July 2013 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH] ASoC: generic: add simple card with devicetree support
by Mark Brown 22 Jul '13

22 Jul '13

On Sat, Jul 20, 2013 at 07:25:54PM +0200, Jean-Francois Moine wrote: > Documentation/devicetree/bindings/sound/simple-dt-card.txt | 18 ++++ > sound/soc/generic/Kconfig | 7 ++ > sound/soc/generic/Makefile | 2 + > sound/soc/generic/simple-dt-card.c | 137 ++++++++++++++++++++++++ > 4 files changed, 163 insertions(+) The obvious question here is why is this a separate driver to the existing simple card driver rather than DT bindings for that? I don't see that mentioned here.

1 0

Re: [alsa-devel] [PATCH] ASoC: tegra: correct playback_dma_data setup
by Lucas Stach 21 Jul '13

21 Jul '13

Am Sonntag, den 21.07.2013, 10:34 +0800 schrieb Richard Zhao: > The errors were caused by copy/paste mistake in below commit > since v3.10: > 3489d50 ASoC: tegra: Use common DAI DMA data struct > > It also corrects slave_id initialization in tegra20_ac97 driver. > > Signed-off-by: Richard Zhao <rizhao(a)nvidia.com> > Cc: Lars-Peter Clausen <lars(a)metafoo.de> > Cc: <stable(a)vger.kernel.org> # 3.10 Acked-by: Lucas Stach <dev(a)lynxeye.de> > --- > sound/soc/tegra/tegra20_ac97.c | 6 +++--- > sound/soc/tegra/tegra20_spdif.c | 4 ++-- > 2 files changed, 5 insertions(+), 5 deletions(-) > > diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c > index e58233f..6c48662 100644 > --- a/sound/soc/tegra/tegra20_ac97.c > +++ b/sound/soc/tegra/tegra20_ac97.c > @@ -389,9 +389,9 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) > ac97->capture_dma_data.slave_id = of_dma[1]; > > ac97->playback_dma_data.addr = mem->start + TEGRA20_AC97_FIFO_TX1; > - ac97->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; > - ac97->capture_dma_data.maxburst = 4; > - ac97->capture_dma_data.slave_id = of_dma[0]; > + ac97->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; > + ac97->playback_dma_data.maxburst = 4; > + ac97->playback_dma_data.slave_id = of_dma[1]; > > ret = tegra_asoc_utils_init(&ac97->util_data, &pdev->dev); > if (ret) > diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c > index 5eaa12c..551b3c9 100644 > --- a/sound/soc/tegra/tegra20_spdif.c > +++ b/sound/soc/tegra/tegra20_spdif.c > @@ -323,8 +323,8 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) > } > > spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; > - spdif->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; > - spdif->capture_dma_data.maxburst = 4; > + spdif->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; > + spdif->playback_dma_data.maxburst = 4; > spdif->playback_dma_data.slave_id = dmareq->start; > > pm_runtime_enable(&pdev->dev);

1 0

Re: [alsa-devel] [PATCH] sound: replace strict_strto*() with kstrto*()
by Takashi Iwai 21 Jul '13

