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

[alsa-devel] [PATCH 0/4] ASoC: core: Update for DSP systems
by Vinod Koul 04 Jan '16

04 Jan '16

This patch series tried to do two things 1. Add support to have cpu based loopback dai_links supported, so the CPU based DSPs can have non hostless streams 2. Add kcontrol to bytes tlv callbacks and fix the topology core info callback for TLV byte controls Comments welcome... Jeeja KP (1): ASoC: core: Adds support for cpu loopback dai_link Mythri P K (1): ASoC: core: Pass kcontrol to bytes tlv callbacks Omair M Abdullah (1): ASoC: topology: fix info callback for TLV byte control Vinod Koul (1): ASoC: core: refactor soc_link_dai_widgets() include/sound/soc-topology.h | 6 ++++-- include/sound/soc.h | 9 +++++++-- sound/soc/soc-core.c | 38 +++++++++++++++++++++++++------------- sound/soc/soc-ops.c | 4 ++-- sound/soc/soc-topology.c | 2 +- 5 files changed, 39 insertions(+), 20 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ALSA: dummy: constify dummy_timer_ops structures
by Julia Lawall 30 Dec '15

30 Dec '15

The dummy_timer_ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/drivers/dummy.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 016e451..75b7485 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -351,7 +351,7 @@ static void dummy_systimer_free(struct snd_pcm_substream *substream) kfree(substream->runtime->private_data); } -static struct dummy_timer_ops dummy_systimer_ops = { +static const struct dummy_timer_ops dummy_systimer_ops = { .create = dummy_systimer_create, .free = dummy_systimer_free, .prepare = dummy_systimer_prepare, @@ -475,7 +475,7 @@ static void dummy_hrtimer_free(struct snd_pcm_substream *substream) kfree(dpcm); } -static struct dummy_timer_ops dummy_hrtimer_ops = { +static const struct dummy_timer_ops dummy_hrtimer_ops = { .create = dummy_hrtimer_create, .free = dummy_hrtimer_free, .prepare = dummy_hrtimer_prepare,

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[alsa-devel] [PATCH] ALSA: cs5535audio: constify cs5535audio_dma_ops structures
by Julia Lawall 30 Dec '15

30 Dec '15

The cs5535audio_dma_ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/pci/cs5535audio/cs5535audio_pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index 9c2dc91..27fa57d 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -402,7 +402,7 @@ static struct snd_pcm_ops snd_cs5535audio_capture_ops = { .pointer = snd_cs5535audio_pcm_pointer, }; -static struct cs5535audio_dma_ops snd_cs5535audio_playback_dma_ops = { +static const struct cs5535audio_dma_ops snd_cs5535audio_playback_dma_ops = { .type = CS5535AUDIO_DMA_PLAYBACK, .enable_dma = cs5535audio_playback_enable_dma, .disable_dma = cs5535audio_playback_disable_dma, @@ -412,7 +412,7 @@ static struct cs5535audio_dma_ops snd_cs5535audio_playback_dma_ops = { .read_dma_pntr = cs5535audio_playback_read_dma_pntr, }; -static struct cs5535audio_dma_ops snd_cs5535audio_capture_dma_ops = { +static const struct cs5535audio_dma_ops snd_cs5535audio_capture_dma_ops = { .type = CS5535AUDIO_DMA_CAPTURE, .enable_dma = cs5535audio_capture_enable_dma, .disable_dma = cs5535audio_capture_disable_dma,

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[alsa-devel] [PATCH] ALSA: atiixp: constify atiixp_dma_ops structures
by Julia Lawall 30 Dec '15

30 Dec '15

The atiixp_dma_ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/pci/atiixp.c | 6 +++--- sound/pci/atiixp_modem.c | 4 ++-- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 1028fc8..2ce0022 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1219,7 +1219,7 @@ static struct ac97_pcm atiixp_pcm_defs[] = { }, }; -static struct atiixp_dma_ops snd_atiixp_playback_dma_ops = { +static const struct atiixp_dma_ops snd_atiixp_playback_dma_ops = { .type = ATI_DMA_PLAYBACK, .llp_offset = ATI_REG_OUT_DMA_LINKPTR, .dt_cur = ATI_REG_OUT_DMA_DT_CUR, @@ -1228,7 +1228,7 @@ static struct atiixp_dma_ops snd_atiixp_playback_dma_ops = { .flush_dma = atiixp_out_flush_dma, }; -static struct atiixp_dma_ops snd_atiixp_capture_dma_ops = { +static const struct atiixp_dma_ops snd_atiixp_capture_dma_ops = { .type = ATI_DMA_CAPTURE, .llp_offset = ATI_REG_IN_DMA_LINKPTR, .dt_cur = ATI_REG_IN_DMA_DT_CUR, @@ -1237,7 +1237,7 @@ static struct atiixp_dma_ops snd_atiixp_capture_dma_ops = { .flush_dma = atiixp_in_flush_dma, }; -static struct atiixp_dma_ops snd_atiixp_spdif_dma_ops = { +static const struct atiixp_dma_ops snd_atiixp_spdif_dma_ops = { .type = ATI_DMA_SPDIF, .llp_offset = ATI_REG_SPDF_DMA_LINKPTR, .dt_cur = ATI_REG_SPDF_DMA_DT_CUR, diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 27ed678..c534552 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -970,7 +970,7 @@ static struct snd_pcm_ops snd_atiixp_capture_ops = { .pointer = snd_atiixp_pcm_pointer, }; -static struct atiixp_dma_ops snd_atiixp_playback_dma_ops = { +static const struct atiixp_dma_ops snd_atiixp_playback_dma_ops = { .type = ATI_DMA_PLAYBACK, .llp_offset = ATI_REG_MODEM_OUT_DMA1_LINKPTR, .dt_cur = ATI_REG_MODEM_OUT_DMA1_DT_CUR, @@ -979,7 +979,7 @@ static struct atiixp_dma_ops snd_atiixp_playback_dma_ops = { .flush_dma = atiixp_out_flush_dma, }; -static struct atiixp_dma_ops snd_atiixp_capture_dma_ops = { +static const struct atiixp_dma_ops snd_atiixp_capture_dma_ops = { .type = ATI_DMA_CAPTURE, .llp_offset = ATI_REG_MODEM_IN_DMA_LINKPTR, .dt_cur = ATI_REG_MODEM_IN_DMA_DT_CUR,

