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Re: [alsa-devel] is 440 Hz the true harmonic wave
by Joël Krähemann 31 Mar '14

31 Mar '14

On Mon, 2014-03-31 at 10:27 +0200, Felix Homann wrote: > 2014-03-31 10:20 GMT+02:00 Joël Krähemann <joel(a)weedlight.ch>: > > > I believe that 440 Hz is an approximation rather than true > harmonic on > it's pressure. Or am I wrong? Sine is harmonic in default arc. > > > > What are you talking about??? About scale. Am I wrong when I believe 440 Hz has physical property to be harmonic?

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[alsa-devel] [PATCH] ALSA: hda - Enable beep for ASUS 1015E
by W. Trevor King 31 Mar '14

31 Mar '14

The `lspci -nnvv` output contains (wrapped for line length): 00:1b.0 Audio device [0403]: Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller [8086:1e20] (rev 04) Subsystem: ASUSTeK Computer Inc. Device [1043:115d] Signed-off-by: W. Trevor King <wking(a)tremily.us> --- This patch is against git://git.alsa-project.org/alsa-kernel.git master (currently e38d6d3, ALSA: hda - Add Dell headset detection quirk for three laptop models, 2013-12-13), but I imagine it will apply cleanly to any recent branch. I'm happy to fill in any other information folks want, in case there's uncertainty about whether it's just a one-off quirk for my hardware instance, or if it's a general quirk for all 1043:115d devices. Cheers, Trevor 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 3578f11..7cfe209 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1001,6 +1001,7 @@ static int alc_codec_rename_from_preset(struct hda_codec *codec) static const struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x103c, "ASUS", 1), + SND_PCI_QUIRK(0x1043, 0x115d, "ASUS", 1), SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1), SND_PCI_QUIRK(0x1043, 0x8376, "EeePC", 1), SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1), -- 1.9.1.353.gc66d89d

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[alsa-devel] [patch] ALSA: asihpi: fix some indenting in snd_card_asihpi_pcm_new()
by Dan Carpenter 31 Mar '14

31 Mar '14

This used to be a part of a condition until f3d145aac913 ('ALSA: asihpi: MMAP for non-busmaster cards') but now it's not and we can remove an indent level. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index e4e42f2..901c949 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1253,11 +1253,12 @@ static int snd_card_asihpi_pcm_new(struct snd_card_asihpi *asihpi, int device) num_outstreams, num_instreams, &pcm); if (err < 0) return err; + /* pointer to ops struct is stored, dont change ops afterwards! */ - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, - &snd_card_asihpi_playback_mmap_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, - &snd_card_asihpi_capture_mmap_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_card_asihpi_playback_mmap_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_card_asihpi_capture_mmap_ops); pcm->private_data = asihpi; pcm->info_flags = 0;

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[alsa-devel] is 440 Hz the true harmonic wave
by Joël Krähemann 31 Mar '14

31 Mar '14

Is 440 Hz the true harmonic wave? Is there a constant for it in glibc, like M_PI?

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[alsa-devel] RT priority 99 on soundcard
by Joël Krähemann 31 Mar '14

31 Mar '14

Is running on top of soundcard with realtime priority 99 problematic? /* and now async */ { static const struct timespec req = { 0, (250000000 * (1 / 45)), }; if(!AGS_IS_DEVOUT_THREAD(thread)){ nanosleep(&req, NULL); } }

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[alsa-devel] [PATCH 1/6] ASoC: rt5640: Remove the pre-allocated size of reg_default
by Oder Chiou 31 Mar '14

31 Mar '14

In order to prevent the redundant memory usage, the pre-allocated size of reg_default should be remove. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index cf041e3..9427511 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -59,7 +59,7 @@ static struct reg_default init_list[] = { }; #define RT5640_INIT_REG_LEN ARRAY_SIZE(init_list) -static const struct reg_default rt5640_reg[RT5640_VENDOR_ID2 + 1] = { +static const struct reg_default rt5640_reg[] = { { 0x00, 0x000e }, { 0x01, 0xc8c8 }, { 0x02, 0xc8c8 }, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: rsnd: use fast_io for bus
by Ben Dooks 29 Mar '14

29 Mar '14

The rcar sound driver uses regmap to access registers in various parts of the block and uses regmap to manage mappings. The regmap is created without fast_io set, which means it locks with a mutex rather than a lighter-spinlock. The use of the mutex lock causes issues when the IRQ handler is entered as the code needs to read/write register values and thus with lock debugging enabled the system outputs a number of warnings such as: BUG: sleeping function called from invalid context at rnel/locking/mutex.c:616 The rcar registers are all connected via APB bus and thus not that slow to access. The fix is to set the fast_io in the code so that regmap creates the regmap structures using the faster spinlock functions. Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> --- sound/soc/sh/rcar/gen.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 9094970..6468962 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -66,6 +66,7 @@ static int rsnd_regmap_read32(void *context, } static struct regmap_bus rsnd_regmap_bus = { + .fast_io = true, .write = rsnd_regmap_write32, .read = rsnd_regmap_read32, .reg_format_endian_default = REGMAP_ENDIAN_NATIVE, -- 1.9.0

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[alsa-devel] Input/output error Alsa api
by Thiago Carísio 29 Mar '14

29 Mar '14

Hi! I am Thiago and i'm starting to use the alsa api. I'm testing one function to make the sound capture and i got this input/output error. I don't get it why this is happening...so, please...some kind soul can help me? Thanks

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[alsa-devel] Advanced Gtk+ Sequencer - Beta Release
by Joël Krähemann 29 Mar '14

29 Mar '14

Hi, I just released the very first beta, 3 hours ago. There are 2 weeks towards final release 0.4.0 and any suggestions are welcome. https://sourceforge.net/projects/ags/

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Re: [alsa-devel] [PATCH] ASoC: Add support for multi register mux
by Songhee Baek 28 Mar '14

28 Mar '14

> > On 03/26/2014 11:41 PM, Songhee Baek wrote: > > >> On 03/26/2014 01:02 AM, Arun Shamanna Lakshmi wrote: > > >> > > >> The way you describe this it seems to me that a value array for > > >> this kind of mux would look like. > > >> > > >> 0x00000000, 0x00000000, 0x00000001 > > >> 0x00000000, 0x00000000, 0x00000002 > > >> 0x00000000, 0x00000000, 0x00000003 > > >> 0x00000000, 0x00000000, 0x00000004 > > >> 0x00000000, 0x00000000, 0x00000008 > > >> ... > > >> > > >> That seems to be extremely tedious. If the MUX uses a one hot > > >> encoding how about storing the index of the bit in the values array > > >> and use (1 << value) when writing the value to the register? > > > > > > If we store the index of the bit, the value will be duplicated for > > > each > > registers inputs since register has 0 to 31bits to shift, then we need > > to decode the index to interpret value for which registers to set. If > > we need to interpret the decoded value of index, it is better to have > > custom put/get function in our driver, isn't it? > > > > > > > I'm not sure I understand. If you use (val / 32) to pick the register > > and (val % > > 32) to pick the bit in the register this should work just fine. Maybe > > I'm missing something. Do you have a real world code example of of the > > this type of enum is used? > > > > I can use val/32 and val%32 for this multi register mux. With this logic, put function would be easy however get function becomes tedious due to looping on each bit per register for 3 of them in our case. Rather, it would be easy to identify a unique MUX_OFFSET to distinguish between the 3 registers as shown in the code below. #define MULTI_MUX_INPUT_OFFSET(n) (5 * n) int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val, reg_idx; if (e->reg[0] != SND_SOC_NOPM) { for (reg_idx = 0; reg_idx < e->num_regs; reg_idx) { reg_val = snd_soc_read(codec, e->reg[reg_idx]); val = (reg_val >> e->shift_l) & e->mask[reg_idx]; if (val) val += MULTI_MUX_INPUT_OFFSET(reg_idx); } } else { reg_val = dapm_kcontrol_get_value(kcontrol); val = (reg_val >> e->shift_l) & e->mask[0]; } ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); if (e->shift_l != e->shift_r) { val = (reg_val >> e->shift_r) & e->mask[0]; val = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = val; } return 0; } int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int *item = ucontrol->value.enumerated.item; unsigned int change, i, value, update_idx = 0; unsigned int mask; struct snd_soc_dapm_update update; int ret = 0, reg_val; if (item[0] >= e->items) return -EINVAL; value = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] > e->items) return -EINVAL; value |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l; mask |= e->mask[0] << e->shift_r; } if (e->num_regs < 2) { goto update_reg; } for (i = 0; i < e->num_regs; i++) { reg_val = value - MULTI_MUX_INPUT_OFFSET(i); /* checking reg_val is power of 2 : one-hot code */ /* if reg_val after subtract MULTI_MUX_INPUT_OFFSET is not power of 2, reg[i] should be zero */ if (reg_val & (reg_val - 1)) { /* clear the current input register */ snd_soc_write(codec, e->reg[i], 0); } else { /* reg_val is power of 2, store updated info */ value = reg_val; mask = e->mask[i]; update_idx = i; } } update_reg: mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); if (e->reg[update_idx] != SND_SOC_NOPM) change = snd_soc_test_bits(codec, e->reg[update_idx], mask, value); else change = dapm_kcontrol_set_value(kcontrol, value); if (change) { if (e->reg[update_idx] != SND_SOC_NOPM) { update.kcontrol = kcontrol; update.reg = e->reg[update_idx]; update.mask= mask; update.val = value; card->update = &update; } ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); card->update = NULL; } mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); return change; } > > >>> - int reg; > > >>> + int reg[SOC_ENUM_MAX_REGS]; > > >>> unsigned char shift_l; > > >>> unsigned char shift_r; > > >>> unsigned int items; > > >>> - unsigned int mask; > > >>> + unsigned int mask[SOC_ENUM_MAX_REGS]; > > >> > > >> If you make mask and reg pointers instead of arrays this should be > > >> much more flexible and not be limited to 3 registers. We will make reg* and mask* instead of arrays and since we use the same structure, the plan is to share the get and put function code. Thanks. Songhee.

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[alsa-devel] [PATCH] ALSA: hda - verify pin:cvt connection on preparing a stream for Intel HDMI codec
by mengdong.lin＠intel.com 28 Mar '14

28 Mar '14

From: Mengdong Lin <mengdong.lin(a)intel.com> The pin:converter connection may get reset after S3 on Intel Broadwell HDMI codec. And this can cause multiple pins share a same convertor and mute control will affect each other. We observed a resumed audio playback become silent after S3. This patch verifies pin:cvt connection on preparing a stream, to assure the pin selects the right convetor and an assigned convertor is not shared by other unused pins. Apply this fix-up on Haswell, Broadwell and Baytrail. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3ab7063..6000ce9 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -68,6 +68,7 @@ struct hdmi_spec_per_pin { hda_nid_t pin_nid; int num_mux_nids; hda_nid_t mux_nids[HDA_MAX_CONNECTIONS]; + int mux_idx; hda_nid_t cvt_nid; struct hda_codec *codec; @@ -1342,6 +1343,8 @@ static int hdmi_choose_cvt(struct hda_codec *codec, if (cvt_idx == spec->num_cvts) return -ENODEV; + per_pin->mux_idx = mux_idx; + if (cvt_id) *cvt_id = cvt_idx; if (mux_id) @@ -1350,6 +1353,22 @@ static int hdmi_choose_cvt(struct hda_codec *codec, return 0; } +/* Assure the pin select the right convetor */ +static void intel_verify_pin_cvt_connect(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin) +{ + hda_nid_t pin_nid = per_pin->pin_nid; + int mux_idx, curr; + + mux_idx = per_pin->mux_idx; + curr = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_CONNECT_SEL, 0); + if (curr != mux_idx) + snd_hda_codec_write_cache(codec, pin_nid, 0, + AC_VERB_SET_CONNECT_SEL, + mux_idx); +} + /* Intel HDMI workaround to fix audio routing issue: * For some Intel display codecs, pins share the same connection list. * So a conveter can be selected by multiple pins and playback on any of these @@ -1751,6 +1770,12 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, bool non_pcm; int pinctl; + if (is_haswell_plus(codec) || is_valleyview(codec)) { + /* the pin:cvt connection may get reset after S3 */ + intel_verify_pin_cvt_connect(codec, per_pin); + intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx); + } + non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); mutex_lock(&per_pin->lock); per_pin->channels = substream->runtime->channels; -- 1.8.1.2

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Re: [alsa-devel] Invalid PCM pointer with CMI8738 and pcm.c buffer underuns
by Raymond Yau 28 Mar '14

28 Mar '14

> > I've previously posted under ALSA's user mailing list so to avoid the > clutter, the details about what's happening is here: > http://article.gmane.org/gmane.linux.alsa.user/38258 > http://article.gmane.org/gmane.linux.alsa.user/38260 > > There was no debug info under /proc/asound/card0/ so nothing to give from > there unless you guys need something specific. > > Output of alsa-debug.sh: > http://www.alsa-project.org/db/?f=26187faadd47f3626029ba667cb004e74695f2b2 https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Seem cannot read valid value from register within three trials state.CMI8738 { control.1 { iface PCM device 2 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 comment { access read type INTEGER count 6 range '0 - 36' } } Seem bug in adding channel map to iec958 instead of analog https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… pcm.!default { type plug slave.pcm "dmixer" } You need asym plugin for playback and capture when overwrite pcm.default

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[alsa-devel] tlv320aic32x4 Codec ADC and DAC must shutdown to alter clocks
by Mike Looijmans 28 Mar '14

28 Mar '14

My general question: If "my" codec needs to alter the state of the DAPM tree, how can the codec ask or inform the DAPM components of this? This problem occurs on tlv320aic32x4 codecs, but probably applies to many others as well. Using the tlv320aic32x4 a simple loop like this may "crash" the codec: while true do speaker-test -c 2 -t sine -l 1 -r 48000 > /dev/null speaker-test -c 2 -t sine -l 1 -r 44100 > /dev/null done A "crash" means, in this respect that the chip loses some of its settings, malfunctions and often it results in NACKs on I2C commands. I've been trying to properly fix this, but it just won't fit in the framework. The cause is that the DAPM starts the DAC and/or ADC cores while the clocks aren't running, and stops the clocks before shutting down the cores. There is the overall bug of the driver that starts and stops all the clocks in the aic32x4_set_bias_level in the wrong order - it should start them from source to sink, and stop them in the opposite direction, so the PLL must be the first to start and last to stop. Apparently this isn't really critical as I've never observed failure as a result of this. What I cannot fix is that changing the parameters (like clock frequency) must stop the DAC core. The DAPM framework just sets the BIAS to "ON" only once, and does not shut down until after a time out. Which is good. But if the next stream to play needs a different clock setting, the DAC must be stopped first, the code must wait until the DAC has powered down, and then the clocks can be stopped and altered. When all settings are done, the DAC can be restarted. Starting the DAC is the DAPM's task, it's this line that does the job: SND_SOC_DAPM_DAC("Left DAC", "Left Playback", AIC32X4_DACSETUP, 7, 0), I can detect the change in the hw_params, and on the first stream to play, this is called while still in STANDBY or lower levels. However, when the codec is already in ON bias level, how do I tell DAPM that I had to shut down the DAC in order to change the parameters? Just toggling the related bits will keep the DAC powered down. A routine to store and restore the power state is possible, but very ineffective. Especially since the "I must shutdown" conclusion can also be in the clock setting routine. That would result in multiple down-up cycles. -- Mike Looijmans

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[alsa-devel] [PATCH v6 0/2] ASoC: Add SiRF I2S controller and hdmi card drivers
by RongJun Ying 28 Mar '14

28 Mar '14

From: Rongjun Ying <rongjun.ying(a)csr.com> This patchset adds CSR SiRFSoC sound drivers including: 4. The SiRF I2S controller driver 5. The SiRF hdmi card driver Rongjun Ying (2): ASoC: sirf: Add SiRF I2S driver ASoC: sirf: Add hdmi support .../devicetree/bindings/sound/sirf-i2s.txt | 27 ++ sound/soc/sirf/Kconfig | 11 + sound/soc/sirf/Makefile | 2 + sound/soc/sirf/sirf-i2s.c | 445 ++++++++++++++++++++ sound/soc/sirf/sirf-i2s.h | 93 ++++ 5 files changed, 578 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sirf-i2s.txt create mode 100644 sound/soc/sirf/sirf-i2s.c create mode 100644 sound/soc/sirf/sirf-i2s.h -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: fsl_sai: Improve fsl_sai_isr()
by Nicolin Chen 28 Mar '14

28 Mar '14

This patch improves fsl_sai_isr() in these ways: 1, Add comment for mask fetching code. 2, Return IRQ_NONE if the IRQ is not for the device. 3, Use regmap_write() instead of regmap_update_bits(). Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 64 ++++++++++++++++++++++++++++++++++--------------- 1 file changed, 45 insertions(+), 19 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 0bc98bb..f088545 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -30,62 +30,88 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid) { struct fsl_sai *sai = (struct fsl_sai *)devid; struct device *dev = &sai->pdev->dev; - u32 xcsr, mask; + u32 flags, xcsr, mask; + bool irq_none = true; - /* Only handle those what we enabled */ + /* + * Both IRQ status bits and IRQ mask bits are in the xCSR but + * different shifts. And we here create a mask only for those + * IRQs that we activated. + */ mask = (FSL_SAI_FLAGS >> FSL_SAI_CSR_xIE_SHIFT) << FSL_SAI_CSR_xF_SHIFT; /* Tx IRQ */ regmap_read(sai->regmap, FSL_SAI_TCSR, &xcsr); - xcsr &= mask; + flags = xcsr & mask; - if (xcsr & FSL_SAI_CSR_WSF) + if (flags) + irq_none = false; + else + goto irq_rx; + + if (flags & FSL_SAI_CSR_WSF) dev_dbg(dev, "isr: Start of Tx word detected\n"); - if (xcsr & FSL_SAI_CSR_SEF) + if (flags & FSL_SAI_CSR_SEF) dev_warn(dev, "isr: Tx Frame sync error detected\n"); - if (xcsr & FSL_SAI_CSR_FEF) { + if (flags & FSL_SAI_CSR_FEF) { dev_warn(dev, "isr: Transmit underrun detected\n"); /* FIFO reset for safety */ xcsr |= FSL_SAI_CSR_FR; } - if (xcsr & FSL_SAI_CSR_FWF) + if (flags & FSL_SAI_CSR_FWF) dev_dbg(dev, "isr: Enabled transmit FIFO is empty\n"); - if (xcsr & FSL_SAI_CSR_FRF) + if (flags & FSL_SAI_CSR_FRF) dev_dbg(dev, "isr: Transmit FIFO watermark has been reached\n"); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, - FSL_SAI_CSR_xF_W_MASK | FSL_SAI_CSR_FR, xcsr); + flags &= FSL_SAI_CSR_xF_W_MASK; + xcsr &= ~FSL_SAI_CSR_xF_MASK; + + if (flags) + regmap_write(sai->regmap, FSL_SAI_TCSR, flags | xcsr); +irq_rx: /* Rx IRQ */ regmap_read(sai->regmap, FSL_SAI_RCSR, &xcsr); - xcsr &= mask; + flags = xcsr & mask; - if (xcsr & FSL_SAI_CSR_WSF) + if (flags) + irq_none = false; + else + goto out; + + if (flags & FSL_SAI_CSR_WSF) dev_dbg(dev, "isr: Start of Rx word detected\n"); - if (xcsr & FSL_SAI_CSR_SEF) + if (flags & FSL_SAI_CSR_SEF) dev_warn(dev, "isr: Rx Frame sync error detected\n"); - if (xcsr & FSL_SAI_CSR_FEF) { + if (flags & FSL_SAI_CSR_FEF) { dev_warn(dev, "isr: Receive overflow detected\n"); /* FIFO reset for safety */ xcsr |= FSL_SAI_CSR_FR; } - if (xcsr & FSL_SAI_CSR_FWF) + if (flags & FSL_SAI_CSR_FWF) dev_dbg(dev, "isr: Enabled receive FIFO is full\n"); - if (xcsr & FSL_SAI_CSR_FRF) + if (flags & FSL_SAI_CSR_FRF) dev_dbg(dev, "isr: Receive FIFO watermark has been reached\n"); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, - FSL_SAI_CSR_xF_W_MASK | FSL_SAI_CSR_FR, xcsr); + flags &= FSL_SAI_CSR_xF_W_MASK; + xcsr &= ~FSL_SAI_CSR_xF_MASK; - return IRQ_HANDLED; + if (flags) + regmap_write(sai->regmap, FSL_SAI_TCSR, flags | xcsr); + +out: + if (irq_none) + return IRQ_NONE; + else + return IRQ_HANDLED; } static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, -- 1.8.4

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[alsa-devel] [PATCH 1/3 v2] ASoC: ak5386: add regulator consumer support
by Daniel Mack 28 Mar '14

28 Mar '14

The chip has two power supplies, VA and VDD. Enable them both as long as the codec is in use. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/ak5386.c | 50 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 50 insertions(+) diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index 72e953b..a30be5c 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -14,12 +14,18 @@ #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/of_device.h> +#include <linux/regulator/consumer.h> #include <sound/soc.h> #include <sound/pcm.h> #include <sound/initval.h> +static const char const *supply_names[] = { + "va", "vd" +}; + struct ak5386_priv { int reset_gpio; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = { @@ -32,7 +38,42 @@ static const struct snd_soc_dapm_route ak5386_dapm_routes[] = { { "Capture", NULL, "AINR" }, }; +static int ak5386_soc_probe(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); +} + +static int ak5386_soc_remove(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; +} + +#ifdef CONFIG_PM +static int ak5386_soc_suspend(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; +} + +static int ak5386_soc_resume(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); +} +#else +#define ak5386_soc_suspend NULL +#define ak5386_soc_resume NULL +#endif /* CONFIG_PM */ + static struct snd_soc_codec_driver soc_codec_ak5386 = { + .probe = ak5386_soc_probe, + .remove = ak5386_soc_remove, + .suspend = ak5386_soc_suspend, + .resume = ak5386_soc_resume, .dapm_widgets = ak5386_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak5386_dapm_widgets), .dapm_routes = ak5386_dapm_routes, @@ -122,6 +163,7 @@ static int ak5386_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct ak5386_priv *priv; + int ret, i; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -130,6 +172,14 @@ static int ak5386_probe(struct platform_device *pdev) priv->reset_gpio = -EINVAL; dev_set_drvdata(dev, priv); + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) + return ret; + if (of_match_device(of_match_ptr(ak5386_dt_ids), dev)) priv->reset_gpio = of_get_named_gpio(dev->of_node, "reset-gpio", 0); -- 1.8.5.3

2 9

[alsa-devel] [PATCH v2 0/4] ASoC: davinci-mcasp: Correct start/stop sequences
by Peter Ujfalusi 28 Mar '14

28 Mar '14

Hi, Changes since v1: - Correction for the stop sequence is added - start code cleanup patch added The startup sequence for TX and RX was not correct, not following the programming sequence from the TRM. This could cause initial channel swap when starting TX. When stopping the stream the AFIFO should be stopped as the last step to avoid unpredictable behavior. The change has been tested on AM335x and AM437x but it should be valid for all devices. Regards, Peter --- Peter Ujfalusi (4): ASoC: davinci-mcasp: Correct TX start sequence ASoC: davinci-mcasp: Correct RX start sequence ASoC: davinci-mcasp: When stopping TX/RX stop the AFIFO as the last step ASoC: davinci-mcasp: Move the AFIFO related code under start_tx/rx functions sound/soc/davinci/davinci-mcasp.c | 95 ++++++++++++++++++--------------------- sound/soc/davinci/davinci-mcasp.h | 6 +++ 2 files changed, 49 insertions(+), 52 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH v2] ASoC: sta350: Add codec driver
by Daniel Mack 28 Mar '14

28 Mar '14

From: Sven Brandau <brandau(a)gmx.de> The TI STA350 is an integrated 2.1-channel power amplifier that is controllable over I2C. This patch adds an ASoC driver for it. At a glance, this chip is very similar to the STA320 for which a driver already exists. In details, however, the register maps contain subtle differences which made a whole new driver easier to write and maintain. [daniel(a)zonque.org: cleanups, DT property rework, rebased on asoc-next] Signed-off-by: Sven Brandau <brandau(a)gmx.de> Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- v1 -> v2: * Added my S-o-b * Made the Kconfig symbol visible * Documented regulator notes in DT bindings doc * Fixed spelling and other style nits * Converted to SOC_ENUM_SINGLE_DECL * Added a lock around the logic in sta350_coefficient_get() * Added a DAPM route from "DAC" to "Playback" * Use dev_dbg() rather than pr_debug() * Changed the code to make use of regmap_update_bits rather than open-coding the same functionality * Cleaned up the reset sequence logic * If a gpio was requested, the claiming has to succeed (handles EPROBE_DEFER cases) * Kick codec->control_data initialization * Skip initialization of default values in regcache * Apply power before running init sequences * Parse the DT without consulting of_match_device() .../devicetree/bindings/sound/st,sta350.txt | 107 ++ include/sound/sta350.h | 52 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta350.c | 1266 ++++++++++++++++++++ sound/soc/codecs/sta350.h | 228 ++++ 6 files changed, 1660 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,sta350.txt create mode 100644 include/sound/sta350.h create mode 100644 sound/soc/codecs/sta350.c create mode 100644 sound/soc/codecs/sta350.h diff --git a/Documentation/devicetree/bindings/sound/st,sta350.txt b/Documentation/devicetree/bindings/sound/st,sta350.txt new file mode 100644 index 0000000..07f9c80 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,sta350.txt @@ -0,0 +1,107 @@ +STA350 audio CODEC + +The driver for this device only supports I2C. + +Required properties: + + - compatible: "st,sta350" + - reg: the I2C address of the device for I2C + - reset-gpio: a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + + - power-down-gpio: a GPIO spec for the power down pin. If specified, + it will be deasserted before communication to the codec + starts. + + - vdd-dig-supply: regulator spec, providing 3.3V + - vdd-pll-supply: regulator spec, providing 3.3V + - vcc-supply: regulator spec, providing 5V - 26V + +Optional properties: + + - st,output-conf: number, Selects the output configuration: + 0: 2-channel (full-bridge) power, 2-channel data-out + 1: 2 (half-bridge). 1 (full-bridge) on-board power + 2: 2 Channel (Full-Bridge) Power, 1 Channel FFX + 3: 1 Channel Mono-Parallel + If parameter is missing, mode 0 will be enabled. + + - st,ch1-output-mapping: Channel 1 output mapping + - st,ch2-output-mapping: Channel 2 output mapping + - st,ch3-output-mapping: Channel 3 output mapping + 0: Channel 1 + 1: Channel 2 + 2: Channel 3 + If parameter is missing, channel 1 is choosen. + + - st,thermal-warning-recover: + If present, thermal warning recovery is enabled. + + - st,thermal-warning-adjustment: + If present, thermal warning adjustment is enabled. + + - st,fault-detect-recovery: + If present, then fault recovery will be enabled. + + - st,ffx-power-output-mode: string + The FFX power output mode selects how the FFX output timing is + configured. Must be one of these values: + - "drop-compensation" + - "tapered-compensation" + - "full-power-mode" + - "variable-drop-compensation" (default) + + - st,drop-compensation-ns: number + Only required for "st,ffx-power-output-mode" == + "variable-drop-compensation". + Specifies the drop compensation in nanoseconds. + The value must be in the range of 0..300, and only + multiples of 20 are allowed. Default is 140ns. + + - st,overcurrent-warning-adjustment: + If present, overcurrent warning adjustment is enabled. + + - st,max-power-use-mpcc: + If present, then MPCC bits are used for MPC coefficients, + otherwise standard MPC coefficients are used. + + - st,max-power-corr: + If present, power bridge correction for THD reduction near maximum + power output is enabled. + + - st,am-reduction-mode: + If present, FFX mode runs in AM reduction mode, otherwise normal + FFX mode is used. + + - st,odd-pwm-speed-mode: + If present, PWM speed mode run on odd speed mode (341.3 kHz) on all + channels. If not present, normal PWM spped mode (384 kHz) will be used. + + - st,distortion-compensation: + If present, distortion compensation variable uses DCC coefficient. + If not present, preset DC coefficient is used. + + - st,invalid-input-detect-mute: + If not present, automatic invalid input detect mute is enabled. + + + +Example: + +codec: sta350@38 { + compatible = "st,sta350"; + reg = <0x1c>; + reset-gpio = <&gpio1 19 0>; + power-down-gpio = <&gpio1 16 0>; + st,output-conf = <0x3>; // set output to 2-channel + // (full-bridge) power, + // 2-channel data-out + st,ch1-output-mapping = <0>; // set channel 1 output ch 1 + st,ch2-output-mapping = <0>; // set channel 2 output ch 1 + st,ch3-output-mapping = <0>; // set channel 3 output ch 1 + st,max-power-correction; // enables power bridge + // correction for THD reduction + // near maximum power output + st,invalid-input-detect-mute; // mute if no valid digital + // audio signal is provided. +}; diff --git a/include/sound/sta350.h b/include/sound/sta350.h new file mode 100644 index 0000000..3a329810 --- /dev/null +++ b/include/sound/sta350.h @@ -0,0 +1,52 @@ +/* + * Platform data for ST STA350 ASoC codec driver. + * + * Copyright: 2014 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#ifndef __LINUX_SND__STA350_H +#define __LINUX_SND__STA350_H + +#define STA350_OCFG_2CH 0 +#define STA350_OCFG_2_1CH 1 +#define STA350_OCFG_1CH 3 + +#define STA350_OM_CH1 0 +#define STA350_OM_CH2 1 +#define STA350_OM_CH3 2 + +#define STA350_THERMAL_ADJUSTMENT_ENABLE 1 +#define STA350_THERMAL_RECOVERY_ENABLE 2 +#define STA350_FAULT_DETECT_RECOVERY_BYPASS 1 + +#define STA350_FFX_PM_DROP_COMP 0 +#define STA350_FFX_PM_TAPERED_COMP 1 +#define STA350_FFX_PM_FULL_POWER 2 +#define STA350_FFX_PM_VARIABLE_DROP_COMP 3 + + +struct sta350_platform_data { + u8 output_conf; + u8 ch1_output_mapping; + u8 ch2_output_mapping; + u8 ch3_output_mapping; + u8 ffx_power_output_mode; + u8 drop_compensation_ns; + unsigned int thermal_warning_recovery:1; + unsigned int thermal_warning_adjustment:1; + unsigned int fault_detect_recovery:1; + unsigned int oc_warning_adjustment:1; + unsigned int max_power_use_mpcc:1; + unsigned int max_power_correction:1; + unsigned int am_reduction_mode:1; + unsigned int odd_pwm_speed_mode:1; + unsigned int distortion_compensation:1; + unsigned int invalid_input_detect_mute:1; +}; + +#endif /* __LINUX_SND__STA350_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f0e8401..c7b853f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -80,6 +80,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SSM2602_SPI if SPI_MASTER select SND_SOC_SSM2602_I2C if I2C select SND_SOC_STA32X if I2C + select SND_SOC_STA350 if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TAS5086 if I2C @@ -435,6 +436,10 @@ config SND_SOC_SSM2602_I2C config SND_SOC_STA32X tristate +config SND_SOC_STA350 + tristate "STA350 speaker amplifier" + depends on I2C + config SND_SOC_STA529 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3c4d275..efdb4d0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -74,6 +74,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-ssm2602-spi-objs := ssm2602-spi.o snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o snd-soc-sta32x-objs := sta32x.o +snd-soc-sta350-objs := sta350.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o @@ -221,6 +222,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o obj-$(CONFIG_SND_SOC_SSM2602_I2C) += snd-soc-ssm2602-i2c.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o +obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c new file mode 100644 index 0000000..915b8769 --- /dev/null +++ b/sound/soc/codecs/sta350.c @@ -0,0 +1,1266 @@ +/* + * Codec driver for ST STA350 2.1-channel high-efficiency digital audio system + * + * Copyright: 2014 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * based on code from: + * Raumfeld GmbH + * Johannes Stezenbach <js(a)sig21.net> + * Wolfson Microelectronics PLC. + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * Freescale Semiconductor, Inc. + * Timur Tabi <timur(a)freescale.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__ + +#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/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.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 <sound/sta350.h> +#include "sta350.h" + +#define STA350_RATES (SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +#define STA350_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + +/* Power-up register defaults */ +static const struct reg_default sta350_regs[] = { + { 0x0, 0x63 }, + { 0x1, 0x80 }, + { 0x2, 0xdf }, + { 0x3, 0x40 }, + { 0x4, 0xc2 }, + { 0x5, 0x5c }, + { 0x6, 0x00 }, + { 0x7, 0xff }, + { 0x8, 0x60 }, + { 0x9, 0x60 }, + { 0xa, 0x60 }, + { 0xb, 0x00 }, + { 0xc, 0x00 }, + { 0xd, 0x00 }, + { 0xe, 0x00 }, + { 0xf, 0x40 }, + { 0x10, 0x80 }, + { 0x11, 0x77 }, + { 0x12, 0x6a }, + { 0x13, 0x69 }, + { 0x14, 0x6a }, + { 0x15, 0x69 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1a, 0x00 }, + { 0x1b, 0x00 }, + { 0x1c, 0x00 }, + { 0x1d, 0x00 }, + { 0x1e, 0x00 }, + { 0x1f, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x2a }, + { 0x28, 0xc0 }, + { 0x29, 0xf3 }, + { 0x2a, 0x33 }, + { 0x2b, 0x00 }, + { 0x2c, 0x0c }, + { 0x31, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x01 }, + { 0x3a, 0xee }, + { 0x3b, 0xff }, + { 0x3c, 0x7e }, + { 0x3d, 0xc0 }, + { 0x3e, 0x26 }, + { 0x3f, 0x00 }, + { 0x48, 0x00 }, + { 0x49, 0x00 }, + { 0x4a, 0x00 }, + { 0x4b, 0x04 }, + { 0x4c, 0x00 }, +}; + +static const struct regmap_range sta350_write_regs_range[] = { + regmap_reg_range(STA350_CONFA, STA350_AUTO2), + regmap_reg_range(STA350_C1CFG, STA350_FDRC2), + regmap_reg_range(STA350_EQCFG, STA350_EVOLRES), + regmap_reg_range(STA350_NSHAPE, STA350_MISC2), +}; + +static const struct regmap_range sta350_read_regs_range[] = { + regmap_reg_range(STA350_CONFA, STA350_AUTO2), + regmap_reg_range(STA350_C1CFG, STA350_STATUS), + regmap_reg_range(STA350_EQCFG, STA350_EVOLRES), + regmap_reg_range(STA350_NSHAPE, STA350_MISC2), +}; + +static const struct regmap_range sta350_volatile_regs_range[] = { + regmap_reg_range(STA350_CFADDR2, STA350_CFUD), + regmap_reg_range(STA350_STATUS, STA350_STATUS), +}; + +static const struct regmap_access_table sta350_write_regs = { + .yes_ranges = sta350_write_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_write_regs_range), +}; + +static const struct regmap_access_table sta350_read_regs = { + .yes_ranges = sta350_read_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_read_regs_range), +}; + +static const struct regmap_access_table sta350_volatile_regs = { + .yes_ranges = sta350_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_volatile_regs_range), +}; + +/* regulator power supply names */ +static const char const *sta350_supply_names[] = { + "vdd-dig", /* digital supply, 3.3V */ + "vdd-pll", /* pll supply, 3.3V */ + "vcc" /* power amp supply, 5V - 26V */ +}; + +/* codec private data */ +struct sta350_priv { + struct regmap *regmap; + struct regulator_bulk_data supplies[ARRAY_SIZE(sta350_supply_names)]; + struct sta350_platform_data *pdata; + + unsigned int mclk; + unsigned int format; + + u32 coef_shadow[STA350_COEF_COUNT]; + int shutdown; + int gpio_nreset; + int gpio_power_down; + + struct mutex coeff_lock; +}; + +static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1); +static const DECLARE_TLV_DB_SCALE(tone_tlv, -1200, 200, 0); + +static const char const *sta350_drc_ac[] = { + "Anti-Clipping", "Dynamic Range Compression" +}; +static const char const *sta350_auto_gc_mode[] = { + "User", "AC no clipping", "AC limited clipping (10%)", + "DRC nighttime listening mode" +}; +static const char const *sta350_auto_xo_mode[] = { + "User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", + "200Hz", "220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", + "340Hz", "360Hz" +}; +static const char const *sta350_binary_output[] = { + "FFX 3-state output - normal operation", "Binary output" +}; +static const char const *sta350_limiter_select[] = { + "Limiter Disabled", "Limiter #1", "Limiter #2" +}; +static const char const *sta350_limiter_attack_rate[] = { + "3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024", + "0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752", + "0.0645", "0.0564", "0.0501", "0.0451" +}; +static const char const *sta350_limiter_release_rate[] = { + "0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299", + "0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137", + "0.0134", "0.0117", "0.0110", "0.0104" +}; +static const char const *sta350_noise_shaper_type[] = { + "Third order", "Fourth order" +}; + +static const unsigned int sta350_limiter_ac_attack_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 7, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(300, 100, 0), +}; + +static const unsigned int sta350_limiter_ac_release_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-2900, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-2000, 0, 0), + 3, 8, TLV_DB_SCALE_ITEM(-1400, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const unsigned int sta350_limiter_drc_attack_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0), + 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0), + 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0), +}; + +static const unsigned int sta350_limiter_drc_release_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 2, TLV_DB_SCALE_ITEM(-3800, 200, 0), + 3, 4, TLV_DB_SCALE_ITEM(-3300, 200, 0), + 5, 12, TLV_DB_SCALE_ITEM(-3000, 200, 0), + 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0), +}; + +static SOC_ENUM_SINGLE_DECL(sta350_drc_ac_enum, + STA350_CONFD, STA350_CONFD_DRC_SHIFT, + sta350_drc_ac); +static SOC_ENUM_SINGLE_DECL(sta350_noise_shaper_enum, + STA350_CONFE, STA350_CONFE_NSBW_SHIFT, + sta350_noise_shaper_type); +static SOC_ENUM_SINGLE_DECL(sta350_auto_gc_enum, + STA350_AUTO1, STA350_AUTO1_AMGC_SHIFT, + sta350_auto_gc_mode); +static SOC_ENUM_SINGLE_DECL(sta350_auto_xo_enum, + STA350_AUTO2, STA350_AUTO2_XO_SHIFT, + sta350_auto_xo_mode); +static SOC_ENUM_SINGLE_DECL(sta350_binary_output_ch1_enum, + STA350_C1CFG, STA350_CxCFG_BO_SHIFT, + sta350_binary_output); +static SOC_ENUM_SINGLE_DECL(sta350_binary_output_ch2_enum, + STA350_C2CFG, STA350_CxCFG_BO_SHIFT, + sta350_binary_output); +static SOC_ENUM_SINGLE_DECL(sta350_binary_output_ch3_enum, + STA350_C3CFG, STA350_CxCFG_BO_SHIFT, + sta350_binary_output); +static SOC_ENUM_SINGLE_DECL(sta350_limiter_ch1_enum, + STA350_C1CFG, STA350_CxCFG_LS_SHIFT, + sta350_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta350_limiter_ch2_enum, + STA350_C2CFG, STA350_CxCFG_LS_SHIFT, + sta350_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta350_limiter_ch3_enum, + STA350_C3CFG, STA350_CxCFG_LS_SHIFT, + sta350_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta350_limiter1_attack_rate_enum, + STA350_L1AR, STA350_LxA_SHIFT, + sta350_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta350_limiter2_attack_rate_enum, + STA350_L2AR, STA350_LxA_SHIFT, + sta350_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta350_limiter1_release_rate_enum, + STA350_L1AR, STA350_LxR_SHIFT, + sta350_limiter_release_rate); +static SOC_ENUM_SINGLE_DECL(sta350_limiter2_release_rate_enum, + STA350_L2AR, STA350_LxR_SHIFT, + sta350_limiter_release_rate); + +/* + * byte array controls for setting biquad, mixer, scaling coefficients; + * for biquads all five coefficients need to be set in one go, + * mixer and pre/postscale coefs can be set individually; + * each coef is 24bit, the bytes are ordered in the same way + * as given in the STA350 data sheet (big endian; b1, b2, a1, a2, b0) + */ + +static int sta350_coefficient_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int numcoef = kcontrol->private_value >> 16; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = 3 * numcoef; + return 0; +} + +static int sta350_coefficient_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int numcoef = kcontrol->private_value >> 16; + int index = kcontrol->private_value & 0xffff; + unsigned int cfud, val; + int i, ret = 0; + + mutex_lock(&sta350->coeff_lock); + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + /* + * chip documentation does not say if the bits are self clearing, + * so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + + regmap_write(sta350->regmap, STA350_CFADDR2, index); + if (numcoef == 1) { + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x04); + } else if (numcoef == 5) { + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x08); + } else { + ret = -EINVAL; + goto exit_unlock; + } + + for (i = 0; i < 3 * numcoef; i++) { + regmap_read(sta350->regmap, STA350_B1CF1 + i, &val); + ucontrol->value.bytes.data[i] = val; + } + +exit_unlock: + mutex_unlock(&sta350->coeff_lock); + + return ret; +} + +static int sta350_coefficient_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int numcoef = kcontrol->private_value >> 16; + int index = kcontrol->private_value & 0xffff; + unsigned int cfud; + int i; + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + /* + * chip documentation does not say if the bits are self clearing, + * so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + + regmap_write(sta350->regmap, STA350_CFADDR2, index); + for (i = 0; i < numcoef && (index + i < STA350_COEF_COUNT); i++) + sta350->coef_shadow[index + i] = + (ucontrol->value.bytes.data[3 * i] << 16) + | (ucontrol->value.bytes.data[3 * i + 1] << 8) + | (ucontrol->value.bytes.data[3 * i + 2]); + for (i = 0; i < 3 * numcoef; i++) + regmap_write(sta350->regmap, STA350_B1CF1 + i, + ucontrol->value.bytes.data[i]); + if (numcoef == 1) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x01); + else if (numcoef == 5) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x02); + else + return -EINVAL; + + return 0; +} + +static int sta350_sync_coef_shadow(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int cfud; + int i; + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + + for (i = 0; i < STA350_COEF_COUNT; i++) { + regmap_write(sta350->regmap, STA350_CFADDR2, i); + regmap_write(sta350->regmap, STA350_B1CF1, + (sta350->coef_shadow[i] >> 16) & 0xff); + regmap_write(sta350->regmap, STA350_B1CF2, + (sta350->coef_shadow[i] >> 8) & 0xff); + regmap_write(sta350->regmap, STA350_B1CF3, + (sta350->coef_shadow[i]) & 0xff); + /* + * chip documentation does not say if the bits are + * self-clearing, so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x01); + } + return 0; +} + +static int sta350_cache_sync(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int mute; + int rc; + + /* mute during register sync */ + regmap_read(sta350->regmap, STA350_CFUD, &mute); + regmap_write(sta350->regmap, STA350_MMUTE, mute | STA350_MMUTE_MMUTE); + sta350_sync_coef_shadow(codec); + rc = regcache_sync(sta350->regmap); + regmap_write(sta350->regmap, STA350_MMUTE, mute); + return rc; +} + +#define SINGLE_COEF(xname, index) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sta350_coefficient_info, \ + .get = sta350_coefficient_get,\ + .put = sta350_coefficient_put, \ + .private_value = index | (1 << 16) } + +#define BIQUAD_COEFS(xname, index) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sta350_coefficient_info, \ + .get = sta350_coefficient_get,\ + .put = sta350_coefficient_put, \ + .private_value = index | (5 << 16) } + +static const struct snd_kcontrol_new sta350_snd_controls[] = { +SOC_SINGLE_TLV("Master Volume", STA350_MVOL, 0, 0xff, 1, mvol_tlv), +/* VOL */ +SOC_SINGLE_TLV("Ch1 Volume", STA350_C1VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch2 Volume", STA350_C2VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch3 Volume", STA350_C3VOL, 0, 0xff, 1, chvol_tlv), +/* CONFD */ +SOC_SINGLE("High Pass Filter Bypass Switch", + STA350_CONFD, STA350_CONFD_HPB_SHIFT, 1, 1), +SOC_SINGLE("De-emphasis Filter Switch", + STA350_CONFD, STA350_CONFD_DEMP_SHIFT, 1, 0), +SOC_SINGLE("DSP Bypass Switch", + STA350_CONFD, STA350_CONFD_DSPB_SHIFT, 1, 0), +SOC_SINGLE("Post-scale Link Switch", + STA350_CONFD, STA350_CONFD_PSL_SHIFT, 1, 0), +SOC_SINGLE("Biquad Coefficient Link Switch", + STA350_CONFD, STA350_CONFD_BQL_SHIFT, 1, 0), +SOC_ENUM("Compressor/Limiter Switch", sta350_drc_ac_enum), +SOC_ENUM("Noise Shaper Bandwidth", sta350_noise_shaper_enum), +SOC_SINGLE("Zero-detect Mute Enable Switch", + STA350_CONFD, STA350_CONFD_ZDE_SHIFT, 1, 0), +SOC_SINGLE("Submix Mode Switch", + STA350_CONFD, STA350_CONFD_SME_SHIFT, 1, 0), +/* CONFE */ +SOC_SINGLE("Zero Cross Switch", STA350_CONFE, STA350_CONFE_ZCE_SHIFT, 1, 0), +SOC_SINGLE("Soft Ramp Switch", STA350_CONFE, STA350_CONFE_SVE_SHIFT, 1, 0), +/* MUTE */ +SOC_SINGLE("Master Switch", STA350_MMUTE, STA350_MMUTE_MMUTE_SHIFT, 1, 1), +SOC_SINGLE("Ch1 Switch", STA350_MMUTE, STA350_MMUTE_C1M_SHIFT, 1, 1), +SOC_SINGLE("Ch2 Switch", STA350_MMUTE, STA350_MMUTE_C2M_SHIFT, 1, 1), +SOC_SINGLE("Ch3 Switch", STA350_MMUTE, STA350_MMUTE_C3M_SHIFT, 1, 1), +/* AUTOx */ +SOC_ENUM("Automode GC", sta350_auto_gc_enum), +SOC_ENUM("Automode XO", sta350_auto_xo_enum), +/* CxCFG */ +SOC_SINGLE("Ch1 Tone Control Bypass Switch", + STA350_C1CFG, STA350_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Tone Control Bypass Switch", + STA350_C2CFG, STA350_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch1 EQ Bypass Switch", + STA350_C1CFG, STA350_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 EQ Bypass Switch", + STA350_C2CFG, STA350_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch1 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch3 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_ENUM("Ch1 Binary Output Select", sta350_binary_output_ch1_enum), +SOC_ENUM("Ch2 Binary Output Select", sta350_binary_output_ch2_enum), +SOC_ENUM("Ch3 Binary Output Select", sta350_binary_output_ch3_enum), +SOC_ENUM("Ch1 Limiter Select", sta350_limiter_ch1_enum), +SOC_ENUM("Ch2 Limiter Select", sta350_limiter_ch2_enum), +SOC_ENUM("Ch3 Limiter Select", sta350_limiter_ch3_enum), +/* TONE */ +SOC_SINGLE_RANGE_TLV("Bass Tone Control Volume", + STA350_TONE, STA350_TONE_BTC_SHIFT, 1, 13, 0, tone_tlv), +SOC_SINGLE_RANGE_TLV("Treble Tone Control Volume", + STA350_TONE, STA350_TONE_TTC_SHIFT, 1, 13, 0, tone_tlv), +SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta350_limiter1_attack_rate_enum), +SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta350_limiter2_attack_rate_enum), +SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta350_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta350_limiter2_release_rate_enum), + +/* + * depending on mode, the attack/release thresholds have + * two different enum definitions; provide both + */ +SOC_SINGLE_TLV("Limiter1 Attack Threshold (AC Mode)", + STA350_L1ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (AC Mode)", + STA350_L2ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (AC Mode)", + STA350_L1ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (AC Mode)", + STA350_L2ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter1 Attack Threshold (DRC Mode)", + STA350_L1ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (DRC Mode)", + STA350_L2ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (DRC Mode)", + STA350_L1ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_drc_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (DRC Mode)", + STA350_L2ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_drc_release_tlv), + +BIQUAD_COEFS("Ch1 - Biquad 1", 0), +BIQUAD_COEFS("Ch1 - Biquad 2", 5), +BIQUAD_COEFS("Ch1 - Biquad 3", 10), +BIQUAD_COEFS("Ch1 - Biquad 4", 15), +BIQUAD_COEFS("Ch2 - Biquad 1", 20), +BIQUAD_COEFS("Ch2 - Biquad 2", 25), +BIQUAD_COEFS("Ch2 - Biquad 3", 30), +BIQUAD_COEFS("Ch2 - Biquad 4", 35), +BIQUAD_COEFS("High-pass", 40), +BIQUAD_COEFS("Low-pass", 45), +SINGLE_COEF("Ch1 - Prescale", 50), +SINGLE_COEF("Ch2 - Prescale", 51), +SINGLE_COEF("Ch1 - Postscale", 52), +SINGLE_COEF("Ch2 - Postscale", 53), +SINGLE_COEF("Ch3 - Postscale", 54), +SINGLE_COEF("Thermal warning - Postscale", 55), +SINGLE_COEF("Ch1 - Mix 1", 56), +SINGLE_COEF("Ch1 - Mix 2", 57), +SINGLE_COEF("Ch2 - Mix 1", 58), +SINGLE_COEF("Ch2 - Mix 2", 59), +SINGLE_COEF("Ch3 - Mix 1", 60), +SINGLE_COEF("Ch3 - Mix 2", 61), +}; + +static const struct snd_soc_dapm_widget sta350_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LEFT"), +SND_SOC_DAPM_OUTPUT("RIGHT"), +SND_SOC_DAPM_OUTPUT("SUB"), +}; + +static const struct snd_soc_dapm_route sta350_dapm_routes[] = { + { "LEFT", NULL, "DAC" }, + { "RIGHT", NULL, "DAC" }, + { "SUB", NULL, "DAC" }, + { "DAC", NULL, "Playback" }, +}; + +/* MCLK interpolation ratio per fs */ +static struct { + int fs; + int ir; +} interpolation_ratios[] = { + { 32000, 0 }, + { 44100, 0 }, + { 48000, 0 }, + { 88200, 1 }, + { 96000, 1 }, + { 176400, 2 }, + { 192000, 2 }, +}; + +/* MCLK to fs clock ratios */ +static int mcs_ratio_table[3][6] = { + { 768, 512, 384, 256, 128, 576 }, + { 384, 256, 192, 128, 64, 0 }, + { 192, 128, 96, 64, 32, 0 }, +}; + +/** + * sta350_set_dai_sysclk - configure MCLK + * @codec_dai: the codec DAI + * @clk_id: the clock ID (ignored) + * @freq: the MCLK input frequency + * @dir: the clock direction (ignored) + * + * The value of MCLK is used to determine which sample rates are supported + * by the STA350, based on the mcs_ratio_table. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + */ +static int sta350_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "mclk=%u\n", freq); + sta350->mclk = freq; + + return 0; +} + +/** + * sta350_set_dai_fmt - configure the codec for the selected audio format + * @codec_dai: the codec DAI + * @fmt: a SND_SOC_DAIFMT_x value indicating the data format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + */ +static int sta350_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int confb = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + sta350->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + confb |= STA350_CONFB_C2IM; + break; + case SND_SOC_DAIFMT_NB_IF: + confb |= STA350_CONFB_C1IM; + break; + default: + return -EINVAL; + } + + return regmap_update_bits(sta350->regmap, STA350_CONFB, + STA350_CONFB_C1IM | STA350_CONFB_C2IM, confb); +} + +/** + * sta350_hw_params - program the STA350 with the given hardware parameters. + * @substream: the audio stream + * @params: the hardware parameters to set + * @dai: the SOC DAI (ignored) + * + * This function programs the hardware with the values provided. + * Specifically, the sample rate and the data format. + */ +static int sta350_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 sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int i, mcs = -EINVAL, ir = -EINVAL; + unsigned int confa, confb; + unsigned int rate, ratio; + int ret; + + if (!sta350->mclk) { + dev_err(codec->dev, + "sta350->mclk is unset. Unable to determine ratio\n"); + return -EIO; + } + + rate = params_rate(params); + ratio = sta350->mclk / rate; + dev_dbg(codec->dev, "rate: %u, ratio: %u\n", rate, ratio); + + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) { + if (interpolation_ratios[i].fs == rate) { + ir = interpolation_ratios[i].ir; + break; + } + } + + if (ir < 0) { + dev_err(codec->dev, "Unsupported samplerate: %u\n", rate); + return -EINVAL; + } + + for (i = 0; i < 6; i++) { + if (mcs_ratio_table[ir][i] == ratio) { + mcs = i; + break; + } + } + + if (mcs < 0) { + dev_err(codec->dev, "Unresolvable ratio: %u\n", ratio); + return -EINVAL; + } + + confa = (ir << STA350_CONFA_IR_SHIFT) | + (mcs << STA350_CONFA_MCS_SHIFT); + confb = 0; + + switch (params_width(params)) { + case 24: + dev_dbg(codec->dev, "24bit\n"); + /* fall through */ + case 32: + dev_dbg(codec->dev, "24bit or 32bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x1; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x2; + break; + } + + break; + case 20: + dev_dbg(codec->dev, "20bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x4; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x5; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x6; + break; + } + + break; + case 18: + dev_dbg(codec->dev, "18bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x8; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x9; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xa; + break; + } + + break; + case 16: + dev_dbg(codec->dev, "16bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0xd; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xe; + break; + } + + break; + default: + return -EINVAL; + } + + ret = regmap_update_bits(sta350->regmap, STA350_CONFA, + STA350_CONFA_MCS_MASK | STA350_CONFA_IR_MASK, + confa); + if (ret < 0) + return ret; + + ret = regmap_update_bits(sta350->regmap, STA350_CONFB, + STA350_CONFB_SAI_MASK | STA350_CONFB_SAIFB, + confb); + if (ret < 0) + return ret; + + return 0; +} + +static int sta350_startup_sequence(struct sta350_priv *sta350) +{ + if (gpio_is_valid(sta350->gpio_power_down)) + gpio_set_value(sta350->gpio_power_down, 1); + + if (gpio_is_valid(sta350->gpio_nreset)) { + gpio_set_value(sta350->gpio_nreset, 0); + mdelay(1); + gpio_set_value(sta350->gpio_nreset, 1); + mdelay(1); + } + + return 0; +} + +/** + * sta350_set_bias_level - DAPM callback + * @codec: the codec device + * @level: DAPM power level + * + * This is called by ALSA to put the codec into low power mode + * or to wake it up. If the codec is powered off completely + * all registers must be restored after power on. + */ +static int sta350_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int ret; + + dev_dbg(codec->dev, "level = %d\n", level); + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Full power on */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, + STA350_CONFF_PWDN | STA350_CONFF_EAPD); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable( + ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + sta350_startup_sequence(sta350); + sta350_cache_sync(codec); + } + + /* Power down */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, + 0); + + break; + + case SND_SOC_BIAS_OFF: + /* The chip runs through the power down sequence for us */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, 0); + + /* power down: low */ + if (gpio_is_valid(sta350->gpio_power_down)) + gpio_set_value(sta350->gpio_power_down, 0); + + if (gpio_is_valid(sta350->gpio_nreset)) + gpio_set_value(sta350->gpio_nreset, 0); + + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), + sta350->supplies); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const struct snd_soc_dai_ops sta350_dai_ops = { + .hw_params = sta350_hw_params, + .set_sysclk = sta350_set_dai_sysclk, + .set_fmt = sta350_set_dai_fmt, +}; + +static struct snd_soc_dai_driver sta350_dai = { + .name = "sta350-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA350_RATES, + .formats = STA350_FORMATS, + }, + .ops = &sta350_dai_ops, +}; + +#ifdef CONFIG_PM +static int sta350_suspend(struct snd_soc_codec *codec) +{ + sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int sta350_resume(struct snd_soc_codec *codec) +{ + sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define sta350_suspend NULL +#define sta350_resume NULL +#endif + +static int sta350_probe(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + struct sta350_platform_data *pdata = sta350->pdata; + int i, ret = 0, thermal = 0; + + ret = regulator_bulk_enable(ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + ret = sta350_startup_sequence(sta350); + if (ret < 0) { + dev_err(codec->dev, "Failed to startup device\n"); + return ret; + } + + /* CONFA */ + if (!pdata->thermal_warning_recovery) + thermal |= STA350_CONFA_TWAB; + if (!pdata->thermal_warning_adjustment) + thermal |= STA350_CONFA_TWRB; + if (!pdata->fault_detect_recovery) + thermal |= STA350_CONFA_FDRB; + regmap_update_bits(sta350->regmap, STA350_CONFA, + STA350_CONFA_TWAB | STA350_CONFA_TWRB | + STA350_CONFA_FDRB, + thermal); + + /* CONFC */ + regmap_update_bits(sta350->regmap, STA350_CONFC, + STA350_CONFC_OM_MASK, + pdata->ffx_power_output_mode + << STA350_CONFC_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_CONFC, + STA350_CONFC_CSZ_MASK, + pdata->drop_compensation_ns + << STA350_CONFC_CSZ_SHIFT); + regmap_update_bits(sta350->regmap, + STA350_CONFC, + STA350_CONFC_OCRB, + pdata->oc_warning_adjustment ? + STA350_CONFC_OCRB : 0); + + /* CONFE */ + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_MPCV, + pdata->max_power_use_mpcc ? + STA350_CONFE_MPCV : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_MPC, + pdata->max_power_correction ? + STA350_CONFE_MPC : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_AME, + pdata->am_reduction_mode ? + STA350_CONFE_AME : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_PWMS, + pdata->odd_pwm_speed_mode ? + STA350_CONFE_PWMS : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_DCCV, + pdata->distortion_compensation ? + STA350_CONFE_DCCV : 0); + /* CONFF */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_IDE, + pdata->invalid_input_detect_mute ? + STA350_CONFF_IDE : 0); + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_OCFG_MASK, + pdata->output_conf + << STA350_CONFF_OCFG_SHIFT); + + /* channel to output mapping */ + regmap_update_bits(sta350->regmap, STA350_C1CFG, + STA350_CxCFG_OM_MASK, + pdata->ch1_output_mapping + << STA350_CxCFG_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_C2CFG, + STA350_CxCFG_OM_MASK, + pdata->ch2_output_mapping + << STA350_CxCFG_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_C3CFG, + STA350_CxCFG_OM_MASK, + pdata->ch3_output_mapping + << STA350_CxCFG_OM_SHIFT); + + /* initialize coefficient shadow RAM with reset values */ + for (i = 4; i <= 49; i += 5) + sta350->coef_shadow[i] = 0x400000; + for (i = 50; i <= 54; i++) + sta350->coef_shadow[i] = 0x7fffff; + sta350->coef_shadow[55] = 0x5a9df7; + sta350->coef_shadow[56] = 0x7fffff; + sta350->coef_shadow[59] = 0x7fffff; + sta350->coef_shadow[60] = 0x400000; + sta350->coef_shadow[61] = 0x400000; + + sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies); + + return 0; +} + +static int sta350_remove(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + + sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies); + + return 0; +} + +static const struct snd_soc_codec_driver sta350_codec = { + .probe = sta350_probe, + .remove = sta350_remove, + .suspend = sta350_suspend, + .resume = sta350_resume, + .set_bias_level = sta350_set_bias_level, + .controls = sta350_snd_controls, + .num_controls = ARRAY_SIZE(sta350_snd_controls), + .dapm_widgets = sta350_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sta350_dapm_widgets), + .dapm_routes = sta350_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sta350_dapm_routes), +}; + +static const struct regmap_config sta350_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = STA350_MISC2, + .reg_defaults = sta350_regs, + .num_reg_defaults = ARRAY_SIZE(sta350_regs), + .cache_type = REGCACHE_RBTREE, + .wr_table = &sta350_write_regs, + .rd_table = &sta350_read_regs, + .volatile_table = &sta350_volatile_regs, +}; + +#ifdef CONFIG_OF +static const struct of_device_id st350_dt_ids[] = { + { .compatible = "st,sta350", }, + { } +}; +MODULE_DEVICE_TABLE(of, st350_dt_ids); + +static const char const *sta350_ffx_modes[] = { + [STA350_FFX_PM_DROP_COMP] = "drop-compensation", + [STA350_FFX_PM_TAPERED_COMP] = "tapered-compensation", + [STA350_FFX_PM_FULL_POWER] = "full-power-mode", + [STA350_FFX_PM_VARIABLE_DROP_COMP] = "variable-drop-compensation", +}; + +static int sta350_probe_dt(struct device *dev, struct sta350_priv *sta350) +{ + struct device_node *np = dev->of_node; + struct sta350_platform_data *pdata; + const char *ffx_power_mode; + u16 tmp; + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + sta350->gpio_nreset = of_get_named_gpio(np, "reset-gpio", 0); + sta350->gpio_power_down = of_get_named_gpio(np, "power-down-gpio", 0); + + of_property_read_u8(np, "st,output-conf", + &pdata->output_conf); + of_property_read_u8(np, "st,ch1-output-mapping", + &pdata->ch1_output_mapping); + of_property_read_u8(np, "st,ch2-output-mapping", + &pdata->ch2_output_mapping); + of_property_read_u8(np, "st,ch3-output-mapping", + &pdata->ch3_output_mapping); + + if (of_get_property(np, "st,thermal-warning-recovery", NULL)) + pdata->thermal_warning_recovery = 1; + if (of_get_property(np, "st,thermal-warning-adjustment", NULL)) + pdata->thermal_warning_adjustment = 1; + if (of_get_property(np, "st,fault-detect-recovery", NULL)) + pdata->fault_detect_recovery = 1; + + pdata->ffx_power_output_mode = STA350_FFX_PM_VARIABLE_DROP_COMP; + if (!of_property_read_string(np, "st,ffx-power-output-mode", + &ffx_power_mode)) { + int i, mode = -EINVAL; + + for (i = 0; i < ARRAY_SIZE(sta350_ffx_modes); i++) + if (!strcasecmp(ffx_power_mode, sta350_ffx_modes[i])) + mode = i; + + if (mode < 0) + dev_warn(dev, "Unsupported ffx output mode: %s\n", + ffx_power_mode); + else + pdata->ffx_power_output_mode = mode; + } + + tmp = 140; + of_property_read_u16(np, "st,drop-compensation-ns", &tmp); + pdata->drop_compensation_ns = clamp_t(u16, tmp, 0, 300) / 20; + + if (of_get_property(np, "st,overcurrent-warning-adjustment", NULL)) + pdata->oc_warning_adjustment = 1; + + /* CONFE */ + if (of_get_property(np, "st,max-power-use-mpcc", NULL)) + pdata->max_power_use_mpcc = 1; + + if (of_get_property(np, "st,max-power-correction", NULL)) + pdata->max_power_correction = 1; + + if (of_get_property(np, "st,am-reduction-mode", NULL)) + pdata->am_reduction_mode = 1; + + if (of_get_property(np, "st,odd-pwm-speed-mode", NULL)) + pdata->odd_pwm_speed_mode = 1; + + if (of_get_property(np, "st,distortion-compensation", NULL)) + pdata->distortion_compensation = 1; + + /* CONFF */ + if (of_get_property(np, "st,invalid-input-detect-mute", NULL)) + pdata->invalid_input_detect_mute = 1; + + sta350->pdata = pdata; + + return 0; +} +#endif + +static int sta350_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct sta350_priv *sta350; + int ret, i; + + sta350 = devm_kzalloc(&i2c->dev, sizeof(struct sta350_priv), + GFP_KERNEL); + if (!sta350) + return -ENOMEM; + + sta350->pdata = dev_get_platdata(&i2c->dev); + sta350->gpio_nreset = -EINVAL; + sta350->gpio_power_down = -EINVAL; + + mutex_init(&sta350->coeff_lock); + +#ifdef CONFIG_OF + ret = sta350_probe_dt(&i2c->dev, sta350); + if (ret < 0) + return ret; +#endif + + /* GPIOs */ + if (gpio_is_valid(sta350->gpio_nreset)) { + ret = devm_gpio_request_one(&i2c->dev, sta350->gpio_nreset, + GPIOF_OUT_INIT_HIGH, + "ST350 Reset"); + if (ret < 0) + return ret; + } + + if (gpio_is_valid(sta350->gpio_power_down)) { + ret = devm_gpio_request_one(&i2c->dev, sta350->gpio_power_down, + GPIOF_OUT_INIT_HIGH, + "ST350 Power-Down"); + if (ret < 0) + return ret; + } + + /* regulators */ + for (i = 0; i < ARRAY_SIZE(sta350->supplies); i++) + sta350->supplies[i].supply = sta350_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + sta350->regmap = devm_regmap_init_i2c(i2c, &sta350_regmap); + if (IS_ERR(sta350->regmap)) { + ret = PTR_ERR(sta350->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, sta350); + + ret = snd_soc_register_codec(&i2c->dev, &sta350_codec, &sta350_dai, 1); + if (ret < 0) + dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); + + return ret; +} + +static int sta350_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id sta350_i2c_id[] = { + { "sta350", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta350_i2c_id); + +static struct i2c_driver sta350_i2c_driver = { + .driver = { + .name = "sta350", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(st350_dt_ids), + }, + .probe = sta350_i2c_probe, + .remove = sta350_i2c_remove, + .id_table = sta350_i2c_id, +}; + +module_i2c_driver(sta350_i2c_driver); + +MODULE_DESCRIPTION("ASoC STA350 driver"); +MODULE_AUTHOR("Sven Brandau <info(a)brandau.biz>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sta350.h b/sound/soc/codecs/sta350.h new file mode 100644 index 0000000..c3248f0 --- /dev/null +++ b/sound/soc/codecs/sta350.h @@ -0,0 +1,228 @@ +/* + * Codec driver for ST STA350 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * based on code from: + * Raumfeld GmbH + * Johannes Stezenbach <js(a)sig21.net> + * Wolfson Microelectronics PLC. + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#ifndef _ASOC_STA_350_H +#define _ASOC_STA_350_H + +/* STA50 register addresses */ + +#define STA350_REGISTER_COUNT 0x4D +#define STA350_COEF_COUNT 62 + +#define STA350_CONFA 0x00 +#define STA350_CONFB 0x01 +#define STA350_CONFC 0x02 +#define STA350_CONFD 0x03 +#define STA350_CONFE 0x04 +#define STA350_CONFF 0x05 +#define STA350_MMUTE 0x06 +#define STA350_MVOL 0x07 +#define STA350_C1VOL 0x08 +#define STA350_C2VOL 0x09 +#define STA350_C3VOL 0x0a +#define STA350_AUTO1 0x0b +#define STA350_AUTO2 0x0c +#define STA350_AUTO3 0x0d +#define STA350_C1CFG 0x0e +#define STA350_C2CFG 0x0f +#define STA350_C3CFG 0x10 +#define STA350_TONE 0x11 +#define STA350_L1AR 0x12 +#define STA350_L1ATRT 0x13 +#define STA350_L2AR 0x14 +#define STA350_L2ATRT 0x15 +#define STA350_CFADDR2 0x16 +#define STA350_B1CF1 0x17 +#define STA350_B1CF2 0x18 +#define STA350_B1CF3 0x19 +#define STA350_B2CF1 0x1a +#define STA350_B2CF2 0x1b +#define STA350_B2CF3 0x1c +#define STA350_A1CF1 0x1d +#define STA350_A1CF2 0x1e +#define STA350_A1CF3 0x1f +#define STA350_A2CF1 0x20 +#define STA350_A2CF2 0x21 +#define STA350_A2CF3 0x22 +#define STA350_B0CF1 0x23 +#define STA350_B0CF2 0x24 +#define STA350_B0CF3 0x25 +#define STA350_CFUD 0x26 +#define STA350_MPCC1 0x27 +#define STA350_MPCC2 0x28 +#define STA350_DCC1 0x29 +#define STA350_DCC2 0x2a +#define STA350_FDRC1 0x2b +#define STA350_FDRC2 0x2c +#define STA350_STATUS 0x2d +/* reserved: 0x2d - 0x30 */ +#define STA350_EQCFG 0x31 +#define STA350_EATH1 0x32 +#define STA350_ERTH1 0x33 +#define STA350_EATH2 0x34 +#define STA350_ERTH2 0x35 +#define STA350_CONFX 0x36 +#define STA350_SVCA 0x37 +#define STA350_SVCB 0x38 +#define STA350_RMS0A 0x39 +#define STA350_RMS0B 0x3a +#define STA350_RMS0C 0x3b +#define STA350_RMS1A 0x3c +#define STA350_RMS1B 0x3d +#define STA350_RMS1C 0x3e +#define STA350_EVOLRES 0x3f +/* reserved: 0x40 - 0x47 */ +#define STA350_NSHAPE 0x48 +#define STA350_CTXB4B1 0x49 +#define STA350_CTXB7B5 0x4a +#define STA350_MISC1 0x4b +#define STA350_MISC2 0x4c + +/* 0x00 CONFA */ +#define STA350_CONFA_MCS_MASK 0x03 +#define STA350_CONFA_MCS_SHIFT 0 +#define STA350_CONFA_IR_MASK 0x18 +#define STA350_CONFA_IR_SHIFT 3 +#define STA350_CONFA_TWRB BIT(5) +#define STA350_CONFA_TWAB BIT(6) +#define STA350_CONFA_FDRB BIT(7) + +/* 0x01 CONFB */ +#define STA350_CONFB_SAI_MASK 0x0f +#define STA350_CONFB_SAI_SHIFT 0 +#define STA350_CONFB_SAIFB BIT(4) +#define STA350_CONFB_DSCKE BIT(5) +#define STA350_CONFB_C1IM BIT(6) +#define STA350_CONFB_C2IM BIT(7) + +/* 0x02 CONFC */ +#define STA350_CONFC_OM_MASK 0x03 +#define STA350_CONFC_OM_SHIFT 0 +#define STA350_CONFC_CSZ_MASK 0x3c +#define STA350_CONFC_CSZ_SHIFT 2 +#define STA350_CONFC_OCRB BIT(7) + +/* 0x03 CONFD */ +#define STA350_CONFD_HPB_SHIFT 0 +#define STA350_CONFD_DEMP_SHIFT 1 +#define STA350_CONFD_DSPB_SHIFT 2 +#define STA350_CONFD_PSL_SHIFT 3 +#define STA350_CONFD_BQL_SHIFT 4 +#define STA350_CONFD_DRC_SHIFT 5 +#define STA350_CONFD_ZDE_SHIFT 6 +#define STA350_CONFD_SME_SHIFT 7 + +/* 0x04 CONFE */ +#define STA350_CONFE_MPCV BIT(0) +#define STA350_CONFE_MPCV_SHIFT 0 +#define STA350_CONFE_MPC BIT(1) +#define STA350_CONFE_MPC_SHIFT 1 +#define STA350_CONFE_NSBW BIT(2) +#define STA350_CONFE_NSBW_SHIFT 2 +#define STA350_CONFE_AME BIT(3) +#define STA350_CONFE_AME_SHIFT 3 +#define STA350_CONFE_PWMS BIT(4) +#define STA350_CONFE_PWMS_SHIFT 4 +#define STA350_CONFE_DCCV BIT(5) +#define STA350_CONFE_DCCV_SHIFT 5 +#define STA350_CONFE_ZCE BIT(6) +#define STA350_CONFE_ZCE_SHIFT 6 +#define STA350_CONFE_SVE BIT(7) +#define STA350_CONFE_SVE_SHIFT 7 + +/* 0x05 CONFF */ +#define STA350_CONFF_OCFG_MASK 0x03 +#define STA350_CONFF_OCFG_SHIFT 0 +#define STA350_CONFF_IDE BIT(2) +#define STA350_CONFF_BCLE BIT(3) +#define STA350_CONFF_LDTE BIT(4) +#define STA350_CONFF_ECLE BIT(5) +#define STA350_CONFF_PWDN BIT(6) +#define STA350_CONFF_EAPD BIT(7) + +/* 0x06 MMUTE */ +#define STA350_MMUTE_MMUTE 0x01 +#define STA350_MMUTE_MMUTE_SHIFT 0 +#define STA350_MMUTE_C1M 0x02 +#define STA350_MMUTE_C1M_SHIFT 1 +#define STA350_MMUTE_C2M 0x04 +#define STA350_MMUTE_C2M_SHIFT 2 +#define STA350_MMUTE_C3M 0x08 +#define STA350_MMUTE_C3M_SHIFT 3 +#define STA350_MMUTE_LOC_MASK 0xC0 +#define STA350_MMUTE_LOC_SHIFT 6 + +/* 0x0b AUTO1 */ +#define STA350_AUTO1_AMGC_MASK 0x30 +#define STA350_AUTO1_AMGC_SHIFT 4 + +/* 0x0c AUTO2 */ +#define STA350_AUTO2_AMAME 0x01 +#define STA350_AUTO2_AMAM_MASK 0x0e +#define STA350_AUTO2_AMAM_SHIFT 1 +#define STA350_AUTO2_XO_MASK 0xf0 +#define STA350_AUTO2_XO_SHIFT 4 + +/* 0x0d AUTO3 */ +#define STA350_AUTO3_PEQ_MASK 0x1f +#define STA350_AUTO3_PEQ_SHIFT 0 + +/* 0x0e 0x0f 0x10 CxCFG */ +#define STA350_CxCFG_TCB_SHIFT 0 +#define STA350_CxCFG_EQBP_SHIFT 1 +#define STA350_CxCFG_VBP_SHIFT 2 +#define STA350_CxCFG_BO_SHIFT 3 +#define STA350_CxCFG_LS_SHIFT 4 +#define STA350_CxCFG_OM_MASK 0xc0 +#define STA350_CxCFG_OM_SHIFT 6 + +/* 0x11 TONE */ +#define STA350_TONE_BTC_SHIFT 0 +#define STA350_TONE_TTC_SHIFT 4 + +/* 0x12 0x13 0x14 0x15 limiter attack/release */ +#define STA350_LxA_SHIFT 0 +#define STA350_LxR_SHIFT 4 + +/* 0x26 CFUD */ +#define STA350_CFUD_W1 0x01 +#define STA350_CFUD_WA 0x02 +#define STA350_CFUD_R1 0x04 +#define STA350_CFUD_RA 0x08 + + +/* biquad filter coefficient table offsets */ +#define STA350_C1_BQ_BASE 0 +#define STA350_C2_BQ_BASE 20 +#define STA350_CH_BQ_NUM 4 +#define STA350_BQ_NUM_COEF 5 +#define STA350_XO_HP_BQ_BASE 40 +#define STA350_XO_LP_BQ_BASE 45 +#define STA350_C1_PRESCALE 50 +#define STA350_C2_PRESCALE 51 +#define STA350_C1_POSTSCALE 52 +#define STA350_C2_POSTSCALE 53 +#define STA350_C3_POSTSCALE 54 +#define STA350_TW_POSTSCALE 55 +#define STA350_C1_MIX1 56 +#define STA350_C1_MIX2 57 +#define STA350_C2_MIX1 58 +#define STA350_C2_MIX2 59 +#define STA350_C3_MIX1 60 +#define STA350_C3_MIX2 61 + +#endif /* _ASOC_STA_350_H */ -- 1.8.5.3

3 2

[alsa-devel] [PATCH 0/2] ASoC: davinci-mcasp: Correct start sequences for TX/RX
by Peter Ujfalusi 28 Mar '14