21 Jul '13

At Fri, 19 Jul 2013 16:24:59 +0900, Jingoo Han wrote: > > The usage of strict_strto*() is not preferred, because > strict_strto*() is obsolete. Thus, kstrto*() should be > used. > > Signed-off-by: Jingoo Han <jg1.han(a)samsung.com> Thanks, applied. Takashi > --- > sound/drivers/dummy.c | 2 +- > sound/pci/hda/hda_hwdep.c | 6 +++--- > sound/soc/codecs/wm8962.c | 2 +- > sound/soc/omap/mcbsp.c | 2 +- > sound/soc/soc-core.c | 8 +++++--- > 5 files changed, 11 insertions(+), 9 deletions(-) > > diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c > index 11048cc..915b4d7 100644 > --- a/sound/drivers/dummy.c > +++ b/sound/drivers/dummy.c > @@ -1022,7 +1022,7 @@ static void dummy_proc_write(struct snd_info_entry *entry, > if (i >= ARRAY_SIZE(fields)) > continue; > snd_info_get_str(item, ptr, sizeof(item)); > - if (strict_strtoull(item, 0, &val)) > + if (kstrtoull(item, 0, &val)) > continue; > if (fields[i].size == sizeof(int)) > *get_dummy_int_ptr(dummy, fields[i].offset) = val; > diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c > index ce67608..fe0bda1 100644 > --- a/sound/pci/hda/hda_hwdep.c > +++ b/sound/pci/hda/hda_hwdep.c > @@ -295,7 +295,7 @@ static ssize_t type##_store(struct device *dev, \ > struct snd_hwdep *hwdep = dev_get_drvdata(dev); \ > struct hda_codec *codec = hwdep->private_data; \ > unsigned long val; \ > - int err = strict_strtoul(buf, 0, &val); \ > + int err = kstrtoul(buf, 0, &val); \ > if (err < 0) \ > return err; \ > codec->type = val; \ > @@ -654,7 +654,7 @@ int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp) > p = snd_hda_get_hint(codec, key); > if (!p) > ret = -ENOENT; > - else if (strict_strtoul(p, 0, &val)) > + else if (kstrtoul(p, 0, &val)) > ret = -EINVAL; > else { > *valp = val; > @@ -751,7 +751,7 @@ static void parse_##name##_mode(char *buf, struct hda_bus *bus, \ > struct hda_codec **codecp) \ > { \ > unsigned long val; \ > - if (!strict_strtoul(buf, 0, &val)) \ > + if (!kstrtoul(buf, 0, &val)) \ > (*codecp)->name = val; \ > } > > diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c > index e2de9ec..e37c06f 100644 > --- a/sound/soc/codecs/wm8962.c > +++ b/sound/soc/codecs/wm8962.c > @@ -3175,7 +3175,7 @@ static ssize_t wm8962_beep_set(struct device *dev, > long int time; > int ret; > > - ret = strict_strtol(buf, 10, &time); > + ret = kstrtol(buf, 10, &time); > if (ret != 0) > return ret; > > diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c > index 361e4c0..83433fd 100644 > --- a/sound/soc/omap/mcbsp.c > +++ b/sound/soc/omap/mcbsp.c > @@ -781,7 +781,7 @@ static ssize_t prop##_store(struct device *dev, \ > unsigned long val; \ > int status; \ > \ > - status = strict_strtoul(buf, 0, &val); \ > + status = kstrtoul(buf, 0, &val); \ > if (status) \ > return status; \ > \ > diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c > index 0ec070c..88daa64 100644 > --- a/sound/soc/soc-core.c > +++ b/sound/soc/soc-core.c > @@ -192,7 +192,7 @@ static ssize_t pmdown_time_set(struct device *dev, > struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); > int ret; > > - ret = strict_strtol(buf, 10, &rtd->pmdown_time); > + ret = kstrtol(buf, 10, &rtd->pmdown_time); > if (ret) > return ret; > > @@ -237,6 +237,7 @@ static ssize_t codec_reg_write_file(struct file *file, > char *start = buf; > unsigned long reg, value; > struct snd_soc_codec *codec = file->private_data; > + int ret; > > buf_size = min(count, (sizeof(buf)-1)); > if (copy_from_user(buf, user_buf, buf_size)) > @@ -248,8 +249,9 @@ static ssize_t codec_reg_write_file(struct file *file, > reg = simple_strtoul(start, &start, 16); > while (*start == ' ') > start++; > - if (strict_strtoul(start, 16, &value)) > - return -EINVAL; > + ret = kstrtoul(start, 16, &value); > + if (ret) > + return ret; > > /* Userspace has been fiddling around behind the kernel's back */ > add_taint(TAINT_USER, LOCKDEP_NOW_UNRELIABLE); > -- > 1.7.10.4 > >

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: 6fire: return correct XRUN indication
by Eldad Zack 21 Jul '13

21 Jul '13

Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer function of 6fire, as expected by snd_pcm_update_hw_ptr0(). Caught by sparse. Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/6fire/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index c5b9cac..60ac8fb 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -543,7 +543,7 @@ static snd_pcm_uframes_t usb6fire_pcm_pointer( snd_pcm_uframes_t ret; if (rt->panic || !sub) - return SNDRV_PCM_STATE_XRUN; + return SNDRV_PCM_POS_XRUN; spin_lock_irqsave(&sub->lock, flags); ret = sub->dma_off; -- 1.8.1.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Add snd_hda_jack_detect_state() helper function
by Takashi Iwai 21 Jul '13

