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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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[alsa-devel] [PATCH] ASoC: arizona: Exit startup early if no runtime
by Charles Keepax 29 Dec '15

29 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, this is perfectly safe as all the startup function does is set the PCM constraints for user-space which arn't relevant to a CODEC to CODEC link anyway. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- 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 fb8f188..33143fe 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 v2 1/2] ALSA: hda - Add mic mute hotkey quirk for Lenovo ThinkCentre AIO
by Hui Wang 29 Dec '15

29 Dec '15

From: Kailang <kailang(a)realtek.com> The Lenovo ThinkCenter AIO uses Line2 (NID 0x1b) to implement the micmute hotkey, here we register an input device and use Line2 unsol event to collect the hotkey pressing or releasing. In the meanwhile, the micmute led is controlled by GPIO2, so we use an existing function alc_fixup_gpio_mic_mute_hook() to control the led. [Hui: And there are two places to register the input device, to make the code simple and clean, move the two same code sections into a function.] Cc: <stable(a)vger.kernel.org> Signed-off-by: Kailang <kailang(a)realtek.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 84 ++++++++++++++++++++++++++++++++++++------- 1 file changed, 71 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fe96428..257b839 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3468,6 +3468,29 @@ static void gpio2_mic_hotkey_event(struct hda_codec *codec, input_sync(spec->kb_dev); } +static int alc_register_micmute_input_device(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->kb_dev = input_allocate_device(); + if (!spec->kb_dev) { + codec_err(codec, "Out of memory (input_allocate_device)\n"); + return -ENOMEM; + } + spec->kb_dev->name = "Microphone Mute Button"; + spec->kb_dev->evbit[0] = BIT_MASK(EV_KEY); + spec->kb_dev->keybit[BIT_WORD(KEY_MICMUTE)] = BIT_MASK(KEY_MICMUTE); + + if (input_register_device(spec->kb_dev)) { + codec_err(codec, "input_register_device failed\n"); + input_free_device(spec->kb_dev); + spec->kb_dev = NULL; + return -ENOMEM; + } + + return 0; +} + static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3485,20 +3508,8 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->kb_dev = input_allocate_device(); - if (!spec->kb_dev) { - codec_err(codec, "Out of memory (input_allocate_device)\n"); + if (alc_register_micmute_input_device(codec) != 0) return; - } - spec->kb_dev->name = "Microphone Mute Button"; - spec->kb_dev->evbit[0] = BIT_MASK(EV_KEY); - spec->kb_dev->keybit[BIT_WORD(KEY_MICMUTE)] = BIT_MASK(KEY_MICMUTE); - if (input_register_device(spec->kb_dev)) { - codec_err(codec, "input_register_device failed\n"); - input_free_device(spec->kb_dev); - spec->kb_dev = NULL; - return; - } snd_hda_add_verbs(codec, gpio_init); snd_hda_codec_write_cache(codec, codec->core.afg, 0, @@ -3528,6 +3539,47 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, } } +static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* Line2 = mic mute hotkey + GPIO2 = mic mute LED */ + static const struct hda_verb gpio_init[] = { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x04 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 }, + {} + }; + + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + if (alc_register_micmute_input_device(codec) != 0) + return; + + snd_hda_add_verbs(codec, gpio_init); + snd_hda_jack_detect_enable_callback(codec, 0x1b, + gpio2_mic_hotkey_event); + + spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook; + spec->gpio_led = 0; + spec->mute_led_polarity = 0; + spec->gpio_mic_led_mask = 0x04; + return; + } + + if (!spec->kb_dev) + return; + + switch (action) { + case HDA_FIXUP_ACT_PROBE: + spec->init_amp = ALC_INIT_DEFAULT; + break; + case HDA_FIXUP_ACT_FREE: + input_unregister_device(spec->kb_dev); + spec->kb_dev = NULL; + } +} + static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4628,6 +4680,7 @@ enum { ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, ALC293_FIXUP_LENOVO_SPK_NOISE, + ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, }; static const struct hda_fixup alc269_fixups[] = { @@ -5237,6 +5290,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI }, + [ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc233_fixup_lenovo_line2_mic_hotkey, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5386,6 +5443,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), + SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 1.9.1

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[alsa-devel] [PATCH 1/2] ALSA: hda - Add mic mute hotkey quirk for Lenovo ThinkCentre AIO
by Hui Wang 28 Dec '15

28 Dec '15

From: Kailang <kailang(a)realtek.com> The Lenovo ThinkCenter AIO uses Line2 (NID 0x1b) to implement the micmute hotkey, here we register an input device and use Line2 unsol event to collect the hotkey pressing or releasing. In the meanwhile, the micmute led is controlled by GPIO2, so we use an existing function alc_fixup_gpio_mic_mute_hook() to control the led. Cc: <stable(a)vger.kernel.org> Signed-off-by: Kailang <kailang(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 59 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 59 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fe96428..8adff4f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3528,6 +3528,59 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, } } +static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* Line2 = mic mute hotkey + GPIO2 = mic mute LED */ + static const struct hda_verb gpio_init[] = { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x04 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 }, + {} + }; + + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->kb_dev = input_allocate_device(); + if (!spec->kb_dev) { + codec_err(codec, "Out of memory (input_allocate_device)\n"); + return; + } + spec->kb_dev->name = "Microphone Mute Button"; + spec->kb_dev->evbit[0] = BIT_MASK(EV_KEY); + spec->kb_dev->keybit[BIT_WORD(KEY_MICMUTE)] = BIT_MASK(KEY_MICMUTE); + if (input_register_device(spec->kb_dev)) { + codec_err(codec, "input_register_device failed\n"); + input_free_device(spec->kb_dev); + spec->kb_dev = NULL; + return; + } + + snd_hda_add_verbs(codec, gpio_init); + snd_hda_jack_detect_enable_callback(codec, 0x1b, + gpio2_mic_hotkey_event); + + spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook; + spec->gpio_led = 0; + spec->mute_led_polarity = 0; + spec->gpio_mic_led_mask = 0x04; + return; + } + + if (!spec->kb_dev) + return; + + switch (action) { + case HDA_FIXUP_ACT_PROBE: + spec->init_amp = ALC_INIT_DEFAULT; + break; + case HDA_FIXUP_ACT_FREE: + input_unregister_device(spec->kb_dev); + spec->kb_dev = NULL; + } +} + static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4628,6 +4681,7 @@ enum { ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, ALC293_FIXUP_LENOVO_SPK_NOISE, + ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, }; static const struct hda_fixup alc269_fixups[] = { @@ -5237,6 +5291,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI }, + [ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc233_fixup_lenovo_line2_mic_hotkey, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5386,6 +5444,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), + SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 1.9.1

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[alsa-devel] [PATCH] ASoC: sun4i-codec: Use proper output for external amp routes
by Hans de Goede 27 Dec '15

27 Dec '15

An external amp (if any) is connected to the external outputs of the SoC of course, rather then directly to the internal amp. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/sunxi/sun4i-codec.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index e6cc6a1..44f170c 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -728,7 +728,8 @@ static const struct snd_soc_dapm_widget sun4i_codec_card_dapm_widgets[] = { }; static const struct snd_soc_dapm_route sun4i_codec_card_dapm_routes[] = { - { "Speaker", NULL, "Power Amplifier" }, + { "Speaker", NULL, "HP Right" }, + { "Speaker", NULL, "HP Left" }, }; static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) -- 2.5.0

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[alsa-devel] [PATCH 0/5 v3] ALSA: dice: improve card registration processing
by Takashi Sakamoto 25 Dec '15

25 Dec '15

Hi, This patchset updates my previous post. [alsa-devel] [PATCH 0/4 v2] ALSA: dice: improve card registration processing http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/102460.h… The previous patchset includes some bugs. * Mutex structure can be destroyed within critical section with it in releasing sound card. * Bus reset handler doesn't execute transaction re-initialization as long as sound card instance is not registered. * Work of sound card registration is not postpone even if bus reset occurs. * Devices in sound card are not released when sound card registration fails. This version fixes these bugs, according to discussions between Takash Iwai and Stefan Richter. Thanks for them. Changes: * Use delayed workqueue API correctly. * Wait for registration work in .remove callback, instead of sound card free callback. * Obsolete mutex_lock()/mutex_unlock() for sound card registration purpose. * Keep devices of sound card and release them at registration failure. In kerneldoc of 'mod_delayed_work', this function call has a side effect to queueing the work. It seems that this function can be used instead of schedule_delayed_work. https://www.kernel.org/doc/htmldocs/device-drivers/API-mod-delayed-work.html But as long as I tested, my Linux 4.2 system is freezed. I have no further investigation. I use both of schedule_delayed_work() and mod_delayed_work() in this patchset. 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 | 199 +++++++++++++++++---------------- sound/firewire/dice/dice.h | 5 + 5 files changed, 191 insertions(+), 139 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH 0/4 v2] ALSA: dice: improve card registration processing
by Takashi Sakamoto 25 Dec '15

25 Dec '15

Hi, This patchset is to update my previous one: [alsa-devel] [PATCH 0/4] ALSA: dice: improve card registration processing http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/102357.h… The main purpose of this patchset is to postpone card registration after successive bus resets, to ensure the processing and related transactions. This change has an advantage for ALSA dice driver to get proper information about hardware, especially for Dice II based models. Changes from v1: * fix jiffies64 handling bug * rename some members and functions related to card registration. * 'struct snd_dice.probed' -> 'registered' * do_probe() -> do_registration() Takashi Sakamoto (4): ALSA: dice: split subaddress check from category check ALSA: dice: postpone card registration ALSA: dice: purge transaction initialization at timeout of Dice notification ALSA: dice: expand timeout to wait for Dice notification sound/firewire/dice/dice-transaction.c | 122 ++++++++++++++-------- sound/firewire/dice/dice.c | 183 ++++++++++++++++----------------- sound/firewire/dice/dice.h | 3 + 3 files changed, 172 insertions(+), 136 deletions(-) -- 2.5.0

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[alsa-devel] [RFC v2][PATCH 00/11] ALSA: fireface: new driver for RME Fireface series
by Takashi Sakamoto 24 Dec '15

24 Dec '15

Hi, This patchset is to update my previous one with mostly full features as an driver in ALSA firewire stack: [alsa-devel] [RFC][PATCH 0/3] ALSA: fireface: new driver for RME Fireface series (MIDI only) http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/101576.h… Changes from RFCv1: * Support own quirk of packet streaming * Add own data block processing layer * Support PCM functionality * Add proc node to help debugging * Add hwdep interface I note that just powering on, the device is muted. Thus, the device generates no sound even if packet streaming is successful. You can de-mute it by write transaction. For example, by using 'firewire-request' command in linux-firewire-utils(https://github.com/cladisch/linux-firewire-utils): $ ./firewire-request /dev/fw1 write 0x0000801c0000 0x000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000001000000010000000100000001000000 Currently, libffado gives no way to do it via its mixer interface. Instead, done in methods to start streaming. I think it better to move de-mute codes to mixer initialization. There's an issue of periodical hissing noise. I observed this with Fireface 400. Interval of the noise differs depending on the situation (10-20 minutes) and continues around 20 seconds. The cause is not clear but I've experienced similar issue with TASCAM FireOne. A common point between these two models is non-blocking transmission. I guess that the cause may be on data block calculation in AMDTP packet streaming layer (calculate_data_blocks() function in sound/firewire/amdtp-stream.c). Further investigation is required. In my previous RFC, I described that zero bits have no effect in write transactions to 0x00008010051c. But this is wrong. The transaction has side effect to set the other options. In this patchset, the state of clock is changed on the way to initialize transaction. This is a bit rude but unavoidable. Fortunately, it's software's responsibility to read from flash memory and set initial configuration, thus I think this change is approval because just powering on the state of device is against user's configuration. The loading and configuring are still userspace responsibility. MIDI functionality can be disabled when any userspace applications run with libffado, as I described. This is a bug of libffado due to node ID handling. In line 199 of libffado/src/rme/fireface_hw.cpp, node id is handled as uint16_t value. This is a fashion of libffado to represent Node ID by lower 6 bits of actual Node ID of IEEE 1394 bus. In the other lines, actual Node Id is constructed with 0xffc0. Just using the raw value returned by ConfigRom::getNodeId() is invalid. For this work, I can borrow test device (Fireface 400) from Syntax Japan Inc. It's my fortune to have an opportunity to contact to the company. I'm graceful to Takeshi Mitsuhashi, Tahiro Hashizume and their kindness. Thank you. Takashi Sakamoto (11): fireface: add skeleton for RME Fireface series fireface: postpone sound card registration fireface: add model specific structure fireface: add transaction support fireface: add support for MIDI functionality fireface: add proc node to help debugging firewire-lib: add no-header packet processing fireface: add data processing layer fireface: add stream management functionality fireface: add PCM functionality fireface: add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 1 + sound/firewire/Kconfig | 7 + sound/firewire/Makefile | 1 + sound/firewire/amdtp-stream.c | 104 +++++-- sound/firewire/amdtp-stream.h | 2 + sound/firewire/fireface/Makefile | 4 + sound/firewire/fireface/amdtp-ff.c | 221 ++++++++++++++ sound/firewire/fireface/fireface-hwdep.c | 191 ++++++++++++ sound/firewire/fireface/fireface-midi.c | 131 ++++++++ sound/firewire/fireface/fireface-pcm.c | 390 ++++++++++++++++++++++++ sound/firewire/fireface/fireface-proc.c | 237 +++++++++++++++ sound/firewire/fireface/fireface-stream.c | 397 +++++++++++++++++++++++++ sound/firewire/fireface/fireface-transaction.c | 301 +++++++++++++++++++ sound/firewire/fireface/fireface.c | 210 +++++++++++++ sound/firewire/fireface/fireface.h | 131 ++++++++ 16 files changed, 2313 insertions(+), 18 deletions(-) create mode 100644 sound/firewire/fireface/Makefile create mode 100644 sound/firewire/fireface/amdtp-ff.c create mode 100644 sound/firewire/fireface/fireface-hwdep.c create mode 100644 sound/firewire/fireface/fireface-midi.c create mode 100644 sound/firewire/fireface/fireface-pcm.c create mode 100644 sound/firewire/fireface/fireface-proc.c create mode 100644 sound/firewire/fireface/fireface-stream.c create mode 100644 sound/firewire/fireface/fireface-transaction.c create mode 100644 sound/firewire/fireface/fireface.c create mode 100644 sound/firewire/fireface/fireface.h -- 2.5.0

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[alsa-devel] [PATCH] ASoC: wm8960: add kcontrol to select ADC data output
by Zidan Wang 24 Dec '15

24 Dec '15

add kcontrol to select ADC data output. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 5380798..162def8 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -147,6 +147,12 @@ static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"}; static const char *wm8960_3d_lower_cutoff[] = {"Low", "High"}; static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"}; static const char *wm8960_alcmode[] = {"ALC", "Limiter"}; +static const char *wm8960_adc_data_output_sel[] = { + "Left Data = Left ADC; Right Data = Right ADC", + "Left Data = Left ADC; Right Data = Left ADC", + "Left Data = Right ADC; Right Data = Right ADC", + "Left Data = Right ADC; Right Data = Left ADC", +}; static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity), @@ -155,6 +161,7 @@ static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_3D, 5, 2, wm8960_3d_lower_cutoff), SOC_ENUM_SINGLE(WM8960_ALC1, 7, 4, wm8960_alcfunc), SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode), + SOC_ENUM_SINGLE(WM8960_ADDCTL1, 2, 4, wm8960_adc_data_output_sel), }; static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; @@ -295,6 +302,8 @@ SOC_SINGLE_TLV("Right Output Mixer Boost Bypass Volume", WM8960_BYPASS2, 4, 7, 1, bypass_tlv), SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume", WM8960_ROUTMIX, 4, 7, 1, bypass_tlv), + +SOC_ENUM("ADC Data Output Select", wm8960_enum[6]), }; static const struct snd_kcontrol_new wm8960_lin_boost[] = { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: twl6040, fsl: use to_platform_device
by Geliang Tang 24 Dec '15

24 Dec '15

Use to_platform_device() instead of open-coding it. Signed-off-by: Geliang Tang <geliangtang(a)163.com> --- sound/soc/codecs/twl6040.c | 3 +-- sound/soc/fsl/mpc8610_hpcd.c | 3 +-- sound/soc/fsl/p1022_ds.c | 3 +-- sound/soc/fsl/p1022_rdk.c | 3 +-- 4 files changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 4cad892..bc3de2e 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1097,8 +1097,7 @@ static int twl6040_probe(struct snd_soc_codec *codec) { struct twl6040_data *priv; struct twl6040 *twl6040 = dev_get_drvdata(codec->dev->parent); - struct platform_device *pdev = container_of(codec->dev, - struct platform_device, dev); + struct platform_device *pdev = to_platform_device(codec->dev); int ret = 0; priv = devm_kzalloc(codec->dev, sizeof(*priv), GFP_KERNEL); diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 6f236f1..ddf49f3 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -189,8 +189,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) { struct device *dev = pdev->dev.parent; /* ssi_pdev is the platform device for the SSI node that probed us */ - struct platform_device *ssi_pdev = - container_of(dev, struct platform_device, dev); + struct platform_device *ssi_pdev = to_platform_device(dev); struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; struct mpc8610_hpcd_data *machine_data; diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 747aab0..a1f780e 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -199,8 +199,7 @@ static int p1022_ds_probe(struct platform_device *pdev) { struct device *dev = pdev->dev.parent; /* ssi_pdev is the platform device for the SSI node that probed us */ - struct platform_device *ssi_pdev = - container_of(dev, struct platform_device, dev); + struct platform_device *ssi_pdev = to_platform_device(dev); struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; struct machine_data *mdata; diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c index 1dd49e5..d4d88a8 100644 --- a/sound/soc/fsl/p1022_rdk.c +++ b/sound/soc/fsl/p1022_rdk.c @@ -203,8 +203,7 @@ static int p1022_rdk_probe(struct platform_device *pdev) { struct device *dev = pdev->dev.parent; /* ssi_pdev is the platform device for the SSI node that probed us */ - struct platform_device *ssi_pdev = - container_of(dev, struct platform_device, dev); + struct platform_device *ssi_pdev = to_platform_device(dev); struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; struct machine_data *mdata; -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: arizona: Fix bclk for sample rates that are multiple of 4kHz
by Charles Keepax 24 Dec '15

24 Dec '15

From: Nikesh Oswal <Nikesh.Oswal(a)cirrus.com> For a sample rate of 12kHz the bclk was taken from the 44.1kHz table as we test for a multiple of 8kHz. This patch fixes this issue by testing for multiples of 4kHz instead. Signed-off-by: Nikesh Oswal <Nikesh.Oswal(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 88e2c74..33143fe 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1659,7 +1659,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, bool reconfig; unsigned int aif_tx_state, aif_rx_state; - if (params_rate(params) % 8000) + if (params_rate(params) % 4000) rates = &arizona_44k1_bclk_rates[0]; else rates = &arizona_48k_bclk_rates[0]; -- 2.1.4

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[alsa-devel] Applied "ASoC: da7219: Add regmap patch to support old silicon" to the asoc tree
by Mark Brown 24 Dec '15

24 Dec '15

The patch ASoC: da7219: Add regmap patch to support old silicon has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From abd7c894fc41a9a674354e10ed6c55413e1db077 Mon Sep 17 00:00:00 2001 From: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Date: Wed, 23 Dec 2015 13:50:04 +0000 Subject: [PATCH] ASoC: da7219: Add regmap patch to support old silicon Initial silicon did not have master bias enabled by default, unlike later HW, so use regmap patch to align with newer defaults. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/da7219.c | 26 +++++++++++++++++++++++++- 1 file changed, 25 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index c6d3b32..9c7e8ec 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1592,9 +1592,14 @@ static void da7219_handle_pdata(struct snd_soc_codec *codec) } } +static struct reg_sequence da7219_rev_aa_patch[] = { + { DA7219_REFERENCES, 0x08 }, +}; + static int da7219_probe(struct snd_soc_codec *codec) { struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + unsigned int rev; int ret; mutex_init(&da7219->lock); @@ -1604,6 +1609,26 @@ static int da7219_probe(struct snd_soc_codec *codec) if (ret) return ret; + ret = regmap_read(da7219->regmap, DA7219_CHIP_REVISION, &rev); + if (ret) { + dev_err(codec->dev, "Failed to read chip revision: %d\n", ret); + goto err_disable_reg; + } + + switch (rev & DA7219_CHIP_MINOR_MASK) { + case 0: + ret = regmap_register_patch(da7219->regmap, da7219_rev_aa_patch, + ARRAY_SIZE(da7219_rev_aa_patch)); + if (ret) { + dev_err(codec->dev, "Failed to register AA patch: %d\n", + ret); + goto err_disable_reg; + } + break; + default: + break; + } + /* Handle DT/Platform data */ if (codec->dev->of_node) da7219->pdata = da7219_of_to_pdata(codec); @@ -1774,7 +1799,6 @@ static struct reg_default da7219_reg_defaults[] = { { DA7219_MIXOUT_R_CTRL, 0x10 }, { DA7219_CHIP_ID1, 0x23 }, { DA7219_CHIP_ID2, 0x93 }, - { DA7219_CHIP_REVISION, 0x00 }, { DA7219_IO_CTRL, 0x00 }, { DA7219_GAIN_RAMP_CTRL, 0x00 }, { DA7219_PC_COUNT, 0x02 }, -- 2.6.4

1 0

[alsa-devel] [PATCH v2] ASoC: fsl_asrc: old version asrc only support even numbered channels
by Zidan Wang 24 Dec '15

24 Dec '15

The latest asrc's channel bits is 4 bit, the old asrc's channel bits is 3. For the old version asrc, channel should be a even number. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v1->v2: 1. add an extra else if for the new condition. sound/soc/fsl/fsl_asrc.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index be78dc9..16ff156 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -116,6 +116,10 @@ static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair) } else if (asrc_priv->channel_avail < channels) { dev_err(dev, "can't afford required channels: %d\n", channels); ret = -EINVAL; + } else if (asrc_priv->channel_bits < 4 && channels % 2 != 0) { + dev_err(dev, "old version asrc can't support an odd \ + channel number\n"); + ret = -EINVAL; } else { asrc_priv->channel_avail -= channels; asrc_priv->pair[index] = pair; -- 1.9.1

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[alsa-devel] [PATCH 0/4] ALSA: dice: improve card registration processing
by Takashi Sakamoto 23 Dec '15

23 Dec '15

Hi, This patchset is to update a part of my previous RFCv2 (patch 01, 03 and 04. 02 is dropped according to Clemens' comment). [alsa-devel] [RFC][PATCH 00/10 v2] ALSA: dice: stabiliza packet streaming http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/101897.h… The main purpose of this patchset is to postpone card registration after successive bus resets, to ensure probe processing and related transactions. This change has effects for ALSA dice driver to get proper information about hardware, especially for Dice II based models. Changes from RFC v2: * Patch 03 is newly added to drop a workaround to ensure transactions under a situation of probe processing with successive bus reset. * Comment updates. Takashi Sakamoto (4): ALSA: dice: split subaddress check from category check ALSA: dice: postpone card registration ALSA: dice: purge transaction initialization at timeout of Dice notification ALSA: dice: expand timeout to wait for Dice notification sound/firewire/dice/dice-transaction.c | 122 +++++++++++++++-------- sound/firewire/dice/dice.c | 177 ++++++++++++++++----------------- sound/firewire/dice/dice.h | 3 + 3 files changed, 166 insertions(+), 136 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH 1/1] ASoC: da7219: Add regmap patch to support old silicon
by Adam Thomson 23 Dec '15

23 Dec '15

Initial silicon did not have master bias enabled by default, unlike later HW, so use regmap patch to align with newer defaults. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- This patch is based on the following commit: 'ASoC: da7219: Remove support for 32KHz PLL mode' (Commit 501f72e9c5205b9d70d5d61e9b186ae7ba873f73, kernel/git/broonie/sound.git) sound/soc/codecs/da7219.c | 26 +++++++++++++++++++++++++- 1 file changed, 25 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 82a7236..341a191 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1592,9 +1592,14 @@ static void da7219_handle_pdata(struct snd_soc_codec *codec) } } +static struct reg_sequence da7219_rev_aa_patch[] = { + { DA7219_REFERENCES, 0x08 }, +}; + static int da7219_probe(struct snd_soc_codec *codec) { struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + unsigned int rev; int ret; mutex_init(&da7219->lock); @@ -1604,6 +1609,26 @@ static int da7219_probe(struct snd_soc_codec *codec) if (ret) return ret; + ret = regmap_read(da7219->regmap, DA7219_CHIP_REVISION, &rev); + if (ret) { + dev_err(codec->dev, "Failed to read chip revision: %d\n", ret); + goto err_disable_reg; + } + + switch (rev & DA7219_CHIP_MINOR_MASK) { + case 0: + ret = regmap_register_patch(da7219->regmap, da7219_rev_aa_patch, + ARRAY_SIZE(da7219_rev_aa_patch)); + if (ret) { + dev_err(codec->dev, "Failed to register AA patch: %d\n", + ret); + goto err_disable_reg; + } + break; + default: + break; + } + /* Handle DT/Platform data */ if (codec->dev->of_node) da7219->pdata = da7219_of_to_pdata(codec); @@ -1774,7 +1799,6 @@ static struct reg_default da7219_reg_defaults[] = { { DA7219_MIXOUT_R_CTRL, 0x10 }, { DA7219_CHIP_ID1, 0x23 }, { DA7219_CHIP_ID2, 0x93 }, - { DA7219_CHIP_REVISION, 0x00 }, { DA7219_IO_CTRL, 0x00 }, { DA7219_GAIN_RAMP_CTRL, 0x00 }, { DA7219_PC_COUNT, 0x02 }, -- 1.9.3

1 0

[alsa-devel] DT soundcard driver with special clock routing
by Petr Kulhavy 23 Dec '15

23 Dec '15

Hi, I'm developing a DT-based driver for an ARM SoC device with the following special clock routing: - WM8758 codec attached to the CPU DAI via I2S - the codec is I2S clock master - the CPU feeds the codec with 12MHz clock (outside of the I2S) - the codec converts the 12MHz internally to 48kHz*256 or 44.1kHz *256 clock and provides it to I2S as MCLK So the special set-up needed here is that the codec needs to be configured to: a) generate the proper MCLK b) enable the respective GPIO pin as clock output What is the best way of representing this set-up using device tree? I'm wondering if this can be done by adding a specific option in the codec DT binding for enabling the clock output and then use the simple audio card. Or do I need to write a specific soundcard driver that sets up the codec registers? I would like to avoid that if possible and use existing code... What would you recommend? Thanks -- Petr Kulhavy, MSc System Architect *BARIX* petr(a)barix.com <mailto:petr@barix.com> | Skype: brain.barix Barix AG, Seefeldstrasse 303 | 8008 Zurich, Switzerland T +41 43 43322 11 | www.barix.com <http://www.barix.com> You have received this email because of your relationship Barix AG and its affiliated companies. Barix AG and its affiliated companies do not sell or exchange email addresses, or any other personal contact information provided by you with any third parties. All email distributions are managed and controlled by Barix AG and its affiliated companies. Barix AG, Seefeldstr. 303, 8008 Zürich, Switzerland. Company Reg. No: CH-020.3.023.869-8, VAT Reg. No: CHE-105.687.663.

2 5

[alsa-devel] [PATCH] ASoC: rt5651: add ACPI and OF support
by Bard Liao 23 Dec '15

23 Dec '15

Add required tables and the binding document for ACPI and OF matching. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5651.txt | 41 ++++++++++++++++++++++ sound/soc/codecs/rt5651.c | 31 ++++++++++++++++ 2 files changed, 72 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5651.txt diff --git a/Documentation/devicetree/bindings/sound/rt5651.txt b/Documentation/devicetree/bindings/sound/rt5651.txt new file mode 100644 index 0000000..3875233 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5651.txt @@ -0,0 +1,41 @@ +RT5651 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5651". + +- reg : The I2C address of the device. + +Optional properties: + +- realtek,in2-differential + Boolean. Indicate MIC2 input are differential, rather than single-ended. + +- realtek,dmic-en + Boolean. true if dmic is used. + +Pins on the device (for linking into audio routes) for RT5651: + + * DMIC L1 + * DMIC R1 + * IN1P + * IN2P + * IN2N + * IN3P + * HPOL + * HPOR + * LOUTL + * LOUTR + * PDML + * PDMR + +Example: + +codec: rt5651@1a { + compatible = "realtek,rt5651"; + reg = <0x1a>; + realtek,dmic-en = "true"; + realtek,in2-diff = "false"; +}; diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 1d40318..7a61970 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -18,6 +18,7 @@ #include <linux/regmap.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/acpi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -1735,12 +1736,38 @@ static const struct regmap_config rt5651_regmap = { .num_ranges = ARRAY_SIZE(rt5651_ranges), }; +#if defined(CONFIG_OF) +static const struct of_device_id rt5651_of_match[] = { + { .compatible = "realtek,rt5651", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5651_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt5651_acpi_match[] = { + { "10EC5651", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt5651_acpi_match); +#endif + static const struct i2c_device_id rt5651_i2c_id[] = { { "rt5651", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5651_i2c_id); +static int rt5651_parse_dt(struct rt5651_priv *rt5651, struct device_node *np) +{ + rt5651->pdata.in2_diff = of_property_read_bool(np, + "realtek,in2-differential"); + rt5651->pdata.dmic_en = of_property_read_bool(np, + "realtek,dmic-en"); + + return 0; +} + static int rt5651_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1757,6 +1784,8 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, if (pdata) rt5651->pdata = *pdata; + else if (i2c->dev.of_node) + rt5651_parse_dt(rt5651, i2c->dev.of_node); rt5651->regmap = devm_regmap_init_i2c(i2c, &rt5651_regmap); if (IS_ERR(rt5651->regmap)) { @@ -1806,6 +1835,8 @@ static int rt5651_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt5651_i2c_driver = { .driver = { .name = "rt5651", + .acpi_match_table = ACPI_PTR(rt5651_acpi_match), + .of_match_table = of_match_ptr(rt5651_of_match), }, .probe = rt5651_i2c_probe, .remove = rt5651_i2c_remove, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 0/6] ASoC: da7219: AB Silicon related updates and small fixes.
by Adam Thomson 23 Dec '15

23 Dec '15

This is a small collection of updates & fixes, put together for ease. For changes relating to AB silicon, previous silicon revision is obsolete and was unavailable outside of Dialog, so does not require driver support in the future. Changes are based on v4.4-rc6 kernel. Adam Thomson (6): ASoC: da7219: Disable regulators on probe() failure ASoC: da7219: Fix Sidetone to work regardless of DAI capture ASoC: da7219: Update REFERENCES reg default, in-line with HW ASoC: da7219: Remove internal LDO features of codec ASoC: da7219: Add support for 1.6V micbias level ASoC: da7219: Remove support for 32KHz PLL mode Documentation/devicetree/bindings/sound/da7219.txt | 8 +- include/sound/da7219.h | 14 +--- sound/soc/codecs/da7219.c | 87 +++++++--------------- sound/soc/codecs/da7219.h | 9 --- 4 files changed, 32 insertions(+), 86 deletions(-) -- 1.9.3

4 17

[alsa-devel] [PATCH v4 1/2] ASoC: wm_adsp: Add a handler for the compressed IRQ
by Charles Keepax 23 Dec '15

23 Dec '15

Here support is added for responding to DSP IRQs that are used to indicate data being available on the DSP. The idea is that we check the amount of data available upon receipt of an IRQ and on subsequent calls to the pointer callback we recheck once less than one fragment is available (to avoid excessive SPI traffic), if there is truely less than one fragment available we ack the last IRQ and wait for a new one. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v3: - Make spurious IRQs return IRQ_NONE - Add a comment to explain the usage of wm_adsp_buffer_ack_irq Thanks, Charles sound/soc/codecs/wm5110.c | 27 +++++++ sound/soc/codecs/wm_adsp.c | 192 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 3 + 3 files changed, 222 insertions(+) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c364096..dde94c4 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2177,10 +2177,23 @@ static int wm5110_open(struct snd_compr_stream *stream) return wm_adsp_compr_open(&priv->core.adsp[n_adsp], stream); } +static irqreturn_t wm5110_adsp2_irq(int irq, void *data) +{ + struct wm5110_priv *florida = data; + int ret; + + ret = wm_adsp_compr_handle_irq(&florida->core.adsp[2]); + if (ret == -ENODEV) + return IRQ_NONE; + + return IRQ_HANDLED; +} + static int wm5110_codec_probe(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->core.arizona; int i, ret; priv->core.arizona->dapm = dapm; @@ -2189,6 +2202,14 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_gpio(codec); arizona_init_mono(codec); + ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, + "ADSP2 Compressed IRQ", wm5110_adsp2_irq, + priv); + if (ret != 0) { + dev_err(codec->dev, "Failed to request DSP IRQ: %d\n", ret); + return ret; + } + for (i = 0; i < WM5110_NUM_ADSP; ++i) { ret = wm_adsp2_codec_probe(&priv->core.adsp[i], codec); if (ret) @@ -2209,12 +2230,15 @@ err_adsp2_codec_probe: for (--i; i >= 0; --i) wm_adsp2_codec_remove(&priv->core.adsp[i], codec); + arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv); + return ret; } static int wm5110_codec_remove(struct snd_soc_codec *codec) { struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->core.arizona; int i; for (i = 0; i < WM5110_NUM_ADSP; ++i) @@ -2222,6 +2246,8 @@ static int wm5110_codec_remove(struct snd_soc_codec *codec) priv->core.arizona->dapm = NULL; + arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv); + return 0; } @@ -2273,6 +2299,7 @@ static struct snd_compr_ops wm5110_compr_ops = { .set_params = wm_adsp_compr_set_params, .get_caps = wm_adsp_compr_get_caps, .trigger = wm_adsp_compr_trigger, + .pointer = wm_adsp_compr_pointer, }; static struct snd_soc_platform_driver wm5110_compr_platform = { diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ac879d1..edf96b5 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -279,6 +279,11 @@ struct wm_adsp_compr_buf { struct wm_adsp_buffer_region *regions; u32 host_buf_ptr; + + u32 error; + u32 irq_ack; + int read_index; + int avail; }; struct wm_adsp_compr { @@ -287,6 +292,8 @@ struct wm_adsp_compr { struct snd_compr_stream *stream; struct snd_compressed_buffer size; + + unsigned int copied_total; }; #define WM_ADSP_DATA_WORD_SIZE 3 @@ -2436,6 +2443,11 @@ static int wm_adsp_compr_check_params(struct snd_compr_stream *stream, return -EINVAL; } +static inline unsigned int wm_adsp_compr_frag_words(struct wm_adsp_compr *compr) +{ + return compr->size.fragment_size / WM_ADSP_DATA_WORD_SIZE; +} + int wm_adsp_compr_set_params(struct snd_compr_stream *stream, struct snd_compr_params *params) { @@ -2622,6 +2634,8 @@ static int wm_adsp_buffer_init(struct wm_adsp *dsp) return -ENOMEM; buf->dsp = dsp; + buf->read_index = -1; + buf->irq_ack = 0xFFFFFFFF; ret = wm_adsp_buffer_locate(buf); if (ret < 0) { @@ -2705,6 +2719,16 @@ int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) ret); break; } + + /* Trigger the IRQ at one fragment of data */ + ret = wm_adsp_buffer_write(compr->buf, + HOST_BUFFER_FIELD(high_water_mark), + wm_adsp_compr_frag_words(compr)); + if (ret < 0) { + adsp_err(dsp, "Failed to set high water mark: %d\n", + ret); + break; + } break; case SNDRV_PCM_TRIGGER_STOP: break; @@ -2719,4 +2743,172 @@ int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) } EXPORT_SYMBOL_GPL(wm_adsp_compr_trigger); +static inline int wm_adsp_buffer_size(struct wm_adsp_compr_buf *buf) +{ + int last_region = wm_adsp_fw[buf->dsp->fw].caps->num_regions - 1; + + return buf->regions[last_region].cumulative_size; +} + +static int wm_adsp_buffer_update_avail(struct wm_adsp_compr_buf *buf) +{ + u32 next_read_index, next_write_index; + int write_index, read_index, avail; + int ret; + + /* Only sync read index if we haven't already read a valid index */ + if (buf->read_index < 0) { + ret = wm_adsp_buffer_read(buf, + HOST_BUFFER_FIELD(next_read_index), + &next_read_index); + if (ret < 0) + return ret; + + read_index = sign_extend32(next_read_index, 23); + + if (read_index < 0) { + adsp_dbg(buf->dsp, "Avail check on unstarted stream\n"); + return 0; + } + + buf->read_index = read_index; + } + + ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(next_write_index), + &next_write_index); + if (ret < 0) + return ret; + + write_index = sign_extend32(next_write_index, 23); + + avail = write_index - buf->read_index; + if (avail < 0) + avail += wm_adsp_buffer_size(buf); + + adsp_dbg(buf->dsp, "readindex=0x%x, writeindex=0x%x, avail=%d\n", + buf->read_index, write_index, avail); + + buf->avail = avail; + + return 0; +} + +int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) +{ + struct wm_adsp_compr_buf *buf = dsp->buffer; + int ret = 0; + + mutex_lock(&dsp->pwr_lock); + + if (!buf) { + adsp_err(dsp, "Spurious buffer IRQ\n"); + ret = -ENODEV; + goto out; + } + + adsp_dbg(dsp, "Handling buffer IRQ\n"); + + ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(error), &buf->error); + if (ret < 0) { + adsp_err(dsp, "Failed to check buffer error: %d\n", ret); + goto out; + } + if (buf->error != 0) { + adsp_err(dsp, "Buffer error occurred: %d\n", buf->error); + ret = -EIO; + goto out; + } + + ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(irq_count), + &buf->irq_ack); + if (ret < 0) { + adsp_err(dsp, "Failed to get irq_count: %d\n", ret); + goto out; + } + + ret = wm_adsp_buffer_update_avail(buf); + if (ret < 0) { + adsp_err(dsp, "Error reading avail: %d\n", ret); + goto out; + } + +out: + mutex_unlock(&dsp->pwr_lock); + + return ret; +} +EXPORT_SYMBOL_GPL(wm_adsp_compr_handle_irq); + +/* + * We only need to ack the IRQ on the firmware end when we wish to receive + * another, as such we do this in response to running out of data in the + * buffer, rather than in direct response to the IRQ itself. + */ +static int wm_adsp_buffer_ack_irq(struct wm_adsp_compr_buf *buf) +{ + if (buf->irq_ack & 0x01) + return 0; + + adsp_dbg(buf->dsp, "Acking buffer IRQ(0x%x)\n", buf->irq_ack); + + buf->irq_ack |= 0x01; + + return wm_adsp_buffer_write(buf, HOST_BUFFER_FIELD(irq_ack), + buf->irq_ack); +} + +int wm_adsp_compr_pointer(struct snd_compr_stream *stream, + struct snd_compr_tstamp *tstamp) +{ + struct wm_adsp_compr *compr = stream->runtime->private_data; + struct wm_adsp_compr_buf *buf = compr->buf; + struct wm_adsp *dsp = compr->dsp; + int ret = 0; + + adsp_dbg(dsp, "Pointer request\n"); + + mutex_lock(&dsp->pwr_lock); + + if (!compr->buf) { + ret = -ENXIO; + goto out; + } + + if (compr->buf->error) { + ret = -EIO; + goto out; + } + + if (buf->avail < wm_adsp_compr_frag_words(compr)) { + ret = wm_adsp_buffer_update_avail(buf); + if (ret < 0) { + adsp_err(dsp, "Error reading avail: %d\n", ret); + goto out; + } + + /* + * If we really have less than 1 fragment available ack the + * last DSP IRQ and rely on the IRQ to inform us once a whole + * fragment is available. + */ + if (buf->avail < wm_adsp_compr_frag_words(compr)) { + ret = wm_adsp_buffer_ack_irq(buf); + if (ret < 0) { + adsp_err(dsp, "Failed to ack buffer IRQ: %d\n", + ret); + goto out; + } + } + } + + tstamp->copied_total = compr->copied_total; + tstamp->copied_total += buf->avail * WM_ADSP_DATA_WORD_SIZE; + +out: + mutex_unlock(&dsp->pwr_lock); + + return ret; +} +EXPORT_SYMBOL_GPL(wm_adsp_compr_pointer); + MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 43af093..522fa1a 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -112,5 +112,8 @@ extern int wm_adsp_compr_set_params(struct snd_compr_stream *stream, extern int wm_adsp_compr_get_caps(struct snd_compr_stream *stream, struct snd_compr_caps *caps); extern int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd); +extern int wm_adsp_compr_handle_irq(struct wm_adsp *dsp); +extern int wm_adsp_compr_pointer(struct snd_compr_stream *stream, + struct snd_compr_tstamp *tstamp); #endif -- 2.1.4

1 1

[alsa-devel] [PATCH 52/54] sound: soc: wm8903: Be sure to clamp return value
by Linus Walleij 23 Dec '15

23 Dec '15

As we want gpio_chip .get() calls to be able to return negative error codes and propagate to drivers, we need to go over all drivers and make sure their return values are clamped to [0,1]. We do this by using the ret = !!(val) design pattern. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Mark: as mentioned in 00/54: either apply this directly or ACK it and I will take it into the GPIO tree. --- sound/soc/codecs/wm8903.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 2512def0d349..a82b8bc2cfc0 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1804,7 +1804,7 @@ static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) regmap_read(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, &reg); - return (reg & WM8903_GP1_LVL_MASK) >> WM8903_GP1_LVL_SHIFT; + return !!((reg & WM8903_GP1_LVL_MASK) >> WM8903_GP1_LVL_SHIFT); } static int wm8903_gpio_direction_out(struct gpio_chip *chip, -- 2.4.3

3 2

[alsa-devel] [GIT PULL] ASoC updates for v4.5
by Mark Brown 23 Dec '15

23 Dec '15

The following changes since commit 4ef7675344d687a0ef5b0d7c0cee12da005870c0: Linux 4.4-rc6 (2015-12-20 16:06:09 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.5 for you to fetch changes up to 822ad70a2f5c420da5baa9f4354e6b7813ca6da9: Merge remote-tracking branch 'asoc/topic/rt5677' into asoc-next (2015-12-23 00:38:14 +0000) ---------------------------------------------------------------- ASoC: Updates for v4.5 This is quite a busy release on the driver front with a lot of new drivers being added but comparatively quiet on the core side with only one big change going in and that a fairly straightforward refactoring. - Conversion of the array of DAI links to a list by Mengdong Lin, supporting dynamically adding and removing DAI links. - Some more fixes for the topology code, though it is still not final and ready for enabling in production. We really need to get to the point where that can be done. - A pile of changes for Intel SkyLake drivers which hopefully deliver some useful initial functionality for systems with this chipset, though there is more work still to come. - New drivers for a number of Imagination Technologies IPs. - Lots of new features and cleanups for the Renesas drivers. - ANC support for WM5110. - New driver for Atmel class D speaker drivers. - New drivers for Cirrus CS47L24 and WM1831. - New driver for Dialog DA7128. - New drivers for Realtek RT5659 and RT56156. - New driver for Rockchip RK3036. - New driver for TI PC3168A ---------------------------------------------------------------- Adam Thomson (9): ASoC: da7218: Add bindings documentation for DA7218 audio codec ASoC: da7218: Add da7218 codec driver ASoC: da7218: Enable mic level detection reporting to user-space ASoC: da7219: Fix Sidetone to work regardless of DAI capture ASoC: da7219: Disable regulators on probe() failure ASoC: da7219: Update REFERENCES reg default, in-line with HW ASoC: da7219: Remove internal LDO features of codec ASoC: da7219: Add support for 1.6V micbias level ASoC: da7219: Remove support for 32KHz PLL mode Anatol Pomozov (1): ASoC: rt5677: use 'active low' logic for reset pin Arnd Bergmann (8): ASoC: samsung: pass DMA channels as pointers ASoC: s3c24xx-i2s: pass DMA channels as platform data ASoC: pxa: remove incorrect do_div() call ASoC: sh: fix fsi build warnings for 64 bit ASoC: samsung: pass filter function as pointer ASoC: rcar: remove unused variable ASoC: fsl: use correct format string for dma_addr_t ASoC: da7218: avoid 64-bit compile warning Axel Lin (3): ASoC: da7219: Use logical instead of bitwise OR for boolean expression ASoC: wm8904: Make undocumented registers non-readable ASoC: rt5616: Return error if device ID mismatch Bard Liao (7): ASoC: rt298: fix remove unnedded clk setting ASoC: rt5645: Separate regmap for rt5645 and rt5650 ASoC: rt5659: add rt5659 codec driver ASoC: rt5645: merge DMI tables of google projects ASoC: rt5616: add rt5616 codec driver ASoC: rt5616: rename some alsa control names ASoC: rt5677: set PLL_CTRL2 non-volatile Ben Zhang (1): ASoC: rt5677: Reconfigure PLL1 after resume Caesar Wang (5): ASoC: rockchip: i2s: change bclk and lrck according to sample rates ASoC: rockchip-max98090: Allow more sample rates ASoC: rockchip-rt5645: Allow more sample rates ASoC: rt5616: add an of_match table ASoC: rt5616: add devicetree document for rt5616 Charles Keepax (17): ASoC: wm5110: Add DAPM/routing hookup for the ANC block ASoC: wm_adsp: Expand the list of available firmwares ASoC: arizona: Add 32uS delay after putting FLL into freerun ASoC: dapm: Make enable/disable_pin work with always on widgets ASoC: arizona: Correct types of mixer texts and values ASoC: arizona: Fix type of clock rate pointer in arizona_set_sysclk ASoC: wm_adsp: Replace debugfs lock with more general DSP power lock ASoC: wm_adsp: Add power lock for firmware change control ASoC: wm_adsp: Fixup some minor formatting and checkpatch errors ASoC: wm_adsp: Add locking to DSP firmware controls ASoC: wm_adsp: Mimic legacy behaviour of reading controls when DSP is on ASoC: wm5110: Provide basic hookup for voice control ASoC: wm_adsp: Factor out finding the location of an algorithm region ALSA: compress: Add SND_AUDIOCODEC_BESPOKE ASoC: wm_adsp: Add support for opening a compressed stream ASoC: wm_adsp: Add code to locate and initialise compressed buffer ASoC: wm_adsp: Attach buffers and streams together Damien.Horsley (17): ASoC: img: Add binding document for I2S input controller ASoC: img: Add driver for I2S input controller ASoC: img: Add binding document for I2S output controller ASoC: img: Add driver for I2S output controller ASoC: img: Add binding document for parallel output controller ASoC: img: Add driver for parallel output controller ASoC: img: Add binding document for SPDIF input controller ASoC: img: Add driver for SPDIF input controller ASoC: img: Add documentation for SPDIF in controls ASoC: img: Add binding document for SPDIF output controller ASoC: img: Add driver for SPDIF output controller ASoC: img: parallel out: Add missing initialiser ASoC: Add SOC_DOUBLE_STS macro ASoC: pcm3168a: Add binding document for pcm3168a codec ASoC: pcm3168a: Add driver for pcm3168a codec ASoC: img: Add binding document for Pistachio internal DAC ASoC: img: Add driver for Pistachio internal DAC Dan Carpenter (3): ASoC: Intel: pass correct parameter in sst_alloc_stream_mrfld() ASoC: Intel: Skylake: Fix a couple signedness bugs ASoC: Intel: Skylake: pointer math issue Dharageswari R (2): ASoC: Intel: Skylake: Add support for Loadable modules ASoC: Intel: Skylake: Add support for Mic Select module Fang, Yang A (3): ASoC: Intel: Skylake: add adi + nau8825 machine driver entry ASoc: Intel: boards: fix dapm map of nau88l25_ssm4567 machine ASoc: Intel: boards: Add HDMI/DP links for nau88l25_ssm4567 machine Geert Uytterhoeven (2): ASoC: rsnd: Add missing initialization of ADG req_rate ASoC: rsrc-card: Clarify compatible value Hans de Goede (3): ASoC: sun4i-codec: Rename codec dapm widgets and routes ASoC: sun4i-codec: Add support for PA gpio pin ASoC: sun4i-codec: Use proper output for external amp routes Hardik T Shah (3): ASoC: Intel: Skylake: Add multiple pin formats ASoC: Intel: Skylake: Update the topology interface structure ASoC: Intel: Skylake: Add support for module GUIDs Harsha Priya (1): ASoC: Intel: Skylake: Adding nau88l25+ssm4567 machine driver Jean-Michel Hautbois (1): ASoC: sgtl5000: fix VAG power up timing Jeeja KP (39): ASoC: Intel: Skylake: Fix to correct check for non DSP widget ASoC: Intel: Skylake: Fix not to ignore return value in be hw_params ASoC: Intel: Skylake: Fix to add 32 bit in update FE params ASoC: Intel: Skylake: Fix to ignore codec_mask check in probe ASoC: Intel: Skylake: Fix to ignore blob check if link type is HDA ASoC: Intel: Skylake: Fix support for multiple pins in a module ASoC: Intel: Skylake: Fix bit depth when querying the NHLT blob ASoC: Intel: Skylake: use module_pin info for unbind ASoC: Intel: Skylake: Add support for virtual dsp widgets ASoC: Intel: Skylake: Fix DSP pipe underrun/overrun issue ASoC: Intel: Skylake: Fix to remove be copier widget power check ASoC: Intel: Skylake: Ignore rate check for DMIC link ASoC: Intel: Skylake: Fix to remove channel_map calculation ASoC: Intel: Skylake: Fix PM behaviour ASoC: Intel: Skylake: Fix the SSP0 Fmt fixup to 24 bit ASoC: Intel: Skylake: Add support for SSP1 BE cpu dai ASoC: Intel: Skylake: remove pm_runtime_get/put calls ASoC: Intel: Skylake: Don't enable WAKEENABLE on suspend ASoC: Intel: Skylake: Remove redundant init in resume ASoC: Intel: Skylake: Fix cleanup of dma buffer ASoC: Intel: Skylake: Reset the DSP when set D3 fails ASoC: Intel: Skylake: Fix CLDMA buffer wrap case ASoC: Intel: Skylake: Fix null ptr dereferenced in skl_tplg_bind_sinks ASoC: Intel: Skylake: Add pm ops for skl_rt286 machine ASoC: Intel: Skylake: Reconfigure HDA stream register in prepare/resume ASoC: Intel: Skylake: Fix to update bit depth for module params ASoC: Intel: Skylake: Update pcm capability ASoC: Intel: Skylake: Poll CLDMA RUN bit when set ASoC: Intel: Skylake: Update DMIC DAIs and capabilities ASoC: Intel: Skylake: Add helper routines to handle module params ASoC: Intel: Skylake: Add helper routine to handle Algo parameter ASoC: Intel: Skylake: Add support to configure module params ASoC: Intel: Skylake: Add tlv byte kcontrols ASoC: Intel: Skylake: Add memory pages to widget data. ASoC: Intel: Skylake: Fix module init data correctly ASoC: Intel: Skylake: Add dai link for DMIC capture ASoC: Intel: Skylake: Add support for active suspend ASoC: Intel: Skylake: Clear stream registers before stream setup ASoC: Intel: Skylake: Fix to set pipe state to invalid when deleting Jiada Wang (1): ASoC: wm8962: set ALC2 as non-volatile register Jie Yang (2): ASoC: Intel: sst: only select sst-firmware when DW DMAC is built-in ASoC: Intel: sst: fix sst_memcpy32 wrong with non-4x bytes issue John Keeping (4): ASoC: rockchip: i2s: separate capture and playback ASoC: rockchip: i2s: remove unused variables ASoC: es8328: Fix deemphasis values ASoC: es8328: Fix shifts for mixer switches Julia Lawall (4): ASoC: Intel: constify sst_block_ops structures ASoC: imx-pcm-dma: add NULL test ASoC: omap-hdmi-audio: add NULL test ASoC: Intel: add NULL test Koro Chen (4): ASoC: dpcm: Make BE prepare possible in suspend state ASoC: mediatek: Move 22M/24M clock control into I2S ops ASoC: mediatek: Use current HW pointer for pointer callback ASoC: mediatek: Turn AFE on/off in runtime resume/suspend Kuninori Morimoto (82): ASoC: rsnd: DMA become SSI/SRC member ASoC: rsnd: DMA related definition goes to dma.c ASoC: rsnd: rename rsnd_dma_init() to rsnd_dma_attach() ASoC: rsnd: enable to use rsnd_dai_connect() from each mod ASoC: rsnd: remove all modules when PIO fallback ASoC: rsnd: fixup rsnd_dma_of_path method for mod base common method ASoC: rsnd: move rsnd_src_ssi_irq_enable/disable() to ssi.c ASoC: rsnd: disable SRC.out only when stop timing ASoC: rsnd: rsnd_dai_stream has each mod's status insted of rsnd_mod ASoC: rsnd: Don't stop HW even if a large number of errors occur ASoC: rsnd: avoid pointless loop in rsnd_mod_interrupt() ASoC: rsnd: use common rsnd_ssi_status_xxx() ASoC: rsnd: use mod base common method on DMA phase1 ASoC: rsnd: use mod base common method on DMA phase2 ASoC: rsnd: use mod base common method on DMA phase3 ASoC: rsnd: use mod base common method on CMD ASoC: rsnd: use mod base common method on SSIU ASoC: rsnd: use mod base common method on SSI-parent ASoC: rsnd: call rsnd_src_quit() from rsnd_src_quit_gen2() ASoC: rsnd: fixup rsnd_dmapp_stop() return value ASoC: rsnd: audio_clkout0/1/2/3 are optional properties ASoC: rsnd: move CMD related operation to cmd.c ASoC: rsnd: DVC settings matches to datasheet ASoC: rsnd: MIX settings matches to datasheet ASoC: rsnd: tidyup comment position of rsnd_mod_xxx ASoC: rsnd: call clk_prepare()/clk_enable() for AUDIO_CLKx ASoC: ak4613: add single-end optional property for IN/OUT pins ASoC: ak4613: tidyup CTRL1 value selection method ASoC: rsnd: SND_SOC_RCAR doesn't depend on DMA_OF ASoC: rsnd: ssi: 24bit data needs right-aligned settings ASoC: ak4613: don't overwrite CTRL2 register ASoC: rsnd: remove Gen1 support from SRC ASoC: rsnd: cleanup RSND_REG_xxx ASoC: rsnd: SRC settings matches to datasheet ASoC: rsnd: remove platform boot support from core.c ASoC: rsnd: remove platform boot support from ssi.c ASoC: rsnd: remove platform boot support from src.c ASoC: rsnd: remove platform boot support from ctu.c ASoC: rsnd: remove platform boot support from mix.c ASoC: rsnd: remove platform boot support from dvc.c ASoC: rsnd: remove platform boot support from gen.c ASoC: rsnd: remove struct rsnd_of_data ASoC: rsnd: remove struct rcar_snd_info ASoC: rsnd: remove struct platform_device from probe/remove parameter ASoC: ak4613: tidyup vendor prefix from ak4613 to asahi-kasei ASoC: rsnd: tidyup void* cast for 64bit compiler ASoC: rsnd: indicate register name for debug ASoC: rsnd: tidyup semantics of rsnd_src_record_error() ASoC: rsnd: tidyup semantics of rsnd_ssi_record_error() ASoC: rsnd: don't use normal *mod in adg.c ASoC: rsnd: add missing ADINR::CHNUM on DVC/SRC/SSIU ASoC: rsnd: add missing SRC_O_BUSIF_MODE register ASoC: rsnd: src: rename rsnd_src_soft_reset() to rsnd_src_activation() ASoC: rsnd: mix: rename rsnd_mix_soft_reset() to rsnd_mix_activation() ASoC: rsnd: dvc: rename rsnd_dvc_soft_reset() to rsnd_dvc_activation() ASoC: rsnd: src: add rsnd_src_halt() ASoC: rsnd: mix: add rsnd_mix_halt() ASoC: rsnd: dvc: add rsnd_dvc_halt() ASoC: rsnd: add rsnd_ssi_config_init() ASoC: rsnd: set SSIWSR setting on rsnd_ssi_config_init() ASoC: rsnd: ssi enables non-stereo sound ASoC: rsnd: dvc enables non-stereo sound ASoC: rsnd: add TDM Extend Mode support ASoC: rsnd: fixup wrong snd_soc_dai_driver pointer access ASoC: rsnd: rsrc-card: check return value of snd_soc_of_get_dai_name() ASoC: rsnd: care SWSP bit for TDM/non-TDM ASoC: rsnd: tidyup data align position for capture ASoC: rsrc-card: enable to use tdm_slot on DT ASoC: rsrc-card: Remove support for setting differing DAI formats ASoC: rsrc-card: tidyup dai format for DPCM ASoC: rsnd: fixup SSIU control timing ASoC: rsnd: tidyup return value of rsnd_get_adinr_bit() ASoC: rsnd: tidyup debug print position on rsnd_dma_attach() ASoC: rsnd: rsnd_dai_connect() returns error if it connect to existing mod ASoC: rsnd: SSI/SSIU use rsnd_get_slot_extend() to check TDM ASoC: rsnd: remove rsnd_get_slot_runtime() ASoC: rsnd: tidyup rsnd_get_slot_xxx() naming ASoC: rsnd: add rsnd_set_slot() / rsnd_get_slot_num() ASoC: rsnd: add rsnd_parse_connect_common() and remove complex macro ASoC: rsnd: add missing DT example for Simple Card ASoC: rsnd: add missing DT example for Simple Card with TDM ASoC: rsnd: add Multi channel support Liam Girdwood (1): ASoC: Intel: sst: fix the IRQ locked issue Linus Walleij (2): ASoC: wm8903: Be sure to clamp return value ASoC: ac97: Be sure to clamp return value Maciej S. Szmigiero (1): ASoC: fsl-asoc-card: use different route map for AC'97 mode Mans Rullgard (3): ASoC: wm8974: set cache type for regmap ASoC: wm8974: add devicetree support ASoC: wm8974: add binding for WM8974 codec Mark Brown (37): Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into HEAD Merge branch 'fix/arizona' of git://git.kernel.org/.../broonie/sound into asoc-arizona Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-rt5645 Merge branch 'topic/pcm-list' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'topic/hdac' of git://git.kernel.org/.../broonie/sound into asoc-intel ASoC: hdac: Fix Makefile and Kconfig sorting Merge branch 'topic/fsl-sai' of git://git.kernel.org/.../broonie/sound into asoc-fsl Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-rt5645 Merge branch 'topic/sink' of git://git.kernel.org/.../broonie/sound into asoc-dapm Merge branch 'fix/sun4i-codec' of git://git.kernel.org/.../broonie/sound into asoc-sunxi Merge branch 'topic/sink' of git://git.kernel.org/.../broonie/sound into asoc-intel rcar: ctu: Avoid use of ret uninitialised rcar: dvc: Avoid use of ret uninitialised rcar: mux: Avoid use of ret uninitialised Merge branch 'topic/arizona' of git://git.kernel.org/.../broonie/sound into asoc-adsp Merge branch 'topic/cs47l24' of git://git.kernel.org/.../broonie/sound into asoc-adsp Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/es8328', 'asoc/fix/fsl-sai', 'asoc/fix/rockchip', 'asoc/fix/sgtl5000' and 'asoc/fix/wm8974' into asoc-linus Merge remote-tracking branch 'asoc/topic/arizona' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm-list' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm3168a' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branch 'asoc/topic/sunxi' into asoc-next Merge remote-tracking branches 'asoc/topic/ac97', 'asoc/topic/adsp', 'asoc/topic/ak4613' and 'asoc/topic/atmel' into asoc-next Merge remote-tracking branches 'asoc/topic/atmel-classd', 'asoc/topic/const' and 'asoc/topic/da7218' into asoc-next Merge remote-tracking branches 'asoc/topic/da7219', 'asoc/topic/dai-link' and 'asoc/topic/doc' into asoc-next Merge remote-tracking branches 'asoc/topic/dpcm', 'asoc/topic/dwc', 'asoc/topic/fsl', 'asoc/topic/fsl-asrc' and 'asoc/topic/fsl-esai' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl-spdif', 'asoc/topic/img' and 'asoc/topic/intel' into asoc-next Merge remote-tracking branches 'asoc/topic/kcontrol', 'asoc/topic/max98357a' and 'asoc/topic/mtk' into asoc-next Merge remote-tracking branches 'asoc/topic/pxa', 'asoc/topic/qcom', 'asoc/topic/rcar', 'asoc/topic/rk3036' and 'asoc/topic/rockchip' into asoc-next Merge remote-tracking branches 'asoc/topic/rt286', 'asoc/topic/rt5616' and 'asoc/topic/rt5677' into asoc-next Merge remote-tracking branches 'asoc/topic/samsung', 'asoc/topic/sh', 'asoc/topic/ssm2518' and 'asoc/topic/sti' into asoc-next Merge remote-tracking branches 'asoc/topic/tegra', 'asoc/topic/topology', 'asoc/topic/wm8903' and 'asoc/topic/wm8904' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8962', 'asoc/topic/wm8974' and 'asoc/topic/wm9713' into asoc-next Merge tag 'asoc-fix-v4.4-rc6' into asoc-linus Merge tag 'asoc-v4.5' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5677' into asoc-next Markus Elfring (1): ASoC: ssm2518: Use a signed return type for ssm2518_lookup_mcs() Maruthi Srinivas Bayyavarapu (3): ASoC: dwc: add runtime suspend/resume functionality ASoC: dwc: add quirk for different register offset ASoC: dwc: reconfigure dwc in 'resume' from 'suspend' Maxime Ripard (3): ASoC: sun4i: Implement MIC1 capture ASoC: sunxi: Remove useless comments and variable ASoC: sun4i-codec: pass through clk_set_rate error Mengdong Lin (9): ASoC: topology: ABI - Rename dai_elems to pcm_elems in manifest ASoC: Vendor drivers get a link's runtime by snd_soc_get_pcm_runtime() ASoC: Change the PCM runtime array to a list ASoC: Define soc_init_dai_link() to wrap link intialization. ASoC: Change 2nd argument of soc_bind_dai_link() to DAI link pointer ASoC: Implement DAI links in a list & define API to add/remove a link ASoC: Define add/remove_dai_link ops for a soc card ASoC: soc_bind_dai_link() directly returns success for a bound DAI link ASoC: Bind new DAI links after probing components Moise Gergaud (1): ASoC: sti: set iec958 channel status sampling freq Mousami Jana (1): ASoC: Intel: Skylake: add LARGE_CONFIG_GET IPC support Mythri P K (1): ASoC: core: Pass kcontrol to bytes tlv callbacks Oder Chiou (1): ASoC: rt5645: Use the mod_delayed_work instead of the queue_delayed_work and cancel_delayed_work_sync Omair M Abdullah (2): ASoC: Intel: Skylake: update mailbox uplink window offset and size ASoC: Intel: Skylake: read params from DSP if module is on PC Liao (1): ASoC: dpcm: Apply symmetry for DPCM Peter Ujfalusi (1): ASoC: davinci-mcasp: Fix XDATA check in mcasp_start_tx Pierre-Louis Bossart (9): ASoC: Intel: boards: align pin names between byt-rt5640 drivers ASoC: Intel: boards: start merging byt-rt5640 drivers ASoC: Intel: boards: merge DMI-based quirks in bytcr-rt5640 driver ASoC: Intel: tag byt-rt5640 machine driver as deprecated ASoc: Intel: Atom: add deep buffer definitions for atom platforms ASoC: Intel: boards: add DEEP_BUFFER support for BYT/CHT/BSW ASoC: Intel: Atom: add 24-bit support for media playback and capture ASoC: Intel: Atom: clean-up compressed DAI definition ASoC: Intel: Atom: flip logic for gain Switch Praveen Diwakar (1): ASoC: Intel: Skylake: Update ignore suspend for rt286 machine Richard Fitzgerald (3): ASoC: wm8998: Remove duplicated consts ASoC: cs47l24: Add driver for Cirrus Logic CS47L24 and WM1831 codecs ASoC: arizona: In arizona_calc_fratio make new codecs the default case Robert Jarzmik (3): ASoC: wm9713: convert to regmap ASoC: wm9713: use snd_soc_*() calls to update ac97 registers ASoC: ac97: add gpio chip Rohit Ainapure (3): ASoC: max98357a: Add ACPI ID for Maxim ASoC: Intel: Add Nuvoton+Maxim machine driver entry ASoC: Intel: Skylake: Add Nuvoton Maxim machine driver Sathyanarayana Nujella (1): ASoc: Intel: boards: Add WOV as sink for nau88l25_ssm4567 machine Sebastien Guiriec (1): ASoC: Intel: bytcr_rt5640: set SSP to I2S mode 2ch Shengjiu Wang (3): ASoC: fsl_esai: spba clock is needed by esai device ASoC: fsl_spdif: spba clk is needed by spdif device ASoC: fsl_asrc: spba clock is needed by asrc device Simon Horman (1): ASoC: rsnd: Add device tree support for r8a77{79,93,94} Sjoerd Simons (1): ASoC: rockchip: spdif: Set transmit data level to 16 samples Songjun Wu (6): ASoC: Atmel: ClassD: supports mono audio ASoC: Atmel: ClassD: add GCK's parent clock in DT binding ASoC: Atmel: ClassD: unregister codec when error occurs ASoC: atmel-pdmic: add the Pulse Density Modulation Interface Controller ASoC: atmel-classd: DT binding for PDMIC driver ASoC: atmel_wm8904: add snd_soc_pm_ops Stephen Boyd (1): ASoC: qcom: Specify LE device endianness Subhransu S. Prusty (7): ASoC: Intel: Skylake: Fix to cleanup if skl_sst_dsp_init fails ALSA: hdac: structure definition for ext_dma_params ASoC: hdac-hdmi: Add hdmi driver ASoC: hdac_hdmi: Add PM support for HDMI ASoC: hdac_hdmi: Add hdac hdmi dai ops ASoC: hdac_hdmi: Setup and start infoframe ASoC: hdac_hdmi: Use i915 component framework for PM Sudip Mukherjee (3): ASoC: hdac_hdmi: fix possible NULL dereference ASoC: tegra_alc5632: check return value ASoC: hdac_hdmi: check error return Sugar Zhang (2): ASoC: rockchip: i2s: compatible with different chips ASoC: rockchip: add playback property Takashi Iwai (1): ASoC: Fix typo in kernel doc comment for snd_soc_put_volsw_sx() Vincent Stehlé (1): ASoC: Intel: Skylake: fix typo in sizeof Vinod Koul (20): ASoC: Intel: Skylake: Fix substream dereference before check ASoC: Intel: Skylake: Fix resource cleanup on teardown ASoC: Intel: Move apci find machine routines ASoC: Intel: Atom: move atom driver to common acpi match ASoC: Intel: Skylake: Fix skl machine driver creation ASoC: Intel: add fw name to common dsp context ASoC: Intel: Skylake: Use the fw name from ACPI mach table ASoC: rt286: set combo jack for Skylake ASoC: core: refactor soc_link_dai_widgets() ASoC: Intel: Skylake: Add I2C depends for SKL machine ASoC: Intel: Skylake: Constrain the audio devices ASoC: fsl-asoc-card: Update the rtd query ASoC: Intel: Skylake: Update the rtd query ASoC: Intel: Skylake: Fix test of a field address ASoC: hdac-hdmi: make driver select CONFIG_HDMI ASoC: dapm: add a dapm sink widget ASoC: core: mark SND_SOC_BYTES_EXT as deprecated ASoC: Intel: Skylake: Move up pipe mem free ASoC: Intel: Skylake: add wov as int sink ASoC: Intel: Skylake: Fix the dapm machine map Yong Zhi (2): ASoc: Intel: boards: update ignore suspend for nau88l25_ssm4567 machine ASoc: Intel: boards: update constraints for nau88l25_ssm4567 machine ZhengShunQian (2): ASoC: rk3036: Inno codec driver for RK3036 SoC ASoC: rk3036: Add binding doc of inno-rk3036 codec driver Zidan Wang (10): ASoC: fsl_esai: ETDR and TX0~5 registers are non volatile ASoC: fsl-sai: add default register map for regmap cache ASoC: fsl_spdif: STL and STR registers are non volatile ASoC: fsl_ssi: using macro for default register map ASoC: fsl-sai: don't set bclk for Tx/Rx Synchronous with another SAI mode ASoC: fsl-sai: set xCR4/xCR5/xMR for SAI master mode ASoC: fsl_sai: add tdm slots operation support ASoC: fsl: using params_width function to simplify code ASoC: fsl_sai: fix no frame clk in master mode ASoC: fsl_asrc: add support for 8-30kHz output sample rate kbuild test robot (2): ASoC: rk3036: fix platform_no_drv_owner.cocci warnings ASoC: da7218: fix boolreturn.cocci warnings Documentation/devicetree/bindings/sound/ak4613.txt | 10 + .../devicetree/bindings/sound/atmel-classd.txt | 6 + .../devicetree/bindings/sound/atmel-pdmic.txt | 55 + Documentation/devicetree/bindings/sound/da7218.txt | 104 + Documentation/devicetree/bindings/sound/da7219.txt | 8 +- .../devicetree/bindings/sound/fsl,asrc.txt | 5 + .../devicetree/bindings/sound/fsl,esai.txt | 5 + .../devicetree/bindings/sound/fsl,spdif.txt | 5 + .../devicetree/bindings/sound/img,i2s-in.txt | 47 + .../devicetree/bindings/sound/img,i2s-out.txt | 51 + .../devicetree/bindings/sound/img,parallel-out.txt | 44 + .../bindings/sound/img,pistachio-internal-dac.txt | 18 + .../devicetree/bindings/sound/img,spdif-in.txt | 41 + .../devicetree/bindings/sound/img,spdif-out.txt | 44 + .../devicetree/bindings/sound/inno-rk3036.txt | 20 + .../devicetree/bindings/sound/renesas,rsnd.txt | 82 + .../bindings/sound/renesas,rsrc-card.txt | 4 +- .../devicetree/bindings/sound/rockchip-i2s.txt | 2 + Documentation/devicetree/bindings/sound/rt5616.txt | 26 + Documentation/devicetree/bindings/sound/rt5659.txt | 75 + Documentation/devicetree/bindings/sound/rt5677.txt | 2 +- .../devicetree/bindings/sound/sun4i-codec.txt | 3 + .../devicetree/bindings/sound/ti,pcm3168a.txt | 48 + .../devicetree/bindings/sound/wlf,wm8974.txt | 15 + Documentation/sound/alsa/img,spdif-in.txt | 49 + arch/arm/mach-s3c64xx/dev-audio.c | 47 +- arch/arm/mach-s3c64xx/include/mach/dma.h | 52 +- arch/arm/plat-samsung/devs.c | 23 +- arch/x86/include/asm/platform_sst_audio.h | 1 + drivers/dma/Kconfig | 2 +- include/linux/platform_data/asoc-s3c.h | 8 + include/sound/ac97_codec.h | 3 + include/sound/da7218.h | 109 + include/sound/da7219.h | 14 +- include/sound/designware_i2s.h | 5 + include/sound/hdaudio_ext.h | 6 + include/sound/rt5659.h | 49 + include/sound/soc-dapm.h | 4 + include/sound/soc-topology.h | 6 +- include/sound/soc.h | 42 +- include/uapi/sound/asoc.h | 2 +- include/uapi/sound/compress_params.h | 5 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-classd.c | 26 +- sound/soc/atmel/atmel-pdmic.c | 738 ++++ sound/soc/atmel/atmel-pdmic.h | 80 + sound/soc/atmel/atmel_wm8904.c | 1 + sound/soc/codecs/Kconfig | 56 +- sound/soc/codecs/Makefile | 18 + sound/soc/codecs/ak4613.c | 118 +- sound/soc/codecs/arizona.c | 143 +- sound/soc/codecs/arizona.h | 17 +- sound/soc/codecs/cs47l24.c | 1148 ++++++ sound/soc/codecs/cs47l24.h | 23 + sound/soc/codecs/da7218.c | 3314 +++++++++++++++ sound/soc/codecs/da7218.h | 1414 +++++++ sound/soc/codecs/da7219.c | 89 +- sound/soc/codecs/da7219.h | 9 - sound/soc/codecs/es8328.c | 25 +- sound/soc/codecs/es8328.h | 1 + sound/soc/codecs/hdac_hdmi.c | 659 +++ sound/soc/codecs/inno_rk3036.c | 490 +++ sound/soc/codecs/inno_rk3036.h | 123 + sound/soc/codecs/max98357a.c | 10 + sound/soc/codecs/pcm3168a-i2c.c | 66 + sound/soc/codecs/pcm3168a-spi.c | 65 + sound/soc/codecs/pcm3168a.c | 767 ++++ sound/soc/codecs/pcm3168a.h | 100 + sound/soc/codecs/rt286.c | 6 + sound/soc/codecs/rt298.c | 2 - sound/soc/codecs/rt5616.c | 1381 +++++++ sound/soc/codecs/rt5616.h | 1819 +++++++++ sound/soc/codecs/rt5645.c | 292 +- sound/soc/codecs/rt5659.c | 4223 ++++++++++++++++++++ sound/soc/codecs/rt5659.h | 1819 +++++++++ sound/soc/codecs/rt5677.c | 13 +- sound/soc/codecs/sgtl5000.c | 1 + sound/soc/codecs/ssm2518.c | 2 +- sound/soc/codecs/wm5110.c | 206 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8962.c | 3 +- sound/soc/codecs/wm8974.c | 8 + sound/soc/codecs/wm8998.c | 46 +- sound/soc/codecs/wm9713.c | 296 +- sound/soc/codecs/wm_adsp.c | 769 +++- sound/soc/codecs/wm_adsp.h | 23 +- sound/soc/davinci/davinci-mcasp.c | 4 +- sound/soc/dwc/designware_i2s.c | 113 +- sound/soc/fsl/fsl-asoc-card.c | 21 +- sound/soc/fsl/fsl_asrc.c | 55 +- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_esai.c | 63 +- sound/soc/fsl/fsl_sai.c | 116 +- sound/soc/fsl/fsl_sai.h | 3 + sound/soc/fsl/fsl_spdif.c | 35 +- sound/soc/fsl/fsl_ssi.c | 25 +- sound/soc/fsl/imx-pcm-dma.c | 2 + sound/soc/fsl/imx-pcm-fiq.c | 4 +- sound/soc/fsl/imx-wm8962.c | 10 +- sound/soc/generic/simple-card.c | 12 +- sound/soc/img/Kconfig | 52 + sound/soc/img/Makefile | 7 + sound/soc/img/img-i2s-in.c | 516 +++ sound/soc/img/img-i2s-out.c | 565 +++ sound/soc/img/img-parallel-out.c | 327 ++ sound/soc/img/img-spdif-in.c | 806 ++++ sound/soc/img/img-spdif-out.c | 441 ++ sound/soc/img/pistachio-internal-dac.c | 287 ++ sound/soc/intel/Kconfig | 46 +- sound/soc/intel/atom/sst-atom-controls.c | 5 +- sound/soc/intel/atom/sst-atom-controls.h | 1 + sound/soc/intel/atom/sst-mfld-platform-pcm.c | 32 +- sound/soc/intel/atom/sst/sst_acpi.c | 67 +- sound/soc/intel/atom/sst/sst_stream.c | 2 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 2 + sound/soc/intel/boards/Makefile | 4 + sound/soc/intel/boards/bytcr_rt5640.c | 241 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 19 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 19 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 19 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 485 +++ sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 536 +++ sound/soc/intel/boards/skl_rt286.c | 128 +- sound/soc/intel/common/Makefile | 7 +- sound/soc/intel/common/sst-acpi.c | 41 +- sound/soc/intel/common/sst-acpi.h | 33 + sound/soc/intel/common/sst-dsp-priv.h | 8 +- sound/soc/intel/common/sst-dsp.c | 2 +- sound/soc/intel/common/sst-dsp.h | 2 +- sound/soc/intel/common/sst-firmware.c | 20 +- sound/soc/intel/common/sst-match-acpi.c | 43 + sound/soc/intel/haswell/sst-haswell-dsp.c | 2 +- sound/soc/intel/haswell/sst-haswell-ipc.c | 31 +- sound/soc/intel/skylake/skl-messages.c | 280 +- sound/soc/intel/skylake/skl-nhlt.c | 19 +- sound/soc/intel/skylake/skl-pcm.c | 256 +- sound/soc/intel/skylake/skl-sst-cldma.c | 97 +- sound/soc/intel/skylake/skl-sst-dsp.h | 21 +- sound/soc/intel/skylake/skl-sst-ipc.c | 108 +- sound/soc/intel/skylake/skl-sst-ipc.h | 9 + sound/soc/intel/skylake/skl-sst.c | 217 +- sound/soc/intel/skylake/skl-topology.c | 658 ++- sound/soc/intel/skylake/skl-topology.h | 63 +- sound/soc/intel/skylake/skl-tplg-interface.h | 105 +- sound/soc/intel/skylake/skl.c | 158 +- sound/soc/intel/skylake/skl.h | 6 +- sound/soc/mediatek/mtk-afe-common.h | 1 - sound/soc/mediatek/mtk-afe-pcm.c | 59 +- sound/soc/omap/omap-hdmi-audio.c | 2 + sound/soc/pxa/brownstone.c | 3 +- sound/soc/pxa/mioa701_wm9713.c | 6 +- sound/soc/qcom/lpass-cpu.c | 1 + sound/soc/rockchip/rockchip_i2s.c | 130 +- sound/soc/rockchip/rockchip_max98090.c | 6 + sound/soc/rockchip/rockchip_rt5645.c | 6 + sound/soc/rockchip/rockchip_spdif.c | 6 +- sound/soc/rockchip/rockchip_spdif.h | 2 +- sound/soc/samsung/Kconfig | 2 - sound/soc/samsung/ac97.c | 29 +- sound/soc/samsung/bells.c | 40 +- sound/soc/samsung/dma.h | 6 +- sound/soc/samsung/dmaengine.c | 20 +- sound/soc/samsung/i2s.c | 31 +- sound/soc/samsung/littlemill.c | 32 +- sound/soc/samsung/odroidx2_max98090.c | 9 +- sound/soc/samsung/pcm.c | 25 +- sound/soc/samsung/s3c2412-i2s.c | 16 +- sound/soc/samsung/s3c24xx-i2s.c | 16 +- sound/soc/samsung/snow.c | 9 +- sound/soc/samsung/spdif.c | 17 +- sound/soc/samsung/speyside.c | 12 +- sound/soc/samsung/tobermory.c | 21 +- sound/soc/sh/Kconfig | 1 - sound/soc/sh/fsi.c | 11 +- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/adg.c | 118 +- sound/soc/sh/rcar/cmd.c | 171 + sound/soc/sh/rcar/core.c | 586 +-- sound/soc/sh/rcar/ctu.c | 99 +- sound/soc/sh/rcar/dma.c | 245 +- sound/soc/sh/rcar/dvc.c | 273 +- sound/soc/sh/rcar/gen.c | 133 +- sound/soc/sh/rcar/mix.c | 158 +- sound/soc/sh/rcar/rcar_snd.h | 117 - sound/soc/sh/rcar/rsnd.h | 335 +- sound/soc/sh/rcar/rsrc-card.c | 129 +- sound/soc/sh/rcar/src.c | 898 ++--- sound/soc/sh/rcar/ssi.c | 744 ++-- sound/soc/sh/rcar/ssiu.c | 225 ++ sound/soc/soc-ac97.c | 125 + sound/soc/soc-core.c | 602 ++- sound/soc/soc-dapm.c | 14 +- sound/soc/soc-ops.c | 4 +- sound/soc/soc-pcm.c | 82 +- sound/soc/sti/uniperif_player.c | 3 +- sound/soc/sunxi/sun4i-codec.c | 279 +- sound/soc/tegra/tegra_alc5632.c | 12 +- sound/soc/tegra/tegra_wm8903.c | 3 +- 202 files changed, 30532 insertions(+), 4175 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/atmel-pdmic.txt create mode 100644 Documentation/devicetree/bindings/sound/da7218.txt create mode 100644 Documentation/devicetree/bindings/sound/img,i2s-in.txt create mode 100644 Documentation/devicetree/bindings/sound/img,i2s-out.txt create mode 100644 Documentation/devicetree/bindings/sound/img,parallel-out.txt create mode 100644 Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt create mode 100644 Documentation/devicetree/bindings/sound/img,spdif-in.txt create mode 100644 Documentation/devicetree/bindings/sound/img,spdif-out.txt create mode 100644 Documentation/devicetree/bindings/sound/inno-rk3036.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5616.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5659.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.txt create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8974.txt create mode 100644 Documentation/sound/alsa/img,spdif-in.txt create mode 100644 include/sound/da7218.h create mode 100644 include/sound/rt5659.h create mode 100644 sound/soc/atmel/atmel-pdmic.c create mode 100644 sound/soc/atmel/atmel-pdmic.h create mode 100644 sound/soc/codecs/cs47l24.c create mode 100644 sound/soc/codecs/cs47l24.h create mode 100644 sound/soc/codecs/da7218.c create mode 100644 sound/soc/codecs/da7218.h create mode 100644 sound/soc/codecs/hdac_hdmi.c create mode 100644 sound/soc/codecs/inno_rk3036.c create mode 100644 sound/soc/codecs/inno_rk3036.h create mode 100644 sound/soc/codecs/pcm3168a-i2c.c create mode 100644 sound/soc/codecs/pcm3168a-spi.c create mode 100644 sound/soc/codecs/pcm3168a.c create mode 100644 sound/soc/codecs/pcm3168a.h create mode 100644 sound/soc/codecs/rt5616.c create mode 100644 sound/soc/codecs/rt5616.h create mode 100644 sound/soc/codecs/rt5659.c create mode 100644 sound/soc/codecs/rt5659.h create mode 100644 sound/soc/img/Kconfig create mode 100644 sound/soc/img/Makefile create mode 100644 sound/soc/img/img-i2s-in.c create mode 100644 sound/soc/img/img-i2s-out.c create mode 100644 sound/soc/img/img-parallel-out.c create mode 100644 sound/soc/img/img-spdif-in.c create mode 100644 sound/soc/img/img-spdif-out.c create mode 100644 sound/soc/img/pistachio-internal-dac.c create mode 100644 sound/soc/intel/boards/skl_nau88l25_max98357a.c create mode 100644 sound/soc/intel/boards/skl_nau88l25_ssm4567.c create mode 100644 sound/soc/intel/common/sst-acpi.h create mode 100644 sound/soc/intel/common/sst-match-acpi.c create mode 100644 sound/soc/sh/rcar/cmd.c delete mode 100644 sound/soc/sh/rcar/rcar_snd.h create mode 100644 sound/soc/sh/rcar/ssiu.c

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v4.4
by Mark Brown 23 Dec '15

23 Dec '15

The following changes since commit 4ef7675344d687a0ef5b0d7c0cee12da005870c0: Linux 4.4-rc6 (2015-12-20 16:06:09 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-fix-v4.4-rc6 for you to fetch changes up to 3dd5fc0eeb5d7cc07b8e3e383037261799b38897: Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/es8328', 'asoc/fix/fsl-sai', 'asoc/fix/rockchip', 'asoc/fix/sgtl5000' and 'asoc/fix/wm8974' into asoc-linus (2015-12-23 00:23:27 +0000) ---------------------------------------------------------------- ASoC: Fixes for v4.4 A collection of small driver specific fixes here, nothing that'll affect users who don't have the devices concerned. At least the wm8974 bug indicates that there's not too many users of some of these devices. ---------------------------------------------------------------- Jean-Michel Hautbois (1): ASoC: sgtl5000: fix VAG power up timing John Keeping (2): ASoC: es8328: Fix deemphasis values ASoC: es8328: Fix shifts for mixer switches Mans Rullgard (1): ASoC: wm8974: set cache type for regmap Mark Brown (1): Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/es8328', 'asoc/fix/fsl-sai', 'asoc/fix/rockchip', 'asoc/fix/sgtl5000' and 'asoc/fix/wm8974' into asoc-linus Peter Ujfalusi (1): ASoC: davinci-mcasp: Fix XDATA check in mcasp_start_tx Sjoerd Simons (1): ASoC: rockchip: spdif: Set transmit data level to 16 samples Zidan Wang (1): ASoC: fsl_sai: fix no frame clk in master mode sound/soc/codecs/es8328.c | 25 +++++++++++++++++-------- sound/soc/codecs/es8328.h | 1 + sound/soc/codecs/sgtl5000.c | 1 + sound/soc/codecs/wm8974.c | 1 + sound/soc/davinci/davinci-mcasp.c | 4 ++-- sound/soc/fsl/fsl_sai.c | 18 ++++++++++++++++++ sound/soc/rockchip/rockchip_spdif.c | 6 ++++-- sound/soc/rockchip/rockchip_spdif.h | 2 +- 8 files changed, 45 insertions(+), 13 deletions(-)

2 2

[alsa-devel] [PATCH 0/4] add NULL test
by Julia Lawall 23 Dec '15

23 Dec '15

4 10

[alsa-devel] [PATCH] ASoC: rt5677: set PLL_CTRL2 non-volatile
by Bard Liao 23 Dec '15

23 Dec '15

There is a status bit on RT5677_PLL1_CTRL2 and RT5677_PLL2_CTRL2. That's why those registers are set volatile. However, the status bit is currently not used by codec driver. So, it should be no problem if we set them non-volatile. The purpose of setting them non-volatile is to restore the setting after a syspend/resume cycle. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Mark, Please revert the "ASoC: rt5677: Reconfigure PLL1 after resume" patch before applying this patch. Thanks. --- sound/soc/codecs/rt5677.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f73fd12..13fef00 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -297,8 +297,6 @@ static bool rt5677_volatile_register(struct device *dev, unsigned int reg) case RT5677_HAP_GENE_CTRL2: case RT5677_PWR_DSP_ST: case RT5677_PRIV_DATA: - case RT5677_PLL1_CTRL2: - case RT5677_PLL2_CTRL2: case RT5677_ASRC_22: case RT5677_ASRC_23: case RT5677_VAD_CTRL5: -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2 0/8] Add support for voice control on Arizona ADSP
by Charles Keepax 23 Dec '15

23 Dec '15

This series add support for voice control on wm8280/5110. This is done by opening a compressed record channel, upon which data will be available once the voice control functionality has been triggered on the DSP. Changes since v1: - Merged selecting of SND_SOC_COMPRESS into patch adding basic hookup as suggested by Vinod. - Fixed a minor typo in usleep_delay, which was using a longer delay than required. - Use kcalloc to allocate an array rather than caculating size manually. Thanks, Charles Charles Keepax (8): ASoC: wm5110: Provide basic hookup for voice control ASoC: wm_adsp: Factor out finding the location of an algorithm region ALSA: compress: Add SND_AUDIOCODEC_BESPOKE ASoC: wm_adsp: Add support for opening a compressed stream ASoC: wm_adsp: Add code to locate and initialise compressed buffer ASoC: wm_adsp: Attach buffers and streams together ASoC: wm_adsp: Add a handler for the compressed IRQ ASoC: wm_adsp: Pull data through compressed read include/uapi/sound/compress_params.h | 5 +- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/arizona.h | 2 +- sound/soc/codecs/wm5110.c | 95 +++- sound/soc/codecs/wm_adsp.c | 907 ++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 21 + 6 files changed, 1010 insertions(+), 21 deletions(-) -- 2.1.4

2 10

[alsa-devel] [PATCH 01/13] ASoC: wm_adsp: Fixup some minor formatting and checkpatch errors
by Charles Keepax 23 Dec '15

23 Dec '15

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 27 +++++++++++++-------------- sound/soc/codecs/wm_adsp.h | 4 ++-- 2 files changed, 15 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 905ae99..3808190 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -544,10 +544,10 @@ static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) be16_to_cpu(scratch[3])); } -static int wm_coeff_info(struct snd_kcontrol *kcontrol, +static int wm_coeff_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; uinfo->count = ctl->len; @@ -593,10 +593,10 @@ static int wm_coeff_write_control(struct wm_coeff_ctl *ctl, return 0; } -static int wm_coeff_put(struct snd_kcontrol *kcontrol, +static int wm_coeff_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; char *p = ucontrol->value.bytes.data; memcpy(ctl->cache, p, ctl->len); @@ -647,10 +647,10 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, return 0; } -static int wm_coeff_get(struct snd_kcontrol *kcontrol, +static int wm_coeff_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; char *p = ucontrol->value.bytes.data; if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) { @@ -829,8 +829,7 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, break; } - list_for_each_entry(ctl, &dsp->ctl_list, - list) { + list_for_each_entry(ctl, &dsp->ctl_list, list) { if (!strcmp(ctl->name, name)) { if (!ctl->enabled) ctl->enabled = 1; @@ -1109,7 +1108,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) goto out_fw; } - header = (void*)&firmware->data[0]; + header = (void *)&firmware->data[0]; if (memcmp(&header->magic[0], "WMFW", 4) != 0) { adsp_err(dsp, "%s: invalid magic\n", file); @@ -1189,7 +1188,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) offset = le32_to_cpu(region->offset) & 0xffffff; type = be32_to_cpu(region->type) & 0xff; mem = wm_adsp_find_region(dsp, type); - + switch (type) { case WMFW_NAME_TEXT: region_name = "Firmware name"; @@ -1646,7 +1645,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) goto out_fw; } - hdr = (void*)&firmware->data[0]; + hdr = (void *)&firmware->data[0]; if (memcmp(hdr->magic, "WMDR", 4) != 0) { adsp_err(dsp, "%s: invalid magic\n", file); goto out_fw; @@ -1672,7 +1671,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) blocks = 0; while (pos < firmware->size && pos - firmware->size > sizeof(*blk)) { - blk = (void*)(&firmware->data[pos]); + blk = (void *)(&firmware->data[pos]); type = le16_to_cpu(blk->type); offset = le16_to_cpu(blk->offset); @@ -1816,7 +1815,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct wm_adsp_alg_region *alg_region; struct wm_coeff_ctl *ctl; int ret; - int val; + unsigned int val; dsp->card = codec->component.card; @@ -1829,7 +1828,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, * For simplicity set the DSP clock rate to be the * SYSCLK rate rather than making it configurable. */ - if(dsp->sysclk_reg) { + if (dsp->sysclk_reg) { ret = regmap_read(dsp->regmap, dsp->sysclk_reg, &val); if (ret != 0) { adsp_err(dsp, "Failed to read SYSCLK state: %d\n", diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 2d117cf..89b7d71 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -45,8 +45,8 @@ struct wm_adsp { struct list_head alg_regions; - int fw_id; - int fw_id_version; + unsigned int fw_id; + unsigned int fw_id_version; const struct wm_adsp_region *mem; int num_mems; -- 2.1.4

3 22

[alsa-devel] [PATCH v2] ALSA: compress: Add SND_AUDIOCODEC_BESPOKE
by Charles Keepax 23 Dec '15

23 Dec '15

When working with the compressed framework occasionally vendors will use esoteric internal audio formats. For such formats it doesn't really make sense to add an new define to the kernel as their use is not sufficiently general. This patch adds a new define SND_AUDIOCODEC_BESPOKE that vendors can use in such situations. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Added a comment to note that users of BESPOKE are free to use the reserved space in snd_enc_generic. Thanks, Charles include/uapi/sound/compress_params.h | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index d9bd9ca..9625484 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -73,7 +73,8 @@ #define SND_AUDIOCODEC_IEC61937 ((__u32) 0x0000000B) #define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C) #define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D) -#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_G729 +#define SND_AUDIOCODEC_BESPOKE ((__u32) 0x0000000E) +#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_BESPOKE /* * Profile and modes are listed with bit masks. This allows for a @@ -312,7 +313,7 @@ struct snd_enc_flac { struct snd_enc_generic { __u32 bw; /* encoder bandwidth */ - __s32 reserved[15]; + __s32 reserved[15]; /* Can be used for SND_AUDIOCODEC_BESPOKE */ } __attribute__((packed, aligned(4))); union snd_codec_options { -- 1.7.2.5

3 2

[alsa-devel] [PATCH 53/54] sound: soc: ac97: Be sure to clamp return value
by Linus Walleij 23 Dec '15

23 Dec '15

As we want gpio_chip .get() calls to be able to return negative error codes and propagate to drivers, we need to go over all drivers and make sure their return values are clamped to [0,1]. We do this by using the ret = !!(val) design pattern. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Mark: as mentioned in 00/54: either apply this directly or ACK it and I will take it into the GPIO tree. --- sound/soc/soc-ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index a2012652f212..7e0acd83b0e6 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -92,7 +92,7 @@ static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) dev_dbg(codec->dev, "get gpio %d : %d\n", offset, ret < 0 ? ret : ret & (1 << offset)); - return ret < 0 ? ret : ret & (1 << offset); + return ret < 0 ? ret : !!(ret & (1 << offset)); } static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned offset, -- 2.4.3

2 1

[alsa-devel] [PATCH] ASoC: rt5616: Return error if device ID mismatch
by Axel Lin 23 Dec '15

23 Dec '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5616.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index f4005cb..0e9414a 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1314,7 +1314,7 @@ static int rt5616_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Device with ID register %#x is not rt5616\n", val); - ret = -ENODEV; + return -ENODEV; } regmap_write(rt5616->regmap, RT5616_RESET, 0); regmap_update_bits(rt5616->regmap, RT5616_PWR_ANLG1, -- 2.1.4

3 2

[alsa-devel] [PATCH] ASoC: mediatek: Turn AFE on/off in runtime resume/suspend
by Koro Chen 23 Dec '15