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[alsa-devel] [PATCH 1/2] ASoC: rt5645: use polling to support HS button
by Bard Liao 30 Dec '15

30 Dec '15

The IRQ pin will keep high when the headset button is pressed. And keep low when the headset button is released. So, we need irq trigger at both edges. However, some platform can't support it. Therefore, we polling the register to report the button release event once a button presse event is received. To support the headset button detection function for those can't support both edges trigger platforms, we also need to invert the polarity of jack detection irq since we need to keep the IRQ pin low in normal case. Signed-off-by: John Lin <john.lin(a)realtek.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 26 ++++++++++++++++++++++++-- 1 file changed, 24 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3e8d666..57c8d9e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -405,6 +405,7 @@ struct rt5645_priv { struct delayed_work jack_detect_work, rcclock_work; struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)]; struct rt5645_eq_param_s *eq_param; + struct timer_list btn_check_timer; int codec_type; int sysclk; @@ -3130,7 +3131,7 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) } if (rt5645->pdata.jd_invert) regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, - RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); + RT5645_JD_1_1_MASK, RT5645_JD_1_1_NOR); } else { /* jack out */ rt5645->jack_type = 0; @@ -3151,7 +3152,7 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) snd_soc_dapm_sync(dapm); if (rt5645->pdata.jd_invert) regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, - RT5645_JD_1_1_MASK, RT5645_JD_1_1_NOR); + RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); } return rt5645->jack_type; @@ -3275,6 +3276,12 @@ static void rt5645_jack_detect_work(struct work_struct *work) } if (btn_type == 0)/* button release */ report = rt5645->jack_type; + else { + if (rt5645->pdata.jd_invert) { + mod_timer(&rt5645->btn_check_timer, + msecs_to_jiffies(100)); + } + } break; /* jack out */ @@ -3317,6 +3324,14 @@ static irqreturn_t rt5645_irq(int irq, void *data) return IRQ_HANDLED; } +static void rt5645_btn_check_callback(unsigned long data) +{ + struct rt5645_priv *rt5645 = (struct rt5645_priv *)data; + + queue_delayed_work(system_power_efficient_wq, + &rt5645->jack_detect_work, msecs_to_jiffies(5)); +} + static int rt5645_probe(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); @@ -3783,6 +3798,13 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } } + if (rt5645->pdata.jd_invert) { + regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, + RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); + setup_timer(&rt5645->btn_check_timer, + rt5645_btn_check_callback, (unsigned long)rt5645); + } + INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work); -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 1/2] ASoC: Use nested lock for snd_soc_dapm_mutex_lock
by Charles Keepax 30 Dec '15

30 Dec '15

snd_soc_dapm_mutex_lock currently uses the un-nested call which can cause lockdep warnings when called from control handlers (a relatively common usage) and using modules. As creating the control causes a potential mutex inversion with the handler, creating the control will take the controls_rwsem under the dapm_mutex and accessing the control will take the dapm_mutex under controls_rwsem. All the users look like they want to be using the runtime class of the lock anyway, so this patch just changes snd_soc_dapm_mutex_lock to use the nested call, with the SND_SOC_DAPM_CLASS_RUNTIME class. Fixes: f6d5e586b416 ("ASoC: dapm: Add helpers to lock/unlock DAPM mutex") Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a2deeb9..aae55d0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1687,7 +1687,7 @@ extern const struct dev_pm_ops snd_soc_pm_ops; /* Helper functions */ static inline void snd_soc_dapm_mutex_lock(struct snd_soc_dapm_context *dapm) { - mutex_lock(&dapm->card->dapm_mutex); + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); } static inline void snd_soc_dapm_mutex_unlock(struct snd_soc_dapm_context *dapm) -- 2.1.4

