + unsigned int reg_val = 0, pol_sft; + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + pol_sft = 8; + break; + default: + pol_sft = 7; + break; + } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -2134,7 +2153,7 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_IB_NF: - reg_val |= RT5645_I2S_BP_INV; + reg_val |= (1 << pol_sft); break; default: return -EINVAL; @@ -2158,12 +2177,12 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (dai->id) { case RT5645_AIF1: snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; case RT5645_AIF2: snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; default: -- 1.8.1.1.439.g50a6b54

1 0

[alsa-devel] [PATCH v4 0/15] Those patches is used for dw_hdmi audio support
by Yakir Yang 27 Mar '15

27 Mar '15

We found Designware hdmi driver only support audio clock config, we can not play sound through it. To add Designware HDMI Audio support, we make those patch set: 1): fixed dw_hdmi irq bug, add irq control to suspend/resume interfaces. 2): add suspend/resume callback for dw_hdmi rockchip driver. 3): Warp irq control in functions. 4): Combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() 5): Adjust n/cts config order. 6): Set ncts_atomic_write & cts_manual, according to dw_hdmi document. 7): Add audio support for more display resolutions(eg. 800x600). 8): Add audio support for No-CEA display resolutions. 9): Add Audio Sample Channel Status config interfaces to dw_hdmi driver. 10): add audio clock control interfaces to dw_hdmi driver. 11): creat "dw_hdmi-audio" platform device in dw_hdmi driver. 12): add codec driver for hdmi audio, callback dw_hdmi audio config functions. 13): add sound driver for hdmi audio, creat hdmi audio sound card. 14): add dt-bings file and add hdmi_audio node to corresponding dt file. Changes in v4: -Correct phy_type assignment bug - Combine CTS3 registers setting, reduce register operate times - Combine N3 registers setting - Add hdmi audio support when monitor support audio - Give HDMI_FC_AUD_SCHNL8 an readable value - Rename "hdmi_audio_*" to "dw_hdmi_audio_*" - Replace delaywork with irq thread, and add suspend/resume interfaces, Replace "dw-hdmi-audio" with consecutive strings. - Add ".pm = &snd_soc_pm_ops," Changes in v3: - Clear Hotplug interrupts before dw_hdmi_fb_register - Wrap irq control in fucntions - Setting the .pm member instead of suspend/resume - Add ID registers parse and record - Combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() - Only adjust the n/cts setting order - Set ncts_atomic_write & cts_manual - Determine whether sample channel should set by desig_id. - Delete hdmi_audio_config interface and modify audio clock control functions - Remove audio_config & get_connect_status callback functions and add write/read/mod register callback functions - Keep audio format config function in dw-hdmi-audio driver and remove audio_config & get_connect_status functions, move jack control to dw-hdmi-audio completely. - Delete the operation of jack in rockchip-hdmi-audio driver, get ready to switch to simple-audio-card driver. - modify cpu-of-node to i2s-controller Changes in v2: - Add irq control to suspend/resume interfaces - Add suspend/resume support for dw_hdmi_rockchip driver - add more n/cts combinations for more display resolutions - Enable audio support for No-CEA display mode - Add audio sample channel status setting - Add audio config interfaces to dw_hdmi driver - Update the audio control interfaces - Update dw_hdmi audio control interfaces, and adjust jack report process - give "codec-name" & "codec-dai-name" an const name - remove codec-name and codec-dai-name - rename rockchip,rockchip-hdmi-audio.txt to rockchip,rockchip-dw-hdmi-audio.txt Daniel Kurtz (3): drm: bridge/dw_hdmi: adjust n/cts setting order drm: bridge/dw_hdmi: set ncts_atomic_write & cts_manual drm: bridge/dw_hdmi: add audio sample channel status setting Yakir Yang (12): drm: bridge/dw_hdmi: add irq control to suspend/resume drm: bridge/dw_hdmi: wrap irq control in fucntions drm: rockchip/dw_hdmi_rockchip: add resume/suspend support drm: bridge/dw_hdmi: add identification registers parse and record drm: bridge/dw_hdmi: combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() drm: bridge/dw_hdmi: add audio support for more display resolutions drm: bridge/dw_hdmi: enable audio support for No-CEA display resolutions drm: bridge/dw_hdmi: add enable/disable to dw_hdmi_audio callbacks drm: bridge/dw_hdmi: creat dw-hdmi-audio platform device ASoC: codec/dw-hdmi-audio: add codec driver for dw hdmi audio ASoC: rockchip/rockchip-hdmi-audio: add sound driver for hdmi audio dt-bindings: Add documentation for Rockchip dw-hdmi-audio .../sound/rockchip,rockchip-dw-hdmi-audio.txt | 12 + drivers/gpu/drm/bridge/dw_hdmi.c | 385 ++++++++++++++++++--- drivers/gpu/drm/bridge/dw_hdmi.h | 53 ++- drivers/gpu/drm/rockchip/dw_hdmi-rockchip.c | 16 + include/drm/bridge/dw_hdmi.h | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dw-hdmi-audio.c | 379 ++++++++++++++++++++ sound/soc/codecs/dw-hdmi-audio.h | 74 ++++ sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_hdmi_audio.c | 169 +++++++++ 12 files changed, 1076 insertions(+), 46 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt create mode 100644 sound/soc/codecs/dw-hdmi-audio.c create mode 100644 sound/soc/codecs/dw-hdmi-audio.h create mode 100644 sound/soc/rockchip/rockchip_hdmi_audio.c -- 2.1.2

4 11

[alsa-devel] [PATCH] ASoC: Add support for NAU8824 codec to ASoC
by Wan Zongshun 27 Mar '15

27 Mar '15

Signed-off-by: Wan Zongshun <mcuos.com(a)gmail.com> Signed-off-by: Chih-Chiang Chang <ccchang12(a)nuvoton.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8824.c | 770 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8824.h | 379 ++++++++++++++++++++++ 4 files changed, 1156 insertions(+) create mode 100644 sound/soc/codecs/nau8824.c create mode 100644 sound/soc/codecs/nau8824.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 064e6c1..862b7bd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8824 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER @@ -463,6 +464,10 @@ config SND_SOC_MAX98357A config SND_SOC_MAX9850 tristate +config SND_SOC_NAU8824 + tristate "Nuvoton NAU8824 CODEC" + depends on I2C + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 69b8666..acb7bda 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -68,6 +68,7 @@ snd-soc-max98357a-objs := max98357a.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8824-objs := nau8824.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o @@ -250,6 +251,7 @@ obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c new file mode 100644 index 0000000..80c2283 --- /dev/null +++ b/sound/soc/codecs/nau8824.c @@ -0,0 +1,770 @@ +/* + * linux/sound/soc/codecs/nau8824.c + * + * Copyright 2015 Nuvoton Technology Corp. + * Author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/regmap.h> +#include <linux/acpi.h> +#include <asm/div64.h> +#include <sound/jack.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/delay.h> + +#include "nau8824.h" + +static int nau8824_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); +static int nau8824_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt); +static int nau8824_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level); +static int nau8824_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir); +static const DECLARE_TLV_DB_SCALE(out_spk_vol_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(out_hp_vol_tlv, -3000, 100, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -12800, 50, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -12800, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_left_vol_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_right_vol_tlv, -12750, 50, 1); + +static const struct snd_kcontrol_new nau8824_snd_controls[] = { + /* SP Class-D mute control */ + SOC_DOUBLE("SP Playback Switch", NAU8824_CLASS_D_GAIN1, + NAU8824_CLASS_D_SFT, NAU8824_CLASS_D_SFT, 1, 0), + /* SP Class-D driver output stage gain control */ + SOC_SINGLE_TLV("SP Left Volume", NAU8824_CLASS_D_GAIN2, + NAU8824_CLASS_D_GAIN_L_SFT, + NAU8824_SPK_VOL_RSCL_RANGE, 0, out_spk_vol_tlv), + SOC_SINGLE_TLV("SP Right Volume", NAU8824_CLASS_D_GAIN1, + NAU8824_CLASS_D_GAIN_R_SFT, + NAU8824_SPK_VOL_RSCL_RANGE, 0, + out_spk_vol_tlv), + /* SP Class-D mute control */ + SOC_DOUBLE("HP Playback Switch", NAU8824_HP_MUTE, NAU8824_L_MUTE_SFT, + NAU8824_R_MUTE_SFT, 1, 1), + SOC_SINGLE_TLV("HP Left Volume", NAU8824_HP_VOL, NAU8824_L_VOL_SFT, + NAU8824_VOL_RSCL_RANGE, 1, out_hp_vol_tlv), + SOC_SINGLE_TLV("HP Right Volume", NAU8824_HP_VOL, NAU8824_R_VOL_SFT, + NAU8824_VOL_RSCL_RANGE, 1, out_hp_vol_tlv), + /* DMIC control */ + SOC_DOUBLE("DMIC L R Switch", NAU8824_ENA_CTRL, NAU8824_DMIC_CH0_SFT, + NAU8824_DMIC_CH1_SFT, 1, 0), + SOC_SINGLE("DMIC Enable", NAU8824_BIAS_ADJ, NAU8824_DMIC1_SFT, 1, 0), + SOC_SINGLE("VMID Switch", NAU8824_BIAS_ADJ, NAU8824_VMID_SFT, 1, 0), + SOC_SINGLE("BIAS Switch", NAU8824_BOOST, NAU8824_G_BIAS_SFT, 1, 0), + SOC_DOUBLE_R_TLV("MIC L R Gain", NAU8824_ADC_CH0_DGAIN_CTRL, + NAU8824_ADC_CH1_DGAIN_CTRL, 0, + NAU8824_ADC_VOL_RSCL_RANGE, 0, adc_vol_tlv), +}; + +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, NAU8824_CLK_DIVIDER, + NAU8824_CLK_DMIC_SRC_MASK, + NAU8824_CLK_DMIC_DIV_4); + break; + default: + break; + } + return 0; +} + +static const struct snd_kcontrol_new nau8824_rec_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", SND_SOC_NOPM, + 0, 0, 0), +}; + +static const struct snd_kcontrol_new nau8824_rec_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", SND_SOC_NOPM, + 0, 0, 0), +}; + +static const struct snd_kcontrol_new nau8824_spo_mix[] = { + SOC_DAPM_SINGLE("SP L Switch", SND_SOC_NOPM, + 0, 0, 0), + SOC_DAPM_SINGLE("SP R Switch", SND_SOC_NOPM, + 0, 0, 0), +}; + +static const struct snd_kcontrol_new nau8824_hpo_mix[] = { + SOC_DAPM_SINGLE("HP L Switch", + NAU8824_HPO_MIXER, NAU8824_M_HPVOL_L_HM_SFT, + 1, 0), + SOC_DAPM_SINGLE("HP R Switch", + NAU8824_HPO_MIXER, NAU8824_M_HPVOL_R_HM_SFT, + 1, 0), +}; + +static const char * const nau8824_stereo_adc1_src[] = { "ADC", "DMIC" }; + +static const SOC_ENUM_SINGLE_DECL(nau8824_stereo_adc_l1_enum, NAU8824_ENA_CTRL, + NAU8824_DMIC_CH0_SFT, nau8824_stereo_adc1_src); + +static const SOC_ENUM_SINGLE_DECL(nau8824_stereo_adc_r1_enum, NAU8824_ENA_CTRL, + NAU8824_DMIC_CH1_SFT, nau8824_stereo_adc1_src); + +static const struct snd_kcontrol_new nau8824_sto_adc_l1_mux = + SOC_DAPM_ENUM("Stereo ADC L1 source", + nau8824_stereo_adc_l1_enum); + +static const struct snd_kcontrol_new nau8824_sto_adc_r1_mux = + SOC_DAPM_ENUM("Stereo ADC R1 source", + nau8824_stereo_adc_r1_enum); + +static int nau8824_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, NAU8824_ANALOG_CTRL2, + NAU8824_CLASS_D_CLAMP_MSK, + NAU8824_CLASS_D_CLAMP_BETTER); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, NAU8824_CLASS_D_GAIN1, + NAU8824_CLASS_D_MASK, NAU8824_CLASS_D_DIS); + default: + break; + } + return 0; +} + +static int nau8824_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + return 0; +} + +static int nau8824_dac1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + return 0; +} + +static int micbias_power_on_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + if (SND_SOC_DAPM_EVENT_ON(event)) + snd_soc_update_bits(codec, NAU8824_BOOST, NAU8824_G_BIAS_MASK, + NAU8824_G_BIAS_EN); + else if (SND_SOC_DAPM_EVENT_OFF(event)) + snd_soc_update_bits(codec, NAU8824_BOOST, NAU8824_G_BIAS_MASK, + NAU8824_G_BIAS_DIS); + return 0; +} + +static int vmid_power_on_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + if (SND_SOC_DAPM_EVENT_ON(event)) + snd_soc_update_bits(codec, NAU8824_BIAS_ADJ, + NAU8824_VMID_MASK, NAU8824_VMID_EN); + else if (SND_SOC_DAPM_EVENT_OFF(event)) + snd_soc_update_bits(codec, NAU8824_BIAS_ADJ, + NAU8824_VMID_MASK, NAU8824_VMID_DIS); + return 0; +} + +static const struct snd_soc_dapm_widget nau8824_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + SND_SOC_DAPM_SUPPLY("micbias", SND_SOC_NOPM, 0, 0, + micbias_power_on_event, SND_SOC_DAPM_PRE_PMU + | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SUPPLY("vmid", SND_SOC_NOPM, 0, 0, vmid_power_on_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA("DMIC L1", NAU8824_ENA_CTRL, NAU8824_DMIC_CH0_SFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC R1", NAU8824_ENA_CTRL, NAU8824_DMIC_CH1_SFT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, set_dmic_clk, + SND_SOC_DAPM_PRE_PMU), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", SND_SOC_NOPM, 0, 0, nau8824_rec_l_mix, + ARRAY_SIZE(nau8824_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", SND_SOC_NOPM, 0, 0, nau8824_rec_r_mix, + ARRAY_SIZE(nau8824_rec_r_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &nau8824_sto_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &nau8824_sto_adc_r1_mux), + /* ADC IF1 */ + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC L1", NULL, NAU8824_DAC_CTRL, + NAU8824_DAC_L_SFT, 0, nau8824_dac1_event, + SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DAC R1", NULL, NAU8824_DAC_CTRL, + NAU8824_DAC_R_SFT, 0, nau8824_dac1_event, + SND_SOC_DAPM_PRE_PMD), + + /* SPO/HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("SPO MIX", SND_SOC_NOPM, 0, 0, nau8824_spo_mix, + ARRAY_SIZE(nau8824_spo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0, nau8824_hpo_mix, + ARRAY_SIZE(nau8824_hpo_mix)), + + SND_SOC_DAPM_PGA_S("HP amp", 1, NAU8824_CLASS_G_CTRL, + NAU8824_CLASS_G_SHIFT, 0, nau8824_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("SPK amp", 1, NAU8824_CLASS_D_GAIN1, + NAU8824_CLASS_D_SFT, 0, nau8824_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8824_dapm_routes[] = { + + {"DMIC L1", NULL, "DMIC1"}, + {"DMIC R1", NULL, "DMIC1"}, + + {"BST1", NULL, "IN1P"}, + {"BST1", NULL, "IN1N"}, + {"BST2", NULL, "IN2P"}, + {"BST2", NULL, "IN2N"}, + {"IN2P", NULL, "micbias"}, + {"IN2P", NULL, "vmid"}, + + {"RECMIXL", "BST1 Switch", "BST1"}, + {"RECMIXR", "BST2 Switch", "BST2"}, + + {"ADC L", NULL, "RECMIXL"}, + {"ADC R", NULL, "RECMIXR"}, + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC CLK"}, + + {"Stereo ADC L1 Mux", "ADC", "ADC L"}, + {"Stereo ADC L1 Mux", "DMIC", "DMIC L1"}, + {"Stereo ADC R1 Mux", "ADC", "ADC R"}, + {"Stereo ADC R1 Mux", "DMIC", "DMIC R1"}, + {"AIF1TX", NULL, "IF1 ADC"}, + + {"DAC L1", NULL, "AIF1RX"}, + {"DAC R1", NULL, "AIF1RX"}, + {"SPO MIX", "SP L Switch", "DAC L1"}, + {"SPO MIX", "SP R Switch", "DAC R1"}, + {"SPK amp", NULL, "SPO MIX"}, + {"SPK", NULL, "SPK amp"}, + + {"HPO MIX", "HP L Switch", "DAC L1"}, + {"HPO MIX", "HP R Switch", "DAC R1"}, + {"HP amp", NULL, "HPO MIX"}, + {"HPOL", NULL, "HP amp"}, + {"HPOR", NULL, "HP amp"}, +}; + +static int nau8824_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *tmp) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + unsigned int val_len = 0; + + dev_dbg(codec->dev, "%s\n", __func__); + dev_dbg(codec->dev, "##- Data length: %d\n", params_format(params)); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= NAU8824_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= NAU8824_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + case SNDRV_PCM_FORMAT_S32_LE: + val_len |= NAU8824_I2S_DL_32; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, NAU8824_I2S_PCM_CTRL_1, NAU8824_I2S_DL_MASK, + val_len); + return 0; +} + +static int nau8824_dac_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + if (mute) + snd_soc_update_bits(codec, NAU8824_DAC_MUTE_CTRL, + NAU8824_SOFT_MUTE_MASK, NAU8824_SOFT_MUTE_EN); + else + snd_soc_update_bits(codec, NAU8824_DAC_MUTE_CTRL, + NAU8824_SOFT_MUTE_MASK, NAU8824_SOFT_MUTE_DIS); + return 0; +} + +static int nau8824_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int reg_val = 0, reg_val_2 = 0; + + dev_dbg(codec->dev, "%s: Entered\n", __func__); + dev_dbg(codec->dev, "###nau8824_set_dai_fmt %x\n", fmt); + dev_dbg(codec->dev, "##+ nau8824_set_dai_fmt (%x)\n", fmt); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + reg_val_2 |= NAU8824_I2S_MS_M; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + break; + default: + dev_alert(codec->dev, "Invalid DAI master/slave interface\n"); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= NAU8824_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg_val |= NAU8824_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= NAU8824_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + reg_val |= NAU8824_I2S_DF_RIGHT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= NAU8824_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= NAU8824_I2S_DF_PCM_B; + reg_val |= NAU8824_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, NAU8824_I2S_PCM_CTRL_1, + NAU8824_I2S_DL_MASK | NAU8824_I2S_DF_MASK + | NAU8824_I2S_BP_MASK | NAU8824_I2S_PCMB_MASK, + reg_val); + snd_soc_update_bits(codec, NAU8824_I2S_PCM_CTRL_2, NAU8824_I2S_MS_MASK, + reg_val_2); + + dev_dbg(codec->dev, "##-nau8824_set_dai_fmt Master\n"); + dev_dbg(codec->dev, "%s: Exiting\n", __func__); + return 0; +} + +static int nau8824_FLL_freerun_mode(struct snd_soc_codec *codec, int IsEnable) +{ + if (IsEnable == true) { + snd_soc_write(codec, 0x03, 0x0853); + snd_soc_write(codec, 0x09, 0xE000); + snd_soc_write(codec, 0x03, 0x8853); + } else { + snd_soc_write(codec, 0x03, 0x0853); + snd_soc_write(codec, 0x09, 0x6000); + snd_soc_write(codec, 0x03, 0x8853); + } + return 0; +} + +void set_sys_clk(struct snd_soc_codec *codec, int sys_clk) +{ + pr_debug("%s sys_clk=%x\n", __func__, sys_clk); + switch (sys_clk) { + case NAU8824_INTERNALCLOCK: + nau8824_FLL_freerun_mode(codec, true); + break; + + case NAU8824_MCLK: + default: + nau8824_FLL_freerun_mode(codec, false); + break; + } +} + +static int nau8824_set_dai_pll(struct snd_soc_dai *dai, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "In nau8824: dai_set_pll\n"); + return 0; +} + +static int nau8824_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + int divider = 1; + + if (clk_id == NAU8824_MCLK) { + set_sys_clk(codec, NAU8824_MCLK); + dev_dbg(codec->dev, "%s: input freq = %d divider = %d", + __func__, freq, divider); + + } else if (clk_id == NAU8824_INTERNALCLOCK) { + set_sys_clk(codec, NAU8824_INTERNALCLOCK); + } else { + dev_err(codec->dev, "Wrong clock src\n"); + return -EINVAL; + } + return 0; +} + +static int nau8824_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + dev_dbg(codec->dev, "## nau8824_set_bias_level %d\n", level); + if (level == codec->dapm.bias_level) { + dev_dbg(codec->dev, "##set_bias_level: level returning...\r\n"); + return -EINVAL; + } + + switch (level) { + case SND_SOC_BIAS_ON: + /* All power is driven by DAPM system*/ + dev_dbg(codec->dev, "###nau8824_set_bias_level BIAS_ON\n"); + snd_soc_update_bits(codec, NAU8824_BIAS_ADJ, + NAU8824_VMID_MASK, NAU8824_VMID_EN); + snd_soc_update_bits(codec, NAU8824_BOOST, + NAU8824_G_BIAS_MASK, NAU8824_G_BIAS_EN); + break; + + case SND_SOC_BIAS_PREPARE: + dev_dbg(codec->dev, "###nau8824_set_bias_level BIAS_PREPARE\n"); + break; + + case SND_SOC_BIAS_STANDBY: + dev_dbg(codec->dev, "###nau8824_set_bias_level STANDBY\n"); + break; + + case SND_SOC_BIAS_OFF: + dev_dbg(codec->dev, "###nau8824_set_bias_level OFF\n"); + set_sys_clk(codec, NAU8824_INTERNALCLOCK); + snd_soc_update_bits(codec, NAU8824_BIAS_ADJ, NAU8824_VMID_MASK, + NAU8824_VMID_DIS); + snd_soc_update_bits(codec, NAU8824_BOOST, + NAU8824_G_BIAS_MASK, NAU8824_G_BIAS_DIS); + break; + default: + break; + } + codec->dapm.bias_level = level; + dev_dbg(codec->dev, "## nau8824_set_bias_level %d\n", level); + + return 0; +} + +static int nau8824_suspend(struct snd_soc_codec *codec) +{ + dev_dbg(codec->dev, "%s: Entered\n", __func__); + nau8824_set_bias_level(codec, SND_SOC_BIAS_OFF); + dev_dbg(codec->dev, "%s: Exiting\n", __func__); + return 0; +} + +static int nau8824_resume(struct snd_soc_codec *codec) +{ + dev_dbg(codec->dev, "%s: Entered\n", __func__); + nau8824_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + dev_dbg(codec->dev, "%s: Exiting\n", __func__); + return 0; +} + +struct nau8824_init_reg { + u8 reg; + u16 val; +}; + +static struct nau8824_init_reg init_list[] = { + {0x0000, 0x0001}, + {0x0000, 0x0000}, + {0x0066, 0x0060}, + {0x0076, 0x3000}, + {0x0004, 0x0002}, + {0x0005, 0xDD44}, + {0x0006, 0x0007}, + {0x0007, 0x0810}, + {0x0008, 0xC000}, + {0x0074, 0x1106}, + {0x0073, 0x330C}, + {0x0080, 0x0000}, + {0x006D, 0x0100}, + {0x006E, 0x0001}, + {0x006A, 0x0008}, + {0x006D, 0x0180}, + {0x006D, 0x1480}, + {0x006E, 0x0014}, + {0x0050, 0x2007}, + {0x0080, 0x3020}, + {0x006B, 0xC000}, + {0x007B, 0x1E1E}, + {0x006B, 0xC005}, + {0x0068, 0x8000}, + {0x0076, 0x300F}, + {0x007F, 0x000F}, + {0x0080, 0x0020}, + {0x006B, 0x0005}, + {0x007B, 0x0000}, + {0x0002, 0x0800}, + {0x0025, 0x1000}, + {0x0024, 0x0002}, + {0x0002, 0x0AA5}, + {0x0020, 0x0000}, + {0x0025, 0x1082}, + {0x0031, 0x0800}, + {0x0032, 0x0100}, + {0x0033, 0x0300}, + {0x0034, 0x0000}, + {0x001C, 0x000A}, + {0x001F, 0x2000}, + {0x0001, 0x0030}, + {0x0002, 0x3AA5}, + {0x0067, 0x4040}, + {0x007F, 0x300F}, + {0x0077, 0x0010}, + {0x0078, 0x3C00}, + {0x0076, 0x380F}, + {0x0076, 0x300F}, + {0x007A, 0x0202}, + {0x0067, 0x0000}, + {0x0072, 0x40A0}, + {0x0079, 0x0300}, + {0x0023, 0x1010}, + {0x0024, 0x0002}, + {0x0038, 0x1486}, + {0x003C, 0x1486}, + {0x0066, 0x0064}, + {0x0001, 0x40F0}, + {0x0001, 0x48F0}, + {0x0025, 0x3082}, + {0x0020, 0x0003}, + {0x0024, 0x8102}, + {0x002D, 0x0108}, + {0x002E, 0x0308}, + {0x002F, 0x0500}, + {0x0030, 0x0700}, + {0x001C, 0x000A}, + {0x001F, 0x2000}, + {0x0002, 0x3BA5}, + {0x0001, 0x48F3}, + {0x0002, 0xFBA5}, + {0x0014, 0x2210}, + {0x007c, 0x1e1e}, +}; + +#define NAU8824_INIT_REG_LEN ARRAY_SIZE(init_list) +static int nau8824_reg_init(struct snd_soc_codec *codec) +{ + int i; + + mdelay(1); + for (i = 0; i < NAU8824_INIT_REG_LEN; i++) { + if (init_list[i].reg == 0xFFFF) + mdelay(1); + else + snd_soc_write(codec, init_list[i].reg, + init_list[i].val); + } + return 0; +} + +static int nau8824_codec_probe(struct snd_soc_codec *codec) +{ + int ret = 0; + struct nau8824_priv *nau8824; + + nau8824 = snd_soc_codec_get_drvdata(codec); + codec->control_data = nau8824->regmap; + mutex_init(&nau8824->mutex); + nau8824_reg_init(codec); + + /* Dynamic Headset detection enabled */ + snd_soc_update_bits(codec, 0x01, 0x400, 0x400); + snd_soc_update_bits(codec, 0x02, 0x0008, 0x0008); + snd_soc_update_bits(codec, 0x0f, 0x0300, 0x0100); + snd_soc_write(codec, 0x09, 0xE000); + snd_soc_write(codec, NAU8824_IRQ_SETTING, 0x1006); + snd_soc_write(codec, 0x13, 0x1615); + snd_soc_write(codec, 0x15, 0x0414); + snd_soc_update_bits(codec, 0x16, 0xFF00, 0x5900); + snd_soc_update_bits(codec, 0x66, 0x0070, 0x0060); + + /* Program codec to use internal clock */ + nau8824_FLL_freerun_mode(codec, true); + return ret; +} + +static int nau8824_codec_remove(struct snd_soc_codec *codec) +{ + nau8824_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_driver_nau8824 = { + .probe = nau8824_codec_probe, + .remove = nau8824_codec_remove, + .suspend = nau8824_suspend, + .resume = nau8824_resume, + .set_bias_level = nau8824_set_bias_level, + .controls = nau8824_snd_controls, + .num_controls = ARRAY_SIZE(nau8824_snd_controls), + .dapm_widgets = nau8824_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8824_dapm_widgets), + .dapm_routes = nau8824_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8824_dapm_routes), + .reg_word_size = sizeof(u16), + .reg_cache_default = init_list, + .set_sysclk = nau8824_set_sysclk, +}; + +static struct snd_soc_dai_ops nau8824_dai_ops = { + .hw_params = nau8824_hw_params, + .set_pll = nau8824_set_dai_pll, + .set_fmt = nau8824_set_dai_fmt, + .digital_mute = nau8824_dac_mute, +}; + +static struct snd_soc_dai_driver nau8824_dai_driver[] = { +{ + .name = "nau8824-aif1", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8824_RATES, + .formats = NAU8824_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8824_RATES, + .formats = NAU8824_FORMATS, + }, + .ops = &nau8824_dai_ops, +} +}; + +static int nau8824_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct nau8824_priv *nau8824; + int ret; + + pr_debug("%s enter", __func__); + nau8824 = kzalloc(sizeof(*nau8824), GFP_KERNEL); + if (nau8824 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, nau8824); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_nau8824, + nau8824_dai_driver, + ARRAY_SIZE(nau8824_dai_driver)); + if (ret < 0) + kfree(nau8824); + + return ret; +} + +static int nau8824_i2c_remove(struct i2c_client *i2c) +{ + struct nau8824_priv *nau8824 = dev_get_drvdata(&i2c->dev); + + snd_soc_unregister_codec(nau8824->dev); + + return 0; +} + +static const struct i2c_device_id nau8824_i2c_id[] = { + { "nau8824", 0}, + {"10508824:00", 0}, + {"10508824", 0}, + { } +}; + +MODULE_DEVICE_TABLE(i2c, nau8824_i2c_id); +static struct i2c_driver nau8824_i2c_driver = { + .driver = { + .name = "nau8824", + .owner = THIS_MODULE, + }, + .probe = nau8824_i2c_probe, + .remove = (nau8824_i2c_remove), + .id_table = nau8824_i2c_id, +}; +module_i2c_driver(nau8824_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8824 codec driver"); +MODULE_AUTHOR("Nuvoton"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/codecs/nau8824.h b/sound/soc/codecs/nau8824.h new file mode 100644 index 0000000..1745dac --- /dev/null +++ b/sound/soc/codecs/nau8824.h @@ -0,0 +1,379 @@ +/* + * linux/sound/soc/codecs/nau8824.h + * + * Copyright 2015 Nuvoton Technology Corp. + * Author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _NAU8824_H +#define _NAU8824_H + +#define NAU8824_RESET 0x00 +#define NAU8824_ENA_CTRL 0x01 +#define NAU8824_CLK_EN 0x02 +#define NAU8824_CLK_DIVIDER 0x03 +#define NAU8824_FL_1 0x04 +#define NAU8824_FL_2 0x05 +#define NAU8824_FL_3 0x06 +#define NAU8824_FL_4 0x07 +#define NAU8824_FL_5 0x08 +#define NAU8824_FL_6 0x09 +#define NAU8824_IRQ_STATUS 0x10 +#define NAU8824_IRQ_CLEAR 0x11 +#define NAU8824_IRQ_SETTING 0x12 +#define NAU8824_SAR_ADC 0x13 +#define NAU8824_VDET_COEFFICIENT 0x14 +#define NAU8824_I2S_PCM_CTRL_1 0x1C +#define NAU8824_I2S_PCM_CTRL_2 0x1D +#define NAU8824_ADC_CH0_DGAIN_CTRL 0x2D +#define NAU8824_ADC_CH1_DGAIN_CTRL 0x2E +#define NAU8824_DAC_MUTE_CTRL 0x31 +#define NAU8824_DAC0_DIG_VOL 0x32 +#define NAU8824_DAC1_DIG_VOL 0x33 +#define NAU8824_CLASS_G_CTRL 0x50 +#define NAU8824_SAR_ADC_OUTPUT 0x59 +#define NAU8824_BIAS_ADJ 0x66 +#define NAU8824_ANALOG_CTRL2 0x6A +#define NAU8824_HPO_MIXER 0x6B +#define NAU8824_CLASS_D_GAIN1 0x6D +#define NAU8824_CLASS_D_GAIN2 0x6E +#define NAU8824_DAC_CTRL 0x73 +#define NAU8824_MIC_BIAS 0x74 +#define NAU8824_HP_MUTE 0x75 +#define NAU8824_BOOST 0x76 +#define NAU8824_HP_VOL 0x7B + +/* reg. NAU8824_ENA_CTRL (0x01) */ +#define NAU8824_DMIC_EDGE_CH23_MASK (0x01 << 15) +#define NAU8824_DMIC_EDGE_CH23_SFT 15 +#define NAU8824_DMIC_EDGE_CH01_MASK (0x01 << 14) +#define NAU8824_DMIC_EDGE_CH01_SFT 14 +#define NAU8824_ADC_OP_EN_MASK (0x01 << 11) +#define NAU8824_ADC_OP_EN_SFT 11 +#define NAU8824_ADC_OP_EN (0x01 << 11) +#define NAU8824_ADC_OP_DIS (0x00 << 11) +#define NAU8824_INT_SLEEP_MASK (0x01 << 10) +#define NAU8824_INT_SLEEP_SFT 10 +#define NAU8824_INT_SLEEP_ON (0x01 << 10) +#define NAU8824_INT_SLEEP_OFF (0x00 << 10) +#define NAU8824_DMIC_CH3_MASK (0x01 << 9) +#define NAU8824_DMIC_CH3_SFT 9 +#define NAU8824_DMIC_CH3_ON (0x01 << 9) +#define NAU8824_DMIC_CH3_OFF (0x00 << 9) +#define NAU8824_DMIC_CH2_MASK (0x01 << 8) +#define NAU8824_DMIC_CH2_SFT 8 +#define NAU8824_DMIC_CH2_ON (0x01 << 8) +#define NAU8824_DMIC_CH2_OFF (0x00 << 8) +#define NAU8824_DMIC_CH1_MASK (0x01 << 7) +#define NAU8824_DMIC_CH1_SFT 7 +#define NAU8824_DMIC_CH1_ON (0x01 << 7) +#define NAU8824_DMIC_CH1_OFF (0x00 << 7) +#define NAU8824_DMIC_CH0_MASK (0x01 << 6) +#define NAU8824_DMIC_CH0_SFT 6 +#define NAU8824_DMIC_CH0_ON (0x01 << 6) +#define NAU8824_DMIC_CH0_OFF (0x00 << 6) +#define NAU8824_DAC_CH1_EN_MASK (0x01 << 5) +#define NAU8824_DAC_CH1_EN_SFT 5 +#define NAU8824_DAC_CH1_EN (0x01 << 5) +#define NAU8824_DAC_CH1_DIS (0x00 << 5) +#define NAU8824_DAC_CH0_EN_MASK (0x01 << 4) +#define NAU8824_DAC_CH0_EN_SFT 4 +#define NAU8824_DAC_CH0_EN (0x01 << 4) +#define NAU8824_DAC_CH0_DIS (0x00 << 4) +#define NAU8824_ADC_CH3_EN_MASK (0x01 << 3) +#define NAU8824_ADC_CH3_EN_SFT 3 +#define NAU8824_ADC_CH3_EN (0x01 << 3) +#define NAU8824_ADC_CH3_DIS (0x00 << 3) +#define NAU8824_ADC_CH2_EN_MASK (0x01 << 2) +#define NAU8824_ADC_CH2_EN_SFT 2 +#define NAU8824_ADC_CH2_EN (0x01 << 2) +#define NAU8824_ADC_CH2_DIS (0x00 << 2) +#define NAU8824_ADC_CH1_EN_MASK (0x01 << 1) +#define NAU8824_ADC_CH1_EN_SFT 1 +#define NAU8824_ADC_CH1_EN (0x01 << 1) +#define NAU8824_ADC_CH1_DIS (0x00 << 1) +#define NAU8824_ADC_CH0_EN_MASK (0x01 << 0) +#define NAU8824_ADC_CH0_EN_SFT 0 +#define NAU8824_ADC_CH0_EN (0x01 << 0) +#define NAU8824_ADC_CH0_DIS (0x00 << 0) + +/* reg. NAU8824_CLK_EN (0x02) */ +#define NAU8824_CLK_SAR_EN_MASK (0x1 << 3) +#define NAU8824_CLK_SAR_EN_SFT 3 +#define NAU8824_CLK_SAR_EN (0x1 << 3) +#define NAU8824_CLK_SAR_DIS (0x0 << 3) + +/* reg. NAU8824_CLK_DIVIDER (0x03) */ +#define NAU8824_SYSCLK_SRC_MASK (0x1 << 15) +#define NAU8824_SYSCLK_SRC_SFT 15 +#define NAU8824_SYSCLK_SRC_VCO (0x1 << 15) +#define NAU8824_SYSCLK_SRC_MCLK (0x0 << 15) +#define NAU8824_CLK_CODEC_SRC_MASK (0x1 << 13) +#define NAU8824_CLK_CODEC_SRC_SFT 13 +#define NAU8824_CODEC_SRC_SYSCLK (0x1 << 13) +#define NAU8824_CODEC_SRC_MCLK (0x0 << 13) +#define NAU8824_CLK_DMIC_SRC_MASK (0x7 << 10) +#define NAU8824_CLK_DMIC_SRC_SFT 10 +#define NAU8824_CLK_DMIC_DIV_4 (0x2 << 10) + +/* reg. NAU8824_IRQ_STATUS (0x10) */ +#define NAU8824_IRQ_KEY_RELEASE (0x1 << 5) +#define NAU8824_IRQ_LONGKEY (0x1 << 4) +#define NAU8824_IRQ_SHORTKEY (0x1 << 3) +#define NAU8824_IRQ_MIC_INSERTED (0x1 << 2) +#define NAU8824_IRQ_JACK_EJECTED (0x1 << 1) +#define NAU8824_IRQ_JACK_INSERTED (0x1 << 0) + +/* reg. NAU8824_IRQ_CLEAR (0x11) */ +#define NAU8824_IRQ_KEY_RELEASE_CLR (0x1 << 5) +#define NAU8824_IRQ_LONGKEY_CLR (0x1 << 4) +#define NAU8824_IRQ_SHORTKEY_CLR (0x1 << 3) +#define NAU8824_IRQ_MIC_INSERTED_CLR (0x1 << 2) +#define NAU8824_IRQ_JACK_EJECTED_CLR (0x1 << 1) +#define NAU8824_IRQ_JACK_INSERTED_CLR (0x1 << 0) + +/* reg NAU8824_SAR_ADC (0x13) */ +#define NAU8824_SAR_EN_MASK (0x1 << 12) +#define NAU8824_SAR_EN_SFT 12 +#define NAU8824_SAR_EN (0x1 << 12) +#define NAU8824_SAR_DIS (0x0 << 12) + +/* reg. NAU8824_I2S_PCM_CTRL_1 (0x1C) */ +#define NAU8824_I2S_BP_MASK (0x1 << 7) +#define NAU8824_I2S_BP_SFT 7 +#define NAU8824_I2S_BP_NOR (0x0 << 7) +#define NAU8824_I2S_BP_INV (0x1 << 7) +#define NAU8824_I2S_PCMB_MASK (0x1 << 6) +#define NAU8824_I2S_PCMB_SFT 6 +#define NAU8824_I2S_PCMB_EN (0x1 << 6) +#define NAU8824_I2S_PCMB_DIS (0x0 << 6) +#define NAU8824_I2S_DL_MASK (0x3 << 2) +#define NAU8824_I2S_DL_SFT 2 +#define NAU8824_I2S_DF_MASK (0x3) +#define NAU8824_I2S_DF_SFT 0 +#define NAU8824_I2S_DF_RIGHT (0x0) +#define NAU8824_I2S_DF_LEFT (0x1) +#define NAU8824_I2S_DF_I2S (0x2) +#define NAU8824_I2S_DF_PCM_A (0x3) +#define NAU8824_I2S_DF_PCM_B (0x3) + +/* reg. NAU8824_I2S_PCM_CTRL_2 (0x1D) */ +#define NAU8824_I2S_BCLKDIV_MASK 7 +#define NAU8824_I2S_BCLKDIV_SFT 0 + +#define NAU8824_I2S_MS_MASK (0x1 << 3) +#define NAU8824_I2S_MS_SFT 3 +#define NAU8824_I2S_MS_M (0x1 << 3) +#define NAU8824_I2S_MS_S (0x0 << 3) + +/* reg. NAU8824_DAC_MUTE_CTRL (0x31) */ +#define NAU8824_SOFT_MUTE_MASK (0x1 << 13) +#define NAU8824_SOFT_MUTE_SFT 13 +#define NAU8824_SOFT_MUTE_EN (0x1 << 13) +#define NAU8824_SOFT_MUTE_DIS (0x0 << 13) + +/* reg NAU8824_DAC0_DIG_VOL (0x32) */ +#define NAU8824_DAC0_GAIN_MASK (0x1FF << 0) + +/* reg NAU8824_DAC1_DIG_VOL (0x33) */ +#define NAU8824_DAC1_GAIN_MASK (0x1FF << 0) + +/* reg. NAU8824_CLASS_G_CTRL (0x50) */ +#define NAU8824_CLASS_G_CLK_SRC_MSK (0x3 << 14) +#define NAU8824_CLASS_G_CLK_DIS (0x3 << 14) +#define NAU8824_CLASS_G_CLK_SRC_MCLK (0x2 << 14) +#define NAU8824_CLASS_G_CLK_SRC_660K (0x1 << 14) +#define NAU8824_CLASS_G_CLK_SRC_2M (0x0 << 14) +#define NAU8824_CLASS_G_TIMER_MSK (0x3F << 8) +#define NAU8824_CLASS_G_TIMER_SFT 8 +#define NAU8824_CLASS_G_THRSLD_MSK (0x3 << 4) +#define NAU8824_CLASS_G_THRSLD_SFT 4 +#define NAU8824_CLASS_G_RIGHT_EN_MSK (0x1 << 2) +#define NAU8824_CLASS_G_RIGHT_EN (0x1 << 2) +#define NAU8824_CLASS_G_RIGHT_DIS (0x0 << 2) +#define NAU8824_CLASS_G_LEFT_EN_MSK (0x1 << 1) +#define NAU8824_CLASS_G_LEFT_EN (0x1 << 1) +#define NAU8824_CLASS_G_LEFT_DIS (0x0 << 1) +#define NAU8824_CLASS_G_EN_MSK (0x1 << 0) +#define NAU8824_CLASS_G_SHIFT 0 +#define NAU8824_CLASS_G_EN (0x1 << 0) +#define NAU8824_CLASS_G_DIS (0x0 << 0) + +/* reg NAU8824_SAR_ADC_OUTPUT (0x59) */ +#define NAU8824_ADC_OUTPUT_MASK (0xFF << 0) +#define NAU8824_ADC_OUTPUT_SFT 0 + +/* reg. NAU8824_BIAS_ADJ_2 (0x66) */ +#define NAU8824_VMID_MASK (0x1 << 6) +#define NAU8824_VMID_SFT 6 +#define NAU8824_VMID_EN (0x1 << 6) +#define NAU8824_VMID_DIS (0x0 << 6) +#define NAU8824_VMID_OPTION_MASK (0x1 << 4) +#define NAU8824_VMID_OPTION_SFT 4 +#define NAU8824_DMIC2_MASK (0x1 << 3) +#define NAU8824_DMIC2_SFT 3 +#define NAU8824_DMIC2_EN (0x1 << 3) +#define NAU8824_DMIC2_DIS (0x0 << 3) +#define NAU8824_DMIC1_MASK (0x1 << 2) +#define NAU8824_DMIC1_SFT 2 +#define NAU8824_DMIC1_EN (0x1 << 2) +#define NAU8824_DMIC1_DIS (0x0 << 2) + +/* reg. NAU8824_ANALOG_CTRL2 (0x6A) */ +#define NAU8824_CLASS_D_CLAMP_MSK (0x1 << 3) +#define NAU8824_CLASS_D_CLAMP_SFT 3 +#define NAU8824_CLASS_D_CLAMP_BETTER (0x1 << 3) +#define NAU8824_CLASS_D_CLAMP_DEFAULT (0x0 << 3) + +/* reg. NAU8824_HPO_MIXER (0x6B) */ +#define NAU8824_M_HPVOL_R_HM (0x1 << 2) +#define NAU8824_M_HPVOL_R_HM_SFT 2 +#define NAU8824_M_HPVOL_L_HM (0x1 << 0) +#define NAU8824_M_HPVOL_L_HM_SFT 0 + +/* reg. NAU8824_CLASS_D_GAIN1 (0x6D) */ +#define NAU8824_CLASS_D_GAIN_R_MSK (0x1F << 8) +#define NAU8824_CLASS_D_GAIN_R_SFT 8 +#define NAU8824_CLASS_D_MASK (0x1 << 7) +#define NAU8824_CLASS_D_SFT 7 +#define NAU8824_CLASS_D_EN_MSK (0x1 << 7) +#define NAU8824_CLASS_D_EN (0x1 << 7) +#define NAU8824_CLASS_D_DIS (0x0 << 7) + +/* reg. NAU8824_CLASS_D_GAIN2 (0x6E) */ +#define NAU8824_CLASS_D_GAIN_L_MSK (0x1F << 0) +#define NAU8824_CLASS_D_GAIN_L_SFT 0 + +/* reg. NAU8824_DAC_CTRL (0x73) */ +#define NAU8824_DAC_R_MASK (0x1 << 13) +#define NAU8824_DAC_R_SFT 13 +#define NAU8824_DAC_R_EN (0x1 << 13) +#define NAU8824_DAC_R_DIS (0x0 << 13) +#define NAU8824_DAC_L_MASK (0x1 << 12) +#define NAU8824_DAC_L_SFT 12 +#define NAU8824_DAC_L_EN (0x1 << 12) +#define NAU8824_DAC_L_DIS (0x0 << 12) + +/* reg. NAU8824_MIC_BIAS (0x74) */ +#define NAU8824_BIAS_POWER_MASK (0x1 << 8) +#define NAU8824_BIAS_POWER_SFT 8 +#define NAU8824_BIAS_POWER_ON (0x1 << 8) +#define NAU8824_BIAS_POWER_DOWN (0x0 << 8) +#define NAU8824_BIAS_LEVEL_MASK (0x7 << 0) +#define NAU8824_BIAS_LEVEL_SFT 0 +#define NAU8824_BIAS_LEVEL_VDDA (0x0 << 0) + +/* reg. NAU8824_HP_MUTE (0x75) */ +#define NAU8824_R_MUTE (0x1 << 14) +#define NAU8824_R_MUTE_SFT 14 +#define NAU8824_L_MUTE (0x1 << 6) +#define NAU8824_L_MUTE_SFT 6 + +/* reg. NAU8824_BOOST (0x76) */ +#define NAU8824_VMID_PRECHARGE_MASK (0x1 << 13) +#define NAU8824_VMID_PRECHARGE_SFT 13 +#define NAU8824_VMID_PRECHARGE_DIS (0x1 << 13) +#define NAU8824_VMID_PRECHARGE_EN (0x0 << 13) +#define NAU8824_G_BIAS_MASK (0x1 << 12) +#define NAU8824_G_BIAS_SFT 12 +#define NAU8824_G_BIAS_EN (0x1 << 12) +#define NAU8824_G_BIAS_DIS (0x0 << 12) + +/* reg. NAU8824_HP_VOL (0x7B) */ +#define NAU8824_R_VOL_MASK (0x1f << 8) +#define NAU8824_R_VOL_SFT 8 +#define NAU8824_SPK_L_VOL_MASK (0x1f) +#define NAU8824_SPK_R_VOL_MASK (0x1f << 8) +#define NAU8824_L_VOL_MASK (0x1f) +#define NAU8824_L_VOL_SFT 0 + +/* Volume Rescale */ +#define NAU8824_VOL_RSCL_MAX 0x1E +#define NAU8824_VOL_RSCL_RANGE 0x1E +#define NAU8824_SPK_VOL_RSCL_MAX 0x19 +#define NAU8824_SPK_VOL_RSCL_RANGE 0x19 +#define NAU8824_DAC_VOL_RSCL_RANGE 0x164 +#define NAU8824_ADC_VOL_RSCL_RANGE 0x164 + +/* Data Format */ +#define NAU8824_I2S_DL_32 (0x3 << 2) +#define NAU8824_I2S_DL_24 (0x2 << 2) +#define NAU8824_I2S_DL_20 (0x1 << 2) +#define NAU8824_I2S_DL_16 (0x0 << 2) + +#define NAU8824_FREQ_25000000 25000000 +#define NAU8824_PLL_CLKIN_MCLK 0 +#define NAU8824_BUTTONPRESS_MASK 0x20 +#define NAU8824_HSPLUG_MASK 0x10 + +enum { + NAU8824_INTERNALCLOCK = 0, + NAU8824_MCLK, +}; + +enum { + NAU8824_J_IN_EVENT, + NAU8824_J_OUT_EVENT, + NAU8824_BP_EVENT, + NAU8824_BR_EVENT, + NAU8824_UN_EVENT, +}; + +enum { + S_NAU8824_J_OUT, + S_NAU8824_J_IN_TEMP, + S_NAU8824_J_IN, + S_NAU8824_BUTTON_DECT, + S_NAU8824_NO_BUTTON_DECT, +}; + +#define NAU8824_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8824_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct nau8824_jack_data { + struct snd_soc_jack *jack; + int report; +}; + +struct nau8824_gpio_setup { + unsigned int reg; + u8 value; +}; + +struct nau8824_pdata { + unsigned int audio_mclk1; + unsigned int gpio_irq; + int naudint_irq; + int headset_detect; + int button_press_detect; +}; + +struct nau8824_priv { + struct nau8824_jack_data hs_jack; + struct workqueue_struct *workqueue; + struct delayed_work delayed_work; + struct snd_soc_codec *codec; + u8 i2c_regs_status; + struct device *dev; + struct regmap *regmap; + struct mutex mutex; + struct nau8824_pdata pdata; + unsigned int irq; + bool jd_status; + bool bp_status; + int jack_type; +}; + +int nau8824_device_init(struct nau8824_priv *nau8824); +void nau8824_device_exit(struct nau8824_priv *nau8824); +void nau8824_enable_mic_bias(struct snd_soc_codec *codec, int enable); +int nau8824_query_jack_status(struct snd_soc_codec *codec); +int nau8824_query_btn_press(struct snd_soc_codec *codec); +void nau8824_btn_press_intr_enable(struct snd_soc_codec *codec, int enable); + +#endif /* _NAU8824_H */ -- 1.8.1.2