23 Dec '15

AFE is actually allowed to be turn on before configuration of DAIs since each DAI has its own enabling control. Turn on/off AFE in runtime resume/suspend to avoid AFE being shut down when closing a DAI while other DAIs are still active. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-pcm.c | 24 ++++++------------------ 1 file changed, 6 insertions(+), 18 deletions(-) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index 5399a0e..08af9f5 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -382,9 +382,6 @@ static void mtk_afe_i2s_shutdown(struct snd_pcm_substream *substream, AUD_TCON0_PDN_22M | AUD_TCON0_PDN_24M, AUD_TCON0_PDN_22M | AUD_TCON0_PDN_24M); mtk_afe_dais_disable_clks(afe, afe->clocks[MTK_CLK_I2S1_M], NULL); - - /* disable AFE */ - regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0); } static int mtk_afe_i2s_prepare(struct snd_pcm_substream *substream, @@ -433,9 +430,6 @@ static void mtk_afe_hdmi_shutdown(struct snd_pcm_substream *substream, mtk_afe_dais_disable_clks(afe, afe->clocks[MTK_CLK_I2S3_M], afe->clocks[MTK_CLK_I2S3_B]); - - /* disable AFE */ - regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0); } static int mtk_afe_hdmi_prepare(struct snd_pcm_substream *substream, @@ -679,17 +673,6 @@ static int mtk_afe_dais_hw_free(struct snd_pcm_substream *substream, return snd_pcm_lib_free_pages(substream); } -static int mtk_afe_dais_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct mtk_afe *afe = snd_soc_platform_get_drvdata(rtd->platform); - - /* enable AFE */ - regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0x1); - return 0; -} - static int mtk_afe_dais_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -757,7 +740,6 @@ static const struct snd_soc_dai_ops mtk_afe_dai_ops = { .shutdown = mtk_afe_dais_shutdown, .hw_params = mtk_afe_dais_hw_params, .hw_free = mtk_afe_dais_hw_free, - .prepare = mtk_afe_dais_prepare, .trigger = mtk_afe_dais_trigger, }; @@ -1118,6 +1100,9 @@ static int mtk_afe_runtime_suspend(struct device *dev) { struct mtk_afe *afe = dev_get_drvdata(dev); + /* disable AFE */ + regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0); + /* disable AFE clk */ regmap_update_bits(afe->regmap, AUDIO_TOP_CON0, AUD_TCON0_PDN_AFE, AUD_TCON0_PDN_AFE); @@ -1164,6 +1149,9 @@ static int mtk_afe_runtime_resume(struct device *dev) /* unmask all IRQs */ regmap_update_bits(afe->regmap, AFE_IRQ_MCU_EN, 0xff, 0xff); + + /* enable AFE */ + regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0x1); return 0; err_bck0: -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5616: add an of_match table
by Caesar Wang 23 Dec '15

23 Dec '15

Add a device tree match table. This serves to make the driver's support of device tree more explicit. Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> --- sound/soc/codecs/rt5616.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index f4005cb..e01a32d 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1287,6 +1287,14 @@ static const struct i2c_device_id rt5616_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5616_i2c_id); +#if defined(CONFIG_OF) +static const struct of_device_id rt5616_of_match[] = { + { .compatible = "realtek,rt5616", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5616_of_match); +#endif + static int rt5616_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1359,6 +1367,7 @@ static void rt5616_i2c_shutdown(struct i2c_client *client) static struct i2c_driver rt5616_i2c_driver = { .driver = { .name = "rt5616", + .of_match_table = of_match_ptr(rt5616_of_match), }, .probe = rt5616_i2c_probe, .remove = rt5616_i2c_remove, -- 1.9.1

4 6

[alsa-devel] Applied "ASoC: rt5616: add devicetree document for rt5616" to the asoc tree
by Mark Brown 23 Dec '15

23 Dec '15

The patch ASoC: rt5616: add devicetree document for rt5616 has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 27af4e488c6a2a8c4b6102e777c29fff467a3895 Mon Sep 17 00:00:00 2001 From: Caesar Wang <wxt(a)rock-chips.com> Date: Tue, 22 Dec 2015 13:45:03 +0800 Subject: [PATCH] ASoC: rt5616: add devicetree document for rt5616 Add the description for rt5616 codec. Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/rt5616.txt | 26 ++++++++++++++++++++++ 1 file changed, 26 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5616.txt diff --git a/Documentation/devicetree/bindings/sound/rt5616.txt b/Documentation/devicetree/bindings/sound/rt5616.txt new file mode 100644 index 000000000000..efc48c65198d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5616.txt @@ -0,0 +1,26 @@ +RT5616 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5616". + +- reg : The I2C address of the device. + +Pins on the device (for linking into audio routes) for RT5616: + + * IN1P + * IN2P + * IN2N + * LOUTL + * LOUTR + * HPOL + * HPOR + +Example: + +codec: rt5616@1b { + compatible = "realtek,rt5616"; + reg = <0x1b>; +}; -- 2.6.2

1 0

[alsa-devel] Applied "ASoC: da7218: Enable mic level detection reporting to user-space" to the asoc tree
by Mark Brown 23 Dec '15

23 Dec '15

The patch ASoC: da7218: Enable mic level detection reporting to user-space has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e05c25a1af29d65260ed1458f2cc4a959030ebd2 Mon Sep 17 00:00:00 2001 From: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Date: Thu, 3 Dec 2015 17:10:07 +0000 Subject: [PATCH] ASoC: da7218: Enable mic level detection reporting to user-space This patch adds support to the codec driver to handle mic level detect related IRQs, and report these to user-space using a uevent variable. The uevent variable string "EVENT=MIC_LEVEL_DETECT" is sent to user-space, if the mic level detect feature is enabled, and the audio captured at the chosen mic(s) is above a certain threshold. User-space can then handle the event accordingly (e.g. process audio capture stream). This method was chosen over ALSA control notification for a couple of reasons: 1) There's no requirement here for a control to read state from. The event is the only thing that's required and of interest. 2) tinyalsa support for control notifications does not exist so on platforms using this over alsa-lib there is a need to add code to support this event handling. Another possible option would be to use the standard Jack reporting framework but this really does not fit for this kind of event. Finally, use of the input device framework is not being encouraged, due to difficulties in enabling apps to access input devices, so this has also been avoided. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/da7218.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index eacde128c4d6..72686517ff54 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -2202,6 +2202,16 @@ int da7218_hpldet(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(da7218_hpldet); +static void da7218_micldet_irq(struct snd_soc_codec *codec) +{ + char *envp[] = { + "EVENT=MIC_LEVEL_DETECT", + NULL, + }; + + kobject_uevent_env(&codec->dev->kobj, KOBJ_CHANGE, envp); +} + static void da7218_hpldet_irq(struct snd_soc_codec *codec) { struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); @@ -2232,6 +2242,10 @@ static irqreturn_t da7218_irq_thread(int irq, void *data) if (!status) return IRQ_NONE; + /* Mic level detect */ + if (status & DA7218_LVL_DET_EVENT_MASK) + da7218_micldet_irq(codec); + /* HP detect */ if (status & DA7218_HPLDET_JACK_EVENT_MASK) da7218_hpldet_irq(codec); @@ -2936,11 +2950,6 @@ static int da7218_probe(struct snd_soc_codec *codec) } if (da7218->irq) { - /* Mask off mic level events, currently not handled */ - snd_soc_update_bits(codec, DA7218_EVENT_MASK, - DA7218_LVL_DET_EVENT_MSK_MASK, - DA7218_LVL_DET_EVENT_MSK_MASK); - ret = devm_request_threaded_irq(codec->dev, da7218->irq, NULL, da7218_irq_thread, IRQF_TRIGGER_LOW | IRQF_ONESHOT, -- 2.6.2

1 0

[alsa-devel] [PATCH 0/12] ASoC: rsnd: add Multi channel support
by Kuninori Morimoto 23 Dec '15

23 Dec '15

Hi Mark These are Multi channel support for Renesas R-Car sound. These patches are tested on Salvator-X board, but it can't use Multi channel, because supported format is different between codec chip and R-Car. Thus, it was tested on board which doesn't mount codec chip, with oscilloscope. Kuninori Morimoto (12): 1) ASoC: rsnd: fixup SSIU control timing 2) ASoC: rsnd: tidyup return value of rsnd_get_adinr_bit() 3) ASoC: rsnd: tidyup debug print position on rsnd_dma_attach() 4) ASoC: rsnd: rsnd_dai_connect() returns error if it connect to existing mod 5) ASoC: rsnd: SSI/SSIU use rsnd_get_slot_extend() to check TDM 6) ASoC: rsnd: remove rsnd_get_slot_runtime() 7) ASoC: rsnd: tidyup rsnd_get_slot_xxx() naming 8) ASoC: rsnd: add rsnd_set_slot() / rsnd_get_slot_num() 9) ASoC: rsnd: add rsnd_parse_connect_common() and remove complex macro 10) ASoC: rsnd: add missing DT example for Simple Card 11) ASoC: rsnd: add missing DT example for Simple Card with TDM 12) ASoC: rsnd: add Multi channel support Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 77 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/core.c | 111 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++---------------------------------------- sound/soc/sh/rcar/dma.c | 10 ++++----- sound/soc/sh/rcar/dvc.c | 3 +-- sound/soc/sh/rcar/gen.c | 3 +++ sound/soc/sh/rcar/rsnd.h | 45 +++++++++++++++++++++++++++++++++------ sound/soc/sh/rcar/ssi.c | 118 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++---- sound/soc/sh/rcar/ssiu.c | 57 +++++++++++++++++++++++++++++++++++++++++--------- 8 files changed, 351 insertions(+), 73 deletions(-) Best regards --- Kuninori Morimoto

3 32

[alsa-devel] [PATCH] ASoC: ssm2518: Use a signed return type for ssm2518_lookup_mcs()
by SF Markus Elfring 23 Dec '15

23 Dec '15

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 20 Dec 2015 10:34:25 +0100 The return type "unsigned int" was used by the ssm2518_lookup_mcs() function even though it will eventually return a negative error code. Improve this implementation detail by deletion of the type modifier then. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/codecs/ssm2518.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 86b81a6..e2e0bfa 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -309,7 +309,7 @@ static const struct snd_pcm_hw_constraint_list ssm2518_constraints_12288000 = { .count = ARRAY_SIZE(ssm2518_rates_12288000), }; -static unsigned int ssm2518_lookup_mcs(struct ssm2518 *ssm2518, +static int ssm2518_lookup_mcs(struct ssm2518 *ssm2518, unsigned int rate) { const unsigned int *sysclks = NULL; -- 2.6.3

2 1

[alsa-devel] [PATCH] ASoC: rt5616: rename some alsa control names
by Bard Liao 23 Dec '15

23 Dec '15

Rename some alsa control name as what they should be. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5616.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index f4005cb..4de8c86 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -323,9 +323,9 @@ static const struct snd_kcontrol_new rt5616_snd_controls[] = { RT5616_L_VOL_SFT, RT5616_R_VOL_SFT, 175, 0, dac_vol_tlv), /* IN1/IN2 Control */ - SOC_SINGLE_TLV("IN1 Boost", RT5616_IN1_IN2, + SOC_SINGLE_TLV("IN1 Boost Volume", RT5616_IN1_IN2, RT5616_BST_SFT1, 8, 0, bst_tlv), - SOC_SINGLE_TLV("IN2 Boost", RT5616_IN1_IN2, + SOC_SINGLE_TLV("IN2 Boost Volume", RT5616_IN1_IN2, RT5616_BST_SFT2, 8, 0, bst_tlv), /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5616_INL1_INR1_VOL, @@ -339,7 +339,7 @@ static const struct snd_kcontrol_new rt5616_snd_controls[] = { 127, 0, adc_vol_tlv), /* ADC Boost Volume Control */ - SOC_DOUBLE_TLV("ADC Boost Gain", RT5616_ADC_BST_VOL, + SOC_DOUBLE_TLV("ADC Boost Volume", RT5616_ADC_BST_VOL, RT5616_ADC_L_BST_SFT, RT5616_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), }; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: atmel_wm8904: add snd_soc_pm_ops
by Songjun Wu 23 Dec '15

23 Dec '15

Sometimes the audio play can not be resumed after it is suspended. Add snd_soc_pm_ops to execute power management operations, then this issue is fixed. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> --- sound/soc/atmel/atmel_wm8904.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index 1933bcd..fdd28ed 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -183,6 +183,7 @@ static struct platform_driver atmel_asoc_wm8904_driver = { .driver = { .name = "atmel-wm8904-audio", .of_match_table = of_match_ptr(atmel_asoc_wm8904_dt_ids), + .pm = &snd_soc_pm_ops, }, .probe = atmel_asoc_wm8904_probe, .remove = atmel_asoc_wm8904_remove, -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2] ASoC: fsl_asrc: add support for 8-30kHz output sample rate
by Zidan Wang 23 Dec '15

23 Dec '15

Add 8kHz, 11.025kHz, 16kHz, 22.05kHz output sample rate support. According referance menual, "Limited support for the case when output sampling rates is between 8kHz and 30kHz. The limitation is the supported ratio (Fsin/Fsout) range as between 1/24 to 8." Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v1->v2: 1. fix indentation issue. sound/soc/fsl/fsl_asrc.c | 39 +++++++++++++++++++++++---------------- 1 file changed, 23 insertions(+), 16 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index d964016..bd32c30 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -31,21 +31,21 @@ dev_dbg(&asrc_priv->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) /* Sample rates are aligned with that defined in pcm.h file */ -static const u8 process_option[][8][2] = { - /* 32kHz 44.1kHz 48kHz 64kHz 88.2kHz 96kHz 176kHz 192kHz */ - {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 5512Hz */ - {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 8kHz */ - {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 11025Hz */ - {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 16kHz */ - {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 22050Hz */ - {{0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0},}, /* 32kHz */ - {{0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 44.1kHz */ - {{0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 48kHz */ - {{1, 2}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0},}, /* 64kHz */ - {{1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 88.2kHz */ - {{1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 96kHz */ - {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 176kHz */ - {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 192kHz */ +static const u8 process_option[][12][2] = { + /* 8kHz 11.025kHz 16kHz 22.05kHz 32kHz 44.1kHz 48kHz 64kHz 88.2kHz 96kHz 176kHz 192kHz */ + {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 5512Hz */ + {{0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 8kHz */ + {{0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 11025Hz */ + {{1, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 16kHz */ + {{1, 2}, {1, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 22050Hz */ + {{1, 2}, {2, 1}, {2, 1}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0},}, /* 32kHz */ + {{2, 2}, {2, 2}, {2, 1}, {2, 1}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 44.1kHz */ + {{2, 2}, {2, 2}, {2, 1}, {2, 1}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 48kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {1, 2}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0},}, /* 64kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 88.2kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 96kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 176kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 192kHz */ }; /* Corresponding to process_option */ @@ -55,7 +55,7 @@ static int supported_input_rate[] = { }; static int supported_asrc_rate[] = { - 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, + 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, }; /** @@ -287,6 +287,13 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) return -EINVAL; } + if ((outrate > 8000 && outrate < 30000) && + (outrate/inrate > 24 || inrate/outrate > 8)) { + pair_err("exceed supported ratio range [1/24, 8] for \ + inrate/outrate: %d/%d\n", inrate, outrate); + return -EINVAL; + } + /* Validate input and output clock sources */ clk_index[IN] = clk_map[IN][config->inclk]; clk_index[OUT] = clk_map[OUT][config->outclk]; -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: fsl-asoc-card: use different route map for AC'97 mode
by Maciej S. Szmigiero 23 Dec '15

23 Dec '15

fsl_ssi uses different stream names ("AC97 Playback" / "AC97 Capture") in AC'97 mode so in this case fsl-asoc-card route map should also be using them. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl-asoc-card.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index c63d89da51f1..562b3bd22d9a 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -107,6 +107,13 @@ static const struct snd_soc_dapm_route audio_map[] = { {"CPU-Capture", NULL, "Capture"}, }; +static const struct snd_soc_dapm_route audio_map_ac97[] = { + {"AC97 Playback", NULL, "ASRC-Playback"}, + {"Playback", NULL, "AC97 Playback"}, + {"ASRC-Capture", NULL, "AC97 Capture"}, + {"AC97 Capture", NULL, "Capture"}, +}; + /* Add all possible widgets into here without being redundant */ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_LINE("Line Out Jack", NULL), @@ -579,7 +586,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.dev = &pdev->dev; priv->card.name = priv->name; priv->card.dai_link = priv->dai_link; - priv->card.dapm_routes = audio_map; + priv->card.dapm_routes = fsl_asoc_card_is_ac97(priv) ? + audio_map_ac97 : audio_map; priv->card.late_probe = fsl_asoc_card_late_probe; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;

3 2

[alsa-devel] [PATCH 0/5] ASoC: Allow topology to create DAI links
by mengdong.lin＠linux.intel.com 23 Dec '15

23 Dec '15

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series allows DAI links to be created from topology info. The components with topology info can come in two ways: - As a component for cpu dai or codec dai of a DAI link predefined by machine driver. - As a auxiliary components defined by the machine driver for a soc card. For this case, the machine driver even need not define any DAI links. The ASoC core will find and probe these components when instantiating the soc card. When probing the components, topology info will be loaded and DAI links can be created by the topology core. And then ASoC core will find and bind new DAI links after probing the components. We implement a DAI link list and define API for adding/removing DAI links from topology. Mengdong Lin (5): ASoC: Implement DAI links in a list & define API to add/remove a link ASoC: Define add/remove_dai_link ops for a soc card ASoC: soc_bind_dai_link() directly returns success for a bound DAI link ASoC: Bind new DAI links after probing components ASoC: The soc card can have auxiliary components include/sound/soc.h | 29 +++++++++- sound/soc/soc-core.c | 153 +++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 180 insertions(+), 2 deletions(-) -- 2.5.0

3 19

[alsa-devel] rawmidi virtual and long sysexes
by Ryan C. Underwood 22 Dec '15

22 Dec '15

Hi alsa-devel, Recently I noticed this problem and fixed it in the client consuming sequencer events: https://github.com/munt/munt/commit/0ea0846d0849872acf4384bdf5edeff026d12f42 Basically, the rawmidi virtual plugin enforces a 256-byte limit on events, and emits multiple events if e.g. a >256 byte sysex is encountered. This does not seem to be documented behavior for snd_seq_event_input(), nor is the fixed limit described anywhere. Would it be better if the maximum event size could be configured through params? That would avoid needing this clumsy workaround by sequencer clients. In any case, it seems that the documentation should be updated to reflect the need to receive multiple sequencer events to complete a sysex if EOX is not located in the first. Please cc me as I'm not subscribed. -- Ryan C. Underwood, <nemesis(a)icequake.net>

3 3

[alsa-devel] [PATCH] sound: hda_intel: add card number to irq description
by Heiner Kallweit 22 Dec '15

22 Dec '15

Currently the info in /proc/interrupts doesn't allow to figure out which interrupt belongs to which card (HDMI, PCH, ..). Therefore add card details to the interrupt description. With the patch the info in /proc/interrupts looks like this: PCI-MSI 442368-edge snd_hda_intel:card1 PCI-MSI 49152-edge snd_hda_intel:card0 This could be partially reused for the hda_tegra driver as it also uses struct azx. Signed-off-by: Heiner Kallweit <hkallweit1(a)gmail.com> --- sound/pci/hda/hda_controller.h | 2 ++ sound/pci/hda/hda_intel.c | 4 +++- 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index ec63bbf..9c6344f 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -125,6 +125,8 @@ struct azx { int num_streams; const int *jackpoll_ms; /* per-card jack poll interval */ + char irq_descr[32]; /* Interrupt description */ + /* Register interaction. */ const struct hda_controller_ops *ops; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 83800ac..ef4e06b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -725,7 +725,7 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect) if (request_irq(chip->pci->irq, azx_interrupt, chip->msi ? 0 : IRQF_SHARED, - KBUILD_MODNAME, chip)) { + chip->irq_descr, chip)) { dev_err(chip->card->dev, "unable to grab IRQ %d, disabling device\n", chip->pci->irq); @@ -1605,6 +1605,8 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, check_msi(chip); chip->dev_index = dev; chip->jackpoll_ms = jackpoll_ms; + snprintf(chip->irq_descr, sizeof(chip->irq_descr), "%s:%s", + KBUILD_MODNAME, dev_name(&card->card_dev)); INIT_LIST_HEAD(&chip->pcm_list); INIT_WORK(&hda->irq_pending_work, azx_irq_pending_work); INIT_LIST_HEAD(&hda->list); -- 2.6.4

2 2

[alsa-devel] [RFC PATCH 0/2] ALSA: hda - DP MST audio for Jack support
by libin.yang＠linux.intel.com 22 Dec '15

22 Dec '15

From: Libin Yang <libin.yang(a)linux.intel.com> This is the patch set for Jack support for DP MST audio Libin Yang (2): ALSA: hda - jack support DP MST audio ALSA: hda - hdmi audio add dynamically jack binding to pin sound/pci/hda/hda_jack.c | 186 ++++++++++++++++++++++++++++++++++++++++----- sound/pci/hda/hda_jack.h | 32 +++++++- sound/pci/hda/patch_hdmi.c | 116 ++++++++++++++++++++++------ 3 files changed, 288 insertions(+), 46 deletions(-) -- 1.9.1

4 12

[alsa-devel] [PATCH 0/8] ALSA: oxfw: obsolete snd-scs1x with snd-oxfw
by Takashi Sakamoto 22 Dec '15

22 Dec '15

Hi, This patchset update a part of my previous RFCv2 (patch 10-17). [alsa-devel] [RFC][PATCH 00/17 v2] ALSA: oxfw: refactoring and merging scs1x module http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/101558.h… This patchset should be accumurated to the other patches under reviewing. [alsa-devel] [PATCH 0/4] ALSA: oxfw: localize stuffs of old firewire-speakers module http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/102071.h… This patchset obsoletes snd-scs1x module with snd-oxfw module because the models supported by snd-scs1x module are based on OXFW971, which snd-oxfw supports, too. So snd-oxfw should support them. As a result, one of the models gets packet streaming functionality to receive PCM samples. This is a lack of functionalities in snd-scs1x. Usually I won't post additional patches during the others are under reviewing. But it's near the end of year. I'd like to post these patches in advance to avoid year-end busyness. I'm happy if you review them and these patches go forward, then I can get a short rest in the year-end. I note that I have the other patches for ALSA dice module, too. I'm also happy to start reviewing process for them after this patchset goes forward... Changes from RFCv2: * remove sound/firewire/scs1x.c (I forget it) * revise operations in rawmidi callbacks Takashi Sakamoto (8): ALSA: oxfw: add scs1x layer ALSA: oxfw: allocate own address region for SCS.1 series ALSA: oxfw: copy handlers of asynchronous transaction for MIDI capture ALSA: oxfw: add MIDI capture port for SCS.1 models ALSA: oxfw: copy handlers of asynchronous transaction for MIDI playback ALSA: oxfw: add MIDI playback port for SCS.1 models ALSA: oxfw: obsolete scs1x module ALSA: oxfw: add stream format quirk for SCS.1 models sound/firewire/Kconfig | 12 +- sound/firewire/Makefile | 2 - sound/firewire/oxfw/Makefile | 2 +- sound/firewire/oxfw/oxfw-scs1x.c | 406 ++++++++++++++++++++++++++++++ sound/firewire/oxfw/oxfw.c | 38 ++- sound/firewire/oxfw/oxfw.h | 2 + sound/firewire/scs1x.c | 530 --------------------------------------- 7 files changed, 446 insertions(+), 546 deletions(-) create mode 100644 sound/firewire/oxfw/oxfw-scs1x.c delete mode 100644 sound/firewire/scs1x.c -- 2.5.0

2 12

[alsa-devel] [PATCH] ALSA: usb-audio: use list_for_each_entry_continue_reverse
by Geliang Tang 22 Dec '15

22 Dec '15

For better readability, use list_for_each_entry_continue_reverse() in have_dup_chmap(). Signed-off-by: Geliang Tang <geliangtang(a)163.com> --- sound/usb/stream.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 8ee14f2..c4dc577 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -125,11 +125,9 @@ static int usb_chmap_ctl_info(struct snd_kcontrol *kcontrol, static bool have_dup_chmap(struct snd_usb_substream *subs, struct audioformat *fp) { - struct list_head *p; + struct audioformat *prev = fp; - for (p = fp->list.prev; p != &subs->fmt_list; p = p->prev) { - struct audioformat *prev; - prev = list_entry(p, struct audioformat, list); + list_for_each_entry_continue_reverse(prev, &subs->fmt_list, list) { if (prev->chmap && !memcmp(prev->chmap, fp->chmap, sizeof(*fp->chmap))) return true; -- 2.5.0

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix silent headphone output on MacPro 4, 1 (v2)
by Mario Kleiner 22 Dec '15

22 Dec '15

Without this patch, internal speaker and line-out work, but front headphone output jack stays silent on the Mac Pro 4,1. This code path also gets executed on the MacPro 5,1 due to identical codec SSID, but i don't know if it has any positive or adverse effects there or not. (v2) Implement feedback from Takashi Iwai: Reuse alc889_fixup_mbp_vref and just add a new nid 0x19 for the MacPro 4,1. Signed-off-by: Mario Kleiner <mario.kleiner.de(a)gmail.com> Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9bedf7c..0e654e4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1755,6 +1755,7 @@ enum { ALC889_FIXUP_MBA11_VREF, ALC889_FIXUP_MBA21_VREF, ALC889_FIXUP_MP11_VREF, + ALC889_FIXUP_MP41_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, ALC887_FIXUP_ASUS_BASS, @@ -1843,7 +1844,7 @@ static void alc889_fixup_mbp_vref(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; - static hda_nid_t nids[2] = { 0x14, 0x15 }; + static hda_nid_t nids[3] = { 0x14, 0x15, 0x19 }; int i; if (action != HDA_FIXUP_ACT_INIT) @@ -2133,6 +2134,12 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC885_FIXUP_MACPRO_GPIO, }, + [ALC889_FIXUP_MP41_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mbp_vref, + .chained = true, + .chain_id = ALC885_FIXUP_MACPRO_GPIO, + }, [ALC882_FIXUP_INV_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic, @@ -2215,7 +2222,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF), - SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 5,1", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 4,1/5,1", ALC889_FIXUP_MP41_VREF), SND_PCI_QUIRK(0x106b, 0x4300, "iMac 9,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), -- 1.9.1

2 1

Re: [alsa-devel] [PATCH v1 2/2] dt-bindings: sound: add devicetree document for rt5616
by Caesar Wang 22 Dec '15

22 Dec '15

Hi 在 2015年12月22日 05:15, Frank Rowand 写道: > On 12/21/2015 6:56 AM, Caesar Wang wrote: >> Add the description for rt5616 codec. >> >> Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> >> >> --- >> >> Changes in v1: >> - As Heiko comments, remove the not exist option properties. >> >> Documentation/devicetree/bindings/sound/rt5616.txt | 36 ++++++++++++++++++++++ >> 1 file changed, 36 insertions(+) >> create mode 100644 Documentation/devicetree/bindings/sound/rt5616.txt >> >> diff --git a/Documentation/devicetree/bindings/sound/rt5616.txt b/Documentation/devicetree/bindings/sound/rt5616.txt >> new file mode 100644 >> index 0000000..2030a22 >> --- /dev/null >> +++ b/Documentation/devicetree/bindings/sound/rt5616.txt >> @@ -0,0 +1,36 @@ >> +RT5616 audio CODEC >> + >> +This device supports I2C only. >> + >> +Required properties: >> + >> +- compatible : "realtek,rt5616". >> + >> +- reg : The I2C address of the device. >> + >> +Pins on the device (for linking into audio routes) for RT5616: >> + >> + * IN1P >> + * IN2P >> + * IN2N >> + * LOUTL >> + * LOUTR >> + * CPN2 >> + * CPP2 >> + * CPN1 >> + * CPP1 >> + * HPO_R >> + * HPO_L >> + * ADCDAT1 >> + * DACDAT1 >> + * LRCK1 >> + * BCLK1 >> + * MCLK >> + * GPIO1 >> + >> +Example: >> + >> +rt5616 { > The node name should be rt5616@1b (including the reg value > from the node). In general, that's right. Okay, fixed in next version. > >> + compatible = "realtek,rt5616"; >> + reg = <0x1b>; >> +}; >> > > > -- caesar wang | software engineer | wxt(a)rock-chip.com

1 0

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix silent headphone output on MacPro 4, 1
by Mario Kleiner 22 Dec '15

22 Dec '15

Without this patch, internal speaker and line-out work, but front headphone output jack stays silent on the Mac Pro 4,1. This code path also gets executed on the MacPro 5,1 due to identical codec SSID, but i don't know if it has any positive or adverse effects there or not. Signed-off-by: Mario Kleiner <mario.kleiner.de(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 32 +++++++++++++++++++++++++++++++- 1 file changed, 31 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9bedf7c..6224bdd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1755,6 +1755,7 @@ enum { ALC889_FIXUP_MBA11_VREF, ALC889_FIXUP_MBA21_VREF, ALC889_FIXUP_MP11_VREF, + ALC889_FIXUP_MP41_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, ALC887_FIXUP_ASUS_BASS, @@ -1905,6 +1906,29 @@ static void alc889_fixup_mba21_vref(struct hda_codec *codec, alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids)); } +/* Set VREF on HP pin on Mac Pro 4,1 */ +static void alc889_fixup_mp41_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + static hda_nid_t nids[1] = { 0x19 }; + int i; + + if (action != HDA_FIXUP_ACT_INIT) + return; + + for (i = 0; i < ARRAY_SIZE(nids); i++) { + unsigned int val = snd_hda_codec_get_pincfg(codec, nids[i]); + if (get_defcfg_device(val) != AC_JACK_HP_OUT) + continue; + val = snd_hda_codec_get_pin_target(codec, nids[i]); + val |= AC_PINCTL_VREF_80; + snd_hda_set_pin_ctl(codec, nids[i], val); + spec->gen.keep_vref_in_automute = 1; + break; + } +} + /* Don't take HP output as primary * Strangely, the speaker output doesn't work on Vaio Z and some Vaio * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05 @@ -2133,6 +2157,12 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC885_FIXUP_MACPRO_GPIO, }, + [ALC889_FIXUP_MP41_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mp41_vref, + .chained = true, + .chain_id = ALC885_FIXUP_MACPRO_GPIO, + }, [ALC882_FIXUP_INV_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic, @@ -2215,7 +2245,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF), - SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 5,1", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 4,1/5,1", ALC889_FIXUP_MP41_VREF), SND_PCI_QUIRK(0x106b, 0x4300, "iMac 9,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), -- 1.9.1

2 2

[alsa-devel] [PATCH 00/13] Add ASoC support for AMD APUs [v5]
by Alex Deucher 21 Dec '15

21 Dec '15

This patch set implements support for i2s audio and new AMD GPUs. The i2s codec is fed by a DMA engine on the GPU. To handle this we create mfd cells which we hang the i2s codec and DMA engine on. Because of this, this patch set covers two subsystems: drm and alsa. The drm patches add support for the ACP hw block which provides the DMA engine for the i2s codec. The alsa patches add the ASoC driver for the i2s codec. Since the alsa changes depend on the drm changes, I'd like to take the alsa patches in via the drm tree. V2 changes: - Use the MFD subsystem rather than adding our own bus - Squash all sub-feature patches together - fix comments mentioned in previous review V3 changes: - Update the designware driver to handle slave mode, amd specific features - Use the designware driver directly for i2s - Move the DMA handling from the GPU driver into the AMD ASoC driver - Change the license on the ASoC driver to GPL v4 changes: - patch "ASoC : dwc : support dw i2s in slave mode" accepted - Add a _dai_fmt() operation that checks to make sure that the mode we're setting corresponds to what we read back from the hardware. - Split specific quirks into separate patches - Set the specific quirks that apply to AMD chips in the acp driver v5 changes: - patch "ASoC : dwc : add check for master/slave format" accepted - Fix MFD_CORE selection in ACP Kconfig - Add irq domain support to amdgpu driver - Use genirq in ACP DMA driver - Export some genpd symbols to support ACP powergating (Acked by PM maintainer) - Use genpd for ACP powergating - add separate capture and playback instances of dws in ACP init - add runtime suspend support for dws in master mode Patch 9 adds the register headers for the ACP block which is a pretty big patch so I've excluded it from email. The entire patch set can be viewed here: http://cgit.freedesktop.org/~agd5f/linux/log/?h=acp-upstream8 Thanks, Alex Alex Deucher (3): drm/amdgpu/cgs: add an interface to access PCI resources drm/amdgpu: add irq domain support drm/amd: add pm domain for ACP IP sub blocks Maruthi Bayyavarapu (1): drm/amd: add ACP driver support Maruthi Srinivas Bayyavarapu (9): ASoC: dwc: add runtime suspend/resume functionality ASoC: dwc: add quirk for different register offset ASoC: dwc: reconfigure dwc in 'resume' from 'suspend' PM / Domains: export symbols to add/remove devices from genpd ASoC : AMD : add ACP 2.2 register headers ASoC: AMD: add ACP 2.x IP DMA abstraction layer ASoC: AMD: add AMD ASoC ACP 2.x DMA driver ASoC: AMD: add pm ops ASoC: AMD: Manage ACP 2.x SRAM banks power drivers/base/power/domain.c | 2 + drivers/gpu/drm/Kconfig | 2 + drivers/gpu/drm/amd/acp/Kconfig | 11 + drivers/gpu/drm/amd/acp/Makefile | 9 + drivers/gpu/drm/amd/acp/acp_hw.c | 50 + drivers/gpu/drm/amd/acp/include/acp_gfx_if.h | 34 + drivers/gpu/drm/amd/amdgpu/Makefile | 13 +- drivers/gpu/drm/amd/amdgpu/amdgpu.h | 12 + drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 501 ++++++ drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h | 42 + drivers/gpu/drm/amd/amdgpu/amdgpu_cgs.c | 36 + drivers/gpu/drm/amd/amdgpu/amdgpu_irq.c | 108 +- drivers/gpu/drm/amd/amdgpu/amdgpu_irq.h | 9 + drivers/gpu/drm/amd/amdgpu/cik_ih.c | 6 + drivers/gpu/drm/amd/amdgpu/cz_ih.c | 7 + drivers/gpu/drm/amd/amdgpu/iceland_ih.c | 7 + drivers/gpu/drm/amd/amdgpu/tonga_ih.c | 7 + drivers/gpu/drm/amd/amdgpu/vi.c | 12 + drivers/gpu/drm/amd/include/amd_shared.h | 1 + drivers/gpu/drm/amd/include/cgs_common.h | 34 + include/sound/designware_i2s.h | 5 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 4 + sound/soc/amd/Makefile | 3 + sound/soc/amd/acp-pcm-dma.c | 560 +++++++ sound/soc/amd/acp.c | 666 ++++++++ sound/soc/amd/acp.h | 140 ++ sound/soc/amd/include/acp_2_2_d.h | 609 +++++++ sound/soc/amd/include/acp_2_2_enum.h | 1068 ++++++++++++ sound/soc/amd/include/acp_2_2_sh_mask.h | 2292 ++++++++++++++++++++++++++ sound/soc/dwc/designware_i2s.c | 113 +- 32 files changed, 6325 insertions(+), 40 deletions(-) create mode 100644 drivers/gpu/drm/amd/acp/Kconfig create mode 100644 drivers/gpu/drm/amd/acp/Makefile create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.c create mode 100644 drivers/gpu/drm/amd/acp/include/acp_gfx_if.h create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c create mode 100644 sound/soc/amd/acp.c create mode 100644 sound/soc/amd/acp.h create mode 100644 sound/soc/amd/include/acp_2_2_d.h create mode 100644 sound/soc/amd/include/acp_2_2_enum.h create mode 100644 sound/soc/amd/include/acp_2_2_sh_mask.h -- 1.8.3.1

3 21

[alsa-devel] [PATCH v1 1/5] ALSA: fm801: propagate TUNER_ONLY bit when autodetected
by Andy Shevchenko 21 Dec '15

21 Dec '15

The commit d7ba858a7f7a (ALSA: fm801: implement TEA575x tuner autodetection) brings autodetection to the driver. However the autodetection algorithm misses the TUNER_ONLY bit if it is supplied by the user. Thus, user gets weird messages and no card registered. snd_fm801 0000:0d:01.0: detected TEA575x radio type SF64-PCR snd_fm801 0000:0d:01.0: AC'97 interface is busy (1) snd_fm801 0000:0d:01.0: AC'97 interface is busy (1) ... snd_fm801 0000:0d:01.0: AC'97 0 does not respond - RESET snd_fm801 0000:0d:01.0: AC'97 interface is busy (1) snd_fm801 0000:0d:01.0: AC'97 interface is busy (1) snd_fm801 0000:0d:01.0: AC'97 0 access is not valid [0x0], removing mixer. snd_fm801: probe of 0000:0d:01.0 failed with error -5 Do a copy of TUNER_ONLY bit to be applied after autodetection is done. Fixes: d7ba858a7f7a (ALSA: fm801: implement TEA575x tuner autodetection) Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Cc: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/fm801.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 4c9d804..6114d81 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1269,6 +1269,8 @@ static int snd_fm801_create(struct snd_card *card, return -ENODEV; } } else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) { + unsigned int tuner_only = tea575x_tuner & TUNER_ONLY; + /* autodetect tuner connection */ for (tea575x_tuner = 1; tea575x_tuner <= 3; tea575x_tuner++) { chip->tea575x_tuner = tea575x_tuner; @@ -1283,6 +1285,8 @@ static int snd_fm801_create(struct snd_card *card, dev_err(card->dev, "TEA575x radio not found\n"); chip->tea575x_tuner = TUNER_DISABLED; } + + chip->tea575x_tuner |= tuner_only; } if (!(chip->tea575x_tuner & TUNER_DISABLED)) { strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name, -- 2.6.4

2 5

[alsa-devel] [PATCH v6] sun4i-codec: Add FM, Line and Mic inputs
by Danny Milosavljevic 21 Dec '15

21 Dec '15

Add inputs to sun4i-codec: - FM-in Left and Right - Line-in Left and Right - Mic1-in - Mic2-in Signed-off-by: Danny Milosavljevic <dannym+a(a)scratchpost.org> Tested-by: Danny Milosavljevic <dannym+a(a)scratchpost.org> (ONLY ON A20!) --- Hi, this is the sixth version of the patch that adds inputs to sun4i-codec. Changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. I successfully tested it again on an A20 board using alsamixer, headphones, a radio and my ears. Note that because of missing capturing support I tested only the mixing, for Mic, Line, and FM. The patches are on top of <git://git.kernel.org/pub/scm/linux/kernel/git/mripard/linux.git>, branch "sunxi/for-next". Regards, Danny --- b/sound/soc/sunxi/sun4i-codec.c | 179 +++++++++++++++++++++++++++++++++++++--- 1 file changed, 166 insertions(+), 13 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index bcbf4da..4241999 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -57,9 +57,20 @@ #define SUN4I_CODEC_DAC_ACTL_DACAENR (31) #define SUN4I_CODEC_DAC_ACTL_DACAENL (30) #define SUN4I_CODEC_DAC_ACTL_MIXEN (29) +#define SUN4I_CODEC_DAC_ACTL_LNG (26) +#define SUN4I_CODEC_DAC_ACTL_FMG (23) +#define SUN4I_CODEC_DAC_ACTL_MICG (20) +#define SUN4I_CODEC_DAC_ACTL_LLNS (19) +#define SUN4I_CODEC_DAC_ACTL_RLNS (18) +#define SUN4I_CODEC_DAC_ACTL_LFMS (17) +#define SUN4I_CODEC_DAC_ACTL_RFMS (16) #define SUN4I_CODEC_DAC_ACTL_LDACLMIXS (15) #define SUN4I_CODEC_DAC_ACTL_RDACRMIXS (14) #define SUN4I_CODEC_DAC_ACTL_LDACRMIXS (13) +#define SUN4I_CODEC_DAC_ACTL_MIC1LS (12) +#define SUN4I_CODEC_DAC_ACTL_MIC1RS (11) +#define SUN4I_CODEC_DAC_ACTL_MIC2LS (10) +#define SUN4I_CODEC_DAC_ACTL_MIC2RS (9) #define SUN4I_CODEC_DAC_ACTL_DACPAS (8) #define SUN4I_CODEC_DAC_ACTL_MIXPAS (7) #define SUN4I_CODEC_DAC_ACTL_PA_MUTE (6) @@ -84,8 +95,11 @@ #define SUN4I_CODEC_ADC_ACTL_PREG1EN (29) #define SUN4I_CODEC_ADC_ACTL_PREG2EN (28) #define SUN4I_CODEC_ADC_ACTL_VMICEN (27) +#define SUN4I_CODEC_ADC_ACTL_PREG1_A10 (25) +#define SUN4I_CODEC_ADC_ACTL_PREG2_A10 (23) #define SUN4I_CODEC_ADC_ACTL_VADCG (20) #define SUN4I_CODEC_ADC_ACTL_ADCIS (17) +#define SUN4I_CODEC_ADC_ACTL_LNRDF (16) #define SUN4I_CODEC_ADC_ACTL_PA_EN (4) #define SUN4I_CODEC_ADC_ACTL_DDE (3) #define SUN4I_CODEC_ADC_DEBUG (0x2c) @@ -94,7 +108,6 @@ #define SUN4I_CODEC_DAC_TXCNT (0x30) #define SUN4I_CODEC_ADC_RXCNT (0x34) #define SUN4I_CODEC_AC_SYS_VERI (0x38) -#define SUN4I_CODEC_AC_MIC_PHONE_CAL (0x3c) struct sun4i_codec { struct device *dev; @@ -402,16 +415,77 @@ static const struct snd_kcontrol_new sun4i_codec_pa_mute = SUN4I_CODEC_DAC_ACTL_PA_MUTE, 1, 0); static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1); +static DECLARE_TLV_DB_SCALE(sun4i_codec_linein_loopback_gain_scale, -150, 150, 0); +static DECLARE_TLV_DB_SCALE(sun4i_codec_fmin_loopback_gain_scale, -450, 150, 0); +static DECLARE_TLV_DB_SCALE(sun4i_codec_micin_loopback_gain_scale, -450, 150, 0); +static DECLARE_TLV_DB_RANGE(sun4i_codec_micin_preamp_gain_scale_a10, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(3500, 300, 0) +); + +#define SUN4I_COMMON_CODEC_WIDGETS \ + SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL, \ + SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0, \ + sun4i_codec_pa_volume_scale), \ + /* Line-In, FM-In, Mic1-In, Mic2-In */ \ + SOC_SINGLE_TLV("Line-In Playback Gain", \ + SUN4I_CODEC_DAC_ACTL, \ + SUN4I_CODEC_DAC_ACTL_LNG, \ + 1, \ + 0, \ + sun4i_codec_linein_loopback_gain_scale), \ + SOC_SINGLE_TLV("FM-In Playback Gain", \ + SUN4I_CODEC_DAC_ACTL, \ + SUN4I_CODEC_DAC_ACTL_FMG, \ + 3, \ + 0, \ + sun4i_codec_fmin_loopback_gain_scale), \ + SOC_SINGLE_TLV("Mic-In Playback Gain", \ + SUN4I_CODEC_DAC_ACTL, \ + SUN4I_CODEC_DAC_ACTL_MICG, \ + 7, \ + 0, \ + sun4i_codec_micin_loopback_gain_scale), \ + SOC_SINGLE_TLV("Mic1-In Boost Switch", \ + SUN4I_CODEC_ADC_ACTL, \ + SUN4I_CODEC_ADC_ACTL_PREG1EN, \ + 1, \ + 0, \ + NULL), \ + SOC_SINGLE_TLV("Mic2-In Boost Switch", \ + SUN4I_CODEC_ADC_ACTL, \ + SUN4I_CODEC_ADC_ACTL_PREG2EN, \ + 1, \ + 0, \ + NULL) static const struct snd_kcontrol_new sun4i_codec_widgets[] = { - SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL, - SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0, - sun4i_codec_pa_volume_scale), + SUN4I_COMMON_CODEC_WIDGETS, + SOC_SINGLE_TLV("Mic1-In Boost Gain", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG1_A10, + 3, + 0, + sun4i_codec_micin_preamp_gain_scale_a10), + SOC_SINGLE_TLV("Mic2-In Boost Gain", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG2_A10, + 3, + 0, + sun4i_codec_micin_preamp_gain_scale_a10), }; static const struct snd_kcontrol_new sun4i_codec_left_mixer_controls[] = { SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_LDACLMIXS, 1, 0), + SOC_DAPM_SINGLE("Left Line-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LLNS, 1, 0), + SOC_DAPM_SINGLE("Left FM-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LFMS, 1, 0), + SOC_DAPM_SINGLE("Mic1-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC1LS, 1, 0), + SOC_DAPM_SINGLE("Mic2-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC2LS, 1, 0), }; static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = { @@ -419,6 +493,14 @@ static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = { SUN4I_CODEC_DAC_ACTL_RDACRMIXS, 1, 0), SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_LDACRMIXS, 1, 0), + SOC_DAPM_SINGLE("Right Line-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_RLNS, 1, 0), + SOC_DAPM_SINGLE("Right FM-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_RFMS, 1, 0), + SOC_DAPM_SINGLE("Mic1-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC1RS, 1, 0), + SOC_DAPM_SINGLE("Mic2-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC2RS, 1, 0), }; static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = { @@ -462,8 +544,16 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HP Right"), SND_SOC_DAPM_OUTPUT("HP Left"), + + SND_SOC_DAPM_INPUT("Line-In Right"), + SND_SOC_DAPM_INPUT("Line-In Left"), + SND_SOC_DAPM_INPUT("FM-In Right"), + SND_SOC_DAPM_INPUT("FM-In Left"), + SND_SOC_DAPM_INPUT("Mic1-In"), + SND_SOC_DAPM_INPUT("Mic2-In"), }; +/* {sink, control, source} */ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { /* Left DAC Routes */ { "Left DAC", NULL, "DAC" }, @@ -490,9 +580,23 @@ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" }, { "HP Right", NULL, "Pre-Amplifier Mute" }, { "HP Left", NULL, "Pre-Amplifier Mute" }, + + /* Line-In, FM-In */ + { "Right Mixer", "Right Line-In Playback Switch", "Line-In Right" }, + { "Left Mixer", "Left Line-In Playback Switch", "Line-In Left" }, + { "Right Mixer", "Right FM-In Playback Switch", "FM-In Right" }, + { "Left Mixer", "Left FM-In Playback Switch", "FM-In Left" }, + + /* Mic1-In */ + { "Right Mixer", "Mic1-In Playback Switch", "Mic1-In" }, + { "Left Mixer", "Mic1-In Playback Switch", "Mic1-In" }, + + /* Mic2-In */ + { "Right Mixer", "Mic2-In Playback Switch", "Mic2-In" }, + { "Left Mixer", "Mic2-In Playback Switch", "Mic2-In" }, }; -static struct snd_soc_codec_driver sun4i_codec_codec = { +static const struct snd_soc_codec_driver sun4i_codec_codec = { .controls = sun4i_codec_widgets, .num_controls = ARRAY_SIZE(sun4i_codec_widgets), .dapm_widgets = sun4i_codec_dapm_widgets, @@ -533,13 +637,6 @@ static struct snd_soc_dai_driver dummy_cpu_dai = { }, }; -static const struct regmap_config sun4i_codec_regmap_config = { - .reg_bits = 32, - .reg_stride = 4, - .val_bits = 32, - .max_register = SUN4I_CODEC_AC_MIC_PHONE_CAL, -}; - static const struct of_device_id sun4i_codec_of_match[] = { { .compatible = "allwinner,sun4i-a10-codec" }, { .compatible = "allwinner,sun7i-a20-codec" }, @@ -586,6 +683,56 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) return card; }; +/* sun7i-specific things: */ +/* MIC_PHONE_CAL register offsets and bit fields (A20 only) */ +#define SUN7I_CODEC_AC_MIC_PHONE_CAL (0x3c) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG1 (29) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG2 (26) +/* note: no idea where the output pins for the following are. */ +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTG (5) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTEN (4) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS3 (3) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS2 (2) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS1 (1) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS0 (0) + +static DECLARE_TLV_DB_RANGE(sun7i_codec_micin_preamp_gain_scale, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0) +); + +static const struct snd_kcontrol_new sun7i_codec_widgets[] = { + SUN4I_COMMON_CODEC_WIDGETS, + SOC_SINGLE_TLV("Mic1-In Boost Gain", + SUN7I_CODEC_AC_MIC_PHONE_CAL, + SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG1, + 7, + 0, + sun7i_codec_micin_preamp_gain_scale), + SOC_SINGLE_TLV("Mic2-In Boost Gain", + SUN7I_CODEC_AC_MIC_PHONE_CAL, + SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG2, + 7, + 0, + sun7i_codec_micin_preamp_gain_scale), +}; + +static const struct snd_soc_codec_driver sun7i_codec_codec = { + .controls = sun7i_codec_widgets, + .num_controls = ARRAY_SIZE(sun7i_codec_widgets), + .dapm_widgets = sun4i_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sun4i_codec_dapm_widgets), + .dapm_routes = sun4i_codec_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sun4i_codec_dapm_routes), +}; +static const struct regmap_config sun4i_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN7I_CODEC_AC_MIC_PHONE_CAL, +}; +/* end sun7i-specific things */ + static int sun4i_codec_probe(struct platform_device *pdev) { struct snd_soc_card *card; @@ -593,6 +740,7 @@ static int sun4i_codec_probe(struct platform_device *pdev) struct resource *res; void __iomem *base; int ret; + const struct snd_soc_codec_driver* codec_codec; scodec = devm_kzalloc(&pdev->dev, sizeof(*scodec), GFP_KERNEL); if (!scodec) @@ -638,7 +786,12 @@ static int sun4i_codec_probe(struct platform_device *pdev) scodec->playback_dma_data.maxburst = 4; scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; - ret = snd_soc_register_codec(&pdev->dev, &sun4i_codec_codec, + if (of_device_is_compatible(pdev->dev.of_node, + "allwinner,sun7i-a20-codec")) + codec_codec = &sun7i_codec_codec; + else + codec_codec = &sun4i_codec_codec; + ret = snd_soc_register_codec(&pdev->dev, codec_codec, &sun4i_codec_dai, 1); if (ret) { dev_err(&pdev->dev, "Failed to register our codec\n");

2 6

[alsa-devel] [PATCH 1/8] ASoC: max98357a: Add ACPI ID for Maxim
by yang.a.fang＠intel.com 21 Dec '15

21 Dec '15

From: Rohit Ainapure <rohit.m.ainapure(a)intel.com> Adding ACPI ID "MX98357A" for the MAXIM 98357A amp. Signed-off-by: Rohit Ainapure <rohit.m.ainapure(a)intel.com> Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/max98357a.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index f5e3dce..5b1dfb1 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -12,6 +12,7 @@ * max98357a.c -- MAX98357A ALSA SoC Codec driver */ +#include <linux/acpi.h> #include <linux/device.h> #include <linux/err.h> #include <linux/gpio.h> @@ -123,10 +124,19 @@ static const struct of_device_id max98357a_device_id[] = { MODULE_DEVICE_TABLE(of, max98357a_device_id); #endif +#ifdef CONFIG_ACPI +static const struct acpi_device_id max98357a_acpi_match[] = { + { "MX98357A", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, max98357a_acpi_match); +#endif + static struct platform_driver max98357a_platform_driver = { .driver = { .name = "max98357a", .of_match_table = of_match_ptr(max98357a_device_id), + .acpi_match_table = ACPI_PTR(max98357a_acpi_match), }, .probe = max98357a_platform_probe, .remove = max98357a_platform_remove, -- 1.7.9.5

4 17

[alsa-devel] [PATCH v8 0/2] ASoc: sun4i-codec: Add FM, Line and Mic inputs
by Danny Milosavljevic 21 Dec '15

21 Dec '15

Hi, this is the eighth version of the patch set that adds inputs to sun4i-codec. Changes compared to v7 are: - fix the routes for line and mic capturing. The patches are on top of the branch "sunxi-next" in <git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git>. Regards, Danny --- Danny Milosavljevic (2): sun4i-codec: Add FM, Line and Mic inputs sun4i-codec: make it possible to use different codec_widgets for A10 and A20. b/sound/soc/sunxi/sun4i-codec.c | 230 +++++++++++++++++++++++++++++++++++++--- 1 file changed, 215 insertions(+), 15 deletions(-)

1 0

[alsa-devel] [GIT PULL] Add get_eld audio component for i915/HD-audio
by Takashi Iwai 21 Dec '15

21 Dec '15

Hi Daniel, please pull the get_eld op addition from: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git tags/drm-i915-get-eld The topmost commit is 0bdf5a05647a66dcc6394986e061daeac9b1cf96 ---------------------------------------------------------------- Add get_eld audio component for i915/HD-audio Here are the patchset to add get_eld op to audio component for communicating more directly between i915 and HD-audio. Currently, the HDMI/DP audio status and ELD are notified and obtained via the hardware-level communication over HD-audio unsolicited event and verbs although the graphics driver holds the exactly same information. As we already have a notification via audio component, this is another step forward; namely, the audio driver may fetch directly the audio status and ELD via the new component op. The commits are based on Dave's latest drm-next branch. ---------------------------------------------------------------- Takashi Iwai (2): drm/i915: Add get_eld audio component drm/i915: Add reverse mapping between port and intel_encoder --- drivers/gpu/drm/i915/i915_drv.h | 2 + drivers/gpu/drm/i915/intel_audio.c | 79 ++++++++++++++++++++++++++------------ drivers/gpu/drm/i915/intel_ddi.c | 1 + drivers/gpu/drm/i915/intel_dp.c | 1 + drivers/gpu/drm/i915/intel_drv.h | 2 + drivers/gpu/drm/i915/intel_hdmi.c | 2 + include/drm/i915_component.h | 14 +++++++ 7 files changed, 77 insertions(+), 24 deletions(-)

2 3

[alsa-devel] [PATCH v4 0/5] ALSA: hda - hdmi support dynamic pcm assignment
by libin.yang＠linux.intel.com 21 Dec '15

21 Dec '15

From: Libin Yang <libin.yang(a)linux.intel.com> The patches are trying to support dynamically binding PCM to pin in HDMI/DP audio. The pin will not bind to any PCM if there is not monitor connected. When there is a monitor connected to the port (pin), driver will try to find a proper PCM to assign to the port and setup the pin if necessary. Libin Yang (5): ALSA: hda - hdmi begin to support dynamic PCM assignment ALSA: hda - hdmi playback without monitor in dynamic pcm bind mode ALSA: hda - hdmi operate spdif based on pcm ALSA: hda - hdmi dynamically bind PCM to pin when monitor hotplug ALSA: hda - hdmi setup pin when monitor hotplug in pcm dynamic assignment mode sound/pci/hda/patch_hdmi.c | 374 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 351 insertions(+), 23 deletions(-) -- 1.9.1

4 18

Re: [alsa-devel] [Intel-gfx] [PATCH V4 2/2] drm/i915: start adding dp mst audio
by Libin Yang 21 Dec '15

21 Dec '15

Add Takashi and ALSA mail list. On 12/10/2015 05:02 PM, Daniel Vetter wrote: > On Tue, Dec 08, 2015 at 04:01:20PM +0800, Libin Yang wrote: >> Hi all, >> >> Any comments on the patches? > > Sorry, simply fell through the cracks since Ander is on vacation. Takashi > is working on some cleanup patches to have a port->encoder mapping for the > audio side of i915. His patch cleans up all the existing audio code in > i915, but please work together with him to align mst code with the new > style. > > Both patches queued for next. Yes, I have seen Takashi's patches. I will check the patches. Best Regards, Libin > > Thanks, Daniel > >> >> Best Regards, >> Libin >> >> On 12/02/2015 02:09 PM, libin.yang(a)linux.intel.com wrote: >>> From: Libin Yang <libin.yang(a)linux.intel.com> >>> >>> This patch adds support for DP MST audio in i915. >>> >>> Enable audio codec when DP MST is enabled if has_audio flag is set. >>> Disable audio codec when DP MST is disabled if has_audio flag is set. >>> >>> Another separated patches to support DP MST audio will be implemented >>> in audio driver. >>> >>> Reviewed-by: Ander Conselvan de Oliveira <conselvan2(a)gmail.com> >>> Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> >>> Signed-off-by: Dave Airlie <airlied(a)redhat.com> >>> --- >>> drivers/gpu/drm/i915/i915_debugfs.c | 16 ++++++++++++++++ >>> drivers/gpu/drm/i915/intel_audio.c | 9 ++++++--- >>> drivers/gpu/drm/i915/intel_ddi.c | 20 +++++++++++++++----- >>> drivers/gpu/drm/i915/intel_dp_mst.c | 22 ++++++++++++++++++++++ >>> drivers/gpu/drm/i915/intel_drv.h | 2 ++ >>> 5 files changed, 61 insertions(+), 8 deletions(-) >>> >>> diff --git a/drivers/gpu/drm/i915/i915_debugfs.c b/drivers/gpu/drm/i915/i915_debugfs.c >>> index bfd57fb..8387638 100644 >>> --- a/drivers/gpu/drm/i915/i915_debugfs.c >>> +++ b/drivers/gpu/drm/i915/i915_debugfs.c >>> @@ -2850,6 +2850,20 @@ static void intel_dp_info(struct seq_file *m, >>> intel_panel_info(m, &intel_connector->panel); >>> } >>> >>> +static void intel_dp_mst_info(struct seq_file *m, >>> + struct intel_connector *intel_connector) >>> +{ >>> + struct intel_encoder *intel_encoder = intel_connector->encoder; >>> + struct intel_dp_mst_encoder *intel_mst = >>> + enc_to_mst(&intel_encoder->base); >>> + struct intel_digital_port *intel_dig_port = intel_mst->primary; >>> + struct intel_dp *intel_dp = &intel_dig_port->dp; >>> + bool has_audio = drm_dp_mst_port_has_audio(&intel_dp->mst_mgr, >>> + intel_connector->port); >>> + >>> + seq_printf(m, "\taudio support: %s\n", yesno(has_audio)); >>> +} >>> + >>> static void intel_hdmi_info(struct seq_file *m, >>> struct intel_connector *intel_connector) >>> { >>> @@ -2893,6 +2907,8 @@ static void intel_connector_info(struct seq_file *m, >>> intel_hdmi_info(m, intel_connector); >>> else if (intel_encoder->type == INTEL_OUTPUT_LVDS) >>> intel_lvds_info(m, intel_connector); >>> + else if (intel_encoder->type == INTEL_OUTPUT_DP_MST) >>> + intel_dp_mst_info(m, intel_connector); >>> } >>> >>> seq_printf(m, "\tmodes:\n"); >>> diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c >>> index 9aa83e7..5ad2e66 100644 >>> --- a/drivers/gpu/drm/i915/intel_audio.c >>> +++ b/drivers/gpu/drm/i915/intel_audio.c >>> @@ -262,7 +262,8 @@ static void hsw_audio_codec_disable(struct intel_encoder *encoder) >>> tmp |= AUD_CONFIG_N_PROG_ENABLE; >>> tmp &= ~AUD_CONFIG_UPPER_N_MASK; >>> tmp &= ~AUD_CONFIG_LOWER_N_MASK; >>> - if (intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DISPLAYPORT)) >>> + if (intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DISPLAYPORT) || >>> + intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DP_MST)) >>> tmp |= AUD_CONFIG_N_VALUE_INDEX; >>> I915_WRITE(HSW_AUD_CFG(pipe), tmp); >>> >>> @@ -474,7 +475,8 @@ static void ilk_audio_codec_enable(struct drm_connector *connector, >>> tmp &= ~AUD_CONFIG_N_VALUE_INDEX; >>> tmp &= ~AUD_CONFIG_N_PROG_ENABLE; >>> tmp &= ~AUD_CONFIG_PIXEL_CLOCK_HDMI_MASK; >>> - if (intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DISPLAYPORT)) >>> + if (intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DISPLAYPORT) || >>> + intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DP_MST)) >>> tmp |= AUD_CONFIG_N_VALUE_INDEX; >>> else >>> tmp |= audio_config_hdmi_pixel_clock(adjusted_mode); >>> @@ -512,7 +514,8 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) >>> >>> /* ELD Conn_Type */ >>> connector->eld[5] &= ~(3 << 2); >>> - if (intel_pipe_has_type(crtc, INTEL_OUTPUT_DISPLAYPORT)) >>> + if (intel_pipe_has_type(crtc, INTEL_OUTPUT_DISPLAYPORT) || >>> + intel_pipe_has_type(crtc, INTEL_OUTPUT_DP_MST)) >>> connector->eld[5] |= (1 << 2); >>> >>> connector->eld[6] = drm_av_sync_delay(connector, adjusted_mode) / 2; >>> diff --git a/drivers/gpu/drm/i915/intel_ddi.c b/drivers/gpu/drm/i915/intel_ddi.c >>> index 76ce7c2..bc7fffb 100644 >>> --- a/drivers/gpu/drm/i915/intel_ddi.c >>> +++ b/drivers/gpu/drm/i915/intel_ddi.c >>> @@ -3108,6 +3108,19 @@ void intel_ddi_fdi_disable(struct drm_crtc *crtc) >>> I915_WRITE(FDI_RX_CTL(PIPE_A), val); >>> } >>> >>> +bool intel_ddi_is_audio_enabled(struct drm_i915_private *dev_priv, >>> + struct intel_crtc *intel_crtc) >>> +{ >>> + u32 temp; >>> + >>> + if (intel_display_power_is_enabled(dev_priv, POWER_DOMAIN_AUDIO)) { >>> + temp = I915_READ(HSW_AUD_PIN_ELD_CP_VLD); >>> + if (temp & AUDIO_OUTPUT_ENABLE(intel_crtc->pipe)) >>> + return true; >>> + } >>> + return false; >>> +} >>> + >>> void intel_ddi_get_config(struct intel_encoder *encoder, >>> struct intel_crtc_state *pipe_config) >>> { >>> @@ -3168,11 +3181,8 @@ void intel_ddi_get_config(struct intel_encoder *encoder, >>> break; >>> } >>> >>> - if (intel_display_power_is_enabled(dev_priv, POWER_DOMAIN_AUDIO)) { >>> - temp = I915_READ(HSW_AUD_PIN_ELD_CP_VLD); >>> - if (temp & AUDIO_OUTPUT_ENABLE(intel_crtc->pipe)) >>> - pipe_config->has_audio = true; >>> - } >>> + pipe_config->has_audio = >>> + intel_ddi_is_audio_enabled(dev_priv, intel_crtc); >>> >>> if (encoder->type == INTEL_OUTPUT_EDP && dev_priv->vbt.edp_bpp && >>> pipe_config->pipe_bpp > dev_priv->vbt.edp_bpp) { >>> diff --git a/drivers/gpu/drm/i915/intel_dp_mst.c b/drivers/gpu/drm/i915/intel_dp_mst.c >>> index 8c4e7df..8b608c2 100644 >>> --- a/drivers/gpu/drm/i915/intel_dp_mst.c >>> +++ b/drivers/gpu/drm/i915/intel_dp_mst.c >>> @@ -78,6 +78,8 @@ static bool intel_dp_mst_compute_config(struct intel_encoder *encoder, >>> return false; >>> } >>> >>> + if (drm_dp_mst_port_has_audio(&intel_dp->mst_mgr, found->port)) >>> + pipe_config->has_audio = true; >>> mst_pbn = drm_dp_calc_pbn_mode(adjusted_mode->crtc_clock, bpp); >>> >>> pipe_config->pbn = mst_pbn; >>> @@ -102,6 +104,11 @@ static void intel_mst_disable_dp(struct intel_encoder *encoder) >>> struct intel_dp_mst_encoder *intel_mst = enc_to_mst(&encoder->base); >>> struct intel_digital_port *intel_dig_port = intel_mst->primary; >>> struct intel_dp *intel_dp = &intel_dig_port->dp; >>> + struct drm_device *dev = encoder->base.dev; >>> + struct drm_i915_private *dev_priv = dev->dev_private; >>> + struct drm_crtc *crtc = encoder->base.crtc; >>> + struct intel_crtc *intel_crtc = to_intel_crtc(crtc); >>> + >>> int ret; >>> >>> DRM_DEBUG_KMS("%d\n", intel_dp->active_mst_links); >>> @@ -112,6 +119,10 @@ static void intel_mst_disable_dp(struct intel_encoder *encoder) >>> if (ret) { >>> DRM_ERROR("failed to update payload %d\n", ret); >>> } >>> + if (intel_crtc->config->has_audio) { >>> + intel_audio_codec_disable(encoder); >>> + intel_display_power_put(dev_priv, POWER_DOMAIN_AUDIO); >>> + } >>> } >>> >>> static void intel_mst_post_disable_dp(struct intel_encoder *encoder) >>> @@ -208,6 +219,7 @@ static void intel_mst_enable_dp(struct intel_encoder *encoder) >>> struct intel_dp *intel_dp = &intel_dig_port->dp; >>> struct drm_device *dev = intel_dig_port->base.base.dev; >>> struct drm_i915_private *dev_priv = dev->dev_private; >>> + struct intel_crtc *crtc = to_intel_crtc(encoder->base.crtc); >>> enum port port = intel_dig_port->port; >>> int ret; >>> >>> @@ -220,6 +232,13 @@ static void intel_mst_enable_dp(struct intel_encoder *encoder) >>> ret = drm_dp_check_act_status(&intel_dp->mst_mgr); >>> >>> ret = drm_dp_update_payload_part2(&intel_dp->mst_mgr); >>> + >>> + if (crtc->config->has_audio) { >>> + DRM_DEBUG_DRIVER("Enabling DP audio on pipe %c\n", >>> + pipe_name(crtc->pipe)); >>> + intel_display_power_get(dev_priv, POWER_DOMAIN_AUDIO); >>> + intel_audio_codec_enable(encoder); >>> + } >>> } >>> >>> static bool intel_dp_mst_enc_get_hw_state(struct intel_encoder *encoder, >>> @@ -245,6 +264,9 @@ static void intel_dp_mst_enc_get_config(struct intel_encoder *encoder, >>> >>> pipe_config->has_dp_encoder = true; >>> >>> + pipe_config->has_audio = >>> + intel_ddi_is_audio_enabled(dev_priv, crtc); >>> + >>> temp = I915_READ(TRANS_DDI_FUNC_CTL(cpu_transcoder)); >>> if (temp & TRANS_DDI_PHSYNC) >>> flags |= DRM_MODE_FLAG_PHSYNC; >>> diff --git a/drivers/gpu/drm/i915/intel_drv.h b/drivers/gpu/drm/i915/intel_drv.h >>> index 1ffd8d5..0d2c59f 100644 >>> --- a/drivers/gpu/drm/i915/intel_drv.h >>> +++ b/drivers/gpu/drm/i915/intel_drv.h >>> @@ -1011,6 +1011,8 @@ void intel_ddi_set_pipe_settings(struct drm_crtc *crtc); >>> void intel_ddi_prepare_link_retrain(struct intel_dp *intel_dp); >>> bool intel_ddi_connector_get_hw_state(struct intel_connector *intel_connector); >>> void intel_ddi_fdi_disable(struct drm_crtc *crtc); >>> +bool intel_ddi_is_audio_enabled(struct drm_i915_private *dev_priv, >>> + struct intel_crtc *intel_crtc); >>> void intel_ddi_get_config(struct intel_encoder *encoder, >>> struct intel_crtc_state *pipe_config); >>> struct intel_encoder * >>> >> _______________________________________________ >> Intel-gfx mailing list >> Intel-gfx(a)lists.freedesktop.org >> http://lists.freedesktop.org/mailman/listinfo/intel-gfx >

3 9

[alsa-devel] [PATCH] ASoC: fsl_asrc: old version asrc only support even numbered channels
by Zidan Wang 21 Dec '15

21 Dec '15

The latest asrc's channel bits is 4 bit, the old asrc's channel bits is 3. For the old version asrc, channel should be a even number. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_asrc.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index be78dc9..484e46ca 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -113,7 +113,8 @@ static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair) if (index == ASRC_INVALID_PAIR) { dev_err(dev, "all pairs are busy now\n"); ret = -EBUSY; - } else if (asrc_priv->channel_avail < channels) { + } else if (asrc_priv->channel_avail < channels || + (asrc_priv->channel_bits < 4 && channels % 2 != 0)) { dev_err(dev, "can't afford required channels: %d\n", channels); ret = -EINVAL; } else { -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ASoC: fsl_asrc: sound is wrong after suspend/resume
by Zidan Wang 21 Dec '15

21 Dec '15

The register ASRCFG is volatile, but some bits need to be recovered after suspend/resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v1->v2: 1. fix indentation issue. 2. add a macro for the magic number. sound/soc/fsl/fsl_asrc.c | 7 +++++++ sound/soc/fsl/fsl_asrc.h | 6 ++++++ 2 files changed, 13 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 484e46ca..d964016 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -990,6 +990,9 @@ static int fsl_asrc_suspend(struct device *dev) { struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + regmap_read(asrc_priv->regmap, REG_ASRCFG, + &asrc_priv->regcache_cfg); + regcache_cache_only(asrc_priv->regmap, true); regcache_mark_dirty(asrc_priv->regmap); @@ -1010,6 +1013,10 @@ static int fsl_asrc_resume(struct device *dev) regcache_cache_only(asrc_priv->regmap, false); regcache_sync(asrc_priv->regmap); + regmap_update_bits(asrc_priv->regmap, REG_ASRCFG, + ASRCFG_NDPRi_ALL_MASK | ASRCFG_POSTMODi_ALL_MASK | + ASRCFG_PREMODi_ALL_MASK, asrc_priv->regcache_cfg); + /* Restart enabled pairs */ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, ASRCTR_ASRCEi_ALL_MASK, asrctr); diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index 68802cd..c119cb0 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -132,10 +132,13 @@ #define ASRCFG_INIRQi (1 << ASRCFG_INIRQi_SHIFT(i)) #define ASRCFG_NDPRi_SHIFT(i) (18 + i) #define ASRCFG_NDPRi_MASK(i) (1 << ASRCFG_NDPRi_SHIFT(i)) +#define ASRCFG_NDPRi_ALL_SHIFT 18 +#define ASRCFG_NDPRi_ALL_MASK (7 << ASRCFG_NDPRi_ALL_SHIFT) #define ASRCFG_NDPRi (1 << ASRCFG_NDPRi_SHIFT(i)) #define ASRCFG_POSTMODi_SHIFT(i) (8 + (i << 2)) #define ASRCFG_POSTMODi_WIDTH 2 #define ASRCFG_POSTMODi_MASK(i) (((1 << ASRCFG_POSTMODi_WIDTH) - 1) << ASRCFG_POSTMODi_SHIFT(i)) +#define ASRCFG_POSTMODi_ALL_MASK (ASRCFG_POSTMODi_MASK(0) | ASRCFG_POSTMODi_MASK(1) | ASRCFG_POSTMODi_MASK(2)) #define ASRCFG_POSTMOD(i, v) ((v) << ASRCFG_POSTMODi_SHIFT(i)) #define ASRCFG_POSTMODi_UP(i) (0 << ASRCFG_POSTMODi_SHIFT(i)) #define ASRCFG_POSTMODi_DCON(i) (1 << ASRCFG_POSTMODi_SHIFT(i)) @@ -143,6 +146,7 @@ #define ASRCFG_PREMODi_SHIFT(i) (6 + (i << 2)) #define ASRCFG_PREMODi_WIDTH 2 #define ASRCFG_PREMODi_MASK(i) (((1 << ASRCFG_PREMODi_WIDTH) - 1) << ASRCFG_PREMODi_SHIFT(i)) +#define ASRCFG_PREMODi_ALL_MASK (ASRCFG_PREMODi_MASK(0) | ASRCFG_PREMODi_MASK(1) | ASRCFG_PREMODi_MASK(2)) #define ASRCFG_PREMOD(i, v) ((v) << ASRCFG_PREMODi_SHIFT(i)) #define ASRCFG_PREMODi_UP(i) (0 << ASRCFG_PREMODi_SHIFT(i)) #define ASRCFG_PREMODi_DCON(i) (1 << ASRCFG_PREMODi_SHIFT(i)) @@ -453,6 +457,8 @@ struct fsl_asrc { int asrc_rate; int asrc_width; + + u32 regcache_cfg; }; extern struct snd_soc_platform_driver fsl_asrc_platform; -- 1.9.1

2 1

[alsa-devel] [PATCH v7 0/2] sun4i-codec: Add FM, Line and Mic inputs
by Danny Milosavljevic 21 Dec '15

21 Dec '15

Hi, this is the seventh version of the patch set that adds inputs to sun4i-codec. Changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top - sun7i-specific things (A20-specific things) are now less grouped-together. The patches are on top of the branch "sunxi-next" in <git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git>. Regards, Danny --- Danny Milosavljevic (2): sun4i-codec: Add FM, Line and Mic inputs sun4i-codec: make it possible to use different codec_widgets for A10 and A20. sound/soc/sunxi/sun4i-codec.c | 223 +++++++++++++++++++++++++++++++++++++---

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[alsa-devel] [PATCH v1 1/4] ALSA: fm801: explicitly free IRQ line
by Andy Shevchenko 20 Dec '15

20 Dec '15

Otherwise we will have a warning on ->remove() since device is a PCI one. WARNING: CPU: 4 PID: 1411 at /home/andy/prj/linux/fs/proc/generic.c:575 remove_proc_entry+0x137/0x160() remove_proc_entry: removing non-empty directory 'irq/21', leaking at least 'snd_fm801' Fixes: 5618955c4269 (ALSA: fm801: move to pcim_* and devm_* functions) Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- The patch series has been tested on SF64-PCR card. sound/pci/fm801.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 1fdd92b6..f57847c 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1165,6 +1165,8 @@ static int snd_fm801_free(struct fm801 *chip) cmdw |= 0x00c3; fm801_writew(chip, IRQ_MASK, cmdw); + devm_free_irq(&chip->pci->dev, chip->irq, chip); + __end_hw: #ifdef CONFIG_SND_FM801_TEA575X_BOOL if (!(chip->tea575x_tuner & TUNER_DISABLED)) { -- 2.6.4

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[alsa-devel] [PATCH V2] ALSA: oss: consolidate kmalloc/memset 0 call to kzalloc
by Nicholas Mc Guire 20 Dec '15

20 Dec '15

This is an API consolidation only. The use of kmalloc + memset to 0 is equivalent to kzalloc. Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- V2: somehow managed to insert exactly the same line being removed so this time actually do the intended kmalloc->kzalloc replacement Found by coccinelle script (relaxed version of scripts/coccinelle/api/alloc/kzalloc-simple.cocci) Patch was compile tested with: x86_64_defconfig CONFIG_SND_PCM_OSS=y Patch is against linux-next (localversion-next is -next-20150518) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index e557dbe..8cdd06f 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -851,7 +851,7 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) if (mutex_lock_interruptible(&runtime->oss.params_lock)) return -EINTR; - sw_params = kmalloc(sizeof(*sw_params), GFP_KERNEL); + sw_params = kzalloc(sizeof(*sw_params), GFP_KERNEL); params = kmalloc(sizeof(*params), GFP_KERNEL); sparams = kmalloc(sizeof(*sparams), GFP_KERNEL); if (!sw_params || !params || !sparams) { @@ -989,7 +989,6 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) goto failure; } - memset(sw_params, 0, sizeof(*sw_params)); if (runtime->oss.trigger) { sw_params->start_threshold = 1; } else { -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: rt5616: add rt5616 codec driver
by Bard Liao 19 Dec '15