28 Mar '14

Hi, The startup sequence for TX and RX was not correct, not following the programming sequence from the TRM. This could cause initial channel swap when starting TX. The change has been tested on AM335x and AM437x but it should be valid for all devices. Regards, Peter --- Peter Ujfalusi (2): ASoC: davinci-mcasp: Correct TX start sequence ASoC: davinci-mcasp: Correct RX start sequence sound/soc/davinci/davinci-mcasp.c | 37 +++++++++++++------------------------ sound/soc/davinci/davinci-mcasp.h | 6 ++++++ 2 files changed, 19 insertions(+), 24 deletions(-) -- 1.9.1

1 3

[alsa-devel] [PATCH v2] ASoC: fsl_sai: Add isr to deal with error flag
by Nicolin Chen 28 Mar '14

28 Mar '14

It's quite cricial to clear error flags because SAI might hang if getting FIFO underrun during playback (I haven't confirmed the same issue on Rx overflow though). So this patch enables those irq and adds isr() to clear the flags so as to keep playback entirely safe. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v2: * Mask the active flags only for the following handler. * Reset FIFO for FIFO underrun/overflow cases. * Enable two error flags only as default. * Use dev_warn for two error flags. * Only clear those W1C bits. sound/soc/fsl/fsl_sai.c | 85 +++++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 15 +++++++++ 2 files changed, 97 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index c4a4231..0bc98bb 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -23,6 +23,71 @@ #include "fsl_sai.h" +#define FSL_SAI_FLAGS (FSL_SAI_CSR_SEIE |\ + FSL_SAI_CSR_FEIE) + +static irqreturn_t fsl_sai_isr(int irq, void *devid) +{ + struct fsl_sai *sai = (struct fsl_sai *)devid; + struct device *dev = &sai->pdev->dev; + u32 xcsr, mask; + + /* Only handle those what we enabled */ + mask = (FSL_SAI_FLAGS >> FSL_SAI_CSR_xIE_SHIFT) << FSL_SAI_CSR_xF_SHIFT; + + /* Tx IRQ */ + regmap_read(sai->regmap, FSL_SAI_TCSR, &xcsr); + xcsr &= mask; + + if (xcsr & FSL_SAI_CSR_WSF) + dev_dbg(dev, "isr: Start of Tx word detected\n"); + + if (xcsr & FSL_SAI_CSR_SEF) + dev_warn(dev, "isr: Tx Frame sync error detected\n"); + + if (xcsr & FSL_SAI_CSR_FEF) { + dev_warn(dev, "isr: Transmit underrun detected\n"); + /* FIFO reset for safety */ + xcsr |= FSL_SAI_CSR_FR; + } + + if (xcsr & FSL_SAI_CSR_FWF) + dev_dbg(dev, "isr: Enabled transmit FIFO is empty\n"); + + if (xcsr & FSL_SAI_CSR_FRF) + dev_dbg(dev, "isr: Transmit FIFO watermark has been reached\n"); + + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, + FSL_SAI_CSR_xF_W_MASK | FSL_SAI_CSR_FR, xcsr); + + /* Rx IRQ */ + regmap_read(sai->regmap, FSL_SAI_RCSR, &xcsr); + xcsr &= mask; + + if (xcsr & FSL_SAI_CSR_WSF) + dev_dbg(dev, "isr: Start of Rx word detected\n"); + + if (xcsr & FSL_SAI_CSR_SEF) + dev_warn(dev, "isr: Rx Frame sync error detected\n"); + + if (xcsr & FSL_SAI_CSR_FEF) { + dev_warn(dev, "isr: Receive overflow detected\n"); + /* FIFO reset for safety */ + xcsr |= FSL_SAI_CSR_FR; + } + + if (xcsr & FSL_SAI_CSR_FWF) + dev_dbg(dev, "isr: Enabled receive FIFO is full\n"); + + if (xcsr & FSL_SAI_CSR_FRF) + dev_dbg(dev, "isr: Receive FIFO watermark has been reached\n"); + + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, + FSL_SAI_CSR_xF_W_MASK | FSL_SAI_CSR_FR, xcsr); + + return IRQ_HANDLED; +} + static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { @@ -373,8 +438,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, FSL_SAI_FLAGS); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, FSL_SAI_FLAGS); regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_TX * 2); regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK, @@ -490,12 +555,14 @@ static int fsl_sai_probe(struct platform_device *pdev) struct fsl_sai *sai; struct resource *res; void __iomem *base; - int ret; + int irq, ret; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) return -ENOMEM; + sai->pdev = pdev; + sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); if (sai->big_endian_regs) fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; @@ -514,6 +581,18 @@ static int fsl_sai_probe(struct platform_device *pdev) return PTR_ERR(sai->regmap); } + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + return irq; + } + + ret = devm_request_irq(&pdev->dev, irq, fsl_sai_isr, 0, np->name, sai); + if (ret) { + dev_err(&pdev->dev, "failed to claim irq %u\n", irq); + return ret; + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index e432260..a264185 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -37,7 +37,21 @@ /* SAI Transmit/Recieve Control Register */ #define FSL_SAI_CSR_TERE BIT(31) +#define FSL_SAI_CSR_FR BIT(25) +#define FSL_SAI_CSR_xF_SHIFT 16 +#define FSL_SAI_CSR_xF_W_SHIFT 18 +#define FSL_SAI_CSR_xF_MASK (0x1f << FSL_SAI_CSR_xF_SHIFT) +#define FSL_SAI_CSR_xF_W_MASK (0x7 << FSL_SAI_CSR_xF_W_SHIFT) +#define FSL_SAI_CSR_WSF BIT(20) +#define FSL_SAI_CSR_SEF BIT(19) +#define FSL_SAI_CSR_FEF BIT(18) #define FSL_SAI_CSR_FWF BIT(17) +#define FSL_SAI_CSR_FRF BIT(16) +#define FSL_SAI_CSR_xIE_SHIFT 8 +#define FSL_SAI_CSR_WSIE BIT(12) +#define FSL_SAI_CSR_SEIE BIT(11) +#define FSL_SAI_CSR_FEIE BIT(10) +#define FSL_SAI_CSR_FWIE BIT(9) #define FSL_SAI_CSR_FRIE BIT(8) #define FSL_SAI_CSR_FRDE BIT(0) @@ -99,6 +113,7 @@ #define FSL_SAI_MAXBURST_RX 6 struct fsl_sai { + struct platform_device *pdev; struct regmap *regmap; bool big_endian_regs; -- 1.8.4

2 2

[alsa-devel] [PATCH] sound: rcar: use fast_io for bus
by Ben Dooks 28 Mar '14

28 Mar '14

The rcar sound driver uses regmap to access registers in various parts of the block and uses regmap to manage mappings. The regmap is created without fast_io set, which means it locks with a mutex rather than a lighter-spinlock. The use of the mutex lock causes issues when the IRQ handler is entered as the code needs to read/write register values and thus with lock debugging enabled the system outputs a number of warnings such as: BUG: sleeping function called from invalid context at rnel/locking/mutex.c:616 The rcar registers are all connected via APB bus and thus not that slow to access. The fix is to set the fast_io in the code so that regmap creates the regmap structures using the faster spinlock functions. Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> --- sound/soc/sh/rcar/gen.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 9094970..6468962 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -66,6 +66,7 @@ static int rsnd_regmap_read32(void *context, } static struct regmap_bus rsnd_regmap_bus = { + .fast_io = true, .write = rsnd_regmap_write32, .read = rsnd_regmap_read32, .reg_format_endian_default = REGMAP_ENDIAN_NATIVE, -- 1.9.0

1 0

[alsa-devel] [PATCH] ASoC: sta350: Add codec driver
by Daniel Mack 27 Mar '14

27 Mar '14

From: Sven Brandau <brandau(a)gmx.de> The TI STA350 is an integrated 2.1-channel power amplifier that is controllable over I2C. This patch adds an ASoC driver for it. At a glance, this chip is very similar to the STA320 for which a driver already exists. In details, however, the register maps contain subtle differences which made a whole new driver easier to write and maintain. [daniel(a)zonque.org: cleanups, DT property rework, rebased on asoc-next] Signed-off-by: Sven Brandau <brandau(a)gmx.de> --- I'm sending this patch on behalf of Sven Brandau, who wrote the driver, based on earlier bits for the STA320. In case of any pending issue, I will amend and resubmit. Thanks, Daniel .../devicetree/bindings/sound/st,sta350.txt | 103 ++ include/sound/sta350.h | 52 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta350.c | 1277 ++++++++++++++++++++ sound/soc/codecs/sta350.h | 228 ++++ 6 files changed, 1666 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,sta350.txt create mode 100644 include/sound/sta350.h create mode 100644 sound/soc/codecs/sta350.c create mode 100644 sound/soc/codecs/sta350.h diff --git a/Documentation/devicetree/bindings/sound/st,sta350.txt b/Documentation/devicetree/bindings/sound/st,sta350.txt new file mode 100644 index 0000000..ae0630b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,sta350.txt @@ -0,0 +1,103 @@ +STA350 audio CODEC + +The driver for this device only supports I2C. + +Required properties: + + - compatible: "st,sta350" + - reg: the I2C address of the device for I2C + - reset-gpio: a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + + - power-down-gpio: a GPIO spec for the power down pin. If specified, + it will be deasserted before communication to the codec + starts. + +Optional properties: + + - st,output-conf: number, Selects the output configuration: + 0: 2-channel (full-bridge) power, 2-channel data-out + 1: 2 (half-bridge). 1 (full-bridge) on-board power + 2: 2 Channel (Full-Bridge) Power, 1 Channel FFX + 3: 1 Channel Mono-Parallel + If parameter is missing, mode 0 will be enabled. + + - st,ch1-output-mapping: Channel 1 output mapping + - st,ch2-output-mapping: Channel 2 output mapping + - st,ch3-output-mapping: Channel 3 output mapping + 0: Channel 1 + 1: Channel 2 + 2: Channel 3 + If parameter is missing, channel 1 is choosen. + + - st,thermal-warning-recover: + If present, thermal warning recovery is enabled. + + - st,thermal-warning-adjustment: + If present, thermal warning adjustment is enabled. + + - st,fault-detect-recovery: + If present, then fault recovery will be enabled. + + - st,ffx-power-output-mode: string + The FFX power output mode selects how the FFX output timing is + configured. Must be one of these values: + - "drop-compensation" + - "tapered-compensation" + - "full-power-mode" + - "variable-drop-compensation" (default) + + - st,drop-compensation-ns: number + Only required for "st,ffx-power-output-mode" == + "variable-drop-compensation". + Specifies the drop compensation in nanoseconds. + The value must be in the range of 0..300, and only + multiples of 20 are allowed. Default is 140ns. + + - st,overcurrent-warning-adjustment: + If present, overcurrent warning adjustment is enabled. + + - st,max-power-use-mpcc: + If present, then MPCC bits are used for MPC coefficients, + otherwise standard MPC coefficients are used. + + - st,max-power-corr: + If present, power bridge correction for THD reduction near maximum + power output is enabled. + + - st,am-reduction-mode: + If present, FFX mode runs in AM reduction mode, otherwise normal + FFX mode is used. + + - st,odd-pwm-speed-mode: + If present, PWM speed mode run on odd speed mode (341.3 kHz) on all + channels. If not present, normal PWM spped mode (384 kHz) will be used. + + - st,distortion-compensation: + If present, distortion compensation variable uses DCC coefficient. + If not present, preset DC coefficient is used. + + - st,invalid-input-detect-mute: + If not present, automatic invalid input detect mute is enabled. + + + +Example: + +codec: sta350@38 { + compatible = "st,sta350"; + reg = <0x1c>; + reset-gpio = <&gpio1 19 0>; + power-down-gpio = <&gpio1 16 0>; + st,output-conf = <0x3>; // set output to 2-channel + // (full-bridge) power, + // 2-channel data-out + st,ch1-output-mapping = <0>; // set channel 1 output ch 1 + st,ch2-output-mapping = <0>; // set channel 2 output ch 1 + st,ch3-output-mapping = <0>; // set channel 3 output ch 1 + st,max-power-correction; // enables power bridge + // correction for THD reduction + // near maximum power output + st,invalid-input-detect-mute; // mute if no valid digital + // audio signal is provided. +}; diff --git a/include/sound/sta350.h b/include/sound/sta350.h new file mode 100644 index 0000000..3a329810 --- /dev/null +++ b/include/sound/sta350.h @@ -0,0 +1,52 @@ +/* + * Platform data for ST STA350 ASoC codec driver. + * + * Copyright: 2014 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#ifndef __LINUX_SND__STA350_H +#define __LINUX_SND__STA350_H + +#define STA350_OCFG_2CH 0 +#define STA350_OCFG_2_1CH 1 +#define STA350_OCFG_1CH 3 + +#define STA350_OM_CH1 0 +#define STA350_OM_CH2 1 +#define STA350_OM_CH3 2 + +#define STA350_THERMAL_ADJUSTMENT_ENABLE 1 +#define STA350_THERMAL_RECOVERY_ENABLE 2 +#define STA350_FAULT_DETECT_RECOVERY_BYPASS 1 + +#define STA350_FFX_PM_DROP_COMP 0 +#define STA350_FFX_PM_TAPERED_COMP 1 +#define STA350_FFX_PM_FULL_POWER 2 +#define STA350_FFX_PM_VARIABLE_DROP_COMP 3 + + +struct sta350_platform_data { + u8 output_conf; + u8 ch1_output_mapping; + u8 ch2_output_mapping; + u8 ch3_output_mapping; + u8 ffx_power_output_mode; + u8 drop_compensation_ns; + unsigned int thermal_warning_recovery:1; + unsigned int thermal_warning_adjustment:1; + unsigned int fault_detect_recovery:1; + unsigned int oc_warning_adjustment:1; + unsigned int max_power_use_mpcc:1; + unsigned int max_power_correction:1; + unsigned int am_reduction_mode:1; + unsigned int odd_pwm_speed_mode:1; + unsigned int distortion_compensation:1; + unsigned int invalid_input_detect_mute:1; +}; + +#endif /* __LINUX_SND__STA350_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f0e8401..6211889 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -80,6 +80,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SSM2602_SPI if SPI_MASTER select SND_SOC_SSM2602_I2C if I2C select SND_SOC_STA32X if I2C + select SND_SOC_STA350 if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TAS5086 if I2C @@ -435,6 +436,9 @@ config SND_SOC_SSM2602_I2C config SND_SOC_STA32X tristate +config SND_SOC_STA350 + tristate + config SND_SOC_STA529 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3c4d275..efdb4d0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -74,6 +74,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-ssm2602-spi-objs := ssm2602-spi.o snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o snd-soc-sta32x-objs := sta32x.o +snd-soc-sta350-objs := sta350.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o @@ -221,6 +222,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o obj-$(CONFIG_SND_SOC_SSM2602_I2C) += snd-soc-ssm2602-i2c.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o +obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c new file mode 100644 index 0000000..c78d1c5 --- /dev/null +++ b/sound/soc/codecs/sta350.c @@ -0,0 +1,1277 @@ +/* + * Codec driver for ST STA350 2.1-channel high-efficiency digital audio system + * + * Copyright: 2014 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * based on code from: + * Raumfeld GmbH + * Johannes Stezenbach <js(a)sig21.net> + * Wolfson Microelectronics PLC. + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * Freescale Semiconductor, Inc. + * Timur Tabi <timur(a)freescale.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__ + +#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/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.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 <sound/sta350.h> +#include "sta350.h" + +#define STA350_RATES (SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +#define STA350_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + +/* Power-up register defaults */ +static const struct reg_default sta350_regs[] = { + { 0x0, 0x63 }, + { 0x1, 0x80 }, + { 0x2, 0xdf }, + { 0x3, 0x40 }, + { 0x4, 0xc2 }, + { 0x5, 0x5c }, + { 0x6, 0x00 }, + { 0x7, 0xff }, + { 0x8, 0x60 }, + { 0x9, 0x60 }, + { 0xa, 0x60 }, + { 0xb, 0x00 }, + { 0xc, 0x00 }, + { 0xd, 0x00 }, + { 0xe, 0x00 }, + { 0xf, 0x40 }, + { 0x10, 0x80 }, + { 0x11, 0x77 }, + { 0x12, 0x6a }, + { 0x13, 0x69 }, + { 0x14, 0x6a }, + { 0x15, 0x69 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1a, 0x00 }, + { 0x1b, 0x00 }, + { 0x1c, 0x00 }, + { 0x1d, 0x00 }, + { 0x1e, 0x00 }, + { 0x1f, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x2a }, + { 0x28, 0xc0 }, + { 0x29, 0xf3 }, + { 0x2a, 0x33 }, + { 0x2b, 0x00 }, + { 0x2c, 0x0c }, + { 0x31, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x01 }, + { 0x3a, 0xee }, + { 0x3b, 0xff }, + { 0x3c, 0x7e }, + { 0x3d, 0xc0 }, + { 0x3e, 0x26 }, + { 0x3f, 0x00 }, + { 0x48, 0x00 }, + { 0x49, 0x00 }, + { 0x4a, 0x00 }, + { 0x4b, 0x04 }, + { 0x4c, 0x00 }, +}; + +static const struct regmap_range sta350_write_regs_range[] = { + regmap_reg_range(STA350_CONFA, STA350_AUTO2), + regmap_reg_range(STA350_C1CFG, STA350_FDRC2), + regmap_reg_range(STA350_EQCFG, STA350_EVOLRES), + regmap_reg_range(STA350_NSHAPE, STA350_MISC2), +}; + +static const struct regmap_range sta350_read_regs_range[] = { + regmap_reg_range(STA350_CONFA, STA350_AUTO2), + regmap_reg_range(STA350_C1CFG, STA350_STATUS), + regmap_reg_range(STA350_EQCFG, STA350_EVOLRES), + regmap_reg_range(STA350_NSHAPE, STA350_MISC2), +}; + +static const struct regmap_range sta350_volatile_regs_range[] = { + regmap_reg_range(STA350_CFADDR2, STA350_CFUD), + regmap_reg_range(STA350_STATUS, STA350_STATUS), +}; + +static const struct regmap_access_table sta350_write_regs = { + .yes_ranges = sta350_write_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_write_regs_range), +}; + +static const struct regmap_access_table sta350_read_regs = { + .yes_ranges = sta350_read_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_read_regs_range), +}; + +static const struct regmap_access_table sta350_volatile_regs = { + .yes_ranges = sta350_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_volatile_regs_range), +}; + +/* regulator power supply names */ +static const char const *sta350_supply_names[] = { + "vdd-dig", /* digital supply, 3.3V */ + "vdd-pll", /* pll supply, 3.3V */ + "vcc" /* power amp supply, 5V - 26V */ +}; + +/* codec private data */ +struct sta350_priv { + struct regmap *regmap; + struct regulator_bulk_data supplies[ARRAY_SIZE(sta350_supply_names)]; + struct sta350_platform_data *pdata; + + unsigned int mclk; + unsigned int format; + + u32 coef_shadow[STA350_COEF_COUNT]; + int shutdown; + int gpio_nreset; + int gpio_power_down; +}; + +static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1); +static const DECLARE_TLV_DB_SCALE(tone_tlv, -1200, 200, 0); + +static const char const *sta350_drc_ac[] = { + "Anti-Clipping", "Dynamic Range Compression" +}; +static const char const *sta350_auto_gc_mode[] = { + "User", "AC no clipping", "AC limited clipping (10%)", + "DRC nighttime listening mode" +}; +static const char const *sta350_auto_xo_mode[] = { + "User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", + "200Hz", "220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", + "340Hz", "360Hz" +}; +static const char const *sta350_binary_output[] = { + "FFX 3-state output - normal operation", "Binary output" +}; +static const char const *sta350_limiter_select[] = { + "Limiter Disabled", "Limiter #1", "Limiter #2" +}; +static const char const *sta350_limiter_attack_rate[] = { + "3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024", + "0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752", + "0.0645", "0.0564", "0.0501", "0.0451" +}; +static const char const *sta350_limiter_release_rate[] = { + "0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299", + "0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137", + "0.0134", "0.0117", "0.0110", "0.0104" +}; +static const char const *sta350_noise_shaper_type[] = { + "Third order", "Forth order" +}; + +static const unsigned int sta350_limiter_ac_attack_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 7, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(300, 100, 0), +}; + +static const unsigned int sta350_limiter_ac_release_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-2900, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-2000, 0, 0), + 3, 8, TLV_DB_SCALE_ITEM(-1400, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const unsigned int sta350_limiter_drc_attack_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0), + 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0), + 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0), +}; + +static const unsigned int sta350_limiter_drc_release_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 2, TLV_DB_SCALE_ITEM(-3800, 200, 0), + 3, 4, TLV_DB_SCALE_ITEM(-3300, 200, 0), + 5, 12, TLV_DB_SCALE_ITEM(-3000, 200, 0), + 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0), +}; + +static const struct soc_enum sta350_drc_ac_enum = + SOC_ENUM_SINGLE(STA350_CONFD, STA350_CONFD_DRC_SHIFT, + 2, sta350_drc_ac); +static const struct soc_enum sta350_noise_shaper_enum = + SOC_ENUM_SINGLE(STA350_CONFE, STA350_CONFE_NSBW_SHIFT, + 2, sta350_noise_shaper_type); +static const struct soc_enum sta350_auto_gc_enum = + SOC_ENUM_SINGLE(STA350_AUTO1, STA350_AUTO1_AMGC_SHIFT, + 4, sta350_auto_gc_mode); +static const struct soc_enum sta350_auto_xo_enum = + SOC_ENUM_SINGLE(STA350_AUTO2, STA350_AUTO2_XO_SHIFT, + 16, sta350_auto_xo_mode); +static const struct soc_enum sta350_binary_output_ch1_enum = + SOC_ENUM_SINGLE(STA350_C1CFG, STA350_CxCFG_BO_SHIFT, + 2, sta350_binary_output); +static const struct soc_enum sta350_binary_output_ch2_enum = + SOC_ENUM_SINGLE(STA350_C2CFG, STA350_CxCFG_BO_SHIFT, + 2, sta350_binary_output); +static const struct soc_enum sta350_binary_output_ch3_enum = + SOC_ENUM_SINGLE(STA350_C3CFG, STA350_CxCFG_BO_SHIFT, + 2, sta350_binary_output); +static const struct soc_enum sta350_limiter_ch1_enum = + SOC_ENUM_SINGLE(STA350_C1CFG, STA350_CxCFG_LS_SHIFT, + 3, sta350_limiter_select); +static const struct soc_enum sta350_limiter_ch2_enum = + SOC_ENUM_SINGLE(STA350_C2CFG, STA350_CxCFG_LS_SHIFT, + 3, sta350_limiter_select); +static const struct soc_enum sta350_limiter_ch3_enum = + SOC_ENUM_SINGLE(STA350_C3CFG, STA350_CxCFG_LS_SHIFT, + 3, sta350_limiter_select); +static const struct soc_enum sta350_limiter1_attack_rate_enum = + SOC_ENUM_SINGLE(STA350_L1AR, STA350_LxA_SHIFT, + 16, sta350_limiter_attack_rate); +static const struct soc_enum sta350_limiter2_attack_rate_enum = + SOC_ENUM_SINGLE(STA350_L2AR, STA350_LxA_SHIFT, + 16, sta350_limiter_attack_rate); +static const struct soc_enum sta350_limiter1_release_rate_enum = + SOC_ENUM_SINGLE(STA350_L1AR, STA350_LxR_SHIFT, + 16, sta350_limiter_release_rate); +static const struct soc_enum sta350_limiter2_release_rate_enum = + SOC_ENUM_SINGLE(STA350_L2AR, STA350_LxR_SHIFT, + 16, sta350_limiter_release_rate); + +/* + * byte array controls for setting biquad, mixer, scaling coefficients; + * for biquads all five coefficients need to be set in one go, + * mixer and pre/postscale coefs can be set individually; + * each coef is 24bit, the bytes are ordered in the same way + * as given in the STA350 data sheet (big endian; b1, b2, a1, a2, b0) + */ + +static int sta350_coefficient_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int numcoef = kcontrol->private_value >> 16; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = 3 * numcoef; + return 0; +} + +static int sta350_coefficient_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int numcoef = kcontrol->private_value >> 16; + int index = kcontrol->private_value & 0xffff; + unsigned int cfud, val; + int i; + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + /* + * chip documentation does not say if the bits are self clearing, + * so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + + regmap_write(sta350->regmap, STA350_CFADDR2, index); + if (numcoef == 1) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x04); + else if (numcoef == 5) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x08); + else + return -EINVAL; + + for (i = 0; i < 3 * numcoef; i++) { + regmap_read(sta350->regmap, STA350_B1CF1 + i, &val); + ucontrol->value.bytes.data[i] = val; + } + + return 0; +} + +static int sta350_coefficient_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int numcoef = kcontrol->private_value >> 16; + int index = kcontrol->private_value & 0xffff; + unsigned int cfud; + int i; + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + /* + * chip documentation does not say if the bits are self clearing, + * so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + + regmap_write(sta350->regmap, STA350_CFADDR2, index); + for (i = 0; i < numcoef && (index + i < STA350_COEF_COUNT); i++) + sta350->coef_shadow[index + i] = + (ucontrol->value.bytes.data[3 * i] << 16) + | (ucontrol->value.bytes.data[3 * i + 1] << 8) + | (ucontrol->value.bytes.data[3 * i + 2]); + for (i = 0; i < 3 * numcoef; i++) + regmap_write(sta350->regmap, STA350_B1CF1 + i, + ucontrol->value.bytes.data[i]); + if (numcoef == 1) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x01); + else if (numcoef == 5) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x02); + else + return -EINVAL; + + return 0; +} + +static int sta350_sync_coef_shadow(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int cfud; + int i; + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + + for (i = 0; i < STA350_COEF_COUNT; i++) { + regmap_write(sta350->regmap, STA350_CFADDR2, i); + regmap_write(sta350->regmap, STA350_B1CF1, + (sta350->coef_shadow[i] >> 16) & 0xff); + regmap_write(sta350->regmap, STA350_B1CF2, + (sta350->coef_shadow[i] >> 8) & 0xff); + regmap_write(sta350->regmap, STA350_B1CF3, + (sta350->coef_shadow[i]) & 0xff); + /* + * chip documentation does not say if the bits are + * self-clearing, so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x01); + } + return 0; +} + +static int sta350_cache_sync(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int mute; + int rc; + + /* mute during register sync */ + regmap_read(sta350->regmap, STA350_CFUD, &mute); + regmap_write(sta350->regmap, STA350_MMUTE, mute | STA350_MMUTE_MMUTE); + sta350_sync_coef_shadow(codec); + rc = regcache_sync(sta350->regmap); + regmap_write(sta350->regmap, STA350_MMUTE, mute); + return rc; +} + +#define SINGLE_COEF(xname, index) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sta350_coefficient_info, \ + .get = sta350_coefficient_get,\ + .put = sta350_coefficient_put, \ + .private_value = index | (1 << 16) } + +#define BIQUAD_COEFS(xname, index) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sta350_coefficient_info, \ + .get = sta350_coefficient_get,\ + .put = sta350_coefficient_put, \ + .private_value = index | (5 << 16) } + +static const struct snd_kcontrol_new sta350_snd_controls[] = { +SOC_SINGLE_TLV("Master Volume", STA350_MVOL, 0, 0xff, 1, mvol_tlv), +/* VOL */ +SOC_SINGLE_TLV("Ch1 Volume", STA350_C1VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch2 Volume", STA350_C2VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch3 Volume", STA350_C3VOL, 0, 0xff, 1, chvol_tlv), +/* CONFD */ +SOC_SINGLE("High Pass Filter Bypass Switch", + STA350_CONFD, STA350_CONFD_HPB_SHIFT, 1, 1), +SOC_SINGLE("De-emphasis Filter Switch", + STA350_CONFD, STA350_CONFD_DEMP_SHIFT, 1, 0), +SOC_SINGLE("DSP Bypass Switch", + STA350_CONFD, STA350_CONFD_DSPB_SHIFT, 1, 0), +SOC_SINGLE("Post-scale Link Switch", + STA350_CONFD, STA350_CONFD_PSL_SHIFT, 1, 0), +SOC_SINGLE("Biquad Coefficient Link Switch", + STA350_CONFD, STA350_CONFD_BQL_SHIFT, 1, 0), +SOC_ENUM("Compressor/Limiter Switch", sta350_drc_ac_enum), +SOC_ENUM("Noise Shaper Bandwidth", sta350_noise_shaper_enum), +SOC_SINGLE("Zero-detect Mute Enable Switch", + STA350_CONFD, STA350_CONFD_ZDE_SHIFT, 1, 0), +SOC_SINGLE("Submix Mode Switch", + STA350_CONFD, STA350_CONFD_SME_SHIFT, 1, 0), +/* CONFE */ +SOC_SINGLE("Zero Cross Switch", STA350_CONFE, STA350_CONFE_ZCE_SHIFT, 1, 0), +SOC_SINGLE("Soft Ramp Switch", STA350_CONFE, STA350_CONFE_SVE_SHIFT, 1, 0), +/* MUTE */ +SOC_SINGLE("Master Switch", STA350_MMUTE, STA350_MMUTE_MMUTE_SHIFT, 1, 1), +SOC_SINGLE("Ch1 Switch", STA350_MMUTE, STA350_MMUTE_C1M_SHIFT, 1, 1), +SOC_SINGLE("Ch2 Switch", STA350_MMUTE, STA350_MMUTE_C2M_SHIFT, 1, 1), +SOC_SINGLE("Ch3 Switch", STA350_MMUTE, STA350_MMUTE_C3M_SHIFT, 1, 1), +/* AUTOx */ +SOC_ENUM("Automode GC", sta350_auto_gc_enum), +SOC_ENUM("Automode XO", sta350_auto_xo_enum), +/* CxCFG */ +SOC_SINGLE("Ch1 Tone Control Bypass Switch", + STA350_C1CFG, STA350_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Tone Control Bypass Switch", + STA350_C2CFG, STA350_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch1 EQ Bypass Switch", + STA350_C1CFG, STA350_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 EQ Bypass Switch", + STA350_C2CFG, STA350_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch1 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch3 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_ENUM("Ch1 Binary Output Select", sta350_binary_output_ch1_enum), +SOC_ENUM("Ch2 Binary Output Select", sta350_binary_output_ch2_enum), +SOC_ENUM("Ch3 Binary Output Select", sta350_binary_output_ch3_enum), +SOC_ENUM("Ch1 Limiter Select", sta350_limiter_ch1_enum), +SOC_ENUM("Ch2 Limiter Select", sta350_limiter_ch2_enum), +SOC_ENUM("Ch3 Limiter Select", sta350_limiter_ch3_enum), +/* TONE */ +SOC_SINGLE_RANGE_TLV("Bass Tone Control Volume", + STA350_TONE, STA350_TONE_BTC_SHIFT, 1, 13, 0, tone_tlv), +SOC_SINGLE_RANGE_TLV("Treble Tone Control Volume", + STA350_TONE, STA350_TONE_TTC_SHIFT, 1, 13, 0, tone_tlv), +SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta350_limiter1_attack_rate_enum), +SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta350_limiter2_attack_rate_enum), +SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta350_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta350_limiter2_release_rate_enum), + +/* + * depending on mode, the attack/release thresholds have + * two different enum definitions; provide both + */ +SOC_SINGLE_TLV("Limiter1 Attack Threshold (AC Mode)", + STA350_L1ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (AC Mode)", + STA350_L2ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (AC Mode)", + STA350_L1ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (AC Mode)", + STA350_L2ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter1 Attack Threshold (DRC Mode)", + STA350_L1ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (DRC Mode)", + STA350_L2ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (DRC Mode)", + STA350_L1ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_drc_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (DRC Mode)", + STA350_L2ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_drc_release_tlv), + +BIQUAD_COEFS("Ch1 - Biquad 1", 0), +BIQUAD_COEFS("Ch1 - Biquad 2", 5), +BIQUAD_COEFS("Ch1 - Biquad 3", 10), +BIQUAD_COEFS("Ch1 - Biquad 4", 15), +BIQUAD_COEFS("Ch2 - Biquad 1", 20), +BIQUAD_COEFS("Ch2 - Biquad 2", 25), +BIQUAD_COEFS("Ch2 - Biquad 3", 30), +BIQUAD_COEFS("Ch2 - Biquad 4", 35), +BIQUAD_COEFS("High-pass", 40), +BIQUAD_COEFS("Low-pass", 45), +SINGLE_COEF("Ch1 - Prescale", 50), +SINGLE_COEF("Ch2 - Prescale", 51), +SINGLE_COEF("Ch1 - Postscale", 52), +SINGLE_COEF("Ch2 - Postscale", 53), +SINGLE_COEF("Ch3 - Postscale", 54), +SINGLE_COEF("Thermal warning - Postscale", 55), +SINGLE_COEF("Ch1 - Mix 1", 56), +SINGLE_COEF("Ch1 - Mix 2", 57), +SINGLE_COEF("Ch2 - Mix 1", 58), +SINGLE_COEF("Ch2 - Mix 2", 59), +SINGLE_COEF("Ch3 - Mix 1", 60), +SINGLE_COEF("Ch3 - Mix 2", 61), +}; + +static const struct snd_soc_dapm_widget sta350_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LEFT"), +SND_SOC_DAPM_OUTPUT("RIGHT"), +SND_SOC_DAPM_OUTPUT("SUB"), +}; + +static const struct snd_soc_dapm_route sta350_dapm_routes[] = { + { "LEFT", NULL, "DAC" }, + { "RIGHT", NULL, "DAC" }, + { "SUB", NULL, "DAC" }, +}; + +/* MCLK interpolation ratio per fs */ +static struct { + int fs; + int ir; +} interpolation_ratios[] = { + { 32000, 0 }, + { 44100, 0 }, + { 48000, 0 }, + { 88200, 1 }, + { 96000, 1 }, + { 176400, 2 }, + { 192000, 2 }, +}; + +/* MCLK to fs clock ratios */ +static int mcs_ratio_table[3][6] = { + { 768, 512, 384, 256, 128, 576 }, + { 384, 256, 192, 128, 64, 0 }, + { 192, 128, 96, 64, 32, 0 }, +}; + +/** + * sta350_set_dai_sysclk - configure MCLK + * @codec_dai: the codec DAI + * @clk_id: the clock ID (ignored) + * @freq: the MCLK input frequency + * @dir: the clock direction (ignored) + * + * The value of MCLK is used to determine which sample rates are supported + * by the STA350, based on the mcs_ratio_table. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + */ +static int sta350_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + + pr_debug("mclk=%u\n", freq); + sta350->mclk = freq; + + return 0; +} + +/** + * sta350_set_dai_fmt - configure the codec for the selected audio format + * @codec_dai: the codec DAI + * @fmt: a SND_SOC_DAIFMT_x value indicating the data format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + */ +static int sta350_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int confb; + + regmap_read(sta350->regmap, STA350_CONFB, &confb); + + pr_debug("\n"); + confb &= ~(STA350_CONFB_C1IM | STA350_CONFB_C2IM); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + sta350->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + confb |= STA350_CONFB_C2IM; + break; + case SND_SOC_DAIFMT_NB_IF: + confb |= STA350_CONFB_C1IM; + break; + default: + return -EINVAL; + } + + regmap_write(sta350->regmap, STA350_CONFB, confb); + return 0; +} + +/** + * sta350_hw_params - program the STA350 with the given hardware parameters. + * @substream: the audio stream + * @params: the hardware parameters to set + * @dai: the SOC DAI (ignored) + * + * This function programs the hardware with the values provided. + * Specifically, the sample rate and the data format. + */ +static int sta350_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 sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int i, mcs = -EINVAL, ir = -EINVAL; + unsigned int confa, confb; + unsigned int rate, ratio; + + if (!sta350->mclk) { + dev_err(codec->dev, + "sta350->mclk is unset. Unable to determine ratio\n"); + return -EIO; + } + + rate = params_rate(params); + ratio = sta350->mclk / rate; + pr_debug("rate: %u, ratio: %u\n", rate, ratio); + + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) + if (interpolation_ratios[i].fs == rate) { + ir = interpolation_ratios[i].ir; + break; + } + if (ir < 0) { + dev_err(codec->dev, "Unsupported samplerate: %u\n", rate); + return -EINVAL; + } + + for (i = 0; i < 6; i++) + if (mcs_ratio_table[ir][i] == ratio) { + mcs = i; + break; + } + if (mcs < 0) { + dev_err(codec->dev, "Unresolvable ratio: %u\n", ratio); + return -EINVAL; + } + + regmap_read(sta350->regmap, STA350_CONFA, &confa); + confa &= ~(STA350_CONFA_MCS_MASK | STA350_CONFA_IR_MASK); + confa |= (ir << STA350_CONFA_IR_SHIFT) | + (mcs << STA350_CONFA_MCS_SHIFT); + + regmap_read(sta350->regmap, STA350_CONFB, &confb); + confb &= ~(STA350_CONFB_SAI_MASK | STA350_CONFB_SAIFB); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_BE: + case SNDRV_PCM_FORMAT_S24_3LE: + case SNDRV_PCM_FORMAT_S24_3BE: + pr_debug("24bit\n"); + /* fall through */ + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S32_BE: + pr_debug("24bit or 32bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x1; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x2; + break; + } + + break; + case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S20_3BE: + pr_debug("20bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x4; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x5; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x6; + break; + } + + break; + case SNDRV_PCM_FORMAT_S18_3LE: + case SNDRV_PCM_FORMAT_S18_3BE: + pr_debug("18bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x8; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x9; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xa; + break; + } + + break; + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + pr_debug("16bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0xd; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xe; + break; + } + + break; + default: + return -EINVAL; + } + + regmap_write(sta350->regmap, STA350_CONFA, confa); + regmap_write(sta350->regmap, STA350_CONFB, confb); + return 0; +} + +static int sta350_startup_sequence(struct sta350_priv *sta350) +{ + if (gpio_is_valid(sta350->gpio_power_down)) + gpio_set_value(sta350->gpio_power_down, 1); + + if (gpio_is_valid(sta350->gpio_nreset)) { + gpio_set_value(sta350->gpio_nreset, 1); + mdelay(1); + gpio_set_value(sta350->gpio_nreset, 0); + mdelay(1); + gpio_set_value(sta350->gpio_nreset, 1); + mdelay(1); + } + + return 0; +} + +/** + * sta350_set_bias_level - DAPM callback + * @codec: the codec device + * @level: DAPM power level + * + * This is called by ALSA to put the codec into low power mode + * or to wake it up. If the codec is powered off completely + * all registers must be restored after power on. + */ +static int sta350_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int ret; + + pr_debug("level = %d\n", level); + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Full power on */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, + STA350_CONFF_PWDN | STA350_CONFF_EAPD); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable( + ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + sta350_startup_sequence(sta350); + sta350_cache_sync(codec); + } + + /* Power down */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, + 0); + + break; + + case SND_SOC_BIAS_OFF: + /* The chip runs through the power down sequence for us. */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, 0); + /* power down: low */ + if (gpio_is_valid(sta350->gpio_power_down)) + gpio_set_value(sta350->gpio_power_down, 0); + + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), + sta350->supplies); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const struct snd_soc_dai_ops sta350_dai_ops = { + .hw_params = sta350_hw_params, + .set_sysclk = sta350_set_dai_sysclk, + .set_fmt = sta350_set_dai_fmt, +}; + +static struct snd_soc_dai_driver sta350_dai = { + .name = "sta350-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA350_RATES, + .formats = STA350_FORMATS, + }, + .ops = &sta350_dai_ops, +}; + +#ifdef CONFIG_PM +static int sta350_suspend(struct snd_soc_codec *codec) +{ + sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int sta350_resume(struct snd_soc_codec *codec) +{ + sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define sta350_suspend NULL +#define sta350_resume NULL +#endif + +static int sta350_probe(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + struct sta350_platform_data *pdata = sta350->pdata; + int i, ret = 0, thermal = 0; + + if (gpio_is_valid(sta350->gpio_nreset)) + if (devm_gpio_request_one(codec->dev, sta350->gpio_nreset, + GPIOF_OUT_INIT_HIGH, + "ST350 Reset")) + sta350->gpio_nreset = -EINVAL; + + if (gpio_is_valid(sta350->gpio_power_down)) + if (devm_gpio_request_one(codec->dev, sta350->gpio_power_down, + GPIOF_OUT_INIT_HIGH, + "ST350 Power-Down")) + sta350->gpio_power_down = -EINVAL; + + ret = sta350_startup_sequence(sta350); + if (ret < 0) { + dev_err(codec->dev, "Failed to startup device\n"); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + /* + * Tell ASoC what kind of I/O to use to read the registers. ASoC will + * then do the I2C transactions itself. + */ + codec->control_data = sta350->regmap; + + /* + * Chip documentation explicitly requires that the reset values + * of reserved register bits are left untouched. + * Write the register default value to cache for reserved registers, + * so the write to the these registers are suppressed by the cache + * restore code when it skips writes of default registers. + */ + regcache_cache_only(sta350->regmap, true); + regmap_write(sta350->regmap, STA350_CONFC, 0xc2); + regmap_write(sta350->regmap, STA350_CONFE, 0xc2); + regmap_write(sta350->regmap, STA350_CONFF, 0x5c); + regmap_write(sta350->regmap, STA350_MMUTE, 0x10); + regmap_write(sta350->regmap, STA350_AUTO1, 0x60); + regmap_write(sta350->regmap, STA350_AUTO3, 0x00); + regmap_write(sta350->regmap, STA350_C3CFG, 0x40); + regcache_cache_only(sta350->regmap, false); + + /* CONFA */ + if (!pdata->thermal_warning_recovery) + thermal |= STA350_CONFA_TWAB; + if (!pdata->thermal_warning_adjustment) + thermal |= STA350_CONFA_TWRB; + if (!pdata->fault_detect_recovery) + thermal |= STA350_CONFA_FDRB; + regmap_update_bits(sta350->regmap, STA350_CONFA, + STA350_CONFA_TWAB | STA350_CONFA_TWRB | + STA350_CONFA_FDRB, + thermal); + + /* CONFC */ + regmap_update_bits(sta350->regmap, STA350_CONFC, + STA350_CONFC_OM_MASK, + pdata->ffx_power_output_mode + << STA350_CONFC_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_CONFC, + STA350_CONFC_CSZ_MASK, + pdata->drop_compensation_ns + << STA350_CONFC_CSZ_SHIFT); + regmap_update_bits(sta350->regmap, + STA350_CONFC, + STA350_CONFC_OCRB, + pdata->oc_warning_adjustment ? + STA350_CONFC_OCRB : 0); + + /* CONFE */ + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_MPCV, + pdata->max_power_use_mpcc ? + STA350_CONFE_MPCV : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_MPC, + pdata->max_power_correction ? + STA350_CONFE_MPC : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_AME, + pdata->am_reduction_mode ? + STA350_CONFE_AME : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_PWMS, + pdata->odd_pwm_speed_mode ? + STA350_CONFE_PWMS : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_DCCV, + pdata->distortion_compensation ? + STA350_CONFE_DCCV : 0); + /* CONFF */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_IDE, + pdata->invalid_input_detect_mute ? + STA350_CONFF_IDE : 0); + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_OCFG_MASK, + pdata->output_conf + << STA350_CONFF_OCFG_SHIFT); + + /* channel to output mapping */ + regmap_update_bits(sta350->regmap, STA350_C1CFG, + STA350_CxCFG_OM_MASK, + pdata->ch1_output_mapping + << STA350_CxCFG_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_C2CFG, + STA350_CxCFG_OM_MASK, + pdata->ch2_output_mapping + << STA350_CxCFG_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_C3CFG, + STA350_CxCFG_OM_MASK, + pdata->ch3_output_mapping + << STA350_CxCFG_OM_SHIFT); + + /* initialize coefficient shadow RAM with reset values */ + for (i = 4; i <= 49; i += 5) + sta350->coef_shadow[i] = 0x400000; + for (i = 50; i <= 54; i++) + sta350->coef_shadow[i] = 0x7fffff; + sta350->coef_shadow[55] = 0x5a9df7; + sta350->coef_shadow[56] = 0x7fffff; + sta350->coef_shadow[59] = 0x7fffff; + sta350->coef_shadow[60] = 0x400000; + sta350->coef_shadow[61] = 0x400000; + + sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies); + + return 0; +} + +static int sta350_remove(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(sta350->gpio_nreset)) + gpio_set_value(sta350->gpio_nreset, 0); + + if (gpio_is_valid(sta350->gpio_power_down)) + gpio_set_value(sta350->gpio_power_down, 0); + + sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies); + + return 0; +} + +static const struct snd_soc_codec_driver sta350_codec = { + .probe = sta350_probe, + .remove = sta350_remove, + .suspend = sta350_suspend, + .resume = sta350_resume, + .set_bias_level = sta350_set_bias_level, + .controls = sta350_snd_controls, + .num_controls = ARRAY_SIZE(sta350_snd_controls), + .dapm_widgets = sta350_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sta350_dapm_widgets), + .dapm_routes = sta350_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sta350_dapm_routes), +}; + +static const struct regmap_config sta350_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = STA350_MISC2, + .reg_defaults = sta350_regs, + .num_reg_defaults = ARRAY_SIZE(sta350_regs), + .cache_type = REGCACHE_RBTREE, + .wr_table = &sta350_write_regs, + .rd_table = &sta350_read_regs, + .volatile_table = &sta350_volatile_regs, +}; + +#ifdef CONFIG_OF +static const struct of_device_id st350_dt_ids[] = { + { .compatible = "st,sta350", }, + { } +}; +MODULE_DEVICE_TABLE(of, st350_dt_ids); + +static const char const *sta350_ffx_modes[] = { + [STA350_FFX_PM_DROP_COMP] = "drop-compensation", + [STA350_FFX_PM_TAPERED_COMP] = "tapered-compensation", + [STA350_FFX_PM_FULL_POWER] = "full-power-mode", + [STA350_FFX_PM_VARIABLE_DROP_COMP] = "variable-drop-compensation", +}; + +static int sta350_probe_dt(struct device *dev, struct sta350_priv *sta350) +{ + struct device_node *np = dev->of_node; + struct sta350_platform_data *pdata; + const char *ffx_power_mode; + u16 tmp; + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + sta350->gpio_nreset = of_get_named_gpio(np, "reset-gpio", 0); + sta350->gpio_power_down = of_get_named_gpio(np, "power-down-gpio", 0); + + of_property_read_u8(np, "st,output-conf", + &pdata->output_conf); + of_property_read_u8(np, "st,ch1-output-mapping", + &pdata->ch1_output_mapping); + of_property_read_u8(np, "st,ch2-output-mapping", + &pdata->ch2_output_mapping); + of_property_read_u8(np, "st,ch3-output-mapping", + &pdata->ch3_output_mapping); + + if (of_get_property(np, "st,thermal-warning-recovery", NULL)) + pdata->thermal_warning_recovery = 1; + if (of_get_property(np, "st,thermal-warning-adjustment", NULL)) + pdata->thermal_warning_adjustment = 1; + if (of_get_property(np, "st,fault-detect-recovery", NULL)) + pdata->fault_detect_recovery = 1; + + pdata->ffx_power_output_mode = STA350_FFX_PM_VARIABLE_DROP_COMP; + if (!of_property_read_string(np, "st,ffx-power-output-mode", + &ffx_power_mode)) { + int i, mode = -EINVAL; + + for (i = 0; i < ARRAY_SIZE(sta350_ffx_modes); i++) + if (!strcasecmp(ffx_power_mode, sta350_ffx_modes[i])) + mode = i; + + if (mode < 0) + dev_warn(dev, "Unsupported ffx output mode: %s\n", + ffx_power_mode); + else + pdata->ffx_power_output_mode = mode; + } + + tmp = 140; + of_property_read_u16(np, "st,drop-compensation-ns", &tmp); + pdata->drop_compensation_ns = clamp_t(u16, tmp, 0, 300) / 20; + + if (of_get_property(np, "st,overcurrent-warning-adjustment", NULL)) + pdata->oc_warning_adjustment = 1; + + /* CONFE */ + if (of_get_property(np, "st,max-power-use-mpcc", NULL)) + pdata->max_power_use_mpcc = 1; + + if (of_get_property(np, "st,max-power-correction", NULL)) + pdata->max_power_correction = 1; + + if (of_get_property(np, "st,am-reduction-mode", NULL)) + pdata->am_reduction_mode = 1; + + if (of_get_property(np, "st,odd-pwm-speed-mode", NULL)) + pdata->odd_pwm_speed_mode = 1; + + if (of_get_property(np, "st,distortion-compensation", NULL)) + pdata->distortion_compensation = 1; + + /* CONFF */ + if (of_get_property(np, "st,invalid-input-detect-mute", NULL)) + pdata->invalid_input_detect_mute = 1; + + sta350->pdata = pdata; + + return 0; +} +#endif + +static int sta350_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct sta350_priv *sta350; + int ret, i; + + sta350 = devm_kzalloc(&i2c->dev, sizeof(struct sta350_priv), + GFP_KERNEL); + if (!sta350) + return -ENOMEM; + + sta350->pdata = dev_get_platdata(&i2c->dev); + sta350->gpio_nreset = -EINVAL; + +#ifdef CONFIG_OF + if (of_match_device(st350_dt_ids, &i2c->dev)) { + ret = sta350_probe_dt(&i2c->dev, sta350); + if (ret < 0) + return ret; + } +#endif + + /* regulators */ + for (i = 0; i < ARRAY_SIZE(sta350->supplies); i++) + sta350->supplies[i].supply = sta350_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + sta350->regmap = devm_regmap_init_i2c(i2c, &sta350_regmap); + if (IS_ERR(sta350->regmap)) { + ret = PTR_ERR(sta350->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, sta350); + + ret = snd_soc_register_codec(&i2c->dev, &sta350_codec, &sta350_dai, 1); + if (ret < 0) + dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); + + return ret; +} + +static int sta350_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id sta350_i2c_id[] = { + { "sta350", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta350_i2c_id); + +static struct i2c_driver sta350_i2c_driver = { + .driver = { + .name = "sta350", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(st350_dt_ids), + }, + .probe = sta350_i2c_probe, + .remove = sta350_i2c_remove, + .id_table = sta350_i2c_id, +}; + +module_i2c_driver(sta350_i2c_driver); + +MODULE_DESCRIPTION("ASoC STA350 driver"); +MODULE_AUTHOR("Sven Brandau <info(a)brandau.biz>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sta350.h b/sound/soc/codecs/sta350.h new file mode 100644 index 0000000..c3248f0 --- /dev/null +++ b/sound/soc/codecs/sta350.h @@ -0,0 +1,228 @@ +/* + * Codec driver for ST STA350 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * based on code from: + * Raumfeld GmbH + * Johannes Stezenbach <js(a)sig21.net> + * Wolfson Microelectronics PLC. + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#ifndef _ASOC_STA_350_H +#define _ASOC_STA_350_H + +/* STA50 register addresses */ + +#define STA350_REGISTER_COUNT 0x4D +#define STA350_COEF_COUNT 62 + +#define STA350_CONFA 0x00 +#define STA350_CONFB 0x01 +#define STA350_CONFC 0x02 +#define STA350_CONFD 0x03 +#define STA350_CONFE 0x04 +#define STA350_CONFF 0x05 +#define STA350_MMUTE 0x06 +#define STA350_MVOL 0x07 +#define STA350_C1VOL 0x08 +#define STA350_C2VOL 0x09 +#define STA350_C3VOL 0x0a +#define STA350_AUTO1 0x0b +#define STA350_AUTO2 0x0c +#define STA350_AUTO3 0x0d +#define STA350_C1CFG 0x0e +#define STA350_C2CFG 0x0f +#define STA350_C3CFG 0x10 +#define STA350_TONE 0x11 +#define STA350_L1AR 0x12 +#define STA350_L1ATRT 0x13 +#define STA350_L2AR 0x14 +#define STA350_L2ATRT 0x15 +#define STA350_CFADDR2 0x16 +#define STA350_B1CF1 0x17 +#define STA350_B1CF2 0x18 +#define STA350_B1CF3 0x19 +#define STA350_B2CF1 0x1a +#define STA350_B2CF2 0x1b +#define STA350_B2CF3 0x1c +#define STA350_A1CF1 0x1d +#define STA350_A1CF2 0x1e +#define STA350_A1CF3 0x1f +#define STA350_A2CF1 0x20 +#define STA350_A2CF2 0x21 +#define STA350_A2CF3 0x22 +#define STA350_B0CF1 0x23 +#define STA350_B0CF2 0x24 +#define STA350_B0CF3 0x25 +#define STA350_CFUD 0x26 +#define STA350_MPCC1 0x27 +#define STA350_MPCC2 0x28 +#define STA350_DCC1 0x29 +#define STA350_DCC2 0x2a +#define STA350_FDRC1 0x2b +#define STA350_FDRC2 0x2c +#define STA350_STATUS 0x2d +/* reserved: 0x2d - 0x30 */ +#define STA350_EQCFG 0x31 +#define STA350_EATH1 0x32 +#define STA350_ERTH1 0x33 +#define STA350_EATH2 0x34 +#define STA350_ERTH2 0x35 +#define STA350_CONFX 0x36 +#define STA350_SVCA 0x37 +#define STA350_SVCB 0x38 +#define STA350_RMS0A 0x39 +#define STA350_RMS0B 0x3a +#define STA350_RMS0C 0x3b +#define STA350_RMS1A 0x3c +#define STA350_RMS1B 0x3d +#define STA350_RMS1C 0x3e +#define STA350_EVOLRES 0x3f +/* reserved: 0x40 - 0x47 */ +#define STA350_NSHAPE 0x48 +#define STA350_CTXB4B1 0x49 +#define STA350_CTXB7B5 0x4a +#define STA350_MISC1 0x4b +#define STA350_MISC2 0x4c + +/* 0x00 CONFA */ +#define STA350_CONFA_MCS_MASK 0x03 +#define STA350_CONFA_MCS_SHIFT 0 +#define STA350_CONFA_IR_MASK 0x18 +#define STA350_CONFA_IR_SHIFT 3 +#define STA350_CONFA_TWRB BIT(5) +#define STA350_CONFA_TWAB BIT(6) +#define STA350_CONFA_FDRB BIT(7) + +/* 0x01 CONFB */ +#define STA350_CONFB_SAI_MASK 0x0f +#define STA350_CONFB_SAI_SHIFT 0 +#define STA350_CONFB_SAIFB BIT(4) +#define STA350_CONFB_DSCKE BIT(5) +#define STA350_CONFB_C1IM BIT(6) +#define STA350_CONFB_C2IM BIT(7) + +/* 0x02 CONFC */ +#define STA350_CONFC_OM_MASK 0x03 +#define STA350_CONFC_OM_SHIFT 0 +#define STA350_CONFC_CSZ_MASK 0x3c +#define STA350_CONFC_CSZ_SHIFT 2 +#define STA350_CONFC_OCRB BIT(7) + +/* 0x03 CONFD */ +#define STA350_CONFD_HPB_SHIFT 0 +#define STA350_CONFD_DEMP_SHIFT 1 +#define STA350_CONFD_DSPB_SHIFT 2 +#define STA350_CONFD_PSL_SHIFT 3 +#define STA350_CONFD_BQL_SHIFT 4 +#define STA350_CONFD_DRC_SHIFT 5 +#define STA350_CONFD_ZDE_SHIFT 6 +#define STA350_CONFD_SME_SHIFT 7 + +/* 0x04 CONFE */ +#define STA350_CONFE_MPCV BIT(0) +#define STA350_CONFE_MPCV_SHIFT 0 +#define STA350_CONFE_MPC BIT(1) +#define STA350_CONFE_MPC_SHIFT 1 +#define STA350_CONFE_NSBW BIT(2) +#define STA350_CONFE_NSBW_SHIFT 2 +#define STA350_CONFE_AME BIT(3) +#define STA350_CONFE_AME_SHIFT 3 +#define STA350_CONFE_PWMS BIT(4) +#define STA350_CONFE_PWMS_SHIFT 4 +#define STA350_CONFE_DCCV BIT(5) +#define STA350_CONFE_DCCV_SHIFT 5 +#define STA350_CONFE_ZCE BIT(6) +#define STA350_CONFE_ZCE_SHIFT 6 +#define STA350_CONFE_SVE BIT(7) +#define STA350_CONFE_SVE_SHIFT 7 + +/* 0x05 CONFF */ +#define STA350_CONFF_OCFG_MASK 0x03 +#define STA350_CONFF_OCFG_SHIFT 0 +#define STA350_CONFF_IDE BIT(2) +#define STA350_CONFF_BCLE BIT(3) +#define STA350_CONFF_LDTE BIT(4) +#define STA350_CONFF_ECLE BIT(5) +#define STA350_CONFF_PWDN BIT(6) +#define STA350_CONFF_EAPD BIT(7) + +/* 0x06 MMUTE */ +#define STA350_MMUTE_MMUTE 0x01 +#define STA350_MMUTE_MMUTE_SHIFT 0 +#define STA350_MMUTE_C1M 0x02 +#define STA350_MMUTE_C1M_SHIFT 1 +#define STA350_MMUTE_C2M 0x04 +#define STA350_MMUTE_C2M_SHIFT 2 +#define STA350_MMUTE_C3M 0x08 +#define STA350_MMUTE_C3M_SHIFT 3 +#define STA350_MMUTE_LOC_MASK 0xC0 +#define STA350_MMUTE_LOC_SHIFT 6 + +/* 0x0b AUTO1 */ +#define STA350_AUTO1_AMGC_MASK 0x30 +#define STA350_AUTO1_AMGC_SHIFT 4 + +/* 0x0c AUTO2 */ +#define STA350_AUTO2_AMAME 0x01 +#define STA350_AUTO2_AMAM_MASK 0x0e +#define STA350_AUTO2_AMAM_SHIFT 1 +#define STA350_AUTO2_XO_MASK 0xf0 +#define STA350_AUTO2_XO_SHIFT 4 + +/* 0x0d AUTO3 */ +#define STA350_AUTO3_PEQ_MASK 0x1f +#define STA350_AUTO3_PEQ_SHIFT 0 + +/* 0x0e 0x0f 0x10 CxCFG */ +#define STA350_CxCFG_TCB_SHIFT 0 +#define STA350_CxCFG_EQBP_SHIFT 1 +#define STA350_CxCFG_VBP_SHIFT 2 +#define STA350_CxCFG_BO_SHIFT 3 +#define STA350_CxCFG_LS_SHIFT 4 +#define STA350_CxCFG_OM_MASK 0xc0 +#define STA350_CxCFG_OM_SHIFT 6 + +/* 0x11 TONE */ +#define STA350_TONE_BTC_SHIFT 0 +#define STA350_TONE_TTC_SHIFT 4 + +/* 0x12 0x13 0x14 0x15 limiter attack/release */ +#define STA350_LxA_SHIFT 0 +#define STA350_LxR_SHIFT 4 + +/* 0x26 CFUD */ +#define STA350_CFUD_W1 0x01 +#define STA350_CFUD_WA 0x02 +#define STA350_CFUD_R1 0x04 +#define STA350_CFUD_RA 0x08 + + +/* biquad filter coefficient table offsets */ +#define STA350_C1_BQ_BASE 0 +#define STA350_C2_BQ_BASE 20 +#define STA350_CH_BQ_NUM 4 +#define STA350_BQ_NUM_COEF 5 +#define STA350_XO_HP_BQ_BASE 40 +#define STA350_XO_LP_BQ_BASE 45 +#define STA350_C1_PRESCALE 50 +#define STA350_C2_PRESCALE 51 +#define STA350_C1_POSTSCALE 52 +#define STA350_C2_POSTSCALE 53 +#define STA350_C3_POSTSCALE 54 +#define STA350_TW_POSTSCALE 55 +#define STA350_C1_MIX1 56 +#define STA350_C1_MIX2 57 +#define STA350_C2_MIX1 58 +#define STA350_C2_MIX2 59 +#define STA350_C3_MIX1 60 +#define STA350_C3_MIX2 61 + +#endif /* _ASOC_STA_350_H */ -- 1.8.5.3

2 2

[alsa-devel] [PATCH] ASoC: Add support for multi register mux
by Arun Shamanna Lakshmi 27 Mar '14

27 Mar '14

If the mux uses 1 bit position per input, and requires to set one single bit at a time, then an N bit register can support up to N inputs. In more recent Tegra chips, we have at least greater than 64 inputs which requires at least 2 .reg fields in struct soc_enum. Signed-off-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> --- include/sound/soc-dapm.h | 20 +++++- include/sound/soc.h | 44 ++++++++++--- sound/soc/soc-core.c | 118 +++++++++++++++++++++++++++++++++-- sound/soc/soc-dapm.c | 155 +++++++++++++++++++++++++++++++++++++++------- 4 files changed, 300 insertions(+), 37 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ef78f56..324de75 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -315,6 +315,12 @@ struct device; .private_value = (unsigned long)&xenum } #define SOC_DAPM_VALUE_ENUM(xname, xenum) \ SOC_DAPM_ENUM(xname, xenum) +#define SOC_DAPM_VALUE_ENUM_WIDE(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_wide, \ + .get = snd_soc_dapm_get_value_enum_wide, \ + .put = snd_soc_dapm_put_value_enum_wide, \ + .private_value = (unsigned long)&xenum } #define SOC_DAPM_PIN_SWITCH(xname) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname " Switch", \ .info = snd_soc_dapm_info_pin_switch, \ @@ -352,6 +358,9 @@ struct device; /* regulator widget flags */ #define SND_SOC_DAPM_REGULATOR_BYPASS 0x1 /* bypass when disabled */ +/* maximum number of registers to update */ +#define SOC_DAPM_UPDATE_MAX_REGS 3 + struct snd_soc_dapm_widget; enum snd_soc_dapm_type; struct snd_soc_dapm_path; @@ -378,6 +387,10 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_get_value_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_put_value_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, @@ -590,9 +603,10 @@ struct snd_soc_dapm_widget { struct snd_soc_dapm_update { struct snd_kcontrol *kcontrol; - int reg; - int mask; - int val; + int reg[SOC_DAPM_UPDATE_MAX_REGS]; + int mask[SOC_DAPM_UPDATE_MAX_REGS]; + int val[SOC_DAPM_UPDATE_MAX_REGS]; + int num_regs; }; /* DAPM context */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 0b83168..67097c6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -177,16 +177,22 @@ {.reg = xreg, .min = xmin, .max = xmax, \ .platform_max = xmax} } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xitems, xtexts) \ -{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ +{ .reg[0] = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .items = xitems, .texts = xtexts, \ - .mask = xitems ? roundup_pow_of_two(xitems) - 1 : 0} + .mask[0] = xitems ? roundup_pow_of_two(xitems) - 1 : 0, .num_regs = 1 } #define SOC_ENUM_SINGLE(xreg, xshift, xitems, xtexts) \ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xitems, xtexts) #define SOC_ENUM_SINGLE_EXT(xitems, xtexts) \ { .items = xitems, .texts = xtexts } +#define SOC_VALUE_ENUM_TRIPLE(reg1, reg2, reg3, mask1, mask2, mask3, \ + nreg, nmax, xtexts, xvalues) \ +{ .reg[0] = reg1, .reg[1] = reg2, .reg[2] = reg3, \ + .mask[0] = mask1, .mask[1] = mask2, .mask[2] = mask3, \ + .num_regs = nreg, .items = nmax, .texts = xtexts, .values = xvalues } #define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \ -{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues} +{ .reg[0] = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ + .mask[0] = xmask, .items = xitems, .texts = xtexts, .values = xvalues,\ + .num_regs = 1 } #define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \ @@ -198,6 +204,12 @@ .private_value = (unsigned long)&xenum } #define SOC_VALUE_ENUM(xname, xenum) \ SOC_ENUM(xname, xenum) +#define SOC_VALUE_ENUM_WIDE(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\ + .info = snd_soc_info_enum_wide, \ + .get = snd_soc_get_value_enum_wide, \ + .put = snd_soc_put_value_enum_wide, \ + .private_value = (unsigned long)&xenum } #define SOC_SINGLE_EXT(xname, xreg, xshift, xmax, xinvert,\ xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -297,7 +309,13 @@ SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \ const struct soc_enum name = SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(xtexts), xtexts) - +#define SOC_VALUE_ENUM_TRIPLE_DECL(name, reg1, reg2, reg3, \ + mask1, mask2, mask3, \ + xtexts, xvalues) \ + const struct soc_enum name = SOC_VALUE_ENUM_TRIPLE(reg1, reg2, reg3, \ + mask1, mask2, mask3, 3, \ + ARRAY_SIZE(xtexts), \ + xtexts, xvalues) /* * Component probe and remove ordering levels for components with runtime * dependencies. @@ -309,6 +327,11 @@ #define SND_SOC_COMP_ORDER_LAST 2 /* + * The maximum number of registers in soc_enum + */ +#define SOC_ENUM_MAX_REGS 3 + +/* * Bias levels * * @ON: Bias is fully on for audio playback and capture operations. @@ -507,6 +530,12 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_info_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_soc_get_value_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_value_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); #define snd_soc_info_bool_ext snd_ctl_boolean_mono_info @@ -1098,13 +1127,14 @@ struct soc_mreg_control { /* enumerated kcontrol */ struct soc_enum { - int reg; + int reg[SOC_ENUM_MAX_REGS]; unsigned char shift_l; unsigned char shift_r; unsigned int items; - unsigned int mask; + unsigned int mask[SOC_ENUM_MAX_REGS]; const char * const *texts; const unsigned int *values; + unsigned int num_regs; }; /** diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index caebd63..e47479d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2601,12 +2601,12 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, unsigned int val, item; unsigned int reg_val; - reg_val = snd_soc_read(codec, e->reg); - val = (reg_val >> e->shift_l) & e->mask; + reg_val = snd_soc_read(codec, e->reg[0]); + val = (reg_val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[0] = item; if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = item; } @@ -2636,19 +2636,125 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, if (item[0] >= e->items) return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; + mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] >= e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r; - mask |= e->mask << e->shift_r; + mask |= e->mask[0] << e->shift_r; } - return snd_soc_update_bits_locked(codec, e->reg, mask, val); + return snd_soc_update_bits_locked(codec, e->reg[0], mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); /** + * snd_soc_info_enum_wide - enumerated mulitple register mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a enumerated multiple register + * mixer control. + * + * Returns 0 for success. + */ +int snd_soc_info_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = e->items; + + if (uinfo->value.enumerated.item > e->items - 1) + uinfo->value.enumerated.item = e->items - 1; + strcpy(uinfo->value.enumerated.name, + e->texts[uinfo->value.enumerated.item]); + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_enum_wide); + +/** + * snd_soc_get_value_enum_wide - semi enumerated multiple registers mixer + * get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a semi enumerated mutiple registers mixer. + * + * Mutiple semi enumerated mixer: the mixer has multiple registers to set the + * values. The enumerated items are referred as values. Can be + * used for handling bitfield coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_get_value_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int val, i, reg_idx; + bool match = true; + + for (i = 0; i < e->items; i++) { + match = true; + for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { + val = snd_soc_read(codec, e->reg[reg_idx]); + if (val != e->values[i * e->num_regs + reg_idx]) { + match = false; + break; + } + } + if (match) + break; + } + if (!match) { + dev_err(codec->dev, "ASoC: Failed to find matched enum value\n"); + return -EINVAL; + } else + ucontrol->value.enumerated.item[0] = i; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_value_enum_wide); + +/** + * snd_soc_get_value_enum_wide - semi enumerated multiple mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to put the value of a multiple semi enumerated mixer. + * + * Mutiple semi enumerated mixer: the mixer has multiple registers to set the + * values. The enumerated items are referred as values. Can be + * used for handling bitfield coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_put_value_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int val, reg_idx, item; + int ret; + + if (ucontrol->value.enumerated.item[0] > e->items - 1) + return -EINVAL; + item = ucontrol->value.enumerated.item[0]; + for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { + val = e->values[item * e->num_regs + reg_idx]; + ret = snd_soc_update_bits_locked(codec, e->reg[reg_idx], + e->mask[reg_idx], val); + if (ret) + return ret; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_put_value_enum_wide); + +/** * snd_soc_read_signed - Read a codec register and interprete as signed value * @codec: codec * @reg: Register to read diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c8a780d..22ca178 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -514,9 +514,9 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, unsigned int val, item; int i; - if (e->reg != SND_SOC_NOPM) { - soc_widget_read(dest, e->reg, &val); - val = (val >> e->shift_l) & e->mask; + if (e->reg[0] != SND_SOC_NOPM) { + soc_widget_read(dest, e->reg[0], &val); + val = (val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); } else { /* since a virtual mux has no backing registers to @@ -1553,7 +1553,7 @@ static void dapm_widget_update(struct snd_soc_card *card) struct snd_soc_dapm_update *update = card->update; struct snd_soc_dapm_widget_list *wlist; struct snd_soc_dapm_widget *w = NULL; - unsigned int wi; + unsigned int wi, update_idx; int ret; if (!update || !dapm_kcontrol_is_powered(update->kcontrol)) @@ -1575,8 +1575,10 @@ static void dapm_widget_update(struct snd_soc_card *card) if (!w) return; - ret = soc_widget_update_bits_locked(w, update->reg, update->mask, - update->val); + for (update_idx = 0; update_idx < update->num_regs; update_idx++) + ret = soc_widget_update_bits_locked(w, update->reg[update_idx], + update->mask[update_idx], + update->val[update_idx]); if (ret < 0) dev_err(w->dapm->dev, "ASoC: %s DAPM update failed: %d\n", w->name, ret); @@ -2866,9 +2868,10 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, if (change) { if (reg != SND_SOC_NOPM) { update.kcontrol = kcontrol; - update.reg = reg; - update.mask = mask; - update.val = val; + update.reg[0] = reg; + update.mask[0] = mask; + update.val[0] = val; + update.num_regs = 1; card->update = &update; } @@ -2903,15 +2906,15 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val; - if (e->reg != SND_SOC_NOPM) - reg_val = snd_soc_read(codec, e->reg); + if (e->reg[0] != SND_SOC_NOPM) + reg_val = snd_soc_read(codec, e->reg[0]); else reg_val = dapm_kcontrol_get_value(kcontrol); - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_l) & e->mask[0]; ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_r) & e->mask; + val = (reg_val >> e->shift_r) & e->mask[0]; val = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = val; } @@ -2945,27 +2948,28 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; + mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] > e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l; - mask |= e->mask << e->shift_r; + mask |= e->mask[0] << e->shift_r; } mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (e->reg != SND_SOC_NOPM) - change = snd_soc_test_bits(codec, e->reg, mask, val); + if (e->reg[0] != SND_SOC_NOPM) + change = snd_soc_test_bits(codec, e->reg[0], mask, val); else change = dapm_kcontrol_set_value(kcontrol, val); if (change) { - if (e->reg != SND_SOC_NOPM) { + if (e->reg[0] != SND_SOC_NOPM) { update.kcontrol = kcontrol; - update.reg = e->reg; - update.mask = mask; - update.val = val; + update.reg[0] = e->reg[0]; + update.mask[0] = mask; + update.val[0] = val; + update.num_regs = 1; card->update = &update; } @@ -2984,6 +2988,115 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); /** + * snd_soc_dapm_get_value_enum_wide - dapm semi enumerated multiple registers + * mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a dapm semi enumerated multiple register mixer + * control. + * + * semi enumerated multiple registers mixer: + * the mixer has multiple regsters to set the enumerated items. The enumerated + * iteams are referred as values. + * Can be used for handling bitfield coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_dapm_get_value_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int reg_val, i, reg_idx; + bool match = true; + + for (i = 0; i < e->items; i++) { + match = true; + for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { + reg_val = snd_soc_read(codec, e->reg[reg_idx]); + if (reg_val != e->values[i * e->num_regs + reg_idx]) { + match = false; + break; + } + } + if (match) + break; + } + if (!match) { + dev_err(codec->dev, "ASoC: Failed to find matched enum value\n"); + return -EINVAL; + } else + ucontrol->value.enumerated.item[0] = i; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_value_enum_wide); + +/** + * snd_soc_dapm_put_value_enum_wide - dapm semi enumerated multiple registers + * mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to put the value of a dapm semi enumerated multiple register mixer + * control. + * + * semi enumerated multiple registers mixer: + * the mixer has multiple regsters to set the enumerated items. The enumerated + * iteams are referred as values. + * Can be used for handling bitfield coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_dapm_put_value_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_card *card = codec->card; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int value, item, old, reg_idx; + struct snd_soc_dapm_update update; + int wi, update_idx; + int ret = 0; + + if (ucontrol->value.enumerated.item[0] > e->items - 1) + return -EINVAL; + + item = ucontrol->value.enumerated.item[0]; + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + for (reg_idx = 0, update_idx = 0; reg_idx < e->num_regs; reg_idx++) { + value = e->values[item * e->num_regs + reg_idx]; + old = snd_soc_read(codec, e->reg[reg_idx]); + if (value != old) { + update.reg[update_idx] = e->reg[reg_idx]; + update.mask[update_idx] = e->mask[reg_idx]; + update.val[update_idx] = value; + update.num_regs = ++update_idx; + } + } + + if (update_idx) { + update.kcontrol = kcontrol; + card->update = &update; + + ret = soc_dapm_mux_update_power(card, kcontrol, item, e); + + card->update = NULL; + } + + mutex_unlock(&card->dapm_mutex); + + if (ret > 0) + soc_dpcm_runtime_update(card); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_wide); + +/** * snd_soc_dapm_info_pin_switch - Info for a pin switch * * @kcontrol: mixer control -- 1.7.9.5

4 7

[alsa-devel] [PATCH] ASoC: rt5640: Correct the judgement of data length
by Oder Chiou 27 Mar '14

27 Mar '14

The patch corrects the judgement of data length. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 0883c8a..bcd6513 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1622,16 +1622,16 @@ static int rt5640_hw_params(struct snd_pcm_substream *substream, 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: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: val_len |= RT5640_I2S_DL_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: val_len |= RT5640_I2S_DL_24; break; - case SNDRV_PCM_FORMAT_S8: + case 8: val_len |= RT5640_I2S_DL_8; break; default: -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: fsl_sai: Add isr to deal with error flag
by Nicolin Chen 27 Mar '14