3 7

[alsa-devel] [PATCH] ASoC: atmel_pcm: use endian agnostic IO accessors
by Ben Dooks 26 Mar '15

26 Mar '15

The atmel-pcm.h is using __raw_readl and __raw_writel for IO accessors which do not work if the processor is in big endian mode (the peripheral will still expect data in little endian format). Note, I do not think this is being used in the AVR32 which would require a different fix (possible after atmel_io.h is merged). This has not been run-tested as I do not have a board with an audio code on. Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> -- CC: Bo Shen <voice.shen(a)atmel.com> CC: Liam Girdwood <lgirdwood(a)gmail.com> CC: Mark Brown <broonie(a)kernel.org> CC: Jaroslav Kysela <perex(a)perex.cz> CC: Takashi Iwai <tiwai(a)suse.de> CC: alsa-devel(a)alsa-project.org --- sound/soc/atmel/atmel-pcm.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h index 12ae814..8f5f551 100644 --- a/sound/soc/atmel/atmel-pcm.h +++ b/sound/soc/atmel/atmel-pcm.h @@ -80,8 +80,8 @@ struct atmel_pcm_dma_params { /* * SSC register access (since ssc_writel() / ssc_readl() require literal name) */ -#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) -#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) +#define ssc_readx(base, reg) (readl_relaxed((base) + (reg))) +#define ssc_writex(base, reg, value) writel_relaxed((value), (base) + (reg)) int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd); void atmel_pcm_free(struct snd_pcm *pcm); -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: davinci-evm: drop un-necessary remove function
by Jyri Sarha 26 Mar '15

26 Mar '15

From: Manish Badarkhe <manishvb(a)ti.com> As davinci card gets registered using 'devm_' api there is no need to unregister the card in 'remove' function. Hence drop the 'remove' function. Signed-off-by: Manish Badarkhe <manishvb(a)ti.com> Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/davinci/davinci-evm.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index b6bb594..8c2b9be 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -425,18 +425,8 @@ static int davinci_evm_probe(struct platform_device *pdev) return ret; } -static int davinci_evm_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static struct platform_driver davinci_evm_driver = { .probe = davinci_evm_probe, - .remove = davinci_evm_remove, .driver = { .name = "davinci_evm", .pm = &snd_soc_pm_ops, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: snd_soc_tas5086: Reinit register values on probe
by Pascal Huerst 26 Mar '15

26 Mar '15

If the machine driver has been un/reloaded, the register values of the codec driver have to be reinitialized in order to run properly. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- sound/soc/codecs/tas5086.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 249ef5c..cd19171 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -851,10 +851,16 @@ static int tas5086_probe(struct snd_soc_codec *codec) } tas5086_reset(priv); + regcache_mark_dirty(priv->regmap); + ret = tas5086_init(codec->dev, priv); if (ret < 0) goto exit_disable_regulators; + ret = regcache_sync(priv->regmap); + if (ret < 0) + goto exit_disable_regulators; + /* set master volume to 0 dB */ ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30); if (ret < 0) -- 1.9.3

2 4

[alsa-devel] [PATCH] ALSA: hda - Add one more node in the EAPD supporting candidate list
by Hui Wang 26 Mar '15

26 Mar '15

We have a HP machine which use the codec node 0x17 connecting the internal speaker, and from the node capability, we saw the EAPD, if we don't set the EAPD on for this node, the internal speaker can't output any sound. Cc: <stable(a)vger.kernel.org> BugLink: https://bugs.launchpad.net/bugs/1436745 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- picked some content from the alsa-info.txt of this machine: Codec: Realtek ALC221 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0221 Subsystem Id: 0x103c8105 Revision Id: 0x100103 No Modem Function Group found ... ... ... Node 0x17 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00] Pincap 0x00010010: OUT EAPD EAPD 0x0: Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x0f sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8775389..75880fe 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -396,7 +396,7 @@ static void alc_auto_setup_eapd(struct hda_codec *codec, bool on) { /* We currently only handle front, HP */ static hda_nid_t pins[] = { - 0x0f, 0x10, 0x14, 0x15, 0 + 0x0f, 0x10, 0x14, 0x15, 0x17, 0 }; hda_nid_t *p; for (p = pins; *p; p++) -- 1.9.1

2 1

[alsa-devel] Add AZX_DCAPS_I915_POWEREWELL on SKL is OK?
by Yang, Libin 26 Mar '15

26 Mar '15

Hi Takashi, Is it OK to add AZX_DCAPS_I915_POWEREWELL for SKL in hda_intel.c? SKL HDA controller is not in powerwell domain. But its HDMI codec is in the domain. My understanding is if the flag is added here, it means the controller is in the domain. But I can't find a good place to add a flag to indicate the HDMI codec need the power domain. Regards, Libin

2 2

[alsa-devel] Fix a small typo in manpage cspctl.1 of alsa-tools/sb16_csp
by saulery＠legtux.org 26 Mar '15

26 Mar '15

Hello, I found an error in man pages cspctl.1. Could you please integrate this patch ? Regards,

2 2

[alsa-devel] [PATCH] ALSA: hda_intel: apply the Seperate stream_tag for Sunrise Point
by libin.yang＠intel.com 26 Mar '15

26 Mar '15

From: Libin Yang <libin.yang(a)intel.com> The total stream number of Sunrise Point's input and output stream exceeds 15, which will cause some streams do not work because of the overflow on SDxCTL.STRM field if using the legacy stream tag allocation method. This patch uses the new stream tag allocation method by add the flag AZX_DCAPS_SEPARATE_STREAM_TAG for Skylake platform. Signed-off-by: Libin Yang <libin.yang(a)intel.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 feebc1d..15a8299 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1963,7 +1963,7 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Sunrise Point */ { PCI_DEVICE(0x8086, 0xa170), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, -- 1.9.1

2 1

[alsa-devel] [PATCH 0/5] ASoC: rsnd: DPCM base sampling rate convert support
by Kuninori Morimoto 26 Mar '15

26 Mar '15

Hi Mark These are remains of my previous patch set. Basically, these doesn't need soc-core's new feature, but it is assuming DPCM base sampling rate convert has both playback/capture in same time. 1 - 2: new Renesas sound card for DPCM 3 : bug fix patch. it can be applied in linus/master branch 4 - 5: new feature, but based on 3) patch Kuninori Morimoto (5): 1) ASoC: rsrc-card: add Renesas sampling rate convert sound card support 2) ASoC: rsrc-card: add .be_hw_params_fixup support for convert rate 3) ASoC: rsnd: call clk_prepare_enable/unprepare() in probe/remove 4) ASoC: rsnd: remove useless debug message 5) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 67 +++++++++++++ sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 - sound/soc/sh/rcar/core.c | 25 ++++- sound/soc/sh/rcar/dma.c | 4 - sound/soc/sh/rcar/dvc.c | 28 +----- sound/soc/sh/rcar/gen.c | 4 - sound/soc/sh/rcar/rsnd.h | 12 ++- sound/soc/sh/rcar/rsrc-card.c | 512 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/src.c | 84 +++++++++------- sound/soc/sh/rcar/ssi.c | 31 ++---- 12 files changed, 683 insertions(+), 96 deletions(-)

1 6

[alsa-devel] RawMIDI behaviour with MidiFace 4x4
by Kalvas, Taneli 25 Mar '15

25 Mar '15

Dear All, I am developing a patch librarian for a synthesizer. Because the communication happens with Midi sysex and timing is not relevant I chose RawMIDI interface and not the alsa sequencer. The problem I have encountered is that writing to the midi port with snd_rawmidi_write() seems to broken. The function accepts any size of data block and claims to have sent everything. In reality only a part of the message is sent. For a large block size I can see the activity light blinking for fraction of a second. This happens both in blocking and non-blocking mode. I would expect the snd_rawmidi_write() to accept only a part of the message. My hardware is Miditech Midiface 4x4, a USB midi interface: $ cat /proc/asound/cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xe0610000 irq 45 1 [MIDI4x4 ]: USB-Audio - MIDI4x4 MIDIPLUS MIDI4x4 at usb-0000:00:14.0-1, full speed usbid: 1acc:1a0b My alsa is quite recent (aplay shows version 1.0.27.2), I am running Ubuntu with 3.13.0-39-generic kernel. I have tested the system by connecting the output port to an input port on the hardware and having another program output the received bytes on stdout. My observations: - As long as the buffer size is 276 bytes or less it gets sent ok - Larger buffer sizes cause corruption and roughly 300 bytes of received. - Alsa reports midi ring buffer size of 4096. - snd_rawmidi_status_get_xruns() on both ends reports 0 errors. - If I write N=128 byte blocks and usleep for 352*N after each write, everything works ok. Is this due to hardware incompatibility with alsa driver or what is going on? Is there a way to connect two programs via a virtual RawMIDI port so that I could test without using the hardware? Any other diagnostics that I could do? My test programs: write.cpp: int main( void ) { int intstatus; snd_rawmidi_t *midiout; if( (intstatus = snd_rawmidi_open(NULL, &midiout, "hw:1,0,1", 0)) ) { std::cerr << "unable to open MIDI out port: " << snd_strerror(intstatus) << "\n"; exit( 1 ); } int N = 3000; // 276 bytes gets through ok. unsigned char msg[N]; for( int a = 0; a < N; a++ ) { msg[a] = a % 0x80; } msg[0] = 0xF0; msg[N-1] = 0xF7; ssize_t wrstatus; if( (wrstatus = snd_rawmidi_write(midiout, msg, N)) < 0 ) { std::cerr << "unable to write MIDI port: " << snd_strerror(wrstatus) << "\n"; return( -1 ); } std::cout << "write status = " << wrstatus << "\n"; if( (intstatus = snd_rawmidi_drain( midiout )) ) { std::cerr << "unable to drain: " << snd_strerror(intstatus) << "\n"; exit( 1 ); } // Sleep for the amount of time needed for transfer + 10 % extra + 100 ms usleep( 352*N + 100000 ); snd_rawmidi_status_t *stat; snd_rawmidi_status_malloc( &stat ); snd_rawmidi_status( midiout, stat ); int err = snd_rawmidi_status_get_xruns( stat ); std::cout << "xruns = " << err << "\n"; snd_rawmidi_close( midiout ); return( 0 ); } -------------------- read.cpp: bool quit = false; void signal_int( int ) { quit = true; } int main( void ) { int mode = SND_RAWMIDI_NONBLOCK; int intstat; std::string portname = "hw:1,0,0"; //std::string portname = "virtual"; snd_rawmidi_t* midiin; if( (intstat = snd_rawmidi_open(&midiin, NULL, portname.c_str(), mode)) < 0 ) { std::cerr << "unable to open MIDI port " << portname << ": " << snd_strerror(intstat) << "\n"; exit( 1 ); } int count = 0; unsigned char buf[1]; std::cout << std::dec << std::setfill(' ') << std::setw(6) << count << " "; signal( SIGINT, signal_int ); while( !quit ) { int read = snd_rawmidi_read( midiin, buf, 1 ); if( read == 0 || read == -EBUSY || read == -EAGAIN ) { usleep( 320 ); continue; } else if( read < 0 ) { std::cerr << "unable to read MIDI port" << portname << ": " << snd_strerror(read) << "\n"; exit( 1 ); } std::cout << std::hex << std::setfill('0') << "0x" << std::setw(2) << (int)buf[0] << " " << std::flush; count++; if( count % 8 == 0 ) { std::cout << "\n"; std::cout << std::dec << std::setfill(' ') << std::setw(6) << count << " " << std::flush; } } snd_rawmidi_status_t *stat; snd_rawmidi_status_malloc( &stat ); snd_rawmidi_status( midiin, stat ); int err = snd_rawmidi_status_get_xruns( stat ); std::cout << "\nxruns = " << err << "\n"; snd_rawmidi_close( midiin ); return( 0 ); } Thanks. Taneli -- Taneli Kalvas Ph.D., Researcher Department of Physics, room FL114 P.O. Box 35 (YFL) 40014 University of Jyväskylä, Finland Mobile: +358-44-314-1602 Fax: +358-14-617-411 Email: taneli.kalvas(a)jyu.fi

3 12

[alsa-devel] [PATCH resend] ASoC: fsl-ssi: Fix bitclock calculation for master mode
by Marc Kleine-Budde 25 Mar '15

25 Mar '15

From: Sascha Hauer <s.hauer(a)pengutronix.de> in fsl_ssi_set_bclk we are looking for the best divider by iterating over all possible dividers. The divider is in the range 1..256, so make this clear by using 'div' as name for the loop counter and iterate from 1 to 256 instead of 0 to 255. This makes obvious that tmprate is calculated wrong, it has to be freq * factor * div, not freq * factor * (i + 2). Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> Signed-off-by: Marc Kleine-Budde <mkl(a)pengutronix.de> --- sound/soc/fsl/fsl_ssi.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b9fabbf69db6..b8af08e49085 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -578,7 +578,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); struct regmap *regs = ssi_private->regs; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret; - u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; + u32 pm = 999, div2, psr, stccr, mask, afreq, factor, div; unsigned long clkrate, baudrate, tmprate; u64 sub, savesub = 100000; unsigned int freq; @@ -602,8 +602,8 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, factor = (div2 + 1) * (7 * psr + 1) * 2; - for (i = 0; i < 255; i++) { - tmprate = freq * factor * (i + 2); + for (div = 1; div < 256; div++) { + tmprate = freq * factor * div; if (baudclk_is_used) clkrate = clk_get_rate(ssi_private->baudclk); @@ -618,7 +618,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, continue; clkrate /= factor; - afreq = clkrate / (i + 1); + afreq = clkrate / div; if (freq == afreq) sub = 0; @@ -636,7 +636,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, if (sub < savesub) { baudrate = tmprate; savesub = sub; - pm = i; + pm = div - 1; } /* We are lucky */ -- 2.1.4

2 2

[alsa-devel] [PATCH resend] ASoC: wm8741: Fix rates constraints values
by Sergej Sawazki 25 Mar '15

25 Mar '15

The WM8741 DAC supports the following typical audio sampling rates: 44.1kHz, 88.2kHz, 176.4kHz (eg: with a master clock of 22.5792MHz) 32kHz, 48kHz, 96kHz, 192kHz (eg: with a master clock of 24.576MHz) For the rates lists, we should use 82000 instead of 88235, 176400 instead of 1764000 and 192000 instead of 19200 (seems to be a typo). Signed-off-by: Sergej Sawazki <ce3a(a)gmx.de> --- sound/soc/codecs/wm8741.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 31bb480..9e71c76 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -123,7 +123,7 @@ static struct { }; static const unsigned int rates_11289[] = { - 44100, 88235, + 44100, 88200, }; static const struct snd_pcm_hw_constraint_list constraints_11289 = { @@ -150,7 +150,7 @@ static const struct snd_pcm_hw_constraint_list constraints_16384 = { }; static const unsigned int rates_16934[] = { - 44100, 88235, + 44100, 88200, }; static const struct snd_pcm_hw_constraint_list constraints_16934 = { @@ -168,7 +168,7 @@ static const struct snd_pcm_hw_constraint_list constraints_18432 = { }; static const unsigned int rates_22579[] = { - 44100, 88235, 1764000 + 44100, 88200, 176400 }; static const struct snd_pcm_hw_constraint_list constraints_22579 = { @@ -186,7 +186,7 @@ static const struct snd_pcm_hw_constraint_list constraints_24576 = { }; static const unsigned int rates_36864[] = { - 48000, 96000, 19200 + 48000, 96000, 192000 }; static const struct snd_pcm_hw_constraint_list constraints_36864 = { -- 1.9.1

3 2

[alsa-devel] [PATCH v3] ASoC: dapm: Fix build warning
by Charles Keepax 25 Mar '15

25 Mar '15

commit c66150824b8a ("ASoC: dapm: add code to configure dai link parameters") introduced the following build warning: sound/soc/soc-dapm.c: In function 'snd_soc_dapm_new_pcm': sound/soc/soc-dapm.c:3389:4: warning: passing argument 1 of 'snprintf' discards 'const' qualifier from pointer target type snprintf(w_param_text[count], len, This patch fixes this by switching to using devm_kasprintf. This also saves a couple of lines of code. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v2: - Use devm_kasprintf sound/soc/soc-dapm.c | 13 +++++-------- 1 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 049b300..4bd2a6f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3346,7 +3346,6 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, { struct snd_soc_dapm_widget template; struct snd_soc_dapm_widget *w; - size_t len; char *link_name; int ret, count; unsigned long private_value; @@ -3366,28 +3365,26 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w_param_text) return -ENOMEM; - len = strlen(source->name) + strlen(sink->name) + 2; - link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); + link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", + source->name, sink->name); if (!link_name) { ret = -ENOMEM; goto outfree_w_param; } - snprintf(link_name, len, "%s-%s", source->name, sink->name); for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { dev_warn(card->dapm.dev, "ASoC: anonymous config %d for dai link %s\n", count, link_name); - len = strlen("Anonymous Configuration ") + 3; w_param_text[count] = - devm_kzalloc(card->dev, len, GFP_KERNEL); + devm_kasprintf(card->dev, GFP_KERNEL, + "Anonymous Configuration %d", + count); if (!w_param_text[count]) { ret = -ENOMEM; goto outfree_link_name; } - snprintf(w_param_text[count], len, - "Anonymous Configuration %d", count); } else { w_param_text[count] = devm_kmemdup(card->dev, config->stream_name, -- 1.7.2.5

2 1

[alsa-devel] [ASoC] Free kcontrol memory while unregistering codec
by Vishal T 25 Mar '15

25 Mar '15

Hi, While going through the codec cleanup code (snd_soc_unregister_codec()), I could not find the code which frees memory allocated for snd_kcontrol (while adding the kcontrols to card). Am I missing anything obvious? Sorry if I have created noise. Thanks, V

1 0

[alsa-devel] [PATCH] ASoC: DAPM: Fix build warning
by Charles Keepax 25 Mar '15

25 Mar '15

commit c66150824b8a ("ASoC: dapm: add code to configure dai link parameters") introduced the following build warning: sound/soc/soc-dapm.c: In function 'snd_soc_dapm_new_pcm': sound/soc/soc-dapm.c:3389:4: warning: passing argument 1 of 'snprintf' discards 'const' qualifier from pointer target type snprintf(w_param_text[count], len, This patch fixes this by adding a non-const temporary variable to hold the value as it is created before assigning to the entry in w_param_text. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 10 ++++++---- 1 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 049b300..7640370 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3376,18 +3376,20 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { + char *anon_name; + dev_warn(card->dapm.dev, "ASoC: anonymous config %d for dai link %s\n", count, link_name); len = strlen("Anonymous Configuration ") + 3; - w_param_text[count] = - devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!w_param_text[count]) { + anon_name = devm_kzalloc(card->dev, len, GFP_KERNEL); + if (!anon_name) { ret = -ENOMEM; goto outfree_link_name; } - snprintf(w_param_text[count], len, + snprintf(anon_name, len, "Anonymous Configuration %d", count); + w_param_text[count] = anon_name; } else { w_param_text[count] = devm_kmemdup(card->dev, config->stream_name, -- 1.7.2.5

2 2

[alsa-devel] [RFC v2 0/3] ALSA: hda: add hda controller to hda core
by Vinod Koul 25 Mar '15

25 Mar '15

This series starts adding base hda controller code to hda core. There are few more changes like dsp loader etc planned to be added as well. This is based on Takashi's test/hda-core branch with patches posted last week, we have modified naming convention loosely based on takashi's work, Takshi feel free to recommend more changes if you think they are required Jeeja KP (2): ALSA: hda: move common hda controller register defines up ALSA: hda: add hda controller to hda Vinod Koul (1): ALSA: hda: add HDA_MAX_CODECS include/sound/hda_registers.h | 200 +++++++ include/sound/hda_verbs.h | 1 + include/sound/hdaudio.h | 215 +++++++ sound/hda/Makefile | 2 +- sound/hda/hdac_controller.c | 1284 ++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_controller.c | 4 +- sound/pci/hda/hda_controller.h | 203 +------ 7 files changed, 1707 insertions(+), 202 deletions(-) create mode 100644 include/sound/hda_registers.h create mode 100644 sound/hda/hdac_controller.c -- 1.7.9.5

2 9

[alsa-devel] [PATCH v2] ASoC: dapm: Fix build warning
by Charles Keepax 25 Mar '15

25 Mar '15

commit c66150824b8a ("ASoC: dapm: add code to configure dai link parameters") introduced the following build warning: sound/soc/soc-dapm.c: In function 'snd_soc_dapm_new_pcm': sound/soc/soc-dapm.c:3389:4: warning: passing argument 1 of 'snprintf' discards 'const' qualifier from pointer target type snprintf(w_param_text[count], len, This patch fixes this by adding a non-const temporary variable to hold the value as it is created before assigning to the entry in w_param_text. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Just small fix to commit message sound/soc/soc-dapm.c | 10 ++++++---- 1 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 049b300..7640370 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3376,18 +3376,20 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { + char *anon_name; + dev_warn(card->dapm.dev, "ASoC: anonymous config %d for dai link %s\n", count, link_name); len = strlen("Anonymous Configuration ") + 3; - w_param_text[count] = - devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!w_param_text[count]) { + anon_name = devm_kzalloc(card->dev, len, GFP_KERNEL); + if (!anon_name) { ret = -ENOMEM; goto outfree_link_name; } - snprintf(w_param_text[count], len, + snprintf(anon_name, len, "Anonymous Configuration %d", count); + w_param_text[count] = anon_name; } else { w_param_text[count] = devm_kmemdup(card->dev, config->stream_name, -- 1.7.2.5

1 0

[alsa-devel] Missing beep controls for Via HDA codec (snd_hda_codec_via)
by W. Trevor King 25 Mar '15

25 Mar '15

My VT1802 chip has a beep-generating node: $ cat /proc/asound/card1/codec#0 Codec: VIA VT1802 ... Vendor Id: 0x11068446 Subsystem Id: 0x15587410 Revision Id: 0x100000 ... Node 0x22 [Beep Generator Widget] wcaps 0x70040c: Mono Amp-Out Amp-Out caps: ofs=0x0a, nsteps=0x12, stepsize=0x05, mute=1 Amp-Out vals: [0x0a] Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 ... But I'm missing the: Control: name=... entries that I think I need to manage this widget from alsamixer. For example, sambrian posts codec output for a CX20561 chip which has [1]: Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] I can patch with something like this: diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 3de6d3d..85a0c08 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1520,8 +1520,10 @@ static int patch_vt2002P(struct hda_codec *codec) spec->gen.mixer_nid = 0x21; override_mic_boost(codec, 0x2b, 0, 3, 40); override_mic_boost(codec, 0x29, 0, 3, 40); - if (spec->codec_type == VT1802) + if (spec->codec_type == VT1802) { + spec->gen.beep_nid = 0x22; fix_vt1802_connections(codec); + } add_secret_dac_path(codec); snd_hda_pick_fixup(codec, NULL, vt2002p_fixups, via_fixups); to get a dmesg entry like: input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/sound/card1/hdaudioC1D0/input15 but that doesn't add the beep controls. Looking at patch_analog.c, patch_conexant.c, and patch_realtek.c for inspiration, I expect we'll need an HDA_CODEC_MUTE_BEEP(_MONO) entry, and possibly an HDA_CODEC_VOLUME(_MONO) entry, but I'm not sure if these should be mono or not, or if they should be inputs (following patch_realtek.c) or outputs (following patch_analog.c and patch_conexant.c). I've filled out Via's datasheet-request form [2], but I don't know how likely that is to be accepted. Does anyone have a VT1802 datasheet that they're allowed to share (or some free time to fill in Via beep support on their own)? Or am I barking up the wrong tree? My only previous experience with this code-base was a beep-tweak for a previous box's Realtek chip [3], but I love my beeps and am happy to poke around with a datasheet until I get this working ;). Maybe folks with more experience can just guess the right code without needing to see specs for the VT1802? Cheers, Trevor [1]: http://forums.fedoraforum.org/archive/index.php/t-265822.html [2]: http://www.via.com.tw/en/support/contact_us/ [3]: http://thread.gmane.org/gmane.linux.alsa.devel/121300 -- This email may be signed or encrypted with GnuPG (http://www.gnupg.org). For more information, see http://en.wikipedia.org/wiki/Pretty_Good_Privacy

3 11

Re: [alsa-devel] ASoC: Max98090 Codec DAPM Widget setup in Open Embedded
by noman pouigt 25 Mar '15

25 Mar '15

On Mon, Mar 23, 2015 at 11:53 PM, noman pouigt <variksla(a)gmail.com> wrote: >> I would like to know which file I can use to connect/enable >> DPAM widgets permanently so a simple aplay/record can work? >> >> My environment is: >> - Yocto Linux Kernel 3.10 - (DTS based) >> - Max98090 Codec >> - imx6SL SoC >> >> I thought about doing this in alsa.conf or asound.conf but I couldn't >> find a single example with DAPM widgets . > > I am also stuck with the same problem. I am sure ALSA experts > can help here. > > What i want to know is how to set the dapm widgets in open > embedded? Is it alsa.conf or asound.conf ? > > I need to set something like this: > amixer cset name="LEFT MIXER DAC" 1 > > Using amixer i am able to playback but i don't know how can i > modify conf file to have this dapm widgets associated with > a device. > > Any example would suffice.