19 Dec '15

This is the initial codec driver for rt5616. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5616.c | 1372 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5616.h | 1819 +++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 3199 insertions(+) create mode 100644 sound/soc/codecs/rt5616.c create mode 100644 sound/soc/codecs/rt5616.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c971371..38102a0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -95,6 +95,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM512x_SPI if SPI_MASTER select SND_SOC_RT286 if I2C select SND_SOC_RT298 if I2C + select SND_SOC_RT5616 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_RT5645 if I2C @@ -563,12 +564,14 @@ config SND_SOC_PCM512x_SPI config SND_SOC_RL6231 tristate + default y if SND_SOC_RT5616=y default y if SND_SOC_RT5640=y default y if SND_SOC_RT5645=y default y if SND_SOC_RT5651=y default y if SND_SOC_RT5659=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y + default m if SND_SOC_RT5616=m default m if SND_SOC_RT5640=m default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m @@ -591,6 +594,9 @@ config SND_SOC_RT298 tristate depends on I2C +config SND_SOC_RT5616 + tristate + config SND_SOC_RT5631 tristate "Realtek ALC5631/RT5631 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 813dd6d..fb846e2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -92,6 +92,7 @@ snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o +snd-soc-rt5616-objs := rt5616.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o @@ -295,6 +296,7 @@ obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o +obj-$(CONFIG_SND_SOC_RT5616) += snd-soc-rt5616.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c new file mode 100644 index 0000000..f4005cb --- /dev/null +++ b/sound/soc/codecs/rt5616.c @@ -0,0 +1,1372 @@ +/* + * rt5616.c -- RT5616 ALSA SoC audio codec driver + * + * Copyright 2015 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.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/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.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 "rl6231.h" +#include "rt5616.h" + +#define RT5616_PR_RANGE_BASE (0xff + 1) +#define RT5616_PR_SPACING 0x100 + +#define RT5616_PR_BASE (RT5616_PR_RANGE_BASE + (0 * RT5616_PR_SPACING)) + +static const struct regmap_range_cfg rt5616_ranges[] = { + { + .name = "PR", + .range_min = RT5616_PR_BASE, + .range_max = RT5616_PR_BASE + 0xf8, + .selector_reg = RT5616_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5616_PRIV_DATA, + .window_len = 0x1, + }, +}; + +static const struct reg_sequence init_list[] = { + {RT5616_PR_BASE + 0x3d, 0x3e00}, + {RT5616_PR_BASE + 0x25, 0x6110}, + {RT5616_PR_BASE + 0x20, 0x611f}, + {RT5616_PR_BASE + 0x21, 0x4040}, + {RT5616_PR_BASE + 0x23, 0x0004}, +}; +#define RT5616_INIT_REG_LEN ARRAY_SIZE(init_list) + +static const struct reg_default rt5616_reg[] = { + { 0x00, 0x0021 }, + { 0x02, 0xc8c8 }, + { 0x03, 0xc8c8 }, + { 0x05, 0x0000 }, + { 0x0d, 0x0000 }, + { 0x0f, 0x0808 }, + { 0x19, 0xafaf }, + { 0x1c, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x27, 0x7860 }, + { 0x29, 0x8080 }, + { 0x2a, 0x5252 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x006f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x006f }, + { 0x45, 0x6000 }, + { 0x4d, 0x0000 }, + { 0x4e, 0x0000 }, + { 0x4f, 0x0279 }, + { 0x50, 0x0000 }, + { 0x51, 0x0000 }, + { 0x52, 0x0279 }, + { 0x53, 0xf000 }, + { 0x61, 0x0000 }, + { 0x62, 0x0000 }, + { 0x63, 0x00c0 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x70, 0x8000 }, + { 0x73, 0x1104 }, + { 0x74, 0x0c00 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x8b, 0x0600 }, + { 0x8e, 0x0004 }, + { 0x8f, 0x1100 }, + { 0x90, 0x0000 }, + { 0x91, 0x0000 }, + { 0x92, 0x0000 }, + { 0x93, 0x2000 }, + { 0x94, 0x0200 }, + { 0x95, 0x0000 }, + { 0xb0, 0x2080 }, + { 0xb1, 0x0000 }, + { 0xb2, 0x0000 }, + { 0xb4, 0x2206 }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xb7, 0x0000 }, + { 0xbb, 0x0000 }, + { 0xbc, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x0000 }, + { 0xc0, 0x0100 }, + { 0xc1, 0x0000 }, + { 0xc2, 0x0000 }, + { 0xc8, 0x0000 }, + { 0xc9, 0x0000 }, + { 0xca, 0x0000 }, + { 0xcb, 0x0000 }, + { 0xcc, 0x0000 }, + { 0xcd, 0x0000 }, + { 0xce, 0x0000 }, + { 0xcf, 0x0013 }, + { 0xd0, 0x0680 }, + { 0xd1, 0x1c17 }, + { 0xd3, 0xb320 }, + { 0xd4, 0x0000 }, + { 0xd6, 0x0000 }, + { 0xd7, 0x0000 }, + { 0xd9, 0x0809 }, + { 0xda, 0x0000 }, + { 0xfa, 0x0010 }, + { 0xfb, 0x0000 }, + { 0xfc, 0x0000 }, + { 0xfe, 0x10ec }, + { 0xff, 0x6281 }, +}; + +struct rt5616_priv { + struct snd_soc_codec *codec; + struct delayed_work patch_work; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck[RT5616_AIFS]; + int bclk[RT5616_AIFS]; + int master[RT5616_AIFS]; + + int pll_src; + int pll_in; + int pll_out; + +}; + +static bool rt5616_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5616_ranges); i++) { + if (reg >= rt5616_ranges[i].range_min && + reg <= rt5616_ranges[i].range_max) { + return true; + } + } + + switch (reg) { + case RT5616_RESET: + case RT5616_PRIV_DATA: + case RT5616_EQ_CTRL1: + case RT5616_DRC_AGC_1: + case RT5616_IRQ_CTRL2: + case RT5616_INT_IRQ_ST: + case RT5616_PGM_REG_ARR1: + case RT5616_PGM_REG_ARR3: + case RT5616_VENDOR_ID: + case RT5616_DEVICE_ID: + return true; + default: + return false; + } +} + +static bool rt5616_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5616_ranges); i++) { + if (reg >= rt5616_ranges[i].range_min && + reg <= rt5616_ranges[i].range_max) { + return true; + } + } + + switch (reg) { + case RT5616_RESET: + case RT5616_VERSION_ID: + case RT5616_VENDOR_ID: + case RT5616_DEVICE_ID: + case RT5616_HP_VOL: + case RT5616_LOUT_CTRL1: + case RT5616_LOUT_CTRL2: + case RT5616_IN1_IN2: + case RT5616_INL1_INR1_VOL: + case RT5616_DAC1_DIG_VOL: + case RT5616_ADC_DIG_VOL: + case RT5616_ADC_BST_VOL: + case RT5616_STO1_ADC_MIXER: + case RT5616_AD_DA_MIXER: + case RT5616_STO_DAC_MIXER: + case RT5616_REC_L1_MIXER: + case RT5616_REC_L2_MIXER: + case RT5616_REC_R1_MIXER: + case RT5616_REC_R2_MIXER: + case RT5616_HPO_MIXER: + case RT5616_OUT_L1_MIXER: + case RT5616_OUT_L2_MIXER: + case RT5616_OUT_L3_MIXER: + case RT5616_OUT_R1_MIXER: + case RT5616_OUT_R2_MIXER: + case RT5616_OUT_R3_MIXER: + case RT5616_LOUT_MIXER: + case RT5616_PWR_DIG1: + case RT5616_PWR_DIG2: + case RT5616_PWR_ANLG1: + case RT5616_PWR_ANLG2: + case RT5616_PWR_MIXER: + case RT5616_PWR_VOL: + case RT5616_PRIV_INDEX: + case RT5616_PRIV_DATA: + case RT5616_I2S1_SDP: + case RT5616_ADDA_CLK1: + case RT5616_ADDA_CLK2: + case RT5616_GLB_CLK: + case RT5616_PLL_CTRL1: + case RT5616_PLL_CTRL2: + case RT5616_HP_OVCD: + case RT5616_DEPOP_M1: + case RT5616_DEPOP_M2: + case RT5616_DEPOP_M3: + case RT5616_CHARGE_PUMP: + case RT5616_PV_DET_SPK_G: + case RT5616_MICBIAS: + case RT5616_A_JD_CTL1: + case RT5616_A_JD_CTL2: + case RT5616_EQ_CTRL1: + case RT5616_EQ_CTRL2: + case RT5616_WIND_FILTER: + case RT5616_DRC_AGC_1: + case RT5616_DRC_AGC_2: + case RT5616_DRC_AGC_3: + case RT5616_SVOL_ZC: + case RT5616_JD_CTRL1: + case RT5616_JD_CTRL2: + case RT5616_IRQ_CTRL1: + case RT5616_IRQ_CTRL2: + case RT5616_INT_IRQ_ST: + case RT5616_GPIO_CTRL1: + case RT5616_GPIO_CTRL2: + case RT5616_GPIO_CTRL3: + case RT5616_PGM_REG_ARR1: + case RT5616_PGM_REG_ARR2: + case RT5616_PGM_REG_ARR3: + case RT5616_PGM_REG_ARR4: + case RT5616_PGM_REG_ARR5: + case RT5616_SCB_FUNC: + case RT5616_SCB_CTRL: + case RT5616_BASE_BACK: + case RT5616_MP3_PLUS1: + case RT5616_MP3_PLUS2: + case RT5616_ADJ_HPF_CTRL1: + case RT5616_ADJ_HPF_CTRL2: + case RT5616_HP_CALIB_AMP_DET: + case RT5616_HP_CALIB2: + case RT5616_SV_ZCD1: + case RT5616_SV_ZCD2: + case RT5616_D_MISC: + case RT5616_DUMMY2: + case RT5616_DUMMY3: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static unsigned int bst_tlv[] = { + TLV_DB_RANGE_HEAD(7), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), +}; + +static const struct snd_kcontrol_new rt5616_snd_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE("HP Playback Switch", RT5616_HP_VOL, + RT5616_L_MUTE_SFT, RT5616_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("HP Playback Volume", RT5616_HP_VOL, + RT5616_L_VOL_SFT, RT5616_R_VOL_SFT, 39, 1, out_vol_tlv), + /* OUTPUT Control */ + SOC_DOUBLE("OUT Playback Switch", RT5616_LOUT_CTRL1, + RT5616_L_MUTE_SFT, RT5616_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("OUT Channel Switch", RT5616_LOUT_CTRL1, + RT5616_VOL_L_SFT, RT5616_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5616_LOUT_CTRL1, + RT5616_L_VOL_SFT, RT5616_R_VOL_SFT, 39, 1, out_vol_tlv), + + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5616_DAC1_DIG_VOL, + RT5616_L_VOL_SFT, RT5616_R_VOL_SFT, + 175, 0, dac_vol_tlv), + /* IN1/IN2 Control */ + SOC_SINGLE_TLV("IN1 Boost", RT5616_IN1_IN2, + RT5616_BST_SFT1, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN2 Boost", RT5616_IN1_IN2, + RT5616_BST_SFT2, 8, 0, bst_tlv), + /* INL/INR Volume Control */ + SOC_DOUBLE_TLV("IN Capture Volume", RT5616_INL1_INR1_VOL, + RT5616_INL_VOL_SFT, RT5616_INR_VOL_SFT, + 31, 1, in_vol_tlv), + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5616_ADC_DIG_VOL, + RT5616_L_MUTE_SFT, RT5616_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5616_ADC_DIG_VOL, + RT5616_L_VOL_SFT, RT5616_R_VOL_SFT, + 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("ADC Boost Gain", RT5616_ADC_BST_VOL, + RT5616_ADC_L_BST_SFT, RT5616_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), +}; + +static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + + val = snd_soc_read(snd_soc_dapm_to_codec(source->dapm), RT5616_GLB_CLK); + val &= RT5616_SCLK_SRC_MASK; + if (val == RT5616_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5616_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5616_STO1_ADC_MIXER, + RT5616_M_STO1_ADC_L1_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5616_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5616_STO1_ADC_MIXER, + RT5616_M_STO1_ADC_R1_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5616_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5616_AD_DA_MIXER, + RT5616_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INF1 Switch", RT5616_AD_DA_MIXER, + RT5616_M_IF1_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5616_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5616_AD_DA_MIXER, + RT5616_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INF1 Switch", RT5616_AD_DA_MIXER, + RT5616_M_IF1_DAC_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5616_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5616_STO_DAC_MIXER, + RT5616_M_DAC_L1_MIXL_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5616_STO_DAC_MIXER, + RT5616_M_DAC_R1_MIXL_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5616_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5616_STO_DAC_MIXER, + RT5616_M_DAC_R1_MIXR_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5616_STO_DAC_MIXER, + RT5616_M_DAC_L1_MIXR_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5616_rec_l_mix[] = { + SOC_DAPM_SINGLE("INL1 Switch", RT5616_REC_L2_MIXER, + RT5616_M_IN1_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5616_REC_L2_MIXER, + RT5616_M_BST2_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5616_REC_L2_MIXER, + RT5616_M_BST1_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5616_rec_r_mix[] = { + SOC_DAPM_SINGLE("INR1 Switch", RT5616_REC_R2_MIXER, + RT5616_M_IN1_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5616_REC_R2_MIXER, + RT5616_M_BST2_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5616_REC_R2_MIXER, + RT5616_M_BST1_RM_R_SFT, 1, 1), +}; + +/* Analog Output Mixer */ + +static const struct snd_kcontrol_new rt5616_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5616_OUT_L3_MIXER, + RT5616_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5616_OUT_L3_MIXER, + RT5616_M_BST2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL1 Switch", RT5616_OUT_L3_MIXER, + RT5616_M_IN1_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXL Switch", RT5616_OUT_L3_MIXER, + RT5616_M_RM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5616_OUT_L3_MIXER, + RT5616_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5616_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5616_OUT_R3_MIXER, + RT5616_M_BST2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5616_OUT_R3_MIXER, + RT5616_M_BST1_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR1 Switch", RT5616_OUT_R3_MIXER, + RT5616_M_IN1_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXR Switch", RT5616_OUT_R3_MIXER, + RT5616_M_RM_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5616_OUT_R3_MIXER, + RT5616_M_DAC_R1_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5616_hpo_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5616_HPO_MIXER, + RT5616_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPVOL Switch", RT5616_HPO_MIXER, + RT5616_M_HPVOL_HM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5616_lout_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5616_LOUT_MIXER, + RT5616_M_DAC_L1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5616_LOUT_MIXER, + RT5616_M_DAC_R1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTVOL L Switch", RT5616_LOUT_MIXER, + RT5616_M_OV_L_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTVOL R Switch", RT5616_LOUT_MIXER, + RT5616_M_OV_R_LM_SFT, 1, 1), +}; + +static int rt5616_adc_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_POST_PMU: + snd_soc_update_bits(codec, RT5616_ADC_DIG_VOL, + RT5616_L_MUTE | RT5616_R_MUTE, 0); + break; + + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, RT5616_ADC_DIG_VOL, + RT5616_L_MUTE | RT5616_R_MUTE, + RT5616_L_MUTE | RT5616_R_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5616_charge_pump_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_POST_PMU: + /* depop parameters */ + snd_soc_update_bits(codec, RT5616_DEPOP_M2, + RT5616_DEPOP_MASK, RT5616_DEPOP_MAN); + snd_soc_update_bits(codec, RT5616_DEPOP_M1, + RT5616_HP_CP_MASK | RT5616_HP_SG_MASK | + RT5616_HP_CB_MASK, RT5616_HP_CP_PU | + RT5616_HP_SG_DIS | RT5616_HP_CB_PU); + snd_soc_write(codec, RT5616_PR_BASE + + RT5616_HP_DCC_INT1, 0x9f00); + /* headphone amp power on */ + snd_soc_update_bits(codec, RT5616_PWR_ANLG1, + RT5616_PWR_FV1 | RT5616_PWR_FV2, 0); + snd_soc_update_bits(codec, RT5616_PWR_VOL, + RT5616_PWR_HV_L | RT5616_PWR_HV_R, + RT5616_PWR_HV_L | RT5616_PWR_HV_R); + snd_soc_update_bits(codec, RT5616_PWR_ANLG1, + RT5616_PWR_HP_L | RT5616_PWR_HP_R | + RT5616_PWR_HA, RT5616_PWR_HP_L | + RT5616_PWR_HP_R | RT5616_PWR_HA); + msleep(50); + snd_soc_update_bits(codec, RT5616_PWR_ANLG1, + RT5616_PWR_FV1 | RT5616_PWR_FV2, + RT5616_PWR_FV1 | RT5616_PWR_FV2); + + snd_soc_update_bits(codec, RT5616_CHARGE_PUMP, + RT5616_PM_HP_MASK, RT5616_PM_HP_HV); + snd_soc_update_bits(codec, RT5616_PR_BASE + + RT5616_CHOP_DAC_ADC, 0x0200, 0x0200); + snd_soc_update_bits(codec, RT5616_DEPOP_M1, + RT5616_HP_CO_MASK | RT5616_HP_SG_MASK, + RT5616_HP_CO_EN | RT5616_HP_SG_EN); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5616_PR_BASE + + RT5616_CHOP_DAC_ADC, 0x0200, 0x0); + snd_soc_update_bits(codec, RT5616_DEPOP_M1, + RT5616_HP_SG_MASK | RT5616_HP_L_SMT_MASK | + RT5616_HP_R_SMT_MASK, RT5616_HP_SG_DIS | + RT5616_HP_L_SMT_DIS | RT5616_HP_R_SMT_DIS); + /* headphone amp power down */ + snd_soc_update_bits(codec, RT5616_DEPOP_M1, + RT5616_SMT_TRIG_MASK | RT5616_HP_CD_PD_MASK | + RT5616_HP_CO_MASK | RT5616_HP_CP_MASK | + RT5616_HP_SG_MASK | RT5616_HP_CB_MASK, + RT5616_SMT_TRIG_DIS | RT5616_HP_CD_PD_EN | + RT5616_HP_CO_DIS | RT5616_HP_CP_PD | + RT5616_HP_SG_EN | RT5616_HP_CB_PD); + snd_soc_update_bits(codec, RT5616_PWR_ANLG1, + RT5616_PWR_HP_L | RT5616_PWR_HP_R | + RT5616_PWR_HA, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt5616_hp_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_POST_PMU: + /* headphone unmute sequence */ + snd_soc_update_bits(codec, RT5616_DEPOP_M3, + RT5616_CP_FQ1_MASK | RT5616_CP_FQ2_MASK | + RT5616_CP_FQ3_MASK, + (RT5616_CP_FQ_192_KHZ << RT5616_CP_FQ1_SFT) | + (RT5616_CP_FQ_12_KHZ << RT5616_CP_FQ2_SFT) | + (RT5616_CP_FQ_192_KHZ << RT5616_CP_FQ3_SFT)); + snd_soc_write(codec, RT5616_PR_BASE + + RT5616_MAMP_INT_REG2, 0xfc00); + snd_soc_update_bits(codec, RT5616_DEPOP_M1, + RT5616_SMT_TRIG_MASK, RT5616_SMT_TRIG_EN); + snd_soc_update_bits(codec, RT5616_DEPOP_M1, + RT5616_RSTN_MASK, RT5616_RSTN_EN); + snd_soc_update_bits(codec, RT5616_DEPOP_M1, + RT5616_RSTN_MASK | RT5616_HP_L_SMT_MASK | + RT5616_HP_R_SMT_MASK, RT5616_RSTN_DIS | + RT5616_HP_L_SMT_EN | RT5616_HP_R_SMT_EN); + snd_soc_update_bits(codec, RT5616_HP_VOL, + RT5616_L_MUTE | RT5616_R_MUTE, 0); + msleep(100); + snd_soc_update_bits(codec, RT5616_DEPOP_M1, + RT5616_HP_SG_MASK | RT5616_HP_L_SMT_MASK | + RT5616_HP_R_SMT_MASK, RT5616_HP_SG_DIS | + RT5616_HP_L_SMT_DIS | RT5616_HP_R_SMT_DIS); + msleep(20); + snd_soc_update_bits(codec, RT5616_HP_CALIB_AMP_DET, + RT5616_HPD_PS_MASK, RT5616_HPD_PS_EN); + break; + + case SND_SOC_DAPM_PRE_PMD: + /* headphone mute sequence */ + snd_soc_update_bits(codec, RT5616_DEPOP_M3, + RT5616_CP_FQ1_MASK | RT5616_CP_FQ2_MASK | + RT5616_CP_FQ3_MASK, + (RT5616_CP_FQ_96_KHZ << RT5616_CP_FQ1_SFT) | + (RT5616_CP_FQ_12_KHZ << RT5616_CP_FQ2_SFT) | + (RT5616_CP_FQ_96_KHZ << RT5616_CP_FQ3_SFT)); + snd_soc_write(codec, RT5616_PR_BASE + + RT5616_MAMP_INT_REG2, 0xfc00); + snd_soc_update_bits(codec, RT5616_DEPOP_M1, + RT5616_HP_SG_MASK, RT5616_HP_SG_EN); + snd_soc_update_bits(codec, RT5616_DEPOP_M1, + RT5616_RSTP_MASK, RT5616_RSTP_EN); + snd_soc_update_bits(codec, RT5616_DEPOP_M1, + RT5616_RSTP_MASK | RT5616_HP_L_SMT_MASK | + RT5616_HP_R_SMT_MASK, RT5616_RSTP_DIS | + RT5616_HP_L_SMT_EN | RT5616_HP_R_SMT_EN); + snd_soc_update_bits(codec, RT5616_HP_CALIB_AMP_DET, + RT5616_HPD_PS_MASK, RT5616_HPD_PS_DIS); + msleep(90); + snd_soc_update_bits(codec, RT5616_HP_VOL, + RT5616_L_MUTE | RT5616_R_MUTE, + RT5616_L_MUTE | RT5616_R_MUTE); + msleep(30); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5616_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_POST_PMU: + snd_soc_update_bits(codec, RT5616_PWR_ANLG1, + RT5616_PWR_LM, RT5616_PWR_LM); + snd_soc_update_bits(codec, RT5616_LOUT_CTRL1, + RT5616_L_MUTE | RT5616_R_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5616_LOUT_CTRL1, + RT5616_L_MUTE | RT5616_R_MUTE, + RT5616_L_MUTE | RT5616_R_MUTE); + snd_soc_update_bits(codec, RT5616_PWR_ANLG1, + RT5616_PWR_LM, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5616_bst1_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_POST_PMU: + snd_soc_update_bits(codec, RT5616_PWR_ANLG2, + RT5616_PWR_BST1_OP2, RT5616_PWR_BST1_OP2); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5616_PWR_ANLG2, + RT5616_PWR_BST1_OP2, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5616_bst2_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_POST_PMU: + snd_soc_update_bits(codec, RT5616_PWR_ANLG2, + RT5616_PWR_BST2_OP2, RT5616_PWR_BST2_OP2); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5616_PWR_ANLG2, + RT5616_PWR_BST2_OP2, 0); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5616_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL1", RT5616_PWR_ANLG2, + RT5616_PWR_PLL_BIT, 0, NULL, 0), + /* Input Side */ + /* micbias */ + SND_SOC_DAPM_SUPPLY("LDO", RT5616_PWR_ANLG1, + RT5616_PWR_LDO_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("micbias1", RT5616_PWR_ANLG2, + RT5616_PWR_MB1_BIT, 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + /* Boost */ + SND_SOC_DAPM_PGA_E("BST1", RT5616_PWR_ANLG2, + RT5616_PWR_BST1_BIT, 0, NULL, 0, rt5616_bst1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("BST2", RT5616_PWR_ANLG2, + RT5616_PWR_BST2_BIT, 0, NULL, 0, rt5616_bst2_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + /* Input Volume */ + SND_SOC_DAPM_PGA("INL1 VOL", RT5616_PWR_VOL, + RT5616_PWR_IN1_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR1 VOL", RT5616_PWR_VOL, + RT5616_PWR_IN1_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INL2 VOL", RT5616_PWR_VOL, + RT5616_PWR_IN2_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR2 VOL", RT5616_PWR_VOL, + RT5616_PWR_IN2_R_BIT, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5616_PWR_MIXER, RT5616_PWR_RM_L_BIT, 0, + rt5616_rec_l_mix, ARRAY_SIZE(rt5616_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5616_PWR_MIXER, RT5616_PWR_RM_R_BIT, 0, + rt5616_rec_r_mix, ARRAY_SIZE(rt5616_rec_r_mix)), + /* ADCs */ + SND_SOC_DAPM_ADC_E("ADC L", NULL, RT5616_PWR_DIG1, + RT5616_PWR_ADC_L_BIT, 0, rt5616_adc_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC_E("ADC R", NULL, RT5616_PWR_DIG1, + RT5616_PWR_ADC_R_BIT, 0, rt5616_adc_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("stereo1 filter", RT5616_PWR_DIG2, + RT5616_PWR_ADC_STO1_F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5616_sto1_adc_l_mix, ARRAY_SIZE(rt5616_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5616_sto1_adc_r_mix, ARRAY_SIZE(rt5616_sto1_adc_r_mix)), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5616_PWR_DIG1, + RT5616_PWR_I2S1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + + /* 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), + + /* Audio DSP */ + SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5616_dac_l_mix, ARRAY_SIZE(rt5616_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5616_dac_r_mix, ARRAY_SIZE(rt5616_dac_r_mix)), + + SND_SOC_DAPM_SUPPLY("Stero1 DAC Power", RT5616_PWR_DIG2, + RT5616_PWR_DAC_STO1_F_BIT, 0, NULL, 0), + + /* DAC Mixer */ + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5616_sto_dac_l_mix, ARRAY_SIZE(rt5616_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5616_sto_dac_r_mix, ARRAY_SIZE(rt5616_sto_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5616_PWR_DIG1, + RT5616_PWR_DAC_L1_BIT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5616_PWR_DIG1, + RT5616_PWR_DAC_R1_BIT, 0), + /* OUT Mixer */ + SND_SOC_DAPM_MIXER("OUT MIXL", RT5616_PWR_MIXER, RT5616_PWR_OM_L_BIT, + 0, rt5616_out_l_mix, ARRAY_SIZE(rt5616_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5616_PWR_MIXER, RT5616_PWR_OM_R_BIT, + 0, rt5616_out_r_mix, ARRAY_SIZE(rt5616_out_r_mix)), + /* Output Volume */ + SND_SOC_DAPM_PGA("OUTVOL L", RT5616_PWR_VOL, + RT5616_PWR_OV_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTVOL R", RT5616_PWR_VOL, + RT5616_PWR_OV_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL L", RT5616_PWR_VOL, + RT5616_PWR_HV_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL R", RT5616_PWR_VOL, + RT5616_PWR_HV_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC 1", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC 2", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("INL1", RT5616_PWR_VOL, + RT5616_PWR_IN1_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR1", RT5616_PWR_VOL, + RT5616_PWR_IN1_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INL2", RT5616_PWR_VOL, + RT5616_PWR_IN2_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR2", RT5616_PWR_VOL, + RT5616_PWR_IN2_R_BIT, 0, NULL, 0), + /* HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0, + rt5616_hpo_mix, ARRAY_SIZE(rt5616_hpo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", SND_SOC_NOPM, 0, 0, + rt5616_lout_mix, ARRAY_SIZE(rt5616_lout_mix)), + + SND_SOC_DAPM_PGA_S("HP amp", 1, SND_SOC_NOPM, 0, 0, + rt5616_hp_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("LOUT amp", 1, SND_SOC_NOPM, 0, 0, + rt5616_lout_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, SND_SOC_NOPM, 0, 0, + rt5616_charge_pump_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), +}; + +static const struct snd_soc_dapm_route rt5616_dapm_routes[] = { + {"IN1P", NULL, "LDO"}, + {"IN2P", NULL, "LDO"}, + + {"IN1P", NULL, "MIC1"}, + {"IN2P", NULL, "MIC2"}, + {"IN2N", NULL, "MIC2"}, + + {"BST1", NULL, "IN1P"}, + {"BST2", NULL, "IN2P"}, + {"BST2", NULL, "IN2N"}, + {"BST1", NULL, "micbias1"}, + {"BST2", NULL, "micbias1"}, + + {"INL1 VOL", NULL, "IN2P"}, + {"INR1 VOL", NULL, "IN2N"}, + + {"RECMIXL", "INL1 Switch", "INL1 VOL"}, + {"RECMIXL", "BST2 Switch", "BST2"}, + {"RECMIXL", "BST1 Switch", "BST1"}, + + {"RECMIXR", "INR1 Switch", "INR1 VOL"}, + {"RECMIXR", "BST2 Switch", "BST2"}, + {"RECMIXR", "BST1 Switch", "BST1"}, + + {"ADC L", NULL, "RECMIXL"}, + {"ADC R", NULL, "RECMIXR"}, + + {"Stereo1 ADC MIXL", "ADC1 Switch", "ADC L"}, + {"Stereo1 ADC MIXL", NULL, "stereo1 filter"}, + {"stereo1 filter", NULL, "PLL1", is_sys_clk_from_pll}, + + {"Stereo1 ADC MIXR", "ADC1 Switch", "ADC R"}, + {"Stereo1 ADC MIXR", NULL, "stereo1 filter"}, + {"stereo1 filter", NULL, "PLL1", is_sys_clk_from_pll}, + + {"IF1 ADC1", NULL, "Stereo1 ADC MIXL"}, + {"IF1 ADC1", NULL, "Stereo1 ADC MIXR"}, + {"IF1 ADC1", NULL, "I2S1"}, + + {"AIF1TX", NULL, "IF1 ADC1"}, + + {"IF1 DAC", NULL, "AIF1RX"}, + {"IF1 DAC", NULL, "I2S1"}, + + {"IF1 DAC1 L", NULL, "IF1 DAC"}, + {"IF1 DAC1 R", NULL, "IF1 DAC"}, + + {"DAC MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"}, + {"DAC MIXL", "INF1 Switch", "IF1 DAC1 L"}, + {"DAC MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"}, + {"DAC MIXR", "INF1 Switch", "IF1 DAC1 R"}, + + {"Audio DSP", NULL, "DAC MIXL"}, + {"Audio DSP", NULL, "DAC MIXR"}, + + {"Stereo DAC MIXL", "DAC L1 Switch", "Audio DSP"}, + {"Stereo DAC MIXL", "DAC R1 Switch", "DAC MIXR"}, + {"Stereo DAC MIXL", NULL, "Stero1 DAC Power"}, + {"Stereo DAC MIXR", "DAC R1 Switch", "Audio DSP"}, + {"Stereo DAC MIXR", "DAC L1 Switch", "DAC MIXL"}, + {"Stereo DAC MIXR", NULL, "Stero1 DAC Power"}, + + {"DAC L1", NULL, "Stereo DAC MIXL"}, + {"DAC L1", NULL, "PLL1", is_sys_clk_from_pll}, + {"DAC R1", NULL, "Stereo DAC MIXR"}, + {"DAC R1", NULL, "PLL1", is_sys_clk_from_pll}, + + {"OUT MIXL", "BST1 Switch", "BST1"}, + {"OUT MIXL", "BST2 Switch", "BST2"}, + {"OUT MIXL", "INL1 Switch", "INL1 VOL"}, + {"OUT MIXL", "REC MIXL Switch", "RECMIXL"}, + {"OUT MIXL", "DAC L1 Switch", "DAC L1"}, + + {"OUT MIXR", "BST2 Switch", "BST2"}, + {"OUT MIXR", "BST1 Switch", "BST1"}, + {"OUT MIXR", "INR1 Switch", "INR1 VOL"}, + {"OUT MIXR", "REC MIXR Switch", "RECMIXR"}, + {"OUT MIXR", "DAC R1 Switch", "DAC R1"}, + + {"HPOVOL L", NULL, "OUT MIXL"}, + {"HPOVOL R", NULL, "OUT MIXR"}, + {"OUTVOL L", NULL, "OUT MIXL"}, + {"OUTVOL R", NULL, "OUT MIXR"}, + + {"DAC 1", NULL, "DAC L1"}, + {"DAC 1", NULL, "DAC R1"}, + {"HPOVOL", NULL, "HPOVOL L"}, + {"HPOVOL", NULL, "HPOVOL R"}, + {"HPO MIX", "DAC1 Switch", "DAC 1"}, + {"HPO MIX", "HPVOL Switch", "HPOVOL"}, + + {"LOUT MIX", "DAC L1 Switch", "DAC L1"}, + {"LOUT MIX", "DAC R1 Switch", "DAC R1"}, + {"LOUT MIX", "OUTVOL L Switch", "OUTVOL L"}, + {"LOUT MIX", "OUTVOL R Switch", "OUTVOL R"}, + + {"HP amp", NULL, "HPO MIX"}, + {"HP amp", NULL, "Charge Pump"}, + {"HPOL", NULL, "HP amp"}, + {"HPOR", NULL, "HP amp"}, + + {"LOUT amp", NULL, "LOUT MIX"}, + {"LOUT amp", NULL, "Charge Pump"}, + {"LOUTL", NULL, "LOUT amp"}, + {"LOUTR", NULL, "LOUT amp"}, + +}; + +static int rt5616_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct rt5616_priv *rt5616 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + rt5616->lrck[dai->id] = params_rate(params); + + pre_div = rl6231_get_clk_info(rt5616->sysclk, rt5616->lrck[dai->id]); + + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting\n"); + return -EINVAL; + } + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + bclk_ms = frame_size > 32 ? 1 : 0; + rt5616->bclk[dai->id] = rt5616->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5616->bclk[dai->id], rt5616->lrck[dai->id]); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= RT5616_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= RT5616_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + val_len |= RT5616_I2S_DL_8; + break; + default: + return -EINVAL; + } + + mask_clk = RT5616_I2S_PD1_MASK; + val_clk = pre_div << RT5616_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5616_I2S1_SDP, + RT5616_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5616_ADDA_CLK1, mask_clk, val_clk); + + + return 0; +} + +static int rt5616_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5616_priv *rt5616 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5616->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5616_I2S_MS_S; + rt5616->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5616_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5616_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5616_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5616_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, RT5616_I2S1_SDP, + RT5616_I2S_MS_MASK | RT5616_I2S_BP_MASK | + RT5616_I2S_DF_MASK, reg_val); + + + return 0; +} + +static int rt5616_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5616_priv *rt5616 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5616->sysclk && clk_id == rt5616->sysclk_src) + return 0; + + switch (clk_id) { + case RT5616_SCLK_S_MCLK: + reg_val |= RT5616_SCLK_SRC_MCLK; + break; + case RT5616_SCLK_S_PLL1: + reg_val |= RT5616_SCLK_SRC_PLL1; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5616_GLB_CLK, + RT5616_SCLK_SRC_MASK, reg_val); + rt5616->sysclk = freq; + rt5616->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + return 0; +} + +static int rt5616_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; + struct rt5616_priv *rt5616 = snd_soc_codec_get_drvdata(codec); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt5616->pll_src && freq_in == rt5616->pll_in && + freq_out == rt5616->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5616->pll_in = 0; + rt5616->pll_out = 0; + snd_soc_update_bits(codec, RT5616_GLB_CLK, + RT5616_SCLK_SRC_MASK, RT5616_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5616_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5616_GLB_CLK, + RT5616_PLL1_SRC_MASK, RT5616_PLL1_SRC_MCLK); + break; + case RT5616_PLL1_S_BCLK1: + case RT5616_PLL1_S_BCLK2: + snd_soc_update_bits(codec, RT5616_GLB_CLK, + RT5616_PLL1_SRC_MASK, RT5616_PLL1_SRC_BCLK1); + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_write(codec, RT5616_PLL_CTRL1, + pll_code.n_code << RT5616_PLL_N_SFT | pll_code.k_code); + snd_soc_write(codec, RT5616_PLL_CTRL2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5616_PLL_M_SFT | + pll_code.m_bp << RT5616_PLL_M_BP_SFT); + + rt5616->pll_in = freq_in; + rt5616->pll_out = freq_out; + rt5616->pll_src = source; + + return 0; +} + +static int rt5616_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + snd_soc_update_bits(codec, RT5616_PWR_ANLG1, + RT5616_PWR_VREF1 | RT5616_PWR_MB | + RT5616_PWR_BG | RT5616_PWR_VREF2, + RT5616_PWR_VREF1 | RT5616_PWR_MB | + RT5616_PWR_BG | RT5616_PWR_VREF2); + mdelay(10); + snd_soc_update_bits(codec, RT5616_PWR_ANLG1, + RT5616_PWR_FV1 | RT5616_PWR_FV2, + RT5616_PWR_FV1 | RT5616_PWR_FV2); + snd_soc_update_bits(codec, RT5616_D_MISC, + RT5616_D_GATE_EN, RT5616_D_GATE_EN); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, RT5616_D_MISC, RT5616_D_GATE_EN, 0); + snd_soc_write(codec, RT5616_PWR_DIG1, 0x0000); + snd_soc_write(codec, RT5616_PWR_DIG2, 0x0000); + snd_soc_write(codec, RT5616_PWR_VOL, 0x0000); + snd_soc_write(codec, RT5616_PWR_MIXER, 0x0000); + snd_soc_write(codec, RT5616_PWR_ANLG1, 0x0000); + snd_soc_write(codec, RT5616_PWR_ANLG2, 0x0000); + break; + + default: + break; + } + + return 0; +} + +static int rt5616_probe(struct snd_soc_codec *codec) +{ + struct rt5616_priv *rt5616 = snd_soc_codec_get_drvdata(codec); + + rt5616->codec = codec; + + return 0; +} + +#ifdef CONFIG_PM +static int rt5616_suspend(struct snd_soc_codec *codec) +{ + struct rt5616_priv *rt5616 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5616->regmap, true); + regcache_mark_dirty(rt5616->regmap); + + return 0; +} + +static int rt5616_resume(struct snd_soc_codec *codec) +{ + struct rt5616_priv *rt5616 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5616->regmap, false); + regcache_sync(rt5616->regmap); + return 0; +} +#else +#define rt5616_suspend NULL +#define rt5616_resume NULL +#endif + +#define RT5616_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5616_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + + +struct snd_soc_dai_ops rt5616_aif_dai_ops = { + .hw_params = rt5616_hw_params, + .set_fmt = rt5616_set_dai_fmt, + .set_sysclk = rt5616_set_dai_sysclk, + .set_pll = rt5616_set_dai_pll, +}; + +struct snd_soc_dai_driver rt5616_dai[] = { + { + .name = "rt5616-aif1", + .id = RT5616_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5616_STEREO_RATES, + .formats = RT5616_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5616_STEREO_RATES, + .formats = RT5616_FORMATS, + }, + .ops = &rt5616_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5616 = { + .probe = rt5616_probe, + .suspend = rt5616_suspend, + .resume = rt5616_resume, + .set_bias_level = rt5616_set_bias_level, + .idle_bias_off = true, + .controls = rt5616_snd_controls, + .num_controls = ARRAY_SIZE(rt5616_snd_controls), + .dapm_widgets = rt5616_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5616_dapm_widgets), + .dapm_routes = rt5616_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5616_dapm_routes), +}; + +static const struct regmap_config rt5616_regmap = { + .reg_bits = 8, + .val_bits = 16, + .use_single_rw = true, + .max_register = RT5616_DEVICE_ID + 1 + (ARRAY_SIZE(rt5616_ranges) * + RT5616_PR_SPACING), + .volatile_reg = rt5616_volatile_register, + .readable_reg = rt5616_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5616_reg, + .num_reg_defaults = ARRAY_SIZE(rt5616_reg), + .ranges = rt5616_ranges, + .num_ranges = ARRAY_SIZE(rt5616_ranges), +}; + +static const struct i2c_device_id rt5616_i2c_id[] = { + { "rt5616", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5616_i2c_id); + +static int rt5616_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5616_priv *rt5616; + unsigned int val; + int ret; + + rt5616 = devm_kzalloc(&i2c->dev, sizeof(struct rt5616_priv), + GFP_KERNEL); + if (rt5616 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5616); + + rt5616->regmap = devm_regmap_init_i2c(i2c, &rt5616_regmap); + if (IS_ERR(rt5616->regmap)) { + ret = PTR_ERR(rt5616->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5616->regmap, RT5616_DEVICE_ID, &val); + if (val != 0x6281) { + dev_err(&i2c->dev, + "Device with ID register %#x is not rt5616\n", + val); + ret = -ENODEV; + } + regmap_write(rt5616->regmap, RT5616_RESET, 0); + regmap_update_bits(rt5616->regmap, RT5616_PWR_ANLG1, + RT5616_PWR_VREF1 | RT5616_PWR_MB | + RT5616_PWR_BG | RT5616_PWR_VREF2, + RT5616_PWR_VREF1 | RT5616_PWR_MB | + RT5616_PWR_BG | RT5616_PWR_VREF2); + mdelay(10); + regmap_update_bits(rt5616->regmap, RT5616_PWR_ANLG1, + RT5616_PWR_FV1 | RT5616_PWR_FV2, + RT5616_PWR_FV1 | RT5616_PWR_FV2); + + ret = regmap_register_patch(rt5616->regmap, init_list, + ARRAY_SIZE(init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + regmap_update_bits(rt5616->regmap, RT5616_PWR_ANLG1, + RT5616_PWR_LDO_DVO_MASK, RT5616_PWR_LDO_DVO_1_2V); + + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5616, + rt5616_dai, ARRAY_SIZE(rt5616_dai)); + +} + +static int rt5616_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +static void rt5616_i2c_shutdown(struct i2c_client *client) +{ + struct rt5616_priv *rt5616 = i2c_get_clientdata(client); + + regmap_write(rt5616->regmap, RT5616_HP_VOL, 0xc8c8); + regmap_write(rt5616->regmap, RT5616_LOUT_CTRL1, 0xc8c8); + +} + +static struct i2c_driver rt5616_i2c_driver = { + .driver = { + .name = "rt5616", + }, + .probe = rt5616_i2c_probe, + .remove = rt5616_i2c_remove, + .shutdown = rt5616_i2c_shutdown, + .id_table = rt5616_i2c_id, +}; +module_i2c_driver(rt5616_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5616 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5616.h b/sound/soc/codecs/rt5616.h new file mode 100644 index 0000000..f88cddd --- /dev/null +++ b/sound/soc/codecs/rt5616.h @@ -0,0 +1,1819 @@ +/* + * rt5616.h -- RT5616 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu(a)realtek.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 __RT5616_H__ +#define __RT5616_H__ + +/* Info */ +#define RT5616_RESET 0x00 +#define RT5616_VERSION_ID 0xfd +#define RT5616_VENDOR_ID 0xfe +#define RT5616_DEVICE_ID 0xff +/* I/O - Output */ +#define RT5616_HP_VOL 0x02 +#define RT5616_LOUT_CTRL1 0x03 +#define RT5616_LOUT_CTRL2 0x05 +/* I/O - Input */ +#define RT5616_IN1_IN2 0x0d +#define RT5616_INL1_INR1_VOL 0x0f +/* I/O - ADC/DAC/DMIC */ +#define RT5616_DAC1_DIG_VOL 0x19 +#define RT5616_ADC_DIG_VOL 0x1c +#define RT5616_ADC_BST_VOL 0x1e +/* Mixer - D-D */ +#define RT5616_STO1_ADC_MIXER 0x27 +#define RT5616_AD_DA_MIXER 0x29 +#define RT5616_STO_DAC_MIXER 0x2a + +/* Mixer - ADC */ +#define RT5616_REC_L1_MIXER 0x3b +#define RT5616_REC_L2_MIXER 0x3c +#define RT5616_REC_R1_MIXER 0x3d +#define RT5616_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5616_HPO_MIXER 0x45 +#define RT5616_OUT_L1_MIXER 0x4d +#define RT5616_OUT_L2_MIXER 0x4e +#define RT5616_OUT_L3_MIXER 0x4f +#define RT5616_OUT_R1_MIXER 0x50 +#define RT5616_OUT_R2_MIXER 0x51 +#define RT5616_OUT_R3_MIXER 0x52 +#define RT5616_LOUT_MIXER 0x53 +/* Power */ +#define RT5616_PWR_DIG1 0x61 +#define RT5616_PWR_DIG2 0x62 +#define RT5616_PWR_ANLG1 0x63 +#define RT5616_PWR_ANLG2 0x64 +#define RT5616_PWR_MIXER 0x65 +#define RT5616_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5616_PRIV_INDEX 0x6a +#define RT5616_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5616_I2S1_SDP 0x70 +#define RT5616_ADDA_CLK1 0x73 +#define RT5616_ADDA_CLK2 0x74 + +/* Function - Analog */ +#define RT5616_GLB_CLK 0x80 +#define RT5616_PLL_CTRL1 0x81 +#define RT5616_PLL_CTRL2 0x82 +#define RT5616_HP_OVCD 0x8b +#define RT5616_DEPOP_M1 0x8e +#define RT5616_DEPOP_M2 0x8f +#define RT5616_DEPOP_M3 0x90 +#define RT5616_CHARGE_PUMP 0x91 +#define RT5616_PV_DET_SPK_G 0x92 +#define RT5616_MICBIAS 0x93 +#define RT5616_A_JD_CTL1 0x94 +#define RT5616_A_JD_CTL2 0x95 +/* Function - Digital */ +#define RT5616_EQ_CTRL1 0xb0 +#define RT5616_EQ_CTRL2 0xb1 +#define RT5616_WIND_FILTER 0xb2 +#define RT5616_DRC_AGC_1 0xb4 +#define RT5616_DRC_AGC_2 0xb5 +#define RT5616_DRC_AGC_3 0xb6 +#define RT5616_SVOL_ZC 0xb7 +#define RT5616_JD_CTRL1 0xbb +#define RT5616_JD_CTRL2 0xbc +#define RT5616_IRQ_CTRL1 0xbd +#define RT5616_IRQ_CTRL2 0xbe +#define RT5616_INT_IRQ_ST 0xbf +#define RT5616_GPIO_CTRL1 0xc0 +#define RT5616_GPIO_CTRL2 0xc1 +#define RT5616_GPIO_CTRL3 0xc2 +#define RT5616_PGM_REG_ARR1 0xc8 +#define RT5616_PGM_REG_ARR2 0xc9 +#define RT5616_PGM_REG_ARR3 0xca +#define RT5616_PGM_REG_ARR4 0xcb +#define RT5616_PGM_REG_ARR5 0xcc +#define RT5616_SCB_FUNC 0xcd +#define RT5616_SCB_CTRL 0xce +#define RT5616_BASE_BACK 0xcf +#define RT5616_MP3_PLUS1 0xd0 +#define RT5616_MP3_PLUS2 0xd1 +#define RT5616_ADJ_HPF_CTRL1 0xd3 +#define RT5616_ADJ_HPF_CTRL2 0xd4 +#define RT5616_HP_CALIB_AMP_DET 0xd6 +#define RT5616_HP_CALIB2 0xd7 +#define RT5616_SV_ZCD1 0xd9 +#define RT5616_SV_ZCD2 0xda +#define RT5616_D_MISC 0xfa +/* Dummy Register */ +#define RT5616_DUMMY2 0xfb +#define RT5616_DUMMY3 0xfc + + +/* Index of Codec Private Register definition */ +#define RT5616_BIAS_CUR1 0x12 +#define RT5616_BIAS_CUR3 0x14 +#define RT5616_CLSD_INT_REG1 0x1c +#define RT5616_MAMP_INT_REG2 0x37 +#define RT5616_CHOP_DAC_ADC 0x3d +#define RT5616_3D_SPK 0x63 +#define RT5616_WND_1 0x6c +#define RT5616_WND_2 0x6d +#define RT5616_WND_3 0x6e +#define RT5616_WND_4 0x6f +#define RT5616_WND_5 0x70 +#define RT5616_WND_8 0x73 +#define RT5616_DIP_SPK_INF 0x75 +#define RT5616_HP_DCC_INT1 0x77 +#define RT5616_EQ_BW_LOP 0xa0 +#define RT5616_EQ_GN_LOP 0xa1 +#define RT5616_EQ_FC_BP1 0xa2 +#define RT5616_EQ_BW_BP1 0xa3 +#define RT5616_EQ_GN_BP1 0xa4 +#define RT5616_EQ_FC_BP2 0xa5 +#define RT5616_EQ_BW_BP2 0xa6 +#define RT5616_EQ_GN_BP2 0xa7 +#define RT5616_EQ_FC_BP3 0xa8 +#define RT5616_EQ_BW_BP3 0xa9 +#define RT5616_EQ_GN_BP3 0xaa +#define RT5616_EQ_FC_BP4 0xab +#define RT5616_EQ_BW_BP4 0xac +#define RT5616_EQ_GN_BP4 0xad +#define RT5616_EQ_FC_HIP1 0xae +#define RT5616_EQ_GN_HIP1 0xaf +#define RT5616_EQ_FC_HIP2 0xb0 +#define RT5616_EQ_BW_HIP2 0xb1 +#define RT5616_EQ_GN_HIP2 0xb2 +#define RT5616_EQ_PRE_VOL 0xb3 +#define RT5616_EQ_PST_VOL 0xb4 + + +/* global definition */ +#define RT5616_L_MUTE (0x1 << 15) +#define RT5616_L_MUTE_SFT 15 +#define RT5616_VOL_L_MUTE (0x1 << 14) +#define RT5616_VOL_L_SFT 14 +#define RT5616_R_MUTE (0x1 << 7) +#define RT5616_R_MUTE_SFT 7 +#define RT5616_VOL_R_MUTE (0x1 << 6) +#define RT5616_VOL_R_SFT 6 +#define RT5616_L_VOL_MASK (0x3f << 8) +#define RT5616_L_VOL_SFT 8 +#define RT5616_R_VOL_MASK (0x3f) +#define RT5616_R_VOL_SFT 0 + +/* LOUT Control 2(0x05) */ +#define RT5616_EN_DFO (0x1 << 15) + +/* IN1 and IN2 Control (0x0d) */ +/* IN3 and IN4 Control (0x0e) */ +#define RT5616_BST_MASK1 (0xf<<12) +#define RT5616_BST_SFT1 12 +#define RT5616_BST_MASK2 (0xf<<8) +#define RT5616_BST_SFT2 8 +#define RT5616_IN_DF1 (0x1 << 7) +#define RT5616_IN_SFT1 7 +#define RT5616_IN_DF2 (0x1 << 6) +#define RT5616_IN_SFT2 6 + +/* INL1 and INR1 Volume Control (0x0f) */ +#define RT5616_INL_VOL_MASK (0x1f << 8) +#define RT5616_INL_VOL_SFT 8 +#define RT5616_INR_SEL_MASK (0x1 << 7) +#define RT5616_INR_SEL_SFT 7 +#define RT5616_INR_SEL_IN4N (0x0 << 7) +#define RT5616_INR_SEL_MONON (0x1 << 7) +#define RT5616_INR_VOL_MASK (0x1f) +#define RT5616_INR_VOL_SFT 0 + +/* DAC1 Digital Volume (0x19) */ +#define RT5616_DAC_L1_VOL_MASK (0xff << 8) +#define RT5616_DAC_L1_VOL_SFT 8 +#define RT5616_DAC_R1_VOL_MASK (0xff) +#define RT5616_DAC_R1_VOL_SFT 0 + +/* DAC2 Digital Volume (0x1a) */ +#define RT5616_DAC_L2_VOL_MASK (0xff << 8) +#define RT5616_DAC_L2_VOL_SFT 8 +#define RT5616_DAC_R2_VOL_MASK (0xff) +#define RT5616_DAC_R2_VOL_SFT 0 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5616_ADC_L_VOL_MASK (0x7f << 8) +#define RT5616_ADC_L_VOL_SFT 8 +#define RT5616_ADC_R_VOL_MASK (0x7f) +#define RT5616_ADC_R_VOL_SFT 0 + +/* Mono ADC Digital Volume Control (0x1d) */ +#define RT5616_M_MONO_ADC_L (0x1 << 15) +#define RT5616_M_MONO_ADC_L_SFT 15 +#define RT5616_MONO_ADC_L_VOL_MASK (0x7f << 8) +#define RT5616_MONO_ADC_L_VOL_SFT 8 +#define RT5616_M_MONO_ADC_R (0x1 << 7) +#define RT5616_M_MONO_ADC_R_SFT 7 +#define RT5616_MONO_ADC_R_VOL_MASK (0x7f) +#define RT5616_MONO_ADC_R_VOL_SFT 0 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5616_ADC_L_BST_MASK (0x3 << 14) +#define RT5616_ADC_L_BST_SFT 14 +#define RT5616_ADC_R_BST_MASK (0x3 << 12) +#define RT5616_ADC_R_BST_SFT 12 +#define RT5616_ADC_COMP_MASK (0x3 << 10) +#define RT5616_ADC_COMP_SFT 10 + +/* Stereo ADC1 Mixer Control (0x27) */ +#define RT5616_M_STO1_ADC_L1 (0x1 << 14) +#define RT5616_M_STO1_ADC_L1_SFT 14 +#define RT5616_M_STO1_ADC_R1 (0x1 << 6) +#define RT5616_M_STO1_ADC_R1_SFT 6 + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5616_M_ADCMIX_L (0x1 << 15) +#define RT5616_M_ADCMIX_L_SFT 15 +#define RT5616_M_IF1_DAC_L (0x1 << 14) +#define RT5616_M_IF1_DAC_L_SFT 14 +#define RT5616_M_ADCMIX_R (0x1 << 7) +#define RT5616_M_ADCMIX_R_SFT 7 +#define RT5616_M_IF1_DAC_R (0x1 << 6) +#define RT5616_M_IF1_DAC_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5616_M_DAC_L1_MIXL (0x1 << 14) +#define RT5616_M_DAC_L1_MIXL_SFT 14 +#define RT5616_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5616_DAC_L1_STO_L_VOL_SFT 13 +#define RT5616_M_DAC_R1_MIXL (0x1 << 9) +#define RT5616_M_DAC_R1_MIXL_SFT 9 +#define RT5616_DAC_R1_STO_L_VOL_MASK (0x1 << 8) +#define RT5616_DAC_R1_STO_L_VOL_SFT 8 +#define RT5616_M_DAC_R1_MIXR (0x1 << 6) +#define RT5616_M_DAC_R1_MIXR_SFT 6 +#define RT5616_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5616_DAC_R1_STO_R_VOL_SFT 5 +#define RT5616_M_DAC_L1_MIXR (0x1 << 1) +#define RT5616_M_DAC_L1_MIXR_SFT 1 +#define RT5616_DAC_L1_STO_R_VOL_MASK (0x1) +#define RT5616_DAC_L1_STO_R_VOL_SFT 0 + +/* DD Mixer Control (0x2b) */ +#define RT5616_M_STO_DD_L1 (0x1 << 14) +#define RT5616_M_STO_DD_L1_SFT 14 +#define RT5616_STO_DD_L1_VOL_MASK (0x1 << 13) +#define RT5616_DAC_DD_L1_VOL_SFT 13 +#define RT5616_M_STO_DD_L2 (0x1 << 12) +#define RT5616_M_STO_DD_L2_SFT 12 +#define RT5616_STO_DD_L2_VOL_MASK (0x1 << 11) +#define RT5616_STO_DD_L2_VOL_SFT 11 +#define RT5616_M_STO_DD_R2_L (0x1 << 10) +#define RT5616_M_STO_DD_R2_L_SFT 10 +#define RT5616_STO_DD_R2_L_VOL_MASK (0x1 << 9) +#define RT5616_STO_DD_R2_L_VOL_SFT 9 +#define RT5616_M_STO_DD_R1 (0x1 << 6) +#define RT5616_M_STO_DD_R1_SFT 6 +#define RT5616_STO_DD_R1_VOL_MASK (0x1 << 5) +#define RT5616_STO_DD_R1_VOL_SFT 5 +#define RT5616_M_STO_DD_R2 (0x1 << 4) +#define RT5616_M_STO_DD_R2_SFT 4 +#define RT5616_STO_DD_R2_VOL_MASK (0x1 << 3) +#define RT5616_STO_DD_R2_VOL_SFT 3 +#define RT5616_M_STO_DD_L2_R (0x1 << 2) +#define RT5616_M_STO_DD_L2_R_SFT 2 +#define RT5616_STO_DD_L2_R_VOL_MASK (0x1 << 1) +#define RT5616_STO_DD_L2_R_VOL_SFT 1 + +/* Digital Mixer Control (0x2c) */ +#define RT5616_M_STO_L_DAC_L (0x1 << 15) +#define RT5616_M_STO_L_DAC_L_SFT 15 +#define RT5616_STO_L_DAC_L_VOL_MASK (0x1 << 14) +#define RT5616_STO_L_DAC_L_VOL_SFT 14 +#define RT5616_M_DAC_L2_DAC_L (0x1 << 13) +#define RT5616_M_DAC_L2_DAC_L_SFT 13 +#define RT5616_DAC_L2_DAC_L_VOL_MASK (0x1 << 12) +#define RT5616_DAC_L2_DAC_L_VOL_SFT 12 +#define RT5616_M_STO_R_DAC_R (0x1 << 11) +#define RT5616_M_STO_R_DAC_R_SFT 11 +#define RT5616_STO_R_DAC_R_VOL_MASK (0x1 << 10) +#define RT5616_STO_R_DAC_R_VOL_SFT 10 +#define RT5616_M_DAC_R2_DAC_R (0x1 << 9) +#define RT5616_M_DAC_R2_DAC_R_SFT 9 +#define RT5616_DAC_R2_DAC_R_VOL_MASK (0x1 << 8) +#define RT5616_DAC_R2_DAC_R_VOL_SFT 8 + +/* DSP Path Control 1 (0x2d) */ +#define RT5616_RXDP_SRC_MASK (0x1 << 15) +#define RT5616_RXDP_SRC_SFT 15 +#define RT5616_RXDP_SRC_NOR (0x0 << 15) +#define RT5616_RXDP_SRC_DIV3 (0x1 << 15) +#define RT5616_TXDP_SRC_MASK (0x1 << 14) +#define RT5616_TXDP_SRC_SFT 14 +#define RT5616_TXDP_SRC_NOR (0x0 << 14) +#define RT5616_TXDP_SRC_DIV3 (0x1 << 14) + +/* DSP Path Control 2 (0x2e) */ +#define RT5616_DAC_L2_SEL_MASK (0x3 << 14) +#define RT5616_DAC_L2_SEL_SFT 14 +#define RT5616_DAC_L2_SEL_IF2 (0x0 << 14) +#define RT5616_DAC_L2_SEL_IF3 (0x1 << 14) +#define RT5616_DAC_L2_SEL_TXDC (0x2 << 14) +#define RT5616_DAC_L2_SEL_BASS (0x3 << 14) +#define RT5616_DAC_R2_SEL_MASK (0x3 << 12) +#define RT5616_DAC_R2_SEL_SFT 12 +#define RT5616_DAC_R2_SEL_IF2 (0x0 << 12) +#define RT5616_DAC_R2_SEL_IF3 (0x1 << 12) +#define RT5616_DAC_R2_SEL_TXDC (0x2 << 12) +#define RT5616_IF2_ADC_L_SEL_MASK (0x1 << 11) +#define RT5616_IF2_ADC_L_SEL_SFT 11 +#define RT5616_IF2_ADC_L_SEL_TXDP (0x0 << 11) +#define RT5616_IF2_ADC_L_SEL_PASS (0x1 << 11) +#define RT5616_IF2_ADC_R_SEL_MASK (0x1 << 10) +#define RT5616_IF2_ADC_R_SEL_SFT 10 +#define RT5616_IF2_ADC_R_SEL_TXDP (0x0 << 10) +#define RT5616_IF2_ADC_R_SEL_PASS (0x1 << 10) +#define RT5616_RXDC_SEL_MASK (0x3 << 8) +#define RT5616_RXDC_SEL_SFT 8 +#define RT5616_RXDC_SEL_NOR (0x0 << 8) +#define RT5616_RXDC_SEL_L2R (0x1 << 8) +#define RT5616_RXDC_SEL_R2L (0x2 << 8) +#define RT5616_RXDC_SEL_SWAP (0x3 << 8) +#define RT5616_RXDP_SEL_MASK (0x3 << 6) +#define RT5616_RXDP_SEL_SFT 6 +#define RT5616_RXDP_SEL_NOR (0x0 << 6) +#define RT5616_RXDP_SEL_L2R (0x1 << 6) +#define RT5616_RXDP_SEL_R2L (0x2 << 6) +#define RT5616_RXDP_SEL_SWAP (0x3 << 6) +#define RT5616_TXDC_SEL_MASK (0x3 << 4) +#define RT5616_TXDC_SEL_SFT 4 +#define RT5616_TXDC_SEL_NOR (0x0 << 4) +#define RT5616_TXDC_SEL_L2R (0x1 << 4) +#define RT5616_TXDC_SEL_R2L (0x2 << 4) +#define RT5616_TXDC_SEL_SWAP (0x3 << 4) +#define RT5616_TXDP_SEL_MASK (0x3 << 2) +#define RT5616_TXDP_SEL_SFT 2 +#define RT5616_TXDP_SEL_NOR (0x0 << 2) +#define RT5616_TXDP_SEL_L2R (0x1 << 2) +#define RT5616_TXDP_SEL_R2L (0x2 << 2) +#define RT5616_TRXDP_SEL_SWAP (0x3 << 2) + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5616_G_LN_L2_RM_L_MASK (0x7 << 13) +#define RT5616_G_IN_L2_RM_L_SFT 13 +#define RT5616_G_LN_L1_RM_L_MASK (0x7 << 10) +#define RT5616_G_IN_L1_RM_L_SFT 10 +#define RT5616_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5616_G_BST3_RM_L_SFT 4 +#define RT5616_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5616_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5616_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5616_G_BST1_RM_L_SFT 13 +#define RT5616_G_OM_L_RM_L_MASK (0x7 << 10) +#define RT5616_G_OM_L_RM_L_SFT 10 +#define RT5616_M_IN2_L_RM_L (0x1 << 6) +#define RT5616_M_IN2_L_RM_L_SFT 6 +#define RT5616_M_IN1_L_RM_L (0x1 << 5) +#define RT5616_M_IN1_L_RM_L_SFT 5 +#define RT5616_M_BST3_RM_L (0x1 << 3) +#define RT5616_M_BST3_RM_L_SFT 3 +#define RT5616_M_BST2_RM_L (0x1 << 2) +#define RT5616_M_BST2_RM_L_SFT 2 +#define RT5616_M_BST1_RM_L (0x1 << 1) +#define RT5616_M_BST1_RM_L_SFT 1 +#define RT5616_M_OM_L_RM_L (0x1) +#define RT5616_M_OM_L_RM_L_SFT 0 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5616_G_IN2_R_RM_R_MASK (0x7 << 13) +#define RT5616_G_IN2_R_RM_R_SFT 13 +#define RT5616_G_IN1_R_RM_R_MASK (0x7 << 10) +#define RT5616_G_IN1_R_RM_R_SFT 10 +#define RT5616_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5616_G_BST3_RM_R_SFT 4 +#define RT5616_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5616_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5616_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5616_G_BST1_RM_R_SFT 13 +#define RT5616_G_OM_R_RM_R_MASK (0x7 << 10) +#define RT5616_G_OM_R_RM_R_SFT 10 +#define RT5616_M_IN2_R_RM_R (0x1 << 6) +#define RT5616_M_IN2_R_RM_R_SFT 6 +#define RT5616_M_IN1_R_RM_R (0x1 << 5) +#define RT5616_M_IN1_R_RM_R_SFT 5 +#define RT5616_M_BST3_RM_R (0x1 << 3) +#define RT5616_M_BST3_RM_R_SFT 3 +#define RT5616_M_BST2_RM_R (0x1 << 2) +#define RT5616_M_BST2_RM_R_SFT 2 +#define RT5616_M_BST1_RM_R (0x1 << 1) +#define RT5616_M_BST1_RM_R_SFT 1 +#define RT5616_M_OM_R_RM_R (0x1) +#define RT5616_M_OM_R_RM_R_SFT 0 + +/* HPMIX Control (0x45) */ +#define RT5616_M_DAC1_HM (0x1 << 14) +#define RT5616_M_DAC1_HM_SFT 14 +#define RT5616_M_HPVOL_HM (0x1 << 13) +#define RT5616_M_HPVOL_HM_SFT 13 +#define RT5616_G_HPOMIX_MASK (0x1 << 12) +#define RT5616_G_HPOMIX_SFT 12 + +/* SPK Left Mixer Control (0x46) */ +#define RT5616_G_RM_L_SM_L_MASK (0x3 << 14) +#define RT5616_G_RM_L_SM_L_SFT 14 +#define RT5616_G_IN_L_SM_L_MASK (0x3 << 12) +#define RT5616_G_IN_L_SM_L_SFT 12 +#define RT5616_G_DAC_L1_SM_L_MASK (0x3 << 10) +#define RT5616_G_DAC_L1_SM_L_SFT 10 +#define RT5616_G_DAC_L2_SM_L_MASK (0x3 << 8) +#define RT5616_G_DAC_L2_SM_L_SFT 8 +#define RT5616_G_OM_L_SM_L_MASK (0x3 << 6) +#define RT5616_G_OM_L_SM_L_SFT 6 +#define RT5616_M_RM_L_SM_L (0x1 << 5) +#define RT5616_M_RM_L_SM_L_SFT 5 +#define RT5616_M_IN_L_SM_L (0x1 << 4) +#define RT5616_M_IN_L_SM_L_SFT 4 +#define RT5616_M_DAC_L1_SM_L (0x1 << 3) +#define RT5616_M_DAC_L1_SM_L_SFT 3 +#define RT5616_M_DAC_L2_SM_L (0x1 << 2) +#define RT5616_M_DAC_L2_SM_L_SFT 2 +#define RT5616_M_OM_L_SM_L (0x1 << 1) +#define RT5616_M_OM_L_SM_L_SFT 1 + +/* SPK Right Mixer Control (0x47) */ +#define RT5616_G_RM_R_SM_R_MASK (0x3 << 14) +#define RT5616_G_RM_R_SM_R_SFT 14 +#define RT5616_G_IN_R_SM_R_MASK (0x3 << 12) +#define RT5616_G_IN_R_SM_R_SFT 12 +#define RT5616_G_DAC_R1_SM_R_MASK (0x3 << 10) +#define RT5616_G_DAC_R1_SM_R_SFT 10 +#define RT5616_G_DAC_R2_SM_R_MASK (0x3 << 8) +#define RT5616_G_DAC_R2_SM_R_SFT 8 +#define RT5616_G_OM_R_SM_R_MASK (0x3 << 6) +#define RT5616_G_OM_R_SM_R_SFT 6 +#define RT5616_M_RM_R_SM_R (0x1 << 5) +#define RT5616_M_RM_R_SM_R_SFT 5 +#define RT5616_M_IN_R_SM_R (0x1 << 4) +#define RT5616_M_IN_R_SM_R_SFT 4 +#define RT5616_M_DAC_R1_SM_R (0x1 << 3) +#define RT5616_M_DAC_R1_SM_R_SFT 3 +#define RT5616_M_DAC_R2_SM_R (0x1 << 2) +#define RT5616_M_DAC_R2_SM_R_SFT 2 +#define RT5616_M_OM_R_SM_R (0x1 << 1) +#define RT5616_M_OM_R_SM_R_SFT 1 + +/* SPOLMIX Control (0x48) */ +#define RT5616_M_DAC_R1_SPM_L (0x1 << 15) +#define RT5616_M_DAC_R1_SPM_L_SFT 15 +#define RT5616_M_DAC_L1_SPM_L (0x1 << 14) +#define RT5616_M_DAC_L1_SPM_L_SFT 14 +#define RT5616_M_SV_R_SPM_L (0x1 << 13) +#define RT5616_M_SV_R_SPM_L_SFT 13 +#define RT5616_M_SV_L_SPM_L (0x1 << 12) +#define RT5616_M_SV_L_SPM_L_SFT 12 +#define RT5616_M_BST1_SPM_L (0x1 << 11) +#define RT5616_M_BST1_SPM_L_SFT 11 + +/* SPORMIX Control (0x49) */ +#define RT5616_M_DAC_R1_SPM_R (0x1 << 13) +#define RT5616_M_DAC_R1_SPM_R_SFT 13 +#define RT5616_M_SV_R_SPM_R (0x1 << 12) +#define RT5616_M_SV_R_SPM_R_SFT 12 +#define RT5616_M_BST1_SPM_R (0x1 << 11) +#define RT5616_M_BST1_SPM_R_SFT 11 + +/* SPOLMIX / SPORMIX Ratio Control (0x4a) */ +#define RT5616_SPO_CLSD_RATIO_MASK (0x7) +#define RT5616_SPO_CLSD_RATIO_SFT 0 + +/* Mono Output Mixer Control (0x4c) */ +#define RT5616_M_DAC_R2_MM (0x1 << 15) +#define RT5616_M_DAC_R2_MM_SFT 15 +#define RT5616_M_DAC_L2_MM (0x1 << 14) +#define RT5616_M_DAC_L2_MM_SFT 14 +#define RT5616_M_OV_R_MM (0x1 << 13) +#define RT5616_M_OV_R_MM_SFT 13 +#define RT5616_M_OV_L_MM (0x1 << 12) +#define RT5616_M_OV_L_MM_SFT 12 +#define RT5616_M_BST1_MM (0x1 << 11) +#define RT5616_M_BST1_MM_SFT 11 +#define RT5616_G_MONOMIX_MASK (0x1 << 10) +#define RT5616_G_MONOMIX_SFT 10 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5616_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5616_G_BST2_OM_L_SFT 10 +#define RT5616_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5616_G_BST1_OM_L_SFT 7 +#define RT5616_G_IN1_L_OM_L_MASK (0x7 << 4) +#define RT5616_G_IN1_L_OM_L_SFT 4 +#define RT5616_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5616_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5616_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5616_G_DAC_L1_OM_L_SFT 7 +#define RT5616_G_IN2_L_OM_L_MASK (0x7 << 4) +#define RT5616_G_IN2_L_OM_L_SFT 4 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5616_M_IN2_L_OM_L (0x1 << 9) +#define RT5616_M_IN2_L_OM_L_SFT 9 +#define RT5616_M_BST2_OM_L (0x1 << 6) +#define RT5616_M_BST2_OM_L_SFT 6 +#define RT5616_M_BST1_OM_L (0x1 << 5) +#define RT5616_M_BST1_OM_L_SFT 5 +#define RT5616_M_IN1_L_OM_L (0x1 << 4) +#define RT5616_M_IN1_L_OM_L_SFT 4 +#define RT5616_M_RM_L_OM_L (0x1 << 3) +#define RT5616_M_RM_L_OM_L_SFT 3 +#define RT5616_M_DAC_L1_OM_L (0x1) +#define RT5616_M_DAC_L1_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5616_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5616_G_BST2_OM_R_SFT 10 +#define RT5616_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5616_G_BST1_OM_R_SFT 7 +#define RT5616_G_IN1_R_OM_R_MASK (0x7 << 4) +#define RT5616_G_IN1_R_OM_R_SFT 4 +#define RT5616_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5616_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5616_G_DAC_R1_OM_R_MASK (0x7 << 7) +#define RT5616_G_DAC_R1_OM_R_SFT 7 +#define RT5616_G_IN2_R_OM_R_MASK (0x7 << 4) +#define RT5616_G_IN2_R_OM_R_SFT 4 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5616_M_IN2_R_OM_R (0x1 << 9) +#define RT5616_M_IN2_R_OM_R_SFT 9 +#define RT5616_M_BST2_OM_R (0x1 << 6) +#define RT5616_M_BST2_OM_R_SFT 6 +#define RT5616_M_BST1_OM_R (0x1 << 5) +#define RT5616_M_BST1_OM_R_SFT 5 +#define RT5616_M_IN1_R_OM_R (0x1 << 4) +#define RT5616_M_IN1_R_OM_R_SFT 4 +#define RT5616_M_RM_R_OM_R (0x1 << 3) +#define RT5616_M_RM_R_OM_R_SFT 3 +#define RT5616_M_DAC_R1_OM_R (0x1) +#define RT5616_M_DAC_R1_OM_R_SFT 0 + +/* LOUT Mixer Control (0x53) */ +#define RT5616_M_DAC_L1_LM (0x1 << 15) +#define RT5616_M_DAC_L1_LM_SFT 15 +#define RT5616_M_DAC_R1_LM (0x1 << 14) +#define RT5616_M_DAC_R1_LM_SFT 14 +#define RT5616_M_OV_L_LM (0x1 << 13) +#define RT5616_M_OV_L_LM_SFT 13 +#define RT5616_M_OV_R_LM (0x1 << 12) +#define RT5616_M_OV_R_LM_SFT 12 +#define RT5616_G_LOUTMIX_MASK (0x1 << 11) +#define RT5616_G_LOUTMIX_SFT 11 + +/* Power Management for Digital 1 (0x61) */ +#define RT5616_PWR_I2S1 (0x1 << 15) +#define RT5616_PWR_I2S1_BIT 15 +#define RT5616_PWR_I2S2 (0x1 << 14) +#define RT5616_PWR_I2S2_BIT 14 +#define RT5616_PWR_DAC_L1 (0x1 << 12) +#define RT5616_PWR_DAC_L1_BIT 12 +#define RT5616_PWR_DAC_R1 (0x1 << 11) +#define RT5616_PWR_DAC_R1_BIT 11 +#define RT5616_PWR_ADC_L (0x1 << 2) +#define RT5616_PWR_ADC_L_BIT 2 +#define RT5616_PWR_ADC_R (0x1 << 1) +#define RT5616_PWR_ADC_R_BIT 1 + +/* Power Management for Digital 2 (0x62) */ +#define RT5616_PWR_ADC_STO1_F (0x1 << 15) +#define RT5616_PWR_ADC_STO1_F_BIT 15 +#define RT5616_PWR_DAC_STO1_F (0x1 << 11) +#define RT5616_PWR_DAC_STO1_F_BIT 11 + +/* Power Management for Analog 1 (0x63) */ +#define RT5616_PWR_VREF1 (0x1 << 15) +#define RT5616_PWR_VREF1_BIT 15 +#define RT5616_PWR_FV1 (0x1 << 14) +#define RT5616_PWR_FV1_BIT 14 +#define RT5616_PWR_MB (0x1 << 13) +#define RT5616_PWR_MB_BIT 13 +#define RT5616_PWR_LM (0x1 << 12) +#define RT5616_PWR_LM_BIT 12 +#define RT5616_PWR_BG (0x1 << 11) +#define RT5616_PWR_BG_BIT 11 +#define RT5616_PWR_HP_L (0x1 << 7) +#define RT5616_PWR_HP_L_BIT 7 +#define RT5616_PWR_HP_R (0x1 << 6) +#define RT5616_PWR_HP_R_BIT 6 +#define RT5616_PWR_HA (0x1 << 5) +#define RT5616_PWR_HA_BIT 5 +#define RT5616_PWR_VREF2 (0x1 << 4) +#define RT5616_PWR_VREF2_BIT 4 +#define RT5616_PWR_FV2 (0x1 << 3) +#define RT5616_PWR_FV2_BIT 3 +#define RT5616_PWR_LDO (0x1 << 2) +#define RT5616_PWR_LDO_BIT 2 +#define RT5616_PWR_LDO_DVO_MASK (0x3) +#define RT5616_PWR_LDO_DVO_1_0V 0 +#define RT5616_PWR_LDO_DVO_1_1V 1 +#define RT5616_PWR_LDO_DVO_1_2V 2 +#define RT5616_PWR_LDO_DVO_1_3V 3 + +/* Power Management for Analog 2 (0x64) */ +#define RT5616_PWR_BST1 (0x1 << 15) +#define RT5616_PWR_BST1_BIT 15 +#define RT5616_PWR_BST2 (0x1 << 14) +#define RT5616_PWR_BST2_BIT 14 +#define RT5616_PWR_MB1 (0x1 << 11) +#define RT5616_PWR_MB1_BIT 11 +#define RT5616_PWR_PLL (0x1 << 9) +#define RT5616_PWR_PLL_BIT 9 +#define RT5616_PWR_BST1_OP2 (0x1 << 5) +#define RT5616_PWR_BST1_OP2_BIT 5 +#define RT5616_PWR_BST2_OP2 (0x1 << 4) +#define RT5616_PWR_BST2_OP2_BIT 4 +#define RT5616_PWR_BST3_OP2 (0x1 << 3) +#define RT5616_PWR_BST3_OP2_BIT 3 +#define RT5616_PWR_JD_M (0x1 << 2) +#define RT5616_PWM_JD_M_BIT 2 +#define RT5616_PWR_JD2 (0x1 << 1) +#define RT5616_PWM_JD2_BIT 1 +#define RT5616_PWR_JD3 (0x1) +#define RT5616_PWM_JD3_BIT 0 + +/* Power Management for Mixer (0x65) */ +#define RT5616_PWR_OM_L (0x1 << 15) +#define RT5616_PWR_OM_L_BIT 15 +#define RT5616_PWR_OM_R (0x1 << 14) +#define RT5616_PWR_OM_R_BIT 14 +#define RT5616_PWR_RM_L (0x1 << 11) +#define RT5616_PWR_RM_L_BIT 11 +#define RT5616_PWR_RM_R (0x1 << 10) +#define RT5616_PWR_RM_R_BIT 10 + +/* Power Management for Volume (0x66) */ +#define RT5616_PWR_OV_L (0x1 << 13) +#define RT5616_PWR_OV_L_BIT 13 +#define RT5616_PWR_OV_R (0x1 << 12) +#define RT5616_PWR_OV_R_BIT 12 +#define RT5616_PWR_HV_L (0x1 << 11) +#define RT5616_PWR_HV_L_BIT 11 +#define RT5616_PWR_HV_R (0x1 << 10) +#define RT5616_PWR_HV_R_BIT 10 +#define RT5616_PWR_IN1_L (0x1 << 9) +#define RT5616_PWR_IN1_L_BIT 9 +#define RT5616_PWR_IN1_R (0x1 << 8) +#define RT5616_PWR_IN1_R_BIT 8 +#define RT5616_PWR_IN2_L (0x1 << 7) +#define RT5616_PWR_IN2_L_BIT 7 +#define RT5616_PWR_IN2_R (0x1 << 6) +#define RT5616_PWR_IN2_R_BIT 6 + +/* I2S1/2/3 Audio Serial Data Port Control (0x70 0x71) */ +#define RT5616_I2S_MS_MASK (0x1 << 15) +#define RT5616_I2S_MS_SFT 15 +#define RT5616_I2S_MS_M (0x0 << 15) +#define RT5616_I2S_MS_S (0x1 << 15) +#define RT5616_I2S_O_CP_MASK (0x3 << 10) +#define RT5616_I2S_O_CP_SFT 10 +#define RT5616_I2S_O_CP_OFF (0x0 << 10) +#define RT5616_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5616_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5616_I2S_I_CP_MASK (0x3 << 8) +#define RT5616_I2S_I_CP_SFT 8 +#define RT5616_I2S_I_CP_OFF (0x0 << 8) +#define RT5616_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5616_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5616_I2S_BP_MASK (0x1 << 7) +#define RT5616_I2S_BP_SFT 7 +#define RT5616_I2S_BP_NOR (0x0 << 7) +#define RT5616_I2S_BP_INV (0x1 << 7) +#define RT5616_I2S_DL_MASK (0x3 << 2) +#define RT5616_I2S_DL_SFT 2 +#define RT5616_I2S_DL_16 (0x0 << 2) +#define RT5616_I2S_DL_20 (0x1 << 2) +#define RT5616_I2S_DL_24 (0x2 << 2) +#define RT5616_I2S_DL_8 (0x3 << 2) +#define RT5616_I2S_DF_MASK (0x3) +#define RT5616_I2S_DF_SFT 0 +#define RT5616_I2S_DF_I2S (0x0) +#define RT5616_I2S_DF_LEFT (0x1) +#define RT5616_I2S_DF_PCM_A (0x2) +#define RT5616_I2S_DF_PCM_B (0x3) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5616_I2S_PD1_MASK (0x7 << 12) +#define RT5616_I2S_PD1_SFT 12 +#define RT5616_I2S_PD1_1 (0x0 << 12) +#define RT5616_I2S_PD1_2 (0x1 << 12) +#define RT5616_I2S_PD1_3 (0x2 << 12) +#define RT5616_I2S_PD1_4 (0x3 << 12) +#define RT5616_I2S_PD1_6 (0x4 << 12) +#define RT5616_I2S_PD1_8 (0x5 << 12) +#define RT5616_I2S_PD1_12 (0x6 << 12) +#define RT5616_I2S_PD1_16 (0x7 << 12) +#define RT5616_I2S_BCLK_MS2_MASK (0x1 << 11) +#define RT5616_DAC_OSR_MASK (0x3 << 2) +#define RT5616_DAC_OSR_SFT 2 +#define RT5616_DAC_OSR_128 (0x0 << 2) +#define RT5616_DAC_OSR_64 (0x1 << 2) +#define RT5616_DAC_OSR_32 (0x2 << 2) +#define RT5616_DAC_OSR_128_3 (0x3 << 2) +#define RT5616_ADC_OSR_MASK (0x3) +#define RT5616_ADC_OSR_SFT 0 +#define RT5616_ADC_OSR_128 (0x0) +#define RT5616_ADC_OSR_64 (0x1) +#define RT5616_ADC_OSR_32 (0x2) +#define RT5616_ADC_OSR_128_3 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5616_DAHPF_EN (0x1 << 11) +#define RT5616_DAHPF_EN_SFT 11 +#define RT5616_ADHPF_EN (0x1 << 10) +#define RT5616_ADHPF_EN_SFT 10 + +/* TDM Control 1 (0x77) */ +#define RT5616_TDM_INTEL_SEL_MASK (0x1 << 15) +#define RT5616_TDM_INTEL_SEL_SFT 15 +#define RT5616_TDM_INTEL_SEL_64 (0x0 << 15) +#define RT5616_TDM_INTEL_SEL_50 (0x1 << 15) +#define RT5616_TDM_MODE_SEL_MASK (0x1 << 14) +#define RT5616_TDM_MODE_SEL_SFT 14 +#define RT5616_TDM_MODE_SEL_NOR (0x0 << 14) +#define RT5616_TDM_MODE_SEL_TDM (0x1 << 14) +#define RT5616_TDM_CH_NUM_SEL_MASK (0x3 << 12) +#define RT5616_TDM_CH_NUM_SEL_SFT 12 +#define RT5616_TDM_CH_NUM_SEL_2 (0x0 << 12) +#define RT5616_TDM_CH_NUM_SEL_4 (0x1 << 12) +#define RT5616_TDM_CH_NUM_SEL_6 (0x2 << 12) +#define RT5616_TDM_CH_NUM_SEL_8 (0x3 << 12) +#define RT5616_TDM_CH_LEN_SEL_MASK (0x3 << 10) +#define RT5616_TDM_CH_LEN_SEL_SFT 10 +#define RT5616_TDM_CH_LEN_SEL_16 (0x0 << 10) +#define RT5616_TDM_CH_LEN_SEL_20 (0x1 << 10) +#define RT5616_TDM_CH_LEN_SEL_24 (0x2 << 10) +#define RT5616_TDM_CH_LEN_SEL_32 (0x3 << 10) +#define RT5616_TDM_ADC_SEL_MASK (0x1 << 9) +#define RT5616_TDM_ADC_SEL_SFT 9 +#define RT5616_TDM_ADC_SEL_NOR (0x0 << 9) +#define RT5616_TDM_ADC_SEL_SWAP (0x1 << 9) +#define RT5616_TDM_ADC_START_SEL_MASK (0x1 << 8) +#define RT5616_TDM_ADC_START_SEL_SFT 8 +#define RT5616_TDM_ADC_START_SEL_SL0 (0x0 << 8) +#define RT5616_TDM_ADC_START_SEL_SL4 (0x1 << 8) +#define RT5616_TDM_I2S_CH2_SEL_MASK (0x3 << 6) +#define RT5616_TDM_I2S_CH2_SEL_SFT 6 +#define RT5616_TDM_I2S_CH2_SEL_LR (0x0 << 6) +#define RT5616_TDM_I2S_CH2_SEL_RL (0x1 << 6) +#define RT5616_TDM_I2S_CH2_SEL_LL (0x2 << 6) +#define RT5616_TDM_I2S_CH2_SEL_RR (0x3 << 6) +#define RT5616_TDM_I2S_CH4_SEL_MASK (0x3 << 4) +#define RT5616_TDM_I2S_CH4_SEL_SFT 4 +#define RT5616_TDM_I2S_CH4_SEL_LR (0x0 << 4) +#define RT5616_TDM_I2S_CH4_SEL_RL (0x1 << 4) +#define RT5616_TDM_I2S_CH4_SEL_LL (0x2 << 4) +#define RT5616_TDM_I2S_CH4_SEL_RR (0x3 << 4) +#define RT5616_TDM_I2S_CH6_SEL_MASK (0x3 << 2) +#define RT5616_TDM_I2S_CH6_SEL_SFT 2 +#define RT5616_TDM_I2S_CH6_SEL_LR (0x0 << 2) +#define RT5616_TDM_I2S_CH6_SEL_RL (0x1 << 2) +#define RT5616_TDM_I2S_CH6_SEL_LL (0x2 << 2) +#define RT5616_TDM_I2S_CH6_SEL_RR (0x3 << 2) +#define RT5616_TDM_I2S_CH8_SEL_MASK (0x3) +#define RT5616_TDM_I2S_CH8_SEL_SFT 0 +#define RT5616_TDM_I2S_CH8_SEL_LR (0x0) +#define RT5616_TDM_I2S_CH8_SEL_RL (0x1) +#define RT5616_TDM_I2S_CH8_SEL_LL (0x2) +#define RT5616_TDM_I2S_CH8_SEL_RR (0x3) + +/* TDM Control 2 (0x78) */ +#define RT5616_TDM_LRCK_POL_SEL_MASK (0x1 << 15) +#define RT5616_TDM_LRCK_POL_SEL_SFT 15 +#define RT5616_TDM_LRCK_POL_SEL_NOR (0x0 << 15) +#define RT5616_TDM_LRCK_POL_SEL_INV (0x1 << 15) +#define RT5616_TDM_CH_VAL_SEL_MASK (0x1 << 14) +#define RT5616_TDM_CH_VAL_SEL_SFT 14 +#define RT5616_TDM_CH_VAL_SEL_CH01 (0x0 << 14) +#define RT5616_TDM_CH_VAL_SEL_CH0123 (0x1 << 14) +#define RT5616_TDM_CH_VAL_EN (0x1 << 13) +#define RT5616_TDM_CH_VAL_SFT 13 +#define RT5616_TDM_LPBK_EN (0x1 << 12) +#define RT5616_TDM_LPBK_SFT 12 +#define RT5616_TDM_LRCK_PULSE_SEL_MASK (0x1 << 11) +#define RT5616_TDM_LRCK_PULSE_SEL_SFT 11 +#define RT5616_TDM_LRCK_PULSE_SEL_BCLK (0x0 << 11) +#define RT5616_TDM_LRCK_PULSE_SEL_CH (0x1 << 11) +#define RT5616_TDM_END_EDGE_SEL_MASK (0x1 << 10) +#define RT5616_TDM_END_EDGE_SEL_SFT 10 +#define RT5616_TDM_END_EDGE_SEL_POS (0x0 << 10) +#define RT5616_TDM_END_EDGE_SEL_NEG (0x1 << 10) +#define RT5616_TDM_END_EDGE_EN (0x1 << 9) +#define RT5616_TDM_END_EDGE_EN_SFT 9 +#define RT5616_TDM_TRAN_EDGE_SEL_MASK (0x1 << 8) +#define RT5616_TDM_TRAN_EDGE_SEL_SFT 8 +#define RT5616_TDM_TRAN_EDGE_SEL_POS (0x0 << 8) +#define RT5616_TDM_TRAN_EDGE_SEL_NEG (0x1 << 8) +#define RT5616_M_TDM2_L (0x1 << 7) +#define RT5616_M_TDM2_L_SFT 7 +#define RT5616_M_TDM2_R (0x1 << 6) +#define RT5616_M_TDM2_R_SFT 6 +#define RT5616_M_TDM4_L (0x1 << 5) +#define RT5616_M_TDM4_L_SFT 5 +#define RT5616_M_TDM4_R (0x1 << 4) +#define RT5616_M_TDM4_R_SFT 4 + +/* Global Clock Control (0x80) */ +#define RT5616_SCLK_SRC_MASK (0x3 << 14) +#define RT5616_SCLK_SRC_SFT 14 +#define RT5616_SCLK_SRC_MCLK (0x0 << 14) +#define RT5616_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5616_PLL1_SRC_MASK (0x3 << 12) +#define RT5616_PLL1_SRC_SFT 12 +#define RT5616_PLL1_SRC_MCLK (0x0 << 12) +#define RT5616_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5616_PLL1_SRC_BCLK2 (0x2 << 12) +#define RT5616_PLL1_PD_MASK (0x1 << 3) +#define RT5616_PLL1_PD_SFT 3 +#define RT5616_PLL1_PD_1 (0x0 << 3) +#define RT5616_PLL1_PD_2 (0x1 << 3) + +#define RT5616_PLL_INP_MAX 40000000 +#define RT5616_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5616_PLL_N_MAX 0x1ff +#define RT5616_PLL_N_MASK (RT5616_PLL_N_MAX << 7) +#define RT5616_PLL_N_SFT 7 +#define RT5616_PLL_K_MAX 0x1f +#define RT5616_PLL_K_MASK (RT5616_PLL_K_MAX) +#define RT5616_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5616_PLL_M_MAX 0xf +#define RT5616_PLL_M_MASK (RT5616_PLL_M_MAX << 12) +#define RT5616_PLL_M_SFT 12 +#define RT5616_PLL_M_BP (0x1 << 11) +#define RT5616_PLL_M_BP_SFT 11 + +/* PLL tracking mode 1 (0x83) */ +#define RT5616_STO1_T_MASK (0x1 << 15) +#define RT5616_STO1_T_SFT 15 +#define RT5616_STO1_T_SCLK (0x0 << 15) +#define RT5616_STO1_T_LRCK1 (0x1 << 15) +#define RT5616_STO2_T_MASK (0x1 << 12) +#define RT5616_STO2_T_SFT 12 +#define RT5616_STO2_T_I2S2 (0x0 << 12) +#define RT5616_STO2_T_LRCK2 (0x1 << 12) +#define RT5616_ASRC2_REF_MASK (0x1 << 11) +#define RT5616_ASRC2_REF_SFT 11 +#define RT5616_ASRC2_REF_LRCK2 (0x0 << 11) +#define RT5616_ASRC2_REF_LRCK1 (0x1 << 11) +#define RT5616_DMIC_1_M_MASK (0x1 << 9) +#define RT5616_DMIC_1_M_SFT 9 +#define RT5616_DMIC_1_M_NOR (0x0 << 9) +#define RT5616_DMIC_1_M_ASYN (0x1 << 9) + +/* PLL tracking mode 2 (0x84) */ +#define RT5616_STO1_ASRC_EN (0x1 << 15) +#define RT5616_STO1_ASRC_EN_SFT 15 +#define RT5616_STO2_ASRC_EN (0x1 << 14) +#define RT5616_STO2_ASRC_EN_SFT 14 +#define RT5616_STO1_DAC_M_MASK (0x1 << 13) +#define RT5616_STO1_DAC_M_SFT 13 +#define RT5616_STO1_DAC_M_NOR (0x0 << 13) +#define RT5616_STO1_DAC_M_ASRC (0x1 << 13) +#define RT5616_STO2_DAC_M_MASK (0x1 << 12) +#define RT5616_STO2_DAC_M_SFT 12 +#define RT5616_STO2_DAC_M_NOR (0x0 << 12) +#define RT5616_STO2_DAC_M_ASRC (0x1 << 12) +#define RT5616_ADC_M_MASK (0x1 << 11) +#define RT5616_ADC_M_SFT 11 +#define RT5616_ADC_M_NOR (0x0 << 11) +#define RT5616_ADC_M_ASRC (0x1 << 11) +#define RT5616_I2S1_R_D_MASK (0x1 << 4) +#define RT5616_I2S1_R_D_SFT 4 +#define RT5616_I2S1_R_D_DIS (0x0 << 4) +#define RT5616_I2S1_R_D_EN (0x1 << 4) +#define RT5616_I2S2_R_D_MASK (0x1 << 3) +#define RT5616_I2S2_R_D_SFT 3 +#define RT5616_I2S2_R_D_DIS (0x0 << 3) +#define RT5616_I2S2_R_D_EN (0x1 << 3) +#define RT5616_PRE_SCLK_MASK (0x3) +#define RT5616_PRE_SCLK_SFT 0 +#define RT5616_PRE_SCLK_512 (0x0) +#define RT5616_PRE_SCLK_1024 (0x1) +#define RT5616_PRE_SCLK_2048 (0x2) + +/* PLL tracking mode 3 (0x85) */ +#define RT5616_I2S1_RATE_MASK (0xf << 12) +#define RT5616_I2S1_RATE_SFT 12 +#define RT5616_I2S2_RATE_MASK (0xf << 8) +#define RT5616_I2S2_RATE_SFT 8 +#define RT5616_G_ASRC_LP_MASK (0x1 << 3) +#define RT5616_G_ASRC_LP_SFT 3 +#define RT5616_ASRC_LP_F_M (0x1 << 2) +#define RT5616_ASRC_LP_F_SFT 2 +#define RT5616_ASRC_LP_F_NOR (0x0 << 2) +#define RT5616_ASRC_LP_F_SB (0x1 << 2) +#define RT5616_FTK_PH_DET_MASK (0x3) +#define RT5616_FTK_PH_DET_SFT 0 +#define RT5616_FTK_PH_DET_DIV1 (0x0) +#define RT5616_FTK_PH_DET_DIV2 (0x1) +#define RT5616_FTK_PH_DET_DIV4 (0x2) +#define RT5616_FTK_PH_DET_DIV8 (0x3) + +/*PLL tracking mode 6 (0x89) */ +#define RT5616_I2S1_PD_MASK (0x7 << 12) +#define RT5616_I2S1_PD_SFT 12 +#define RT5616_I2S2_PD_MASK (0x7 << 8) +#define RT5616_I2S2_PD_SFT 8 + +/*PLL tracking mode 7 (0x8a) */ +#define RT5616_FSI1_RATE_MASK (0xf << 12) +#define RT5616_FSI1_RATE_SFT 12 +#define RT5616_FSI2_RATE_MASK (0xf << 8) +#define RT5616_FSI2_RATE_SFT 8 + +/* HPOUT Over Current Detection (0x8b) */ +#define RT5616_HP_OVCD_MASK (0x1 << 10) +#define RT5616_HP_OVCD_SFT 10 +#define RT5616_HP_OVCD_DIS (0x0 << 10) +#define RT5616_HP_OVCD_EN (0x1 << 10) +#define RT5616_HP_OC_TH_MASK (0x3 << 8) +#define RT5616_HP_OC_TH_SFT 8 +#define RT5616_HP_OC_TH_90 (0x0 << 8) +#define RT5616_HP_OC_TH_105 (0x1 << 8) +#define RT5616_HP_OC_TH_120 (0x2 << 8) +#define RT5616_HP_OC_TH_135 (0x3 << 8) + +/* Depop Mode Control 1 (0x8e) */ +#define RT5616_SMT_TRIG_MASK (0x1 << 15) +#define RT5616_SMT_TRIG_SFT 15 +#define RT5616_SMT_TRIG_DIS (0x0 << 15) +#define RT5616_SMT_TRIG_EN (0x1 << 15) +#define RT5616_HP_L_SMT_MASK (0x1 << 9) +#define RT5616_HP_L_SMT_SFT 9 +#define RT5616_HP_L_SMT_DIS (0x0 << 9) +#define RT5616_HP_L_SMT_EN (0x1 << 9) +#define RT5616_HP_R_SMT_MASK (0x1 << 8) +#define RT5616_HP_R_SMT_SFT 8 +#define RT5616_HP_R_SMT_DIS (0x0 << 8) +#define RT5616_HP_R_SMT_EN (0x1 << 8) +#define RT5616_HP_CD_PD_MASK (0x1 << 7) +#define RT5616_HP_CD_PD_SFT 7 +#define RT5616_HP_CD_PD_DIS (0x0 << 7) +#define RT5616_HP_CD_PD_EN (0x1 << 7) +#define RT5616_RSTN_MASK (0x1 << 6) +#define RT5616_RSTN_SFT 6 +#define RT5616_RSTN_DIS (0x0 << 6) +#define RT5616_RSTN_EN (0x1 << 6) +#define RT5616_RSTP_MASK (0x1 << 5) +#define RT5616_RSTP_SFT 5 +#define RT5616_RSTP_DIS (0x0 << 5) +#define RT5616_RSTP_EN (0x1 << 5) +#define RT5616_HP_CO_MASK (0x1 << 4) +#define RT5616_HP_CO_SFT 4 +#define RT5616_HP_CO_DIS (0x0 << 4) +#define RT5616_HP_CO_EN (0x1 << 4) +#define RT5616_HP_CP_MASK (0x1 << 3) +#define RT5616_HP_CP_SFT 3 +#define RT5616_HP_CP_PD (0x0 << 3) +#define RT5616_HP_CP_PU (0x1 << 3) +#define RT5616_HP_SG_MASK (0x1 << 2) +#define RT5616_HP_SG_SFT 2 +#define RT5616_HP_SG_DIS (0x0 << 2) +#define RT5616_HP_SG_EN (0x1 << 2) +#define RT5616_HP_DP_MASK (0x1 << 1) +#define RT5616_HP_DP_SFT 1 +#define RT5616_HP_DP_PD (0x0 << 1) +#define RT5616_HP_DP_PU (0x1 << 1) +#define RT5616_HP_CB_MASK (0x1) +#define RT5616_HP_CB_SFT 0 +#define RT5616_HP_CB_PD (0x0) +#define RT5616_HP_CB_PU (0x1) + +/* Depop Mode Control 2 (0x8f) */ +#define RT5616_DEPOP_MASK (0x1 << 13) +#define RT5616_DEPOP_SFT 13 +#define RT5616_DEPOP_AUTO (0x0 << 13) +#define RT5616_DEPOP_MAN (0x1 << 13) +#define RT5616_RAMP_MASK (0x1 << 12) +#define RT5616_RAMP_SFT 12 +#define RT5616_RAMP_DIS (0x0 << 12) +#define RT5616_RAMP_EN (0x1 << 12) +#define RT5616_BPS_MASK (0x1 << 11) +#define RT5616_BPS_SFT 11 +#define RT5616_BPS_DIS (0x0 << 11) +#define RT5616_BPS_EN (0x1 << 11) +#define RT5616_FAST_UPDN_MASK (0x1 << 10) +#define RT5616_FAST_UPDN_SFT 10 +#define RT5616_FAST_UPDN_DIS (0x0 << 10) +#define RT5616_FAST_UPDN_EN (0x1 << 10) +#define RT5616_MRES_MASK (0x3 << 8) +#define RT5616_MRES_SFT 8 +#define RT5616_MRES_15MO (0x0 << 8) +#define RT5616_MRES_25MO (0x1 << 8) +#define RT5616_MRES_35MO (0x2 << 8) +#define RT5616_MRES_45MO (0x3 << 8) +#define RT5616_VLO_MASK (0x1 << 7) +#define RT5616_VLO_SFT 7 +#define RT5616_VLO_3V (0x0 << 7) +#define RT5616_VLO_32V (0x1 << 7) +#define RT5616_DIG_DP_MASK (0x1 << 6) +#define RT5616_DIG_DP_SFT 6 +#define RT5616_DIG_DP_DIS (0x0 << 6) +#define RT5616_DIG_DP_EN (0x1 << 6) +#define RT5616_DP_TH_MASK (0x3 << 4) +#define RT5616_DP_TH_SFT 4 + +/* Depop Mode Control 3 (0x90) */ +#define RT5616_CP_SYS_MASK (0x7 << 12) +#define RT5616_CP_SYS_SFT 12 +#define RT5616_CP_FQ1_MASK (0x7 << 8) +#define RT5616_CP_FQ1_SFT 8 +#define RT5616_CP_FQ2_MASK (0x7 << 4) +#define RT5616_CP_FQ2_SFT 4 +#define RT5616_CP_FQ3_MASK (0x7) +#define RT5616_CP_FQ3_SFT 0 +#define RT5616_CP_FQ_1_5_KHZ 0 +#define RT5616_CP_FQ_3_KHZ 1 +#define RT5616_CP_FQ_6_KHZ 2 +#define RT5616_CP_FQ_12_KHZ 3 +#define RT5616_CP_FQ_24_KHZ 4 +#define RT5616_CP_FQ_48_KHZ 5 +#define RT5616_CP_FQ_96_KHZ 6 +#define RT5616_CP_FQ_192_KHZ 7 + +/* HPOUT charge pump (0x91) */ +#define RT5616_OSW_L_MASK (0x1 << 11) +#define RT5616_OSW_L_SFT 11 +#define RT5616_OSW_L_DIS (0x0 << 11) +#define RT5616_OSW_L_EN (0x1 << 11) +#define RT5616_OSW_R_MASK (0x1 << 10) +#define RT5616_OSW_R_SFT 10 +#define RT5616_OSW_R_DIS (0x0 << 10) +#define RT5616_OSW_R_EN (0x1 << 10) +#define RT5616_PM_HP_MASK (0x3 << 8) +#define RT5616_PM_HP_SFT 8 +#define RT5616_PM_HP_LV (0x0 << 8) +#define RT5616_PM_HP_MV (0x1 << 8) +#define RT5616_PM_HP_HV (0x2 << 8) +#define RT5616_IB_HP_MASK (0x3 << 6) +#define RT5616_IB_HP_SFT 6 +#define RT5616_IB_HP_125IL (0x0 << 6) +#define RT5616_IB_HP_25IL (0x1 << 6) +#define RT5616_IB_HP_5IL (0x2 << 6) +#define RT5616_IB_HP_1IL (0x3 << 6) + +/* Micbias Control (0x93) */ +#define RT5616_MIC1_BS_MASK (0x1 << 15) +#define RT5616_MIC1_BS_SFT 15 +#define RT5616_MIC1_BS_9AV (0x0 << 15) +#define RT5616_MIC1_BS_75AV (0x1 << 15) +#define RT5616_MIC1_CLK_MASK (0x1 << 13) +#define RT5616_MIC1_CLK_SFT 13 +#define RT5616_MIC1_CLK_DIS (0x0 << 13) +#define RT5616_MIC1_CLK_EN (0x1 << 13) +#define RT5616_MIC1_OVCD_MASK (0x1 << 11) +#define RT5616_MIC1_OVCD_SFT 11 +#define RT5616_MIC1_OVCD_DIS (0x0 << 11) +#define RT5616_MIC1_OVCD_EN (0x1 << 11) +#define RT5616_MIC1_OVTH_MASK (0x3 << 9) +#define RT5616_MIC1_OVTH_SFT 9 +#define RT5616_MIC1_OVTH_600UA (0x0 << 9) +#define RT5616_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5616_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5616_PWR_MB_MASK (0x1 << 5) +#define RT5616_PWR_MB_SFT 5 +#define RT5616_PWR_MB_PD (0x0 << 5) +#define RT5616_PWR_MB_PU (0x1 << 5) +#define RT5616_PWR_CLK12M_MASK (0x1 << 4) +#define RT5616_PWR_CLK12M_SFT 4 +#define RT5616_PWR_CLK12M_PD (0x0 << 4) +#define RT5616_PWR_CLK12M_PU (0x1 << 4) + +/* Analog JD Control 1 (0x94) */ +#define RT5616_JD2_CMP_MASK (0x7 << 12) +#define RT5616_JD2_CMP_SFT 12 +#define RT5616_JD_PU (0x1 << 11) +#define RT5616_JD_PU_SFT 11 +#define RT5616_JD_PD (0x1 << 10) +#define RT5616_JD_PD_SFT 10 +#define RT5616_JD_MODE_SEL_MASK (0x3 << 8) +#define RT5616_JD_MODE_SEL_SFT 8 +#define RT5616_JD_MODE_SEL_M0 (0x0 << 8) +#define RT5616_JD_MODE_SEL_M1 (0x1 << 8) +#define RT5616_JD_MODE_SEL_M2 (0x2 << 8) +#define RT5616_JD_M_CMP (0x7 << 4) +#define RT5616_JD_M_CMP_SFT 4 +#define RT5616_JD_M_PU (0x1 << 3) +#define RT5616_JD_M_PU_SFT 3 +#define RT5616_JD_M_PD (0x1 << 2) +#define RT5616_JD_M_PD_SFT 2 +#define RT5616_JD_M_MODE_SEL_MASK (0x3) +#define RT5616_JD_M_MODE_SEL_SFT 0 +#define RT5616_JD_M_MODE_SEL_M0 (0x0) +#define RT5616_JD_M_MODE_SEL_M1 (0x1) +#define RT5616_JD_M_MODE_SEL_M2 (0x2) + +/* Analog JD Control 2 (0x95) */ +#define RT5616_JD3_CMP_MASK (0x7 << 12) +#define RT5616_JD3_CMP_SFT 12 + +/* EQ Control 1 (0xb0) */ +#define RT5616_EQ_SRC_MASK (0x1 << 15) +#define RT5616_EQ_SRC_SFT 15 +#define RT5616_EQ_SRC_DAC (0x0 << 15) +#define RT5616_EQ_SRC_ADC (0x1 << 15) +#define RT5616_EQ_UPD (0x1 << 14) +#define RT5616_EQ_UPD_BIT 14 +#define RT5616_EQ_CD_MASK (0x1 << 13) +#define RT5616_EQ_CD_SFT 13 +#define RT5616_EQ_CD_DIS (0x0 << 13) +#define RT5616_EQ_CD_EN (0x1 << 13) +#define RT5616_EQ_DITH_MASK (0x3 << 8) +#define RT5616_EQ_DITH_SFT 8 +#define RT5616_EQ_DITH_NOR (0x0 << 8) +#define RT5616_EQ_DITH_LSB (0x1 << 8) +#define RT5616_EQ_DITH_LSB_1 (0x2 << 8) +#define RT5616_EQ_DITH_LSB_2 (0x3 << 8) +#define RT5616_EQ_CD_F (0x1 << 7) +#define RT5616_EQ_CD_F_BIT 7 +#define RT5616_EQ_STA_HP2 (0x1 << 6) +#define RT5616_EQ_STA_HP2_BIT 6 +#define RT5616_EQ_STA_HP1 (0x1 << 5) +#define RT5616_EQ_STA_HP1_BIT 5 +#define RT5616_EQ_STA_BP4 (0x1 << 4) +#define RT5616_EQ_STA_BP4_BIT 4 +#define RT5616_EQ_STA_BP3 (0x1 << 3) +#define RT5616_EQ_STA_BP3_BIT 3 +#define RT5616_EQ_STA_BP2 (0x1 << 2) +#define RT5616_EQ_STA_BP2_BIT 2 +#define RT5616_EQ_STA_BP1 (0x1 << 1) +#define RT5616_EQ_STA_BP1_BIT 1 +#define RT5616_EQ_STA_LP (0x1) +#define RT5616_EQ_STA_LP_BIT 0 + +/* EQ Control 2 (0xb1) */ +#define RT5616_EQ_HPF1_M_MASK (0x1 << 8) +#define RT5616_EQ_HPF1_M_SFT 8 +#define RT5616_EQ_HPF1_M_HI (0x0 << 8) +#define RT5616_EQ_HPF1_M_1ST (0x1 << 8) +#define RT5616_EQ_LPF1_M_MASK (0x1 << 7) +#define RT5616_EQ_LPF1_M_SFT 7 +#define RT5616_EQ_LPF1_M_LO (0x0 << 7) +#define RT5616_EQ_LPF1_M_1ST (0x1 << 7) +#define RT5616_EQ_HPF2_MASK (0x1 << 6) +#define RT5616_EQ_HPF2_SFT 6 +#define RT5616_EQ_HPF2_DIS (0x0 << 6) +#define RT5616_EQ_HPF2_EN (0x1 << 6) +#define RT5616_EQ_HPF1_MASK (0x1 << 5) +#define RT5616_EQ_HPF1_SFT 5 +#define RT5616_EQ_HPF1_DIS (0x0 << 5) +#define RT5616_EQ_HPF1_EN (0x1 << 5) +#define RT5616_EQ_BPF4_MASK (0x1 << 4) +#define RT5616_EQ_BPF4_SFT 4 +#define RT5616_EQ_BPF4_DIS (0x0 << 4) +#define RT5616_EQ_BPF4_EN (0x1 << 4) +#define RT5616_EQ_BPF3_MASK (0x1 << 3) +#define RT5616_EQ_BPF3_SFT 3 +#define RT5616_EQ_BPF3_DIS (0x0 << 3) +#define RT5616_EQ_BPF3_EN (0x1 << 3) +#define RT5616_EQ_BPF2_MASK (0x1 << 2) +#define RT5616_EQ_BPF2_SFT 2 +#define RT5616_EQ_BPF2_DIS (0x0 << 2) +#define RT5616_EQ_BPF2_EN (0x1 << 2) +#define RT5616_EQ_BPF1_MASK (0x1 << 1) +#define RT5616_EQ_BPF1_SFT 1 +#define RT5616_EQ_BPF1_DIS (0x0 << 1) +#define RT5616_EQ_BPF1_EN (0x1 << 1) +#define RT5616_EQ_LPF_MASK (0x1) +#define RT5616_EQ_LPF_SFT 0 +#define RT5616_EQ_LPF_DIS (0x0) +#define RT5616_EQ_LPF_EN (0x1) +#define RT5616_EQ_CTRL_MASK (0x7f) + +/* Memory Test (0xb2) */ +#define RT5616_MT_MASK (0x1 << 15) +#define RT5616_MT_SFT 15 +#define RT5616_MT_DIS (0x0 << 15) +#define RT5616_MT_EN (0x1 << 15) + +/* DRC/AGC Control 1 (0xb4) */ +#define RT5616_DRC_AGC_P_MASK (0x1 << 15) +#define RT5616_DRC_AGC_P_SFT 15 +#define RT5616_DRC_AGC_P_DAC (0x0 << 15) +#define RT5616_DRC_AGC_P_ADC (0x1 << 15) +#define RT5616_DRC_AGC_MASK (0x1 << 14) +#define RT5616_DRC_AGC_SFT 14 +#define RT5616_DRC_AGC_DIS (0x0 << 14) +#define RT5616_DRC_AGC_EN (0x1 << 14) +#define RT5616_DRC_AGC_UPD (0x1 << 13) +#define RT5616_DRC_AGC_UPD_BIT 13 +#define RT5616_DRC_AGC_AR_MASK (0x1f << 8) +#define RT5616_DRC_AGC_AR_SFT 8 +#define RT5616_DRC_AGC_R_MASK (0x7 << 5) +#define RT5616_DRC_AGC_R_SFT 5 +#define RT5616_DRC_AGC_R_48K (0x1 << 5) +#define RT5616_DRC_AGC_R_96K (0x2 << 5) +#define RT5616_DRC_AGC_R_192K (0x3 << 5) +#define RT5616_DRC_AGC_R_441K (0x5 << 5) +#define RT5616_DRC_AGC_R_882K (0x6 << 5) +#define RT5616_DRC_AGC_R_1764K (0x7 << 5) +#define RT5616_DRC_AGC_RC_MASK (0x1f) +#define RT5616_DRC_AGC_RC_SFT 0 + +/* DRC/AGC Control 2 (0xb5) */ +#define RT5616_DRC_AGC_POB_MASK (0x3f << 8) +#define RT5616_DRC_AGC_POB_SFT 8 +#define RT5616_DRC_AGC_CP_MASK (0x1 << 7) +#define RT5616_DRC_AGC_CP_SFT 7 +#define RT5616_DRC_AGC_CP_DIS (0x0 << 7) +#define RT5616_DRC_AGC_CP_EN (0x1 << 7) +#define RT5616_DRC_AGC_CPR_MASK (0x3 << 5) +#define RT5616_DRC_AGC_CPR_SFT 5 +#define RT5616_DRC_AGC_CPR_1_1 (0x0 << 5) +#define RT5616_DRC_AGC_CPR_1_2 (0x1 << 5) +#define RT5616_DRC_AGC_CPR_1_3 (0x2 << 5) +#define RT5616_DRC_AGC_CPR_1_4 (0x3 << 5) +#define RT5616_DRC_AGC_PRB_MASK (0x1f) +#define RT5616_DRC_AGC_PRB_SFT 0 + +/* DRC/AGC Control 3 (0xb6) */ +#define RT5616_DRC_AGC_NGB_MASK (0xf << 12) +#define RT5616_DRC_AGC_NGB_SFT 12 +#define RT5616_DRC_AGC_TAR_MASK (0x1f << 7) +#define RT5616_DRC_AGC_TAR_SFT 7 +#define RT5616_DRC_AGC_NG_MASK (0x1 << 6) +#define RT5616_DRC_AGC_NG_SFT 6 +#define RT5616_DRC_AGC_NG_DIS (0x0 << 6) +#define RT5616_DRC_AGC_NG_EN (0x1 << 6) +#define RT5616_DRC_AGC_NGH_MASK (0x1 << 5) +#define RT5616_DRC_AGC_NGH_SFT 5 +#define RT5616_DRC_AGC_NGH_DIS (0x0 << 5) +#define RT5616_DRC_AGC_NGH_EN (0x1 << 5) +#define RT5616_DRC_AGC_NGT_MASK (0x1f) +#define RT5616_DRC_AGC_NGT_SFT 0 + +/* Jack Detect Control 1 (0xbb) */ +#define RT5616_JD_MASK (0x7 << 13) +#define RT5616_JD_SFT 13 +#define RT5616_JD_DIS (0x0 << 13) +#define RT5616_JD_GPIO1 (0x1 << 13) +#define RT5616_JD_GPIO2 (0x2 << 13) +#define RT5616_JD_GPIO3 (0x3 << 13) +#define RT5616_JD_GPIO4 (0x4 << 13) +#define RT5616_JD_GPIO5 (0x5 << 13) +#define RT5616_JD_GPIO6 (0x6 << 13) +#define RT5616_JD_HP_MASK (0x1 << 11) +#define RT5616_JD_HP_SFT 11 +#define RT5616_JD_HP_DIS (0x0 << 11) +#define RT5616_JD_HP_EN (0x1 << 11) +#define RT5616_JD_HP_TRG_MASK (0x1 << 10) +#define RT5616_JD_HP_TRG_SFT 10 +#define RT5616_JD_HP_TRG_LO (0x0 << 10) +#define RT5616_JD_HP_TRG_HI (0x1 << 10) +#define RT5616_JD_SPL_MASK (0x1 << 9) +#define RT5616_JD_SPL_SFT 9 +#define RT5616_JD_SPL_DIS (0x0 << 9) +#define RT5616_JD_SPL_EN (0x1 << 9) +#define RT5616_JD_SPL_TRG_MASK (0x1 << 8) +#define RT5616_JD_SPL_TRG_SFT 8 +#define RT5616_JD_SPL_TRG_LO (0x0 << 8) +#define RT5616_JD_SPL_TRG_HI (0x1 << 8) +#define RT5616_JD_SPR_MASK (0x1 << 7) +#define RT5616_JD_SPR_SFT 7 +#define RT5616_JD_SPR_DIS (0x0 << 7) +#define RT5616_JD_SPR_EN (0x1 << 7) +#define RT5616_JD_SPR_TRG_MASK (0x1 << 6) +#define RT5616_JD_SPR_TRG_SFT 6 +#define RT5616_JD_SPR_TRG_LO (0x0 << 6) +#define RT5616_JD_SPR_TRG_HI (0x1 << 6) +#define RT5616_JD_LO_MASK (0x1 << 3) +#define RT5616_JD_LO_SFT 3 +#define RT5616_JD_LO_DIS (0x0 << 3) +#define RT5616_JD_LO_EN (0x1 << 3) +#define RT5616_JD_LO_TRG_MASK (0x1 << 2) +#define RT5616_JD_LO_TRG_SFT 2 +#define RT5616_JD_LO_TRG_LO (0x0 << 2) +#define RT5616_JD_LO_TRG_HI (0x1 << 2) + +/* Jack Detect Control 2 (0xbc) */ +#define RT5616_JD_TRG_SEL_MASK (0x7 << 9) +#define RT5616_JD_TRG_SEL_SFT 9 +#define RT5616_JD_TRG_SEL_GPIO (0x0 << 9) +#define RT5616_JD_TRG_SEL_JD1_1 (0x1 << 9) +#define RT5616_JD_TRG_SEL_JD1_2 (0x2 << 9) +#define RT5616_JD_TRG_SEL_JD2 (0x3 << 9) +#define RT5616_JD_TRG_SEL_JD3 (0x4 << 9) +#define RT5616_JD3_IRQ_EN (0x1 << 8) +#define RT5616_JD3_IRQ_EN_SFT 8 +#define RT5616_JD3_EN_STKY (0x1 << 7) +#define RT5616_JD3_EN_STKY_SFT 7 +#define RT5616_JD3_INV (0x1 << 6) +#define RT5616_JD3_INV_SFT 6 + +/* IRQ Control 1 (0xbd) */ +#define RT5616_IRQ_JD_MASK (0x1 << 15) +#define RT5616_IRQ_JD_SFT 15 +#define RT5616_IRQ_JD_BP (0x0 << 15) +#define RT5616_IRQ_JD_NOR (0x1 << 15) +#define RT5616_JD_STKY_MASK (0x1 << 13) +#define RT5616_JD_STKY_SFT 13 +#define RT5616_JD_STKY_DIS (0x0 << 13) +#define RT5616_JD_STKY_EN (0x1 << 13) +#define RT5616_JD_P_MASK (0x1 << 11) +#define RT5616_JD_P_SFT 11 +#define RT5616_JD_P_NOR (0x0 << 11) +#define RT5616_JD_P_INV (0x1 << 11) +#define RT5616_JD1_1_IRQ_EN (0x1 << 9) +#define RT5616_JD1_1_IRQ_EN_SFT 9 +#define RT5616_JD1_1_EN_STKY (0x1 << 8) +#define RT5616_JD1_1_EN_STKY_SFT 8 +#define RT5616_JD1_1_INV (0x1 << 7) +#define RT5616_JD1_1_INV_SFT 7 +#define RT5616_JD1_2_IRQ_EN (0x1 << 6) +#define RT5616_JD1_2_IRQ_EN_SFT 6 +#define RT5616_JD1_2_EN_STKY (0x1 << 5) +#define RT5616_JD1_2_EN_STKY_SFT 5 +#define RT5616_JD1_2_INV (0x1 << 4) +#define RT5616_JD1_2_INV_SFT 4 +#define RT5616_JD2_IRQ_EN (0x1 << 3) +#define RT5616_JD2_IRQ_EN_SFT 3 +#define RT5616_JD2_EN_STKY (0x1 << 2) +#define RT5616_JD2_EN_STKY_SFT 2 +#define RT5616_JD2_INV (0x1 << 1) +#define RT5616_JD2_INV_SFT 1 + +/* IRQ Control 2 (0xbe) */ +#define RT5616_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5616_IRQ_MB1_OC_SFT 15 +#define RT5616_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5616_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5616_MB1_OC_STKY_MASK (0x1 << 11) +#define RT5616_MB1_OC_STKY_SFT 11 +#define RT5616_MB1_OC_STKY_DIS (0x0 << 11) +#define RT5616_MB1_OC_STKY_EN (0x1 << 11) +#define RT5616_MB1_OC_P_MASK (0x1 << 7) +#define RT5616_MB1_OC_P_SFT 7 +#define RT5616_MB1_OC_P_NOR (0x0 << 7) +#define RT5616_MB1_OC_P_INV (0x1 << 7) +#define RT5616_MB2_OC_P_MASK (0x1 << 6) +#define RT5616_MB1_OC_CLR (0x1 << 3) +#define RT5616_MB1_OC_CLR_SFT 3 +#define RT5616_STA_GPIO8 (0x1) +#define RT5616_STA_GPIO8_BIT 0 + +/* Internal Status and GPIO status (0xbf) */ +#define RT5616_STA_JD3 (0x1 << 15) +#define RT5616_STA_JD3_BIT 15 +#define RT5616_STA_JD2 (0x1 << 14) +#define RT5616_STA_JD2_BIT 14 +#define RT5616_STA_JD1_2 (0x1 << 13) +#define RT5616_STA_JD1_2_BIT 13 +#define RT5616_STA_JD1_1 (0x1 << 12) +#define RT5616_STA_JD1_1_BIT 12 +#define RT5616_STA_GP7 (0x1 << 11) +#define RT5616_STA_GP7_BIT 11 +#define RT5616_STA_GP6 (0x1 << 10) +#define RT5616_STA_GP6_BIT 10 +#define RT5616_STA_GP5 (0x1 << 9) +#define RT5616_STA_GP5_BIT 9 +#define RT5616_STA_GP1 (0x1 << 8) +#define RT5616_STA_GP1_BIT 8 +#define RT5616_STA_GP2 (0x1 << 7) +#define RT5616_STA_GP2_BIT 7 +#define RT5616_STA_GP3 (0x1 << 6) +#define RT5616_STA_GP3_BIT 6 +#define RT5616_STA_GP4 (0x1 << 5) +#define RT5616_STA_GP4_BIT 5 +#define RT5616_STA_GP_JD (0x1 << 4) +#define RT5616_STA_GP_JD_BIT 4 + +/* GPIO Control 1 (0xc0) */ +#define RT5616_GP1_PIN_MASK (0x1 << 15) +#define RT5616_GP1_PIN_SFT 15 +#define RT5616_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5616_GP1_PIN_IRQ (0x1 << 15) +#define RT5616_GP2_PIN_MASK (0x1 << 14) +#define RT5616_GP2_PIN_SFT 14 +#define RT5616_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5616_GP2_PIN_DMIC1_SCL (0x1 << 14) +#define RT5616_GPIO_M_MASK (0x1 << 9) +#define RT5616_GPIO_M_SFT 9 +#define RT5616_GPIO_M_FLT (0x0 << 9) +#define RT5616_GPIO_M_PH (0x1 << 9) +#define RT5616_I2S2_SEL_MASK (0x1 << 8) +#define RT5616_I2S2_SEL_SFT 8 +#define RT5616_I2S2_SEL_I2S (0x0 << 8) +#define RT5616_I2S2_SEL_GPIO (0x1 << 8) +#define RT5616_GP5_PIN_MASK (0x1 << 7) +#define RT5616_GP5_PIN_SFT 7 +#define RT5616_GP5_PIN_GPIO5 (0x0 << 7) +#define RT5616_GP5_PIN_IRQ (0x1 << 7) +#define RT5616_GP6_PIN_MASK (0x1 << 6) +#define RT5616_GP6_PIN_SFT 6 +#define RT5616_GP6_PIN_GPIO6 (0x0 << 6) +#define RT5616_GP6_PIN_DMIC_SDA (0x1 << 6) +#define RT5616_GP7_PIN_MASK (0x1 << 5) +#define RT5616_GP7_PIN_SFT 5 +#define RT5616_GP7_PIN_GPIO7 (0x0 << 5) +#define RT5616_GP7_PIN_IRQ (0x1 << 5) +#define RT5616_GP8_PIN_MASK (0x1 << 4) +#define RT5616_GP8_PIN_SFT 4 +#define RT5616_GP8_PIN_GPIO8 (0x0 << 4) +#define RT5616_GP8_PIN_DMIC_SDA (0x1 << 4) +#define RT5616_GPIO_PDM_SEL_MASK (0x1 << 3) +#define RT5616_GPIO_PDM_SEL_SFT 3 +#define RT5616_GPIO_PDM_SEL_GPIO (0x0 << 3) +#define RT5616_GPIO_PDM_SEL_PDM (0x1 << 3) + +/* GPIO Control 2 (0xc1) */ +#define RT5616_GP5_DR_MASK (0x1 << 14) +#define RT5616_GP5_DR_SFT 14 +#define RT5616_GP5_DR_IN (0x0 << 14) +#define RT5616_GP5_DR_OUT (0x1 << 14) +#define RT5616_GP5_OUT_MASK (0x1 << 13) +#define RT5616_GP5_OUT_SFT 13 +#define RT5616_GP5_OUT_LO (0x0 << 13) +#define RT5616_GP5_OUT_HI (0x1 << 13) +#define RT5616_GP5_P_MASK (0x1 << 12) +#define RT5616_GP5_P_SFT 12 +#define RT5616_GP5_P_NOR (0x0 << 12) +#define RT5616_GP5_P_INV (0x1 << 12) +#define RT5616_GP4_DR_MASK (0x1 << 11) +#define RT5616_GP4_DR_SFT 11 +#define RT5616_GP4_DR_IN (0x0 << 11) +#define RT5616_GP4_DR_OUT (0x1 << 11) +#define RT5616_GP4_OUT_MASK (0x1 << 10) +#define RT5616_GP4_OUT_SFT 10 +#define RT5616_GP4_OUT_LO (0x0 << 10) +#define RT5616_GP4_OUT_HI (0x1 << 10) +#define RT5616_GP4_P_MASK (0x1 << 9) +#define RT5616_GP4_P_SFT 9 +#define RT5616_GP4_P_NOR (0x0 << 9) +#define RT5616_GP4_P_INV (0x1 << 9) +#define RT5616_GP3_DR_MASK (0x1 << 8) +#define RT5616_GP3_DR_SFT 8 +#define RT5616_GP3_DR_IN (0x0 << 8) +#define RT5616_GP3_DR_OUT (0x1 << 8) +#define RT5616_GP3_OUT_MASK (0x1 << 7) +#define RT5616_GP3_OUT_SFT 7 +#define RT5616_GP3_OUT_LO (0x0 << 7) +#define RT5616_GP3_OUT_HI (0x1 << 7) +#define RT5616_GP3_P_MASK (0x1 << 6) +#define RT5616_GP3_P_SFT 6 +#define RT5616_GP3_P_NOR (0x0 << 6) +#define RT5616_GP3_P_INV (0x1 << 6) +#define RT5616_GP2_DR_MASK (0x1 << 5) +#define RT5616_GP2_DR_SFT 5 +#define RT5616_GP2_DR_IN (0x0 << 5) +#define RT5616_GP2_DR_OUT (0x1 << 5) +#define RT5616_GP2_OUT_MASK (0x1 << 4) +#define RT5616_GP2_OUT_SFT 4 +#define RT5616_GP2_OUT_LO (0x0 << 4) +#define RT5616_GP2_OUT_HI (0x1 << 4) +#define RT5616_GP2_P_MASK (0x1 << 3) +#define RT5616_GP2_P_SFT 3 +#define RT5616_GP2_P_NOR (0x0 << 3) +#define RT5616_GP2_P_INV (0x1 << 3) +#define RT5616_GP1_DR_MASK (0x1 << 2) +#define RT5616_GP1_DR_SFT 2 +#define RT5616_GP1_DR_IN (0x0 << 2) +#define RT5616_GP1_DR_OUT (0x1 << 2) +#define RT5616_GP1_OUT_MASK (0x1 << 1) +#define RT5616_GP1_OUT_SFT 1 +#define RT5616_GP1_OUT_LO (0x0 << 1) +#define RT5616_GP1_OUT_HI (0x1 << 1) +#define RT5616_GP1_P_MASK (0x1) +#define RT5616_GP1_P_SFT 0 +#define RT5616_GP1_P_NOR (0x0) +#define RT5616_GP1_P_INV (0x1) + +/* GPIO Control 3 (0xc2) */ +#define RT5616_GP8_DR_MASK (0x1 << 8) +#define RT5616_GP8_DR_SFT 8 +#define RT5616_GP8_DR_IN (0x0 << 8) +#define RT5616_GP8_DR_OUT (0x1 << 8) +#define RT5616_GP8_OUT_MASK (0x1 << 7) +#define RT5616_GP8_OUT_SFT 7 +#define RT5616_GP8_OUT_LO (0x0 << 7) +#define RT5616_GP8_OUT_HI (0x1 << 7) +#define RT5616_GP8_P_MASK (0x1 << 6) +#define RT5616_GP8_P_SFT 6 +#define RT5616_GP8_P_NOR (0x0 << 6) +#define RT5616_GP8_P_INV (0x1 << 6) +#define RT5616_GP7_DR_MASK (0x1 << 5) +#define RT5616_GP7_DR_SFT 5 +#define RT5616_GP7_DR_IN (0x0 << 5) +#define RT5616_GP7_DR_OUT (0x1 << 5) +#define RT5616_GP7_OUT_MASK (0x1 << 4) +#define RT5616_GP7_OUT_SFT 4 +#define RT5616_GP7_OUT_LO (0x0 << 4) +#define RT5616_GP7_OUT_HI (0x1 << 4) +#define RT5616_GP7_P_MASK (0x1 << 3) +#define RT5616_GP7_P_SFT 3 +#define RT5616_GP7_P_NOR (0x0 << 3) +#define RT5616_GP7_P_INV (0x1 << 3) +#define RT5616_GP6_DR_MASK (0x1 << 2) +#define RT5616_GP6_DR_SFT 2 +#define RT5616_GP6_DR_IN (0x0 << 2) +#define RT5616_GP6_DR_OUT (0x1 << 2) +#define RT5616_GP6_OUT_MASK (0x1 << 1) +#define RT5616_GP6_OUT_SFT 1 +#define RT5616_GP6_OUT_LO (0x0 << 1) +#define RT5616_GP6_OUT_HI (0x1 << 1) +#define RT5616_GP6_P_MASK (0x1) +#define RT5616_GP6_P_SFT 0 +#define RT5616_GP6_P_NOR (0x0) +#define RT5616_GP6_P_INV (0x1) + +/* Scramble Control (0xce) */ +#define RT5616_SCB_SWAP_MASK (0x1 << 15) +#define RT5616_SCB_SWAP_SFT 15 +#define RT5616_SCB_SWAP_DIS (0x0 << 15) +#define RT5616_SCB_SWAP_EN (0x1 << 15) +#define RT5616_SCB_MASK (0x1 << 14) +#define RT5616_SCB_SFT 14 +#define RT5616_SCB_DIS (0x0 << 14) +#define RT5616_SCB_EN (0x1 << 14) + +/* Baseback Control (0xcf) */ +#define RT5616_BB_MASK (0x1 << 15) +#define RT5616_BB_SFT 15 +#define RT5616_BB_DIS (0x0 << 15) +#define RT5616_BB_EN (0x1 << 15) +#define RT5616_BB_CT_MASK (0x7 << 12) +#define RT5616_BB_CT_SFT 12 +#define RT5616_BB_CT_A (0x0 << 12) +#define RT5616_BB_CT_B (0x1 << 12) +#define RT5616_BB_CT_C (0x2 << 12) +#define RT5616_BB_CT_D (0x3 << 12) +#define RT5616_M_BB_L_MASK (0x1 << 9) +#define RT5616_M_BB_L_SFT 9 +#define RT5616_M_BB_R_MASK (0x1 << 8) +#define RT5616_M_BB_R_SFT 8 +#define RT5616_M_BB_HPF_L_MASK (0x1 << 7) +#define RT5616_M_BB_HPF_L_SFT 7 +#define RT5616_M_BB_HPF_R_MASK (0x1 << 6) +#define RT5616_M_BB_HPF_R_SFT 6 +#define RT5616_G_BB_BST_MASK (0x3f) +#define RT5616_G_BB_BST_SFT 0 + +/* MP3 Plus Control 1 (0xd0) */ +#define RT5616_M_MP3_L_MASK (0x1 << 15) +#define RT5616_M_MP3_L_SFT 15 +#define RT5616_M_MP3_R_MASK (0x1 << 14) +#define RT5616_M_MP3_R_SFT 14 +#define RT5616_M_MP3_MASK (0x1 << 13) +#define RT5616_M_MP3_SFT 13 +#define RT5616_M_MP3_DIS (0x0 << 13) +#define RT5616_M_MP3_EN (0x1 << 13) +#define RT5616_EG_MP3_MASK (0x1f << 8) +#define RT5616_EG_MP3_SFT 8 +#define RT5616_MP3_HLP_MASK (0x1 << 7) +#define RT5616_MP3_HLP_SFT 7 +#define RT5616_MP3_HLP_DIS (0x0 << 7) +#define RT5616_MP3_HLP_EN (0x1 << 7) +#define RT5616_M_MP3_ORG_L_MASK (0x1 << 6) +#define RT5616_M_MP3_ORG_L_SFT 6 +#define RT5616_M_MP3_ORG_R_MASK (0x1 << 5) +#define RT5616_M_MP3_ORG_R_SFT 5 + +/* MP3 Plus Control 2 (0xd1) */ +#define RT5616_MP3_WT_MASK (0x1 << 13) +#define RT5616_MP3_WT_SFT 13 +#define RT5616_MP3_WT_1_4 (0x0 << 13) +#define RT5616_MP3_WT_1_2 (0x1 << 13) +#define RT5616_OG_MP3_MASK (0x1f << 8) +#define RT5616_OG_MP3_SFT 8 +#define RT5616_HG_MP3_MASK (0x3f) +#define RT5616_HG_MP3_SFT 0 + +/* 3D HP Control 1 (0xd2) */ +#define RT5616_3D_CF_MASK (0x1 << 15) +#define RT5616_3D_CF_SFT 15 +#define RT5616_3D_CF_DIS (0x0 << 15) +#define RT5616_3D_CF_EN (0x1 << 15) +#define RT5616_3D_HP_MASK (0x1 << 14) +#define RT5616_3D_HP_SFT 14 +#define RT5616_3D_HP_DIS (0x0 << 14) +#define RT5616_3D_HP_EN (0x1 << 14) +#define RT5616_3D_BT_MASK (0x1 << 13) +#define RT5616_3D_BT_SFT 13 +#define RT5616_3D_BT_DIS (0x0 << 13) +#define RT5616_3D_BT_EN (0x1 << 13) +#define RT5616_3D_1F_MIX_MASK (0x3 << 11) +#define RT5616_3D_1F_MIX_SFT 11 +#define RT5616_3D_HP_M_MASK (0x1 << 10) +#define RT5616_3D_HP_M_SFT 10 +#define RT5616_3D_HP_M_SUR (0x0 << 10) +#define RT5616_3D_HP_M_FRO (0x1 << 10) +#define RT5616_M_3D_HRTF_MASK (0x1 << 9) +#define RT5616_M_3D_HRTF_SFT 9 +#define RT5616_M_3D_D2H_MASK (0x1 << 8) +#define RT5616_M_3D_D2H_SFT 8 +#define RT5616_M_3D_D2R_MASK (0x1 << 7) +#define RT5616_M_3D_D2R_SFT 7 +#define RT5616_M_3D_REVB_MASK (0x1 << 6) +#define RT5616_M_3D_REVB_SFT 6 + +/* Adjustable high pass filter control 1 (0xd3) */ +#define RT5616_2ND_HPF_MASK (0x1 << 15) +#define RT5616_2ND_HPF_SFT 15 +#define RT5616_2ND_HPF_DIS (0x0 << 15) +#define RT5616_2ND_HPF_EN (0x1 << 15) +#define RT5616_HPF_CF_L_MASK (0x7 << 12) +#define RT5616_HPF_CF_L_SFT 12 +#define RT5616_HPF_CF_R_MASK (0x7 << 8) +#define RT5616_HPF_CF_R_SFT 8 +#define RT5616_ZD_T_MASK (0x3 << 6) +#define RT5616_ZD_T_SFT 6 +#define RT5616_ZD_F_MASK (0x3 << 4) +#define RT5616_ZD_F_SFT 4 +#define RT5616_ZD_F_IM (0x0 << 4) +#define RT5616_ZD_F_ZC_IM (0x1 << 4) +#define RT5616_ZD_F_ZC_IOD (0x2 << 4) +#define RT5616_ZD_F_UN (0x3 << 4) + +/* Adjustable high pass filter control 2 (0xd4) */ +#define RT5616_HPF_CF_L_NUM_MASK (0x3f << 8) +#define RT5616_HPF_CF_L_NUM_SFT 8 +#define RT5616_HPF_CF_R_NUM_MASK (0x3f) +#define RT5616_HPF_CF_R_NUM_SFT 0 + +/* HP calibration control and Amp detection (0xd6) */ +#define RT5616_SI_DAC_MASK (0x1 << 11) +#define RT5616_SI_DAC_SFT 11 +#define RT5616_SI_DAC_AUTO (0x0 << 11) +#define RT5616_SI_DAC_TEST (0x1 << 11) +#define RT5616_DC_CAL_M_MASK (0x1 << 10) +#define RT5616_DC_CAL_M_SFT 10 +#define RT5616_DC_CAL_M_NOR (0x0 << 10) +#define RT5616_DC_CAL_M_CAL (0x1 << 10) +#define RT5616_DC_CAL_MASK (0x1 << 9) +#define RT5616_DC_CAL_SFT 9 +#define RT5616_DC_CAL_DIS (0x0 << 9) +#define RT5616_DC_CAL_EN (0x1 << 9) +#define RT5616_HPD_RCV_MASK (0x7 << 6) +#define RT5616_HPD_RCV_SFT 6 +#define RT5616_HPD_PS_MASK (0x1 << 5) +#define RT5616_HPD_PS_SFT 5 +#define RT5616_HPD_PS_DIS (0x0 << 5) +#define RT5616_HPD_PS_EN (0x1 << 5) +#define RT5616_CAL_M_MASK (0x1 << 4) +#define RT5616_CAL_M_SFT 4 +#define RT5616_CAL_M_DEP (0x0 << 4) +#define RT5616_CAL_M_CAL (0x1 << 4) +#define RT5616_CAL_MASK (0x1 << 3) +#define RT5616_CAL_SFT 3 +#define RT5616_CAL_DIS (0x0 << 3) +#define RT5616_CAL_EN (0x1 << 3) +#define RT5616_CAL_TEST_MASK (0x1 << 2) +#define RT5616_CAL_TEST_SFT 2 +#define RT5616_CAL_TEST_DIS (0x0 << 2) +#define RT5616_CAL_TEST_EN (0x1 << 2) +#define RT5616_CAL_P_MASK (0x3) +#define RT5616_CAL_P_SFT 0 +#define RT5616_CAL_P_NONE (0x0) +#define RT5616_CAL_P_CAL (0x1) +#define RT5616_CAL_P_DAC_CAL (0x2) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5616_SV_MASK (0x1 << 15) +#define RT5616_SV_SFT 15 +#define RT5616_SV_DIS (0x0 << 15) +#define RT5616_SV_EN (0x1 << 15) +#define RT5616_OUT_SV_MASK (0x1 << 13) +#define RT5616_OUT_SV_SFT 13 +#define RT5616_OUT_SV_DIS (0x0 << 13) +#define RT5616_OUT_SV_EN (0x1 << 13) +#define RT5616_HP_SV_MASK (0x1 << 12) +#define RT5616_HP_SV_SFT 12 +#define RT5616_HP_SV_DIS (0x0 << 12) +#define RT5616_HP_SV_EN (0x1 << 12) +#define RT5616_ZCD_DIG_MASK (0x1 << 11) +#define RT5616_ZCD_DIG_SFT 11 +#define RT5616_ZCD_DIG_DIS (0x0 << 11) +#define RT5616_ZCD_DIG_EN (0x1 << 11) +#define RT5616_ZCD_MASK (0x1 << 10) +#define RT5616_ZCD_SFT 10 +#define RT5616_ZCD_PD (0x0 << 10) +#define RT5616_ZCD_PU (0x1 << 10) +#define RT5616_M_ZCD_MASK (0x3f << 4) +#define RT5616_M_ZCD_SFT 4 +#define RT5616_M_ZCD_OM_L (0x1 << 7) +#define RT5616_M_ZCD_OM_R (0x1 << 6) +#define RT5616_M_ZCD_RM_L (0x1 << 5) +#define RT5616_M_ZCD_RM_R (0x1 << 4) +#define RT5616_SV_DLY_MASK (0xf) +#define RT5616_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5616_ZCD_HP_MASK (0x1 << 15) +#define RT5616_ZCD_HP_SFT 15 +#define RT5616_ZCD_HP_DIS (0x0 << 15) +#define RT5616_ZCD_HP_EN (0x1 << 15) + +/* Digital Misc Control (0xfa) */ +#define RT5616_I2S2_MS_SP_MASK (0x1 << 8) +#define RT5616_I2S2_MS_SP_SEL 8 +#define RT5616_I2S2_MS_SP_64 (0x0 << 8) +#define RT5616_I2S2_MS_SP_50 (0x1 << 8) +#define RT5616_CLK_DET_EN (0x1 << 3) +#define RT5616_CLK_DET_EN_SFT 3 +#define RT5616_AMP_DET_EN (0x1 << 1) +#define RT5616_AMP_DET_EN_SFT 1 +#define RT5616_D_GATE_EN (0x1) +#define RT5616_D_GATE_EN_SFT 0 + +/* Codec Private Register definition */ +/* 3D Speaker Control (0x63) */ +#define RT5616_3D_SPK_MASK (0x1 << 15) +#define RT5616_3D_SPK_SFT 15 +#define RT5616_3D_SPK_DIS (0x0 << 15) +#define RT5616_3D_SPK_EN (0x1 << 15) +#define RT5616_3D_SPK_M_MASK (0x3 << 13) +#define RT5616_3D_SPK_M_SFT 13 +#define RT5616_3D_SPK_CG_MASK (0x1f << 8) +#define RT5616_3D_SPK_CG_SFT 8 +#define RT5616_3D_SPK_SG_MASK (0x1f) +#define RT5616_3D_SPK_SG_SFT 0 + +/* Wind Noise Detection Control 1 (0x6c) */ +#define RT5616_WND_MASK (0x1 << 15) +#define RT5616_WND_SFT 15 +#define RT5616_WND_DIS (0x0 << 15) +#define RT5616_WND_EN (0x1 << 15) + +/* Wind Noise Detection Control 2 (0x6d) */ +#define RT5616_WND_FC_NW_MASK (0x3f << 10) +#define RT5616_WND_FC_NW_SFT 10 +#define RT5616_WND_FC_WK_MASK (0x3f << 4) +#define RT5616_WND_FC_WK_SFT 4 + +/* Wind Noise Detection Control 3 (0x6e) */ +#define RT5616_HPF_FC_MASK (0x3f << 6) +#define RT5616_HPF_FC_SFT 6 +#define RT5616_WND_FC_ST_MASK (0x3f) +#define RT5616_WND_FC_ST_SFT 0 + +/* Wind Noise Detection Control 4 (0x6f) */ +#define RT5616_WND_TH_LO_MASK (0x3ff) +#define RT5616_WND_TH_LO_SFT 0 + +/* Wind Noise Detection Control 5 (0x70) */ +#define RT5616_WND_TH_HI_MASK (0x3ff) +#define RT5616_WND_TH_HI_SFT 0 + +/* Wind Noise Detection Control 8 (0x73) */ +#define RT5616_WND_WIND_MASK (0x1 << 13) /* Read-Only */ +#define RT5616_WND_WIND_SFT 13 +#define RT5616_WND_STRONG_MASK (0x1 << 12) /* Read-Only */ +#define RT5616_WND_STRONG_SFT 12 +enum { + RT5616_NO_WIND, + RT5616_BREEZE, + RT5616_STORM, +}; + +/* Dipole Speaker Interface (0x75) */ +#define RT5616_DP_ATT_MASK (0x3 << 14) +#define RT5616_DP_ATT_SFT 14 +#define RT5616_DP_SPK_MASK (0x1 << 10) +#define RT5616_DP_SPK_SFT 10 +#define RT5616_DP_SPK_DIS (0x0 << 10) +#define RT5616_DP_SPK_EN (0x1 << 10) + +/* EQ Pre Volume Control (0xb3) */ +#define RT5616_EQ_PRE_VOL_MASK (0xffff) +#define RT5616_EQ_PRE_VOL_SFT 0 + +/* EQ Post Volume Control (0xb4) */ +#define RT5616_EQ_PST_VOL_MASK (0xffff) +#define RT5616_EQ_PST_VOL_SFT 0 + +/* System Clock Source */ +enum { + RT5616_SCLK_S_MCLK, + RT5616_SCLK_S_PLL1, +}; + +/* PLL1 Source */ +enum { + RT5616_PLL1_S_MCLK, + RT5616_PLL1_S_BCLK1, + RT5616_PLL1_S_BCLK2, +}; + +enum { + RT5616_AIF1, + RT5616_AIFS, +}; + +#endif /* __RT5616_H__ */ -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH] ALSA: oss: consolidate kmalloc/memset 0 call to kzalloc
by Nicholas Mc Guire 18 Dec '15