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[alsa-devel] [PATCH] ASoC: rsnd: fix usrcnt decrementing bug
by Andrzej Hajda 30 Dec '15

30 Dec '15

Field usrcnt is unsigned so it cannot be lesser than zero. The problem has been detected using proposed semantic patch scripts/coccinelle/tests/unsigned_lesser_than_zero.cocci [1]. [1]: http://permalink.gmane.org/gmane.linux.kernel/2038576 Signed-off-by: Andrzej Hajda <a.hajda(a)samsung.com> --- sound/soc/sh/rcar/ssi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 0b91692..8ca30fd 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -381,9 +381,9 @@ rsnd_ssi_quit_end: rsnd_mod_power_off(mod); - ssi->usrcnt--; - - if (ssi->usrcnt < 0) + if (ssi->usrcnt > 0) + ssi->usrcnt--; + else dev_err(dev, "%s[%d] usrcnt error\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: arizona: Exit startup early if no runtime
by Charles Keepax 30 Dec '15

30 Dec '15

commit 9b8ef9f6b3fc ("ASoC: dapm: Add startup & shutdown for dai_links") Added support for calling startup on CODEC to CODEC links, however this is called with a NULL runtime pointer. There isn't really a sensible way to pass a valid runtime pointer to a CODEC to CODEC link at the moment, so we need to make the startup function safe for NULL runtimes. This patch returns from the Arizona startup function early if there is no runtime. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- I will fixup the other users as well if everyone is happy with this approach? Thanks, Charles sound/soc/codecs/arizona.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 38a73e3..88e2c74 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1494,6 +1494,9 @@ static int arizona_startup(struct snd_pcm_substream *substream, const struct snd_pcm_hw_constraint_list *constraint; unsigned int base_rate; + if (!substream->runtime) + return 0; + switch (dai_priv->clk) { case ARIZONA_CLK_SYSCLK: base_rate = priv->sysclk; -- 2.1.4

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[alsa-devel] [PATCH 0/5 v4] ALSA: dice: improve card registration processing
by Takashi Sakamoto 30 Dec '15

30 Dec '15

Hi, This patchset updates my previous one: [alsa-devel] [PATCH 0/5 v3] ALSA: dice: improve card registration processing http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/102501.h… The patch 02 included a bug of missing 'end' tag. This v4 patchset includes fixes of the bug. Additionally, this patchset adds to output information message when registration work encounters any errors. Furthermore, I use mod_delayed_work() to queue and postpone work for sound card registration. I've reported system freeze in previous patchset, but I had a mistake not to use any actual workqueus as a first argument of the function. Takashi Sakamoto (5): ALSA: dice: split subaddress check from category check ALSA: dice: postpone card registration ALSA: dice: ensure to release sound devices after sound card registration fails ALSA: dice: purge transaction initialization at timeout of Dice notification ALSA: dice: expand timeout to wait for Dice notification sound/firewire/dice/dice-midi.c | 2 + sound/firewire/dice/dice-pcm.c | 2 + sound/firewire/dice/dice-transaction.c | 122 ++++++++++++-------- sound/firewire/dice/dice.c | 197 +++++++++++++++++---------------- sound/firewire/dice/dice.h | 5 + 5 files changed, 189 insertions(+), 139 deletions(-) -- 2.5.0

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[alsa-devel] Rawmidi
by Daniel Viksporre 29 Dec '15

29 Dec '15

I cant compile example programs that use rawmidi. When I look on my harddrive, I have many rawmidi.h files, but since 2015 they are all 0 bytes. Can that be the reason I cant compile. But why is those files 0 bytes from the repository Im using? I get the following error messages... test.c:(.text+0x507): undefined reference to `snd_rawmidi_open' test.c:(.text+0x5d8): undefined reference to `snd_rawmidi_open' test.c:(.text+0x75e): undefined reference to `snd_rawmidi_read' test.c:(.text+0x846): undefined reference to `snd_rawmidi_write' test.c:(.text+0x867): undefined reference to `snd_rawmidi_write' test.c:(.text+0x888): undefined reference to `snd_rawmidi_write' test.c:(.text+0x894): undefined reference to `snd_rawmidi_drain' test.c:(.text+0x8c1): undefined reference to `snd_rawmidi_write' test.c:(.text+0x8e2): undefined reference to `snd_rawmidi_write' test.c:(.text+0x903): undefined reference to `snd_rawmidi_write' test.c:(.text+0x90f): undefined reference to `snd_rawmidi_drain' test.c:(.text+0xa71): undefined reference to `snd_rawmidi_read' test.c:(.text+0xae0): undefined reference to `snd_rawmidi_write' test.c:(.text+0xaec): undefined reference to `snd_rawmidi_drain' test.c:(.text+0xb8f): undefined reference to `snd_rawmidi_drain' test.c:(.text+0xb9b): undefined reference to `snd_rawmidi_close' test.c:(.text+0xbaf): undefined reference to `snd_rawmidi_drain' test.c:(.text+0xbbb): undefined reference to `snd_rawmidi_close'

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