4 3

[alsa-devel] [PATCH v3 0/2] ALSA: jack: Refactoring for jack kctls
by Jie Yang 25 Mar '15

25 Mar '15

Currently only hda will create kctls for hda jacks, for asoc, people may need create jack kctls in specific driver. Here we are introducing kctls for each jack, by creating kctls and add them to jack controls list (considering exist of combo jack). At the same time, we will report events for each control in the list. With this implement, we are no longer need hda jack specific kctls, so here also remove jack kctls for hda. Changes in v3: 1. replace bit index with bit mask in jack_kctl; 2. add exception for SND_JACK_HEADSET and SND_JACK_AVOUT, only create one jack kctl for these two combo jacks, respectively. 3. add NULL check, mem kfree, and fix some potential risk. Change in v2: 1. define jack_kctl struct, and put jack kctl related stuff there; 2. add a patch to remove the existing controls for HDA jack. Jie Yang (2): ALSA: jack: create jack kcontrols for every jack input device ALSA: hda - Remove jack kctls include/sound/jack.h | 8 ++++ sound/core/Kconfig | 3 -- sound/core/Makefile | 3 +- sound/core/jack.c | 118 ++++++++++++++++++++++++++++++++++++++++++++++- sound/pci/hda/Kconfig | 10 +--- sound/pci/hda/hda_jack.c | 45 ++---------------- sound/pci/hda/hda_jack.h | 3 -- 7 files changed, 129 insertions(+), 61 deletions(-) -- 1.9.1

7 19

[alsa-devel] 4.0-rc3: sound broken in chromium but not in mpg123
by Pavel Machek 24 Mar '15

24 Mar '15

Hi! Intel desktop machine: 00:1b.0 Audio device: Intel Corporation NM10/ICH7 Family High Definition Audio Controller (rev 01) Sound worked ok in both in previous kernel versions. Any ideas? Thanks, Pavel -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

2 6

[alsa-devel] [PATCH] ASoC: soc-core: call soc_cleanup_card_debugfs() from snd_soc_unregister_card()
by Kuninori Morimoto 24 Mar '15

24 Mar '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Card debugfs is created in snd_soc_register_card(), but soc_cleanup_card_debugfs() is called from soc_cleanup_card_resources(). Cleanup function should be called from paired unregister function. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c26074..211783f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1751,8 +1751,6 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) /* remove and free each DAI */ soc_remove_dai_links(card); - soc_cleanup_card_debugfs(card); - /* remove the card */ if (card->remove) card->remove(card); @@ -2447,6 +2445,7 @@ int snd_soc_unregister_card(struct snd_soc_card *card) card->instantiated = false; snd_soc_dapm_shutdown(card); soc_cleanup_card_resources(card); + soc_cleanup_card_debugfs(card); dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); } -- 1.9.1

3 7

[alsa-devel] [PATCH v3] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used
by Jyri Sarha 24 Mar '15

24 Mar '15

Set rule constraints to allow only combinations of sample-rate, sample-format, and channels counts that can be played/captured with reasonable sample-rate accuracy. The logic with tdm-slots and serializers (=i2s data wires) goes like this: The first wire will take all channels up to number of tdm-slots, before following wires (if any) are used. If the first wire is used fully, the remaining wires share the same clocks and the divider can be calculated for the first wire. Also, takes the number of tdm-slots into account when implicitly selecting the BLCK divider. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Previous version of the patch: http://mailman.alsa-project.org/pipermail/alsa-devel/2015-March/089319.html Since v2 I changed the SNDRV_PCM_HW_PARAM_SAMPLE_BITS rule in the to SNDRV_PCM_HW_PARAM_FORMAT rule. Somehow the constraints with a SNDRV_PCM_HW_PARAM_SAMPLE_BITS rule using snd_interval_list() converged to an empty set when ever I tried to play a format that could not pass the rule with any channel count or rate, resulting all too familiar assertion in alsa-lib. However, the direct tampering with SNDRV_PCM_HW_PARAM_FORMAT mask appears to work as it should. In addition to SAMPLE_BITS- to FORMAT-rule change I optimized the rate rule a bit so that it does not try the rates that are outside current range. According to my tests this does not affect the behavior in any other way but slightly speeds up the decision making. sound/soc/davinci/davinci-mcasp.c | 207 ++++++++++++++++++++++++++++++++++++-- 1 file changed, 197 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d40b392..76156d18 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include <linux/of_platform.h> #include <linux/of_device.h> #include <linux/platform_data/davinci_asp.h> +#include <linux/math64.h> #include <sound/asoundef.h> #include <sound/core.h> @@ -65,6 +66,11 @@ struct davinci_mcasp_context { bool pm_state; }; +struct davinci_mcasp_ruledata { + struct davinci_mcasp *mcasp; + int serializers; +}; + struct davinci_mcasp { struct snd_dmaengine_dai_dma_data dma_data[2]; void __iomem *base; @@ -99,6 +105,8 @@ struct davinci_mcasp { #ifdef CONFIG_PM_SLEEP struct davinci_mcasp_context context; #endif + + struct davinci_mcasp_ruledata ruledata[2]; }; static inline void mcasp_set_bits(struct davinci_mcasp *mcasp, u32 offset, @@ -868,6 +876,30 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, return 0; } +static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, + unsigned int bclk_freq, + int *error_ppm) +{ + int div = mcasp->sysclk_freq / bclk_freq; + int rem = mcasp->sysclk_freq % bclk_freq; + + if (rem != 0) { + if (div == 0 || + ((mcasp->sysclk_freq / div) - bclk_freq) > + (bclk_freq - (mcasp->sysclk_freq / (div+1)))) { + div++; + rem = rem - bclk_freq; + } + } + if (error_ppm) + *error_ppm = + (div*1000000 + (int)div64_long(1000000LL*rem, + (int)bclk_freq)) + /div - 1000000; + + return div; +} + static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -883,16 +915,20 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, * the machine driver, we need to calculate the ratio. */ if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { - unsigned int bclk_freq = snd_soc_params_to_bclk(params); - unsigned int div = mcasp->sysclk_freq / bclk_freq; - if (mcasp->sysclk_freq % bclk_freq != 0) { - if (((mcasp->sysclk_freq / div) - bclk_freq) > - (bclk_freq - (mcasp->sysclk_freq / (div+1)))) - div++; - dev_warn(mcasp->dev, - "Inaccurate BCLK: %u Hz / %u != %u Hz\n", - mcasp->sysclk_freq, div, bclk_freq); - } + int channels = params_channels(params); + int rate = params_rate(params); + int sbits = params_width(params); + int ppm, div; + + if (channels > mcasp->tdm_slots) + channels = mcasp->tdm_slots; + + div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*channels, + &ppm); + if (ppm) + dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", + ppm); + __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); } @@ -974,6 +1010,120 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, return ret; } +static const unsigned int davinci_mcasp_dai_rates[] = { + 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, + 88200, 96000, 176400, 192000, +}; + +#define DAVINCI_MAX_RATE_ERROR_PPM 1000 + +static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_interval *ri = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int sbits = params_width(params); + int channels = params_channels(params); + unsigned int list[ARRAY_SIZE(davinci_mcasp_dai_rates)]; + int i, count = 0; + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (i = 0; i < ARRAY_SIZE(davinci_mcasp_dai_rates); i++) { + if (ri->min <= davinci_mcasp_dai_rates[i] && + ri->max >= davinci_mcasp_dai_rates[i]) { + uint bclk_freq = sbits*channels* + davinci_mcasp_dai_rates[i]; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = davinci_mcasp_dai_rates[i]; + } + } + dev_dbg(rd->mcasp->dev, + "%d frequencies (%d-%d) for %d sbits and %d channels\n", + count, ri->min, ri->max, sbits, channels); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_mask nfmt; + int rate = params_rate(params); + int channels = params_channels(params); + int i, count = 0; + + snd_mask_none(&nfmt); + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { + if (snd_mask_test(fmt, i)) { + uint bclk_freq = snd_pcm_format_width(i)*channels*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { + snd_mask_set(&nfmt, i); + count++; + } + } + } + dev_dbg(rd->mcasp->dev, + "%d possible sample format for %d Hz and %d channels\n", + count, rate, channels); + + return snd_mask_refine(fmt, &nfmt); +} + +static int davinci_mcasp_hw_rule_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_interval *ci = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + int sbits = params_width(params); + int rate = params_rate(params); + int max_chan_per_wire = rd->mcasp->tdm_slots < ci->max ? + rd->mcasp->tdm_slots : ci->max; + unsigned int list[ci->max - ci->min + 1]; + int c1, c, count = 0; + + for (c1 = ci->min; c1 <= max_chan_per_wire; c1++) { + uint bclk_freq = c1*sbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { + /* If we can use all tdm_slots, we can put any + amount of channels to remaining wires as + long as they fit in. */ + if (c1 == rd->mcasp->tdm_slots) { + for (c = c1; c <= rd->serializers*c1 && + c <= ci->max; c++) + list[count++] = c; + } else { + list[count++] = c1; + } + } + } + dev_dbg(rd->mcasp->dev, + "%d possible channel counts (%d-%d) for %d Hz and %d sbits\n", + count, ci->min, ci->max, rate, sbits); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + static int davinci_mcasp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { @@ -999,6 +1149,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->serial_dir[i] == dir) max_channels++; } + mcasp->ruledata[dir].serializers = max_channels; max_channels *= mcasp->tdm_slots; /* * If the already active stream has less channels than the calculated @@ -1013,6 +1164,42 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, max_channels); + + /* + * If we rely on implicit BCLK divider setting we should + * set constraints based on what we can provide. + */ + if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { + int ret; + + mcasp->ruledata[dir].mcasp = mcasp; + + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + davinci_mcasp_hw_rule_rate, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_FORMAT, + davinci_mcasp_hw_rule_format, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + davinci_mcasp_hw_rule_channels, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (ret) + return ret; + } + return 0; } -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: rsnd: Fix duplicate const for DVC ramp rates
by Krzysztof Kozlowski 24 Mar '15

24 Mar '15

Replace duplicated const keyword for 'dvc_ramp_rate' with proper array of const pointers to const strings. Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> --- sound/soc/sh/rcar/dvc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index aeeef1352eee..6d85e36c6c2b 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -36,7 +36,7 @@ struct rsnd_dvc { ((pos) = (struct rsnd_dvc *)(priv)->dvc + i); \ i++) -static const char const *dvc_ramp_rate[] = { +static const char * const dvc_ramp_rate[] = { "128 dB/1 step", /* 00000 */ "64 dB/1 step", /* 00001 */ "32 dB/1 step", /* 00010 */ -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: ak4642: enable stereo line output power-save mode
by Kuninori Morimoto 24 Mar '15

24 Mar '15

From: Takeshi Kihara <takeshi.kihara.df(a)renesas.com> ak4642 has power-save mode for stereo line to reduce pop noise. This patch enables it. Signed-off-by: Takeshi Kihara <takeshi.kihara.df(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 33 +++++++++++++++++++++++++++++++-- 1 file changed, 31 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 7255f69..24ceb36 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -97,6 +97,9 @@ #define PMMP (1 << 2) /* MPWR pin Power Management */ #define MGAIN0 (1 << 0) /* MIC amp gain*/ +/* SG_SL2 */ +#define LOPS (1 << 6) /* Stero Line-out Power Save Mode */ + /* TIMER */ #define ZTM(param) ((param & 0x3) << 4) /* ALC Zero Crossing TimeOut */ #define WTM(param) (((param & 0x4) << 4) | ((param & 0x3) << 2)) @@ -168,6 +171,29 @@ static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = { SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0), }; +/* event handlers */ +static int ak4642_lout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_PRE_PMU: + /* Power save mode ON */ + snd_soc_update_bits(codec, SG_SL2, LOPS, LOPS); + break; + case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_POST_PMD: + /* Power save mode OFF */ + mdelay(300); + snd_soc_update_bits(codec, SG_SL2, LOPS, 0); + break; + } + + return 0; +} + static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { /* Outputs */ @@ -182,9 +208,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0, + SND_SOC_DAPM_MIXER_E("LINEOUT Mixer", PW_MGMT1, 3, 0, &ak4642_lout_mixer_controls[0], - ARRAY_SIZE(ak4642_lout_mixer_controls)), + ARRAY_SIZE(ak4642_lout_mixer_controls), + ak4642_lout_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), /* DAC */ SND_SOC_DAPM_DAC("DAC", NULL, PW_MGMT1, 2, 0), -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt286: check regmap_read result for ID check
by Bard Liao 24 Mar '15

24 Mar '15

It is worth to check the regmap_read result for ID check since it is the first regmap_read. And we can check if there is any i2c issue. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 098fc78..ccc0fa0 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1220,7 +1220,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, { struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt286_priv *rt286; - int i, ret; + int i, ret, val; rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), GFP_KERNEL); @@ -1235,11 +1235,15 @@ static int rt286_i2c_probe(struct i2c_client *i2c, return ret; } - regmap_read(rt286->regmap, - RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); - if (ret != RT286_VENDOR_ID && ret != RT288_VENDOR_ID) { + ret = regmap_read(rt286->regmap, + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + if (ret != 0) { + dev_err(&i2c->dev, "I2C error %d\n", ret); + return ret; + } + if (val != RT286_VENDOR_ID && val != RT288_VENDOR_ID) { dev_err(&i2c->dev, - "Device with ID register %x is not rt286\n", ret); + "Device with ID register %x is not rt286\n", val); return -ENODEV; } -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH RESEND] ARM: dts: Support audio on Exynos5422-odroidxu3 using simple-audio-card
by Inha Song 24 Mar '15

24 Mar '15

Add MAX98090 audio codec, I2S interface and the sound nodes to support audio on Exynos5422 SoC Based Odroid-XU3 board. Now we can support audio in Odroid-XU3 board using simple-audio-card DT binding. Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- arch/arm/boot/dts/exynos5420.dtsi | 9 +++++ arch/arm/boot/dts/exynos5422-odroidxu3.dts | 57 ++++++++++++++++++++++++++++++ 2 files changed, 66 insertions(+) diff --git a/arch/arm/boot/dts/exynos5420.dtsi b/arch/arm/boot/dts/exynos5420.dtsi index 4eaeabe..e459c1d 100644 --- a/arch/arm/boot/dts/exynos5420.dtsi +++ b/arch/arm/boot/dts/exynos5420.dtsi @@ -415,6 +415,9 @@ <&clock_audss EXYNOS_I2S_BUS>, <&clock_audss EXYNOS_SCLK_I2S>; clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk0"; + #sound-dai-cells = <1>; samsung,idma-addr = <0x03000000>; pinctrl-names = "default"; pinctrl-0 = <&i2s0_bus>; @@ -429,6 +432,9 @@ dma-names = "tx", "rx"; clocks = <&clock CLK_I2S1>, <&clock CLK_SCLK_I2S1>; clock-names = "iis", "i2s_opclk0"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk1"; + #sound-dai-cells = <1>; pinctrl-names = "default"; pinctrl-0 = <&i2s1_bus>; status = "disabled"; @@ -442,6 +448,9 @@ dma-names = "tx", "rx"; clocks = <&clock CLK_I2S2>, <&clock CLK_SCLK_I2S2>; clock-names = "iis", "i2s_opclk0"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk2"; + #sound-dai-cells = <1>; pinctrl-names = "default"; pinctrl-0 = <&i2s2_bus>; status = "disabled"; diff --git a/arch/arm/boot/dts/exynos5422-odroidxu3.dts b/arch/arm/boot/dts/exynos5422-odroidxu3.dts index edc25cf..9275ad6 100644 --- a/arch/arm/boot/dts/exynos5422-odroidxu3.dts +++ b/arch/arm/boot/dts/exynos5422-odroidxu3.dts @@ -11,6 +11,7 @@ */ /dts-v1/; +#include <dt-bindings/sound/samsung-i2s.h> #include "exynos5800.dtsi" / { @@ -285,6 +286,62 @@ rtc@101E0000 { status = "okay"; }; + + sound: sound { + compatible = "simple-audio-card"; + assigned-clocks = <&clock_audss EXYNOS_MOUT_AUDSS>, + <&clock_audss EXYNOS_MOUT_I2S>, + <&clock_audss EXYNOS_DOUT_AUD_BUS>; + assigned-clock-parents = <&clock CLK_FIN_PLL>, + <&clock_audss EXYNOS_MOUT_AUDSS>; + assigned-clock-rates = <0>, + <0>, + <19200000>; + + simple-audio-card,name = "Odroid-XU3"; + + simple-audio-card,widgets = + "Headphone", "Headphone Jack", + "Speakers", "Speakers"; + simple-audio-card,routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "Headphone Jack", "MICBIAS", + "IN1", "Headphone Jack", + "Speakers", "SPKL", + "Speakers", "SPKR"; + + simple-audio-card,format = "i2s"; + simple-audio-card,bitclock-master = <&link0_codec>; + simple-audio-card,frame-master = <&link0_codec>; + + simple-audio-card,cpu { + sound-dai = <&i2s0 0>; + system-clock-frequency = <19200000>; + }; + + link0_codec: simple-audio-card,codec { + sound-dai = <&max98090>; + clocks = <&i2s0 CLK_I2S_CDCLK>; + }; + }; +}; + +&hsi2c_5 { + status = "okay"; + max98090: max98090@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + interrupt-parent = <&gpx3>; + interrupts = <2 0>; + clocks = <&i2s0 CLK_I2S_CDCLK>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + }; +}; + +&i2s0 { + status = "okay"; }; &hdmi { -- 2.0.0.390.gcb682f8

3 3

[alsa-devel] [PATCH] ALSA: asihpi: Fix duplicate const for clock sources
by Krzysztof Kozlowski 24 Mar '15

24 Mar '15

Replace duplicated const keyword for 'sampleclock_sources' with proper array of const pointers to const strings. Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> --- sound/pci/asihpi/asihpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index e5cd7be85355..1039eccbb895 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -2376,7 +2376,7 @@ static int snd_asihpi_cmode_add(struct snd_card_asihpi *asihpi, /*------------------------------------------------------------ Sampleclock source controls ------------------------------------------------------------*/ -static const char const *sampleclock_sources[] = { +static const char * const sampleclock_sources[] = { "N/A", "Local PLL", "Digital Sync", "Word External", "Word Header", "SMPTE", "Digital1", "Auto", "Network", "Invalid", "Prev Module", "BLU-Link", -- 1.9.1

2 1

[alsa-devel] [trivial PATCH] sound: Deparenthesize negative error returns
by Joe Perches 24 Mar '15

24 Mar '15

Make the returns a bit more kernel standard style. Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/isa/wavefront/wavefront_fx.c | 6 +++--- sound/isa/wavefront/wavefront_synth.c | 26 +++++++++++++------------- sound/oss/dev_table.c | 6 +++--- sound/oss/v_midi.c | 4 ++-- 4 files changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/isa/wavefront/wavefront_fx.c b/sound/isa/wavefront/wavefront_fx.c index b5a1970..0608a5a 100644 --- a/sound/isa/wavefront/wavefront_fx.c +++ b/sound/isa/wavefront/wavefront_fx.c @@ -79,13 +79,13 @@ wavefront_fx_memset (snd_wavefront_t *dev, if (page < 0 || page > 7) { snd_printk ("FX memset: " "page must be >= 0 and <= 7\n"); - return -(EINVAL); + return -EINVAL; } if (addr < 0 || addr > 0x7f) { snd_printk ("FX memset: " "addr must be >= 0 and <= 7f\n"); - return -(EINVAL); + return -EINVAL; } if (cnt == 1) { @@ -118,7 +118,7 @@ wavefront_fx_memset (snd_wavefront_t *dev, snd_printk ("FX memset " "(0x%x, 0x%x, 0x%lx, %d) incomplete\n", page, addr, (unsigned long) data, cnt); - return -(EIO); + return -EIO; } } diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 33f5ec1..69f76ff 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -793,7 +793,7 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header) if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_PATCH, NULL, buf)) { snd_printk ("download patch failed\n"); - return -(EIO); + return -EIO; } return (0); @@ -831,7 +831,7 @@ wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header) if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_PROGRAM, NULL, buf)) { snd_printk ("download patch failed\n"); - return -(EIO); + return -EIO; } return (0); @@ -952,7 +952,7 @@ wavefront_send_sample (snd_wavefront_t *dev, if (skip > 0 && header->hdr.s.SampleResolution != LINEAR_16BIT) { snd_printk ("channel selection only " "possible on 16-bit samples"); - return -(EINVAL); + return -EINVAL; } switch (skip) { @@ -1049,7 +1049,7 @@ wavefront_send_sample (snd_wavefront_t *dev, NULL, sample_hdr)) { snd_printk ("sample %sdownload refused.\n", header->size ? "" : "header "); - return -(EIO); + return -EIO; } if (header->size == 0) { @@ -1075,7 +1075,7 @@ wavefront_send_sample (snd_wavefront_t *dev, if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_BLOCK, NULL, NULL)) { snd_printk ("download block " "request refused.\n"); - return -(EIO); + return -EIO; } for (i = 0; i < blocksize; i++) { @@ -1135,12 +1135,12 @@ wavefront_send_sample (snd_wavefront_t *dev, if (dma_ack == -1) { snd_printk ("upload sample " "DMA ack timeout\n"); - return -(EIO); + return -EIO; } else { snd_printk ("upload sample " "DMA ack error 0x%x\n", dma_ack); - return -(EIO); + return -EIO; } } } @@ -1181,7 +1181,7 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_SAMPLE_ALIAS, NULL, alias_hdr)) { snd_printk ("download alias failed.\n"); - return -(EIO); + return -EIO; } dev->sample_status[header->number] = (WF_SLOT_FILLED|WF_ST_ALIAS); @@ -1232,7 +1232,7 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) msample_hdr)) { snd_printk ("download of multisample failed.\n"); kfree(msample_hdr); - return -(EIO); + return -EIO; } dev->sample_status[header->number] = (WF_SLOT_FILLED|WF_ST_MULTISAMPLE); @@ -1254,7 +1254,7 @@ wavefront_fetch_multisample (snd_wavefront_t *dev, if (snd_wavefront_cmd (dev, WFC_UPLOAD_MULTISAMPLE, log_ns, number)) { snd_printk ("upload multisample failed.\n"); - return -(EIO); + return -EIO; } DPRINT (WF_DEBUG_DATA, "msample %d has %d samples\n", @@ -1273,14 +1273,14 @@ wavefront_fetch_multisample (snd_wavefront_t *dev, if ((val = wavefront_read (dev)) == -1) { snd_printk ("upload multisample failed " "during sample loop.\n"); - return -(EIO); + return -EIO; } d[0] = val; if ((val = wavefront_read (dev)) == -1) { snd_printk ("upload multisample failed " "during sample loop.\n"); - return -(EIO); + return -EIO; } d[1] = val; @@ -1315,7 +1315,7 @@ wavefront_send_drum (snd_wavefront_t *dev, wavefront_patch_info *header) if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_EDRUM_PROGRAM, NULL, drumbuf)) { snd_printk ("download drum failed.\n"); - return -(EIO); + return -EIO; } return (0); diff --git a/sound/oss/dev_table.c b/sound/oss/dev_table.c index d8cf3e5..6dad515 100644 --- a/sound/oss/dev_table.c +++ b/sound/oss/dev_table.c @@ -58,13 +58,13 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver, if (vers != AUDIO_DRIVER_VERSION || driver_size > sizeof(struct audio_driver)) { printk(KERN_ERR "Sound: Incompatible audio driver for %s\n", name); - return -(EINVAL); + return -EINVAL; } num = sound_alloc_audiodev(); if (num == -1) { printk(KERN_ERR "sound: Too many audio drivers\n"); - return -(EBUSY); + return -EBUSY; } d = (struct audio_driver *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_driver))); sound_nblocks++; @@ -79,7 +79,7 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver, if (d == NULL || op == NULL) { printk(KERN_ERR "Sound: Can't allocate driver for (%s)\n", name); sound_unload_audiodev(num); - return -(ENOMEM); + return -ENOMEM; } init_waitqueue_head(&op->in_sleeper); init_waitqueue_head(&op->out_sleeper); diff --git a/sound/oss/v_midi.c b/sound/oss/v_midi.c index f0b4151..fc0ba27 100644 --- a/sound/oss/v_midi.c +++ b/sound/oss/v_midi.c @@ -49,13 +49,13 @@ static int v_midi_open (int dev, int mode, unsigned long flags; if (devc == NULL) - return -(ENXIO); + return -ENXIO; spin_lock_irqsave(&devc->lock,flags); if (devc->opened) { spin_unlock_irqrestore(&devc->lock,flags); - return -(EBUSY); + return -EBUSY; } devc->opened = 1; spin_unlock_irqrestore(&devc->lock,flags);

2 1

[alsa-devel] [PATCH] ASoC: pcm512x: Remove hardcoding of pll-lock to GPIO4
by Howard Mitchell 24 Mar '15

24 Mar '15

Currently GPIO4 is hardcoded to output the pll-lock signal. Unfortunately this is after the pll-out GPIO is configured which is selectable in the device tree. Therefore it is not possible to use GPIO4 for pll-out. Therefore this patch removes the configuration of GPIO4. Signed-off-by: Howard Mitchell <hm(a)hmbedded.co.uk> --- sound/soc/codecs/pcm512x.c | 19 ------------------- 1 file changed, 19 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 8472099..5a30fdd 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1296,25 +1296,6 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, ret, pcm512x->pll_out); return ret; } - - gpio = PCM512x_G1OE << (4 - 1); - ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_EN, - gpio, gpio); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable gpio %d: %d\n", - 4, ret); - return ret; - } - - gpio = PCM512x_GPIO_OUTPUT_1 + 4 - 1; - ret = regmap_update_bits(pcm512x->regmap, gpio, - PCM512x_GxSL, PCM512x_GxSL_PLLLK); - if (ret != 0) { - dev_err(codec->dev, - "Failed to output pll lock on %d: %d\n", - ret, 4); - return ret; - } } ret = regmap_update_bits(pcm512x->regmap, PCM512x_SYNCHRONIZE, -- 1.7.9.5

3 3

[alsa-devel] [TINYCOMPRESS][PATCH 0/6] Convert build system to autotools
by Qais Yousef 24 Mar '15

24 Mar '15

This series converts the build system to use autotools and splits the files into lib and utils. Also patch 5 fixes error handling code. I can't test Android builds but both cross and native compiling worked for me. I left AUTHORS, ChangeLog and NEWS files empty. I used the license header to fill COPYING. Qais Yousef (6): Convert the build system to autotools add .gitignore file makefile.linux: delete as no longer necessary/used Android.mk: Update to use the new location of source files compress.c: fix check for errors from poll(), read() and write() src/lib/utils.c: remove this empty file .gitignore | 39 ++++ AUTHORS | 0 Android.mk | 6 +- COPYING | 51 ++++ ChangeLog | 0 INSTALL | 365 +++++++++++++++++++++++++++++ Makefile.am | 3 + NEWS | 0 autogen.sh | 6 + compress.c | 632 ------------------------------------------------- configure.ac | 31 +++ cplay.c | 330 -------------------------- crec.c | 489 -------------------------------------- m4/.place_holder | 0 makefile.linux | 41 ---- src/Makefile.am | 1 + src/lib/Makefile.am | 5 + src/lib/compress.c | 634 ++++++++++++++++++++++++++++++++++++++++++++++++++ src/utils/Makefile.am | 11 + src/utils/cplay.c | 330 ++++++++++++++++++++++++++ src/utils/crec.c | 489 ++++++++++++++++++++++++++++++++++++++ utils.c | 62 ----- 22 files changed, 1968 insertions(+), 1557 deletions(-) create mode 100644 .gitignore create mode 100644 AUTHORS create mode 100644 COPYING create mode 100644 ChangeLog create mode 100644 INSTALL create mode 100644 Makefile.am create mode 100644 NEWS create mode 100755 autogen.sh delete mode 100644 compress.c create mode 100644 configure.ac delete mode 100644 cplay.c delete mode 100644 crec.c create mode 100644 m4/.place_holder delete mode 100644 makefile.linux create mode 100644 src/Makefile.am create mode 100644 src/lib/Makefile.am create mode 100644 src/lib/compress.c create mode 100644 src/utils/Makefile.am create mode 100644 src/utils/cplay.c create mode 100644 src/utils/crec.c delete mode 100644 utils.c -- 2.1.0

2 8

[alsa-devel] [PATCH] MAINTAINERS: Add maintainer for maxim codecs
by Anish Kumar 24 Mar '15

24 Mar '15

This patch adds maintainer for maxim audio codecs. Signed-off-by: Anish Kumar <yesanishhere(a)gmail.com> --- MAINTAINERS | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 8bdd7a7..2128586 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7387,6 +7387,13 @@ S: Supported F: sound/soc/ F: include/sound/soc* +MAXIM INTEGRATED SOC CODEC DRIVER +M: Anish Kumar <yesanishhere(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: sound/soc/codecs/max* +F: include/sound/max* + SPARC + UltraSPARC (sparc/sparc64) M: "David S. Miller" <davem(a)davemloft.net> L: sparclinux(a)vger.kernel.org -- 1.9.3

3 5

Re: [alsa-devel] Alsa-devel Digest, Vol 97, Issue 173
by noman pouigt 24 Mar '15

24 Mar '15

> I would like to know which file I can use to connect/enable > DPAM widgets permanently so a simple aplay/record can work? > > My environment is: > - Yocto Linux Kernel 3.10 - (DTS based) > - Max98090 Codec > - imx6SL SoC > > I thought about doing this in alsa.conf or asound.conf but I couldn't > find a single example with DAPM widgets . I am also stuck with the same problem. I am sure ALSA experts can help here. What i want to know is how to set the dapm widgets in open embedded? Is it alsa.conf or asound.conf ? I need to set something like this: amixer cset name="LEFT MIXER DAC" 1 Using amixer i am able to playback but i don't know how can i modify conf file to have this dapm widgets associated with a device. Any example would suffice.

1 0

[alsa-devel] [RFC][PATCH 00/11] digi00x: new driver for Digidesign 002/003 family
by Takashi Sakamoto 24 Mar '15

24 Mar '15

This patchset adds an experimental driver for Digidesign 002/003 family. If you have any models, please test with this driver and report the result. Currently, this driver correctly supports these major functionalities: (as long as I tested) * PCM sample capture * MIDI messages capture * MIDI messages playback And functionalities which need to be tested: * streaming with S/PDIF, ADAT and work clock source * MIDI device control And includes a bug which should be fixed: * PCM sample playback For a driver for these models, there was a preceding implementation by Robin Gareus and Damien Zammit. [alsa-devel] draft -- ALSA firewire + digi003 http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/058177.ht… This patchset includes their implementation, called-as double-oh-three, to multiplex PCM samples into CIP packet. As long as I test, their implementation may includes any bugs and this driver still causes noisy sound in several cases, while I confirm to get the same data block pattern which Robin explains. http://gareus.org/wiki/digi003 If you interested in this bug and tries to fix it, my packet dump may be helpfull. Please see: http://subversion.ffado.org/wiki/digi003rack * 96khz-24bit-1ch.log (27.5 kB) * 44khz-24bit-1ch.log (49.8 kB) * 44khz-24bit-2ch.log (49.8 kB) And DKMS package is available on my github repository. https://github.com/takaswie/snd-firewire-improve Takashi Sakamoto (11): ALSA: digi00x: add skelton for Digi 002/003 device driver ALSA: digi00x: add streaming functionality ALSA: digi00x: add proc node for clock status ALSA: digi00x: add PCM functionality ALSA: digi00x: add MIDI functionality ALSA: digi00x: add hwdep interface ALSA: digi00x: support unknown asynchronous message ALSA: digi00x: support MIDI ports for device control ALSA: firewire-lib: allows to implement external MIDI callback function digi00x: improve MIDI capture/playback ALSA: digi00x: apply double-oh-three algorism to multiplex PCM samples include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 1 + sound/firewire/Kconfig | 10 + sound/firewire/Makefile | 1 + sound/firewire/amdtp.c | 30 ++- sound/firewire/amdtp.h | 5 + sound/firewire/digi00x/Makefile | 3 + sound/firewire/digi00x/digi00x-hwdep.c | 192 ++++++++++++++ sound/firewire/digi00x/digi00x-midi.c | 199 +++++++++++++++ sound/firewire/digi00x/digi00x-pcm.c | 336 +++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-proc.c | 71 ++++++ sound/firewire/digi00x/digi00x-protocol.c | 361 +++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-stream.c | 398 ++++++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x.c | 179 ++++++++++++++ sound/firewire/digi00x/digi00x.h | 126 ++++++++++ 15 files changed, 1907 insertions(+), 8 deletions(-) create mode 100644 sound/firewire/digi00x/Makefile create mode 100644 sound/firewire/digi00x/digi00x-hwdep.c create mode 100644 sound/firewire/digi00x/digi00x-midi.c create mode 100644 sound/firewire/digi00x/digi00x-pcm.c create mode 100644 sound/firewire/digi00x/digi00x-proc.c create mode 100644 sound/firewire/digi00x/digi00x-protocol.c create mode 100644 sound/firewire/digi00x/digi00x-stream.c create mode 100644 sound/firewire/digi00x/digi00x.c create mode 100644 sound/firewire/digi00x/digi00x.h -- 2.1.0

7 39

[alsa-devel] [PATCH v2] ARM: dts: Support audio on Exynos5422-odroidxu3 using simple-audio-card
by Inha Song 24 Mar '15

24 Mar '15

Add MAX98090 audio codec, I2S interface and the sound nodes to support audio on Exynos5422 SoC Based Odroid-XU3 board. Now we can support audio in Odroid-XU3 board using simple-audio-card DT binding. Signed-off-by: Inha Song <ideal.song(a)samsung.com> Reviewed-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- arch/arm/boot/dts/exynos5420.dtsi | 9 +++++ arch/arm/boot/dts/exynos5422-odroidxu3.dts | 59 ++++++++++++++++++++++++++++++ 2 files changed, 68 insertions(+) diff --git a/arch/arm/boot/dts/exynos5420.dtsi b/arch/arm/boot/dts/exynos5420.dtsi index 4eaeabe..e459c1d 100644 --- a/arch/arm/boot/dts/exynos5420.dtsi +++ b/arch/arm/boot/dts/exynos5420.dtsi @@ -415,6 +415,9 @@ <&clock_audss EXYNOS_I2S_BUS>, <&clock_audss EXYNOS_SCLK_I2S>; clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk0"; + #sound-dai-cells = <1>; samsung,idma-addr = <0x03000000>; pinctrl-names = "default"; pinctrl-0 = <&i2s0_bus>; @@ -429,6 +432,9 @@ dma-names = "tx", "rx"; clocks = <&clock CLK_I2S1>, <&clock CLK_SCLK_I2S1>; clock-names = "iis", "i2s_opclk0"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk1"; + #sound-dai-cells = <1>; pinctrl-names = "default"; pinctrl-0 = <&i2s1_bus>; status = "disabled"; @@ -442,6 +448,9 @@ dma-names = "tx", "rx"; clocks = <&clock CLK_I2S2>, <&clock CLK_SCLK_I2S2>; clock-names = "iis", "i2s_opclk0"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk2"; + #sound-dai-cells = <1>; pinctrl-names = "default"; pinctrl-0 = <&i2s2_bus>; status = "disabled"; diff --git a/arch/arm/boot/dts/exynos5422-odroidxu3.dts b/arch/arm/boot/dts/exynos5422-odroidxu3.dts index edc25cf..c038f0d 100644 --- a/arch/arm/boot/dts/exynos5422-odroidxu3.dts +++ b/arch/arm/boot/dts/exynos5422-odroidxu3.dts @@ -11,6 +11,7 @@ */ /dts-v1/; +#include <dt-bindings/sound/samsung-i2s.h> #include "exynos5800.dtsi" / { @@ -285,6 +286,64 @@ rtc@101E0000 { status = "okay"; }; + + sound: sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "Odroid-XU3"; + simple-audio-card,widgets = + "Headphone", "Headphone Jack", + "Speakers", "Speakers"; + simple-audio-card,routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "Headphone Jack", "MICBIAS", + "IN1", "Headphone Jack", + "Speakers", "SPKL", + "Speakers", "SPKR"; + + simple-audio-card,format = "i2s"; + simple-audio-card,bitclock-master = <&link0_codec>; + simple-audio-card,frame-master = <&link0_codec>; + + simple-audio-card,cpu { + sound-dai = <&i2s0 0>; + system-clock-frequency = <19200000>; + }; + + link0_codec: simple-audio-card,codec { + sound-dai = <&max98090>; + clocks = <&i2s0 CLK_I2S_CDCLK>; + }; + }; +}; + +&clock_audss { + assigned-clocks = <&clock_audss EXYNOS_MOUT_AUDSS>, + <&clock_audss EXYNOS_MOUT_I2S>, + <&clock_audss EXYNOS_DOUT_AUD_BUS>; + assigned-clock-parents = <&clock CLK_FIN_PLL>, + <&clock_audss EXYNOS_MOUT_AUDSS>; + assigned-clock-rates = <0>, + <0>, + <19200000>; +}; + +&hsi2c_5 { + status = "okay"; + max98090: max98090@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + interrupt-parent = <&gpx3>; + interrupts = <2 0>; + clocks = <&i2s0 CLK_I2S_CDCLK>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + }; +}; + +&i2s0 { + status = "okay"; }; &hdmi { -- 2.0.0.390.gcb682f8

1 0

[alsa-devel] [PATCH 00/10 v3] ASoC: Renesas adds DPCM based sampling rate convert
by Kuninori Morimoto 24 Mar '15

24 Mar '15

Hi Mark These are v3 of DPCM based sampling rate convert support patch-set for Rensas sound driver. Then, it doesn't use simple-card. it uses new Renesas specific sound card driver, basically it was copied from simple-card. v2 patch-set had ak4642 patch, but, it is already applied. v3 doesn't include it. 1 : prepare patch for Renesas new sound card driver 2 - 3 : new Renesas sound card driver 4 - 8 : fixup/tidyup patches for rsnd 9 - 10 : DPCM support patch for rsnd Kuninori Morimoto (10): 1) ASoC: core: add snd_soc_get_dai_drv() to get DAI driver 2) ASoC: rsrc-card: add Renesas sampling rate convert sound card support 3) ASoC: rsrc-card: add .be_hw_params_fixup support for convert rate 4) ASoC: rsnd: no more SSI restart when unusual situation 5) ASoC: rsnd: no more SRC restart when unusual situation 6) ASoC: rsnd: tidyup error message format 7) ASoC: rsnd: show debug info for sampling rate convert 8) ASoC: rsnd: don't call clk_prepare_enable/unprepare() from inside spin lock 9) ASoC: rsnd: add dai_link stream name 10) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 67 +++++++++++++ include/sound/soc.h | 1 + sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 + sound/soc/sh/rcar/core.c | 48 +++++++++- sound/soc/sh/rcar/dvc.c | 12 +-- sound/soc/sh/rcar/rsnd.h | 16 +++- sound/soc/sh/rcar/rsrc-card.c | 512 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/src.c | 63 ++++++++++-- sound/soc/sh/rcar/ssi.c | 31 ++++-- sound/soc/soc-core.c | 69 ++++++++++---- 12 files changed, 777 insertions(+), 54 deletions(-) Best regards --- Kuninori Morimoto _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

2 22

[alsa-devel] [PATCH] ASoC:pcm512x: Make PLL lock output selectable via device tree.
by Howard Mitchell 23 Mar '15