18 Dec '15

This is an API consolidation only. The use of kmalloc + memset to 0 is equivalent to kzalloc. Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- Found by coccinelle script (relaxed version of scripts/coccinelle/api/alloc/kzalloc-simple.cocci) Patch was compile tested with: x86_64_defconfig CONFIG_SND_PCM_OSS=y Patch is against linux-next (localversion-next is -next-20150518) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index e557dbe..8cdd06f 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -850,7 +850,7 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) if (mutex_lock_interruptible(&runtime->oss.params_lock)) return -EINTR; - sw_params = kmalloc(sizeof(*sw_params), GFP_KERNEL); + sw_params = kmalloc(sizeof(*sw_params), GFP_KERNEL); params = kmalloc(sizeof(*params), GFP_KERNEL); sparams = kmalloc(sizeof(*sparams), GFP_KERNEL); if (!sw_params || !params || !sparams) { @@ -988,7 +988,6 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) goto failure; } - memset(sw_params, 0, sizeof(*sw_params)); if (runtime->oss.trigger) { sw_params->start_threshold = 1; } else { -- 1.7.10.4

2 1

[alsa-devel] [PATCH v2 1/4] ASoC: sun4i-codec: Rename codec dapm widgets and routes
by Hans de Goede 18 Dec '15

18 Dec '15

Rename the codec dapm widgets and routes with a _codec prefix. This is a preparation patch for adding card dapm widgets and routes. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Changes in v2: -New patch in v2 of this patchset --- sound/soc/sunxi/sun4i-codec.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 7a3fe1d..519ccb3 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -532,7 +532,7 @@ static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = { SUN4I_CODEC_DAC_ACTL_MIXPAS, 1, 0), }; -static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { +static const struct snd_soc_dapm_widget sun4i_codec_codec_dapm_widgets[] = { /* Digital parts of the ADCs */ SND_SOC_DAPM_SUPPLY("ADC", SUN4I_CODEC_ADC_FIFOC, SUN4I_CODEC_ADC_FIFOC_EN_AD, 0, @@ -589,7 +589,7 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HP Left"), }; -static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { +static const struct snd_soc_dapm_route sun4i_codec_codec_dapm_routes[] = { /* Left ADC / DAC Routes */ { "Left ADC", NULL, "ADC" }, { "Left DAC", NULL, "DAC" }, @@ -628,10 +628,10 @@ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { static struct snd_soc_codec_driver sun4i_codec_codec = { .controls = sun4i_codec_widgets, .num_controls = ARRAY_SIZE(sun4i_codec_widgets), - .dapm_widgets = sun4i_codec_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(sun4i_codec_dapm_widgets), - .dapm_routes = sun4i_codec_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(sun4i_codec_dapm_routes), + .dapm_widgets = sun4i_codec_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sun4i_codec_codec_dapm_widgets), + .dapm_routes = sun4i_codec_codec_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sun4i_codec_codec_dapm_routes), }; static const struct snd_soc_component_driver sun4i_codec_component = { -- 2.5.0

4 12

[alsa-devel] [PATCH] ALSA: consolidate allocation API
by Nicholas Mc Guire 18 Dec '15

18 Dec '15

This is an API consolidation only. The use of kmalloc + memset to 0 is equivalent to kzalloc. Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- Found by coccinelle script (relaxed version of scripts/coccinelle/api/alloc/kzalloc-simple.cocci) Patch was compile tested with: x86_64_defconfig CONFIG_SND_SEQUENCER=y, CONFIG_SND_SEQUENCER_OSS=y Patch is against linux-next (localversion-next is -next-20150518) sound/core/seq/oss/seq_oss_init.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index b1221b2..9babc9e 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -79,7 +79,7 @@ snd_seq_oss_create_client(void) struct snd_seq_port_info *port; struct snd_seq_port_callback port_callback; - port = kmalloc(sizeof(*port), GFP_KERNEL); + port = kzalloc(sizeof(*port), GFP_KERNEL); if (!port) { rc = -ENOMEM; goto __error; @@ -94,7 +94,6 @@ snd_seq_oss_create_client(void) system_client = rc; /* create annoucement receiver port */ - memset(port, 0, sizeof(*port)); strcpy(port->name, "Receiver"); port->addr.client = system_client; port->capability = SNDRV_SEQ_PORT_CAP_WRITE; /* receive only */ -- 1.7.10.4

1 0

[alsa-devel] [PATCH v3] ASoC: fsl_sai: fix no frame clk in master mode
by Zidan Wang 18 Dec '15

18 Dec '15

After several open/close sai test with ctrl+c, there will be I/O error. The SAI can't work anymore, can't recover. There will be no frame clock. With adding the software reset in trigger stop, the issue can be fixed. This is a hardware bug/errata and reset is the only option. According to the reference manual, the software reset doesn't reset any control register but only internal hardware logics such as bit clock generator, status flags, and FIFO pointers. (Our purpose is just to reset the clock generator while the software reset is the only way to do that.) Since slave mode doesn't use the clock generator, only apply the reset procedure to the master mode. For asynchronous mode, TX will not be reset when RX is still running. In this case, i can't reproduce this issue. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- v1->v2: 1) refine the commit log and comments 2) Since slave mode doesn't use the clock generator, only apply the reset procedure to the master mode. v2->v3: add a simple comment in code to describe the hardware bug. sound/soc/fsl/fsl_sai.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 8832901..fef264d 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -564,6 +564,24 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); regmap_update_bits(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); + + /* + * For sai master mode, after several open/close sai, + * there will be no frame clock, and can't recover + * anymore. Add software reset to fix this issue. + * This is a hardware bug, and will be fix in the + * next sai version. + */ + if (!sai->is_slave_mode) { + /* Software Reset for both Tx and Rx */ + regmap_write(sai->regmap, + FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, + FSL_SAI_RCSR, FSL_SAI_CSR_SR); + /* Clear SR bit to finish the reset */ + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + } } break; default: -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Mimic legacy behaviour of reading controls when DSP is on
by Charles Keepax 18 Dec '15

18 Dec '15

Older firmwares don't specify access flags for the controls, unfortunately the usage of some of these firmware relies on being able to read back values from the DSP. The current control code will only do this for volatile controls. This patch will read the control from the hardware if no flags are specified and the control is currently enabled, which should cover these legacy use-cases. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b083642..d1e0826 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -666,6 +666,9 @@ static int wm_coeff_get(struct snd_kcontrol *kctl, else ret = -EPERM; } else { + if (!ctl->flags && ctl->enabled) + ret = wm_coeff_read_control(ctl, ctl->cache, ctl->len); + memcpy(p, ctl->cache, ctl->len); } -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: fix VAG power up timing
by Jean-Michel Hautbois 18 Dec '15

18 Dec '15

When power up, a "pop" is heard on line-in and mic-in. An analysis of the PCM shows it lasts ~400ms and looks like a filter response. VAG power up should be delayed by 400ms as VAG power down is. Signed-off-by: Jean-Michel Hautbois <jean-michel.hautbois(a)veo-labs.com> --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index f540f82..08b4046 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -189,6 +189,7 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); + msleep(400); break; case SND_SOC_DAPM_PRE_PMD: -- 2.6.4

2 1

[alsa-devel] [PATCH] ASoC: rsrc-card: enable to use tdm_slot on DT
by Kuninori Morimoto 18 Dec '15

18 Dec '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Renesas sound driver will use tdm slot on TDM Multi Mode support. This patch enables tdm slot on rsrc card driver on DT. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsrc-card.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/soc/sh/rcar/rsrc-card.c b/sound/soc/sh/rcar/rsrc-card.c index a3ec13f..3c308e2 100644 --- a/sound/soc/sh/rcar/rsrc-card.c +++ b/sound/soc/sh/rcar/rsrc-card.c @@ -50,6 +50,10 @@ MODULE_DEVICE_TABLE(of, rsrc_card_of_match); struct rsrc_card_dai { unsigned int fmt; unsigned int sysclk; + unsigned int tx_slot_mask; + unsigned int rx_slot_mask; + int slots; + int slot_width; struct clk *clk; char dai_name[DAI_NAME_NUM]; }; @@ -126,6 +130,18 @@ static int rsrc_card_dai_init(struct snd_soc_pcm_runtime *rtd) } } + if (dai_props->slots) { + ret = snd_soc_dai_set_tdm_slot(dai, + dai_props->tx_slot_mask, + dai_props->rx_slot_mask, + dai_props->slots, + dai_props->slot_width); + if (ret && ret != -ENOTSUPP) { + dev_err(dai->dev, "set_tdm_slot error\n"); + goto err; + } + } + ret = 0; err: @@ -198,6 +214,15 @@ static int rsrc_card_parse_links(struct device_node *np, if (ret) return ret; + /* Parse TDM slot */ + ret = snd_soc_of_parse_tdm_slot(np, + &dai_props->tx_slot_mask, + &dai_props->rx_slot_mask, + &dai_props->slots, + &dai_props->slot_width); + if (ret) + return ret; + if (is_fe) { /* BE is dummy */ dai_link->codec_of_node = NULL; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsrc-card: Remove support for setting differing DAI formats
by Kuninori Morimoto 18 Dec '15

18 Dec '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> 1efb53a220 ("ASoC: simple-card: Remove support for setting differing DAI formats") removed set_fmt support from simple-card. rsrc-card follows same style, because it is based on simple-card. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsrc-card.c | 16 ++++------------ 1 file changed, 4 insertions(+), 12 deletions(-) diff --git a/sound/soc/sh/rcar/rsrc-card.c b/sound/soc/sh/rcar/rsrc-card.c index 3c308e2..9f522ba 100644 --- a/sound/soc/sh/rcar/rsrc-card.c +++ b/sound/soc/sh/rcar/rsrc-card.c @@ -48,7 +48,6 @@ MODULE_DEVICE_TABLE(of, rsrc_card_of_match); #define DAI_NAME_NUM 32 struct rsrc_card_dai { - unsigned int fmt; unsigned int sysclk; unsigned int tx_slot_mask; unsigned int rx_slot_mask; @@ -114,14 +113,6 @@ static int rsrc_card_dai_init(struct snd_soc_pcm_runtime *rtd) rtd->cpu_dai : rtd->codec_dai; - if (dai_props->fmt) { - ret = snd_soc_dai_set_fmt(dai, dai_props->fmt); - if (ret && ret != -ENOTSUPP) { - dev_err(dai->dev, "set_fmt error\n"); - goto err; - } - } - if (dai_props->sysclk) { ret = snd_soc_dai_set_sysclk(dai, 0, dai_props->sysclk, 0); if (ret && ret != -ENOTSUPP) { @@ -168,7 +159,7 @@ static int rsrc_card_parse_daifmt(struct device_node *node, struct rsrc_card_priv *priv, int idx, bool is_fe) { - struct rsrc_card_dai *dai_props = rsrc_priv_to_props(priv, idx); + struct snd_soc_dai_link *dai_link = rsrc_priv_to_link(priv, idx); struct device_node *bitclkmaster = NULL; struct device_node *framemaster = NULL; struct device_node *codec = is_fe ? NULL : np; @@ -188,7 +179,7 @@ static int rsrc_card_parse_daifmt(struct device_node *node, daifmt |= (codec == framemaster) ? SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS; - dai_props->fmt = daifmt; + dai_link->dai_fmt = daifmt; of_node_put(bitclkmaster); of_node_put(framemaster); @@ -340,6 +331,7 @@ static int rsrc_card_dai_link_of(struct device_node *node, int idx) { struct device *dev = rsrc_priv_to_dev(priv); + struct snd_soc_dai_link *dai_link = rsrc_priv_to_link(priv, idx); struct rsrc_card_dai *dai_props = rsrc_priv_to_props(priv, idx); bool is_fe = false; int ret; @@ -361,7 +353,7 @@ static int rsrc_card_dai_link_of(struct device_node *node, dev_dbg(dev, "\t%s / %04x / %d\n", dai_props->dai_name, - dai_props->fmt, + dai_link->dai_fmt, dai_props->sysclk); return ret; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsrc-card: tidyup dai format for DPCM
by Kuninori Morimoto 18 Dec '15

18 Dec '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsrc-card is DPCM supported version of simple-card. Thus it has similar DT format. OTOH, snd_soc_dai_link requests cpu/codec, but one of them will be snd-soc-dummy in DPCM case, and DPCM requests frontend/backend dai_link. This means it might have multi backend/codec. And, SND_SOC_DAIFMT_xxx is based on "codec". Because of these difference, current rsrc card can't detect correct dai_fmt. This patch detect correct dai fmt from 1st "codec". Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsrc-card.c | 82 +++++++++++++++++++++++++++++-------------- 1 file changed, 55 insertions(+), 27 deletions(-) diff --git a/sound/soc/sh/rcar/rsrc-card.c b/sound/soc/sh/rcar/rsrc-card.c index 9f522ba..5fe0b51 100644 --- a/sound/soc/sh/rcar/rsrc-card.c +++ b/sound/soc/sh/rcar/rsrc-card.c @@ -155,14 +155,13 @@ static int rsrc_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, } static int rsrc_card_parse_daifmt(struct device_node *node, - struct device_node *np, + struct device_node *codec, struct rsrc_card_priv *priv, - int idx, bool is_fe) + struct snd_soc_dai_link *dai_link, + unsigned int *retfmt) { - struct snd_soc_dai_link *dai_link = rsrc_priv_to_link(priv, idx); struct device_node *bitclkmaster = NULL; struct device_node *framemaster = NULL; - struct device_node *codec = is_fe ? NULL : np; unsigned int daifmt; daifmt = snd_soc_of_parse_daifmt(node, NULL, @@ -179,11 +178,11 @@ static int rsrc_card_parse_daifmt(struct device_node *node, daifmt |= (codec == framemaster) ? SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS; - dai_link->dai_fmt = daifmt; - of_node_put(bitclkmaster); of_node_put(framemaster); + *retfmt = daifmt; + return 0; } @@ -325,24 +324,16 @@ static int rsrc_card_parse_clk(struct device_node *np, return 0; } -static int rsrc_card_dai_link_of(struct device_node *node, - struct device_node *np, - struct rsrc_card_priv *priv, - int idx) +static int rsrc_card_dai_sub_link_of(struct device_node *node, + struct device_node *np, + struct rsrc_card_priv *priv, + int idx, bool is_fe) { struct device *dev = rsrc_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = rsrc_priv_to_link(priv, idx); struct rsrc_card_dai *dai_props = rsrc_priv_to_props(priv, idx); - bool is_fe = false; int ret; - if (0 == strcmp(np->name, "cpu")) - is_fe = true; - - ret = rsrc_card_parse_daifmt(node, np, priv, idx, is_fe); - if (ret < 0) - return ret; - ret = rsrc_card_parse_links(np, priv, idx, is_fe); if (ret < 0) return ret; @@ -359,6 +350,48 @@ static int rsrc_card_dai_link_of(struct device_node *node, return ret; } +static int rsrc_card_dai_link_of(struct device_node *node, + struct rsrc_card_priv *priv) +{ + struct snd_soc_dai_link *dai_link; + struct device_node *np; + unsigned int daifmt = 0; + int ret, i; + bool is_fe; + + /* find 1st codec */ + i = 0; + for_each_child_of_node(node, np) { + dai_link = rsrc_priv_to_link(priv, i); + + if (strcmp(np->name, "codec") == 0) { + ret = rsrc_card_parse_daifmt(node, np, priv, + dai_link, &daifmt); + if (ret < 0) + return ret; + break; + } + i++; + } + + i = 0; + for_each_child_of_node(node, np) { + dai_link = rsrc_priv_to_link(priv, i); + dai_link->dai_fmt = daifmt; + + is_fe = false; + if (strcmp(np->name, "cpu") == 0) + is_fe = true; + + ret = rsrc_card_dai_sub_link_of(node, np, priv, i, is_fe); + if (ret < 0) + return ret; + i++; + } + + return 0; +} + static int rsrc_card_parse_of(struct device_node *node, struct rsrc_card_priv *priv, struct device *dev) @@ -366,9 +399,8 @@ static int rsrc_card_parse_of(struct device_node *node, const struct rsrc_card_of_data *of_data = rsrc_dev_to_of_data(dev); struct rsrc_card_dai *props; struct snd_soc_dai_link *links; - struct device_node *np; int ret; - int i, num; + int num; if (!node) return -EINVAL; @@ -409,13 +441,9 @@ static int rsrc_card_parse_of(struct device_node *node, priv->snd_card.name ? priv->snd_card.name : "", priv->convert_rate); - i = 0; - for_each_child_of_node(node, np) { - ret = rsrc_card_dai_link_of(node, np, priv, i); - if (ret < 0) - return ret; - i++; - } + ret = rsrc_card_dai_link_of(node, priv); + if (ret < 0) + return ret; if (!priv->snd_card.name) priv->snd_card.name = priv->snd_card.dai_link->name; -- 1.9.1

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: atmel-pdmic: add the Pulse Density Modulation Interface Controller
by Songjun Wu 18 Dec '15

18 Dec '15

Add driver for the Pulse Density Modulation Interface Controller. It comes with digitallly controlled gain, a High-Pass and a SINCC filter. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> --- Changes in v2: - Remove the function 'atmel_pdmic_codec_get_remap', since the regmap is initialized in the core. - Replace 'if' with 'switch' in function 'atmel_pdmic_codec_dai_hw_params'. - Fix the coding style issue in function 'atmel_pdmic_dt_init', 'atmel_pdmic_interrupt','pdmic_put_mic_volsw', 'atmel_pdmic_codec_probe', 'pdmic_get_mic_volsw' and 'atmel_pdmic_codec_dai_hw_params'. sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-pdmic.c | 738 +++++++++++++++++++++++++++++++++++++++++ sound/soc/atmel/atmel-pdmic.h | 80 +++++ 4 files changed, 829 insertions(+) create mode 100644 sound/soc/atmel/atmel-pdmic.c create mode 100644 sound/soc/atmel/atmel-pdmic.h diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 2d30464..06e099e 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -68,4 +68,13 @@ config SND_ATMEL_SOC_CLASSD help Say Y if you want to add support for Atmel ASoC driver for boards using CLASSD. + +config SND_ATMEL_SOC_PDMIC + tristate "Atmel ASoC driver for boards using PDMIC" + depends on OF && (ARCH_AT91 || COMPILE_TEST) + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + Say Y if you want to add support for Atmel ASoC driver for boards using + PDMIC. endif diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index f6f7db4..a2b127b 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -12,8 +12,10 @@ snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o snd-atmel-soc-wm8904-objs := atmel_wm8904.o snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o snd-atmel-soc-classd-objs := atmel-classd.o +snd-atmel-soc-pdmic-objs := atmel-pdmic.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_CLASSD) += snd-atmel-soc-classd.o +obj-$(CONFIG_SND_ATMEL_SOC_PDMIC) += snd-atmel-soc-pdmic.o diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c new file mode 100644 index 0000000..aee4787 --- /dev/null +++ b/sound/soc/atmel/atmel-pdmic.c @@ -0,0 +1,738 @@ +/* Atmel PDMIC driver + * + * Copyright (C) 2015 Atmel + * + * Author: Songjun Wu <songjun.wu(a)atmel.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 or later + * as published by the Free Software Foundation. + */ + +#include <linux/of.h> +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include "atmel-pdmic.h" + +struct atmel_pdmic_pdata { + u32 mic_min_freq; + u32 mic_max_freq; + s32 mic_offset; + const char *card_name; +}; + +struct atmel_pdmic { + dma_addr_t phy_base; + struct regmap *regmap; + struct clk *pclk; + struct clk *gclk; + int irq; + struct snd_pcm_substream *substream; + const struct atmel_pdmic_pdata *pdata; +}; + +static const struct of_device_id atmel_pdmic_of_match[] = { + { + .compatible = "atmel,sama5d2-pdmic", + }, { + /* sentinel */ + } +}; +MODULE_DEVICE_TABLE(of, atmel_pdmic_of_match); + +#define PDMIC_OFFSET_MAX_VAL S16_MAX +#define PDMIC_OFFSET_MIN_VAL S16_MIN + +static struct atmel_pdmic_pdata *atmel_pdmic_dt_init(struct device *dev) +{ + struct device_node *np = dev->of_node; + struct atmel_pdmic_pdata *pdata; + + if (!np) { + dev_err(dev, "device node not found\n"); + return ERR_PTR(-EINVAL); + } + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return ERR_PTR(-ENOMEM); + + if (of_property_read_string(np, "atmel,model", &pdata->card_name)) + pdata->card_name = "PDMIC"; + + if (of_property_read_u32(np, "atmel,mic-min-freq", + &pdata->mic_min_freq)) { + dev_err(dev, "failed to get mic-min-freq\n"); + return ERR_PTR(-EINVAL); + } + + if (of_property_read_u32(np, "atmel,mic-max-freq", + &pdata->mic_max_freq)) { + dev_err(dev, "failed to get mic-max-freq\n"); + return ERR_PTR(-EINVAL); + } + + if (pdata->mic_max_freq < pdata->mic_min_freq) { + dev_err(dev, + "mic-max-freq should not less than mic-min-freq\n"); + return ERR_PTR(-EINVAL); + } + + if (of_property_read_s32(np, "atmel,mic-offset", &pdata->mic_offset)) + pdata->mic_offset = 0; + + if (pdata->mic_offset > PDMIC_OFFSET_MAX_VAL) { + dev_warn(dev, + "mic-offset value %d is larger than the max value %d, the max value is specified\n", + pdata->mic_offset, PDMIC_OFFSET_MAX_VAL); + pdata->mic_offset = PDMIC_OFFSET_MAX_VAL; + } else if (pdata->mic_offset < PDMIC_OFFSET_MIN_VAL) { + dev_warn(dev, + "mic-offset value %d is less than the min value %d, the min value is specified\n", + pdata->mic_offset, PDMIC_OFFSET_MIN_VAL); + pdata->mic_offset = PDMIC_OFFSET_MIN_VAL; + } + + return pdata; +} + +/* cpu dai component */ +static int atmel_pdmic_cpu_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + int ret; + + ret = clk_prepare_enable(dd->gclk); + if (ret) + return ret; + + ret = clk_prepare_enable(dd->pclk); + if (ret) + return ret; + + /* Clear all bits in the Control Register(PDMIC_CR) */ + regmap_write(dd->regmap, PDMIC_CR, 0); + + dd->substream = substream; + + /* Enable the overrun error interrupt */ + regmap_write(dd->regmap, PDMIC_IER, PDMIC_IER_OVRE); + + return 0; +} + +static void atmel_pdmic_cpu_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + + /* Disable the overrun error interrupt */ + regmap_write(dd->regmap, PDMIC_IDR, PDMIC_IDR_OVRE); + + clk_disable_unprepare(dd->gclk); + clk_disable_unprepare(dd->pclk); +} + +static int atmel_pdmic_cpu_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + u32 val; + + /* Clean the PDMIC Converted Data Register */ + return regmap_read(dd->regmap, PDMIC_CDR, &val); +} + +static const struct snd_soc_dai_ops atmel_pdmic_cpu_dai_ops = { + .startup = atmel_pdmic_cpu_dai_startup, + .shutdown = atmel_pdmic_cpu_dai_shutdown, + .prepare = atmel_pdmic_cpu_dai_prepare, +}; + +#define ATMEL_PDMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver atmel_pdmic_cpu_dai = { + .capture = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = ATMEL_PDMIC_FORMATS,}, + .ops = &atmel_pdmic_cpu_dai_ops, +}; + +static const struct snd_soc_component_driver atmel_pdmic_cpu_dai_component = { + .name = "atmel-pdmic", +}; + +/* platform */ +#define ATMEL_PDMIC_MAX_BUF_SIZE (64 * 1024) +#define ATMEL_PDMIC_PREALLOC_BUF_SIZE ATMEL_PDMIC_MAX_BUF_SIZE + +static const struct snd_pcm_hardware atmel_pdmic_hw = { + .info = SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_INTERLEAVED + | SNDRV_PCM_INFO_RESUME + | SNDRV_PCM_INFO_PAUSE, + .formats = ATMEL_PDMIC_FORMATS, + .buffer_bytes_max = ATMEL_PDMIC_MAX_BUF_SIZE, + .period_bytes_min = 256, + .period_bytes_max = 32 * 1024, + .periods_min = 2, + .periods_max = 256, +}; + +static int +atmel_pdmic_platform_configure_dma(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct dma_slave_config *slave_config) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + int ret; + + ret = snd_hwparams_to_dma_slave_config(substream, params, + slave_config); + if (ret) { + dev_err(rtd->platform->dev, + "hw params to dma slave configure failed\n"); + return ret; + } + + slave_config->src_addr = dd->phy_base + PDMIC_CDR; + slave_config->src_maxburst = 1; + slave_config->dst_maxburst = 1; + + return 0; +} + +static const struct snd_dmaengine_pcm_config +atmel_pdmic_dmaengine_pcm_config = { + .prepare_slave_config = atmel_pdmic_platform_configure_dma, + .pcm_hardware = &atmel_pdmic_hw, + .prealloc_buffer_size = ATMEL_PDMIC_PREALLOC_BUF_SIZE, +}; + +/* codec */ +/* Mic Gain = dgain * 2^(-scale) */ +struct mic_gain { + unsigned int dgain; + unsigned int scale; +}; + +/* range from -90 dB to 90 dB */ +static const struct mic_gain mic_gain_table[] = { +{ 1, 15}, { 1, 14}, /* -90, -84 dB */ +{ 3, 15}, { 1, 13}, { 3, 14}, { 1, 12}, /* -81, -78, -75, -72 dB */ +{ 5, 14}, { 13, 15}, /* -70, -68 dB */ +{ 9, 14}, { 21, 15}, { 23, 15}, { 13, 14}, /* -65 ~ -62 dB */ +{ 29, 15}, { 33, 15}, { 37, 15}, { 41, 15}, /* -61 ~ -58 dB */ +{ 23, 14}, { 13, 13}, { 58, 15}, { 65, 15}, /* -57 ~ -54 dB */ +{ 73, 15}, { 41, 14}, { 23, 13}, { 13, 12}, /* -53 ~ -50 dB */ +{ 29, 13}, { 65, 14}, { 73, 14}, { 41, 13}, /* -49 ~ -46 dB */ +{ 23, 12}, { 207, 15}, { 29, 12}, { 65, 13}, /* -45 ~ -42 dB */ +{ 73, 13}, { 41, 12}, { 23, 11}, { 413, 15}, /* -41 ~ -38 dB */ +{ 463, 15}, { 519, 15}, { 583, 15}, { 327, 14}, /* -37 ~ -34 dB */ +{ 367, 14}, { 823, 15}, { 231, 13}, { 1036, 15}, /* -33 ~ -30 dB */ +{ 1163, 15}, { 1305, 15}, { 183, 12}, { 1642, 15}, /* -29 ~ -26 dB */ +{ 1843, 15}, { 2068, 15}, { 145, 11}, { 2603, 15}, /* -25 ~ -22 dB */ +{ 365, 12}, { 3277, 15}, { 3677, 15}, { 4125, 15}, /* -21 ~ -18 dB */ +{ 4629, 15}, { 5193, 15}, { 5827, 15}, { 3269, 14}, /* -17 ~ -14 dB */ +{ 917, 12}, { 8231, 15}, { 9235, 15}, { 5181, 14}, /* -13 ~ -10 dB */ +{11627, 15}, {13045, 15}, {14637, 15}, {16423, 15}, /* -9 ~ -6 dB */ +{18427, 15}, {20675, 15}, { 5799, 13}, {26029, 15}, /* -5 ~ -2 dB */ +{ 7301, 13}, { 1, 0}, {18383, 14}, {10313, 13}, /* -1 ~ 2 dB */ +{23143, 14}, {25967, 14}, {29135, 14}, {16345, 13}, /* 3 ~ 6 dB */ +{ 4585, 11}, {20577, 13}, { 1443, 9}, {25905, 13}, /* 7 ~ 10 dB */ +{14533, 12}, { 8153, 11}, { 2287, 9}, {20529, 12}, /* 11 ~ 14 dB */ +{11517, 11}, { 6461, 10}, {28997, 12}, { 4067, 9}, /* 15 ~ 18 dB */ +{18253, 11}, { 10, 0}, {22979, 11}, {25783, 11}, /* 19 ~ 22 dB */ +{28929, 11}, {32459, 11}, { 9105, 9}, {20431, 10}, /* 23 ~ 26 dB */ +{22925, 10}, {12861, 9}, { 7215, 8}, {16191, 9}, /* 27 ~ 30 dB */ +{ 9083, 8}, {20383, 9}, {11435, 8}, { 6145, 7}, /* 31 ~ 34 dB */ +{ 3599, 6}, {32305, 9}, {18123, 8}, {20335, 8}, /* 35 ~ 38 dB */ +{ 713, 3}, { 100, 0}, { 7181, 6}, { 8057, 6}, /* 39 ~ 42 dB */ +{ 565, 2}, {20287, 7}, {11381, 6}, {25539, 7}, /* 43 ~ 46 dB */ +{ 1791, 3}, { 4019, 4}, { 9019, 5}, {20239, 6}, /* 47 ~ 50 dB */ +{ 5677, 4}, {25479, 6}, { 7147, 4}, { 8019, 4}, /* 51 ~ 54 dB */ +{17995, 5}, {20191, 5}, {11327, 4}, {12709, 4}, /* 55 ~ 58 dB */ +{ 3565, 2}, { 1000, 0}, { 1122, 0}, { 1259, 0}, /* 59 ~ 62 dB */ +{ 2825, 1}, {12679, 3}, { 7113, 2}, { 7981, 2}, /* 63 ~ 66 dB */ +{ 8955, 2}, {20095, 3}, {22547, 3}, {12649, 2}, /* 67 ~ 70 dB */ +{28385, 3}, { 3981, 0}, {17867, 2}, {20047, 2}, /* 71 ~ 74 dB */ +{11247, 1}, {12619, 1}, {14159, 1}, {31773, 2}, /* 75 ~ 78 dB */ +{17825, 1}, {10000, 0}, {11220, 0}, {12589, 0}, /* 79 ~ 82 dB */ +{28251, 1}, {15849, 0}, {17783, 0}, {19953, 0}, /* 83 ~ 86 dB */ +{22387, 0}, {25119, 0}, {28184, 0}, {31623, 0}, /* 87 ~ 90 dB */ +}; + +static const DECLARE_TLV_DB_RANGE(mic_gain_tlv, + 0, 1, TLV_DB_SCALE_ITEM(-9000, 600, 0), + 2, 5, TLV_DB_SCALE_ITEM(-8100, 300, 0), + 6, 7, TLV_DB_SCALE_ITEM(-7000, 200, 0), + 8, ARRAY_SIZE(mic_gain_table)-1, TLV_DB_SCALE_ITEM(-6500, 100, 0), +); + +int pdmic_get_mic_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + unsigned int dgain_val, scale_val; + int i; + + dgain_val = (snd_soc_read(codec, PDMIC_DSPR1) & PDMIC_DSPR1_DGAIN_MASK) + >> PDMIC_DSPR1_DGAIN_SHIFT; + + scale_val = (snd_soc_read(codec, PDMIC_DSPR0) & PDMIC_DSPR0_SCALE_MASK) + >> PDMIC_DSPR0_SCALE_SHIFT; + + for (i = 0; i < ARRAY_SIZE(mic_gain_table); i++) { + if ((mic_gain_table[i].dgain == dgain_val) && + (mic_gain_table[i].scale == scale_val)) + ucontrol->value.integer.value[0] = i; + } + + return 0; +} + +static int pdmic_put_mic_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + int max = mc->max; + unsigned int val; + int ret; + + val = ucontrol->value.integer.value[0]; + + if (val > max) + return -EINVAL; + + ret = snd_soc_update_bits(codec, PDMIC_DSPR1, PDMIC_DSPR1_DGAIN_MASK, + mic_gain_table[val].dgain << PDMIC_DSPR1_DGAIN_SHIFT); + if (ret < 0) + return ret; + + ret = snd_soc_update_bits(codec, PDMIC_DSPR0, PDMIC_DSPR0_SCALE_MASK, + mic_gain_table[val].scale << PDMIC_DSPR0_SCALE_SHIFT); + if (ret < 0) + return ret; + + return 0; +} + +static const struct snd_kcontrol_new atmel_pdmic_snd_controls[] = { +SOC_SINGLE_EXT_TLV("Mic Capture Volume", PDMIC_DSPR1, PDMIC_DSPR1_DGAIN_SHIFT, + ARRAY_SIZE(mic_gain_table)-1, 0, + pdmic_get_mic_volsw, pdmic_put_mic_volsw, mic_gain_tlv), + +SOC_SINGLE("High Pass Filter Switch", PDMIC_DSPR0, + PDMIC_DSPR0_HPFBYP_SHIFT, 1, 1), + +SOC_SINGLE("SINCC Filter Switch", PDMIC_DSPR0, PDMIC_DSPR0_SINBYP_SHIFT, 1, 1), +}; + +static int atmel_pdmic_codec_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_card *card = snd_soc_codec_get_drvdata(codec); + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(card); + + snd_soc_update_bits(codec, PDMIC_DSPR1, PDMIC_DSPR1_OFFSET_MASK, + (u32)(dd->pdata->mic_offset << PDMIC_DSPR1_OFFSET_SHIFT)); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_pdmic = { + .probe = atmel_pdmic_codec_probe, + .controls = atmel_pdmic_snd_controls, + .num_controls = ARRAY_SIZE(atmel_pdmic_snd_controls), +}; + +/* codec dai component */ +#define PDMIC_MR_PRESCAL_MAX_VAL 127 + +static int +atmel_pdmic_codec_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int rate_min = substream->runtime->hw.rate_min; + unsigned int rate_max = substream->runtime->hw.rate_max; + int fs = params_rate(params); + int bits = params_width(params); + unsigned long pclk_rate, gclk_rate; + unsigned int f_pdmic; + u32 mr_val, dspr0_val, pclk_prescal, gclk_prescal; + + if (params_channels(params) != 1) { + dev_err(codec->dev, + "only supports one channel\n"); + return -EINVAL; + } + + if ((fs < rate_min) || (fs > rate_max)) { + dev_err(codec->dev, + "sample rate is %dHz, min rate is %dHz, max rate is %dHz\n", + fs, rate_min, rate_max); + + return -EINVAL; + } + + switch (bits) { + case 16: + dspr0_val = (PDMIC_DSPR0_SIZE_16_BITS + << PDMIC_DSPR0_SIZE_SHIFT); + break; + case 32: + dspr0_val = (PDMIC_DSPR0_SIZE_32_BITS + << PDMIC_DSPR0_SIZE_SHIFT); + break; + default: + return -EINVAL; + } + + if ((fs << 7) > (rate_max << 6)) { + f_pdmic = fs << 6; + dspr0_val |= PDMIC_DSPR0_OSR_64 << PDMIC_DSPR0_OSR_SHIFT; + } else { + f_pdmic = fs << 7; + dspr0_val |= PDMIC_DSPR0_OSR_128 << PDMIC_DSPR0_OSR_SHIFT; + } + + pclk_rate = clk_get_rate(dd->pclk); + gclk_rate = clk_get_rate(dd->gclk); + + /* PRESCAL = SELCK/(2*f_pdmic) - 1*/ + pclk_prescal = (u32)(pclk_rate/(f_pdmic << 1)) - 1; + gclk_prescal = (u32)(gclk_rate/(f_pdmic << 1)) - 1; + + if ((pclk_prescal > PDMIC_MR_PRESCAL_MAX_VAL) || + (gclk_rate/((gclk_prescal + 1) << 1) < + pclk_rate/((pclk_prescal + 1) << 1))) { + mr_val = gclk_prescal << PDMIC_MR_PRESCAL_SHIFT; + mr_val |= PDMIC_MR_CLKS_GCK << PDMIC_MR_CLKS_SHIFT; + } else { + mr_val = pclk_prescal << PDMIC_MR_PRESCAL_SHIFT; + mr_val |= PDMIC_MR_CLKS_PCK << PDMIC_MR_CLKS_SHIFT; + } + + snd_soc_update_bits(codec, PDMIC_MR, + PDMIC_MR_PRESCAL_MASK | PDMIC_MR_CLKS_MASK, mr_val); + + snd_soc_update_bits(codec, PDMIC_DSPR0, + PDMIC_DSPR0_OSR_MASK | PDMIC_DSPR0_SIZE_MASK, dspr0_val); + + return 0; +} + +static int atmel_pdmic_codec_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + snd_soc_update_bits(codec, PDMIC_CR, PDMIC_CR_ENPDM_MASK, + PDMIC_CR_ENPDM_DIS << PDMIC_CR_ENPDM_SHIFT); + + return 0; +} + +static int atmel_pdmic_codec_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u32 val; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = PDMIC_CR_ENPDM_EN << PDMIC_CR_ENPDM_SHIFT; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + val = PDMIC_CR_ENPDM_DIS << PDMIC_CR_ENPDM_SHIFT; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, PDMIC_CR, PDMIC_CR_ENPDM_MASK, val); + + return 0; +} + +static const struct snd_soc_dai_ops atmel_pdmic_codec_dai_ops = { + .hw_params = atmel_pdmic_codec_dai_hw_params, + .prepare = atmel_pdmic_codec_dai_prepare, + .trigger = atmel_pdmic_codec_dai_trigger, +}; + +#define ATMEL_PDMIC_CODEC_DAI_NAME "atmel-pdmic-hifi" + +static struct snd_soc_dai_driver atmel_pdmic_codec_dai = { + .name = ATMEL_PDMIC_CODEC_DAI_NAME, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = ATMEL_PDMIC_FORMATS, + }, + .ops = &atmel_pdmic_codec_dai_ops, +}; + +/* ASoC sound card */ +static int atmel_pdmic_asoc_card_init(struct device *dev, + struct snd_soc_card *card) +{ + struct snd_soc_dai_link *dai_link; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(card); + + dai_link = devm_kzalloc(dev, sizeof(*dai_link), GFP_KERNEL); + if (!dai_link) + return -ENOMEM; + + dai_link->name = "PDMIC"; + dai_link->stream_name = "PDMIC PCM"; + dai_link->codec_dai_name = ATMEL_PDMIC_CODEC_DAI_NAME; + dai_link->cpu_dai_name = dev_name(dev); + dai_link->codec_name = dev_name(dev); + dai_link->platform_name = dev_name(dev); + + card->dai_link = dai_link; + card->num_links = 1; + card->name = dd->pdata->card_name; + card->dev = dev; + + return 0; +} + +static void atmel_pdmic_get_sample_rate(struct atmel_pdmic *dd, + unsigned int *rate_min, unsigned int *rate_max) +{ + u32 mic_min_freq = dd->pdata->mic_min_freq; + u32 mic_max_freq = dd->pdata->mic_max_freq; + u32 clk_max_rate = (u32)(clk_get_rate(dd->pclk) >> 1); + u32 clk_min_rate = (u32)(clk_get_rate(dd->gclk) >> 8); + + if (mic_max_freq > clk_max_rate) + mic_max_freq = clk_max_rate; + + if (mic_min_freq < clk_min_rate) + mic_min_freq = clk_min_rate; + + *rate_min = DIV_ROUND_CLOSEST(mic_min_freq, 128); + *rate_max = mic_max_freq >> 6; +} + +/* PDMIC interrupt handler */ +static irqreturn_t atmel_pdmic_interrupt(int irq, void *dev_id) +{ + struct atmel_pdmic *dd = (struct atmel_pdmic *)dev_id; + u32 pdmic_isr; + irqreturn_t ret = IRQ_NONE; + + regmap_read(dd->regmap, PDMIC_ISR, &pdmic_isr); + + if (pdmic_isr & PDMIC_ISR_OVRE) { + regmap_update_bits(dd->regmap, PDMIC_CR, PDMIC_CR_ENPDM_MASK, + PDMIC_CR_ENPDM_DIS << PDMIC_CR_ENPDM_SHIFT); + + snd_pcm_stop_xrun(dd->substream); + + ret = IRQ_HANDLED; + } + + return ret; +} + +/* regmap configuration */ +#define ATMEL_PDMIC_REG_MAX 0x124 +static const struct regmap_config atmel_pdmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = ATMEL_PDMIC_REG_MAX, +}; + +static int atmel_pdmic_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct atmel_pdmic *dd; + struct resource *res; + void __iomem *io_base; + const struct atmel_pdmic_pdata *pdata; + struct snd_soc_card *card; + unsigned int rate_min, rate_max; + int ret; + + pdata = atmel_pdmic_dt_init(dev); + if (IS_ERR(pdata)) + return PTR_ERR(pdata); + + dd = devm_kzalloc(dev, sizeof(*dd), GFP_KERNEL); + if (!dd) + return -ENOMEM; + + dd->pdata = pdata; + + dd->irq = platform_get_irq(pdev, 0); + if (dd->irq < 0) { + ret = dd->irq; + dev_err(dev, "failed to could not get irq: %d\n", ret); + return ret; + } + + dd->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(dd->pclk)) { + ret = PTR_ERR(dd->pclk); + dev_err(dev, "failed to get peripheral clock: %d\n", ret); + return ret; + } + + dd->gclk = devm_clk_get(dev, "gclk"); + if (IS_ERR(dd->gclk)) { + ret = PTR_ERR(dd->gclk); + dev_err(dev, "failed to get GCK: %d\n", ret); + return ret; + } + + /* The gclk clock frequency must always be tree times + * lower than the pclk clock frequency + */ + ret = clk_set_rate(dd->gclk, clk_get_rate(dd->pclk)/3); + if (ret) { + dev_err(dev, "failed to set GCK clock rate: %d\n", ret); + return ret; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(dev, "no memory resource\n"); + return -ENXIO; + } + + io_base = devm_ioremap_resource(dev, res); + if (IS_ERR(io_base)) { + ret = PTR_ERR(io_base); + dev_err(dev, "failed to remap register memory: %d\n", ret); + return ret; + } + + dd->phy_base = res->start; + + dd->regmap = devm_regmap_init_mmio(dev, io_base, + &atmel_pdmic_regmap_config); + if (IS_ERR(dd->regmap)) { + ret = PTR_ERR(dd->regmap); + dev_err(dev, "failed to init register map: %d\n", ret); + return ret; + } + + ret = devm_request_irq(dev, dd->irq, atmel_pdmic_interrupt, 0, + "PDMIC", (void *)dd); + if (ret < 0) { + dev_err(dev, "can't register ISR for IRQ %u (ret=%i)\n", + dd->irq, ret); + return ret; + } + + /* Get the minimal and maximal sample rate that micphone supports */ + atmel_pdmic_get_sample_rate(dd, &rate_min, &rate_max); + + /* register cpu dai */ + atmel_pdmic_cpu_dai.capture.rate_min = rate_min; + atmel_pdmic_cpu_dai.capture.rate_max = rate_max; + ret = devm_snd_soc_register_component(dev, + &atmel_pdmic_cpu_dai_component, + &atmel_pdmic_cpu_dai, 1); + if (ret) { + dev_err(dev, "could not register CPU DAI: %d\n", ret); + return ret; + } + + /* register platform */ + ret = devm_snd_dmaengine_pcm_register(dev, + &atmel_pdmic_dmaengine_pcm_config, + 0); + if (ret) { + dev_err(dev, "could not register platform: %d\n", ret); + return ret; + } + + /* register codec and codec dai */ + atmel_pdmic_codec_dai.capture.rate_min = rate_min; + atmel_pdmic_codec_dai.capture.rate_max = rate_max; + ret = snd_soc_register_codec(dev, &soc_codec_dev_pdmic, + &atmel_pdmic_codec_dai, 1); + if (ret) { + dev_err(dev, "could not register codec: %d\n", ret); + return ret; + } + + /* register sound card */ + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) { + ret = -ENOMEM; + goto unregister_codec; + } + + snd_soc_card_set_drvdata(card, dd); + platform_set_drvdata(pdev, card); + + ret = atmel_pdmic_asoc_card_init(dev, card); + if (ret) { + dev_err(dev, "failed to init sound card: %d\n", ret); + goto unregister_codec; + } + + ret = devm_snd_soc_register_card(dev, card); + if (ret) { + dev_err(dev, "failed to register sound card: %d\n", ret); + goto unregister_codec; + } + + return 0; + +unregister_codec: + snd_soc_unregister_codec(dev); + return ret; +} + +static int atmel_pdmic_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver atmel_pdmic_driver = { + .driver = { + .name = "atmel-pdmic", + .of_match_table = of_match_ptr(atmel_pdmic_of_match), + .pm = &snd_soc_pm_ops, + }, + .probe = atmel_pdmic_probe, + .remove = atmel_pdmic_remove, +}; +module_platform_driver(atmel_pdmic_driver); + +MODULE_DESCRIPTION("Atmel PDMIC driver under ALSA SoC architecture"); +MODULE_AUTHOR("Songjun Wu <songjun.wu(a)atmel.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/atmel/atmel-pdmic.h b/sound/soc/atmel/atmel-pdmic.h new file mode 100644 index 0000000..4527ac7 --- /dev/null +++ b/sound/soc/atmel/atmel-pdmic.h @@ -0,0 +1,80 @@ +#ifndef __ATMEL_PDMIC_H_ +#define __ATMEL_PDMIC_H_ + +#include <linux/bitops.h> + +#define PDMIC_CR 0x00000000 + +#define PDMIC_CR_SWRST 0x1 +#define PDMIC_CR_SWRST_MASK BIT(0) +#define PDMIC_CR_SWRST_SHIFT (0) + +#define PDMIC_CR_ENPDM_DIS 0x0 +#define PDMIC_CR_ENPDM_EN 0x1 +#define PDMIC_CR_ENPDM_MASK BIT(4) +#define PDMIC_CR_ENPDM_SHIFT (4) + +#define PDMIC_MR 0x00000004 + +#define PDMIC_MR_CLKS_PCK 0x0 +#define PDMIC_MR_CLKS_GCK 0x1 +#define PDMIC_MR_CLKS_MASK BIT(4) +#define PDMIC_MR_CLKS_SHIFT (4) + +#define PDMIC_MR_PRESCAL_MASK GENMASK(14, 8) +#define PDMIC_MR_PRESCAL_SHIFT (8) + +#define PDMIC_CDR 0x00000014 + +#define PDMIC_IER 0x00000018 +#define PDMIC_IER_OVRE BIT(25) + +#define PDMIC_IDR 0x0000001c +#define PDMIC_IDR_OVRE BIT(25) + +#define PDMIC_IMR 0x00000020 + +#define PDMIC_ISR 0x00000024 +#define PDMIC_ISR_OVRE BIT(25) + +#define PDMIC_DSPR0 0x00000058 + +#define PDMIC_DSPR0_HPFBYP_DIS 0x1 +#define PDMIC_DSPR0_HPFBYP_EN 0x0 +#define PDMIC_DSPR0_HPFBYP_MASK BIT(1) +#define PDMIC_DSPR0_HPFBYP_SHIFT (1) + +#define PDMIC_DSPR0_SINBYP_DIS 0x1 +#define PDMIC_DSPR0_SINBYP_EN 0x0 +#define PDMIC_DSPR0_SINBYP_MASK BIT(2) +#define PDMIC_DSPR0_SINBYP_SHIFT (2) + +#define PDMIC_DSPR0_SIZE_16_BITS 0x0 +#define PDMIC_DSPR0_SIZE_32_BITS 0x1 +#define PDMIC_DSPR0_SIZE_MASK BIT(3) +#define PDMIC_DSPR0_SIZE_SHIFT (3) + +#define PDMIC_DSPR0_OSR_128 0x0 +#define PDMIC_DSPR0_OSR_64 0x1 +#define PDMIC_DSPR0_OSR_MASK GENMASK(6, 4) +#define PDMIC_DSPR0_OSR_SHIFT (4) + +#define PDMIC_DSPR0_SCALE_MASK GENMASK(11, 8) +#define PDMIC_DSPR0_SCALE_SHIFT (8) + +#define PDMIC_DSPR0_SHIFT_MASK GENMASK(15, 12) +#define PDMIC_DSPR0_SHIFT_SHIFT (12) + +#define PDMIC_DSPR1 0x0000005c + +#define PDMIC_DSPR1_DGAIN_MASK GENMASK(14, 0) +#define PDMIC_DSPR1_DGAIN_SHIFT (0) + +#define PDMIC_DSPR1_OFFSET_MASK GENMASK(31, 16) +#define PDMIC_DSPR1_OFFSET_SHIFT (16) + +#define PDMIC_WPMR 0x000000e4 + +#define PDMIC_WPSR 0x000000e8 + +#endif -- 1.7.9.5

2 1

[alsa-devel] Applied "ASoC: atmel-classd: DT binding for PDMIC driver" to the asoc tree
by Mark Brown 18 Dec '15

18 Dec '15

The patch ASoC: atmel-classd: DT binding for PDMIC driver has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4bc5145cec89b3df719d87d0a6876aa00941d66c Mon Sep 17 00:00:00 2001 From: Songjun Wu <songjun.wu(a)atmel.com> Date: Thu, 17 Dec 2015 17:50:00 +0800 Subject: [PATCH] ASoC: atmel-classd: DT binding for PDMIC driver DT binding documentation for this new ASoC driver. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/atmel-pdmic.txt | 55 ++++++++++++++++++++++ 1 file changed, 55 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-pdmic.txt diff --git a/Documentation/devicetree/bindings/sound/atmel-pdmic.txt b/Documentation/devicetree/bindings/sound/atmel-pdmic.txt new file mode 100644 index 000000000000..e0875f17c229 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel-pdmic.txt @@ -0,0 +1,55 @@ +* Atmel PDMIC driver under ALSA SoC architecture + +Required properties: +- compatible + Should be "atmel,sama5d2-pdmic". +- reg + Should contain PDMIC registers location and length. +- interrupts + Should contain the IRQ line for the PDMIC. +- dmas + One DMA specifiers as described in atmel-dma.txt and dma.txt files. +- dma-names + Must be "rx". +- clock-names + Required elements: + - "pclk" peripheral clock + - "gclk" generated clock +- clocks + Must contain an entry for each required entry in clock-names. + Please refer to clock-bindings.txt. +- atmel,mic-min-freq + The minimal frequency that the micphone supports. +- atmel,mic-max-freq + The maximal frequency that the micphone supports. + +Optional properties: +- pinctrl-names, pinctrl-0 + Please refer to pinctrl-bindings.txt. +- atmel,model + The user-visible name of this sound card. + The default value is "PDMIC". +- atmel,mic-offset + The offset that should be added. + The range is from -32768 to 32767. + The default value is 0. + +Example: + pdmic@f8018000 { + compatible = "atmel,sama5d2-pdmic"; + reg = <0xf8018000 0x124>; + interrupts = <48 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) + | AT91_XDMAC_DT_PERID(50))>; + dma-names = "rx"; + clocks = <&pdmic_clk>, <&pdmic_gclk>; + clock-names = "pclk", "gclk"; + + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pdmic_default>; + atmel,model = "PDMIC @ sama5d2_xplained"; + atmel,mic-min-freq = <1000000>; + atmel,mic-max-freq = <3246000>; + atmel,mic-offset = <0x0>; + }; -- 2.6.2

1 0

[alsa-devel] [PATCH 2/2] ASoC: wm8974: add devicetree support
by Mans Rullgard 18 Dec '15

18 Dec '15

This adds devicetree support to the wm8974 codec driver. With a DT-based kernel, there is no board-specific setting to select the driver so allow it to be manually chosen. Signed-off-by: Mans Rullgard <mans(a)mansr.com> --- sound/soc/codecs/Kconfig | 3 ++- sound/soc/codecs/wm8974.c | 7 +++++++ 2 files changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cfdafc4..e36b14c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -838,7 +838,8 @@ config SND_SOC_WM8971 tristate config SND_SOC_WM8974 - tristate + tristate "Wolfson Microelectronics WM8974 codec" + depends on I2C config SND_SOC_WM8978 tristate "Wolfson Microelectronics WM8978 codec" diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 4c29bd2..c284c7b 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -632,9 +632,16 @@ static const struct i2c_device_id wm8974_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8974_i2c_id); +static const struct of_device_id wm8974_of_match[] = { + { .compatible = "wlf,wm8974", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8974_of_match); + static struct i2c_driver wm8974_i2c_driver = { .driver = { .name = "wm8974", + .of_match_table = wm8974_of_match, }, .probe = wm8974_i2c_probe, .remove = wm8974_i2c_remove, -- 2.6.3

3 3

[alsa-devel] Applied "ASoC: wm8974: add binding for WM8974 codec" to the asoc tree
by Mark Brown 18 Dec '15

18 Dec '15

The patch ASoC: wm8974: add binding for WM8974 codec has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 47d358bbf22d504a41007e7fbf71b5b4b1b45b79 Mon Sep 17 00:00:00 2001 From: Mans Rullgard <mans(a)mansr.com> Date: Wed, 16 Dec 2015 13:55:13 +0000 Subject: [PATCH] ASoC: wm8974: add binding for WM8974 codec This adds a binding for the Wolfson WM8974 mono audio codec. Signed-off-by: Mans Rullgard <mans(a)mansr.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/wlf,wm8974.txt | 15 +++++++++++++++ 1 file changed, 15 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8974.txt diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8974.txt b/Documentation/devicetree/bindings/sound/wlf,wm8974.txt new file mode 100644 index 000000000000..01d3a7c83419 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8974.txt @@ -0,0 +1,15 @@ +WM8974 audio CODEC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + - compatible: "wlf,wm8974" + - reg: the I2C address or SPI chip select number of the device + +Examples: + +codec: wm8974@1a { + compatible = "wlf,wm8974"; + reg = <0x1a>; +}; -- 2.6.2

1 0

[alsa-devel] [PATCH] ALSA: hda - Set intel hda controller power at freeze() and thaw()
by Xiong Zhang 18 Dec '15

18 Dec '15

It takes three minutes to enter into hibernation on some OEM SKL machines and we see many codec spurious response after thaw() opertion. This is because HDA is still in D0 state after freeze() call and pci_pm_freeze/pci_pm_freeze_noirq() don't set D3 hot in pci_bus driver. It seems bios still access HDA when system enter into freeze state, HDA will receive codec response interrupt immediately after thaw() call. Because of this unexpected interrupt, HDA enter into a abnormal state and slow down the system enter into hibernation. In this patch, we put HDA into D3 hot state in azx_freeze_noirq() and put HDA into D0 state in azx_thaw_noirq(). Signed-off-by: Xiong Zhang <xiong.y.zhang(a)intel.com> --- sound/pci/hda/hda_intel.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c38c68f..3ba7cbc 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -890,6 +890,16 @@ static int azx_suspend(struct device *dev) return 0; } +static int azx_freeze_noirq(struct device *dev) +{ + struct pci_dev *pci = to_pci_dev(dev); + + if (pci->vendor == PCI_VENDOR_ID_INTEL) + pci_set_power_state(pci, PCI_D3hot); + + return 0; +} + static int azx_resume(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); @@ -924,6 +934,14 @@ static int azx_resume(struct device *dev) trace_azx_resume(chip); return 0; } + +static int azx_thaw_noirq(struct device *dev) +{ + struct pci_dev *pci = to_pci_dev(dev); + + if (pci->vendor == PCI_VENDOR_ID_INTEL) + pci_set_power_state(pci, PCI_D0); +} #endif /* CONFIG_PM_SLEEP || SUPPORT_VGA_SWITCHEROO */ #ifdef CONFIG_PM @@ -1035,6 +1053,8 @@ static int azx_runtime_idle(struct device *dev) static const struct dev_pm_ops azx_pm = { SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + .freeze_noirq = azx_freeze_noirq, + .thaw_noirq = azx_thaw_noirq, SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, azx_runtime_idle) }; -- 1.8.2.1

4 7

[alsa-devel] HT Omega eCLARO sound card
by Patricio Rossi 18 Dec '15

18 Dec '15

I got this card few days ago, after connected it into my PC got detected by lspci, but I cannot get a module to load, looks like the oxigen is the correct one but even when I load it manually nothing happens. I did some search on the web and I found people that was able to use it but I can't find how, I read the oxigen source code and the card is not there, is very similar to the ASUS Xonar. Is this the correct place to look for help? What is the status of this driver? where I can get more information? Thank you!

2 6

[alsa-devel] mixer controls not detected for class compliant USB audio interface (RME Babyface Pro)
by Be 17 Dec '15

17 Dec '15

Hello, I recently got a USB audio class compliant RME Babyface Pro interface. I previously posted about an issue getting all of the channels to work, but that ended up being an issue with PortAudio that is trivial to work around. Anyway, there is still the issue that the device's mixer controls are not detected by Linux. alsamixer sees no controls at all. The mixer controls must be accessible in the device's class compliant mode because the manufacturer's iPad mixer program can control its mixer and onboard DSP effects. I have attached lsusb -vvv output for the device. Below is dmesg output when the device is plugged in: [191013.268711] usb 2-1.2: new high-speed USB device number 11 using ehci-pci [191013.355392] usb 2-1.2: New USB device found, idVendor=2a39, idProduct=3fb0 [191013.355403] usb 2-1.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [191013.355409] usb 2-1.2: Product: Babyface Pro (70785713) [191013.355413] usb 2-1.2: Manufacturer: RME [191013.355417] usb 2-1.2: SerialNumber: 9399A998B9CB3C8 [191013.360347] usb 2-1.2: unit 2 not found!