21 Jul '13

snd_hda_jack_detect() function returns a boolean value for a jack plugged in or not, but it also returns always true when the corresponding pin is phantom (i.e. fixed). This is OK in most cases, but it makes the generic parser misbehaving about the auto-mute or auto-mic switching, e.g. when one of headphone pins is a fixed. Namely, the driver decides whether to mute the speaker or not, just depending on the headphone plug state: if one of the headphone jacks is seen as active, then the speaker is muted. Thus this will result always in the muted speaker output. So, the problem is the function returns a boolean, after all, although we need to think of "phantom" jack. Now a new function, snd_hda_jack_detect_state() is introduced to return these tristates. The generic parser uses this function for checking the headphone or mic jack states. Meanwhile, the behavior of snd_hda_jack_detect() is kept as is, for keeping compatibility in other driver codes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 8 +++++--- sound/pci/hda/hda_jack.c | 18 ++++++++++++------ sound/pci/hda/hda_jack.h | 13 ++++++++++++- 3 files changed, 29 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8e77cbb..f5c2d1f 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3724,7 +3724,8 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx, /* check each pin in the given array; returns true if any of them is plugged */ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) { - int i, present = 0; + int i; + bool present = false; for (i = 0; i < num_pins; i++) { hda_nid_t nid = pins[i]; @@ -3733,7 +3734,8 @@ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) /* don't detect pins retasked as inputs */ if (snd_hda_codec_get_pin_target(codec, nid) & AC_PINCTL_IN_EN) continue; - present |= snd_hda_jack_detect(codec, nid); + if (snd_hda_jack_detect_state(codec, nid) == HDA_JACK_PRESENT) + present = true; } return present; } @@ -3887,7 +3889,7 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *ja /* don't detect pins retasked as outputs */ if (snd_hda_codec_get_pin_target(codec, pin) & AC_PINCTL_OUT_EN) continue; - if (snd_hda_jack_detect(codec, pin)) { + if (snd_hda_jack_detect_state(codec, pin) == HDA_JACK_PRESENT) { mux_select(codec, 0, spec->am_entry[i].idx); return; } diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 3fd2973..dc93761 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -194,18 +194,24 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) EXPORT_SYMBOL_HDA(snd_hda_pin_sense); /** - * snd_hda_jack_detect - query pin Presence Detect status + * snd_hda_jack_detect_state - query pin Presence Detect status * @codec: the CODEC to sense * @nid: the pin NID to sense * - * Query and return the pin's Presence Detect status. + * Query and return the pin's Presence Detect status, as either + * HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT or HDA_JACK_PHANTOM. */ -int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) +int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid) { - u32 sense = snd_hda_pin_sense(codec, nid); - return get_jack_plug_state(sense); + struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); + if (jack && jack->phantom_jack) + return HDA_JACK_PHANTOM; + else if (snd_hda_pin_sense(codec, nid) & AC_PINSENSE_PRESENCE) + return HDA_JACK_PRESENT; + else + return HDA_JACK_NOT_PRESENT; } -EXPORT_SYMBOL_HDA(snd_hda_jack_detect); +EXPORT_SYMBOL_HDA(snd_hda_jack_detect_state); /** * snd_hda_jack_detect_enable - enable the jack-detection diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index ec12abd..379420c 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -75,7 +75,18 @@ int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid); u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); -int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); + +/* the jack state returned from snd_hda_jack_detect_state() */ +enum { + HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT, HDA_JACK_PHANTOM, +}; + +int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid); + +static inline bool snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) +{ + return snd_hda_jack_detect_state(codec, nid) != HDA_JACK_NOT_PRESENT; +} bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid); -- 1.8.3.1

2 2

Re: [alsa-devel] [PATCH] ASoC: tegra: correct playback_dma_data setup
by Stephen Warren 21 Jul '13

21 Jul '13

On 07/20/2013 08:34 PM, Richard Zhao wrote: > The errors were caused by copy/paste mistake in below commit > since v3.10: > 3489d50 ASoC: tegra: Use common DAI DMA data struct > > It also corrects slave_id initialization in tegra20_ac97 driver. Acked-by: Stephen Warren <swarren(a)nvidia.com> Hmm, that's what I get for only testing I2S and not SPDIF or AC'97 (not that I can test the latter).