27 Mar '14

It's quite cricial to clear error flags because SAI might hang if getting FIFO underrun during playback (I haven't confirmed the same issue on Rx overflow though). So this patch enables those irq and adds isr() to clear the flags so as to keep playback entirely safe. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 69 ++++++++++++++++++++++++++++++++++++++++++++++--- sound/soc/fsl/fsl_sai.h | 9 +++++++ 2 files changed, 75 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index c4a4231..5f91aff 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -23,6 +23,55 @@ #include "fsl_sai.h" +#define FSL_SAI_FLAGS (FSL_SAI_CSR_SEIE |\ + FSL_SAI_CSR_FEIE |\ + FSL_SAI_CSR_FWIE) + +static irqreturn_t fsl_sai_isr(int irq, void *devid) +{ + struct fsl_sai *sai = (struct fsl_sai *)devid; + struct device *dev = &sai->pdev->dev; + u32 xcsr; + + regmap_read(sai->regmap, FSL_SAI_TCSR, &xcsr); + regmap_write(sai->regmap, FSL_SAI_TCSR, xcsr); + + if (xcsr & FSL_SAI_CSR_WSF) + dev_dbg(dev, "isr: Start of Tx word detected\n"); + + if (xcsr & FSL_SAI_CSR_SEF) + dev_dbg(dev, "isr: Tx Frame sync error detected\n"); + + if (xcsr & FSL_SAI_CSR_FEF) + dev_dbg(dev, "isr: Transmit underrun detected\n"); + + if (xcsr & FSL_SAI_CSR_FWF) + dev_dbg(dev, "isr: Enabled transmit FIFO is empty\n"); + + if (xcsr & FSL_SAI_CSR_FRF) + dev_dbg(dev, "isr: Transmit FIFO watermark has been reached\n"); + + regmap_read(sai->regmap, FSL_SAI_RCSR, &xcsr); + regmap_write(sai->regmap, FSL_SAI_RCSR, xcsr); + + if (xcsr & FSL_SAI_CSR_WSF) + dev_dbg(dev, "isr: Start of Rx word detected\n"); + + if (xcsr & FSL_SAI_CSR_SEF) + dev_dbg(dev, "isr: Rx Frame sync error detected\n"); + + if (xcsr & FSL_SAI_CSR_FEF) + dev_dbg(dev, "isr: Receive overflow detected\n"); + + if (xcsr & FSL_SAI_CSR_FWF) + dev_dbg(dev, "isr: Enabled receive FIFO is full\n"); + + if (xcsr & FSL_SAI_CSR_FRF) + dev_dbg(dev, "isr: Receive FIFO watermark has been reached\n"); + + return IRQ_HANDLED; +} + static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { @@ -373,8 +422,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, FSL_SAI_FLAGS); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, FSL_SAI_FLAGS); regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_TX * 2); regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK, @@ -490,12 +539,14 @@ static int fsl_sai_probe(struct platform_device *pdev) struct fsl_sai *sai; struct resource *res; void __iomem *base; - int ret; + int irq, ret; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) return -ENOMEM; + sai->pdev = pdev; + sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); if (sai->big_endian_regs) fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; @@ -514,6 +565,18 @@ static int fsl_sai_probe(struct platform_device *pdev) return PTR_ERR(sai->regmap); } + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + return irq; + } + + ret = devm_request_irq(&pdev->dev, irq, fsl_sai_isr, 0, np->name, sai); + if (ret) { + dev_err(&pdev->dev, "failed to claim irq %u\n", irq); + return ret; + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index e432260..05d1a1f 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -37,7 +37,15 @@ /* SAI Transmit/Recieve Control Register */ #define FSL_SAI_CSR_TERE BIT(31) +#define FSL_SAI_CSR_WSF BIT(20) +#define FSL_SAI_CSR_SEF BIT(19) +#define FSL_SAI_CSR_FEF BIT(18) #define FSL_SAI_CSR_FWF BIT(17) +#define FSL_SAI_CSR_FRF BIT(16) +#define FSL_SAI_CSR_WSIE BIT(12) +#define FSL_SAI_CSR_SEIE BIT(11) +#define FSL_SAI_CSR_FEIE BIT(10) +#define FSL_SAI_CSR_FWIE BIT(9) #define FSL_SAI_CSR_FRIE BIT(8) #define FSL_SAI_CSR_FRDE BIT(0) @@ -99,6 +107,7 @@ #define FSL_SAI_MAXBURST_RX 6 struct fsl_sai { + struct platform_device *pdev; struct regmap *regmap; bool big_endian_regs; -- 1.8.4

4 15

[alsa-devel] [PATCH 1/1] ASoC: Samsung: Fix build on multiplatform
by Sachin Kamat 27 Mar '14

27 Mar '14

PCM and S/PDIF drivers referenced mach headers for a trivial data structure. This caused build errors on multiplatform builds as machine headers are not accessible from driver files. Move the data structure definition to the driver header and remove the dependency. While at it rename the structure to avoid multiple definition errors as the same structure is also used by the platform code. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/ac97.c | 6 +++--- sound/soc/samsung/dma.h | 6 +++++- sound/soc/samsung/i2s.c | 6 +++--- sound/soc/samsung/pcm.c | 5 ++--- sound/soc/samsung/s3c2412-i2s.c | 4 ++-- sound/soc/samsung/s3c24xx-i2s.c | 4 ++-- sound/soc/samsung/spdif.c | 3 +-- 7 files changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 4a88e36c82ec..76b072bd4ba2 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -39,15 +39,15 @@ struct s3c_ac97_info { }; static struct s3c_ac97_info s3c_ac97; -static struct s3c2410_dma_client s3c_dma_client_out = { +static struct s3c_dma_client s3c_dma_client_out = { .name = "AC97 PCMOut" }; -static struct s3c2410_dma_client s3c_dma_client_in = { +static struct s3c_dma_client s3c_dma_client_in = { .name = "AC97 PCMIn" }; -static struct s3c2410_dma_client s3c_dma_client_micin = { +static struct s3c_dma_client s3c_dma_client_micin = { .name = "AC97 MicIn" }; diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 225e5378014e..ad7c0f04f00d 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -14,8 +14,12 @@ #include <sound/dmaengine_pcm.h> +struct s3c_dma_client { + char *name; +}; + struct s3c_dma_params { - struct s3c2410_dma_client *client; /* stream identifier */ + struct s3c_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ dma_addr_t dma_addr; int dma_size; /* Size of the DMA transfer */ diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 0a9b44c940ce..048ead967199 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1211,10 +1211,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_playback.dma_addr = regs_base + I2STXD; pri_dai->dma_capture.dma_addr = regs_base + I2SRXD; pri_dai->dma_playback.client = - (struct s3c2410_dma_client *)&pri_dai->dma_playback; + (struct s3c_dma_client *)&pri_dai->dma_playback; pri_dai->dma_playback.ch_name = "tx"; pri_dai->dma_capture.client = - (struct s3c2410_dma_client *)&pri_dai->dma_capture; + (struct s3c_dma_client *)&pri_dai->dma_capture; pri_dai->dma_capture.ch_name = "rx"; pri_dai->dma_playback.dma_size = 4; pri_dai->dma_capture.dma_size = 4; @@ -1233,7 +1233,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) } sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; sec_dai->dma_playback.client = - (struct s3c2410_dma_client *)&sec_dai->dma_playback; + (struct s3c_dma_client *)&sec_dai->dma_playback; sec_dai->dma_playback.ch_name = "tx-sec"; if (!np) { diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 6a5e4bf6ac96..ab54e297957c 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -20,7 +20,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "pcm.h" @@ -132,11 +131,11 @@ struct s3c_pcm_info { struct s3c_dma_params *dma_capture; }; -static struct s3c2410_dma_client s3c_pcm_dma_client_out = { +static struct s3c_dma_client s3c_pcm_dma_client_out = { .name = "PCM Stereo out" }; -static struct s3c2410_dma_client s3c_pcm_dma_client_in = { +static struct s3c_dma_client s3c_pcm_dma_client_in = { .name = "PCM Stereo in" }; diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index d0794458963a..e9bb5d7a71ee 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -33,11 +33,11 @@ #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" -static struct s3c2410_dma_client s3c2412_dma_client_out = { +static struct s3c_dma_client s3c2412_dma_client_out = { .name = "I2S PCM Stereo out" }; -static struct s3c2410_dma_client s3c2412_dma_client_in = { +static struct s3c_dma_client s3c2412_dma_client_in = { .name = "I2S PCM Stereo in" }; diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index f31e916dd8c4..d7b8457b5650 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -31,11 +31,11 @@ #include "dma.h" #include "s3c24xx-i2s.h" -static struct s3c2410_dma_client s3c24xx_dma_client_out = { +static struct s3c_dma_client s3c24xx_dma_client_out = { .name = "I2S PCM Stereo out" }; -static struct s3c2410_dma_client s3c24xx_dma_client_in = { +static struct s3c_dma_client s3c24xx_dma_client_in = { .name = "I2S PCM Stereo in" }; diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 28487dcc4538..cfe63b7bcc9f 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -18,7 +18,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "spdif.h" @@ -94,7 +93,7 @@ struct samsung_spdif_info { struct s3c_dma_params *dma_playback; }; -static struct s3c2410_dma_client spdif_dma_client_out = { +static struct s3c_dma_client spdif_dma_client_out = { .name = "S/PDIF Stereo out", }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rt5640: Move cache sync() to resume()
by Oder Chiou 27 Mar '14

27 Mar '14

The patch fixes the defect in case of resume which doesn't sync the cache. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 0061ae6..0883c8a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1890,11 +1890,9 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, static int rt5640_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); switch (level) { case SND_SOC_BIAS_STANDBY: if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { - regcache_cache_only(rt5640->regmap, false); snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_VREF1 | RT5640_PWR_MB | RT5640_PWR_BG | RT5640_PWR_VREF2, @@ -1904,7 +1902,6 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_FV1 | RT5640_PWR_FV2, RT5640_PWR_FV1 | RT5640_PWR_FV2); - regcache_sync(rt5640->regmap); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); snd_soc_update_bits(codec, RT5640_MICBIAS, @@ -1979,6 +1976,9 @@ static int rt5640_resume(struct snd_soc_codec *codec) msleep(400); } + regcache_cache_only(rt5640->regmap, false); + regcache_sync(rt5640->regmap); + return 0; } #else -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2] ASoC: davinci-mcasp: set up channel status bits for S/PDIF mode
by Daniel Mack 27 Mar '14

27 Mar '14

In DIT (S/PDIF) mode, program the transmitted user bits to reflect the configured sample rate, along with some other details. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- Thanks to Peter for the review and bearing with my confusion :) sound/soc/davinci/davinci-mcasp.c | 49 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 47 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 375e197..1fbded0 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include <linux/of_platform.h> #include <linux/of_device.h> +#include <sound/asoundef.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -576,8 +577,12 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) } /* S/PDIF */ -static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) +static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, + unsigned int rate) { + u32 val = 0; + u8 *bytes = (u8*) &val; + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); @@ -599,6 +604,46 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) /* Enable the DIT */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); + /* Set S/PDIF channel status bits */ + bytes[0] = IEC958_AES0_CON_NOT_COPYRIGHT; + bytes[1] = IEC958_AES1_CON_PCM_CODER; + + switch (rate) { + case 22050: + bytes[3] |= IEC958_AES3_CON_FS_22050; + break; + case 24000: + bytes[3] |= IEC958_AES3_CON_FS_24000; + break; + case 32000: + bytes[3] |= IEC958_AES3_CON_FS_32000; + break; + case 44100: + bytes[3] |= IEC958_AES3_CON_FS_44100; + break; + case 48000: + bytes[3] |= IEC958_AES3_CON_FS_48000; + break; + case 88200: + bytes[3] |= IEC958_AES3_CON_FS_88200; + break; + case 96000: + bytes[3] |= IEC958_AES3_CON_FS_96000; + break; + case 176400: + bytes[3] |= IEC958_AES3_CON_FS_176400; + break; + case 192000: + bytes[3] |= IEC958_AES3_CON_FS_192000; + break; + default: + printk(KERN_WARNING "unsupported sampling rate: %d\n", rate); + return -EINVAL; + } + + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, val); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, val); + return 0; } @@ -634,7 +679,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return ret; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) - ret = mcasp_dit_hw_param(mcasp); + ret = mcasp_dit_hw_param(mcasp, params_rate(params)); else ret = mcasp_i2s_hw_param(mcasp, substream->stream); -- 1.8.5.3

2 2

[alsa-devel] [PATCH] ASoC: davinci-mcasp: set up user bits for S/PDIF mode
by Daniel Mack 27 Mar '14

27 Mar '14

In DIT (S/PDIF) mode, program the transmitted user bits to reflect the configured sample rate, along with some other details. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/davinci/davinci-mcasp.c | 55 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 53 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 712a7cd..ec0463a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include <linux/of_platform.h> #include <linux/of_device.h> +#include <sound/asoundef.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -566,8 +567,11 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) } /* S/PDIF */ -static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) +static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, + unsigned int rate) { + u32 val = 0; + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); @@ -589,6 +593,53 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) /* Enable the DIT */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); + /* Set S/PDIF channel status bits */ + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG + 0, + IEC958_AES0_CON_NOT_COPYRIGHT); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG + 0, + IEC958_AES0_CON_NOT_COPYRIGHT); + + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG + 1, + IEC958_AES1_CON_PCM_CODER); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG + 1, + IEC958_AES1_CON_PCM_CODER); + + switch (rate) { + case 22050: + val |= IEC958_AES3_CON_FS_22050; + break; + case 24000: + val |= IEC958_AES3_CON_FS_24000; + break; + case 32000: + val |= IEC958_AES3_CON_FS_32000; + break; + case 44100: + val |= IEC958_AES3_CON_FS_44100; + break; + case 48000: + val |= IEC958_AES3_CON_FS_48000; + break; + case 88200: + val |= IEC958_AES3_CON_FS_88200; + break; + case 96000: + val |= IEC958_AES3_CON_FS_96000; + break; + case 176400: + val |= IEC958_AES3_CON_FS_176400; + break; + case 192000: + val |= IEC958_AES3_CON_FS_192000; + break; + default: + printk(KERN_WARNING "unsupported sampling rate: %d\n", rate); + return -EINVAL; + } + + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG + 3, val); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG + 3, val); + return 0; } @@ -621,7 +672,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, fifo_level = mcasp->rxnumevt * active_serializers; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) - ret = mcasp_dit_hw_param(mcasp); + ret = mcasp_dit_hw_param(mcasp, params_rate(params)); else ret = mcasp_i2s_hw_param(mcasp, substream->stream); -- 1.8.5.3

3 5

[alsa-devel] Lenovo G580 mic noise when recording
by Nevyan Neykov 27 Mar '14

27 Mar '14

Hello, my lenovo g580(2189, 20150) laptop is using intel hd audio conexant driver( Intel Corporation 7 Series/C210 ) and experiencing lots of noise when trying to record from either external or internal microphone. If I lower mic volume / boost the voice get barely recorded. I am using this line in my alsa conf: options snd-hda-intel model=auto position_fix=2 also tried with model=asus etc..., when using position_fix=1 recording receives lots of cracking noise. I must mention that both mics are working very well under win7/8. Regards.

1 0

[alsa-devel] [PATCH RFC v3 0/2] Move dai-link level properties away from dai subnodes
by Jyri Sarha 27 Mar '14

27 Mar '14

This patch is implemented on top of late patches from Jean-Francois Moine [1]. These patches implement the main part of the simple-card changes discussed in alsa-devel mailing list [2]. Since RFC v2 version of the patches, address remaining comments from Jean-Francois Moine [4]: - ASoC: simple-card: Move dai-link level properties away from dai subnodes - Only allow legacy parsing if there is no dai-link node. - Fix double unref in of_card parsing loop. Since RFC version of the patches: - ASoC: core: Update snd_soc_of_parse_daifmt() interface - Make minimal changes to simple-card for it to work with the updated snd_soc_of_parse_daifmt(). (Kuninori Morimoto's comment [3]). - ASoC: simple-card: Move dai-link level properties away from dai subnodes - Fix typo in DT binding document (Jean-Francois Moine's comment [4]). - Remove #define DEBUG - Remove accidental line wrap at simple-audio-card,name property descripition in the DT binding document [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074592.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074388.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074699.html [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074697.html Best regards, Jyri Jyri Sarha (2): ASoC: core: Update snd_soc_of_parse_daifmt() interface ASoC: simple-card: Move dai-link level properties away from dai subnodes .../devicetree/bindings/sound/simple-card.txt | 88 ++++---- include/sound/soc.h | 4 +- sound/soc/generic/simple-card.c | 238 ++++++++++++-------- sound/soc/soc-core.c | 8 +- 4 files changed, 200 insertions(+), 138 deletions(-) -- 1.7.9.5

3 4

[alsa-devel] [PATCH] ASoC: jack: export gpio detect
by cfreeman＠nvidia.com 26 Mar '14

26 Mar '14

From: Christopher Freeman <cfreeman(a)nvidia.com> Export the gpio detect function so machine drivers may call it. Interrupts for the jack may be disabled during sleep, so this allows a machine driver to have the jack status updated during resume. Signed-off-by: Christopher Freeman <cfreeman(a)nvidia.com> --- include/sound/soc.h | 1 + sound/soc/soc-jack.c | 3 ++- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a6a059c..36383cb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -442,6 +442,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); +void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio); #endif /* codec register bit access */ diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0bb5ccc..fd97cc5 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -231,7 +231,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_notifier_unregister); #ifdef CONFIG_GPIOLIB /* gpio detect */ -static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) +void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) { struct snd_soc_jack *jack = gpio->jack; int enable; @@ -251,6 +251,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) snd_soc_jack_report(jack, report, gpio->report); } +EXPORT_SYMBOL_GPL(snd_soc_jack_gpio_detect); /* irq handler for gpio pin */ static irqreturn_t gpio_handler(int irq, void *data) -- 1.8.3.2 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

1 0

Re: [alsa-devel] linux-next: Tree for Mar 26 (sound/soc/codecs/cs42l52.c)
by Randy Dunlap 26 Mar '14

26 Mar '14

On 03/25/2014 11:35 PM, Stephen Rothwell wrote: > Hi all, > > This tree still fails (more than usual) the powerpc allyesconfig build. > > Changes since 20140325: > on x86_64: when CONFIG_INPUT=m and SNC_SOC_CS42L42=y: sound/built-in.o: In function `cs42l52_beep_set': cs42l52.c:(.text+0xd4b01): undefined reference to `input_event' sound/built-in.o: In function `cs42l52_probe': cs42l52.c:(.text+0xd4beb): undefined reference to `devm_input_allocate_device' cs42l52.c:(.text+0xd4d0c): undefined reference to `input_register_device' sound/built-in.o: In function `wm8962_beep_set': wm8962.c:(.text+0x116240): undefined reference to `input_event' sound/built-in.o: In function `wm8962_probe': wm8962.c:(.text+0x116cae): undefined reference to `devm_input_allocate_device' wm8962.c:(.text+0x116e0e): undefined reference to `input_register_device' Full randconfig file is attached. -- ~Randy

1 1

[alsa-devel] Fwd: Re: no reset_resume for driver snd-usb-audio for logitech headset H600
by baumber＠hotmail.com 26 Mar '14

26 Mar '14

FWD: Any news to the snd-usb-audio issue? Thanks, Bernhard Old Message: ------------- I have tested the patch with kernel 3.12.0 with suspend and hibernate, and the problem is gone. No messages "no reset_resume for driver snd-usb-audio?" or other negative effects/messages. After resume, the usb_audio_device (Headset) is the master channel in KDE Mixer, as it should be. => So the problem is solved for me. Can you please add the patch to the latest kernels and LTS-kernels or whitelist the workaround for my device (ID 046d:0a29 Logitech, Inc. H600 [Wireless Headset]) Thanks, Bernhard On 2013-10-20 23:35, Takashi Iwai wrote: > At Sun, 20 Oct 2013 23:18:13 +0200, > <baumber(a)hotmail.com> wrote: >> >> On 2013-10-09 11:16 (CEST,UTC+2), Takashi Iwai wrote: >>>>>> diff --git a/sound/usb/card.c b/sound/usb/card.c >>>>>> index 9d9de8d..c591a83 100644 >>>>>> --- a/sound/usb/card.c >>>>>> +++ b/sound/usb/card.c >>>>>> @@ -747,6 +747,7 @@ static struct usb_driver usb_audio_driver = { >>>>>> .disconnect = usb_audio_disconnect, >>>>>> .suspend = usb_audio_suspend, >>>>>> .resume = usb_audio_resume, >>>>>> + .reset_resume = usb_audio_resume, >>>>>> .id_table = usb_audio_ids, >>>>>> .supports_autosuspend = 1, >>>>>> }; >> >> Shall I add the line in b/sound/usb/card.c and try it? Which kernel version shall I use? > > It doesn't matter which kernel version is. > The error message should disappear, but I don't know whether any other > positive effect you get. As Clemens pointed, there can be some > negative effects by this, so it cannot be taken as is. But at least, > it'd be good to know the result... > > > thanks, > > Takashi

3 10

[alsa-devel] [PATCH 1/3] ASoC: ak5386: add regulator consumer support
by Daniel Mack 26 Mar '14

26 Mar '14

The chip has two power supplies, VA and VDD. Register and enable them both. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/ak5386.c | 35 ++++++++++++++++++++++++++++++++--- 1 file changed, 32 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index 72e953b..99fb5db 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -14,12 +14,18 @@ #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/of_device.h> +#include <linux/regulator/consumer.h> #include <sound/soc.h> #include <sound/pcm.h> #include <sound/initval.h> +static const char *supply_names[] = { + "va", "vd" +}; + struct ak5386_priv { int reset_gpio; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = { @@ -122,6 +128,7 @@ static int ak5386_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct ak5386_priv *priv; + int ret, i; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -130,6 +137,19 @@ static int ak5386_probe(struct platform_device *pdev) priv->reset_gpio = -EINVAL; dev_set_drvdata(dev, priv); + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) + return ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) + return ret; + if (of_match_device(of_match_ptr(ak5386_dt_ids), dev)) priv->reset_gpio = of_get_named_gpio(dev->of_node, "reset-gpio", 0); @@ -140,13 +160,22 @@ static int ak5386_probe(struct platform_device *pdev) "AK5386 Reset")) priv->reset_gpio = -EINVAL; - return snd_soc_register_codec(dev, &soc_codec_ak5386, - &ak5386_dai, 1); + ret = snd_soc_register_codec(dev, &soc_codec_ak5386, + &ak5386_dai, 1); + if (ret < 0) + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), + priv->supplies); + return ret; } static int ak5386_remove(struct platform_device *pdev) { - snd_soc_unregister_codec(&pdev->dev); + struct device *dev = &pdev->dev; + struct ak5386_priv *priv = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; } -- 1.8.5.3

2 5

[alsa-devel] Invalid PCM pointer with CMI8738 and pcm.c buffer underuns
by andoru 26 Mar '14

26 Mar '14

I've previously posted under ALSA's user mailing list so to avoid the clutter, the details about what's happening is here: http://article.gmane.org/gmane.linux.alsa.user/38258 http://article.gmane.org/gmane.linux.alsa.user/38260 There was no debug info under /proc/asound/card0/ so nothing to give from there unless you guys need something specific. Output of alsa-debug.sh: http://www.alsa-project.org/db/?f=26187faadd47f3626029ba667cb004e74695f2b2 aplay returns: ________________________________________________________________________ $ aplay -vv 03\ paris\ match-\ Bikeride.wav Playing WAVE '03 paris match- Bikeride.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 4096 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 4096 stop_threshold : 4096 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'C-Media CMI8738' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 4096 period_size : 1024 period_time : 23219 tstamp_mode : ENABLE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 1073741824 silence_threshold: 0 silence_size : 1073741824 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 #+ | 00% _____________________________________________________________________

1 0

[alsa-devel] twl4030 latency update
by Leonardo Gabrielli 26 Mar '14

26 Mar '14

Dear Peter, I was investigating on TWL4030 high playback latency and stumbled in an old thread started by Edgar http://mailman.alsa-project.org/pipermail/alsa-devel/2011-October/045173.ht… where I read this is related to McBSP2 buffer length Recent kernels seems to have the same behavior (I have a debian beagleboardxM with 3.13.3-armv7-x10) Did you manage to get a fix to this problem? Would it be possible? Regards Leonardo -- Dr. Leonardo Gabrielli, PhD student A3Lab - Dept. Information Engineering Università Politecnica delle Marche via Brecce Bianche 12, 60131, Ancona, Italy Skype: leonardo.gabrielli Web: a3lab.dii.univpm.it/people/leonardo-gabrielli <http://a3lab.dii.univpm.it/people/leonardo-gabrielli>

2 11

[alsa-devel] [PATCH] ASoC: add RT5639 CODEC driver
by oder_chiou＠realtek.com 26 Mar '14