2 3

[alsa-devel] [PATCH] ASoC: rt5677: Reconfigure PLL1 after resume
by Ben Zhang 17 Dec '15

17 Dec '15

Sometimes PLL1 stops working if the codec loses power during suspend (when pow-ldo2 or reset gpio is used). MX-7Bh(RT5677_PLL1_CTRL2) is cleared and won't be restored by regcache since it's volatile. MX-7Bh has one status bit and M code for PLL1. rt5677_set_dai_pll doesn't reconfigure PLL1 after resume because it thinks the PLL params are not changed. This patch clears the cached PLL params at resume so that rt5677_set_dai_pll can reconfigure the PLL after resume. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/rt5677.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f73fd12..c404f51 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4792,6 +4792,9 @@ static int rt5677_resume(struct snd_soc_codec *codec) struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); if (!rt5677->dsp_vad_en) { + rt5677->pll_src = 0; + rt5677->pll_in = 0; + rt5677->pll_out = 0; gpiod_set_value_cansleep(rt5677->pow_ldo2, 1); gpiod_set_value_cansleep(rt5677->reset_pin, 0); if (rt5677->pow_ldo2 || rt5677->reset_pin) -- 2.6.0.rc2.230.g3dd15c0

3 4

[alsa-devel] [sound:for-next 38/40] sound/pci/hda/hda_tegra.c:484:36: error: 'AZX_DCAPS_RIRB_DELAY' undeclared
by kbuild test robot 17 Dec '15

17 Dec '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 64cfd2fd5325dbe24b990dbe003c22f5675d9742 commit: 119a42ac2b249abd6b5195e0606a0d70af2953eb [38/40] ALSA: hda - Move Nvidia-specific check into hda_intel.c config: arm-allyesconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 119a42ac2b249abd6b5195e0606a0d70af2953eb # save the attached .config to linux build tree make.cross ARCH=arm All errors (new ones prefixed by >>): sound/pci/hda/hda_tegra.c: In function 'hda_tegra_probe': >> sound/pci/hda/hda_tegra.c:484:36: error: 'AZX_DCAPS_RIRB_DELAY' undeclared (first use in this function) const unsigned int driver_flags = AZX_DCAPS_RIRB_DELAY | ^ sound/pci/hda/hda_tegra.c:484:36: note: each undeclared identifier is reported only once for each function it appears in vim +/AZX_DCAPS_RIRB_DELAY +484 sound/pci/hda/hda_tegra.c 3c320f3f Dylan Reid 2014-05-19 478 {}, 3c320f3f Dylan Reid 2014-05-19 479 }; f73387cb Dylan Reid 2014-05-20 480 MODULE_DEVICE_TABLE(of, hda_tegra_match); 3c320f3f Dylan Reid 2014-05-19 481 3c320f3f Dylan Reid 2014-05-19 482 static int hda_tegra_probe(struct platform_device *pdev) 3c320f3f Dylan Reid 2014-05-19 483 { 88871dd2 Thierry Reding 2015-05-05 @484 const unsigned int driver_flags = AZX_DCAPS_RIRB_DELAY | 88871dd2 Thierry Reding 2015-05-05 485 AZX_DCAPS_CORBRP_SELF_CLEAR; 3c320f3f Dylan Reid 2014-05-19 486 struct snd_card *card; 3c320f3f Dylan Reid 2014-05-19 487 struct azx *chip; :::::: The code at line 484 was first introduced by commit :::::: 88871dd25f64b9bf16222e829ad766fac201866a ALSA: hda/tegra - Set CORBRP self-clear flag :::::: TO: Thierry Reding <treding(a)nvidia.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] [RESEND PATCH v2 0/2] ASoC: atmel-pdmic: add driver for Atmel PDMIC
by Songjun Wu 17 Dec '15

17 Dec '15

The Pulse Density Modulation Inteface Controller driver includes two parts. 1) Driver code to implement the PDMIC function. 2) Device tree binding documentation, it describes how to add the PDMIC in device tree. Changes in v2: - Remove the function 'atmel_pdmic_codec_get_remap', since the regmap is initialized in the core. - Replace 'if' with 'switch' in function 'atmel_pdmic_codec_dai_hw_params'. - Fix the coding style issue in function 'atmel_pdmic_dt_init', 'atmel_pdmic_interrupt','pdmic_put_mic_volsw', 'atmel_pdmic_codec_probe', 'pdmic_get_mic_volsw' and 'atmel_pdmic_codec_dai_hw_params'. Songjun Wu (2): ASoC: atmel-pdmic: add the Pulse Density Modulation Interface Controller ASoC: atmel-classd: DT binding for PDMIC driver .../devicetree/bindings/sound/atmel-pdmic.txt | 55 ++ sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-pdmic.c | 738 ++++++++++++++++++++ sound/soc/atmel/atmel-pdmic.h | 80 +++ 5 files changed, 884 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-pdmic.txt create mode 100644 sound/soc/atmel/atmel-pdmic.c create mode 100644 sound/soc/atmel/atmel-pdmic.h -- 1.7.9.5

1 1

[alsa-devel] [PATCH 0/4] Be ecologist; save more bits in AZX_DCAPS_*
by Takashi Iwai 17 Dec '15

17 Dec '15

Hi, this patchset cleans up messy AZX_DCAPS_* definitions and remove a few usages there. Takashi === Takashi Iwai (4): ALSA: hda - Merge RIRB_PRE_DELAY into CTX_WORKAROUND caps ALSA: hda - Move Nvidia-specific check into hda_intel.c ALSA: hda - Drop AZX_DCAPS_POSFIX_VIA bit ALSA: hda - Drop unused AZX_DCAPS_REVERSE_ASSIGN sound/pci/hda/hda_controller.c | 7 +------ sound/pci/hda/hda_controller.h | 8 ++++---- sound/pci/hda/hda_intel.c | 20 +++++++++++--------- 3 files changed, 16 insertions(+), 19 deletions(-) -- 2.6.4

1 4

[alsa-devel] [PATCH 0/2] ASoC: atmel-pdmic: add driver for Atmel PDMIC
by Songjun Wu 17 Dec '15

17 Dec '15

The Pulse Density Modulation Inteface Controller driver includes two parts. 1) Driver code to implement the PDMIC function. 2) Device tree binding documentation, it describes how to add the PDMIC in device tree. Songjun Wu (2): ASoC: atmel-pdmic: add the Pulse Density Modulation Interface Controller ASoC: atmel-pdmic: DT binding for PDMIC driver .../devicetree/bindings/sound/atmel-pdmic.txt | 55 ++ sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-pdmic.c | 745 ++++++++++++++++++++ sound/soc/atmel/atmel-pdmic.h | 80 +++ 5 files changed, 891 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-pdmic.txt create mode 100644 sound/soc/atmel/atmel-pdmic.c create mode 100644 sound/soc/atmel/atmel-pdmic.h -- 1.7.9.5

3 3

[alsa-devel] [patch] ASoC: Intel: Skylake: Fix a couple signedness bugs
by Dan Carpenter 16 Dec '15

16 Dec '15

These need to be unsigned because they hold negative error codes. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 8cd5cdb..e26f474 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -208,7 +208,7 @@ static unsigned int skl_get_errorcode(struct sst_dsp *ctx) * since get/set_module are called from DAPM context, * we don't need lock for usage count */ -static unsigned int skl_get_module(struct sst_dsp *ctx, u16 mod_id) +static int skl_get_module(struct sst_dsp *ctx, u16 mod_id) { struct skl_module_table *module; @@ -220,7 +220,7 @@ static unsigned int skl_get_module(struct sst_dsp *ctx, u16 mod_id) return -EINVAL; } -static unsigned int skl_put_module(struct sst_dsp *ctx, u16 mod_id) +static int skl_put_module(struct sst_dsp *ctx, u16 mod_id) { struct skl_module_table *module; @@ -340,7 +340,7 @@ static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, char *guid) static int skl_unload_module(struct sst_dsp *ctx, u16 mod_id) { - unsigned int usage_cnt; + int usage_cnt; struct skl_sst *skl = ctx->thread_context; int ret = 0;

3 4

[alsa-devel] [PATCH] ASoC: arizona: In arizona_calc_fratio make new codecs the default case
by Richard Fitzgerald 16 Dec '15

16 Dec '15

This patch rearranges the switch statement in arizona_calc_fratio so that older codecs are the special cases, with the default case applying to newer codecs (WM8998 and later). This is preferable because it avoids having to patch new cases in every time a new codec is added. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index d90b3c5..38a73e3 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2020,18 +2020,18 @@ static int arizona_calc_fratio(struct arizona_fll *fll, } switch (fll->arizona->type) { + case WM5102: + case WM8997: + return init_ratio; case WM5110: case WM8280: if (fll->arizona->rev < 3 || sync) return init_ratio; break; - case WM8998: - case WM1814: + default: if (sync) return init_ratio; break; - default: - return init_ratio; } cfg->fratio = init_ratio - 1; -- 1.9.1

2 1

[alsa-devel] [patch] ASoC: Intel: Skylake: pointer math issue
by Dan Carpenter 16 Dec '15

16 Dec '15

"data" is a u32 pointer so this copies the information to wrong place entirely. Fixes: 140adfba5280 ('ASoC: Intel: Skylake: Add tlv byte kcontrols') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index dde8d3c..2c4405e 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -885,9 +885,9 @@ static int skl_tplg_tlv_control_get(struct snd_kcontrol *kcontrol, if (bc->params) { if (copy_to_user(data, &bc->param_id, sizeof(u32))) return -EFAULT; - if (copy_to_user(data + sizeof(u32), &size, sizeof(u32))) + if (copy_to_user(data + 1, &size, sizeof(u32))) return -EFAULT; - if (copy_to_user(data + 2 * sizeof(u32), bc->params, size)) + if (copy_to_user(data + 2, bc->params, size)) return -EFAULT; }

3 3

[alsa-devel] [patch v2] ASoC: Intel: Skylake: Fix a couple signedness bugs
by Dan Carpenter 16 Dec '15

16 Dec '15

These need to be signed because they hold negative error codes. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Fix a typo in the change log. s/unsigned/signed/ diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 8cd5cdb..e26f474 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -208,7 +208,7 @@ static unsigned int skl_get_errorcode(struct sst_dsp *ctx) * since get/set_module are called from DAPM context, * we don't need lock for usage count */ -static unsigned int skl_get_module(struct sst_dsp *ctx, u16 mod_id) +static int skl_get_module(struct sst_dsp *ctx, u16 mod_id) { struct skl_module_table *module; @@ -220,7 +220,7 @@ static unsigned int skl_get_module(struct sst_dsp *ctx, u16 mod_id) return -EINVAL; } -static unsigned int skl_put_module(struct sst_dsp *ctx, u16 mod_id) +static int skl_put_module(struct sst_dsp *ctx, u16 mod_id) { struct skl_module_table *module; @@ -340,7 +340,7 @@ static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, char *guid) static int skl_unload_module(struct sst_dsp *ctx, u16 mod_id) { - unsigned int usage_cnt; + int usage_cnt; struct skl_sst *skl = ctx->thread_context; int ret = 0;

2 1

[alsa-devel] [PATCH v5 1/2] ALSA: hda - hdmi dynamically bind PCM to pin when monitor hotplug
by libin.yang＠linux.intel.com 16 Dec '15

16 Dec '15

From: Libin Yang <libin.yang(a)linux.intel.com> Dynamically bind/unbind the PCM to pin when HDMI/DP monitor hotplug. When monitor is connected, find a proper PCM for the monitor. When monitor is disconnected, unbind the PCM from the pin. The binding policy (use Intel platform as example) is: 1. Try to use the legacy pin-pcm mapping for the device entry 0 of the pin. 2. If step 1 fails, try to bind pin to the backup PCMs. For example, on Intel platform, if DP MST is enabled, 5 PCMs will be created. PCM 3, PCM 7, PCM 8 are supposed to be used by device entry 0 of pin 5, pin 6 and pin 7. PCM 9 and PCM 10 are the backup PCMs. 3. If step 2 fails, try to find any PCM to bind to the pin. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 82 ++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 80 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 19bf782..56b7625 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -73,6 +73,8 @@ struct hdmi_spec_per_cvt { struct hdmi_spec_per_pin { hda_nid_t pin_nid; + /* pin idx, different device entries on the same pin use the same idx */ + int pin_nid_idx; int num_mux_nids; hda_nid_t mux_nids[HDA_MAX_CONNECTIONS]; int mux_idx; @@ -85,6 +87,7 @@ struct hdmi_spec_per_pin { struct snd_kcontrol *eld_ctl; struct snd_jack *acomp_jack; /* jack via audio component */ struct hda_pcm *pcm; /* pointer to spec->pcm_rec[n] dynamically*/ + int pcm_idx; /* which pcm is attached. -1 means no pcm is attached */ int repoll_count; bool setup; /* the stream has been set up by prepare callback */ int channels; /* current number of channels */ @@ -138,6 +141,8 @@ struct hdmi_spec { struct snd_array pins; /* struct hdmi_spec_per_pin */ struct hda_pcm *pcm_rec[16]; struct mutex pcm_lock; + /* pcm_bitmap means which pcms have been assigned to pins*/ + unsigned long pcm_bitmap; int pcm_used; /* counter of pcm_rec[] */ unsigned int channels_max; /* max over all cvts */ @@ -1671,6 +1676,60 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) return 0; } +static int hdmi_find_pcm_slot(struct hdmi_spec *spec, + struct hdmi_spec_per_pin *per_pin) +{ + int i; + + /* try the prefer PCM */ + if (!test_bit(per_pin->pin_nid_idx, &spec->pcm_bitmap)) + return per_pin->pin_nid_idx; + + /* have a second try; check the "reserved area" over num_pins */ + for (i = spec->num_pins; i < spec->pcm_used; i++) { + if (!test_bit(i, &spec->pcm_bitmap)) + return i; + } + + /* the last try; check the empty slots in pins */ + for (i = 0; i < spec->num_pins; i++) { + if (!test_bit(i, &spec->pcm_bitmap)) + return i; + } + return -EBUSY; +} + +static void hdmi_attach_hda_pcm(struct hdmi_spec *spec, + struct hdmi_spec_per_pin *per_pin) +{ + int idx; + + /* pcm already be attached to the pin */ + if (per_pin->pcm) + return; + idx = hdmi_find_pcm_slot(spec, per_pin); + if (idx == -ENODEV) + return; + per_pin->pcm_idx = idx; + per_pin->pcm = spec->pcm_rec[idx]; + set_bit(idx, &spec->pcm_bitmap); +} + +static void hdmi_detach_hda_pcm(struct hdmi_spec *spec, + struct hdmi_spec_per_pin *per_pin) +{ + int idx; + + /* pcm already be detached from the pin */ + if (!per_pin->pcm) + return; + idx = per_pin->pcm_idx; + per_pin->pcm_idx = -1; + per_pin->pcm = NULL; + if (idx >= 0 && idx < spec->pcm_used) + clear_bit(idx, &spec->pcm_bitmap); +} + /* update per_pin ELD from the given new ELD; * setup info frame and notification accordingly */ @@ -1679,9 +1738,17 @@ static void update_eld(struct hda_codec *codec, struct hdmi_eld *eld) { struct hdmi_eld *pin_eld = &per_pin->sink_eld; + struct hdmi_spec *spec = codec->spec; bool old_eld_valid = pin_eld->eld_valid; bool eld_changed; + if (spec->dyn_pcm_assign) { + if (eld->eld_valid) + hdmi_attach_hda_pcm(spec, per_pin); + else + hdmi_detach_hda_pcm(spec, per_pin); + } + if (eld->eld_valid) snd_hdmi_show_eld(codec, &eld->info); @@ -1825,13 +1892,19 @@ static void sync_eld_via_acomp(struct hda_codec *codec, static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { struct hda_codec *codec = per_pin->codec; + struct hdmi_spec *spec = codec->spec; + int ret; + mutex_lock(&spec->pcm_lock); if (codec_has_acomp(codec)) { sync_eld_via_acomp(codec, per_pin); - return false; /* don't call snd_hda_jack_report_sync() */ + ret = false; /* don't call snd_hda_jack_report_sync() */ } else { - return hdmi_present_sense_via_verbs(per_pin, repoll); + ret = hdmi_present_sense_via_verbs(per_pin, repoll); } + mutex_unlock(&spec->pcm_lock); + + return ret; } static void hdmi_repoll_eld(struct work_struct *work) @@ -1875,6 +1948,11 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) per_pin->pin_nid = pin_nid; per_pin->non_pcm = false; + if (spec->dyn_pcm_assign) + per_pin->pcm_idx = -1; + else + per_pin->pcm_idx = pin_idx; + per_pin->pin_nid_idx = pin_idx; err = hdmi_read_pin_conn(codec, pin_idx); if (err < 0) -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: Intel: sst: fix sst_memcpy32 wrong with non-4x bytes issue
by Jie Yang 16 Dec '15

16 Dec '15

sst_memcpy32() only copied bytes/4 32bits, which means it dropped the remaining bytes%4 bytes wrongly. Here add copying those missing bytes, first to a 32bits tmp, and then write the tmp to 32bits iomem. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/common/sst-firmware.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 60652cb..d745a78 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -51,8 +51,22 @@ struct sst_dma { static inline void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) { + u32 tmp = 0; + int i, m, n; + const u8 *src_byte = src; + + m = bytes / 4; + n = bytes % 4; + /* __iowrite32_copy use 32bit size values so divide by 4 */ - __iowrite32_copy((void *)dest, src, bytes/4); + __iowrite32_copy((void *)dest, src, m); + + if (n) { + for (i = 0; i < n; i++) + tmp |= (u32)*(src_byte + m * 4 + i) << (i * 8); + __iowrite32_copy((void *)(dest + m * 4), &tmp, 1); + } + } static void sst_dma_transfer_complete(void *arg) -- 2.5.0

2 1

[alsa-devel] [PATCH 0/4] ALSA: oxfw: localize stuffs of old firewire-speakers module
by Takashi Sakamoto 16 Dec '15

16 Dec '15

Hi, ALSA oxfw module inherits some stuffs from old firewire-speakers module. Currently, the module's common structure has some members for the stuffs. This is not good because these stuffs are not common between all of supported models. It's better to localize them. The purpose of this patchset is to enable the module to have memory block and push model-specific stuffs to each structure. This patchset is a part of my previous RFCv2 (patch 06-09) and a preparation for rest of the patches: [alsa-devel] [RFC][PATCH 00/17 v2] ALSA: oxfw: refactoring and merging scs1x module http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/101558.h… Takashi Sakamoto (4): ALSA: oxfw: enable to keep memory block for model-specific structure ALSA: oxfw: move model-specific members from common structure ALSA: oxfw: move model-specific parameters from common structure ALSA: oxfw: rename a structure so that it means backward compatibility to old drivers sound/firewire/oxfw/oxfw-spkr.c | 96 ++++++++++++++++++++++++++--------------- sound/firewire/oxfw/oxfw.c | 31 +++++++------ sound/firewire/oxfw/oxfw.h | 17 +------- 3 files changed, 79 insertions(+), 65 deletions(-) -- 2.5.0

1 4

[alsa-devel] [PATCH 0/5] ALSA: oxfw: code refactoring for model-dependent quirks
by Takashi Sakamoto 16 Dec '15

16 Dec '15

Hi, ALSA oxfw driver has got some patches to expand its functionalities. As a result, current code includes some inefficiencies and confusions. This patchset is a code refactoring about model-dependent quirks. This patchset is a part of my previous RFCv2 (patch 01-05) and a preparation for the rest of patches: [alsa-devel] [RFC][PATCH 00/17 v2] ALSA: oxfw: refactoring and merging scs1x module http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/101558.h… Takashi Sakamoto (5): ALSA: oxfw: rename a file for control elements so that it's for model-specific ALSA: oxfw: rename local functions for control elements so that they represent as local ALSA: oxfw: change function prototype for AV/C Audio Subunit command ALSA: oxfw: reuse driver entry to detect quirk ALSA: oxfw: gather model-dependent conditions to a function sound/firewire/oxfw/Makefile | 4 +- .../firewire/oxfw/{oxfw-control.c => oxfw-spkr.c} | 80 ++++++++++++---------- sound/firewire/oxfw/oxfw.c | 59 +++++++++------- sound/firewire/oxfw/oxfw.h | 6 +- 4 files changed, 86 insertions(+), 63 deletions(-) rename sound/firewire/oxfw/{oxfw-control.c => oxfw-spkr.c} (71%) -- 2.5.0

2 8

[alsa-devel] [PATCH] ASoC: Intel: sst: fix the IRQ locked issue
by Jie Yang 15 Dec '15

15 Dec '15

From: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> If driver received a message that it can't handle, it won't clear the corresponding bit and unmask interrupt, this may lock the IRQ and DSP can't send message anymore. To fix the issue, we should Always update IMRX after IPC. Here we always clear the DONE/BUSY bit and unmask the IRQ source, even when IPC failures have occurred previously. 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> --- sound/soc/intel/haswell/sst-haswell-ipc.c | 31 +++++++++++++------------------ 1 file changed, 13 insertions(+), 18 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index b27f25f..ac60f13 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -778,7 +778,6 @@ static irqreturn_t hsw_irq_thread(int irq, void *context) struct sst_hsw *hsw = sst_dsp_get_thread_context(sst); struct sst_generic_ipc *ipc = &hsw->ipc; u32 ipcx, ipcd; - int handled; unsigned long flags; spin_lock_irqsave(&sst->spinlock, flags); @@ -790,34 +789,30 @@ static irqreturn_t hsw_irq_thread(int irq, void *context) if (ipcx & SST_IPCX_DONE) { /* Handle Immediate reply from DSP Core */ - handled = hsw_process_reply(hsw, ipcx); + hsw_process_reply(hsw, ipcx); - if (handled > 0) { - /* clear DONE bit - tell DSP we have completed */ - sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX, - SST_IPCX_DONE, 0); + /* clear DONE bit - tell DSP we have completed */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX, + SST_IPCX_DONE, 0); - /* unmask Done interrupt */ - sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, - SST_IMRX_DONE, 0); - } + /* unmask Done interrupt */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, + SST_IMRX_DONE, 0); } /* new message from DSP */ if (ipcd & SST_IPCD_BUSY) { /* Handle Notification and Delayed reply from DSP Core */ - handled = hsw_process_notification(hsw); + hsw_process_notification(hsw); /* clear BUSY bit and set DONE bit - accept new messages */ - if (handled > 0) { - sst_dsp_shim_update_bits_unlocked(sst, SST_IPCD, - SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); + sst_dsp_shim_update_bits_unlocked(sst, SST_IPCD, + SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); - /* unmask busy interrupt */ - sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, - SST_IMRX_BUSY, 0); - } + /* unmask busy interrupt */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, + SST_IMRX_BUSY, 0); } spin_unlock_irqrestore(&sst->spinlock, flags); -- 2.5.0

2 1

[alsa-devel] [PATCH 1/3] rcar: ctu: Avoid use of ret uninitialised
by Mark Brown 15 Dec '15

15 Dec '15

We use ret as the return value from the rsnd_ctu_probe() but if there are no child nodes and no errors then we will never initialize ret leading to build warnings. Ensure ret is initialized before we iterate over the child nodes to avoid this. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Perhaps this should be an error instead? It's not like it's a sensible configuration... sound/soc/sh/rcar/ctu.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 7c1e190cd389..d53a225d19e9 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -111,6 +111,7 @@ int rsnd_ctu_probe(struct rsnd_priv *priv) priv->ctu = ctu; i = 0; + ret = 0; for_each_child_of_node(node, np) { ctu = rsnd_ctu_get(priv, i); -- 2.6.2

2 7

[alsa-devel] [RFC][PATCH 00/10 v2] ALSA: dice: stabiliza packet streaming
by Takashi Sakamoto 15 Dec '15

15 Dec '15

This patchset updates my previous one: [alsa-devel] [RFC][PATCH 0/8] ALSA: dice: constrain PCM substreams to current sampling transfer frequency http://mailman.alsa-project.org/pipermail/alsa-devel/2015-November/100670.h… I hope more users to test this patchset in this developing cycle. [Background] I've frequently got private messages from owners of dice-based models since below patchset was merged to mainline: [alsa-devel] [PATCH 00/15 v5] ALSA: Enhancement for existed FireWire drivers http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085142.h… Some of them addressed that ALSA dice driver fails to handle their models, while their experiences are not the same. The summary: * ETIMEOUT at starting streaming * detecting packet discontinuity at starting streaming * ETIMEOUT at probing snd-dice * no sound even if starting streaming This patchset is my solutions for these issues. It consists of four parts; [Patch 01] ETIMEDOUT at starting streaming means that ALSA dice driver has never received dice notification within current timeout (200 msec). This patch expands the timeout up to 2 sec. There's some disadvantages of this change in usage of ALSA PCM API, see commit comment. [Patch 02] Packet discontinuity at starting streaming means that the device fail to get constant clock signals from phase lock loop (PLL) circuit. In this case, ALSA dice driver successfully starts streaming and receives AMDTP packet from the device, therefore any dice notifications are already transferred. It's reasonable to judge that the device has a quirk to transfer the notification before set its PLL circuit steadily. This patch continue to check a register so that the device get phase lock after receiving dice notification. [Patch 03-04] When connected to IEEE 1394 bus, some models generate bus reset after appearing on bus topology. Usually, boot processing of Dice based models consists of three stages; boot loader (RedBoot) is loaded, it loads eCos, the operating system executes vendor's configurations for hardware initialization. It's reasonable to assume that bus reset occurs at each of these stages. Furthermore, I observed that ImpactTwin (TC Electronic produced) changes some registers before/after ALSA dice driver is probed with my RFC v1. This means that the hardware initialization requires a bit time. These patches postpone unit probe processing after continuous bus resets. These patches have advantages to start communication to the device after the hardware initialization. These patches may improve handling Dice II based models such as ImpactTwin and IO26 (Alesis produces). [Patch 05-10] When seeing sound/firewire/dice/dice-interface.h, Dice interface gives no ways for drivers to get all formats of AMDTP stream. Current ALSA dice driver manage to generate 'format cache' at unit probing processing, while this requires to change the state of actual clock on devices. This idea is not good because of ignoring hardware design. This idea is not good in a case that users save their favorite configuration to the device. Furthermore, this is not also good in a case that the stream format is not decided according to sampling transfer frequency only, i.e. data format of digital interface is also dominant to the stream format. These patches add limitation to constrain PCM substream at current sampling transfer frequency. As a result, userspace applications are allowed to request according to current sampling transfer frequency. When users change the frequency, they should use userspace applications with fw character devices to set it in advance. ALSA dice driver gives no way to achieve it in describes reasons. Although ALSA dice driver goes backward with these patchset, we don't beat any actual hardware designs. [Changes from RFC v1] * Patch 03-04 are newly added. This enables to handle bus reset at unit probe processing, and drops one of reasons to constrain PCM substreams at current sampling transfer frequency. * Minor arrangements Takashi Sakamoto (10): ALSA: dice: expand timeout to wait for Dice notification ALSA: dice: wait for ensuring phase lock ALSA: dice: split subaddress check from category check ALSA: dice: postpone probe processing ALSA: dice: limit to current sampling transfer frequency ALSA: dice: limit stream to current sampling transfer frequency. ALSA: dice: add MIDI ports according to current number of MIDI substreams ALSA: dice: get the number of MBLA data channel at opening PCM substream ALSA: dice: purge generating channel cache ALSA: dice: ensure phase lock before starting streaming sound/firewire/dice/dice-midi.c | 25 +++- sound/firewire/dice/dice-pcm.c | 201 ++++++++------------------- sound/firewire/dice/dice-stream.c | 84 ++++++++---- sound/firewire/dice/dice-transaction.c | 158 ++++++++++----------- sound/firewire/dice/dice.c | 242 ++++++++++++--------------------- sound/firewire/dice/dice.h | 11 +- 6 files changed, 293 insertions(+), 428 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH 0/3] Yet more noise fixes for Thinkpad
by Takashi Iwai 15 Dec '15

15 Dec '15

Hi, here is a series of small patches to reduce still existing crackling noises on Thinkpad with Realtek codecs. Takashi Iwai (3): ALSA: hda - Apply click noise workaround for Thinkpads generically ALSA: hda - Set codec to D3 at reboot/shutdown on Thinkpads ALSA: hda - Add a fixup for Thinkpad X1 Carbon 2nd sound/pci/hda/patch_realtek.c | 39 +++++++++++++++++++++++++-------------- 1 file changed, 25 insertions(+), 14 deletions(-) -- 2.6.4

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[alsa-devel] [PATCH] ALSA: hda - Fix headphone mic input on a few Dell ALC293 machines
by David Henningsson 15 Dec '15

15 Dec '15

These laptops support both headphone, headset and mic modes for the 3.5mm jack. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1526330 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8dd2ac1..b745a72 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4606,6 +4606,7 @@ enum { ALC288_FIXUP_DISABLE_AAMIX, ALC292_FIXUP_DELL_E7X, ALC292_FIXUP_DISABLE_AAMIX, + ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK, ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, @@ -5169,6 +5170,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE }, + [ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE + }, [ALC292_FIXUP_DELL_E7X] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_dell_xps13, @@ -5247,11 +5254,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC292_FIXUP_DISABLE_AAMIX), - SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC292_FIXUP_DISABLE_AAMIX), - SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX), - SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC292_FIXUP_DISABLE_AAMIX), - SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC292_FIXUP_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Clean up the code to check bdl_pos_adj option
by Takashi Iwai 15 Dec '15

15 Dec '15

Just a minor cleanup; instead of passing an array, pass the assigned bdl_pos_adj option value directory in struct azx. Also split the code to get the default bdl_pos_adj value for the change that will follow after this. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.c | 3 +-- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 29 ++++++++++++++++------------- 3 files changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 22dbfa563919..10c77074b4dc 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1050,8 +1050,7 @@ int azx_bus_init(struct azx *chip, const char *model, if (chip->get_position[0] != azx_get_pos_lpib || chip->get_position[1] != azx_get_pos_lpib) bus->core.use_posbuf = true; - if (chip->bdl_pos_adj) - bus->core.bdl_pos_adj = chip->bdl_pos_adj[chip->dev_index]; + bus->core.bdl_pos_adj = chip->bdl_pos_adj; if (chip->driver_caps & AZX_DCAPS_CORBRP_SELF_CLEAR) bus->core.corbrp_self_clear = true; diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index c1d28a657f19..a32ec9004edd 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -147,7 +147,7 @@ struct azx { #endif /* flags */ - const int *bdl_pos_adj; + int bdl_pos_adj; int poll_count; unsigned int running:1; unsigned int single_cmd:1; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fe9bef339cea..a17bf0467edc 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -632,7 +632,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) if (wallclk < (azx_dev->core.period_wallclk * 5) / 4 && pos % azx_dev->core.period_bytes > azx_dev->core.period_bytes / 2) /* NG - it's below the first next period boundary */ - return chip->bdl_pos_adj[chip->dev_index] ? 0 : -1; + return chip->bdl_pos_adj ? 0 : -1; azx_dev->core.start_wallclk += wallclk; return 1; /* OK, it's fine */ } @@ -1488,6 +1488,17 @@ static void azx_probe_work(struct work_struct *work) azx_probe_continue(&hda->chip); } +static int default_bdl_pos_adj(struct azx *chip) +{ + switch (chip->driver_type) { + case AZX_DRIVER_ICH: + case AZX_DRIVER_PCH: + return 1; + default: + return 32; + } +} + /* * constructor */ @@ -1541,18 +1552,10 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, chip->single_cmd = single_cmd; azx_check_snoop_available(chip); - if (bdl_pos_adj[dev] < 0) { - switch (chip->driver_type) { - case AZX_DRIVER_ICH: - case AZX_DRIVER_PCH: - bdl_pos_adj[dev] = 1; - break; - default: - bdl_pos_adj[dev] = 32; - break; - } - } - chip->bdl_pos_adj = bdl_pos_adj; + if (bdl_pos_adj[dev] < 0) + chip->bdl_pos_adj = default_bdl_pos_adj(chip); + else + chip->bdl_pos_adj = bdl_pos_adj[dev]; err = azx_bus_init(chip, model[dev], &pci_hda_io_ops); if (err < 0) { -- 2.6.4

1 1

[alsa-devel] [PATCH] ASoC: Atmel: ClassD: unregister codec when error occurs
by Songjun Wu 15 Dec '15

15 Dec '15

Add code to unregister codec in probe function, when the error occurs after the codec is registered. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> --- sound/soc/atmel/atmel-classd.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index dca6141..6107de9 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -640,8 +640,10 @@ static int atmel_classd_probe(struct platform_device *pdev) /* register sound card */ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); - if (!card) - return -ENOMEM; + if (!card) { + ret = -ENOMEM; + goto unregister_codec; + } snd_soc_card_set_drvdata(card, dd); platform_set_drvdata(pdev, card); @@ -649,16 +651,20 @@ static int atmel_classd_probe(struct platform_device *pdev) ret = atmel_classd_asoc_card_init(dev, card); if (ret) { dev_err(dev, "failed to init sound card\n"); - return ret; + goto unregister_codec; } ret = devm_snd_soc_register_card(dev, card); if (ret) { dev_err(dev, "failed to register sound card: %d\n", ret); - return ret; + goto unregister_codec; } return 0; + +unregister_codec: + snd_soc_unregister_codec(dev); + return ret; } static int atmel_classd_remove(struct platform_device *pdev) -- 1.7.9.5

3 2

[alsa-devel] Can I stop sound from driver as system error ?
by Kuninori Morimoto 15 Dec '15

15 Dec '15

Hi I wonder can I stop sound from driver side (not from user-land) as system error during playback or capture ? How to do it if we can ? Best regards --- Kuninori Morimoto

2 3

[alsa-devel] Intel Cherry Trail -- No sound in linux
by Zhihe Tu 14 Dec '15

14 Dec '15

Hi All, I'm trying to make the CherryTrail tablet works in ubuntu14.04.3, there's no sound at all, but it works in win8 & android. ... [ 5.463371] cht-bsw-rt5672 cht-bsw-rt5672: snd-soc-dummy-dai <-> media-cpu-dai mapping ok [ 5.463406] compress asoc: snd-soc-dummy-dai <-> compress-cpu-dai mapping ok [ 5.475786] cht-bsw-rt5672 cht-bsw-rt5672: rt5670-aif1 <-> ssp2-port mapping ok [ 5.507355] hid-generic 0003:413C:2003.0002: input,hidraw1: USB HID v1.10 Keyboard [Dell Dell USB Keyboard] on usb-0000:00:14.0-4.2/input0 [ 5.695892] Console: switching to colour frame buffer device 170x48 [ 5.712458] i915_bpo 0000:00:02.0: fb0: inteldrmfb frame buffer device [ 5.712461] i915_bpo 0000:00:02.0: registered panic notifier [ 5.725983] cfg80211: Calling CRDA to update world regulatory domain [ 5.872710] scsi 0:0:0:0: Direct-Access ASMT 2115 0 PQ: 0 ANSI: 6 [ 5.873522] sd 0:0:0:0: Attached scsi generic sg0 type 0 [ 5.875671] sd 0:0:0:0: [sda] Attached SCSI removable disk [ 5.907038] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 0 doesn't have DAPM widget!!! [ 5.922073] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 1 doesn't have DAPM widget!!! [ 5.935876] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 2 doesn't have DAPM widget!!! [ 5.949941] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 3 doesn't have DAPM widget!!! [ 5.968241] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_0 doesn't have DAPM widget!!! [ 5.968267] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_1 doesn't have DAPM widget!!! [ 5.968292] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_0 doesn't have DAPM widget!!! [ 5.968318] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_1 doesn't have DAPM widget!!! ... [ 6.617144] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 0 doesn't have DAPM widget!!! [ 6.632519] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 1 doesn't have DAPM widget!!! [ 6.647501] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 2 doesn't have DAPM widget!!! [ 6.662176] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 3 doesn't have DAPM widget!!! [ 6.677049] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_0 doesn't have DAPM widget!!! [ 6.692739] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_1 doesn't have DAPM widget!!! [ 6.692765] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_0 doesn't have DAPM widget!!! [ 6.692792] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_1 doesn't have DAPM widget!!! [ 7.541669] init: plymouth-upstart-bridge main process ended, respawning [ 9.804555] Audio Port: ASoC: no backend DAIs enabled for Audio Port [ 10.139952] Audio Port: ASoC: no backend DAIs enabled for Audio Port [ 10.147834] Audio Port: ASoC: no backend DAIs enabled for Audio Port [ 10.159472] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 10.162179] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 10.163874] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 10.164488] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 10.165994] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 10.166626] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 10.168438] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 10.169549] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 10.171607] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 10.189856] intel_sst_acpi 808622A8:00: Alloc for str 1 pipe 0x90 [ 10.204244] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 10.205026] intel_sst_acpi 808622A8:00: Free for str 1 pipe 0x90 [ 10.215195] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 12.182265] intel_sst_acpi 808622A8:00: Alloc for str 1 pipe 0x90 [ 13.181361] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x2 [ 13.181374] intel_sst_acpi 808622A8:00: fw returned err -16 [ 14.181735] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 14.181764] intel_sst_acpi 808622A8:00: fw returned err -16 [ 15.181611] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 15.181641] intel_sst_acpi 808622A8:00: fw returned err -16 [ 16.181346] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 16.181376] intel_sst_acpi 808622A8:00: fw returned err -16 [ 16.210605] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 17.209115] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 17.209144] intel_sst_acpi 808622A8:00: fw returned err -16 [ 17.209183] intel_sst_acpi 808622A8:00: free stream returned err -19 [ 18.262035] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 18.262053] intel_sst_acpi 808622A8:00: fw returned err -16 [ 19.260719] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 19.260749] intel_sst_acpi 808622A8:00: fw returned err -16 [ 19.313903] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 19.445415] intel_sst_acpi 808622A8:00: Alloc for str 1 pipe 0x90 [ 19.445862] intel_sst_acpi 808622A8:00: FW sent error response 0x40034 [ 19.445883] intel_sst_acpi 808622A8:00: FW alloc failed ret -4 [ 19.445888] intel_sst_acpi 808622A8:00: sst_get_stream returned err -5 [ 19.445893] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 19.445898] Audio Port: ASoC: prepare FE Audio Port failed [ 20.444493] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 20.444522] intel_sst_acpi 808622A8:00: fw returned err -16 [ 21.444427] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 21.444456] intel_sst_acpi 808622A8:00: fw returned err -16 [ 22.444126] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 22.444155] intel_sst_acpi 808622A8:00: fw returned err -16 [ 23.443906] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 23.443934] intel_sst_acpi 808622A8:00: fw returned err -16 [ 23.474492] Audio Port: sst: error code = -22 [ 23.474504] snd_pcm_update_hw_ptr0: 123 callbacks suppressed [ 23.474512] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.474589] Audio Port: sst: error code = -22 [ 23.474596] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.474612] Audio Port: sst: error code = -22 [ 23.474617] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.741085] Audio Port: sst: error code = -22 [ 23.741104] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.741140] Audio Port: sst: error code = -22 [ 23.741147] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.741206] Audio Port: sst: error code = -22 [ 23.741213] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.741233] Audio Port: sst: error code = -22 [ 23.741240] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.741350] Audio Port: sst: error code = -22 [ 23.741358] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.741379] Audio Port: sst: error code = -22 [ 23.741385] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.741427] Audio Port: sst: error code = -22 [ 23.741434] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.741455] Audio Port: sst: error code = -22 [ 23.785062] Audio Port: sst: error code = -22 [ 23.785108] Audio Port: sst: error code = -22 [ 23.785185] Audio Port: sst: error code = -22 [ 23.785210] Audio Port: sst: error code = -22 [ 23.785343] Audio Port: sst: error code = -22 [ 23.785367] Audio Port: sst: error code = -22 [ 23.785416] Audio Port: sst: error code = -22 [ 28.496573] snd_pcm_update_hw_ptr0: 13 callbacks suppressed [ 28.496582] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 28.496707] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 29.494259] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 29.494269] intel_sst_acpi 808622A8:00: fw returned err -16 [ 29.494284] intel_sst_acpi 808622A8:00: free stream returned err -19 [ 30.562054] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 30.562065] intel_sst_acpi 808622A8:00: fw returned err -16 [ 31.561892] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 31.561904] intel_sst_acpi 808622A8:00: fw returned err -16 [ 906.106300] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 906.249375] intel_sst_acpi 808622A8:00: Alloc for str 1 pipe 0x90 [ 907.248864] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x2 [ 907.248894] intel_sst_acpi 808622A8:00: fw returned err -16 [ 908.248393] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 908.248421] intel_sst_acpi 808622A8:00: fw returned err -16 [ 909.248611] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 909.248641] intel_sst_acpi 808622A8:00: fw returned err -16 [ 910.248108] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 910.248137] intel_sst_acpi 808622A8:00: fw returned err -16 [ 1053.563879] isofs_fill_super: bread failed, dev=sdb1, iso_blknum=16, block=32 [ 1185.717932] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 1186.716247] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 1186.716277] intel_sst_acpi 808622A8:00: fw returned err -16 [ 1186.716315] intel_sst_acpi 808622A8:00: free stream returned err -19 [ 1187.783992] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 1187.784021] intel_sst_acpi 808622A8:00: fw returned err -16 [ 1188.783551] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 1188.783584] intel_sst_acpi 808622A8:00: fw returned err -16 [ 1188.821911] rt5670 i2c-10EC5670:00: Sysclk is 0Hz and clock id is 2 [ 1207.389279] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 1207.404620] rt5670 i2c-10EC5670:00: Sysclk is 24576000Hz and clock id is 1 [ 1207.404646] rt5670 i2c-10EC5670:00: bclk is 1536000Hz and lrck is 48000Hz [ 1207.404660] rt5670 i2c-10EC5670:00: bclk_ms is 0 and pre_div is 1 for iis 1 [ 1207.533606] intel_sst_acpi 808622A8:00: Alloc for str 1 pipe 0x90 [ 1208.535662] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x2 [ 1208.535691] intel_sst_acpi 808622A8:00: fw returned err -16 [ 1209.535853] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 1209.535883] intel_sst_acpi 808622A8:00: fw returned err -16 [ 1210.535333] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 1210.535361] intel_sst_acpi 808622A8:00: fw returned err -16 [ 1211.535576] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 1211.535606] intel_sst_acpi 808622A8:00: fw returned err -16 [ 1286.976450] usb 1-4.1: USB disconnect, device number 8 [11996.101738] r8169 0000:01:00.0 eth0: link up [11996.101780] IPv6: ADDRCONF(NETDEV_CHANGE): eth0: link becomes ready [12019.249846] [drm:intel_pipe_update_end [i915_bpo]] *ERROR* Atomic update failure on pipe A (start=93814 end=93815) [12157.754218] [drm:intel_pipe_update_end [i915_bpo]] *ERROR* Atomic update failure on pipe A (start=102079 end=102080) [12194.085543] [drm:intel_pipe_update_end [i915_bpo]] *ERROR* Atomic update failure on pipe A (start=104247 end=104248) [13567.332608] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [13568.332857] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [13568.332886] intel_sst_acpi 808622A8:00: fw returned err -16 [13568.332924] intel_sst_acpi 808622A8:00: free stream returned err -19 [13569.400219] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [13569.400247] intel_sst_acpi 808622A8:00: fw returned err -16 [13570.399826] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [13570.399837] intel_sst_acpi 808622A8:00: fw returned err -16 [13570.437188] rt5670 i2c-10EC5670:00: Sysclk is 0Hz and clock id is 2 >From /sys/kernel/debug/asoc/cherrytrailcraudio/, some necessary widgets are all power on, from media1_in to ssp2 Tx end with Headphone. I really don't know how to fix the sst fw error, any anyone give a look at this issue? ................ Thanks Tu __________ Information from ESET Smart Security, version of virus signature database 4468 (20090929) __________ The message was checked by ESET Smart Security. http://www.eset.com

3 7

[alsa-devel] [PATCH v3] ASoC: cs53l30: Add support for Cirrus Logic CS53L30
by me＠mx0a-001ae601.pphosted.com 14 Dec '15

14 Dec '15

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- .../devicetree/bindings/sound/cs53l30.txt | 21 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs53l30.c | 933 +++++++++++++++++++++ sound/soc/codecs/cs53l30.h | 272 ++++++ 5 files changed, 1233 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs53l30.txt create mode 100644 sound/soc/codecs/cs53l30.c create mode 100644 sound/soc/codecs/cs53l30.h diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt new file mode 100644 index 0000000..a2dff95 +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -0,0 +1,21 @@ +CS53L30 audio CODEC + +Required properties: + + - compatible : "cirrus,cs53l30" + + - reg : the I2C address of the device + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + + +Example: + +codec: cs53l30@48 { + compatible = "cirrus,cs53l30"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b3317da..f59aa4b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -59,6 +59,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CS4349 if I2C + select SND_SOC_CS53L30 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -444,6 +445,10 @@ config SND_SOC_CS4349 tristate "Cirrus Logic CS4349 CODEC" depends on I2C +config SND_SOC_CS53L30 + tristate "Cirrus Logic CS53L30 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cdadb9b..4f1ab1d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -51,6 +51,7 @@ snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cs4349-objs := cs4349.o +snd-soc-cs53l30-objs := cs53l30.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -250,6 +251,7 @@ obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o +obj-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c new file mode 100644 index 0000000..3d5cac9 +++ b/sound/soc/codecs/cs53l30.c @@ -0,0 +1,933 @@ +/* + * cs53l30.c -- CS53l30 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * Tim Howe <Tim.Howe(a)cirrus.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/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.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 "cs53l30.h" + +struct cs53l30_private { + struct regmap *regmap; + struct gpio_desc *reset_gpio; + u8 asp_config_ctl; + u32 mclk; +}; + +static const struct reg_default cs53l30_reg_defaults[] = { + { CS53L30_PWRCTL, THMS_PDN }, + { CS53L30_MCLKCTL, MCLK_DIV_DFLT }, + { CS53L30_INT_SR_CTL, INTERNAL_FS_DFLT }, + { CS53L30_MICBIAS_CTL, MIC_BIAS_DFLT }, + { CS53L30_ASPCFG_CTL, ASP_RATE_48K }, + { CS53L30_ASP1_CTL, ASP1_TDM_PDN }, + { CS53L30_ASP1_TDMTX_CTL1, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL2, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL3, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL4, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_EN1, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN2, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN3, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN4, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN5, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN6, ASP_TX_DISABLED }, + { CS53L30_ASP2_CTL, ASP2_CTRL_DFLT }, + { CS53L30_SFT_RAMP, SFT_RMP_DFLT }, + { CS53L30_LRCLK_CTL1, LRCK_CTLX_DFLT }, + { CS53L30_LRCLK_CTL2, LRCK_CTLX_DFLT }, + { CS53L30_MUTEP_CTL1, MUTEP_CTRL1_DFLT }, + { CS53L30_MUTEP_CTL2, MUTE_PDN_ULP }, + { CS53L30_INBIAS_CTL1, INBIAS_CTLX_DFLT }, + { CS53L30_INBIAS_CTL2, INBIAS_CTLX_DFLT }, + { CS53L30_DMIC1_STR_CTL, DMIC1_STEREO_DFLT }, + { CS53L30_DMIC2_STR_CTL, DMIC2_STEREO_DFLT }, + { CS53L30_ADCDMIC1_CTL1, ADC1_ON_CH_AB_IN }, + { CS53L30_ADCDMIC1_CTL2, ADC1_DMIC1_CTL2_DFLT }, + { CS53L30_ADC1_CTL3, ADC1_HPF_EN }, + { CS53L30_ADC1_NG_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1A_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1B_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1A_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1B_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADCDMIC2_CTL1, ADC2_ON_CH_AB_IN }, + { CS53L30_ADCDMIC2_CTL2, ADCX_ZERO_DFLT }, + { CS53L30_ADC2_CTL3, ADC2_HPF_EN }, + { CS53L30_ADC2_NG_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2A_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2B_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2A_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2B_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_INT_MASK, DEVICE_INT_MASK }, +}; + +static bool cs53l30_volatile_register(struct device *dev, unsigned int reg) +{ + if (reg == CS53L30_IS) + return true; + else + return false; +} + +static bool cs53l30_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS53L30_DEVID_AB: + case CS53L30_DEVID_CD: + case CS53L30_DEVID_E: + case CS53L30_REVID: + case CS53L30_PWRCTL: + case CS53L30_MCLKCTL: + case CS53L30_INT_SR_CTL: + case CS53L30_MICBIAS_CTL: + case CS53L30_ASPCFG_CTL: + case CS53L30_ASP1_CTL: + case CS53L30_ASP1_TDMTX_CTL1: + case CS53L30_ASP1_TDMTX_CTL2: + case CS53L30_ASP1_TDMTX_CTL3: + case CS53L30_ASP1_TDMTX_CTL4: + case CS53L30_ASP1_TDMTX_EN1: + case CS53L30_ASP1_TDMTX_EN2: + case CS53L30_ASP1_TDMTX_EN3: + case CS53L30_ASP1_TDMTX_EN4: + case CS53L30_ASP1_TDMTX_EN5: + case CS53L30_ASP1_TDMTX_EN6: + case CS53L30_ASP2_CTL: + case CS53L30_SFT_RAMP: + case CS53L30_LRCLK_CTL1: + case CS53L30_LRCLK_CTL2: + case CS53L30_MUTEP_CTL1: + case CS53L30_MUTEP_CTL2: + case CS53L30_INBIAS_CTL1: + case CS53L30_INBIAS_CTL2: + case CS53L30_DMIC1_STR_CTL: + case CS53L30_DMIC2_STR_CTL: + case CS53L30_ADCDMIC1_CTL1: + case CS53L30_ADCDMIC1_CTL2: + case CS53L30_ADC1_CTL3: + case CS53L30_ADC1_NG_CTL: + case CS53L30_ADC1A_AFE_CTL: + case CS53L30_ADC1B_AFE_CTL: + case CS53L30_ADC1A_DIG_VOL: + case CS53L30_ADC1B_DIG_VOL: + case CS53L30_ADCDMIC2_CTL1: + case CS53L30_ADCDMIC2_CTL2: + case CS53L30_ADC2_CTL3: + case CS53L30_ADC2_NG_CTL: + case CS53L30_ADC2A_AFE_CTL: + case CS53L30_ADC2B_AFE_CTL: + case CS53L30_ADC2A_DIG_VOL: + case CS53L30_ADC2B_DIG_VOL: + case CS53L30_INT_MASK: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2000, 0); +static DECLARE_TLV_DB_SCALE(adc_ng_boost_tlv, 0, 3000, 0); +static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); + +static DECLARE_TLV_DB_SCALE(dig_tlv, -9600, 100, 1); + +static const char * const input1_sel_text[] = { "DMIC1 On AB In", + "DMIC1 On A In", "DMIC1 On B In", "ADC1 On AB In", "ADC1 On A In", + "ADC1 On B In", "DMIC1 Off ADC1 Off", }; + +unsigned int const input1_sel_values[] = { DMIC1_ON_CH_AB_IN, DMIC1_ON_CH_A_IN, + DMIC1_ON_CH_B_IN, ADC1_ON_CH_AB_IN, ADC1_ON_CH_A_IN, ADC1_ON_CH_B_IN, + DMIC1_OFF_ADC1_OFF, }; + +static const char * const input2_sel_text[] = { "DMIC2 On AB In", + "DMIC2 On A In", "DMIC2 On B In", "ADC2 On AB In", "ADC2 On A In", + "ADC2 On B In", "DMIC2 Off ADC2 Off", }; + +unsigned int const input2_sel_values[] = { DMIC2_ON_CH_AB_IN, DMIC2_ON_CH_A_IN, + DMIC2_ON_CH_B_IN, ADC2_ON_CH_AB_IN, ADC2_ON_CH_A_IN, ADC2_ON_CH_B_IN, + DMIC2_OFF_ADC2_OFF, }; + +static const char * const input1_route_sel_text[] = { "ADC1_SEL", + "DMIC1_SEL" }; + +static const struct soc_enum input1_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input1_route_sel_text), input1_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input1_sel_enum, CS53L30_ADCDMIC1_CTL1, 0, + ADC1_DMIC1_PDN_MASK, input1_sel_text, input1_sel_values); + +static const struct snd_kcontrol_new input1_route_sel_mux = + SOC_DAPM_ENUM("Input 1 Route", input1_route_sel_enum); + +static const char * const input2_route_sel_text[] = { "ADC2_SEL", + "DMIC2_SEL" }; + +/* Note: CS53L30_ADCDMIC1_CTL1 CH_TYPE controls inputs 1 and 2 */ +static const struct soc_enum input2_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input2_route_sel_text), input2_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input2_sel_enum, CS53L30_ADCDMIC2_CTL1, 0, + ADC1_DMIC1_PDN_MASK, input2_sel_text, input2_sel_values); + +static const struct snd_kcontrol_new input2_route_sel_mux = + SOC_DAPM_ENUM("Input 2 Route", input2_route_sel_enum); + +/* + * TB = 6144*(MCLK(int) scaling factor)/MCLK(internal) + * TB - Time base + * NOTE: If MCLK_INT_SCALE = 0, then TB=1 + */ +static const char * const cs53l30_ng_delay_text[] = { + "TB*50ms", "TB*100ms", "TB*150ms", "TB*200ms" }; + +static const struct soc_enum adc1_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, 0, + ARRAY_SIZE(cs53l30_ng_delay_text), cs53l30_ng_delay_text); + +static const struct soc_enum adc2_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, 0, + ARRAY_SIZE(cs53l30_ng_delay_text), cs53l30_ng_delay_text); + +/* The noise gate threshold selected will depend on NG Boost */ +static const char * const cs53l30_ng_thres_text[] = { + "-64dB/-34dB", "-66dB/-36dB", "-70dB/-40dB", "-73dB/-43dB", + "-76dB/-46dB", "-82dB/-52dB", "-58dB", "-64dB"}; + +static const struct soc_enum adc1_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, 2, + ARRAY_SIZE(cs53l30_ng_thres_text), cs53l30_ng_thres_text); + +static const struct soc_enum adc2_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, 2, + ARRAY_SIZE(cs53l30_ng_thres_text), cs53l30_ng_thres_text); + +/* ADC Preamp gain select */ +static const char * const cs53l30_preamp_gain_sel_text[] = { + "0dB", "10dB", "20dB"}; + +static const struct soc_enum adc1a_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1A_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc1b_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1B_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc2a_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2A_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc2b_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2B_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +/* Set MIC Bias Voltage Control */ +static const char * const cs53l30_micbias_text[] = { + "HiZ", "1.8V", "2.75V"}; + +static const struct soc_enum micbias_enum = + SOC_ENUM_SINGLE(CS53L30_MICBIAS_CTL, 0, + ARRAY_SIZE(cs53l30_micbias_text), + cs53l30_micbias_text); + +/* Corner frequencies are with an Fs of 48kHz. */ +static const char * const hpf_corner_freq_text[] = { + "1.86Hz", "120Hz", "235Hz", "466Hz"}; + +static const struct soc_enum adc1_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_CTL3, 1, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct soc_enum adc2_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_CTL3, 1, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct snd_kcontrol_new cs53l30_snd_controls[] = { + SOC_SINGLE("Digital Soft-Ramp Switch", CS53L30_SFT_RAMP, 5, 1, 0), + SOC_SINGLE("ADC1 Noise Gate Ganging Switch", + CS53L30_ADC1_CTL3, 0, 1, 0), + SOC_SINGLE("ADC2 Noise Gate Ganging Switch", + CS53L30_ADC2_CTL3, 0, 1, 0), + SOC_SINGLE("ADC1A Noise Gate Enable Switch", + CS53L30_ADC1_NG_CTL, 6, 1, 0), + SOC_SINGLE("ADC1B Noise Gate Enable Switch", + CS53L30_ADC1_NG_CTL, 7, 1, 0), + SOC_SINGLE("ADC2A Noise Gate Enable Switch", + CS53L30_ADC2_NG_CTL, 6, 1, 0), + SOC_SINGLE("ADC2B Noise Gate Enable Switch", + CS53L30_ADC2_NG_CTL, 7, 1, 0), + SOC_SINGLE("ADC1 Notch Filter Switch", + CS53L30_ADCDMIC1_CTL2, 7, 1, 1), + SOC_SINGLE("ADC2 Notch Filter Switch", + CS53L30_ADCDMIC2_CTL2, 7, 1, 1), + SOC_SINGLE("ADC1A Invert Switch", + CS53L30_ADCDMIC1_CTL2, 4, 1, 0), + SOC_SINGLE("ADC1B Invert Switch", + CS53L30_ADCDMIC1_CTL2, 5, 1, 0), + SOC_SINGLE("ADC2A Invert Switch", + CS53L30_ADCDMIC2_CTL2, 4, 1, 0), + SOC_SINGLE("ADC2B Invert Switch", + CS53L30_ADCDMIC2_CTL2, 5, 1, 0), + + SOC_SINGLE_TLV("ADC1A Digital Boost Volume", + CS53L30_ADCDMIC1_CTL2, 0, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1B Digital Boost Volume", + CS53L30_ADCDMIC1_CTL2, 1, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2A Digital Boost Volume", + CS53L30_ADCDMIC2_CTL2, 0, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2B Digital Boost Volume", + CS53L30_ADCDMIC2_CTL2, 1, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1 NG Boost Volume", + CS53L30_ADC1_NG_CTL, 5, 1, 0, adc_ng_boost_tlv), + SOC_SINGLE_TLV("ADC2 NG Boost Volume", + CS53L30_ADC2_NG_CTL, 5, 1, 0, adc_ng_boost_tlv), + + SOC_ENUM("Input 1 Channel Select", input1_sel_enum), + SOC_ENUM("Input 2 Channel Select", input2_sel_enum), + + SOC_ENUM("ADC1 HPF Select", adc1_hpf_enum), + SOC_ENUM("ADC2 HPF Select", adc2_hpf_enum), + SOC_ENUM("ADC1 NG Threshold", adc1_ng_thres_enum), + SOC_ENUM("ADC2 NG Threshold", adc2_ng_thres_enum), + SOC_ENUM("ADC1 NG Delay", adc1_ng_delay_enum), + SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), + SOC_ENUM("ADC1A Pre Amp Gain", adc1a_preamp_gain_enum), + SOC_ENUM("ADC1B Pre Amp Gain", adc1b_preamp_gain_enum), + SOC_ENUM("ADC2A Pre Amp Gain", adc2a_preamp_gain_enum), + SOC_ENUM("ADC2B Pre Amp Gain", adc2b_preamp_gain_enum), + SOC_ENUM("Mic Bias Voltage Select", micbias_enum), + + SOC_SINGLE_SX_TLV("ADC1A PGA Volume", + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC1B PGA Volume", + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2A PGA Volume", + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2B PGA Volume", + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + + SOC_SINGLE_SX_TLV("ADC1A Digital Volume", + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC1B Digital Volume", + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2A Digital Volume", + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2B Digital Volume", + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), +}; + +static int cs53l30_asp_sdout_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); + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + ASP1_3ST, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + ASP1_3ST, 1); + break; + default: + pr_err("Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("IN1_DMIC1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3_DMIC2"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS53L30_MICBIAS_CTL, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS53L30_MICBIAS_CTL, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC3 Bias", CS53L30_MICBIAS_CTL, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC4 Bias", CS53L30_MICBIAS_CTL, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT_E("ASP_SDOUT1", NULL, 0, + CS53L30_ASP1_CTL, ASP1_SDOUT_PDN, + 1, cs53l30_asp_sdout_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + SND_SOC_DAPM_AIF_OUT_E("ASP_SDOUT2", NULL, 0, + CS53L30_ASP2_CTL, ASP2_SDOUT_PDN, + 1, cs53l30_asp_sdout_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + + SND_SOC_DAPM_MUX("Input Mux 1", SND_SOC_NOPM, 0, 0, + &input1_route_sel_mux), + SND_SOC_DAPM_MUX("Input Mux 2", SND_SOC_NOPM, 0, 0, + &input2_route_sel_mux), + + SND_SOC_DAPM_ADC("ADC1A", NULL, CS53L30_ADCDMIC1_CTL1, 6, 1), + SND_SOC_DAPM_ADC("ADC1B", NULL, CS53L30_ADCDMIC1_CTL1, 7, 1), + SND_SOC_DAPM_ADC("ADC2A", NULL, CS53L30_ADCDMIC2_CTL1, 6, 1), + SND_SOC_DAPM_ADC("ADC2B", NULL, CS53L30_ADCDMIC2_CTL1, 7, 1), + SND_SOC_DAPM_ADC("DMIC1", NULL, CS53L30_ADCDMIC1_CTL1, 2, 1), + SND_SOC_DAPM_ADC("DMIC2", NULL, CS53L30_ADCDMIC2_CTL1, 2, 1), +}; + +static const struct snd_soc_dapm_route cs53l30_audio_map[] = { + + /* ADC Input Paths */ + {"ADC1A", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "ADC1_SEL", "ADC1A"}, + {"ADC1B", NULL, "IN2"}, + + {"ADC2A", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "ADC2_SEL", "ADC2A"}, + {"ADC2B", NULL, "IN4"}, + + /* MIC Bias Paths */ + {"ADC1A", NULL, "MIC1 Bias"}, + {"ADC1B", NULL, "MIC2 Bias"}, + {"ADC2A", NULL, "MIC3 Bias"}, + {"ADC2B", NULL, "MIC4 Bias"}, + + /* DMIC Paths */ + {"DMIC1", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "DMIC1_SEL", "DMIC1"}, + + {"DMIC2", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "DMIC2_SEL", "DMIC2"}, + + /* Output Paths */ + {"ASP_SDOUT1", NULL, "ADC1A" }, + {"ASP_SDOUT1", NULL, "Input Mux 1"}, + {"ASP_SDOUT1", NULL, "ADC1B"}, + + {"ASP_SDOUT2", NULL, "ADC2A"}, + {"ASP_SDOUT2", NULL, "Input Mux 2"}, + {"ASP_SDOUT2", NULL, "ADC2B"}, + + {"ASP1 Capture", NULL, "ASP_SDOUT1"}, + {"ASP2 Capture", NULL, "ASP_SDOUT2"}, +}; + +struct cs53l30_mclk_div { + u32 mclk; + u32 srate; + u8 asp_rate; + u8 internal_fs_ratio; + u8 mclk_int_scale; +}; + +static struct cs53l30_mclk_div cs53l30_mclk_coeffs[] = { + /* NOTE: Enable MCLK_INT_SCALE to save power. */ + + /* MCLK, Sample Rate, asp_rate, internal_fs_ratio, mclk_int_scale */ + {5644800, 11025, 0x4, 1, 1}, + {5644800, 22050, 0x8, 1, 1}, + {5644800, 44100, 0xC, 1, 1}, + + {6000000, 8000, 0x1, 0, 1}, + {6000000, 11025, 0x2, 0, 1}, + {6000000, 12000, 0x4, 0, 1}, + {6000000, 16000, 0x5, 0, 1}, + {6000000, 22050, 0x6, 0, 1}, + {6000000, 24000, 0x8, 0, 1}, + {6000000, 32000, 0x9, 0, 1}, + {6000000, 44100, 0xA, 0, 1}, + {6000000, 48000, 0xC, 0, 1}, + + {6144000, 8000, 0x1, 1, 1}, + {6144000, 11025, 0x2, 1, 1}, + {6144000, 12000, 0x4, 1, 1}, + {6144000, 16000, 0x5, 1, 1}, + {6144000, 22050, 0x6, 1, 1}, + {6144000, 24000, 0x8, 1, 1}, + {6144000, 32000, 0x9, 1, 1}, + {6144000, 44100, 0xA, 1, 1}, + {6144000, 48000, 0xC, 1, 1}, + + {6400000, 8000, 0x1, 1, 1}, + {6400000, 11025, 0x2, 1, 1}, + {6400000, 12000, 0x4, 1, 1}, + {6400000, 16000, 0x5, 1, 1}, + {6400000, 22050, 0x6, 1, 1}, + {6400000, 24000, 0x8, 1, 1}, + {6400000, 32000, 0x9, 1, 1}, + {6400000, 44100, 0xA, 1, 1}, + {6400000, 48000, 0xC, 1, 1}, +}; + +struct cs53l30_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs53l30_mclkx_div cs53l30_mclkx_coeffs[] = { + {5644800, 1, 0}, + {6000000, 1, 0}, + {6144000, 1, 0}, + {11289600, 2, 1}, + {12288000, 2, 1}, + {12000000, 2, 1}, + {19200000, 3, 2}, +}; + +static int cs53l30_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs53l30_mclkx_coeffs); i++) { + if (cs53l30_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs53l30_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs53l30_mclk_coeffs); i++) { + if (cs53l30_mclk_coeffs[i].mclk == mclk && + cs53l30_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs53l30_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk; + unsigned int mclk_ctl; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs53l30_get_mclkx_coeff(freq); + if (mclkx_coeff < 0) + return mclkx_coeff; + + mclk = cs53l30_mclkx_coeffs[mclkx_coeff].mclkx / + cs53l30_mclkx_coeffs[mclkx_coeff].ratio; + + regmap_read(priv->regmap, CS53L30_MCLKCTL, &mclk_ctl); + mclk_ctl &= ~MCLK_DIV; + mclk_ctl |= cs53l30_mclkx_coeffs[mclkx_coeff].mclkdiv; + + regmap_write(priv->regmap, CS53L30_MCLKCTL, mclk_ctl); + priv->mclk = mclk; + return 0; +} + +static int cs53l30_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int asp_config_ctl; + + regmap_read(priv->regmap, CS53L30_ASPCFG_CTL, &asp_config_ctl); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + asp_config_ctl |= ASP_MS; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + asp_config_ctl &= ~ASP_MS; + break; + + default: + return -EINVAL; + } + + /* Check to see if the SCLK is inverted */ + if (fmt & (SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF)) + asp_config_ctl |= ASP_SCLK_INV; + else + asp_config_ctl &= ~ASP_SCLK_INV; + + priv->asp_config_ctl = asp_config_ctl; + return 0; +} + +static int cs53l30_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + int mclk_coeff; + int srate = params_rate(params); + unsigned int int_sr_ctl, mclk_ctl; + + /* MCLK -> srate */ + mclk_coeff = + cs53l30_get_mclk_coeff(priv->mclk, srate); + if (mclk_coeff < 0) + return -EINVAL; + + regmap_write(priv->regmap, CS53L30_ASPCFG_CTL, priv->asp_config_ctl); + regmap_read(priv->regmap, CS53L30_INT_SR_CTL, &int_sr_ctl); + if (cs53l30_mclk_coeffs[mclk_coeff].internal_fs_ratio) + int_sr_ctl |= INTERNAL_FS_RATIO; + else + int_sr_ctl &= ~INTERNAL_FS_RATIO; + regmap_write(priv->regmap, CS53L30_INT_SR_CTL, int_sr_ctl); + + regmap_read(priv->regmap, CS53L30_MCLKCTL, &mclk_ctl); + if (cs53l30_mclk_coeffs[mclk_coeff].mclk_int_scale) + mclk_ctl |= MCLK_INT_SCALE; + else + mclk_ctl &= ~MCLK_INT_SCALE; + + regmap_write(priv->regmap, CS53L30_MCLKCTL, mclk_ctl); + priv->asp_config_ctl &= ASP_CNFG_MASK; + priv->asp_config_ctl |= (cs53l30_mclk_coeffs[mclk_coeff].asp_rate + & ASP_RATE); + regmap_write(priv->regmap, CS53L30_MCLKCTL, priv->asp_config_ctl); + + return 0; +} + +static int cs53l30_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + int i, inter_max_check; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_LP, 0); + break; + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_OFF) { + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + MCLK_DIS, 0); + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_ULP, 0); + msleep(50); + } else { + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_LP, PDN_LP); + } + break; + + case SND_SOC_BIAS_OFF: + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + PDN_DONE, 0); + /* If digital softramp is set, the amount of time required + * for power down increases and depends on the digital + * volume setting. + */ + + /* Set the max possible time if digsft is set */ + regmap_read(priv->regmap, CS53L30_SFT_RAMP, &reg); + if (reg & DIGSFT) + inter_max_check = PDN_POLL_MAX; + else + inter_max_check = 10; + + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_ULP, PDN_ULP); + msleep(20); /* PDN_DONE will take a min of 20ms to be set.*/ + regmap_read(priv->regmap, CS53L30_IS, &reg); /* Clr status */ + for (i = 0; i < inter_max_check; i++) { + usleep_range(1000, 1000); + msleep(1); + regmap_read(priv->regmap, CS53L30_IS, &reg); + if (reg & PDN_DONE) + break; + } + /* PDN_DONE is set. We now can disable the MCLK */ + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + PDN_DONE, PDN_DONE); + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + MCLK_DIS, MCLK_DIS); + break; + } + dapm->bias_level = level; + + return 0; +} + +static int cs53l30_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + return regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, ASP1_3ST, + (ASP1_3ST_VAL(tristate) & ASP1_3ST)); +} + +unsigned int const cs53l30_src_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static struct snd_pcm_hw_constraint_list src_constraints = { + .count = ARRAY_SIZE(cs53l30_src_rates), + .list = cs53l30_src_rates, +}; + +static int cs53l30_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &src_constraints); + + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS53L30_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + +#define CS53L30_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops cs53l30_ops = { + .startup = cs53l30_pcm_startup, + .hw_params = cs53l30_pcm_hw_params, + .set_fmt = cs53l30_set_dai_fmt, + .set_sysclk = cs53l30_set_sysclk, + .set_tristate = cs53l30_set_tristate, +}; + +static struct snd_soc_dai_driver cs53l30_dai[] = { + { + .name = "cs53l30-asp1", + .id = CS53L30_ASP1, + .capture = { + .stream_name = "ASP1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, + }, + { + .name = "cs53l30-asp2", + .id = CS53L30_ASP2, + .capture = { + .stream_name = "ASP2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, + } +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs53l30 = { + .set_bias_level = cs53l30_set_bias_level, + + .dapm_widgets = cs53l30_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs53l30_dapm_widgets), + .dapm_routes = cs53l30_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs53l30_audio_map), + + .controls = cs53l30_snd_controls, + .num_controls = ARRAY_SIZE(cs53l30_snd_controls), +}; + +static struct regmap_config cs53l30_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS53L30_MAX_REGISTER, + .reg_defaults = cs53l30_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs53l30_reg_defaults), + .volatile_reg = cs53l30_volatile_register, + .readable_reg = cs53l30_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs53l30_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs53l30_private *cs53l30; + int ret = 0; + unsigned int devid = 0; + unsigned int reg; + + cs53l30 = devm_kzalloc(&client->dev, + sizeof(struct cs53l30_private), GFP_KERNEL); + if (!cs53l30) { + dev_err(&client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + /* Reset the Device */ + cs53l30->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs53l30->reset_gpio)) + return PTR_ERR(cs53l30->reset_gpio); + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + i2c_set_clientdata(client, cs53l30); + + cs53l30->mclk = 0; + + cs53l30->regmap = devm_regmap_init_i2c(client, &cs53l30_regmap); + if (IS_ERR(cs53l30->regmap)) { + ret = PTR_ERR(cs53l30->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + /* initialize codec */ + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_AB, &reg); + devid = reg << 12; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_CD, &reg); + devid |= reg << 4; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS53L30_DEVID) { + ret = -ENODEV; + dev_err(&client->dev, + "CS53L30 Device ID (%X). Expected %X\n", + devid, CS53L30_DEVID); + return ret; + } + + ret = regmap_read(cs53l30->regmap, CS53L30_REVID, &reg); + if (ret < 0) { + dev_err(&client->dev, "Get Revision ID failed\n"); + return ret; + } + + dev_info(&client->dev, + "Cirrus Logic CS53L30, Revision: %02X\n", reg & 0xFF); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_cs53l30, cs53l30_dai, + ARRAY_SIZE(cs53l30_dai)); + return ret; +} + +static int cs53l30_i2c_remove(struct i2c_client *client) +{ + struct cs53l30_private *cs53l30 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs53l30_runtime_suspend(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + regcache_cache_only(cs53l30->regmap, true); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + return 0; +} + +static int cs53l30_runtime_resume(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + regcache_cache_only(cs53l30->regmap, false); + regcache_sync(cs53l30->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs53l30_runtime_pm = { + SET_RUNTIME_PM_OPS(cs53l30_runtime_suspend, cs53l30_runtime_resume, + NULL) +}; + +static const struct of_device_id cs53l30_of_match[] = { + { .compatible = "cirrus,cs53l30", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs53l30_of_match); + +static const struct i2c_device_id cs53l30_id[] = { + {"cs53l30", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs53l30_id); + +static struct i2c_driver cs53l30_i2c_driver = { + .driver = { + .name = "cs53l30", + .owner = THIS_MODULE, + }, + .id_table = cs53l30_id, + .probe = cs53l30_i2c_probe, + .remove = cs53l30_i2c_remove, + +}; + +module_i2c_driver(cs53l30_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS53L30 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <Paul.Handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs53l30.h b/sound/soc/codecs/cs53l30.h new file mode 100644 index 0000000..f45e91f +++ b/sound/soc/codecs/cs53l30.h @@ -0,0 +1,272 @@ +/* + * ALSA SoC CS53L30 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * Tim Howe <Tim.Howe(a)cirrus.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 __CS53L30_H__ +#define __CS53L30_H__ + +/* I2C Registers */ +#define CS53L30_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS53L30_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS53L30_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS53L30_REVID 0x05 /* Revision ID [RO]. */ +#define CS53L30_PWRCTL 0x06 /* Power Control. */ +#define CS53L30_MCLKCTL 0x07 /* MCLK Control. */ +#define CS53L30_INT_SR_CTL 0x08 /* Internal Sample Rate Control. */ +#define CS53L30_MICBIAS_CTL 0x0A /* Mic Bias Control. */ +#define CS53L30_ASPCFG_CTL 0x0C /* ASP Config Control. */ +#define CS53L30_ASP1_CTL 0x0D /* ASP1 Control. */ +#define CS53L30_ASP1_TDMTX_CTL1 0x0E /* ASP1 TDM TX Control 1 */ +#define CS53L30_ASP1_TDMTX_CTL2 0x0F /* ASP1 TDM TX Control 2 */ +#define CS53L30_ASP1_TDMTX_CTL3 0x10 /* ASP1 TDM TX Control 3 */ +#define CS53L30_ASP1_TDMTX_CTL4 0x11 /* ASP1 TDM TX Control 4 */ +#define CS53L30_ASP1_TDMTX_EN1 0x12 /* ASP1 TDM TX Enable 1 */ +#define CS53L30_ASP1_TDMTX_EN2 0x13 /* ASP1 TDM TX Enable 2 */ +#define CS53L30_ASP1_TDMTX_EN3 0x14 /* ASP1 TDM TX Enable 3 */ +#define CS53L30_ASP1_TDMTX_EN4 0x15 /* ASP1 TDM TX Enable 4 */ +#define CS53L30_ASP1_TDMTX_EN5 0x16 /* ASP1 TDM TX Enable 5 */ +#define CS53L30_ASP1_TDMTX_EN6 0x17 /* ASP1 TDM TX Enable 6 */ +#define CS53L30_ASP2_CTL 0x18 /* ASP2 Control. */ +#define CS53L30_SFT_RAMP 0x1A /* Soft Ramp Control. */ +#define CS53L30_LRCLK_CTL1 0x1B /* LRCLK Control 1. */ +#define CS53L30_LRCLK_CTL2 0x1C /* LRCLK Control 2. */ +#define CS53L30_MUTEP_CTL1 0x1F /* Mute Pin Control 1. */ +#define CS53L30_MUTEP_CTL2 0x20 /* Mute Pin Control 2. */ +#define CS53L30_INBIAS_CTL1 0x21 /* Input Bias Control 1. */ +#define CS53L30_INBIAS_CTL2 0x22 /* Input Bias Control 2. */ +#define CS53L30_DMIC1_STR_CTL 0x23 /* DMIC1 Stereo Control. */ +#define CS53L30_DMIC2_STR_CTL 0x24 /* DMIC2 Stereo Control. */ +#define CS53L30_ADCDMIC1_CTL1 0x25 /* ADC1/DMIC1 Control 1. */ +#define CS53L30_ADCDMIC1_CTL2 0x26 /* ADC1/DMIC1 Control 2. */ +#define CS53L30_ADC1_CTL3 0x27 /* ADC1 Control 3. */ +#define CS53L30_ADC1_NG_CTL 0x28 /* ADC1 Noise Gate Control. */ +#define CS53L30_ADC1A_AFE_CTL 0x29 /* ADC1A AFE Control. */ +#define CS53L30_ADC1B_AFE_CTL 0x2A /* ADC1B AFE Control. */ +#define CS53L30_ADC1A_DIG_VOL 0x2B /* ADC1A Digital Volume. */ +#define CS53L30_ADC1B_DIG_VOL 0x2C /* ADC1B Digital Volume. */ +#define CS53L30_ADCDMIC2_CTL1 0x2D /* ADC2/DMIC2 Control 1. */ +#define CS53L30_ADCDMIC2_CTL2 0x2E /* ADC2/DMIC2 Control 2. */ +#define CS53L30_ADC2_CTL3 0x2F /* ADC2 Control 3. */ +#define CS53L30_ADC2_NG_CTL 0x30 /* ADC2 Noise Gate Control. */ +#define CS53L30_ADC2A_AFE_CTL 0x31 /* ADC2A AFE Control. */ +#define CS53L30_ADC2B_AFE_CTL 0x32 /* ADC2B AFE Control. */ +#define CS53L30_ADC2A_DIG_VOL 0x33 /* ADC2A Digital Volume. */ +#define CS53L30_ADC2B_DIG_VOL 0x34 /* ADC2B Digital Volume. */ +#define CS53L30_INT_MASK 0x35 /* Interrupt Mask. */ +#define CS53L30_IS 0x36 /* Interrupt Status. */ +#define CS53L30_MAX_REGISTER 0x36 + +/* Device ID */ +#define CS53L30_DEVID 0x53A30 + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define PDN_POLL_MAX 900 + +/* Bitfield Definitions */ + +/* CS53L30_PWRCTL */ +#define PDN_ULP (1 << 7) +#define PDN_LP (1 << 6) +#define DISCHARGE_FILT (1 << 5) +#define THMS_PDN (1 << 4) + +/* CS53L30_MCLKCTL */ +#define MCLK_DIS (1 << 7) +#define MCLK_INT_SCALE (1 << 6) +#define DMIC_DRIVE (1 << 5) +#define MCLK_DIV (3 << 2) +#define MCLK_DIV_DFLT (1 << 2) +#define SYNC_EN (1 << 1) + +/* CS53L30_INT_SR_CTL */ +#define INTERNAL_FS_RATIO (1 << 4) +#define INTERNAL_FS_DFLT (7 << 2) +#define MCLK_19MHZ_EN (1 << 0) + +/* CS53L30_MICBIAS_CTL */ +#define MIC4_BIAS_PDN (1 << 7) +#define MIC3_BIAS_PDN (1 << 6) +#define MIC2_BIAS_PDN (1 << 5) +#define MIC1_BIAS_PDN (1 << 4) +#define VP_MIN (1 << 2) +#define MIC_BIAS_CTRL (3 << 0) +#define MIC_BIAS_ALL_PDN 0xF0 +#define MIC_BIAS_DFLT (MIC_BIAS_ALL_PDN | VP_MIN) + +/* CS53L30_ASPCFG_CTL */ +#define ASP_MS (1 << 7) +#define ASP_SCLK_INV (1 << 4) +#define ASP_RATE_48K (3 << 2) +#define ASP_RATE 0x0F +#define ASP_CNFG_MASK 0xF0 + +/* CS53L30_ASP1_CTL */ +#define ASP1_TDM_PDN (1 << 7) +#define ASP1_SDOUT_PDN (1 << 6) +#define ASP1_3ST (1 << 5) +#define SHIFT_LEFT (1 << 4) +#define ASP1_DRIVE (1 << 0) +#define ASP1_3ST_VAL(x) ((x) << 5) + +/* CS53L30_ASP1_TDMTX_CTL */ +#define ASP1_CHX_TX_STATE (1 << 7) +#define ASP1_CHX_TX_LOC 0x3F +#define ASP1_CHX_TX_DFLT_SLT47 0x2F +#define ASP_TX_DISABLED 0x00 + +/* CS53L30_ASP2_CTL */ +#define ASP2_SDOUT_PDN (1 << 6) +#define ASP2_DRIVE (1 << 0) +#define ASP2_CTRL_DFLT 0x00 + +/* CS53L30_SFT_RAMP */ +#define DIGSFT (1 << 5) +#define SFT_RMP_DFLT 0x00 + +/* CS53L30_LRCLK_CTL2 */ +#define LRCK_50_NPW (1 << 3) +#define LRCK_TPWH (7 << 0) +#define LRCK_CTLX_DFLT 0x00 + +/* CS53L30_MUTEP_CTL */ +#define MUTE_PDN_ULP (1 << 7) +#define MUTE_PDN_LP (1 << 6) +#define MUTE_M4B_PDN (1 << 4) +#define MUTE_M3B_PDN (1 << 3) +#define MUTE_M2B_PDN (1 << 2) +#define MUTE_M1B_PDN (1 << 1) +#define MUTE_MB_ALL_PDN (1 << 0) +#define MUTEP_CTRL1_DFLT 0x00 + +/* CS53L30_MUTEP_CTL2 */ +#define MUTE_PIN_POLARITY (1 << 7) +#define MUTE_ASP_TDM_PDN (1 << 6) +#define MUTE_ASP_SDOUT2_PDN (1 << 5) +#define MUTE_ASP_SDOUT1_PDN (1 << 4) +#define MUTE_ADC2B_PDN (1 << 3) +#define MUTE_ADC2A_PDN (1 << 2) +#define MUTE_ADC1B_PDN (1 << 1) +#define MUTE_ADC1A_PDN (1 << 0) + +/* CS53L30_INBIAS_CTL1 */ +#define IN4M_BIAS (3 << 6) +#define IN4P_BIAS (3 << 4) +#define IN3M_BIAS (3 << 2) +#define IN3P_BIAS (3 << 0) + +/* CS53L30_INBIAS_CTL2 */ +#define IN2M_BIAS (3 << 6) +#define IN2P_BIAS (3 << 4) +#define IN1M_BIAS (3 << 2) +#define IN1P_BIAS (3 << 0) +#define INBIAS_CTLX_DFLT 0xAA + +/* CS53L30_DMIC1_STR_CTL */ +#define DMIC1_STEREO_ENB (1 << 5) +#define DMIC1_STEREO_DFLT 0xA8 + +/* CS53L30_DMIC2_STR_CTL */ +#define DMIC2_STEREO_EN (1 << 5) +#define DMIC2_STEREO_DFLT 0xEC + +/* CS53L30_ADCDMIC1_CTL1 */ +#define ADC1B_PDN (1 << 7) +#define ADC1A_PDN (1 << 6) +#define DMIC1_PDN (1 << 2) +#define DMIC1_SCLK_DIV (1 << 1) +#define CH_TYPE (1 << 0) +#define DMIC1_ON_CH_AB_IN (CH_TYPE) +#define DMIC1_ON_CH_A_IN (ADC1B_PDN | CH_TYPE) +#define DMIC1_ON_CH_B_IN (ADC1A_PDN | CH_TYPE) +#define ADC1_ON_CH_AB_IN (DMIC1_PDN) +#define ADC1_ON_CH_A_IN (ADC1B_PDN | DMIC1_PDN) +#define ADC1_ON_CH_B_IN (ADC1A_PDN | DMIC1_PDN) +#define DMIC1_OFF_ADC1_OFF (ADC1A_PDN | ADC1B_PDN | DMIC1_PDN) +#define ADC1_DMIC1_PDN_MASK 0xFF + +/* CS53L30_ADCDMIC1_CTL2 */ +#define ADC1_NOTCH_DIS (1 << 7) +#define ADC1B_INV (1 << 5) +#define ADC1A_INV (1 << 4) +#define ADC1B_DIG_BOOST (1 << 1) +#define ADC1A_DIG_BOOST (1 << 0) +#define ADC1_DMIC1_CTL2_DFLT 0x00 + +/* CS53L30_ADC1_CTL3 */ +#define ADC1_HPF_EN (1 << 3) +#define ADC1_HPF_CF (3 << 1) +#define ADC1_NG_ALL (1 << 0) + +/* CS53L30_ADC1_NG_CTL */ +#define ADC1B_NG (1 << 7) +#define ADC1A_NG (1 << 6) +#define ADC1_NG_BOOST (1 << 5) +#define ADC1_NG_THRESH (7 << 2) +#define ADC1_NG_DELAY (3 << 0) +#define ADCX_ZERO_DFLT 0x00 + +/* CS53L30_ADC1A_AFE_CTL */ +#define ADC1A_PREAMP (3 << 6) +#define ADC1A_PGA_VOL 0x3F + +/* CS53L30_ADC1B_AFE_CTL */ +#define ADC1B_PREAMP (3 << 6) +#define ADC1B_PGA_VOL 0x3F + +/* CS53L30_ADCXX_DIG_VOL */ +#define MUTE_DIG_OUT (1 << 7) + +/* CS53L30_ADCDMIC2_CTL1 */ +#define ADC2B_PDN (1 << 7) +#define ADC2A_PDN (1 << 6) +#define DMIC2_PDN (1 << 2) +#define DMIC2_CLKDIV (1 << 1) +#define DMIC2_ON_CH_AB_IN 0x00 /* CH_TYPE must = 1 */ +#define DMIC2_ON_CH_A_IN (ADC2B_PDN) /* CH_TYPE must = 1 */ +#define DMIC2_ON_CH_B_IN (ADC2A_PDN) /* CH_TYPE must = 1 */ +#define ADC2_ON_CH_AB_IN (DMIC2_PDN) /* CH_TYPE must = 0 */ +#define ADC2_ON_CH_A_IN (ADC2B_PDN | DMIC2_PDN) /* CH_TYPE must = 0 */ +#define ADC2_ON_CH_B_IN (ADC2A_PDN | DMIC2_PDN) /* CH_TYPE must = 0 */ +#define DMIC2_OFF_ADC2_OFF (ADC2A_PDN | ADC2B_PDN | DMIC2_PDN) + +/* CS53L30_ADCDMIC2_CTL2 */ +#define ADC2_NOTCH_DIS (1 << 7) +#define ADC2B_INV (1 << 5) +#define ADC2A_INV (1 << 4) +#define ADC2B_DIG_BOOST (1 << 1) +#define ADC2A_DIG_BOOST (1 << 0) + +/* CS53L30_ADC2_CTL3 */ +#define ADC2_HPF_EN (1 << 3) +#define ADC2_HPF_CF (3 << 1) +#define ADC2_NG_ALL (1 << 0) + +/* CS53L30_INT */ +#define PDN_DONE (1 << 7) +#define THMS_TRIP (1 << 6) +#define SYNC_DONE (1 << 5) +#define ADC2B_OVFL (1 << 4) +#define ADC2A_OVFL (1 << 3) +#define ADC1B_OVFL (1 << 2) +#define ADC1A_OVFL (1 << 1) +#define MUTE_PIN (1 << 0) +#define DEVICE_INT_MASK 0xFF + +/* Serial Ports */ +#define CS53L30_ASP1 0 +#define CS53L30_ASP2 1 + +#endif /* __CS53L30_H__ */ -- 2.4.5

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[alsa-devel] vsp1 BUG_ON() and crash (Re: [PATCH v9 03/12] media: Entities, pads and links)
by Geert Uytterhoeven 14 Dec '15

14 Dec '15

On Mon, Feb 14, 2011 at 1:20 PM, Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> wrote: > +media_entity_create_link(struct media_entity *source, u16 source_pad, > + struct media_entity *sink, u16 sink_pad, u32 flags) > +{ > + struct media_link *link; > + struct media_link *backlink; > + > + BUG_ON(source == NULL || sink == NULL); > + BUG_ON(source_pad >= source->num_pads); > + BUG_ON(sink_pad >= sink->num_pads); This triggers on r8a7791/koelsch: vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 bru was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 hsi was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 hst was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 lut was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 rpf.0 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 rpf.1 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 rpf.2 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 rpf.3 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 rpf.4 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 sru was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 uds.0 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 uds.1 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 uds.2 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 wpf.0 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 wpf.1 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 wpf.2 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 wpf.3 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 bru was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 hsi was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 hst was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 lif was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 lut was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 rpf.0 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 rpf.1 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 rpf.2 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 rpf.3 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 uds.0 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 wpf.0 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 wpf.1 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 wpf.2 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 wpf.3 was not initialized! ------------[ cut here ]------------ kernel BUG at drivers/media/media-entity.c:521! Internal error: Oops - BUG: 0 [#1] SMP ARM CPU: 0 PID: 1 Comm: swapper/0 Not tainted 4.4.0-rc5-03225-g596de581263544d2 #400 Hardware name: Generic R8A7791 (Flattened Device Tree) task: ee84fb40 ti: ee850000 task.ti: ee850000 PC is at media_create_pad_link+0x50/0x13c LR is at vsp1_wpf_create_links+0x4c/0x54 pc : [<c034fd90>] lr : [<c03761cc>] psr: 60000013 sp : ee851d50 ip : ee851d80 fp : ee851d7c r10: 00000001 r9 : ee9e9600 r8 : 00000000 r7 : ee111c80 r6 : 00000000 r5 : ee114a40 r4 : ee114b18 r3 : 00000000 r2 : ee114b18 r1 : 00000001 r0 : ee114a40 Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment none Control: 10c5307d Table: 4000406a DAC: 00000051 Process swapper/0 (pid: 1, stack limit = 0xee850210) Stack: (0xee851d50 to 0xee852000) 1d40: 00000000 ee111c10 ee114a10 ee111c90 1d60: ee111c80 ee9e9610 ee9e9600 ee111c98 ee851d94 ee851d80 c03761cc c034fd4c 1d80: 00000001 ee111c98 ee851de4 ee851d98 c03730c0 c037618c 00000000 ee9eac00 1da0: ee111c10 ee851db0 00000000 ee114840 ee114c40 00000000 c071b678 00000000 1dc0: ee9e9610 c06d3414 c071b678 c06d3414 00000000 00000000 ee851e04 ee851de8 1de0: c028f2b0 c0372bbc c028f258 ee9e9610 c071b668 00000000 ee851e2c ee851e08 1e00: c028dd4c c028f264 ee9e9610 ee9e9644 c06d3414 c06cb3a8 c069e010 c06e2000 1e20: ee851e4c ee851e30 c028df04 c028dc1c 00000001 00000000 c06d3414 c028de94 1e40: ee851e74 ee851e50 c028c47c c028dea0 ee884f5c ee9e7bb4 ee884f70 c06d3414 1e60: 00000000 eeaeb980 ee851e84 ee851e78 c028e138 c028c414 ee851eac ee851e88 1e80: c028cc18 c028e124 c0646a6b ee851e98 c06d3414 c0681dc8 00000000 c069e010 1ea0: ee851ec4 ee851eb0 c028e848 c028cb50 ee0bd380 c0681dc8 ee851ed4 ee851ec8 1ec0: c028fcbc c028e7b0 ee851ee4 ee851ed8 c0681de0 c028fc90 ee851f5c ee851ee8 1ee0: c0664df8 c0681dd4 ee851f0c ee851ef8 c0664604 c01c94bc efffcc00 efffcc10 1f00: ee851f5c ee851f10 c003f90c c06645f4 00000000 c0661740 c0661740 00000006 1f20: 00000006 000000b6 c0660908 efffcc38 ee851f5c 00000006 00000006 000000b6 1f40: c069e83c c06aad08 c06e2000 c06e2000 ee851f94 ee851f60 c0664fd4 c0664cfc 1f60: 00000006 00000006 00000000 c06645e8 00000000 c04e752c 00000000 00000000 1f80: 00000000 00000000 ee851fac ee851f98 c04e753c c0664eb8 ee850000 00000000 1fa0: 00000000 ee851fb0 c000fe98 c04e7538 00000000 00000000 00000000 00000000 1fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 1fe0: 00000000 00000000 00000000 00000000 00000013 00000000 00000000 00000000 Backtrace: [<c034fd40>] (media_create_pad_link) from [<c03761cc>] (vsp1_wpf_create_links+0x4c/0x54) r10:ee111c98 r9:ee9e9600 r8:ee9e9610 r7:ee111c80 r6:ee111c90 r5:ee114a10 r4:ee111c10 r3:00000000 [<c0376180>] (vsp1_wpf_create_links) from [<c03730c0>] (vsp1_probe+0x510/0x700) [<c0372bb0>] (vsp1_probe) from [<c028f2b0>] (platform_drv_probe+0x58/0xa8) r10:00000000 r9:00000000 r8:c06d3414 r7:c071b678 r6:c06d3414 r5:ee9e9610 r4:00000000 While bisecting, I arrived at a NULL pointer dereference: Unable to handle kernel NULL pointer dereference at virtual address 00000250 pgd = c0004000 [00000250] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM CPU: 0 PID: 1 Comm: swapper/0 Not tainted 4.4.0-rc2-00170-gf9836d92a3c8c143 #417 ata1: SATA link down (SStatus 0 SControl 300) Hardware name: Generic R8A7791 (Flattened Device Tree) task: ee84fb40 ti: ee850000 task.ti: ee850000 PC is at media_gobj_init+0x20/0x68 LR is at media_entity_create_link+0xe4/0x108 pc : [<c0342428>] lr : [<c0342a80>] psr: 60000013 sp : ee851d10 ip : ee851d20 fp : ee851d1c r10: 00000000 r9 : ee103890 r8 : 00000000 r7 : ee159920 r6 : ee103740 r5 : ee103840 r4 : ee159840 r3 : 00000003 r2 : ee103840 r1 : 00000002 r0 : 00000000 Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment none Control: 10c5307d Table: 4000406a DAC: 00000051 Process swapper/0 (pid: 1, stack limit = 0xee850210) Stack: (0xee851d10 to 0xee852000) 1d00: ee851d4c ee851d20 c0342a80 c0342414 1d20: 00000000 ee159810 00000000 ee159840 ee159910 00000000 ee9e6800 ee102898 1d40: ee851d94 ee851d50 c036506c c03429a8 00000003 00000000 000000ff 00000000 1d60: 00000001 00000000 000000ff 00000000 c0366740 ee102810 ee102890 00000000 1d80: ee102858 ee9e6810 ee851de4 ee851d98 c0362a44 c0364ef4 00000080 ee9e8f00 1da0: ee102810 ee851db0 c012bcac ee9e6810 ee9e6810 c06b1f88 c06e76b4 00000000 1dc0: ee9e6810 c06b1f88 c06e76b4 00000000 c06b1f88 00000000 ee851e04 ee851de8 1de0: c0285b14 c0362694 c0285abc ee9e6810 00000000 c06e76a8 ee851e2c ee851e08 1e00: c028457c c0285ac8 ee9e6810 ee9e6844 c06b1f88 c06aab38 c067d010 c06c1000 1e20: ee851e4c ee851e30 c0284770 c028446c 00000001 00000000 c06b1f88 c0284700 1e40: ee851e74 ee851e50 c0282d78 c028470c ee884f5c ee9e4cb4 ee884f70 c06b1f88 1e60: 00000000 ee9c4700 ee851e84 ee851e78 c028499c c0282d10 ee851eac ee851e88 1e80: c0283514 c0284988 c0627e00 ee851e98 c06b1f88 c06618a8 00000000 c067d010 1ea0: ee851ec4 ee851eb0 c02850ac c028344c eea9a200 c06618a8 ee851ed4 ee851ec8 1ec0: c02864e0 c0285014 ee851ee4 ee851ed8 c06618c0 c02864b4 ee851f5c ee851ee8 1ee0: c0644df8 c06618b4 ee851f0c ee851ef8 c0644604 c01c91bc efffcc00 efffcc10 1f00: ee851f5c ee851f10 c003f908 c06445f4 00000000 c0641c1c c0641c1c 00000006 1f20: 00000006 000000b1 c0640e48 efffcc38 ee851f5c 00000006 00000006 000000b1 1f40: c067d83c c0689c84 c06c1000 c06c1000 ee851f94 ee851f60 c0644fd4 c0644cfc 1f60: 00000006 00000006 00000000 c06445e8 00000000 c04d00ac 00000000 00000000 1f80: 00000000 00000000 ee851fac ee851f98 c04d00bc c0644eb8 ee850000 00000000 1fa0: 00000000 ee851fb0 c000fe98 c04d00b8 00000000 00000000 00000000 00000000 1fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 1fe0: 00000000 00000000 00000000 00000000 00000013 00000000 00000000 00000000 Backtrace: [<c0342408>] (media_gobj_init) from [<c0342a80>] (media_entity_create_link+0xe4/0x108) [<c034299c>] (media_entity_create_link) from [<c036506c>] (vsp1_rpf_create+0x184/0x1c8) r10:ee102898 r9:ee9e6800 r8:00000000 r7:ee159910 r6:ee159840 r5:00000000 r4:ee159810 r3:00000000 [<c0364ee8>] (vsp1_rpf_create) from [<c0362a44>] (vsp1_probe+0x3bc/0x6c0) r8:ee9e6810 r7:ee102858 r6:00000000 r5:ee102890 r4:ee102810 [<c0362688>] (vsp1_probe) from [<c0285b14>] (platform_drv_probe+0x58/0xa8) r10:00000000 r9:c06b1f88 r8:00000000 r7:c06e76b4 r6:c06b1f88 r5:ee9e6810 r4:00000000 The NULL pointer dereference was introduced by: commit f9836d92a3c8c143c6f5fe89c558c37767628fa6 Author: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> Date: Fri Aug 14 12:54:36 2015 -0300 [media] media: use media_gobj inside links and triggers until the following commit, after which the BUG_ON() takes over: commit eb9e998983338811c3328294ed2f093367161df1 Author: Javier Martinez Canillas <javier(a)osg.samsung.com> Date: Thu Sep 3 12:19:25 2015 -0300 [media] v4l: vsp1: separate links creation from entities init Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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[alsa-devel] [PATCH v2] sound: codecs: pcm1792a: clean function exit
by Raphael Poggi 14 Dec '15

