April 2015 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC.
by Mark Brown 10 Apr '15

10 Apr '15

On Thu, Apr 02, 2015 at 11:47:18AM -0700, Lori Hikichi wrote: > On 15-03-30 11:42 PM, Mark Brown wrote: > >>+config SND_SOC_CYGNUS > >>+ tristate "SoC platform audio for Broadcom Cygnus chips" > >>+ depends on ARCH_BCM_CYGNUS || COMPILE_TEST > >>+ default ARCH_BCM_CYGNUS > Okay. You don't need to reply to every single comment, the general assumption will be that if there's no other followup all review comments will be addressed. It's better to just reply to things where there's something more detailed to say, if you explicitly reply to everything then that makes it easier for actual replies to be missed since there's a lot of there's a lot of the mail that's just going to be skipped through. > >>+static void ringbuf_inc(void __iomem *audio_io, bool is_playback, > >>+ const struct ringbuf_regs *p_rbuf) > >So it looks like we're getting an interrupt per period and we have to > >manually advance to the next one? > Yes. OK, so that seems fragile - what happens if we're slightly late processing an interrupt or miss one entirely? Most hardware has some way to read back the current position which tends to be more reliable, is that not an option here?

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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

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[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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