26 Mar '14

From: Oder Chiou <oder_chiou(a)realtek.com> This patch adds the RT5639 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- It is migrated from RT5640 codec driver, and modified for RT5639's specification. --- include/sound/rt5639.h | 22 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5639.c | 2092 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5639.h | 1928 +++++++++++++++++++++++++++++++++++++++++ 5 files changed, 4048 insertions(+) create mode 100644 include/sound/rt5639.h create mode 100644 sound/soc/codecs/rt5639.c create mode 100644 sound/soc/codecs/rt5639.h diff --git a/include/sound/rt5639.h b/include/sound/rt5639.h new file mode 100644 index 0000000..7101120 --- /dev/null +++ b/include/sound/rt5639.h @@ -0,0 +1,22 @@ +/* + * linux/sound/rt5639.h -- Platform data for RT5639 + * + * Copyright 2011 Realtek Microelectronics + * + * 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 __LINUX_SND_RT5639_H +#define __LINUX_SND_RT5639_H + +struct rt5639_platform_data { + /* IN1 & IN2 can optionally be differential */ + bool in1_diff; + bool in2_diff; + + int ldo1_en; /* GPIO for LDO1_EN */ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..ed791fb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -60,6 +60,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C + select SND_SOC_RT5639 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE @@ -316,6 +317,9 @@ config SND_SOC_PCM3008 config SND_SOC_RT5631 tristate +config SND_SOC_RT5639 + tristate + config SND_SOC_RT5640 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc12676..051f3fb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -47,6 +47,7 @@ snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o +snd-soc-rt5639-objs := rt5639.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -180,6 +181,7 @@ obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o +obj-$(CONFIG_SND_SOC_RT5639) += snd-soc-rt5639.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o diff --git a/sound/soc/codecs/rt5639.c b/sound/soc/codecs/rt5639.c new file mode 100644 index 0000000..df8d307 --- /dev/null +++ b/sound/soc/codecs/rt5639.c @@ -0,0 +1,2092 @@ +/* + * rt5639.c -- RT5639 ALSA SoC audio codec driver + * + * Copyright 2011 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * Copyright (c) 2013, NVIDIA 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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of_gpio.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> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5639.h" + +#define RT5639_DEVICE_ID 0x6231 + +#define RT5639_PR_RANGE_BASE (0xff + 1) +#define RT5639_PR_SPACING 0x100 + +#define RT5639_PR_BASE (RT5639_PR_RANGE_BASE + (0 * RT5639_PR_SPACING)) + +static const struct regmap_range_cfg rt5639_ranges[] = { + { .name = "PR", .range_min = RT5639_PR_BASE, + .range_max = RT5639_PR_BASE + 0xb4, + .selector_reg = RT5639_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5639_PRIV_DATA, + .window_len = 0x1, }, +}; + +static struct reg_default init_list[] = { + {0x2e, 0x0c00}, + {RT5639_PR_BASE + 0x3d, 0x3600}, + {RT5639_PR_BASE + 0x12, 0x0aa8}, + {RT5639_PR_BASE + 0x14, 0x0aaa}, + {RT5639_PR_BASE + 0x20, 0x6110}, + {RT5639_PR_BASE + 0x21, 0xe0e0}, + {RT5639_PR_BASE + 0x23, 0x1804}, +}; +#define RT5639_INIT_REG_LEN ARRAY_SIZE(init_list) + +static const struct reg_default rt5639_reg[] = { + { 0x00, 0x000e }, + { 0x01, 0xc8c8 }, + { 0x02, 0xc8c8 }, + { 0x03, 0xc8c8 }, + { 0x04, 0x8000 }, + { 0x0d, 0x0000 }, + { 0x0e, 0x0000 }, + { 0x0f, 0x0808 }, + { 0x19, 0xafaf }, + { 0x1a, 0xafaf }, + { 0x1b, 0x0000 }, + { 0x1c, 0x2f2f }, + { 0x1d, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x27, 0x7060 }, + { 0x28, 0x7070 }, + { 0x29, 0x8080 }, + { 0x2a, 0x5454 }, + { 0x2b, 0x5454 }, + { 0x2c, 0xaa00 }, + { 0x2d, 0x0000 }, + { 0x2e, 0xa000 }, + { 0x2f, 0x0000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x007f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x007f }, + { 0x45, 0xe000 }, + { 0x46, 0x003e }, + { 0x47, 0x003e }, + { 0x48, 0xf800 }, + { 0x49, 0x3800 }, + { 0x4a, 0x0004 }, + { 0x4c, 0xfc00 }, + { 0x4d, 0x0000 }, + { 0x4f, 0x01ff }, + { 0x50, 0x0000 }, + { 0x51, 0x0000 }, + { 0x52, 0x01ff }, + { 0x53, 0xf000 }, + { 0x61, 0x0000 }, + { 0x62, 0x0000 }, + { 0x63, 0x00c0 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x6a, 0x0000 }, + { 0x6c, 0x0000 }, + { 0x70, 0x8000 }, + { 0x71, 0x8000 }, + { 0x72, 0x8000 }, + { 0x73, 0x1114 }, + { 0x74, 0x0c00 }, + { 0x75, 0x1d00 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x83, 0x0000 }, + { 0x84, 0x0000 }, + { 0x85, 0x0008 }, + { 0x89, 0x0000 }, + { 0x8a, 0x0000 }, + { 0x8b, 0x0600 }, + { 0x8c, 0x0228 }, + { 0x8d, 0xa000 }, + { 0x8e, 0x0004 }, + { 0x8f, 0x1100 }, + { 0x90, 0x0646 }, + { 0x91, 0x0c00 }, + { 0x92, 0x0000 }, + { 0x93, 0x3000 }, + { 0xb0, 0x2080 }, + { 0xb1, 0x0000 }, + { 0xb4, 0x2206 }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xb8, 0x034b }, + { 0xb9, 0x0066 }, + { 0xba, 0x000b }, + { 0xbb, 0x0000 }, + { 0xbc, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x0000 }, + { 0xc0, 0x0400 }, + { 0xc2, 0x0000 }, + { 0xc4, 0x0000 }, + { 0xc5, 0x0000 }, + { 0xc6, 0x2000 }, + { 0xc8, 0x0000 }, + { 0xc9, 0x0000 }, + { 0xca, 0x0000 }, + { 0xcb, 0x0000 }, + { 0xcc, 0x0000 }, + { 0xcf, 0x0013 }, + { 0xd0, 0x0680 }, + { 0xd1, 0x1c17 }, + { 0xd2, 0x8c00 }, + { 0xd3, 0xaa20 }, + { 0xd6, 0x0400 }, + { 0xd9, 0x0809 }, + { 0xfe, 0x10ec }, + { 0xff, 0x6231 }, +}; + +static int rt5639_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, RT5639_RESET, 0); +} + +static bool rt5639_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5639_ranges); i++) + if ((reg >= rt5639_ranges[i].window_start && + reg <= rt5639_ranges[i].window_start + + rt5639_ranges[i].window_len) || + (reg >= rt5639_ranges[i].range_min && + reg <= rt5639_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5639_RESET: + case RT5639_ASRC_5: + case RT5639_EQ_CTRL1: + case RT5639_DRC_AGC_1: + case RT5639_IRQ_CTRL2: + case RT5639_INT_IRQ_ST: + case RT5639_PRIV_INDEX: + case RT5639_PRIV_DATA: + case RT5639_PGM_REG_ARR1: + case RT5639_PGM_REG_ARR3: + case RT5639_VENDOR_ID: + case RT5639_VENDOR_ID1: + case RT5639_VENDOR_ID2: + return true; + default: + return false; + } +} + +static bool rt5639_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5639_ranges); i++) + if ((reg >= rt5639_ranges[i].window_start && + reg <= rt5639_ranges[i].window_start + + rt5639_ranges[i].window_len) || + (reg >= rt5639_ranges[i].range_min && + reg <= rt5639_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5639_RESET: + case RT5639_SPK_VOL: + case RT5639_HP_VOL: + case RT5639_OUTPUT: + case RT5639_IN1_IN2: + case RT5639_IN3_IN4: + case RT5639_INL_INR_VOL: + case RT5639_DAC1_DIG_VOL: + case RT5639_DAC2_DIG_VOL: + case RT5639_DAC2_CTRL: + case RT5639_ADC_DIG_VOL: + case RT5639_ADC_DATA: + case RT5639_ADC_BST_VOL: + case RT5639_STO_ADC_MIXER: + case RT5639_MONO_ADC_MIXER: + case RT5639_AD_DA_MIXER: + case RT5639_STO_DAC_MIXER: + case RT5639_MONO_DAC_MIXER: + case RT5639_DIG_MIXER: + case RT5639_DIG_INF_DATA: + case RT5639_REC_L1_MIXER: + case RT5639_REC_L2_MIXER: + case RT5639_REC_R1_MIXER: + case RT5639_REC_R2_MIXER: + case RT5639_HPO_MIXER: + case RT5639_SPK_L_MIXER: + case RT5639_SPK_R_MIXER: + case RT5639_SPO_L_MIXER: + case RT5639_SPO_R_MIXER: + case RT5639_SPO_CLSD_RATIO: + case RT5639_OUT_L1_MIXER: + case RT5639_OUT_L2_MIXER: + case RT5639_OUT_L3_MIXER: + case RT5639_OUT_R1_MIXER: + case RT5639_OUT_R2_MIXER: + case RT5639_OUT_R3_MIXER: + case RT5639_LOUT_MIXER: + case RT5639_PWR_DIG1: + case RT5639_PWR_DIG2: + case RT5639_PWR_ANLG1: + case RT5639_PWR_ANLG2: + case RT5639_PWR_MIXER: + case RT5639_PWR_VOL: + case RT5639_PRIV_INDEX: + case RT5639_PRIV_DATA: + case RT5639_I2S1_SDP: + case RT5639_I2S2_SDP: + case RT5639_ADDA_CLK1: + case RT5639_DMIC: + case RT5639_GLB_CLK: + case RT5639_PLL_CTRL1: + case RT5639_PLL_CTRL2: + case RT5639_ASRC_1: + case RT5639_ASRC_2: + case RT5639_ASRC_3: + case RT5639_ASRC_4: + case RT5639_ASRC_5: + case RT5639_CLS_D_OVCD: + case RT5639_CLS_D_OUT: + case RT5639_DEPOP_M1: + case RT5639_DEPOP_M2: + case RT5639_DEPOP_M3: + case RT5639_CHARGE_PUMP: + case RT5639_PV_DET_SPK_G: + case RT5639_MICBIAS: + case RT5639_EQ_CTRL1: + case RT5639_EQ_CTRL2: + case RT5639_WIND_FILTER: + case RT5639_DRC_AGC_1: + case RT5639_DRC_AGC_2: + case RT5639_DRC_AGC_3: + case RT5639_JD_CTRL: + case RT5639_IRQ_CTRL1: + case RT5639_IRQ_CTRL2: + case RT5639_INT_IRQ_ST: + case RT5639_GPIO_CTRL1: + case RT5639_GPIO_CTRL2: + case RT5639_PGM_REG_ARR1: + case RT5639_PGM_REG_ARR2: + case RT5639_PGM_REG_ARR3: + case RT5639_PGM_REG_ARR4: + case RT5639_PGM_REG_ARR5: + case RT5639_ADJ_HPF: + case RT5639_HP_CALIB_AMP_DET: + case RT5639_SV_ZCD1: + case RT5639_DUMMY1: + case RT5639_DUMMY2: + case RT5639_DUMMY3: + case RT5639_VENDOR_ID: + case RT5639_VENDOR_ID1: + case RT5639_VENDOR_ID2: + 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), +}; + +/* Interface data select */ +static const char * const rt5639_data_select[] = { + "Normal", "left copy to right", "right copy to left", "Swap"}; + +static SOC_ENUM_SINGLE_DECL(rt5639_if1_dac_enum, RT5639_DIG_INF_DATA, + RT5639_IF1_DAC_SEL_SFT, rt5639_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5639_if1_adc_enum, RT5639_DIG_INF_DATA, + RT5639_IF1_ADC_SEL_SFT, rt5639_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5639_if2_dac_enum, RT5639_DIG_INF_DATA, + RT5639_IF2_DAC_SEL_SFT, rt5639_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5639_if2_adc_enum, RT5639_DIG_INF_DATA, + RT5639_IF2_ADC_SEL_SFT, rt5639_data_select); + +/* Class D speaker gain ratio */ +static const char * const rt5639_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x", + "2x", "2.11x", "2.22x", "2.33x", "2.44x", "2.55x", "2.66x", "2.77x"}; + +static SOC_ENUM_SINGLE_DECL(rt5639_clsd_spk_ratio_enum, + RT5639_CLS_D_OUT, RT5639_CLSD_RATIO_SFT, rt5639_clsd_spk_ratio); + +static const struct snd_kcontrol_new rt5639_snd_controls[] = { + /* Speaker Output Volume */ + SOC_DOUBLE("Speaker Channel Switch", RT5639_SPK_VOL, + RT5639_VOL_L_SFT, RT5639_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("Speaker Playback Volume", RT5639_SPK_VOL, + RT5639_L_VOL_SFT, RT5639_R_VOL_SFT, 39, 1, out_vol_tlv), + + /* Headphone Output Volume */ + SOC_DOUBLE("HP Channel Switch", RT5639_HP_VOL, + RT5639_VOL_L_SFT, RT5639_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("HP Playback Volume", RT5639_HP_VOL, + RT5639_L_VOL_SFT, RT5639_R_VOL_SFT, 39, 1, out_vol_tlv), + + /* OUTPUT Control */ + SOC_DOUBLE("OUT Playback Switch", RT5639_OUTPUT, + RT5639_L_MUTE_SFT, RT5639_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("OUT Channel Switch", RT5639_OUTPUT, + RT5639_VOL_L_SFT, RT5639_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5639_OUTPUT, + RT5639_L_VOL_SFT, RT5639_R_VOL_SFT, 39, 1, out_vol_tlv), + + /* DAC Digital Volume */ + SOC_DOUBLE("DAC2 Playback Switch", RT5639_DAC2_CTRL, + RT5639_M_DAC_L2_VOL_SFT, RT5639_M_DAC_R2_VOL_SFT, 1, 1), + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5639_DAC1_DIG_VOL, + RT5639_L_VOL_SFT, RT5639_R_VOL_SFT, 175, 0, dac_vol_tlv), + + /* IN1/IN2 Control */ + SOC_SINGLE_TLV("IN1 Boost", RT5639_IN1_IN2, RT5639_BST_SFT1, 8, 0, + bst_tlv), + SOC_SINGLE_TLV("IN2 Boost", RT5639_IN3_IN4, RT5639_BST_SFT2, 8, 0, + bst_tlv), + + /* INL/INR Volume Control */ + SOC_DOUBLE_TLV("IN Capture Volume", RT5639_INL_INR_VOL, + RT5639_INL_VOL_SFT, RT5639_INR_VOL_SFT, 31, 1, in_vol_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5639_ADC_DIG_VOL, + RT5639_L_MUTE_SFT, RT5639_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5639_ADC_DIG_VOL, + RT5639_L_VOL_SFT, RT5639_R_VOL_SFT, 127, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5639_ADC_DATA, + RT5639_L_VOL_SFT, RT5639_R_VOL_SFT, 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("ADC Boost Gain", RT5639_ADC_BST_VOL, + RT5639_ADC_L_BST_SFT, RT5639_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), + + /* Class D speaker gain ratio */ + SOC_ENUM("Class D SPK Ratio Control", rt5639_clsd_spk_ratio_enum), + + /* Data Swap */ + SOC_ENUM("ADC IF1 Data Switch", rt5639_if1_adc_enum), + SOC_ENUM("DAC IF1 Data Switch", rt5639_if1_dac_enum), + SOC_ENUM("ADC IF2 Data Switch", rt5639_if2_adc_enum), + SOC_ENUM("DAC IF2 Data Switch", rt5639_if2_dac_enum), +}; + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + int div[] = {2, 3, 4, 6, 8, 12}; + int i, idx = -EINVAL; + int rate, red, bound, temp; + + rate = rt5639->sysclk; + red = 3000000 * 12; + for (i = 0; i < ARRAY_SIZE(div); i++) { + bound = div[i] * 3000000; + if (rate > bound) + continue; + temp = bound - rate; + if (temp < red) { + red = temp; + idx = i; + } + } + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5639_DMIC, RT5639_DMIC_CLK_MASK, + idx << RT5639_DMIC_CLK_SFT); + return idx; +} + +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(source->codec, RT5639_GLB_CLK); + val &= RT5639_SCLK_SRC_MASK; + if (val == RT5639_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5639_sto_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5639_STO_ADC_MIXER, + RT5639_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5639_STO_ADC_MIXER, + RT5639_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_sto_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5639_STO_ADC_MIXER, + RT5639_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5639_STO_ADC_MIXER, + RT5639_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_mono_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5639_MONO_ADC_MIXER, + RT5639_M_MONO_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5639_MONO_ADC_MIXER, + RT5639_M_MONO_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_mono_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5639_MONO_ADC_MIXER, + RT5639_M_MONO_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5639_MONO_ADC_MIXER, + RT5639_M_MONO_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5639_AD_DA_MIXER, + RT5639_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INF1 Switch", RT5639_AD_DA_MIXER, + RT5639_M_IF1_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5639_AD_DA_MIXER, + RT5639_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INF1 Switch", RT5639_AD_DA_MIXER, + RT5639_M_IF1_DAC_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5639_STO_DAC_MIXER, + RT5639_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5639_STO_DAC_MIXER, + RT5639_M_DAC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5639_STO_DAC_MIXER, + RT5639_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5639_STO_DAC_MIXER, + RT5639_M_DAC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_mono_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5639_MONO_DAC_MIXER, + RT5639_M_DAC_L1_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5639_MONO_DAC_MIXER, + RT5639_M_DAC_L2_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5639_MONO_DAC_MIXER, + RT5639_M_DAC_R2_MONO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_mono_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5639_MONO_DAC_MIXER, + RT5639_M_DAC_R1_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5639_MONO_DAC_MIXER, + RT5639_M_DAC_R2_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5639_MONO_DAC_MIXER, + RT5639_M_DAC_L2_MONO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_dig_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5639_DIG_MIXER, + RT5639_M_STO_L_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5639_DIG_MIXER, + RT5639_M_DAC_L2_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_dig_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5639_DIG_MIXER, + RT5639_M_STO_R_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5639_DIG_MIXER, + RT5639_M_DAC_R2_DAC_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5639_rec_l_mix[] = { + SOC_DAPM_SINGLE("HPOL Switch", RT5639_REC_L2_MIXER, + RT5639_M_HP_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5639_REC_L2_MIXER, + RT5639_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST3 Switch", RT5639_REC_L2_MIXER, + RT5639_M_BST2_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5639_REC_L2_MIXER, + RT5639_M_BST4_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5639_REC_L2_MIXER, + RT5639_M_BST1_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXL Switch", RT5639_REC_L2_MIXER, + RT5639_M_OM_L_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_rec_r_mix[] = { + SOC_DAPM_SINGLE("HPOR Switch", RT5639_REC_R2_MIXER, + RT5639_M_HP_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5639_REC_R2_MIXER, + RT5639_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST3 Switch", RT5639_REC_R2_MIXER, + RT5639_M_BST2_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5639_REC_R2_MIXER, + RT5639_M_BST4_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5639_REC_R2_MIXER, + RT5639_M_BST1_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXR Switch", RT5639_REC_R2_MIXER, + RT5639_M_OM_R_RM_R_SFT, 1, 1), +}; + +/* Analog Output Mixer */ +static const struct snd_kcontrol_new rt5639_spk_l_mix[] = { + SOC_DAPM_SINGLE("REC MIXL Switch", RT5639_SPK_L_MIXER, + RT5639_M_RM_L_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5639_SPK_L_MIXER, + RT5639_M_IN_L_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5639_SPK_L_MIXER, + RT5639_M_DAC_L1_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5639_SPK_L_MIXER, + RT5639_M_DAC_L2_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXL Switch", RT5639_SPK_L_MIXER, + RT5639_M_OM_L_SM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_spk_r_mix[] = { + SOC_DAPM_SINGLE("REC MIXR Switch", RT5639_SPK_R_MIXER, + RT5639_M_RM_R_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5639_SPK_R_MIXER, + RT5639_M_IN_R_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5639_SPK_R_MIXER, + RT5639_M_DAC_R1_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5639_SPK_R_MIXER, + RT5639_M_DAC_R2_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXR Switch", RT5639_SPK_R_MIXER, + RT5639_M_OM_R_SM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5639_OUT_L3_MIXER, + RT5639_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5639_OUT_L3_MIXER, + RT5639_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXL Switch", RT5639_OUT_L3_MIXER, + RT5639_M_RM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5639_OUT_L3_MIXER, + RT5639_M_DAC_R2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5639_OUT_L3_MIXER, + RT5639_M_DAC_L2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5639_OUT_L3_MIXER, + RT5639_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5639_OUT_R3_MIXER, + RT5639_M_BST4_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5639_OUT_R3_MIXER, + RT5639_M_BST1_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5639_OUT_R3_MIXER, + RT5639_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXR Switch", RT5639_OUT_R3_MIXER, + RT5639_M_RM_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5639_OUT_R3_MIXER, + RT5639_M_DAC_L2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5639_OUT_R3_MIXER, + RT5639_M_DAC_R2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5639_OUT_R3_MIXER, + RT5639_M_DAC_R1_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_spo_l_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5639_SPO_L_MIXER, + RT5639_M_DAC_R1_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5639_SPO_L_MIXER, + RT5639_M_DAC_L1_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL R Switch", RT5639_SPO_L_MIXER, + RT5639_M_SV_R_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL L Switch", RT5639_SPO_L_MIXER, + RT5639_M_SV_L_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5639_SPO_L_MIXER, + RT5639_M_BST1_SPM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_spo_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5639_SPO_R_MIXER, + RT5639_M_DAC_R1_SPM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL R Switch", RT5639_SPO_R_MIXER, + RT5639_M_SV_R_SPM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5639_SPO_R_MIXER, + RT5639_M_BST1_SPM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_hpo_mix[] = { + SOC_DAPM_SINGLE("HPO MIX DAC1 Switch", RT5639_HPO_MIXER, + RT5639_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPO MIX HPVOL Switch", RT5639_HPO_MIXER, + RT5639_M_HPVOL_HM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_lout_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5639_LOUT_MIXER, + RT5639_M_DAC_L1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5639_LOUT_MIXER, + RT5639_M_DAC_R1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTVOL L Switch", RT5639_LOUT_MIXER, + RT5639_M_OV_L_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTVOL R Switch", RT5639_LOUT_MIXER, + RT5639_M_OV_R_LM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spk_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5639_SPK_VOL, + RT5639_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new spk_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5639_SPK_VOL, + RT5639_R_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5639_HP_VOL, + RT5639_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5639_HP_VOL, + RT5639_R_MUTE_SFT, 1, 1); + +/* Stereo ADC source */ +static const char * const rt5639_stereo_adc1_src[] = { + "DIG MIX", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5639_stereo_adc1_enum, RT5639_STO_ADC_MIXER, + RT5639_ADC_1_SRC_SFT, rt5639_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5639_sto_adc_1_mux = + SOC_DAPM_ENUM("Stereo ADC1 Mux", rt5639_stereo_adc1_enum); + +static const char * const rt5639_stereo_adc2_src[] = { + "DMIC1", "DMIC2", "DIG MIX" +}; + +static SOC_ENUM_SINGLE_DECL(rt5639_stereo_adc2_enum, RT5639_STO_ADC_MIXER, + RT5639_ADC_2_SRC_SFT, rt5639_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5639_sto_adc_2_mux = + SOC_DAPM_ENUM("Stereo ADC2 Mux", rt5639_stereo_adc2_enum); + +/* Mono ADC source */ +static const char * const rt5639_mono_adc_l1_src[] = { + "Mono DAC MIXL", "ADCL" +}; + +static SOC_ENUM_SINGLE_DECL(rt5639_mono_adc_l1_enum, RT5639_MONO_ADC_MIXER, + RT5639_MONO_ADC_L1_SRC_SFT, rt5639_mono_adc_l1_src); + +static const struct snd_kcontrol_new rt5639_mono_adc_l1_mux = + SOC_DAPM_ENUM("Mono ADC1 left source", rt5639_mono_adc_l1_enum); + +static const char * const rt5639_mono_adc_l2_src[] = { + "DMIC L1", "DMIC L2", "Mono DAC MIXL" +}; + +static SOC_ENUM_SINGLE_DECL(rt5639_mono_adc_l2_enum, RT5639_MONO_ADC_MIXER, + RT5639_MONO_ADC_L2_SRC_SFT, rt5639_mono_adc_l2_src); + +static const struct snd_kcontrol_new rt5639_mono_adc_l2_mux = + SOC_DAPM_ENUM("Mono ADC2 left source", rt5639_mono_adc_l2_enum); + +static const char * const rt5639_mono_adc_r1_src[] = { + "Mono DAC MIXR", "ADCR" +}; + +static SOC_ENUM_SINGLE_DECL(rt5639_mono_adc_r1_enum, RT5639_MONO_ADC_MIXER, + RT5639_MONO_ADC_R1_SRC_SFT, rt5639_mono_adc_r1_src); + +static const struct snd_kcontrol_new rt5639_mono_adc_r1_mux = + SOC_DAPM_ENUM("Mono ADC1 right source", rt5639_mono_adc_r1_enum); + +static const char * const rt5639_mono_adc_r2_src[] = { + "DMIC R1", "DMIC R2", "Mono DAC MIXR" +}; + +static SOC_ENUM_SINGLE_DECL(rt5639_mono_adc_r2_enum, RT5639_MONO_ADC_MIXER, + RT5639_MONO_ADC_R2_SRC_SFT, rt5639_mono_adc_r2_src); + +static const struct snd_kcontrol_new rt5639_mono_adc_r2_mux = + SOC_DAPM_ENUM("Mono ADC2 right source", rt5639_mono_adc_r2_enum); + +/* digital interface and iis interface map */ +static const char * const rt5639_dai_iis_map[] = { + "1:1|2:2", "1:2|2:1", "1:1|2:1", "1:2|2:2" +}; + +static int rt5639_dai_iis_map_values[] = { + 0, + 5, + 6, + 7, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5639_dai_iis_map_enum, + RT5639_I2S1_SDP, RT5639_I2S_IF_SFT, + 0x7, rt5639_dai_iis_map, + rt5639_dai_iis_map_values); + +static const struct snd_kcontrol_new rt5639_dai_mux = + SOC_DAPM_VALUE_ENUM("DAI select", rt5639_dai_iis_map_enum); + +/* SDI select */ +static const char * const rt5639_sdi_sel[] = { + "IF1", "IF2" +}; + +static SOC_ENUM_SINGLE_DECL(rt5639_sdi_sel_enum, RT5639_I2S2_SDP, + RT5639_I2S2_SDI_SFT, rt5639_sdi_sel); + +static const struct snd_kcontrol_new rt5639_sdi_mux = + SOC_DAPM_ENUM("SDI select", rt5639_sdi_sel_enum); + +static int rt5639_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, RT5639_GPIO_CTRL1, + RT5639_GP2_PIN_MASK | RT5639_GP3_PIN_MASK, + RT5639_GP2_PIN_DMIC1_SCL | RT5639_GP3_PIN_DMIC1_SDA); + snd_soc_update_bits(codec, RT5639_DMIC, + RT5639_DMIC_1L_LH_MASK | RT5639_DMIC_1R_LH_MASK | + RT5639_DMIC_1_DP_MASK, + RT5639_DMIC_1L_LH_FALLING | RT5639_DMIC_1R_LH_RISING | + RT5639_DMIC_1_DP_IN1P); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5639_set_dmic2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, RT5639_GPIO_CTRL1, + RT5639_GP2_PIN_MASK | RT5639_GP4_PIN_MASK, + RT5639_GP2_PIN_DMIC1_SCL | RT5639_GP4_PIN_DMIC2_SDA); + snd_soc_update_bits(codec, RT5639_DMIC, + RT5639_DMIC_2L_LH_MASK | RT5639_DMIC_2R_LH_MASK | + RT5639_DMIC_2_DP_MASK, + RT5639_DMIC_2L_LH_FALLING | RT5639_DMIC_2R_LH_RISING | + RT5639_DMIC_2_DP_IN1N); + break; + + default: + return 0; + } + + return 0; +} + +static void hp_amp_power_on(struct snd_soc_codec *codec) +{ + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + + /* depop parameters */ + regmap_update_bits(rt5639->regmap, RT5639_PR_BASE + + RT5639_CHPUMP_INT_REG1, 0x0700, 0x0200); + regmap_update_bits(rt5639->regmap, RT5639_DEPOP_M2, + RT5639_DEPOP_MASK, RT5639_DEPOP_MAN); + regmap_update_bits(rt5639->regmap, RT5639_DEPOP_M1, + RT5639_HP_CP_MASK | RT5639_HP_SG_MASK | RT5639_HP_CB_MASK, + RT5639_HP_CP_PU | RT5639_HP_SG_DIS | RT5639_HP_CB_PU); + regmap_write(rt5639->regmap, RT5639_PR_BASE + RT5639_HP_DCC_INT1, + 0x9f00); + /* headphone amp power on */ + regmap_update_bits(rt5639->regmap, RT5639_PWR_ANLG1, + RT5639_PWR_FV1 | RT5639_PWR_FV2, 0); + regmap_update_bits(rt5639->regmap, RT5639_PWR_ANLG1, + RT5639_PWR_HA, + RT5639_PWR_HA); + usleep_range(10000, 15000); + regmap_update_bits(rt5639->regmap, RT5639_PWR_ANLG1, + RT5639_PWR_FV1 | RT5639_PWR_FV2 , + RT5639_PWR_FV1 | RT5639_PWR_FV2); +} + +static void rt5639_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(rt5639->regmap, RT5639_DEPOP_M2, + RT5639_DEPOP_MASK | RT5639_DIG_DP_MASK, + RT5639_DEPOP_AUTO | RT5639_DIG_DP_EN); + regmap_update_bits(rt5639->regmap, RT5639_CHARGE_PUMP, + RT5639_PM_HP_MASK, RT5639_PM_HP_HV); + + regmap_update_bits(rt5639->regmap, RT5639_DEPOP_M3, + RT5639_CP_FQ1_MASK | RT5639_CP_FQ2_MASK | RT5639_CP_FQ3_MASK, + (RT5639_CP_FQ_192_KHZ << RT5639_CP_FQ1_SFT) | + (RT5639_CP_FQ_12_KHZ << RT5639_CP_FQ2_SFT) | + (RT5639_CP_FQ_192_KHZ << RT5639_CP_FQ3_SFT)); + + regmap_write(rt5639->regmap, RT5639_PR_BASE + + RT5639_MAMP_INT_REG2, 0x1c00); + regmap_update_bits(rt5639->regmap, RT5639_DEPOP_M1, + RT5639_HP_CP_MASK | RT5639_HP_SG_MASK, + RT5639_HP_CP_PD | RT5639_HP_SG_EN); + regmap_update_bits(rt5639->regmap, RT5639_PR_BASE + + RT5639_CHPUMP_INT_REG1, 0x0700, 0x0400); +} + +static int rt5639_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt5639_pmu_depop(codec); + rt5639->hp_mute = 0; + break; + + case SND_SOC_DAPM_PRE_PMD: + rt5639->hp_mute = 1; + usleep_range(70000, 75000); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5639_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power_on(codec); + break; + default: + return 0; + } + + return 0; +} + +static int rt5639_hp_post_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5639->hp_mute) + usleep_range(80000, 85000); + + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5639_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL1", RT5639_PWR_ANLG2, RT5639_PWR_PLL_BIT, + 0, NULL, 0), + /* Input Side */ + /* micbias */ + SND_SOC_DAPM_SUPPLY("LDO2", RT5639_PWR_ANLG1, RT5639_PWR_LDO2_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5639_PWR_ANLG2, RT5639_PWR_MB1_BIT, + 0, NULL, 0), + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("DMIC2"), + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + SND_SOC_DAPM_INPUT("IN3P"), + SND_SOC_DAPM_INPUT("IN3N"), + SND_SOC_DAPM_PGA("DMIC L1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC R1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC L2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC R2", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5639_DMIC, + RT5639_DMIC_1_EN_SFT, 0, rt5639_set_dmic1_event, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5639_DMIC, + RT5639_DMIC_2_EN_SFT, 0, rt5639_set_dmic2_event, + SND_SOC_DAPM_PRE_PMU), + /* Boost */ + SND_SOC_DAPM_PGA("BST1", RT5639_PWR_ANLG2, + RT5639_PWR_BST1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("BST2", RT5639_PWR_ANLG2, + RT5639_PWR_BST4_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("BST3", RT5639_PWR_ANLG2, + RT5639_PWR_BST2_BIT, 0, NULL, 0), + + /* Input Volume */ + SND_SOC_DAPM_PGA("INL VOL", RT5639_PWR_VOL, + RT5639_PWR_IN_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR VOL", RT5639_PWR_VOL, + RT5639_PWR_IN_R_BIT, 0, NULL, 0), + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5639_PWR_MIXER, RT5639_PWR_RM_L_BIT, 0, + rt5639_rec_l_mix, ARRAY_SIZE(rt5639_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5639_PWR_MIXER, RT5639_PWR_RM_R_BIT, 0, + rt5639_rec_r_mix, ARRAY_SIZE(rt5639_rec_r_mix)), + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, RT5639_PWR_DIG1, RT5639_PWR_ADC_L_BIT, + 0), + SND_SOC_DAPM_ADC("ADC R", NULL, RT5639_PWR_DIG1, RT5639_PWR_ADC_R_BIT, + 0), + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5639_sto_adc_2_mux), + SND_SOC_DAPM_MUX("Stereo ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5639_sto_adc_2_mux), + SND_SOC_DAPM_MUX("Stereo ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5639_sto_adc_1_mux), + SND_SOC_DAPM_MUX("Stereo ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5639_sto_adc_1_mux), + SND_SOC_DAPM_MUX("Mono ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5639_mono_adc_l2_mux), + SND_SOC_DAPM_MUX("Mono ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5639_mono_adc_l1_mux), + SND_SOC_DAPM_MUX("Mono ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5639_mono_adc_r1_mux), + SND_SOC_DAPM_MUX("Mono ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5639_mono_adc_r2_mux), + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("Stereo Filter", RT5639_PWR_DIG2, + RT5639_PWR_ADC_SF_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5639_sto_adc_l_mix, ARRAY_SIZE(rt5639_sto_adc_l_mix)), + SND_SOC_DAPM_MIXER("Stereo ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5639_sto_adc_r_mix, ARRAY_SIZE(rt5639_sto_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("Mono Left Filter", RT5639_PWR_DIG2, + RT5639_PWR_ADC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5639_mono_adc_l_mix, ARRAY_SIZE(rt5639_mono_adc_l_mix)), + SND_SOC_DAPM_SUPPLY("Mono Right Filter", RT5639_PWR_DIG2, + RT5639_PWR_ADC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5639_mono_adc_r_mix, ARRAY_SIZE(rt5639_mono_adc_r_mix)), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5639_PWR_DIG1, + RT5639_PWR_I2S1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5639_PWR_DIG1, + RT5639_PWR_I2S2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("DAI1 RX Mux", SND_SOC_NOPM, 0, 0, &rt5639_dai_mux), + SND_SOC_DAPM_MUX("DAI1 TX Mux", SND_SOC_NOPM, 0, 0, &rt5639_dai_mux), + SND_SOC_DAPM_MUX("DAI1 IF1 Mux", SND_SOC_NOPM, 0, 0, &rt5639_dai_mux), + SND_SOC_DAPM_MUX("DAI1 IF2 Mux", SND_SOC_NOPM, 0, 0, &rt5639_dai_mux), + SND_SOC_DAPM_MUX("SDI1 TX Mux", SND_SOC_NOPM, 0, 0, &rt5639_sdi_mux), + SND_SOC_DAPM_MUX("DAI2 RX Mux", SND_SOC_NOPM, 0, 0, &rt5639_dai_mux), + SND_SOC_DAPM_MUX("DAI2 TX Mux", SND_SOC_NOPM, 0, 0, &rt5639_dai_mux), + SND_SOC_DAPM_MUX("DAI2 IF1 Mux", SND_SOC_NOPM, 0, 0, &rt5639_dai_mux), + SND_SOC_DAPM_MUX("DAI2 IF2 Mux", SND_SOC_NOPM, 0, 0, &rt5639_dai_mux), + SND_SOC_DAPM_MUX("SDI2 TX Mux", SND_SOC_NOPM, 0, 0, &rt5639_sdi_mux), + /* 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), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5639_dac_l_mix, ARRAY_SIZE(rt5639_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5639_dac_r_mix, ARRAY_SIZE(rt5639_dac_r_mix)), + + /* DAC Mixer */ + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5639_sto_dac_l_mix, ARRAY_SIZE(rt5639_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5639_sto_dac_r_mix, ARRAY_SIZE(rt5639_sto_dac_r_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5639_mono_dac_l_mix, ARRAY_SIZE(rt5639_mono_dac_l_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5639_mono_dac_r_mix, ARRAY_SIZE(rt5639_mono_dac_r_mix)), + SND_SOC_DAPM_MIXER("DIG MIXL", SND_SOC_NOPM, 0, 0, + rt5639_dig_l_mix, ARRAY_SIZE(rt5639_dig_l_mix)), + SND_SOC_DAPM_MIXER("DIG MIXR", SND_SOC_NOPM, 0, 0, + rt5639_dig_r_mix, ARRAY_SIZE(rt5639_dig_r_mix)), + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5639_PWR_DIG1, RT5639_PWR_DAC_L1_BIT, + 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5639_PWR_DIG1, RT5639_PWR_DAC_R1_BIT, + 0), + SND_SOC_DAPM_SUPPLY("DAC L2 Filter", RT5639_PWR_DIG1, + RT5639_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R2 Filter", RT5639_PWR_DIG1, + RT5639_PWR_DAC_R2_BIT, 0, NULL, 0), + /* SPK/OUT Mixer */ + SND_SOC_DAPM_MIXER("SPK MIXL", RT5639_PWR_MIXER, RT5639_PWR_SM_L_BIT, + 0, rt5639_spk_l_mix, ARRAY_SIZE(rt5639_spk_l_mix)), + SND_SOC_DAPM_MIXER("SPK MIXR", RT5639_PWR_MIXER, RT5639_PWR_SM_R_BIT, + 0, rt5639_spk_r_mix, ARRAY_SIZE(rt5639_spk_r_mix)), + SND_SOC_DAPM_MIXER("OUT MIXL", RT5639_PWR_MIXER, RT5639_PWR_OM_L_BIT, + 0, rt5639_out_l_mix, ARRAY_SIZE(rt5639_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5639_PWR_MIXER, RT5639_PWR_OM_R_BIT, + 0, rt5639_out_r_mix, ARRAY_SIZE(rt5639_out_r_mix)), + /* Ouput Volume */ + SND_SOC_DAPM_PGA("SPKVOL L", RT5639_PWR_VOL, + RT5639_PWR_SV_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPKVOL R", RT5639_PWR_VOL, + RT5639_PWR_SV_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTVOL L", RT5639_PWR_VOL, + RT5639_PWR_OV_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTVOL R", RT5639_PWR_VOL, + RT5639_PWR_OV_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL L", RT5639_PWR_VOL, + RT5639_PWR_HV_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL R", RT5639_PWR_VOL, + RT5639_PWR_HV_R_BIT, 0, NULL, 0), + /* SPO/HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("SPOL MIX", SND_SOC_NOPM, 0, + 0, rt5639_spo_l_mix, ARRAY_SIZE(rt5639_spo_l_mix)), + SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0, + 0, rt5639_spo_r_mix, ARRAY_SIZE(rt5639_spo_r_mix)), + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, + rt5639_hpo_mix, ARRAY_SIZE(rt5639_hpo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, + rt5639_hpo_mix, ARRAY_SIZE(rt5639_hpo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", RT5639_PWR_ANLG1, RT5639_PWR_LM_BIT, 0, + rt5639_lout_mix, ARRAY_SIZE(rt5639_lout_mix)), + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5639_hp_power_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5639_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5639_PWR_ANLG1, + RT5639_PWR_HP_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5639_PWR_ANLG1, + RT5639_PWR_HP_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5639_PWR_DIG1, + RT5639_PWR_CLS_D_BIT, 0, NULL, 0), + + /* Output Switch */ + SND_SOC_DAPM_SWITCH("Speaker L Playback", SND_SOC_NOPM, 0, 0, + &spk_l_enable_control), + SND_SOC_DAPM_SWITCH("Speaker R Playback", SND_SOC_NOPM, 0, 0, + &spk_r_enable_control), + SND_SOC_DAPM_SWITCH("HP L Playback", SND_SOC_NOPM, 0, 0, + &hp_l_enable_control), + SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0, + &hp_r_enable_control), + SND_SOC_DAPM_POST("HP Post", rt5639_hp_post_event), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOLP"), + SND_SOC_DAPM_OUTPUT("SPOLN"), + SND_SOC_DAPM_OUTPUT("SPORP"), + SND_SOC_DAPM_OUTPUT("SPORN"), + 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 rt5639_dapm_routes[] = { + {"IN1P", NULL, "LDO2"}, + {"IN2P", NULL, "LDO2"}, + {"IN3P", NULL, "LDO2"}, + + {"DMIC L1", NULL, "DMIC1"}, + {"DMIC R1", NULL, "DMIC1"}, + {"DMIC L2", NULL, "DMIC2"}, + {"DMIC R2", NULL, "DMIC2"}, + + {"BST1", NULL, "IN1P"}, + {"BST1", NULL, "IN1N"}, + {"BST2", NULL, "IN2P"}, + {"BST2", NULL, "IN2N"}, + {"BST3", NULL, "IN3P"}, + {"BST3", NULL, "IN3N"}, + + {"INL VOL", NULL, "IN2P"}, + {"INR VOL", NULL, "IN2N"}, + + {"RECMIXL", "HPOL Switch", "HPOL"}, + {"RECMIXL", "INL Switch", "INL VOL"}, + {"RECMIXL", "BST3 Switch", "BST3"}, + {"RECMIXL", "BST2 Switch", "BST2"}, + {"RECMIXL", "BST1 Switch", "BST1"}, + {"RECMIXL", "OUT MIXL Switch", "OUT MIXL"}, + + {"RECMIXR", "HPOR Switch", "HPOR"}, + {"RECMIXR", "INR Switch", "INR VOL"}, + {"RECMIXR", "BST3 Switch", "BST3"}, + {"RECMIXR", "BST2 Switch", "BST2"}, + {"RECMIXR", "BST1 Switch", "BST1"}, + {"RECMIXR", "OUT MIXR Switch", "OUT MIXR"}, + + {"ADC L", NULL, "RECMIXL"}, + {"ADC R", NULL, "RECMIXR"}, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC1 Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC1 Power"}, + {"DMIC L2", NULL, "DMIC CLK"}, + {"DMIC L2", NULL, "DMIC2 Power"}, + {"DMIC R2", NULL, "DMIC CLK"}, + {"DMIC R2", NULL, "DMIC2 Power"}, + + {"Stereo ADC L2 Mux", "DMIC1", "DMIC L1"}, + {"Stereo ADC L2 Mux", "DMIC2", "DMIC L2"}, + {"Stereo ADC L2 Mux", "DIG MIX", "DIG MIXL"}, + {"Stereo ADC L1 Mux", "ADC", "ADC L"}, + {"Stereo ADC L1 Mux", "DIG MIX", "DIG MIXL"}, + + {"Stereo ADC R1 Mux", "ADC", "ADC R"}, + {"Stereo ADC R1 Mux", "DIG MIX", "DIG MIXR"}, + {"Stereo ADC R2 Mux", "DMIC1", "DMIC R1"}, + {"Stereo ADC R2 Mux", "DMIC2", "DMIC R2"}, + {"Stereo ADC R2 Mux", "DIG MIX", "DIG MIXR"}, + + {"Mono ADC L2 Mux", "DMIC L1", "DMIC L1"}, + {"Mono ADC L2 Mux", "DMIC L2", "DMIC L2"}, + {"Mono ADC L2 Mux", "Mono DAC MIXL", "Mono DAC MIXL"}, + {"Mono ADC L1 Mux", "Mono DAC MIXL", "Mono DAC MIXL"}, + {"Mono ADC L1 Mux", "ADCL", "ADC L"}, + + {"Mono ADC R1 Mux", "Mono DAC MIXR", "Mono DAC MIXR"}, + {"Mono ADC R1 Mux", "ADCR", "ADC R"}, + {"Mono ADC R2 Mux", "DMIC R1", "DMIC R1"}, + {"Mono ADC R2 Mux", "DMIC R2", "DMIC R2"}, + {"Mono ADC R2 Mux", "Mono DAC MIXR", "Mono DAC MIXR"}, + + {"Stereo ADC MIXL", "ADC1 Switch", "Stereo ADC L1 Mux"}, + {"Stereo ADC MIXL", "ADC2 Switch", "Stereo ADC L2 Mux"}, + {"Stereo ADC MIXL", NULL, "Stereo Filter"}, + {"Stereo Filter", NULL, "PLL1", is_sys_clk_from_pll}, + + {"Stereo ADC MIXR", "ADC1 Switch", "Stereo ADC R1 Mux"}, + {"Stereo ADC MIXR", "ADC2 Switch", "Stereo ADC R2 Mux"}, + {"Stereo ADC MIXR", NULL, "Stereo Filter"}, + {"Stereo Filter", NULL, "PLL1", is_sys_clk_from_pll}, + + {"Mono ADC MIXL", "ADC1 Switch", "Mono ADC L1 Mux"}, + {"Mono ADC MIXL", "ADC2 Switch", "Mono ADC L2 Mux"}, + {"Mono ADC MIXL", NULL, "Mono Left Filter"}, + {"Mono Left Filter", NULL, "PLL1", is_sys_clk_from_pll}, + + {"Mono ADC MIXR", "ADC1 Switch", "Mono ADC R1 Mux"}, + {"Mono ADC MIXR", "ADC2 Switch", "Mono ADC R2 Mux"}, + {"Mono ADC MIXR", NULL, "Mono Right Filter"}, + {"Mono Right Filter", NULL, "PLL1", is_sys_clk_from_pll}, + + {"IF2 ADC L", NULL, "Mono ADC MIXL"}, + {"IF2 ADC R", NULL, "Mono ADC MIXR"}, + {"IF1 ADC L", NULL, "Stereo ADC MIXL"}, + {"IF1 ADC R", NULL, "Stereo ADC MIXR"}, + + {"IF1 ADC", NULL, "I2S1"}, + {"IF1 ADC", NULL, "IF1 ADC L"}, + {"IF1 ADC", NULL, "IF1 ADC R"}, + {"IF2 ADC", NULL, "I2S2"}, + {"IF2 ADC", NULL, "IF2 ADC L"}, + {"IF2 ADC", NULL, "IF2 ADC R"}, + + {"DAI1 TX Mux", "1:1|2:2", "IF1 ADC"}, + {"DAI1 TX Mux", "1:2|2:1", "IF2 ADC"}, + {"DAI1 IF1 Mux", "1:1|2:1", "IF1 ADC"}, + {"DAI1 IF2 Mux", "1:1|2:1", "IF2 ADC"}, + {"SDI1 TX Mux", "IF1", "DAI1 IF1 Mux"}, + {"SDI1 TX Mux", "IF2", "DAI1 IF2 Mux"}, + + {"DAI2 TX Mux", "1:2|2:1", "IF1 ADC"}, + {"DAI2 TX Mux", "1:1|2:2", "IF2 ADC"}, + {"DAI2 IF1 Mux", "1:2|2:2", "IF1 ADC"}, + {"DAI2 IF2 Mux", "1:2|2:2", "IF2 ADC"}, + {"SDI2 TX Mux", "IF1", "DAI2 IF1 Mux"}, + {"SDI2 TX Mux", "IF2", "DAI2 IF2 Mux"}, + + {"AIF1TX", NULL, "DAI1 TX Mux"}, + {"AIF1TX", NULL, "SDI1 TX Mux"}, + {"AIF2TX", NULL, "DAI2 TX Mux"}, + {"AIF2TX", NULL, "SDI2 TX Mux"}, + + {"DAI1 RX Mux", "1:1|2:2", "AIF1RX"}, + {"DAI1 RX Mux", "1:1|2:1", "AIF1RX"}, + {"DAI1 RX Mux", "1:2|2:1", "AIF2RX"}, + {"DAI1 RX Mux", "1:2|2:2", "AIF2RX"}, + + {"DAI2 RX Mux", "1:2|2:1", "AIF1RX"}, + {"DAI2 RX Mux", "1:1|2:1", "AIF1RX"}, + {"DAI2 RX Mux", "1:1|2:2", "AIF2RX"}, + {"DAI2 RX Mux", "1:2|2:2", "AIF2RX"}, + + {"IF1 DAC", NULL, "I2S1"}, + {"IF1 DAC", NULL, "DAI1 RX Mux"}, + {"IF2 DAC", NULL, "I2S2"}, + {"IF2 DAC", NULL, "DAI2 RX Mux"}, + + {"IF1 DAC L", NULL, "IF1 DAC"}, + {"IF1 DAC R", NULL, "IF1 DAC"}, + {"IF2 DAC L", NULL, "IF2 DAC"}, + {"IF2 DAC R", NULL, "IF2 DAC"}, + + {"DAC MIXL", "Stereo ADC Switch", "Stereo ADC MIXL"}, + {"DAC MIXL", "INF1 Switch", "IF1 DAC L"}, + {"DAC MIXR", "Stereo ADC Switch", "Stereo ADC MIXR"}, + {"DAC MIXR", "INF1 Switch", "IF1 DAC R"}, + + {"Stereo DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, + {"Stereo DAC MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"Stereo DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "IF2 DAC R"}, + + {"Mono DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, + {"Mono DAC MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"Mono DAC MIXL", "DAC R2 Switch", "IF2 DAC R"}, + {"Mono DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, + {"Mono DAC MIXR", "DAC R2 Switch", "IF2 DAC R"}, + {"Mono DAC MIXR", "DAC L2 Switch", "IF2 DAC L"}, + + {"DIG MIXL", "DAC L1 Switch", "DAC MIXL"}, + {"DIG MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"DIG MIXR", "DAC R1 Switch", "DAC MIXR"}, + {"DIG MIXR", "DAC R2 Switch", "IF2 DAC R"}, + + {"IF2 DAC L", NULL, "DAC L2 Filter"}, + {"IF2 DAC R", NULL, "DAC R2 Filter"}, + + {"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}, + + {"SPK MIXL", "REC MIXL Switch", "RECMIXL"}, + {"SPK MIXL", "INL Switch", "INL VOL"}, + {"SPK MIXL", "DAC L1 Switch", "DAC L1"}, + {"SPK MIXL", "OUT MIXL Switch", "OUT MIXL"}, + {"SPK MIXR", "REC MIXR Switch", "RECMIXR"}, + {"SPK MIXR", "INR Switch", "INR VOL"}, + {"SPK MIXR", "DAC R1 Switch", "DAC R1"}, + {"SPK MIXR", "OUT MIXR Switch", "OUT MIXR"}, + + {"OUT MIXL", "BST1 Switch", "BST1"}, + {"OUT MIXL", "INL Switch", "INL 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", "INR Switch", "INR VOL"}, + {"OUT MIXR", "REC MIXR Switch", "RECMIXR"}, + {"OUT MIXR", "DAC R1 Switch", "DAC R1"}, + + {"SPKVOL L", NULL, "SPK MIXL"}, + {"SPKVOL R", NULL, "SPK MIXR"}, + {"HPOVOL L", NULL, "OUT MIXL"}, + {"HPOVOL R", NULL, "OUT MIXR"}, + {"OUTVOL L", NULL, "OUT MIXL"}, + {"OUTVOL R", NULL, "OUT MIXR"}, + + {"SPOL MIX", "DAC R1 Switch", "DAC R1"}, + {"SPOL MIX", "DAC L1 Switch", "DAC L1"}, + {"SPOL MIX", "SPKVOL R Switch", "SPKVOL R"}, + {"SPOL MIX", "SPKVOL L Switch", "SPKVOL L"}, + {"SPOL MIX", "BST1 Switch", "BST1"}, + {"SPOR MIX", "DAC R1 Switch", "DAC R1"}, + {"SPOR MIX", "SPKVOL R Switch", "SPKVOL R"}, + {"SPOR MIX", "BST1 Switch", "BST1"}, + + {"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"}, + {"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"}, + {"HPO MIX L", NULL, "HP L Amp"}, + {"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"}, + {"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"}, + {"HPO MIX R", NULL, "HP R Amp"}, + + {"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 L"}, + {"HP Amp", NULL, "HPO MIX R"}, + + {"Speaker L Playback", "Switch", "SPOL MIX"}, + {"Speaker R Playback", "Switch", "SPOR MIX"}, + {"SPOLP", NULL, "Speaker L Playback"}, + {"SPOLN", NULL, "Speaker L Playback"}, + {"SPORP", NULL, "Speaker R Playback"}, + {"SPORN", NULL, "Speaker R Playback"}, + + {"SPOLP", NULL, "Improve SPK Amp Drv"}, + {"SPOLN", NULL, "Improve SPK Amp Drv"}, + {"SPORP", NULL, "Improve SPK Amp Drv"}, + {"SPORN", NULL, "Improve SPK Amp Drv"}, + + {"HPOL", NULL, "Improve HP Amp Drv"}, + {"HPOR", NULL, "Improve HP Amp Drv"}, + + {"HP L Playback", "Switch", "HP Amp"}, + {"HP R Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HP L Playback"}, + {"HPOR", NULL, "HP R Playback"}, + {"LOUTL", NULL, "LOUT MIX"}, + {"LOUTR", NULL, "LOUT MIX"}, +}; + +static int get_sdp_info(struct snd_soc_codec *codec, int dai_id) +{ + int ret = 0, val; + + if (codec == NULL) + return -EINVAL; + + val = snd_soc_read(codec, RT5639_I2S1_SDP); + val = (val & RT5639_I2S_IF_MASK) >> RT5639_I2S_IF_SFT; + switch (dai_id) { + case RT5639_AIF1: + switch (val) { + case RT5639_IF_123: + case RT5639_IF_132: + ret |= RT5639_U_IF1; + break; + case RT5639_IF_113: + ret |= RT5639_U_IF1; + case RT5639_IF_312: + case RT5639_IF_213: + ret |= RT5639_U_IF2; + break; + } + break; + + case RT5639_AIF2: + switch (val) { + case RT5639_IF_231: + case RT5639_IF_213: + ret |= RT5639_U_IF1; + break; + case RT5639_IF_223: + ret |= RT5639_U_IF1; + case RT5639_IF_123: + case RT5639_IF_321: + ret |= RT5639_U_IF2; + break; + } + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int get_clk_info(int sclk, int rate) +{ + int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + + if (sclk <= 0 || rate <= 0) + return -EINVAL; + + rate = rate << 8; + for (i = 0; i < ARRAY_SIZE(pd); i++) + if (sclk == rate * pd[i]) + return i; + + return -EINVAL; +} + +static int rt5639_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 rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int dai_sel, pre_div, bclk_ms, frame_size; + + rt5639->lrck[dai->id] = params_rate(params); + pre_div = get_clk_info(rt5639->sysclk, rt5639->lrck[dai->id]); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5639->lrck[dai->id], dai->id); + 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 frame_size; + } + if (frame_size > 32) + bclk_ms = 1; + else + bclk_ms = 0; + rt5639->bclk[dai->id] = rt5639->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5639->bclk[dai->id], rt5639->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 |= RT5639_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= RT5639_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + val_len |= RT5639_I2S_DL_8; + break; + default: + return -EINVAL; + } + + dai_sel = get_sdp_info(codec, dai->id); + if (dai_sel < 0) { + dev_err(codec->dev, "Failed to get sdp info: %d\n", dai_sel); + return -EINVAL; + } + if (dai_sel & RT5639_U_IF1) { + mask_clk = RT5639_I2S_BCLK_MS1_MASK | RT5639_I2S_PD1_MASK; + val_clk = bclk_ms << RT5639_I2S_BCLK_MS1_SFT | + pre_div << RT5639_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5639_I2S1_SDP, + RT5639_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5639_ADDA_CLK1, mask_clk, val_clk); + } + if (dai_sel & RT5639_U_IF2) { + mask_clk = RT5639_I2S_BCLK_MS2_MASK | RT5639_I2S_PD2_MASK; + val_clk = bclk_ms << RT5639_I2S_BCLK_MS2_SFT | + pre_div << RT5639_I2S_PD2_SFT; + snd_soc_update_bits(codec, RT5639_I2S2_SDP, + RT5639_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5639_ADDA_CLK1, mask_clk, val_clk); + } + + return 0; +} + +static int rt5639_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + int dai_sel; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5639->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5639_I2S_MS_S; + rt5639->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 |= RT5639_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 |= RT5639_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5639_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5639_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + dai_sel = get_sdp_info(codec, dai->id); + if (dai_sel < 0) { + dev_err(codec->dev, "Failed to get sdp info: %d\n", dai_sel); + return -EINVAL; + } + if (dai_sel & RT5639_U_IF1) { + snd_soc_update_bits(codec, RT5639_I2S1_SDP, + RT5639_I2S_MS_MASK | RT5639_I2S_BP_MASK | + RT5639_I2S_DF_MASK, reg_val); + } + if (dai_sel & RT5639_U_IF2) { + snd_soc_update_bits(codec, RT5639_I2S2_SDP, + RT5639_I2S_MS_MASK | RT5639_I2S_BP_MASK | + RT5639_I2S_DF_MASK, reg_val); + } + + return 0; +} + +static int rt5639_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5639->sysclk && clk_id == rt5639->sysclk_src) + return 0; + + switch (clk_id) { + case RT5639_SCLK_S_MCLK: + reg_val |= RT5639_SCLK_SRC_MCLK; + break; + case RT5639_SCLK_S_PLL1: + reg_val |= RT5639_SCLK_SRC_PLL1; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5639_GLB_CLK, + RT5639_SCLK_SRC_MASK, reg_val); + rt5639->sysclk = freq; + rt5639->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + return 0; +} + +/** + * rt5639_pll_calc - Calculate PLL M/N/K code. + * @freq_in: external clock provided to codec. + * @freq_out: target clock which codec works on. + * @pll_code: Pointer to structure with M, N, K and bypass flag. + * + * Calculate M/N/K code to configure PLL for codec. And K is assigned to 2 + * which make calculation more efficiently. + * + * Returns 0 for success or negative error code. + */ +static int rt5639_pll_calc(const unsigned int freq_in, + const unsigned int freq_out, struct rt5639_pll_code *pll_code) +{ + int max_n = RT5639_PLL_N_MAX, max_m = RT5639_PLL_M_MAX; + int n = 0, m = 0, red, n_t, m_t, in_t, out_t; + int red_t = abs(freq_out - freq_in); + bool bypass = false; + + if (RT5639_PLL_INP_MAX < freq_in || RT5639_PLL_INP_MIN > freq_in) + return -EINVAL; + + for (n_t = 0; n_t <= max_n; n_t++) { + in_t = (freq_in >> 1) + (freq_in >> 2) * n_t; + if (in_t < 0) + continue; + if (in_t == freq_out) { + bypass = true; + n = n_t; + goto code_find; + } + for (m_t = 0; m_t <= max_m; m_t++) { + out_t = in_t / (m_t + 2); + red = abs(out_t - freq_out); + if (red < red_t) { + n = n_t; + m = m_t; + if (red == 0) + goto code_find; + red_t = red; + } + } + } + pr_debug("Only get approximation about PLL\n"); + +code_find: + pll_code->m_bp = bypass; + pll_code->m_code = m; + pll_code->n_code = n; + pll_code->k_code = 2; + return 0; +} + +static int rt5639_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 rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + struct rt5639_pll_code *pll_code = &rt5639->pll_code; + int ret, dai_sel; + + if (source == rt5639->pll_src && freq_in == rt5639->pll_in && + freq_out == rt5639->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5639->pll_in = 0; + rt5639->pll_out = 0; + snd_soc_update_bits(codec, RT5639_GLB_CLK, + RT5639_SCLK_SRC_MASK, RT5639_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5639_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5639_GLB_CLK, + RT5639_PLL1_SRC_MASK, RT5639_PLL1_SRC_MCLK); + break; + case RT5639_PLL1_S_BCLK1: + case RT5639_PLL1_S_BCLK2: + dai_sel = get_sdp_info(codec, dai->id); + if (dai_sel < 0) { + dev_err(codec->dev, + "Failed to get sdp info: %d\n", dai_sel); + return -EINVAL; + } + if (dai_sel & RT5639_U_IF1) { + snd_soc_update_bits(codec, RT5639_GLB_CLK, + RT5639_PLL1_SRC_MASK, RT5639_PLL1_SRC_BCLK1); + } + if (dai_sel & RT5639_U_IF2) { + snd_soc_update_bits(codec, RT5639_GLB_CLK, + RT5639_PLL1_SRC_MASK, RT5639_PLL1_SRC_BCLK2); + } + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rt5639_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=2\n", pll_code->m_bp, + (pll_code->m_bp ? 0 : pll_code->m_code), pll_code->n_code); + + snd_soc_write(codec, RT5639_PLL_CTRL1, + pll_code->n_code << RT5639_PLL_N_SFT | pll_code->k_code); + snd_soc_write(codec, RT5639_PLL_CTRL2, + (pll_code->m_bp ? 0 : pll_code->m_code) << RT5639_PLL_M_SFT | + pll_code->m_bp << RT5639_PLL_M_BP_SFT); + + rt5639->pll_in = freq_in; + rt5639->pll_out = freq_out; + rt5639->pll_src = source; + + return 0; +} + +static int rt5639_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + switch (level) { + case SND_SOC_BIAS_STANDBY: + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { + regcache_cache_only(rt5639->regmap, false); + snd_soc_update_bits(codec, RT5639_PWR_ANLG1, + RT5639_PWR_VREF1 | RT5639_PWR_MB | + RT5639_PWR_BG | RT5639_PWR_VREF2, + RT5639_PWR_VREF1 | RT5639_PWR_MB | + RT5639_PWR_BG | RT5639_PWR_VREF2); + usleep_range(10000, 15000); + snd_soc_update_bits(codec, RT5639_PWR_ANLG1, + RT5639_PWR_FV1 | RT5639_PWR_FV2, + RT5639_PWR_FV1 | RT5639_PWR_FV2); + regcache_sync(rt5639->regmap); + snd_soc_update_bits(codec, RT5639_DUMMY1, + 0x0301, 0x0301); + snd_soc_update_bits(codec, RT5639_MICBIAS, + 0x0030, 0x0030); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, RT5639_DEPOP_M1, 0x0004); + snd_soc_write(codec, RT5639_DEPOP_M2, 0x1100); + snd_soc_update_bits(codec, RT5639_DUMMY1, 0x1, 0); + snd_soc_write(codec, RT5639_PWR_DIG1, 0x0000); + snd_soc_write(codec, RT5639_PWR_DIG2, 0x0000); + snd_soc_write(codec, RT5639_PWR_VOL, 0x0000); + snd_soc_write(codec, RT5639_PWR_MIXER, 0x0000); + snd_soc_write(codec, RT5639_PWR_ANLG1, 0x0000); + snd_soc_write(codec, RT5639_PWR_ANLG2, 0x0000); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt5639_probe(struct snd_soc_codec *codec) +{ + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + int ret; + + rt5639->codec = codec; + codec->control_data = rt5639->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + rt5639_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_update_bits(codec, RT5639_DUMMY1, 0x0301, 0x0301); + snd_soc_update_bits(codec, RT5639_MICBIAS, 0x0030, 0x0030); + + return 0; +} + +static int rt5639_remove(struct snd_soc_codec *codec) +{ + rt5639_reset(codec); + + return 0; +} + +#ifdef CONFIG_PM +static int rt5639_suspend(struct snd_soc_codec *codec) +{ + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + + rt5639_set_bias_level(codec, SND_SOC_BIAS_OFF); + rt5639_reset(codec); + regcache_cache_only(rt5639->regmap, true); + regcache_mark_dirty(rt5639->regmap); + if (gpio_is_valid(rt5639->pdata.ldo1_en)) + gpio_set_value_cansleep(rt5639->pdata.ldo1_en, 0); + + return 0; +} + +static int rt5639_resume(struct snd_soc_codec *codec) +{ + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(rt5639->pdata.ldo1_en)) { + gpio_set_value_cansleep(rt5639->pdata.ldo1_en, 1); + msleep(400); + } + + return 0; +} +#else +#define rt5639_suspend NULL +#define rt5639_resume NULL +#endif + +#define RT5639_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5639_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt5639_aif_dai_ops = { + .hw_params = rt5639_hw_params, + .set_fmt = rt5639_set_dai_fmt, + .set_sysclk = rt5639_set_dai_sysclk, + .set_pll = rt5639_set_dai_pll, +}; + +static struct snd_soc_dai_driver rt5639_dai[] = { + { + .name = "rt5639-aif1", + .id = RT5639_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5639_STEREO_RATES, + .formats = RT5639_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5639_STEREO_RATES, + .formats = RT5639_FORMATS, + }, + .ops = &rt5639_aif_dai_ops, + }, + { + .name = "rt5639-aif2", + .id = RT5639_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5639_STEREO_RATES, + .formats = RT5639_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5639_STEREO_RATES, + .formats = RT5639_FORMATS, + }, + .ops = &rt5639_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5639 = { + .probe = rt5639_probe, + .remove = rt5639_remove, + .suspend = rt5639_suspend, + .resume = rt5639_resume, + .set_bias_level = rt5639_set_bias_level, + .idle_bias_off = true, + .controls = rt5639_snd_controls, + .num_controls = ARRAY_SIZE(rt5639_snd_controls), + .dapm_widgets = rt5639_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5639_dapm_widgets), + .dapm_routes = rt5639_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5639_dapm_routes), +}; + +static const struct regmap_config rt5639_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = RT5639_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5639_ranges) * + RT5639_PR_SPACING), + .volatile_reg = rt5639_volatile_register, + .readable_reg = rt5639_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5639_reg, + .num_reg_defaults = ARRAY_SIZE(rt5639_reg), + .ranges = rt5639_ranges, + .num_ranges = ARRAY_SIZE(rt5639_ranges), +}; + +static const struct i2c_device_id rt5639_i2c_id[] = { + { "rt5639", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5639_i2c_id); + +static int rt5639_parse_dt(struct rt5639_priv *rt5639, struct device_node *np) +{ + rt5639->pdata.in1_diff = of_property_read_bool(np, + "realtek,in1-differential"); + rt5639->pdata.in2_diff = of_property_read_bool(np, + "realtek,in2-differential"); + + rt5639->pdata.ldo1_en = of_get_named_gpio(np, + "realtek,ldo1-en-gpios", 0); + /* + * LDO1_EN is optional (it may be statically tied on the board). + * -ENOENT means that the property doesn't exist, i.e. there is no + * GPIO, so is not an error. Any other error code means the property + * exists, but could not be parsed. + */ + if (!gpio_is_valid(rt5639->pdata.ldo1_en) && + (rt5639->pdata.ldo1_en != -ENOENT)) + return rt5639->pdata.ldo1_en; + + return 0; +} + +static int rt5639_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5639_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5639_priv *rt5639; + int ret; + unsigned int val; + + rt5639 = devm_kzalloc(&i2c->dev, + sizeof(struct rt5639_priv), + GFP_KERNEL); + if (NULL == rt5639) + return -ENOMEM; + i2c_set_clientdata(i2c, rt5639); + + if (pdata) { + rt5639->pdata = *pdata; + /* + * Translate zero'd out (default) pdata value to an invalid + * GPIO ID. This makes the pdata and DT paths consistent in + * terms of the value left in this field when no GPIO is + * specified, but means we can't actually use GPIO 0. + */ + if (!rt5639->pdata.ldo1_en) + rt5639->pdata.ldo1_en = -EINVAL; + } else if (i2c->dev.of_node) { + ret = rt5639_parse_dt(rt5639, i2c->dev.of_node); + if (ret) + return ret; + } else + rt5639->pdata.ldo1_en = -EINVAL; + + rt5639->regmap = devm_regmap_init_i2c(i2c, &rt5639_regmap); + if (IS_ERR(rt5639->regmap)) { + ret = PTR_ERR(rt5639->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + if (gpio_is_valid(rt5639->pdata.ldo1_en)) { + ret = devm_gpio_request_one(&i2c->dev, rt5639->pdata.ldo1_en, + GPIOF_OUT_INIT_HIGH, + "RT5639 LDO1_EN"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request LDO1_EN %d: %d\n", + rt5639->pdata.ldo1_en, ret); + return ret; + } + msleep(400); + } + + regmap_read(rt5639->regmap, RT5639_VENDOR_ID2, &val); + if (val != RT5639_DEVICE_ID) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt5640/39\n", val); + return -ENODEV; + } + + regmap_write(rt5639->regmap, RT5639_RESET, 0); + + ret = regmap_register_patch(rt5639->regmap, init_list, + ARRAY_SIZE(init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5639->pdata.in1_diff) + regmap_update_bits(rt5639->regmap, RT5639_IN1_IN2, + RT5639_IN_DF1, RT5639_IN_DF1); + + if (rt5639->pdata.in2_diff) + regmap_update_bits(rt5639->regmap, RT5639_IN3_IN4, + RT5639_IN_DF2, RT5639_IN_DF2); + + rt5639->hp_mute = 1; + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5639, + rt5639_dai, ARRAY_SIZE(rt5639_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt5639_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +static struct i2c_driver rt5639_i2c_driver = { + .driver = { + .name = "rt5639", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5639_acpi_match), + }, + .probe = rt5639_i2c_probe, + .remove = rt5639_i2c_remove, + .id_table = rt5639_i2c_id, +}; +module_i2c_driver(rt5639_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5639 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5639.h b/sound/soc/codecs/rt5639.h new file mode 100644 index 0000000..b9b7f4a --- /dev/null +++ b/sound/soc/codecs/rt5639.h @@ -0,0 +1,1928 @@ +/* + * rt5639.h -- RT5639 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * 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. + */ + +#ifndef _RT5639_H +#define _RT5639_H + +#include <sound/rt5639.h> + +/* Info */ +#define RT5639_RESET 0x00 +#define RT5639_VENDOR_ID 0xfd +#define RT5639_VENDOR_ID1 0xfe +#define RT5639_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5639_SPK_VOL 0x01 +#define RT5639_HP_VOL 0x02 +#define RT5639_OUTPUT 0x03 +/* I/O - Input */ +#define RT5639_IN1_IN2 0x0d +#define RT5639_IN3_IN4 0x0e +#define RT5639_INL_INR_VOL 0x0f +/* I/O - ADC/DAC/DMIC */ +#define RT5639_DAC1_DIG_VOL 0x19 +#define RT5639_DAC2_DIG_VOL 0x1a +#define RT5639_DAC2_CTRL 0x1b +#define RT5639_ADC_DIG_VOL 0x1c +#define RT5639_ADC_DATA 0x1d +#define RT5639_ADC_BST_VOL 0x1e +/* Mixer - D-D */ +#define RT5639_STO_ADC_MIXER 0x27 +#define RT5639_MONO_ADC_MIXER 0x28 +#define RT5639_AD_DA_MIXER 0x29 +#define RT5639_STO_DAC_MIXER 0x2a +#define RT5639_MONO_DAC_MIXER 0x2b +#define RT5639_DIG_MIXER 0x2c +#define RT5639_DIG_INF_DATA 0x2f +/* Mixer - ADC */ +#define RT5639_REC_L1_MIXER 0x3b +#define RT5639_REC_L2_MIXER 0x3c +#define RT5639_REC_R1_MIXER 0x3d +#define RT5639_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5639_HPO_MIXER 0x45 +#define RT5639_SPK_L_MIXER 0x46 +#define RT5639_SPK_R_MIXER 0x47 +#define RT5639_SPO_L_MIXER 0x48 +#define RT5639_SPO_R_MIXER 0x49 +#define RT5639_SPO_CLSD_RATIO 0x4a +#define RT5639_OUT_L1_MIXER 0x4d +#define RT5639_OUT_L2_MIXER 0x4e +#define RT5639_OUT_L3_MIXER 0x4f +#define RT5639_OUT_R1_MIXER 0x50 +#define RT5639_OUT_R2_MIXER 0x51 +#define RT5639_OUT_R3_MIXER 0x52 +#define RT5639_LOUT_MIXER 0x53 +/* Power */ +#define RT5639_PWR_DIG1 0x61 +#define RT5639_PWR_DIG2 0x62 +#define RT5639_PWR_ANLG1 0x63 +#define RT5639_PWR_ANLG2 0x64 +#define RT5639_PWR_MIXER 0x65 +#define RT5639_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5639_PRIV_INDEX 0x6a +#define RT5639_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5639_I2S1_SDP 0x70 +#define RT5639_I2S2_SDP 0x71 +#define RT5639_ADDA_CLK1 0x73 +#define RT5639_DMIC 0x75 +/* Function - Analog */ +#define RT5639_GLB_CLK 0x80 +#define RT5639_PLL_CTRL1 0x81 +#define RT5639_PLL_CTRL2 0x82 +#define RT5639_ASRC_1 0x83 +#define RT5639_ASRC_2 0x84 +#define RT5639_ASRC_3 0x85 +#define RT5639_ASRC_4 0x89 +#define RT5639_ASRC_5 0x8a +#define RT5639_CLS_D_OVCD 0x8c +#define RT5639_CLS_D_OUT 0x8d +#define RT5639_DEPOP_M1 0x8e +#define RT5639_DEPOP_M2 0x8f +#define RT5639_DEPOP_M3 0x90 +#define RT5639_CHARGE_PUMP 0x91 +#define RT5639_PV_DET_SPK_G 0x92 +#define RT5639_MICBIAS 0x93 +/* Function - Digital */ +#define RT5639_EQ_CTRL1 0xb0 +#define RT5639_EQ_CTRL2 0xb1 +#define RT5639_WIND_FILTER 0xb2 +#define RT5639_DRC_AGC_1 0xb4 +#define RT5639_DRC_AGC_2 0xb5 +#define RT5639_DRC_AGC_3 0xb6 +#define RT5639_JD_CTRL 0xbb +#define RT5639_IRQ_CTRL1 0xbd +#define RT5639_IRQ_CTRL2 0xbe +#define RT5639_INT_IRQ_ST 0xbf +#define RT5639_GPIO_CTRL1 0xc0 +#define RT5639_GPIO_CTRL2 0xc2 +#define RT5639_PGM_REG_ARR1 0xc8 +#define RT5639_PGM_REG_ARR2 0xc9 +#define RT5639_PGM_REG_ARR3 0xca +#define RT5639_PGM_REG_ARR4 0xcb +#define RT5639_PGM_REG_ARR5 0xcc +#define RT5639_ADJ_HPF 0xd3 +#define RT5639_HP_CALIB_AMP_DET 0xd6 +#define RT5639_SV_ZCD1 0xd9 +/* Dummy Register */ +#define RT5639_DUMMY1 0xfa +#define RT5639_DUMMY2 0xfb +#define RT5639_DUMMY3 0xfc + + +/* Index of Codec Private Register definition */ +#define RT5639_CHPUMP_INT_REG1 0x24 +#define RT5639_MAMP_INT_REG2 0x37 +#define RT5639_3D_SPK 0x63 +#define RT5639_WND_1 0x6c +#define RT5639_WND_2 0x6d +#define RT5639_WND_3 0x6e +#define RT5639_WND_4 0x6f +#define RT5639_WND_5 0x70 +#define RT5639_WND_8 0x73 +#define RT5639_DIP_SPK_INF 0x75 +#define RT5639_HP_DCC_INT1 0x77 +#define RT5639_EQ_BW_LOP 0xa0 +#define RT5639_EQ_GN_LOP 0xa1 +#define RT5639_EQ_FC_BP1 0xa2 +#define RT5639_EQ_BW_BP1 0xa3 +#define RT5639_EQ_GN_BP1 0xa4 +#define RT5639_EQ_FC_BP2 0xa5 +#define RT5639_EQ_BW_BP2 0xa6 +#define RT5639_EQ_GN_BP2 0xa7 +#define RT5639_EQ_FC_BP3 0xa8 +#define RT5639_EQ_BW_BP3 0xa9 +#define RT5639_EQ_GN_BP3 0xaa +#define RT5639_EQ_FC_BP4 0xab +#define RT5639_EQ_BW_BP4 0xac +#define RT5639_EQ_GN_BP4 0xad +#define RT5639_EQ_FC_HIP1 0xae +#define RT5639_EQ_GN_HIP1 0xaf +#define RT5639_EQ_FC_HIP2 0xb0 +#define RT5639_EQ_BW_HIP2 0xb1 +#define RT5639_EQ_GN_HIP2 0xb2 +#define RT5639_EQ_PRE_VOL 0xb3 +#define RT5639_EQ_PST_VOL 0xb4 + +/* global definition */ +#define RT5639_L_MUTE (0x1 << 15) +#define RT5639_L_MUTE_SFT 15 +#define RT5639_VOL_L_MUTE (0x1 << 14) +#define RT5639_VOL_L_SFT 14 +#define RT5639_R_MUTE (0x1 << 7) +#define RT5639_R_MUTE_SFT 7 +#define RT5639_VOL_R_MUTE (0x1 << 6) +#define RT5639_VOL_R_SFT 6 +#define RT5639_L_VOL_MASK (0x3f << 8) +#define RT5639_L_VOL_SFT 8 +#define RT5639_R_VOL_MASK (0x3f) +#define RT5639_R_VOL_SFT 0 + +/* IN1 and IN2 Control (0x0d) */ +/* IN3 and IN4 Control (0x0e) */ +#define RT5639_BST_SFT1 12 +#define RT5639_BST_SFT2 8 +#define RT5639_IN_DF1 (0x1 << 7) +#define RT5639_IN_SFT1 7 +#define RT5639_IN_DF2 (0x1 << 6) +#define RT5639_IN_SFT2 6 + +/* INL and INR Volume Control (0x0f) */ +#define RT5639_INL_SEL_MASK (0x1 << 15) +#define RT5639_INL_SEL_SFT 15 +#define RT5639_INL_SEL_IN4P (0x0 << 15) +#define RT5639_INL_SEL_MONOP (0x1 << 15) +#define RT5639_INL_VOL_MASK (0x1f << 8) +#define RT5639_INL_VOL_SFT 8 +#define RT5639_INR_SEL_MASK (0x1 << 7) +#define RT5639_INR_SEL_SFT 7 +#define RT5639_INR_SEL_IN4N (0x0 << 7) +#define RT5639_INR_SEL_MONON (0x1 << 7) +#define RT5639_INR_VOL_MASK (0x1f) +#define RT5639_INR_VOL_SFT 0 + +/* DAC1 Digital Volume (0x19) */ +#define RT5639_DAC_L1_VOL_MASK (0xff << 8) +#define RT5639_DAC_L1_VOL_SFT 8 +#define RT5639_DAC_R1_VOL_MASK (0xff) +#define RT5639_DAC_R1_VOL_SFT 0 + +/* DAC2 Digital Volume (0x1a) */ +#define RT5639_DAC_L2_VOL_MASK (0xff << 8) +#define RT5639_DAC_L2_VOL_SFT 8 +#define RT5639_DAC_R2_VOL_MASK (0xff) +#define RT5639_DAC_R2_VOL_SFT 0 + +/* DAC2 Control (0x1b) */ +#define RT5639_M_DAC_L2_VOL (0x1 << 13) +#define RT5639_M_DAC_L2_VOL_SFT 13 +#define RT5639_M_DAC_R2_VOL (0x1 << 12) +#define RT5639_M_DAC_R2_VOL_SFT 12 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5639_ADC_L_VOL_MASK (0x7f << 8) +#define RT5639_ADC_L_VOL_SFT 8 +#define RT5639_ADC_R_VOL_MASK (0x7f) +#define RT5639_ADC_R_VOL_SFT 0 + +/* Mono ADC Digital Volume Control (0x1d) */ +#define RT5639_MONO_ADC_L_VOL_MASK (0x7f << 8) +#define RT5639_MONO_ADC_L_VOL_SFT 8 +#define RT5639_MONO_ADC_R_VOL_MASK (0x7f) +#define RT5639_MONO_ADC_R_VOL_SFT 0 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5639_ADC_L_BST_MASK (0x3 << 14) +#define RT5639_ADC_L_BST_SFT 14 +#define RT5639_ADC_R_BST_MASK (0x3 << 12) +#define RT5639_ADC_R_BST_SFT 12 +#define RT5639_ADC_COMP_MASK (0x3 << 10) +#define RT5639_ADC_COMP_SFT 10 + +/* Stereo ADC Mixer Control (0x27) */ +#define RT5639_M_ADC_L1 (0x1 << 14) +#define RT5639_M_ADC_L1_SFT 14 +#define RT5639_M_ADC_L2 (0x1 << 13) +#define RT5639_M_ADC_L2_SFT 13 +#define RT5639_ADC_1_SRC_MASK (0x1 << 12) +#define RT5639_ADC_1_SRC_SFT 12 +#define RT5639_ADC_1_SRC_ADC (0x1 << 12) +#define RT5639_ADC_1_SRC_DACMIX (0x0 << 12) +#define RT5639_ADC_2_SRC_MASK (0x3 << 10) +#define RT5639_ADC_2_SRC_SFT 10 +#define RT5639_ADC_2_SRC_DMIC1 (0x0 << 10) +#define RT5639_ADC_2_SRC_DMIC2 (0x1 << 10) +#define RT5639_ADC_2_SRC_DACMIX (0x2 << 10) +#define RT5639_M_ADC_R1 (0x1 << 6) +#define RT5639_M_ADC_R1_SFT 6 +#define RT5639_M_ADC_R2 (0x1 << 5) +#define RT5639_M_ADC_R2_SFT 5 + +/* Mono ADC Mixer Control (0x28) */ +#define RT5639_M_MONO_ADC_L1 (0x1 << 14) +#define RT5639_M_MONO_ADC_L1_SFT 14 +#define RT5639_M_MONO_ADC_L2 (0x1 << 13) +#define RT5639_M_MONO_ADC_L2_SFT 13 +#define RT5639_MONO_ADC_L1_SRC_MASK (0x1 << 12) +#define RT5639_MONO_ADC_L1_SRC_SFT 12 +#define RT5639_MONO_ADC_L1_SRC_DACMIXL (0x0 << 12) +#define RT5639_MONO_ADC_L1_SRC_ADCL (0x1 << 12) +#define RT5639_MONO_ADC_L2_SRC_MASK (0x3 << 10) +#define RT5639_MONO_ADC_L2_SRC_SFT 10 +#define RT5639_MONO_ADC_L2_SRC_DMIC_L1 (0x0 << 10) +#define RT5639_MONO_ADC_L2_SRC_DMIC_L2 (0x1 << 10) +#define RT5639_MONO_ADC_L2_SRC_DACMIXL (0x2 << 10) +#define RT5639_M_MONO_ADC_R1 (0x1 << 6) +#define RT5639_M_MONO_ADC_R1_SFT 6 +#define RT5639_M_MONO_ADC_R2 (0x1 << 5) +#define RT5639_M_MONO_ADC_R2_SFT 5 +#define RT5639_MONO_ADC_R1_SRC_MASK (0x1 << 4) +#define RT5639_MONO_ADC_R1_SRC_SFT 4 +#define RT5639_MONO_ADC_R1_SRC_ADCR (0x1 << 4) +#define RT5639_MONO_ADC_R1_SRC_DACMIXR (0x0 << 4) +#define RT5639_MONO_ADC_R2_SRC_MASK (0x3 << 2) +#define RT5639_MONO_ADC_R2_SRC_SFT 2 +#define RT5639_MONO_ADC_R2_SRC_DMIC_R1 (0x0 << 2) +#define RT5639_MONO_ADC_R2_SRC_DMIC_R2 (0x1 << 2) +#define RT5639_MONO_ADC_R2_SRC_DACMIXR (0x2 << 2) + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5639_M_ADCMIX_L (0x1 << 15) +#define RT5639_M_ADCMIX_L_SFT 15 +#define RT5639_M_IF1_DAC_L (0x1 << 14) +#define RT5639_M_IF1_DAC_L_SFT 14 +#define RT5639_M_ADCMIX_R (0x1 << 7) +#define RT5639_M_ADCMIX_R_SFT 7 +#define RT5639_M_IF1_DAC_R (0x1 << 6) +#define RT5639_M_IF1_DAC_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5639_M_DAC_L1 (0x1 << 14) +#define RT5639_M_DAC_L1_SFT 14 +#define RT5639_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5639_DAC_L1_STO_L_VOL_SFT 13 +#define RT5639_M_DAC_L2 (0x1 << 12) +#define RT5639_M_DAC_L2_SFT 12 +#define RT5639_DAC_L2_STO_L_VOL_MASK (0x1 << 11) +#define RT5639_DAC_L2_STO_L_VOL_SFT 11 +#define RT5639_M_DAC_R1 (0x1 << 6) +#define RT5639_M_DAC_R1_SFT 6 +#define RT5639_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5639_DAC_R1_STO_R_VOL_SFT 5 +#define RT5639_M_DAC_R2 (0x1 << 4) +#define RT5639_M_DAC_R2_SFT 4 +#define RT5639_DAC_R2_STO_R_VOL_MASK (0x1 << 3) +#define RT5639_DAC_R2_STO_R_VOL_SFT 3 + +/* Mono DAC Mixer Control (0x2b) */ +#define RT5639_M_DAC_L1_MONO_L (0x1 << 14) +#define RT5639_M_DAC_L1_MONO_L_SFT 14 +#define RT5639_DAC_L1_MONO_L_VOL_MASK (0x1 << 13) +#define RT5639_DAC_L1_MONO_L_VOL_SFT 13 +#define RT5639_M_DAC_L2_MONO_L (0x1 << 12) +#define RT5639_M_DAC_L2_MONO_L_SFT 12 +#define RT5639_DAC_L2_MONO_L_VOL_MASK (0x1 << 11) +#define RT5639_DAC_L2_MONO_L_VOL_SFT 11 +#define RT5639_M_DAC_R2_MONO_L (0x1 << 10) +#define RT5639_M_DAC_R2_MONO_L_SFT 10 +#define RT5639_DAC_R2_MONO_L_VOL_MASK (0x1 << 9) +#define RT5639_DAC_R2_MONO_L_VOL_SFT 9 +#define RT5639_M_DAC_R1_MONO_R (0x1 << 6) +#define RT5639_M_DAC_R1_MONO_R_SFT 6 +#define RT5639_DAC_R1_MONO_R_VOL_MASK (0x1 << 5) +#define RT5639_DAC_R1_MONO_R_VOL_SFT 5 +#define RT5639_M_DAC_R2_MONO_R (0x1 << 4) +#define RT5639_M_DAC_R2_MONO_R_SFT 4 +#define RT5639_DAC_R2_MONO_R_VOL_MASK (0x1 << 3) +#define RT5639_DAC_R2_MONO_R_VOL_SFT 3 +#define RT5639_M_DAC_L2_MONO_R (0x1 << 2) +#define RT5639_M_DAC_L2_MONO_R_SFT 2 +#define RT5639_DAC_L2_MONO_R_VOL_MASK (0x1 << 1) +#define RT5639_DAC_L2_MONO_R_VOL_SFT 1 + +/* Digital Mixer Control (0x2c) */ +#define RT5639_M_STO_L_DAC_L (0x1 << 15) +#define RT5639_M_STO_L_DAC_L_SFT 15 +#define RT5639_STO_L_DAC_L_VOL_MASK (0x1 << 14) +#define RT5639_STO_L_DAC_L_VOL_SFT 14 +#define RT5639_M_DAC_L2_DAC_L (0x1 << 13) +#define RT5639_M_DAC_L2_DAC_L_SFT 13 +#define RT5639_DAC_L2_DAC_L_VOL_MASK (0x1 << 12) +#define RT5639_DAC_L2_DAC_L_VOL_SFT 12 +#define RT5639_M_STO_R_DAC_R (0x1 << 11) +#define RT5639_M_STO_R_DAC_R_SFT 11 +#define RT5639_STO_R_DAC_R_VOL_MASK (0x1 << 10) +#define RT5639_STO_R_DAC_R_VOL_SFT 10 +#define RT5639_M_DAC_R2_DAC_R (0x1 << 9) +#define RT5639_M_DAC_R2_DAC_R_SFT 9 +#define RT5639_DAC_R2_DAC_R_VOL_MASK (0x1 << 8) +#define RT5639_DAC_R2_DAC_R_VOL_SFT 8 + +/* DSP Path Control 2 (0x2e) */ +#define RT5639_DAC_L2_SEL_MASK (0x3 << 14) +#define RT5639_DAC_L2_SEL_SFT 14 +#define RT5639_DAC_L2_SEL_IF2 (0x0 << 14) +#define RT5639_DAC_R2_SEL_MASK (0x3 << 12) +#define RT5639_DAC_R2_SEL_SFT 12 +#define RT5639_DAC_R2_SEL_IF2 (0x0 << 12) + +/* Digital Interface Data Control (0x2f) */ +#define RT5639_IF1_DAC_SEL_MASK (0x3 << 14) +#define RT5639_IF1_DAC_SEL_SFT 14 +#define RT5639_IF1_DAC_SEL_NOR (0x0 << 14) +#define RT5639_IF1_DAC_SEL_L2R (0x1 << 14) +#define RT5639_IF1_DAC_SEL_R2L (0x2 << 14) +#define RT5639_IF1_DAC_SEL_SWAP (0x3 << 14) +#define RT5639_IF1_ADC_SEL_MASK (0x3 << 12) +#define RT5639_IF1_ADC_SEL_SFT 12 +#define RT5639_IF1_ADC_SEL_NOR (0x0 << 12) +#define RT5639_IF1_ADC_SEL_L2R (0x1 << 12) +#define RT5639_IF1_ADC_SEL_R2L (0x2 << 12) +#define RT5639_IF1_ADC_SEL_SWAP (0x3 << 12) +#define RT5639_IF2_DAC_SEL_MASK (0x3 << 10) +#define RT5639_IF2_DAC_SEL_SFT 10 +#define RT5639_IF2_DAC_SEL_NOR (0x0 << 10) +#define RT5639_IF2_DAC_SEL_L2R (0x1 << 10) +#define RT5639_IF2_DAC_SEL_R2L (0x2 << 10) +#define RT5639_IF2_DAC_SEL_SWAP (0x3 << 10) +#define RT5639_IF2_ADC_SEL_MASK (0x3 << 8) +#define RT5639_IF2_ADC_SEL_SFT 8 +#define RT5639_IF2_ADC_SEL_NOR (0x0 << 8) +#define RT5639_IF2_ADC_SEL_L2R (0x1 << 8) +#define RT5639_IF2_ADC_SEL_R2L (0x2 << 8) +#define RT5639_IF2_ADC_SEL_SWAP (0x3 << 8) +#define RT5639_IF3_DAC_SEL_MASK (0x3 << 6) +#define RT5639_IF3_DAC_SEL_SFT 6 +#define RT5639_IF3_DAC_SEL_NOR (0x0 << 6) +#define RT5639_IF3_DAC_SEL_L2R (0x1 << 6) +#define RT5639_IF3_DAC_SEL_R2L (0x2 << 6) +#define RT5639_IF3_DAC_SEL_SWAP (0x3 << 6) +#define RT5639_IF3_ADC_SEL_MASK (0x3 << 4) +#define RT5639_IF3_ADC_SEL_SFT 4 +#define RT5639_IF3_ADC_SEL_NOR (0x0 << 4) +#define RT5639_IF3_ADC_SEL_L2R (0x1 << 4) +#define RT5639_IF3_ADC_SEL_R2L (0x2 << 4) +#define RT5639_IF3_ADC_SEL_SWAP (0x3 << 4) + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5639_G_HP_L_RM_L_MASK (0x7 << 13) +#define RT5639_G_HP_L_RM_L_SFT 13 +#define RT5639_G_IN_L_RM_L_MASK (0x7 << 10) +#define RT5639_G_IN_L_RM_L_SFT 10 +#define RT5639_G_BST4_RM_L_MASK (0x7 << 7) +#define RT5639_G_BST4_RM_L_SFT 7 +#define RT5639_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5639_G_BST3_RM_L_SFT 4 +#define RT5639_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5639_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5639_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5639_G_BST1_RM_L_SFT 13 +#define RT5639_G_OM_L_RM_L_MASK (0x7 << 10) +#define RT5639_G_OM_L_RM_L_SFT 10 +#define RT5639_M_HP_L_RM_L (0x1 << 6) +#define RT5639_M_HP_L_RM_L_SFT 6 +#define RT5639_M_IN_L_RM_L (0x1 << 5) +#define RT5639_M_IN_L_RM_L_SFT 5 +#define RT5639_M_BST4_RM_L (0x1 << 4) +#define RT5639_M_BST4_RM_L_SFT 4 +#define RT5639_M_BST3_RM_L (0x1 << 3) +#define RT5639_M_BST3_RM_L_SFT 3 +#define RT5639_M_BST2_RM_L (0x1 << 2) +#define RT5639_M_BST2_RM_L_SFT 2 +#define RT5639_M_BST1_RM_L (0x1 << 1) +#define RT5639_M_BST1_RM_L_SFT 1 +#define RT5639_M_OM_L_RM_L (0x1) +#define RT5639_M_OM_L_RM_L_SFT 0 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5639_G_HP_R_RM_R_MASK (0x7 << 13) +#define RT5639_G_HP_R_RM_R_SFT 13 +#define RT5639_G_IN_R_RM_R_MASK (0x7 << 10) +#define RT5639_G_IN_R_RM_R_SFT 10 +#define RT5639_G_BST4_RM_R_MASK (0x7 << 7) +#define RT5639_G_BST4_RM_R_SFT 7 +#define RT5639_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5639_G_BST3_RM_R_SFT 4 +#define RT5639_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5639_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5639_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5639_G_BST1_RM_R_SFT 13 +#define RT5639_G_OM_R_RM_R_MASK (0x7 << 10) +#define RT5639_G_OM_R_RM_R_SFT 10 +#define RT5639_M_HP_R_RM_R (0x1 << 6) +#define RT5639_M_HP_R_RM_R_SFT 6 +#define RT5639_M_IN_R_RM_R (0x1 << 5) +#define RT5639_M_IN_R_RM_R_SFT 5 +#define RT5639_M_BST4_RM_R (0x1 << 4) +#define RT5639_M_BST4_RM_R_SFT 4 +#define RT5639_M_BST3_RM_R (0x1 << 3) +#define RT5639_M_BST3_RM_R_SFT 3 +#define RT5639_M_BST2_RM_R (0x1 << 2) +#define RT5639_M_BST2_RM_R_SFT 2 +#define RT5639_M_BST1_RM_R (0x1 << 1) +#define RT5639_M_BST1_RM_R_SFT 1 +#define RT5639_M_OM_R_RM_R (0x1) +#define RT5639_M_OM_R_RM_R_SFT 0 + +/* HPMIX Control (0x45) */ +#define RT5639_M_DAC2_HM (0x1 << 15) +#define RT5639_M_DAC2_HM_SFT 15 +#define RT5639_M_DAC1_HM (0x1 << 14) +#define RT5639_M_DAC1_HM_SFT 14 +#define RT5639_M_HPVOL_HM (0x1 << 13) +#define RT5639_M_HPVOL_HM_SFT 13 +#define RT5639_G_HPOMIX_MASK (0x1 << 12) +#define RT5639_G_HPOMIX_SFT 12 + +/* SPK Left Mixer Control (0x46) */ +#define RT5639_G_RM_L_SM_L_MASK (0x3 << 14) +#define RT5639_G_RM_L_SM_L_SFT 14 +#define RT5639_G_IN_L_SM_L_MASK (0x3 << 12) +#define RT5639_G_IN_L_SM_L_SFT 12 +#define RT5639_G_DAC_L1_SM_L_MASK (0x3 << 10) +#define RT5639_G_DAC_L1_SM_L_SFT 10 +#define RT5639_G_DAC_L2_SM_L_MASK (0x3 << 8) +#define RT5639_G_DAC_L2_SM_L_SFT 8 +#define RT5639_G_OM_L_SM_L_MASK (0x3 << 6) +#define RT5639_G_OM_L_SM_L_SFT 6 +#define RT5639_M_RM_L_SM_L (0x1 << 5) +#define RT5639_M_RM_L_SM_L_SFT 5 +#define RT5639_M_IN_L_SM_L (0x1 << 4) +#define RT5639_M_IN_L_SM_L_SFT 4 +#define RT5639_M_DAC_L1_SM_L (0x1 << 3) +#define RT5639_M_DAC_L1_SM_L_SFT 3 +#define RT5639_M_DAC_L2_SM_L (0x1 << 2) +#define RT5639_M_DAC_L2_SM_L_SFT 2 +#define RT5639_M_OM_L_SM_L (0x1 << 1) +#define RT5639_M_OM_L_SM_L_SFT 1 + +/* SPK Right Mixer Control (0x47) */ +#define RT5639_G_RM_R_SM_R_MASK (0x3 << 14) +#define RT5639_G_RM_R_SM_R_SFT 14 +#define RT5639_G_IN_R_SM_R_MASK (0x3 << 12) +#define RT5639_G_IN_R_SM_R_SFT 12 +#define RT5639_G_DAC_R1_SM_R_MASK (0x3 << 10) +#define RT5639_G_DAC_R1_SM_R_SFT 10 +#define RT5639_G_DAC_R2_SM_R_MASK (0x3 << 8) +#define RT5639_G_DAC_R2_SM_R_SFT 8 +#define RT5639_G_OM_R_SM_R_MASK (0x3 << 6) +#define RT5639_G_OM_R_SM_R_SFT 6 +#define RT5639_M_RM_R_SM_R (0x1 << 5) +#define RT5639_M_RM_R_SM_R_SFT 5 +#define RT5639_M_IN_R_SM_R (0x1 << 4) +#define RT5639_M_IN_R_SM_R_SFT 4 +#define RT5639_M_DAC_R1_SM_R (0x1 << 3) +#define RT5639_M_DAC_R1_SM_R_SFT 3 +#define RT5639_M_DAC_R2_SM_R (0x1 << 2) +#define RT5639_M_DAC_R2_SM_R_SFT 2 +#define RT5639_M_OM_R_SM_R (0x1 << 1) +#define RT5639_M_OM_R_SM_R_SFT 1 + +/* SPOLMIX Control (0x48) */ +#define RT5639_M_DAC_R1_SPM_L (0x1 << 15) +#define RT5639_M_DAC_R1_SPM_L_SFT 15 +#define RT5639_M_DAC_L1_SPM_L (0x1 << 14) +#define RT5639_M_DAC_L1_SPM_L_SFT 14 +#define RT5639_M_SV_R_SPM_L (0x1 << 13) +#define RT5639_M_SV_R_SPM_L_SFT 13 +#define RT5639_M_SV_L_SPM_L (0x1 << 12) +#define RT5639_M_SV_L_SPM_L_SFT 12 +#define RT5639_M_BST1_SPM_L (0x1 << 11) +#define RT5639_M_BST1_SPM_L_SFT 11 + +/* SPORMIX Control (0x49) */ +#define RT5639_M_DAC_R1_SPM_R (0x1 << 13) +#define RT5639_M_DAC_R1_SPM_R_SFT 13 +#define RT5639_M_SV_R_SPM_R (0x1 << 12) +#define RT5639_M_SV_R_SPM_R_SFT 12 +#define RT5639_M_BST1_SPM_R (0x1 << 11) +#define RT5639_M_BST1_SPM_R_SFT 11 + +/* SPOLMIX / SPORMIX Ratio Control (0x4a) */ +#define RT5639_SPO_CLSD_RATIO_MASK (0x7) +#define RT5639_SPO_CLSD_RATIO_SFT 0 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5639_G_BST3_OM_L_MASK (0x7 << 13) +#define RT5639_G_BST3_OM_L_SFT 13 +#define RT5639_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5639_G_BST2_OM_L_SFT 10 +#define RT5639_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5639_G_BST1_OM_L_SFT 7 +#define RT5639_G_IN_L_OM_L_MASK (0x7 << 4) +#define RT5639_G_IN_L_OM_L_SFT 4 +#define RT5639_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5639_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5639_G_DAC_R2_OM_L_MASK (0x7 << 13) +#define RT5639_G_DAC_R2_OM_L_SFT 13 +#define RT5639_G_DAC_L2_OM_L_MASK (0x7 << 10) +#define RT5639_G_DAC_L2_OM_L_SFT 10 +#define RT5639_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5639_G_DAC_L1_OM_L_SFT 7 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5639_M_BST3_OM_L (0x1 << 7) +#define RT5639_M_BST3_OM_L_SFT 7 +#define RT5639_M_BST2_OM_L (0x1 << 6) +#define RT5639_M_BST2_OM_L_SFT 6 +#define RT5639_M_BST1_OM_L (0x1 << 5) +#define RT5639_M_BST1_OM_L_SFT 5 +#define RT5639_M_IN_L_OM_L (0x1 << 4) +#define RT5639_M_IN_L_OM_L_SFT 4 +#define RT5639_M_RM_L_OM_L (0x1 << 3) +#define RT5639_M_RM_L_OM_L_SFT 3 +#define RT5639_M_DAC_R2_OM_L (0x1 << 2) +#define RT5639_M_DAC_R2_OM_L_SFT 2 +#define RT5639_M_DAC_L2_OM_L (0x1 << 1) +#define RT5639_M_DAC_L2_OM_L_SFT 1 +#define RT5639_M_DAC_L1_OM_L (0x1) +#define RT5639_M_DAC_L1_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5639_G_BST4_OM_R_MASK (0x7 << 13) +#define RT5639_G_BST4_OM_R_SFT 13 +#define RT5639_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5639_G_BST2_OM_R_SFT 10 +#define RT5639_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5639_G_BST1_OM_R_SFT 7 +#define RT5639_G_IN_R_OM_R_MASK (0x7 << 4) +#define RT5639_G_IN_R_OM_R_SFT 4 +#define RT5639_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5639_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5639_G_DAC_L2_OM_R_MASK (0x7 << 13) +#define RT5639_G_DAC_L2_OM_R_SFT 13 +#define RT5639_G_DAC_R2_OM_R_MASK (0x7 << 10) +#define RT5639_G_DAC_R2_OM_R_SFT 10 +#define RT5639_G_DAC_R1_OM_R_MASK (0x7 << 7) +#define RT5639_G_DAC_R1_OM_R_SFT 7 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5639_M_BST4_OM_R (0x1 << 7) +#define RT5639_M_BST4_OM_R_SFT 7 +#define RT5639_M_BST2_OM_R (0x1 << 6) +#define RT5639_M_BST2_OM_R_SFT 6 +#define RT5639_M_BST1_OM_R (0x1 << 5) +#define RT5639_M_BST1_OM_R_SFT 5 +#define RT5639_M_IN_R_OM_R (0x1 << 4) +#define RT5639_M_IN_R_OM_R_SFT 4 +#define RT5639_M_RM_R_OM_R (0x1 << 3) +#define RT5639_M_RM_R_OM_R_SFT 3 +#define RT5639_M_DAC_L2_OM_R (0x1 << 2) +#define RT5639_M_DAC_L2_OM_R_SFT 2 +#define RT5639_M_DAC_R2_OM_R (0x1 << 1) +#define RT5639_M_DAC_R2_OM_R_SFT 1 +#define RT5639_M_DAC_R1_OM_R (0x1) +#define RT5639_M_DAC_R1_OM_R_SFT 0 + +/* LOUT Mixer Control (0x53) */ +#define RT5639_M_DAC_L1_LM (0x1 << 15) +#define RT5639_M_DAC_L1_LM_SFT 15 +#define RT5639_M_DAC_R1_LM (0x1 << 14) +#define RT5639_M_DAC_R1_LM_SFT 14 +#define RT5639_M_OV_L_LM (0x1 << 13) +#define RT5639_M_OV_L_LM_SFT 13 +#define RT5639_M_OV_R_LM (0x1 << 12) +#define RT5639_M_OV_R_LM_SFT 12 +#define RT5639_G_LOUTMIX_MASK (0x1 << 11) +#define RT5639_G_LOUTMIX_SFT 11 + +/* Power Management for Digital 1 (0x61) */ +#define RT5639_PWR_I2S1 (0x1 << 15) +#define RT5639_PWR_I2S1_BIT 15 +#define RT5639_PWR_I2S2 (0x1 << 14) +#define RT5639_PWR_I2S2_BIT 14 +#define RT5639_PWR_DAC_L1 (0x1 << 12) +#define RT5639_PWR_DAC_L1_BIT 12 +#define RT5639_PWR_DAC_R1 (0x1 << 11) +#define RT5639_PWR_DAC_R1_BIT 11 +#define RT5639_PWR_DAC_L2 (0x1 << 7) +#define RT5639_PWR_DAC_L2_BIT 7 +#define RT5639_PWR_DAC_R2 (0x1 << 6) +#define RT5639_PWR_DAC_R2_BIT 6 +#define RT5639_PWR_ADC_L (0x1 << 2) +#define RT5639_PWR_ADC_L_BIT 2 +#define RT5639_PWR_ADC_R (0x1 << 1) +#define RT5639_PWR_ADC_R_BIT 1 +#define RT5639_PWR_CLS_D (0x1) +#define RT5639_PWR_CLS_D_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5639_PWR_ADC_SF (0x1 << 15) +#define RT5639_PWR_ADC_SF_BIT 15 +#define RT5639_PWR_ADC_MF_L (0x1 << 14) +#define RT5639_PWR_ADC_MF_L_BIT 14 +#define RT5639_PWR_ADC_MF_R (0x1 << 13) +#define RT5639_PWR_ADC_MF_R_BIT 13 +#define RT5639_PWR_I2S_DSP (0x1 << 12) +#define RT5639_PWR_I2S_DSP_BIT 12 + +/* Power Management for Analog 1 (0x63) */ +#define RT5639_PWR_VREF1 (0x1 << 15) +#define RT5639_PWR_VREF1_BIT 15 +#define RT5639_PWR_FV1 (0x1 << 14) +#define RT5639_PWR_FV1_BIT 14 +#define RT5639_PWR_MB (0x1 << 13) +#define RT5639_PWR_MB_BIT 13 +#define RT5639_PWR_LM (0x1 << 12) +#define RT5639_PWR_LM_BIT 12 +#define RT5639_PWR_BG (0x1 << 11) +#define RT5639_PWR_BG_BIT 11 +#define RT5639_PWR_MM (0x1 << 10) +#define RT5639_PWR_MM_BIT 10 +#define RT5639_PWR_MA (0x1 << 8) +#define RT5639_PWR_MA_BIT 8 +#define RT5639_PWR_HP_L (0x1 << 7) +#define RT5639_PWR_HP_L_BIT 7 +#define RT5639_PWR_HP_R (0x1 << 6) +#define RT5639_PWR_HP_R_BIT 6 +#define RT5639_PWR_HA (0x1 << 5) +#define RT5639_PWR_HA_BIT 5 +#define RT5639_PWR_VREF2 (0x1 << 4) +#define RT5639_PWR_VREF2_BIT 4 +#define RT5639_PWR_FV2 (0x1 << 3) +#define RT5639_PWR_FV2_BIT 3 +#define RT5639_PWR_LDO2 (0x1 << 2) +#define RT5639_PWR_LDO2_BIT 2 + +/* Power Management for Analog 2 (0x64) */ +#define RT5639_PWR_BST1 (0x1 << 15) +#define RT5639_PWR_BST1_BIT 15 +#define RT5639_PWR_BST2 (0x1 << 14) +#define RT5639_PWR_BST2_BIT 14 +#define RT5639_PWR_BST3 (0x1 << 13) +#define RT5639_PWR_BST3_BIT 13 +#define RT5639_PWR_BST4 (0x1 << 12) +#define RT5639_PWR_BST4_BIT 12 +#define RT5639_PWR_MB1 (0x1 << 11) +#define RT5639_PWR_MB1_BIT 11 +#define RT5639_PWR_PLL (0x1 << 9) +#define RT5639_PWR_PLL_BIT 9 + +/* Power Management for Mixer (0x65) */ +#define RT5639_PWR_OM_L (0x1 << 15) +#define RT5639_PWR_OM_L_BIT 15 +#define RT5639_PWR_OM_R (0x1 << 14) +#define RT5639_PWR_OM_R_BIT 14 +#define RT5639_PWR_SM_L (0x1 << 13) +#define RT5639_PWR_SM_L_BIT 13 +#define RT5639_PWR_SM_R (0x1 << 12) +#define RT5639_PWR_SM_R_BIT 12 +#define RT5639_PWR_RM_L (0x1 << 11) +#define RT5639_PWR_RM_L_BIT 11 +#define RT5639_PWR_RM_R (0x1 << 10) +#define RT5639_PWR_RM_R_BIT 10 + +/* Power Management for Volume (0x66) */ +#define RT5639_PWR_SV_L (0x1 << 15) +#define RT5639_PWR_SV_L_BIT 15 +#define RT5639_PWR_SV_R (0x1 << 14) +#define RT5639_PWR_SV_R_BIT 14 +#define RT5639_PWR_OV_L (0x1 << 13) +#define RT5639_PWR_OV_L_BIT 13 +#define RT5639_PWR_OV_R (0x1 << 12) +#define RT5639_PWR_OV_R_BIT 12 +#define RT5639_PWR_HV_L (0x1 << 11) +#define RT5639_PWR_HV_L_BIT 11 +#define RT5639_PWR_HV_R (0x1 << 10) +#define RT5639_PWR_HV_R_BIT 10 +#define RT5639_PWR_IN_L (0x1 << 9) +#define RT5639_PWR_IN_L_BIT 9 +#define RT5639_PWR_IN_R (0x1 << 8) +#define RT5639_PWR_IN_R_BIT 8 + +/* I2S1/2/3 Audio Serial Data Port Control (0x70 0x71 0x72) */ +#define RT5639_I2S_MS_MASK (0x1 << 15) +#define RT5639_I2S_MS_SFT 15 +#define RT5639_I2S_MS_M (0x0 << 15) +#define RT5639_I2S_MS_S (0x1 << 15) +#define RT5639_I2S_IF_MASK (0x7 << 12) +#define RT5639_I2S_IF_SFT 12 +#define RT5639_I2S_O_CP_MASK (0x3 << 10) +#define RT5639_I2S_O_CP_SFT 10 +#define RT5639_I2S_O_CP_OFF (0x0 << 10) +#define RT5639_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5639_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5639_I2S_I_CP_MASK (0x3 << 8) +#define RT5639_I2S_I_CP_SFT 8 +#define RT5639_I2S_I_CP_OFF (0x0 << 8) +#define RT5639_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5639_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5639_I2S_BP_MASK (0x1 << 7) +#define RT5639_I2S_BP_SFT 7 +#define RT5639_I2S_BP_NOR (0x0 << 7) +#define RT5639_I2S_BP_INV (0x1 << 7) +#define RT5639_I2S_DL_MASK (0x3 << 2) +#define RT5639_I2S_DL_SFT 2 +#define RT5639_I2S_DL_16 (0x0 << 2) +#define RT5639_I2S_DL_20 (0x1 << 2) +#define RT5639_I2S_DL_24 (0x2 << 2) +#define RT5639_I2S_DL_8 (0x3 << 2) +#define RT5639_I2S_DF_MASK (0x3) +#define RT5639_I2S_DF_SFT 0 +#define RT5639_I2S_DF_I2S (0x0) +#define RT5639_I2S_DF_LEFT (0x1) +#define RT5639_I2S_DF_PCM_A (0x2) +#define RT5639_I2S_DF_PCM_B (0x3) + +/* I2S2 Audio Serial Data Port Control (0x71) */ +#define RT5639_I2S2_SDI_MASK (0x1 << 6) +#define RT5639_I2S2_SDI_SFT 6 +#define RT5639_I2S2_SDI_I2S1 (0x0 << 6) +#define RT5639_I2S2_SDI_I2S2 (0x1 << 6) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5639_I2S_BCLK_MS1_MASK (0x1 << 15) +#define RT5639_I2S_BCLK_MS1_SFT 15 +#define RT5639_I2S_BCLK_MS1_32 (0x0 << 15) +#define RT5639_I2S_BCLK_MS1_64 (0x1 << 15) +#define RT5639_I2S_PD1_MASK (0x7 << 12) +#define RT5639_I2S_PD1_SFT 12 +#define RT5639_I2S_PD1_1 (0x0 << 12) +#define RT5639_I2S_PD1_2 (0x1 << 12) +#define RT5639_I2S_PD1_3 (0x2 << 12) +#define RT5639_I2S_PD1_4 (0x3 << 12) +#define RT5639_I2S_PD1_6 (0x4 << 12) +#define RT5639_I2S_PD1_8 (0x5 << 12) +#define RT5639_I2S_PD1_12 (0x6 << 12) +#define RT5639_I2S_PD1_16 (0x7 << 12) +#define RT5639_I2S_BCLK_MS2_MASK (0x1 << 11) +#define RT5639_I2S_BCLK_MS2_SFT 11 +#define RT5639_I2S_BCLK_MS2_32 (0x0 << 11) +#define RT5639_I2S_BCLK_MS2_64 (0x1 << 11) +#define RT5639_I2S_PD2_MASK (0x7 << 8) +#define RT5639_I2S_PD2_SFT 8 +#define RT5639_I2S_PD2_1 (0x0 << 8) +#define RT5639_I2S_PD2_2 (0x1 << 8) +#define RT5639_I2S_PD2_3 (0x2 << 8) +#define RT5639_I2S_PD2_4 (0x3 << 8) +#define RT5639_I2S_PD2_6 (0x4 << 8) +#define RT5639_I2S_PD2_8 (0x5 << 8) +#define RT5639_I2S_PD2_12 (0x6 << 8) +#define RT5639_I2S_PD2_16 (0x7 << 8) +#define RT5639_I2S_BCLK_MS3_MASK (0x1 << 7) +#define RT5639_I2S_BCLK_MS3_SFT 7 +#define RT5639_I2S_BCLK_MS3_32 (0x0 << 7) +#define RT5639_I2S_BCLK_MS3_64 (0x1 << 7) +#define RT5639_I2S_PD3_MASK (0x7 << 4) +#define RT5639_I2S_PD3_SFT 4 +#define RT5639_I2S_PD3_1 (0x0 << 4) +#define RT5639_I2S_PD3_2 (0x1 << 4) +#define RT5639_I2S_PD3_3 (0x2 << 4) +#define RT5639_I2S_PD3_4 (0x3 << 4) +#define RT5639_I2S_PD3_6 (0x4 << 4) +#define RT5639_I2S_PD3_8 (0x5 << 4) +#define RT5639_I2S_PD3_12 (0x6 << 4) +#define RT5639_I2S_PD3_16 (0x7 << 4) +#define RT5639_DAC_OSR_MASK (0x3 << 2) +#define RT5639_DAC_OSR_SFT 2 +#define RT5639_DAC_OSR_128 (0x0 << 2) +#define RT5639_DAC_OSR_64 (0x1 << 2) +#define RT5639_DAC_OSR_32 (0x2 << 2) +#define RT5639_DAC_OSR_16 (0x3 << 2) +#define RT5639_ADC_OSR_MASK (0x3) +#define RT5639_ADC_OSR_SFT 0 +#define RT5639_ADC_OSR_128 (0x0) +#define RT5639_ADC_OSR_64 (0x1) +#define RT5639_ADC_OSR_32 (0x2) +#define RT5639_ADC_OSR_16 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5639_DAC_L_OSR_MASK (0x3 << 14) +#define RT5639_DAC_L_OSR_SFT 14 +#define RT5639_DAC_L_OSR_128 (0x0 << 14) +#define RT5639_DAC_L_OSR_64 (0x1 << 14) +#define RT5639_DAC_L_OSR_32 (0x2 << 14) +#define RT5639_DAC_L_OSR_16 (0x3 << 14) +#define RT5639_ADC_R_OSR_MASK (0x3 << 12) +#define RT5639_ADC_R_OSR_SFT 12 +#define RT5639_ADC_R_OSR_128 (0x0 << 12) +#define RT5639_ADC_R_OSR_64 (0x1 << 12) +#define RT5639_ADC_R_OSR_32 (0x2 << 12) +#define RT5639_ADC_R_OSR_16 (0x3 << 12) +#define RT5639_DAHPF_EN (0x1 << 11) +#define RT5639_DAHPF_EN_SFT 11 +#define RT5639_ADHPF_EN (0x1 << 10) +#define RT5639_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5639_DMIC_1_EN_MASK (0x1 << 15) +#define RT5639_DMIC_1_EN_SFT 15 +#define RT5639_DMIC_1_DIS (0x0 << 15) +#define RT5639_DMIC_1_EN (0x1 << 15) +#define RT5639_DMIC_2_EN_MASK (0x1 << 14) +#define RT5639_DMIC_2_EN_SFT 14 +#define RT5639_DMIC_2_DIS (0x0 << 14) +#define RT5639_DMIC_2_EN (0x1 << 14) +#define RT5639_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5639_DMIC_1L_LH_SFT 13 +#define RT5639_DMIC_1L_LH_FALLING (0x0 << 13) +#define RT5639_DMIC_1L_LH_RISING (0x1 << 13) +#define RT5639_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5639_DMIC_1R_LH_SFT 12 +#define RT5639_DMIC_1R_LH_FALLING (0x0 << 12) +#define RT5639_DMIC_1R_LH_RISING (0x1 << 12) +#define RT5639_DMIC_1_DP_MASK (0x1 << 11) +#define RT5639_DMIC_1_DP_SFT 11 +#define RT5639_DMIC_1_DP_GPIO3 (0x0 << 11) +#define RT5639_DMIC_1_DP_IN1P (0x1 << 11) +#define RT5639_DMIC_2_DP_MASK (0x1 << 10) +#define RT5639_DMIC_2_DP_SFT 10 +#define RT5639_DMIC_2_DP_GPIO4 (0x0 << 10) +#define RT5639_DMIC_2_DP_IN1N (0x1 << 10) +#define RT5639_DMIC_2L_LH_MASK (0x1 << 9) +#define RT5639_DMIC_2L_LH_SFT 9 +#define RT5639_DMIC_2L_LH_FALLING (0x0 << 9) +#define RT5639_DMIC_2L_LH_RISING (0x1 << 9) +#define RT5639_DMIC_2R_LH_MASK (0x1 << 8) +#define RT5639_DMIC_2R_LH_SFT 8 +#define RT5639_DMIC_2R_LH_FALLING (0x0 << 8) +#define RT5639_DMIC_2R_LH_RISING (0x1 << 8) +#define RT5639_DMIC_CLK_MASK (0x7 << 5) +#define RT5639_DMIC_CLK_SFT 5 + +/* Global Clock Control (0x80) */ +#define RT5639_SCLK_SRC_MASK (0x3 << 14) +#define RT5639_SCLK_SRC_SFT 14 +#define RT5639_SCLK_SRC_MCLK (0x0 << 14) +#define RT5639_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5639_PLL1_SRC_MASK (0x3 << 12) +#define RT5639_PLL1_SRC_SFT 12 +#define RT5639_PLL1_SRC_MCLK (0x0 << 12) +#define RT5639_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5639_PLL1_SRC_BCLK2 (0x2 << 12) +#define RT5639_PLL1_SRC_BCLK3 (0x3 << 12) +#define RT5639_PLL1_PD_MASK (0x1 << 3) +#define RT5639_PLL1_PD_SFT 3 +#define RT5639_PLL1_PD_1 (0x0 << 3) +#define RT5639_PLL1_PD_2 (0x1 << 3) + +#define RT5639_PLL_INP_MAX 40000000 +#define RT5639_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5639_PLL_N_MAX 0x1ff +#define RT5639_PLL_N_MASK (RT5639_PLL_N_MAX << 7) +#define RT5639_PLL_N_SFT 7 +#define RT5639_PLL_K_MAX 0x1f +#define RT5639_PLL_K_MASK (RT5639_PLL_K_MAX) +#define RT5639_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5639_PLL_M_MAX 0xf +#define RT5639_PLL_M_MASK (RT5639_PLL_M_MAX << 12) +#define RT5639_PLL_M_SFT 12 +#define RT5639_PLL_M_BP (0x1 << 11) +#define RT5639_PLL_M_BP_SFT 11 + +/* ASRC Control 1 (0x83) */ +#define RT5639_STO_T_MASK (0x1 << 15) +#define RT5639_STO_T_SFT 15 +#define RT5639_STO_T_SCLK (0x0 << 15) +#define RT5639_STO_T_LRCK1 (0x1 << 15) +#define RT5639_M1_T_MASK (0x1 << 14) +#define RT5639_M1_T_SFT 14 +#define RT5639_M1_T_I2S2 (0x0 << 14) +#define RT5639_M1_T_I2S2_D3 (0x1 << 14) +#define RT5639_I2S2_F_MASK (0x1 << 12) +#define RT5639_I2S2_F_SFT 12 +#define RT5639_I2S2_F_I2S2_D2 (0x0 << 12) +#define RT5639_I2S2_F_I2S1_TCLK (0x1 << 12) +#define RT5639_DMIC_1_M_MASK (0x1 << 9) +#define RT5639_DMIC_1_M_SFT 9 +#define RT5639_DMIC_1_M_NOR (0x0 << 9) +#define RT5639_DMIC_1_M_ASYN (0x1 << 9) +#define RT5639_DMIC_2_M_MASK (0x1 << 8) +#define RT5639_DMIC_2_M_SFT 8 +#define RT5639_DMIC_2_M_NOR (0x0 << 8) +#define RT5639_DMIC_2_M_ASYN (0x1 << 8) + +/* ASRC Control 2 (0x84) */ +#define RT5639_MDA_L_M_MASK (0x1 << 15) +#define RT5639_MDA_L_M_SFT 15 +#define RT5639_MDA_L_M_NOR (0x0 << 15) +#define RT5639_MDA_L_M_ASYN (0x1 << 15) +#define RT5639_MDA_R_M_MASK (0x1 << 14) +#define RT5639_MDA_R_M_SFT 14 +#define RT5639_MDA_R_M_NOR (0x0 << 14) +#define RT5639_MDA_R_M_ASYN (0x1 << 14) +#define RT5639_MAD_L_M_MASK (0x1 << 13) +#define RT5639_MAD_L_M_SFT 13 +#define RT5639_MAD_L_M_NOR (0x0 << 13) +#define RT5639_MAD_L_M_ASYN (0x1 << 13) +#define RT5639_MAD_R_M_MASK (0x1 << 12) +#define RT5639_MAD_R_M_SFT 12 +#define RT5639_MAD_R_M_NOR (0x0 << 12) +#define RT5639_MAD_R_M_ASYN (0x1 << 12) +#define RT5639_ADC_M_MASK (0x1 << 11) +#define RT5639_ADC_M_SFT 11 +#define RT5639_ADC_M_NOR (0x0 << 11) +#define RT5639_ADC_M_ASYN (0x1 << 11) +#define RT5639_STO_DAC_M_MASK (0x1 << 5) +#define RT5639_STO_DAC_M_SFT 5 +#define RT5639_STO_DAC_M_NOR (0x0 << 5) +#define RT5639_STO_DAC_M_ASYN (0x1 << 5) +#define RT5639_I2S1_R_D_MASK (0x1 << 4) +#define RT5639_I2S1_R_D_SFT 4 +#define RT5639_I2S1_R_D_DIS (0x0 << 4) +#define RT5639_I2S1_R_D_EN (0x1 << 4) +#define RT5639_I2S2_R_D_MASK (0x1 << 3) +#define RT5639_I2S2_R_D_SFT 3 +#define RT5639_I2S2_R_D_DIS (0x0 << 3) +#define RT5639_I2S2_R_D_EN (0x1 << 3) +#define RT5639_PRE_SCLK_MASK (0x3) +#define RT5639_PRE_SCLK_SFT 0 +#define RT5639_PRE_SCLK_512 (0x0) +#define RT5639_PRE_SCLK_1024 (0x1) +#define RT5639_PRE_SCLK_2048 (0x2) + +/* ASRC Control 3 (0x85) */ +#define RT5639_I2S1_RATE_MASK (0xf << 12) +#define RT5639_I2S1_RATE_SFT 12 +#define RT5639_I2S2_RATE_MASK (0xf << 8) +#define RT5639_I2S2_RATE_SFT 8 + +/* ASRC Control 4 (0x89) */ +#define RT5639_I2S1_PD_MASK (0x7 << 12) +#define RT5639_I2S1_PD_SFT 12 +#define RT5639_I2S2_PD_MASK (0x7 << 8) +#define RT5639_I2S2_PD_SFT 8 + +/* HPOUT Over Current Detection (0x8b) */ +#define RT5639_HP_OVCD_MASK (0x1 << 10) +#define RT5639_HP_OVCD_SFT 10 +#define RT5639_HP_OVCD_DIS (0x0 << 10) +#define RT5639_HP_OVCD_EN (0x1 << 10) +#define RT5639_HP_OC_TH_MASK (0x3 << 8) +#define RT5639_HP_OC_TH_SFT 8 +#define RT5639_HP_OC_TH_90 (0x0 << 8) +#define RT5639_HP_OC_TH_105 (0x1 << 8) +#define RT5639_HP_OC_TH_120 (0x2 << 8) +#define RT5639_HP_OC_TH_135 (0x3 << 8) + +/* Class D Over Current Control (0x8c) */ +#define RT5639_CLSD_OC_MASK (0x1 << 9) +#define RT5639_CLSD_OC_SFT 9 +#define RT5639_CLSD_OC_PU (0x0 << 9) +#define RT5639_CLSD_OC_PD (0x1 << 9) +#define RT5639_AUTO_PD_MASK (0x1 << 8) +#define RT5639_AUTO_PD_SFT 8 +#define RT5639_AUTO_PD_DIS (0x0 << 8) +#define RT5639_AUTO_PD_EN (0x1 << 8) +#define RT5639_CLSD_OC_TH_MASK (0x3f) +#define RT5639_CLSD_OC_TH_SFT 0 + +/* Class D Output Control (0x8d) */ +#define RT5639_CLSD_RATIO_MASK (0xf << 12) +#define RT5639_CLSD_RATIO_SFT 12 +#define RT5639_CLSD_OM_MASK (0x1 << 11) +#define RT5639_CLSD_OM_SFT 11 +#define RT5639_CLSD_OM_MONO (0x0 << 11) +#define RT5639_CLSD_OM_STO (0x1 << 11) +#define RT5639_CLSD_SCH_MASK (0x1 << 10) +#define RT5639_CLSD_SCH_SFT 10 +#define RT5639_CLSD_SCH_L (0x0 << 10) +#define RT5639_CLSD_SCH_S (0x1 << 10) + +/* Depop Mode Control 1 (0x8e) */ +#define RT5639_SMT_TRIG_MASK (0x1 << 15) +#define RT5639_SMT_TRIG_SFT 15 +#define RT5639_SMT_TRIG_DIS (0x0 << 15) +#define RT5639_SMT_TRIG_EN (0x1 << 15) +#define RT5639_HP_L_SMT_MASK (0x1 << 9) +#define RT5639_HP_L_SMT_SFT 9 +#define RT5639_HP_L_SMT_DIS (0x0 << 9) +#define RT5639_HP_L_SMT_EN (0x1 << 9) +#define RT5639_HP_R_SMT_MASK (0x1 << 8) +#define RT5639_HP_R_SMT_SFT 8 +#define RT5639_HP_R_SMT_DIS (0x0 << 8) +#define RT5639_HP_R_SMT_EN (0x1 << 8) +#define RT5639_HP_CD_PD_MASK (0x1 << 7) +#define RT5639_HP_CD_PD_SFT 7 +#define RT5639_HP_CD_PD_DIS (0x0 << 7) +#define RT5639_HP_CD_PD_EN (0x1 << 7) +#define RT5639_RSTN_MASK (0x1 << 6) +#define RT5639_RSTN_SFT 6 +#define RT5639_RSTN_DIS (0x0 << 6) +#define RT5639_RSTN_EN (0x1 << 6) +#define RT5639_RSTP_MASK (0x1 << 5) +#define RT5639_RSTP_SFT 5 +#define RT5639_RSTP_DIS (0x0 << 5) +#define RT5639_RSTP_EN (0x1 << 5) +#define RT5639_HP_CO_MASK (0x1 << 4) +#define RT5639_HP_CO_SFT 4 +#define RT5639_HP_CO_DIS (0x0 << 4) +#define RT5639_HP_CO_EN (0x1 << 4) +#define RT5639_HP_CP_MASK (0x1 << 3) +#define RT5639_HP_CP_SFT 3 +#define RT5639_HP_CP_PD (0x0 << 3) +#define RT5639_HP_CP_PU (0x1 << 3) +#define RT5639_HP_SG_MASK (0x1 << 2) +#define RT5639_HP_SG_SFT 2 +#define RT5639_HP_SG_DIS (0x0 << 2) +#define RT5639_HP_SG_EN (0x1 << 2) +#define RT5639_HP_DP_MASK (0x1 << 1) +#define RT5639_HP_DP_SFT 1 +#define RT5639_HP_DP_PD (0x0 << 1) +#define RT5639_HP_DP_PU (0x1 << 1) +#define RT5639_HP_CB_MASK (0x1) +#define RT5639_HP_CB_SFT 0 +#define RT5639_HP_CB_PD (0x0) +#define RT5639_HP_CB_PU (0x1) + +/* Depop Mode Control 2 (0x8f) */ +#define RT5639_DEPOP_MASK (0x1 << 13) +#define RT5639_DEPOP_SFT 13 +#define RT5639_DEPOP_AUTO (0x0 << 13) +#define RT5639_DEPOP_MAN (0x1 << 13) +#define RT5639_RAMP_MASK (0x1 << 12) +#define RT5639_RAMP_SFT 12 +#define RT5639_RAMP_DIS (0x0 << 12) +#define RT5639_RAMP_EN (0x1 << 12) +#define RT5639_BPS_MASK (0x1 << 11) +#define RT5639_BPS_SFT 11 +#define RT5639_BPS_DIS (0x0 << 11) +#define RT5639_BPS_EN (0x1 << 11) +#define RT5639_FAST_UPDN_MASK (0x1 << 10) +#define RT5639_FAST_UPDN_SFT 10 +#define RT5639_FAST_UPDN_DIS (0x0 << 10) +#define RT5639_FAST_UPDN_EN (0x1 << 10) +#define RT5639_MRES_MASK (0x3 << 8) +#define RT5639_MRES_SFT 8 +#define RT5639_MRES_15MO (0x0 << 8) +#define RT5639_MRES_25MO (0x1 << 8) +#define RT5639_MRES_35MO (0x2 << 8) +#define RT5639_MRES_45MO (0x3 << 8) +#define RT5639_VLO_MASK (0x1 << 7) +#define RT5639_VLO_SFT 7 +#define RT5639_VLO_3V (0x0 << 7) +#define RT5639_VLO_32V (0x1 << 7) +#define RT5639_DIG_DP_MASK (0x1 << 6) +#define RT5639_DIG_DP_SFT 6 +#define RT5639_DIG_DP_DIS (0x0 << 6) +#define RT5639_DIG_DP_EN (0x1 << 6) +#define RT5639_DP_TH_MASK (0x3 << 4) +#define RT5639_DP_TH_SFT 4 + +/* Depop Mode Control 3 (0x90) */ +#define RT5639_CP_SYS_MASK (0x7 << 12) +#define RT5639_CP_SYS_SFT 12 +#define RT5639_CP_FQ1_MASK (0x7 << 8) +#define RT5639_CP_FQ1_SFT 8 +#define RT5639_CP_FQ2_MASK (0x7 << 4) +#define RT5639_CP_FQ2_SFT 4 +#define RT5639_CP_FQ3_MASK (0x7) +#define RT5639_CP_FQ3_SFT 0 +#define RT5639_CP_FQ_1_5_KHZ 0 +#define RT5639_CP_FQ_3_KHZ 1 +#define RT5639_CP_FQ_6_KHZ 2 +#define RT5639_CP_FQ_12_KHZ 3 +#define RT5639_CP_FQ_24_KHZ 4 +#define RT5639_CP_FQ_48_KHZ 5 +#define RT5639_CP_FQ_96_KHZ 6 +#define RT5639_CP_FQ_192_KHZ 7 + +/* HPOUT charge pump (0x91) */ +#define RT5639_OSW_L_MASK (0x1 << 11) +#define RT5639_OSW_L_SFT 11 +#define RT5639_OSW_L_DIS (0x0 << 11) +#define RT5639_OSW_L_EN (0x1 << 11) +#define RT5639_OSW_R_MASK (0x1 << 10) +#define RT5639_OSW_R_SFT 10 +#define RT5639_OSW_R_DIS (0x0 << 10) +#define RT5639_OSW_R_EN (0x1 << 10) +#define RT5639_PM_HP_MASK (0x3 << 8) +#define RT5639_PM_HP_SFT 8 +#define RT5639_PM_HP_LV (0x0 << 8) +#define RT5639_PM_HP_MV (0x1 << 8) +#define RT5639_PM_HP_HV (0x2 << 8) +#define RT5639_IB_HP_MASK (0x3 << 6) +#define RT5639_IB_HP_SFT 6 +#define RT5639_IB_HP_125IL (0x0 << 6) +#define RT5639_IB_HP_25IL (0x1 << 6) +#define RT5639_IB_HP_5IL (0x2 << 6) +#define RT5639_IB_HP_1IL (0x3 << 6) + +/* PV detection and SPK gain control (0x92) */ +#define RT5639_PVDD_DET_MASK (0x1 << 15) +#define RT5639_PVDD_DET_SFT 15 +#define RT5639_PVDD_DET_DIS (0x0 << 15) +#define RT5639_PVDD_DET_EN (0x1 << 15) +#define RT5639_SPK_AG_MASK (0x1 << 14) +#define RT5639_SPK_AG_SFT 14 +#define RT5639_SPK_AG_DIS (0x0 << 14) +#define RT5639_SPK_AG_EN (0x1 << 14) + +/* Micbias Control (0x93) */ +#define RT5639_MIC1_BS_MASK (0x1 << 15) +#define RT5639_MIC1_BS_SFT 15 +#define RT5639_MIC1_BS_9AV (0x0 << 15) +#define RT5639_MIC1_BS_75AV (0x1 << 15) +#define RT5639_MIC2_BS_MASK (0x1 << 14) +#define RT5639_MIC2_BS_SFT 14 +#define RT5639_MIC2_BS_9AV (0x0 << 14) +#define RT5639_MIC2_BS_75AV (0x1 << 14) +#define RT5639_MIC1_CLK_MASK (0x1 << 13) +#define RT5639_MIC1_CLK_SFT 13 +#define RT5639_MIC1_CLK_DIS (0x0 << 13) +#define RT5639_MIC1_CLK_EN (0x1 << 13) +#define RT5639_MIC2_CLK_MASK (0x1 << 12) +#define RT5639_MIC2_CLK_SFT 12 +#define RT5639_MIC2_CLK_DIS (0x0 << 12) +#define RT5639_MIC2_CLK_EN (0x1 << 12) +#define RT5639_MIC1_OVCD_MASK (0x1 << 11) +#define RT5639_MIC1_OVCD_SFT 11 +#define RT5639_MIC1_OVCD_DIS (0x0 << 11) +#define RT5639_MIC1_OVCD_EN (0x1 << 11) +#define RT5639_MIC1_OVTH_MASK (0x3 << 9) +#define RT5639_MIC1_OVTH_SFT 9 +#define RT5639_MIC1_OVTH_600UA (0x0 << 9) +#define RT5639_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5639_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5639_MIC2_OVCD_MASK (0x1 << 8) +#define RT5639_MIC2_OVCD_SFT 8 +#define RT5639_MIC2_OVCD_DIS (0x0 << 8) +#define RT5639_MIC2_OVCD_EN (0x1 << 8) +#define RT5639_MIC2_OVTH_MASK (0x3 << 6) +#define RT5639_MIC2_OVTH_SFT 6 +#define RT5639_MIC2_OVTH_600UA (0x0 << 6) +#define RT5639_MIC2_OVTH_1500UA (0x1 << 6) +#define RT5639_MIC2_OVTH_2000UA (0x2 << 6) +#define RT5639_PWR_MB_MASK (0x1 << 5) +#define RT5639_PWR_MB_SFT 5 +#define RT5639_PWR_MB_PD (0x0 << 5) +#define RT5639_PWR_MB_PU (0x1 << 5) +#define RT5639_PWR_CLK25M_MASK (0x1 << 4) +#define RT5639_PWR_CLK25M_SFT 4 +#define RT5639_PWR_CLK25M_PD (0x0 << 4) +#define RT5639_PWR_CLK25M_PU (0x1 << 4) + +/* EQ Control 1 (0xb0) */ +#define RT5639_EQ_SRC_MASK (0x1 << 15) +#define RT5639_EQ_SRC_SFT 15 +#define RT5639_EQ_SRC_DAC (0x0 << 15) +#define RT5639_EQ_SRC_ADC (0x1 << 15) +#define RT5639_EQ_UPD (0x1 << 14) +#define RT5639_EQ_UPD_BIT 14 +#define RT5639_EQ_CD_MASK (0x1 << 13) +#define RT5639_EQ_CD_SFT 13 +#define RT5639_EQ_CD_DIS (0x0 << 13) +#define RT5639_EQ_CD_EN (0x1 << 13) +#define RT5639_EQ_DITH_MASK (0x3 << 8) +#define RT5639_EQ_DITH_SFT 8 +#define RT5639_EQ_DITH_NOR (0x0 << 8) +#define RT5639_EQ_DITH_LSB (0x1 << 8) +#define RT5639_EQ_DITH_LSB_1 (0x2 << 8) +#define RT5639_EQ_DITH_LSB_2 (0x3 << 8) + +/* EQ Control 2 (0xb1) */ +#define RT5639_EQ_HPF1_M_MASK (0x1 << 8) +#define RT5639_EQ_HPF1_M_SFT 8 +#define RT5639_EQ_HPF1_M_HI (0x0 << 8) +#define RT5639_EQ_HPF1_M_1ST (0x1 << 8) +#define RT5639_EQ_LPF1_M_MASK (0x1 << 7) +#define RT5639_EQ_LPF1_M_SFT 7 +#define RT5639_EQ_LPF1_M_LO (0x0 << 7) +#define RT5639_EQ_LPF1_M_1ST (0x1 << 7) +#define RT5639_EQ_HPF2_MASK (0x1 << 6) +#define RT5639_EQ_HPF2_SFT 6 +#define RT5639_EQ_HPF2_DIS (0x0 << 6) +#define RT5639_EQ_HPF2_EN (0x1 << 6) +#define RT5639_EQ_HPF1_MASK (0x1 << 5) +#define RT5639_EQ_HPF1_SFT 5 +#define RT5639_EQ_HPF1_DIS (0x0 << 5) +#define RT5639_EQ_HPF1_EN (0x1 << 5) +#define RT5639_EQ_BPF4_MASK (0x1 << 4) +#define RT5639_EQ_BPF4_SFT 4 +#define RT5639_EQ_BPF4_DIS (0x0 << 4) +#define RT5639_EQ_BPF4_EN (0x1 << 4) +#define RT5639_EQ_BPF3_MASK (0x1 << 3) +#define RT5639_EQ_BPF3_SFT 3 +#define RT5639_EQ_BPF3_DIS (0x0 << 3) +#define RT5639_EQ_BPF3_EN (0x1 << 3) +#define RT5639_EQ_BPF2_MASK (0x1 << 2) +#define RT5639_EQ_BPF2_SFT 2 +#define RT5639_EQ_BPF2_DIS (0x0 << 2) +#define RT5639_EQ_BPF2_EN (0x1 << 2) +#define RT5639_EQ_BPF1_MASK (0x1 << 1) +#define RT5639_EQ_BPF1_SFT 1 +#define RT5639_EQ_BPF1_DIS (0x0 << 1) +#define RT5639_EQ_BPF1_EN (0x1 << 1) +#define RT5639_EQ_LPF_MASK (0x1) +#define RT5639_EQ_LPF_SFT 0 +#define RT5639_EQ_LPF_DIS (0x0) +#define RT5639_EQ_LPF_EN (0x1) + +/* Memory Test (0xb2) */ +#define RT5639_MT_MASK (0x1 << 15) +#define RT5639_MT_SFT 15 +#define RT5639_MT_DIS (0x0 << 15) +#define RT5639_MT_EN (0x1 << 15) + +/* DRC/AGC Control 1 (0xb4) */ +#define RT5639_DRC_AGC_P_MASK (0x1 << 15) +#define RT5639_DRC_AGC_P_SFT 15 +#define RT5639_DRC_AGC_P_DAC (0x0 << 15) +#define RT5639_DRC_AGC_P_ADC (0x1 << 15) +#define RT5639_DRC_AGC_MASK (0x1 << 14) +#define RT5639_DRC_AGC_SFT 14 +#define RT5639_DRC_AGC_DIS (0x0 << 14) +#define RT5639_DRC_AGC_EN (0x1 << 14) +#define RT5639_DRC_AGC_UPD (0x1 << 13) +#define RT5639_DRC_AGC_UPD_BIT 13 +#define RT5639_DRC_AGC_AR_MASK (0x1f << 8) +#define RT5639_DRC_AGC_AR_SFT 8 +#define RT5639_DRC_AGC_R_MASK (0x7 << 5) +#define RT5639_DRC_AGC_R_SFT 5 +#define RT5639_DRC_AGC_R_48K (0x1 << 5) +#define RT5639_DRC_AGC_R_96K (0x2 << 5) +#define RT5639_DRC_AGC_R_192K (0x3 << 5) +#define RT5639_DRC_AGC_R_441K (0x5 << 5) +#define RT5639_DRC_AGC_R_882K (0x6 << 5) +#define RT5639_DRC_AGC_R_1764K (0x7 << 5) +#define RT5639_DRC_AGC_RC_MASK (0x1f) +#define RT5639_DRC_AGC_RC_SFT 0 + +/* DRC/AGC Control 2 (0xb5) */ +#define RT5639_DRC_AGC_POB_MASK (0x3f << 8) +#define RT5639_DRC_AGC_POB_SFT 8 +#define RT5639_DRC_AGC_CP_MASK (0x1 << 7) +#define RT5639_DRC_AGC_CP_SFT 7 +#define RT5639_DRC_AGC_CP_DIS (0x0 << 7) +#define RT5639_DRC_AGC_CP_EN (0x1 << 7) +#define RT5639_DRC_AGC_CPR_MASK (0x3 << 5) +#define RT5639_DRC_AGC_CPR_SFT 5 +#define RT5639_DRC_AGC_CPR_1_1 (0x0 << 5) +#define RT5639_DRC_AGC_CPR_1_2 (0x1 << 5) +#define RT5639_DRC_AGC_CPR_1_3 (0x2 << 5) +#define RT5639_DRC_AGC_CPR_1_4 (0x3 << 5) +#define RT5639_DRC_AGC_PRB_MASK (0x1f) +#define RT5639_DRC_AGC_PRB_SFT 0 + +/* DRC/AGC Control 3 (0xb6) */ +#define RT5639_DRC_AGC_NGB_MASK (0xf << 12) +#define RT5639_DRC_AGC_NGB_SFT 12 +#define RT5639_DRC_AGC_TAR_MASK (0x1f << 7) +#define RT5639_DRC_AGC_TAR_SFT 7 +#define RT5639_DRC_AGC_NG_MASK (0x1 << 6) +#define RT5639_DRC_AGC_NG_SFT 6 +#define RT5639_DRC_AGC_NG_DIS (0x0 << 6) +#define RT5639_DRC_AGC_NG_EN (0x1 << 6) +#define RT5639_DRC_AGC_NGH_MASK (0x1 << 5) +#define RT5639_DRC_AGC_NGH_SFT 5 +#define RT5639_DRC_AGC_NGH_DIS (0x0 << 5) +#define RT5639_DRC_AGC_NGH_EN (0x1 << 5) +#define RT5639_DRC_AGC_NGT_MASK (0x1f) +#define RT5639_DRC_AGC_NGT_SFT 0 + +/* Jack Detect Control (0xbb) */ +#define RT5639_JD_MASK (0x7 << 13) +#define RT5639_JD_SFT 13 +#define RT5639_JD_DIS (0x0 << 13) +#define RT5639_JD_GPIO1 (0x1 << 13) +#define RT5639_JD_JD1_IN4P (0x2 << 13) +#define RT5639_JD_JD2_IN4N (0x3 << 13) +#define RT5639_JD_GPIO2 (0x4 << 13) +#define RT5639_JD_GPIO3 (0x5 << 13) +#define RT5639_JD_GPIO4 (0x6 << 13) +#define RT5639_JD_HP_MASK (0x1 << 11) +#define RT5639_JD_HP_SFT 11 +#define RT5639_JD_HP_DIS (0x0 << 11) +#define RT5639_JD_HP_EN (0x1 << 11) +#define RT5639_JD_HP_TRG_MASK (0x1 << 10) +#define RT5639_JD_HP_TRG_SFT 10 +#define RT5639_JD_HP_TRG_LO (0x0 << 10) +#define RT5639_JD_HP_TRG_HI (0x1 << 10) +#define RT5639_JD_SPL_MASK (0x1 << 9) +#define RT5639_JD_SPL_SFT 9 +#define RT5639_JD_SPL_DIS (0x0 << 9) +#define RT5639_JD_SPL_EN (0x1 << 9) +#define RT5639_JD_SPL_TRG_MASK (0x1 << 8) +#define RT5639_JD_SPL_TRG_SFT 8 +#define RT5639_JD_SPL_TRG_LO (0x0 << 8) +#define RT5639_JD_SPL_TRG_HI (0x1 << 8) +#define RT5639_JD_SPR_MASK (0x1 << 7) +#define RT5639_JD_SPR_SFT 7 +#define RT5639_JD_SPR_DIS (0x0 << 7) +#define RT5639_JD_SPR_EN (0x1 << 7) +#define RT5639_JD_SPR_TRG_MASK (0x1 << 6) +#define RT5639_JD_SPR_TRG_SFT 6 +#define RT5639_JD_SPR_TRG_LO (0x0 << 6) +#define RT5639_JD_SPR_TRG_HI (0x1 << 6) +#define RT5639_JD_MO_MASK (0x1 << 5) +#define RT5639_JD_MO_SFT 5 +#define RT5639_JD_MO_DIS (0x0 << 5) +#define RT5639_JD_MO_EN (0x1 << 5) +#define RT5639_JD_MO_TRG_MASK (0x1 << 4) +#define RT5639_JD_MO_TRG_SFT 4 +#define RT5639_JD_MO_TRG_LO (0x0 << 4) +#define RT5639_JD_MO_TRG_HI (0x1 << 4) +#define RT5639_JD_LO_MASK (0x1 << 3) +#define RT5639_JD_LO_SFT 3 +#define RT5639_JD_LO_DIS (0x0 << 3) +#define RT5639_JD_LO_EN (0x1 << 3) +#define RT5639_JD_LO_TRG_MASK (0x1 << 2) +#define RT5639_JD_LO_TRG_SFT 2 +#define RT5639_JD_LO_TRG_LO (0x0 << 2) +#define RT5639_JD_LO_TRG_HI (0x1 << 2) +#define RT5639_JD1_IN4P_MASK (0x1 << 1) +#define RT5639_JD1_IN4P_SFT 1 +#define RT5639_JD1_IN4P_DIS (0x0 << 1) +#define RT5639_JD1_IN4P_EN (0x1 << 1) +#define RT5639_JD2_IN4N_MASK (0x1) +#define RT5639_JD2_IN4N_SFT 0 +#define RT5639_JD2_IN4N_DIS (0x0) +#define RT5639_JD2_IN4N_EN (0x1) + +/* IRQ Control 1 (0xbd) */ +#define RT5639_IRQ_JD_MASK (0x1 << 15) +#define RT5639_IRQ_JD_SFT 15 +#define RT5639_IRQ_JD_BP (0x0 << 15) +#define RT5639_IRQ_JD_NOR (0x1 << 15) +#define RT5639_IRQ_OT_MASK (0x1 << 14) +#define RT5639_IRQ_OT_SFT 14 +#define RT5639_IRQ_OT_BP (0x0 << 14) +#define RT5639_IRQ_OT_NOR (0x1 << 14) +#define RT5639_JD_STKY_MASK (0x1 << 13) +#define RT5639_JD_STKY_SFT 13 +#define RT5639_JD_STKY_DIS (0x0 << 13) +#define RT5639_JD_STKY_EN (0x1 << 13) +#define RT5639_OT_STKY_MASK (0x1 << 12) +#define RT5639_OT_STKY_SFT 12 +#define RT5639_OT_STKY_DIS (0x0 << 12) +#define RT5639_OT_STKY_EN (0x1 << 12) +#define RT5639_JD_P_MASK (0x1 << 11) +#define RT5639_JD_P_SFT 11 +#define RT5639_JD_P_NOR (0x0 << 11) +#define RT5639_JD_P_INV (0x1 << 11) +#define RT5639_OT_P_MASK (0x1 << 10) +#define RT5639_OT_P_SFT 10 +#define RT5639_OT_P_NOR (0x0 << 10) +#define RT5639_OT_P_INV (0x1 << 10) + +/* IRQ Control 2 (0xbe) */ +#define RT5639_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5639_IRQ_MB1_OC_SFT 15 +#define RT5639_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5639_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5639_IRQ_MB2_OC_MASK (0x1 << 14) +#define RT5639_IRQ_MB2_OC_SFT 14 +#define RT5639_IRQ_MB2_OC_BP (0x0 << 14) +#define RT5639_IRQ_MB2_OC_NOR (0x1 << 14) +#define RT5639_MB1_OC_STKY_MASK (0x1 << 11) +#define RT5639_MB1_OC_STKY_SFT 11 +#define RT5639_MB1_OC_STKY_DIS (0x0 << 11) +#define RT5639_MB1_OC_STKY_EN (0x1 << 11) +#define RT5639_MB2_OC_STKY_MASK (0x1 << 10) +#define RT5639_MB2_OC_STKY_SFT 10 +#define RT5639_MB2_OC_STKY_DIS (0x0 << 10) +#define RT5639_MB2_OC_STKY_EN (0x1 << 10) +#define RT5639_MB1_OC_P_MASK (0x1 << 7) +#define RT5639_MB1_OC_P_SFT 7 +#define RT5639_MB1_OC_P_NOR (0x0 << 7) +#define RT5639_MB1_OC_P_INV (0x1 << 7) +#define RT5639_MB2_OC_P_MASK (0x1 << 6) +#define RT5639_MB2_OC_P_SFT 6 +#define RT5639_MB2_OC_P_NOR (0x0 << 6) +#define RT5639_MB2_OC_P_INV (0x1 << 6) +#define RT5639_MB1_OC_CLR (0x1 << 3) +#define RT5639_MB1_OC_CLR_SFT 3 +#define RT5639_MB2_OC_CLR (0x1 << 2) +#define RT5639_MB2_OC_CLR_SFT 2 + +/* GPIO Control 1 (0xc0) */ +#define RT5639_GP1_PIN_MASK (0x1 << 15) +#define RT5639_GP1_PIN_SFT 15 +#define RT5639_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5639_GP1_PIN_IRQ (0x1 << 15) +#define RT5639_GP2_PIN_MASK (0x1 << 14) +#define RT5639_GP2_PIN_SFT 14 +#define RT5639_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5639_GP2_PIN_DMIC1_SCL (0x1 << 14) +#define RT5639_GP3_PIN_MASK (0x3 << 12) +#define RT5639_GP3_PIN_SFT 12 +#define RT5639_GP3_PIN_GPIO3 (0x0 << 12) +#define RT5639_GP3_PIN_DMIC1_SDA (0x1 << 12) +#define RT5639_GP3_PIN_IRQ (0x2 << 12) +#define RT5639_GP4_PIN_MASK (0x1 << 11) +#define RT5639_GP4_PIN_SFT 11 +#define RT5639_GP4_PIN_GPIO4 (0x0 << 11) +#define RT5639_GP4_PIN_DMIC2_SDA (0x1 << 11) +#define RT5639_DP_SIG_MASK (0x1 << 10) +#define RT5639_DP_SIG_SFT 10 +#define RT5639_DP_SIG_TEST (0x0 << 10) +#define RT5639_DP_SIG_AP (0x1 << 10) +#define RT5639_GPIO_M_MASK (0x1 << 9) +#define RT5639_GPIO_M_SFT 9 +#define RT5639_GPIO_M_FLT (0x0 << 9) +#define RT5639_GPIO_M_PH (0x1 << 9) + +/* GPIO Control 3 (0xc2) */ +#define RT5639_GP4_PF_MASK (0x1 << 11) +#define RT5639_GP4_PF_SFT 11 +#define RT5639_GP4_PF_IN (0x0 << 11) +#define RT5639_GP4_PF_OUT (0x1 << 11) +#define RT5639_GP4_OUT_MASK (0x1 << 10) +#define RT5639_GP4_OUT_SFT 10 +#define RT5639_GP4_OUT_LO (0x0 << 10) +#define RT5639_GP4_OUT_HI (0x1 << 10) +#define RT5639_GP4_P_MASK (0x1 << 9) +#define RT5639_GP4_P_SFT 9 +#define RT5639_GP4_P_NOR (0x0 << 9) +#define RT5639_GP4_P_INV (0x1 << 9) +#define RT5639_GP3_PF_MASK (0x1 << 8) +#define RT5639_GP3_PF_SFT 8 +#define RT5639_GP3_PF_IN (0x0 << 8) +#define RT5639_GP3_PF_OUT (0x1 << 8) +#define RT5639_GP3_OUT_MASK (0x1 << 7) +#define RT5639_GP3_OUT_SFT 7 +#define RT5639_GP3_OUT_LO (0x0 << 7) +#define RT5639_GP3_OUT_HI (0x1 << 7) +#define RT5639_GP3_P_MASK (0x1 << 6) +#define RT5639_GP3_P_SFT 6 +#define RT5639_GP3_P_NOR (0x0 << 6) +#define RT5639_GP3_P_INV (0x1 << 6) +#define RT5639_GP2_PF_MASK (0x1 << 5) +#define RT5639_GP2_PF_SFT 5 +#define RT5639_GP2_PF_IN (0x0 << 5) +#define RT5639_GP2_PF_OUT (0x1 << 5) +#define RT5639_GP2_OUT_MASK (0x1 << 4) +#define RT5639_GP2_OUT_SFT 4 +#define RT5639_GP2_OUT_LO (0x0 << 4) +#define RT5639_GP2_OUT_HI (0x1 << 4) +#define RT5639_GP2_P_MASK (0x1 << 3) +#define RT5639_GP2_P_SFT 3 +#define RT5639_GP2_P_NOR (0x0 << 3) +#define RT5639_GP2_P_INV (0x1 << 3) +#define RT5639_GP1_PF_MASK (0x1 << 2) +#define RT5639_GP1_PF_SFT 2 +#define RT5639_GP1_PF_IN (0x0 << 2) +#define RT5639_GP1_PF_OUT (0x1 << 2) +#define RT5639_GP1_OUT_MASK (0x1 << 1) +#define RT5639_GP1_OUT_SFT 1 +#define RT5639_GP1_OUT_LO (0x0 << 1) +#define RT5639_GP1_OUT_HI (0x1 << 1) +#define RT5639_GP1_P_MASK (0x1) +#define RT5639_GP1_P_SFT 0 +#define RT5639_GP1_P_NOR (0x0) +#define RT5639_GP1_P_INV (0x1) + +/* FM34-500 Register Control 1 (0xc4) */ +#define RT5639_DSP_ADD_SFT 0 + +/* FM34-500 Register Control 2 (0xc5) */ +#define RT5639_DSP_DAT_SFT 0 + +/* FM34-500 Register Control 3 (0xc6) */ +#define RT5639_DSP_BUSY_MASK (0x1 << 15) +#define RT5639_DSP_BUSY_BIT 15 +#define RT5639_DSP_DS_MASK (0x1 << 14) +#define RT5639_DSP_DS_SFT 14 +#define RT5639_DSP_DS_FM3010 (0x1 << 14) +#define RT5639_DSP_DS_TEMP (0x1 << 14) +#define RT5639_DSP_CLK_MASK (0x3 << 12) +#define RT5639_DSP_CLK_SFT 12 +#define RT5639_DSP_CLK_384K (0x0 << 12) +#define RT5639_DSP_CLK_192K (0x1 << 12) +#define RT5639_DSP_CLK_96K (0x2 << 12) +#define RT5639_DSP_CLK_64K (0x3 << 12) +#define RT5639_DSP_PD_PIN_MASK (0x1 << 11) +#define RT5639_DSP_PD_PIN_SFT 11 +#define RT5639_DSP_PD_PIN_LO (0x0 << 11) +#define RT5639_DSP_PD_PIN_HI (0x1 << 11) +#define RT5639_DSP_RST_PIN_MASK (0x1 << 10) +#define RT5639_DSP_RST_PIN_SFT 10 +#define RT5639_DSP_RST_PIN_LO (0x0 << 10) +#define RT5639_DSP_RST_PIN_HI (0x1 << 10) +#define RT5639_DSP_R_EN (0x1 << 9) +#define RT5639_DSP_R_EN_BIT 9 +#define RT5639_DSP_W_EN (0x1 << 8) +#define RT5639_DSP_W_EN_BIT 8 +#define RT5639_DSP_CMD_MASK (0xff) +#define RT5639_DSP_CMD_SFT 0 +#define RT5639_DSP_CMD_MW (0x3B) /* Memory Write */ +#define RT5639_DSP_CMD_MR (0x37) /* Memory Read */ +#define RT5639_DSP_CMD_RR (0x60) /* Register Read */ +#define RT5639_DSP_CMD_RW (0x68) /* Register Write */ + +/* Programmable Register Array Control 1 (0xc8) */ +#define RT5639_REG_SEQ_MASK (0xf << 12) +#define RT5639_REG_SEQ_SFT 12 +#define RT5639_SEQ1_ST_MASK (0x1 << 11) /*RO*/ +#define RT5639_SEQ1_ST_SFT 11 +#define RT5639_SEQ1_ST_RUN (0x0 << 11) +#define RT5639_SEQ1_ST_FIN (0x1 << 11) +#define RT5639_SEQ2_ST_MASK (0x1 << 10) /*RO*/ +#define RT5639_SEQ2_ST_SFT 10 +#define RT5639_SEQ2_ST_RUN (0x0 << 10) +#define RT5639_SEQ2_ST_FIN (0x1 << 10) +#define RT5639_REG_LV_MASK (0x1 << 9) +#define RT5639_REG_LV_SFT 9 +#define RT5639_REG_LV_MX (0x0 << 9) +#define RT5639_REG_LV_PR (0x1 << 9) +#define RT5639_SEQ_2_PT_MASK (0x1 << 8) +#define RT5639_SEQ_2_PT_BIT 8 +#define RT5639_REG_IDX_MASK (0xff) +#define RT5639_REG_IDX_SFT 0 + +/* Programmable Register Array Control 2 (0xc9) */ +#define RT5639_REG_DAT_MASK (0xffff) +#define RT5639_REG_DAT_SFT 0 + +/* Programmable Register Array Control 3 (0xca) */ +#define RT5639_SEQ_DLY_MASK (0xff << 8) +#define RT5639_SEQ_DLY_SFT 8 +#define RT5639_PROG_MASK (0x1 << 7) +#define RT5639_PROG_SFT 7 +#define RT5639_PROG_DIS (0x0 << 7) +#define RT5639_PROG_EN (0x1 << 7) +#define RT5639_SEQ1_PT_RUN (0x1 << 6) +#define RT5639_SEQ1_PT_RUN_BIT 6 +#define RT5639_SEQ2_PT_RUN (0x1 << 5) +#define RT5639_SEQ2_PT_RUN_BIT 5 + +/* Programmable Register Array Control 4 (0xcb) */ +#define RT5639_SEQ1_START_MASK (0xf << 8) +#define RT5639_SEQ1_START_SFT 8 +#define RT5639_SEQ1_END_MASK (0xf) +#define RT5639_SEQ1_END_SFT 0 + +/* Programmable Register Array Control 5 (0xcc) */ +#define RT5639_SEQ2_START_MASK (0xf << 8) +#define RT5639_SEQ2_START_SFT 8 +#define RT5639_SEQ2_END_MASK (0xf) +#define RT5639_SEQ2_END_SFT 0 + +/* Scramble Function (0xcd) */ +#define RT5639_SCB_KEY_MASK (0xff) +#define RT5639_SCB_KEY_SFT 0 + +/* Scramble Control (0xce) */ +#define RT5639_SCB_SWAP_MASK (0x1 << 15) +#define RT5639_SCB_SWAP_SFT 15 +#define RT5639_SCB_SWAP_DIS (0x0 << 15) +#define RT5639_SCB_SWAP_EN (0x1 << 15) +#define RT5639_SCB_MASK (0x1 << 14) +#define RT5639_SCB_SFT 14 +#define RT5639_SCB_DIS (0x0 << 14) +#define RT5639_SCB_EN (0x1 << 14) + +/* Baseback Control (0xcf) */ +#define RT5639_BB_MASK (0x1 << 15) +#define RT5639_BB_SFT 15 +#define RT5639_BB_DIS (0x0 << 15) +#define RT5639_BB_EN (0x1 << 15) +#define RT5639_BB_CT_MASK (0x7 << 12) +#define RT5639_BB_CT_SFT 12 +#define RT5639_BB_CT_A (0x0 << 12) +#define RT5639_BB_CT_B (0x1 << 12) +#define RT5639_BB_CT_C (0x2 << 12) +#define RT5639_BB_CT_D (0x3 << 12) +#define RT5639_M_BB_L_MASK (0x1 << 9) +#define RT5639_M_BB_L_SFT 9 +#define RT5639_M_BB_R_MASK (0x1 << 8) +#define RT5639_M_BB_R_SFT 8 +#define RT5639_M_BB_HPF_L_MASK (0x1 << 7) +#define RT5639_M_BB_HPF_L_SFT 7 +#define RT5639_M_BB_HPF_R_MASK (0x1 << 6) +#define RT5639_M_BB_HPF_R_SFT 6 +#define RT5639_G_BB_BST_MASK (0x3f) +#define RT5639_G_BB_BST_SFT 0 + +/* MP3 Plus Control 1 (0xd0) */ +#define RT5639_M_MP3_L_MASK (0x1 << 15) +#define RT5639_M_MP3_L_SFT 15 +#define RT5639_M_MP3_R_MASK (0x1 << 14) +#define RT5639_M_MP3_R_SFT 14 +#define RT5639_M_MP3_MASK (0x1 << 13) +#define RT5639_M_MP3_SFT 13 +#define RT5639_M_MP3_DIS (0x0 << 13) +#define RT5639_M_MP3_EN (0x1 << 13) +#define RT5639_EG_MP3_MASK (0x1f << 8) +#define RT5639_EG_MP3_SFT 8 +#define RT5639_MP3_HLP_MASK (0x1 << 7) +#define RT5639_MP3_HLP_SFT 7 +#define RT5639_MP3_HLP_DIS (0x0 << 7) +#define RT5639_MP3_HLP_EN (0x1 << 7) +#define RT5639_M_MP3_ORG_L_MASK (0x1 << 6) +#define RT5639_M_MP3_ORG_L_SFT 6 +#define RT5639_M_MP3_ORG_R_MASK (0x1 << 5) +#define RT5639_M_MP3_ORG_R_SFT 5 + +/* MP3 Plus Control 2 (0xd1) */ +#define RT5639_MP3_WT_MASK (0x1 << 13) +#define RT5639_MP3_WT_SFT 13 +#define RT5639_MP3_WT_1_4 (0x0 << 13) +#define RT5639_MP3_WT_1_2 (0x1 << 13) +#define RT5639_OG_MP3_MASK (0x1f << 8) +#define RT5639_OG_MP3_SFT 8 +#define RT5639_HG_MP3_MASK (0x3f) +#define RT5639_HG_MP3_SFT 0 + +/* 3D HP Control 1 (0xd2) */ +#define RT5639_3D_CF_MASK (0x1 << 15) +#define RT5639_3D_CF_SFT 15 +#define RT5639_3D_CF_DIS (0x0 << 15) +#define RT5639_3D_CF_EN (0x1 << 15) +#define RT5639_3D_HP_MASK (0x1 << 14) +#define RT5639_3D_HP_SFT 14 +#define RT5639_3D_HP_DIS (0x0 << 14) +#define RT5639_3D_HP_EN (0x1 << 14) +#define RT5639_3D_BT_MASK (0x1 << 13) +#define RT5639_3D_BT_SFT 13 +#define RT5639_3D_BT_DIS (0x0 << 13) +#define RT5639_3D_BT_EN (0x1 << 13) +#define RT5639_3D_1F_MIX_MASK (0x3 << 11) +#define RT5639_3D_1F_MIX_SFT 11 +#define RT5639_3D_HP_M_MASK (0x1 << 10) +#define RT5639_3D_HP_M_SFT 10 +#define RT5639_3D_HP_M_SUR (0x0 << 10) +#define RT5639_3D_HP_M_FRO (0x1 << 10) +#define RT5639_M_3D_HRTF_MASK (0x1 << 9) +#define RT5639_M_3D_HRTF_SFT 9 +#define RT5639_M_3D_D2H_MASK (0x1 << 8) +#define RT5639_M_3D_D2H_SFT 8 +#define RT5639_M_3D_D2R_MASK (0x1 << 7) +#define RT5639_M_3D_D2R_SFT 7 +#define RT5639_M_3D_REVB_MASK (0x1 << 6) +#define RT5639_M_3D_REVB_SFT 6 + +/* Adjustable high pass filter control 1 (0xd3) */ +#define RT5639_2ND_HPF_MASK (0x1 << 15) +#define RT5639_2ND_HPF_SFT 15 +#define RT5639_2ND_HPF_DIS (0x0 << 15) +#define RT5639_2ND_HPF_EN (0x1 << 15) +#define RT5639_HPF_CF_L_MASK (0x7 << 12) +#define RT5639_HPF_CF_L_SFT 12 +#define RT5639_1ST_HPF_MASK (0x1 << 11) +#define RT5639_1ST_HPF_SFT 11 +#define RT5639_1ST_HPF_DIS (0x0 << 11) +#define RT5639_1ST_HPF_EN (0x1 << 11) +#define RT5639_HPF_CF_R_MASK (0x7 << 8) +#define RT5639_HPF_CF_R_SFT 8 +#define RT5639_ZD_T_MASK (0x3 << 6) +#define RT5639_ZD_T_SFT 6 +#define RT5639_ZD_F_MASK (0x3 << 4) +#define RT5639_ZD_F_SFT 4 +#define RT5639_ZD_F_IM (0x0 << 4) +#define RT5639_ZD_F_ZC_IM (0x1 << 4) +#define RT5639_ZD_F_ZC_IOD (0x2 << 4) +#define RT5639_ZD_F_UN (0x3 << 4) + +/* HP calibration control and Amp detection (0xd6) */ +#define RT5639_SI_DAC_MASK (0x1 << 11) +#define RT5639_SI_DAC_SFT 11 +#define RT5639_SI_DAC_AUTO (0x0 << 11) +#define RT5639_SI_DAC_TEST (0x1 << 11) +#define RT5639_DC_CAL_M_MASK (0x1 << 10) +#define RT5639_DC_CAL_M_SFT 10 +#define RT5639_DC_CAL_M_CAL (0x0 << 10) +#define RT5639_DC_CAL_M_NOR (0x1 << 10) +#define RT5639_DC_CAL_MASK (0x1 << 9) +#define RT5639_DC_CAL_SFT 9 +#define RT5639_DC_CAL_DIS (0x0 << 9) +#define RT5639_DC_CAL_EN (0x1 << 9) +#define RT5639_HPD_RCV_MASK (0x7 << 6) +#define RT5639_HPD_RCV_SFT 6 +#define RT5639_HPD_PS_MASK (0x1 << 5) +#define RT5639_HPD_PS_SFT 5 +#define RT5639_HPD_PS_DIS (0x0 << 5) +#define RT5639_HPD_PS_EN (0x1 << 5) +#define RT5639_CAL_M_MASK (0x1 << 4) +#define RT5639_CAL_M_SFT 4 +#define RT5639_CAL_M_DEP (0x0 << 4) +#define RT5639_CAL_M_CAL (0x1 << 4) +#define RT5639_CAL_MASK (0x1 << 3) +#define RT5639_CAL_SFT 3 +#define RT5639_CAL_DIS (0x0 << 3) +#define RT5639_CAL_EN (0x1 << 3) +#define RT5639_CAL_TEST_MASK (0x1 << 2) +#define RT5639_CAL_TEST_SFT 2 +#define RT5639_CAL_TEST_DIS (0x0 << 2) +#define RT5639_CAL_TEST_EN (0x1 << 2) +#define RT5639_CAL_P_MASK (0x3) +#define RT5639_CAL_P_SFT 0 +#define RT5639_CAL_P_NONE (0x0) +#define RT5639_CAL_P_CAL (0x1) +#define RT5639_CAL_P_DAC_CAL (0x2) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5639_SV_MASK (0x1 << 15) +#define RT5639_SV_SFT 15 +#define RT5639_SV_DIS (0x0 << 15) +#define RT5639_SV_EN (0x1 << 15) +#define RT5639_SPO_SV_MASK (0x1 << 14) +#define RT5639_SPO_SV_SFT 14 +#define RT5639_SPO_SV_DIS (0x0 << 14) +#define RT5639_SPO_SV_EN (0x1 << 14) +#define RT5639_OUT_SV_MASK (0x1 << 13) +#define RT5639_OUT_SV_SFT 13 +#define RT5639_OUT_SV_DIS (0x0 << 13) +#define RT5639_OUT_SV_EN (0x1 << 13) +#define RT5639_HP_SV_MASK (0x1 << 12) +#define RT5639_HP_SV_SFT 12 +#define RT5639_HP_SV_DIS (0x0 << 12) +#define RT5639_HP_SV_EN (0x1 << 12) +#define RT5639_ZCD_DIG_MASK (0x1 << 11) +#define RT5639_ZCD_DIG_SFT 11 +#define RT5639_ZCD_DIG_DIS (0x0 << 11) +#define RT5639_ZCD_DIG_EN (0x1 << 11) +#define RT5639_ZCD_MASK (0x1 << 10) +#define RT5639_ZCD_SFT 10 +#define RT5639_ZCD_PD (0x0 << 10) +#define RT5639_ZCD_PU (0x1 << 10) +#define RT5639_M_ZCD_MASK (0x3f << 4) +#define RT5639_M_ZCD_SFT 4 +#define RT5639_M_ZCD_RM_L (0x1 << 9) +#define RT5639_M_ZCD_RM_R (0x1 << 8) +#define RT5639_M_ZCD_SM_L (0x1 << 7) +#define RT5639_M_ZCD_SM_R (0x1 << 6) +#define RT5639_M_ZCD_OM_L (0x1 << 5) +#define RT5639_M_ZCD_OM_R (0x1 << 4) +#define RT5639_SV_DLY_MASK (0xf) +#define RT5639_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5639_ZCD_HP_MASK (0x1 << 15) +#define RT5639_ZCD_HP_SFT 15 +#define RT5639_ZCD_HP_DIS (0x0 << 15) +#define RT5639_ZCD_HP_EN (0x1 << 15) + + +/* Codec Private Register definition */ +/* 3D Speaker Control (0x63) */ +#define RT5639_3D_SPK_MASK (0x1 << 15) +#define RT5639_3D_SPK_SFT 15 +#define RT5639_3D_SPK_DIS (0x0 << 15) +#define RT5639_3D_SPK_EN (0x1 << 15) +#define RT5639_3D_SPK_M_MASK (0x3 << 13) +#define RT5639_3D_SPK_M_SFT 13 +#define RT5639_3D_SPK_CG_MASK (0x1f << 8) +#define RT5639_3D_SPK_CG_SFT 8 +#define RT5639_3D_SPK_SG_MASK (0x1f) +#define RT5639_3D_SPK_SG_SFT 0 + +/* Wind Noise Detection Control 1 (0x6c) */ +#define RT5639_WND_MASK (0x1 << 15) +#define RT5639_WND_SFT 15 +#define RT5639_WND_DIS (0x0 << 15) +#define RT5639_WND_EN (0x1 << 15) + +/* Wind Noise Detection Control 2 (0x6d) */ +#define RT5639_WND_FC_NW_MASK (0x3f << 10) +#define RT5639_WND_FC_NW_SFT 10 +#define RT5639_WND_FC_WK_MASK (0x3f << 4) +#define RT5639_WND_FC_WK_SFT 4 + +/* Wind Noise Detection Control 3 (0x6e) */ +#define RT5639_HPF_FC_MASK (0x3f << 6) +#define RT5639_HPF_FC_SFT 6 +#define RT5639_WND_FC_ST_MASK (0x3f) +#define RT5639_WND_FC_ST_SFT 0 + +/* Wind Noise Detection Control 4 (0x6f) */ +#define RT5639_WND_TH_LO_MASK (0x3ff) +#define RT5639_WND_TH_LO_SFT 0 + +/* Wind Noise Detection Control 5 (0x70) */ +#define RT5639_WND_TH_HI_MASK (0x3ff) +#define RT5639_WND_TH_HI_SFT 0 + +/* Wind Noise Detection Control 8 (0x73) */ +#define RT5639_WND_WIND_MASK (0x1 << 13) /* Read-Only */ +#define RT5639_WND_WIND_SFT 13 +#define RT5639_WND_STRONG_MASK (0x1 << 12) /* Read-Only */ +#define RT5639_WND_STRONG_SFT 12 +enum { + RT5639_NO_WIND, + RT5639_BREEZE, + RT5639_STORM, +}; + +/* Dipole Speaker Interface (0x75) */ +#define RT5639_DP_ATT_MASK (0x3 << 14) +#define RT5639_DP_ATT_SFT 14 +#define RT5639_DP_SPK_MASK (0x1 << 10) +#define RT5639_DP_SPK_SFT 10 +#define RT5639_DP_SPK_DIS (0x0 << 10) +#define RT5639_DP_SPK_EN (0x1 << 10) + +/* EQ Pre Volume Control (0xb3) */ +#define RT5639_EQ_PRE_VOL_MASK (0xffff) +#define RT5639_EQ_PRE_VOL_SFT 0 + +/* EQ Post Volume Control (0xb4) */ +#define RT5639_EQ_PST_VOL_MASK (0xffff) +#define RT5639_EQ_PST_VOL_SFT 0 + +#define RT5639_NO_JACK BIT(0) +#define RT5639_HEADSET_DET BIT(1) +#define RT5639_HEADPHO_DET BIT(2) + +/* System Clock Source */ +#define RT5639_SCLK_S_MCLK 0 +#define RT5639_SCLK_S_PLL1 1 +#define RT5639_SCLK_S_PLL1_TK 2 +#define RT5639_SCLK_S_RCCLK 3 + +/* PLL1 Source */ +#define RT5639_PLL1_S_MCLK 0 +#define RT5639_PLL1_S_BCLK1 1 +#define RT5639_PLL1_S_BCLK2 2 +#define RT5639_PLL1_S_BCLK3 3 + + +enum { + RT5639_AIF1, + RT5639_AIF2, + RT5639_AIF3, + RT5639_AIFS, +}; + +enum { + RT5639_U_IF1 = 0x1, + RT5639_U_IF2 = 0x2, + RT5639_U_IF3 = 0x4, +}; + +enum { + RT5639_IF_123, + RT5639_IF_132, + RT5639_IF_312, + RT5639_IF_321, + RT5639_IF_231, + RT5639_IF_213, + RT5639_IF_113, + RT5639_IF_223, + RT5639_IF_ALL, +}; + +enum { + RT5639_DMIC_DIS, + RT5639_DMIC1, + RT5639_DMIC2, +}; + +struct rt5639_pll_code { + bool m_bp; /* Indicates bypass m code or not. */ + int m_code; + int n_code; + int k_code; +}; + +struct rt5639_priv { + struct snd_soc_codec *codec; + struct rt5639_platform_data pdata; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck[RT5639_AIFS]; + int bclk[RT5639_AIFS]; + int master[RT5639_AIFS]; + + struct rt5639_pll_code pll_code; + int pll_src; + int pll_in; + int pll_out; + + bool hp_mute; +}; + +#endif -- 1.8.1.1.439.g50a6b54