14 Dec '15

Signed-off-by: Raphael Poggi <poggi.raph(a)gmail.com> --- sound/soc/codecs/pcm1792a.c | 15 +++------------ 1 file changed, 3 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 08bb486..27d54bc 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -91,14 +91,9 @@ static int pcm1792a_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); - int ret; - ret = regmap_update_bits(priv->regmap, PCM1792A_SOFT_MUTE, + return regmap_update_bits(priv->regmap, PCM1792A_SOFT_MUTE, PCM1792A_MUTE_MASK, !!mute); - if (ret < 0) - return ret; - - return 0; } static int pcm1792a_hw_params(struct snd_pcm_substream *substream, @@ -107,7 +102,7 @@ static int pcm1792a_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); - int val = 0, ret; + int val = 0; priv->rate = params_rate(params); @@ -145,12 +140,8 @@ static int pcm1792a_hw_params(struct snd_pcm_substream *substream, val = val << PCM1792A_FMT_SHIFT | PCM1792A_ATLD_ENABLE; - ret = regmap_update_bits(priv->regmap, PCM1792A_FMT_CONTROL, + return regmap_update_bits(priv->regmap, PCM1792A_FMT_CONTROL, PCM1792A_FMT_MASK | PCM1792A_ATLD_ENABLE, val); - if (ret < 0) - return ret; - - return 0; } static const struct snd_soc_dai_ops pcm1792a_dai_ops = { -- 2.1.0

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[alsa-devel] [PATCH 1/2] ALSA: usb-audio: Add a volume scale quirk for AudioQuest DragonFly
by Anssi Hannula 14 Dec '15

14 Dec '15

AudioQuest DragonFly DAC reports a volume control range of 0..50 (0x0000..0x0032) which in USB Audio means a range of 0 .. 0.2dB, which is obviously incorrect and causes software using the dB information in e.g. volume sliders to have a massive volume difference in 100..102% range. The actual volume mapping seems to be neither linear volume nor linear dB scale, but instead quite close to the cubic mapping e.g. alsamixer uses, with a range of -53...0 dB. Add a quirk for DragonFly to use a custom dB mapping, based on my measurements, using a 10-item range TLV (so it still fits in alsa-lib MAX_TLV_RANGE_SIZE). Tested on AudioQuest DragonFly HW v1.2. The quirk is only applied if the range is 0..50, so if this gets fixed/changed in later HW revisions it will no longer be applied. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: <stable(a)vger.kernel.org> --- sound/usb/mixer_quirks.c | 37 +++++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 337c317ead6f..39d7f34e44e6 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -37,6 +37,7 @@ #include <sound/control.h> #include <sound/hwdep.h> #include <sound/info.h> +#include <sound/tlv.h> #include "usbaudio.h" #include "mixer.h" @@ -1733,6 +1734,38 @@ static int snd_microii_controls_create(struct usb_mixer_interface *mixer) return 0; } +static void snd_dragonfly_quirk_db_scale(struct usb_mixer_interface *mixer) +{ + struct usb_mixer_elem_list *list; + struct usb_mixer_elem_info *cval; + static const int unit_id = 7; + + /* approximation using 10 ranges based on output measurement on hw v1.2 */ + static const DECLARE_TLV_DB_RANGE(scale, + 0, 1, TLV_DB_MINMAX_ITEM(-5300, -4970), + 2, 5, TLV_DB_MINMAX_ITEM(-4710, -4160), + 6, 7, TLV_DB_MINMAX_ITEM(-3884, -3710), + 8, 14, TLV_DB_MINMAX_ITEM(-3443, -2560), + 15, 16, TLV_DB_MINMAX_ITEM(-2475, -2324), + 17, 19, TLV_DB_MINMAX_ITEM(-2228, -2031), + 20, 26, TLV_DB_MINMAX_ITEM(-1910, -1393), + 27, 31, TLV_DB_MINMAX_ITEM(-1322, -1032), + 32, 40, TLV_DB_MINMAX_ITEM(-968, -490), + 41, 50, TLV_DB_MINMAX_ITEM(-441, 0), + ); + + for (list = mixer->id_elems[unit_id]; list; list = list->next_id_elem) { + cval = (struct usb_mixer_elem_info *)list; + if (cval->control == UAC_FU_VOLUME && + cval->min == 0 && cval->max == 50) { + usb_audio_info(mixer->chip, "applying DragonFly dB scale quirk\n"); + list->kctl->tlv.p = scale; + list->kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + list->kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } + } +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1810,6 +1843,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x1235, 0x800c): /* Focusrite Scarlett 18i20 */ err = snd_scarlett_controls_create(mixer); break; + + case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ + snd_dragonfly_quirk_db_scale(mixer); + break; } return err; -- 2.3.8

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[alsa-devel] [PATCH v3 0/5] ALSA: hda - hdmi support dynmaic pcm assignment
by libin.yang＠linux.intel.com 14 Dec '15

14 Dec '15

From: Libin Yang <libin.yang(a)linux.intel.com> The patches are trying to support dynamically binding PCM to pin in HDMI/DP audio. The pin will not bind to any PCM if there is not monitor connected. When there is a monitor connected to the port (pin), driver will try to find a proper PCM to assign to the port and setup the pin if necessary. Libin Yang (5): ALSA: hda - hdmi begin to support dynamic PCM assignment ALSA: hda - hdmi playback without monitor in dynamic pcm bind mode ALSA: hda - hdmi operate spdif based on pcm ALSA: hda - hdmi dynamically bind PCM to pin when monitor hotplug ALSA: hda - hdmi setup pin when monitor hotplug in pcm dynamic assignment mode sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/patch_hdmi.c | 369 ++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 347 insertions(+), 23 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v3 00/13] ASoC: Intel: Skylake: Update to SKL driver
by Vinod Koul 14 Dec '15

14 Dec '15

This update bring in new APIs in core and usage of those in SKL driver. The series is broadly of three parts - first two are fixes on the driver - then we add DMA resume capability in HDA core and use it to save restore DMA link values and add DMA resume capability in driver - lastly, we add support for MISCBDCGE configuration in core and use it while resetting and also for phrase detection per HW recommendation This should be merged thru ASoC tree due to obvious dependencies. Takashi would need your ack on core patches change is v3: - fix the missing export symbol, reported by lkp changes in v2: - move msicbdge api to driver and as a consequence sqash into into first use - fix lkp warning Dharageswari.R (1): ASoC: Intel: Skylake: Use CGCTL.MISCBDCGE for Phrase detection notification Jayachandran B (2): ALSA: hdac: Increase timeout value for link power check ASoC: Intel: Skylake: fix reset controller sequencing Jeeja KP (8): ASoC: Intel: Skylake: Clear stream registers before stream setup ASoC: Intel: Skylake: Fix to set pipe state to invalid when deleting ALSA: hdac: Add support for hda DMA Resume capability ALSA: hdac: couple the hda DMA stream in cleanup ASoC: Intel: Skylake: enable interrupt as wake source in active suspend ASoC: Intel: Skylake: Add DMA resume position in Trigger resume/suspend ASoC: Intel: Skylake: Reconfigure Link stream on suspend/resume ASoC: Intel: Skylake: Add Resume capability in PCM info. Vinod Koul (2): ALSA: hdac: add snd_hdac_ext_bus_link_power_up_all ASoC: Intel: Skylake: manage link power in active suspend include/sound/hda_register.h | 9 +++++ include/sound/hdaudio_ext.h | 15 +++++++ sound/hda/ext/hdac_ext_controller.c | 29 ++++++++++++- sound/hda/ext/hdac_ext_stream.c | 72 +++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-messages.c | 2 + sound/soc/intel/skylake/skl-pcm.c | 56 ++++++++++++++++++++----- sound/soc/intel/skylake/skl-sst-cldma.c | 38 ++++++++++------- sound/soc/intel/skylake/skl-sst-ipc.c | 15 +++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 3 ++ sound/soc/intel/skylake/skl.c | 58 ++++++++++++++++++++++++-- sound/soc/intel/skylake/skl.h | 5 +++ 11 files changed, 272 insertions(+), 30 deletions(-) -- 1.9.1

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Re: [alsa-devel] [PATCH v2 08/13] ASoC: Intel: Skylake: Use CGCTL.MISCBDCGE for Phrase detection notification
by Vinod Koul 14 Dec '15

14 Dec '15

On Sun, Dec 13, 2015 at 02:46:08AM +0800, kbuild test robot wrote: > Hi Dharageswari.R, > > [auto build test ERROR on asoc/for-next] > [also build test ERROR on next-20151211] > [cannot apply to v4.4-rc4] > > url: https://github.com/0day-ci/linux/commits/Vinod-Koul/ASoC-Intel-Skylake-Upda… > base: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next > config: i386-allmodconfig (attached as .config) > reproduce: > # save the attached .config to linux build tree > make ARCH=i386 > > All errors (new ones prefixed by >>): > > >> ERROR: "skl_enable_miscbdcge" undefined! I will update this one, thanks for reporting -- ~Vinod

1 0

[alsa-devel] [PATCH] ASoC: es8328: Fix shifts for mixer switches
by John Keeping 13 Dec '15

13 Dec '15

These are all off by one; the playback and bypass switches are the top two bits of the registers, which are at shifts 7 and 6 not 8 and 7. Signed-off-by: John Keeping <john(a)metanate.com> --- sound/soc/codecs/es8328.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 969e337..84f5eb0 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -205,18 +205,18 @@ static const struct snd_kcontrol_new es8328_right_line_controls = /* Left Mixer */ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { - SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 8, 1, 0), - SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 7, 1, 0), - SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 8, 1, 0), - SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 7, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 7, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 6, 1, 0), + SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 7, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 6, 1, 0), }; /* Right Mixer */ static const struct snd_kcontrol_new es8328_right_mixer_controls[] = { - SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 8, 1, 0), - SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 7, 1, 0), - SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 8, 1, 0), - SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 7, 1, 0), + SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 7, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 6, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 7, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 6, 1, 0), }; static const char * const es8328_pga_sel[] = { -- 2.6.3.462.gbe2c914

2 1

[alsa-devel] [PATCH] ASoC: wm8974: set cache type for regmap
by Mans Rullgard 13 Dec '15

13 Dec '15

Attempting to use this codec driver triggers a BUG() in regcache_sync() since no cache type is set. The register map of this device is fairly small and has few holes so a flat cache is suitable. Signed-off-by: Mans Rullgard <mans(a)mansr.com> --- sound/soc/codecs/wm8974.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 45ba828..c284c7b 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -574,6 +574,7 @@ static const struct regmap_config wm8974_regmap = { .max_register = WM8974_MONOMIX, .reg_defaults = wm8974_reg_defaults, .num_reg_defaults = ARRAY_SIZE(wm8974_reg_defaults), + .cache_type = REGCACHE_FLAT, }; static int wm8974_probe(struct snd_soc_codec *codec) -- 2.6.3

3 2

[alsa-devel] [PATCH] ASoC: rockchip: spdif: Set transmit data level to 16 samples
by Sjoerd Simons 13 Dec '15

13 Dec '15

Explicitly set the transmit data level on the transceiver to 16 samples rather then the default 0. This matches both the level set in the vendor kernel and the (seemingly very similar) i2s engine. This fixes audio glitches when playing back at 192k rate. At the same time, fix a trivial typo in the TDL mask definition Signed-off-by: Sjoerd Simons <sjoerd.simons(a)collabora.co.uk> --- sound/soc/rockchip/rockchip_spdif.c | 6 ++++-- sound/soc/rockchip/rockchip_spdif.h | 2 +- 2 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index ac72ff5..5a806da 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -152,8 +152,10 @@ static int rk_spdif_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ret = regmap_update_bits(spdif->regmap, SPDIF_DMACR, - SPDIF_DMACR_TDE_ENABLE, - SPDIF_DMACR_TDE_ENABLE); + SPDIF_DMACR_TDE_ENABLE | + SPDIF_DMACR_TDL_MASK, + SPDIF_DMACR_TDE_ENABLE | + SPDIF_DMACR_TDL(16)); if (ret != 0) return ret; diff --git a/sound/soc/rockchip/rockchip_spdif.h b/sound/soc/rockchip/rockchip_spdif.h index 921b409..3ef1277 100644 --- a/sound/soc/rockchip/rockchip_spdif.h +++ b/sound/soc/rockchip/rockchip_spdif.h @@ -42,7 +42,7 @@ #define SPDIF_DMACR_TDL_SHIFT 0 #define SPDIF_DMACR_TDL(x) ((x) << SPDIF_DMACR_TDL_SHIFT) -#define SPDIF_DMACR_TDL_MASK (0x1f << SDPIF_DMACR_TDL_SHIFT) +#define SPDIF_DMACR_TDL_MASK (0x1f << SPDIF_DMACR_TDL_SHIFT) /* * XFER -- 2.6.2

2 1

[alsa-devel] [PATCH V2 0/3] Add support for Texas Instruments pcm3168a codec
by Damien Horsley 13 Dec '15

13 Dec '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add SOC_DOUBLE_STS macro Add driver and binding docuemnt for Texas Instruments pcm3168a codec Changes in V2: Addressed comments from Mark Brown Damien.Horsley (3): ASoC: Add SOC_DOUBLE_STS macro ASoC: pcm3168a: Add binding document for pcm3168a codec ASoC: pcm3168a: Add driver for pcm3168a codec .../devicetree/bindings/sound/ti,pcm3168a.txt | 48 ++ include/sound/soc.h | 8 + sound/soc/codecs/Kconfig | 17 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/pcm3168a-i2c.c | 66 ++ sound/soc/codecs/pcm3168a-spi.c | 65 ++ sound/soc/codecs/pcm3168a.c | 767 +++++++++++++++++++++ sound/soc/codecs/pcm3168a.h | 100 +++ 8 files changed, 1077 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.txt create mode 100644 sound/soc/codecs/pcm3168a-i2c.c create mode 100644 sound/soc/codecs/pcm3168a-spi.c create mode 100644 sound/soc/codecs/pcm3168a.c create mode 100644 sound/soc/codecs/pcm3168a.h -- 2.1.4

2 5

[alsa-devel] [PATCH 0/2] Add support for Texas Instruments pcm3168a codec
by Damien Horsley 13 Dec '15

13 Dec '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add driver and binding docuemnt for Texas Instruments pcm3168a codec Damien.Horsley (2): ASoC: pcm3168a: Add binding document for pcm3168a codec ASoC: pcm3168a: Add driver for pcm3168a codec .../devicetree/bindings/sound/ti,pcm3168a.txt | 48 ++ sound/soc/codecs/Kconfig | 17 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/pcm3168a-i2c.c | 66 ++ sound/soc/codecs/pcm3168a-spi.c | 65 ++ sound/soc/codecs/pcm3168a.c | 766 +++++++++++++++++++++ sound/soc/codecs/pcm3168a.h | 111 +++ 7 files changed, 1079 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.txt create mode 100644 sound/soc/codecs/pcm3168a-i2c.c create mode 100644 sound/soc/codecs/pcm3168a-spi.c create mode 100644 sound/soc/codecs/pcm3168a.c create mode 100644 sound/soc/codecs/pcm3168a.h -- 2.1.4

2 6

[alsa-devel] [PATCH 0/2] Add support for Pistachio internal DAC
by Damien Horsley 13 Dec '15

13 Dec '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add binding document and driver for Pistachio internal DAC Damien.Horsley (2): ASoC: img: Add binding document for Pistachio internal DAC ASoC: img: Add driver for Pistachio internal DAC .../bindings/sound/img,pistachio-internal-dac.txt | 18 ++ sound/soc/img/Kconfig | 8 + sound/soc/img/Makefile | 2 + sound/soc/img/pistachio-internal-dac.c | 350 +++++++++++++++++++++ 4 files changed, 378 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt create mode 100644 sound/soc/img/pistachio-internal-dac.c -- 2.1.4

2 6

[alsa-devel] [PATCH V4 0/2] Add driver for Pistachio internal DAC
by Damien Horsley 13 Dec '15

13 Dec '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add binding document and driver for Pistachio internal DAC Changes in V4: Renamed Mute control and change definition to inverted Damien.Horsley (2): ASoC: img: Add binding document for Pistachio internal DAC ASoC: img: Add driver for Pistachio internal DAC .../bindings/sound/img,pistachio-internal-dac.txt | 18 ++ sound/soc/img/Kconfig | 8 + sound/soc/img/Makefile | 2 + sound/soc/img/pistachio-internal-dac.c | 287 +++++++++++++++++++++ 4 files changed, 315 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt create mode 100644 sound/soc/img/pistachio-internal-dac.c -- 2.1.4

2 3

[alsa-devel] [RFC PATCH] ASoC: dpcm: Apply symmetry for DPCM
by Koro Chen 13 Dec '15

13 Dec '15

From: PC Liao <pc.liao(a)mediatek.com> DPCM does not fully support symmetry attributes. soc_pcm_apply_symmetry() is skipped in soc_pcm_open() for DPCM, without being applied elsewhere. So HW parameters cannot be correctly limited, and user space can do playback/capture at different rates while HW actually does not support it. soc_pcm_params_symmetry() will return error and the second stream stops. This patch adds soc_pcm_apply_symmetry() for FE, BE, and codec DAIs in DPCM path that was skipped in soc_pcm_open(). Signed-off-by: PC Liao <pc.liao(a)mediatek.com> Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/soc-pcm.c | 57 +++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 67ef1a0..2a2ca22 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1614,6 +1614,56 @@ static void dpcm_set_fe_update_state(struct snd_soc_pcm_runtime *fe, snd_pcm_stream_unlock_irq(substream); } +static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, + int stream) +{ + struct snd_soc_dpcm *dpcm; + struct snd_soc_pcm_runtime *fe = fe_substream->private_data; + struct snd_soc_dai *fe_cpu_dai = fe->cpu_dai; + int err; + + /* apply symmetry for FE */ + if (soc_pcm_has_symmetry(fe_substream)) + fe_substream->runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; + + /* Symmetry only applies if we've got an active stream. */ + if (fe_cpu_dai->active) { + err = soc_pcm_apply_symmetry(fe_substream, fe_cpu_dai); + if (err < 0) + return err; + } + + /* apply symmetry for BE */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + struct snd_soc_pcm_runtime *rtd = be_substream->private_data; + int i; + + if (soc_pcm_has_symmetry(be_substream)) + be_substream->runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; + + /* Symmetry only applies if we've got an active stream. */ + if (rtd->cpu_dai->active) { + err = soc_pcm_apply_symmetry(be_substream, rtd->cpu_dai); + if (err < 0) + return err; + } + + for (i = 0; i < rtd->num_codecs; i++) { + if (rtd->codec_dais[i]->active) { + err = soc_pcm_apply_symmetry(be_substream, + rtd->codec_dais[i]); + if (err < 0) + return err; + } + } + } + + return 0; +} + static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) { struct snd_soc_pcm_runtime *fe = fe_substream->private_data; @@ -1642,6 +1692,13 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) dpcm_set_fe_runtime(fe_substream); snd_pcm_limit_hw_rates(runtime); + ret = dpcm_apply_symmetry(fe_substream, stream); + if (ret < 0) { + dev_err(fe->dev, "ASoC: failed to apply dpcm symmetry %d\n", + ret); + goto unwind; + } + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return 0; -- 1.7.9.5

3 2

[alsa-devel] [PATCH 181/182] ASoC: ac97: use gpiochip data pointer
by Linus Walleij 12 Dec '15

12 Dec '15

This makes the driver use the data pointer added to the gpio_chip to store a pointer to the state container instead of relying on container_of(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Liam/Mark: please ACK this so I can merge it into the GPIO tree. --- sound/soc/soc-ac97.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index a2012652f212..eaec7177eab2 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -59,8 +59,7 @@ static void soc_ac97_device_release(struct device *dev) #ifdef CONFIG_GPIOLIB static inline struct snd_soc_codec *gpio_to_codec(struct gpio_chip *chip) { - struct snd_ac97_gpio_priv *gpio_priv = - container_of(chip, struct snd_ac97_gpio_priv, gpio_chip); + struct snd_ac97_gpio_priv *gpio_priv = gpiochip_get_data(chip); return gpio_priv->codec; } @@ -98,8 +97,7 @@ static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct snd_ac97_gpio_priv *gpio_priv = - container_of(chip, struct snd_ac97_gpio_priv, gpio_chip); + struct snd_ac97_gpio_priv *gpio_priv = gpiochip_get_data(chip); struct snd_soc_codec *codec = gpio_to_codec(chip); gpio_priv->gpios_set &= ~(1 << offset); @@ -145,7 +143,7 @@ static int snd_soc_ac97_init_gpio(struct snd_ac97 *ac97, gpio_priv->gpio_chip.parent = codec->dev; gpio_priv->gpio_chip.base = -1; - ret = gpiochip_add(&gpio_priv->gpio_chip); + ret = gpiochip_add_data(&gpio_priv->gpio_chip, gpio_priv); if (ret != 0) dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); return ret; -- 2.4.3

2 1

[alsa-devel] [PATCH 180/182] ASoC: wm8996: use gpiochip data pointer
by Linus Walleij 12 Dec '15

12 Dec '15

This makes the driver use the data pointer added to the gpio_chip to store a pointer to the state container instead of relying on container_of(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Liam/Mark: please ACK this so I can merge it into the GPIO tree. --- sound/soc/codecs/wm8996.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8d7d6c01a2f7..f9656df9dccc 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -17,6 +17,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/gcd.h> +#include <linux/gpio/driver.h> #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -2139,14 +2140,9 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } #ifdef CONFIG_GPIOLIB -static inline struct wm8996_priv *gpio_to_wm8996(struct gpio_chip *chip) -{ - return container_of(chip, struct wm8996_priv, gpio_chip); -} - static void wm8996_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8996_priv *wm8996 = gpio_to_wm8996(chip); + struct wm8996_priv *wm8996 = gpiochip_get_data(chip); regmap_update_bits(wm8996->regmap, WM8996_GPIO_1 + offset, WM8996_GP1_LVL, !!value << WM8996_GP1_LVL_SHIFT); @@ -2155,7 +2151,7 @@ static void wm8996_gpio_set(struct gpio_chip *chip, unsigned offset, int value) static int wm8996_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8996_priv *wm8996 = gpio_to_wm8996(chip); + struct wm8996_priv *wm8996 = gpiochip_get_data(chip); int val; val = (1 << WM8996_GP1_FN_SHIFT) | (!!value << WM8996_GP1_LVL_SHIFT); @@ -2167,7 +2163,7 @@ static int wm8996_gpio_direction_out(struct gpio_chip *chip, static int wm8996_gpio_get(struct gpio_chip *chip, unsigned offset) { - struct wm8996_priv *wm8996 = gpio_to_wm8996(chip); + struct wm8996_priv *wm8996 = gpiochip_get_data(chip); unsigned int reg; int ret; @@ -2180,7 +2176,7 @@ static int wm8996_gpio_get(struct gpio_chip *chip, unsigned offset) static int wm8996_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { - struct wm8996_priv *wm8996 = gpio_to_wm8996(chip); + struct wm8996_priv *wm8996 = gpiochip_get_data(chip); return regmap_update_bits(wm8996->regmap, WM8996_GPIO_1 + offset, WM8996_GP1_FN_MASK | WM8996_GP1_DIR, @@ -2211,7 +2207,7 @@ static void wm8996_init_gpio(struct wm8996_priv *wm8996) else wm8996->gpio_chip.base = -1; - ret = gpiochip_add(&wm8996->gpio_chip); + ret = gpiochip_add_data(&wm8996->gpio_chip, wm8996); if (ret != 0) dev_err(wm8996->dev, "Failed to add GPIOs: %d\n", ret); } -- 2.4.3

3 2

[alsa-devel] [PATCH 179/182] ASoC: wm8962: use gpiochip data pointer
by Linus Walleij 12 Dec '15

12 Dec '15

This makes the driver use the data pointer added to the gpio_chip to store a pointer to the state container instead of relying on container_of(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Liam/Mark: please ACK this so I can merge it into the GPIO tree. --- sound/soc/codecs/wm8962.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b563d6746ac4..1577d2e62e76 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -18,7 +18,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/gcd.h> -#include <linux/gpio.h> +#include <linux/gpio/driver.h> #include <linux/i2c.h> #include <linux/input.h> #include <linux/pm_runtime.h> @@ -3308,14 +3308,9 @@ static void wm8962_set_gpio_mode(struct wm8962_priv *wm8962, int gpio) } #ifdef CONFIG_GPIOLIB -static inline struct wm8962_priv *gpio_to_wm8962(struct gpio_chip *chip) -{ - return container_of(chip, struct wm8962_priv, gpio_chip); -} - static int wm8962_gpio_request(struct gpio_chip *chip, unsigned offset) { - struct wm8962_priv *wm8962 = gpio_to_wm8962(chip); + struct wm8962_priv *wm8962 = gpiochip_get_data(chip); /* The WM8962 GPIOs aren't linearly numbered. For simplicity * we export linear numbers and error out if the unsupported @@ -3338,7 +3333,7 @@ static int wm8962_gpio_request(struct gpio_chip *chip, unsigned offset) static void wm8962_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8962_priv *wm8962 = gpio_to_wm8962(chip); + struct wm8962_priv *wm8962 = gpiochip_get_data(chip); struct snd_soc_codec *codec = wm8962->codec; snd_soc_update_bits(codec, WM8962_GPIO_BASE + offset, @@ -3348,7 +3343,7 @@ static void wm8962_gpio_set(struct gpio_chip *chip, unsigned offset, int value) static int wm8962_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8962_priv *wm8962 = gpio_to_wm8962(chip); + struct wm8962_priv *wm8962 = gpiochip_get_data(chip); struct snd_soc_codec *codec = wm8962->codec; int ret, val; @@ -3387,7 +3382,7 @@ static void wm8962_init_gpio(struct snd_soc_codec *codec) else wm8962->gpio_chip.base = -1; - ret = gpiochip_add(&wm8962->gpio_chip); + ret = gpiochip_add_data(&wm8962->gpio_chip, wm8962); if (ret != 0) dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); } -- 2.4.3

3 2

[alsa-devel] [PATCH 178/182] ASoC: wm8903: use gpiochip data pointer
by Linus Walleij 12 Dec '15

12 Dec '15

This makes the driver use the data pointer added to the gpio_chip to store a pointer to the state container instead of relying on container_of(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Liam/Mark: please ACK this so I can merge it into the GPIO tree. --- sound/soc/codecs/wm8903.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 2512def0d349..d1315ce5f0a9 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -20,7 +20,7 @@ #include <linux/init.h> #include <linux/completion.h> #include <linux/delay.h> -#include <linux/gpio.h> +#include <linux/gpio/driver.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -1766,11 +1766,6 @@ static int wm8903_resume(struct snd_soc_codec *codec) } #ifdef CONFIG_GPIOLIB -static inline struct wm8903_priv *gpio_to_wm8903(struct gpio_chip *chip) -{ - return container_of(chip, struct wm8903_priv, gpio_chip); -} - static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset) { if (offset >= WM8903_NUM_GPIO) @@ -1781,7 +1776,7 @@ static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset) static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { - struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct wm8903_priv *wm8903 = gpiochip_get_data(chip); unsigned int mask, val; int ret; @@ -1799,7 +1794,7 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) { - struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct wm8903_priv *wm8903 = gpiochip_get_data(chip); unsigned int reg; regmap_read(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, &reg); @@ -1810,7 +1805,7 @@ static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) static int wm8903_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct wm8903_priv *wm8903 = gpiochip_get_data(chip); unsigned int mask, val; int ret; @@ -1828,7 +1823,7 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct wm8903_priv *wm8903 = gpiochip_get_data(chip); regmap_update_bits(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, WM8903_GP1_LVL_MASK, @@ -1860,7 +1855,7 @@ static void wm8903_init_gpio(struct wm8903_priv *wm8903) else wm8903->gpio_chip.base = -1; - ret = gpiochip_add(&wm8903->gpio_chip); + ret = gpiochip_add_data(&wm8903->gpio_chip, wm8903); if (ret != 0) dev_err(wm8903->dev, "Failed to add GPIOs: %d\n", ret); } -- 2.4.3

3 2

[alsa-devel] [PATCH 177/182] ASoC: wm5100: use gpiochip data pointer
by Linus Walleij 12 Dec '15

12 Dec '15

This makes the driver use the data pointer added to the gpio_chip to store a pointer to the state container instead of relying on container_of(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Liam/Mark: please ACK this so I can merge it into the GPIO tree. --- sound/soc/codecs/wm5100.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 171a23ddd15d..512a9d25fe6f 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -17,6 +17,7 @@ #include <linux/export.h> #include <linux/pm.h> #include <linux/gcd.h> +#include <linux/gpio/driver.h> #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/pm_runtime.h> @@ -2236,14 +2237,9 @@ static irqreturn_t wm5100_edge_irq(int irq, void *data) } #ifdef CONFIG_GPIOLIB -static inline struct wm5100_priv *gpio_to_wm5100(struct gpio_chip *chip) -{ - return container_of(chip, struct wm5100_priv, gpio_chip); -} - static void wm5100_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct wm5100_priv *wm5100 = gpiochip_get_data(chip); regmap_update_bits(wm5100->regmap, WM5100_GPIO_CTRL_1 + offset, WM5100_GP1_LVL, !!value << WM5100_GP1_LVL_SHIFT); @@ -2252,7 +2248,7 @@ static void wm5100_gpio_set(struct gpio_chip *chip, unsigned offset, int value) static int wm5100_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct wm5100_priv *wm5100 = gpiochip_get_data(chip); int val, ret; val = (1 << WM5100_GP1_FN_SHIFT) | (!!value << WM5100_GP1_LVL_SHIFT); @@ -2268,7 +2264,7 @@ static int wm5100_gpio_direction_out(struct gpio_chip *chip, static int wm5100_gpio_get(struct gpio_chip *chip, unsigned offset) { - struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct wm5100_priv *wm5100 = gpiochip_get_data(chip); unsigned int reg; int ret; @@ -2281,7 +2277,7 @@ static int wm5100_gpio_get(struct gpio_chip *chip, unsigned offset) static int wm5100_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { - struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct wm5100_priv *wm5100 = gpiochip_get_data(chip); return regmap_update_bits(wm5100->regmap, WM5100_GPIO_CTRL_1 + offset, WM5100_GP1_FN_MASK | WM5100_GP1_DIR, @@ -2313,7 +2309,7 @@ static void wm5100_init_gpio(struct i2c_client *i2c) else wm5100->gpio_chip.base = -1; - ret = gpiochip_add(&wm5100->gpio_chip); + ret = gpiochip_add_data(&wm5100->gpio_chip, wm5100); if (ret != 0) dev_err(&i2c->dev, "Failed to add GPIOs: %d\n", ret); } -- 2.4.3

3 2

[alsa-devel] [PATCH 176/182] ASoC: rt5677: use gpiochip data pointer
by Linus Walleij 12 Dec '15

12 Dec '15

This makes the driver use the data pointer added to the gpio_chip to store a pointer to the state container instead of relying on container_of(). Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Liam/Mark: please ACK this so I can merge it into the GPIO tree. --- sound/soc/codecs/rt5677.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 1f590b5a6718..89dda8962c7c 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4500,14 +4500,9 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec, } #ifdef CONFIG_GPIOLIB -static inline struct rt5677_priv *gpio_to_rt5677(struct gpio_chip *chip) -{ - return container_of(chip, struct rt5677_priv, gpio_chip); -} - static void rt5677_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct rt5677_priv *rt5677 = gpiochip_get_data(chip); switch (offset) { case RT5677_GPIO1 ... RT5677_GPIO5: @@ -4528,7 +4523,7 @@ static void rt5677_gpio_set(struct gpio_chip *chip, unsigned offset, int value) static int rt5677_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct rt5677_priv *rt5677 = gpiochip_get_data(chip); switch (offset) { case RT5677_GPIO1 ... RT5677_GPIO5: @@ -4552,7 +4547,7 @@ static int rt5677_gpio_direction_out(struct gpio_chip *chip, static int rt5677_gpio_get(struct gpio_chip *chip, unsigned offset) { - struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct rt5677_priv *rt5677 = gpiochip_get_data(chip); int value, ret; ret = regmap_read(rt5677->regmap, RT5677_GPIO_ST, &value); @@ -4564,7 +4559,7 @@ static int rt5677_gpio_get(struct gpio_chip *chip, unsigned offset) static int rt5677_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { - struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct rt5677_priv *rt5677 = gpiochip_get_data(chip); switch (offset) { case RT5677_GPIO1 ... RT5677_GPIO5: @@ -4618,7 +4613,7 @@ static void rt5677_gpio_config(struct rt5677_priv *rt5677, unsigned offset, static int rt5677_to_irq(struct gpio_chip *chip, unsigned offset) { - struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct rt5677_priv *rt5677 = gpiochip_get_data(chip); struct regmap_irq_chip_data *data = rt5677->irq_data; int irq; @@ -4677,7 +4672,7 @@ static void rt5677_init_gpio(struct i2c_client *i2c) rt5677->gpio_chip.parent = &i2c->dev; rt5677->gpio_chip.base = -1; - ret = gpiochip_add(&rt5677->gpio_chip); + ret = gpiochip_add_data(&rt5677->gpio_chip, rt5677); if (ret != 0) dev_err(&i2c->dev, "Failed to add GPIOs: %d\n", ret); } -- 2.4.3

2 1

[alsa-devel] [PATCH v2 00/13] ASoC: Intel: Skylake: Update to SKL driver
by Vinod Koul 12 Dec '15

12 Dec '15

This update bring in new APIs in core and usage of those in SKL driver. The series is broadly of three parts - first two are fixes on the driver - then we add DMA resume capability in HDA core and use it to save restore DMA link values and add DMA resume capability in driver - lastly, we add support for MISCBDCGE configuration in core and use it while resetting and also for phrase detection per HW recommendation This should be merged thru ASoC tree due to obvious dependencies. Takashi would need your ack on core patches changes in v2: - move msicbdge api to driver and as a consequence sqash into into first use - fix lkp warning Dharageswari.R (1): ASoC: Intel: Skylake: Use CGCTL.MISCBDCGE for Phrase detection notification Jayachandran B (2): ALSA: hdac: Increase timeout value for link power check ASoC: Intel: Skylake: fix reset controller sequencing Jeeja KP (8): ASoC: Intel: Skylake: Clear stream registers before stream setup ASoC: Intel: Skylake: Fix to set pipe state to invalid when deleting ALSA: hdac: Add support for hda DMA Resume capability ALSA: hdac: couple the hda DMA stream in cleanup ASoC: Intel: Skylake: enable interrupt as wake source in active suspend ASoC: Intel: Skylake: Add DMA resume position in Trigger resume/suspend ASoC: Intel: Skylake: Reconfigure Link stream on suspend/resume ASoC: Intel: Skylake: Add Resume capability in PCM info. Vinod Koul (2): ALSA: hdac: add snd_hdac_ext_bus_link_power_up_all ASoC: Intel: Skylake: manage link power in active suspend include/sound/hda_register.h | 9 +++++ include/sound/hdaudio_ext.h | 15 +++++++ sound/hda/ext/hdac_ext_controller.c | 29 ++++++++++++- sound/hda/ext/hdac_ext_stream.c | 72 +++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-messages.c | 2 + sound/soc/intel/skylake/skl-pcm.c | 56 ++++++++++++++++++++----- sound/soc/intel/skylake/skl-sst-cldma.c | 38 ++++++++++------- sound/soc/intel/skylake/skl-sst-ipc.c | 15 +++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 3 ++ sound/soc/intel/skylake/skl.c | 51 +++++++++++++++++++++-- sound/soc/intel/skylake/skl.h | 5 +++ 11 files changed, 265 insertions(+), 30 deletions(-) -- 1.9.1

1 13

Re: [alsa-devel] [PATCH 05/14] ALSA: hdac: Add MISCBDCGE support
by Vinod Koul 12 Dec '15

12 Dec '15

On Sat, Dec 12, 2015 at 05:14:48AM +0800, kbuild test robot wrote: > Hi Jayachandran, > > [auto build test WARNING on asoc/for-next] > [also build test WARNING on next-20151211] > [cannot apply to sound/for-next v4.4-rc4] > > url: https://github.com/0day-ci/linux/commits/Vinod-Koul/ASoC-Intel-Skylake-Upda… > base: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next > config: mn10300-allmodconfig (attached as .config) > reproduce: > wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross > chmod +x ~/bin/make.cross > # save the attached .config to linux build tree > make.cross ARCH=mn10300 > > All warnings (new ones prefixed by >>): > > sound/hda/ext/hdac_ext_controller.c: In function 'update_pci_dword': > >> sound/hda/ext/hdac_ext_controller.c:317:7: warning: 'data' is used uninitialized in this function [-Wuninitialized] > data &= ~mask; Thanks, will fix it in v2 -- ~Vinod

1 0

[alsa-devel] [PATCH] ASoC: fsl_asrc: sound is wrong after suspend/resume
by Zidan Wang 12 Dec '15

12 Dec '15

The register ASRCFG is volatile, but some bits need to be recovered after suspend/resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_asrc.c | 6 ++++++ sound/soc/fsl/fsl_asrc.h | 2 ++ 2 files changed, 8 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 3e404ba..942e74bc 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -990,6 +990,9 @@ static int fsl_asrc_suspend(struct device *dev) { struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + regmap_read(asrc_priv->regmap, REG_ASRCFG, + &asrc_priv->regcache_cfg); + regcache_cache_only(asrc_priv->regmap, true); regcache_mark_dirty(asrc_priv->regmap); @@ -1010,6 +1013,9 @@ static int fsl_asrc_resume(struct device *dev) regcache_cache_only(asrc_priv->regmap, false); regcache_sync(asrc_priv->regmap); + regmap_update_bits(asrc_priv->regmap, REG_ASRCFG, + 0x1FFFC0, asrc_priv->regcache_cfg); + /* Restart enabled pairs */ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, ASRCTR_ASRCEi_ALL_MASK, asrctr); diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index 68802cd..3695ca9 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -453,6 +453,8 @@ struct fsl_asrc { int asrc_rate; int asrc_width; + + u32 regcache_cfg; }; extern struct snd_soc_platform_driver fsl_asrc_platform; -- 1.9.1

2 5

[alsa-devel] [PATCH v2] ASoC: fsl_sai: fix no frame clk in master mode
by Zidan Wang 12 Dec '15

12 Dec '15

After several open/close sai test with ctrl+c, there will be I/O error. The SAI can't work anymore, can't recover. There will be no frame clock. With adding the software reset in trigger stop, the issue can be fixed. This is a hardware bug/errata and reset is the only option. According to the reference manual, the software reset doesn't reset any control register but only internal hardware logics such as bit clock generator, status flags, and FIFO pointers. (Our purpose is just to reset the clock generator while the software reset is the only way to do that.) Since slave mode doesn't use the clock generator, only apply the reset procedure to the master mode. For asynchronous mode, TX will not be reset when RX is still running. In this case, i can't reproduce this issue. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v1->v2: 1) refine the commit log and comments 2) Since slave mode doesn't use the clock generator, only apply the reset procedure to the master mode. sound/soc/fsl/fsl_sai.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 8832901..a55a7de 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -564,6 +564,17 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); regmap_update_bits(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); + + if (!sai->is_slave_mode) { + /* Software Reset for both Tx and Rx */ + regmap_write(sai->regmap, + FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, + FSL_SAI_RCSR, FSL_SAI_CSR_SR); + /* Clear SR bit to finish the reset */ + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + } } break; default: -- 1.9.1

2 1

[alsa-devel] Available frames > buffer size
by Rob Duncan 11 Dec '15

11 Dec '15

Is it ever correct for snd_pcm_avail() to return a value that’s greater than that returned by snd_pcm_hw_params_get_buffer_size()? I’m trying to track down some unexpected underruns in my application and I notice that when they occur I also see that the available frames count is wrong. For example here are a couple of lines from my logging: 2015-12-08T23:17:23.118817-08:00 X ? 2080 + 2080 2015-12-08T23:17:23.119524-08:00 X = 2080 ? 4126 The first log line: here I asked the device how many frames are available and it responds with 2080; I accordingly write 2080 frames. The second log line: the writei() call returns 2080. I immediately ask how many frames are available now (I usually get a small number in response ~ 20-30) but this time I get an XRUN. I call snd_pcm_recover() and then snd_pcm_avail() again and now I get 4126 frames. As you can see the time between these two logs is less than 1 millisecond, so I don’t understand how the PCM can go from 2080 frames available to >4096 frames available so quickly. Is this evidence of a bug in the device driver? My application is using a single channel dshare PCM (rt) that has an 8-channel device underneath. Here’s the configuration fragment: pcm_slave.cid_dsp { pcm "hw:0,0" channels 8 rate 44100 } pcm.rt_channel { type dshare ipc_key 1025 ipc_gid audio ipc_perm 0660 slave cid_dsp bindings { 0 5 } } pcm.rt { type plug slave.pcm rt_channel } The underlying hardware looks like this: access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 8 rate: 44100 (44100/1) period_size: 1024 buffer_size: 4096 tstamp_mode: ENABLE period_step: 1 avail_min: 1 start_threshold: 1 stop_threshold: 1073741824 silence_threshold: 0 silence_size: 0 boundary: 1073741824 Thanks for any insight and help, Rob.

2 3

[alsa-devel] [PATCH] ALSA: usb-audio: constify usb_protocol_ops structures
by Julia Lawall 11 Dec '15

11 Dec '15

The usb_protocol_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/usb/midi.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index ee212e7..cc39f63 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -112,7 +112,7 @@ struct snd_usb_midi { struct usb_interface *iface; const struct snd_usb_audio_quirk *quirk; struct snd_rawmidi *rmidi; - struct usb_protocol_ops *usb_protocol_ops; + const struct usb_protocol_ops *usb_protocol_ops; struct list_head list; struct timer_list error_timer; spinlock_t disc_lock; @@ -671,31 +671,32 @@ static void snd_usbmidi_standard_output(struct snd_usb_midi_out_endpoint *ep, } } -static struct usb_protocol_ops snd_usbmidi_standard_ops = { +static const struct usb_protocol_ops snd_usbmidi_standard_ops = { .input = snd_usbmidi_standard_input, .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, }; -static struct usb_protocol_ops snd_usbmidi_midiman_ops = { +static const struct usb_protocol_ops snd_usbmidi_midiman_ops = { .input = snd_usbmidi_midiman_input, .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_midiman_packet, }; -static struct usb_protocol_ops snd_usbmidi_maudio_broken_running_status_ops = { +static const +struct usb_protocol_ops snd_usbmidi_maudio_broken_running_status_ops = { .input = snd_usbmidi_maudio_broken_running_status_input, .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, }; -static struct usb_protocol_ops snd_usbmidi_cme_ops = { +static const struct usb_protocol_ops snd_usbmidi_cme_ops = { .input = snd_usbmidi_cme_input, .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, }; -static struct usb_protocol_ops snd_usbmidi_ch345_broken_sysex_ops = { +static const struct usb_protocol_ops snd_usbmidi_ch345_broken_sysex_ops = { .input = ch345_broken_sysex_input, .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, @@ -795,7 +796,7 @@ static void snd_usbmidi_akai_output(struct snd_usb_midi_out_endpoint *ep, } } -static struct usb_protocol_ops snd_usbmidi_akai_ops = { +static const struct usb_protocol_ops snd_usbmidi_akai_ops = { .input = snd_usbmidi_akai_input, .output = snd_usbmidi_akai_output, }; @@ -835,7 +836,7 @@ static void snd_usbmidi_novation_output(struct snd_usb_midi_out_endpoint *ep, urb->transfer_buffer_length = 2 + count; } -static struct usb_protocol_ops snd_usbmidi_novation_ops = { +static const struct usb_protocol_ops snd_usbmidi_novation_ops = { .input = snd_usbmidi_novation_input, .output = snd_usbmidi_novation_output, }; @@ -867,7 +868,7 @@ static void snd_usbmidi_raw_output(struct snd_usb_midi_out_endpoint *ep, urb->transfer_buffer_length = count; } -static struct usb_protocol_ops snd_usbmidi_raw_ops = { +static const struct usb_protocol_ops snd_usbmidi_raw_ops = { .input = snd_usbmidi_raw_input, .output = snd_usbmidi_raw_output, }; @@ -883,7 +884,7 @@ static void snd_usbmidi_ftdi_input(struct snd_usb_midi_in_endpoint *ep, snd_usbmidi_input_data(ep, 0, buffer + 2, buffer_length - 2); } -static struct usb_protocol_ops snd_usbmidi_ftdi_ops = { +static const struct usb_protocol_ops snd_usbmidi_ftdi_ops = { .input = snd_usbmidi_ftdi_input, .output = snd_usbmidi_raw_output, }; @@ -927,7 +928,7 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, urb->transfer_buffer_length = ep->max_transfer; } -static struct usb_protocol_ops snd_usbmidi_122l_ops = { +static const struct usb_protocol_ops snd_usbmidi_122l_ops = { .input = snd_usbmidi_us122l_input, .output = snd_usbmidi_us122l_output, }; @@ -1060,7 +1061,7 @@ static void snd_usbmidi_emagic_output(struct snd_usb_midi_out_endpoint *ep, urb->transfer_buffer_length = ep->max_transfer - buf_free; } -static struct usb_protocol_ops snd_usbmidi_emagic_ops = { +static const struct usb_protocol_ops snd_usbmidi_emagic_ops = { .input = snd_usbmidi_emagic_input, .output = snd_usbmidi_emagic_output, .init_out_endpoint = snd_usbmidi_emagic_init_out,

3 2

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Fix XDATA check in mcasp_start_tx
by Peter Ujfalusi 11 Dec '15

11 Dec '15

The condition for checking for XDAT being cleared was not correct. Fixes: 36bcecd0a73eb ("ASoC: davinci-mcasp: Correct TX start sequence") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c1c9c2e3525b..2ccb8bccc9d4 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -223,8 +223,8 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) /* wait for XDATA to be cleared */ cnt = 0; - while (!(mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG) & - ~XRDATA) && (cnt < 100000)) + while ((mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG) & XRDATA) && + (cnt < 100000)) cnt++; /* Release TX state machine */ -- 2.6.4

1 0

Re: [alsa-devel] [Intel-gfx] [PATCH V4 1/2] dp/mst: add SDP stream support
by Libin Yang 11 Dec '15

11 Dec '15

Add Takashi and ALSA mailing list Best Regards, Libin On 12/02/2015 02:09 PM, libin.yang(a)linux.intel.com wrote: > From: Libin Yang <libin.yang(a)linux.intel.com> > > This adds code to initialise the SDP streams > for a sink in the simplest ordering. > > I've no idea how you'd want to control the > ordering at this level, so don't bother > until someone comes up with a use case. > > Reviewed-by: Ander Conselvan de Oliveira <conselvan2(a)gmail.com> > Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> > Signed-off-by: Dave Airlie <airlied(a)redhat.com> > --- > drivers/gpu/drm/drm_dp_mst_topology.c | 38 ++++++++++++++++++++++++++++++++--- > include/drm/drm_dp_mst_helper.h | 7 +++++-- > 2 files changed, 40 insertions(+), 5 deletions(-) > > diff --git a/drivers/gpu/drm/drm_dp_mst_topology.c b/drivers/gpu/drm/drm_dp_mst_topology.c > index 809959d..f50eb7b 100644 > --- a/drivers/gpu/drm/drm_dp_mst_topology.c > +++ b/drivers/gpu/drm/drm_dp_mst_topology.c > @@ -666,7 +666,9 @@ static int build_enum_path_resources(struct drm_dp_sideband_msg_tx *msg, int por > } > > static int build_allocate_payload(struct drm_dp_sideband_msg_tx *msg, int port_num, > - u8 vcpi, uint16_t pbn) > + u8 vcpi, uint16_t pbn, > + u8 number_sdp_streams, > + u8 *sdp_stream_sink) > { > struct drm_dp_sideband_msg_req_body req; > memset(&req, 0, sizeof(req)); > @@ -674,6 +676,9 @@ static int build_allocate_payload(struct drm_dp_sideband_msg_tx *msg, int port_n > req.u.allocate_payload.port_number = port_num; > req.u.allocate_payload.vcpi = vcpi; > req.u.allocate_payload.pbn = pbn; > + req.u.allocate_payload.number_sdp_streams = number_sdp_streams; > + memcpy(req.u.allocate_payload.sdp_stream_sink, sdp_stream_sink, > + number_sdp_streams); > drm_dp_encode_sideband_req(&req, msg); > msg->path_msg = true; > return 0; > @@ -1562,6 +1567,8 @@ static int drm_dp_payload_send_msg(struct drm_dp_mst_topology_mgr *mgr, > struct drm_dp_sideband_msg_tx *txmsg; > struct drm_dp_mst_branch *mstb; > int len, ret; > + u8 sinks[DRM_DP_MAX_SDP_STREAMS]; > + int i; > > mstb = drm_dp_get_validated_mstb_ref(mgr, port->parent); > if (!mstb) > @@ -1573,10 +1580,13 @@ static int drm_dp_payload_send_msg(struct drm_dp_mst_topology_mgr *mgr, > goto fail_put; > } > > + for (i = 0; i < port->num_sdp_streams; i++) > + sinks[i] = i; > + > txmsg->dst = mstb; > len = build_allocate_payload(txmsg, port->port_num, > id, > - pbn); > + pbn, port->num_sdp_streams, sinks); > > drm_dp_queue_down_tx(mgr, txmsg); > > @@ -2259,6 +2269,27 @@ out: > EXPORT_SYMBOL(drm_dp_mst_detect_port); > > /** > + * drm_dp_mst_port_has_audio() - Check whether port has audio capability or not > + * @mgr: manager for this port > + * @port: unverified pointer to a port. > + * > + * This returns whether the port supports audio or not. > + */ > +bool drm_dp_mst_port_has_audio(struct drm_dp_mst_topology_mgr *mgr, > + struct drm_dp_mst_port *port) > +{ > + bool ret = false; > + > + port = drm_dp_get_validated_port_ref(mgr, port); > + if (!port) > + return ret; > + ret = port->has_audio; > + drm_dp_put_port(port); > + return ret; > +} > +EXPORT_SYMBOL(drm_dp_mst_port_has_audio); > + > +/** > * drm_dp_mst_get_edid() - get EDID for an MST port > * @connector: toplevel connector to get EDID for > * @mgr: manager for this port > @@ -2283,6 +2314,7 @@ struct edid *drm_dp_mst_get_edid(struct drm_connector *connector, struct drm_dp_ > edid = drm_get_edid(connector, &port->aux.ddc); > drm_mode_connector_set_tile_property(connector); > } > + port->has_audio = drm_detect_monitor_audio(edid); > drm_dp_put_port(port); > return edid; > } > @@ -2566,7 +2598,7 @@ static void drm_dp_mst_dump_mstb(struct seq_file *m, > > seq_printf(m, "%smst: %p, %d\n", prefix, mstb, mstb->num_ports); > list_for_each_entry(port, &mstb->ports, next) { > - seq_printf(m, "%sport: %d: ddps: %d ldps: %d, %p, conn: %p\n", prefix, port->port_num, port->ddps, port->ldps, port, port->connector); > + seq_printf(m, "%sport: %d: ddps: %d ldps: %d, sdp: %d/%d, %p, conn: %p\n", prefix, port->port_num, port->ddps, port->ldps, port->num_sdp_streams, port->num_sdp_stream_sinks, port, port->connector); > if (port->mstb) > drm_dp_mst_dump_mstb(m, port->mstb); > } > diff --git a/include/drm/drm_dp_mst_helper.h b/include/drm/drm_dp_mst_helper.h > index 5340099..74b5888 100644 > --- a/include/drm/drm_dp_mst_helper.h > +++ b/include/drm/drm_dp_mst_helper.h > @@ -94,6 +94,7 @@ struct drm_dp_mst_port { > struct drm_dp_mst_topology_mgr *mgr; > > struct edid *cached_edid; /* for DP logical ports - make tiling work */ > + bool has_audio; > }; > > /** > @@ -215,13 +216,13 @@ struct drm_dp_sideband_msg_rx { > struct drm_dp_sideband_msg_hdr initial_hdr; > }; > > - > +#define DRM_DP_MAX_SDP_STREAMS 16 > struct drm_dp_allocate_payload { > u8 port_number; > u8 number_sdp_streams; > u8 vcpi; > u16 pbn; > - u8 sdp_stream_sink[8]; > + u8 sdp_stream_sink[DRM_DP_MAX_SDP_STREAMS]; > }; > > struct drm_dp_allocate_payload_ack_reply { > @@ -484,6 +485,8 @@ int drm_dp_mst_hpd_irq(struct drm_dp_mst_topology_mgr *mgr, u8 *esi, bool *handl > > enum drm_connector_status drm_dp_mst_detect_port(struct drm_connector *connector, struct drm_dp_mst_topology_mgr *mgr, struct drm_dp_mst_port *port); > > +bool drm_dp_mst_port_has_audio(struct drm_dp_mst_topology_mgr *mgr, > + struct drm_dp_mst_port *port); > struct edid *drm_dp_mst_get_edid(struct drm_connector *connector, struct drm_dp_mst_topology_mgr *mgr, struct drm_dp_mst_port *port); > > >

1 0

[alsa-devel] [PATCH v5] sun4i-codec: add inputs
by Danny Milosavljevic 11 Dec '15

11 Dec '15

Hi, this is the fifth version of the patch that adds inputs to sun4i-codec. (Previous versions have been on the linux-sunxi mailing list only) The inputs added are: - FM-In Left and Right - Line-In Left and Right - Mic1-In - Mic2-In Changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. I successfully tested it on an A20 board using alsamixer, headphones, a radio and my ears. Note that because of missing capturing support I tested only the mixing, for Mic, Line, and FM. The patches are on top of <git://git.kernel.org/pub/scm/linux/kernel/git/mripard/linux.git>, branch "sunxi/for-next". Regards, Danny Danny (1): b/sound/soc/sunxi/sun4i-codec.c | 153 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 150 insertions(+), 3 deletions(-) Signed-off-by: Danny Milosavljevic <dannym+a(a)scratchpost.org> --- diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index bcbf4da..106b568 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -57,9 +57,20 @@ #define SUN4I_CODEC_DAC_ACTL_DACAENR (31) #define SUN4I_CODEC_DAC_ACTL_DACAENL (30) #define SUN4I_CODEC_DAC_ACTL_MIXEN (29) +#define SUN4I_CODEC_DAC_ACTL_LNG (26) +#define SUN4I_CODEC_DAC_ACTL_FMG (23) +#define SUN4I_CODEC_DAC_ACTL_MICG (20) +#define SUN4I_CODEC_DAC_ACTL_LLNS (19) +#define SUN4I_CODEC_DAC_ACTL_RLNS (18) +#define SUN4I_CODEC_DAC_ACTL_LFMS (17) +#define SUN4I_CODEC_DAC_ACTL_RFMS (16) #define SUN4I_CODEC_DAC_ACTL_LDACLMIXS (15) #define SUN4I_CODEC_DAC_ACTL_RDACRMIXS (14) #define SUN4I_CODEC_DAC_ACTL_LDACRMIXS (13) +#define SUN4I_CODEC_DAC_ACTL_MIC1LS (12) +#define SUN4I_CODEC_DAC_ACTL_MIC1RS (11) +#define SUN4I_CODEC_DAC_ACTL_MIC2LS (10) +#define SUN4I_CODEC_DAC_ACTL_MIC2RS (9) #define SUN4I_CODEC_DAC_ACTL_DACPAS (8) #define SUN4I_CODEC_DAC_ACTL_MIXPAS (7) #define SUN4I_CODEC_DAC_ACTL_PA_MUTE (6) @@ -84,8 +95,11 @@ #define SUN4I_CODEC_ADC_ACTL_PREG1EN (29) #define SUN4I_CODEC_ADC_ACTL_PREG2EN (28) #define SUN4I_CODEC_ADC_ACTL_VMICEN (27) +#define SUN4I_CODEC_ADC_ACTL_PREG1_A10 (25) +#define SUN4I_CODEC_ADC_ACTL_PREG2_A10 (23) #define SUN4I_CODEC_ADC_ACTL_VADCG (20) #define SUN4I_CODEC_ADC_ACTL_ADCIS (17) +#define SUN4I_CODEC_ADC_ACTL_LNRDF (16) #define SUN4I_CODEC_ADC_ACTL_PA_EN (4) #define SUN4I_CODEC_ADC_ACTL_DDE (3) #define SUN4I_CODEC_ADC_DEBUG (0x2c) @@ -94,7 +108,18 @@ #define SUN4I_CODEC_DAC_TXCNT (0x30) #define SUN4I_CODEC_ADC_RXCNT (0x34) #define SUN4I_CODEC_AC_SYS_VERI (0x38) -#define SUN4I_CODEC_AC_MIC_PHONE_CAL (0x3c) + +/* MIC_PHONE_CAL register offsets and bit fields (A20 only) */ +#define SUN7I_CODEC_AC_MIC_PHONE_CAL (0x3c) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG1 (29) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG2 (26) +/* note: no idea where the output pins for the following are. */ +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTG (5) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTEN (4) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS3 (3) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS2 (2) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS1 (1) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS0 (0) struct sun4i_codec { struct device *dev; @@ -402,16 +427,94 @@ static const struct snd_kcontrol_new sun4i_codec_pa_mute = SUN4I_CODEC_DAC_ACTL_PA_MUTE, 1, 0); static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1); +static DECLARE_TLV_DB_SCALE(sun4i_codec_linein_loopback_gain_scale, -150, 150, 0); +static DECLARE_TLV_DB_SCALE(sun4i_codec_fmin_loopback_gain_scale, -450, 150, 0); +static DECLARE_TLV_DB_SCALE(sun4i_codec_micin_loopback_gain_scale, -450, 150, 0); +static DECLARE_TLV_DB_RANGE(sun7i_codec_micin_preamp_gain_scale, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0) +); +static DECLARE_TLV_DB_RANGE(sun4i_codec_micin_preamp_gain_scale_a10, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(3500, 300, 0) +); + +struct snd_kcontrol_new sun7ionly_codec_widgets[] = { + SOC_SINGLE_TLV("Mic1-In Boost Gain", + SUN7I_CODEC_AC_MIC_PHONE_CAL, + SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG1, + 7, + 0, + sun7i_codec_micin_preamp_gain_scale), + SOC_SINGLE_TLV("Mic2-In Boost Gain", + SUN7I_CODEC_AC_MIC_PHONE_CAL, + SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG2, + 7, + 0, + sun7i_codec_micin_preamp_gain_scale), +}; -static const struct snd_kcontrol_new sun4i_codec_widgets[] = { +/* Note: this is modified by the sun4i_codec_probe() function. */ +static struct snd_kcontrol_new sun4i_codec_widgets[] = { SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0, sun4i_codec_pa_volume_scale), + /* Line-In, FM-In, Mic1-In, Mic2-In */ + SOC_SINGLE_TLV("Line-In Playback Gain", + SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LNG, + 1, + 0, + sun4i_codec_linein_loopback_gain_scale), + SOC_SINGLE_TLV("FM-In Playback Gain", + SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_FMG, + 3, + 0, + sun4i_codec_fmin_loopback_gain_scale), + SOC_SINGLE_TLV("Mic-In Playback Gain", + SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MICG, + 7, + 0, + sun4i_codec_micin_loopback_gain_scale), + SOC_SINGLE_TLV("Mic1-In Boost Gain", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG1_A10, + 3, + 0, + sun4i_codec_micin_preamp_gain_scale_a10), + SOC_SINGLE_TLV("Mic2-In Boost Gain", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG2_A10, + 3, + 0, + sun4i_codec_micin_preamp_gain_scale_a10), + SOC_SINGLE_TLV("Mic1-In Boost Switch", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG1EN, + 1, + 0, + NULL), + SOC_SINGLE_TLV("Mic2-In Boost Switch", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG2EN, + 1, + 0, + NULL), }; static const struct snd_kcontrol_new sun4i_codec_left_mixer_controls[] = { SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_LDACLMIXS, 1, 0), + SOC_DAPM_SINGLE("Left Line-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LLNS, 1, 0), + SOC_DAPM_SINGLE("Left FM-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LFMS, 1, 0), + SOC_DAPM_SINGLE("Mic1-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC1LS, 1, 0), + SOC_DAPM_SINGLE("Mic2-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC2LS, 1, 0), }; static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = { @@ -419,6 +522,14 @@ static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = { SUN4I_CODEC_DAC_ACTL_RDACRMIXS, 1, 0), SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_LDACRMIXS, 1, 0), + SOC_DAPM_SINGLE("Right Line-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_RLNS, 1, 0), + SOC_DAPM_SINGLE("Right FM-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_RFMS, 1, 0), + SOC_DAPM_SINGLE("Mic1-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC1RS, 1, 0), + SOC_DAPM_SINGLE("Mic2-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC2RS, 1, 0), }; static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = { @@ -462,8 +573,16 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HP Right"), SND_SOC_DAPM_OUTPUT("HP Left"), + + SND_SOC_DAPM_INPUT("Line-In Right"), + SND_SOC_DAPM_INPUT("Line-In Left"), + SND_SOC_DAPM_INPUT("FM-In Right"), + SND_SOC_DAPM_INPUT("FM-In Left"), + SND_SOC_DAPM_INPUT("Mic1-In"), + SND_SOC_DAPM_INPUT("Mic2-In"), }; +/* {sink, control, source} */ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { /* Left DAC Routes */ { "Left DAC", NULL, "DAC" }, @@ -490,6 +609,20 @@ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" }, { "HP Right", NULL, "Pre-Amplifier Mute" }, { "HP Left", NULL, "Pre-Amplifier Mute" }, + + /* Line-In, FM-In */ + { "Right Mixer", "Right Line-In Playback Switch", "Line-In Right" }, + { "Left Mixer", "Left Line-In Playback Switch", "Line-In Left" }, + { "Right Mixer", "Right FM-In Playback Switch", "FM-In Right" }, + { "Left Mixer", "Left FM-In Playback Switch", "FM-In Left" }, + + /* Mic1-In */ + { "Right Mixer", "Mic1-In Playback Switch", "Mic1-In" }, + { "Left Mixer", "Mic1-In Playback Switch", "Mic1-In" }, + + /* Mic2-In */ + { "Right Mixer", "Mic2-In Playback Switch", "Mic2-In" }, + { "Left Mixer", "Mic2-In Playback Switch", "Mic2-In" }, }; static struct snd_soc_codec_driver sun4i_codec_codec = { @@ -537,7 +670,7 @@ static const struct regmap_config sun4i_codec_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, - .max_register = SUN4I_CODEC_AC_MIC_PHONE_CAL, + .max_register = SUN7I_CODEC_AC_MIC_PHONE_CAL, }; static const struct of_device_id sun4i_codec_of_match[] = { @@ -594,6 +727,20 @@ static int sun4i_codec_probe(struct platform_device *pdev) void __iomem *base; int ret; + if (of_device_is_compatible(pdev->dev.of_node, "allwinner,sun7i-a20-codec")) { + size_t i, j; + for (i = 0; i < ARRAY_SIZE(sun7ionly_codec_widgets); ++i) { + struct snd_kcontrol_new *new_kcontrol; + new_kcontrol = &sun7ionly_codec_widgets[i]; + for (j = 0; j < ARRAY_SIZE(sun4i_codec_widgets); ++j) { + struct snd_kcontrol_new *kcontrol; + kcontrol = &sun4i_codec_widgets[j]; + if (strcmp(kcontrol->name, new_kcontrol->name) == 0) + memcpy(kcontrol, new_kcontrol, sizeof(*kcontrol)); + } + } + } + scodec = devm_kzalloc(&pdev->dev, sizeof(*scodec), GFP_KERNEL); if (!scodec) return -ENOMEM;

3 3

[alsa-devel] HDMI Repoll after pm?
by David Henningsson 10 Dec '15

10 Dec '15

Hi Takashi, A recent change in sound.git seems to have introduced some repoll after pm. Is this an intentional change? I don't think it's wrong per se (it stops after 6 repolls), but it breaks my hda-emu tester when "list" does not return the expected input. To reproduce, try running: ./hda-emu codecs/canonical/92hd91bxx-dell-inspiron-5423-ccert-201208-11456 -i 1 > pm .... lots of data .... ** TEST_PM DONE ** HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 JACK report HDMI/DP, status 0 > list 1: HDMI/DP Jack:0 (1) 2: IEC958 Playback Con Mask:0 (1) 3: IEC958 Playback Pro Mask:0 (1) 4: IEC958 Playback Default:0 (1) 5: IEC958 Playback Switch:0 (1) 6: ELD:0 (1) 7: Playback Channel Map:0 (1) HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 JACK report HDMI/DP, status 0 > ...notice how the last two lines are somewhat unexpected output for a "list" command, but I can make my tool more robust in case you confirm this is an expected change. // David

2 5

[alsa-devel] [PATCH 1/2] ALSA: hda - Optimize audio component check in patch_hdmi.c
by Takashi Iwai 10 Dec '15

10 Dec '15

The audio component is enabled only when CONFIG_SND_HDA_I915 is set. Give a dummy macro for allowing the compiler optimize out the relevant codes when this Kconfig isn't set. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 85342d261043..44d0d2374bb6 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -152,8 +152,12 @@ struct hdmi_spec { struct i915_audio_component_audio_ops i915_audio_ops; }; +#ifdef CONFIG_SND_HDA_I915 #define codec_has_acomp(codec) \ ((codec)->bus->core.audio_component != NULL) +#else +#define codec_has_acomp(codec) false +#endif struct hdmi_audio_infoframe { u8 type; /* 0x84 */ -- 2.6.3

1 1

[alsa-devel] [RFC][PATCH 0/3] ALSA: fireface: new driver for RME Fireface series (MIDI only)
by Takashi Sakamoto 10 Dec '15

10 Dec '15

Hi, This patchset is to add a new driver for RME Fireface series. This is still working in progress but currently the new driver support MIDI functionality. Unfortunately, ffado library can disturb this functionality. In RME::Device::init_hardware() function, the library sends a write transaction to 0x0000801003f4 with invalid value as higher part of address in IEEE 1212 or ISO/IEC 13213. This is a worst case I describe in patch 03. A workaround is to load this module after running ffado library. This module sends write transactions with valid values and regain MIDI functionality. I think it better that FFADO developers fixes the bug as long as they doesn't support MIDI functionality. Takashi Sakamoto (3): fireface: add skeleton for RME Fireface series fireface: add transaction support fireface: add support for MIDI functionality sound/firewire/Kconfig | 7 + sound/firewire/Makefile | 1 + sound/firewire/fireface/Makefile | 2 + sound/firewire/fireface/fireface-midi.c | 129 ++++++++ sound/firewire/fireface/fireface-transaction.c | 388 +++++++++++++++++++++++++ sound/firewire/fireface/fireface.c | 151 ++++++++++ sound/firewire/fireface/fireface.h | 73 +++++ 7 files changed, 751 insertions(+) create mode 100644 sound/firewire/fireface/Makefile create mode 100644 sound/firewire/fireface/fireface-midi.c create mode 100644 sound/firewire/fireface/fireface-transaction.c create mode 100644 sound/firewire/fireface/fireface.c create mode 100644 sound/firewire/fireface/fireface.h -- 2.5.0

4 23

[alsa-devel] A root cause of "unable to create IPC shm instance"
by IceBsi 10 Dec '15

10 Dec '15

Dear ALSA developers: I found a root cause of the following issue by using strace: unable to create IPC shm instance Firstly, please see the test code below: 1 #include <stdio.h> 2 #include <unistd.h> 3 #include <sys/ipc.h> 4 #include <sys/shm.h> 5 #include <errno.h> 6 #include <string.h> 7 8 int main(int argc, char *argv[]) 9 { 10 int r = 0; 11 struct shmid_ds buf; 12 key_t ipc_key = (key_t)325559; 13 size_t shmsize = 488; 14 int shmid = -1; 15 mode_t ipc_perm = 0666; 16 void *shmptr = NULL; 17 pid_t pid; 18 19 printf("___%d:%d call shmget(%d,%u,%d)\n", __LINE__, getpid(), (int)(ipc_key), shmsize, IPC_CREAT | ipc_perm); 20 shmid = shmget(ipc_key, shmsize, IPC_CREAT | ipc_perm); 21 printf("___%d:%d ret shmget=%d\n", __LINE__, getpid(), shmid); 22 23 printf("___%d:%d call shmat(%d,0,0)\n", __LINE__, getpid(), shmid); 24 shmptr = shmat(shmid, 0, 0); 25 printf("___%d:%d ret shmat=%p\n", __LINE__, getpid(), shmptr); 26 27 printf("+++%d:%d call fork\n", __LINE__, getpid()); 28 pid = fork(); 29 printf("+++%d:%d ret fork=%d\n", __LINE__, getpid(), (int)pid); 30 31 printf("___%d:%d call shmctl(%d,IPC_STAT,%p)\n", __LINE__, getpid(), shmid, &buf); 32 r = shmctl(shmid, IPC_STAT, &buf); 33 printf("___%d:%d ret shmctl=%d\n", __LINE__, getpid(), r); 34 printf("___%d:%d nattch=%lu\n", __LINE__, getpid(), buf.shm_nattch); 35 36 printf("___%d:%d call shmdt(%p)\n", __LINE__, getpid(), shmptr); 37 r = shmdt(shmptr); 38 printf("___%d:%d ret shmdt=%d\n", __LINE__, getpid(), r); 39 40 printf("___%d:%d call shmctl(%d,IPC_RMID,NULL)\n", __LINE__, getpid(), shmid); 41 r = shmctl(shmid, IPC_RMID, NULL); 42 printf("___%d:%d ret shmctl=%d, %s\n", __LINE__, getpid(), r, strerror(errno)); 43 44 if ((pid_t)0 != pid) { 45 printf("+++%d:%d call wait(NULL)\n", __LINE__, getpid()); 46 pid = wait(NULL); 47 printf("+++%d:%d ret wait=%d\n", __LINE__, getpid(), (int)pid); 48 } 49 50 return 0; 51 } The output of the test code is: ___19:11782 call shmget(325559,488,950) ___21:11782 ret shmget=2850820 ___23:11782 call shmat(2850820,0,0) ___25:11782 ret shmat=0x7f452c223000 +++27:11782 call fork +++29:11782 ret fork=11783 ___31:11782 call shmctl(2850820,IPC_STAT,0x7fffdaf0a750) ___33:11782 ret shmctl=0 ___34:11782 nattch=2 ___36:11782 call shmdt(0x7f452c223000) +++29:11783 ret fork=0 ___38:11782 ret shmdt=0 ___40:11782 call shmctl(2850820,IPC_RMID,NULL) ___31:11783 call shmctl(2850820,IPC_STAT,0x7fffdaf0a750) ___42:11782 ret shmctl=0, Success +++45:11782 call wait(NULL) ___33:11783 ret shmctl=0 ___34:11783 nattch=1 ___36:11783 call shmdt(0x7f452c223000) ___38:11783 ret shmdt=0 ___40:11783 call shmctl(2850820,IPC_RMID,NULL) ___42:11783 ret shmctl=-1, Invalid argument +++47:11782 ret wait=11783 And sometimes output is: ___19:11798 call shmget(325559,488,950) ___21:11798 ret shmget=3112964 ___23:11798 call shmat(3112964,0,0) ___25:11798 ret shmat=0x7f6b39378000 +++27:11798 call fork +++29:11798 ret fork=11799 ___31:11798 call shmctl(3112964,IPC_STAT,0x7fff3a6472a0) ___33:11798 ret shmctl=0 +++29:11799 ret fork=0 ___31:11799 call shmctl(3112964,IPC_STAT,0x7fff3a6472a0) ___34:11798 nattch=2 ___36:11798 call shmdt(0x7f6b39378000) ___33:11799 ret shmctl=0 ___38:11798 ret shmdt=0 ___34:11799 nattch=2 ___40:11798 call shmctl(3112964,IPC_RMID,NULL) ___36:11799 call shmdt(0x7f6b39378000) ___42:11798 ret shmctl=0, Success +++45:11798 call wait(NULL) ___38:11799 ret shmdt=0 ___40:11799 call shmctl(3112964,IPC_RMID,NULL) ___42:11799 ret shmctl=-1, Invalid argument +++47:11798 ret wait=11799 >From the outputs the parent process always get nattch=2, the child process get nattch=1 or sometimes get nattch=2. Then please see the following lines of src/pcm/pcm_direct.c (from alsa-lib-1.1.0.tar.bz2): 110 dmix->shmptr = shmat(dmix->shmid, 0, 0); 111 if (dmix->shmptr == (void *) -1) { 112 err = -errno; 113 snd_pcm_direct_shm_discard(dmix); 114 return err; 115 } 116 mlock(dmix->shmptr, sizeof(snd_pcm_direct_share_t)); 117 if (shmctl(dmix->shmid, IPC_STAT, &buf) < 0) { 118 err = -errno; 119 snd_pcm_direct_shm_discard(dmix); 120 return err; 121 } 122 if (buf.shm_nattch == 1) { /* we're the first user, clear the segment */ 123 memset(dmix->shmptr, 0, sizeof(snd_pcm_direct_share_t)); 124 if (dmix->ipc_gid>= 0) { 125 buf.shm_perm.gid = dmix->ipc_gid; 126 shmctl(dmix->shmid, IPC_SET, &buf); 127 } 128 dmix->shmptr->magic = SND_PCM_DIRECT_MAGIC; 129 return 1; 130 } else { 131 if (dmix->shmptr->magic != SND_PCM_DIRECT_MAGIC) { 132 snd_pcm_direct_shm_discard(dmix); 133 return -EINVAL; 134 } 135 } If an applicatioin that uses alsa-lib invokes fork() and the fork() is invoked just after shmat(line 110) and just before shmctl(line 117), buf.shm_nattch will be 2, so "buf.shm_nattch == 1" will be false. But while buf.shm_nattch is 2, we're still the first user this time, and we should executes line 123 to line 129 of the codes above. There is the following statement in `man shmget`: When a new shared memory segment is created, its contents are initialized to zero values According to the statement above, dmix->shmptr->magic is 0 on shm first created. So, to fix this issue, we should change the following line: 122 if (buf.shm_nattch == 1) { /* we're the first user, clear the segment */ to: 122 if (dmix->shmptr->magic != SND_PCM_DIRECT_MAGIC) { /* we're the first user, clear the segment */ Sincerely, Sisrnb

2 1

[alsa-devel] [PATCH v3 0/4] Add get_eld audio component for i915/HD-audio
by Takashi Iwai 10 Dec '15

10 Dec '15

Hi, this is the third revision of the patchset to add get_eld op to audio component for communicating more directly between i915 and HD-audio. Currently, the HDMI/DP audio status and ELD are notified and obtained via the hardware-level communication over HD-audio unsolicited event and verbs although the graphics driver holds the exactly same information. As we already have a notification via audio component, this is another step forward; the audio driver fetches directly the audio status and ELD via the new component op. A few patches have been dropped from this patchset as they were already applied individually beforehand. Also, two cleanup patches for i915 were squashed into one patch. The current patchset is found in sound git tree test/hdmi-jack branch. thanks, Takashi === v2->v3: * Track drm_connector for easier ELD retrieval, remove superfluous audio_enabled flag instead * Back to av_mutex * Adapt the earlier applied preliminary patches * Direct invocation of get_eld now instead of deferred call, as we have no longer deadlocks * Rebase to drm-next, update get_eld documentation for new kernel doc v1->v2: * Use modeset lock for get_eld lock, drop av mutex * Return the expected size from get_eld, not the copied size * Add reverse map from a port number to the encoder * Deferred invocation of get_eld from the eld_notify in HDA side * A bit more code refactoring in HDA side * Move audio component accessors to hdac_i915.c Takashi Iwai (4): drm/i915: Add get_eld audio component drm/i915: Add reverse mapping between port and intel_encoder ALSA: hda - Use component ops for i915 HDMI/DP audio jack handling ALSA: hda - Move audio component accesses to hdac_i915.c drivers/gpu/drm/i915/i915_drv.h | 2 + drivers/gpu/drm/i915/intel_audio.c | 59 +++++++++++++++------ drivers/gpu/drm/i915/intel_ddi.c | 1 + drivers/gpu/drm/i915/intel_dp.c | 1 + drivers/gpu/drm/i915/intel_drv.h | 2 + drivers/gpu/drm/i915/intel_hdmi.c | 2 + include/drm/i915_component.h | 14 +++++ include/sound/hda_i915.h | 14 +++++ sound/hda/hdac_i915.c | 66 +++++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 104 ++++++++++++++++++++++++++++++------- 10 files changed, 229 insertions(+), 36 deletions(-) -- 2.6.3

4 14

[alsa-devel] using all channels of RME Babyface Pro in class compliant mode
by Be 10 Dec '15

10 Dec '15

Hello, I just got an RME Babyface Pro USB sound card. It can operate either as a USB audio class compliant device (with mixing controlled by the device's physical controls) or make use of a proprietary driver on Windows and Mac OS X. The proprietary driver allows for control of the mixing capabilities of the device and using onboard DSP EQ. It would be great if someone wrote a Linux driver to make use of these. I'd be happy to send debugging info or whatever about the device and test kernel patches. I could set up a Windows VM to intercept USB communication. In the meantime, I am trying to use its class compliant mode to use with Mixxx for DJing. However, I cannot figure out how to use channels 3-4 for my headphones. The device has 4 analog channels out and in plus 8 digital out and in (via ADAT). Its manual says, regarding class compliant mode, "in Mac OS X, either the first two or all channels will work, depending on the choice in the Audio MIDI Setup." I believe the two channel operation exists so when the Babyface Pro is plugged into an iPad, the iPad sends stereo output from iTunes and other programs to the Babyface Pro's main output. However, I do not know how to select between this 2 channel mode and the 12 channel mode with Linux. PulseAudio only shows it as a stereo device. Mixxx, which uses the PortAudio library, displays all 12 channels as available but doesn't work when any channels other than 1-2 are selected. Here is debugging output of Mixxx: Debug [Main]: SoundManager::setupDevices() Debug [Main]: SoundDevicePortAudio::open() "2, Babyface Pro (70785713): USB Audio (hw:1,0)" Debug [Main]: framesPerBuffer: 256 Debug [Main]: Requested sample rate: 44100 Hz, latency: 5.80499 ms Debug [Main]: Output channels: 4 | Input channels: 0 Debug [Main]: Opening stream with id 2 Expression 'alsa_snd_pcm_hw_params_set_channels( pcm, hwParams, self->numHostChannels )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2051 Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2717 Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2838 Warning [Main]: Error opening stream: Invalid number of channels aplay -l output: **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC272 Analog [ALC272 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Pro70785713 [Babyface Pro (70785713)], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 aplay -L output: null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server default Default ALSA Output (currently PulseAudio Sound Server) sysdefault:CARD=PCH HDA Intel PCH, ALC272 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog Front speakers surround21:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=PCH,DEV=0 HDA Intel PCH, HDMI 0 HDMI Audio Output sysdefault:CARD=Pro70785713 Babyface Pro (70785713), USB Audio Default Audio Device front:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio Front speakers surround21:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio IEC958 (S/PDIF) Digital Audio Output /proc/asound/card1/stream0 when playing to channels 1-2 with PulseAudio: RME Babyface Pro (70785713) at usb-0000:00:1d.0-1.2, high speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 47 Momentary freq = 44118 Hz (0x5.83c9) Feedback Format = 16.16 Interface 1 Altset 1 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Interface 1 Altset 2 Format: S24_3LE Channels: 12 Endpoint: 3 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Capture: Status: Stop Interface 2 Altset 1 Format: S24_3LE Channels: 12 Endpoint: 4 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Interface 2 Altset 2 Format: S24_3LE Channels: 2 Endpoint: 4 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us So I guess the question is how to get PortAudio to use the "Altset 2" option. Could this be accomplished with some asoundrc sorcery? Does something need to change in snd-usb-audio? Is the issue with PortAudio and PulseAudio? How can I test channels 3-4 with aplay or another program?

2 2

[alsa-devel] [PATCH] ALSA: hda - Implement loopback control switch for Realtek and other codecs
by Takashi Iwai 10 Dec '15

10 Dec '15

Many codecs, typically found on Realtek codecs, have the analog loopback path merged to the secondary input of the middle of the output paths. Currently, we don't offer the dynamic switching in such configuration but let each loopback path mute by itself. This should work well in theory, but in reality, we often see that such a dead loopback path causes some background noises even if all the elements get muted. Such a problem has been fixed by adding the quirk accordingly to disable aamix, and it's the right fix, per se. The only problem is that it's not so trivial to achieve it; user needs to pass a hint string via patch module option or sysfs. This patch gives a bit improvement on the situation: it adds "Loopback Mixing" control element for such codecs like other codecs (e.g. IDT or VIA codecs) with the individual loopback paths. User can turn on/off the loopback path simply via a mixer app. For keeping the compatibility, the loopback is still enabled on these codecs. But user can try to turn it off if experiencing a suspicious background or click noise on the fly, then build a static fixup later once after the problem is addressed. Other than the addition of the loopback enable/disablement control, there should be no changes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 87 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 68 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f3c058f6c831..30c8efe0f80a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -754,9 +754,6 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps; unsigned int mask, val; - if (!enable && is_active_nid(codec, nid, dir, idx_to_check)) - return; - caps = query_amp_caps(codec, nid, dir); val = get_amp_val_to_activate(codec, nid, dir, caps, enable); mask = get_amp_mask_to_modify(codec, nid, dir, idx_to_check, caps); @@ -767,12 +764,22 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir, update_amp(codec, nid, dir, idx, mask, val); } +static void check_and_activate_amp(struct hda_codec *codec, hda_nid_t nid, + int dir, int idx, int idx_to_check, + bool enable) +{ + /* check whether the given amp is still used by others */ + if (!enable && is_active_nid(codec, nid, dir, idx_to_check)) + return; + activate_amp(codec, nid, dir, idx, idx_to_check, enable); +} + static void activate_amp_out(struct hda_codec *codec, struct nid_path *path, int i, bool enable) { hda_nid_t nid = path->path[i]; init_amp(codec, nid, HDA_OUTPUT, 0); - activate_amp(codec, nid, HDA_OUTPUT, 0, 0, enable); + check_and_activate_amp(codec, nid, HDA_OUTPUT, 0, 0, enable); } static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, @@ -800,9 +807,16 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, * when aa-mixer is available, we need to enable the path as well */ for (n = 0; n < nums; n++) { - if (n != idx && (!add_aamix || conn[n] != spec->mixer_merge_nid)) - continue; - activate_amp(codec, nid, HDA_INPUT, n, idx, enable); + if (n != idx) { + if (conn[n] != spec->mixer_merge_nid) + continue; + /* when aamix is disabled, force to off */ + if (!add_aamix) { + activate_amp(codec, nid, HDA_INPUT, n, n, false); + continue; + } + } + check_and_activate_amp(codec, nid, HDA_INPUT, n, idx, enable); } } @@ -1563,6 +1577,12 @@ static bool map_singles(struct hda_codec *codec, int outs, return found; } +static inline bool has_aamix_out_paths(struct hda_gen_spec *spec) +{ + return spec->aamix_out_paths[0] || spec->aamix_out_paths[1] || + spec->aamix_out_paths[2]; +} + /* create a new path including aamix if available, and return its index */ static int check_aamix_out_path(struct hda_codec *codec, int path_idx) { @@ -2405,25 +2425,51 @@ static void update_aamix_paths(struct hda_codec *codec, bool do_mix, } } +/* re-initialize the output paths; only called from loopback_mixing_put() */ +static void update_output_paths(struct hda_codec *codec, int num_outs, + const int *paths) +{ + struct hda_gen_spec *spec = codec->spec; + struct nid_path *path; + int i; + + for (i = 0; i < num_outs; i++) { + path = snd_hda_get_path_from_idx(codec, paths[i]); + if (path) + snd_hda_activate_path(codec, path, path->active, + spec->aamix_mode); + } +} + static int loopback_mixing_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_gen_spec *spec = codec->spec; + const struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int val = ucontrol->value.enumerated.item[0]; if (val == spec->aamix_mode) return 0; spec->aamix_mode = val; - update_aamix_paths(codec, val, spec->out_paths[0], - spec->aamix_out_paths[0], - spec->autocfg.line_out_type); - update_aamix_paths(codec, val, spec->hp_paths[0], - spec->aamix_out_paths[1], - AUTO_PIN_HP_OUT); - update_aamix_paths(codec, val, spec->speaker_paths[0], - spec->aamix_out_paths[2], - AUTO_PIN_SPEAKER_OUT); + if (has_aamix_out_paths(spec)) { + update_aamix_paths(codec, val, spec->out_paths[0], + spec->aamix_out_paths[0], + cfg->line_out_type); + update_aamix_paths(codec, val, spec->hp_paths[0], + spec->aamix_out_paths[1], + AUTO_PIN_HP_OUT); + update_aamix_paths(codec, val, spec->speaker_paths[0], + spec->aamix_out_paths[2], + AUTO_PIN_SPEAKER_OUT); + } else { + update_output_paths(codec, cfg->line_outs, spec->out_paths); + if (cfg->line_out_type != AUTO_PIN_HP_OUT) + update_output_paths(codec, cfg->hp_outs, spec->hp_paths); + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) + update_output_paths(codec, cfg->speaker_outs, + spec->speaker_paths); + } return 1; } @@ -2441,12 +2487,13 @@ static int create_loopback_mixing_ctl(struct hda_codec *codec) if (!spec->mixer_nid) return 0; - if (!(spec->aamix_out_paths[0] || spec->aamix_out_paths[1] || - spec->aamix_out_paths[2])) - return 0; if (!snd_hda_gen_add_kctl(spec, NULL, &loopback_mixing_enum)) return -ENOMEM; spec->have_aamix_ctl = 1; + /* if no explicit aamix path is present (e.g. for Realtek codecs), + * enable aamix as default -- just for compatibility + */ + spec->aamix_mode = !has_aamix_out_paths(spec); return 0; } @@ -5647,6 +5694,8 @@ static void init_aamix_paths(struct hda_codec *codec) if (!spec->have_aamix_ctl) return; + if (!has_aamix_out_paths(spec)) + return; update_aamix_paths(codec, spec->aamix_mode, spec->out_paths[0], spec->aamix_out_paths[0], spec->autocfg.line_out_type); -- 2.6.3

2 2

[alsa-devel] [PATCH] ASoC: es8328: Fix deemphasis values
by John Keeping 09 Dec '15

09 Dec '15

This is using completely the wrong mask and value when updating the register. Since the correct values are already defined in the header, switch to using a table with explicit constants rather than shifting the array index. Signed-off-by: John Keeping <john(a)metanate.com> --- sound/soc/codecs/es8328.c | 25 +++++++++++++++++-------- sound/soc/codecs/es8328.h | 1 + 2 files changed, 18 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 84f5eb0..afa6c5d 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -85,7 +85,15 @@ static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0); static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 300, 0); -static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; +static const struct { + int rate; + unsigned int val; +} deemph_settings[] = { + { 0, ES8328_DACCONTROL6_DEEMPH_OFF }, + { 32000, ES8328_DACCONTROL6_DEEMPH_32k }, + { 44100, ES8328_DACCONTROL6_DEEMPH_44_1k }, + { 48000, ES8328_DACCONTROL6_DEEMPH_48k }, +}; static int es8328_set_deemph(struct snd_soc_codec *codec) { @@ -97,21 +105,22 @@ static int es8328_set_deemph(struct snd_soc_codec *codec) * rate. */ if (es8328->deemph) { - best = 1; - for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { - if (abs(deemph_settings[i] - es8328->playback_fs) < - abs(deemph_settings[best] - es8328->playback_fs)) + best = 0; + for (i = 1; i < ARRAY_SIZE(deemph_settings); i++) { + if (abs(deemph_settings[i].rate - es8328->playback_fs) < + abs(deemph_settings[best].rate - es8328->playback_fs)) best = i; } - val = best << 1; + val = deemph_settings[best].val; } else { - val = 0; + val = ES8328_DACCONTROL6_DEEMPH_OFF; } dev_dbg(codec->dev, "Set deemphasis %d\n", val); - return snd_soc_update_bits(codec, ES8328_DACCONTROL6, 0x6, val); + return snd_soc_update_bits(codec, ES8328_DACCONTROL6, + ES8328_DACCONTROL6_DEEMPH_MASK, val); } static int es8328_get_deemph(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h index cb36afe..156c748 100644 --- a/sound/soc/codecs/es8328.h +++ b/sound/soc/codecs/es8328.h @@ -153,6 +153,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_DACCONTROL6_CLICKFREE (1 << 3) #define ES8328_DACCONTROL6_DAC_INVR (1 << 4) #define ES8328_DACCONTROL6_DAC_INVL (1 << 5) +#define ES8328_DACCONTROL6_DEEMPH_MASK (3 << 6) #define ES8328_DACCONTROL6_DEEMPH_OFF (0 << 6) #define ES8328_DACCONTROL6_DEEMPH_32k (1 << 6) #define ES8328_DACCONTROL6_DEEMPH_44_1k (2 << 6) -- 2.6.3.462.gbe2c914

2 1

[alsa-devel] [PATCH V3 0/2] Add support for Pistachio internal DAC
by Damien Horsley 09 Dec '15

09 Dec '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add binding document and driver for Pistachio internal DAC Changes in V3: Mute control definition changed to SOC_SINGLE Codec driver probe function added to associate regmap with codec Damien.Horsley (2): ASoC: img: Add binding document for Pistachio internal DAC ASoC: img: Add driver for Pistachio internal DAC .../bindings/sound/img,pistachio-internal-dac.txt | 18 ++ sound/soc/img/Kconfig | 8 + sound/soc/img/Makefile | 2 + sound/soc/img/pistachio-internal-dac.c | 287 +++++++++++++++++++++ 4 files changed, 315 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt create mode 100644 sound/soc/img/pistachio-internal-dac.c -- 2.1.4

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[alsa-devel] [asoc:topic/intel 75/78] sound/soc/intel/boards/skl_rt286.c:55:2: error: implicit declaration of function 'SND_SOC_DAPM_SINK'
by kbuild test robot 09 Dec '15

09 Dec '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 9ec2053b13f75d7ad9c0e6db9763954bd1a1b9ae commit: b34e24d2406f123d5dbbff4fdeebbc2af76b0acd [75/78] ASoC: Intel: Skylake: add wov as int sink config: i386-allmodconfig (attached as .config) reproduce: git checkout b34e24d2406f123d5dbbff4fdeebbc2af76b0acd # save the attached .config to linux build tree make ARCH=i386 All error/warnings (new ones prefixed by >>): >> sound/soc/intel/boards/skl_rt286.c:55:2: error: implicit declaration of function 'SND_SOC_DAPM_SINK' [-Werror=implicit-function-declaration] SND_SOC_DAPM_SINK("WoV Sink"), ^ >> sound/soc/intel/boards/skl_rt286.c:55:2: error: initializer element is not constant sound/soc/intel/boards/skl_rt286.c:55:2: note: (near initialization for 'skylake_widgets[5].id') >> sound/soc/intel/boards/skl_rt286.c:49:61: warning: missing braces around initializer [-Wmissing-braces] static const struct snd_soc_dapm_widget skylake_widgets[] = { ^ sound/soc/intel/boards/skl_rt286.c:49:61: note: (near initialization for 'skylake_widgets') cc1: some warnings being treated as errors vim +/SND_SOC_DAPM_SINK +55 sound/soc/intel/boards/skl_rt286.c 43 static const struct snd_kcontrol_new skylake_controls[] = { 44 SOC_DAPM_PIN_SWITCH("Speaker"), 45 SOC_DAPM_PIN_SWITCH("Headphone Jack"), 46 SOC_DAPM_PIN_SWITCH("Mic Jack"), 47 }; 48 > 49 static const struct snd_soc_dapm_widget skylake_widgets[] = { 50 SND_SOC_DAPM_HP("Headphone Jack", NULL), 51 SND_SOC_DAPM_SPK("Speaker", NULL), 52 SND_SOC_DAPM_MIC("Mic Jack", NULL), 53 SND_SOC_DAPM_MIC("DMIC2", NULL), 54 SND_SOC_DAPM_MIC("SoC DMIC", NULL), > 55 SND_SOC_DAPM_SINK("WoV Sink"), 56 }; 57 58 static const struct snd_soc_dapm_route skylake_rt286_map[] = { --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] Intel HDA / ca0132: quirk for Alienware 17 2015
by Gabriele Martino 09 Dec '15

09 Dec '15

The Alienware 17 (2015) has the same card and pin configuration of the Alienware 15, so the same quirks must be applied. Signed-off-by: Gabriele Martino <g.martino(a)gmx.com>

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[alsa-devel] [PATCH] ALSA: hda - Fix noise problems on Thinkpad T440s
by Takashi Iwai 09 Dec '15

09 Dec '15

Lenovo Thinkpad T440s suffers from constant background noises, and it seems to be a generic hardware issue on this model: https://forums.lenovo.com/t5/ThinkPad-T400-T500-and-newer-T/T440s-speaker-n… As the noise comes from the analog loopback path, disabling the path is the easy workaround. Also, the machine gives significant cracking noises at PM suspend. A workaround found by trial-and-error is to disable the shutup callback currently used for ALC269-variant. This patch addresses these noise issues by introducing a new fixup chain. Although the same workaround might be applicable to other Thinkpad models, it's applied only to T440s (17aa:220c) in this patch, so far, just to be safe (you chicken!). As a compromise, a new model option string "tp440" is provided now, though, so that owners of other Thinkpad models can test it more easily. Bugzilla: https://bugzilla.opensuse.org/show_bug.cgi?id=958504 Reported-and-tested-by: Tim Hardeck <thardeck(a)suse.de> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 49b1d366f82a..8dd2ac13b3af 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4204,6 +4204,18 @@ static void alc_fixup_tpt440_dock(struct hda_codec *codec, } } +/* additional fixup for Thinkpad T440s noise problem */ +static void alc_fixup_tpt440(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->shutup = alc_no_shutup; /* reduce click noise */ + spec->gen.mixer_nid = 0; /* reduce background noise */ + } +} + static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -4578,6 +4590,7 @@ enum { ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, + ALC292_FIXUP_TPT440, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, @@ -5051,6 +5064,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, + [ALC292_FIXUP_TPT440] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_tpt440, + .chained = true, + .chain_id = ALC292_FIXUP_TPT440_DOCK, + }, [ALC283_FIXUP_BXBT2807_MIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5332,7 +5351,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), - SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440), SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2211, "Thinkpad W541", ALC292_FIXUP_TPT440_DOCK), @@ -5432,6 +5451,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"}, {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, + {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, {} }; -- 2.6.3

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[alsa-devel] [PATCH] sound/oss: remove VIRT_TO_BUS dependency
by Arnd Bergmann 09 Dec '15

09 Dec '15

The OSS sound drivers used to rely on virt_to_bus(), but don't any more, so we can remove the Kconfig dependency. As a lot of architectures don't provide VIRT_TO_BUS any more, removing the dependency in sounds/oss/ would make the deprecated drivers appear there, which we probably don't want. Instead I'm replacing the simple dependency with 'VIRT_TO_BUS || RPC || NETWINDER' so we can still build these sound drivers for the platforms that need them, but don't change anything on other architectures. As a follow-up, we can remove the virt_to_bus() implementation and Kconfig symbol in the ARM architecture. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index 48568fdf847f..4033fe58f0cf 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -240,7 +240,7 @@ config MSND_FIFOSIZE menuconfig SOUND_OSS tristate "OSS sound modules" - depends on ISA_DMA_API && VIRT_TO_BUS + depends on ISA_DMA_API && (VIRT_TO_BUS || ARCH_RPC || ARCH_NETWINDER) depends on !GENERIC_ISA_DMA_SUPPORT_BROKEN help OSS is the Open Sound System suite of sound card drivers. They make

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[alsa-devel] [PATCH 0/3] ASoC: sti: correction for HBRA support
by Moise Gergaud 09 Dec '15

09 Dec '15

This patchset is V2 of "[PATCH 2/2] ASoC: sti: correction for HBRA support" Moise Gergaud (3): ALSA: pcm: add IEC958 channel status update ASoC: sti: use pcm iec958 channel status helper ASoC: sti: correction for HBRA (High Bit Rate Audio) support include/sound/pcm_iec958.h | 19 ++++++++- sound/core/pcm_iec958.c | 78 +++++++++++++++++++++++++------------ sound/soc/sti/Kconfig | 1 + sound/soc/sti/uniperif_player.c | 86 +++++++++++++++++------------------------ 4 files changed, 107 insertions(+), 77 deletions(-) -- 1.9.1

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[alsa-devel] [RFC] ARM: remove VIRT_TO_BUS
by Arnd Bergmann 09 Dec '15

09 Dec '15

All drivers that are relevant for rpc or footbridge have stopped using virt_to_bus a while ago, so we can remove it and avoid some harmless randconfig warnings for drivers that we do not care about: drivers/atm/zatm.c: In function 'poll_rx': drivers/atm/zatm.c:401:18: warning: 'bus_to_virt' is deprecated [-Wdeprecated-declarations] skb = ((struct rx_buffer_head *) bus_to_virt(here[2]))->skb; FWIW, the remaining drivers using this are: ATM: firestream, zatm, ambassador, horizon ISDN: hisax/netjet V4L: STA2X11, zoran Net: Appletalk LTPC, Tulip DE4x5, Toshiba IrDA WAN: comtrol sv11, cosa, lanmedia, sealevel SCSI: DPT_I2O, buslogic VME: CA91C142 My best guess is that all of the above are so hopelessly obsolete that we are best off removing all of them form the kernel, but that can be done another time. The sound drivers used to rely on virt_to_bus, but don't any more. As a lot of architectures don't provide VIRT_TO_BUS any more, removing the dependency in sounds/oss/ would make the deprecated drivers appear there, which we probably don't want. Instead I'm replacing the simple dependency with 'VIRT_TO_BUS || RPC || NETWINDER' so we can still build these sound drivers for the platforms that need them, but don't change anything on other architectures. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- No hurry for this one. If we can agree that it's a good idea, we can probably do the sound/oss change first, and then do the rest for the following merge window. If nobody cares about picking up the sound/oss patches, we can also just do it as one patch. diff --git a/arch/arm/Kconfig b/arch/arm/Kconfig index a077f59350a2..b03892cecd65 100644 --- a/arch/arm/Kconfig +++ b/arch/arm/Kconfig @@ -556,7 +556,6 @@ config ARCH_RPC select NEED_MACH_IO_H select NEED_MACH_MEMORY_H select NO_IOPORT_MAP - select VIRT_TO_BUS help On the Acorn Risc-PC, Linux can support the internal IDE disk and CD-ROM interface, serial and parallel port, and the floppy drive. diff --git a/arch/arm/include/asm/memory.h b/arch/arm/include/asm/memory.h index c79b57bf71c4..8f8c3a0571a6 100644 --- a/arch/arm/include/asm/memory.h +++ b/arch/arm/include/asm/memory.h @@ -303,20 +303,6 @@ static inline phys_addr_t __virt_to_idmap(unsigned long x) #define __bus_to_pfn(x) __phys_to_pfn(x) #endif -#ifdef CONFIG_VIRT_TO_BUS -#define virt_to_bus virt_to_bus -static inline __deprecated unsigned long virt_to_bus(void *x) -{ - return __virt_to_bus((unsigned long)x); -} - -#define bus_to_virt bus_to_virt -static inline __deprecated void *bus_to_virt(unsigned long x) -{ - return (void *)__bus_to_virt(x); -} -#endif - /* * Conversion between a struct page and a physical address. * diff --git a/arch/arm/mach-footbridge/Kconfig b/arch/arm/mach-footbridge/Kconfig index 735909f90349..7b87bac114a2 100644 --- a/arch/arm/mach-footbridge/Kconfig +++ b/arch/arm/mach-footbridge/Kconfig @@ -68,7 +68,6 @@ config ARCH_NETWINDER select ISA select ISA_DMA select PCI - select VIRT_TO_BUS help Say Y here if you intend to run this kernel on the Rebel.COM NetWinder. Information about this machine can be found at: diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index 48568fdf847f..4033fe58f0cf 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -240,7 +240,7 @@ config MSND_FIFOSIZE menuconfig SOUND_OSS tristate "OSS sound modules" - depends on ISA_DMA_API && VIRT_TO_BUS + depends on ISA_DMA_API && (VIRT_TO_BUS || ARCH_RPC || ARCH_NETWINDER) depends on !GENERIC_ISA_DMA_SUPPORT_BROKEN help OSS is the Open Sound System suite of sound card drivers. They make

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[alsa-devel] [PATCH 1/2] ALSA: hda - Less grumbling about lack of i915 binding
by Takashi Iwai 09 Dec '15

09 Dec '15

The recent commit [6603249dcdbb: ALSA: hda - Enable audio component for old Intel PCH devices] enabled the i915 binding for HDMI/DP on old Intel PCHs. But many boards are without HDMI/DP, and they actually don't need i915 binding, and yet the driver has a check of i915 binding and complains like Haswell must be built with CONFIG_SND_HDA_I915 This error is false-positive, and it should be put only for HSW/BDW, instead of all devices that may be bound with i915. This patch fixes the condition to check, as well as rephrasing the message specific to HSW/BDW HDMI/DP. Fixes: 6603249dcdbb ('ALSA: hda - Enable audio component for old Intel PCH devices') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ee0e316401f9..b49547f3c2e9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1954,8 +1954,8 @@ static int azx_probe(struct pci_dev *pci, #endif /* CONFIG_SND_HDA_PATCH_LOADER */ #ifndef CONFIG_SND_HDA_I915 - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - dev_err(card->dev, "Haswell must build in CONFIG_SND_HDA_I915\n"); + if (CONTROLLER_IN_GPU(pci)) + dev_err(card->dev, "Haswell/Broadwell HDMI/DP must build in CONFIG_SND_HDA_I915\n"); #endif if (schedule_probe) -- 2.6.3

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[alsa-devel] next-20151208 - CONFIG_SND_HDA_I915
by Valdis Kletnieks 09 Dec '15

09 Dec '15

I'm experiencing a spot of trouble with this commit: commit aad730d0704545ad97654bd929b0aba05adb1436 Author: Takashi Iwai <tiwai(a)suse.de> Date: Mon Dec 2 13:33:57 2013 +0100 ALSA: hda - Always do delayed probes for HD-audio devices That adds some code: #ifndef CONFIG_SND_HDA_I915 if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) dev_err(card->dev, "Haswell must build in CONFIG_SND_HDA_I915\n"); "Oh, bother" -- Winnie the Pooh So I go looking how to fix this, and 'make menuconfig' tells me: Symbol: SND_HDA_I915 [=n] x x Type : boolean x x Defined at sound/hda/Kconfig:8 x x Depends on: SOUND [=y] && !M68K && !UML && SND [=y] && DRM_I915 [=n] && SND_HDA_CORE [=y] And why is DRM_I915=n? # lspci | grep NV 01:00.0 VGA compatible controller: NVIDIA Corporation GF108GLM [NVS 5200M] (rev a1) Because this Dell laptop has NVidia graphics onboard, and no sign of I915 graphics that needs I915 DRM. I think the infrastructure that you're trying to re-use from DRM_I915 needs to be moved to another symbol which can be SELECT/DEPEND by both DRM_I915 and SND_HSA_I915. Comments/suggestions?