23 Mar '15

Currently the PLL Lock output signal is hardcoded to GPIO4. This makes it seletable in the same way as pll-in and pll-out. Signed-off-by: Howard Mitchell <hm(a)hmbedded.co.uk> --- .../devicetree/bindings/sound/pcm512x.txt | 3 ++ sound/soc/codecs/pcm512x.c | 47 +++++++++++++------- 2 files changed, 33 insertions(+), 17 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/pcm512x.txt b/Documentation/devicetree/bindings/sound/pcm512x.txt index 3aae3b4..432f186 100644 --- a/Documentation/devicetree/bindings/sound/pcm512x.txt +++ b/Documentation/devicetree/bindings/sound/pcm512x.txt @@ -26,6 +26,8 @@ Optional properties: given pll-in pin and PLL output on the given pll-out pin. An external connection from the pll-out pin to the SCLK pin is assumed. + - pll-lock : gpio pin used to output the PLL lock flag. + Examples: pcm5122: pcm5122@4c { @@ -49,4 +51,5 @@ Examples: clocks = <&sck>; pll-in = <3>; pll-out = <6>; + pll-lock = <4>; }; diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 8472099..a4217d7 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -49,6 +49,7 @@ struct pcm512x_priv { int fmt; int pll_in; int pll_out; + int pll_lock; int pll_r; int pll_j; int pll_d; @@ -1296,24 +1297,26 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, ret, pcm512x->pll_out); return ret; } + } - gpio = PCM512x_G1OE << (4 - 1); - ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_EN, - gpio, gpio); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable gpio %d: %d\n", - 4, ret); - return ret; - } - - gpio = PCM512x_GPIO_OUTPUT_1 + 4 - 1; - ret = regmap_update_bits(pcm512x->regmap, gpio, - PCM512x_GxSL, PCM512x_GxSL_PLLLK); - if (ret != 0) { - dev_err(codec->dev, - "Failed to output pll lock on %d: %d\n", - ret, 4); - return ret; + if (pcm512x->pll_lock) { + gpio = PCM512x_G1OE << (pcm512x->pll_lock - 1); + ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_EN, + gpio, gpio); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable gpio %d: %d\n", + pcm512x->pll_lock, ret); + return ret; + } + + gpio = PCM512x_GPIO_OUTPUT_1 + pcm512x->pll_lock - 1; + ret = regmap_update_bits(pcm512x->regmap, gpio, + PCM512x_GxSL, PCM512x_GxSL_PLLLK); + if (ret != 0) { + dev_err(codec->dev, + "Failed to output pll lock on %d: %d\n", + ret, pcm512x->pll_lock); + return ret; } } @@ -1518,6 +1521,16 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) ret = -EINVAL; goto err_clk; } + + if (of_property_read_u32(np, "pll-lock", &val) >= 0) { + if (val > 6) { + dev_err(dev, "Invalid pll-lock\n"); + ret = -EINVAL; + goto err_clk; + } + pcm512x->pll_lock = val; + } + } #endif -- 1.7.9.5

3 7

[alsa-devel] [PATCH v2] ALSA: hda - Add dock support for Thinkpad T450s (17aa:5036)
by Sebastian Wicki 23 Mar '15

23 Mar '15

This model uses the same dock port as the previous generation. Signed-off-by: Sebastian Wicki <gandro(a)gmx.net> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eee4532..8775389 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4995,6 +4995,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), -- 2.3.3

2 1

[alsa-devel] [PATCH] ALSA: hda - Add dock support for Thinkpad T450s (17aa:5036)
by Sebastian Wicki 23 Mar '15

23 Mar '15

This model uses the same dock port as the previous generation. Signed-off-by: Sebastian Wicki <gandro(a)gmx.net> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eee4532..a9e2e6e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4988,6 +4988,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), -- 2.3.3

2 2

[alsa-devel] [PATCH 0/7] Yet more HD-audio restrucruing
by Takashi Iwai 23 Mar '15

23 Mar '15

Hi, this is the patchset I mentioned in the previous post -- moving some HD-audio core stuff into sound/hda/ directory. As of now, only the codec drivers are split to sound/hda/. The legacy bus (controller) driver still remains almost intact, but the split of the core controller driver is also planned, and will be finished hopefully before 4.1 merge window. The original patches are found in topic/hda-core branch of sound git tree. The patches posted here are the ones rebased on top of the previous widget PM patches, and found in test/hda-core branch. Once when the widget PM patches get merged, I'll adjust this patchset on top of it. Takashi === Takashi Iwai (7): ALSA: hda - Make snd_hda_bus_type public ALSA: hda - Move some codes up to hdac_bus struct ALSA: hda - Move a part of hda_codec stuff into hdac_device ALSA: hda - Add widget sysfs tree ALSA: hda - Support indirect execution of verbs ALSA: hda - Fix possible runtime PM refcount unbalance ALSA: hda - Re-add tracepoints to HD-audio core driver include/sound/hdaudio.h | 181 ++++++++++++ sound/Kconfig | 2 + sound/Makefile | 2 +- sound/hda/Kconfig | 2 + sound/hda/Makefile | 6 + sound/hda/hda_bus_type.c | 42 +++ sound/hda/hdac_bus.c | 186 +++++++++++++ sound/hda/hdac_device.c | 527 +++++++++++++++++++++++++++++++++++ sound/hda/hdac_sysfs.c | 404 +++++++++++++++++++++++++++ sound/hda/local.h | 23 ++ sound/hda/trace.c | 6 + sound/hda/trace.h | 62 +++++ sound/pci/hda/Kconfig | 1 + sound/pci/hda/Makefile | 1 - sound/pci/hda/hda_auto_parser.c | 33 ++- sound/pci/hda/hda_beep.c | 4 +- sound/pci/hda/hda_bind.c | 157 +++-------- sound/pci/hda/hda_codec.c | 600 +++++++++------------------------------- sound/pci/hda/hda_codec.h | 87 ++---- sound/pci/hda/hda_controller.c | 8 +- sound/pci/hda/hda_generic.c | 25 +- sound/pci/hda/hda_intel.c | 4 +- sound/pci/hda/hda_local.h | 16 +- sound/pci/hda/hda_proc.c | 42 +-- sound/pci/hda/hda_sysfs.c | 60 ++-- sound/pci/hda/hda_trace.h | 119 -------- sound/pci/hda/local.h | 39 +++ sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_ca0132.c | 6 +- sound/pci/hda/patch_conexant.c | 20 +- sound/pci/hda/patch_hdmi.c | 20 +- sound/pci/hda/patch_realtek.c | 78 +++--- sound/pci/hda/patch_si3054.c | 6 +- sound/pci/hda/patch_sigmatel.c | 84 +++--- sound/pci/hda/patch_via.c | 23 +- sound/pci/hda/thinkpad_helper.c | 2 +- 36 files changed, 1895 insertions(+), 985 deletions(-) create mode 100644 include/sound/hdaudio.h create mode 100644 sound/hda/Kconfig create mode 100644 sound/hda/Makefile create mode 100644 sound/hda/hda_bus_type.c create mode 100644 sound/hda/hdac_bus.c create mode 100644 sound/hda/hdac_device.c create mode 100644 sound/hda/hdac_sysfs.c create mode 100644 sound/hda/local.h create mode 100644 sound/hda/trace.c create mode 100644 sound/hda/trace.h delete mode 100644 sound/pci/hda/hda_trace.h create mode 100644 sound/pci/hda/local.h -- 2.3.3

1 8

[alsa-devel] [patch] ALSA: echoaudio: read past end of array
by Dan Carpenter 23 Mar '15

23 Mar '15

We need to cap "ucontrol->id.index / num_busses_in(chip)" so the we don't read beyond the end of the array. I also adding a check on "in" and changing the type in snd_echo_mixer_put() from short to unsigned int. Those changes are done for symmetry and are cosmetic. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index a962de0..862db9a 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1283,12 +1283,14 @@ static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol, static int snd_echo_mixer_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct echoaudio *chip; + struct echoaudio *chip = snd_kcontrol_chip(kcontrol); + unsigned int out = ucontrol->id.index / num_busses_in(chip); + unsigned int in = ucontrol->id.index % num_busses_in(chip); - chip = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = - chip->monitor_gain[ucontrol->id.index / num_busses_in(chip)] - [ucontrol->id.index % num_busses_in(chip)]; + if (out >= ECHO_MAXAUDIOOUTPUTS || in >= ECHO_MAXAUDIOINPUTS) + return -EINVAL; + + ucontrol->value.integer.value[0] = chip->monitor_gain[out][in]; return 0; } @@ -1297,12 +1299,14 @@ static int snd_echo_mixer_put(struct snd_kcontrol *kcontrol, { struct echoaudio *chip; int changed, gain; - short out, in; + unsigned int out, in; changed = 0; chip = snd_kcontrol_chip(kcontrol); out = ucontrol->id.index / num_busses_in(chip); in = ucontrol->id.index % num_busses_in(chip); + if (out >= ECHO_MAXAUDIOOUTPUTS || in >= ECHO_MAXAUDIOINPUTS) + return -EINVAL; gain = ucontrol->value.integer.value[0]; if (gain < ECHOGAIN_MINOUT || gain > ECHOGAIN_MAXOUT) return -EINVAL;

2 1

[alsa-devel] [patch] ALSA: ak411x: simplify snd_ak4113_create() a bit
by Dan Carpenter 23 Mar '15

23 Mar '15

"err" is always a negative error code here, so there is no point in checking. Removing the check silences a static checker warning and makes the code a bit more clear. Also we don't need to initialize "err". Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c index 8884488..2183e9e 100644 --- a/sound/i2c/other/ak4113.c +++ b/sound/i2c/other/ak4113.c @@ -73,7 +73,7 @@ int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read, void *private_data, struct ak4113 **r_ak4113) { struct ak4113 *chip; - int err = 0; + int err; unsigned char reg; static struct snd_device_ops ops = { .dev_free = snd_ak4113_dev_free, @@ -109,7 +109,7 @@ int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read, __fail: snd_ak4113_free(chip); - return err < 0 ? err : -EIO; + return err; } EXPORT_SYMBOL_GPL(snd_ak4113_create);

2 1

[alsa-devel] [PATCH 0/5 v4] ASoC: rsnd: tidyup .remove method
by Kuninori Morimoto 23 Mar '15

23 Mar '15

Hi Mark These are v4 of unbind/bind issue fixup patches. It adds new snd_soc_remove_card() and temporally remove card from system. and rebind it again if next component was binded. Then, it indicates warning Kuninori Morimoto (5): ASoC: soc-core: add snd_soc_remove_card() ASoC: soc-core: deactivate pins in snd_soc_instantiate_card() ASoC: soc-core: call soc_cleanup_card_debugfs() from snd_soc_unregister_card() ASoC: soc-core: call snd_soc_remove_card() when component del ASoC: soc-core: indicate warning for unbind/bind issue include/sound/soc.h | 1 + sound/soc/soc-core.c | 81 +++++++++++++++++++++++++++++++++++++------------- 2 files changed, 61 insertions(+), 21 deletions(-) _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

2 13

[alsa-devel] simplify simple-card
by Kuninori Morimoto 23 Mar '15

23 Mar '15

Hi Mark, all I want to create new sound card driver which supports DAPM. But I can't use simple-card driver for this purpose because DT binding issue. I guess, this new sound card can use/share simple-card functions. OTOH, simple-card is not "simple" today :( Basically, main purpose of simple-card / new sound card is binding CPU/Codec settings to ALSA SoC framework. But we have many pattern for it. From DT bindings point of view, we can/should use same DT bindings for same settings. Now, simple-card is supporting - single CPU/Codec - multi CPU/Codec - more feature... My opinion is that complex code has un-understandable bug. So, how about simplify simple-card, what do you think about this idea ? We separate current simple-card to common parts, and, simple-card specific parts. basically, common parts controls DT bindings and ALSA SoC settings. sound card specific parts cares each own feature. I don't know about naming, but these audio card can use same bindings via common functions. - common : basic common functions - simple-audio-card : existing driver, keep as-is for compatible reason - single-audio-card : single CPU/Codec sound card. simplified from simple-audio-card - multi-audio-card : multi CPU/Codec sound card. simplified from simple-audio-card - fixrate-audio-card : new sound card same as single-audio-card, but it needs sampling-rate-convert via DAPM, - chain-audio-card : DAPM card (?) - .... My idea is taht we can replace current "simple-audio-card" to "single-audio-card", or, "simple-audio-card" to "multi-audio-card" on DT. Best regards --- Kuninori Morimoto

2 6

[alsa-devel] [PATCH 1/2] [resend] snd-waveartist: Introduce Rockwell WaveArtist RWA010 driver
by Ondrej Zary 22 Mar '15

22 Mar '15

This is a new ALSA driver driver for Rockwell WaveArtist RWA010 chips found on some (rare) ISA sound cards, such as DCS Multimedia S717. I wasn't able to get the old OSS WaveArtist driver to work with my card but it was a great source of information about the chip (as the full datasheet is not available - only a brief one and a design guide). However, the OSS driver only supports few of the mixer registers so I had to install the card in Windows and dump mixer registers while changing the mixer settings. Then tried what the rest of the registers do and the result is a fully working mixer which even supports more controls than the Windows driver. Someone with a NetWinder can add support for it to this driver and then remove the old OSS one. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- include/sound/mpu401.h | 1 + sound/isa/Kconfig | 11 + sound/isa/Makefile | 2 + sound/isa/waveartist.c | 1399 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/isa/waveartist.h | 74 +++ 5 files changed, 1487 insertions(+) create mode 100644 sound/isa/waveartist.c create mode 100644 sound/isa/waveartist.h diff --git a/include/sound/mpu401.h b/include/sound/mpu401.h index e942096..8fceeba 100644 --- a/include/sound/mpu401.h +++ b/include/sound/mpu401.h @@ -44,6 +44,7 @@ #define MPU401_HW_INTEL8X0 17 /* Intel8x0 driver */ #define MPU401_HW_PC98II 18 /* Roland PC98II */ #define MPU401_HW_AUREAL 19 /* Aureal Vortex */ +#define MPU401_HW_WAVEARTIST 20 /* Rockwell WaveArtist */ #define MPU401_INFO_INPUT (1 << 0) /* input stream */ #define MPU401_INFO_OUTPUT (1 << 1) /* output stream */ diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 0216475..7a3b4a2 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -454,5 +454,16 @@ config SND_MSND_CLASSIC To compile this driver as a module, choose M here: the module will be called snd-msnd-classic. +config SND_WAVEARTIST + tristate "Rockwell WaveArtist RWA010" + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_PCM + help + Say Y here to include support for Rockwell WaveArtist RWA010 chips. + + To compile this driver as a module, choose M here: the module + will be called snd-waveartist. + endif # SND_ISA diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 9a15f14..23e11e2 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -12,6 +12,7 @@ snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o snd-sc6000-objs := sc6000.o snd-sscape-objs := sscape.o +snd-waveartist-objs := waveartist.o # Toplevel Module Dependency obj-$(CONFIG_SND_ADLIB) += snd-adlib.o @@ -23,6 +24,7 @@ obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o obj-$(CONFIG_SND_SC6000) += snd-sc6000.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o +obj-$(CONFIG_SND_WAVEARTIST) += snd-waveartist.o obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ galaxy/ gus/ msnd/ opti9xx/ \ sb/ wavefront/ wss/ diff --git a/sound/isa/waveartist.c b/sound/isa/waveartist.c new file mode 100644 index 0000000..e5bda5c --- /dev/null +++ b/sound/isa/waveartist.c @@ -0,0 +1,1399 @@ +/* + * Driver for Rockwell WaveArtist RWA010 soundcards + * + * Copyright (c) 2015 Ondrej Zary + * + * HW-related parts based on OSS WaveArtist driver by Hannu Savolainen + * + * ALSA code based on ES18xx driver by Christian Fischbach & Abramo Bagnara + * and also by OPL3-SA2 driver by Jaroslav Kysela + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/isa.h> +#include <linux/pnp.h> +#include <linux/isapnp.h> +#include <asm/dma.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> +#include <sound/initval.h> +#include "waveartist.h" + +MODULE_AUTHOR("Ondrej Zary"); +MODULE_DESCRIPTION("Driver for Rockwell WaveArtist RWA010 sound cards"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; +#ifdef CONFIG_PNP +static bool isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; +#endif +static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x250-0x3f0 */ +static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x388-0x3f0 */ +static long midi_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;/* 0x300-0x3f0 */ +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,10,11 */ +static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 5,6,7 */ +static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3 */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for WaveArtist soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for WaveArtist soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable WaveArtist soundcard."); +#ifdef CONFIG_PNP +module_param_array(isapnp, bool, NULL, 0444); +MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); +#endif +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for WaveArtist driver."); +module_param_array(fm_port, long, NULL, 0444); +MODULE_PARM_DESC(fm_port, "FM port # for WaveArtist driver."); +module_param_array(midi_port, long, NULL, 0444); +MODULE_PARM_DESC(midi_port, "MIDI port # for WaveArtist driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for WaveArtist driver."); +module_param_array(dma1, int, NULL, 0444); +MODULE_PARM_DESC(dma1, "DMA1 # for WaveArtist driver."); +module_param_array(dma2, int, NULL, 0444); +MODULE_PARM_DESC(dma2, "DMA2 # for WaveArtist driver."); + +#ifdef CONFIG_PNP +static int isa_registered; +static int pnp_registered; +#endif + +#define PFX "waveartist: " + +#ifdef CONFIG_PNP +#define WA_DEVICE(pnpid) { \ + .id = pnpid, \ + .devs = { {"RSS5000"}, {"RSS5001"}, {"RSS5002"} } \ +} +/* + * RSS5000 = WaveArtist, RSS5001 = SB, RSS5002 = MPU-401, RSS5003 = IDE, + * RSS5004 = gameport, RSS5005 = modem, RSS5006 = 3D + */ + +static struct pnp_card_device_id snd_waveartist_pnpids[] = { + WA_DEVICE("RSS5000"), /* 16-bit decode */ + WA_DEVICE("RSS5100"), /* 16-bit decode + modem */ + WA_DEVICE("RSS5200"), /* 10-bit decode + modem */ + WA_DEVICE("RSS5300"), /* 10-bit decode */ + WA_DEVICE("RSS5400"), /* 16-bit decode + IDE */ + WA_DEVICE("RSS5500"), /* 16-bit decode + modem + IDE */ + WA_DEVICE("RSS5600"), /* 10-bit decode + IDE */ + WA_DEVICE("RSS5700"), /* 10-bit decode + modem + IDE */ + WA_DEVICE("RSS5800"), /* 10-bit decode + modem + 3D */ + WA_DEVICE("RSS5900"), /* 10-bit decode + 3D */ + WA_DEVICE("RSS5A00"), /* 16-bit decode + modem + 3D */ + WA_DEVICE("RSS5B00"), /* 16-bit decode + 3D */ + { .id = "" } /* end */ +}; +MODULE_DEVICE_TABLE(pnp_card, snd_waveartist_pnpids); +#endif /* CONFIG_PNP */ + +struct snd_waveartist { +#ifdef CONFIG_PNP + struct pnp_dev *wa; /* WaveArtist device */ + struct pnp_dev *sb; /* SB emulation device */ + struct pnp_dev *mpu; /* MPU-401 device */ +#endif + unsigned long port; /* base port */ + struct resource *res_port; /* base port resource */ + int irq; + int dma_playback; + int dma_capture; + + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + + spinlock_t reg_lock; + struct snd_hwdep *synth; + struct snd_rawmidi *rmidi; + + u16 image[20]; /* mixer registers image */ +}; + +static inline void wa_outb(struct snd_waveartist *chip, u8 reg, u8 val) +{ + outb(val, chip->port + reg); +} + +static inline u8 wa_inb(struct snd_waveartist *chip, u8 reg) +{ + return inb(chip->port + reg); +} + +static inline void wa_outw(struct snd_waveartist *chip, u8 reg, u16 val) +{ + outw(val, chip->port + reg); +} + +static inline u16 wa_inw(struct snd_waveartist *chip, u8 reg) +{ + return inw(chip->port + reg); +} + +static inline void waveartist_set_ctlr(struct snd_waveartist *chip, u8 clear, + u8 set) +{ + clear = ~clear & wa_inb(chip, CTLR); + wa_outb(chip, CTLR, clear | set); +} + +/* acknowledge IRQ */ +static inline void waveartist_iack(struct snd_waveartist *chip) +{ + u8 old_ctlr = wa_inb(chip, CTLR) & ~IRQ_ACK; + + wa_outb(chip, CTLR, old_ctlr | IRQ_ACK); + wa_outb(chip, CTLR, old_ctlr); +} + +static int waveartist_reset(struct snd_waveartist *chip) +{ + unsigned int timeout, res = -1; + + waveartist_set_ctlr(chip, -1, RESET); + msleep(200); + waveartist_set_ctlr(chip, RESET, 0); + + timeout = 500; + do { + mdelay(2); + + if (wa_inb(chip, STATR) & CMD_RF) { + res = wa_inw(chip, CMDR); + if (res == 0x55aa) + break; + } + } while (--timeout); + + if (timeout == 0) { + dev_warn(chip->card->dev, "WaveArtist: reset timeout (res=0x%x)\n", + res); + return 1; + } + + return 0; +} + +/* Helper function to send and receive words + * from WaveArtist. It handles all the handshaking + * and can send or receive multiple words. + */ +static int waveartist_cmd(struct snd_waveartist *chip, + int nr_cmd, u16 *cmd, + int nr_resp, u16 *resp) +{ + unsigned long flags; + unsigned int timed_out = 0, i; + + spin_lock_irqsave(&chip->reg_lock, flags); + /* + * The chip can hang if we access the STATR register too quickly + * after a write. Do a dummy read to slow down. + */ + wa_inb(chip, CTLR); + + if (wa_inb(chip, STATR) & CMD_RF) { + /* flush the port */ + wa_inw(chip, CMDR); + udelay(10); + } + + for (i = 0; !timed_out && i < nr_cmd; i++) { + int count; + + for (count = 5000; count; count--) + if (wa_inb(chip, STATR) & CMD_WE) + break; + + if (!count) + timed_out = 1; + else + wa_outw(chip, CMDR, cmd[i]); + /* Another dummy read */ + wa_inb(chip, CTLR); + } + + for (i = 0; !timed_out && i < nr_resp; i++) { + int count; + + for (count = 5000; count; count--) + if (wa_inb(chip, STATR) & CMD_RF) + break; + + if (!count) + timed_out = 1; + else + resp[i] = wa_inw(chip, CMDR); + } + spin_unlock_irqrestore(&chip->reg_lock, flags); + + return timed_out ? 1 : 0; +} + +/* Send one command word */ +static inline int waveartist_cmd1(struct snd_waveartist *chip, u16 cmd) +{ + return waveartist_cmd(chip, 1, &cmd, 0, NULL); +} + +/* Send one command, receive one word */ +static inline u16 waveartist_cmd1_r(struct snd_waveartist *chip, u16 cmd) +{ + u16 ret; + + waveartist_cmd(chip, 1, &cmd, 1, &ret); + + return ret; +} + +/* Send a double command, receive one word (and throw it away) */ +static inline int waveartist_cmd2(struct snd_waveartist *chip, u16 cmd, u16 arg) +{ + u16 vals[2] = { cmd, arg }; + + return waveartist_cmd(chip, 2, vals, 1, vals); +} + +/* Send a triple command */ +static inline int waveartist_cmd3(struct snd_waveartist *chip, u16 cmd, + u16 arg1, u16 arg2) +{ + u16 vals[3] = { cmd, arg1, arg2 }; + + return waveartist_cmd(chip, 3, vals, 0, NULL); +} + +static u16 waveartist_getrev(struct snd_waveartist *chip) +{ + u16 temp[2]; + u16 cmd = WACMD_GETREV; + + waveartist_cmd(chip, 1, &cmd, 2, temp); + + return temp[0]; +} + +static irqreturn_t snd_waveartist_interrupt(int irq, void *dev_id) +{ + u8 status, irqstatus; + struct snd_card *card = dev_id; + struct snd_waveartist *chip; + + if (card == NULL) + return IRQ_NONE; + + chip = card->private_data; + + irqstatus = wa_inb(chip, IRQSTAT); + status = wa_inb(chip, STATR); + + if (status & IRQ_REQ) /* clear interrupt */ + waveartist_iack(chip); + + if (irqstatus & IRQ_PCM) { /* PCM buffer done */ + if ((status & DMA1) && chip->playback_substream) + snd_pcm_period_elapsed(chip->playback_substream); + if ((status & DMA0) && chip->capture_substream) + snd_pcm_period_elapsed(chip->capture_substream); + if (!(status & (DMA0 | DMA1))) + dev_warn(chip->card->dev, "Unknown PCM interrupt\n"); + } + + if (irqstatus & IRQ_SB) /* we do not use SB mode */ + dev_warn(chip->card->dev, "Unexpected SB interrupt\n"); + + if ((irqstatus & IRQ_MPU) && chip->rmidi) + snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data); + + return IRQ_HANDLED; +} + +#ifdef CONFIG_PM +static int snd_waveartist_suspend(struct snd_card *card, pm_message_t state) +{ + struct snd_waveartist *chip = card->private_data; + int i; + + if (!card) + return 0; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(chip->pcm); + + /* save mixer registers */ + for (i = 0; i < ARRAY_SIZE(chip->image); i++) + chip->image[i] = waveartist_cmd1_r(chip, + WACMD_GET_LEVEL | i << 8); + + return 0; +} + +static int snd_waveartist_resume(struct snd_card *card) +{ + struct snd_waveartist *chip; + int i; + + if (!card) + return 0; + + chip = card->private_data; + + /* restore mixer registers */ + for (i = 0; i < 10; i += 2) + waveartist_cmd3(chip, WACMD_SET_MIXER, + chip->image[i], chip->image[i + 1]); + for (i = 10; i < ARRAY_SIZE(chip->image); i += 2) + waveartist_cmd3(chip, WACMD_SET_LEVEL | + ((i - 10) / 2) << 8, + chip->image[i], chip->image[i + 1]); + + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + return 0; +} +#endif /* CONFIG_PM */ + +#ifdef CONFIG_PNP +static void snd_waveartist_set_irq(struct pnp_dev *pdev, int irq) +{ + if (!pdev->active) + return; + isapnp_cfg_begin(isapnp_card_number(pdev), isapnp_csn_number(pdev)); + isapnp_write_byte(0x70, irq); /* ISAPNP_CFG_IRQ */ + isapnp_cfg_end(); +} + +static int snd_waveartist_pnp(int dev, struct snd_waveartist *chip, + struct pnp_card_link *card, + const struct pnp_card_device_id *id) +{ + chip->wa = pnp_request_card_device(card, id->devs[0].id, NULL); + if (!chip->wa) + return -EBUSY; + + chip->sb = pnp_request_card_device(card, id->devs[1].id, NULL); + if (!chip->sb) + return -EBUSY; + + chip->mpu = pnp_request_card_device(card, id->devs[2].id, NULL); + if (!chip->mpu) + return -EBUSY; + + if (pnp_activate_dev(chip->wa) < 0) { + dev_err(chip->card->dev, "WA PnP configure failure\n"); + return -EBUSY; + } + if (pnp_activate_dev(chip->sb) < 0) { + dev_err(chip->card->dev, "SB PnP configure failure\n"); + return -EBUSY; + } + port[dev] = pnp_port_start(chip->wa, 0); + dma2[dev] = pnp_dma(chip->wa, 0); + fm_port[dev] = pnp_port_start(chip->sb, 1); + dma1[dev] = pnp_dma(chip->sb, 0); + irq[dev] = pnp_irq(chip->sb, 0); + + /* + * The card uses only one IRQ (listed in the resources of SB device) + * which needs to be shared by WaveArtist and MPU-401 devices. They + * don't have an IRQ resource so it must be forced. + */ + snd_waveartist_set_irq(chip->wa, irq[dev]); + + /* allocate MPU-401 resources */ + if (pnp_activate_dev(chip->mpu) < 0) + dev_err(chip->card->dev, "MPU-401 PnP configure failure: will be disabled\n"); + else { + midi_port[dev] = pnp_port_start(chip->mpu, 0); + snd_waveartist_set_irq(chip->mpu, irq[dev]); + } + + dev_dbg(chip->card->dev, "PnP WaveArtist: port=0x%lx, fm port=0x%lx, midi port=0x%lx\n", + port[dev], fm_port[dev], midi_port[dev]); + dev_dbg(chip->card->dev, "PnP WaveArtist: dma1=%i, dma2=%i, irq=%i\n", + dma1[dev], dma2[dev], irq[dev]); + + return 0; +} +#endif /* CONFIG_PNP */ + +static int snd_waveartist_playback_hw_params( + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + + if (err < 0) + return err; + + return 0; +} + +static int snd_waveartist_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static enum wa_format waveartist_format(snd_pcm_format_t format) +{ + if (snd_pcm_format_width(format) == 16) + return WA_FMT_S16; + if (snd_pcm_format_unsigned(format)) + return WA_FMT_U8; + else + return WA_FMT_S8; +} + +static u16 waveartist_rate(struct snd_pcm_runtime *runtime) +{ + return (runtime->rate << 16) / 44100; +} + +static int snd_waveartist_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int size = snd_pcm_lib_buffer_bytes(substream); + unsigned int count = snd_pcm_lib_period_bytes(substream); + + /* Set rate */ + if (waveartist_cmd2(chip, WACMD_OUTPUTSPEED, waveartist_rate(runtime))) + dev_warn(chip->card->dev, "error setting playback rate %dHz\n", + runtime->rate); + /* Set channel count */ + if (waveartist_cmd2(chip, WACMD_OUTPUTCHANNELS, runtime->channels)) + dev_warn(chip->card->dev, "error setting playback %d channels\n", + runtime->channels); + /* Set DMA channel */ + if (waveartist_cmd2(chip, WACMD_OUTPUTDMA, + chip->dma_playback > 3 ? WA_DMA_16BIT : WA_DMA_8BIT)) + dev_warn(chip->card->dev, "error setting playback data path\n"); + /* Set format */ + if (waveartist_cmd2(chip, WACMD_OUTPUTFORMAT, + waveartist_format(runtime->format))) + dev_warn(chip->card->dev, "error setting playback format %d\n", + runtime->format); + /* Set sample count */ + if (waveartist_cmd2(chip, WACMD_OUTPUTSIZE, count - 1)) + dev_warn(chip->card->dev, "error setting playback count %d\n", + count); + /* Configure DMA controller */ + snd_dma_program(chip->dma_playback, runtime->dma_addr, size, + DMA_MODE_WRITE | DMA_AUTOINIT); + + return 0; +} + +static int snd_waveartist_playback_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + waveartist_cmd1(chip, WACMD_OUTPUTSTART); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + waveartist_cmd1(chip, WACMD_OUTPUTSTOP); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + waveartist_cmd1(chip, WACMD_OUTPUTPAUSE); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + waveartist_cmd1(chip, WACMD_OUTPUTRESUME); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int snd_waveartist_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + + if (err < 0) + return err; + + return 0; +} + +static int snd_waveartist_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int size = snd_pcm_lib_buffer_bytes(substream); + unsigned int count = snd_pcm_lib_period_bytes(substream); + + /* Set rate */ + if (waveartist_cmd2(chip, WACMD_INPUTSPEED, waveartist_rate(runtime))) + dev_warn(chip->card->dev, "error setting capture rate %dHz\n", + runtime->rate); + /* Set channel count */ + if (waveartist_cmd2(chip, WACMD_INPUTCHANNELS, runtime->channels)) + dev_warn(chip->card->dev, "error setting capture %d channels\n", + runtime->channels); + /* Set DMA channel */ + if (waveartist_cmd2(chip, WACMD_INPUTDMA, + chip->dma_capture > 3 ? WA_DMA_16BIT : WA_DMA_8BIT)) + dev_warn(chip->card->dev, "error setting capture data path\n"); + /* Set format */ + if (waveartist_cmd2(chip, WACMD_INPUTFORMAT, + waveartist_format(runtime->format))) + dev_warn(chip->card->dev, "error setting capture format %d\n", + runtime->format); + /* Set sample count */ + if (waveartist_cmd2(chip, WACMD_INPUTSIZE, count - 1)) + dev_warn(chip->card->dev, "error setting capture count %d\n", + count); + /* Configure DMA controller */ + snd_dma_program(chip->dma_capture, runtime->dma_addr, size, + DMA_MODE_READ | DMA_AUTOINIT); + + return 0; +} + +static int snd_waveartist_capture_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + waveartist_cmd1(chip, WACMD_INPUTSTART); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + waveartist_cmd1(chip, WACMD_INPUTSTOP); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + waveartist_cmd1(chip, WACMD_INPUTPAUSE); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + waveartist_cmd1(chip, WACMD_INPUTRESUME); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t snd_waveartist_playback_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + size_t size = snd_pcm_lib_buffer_bytes(substream); + size_t ptr = snd_dma_pointer(chip->dma_playback, size); + + return bytes_to_frames(substream->runtime, ptr); +} + +static snd_pcm_uframes_t snd_waveartist_capture_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + size_t size = snd_pcm_lib_buffer_bytes(substream); + size_t ptr = snd_dma_pointer(chip->dma_capture, size); + + return bytes_to_frames(substream->runtime, ptr); +} + +static struct snd_pcm_hardware snd_waveartist_playback = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_44100, + .rate_min = 4000, + .rate_max = 44100, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = 64, + .period_bytes_max = (128*1024), + .periods_min = 1, + .periods_max = 1024, +}; + +static struct snd_pcm_hardware snd_waveartist_capture = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_44100, + .rate_min = 4000, + .rate_max = 44100, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = 64, + .period_bytes_max = (128*1024), + .periods_min = 1, + .periods_max = 1024, +}; + +static int snd_waveartist_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + chip->playback_substream = substream; + substream->runtime->hw = snd_waveartist_playback; + + snd_pcm_limit_isa_dma_size(chip->dma_playback, + &substream->runtime->hw.buffer_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma_playback, + &substream->runtime->hw.period_bytes_max); + + return 0; +} + +static int snd_waveartist_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + chip->capture_substream = substream; + substream->runtime->hw = snd_waveartist_capture; + + snd_pcm_limit_isa_dma_size(chip->dma_capture, + &substream->runtime->hw.buffer_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma_capture, + &substream->runtime->hw.period_bytes_max); + + return 0; +} + +static int snd_waveartist_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + chip->playback_substream = NULL; + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static int snd_waveartist_capture_close(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + chip->capture_substream = NULL; + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static struct snd_pcm_ops snd_waveartist_playback_ops = { + .open = snd_waveartist_playback_open, + .close = snd_waveartist_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_waveartist_playback_hw_params, + .hw_free = snd_waveartist_pcm_hw_free, + .prepare = snd_waveartist_playback_prepare, + .trigger = snd_waveartist_playback_trigger, + .pointer = snd_waveartist_playback_pointer, +}; + +static struct snd_pcm_ops snd_waveartist_capture_ops = { + .open = snd_waveartist_capture_open, + .close = snd_waveartist_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_waveartist_capture_hw_params, + .hw_free = snd_waveartist_pcm_hw_free, + .prepare = snd_waveartist_capture_prepare, + .trigger = snd_waveartist_capture_trigger, + .pointer = snd_waveartist_capture_pointer, +}; + +static int snd_waveartist_pcm(struct snd_card *card) +{ + struct snd_waveartist *chip = card->private_data; + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(card, "WaveArtist", 0, 1, 1, &pcm); + if (err < 0) + return err; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_waveartist_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_waveartist_capture_ops); + + /* global setup */ + pcm->private_data = chip; + pcm->info_flags = 0; + if (chip->dma_playback == chip->dma_capture) + pcm->info_flags |= SNDRV_PCM_INFO_HALF_DUPLEX; + strcpy(pcm->name, card->shortname); + chip->pcm = pcm; + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_isa_data(), 64 * 1024, + chip->dma_playback > 3 || chip->dma_capture > 3 ? + 128 * 1024 : 64 * 1024); + + return 0; +} + +static void snd_waveartist_free(struct snd_card *card) +{ + struct snd_waveartist *chip = card->private_data; + + release_and_free_resource(chip->res_port); + if (chip->irq >= 0) + free_irq(chip->irq, card); + if (chip->dma_playback >= 0) { + disable_dma(chip->dma_playback); + free_dma(chip->dma_playback); + } + if (chip->dma_capture >= 0 && chip->dma_capture != chip->dma_playback) { + disable_dma(chip->dma_capture); + free_dma(chip->dma_capture); + } +} + +static int snd_waveartist_dev_free(struct snd_device *device) +{ + snd_waveartist_free(device->card); + + return 0; +} + +static int snd_waveartist_init(struct snd_waveartist *chip) +{ + if (waveartist_reset(chip)) + return -ENODEV; + dev_dbg(chip->card->dev, "chip rev. 0x%x\n", waveartist_getrev(chip)); + + waveartist_cmd1(chip, WACMD_RST_MIXER); /* reset mixer */ + waveartist_iack(chip); /* clear any pending interrupt */ + waveartist_set_ctlr(chip, 0, DMA1_IE | DMA0_IE); /* enable DMA IRQs */ + waveartist_iack(chip); /* clear any pending interrupt */ + + return 0; +} + +static int snd_waveartist_new_device(struct snd_card *card, + unsigned long port, + unsigned long mpu_port, + unsigned long fm_port, + int irq, int dma1, int dma2) +{ + struct snd_waveartist *chip = card->private_data; + static struct snd_device_ops ops = { + .dev_free = snd_waveartist_dev_free, + }; + int err; + + spin_lock_init(&chip->reg_lock); + chip->card = card; + chip->port = port; + chip->irq = -1; + chip->dma_playback = -1; + chip->dma_capture = -1; + + chip->res_port = request_region(port, 16, "WaveArtist"); + if (!chip->res_port) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab ports 0x%lx-0x%lx\n", + port, port + 16 - 1); + return -EBUSY; + } + + if (snd_waveartist_init(chip) < 0) { + snd_waveartist_free(card); + return -ENODEV; + } + + if (request_irq(irq, snd_waveartist_interrupt, 0, "WaveArtist", card)) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab IRQ %d\n", irq); + return -EBUSY; + } + chip->irq = irq; + + if (dma1 == dma2 || dma1 == SNDRV_AUTO_DMA || dma2 == SNDRV_AUTO_DMA) { + /* we have only one DMA channel */ + if (dma1 == SNDRV_AUTO_DMA) + dma1 = dma2; + if (request_dma(dma1, "WaveArtist")) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab DMA %d\n", + dma1); + return -EBUSY; + } + chip->dma_playback = chip->dma_capture = dma1; + } else { + /* + * 2 channels: use 16-bit for playback and 8-bit for capture as + * full-duplex works better this way. However, the chip seems to + * have some band-width limit so full-duplex at 44kHz/16-bit/2ch + * is not possible - some frames are dropped during capture, + * resulting in too-fast recording. If capture is done using + * lower rate or 8-bit or mono, everything is fine. + */ + if (dma2 > 3) + swap(dma1, dma2); + + if (request_dma(dma1, "WaveArtist playback")) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab playback DMA %d\n", + dma1); + return -EBUSY; + } + chip->dma_playback = dma1; + + if (dma2 != dma1 && request_dma(dma2, "WaveArtist capture")) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab capture DMA %d\n", + dma2); + return -EBUSY; + } + chip->dma_capture = dma2; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_waveartist_free(card); + return err; + } + + return 0; +} + +static int snd_waveartist_card_new(struct device *pdev, int dev, + struct snd_card **cardp) +{ + return snd_card_new(pdev, index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_waveartist), cardp); +} + +static int snd_waveartist_info_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 16) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : + SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + + return 0; +} + +static int snd_waveartist_get_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u16 val; + + val = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | reg << 8); + ucontrol->value.integer.value[0] = (val >> shift) & mask; + + return 0; +} + +static int snd_waveartist_put_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u16 val, old_val, new_val; + + val = (ucontrol->value.integer.value[0] & mask); + mask <<= shift; + val <<= shift; + + old_val = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | reg << 8); + new_val = (old_val & ~mask) | (val & mask); + + if (new_val == old_val) + return 0; + + /* new_val already contains register number (bits 12..15), bit 11 set */ + waveartist_cmd3(chip, WACMD_SET_MIXER, new_val, new_val); + + return 1; +} + +static int snd_waveartist_info_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 24) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : + SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask ? mask : 0x7fff; + + return 0; +} + +static int snd_waveartist_get_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int left_reg = kcontrol->private_value & 0xff; + int right_reg = (kcontrol->private_value >> 8) & 0xff; + int shift_left = (kcontrol->private_value >> 16) & 0x0f; + int shift_right = (kcontrol->private_value >> 20) & 0x0f; + int mask = (kcontrol->private_value >> 24) & 0xff; + u16 left, right; + + if (mask == 0) + mask = 0x7fff; + + left = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | left_reg << 8); + if (left_reg != right_reg) + right = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | + right_reg << 8); + else + right = left; + + ucontrol->value.integer.value[0] = (left >> shift_left) & mask; + ucontrol->value.integer.value[1] = (right >> shift_right) & mask; + + return 0; +} + +static int snd_waveartist_put_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int left_reg = kcontrol->private_value & 0xff; + int right_reg = (kcontrol->private_value >> 8) & 0xff; + int shift_left = (kcontrol->private_value >> 16) & 0x0f; + int shift_right = (kcontrol->private_value >> 20) & 0x0f; + int mask = (kcontrol->private_value >> 24) & 0xff; + u16 val1, val2, mask1, mask2; + u16 old_left, old_right, new_left, new_right; + + if (mask == 0) + mask = 0x7fff; + + val1 = ucontrol->value.integer.value[0] & mask; + val2 = ucontrol->value.integer.value[1] & mask; + val1 <<= shift_left; + val2 <<= shift_right; + mask1 = mask << shift_left; + mask2 = mask << shift_right; + + old_left = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | left_reg << 8); + if (left_reg != right_reg) + old_right = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | + right_reg << 8); + else + old_right = old_left; + + new_left = (old_left & ~mask1) | (val1 & mask1); + new_right = (old_right & ~mask2) | (val2 & mask2); + + if (new_left == old_left && new_right == old_right) + return 0; + + if (left_reg < 10) + /* new_* already contain reg. num. (bits 12..15), bit 11 set */ + waveartist_cmd3(chip, WACMD_SET_MIXER, new_left, new_right); + else + waveartist_cmd3(chip, WACMD_SET_LEVEL | + ((left_reg - 10) / 2) << 8, + new_left, new_right); + + return 1; +} + +static int snd_waveartist_info_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const mux_texts[] = { + "None", "Mix", "Line", "Phone", "CD", "Mic" + }; + + return snd_ctl_enum_info(uinfo, 2, ARRAY_SIZE(mux_texts), mux_texts); +} + +static int snd_waveartist_get_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int mux = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | 0x08 << 8); + + ucontrol->value.enumerated.item[0] = mux & 0x07; + ucontrol->value.enumerated.item[1] = (mux >> 3) & 0x07; + + return 0; +} + +static int snd_waveartist_put_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + u16 old_val, new_val; + u16 val1 = ucontrol->value.enumerated.item[0]; + u16 val2 = ucontrol->value.enumerated.item[1]; + + old_val = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | 0x08 << 8); + new_val = (old_val & ~0x3f) | (val1 & 0x07) | ((val2 & 0x07) << 3); + if (new_val == old_val) + return 0; + + /* new_val already contains register number (bits 12..15), bit 11 set */ + waveartist_cmd3(chip, WACMD_SET_MIXER, new_val, new_val); + + return 1; +} + +#define WAVEARTIST_SINGLE(xname, reg, shift, mask) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_waveartist_info_single, \ + .get = snd_waveartist_get_single, .put = snd_waveartist_put_single, \ + .private_value = reg | (shift << 8) | (mask << 16) \ +} + +#define WAVEARTIST_DOUBLE(xname, left_reg, right_reg, shift_left, shift_right, \ + mask) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_waveartist_info_double, \ + .get = snd_waveartist_get_double, .put = snd_waveartist_put_double, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ + (shift_right << 20) | (mask << 24) \ +} + +static struct snd_kcontrol_new snd_waveartist_controls[] = { +WAVEARTIST_DOUBLE("Master Playback Volume", 2, 6, 1, 1, 0x07), +WAVEARTIST_DOUBLE("Master Playback Switch", 0, 4, 0, 0, 1), +WAVEARTIST_DOUBLE("PCM Playback Volume", 10, 11, 0, 0, 0x00),/* 0x00 = 0x7fff */ +WAVEARTIST_DOUBLE("FM Playback Volume", 12, 13, 0, 0, 0x00),/* 0x00 = 0x7fff */ +/* WAVEARTIST_DOUBLE("Wavetable? Playback Volume", 14, 15, 0, 0, 0x00), */ +/* WAVEARTIST_DOUBLE("? Playback Volume", 16, 17, 0, 0, 0x00), */ +/* WAVEARTIST_DOUBLE("? Playback Volume", 18, 19, 0, 0, 0x00), */ +WAVEARTIST_DOUBLE("Digital Playback Switch", 3, 7, 10, 10, 1), /* PCM + FM */ +WAVEARTIST_DOUBLE("CD Playback Volume", 0, 4, 1, 1, 0x1f), +WAVEARTIST_DOUBLE("CD Playback Switch", 3, 7, 6, 6, 1), +WAVEARTIST_DOUBLE("Line Playback Volume", 0, 4, 6, 6, 0x1f), +WAVEARTIST_DOUBLE("Line Playback Switch", 3, 7, 4, 4, 1), +WAVEARTIST_DOUBLE("Phone Playback Volume", 1, 5, 6, 6, 0x1f), +WAVEARTIST_DOUBLE("Phone Playback Switch", 3, 7, 5, 5, 1), +WAVEARTIST_SINGLE("Mono Playback Volume", 8, 5, 0x1f), +WAVEARTIST_DOUBLE("Mono Playback Switch", 3, 7, 9, 9, 1), +WAVEARTIST_DOUBLE("Mic Playback Volume", 2, 6, 6, 6, 0x1f), +WAVEARTIST_DOUBLE("Mic 2 Playback Volume", 1, 5, 1, 1, 0x1f), +WAVEARTIST_DOUBLE("Mic Playback Switch", 3, 7, 7, 7, 1), +WAVEARTIST_DOUBLE("Mic 2 Playback Switch", 3, 7, 8, 8, 1), +WAVEARTIST_DOUBLE("Mic Gain", 2, 6, 4, 4, 0x03), +WAVEARTIST_DOUBLE("Capture Volume", 3, 7, 0, 0, 0x0f), +WAVEARTIST_SINGLE("Mono Output Playback Switch", 1, 0, 1), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_waveartist_info_mux, + .get = snd_waveartist_get_mux, + .put = snd_waveartist_put_mux, +} +}; + +static int snd_waveartist_mixer(struct snd_card *card) +{ + struct snd_waveartist *chip = card->private_data; + int err; + unsigned int idx; + + strcpy(card->mixername, chip->pcm->name); + + for (idx = 0; idx < ARRAY_SIZE(snd_waveartist_controls); idx++) { + struct snd_kcontrol *kctl; + + kctl = snd_ctl_new1(&snd_waveartist_controls[idx], chip); + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + } + + return 0; +} + +static int snd_waveartist_probe(struct snd_card *card, int dev) +{ + struct snd_waveartist *chip = card->private_data; + struct snd_opl3 *opl3; + int err; + + err = snd_waveartist_new_device(card, + port[dev], midi_port[dev], fm_port[dev], + irq[dev], dma1[dev], dma2[dev]); + if (err < 0) + return err; + + strcpy(card->driver, "WaveArtist"); + strcpy(card->shortname, "Rockwell WaveArtist RWA010"); + sprintf(card->longname, "%s at 0x%lx, irq %d, dma1 %d, dma2 %d", + card->shortname, chip->port, irq[dev], dma1[dev], dma2[dev]); + + err = snd_waveartist_pcm(card); + if (err < 0) + return err; + err = snd_waveartist_mixer(card); + if (err < 0) + return err; + + if (fm_port[dev] >= 0x388 && fm_port[dev] < 0x3f0) { + err = snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2, + OPL3_HW_OPL3, 0, &opl3); + if (err < 0) + return err; + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) + return err; + err = snd_opl3_hwdep_new(opl3, 0, 1, &chip->synth); + if (err < 0) + return err; + } + if (midi_port[dev] >= 0x300 && midi_port[dev] < 0x3f0) { + err = snd_mpu401_uart_new(card, 0, MPU401_HW_WAVEARTIST, + midi_port[dev], MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi); + if (err < 0) + return err; + } + + return snd_card_register(card); +} + +static int snd_waveartist_isa_match(struct device *pdev, + unsigned int dev) +{ + if (!enable[dev]) + return 0; +#ifdef CONFIG_PNP + if (isapnp[dev]) + return 0; +#endif + if (port[dev] == SNDRV_AUTO_PORT) { + dev_err(pdev, "specify port\n"); + return 0; + } + if (irq[dev] == SNDRV_AUTO_IRQ) { + dev_err(pdev, "specify irq\n"); + return 0; + } + if (dma1[dev] == SNDRV_AUTO_DMA) { + dev_err(pdev, "specify dma1\n"); + return 0; + } + + return 1; +} + +static int snd_waveartist_isa_probe(struct device *pdev, + unsigned int dev) +{ + struct snd_card *card; + int err; + + err = snd_waveartist_card_new(pdev, dev, &card); + if (err < 0) + return err; + err = snd_waveartist_probe(card, dev); + if (err < 0) { + snd_card_free(card); + return err; + } + dev_set_drvdata(pdev, card); + + return 0; +} + +static int snd_waveartist_isa_remove(struct device *devptr, + unsigned int dev) +{ + snd_card_free(dev_get_drvdata(devptr)); + + return 0; +} + +#ifdef CONFIG_PM +static int snd_waveartist_isa_suspend(struct device *dev, unsigned int n, + pm_message_t state) +{ + return snd_waveartist_suspend(dev_get_drvdata(dev), state); +} + +static int snd_waveartist_isa_resume(struct device *dev, unsigned int n) +{ + return snd_waveartist_resume(dev_get_drvdata(dev)); +} +#endif + +static struct isa_driver waveartist_isa_driver = { + .match = snd_waveartist_isa_match, + .probe = snd_waveartist_isa_probe, + .remove = snd_waveartist_isa_remove, +#ifdef CONFIG_PM + .suspend = snd_waveartist_isa_suspend, + .resume = snd_waveartist_isa_resume, +#endif + .driver = { .name = "waveartist" }, +}; + +#ifdef CONFIG_PNP +static int snd_waveartist_pnp_detect(struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + static int dev; + int err; + struct snd_card *card; + + for (; dev < SNDRV_CARDS; dev++) { + if (enable[dev] && isapnp[dev]) + break; + } + if (dev >= SNDRV_CARDS) + return -ENODEV; + + err = snd_waveartist_card_new(&pcard->card->dev, dev, &card); + if (err < 0) + return err; + err = snd_waveartist_pnp(dev, card->private_data, pcard, pid); + if (err < 0) { + dev_err(&pcard->card->dev, "PnP detection failed\n"); + snd_card_free(card); + return err; + } + err = snd_waveartist_probe(card, dev); + if (err < 0) { + snd_card_free(card); + return err; + } + pnp_set_card_drvdata(pcard, card); + dev++; + + return 0; +} + +static void snd_waveartist_pnp_remove(struct pnp_card_link *pcard) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + struct snd_waveartist *chip = card->private_data; + + /* disable forced IRQs */ + snd_waveartist_set_irq(chip->wa, 0); + snd_waveartist_set_irq(chip->mpu, 0); + + snd_card_free(pnp_get_card_drvdata(pcard)); + pnp_set_card_drvdata(pcard, NULL); +} + +#ifdef CONFIG_PM +static int snd_waveartist_pnp_suspend(struct pnp_card_link *pcard, + pm_message_t state) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + struct snd_waveartist *chip = card->private_data; + + snd_waveartist_suspend(card, state); + + /* disable forced IRQs to prevent opps */ + snd_waveartist_set_irq(chip->wa, 0); + snd_waveartist_set_irq(chip->mpu, 0); + + return 0; +} +static int snd_waveartist_pnp_resume(struct pnp_card_link *pcard) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + struct snd_waveartist *chip = card->private_data; + + /* re-enable forced IRQs */ + snd_waveartist_set_irq(chip->wa, chip->irq); + snd_waveartist_set_irq(chip->mpu, chip->irq); + + return snd_waveartist_resume(pnp_get_card_drvdata(pcard)); +} +#endif + +static struct pnp_card_driver waveartist_pnpc_driver = { + .flags = PNP_DRIVER_RES_DISABLE, + .name = "waveartist", + .id_table = snd_waveartist_pnpids, + .probe = snd_waveartist_pnp_detect, + .remove = snd_waveartist_pnp_remove, +#ifdef CONFIG_PM + .suspend = snd_waveartist_pnp_suspend, + .resume = snd_waveartist_pnp_resume, +#endif +}; +#endif /* CONFIG_PNP */ + +static int __init alsa_card_waveartist_init(void) +{ + int err; + + err = isa_register_driver(&waveartist_isa_driver, SNDRV_CARDS); +#ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + + err = pnp_register_card_driver(&waveartist_pnpc_driver); + if (!err) + pnp_registered = 1; + + if (isa_registered) + err = 0; +#endif + return err; +} + +static void __exit alsa_card_waveartist_exit(void) +{ +#ifdef CONFIG_PNP + if (pnp_registered) + pnp_unregister_card_driver(&waveartist_pnpc_driver); + + if (isa_registered) +#endif + isa_unregister_driver(&waveartist_isa_driver); +} + +module_init(alsa_card_waveartist_init) +module_exit(alsa_card_waveartist_exit) diff --git a/sound/isa/waveartist.h b/sound/isa/waveartist.h new file mode 100644 index 0000000..72254cf --- /dev/null +++ b/sound/isa/waveartist.h @@ -0,0 +1,74 @@ +/* + * Rockwell WaveArtist RWA010 chip register definitions + * + * Based on OSS WaveArtist driver by Hannu Savolainen + */ + +/* registers */ +#define CMDR 0 /* command (16-bit) */ +#define DATR 2 /* data (for PIO?) (16-bit) */ +#define CTLR 4 /* control */ +#define STATR 5 /* status */ +#define EXPCR1 6 /* expansion control 1 */ +#define EXPCR2 7 /* expansion control 2 */ +#define EXPDAT1 8 /* expansion data 1 (16-bit) */ +#define EXPDAT2 10 /* expansion data 2 (16-bit) */ +#define IRQSTAT 12 /* IRQ status */ + +/* STATR register bit definitions */ +#define CMD_WE BIT(7) /* CMDR write empty (ready for write) */ +#define CMD_RF BIT(6) /* CMDR read full (ready for read) */ +#define DAT_WE BIT(5) /* DATR write empty (ready for write) */ +#define DAT_RF BIT(4) /* DATR read full (ready for read) */ +#define IRQ_REQ BIT(3) /* IRQ was requested */ +#define DMA1 BIT(2) /* DMA1 IRQ was requested */ +#define DMA0 BIT(1) /* DMA0 IRQ was requested */ + +/* CTLR register bit definitions */ +#define CMD_WEIE BIT(7) /* CMDR write empty interrupt enable */ +#define CMD_RFIE BIT(6) /* CMDR read full interrupt enable */ +#define DAT_WEIE BIT(5) /* DATR write empty interrupt enable */ +#define DAT_RFIE BIT(4) /* DATR read full interrupt enable */ +#define RESET BIT(3) /* chip reset */ +#define DMA1_IE BIT(2) /* DMA1 interrupt enable */ +#define DMA0_IE BIT(1) /* DMA0 interrupt enable */ +#define IRQ_ACK BIT(0) /* IRQ acknowlege */ + +/* IRQSTAT register bit definitions */ +#define IRQ_MPU BIT(2) /* MPU-401 */ +#define IRQ_SB BIT(1) /* SB emulation */ +#define IRQ_PCM BIT(0) /* native PCM */ + +/* commands */ +#define WACMD_GETREV 0x00 + +#define WACMD_INPUTFORMAT 0x10 /* 0=S8, 1=S16, 2=U8 */ +#define WACMD_INPUTCHANNELS 0x11 /* 1=mono, 2=Stereo */ +#define WACMD_INPUTSPEED 0x12 /* sampling rate */ +#define WACMD_INPUTDMA 0x13 /* 0=8bit, 1=16bit, 2=PIO */ +#define WACMD_INPUTSIZE 0x14 /* samples to interrupt */ +#define WACMD_INPUTSTART 0x15 /* start ADC */ +#define WACMD_INPUTPAUSE 0x16 /* pause ADC */ +#define WACMD_INPUTSTOP 0x17 /* stop ADC */ +#define WACMD_INPUTRESUME 0x18 /* resume ADC */ +#define WACMD_INPUTPIO 0x19 /* PIO ADC */ + +#define WACMD_OUTPUTFORMAT 0x20 /* 0=S8, 1=S16, 2=U8 */ +#define WACMD_OUTPUTCHANNELS 0x21 /* 1=mono, 2=stereo */ +#define WACMD_OUTPUTSPEED 0x22 /* sampling rate */ +#define WACMD_OUTPUTDMA 0x23 /* 0=8bit, 1=16bit, 2=PIO */ +#define WACMD_OUTPUTSIZE 0x24 /* samples to interrupt */ +#define WACMD_OUTPUTSTART 0x25 /* start DAC */ +#define WACMD_OUTPUTPAUSE 0x26 /* pause DAC */ +#define WACMD_OUTPUTSTOP 0x27 /* stop DAC */ +#define WACMD_OUTPUTRESUME 0x28 /* resume DAC */ +#define WACMD_OUTPUTPIO 0x29 /* PIO DAC */ + +#define WACMD_GET_LEVEL 0x30 /* read mixer reg */ +#define WACMD_SET_LEVEL 0x31 /* set mixer regs (10..19) */ +#define WACMD_SET_MIXER 0x32 /* set mixer regs (0..9) */ +#define WACMD_RST_MIXER 0x33 /* mixer reset */ +#define WACMD_SET_MONO 0x34 /* set mono mode (|0x000=L, |0x100=R) */ + +enum wa_format { WA_FMT_S8 = 0, WA_FMT_S16, WA_FMT_U8 }; +enum wa_dma { WA_DMA_8BIT = 0, WA_DMA_16BIT, WA_DMA_PIO }; -- Ondrej Zary