5 9

[alsa-devel] [PATCH] ALSA: hda - add headset mic detect quirks for three Dell laptops
by Hui Wang 26 Mar '14

26 Mar '14

When we plug a 3-ring headset on the Dell machines (VID: 0x10ec0255, SID: 0x10280632; VID: 0x10ec0293, SID: 0x1028062c; VID: 0x10ec0293, SID: 0x1028062e), the headset mic can't be detected, after apply this patch, the headset mic can work well. BugLink: https://bugs.launchpad.net/bugs/1297581 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b0e25e5..af5f903 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4586,6 +4586,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0629, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x062c, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x062e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0632, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK), SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.8.1.2

2 1

[alsa-devel] [Audio Driver Low power state]
by Gomathi Kumar 26 Mar '14

26 Mar '14

Hi all, I am new to kernel work and am trying to understand the low power state of audio driver. I am using ALC262 codec, hda_intel and 3.10 kernel code. In this codec, d3_stop_clk flag is not set. So in the code flow I was able to understand the following. if (!codec->pm_down_notified && !bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) this particular condition in hda_power_work function fails, as power_keep_link_on flag is directly based on d3_stop_clk flag. codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_CLKSTOP); I modified the if condition to not check the power_keep_link_on flag. After this change the audio is consuming lesser power, but I am not able to see it going to suspend/runtime suspend. *Please clarify what is happening when I modify the if condition and how audio device is consuming low power?* *Also please help me understand, the significance of flags EPSS and D3_STOP_CLK and if they can be set by software code. * Thanks, Gomathi

1 0

[alsa-devel] [RFC v2 PATCH] ASoC: wm8904: add CCF support
by Bo Shen 26 Mar '14

26 Mar '14

Enable WM8904 to support common clock framework. - Now it supports use SoC provided clock or external oscillator as MCLK. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Documentation/devicetree/bindings/sound/wm8904.txt | 48 ++++++++++++++++++++++ sound/soc/codecs/wm8904.c | 17 ++++++++ 2 files changed, 65 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8904.txt diff --git a/Documentation/devicetree/bindings/sound/wm8904.txt b/Documentation/devicetree/bindings/sound/wm8904.txt new file mode 100644 index 0000000..bcf4aa0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8904.txt @@ -0,0 +1,48 @@ +WM8904 audio CODEC + +This device supports I2C only. + +Required properties: + - compatible: "wlf,wm8904" + - reg: the I2C address of the device. + - clocks and clock-names: reference to + <Documentation/devicetree/bindings/clock/clock-bindings.txt> + +Pins on the device (for linking into audio routes): + + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * MICBIAS + +Examples: + +1. Using SoC provided clock as mclk: +codec: wm8904@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; + clocks = <&pck0>; + clock-names = "mclk"; +}; + +2. Using external oscillator as mclk: +codec: wm8904@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; + + wm8904_osc { + compatible = "fixed-clock"; + #clock-cells = <0>; + #clock-frequency = <12000000>; + } + + clocks = <&wm8904_osc>; + clock-names = "mclk"; +}; diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 49c35c3..4db6a26 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -11,6 +11,7 @@ * published by the Free Software Foundation. */ +#include <linux/clk.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -49,6 +50,7 @@ static const char *wm8904_supply_names[WM8904_NUM_SUPPLIES] = { /* codec private data */ struct wm8904_priv { struct regmap *regmap; + struct clk *mclk; enum wm8904_type devtype; @@ -1828,6 +1830,9 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + if (IS_ENABLED(CONFIG_COMMON_CLK)) + if (wm8904->mclk) + clk_prepare_enable(wm8904->mclk); break; case SND_SOC_BIAS_PREPARE: @@ -1894,6 +1899,9 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); + if (IS_ENABLED(CONFIG_COMMON_CLK)) + if (wm8904->mclk) + clk_disable_unprepare(wm8904->mclk); break; } codec->dapm.bias_level = level; @@ -2139,6 +2147,15 @@ static int wm8904_i2c_probe(struct i2c_client *i2c, return ret; } + if (IS_ENABLED(CONFIG_COMMON_CLK)) { + wm8904->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8904->mclk)) { + dev_err(&i2c->dev, "Failed to get MCLK\n"); + ret = PTR_ERR(wm8904->mclk); + goto err_enable; + } + } + ret = regmap_read(wm8904->regmap, WM8904_SW_RESET_AND_ID, &val); if (ret < 0) { dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret); -- 1.8.5.2

4 4

[alsa-devel] [PATCH v3] ASOC: tegra: move AC97 clock handling to the machine driver
by Lucas Stach 25 Mar '14

25 Mar '14

On Tegra the convention is to have a single machine driver that's controlling the whole audio subsystem. This was introduced after the AC97 driver was merged and ever since AC97 has been broken. Move the clock handling to the machine driver, to be in line with the other Tegra drivers and give the machine driver full control over the single Tegra audio PLL. Signed-off-by: Lucas Stach <dev(a)lynxeye.de> Acked-by: Stephen Warren <swarren(a)nvidia.com> -- v2: - remove redundant clock rate setting - defer removal of err_clk_put label to avoid conflict with Stephens dma channel cleanup v3: - rebased to 3.14-rc1 - no functional change - reword commit message CC: Liam Girdwood <lgirdwood(a)gmail.com> CC: Mark Brown <broonie(a)kernel.org> CC: Thierry Reding <thierry.reding(a)gmail.com> CC: Stephen Warren <swarren(a)wwwdotorg.org> CC: Stefan Agner <stefan(a)agner.ch> CC: alsa-devel(a)alsa-project.org --- sound/soc/tegra/tegra20_ac97.c | 15 +-------------- sound/soc/tegra/tegra20_ac97.h | 1 - sound/soc/tegra/tegra_wm9712.c | 17 ++++++++++++++++- 3 files changed, 17 insertions(+), 16 deletions(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index cf5e1cf..29f8832 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -37,7 +37,6 @@ #include <sound/soc.h> #include <sound/dmaengine_pcm.h> -#include "tegra_asoc_utils.h" #include "tegra20_ac97.h" #define DRV_NAME "tegra20-ac97" @@ -376,18 +375,10 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ac97->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; ac97->playback_dma_data.maxburst = 4; - ret = tegra_asoc_utils_init(&ac97->util_data, &pdev->dev); - if (ret) - goto err_clk_put; - - ret = tegra_asoc_utils_set_ac97_rate(&ac97->util_data); - if (ret) - goto err_asoc_utils_fini; - ret = clk_prepare_enable(ac97->clk_ac97); if (ret) { dev_err(&pdev->dev, "clk_enable failed: %d\n", ret); - goto err_asoc_utils_fini; + goto err; } ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops); @@ -419,8 +410,6 @@ err_unregister_component: snd_soc_unregister_component(&pdev->dev); err_clk_disable_unprepare: clk_disable_unprepare(ac97->clk_ac97); -err_asoc_utils_fini: - tegra_asoc_utils_fini(&ac97->util_data); err_clk_put: err: snd_soc_set_ac97_ops(NULL); @@ -434,8 +423,6 @@ static int tegra20_ac97_platform_remove(struct platform_device *pdev) tegra_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); - tegra_asoc_utils_fini(&ac97->util_data); - clk_disable_unprepare(ac97->clk_ac97); snd_soc_set_ac97_ops(NULL); diff --git a/sound/soc/tegra/tegra20_ac97.h b/sound/soc/tegra/tegra20_ac97.h index 4acb3aa..0a39d82 100644 --- a/sound/soc/tegra/tegra20_ac97.h +++ b/sound/soc/tegra/tegra20_ac97.h @@ -90,6 +90,5 @@ struct tegra20_ac97 { struct regmap *regmap; int reset_gpio; int sync_gpio; - struct tegra_asoc_utils_data util_data; }; #endif /* __TEGRA20_AC97_H__ */ diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 45b5789..25a7f82 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -29,10 +29,13 @@ #include <sound/pcm_params.h> #include <sound/soc.h> +#include "tegra_asoc_utils.h" + #define DRV_NAME "tegra-snd-wm9712" struct tegra_wm9712 { struct platform_device *codec; + struct tegra_asoc_utils_data util_data; }; static const struct snd_soc_dapm_widget tegra_wm9712_dapm_widgets[] = { @@ -118,15 +121,25 @@ static int tegra_wm9712_driver_probe(struct platform_device *pdev) tegra_wm9712_dai.platform_of_node = tegra_wm9712_dai.cpu_of_node; + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto codec_unregister; + + ret = tegra_asoc_utils_set_ac97_rate(&machine->util_data); + if (ret) + goto asoc_utils_fini; + ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto codec_unregister; + goto asoc_utils_fini; } return 0; +asoc_utils_fini: + tegra_asoc_utils_fini(&machine->util_data); codec_unregister: platform_device_del(machine->codec); codec_put: @@ -141,6 +154,8 @@ static int tegra_wm9712_driver_remove(struct platform_device *pdev) snd_soc_unregister_card(card); + tegra_asoc_utils_fini(&machine->util_data); + platform_device_unregister(machine->codec); return 0; -- 1.8.5.3

4 4

[alsa-devel] [PARTIAL PATCH 0/3] ASoC: simple-card: multi DAI links extension
by Jean-Francois Moine 25 Mar '14

25 Mar '14

This patch series extends the simple card driver to handle many DAI links as this exists in the Cubox audio subsystem. ** this is a partial patch ** Jyri, is this OK as a base for you? - partial patch - remove 'simple-card: Simplify code' which is applied (Mark Brown) - add a DAI link container in the DT - v3 - remove 'Fix the reference count of device nodes' which is applied (Mark Brown) - new patch 'Simplify code' - dynamically allocate and use properties for all DAI links (Jyri Sarha and Li Xiubo) - v2 - change subject/comment about device node reference count (Mark Brown) - use a null size array instead of an implicit area for the DAI links (Li Xiubo) - update the reference count of the device node at end of probe Jean-Francois Moine (3): ASoC: simple-card: dynamically allocate the DAI link and properties ASoC: simple-card: Add DT documentation for multi-DAI links ASoC: simple-card: Handle many DAI links .../devicetree/bindings/sound/simple-card.txt | 42 +++- sound/soc/generic/simple-card.c | 220 +++++++++++++-------- 2 files changed, 183 insertions(+), 79 deletions(-) -- 1.9.1

3 5

[alsa-devel] [RFC PATCH] alsa-lib: Call LADSPA activate after all ports are setup
by Matthias Larisch 25 Mar '14

25 Mar '14

Hey there, I notices mainly swh-plugins (like butterworth_1902) to crash on seek with mplayer or mpd. This is due to their behaviour on relying on the activate call after the control inputs are set. This is not right by LADSPA API description but I see no point in NOT calling activate as late as possible in alsa-lib. Comments? Regards, Matthias From f7f5d920c845fc54ffef208f68f568b06e8fadd1 Mon Sep 17 00:00:00 2001 From: Matthias Larisch <mail(a)matthias-larisch.de> Date: Mon, 17 Mar 2014 19:37:01 +0100 Subject: [PATCH] call ladspa activate as late as possible to work around wrong plugin behaviour --- src/pcm/pcm_ladspa.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c index 9ce5242..7d1e3df 100644 --- a/src/pcm/pcm_ladspa.c +++ b/src/pcm/pcm_ladspa.c @@ -645,8 +645,6 @@ static int snd_pcm_ladspa_allocate_instances(snd_pcm_t *pcm, snd_pcm_ladspa_t *l return -EINVAL; } list_add_tail(&instance->list, &plugin->instances); - if (plugin->desc->activate) - plugin->desc->activate(instance->handle); if (plugin->policy == SND_PCM_LADSPA_POLICY_DUPLICATE) { err = snd_pcm_ladspa_connect_plugin_duplicate(plugin, &plugin->input, &plugin->output, instance, idx); if (err < 0) { @@ -664,6 +662,8 @@ static int snd_pcm_ladspa_allocate_instances(snd_pcm_t *pcm, snd_pcm_ladspa_t *l assert(err >= 0); err = snd_pcm_ladspa_connect_controls(plugin, &plugin->output, instance); assert(err >= 0); + if (plugin->desc->activate) + plugin->desc->activate(instance->handle); } err = snd_pcm_ladspa_check_connect(plugin, &plugin->input, &instance->input, depth); if (err < 0) -- 1.9.0

2 1

[alsa-devel] [PATCH 0/4] Add fsl_asoc_xlate_tdm_slot_mask()
by Xiubo Li 25 Mar '14

25 Mar '14

Xiubo Li (4): ASoC: core: remove the 'of_' prefix of of_xlate_tdm_slot_mask. ASoC: fsl-utils: Add fsl_asoc_xlate_tdm_slot_mask() support. ASoC: fsl-esai: Add .xlate_tdm_slot_mask() support. ASoC: imx-ssi: Add .xlate_tdm_slot_mask() support. include/sound/soc-dai.h | 2 +- sound/soc/fsl/Kconfig | 2 ++ sound/soc/fsl/fsl_esai.c | 2 ++ sound/soc/fsl/fsl_utils.c | 27 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_utils.h | 4 +++- sound/soc/fsl/imx-ssi.c | 2 ++ sound/soc/soc-core.c | 10 +++++----- 7 files changed, 42 insertions(+), 7 deletions(-) -- 1.8.4

3 7

[alsa-devel] [PATCH 0/19] ASoC: rsnd: new module path method and DT support
by Kuninori Morimoto 25 Mar '14

25 Mar '14

Hi Mark These are Renesas R-Car sound driver patches. #1 - #7 are clean up patches. #8 - #18 adds new module path method #19 adds DT support Now, Mark's ALSA branch doesn't include SH-ARM platform branch. Thus, these patches are keeping compatible for it at this point. Compatible code will be removed after merge. Kuninori Morimoto (19): 1) ASoC: rsnd: move priv member settings to upper side 2) ASoC: rsnd: move rsnd_mod_call() macro 3) ASoC: rsnd: remove verbose function parameter 4) ASoC: rsnd: remove verbose debug message from scu/ssi 5) ASoC: rsnd: unify rdai naming 6) ASoC: rsnd: add .set_sysclk on snd_soc_dai_ops 7) ASoC: rsnd: tidyup RSND_SSI_xxx flags 8) ASoC: rsnd: run rsnd_path_init() when probe() timing 9) ASoC: rsnd: use mod array instead of list on rdai 10) ASoC: rsnd: get ssi/scu from rsnd_dai_stream 11) ASoC: rsnd: add struct rsnd_dai_platform_info 12) ASoC: rsnd: use function pointer for each probe 13) ASoC: rsnd: Get correct SCU ID 14) ASoC: rsnd: add rsnd_scu_enable_ssi_irq() 15) ASoC: rsnd: call rsnd_scu_ssi_mode_init() from SSI 16) ASoC: rsnd: add probe/remove callback on rsnd_mod_ops 17) ASoC: rsnd: use mod probe method on SCU 18) ASoC: rsnd: use mod probe method on SSI 19) ASoC: rsnd: add DeviceTree support .../devicetree/bindings/sound/renesas,rsnd.txt | 101 ++++++ include/sound/rcar_snd.h | 21 +- sound/soc/sh/rcar/adg.c | 2 +- sound/soc/sh/rcar/core.c | 380 ++++++++++++++++---- sound/soc/sh/rcar/gen.c | 78 +--- sound/soc/sh/rcar/rsnd.h | 86 +++-- sound/soc/sh/rcar/scu.c | 188 ++++++---- sound/soc/sh/rcar/ssi.c | 191 +++++++--- 8 files changed, 759 insertions(+), 288 deletions(-)

3 67

[alsa-devel] [PATCH v2 00/12] Cleanup card DAPM and control setup
by Lars-Peter Clausen 25 Mar '14

25 Mar '14

This patch series converts all remaining machine drivers that still register their controls and DAPM elements with the CODEC to registering them with the card. The reason for doing this is to be able to remove the hack in the ASoC core that un-sets the name_prefix of a CODEC before calling the init callback for a DAI link. This comes from a time where it was not possible to register controls or DAPM elements at a card level. And removing this hack makes going forward with the componentisation easier and it also looks like the DAI multi-CODEC patch could benefit from this as well. The series depends on topic/pxa and topic/dapm - Lars Lars-Peter Clausen (12): ASoC: ams-delta: Fix compile error ASoC: ams-delta: Convert to table based DAPM and control setup ASoC: mfld_machine: Convert to table based DAPM and control setup ASoC: omap-abe-twl6040: Register machine level DMIC DAPM routes with the card ASoC: omap: rx51: Convert to table based control and DAPM setup ASoC: omap3pandora: Convert to table based DAPM setup ASoC: davinci-evm: Register machine level DAPM elements with the card ASoC: snappercl15: Convert to table based DAPM setup ASoC: s6105-ipcam: Convert to table based DAPM setup ASoC: pxa: magician: Convert to table based DAPM and control setup ASoC: pxa: tosa: Convert to table based DAPM and control setup ASoC: Remove name_prefix un-set during DAI link init hack sound/soc/cirrus/snappercl15.c | 18 +++-------- sound/soc/davinci/davinci-evm.c | 22 +++++-------- sound/soc/intel/mfld_machine.c | 49 ++++++++++++----------------- sound/soc/omap/ams-delta.c | 66 +++++++++++++-------------------------- sound/soc/omap/omap-abe-twl6040.c | 3 +- sound/soc/omap/omap3pandora.c | 31 ++++++------------ sound/soc/omap/rx51.c | 47 +++++----------------------- sound/soc/pxa/magician.c | 34 +++++++------------- sound/soc/pxa/tosa.c | 35 ++++++++------------- sound/soc/s6000/s6105-ipcam.c | 28 ++++++----------- sound/soc/soc-core.c | 7 +---- 11 files changed, 108 insertions(+), 232 deletions(-) -- 1.8.0

3 25

[alsa-devel] [PATCH v3 0/3] ASoC: davinci: perparation for eDMA dmaengine PCM
by Peter Ujfalusi 25 Mar '14

25 Mar '14

Hi, Changes since v2: - Constraint only placed for the period size [1] - Patch for davinci-pcm.h to add empty function in case of !DAVINCI platforms When the edma arch and dmaengine patches has been taken only one patch is needed to switch AM335x/AM437x to use the edma-pcm. [1]. I was not able to reporoduce the issue I have seen with mplayer anymore. As it was thought constraint for the period size is enough and there's no need to place the same constraint on buffer size. Regards, Peter --- Peter Ujfalusi (3): ASoC: davinci-mcasp: Constraint on the period and buffer size based on FIFO usage ASoC: davinci-mcasp: Assign the dma_data earlier in dai_probe callback ASoC: davinci-pcm: Add empty functions for !CONFIG_SND_DAVINCI_SOC builds sound/soc/davinci/davinci-mcasp.c | 33 +++++++++++++++++++++++++++++---- sound/soc/davinci/davinci-pcm.h | 10 ++++++++++ 2 files changed, 39 insertions(+), 4 deletions(-) -- 1.9.0

3 13

[alsa-devel] [PATCH] ALSA: hda - Inform the unexpectedly ignored pins by auto-parser
by Takashi Iwai 25 Mar '14

25 Mar '14

The auto-parser may ignore some pins that could be valid when they don't match with the assumption or if there are way too many pins assigned to the same output type. So far, such a pin has been silently ignored, but it's better to leave a message, which would help for debugging and understanding the problem. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_auto_parser.c | 30 +++++++++++++++++++++++++----- 1 file changed, 25 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 8de7cfa460b9..90d2fda6c8f9 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -227,10 +227,18 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, continue; if (!assoc_line_out) assoc_line_out = assoc; - else if (assoc_line_out != assoc) + else if (assoc_line_out != assoc) { + codec_info(codec, + "ignore pin 0x%x with mismatching assoc# 0x%x vs 0x%x\n", + nid, assoc, assoc_line_out); continue; - if (cfg->line_outs >= ARRAY_SIZE(cfg->line_out_pins)) + } + if (cfg->line_outs >= ARRAY_SIZE(cfg->line_out_pins)) { + codec_info(codec, + "ignore pin 0x%x, too many assigned pins\n", + nid); continue; + } line_out[cfg->line_outs].pin = nid; line_out[cfg->line_outs].seq = seq; cfg->line_outs++; @@ -238,8 +246,12 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, case AC_JACK_SPEAKER: seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); - if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins)) + if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins)) { + codec_info(codec, + "ignore pin 0x%x, too many assigned pins\n", + nid); continue; + } speaker_out[cfg->speaker_outs].pin = nid; speaker_out[cfg->speaker_outs].seq = (assoc << 4) | seq; cfg->speaker_outs++; @@ -247,8 +259,12 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, case AC_JACK_HP_OUT: seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); - if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins)) + if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins)) { + codec_info(codec, + "ignore pin 0x%x, too many assigned pins\n", + nid); continue; + } hp_out[cfg->hp_outs].pin = nid; hp_out[cfg->hp_outs].seq = (assoc << 4) | seq; cfg->hp_outs++; @@ -267,8 +283,12 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, break; case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: - if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins)) + if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins)) { + codec_info(codec, + "ignore pin 0x%x, too many assigned pins\n", + nid); continue; + } cfg->dig_out_pins[cfg->dig_outs] = nid; cfg->dig_out_type[cfg->dig_outs] = (loc == AC_JACK_LOC_HDMI) ? -- 1.9.1

1 0

[alsa-devel] [RFC PATCH] ASoC: wm8904: add CCF support
by Bo Shen 25 Mar '14

25 Mar '14

Enable WM8904 to support common clock framework. And also supports different clk configuration. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Documentation/devicetree/bindings/sound/wm8904.txt | 57 ++++++++++++++++++++++ sound/soc/codecs/wm8904.c | 39 +++++++++++++++ 2 files changed, 96 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8904.txt diff --git a/Documentation/devicetree/bindings/sound/wm8904.txt b/Documentation/devicetree/bindings/sound/wm8904.txt new file mode 100644 index 0000000..039374d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8904.txt @@ -0,0 +1,57 @@ +WM8904 audio CODEC + +This device supports I2C only. + +Required properties: + - compatible: "wlf,wm8904" + - reg: the I2C address of the device. + + if work with CCF, the following properties as required: + - wlf,sysclk-from-mclk: set the sys clock is driven from mclk, + - wlf,mclk-use-xtal: if the mclk is generated by crystal. + if without this property, the mclk is generated from SOC. + - clocks: which clock generated clock to mclk. + - clock-names: the clock names use for retrieve. + - wlf,mclk-freq: mclk's frequency + + +Pins on the device (for linking into audio routes): + + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * MICBIAS + +Examples: + +1. General usage: +codec: wm8904@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; +}; + +2. Working with CCF supports and using crystall provide mclk: +codec: wm8904@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; + wlf,sysclk-from-mclk; + wlf,mclk-use-xtal; + wlf,mclk-freq = <12000000>; +}; + +3. Working with CCF supports and using SoC provide mclk: +codec: wm8904@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; + wlf,sysclk-from-mclk; + clocks = <&pck0>; + clock-names = "mclk"; + wlf,mclk-freq = <32768>; +}; diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 49c35c3..99e3e90 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -11,6 +11,7 @@ * published by the Free Software Foundation. */ +#include <linux/clk.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -49,6 +50,10 @@ static const char *wm8904_supply_names[WM8904_NUM_SUPPLIES] = { /* codec private data */ struct wm8904_priv { struct regmap *regmap; + struct clk *mclk; + bool sysclk_from_mclk; + bool mclk_use_xtal; + u32 mclk_freq; enum wm8904_type devtype; @@ -1828,6 +1833,9 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + if (IS_ENABLED(CONFIG_COMMON_CLK)) + if (!wm8904->mclk_use_xtal) + clk_prepare_enable(wm8904->mclk); break; case SND_SOC_BIAS_PREPARE: @@ -1894,6 +1902,9 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); + if (IS_ENABLED(CONFIG_COMMON_CLK)) + if (!wm8904->mclk_use_xtal) + clk_disable_unprepare(wm8904->mclk); break; } codec->dapm.bias_level = level; @@ -2139,6 +2150,34 @@ static int wm8904_i2c_probe(struct i2c_client *i2c, return ret; } + if (IS_ENABLED(CONFIG_COMMON_CLK)) { + struct device_node *np = i2c->dev.of_node; + wm8904->sysclk_from_mclk = + of_property_read_bool(np, "wlf,sysclk-from-mclk"); + if (wm8904->sysclk_from_mclk) { + wm8904->mclk_use_xtal = + of_property_read_bool(np, "wlf,mclk-use-xtal"); + + ret = of_property_read_u32(np, "wlf,mclk-freq", + &wm8904->mclk_freq); + if (ret) { + dev_err(&i2c->dev, "Failed to read mclk freq"); + goto err_enable; + } + + if (!wm8904->mclk_use_xtal) { + wm8904->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8904->mclk)) { + dev_err(&i2c->dev, "Failed to get MCLK\n"); + ret = PTR_ERR(wm8904->mclk); + goto err_enable; + } + + clk_set_rate(wm8904->mclk, wm8904->mclk_freq); + } + } + } + ret = regmap_read(wm8904->regmap, WM8904_SW_RESET_AND_ID, &val); if (ret < 0) { dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret); -- 1.8.5.2

4 8

[alsa-devel] [PATCH] ASoC: rcar: bugfix: it cares about the non-src case
by Kuninori Morimoto 24 Mar '14

24 Mar '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> src might not exist. kernel will be hung-up without this patch in such case. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- >> Mark This is for mark/topic/rcar branch sound/soc/sh/rcar/src.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index eee75eb..6232b7d 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -182,16 +182,20 @@ unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv, struct rsnd_dai_stream *io, struct snd_pcm_runtime *runtime) { + struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io); struct rsnd_src *src; - unsigned int rate; + unsigned int rate = 0; - src = rsnd_mod_to_src(rsnd_io_to_mod_src(io)); + if (src_mod) { + src = rsnd_mod_to_src(src_mod); + + /* + * return convert rate if SRC is used, + * otherwise, return runtime->rate as usual + */ + rate = rsnd_src_convert_rate(src); + } - /* - * return convert rate if SRC is used, - * otherwise, return runtime->rate as usual - */ - rate = rsnd_src_convert_rate(src); if (!rate) rate = runtime->rate; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ARM: bockw: fixup SND_SOC_DAIFMT_CBx_CFx flags
by Kuninori Morimoto 24 Mar '14