1 0

Re: [alsa-devel] [PATCH] ASoC: generic: add simple card with devicetree support
by Stephen Warren 21 Jul '13

21 Jul '13

On 07/20/2013 11:25 AM, Jean-Francois Moine wrote: > This generic simple card driver uses DT values to instanciate an audio > system in which the real work is done by the subdrivers (audio > controller, audio codec and pcm/dma controller). New DT bindings should be sent to devicetree(a)vger.kernel.org for review. Note that's a branch new list (it moved from a different server), so it might be best to wait a few days for people to subscribe before sending mail to it.

1 0

[alsa-devel] [PATCH RFC 00/10] Refactor set_format()
by Eldad Zack 20 Jul '13

20 Jul '13

Hi, This is a low-priority series (applies against current for-next). I had the idea of refactoring set_format to make it easier to read (I actually indended to look at the implicit feedback setup race, but stumbled on this again). This series is not intended to be applied as-is (it has some whitespace issues and I wrote the commit messages quickly), but to get some feedback - if you think it's a good idea in general, and also specific feedback to individual patches. Assuming it makes sense, I'll fix the whitespace issues, commit messages and anything else before resubmitting. Cheers, Eldad Eldad Zack (10): ALSA: usb-audio: remove disabled debug code in set_format ALSA: usb-audio: remove assignment from if condition ALSA: usb-audio: separate sync endpoint setting from set_format ALSA: usb-audio: move implicit fb quirks to separate function ALSA: usb-audio: reverse condition logic in set_sync_endpoint ALSA: usb-audio: do not initialize and check implicit_fb ALSA: usb-audio: remove is_playback from implicit feedback quirks ALSA: usb-audio: remove implicit_fb from quirk ALSA: usb-audio: clarify implicit fb quirk for Roland ALSA: usb-audio: check alts is not NULL before adding ep sound/usb/pcm.c | 249 ++++++++++++++++++++++++++++++++------------------------ 1 file changed, 144 insertions(+), 105 deletions(-) -- 1.8.1.5

2 12

Re: [alsa-devel] [ecasound] Garbage at end of playback: ecasound or alsa broken?
by Knut Petersen 20 Jul '13

20 Jul '13

On 20.07.2013 11:57, Julien Claassen wrote: > Hello Knut! > No idea yet, who's to blame. but have you tried repeating the experiment > with another tool. You can use audacity, so you can use the GUI. Audacity is even more broken. That´s true for the version from opensuse (2.03), and it´s true for the current development code, If I mark that 1s of audio described in the original post, a lot of audio is missing at the beginning and a bit of audio is missing at end. In total only about 460 msec of audio is captured, 540 msec is lost. Audacity currently does not playback correctly with non-memory-mapped alsa devices. But that´s another story ;-( > There are other recording applications. Or you might even try with the alsa utilities > aplay and arecord. this will require two console or terminals. If I put arecord -f cd -D hw:0,0 -d 2 testarec.wav & aplay testin.wav -D hw:0,0 into a script I get a perfect recording: 47 msec silence, exactly 1s of audio data, silence up to the end of the file. Well, that result seems to indicate that I found an ecasound bug. Unless it´s some alsa code unused by aplay ;-) > You might also try the same experiment with JACK, if you're not averse to it. In the best case that could only mask the bug. > I did digitise a few tapes and prelistened to the volume and audio quality. > So it's not a problem in general. > Half a second silence at the end of the audio masks the bug. Because of that I agree that it´s not a general problem, most people will never notice it. But it´s still a bug. cu, Knut

1 0

[alsa-devel] [PATCH] ASoC: twl6040: Use power efficient workqueue
by Mark Brown 20 Jul '13

20 Jul '13

From: Mark Brown <broonie(a)linaro.org> The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/twl6040.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 44621dd..caf8784 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -429,7 +429,8 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) struct snd_soc_codec *codec = data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - schedule_delayed_work(&priv->hs_jack.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hs_jack.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } -- 1.8.3.2

3 2