1 1

[alsa-devel] [PATCH 0/7 linux-next] sound: constify of_device_id array
by Fabian Frederick 22 Mar '15

22 Mar '15

This small patchset adds const to of_device_id arrays in sound branch. Fabian Frederick (7): ALSA: aoa: constify of_device_id array ASoC: fsl: constify of_device_id array ASoC: kirkwood: constify of_device_id array ASoC: rt5631: constify of_device_id array ASoC: ak4554: constify of_device_id array ASoC: fsi: constify of_device_id array ASoC: rsnd: constify of_device_id array sound/aoa/soundbus/i2sbus/core.c | 2 +- sound/soc/codecs/ak4554.c | 2 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/fsl/mpc5200_psc_ac97.c | 2 +- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/pcm030-audio-fabric.c | 2 +- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/sh/fsi.c | 4 ++-- sound/soc/sh/rcar/core.c | 2 +- 9 files changed, 10 insertions(+), 10 deletions(-) -- 1.9.1

3 14

Re: [alsa-devel] Patches to bind the SGTL5000 chip to AM33XX McASP
by Peter Ujfalusi 22 Mar '15

22 Mar '15

On 03/19/2015 04:34 PM, Greg Knight wrote: >> SO is it so that the codec's MCLK is coming from McASP AHCLKX (or R) and this >> clock need to be present all the time? W/o the MCLK the registers are not >> accessible? > > Correct, the SGTL5000's I2C registers are not accessible without MCLK. Yeah, according to the datasheet, it needs MCLK in order to communicate with it. > From my testing, the clock need only be present when the audio chip is > in use. When not in use, the chip appears perfectly happy to not be > clocked. If you change audio controls while you don't have audio activity, you will still need to have the MCLK running. > Is there a better place to pull a ~12MHz clock signal for use as a > master clock from an AM335X? For a codec such as the SGTL5000 I would connect a clock, which is running all the time. If you take a look at the driver, it enables the clock at probe and leaves it running as long as the driver is loaded. But AM335x has CLKOUT1,2 outputs as well. They are not high quality clocks, but can be enabled/disabled. >> This will not work in mainline kernel. The McASP is turned off when it is not >> in use (no audio activity) so the AHCLKX/R is not going to be there on the >> ball output. > > The clock-to-talk hack in the patch enables the AHCLKX line during > codec initialization. From my testing, we don't seem to spend much > time communicating with the device when we're not using it. Disabling > the clock when it's not used should not be a problem. As I said, this is not going to work in upstream kernel. The McASP is not even powered up during probe time anymore. It is only powered up when it is needed, this means that you will not have clocks for the codec coming from McASP pin. The driver is not designed to support such a use case, when you use one of it's clock as a master clock for external chip. In other words the McASP driver is not a clock driver. It might be possible to create a clock driver on top of the McASP driver as child, but it is really something which should not be supported. > >> How do you configure the frequency of the AHCLKX/R, this hack only enables it, >> so you are going to have _some_ clock going out. > > The frequency is the 12 MHz (I think) I get out from AHCLKX without > any further configuration; I have not investigated changing that > frequency as it suited my needs. But I can look into doing > set_clkdiv/set_sysclk during the clock-to-talk enable if that's more > appropriate? > >> You should be using set_clkdiv and set_sysclk calls from the machine driver to >> configure the AHCLKX (we do not have AHCLKR supported in the driver ATM). > > So, ehm, the SGTL5000 component logic (is this the machine driver?) > should adjust the McASP's clocks? Am I misunderstanding? Would this > not impact binding the SGTL5000 to devices that are *not* McASPs/break > abstraction? > > TBH I'm not clear on how the SGTL5000 bindings to other hardware work. It works in other HW because the designers connected the SGTL5000's MCLK to a clock source which can be controlled or it is running all the time. In your case the designers chosen a clock line, which is not working as 'normal' clock. > Is it safe to assume set_sysclk is always going to adjust the I2S MCLK > speed? Please pardon my crude questions, I'm not deeply familiar with > the structure of ALSA. With the setclkdiv/sysclk you can configure the AHCLKX direction and configure the divider to be used to generate the clock which goes out, or to divide down the clock which is coming in. Another issue this hack has is: what happens if for some reason the SGTL5000 driver loads first? McASP will not enable the AHCLKX so the codec driver will fail reading the CHIP_ID and fails to load. You need to provide a dummy clock node to the codec, so that will not fail, but you will not have the MCLK. > I'll reformat the patches and try from a mainline kernel when I get > some spare time. I'm not sure if this can be accepted, sorry. It is really a hack and there are too many corner cases where the system can fail. -- Péter

3 4

Re: [alsa-devel] Patches to bind the SGTL5000 chip to AM33XX McASP
by Mark Brown 22 Mar '15

22 Mar '15

On Thu, Mar 19, 2015 at 01:48:08PM -0400, Greg Knight wrote: > Is there a straightforward way we could enable the clock on-demand - > when we're communicating with the codec/mixer or actually using the > audio subsystem - while keeping it disabled when not in use? We don't > expect to be using the audio subsystem frequently. This is what the clock API is there for - the driver can enable the clock whenver it needs it and disable it when it doesn't.

1 0

[alsa-devel] [PATCH] ASoC: Intel: remove misleading DMA error messages on Baytrail platforms
by Pierre-Louis Bossart 22 Mar '15

22 Mar '15

During probe, the Baytrail audio driver reports errors such as: [44.172040] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [44.172137] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed Those error messages are misleading, there is no error since the DMA is explicitly not configured for Baytrail. Add a test to remove DMA error checks when DMA is not configured and return silently. Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- With Mark in cc: this time, sorry for the noise. sound/soc/intel/sst-acpi.c | 1 + sound/soc/intel/sst-dsp.h | 1 + sound/soc/intel/sst-firmware.c | 4 ++++ 3 files changed, 6 insertions(+) diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index b3d8456..42f293f 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -142,6 +142,7 @@ static int sst_acpi_probe(struct platform_device *pdev) sst_acpi->desc = desc; sst_acpi->mach = mach; + sst_pdata->resindex_dma_base = desc->resindex_dma_base; if (desc->resindex_dma_base >= 0) { sst_pdata->dma_engine = desc->dma_engine; sst_pdata->dma_base = desc->resindex_dma_base; diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index f291e32..148d8c5 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -206,6 +206,7 @@ struct sst_pdata { const struct firmware *fw; /* DMA */ + int resindex_dma_base; /* other fields invalid if equals to -1 */ u32 dma_base; u32 dma_size; int dma_engine; diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 5f71ef6..4a993d1 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -271,6 +271,10 @@ int sst_dma_new(struct sst_dsp *sst) const char *dma_dev_name; int ret = 0; + if (sst->pdata->resindex_dma_base == -1) + /* DMA is not used, return and squelsh error messages */ + return 0; + /* configure the correct platform data for whatever DMA engine * is attached to the ADSP IP. */ switch (sst->pdata->dma_engine) { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC:pcm512x: Fix divide by zero issue.
by Howard Mitchell 22 Mar '15

22 Mar '15

If den=1 and pllin_rate>20MHz then den and num are adjusted to 0 causing a divide by zero error a few lines further on. Therefore this patch correctly scales num and den such that pllin_rate/den < 20MHz as required in the device data sheet. Signed-off-by: Howard Mitchell <hm(a)hmbedded.co.uk> --- sound/soc/codecs/pcm512x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 5301c4a..8472099 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -713,8 +713,8 @@ static int pcm512x_find_pll_coeff(struct snd_soc_dai *dai, /* pllin_rate / P (or here, den) cannot be greater than 20 MHz */ if (pllin_rate / den > 20000000 && num < 8) { - num *= 20000000 / (pllin_rate / den); - den *= 20000000 / (pllin_rate / den); + num *= DIV_ROUND_UP(pllin_rate / den, 20000000); + den *= DIV_ROUND_UP(pllin_rate / den, 20000000); } dev_dbg(dev, "num / den = %lu / %lu\n", num, den); -- 1.7.9.5

3 2

Re: [alsa-devel] dummy codec and the dummy platform use the same name: snd-soc-dummy
by Mark Brown 22 Mar '15

22 Mar '15

On Fri, Mar 20, 2015 at 11:27:35PM -0700, Zhao Ye wrote: > Then in > static void soc_init_codec_debugfs(struct snd_soc_codec *codec) > and, > static void soc_init_platform_debugfs(struct snd_soc_platform *platform) > this two place will make two debugfs_create_dir() with the " snd-soc-dummy" name twice. > So I think we should change one of the names. Well, upstream what I'd expect to be happening here is that we merge everything into a single dummy component which avoids the problem another way.

1 0

Re: [alsa-devel] [PATCH] ASoC: wm8741: Fix rates constraints values
by Charles Keepax 22 Mar '15

22 Mar '15

On Fri, Mar 13, 2015 at 10:20:14PM +0100, Sergej Sawazki wrote: > According to table 11 in the datasheet (Rev 4.3): > Example: With a master clock (MCLK) of 11.2896MHz, the following sampling rates (LRCLK) are supported: > 44.1kHz (MCLK/256), 88.2kHz (MCLK/128) > > For the rates lists, we should use 82000 instead of 88235 and 176400 instead of 1764000 (seems to be a typo). > > Signed-off-by: Sergej Sawazki <ce3a(a)gmx.de> > --- Acked-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Thanks, Charles

2 1

[alsa-devel] [PATCH 1/2] ALSA:ControlName.txt: Add 'Analogue' as a prefix for volume controls
by Howard Mitchell 21 Mar '15

21 Mar '15

Currently 'Playback Volume' is the correct way to express an analogue volume control. However, this control name has initialisation defaults applied when using 'alsactl restore' and in some cases this is not appropriate. An example would be a control that has a selection of 0db and -6dB of gain that is intended to set the fullscale ouput voltage of a DAC. The TI pcm512x family of DAcs have such a control. In this case the device/driver reset defaults are preferred. Signed-off-by: Howard Mitchell <hm(a)hmbedded.co.uk> --- Documentation/sound/alsa/ControlNames.txt | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt index 79a6127..3fc1cf50 100644 --- a/Documentation/sound/alsa/ControlNames.txt +++ b/Documentation/sound/alsa/ControlNames.txt @@ -71,11 +71,11 @@ SOURCE: HDMI/DP (either HDMI or DisplayPort) Exceptions (deprecated): - [Digital] Capture Source - [Digital] Capture Switch (aka input gain switch) - [Digital] Capture Volume (aka input gain volume) - [Digital] Playback Switch (aka output gain switch) - [Digital] Playback Volume (aka output gain volume) + [Analogue|Digital] Capture Source + [Analogue|Digital] Capture Switch (aka input gain switch) + [Analogue|Digital] Capture Volume (aka input gain volume) + [Analogue|Digital] Playback Switch (aka output gain switch) + [Analogue|Digital] Playback Volume (aka output gain volume) Tone Control - Switch Tone Control - Bass Tone Control - Treble -- 1.7.9.5

3 3

[alsa-devel] [PATCH 0/4] More aggressive PM for HD-audio
by Takashi Iwai 21 Mar '15

21 Mar '15

Hi, here is a patchset for supporting more aggressive PM for HD-audio. This allows to change the power state of each widget more dynamically with jack and stream states. It's activated only when the codec driver (or via sysfs or f/w patch) sets codec->power_mgmt flag. In theory, this should work for the recent Realtek codecs, but currently I have no machine for test. David, could you or your team check whether this works for ALC282 or such? Just add like: --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5415,6 +5415,7 @@ static int patch_alc269(struct hda_codec *codec) spec = codec->spec; spec->gen.shared_mic_vref_pin = 0x18; + codec->power_mgmt = 1; snd_hda_pick_fixup(codec, alc269_fixup_models, alc269_fixup_tbl, alc269_fixups); The patchset is for for-next branch of sound git tree, but they might be applicable to 4.0-rc (or even older), too. The current patches are found in topic/hda-power branch. thanks, Takashi === Takashi Iwai (4): ALSA: hda - Simplify PCM setup overrides ALSA: hda - Support advanced power state controls ALSA: hda - Use the new power control for VIA codecs ALSA: hda - Adjust power of beep widget and outputs sound/pci/hda/hda_beep.c | 29 +- sound/pci/hda/hda_beep.h | 1 + sound/pci/hda/hda_codec.c | 4 + sound/pci/hda/hda_codec.h | 2 + sound/pci/hda/hda_generic.c | 480 ++++++++++++++++++++++++------ sound/pci/hda/hda_generic.h | 5 +- sound/pci/hda/patch_realtek.c | 41 --- sound/pci/hda/patch_sigmatel.c | 5 + sound/pci/hda/patch_via.c | 662 +---------------------------------------- 9 files changed, 427 insertions(+), 802 deletions(-) -- 2.3.3

3 11

[alsa-devel] ASoC: Max98090 Codec DAPM Widget setup in Open Embedded
by C. Omer Rafique 20 Mar '15

20 Mar '15

I would like to know which file I can use to connect/enable DPAM widgets permanently so a simple aplay/record can work? My environment is: - Yocto Linux Kernel 3.10 - (DTS based) - Max98090 Codec - imx6SL SoC I thought about doing this in alsa.conf or asound.conf but I couldn't find a single example with DAPM widgets . Thank you Choudhary

1 0

[alsa-devel] [PATCH] ALSA: hda - Rename power_mgmt flag with power_save_node
by Takashi Iwai 20 Mar '15

20 Mar '15

David suggested that the name "power_mgmt" is too ambiguous. Rename the flag with a bit clearer one "power_save_node". Also, add the corresponding description to HD-Audio.txt, too. Reported-by: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa/HD-Audio.txt | 6 +++++- sound/pci/hda/hda_codec.h | 2 +- sound/pci/hda/hda_generic.c | 30 +++++++++++++++--------------- sound/pci/hda/patch_sigmatel.c | 8 ++++---- sound/pci/hda/patch_via.c | 12 ++++++------ 5 files changed, 31 insertions(+), 27 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 42a0a39b77e6..e7193aac669c 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -466,7 +466,11 @@ The generic parser supports the following hints: - add_jack_modes (bool): add "xxx Jack Mode" enum controls to each I/O jack for allowing to change the headphone amp and mic bias VREF capabilities -- power_down_unused (bool): power down the unused widgets +- power_save_node (bool): advanced power management for each widget, + controlling the power sate (D0/D3) of each widget node depending on + the actual pin and stream states +- power_down_unused (bool): power down the unused widgets, a subset of + power_save_node, and will be dropped in future - add_hp_mic (bool): add the headphone to capture source if possible - hp_mic_detect (bool): enable/disable the hp/mic shared input for a single built-in mic case; default true diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 148e84ce61cf..ccf355d4a8fa 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -371,7 +371,7 @@ struct hda_codec { unsigned int cached_write:1; /* write only to caches */ unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ - unsigned int power_mgmt:1; /* advanced PM for each widget */ + unsigned int power_save_node:1; /* advanced PM for each widget */ #ifdef CONFIG_PM unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ atomic_t in_pm; /* suspend/resume being performed */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 94c7f43ad182..de4024db27b4 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -140,9 +140,9 @@ static void parse_user_hints(struct hda_codec *codec) val = snd_hda_get_bool_hint(codec, "single_adc_amp"); if (val >= 0) codec->single_adc_amp = !!val; - val = snd_hda_get_bool_hint(codec, "power_mgmt"); + val = snd_hda_get_bool_hint(codec, "power_save_node"); if (val >= 0) - codec->power_mgmt = !!val; + codec->power_save_node = !!val; val = snd_hda_get_bool_hint(codec, "auto_mute"); if (val >= 0) @@ -661,7 +661,7 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, struct nid_path *path = snd_array_elem(&spec->paths, n); if (!path->active) continue; - if (codec->power_mgmt) { + if (codec->power_save_node) { if (!path->stream_enabled && !path->pin_fixed) continue; /* ignore unplugged paths except for DAC/ADC */ @@ -879,8 +879,8 @@ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path, path->active = false; /* make sure the widget is powered up */ - if (enable && (spec->power_down_unused || codec->power_mgmt)) - path_power_update(codec, path, codec->power_mgmt); + if (enable && (spec->power_down_unused || codec->power_save_node)) + path_power_update(codec, path, codec->power_save_node); for (i = path->depth - 1; i >= 0; i--) { hda_nid_t nid = path->path[i]; @@ -905,7 +905,7 @@ static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path) { struct hda_gen_spec *spec = codec->spec; - if (!(spec->power_down_unused || codec->power_mgmt) || path->active) + if (!(spec->power_down_unused || codec->power_save_node) || path->active) return; sync_power_state_change(codec, path_power_update(codec, path, true)); } @@ -3981,7 +3981,7 @@ static hda_nid_t set_pin_power_jack(struct hda_codec *codec, hda_nid_t pin, { bool on; - if (!codec->power_mgmt) + if (!codec->power_save_node) return 0; on = snd_hda_jack_detect_state(codec, pin) != HDA_JACK_NOT_PRESENT; @@ -4038,7 +4038,7 @@ static void add_all_pin_power_ctls(struct hda_codec *codec, bool on) struct auto_pin_cfg *cfg = &spec->autocfg; int i; - if (!codec->power_mgmt) + if (!codec->power_save_node) return; add_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins, on); if (cfg->line_out_type != AUTO_PIN_HP_OUT) @@ -4067,7 +4067,7 @@ static void sync_all_pin_power_ctls(struct hda_codec *codec) struct auto_pin_cfg *cfg = &spec->autocfg; int i; - if (!codec->power_mgmt) + if (!codec->power_save_node) return; sync_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins); if (cfg->line_out_type != AUTO_PIN_HP_OUT) @@ -4111,7 +4111,7 @@ static int add_fake_beep_paths(struct hda_codec *codec) hda_nid_t nid = spec->beep_nid; int err; - if (!codec->power_mgmt || !nid) + if (!codec->power_save_node || !nid) return 0; err = add_fake_paths(codec, nid, cfg->line_outs, cfg->line_out_pins); if (err < 0) @@ -4232,7 +4232,7 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, } set_pin_eapd(codec, nid, !mute); - if (codec->power_mgmt) { + if (codec->power_save_node) { bool on = !mute; if (on) on = snd_hda_jack_detect_state(codec, nid) @@ -4740,11 +4740,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) * @nid: audio widget * @on: power on/off flag * - * Set this in patch_ops.stream_pm. Only valid with power_mgmt flag. + * Set this in patch_ops.stream_pm. Only valid with power_save_node flag. */ void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on) { - if (codec->power_mgmt) + if (codec->power_save_node) set_path_power(codec, nid, -1, on); } EXPORT_SYMBOL_GPL(snd_hda_gen_stream_pm); @@ -4915,14 +4915,14 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, dig_only: parse_digital(codec); - if (spec->power_down_unused || codec->power_mgmt) + if (spec->power_down_unused || codec->power_save_node) codec->power_filter = snd_hda_gen_path_power_filter; if (!spec->no_analog && spec->beep_nid) { err = snd_hda_attach_beep_device(codec, spec->beep_nid); if (err < 0) return err; - if (codec->beep && codec->power_mgmt) { + if (codec->beep && codec->power_save_node) { err = add_fake_beep_paths(codec); if (err < 0) return err; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7e531d5cde51..5b7c173adcb8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4494,7 +4494,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return err; spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.mixer_nid = 0x1d; spec->have_spdif_mux = 1; @@ -4600,7 +4600,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) codec->epss = 0; /* longer delay needed for D3 */ spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -4650,7 +4650,7 @@ static int patch_stac92hd95(struct hda_codec *codec) codec->epss = 0; /* longer delay needed for D3 */ spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -4692,7 +4692,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) return err; spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index d5d1dca4f11b..485663bb9101 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -133,7 +133,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) spec->gen.keep_eapd_on = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->gen.power_down_unused = 1; return spec; } @@ -236,7 +236,7 @@ static int via_pin_power_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = codec->power_mgmt; + ucontrol->value.enumerated.item[0] = codec->power_save_node; return 0; } @@ -247,9 +247,9 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol, struct via_spec *spec = codec->spec; bool val = !!ucontrol->value.enumerated.item[0]; - if (val == codec->power_mgmt) + if (val == codec->power_save_node) return 0; - codec->power_mgmt = val; + codec->power_save_node = val; spec->gen.power_down_unused = val; analog_low_current_mode(codec); return 1; @@ -295,7 +295,7 @@ static void __analog_low_current_mode(struct hda_codec *codec, bool force) bool enable; unsigned int verb, parm; - if (!codec->power_mgmt) + if (!codec->power_save_node) enable = false; else enable = is_aa_path_mute(codec) && !spec->gen.active_streams; @@ -517,7 +517,7 @@ static int via_parse_auto_config(struct hda_codec *codec) return err; /* disable widget PM at start for compatibility */ - codec->power_mgmt = 0; + codec->power_save_node = 0; spec->gen.power_down_unused = 0; return 0; } -- 2.3.3