24 Mar '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> SND_SOC_DAIFMT_CBx_CFx means "codec" side master/slave mode. Then, rcar will be master mode if it was SND_SOC_DAIFMT_CBS_CFS. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- >> Mark, Simon I'm so sorry. current Mark's fix/rcar branch has SND_SOC_DAIFMT_CBx_CFx fixup patches, but it didn't care about bockw board. This patch is for it. This Subject is using "ARM: bockw:" prefix, but, should it be "ASoC: rcar" ? arch/arm/mach-shmobile/board-bockw.c | 16 +++++----------- 1 file changed, 5 insertions(+), 11 deletions(-) diff --git a/arch/arm/mach-shmobile/board-bockw.c b/arch/arm/mach-shmobile/board-bockw.c index b5867b7..f444be2 100644 --- a/arch/arm/mach-shmobile/board-bockw.c +++ b/arch/arm/mach-shmobile/board-bockw.c @@ -467,14 +467,12 @@ static struct asoc_simple_card_info rsnd_card_info[] = { .card = "SSI56-AK4643", .codec = "ak4642-codec.0-0012", .platform = "rcar_sound", - .daifmt = SND_SOC_DAIFMT_LEFT_J, + .daifmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM, .cpu_dai = { .name = "rsnd-dai.0", - .fmt = SND_SOC_DAIFMT_CBS_CFS, }, .codec_dai = { .name = "ak4642-hifi", - .fmt = SND_SOC_DAIFMT_CBM_CFM, .sysclk = 11289600, }, }, @@ -484,10 +482,9 @@ static struct asoc_simple_card_info rsnd_card_info[] = { .card = "SSI3-AK4554(playback)", .codec = "ak4554-adc-dac.0", .platform = "rcar_sound", + .daifmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_RIGHT_J, .cpu_dai = { .name = "rsnd-dai.1", - .fmt = SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_RIGHT_J, }, .codec_dai = { .name = "ak4554-hifi", @@ -499,10 +496,9 @@ static struct asoc_simple_card_info rsnd_card_info[] = { .card = "SSI4-AK4554(capture)", .codec = "ak4554-adc-dac.0", .platform = "rcar_sound", + .daifmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_LEFT_J, .cpu_dai = { .name = "rsnd-dai.2", - .fmt = SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_LEFT_J, }, .codec_dai = { .name = "ak4554-hifi", @@ -514,10 +510,9 @@ static struct asoc_simple_card_info rsnd_card_info[] = { .card = "SSI7-AK4554(playback)", .codec = "ak4554-adc-dac.1", .platform = "rcar_sound", + .daifmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_RIGHT_J, .cpu_dai = { .name = "rsnd-dai.3", - .fmt = SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_RIGHT_J, }, .codec_dai = { .name = "ak4554-hifi", @@ -529,10 +524,9 @@ static struct asoc_simple_card_info rsnd_card_info[] = { .card = "SSI8-AK4554(capture)", .codec = "ak4554-adc-dac.1", .platform = "rcar_sound", + .daifmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_LEFT_J, .cpu_dai = { .name = "rsnd-dai.4", - .fmt = SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_LEFT_J, }, .codec_dai = { .name = "ak4554-hifi", -- 1.7.9.5

2 2

[alsa-devel] [PATCH RFC 0/2] Move dai-link level properties away from dai subnodes
by Jyri Sarha 24 Mar '14

24 Mar '14

This patch is implemented on top of late patches from Jean-Francois Moine [1]. The patches implement the main part of the simple-card changes discussed in alsa-devel mailing list. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074592.html Best regards, Jyri Jyri Sarha (2): ASoC: core: Update snd_soc_of_parse_daifmt() interface ASoC: simple-card: Move dai-link level properties away from dai subnodes .../devicetree/bindings/sound/simple-card.txt | 91 ++++---- include/sound/soc.h | 4 +- sound/soc/generic/simple-card.c | 237 ++++++++++++-------- sound/soc/soc-core.c | 8 +- 4 files changed, 201 insertions(+), 139 deletions(-) -- 1.7.9.5

3 6

[alsa-devel] [PATCH RFC v2 0/2] Move dai-link level properties away from dai subnodes
by Jyri Sarha 24 Mar '14

24 Mar '14

This patch is implemented on top of late patches from Jean-Francois Moine [1]. These patches implement the main part of the simple-card changes discussed in alsa-devel mailing list [2]. Since RFC version of the patches: - ASoC: core: Update snd_soc_of_parse_daifmt() interface - Make minimal changes to simple-card for it to work with the updated snd_soc_of_parse_daifmt(). (Kuninori Morimoto's comment [3]). - ASoC: simple-card: Move dai-link level properties away from dai subnodes - Fix typo in DT binding document (Jean-Francois Moine's comment [4]). - Remove #define DEBUG - Remove accidental line wrap at simple-audio-card,name property descripition in the DT binding document [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074592.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074388.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074699.html [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074697.html Best regards, Jyri Jyri Sarha (2): ASoC: core: Update snd_soc_of_parse_daifmt() interface ASoC: simple-card: Move dai-link level properties away from dai subnodes .../devicetree/bindings/sound/simple-card.txt | 88 ++++---- include/sound/soc.h | 4 +- sound/soc/generic/simple-card.c | 236 ++++++++++++-------- sound/soc/soc-core.c | 8 +- 4 files changed, 198 insertions(+), 138 deletions(-) -- 1.7.9.5

1 2

[alsa-devel] [hsw dp audio] Muting S/PDIF turns off DP
by Daniel Martin 24 Mar '14

24 Mar '14

Hi, I've a bug here: when playing audio via the speakers of a monitor attached to a DisplayPort and I mute the corresponding S/PDIF (i.e. with alsamixer) then the monitor switches off into standby, unmuting the S/PDIF turns the monitor back on. When muting the S/PDIF while no audio is playing everything is fine. There's no audio daemon running and my ~/.asoundrc contains only: pcm.!default { type hw card 0 device 3 } I've already filed a fdo bug report (drm/intel) for this as I thought it might be related to it: Bug 75729 - [hsw dp audio] Muting S/PDIF turns off DP https://bugs.freedesktop.org/show_bug.cgi?id=75729 Attached to this bug is a `alsa-info.log` too. As stated in the bug report it's not a showstopper for me, but obviously something seems to go wrong and maybe someone has an idea. Thanks, Daniel Martin

2 1

[alsa-devel] [GIT PULL] More ASoC updates for v3.15
by Mark Brown 24 Mar '14

24 Mar '14

The following changes since commit 9296f4da3bafa23d8b9abc5cd271a66ea8f90cd2: ASoC: tlv320aic31xx: Staticise non-exported symbols (2014-03-13 17:44:22 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.15-3 for you to fetch changes up to e090d5b6ad20056ec0ef58727e3ae95fd82be090: Merge remote-tracking branches 'asoc/topic/simple' and 'asoc/topic/sirf' into asoc-next (2014-03-23 14:00:59 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.15 A few more updates for the merge window: - Fixes for the simple-card DAI format DT mess. - A new driver for Cirrus cs42xx8 devices. - DT support for a couple more devices. - A revert of a previous buggy fix for soc-pcm, plus a few more fixes and cleanups. ---------------------------------------------------------------- Al Viro (3): ocfs2 syncs the wrong range... sockfd_lookup_light(): switch to fdget^W^Waway from fget_light get rid of fget_light() Ales Novak (1): [SCSI] storvsc: NULL pointer dereference fix Alex Deucher (4): drm/radeon: fix runpm disabling on non-PX harder drm/radeon/cik: properly set sdma ring status on disable drm/radeon/cik: stop the sdma engines in the enable() function drm/radeon/cik: properly set compute ring status on disable Alexander Aring (1): at86rf230: fix lockdep splats Alexei Starovoitov (1): x86: bpf_jit: support negative offsets Amir Vadai (2): net/mlx4_core: Fix memory access error in mlx4_QUERY_DEV_CAP_wrapper() net/mlx4_core: mlx4_init_slave() shouldn't access comm channel before PF is ready Amitkumar Karwar (2): mwifiex: copy AP's HT capability info correctly mwifiex: save and copy AP's VHT capability info correctly Andrew Lutomirski (1): net: Improve SO_TIMESTAMPING documentation and fix a minor code bug Andrew Morton (1): revert "kallsyms: fix absolute addresses for kASLR" Annie Li (1): Xen-netback: Fix issue caused by using gso_type wrongly Anton Blanchard (2): net: unix socket code abuses csum_partial ibmveth: Fix endian issues with MAC addresses Anton Nayshtut (1): ipv6: Fix exthdrs offload registration. Arnd Bergmann (1): vmxnet3: fix building without CONFIG_PCI_MSI Artem Fetishev (1): fs/proc/base.c: fix GPF in /proc/$PID/map_files Axel Lin (1): ASoC: tlv320aic31xx: Don't call kfree for memory allocated by devm_kzalloc Ben Hutchings (3): perf trace: Decode architecture-specific signal numbers bna: Replace large udelay() with mdelay() mm/Kconfig: fix URL for zsmalloc benchmark Bjorn Helgaas (3): phy: fix compiler array bounds warning on settings[] PCI: Enable INTx in pci_reenable_device() only when MSI/MSI-X not enabled PCI: Don't check resource_size() in pci_bus_alloc_resource() Brian Austin (3): ASoC: cs42l73: Fix mask bits for SOC_VALUE_ENUM_SINGLE ASoC: cs42l52: Fix mask bits for SOC_VALUE_ENUM_SINGLE ASoC: cs42l51: Fix SOC_DOUBLE_R_SX_TLV shift values for ADC, PCM, and Analog kcontrols Chad Dupuis (1): [SCSI] qla2xxx: Fix multiqueue MSI-X registration. Colin Ian King (1): tools/testing/selftests/ipc/msgque.c: handle msgget failure return correctly Dan Williams (1): [SCSI] isci: fix reset timeout handling Daniel Borkmann (2): net: sctp: fix skb leakage in COOKIE ECHO path of chunk->auth_chunk MAINTAINERS: add networking selftests to NETWORKING Daniel J Blueman (1): x86/amd/numa: Fix northbridge quirk to assign correct NUMA node Dave Airlie (3): Merge tag 'vmwgfx-fixes-3.14-2014-03-13' of git://people.freedesktop.org/~thomash/linux into drm-fixes Merge branch 'drm-fixes-3.14' of git://people.freedesktop.org/~agd5f/linux into drm-fixes Merge tag 'ttm-fixes-3.14-2014-03-12' of git://people.freedesktop.org/~thomash/linux into drm-fixes Dave Jones (2): perf/x86: Fix leak in uncore_type_init failure paths x86: Remove CONFIG_X86_OOSTORE David S. Miller (5): Merge branch 'tipc' Merge branch 'for-davem' of git://git.kernel.org/.../linville/wireless Merge branch 'stmmac' Merge branch 'skb_frags' Merge branch 'mlx4' Don Zickus (1): perf machine: Use map as success in ip__resolve_ams Dylan Reid (1): ASoC: tegra: Use flat regcache Eliad Peller (1): mac80211: consider virtual mon when calculating min_def Emmanuel Grumbach (1): iwlwifi: mvm: don't WARN when statistics are handled late Eric Dumazet (3): pkt_sched: move the sanity test in qdisc_list_add() pkt_sched: fq: do not hold qdisc lock while allocating memory tcp: tcp_release_cb() should release socket ownership Eric W. Biederman (3): audit: Use struct net not pid_t to remember the network namespce to reply in audit: Send replies in the proper network namespace. audit: Update kdoc for audit_send_reply and audit_list_rules_send Erik Hugne (3): tipc: drop subscriber connection id invalidation tipc: fix memory leak during module removal tipc: don't log disabled tasklet handler errors Fernando Luis Vazquez Cao (1): sched/clock: Prevent tracing recursion in sched_clock_cpu() Florian Westphal (1): inet: frag: make sure forced eviction removes all frags Gavin Shan (1): net/mlx4: Support shutdown() interface Geert Uytterhoeven (2): cris: convert ffs from an object-like macro to a function-like macro packet: doc: Spelling s/than/that/ Giridhar Malavali (1): [SCSI] qla2xxx: Poll during initialization for ISP25xx and ISP83xx Giuseppe CAVALLARO (4): stmmac: disable at run-time the EEE if not supported stmmac: fix and better tune the default buffer sizes stmmac: fix chained mode stmmac: dwmac-sti: fix broken STiD127 compatibility Guillaume Nault (2): l2tp: fix manual sequencing (de)activation in L2TPv2 l2tp: fix userspace reception on plain L2TP sockets Haiyang Zhang (1): hyperv: Move state setting for link query Heiner Kallweit (1): ipv6: Avoid unnecessary temporary addresses being generated Heinz Mauelshagen (2): dm cache: fix truncation bug when copying a block to/from >2TB fast device dm cache: fix access beyond end of origin device Ingo Molnar (1): Merge tag 'perf-urgent-for-mingo' of git://git.kernel.org/.../acme/linux into perf/urgent Ivaylo Dimitrov (1): wl1251: use skb_trim to make skb shorter Jean-Francois Moine (2): ASoC: simple-card: Fix the reference count of device nodes ASoC: simple-card: Simplify code Jeff Layton (2): cifs: sanity check length of data to send before sending cifs: mask off top byte in get_rfc1002_length() Jiri Olsa (1): perf symbols: Fix crash in elf_section_by_name Johannes Berg (2): cfg80211: remove racy beacon_interval assignment mac80211: clear sequence/fragment number in QoS-null frames Johannes Weiner (1): mm: fix GFP_THISNODE callers and clarify John W. Linville (3): Merge branch 'for-john' of git://git.kernel.org/.../iwlwifi/iwlwifi-fixes Merge branch 'for-john' of git://git.kernel.org/.../jberg/mac80211 Merge branch 'master' of git://git.kernel.org/.../linville/wireless into for-davem Juri Lelli (1): sched/deadline: Deny unprivileged users to set/change SCHED_DEADLINE policy Jyri Sarha (1): ASoC: tlv320aic31xx: Turn power off only once. Kuninori Morimoto (4): ASoC: fsi: fixup SND_SOC_DAIFMT_CBx_CFx flags ASoC: rcar: fixup SND_SOC_DAIFMT_CBx_CFx flags ASoC: fsi: fixup compile error for simple-card ASoC: rsnd: add DeviceTree support Lars-Peter Clausen (8): ASoC: omap-abe-twl6040: Register machine level DMIC DAPM routes with the card ASoC: davinci-evm: Register machine level DAPM elements with the card ASoC: max98090: Remove unused control_data field ASoC: sirf-audio-codec: Remove snd_soc_codec_set_cache_io() call ASoC: codecs: Replace instances of rtd->codec with dai->codec ASoC: Add helper function to cast component back to CODEC ASoC: mfld_machine: Convert to table based DAPM and control setup ASoC: mfld_machine: Fix compile error Laura Abbott (1): mm/compaction: break out of loop on !PageBuddy in isolate_freepages_block Linus Lüssing (3): bridge: multicast: add sanity check for query source addresses bridge: multicast: add sanity check for general query destination bridge: multicast: enable snooping on general queries only Linus Torvalds (19): vfs: atomic f_pos accesses as per POSIX Merge branch 'for-3.14-fixes' of git://git.kernel.org/.../tj/libata Merge branch 'for-linus' of git://git.kernel.org/.../viro/vfs Merge branch 'akpm' (patches from Andrew Morton) Merge branch 'for-linus' of git://git.kernel.org/.../ebiederm/user-namespace Merge branch 'for-next' of git://git.samba.org/sfrench/cifs-2.6 Merge tag 'sound-3.14-rc7' of git://git.kernel.org/.../tiwai/sound Merge tag 'for-linus' of git://git.kernel.org/pub/scm/virt/kvm/kvm Merge tag 'pci-v3.14-fixes-2' of git://git.kernel.org/.../helgaas/pci Merge git://git.kernel.org/.../davem/net Merge branch 'i2c/for-current' of git://git.kernel.org/.../wsa/linux Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge tag 'dm-3.14-fixes-4' of git://git.kernel.org/.../device-mapper/linux-dm Merge tag 'pm+acpi-3.14-rc7' of git://git.kernel.org/.../rafael/linux-pm Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge tag 'scsi-fixes' of git://git.kernel.org/.../jejb/scsi Merge branch 'perf-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'sched-urgent-for-linus' of git://git.kernel.org/.../tip/tip Linux 3.14-rc7 Lukasz Dorau (1): [SCSI] isci: correct erroneous for_each_isci_host macro Marc Zyngier (1): ARM: KVM: fix non-VGIC compilation Mark Brown (13): ASoC: io: Remove support for ASoC cache in conjunction with regmap ASoC: pcm: Drop incorrect double/extra frees Merge remote-tracking branches 'asoc/fix/cs42l51', 'asoc/fix/cs42l52', 'asoc/fix/cs42l73', 'asoc/fix/rcar', 'asoc/fix/spear' and 'asoc/fix/tegra' into asoc-linus Merge tag 'asoc-v3.15' into asoc-next Merge tag 'asoc-v3.15-2' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/codecs' into asoc-next Merge remote-tracking branch 'asoc/topic/cs42xx8' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branches 'asoc/topic/intel', 'asoc/topic/io' and 'asoc/topic/kirkwood' into asoc-next Merge remote-tracking branches 'asoc/topic/max98090' and 'asoc/topic/omap' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/simple' and 'asoc/topic/sirf' into asoc-next Markos Chandras (1): tools/net/Makefile: Define PACKAGE to fix build problems Matthew Leach (1): net: socket: error on a negative msg_namelen Michael Chan (1): bnx2: Fix shutdown sequence Michael Kerrisk (1): ipc: Fix 2 bugs in msgrcv() MSG_COPY implementation Michael Opdenacker (1): MAINTAINERS: blackfin: add git repository Michael S. Tsirkin (5): skbuff: skb_segment: s/frag/nskb_frag/ skbuff: skb_segment: s/skb_frag/frag/ skbuff: skb_segment: s/skb/head_skb/ skbuff: skb_segment: s/fskb/list_skb/ skbuff: skb_segment: orphan frags before copying Mike Christie (1): [SCSI] be2iscsi: fix bad if expression Neil Horman (1): vmxnet3: fix netpoll race condition Nenghua Cao (1): asoc: soc-core: fix coccinelle warnings Nicolin Chen (2): ASoC: simple-card: overwrite cpu_dai->fmt with codec_dai->fmt ASoC: cs42xx8: Add codec driver support for CS42448/CS42888 Nikolay Aleksandrov (1): net: fix for a race condition in the inet frag code Or Gerlitz (4): net/mlx4_core: Fix wrong dump of the vxlan offloads device capability net/mlx4_en: Handle vxlan steering rules for mac address changes net/mlx4_core: Load the IB driver when the device supports IBoE net/mlx4_en: Deregister multicast vxlan steering rules when going down Oren Givon (1): iwlwifi: fix and add 7265 series HW IDs Pavel Shilovsky (1): CIFS: Fix wrong pos argument of cifs_find_lock_conflict Peter Boström (1): vlan: Set correct source MAC address with TX VLAN offload enabled Peter Ujfalusi (2): ASoC: davinci: Add edma dmaengine platform driver ASoC: davinci-mcasp: Provide correct filter_data for dmaengine for non-DT boot Peter Zijlstra (1): stop_machine: Fix^2 race between stop_two_cpus() and stop_cpus() Radim Krčmář (1): KVM: SVM: fix cr8 intercept window Rafael J. Wysocki (4): cpufreq: Skip current frequency initialization for ->setpolicy drivers ACPI / init: Invoke early ACPI initialization later ACPI / sleep: Add extra checks for HW Reduced ACPI mode sleep states Merge branches 'pnp', 'acpi-init', 'acpi-sleep' and 'pm-cpufreq' Richard Weinberger (1): i2c: Remove usage of orphaned symbol OF_I2C Rob Clark (1): drm/ttm: don't oops if no invalidate_caches() Sabrina Dubroca (1): ipv6: don't set DST_NOCOUNT for remotely added routes Sasha Levin (1): bonding: correctly handle out of range parameters for lp_interval Sergei Antonov (1): hfsplus: add HFSX subfolder count support Soren Brinkmann (2): net: macb: Check DMA mappings for error net: macb: DMA-unmap full rx-buffer Stefan Wahren (1): eth: fec: Fix lost promiscuous mode after reconnecting cable Suresh Siddha (1): x86, fpu: Check tsk_used_math() in kernel_fpu_end() for eager FPU Tejun Heo (1): libata: use wider match for blacklisting Crucial M500 Thomas Hellstrom (2): drm/ttm: Work around performance regression with VM_PFNMAP drm/vmwgfx: Fix a surface reference corner-case in legacy emulation mode Thomas Petazzoni (1): ASoC: add S/PDIF support to Armada 370 DB ASoC driver Tobias Klauser (1): MAINTAINERS: Add tools/net to NETWORKING [GENERAL] Wei Liu (1): xen-netback: use skb_is_gso in xenvif_start_xmit Ying Xue (3): tipc: allow connection shutdown callback to be invoked in advance tipc: fix connection refcount leak tipc: avoid to unnecessary process switch under non-block mode Zhang Rui (1): PNP / ACPI: proper handling of ACPI IO/Memory resource parsing failures dingtianhong (1): bonding: set correct vlan id for alb xmit path hayeswang (2): r8152: disable the ECM mode r8169: fix the incorrect tx descriptor version .gitignore | 3 + Documentation/00-INDEX | 24 +- Documentation/PCI/MSI-HOWTO.txt | 119 +- Documentation/RCU/00-INDEX | 2 + Documentation/arm/00-INDEX | 14 + Documentation/blackfin/00-INDEX | 6 +- Documentation/block/00-INDEX | 2 + 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+- arch/powerpc/mm/hash_utils_64.c | 14 + arch/powerpc/mm/pgtable_64.c | 12 +- arch/powerpc/mm/subpage-prot.c | 2 +- arch/powerpc/perf/core-book3s.c | 5 +- arch/powerpc/perf/power8-pmu.c | 144 ++ arch/powerpc/platforms/cell/ras.c | 3 +- arch/powerpc/platforms/powernv/eeh-ioda.c | 128 +- arch/powerpc/platforms/powernv/eeh-powernv.c | 2 +- arch/powerpc/platforms/powernv/opal-xscom.c | 21 +- arch/powerpc/platforms/powernv/pci-ioda.c | 84 + arch/powerpc/platforms/powernv/pci.c | 230 +-- arch/powerpc/platforms/powernv/pci.h | 6 +- arch/powerpc/platforms/powernv/powernv.h | 8 + arch/powerpc/platforms/powernv/setup.c | 9 + arch/powerpc/platforms/pseries/Kconfig | 1 + arch/powerpc/platforms/pseries/eeh_pseries.c | 2 +- arch/powerpc/platforms/pseries/hotplug-cpu.c | 22 +- arch/powerpc/platforms/pseries/pci.c | 22 +- arch/powerpc/platforms/pseries/setup.c | 3 +- arch/powerpc/sysdev/mpic.c | 38 +- arch/powerpc/xmon/xmon.c | 24 +- arch/s390/appldata/appldata_base.c | 1 + arch/s390/crypto/aes_s390.c | 65 +- arch/s390/crypto/des_s390.c | 95 +- arch/s390/kernel/compat_wrapper.S | 2 +- arch/s390/kernel/head64.S | 7 +- arch/s390/mm/page-states.c | 10 + arch/s390/pci/pci_dma.c | 8 +- arch/sh/boards/mach-ecovec24/setup.c | 4 +- arch/sh/boards/mach-se/7724/setup.c | 4 +- arch/sh/include/cpu-sh2/cpu/cache.h | 2 +- arch/sh/include/cpu-sh2a/cpu/cache.h | 4 +- arch/sh/include/cpu-sh3/cpu/cache.h | 2 +- arch/sh/include/cpu-sh4/cpu/cache.h | 2 +- arch/sh/kernel/cpu/init.c | 4 +- arch/sh/mm/cache-debugfs.c | 2 +- arch/sh/mm/cache-sh2.c | 4 +- arch/sh/mm/cache-sh2a.c | 6 +- arch/sh/mm/cache-sh4.c | 4 +- arch/sh/mm/cache-shx3.c | 4 +- arch/sh/mm/cache.c | 4 +- arch/sparc/Kconfig | 2 +- arch/sparc/mm/srmmu.c | 2 + arch/x86/Kconfig | 7 +- arch/x86/Kconfig.cpu | 4 - arch/x86/Kconfig.debug | 1 + arch/x86/boot/compressed/aslr.c | 9 +- arch/x86/include/asm/amd_nb.h | 2 +- arch/x86/include/asm/barrier.h | 8 +- arch/x86/include/asm/efi.h | 3 + arch/x86/include/asm/io.h | 2 +- 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arch/xtensa/kernel/xtensa_ksyms.c | 2 - arch/xtensa/mm/init.c | 13 +- arch/xtensa/mm/mmu.c | 2 +- arch/xtensa/platforms/xtfpga/setup.c | 7 +- arch/xtensa/variants/fsf/include/variant/tie.h | 9 +- block/blk-core.c | 20 +- block/blk-exec.c | 2 +- block/blk-flush.c | 105 +- block/blk-lib.c | 8 + block/blk-merge.c | 91 +- block/blk-mq-cpu.c | 14 +- block/blk-mq-tag.c | 2 +- block/blk-mq.c | 237 ++- block/blk-mq.h | 5 +- block/blk-sysfs.c | 2 + block/blk-timeout.c | 2 +- block/blk.h | 2 +- drivers/acpi/ac.c | 2 + drivers/acpi/battery.c | 4 +- drivers/acpi/blacklist.c | 58 - drivers/acpi/button.c | 2 + drivers/acpi/container.c | 5 +- drivers/acpi/dock.c | 13 +- drivers/acpi/ec.c | 64 + drivers/acpi/fan.c | 3 + drivers/acpi/pci_irq.c | 1 + drivers/acpi/proc.c | 2 +- drivers/acpi/processor_throttling.c | 69 +- drivers/acpi/resource.c | 10 + drivers/acpi/sbs.c | 4 +- drivers/acpi/scan.c | 6 +- drivers/acpi/sleep.c | 32 +- drivers/acpi/thermal.c | 2 + drivers/acpi/utils.c | 4 - drivers/acpi/video.c | 147 +- drivers/acpi/video_detect.c | 8 - drivers/ata/Kconfig | 1 + drivers/ata/ahci.c | 18 +- drivers/ata/libata-core.c | 3 +- drivers/ata/libata-pmp.c | 7 +- drivers/ata/pata_imx.c | 8 +- drivers/ata/sata_mv.c | 16 +- drivers/ata/sata_sil.c | 1 + drivers/base/component.c | 8 + drivers/base/dma-buf.c | 25 +- drivers/base/firmware_class.c | 1 + drivers/base/regmap/regmap.c | 12 + drivers/block/aoe/aoecmd.c | 4 +- drivers/block/mtip32xx/mtip32xx.h | 2 +- drivers/block/null_blk.c | 97 +- drivers/block/nvme-core.c | 610 +++++-- drivers/block/nvme-scsi.c | 147 ++ drivers/block/virtio_blk.c | 7 +- drivers/block/xen-blkback/blkback.c | 81 +- drivers/block/xen-blkback/common.h | 5 +- drivers/block/xen-blkback/xenbus.c | 14 + drivers/block/xen-blkfront.c | 11 +- drivers/block/zram/zram_drv.c | 2 + drivers/char/Kconfig | 1 + drivers/char/raw.c | 2 +- drivers/char/virtio_console.c | 9 +- drivers/clk/at91/clk-master.c | 2 +- drivers/clk/clk-nomadik.c | 3 + drivers/clk/clk.c | 13 +- drivers/clk/keystone/gate.c | 1 + drivers/clk/mvebu/armada-370.c | 21 +- drivers/clk/mvebu/armada-xp.c | 20 +- drivers/clk/mvebu/dove.c | 19 +- drivers/clk/mvebu/kirkwood.c | 34 +- drivers/clk/shmobile/clk-rcar-gen2.c | 48 +- drivers/clk/tegra/clk-divider.c | 2 +- drivers/clk/tegra/clk-id.h | 4 + drivers/clk/tegra/clk-tegra-periph.c | 10 +- drivers/clk/tegra/clk-tegra-super-gen4.c | 2 +- drivers/clk/tegra/clk-tegra114.c | 8 +- drivers/clk/tegra/clk-tegra124.c | 48 +- drivers/clk/tegra/clk-tegra20.c | 2 + drivers/clocksource/bcm_kona_timer.c | 54 +- drivers/cpufreq/cpufreq.c | 58 +- drivers/cpufreq/intel_pstate.c | 61 +- drivers/cpufreq/powernow-k8.c | 10 +- drivers/crypto/nx/nx-842.c | 29 +- drivers/dma/Kconfig | 1 + drivers/dma/imx-sdma.c | 1 + drivers/dma/ioat/dma.c | 52 +- drivers/dma/ioat/dma.h | 1 + drivers/dma/ioat/dma_v2.c | 11 +- drivers/dma/ioat/dma_v3.c | 3 + drivers/dma/mv_xor.c | 24 +- drivers/dma/ste_dma40.c | 4 +- drivers/edac/edac_mc.c | 13 +- drivers/edac/edac_mc_sysfs.c | 10 +- drivers/edac/edac_module.h | 2 +- drivers/edac/i7300_edac.c | 38 +- drivers/edac/i7core_edac.c | 9 +- drivers/extcon/extcon-arizona.c | 12 - drivers/firewire/core-device.c | 22 +- drivers/firewire/net.c | 6 +- drivers/firewire/ohci.c | 15 +- drivers/firewire/sbp2.c | 17 +- drivers/fmc/fmc-write-eeprom.c | 2 +- drivers/gpio/Kconfig | 1 + drivers/gpio/gpio-bcm-kona.c | 4 +- drivers/gpio/gpio-clps711x.c | 1 + drivers/gpio/gpio-intel-mid.c | 4 +- drivers/gpio/gpio-xtensa.c | 16 + drivers/gpu/drm/armada/armada_drv.c | 10 +- drivers/gpu/drm/ast/ast_fb.c | 2 +- drivers/gpu/drm/bochs/Kconfig | 1 + drivers/gpu/drm/cirrus/cirrus_fbdev.c | 2 +- drivers/gpu/drm/drm_ioctl.c | 12 + drivers/gpu/drm/exynos/Kconfig | 4 +- drivers/gpu/drm/exynos/exynos_drm_drv.c | 14 +- drivers/gpu/drm/exynos/exynos_drm_g2d.c | 2 +- drivers/gpu/drm/exynos/exynos_drm_ipp.c | 3 +- drivers/gpu/drm/exynos/exynos_hdmi.c | 66 +- drivers/gpu/drm/i2c/tda998x_drv.c | 19 +- drivers/gpu/drm/i915/i915_drv.c | 23 +- drivers/gpu/drm/i915/i915_drv.h | 8 + drivers/gpu/drm/i915/i915_gem_stolen.c | 19 +- drivers/gpu/drm/i915/i915_gpu_error.c | 5 +- drivers/gpu/drm/i915/i915_irq.c | 3 +- drivers/gpu/drm/i915/intel_display.c | 22 +- drivers/gpu/drm/i915/intel_dp.c | 31 +- drivers/gpu/drm/i915/intel_hdmi.c | 6 +- drivers/gpu/drm/i915/intel_i2c.c | 7 - drivers/gpu/drm/i915/intel_opregion.c | 9 +- drivers/gpu/drm/i915/intel_panel.c | 4 +- drivers/gpu/drm/i915/intel_pm.c | 6 +- drivers/gpu/drm/i915/intel_ringbuffer.c | 21 + drivers/gpu/drm/i915/intel_ringbuffer.h | 1 + drivers/gpu/drm/mgag200/mgag200_fb.c | 2 +- drivers/gpu/drm/mgag200/mgag200_mode.c | 4 +- drivers/gpu/drm/msm/mdp/mdp4/mdp4_crtc.c | 179 +- drivers/gpu/drm/msm/mdp/mdp4/mdp4_plane.c | 4 +- drivers/gpu/drm/msm/mdp/mdp5/mdp5_crtc.c | 7 +- drivers/gpu/drm/msm/msm_gem.c | 2 +- drivers/gpu/drm/msm/msm_gem_submit.c | 9 +- drivers/gpu/drm/msm/msm_gpu.c | 3 - drivers/gpu/drm/nouveau/Makefile | 1 + drivers/gpu/drm/nouveau/core/engine/device/nv40.c | 10 +- drivers/gpu/drm/nouveau/core/engine/disp/nv50.c | 2 +- drivers/gpu/drm/nouveau/core/engine/fifo/nve0.c | 2 +- drivers/gpu/drm/nouveau/core/engine/graph/nv50.c | 2 +- drivers/gpu/drm/nouveau/core/include/subdev/mc.h | 1 + drivers/gpu/drm/nouveau/core/subdev/bios/base.c | 4 + drivers/gpu/drm/nouveau/core/subdev/fb/nv1a.c | 2 +- drivers/gpu/drm/nouveau/core/subdev/mc/nv04.h | 1 + drivers/gpu/drm/nouveau/core/subdev/mc/nv44.c | 2 +- drivers/gpu/drm/nouveau/core/subdev/mc/nv4c.c | 45 + drivers/gpu/drm/nouveau/nouveau_acpi.c | 26 +- drivers/gpu/drm/nouveau/nouveau_bo.c | 2 +- drivers/gpu/drm/nouveau/nouveau_drm.c | 3 + drivers/gpu/drm/nouveau/nouveau_vga.c | 4 +- drivers/gpu/drm/radeon/atombios_crtc.c | 31 +- drivers/gpu/drm/radeon/atombios_encoders.c | 9 +- drivers/gpu/drm/radeon/btc_dpm.c | 32 + drivers/gpu/drm/radeon/btcd.h | 4 + drivers/gpu/drm/radeon/cik.c | 10 +- drivers/gpu/drm/radeon/cik_sdma.c | 14 +- drivers/gpu/drm/radeon/dce6_afmt.c | 15 +- drivers/gpu/drm/radeon/evergreen.c | 7 +- drivers/gpu/drm/radeon/evergreen_hdmi.c | 26 +- drivers/gpu/drm/radeon/evergreen_smc.h | 2 +- drivers/gpu/drm/radeon/kv_dpm.c | 2 +- drivers/gpu/drm/radeon/ni.c | 3 +- drivers/gpu/drm/radeon/ni_dpm.c | 10 +- drivers/gpu/drm/radeon/r100.c | 2 - drivers/gpu/drm/radeon/r300.c | 2 - drivers/gpu/drm/radeon/r420.c | 2 - drivers/gpu/drm/radeon/r520.c | 2 - drivers/gpu/drm/radeon/r600.c | 7 +- drivers/gpu/drm/radeon/r600_audio.c | 14 +- drivers/gpu/drm/radeon/r600_cs.c | 18 +- drivers/gpu/drm/radeon/r600_hdmi.c | 15 +- drivers/gpu/drm/radeon/radeon.h | 10 +- drivers/gpu/drm/radeon/radeon_asic.c | 2 +- drivers/gpu/drm/radeon/radeon_asic.h | 2 + drivers/gpu/drm/radeon/radeon_atpx_handler.c | 3 +- drivers/gpu/drm/radeon/radeon_device.c | 5 +- drivers/gpu/drm/radeon/radeon_display.c | 2 + drivers/gpu/drm/radeon/radeon_drv.c | 3 +- drivers/gpu/drm/radeon/radeon_kms.c | 16 +- drivers/gpu/drm/radeon/radeon_ring.c | 2 +- drivers/gpu/drm/radeon/radeon_semaphore.c | 19 +- drivers/gpu/drm/radeon/radeon_ttm.c | 5 +- drivers/gpu/drm/radeon/radeon_uvd.c | 2 + drivers/gpu/drm/radeon/reg_srcs/r600 | 1 + drivers/gpu/drm/radeon/rs400.c | 2 - drivers/gpu/drm/radeon/rs600.c | 2 - drivers/gpu/drm/radeon/rs690.c | 2 - drivers/gpu/drm/radeon/rv515.c | 2 - drivers/gpu/drm/radeon/rv770.c | 5 +- drivers/gpu/drm/radeon/rv770_dpm.c | 14 +- drivers/gpu/drm/radeon/si.c | 7 +- drivers/gpu/drm/radeon/si_dpm.c | 5 +- drivers/gpu/drm/radeon/sumo_dpm.c | 2 +- drivers/gpu/drm/radeon/trinity_dpm.c | 3 +- drivers/gpu/drm/radeon/uvd_v2_2.c | 1 - drivers/gpu/drm/tegra/drm.c | 2 +- drivers/gpu/drm/tegra/rgb.c | 11 + drivers/gpu/drm/ttm/ttm_agp_backend.c | 1 + drivers/gpu/drm/ttm/ttm_bo.c | 8 +- drivers/gpu/drm/ttm/ttm_bo_vm.c | 12 +- drivers/gpu/drm/ttm/ttm_object.c | 2 +- drivers/gpu/drm/ttm/ttm_tt.c | 3 + drivers/gpu/drm/vmwgfx/svga3d_reg.h | 153 +- drivers/gpu/drm/vmwgfx/svga3d_surfacedefs.h | 11 +- drivers/gpu/drm/vmwgfx/svga_reg.h | 9 +- drivers/gpu/drm/vmwgfx/vmwgfx_context.c | 141 +- drivers/gpu/drm/vmwgfx/vmwgfx_drv.c | 10 + drivers/gpu/drm/vmwgfx/vmwgfx_drv.h | 38 +- drivers/gpu/drm/vmwgfx/vmwgfx_execbuf.c | 337 +++- drivers/gpu/drm/vmwgfx/vmwgfx_ioctl.c | 96 +- drivers/gpu/drm/vmwgfx/vmwgfx_mob.c | 36 +- drivers/gpu/drm/vmwgfx/vmwgfx_resource.c | 14 +- drivers/gpu/drm/vmwgfx/vmwgfx_shader.c | 469 ++++- drivers/gpu/drm/vmwgfx/vmwgfx_surface.c | 24 +- drivers/gpu/host1x/job.c | 2 +- drivers/hid/hid-apple.c | 3 + drivers/hid/hid-core.c | 3 + drivers/hid/hid-hyperv.c | 11 +- drivers/hid/hid-ids.h | 8 + drivers/hid/hid-input.c | 2 +- drivers/hid/hid-microsoft.c | 4 + drivers/hid/hid-multitouch.c | 5 + drivers/hid/hid-sensor-hub.c | 3 + drivers/hid/i2c-hid/i2c-hid.c | 2 +- drivers/hid/usbhid/hid-quirks.c | 1 + drivers/hv/connection.c | 13 +- drivers/hwmon/da9055-hwmon.c | 4 - drivers/hwmon/max1668.c | 2 +- drivers/hwmon/ntc_thermistor.c | 6 +- drivers/hwmon/pmbus/pmbus_core.c | 68 +- drivers/i2c/busses/Kconfig | 2 +- drivers/i2c/busses/i2c-mv64xxx.c | 33 +- drivers/iio/accel/bma180.c | 16 +- drivers/iio/adc/max1363.c | 2 +- drivers/iio/gyro/Kconfig | 2 +- drivers/iio/gyro/st_gyro.h | 1 - drivers/iio/gyro/st_gyro_core.c | 9 +- drivers/iio/gyro/st_gyro_i2c.c | 1 - drivers/iio/gyro/st_gyro_spi.c | 1 - drivers/iio/imu/adis16400.h | 1 + drivers/iio/imu/adis16400_core.c | 10 +- drivers/iio/light/cm32181.c | 16 +- drivers/iio/light/cm36651.c | 45 +- drivers/iio/light/tsl2563.c | 16 +- drivers/iio/magnetometer/ak8975.c | 16 +- drivers/iio/magnetometer/mag3110.c | 8 +- drivers/infiniband/hw/amso1100/c2.c | 4 +- drivers/infiniband/hw/amso1100/c2_rnic.c | 3 +- drivers/infiniband/hw/cxgb4/cm.c | 1 + drivers/infiniband/hw/mlx4/main.c | 189 ++- drivers/infiniband/hw/mlx5/Kconfig | 2 +- drivers/infiniband/hw/mlx5/main.c | 26 +- drivers/infiniband/hw/mlx5/qp.c | 18 +- drivers/infiniband/hw/mlx5/user.h | 7 + drivers/infiniband/hw/nes/nes.c | 5 +- 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| 6 +- include/uapi/asm-generic/unistd.h | 6 +- include/uapi/drm/drm.h | 2 + include/uapi/drm/vmwgfx_drm.h | 1 + include/uapi/linux/btrfs.h | 1 - include/uapi/linux/in6.h | 23 +- include/uapi/linux/mic_ioctl.h | 2 +- include/uapi/linux/nvme.h | 11 + include/uapi/xen/Kbuild | 2 + include/{ => uapi}/xen/gntalloc.h | 0 include/{ => uapi}/xen/gntdev.h | 0 include/xen/grant_table.h | 8 +- include/xen/interface/io/blkif.h | 34 +- include/xen/interface/xencomm.h | 41 - include/xen/xencomm.h | 77 - init/main.c | 4 +- ipc/mq_sysctl.c | 18 +- ipc/mqueue.c | 6 +- ipc/msg.c | 2 + kernel/audit.c | 31 +- kernel/audit.h | 2 +- kernel/audit_tree.c | 2 +- kernel/audit_watch.c | 2 +- kernel/auditfilter.c | 10 +- kernel/auditsc.c | 2 +- kernel/cgroup.c | 60 +- kernel/cpuset.c | 10 +- kernel/events/core.c | 12 +- kernel/irq/Kconfig | 1 + kernel/irq/devres.c | 45 + kernel/irq/irqdesc.c | 1 + kernel/irq/irqdomain.c | 1 + kernel/irq/manage.c | 3 +- kernel/kmod.c | 2 +- kernel/power/console.c | 1 + kernel/printk/printk.c | 2 - kernel/profile.c | 4 +- kernel/sched/clock.c | 4 +- kernel/sched/core.c | 37 +- kernel/sched/cpudeadline.c | 6 +- kernel/sched/deadline.c | 20 +- kernel/sched/fair.c | 10 +- kernel/sched/rt.c | 8 + kernel/sched/sched.h | 1 - kernel/stop_machine.c | 2 +- kernel/time/jiffies.c | 6 + kernel/time/sched_clock.c | 46 +- kernel/time/tick-broadcast.c | 1 + kernel/trace/ring_buffer.c | 7 + kernel/trace/trace_events.c | 10 + kernel/tracepoint.c | 7 +- kernel/user_namespace.c | 2 +- kernel/workqueue.c | 7 + lib/Kconfig.debug | 2 +- lib/Makefile | 1 + lib/dma-debug.c | 131 +- lib/percpu_ida.c | 7 +- lib/radix-tree.c | 4 +- mm/Kconfig | 4 +- mm/compaction.c | 20 +- mm/filemap.c | 4 +- mm/huge_memory.c | 20 +- mm/ksm.c | 2 +- mm/memcontrol.c | 20 +- mm/memory-failure.c | 8 +- mm/memory.c | 15 +- mm/migrate.c | 11 +- mm/mprotect.c | 25 +- mm/page-writeback.c | 5 +- mm/page_alloc.c | 30 +- mm/slub.c | 38 +- mm/swap.c | 4 +- mm/swap_state.c | 63 +- mm/swapfile.c | 11 +- mm/vmpressure.c | 1 + mm/vmstat.c | 4 +- net/8021q/vlan_dev.c | 3 + net/9p/client.c | 2 +- net/9p/trans_virtio.c | 5 +- net/batman-adv/bat_iv_ogm.c | 30 +- net/batman-adv/hard-interface.c | 22 +- net/batman-adv/originator.c | 36 + net/batman-adv/originator.h | 4 + net/batman-adv/routing.c | 4 +- net/batman-adv/send.c | 9 +- net/batman-adv/translation-table.c | 23 +- net/bluetooth/hidp/core.c | 16 +- net/bluetooth/hidp/hidp.h | 4 + net/bridge/br_device.c | 54 +- net/bridge/br_fdb.c | 137 +- net/bridge/br_if.c | 6 +- net/bridge/br_input.c | 4 +- net/bridge/br_multicast.c | 33 +- net/bridge/br_private.h | 13 +- net/bridge/br_stp_if.c | 2 + net/bridge/br_vlan.c | 27 +- net/caif/caif_dev.c | 1 + net/caif/cfsrvl.c | 1 + net/can/af_can.c | 3 +- net/can/bcm.c | 4 +- net/can/raw.c | 27 +- net/ceph/messenger.c | 8 +- net/ceph/osd_client.c | 84 +- net/core/dev.c | 28 +- net/core/fib_rules.c | 7 + net/core/flow_dissector.c | 20 +- net/core/neighbour.c | 8 +- net/core/netpoll.c | 4 +- net/core/rtnetlink.c | 21 +- net/core/skbuff.c | 103 +- net/core/sock.c | 11 +- net/dccp/ccids/lib/tfrc.c | 2 +- net/dccp/ccids/lib/tfrc.h | 1 + net/decnet/af_decnet.c | 5 - net/hsr/hsr_framereg.c | 2 +- net/ieee802154/6lowpan.c | 23 +- net/ipv4/af_inet.c | 7 +- net/ipv4/devinet.c | 3 +- net/ipv4/inet_fragment.c | 5 +- net/ipv4/ip_forward.c | 71 +- net/ipv4/ip_output.c | 3 - net/ipv4/ip_tunnel.c | 82 +- net/ipv4/ip_tunnel_core.c | 47 +- net/ipv4/ipconfig.c | 2 +- net/ipv4/netfilter/Kconfig | 5 + net/ipv4/netfilter/Makefile | 1 + net/ipv4/netfilter/nf_nat_h323.c | 5 +- net/ipv4/netfilter/nf_nat_snmp_basic.c | 4 +- net/ipv4/netfilter/nft_reject_ipv4.c | 75 + net/ipv4/route.c | 13 +- net/ipv4/tcp.c | 10 +- net/ipv4/tcp_cong.c | 3 +- net/ipv4/tcp_input.c | 21 +- net/ipv4/tcp_output.c | 48 +- net/ipv4/udp_offload.c | 17 +- net/ipv6/Kconfig | 1 + net/ipv6/addrconf.c | 7 +- net/ipv6/exthdrs_core.c | 2 +- net/ipv6/exthdrs_offload.c | 4 +- net/ipv6/icmp.c | 2 +- net/ipv6/ip6_offload.c | 20 +- net/ipv6/ip6_output.c | 20 +- net/ipv6/netfilter/Kconfig | 5 + net/ipv6/netfilter/Makefile | 1 + net/ipv6/netfilter/nft_reject_ipv6.c | 76 + net/ipv6/ping.c | 1 + net/ipv6/route.c | 2 +- net/ipv6/sit.c | 19 + net/ipv6/udp_offload.c | 2 +- net/ipx/af_ipx.c | 22 +- net/ipx/ipx_route.c | 4 - net/l2tp/l2tp_core.c | 4 +- net/l2tp/l2tp_core.h | 1 + net/l2tp/l2tp_netlink.c | 4 +- net/l2tp/l2tp_ppp.c | 13 +- net/mac80211/cfg.c | 44 +- net/mac80211/chan.c | 6 + net/mac80211/ht.c | 4 +- net/mac80211/ibss.c | 5 +- net/mac80211/ieee80211_i.h | 10 +- net/mac80211/iface.c | 33 +- net/mac80211/mesh_ps.c | 1 + net/mac80211/mlme.c | 24 + net/mac80211/rx.c | 7 + net/mac80211/sta_info.c | 67 +- net/mac80211/sta_info.h | 7 +- net/mac80211/tx.c | 17 +- net/mac80211/util.c | 48 +- net/mac80211/wme.c | 5 + net/netfilter/Kconfig | 6 +- net/netfilter/Makefile | 1 + net/netfilter/ipvs/ip_vs_conn.c | 8 +- net/netfilter/nf_conntrack_core.c | 55 +- net/netfilter/nf_conntrack_netlink.c | 35 +- net/netfilter/nf_nat_core.c | 56 +- net/netfilter/nf_synproxy_core.c | 5 +- net/netfilter/nf_tables_api.c | 82 +- net/netfilter/nf_tables_core.c | 6 +- net/netfilter/nft_ct.c | 16 +- net/netfilter/nft_log.c | 5 +- net/netfilter/nft_lookup.c | 1 + net/netfilter/nft_meta.c | 4 +- net/netfilter/nft_payload.c | 3 +- net/netfilter/nft_queue.c | 4 +- net/netfilter/nft_rbtree.c | 16 +- net/netfilter/nft_reject.c | 89 +- net/netfilter/nft_reject_inet.c | 63 + net/netfilter/xt_CT.c | 7 +- net/netlink/af_netlink.c | 4 +- net/nfc/nci/core.c | 2 +- net/openvswitch/datapath.c | 23 +- net/openvswitch/flow_table.c | 88 +- net/openvswitch/flow_table.h | 2 +- net/packet/af_packet.c | 26 +- net/sched/sch_api.c | 7 +- net/sched/sch_fq.c | 21 +- net/sched/sch_pie.c | 21 +- net/sched/sch_tbf.c | 24 +- net/sctp/associola.c | 211 ++- net/sctp/ipv6.c | 2 + net/sctp/sm_make_chunk.c | 4 +- net/sctp/sm_sideeffect.c | 7 +- net/sctp/sm_statefuns.c | 12 +- net/sctp/socket.c | 47 +- net/sctp/sysctl.c | 18 +- net/sctp/ulpevent.c | 8 +- net/socket.c | 17 +- net/sunrpc/auth_gss/auth_gss.c | 19 +- net/sunrpc/backchannel_rqst.c | 6 +- net/sunrpc/svc_xprt.c | 6 +- net/sunrpc/xprtsock.c | 6 +- net/tipc/bearer.c | 7 +- net/tipc/config.c | 11 +- net/tipc/core.c | 109 +- net/tipc/core.h | 2 +- net/tipc/handler.c | 1 - net/tipc/link.c | 7 + net/tipc/name_table.c | 40 +- net/tipc/netlink.c | 8 - net/tipc/ref.c | 3 - net/tipc/server.c | 19 +- net/tipc/server.h | 2 - net/tipc/socket.c | 12 +- net/tipc/subscr.c | 19 +- net/unix/af_unix.c | 3 +- net/wireless/core.c | 19 +- net/wireless/core.h | 4 +- net/wireless/nl80211.c | 32 +- net/wireless/nl80211.h | 8 +- net/wireless/reg.c | 12 +- net/wireless/scan.c | 40 +- net/wireless/sme.c | 2 +- net/xfrm/xfrm_policy.c | 2 +- net/xfrm/xfrm_state.c | 23 +- net/xfrm/xfrm_user.c | 5 - scripts/Makefile.lib | 1 + scripts/checkpatch.pl | 4 +- scripts/gen_initramfs_list.sh | 2 +- scripts/get_maintainer.pl | 2 +- scripts/mod/file2alias.c | 4 +- scripts/mod/modpost.c | 13 + security/Kconfig | 2 +- security/keys/keyring.c | 6 +- security/selinux/nlmsgtab.c | 2 + security/selinux/ss/policydb.c | 8 +- security/selinux/ss/services.c | 4 + sound/core/pcm_lib.c | 1 + sound/pci/hda/hda_codec.c | 34 +- sound/pci/hda/hda_generic.c | 8 +- sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/hda_intel.c | 2 +- sound/pci/hda/patch_analog.c | 31 + sound/pci/hda/patch_ca0132.c | 68 +- sound/pci/hda/patch_conexant.c | 3 +- sound/pci/hda/patch_realtek.c | 92 +- sound/pci/hda/patch_sigmatel.c | 60 +- sound/pci/hda/thinkpad_helper.c | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/Kconfig | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 13 +- sound/soc/atmel/sam9g20_wm8731.c | 20 +- sound/soc/blackfin/Kconfig | 20 +- sound/soc/cirrus/Kconfig | 4 +- sound/soc/cirrus/snappercl15.c | 18 +- sound/soc/codecs/88pm860x-codec.c | 120 +- sound/soc/codecs/Kconfig | 191 ++- sound/soc/codecs/Makefile | 33 + sound/soc/codecs/ad1836.c | 4 +- sound/soc/codecs/ad193x-i2c.c | 54 + sound/soc/codecs/ad193x-spi.c | 48 + sound/soc/codecs/ad193x.c | 154 +- sound/soc/codecs/ad193x.h | 7 + sound/soc/codecs/ad1980.c | 4 +- sound/soc/codecs/adau1373.c | 39 +- sound/soc/codecs/adau1977-i2c.c | 59 + sound/soc/codecs/adau1977-spi.c | 76 + sound/soc/codecs/adau1977.c | 1018 +++++++++++ sound/soc/codecs/adau1977.h | 37 + sound/soc/codecs/adav801.c | 53 + sound/soc/codecs/adav803.c | 50 + sound/soc/codecs/adav80x.c | 159 +- sound/soc/codecs/adav80x.h | 7 + sound/soc/codecs/ak4104.c | 2 - sound/soc/codecs/ak4535.c | 9 - sound/soc/codecs/ak4641.c | 24 +- sound/soc/codecs/ak4642.c | 8 - sound/soc/codecs/ak4671.c | 250 ++- sound/soc/codecs/ak4671.h | 2 - sound/soc/codecs/alc5623.c | 120 +- sound/soc/codecs/alc5632.c | 48 +- sound/soc/codecs/arizona.c | 325 ++-- sound/soc/codecs/cq93vc.c | 3 +- sound/soc/codecs/cs4270.c | 9 - sound/soc/codecs/cs4271.c | 63 +- sound/soc/codecs/cs42l51.c | 99 +- sound/soc/codecs/cs42l52.c | 109 +- sound/soc/codecs/cs42l73.c | 72 +- sound/soc/codecs/cs42xx8-i2c.c | 64 + sound/soc/codecs/cs42xx8.c | 602 +++++++ sound/soc/codecs/cs42xx8.h | 238 +++ sound/soc/codecs/da7210.c | 28 +- sound/soc/codecs/da7213.c | 159 +- sound/soc/codecs/da732x.c | 208 ++- sound/soc/codecs/da732x.h | 3 - sound/soc/codecs/da9055.c | 111 +- sound/soc/codecs/isabelle.c | 71 +- sound/soc/codecs/lm4857.c | 3 +- sound/soc/codecs/lm49453.c | 47 +- sound/soc/codecs/max9768.c | 5 - sound/soc/codecs/max98088.c | 47 +- sound/soc/codecs/max98090.c | 202 +-- sound/soc/codecs/max98090.h | 1 - sound/soc/codecs/max98095.c | 60 +- sound/soc/codecs/max9850.c | 8 - sound/soc/codecs/mc13783.c | 30 +- sound/soc/codecs/ml26124.c | 22 +- sound/soc/codecs/pcm1681.c | 15 +- sound/soc/codecs/pcm1792a.c | 33 +- sound/soc/codecs/pcm512x-i2c.c | 71 + sound/soc/codecs/pcm512x-spi.c | 69 + sound/soc/codecs/pcm512x.c | 589 +++++++ sound/soc/codecs/pcm512x.h | 171 ++ sound/soc/codecs/rt5631.c | 84 +- sound/soc/codecs/rt5640.c | 87 +- sound/soc/codecs/sgtl5000.c | 18 +- sound/soc/codecs/si476x.c | 6 +- sound/soc/codecs/sirf-audio-codec.c | 524 ++++++ sound/soc/codecs/sirf-audio-codec.h | 75 + sound/soc/codecs/sn95031.c | 46 +- sound/soc/codecs/ssm2518.c | 24 +- sound/soc/codecs/ssm2602-i2c.c | 57 + sound/soc/codecs/ssm2602-spi.c | 41 + sound/soc/codecs/ssm2602.c | 180 +- sound/soc/codecs/ssm2602.h | 14 + sound/soc/codecs/sta32x.c | 90 +- sound/soc/codecs/sta529.c | 15 +- sound/soc/codecs/stac9766.c | 38 +- sound/soc/codecs/tlv320aic23-i2c.c | 59 + sound/soc/codecs/tlv320aic23-spi.c | 56 + sound/soc/codecs/tlv320aic23.c | 79 +- sound/soc/codecs/tlv320aic23.h | 6 + sound/soc/codecs/tlv320aic26.c | 7 +- sound/soc/codecs/tlv320aic31xx.c | 19 +- sound/soc/codecs/tlv320aic32x4.c | 2 - sound/soc/codecs/tlv320aic3x.c | 6 - sound/soc/codecs/tlv320dac33.c | 35 +- sound/soc/codecs/twl4030.c | 101 +- sound/soc/codecs/twl6040.c | 17 +- sound/soc/codecs/uda134x.c | 3 +- sound/soc/codecs/uda1380.c | 43 +- sound/soc/codecs/wl1273.c | 9 +- sound/soc/codecs/wm2000.c | 2 - sound/soc/codecs/wm2200.c | 25 +- sound/soc/codecs/wm5100.c | 43 +- sound/soc/codecs/wm5102.c | 32 +- sound/soc/codecs/wm5110.c | 22 +- sound/soc/codecs/wm8350.c | 4 +- sound/soc/codecs/wm8400.c | 37 +- sound/soc/codecs/wm8510.c | 10 +- sound/soc/codecs/wm8523.c | 11 +- sound/soc/codecs/wm8580.c | 9 +- sound/soc/codecs/wm8711.c | 8 +- sound/soc/codecs/wm8728.c | 11 +- sound/soc/codecs/wm8731.c | 11 +- sound/soc/codecs/wm8737.c | 56 +- sound/soc/codecs/wm8741.c | 40 +- sound/soc/codecs/wm8750.c | 6 - sound/soc/codecs/wm8753.c | 12 +- sound/soc/codecs/wm8770.c | 10 +- sound/soc/codecs/wm8776.c | 6 - sound/soc/codecs/wm8804.c | 10 +- sound/soc/codecs/wm8900.c | 52 +- sound/soc/codecs/wm8903.c | 118 +- sound/soc/codecs/wm8904.c | 86 +- sound/soc/codecs/wm8940.c | 26 +- sound/soc/codecs/wm8955.c | 19 +- sound/soc/codecs/wm8958-dsp2.c | 10 +- sound/soc/codecs/wm8960.c | 6 - sound/soc/codecs/wm8961.c | 23 +- sound/soc/codecs/wm8962.c | 87 +- sound/soc/codecs/wm8971.c | 6 - sound/soc/codecs/wm8974.c | 10 +- sound/soc/codecs/wm8978.c | 38 +- sound/soc/codecs/wm8983.c | 51 +- sound/soc/codecs/wm8985.c | 46 +- sound/soc/codecs/wm8988.c | 70 +- sound/soc/codecs/wm8990.c | 49 +- sound/soc/codecs/wm8991.c | 52 +- sound/soc/codecs/wm8993.c | 74 +- sound/soc/codecs/wm8994.c | 186 +- sound/soc/codecs/wm8995.c | 50 +- sound/soc/codecs/wm8996.c | 87 +- sound/soc/codecs/wm8997.c | 29 +- sound/soc/codecs/wm9081.c | 34 +- sound/soc/codecs/wm9090.c | 10 - sound/soc/codecs/wm9705.c | 12 +- sound/soc/codecs/wm_adsp.c | 50 +- sound/soc/codecs/wm_hubs.c | 16 +- sound/soc/davinci/davinci-evm.c | 81 +- sound/soc/davinci/davinci-mcasp.c | 294 ++-- sound/soc/davinci/edma-pcm.c | 57 + sound/soc/davinci/edma-pcm.h | 25 + sound/soc/fsl/Kconfig | 12 +- sound/soc/fsl/eukrea-tlv320.c | 108 +- sound/soc/fsl/fsl_esai.c | 36 +- sound/soc/fsl/fsl_esai.h | 2 +- sound/soc/fsl/fsl_sai.c | 332 ++-- sound/soc/fsl/fsl_sai.h | 48 +- sound/soc/fsl/fsl_spdif.c | 9 +- sound/soc/fsl/imx-mc13783.c | 1 - sound/soc/fsl/imx-pcm-fiq.c | 7 +- sound/soc/fsl/imx-sgtl5000.c | 10 +- sound/soc/fsl/imx-wm8962.c | 11 +- sound/soc/fsl/wm1133-ev1.c | 11 +- sound/soc/generic/simple-card.c | 295 ++-- sound/soc/intel/Kconfig | 42 +- sound/soc/intel/Makefile | 27 +- sound/soc/intel/byt-rt5640.c | 187 ++ sound/soc/intel/haswell.c | 235 +++ sound/soc/intel/mfld_machine.c | 108 +- sound/soc/intel/sst-acpi.c | 284 ++++ sound/soc/intel/sst-baytrail-dsp.c | 372 ++++ sound/soc/intel/sst-baytrail-ipc.c | 867 ++++++++++ sound/soc/intel/sst-baytrail-ipc.h | 69 + sound/soc/intel/sst-baytrail-pcm.c | 422 +++++ sound/soc/intel/sst-dsp-priv.h | 309 ++++ sound/soc/intel/sst-dsp.c | 385 +++++ sound/soc/intel/sst-dsp.h | 233 +++ sound/soc/intel/sst-firmware.c | 587 +++++++ sound/soc/intel/sst-haswell-dsp.c | 517 ++++++ sound/soc/intel/sst-haswell-ipc.c | 1785 ++++++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 488 ++++++ sound/soc/intel/sst-haswell-pcm.c | 872 ++++++++++ sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} | 8 +- .../intel/{sst_platform.c => sst-mfld-platform.c} | 8 +- .../intel/{sst_platform.h => sst-mfld-platform.h} | 4 +- sound/soc/kirkwood/Kconfig | 11 +- sound/soc/kirkwood/Makefile | 2 + sound/soc/kirkwood/armada-370-db.c | 148 ++ sound/soc/kirkwood/kirkwood-i2s.c | 1 + sound/soc/omap/Kconfig | 4 +- sound/soc/omap/ams-delta.c | 55 +- sound/soc/omap/n810.c | 26 +- sound/soc/omap/omap-abe-twl6040.c | 3 +- sound/soc/omap/rx51.c | 22 +- sound/soc/pxa/corgi.c | 49 +- sound/soc/pxa/e740_wm9705.c | 10 +- sound/soc/pxa/e750_wm9705.c | 10 +- sound/soc/pxa/e800_wm9712.c | 19 +- sound/soc/pxa/magician.c | 60 +- sound/soc/pxa/mioa701_wm9713.c | 19 +- sound/soc/pxa/poodle.c | 7 +- sound/soc/pxa/spitz.c | 58 +- sound/soc/pxa/tosa.c | 67 +- sound/soc/pxa/zylonite.c | 17 +- sound/soc/s6000/s6105-ipcam.c | 28 +- sound/soc/samsung/Kconfig | 8 +- sound/soc/samsung/h1940_uda1380.c | 7 +- sound/soc/samsung/neo1973_wm8753.c | 168 +- sound/soc/samsung/rx1950_uda1380.c | 5 - sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/tobermory.c | 2 + sound/soc/sh/fsi.c | 2 +- sound/soc/sh/migor.c | 19 +- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/adg.c | 229 ++- sound/soc/sh/rcar/core.c | 426 +++-- sound/soc/sh/rcar/gen.c | 113 +- sound/soc/sh/rcar/rsnd.h | 206 ++- sound/soc/sh/rcar/scu.c | 384 ----- sound/soc/sh/rcar/src.c | 723 ++++++++ sound/soc/sh/rcar/ssi.c | 388 ++--- sound/soc/sirf/Kconfig | 14 + sound/soc/sirf/Makefile | 5 + sound/soc/sirf/sirf-audio-port.c | 194 +++ sound/soc/sirf/sirf-audio-port.h | 62 + sound/soc/sirf/sirf-audio.c | 156 ++ sound/soc/soc-cache.c | 13 +- sound/soc/soc-compress.c | 65 +- sound/soc/soc-core.c | 552 +++--- sound/soc/soc-dapm.c | 604 +++---- sound/soc/soc-io.c | 99 +- sound/soc/soc-jack.c | 5 +- sound/soc/soc-pcm.c | 109 +- sound/soc/spear/spdif_out.c | 10 +- sound/soc/tegra/Kconfig | 2 +- sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra20_das.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra20_spdif.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 4 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/txx9/txx9aclc-ac97.c | 8 +- sound/usb/Kconfig | 1 + sound/usb/mixer.c | 1 + sound/usb/mixer_maps.c | 9 + tools/lib/lockdep/Makefile | 6 +- tools/lib/lockdep/preload.c | 2 +- tools/lib/lockdep/run_tests.sh | 0 tools/lib/lockdep/uinclude/asm/hash.h | 6 + tools/lib/lockdep/uinclude/linux/rcu.h | 5 + tools/net/Makefile | 2 +- tools/perf/builtin-buildid-cache.c | 33 +- tools/perf/builtin-record.c | 10 +- tools/perf/builtin-report.c | 40 +- tools/perf/builtin-top.c | 6 +- tools/perf/builtin-trace.c | 32 +- tools/perf/config/Makefile | 2 +- tools/perf/config/feature-checks/Makefile | 2 +- tools/perf/design.txt | 1 - tools/perf/perf.h | 4 +- tools/perf/tests/vmlinux-kallsyms.c | 10 - tools/perf/util/annotate.c | 9 +- tools/perf/util/annotate.h | 2 + tools/perf/util/event.c | 36 +- tools/perf/util/event.h | 6 +- tools/perf/util/include/asm/hash.h | 6 + tools/perf/util/include/linux/bitops.h | 4 +- tools/perf/util/machine.c | 44 +- tools/perf/util/machine.h | 2 + tools/perf/util/map.c | 5 +- tools/perf/util/map.h | 1 + tools/perf/util/parse-events.c | 17 +- tools/perf/util/probe-event.c | 2 +- tools/perf/util/session.c | 6 + tools/perf/util/symbol-elf.c | 10 +- tools/perf/util/symbol.c | 67 +- tools/testing/selftests/ipc/msgque.c | 1 + virt/kvm/arm/vgic.c | 1 + virt/kvm/coalesced_mmio.c | 8 +- 1779 files changed, 37413 insertions(+), 15444 deletions(-) create mode 100644 Documentation/devicetree/bindings/interrupt-controller/lsi,zevio-intc.txt create mode 100644 Documentation/devicetree/bindings/net/opencores-ethoc.txt create mode 100644 Documentation/devicetree/bindings/net/sti-dwmac.txt rename Documentation/devicetree/bindings/pinctrl/{brcm,capri-pinctrl.txt => brcm,bcm11351-pinctrl.txt} (98%) create mode 100644 Documentation/devicetree/bindings/power/bq2415x.txt create mode 100644 Documentation/devicetree/bindings/sound/armada-370db-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/cs42xx8.txt create mode 100644 Documentation/devicetree/bindings/sound/da9055.txt create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm512x.txt create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-port.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/tdm-slot.txt create mode 100644 Documentation/devicetree/bindings/sound/widgets.txt delete mode 100644 Documentation/networking/3c505.txt create mode 100644 Documentation/spi/00-INDEX create mode 100644 Documentation/zh_CN/arm64/tagged-pointers.txt create mode 100644 arch/arm/boot/dts/at91-sama5d3_xplained.dts create mode 100644 arch/arm/boot/dts/omap3-overo-storm-tobi.dts rename arch/arm/boot/dts/{omap3-tobi.dts => omap3-overo-tobi-common.dtsi} (94%) create mode 100644 arch/arm/boot/dts/omap3-overo-tobi.dts delete mode 100644 arch/arm/boot/dts/testcases/tests.dtsi delete mode 100644 arch/m68k/include/asm/barrier.h create mode 100644 drivers/gpu/drm/nouveau/core/subdev/mc/nv4c.c create mode 100644 drivers/irqchip/irq-zevio.c create mode 100644 drivers/net/ethernet/stmicro/stmmac/dwmac-sti.c delete mode 100644 drivers/net/irda/ep7211-sir.c delete mode 100644 drivers/net/usb/r815x.c create mode 100644 drivers/net/usb/sr9800.c create mode 100644 drivers/net/usb/sr9800.h create mode 100644 drivers/of/testcase-data/testcases.dtsi rename {arch/arm/boot/dts/testcases => drivers/of/testcase-data}/tests-interrupts.dtsi (100%) create mode 100644 drivers/of/testcase-data/tests-match.dtsi rename {arch/arm/boot/dts/testcases => drivers/of/testcase-data}/tests-phandle.dtsi (100%) delete mode 100644 drivers/xen/xencomm.c create mode 100644 include/linux/platform_data/adau1977.h create mode 100644 include/net/netfilter/nft_reject.h create mode 100644 include/trace/events/hswadsp.h create mode 100644 include/trace/events/intel-sst.h rename include/{ => uapi}/xen/gntalloc.h (100%) rename include/{ => uapi}/xen/gntdev.h (100%) delete mode 100644 include/xen/interface/xencomm.h delete mode 100644 include/xen/xencomm.h create mode 100644 net/ipv4/netfilter/nft_reject_ipv4.c create mode 100644 net/ipv6/netfilter/nft_reject_ipv6.c create mode 100644 net/netfilter/nft_reject_inet.c create mode 100644 sound/soc/codecs/ad193x-i2c.c create mode 100644 sound/soc/codecs/ad193x-spi.c create mode 100644 sound/soc/codecs/adau1977-i2c.c create mode 100644 sound/soc/codecs/adau1977-spi.c create mode 100644 sound/soc/codecs/adau1977.c create mode 100644 sound/soc/codecs/adau1977.h create mode 100644 sound/soc/codecs/adav801.c create mode 100644 sound/soc/codecs/adav803.c create mode 100644 sound/soc/codecs/cs42xx8-i2c.c create mode 100644 sound/soc/codecs/cs42xx8.c create mode 100644 sound/soc/codecs/cs42xx8.h create mode 100644 sound/soc/codecs/pcm512x-i2c.c create mode 100644 sound/soc/codecs/pcm512x-spi.c create mode 100644 sound/soc/codecs/pcm512x.c create mode 100644 sound/soc/codecs/pcm512x.h create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h create mode 100644 sound/soc/codecs/ssm2602-i2c.c create mode 100644 sound/soc/codecs/ssm2602-spi.c create mode 100644 sound/soc/codecs/tlv320aic23-i2c.c create mode 100644 sound/soc/codecs/tlv320aic23-spi.c create mode 100644 sound/soc/davinci/edma-pcm.c create mode 100644 sound/soc/davinci/edma-pcm.h create mode 100644 sound/soc/intel/byt-rt5640.c create mode 100644 sound/soc/intel/haswell.c create mode 100644 sound/soc/intel/sst-acpi.c create mode 100644 sound/soc/intel/sst-baytrail-dsp.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.h create mode 100644 sound/soc/intel/sst-baytrail-pcm.c create mode 100644 sound/soc/intel/sst-dsp-priv.h create mode 100644 sound/soc/intel/sst-dsp.c create mode 100644 sound/soc/intel/sst-dsp.h create mode 100644 sound/soc/intel/sst-firmware.c create mode 100644 sound/soc/intel/sst-haswell-dsp.c create mode 100644 sound/soc/intel/sst-haswell-ipc.c create mode 100644 sound/soc/intel/sst-haswell-ipc.h create mode 100644 sound/soc/intel/sst-haswell-pcm.c rename sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} (96%) rename sound/soc/intel/{sst_platform.c => sst-mfld-platform.c} (99%) rename sound/soc/intel/{sst_platform.h => sst-mfld-platform.h} (97%) create mode 100644 sound/soc/kirkwood/armada-370-db.c delete mode 100644 sound/soc/sh/rcar/scu.c create mode 100644 sound/soc/sh/rcar/src.c create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio-port.c create mode 100644 sound/soc/sirf/sirf-audio-port.h create mode 100644 sound/soc/sirf/sirf-audio.c mode change 100644 => 100755 tools/lib/lockdep/run_tests.sh create mode 100644 tools/lib/lockdep/uinclude/asm/hash.h create mode 100644 tools/perf/util/include/asm/hash.h

2 1

Re: [alsa-devel] [PATCH 42/44] bebob/firewire-lib: Add a quirk of wrong dbc in empty packet for M-Audio special Firewire series
by Takashi Sakamoto 24 Mar '14

24 Mar '14

Hi Euan, Thanks for your testing. (Mar 24 2014 10:41), Euan de Kock wrote: > Thanks for the update. This bit me last night, and I just commented out > the check to keep working. > > I'll try and retest it tonight with this version. (On a Echo Audiofire > Pre8 with firmware < 5) > > Regards, > > Euan deKock. When reading this message, I wondered why this bug affects driving on Echo AudioFirePre8. As describing the title, this affects a part of BeBoB based devices. Did you test with M-Audio Firewire 1814 or M-Audio ProjectMix I/O? If not, it may be a bug for Fireworks driver. Then please send me output from: /proc/asound/cardX/firewire/*. (The 'cardX' should be for your device.) Thanks Takashi Sakamoto o-takashi(a)sakamocchi.jp

1 0

Re: [alsa-devel] [PATCH 3.13 110/149] ASoC: pcm: free path list before exiting from error conditions
by Mark Brown 22 Mar '14

22 Mar '14

On Sat, Mar 22, 2014 at 03:53:08PM +0000, Ben Hutchings wrote: > On Thu, 2014-03-20 at 17:04 -0700, Greg Kroah-Hartman wrote: If you spot something on a stable patch that applies upstream (as opposed to being stable process or to do with the older kernel which are the most common things with these mails) please add both the maintainers and relevant lists. -stable mail only goes to signoffs so misses both lists and additional maintainers meaning things may not be going to the people who should see them (in this case it's Liam who mostly looks at DPCM for example). It might make sense for the -stable mails to do this in general, at least with the lists if not the maintainers. Ideally it's helpful to also send it in a separate thread since the volume of mail about -stable stuff is extremely high so things are easily missed but that's a bit more fun to do, I'm really not sure there's any general way to resolve that one sensibly. > > dpcm_path_get() allocates dynamic memory to hold path list. > > Corresponding dpcm_path_put() must be called to free the memory. > > dpcm_path_put() is not called under several error conditions. > > This leads to memory leak. > This is broken. dpcm_path_get() may return -ENOMEM and not initialise > the list at all. > If snd_soc_dapm_dai_get_connected_widgets() can fail (I don't think it > can) then dpcm_path_get() should be responsible for freeing the list > before returning. It can't fail without memory corruption or internal bugs and would only fail with a crash. > [...] > > --- a/sound/soc/soc-pcm.c > > +++ b/sound/soc/soc-pcm.c > [...] > > @@ -1979,6 +1981,7 @@ static int dpcm_fe_dai_open(struct snd_p > > fe->dpcm[stream].runtime = fe_substream->runtime; > > > > if (dpcm_path_get(fe, stream, &list) <= 0) { > > + dpcm_path_put(&list); > This is the one place where a memory leak seems to be possible, but the > < 0 and == 0 cases need to be distinguished. > Greg, please drop this until it is fixed properly upstream. It's actually not going to cause a leak there at all since we have an unconditional run through the rest of the function to a double free with references to the list that gets freed, AFAICT there wasn't a leak to start off with. I think what the function needs to do here is bomb out early on -ENOMEM and otherwise trundle on. Anyway, I'll revert this upstream. Thanks for noticing this.