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[alsa-devel] [PATCH] ASoC: rsnd: tidyup data align position for capture
by Kuninori Morimoto 08 Dec '15

08 Dec '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> L/R channel data has been treated as inverted on R-Car sound 16bit mode, Thus, 4689032b1("ASoC: rsnd: tidyup data align position") tidyuped data align position. But it couldn't care about capture case. This patch cares both playback/capture Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/cmd.c | 1 + sound/soc/sh/rcar/core.c | 13 +++++++++++-- sound/soc/sh/rcar/gen.c | 1 + sound/soc/sh/rcar/rsnd.h | 2 ++ 4 files changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index ab904c3..cd1f064 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -80,6 +80,7 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, dev_dbg(dev, "ctu/mix path = 0x%08x", data); rsnd_mod_write(mod, CMD_ROUTE_SLCT, data); + rsnd_mod_write(mod, CMD_BUSIF_DALIGN, rsnd_get_dalign(mod, io)); rsnd_adg_set_cmd_timsel_gen2(mod, io); diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index bf596ff..0945cff 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -273,13 +273,22 @@ u32 rsnd_get_adinr_chan(struct rsnd_mod *mod, struct rsnd_dai_stream *io) */ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { - struct rsnd_mod *src = rsnd_io_to_mod_src(io); struct rsnd_mod *ssi = rsnd_io_to_mod_ssi(io); - struct rsnd_mod *target = src ? src : ssi; + struct rsnd_mod *target; struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); u32 val = 0x76543210; u32 mask = ~0; + if (rsnd_io_is_play(io)) { + struct rsnd_mod *src = rsnd_io_to_mod_src(io); + + target = src ? src : ssi; + } else { + struct rsnd_mod *cmd = rsnd_io_to_mod_cmd(io); + + target = cmd ? cmd : ssi; + } + mask <<= runtime->channels * 4; val = val & mask; diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 50fc730..7c5485e 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -242,6 +242,7 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_M_REG(SRC_ROUTE_MODE0, 0xc, 0x20), RSND_GEN_M_REG(SRC_CTRL, 0x10, 0x20), RSND_GEN_M_REG(SRC_INT_ENABLE0, 0x18, 0x20), + RSND_GEN_M_REG(CMD_BUSIF_DALIGN,0x188, 0x20), RSND_GEN_M_REG(CMD_ROUTE_SLCT, 0x18c, 0x20), RSND_GEN_M_REG(CMD_CTRL, 0x190, 0x20), RSND_GEN_S_REG(SCU_SYS_STATUS0, 0x1c8), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 970e130..ad854d6 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -81,6 +81,7 @@ enum rsnd_reg { RSND_REG_SCU_SYS_INT_EN0, RSND_REG_SCU_SYS_INT_EN1, /* Gen2 only */ RSND_REG_CMD_CTRL, /* Gen2 only */ + RSND_REG_CMD_BUSIF_DALIGN, /* Gen2 only */ RSND_REG_CMD_ROUTE_SLCT, RSND_REG_CMDOUT_TIMSEL, /* Gen2 only */ RSND_REG_CTU_CTUIR, @@ -319,6 +320,7 @@ struct rsnd_dai_stream { #define rsnd_io_to_mod_ctu(io) rsnd_io_to_mod((io), RSND_MOD_CTU) #define rsnd_io_to_mod_mix(io) rsnd_io_to_mod((io), RSND_MOD_MIX) #define rsnd_io_to_mod_dvc(io) rsnd_io_to_mod((io), RSND_MOD_DVC) +#define rsnd_io_to_mod_cmd(io) rsnd_io_to_mod((io), RSND_MOD_CMD) #define rsnd_io_to_rdai(io) ((io)->rdai) #define rsnd_io_to_priv(io) (rsnd_rdai_to_priv(rsnd_io_to_rdai(io))) #define rsnd_io_is_play(io) (&rsnd_io_to_rdai(io)->playback == io) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: mediatek: Use current HW pointer for pointer callback
by Koro Chen 08 Dec '15

08 Dec '15

Previously we recorded "last interrupt position" and used it in pointer callback. This is not correct implementation, and it causes underruns when user space monitors buffer level to decide when to send next data chunk in low latency application. Remove position recording in IRQ handler and also hw_ptr in struct mtk_afe_memif used to record that, and let pointer callback reports current HW pointer instead. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-common.h | 1 - sound/soc/mediatek/mtk-afe-pcm.c | 22 +++++++++++----------- 2 files changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/mediatek/mtk-afe-common.h b/sound/soc/mediatek/mtk-afe-common.h index cc4393c..9b1af1a 100644 --- a/sound/soc/mediatek/mtk-afe-common.h +++ b/sound/soc/mediatek/mtk-afe-common.h @@ -92,7 +92,6 @@ struct mtk_afe_memif_data { struct mtk_afe_memif { unsigned int phys_buf_addr; int buffer_size; - unsigned int hw_ptr; /* Previous IRQ's HW ptr */ struct snd_pcm_substream *substream; const struct mtk_afe_memif_data *data; const struct mtk_afe_irq_data *irqdata; diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index 7f71343..5399a0e 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -175,8 +175,17 @@ static snd_pcm_uframes_t mtk_afe_pcm_pointer struct snd_soc_pcm_runtime *rtd = substream->private_data; struct mtk_afe *afe = snd_soc_platform_get_drvdata(rtd->platform); struct mtk_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; + unsigned int hw_ptr; + int ret; + + ret = regmap_read(afe->regmap, memif->data->reg_ofs_cur, &hw_ptr); + if (ret || hw_ptr == 0) { + dev_err(afe->dev, "%s hw_ptr err\n", __func__); + hw_ptr = memif->phys_buf_addr; + } - return bytes_to_frames(substream->runtime, memif->hw_ptr); + return bytes_to_frames(substream->runtime, + hw_ptr - memif->phys_buf_addr); } static const struct snd_pcm_ops mtk_afe_pcm_ops = { @@ -602,7 +611,6 @@ static int mtk_afe_dais_hw_params(struct snd_pcm_substream *substream, memif->phys_buf_addr = substream->runtime->dma_addr; memif->buffer_size = substream->runtime->dma_bytes; - memif->hw_ptr = 0; /* start */ regmap_write(afe->regmap, @@ -737,7 +745,6 @@ static int mtk_afe_dais_trigger(struct snd_pcm_substream *substream, int cmd, /* and clear pending IRQ */ regmap_write(afe->regmap, AFE_IRQ_CLR, 1 << memif->data->irq_clr_shift); - memif->hw_ptr = 0; return 0; default: return -EINVAL; @@ -1081,7 +1088,7 @@ static const struct regmap_config mtk_afe_regmap_config = { static irqreturn_t mtk_afe_irq_handler(int irq, void *dev_id) { struct mtk_afe *afe = dev_id; - unsigned int reg_value, hw_ptr; + unsigned int reg_value; int i, ret; ret = regmap_read(afe->regmap, AFE_IRQ_STATUS, &reg_value); @@ -1097,13 +1104,6 @@ static irqreturn_t mtk_afe_irq_handler(int irq, void *dev_id) if (!(reg_value & (1 << memif->data->irq_clr_shift))) continue; - ret = regmap_read(afe->regmap, memif->data->reg_ofs_cur, - &hw_ptr); - if (ret || hw_ptr == 0) { - dev_err(afe->dev, "%s hw_ptr err\n", __func__); - hw_ptr = memif->phys_buf_addr; - } - memif->hw_ptr = hw_ptr - memif->phys_buf_addr; snd_pcm_period_elapsed(memif->substream); } -- 1.7.9.5

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[alsa-devel] [PATCH v2 00/12] ASoC: Intel: Skylake: Add support for loadable modules
by Vinod Koul 08 Dec '15

08 Dec '15

This series adds first the infrastructure for loadable modules in SKL aDSP. The loadable modules are loaded using load/unload IPC to aDSP thru code load DMA and can be configured by user thru TLV byte kcontrols for these modules. This also adds the implementation of new DAPM sinks for phrase detection modules, and adds support for keeping controller active while is suspend to support ignore_suspend DAIs Changes on v2: - squash two load module IPCs and load modules usgae patches - add comments for module loading - add active suspend two patches Dharageswari R (2): ASoC: Intel: Skylake: Add support for Loadable modules ASoC: Intel: Skylake: Add support for Mic Select module Jeeja KP (4): ASoC: Intel: Skylake: Add memory pages to widget data. ASoC: Intel: Skylake: Fix module init data correctly ASoC: Intel: Skylake: Add dai link for DMIC capture ASoC: Intel: Skylake: Add support for active suspend Mousami Jana (1): ASoC: Intel: Skylake: add LARGE_CONFIG_GET IPC support Omair M Abdullah (2): ASoC: Intel: Skylake: update mailbox uplink window offset and size ASoC: Intel: Skylake: read params from DSP if module is on Praveen Diwakar (1): ASoC: Intel: Skylake: Update ignore suspend for rt286 machine Vinod Koul (2): ASoC: Intel: Skylake: add wov as int sink ASoc: Intel: Skylake: Fix the dapm machine map sound/soc/intel/boards/skl_rt286.c | 73 ++++++++++- sound/soc/intel/skylake/skl-messages.c | 40 ++++++ sound/soc/intel/skylake/skl-pcm.c | 27 +++++ sound/soc/intel/skylake/skl-sst-dsp.h | 18 ++- sound/soc/intel/skylake/skl-sst-ipc.c | 106 ++++++++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 9 ++ sound/soc/intel/skylake/skl-sst.c | 175 +++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.c | 43 ++++++- sound/soc/intel/skylake/skl-topology.h | 12 +- sound/soc/intel/skylake/skl-tplg-interface.h | 19 ++- sound/soc/intel/skylake/skl.c | 28 ++++- sound/soc/intel/skylake/skl.h | 2 + 12 files changed, 533 insertions(+), 19 deletions(-) -- 1.9.1

2 16

[alsa-devel] [PATCH 00/16] ASoC: Intel: Skylake: Add support for loadable modules
by Vinod Koul 08 Dec '15

08 Dec '15

This series adds first the infrastructure for loadable modules in SKL aDSP. The loadable modules are loaded using load/unload IPC to aDSP thru code load DMA and can be configured by user thru TLV byte kcontrols for these modules. This also adds the implementation of new DAPM sinks for phrase detection modules. Dharageswari R (2): ASoC: Intel: Skylake: Add support for Load/Unload IPCs ASoC: Intel: Skylake: Add support for Loadable modules Dharageswari.R (1): ASoC: Intel: Skylake: Add support for Mic Select module Jeeja KP (8): ASoC: Intel: Skylake: Update DMIC DAIs and capabilities ASoC: Intel: Skylake: Add helper routines to handle module params ASoC: Intel: Skylake: Add helper routine to handle Algo parameter ASoC: Intel: Skylake: Add support to configure module params ASoC: Intel: Skylake: Add tlv byte kcontrols ASoC: Intel: Skylake: Add memory pages to widget data. ASoC: Intel: Skylake: Fix module init data correctly ASoC: Intel: Skylake: Add dai link for DMIC capture Mousami Jana (1): ASoC: Intel: Skylake: add LARGE_CONFIG_GET IPC support Omair M Abdullah (2): ASoC: Intel: Skylake: update mailbox uplink window offset and size ASoC: Intel: Skylake: read params from DSP if module is on Vinod Koul (2): ASoC: Intel: Skylake: add wov as int sink ASoc: Intel: Skylake: Fix the dapm machine map sound/soc/intel/boards/skl_rt286.c | 58 ++++++- sound/soc/intel/skylake/skl-messages.c | 105 +++++++++++- sound/soc/intel/skylake/skl-pcm.c | 23 ++- sound/soc/intel/skylake/skl-sst-dsp.h | 18 +- sound/soc/intel/skylake/skl-sst-ipc.c | 100 +++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 9 + sound/soc/intel/skylake/skl-sst.c | 175 +++++++++++++++++++ sound/soc/intel/skylake/skl-topology.c | 242 ++++++++++++++++++++++++++- sound/soc/intel/skylake/skl-topology.h | 32 +++- sound/soc/intel/skylake/skl-tplg-interface.h | 23 ++- 10 files changed, 761 insertions(+), 24 deletions(-) -- 2.4.3

2 36

[alsa-devel] [PATCH v2 00/13] ASoC: topology: Add support for PCM, BE & CC links
by mengdong.lin＠linux.intel.com 08 Dec '15

08 Dec '15

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series allows the topology core to create PCM devices dynamically, as well as configure BE and CC DAI links. Mengdong Lin (12): ASoC: Vendor drivers get a link's runtime by snd_soc_get_pcm_runtime() ASoC: Change the PCM runtime array to a list ASoC: Define soc_init_dai_link() to wrap link intialization. ASoC: Change 2nd argument of soc_bind_dai_link() to DAI link pointer ASoC: Implement DAI links in a list & define API to add a link ASoC: Add add_dai_link ops for a soc card ASoC: soc_bind_dai_link() directly returns success for a bound DAI link ASoC: Bind new DAI links after probing components ASoC: The soc card can have auxiliary components ASoC: Support adding a DAI dynamically ASoC: topology: Add PCM DAIs dynamically when loading them ASoC: topology: Add support for FE DAI links Vedang Patel (1): ASoC: topology: Add support for BE and CC DAI Links include/sound/soc-dai.h | 1 + include/sound/soc-topology.h | 22 +- include/sound/soc.h | 38 +- sound/soc/fsl/fsl-asoc-card.c | 5 +- sound/soc/fsl/imx-wm8962.c | 10 +- sound/soc/generic/simple-card.c | 12 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 12 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 7 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 7 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 7 +- sound/soc/pxa/mioa701_wm9713.c | 6 +- sound/soc/samsung/bells.c | 40 +- sound/soc/samsung/littlemill.c | 32 +- sound/soc/samsung/odroidx2_max98090.c | 9 +- sound/soc/samsung/snow.c | 9 +- sound/soc/samsung/speyside.c | 12 +- sound/soc/samsung/tobermory.c | 21 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/rcar/rsrc-card.c | 6 +- sound/soc/soc-core.c | 629 ++++++++++++++++++--------- sound/soc/soc-dapm.c | 7 +- sound/soc/soc-pcm.c | 22 +- sound/soc/soc-topology.c | 284 ++++++++++-- sound/soc/tegra/tegra_wm8903.c | 3 +- 24 files changed, 879 insertions(+), 324 deletions(-) -- 1.9.1

4 18

[alsa-devel] [PATCH] ASoC: fsl: use correct format string for dma_addr_t
by Arnd Bergmann 08 Dec '15

08 Dec '15

We get a warning for the imx-pcm-fiq driver when CONFIG_LPAE is enabled on ARM, because dma_addr_t is 64-bit then: sound/soc/fsl/imx-pcm-fiq.c: In function 'snd_imx_pcm_mmap': sound/soc/fsl/imx-pcm-fiq.c:223:107: warning: format '%x' expects argument of type 'unsigned int', but argument 6 has type 'dma_addr_t {aka long long unsigned int}' [-Wformat=] This changes the printk to use the correct format string for printing a dma_addr_t. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/fsl/imx-pcm-fiq.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 7abf6a079574..49d7513f429e 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -220,9 +220,9 @@ static int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, ret = dma_mmap_writecombine(substream->pcm->card->dev, vma, runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); - pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, + pr_debug("%s: ret: %d %p %pad 0x%08x\n", __func__, ret, runtime->dma_area, - runtime->dma_addr, + &runtime->dma_addr, runtime->dma_bytes); return ret; } -- 2.1.0.rc2

2 1

[alsa-devel] [PATCH 01/19] ASoC: da7218: avoid 64-bit compile warning
by Arnd Bergmann 08 Dec '15

08 Dec '15

When building the da7218 driver on a 64-bit architecture, we get a harmless warning: sound/soc/codecs/da7218.c: In function 'da7218_of_get_id': sound/soc/codecs/da7218.c:2261:10: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast] This changes the code to use uintptr_t to ensure we have an integer type of the same size as a pointer and won't get a warning on any architecture. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 4d50934abd22 ("ASoC: da7218: Add da7218 codec driver") --- sound/soc/codecs/da7218.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index 4fee7aeaadc7..eacde128c4d6 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -2258,7 +2258,7 @@ static inline int da7218_of_get_id(struct device *dev) const struct of_device_id *id = of_match_device(da7218_of_match, dev); if (id) - return (int) id->data; + return (uintptr_t)id->data; else return -EINVAL; } -- 2.1.0.rc2

2 2

[alsa-devel] [PATCH] ASoC: Add max98926 codec driver
by anish kumar 08 Dec '15

08 Dec '15

Changes since v1: Made some widgets as supply widgets Changed the names a bit Moved some registers to volatile struct Some coding style changes added the print for diagnostics Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- .../devicetree/bindings/sound/max98926.txt | 32 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98926.c | 599 +++++++++++++++ sound/soc/codecs/max98926.h | 848 +++++++++++++++++++++ 5 files changed, 1486 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98926.txt create mode 100644 sound/soc/codecs/max98926.c create mode 100644 sound/soc/codecs/max98926.h diff --git a/Documentation/devicetree/bindings/sound/max98926.txt b/Documentation/devicetree/bindings/sound/max98926.txt new file mode 100644 index 0000000..0b7f4e4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98926.txt @@ -0,0 +1,32 @@ +max98926 audio CODEC + +This device supports I2C. + +Required properties: + + - compatible : "maxim,max98926" + + - vmon-slot-no : slot number used to send voltage information + or in inteleave mode this will be used as + interleave slot. + + - imon-slot-no : slot number used to send current information + + - interleave-mode : When using two MAX98926 in a system it is + possible to create ADC data that that will + overflow the frame size. Digital Audio Interleave + mode provides a means to output VMON and IMON data + from two devices on a single DOUT line when running + smaller frames sizes such as 32 BCLKS per LRCLK or + 48 BCLKS per LRCLK. + + - reg : the I2C address of the device for I2C + +Example: + +codec: max98926@1a { + compatible = "maxim,max98926"; + vmon-slot-no = <0>; + imon-slot-no = <2>; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cfdafc4..3fe8bd3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -39,6 +39,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4613 if I2C select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C + select SND_SOC_MAX98926 if I2C select SND_SOC_AK4671 if I2C select SND_SOC_AK5386 select SND_SOC_ALC5623 if I2C @@ -331,6 +332,10 @@ config SND_SOC_AK4613 config SND_SOC_AK4641 tristate +config SND_SOC_MAX98926 + tristate "MAX98926 CODEC" + depends on I2C + config SND_SOC_AK4642 tristate "AKM AK4642 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f632fc4..a38b536 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -29,6 +29,7 @@ snd-soc-ak4554-objs := ak4554.o snd-soc-ak4613-objs := ak4613.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o +snd-soc-max98926-objs := max98926.o snd-soc-ak4671-objs := ak4671.o snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o @@ -222,6 +223,7 @@ obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o obj-$(CONFIG_SND_SOC_AK4613) += snd-soc-ak4613.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o +obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_AK5386) += snd-soc-ak5386.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c new file mode 100644 index 0000000..3ec54ce --- /dev/null +++ b/sound/soc/codecs/max98926.c @@ -0,0 +1,599 @@ +/* + * max98926.c -- ALSA SoC MAX98926 driver + * Copyright 2013-15 Maxim Integrated Products + * 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/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/cdev.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98926.h" + +static const char * const max98926_boost_voltage_txt[] = { + "8.5V", "8.25V", "8.0V", "7.75V", "7.5V", "7.25V", "7.0V", "6.75V", + "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V" +}; + +static const char *const max98926_dai_txt[] = { + "Left", "Right", "LeftRight", "LeftRightDiv2", +}; + +static const char *const max98926_pdm_ch_text[] = { + "Current", "Voltage", +}; + +static const char *const max98926_hpf_cutoff_txt[] = { + "Disable", "DC Block", "100Hz", + "200Hz", "400Hz", "800Hz", +}; + +static struct reg_default max98926_reg[] = { + { 0x0B, 0x00 }, /* IRQ Enable0 */ + { 0x0C, 0x00 }, /* IRQ Enable1 */ + { 0x0D, 0x00 }, /* IRQ Enable2 */ + { 0x0E, 0x00 }, /* IRQ Clear0 */ + { 0x0F, 0x00 }, /* IRQ Clear1 */ + { 0x10, 0x00 }, /* IRQ Clear2 */ + { 0x11, 0xC0 }, /* Map0 */ + { 0x12, 0x00 }, /* Map1 */ + { 0x13, 0x00 }, /* Map2 */ + { 0x14, 0xF0 }, /* Map3 */ + { 0x15, 0x00 }, /* Map4 */ + { 0x16, 0xAB }, /* Map5 */ + { 0x17, 0x89 }, /* Map6 */ + { 0x18, 0x00 }, /* Map7 */ + { 0x19, 0x00 }, /* Map8 */ + { 0x1A, 0x04 }, /* DAI Clock Mode 1 */ + { 0x1B, 0x00 }, /* DAI Clock Mode 2 */ + { 0x1C, 0x00 }, /* DAI Clock Divider Denominator MSBs */ + { 0x1D, 0x00 }, /* DAI Clock Divider Denominator LSBs */ + { 0x1E, 0xF0 }, /* DAI Clock Divider Numerator MSBs */ + { 0x1F, 0x00 }, /* DAI Clock Divider Numerator LSBs */ + { 0x20, 0x50 }, /* Format */ + { 0x21, 0x00 }, /* TDM Slot Select */ + { 0x22, 0x00 }, /* DOUT Configuration VMON */ + { 0x23, 0x00 }, /* DOUT Configuration IMON */ + { 0x24, 0x00 }, /* DOUT Configuration VBAT */ + { 0x25, 0x00 }, /* DOUT Configuration VBST */ + { 0x26, 0x00 }, /* DOUT Configuration FLAG */ + { 0x27, 0xFF }, /* DOUT HiZ Configuration 1 */ + { 0x28, 0xFF }, /* DOUT HiZ Configuration 2 */ + { 0x29, 0xFF }, /* DOUT HiZ Configuration 3 */ + { 0x2A, 0xFF }, /* DOUT HiZ Configuration 4 */ + { 0x2B, 0x02 }, /* DOUT Drive Strength */ + { 0x2C, 0x90 }, /* Filters */ + { 0x2D, 0x00 }, /* Gain */ + { 0x2E, 0x02 }, /* Gain Ramping */ + { 0x2F, 0x00 }, /* Speaker Amplifier */ + { 0x30, 0x0A }, /* Threshold */ + { 0x31, 0x00 }, /* ALC Attack */ + { 0x32, 0x80 }, /* ALC Atten and Release */ + { 0x33, 0x00 }, /* ALC Infinite Hold Release */ + { 0x34, 0x92 }, /* ALC Configuration */ + { 0x35, 0x01 }, /* Boost Converter */ + { 0x36, 0x00 }, /* Block Enable */ + { 0x37, 0x00 }, /* Configuration */ + { 0x38, 0x00 }, /* Global Enable */ + { 0x3A, 0x00 }, /* Boost Limiter */ +}; + +static const struct soc_enum max98926_voltage_enum[] = { + SOC_ENUM_SINGLE(MAX98926_DAI_CLK_DIV_N_LSBS, 0, + ARRAY_SIZE(max98926_pdm_ch_text), + max98926_pdm_ch_text), +}; + +static const struct snd_kcontrol_new max98926_voltage_control = + SOC_DAPM_ENUM("Route", max98926_voltage_enum); + +static const struct soc_enum max98926_current_enum[] = { + SOC_ENUM_SINGLE(MAX98926_DAI_CLK_DIV_N_LSBS, + MAX98926_PDM_SOURCE_1_SHIFT, + ARRAY_SIZE(max98926_pdm_ch_text), + max98926_pdm_ch_text), +}; + +static const struct snd_kcontrol_new max98926_current_control = + SOC_DAPM_ENUM("Route", max98926_current_enum); + +static const struct snd_kcontrol_new max98926_mixer_controls[] = { + SOC_DAPM_SINGLE("PCM", MAX98926_SPK_AMP, + MAX98926_INSELECT_MODE_SHIFT, 0, 0), + SOC_DAPM_SINGLE("PDM", MAX98926_SPK_AMP, + MAX98926_INSELECT_MODE_SHIFT, 1, 0), +}; + +static const struct snd_kcontrol_new max98926_dai_controls[] = { + SOC_DAPM_SINGLE("Left", MAX98926_GAIN, + MAX98926_DAC_IN_SEL_SHIFT, 0, 0), + SOC_DAPM_SINGLE("Right", MAX98926_GAIN, + MAX98926_DAC_IN_SEL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LeftRight", MAX98926_GAIN, + MAX98926_DAC_IN_SEL_SHIFT, 2, 0), + SOC_DAPM_SINGLE("LeftRightDiv2", MAX98926_GAIN, + MAX98926_DAC_IN_SEL_SHIFT, 3, 0), +}; + +static const struct snd_soc_dapm_widget max98926_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("Amp Enable", NULL, MAX98926_BLOCK_ENABLE, + MAX98926_SPK_EN_SHIFT, 0), + SND_SOC_DAPM_SUPPLY("Global Enable", MAX98926_GLOBAL_ENABLE, + MAX98926_EN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("VI Enable", MAX98926_BLOCK_ENABLE, + MAX98926_ADC_IMON_EN_WIDTH | + MAX98926_ADC_VMON_EN_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("BST Enable", MAX98926_BLOCK_ENABLE, + MAX98926_BST_EN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("BE_OUT"), + SND_SOC_DAPM_MIXER("PCM Sel", MAX98926_SPK_AMP, + MAX98926_INSELECT_MODE_SHIFT, 0, + &max98926_mixer_controls[0], + ARRAY_SIZE(max98926_mixer_controls)), + SND_SOC_DAPM_MIXER("DAI Sel", + MAX98926_GAIN, MAX98926_DAC_IN_SEL_SHIFT, 0, + &max98926_dai_controls[0], + ARRAY_SIZE(max98926_dai_controls)), + SND_SOC_DAPM_MUX("PDM CH1 Source", + MAX98926_DAI_CLK_DIV_N_LSBS, + MAX98926_PDM_CURRENT_SHIFT, + 0, &max98926_current_control), + SND_SOC_DAPM_MUX("PDM CH0 Source", + MAX98926_DAI_CLK_DIV_N_LSBS, + MAX98926_PDM_VOLTAGE_SHIFT, + 0, &max98926_voltage_control), +}; + +static const struct snd_soc_dapm_route max98926_audio_map[] = { + {"VI Enable", NULL, "DAI_OUT"}, + {"DAI Sel", "Left", "VI Enable"}, + {"DAI Sel", "Right", "VI Enable"}, + {"DAI Sel", "LeftRight", "VI Enable"}, + {"DAI Sel", "LeftRightDiv2", "VI Enable"}, + {"PCM Sel", "PCM", "DAI Sel"}, + + {"PDM CH1 Source", "Current", "DAI_OUT"}, + {"PDM CH1 Source", "Voltage", "DAI_OUT"}, + {"PDM CH0 Source", "Current", "DAI_OUT"}, + {"PDM CH0 Source", "Voltage", "DAI_OUT"}, + {"PCM Sel", "Analog", "PDM CH1 Source"}, + {"PCM Sel", "Analog", "PDM CH0 Source"}, + {"Amp Enable", NULL, "PCM Sel"}, + + {"BST Enable", NULL, "Amp Enable"}, + {"BE_OUT", NULL, "BST Enable"}, +}; + +static bool max98926_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98926_VBAT_DATA: + case MAX98926_VBST_DATA: + case MAX98926_LIVE_STATUS0: + case MAX98926_LIVE_STATUS1: + case MAX98926_LIVE_STATUS2: + case MAX98926_STATE0: + case MAX98926_STATE1: + case MAX98926_STATE2: + case MAX98926_FLAG0: + case MAX98926_FLAG1: + case MAX98926_FLAG2: + case MAX98926_VERSION: + return true; + default: + return false; + } +} + +static bool max98926_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98926_IRQ_CLEAR0: + case MAX98926_IRQ_CLEAR1: + case MAX98926_IRQ_CLEAR2: + case MAX98926_ALC_HOLD_RLS: + return false; + default: + return true; + } +}; + +DECLARE_TLV_DB_SCALE(max98926_spk_tlv, -600, 100, 0); + +static SOC_ENUM_SINGLE_DECL(max98926_dac_hpf_cutoff, + MAX98926_FILTERS, MAX98926_DAC_HPF_SHIFT, + max98926_hpf_cutoff_txt); + +static SOC_ENUM_SINGLE_DECL(max98926_boost_voltage, + MAX98926_CONFIGURATION, MAX98926_BST_VOUT_SHIFT, + max98926_boost_voltage_txt); + +static const struct snd_kcontrol_new max98926_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98926_GAIN, + MAX98926_SPK_GAIN_SHIFT, + (1<<MAX98926_SPK_GAIN_WIDTH)-1, 0, + max98926_spk_tlv), + SOC_SINGLE("Ramp Switch", MAX98926_GAIN_RAMPING, + MAX98926_SPK_RMP_EN_SHIFT, 1, 0), + SOC_SINGLE("ZCD Switch", MAX98926_GAIN_RAMPING, + MAX98926_SPK_ZCD_EN_SHIFT, 1, 0), + SOC_SINGLE("ALC Switch", MAX98926_THRESHOLD, + MAX98926_ALC_EN_SHIFT, 1, 0), + SOC_SINGLE("ALC Threshold", MAX98926_THRESHOLD, + MAX98926_ALC_TH_SHIFT, + (1<<MAX98926_ALC_TH_WIDTH)-1, 0), + SOC_ENUM("Boost Output Voltage", max98926_boost_voltage), + SOC_ENUM("DAC HPF Cutoff", max98926_dac_hpf_cutoff), + SOC_DOUBLE("PDM Channel One", MAX98926_DAI_CLK_DIV_N_LSBS, + MAX98926_PDM_CHANNEL_1_SHIFT, + MAX98926_PDM_CHANNEL_1_HIZ, 1, 0), + SOC_DOUBLE("PDM Channel Zero", MAX98926_DAI_CLK_DIV_N_LSBS, + MAX98926_PDM_CHANNEL_0_SHIFT, + MAX98926_PDM_CHANNEL_0_HIZ, 1, 0), +}; + +static const struct { + int rate; + int sr; +} rate_table[] = { + { + .rate = 8000, + .sr = 0, + }, + { + .rate = 11025, + .sr = 1, + }, + { + .rate = 12000, + .sr = 2, + }, + { + .rate = 16000, + .sr = 3, + }, + { + .rate = 22050, + .sr = 4, + }, + { + .rate = 24000, + .sr = 5, + }, + { + .rate = 32000, + .sr = 6, + }, + { + .rate = 44100, + .sr = 7, + }, + { + .rate = 48000, + .sr = 8, + }, +}; + +static void max98926_set_sense_data(struct max98926_priv *max98926) +{ + if (max98926->interleave_mode) { + /* set VMON slots */ + regmap_update_bits(max98926->regmap, + MAX98926_DOUT_CFG_VMON, + MAX98926_DAI_VMON_EN_MASK, + MAX98926_DAI_VMON_EN_MASK); + regmap_update_bits(max98926->regmap, + MAX98926_DOUT_CFG_VMON, + MAX98926_DAI_VMON_SLOT_MASK, + max98926->v_slot); + /* set IMON slots */ + regmap_update_bits(max98926->regmap, + MAX98926_DOUT_CFG_IMON, + MAX98926_DAI_IMON_EN_MASK, + MAX98926_DAI_IMON_EN_MASK); + regmap_update_bits(max98926->regmap, + MAX98926_DOUT_CFG_IMON, + MAX98926_DAI_IMON_SLOT_MASK, + max98926->i_slot); + } else { + /* enable interleave mode */ + regmap_update_bits(max98926->regmap, + MAX98926_FORMAT, + MAX98926_DAI_INTERLEAVE_MASK, + MAX98926_DAI_INTERLEAVE_MASK); + /* set interleave slots */ + regmap_update_bits(max98926->regmap, + MAX98926_DOUT_CFG_VBAT, + MAX98926_DAI_INTERLEAVE_SLOT_MASK, + max98926->v_slot); + } +} + +static int max98926_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98926_priv *max98926 = snd_soc_codec_get_drvdata(codec); + unsigned int invert = 0; + + dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + max98926_set_sense_data(max98926); + break; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + invert = MAX98926_DAI_WCI_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + invert = MAX98926_DAI_BCI_MASK; + break; + case SND_SOC_DAIFMT_IB_IF: + invert = MAX98926_DAI_BCI_MASK | MAX98926_DAI_WCI_MASK; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + regmap_write(max98926->regmap, + MAX98926_FORMAT, MAX98926_DAI_DLY_MASK); + regmap_update_bits(max98926->regmap, MAX98926_FORMAT, + MAX98926_DAI_BCI_MASK, invert); + return 0; +} + +static int max98926_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int dai_sr = -EINVAL; + int rate = params_rate(params), i; + struct snd_soc_codec *codec = dai->codec; + struct max98926_priv *max98926 = snd_soc_codec_get_drvdata(codec); + /* BCLK/LRCLK ratio calculation */ + int blr_clk_ratio = params_channels(params) * max98926->ch_size; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98926->regmap, + MAX98926_FORMAT, + MAX98926_DAI_CHANSZ_MASK, + MAX98926_DAI_CHANSZ_16); + max98926->ch_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98926->regmap, + MAX98926_FORMAT, + MAX98926_DAI_CHANSZ_MASK, + MAX98926_DAI_CHANSZ_24); + max98926->ch_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98926->regmap, + MAX98926_FORMAT, + MAX98926_DAI_CHANSZ_MASK, + MAX98926_DAI_CHANSZ_32); + max98926->ch_size = 32; + break; + default: + dev_dbg(codec->dev, "format unsupported %d", + params_format(params)); + return -EINVAL; + } + + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98926->regmap, + MAX98926_DAI_CLK_MODE2, + MAX98926_DAI_BSEL_MASK, + MAX98926_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98926->regmap, + MAX98926_DAI_CLK_MODE2, + MAX98926_DAI_BSEL_MASK, + MAX98926_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98926->regmap, + MAX98926_DAI_CLK_MODE2, + MAX98926_DAI_BSEL_MASK, + MAX98926_DAI_BSEL_64); + break; + default: + return -EINVAL; + } + + /* find the closest rate */ + for (i = 0; i < ARRAY_SIZE(rate_table); i++) { + if (rate_table[i].rate >= rate) { + dai_sr = rate_table[i].sr; + break; + } + } + if (dai_sr < 0) + return -EINVAL; + + /* set DAI_SR to correct LRCLK frequency */ + regmap_update_bits(max98926->regmap, + MAX98926_DAI_CLK_MODE2, + MAX98926_DAI_SR_MASK, dai_sr << MAX98926_DAI_SR_SHIFT); + return 0; +} + +#define MAX98926_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops max98926_dai_ops = { + .set_fmt = max98926_dai_set_fmt, + .hw_params = max98926_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98926_dai[] = { +{ + .name = "max98926-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98926_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98926_FORMATS, + }, + .ops = &max98926_dai_ops, +} +}; + +static int max98926_probe(struct snd_soc_codec *codec) +{ + struct max98926_priv *max98926 = snd_soc_codec_get_drvdata(codec); + + max98926->codec = codec; + codec->control_data = max98926->regmap; + regmap_write(max98926->regmap, MAX98926_GLOBAL_ENABLE, 0x00); + regmap_write(max98926->regmap, MAX98926_TDM_SLOT_SELECT, 0xC8); + regmap_write(max98926->regmap, MAX98926_DOUT_HIZ_CFG1, 0xFF); + regmap_write(max98926->regmap, MAX98926_DOUT_HIZ_CFG2, 0xFF); + regmap_write(max98926->regmap, MAX98926_DOUT_HIZ_CFG3, 0xFF); + regmap_write(max98926->regmap, MAX98926_DOUT_HIZ_CFG4, 0xF0); + regmap_write(max98926->regmap, MAX98926_FILTERS, 0xD8); + regmap_write(max98926->regmap, MAX98926_ALC_CONFIGURATION, 0xF8); + regmap_write(max98926->regmap, MAX98926_CONFIGURATION, 0xF0); + /* Disable ALC muting */ + regmap_write(max98926->regmap, MAX98926_BOOST_LIMITER, 0xF8); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max98926 = { + .probe = max98926_probe, + .controls = max98926_snd_controls, + .num_controls = ARRAY_SIZE(max98926_snd_controls), + .dapm_routes = max98926_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98926_audio_map), + .dapm_widgets = max98926_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98926_dapm_widgets), +}; + +static struct regmap_config max98926_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98926_VERSION, + .reg_defaults = max98926_reg, + .num_reg_defaults = ARRAY_SIZE(max98926_reg), + .volatile_reg = max98926_volatile_register, + .readable_reg = max98926_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98926_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + int ret, reg; + u32 value; + struct max98926_priv *max98926; + + max98926 = devm_kzalloc(&i2c->dev, + sizeof(*max98926), GFP_KERNEL); + if (!max98926) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98926); + max98926->regmap = devm_regmap_init_i2c(i2c, &max98926_regmap); + if (IS_ERR(max98926->regmap)) { + ret = PTR_ERR(max98926->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + goto err_out; + } + if (of_property_read_bool(i2c->dev.of_node, "interleave-mode")) + max98926->interleave_mode = true; + + if (!of_property_read_u32(i2c->dev.of_node, "vmon-slot-no", &value)) { + if (value > MAX98926_DAI_VMON_SLOT_1E_1F) { + dev_err(&i2c->dev, "vmon slot number is wrong:\n"); + return -EINVAL; + } + max98926->v_slot = value; + } + if (!of_property_read_u32(i2c->dev.of_node, "imon-slot-no", &value)) { + if (value > MAX98926_DAI_IMON_SLOT_1E_1F) { + dev_err(&i2c->dev, "imon slot number is wrong:\n"); + return -EINVAL; + } + max98926->i_slot = value; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98926, + max98926_dai, ARRAY_SIZE(max98926_dai)); + if (ret < 0) + dev_err(&i2c->dev, + "Failed to register codec: %d\n", ret); + ret = regmap_read(max98926->regmap, + MAX98926_VERSION, &reg); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read: %x\n", reg); + return -EINVAL; + } + dev_info(&i2c->dev, "device version: %x\n", reg); +err_out: + return ret; +} + +static int max98926_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98926_i2c_id[] = { + { "max98926", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max98926_i2c_id); + +static const struct of_device_id max98926_of_match[] = { + { .compatible = "maxim, max98926", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98926_of_match); + +static struct i2c_driver max98926_i2c_driver = { + .driver = { + .name = "max98926", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(max98926_of_match), + .pm = NULL, + }, + .probe = max98926_i2c_probe, + .remove = max98926_i2c_remove, + .id_table = max98926_i2c_id, +}; + +module_i2c_driver(max98926_i2c_driver) +MODULE_DESCRIPTION("ALSA SoC MAX98926 driver"); +MODULE_AUTHOR("Anish kumar <anish.kumar(a)maximintegrated.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98926.h b/sound/soc/codecs/max98926.h new file mode 100644 index 0000000..5243f82 --- /dev/null +++ b/sound/soc/codecs/max98926.h @@ -0,0 +1,848 @@ +/* + * max98926.h -- MAX98926 ALSA SoC Audio driver + * Copyright 2013-2015 Maxim Integrated Products + * 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 _MAX98926_H +#define _MAX98926_H + +#define MAX98926_CHIP_VERSION 0x40 +#define MAX98926_CHIP_VERSION1 0x50 + +#define MAX98926_VBAT_DATA 0x00 +#define MAX98926_VBST_DATA 0x01 +#define MAX98926_LIVE_STATUS0 0x02 +#define MAX98926_LIVE_STATUS1 0x03 +#define MAX98926_LIVE_STATUS2 0x04 +#define MAX98926_STATE0 0x05 +#define MAX98926_STATE1 0x06 +#define MAX98926_STATE2 0x07 +#define MAX98926_FLAG0 0x08 +#define MAX98926_FLAG1 0x09 +#define MAX98926_FLAG2 0x0A +#define MAX98926_IRQ_ENABLE0 0x0B +#define MAX98926_IRQ_ENABLE1 0x0C +#define MAX98926_IRQ_ENABLE2 0x0D +#define MAX98926_IRQ_CLEAR0 0x0E +#define MAX98926_IRQ_CLEAR1 0x0F +#define MAX98926_IRQ_CLEAR2 0x10 +#define MAX98926_MAP0 0x11 +#define MAX98926_MAP1 0x12 +#define MAX98926_MAP2 0x13 +#define MAX98926_MAP3 0x14 +#define MAX98926_MAP4 0x15 +#define MAX98926_MAP5 0x16 +#define MAX98926_MAP6 0x17 +#define MAX98926_MAP7 0x18 +#define MAX98926_MAP8 0x19 +#define MAX98926_DAI_CLK_MODE1 0x1A +#define MAX98926_DAI_CLK_MODE2 0x1B +#define MAX98926_DAI_CLK_DIV_M_MSBS 0x1C +#define MAX98926_DAI_CLK_DIV_M_LSBS 0x1D +#define MAX98926_DAI_CLK_DIV_N_MSBS 0x1E +#define MAX98926_DAI_CLK_DIV_N_LSBS 0x1F +#define MAX98926_FORMAT 0x20 +#define MAX98926_TDM_SLOT_SELECT 0x21 +#define MAX98926_DOUT_CFG_VMON 0x22 +#define MAX98926_DOUT_CFG_IMON 0x23 +#define MAX98926_DOUT_CFG_VBAT 0x24 +#define MAX98926_DOUT_CFG_VBST 0x25 +#define MAX98926_DOUT_CFG_FLAG 0x26 +#define MAX98926_DOUT_HIZ_CFG1 0x27 +#define MAX98926_DOUT_HIZ_CFG2 0x28 +#define MAX98926_DOUT_HIZ_CFG3 0x29 +#define MAX98926_DOUT_HIZ_CFG4 0x2A +#define MAX98926_DOUT_DRV_STRENGTH 0x2B +#define MAX98926_FILTERS 0x2C +#define MAX98926_GAIN 0x2D +#define MAX98926_GAIN_RAMPING 0x2E +#define MAX98926_SPK_AMP 0x2F +#define MAX98926_THRESHOLD 0x30 +#define MAX98926_ALC_ATTACK 0x31 +#define MAX98926_ALC_ATTEN_RLS 0x32 +#define MAX98926_ALC_HOLD_RLS 0x33 +#define MAX98926_ALC_CONFIGURATION 0x34 +#define MAX98926_BOOST_CONVERTER 0x35 +#define MAX98926_BLOCK_ENABLE 0x36 +#define MAX98926_CONFIGURATION 0x37 +#define MAX98926_GLOBAL_ENABLE 0x38 +#define MAX98926_BOOST_LIMITER 0x3A +#define MAX98926_VERSION 0xFF + +#define MAX98926_REG_CNT (MAX98926_R03A_BOOST_LIMITER+1) + +#define MAX98926_PDM_CURRENT_MASK (1<<7) +#define MAX98926_PDM_CURRENT_SHIFT 7 +#define MAX98926_PDM_VOLTAGE_MASK (1<<3) +#define MAX98926_PDM_VOLTAGE_SHIFT 3 +#define MAX98926_PDM_CHANNEL_0_MASK (1<<2) +#define MAX98926_PDM_CHANNEL_0_SHIFT 2 +#define MAX98926_PDM_CHANNEL_1_MASK (1<<6) +#define MAX98926_PDM_CHANNEL_1_SHIFT 6 +#define MAX98926_PDM_CHANNEL_1_HIZ 5 +#define MAX98926_PDM_CHANNEL_0_HIZ 1 +#define MAX98926_PDM_SOURCE_0_SHIFT 0 +#define MAX98926_PDM_SOURCE_0_MASK (1<<0) +#define MAX98926_PDM_SOURCE_1_MASK (1<<4) +#define MAX98926_PDM_SOURCE_1_SHIFT 4 + +/* MAX98926 Register Bit Fields */ + +/* MAX98926_R002_LIVE_STATUS0 */ +#define MAX98926_THERMWARN_STATUS_MASK (1<<3) +#define MAX98926_THERMWARN_STATUS_SHIFT 3 +#define MAX98926_THERMWARN_STATUS_WIDTH 1 +#define MAX98926_THERMSHDN_STATUS_MASK (1<<1) +#define MAX98926_THERMSHDN_STATUS_SHIFT 1 +#define MAX98926_THERMSHDN_STATUS_WIDTH 1 + +/* MAX98926_R003_LIVE_STATUS1 */ +#define MAX98926_SPKCURNT_STATUS_MASK (1<<5) +#define MAX98926_SPKCURNT_STATUS_SHIFT 5 +#define MAX98926_SPKCURNT_STATUS_WIDTH 1 +#define MAX98926_WATCHFAIL_STATUS_MASK (1<<4) +#define MAX98926_WATCHFAIL_STATUS_SHIFT 4 +#define MAX98926_WATCHFAIL_STATUS_WIDTH 1 +#define MAX98926_ALCINFH_STATUS_MASK (1<<3) +#define MAX98926_ALCINFH_STATUS_SHIFT 3 +#define MAX98926_ALCINFH_STATUS_WIDTH 1 +#define MAX98926_ALCACT_STATUS_MASK (1<<2) +#define MAX98926_ALCACT_STATUS_SHIFT 2 +#define MAX98926_ALCACT_STATUS_WIDTH 1 +#define MAX98926_ALCMUT_STATUS_MASK (1<<1) +#define MAX98926_ALCMUT_STATUS_SHIFT 1 +#define MAX98926_ALCMUT_STATUS_WIDTH 1 +#define MAX98926_ACLP_STATUS_MASK (1<<0) +#define MAX98926_ACLP_STATUS_SHIFT 0 +#define MAX98926_ACLP_STATUS_WIDTH 1 + +/* MAX98926_R004_LIVE_STATUS2 */ +#define MAX98926_SLOTOVRN_STATUS_MASK (1<<6) +#define MAX98926_SLOTOVRN_STATUS_SHIFT 6 +#define MAX98926_SLOTOVRN_STATUS_WIDTH 1 +#define MAX98926_INVALSLOT_STATUS_MASK (1<<5) +#define MAX98926_INVALSLOT_STATUS_SHIFT 5 +#define MAX98926_INVALSLOT_STATUS_WIDTH 1 +#define MAX98926_SLOTCNFLT_STATUS_MASK (1<<4) +#define MAX98926_SLOTCNFLT_STATUS_SHIFT 4 +#define MAX98926_SLOTCNFLT_STATUS_WIDTH 1 +#define MAX98926_VBSTOVFL_STATUS_MASK (1<<3) +#define MAX98926_VBSTOVFL_STATUS_SHIFT 3 +#define MAX98926_VBSTOVFL_STATUS_WIDTH 1 +#define MAX98926_VBATOVFL_STATUS_MASK (1<<2) +#define MAX98926_VBATOVFL_STATUS_SHIFT 2 +#define MAX98926_VBATOVFL_STATUS_WIDTH 1 +#define MAX98926_IMONOVFL_STATUS_MASK (1<<1) +#define MAX98926_IMONOVFL_STATUS_SHIFT 1 +#define MAX98926_IMONOVFL_STATUS_WIDTH 1 +#define MAX98926_VMONOVFL_STATUS_MASK (1<<0) +#define MAX98926_VMONOVFL_STATUS_SHIFT 0 +#define MAX98926_VMONOVFL_STATUS_WIDTH 1 + +/* MAX98926_R005_STATE0 */ +#define MAX98926_THERMWARN_END_STATE_MASK (1<<3) +#define MAX98926_THERMWARN_END_STATE_SHIFT 3 +#define MAX98926_THERMWARN_END_STATE_WIDTH 1 +#define MAX98926_THERMWARN_BGN_STATE_MASK (1<<2) +#define MAX98926_THERMWARN_BGN_STATE_SHIFT 1 +#define MAX98926_THERMWARN_BGN_STATE_WIDTH 1 +#define MAX98926_THERMSHDN_END_STATE_MASK (1<<1) +#define MAX98926_THERMSHDN_END_STATE_SHIFT 1 +#define MAX98926_THERMSHDN_END_STATE_WIDTH 1 +#define MAX98926_THERMSHDN_BGN_STATE_MASK (1<<0) +#define MAX98926_THERMSHDN_BGN_STATE_SHIFT 0 +#define MAX98926_THERMSHDN_BGN_STATE_WIDTH 1 + +/* MAX98926_R006_STATE1 */ +#define MAX98926_SPRCURNT_STATE_MASK (1<<5) +#define MAX98926_SPRCURNT_STATE_SHIFT 5 +#define MAX98926_SPRCURNT_STATE_WIDTH 1 +#define MAX98926_WATCHFAIL_STATE_MASK (1<<4) +#define MAX98926_WATCHFAIL_STATE_SHIFT 4 +#define MAX98926_WATCHFAIL_STATE_WIDTH 1 +#define MAX98926_ALCINFH_STATE_MASK (1<<3) +#define MAX98926_ALCINFH_STATE_SHIFT 3 +#define MAX98926_ALCINFH_STATE_WIDTH 1 +#define MAX98926_ALCACT_STATE_MASK (1<<2) +#define MAX98926_ALCACT_STATE_SHIFT 2 +#define MAX98926_ALCACT_STATE_WIDTH 1 +#define MAX98926_ALCMUT_STATE_MASK (1<<1) +#define MAX98926_ALCMUT_STATE_SHIFT 1 +#define MAX98926_ALCMUT_STATE_WIDTH 1 +#define MAX98926_ALCP_STATE_MASK (1<<0) +#define MAX98926_ALCP_STATE_SHIFT 0 +#define MAX98926_ALCP_STATE_WIDTH 1 + +/* MAX98926_R007_STATE2 */ +#define MAX98926_SLOTOVRN_STATE_MASK (1<<6) +#define MAX98926_SLOTOVRN_STATE_SHIFT 6 +#define MAX98926_SLOTOVRN_STATE_WIDTH 1 +#define MAX98926_INVALSLOT_STATE_MASK (1<<5) +#define MAX98926_INVALSLOT_STATE_SHIFT 5 +#define MAX98926_INVALSLOT_STATE_WIDTH 1 +#define MAX98926_SLOTCNFLT_STATE_MASK (1<<4) +#define MAX98926_SLOTCNFLT_STATE_SHIFT 4 +#define MAX98926_SLOTCNFLT_STATE_WIDTH 1 +#define MAX98926_VBSTOVFL_STATE_MASK (1<<3) +#define MAX98926_VBSTOVFL_STATE_SHIFT 3 +#define MAX98926_VBSTOVFL_STATE_WIDTH 1 +#define MAX98926_VBATOVFL_STATE_MASK (1<<2) +#define MAX98926_VBATOVFL_STATE_SHIFT 2 +#define MAX98926_VBATOVFL_STATE_WIDTH 1 +#define MAX98926_IMONOVFL_STATE_MASK (1<<1) +#define MAX98926_IMONOVFL_STATE_SHIFT 1 +#define MAX98926_IMONOVFL_STATE_WIDTH 1 +#define MAX98926_VMONOVFL_STATE_MASK (1<<0) +#define MAX98926_VMONOVFL_STATE_SHIFT 0 +#define MAX98926_VMONOVFL_STATE_WIDTH 1 + +/* MAX98926_R008_FLAG0 */ +#define MAX98926_THERMWARN_END_FLAG_MASK (1<<3) +#define MAX98926_THERMWARN_END_FLAG_SHIFT 3 +#define MAX98926_THERMWARN_END_FLAG_WIDTH 1 +#define MAX98926_THERMWARN_BGN_FLAG_MASK (1<<2) +#define MAX98926_THERMWARN_BGN_FLAG_SHIFT 2 +#define MAX98926_THERMWARN_BGN_FLAG_WIDTH 1 +#define MAX98926_THERMSHDN_END_FLAG_MASK (1<<1) +#define MAX98926_THERMSHDN_END_FLAG_SHIFT 1 +#define MAX98926_THERMSHDN_END_FLAG_WIDTH 1 +#define MAX98926_THERMSHDN_BGN_FLAG_MASK (1<<0) +#define MAX98926_THERMSHDN_BGN_FLAG_SHIFT 0 +#define MAX98926_THERMSHDN_BGN_FLAG_WIDTH 1 + +/* MAX98926_R009_FLAG1 */ +#define MAX98926_SPKCURNT_FLAG_MASK (1<<5) +#define MAX98926_SPKCURNT_FLAG_SHIFT 5 +#define MAX98926_SPKCURNT_FLAG_WIDTH 1 +#define MAX98926_WATCHFAIL_FLAG_MASK (1<<4) +#define MAX98926_WATCHFAIL_FLAG_SHIFT 4 +#define MAX98926_WATCHFAIL_FLAG_WIDTH 1 +#define MAX98926_ALCINFH_FLAG_MASK (1<<3) +#define MAX98926_ALCINFH_FLAG_SHIFT 3 +#define MAX98926_ALCINFH_FLAG_WIDTH 1 +#define MAX98926_ALCACT_FLAG_MASK (1<<2) +#define MAX98926_ALCACT_FLAG_SHIFT 2 +#define MAX98926_ALCACT_FLAG_WIDTH 1 +#define MAX98926_ALCMUT_FLAG_MASK (1<<1) +#define MAX98926_ALCMUT_FLAG_SHIFT 1 +#define MAX98926_ALCMUT_FLAG_WIDTH 1 +#define MAX98926_ALCP_FLAG_MASK (1<<0) +#define MAX98926_ALCP_FLAG_SHIFT 0 +#define MAX98926_ALCP_FLAG_WIDTH 1 + +/* MAX98926_R00A_FLAG2 */ +#define MAX98926_SLOTOVRN_FLAG_MASK (1<<6) +#define MAX98926_SLOTOVRN_FLAG_SHIFT 6 +#define MAX98926_SLOTOVRN_FLAG_WIDTH 1 +#define MAX98926_INVALSLOT_FLAG_MASK (1<<5) +#define MAX98926_INVALSLOT_FLAG_SHIFT 5 +#define MAX98926_INVALSLOT_FLAG_WIDTH 1 +#define MAX98926_SLOTCNFLT_FLAG_MASK (1<<4) +#define MAX98926_SLOTCNFLT_FLAG_SHIFT 4 +#define MAX98926_SLOTCNFLT_FLAG_WIDTH 1 +#define MAX98926_VBSTOVFL_FLAG_MASK (1<<3) +#define MAX98926_VBSTOVFL_FLAG_SHIFT 3 +#define MAX98926_VBSTOVFL_FLAG_WIDTH 1 +#define MAX98926_VBATOVFL_FLAG_MASK (1<<2) +#define MAX98926_VBATOVFL_FLAG_SHIFT 2 +#define MAX98926_VBATOVFL_FLAG_WIDTH 1 +#define MAX98926_IMONOVFL_FLAG_MASK (1<<1) +#define MAX98926_IMONOVFL_FLAG_SHIFT 1 +#define MAX98926_IMONOVFL_FLAG_WIDTH 1 +#define MAX98926_VMONOVFL_FLAG_MASK (1<<0) +#define MAX98926_VMONOVFL_FLAG_SHIFT 0 +#define MAX98926_VMONOVFL_FLAG_WIDTH 1 + +/* MAX98926_R00B_IRQ_ENABLE0 */ +#define MAX98926_THERMWARN_END_EN_MASK (1<<3) +#define MAX98926_THERMWARN_END_EN_SHIFT 3 +#define MAX98926_THERMWARN_END_EN_WIDTH 1 +#define MAX98926_THERMWARN_BGN_EN_MASK (1<<2) +#define MAX98926_THERMWARN_BGN_EN_SHIFT 2 +#define MAX98926_THERMWARN_BGN_EN_WIDTH 1 +#define MAX98926_THERMSHDN_END_EN_MASK (1<<1) +#define MAX98926_THERMSHDN_END_EN_SHIFT 1 +#define MAX98926_THERMSHDN_END_EN_WIDTH 1 +#define MAX98926_THERMSHDN_BGN_EN_MASK (1<<0) +#define MAX98926_THERMSHDN_BGN_EN_SHIFT 0 +#define MAX98926_THERMSHDN_BGN_EN_WIDTH 1 + +/* MAX98926_R00C_IRQ_ENABLE1 */ +#define MAX98926_SPKCURNT_EN_MASK (1<<5) +#define MAX98926_SPKCURNT_EN_SHIFT 5 +#define MAX98926_SPKCURNT_EN_WIDTH 1 +#define MAX98926_WATCHFAIL_EN_MASK (1<<4) +#define MAX98926_WATCHFAIL_EN_SHIFT 4 +#define MAX98926_WATCHFAIL_EN_WIDTH 1 +#define MAX98926_ALCINFH_EN_MASK (1<<3) +#define MAX98926_ALCINFH_EN_SHIFT 3 +#define MAX98926_ALCINFH_EN_WIDTH 1 +#define MAX98926_ALCACT_EN_MASK (1<<2) +#define MAX98926_ALCACT_EN_SHIFT 2 +#define MAX98926_ALCACT_EN_WIDTH 1 +#define MAX98926_ALCMUT_EN_MASK (1<<1) +#define MAX98926_ALCMUT_EN_SHIFT 1 +#define MAX98926_ALCMUT_EN_WIDTH 1 +#define MAX98926_ALCP_EN_MASK (1<<0) +#define MAX98926_ALCP_EN_SHIFT 0 +#define MAX98926_ALCP_EN_WIDTH 1 + +/* MAX98926_R00D_IRQ_ENABLE2 */ +#define MAX98926_SLOTOVRN_EN_MASK (1<<6) +#define MAX98926_SLOTOVRN_EN_SHIFT 6 +#define MAX98926_SLOTOVRN_EN_WIDTH 1 +#define MAX98926_INVALSLOT_EN_MASK (1<<5) +#define MAX98926_INVALSLOT_EN_SHIFT 5 +#define MAX98926_INVALSLOT_EN_WIDTH 1 +#define MAX98926_SLOTCNFLT_EN_MASK (1<<4) +#define MAX98926_SLOTCNFLT_EN_SHIFT 4 +#define MAX98926_SLOTCNFLT_EN_WIDTH 1 +#define MAX98926_VBSTOVFL_EN_MASK (1<<3) +#define MAX98926_VBSTOVFL_EN_SHIFT 3 +#define MAX98926_VBSTOVFL_EN_WIDTH 1 +#define MAX98926_VBATOVFL_EN_MASK (1<<2) +#define MAX98926_VBATOVFL_EN_SHIFT 2 +#define MAX98926_VBATOVFL_EN_WIDTH 1 +#define MAX98926_IMONOVFL_EN_MASK (1<<1) +#define MAX98926_IMONOVFL_EN_SHIFT 1 +#define MAX98926_IMONOVFL_EN_WIDTH 1 +#define MAX98926_VMONOVFL_EN_MASK (1<<0) +#define MAX98926_VMONOVFL_EN_SHIFT 0 +#define MAX98926_VMONOVFL_EN_WIDTH 1 + +/* MAX98926_R00E_IRQ_CLEAR0 */ +#define MAX98926_THERMWARN_END_CLR_MASK (1<<3) +#define MAX98926_THERMWARN_END_CLR_SHIFT 3 +#define MAX98926_THERMWARN_END_CLR_WIDTH 1 +#define MAX98926_THERMWARN_BGN_CLR_MASK (1<<2) +#define MAX98926_THERMWARN_BGN_CLR_SHIFT 2 +#define MAX98926_THERMWARN_BGN_CLR_WIDTH 1 +#define MAX98926_THERMSHDN_END_CLR_MASK (1<<1) +#define MAX98926_THERMSHDN_END_CLR_SHIFT 1 +#define MAX98926_THERMSHDN_END_CLR_WIDTH 1 +#define MAX98926_THERMSHDN_BGN_CLR_MASK (1<<0) +#define MAX98926_THERMSHDN_BGN_CLR_SHIFT 0 +#define MAX98926_THERMSHDN_BGN_CLR_WIDTH 1 + +/* MAX98926_R00F_IRQ_CLEAR1 */ +#define MAX98926_SPKCURNT_CLR_MASK (1<<5) +#define MAX98926_SPKCURNT_CLR_SHIFT 5 +#define MAX98926_SPKCURNT_CLR_WIDTH 1 +#define MAX98926_WATCHFAIL_CLR_MASK (1<<4) +#define MAX98926_WATCHFAIL_CLR_SHIFT 4 +#define MAX98926_WATCHFAIL_CLR_WIDTH 1 +#define MAX98926_ALCINFH_CLR_MASK (1<<3) +#define MAX98926_ALCINFH_CLR_SHIFT 3 +#define MAX98926_ALCINFH_CLR_WIDTH 1 +#define MAX98926_ALCACT_CLR_MASK (1<<2) +#define MAX98926_ALCACT_CLR_SHIFT 2 +#define MAX98926_ALCACT_CLR_WIDTH 1 +#define MAX98926_ALCMUT_CLR_MASK (1<<1) +#define MAX98926_ALCMUT_CLR_SHIFT 1 +#define MAX98926_ALCMUT_CLR_WIDTH 1 +#define MAX98926_ALCP_CLR_MASK (1<<0) +#define MAX98926_ALCP_CLR_SHIFT 0 +#define MAX98926_ALCP_CLR_WIDTH 1 + +/* MAX98926_R010_IRQ_CLEAR2 */ +#define MAX98926_SLOTOVRN_CLR_MASK (1<<6) +#define MAX98926_SLOTOVRN_CLR_SHIFT 6 +#define MAX98926_SLOTOVRN_CLR_WIDTH 1 +#define MAX98926_INVALSLOT_CLR_MASK (1<<5) +#define MAX98926_INVALSLOT_CLR_SHIFT 5 +#define MAX98926_INVALSLOT_CLR_WIDTH 1 +#define MAX98926_SLOTCNFLT_CLR_MASK (1<<4) +#define MAX98926_SLOTCNFLT_CLR_SHIFT 4 +#define MAX98926_SLOTCNFLT_CLR_WIDTH 1 +#define MAX98926_VBSTOVFL_CLR_MASK (1<<3) +#define MAX98926_VBSTOVFL_CLR_SHIFT 3 +#define MAX98926_VBSTOVFL_CLR_WIDTH 1 +#define MAX98926_VBATOVFL_CLR_MASK (1<<2) +#define MAX98926_VBATOVFL_CLR_SHIFT 2 +#define MAX98926_VBATOVFL_CLR_WIDTH 1 +#define MAX98926_IMONOVFL_CLR_MASK (1<<1) +#define MAX98926_IMONOVFL_CLR_SHIFT 1 +#define MAX98926_IMONOVFL_CLR_WIDTH 1 +#define MAX98926_VMONOVFL_CLR_MASK (1<<0) +#define MAX98926_VMONOVFL_CLR_SHIFT 0 +#define MAX98926_VMONOVFL_CLR_WIDTH 1 + +/* MAX98926_R011_MAP0 */ +#define MAX98926_ER_THERMWARN_EN_MASK (1<<7) +#define MAX98926_ER_THERMWARN_EN_SHIFT 7 +#define MAX98926_ER_THERMWARN_EN_WIDTH 1 +#define MAX98926_ER_THERMWARN_MAP_MASK (0x07<<4) +#define MAX98926_ER_THERMWARN_MAP_SHIFT 4 +#define MAX98926_ER_THERMWARN_MAP_WIDTH 3 + +/* MAX98926_R012_MAP1 */ +#define MAX98926_ER_ALCMUT_EN_MASK (1<<7) +#define MAX98926_ER_ALCMUT_EN_SHIFT 7 +#define MAX98926_ER_ALCMUT_EN_WIDTH 1 +#define MAX98926_ER_ALCMUT_MAP_MASK (0x07<<4) +#define MAX98926_ER_ALCMUT_MAP_SHIFT 4 +#define MAX98926_ER_ALCMUT_MAP_WIDTH 3 +#define MAX98926_ER_ALCP_EN_MASK (1<<3) +#define MAX98926_ER_ALCP_EN_SHIFT 3 +#define MAX98926_ER_ALCP_EN_WIDTH 1 +#define MAX98926_ER_ALCP_MAP_MASK (0x07<<0) +#define MAX98926_ER_ALCP_MAP_SHIFT 0 +#define MAX98926_ER_ALCP_MAP_WIDTH 3 + +/* MAX98926_R013_MAP2 */ +#define MAX98926_ER_ALCINFH_EN_MASK (1<<7) +#define MAX98926_ER_ALCINFH_EN_SHIFT 7 +#define MAX98926_ER_ALCINFH_EN_WIDTH 1 +#define MAX98926_ER_ALCINFH_MAP_MASK (0x07<<4) +#define MAX98926_ER_ALCINFH_MAP_SHIFT 4 +#define MAX98926_ER_ALCINFH_MAP_WIDTH 3 +#define MAX98926_ER_ALCACT_EN_MASK (1<<3) +#define MAX98926_ER_ALCACT_EN_SHIFT 3 +#define MAX98926_ER_ALCACT_EN_WIDTH 1 +#define MAX98926_ER_ALCACT_MAP_MASK (0x07<<0) +#define MAX98926_ER_ALCACT_MAP_SHIFT 0 +#define MAX98926_ER_ALCACT_MAP_WIDTH 3 + +/* MAX98926_R014_MAP3 */ +#define MAX98926_ER_SPKCURNT_EN_MASK (1<<7) +#define MAX98926_ER_SPKCURNT_EN_SHIFT 7 +#define MAX98926_ER_SPKCURNT_EN_WIDTH 1 +#define MAX98926_ER_SPKCURNT_MAP_MASK (0x07<<4) +#define MAX98926_ER_SPKCURNT_MAP_SHIFT 4 +#define MAX98926_ER_SPKCURNT_MAP_WIDTH 3 + +/* MAX98926_R015_MAP4 */ +/* RESERVED */ + +/* MAX98926_R016_MAP5 */ +#define MAX98926_ER_IMONOVFL_EN_MASK (1<<7) +#define MAX98926_ER_IMONOVFL_EN_SHIFT 7 +#define MAX98926_ER_IMONOVFL_EN_WIDTH 1 +#define MAX98926_ER_IMONOVFL_MAP_MASK (0x07<<4) +#define MAX98926_ER_IMONOVFL_MAP_SHIFT 4 +#define MAX98926_ER_IMONOVFL_MAP_WIDTH 3 +#define MAX98926_ER_VMONOVFL_EN_MASK (1<<3) +#define MAX98926_ER_VMONOVFL_EN_SHIFT 3 +#define MAX98926_ER_VMONOVFL_EN_WIDTH 1 +#define MAX98926_ER_VMONOVFL_MAP_MASK (0x07<<0) +#define MAX98926_ER_VMONOVFL_MAP_SHIFT 0 +#define MAX98926_ER_VMONOVFL_MAP_WIDTH 3 + +/* MAX98926_R017_MAP6 */ +#define MAX98926_ER_VBSTOVFL_EN_MASK (1<<7) +#define MAX98926_ER_VBSTOVFL_EN_SHIFT 7 +#define MAX98926_ER_VBSTOVFL_EN_WIDTH 1 +#define MAX98926_ER_VBSTOVFL_MAP_MASK (0x07<<4) +#define MAX98926_ER_VBSTOVFL_MAP_SHIFT 4 +#define MAX98926_ER_VBSTOVFL_MAP_WIDTH 3 +#define MAX98926_ER_VBATOVFL_EN_MASK (1<<3) +#define MAX98926_ER_VBATOVFL_EN_SHIFT 3 +#define MAX98926_ER_VBATOVFL_EN_WIDTH 1 +#define MAX98926_ER_VBATOVFL_MAP_MASK (0x07<<0) +#define MAX98926_ER_VBATOVFL_MAP_SHIFT 0 +#define MAX98926_ER_VBATOVFL_MAP_WIDTH 3 + +/* MAX98926_R018_MAP7 */ +#define MAX98926_ER_INVALSLOT_EN_MASK (1<<7) +#define MAX98926_ER_INVALSLOT_EN_SHIFT 7 +#define MAX98926_ER_INVALSLOT_EN_WIDTH 1 +#define MAX98926_ER_INVALSLOT_MAP_MASK (0x07<<4) +#define MAX98926_ER_INVALSLOT_MAP_SHIFT 4 +#define MAX98926_ER_INVALSLOT_MAP_WIDTH 3 +#define MAX98926_ER_SLOTCNFLT_EN_MASK (1<<3) +#define MAX98926_ER_SLOTCNFLT_EN_SHIFT 3 +#define MAX98926_ER_SLOTCNFLT_EN_WIDTH 1 +#define MAX98926_ER_SLOTCNFLT_MAP_MASK (0x07<<0) +#define MAX98926_ER_SLOTCNFLT_MAP_SHIFT 0 +#define MAX98926_ER_SLOTCNFLT_MAP_WIDTH 3 + +/* MAX98926_R019_MAP8 */ +#define MAX98926_ER_SLOTOVRN_EN_MASK (1<<3) +#define MAX98926_ER_SLOTOVRN_EN_SHIFT 3 +#define MAX98926_ER_SLOTOVRN_EN_WIDTH 1 +#define MAX98926_ER_SLOTOVRN_MAP_MASK (0x07<<0) +#define MAX98926_ER_SLOTOVRN_MAP_SHIFT 0 +#define MAX98926_ER_SLOTOVRN_MAP_WIDTH 3 + +/* MAX98926_R01A_DAI_CLK_MODE1 */ +#define MAX98926_DAI_CLK_SOURCE_MASK (1<<6) +#define MAX98926_DAI_CLK_SOURCE_SHIFT 6 +#define MAX98926_DAI_CLK_SOURCE_WIDTH 1 +#define MAX98926_MDLL_MULT_MASK (0x0F<<0) +#define MAX98926_MDLL_MULT_SHIFT 0 +#define MAX98926_MDLL_MULT_WIDTH 4 + +#define MAX98926_MDLL_MULT_MCLKx8 6 +#define MAX98926_MDLL_MULT_MCLKx16 8 + +/* MAX98926_R01B_DAI_CLK_MODE2 */ +#define MAX98926_DAI_SR_MASK (0x0F<<4) +#define MAX98926_DAI_SR_SHIFT 4 +#define MAX98926_DAI_SR_WIDTH 4 +#define MAX98926_DAI_MAS_MASK (1<<3) +#define MAX98926_DAI_MAS_SHIFT 3 +#define MAX98926_DAI_MAS_WIDTH 1 +#define MAX98926_DAI_BSEL_MASK (0x07<<0) +#define MAX98926_DAI_BSEL_SHIFT 0 +#define MAX98926_DAI_BSEL_WIDTH 3 + +#define MAX98926_DAI_BSEL_32 (0 << MAX98926_DAI_BSEL_SHIFT) +#define MAX98926_DAI_BSEL_48 (1 << MAX98926_DAI_BSEL_SHIFT) +#define MAX98926_DAI_BSEL_64 (2 << MAX98926_DAI_BSEL_SHIFT) +#define MAX98926_DAI_BSEL_256 (6 << MAX98926_DAI_BSEL_SHIFT) + +/* MAX98926_R01C_DAI_CLK_DIV_M_MSBS */ +#define MAX98926_DAI_M_MSBS_MASK (0xFF<<0) +#define MAX98926_DAI_M_MSBS_SHIFT 0 +#define MAX98926_DAI_M_MSBS_WIDTH 8 + +/* MAX98926_R01D_DAI_CLK_DIV_M_LSBS */ +#define MAX98926_DAI_M_LSBS_MASK (0xFF<<0) +#define MAX98926_DAI_M_LSBS_SHIFT 0 +#define MAX98926_DAI_M_LSBS_WIDTH 8 + +/* MAX98926_R01E_DAI_CLK_DIV_N_MSBS */ +#define MAX98926_DAI_N_MSBS_MASK (0x7F<<0) +#define MAX98926_DAI_N_MSBS_SHIFT 0 +#define MAX98926_DAI_N_MSBS_WIDTH 7 + +/* MAX98926_R01F_DAI_CLK_DIV_N_LSBS */ +#define MAX98926_DAI_N_LSBS_MASK (0xFF<<0) +#define MAX98926_DAI_N_LSBS_SHIFT 0 +#define MAX98926_DAI_N_LSBS_WIDTH 8 + +/* MAX98926_R020_FORMAT */ +#define MAX98926_DAI_CHANSZ_MASK (0x03<<6) +#define MAX98926_DAI_CHANSZ_SHIFT 6 +#define MAX98926_DAI_CHANSZ_WIDTH 2 +#define MAX98926_DAI_INTERLEAVE_MASK (1<<5) +#define MAX98926_DAI_INTERLEAVE_SHIFT 5 +#define MAX98926_DAI_INTERLEAVE_WIDTH 1 +#define MAX98926_DAI_EXTBCLK_HIZ_MASK (1<<4) +#define MAX98926_DAI_EXTBCLK_HIZ_SHIFT 4 +#define MAX98926_DAI_EXTBCLK_HIZ_WIDTH 1 +#define MAX98926_DAI_WCI_MASK (1<<3) +#define MAX98926_DAI_WCI_SHIFT 3 +#define MAX98926_DAI_WCI_WIDTH 1 +#define MAX98926_DAI_BCI_MASK (1<<2) +#define MAX98926_DAI_BCI_SHIFT 2 +#define MAX98926_DAI_BCI_WIDTH 1 +#define MAX98926_DAI_DLY_MASK (1<<1) +#define MAX98926_DAI_DLY_SHIFT 1 +#define MAX98926_DAI_DLY_WIDTH 1 +#define MAX98926_DAI_TDM_MASK (1<<0) +#define MAX98926_DAI_TDM_SHIFT 0 +#define MAX98926_DAI_TDM_WIDTH 1 + +#define MAX98926_DAI_CHANSZ_16 (1 << MAX98926_DAI_CHANSZ_SHIFT) +#define MAX98926_DAI_CHANSZ_24 (2 << MAX98926_DAI_CHANSZ_SHIFT) +#define MAX98926_DAI_CHANSZ_32 (3 << MAX98926_DAI_CHANSZ_SHIFT) + +/* MAX98926_R021_TDM_SLOT_SELECT */ +#define MAX98926_DAI_DO_EN_MASK (1<<7) +#define MAX98926_DAI_DO_EN_SHIFT 7 +#define MAX98926_DAI_DO_EN_WIDTH 1 +#define MAX98926_DAI_DIN_EN_MASK (1<<6) +#define MAX98926_DAI_DIN_EN_SHIFT 6 +#define MAX98926_DAI_DIN_EN_WIDTH 1 +#define MAX98926_DAI_INR_SOURCE_MASK (0x07<<3) +#define MAX98926_DAI_INR_SOURCE_SHIFT 3 +#define MAX98926_DAI_INR_SOURCE_WIDTH 3 +#define MAX98926_DAI_INL_SOURCE_MASK (0x07<<0) +#define MAX98926_DAI_INL_SOURCE_SHIFT 0 +#define MAX98926_DAI_INL_SOURCE_WIDTH 3 + +/* MAX98926_R022_DOUT_CFG_VMON */ +#define MAX98926_DAI_VMON_EN_MASK (1<<5) +#define MAX98926_DAI_VMON_EN_SHIFT 5 +#define MAX98926_DAI_VMON_EN_WIDTH 1 +#define MAX98926_DAI_VMON_SLOT_MASK (0x1F<<0) +#define MAX98926_DAI_VMON_SLOT_SHIFT 0 +#define MAX98926_DAI_VMON_SLOT_WIDTH 5 + +#define MAX98926_DAI_VMON_SLOT_00_01 (0 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_01_02 (1 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_02_03 (2 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_03_04 (3 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_04_05 (4 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_05_06 (5 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_06_07 (6 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_07_08 (7 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_08_09 (8 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_09_0A (9 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0A_0B (10 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0B_0C (11 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0C_0D (12 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0D_0E (13 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0E_0F (14 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0F_10 (15 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_10_11 (16 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_11_12 (17 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_12_13 (18 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_13_14 (19 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_14_15 (20 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_15_16 (21 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_16_17 (22 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_17_18 (23 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_18_19 (24 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_19_1A (25 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_1A_1B (26 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_1B_1C (27 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_1C_1D (28 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_1D_1E (29 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_1E_1F (30 << MAX98926_DAI_VMON_SLOT_SHIFT) + +/* MAX98926_R023_DOUT_CFG_IMON */ +#define MAX98926_DAI_IMON_EN_MASK (1<<5) +#define MAX98926_DAI_IMON_EN_SHIFT 5 +#define MAX98926_DAI_IMON_EN_WIDTH 1 +#define MAX98926_DAI_IMON_SLOT_MASK (0x1F<<0) +#define MAX98926_DAI_IMON_SLOT_SHIFT 0 +#define MAX98926_DAI_IMON_SLOT_WIDTH 5 + +#define MAX98926_DAI_IMON_SLOT_00_01 (0 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_01_02 (1 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_02_03 (2 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_03_04 (3 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_04_05 (4 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_05_06 (5 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_06_07 (6 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_07_08 (7 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_08_09 (8 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_09_0A (9 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0A_0B (10 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0B_0C (11 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0C_0D (12 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0D_0E (13 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0E_0F (14 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0F_10 (15 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_10_11 (16 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_11_12 (17 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_12_13 (18 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_13_14 (19 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_14_15 (20 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_15_16 (21 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_16_17 (22 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_17_18 (23 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_18_19 (24 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_19_1A (25 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_1A_1B (26 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_1B_1C (27 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_1C_1D (28 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_1D_1E (29 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_1E_1F (30 << MAX98926_DAI_IMON_SLOT_SHIFT) + +/* MAX98926_R024_DOUT_CFG_VBAT */ +#define MAX98926_DAI_INTERLEAVE_SLOT_MASK (0x1F<<0) +#define MAX98926_DAI_INTERLEAVE_SLOT_SHIFT 0 +#define MAX98926_DAI_INTERLEAVE_SLOT_WIDTH 5 + +/* MAX98926_R025_DOUT_CFG_VBST */ +#define MAX98926_DAI_VBST_EN_MASK (1<<5) +#define MAX98926_DAI_VBST_EN_SHIFT 5 +#define MAX98926_DAI_VBST_EN_WIDTH 1 +#define MAX98926_DAI_VBST_SLOT_MASK (0x1F<<0) +#define MAX98926_DAI_VBST_SLOT_SHIFT 0 +#define MAX98926_DAI_VBST_SLOT_WIDTH 5 + +/* MAX98926_R026_DOUT_CFG_FLAG */ +#define MAX98926_DAI_FLAG_EN_MASK (1<<5) +#define MAX98926_DAI_FLAG_EN_SHIFT 5 +#define MAX98926_DAI_FLAG_EN_WIDTH 1 +#define MAX98926_DAI_FLAG_SLOT_MASK (0x1F<<0) +#define MAX98926_DAI_FLAG_SLOT_SHIFT 0 +#define MAX98926_DAI_FLAG_SLOT_WIDTH 5 + +/* MAX98926_R027_DOUT_HIZ_CFG1 */ +#define MAX98926_DAI_SLOT_HIZ_CFG1_MASK (0xFF<<0) +#define MAX98926_DAI_SLOT_HIZ_CFG1_SHIFT 0 +#define MAX98926_DAI_SLOT_HIZ_CFG1_WIDTH 8 + +/* MAX98926_R028_DOUT_HIZ_CFG2 */ +#define MAX98926_DAI_SLOT_HIZ_CFG2_MASK (0xFF<<0) +#define MAX98926_DAI_SLOT_HIZ_CFG2_SHIFT 0 +#define MAX98926_DAI_SLOT_HIZ_CFG2_WIDTH 8 + +/* MAX98926_R029_DOUT_HIZ_CFG3 */ +#define MAX98926_DAI_SLOT_HIZ_CFG3_MASK (0xFF<<0) +#define MAX98926_DAI_SLOT_HIZ_CFG3_SHIFT 0 +#define MAX98926_DAI_SLOT_HIZ_CFG3_WIDTH 8 + +/* MAX98926_R02A_DOUT_HIZ_CFG4 */ +#define MAX98926_DAI_SLOT_HIZ_CFG4_MASK (0xFF<<0) +#define MAX98926_DAI_SLOT_HIZ_CFG4_SHIFT 0 +#define MAX98926_DAI_SLOT_HIZ_CFG4_WIDTH 8 + +/* MAX98926_R02B_DOUT_DRV_STRENGTH */ +#define MAX98926_DAI_OUT_DRIVE_MASK (0x03<<0) +#define MAX98926_DAI_OUT_DRIVE_SHIFT 0 +#define MAX98926_DAI_OUT_DRIVE_WIDTH 2 + +/* MAX98926_R02C_FILTERS */ +#define MAX98926_ADC_DITHER_EN_MASK (1<<7) +#define MAX98926_ADC_DITHER_EN_SHIFT 7 +#define MAX98926_ADC_DITHER_EN_WIDTH 1 +#define MAX98926_IV_DCB_EN_MASK (1<<6) +#define MAX98926_IV_DCB_EN_SHIFT 6 +#define MAX98926_IV_DCB_EN_WIDTH 1 +#define MAX98926_DAC_DITHER_EN_MASK (1<<4) +#define MAX98926_DAC_DITHER_EN_SHIFT 4 +#define MAX98926_DAC_DITHER_EN_WIDTH 1 +#define MAX98926_DAC_FILTER_MODE_MASK (1<<3) +#define MAX98926_DAC_FILTER_MODE_SHIFT 3 +#define MAX98926_DAC_FILTER_MODE_WIDTH 1 +#define MAX98926_DAC_HPF_MASK (0x07<<0) +#define MAX98926_DAC_HPF_SHIFT 0 +#define MAX98926_DAC_HPF_WIDTH 3 +#define MAX98926_DAC_HPF_DISABLE (0 << MAX98926_DAC_HPF_SHIFT) +#define MAX98926_DAC_HPF_DC_BLOCK (1 << MAX98926_DAC_HPF_SHIFT) +#define MAX98926_DAC_HPF_EN_100 (2 << MAX98926_DAC_HPF_SHIFT) +#define MAX98926_DAC_HPF_EN_200 (3 << MAX98926_DAC_HPF_SHIFT) +#define MAX98926_DAC_HPF_EN_400 (4 << MAX98926_DAC_HPF_SHIFT) +#define MAX98926_DAC_HPF_EN_800 (5 << MAX98926_DAC_HPF_SHIFT) + +/* MAX98926_R02D_GAIN */ +#define MAX98926_DAC_IN_SEL_MASK (0x03<<5) +#define MAX98926_DAC_IN_SEL_SHIFT 5 +#define MAX98926_DAC_IN_SEL_WIDTH 2 +#define MAX98926_SPK_GAIN_MASK (0x1F<<0) +#define MAX98926_SPK_GAIN_SHIFT 0 +#define MAX98926_SPK_GAIN_WIDTH 5 + +#define MAX98926_DAC_IN_SEL_LEFT_DAI (0 << MAX98926_DAC_IN_SEL_SHIFT) +#define MAX98926_DAC_IN_SEL_RIGHT_DAI (1 << MAX98926_DAC_IN_SEL_SHIFT) +#define MAX98926_DAC_IN_SEL_SUMMED_DAI (2 << MAX98926_DAC_IN_SEL_SHIFT) +#define MAX98926_DAC_IN_SEL_DIV2_SUMMED_DAI (3 << MAX98926_DAC_IN_SEL_SHIFT) + +/* MAX98926_R02E_GAIN_RAMPING */ +#define MAX98926_SPK_RMP_EN_MASK (1<<1) +#define MAX98926_SPK_RMP_EN_SHIFT 1 +#define MAX98926_SPK_RMP_EN_WIDTH 1 +#define MAX98926_SPK_ZCD_EN_MASK (1<<0) +#define MAX98926_SPK_ZCD_EN_SHIFT 0 +#define MAX98926_SPK_ZCD_EN_WIDTH 1 + +/* MAX98926_R02F_SPK_AMP */ +#define MAX98926_SPK_MODE_MASK (1<<0) +#define MAX98926_SPK_MODE_SHIFT 0 +#define MAX98926_SPK_MODE_WIDTH 1 +#define MAX98926_INSELECT_MODE_MASK (1<<1) +#define MAX98926_INSELECT_MODE_SHIFT 1 +#define MAX98926_INSELECT_MODE_WIDTH 1 + +/* MAX98926_R030_THRESHOLD */ +#define MAX98926_ALC_EN_MASK (1<<5) +#define MAX98926_ALC_EN_SHIFT 5 +#define MAX98926_ALC_EN_WIDTH 1 +#define MAX98926_ALC_TH_MASK (0x1F<<0) +#define MAX98926_ALC_TH_SHIFT 0 +#define MAX98926_ALC_TH_WIDTH 5 + +/* MAX98926_R031_ALC_ATTACK */ +#define MAX98926_ALC_ATK_STEP_MASK (0x0F<<4) +#define MAX98926_ALC_ATK_STEP_SHIFT 4 +#define MAX98926_ALC_ATK_STEP_WIDTH 4 +#define MAX98926_ALC_ATK_RATE_MASK (0x7<<0) +#define MAX98926_ALC_ATK_RATE_SHIFT 0 +#define MAX98926_ALC_ATK_RATE_WIDTH 3 + +/* MAX98926_R032_ALC_ATTEN_RLS */ +#define MAX98926_ALC_MAX_ATTEN_MASK (0x0F<<4) +#define MAX98926_ALC_MAX_ATTEN_SHIFT 4 +#define MAX98926_ALC_MAX_ATTEN_WIDTH 4 +#define MAX98926_ALC_RLS_RATE_MASK (0x7<<0) +#define MAX98926_ALC_RLS_RATE_SHIFT 0 +#define MAX98926_ALC_RLS_RATE_WIDTH 3 + +/* MAX98926_R033_ALC_HOLD_RLS */ +#define MAX98926_ALC_RLS_TGR_MASK (1<<0) +#define MAX98926_ALC_RLS_TGR_SHIFT 0 +#define MAX98926_ALC_RLS_TGR_WIDTH 1 + +/* MAX98926_R034_ALC_CONFIGURATION */ +#define MAX98926_ALC_MUTE_EN_MASK (1<<7) +#define MAX98926_ALC_MUTE_EN_SHIFT 7 +#define MAX98926_ALC_MUTE_EN_WIDTH 1 +#define MAX98926_ALC_MUTE_DLY_MASK (0x07<<4) +#define MAX98926_ALC_MUTE_DLY_SHIFT 4 +#define MAX98926_ALC_MUTE_DLY_WIDTH 3 +#define MAX98926_ALC_RLS_DBT_MASK (0x07<<0) +#define MAX98926_ALC_RLS_DBT_SHIFT 0 +#define MAX98926_ALC_RLS_DBT_WIDTH 3 + +/* MAX98926_R035_BOOST_CONVERTER */ +#define MAX98926_BST_SYNC_MASK (1<<7) +#define MAX98926_BST_SYNC_SHIFT 7 +#define MAX98926_BST_SYNC_WIDTH 1 +#define MAX98926_BST_PHASE_MASK (0x03<<4) +#define MAX98926_BST_PHASE_SHIFT 4 +#define MAX98926_BST_PHASE_WIDTH 2 +#define MAX98926_BST_SKIP_MODE_MASK (0x03<<0) +#define MAX98926_BST_SKIP_MODE_SHIFT 0 +#define MAX98926_BST_SKIP_MODE_WIDTH 2 + +/* MAX98926_R036_BLOCK_ENABLE */ +#define MAX98926_BST_EN_MASK (1<<7) +#define MAX98926_BST_EN_SHIFT 7 +#define MAX98926_BST_EN_WIDTH 1 +#define MAX98926_WATCH_EN_MASK (1<<6) +#define MAX98926_WATCH_EN_SHIFT 6 +#define MAX98926_WATCH_EN_WIDTH 1 +#define MAX98926_CLKMON_EN_MASK (1<<5) +#define MAX98926_CLKMON_EN_SHIFT 5 +#define MAX98926_CLKMON_EN_WIDTH 1 +#define MAX98926_SPK_EN_MASK (1<<4) +#define MAX98926_SPK_EN_SHIFT 4 +#define MAX98926_SPK_EN_WIDTH 1 +#define MAX98926_ADC_VBST_EN_MASK (1<<3) +#define MAX98926_ADC_VBST_EN_SHIFT 3 +#define MAX98926_ADC_VBST_EN_WIDTH 1 +#define MAX98926_ADC_VBAT_EN_MASK (1<<2) +#define MAX98926_ADC_VBAT_EN_SHIFT 2 +#define MAX98926_ADC_VBAT_EN_WIDTH 1 +#define MAX98926_ADC_IMON_EN_MASK (1<<1) +#define MAX98926_ADC_IMON_EN_SHIFT 1 +#define MAX98926_ADC_IMON_EN_WIDTH 1 +#define MAX98926_ADC_VMON_EN_MASK (1<<0) +#define MAX98926_ADC_VMON_EN_SHIFT 0 +#define MAX98926_ADC_VMON_EN_WIDTH 1 + +/* MAX98926_R037_CONFIGURATION */ +#define MAX98926_BST_VOUT_MASK (0x0F<<4) +#define MAX98926_BST_VOUT_SHIFT 4 +#define MAX98926_BST_VOUT_WIDTH 4 +#define MAX98926_THERMWARN_LEVEL_MASK (0x03<<2) +#define MAX98926_THERMWARN_LEVEL_SHIFT 2 +#define MAX98926_THERMWARN_LEVEL_WIDTH 2 +#define MAX98926_WATCH_TIME_MASK (0x03<<0) +#define MAX98926_WATCH_TIME_SHIFT 0 +#define MAX98926_WATCH_TIME_WIDTH 2 + +/* MAX98926_R038_GLOBAL_ENABLE */ +#define MAX98926_EN_MASK (1<<7) +#define MAX98926_EN_SHIFT 7 +#define MAX98926_EN_WIDTH 1 + +/* MAX98926_R03A_BOOST_LIMITER */ +#define MAX98926_BST_ILIM_MASK (0x1F<<3) +#define MAX98926_BST_ILIM_SHIFT 3 +#define MAX98926_BST_ILIM_WIDTH 5 + +/* MAX98926_R0FF_VERSION */ +#define MAX98926_REV_ID_MASK (0xFF<<0) +#define MAX98926_REV_ID_SHIFT 0 +#define MAX98926_REV_ID_WIDTH 8 + +struct max98926_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + unsigned int sysclk; + unsigned int v_slot; + unsigned int i_slot; + unsigned int ch_size; + unsigned int interleave_mode; +}; +#endif -- 1.9.3