1 0

[alsa-devel] [PATCH] ASoC: max98925: Constify regmap config and other codec data
by Krzysztof Kozlowski 20 Mar '15

20 Mar '15

Constify local structures (snd_soc_dai_ops, snd_soc_codec_driver, regmap_config) and array (reg_defaults) which are not modified by the driver and passed to core as pointer to const. Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> --- sound/soc/codecs/max98925.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 1f8e80315749..0796dfaeb139 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -34,7 +34,7 @@ static const char *const hpf_text[] = { "Disable", "DC Block", "100Hz", "200Hz", "400Hz", "800Hz", }; -static struct reg_default max98925_reg[] = { +static const struct reg_default max98925_reg[] = { { 0x0B, 0x00 }, /* IRQ Enable0 */ { 0x0C, 0x00 }, /* IRQ Enable1 */ { 0x0D, 0x00 }, /* IRQ Enable2 */ @@ -491,7 +491,7 @@ static int max98925_dai_set_sysclk(struct snd_soc_dai *dai, #define MAX98925_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops max98925_dai_ops = { +static const struct snd_soc_dai_ops max98925_dai_ops = { .set_sysclk = max98925_dai_set_sysclk, .set_fmt = max98925_dai_set_fmt, .hw_params = max98925_dai_hw_params, @@ -541,7 +541,7 @@ static int max98925_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_max98925 = { +static const struct snd_soc_codec_driver soc_codec_dev_max98925 = { .probe = max98925_probe, .controls = max98925_snd_controls, .num_controls = ARRAY_SIZE(max98925_snd_controls), @@ -551,7 +551,7 @@ static struct snd_soc_codec_driver soc_codec_dev_max98925 = { .num_dapm_widgets = ARRAY_SIZE(max98925_dapm_widgets), }; -static struct regmap_config max98925_regmap = { +static const struct regmap_config max98925_regmap = { .reg_bits = 8, .val_bits = 8, .max_register = MAX98925_REV_VERSION, -- 1.9.1

2 1

[alsa-devel] [RFC][PATCH 00/13] alsa-tools: libhinawa for control applications of FireWire devices
by Takashi Sakamoto 20 Mar '15

20 Mar '15

This is RFC for a new library into alsa-tools. ALSA in Linux 3.16 or later extends a support for FireWire sound devices. Currently ALSA drivers supports streaming functionality only, while most of these devices require software implementation to control its internal DSP. The way to achieve this is to transfer byte sequence to the unit and wait byte sequence which the unit transfers if required. This library, libhinawa, just support these operations, nothing others. The supported types of transactions are: - IEEE 1394 read transaction - IEEE 1394 write transaction - IEEE 1394 lock transaction - IEC 61883-1 FCP transaction - Echo Fireworks transaction (with a help of snd-fireworks kernel driver) - Dice notification (with a help of snd-dice kernel driver) To help new developers, this library supports GObject Introspection for language bindings. The main logic of applications can be written with preferred languages such as Python, Ruby, Perl and so on. In the end of this patchset, some Python scripts with Gtk+, Qt4 and Qt5 are committed as samples. This is my first development with GNU Autotools, GLib/GObject and GObject Introspection. Furthermore, I'm a beginner of FireWire subsystem programming. I'm happy to receive your comments, especially: - the way to write Linux version dependency in configure.ac - the way of libtool versioning - the value of poll timeout in fw_unit/snd_unit - the value of thread condition timeout in fw_fcp/snd_dice/snd_efw - improvements of 'unit_query' object - any programming mistakes (threading and so on...) Regards Takashi Sakamoto (13): libhinawa: add build definitions libhinawa: add hinawa context libhinawa: support GTK-Doc to generate references libhinawa: add 'fw_unit' object as a listener for FireWire unit libhinawa: support GObject Introspection for language bindings libhinawa: add 'fw_resp' object as a responder for FireWire transaction libhinawa: add 'fw_req' object as requester for FireWire transaction libhinawa: add 'fw_fcp' object as a helper of FCP transaction libhinawa: add 'snd_unit' object as a listener for ALSA FireWire devices libhinawa: add 'snd_dice' object as a helper for Dice notification libhinawa: add 'snd_efw' object as a helper for EFW transaction libhinawa: add 'unit_query' as a query for ALSA FireWire devices libhinawa: add sample scripts Makefile | 2 +- libhinawa/AUTHORS | 1 + libhinawa/COPYING | 504 ++++++++++++++++++++++++++++ libhinawa/ChangeLog | 5 + libhinawa/Makefile.am | 6 + libhinawa/NEWS | 0 libhinawa/README | 30 ++ libhinawa/autogen.sh | 11 + libhinawa/configure.ac | 68 ++++ libhinawa/doc/Makefile.am | 2 + libhinawa/doc/reference/Makefile.am | 46 +++ libhinawa/doc/reference/hinawa-docs.sgml | 47 +++ libhinawa/doc/reference/version.xml.in | 1 + libhinawa/samples/gtk3.py | 190 +++++++++++ libhinawa/samples/qt4.py | 206 ++++++++++++ libhinawa/samples/qt5.py | 192 +++++++++++ libhinawa/samples/run.sh | 15 + libhinawa/src/Makefile.am | 97 ++++++ libhinawa/src/backport.h | 41 +++ libhinawa/src/fw_fcp.c | 282 ++++++++++++++++ libhinawa/src/fw_fcp.h | 58 ++++ libhinawa/src/fw_req.c | 265 +++++++++++++++ libhinawa/src/fw_req.h | 56 ++++ libhinawa/src/fw_resp.c | 232 +++++++++++++ libhinawa/src/fw_resp.h | 51 +++ libhinawa/src/fw_unit.c | 343 +++++++++++++++++++ libhinawa/src/fw_unit.h | 52 +++ libhinawa/src/hinawa_context.c | 60 ++++ libhinawa/src/hinawa_context.h | 10 + libhinawa/src/internal.h | 31 ++ libhinawa/src/snd_dice.c | 195 +++++++++++ libhinawa/src/snd_dice.h | 53 +++ libhinawa/src/snd_efw.c | 315 ++++++++++++++++++ libhinawa/src/snd_efw.h | 54 +++ libhinawa/src/snd_unit.c | 548 +++++++++++++++++++++++++++++++ libhinawa/src/snd_unit.h | 69 ++++ libhinawa/src/unit_query.c | 116 +++++++ libhinawa/src/unit_query.h | 48 +++ 38 files changed, 4301 insertions(+), 1 deletion(-) create mode 100644 libhinawa/AUTHORS create mode 100644 libhinawa/COPYING create mode 100644 libhinawa/ChangeLog create mode 100644 libhinawa/Makefile.am create mode 100644 libhinawa/NEWS create mode 100644 libhinawa/README create mode 100755 libhinawa/autogen.sh create mode 100644 libhinawa/configure.ac create mode 100644 libhinawa/doc/Makefile.am create mode 100644 libhinawa/doc/reference/Makefile.am create mode 100644 libhinawa/doc/reference/hinawa-docs.sgml create mode 100644 libhinawa/doc/reference/version.xml.in create mode 100755 libhinawa/samples/gtk3.py create mode 100755 libhinawa/samples/qt4.py create mode 100755 libhinawa/samples/qt5.py create mode 100755 libhinawa/samples/run.sh create mode 100644 libhinawa/src/Makefile.am create mode 100644 libhinawa/src/backport.h create mode 100644 libhinawa/src/fw_fcp.c create mode 100644 libhinawa/src/fw_fcp.h create mode 100644 libhinawa/src/fw_req.c create mode 100644 libhinawa/src/fw_req.h create mode 100644 libhinawa/src/fw_resp.c create mode 100644 libhinawa/src/fw_resp.h create mode 100644 libhinawa/src/fw_unit.c create mode 100644 libhinawa/src/fw_unit.h create mode 100644 libhinawa/src/hinawa_context.c create mode 100644 libhinawa/src/hinawa_context.h create mode 100644 libhinawa/src/internal.h create mode 100644 libhinawa/src/snd_dice.c create mode 100644 libhinawa/src/snd_dice.h create mode 100644 libhinawa/src/snd_efw.c create mode 100644 libhinawa/src/snd_efw.h create mode 100644 libhinawa/src/snd_unit.c create mode 100644 libhinawa/src/snd_unit.h create mode 100644 libhinawa/src/unit_query.c create mode 100644 libhinawa/src/unit_query.h -- 2.1.0

5 20

Re: [alsa-devel] Patches to bind the SGTL5000 chip to AM33XX McASP
by Peter Ujfalusi 20 Mar '15

20 Mar '15

Greg, On 03/19/2015 07:48 PM, Greg Knight wrote: >> For a codec such as the SGTL5000 I would connect a clock, which is > running all >> the time. If you take a look at the driver, it enables the clock at > probe and >> leaves it running as long as the driver is loaded. > > Speaking briefly to my "electrical" colleague, he's a bit concerned > about the extra power load that a crystal oscillator might incur, while > I'm concerned about audio quality from a source like CLKOUT2. > > Is there a straightforward way we could enable the clock on-demand - > when we're communicating with the codec/mixer or actually using the > audio subsystem - while keeping it disabled when not in use? We don't > expect to be using the audio subsystem frequently. > > Worst-case we can set up a GPIO and enable/disable it as needed in > user-space - acceptable, if annoying, for our application. See my reply to Nikolay. -- Péter

1 0

[alsa-devel] Can iMX6SL supports different sampling rate for different SSIs?
by Xuebing Wang 20 Mar '15

20 Mar '15

Nicolin, Thank you very much for your always help. I am using 3 SSIs (as I2S master mode) on iMX6SL for 3 different codec. 1) If I read iMX6SL reference manual correctly, I am deriving SSI sys_clock from pll4. -- pll4 = Fref * (DIV_SELECT + NUM/DENOM) ** I am using a value of 786.432000 MHz -- pll4_main_clk = pll4 / (PLL_AUDIOn: POST_DIV_SELECT) / CCM_ANALOG_MISC2n: MSB:LSB ** I am using a value of 196.608 MHz -- SSI1_CLK_ROOT = pll4_main_clk / (CS1CDR: ssi1_clk_pred) / CS1CDR: ssi1_clk_podf ** I am using a value of 3.072 MHz (for 48k sampling rate) 2) As pll4_main_clk is global for all 3 SSIs, I think that we can *not* use 8k for ssi1 and 44.1k for ssi2, right? Because 8k and 44.1k requires different pll4 clock, right? -- However, as 48k, 32k, 16k, 8k can use same pll4, thus different SSI can use different sampling rate in this subset (48k, 32k, 16k, 8k). -- Same principle applies for 44.1k, 22.050, 11.025 subset of sampling rates. Is my understanding correct? -- Xuebing

2 3

[alsa-devel] [PATCH v2.1] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used
by Jyri Sarha 19 Mar '15

19 Mar '15

Set rule constraints to allow only combinations of sample-rate, sample-bits, and channels that can be played/captured with reasonable sample-rate accuracy. The logic with tdm-slots and serializers (=i2s data wires) goes like this: The first wire will take all channels up to number of tdm-slots, before following wires (if any) are used. If the first wire is used fully, the remaining wires share the same clocks and the divider can be calculated for the first wire. Also, takes the number of tdm-slots into account when implicitly selecting the BLCK divider. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- This is othetwise the same patch as v2, but I added Peter's ack and rebased on top of: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/davinci sound/soc/davinci/davinci-mcasp.c | 195 ++++++++++++++++++++++++++++++++++++-- 1 file changed, 185 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d40b392..625cca1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include <linux/of_platform.h> #include <linux/of_device.h> #include <linux/platform_data/davinci_asp.h> +#include <linux/math64.h> #include <sound/asoundef.h> #include <sound/core.h> @@ -65,6 +66,11 @@ struct davinci_mcasp_context { bool pm_state; }; +struct davinci_mcasp_ruledata { + struct davinci_mcasp *mcasp; + int serializers; +}; + struct davinci_mcasp { struct snd_dmaengine_dai_dma_data dma_data[2]; void __iomem *base; @@ -99,6 +105,8 @@ struct davinci_mcasp { #ifdef CONFIG_PM_SLEEP struct davinci_mcasp_context context; #endif + + struct davinci_mcasp_ruledata ruledata[2]; }; static inline void mcasp_set_bits(struct davinci_mcasp *mcasp, u32 offset, @@ -868,6 +876,30 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, return 0; } +static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, + unsigned int bclk_freq, + int *error_ppm) +{ + int div = mcasp->sysclk_freq / bclk_freq; + int rem = mcasp->sysclk_freq % bclk_freq; + + if (rem != 0) { + if (div == 0 || + ((mcasp->sysclk_freq / div) - bclk_freq) > + (bclk_freq - (mcasp->sysclk_freq / (div+1)))) { + div++; + rem = rem - bclk_freq; + } + } + if (error_ppm) + *error_ppm = + (div*1000000 + (int)div64_long(1000000LL*rem, + (int)bclk_freq)) + /div - 1000000; + + return div; +} + static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -883,16 +915,20 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, * the machine driver, we need to calculate the ratio. */ if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { - unsigned int bclk_freq = snd_soc_params_to_bclk(params); - unsigned int div = mcasp->sysclk_freq / bclk_freq; - if (mcasp->sysclk_freq % bclk_freq != 0) { - if (((mcasp->sysclk_freq / div) - bclk_freq) > - (bclk_freq - (mcasp->sysclk_freq / (div+1)))) - div++; - dev_warn(mcasp->dev, - "Inaccurate BCLK: %u Hz / %u != %u Hz\n", - mcasp->sysclk_freq, div, bclk_freq); - } + int channels = params_channels(params); + int rate = params_rate(params); + int sbits = params_width(params); + int ppm, div; + + if (channels > mcasp->tdm_slots) + channels = mcasp->tdm_slots; + + div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*channels, + &ppm); + if (ppm) + dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", + ppm); + __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); } @@ -974,6 +1010,108 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, return ret; } +static const unsigned int davinci_mcasp_dai_rates[] = { + 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, + 88200, 96000, 176400, 192000, +}; + +#define DAVINCI_MAX_RATE_ERROR_PPM 1000 + +static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + int sbits = params_width(params); + int channels = params_channels(params); + unsigned int list[ARRAY_SIZE(davinci_mcasp_dai_rates)]; + int i, count = 0; + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (i = 0; i < ARRAY_SIZE(davinci_mcasp_dai_rates); i++) { + uint bclk_freq = sbits*channels*davinci_mcasp_dai_rates[i]; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = davinci_mcasp_dai_rates[i]; + + } + dev_dbg(rd->mcasp->dev, "%d frequencies for %d sbits and %d channels\n", + count, sbits, channels); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_samplebits(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + int channels = params_channels(params); + int rate = params_rate(params); + unsigned int list[4]; + int sbits, count = 0; + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (sbits = 8; sbits <= 32; sbits += 8) { + uint bclk_freq = channels*sbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = sbits; + } + dev_dbg(rd->mcasp->dev, + "%d possible sample size for %d Hz and %d channels\n", + count, rate, channels); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_interval *ci = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + int sbits = params_width(params); + int rate = params_rate(params); + int max_chan_per_wire = rd->mcasp->tdm_slots < ci->max ? + rd->mcasp->tdm_slots : ci->max; + unsigned int list[ci->max - ci->min + 1]; + int c1, c, count = 0; + + for (c1 = ci->min; c1 <= max_chan_per_wire; c1++) { + uint bclk_freq = c1*sbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { + /* If we can use all tdm_slots, we can put any + amount of channels to remaining wires as + long as they fit in. */ + if (c1 == rd->mcasp->tdm_slots) { + for (c = c1; c <= rd->serializers*c1 && + c <= ci->max; c++) + list[count++] = c; + } else { + list[count++] = c1; + } + } + } + dev_dbg(rd->mcasp->dev, + "%d possible channel counts for %d Hz and %d sbits\n", + count, rate, sbits); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + static int davinci_mcasp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { @@ -999,6 +1137,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->serial_dir[i] == dir) max_channels++; } + mcasp->ruledata[dir].serializers = max_channels; max_channels *= mcasp->tdm_slots; /* * If the already active stream has less channels than the calculated @@ -1013,6 +1152,42 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, max_channels); + + /* + * If we rely on implicit BCLK divider setting we should + * set constraints based on what we can provide. + */ + if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { + int ret; + + mcasp->ruledata[dir].mcasp = mcasp; + + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + davinci_mcasp_hw_rule_rate, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + davinci_mcasp_hw_rule_samplebits, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + davinci_mcasp_hw_rule_channels, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); + if (ret) + return ret; + } + return 0; } -- 1.9.1

1 1

[alsa-devel] [PATCH] ASoC: Intel: remove misleading DMA error messages on Baytrail platforms
by Pierre-Louis Bossart 19 Mar '15

19 Mar '15

During probe, the Baytrail audio driver reports errors such as: [44.172040] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [44.172137] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed Those error messages are misleading, there is no error since the DMA is explicitly not configured for Baytrail. Add a test to remove DMA error checks when DMA is not configured and return silently. Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/sst-acpi.c | 1 + sound/soc/intel/sst-dsp.h | 1 + sound/soc/intel/sst-firmware.c | 4 ++++ 3 files changed, 6 insertions(+) diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index b3d8456..42f293f 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -142,6 +142,7 @@ static int sst_acpi_probe(struct platform_device *pdev) sst_acpi->desc = desc; sst_acpi->mach = mach; + sst_pdata->resindex_dma_base = desc->resindex_dma_base; if (desc->resindex_dma_base >= 0) { sst_pdata->dma_engine = desc->dma_engine; sst_pdata->dma_base = desc->resindex_dma_base; diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index f291e32..148d8c5 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -206,6 +206,7 @@ struct sst_pdata { const struct firmware *fw; /* DMA */ + int resindex_dma_base; /* other fields invalid if equals to -1 */ u32 dma_base; u32 dma_size; int dma_engine; diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 5f71ef6..4a993d1 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -271,6 +271,10 @@ int sst_dma_new(struct sst_dsp *sst) const char *dma_dev_name; int ret = 0; + if (sst->pdata->resindex_dma_base == -1) + /* DMA is not used, return and squelsh error messages */ + return 0; + /* configure the correct platform data for whatever DMA engine * is attached to the ADSP IP. */ switch (sst->pdata->dma_engine) { -- 1.9.1

1 0

[alsa-devel] [PATCH] ALSA: jack: create jack kcontrols for every jack input device
by Jie Yang 19 Mar '15

19 Mar '15

From: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Currently the ALSA jack core registers only input devices for each jack registered. These jack input devices are not readable by userspace devices that run as non root. This patch adds support for additionally registering jack kcontrol devices for every input jack registered. This allows non root userspace to read jack status. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Modified-by: Jie Yang <yang.jie(a)intel.com> Signed-off-by: Jie Yang <yang.jie(a)intel.com> Reveiwed-by: Mark Brown <broonie(a)kernel.org> --- include/sound/control.h | 2 ++ include/sound/jack.h | 2 ++ sound/core/jack.c | 89 +++++++++++++++++++++++++++++++++++++++++++++++-- 3 files changed, 91 insertions(+), 2 deletions(-) mode change 100644 => 100755 sound/core/jack.c diff --git a/include/sound/control.h b/include/sound/control.h index 75f3054..023b70a 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -66,6 +66,7 @@ struct snd_kcontrol_volatile { struct snd_kcontrol { struct list_head list; /* list of controls */ + struct list_head jack_list; /* list of controls belong to the same jack*/ struct snd_ctl_elem_id id; unsigned int count; /* count of same elements */ snd_kcontrol_info_t *info; @@ -75,6 +76,7 @@ struct snd_kcontrol { snd_kcontrol_tlv_rw_t *c; const unsigned int *p; } tlv; + unsigned int jack_bit_idx; /*the corresponding jack type bit index */ unsigned long private_value; void *private_data; void (*private_free)(struct snd_kcontrol *kcontrol); diff --git a/include/sound/jack.h b/include/sound/jack.h index 2182350..477dcb4 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -73,6 +73,8 @@ enum snd_jack_types { struct snd_jack { struct input_dev *input_dev; + struct list_head kctl_list; + struct snd_card *card; int registered; int type; const char *id; diff --git a/sound/core/jack.c b/sound/core/jack.c old mode 100644 new mode 100755 index 8658578..c067b6a --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -24,6 +24,7 @@ #include <linux/module.h> #include <sound/jack.h> #include <sound/core.h> +#include <sound/control.h> static int jack_switch_types[SND_JACK_SWITCH_TYPES] = { SW_HEADPHONE_INSERT, @@ -54,7 +55,13 @@ static int snd_jack_dev_disconnect(struct snd_device *device) static int snd_jack_dev_free(struct snd_device *device) { struct snd_jack *jack = device->device_data; + struct snd_card *card = device->card; + struct snd_kcontrol *kctl; + list_for_each_entry(kctl, &jack->kctl_list, jack_list) { + list_del(&kctl->jack_list); + snd_ctl_remove(card, kctl); + } if (jack->private_free) jack->private_free(jack); @@ -100,6 +107,63 @@ static int snd_jack_dev_register(struct snd_device *device) return err; } + +/* get the first unused/available index number for the given kctl name */ +static int get_available_index(struct snd_card *card, const char *name) +{ + struct snd_kcontrol *kctl; + int idx = 0; + int len = strlen(name); + + down_write(&card->controls_rwsem); + next: + list_for_each_entry(kctl, &card->controls, list) { + if (!strncmp(name, kctl->id.name, len) && + !strcmp(" Jack", kctl->id.name + len) && + kctl->id.index == idx) { + idx++; + goto next; + } + } + up_write(&card->controls_rwsem); + return idx; +} + +static void kctl_private_free(struct snd_kcontrol *kctl) +{ + list_del(&kctl->jack_list); +} + +static int snd_jack_new_kctl(struct snd_card *card, struct snd_jack *jack, int type) +{ + struct snd_kcontrol *kctl; + int i, err, index, state = 0 /* use 0 for default state ?? */; + + INIT_LIST_HEAD(&jack->kctl_list); + for (i = 0; i < fls(SND_JACK_BTN_0); i++) { + int testbit = 1 << i; + if (type & testbit) { + index = get_available_index(card,jack->id); + kctl = snd_kctl_jack_new(jack->id, index, card); + if (!kctl) + return -ENOMEM; + kctl->private_free = kctl_private_free; + /* use jack_bit_idx for the kctl type bit */ + kctl->jack_bit_idx = i; + + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + list_add_tail(&kctl->jack_list, &jack->kctl_list); + + /* set initial jack state */ + snd_kctl_jack_report(card, kctl, state & testbit); + } + } + + return 0; +} + /** * snd_jack_new - Create a new jack * @card: the card instance @@ -138,7 +202,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, } jack->input_dev->phys = "ALSA"; - + jack->card = card; jack->type = type; for (i = 0; i < SND_JACK_SWITCH_TYPES; i++) @@ -150,10 +214,17 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (err < 0) goto fail_input; + /* register jack kcontrols */ + err = snd_jack_new_kctl(card, jack, type); + if (err < 0) + goto fail_kctl; + *jjack = jack; return 0; +fail_kctl: + snd_device_free(card, jack); fail_input: input_free_device(jack->input_dev); kfree(jack->id); @@ -230,6 +301,7 @@ EXPORT_SYMBOL(snd_jack_set_key); */ void snd_jack_report(struct snd_jack *jack, int status) { + struct snd_kcontrol *kctl; int i; if (!jack) @@ -245,13 +317,26 @@ void snd_jack_report(struct snd_jack *jack, int status) for (i = 0; i < ARRAY_SIZE(jack_switch_types); i++) { int testbit = 1 << i; - if (jack->type & testbit) + if (jack->type & testbit) { input_report_switch(jack->input_dev, jack_switch_types[i], status & testbit); + } } input_sync(jack->input_dev); + + for (i = 0; i < fls(SND_JACK_BTN_0); i++) { + int testbit = 1 << i; + if (jack->type & testbit) { + list_for_each_entry(kctl, &jack->kctl_list, jack_list) { + if (kctl->jack_bit_idx == i) { + snd_kctl_jack_report(jack->card, kctl, + status & testbit); + } + } + } + } } EXPORT_SYMBOL(snd_jack_report); -- 1.9.1

5 8

Re: [alsa-devel] Patches to bind the SGTL5000 chip to AM33XX McASP
by Peter Ujfalusi 19 Mar '15

19 Mar '15

On 03/19/2015 07:07 AM, Greg Knight wrote: > Hi, Mark, Jaroslav, > > I've attached a couple patches to provide support for the Freescale > SGTL5000 chip when using the McASP on the AM3359. It is more correct to say that this is needed in a custom hardware which uses McASP and SGTL5000 codec. > This adds an optional hack, "clock-to-talk" to the McASP subsystem which > can start AHCLKX/AHCLKR prior to initializing the codec. This is needed > on the SGTL5000, which depends on AHCLKX for its master clock. SO is it so that the codec's MCLK is coming from McASP AHCLKX (or R) and this clock need to be present all the time? W/o the MCLK the registers are not accessible? > These patches are based off of 3.14.31-ti-r49. What is the process for > getting these merged? First off, please send the patches with git send-email so people can comment on them. This will not work in mainline kernel. The McASP is turned off when it is not in use (no audio activity) so the AHCLKX/R is not going to be there on the ball output. How do you configure the frequency of the AHCLKX/R, this hack only enables it, so you are going to have _some_ clock going out. You should be using set_clkdiv and set_sysclk calls from the machine driver to configure the AHCLKX (we do not have AHCLKR supported in the driver ATM). -- Péter

1 0

Re: [alsa-devel] Patches to bind the SGTL5000 chip to AM33XX McASP
by Mark Brown 19 Mar '15

19 Mar '15

On Thu, Mar 19, 2015 at 01:07:39AM -0400, Greg Knight wrote: > These patches are based off of 3.14.31-ti-r49. What is the process for > getting these merged? Please submit patches using the process in SubmittingPatches, you need to work against current development versions of the kernel rather than an old vendor kernel.

1 0

[alsa-devel] [PATCH v2] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used
by Jyri Sarha 19 Mar '15

19 Mar '15

Set rule constraints to allow only combinations of sample-rate, sample-bits, and channels that can be played/captured with reasonable sample-rate accuracy. The logic with tdm-slots and serializers (=i2s data wires) goes like this: The first wire will take all channels up to number of tdm-slots, before following wires (if any) are used. If the first wire is used fully, the remaining wires share the same clocks and the divider can be calculated for the first wire. Also, takes the number of tdm-slots into account when implicitly selecting the BLCK divider. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- I hope the logic with tdm-slots and serializers is now correct. I do not have HW to test multiple serializers case, so please review davinci_mcasp_hw_rule_channels() extra carefully. Best regards, Jyri sound/soc/davinci/davinci-mcasp.c | 195 ++++++++++++++++++++++++++++++++++++-- 1 file changed, 185 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index de3b155..3b2fb56 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -26,6 +26,7 @@ #include <linux/of.h> #include <linux/of_platform.h> #include <linux/of_device.h> +#include <linux/math64.h> #include <sound/asoundef.h> #include <sound/core.h> @@ -64,6 +65,11 @@ struct davinci_mcasp_context { u32 *xrsr_regs; /* for serializer configuration */ }; +struct davinci_mcasp_ruledata { + struct davinci_mcasp *mcasp; + int serializers; +}; + struct davinci_mcasp { struct davinci_pcm_dma_params dma_params[2]; struct snd_dmaengine_dai_dma_data dma_data[2]; @@ -98,6 +104,8 @@ struct davinci_mcasp { #ifdef CONFIG_PM_SLEEP struct davinci_mcasp_context context; #endif + + struct davinci_mcasp_ruledata ruledata[2]; }; static inline void mcasp_set_bits(struct davinci_mcasp *mcasp, u32 offset, @@ -855,6 +863,30 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, return 0; } +static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, + unsigned int bclk_freq, + int *error_ppm) +{ + int div = mcasp->sysclk_freq / bclk_freq; + int rem = mcasp->sysclk_freq % bclk_freq; + + if (rem != 0) { + if (div == 0 || + ((mcasp->sysclk_freq / div) - bclk_freq) > + (bclk_freq - (mcasp->sysclk_freq / (div+1)))) { + div++; + rem = rem - bclk_freq; + } + } + if (error_ppm) + *error_ppm = + (div*1000000 + (int)div64_long(1000000LL*rem, + (int)bclk_freq)) + /div - 1000000; + + return div; +} + static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -872,16 +904,20 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, * the machine driver, we need to calculate the ratio. */ if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { - unsigned int bclk_freq = snd_soc_params_to_bclk(params); - unsigned int div = mcasp->sysclk_freq / bclk_freq; - if (mcasp->sysclk_freq % bclk_freq != 0) { - if (((mcasp->sysclk_freq / div) - bclk_freq) > - (bclk_freq - (mcasp->sysclk_freq / (div+1)))) - div++; - dev_warn(mcasp->dev, - "Inaccurate BCLK: %u Hz / %u != %u Hz\n", - mcasp->sysclk_freq, div, bclk_freq); - } + int channels = params_channels(params); + int rate = params_rate(params); + int sbits = params_width(params); + int ppm, div; + + if (channels > mcasp->tdm_slots) + channels = mcasp->tdm_slots; + + div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*channels, + &ppm); + if (ppm) + dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", + ppm); + __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); } @@ -973,6 +1009,108 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, return ret; } +static const unsigned int davinci_mcasp_dai_rates[] = { + 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, + 88200, 96000, 176400, 192000, +}; + +#define DAVINCI_MAX_RATE_ERROR_PPM 1000 + +static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + int sbits = params_width(params); + int channels = params_channels(params); + unsigned int list[ARRAY_SIZE(davinci_mcasp_dai_rates)]; + int i, count = 0; + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (i = 0; i < ARRAY_SIZE(davinci_mcasp_dai_rates); i++) { + uint bclk_freq = sbits*channels*davinci_mcasp_dai_rates[i]; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = davinci_mcasp_dai_rates[i]; + + } + dev_dbg(rd->mcasp->dev, "%d frequencies for %d sbits and %d channels\n", + count, sbits, channels); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_samplebits(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + int channels = params_channels(params); + int rate = params_rate(params); + unsigned int list[4]; + int sbits, count = 0; + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (sbits = 8; sbits <= 32; sbits += 8) { + uint bclk_freq = channels*sbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = sbits; + } + dev_dbg(rd->mcasp->dev, + "%d possible sample size for %d Hz and %d channels\n", + count, rate, channels); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_interval *ci = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + int sbits = params_width(params); + int rate = params_rate(params); + int max_chan_per_wire = rd->mcasp->tdm_slots < ci->max ? + rd->mcasp->tdm_slots : ci->max; + unsigned int list[ci->max - ci->min + 1]; + int c1, c, count = 0; + + for (c1 = ci->min; c1 <= max_chan_per_wire; c1++) { + uint bclk_freq = c1*sbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { + /* If we can use all tdm_slots, we can put any + amount of channels to remaining wires as + long as they fit in. */ + if (c1 == rd->mcasp->tdm_slots) { + for (c = c1; c <= rd->serializers*c1 && + c <= ci->max; c++) + list[count++] = c; + } else { + list[count++] = c1; + } + } + } + dev_dbg(rd->mcasp->dev, + "%d possible channel counts for %d Hz and %d sbits\n", + count, rate, sbits); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + static int davinci_mcasp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { @@ -998,6 +1136,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->serial_dir[i] == dir) max_channels++; } + mcasp->ruledata[dir].serializers = max_channels; max_channels *= mcasp->tdm_slots; /* * If the already active stream has less channels than the calculated @@ -1012,6 +1151,42 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, max_channels); + + /* + * If we rely on implicit BCLK divider setting we should + * set constraints based on what we can provide. + */ + if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { + int ret; + + mcasp->ruledata[dir].mcasp = mcasp; + + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + davinci_mcasp_hw_rule_rate, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + davinci_mcasp_hw_rule_samplebits, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + davinci_mcasp_hw_rule_channels, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); + if (ret) + return ret; + } + return 0; } -- 1.9.1

3 2

[alsa-devel] snd_usb_audio - siberia elite prism soundcard issues
by Daniel 19 Mar '15

19 Mar '15

Hello, First of all, since this is my first message I hope am not mistaken reporting this two issues about this piece of hardware (I was pointed here by alsa irc support channel) and bug tracker url does not seem to work, so I decided to post it here. Problem: Errors with mixer controls, breaking alsamixer, and not working 7.1 support. Scenario: Siberia Elite Prism soundcard supports 7.1 channels but it seems it can only play FRONT LEFT and FRONT RIGHT channels, no matter the setup. Here is the alsa-info script output: http://www.alsa-project.org/db/?f=301e6045129665d086d2571652c62648de0eac99 Additionally: aplay -L output: front:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio Front speakers surround40:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio IEC958 (S/PDIF) Digital Audio Output dmix:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio Direct sample mixing device dsnoop:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio Direct sample snooping device hw:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio Direct hardware device without any conversions plughw:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio Hardware device with all software conversions and amixer output for this device: # for i in {1..32}; do amixer -cElite cget numid=$i; done numid=1,iface=PCM,name='Capture Channel Map' ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | | chmap-fixed=FL,FR numid=2,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | | chmap-fixed=FL,FR numid=3,iface=MIXER,name='Sidetone Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=4,iface=MIXER,name='Sidetone Playback Volume' ; type=INTEGER,access=rw---R--,values=1,min=0,max=56,step=0 : values=0 | dBminmax-min=-74.00dB,max=-18.00dB numid=5,iface=MIXER,name='PCM Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=6,iface=MIXER,name='PCM Volume',index=1 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=7,iface=MIXER,name='PCM Volume',index=2 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=8,iface=MIXER,name='PCM Volume',index=3 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=9,iface=MIXER,name='PCM Volume',index=4 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=10,iface=MIXER,name='Speaker Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=11,iface=MIXER,name='Speaker Playback Volume' ; type=INTEGER,access=rw---R--,values=8,min=0,max=74,step=0 : values=54,54,54,54,54,54,54,54 | dBminmax-min=-74.00dB,max=0.00dB numid=12,iface=MIXER,name='Mic Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=13,iface=MIXER,name='Mic Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=74,step=0 : values=73,73 | dBminmax-min=-74.00dB,max=0.00dB amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 And hw-params from aplay: aplay --dump-hw-params -d1 -Dhw:Elite /dev/zero Playing raw data '/dev/zero' : Unsigned 8 bit, Rate 8000 Hz, Mono HW Params of device "hw:Elite": -------------------- ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S16_LE S24_3LE SUBFORMAT: STD SAMPLE_BITS: [16 24] FRAME_BITS: [32 192] CHANNELS: [2 8] RATE: [8000 96000] PERIOD_TIME: [1000 16384000] PERIOD_SIZE: [8 131072] PERIOD_BYTES: [64 524288] PERIODS: [2 1024] BUFFER_TIME: (166 32768000] BUFFER_SIZE: [16 262144] BUFFER_BYTES: [64 1048576] TICK_TIME: ALL -------------------- aplay: set_params:1233: Sample format non available Available formats: - S16_LE - S24_3LE Also in syslog I often can see messages related to this device: usb 1-1.3: 10:0: cannot get min/max values for control 3 (id 10) usb 1-1.3: 2:1: cannot get freq at ep 0x1 Loading alsamixer and trying to select the soundcard exits the program with this message: cannot load mixer controls: Invalid argument If anyone wants to check this issue, should you need any type of info regarding this, please do not hesitate asking, I will post anything you require. PS: I don't think it is really necessary to say either but I don't expect anyone to force-commit to this either, I mainly want to know if there is a solution to this or if reporting this issue I can help to get it fixed. Thanks for your time. -- *Daniel Ferradal*

3 5

[alsa-devel] [PATCH] ASoC: Intel: acpi_probe: fix error return path
by Vinod Koul 19 Mar '15

19 Mar '15

Fix the sst_acpi_probe memory allocation error path by setting right error code and initiating the cleanup insteadof just returning Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/intel/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index b782dfdcdbba..b536ddd2517d 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -309,7 +309,7 @@ static int sst_acpi_probe(struct platform_device *pdev) ctx->shim_regs64 = devm_kzalloc(ctx->dev, sizeof(*ctx->shim_regs64), GFP_KERNEL); if (!ctx->shim_regs64) { - return -ENOMEM; + ret = -ENOMEM; goto do_sst_cleanup; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] Intel: ASoC: Add condition check before set param to waves
by han.lu＠intel.com 19 Mar '15

19 Mar '15

From: "Lu, Han" <han.lu(a)intel.com> Check waves state before set parameter through ipc to prevent unexpected operation. Also remove redundant check. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 43fb5f3..20b629a 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -2055,6 +2055,11 @@ int sst_hsw_launch_param_buf(struct sst_hsw *hsw) { int ret, idx; + if (!sst_hsw_is_module_active(hsw, SST_HSW_MODULE_WAVES)) { + dev_dbg(hsw->dev, "module waves is not active\n"); + return 0; + } + /* put all param lines to DSP through ipc */ for (idx = 0; idx < hsw->param_idx_w; idx++) { ret = sst_hsw_module_set_param(hsw, diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index b40ec74..6c6229a 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -399,13 +399,9 @@ static int hsw_waves_param_put(struct snd_kcontrol *kcontrol, if (ret < 0) return ret; - if (sst_hsw_is_module_loaded(hsw, id)) { - if (!sst_hsw_is_module_active(hsw, id)) - return 0; - + if (sst_hsw_is_module_active(hsw, id)) ret = sst_hsw_module_set_param(hsw, id, 0, param_id, param_size, ucontrol->value.bytes.data); - } return ret; } -- 1.9.1

2 1

[alsa-devel] [PATCH 0/12 v2] ASoC: Renesas adds DPCM based sampling rate convert
by Kuninori Morimoto 19 Mar '15

19 Mar '15

Hi Mark These are v2 of DPCM based sampling rate convert support patch-set for Rensas sound driver. Then, it doesn't use simple-card. it uses new Renesas specific sound card driver, but it is copied from simple-card. 1 - 2 : prepare patch for Renesas new sound card driver 3 - 4 : new Renesas sound card driver 5 : tidyup for ak4642 6 - 10 : fixup/tidyup patches for rsnd 11 - 12 : DPCM support patch for rsnd Kuninori Morimoto (12): 1) ASoC: core: remove of_xlate_dai_name from snd_soc_component_driver 2) ASoC: core: add snd_soc_get_dai_info() to get driver infomation 3) ASoC: rsnd-dpcm-card: add Renesas sound card support for DPCM 4) ASoC: rsnd-dpcm-card: add .be_hw_params_fixup support for convert rate 5) ASoC: ak4642: tidyup DAPM route for playback 6) ASoC: rsnd: no more SSI restart when unusual situation 7) ASoC: rsnd: no more SRC restart when unusual situation 8) ASoC: rsnd: tidyup error message format 9) ASoC: rsnd: show debug info for sampling rate convert 10) ASoC: rsnd: don't call clk_prepare_enable/unprepare() from inside spin lock 11) ASoC: rsnd: add dai_link stream name 12) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsnd-dpcm-card.txt | 72 ++++++++++++++ include/sound/soc.h | 6 +- sound/soc/codecs/ak4642.c | 4 +- sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 + sound/soc/sh/rcar/core.c | 48 ++++++++- sound/soc/sh/rcar/dvc.c | 12 +-- sound/soc/sh/rcar/rsnd-dpcm-card.c | 496 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 16 ++- sound/soc/sh/rcar/src.c | 63 ++++++++++-- sound/soc/sh/rcar/ssi.c | 31 ++++-- sound/soc/soc-core.c | 71 ++++++++------ 13 files changed, 765 insertions(+), 66 deletions(-) Best regards --- Kuninori Morimoto _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

3 17

[alsa-devel] bug in alsa-plugins: #if SND_PCM_EXTPLUG_VERSION wrong
by Ethan Grammatikidis 18 Mar '15

18 Mar '15

in alsa-plugins 1.0.28 and 1.0.29, mix/*.c contain the following line: #if SND_PCM_EXTPLUG_VERSION >= 0x102 that 0x102 needs to be 0x10002. i have alsa-lib 1.0.26 installed. its headers define SND_PCM_EXTPLUG_VERSION to 0x10001, corresponding to version number 1.0.1. that #if effectively tests for >= 0.1.2, so attempting to build alsa-plugins 1.0.28 or 1.0.29 results in a long list of errors. SND_CHMAP_MONO, SND_CHMAP_FL, etc are undeclared and snd_pcm_chmap_query_t is an unknown type. in alsa-plugins 1.0.26 SND_PCM_EXTPLUG_VERSION is not tested. i didn't look at 1.0.27 reporting here because the bug tracker is down. -- Wir mussen wissen, wir werden wissen.