1 0

[alsa-devel] [PATCH] ASoC: pcm: Drop incorrect double/extra frees
by Mark Brown 22 Mar '14

22 Mar '14

From: Mark Brown <broonie(a)linaro.org> The changes in "ASoC: pcm: free path list before exiting from error conditions" actually introduced both double frees (in case where the path list was allocated but empty) and frees of unallocated memory (in cases where the error being handled was -ENOMEM. Drop the commit for now. Fixes: e4ad1accb (ASoC: pcm: free path list before exiting from error conditions) Reported-by: Ben Hutchings <ben(a)decadent.org.uk> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-pcm.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 28522bd03b8e..47e1ce771e65 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1989,7 +1989,6 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); if (paths < 0) { - dpcm_path_put(&list); dev_warn(fe->dev, "ASoC: %s no valid %s path\n", fe->dai_link->name, "playback"); mutex_unlock(&card->mutex); @@ -2019,7 +2018,6 @@ capture: paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); if (paths < 0) { - dpcm_path_put(&list); dev_warn(fe->dev, "ASoC: %s no valid %s path\n", fe->dai_link->name, "capture"); mutex_unlock(&card->mutex); @@ -2084,7 +2082,6 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) fe->dpcm[stream].runtime = fe_substream->runtime; if (dpcm_path_get(fe, stream, &list) <= 0) { - dpcm_path_put(&list); dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); } -- 1.9.1

1 0

[alsa-devel] Edirol DA2496 and word clock sync
by Brian McNeil 22 Mar '14

22 Mar '14

Hello all; I'm new to the list, and sitting playing with a pair of Edirol DA2496 cards. These support Word Clock synchronisation, which can be done using S/PDIF, a 75 ohm coax cable on the breakout boxes, or with a internal jumper cable between the PCI cards. Unfortunately neither the driver, or Mudita24, were exposing access to the Word Clock. I've managed to patch support into both; doing so with the 3.12.13-rt21 kernel, and mudita24-1.0.3+svn13 from the Ubuntu repositories. This is, mostly, by treating the cards the same as the Delta1010LT. Forgive me being so rusty with C that I've not provided a patch; going to have to redo this and actually build proper diffs for it. I do have a related query, about Xruns - when using an .asoundrc file to merge the two cards into a single virtual card - is there any way to properly suppress backend Xruns that have no impact on the actual recorded audio? Brian McNeil -- Wikinewsie.org | http://wikinewsie.org | https://en.wikinews.org "Facts don't cease to be facts, but news ceases to be news." GPG Fingerprint: 7C3D FFD5 5ED5 B80F 1D18 A52B E84C 8928 6ABC A7AD

1 0

[alsa-devel] [PATCH] ASoC: mfld_machine: Fix compile error
by Lars-Peter Clausen 21 Mar '14

21 Mar '14

Fixes: 115f3f8 ("ASoC: mfld_machine: Convert to table based DAPM and control setup") Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/intel/mfld_machine.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c index 4616942..031d787 100644 --- a/sound/soc/intel/mfld_machine.c +++ b/sound/soc/intel/mfld_machine.c @@ -254,7 +254,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEINR"); /* Headset and button jack detection */ - ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", + ret_val = snd_soc_jack_new(mfld_codec, "Intel(R) MID Audio Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack); if (ret_val) { -- 1.8.0

2 1

[alsa-devel] [PATCH - ALSA-LIB 1/1] Remove 2.1 , 4.1 and 5.0 from EMU10K1.conf
by Raymond Yau 21 Mar '14

21 Mar '14

- Creative 4.1 Speaker only have green and black jacks - Those creative sound cards using 4.1 speakers does not support 6 channels

2 1

[alsa-devel] Ambiguous DAPM widget names and DPCM
by Daniel Mack 21 Mar '14

21 Mar '14

Hi Mark, Hi Liam, I spent some hours reverse-engineering the DPCM code, and I plan to prepare a patch to add some documentation once I've assembled a fully working setup. However, I have a general question regarding DAPM widget names. DPCM uses DAPM widgets and routes in order to determine the FE <-> BE connections at runtime. The problem with this is that widget names are very ambiguous, as the automatically created widgets for DAIs are named after their stream_name properties, which contain "Playback" or "Capture" for all components in my case. Hence, the code which walks the connections (snd_soc_dapm_dai_get_connected_widgets()) will most likely match the wrong one (which has empty sound/sink lists), which results in a failure like this: [ 16.606268] fe: ASoC: fe no valid playback route I hacked the names of all components so they are unique, but even then, the problem is that snd_soc_dapm_new_dai_widgets() is called twice for the CPU DAI, once from soc_probe_platform(), and later from soc_probe_link_dais(), and both calls would create DAPM widgets with identical names, resulting in the same problem as described above. I don't know how to handle this, but I believe that nobody was really hit by this issue yet, as DPCM doesn't seem to be used widely, at least not by any machine code in mainline. One idea would be to change the automatically generated names of DAI widgets in order to make them unique, or re-factor the lookup routines. Any opinion on that? Do I miss a general consideration here? With more hacks in place, the setup somehow succeeds, but when the stream is opened, we'll hit a BUG() in sound/core/pcm_memory.c, because the FE device's DMA type is undefined. But that's another topic I'll address later. Thanks, Daniel

5 9

[alsa-devel] [PATCH v5] ASoC: add RT286 CODEC driver
by bardliao＠realtek.com 21 Mar '14

21 Mar '14

From: Bard Liao <bardliao(a)realtek.com> This patch adds the ALC286 codec driver. ALC286 is a dual mode codec, which can run as HD-A or I2S mode. It is controlled by HD-A verb commands via I2C protocol. The following is the I/O difference between ALC286 and general I2S codecs. 1. A HD-A verb command contains three parts, NID, VID, and PID. And an I2S command contains only two parts: address and data. 2. Not only the register address is written, but the read command also includes the entire write command. 3. rt286 uses different registers for read and write the same bits. As a result, standard regmap is difficult to be used on ALC286. We don't request a standard I/O by snd_soc_codec_set_cache_io anymore. Now we have ,reg_write and .reg_read functions for ALC286's I/O. And we don't use cache due to item 3 above. Some dummy registers (address <= 0xff) are defined for dapm routing. Thhe dummy registers are cache only. Due to item 2 above, HD-A verb commands are put into the address part of regmap. When we issue HD-A verb write commands, the data part of regmap is zero. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski(a)intel.com> --- The jack detection function is done by Lewandowski Gustaw. So I add Gustaw's Signed-off-by line in this patch The difference between this version and previous version is listed below. * Include <sound/hda_verbs.h> instead of hda_codec.h * Fix rt286_update_bits doesn't write the new value to codec issue. * Apply different settings for system clock = 19.2MHz and 24MHz. * Use .set_bclk_ratio instead of .set_dfs. * Change ACPI number from INT33CA to INT343A. * Other minor changes. --- include/sound/rt286.h | 19 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt286.c | 1379 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 159 ++++++ 5 files changed, 1563 insertions(+) create mode 100644 include/sound/rt286.h create mode 100644 sound/soc/codecs/rt286.c create mode 100644 sound/soc/codecs/rt286.h diff --git a/include/sound/rt286.h b/include/sound/rt286.h new file mode 100644 index 0000000..eb773d1 --- /dev/null +++ b/include/sound/rt286.h @@ -0,0 +1,19 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * 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 __LINUX_SND_RT286_H +#define __LINUX_SND_RT286_H + +struct rt286_platform_data { + bool cbj_en; /*combo jack enable*/ + bool gpio2_en; /*GPIO2 enable*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 32d7a6f..c46e722 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -68,6 +68,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_RT286 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C @@ -379,6 +380,9 @@ config SND_SOC_PCM512x_SPI select SND_SOC_PCM512x select REGMAP_SPI +config SND_SOC_RT286 + tristate + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cb46c4c..ee2f368 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -56,6 +56,7 @@ snd-soc-pcm3008-objs := pcm3008.o snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o +snd-soc-rt286-objs := rt286.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -204,6 +205,7 @@ obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o +obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 0000000..4621b62 --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1379 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 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 <linux/acpi.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 <sound/jack.h> +#include <sound/rt286.h> +#include <sound/hda_verbs.h> + +#include "rt286.h" + +struct rt286_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct rt286_platform_data pdata; + struct snd_soc_jack *jack; + struct i2c_client *i2c; + int sys_clk; +}; + +static unsigned int rt286_reg_cache[] = { + [RT286_AUDIO_FUNCTION_GROUP] = 0x0000, + [RT286_DAC_OUT1] = 0x7f7f, + [RT286_DAC_OUT2] = 0x7f7f, + [RT286_SPDIF] = 0x0000, + [RT286_ADC_IN1] = 0x4343, + [RT286_ADC_IN2] = 0x4343, + [RT286_MIC1] = 0x0000, + [RT286_MIXER_IN] = 0x000b, + [RT286_MIXER_OUT1] = 0x0002, + [RT286_MIXER_OUT2] = 0x0000, + [RT286_DMIC1] = 0x0000, + [RT286_DMIC2] = 0x0000, + [RT286_LINE1] = 0x0000, + [RT286_BEEP] = 0x0000, + [RT286_VENDOR_REGISTERS] = 0x0000, + [RT286_SPK_OUT] = 0x8080, + [RT286_HP_OUT] = 0x8080, + [RT286_MIXER_IN1] = 0x0000, + [RT286_MIXER_IN2] = 0x0000, +}; + +static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + int ret; + unsigned int buf = 0x0; + + if (reg <= 0xff) { /*read cache*/ + *value = rt286_reg_cache[reg]; + return 0; + } + reg = cpu_to_be32(reg); + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + *value = be32_to_cpu(buf); + + return 0; +} + +static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) +{ + struct i2c_client *client = context; + u8 data[4]; + int ret; + + if (reg <= 0xff) { /*write cache*/ + rt286_reg_cache[reg] = value; + return 0; + } + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + data[2] = (reg >> 8) & 0xff; + data[3] = reg & 0xff; + + ret = i2c_master_send(client, data, 4); + if (ret == 4) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static int rt286_update_bits(struct snd_soc_codec *codec, unsigned int vid, + unsigned int nid, unsigned int data, + unsigned int mask, unsigned int value) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int old, new, verb; + int change, ret; + + verb = VERB_CMD((vid | 0x800), nid, data); + regmap_read(rt286->regmap, verb, &old); + new = (old & ~mask) | (value & mask); + change = old != new; + + if (change) { + verb = VERB_CMD(vid, nid, new); + ret = regmap_write(rt286->regmap, verb, 0); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_index_write(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index, unsigned int data) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int ret; + unsigned int verb; + + verb = VERB_CMD(AC_VERB_SET_COEF_INDEX, wid, index); + ret = regmap_write(rt286->regmap, verb, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + goto err; + } + verb = VERB_CMD(AC_VERB_SET_PROC_COEF, wid, data); + ret = regmap_write(rt286->regmap, verb, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + + return 0; +err: + return ret; +} + +static unsigned int rt286_index_read(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int ret; + unsigned int verb, val; + + verb = VERB_CMD(AC_VERB_SET_COEF_INDEX, wid, index); + ret = regmap_write(rt286->regmap, verb, 0); + + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + return ret; + } + + verb = VERB_CMD(AC_VERB_GET_PROC_COEF, wid, index); + regmap_read(rt286->regmap, verb, &val); + return val; +} + +static int rt286_index_update_bits(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index, + unsigned int mask, unsigned int data) +{ + unsigned int old, new; + int change, ret; + + old = rt286_index_read(codec, wid, index); + new = (old & ~mask) | (data & mask); + change = old != new; + + if (change) { + ret = rt286_index_write(codec, wid, index, new); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_support_power_controls[] = { + RT286_DAC_OUT1, + RT286_DAC_OUT2, + RT286_ADC_IN1, + RT286_ADC_IN2, + RT286_MIC1, + RT286_DMIC1, + RT286_DMIC2, + RT286_SPK_OUT, + RT286_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static const struct reg_default rt286_reg[] = { + { RT286_DAC_OUT1, 0x7f7f }, + { RT286_DAC_OUT2, 0x7f7f }, + { RT286_SPDIF, 0x0000 }, + { RT286_ADC_IN1, 0x4343 }, + { RT286_ADC_IN2, 0x4343 }, + { RT286_MIC1, 0x0000 }, + { RT286_MIXER_IN, 0x000b }, + { RT286_MIXER_OUT1, 0x0002 }, + { RT286_MIXER_OUT2, 0x0000 }, + { RT286_SPK_OUT, 0x0000 }, + { RT286_HP_OUT, 0x0000 }, + { RT286_MIXER_IN1, 0x0005 }, + { RT286_MIXER_IN2, 0x0005 }, +}; + +static bool rt286_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT286_DAC_OUT1: + case RT286_DAC_OUT2: + case RT286_ADC_IN1: + case RT286_ADC_IN2: + case RT286_MIC1: + case RT286_SPDIF: + case RT286_MIXER_IN: + case RT286_MIXER_OUT1: + case RT286_MIXER_OUT2: + case RT286_DMIC1: + case RT286_DMIC2: + case RT286_SPK_OUT: + case RT286_HP_OUT: + case RT286_MIXER_IN1: + case RT286_MIXER_IN2: + return true; + default: + return false; + } +} + +static int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val, buf; + int i; + + *hp = false; + *mic = false; + + if (rt286->pdata.cbj_en) { + regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); + + *hp = buf & 0x80000000; + if (*hp) { + /* power on HV,VERF */ + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_POWER_CTRL1, 0x1001, 0x0); + /* power LDO1 */ + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_POWER_CTRL2, 0x4, 0x4); + regmap_write(rt286->regmap, RT286_SET_MIC1_24, 0); + val = rt286_index_read(codec, + RT286_VENDOR_REGISTERS, RT286_CBJ_CTRL2); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + val = rt286_index_read(codec, + RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL2); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *mic = false; + + regmap_write(rt286->regmap, + RT286_SET_MIC1_20, 0); + /* power off HV,VERF */ + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, + RT286_POWER_CTRL1, 0x1001, 0x1001); + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL3, 0xc000, 0x0000); + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL1, 0x0030, 0x0000); + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL2, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *mic = true; + } else { + *mic = false; + } + + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_MISC_CTRL1, + 0x0060, 0x0000); + } else { + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_MISC_CTRL1, + 0x0060, 0x0020); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL3, + 0xc000, 0x8000); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL1, + 0x0030, 0x0020); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL2, + 0xc000, 0x8000); + + *mic = false; + } + } else { + regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); + *hp = buf & 0x80000000; + regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf); + *mic = buf & 0x80000000; + } + + + /* Clear IRQ */ + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_IRQ_CTRL, 0x1, 0x1); + return 0; +} + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + rt286->jack = jack; + + /* Send an initial empty report */ + snd_soc_jack_report(rt286->jack, 0, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + return 0; +} +EXPORT_SYMBOL_GPL(rt286_mic_detect); + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static int rt286_vol_get(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_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, buf, io = 0x8000; + + if (RT286_ADC_IN1 == mc->reg) + io = 0; + verb = VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, mc->reg, (0x2000 | io)); + regmap_read(rt286->regmap, verb, &buf); + ucontrol->value.integer.value[0] = buf & 0x7f; + verb = VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, mc->reg, io); + regmap_read(rt286->regmap, verb, &buf); + ucontrol->value.integer.value[1] = buf & 0x7f; + + return 0; +} + +static int rt286_vol_put(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_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, vall, valr, io; + + if (RT286_ADC_IN1 == mc->reg) + io = 0x4000; + else + io = 0x8000; + vall = ucontrol->value.integer.value[0]; + valr = ucontrol->value.integer.value[1]; + if (vall == valr) { + vall = vall | 0x3000 | io; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + regmap_write(rt286->regmap, verb, 0); + } else { + vall = vall | 0x2000 | io; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + regmap_write(rt286->regmap, verb, 0); + valr = valr | 0x1000 | io; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + regmap_write(rt286->regmap, verb, 0); + } + + return 0; +} + +static int rt286_mic_gain_get(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_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, buf; + + verb = VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, mc->reg, 0x0); + regmap_read(rt286->regmap, verb, &buf); + ucontrol->value.integer.value[0] = buf & 0x3; + + return 0; +} + +static int rt286_mic_gain_put(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_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, val; + + val = ucontrol->value.integer.value[0]; + val = val | 0x7000; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, val); + regmap_write(rt286->regmap, verb, 0); + + return 0; +} + +static int rt286_playback_switch_get(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_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, buf; + + verb = VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, mc->reg, 0xa000); + regmap_read(rt286->regmap, verb, &buf); + ucontrol->value.integer.value[0] = !(buf & 0x80); + verb = VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, mc->reg, 0x8000); + regmap_read(rt286->regmap, verb, &buf); + ucontrol->value.integer.value[1] = !(buf & 0x80); + + return 0; +} + +static int rt286_playback_switch_put(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_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, vall, valr, val = 0; + + vall = (!ucontrol->value.integer.value[0] << 7); + valr = (!ucontrol->value.integer.value[1] << 7); + if (vall) + val |= 0x8000; + if (valr) + val |= 0x0080; + + if (vall == valr) { + vall = vall | 0xb000; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + regmap_write(rt286->regmap, verb, 0); + } else { + vall = vall | 0xa000; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + regmap_write(rt286->regmap, verb, 0); + valr = valr | 0x9000; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + regmap_write(rt286->regmap, verb, 0); + } + snd_soc_update_bits(codec, mc->reg, 0x8080, val); + + return 0; +} + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_EXT_TLV("DAC0 Playback Volume", RT286_DAC_OUT1, + 8, 0, 0x7f, 0, + rt286_vol_get, rt286_vol_put, + out_vol_tlv), + SOC_DOUBLE_EXT_TLV("ADC0 Capture Volume", RT286_ADC_IN1, + 8, 0, 0x7f, 0, + rt286_vol_get, rt286_vol_put, + out_vol_tlv), + SOC_SINGLE_EXT_TLV("AMIC Volume", RT286_MIC1, 0, 0x3, 0, + rt286_mic_gain_get, rt286_mic_gain_put, + mic_vol_tlv), + SOC_DOUBLE_EXT("Headphone Playback Switch", RT286_HP_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), + SOC_DOUBLE_EXT("Speaker Playback Switch", RT286_SPK_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT286_MIXER_OUT1, + RT286_M_FRONT_DAC_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", RT286_MIXER_OUT1, + RT286_M_FRONT_REC_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Beep Switch", RT286_MIXER_IN, + RT286_M_REC_BEEP_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", RT286_MIXER_IN, + RT286_M_REC_LINE1_SFT, 1, 1), + SOC_DAPM_SINGLE("Mic1 Switch", RT286_MIXER_IN, + RT286_M_REC_MIC1_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", RT286_MIXER_IN, + RT286_M_REC_I2S_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spo_enable_control = + SOC_DAPM_SINGLE("Switch", RT286_SPK_OUT, + RT286_M_SPK_MUX_SFT, 1, 1); + +static const struct snd_kcontrol_new hpo_enable_control = + SOC_DAPM_SINGLE("Switch", RT286_HP_OUT, + RT286_M_HP_MUX_SFT, 1, 1); + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "Dmic" +}; + +static int rt286_adc_values[] = { + 0, 5, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, RT286_MIXER_IN1, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_VALUE_ENUM("ADC 0 source", rt286_adc0_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, RT286_MIXER_IN2, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_VALUE_ENUM("ADC 1 source", rt286_adc1_enum); + +/* HP-OUT source */ +static const char * const rt286_hpo_src[] = { + "Front", "Surround" +}; + +static SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, RT286_HP_OUT, + RT286_HP_SEL_SFT, rt286_hpo_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static const char * const rt286_spo_src[] = { + "Front", "Surround" +}; + +static SOC_ENUM_SINGLE_DECL(rt286_spo_enum, RT286_SPK_OUT, + RT286_SPK_SEL_SFT, rt286_spo_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +/* SPDIF source */ +static const char * const rt286_spdif_src[] = { + "PCM-IN 0", "PCM-IN 1", "SP-OUT", "PP" +}; + +static SOC_ENUM_SINGLE_DECL(rt286_spdif_enum, RT286_SPDIF, + RT286_SPDIF_SEL_SFT, rt286_spdif_src); + +static const struct snd_kcontrol_new rt286_spdif_mux = +SOC_DAPM_ENUM("SPDIF source", rt286_spdif_enum); + + +static int rt286_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt286->regmap, RT286_SET_SPK_D0, 0); + regmap_write(rt286->regmap, RT286_ENABLE_SPK, 0); + regmap_write(rt286->regmap, RT286_SPK_EAPD_HIGH, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt286->regmap, RT286_SPK_EAPD_LOW, 0); + regmap_write(rt286->regmap, RT286_DISABLE_SPK, 0); + regmap_write(rt286->regmap, RT286_SET_SPK_D3, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val, buf; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_read(rt286->regmap, RT286_HP_OUT, &buf); + val = buf & 0x8080; + switch (val) { + case 0x0: + regmap_write(rt286->regmap, RT286_UNMUTE_HPO, 0); + break; + case 0x8000: + regmap_write(rt286->regmap, RT286_UNMUTE_HPO_R, 0); + break; + case 0x0080: + regmap_write(rt286->regmap, RT286_UNMUTE_HPO_L, 0); + break; + } + if (val != 0x8080) + mdelay(100); + break; + + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt286->regmap, RT286_MUTE_HPO, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt286->regmap, RT286_SET_D0(w->reg), 0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt286->regmap, RT286_SET_D3(w->reg), 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt286->regmap, RT286_SET_DMIC1_D0, 0); + regmap_write(rt286->regmap, RT286_ENABLE_DMIC1, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt286->regmap, RT286_DISABLE_DMIC1, 0); + regmap_write(rt286->regmap, RT286_SET_DMIC1_D3, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_hp_pow_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt286->regmap, RT286_SET_HPO_D0, 0); + regmap_write(rt286->regmap, RT286_ENABLE_HPO, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt286->regmap, RT286_DISABLE_HPO, 0); + regmap_write(rt286->regmap, RT286_SET_HPO_D3, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e; + unsigned int verb, val, buf; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (rt286->pdata.cbj_en) { + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL3, 0xc000, 0x8000); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL1, 0x0030, 0x0020); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL2, 0xc000, 0x8000); + } + regmap_write(rt286->regmap, RT286_SET_D0(w->reg), 0); + rt286_update_bits(codec, AC_VERB_SET_AMP_GAIN_MUTE, + w->reg, 0, 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_update_bits(codec, AC_VERB_SET_AMP_GAIN_MUTE, + w->reg, 0, 0x7080, 0x7080); + regmap_write(rt286->regmap, RT286_SET_D3(w->reg), 0); + break; + case SND_SOC_DAPM_POST_REG: + e = (struct soc_enum *)kcontrol->private_value; + regmap_read(rt286->regmap, e->reg, &buf); + val = buf & RT286_ADC_SEL_MASK; + verb = VERB_CMD(AC_VERB_SET_CONNECT_SEL, e->reg, val); + regmap_write(rt286->regmap, verb, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_out_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int verb, val, buf; + + switch (event) { + case SND_SOC_DAPM_POST_REG: + regmap_read(rt286->regmap, e->reg, &buf); + val = buf & 0x1; + verb = VERB_CMD(AC_VERB_SET_CONNECT_SEL, e->reg, val); + regmap_write(rt286->regmap, verb, 0); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", RT286_DMIC1, 15, 0, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("DMIC2", RT286_DMIC2, 15, 0, + NULL, 0, rt286_set_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_ADC_IN1, 15, 0, + &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_ADC_IN2, 15, 0, + &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + + + /* 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), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + + /* Output Side */ + + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC 0", NULL, RT286_DAC_OUT1, + 15, 0, rt286_set_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DAC 1", NULL, RT286_DAC_OUT2, + 15, 0, rt286_set_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Output Mux */ + SND_SOC_DAPM_MUX_E("SPK Mux", SND_SOC_NOPM, 0, 0, + &rt286_spo_mux, + rt286_out_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_MUX_E("HPO Mux", SND_SOC_NOPM, 0, 0, + &rt286_hpo_mux, + rt286_out_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_SUPPLY("HP Power", SND_SOC_NOPM, + 0, 0, rt286_hp_pow_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_MUX("SPDIF Mux", SND_SOC_NOPM, 0, 0, + &rt286_spdif_mux), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", SND_SOC_NOPM, 0, 0, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surround", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_SWITCH_E("SPO", SND_SOC_NOPM, 0, 0, + &spo_enable_control, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH_E("HPO", SND_SOC_NOPM, 0, 0, + &hpo_enable_control, rt286_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", "Switch", "SPK Mux"}, + {"HPO", "Switch", "HPO Mux"}, + {"HPO", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO"}, +}; + +static int rt286_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 rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + val |= 0x4000; + break; + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt286->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_format(params)) { + /* bit 6:4 Bits per Sample */ + case SNDRV_PCM_FORMAT_S16_LE: + d_len_code = 0; + val |= (0x1 << 4); + break; + case SNDRV_PCM_FORMAT_S32_LE: + d_len_code = 2; + val |= (0x4 << 4); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + d_len_code = 1; + val |= (0x2 << 4); + break; + case SNDRV_PCM_FORMAT_S24_LE: + d_len_code = 2; + val |= (0x3 << 4); + break; + case SNDRV_PCM_FORMAT_S8: + break; + default: + return -EINVAL; + } + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x0018, d_len_code << 3); + dev_dbg(codec->dev, "format val = 0x%x\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + RT286_DAC_OUT1, 0, 0x407f, val); + else + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + RT286_ADC_IN1, 0, 0x407f, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + RT286_DAC_OUT1, 0, 0x8000, 0x0); + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + RT286_ADC_IN1, 0, 0x8000, 0x0); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq); + + if (RT286_SCLK_S_MCLK == clk_id) { + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL2, 0x0100, 0x0); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_PLL_CTRL1, 0x20, 0x20); + } else { + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL2, 0x0100, 0x0100); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_PLL_CTRL, 0x4, 0x4); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_PLL_CTRL1, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL2, 0x40, 0x40); + break; + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL2, 0x40, 0x0); + break; + case 12288000: + case 11289600: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL2, 0x8, 0x0); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CLK_DIV, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL2, 0x8, 0x8); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CLK_DIV, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt286->sys_clk = freq; + + return 0; +} + +static int rt286_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (50 == ratio) + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x1000, 0x1000); + else + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x1000, 0x0); + + + return 0; +} + +static int rt286_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) + regmap_write(rt286->regmap, RT286_SET_AUDIO_D0, 0); + break; + + case SND_SOC_BIAS_STANDBY: + regmap_write(rt286->regmap, RT286_SET_AUDIO_D3, 0); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static irqreturn_t rt286_irq(int irq, void *data) +{ + struct rt286_priv *rt286 = data; + bool hp = false; + bool mic = false; + int status = 0; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + pm_wakeup_event(&rt286->i2c->dev, 300); + + return IRQ_HANDLED; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i, ret; + + regmap_write(rt286->regmap, RT286_SET_AUDIO_D3, 0); + for (i = 0; i < RT286_POWER_REG_LEN; i++) { + regmap_write(rt286->regmap, + RT286_SET_D3(rt286_support_power_controls[i]), 0); + } + + if (!rt286->pdata.cbj_en) { + rt286_index_write(codec, RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL2, 0x0000); + rt286_index_write(codec, RT286_VENDOR_REGISTERS, + RT286_MIC1_DET_CTRL, 0x0816); + rt286_index_write(codec, RT286_VENDOR_REGISTERS, + RT286_MISC_CTRL1, 0x0000); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL1, 0xf000, 0xb000); + } else { + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL1, 0xf000, 0x5000); + } + + mdelay(10); + + if (!rt286->pdata.gpio2_en) + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT_4000, 0); + else + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT_0, 0); + + mdelay(10); + + /*Power down LDO2*/ + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, RT286_POWER_CTRL2, 0x8, 0x0); + + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + rt286->codec = codec; + + if (rt286->i2c->irq) { + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, RT286_IRQ_CTRL, 0x2, 0x2); + ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); + if (ret != 0) { + dev_err(codec->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + + return 0; +} + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_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 rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_bclk_ratio = rt286_set_bclk_ratio, +}; + +struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct regmap_config rt286_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_write = rt286_hw_write, + .reg_read = rt286_hw_read, + .readable_reg = rt286_readable_register, + .cache_type = REGCACHE_NONE, +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt286", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static struct acpi_device_id rt286_acpi_match[] = { + { "INT343A", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt286_priv *rt286; + struct device *dev = &i2c->dev; + int ret; + + rt286 = devm_kzalloc(&i2c->dev, + sizeof(struct rt286_priv), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->regmap = devm_regmap_init(dev, NULL, i2c, &rt286_regmap); + if (IS_ERR(rt286->regmap)) { + ret = PTR_ERR(rt286->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + rt286->i2c = i2c; + i2c_set_clientdata(i2c, rt286); + + if (pdata) + rt286->pdata = *pdata; + + ret = devm_snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + + return ret; +} + +struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt286", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .id_table = rt286_i2c_id, +}; + +module_i2c_driver(rt286_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 0000000..5f9dc99 --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,159 @@ +/* + * rt286.h -- RT286 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 __RT286_H__ +#define __RT286_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RT286_AUDIO_FUNCTION_GROUP 0x01 +#define RT286_DAC_OUT1 0x02 +#define RT286_DAC_OUT2 0x03 +#define RT286_ADC_IN1 0x09 +#define RT286_ADC_IN2 0x08 +#define RT286_MIXER_IN 0x0b +#define RT286_MIXER_OUT1 0x0c +#define RT286_MIXER_OUT2 0x0d +#define RT286_DMIC1 0x12 +#define RT286_DMIC2 0x13 +#define RT286_SPK_OUT 0x14 +#define RT286_MIC1 0x18 +#define RT286_LINE1 0x1a +#define RT286_BEEP 0x1d +#define RT286_SPDIF 0x1e +#define RT286_VENDOR_REGISTERS 0x20 +#define RT286_HP_OUT 0x21 +#define RT286_MIXER_IN1 0x22 +#define RT286_MIXER_IN2 0x23 + +#define RT286_CONNECTION_INDEX_MIC1 0X0 +#define RT286_CONNECTION_INDEX_DMIC 0X5 + +/* Verb commands */ +#define RT286_SET_D0(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, AC_PWRST_D0) +#define RT286_SET_D3(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, AC_PWRST_D3) + +#define RT286_SET_MIC1_24\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0x24) +#define RT286_SET_MIC1_20\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0x20) +#define RT286_SET_AUDIO_D0 RT286_SET_D0(RT286_AUDIO_FUNCTION_GROUP) +#define RT286_SET_AUDIO_D3 RT286_SET_D3(RT286_AUDIO_FUNCTION_GROUP) +#define RT286_SET_HPO_D0 RT286_SET_D0(RT286_HP_OUT) +#define RT286_SET_HPO_D3 RT286_SET_D3(RT286_HP_OUT) +#define RT286_SET_SPK_D0 RT286_SET_D0(RT286_SPK_OUT) +#define RT286_SET_SPK_D3 RT286_SET_D3(RT286_SPK_OUT) +#define RT286_SET_DMIC1_D0 RT286_SET_D0(RT286_DMIC1) +#define RT286_SET_DMIC1_D3 RT286_SET_D3(RT286_DMIC1) +#define RT286_ENABLE_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0x40) +#define RT286_DISABLE_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0) +#define RT286_ENABLE_SPK\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0x40) +#define RT286_DISABLE_SPK\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0) +#define RT286_ENABLE_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0x20) +#define RT286_DISABLE_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0) +#define RT286_SPK_EAPD_HIGH\ + VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0x2) +#define RT286_SPK_EAPD_LOW\ + VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0) +#define RT286_UNMUTE_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xb000) +#define RT286_UNMUTE_HPO_R\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0x9000) +#define RT286_UNMUTE_HPO_L\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xa000) +#define RT286_MUTE_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xb080) +#define RT286_GET_HP_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_HP_OUT, 0) +#define RT286_GET_MIC1_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_MIC1, 0) +#define RT286_SET_DMIC2_DEFAULT_4000\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0x4000) +#define RT286_SET_DMIC2_DEFAULT_0\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0) + +/* Index registers */ +#define RT286_A_BIAS_CTRL1 0x01 +#define RT286_A_BIAS_CTRL2 0x02 +#define RT286_POWER_CTRL1 0x03 +#define RT286_A_BIAS_CTRL3 0x04 +#define RT286_POWER_CTRL2 0x08 +#define RT286_I2S_CTRL1 0x09 +#define RT286_I2S_CTRL2 0x0a +#define RT286_CLK_DIV 0x0b +#define RT286_POWER_CTRL3 0x0f +#define RT286_MIC1_DET_CTRL 0x19 +#define RT286_MISC_CTRL1 0x20 +#define RT286_IRQ_CTRL 0x33 +#define RT286_PLL_CTRL1 0x49 +#define RT286_CBJ_CTRL1 0x4f +#define RT286_CBJ_CTRL2 0x50 +#define RT286_PLL_CTRL 0x63 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_M_SPK_MUX_SFT 14 +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_M_HP_MUX_SFT 14 +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +#define RT286_SCLK_S_MCLK 0 +#define RT286_SCLK_S_PLL 1 + +enum { + RT286_AIF1, + RT286_AIFS, +}; + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +#endif /* __RT286_H__ */ -- 1.8.1.1.439.g50a6b54

5 14

[alsa-devel] [PATCH] ASoC: Add support for multi register mux
by Arun Shamanna Lakshmi 21 Mar '14

21 Mar '14

Currently soc_enum structure supports only 2 registers (reg, reg2) for kcontrol. However, it is possible to have multiple registers per mux. This change allows us to control these multiple registers. Signed-off-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> --- include/sound/soc.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a00147..ddedfb4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1093,6 +1093,9 @@ struct soc_enum { unsigned int mask; const char * const *texts; const unsigned int *values; + unsigned int *regs; + unsigned int *masks; + unsigned int num_regs; }; /* codec IO */ -- 1.7.9.5

4 8

[alsa-devel] [PATCH 1/2] ASoC: fsl: make sure the regmap-mmio is actually enabled
by Xiubo Li 21 Mar '14

21 Mar '14

Since the Freescale SAI, ESAI and SPDIF are using the regmap-mmio, there needs a select of REGMAP_MMIO in the Kconfig to ensure that regmap-mmio is actually enabled. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> Cc: Nicolin Chen <Guangyu.Chen(a)freescale.com> Cc: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/Kconfig | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 07f8f14..d0914c0 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,5 +1,6 @@ config SND_SOC_FSL_SAI tristate + select REGMAP_MMIO select SND_SOC_GENERIC_DMAENGINE_PCM config SND_SOC_FSL_SSI @@ -7,9 +8,11 @@ config SND_SOC_FSL_SSI config SND_SOC_FSL_SPDIF tristate + select REGMAP_MMIO config SND_SOC_FSL_ESAI tristate + select REGMAP_MMIO config SND_SOC_FSL_UTILS tristate @@ -204,7 +207,6 @@ config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF - select REGMAP_MMIO help SoC Audio support for i.MX boards with S/PDIF Say Y if you want to add support for SoC audio on an i.MX board with -- 1.8.4

4 7

[alsa-devel] [PATCH v4] ASoC: cs42xx8: Add codec driver support for CS42448/CS42888
by Nicolin Chen 20 Mar '14

20 Mar '14

This patch adds support for the Cirrus Logic CS42448/CS42888 Audio CODEC that has six/four 24-bit AD and eight 24-bit DA converters. [ CS42448/CS42888 supports both I2C and SPI control ports. As initial patch, this patch only adds the support for I2C. ] Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> Acked-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- Changelog v4: * Added CS42448 supports * Renamed the driver to a gernal one -- cs42xx8 v3: * Refined document at 'clocks' section. v2: * Separated driver into I2C and main part drivers. * Added error out if failed to get clock. * Enabled Auto-mute function as default. * Disabled runtime-pm in remove(). * Added return value checking for regcache_sync(). * Fixed some trival parts like 'const' as Lars mentioned. .../devicetree/bindings/sound/cs42xx8.txt | 28 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42xx8-i2c.c | 64 +++ sound/soc/codecs/cs42xx8.c | 602 +++++++++++++++++++++ sound/soc/codecs/cs42xx8.h | 238 ++++++++ 6 files changed, 946 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs42xx8.txt create mode 100644 sound/soc/codecs/cs42xx8-i2c.c create mode 100644 sound/soc/codecs/cs42xx8.c create mode 100644 sound/soc/codecs/cs42xx8.h diff --git a/Documentation/devicetree/bindings/sound/cs42xx8.txt b/Documentation/devicetree/bindings/sound/cs42xx8.txt new file mode 100644 index 0000000..f631fbc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42xx8.txt @@ -0,0 +1,28 @@ +CS42448/CS42888 audio CODEC + +Required properties: + + - compatible : must contain one of "cirrus,cs42448" and "cirrus,cs42888" + + - reg : the I2C address of the device for I2C + + - clocks : a list of phandles + clock-specifiers, one for each entry in + clock-names + + - clock-names : must contain "mclk" + + - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +Example: + +codec: cs42888@48 { + compatible = "cirrus,cs42888"; + reg = <0x48>; + clocks = <&codec_mclk 0>; + clock-names = "mclk"; + VA-supply = <&reg_audio>; + VD-supply = <&reg_audio>; + VLS-supply = <&reg_audio>; + VLC-supply = <&reg_audio>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1a8ff1e..f0e8401 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,6 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI + select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if I2C select SND_SOC_DA7213 if I2C @@ -304,6 +305,15 @@ config SND_SOC_CS4271 tristate "Cirrus Logic CS4271 CODEC" depends on SND_SOC_I2C_AND_SPI +config SND_SOC_CS42XX8 + tristate + +config SND_SOC_CS42XX8_I2C + tristate "Cirrus Logic CS42448/CS42888 CODEC (I2C)" + depends on I2C + select SND_SOC_CS42XX8 + select REGMAP_I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 73df822..3c4d275 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -30,6 +30,8 @@ snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o +snd-soc-cs42xx8-objs := cs42xx8.o +snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -179,6 +181,8 @@ obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o +obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o +obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs42xx8-i2c.c b/sound/soc/codecs/cs42xx8-i2c.c new file mode 100644 index 0000000..657dce2 --- /dev/null +++ b/sound/soc/codecs/cs42xx8-i2c.c @@ -0,0 +1,64 @@ +/* + * Cirrus Logic CS42448/CS42888 Audio CODEC DAI I2C driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <sound/soc.h> + +#include "cs42xx8.h" + +static int cs42xx8_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + u32 ret = cs42xx8_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &cs42xx8_regmap_config)); + if (ret) + return ret; + + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + + return 0; +} + +static int cs42xx8_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + pm_runtime_disable(&i2c->dev); + + return 0; +} + +static struct i2c_device_id cs42xx8_i2c_id[] = { + {"cs42448", (kernel_ulong_t)&cs42448_data}, + {"cs42888", (kernel_ulong_t)&cs42888_data}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42xx8_i2c_id); + +static struct i2c_driver cs42xx8_i2c_driver = { + .driver = { + .name = "cs42xx8", + .owner = THIS_MODULE, + .pm = &cs42xx8_pm, + }, + .probe = cs42xx8_i2c_probe, + .remove = cs42xx8_i2c_remove, + .id_table = cs42xx8_i2c_id, +}; + +module_i2c_driver(cs42xx8_i2c_driver); + +MODULE_DESCRIPTION("Cirrus Logic CS42448/CS42888 ALSA SoC Codec I2C Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c new file mode 100644 index 0000000..082299a --- /dev/null +++ b/sound/soc/codecs/cs42xx8.c @@ -0,0 +1,602 @@ +/* + * Cirrus Logic CS42448/CS42888 Audio CODEC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "cs42xx8.h" + +#define CS42XX8_NUM_SUPPLIES 4 +static const char *const cs42xx8_supply_names[CS42XX8_NUM_SUPPLIES] = { + "VA", + "VD", + "VLS", + "VLC", +}; + +#define CS42XX8_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct cs42xx8_priv { + struct regulator_bulk_data supplies[CS42XX8_NUM_SUPPLIES]; + const struct cs42xx8_driver_data *drvdata; + struct regmap *regmap; + struct clk *clk; + + bool slave_mode; + unsigned long sysclk; +}; + +/* -127.5dB to 0dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +/* -64dB to 24dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(adc_tlv, -6400, 50, 0); + +static const char *const cs42xx8_adc_single[] = { "Differential", "Single-Ended" }; +static const char *const cs42xx8_szc[] = { "Immediate Change", "Zero Cross", + "Soft Ramp", "Soft Ramp on Zero Cross" }; + +static const struct soc_enum adc1_single_enum = + SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 4, 2, cs42xx8_adc_single); +static const struct soc_enum adc2_single_enum = + SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 3, 2, cs42xx8_adc_single); +static const struct soc_enum adc3_single_enum = + SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 2, 2, cs42xx8_adc_single); +static const struct soc_enum dac_szc_enum = + SOC_ENUM_SINGLE(CS42XX8_TXCTL, 5, 4, cs42xx8_szc); +static const struct soc_enum adc_szc_enum = + SOC_ENUM_SINGLE(CS42XX8_TXCTL, 0, 4, cs42xx8_szc); + +static const struct snd_kcontrol_new cs42xx8_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", CS42XX8_VOLAOUT1, + CS42XX8_VOLAOUT2, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", CS42XX8_VOLAOUT3, + CS42XX8_VOLAOUT4, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC3 Playback Volume", CS42XX8_VOLAOUT5, + CS42XX8_VOLAOUT6, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC4 Playback Volume", CS42XX8_VOLAOUT7, + CS42XX8_VOLAOUT8, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_S_TLV("ADC1 Capture Volume", CS42XX8_VOLAIN1, + CS42XX8_VOLAIN2, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE_R_S_TLV("ADC2 Capture Volume", CS42XX8_VOLAIN3, + CS42XX8_VOLAIN4, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("DAC1 Invert Switch", CS42XX8_DACINV, 0, 1, 1, 0), + SOC_DOUBLE("DAC2 Invert Switch", CS42XX8_DACINV, 2, 3, 1, 0), + SOC_DOUBLE("DAC3 Invert Switch", CS42XX8_DACINV, 4, 5, 1, 0), + SOC_DOUBLE("DAC4 Invert Switch", CS42XX8_DACINV, 6, 7, 1, 0), + SOC_DOUBLE("ADC1 Invert Switch", CS42XX8_ADCINV, 0, 1, 1, 0), + SOC_DOUBLE("ADC2 Invert Switch", CS42XX8_ADCINV, 2, 3, 1, 0), + SOC_SINGLE("ADC High-Pass Filter Switch", CS42XX8_ADCCTL, 7, 1, 1), + SOC_SINGLE("DAC De-emphasis Switch", CS42XX8_ADCCTL, 5, 1, 0), + SOC_ENUM("ADC1 Single Ended Mode Switch", adc1_single_enum), + SOC_ENUM("ADC2 Single Ended Mode Switch", adc2_single_enum), + SOC_SINGLE("DAC Single Volume Control Switch", CS42XX8_TXCTL, 7, 1, 0), + SOC_ENUM("DAC Soft Ramp & Zero Cross Control Switch", dac_szc_enum), + SOC_SINGLE("DAC Auto Mute Switch", CS42XX8_TXCTL, 4, 1, 0), + SOC_SINGLE("Mute ADC Serial Port Switch", CS42XX8_TXCTL, 3, 1, 0), + SOC_SINGLE("ADC Single Volume Control Switch", CS42XX8_TXCTL, 2, 1, 0), + SOC_ENUM("ADC Soft Ramp & Zero Cross Control Switch", adc_szc_enum), +}; + +static const struct snd_kcontrol_new cs42xx8_adc3_snd_controls[] = { + SOC_DOUBLE_R_S_TLV("ADC3 Capture Volume", CS42XX8_VOLAIN5, + CS42XX8_VOLAIN6, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("ADC3 Invert Switch", CS42XX8_ADCINV, 4, 5, 1, 0), + SOC_ENUM("ADC3 Single Ended Mode Switch", adc3_single_enum), +}; + +static const struct snd_soc_dapm_widget cs42xx8_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC1", "Playback", CS42XX8_PWRCTL, 1, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", CS42XX8_PWRCTL, 2, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", CS42XX8_PWRCTL, 3, 1), + SND_SOC_DAPM_DAC("DAC4", "Playback", CS42XX8_PWRCTL, 4, 1), + + SND_SOC_DAPM_OUTPUT("AOUT1L"), + SND_SOC_DAPM_OUTPUT("AOUT1R"), + SND_SOC_DAPM_OUTPUT("AOUT2L"), + SND_SOC_DAPM_OUTPUT("AOUT2R"), + SND_SOC_DAPM_OUTPUT("AOUT3L"), + SND_SOC_DAPM_OUTPUT("AOUT3R"), + SND_SOC_DAPM_OUTPUT("AOUT4L"), + SND_SOC_DAPM_OUTPUT("AOUT4R"), + + SND_SOC_DAPM_ADC("ADC1", "Capture", CS42XX8_PWRCTL, 5, 1), + SND_SOC_DAPM_ADC("ADC2", "Capture", CS42XX8_PWRCTL, 6, 1), + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + + SND_SOC_DAPM_SUPPLY("PWR", CS42XX8_PWRCTL, 0, 1, NULL, 0), +}; + +static const struct snd_soc_dapm_widget cs42xx8_adc3_dapm_widgets[] = { + SND_SOC_DAPM_ADC("ADC3", "Capture", CS42XX8_PWRCTL, 7, 1), + + SND_SOC_DAPM_INPUT("AIN3L"), + SND_SOC_DAPM_INPUT("AIN3R"), +}; + +static const struct snd_soc_dapm_route cs42xx8_dapm_routes[] = { + /* Playback */ + { "AOUT1L", NULL, "DAC1" }, + { "AOUT1R", NULL, "DAC1" }, + { "DAC1", NULL, "PWR" }, + + { "AOUT2L", NULL, "DAC2" }, + { "AOUT2R", NULL, "DAC2" }, + { "DAC2", NULL, "PWR" }, + + { "AOUT3L", NULL, "DAC3" }, + { "AOUT3R", NULL, "DAC3" }, + { "DAC3", NULL, "PWR" }, + + { "AOUT4L", NULL, "DAC4" }, + { "AOUT4R", NULL, "DAC4" }, + { "DAC4", NULL, "PWR" }, + + /* Capture */ + { "ADC1", NULL, "AIN1L" }, + { "ADC1", NULL, "AIN1R" }, + { "ADC1", NULL, "PWR" }, + + { "ADC2", NULL, "AIN2L" }, + { "ADC2", NULL, "AIN2R" }, + { "ADC2", NULL, "PWR" }, +}; + +static const struct snd_soc_dapm_route cs42xx8_adc3_dapm_routes[] = { + /* Capture */ + { "ADC3", NULL, "AIN3L" }, + { "ADC3", NULL, "AIN3R" }, + { "ADC3", NULL, "PWR" }, +}; + +struct cs42xx8_ratios { + unsigned int ratio; + unsigned char speed; + unsigned char mclk; +}; + +static const struct cs42xx8_ratios cs42xx8_ratios[] = { + { 64, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_256(4) }, + { 96, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_384(4) }, + { 128, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_512(4) }, + { 192, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_768(4) }, + { 256, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_256(1) }, + { 384, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_384(1) }, + { 512, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_512(1) }, + { 768, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_768(1) }, + { 1024, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_1024(1) } +}; + +static int cs42xx8_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + + cs42xx8->sysclk = freq; + + return 0; +} + +static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + u32 val; + + /* Set DAI format */ + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + val = CS42XX8_INTF_DAC_DIF_LEFTJ | CS42XX8_INTF_ADC_DIF_LEFTJ; + break; + case SND_SOC_DAIFMT_I2S: + val = CS42XX8_INTF_DAC_DIF_I2S | CS42XX8_INTF_ADC_DIF_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = CS42XX8_INTF_DAC_DIF_RIGHTJ | CS42XX8_INTF_ADC_DIF_RIGHTJ; + break; + default: + dev_err(codec->dev, "unsupported dai format\n"); + return -EINVAL; + } + + regmap_update_bits(cs42xx8->regmap, CS42XX8_INTF, + CS42XX8_INTF_DAC_DIF_MASK | + CS42XX8_INTF_ADC_DIF_MASK, val); + + /* Set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs42xx8->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs42xx8->slave_mode = false; + break; + default: + dev_err(codec->dev, "unsupported master/slave mode\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42xx8_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 cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 ratio = cs42xx8->sysclk / params_rate(params); + u32 i, fm, val, mask; + + for (i = 0; i < ARRAY_SIZE(cs42xx8_ratios); i++) { + if (cs42xx8_ratios[i].ratio == ratio) + break; + } + + if (i == ARRAY_SIZE(cs42xx8_ratios)) { + dev_err(codec->dev, "unsupported sysclk ratio\n"); + return -EINVAL; + } + + mask = CS42XX8_FUNCMOD_MFREQ_MASK; + val = cs42xx8_ratios[i].mclk; + + fm = cs42xx8->slave_mode ? CS42XX8_FM_AUTO : cs42xx8_ratios[i].speed; + + regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD, + CS42XX8_FUNCMOD_xC_FM_MASK(tx) | mask, + CS42XX8_FUNCMOD_xC_FM(tx, fm) | val); + + return 0; +} + +static int cs42xx8_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(cs42xx8->regmap, CS42XX8_DACMUTE, + CS42XX8_DACMUTE_ALL, mute ? CS42XX8_DACMUTE_ALL : 0); + + return 0; +} + +static const struct snd_soc_dai_ops cs42xx8_dai_ops = { + .set_fmt = cs42xx8_set_dai_fmt, + .set_sysclk = cs42xx8_set_dai_sysclk, + .hw_params = cs42xx8_hw_params, + .digital_mute = cs42xx8_digital_mute, +}; + +static struct snd_soc_dai_driver cs42xx8_dai = { + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42XX8_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42XX8_FORMATS, + }, + .ops = &cs42xx8_dai_ops, +}; + +static const struct reg_default cs42xx8_reg[] = { + { 0x01, 0x01 }, /* Chip I.D. and Revision Register */ + { 0x02, 0x00 }, /* Power Control */ + { 0x03, 0xF0 }, /* Functional Mode */ + { 0x04, 0x46 }, /* Interface Formats */ + { 0x05, 0x00 }, /* ADC Control & DAC De-Emphasis */ + { 0x06, 0x10 }, /* Transition Control */ + { 0x07, 0x00 }, /* DAC Channel Mute */ + { 0x08, 0x00 }, /* Volume Control AOUT1 */ + { 0x09, 0x00 }, /* Volume Control AOUT2 */ + { 0x0a, 0x00 }, /* Volume Control AOUT3 */ + { 0x0b, 0x00 }, /* Volume Control AOUT4 */ + { 0x0c, 0x00 }, /* Volume Control AOUT5 */ + { 0x0d, 0x00 }, /* Volume Control AOUT6 */ + { 0x0e, 0x00 }, /* Volume Control AOUT7 */ + { 0x0f, 0x00 }, /* Volume Control AOUT8 */ + { 0x10, 0x00 }, /* DAC Channel Invert */ + { 0x11, 0x00 }, /* Volume Control AIN1 */ + { 0x12, 0x00 }, /* Volume Control AIN2 */ + { 0x13, 0x00 }, /* Volume Control AIN3 */ + { 0x14, 0x00 }, /* Volume Control AIN4 */ + { 0x15, 0x00 }, /* Volume Control AIN5 */ + { 0x16, 0x00 }, /* Volume Control AIN6 */ + { 0x17, 0x00 }, /* ADC Channel Invert */ + { 0x18, 0x00 }, /* Status Control */ + { 0x1a, 0x00 }, /* Status Mask */ + { 0x1b, 0x00 }, /* MUTEC Pin Control */ +}; + +static bool cs42xx8_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42XX8_STATUS: + return true; + default: + return false; + } +} + +static bool cs42xx8_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42XX8_CHIPID: + case CS42XX8_STATUS: + return false; + default: + return true; + } +} + +const struct regmap_config cs42xx8_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42XX8_LASTREG, + .reg_defaults = cs42xx8_reg, + .num_reg_defaults = ARRAY_SIZE(cs42xx8_reg), + .volatile_reg = cs42xx8_volatile_register, + .writeable_reg = cs42xx8_writeable_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(cs42xx8_regmap_config); + +static int cs42xx8_codec_probe(struct snd_soc_codec *codec) +{ + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + + switch (cs42xx8->drvdata->num_adcs) { + case 3: + snd_soc_add_codec_controls(codec, cs42xx8_adc3_snd_controls, + ARRAY_SIZE(cs42xx8_adc3_snd_controls)); + snd_soc_dapm_new_controls(dapm, cs42xx8_adc3_dapm_widgets, + ARRAY_SIZE(cs42xx8_adc3_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, cs42xx8_adc3_dapm_routes, + ARRAY_SIZE(cs42xx8_adc3_dapm_routes)); + break; + default: + break; + } + + /* Mute all DAC channels */ + regmap_write(cs42xx8->regmap, CS42XX8_DACMUTE, CS42XX8_DACMUTE_ALL); + + return 0; +} + +static const struct snd_soc_codec_driver cs42xx8_driver = { + .probe = cs42xx8_codec_probe, + .idle_bias_off = true, + + .controls = cs42xx8_snd_controls, + .num_controls = ARRAY_SIZE(cs42xx8_snd_controls), + .dapm_widgets = cs42xx8_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42xx8_dapm_widgets), + .dapm_routes = cs42xx8_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs42xx8_dapm_routes), +}; + +const struct cs42xx8_driver_data cs42448_data = { + .name = "cs42448", + .num_adcs = 3, +}; +EXPORT_SYMBOL_GPL(cs42448_data); + +const struct cs42xx8_driver_data cs42888_data = { + .name = "cs42888", + .num_adcs = 2, +}; +EXPORT_SYMBOL_GPL(cs42888_data); + +const struct of_device_id cs42xx8_of_match[] = { + { .compatible = "cirrus,cs42448", .data = &cs42448_data, }, + { .compatible = "cirrus,cs42888", .data = &cs42888_data, }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, cs42xx8_of_match); +EXPORT_SYMBOL_GPL(cs42xx8_of_match); + +int cs42xx8_probe(struct device *dev, struct regmap *regmap) +{ + const struct of_device_id *of_id = of_match_device(cs42xx8_of_match, dev); + struct cs42xx8_priv *cs42xx8; + int ret, val, i; + + cs42xx8 = devm_kzalloc(dev, sizeof(*cs42xx8), GFP_KERNEL); + if (cs42xx8 == NULL) + return -ENOMEM; + + dev_set_drvdata(dev, cs42xx8); + + if (of_id) + cs42xx8->drvdata = of_id->data; + + if (!cs42xx8->drvdata) { + dev_err(dev, "failed to find driver data\n"); + return -EINVAL; + } + + cs42xx8->clk = devm_clk_get(dev, "mclk"); + if (IS_ERR(cs42xx8->clk)) { + dev_err(dev, "failed to get the clock: %ld\n", + PTR_ERR(cs42xx8->clk)); + return -EINVAL; + } + + cs42xx8->sysclk = clk_get_rate(cs42xx8->clk); + + for (i = 0; i < ARRAY_SIZE(cs42xx8->supplies); i++) + cs42xx8->supplies[i].supply = cs42xx8_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, + ARRAY_SIZE(cs42xx8->supplies), cs42xx8->supplies); + if (ret) { + dev_err(dev, "failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + /* Make sure hardware reset done */ + msleep(5); + + cs42xx8->regmap = regmap; + if (IS_ERR(cs42xx8->regmap)) { + ret = PTR_ERR(cs42xx8->regmap); + dev_err(dev, "failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + /* + * We haven't marked the chip revision as volatile due to + * sharing a register with the right input volume; explicitly + * bypass the cache to read it. + */ + regcache_cache_bypass(cs42xx8->regmap, true); + + /* Validate the chip ID */ + regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); + if (val < 0) { + dev_err(dev, "failed to get device ID: %x", val); + ret = -EINVAL; + goto err_enable; + } + + /* The top four bits of the chip ID should be 0000 */ + if ((val & CS42XX8_CHIPID_CHIP_ID_MASK) != 0x00) { + dev_err(dev, "unmatched chip ID: %d\n", + val & CS42XX8_CHIPID_CHIP_ID_MASK); + ret = -EINVAL; + goto err_enable; + } + + dev_info(dev, "found device, revision %X\n", + val & CS42XX8_CHIPID_REV_ID_MASK); + + regcache_cache_bypass(cs42xx8->regmap, false); + + cs42xx8_dai.name = cs42xx8->drvdata->name; + + /* Each adc supports stereo input */ + cs42xx8_dai.capture.channels_max = cs42xx8->drvdata->num_adcs * 2; + + ret = snd_soc_register_codec(dev, &cs42xx8_driver, &cs42xx8_dai, 1); + if (ret) { + dev_err(dev, "failed to register codec:%d\n", ret); + goto err_enable; + } + + regcache_cache_only(cs42xx8->regmap, true); + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + + return ret; +} +EXPORT_SYMBOL_GPL(cs42xx8_probe); + +#ifdef CONFIG_PM_RUNTIME +static int cs42xx8_runtime_resume(struct device *dev) +{ + struct cs42xx8_priv *cs42xx8 = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(cs42xx8->clk); + if (ret) { + dev_err(dev, "failed to enable mclk: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + goto err_clk; + } + + /* Make sure hardware reset done */ + msleep(5); + + regcache_cache_only(cs42xx8->regmap, false); + + ret = regcache_sync(cs42xx8->regmap); + if (ret) { + dev_err(dev, "failed to sync regmap: %d\n", ret); + goto err_bulk; + } + + return 0; + +err_bulk: + regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); +err_clk: + clk_disable_unprepare(cs42xx8->clk); + + return ret; +} + +static int cs42xx8_runtime_suspend(struct device *dev) +{ + struct cs42xx8_priv *cs42xx8 = dev_get_drvdata(dev); + + regcache_cache_only(cs42xx8->regmap, true); + + regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + + clk_disable_unprepare(cs42xx8->clk); + + return 0; +} +#endif + +const struct dev_pm_ops cs42xx8_pm = { + SET_RUNTIME_PM_OPS(cs42xx8_runtime_suspend, cs42xx8_runtime_resume, NULL) +}; +EXPORT_SYMBOL_GPL(cs42xx8_pm); + +MODULE_DESCRIPTION("Cirrus Logic CS42448/CS42888 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42xx8.h b/sound/soc/codecs/cs42xx8.h new file mode 100644 index 0000000..da0b94a --- /dev/null +++ b/sound/soc/codecs/cs42xx8.h @@ -0,0 +1,238 @@ +/* + * cs42xx8.h - Cirrus Logic CS42448/CS42888 Audio CODEC driver header file + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _CS42XX8_H +#define _CS42XX8_H + +struct cs42xx8_driver_data { + char name[32]; + int num_adcs; +}; + +extern const struct dev_pm_ops cs42xx8_pm; +extern const struct cs42xx8_driver_data cs42448_data; +extern const struct cs42xx8_driver_data cs42888_data; +extern const struct regmap_config cs42xx8_regmap_config; +int cs42xx8_probe(struct device *dev, struct regmap *regmap); + +/* CS42888 register map */ +#define CS42XX8_CHIPID 0x01 /* Chip ID */ +#define CS42XX8_PWRCTL 0x02 /* Power Control */ +#define CS42XX8_FUNCMOD 0x03 /* Functional Mode */ +#define CS42XX8_INTF 0x04 /* Interface Formats */ +#define CS42XX8_ADCCTL 0x05 /* ADC Control */ +#define CS42XX8_TXCTL 0x06 /* Transition Control */ +#define CS42XX8_DACMUTE 0x07 /* DAC Mute Control */ +#define CS42XX8_VOLAOUT1 0x08 /* Volume Control AOUT1 */ +#define CS42XX8_VOLAOUT2 0x09 /* Volume Control AOUT2 */ +#define CS42XX8_VOLAOUT3 0x0A /* Volume Control AOUT3 */ +#define CS42XX8_VOLAOUT4 0x0B /* Volume Control AOUT4 */ +#define CS42XX8_VOLAOUT5 0x0C /* Volume Control AOUT5 */ +#define CS42XX8_VOLAOUT6 0x0D /* Volume Control AOUT6 */ +#define CS42XX8_VOLAOUT7 0x0E /* Volume Control AOUT7 */ +#define CS42XX8_VOLAOUT8 0x0F /* Volume Control AOUT8 */ +#define CS42XX8_DACINV 0x10 /* DAC Channel Invert */ +#define CS42XX8_VOLAIN1 0x11 /* Volume Control AIN1 */ +#define CS42XX8_VOLAIN2 0x12 /* Volume Control AIN2 */ +#define CS42XX8_VOLAIN3 0x13 /* Volume Control AIN3 */ +#define CS42XX8_VOLAIN4 0x14 /* Volume Control AIN4 */ +#define CS42XX8_VOLAIN5 0x15 /* Volume Control AIN5 */ +#define CS42XX8_VOLAIN6 0x16 /* Volume Control AIN6 */ +#define CS42XX8_ADCINV 0x17 /* ADC Channel Invert */ +#define CS42XX8_STATUSCTL 0x18 /* Status Control */ +#define CS42XX8_STATUS 0x19 /* Status */ +#define CS42XX8_STATUSM 0x1A /* Status Mask */ +#define CS42XX8_MUTEC 0x1B /* MUTEC Pin Control */ + +#define CS42XX8_FIRSTREG CS42XX8_CHIPID +#define CS42XX8_LASTREG CS42XX8_MUTEC +#define CS42XX8_NUMREGS (CS42XX8_LASTREG - CS42XX8_FIRSTREG + 1) +#define CS42XX8_I2C_INCR 0x80 + +/* Chip I.D. and Revision Register (Address 01h) */ +#define CS42XX8_CHIPID_CHIP_ID_MASK 0xF0 +#define CS42XX8_CHIPID_REV_ID_MASK 0x0F + +/* Power Control (Address 02h) */ +#define CS42XX8_PWRCTL_PDN_ADC3_SHIFT 7 +#define CS42XX8_PWRCTL_PDN_ADC3_MASK (1 << CS42XX8_PWRCTL_PDN_ADC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC3 (1 << CS42XX8_PWRCTL_PDN_ADC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC2_SHIFT 6 +#define CS42XX8_PWRCTL_PDN_ADC2_MASK (1 << CS42XX8_PWRCTL_PDN_ADC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC2 (1 << CS42XX8_PWRCTL_PDN_ADC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC1_SHIFT 5 +#define CS42XX8_PWRCTL_PDN_ADC1_MASK (1 << CS42XX8_PWRCTL_PDN_ADC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC1 (1 << CS42XX8_PWRCTL_PDN_ADC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC4_SHIFT 4 +#define CS42XX8_PWRCTL_PDN_DAC4_MASK (1 << CS42XX8_PWRCTL_PDN_DAC4_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC4 (1 << CS42XX8_PWRCTL_PDN_DAC4_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC3_SHIFT 3 +#define CS42XX8_PWRCTL_PDN_DAC3_MASK (1 << CS42XX8_PWRCTL_PDN_DAC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC3 (1 << CS42XX8_PWRCTL_PDN_DAC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC2_SHIFT 2 +#define CS42XX8_PWRCTL_PDN_DAC2_MASK (1 << CS42XX8_PWRCTL_PDN_DAC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC2 (1 << CS42XX8_PWRCTL_PDN_DAC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC1_SHIFT 1 +#define CS42XX8_PWRCTL_PDN_DAC1_MASK (1 << CS42XX8_PWRCTL_PDN_DAC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC1 (1 << CS42XX8_PWRCTL_PDN_DAC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_SHIFT 0 +#define CS42XX8_PWRCTL_PDN_MASK (1 << CS42XX8_PWRCTL_PDN_SHIFT) +#define CS42XX8_PWRCTL_PDN (1 << CS42XX8_PWRCTL_PDN_SHIFT) + +/* Functional Mode (Address 03h) */ +#define CS42XX8_FUNCMOD_DAC_FM_SHIFT 6 +#define CS42XX8_FUNCMOD_DAC_FM_WIDTH 2 +#define CS42XX8_FUNCMOD_DAC_FM_MASK (((1 << CS42XX8_FUNCMOD_DAC_FM_WIDTH) - 1) << CS42XX8_FUNCMOD_DAC_FM_SHIFT) +#define CS42XX8_FUNCMOD_DAC_FM(v) ((v) << CS42XX8_FUNCMOD_DAC_FM_SHIFT) +#define CS42XX8_FUNCMOD_ADC_FM_SHIFT 4 +#define CS42XX8_FUNCMOD_ADC_FM_WIDTH 2 +#define CS42XX8_FUNCMOD_ADC_FM_MASK (((1 << CS42XX8_FUNCMOD_ADC_FM_WIDTH) - 1) << CS42XX8_FUNCMOD_ADC_FM_SHIFT) +#define CS42XX8_FUNCMOD_ADC_FM(v) ((v) << CS42XX8_FUNCMOD_ADC_FM_SHIFT) +#define CS42XX8_FUNCMOD_xC_FM_MASK(x) ((x) ? CS42XX8_FUNCMOD_DAC_FM_MASK : CS42XX8_FUNCMOD_ADC_FM_MASK) +#define CS42XX8_FUNCMOD_xC_FM(x, v) ((x) ? CS42XX8_FUNCMOD_DAC_FM(v) : CS42XX8_FUNCMOD_ADC_FM(v)) +#define CS42XX8_FUNCMOD_MFREQ_SHIFT 1 +#define CS42XX8_FUNCMOD_MFREQ_WIDTH 3 +#define CS42XX8_FUNCMOD_MFREQ_MASK (((1 << CS42XX8_FUNCMOD_MFREQ_WIDTH) - 1) << CS42XX8_FUNCMOD_MFREQ_SHIFT) +#define CS42XX8_FUNCMOD_MFREQ_256(s) ((0 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_384(s) ((1 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_512(s) ((2 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_768(s) ((3 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_1024(s) ((4 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) + +#define CS42XX8_FM_SINGLE 0 +#define CS42XX8_FM_DOUBLE 1 +#define CS42XX8_FM_QUAD 2 +#define CS42XX8_FM_AUTO 3 + +/* Interface Formats (Address 04h) */ +#define CS42XX8_INTF_FREEZE_SHIFT 7 +#define CS42XX8_INTF_FREEZE_MASK (1 << CS42XX8_INTF_FREEZE_SHIFT) +#define CS42XX8_INTF_FREEZE (1 << CS42XX8_INTF_FREEZE_SHIFT) +#define CS42XX8_INTF_AUX_DIF_SHIFT 6 +#define CS42XX8_INTF_AUX_DIF_MASK (1 << CS42XX8_INTF_AUX_DIF_SHIFT) +#define CS42XX8_INTF_AUX_DIF (1 << CS42XX8_INTF_AUX_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_SHIFT 3 +#define CS42XX8_INTF_DAC_DIF_WIDTH 3 +#define CS42XX8_INTF_DAC_DIF_MASK (((1 << CS42XX8_INTF_DAC_DIF_WIDTH) - 1) << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_LEFTJ (0 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_I2S (1 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_RIGHTJ (2 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_ONELINE_20 (4 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (6 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_TDM (7 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_SHIFT 0 +#define CS42XX8_INTF_ADC_DIF_WIDTH 3 +#define CS42XX8_INTF_ADC_DIF_MASK (((1 << CS42XX8_INTF_ADC_DIF_WIDTH) - 1) << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_LEFTJ (0 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_I2S (1 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_RIGHTJ (2 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_ONELINE_20 (4 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (6 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_TDM (7 << CS42XX8_INTF_ADC_DIF_SHIFT) + +/* ADC Control & DAC De-Emphasis (Address 05h) */ +#define CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 +#define CS42XX8_ADCCTL_ADC_HPF_FREEZE_MASK (1 << CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42XX8_ADCCTL_ADC_HPF_FREEZE (1 << CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42XX8_ADCCTL_DAC_DEM_SHIFT 5 +#define CS42XX8_ADCCTL_DAC_DEM_MASK (1 << CS42XX8_ADCCTL_DAC_DEM_SHIFT) +#define CS42XX8_ADCCTL_DAC_DEM (1 << CS42XX8_ADCCTL_DAC_DEM_SHIFT) +#define CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT 4 +#define CS42XX8_ADCCTL_ADC1_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC1_SINGLE (1 << CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT 3 +#define CS42XX8_ADCCTL_ADC2_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC2_SINGLE (1 << CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT 2 +#define CS42XX8_ADCCTL_ADC3_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC3_SINGLE (1 << CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_AIN5_MUX_SHIFT 1 +#define CS42XX8_ADCCTL_AIN5_MUX_MASK (1 << CS42XX8_ADCCTL_AIN5_MUX_SHIFT) +#define CS42XX8_ADCCTL_AIN5_MUX (1 << CS42XX8_ADCCTL_AIN5_MUX_SHIFT) +#define CS42XX8_ADCCTL_AIN6_MUX_SHIFT 0 +#define CS42XX8_ADCCTL_AIN6_MUX_MASK (1 << CS42XX8_ADCCTL_AIN6_MUX_SHIFT) +#define CS42XX8_ADCCTL_AIN6_MUX (1 << CS42XX8_ADCCTL_AIN6_MUX_SHIFT) + +/* Transition Control (Address 06h) */ +#define CS42XX8_TXCTL_DAC_SNGVOL_SHIFT 7 +#define CS42XX8_TXCTL_DAC_SNGVOL_MASK (1 << CS42XX8_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_DAC_SNGVOL (1 << CS42XX8_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_SHIFT 5 +#define CS42XX8_TXCTL_DAC_SZC_WIDTH 2 +#define CS42XX8_TXCTL_DAC_SZC_MASK (((1 << CS42XX8_TXCTL_DAC_SZC_WIDTH) - 1) << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_IC (0 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_ZC (1 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_SR (2 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_SRZC (3 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_AMUTE_SHIFT 4 +#define CS42XX8_TXCTL_AMUTE_MASK (1 << CS42XX8_TXCTL_AMUTE_SHIFT) +#define CS42XX8_TXCTL_AMUTE (1 << CS42XX8_TXCTL_AMUTE_SHIFT) +#define CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT 3 +#define CS42XX8_TXCTL_MUTE_ADC_SP_MASK (1 << CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42XX8_TXCTL_MUTE_ADC_SP (1 << CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42XX8_TXCTL_ADC_SNGVOL_SHIFT 2 +#define CS42XX8_TXCTL_ADC_SNGVOL_MASK (1 << CS42XX8_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_ADC_SNGVOL (1 << CS42XX8_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_SHIFT 0 +#define CS42XX8_TXCTL_ADC_SZC_MASK (((1 << CS42XX8_TXCTL_ADC_SZC_WIDTH) - 1) << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_IC (0 << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_ZC (1 << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_SR (2 << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_SRZC (3 << CS42XX8_TXCTL_ADC_SZC_SHIFT) + +/* DAC Channel Mute (Address 07h) */ +#define CS42XX8_DACMUTE_AOUT(n) (0x1 << n) +#define CS42XX8_DACMUTE_ALL 0xff + +/* Status Control (Address 18h)*/ +#define CS42XX8_STATUSCTL_INI_SHIFT 2 +#define CS42XX8_STATUSCTL_INI_WIDTH 2 +#define CS42XX8_STATUSCTL_INI_MASK (((1 << CS42XX8_STATUSCTL_INI_WIDTH) - 1) << CS42XX8_STATUSCTL_INI_SHIFT) +#define CS42XX8_STATUSCTL_INT_ACTIVE_HIGH (0 << CS42XX8_STATUSCTL_INI_SHIFT) +#define CS42XX8_STATUSCTL_INT_ACTIVE_LOW (1 << CS42XX8_STATUSCTL_INI_SHIFT) +#define CS42XX8_STATUSCTL_INT_OPEN_DRAIN (2 << CS42XX8_STATUSCTL_INI_SHIFT) + +/* Status (Address 19h)*/ +#define CS42XX8_STATUS_DAC_CLK_ERR_SHIFT 4 +#define CS42XX8_STATUS_DAC_CLK_ERR_MASK (1 << CS42XX8_STATUS_DAC_CLK_ERR_SHIFT) +#define CS42XX8_STATUS_ADC_CLK_ERR_SHIFT 3 +#define CS42XX8_STATUS_ADC_CLK_ERR_MASK (1 << CS42XX8_STATUS_ADC_CLK_ERR_SHIFT) +#define CS42XX8_STATUS_ADC3_OVFL_SHIFT 2 +#define CS42XX8_STATUS_ADC3_OVFL_MASK (1 << CS42XX8_STATUS_ADC3_OVFL_SHIFT) +#define CS42XX8_STATUS_ADC2_OVFL_SHIFT 1 +#define CS42XX8_STATUS_ADC2_OVFL_MASK (1 << CS42XX8_STATUS_ADC2_OVFL_SHIFT) +#define CS42XX8_STATUS_ADC1_OVFL_SHIFT 0 +#define CS42XX8_STATUS_ADC1_OVFL_MASK (1 << CS42XX8_STATUS_ADC1_OVFL_SHIFT) + +/* Status Mask (Address 1Ah) */ +#define CS42XX8_STATUS_DAC_CLK_ERR_M_SHIFT 4 +#define CS42XX8_STATUS_DAC_CLK_ERR_M_MASK (1 << CS42XX8_STATUS_DAC_CLK_ERR_M_SHIFT) +#define CS42XX8_STATUS_ADC_CLK_ERR_M_SHIFT 3 +#define CS42XX8_STATUS_ADC_CLK_ERR_M_MASK (1 << CS42XX8_STATUS_ADC_CLK_ERR_M_SHIFT) +#define CS42XX8_STATUS_ADC3_OVFL_M_SHIFT 2 +#define CS42XX8_STATUS_ADC3_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC3_OVFL_M_SHIFT) +#define CS42XX8_STATUS_ADC2_OVFL_M_SHIFT 1 +#define CS42XX8_STATUS_ADC2_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC2_OVFL_M_SHIFT) +#define CS42XX8_STATUS_ADC1_OVFL_M_SHIFT 0 +#define CS42XX8_STATUS_ADC1_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC1_OVFL_M_SHIFT) + +/* MUTEC Pin Control (Address 1Bh) */ +#define CS42XX8_MUTEC_MCPOLARITY_SHIFT 1 +#define CS42XX8_MUTEC_MCPOLARITY_MASK (1 << CS42XX8_MUTEC_MCPOLARITY_SHIFT) +#define CS42XX8_MUTEC_MCPOLARITY_ACTIVE_LOW (0 << CS42XX8_MUTEC_MCPOLARITY_SHIFT) +#define CS42XX8_MUTEC_MCPOLARITY_ACTIVE_HIGH (1 << CS42XX8_MUTEC_MCPOLARITY_SHIFT) +#define CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT 0 +#define CS42XX8_MUTEC_MUTEC_ACTIVE_MASK (1 << CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT) +#define CS42XX8_MUTEC_MUTEC_ACTIVE (1 << CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT) +#endif /* _CS42XX8_H */ -- 1.8.4

2 1

[alsa-devel] [PATCH v3 0/4] ASoC: simple-card: multi DAI links extension
by Jean-Francois Moine 20 Mar '14

20 Mar '14

This patch series extends the simple card driver to handle many DAI links as this exists in the Cubox audio subsystem. -v3 - remove 'Fix the reference count of device nodes' which is applied (Mark Brown) - new patch 'Simplify code' - dynamically allocate and use properties for all DAI links (Jyri Sarha and Li Xiubo) - v2 - change subject/comment about device node reference count (Mark Brown) - use a null size array instead of an implicit area for the DAI links (Li Xiubo) - update the reference count of the device node at end of probe Jean-Francois Moine (4): ASoC: simple-card: Simplify code ASoC: simple-card: dynamically allocate the DAI link and properties ASoC: simple-card: Handle many DAI links ASoC: simple-card: Add DT documentation for multi-DAI links .../devicetree/bindings/sound/simple-card.txt | 34 +++- sound/soc/generic/simple-card.c | 181 +++++++++++++-------- 2 files changed, 145 insertions(+), 70 deletions(-) -- 1.9.0