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[alsa-devel] [PATCH V2 0/2] Add support for Pistachio internal DAC
by Damien Horsley 08 Dec '15

08 Dec '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add binding document and driver for Pistachio internal DAC Damien.Horsley (2): ASoC: img: Add binding document for Pistachio internal DAC ASoC: img: Add driver for Pistachio internal DAC .../bindings/sound/img,pistachio-internal-dac.txt | 18 ++ sound/soc/img/Kconfig | 8 + sound/soc/img/Makefile | 2 + sound/soc/img/pistachio-internal-dac.c | 316 +++++++++++++++++++++ 4 files changed, 344 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt create mode 100644 sound/soc/img/pistachio-internal-dac.c -- 2.1.4

2 3

[alsa-devel] [PATCH 1/2] ALSA: hda - Remove unused snd_hda_get_nid_path()
by Takashi Iwai 08 Dec '15

08 Dec '15

An exported helper function snd_hda_get_nid_path() is nowhere used. Let's remove it. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 16 ---------------- sound/pci/hda/hda_generic.h | 2 -- 2 files changed, 18 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index c6e8a651cea1..a644fc3302f0 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -279,22 +279,6 @@ static struct nid_path *get_nid_path(struct hda_codec *codec, } /** - * snd_hda_get_nid_path - get the path between the given NIDs - * @codec: the HDA codec - * @from_nid: the NID where the path start from - * @to_nid: the NID where the path ends at - * - * Return the found nid_path object or NULL for error. - * Passing 0 to either @from_nid or @to_nid behaves as a wildcard. - */ -struct nid_path *snd_hda_get_nid_path(struct hda_codec *codec, - hda_nid_t from_nid, hda_nid_t to_nid) -{ - return get_nid_path(codec, from_nid, to_nid, 0); -} -EXPORT_SYMBOL_GPL(snd_hda_get_nid_path); - -/** * snd_hda_get_path_idx - get the index number corresponding to the path * instance * @codec: the HDA codec diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 56e4139b9032..692510e59365 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -306,8 +306,6 @@ int snd_hda_gen_spec_init(struct hda_gen_spec *spec); int snd_hda_gen_init(struct hda_codec *codec); void snd_hda_gen_free(struct hda_codec *codec); -struct nid_path *snd_hda_get_nid_path(struct hda_codec *codec, - hda_nid_t from_nid, hda_nid_t to_nid); int snd_hda_get_path_idx(struct hda_codec *codec, struct nid_path *path); struct nid_path *snd_hda_get_path_from_idx(struct hda_codec *codec, int idx); bool snd_hda_parse_nid_path(struct hda_codec *codec, hda_nid_t from_nid, -- 2.6.3

1 1

[alsa-devel] [PATCH v2] ALSA: hda - Fixing speaker noise on the two latest thinkpad models
by Hui Wang 08 Dec '15

08 Dec '15

We have two latest thinkpad laptop models which are all based on the Intel skylake platforms, and all of them have the codec alc293 on them. When the machines boot to the desktop, an greeting dialogue shows up with the notification sound. But on these two models, there is noise with the notification sound. We have 3 SKUs for each of the models, all of them have this problem. So far, this problem is only specific to these two thinkpad models, we did not find this problem on the old thinkpad models with the codec alc293 or alc292. A workaround for this problem is disabling the aamix. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1523517 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ebc5362..49b1d36 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4596,6 +4596,7 @@ enum { ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, + ALC293_FIXUP_LENOVO_SPK_NOISE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5187,6 +5188,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC293_FIXUP_LENOVO_SPK_NOISE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5334,6 +5341,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -5343,6 +5351,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), 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), -- 1.9.1

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[alsa-devel] [PATCH V2 1/1] audio: Fix playback noise with 24/32 bit sample size on BXT
by han.lu＠intel.com 08 Dec '15

08 Dec '15

From: "Lu, Han" <han.lu(a)intel.com> In BXT-P A0, HD-Audio DMA requests is later than expected, and makes an audio stream sensitive to system latencies when 24/32 bits are playing. Adjusting threshold of DMA fifo to force the DMA request sooner to improve latency tolerance at the expense of power. v2: move Intel specific code to hda_intel.c Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 2ae8812..94dc6a9 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -93,6 +93,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_HSW_EM4 0x100c #define AZX_REG_HSW_EM5 0x1010 +/* Skylake/Broxton display HD-A controller Extended Mode registers */ +#define AZX_REG_SKL_EM4L 0x1040 + /* PCI space */ #define AZX_PCIREG_TCSEL 0x44 diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 963f824..bff5c8b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -355,6 +355,8 @@ enum { ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c)) +#define IS_BROXTON(pci) ((pci)->device == 0x5a98) + static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", [AZX_DRIVER_PCH] = "HDA Intel PCH", @@ -506,15 +508,36 @@ static void azx_init_pci(struct azx *chip) } } +/* + * In BXT-P A0, HD-Audio DMA requests is later than expected, + * and makes an audio stream sensitive to system latencies when + * 24/32 bits are playing. + * Adjusting threshold of DMA fifo to force the DMA request + * sooner to improve latency tolerance at the expense of power. + */ +static void bxt_reduce_dma_latency(struct azx *chip) +{ + u32 val; + + val = azx_readl(chip, SKL_EM4L); + val &= (0x3 << 20); + azx_writel(chip, SKL_EM4L, val); +} + static void hda_intel_init_chip(struct azx *chip, bool full_reset) { struct hdac_bus *bus = azx_bus(chip); + struct pci_dev *pci = chip->pci; if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) snd_hdac_set_codec_wakeup(bus, true); azx_init_chip(chip, full_reset); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) snd_hdac_set_codec_wakeup(bus, false); + + /* reduce dma latency to avoid noise */ + if (IS_BROXTON(pci)) + bxt_reduce_dma_latency(chip); } /* calculate runtime delay from LPIB */ -- 2.5.0

3 4

[alsa-devel] [PATCH] ALSA: hda - Fixing speaker noise on two thinkpad machines
by Hui Wang 08 Dec '15

08 Dec '15

Disable the aamix can fix this problem. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1523517 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ebc5362..49b1d36 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4596,6 +4596,7 @@ enum { ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, + ALC293_FIXUP_LENOVO_SPK_NOISE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5187,6 +5188,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC293_FIXUP_LENOVO_SPK_NOISE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5334,6 +5341,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -5343,6 +5351,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), 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), -- 1.9.1

2 6

[alsa-devel] [PATCH] [RFC] ARM: shmobile: bockw dts: Override #sound-dai-cells to zero
by Geert Uytterhoeven 08 Dec '15

08 Dec '15

The default value of #sound-dai-cells in r8a7778.dtsi is one, while the /sound/simple-audio-card,cpu device node in r8a7778-bockw.dts uses a phandle without any extra cells ("<&rcar_sound>"), causing: /sound/simple-audio-card,cpu: arguments longer than property asoc-simple-card sound: parse error -22 asoc-simple-card: probe of sound failed with error -22 Override #sound-dai-cells to zero to fix this. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- The rcar_sound node is still disabled. So why is this needed?!? --- arch/arm/boot/dts/r8a7778-bockw.dts | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/arch/arm/boot/dts/r8a7778-bockw.dts b/arch/arm/boot/dts/r8a7778-bockw.dts index 3c03e6b8261e9348..158ad94fd4d187a7 100644 --- a/arch/arm/boot/dts/r8a7778-bockw.dts +++ b/arch/arm/boot/dts/r8a7778-bockw.dts @@ -172,6 +172,11 @@ }; }; +&rcar_sound { + /* Single DAI */ + #sound-dai-cells = <0>; +}; + &sdhi0 { pinctrl-0 = <&sdhi0_pins>, <&sdhi0_pup_pins>; pinctrl-names = "default"; -- 1.9.1

2 1

[alsa-devel] [PATCH v2 MC Next Gen] sound/usb: Fix out of bounds access in media_entity_init()
by Shuah Khan 07 Dec '15

07 Dec '15

Fix the out of bounds access in media_entity_init() found by KASan. This is a result of media_mixer_init() failing to allocate memory for all 3 of its pads before calling media_entity_init(). Fix it to allocate memory for the right struct media_mixer_ctl instead of struct media_ctl. Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> --- Changes since v1: Change to address review comment from Takashi Iwai This patch fixes the mixer patch below: https://patchwork.linuxtv.org/patch/31827/ sound/usb/media.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/media.c b/sound/usb/media.c index bebe27b..b0d283f 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -233,8 +233,8 @@ int media_mixer_init(struct snd_usb_audio *chip) if (mixer->media_mixer_ctl) continue; - /* allocate media_ctl */ - mctl = kzalloc(sizeof(struct media_ctl), GFP_KERNEL); + /* allocate media_mixer_ctl */ + mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); if (!mctl) return -ENOMEM; -- 2.5.0

1 0

[alsa-devel] Sound UsbPre : minimal patch for support
by shal＠free.fr 07 Dec '15

07 Dec '15

Hello, I have a Sound UsbPre (the first not the UsbPRe2) sound card. There is no support for this card in alsa. I do a usb packet sniffing on Windows for detect problem. Some control quirk should be added at boot in order to use the card. It works to play and record in 44100 khz. It works only with 44100khz (for example, if you play a 88200 file, it will be played too slow). This patch is not well tested, work on my desktop : Linux Ubuntu with 4.2.19-lowlatency kernel. The button emulation is not done (by a control packet, you can change the card configuration, like phantom power activation). If somebody is interested, I can provide needed information. If Alsa's developer is interested, I can provide support for test patches or develop a second version of this patch in order to integrate the support for this card in the linux kernel Regards, Olivier linux-source-4.2.0/sound/usb$ diff -rub quirks.c.ORIG quirks.c --- quirks.c.ORIG 2015-11-24 22:43:16.878635155 +0100 +++ quirks.c 2015-11-26 20:56:09.492126460 +0100 @@ -601,6 +601,40 @@ return 0; } +/* Add some control messages needed by Sound Devices UsbPre. + * Values here are chosen based on sniffing USB traffic + * under Windows.*/ +static int snd_usb_usbpre_boot_quirk(struct usb_device *dev) +{ + u8 buf = 1; + u8 data1[1] = {0x81}; + u8 data2[1] = {0xc1}; + u8 data3[3] = {0x44,0xac,0x00}; + + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 1, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x00f0, 496, data1, 1); + snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x81 , + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x00f0, 0, &buf, 1); + if (buf == 0) { + return -ENODEV; + } + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 1, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x00f0, 496, data2, 1); + snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x81 , + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x00f0, 0, &buf, 1); + if (buf == 0) { + return -ENODEV; + } + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 1, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_ENDPOINT, + 0x0100, 133, data3, 3); + return 0; +} + static int snd_usb_fasttrackpro_boot_quirk(struct usb_device *dev) { int err; @@ -985,6 +1019,10 @@ /* SB Audigy 2 NX needs its own boot-up magic, too */ return snd_usb_audigy2nx_boot_quirk(dev); + case USB_ID(0x0926, 0x0100): + /* Sound Devices UsbPre needs its own boot-up magic, too */ + return snd_usb_usbpre_boot_quirk(dev); + case USB_ID(0x10f5, 0x0200): /* C-Media CM106 / Turtle Beach Audio Advantage Roadie */ return snd_usb_cm106_boot_quirk(dev);

2 1

[alsa-devel] [PATCH MC Next Gen] sound/usb: Fix out of bounds access in media_entity_init()
by Shuah Khan 07 Dec '15

07 Dec '15

Fix the out of bounds access in media_entity_init() found by KASan. This is a result of media_mixer_init() failing to allocate memory for all 3 of its pads before calling media_entity_init(). Fix it to allocate memory for the right struct media_mixer_ctl instead of struct media_ctl. Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> --- This patch fixes the mixer patch below: https://patchwork.linuxtv.org/patch/31827/ sound/usb/media.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/media.c b/sound/usb/media.c index bebe27b..0cb44b9 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -233,8 +233,8 @@ int media_mixer_init(struct snd_usb_audio *chip) if (mixer->media_mixer_ctl) continue; - /* allocate media_ctl */ - mctl = kzalloc(sizeof(struct media_ctl), GFP_KERNEL); + /* allocate media_mixer_ctl */ + mctl = kzalloc(sizeof(struct media_mixer_ctl), GFP_KERNEL); if (!mctl) return -ENOMEM; @@ -244,6 +244,7 @@ int media_mixer_init(struct snd_usb_audio *chip) mctl->media_pad[0].flags = MEDIA_PAD_FL_SINK; mctl->media_pad[1].flags = MEDIA_PAD_FL_SOURCE; mctl->media_pad[2].flags = MEDIA_PAD_FL_SOURCE; + media_entity_init(&mctl->media_entity, MEDIA_MIXER_PAD_MAX, mctl->media_pad); ret = media_device_register_entity(mctl->media_dev, -- 2.5.0

2 2

[alsa-devel] [GIT PULL] AC'97 gpiochip
by Mark Brown 07 Dec '15

07 Dec '15

The following changes since commit 8005c49d9aea74d382f474ce11afbbc7d7130bec: Linux 4.4-rc1 (2015-11-15 17:00:27 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-ac97-gpio for you to fetch changes up to 9bf5c3d11f1fbaf43399d189f05fb20ceb46ee5d: ASoC: ac97: add gpio chip (2015-11-18 18:08:54 +0000) ---------------------------------------------------------------- ASoC: Add a GPIO chip for AC'97 GPIOs are part of the AC'97 spec, enable their use on embedded platforms using AC'97. ---------------------------------------------------------------- Robert Jarzmik (1): ASoC: ac97: add gpio chip include/sound/ac97_codec.h | 3 ++ sound/soc/soc-ac97.c | 125 +++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 128 insertions(+)

2 1

[alsa-devel] [PATCH] ALSA: hda - Add inverted dmic for Packard Bell DOTS
by David Henningsson 07 Dec '15

07 Dec '15

On the internal mic of the Packard Bell DOTS, one channel has an inverted signal. Add a quirk to fix this up. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1523232 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9bedf7c..ebc5362 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6409,6 +6409,7 @@ static const struct hda_fixup alc662_fixups[] = { static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x1025, 0x022f, "Acer Aspire One", ALC662_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1025, 0x0241, "Packard Bell DOTS", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC), -- 1.9.1

2 1

[alsa-devel] [PATCH v3] ALSA: compress: add support for 32bit calls in a 64bit kernel
by Ravindra Lokhande 07 Dec '15

07 Dec '15

Compress offload does not support ioctl calls from a 32bit userspace in a 64 bit kernel. This patch adds support for ioctls from a 32bit userspace in a 64bit kernel Signed-off-by: Ravindra Lokhande <rlokhande(a)nvidia.com> --- v1 -> v2: use compat_ptr to convert 32 bit address to 64 bit pointer v2 -> v3: add header file compat.h sound/core/compress_offload.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index b123c42..4597820 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -38,6 +38,7 @@ #include <linux/uio.h> #include <linux/uaccess.h> #include <linux/module.h> +#include <linux/compat.h> #include <sound/core.h> #include <sound/initval.h> #include <sound/compress_params.h> @@ -847,6 +848,15 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) return retval; } +/* support of 32bit userspace on 64bit platforms */ +#ifdef CONFIG_COMPAT +static long snd_compr_ioctl_compat(struct file *file, unsigned int cmd, + unsigned long arg) +{ + return snd_compr_ioctl(file, cmd, (unsigned long)compat_ptr(arg)); +} +#endif + static const struct file_operations snd_compr_file_ops = { .owner = THIS_MODULE, .open = snd_compr_open, @@ -854,6 +864,9 @@ static const struct file_operations snd_compr_file_ops = { .write = snd_compr_write, .read = snd_compr_read, .unlocked_ioctl = snd_compr_ioctl, +#ifdef CONFIG_COMPAT + .compat_ioctl = snd_compr_ioctl_compat, +#endif .mmap = snd_compr_mmap, .poll = snd_compr_poll, }; -- 2.1.4

3 3

[alsa-devel] [PATCH 1/1] audio: Fix playback noise with 24/32 bit sample size on BXT
by han.lu＠intel.com 07 Dec '15

07 Dec '15

From: "Lu, Han" <han.lu(a)intel.com> In BXT-P A0, HD-Audio DMA requests is later than expected, and makes an audio stream sensitive to system latencies when 24/32 bits are playing. Adjusting threshold of DMA fifo to force the DMA request sooner to improve latency tolerance at the expense of power. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 2ae8812..94dc6a9 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -93,6 +93,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_HSW_EM4 0x100c #define AZX_REG_HSW_EM5 0x1010 +/* Skylake/Broxton display HD-A controller Extended Mode registers */ +#define AZX_REG_SKL_EM4L 0x1040 + /* PCI space */ #define AZX_PCIREG_TCSEL 0x44 diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 22dbfa5..6c2aa4c 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -27,6 +27,7 @@ #include <linux/module.h> #include <linux/pm_runtime.h> #include <linux/slab.h> +#include <linux/pci.h> #include <sound/core.h> #include <sound/initval.h> #include "hda_controller.h" @@ -34,6 +35,8 @@ #define CREATE_TRACE_POINTS #include "hda_controller_trace.h" +#define is_broxton(pci) ((pci)->device == 0x5a98) + /* DSP lock helpers */ #define dsp_lock(dev) snd_hdac_dsp_lock(azx_stream(dev)) #define dsp_unlock(dev) snd_hdac_dsp_unlock(azx_stream(dev)) @@ -880,14 +883,35 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_load_dsp_cleanup); #endif /* CONFIG_SND_HDA_DSP_LOADER */ /* + * In BXT-P A0, HD-Audio DMA requests is later than expected, + * and makes an audio stream sensitive to system latencies when + * 24/32 bits are playing. + * Adjusting threshold of DMA fifo to force the DMA request + * sooner to improve latency tolerance at the expense of power. + */ +static void bxt_reduce_dma_latency(struct azx *chip) +{ + u32 val; + + val = azx_readl(chip, SKL_EM4L); + val &= (0x3 << 20); + azx_writel(chip, SKL_EM4L, val); +} + +/* * reset and start the controller registers */ void azx_init_chip(struct azx *chip, bool full_reset) { + struct pci_dev *pci = chip->pci; + if (snd_hdac_bus_init_chip(azx_bus(chip), full_reset)) { /* correct RINTCNT for CXT */ if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) azx_writew(chip, RINTCNT, 0xc0); + /* reduce dma latency to avoid noise */ + if (is_broxton(pci)) + bxt_reduce_dma_latency(chip); } } EXPORT_SYMBOL_GPL(azx_init_chip); -- 2.5.0

3 2

[alsa-devel] [PATCH] crec: Always read whole number of fragments when length limit specified
by Richard Fitzgerald 07 Dec '15

07 Dec '15

This patch removes the lines that adjusted the read size to exactly read to the end of the calculated length in bytes. Doing this could lead to invalid fragment sizes being requested which may then cause a read error on some codecs. Instead we keep reading whole fragments until we have >= length bytes of data captured. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- src/utils/crec.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/src/utils/crec.c b/src/utils/crec.c index 258fade..e61e620 100644 --- a/src/utils/crec.c +++ b/src/utils/crec.c @@ -332,9 +332,6 @@ void capture_samples(char *name, unsigned int card, unsigned int device, fprintf(finfo, "%s: Capturing audio NOW!!!\n", __func__); do { - if (length && size > length - total_read) - size = length - total_read; - read = compress_read(compress, buffer, size); if (read < 0) { fprintf(stderr, "Error reading sample\n"); -- 1.9.1

2 1

[alsa-devel] M-audio 410 : complex procedure to get a sound
by ROUGE ETIENNE 06 Dec '15

06 Dec '15

Hello ! I've just bought a M-audio firewire 410 and I'm very thankfull to takashi's work that allows us to use the device almost like an USB one ! I can have sound and have access to the routing controls. However 2 bugs makes it a bit tedious : NOTE : I use an up to date freshly installed ubuntustudio 15.10. The only modifications I made was 1) deleting pulse 2) select the soundcard as default in asoundrc 1) I do not have sound out of the box, but I can manage to have sound by starting ffado-mixer then force restart alsa. After that everything works fine.... but I gotta do it at every restart which isn't nice 2) ffado-mixer suffers several bugs : - if I close the window I can't restart it again unless I kill ffado-dbus-server (it says : "another instance of ffado-mixer is already running). - I got few errors on startup "invalid samplerate for input"... don't know it it has any impact - when I tweak too much ffado-mixer (AFAIK, it's random, but happens within few minuts) the mouse shows a 'hand closed' cursor and I can't click anything. The system remains fully functionnal, but I can't close anything nor click... If I start an emergency terminal I can kill whatever I want, but the problem remains. I'll try with a fedora this evening to see if it's not related with ubuntustudio, but I'd be glad to have some help with those 2 issues (especially the stalling ffado-mixer) thanks ! Etienne

2 2

[alsa-devel] [RFC][PATCH 00/17 v2] ALSA: oxfw: refactoring and merging scs1x module
by Takashi Sakamoto 06 Dec '15

06 Dec '15

Hi, This patchset update my previous post for merging ALSA scs1x module to oxfw module: [alsa-devel] [RFC][PATCH 0/9] ALSA: oxfw: merge scs1x module http://mailman.alsa-project.org/pipermail/alsa-devel/2015-November/100534.h… This patchset consists of three parts: * 01-05: code refactoring for OXFW module * 06-09: code refactoring for old firewire-speaker functionalities in OXFW module * 10-17: merge scs1x module Changes: * split some patches for a fine granularity * drop a small framework I proposed in previous patch * rename functions for AV/C Audio Subunit command After this patchset is applied, OXFW module is loaded instead of scs1x module. Then, some names except for rawmidi ID are changed, see patch 16. Currently, I find no advantage to keep them what they were. I'm welcome to receive your oppositions about it. Takashi Sakamoto (17): ALSA: oxfw: rename a file for control elements so that it's for model-specific ALSA: oxfw: rename local functions for control elements so that they represent as local ALSA: oxfw: change function prototype for AV/C Audio Subunit command ALSA: oxfw: reuse driver entry to detect quirk ALSA: oxfw: gather model-dependent conditions to a function ALSA: oxfw: add memory allocation for model-specific structure ALSA: oxfw: move model-specific members from common structure ALSA: oxfw: move model-specific parameters from common structure ALSA: oxfw: rename a structure so that it means backward compatibility to old drivers ALSA: oxfw: add scs1x layer ALSA: oxfw: allocate own address region for SCS.1 series ALSA: oxfw: copy handlers of asynchronous transaction for MIDI capture ALSA: oxfw: add MIDI capture port for SCS.1 models ALSA: oxfw: copy handlers of asynchronous transaction for MIDI playback ALSA: oxfw: add MIDI playback port for SCS.1 models ALSA: oxfw: obsolete scs1x module ALSA: oxfw: add stream format quirk for SCS.1 models sound/firewire/Kconfig | 12 +- sound/firewire/Makefile | 2 - sound/firewire/oxfw/Makefile | 4 +- sound/firewire/oxfw/oxfw-scs1x.c | 409 +++++++++++++++++++++ .../firewire/oxfw/{oxfw-control.c => oxfw-spkr.c} | 142 ++++--- sound/firewire/oxfw/oxfw.c | 110 ++++-- sound/firewire/oxfw/oxfw.h | 23 +- 7 files changed, 586 insertions(+), 116 deletions(-) create mode 100644 sound/firewire/oxfw/oxfw-scs1x.c rename sound/firewire/oxfw/{oxfw-control.c => oxfw-spkr.c} (55%) -- 2.5.0

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Re: [alsa-devel] [sound:for-linus 29/29] sound/pci/rme96.c:1035:1-7: preceding lock on line 991
by Julia Lawall 04 Dec '15

04 Dec '15

Looks suspicious. Please check. julia On Sat, 5 Dec 2015, kbuild test robot wrote: > CC: kbuild-all(a)01.org > CC: alsa-devel(a)alsa-project.org > TO: Takashi Iwai <tiwai(a)suse.de> > > tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus > head: ec20c9c7a63f7b8676eb574406a8bc8f32b7ba7e > commit: ec20c9c7a63f7b8676eb574406a8bc8f32b7ba7e [29/29] ALSA: rme96: Fix unexpected volume reset after rate changes > :::::: branch date: 2 hours ago > :::::: commit date: 2 hours ago > > >> sound/pci/rme96.c:1035:1-7: preceding lock on line 991 > > git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git > git remote update sound > git checkout ec20c9c7a63f7b8676eb574406a8bc8f32b7ba7e > vim +1035 sound/pci/rme96.c > > ^1da177e Linus Torvalds 2005-04-16 985 > 4d23359b Clemens Ladisch 2005-09-05 986 runtime->dma_area = (void __force *)(rme96->iobase + > 4d23359b Clemens Ladisch 2005-09-05 987 RME96_IO_PLAY_BUFFER); > ^1da177e Linus Torvalds 2005-04-16 988 runtime->dma_addr = rme96->port + RME96_IO_PLAY_BUFFER; > ^1da177e Linus Torvalds 2005-04-16 989 runtime->dma_bytes = RME96_BUFFER_SIZE; > ^1da177e Linus Torvalds 2005-04-16 990 > ^1da177e Linus Torvalds 2005-04-16 @991 spin_lock_irq(&rme96->lock); > ^1da177e Linus Torvalds 2005-04-16 992 if (!(rme96->wcreg & RME96_WCR_MASTER) && > ^1da177e Linus Torvalds 2005-04-16 993 snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG && > ^1da177e Linus Torvalds 2005-04-16 994 (rate = snd_rme96_capture_getrate(rme96, &dummy)) > 0) > ^1da177e Linus Torvalds 2005-04-16 995 { > ^1da177e Linus Torvalds 2005-04-16 996 /* slave clock */ > ^1da177e Linus Torvalds 2005-04-16 997 if ((int)params_rate(params) != rate) { > ec20c9c7 Takashi Iwai 2015-12-04 998 err = -EIO; > ec20c9c7 Takashi Iwai 2015-12-04 999 goto error; > ^1da177e Linus Torvalds 2005-04-16 1000 } > ec20c9c7 Takashi Iwai 2015-12-04 1001 } else { > ec20c9c7 Takashi Iwai 2015-12-04 1002 err = snd_rme96_playback_setrate(rme96, params_rate(params)); > ec20c9c7 Takashi Iwai 2015-12-04 1003 if (err < 0) > ec20c9c7 Takashi Iwai 2015-12-04 1004 goto error; > ec20c9c7 Takashi Iwai 2015-12-04 1005 apply_dac_volume = err > 0; /* need to restore volume later? */ > ^1da177e Linus Torvalds 2005-04-16 1006 } > ec20c9c7 Takashi Iwai 2015-12-04 1007 if ((err = snd_rme96_playback_setformat(rme96, params_format(params))) < 0) > ec20c9c7 Takashi Iwai 2015-12-04 1008 goto error; > ^1da177e Linus Torvalds 2005-04-16 1009 snd_rme96_setframelog(rme96, params_channels(params), 1); > ^1da177e Linus Torvalds 2005-04-16 1010 if (rme96->capture_periodsize != 0) { > ^1da177e Linus Torvalds 2005-04-16 1011 if (params_period_size(params) << rme96->playback_frlog != > ^1da177e Linus Torvalds 2005-04-16 1012 rme96->capture_periodsize) > ^1da177e Linus Torvalds 2005-04-16 1013 { > ec20c9c7 Takashi Iwai 2015-12-04 1014 err = -EBUSY; > ec20c9c7 Takashi Iwai 2015-12-04 1015 goto error; > ^1da177e Linus Torvalds 2005-04-16 1016 } > ^1da177e Linus Torvalds 2005-04-16 1017 } > ^1da177e Linus Torvalds 2005-04-16 1018 rme96->playback_periodsize = > ^1da177e Linus Torvalds 2005-04-16 1019 params_period_size(params) << rme96->playback_frlog; > ^1da177e Linus Torvalds 2005-04-16 1020 snd_rme96_set_period_properties(rme96, rme96->playback_periodsize); > ^1da177e Linus Torvalds 2005-04-16 1021 /* S/PDIF setup */ > ^1da177e Linus Torvalds 2005-04-16 1022 if ((rme96->wcreg & RME96_WCR_ADAT) == 0) { > ^1da177e Linus Torvalds 2005-04-16 1023 rme96->wcreg &= ~(RME96_WCR_PRO | RME96_WCR_DOLBY | RME96_WCR_EMP); > ^1da177e Linus Torvalds 2005-04-16 1024 writel(rme96->wcreg |= rme96->wcreg_spdif_stream, rme96->iobase + RME96_IO_CONTROL_REGISTER); > ^1da177e Linus Torvalds 2005-04-16 1025 } > ^1da177e Linus Torvalds 2005-04-16 1026 spin_unlock_irq(&rme96->lock); > ec20c9c7 Takashi Iwai 2015-12-04 1027 err = 0; > ^1da177e Linus Torvalds 2005-04-16 1028 > ec20c9c7 Takashi Iwai 2015-12-04 1029 error: > ec20c9c7 Takashi Iwai 2015-12-04 1030 if (apply_dac_volume) { > ec20c9c7 Takashi Iwai 2015-12-04 1031 usleep_range(3000, 10000); > ec20c9c7 Takashi Iwai 2015-12-04 1032 snd_rme96_apply_dac_volume(rme96); > ec20c9c7 Takashi Iwai 2015-12-04 1033 } > ec20c9c7 Takashi Iwai 2015-12-04 1034 > ec20c9c7 Takashi Iwai 2015-12-04 @1035 return err; > ^1da177e Linus Torvalds 2005-04-16 1036 } > ^1da177e Linus Torvalds 2005-04-16 1037 > ^1da177e Linus Torvalds 2005-04-16 1038 static int > > --- > 0-DAY kernel test infrastructure Open Source Technology Center > https://lists.01.org/pipermail/kbuild-all Intel Corporation >

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Re: [alsa-devel] [bug] Volume at maximum when track with different frequency is played with my RME sound card
by Maeda 04 Dec '15

04 Dec '15

Hi ! Someone's here ? Thanks. Le 23/10/2015 18:30, Maeda a écrit : > Hi all ! > > I don't know if the bug is known from your team, but some people > advice me to send an email to the Alsa devel list if nobody answer in > the kernel's bugtracking. > > Here is the bug I have : > https://bugzilla.kernel.org/show_bug.cgi?id=105771 > > What do you think about it ? > > If I was wrong sending this mail to this list, I'm sorry :) > > Thanks. > Have a nice day.

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[alsa-devel] [PATCH v2 0/9] Add get_eld audio component for i915/HD-audio
by Takashi Iwai 04 Dec '15

04 Dec '15

Hi, this is a revised patchset to add get_eld op to audio component for communicating more directly between i915 and HD-audio. Currently, the HDMI/DP audio status and ELD are notified and obtained via the hardware-level communication over HD-audio unsolicited event and verbs although the graphics driver holds the exactly same information. As we already have a notification via audio component, this is another step forward; the audio driver fetches directly the audio status and ELD via the new component op. The patch still doesn't follow the recent kernel doc comment changes in i915_component.h. I'll rebase once when I get the steady branch in drm tree. The current patchset is found in sound git tree test/hdmi-jack branch. Takashi === v1->v2: * Use modeset lock for get_eld lock, drop av mutex * Return the expected size from get_eld, not the copied size * Add reverse map from a port number to the encoder * Deferred invocation of get_eld from the eld_notify in HDA side * A bit more code refactoring in HDA side * Move audio component accessors to hdac_i915.c Takashi Iwai (9): drm/i915: Remove superfluous NULL check drm/i915: Add get_eld audio component drm/i915: refactoring audio component functions drm/i915: Add reverse mapping between port and intel_encoder ALSA: hda - Split ELD update code from hdmi_present_sense() ALSA: hda - Use component ops for i915 HDMI/DP audio jack handling ALSA: hda - Do zero-clear in snd_hdmi_parse_eld() itself ALSA: hda - Skip ELD notification during PM process ALSA: hda - Move audio component accesses to hdac_i915.c drivers/gpu/drm/i915/i915_drv.h | 2 + drivers/gpu/drm/i915/intel_audio.c | 59 ++++++--- drivers/gpu/drm/i915/intel_ddi.c | 1 + drivers/gpu/drm/i915/intel_dp.c | 1 + drivers/gpu/drm/i915/intel_drv.h | 1 + drivers/gpu/drm/i915/intel_hdmi.c | 2 + include/drm/i915_component.h | 6 + include/sound/hda_i915.h | 14 +++ sound/hda/hdac_i915.c | 66 +++++++++++ sound/pci/hda/hda_eld.c | 1 + sound/pci/hda/patch_hdmi.c | 237 +++++++++++++++++++++++++------------ 11 files changed, 294 insertions(+), 96 deletions(-) -- 2.6.3

3 34

[alsa-devel] [PATCH v2] ALSA: compress: add support for 32bit calls in a 64bit kernel
by Ravindra Lokhande 04 Dec '15

04 Dec '15

Compress offload does not support ioctl calls from a 32bit userspace in a 64 bit kernel. This patch adds support for ioctls from a 32bit userspace in a 64bit kernel Signed-off-by: Ravindra Lokhande <rlokhande(a)nvidia.com> --- v1 -> v2: use compat_ptr to convert 32 bit address to 64 bit pointer sound/core/compress_offload.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index b123c42..4c156a7 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -847,6 +847,15 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) return retval; } +/* support of 32bit userspace on 64bit platforms */ +#ifdef CONFIG_COMPAT +static long snd_compr_ioctl_compat(struct file *file, unsigned int cmd, + unsigned long arg) +{ + return snd_compr_ioctl(file, cmd, (unsigned long)compat_ptr(arg)); +} +#endif + static const struct file_operations snd_compr_file_ops = { .owner = THIS_MODULE, .open = snd_compr_open, @@ -854,6 +863,9 @@ static const struct file_operations snd_compr_file_ops = { .write = snd_compr_write, .read = snd_compr_read, .unlocked_ioctl = snd_compr_ioctl, +#ifdef CONFIG_COMPAT + .compat_ioctl = snd_compr_ioctl_compat, +#endif .mmap = snd_compr_mmap, .poll = snd_compr_poll, }; -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: hda - Enable audio component for old Intel PCH devices
by Takashi Iwai 04 Dec '15

04 Dec '15

As i915 graphics driver provides the notification via audio component, not only the currently implemented HSW+ and VLV+ platforms but also all other PCH-based platforms (e.g. Cougar Point, Panther Point, etc) can use this infrastructure. It'll improve the reliability and the power consumption significantly, especially once when we implement the ELD notification via component. As a preliminary, this patch enables the usage of audio component for all PCH platforms. The HDA controller just needs to set AZX_DCAPS_I915_POWERWELL flag appropriately. The name of the flag is a bit confusing, but this actually works even on the chips without the powerwell but accesses only the other component ops. In the HDMI/DP codec driver side, we just need to register/unregister the notifier for such chips. This can be identified by checking the audio_component field in the assigned hdac_bus. One caveat is that PCH for Haswell and Broadwell must not be bound with i915 audio component, as there are dedicated HD-audio HDMI controllers on these platforms. Ditto for Poulsbo and Oaktrail as they use gma500 graphics, not i915. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 14 ++++++++++---- sound/pci/hda/patch_hdmi.c | 6 ++++-- 2 files changed, 14 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 963f82430938..ee0e316401f9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -284,13 +284,19 @@ enum { (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE) /* quirks for Intel PCH */ -#define AZX_DCAPS_INTEL_PCH_NOPM \ +#define AZX_DCAPS_INTEL_PCH_BASE \ (AZX_DCAPS_NO_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ AZX_DCAPS_REVERSE_ASSIGN | AZX_DCAPS_SNOOP_TYPE(SCH)) +/* PCH up to IVB; bound with i915 audio component for HDMI, no runtime PM */ +#define AZX_DCAPS_INTEL_PCH_NOPM \ + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_I915_POWERWELL) + +/* PCH for HSW/BDW; with runtime PM, but no i915 binding */ #define AZX_DCAPS_INTEL_PCH \ - (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME) + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME) +/* HSW HDMI */ #define AZX_DCAPS_INTEL_HASWELL \ (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_COUNT_LPIB_DELAY |\ AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ @@ -2146,10 +2152,10 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Poulsbo */ { PCI_DEVICE(0x8086, 0x811b), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_BASE }, /* Oaktrail */ { PCI_DEVICE(0x8086, 0x080a), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_BASE }, /* BayTrail */ { PCI_DEVICE(0x8086, 0x0f04), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BAYTRAIL }, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index a918377d3e9b..85342d261043 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -152,6 +152,8 @@ struct hdmi_spec { struct i915_audio_component_audio_ops i915_audio_ops; }; +#define codec_has_acomp(codec) \ + ((codec)->bus->core.audio_component != NULL) struct hdmi_audio_infoframe { u8 type; /* 0x84 */ @@ -2218,7 +2220,7 @@ static void generic_hdmi_free(struct hda_codec *codec) struct hdmi_spec *spec = codec->spec; int pin_idx; - if (is_haswell_plus(codec) || is_valleyview_plus(codec)) + if (codec_has_acomp(codec)) snd_hdac_i915_register_notifier(NULL); for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { @@ -2390,7 +2392,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) is_broxton(codec)) codec->core.link_power_control = 1; - if (is_haswell_plus(codec) || is_valleyview_plus(codec)) { + if (codec_has_acomp(codec)) { codec->depop_delay = 0; spec->i915_audio_ops.audio_ptr = codec; spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; -- 2.6.3

1 0

[alsa-devel] Clock Rate Clock Rate Adjustment API
by Chas Leichner 04 Dec '15

04 Dec '15

I would like to be able to make small adjustments the playback clock rate of a PCM device as it is playing in order to accurately synchronize music playback across several computers. Looking at the kernel source, this functionality has been previously implemented in both the aloop driver and ST's uniperif driver where the clock rate adjustment was exposed to the user space as a control element. Although the control interface can be used to communicate this information, the naming and properties of the clock adjustment controls don't appear to be standardized or documented and I don't see anything related to clock rate adjustment in Documentation/sound/alsa/ControlNames.txt. In order to make a standardized interface, do you think it makes more sense to standardize on a clock rate adjustment control or introduce a new interface other than the control interface because the control interface is only for mixer controls rather than arbitrary runtime changes to the driver? Regards, Chas

2 1

[alsa-devel] [PATCH 1/2] ASoC: arizona: Correct types of mixer texts and values
by Charles Keepax 03 Dec '15

03 Dec '15

The core expects "const char * const" and "unsigned int" for enum controls, various places in Arizona use "const char *" and "int". This patch corrects the type of these arrays. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 8 ++++---- sound/soc/codecs/arizona.h | 8 ++++---- 2 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a23f7d1..d2731cf 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -310,7 +310,7 @@ int arizona_init_gpio(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(arizona_init_gpio); -const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { +const char * const arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "None", "Tone Generator 1", "Tone Generator 2", @@ -418,7 +418,7 @@ const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { }; EXPORT_SYMBOL_GPL(arizona_mixer_texts); -int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { +unsigned int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { 0x00, /* None */ 0x04, /* Tone */ 0x05, @@ -555,12 +555,12 @@ const char *arizona_sample_rate_val_to_name(unsigned int rate_val) } EXPORT_SYMBOL_GPL(arizona_sample_rate_val_to_name); -const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE] = { +const char * const arizona_rate_text[ARIZONA_RATE_ENUM_SIZE] = { "SYNCCLK rate", "8kHz", "16kHz", "ASYNCCLK rate", }; EXPORT_SYMBOL_GPL(arizona_rate_text); -const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = { +const unsigned int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = { 0, 1, 2, 8, }; EXPORT_SYMBOL_GPL(arizona_rate_val); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 01a367c..b4f1867 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -96,8 +96,8 @@ struct arizona_priv { #define ARIZONA_NUM_MIXER_INPUTS 104 extern const unsigned int arizona_mixer_tlv[]; -extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; -extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; +extern const char * const arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; +extern unsigned int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; #define ARIZONA_GAINMUX_CONTROLS(name, base) \ SOC_SINGLE_RANGE_TLV(name " Input Volume", base + 1, \ @@ -216,8 +216,8 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; #define ARIZONA_RATE_ENUM_SIZE 4 #define ARIZONA_SAMPLE_RATE_ENUM_SIZE 14 -extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; -extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; +extern const char * const arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; +extern const unsigned int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; extern const char * const arizona_sample_rate_text[ARIZONA_SAMPLE_RATE_ENUM_SIZE]; extern const unsigned int arizona_sample_rate_val[ARIZONA_SAMPLE_RATE_ENUM_SIZE]; -- 2.1.4

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[alsa-devel] mx28: bad unlock balance detected
by Fabio Estevam 03 Dec '15

03 Dec '15

Hi, Running linux-next 20151126 on a mx28evk leads to the following bug: sgtl5000 0-000a: sgtl5000 revision 0x11 sgtl5000 0-000a: Using internal LDO instead of VDDD ===================================== [ BUG: bad unlock balance detected! ] 4.4.0-rc2-next-20151126-00001-g44eb6b2 #156 Not tainted ------------------------------------- kworker/u2:0/6 is trying to release lock (&rdev->mutex) at: [<c02e7858>] regulator_get_voltage+0x30/0x44 but there are no more locks to release! other info that might help us debug this: 5 locks held by kworker/u2:0/6: #0: ("%s""deferwq"){++++.+}, at: [<c002f1f8>] process_one_work+0x10c/0x49c #1: (deferred_probe_work){+.+.+.}, at: [<c002f1f8>] process_one_work+0x10c/0x49c #2: (&dev->mutex){......}, at: [<c03197bc>] __device_attach+0x20/0x108 #3: (client_mutex){+.+.+.}, at: [<c042dc34>] snd_soc_register_card+0x280/0x1108 #4: (&card->mutex){+.+.+.}, at: [<c042dc40>] snd_soc_register_card+0x28c/0x1108 stack backtrace: CPU: 0 PID: 6 Comm: kworker/u2:0 Not tainted 4.4.0-rc2-next-20151126-00001-g44eb6b2 #156 Hardware name: Freescale MXS (Device Tree) Workqueue: deferwq deferred_probe_work_func [<c000eea0>] (unwind_backtrace) from [<c000cfa0>] (show_stack+0x10/0x14) [<c000cfa0>] (show_stack) from [<c0049350>] (print_unlock_imbalance_bug+0xac/0xdc) [<c0049350>] (print_unlock_imbalance_bug) from [<c004c2f8>] (lock_release+0x2a4/0x38c) [<c004c2f8>] (lock_release) from [<c0513e70>] (__mutex_unlock_slowpath+0x7c/0x15c) [<c0513e70>] (__mutex_unlock_slowpath) from [<c02e7858>] (regulator_get_voltage+0x30/0x44) [<c02e7858>] (regulator_get_voltage) from [<c043c200>] (sgtl5000_probe+0x1c0/0x5f0) [<c043c200>] (sgtl5000_probe) from [<c0428a74>] (snd_soc_codec_drv_probe+0x14/0x18) [<c0428a74>] (snd_soc_codec_drv_probe) from [<c042cd54>] (soc_probe_component+0x200/0x330) [<c042cd54>] (soc_probe_component) from [<c042e178>] (snd_soc_register_card+0x7c4/0x1108) [<c042e178>] (snd_soc_register_card) from [<c0439cdc>] (devm_snd_soc_register_card+0x30/0x70) [<c0439cdc>] (devm_snd_soc_register_card) from [<c043d1cc>] (mxs_sgtl5000_probe+0xc8/0x138) [<c043d1cc>] (mxs_sgtl5000_probe) from [<c031b1fc>] (platform_drv_probe+0x50/0xac) [<c031b1fc>] (platform_drv_probe) from [<c0319ab0>] (driver_probe_device+0x1e4/0x2b0) [<c0319ab0>] (driver_probe_device) from [<c03180d8>] (bus_for_each_drv+0x44/0x8c) [<c03180d8>] (bus_for_each_drv) from [<c0319840>] (__device_attach+0xa4/0x108) [<c0319840>] (__device_attach) from [<c0318f84>] (bus_probe_device+0x88/0x90) [<c0318f84>] (bus_probe_device) from [<c0319370>] (deferred_probe_work_func+0x64/0x94) [<c0319370>] (deferred_probe_work_func) from [<c002f290>] (process_one_work+0x1a4/0x49c) [<c002f290>] (process_one_work) from [<c002f5e8>] (worker_thread+0x28/0x468) [<c002f5e8>] (worker_thread) from [<c0034dbc>] (kthread+0xc0/0xdc) [<c0034dbc>] (kthread) from [<c000a2ec>] (ret_from_fork+0x14/0x28) mxs-sgtl5000 sound: sgtl5000 <-> 80042000.saif mapping ok By inspecting the sgtl5000 codec driver I could not find where this "bad unlock balance" is happening. Any ideas? Thanks, Fabio Estevam

2 3

[alsa-devel] [PATCH v2] ASoC: da7218: Enable mic level detection reporting to user-space
by Adam Thomson 03 Dec '15

03 Dec '15

This patch adds support to the codec driver to handle mic level detect related IRQs, and report these to user-space using a uevent variable. The uevent variable string "EVENT=MIC_LEVEL_DETECT" is sent to user-space, if the mic level detect feature is enabled, and the audio captured at the chosen mic(s) is above a certain threshold. User-space can then handle the event accordingly (e.g. process audio capture stream). This method was chosen over ALSA control notification for a couple of reasons: 1) There's no requirement here for a control to read state from. The event is the only thing that's required and of interest. 2) tinyalsa support for control notifications does not exist so on platforms using this over alsa-lib there is a need to add code to support this event handling. Another possible option would be to use the standard Jack reporting framework but this really does not fit for this kind of event. Finally, use of the input device framework is not being encouraged, due to difficulties in enabling apps to access input devices, so this has also been avoided. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- Changes in v2: - More verbose and descriptive commit message added. sound/soc/codecs/da7218.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index 4fee7ae..752ed04 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -2202,6 +2202,16 @@ int da7218_hpldet(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(da7218_hpldet); +static void da7218_micldet_irq(struct snd_soc_codec *codec) +{ + char *envp[] = { + "EVENT=MIC_LEVEL_DETECT", + NULL, + }; + + kobject_uevent_env(&codec->dev->kobj, KOBJ_CHANGE, envp); +} + static void da7218_hpldet_irq(struct snd_soc_codec *codec) { struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); @@ -2232,6 +2242,10 @@ static irqreturn_t da7218_irq_thread(int irq, void *data) if (!status) return IRQ_NONE; + /* Mic level detect */ + if (status & DA7218_LVL_DET_EVENT_MASK) + da7218_micldet_irq(codec); + /* HP detect */ if (status & DA7218_HPLDET_JACK_EVENT_MASK) da7218_hpldet_irq(codec); @@ -2936,11 +2950,6 @@ static int da7218_probe(struct snd_soc_codec *codec) } if (da7218->irq) { - /* Mask off mic level events, currently not handled */ - snd_soc_update_bits(codec, DA7218_EVENT_MASK, - DA7218_LVL_DET_EVENT_MSK_MASK, - DA7218_LVL_DET_EVENT_MSK_MASK); - ret = devm_request_threaded_irq(codec->dev, da7218->irq, NULL, da7218_irq_thread, IRQF_TRIGGER_LOW | IRQF_ONESHOT, -- 1.9.3

1 0

[alsa-devel] [PATCH] ALSA: Fix compat_ioctl handling for OSS emulations
by Takashi Iwai 03 Dec '15

03 Dec '15

The ALSA PCM, mixer and sequencer OSS emulations provide the 32bit compatible ioctl, but they just call the 64bit native ioctl as is. Although this works in most cases, passing the argument value as-is isn't guaranteed to work on all architectures. We need to convert it via compat_ptr() instead. This patch addresses the missing conversions. Since all relevant ioctls in these functions take the argument as a pointer, we do the pointer conversion in each compat_ioctl and pass it as a 64bit value to the native ioctl. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/mixer_oss.c | 8 +++++++- sound/core/oss/pcm_oss.c | 7 ++++++- sound/core/seq/oss/seq_oss.c | 7 ++++++- 3 files changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 7a8c79dd9734..2ff9c12d664a 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -24,6 +24,7 @@ #include <linux/time.h> #include <linux/string.h> #include <linux/module.h> +#include <linux/compat.h> #include <sound/core.h> #include <sound/minors.h> #include <sound/control.h> @@ -397,7 +398,12 @@ int snd_mixer_oss_ioctl_card(struct snd_card *card, unsigned int cmd, unsigned l #ifdef CONFIG_COMPAT /* all compatible */ -#define snd_mixer_oss_ioctl_compat snd_mixer_oss_ioctl +static long snd_mixer_oss_ioctl_compat(struct file *file, unsigned int cmd, + unsigned long arg) +{ + return snd_mixer_oss_ioctl1(file->private_data, cmd, + (unsigned long)compat_ptr(arg)); +} #else #define snd_mixer_oss_ioctl_compat NULL #endif diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 58550cc93f28..e557dbe469f4 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -33,6 +33,7 @@ #include <linux/module.h> #include <linux/math64.h> #include <linux/string.h> +#include <linux/compat.h> #include <sound/core.h> #include <sound/minors.h> #include <sound/pcm.h> @@ -2648,7 +2649,11 @@ static long snd_pcm_oss_ioctl(struct file *file, unsigned int cmd, unsigned long #ifdef CONFIG_COMPAT /* all compatible */ -#define snd_pcm_oss_ioctl_compat snd_pcm_oss_ioctl +static long snd_pcm_oss_ioctl_compat(struct file *file, unsigned int cmd, + unsigned long arg) +{ + return snd_pcm_oss_ioctl(file, cmd, (unsigned long)compat_ptr(arg)); +} #else #define snd_pcm_oss_ioctl_compat NULL #endif diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index 7354b8bed860..8db156b207f1 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -23,6 +23,7 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/mutex.h> +#include <linux/compat.h> #include <sound/core.h> #include <sound/minors.h> #include <sound/initval.h> @@ -189,7 +190,11 @@ odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg) } #ifdef CONFIG_COMPAT -#define odev_ioctl_compat odev_ioctl +static long odev_ioctl_compat(struct file *file, unsigned int cmd, + unsigned long arg) +{ + return odev_ioctl(file, cmd, (unsigned long)compat_ptr(arg)); +} #else #define odev_ioctl_compat NULL #endif -- 2.6.3

1 0

[alsa-devel] [PATCH] ALSA: compress: add support for 32bit calls in a 64bit kernel
by Ravindra Lokhande 03 Dec '15

03 Dec '15

Compress offload does not support ioctl calls from a 32bit userspace in a 64 bit kernel. This patch adds support for ioctls from a 32bit userspace in a 64bit kernel Signed-off-by: Ravindra Lokhande <rlokhande(a)nvidia.com> --- sound/core/compress_offload.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index b123c42..98f6b47 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -847,6 +847,15 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) return retval; } +/* support of 32bit userspace on 64bit platforms */ +#ifdef CONFIG_COMPAT +static long snd_compr_ioctl_compat(struct file *file, unsigned int cmd, + unsigned long arg) +{ + return snd_compr_ioctl(file, cmd, arg); +} +#endif + static const struct file_operations snd_compr_file_ops = { .owner = THIS_MODULE, .open = snd_compr_open, @@ -854,6 +863,9 @@ static const struct file_operations snd_compr_file_ops = { .write = snd_compr_write, .read = snd_compr_read, .unlocked_ioctl = snd_compr_ioctl, +#ifdef CONFIG_COMPAT + .compat_ioctl = snd_compr_ioctl_compat, +#endif .mmap = snd_compr_mmap, .poll = snd_compr_poll, }; -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: hda - Enable power_save_node for CX20722
by David Henningsson 03 Dec '15

03 Dec '15

I've tested it on one device and it works fine, no clicks. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Alsa-info for this device is available here: https://launchpadlibrarian.net/228090406/AlsaInfo.txt Possibly it works well for other chips too, but this is the only one I have tested. Btw; I noticed another thing which I wonder whether it's correct or not: If I run "hda-emu AlsaInfo.txt", all nodes are in D3. But after running "P 0 p", some nodes (e g the internal speaker pin) stays in D0. Is that expected? I thought all pins would go down to D3 when playback closes. Same behaviour is seen on the real hardware, too. sound/pci/hda/patch_conexant.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ef19890..6122b8c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -901,6 +901,9 @@ static int patch_conexant_auto(struct hda_codec *codec) snd_hda_pick_fixup(codec, cxt5051_fixup_models, cxt5051_fixups, cxt_fixups); break; + case 0x14f150f2: + codec->power_save_node = 1; + /* Fall through */ default: codec->pin_amp_workaround = 1; snd_hda_pick_fixup(codec, cxt5066_fixup_models, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: da7218: Enable mic level detection reporting to user-space
by Adam Thomson 03 Dec '15

03 Dec '15

This patch adds support to the codec driver to handle mic level detect related IRQs, and report these to user-space using a uevent variable. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7218.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index 4fee7ae..752ed04 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -2202,6 +2202,16 @@ int da7218_hpldet(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(da7218_hpldet); +static void da7218_micldet_irq(struct snd_soc_codec *codec) +{ + char *envp[] = { + "EVENT=MIC_LEVEL_DETECT", + NULL, + }; + + kobject_uevent_env(&codec->dev->kobj, KOBJ_CHANGE, envp); +} + static void da7218_hpldet_irq(struct snd_soc_codec *codec) { struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); @@ -2232,6 +2242,10 @@ static irqreturn_t da7218_irq_thread(int irq, void *data) if (!status) return IRQ_NONE; + /* Mic level detect */ + if (status & DA7218_LVL_DET_EVENT_MASK) + da7218_micldet_irq(codec); + /* HP detect */ if (status & DA7218_HPLDET_JACK_EVENT_MASK) da7218_hpldet_irq(codec); @@ -2936,11 +2950,6 @@ static int da7218_probe(struct snd_soc_codec *codec) } if (da7218->irq) { - /* Mask off mic level events, currently not handled */ - snd_soc_update_bits(codec, DA7218_EVENT_MASK, - DA7218_LVL_DET_EVENT_MSK_MASK, - DA7218_LVL_DET_EVENT_MSK_MASK); - ret = devm_request_threaded_irq(codec->dev, da7218->irq, NULL, da7218_irq_thread, IRQF_TRIGGER_LOW | IRQF_ONESHOT, -- 1.9.3

3 4

[alsa-devel] [PATCH] hda-emu: Add snd_power_get_state
by David Henningsson 03 Dec '15

03 Dec '15

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/sound/core.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index e556765..1196319 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -113,6 +113,8 @@ int snd_card_register(struct snd_card *card) return 0; } +#define snd_power_get_state(card) ({ (void)(card); SNDRV_CTL_POWER_D0; }) + #include <linux/list.h> #include <linux/mutex.h> #include <linux/pci.h> -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Rename NAU88L25->NAU8825 to stay consistent with existing driver name
by Anatol Pomozov 03 Dec '15

03 Dec '15

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/soc/intel/Kconfig | 6 +- sound/soc/intel/boards/Makefile | 4 +- sound/soc/intel/boards/skl_nau8825_ssm4567.c | 366 ++++++++++++++++++++++++++ sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 366 -------------------------- sound/soc/intel/skylake/skl.c | 2 +- 5 files changed, 372 insertions(+), 372 deletions(-) create mode 100644 sound/soc/intel/boards/skl_nau8825_ssm4567.c delete mode 100644 sound/soc/intel/boards/skl_nau88l25_ssm4567.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2d3b124..7108e13 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -156,8 +156,8 @@ config SND_SOC_INTEL_SKL_RT286_MACH Say Y if you have such a device If unsure select "N". -config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH - tristate "ASoC Audio driver for SKL with NAU88L25 and SSM4567 in I2S Mode" +config SND_SOC_INTEL_SKL_NAU8825_SSM4567_MACH + tristate "ASoC Audio driver for SKL with NAU8825 and SSM4567 in I2S Mode" depends on X86_INTEL_LPSS && I2C select SND_SOC_INTEL_SST select SND_SOC_INTEL_SKYLAKE @@ -166,6 +166,6 @@ config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH select SND_SOC_DMIC help This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for NAU88L25 + SSM4567. + create an alsa sound card for NAU8825 + SSM4567. Say Y if you have such a device If unsure select "N". diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index a59f762..fb50457 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -7,7 +7,7 @@ snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o snd-soc-skl_rt286-objs := skl_rt286.o -snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o +snd-soc-skl_nau8825_ssm4567-objs := skl_nau8825_ssm4567.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -18,4 +18,4 @@ obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o -obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o +obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU8825_SSM4567_MACH) += snd-soc-skl_nau8825_ssm4567.o diff --git a/sound/soc/intel/boards/skl_nau8825_ssm4567.c b/sound/soc/intel/boards/skl_nau8825_ssm4567.c new file mode 100644 index 0000000..7c34927 --- /dev/null +++ b/sound/soc/intel/boards/skl_nau8825_ssm4567.c @@ -0,0 +1,366 @@ +/* + * Intel Skylake I2S Machine Driver for NAU8825+SSM4567 + * + * Copyright (C) 2015, Intel Corporation. All rights reserved. + * + * Modified from: + * Intel Skylake I2S Machine Driver for NAU8825 and SSM4567 + * + * Copyright (C) 2015, Intel Corporation. All rights reserved. + * + * 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. + * + * 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. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include "../../codecs/nau8825.h" + +#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" +#define SKL_SSM_CODEC_DAI "ssm4567-hifi" + +static struct snd_soc_jack skylake_headset; +static struct snd_soc_card skylake_audio_card; + +static inline struct snd_soc_dai *skl_get_codec_dai(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + + list_for_each_entry(rtd, &card->rtd_list, list) { + + if (!strncmp(rtd->codec_dai->name, SKL_NUVOTON_CODEC_DAI, + strlen(SKL_NUVOTON_CODEC_DAI))) + return rtd->codec_dai; + } + + return NULL; +} + +static const struct snd_kcontrol_new skylake_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Left Speaker"), + SOC_DAPM_PIN_SWITCH("Right Speaker"), +}; + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret; + + codec_dai = skl_get_codec_dai(card); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, + NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "set sysclk err = %d\n", ret); + return -EIO; + } + } else { + ret = snd_soc_dai_set_sysclk(codec_dai, + NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "set sysclk err = %d\n", ret); + return -EIO; + } + } + return ret; +} + +static const struct snd_soc_dapm_widget skylake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Speaker", NULL), + SND_SOC_DAPM_SPK("Right Speaker", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route skylake_map[] = { + /* HP jack connectors - unknown if we have jack detection */ + {"Headphone Jack", NULL, "HPOL"}, + {"Headphone Jack", NULL, "HPOR"}, + + /* speaker */ + {"Left Speaker", NULL, "Left OUT"}, + {"Right Speaker", NULL, "Right OUT"}, + + /* other jacks */ + {"MIC", NULL, "Headset Mic"}, + {"DMIC AIF", NULL, "SoC DMIC"}, + + /* CODEC BE connections */ + { "Left Playback", NULL, "ssp0 Tx"}, + { "Right Playback", NULL, "ssp0 Tx"}, + { "ssp0 Tx", NULL, "codec0_out"}, + + { "AIF1 Playback", NULL, "ssp1 Tx"}, + { "ssp1 Tx", NULL, "codec1_out"}, + + { "codec0_in", NULL, "ssp1 Rx" }, + { "ssp1 Rx", NULL, "AIF1 Capture" }, + + /* DMIC */ + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "Capture" }, + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, +}; + +static struct snd_soc_codec_conf ssm4567_codec_conf[] = { + { + .dev_name = "i2c-INT343B:00", + .name_prefix = "Left", + }, + { + .dev_name = "i2c-INT343B:01", + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link_component ssm4567_codec_components[] = { + { /* Left */ + .name = "i2c-INT343B:00", + .dai_name = SKL_SSM_CODEC_DAI, + }, + { /* Right */ + .name = "i2c-INT343B:01", + .dai_name = SKL_SSM_CODEC_DAI, + }, +}; + +static int skylake_ssm4567_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + + /* Slot 1 for left */ + ret = snd_soc_dai_set_tdm_slot(rtd->codec_dais[0], 0x01, 0x01, 2, 48); + if (ret < 0) + return ret; + + /* Slot 2 for right */ + ret = snd_soc_dai_set_tdm_slot(rtd->codec_dais[1], 0x02, 0x02, 2, 48); + if (ret < 0) + return ret; + + return ret; +} + +static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_soc_codec *codec = rtd->codec; + + /* + * 4 buttons here map to the google Reference headset + * The use of these buttons can be decided by the user space. + */ + ret = snd_soc_card_jack_new(&skylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, &skylake_headset, + NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); + return ret; + } + + nau8825_enable_jack_detect(codec, &skylake_headset); + + return ret; +} + +static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, + NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); + + if (ret < 0) + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + + return ret; +} + +static struct snd_soc_ops skylake_nau8825_ops = { + .hw_params = skylake_nau8825_hw_params, +}; + +/* skylake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link skylake_dais[] = { + /* Front End DAI links */ + { + .name = "Skl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Skl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + { + .name = "Skl Audio Reference cap", + .stream_name = "refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .ignore_suspend = 1, + .nonatomic = 1, + .dynamic = 1, + }, + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .be_id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codecs = ssm4567_codec_components, + .num_codecs = ARRAY_SIZE(ssm4567_codec_components), + .dai_fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .init = skylake_ssm4567_codec_init, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = skylake_ssp_fixup, + .dpcm_playback = 1, + }, + { + /* SSP1 - Codec */ + .name = "SSP1-Codec", + .be_id = 0, + .cpu_dai_name = "SSP1 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codec_name = "i2c-10508825:00", + .codec_dai_name = SKL_NUVOTON_CODEC_DAI, + .init = skylake_nau8825_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = skylake_ssp_fixup, + .ops = &skylake_nau8825_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .be_id = 1, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, +}; + +/* skylake audio machine driver for SPT + NAU8825 */ +static struct snd_soc_card skylake_audio_card = { + .name = "sklnau8825adi", + .owner = THIS_MODULE, + .dai_link = skylake_dais, + .num_links = ARRAY_SIZE(skylake_dais), + .controls = skylake_controls, + .num_controls = ARRAY_SIZE(skylake_controls), + .dapm_widgets = skylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(skylake_widgets), + .dapm_routes = skylake_map, + .num_dapm_routes = ARRAY_SIZE(skylake_map), + .codec_conf = ssm4567_codec_conf, + .num_configs = ARRAY_SIZE(ssm4567_codec_conf), +}; + +static int skylake_audio_probe(struct platform_device *pdev) +{ + skylake_audio_card.dev = &pdev->dev; + + return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card); +} + +static struct platform_driver skylake_audio = { + .probe = skylake_audio_probe, + .driver = { + .name = "skl_nau8825_ssm4567_i2s", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(skylake_audio) + +/* Module information */ +MODULE_AUTHOR("Conrad Cooke <conrad.cooke(a)intel.com>"); +MODULE_AUTHOR("Harsha Priya <harshapriya.n(a)intel.com>"); +MODULE_AUTHOR("Naveen M <naveen.m(a)intel.com>"); +MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); +MODULE_AUTHOR("Yong Zhi <yong.zhi(a)intel.com>"); +MODULE_DESCRIPTION("Intel Audio Machine driver for SKL with NAU8825 and SSM4567 in I2S Mode"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:skl_nau8825_ssm4567_i2s"); diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c deleted file mode 100644 index 65c65d4..0000000 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ /dev/null @@ -1,366 +0,0 @@ -/* - * Intel Skylake I2S Machine Driver for NAU88L25+SSM4567 - * - * Copyright (C) 2015, Intel Corporation. All rights reserved. - * - * Modified from: - * Intel Skylake I2S Machine Driver for NAU88L25 and SSM4567 - * - * Copyright (C) 2015, Intel Corporation. All rights reserved. - * - * 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. - * - * 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. - */ - -#include <linux/module.h> -#include <linux/platform_device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/jack.h> -#include <sound/pcm_params.h> -#include "../../codecs/nau8825.h" - -#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" -#define SKL_SSM_CODEC_DAI "ssm4567-hifi" - -static struct snd_soc_jack skylake_headset; -static struct snd_soc_card skylake_audio_card; - -static inline struct snd_soc_dai *skl_get_codec_dai(struct snd_soc_card *card) -{ - struct snd_soc_pcm_runtime *rtd; - - list_for_each_entry(rtd, &card->rtd_list, list) { - - if (!strncmp(rtd->codec_dai->name, SKL_NUVOTON_CODEC_DAI, - strlen(SKL_NUVOTON_CODEC_DAI))) - return rtd->codec_dai; - } - - return NULL; -} - -static const struct snd_kcontrol_new skylake_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Left Speaker"), - SOC_DAPM_PIN_SWITCH("Right Speaker"), -}; - -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - int ret; - - codec_dai = skl_get_codec_dai(card); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found\n"); - return -EIO; - } - - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } - return ret; -} - -static const struct snd_soc_dapm_widget skylake_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Left Speaker", NULL), - SND_SOC_DAPM_SPK("Right Speaker", NULL), - SND_SOC_DAPM_MIC("SoC DMIC", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), -}; - -static const struct snd_soc_dapm_route skylake_map[] = { - /* HP jack connectors - unknown if we have jack detection */ - {"Headphone Jack", NULL, "HPOL"}, - {"Headphone Jack", NULL, "HPOR"}, - - /* speaker */ - {"Left Speaker", NULL, "Left OUT"}, - {"Right Speaker", NULL, "Right OUT"}, - - /* other jacks */ - {"MIC", NULL, "Headset Mic"}, - {"DMIC AIF", NULL, "SoC DMIC"}, - - /* CODEC BE connections */ - { "Left Playback", NULL, "ssp0 Tx"}, - { "Right Playback", NULL, "ssp0 Tx"}, - { "ssp0 Tx", NULL, "codec0_out"}, - - { "AIF1 Playback", NULL, "ssp1 Tx"}, - { "ssp1 Tx", NULL, "codec1_out"}, - - { "codec0_in", NULL, "ssp1 Rx" }, - { "ssp1 Rx", NULL, "AIF1 Capture" }, - - /* DMIC */ - { "dmic01_hifi", NULL, "DMIC01 Rx" }, - { "DMIC01 Rx", NULL, "Capture" }, - { "Headphone Jack", NULL, "Platform Clock" }, - { "Headset Mic", NULL, "Platform Clock" }, -}; - -static struct snd_soc_codec_conf ssm4567_codec_conf[] = { - { - .dev_name = "i2c-INT343B:00", - .name_prefix = "Left", - }, - { - .dev_name = "i2c-INT343B:01", - .name_prefix = "Right", - }, -}; - -static struct snd_soc_dai_link_component ssm4567_codec_components[] = { - { /* Left */ - .name = "i2c-INT343B:00", - .dai_name = SKL_SSM_CODEC_DAI, - }, - { /* Right */ - .name = "i2c-INT343B:01", - .dai_name = SKL_SSM_CODEC_DAI, - }, -}; - -static int skylake_ssm4567_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - - /* Slot 1 for left */ - ret = snd_soc_dai_set_tdm_slot(rtd->codec_dais[0], 0x01, 0x01, 2, 48); - if (ret < 0) - return ret; - - /* Slot 2 for right */ - ret = snd_soc_dai_set_tdm_slot(rtd->codec_dais[1], 0x02, 0x02, 2, 48); - if (ret < 0) - return ret; - - return ret; -} - -static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - struct snd_soc_codec *codec = rtd->codec; - - /* - * 4 buttons here map to the google Reference headset - * The use of these buttons can be decided by the user space. - */ - ret = snd_soc_card_jack_new(&skylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3, &skylake_headset, - NULL, 0); - if (ret) { - dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); - return ret; - } - - nau8825_enable_jack_detect(codec, &skylake_headset); - - return ret; -} - -static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - - /* The ADSP will covert the FE rate to 48k, stereo */ - rate->min = rate->max = 48000; - channels->min = channels->max = 2; - - /* set SSP0 to 24 bit */ - snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); - return 0; -} - -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); - - return ret; -} - -static struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, -}; - -/* skylake digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link skylake_dais[] = { - /* Front End DAI links */ - { - .name = "Skl Audio Port", - .stream_name = "Audio", - .cpu_dai_name = "System Pin", - .platform_name = "0000:00:1f.3", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - }, - { - .name = "Skl Audio Capture Port", - .stream_name = "Audio Record", - .cpu_dai_name = "System Pin", - .platform_name = "0000:00:1f.3", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - }, - { - .name = "Skl Audio Reference cap", - .stream_name = "refcap", - .cpu_dai_name = "Reference Pin", - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .platform_name = "0000:00:1f.3", - .init = NULL, - .dpcm_capture = 1, - .ignore_suspend = 1, - .nonatomic = 1, - .dynamic = 1, - }, - /* Back End DAI links */ - { - /* SSP0 - Codec */ - .name = "SSP0-Codec", - .be_id = 0, - .cpu_dai_name = "SSP0 Pin", - .platform_name = "0000:00:1f.3", - .no_pcm = 1, - .codecs = ssm4567_codec_components, - .num_codecs = ARRAY_SIZE(ssm4567_codec_components), - .dai_fmt = SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .init = skylake_ssm4567_codec_init, - .ignore_suspend = 1, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = skylake_ssp_fixup, - .dpcm_playback = 1, - }, - { - /* SSP1 - Codec */ - .name = "SSP1-Codec", - .be_id = 0, - .cpu_dai_name = "SSP1 Pin", - .platform_name = "0000:00:1f.3", - .no_pcm = 1, - .codec_name = "i2c-10508825:00", - .codec_dai_name = SKL_NUVOTON_CODEC_DAI, - .init = skylake_nau8825_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ignore_suspend = 1, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = skylake_ssp_fixup, - .ops = &skylake_nau8825_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - }, - { - .name = "dmic01", - .be_id = 1, - .cpu_dai_name = "DMIC01 Pin", - .codec_name = "dmic-codec", - .codec_dai_name = "dmic-hifi", - .platform_name = "0000:00:1f.3", - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - }, -}; - -/* skylake audio machine driver for SPT + NAU88L25 */ -static struct snd_soc_card skylake_audio_card = { - .name = "sklnau8825adi", - .owner = THIS_MODULE, - .dai_link = skylake_dais, - .num_links = ARRAY_SIZE(skylake_dais), - .controls = skylake_controls, - .num_controls = ARRAY_SIZE(skylake_controls), - .dapm_widgets = skylake_widgets, - .num_dapm_widgets = ARRAY_SIZE(skylake_widgets), - .dapm_routes = skylake_map, - .num_dapm_routes = ARRAY_SIZE(skylake_map), - .codec_conf = ssm4567_codec_conf, - .num_configs = ARRAY_SIZE(ssm4567_codec_conf), -}; - -static int skylake_audio_probe(struct platform_device *pdev) -{ - skylake_audio_card.dev = &pdev->dev; - - return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card); -} - -static struct platform_driver skylake_audio = { - .probe = skylake_audio_probe, - .driver = { - .name = "skl_nau88l25_ssm4567_i2s", - .pm = &snd_soc_pm_ops, - }, -}; - -module_platform_driver(skylake_audio) - -/* Module information */ -MODULE_AUTHOR("Conrad Cooke <conrad.cooke(a)intel.com>"); -MODULE_AUTHOR("Harsha Priya <harshapriya.n(a)intel.com>"); -MODULE_AUTHOR("Naveen M <naveen.m(a)intel.com>"); -MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); -MODULE_AUTHOR("Yong Zhi <yong.zhi(a)intel.com>"); -MODULE_DESCRIPTION("Intel Audio Machine driver for SKL with NAU88L25 and SSM4567 in I2S Mode"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:skl_nau88l25_ssm4567_i2s"); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index d3e87b6..0551322 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -574,7 +574,7 @@ static void skl_remove(struct pci_dev *pci) static struct sst_acpi_mach sst_skl_devdata[] = { { "INT343A", "skl_alc286s_i2s", "intel/dsp_fw_release.bin", NULL, NULL, NULL }, - { "INT343B", "skl_nau88l25_ssm4567_i2s", "intel/dsp_fw_release.bin", + { "INT343B", "skl_nau8825_ssm4567_i2s", "intel/dsp_fw_release.bin", NULL, NULL, NULL }, {} }; -- 2.6.0.rc2.230.g3dd15c0

1 0

[alsa-devel] [PATCH] ASoC: rsnd: care SWSP bit for TDM/non-TDM
by Kuninori Morimoto 02 Dec '15

02 Dec '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> SSICR::SWSP bit controls WS signal low/high, but in case of TDM it is inverted. This patch solves this issue. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 628739f..79c3211 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -265,6 +265,9 @@ static int rsnd_ssi_config_init(struct rsnd_ssi *ssi, u32 cr_own; u32 cr_mode; u32 wsr; + int is_tdm; + + is_tdm = (rsnd_get_slot_runtime(io) >= 6) ? 1 : 0; /* * always use 32bit system word. @@ -274,7 +277,7 @@ static int rsnd_ssi_config_init(struct rsnd_ssi *ssi, if (rdai->bit_clk_inv) cr_own |= SCKP; - if (rdai->frm_clk_inv) + if (rdai->frm_clk_inv ^ is_tdm) cr_own |= SWSP; if (rdai->data_alignment) cr_own |= SDTA; @@ -307,7 +310,7 @@ static int rsnd_ssi_config_init(struct rsnd_ssi *ssi, * rsnd_ssiu_init_gen2() */ wsr = ssi->wsr; - if (rsnd_get_slot_runtime(io) >= 6) { + if (is_tdm) { wsr |= WS_MODE; cr_own |= CHNL_8; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: dapm: Make enable/disable_pin work with always on widgets
by Charles Keepax 02 Dec '15

02 Dec '15

Always on widgets currently have some odd interactions with DAPM. Enabling/disabling a widget (snd_soc_dapm_enable_pin) then connecting it to a path works as expected, ie. when the widget is disabled the path doesn't power up and it does when the widget is enabled. However once in a path enabling the widget does not cause anything to power up, dapm_widget_set_power will return the current power state of the widget as 1, meaning we never check peer power states. This patch updates dapm_always_on_check_power to return w->connected such that it is effected by snd_soc_dapm_enable_pin and the like. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a8156b5..5a2812f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1300,7 +1300,7 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) static int dapm_always_on_check_power(struct snd_soc_dapm_widget *w) { - return 1; + return w->connected; } static int dapm_seq_compare(struct snd_soc_dapm_widget *a, -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: usx2y: fix inconsistent indenting on if statement
by Colin King 02 Dec '15

02 Dec '15

From: Colin Ian King <colin.king(a)canonical.com> minor change, indenting is one tab out. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/usx2y/usbusx2yaudio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 61d5dc2..dd40ca9 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -166,7 +166,7 @@ static int usX2Y_urb_play_prepare(struct snd_usX2Y_substream *subs, /* set the buffer pointer */ urb->transfer_buffer = runtime->dma_area + subs->hwptr * usX2Y->stride; if ((subs->hwptr += count) >= runtime->buffer_size) - subs->hwptr -= runtime->buffer_size; + subs->hwptr -= runtime->buffer_size; } else urb->transfer_buffer = subs->tmpbuf; -- 2.6.2

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[alsa-devel] [PATCH v4 0/3]ASoC: cygnus: Add audio support for Broadcom Cygnus SoC
by Simran Rai 02 Dec '15

02 Dec '15

Hi, This patchset contains audio support for Broadcom's Cygnus SoC. It contains DT bindings and core audio driver. The audio driver supports both capture and playback of Audio PCM samples over I2S/TDM interface and provides playback support over SPDIF interface. This patchset is derived from a previously submitted patchset: http://lkml.iu.edu/hypermail/linux/kernel/1503.3/05434.html This patchset has been tested on Cygnus wireless audio bcm958305K board. It is based on v4.3-rc5 and is available from github: repo: https://github.com/Broadcom/cygnus-linux/tree/cygnus-sound-v4 Changes from v3: - Fix the subject lines to match the style for the subsystem Simran Rai (3): ASoC: cygnus: Add DT bindings for Broadcom Cygnus audio ASoC: cygnus: Add Cygnus audio DAI driver ASoC: cygnus: Add Cygnus audio DMA driver .../bindings/sound/brcm,cygnus-audio.txt | 54 + sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/cygnus-pcm.c | 859 +++++++++++ sound/soc/bcm/cygnus-ssp.c | 1508 ++++++++++++++++++++ sound/soc/bcm/cygnus-ssp.h | 129 ++ 6 files changed, 2564 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt create mode 100644 sound/soc/bcm/cygnus-pcm.c create mode 100644 sound/soc/bcm/cygnus-ssp.c create mode 100644 sound/soc/bcm/cygnus-ssp.h -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: dapm: Replace siggen widget type with mic widgets
by Charles Keepax 02 Dec '15

02 Dec '15

The siggen widget type doesn't really require its own special handling. It can be handled identically to a mic widget, ie. it is a source of a signal and causes things to power up when an output is connected. Additionally, several places (cs42l52, cs42l56, wm8962, arizona_haptics) tie siggens in to the input framework and the pattern for this is usually to use snd_soc_dapm_enable_pin/disable_pin on the siggen widget. However, as siggen widgets present as always on (the power check always returns 1), this leads to some odd interactions with DAPM. Enabling/disabling the widget then connecting it to a path works as expected, however once in a path enabling the widget has no effect. dapm_widget_set_power will return immediately as the current power state of the widget will return 1, meaning we never check peer power states. Treating the widget as a mic widget causes the expected behaviour in all cases. This does cause a minor cosmetic side effect that siggen widgets no longer show as always on through debugfs, although in the case described above that is more representive of their behaviour. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 2 +- sound/soc/soc-dapm.c | 6 +----- 2 files changed, 2 insertions(+), 6 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 9706946..67f176f 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -47,7 +47,7 @@ struct device; /* platform domain */ #define SND_SOC_DAPM_SIGGEN(wname) \ -{ .id = snd_soc_dapm_siggen, .name = wname, .kcontrol_news = NULL, \ +{ .id = snd_soc_dapm_mic, .name = wname, .kcontrol_news = NULL, \ .num_kcontrols = 0, .reg = SND_SOC_NOPM } #define SND_SOC_DAPM_SINK(wname) \ { .id = snd_soc_dapm_sink, .name = wname, .kcontrol_news = NULL, \ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a8156b5..ea6f477 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1817,12 +1817,9 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) /* Supplies and micbiases only bring the * context up to STANDBY as unless something * else is active and passing audio they - * generally don't require full power. Signal - * generators are virtual pins and have no - * power impact themselves. + * generally don't require full power. */ switch (w->id) { - case snd_soc_dapm_siggen: case snd_soc_dapm_vmid: break; case snd_soc_dapm_supply: @@ -3354,7 +3351,6 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->power_check = dapm_generic_check_power; break; case snd_soc_dapm_vmid: - case snd_soc_dapm_siggen: w->is_ep = SND_SOC_DAPM_EP_SOURCE; w->power_check = dapm_always_on_check_power; break; -- 2.1.4

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[alsa-devel] snd-fireworks requires larger period sizes than FFADO
by Dominic Sacré 02 Dec '15

02 Dec '15

Hi, I'm trying to use an Echo AudioFire 4 with the snd-fireworks driver. With FFADO the device works reasonably well at a period size of 64 samples (on a PREEMPT_RT kernel), whereas with snd-fireworks the lowest possible period size seems to be 256 samples. This is what JACK says when I try to start it with a period size of 128 samples: configuring for 48000Hz, period = 128 frames (2.7 ms), buffer = 2 periods ALSA: final selected sample format for capture: 32bit integer little-endian ALSA: cannot set period size to 128 frames for capture The lowest achievable analog roundtrip latency seems to be around 19ms (at 2x256 frames), as opposed to 8.3ms with FFADO (at 2x64). Are there any known limitations of the snd-fireworks driver in this regard? Thanks, Dominic

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[alsa-devel] [PATCH 0/14] ASoC: rsnd: TDM Extend Mode support
by Kuninori Morimoto 02 Dec '15

02 Dec '15

Hi Mark These patch-set adds TDM Extend Mode to Renesas R-Car sound driver. 1) - 9) tidyup / cleanup patches 10) - 14) TDM Extend Mode support Unfortunately, our board can't use this TDM Extend Mode, because our SoC and codec are using different style TDM. Thus I have tested this patches on special board which doesn't have codec chip, and I checked signal via oscilloscope. oKuninori Morimoto (14): 1) ASoC: rsnd: don't use normal *mod in adg.c 2) ASoC: rsnd: add missing ADINR::CHNUM on DVC/SRC/SSIU 3) ASoC: rsnd: add missing SRC_O_BUSIF_MODE register 4) ASoC: rsnd: src: rename rsnd_src_soft_reset() to rsnd_src_activation() 5) ASoC: rsnd: mix: rename rsnd_mix_soft_reset() to rsnd_mix_activation() 6) ASoC: rsnd: dvc: rename rsnd_dvc_soft_reset() to rsnd_dvc_activation() 7) ASoC: rsnd: src: add rsnd_src_halt() 8) ASoC: rsnd: mix: add rsnd_mix_halt() 9) ASoC: rsnd: dvc: add rsnd_dvc_halt() 10) ASoC: rsnd: add rsnd_ssi_config_init() 11) ASoC: rsnd: set SSIWSR setting on rsnd_ssi_config_init() 12) ASoC: rsnd: ssi enables non-stereo sound 13) ASoC: rsnd: dvc enables non-stereo sound 14) ASoC: rsnd: add TDM Extend Mode support sound/soc/sh/rcar/adg.c | 30 +++++++++++------------------- sound/soc/sh/rcar/core.c | 65 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++----- sound/soc/sh/rcar/dvc.c | 32 +++++++++++++++++++++++++++----- sound/soc/sh/rcar/gen.c | 11 ++++++++++- sound/soc/sh/rcar/mix.c | 12 ++++++++++-- sound/soc/sh/rcar/rsnd.h | 19 +++++++++++++++++-- sound/soc/sh/rcar/src.c | 18 ++++++++++++++---- sound/soc/sh/rcar/ssi.c | 126 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++--------------------------------------------------- sound/soc/sh/rcar/ssiu.c | 12 +++++++++++- 9 files changed, 235 insertions(+), 90 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: rsnd: fixup wrong snd_soc_dai_driver pointer access
by Kuninori Morimoto 01 Dec '15

01 Dec '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> drv pointer should be "base + offset" instead of "current + offset". This patch fixup this issue, otherwise third and subsequent pointer will be broken Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index b187a89..f1d7af1 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -603,7 +603,7 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) struct device_node *playback, *capture; struct rsnd_dai_stream *io_playback; struct rsnd_dai_stream *io_capture; - struct snd_soc_dai_driver *drv; + struct snd_soc_dai_driver *rdrv, *drv; struct rsnd_dai *rdai; struct device *dev = rsnd_priv_to_dev(priv); int nr, dai_i, io_i, np_i; @@ -616,15 +616,15 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) goto rsnd_dai_probe_done; } - drv = devm_kzalloc(dev, sizeof(*drv) * nr, GFP_KERNEL); + rdrv = devm_kzalloc(dev, sizeof(*rdrv) * nr, GFP_KERNEL); rdai = devm_kzalloc(dev, sizeof(*rdai) * nr, GFP_KERNEL); - if (!drv || !rdai) { + if (!rdrv || !rdai) { ret = -ENOMEM; goto rsnd_dai_probe_done; } priv->rdai_nr = nr; - priv->daidrv = drv; + priv->daidrv = rdrv; priv->rdai = rdai; /* @@ -633,7 +633,7 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) dai_i = 0; for_each_child_of_node(dai_node, dai_np) { rdai = rsnd_rdai_get(priv, dai_i); - drv = drv + dai_i; + drv = rdrv + dai_i; io_playback = &rdai->playback; io_capture = &rdai->capture; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: rsrc-card: check return value of snd_soc_of_get_dai_name()
by Kuninori Morimoto 01 Dec '15

01 Dec '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch adds missing check of snd_soc_of_get_dai_name(). It might not be able to use sound card, because it might returns -EPROBE_DEFER. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsrc-card.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/rsrc-card.c b/sound/soc/sh/rcar/rsrc-card.c index d61db9c..a3ec13f 100644 --- a/sound/soc/sh/rcar/rsrc-card.c +++ b/sound/soc/sh/rcar/rsrc-card.c @@ -208,7 +208,9 @@ static int rsrc_card_parse_links(struct device_node *np, dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; dai_link->cpu_of_node = args.np; - snd_soc_of_get_dai_name(np, &dai_link->cpu_dai_name); + ret = snd_soc_of_get_dai_name(np, &dai_link->cpu_dai_name); + if (ret < 0) + return ret; /* set dai_name */ snprintf(dai_props->dai_name, DAI_NAME_NUM, "fe.%s", @@ -240,7 +242,9 @@ static int rsrc_card_parse_links(struct device_node *np, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = rsrc_card_be_hw_params_fixup; dai_link->codec_of_node = args.np; - snd_soc_of_get_dai_name(np, &dai_link->codec_dai_name); + ret = snd_soc_of_get_dai_name(np, &dai_link->codec_dai_name); + if (ret < 0) + return ret; /* additional name prefix */ if (of_data) { -- 1.9.1

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