2 1

[alsa-devel] [PATCH 1/6] ESI W192M : Correct copy/paste from prodigy driver
by klem.dev＠gmail.com 18 Mar '15

18 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Correct copy/paste name from prodigy driver, no behaviour change, only name. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index bcf30a3..f65ac19 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -463,7 +463,7 @@ static int wtm_add_controls(struct snd_ice1712 *ice) static int wtm_init(struct snd_ice1712 *ice) { - static unsigned short stac_inits_prodigy[] = { + static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, (unsigned short)-1 }; @@ -475,7 +475,7 @@ static int wtm_init(struct snd_ice1712 *ice) ice->force_rdma1 = 1; /*initialize codec*/ - p = stac_inits_prodigy; + p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); -- 2.1.4

3 2

[alsa-devel] [PATCH 1/2] ASoC: Intel: Remove support for Intel MID DMA from firmware loader
by Jarkko Nikula 18 Mar '15

18 Mar '15

Intel MID DMA driver is going to be removed by the coming commit 36111da7838e ("dmaengine: intel-mid-dma: remove the driver") in spi.git tree. Since there are no users for SST_DMA_TYPE_MID type the support for it can be removed from here in advance. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-dsp.h | 1 - sound/soc/intel/sst-firmware.c | 3 --- 2 files changed, 4 deletions(-) diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index f291e32f0077..3412474083ff 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -28,7 +28,6 @@ /* Supported SST DMA Devices */ #define SST_DMA_TYPE_DW 1 -#define SST_DMA_TYPE_MID 2 /* autosuspend delay 5s*/ #define SST_RUNTIME_SUSPEND_DELAY (5 * 1000) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 5e5800897da2..38881f1fb990 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -277,9 +277,6 @@ int sst_dma_new(struct sst_dsp *sst) case SST_DMA_TYPE_DW: dma_dev_name = "dw_dmac"; break; - case SST_DMA_TYPE_MID: - dma_dev_name = "Intel MID DMA"; - break; default: dev_err(sst->dev, "error: invalid DMA engine %d\n", sst->pdata->dma_engine); -- 2.1.4

3 6

[alsa-devel] [PATCH] ASoC: samsung: Enable SND_SIMPLE_CARD for Odroid X2/U3
by Sylwester Nawrocki 18 Mar '15

18 Mar '15

Odroid X2/U3 sound support can now be specified in device tree using the simple card binding, make SND_SOC_ODROIDX2 select SND_SIMPLE_CARD to ensure there are always required drivers in place. Reported-by: Tobias Jakobi <liquid.acid(a)gmx.net> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 0632a36..25caadf 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -226,6 +226,7 @@ config SND_SOC_ODROIDX2 depends on SND_SOC_SAMSUNG && I2C select SND_SOC_MAX98090 select SND_SAMSUNG_I2S + select SND_SIMPLE_CARD help Say Y here to enable audio support for the Odroid-X2/U3. -- 1.7.9.5

2 1

Re: [alsa-devel] ASoC: Intel: mrfld- add ACPI module
by Dan Carpenter 18 Mar '15

18 Mar '15

Hello Vinod Koul, The patch 336cfbb05edf: "ASoC: Intel: mrfld- add ACPI module" from Nov 11, 2014, leads to the following static checker warning: sound/soc/intel/sst/sst_acpi.c:313 sst_acpi_probe() info: ignoring unreachable code. sound/soc/intel/sst/sst_acpi.c 304 ret = sst_context_init(ctx); 305 if (ret < 0) 306 return ret; 307 308 /* need to save shim registers in BYT */ 309 ctx->shim_regs64 = devm_kzalloc(ctx->dev, sizeof(*ctx->shim_regs64), 310 GFP_KERNEL); 311 if (!ctx->shim_regs64) { 312 return -ENOMEM; ^^^^^^^^^^^^^^^ 313 goto do_sst_cleanup; ^^^^^^^^^^^^^^^^^^^^ 314 } 315 316 sst_configure_runtime_pm(ctx); 317 platform_set_drvdata(pdev, ctx); 318 return ret; 319 320 do_sst_cleanup: 321 sst_context_cleanup(ctx); 322 platform_set_drvdata(pdev, NULL); 323 dev_err(ctx->dev, "failed with %d\n", ret); 324 return ret; 325 } regards, dan carpenter

2 2

[alsa-devel] [GIT PULL] ASoC fixes for v4.0
by Mark Brown 18 Mar '15

18 Mar '15

The following changes since commit 06e5801b8cb3fc057d88cb4dc03c0b64b2744cda: Linux 4.0-rc4 (2015-03-15 17:38:20 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.0-rc4 for you to fetch changes up to 33484c67398295108ab31e17b50b5a6fba1f5eef: Merge remote-tracking branches 'asoc/fix/sgtl5000' and 'asoc/fix/sn95031' into asoc-linus (2015-03-16 12:03:17 +0000) ---------------------------------------------------------------- ASoC: Fixes for v4.0 As well as the usual collection of driver specific fixes there's a few more generic things: - Lots of fixes from Takashi for drivers using the wrong field in the control union to communicate with userspace, leading to potential errors on 64 bit systems. - A fix from Lars for locking of the lists of devices we maintain, mostly only likely to trigger during device probe and removal. ---------------------------------------------------------------- Bard Liao (1): ASoC: rt286: Change the DMI mapping for Dino Eric Nelson (1): ASoC: sgtl5000: remove useless register write clearing CHRGPUMP_POWERUP Lars-Peter Clausen (4): ASoC: ak4671: Fix control-less DAPM routes ASoC: da732x: Fix control-less DAPM routes ASoC: sn95031: Fix control-less DAPM routes ASoC: Fix component lists locking Lu, Han (1): ASoC: Intel: remove conflicts when load/unload multiple firmware images Maciej S. Szmigiero (1): fsl_ssi: fix of_property_read_u32_array return value check Mark Brown (3): Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branches 'asoc/fix/ak4671', 'asoc/fix/control', 'asoc/fix/da732x', 'asoc/fix/fsl-ssi', 'asoc/fix/lock' and 'asoc/fix/rt286' into asoc-linus Merge remote-tracking branches 'asoc/fix/sgtl5000' and 'asoc/fix/sn95031' into asoc-linus Nicolin Chen (1): ASoC: fsl_ssi: Don't try to round-up for PM divisor calculation Takashi Iwai (14): ASoC: adav80x: Fix wrong value references for boolean kctl ASoC: ak4641: Fix wrong value references for boolean kctl ASoC: cs4271: Fix wrong value references for boolean kctl ASoC: es8238: Fix wrong value references for boolean kctl ASoC: pcm1681: Fix wrong value references for boolean kctl ASoC: tas5086: Fix wrong value references for boolean kctl ASoC: wm2000: Fix wrong value references for boolean kctl ASoC: wm8731: Fix wrong value references for boolean kctl ASoC: wm8903: Fix wrong value references for boolean kctl ASoC: wm8904: Fix wrong value references for boolean kctl ASoC: wm8955: Fix wrong value references for boolean kctl ASoC: wm8960: Fix wrong value references for boolean kctl ASoC: wm9712: Fix wrong value references for boolean kctl ASoC: wm9713: Fix wrong value references for boolean kctl sound/soc/codecs/adav80x.c | 4 ++-- sound/soc/codecs/ak4641.c | 4 ++-- sound/soc/codecs/ak4671.c | 44 +++++++++++++++++++-------------------- sound/soc/codecs/cs4271.c | 4 ++-- sound/soc/codecs/da732x.c | 8 +++---- sound/soc/codecs/es8328.c | 4 ++-- sound/soc/codecs/pcm1681.c | 4 ++-- sound/soc/codecs/rt286.c | 2 +- sound/soc/codecs/sgtl5000.c | 8 +------ sound/soc/codecs/sn95031.c | 4 ++-- sound/soc/codecs/tas5086.c | 4 ++-- sound/soc/codecs/wm2000.c | 8 +++---- sound/soc/codecs/wm8731.c | 4 ++-- sound/soc/codecs/wm8903.c | 4 ++-- sound/soc/codecs/wm8904.c | 4 ++-- sound/soc/codecs/wm8955.c | 4 ++-- sound/soc/codecs/wm8960.c | 4 ++-- sound/soc/codecs/wm9712.c | 6 +++--- sound/soc/codecs/wm9713.c | 6 +++--- sound/soc/fsl/fsl_ssi.c | 4 ++-- sound/soc/intel/sst-haswell-dsp.c | 3 --- sound/soc/intel/sst-haswell-ipc.c | 32 +++++++++++++++++++++------- sound/soc/soc-core.c | 41 ++++++++++++++++++++++++++---------- 23 files changed, 118 insertions(+), 92 deletions(-)

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used
by Jyri Sarha 18 Mar '15

18 Mar '15

Set a rule constraints to allow only combinations of sample-rate and number of frame-bits that can be played/captured with a reasonable sample-rate accuracy. Also, takes the number of serializers into account when implicitly selecting the BLCK divider. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Now this works as it should. Thanks, Jyri sound/soc/davinci/davinci-mcasp.c | 150 +++++++++++++++++++++++++++++++++++--- 1 file changed, 140 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index de3b155..3ecee3b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -26,6 +26,7 @@ #include <linux/of.h> #include <linux/of_platform.h> #include <linux/of_device.h> +#include <linux/math64.h> #include <sound/asoundef.h> #include <sound/core.h> @@ -64,6 +65,11 @@ struct davinci_mcasp_context { u32 *xrsr_regs; /* for serializer configuration */ }; +struct davinci_mcasp_ruledata { + struct davinci_mcasp *mcasp; + int serializers; +}; + struct davinci_mcasp { struct davinci_pcm_dma_params dma_params[2]; struct snd_dmaengine_dai_dma_data dma_data[2]; @@ -98,6 +104,8 @@ struct davinci_mcasp { #ifdef CONFIG_PM_SLEEP struct davinci_mcasp_context context; #endif + + struct davinci_mcasp_ruledata ruledata[2]; }; static inline void mcasp_set_bits(struct davinci_mcasp *mcasp, u32 offset, @@ -855,6 +863,30 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, return 0; } +static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, + unsigned int bclk_freq, + int *error_ppm) +{ + int div = mcasp->sysclk_freq / bclk_freq; + int rem = mcasp->sysclk_freq % bclk_freq; + + if (rem != 0) { + if (div == 0 || + ((mcasp->sysclk_freq / div) - bclk_freq) > + (bclk_freq - (mcasp->sysclk_freq / (div+1)))) { + div++; + rem = rem - bclk_freq; + } + } + if (error_ppm) + *error_ppm = + (div*1000000 + (int)div64_long(1000000LL*rem, + (int)bclk_freq)) + /div - 1000000; + + return div; +} + static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -872,16 +904,19 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, * the machine driver, we need to calculate the ratio. */ if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { - unsigned int bclk_freq = snd_soc_params_to_bclk(params); - unsigned int div = mcasp->sysclk_freq / bclk_freq; - if (mcasp->sysclk_freq % bclk_freq != 0) { - if (((mcasp->sysclk_freq / div) - bclk_freq) > - (bclk_freq - (mcasp->sysclk_freq / (div+1)))) - div++; - dev_warn(mcasp->dev, - "Inaccurate BCLK: %u Hz / %u != %u Hz\n", - mcasp->sysclk_freq, div, bclk_freq); - } + uint bclk_freq = snd_soc_params_to_bclk(params); + int dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + TX_MODE : RX_MODE; + int ppm, div; + + bclk_freq /= mcasp->ruledata[dir].serializers; + + div = davinci_mcasp_calc_clk_div(mcasp, bclk_freq, &ppm); + + if (ppm) + dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", + ppm); + __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); } @@ -973,6 +1008,74 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, return ret; } +static const unsigned int davinci_mcasp_dai_rates[] = { + 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, + 88200, 96000, 176400, 192000, +}; + +#define DAVINCI_MAX_RATE_ERROR_PPM 1000 + +static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + unsigned int list[ARRAY_SIZE(davinci_mcasp_dai_rates)]; + int frame_size; + int i, count = 0; + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) + return frame_size; + + for (i = 0; i < ARRAY_SIZE(davinci_mcasp_dai_rates); i++) { + uint bclk_freq = frame_size*davinci_mcasp_dai_rates[i]/ + rd->serializers; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = davinci_mcasp_dai_rates[i]; + + } + dev_dbg(rd->mcasp->dev, "%d frequencies for %u fsize (%d)\n", + count, frame_size, rd->serializers); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_framebits(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_interval *ci = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + unsigned int max_chan_per_wire = ci->max / rd->serializers; + unsigned int list[max_chan_per_wire*4]; + int rate = params_rate(params); + int chans, fbits, count = 0; + + for (chans = ci->min; chans <= max_chan_per_wire; chans++) { + for (fbits = 8; fbits <= 32; fbits += 8) { + uint bclk_freq = chans*fbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { + list[count++] = chans*fbits*rd->serializers; + BUG_ON(count > ARRAY_SIZE(list)); + } + } + } + + dev_dbg(rd->mcasp->dev, + "%d possible frame size for %d Hz (%d)\n", + count, rate, rd->serializers); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + static int davinci_mcasp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { @@ -998,6 +1101,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->serial_dir[i] == dir) max_channels++; } + mcasp->ruledata[dir].serializers = max_channels; max_channels *= mcasp->tdm_slots; /* * If the already active stream has less channels than the calculated @@ -1012,6 +1116,32 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, max_channels); + + /* + * If we rely on implicit BCLK divider setting we should + * set constraints based on what we can provide. + */ + if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { + int ret; + + mcasp->ruledata[dir].mcasp = mcasp; + + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + davinci_mcasp_hw_rule_rate, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + davinci_mcasp_hw_rule_framebits, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, -1); + if (ret) + return ret; + } + return 0; } -- 1.9.1

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[alsa-devel] Support two wm8962 codec in i.mx6 board.
by Shawn Embedded 18 Mar '15

18 Mar '15

Hello everyone, I'm new to ALSA. I have a question here. Now there is a wm8962 on the i.mx6 board, running well, and we want to add an extra wm8962 to the board, connected to i.mx6 on another SSI port through a different i.mx6's AUDMUX port. The two wm8962 will work independently completely, at the same time possiblly. Before sending this mail to the list, I've already googled some materials about my requirements. I've found there is a structure named "struct snd_soc_aux_dev". For my applicantion, if I want to support another wm8962, could I can use this structure for my intention please? But in this link http://www.spinics.net/lists/alsa-devel/msg40127.html , it's said that This makes possible to register auxiliary dailess codecs in a machine driver. Term dailess is used here for amplifiers and codecs without DAI or DAI being unused. then, does it mean that I couldn't use this structure for my application? Or, I should declare another structure "struct snd_soc_card" for the second wm8962, just as the now-working wm8962? or other ways? In fact, I'm a little confused if I could declare two structure snd_soc_card at the same? If it's OK, what should I do next to meet my requirement? Could anyone give me any guidance on this? Any links, and materials will also be much appreciated. Thank you in advance. ------------------- Best Regards, Shawn

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[alsa-devel] [PATCH 6/6] ESI W192M : Add sampling rate control of the ADC/DAC
by klem.dev＠gmail.com 18 Mar '15

18 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add sampling rate control for ADC/DAC for ESI W192M. Allow to switch between 48K/96K/192K sampling rate. All DAC need to be mute when changing samplerate. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 6d3412f..9906119 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -30,12 +30,18 @@ #include <linux/init.h> #include <sound/core.h> #include <sound/tlv.h> +#include <linux/slab.h> #include "ice1712.h" #include "envy24ht.h" #include "wtm.h" #include "stac946x.h" +struct wtm_spec { + /* rate change needs atomic mute/unmute of all dacs*/ + struct mutex mute_mutex; +}; + /* * 2*ADC 6*DAC no1 ringbuffer r/w on i2c bus @@ -69,15 +75,65 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg) /* * DAC mute control */ +static void stac9460_dac_mute_all(struct snd_ice1712 *ice, unsigned char mute, + unsigned short int *change_mask) +{ + unsigned char new, old; + int id, idx, change; + + /*stac9460 1*/ + for (id = 0; id < 7; id++) { + if (*change_mask & (0x01 << id)) { + if (id == 0) + idx = STAC946X_MASTER_VOLUME; + else + idx = STAC946X_LF_VOLUME - 1 + id; + old = stac9460_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) { + stac9460_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + } else { + *change_mask = *change_mask & ~(0x01 << id); + } + } + } + + /*stac9460 2*/ + for (id = 0; id < 3; id++) { + if (*change_mask & (0x01 << (id + 7))) { + if (id == 0) + idx = STAC946X_MASTER_VOLUME; + else + idx = STAC946X_LF_VOLUME - 1 + id; + old = stac9460_2_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) { + stac9460_2_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + } else { + *change_mask = *change_mask & ~(0x01 << id); + } + } + } +} + + + #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct wtm_spec *spec = ice->spec; unsigned char val; int idx, id; + mutex_lock(&spec->mute_mutex); + if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; id = 0; @@ -90,6 +146,8 @@ static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, else val = stac9460_2_get(ice, idx - 6); ucontrol->value.integer.value[0] = (~val >> 7) & 0x1; + + mutex_unlock(&spec->mute_mutex); return 0; } @@ -388,6 +446,44 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, } +/* + * Handler for setting correct codec rate - called when rate change is detected + */ +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned char old, new; + unsigned short int changed; + struct wtm_spec *spec = ice->spec; + + if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + return; + else if (rate <= 48000) + new = 0x08; /* 256x, base rate mode */ + else if (rate <= 96000) + new = 0x11; /* 256x, mid rate mode */ + else + new = 0x12; /* 128x, high rate mode */ + + old = stac9460_get(ice, STAC946X_MASTER_CLOCKING); + if (old == new) + return; + /* change detected, setting master clock, muting first */ + /* due to possible conflicts with mute controls - mutexing */ + mutex_lock(&spec->mute_mutex); + /* we have to remember current mute status for each DAC */ + changed = 0xFFFF; + stac9460_dac_mute_all(ice, 0, &changed); + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); + stac9460_2_put(ice, STAC946X_MASTER_CLOCKING, new); + udelay(10); + /* unmuting - only originally unmuted dacs - + * i.e. those changed when muting */ + stac9460_dac_mute_all(ice, 1, &changed); + mutex_unlock(&spec->mute_mutex); +} + + /*Limits value in dB for fader*/ static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); @@ -487,21 +583,32 @@ static int wtm_init(struct snd_ice1712 *ice) { static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, + STAC946X_MASTER_CLOCKING, 0x11, (unsigned short)-1 }; unsigned short *p; + struct wtm_spec *spec; /*WTM 192M*/ ice->num_total_dacs = 8; ice->num_total_adcs = 4; ice->force_rdma1 = 1; + /*init mutex for dac mute conflict*/ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + mutex_init(&spec->mute_mutex); + + /*initialize codec*/ p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); } + ice->gpio.set_pro_rate = stac9460_set_rate_val; return 0; } -- 2.1.4

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[alsa-devel] [PATCH 5/6] ESI W192M : Add text Line in/Mic for selecting input gain state
by klem.dev＠gmail.com 18 Mar '15

18 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add text Line in/Mic for selecting input gain state in mixer for ESI W192M. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index c7ffafa..6d3412f 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -339,8 +339,14 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, /* * MIC / LINE switch fonction */ +static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { "Line In", "Mic" }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} -#define stac9460_mic_sw_info snd_ctl_boolean_mono_info static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -354,7 +360,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - ucontrol->value.integer.value[0] = ~val>>7 & 0x1; + ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; } @@ -370,7 +376,7 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); + new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) @@ -411,7 +417,7 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "MIC/Line switch", + .name = "MIC/Line Input Enum", .count = 2, .info = stac9460_mic_sw_info, .get = stac9460_mic_sw_get, -- 2.1.4

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[alsa-devel] [PATCH 4/6] ESI W192M : Add TLV support for control value in dB scale
by klem.dev＠gmail.com 18 Mar '15

18 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add TLV support to control volume using dB scale for input and ouput on ESI W192M. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 59483b4..c7ffafa 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -29,6 +29,7 @@ #include <linux/interrupt.h> #include <linux/init.h> #include <sound/core.h> +#include <sound/tlv.h> #include "ice1712.h" #include "envy24ht.h" @@ -380,17 +381,25 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, return change; } + +/*Limits value in dB for fader*/ +static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); + /* * Control tabs */ static struct snd_kcontrol_new stac9640_controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Switch", .info = stac9460_dac_mute_info, .get = stac9460_dac_mute_get, .put = stac9460_dac_mute_put, - .private_value = 1 + .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -419,11 +428,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "DAC Volume", .count = 8, .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -435,12 +448,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "ADC Volume", .count = 2, .info = stac9460_adc_vol_info, .get = stac9460_adc_vol_get, .put = stac9460_adc_vol_put, - + .tlv = { .p = db_scale_adc } } }; -- 2.1.4

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[alsa-devel] [PATCH 3/6] ESI W192M : Enable midi i/o of port envy24 chip as available
by klem.dev＠gmail.com 18 Mar '15

18 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Enable midi i/o port of envy24 chip as their are available on ESI W192M soundcard. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 6e1026e..59483b4 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -485,7 +485,8 @@ static int wtm_init(struct snd_ice1712 *ice) static unsigned char wtm_eeprom[] = { - [ICE_EEP2_SYSCONF] = 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ + [ICE_EEP2_SYSCONF] = 0x67, /*SYSCONF: clock 192KHz, mpu401, + 4ADC, 8DAC */ [ICE_EEP2_ACLINK] = 0x80, /* ACLINK : I2S */ [ICE_EEP2_I2S] = 0xf8, /* I2S: vol; 96k, 24bit, 192k */ [ICE_EEP2_SPDIF] = 0xc1, /*SPDIF: out-en, spidf ext out*/ -- 2.1.4

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[alsa-devel] [PATCH 2/6] ESI W192M : Update eeprom structure to C99 standard
by klem.dev＠gmail.com 18 Mar '15

18 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Update eeprom structure to C99 standard to be compliant with change in alsa. It's just a notation change, no configuration change. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index f65ac19..6e1026e 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -485,19 +485,19 @@ static int wtm_init(struct snd_ice1712 *ice) static unsigned char wtm_eeprom[] = { - 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ - 0x80, /* ACLINK : I2S */ - 0xf8, /* I2S: vol; 96k, 24bit, 192k */ - 0xc1 /*SPDIF: out-en, spidf ext out*/, - 0x9f, /* GPIO_DIR */ - 0xff, /* GPIO_DIR1 */ - 0x7f, /* GPIO_DIR2 */ - 0x9f, /* GPIO_MASK */ - 0xff, /* GPIO_MASK1 */ - 0x7f, /* GPIO_MASK2 */ - 0x16, /* GPIO_STATE */ - 0x80, /* GPIO_STATE1 */ - 0x00, /* GPIO_STATE2 */ + [ICE_EEP2_SYSCONF] = 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ + [ICE_EEP2_ACLINK] = 0x80, /* ACLINK : I2S */ + [ICE_EEP2_I2S] = 0xf8, /* I2S: vol; 96k, 24bit, 192k */ + [ICE_EEP2_SPDIF] = 0xc1, /*SPDIF: out-en, spidf ext out*/ + [ICE_EEP2_GPIO_DIR] = 0x9f, + [ICE_EEP2_GPIO_DIR1] = 0xff, + [ICE_EEP2_GPIO_DIR2] = 0x7f, + [ICE_EEP2_GPIO_MASK] = 0x9f, + [ICE_EEP2_GPIO_MASK1] = 0xff, + [ICE_EEP2_GPIO_MASK2] = 0x7f, + [ICE_EEP2_GPIO_STATE] = 0x16, + [ICE_EEP2_GPIO_STATE1] = 0x80, + [ICE_EEP2_GPIO_STATE2] = 0x00, }; -- 2.1.4

1 0

[alsa-devel] [PATCH v9] ASoC: dapm: add code to configure dai link parameters
by Nikesh Oswal 17 Mar '15

17 Mar '15

dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 164 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 166 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 5c0a798..a2098e4 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -531,6 +532,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade1..4636a05 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -941,6 +941,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ededb97..893a6e5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1230,7 +1230,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1242,7 +1243,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index bc2d7a0..b0c5951 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -853,6 +853,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* create control for links with > 1 config */ + if (w->num_params <= 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* We implement power down on suspend by checking the power state of * the ALSA card - when we are suspending the ALSA state for the card * is set to D3. @@ -2724,6 +2754,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3198,7 +3231,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, { struct snd_soc_dapm_path *source_p, *sink_p; struct snd_soc_dai *source, *sink; - const struct snd_soc_pcm_stream *config = w->params; + const struct snd_soc_pcm_stream *config = w->params + w->params_select; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; u64 fmt; @@ -3290,8 +3323,39 @@ out: return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + /* Can't change the config when widget is already powered */ + if (w->power) + return -EBUSY; + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3299,14 +3363,61 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + if (!link_name) { + ret = -ENOMEM; + goto outfree_w_param; + } snprintf(link_name, len, "%s-%s", source->name, sink->name); + for (count = 0 ; count < num_params; count++) { + if (!config->stream_name) { + dev_warn(card->dapm.dev, + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); + len = strlen("Anonymous Configuration ") + 3; + w_param_text[count] = + devm_kzalloc(card->dev, len, GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + snprintf(w_param_text[count], len, + "Anonymous Configuration %d", count); + } else { + w_param_text[count] = devm_kmemdup(card->dev, + config->stream_name, + strlen(config->stream_name) + 1, + GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3314,6 +3425,30 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + /* duplicate w_param_enum on heap so that memory persists */ + private_value = + (unsigned long) devm_kmemdup(card->dev, + (void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + /* duplicate kcontrol_dai_link on heap so that memory persists */ + template.kcontrol_news = + devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3321,15 +3456,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + devm_kfree(card->dev, w); +outfree_kcontrol_news: + devm_kfree(card->dev, (void *)template.kcontrol_news); +outfree_private_value: + devm_kfree(card->dev, (void *)private_value); +outfree_link_name: + devm_kfree(card->dev, link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + devm_kfree(card->dev, (void *)w_param_text[count]); + devm_kfree(card->dev, w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, -- 1.7.9.5

2 1

[alsa-devel] pcm512x driver: Mixer control naming issue
by Howard Mitchell 17 Mar '15

17 Mar '15

Hi, As you may be aware the Raspbian distribution for the RaspberryPi has recently been upgraded to kernel version 3.18. It was previously at version 3.12. This has had the unfortunate side effect that users of the PCM5122 DAC have found that the output volume has been significantly reduced. The issue is related to this commit <https://git.kernel.org/cgit/linux/kernel/git/stable/linux-stable.git/commit…> which adds support for a couple of analogue gain controls. One control (Playback Volume) sets a gain of either 0db or -6dB and is intended to set the full scale output at 2Vrms or 1Vrms respectively. The other (Playback Boost Volume) provides a boost of +0.8dB. The hardware reset value of both of these gain controls is 0dB, however, in the Raspbian distribution 'Playback Volume' is being defaulted to -6dB. After some investigation I figured out that this was caused by the alsa restore mechanism. If '/var/lib/alsa/asound.state' doesn't exist or there is no saved state for a particular control then the defaults in '/usr/share/alsa/init/default' are used. In this file the default for a control named 'Playback Volume' is -20dB which is forcing the control to it's minimum value of -6dB. This seems like incorrect behaviour for something that is essentially an implementation detail, i.e. a choice that should be made by the board designer not the end user. Therefore I'd like to see what you think of the following suggestions (which I'd be more than happy to implement and submit patches for): * Either change the names of these controls to something that's not affected by the alsa restore mechanism, * or remove them as mixer controls, but make them accessible some other way - e.g. device tree. Howard Mitchell

3 9

[alsa-devel] [PATCH 0/2] ASoC: Intel: fixup patches for Braswell machine driver
by Jin Yao 17 Mar '15

17 Mar '15

Hi Mark, These are Braswell machine driver fixup patches. Jin Yao (2): 1) ASoC: Intel: Add suspend_pre and resume_post for Braswell snd_soc_card 2) ASoC: Intel: move the jack creation to Braswell machine driver For 1), it requires following dependency patches to build: ASoC: rt5670: Add IRQ function ASoC: rt5670: export jack suspend/resume APIs For 2), it requires following dependency patch to build: ASoC: Allow to register jacks at the card level These dependency patches have been integrated in for-next branch but not in intel branch. Thanks Jin Yao sound/soc/intel/cht_bsw_rt5672.c | 55 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 55 insertions(+) -- 1.9.1

2 3

[alsa-devel] [PATCH 1/2] ASoC: rt286: Drop unnecessary dapm bias_level initialization
by Lars-Peter Clausen 17 Mar '15

17 Mar '15

The default value for the bias_level is SND_SOC_BIAS_OFF when probe is being called, there is no need to initialize it explicitly. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/rt286.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index fb3f5d3..7bbe714 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1048,7 +1048,6 @@ static int rt286_probe(struct snd_soc_codec *codec) struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); rt286->codec = codec; - codec->dapm.bias_level = SND_SOC_BIAS_OFF; if (rt286->i2c->irq) { regmap_update_bits(rt286->regmap, -- 1.8.0

3 4

[alsa-devel] [PATCH 1/1] Intel: ASoC: remove duplicated code on merge
by han.lu＠intel.com 17 Mar '15

17 Mar '15

From: "Lu, Han" <han.lu(a)intel.com> Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index d4865a6..43fb5f3 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -2363,11 +2363,6 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) /* init param buffer */ sst_hsw_reset_param_buf(hsw); - /* allocate scratch mem regions */ - ret = sst_block_alloc_scratch(hsw->dsp); - if (ret < 0) - goto boot_err; - /* wait for DSP boot completion */ sst_dsp_boot(hsw->dsp); ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete, -- 1.9.1

3 2

[alsa-devel] HD-audio restructuring (and future changes)
by Takashi Iwai 16 Mar '15

16 Mar '15

Hi, I now merged the branches topic/hda-bus and topic/hda-unbind into for-next branch that is targeted for 4.1 kernel. They are the changes of HD-audio I posted in the last month, moving the HD-audio binding to standard bus and its relevant fixes. At this moment, there should be no big visible (nor audible) changes. It's merely an internal structure change. Now you see more sysfs entries and controls, but the basic functionality via API remains identical. Also, hda-emu was updated together with this merge. Let me know if you hit any issue with these updates. About the future plan: there are already more pending changes for HD-audio and I'd like to merge them for 4.1, too. Namely, many basic HD-audio codes are split to a smaller core library in sound/hda, so that we can build the upcoming ASoC HD-audio driver with a smaller subset of HD-audio codes. You can find the current code in topic/hda-core branch. In this branch, the bus and HD-audio codec core are moved into sound/hda. You may grasp an image from the main header file include/sound/hdaudio.h. The split and cleanup of controller driver code is in development, and some pieces are found in sound-unstable git tree as of now. The patches will be submitted once when all settled down. Meanwhile, there is another patchset I've set previously for the regmap support. This is found in topic/hda-regmap branch, which is built on top of topic/hda-core branch. Unless any serious problem is found, I'm going to merge this as well. Note that this needs some fixes in regmap side, so this branch might be rebased in future to merge the regmap fixes into it. One undecided thing is whether we keep codes in sound/pci/hda directory, or move everything into sound/hda, e.g. sound/hda/legacy/pci/hda_intel.c, sound/hda/legacy/codecs/*.c. The obvious drawback of moving files is that it makes backport difficult, so it's bad for stable kernels. As of now, I'm inclined to keep them as is now, but I'm open about this. Let me know if you have any suggestions. thanks, Takashi

1 0

[alsa-devel] [PATCH v2] ASoC: wm8994: Manage wm8994's MCLK in codec driver
by Inha Song 16 Mar '15

16 Mar '15

Previously, When we use wm8994 codec, we should control its MCLK in machine driver. But, It should be managed by wm8994 codec driver, not machine driver. This patch add MCLK clock DT parsing logic and control own MCLK. Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- sound/soc/codecs/wm8994.c | 22 +++++++++++++++++++++- sound/soc/codecs/wm8994.h | 1 + 2 files changed, 22 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 4fbc768..acb6b44 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -16,6 +16,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> #include <linux/gcd.h> #include <linux/i2c.h> #include <linux/platform_device.h> @@ -2471,6 +2472,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; + int i; wm_hubs_set_bias_level(codec, level); @@ -2492,8 +2494,15 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; } - if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { + for (i = 0; i < ARRAY_SIZE(wm8994->clk_mclk); i++) + clk_prepare_enable(wm8994->clk_mclk[i]); + active_reference(codec); + } else if (codec->dapm.bias_level == SND_SOC_BIAS_ON) { + for (i = 0; i < ARRAY_SIZE(wm8994->clk_mclk); i++) + clk_disable_unprepare(wm8994->clk_mclk[i]); + } break; case SND_SOC_BIAS_STANDBY: @@ -4449,6 +4458,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8994 = { static int wm8994_probe(struct platform_device *pdev) { struct wm8994_priv *wm8994; + int i; wm8994 = devm_kzalloc(&pdev->dev, sizeof(struct wm8994_priv), GFP_KERNEL); @@ -4460,6 +4470,16 @@ static int wm8994_probe(struct platform_device *pdev) wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent); + for (i = 0; i < ARRAY_SIZE(wm8994->clk_mclk); i++) { + char mclk_name[] = "MCLKx"; + + snprintf(mclk_name, sizeof(mclk_name), "MCLK%d", i + 1); + + wm8994->clk_mclk[i] = devm_clk_get(pdev->dev.parent, mclk_name); + if (IS_ERR(wm8994->clk_mclk[i])) + wm8994->clk_mclk[i] = NULL; + } + pm_runtime_enable(&pdev->dev); pm_runtime_idle(&pdev->dev); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index dd73387..d426c44 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -79,6 +79,7 @@ struct wm8994_priv { int sysclk[2]; int sysclk_rate[2]; int mclk[2]; + struct clk *clk_mclk[2]; int aifclk[2]; int aifdiv[2]; int channels[2]; -- 2.0.0.390.gcb682f8

2 1

[alsa-devel] [RFC PATCH] ASoC: wm8731: let codec to manage clock by itself
by Songjun Wu 16 Mar '15

16 Mar '15

Enable WM8731 to support common clock framework. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> --- sound/soc/codecs/wm8731.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 098c143..8df1550 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -25,6 +25,7 @@ #include <linux/spi/spi.h> #include <linux/of_device.h> #include <linux/mutex.h> +#include <linux/clk.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -45,6 +46,7 @@ static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = { /* codec private data */ struct wm8731_priv { struct regmap *regmap; + struct clk *mclk; struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; const struct snd_pcm_hw_constraint_list *constraints; unsigned int sysclk; @@ -390,6 +392,8 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (clk_id) { case WM8731_SYSCLK_XTAL: case WM8731_SYSCLK_MCLK: + if (wm8731->mclk && clk_set_rate(wm8731->mclk, freq)) + return -EINVAL; wm8731->sysclk_type = clk_id; break; default: @@ -491,6 +495,8 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + if (wm8731->mclk) + clk_prepare_enable(wm8731->mclk); break; case SND_SOC_BIAS_PREPARE: break; @@ -509,6 +515,8 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: + if (wm8731->mclk) + clk_disable_unprepare(wm8731->mclk); snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -667,6 +675,19 @@ static int wm8731_spi_probe(struct spi_device *spi) if (wm8731 == NULL) return -ENOMEM; + wm8731->mclk = devm_clk_get(&spi->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + if (ret == -ENOENT) { + wm8731->mclk = NULL; + dev_warn(&spi->dev, "Assuming static MCLK\n"); + } else { + dev_err(&spi->dev, "Failed to get MCLK: %d\n", + ret); + return ret; + } + } + mutex_init(&wm8731->lock); wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); @@ -718,6 +739,19 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, if (wm8731 == NULL) return -ENOMEM; + wm8731->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + if (ret == -ENOENT) { + wm8731->mclk = NULL; + dev_warn(&i2c->dev, "Assuming static MCLK\n"); + } else { + dev_err(&i2c->dev, "Failed to get MCLK: %d\n", + ret); + return ret; + } + } + mutex_init(&wm8731->lock); wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); -- 1.7.9.5

3 2

[alsa-devel] [PATCH 5/6] ESI W192M : Add text Line in/Mic for selecting input gain state
by klem.dev＠gmail.com 16 Mar '15

16 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add text Line in/Mic for selecting input gain state in mixer for ESI W192M. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index c7ffafa..a6e7afa 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -339,8 +339,14 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, /* * MIC / LINE switch fonction */ +static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { "Line In", "Mic" }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} -#define stac9460_mic_sw_info snd_ctl_boolean_mono_info static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -354,7 +360,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - ucontrol->value.integer.value[0] = ~val>>7 & 0x1; + ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; } @@ -370,7 +376,7 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); + new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) -- 2.1.4

3 3

[alsa-devel] [PATCH] ASoC: rt5677: add API to select ASRC clock source
by Oder Chiou 16 Mar '15

16 Mar '15

This patch defines an API to select the clock source for specified filters. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 163 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5677.h | 79 ++++++++++++++++++++++ 2 files changed, 242 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index c2a6e40..af18258 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1034,6 +1034,169 @@ static int can_use_asrc(struct snd_soc_dapm_widget *source, return 0; } +/** + * rt5677_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @codec: SoC audio codec device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5677 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the codec driver will turn on ASRC + * for these filters if ASRC is selected as their clock source. + */ +int rt5677_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src) +{ + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int asrc3_mask = 0, asrc3_value = 0; + unsigned int asrc4_mask = 0, asrc4_value = 0; + unsigned int asrc5_mask = 0, asrc5_value = 0; + unsigned int asrc6_mask = 0, asrc6_value = 0; + unsigned int asrc7_mask = 0, asrc7_value = 0; + + switch (clk_src) { + case RT5677_CLK_SEL_SYS: + case RT5677_CLK_SEL_I2S1_ASRC: + case RT5677_CLK_SEL_I2S2_ASRC: + case RT5677_CLK_SEL_I2S3_ASRC: + case RT5677_CLK_SEL_I2S4_ASRC: + case RT5677_CLK_SEL_I2S5_ASRC: + case RT5677_CLK_SEL_I2S6_ASRC: + case RT5677_CLK_SEL_SYS2: + case RT5677_CLK_SEL_SYS3: + case RT5677_CLK_SEL_SYS4: + case RT5677_CLK_SEL_SYS5: + case RT5677_CLK_SEL_SYS6: + case RT5677_CLK_SEL_SYS7: + break; + + default: + return -EINVAL; + } + + /* ASRC 3 */ + if (filter_mask & RT5677_DA_STEREO_FILTER) { + asrc3_mask |= RT5677_DA_STO_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5677_DA_STO_CLK_SEL_MASK) + | (clk_src << RT5677_DA_STO_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO2_L_FILTER) { + asrc3_mask |= RT5677_DA_MONO2L_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5677_DA_MONO2L_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO2L_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO2_R_FILTER) { + asrc3_mask |= RT5677_DA_MONO2R_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5677_DA_MONO2R_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO2R_CLK_SEL_SFT); + } + + if (asrc3_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_3, asrc3_mask, + asrc3_value); + + /* ASRC 4 */ + if (filter_mask & RT5677_DA_MONO3_L_FILTER) { + asrc4_mask |= RT5677_DA_MONO3L_CLK_SEL_MASK; + asrc4_value = (asrc4_value & ~RT5677_DA_MONO3L_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO3L_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO3_R_FILTER) { + asrc4_mask |= RT5677_DA_MONO3R_CLK_SEL_MASK; + asrc4_value = (asrc4_value & ~RT5677_DA_MONO3R_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO3R_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO4_L_FILTER) { + asrc4_mask |= RT5677_DA_MONO4L_CLK_SEL_MASK; + asrc4_value = (asrc4_value & ~RT5677_DA_MONO4L_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO4L_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO4_R_FILTER) { + asrc4_mask |= RT5677_DA_MONO4R_CLK_SEL_MASK; + asrc4_value = (asrc4_value & ~RT5677_DA_MONO4R_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO4R_CLK_SEL_SFT); + } + + if (asrc4_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_4, asrc4_mask, + asrc4_value); + + /* ASRC 5 */ + if (filter_mask & RT5677_AD_STEREO1_FILTER) { + asrc5_mask |= RT5677_AD_STO1_CLK_SEL_MASK; + asrc5_value = (asrc5_value & ~RT5677_AD_STO1_CLK_SEL_MASK) + | (clk_src << RT5677_AD_STO1_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_AD_STEREO2_FILTER) { + asrc5_mask |= RT5677_AD_STO2_CLK_SEL_MASK; + asrc5_value = (asrc5_value & ~RT5677_AD_STO2_CLK_SEL_MASK) + | (clk_src << RT5677_AD_STO2_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_AD_STEREO3_FILTER) { + asrc5_mask |= RT5677_AD_STO3_CLK_SEL_MASK; + asrc5_value = (asrc5_value & ~RT5677_AD_STO3_CLK_SEL_MASK) + | (clk_src << RT5677_AD_STO3_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_AD_STEREO4_FILTER) { + asrc5_mask |= RT5677_AD_STO4_CLK_SEL_MASK; + asrc5_value = (asrc5_value & ~RT5677_AD_STO4_CLK_SEL_MASK) + | (clk_src << RT5677_AD_STO4_CLK_SEL_SFT); + } + + if (asrc5_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_5, asrc5_mask, + asrc5_value); + + /* ASRC 6 */ + if (filter_mask & RT5677_AD_MONO_L_FILTER) { + asrc6_mask |= RT5677_AD_MONOL_CLK_SEL_MASK; + asrc6_value = (asrc6_value & ~RT5677_AD_MONOL_CLK_SEL_MASK) + | (clk_src << RT5677_AD_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_AD_MONO_R_FILTER) { + asrc6_mask |= RT5677_AD_MONOR_CLK_SEL_MASK; + asrc6_value = (asrc6_value & ~RT5677_AD_MONOR_CLK_SEL_MASK) + | (clk_src << RT5677_AD_MONOR_CLK_SEL_SFT); + } + + if (asrc6_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_6, asrc6_mask, + asrc6_value); + + /* ASRC 7 */ + if (filter_mask & RT5677_DSP_OB_0_3_FILTER) { + asrc7_mask |= RT5677_DSP_OB_0_3_CLK_SEL_MASK; + asrc7_value = (asrc7_value & ~RT5677_DSP_OB_0_3_CLK_SEL_MASK) + | (clk_src << RT5677_DSP_OB_0_3_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DSP_OB_4_7_FILTER) { + asrc7_mask |= RT5677_DSP_OB_4_7_CLK_SEL_MASK; + asrc7_value = (asrc7_value & ~RT5677_DSP_OB_4_7_CLK_SEL_MASK) + | (clk_src << RT5677_DSP_OB_4_7_CLK_SEL_SFT); + } + + if (asrc7_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_7, asrc7_mask, + asrc7_value); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5677_sel_asrc_clk_src); + /* Digital Mixer */ static const struct snd_kcontrol_new rt5677_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5677_STO1_ADC_MIXER, diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 07df96b..9dceb41 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1406,6 +1406,46 @@ #define RT5677_DSP_CLK_SRC_PLL2 (0x0 << 7) #define RT5677_DSP_CLK_SRC_BYPASS (0x1 << 7) +/* ASRC Control 3 (0x85) */ +#define RT5677_DA_STO_CLK_SEL_MASK (0xf << 12) +#define RT5677_DA_STO_CLK_SEL_SFT 12 +#define RT5677_DA_MONO2L_CLK_SEL_MASK (0xf << 4) +#define RT5677_DA_MONO2L_CLK_SEL_SFT 4 +#define RT5677_DA_MONO2R_CLK_SEL_MASK (0xf << 0) +#define RT5677_DA_MONO2R_CLK_SEL_SFT 0 + +/* ASRC Control 4 (0x86) */ +#define RT5677_DA_MONO3L_CLK_SEL_MASK (0xf << 12) +#define RT5677_DA_MONO3L_CLK_SEL_SFT 12 +#define RT5677_DA_MONO3R_CLK_SEL_MASK (0xf << 8) +#define RT5677_DA_MONO3R_CLK_SEL_SFT 8 +#define RT5677_DA_MONO4L_CLK_SEL_MASK (0xf << 4) +#define RT5677_DA_MONO4L_CLK_SEL_SFT 4 +#define RT5677_DA_MONO4R_CLK_SEL_MASK (0xf << 0) +#define RT5677_DA_MONO4R_CLK_SEL_SFT 0 + +/* ASRC Control 5 (0x87) */ +#define RT5677_AD_STO1_CLK_SEL_MASK (0xf << 12) +#define RT5677_AD_STO1_CLK_SEL_SFT 12 +#define RT5677_AD_STO2_CLK_SEL_MASK (0xf << 8) +#define RT5677_AD_STO2_CLK_SEL_SFT 8 +#define RT5677_AD_STO3_CLK_SEL_MASK (0xf << 4) +#define RT5677_AD_STO3_CLK_SEL_SFT 4 +#define RT5677_AD_STO4_CLK_SEL_MASK (0xf << 0) +#define RT5677_AD_STO4_CLK_SEL_SFT 0 + +/* ASRC Control 6 (0x88) */ +#define RT5677_AD_MONOL_CLK_SEL_MASK (0xf << 12) +#define RT5677_AD_MONOL_CLK_SEL_SFT 12 +#define RT5677_AD_MONOR_CLK_SEL_MASK (0xf << 8) +#define RT5677_AD_MONOR_CLK_SEL_SFT 8 + +/* ASRC Control 7 (0x89) */ +#define RT5677_DSP_OB_0_3_CLK_SEL_MASK (0xf << 12) +#define RT5677_DSP_OB_0_3_CLK_SEL_SFT 12 +#define RT5677_DSP_OB_4_7_CLK_SEL_MASK (0xf << 8) +#define RT5677_DSP_OB_4_7_CLK_SEL_SFT 8 + /* VAD Function Control 4 (0x9f) */ #define RT5677_VAD_SRC_MASK (0x7 << 8) #define RT5677_VAD_SRC_SFT 8 @@ -1670,6 +1710,42 @@ enum rt5677_type { RT5676, }; +/* ASRC clock source selection */ +enum { + RT5677_CLK_SEL_SYS, + RT5677_CLK_SEL_I2S1_ASRC, + RT5677_CLK_SEL_I2S2_ASRC, + RT5677_CLK_SEL_I2S3_ASRC, + RT5677_CLK_SEL_I2S4_ASRC, + RT5677_CLK_SEL_I2S5_ASRC, + RT5677_CLK_SEL_I2S6_ASRC, + RT5677_CLK_SEL_SYS2, + RT5677_CLK_SEL_SYS3, + RT5677_CLK_SEL_SYS4, + RT5677_CLK_SEL_SYS5, + RT5677_CLK_SEL_SYS6, + RT5677_CLK_SEL_SYS7, +}; + +/* filter mask */ +enum { + RT5677_DA_STEREO_FILTER = 0x1, + RT5677_DA_MONO2_L_FILTER = (0x1 << 1), + RT5677_DA_MONO2_R_FILTER = (0x1 << 2), + RT5677_DA_MONO3_L_FILTER = (0x1 << 3), + RT5677_DA_MONO3_R_FILTER = (0x1 << 4), + RT5677_DA_MONO4_L_FILTER = (0x1 << 5), + RT5677_DA_MONO4_R_FILTER = (0x1 << 6), + RT5677_AD_STEREO1_FILTER = (0x1 << 7), + RT5677_AD_STEREO2_FILTER = (0x1 << 8), + RT5677_AD_STEREO3_FILTER = (0x1 << 9), + RT5677_AD_STEREO4_FILTER = (0x1 << 10), + RT5677_AD_MONO_L_FILTER = (0x1 << 11), + RT5677_AD_MONO_R_FILTER = (0x1 << 12), + RT5677_DSP_OB_0_3_FILTER = (0x1 << 13), + RT5677_DSP_OB_4_7_FILTER = (0x1 << 14), +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1696,4 +1772,7 @@ struct rt5677_priv { bool is_vref_slow; }; +int rt5677_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src); + #endif /* __RT5677_H__ */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: nuc900: No need to track the dma buffer in the driver state struct
by Lars-Peter Clausen 16 Mar '15

16 Mar '15

The DMA buffer and address can be accessed through the snd_pcm_runtime. There is no need to manually track them in the driver's state struct. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/nuc900/nuc900-audio.h | 3 --- sound/soc/nuc900/nuc900-pcm.c | 31 +++++-------------------------- 2 files changed, 5 insertions(+), 29 deletions(-) diff --git a/sound/soc/nuc900/nuc900-audio.h b/sound/soc/nuc900/nuc900-audio.h index 59f7e8e..d0b7257 100644 --- a/sound/soc/nuc900/nuc900-audio.h +++ b/sound/soc/nuc900/nuc900-audio.h @@ -100,10 +100,7 @@ struct nuc900_audio { void __iomem *mmio; spinlock_t lock; - dma_addr_t dma_addr[2]; - unsigned long buffersize[2]; unsigned long irq_num; - struct snd_pcm_substream *substream; struct resource *res; struct clk *clk; struct device *dev; diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index b809fa9..5ae5ca1 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -42,29 +42,10 @@ static const struct snd_pcm_hardware nuc900_pcm_hardware = { static int nuc900_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct nuc900_audio *nuc900_audio = runtime->private_data; - unsigned long flags; - int ret = 0; - - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - if (ret < 0) - return ret; - - spin_lock_irqsave(&nuc900_audio->lock, flags); - - nuc900_audio->substream = substream; - nuc900_audio->dma_addr[substream->stream] = runtime->dma_addr; - nuc900_audio->buffersize[substream->stream] = - params_buffer_bytes(params); - - spin_unlock_irqrestore(&nuc900_audio->lock, flags); - - return ret; + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); } -static void nuc900_update_dma_register(struct snd_pcm_substream *substream, - dma_addr_t dma_addr, size_t count) +static void nuc900_update_dma_register(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct nuc900_audio *nuc900_audio = runtime->private_data; @@ -78,8 +59,8 @@ static void nuc900_update_dma_register(struct snd_pcm_substream *substream, mmio_len = nuc900_audio->mmio + ACTL_RDST_LENGTH; } - AUDIO_WRITE(mmio_addr, dma_addr); - AUDIO_WRITE(mmio_len, count); + AUDIO_WRITE(mmio_addr, runtime->dma_addr); + AUDIO_WRITE(mmio_len, runtime->dma_bytes); } static void nuc900_dma_start(struct snd_pcm_substream *substream) @@ -170,9 +151,7 @@ static int nuc900_dma_prepare(struct snd_pcm_substream *substream) spin_lock_irqsave(&nuc900_audio->lock, flags); - nuc900_update_dma_register(substream, - nuc900_audio->dma_addr[substream->stream], - nuc900_audio->buffersize[substream->stream]); + nuc900_update_dma_register(substream); val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: Make snd_soc_dapm_kcontrol_codec() inline
by Lars-Peter Clausen 16 Mar '15

16 Mar '15

snd_soc_dapm_kcontrol_codec() is a extremely simple function and inlining it typically results in less code than necessary for calling the non-inlined version of the function. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dapm.h | 1 - include/sound/soc.h | 13 +++++++++++++ sound/soc/soc-dapm.c | 10 ---------- 3 files changed, 13 insertions(+), 11 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 8d7416e..78633ef 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -440,7 +440,6 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card); int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget_list **list); -struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol); struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); diff --git a/include/sound/soc.h b/include/sound/soc.h index bc6cfab..92b92ac 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1268,6 +1268,19 @@ static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm( return component->dapm_ptr; } +/** + * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol + * @kcontrol: The kcontrol + * + * This function must only be used on DAPM contexts that are known to be part of + * a CODEC (e.g. in a CODEC driver). Otherwise the behavior is undefined. + */ +static inline struct snd_soc_codec *snd_soc_dapm_kcontrol_codec( + struct snd_kcontrol *kcontrol) +{ + return snd_soc_dapm_to_codec(snd_soc_dapm_kcontrol_dapm(kcontrol)); +} + /* codec IO */ unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg); int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b6f8820..95337c8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -473,16 +473,6 @@ struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( } EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_dapm); -/** - * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol - * @kcontrol: The kcontrol - */ -struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol) -{ - return snd_soc_dapm_to_codec(snd_soc_dapm_kcontrol_dapm(kcontrol)); -} -EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_codec); - static void dapm_reset(struct snd_soc_card *card) { struct snd_soc_dapm_widget *w; -- 1.8.0

2 1

[alsa-devel] [PATCH V8 0/5] ASoC: QCOM: Add support for ipq806x SOC
by Kenneth Westfield 16 Mar '15

16 Mar '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> This patch series adds support for I2S audio playback on the Storm board, which contains a Qualcomm Technologies ipq806x SOC and a Maxim max98357a DAC/amp. The ipq806x SOC has audio-related hardware blocks in its low-power audio subsystem (or LPASS). One of the relevant blocks in the LPASS is its low-power audio interface (or LPAIF). This contains an MI2S port, which is what these drivers are configured to use. The LPAIF also contains a DMA engine that is dedicated to moving audio samples into the transmit FIFO of the MI2S port. One bus from the MI2S port of the SOC is connected to the DAC/amp for stereo playback. This bus is configured so that the SOC is bus master and consists of DATA, LRCLK, and BCLK. The DAC/amp does not need MCLK to operate. In addition, a single GPIO pin from the SOC is connected to the same DAC/amp, which gives enable/disable control over the DAC/amp. The specific drivers added are: * a codec DAI driver for controlling the DAC/amp (applied from V4) * a CPU DAI driver for controlling the MI2S port (applied from V7) * a platform driver for controlling the LPAIF DMA engine (applied from V7) * a machine driver that instantiates a dai-link for playback The LPASS also contains clocks that need to be controlled. Those drivers have been submitted as several separate patch series: * [PATCH v3 0/8] qcom audio clock control drivers http://lkml.org/lkml/2015/1/19/656 * [PATCH] arm: dts: qcom: Add LCC nodes https://lkml.org/lkml/2015/1/28/557 * [PATCH 1/2] arm: qcom_defconfig: Enable lpass clock driver https://lkml.org/lkml/2015/1/28/582 * [PATCH] clk: qcom: Fix ipq806x LCC frequency tables https://lkml.org/lkml/2015/2/26/774 * [PATCH v2] clk: qcom: Properly change rates for ahbix clock https://lkml.org/lkml/2015/3/6/878 Even though the ipq806x LPASS does not contain an audio DSP, other SOCs do have one. For those SOCs, the audio DSP typically controls the hardware blocks in the LPASS. Hence, different CPU DAI driver(s) would need to be used in order to facilitate audio with the DSP. As such, the LPASS DT binding description alludes to an optional adsp phandle, which is not present for this SOC. For other SOCs that do contain an audio DSP, the phandle should exist and refer to a an adsp node. Not using the audio DSP on an SOC that does have one would require different CPU DAI driver(s), in addition to possible bootloader and/or firmware changes. Note that the use of a phandle was a modification that has been submitted as part of a separate patch series. Corresponding additions to the device tree for the ipq806x SOC and its documentation have also been added. Also, as this is a new directory, the MAINTAINERS file has been updated as well (applied from V7). = Changes since V7 [Patch V7 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2015-March/088546.html * Changed the compatible property of the machine driver to make it specific to audio * Removed the lcc node definition from the DT, as it was submitted separately = Changes since V6 [Patch V6 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/088218.h… * Added REGMAP_MMIO selection for the CPU and platform drivers * Modified the AHBIX frequency to match the corresponding LCC fix * Tweaked the logging in the CPU driver probe. = Changes since V5 [Patch V5 00/12] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/087832.h… * Correctly use Storm as the build target label and DT binding label. * Added audio DSP sub-node to the LPASS device tree, disabled for this SOC. * Added logic to CPU DAI driver to fail the probe() if a DSP is present. * Use the standard naming convention for the DAI link properties. * General code cleanup. = Changes since V4 [Patch V4 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/087499.h… * Replaced simple-card with a machine driver to resolve the system clock configuration, rather than having the CPU DAI driver do it. * Added header files to avoid indirect header dependencies and implicit forward declarations. * Tweaked the ISR to match the conventions of the surrounding code. * Removed the usage of the low-power memory as it is not needed. * Removed the use of the DRV_NAME constant. * Added explicit dependency on gpiolib for the codec driver. * Moved the MODULE_DEVICE_TABLE macro inside the CONFIG_OF conditional. * Modified the documentation to reflect the changes. * General code cleanup. = Changes since V3 [Patch V3 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085694.h… * Placed the content of the inline functions into the callbacks. * Replaced use of readl/writel register access functions with regmap access functions. Notable exception is the ISR, which uses ioread32/iowrite32. * Rearranged the sequencing of the hardware block enables to fit within the ASoC framework callbacks, while remaining functional. REQ 1: The hardware requires the enable sequence to be: LPAIF-DMA[enable],then LPAIF-MI2S[enable], then DAC-GPIO[enable] REQ 2: The hardware requires the disable sequence to be: DAC-GPIO[disable], then LPAIF-MI2S[disable] * Corrected the implementation of the pointer callback. * Utilize the LPM to buffer audio samples, rather than memory external to LPASS. * Corrected the interrupt clearing in the ISR. * Implemented a default system clock (defined by the simple-card DT node), and optional LPASS DT node modifiers that can alter the system clock in order to expand the range of available bit clock frequencies. * Addressed all of the remaining issues raised by Mark Brown. * General code cleanup. = Changes since V2 [Patch v2 00/11] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085186.h… * Removed the PCM platform driver from the DTS platform and tied it to the CPU DAI driver. * Changed I2S pinctrl to use generic naming convention and moved control to CPU DAI driver. It should be controlled now by soc-core's pinctrl_pm_* functionality. * Added stub DAPM support in codec driver. As the DAC GPIO needs to be enabled last when starting playback, and disabled first when stopping playback, it seems as though the trigger function may be the place for this. Suggestions are welcome for a better place to put this. * Removed machine driver and tied DAI drivers to simple-audio-card. * Packaged the build files and Maxim codec files together in one change. * Removed QCOM as vendor from Maxim code and documentation. * Separated the SOC and board definitions into the correct DTS files. * Update device tree documentation to reflect changes. * General code cleanup. = Changes since V1 [PATCH 0/9] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084322.h… * Remove the native LPAIF driver, and move its functionality to the CPU DAI driver. * Add a codec driver to manage the pins going to the external DAC (previously managed by the machine driver). * Use devm_* and dev_* where possible. * ISR only handles relevant DMA channel now. * Update device tree documentation to reflect changes. * General code cleanup. Kenneth Westfield (5): ASoC: qcom: Document Storm bindings ASoC: qcom: Add Storm machine driver ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers ARM: dts: Model IPQ LPASS audio hardware Documentation/devicetree/bindings/sound/storm.txt | 23 +++ arch/arm/boot/dts/qcom-ipq8064.dtsi | 16 +++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/qcom/Kconfig | 25 ++++ sound/soc/qcom/Makefile | 11 ++ sound/soc/qcom/storm.c | 162 ++++++++++++++++++++++ 7 files changed, 239 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/storm.txt create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/storm.c -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 6

[alsa-devel] [PATCH 1/2] ASoC: qcom: Change qcom, adsp in LPASS CPU bindings
by Kenneth Westfield 16 Mar '15

16 Mar '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> Change the representation of the audio DSP, in the LPASS CPU bindings description, from a required subnode to an optional phandle. Signed-off-by: Kenneth Westfield <kwestfie(a)codeaurora.org> --- Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt index e7c6e9321863f022ebf0d51b75d7bb83c10b9062..e00732dac9393e95f0530505c9f38127f58980a3 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt @@ -22,13 +22,9 @@ Required properties: - reg-names : A list which must include the following entries: * "lpass-lpaif" -Required subnodes: +Optional properties: -- qcom,adsp : Audio DSP sub-node - -Optional Audio DSP subnode properties: - -- status : "disabled" indicates the adsp is not available. +- qcom,adsp : Phandle for the audio DSP node Example: @@ -43,7 +39,5 @@ lpass@28100000 { pinctrl-1 = <&mi2s_idle>; reg = <0x28100000 0x10000>; reg-names = "lpass-lpaif"; - qcom,adsp { - status = "disabled"; - }; + qcom,adsp = <&adsp>; }; -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 2

[alsa-devel] [PATCH] ALSA: hda - Don't access stereo amps for mono channel widgets
by Takashi Iwai 16 Mar '15

16 Mar '15

The current HDA generic parser initializes / modifies the amp values always in stereo, but this seems causing the problem on ALC3229 codec that has a few mono channel widgets: namely, these mono widgets react to actions for both channels equally. In the driver code, we do care the mono channel and create a control only for the left channel (as defined in HD-audio spec) for such a node. When the control is updated, only the left channel value is changed. However, in the resume, the right channel value is also restored from the initial value we took as stereo, and this overwrites the left channel value. This ends up being the silent output as the right channel has been never touched and remains muted. This patch covers the places where unconditional stereo amp accesses are done and converts to the conditional accesses. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=94581 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 30 ++++++++++++++++++++++-------- 1 file changed, 22 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b680b4ec6331..fe18071bf93a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -692,7 +692,23 @@ static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx) { unsigned int caps = query_amp_caps(codec, nid, dir); int val = get_amp_val_to_activate(codec, nid, dir, caps, false); - snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val); + + if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val); + else + snd_hda_codec_amp_init(codec, nid, 0, dir, idx, 0xff, val); +} + +/* update the amp, doing in stereo or mono depending on NID */ +static int update_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx, + unsigned int mask, unsigned int val) +{ + if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + return snd_hda_codec_amp_stereo(codec, nid, dir, idx, + mask, val); + else + return snd_hda_codec_amp_update(codec, nid, 0, dir, idx, + mask, val); } /* calculate amp value mask we can modify; @@ -732,7 +748,7 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir, return; val &= mask; - snd_hda_codec_amp_stereo(codec, nid, dir, idx, mask, val); + update_amp(codec, nid, dir, idx, mask, val); } static void activate_amp_out(struct hda_codec *codec, struct nid_path *path, @@ -4424,13 +4440,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) has_amp = nid_has_mute(codec, mix, HDA_INPUT); for (i = 0; i < nums; i++) { if (has_amp) - snd_hda_codec_amp_stereo(codec, mix, - HDA_INPUT, i, - 0xff, HDA_AMP_MUTE); + update_amp(codec, mix, HDA_INPUT, i, + 0xff, HDA_AMP_MUTE); else if (nid_has_volume(codec, conn[i], HDA_OUTPUT)) - snd_hda_codec_amp_stereo(codec, conn[i], - HDA_OUTPUT, 0, - 0xff, HDA_AMP_MUTE); + update_amp(codec, conn[i], HDA_OUTPUT, 0, + 0xff, HDA_AMP_MUTE); } } -- 2.3.2

2 4

[alsa-devel] [PATCH 0/13] ASoC: Renesas adds DPCM based sampling rate convert
by Kuninori Morimoto 16 Mar '15

16 Mar '15

Hi Mark These are DPCM based sampling rate convert support patch-set for Rensas sound driver. Then, it doesn't use simple-card. it uses new Renesas specific sound card driver, but it is copied from simple-card. Current ASoC DPCM needs route settings for FE/BE, but sometimes it will have trouble, because many drivers are using same name for this routing. This patch set solved this issue too. 1 - 2 : prepare patch for Renesas new sound card driver 3 - 4 : DAPM routing issue solution patches 5 - 6 : new Renesas sound card driver 7 - 11 : fixup/tidyup patches for rsnd 12 - 13 : DPCM support patch for rsnd Kuninori Morimoto (13): 1) ASoC: core: remove of_xlate_dai_name from snd_soc_component_driver 2) ASoC: core: add snd_soc_get_dai_info() to get driver infomation 3) ASoC: dapm: add snd_soc_dapm_route_scan macro for DAPM route 4) ASoC: dapm: add driver prefix matching for snd_soc_dapm_add_route() 5) ASoC: rsnd-dpcm-card: add Renesas sound card support for DPCM 6) ASoC: rsnd-dpcm-card: add .be_hw_params_fixup support for convert rate 7) ASoC: rsnd: no more SSI restart when unusual situation 8) ASoC: rsnd: no more SRC restart when unusual situation 9) ASoC: rsnd: tidyup error message format 10) ASoC: rsnd: show debug info for sampling rate convert 11) ASoC: rsnd: don't call clk_prepare_enable/unprepare() from inside spin lock 12) ASoC: rsnd: add dai_link stream name 13) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsnd-dpcm-card.txt | 72 ++++++++++++++ include/sound/soc.h | 6 +- sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 + sound/soc/sh/rcar/core.c | 48 ++++++++- sound/soc/sh/rcar/dvc.c | 12 +-- sound/soc/sh/rcar/rsnd-dpcm-card.c | 491 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 16 ++- sound/soc/sh/rcar/src.c | 63 ++++++++++-- sound/soc/sh/rcar/ssi.c | 31 ++++-- sound/soc/soc-core.c | 71 ++++++++------ sound/soc/soc-dapm.c | 33 +++++-- 13 files changed, 780 insertions(+), 75 deletions(-) Best regards --- Kuninori Morimoto

2 16

[alsa-devel] [PATCH 6/6] ESI W192M : Add sampling rate control of the ADC/DAC
by klem.dev＠gmail.com 16 Mar '15

16 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add sampling rate control for ADC/DAC for ESI W192M. Allow to switch between 48K/96K/192K sampling rate. All DAC need to be mute when changing samplerate. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index a6e7afa..f3e3879 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -30,12 +30,18 @@ #include <linux/init.h> #include <sound/core.h> #include <sound/tlv.h> +#include <linux/slab.h> #include "ice1712.h" #include "envy24ht.h" #include "wtm.h" #include "stac946x.h" +struct wtm_spec { + /* rate change needs atomic mute/unmute of all dacs*/ + struct mutex mute_mutex; +}; + /* * 2*ADC 6*DAC no1 ringbuffer r/w on i2c bus @@ -69,15 +75,65 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg) /* * DAC mute control */ +static void stac9460_dac_mute_all(struct snd_ice1712 *ice, unsigned char mute, + unsigned short int *change_mask) +{ + unsigned char new, old; + int id, idx, change; + + /*stac9460 1*/ + for (id = 0; id < 7; id++) { + if (*change_mask & (0x01 << id)) { + if (id == 0) + idx = STAC946X_MASTER_VOLUME; + else + idx = STAC946X_LF_VOLUME - 1 + id; + old = stac9460_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) { + stac9460_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + } else { + *change_mask = *change_mask & ~(0x01 << id); + } + } + } + + /*stac9460 2*/ + for (id = 0; id < 3; id++) { + if (*change_mask & (0x01 << (id + 7))) { + if (id == 0) + idx = STAC946X_MASTER_VOLUME; + else + idx = STAC946X_LF_VOLUME - 1 + id; + old = stac9460_2_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) { + stac9460_2_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + } else { + *change_mask = *change_mask & ~(0x01 << id); + } + } + } +} + + + #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct wtm_spec *spec = ice->spec; unsigned char val; int idx, id; + mutex_lock(&spec->mute_mutex); + if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; id = 0; @@ -90,6 +146,8 @@ static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, else val = stac9460_2_get(ice, idx - 6); ucontrol->value.integer.value[0] = (~val >> 7) & 0x1; + + mutex_unlock(&spec->mute_mutex); return 0; } @@ -388,6 +446,44 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, } +/* + * Handler for setting correct codec rate - called when rate change is detected + */ +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned char old, new; + unsigned short int changed; + struct wtm_spec *spec = ice->spec; + + if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + return; + else if (rate <= 48000) + new = 0x08; /* 256x, base rate mode */ + else if (rate <= 96000) + new = 0x11; /* 256x, mid rate mode */ + else + new = 0x12; /* 128x, high rate mode */ + + old = stac9460_get(ice, STAC946X_MASTER_CLOCKING); + if (old == new) + return; + /* change detected, setting master clock, muting first */ + /* due to possible conflicts with mute controls - mutexing */ + mutex_lock(&spec->mute_mutex); + /* we have to remember current mute status for each DAC */ + changed = 0xFFFF; + stac9460_dac_mute_all(ice, 0, &changed); + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); + stac9460_2_put(ice, STAC946X_MASTER_CLOCKING, new); + udelay(10); + /* unmuting - only originally unmuted dacs - + * i.e. those changed when muting */ + stac9460_dac_mute_all(ice, 1, &changed); + mutex_unlock(&spec->mute_mutex); +} + + /*Limits value in dB for fader*/ static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); @@ -487,21 +583,32 @@ static int wtm_init(struct snd_ice1712 *ice) { static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, + STAC946X_MASTER_CLOCKING, 0x11, (unsigned short)-1 }; unsigned short *p; + struct wtm_spec *spec; /*WTM 192M*/ ice->num_total_dacs = 8; ice->num_total_adcs = 4; ice->force_rdma1 = 1; + /*init mutex for dac mute conflict*/ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + mutex_init(&spec->mute_mutex); + + /*initialize codec*/ p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); } + ice->gpio.set_pro_rate = stac9460_set_rate_val; return 0; } -- 2.1.4

1 0

[alsa-devel] [PATCH 4/6] ESI W192M : Add TLV support for control value in dB scale
by klem.dev＠gmail.com 16 Mar '15

16 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add TLV support to control volume using dB scale for input and ouput on ESI W192M. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 59483b4..c7ffafa 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -29,6 +29,7 @@ #include <linux/interrupt.h> #include <linux/init.h> #include <sound/core.h> +#include <sound/tlv.h> #include "ice1712.h" #include "envy24ht.h" @@ -380,17 +381,25 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, return change; } + +/*Limits value in dB for fader*/ +static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); + /* * Control tabs */ static struct snd_kcontrol_new stac9640_controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Switch", .info = stac9460_dac_mute_info, .get = stac9460_dac_mute_get, .put = stac9460_dac_mute_put, - .private_value = 1 + .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -419,11 +428,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "DAC Volume", .count = 8, .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -435,12 +448,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "ADC Volume", .count = 2, .info = stac9460_adc_vol_info, .get = stac9460_adc_vol_get, .put = stac9460_adc_vol_put, - + .tlv = { .p = db_scale_adc } } }; -- 2.1.4

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[alsa-devel] [PATCH 3/6] ESI W192M : Enable midi i/o of port envy24 chip as available
by klem.dev＠gmail.com 16 Mar '15

16 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Enable midi i/o port of envy24 chip as their are available on ESI W192M soundcard. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 6e1026e..59483b4 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -485,7 +485,8 @@ static int wtm_init(struct snd_ice1712 *ice) static unsigned char wtm_eeprom[] = { - [ICE_EEP2_SYSCONF] = 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ + [ICE_EEP2_SYSCONF] = 0x67, /*SYSCONF: clock 192KHz, mpu401, + 4ADC, 8DAC */ [ICE_EEP2_ACLINK] = 0x80, /* ACLINK : I2S */ [ICE_EEP2_I2S] = 0xf8, /* I2S: vol; 96k, 24bit, 192k */ [ICE_EEP2_SPDIF] = 0xc1, /*SPDIF: out-en, spidf ext out*/ -- 2.1.4

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