5 26

[alsa-devel] [PATCH v2] ASoC: tegra: Use flat regcache.
by Dylan Reid 20 Mar '14

20 Mar '14

When using an rbtree cache, there can be allocations the first time a register is accessed. This can cause an attempt to schedule while atomic in the case that the regmap is using a spinlock. This could be fixed by either initializing all the registers or using a flat cache. The register maps for tegra30_ahub and tegra30_i2s are dense and don't save much from using a tree so convert them to flat. Tegra30 changes tested on Norrin, Tegra20 changes compile. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Changes v1 to v2: Fix Tegra20 fix as well. sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra20_das.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra20_spdif.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 4 ++-- sound/soc/tegra/tegra30_i2s.c | 2 +- 6 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index cf5e1cf..0a59e23 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -306,7 +306,7 @@ static const struct regmap_config tegra20_ac97_regmap_config = { .readable_reg = tegra20_ac97_wr_rd_reg, .volatile_reg = tegra20_ac97_volatile_reg, .precious_reg = tegra20_ac97_precious_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static int tegra20_ac97_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index e723929..a634f13 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -128,7 +128,7 @@ static const struct regmap_config tegra20_das_regmap_config = { .max_register = LAST_REG(DAC_INPUT_DATA_CLK_SEL), .writeable_reg = tegra20_das_wr_rd_reg, .readable_reg = tegra20_das_wr_rd_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static int tegra20_das_probe(struct platform_device *pdev) diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 42c1f6b..79a9932 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -333,7 +333,7 @@ static const struct regmap_config tegra20_i2s_regmap_config = { .readable_reg = tegra20_i2s_wr_rd_reg, .volatile_reg = tegra20_i2s_volatile_reg, .precious_reg = tegra20_i2s_precious_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static int tegra20_i2s_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 8c7c102..a0ce924 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -259,7 +259,7 @@ static const struct regmap_config tegra20_spdif_regmap_config = { .readable_reg = tegra20_spdif_wr_rd_reg, .volatile_reg = tegra20_spdif_volatile_reg, .precious_reg = tegra20_spdif_precious_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static int tegra20_spdif_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d6f4c99..0db68f4 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -471,7 +471,7 @@ static const struct regmap_config tegra30_ahub_apbif_regmap_config = { .readable_reg = tegra30_ahub_apbif_wr_rd_reg, .volatile_reg = tegra30_ahub_apbif_volatile_reg, .precious_reg = tegra30_ahub_apbif_precious_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static bool tegra30_ahub_ahub_wr_rd_reg(struct device *dev, unsigned int reg) @@ -490,7 +490,7 @@ static const struct regmap_config tegra30_ahub_ahub_regmap_config = { .max_register = LAST_REG(AUDIO_RX), .writeable_reg = tegra30_ahub_ahub_wr_rd_reg, .readable_reg = tegra30_ahub_ahub_wr_rd_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static struct tegra30_ahub_soc_data soc_data_tegra30 = { diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 49ad936..f146c41 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -357,7 +357,7 @@ static const struct regmap_config tegra30_i2s_regmap_config = { .writeable_reg = tegra30_i2s_wr_rd_reg, .readable_reg = tegra30_i2s_wr_rd_reg, .volatile_reg = tegra30_i2s_volatile_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static const struct tegra30_i2s_soc_data tegra30_i2s_config = { -- 1.8.1.3.605.g02339dd

3 2

[alsa-devel] [PATCH 0/2] ASoC: sirf: Add sirf audio hub driver and use it.
by RongJun Ying 19 Mar '14

19 Mar '14

From: Rongjun Ying <rongjun.ying(a)csr.com> This patchset adds including: 1. SiRF audio hub driver 2. The internal codec and port get regmap form audio hub driver 3. Add SiRF audio hub binding document Rongjun Ying (2): ASoC: sirf: Add sirf audio hub driver for sharing same register address space Documentation: Describe the SiRF audio hub Device Tree bindings .../devicetree/bindings/sound/sirf-audio-hub.txt | 52 +++++++++++++++ sound/soc/codecs/sirf-audio-codec.c | 32 ++-------- sound/soc/sirf/Kconfig | 5 ++ sound/soc/sirf/Makefile | 2 + sound/soc/sirf/sirf-audio-hub.c | 66 ++++++++++++++++++++ sound/soc/sirf/sirf-audio-port.c | 26 +-------- 6 files changed, 131 insertions(+), 52 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-hub.txt create mode 100644 sound/soc/sirf/sirf-audio-hub.c -- 1.7.5.4

2 4

[alsa-devel] Regression caused by "ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV"
by Thomas Petazzoni 19 Mar '14

19 Mar '14

Brian, Liam, Mark, I am currently bringing up audio support on the Marvell Armada 370 ARM SoC, for which the development board uses the CS42L51 audio codec. It turns out that the commit 1d99f2436d0d1c7741d6dfd9d27b5376cdbbca40 ("ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV") causes a problem here, and reverting this commit gets things back to normal. Basically, when this commit is applied, the "PCM", "Analog" and "ADC Mixer" volume controls in alsamixer cannot be changed to any other value than 60 or 61. The volume control doesn't allow to go above 61 and below 60. If I remember correctly, at the codec level, the values configured for PCM in registers 0x10 and 0x11 are either 0x0 or 0x1 depending on whether you select 60 or 61 in alsamixer. Reverting the commit mentioned above makes alsamixer behave normally: the volume can be adjusted between 0 and 100 as expected, the values in register 0x10 and 0x11 are meaningful, and the behavior is as expected: with a volume to 0, I hear nothing in my headphones, with a volume set to 100, the volume is maximum, with a nice progressive volume increase in-between. Therefore, I believe that the commit has a problem. I haven't yet investigated where the problem is, maybe just the values passed to SOC_DOUBLE_R_SX_TLV(), or maybe in the implementation of the volume control functions themselves. I can certainly start investigating, but maybe the author of the commit will immediately realize where the problem could be. Thanks a lot! Thomas -- Thomas Petazzoni, CTO, Free Electrons Embedded Linux, Kernel and Android engineering http://free-electrons.com

4 18

[alsa-devel] [PATCH] Use jack_client_open instead of deprecated jack_client_new.
by Maarten Baert 19 Mar '14

19 Mar '14

Signed-off-by: Maarten Baert <maarten-baert(a)hotmail.com> --- jack/pcm_jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 4213708..7a8b24d 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -370,7 +370,7 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, __func__, jack_client_name, (int)strlen(jack_client_name)); } - jack->client = jack_client_new(jack_client_name); + jack->client = jack_client_open(jack_client_name, JackNoStartServer, NULL); if (jack->client == 0) { snd_pcm_jack_free(jack); -- 1.9.0

2 1

[alsa-devel] [PATCH] ALSA: compress: Pass through return value of open ops callback
by Charles Keepax 19 Mar '14

19 Mar '14

The snd_compr_open function would always return 0 even if the compressed ops open function failed, obviously this is incorrect. Looks like this was introduced by a small typo in: commit a0830dbd4e42b38aefdf3fb61ba5019a1a99ea85 ALSA: Add a reference counter to card instance This patch returns the value from the compressed op as it should. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 7a20897..7403f34 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -133,7 +133,7 @@ static int snd_compr_open(struct inode *inode, struct file *f) kfree(data); } snd_card_unref(compr->card); - return 0; + return ret; } static int snd_compr_free(struct inode *inode, struct file *f) -- 1.7.2.5

3 2

[alsa-devel] [PATCH] pcm: route: Don't handle no matching chmap as a serious error
by Takashi Iwai 19 Mar '14

19 Mar '14

When find_matching_chmap() returns an error for the non-matching chmap, the caller, snd_pcm_route_open(), also returns an error although it shouldn't be handled as the fatal error. This results in the probe error with PulseAudio and it gives no real output in the end. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_route.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/src/pcm/pcm_route.c b/src/pcm/pcm_route.c index ab17fa78be2c..ac11bdc8adfd 100644 --- a/src/pcm/pcm_route.c +++ b/src/pcm/pcm_route.c @@ -940,10 +940,8 @@ static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, snd_pcm_free_chmaps(chmaps); - if (*found_chmap == NULL) { + if (*found_chmap == NULL) SNDERR("Found no matching channel map"); - return -EINVAL; - } return 0; } -- 1.9.0

2 6

[alsa-devel] [PATCH v2 1/2] remgap: Fix possible sleep-in-atomic in regmap_bulk_write()
by Takashi Iwai 18 Mar '14

18 Mar '14

regmap deploys the spinlock for the protection when set up in fast_io mode. This may lead to sleep-in-atomic by memory allocation with GFP_KERNEL in regmap_bulk_write(). This patch fixes it by moving the allocation out of the lock. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/base/regmap/regmap.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 6a19515f8a45..6d134a3cbfd3 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -1520,12 +1520,12 @@ int regmap_bulk_write(struct regmap *map, unsigned int reg, const void *val, if (reg % map->reg_stride) return -EINVAL; - map->lock(map->lock_arg); /* * Some devices don't support bulk write, for * them we have a series of single write operations. */ if (!map->bus || map->use_single_rw) { + map->lock(map->lock_arg); for (i = 0; i < val_count; i++) { unsigned int ival; @@ -1554,24 +1554,25 @@ int regmap_bulk_write(struct regmap *map, unsigned int reg, const void *val, if (ret != 0) goto out; } +out: + map->unlock(map->lock_arg); } else { void *wval; wval = kmemdup(val, val_count * val_bytes, GFP_KERNEL); if (!wval) { - ret = -ENOMEM; dev_err(map->dev, "Error in memory allocation\n"); - goto out; + return -ENOMEM; } + map->lock(map->lock_arg); for (i = 0; i < val_count * val_bytes; i += val_bytes) map->format.parse_inplace(wval + i); ret = _regmap_raw_write(map, reg, wval, val_bytes * val_count); + map->unlock(map->lock_arg); kfree(wval); } -out: - map->unlock(map->lock_arg); return ret; } EXPORT_SYMBOL_GPL(regmap_bulk_write); -- 1.9.0

2 7

[alsa-devel] [PATCH v2] ASoC: simple-card: overwrite cpu_dai->fmt with codec_dai->fmt
by Nicolin Chen 18 Mar '14

18 Mar '14

The current simple-card driver separates the daimft for cpu_dai and codec_dai. So we might get different values for them (0x4003 and 0x1003 for example): asoc-simple-card sound-cs42888.12: cpu : 2024000.esai / 4003 / 132000000 asoc-simple-card sound-cs42888.12: codec : cs42888 / 1003 / 24576000 asoc-simple-card sound-cs42888.12: cs42888 <-> 2024000.esai mapping ok This is not allowed at all as we need to keep the DAIFMT settings identical for both the ends of the link. Thus this patch fixes it by overwriting the cpu_dai->fmt with codec_dai->fmt since we defined the DAIFMT_MASTER basing on CODEC at the first place while the other bits are same. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v2: * Keep the fmt identical for both CPU and CODEC sides. * Appended warning to binding doc. .../devicetree/bindings/sound/simple-card.txt | 6 ++++++ sound/soc/generic/simple-card.c | 20 ++++++++++++++------ 2 files changed, 20 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index b30c222..881914b1 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -43,6 +43,12 @@ Optional CPU/CODEC subnodes properties: clock node (= common clock), or "system-clock-frequency" (if system doens't support common clock) +Note: + * For 'format', 'frame-master', 'bitclock-master', 'bitclock-inversion' and + 'frame-inversion', the simple card will use the settings of CODEC for both + CPU and CODEC sides as we need to keep the settings identical for both ends + of the link. + Example: sound { diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 5dd4769..09591ab 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -157,6 +157,8 @@ static int asoc_simple_card_parse_of(struct device_node *node, struct device *dev) { struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; + struct asoc_simple_dai *codec_dai = &priv->codec_dai; + struct asoc_simple_dai *cpu_dai = &priv->cpu_dai; struct device_node *np; char *name; int ret; @@ -189,7 +191,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) ret = asoc_simple_card_sub_parse_of(np, priv->daifmt, - &priv->cpu_dai, + cpu_dai, &dai_link->cpu_of_node, &dai_link->cpu_dai_name); if (ret < 0) @@ -200,12 +202,18 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) ret = asoc_simple_card_sub_parse_of(np, priv->daifmt, - &priv->codec_dai, + codec_dai, &dai_link->codec_of_node, &dai_link->codec_dai_name); if (ret < 0) return ret; + /* + * overwrite cpu_dai->fmt as its DAIFMT_MASTER bit is based on CODEC + * while the other bits should be identical unless buggy SW/HW design. + */ + cpu_dai->fmt = codec_dai->fmt; + if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name) return -EINVAL; @@ -227,12 +235,12 @@ static int asoc_simple_card_parse_of(struct device_node *node, dev_dbg(dev, "platform : %04x\n", priv->daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", dai_link->cpu_dai_name, - priv->cpu_dai.fmt, - priv->cpu_dai.sysclk); + cpu_dai->fmt, + cpu_dai->sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", dai_link->codec_dai_name, - priv->codec_dai.fmt, - priv->codec_dai.sysclk); + codec_dai->fmt, + codec_dai->sysclk); /* * soc_bind_dai_link() will check cpu name -- 1.8.4

3 4

[alsa-devel] [PATCH v3] ASoC: cs42888: Add codec driver support
by Nicolin Chen 18 Mar '14

18 Mar '14

This patch adds support for the Cirrus Logic CS42888 Audio CODEC that has four 24-bit A/D and eight 24-bit D/A converters. [ CS42888 supports both I2C and SPI control ports. As initial patch, this patch only adds the support for I2C. ] Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v3: * Refined document at 'clocks' section. v2: * Separated driver into I2C and main part drivers. * Added error out if failed to get clock. * Enabled Auto-mute function as default. * Disabled runtime-pm in remove(). * Added return value checking for regcache_sync(). * Fixed some trival parts like 'const' as Lars mentioned. .../devicetree/bindings/sound/cs42888.txt | 28 ++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42888-i2c.c | 68 +++ sound/soc/codecs/cs42888.c | 536 +++++++++++++++++++++ sound/soc/codecs/cs42888.h | 213 ++++++++ 6 files changed, 859 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs42888.txt create mode 100644 sound/soc/codecs/cs42888-i2c.c create mode 100644 sound/soc/codecs/cs42888.c create mode 100644 sound/soc/codecs/cs42888.h diff --git a/Documentation/devicetree/bindings/sound/cs42888.txt b/Documentation/devicetree/bindings/sound/cs42888.txt new file mode 100644 index 0000000..b20bd78 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42888.txt @@ -0,0 +1,28 @@ +CS42888 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42888" + + - reg : the I2C address of the device for I2C + + - clocks : a list of phandles + clock-specifiers, one for each entry in + clock-names + + - clock-names : must contain "mclk" + + - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +Example: + +codec: cs42888@48 { + compatible = "cirrus,cs42888"; + reg = <0x48>; + clocks = <&codec_mclk 0>; + clock-names = "mclk"; + VA-supply = <&reg_audio>; + VD-supply = <&reg_audio>; + VLS-supply = <&reg_audio>; + VLC-supply = <&reg_audio>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 32d7a6f..1642578 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,6 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI + select SND_SOC_CS42888_I2C if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if I2C select SND_SOC_DA7213 if I2C @@ -303,6 +304,15 @@ config SND_SOC_CS4271 tristate "Cirrus Logic CS4271 CODEC" depends on SND_SOC_I2C_AND_SPI +config SND_SOC_CS42888 + tristate + +config SND_SOC_CS42888_I2C + tristate "Cirrus Logic CS42888 CODEC (I2C)" + depends on I2C + select SND_SOC_CS42888 + select REGMAP_I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cb46c4c..f5612fb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -30,6 +30,8 @@ snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o +snd-soc-cs42888-objs := cs42888.o +snd-soc-cs42888-i2c-objs := cs42888-i2c.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -178,6 +180,8 @@ obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o +obj-$(CONFIG_SND_SOC_CS42888) += snd-soc-cs42888.o +obj-$(CONFIG_SND_SOC_CS42888_I2C) += snd-soc-cs42888-i2c.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/cs42888-i2c.c b/sound/soc/codecs/cs42888-i2c.c new file mode 100644 index 0000000..54813bc --- /dev/null +++ b/sound/soc/codecs/cs42888-i2c.c @@ -0,0 +1,68 @@ +/* + * Cirrus Logic CS42888 Audio CODEC Digital Audio Interface (DAI) I2C driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "cs42888.h" + +static int cs42888_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + u32 ret = cs42888_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &cs42888_regmap_config)); + if (ret) + return ret; + + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + + return 0; +} + +static int cs42888_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + pm_runtime_disable(&i2c->dev); + + return 0; +} + +static struct i2c_device_id cs42888_i2c_id[] = { + {"cs42888", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42888_i2c_id); + +static struct i2c_driver cs42888_i2c_driver = { + .driver = { + .name = "cs42888", + .owner = THIS_MODULE, + .pm = &cs42888_pm, + }, + .probe = cs42888_i2c_probe, + .remove = cs42888_i2c_remove, + .id_table = cs42888_i2c_id, +}; + +module_i2c_driver(cs42888_i2c_driver); + +MODULE_DESCRIPTION("Cirrus Logic CS42888 ALSA SoC Codec I2C Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42888.c b/sound/soc/codecs/cs42888.c new file mode 100644 index 0000000..8cd3d4a --- /dev/null +++ b/sound/soc/codecs/cs42888.c @@ -0,0 +1,536 @@ +/* + * Cirrus Logic CS42888 Audio CODEC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "cs42888.h" + +#define CS42888_NUM_SUPPLIES 4 +static const char *const cs42888_supply_names[CS42888_NUM_SUPPLIES] = { + "VA", + "VD", + "VLS", + "VLC", +}; + +#define CS42888_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct cs42888_priv { + struct regulator_bulk_data supplies[CS42888_NUM_SUPPLIES]; + struct regmap *regmap; + struct clk *clk; + + bool slave_mode; + unsigned long sysclk; +}; + +/* -127.5dB to 0dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +/* -64dB to 24dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(adc_tlv, -6400, 50, 0); + +static const char *const cs42888_adc_single[] = { "Differential", "Single-Ended" }; +static const char *const cs42888_szc[] = { "Immediate Change", "Zero Cross", + "Soft Ramp", "Soft Ramp on Zero Cross" }; + +static const struct soc_enum adc1_single_enum = + SOC_ENUM_SINGLE(CS42888_ADCCTL, 4, 2, cs42888_adc_single); +static const struct soc_enum adc2_single_enum = + SOC_ENUM_SINGLE(CS42888_ADCCTL, 3, 2, cs42888_adc_single); +static const struct soc_enum dac_szc_enum = + SOC_ENUM_SINGLE(CS42888_TXCTL, 5, 4, cs42888_szc); +static const struct soc_enum adc_szc_enum = + SOC_ENUM_SINGLE(CS42888_TXCTL, 0, 4, cs42888_szc); + +static const struct snd_kcontrol_new cs42888_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", CS42888_VOLAOUT1, + CS42888_VOLAOUT2, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", CS42888_VOLAOUT3, + CS42888_VOLAOUT4, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC3 Playback Volume", CS42888_VOLAOUT5, + CS42888_VOLAOUT6, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC4 Playback Volume", CS42888_VOLAOUT7, + CS42888_VOLAOUT8, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_S_TLV("ADC1 Capture Volume", CS42888_VOLAIN1, + CS42888_VOLAIN2, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE_R_S_TLV("ADC2 Capture Volume", CS42888_VOLAIN3, + CS42888_VOLAIN4, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("DAC1 Invert Switch", CS42888_DACINV, 0, 1, 1, 0), + SOC_DOUBLE("DAC2 Invert Switch", CS42888_DACINV, 2, 3, 1, 0), + SOC_DOUBLE("DAC3 Invert Switch", CS42888_DACINV, 4, 5, 1, 0), + SOC_DOUBLE("DAC4 Invert Switch", CS42888_DACINV, 6, 7, 1, 0), + SOC_DOUBLE("ADC1 Invert Switch", CS42888_ADCINV, 0, 1, 1, 0), + SOC_DOUBLE("ADC2 Invert Switch", CS42888_ADCINV, 2, 3, 1, 0), + SOC_SINGLE("ADC High-Pass Filter Switch", CS42888_ADCCTL, 7, 1, 1), + SOC_SINGLE("DAC De-emphasis Switch", CS42888_ADCCTL, 5, 1, 0), + SOC_ENUM("ADC1 Single Ended Mode Switch", adc1_single_enum), + SOC_ENUM("ADC2 Single Ended Mode Switch", adc2_single_enum), + SOC_SINGLE("DAC Single Volume Control Switch", CS42888_TXCTL, 7, 1, 0), + SOC_ENUM("DAC Soft Ramp & Zero Cross Control Switch", dac_szc_enum), + SOC_SINGLE("DAC Auto Mute Switch", CS42888_TXCTL, 4, 1, 0), + SOC_SINGLE("Mute ADC Serial Port Switch", CS42888_TXCTL, 3, 1, 0), + SOC_SINGLE("ADC Single Volume Control Switch", CS42888_TXCTL, 2, 1, 0), + SOC_ENUM("ADC Soft Ramp & Zero Cross Control Switch", adc_szc_enum), +}; + +static const struct snd_soc_dapm_widget cs42888_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC1", "Playback", CS42888_PWRCTL, 1, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", CS42888_PWRCTL, 2, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", CS42888_PWRCTL, 3, 1), + SND_SOC_DAPM_DAC("DAC4", "Playback", CS42888_PWRCTL, 4, 1), + + SND_SOC_DAPM_OUTPUT("AOUT1L"), + SND_SOC_DAPM_OUTPUT("AOUT1R"), + SND_SOC_DAPM_OUTPUT("AOUT2L"), + SND_SOC_DAPM_OUTPUT("AOUT2R"), + SND_SOC_DAPM_OUTPUT("AOUT3L"), + SND_SOC_DAPM_OUTPUT("AOUT3R"), + SND_SOC_DAPM_OUTPUT("AOUT4L"), + SND_SOC_DAPM_OUTPUT("AOUT4R"), + + SND_SOC_DAPM_ADC("ADC1", "Capture", CS42888_PWRCTL, 5, 1), + SND_SOC_DAPM_ADC("ADC2", "Capture", CS42888_PWRCTL, 6, 1), + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + + SND_SOC_DAPM_SUPPLY("PWR", CS42888_PWRCTL, 0, 1, NULL, 0), +}; + +static const struct snd_soc_dapm_route cs42888_dapm_routes[] = { + /* Playback */ + { "AOUT1L", NULL, "DAC1" }, + { "AOUT1R", NULL, "DAC1" }, + { "DAC1", NULL, "PWR" }, + + { "AOUT2L", NULL, "DAC2" }, + { "AOUT2R", NULL, "DAC2" }, + { "DAC2", NULL, "PWR" }, + + { "AOUT3L", NULL, "DAC3" }, + { "AOUT3R", NULL, "DAC3" }, + { "DAC3", NULL, "PWR" }, + + { "AOUT4L", NULL, "DAC4" }, + { "AOUT4R", NULL, "DAC4" }, + { "DAC4", NULL, "PWR" }, + + /* Capture */ + { "ADC1", NULL, "AIN1L" }, + { "ADC1", NULL, "AIN1R" }, + { "ADC1", NULL, "PWR" }, + + { "ADC2", NULL, "AIN2L" }, + { "ADC2", NULL, "AIN2R" }, + { "ADC2", NULL, "PWR" }, +}; + +struct cs42888_ratios { + unsigned int ratio; + unsigned char speed; + unsigned char mclk; +}; + +static const struct cs42888_ratios cs42888_ratios[] = { + { 64, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_256(4) }, + { 96, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_384(4) }, + { 128, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_512(4) }, + { 192, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_768(4) }, + { 256, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_256(1) }, + { 384, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_384(1) }, + { 512, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_512(1) }, + { 768, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_768(1) }, + { 1024, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_1024(1) } +}; + +static int cs42888_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + cs42888->sysclk = freq; + + return 0; +} + +static int cs42888_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + u32 val; + + /* Set DAI format */ + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + val = CS42888_INTF_DAC_DIF_LEFTJ | CS42888_INTF_ADC_DIF_LEFTJ; + break; + case SND_SOC_DAIFMT_I2S: + val = CS42888_INTF_DAC_DIF_I2S | CS42888_INTF_ADC_DIF_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = CS42888_INTF_DAC_DIF_RIGHTJ | CS42888_INTF_ADC_DIF_RIGHTJ; + break; + default: + dev_err(codec->dev, "unsupported dai format\n"); + return -EINVAL; + } + + regmap_update_bits(cs42888->regmap, CS42888_INTF, + CS42888_INTF_DAC_DIF_MASK | + CS42888_INTF_ADC_DIF_MASK, val); + + /* Set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs42888->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs42888->slave_mode = false; + break; + default: + dev_err(codec->dev, "unsupported master/slave mode\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42888_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 cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 ratio = cs42888->sysclk / params_rate(params); + u32 i, fm, val, mask; + + for (i = 0; i < ARRAY_SIZE(cs42888_ratios); i++) { + if (cs42888_ratios[i].ratio == ratio) + break; + } + + if (i == ARRAY_SIZE(cs42888_ratios)) { + dev_err(codec->dev, "unsupported sysclk ratio\n"); + return -EINVAL; + } + + mask = CS42888_FUNCMOD_MFREQ_MASK; + val = cs42888_ratios[i].mclk; + + fm = cs42888->slave_mode ? CS42888_FM_AUTO : cs42888_ratios[i].speed; + + regmap_update_bits(cs42888->regmap, CS42888_FUNCMOD, + CS42888_FUNCMOD_xC_FM_MASK(tx) | mask, + CS42888_FUNCMOD_xC_FM(tx, fm) | val); + + return 0; +} + +static int cs42888_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(cs42888->regmap, CS42888_DACMUTE, + CS42888_DACMUTE_ALL, mute ? CS42888_DACMUTE_ALL : 0); + + return 0; +} + +static const struct snd_soc_dai_ops cs42888_dai_ops = { + .set_fmt = cs42888_set_dai_fmt, + .set_sysclk = cs42888_set_dai_sysclk, + .hw_params = cs42888_hw_params, + .digital_mute = cs42888_digital_mute, +}; + +static struct snd_soc_dai_driver cs42888_dai = { + .name = "cs42888", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42888_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42888_FORMATS, + }, + .ops = &cs42888_dai_ops, +}; + +static const struct reg_default cs42888_reg[] = { + { 0x01, 0x01 }, /* Chip I.D. and Revision Register */ + { 0x02, 0x00 }, /* Power Control */ + { 0x03, 0xF0 }, /* Functional Mode */ + { 0x04, 0x46 }, /* Interface Formats */ + { 0x05, 0x00 }, /* ADC Control & DAC De-Emphasis */ + { 0x06, 0x10 }, /* Transition Control */ + { 0x07, 0x00 }, /* DAC Channel Mute */ + { 0x08, 0x00 }, /* Volume Control AOUT1 */ + { 0x09, 0x00 }, /* Volume Control AOUT2 */ + { 0x0a, 0x00 }, /* Volume Control AOUT3 */ + { 0x0b, 0x00 }, /* Volume Control AOUT4 */ + { 0x0c, 0x00 }, /* Volume Control AOUT5 */ + { 0x0d, 0x00 }, /* Volume Control AOUT6 */ + { 0x0e, 0x00 }, /* Volume Control AOUT7 */ + { 0x0f, 0x00 }, /* Volume Control AOUT8 */ + { 0x10, 0x00 }, /* DAC Channel Invert */ + { 0x11, 0x00 }, /* Volume Control AIN1 */ + { 0x12, 0x00 }, /* Volume Control AIN2 */ + { 0x13, 0x00 }, /* Volume Control AIN3 */ + { 0x14, 0x00 }, /* Volume Control AIN4 */ + { 0x17, 0x00 }, /* ADC Channel Invert */ + { 0x18, 0x00 }, /* Status Control */ + { 0x1a, 0x00 }, /* Status Mask */ + { 0x1b, 0x00 }, /* MUTEC Pin Control */ +}; + +static bool cs42888_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42888_STATUS: + return true; + default: + return false; + } +} + +static bool cs42888_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42888_CHIPID: + case CS42888_STATUS: + return false; + default: + return true; + } +} + +const struct regmap_config cs42888_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42888_LASTREG, + .reg_defaults = cs42888_reg, + .num_reg_defaults = ARRAY_SIZE(cs42888_reg), + .volatile_reg = cs42888_volatile_register, + .writeable_reg = cs42888_writeable_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(cs42888_regmap_config); + +static int cs42888_codec_probe(struct snd_soc_codec *codec) +{ + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + /* Mute all DAC channels */ + regmap_write(cs42888->regmap, CS42888_DACMUTE, CS42888_DACMUTE_ALL); + + return 0; +} + +static const struct snd_soc_codec_driver cs42888_driver = { + .probe = cs42888_codec_probe, + .idle_bias_off = true, + + .controls = cs42888_snd_controls, + .num_controls = ARRAY_SIZE(cs42888_snd_controls), + .dapm_widgets = cs42888_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42888_dapm_widgets), + .dapm_routes = cs42888_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs42888_dapm_routes), +}; + +int cs42888_probe(struct device *dev, struct regmap *regmap) +{ + struct cs42888_priv *cs42888; + int ret, val, i; + + cs42888 = devm_kzalloc(dev, sizeof(*cs42888), GFP_KERNEL); + if (cs42888 == NULL) + return -ENOMEM; + + dev_set_drvdata(dev, cs42888); + + cs42888->clk = devm_clk_get(dev, "mclk"); + if (IS_ERR(cs42888->clk)) { + dev_err(dev, "failed to get the clock: %ld\n", + PTR_ERR(cs42888->clk)); + return -EINVAL; + } + + cs42888->sysclk = clk_get_rate(cs42888->clk); + + for (i = 0; i < ARRAY_SIZE(cs42888->supplies); i++) + cs42888->supplies[i].supply = cs42888_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, + ARRAY_SIZE(cs42888->supplies), cs42888->supplies); + if (ret) { + dev_err(dev, "failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + /* Make sure hardware reset done */ + msleep(5); + + cs42888->regmap = regmap; + if (IS_ERR(cs42888->regmap)) { + ret = PTR_ERR(cs42888->regmap); + dev_err(dev, "failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + /* + * We haven't marked the chip revision as volatile due to + * sharing a register with the right input volume; explicitly + * bypass the cache to read it. + */ + regcache_cache_bypass(cs42888->regmap, true); + + /* Validate the chip ID */ + regmap_read(cs42888->regmap, CS42888_CHIPID, &val); + if (val < 0) { + dev_err(dev, "failed to get device ID: %x", val); + ret = -EINVAL; + goto err_enable; + } + + /* The top four bits of the chip ID should be 0000 */ + if ((val & CS42888_CHIPID_CHIP_ID_MASK) != 0x00) { + dev_err(dev, "unmatched chip ID: %d\n", + val & CS42888_CHIPID_CHIP_ID_MASK); + ret = -EINVAL; + goto err_enable; + } + + dev_info(dev, "found device, revision %X\n", + val & CS42888_CHIPID_REV_ID_MASK); + + regcache_cache_bypass(cs42888->regmap, false); + + ret = snd_soc_register_codec(dev, &cs42888_driver, &cs42888_dai, 1); + if (ret) { + dev_err(dev, "failed to register codec:%d\n", ret); + goto err_enable; + } + + regcache_cache_only(cs42888->regmap, true); + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + + return ret; +} +EXPORT_SYMBOL_GPL(cs42888_probe); + +#ifdef CONFIG_PM_RUNTIME +static int cs42888_runtime_resume(struct device *dev) +{ + struct cs42888_priv *cs42888 = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(cs42888->clk); + if (ret) { + dev_err(dev, "failed to enable mclk: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + goto err_clk; + } + + /* Make sure hardware reset done */ + msleep(5); + + regcache_cache_only(cs42888->regmap, false); + + ret = regcache_sync(cs42888->regmap); + if (ret) { + dev_err(dev, "failed to sync regmap: %d\n", ret); + goto err_bulk; + } + + return 0; + +err_bulk: + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); +err_clk: + clk_disable_unprepare(cs42888->clk); + + return ret; +} + +static int cs42888_runtime_suspend(struct device *dev) +{ + struct cs42888_priv *cs42888 = dev_get_drvdata(dev); + + regcache_cache_only(cs42888->regmap, true); + + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + + clk_disable_unprepare(cs42888->clk); + + return 0; +} +#endif + +const struct dev_pm_ops cs42888_pm = { + SET_RUNTIME_PM_OPS(cs42888_runtime_suspend, cs42888_runtime_resume, NULL) +}; +EXPORT_SYMBOL_GPL(cs42888_pm); + +const struct of_device_id cs42888_of_match[] = { + { .compatible = "cirrus,cs42888", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, cs42888_of_match); +EXPORT_SYMBOL_GPL(cs42888_of_match); + +MODULE_DESCRIPTION("Cirrus Logic CS42888 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42888.h b/sound/soc/codecs/cs42888.h new file mode 100644 index 0000000..eda55ac --- /dev/null +++ b/sound/soc/codecs/cs42888.h @@ -0,0 +1,213 @@ +/* + * cs42888.h - Cirrus Logic CS42888 Audio CODEC driver header file + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _CS42888_H +#define _CS42888_H + +extern const struct dev_pm_ops cs42888_pm; +extern const struct regmap_config cs42888_regmap_config; +int cs42888_probe(struct device *dev, struct regmap *regmap); + +/* CS42888 register map */ +#define CS42888_CHIPID 0x01 /* Chip ID */ +#define CS42888_PWRCTL 0x02 /* Power Control */ +#define CS42888_FUNCMOD 0x03 /* Functional Mode */ +#define CS42888_INTF 0x04 /* Interface Formats */ +#define CS42888_ADCCTL 0x05 /* ADC Control */ +#define CS42888_TXCTL 0x06 /* Transition Control */ +#define CS42888_DACMUTE 0x07 /* DAC Mute Control */ +#define CS42888_VOLAOUT1 0x08 /* Volume Control AOUT1 */ +#define CS42888_VOLAOUT2 0x09 /* Volume Control AOUT2 */ +#define CS42888_VOLAOUT3 0x0A /* Volume Control AOUT3 */ +#define CS42888_VOLAOUT4 0x0B /* Volume Control AOUT4 */ +#define CS42888_VOLAOUT5 0x0C /* Volume Control AOUT5 */ +#define CS42888_VOLAOUT6 0x0D /* Volume Control AOUT6 */ +#define CS42888_VOLAOUT7 0x0E /* Volume Control AOUT7 */ +#define CS42888_VOLAOUT8 0x0F /* Volume Control AOUT8 */ +#define CS42888_DACINV 0x10 /* DAC Channel Invert */ +#define CS42888_VOLAIN1 0x11 /* Volume Control AIN1 */ +#define CS42888_VOLAIN2 0x12 /* Volume Control AIN2 */ +#define CS42888_VOLAIN3 0x13 /* Volume Control AIN3 */ +#define CS42888_VOLAIN4 0x14 /* Volume Control AIN4 */ +#define CS42888_ADCINV 0x17 /* ADC Channel Invert */ +#define CS42888_STATUSCTL 0x18 /* Status Control */ +#define CS42888_STATUS 0x19 /* Status */ +#define CS42888_STATUSM 0x1A /* Status Mask */ +#define CS42888_MUTEC 0x1B /* MUTEC Pin Control */ + +#define CS42888_FIRSTREG CS42888_CHIPID +#define CS42888_LASTREG CS42888_MUTEC +#define CS42888_NUMREGS (CS42888_LASTREG - CS42888_FIRSTREG + 1) +#define CS42888_I2C_INCR 0x80 + +/* Chip I.D. and Revision Register (Address 01h) */ +#define CS42888_CHIPID_CHIP_ID_MASK 0xF0 +#define CS42888_CHIPID_REV_ID_MASK 0x0F + +/* Power Control (Address 02h) */ +#define CS42888_PWRCTL_PDN_ADC2_SHIFT 6 +#define CS42888_PWRCTL_PDN_ADC2_MASK (1 << CS42888_PWRCTL_PDN_ADC2_SHIFT) +#define CS42888_PWRCTL_PDN_ADC2 (1 << CS42888_PWRCTL_PDN_ADC2_SHIFT) +#define CS42888_PWRCTL_PDN_ADC1_SHIFT 5 +#define CS42888_PWRCTL_PDN_ADC1_MASK (1 << CS42888_PWRCTL_PDN_ADC1_SHIFT) +#define CS42888_PWRCTL_PDN_ADC1 (1 << CS42888_PWRCTL_PDN_ADC1_SHIFT) +#define CS42888_PWRCTL_PDN_DAC4_SHIFT 4 +#define CS42888_PWRCTL_PDN_DAC4_MASK (1 << CS42888_PWRCTL_PDN_DAC4_SHIFT) +#define CS42888_PWRCTL_PDN_DAC4 (1 << CS42888_PWRCTL_PDN_DAC4_SHIFT) +#define CS42888_PWRCTL_PDN_DAC3_SHIFT 3 +#define CS42888_PWRCTL_PDN_DAC3_MASK (1 << CS42888_PWRCTL_PDN_DAC3_SHIFT) +#define CS42888_PWRCTL_PDN_DAC3 (1 << CS42888_PWRCTL_PDN_DAC3_SHIFT) +#define CS42888_PWRCTL_PDN_DAC2_SHIFT 2 +#define CS42888_PWRCTL_PDN_DAC2_MASK (1 << CS42888_PWRCTL_PDN_DAC2_SHIFT) +#define CS42888_PWRCTL_PDN_DAC2 (1 << CS42888_PWRCTL_PDN_DAC2_SHIFT) +#define CS42888_PWRCTL_PDN_DAC1_SHIFT 1 +#define CS42888_PWRCTL_PDN_DAC1_MASK (1 << CS42888_PWRCTL_PDN_DAC1_SHIFT) +#define CS42888_PWRCTL_PDN_DAC1 (1 << CS42888_PWRCTL_PDN_DAC1_SHIFT) +#define CS42888_PWRCTL_PDN_SHIFT 0 +#define CS42888_PWRCTL_PDN_MASK (1 << CS42888_PWRCTL_PDN_SHIFT) +#define CS42888_PWRCTL_PDN (1 << CS42888_PWRCTL_PDN_SHIFT) + +/* Functional Mode (Address 03h) */ +#define CS42888_FUNCMOD_DAC_FM_SHIFT 6 +#define CS42888_FUNCMOD_DAC_FM_WIDTH 2 +#define CS42888_FUNCMOD_DAC_FM_MASK (((1 << CS42888_FUNCMOD_DAC_FM_WIDTH) - 1) << CS42888_FUNCMOD_DAC_FM_SHIFT) +#define CS42888_FUNCMOD_DAC_FM(v) ((v) << CS42888_FUNCMOD_DAC_FM_SHIFT) +#define CS42888_FUNCMOD_ADC_FM_SHIFT 4 +#define CS42888_FUNCMOD_ADC_FM_WIDTH 2 +#define CS42888_FUNCMOD_ADC_FM_MASK (((1 << CS42888_FUNCMOD_ADC_FM_WIDTH) - 1) << CS42888_FUNCMOD_ADC_FM_SHIFT) +#define CS42888_FUNCMOD_ADC_FM(v) ((v) << CS42888_FUNCMOD_ADC_FM_SHIFT) +#define CS42888_FUNCMOD_xC_FM_MASK(x) ((x) ? CS42888_FUNCMOD_DAC_FM_MASK : CS42888_FUNCMOD_ADC_FM_MASK) +#define CS42888_FUNCMOD_xC_FM(x, v) ((x) ? CS42888_FUNCMOD_DAC_FM(v) : CS42888_FUNCMOD_ADC_FM(v)) +#define CS42888_FUNCMOD_MFREQ_SHIFT 1 +#define CS42888_FUNCMOD_MFREQ_WIDTH 3 +#define CS42888_FUNCMOD_MFREQ_MASK (((1 << CS42888_FUNCMOD_MFREQ_WIDTH) - 1) << CS42888_FUNCMOD_MFREQ_SHIFT) +#define CS42888_FUNCMOD_MFREQ_256(s) ((0 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_384(s) ((1 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_512(s) ((2 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_768(s) ((3 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_1024(s) ((4 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) + +#define CS42888_FM_SINGLE 0 +#define CS42888_FM_DOUBLE 1 +#define CS42888_FM_QUAD 2 +#define CS42888_FM_AUTO 3 + +/* Interface Formats (Address 04h) */ +#define CS42888_INTF_FREEZE_SHIFT 7 +#define CS42888_INTF_FREEZE_MASK (1 << CS42888_INTF_FREEZE_SHIFT) +#define CS42888_INTF_FREEZE (1 << CS42888_INTF_FREEZE_SHIFT) +#define CS42888_INTF_AUX_DIF_SHIFT 6 +#define CS42888_INTF_AUX_DIF_MASK (1 << CS42888_INTF_AUX_DIF_SHIFT) +#define CS42888_INTF_AUX_DIF (1 << CS42888_INTF_AUX_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_SHIFT 3 +#define CS42888_INTF_DAC_DIF_WIDTH 3 +#define CS42888_INTF_DAC_DIF_MASK (((1 << CS42888_INTF_DAC_DIF_WIDTH) - 1) << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_LEFTJ (0 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_I2S (1 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_RIGHTJ (2 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_ONELINE_20 (4 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_ONELINE_24 (6 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_TDM (7 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_SHIFT 0 +#define CS42888_INTF_ADC_DIF_WIDTH 3 +#define CS42888_INTF_ADC_DIF_MASK (((1 << CS42888_INTF_ADC_DIF_WIDTH) - 1) << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_LEFTJ (0 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_I2S (1 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_RIGHTJ (2 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_ONELINE_20 (4 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_ONELINE_24 (6 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_TDM (7 << CS42888_INTF_ADC_DIF_SHIFT) + +/* ADC Control & DAC De-Emphasis (Address 05h) */ +#define CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 +#define CS42888_ADCCTL_ADC_HPF_FREEZE_MASK (1 << CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42888_ADCCTL_ADC_HPF_FREEZE (1 << CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42888_ADCCTL_DAC_DEM_SHIFT 5 +#define CS42888_ADCCTL_DAC_DEM_MASK (1 << CS42888_ADCCTL_DAC_DEM_SHIFT) +#define CS42888_ADCCTL_DAC_DEM (1 << CS42888_ADCCTL_DAC_DEM_SHIFT) +#define CS42888_ADCCTL_ADC1_SINGLE_SHIFT 4 +#define CS42888_ADCCTL_ADC1_SINGLE_MASK (1 << CS42888_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC1_SINGLE (1 << CS42888_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC2_SINGLE_SHIFT 3 +#define CS42888_ADCCTL_ADC2_SINGLE_MASK (1 << CS42888_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC2_SINGLE (1 << CS42888_ADCCTL_ADC2_SINGLE_SHIFT) + +/* Transition Control (Address 06h) */ +#define CS42888_TXCTL_DAC_SNGVOL_SHIFT 7 +#define CS42888_TXCTL_DAC_SNGVOL_MASK (1 << CS42888_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42888_TXCTL_DAC_SNGVOL (1 << CS42888_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SHIFT 5 +#define CS42888_TXCTL_DAC_SZC_WIDTH 2 +#define CS42888_TXCTL_DAC_SZC_MASK (((1 << CS42888_TXCTL_DAC_SZC_WIDTH) - 1) << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_IC (0 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_ZC (1 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SR (2 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SRZC (3 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_AMUTE_SHIFT 4 +#define CS42888_TXCTL_AMUTE_MASK (1 << CS42888_TXCTL_AMUTE_SHIFT) +#define CS42888_TXCTL_AMUTE (1 << CS42888_TXCTL_AMUTE_SHIFT) +#define CS42888_TXCTL_MUTE_ADC_SP_SHIFT 3 +#define CS42888_TXCTL_MUTE_ADC_SP_MASK (1 << CS42888_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42888_TXCTL_MUTE_ADC_SP (1 << CS42888_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42888_TXCTL_ADC_SNGVOL_SHIFT 2 +#define CS42888_TXCTL_ADC_SNGVOL_MASK (1 << CS42888_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42888_TXCTL_ADC_SNGVOL (1 << CS42888_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SHIFT 0 +#define CS42888_TXCTL_ADC_SZC_MASK (((1 << CS42888_TXCTL_ADC_SZC_WIDTH) - 1) << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_IC (0 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_ZC (1 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SR (2 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SRZC (3 << CS42888_TXCTL_ADC_SZC_SHIFT) + +/* DAC Channel Mute (Address 07h) */ +#define CS42888_DACMUTE_AOUT(n) (0x1 << n) +#define CS42888_DACMUTE_ALL 0xff + +/* Status Control (Address 18h)*/ +#define CS42888_STATUSCTL_INI_SHIFT 2 +#define CS42888_STATUSCTL_INI_WIDTH 2 +#define CS42888_STATUSCTL_INI_MASK (((1 << CS42888_STATUSCTL_INI_WIDTH) - 1) << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_ACTIVE_HIGH (0 << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_ACTIVE_LOW (1 << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_OPEN_DRAIN (2 << CS42888_STATUSCTL_INI_SHIFT) + +/* Status (Address 19h)*/ +#define CS42888_STATUS_DAC_CLK_ERR_SHIFT 4 +#define CS42888_STATUS_DAC_CLK_ERR_MASK (1 << CS42888_STATUS_DAC_CLK_ERR_SHIFT) +#define CS42888_STATUS_ADC_CLK_ERR_SHIFT 3 +#define CS42888_STATUS_ADC_CLK_ERR_MASK (1 << CS42888_STATUS_ADC_CLK_ERR_SHIFT) +#define CS42888_STATUS_ADC2_OVFL_SHIFT 1 +#define CS42888_STATUS_ADC2_OVFL_MASK (1 << CS42888_STATUS_ADC2_OVFL_SHIFT) +#define CS42888_STATUS_ADC1_OVFL_SHIFT 0 +#define CS42888_STATUS_ADC1_OVFL_MASK (1 << CS42888_STATUS_ADC1_OVFL_SHIFT) + +/* Status Mask (Address 1Ah) */ +#define CS42888_STATUS_DAC_CLK_ERR_M_SHIFT 4 +#define CS42888_STATUS_DAC_CLK_ERR_M_MASK (1 << CS42888_STATUS_DAC_CLK_ERR_M_SHIFT) +#define CS42888_STATUS_ADC_CLK_ERR_M_SHIFT 3 +#define CS42888_STATUS_ADC_CLK_ERR_M_MASK (1 << CS42888_STATUS_ADC_CLK_ERR_M_SHIFT) +#define CS42888_STATUS_ADC2_OVFL_M_SHIFT 1 +#define CS42888_STATUS_ADC2_OVFL_M_MASK (1 << CS42888_STATUS_ADC2_OVFL_M_SHIFT) +#define CS42888_STATUS_ADC1_OVFL_M_SHIFT 0 +#define CS42888_STATUS_ADC1_OVFL_M_MASK (1 << CS42888_STATUS_ADC1_OVFL_M_SHIFT) + +/* MUTEC Pin Control (Address 1Bh) */ +#define CS42888_MUTEC_MCPOLARITY_SHIFT 1 +#define CS42888_MUTEC_MCPOLARITY_MASK (1 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MCPOLARITY_ACTIVE_LOW (0 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MCPOLARITY_ACTIVE_HIGH (1 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MUTEC_ACTIVE_SHIFT 0 +#define CS42888_MUTEC_MUTEC_ACTIVE_MASK (1 << CS42888_MUTEC_MUTEC_ACTIVE_SHIFT) +#define CS42888_MUTEC_MUTEC_ACTIVE (1 << CS42888_MUTEC_MUTEC_ACTIVE_SHIFT) +#endif /* _CS42888_H */ -- 1.8.4

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[alsa-devel] [PATCH] ASoC: cs42l52: Fix mask bits for SOC_VALUE_ENUM_SINGLE
by Brian Austin 18 Mar '14

18 Mar '14

The mask bits values were wrong for the SOC_VALUE_ENUM_SINGLE for the PCM/ADC Swap controls Reported-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l52.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index be2ba1b..ab3ac7b 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -352,7 +352,7 @@ static const char * const right_swap_text[] = { static const unsigned int swap_values[] = { 0, 1, 3 }; static const struct soc_enum adca_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 3, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); @@ -361,7 +361,7 @@ static const struct snd_kcontrol_new adca_mixer = SOC_DAPM_ENUM("Route", adca_swap_enum); static const struct soc_enum pcma_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 3, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); @@ -370,7 +370,7 @@ static const struct snd_kcontrol_new pcma_mixer = SOC_DAPM_ENUM("Route", pcma_swap_enum); static const struct soc_enum adcb_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); @@ -379,7 +379,7 @@ static const struct snd_kcontrol_new adcb_mixer = SOC_DAPM_ENUM("Route", adcb_swap_enum); static const struct soc_enum pcmb_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: cs42l73: Fix mask bits for SOC_VALUE_ENUM_SINGLE
by Brian Austin 18 Mar '14

18 Mar '14

The mask bits values were wrong for the SOC_VALUE_ENUM_SINGLE for the mono mix controls. Reported-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l73.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 61c0f64..4cb7652 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -325,7 +325,7 @@ static const char * const cs42l73_mono_mix_texts[] = { static const unsigned int cs42l73_mono_mix_values[] = { 0, 1, 2 }; static const struct soc_enum spk_asp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 1, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); @@ -343,7 +343,7 @@ static const struct snd_kcontrol_new spk_xsp_mixer = SOC_DAPM_ENUM("Route", spk_xsp_enum); static const struct soc_enum esl_asp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 5, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); @@ -352,7 +352,7 @@ static const struct snd_kcontrol_new esl_asp_mixer = SOC_DAPM_ENUM("Route", esl_asp_enum); static const struct soc_enum esl_xsp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 7, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 0/7] Audio support for Armada 370 DB
by Thomas Petazzoni 18 Mar '14

18 Mar '14

Hello, This series of patches enable audio support on the Marvell Armada 370 Development Board. Since both the I2S controller on the SoC side and the I2C audio codec are already supported by the kernel, the amount of work is fairly limited. Changes since v1: * Drop patches "sound: codec: add Device Tree binding to cs42l51" and " sound: soc: enable Kirkwood driver for mvebu platforms" since they have been applied by Mark Brown. * Set DAI format directly in the snd_soc_dai_link structure instead of separately calling snd_soc_dai_set_fmt(). Suggested by Mark Brown. * Do not call snd_soc_dapm_enable_pin(), as Mark Brown mentionned that DAPM widgets default to enabled. It allowed to entirely remove the dai_init operation. * Use devm_snd_soc_register_card() instead of snd_soc_register_card(), as suggested by Mark Brown. * Wrote a Device Tree binding document for the DT binding introduced by this driver. Requested by Mark Brown. * Use the DT to look up the controller and the codec, as suggested by Mark Brown. * Add S/PDIF support, which has been successfully tested. I've kept it as separate patches (for both the ASoC machine driver, and the Device Tree bits), so that it can be reviewed and applied separately. * Modified the commit title so that they contain "ASoC", as requested by Mark Brown. Patches 1 and 2 are to be reviewed/applied by the ASoC maintainer Mark Brown, while patches 3 to 7 are to be reviewed/applied by the ARM mvebu maintainers. Note that the audio support for Armada 370 also needs a fix to the CS42L51, which is being discussed with the author of the change that apparently introduced the problem (see discussion at http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071916.ht…). Thanks, Thomas Thomas Petazzoni (7): sound: ASoC: add ASoC board driver for Armada 370 DB sound: ASoC: add S/PDIF support to Armada 370 DB ASoC driver ARM: mvebu: add audio I2S controller to Armada 370 Device Tree ARM: mvebu: add I2C0 muxing option for Armada 370 SoC ARM: mvebu: add audio support to Armada 370 DB ARM: mvebu: enable S/PDIF audio in Armada 370 DB Device Tree ARM: mvebu: enable audio options in mvebu_defconfig .../bindings/sound/armada-370db-audio.txt | 27 ++++ arch/arm/boot/dts/armada-370-db.dts | 56 ++++++++ arch/arm/boot/dts/armada-370.dtsi | 28 ++++ arch/arm/configs/mvebu_defconfig | 5 + sound/soc/kirkwood/Kconfig | 9 ++ sound/soc/kirkwood/Makefile | 2 + sound/soc/kirkwood/armada-370-db.c | 148 +++++++++++++++++++++ 7 files changed, 275 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/armada-370db-audio.txt create mode 100644 sound/soc/kirkwood/armada-370-db.c -- 1.8.3.2

3 10

[alsa-devel] [RFC] ASoC: samsung: don't build pcm and spdif on exynos
by Arnd Bergmann 18 Mar '14

18 Mar '14

In commit d37bdf736d9b "ASoC: samsung: Use ASoC dmaengine code where possible", Mark Brown changed the EXYNOS ASoC code to no longer use the private dma interfaces that are standing in the way of multiplatform enablement. However, in 313367e7bfa "ASoC: Samsung: Update Kconfig for I2S,SPDIF and PCM audio", two new users of that interface appeared, since it is now possible to enable SND_SOC_SAMSUNG_SMDK_SPDIF and SND_SOC_SMDK_WM8994_PCM on all Samsung SoCs including EXYNOS. This patch does a partial revert of 313367e7bfa by allowing these drivers on all samsung platforms except EXYNOS, so we can proceed with the multiplatform patches. If support for these drivers is actually needed on EXYNOS machines, the drivers should probably first be moved over to use the dmaengine code. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Sangsu Park <sangsu4u.park(a)samsung.com> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Padmavathi Venna <padma.v(a)samsung.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Padma, could you comment on this? When you did the patch that introduced the usage of these drivers on EXYNOS, did you actually need that for a particular machine that would get broken by this revert? diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 7b610a8..efc01e0 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -167,7 +167,7 @@ config SND_SOC_GONI_AQUILA_WM8994 config SND_SOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && !ARCH_EXYNOS select SND_SAMSUNG_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. @@ -183,7 +183,7 @@ config SND_SOC_SMDK_WM8580_PCM config SND_SOC_SMDK_WM8994_PCM tristate "SoC PCM Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && !ARCH_EXYNOS depends on I2C=y select MFD_WM8994 select SND_SOC_WM8994

1 0

[alsa-devel] [PATCH] ASoC: codecs: Replace instances of rtd->codec with dai->codec
by Lars-Peter Clausen 18 Mar '14

18 Mar '14

With CODEC to CODEC links rtd->codec does not necessarily point to the driver's CODEC. CODEC drivers should always use dai->codec and never even look at the PCM runtime. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/isabelle.c | 3 +-- sound/soc/codecs/mc13783.c | 6 ++---- sound/soc/codecs/rt5640.c | 3 +-- sound/soc/codecs/sta529.c | 3 +-- sound/soc/codecs/tlv320aic31xx.c | 5 ++--- sound/soc/codecs/uda134x.c | 3 +-- sound/soc/codecs/uda1380.c | 3 +-- sound/soc/codecs/wm8580.c | 3 +-- 8 files changed, 10 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 3e264a7..3a89ce6 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -918,8 +918,7 @@ static int isabelle_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 snd_soc_codec *codec = dai->codec; u16 aif = 0; unsigned int fs_val = 0; diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 37d737e..2c59b1f 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -106,8 +106,7 @@ static int mc13783_pcm_hw_params_dac(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 snd_soc_codec *codec = dai->codec; unsigned int rate = params_rate(params); int i; @@ -126,8 +125,7 @@ static int mc13783_pcm_hw_params_codec(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 snd_soc_codec *codec = dai->codec; unsigned int rate = params_rate(params); unsigned int val; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 13ccee4..0061ae6 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1594,8 +1594,7 @@ static int get_clk_info(int sclk, int rate) static int rt5640_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 snd_soc_codec *codec = dai->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); unsigned int val_len = 0, val_clk, mask_clk; int dai_sel, pre_div, bclk_ms, frame_size; diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index a3c61d3..a40c4b0 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -193,8 +193,7 @@ static int sta529_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 snd_soc_codec *codec = dai->codec; int pdata, play_freq_val, record_freq_val; int bclk_to_fs_ratio; diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index d3517a9..cc1c9c2 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -753,10 +753,9 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, static int aic31xx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, - struct snd_soc_dai *tmp) + struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u8 data = 0; dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n", diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index c94d4c1..edf27ac 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -203,8 +203,7 @@ static int uda134x_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 snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); u8 hw_params; diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 4dadaa8..e62e707 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -566,8 +566,7 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* shut down WSPLL power if running from this clock */ diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 7558c83..af7ed8b 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -504,8 +504,7 @@ static int wm8580_paif_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 snd_soc_codec *codec = dai->codec; struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); u16 paifa = 0; u16 paifb = 0; -- 1.8.0

4 3

[alsa-devel] [PATCH v2 0/4] ASoC: davinci: edma dmaengine PCM and mcasp preparation
by Peter Ujfalusi 18 Mar '14

18 Mar '14

Hi, This is a cutdown version of the previous [1] series containing only the ASoC part of the edma dmaengine conversion, but without the actual switch to this new edma-pcm platform for the McASP since the edma arch and dmaengine core changes are needed for it to be usable. Changes since v1: - arch and dmaengine patches has been dropped from the ASoC series - Patches has been reordered so only one patch is going to be needed to switch AM335x/AM447x to use the edma-pcm platform driver - all comments from Lars-Peter Clausen has been addressed for the edma-pcm driver - no changes to Kconfig/Makefile - do not yet build the edma-pcm driver. It will be enabled and integrated when we can do the real switch. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074269.html Regards, Peter --- Peter Ujfalusi (4): ASoC: davinci: Add edma dmaengine platform driver ASoC: davinci-mcasp: Provide correct filter_data for dmaengine for non-DT boot ASoC: davinci-mcasp: Constraint on the period and buffer size based on FIFO usage ASoC: davinci-mcasp: Assign the dma_data earlier in dai_probe callback sound/soc/davinci/davinci-mcasp.c | 59 +++++++++++++++++++++++++++++++++------ sound/soc/davinci/edma-pcm.c | 57 +++++++++++++++++++++++++++++++++++++ sound/soc/davinci/edma-pcm.h | 25 +++++++++++++++++ 3 files changed, 132 insertions(+), 9 deletions(-) create mode 100644 sound/soc/davinci/edma-pcm.c create mode 100644 sound/soc/davinci/edma-pcm.h -- 1.9.0

3 13

[alsa-devel] [PATCH v2] ASoC: tlv320aic31xx: Don't call kfree for memory allocated by devm_kzalloc
by Axel Lin 18 Mar '14

18 Mar '14

The kfree call is not necessary, but we need to call snd_soc_unregister_codec() in remove(). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/tlv320aic31xx.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index d3517a9..d5d6213 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1228,7 +1228,6 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return -ENOMEM; aic31xx->regmap = devm_regmap_init_i2c(i2c, regmap_config); - if (IS_ERR(aic31xx->regmap)) { ret = PTR_ERR(aic31xx->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -1241,18 +1240,14 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, aic31xx_device_init(aic31xx); - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, + return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, aic31xx_dai_driver, ARRAY_SIZE(aic31xx_dai_driver)); - - return ret; } static int aic31xx_i2c_remove(struct i2c_client *i2c) { - struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev); - - kfree(aic31xx); + snd_soc_unregister_codec(&i2c->dev); return 0; } @@ -1274,7 +1269,7 @@ static struct i2c_driver aic31xx_i2c_driver = { .of_match_table = of_match_ptr(tlv320aic31xx_of_match), }, .probe = aic31xx_i2c_probe, - .remove = (aic31xx_i2c_remove), + .remove = aic31xx_i2c_remove, .id_table = aic31xx_i2c_id, }; -- 1.8.1.2

3 2

[alsa-devel] [PATCH] regmap: Fix possible sleep-in-atomic in fast io mode
by Takashi Iwai 18 Mar '14

18 Mar '14

regmap deploys the spinlock for the protection when set up in fast_io mode. But some places are still left with possible sleep in the lock, typically allocating memory with GFP_KERNEL. This patch fixes these calls by moving the allocations out of the lock. The krealloc() call in regmap_register_patch() makes it unsafe for concurrent accesses, however. But such a use case must be very rare, thus we take rather simplicity over complexity, and add a note in the function description mentioning this restriction. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Only compile tested. drivers/base/regmap/regmap.c | 36 ++++++++++++++++++++---------------- 1 file changed, 20 insertions(+), 16 deletions(-) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 6a19515f8a45..e6827cee29a3 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -1520,12 +1520,12 @@ int regmap_bulk_write(struct regmap *map, unsigned int reg, const void *val, if (reg % map->reg_stride) return -EINVAL; - map->lock(map->lock_arg); /* * Some devices don't support bulk write, for * them we have a series of single write operations. */ if (!map->bus || map->use_single_rw) { + map->lock(map->lock_arg); for (i = 0; i < val_count; i++) { unsigned int ival; @@ -1554,24 +1554,25 @@ int regmap_bulk_write(struct regmap *map, unsigned int reg, const void *val, if (ret != 0) goto out; } +out: + map->unlock(map->lock_arg); } else { void *wval; wval = kmemdup(val, val_count * val_bytes, GFP_KERNEL); if (!wval) { - ret = -ENOMEM; dev_err(map->dev, "Error in memory allocation\n"); - goto out; + return -ENOMEM; } + map->lock(map->lock_arg); for (i = 0; i < val_count * val_bytes; i += val_bytes) map->format.parse_inplace(wval + i); ret = _regmap_raw_write(map, reg, wval, val_bytes * val_count); + map->unlock(map->lock_arg); kfree(wval); } -out: - map->unlock(map->lock_arg); return ret; } EXPORT_SYMBOL_GPL(regmap_bulk_write); @@ -2173,6 +2174,10 @@ EXPORT_SYMBOL_GPL(regmap_async_complete); * apply them immediately. Typically this is used to apply * corrections to be applied to the device defaults on startup, such * as the updates some vendors provide to undocumented registers. + * + * Note that the function gives no protection over concurrent access + * to map->patch and map->patch_regs. If needed, apply some lock in + * the caller side. */ int regmap_register_patch(struct regmap *map, const struct reg_default *regs, int num_regs) @@ -2185,6 +2190,16 @@ int regmap_register_patch(struct regmap *map, const struct reg_default *regs, num_regs)) return 0; + p = krealloc(map->patch, + sizeof(struct reg_default) * (map->patch_regs + num_regs), + GFP_KERNEL); + if (!p) + return -ENOMEM; + + memcpy(p + map->patch_regs, regs, num_regs * sizeof(*regs)); + map->patch = p; + map->patch_regs += num_regs; + map->lock(map->lock_arg); bypass = map->cache_bypass; @@ -2202,17 +2217,6 @@ int regmap_register_patch(struct regmap *map, const struct reg_default *regs, } } - p = krealloc(map->patch, - sizeof(struct reg_default) * (map->patch_regs + num_regs), - GFP_KERNEL); - if (p) { - memcpy(p + map->patch_regs, regs, num_regs * sizeof(*regs)); - map->patch = p; - map->patch_regs += num_regs; - } else { - ret = -ENOMEM; - } - out: map->async = false; map->cache_bypass = bypass; -- 1.9.0

2 3

[alsa-devel] [PATCH RFC] ALSA: oxygen: Xonar DG(X): fix Stereo Upmixing regression
by Clemens Ladisch 18 Mar '14

18 Mar '14

The code introduced in commit 1f91ecc14dee ("ALSA: oxygen: modify adjust_dg_dac_routing function") accidentally disregarded the old value of the playback routing register, so it broke the "Stereo Upmixing" mixer control. The unmuted parts of the channel routing are the same for all settings of the output destination, so it suffices to revert that part of the patch. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/pci/oxygen/xonar_dg.c | 30 ++++++++++++++++++++++++------ 1 file changed, 24 insertions(+), 6 deletions(-) Roman, could you please test this patch, and add your Tested-by? diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index ed6f199..4cf3200 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -238,11 +238,21 @@ void set_cs4245_adc_params(struct oxygen *chip, cs4245_write_spi(chip, CS4245_MCLK_FREQ); } +static inline unsigned int shift_bits(unsigned int value, + unsigned int shift_from, + unsigned int shift_to, + unsigned int mask) +{ + if (shift_from < shift_to) + return (value << (shift_to - shift_from)) & mask; + else + return (value >> (shift_from - shift_to)) & mask; +} + unsigned int adjust_dg_dac_routing(struct oxygen *chip, unsigned int play_routing) { struct dg *data = chip->model_data; - unsigned int routing = 0; switch (data->output_sel) { case PLAYBACK_DST_HP: @@ -252,15 +262,23 @@ unsigned int adjust_dg_dac_routing(struct oxygen *chip, OXYGEN_PLAY_MUTE67, OXYGEN_PLAY_MUTE_MASK); break; case PLAYBACK_DST_MULTICH: - routing = (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | - (2 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | - (1 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | - (0 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT); oxygen_write8_masked(chip, OXYGEN_PLAY_ROUTING, OXYGEN_PLAY_MUTE01, OXYGEN_PLAY_MUTE_MASK); break; } - return routing; + return (play_routing & OXYGEN_PLAY_DAC0_SOURCE_MASK) | + shift_bits(play_routing, + OXYGEN_PLAY_DAC2_SOURCE_SHIFT, + OXYGEN_PLAY_DAC1_SOURCE_SHIFT, + OXYGEN_PLAY_DAC1_SOURCE_MASK) | + shift_bits(play_routing, + OXYGEN_PLAY_DAC1_SOURCE_SHIFT, + OXYGEN_PLAY_DAC2_SOURCE_SHIFT, + OXYGEN_PLAY_DAC2_SOURCE_MASK) | + shift_bits(play_routing, + OXYGEN_PLAY_DAC0_SOURCE_SHIFT, + OXYGEN_PLAY_DAC3_SOURCE_SHIFT, + OXYGEN_PLAY_DAC3_SOURCE_MASK); } void dump_cs4245_registers(struct oxygen *chip,

3 4

[alsa-devel] Restore default value for alc282
by Kailang 18 Mar '14

18 Mar '14

Hi Takashi, Attachment is the patch for fix below issue for alc282. Restore the registers to prevent the abnormal digital power supply rising ratio/sequence to the codec and causing the incorrect default codec register restoration during initialization. Many Thanks. BR, Kailang

1 0

[alsa-devel] Restore default value for alc283
by Kailang 18 Mar '14

18 Mar '14

Hi Takashi, Attachment is the patch for fix below issue. Restore the registers to prevent the abnormal digital power supply rising ratio/sequence to the codec and causing the incorrect default codec register restoration during initialization. Many Thanks. BR, Kailang

2 4

[alsa-devel] [PATCH] ASoC: tegra: Use flat regcache.
by Dylan Reid 18 Mar '14

18 Mar '14

When using an rbtree cache, there can be allocations the first time a register is accessed. This can cause an attempt to schedule while atomic in the case that the regmap is using a spinlock. This could be fixed by either initializing all the registers or using a flat cache. The register maps for tegra30_ahub and tegra30_i2s are dense and don't save much from using a tree so convert them to flat. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/tegra/tegra30_ahub.c | 4 ++-- sound/soc/tegra/tegra30_i2s.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d6f4c99..0db68f4 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -471,7 +471,7 @@ static const struct regmap_config tegra30_ahub_apbif_regmap_config = { .readable_reg = tegra30_ahub_apbif_wr_rd_reg, .volatile_reg = tegra30_ahub_apbif_volatile_reg, .precious_reg = tegra30_ahub_apbif_precious_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static bool tegra30_ahub_ahub_wr_rd_reg(struct device *dev, unsigned int reg) @@ -490,7 +490,7 @@ static const struct regmap_config tegra30_ahub_ahub_regmap_config = { .max_register = LAST_REG(AUDIO_RX), .writeable_reg = tegra30_ahub_ahub_wr_rd_reg, .readable_reg = tegra30_ahub_ahub_wr_rd_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static struct tegra30_ahub_soc_data soc_data_tegra30 = { diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 49ad936..f146c41 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -357,7 +357,7 @@ static const struct regmap_config tegra30_i2s_regmap_config = { .writeable_reg = tegra30_i2s_wr_rd_reg, .readable_reg = tegra30_i2s_wr_rd_reg, .volatile_reg = tegra30_i2s_volatile_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static const struct tegra30_i2s_soc_data tegra30_i2s_config = { -- 1.8.1.3.605.g02339dd

5 7

[alsa-devel] [PATCH 1/2] ASoC: sirf-audio-codec: Remove snd_soc_codec_set_cache_io() call
by Lars-Peter Clausen 18 Mar '14

18 Mar '14

There was a overlap between the snd_soc_codec_set_cache_io() cleanup and the addition of the sirf-audio-codec resulting in the sirf-audio-codec driver still using the old signature of snd_soc_codec_set_cache_io(), which will cause a compile error. Since the core is able to automatically setup IO for this driver we can just remove both the snd_soc_set_cache_io() call and the control_data assignment. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/sirf-audio-codec.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 90e3a22..58e7c1f 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -337,18 +337,9 @@ struct snd_soc_dai_driver sirf_audio_codec_dai = { static int sirf_audio_codec_probe(struct snd_soc_codec *codec) { - int ret; struct snd_soc_dapm_context *dapm = &codec->dapm; - struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec); pm_runtime_enable(codec->dev); - codec->control_data = sirf_audio_codec->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } if (of_device_is_compatible(codec->dev->of_node, "sirf,prima2-audio-codec")) { snd_soc_dapm_new_controls(dapm, -- 1.8.0

3 6

[alsa-devel] [PATCH v2 0/4] ASoC: simple-card: DT fix and multi DAI links extension
by Jean-Francois Moine 17 Mar '14

17 Mar '14

This patch series fixes a small problem about node device locks and also extends the simple card driver to handle many DAI links as this exists in the Cubox audio subsystem. - v2 - change subject/comment about device node reference count (Mark Brown) - use a null size array instead of an implicit area for the DAI links (Li Xiubo) - update the reference count of the device node at end of probe Jean-Francois Moine (4): ASoC: simple-card: Fix the reference count of device nodes ASoC: simple-card: dynamically allocate the DAI link array ASoC: simple-card: accept many DAI links ASoC: simple-card: Add DT documentation for multi-DAI links .../devicetree/bindings/sound/simple-card.txt | 34 +++- sound/soc/generic/simple-card.c | 174 ++++++++++++++------- 2 files changed, 152 insertions(+), 56 deletions(-) -- 1.9.0

4 11

[alsa-devel] [PATCH] ASoC: fsi: fixup compile error for simple-card
by Kuninori Morimoto 17 Mar '14

17 Mar '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patches fixes c7a507eea1db1430476289f525f9c853d5d485e8 (ASoC: fsi: fixup SND_SOC_DAIFMT_CBx_CFx flags) commit's compie error arch/arm/mach-shmobile/board-mackerel.c:512:2: \ error: unknown field 'fmt' specified in initializer Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark This patch is for mark/fix/rcar arch/arm/mach-shmobile/board-mackerel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c index 8c43321..0ff4d8e 100644 --- a/arch/arm/mach-shmobile/board-mackerel.c +++ b/arch/arm/mach-shmobile/board-mackerel.c @@ -509,7 +509,7 @@ static struct asoc_simple_card_info fsi2_hdmi_info = { .card = "FSI2B-HDMI", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", - .fmt = SND_SOC_DAIFMT_CBS_CFS, + .daifmt = SND_SOC_DAIFMT_CBS_CFS, .cpu_dai = { .name = "fsib-dai", }, -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Turn power off only once.
by Jyri Sarha 17 Mar '14

17 Mar '14

Regulator code keep count of enables and disables. Double disable causes an ugly warning. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 89f808b..07ddec3 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1021,7 +1021,8 @@ static int aic31xx_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_OFF: - aic31xx_power_off(codec); + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + aic31xx_power_off(codec); break; } codec->dapm.bias_level = level; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/2][RFC] ASoC: fsi/rcar: fixup SND_SOC_DAIFMT_CBx_CFx flags
by Kuninori Morimoto 17 Mar '14

17 Mar '14

Hi Mark, Simon These patches fixup Renesas fsi/rcar driver's SND_SOC_DAIFMT_CBx_CFx flag settings. I had misunderstand these settings... But this is still [RFC], since it modifies DT side code too. >> Mark #1 patch depends on Jyri's "Fix simple-card *-master DT parameter handling" if it was accepted >> Simon Can you check #1 patch which modifys SH-ARM platform ? Kuninori Morimoto (2): 1) ASoC: fsi: fixup SND_SOC_DAIFMT_CBx_CFx flags 2) ASoC: rcar: fixup SND_SOC_DAIFMT_CBx_CFx flags .../boot/dts/r8a7740-armadillo800eva-reference.dts | 3 ++- arch/arm/boot/dts/sh73a0-kzm9g-reference.dts | 2 ++ arch/arm/mach-shmobile/board-armadillo800eva.c | 4 +--- arch/arm/mach-shmobile/board-kzm9g.c | 4 +--- arch/arm/mach-shmobile/board-mackerel.c | 6 ++---- arch/sh/boards/mach-ecovec24/setup.c | 4 +--- arch/sh/boards/mach-se/7724/setup.c | 4 +--- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/rcar/core.c | 4 ++-- 9 files changed, 13 insertions(+), 20 deletions(-) Best regards --- Kuninori Morimoto

3 13

Re: [alsa-devel] compress: adding support for hardware dependent params
by noman pouigt 16 Mar '14

16 Mar '14

> We have a requirement, where we need to set a number of hardware specific > parameters for compressed offload use-cases, before starting a stream. As > these parameters need to be set once for the entire session, we would > like to explore the best way to set these additional parameters somewhere > before calling trigger() function. > Below are few of the proposed solutions that we could think of at this > point for getting this done- As you have not mentioned what kind of metadata you are talking about I am assuming if your meta data can fit in below structure then we already have a code in upstream which takes care of your need. 140 struct snd_compr_metadata { 141 __u32 key; 142 __u32 value[8]; 143 }; Structure snd_compr_ops has set_metadata and get_metadata callbacks which you can use for your need. 88 * struct snd_compr_ops: compressed path DSP operations 89 * @open: Open the compressed stream 90 * This callback is mandatory and shall keep dsp ready to receive the stream 91 * parameter 92 * @free: Close the compressed stream, mandatory 93 * @set_params: Sets the compressed stream parameters, mandatory 94 * This can be called in during stream creation only to set codec params 95 * and the stream properties 96 * @get_params: retrieve the codec parameters, mandatory 97 * @trigger: Trigger operations like start, pause, resume, drain, stop. 98 * This callback is mandatory 99 * @pointer: Retrieve current h/w pointer information. Mandatory 100 * @copy: Copy the compressed data to/from userspace, Optional 101 * Can't be implemented if DSP supports mmap 102 * @mmap: DSP mmap method to mmap DSP memory 103 * @ack: Ack for DSP when data is written to audio buffer, Optional 104 * Not valid if copy is implemented 105 * @get_caps: Retrieve DSP capabilities, mandatory 106 * @get_codec_caps: Retrieve capabilities for a specific codec, mandatory 107 */ It would be nice if someone can send a patch here to describe set and get_metadata callbacks in compress_driver.h > > > 1. Adding new IOCTL call: Add a new ioctl control in compress-offload.c > file to handle hardware specific parameters setting. This new ioctl > control will call into soc-compress.c and eventually call into platform > driver ioctl to set those hardware specific parameters. In this case, we > will add a new structure, let's say "snd_compr_hw_ctrl_params", with the > same fields as "snd_compr_metadata" struct. > > > 2. Adding a function hw_dependant_hw_params(), which will be called after > hw_params() and before prepare() function. The new function will > eventually call into the platform driver to get the hardware specific > parameters set. Once implemented, this would look something like- > hw_params(); > hw_dependant_hw_params(); > prepare(); > > > 3. Add a function "hw_dependant_hw_params()", as _part_ of the hw_params() > function. This new function will be called after the (existing) generic > parameters are set. So once implemented, the proposed function calls would > look like- > hw_params() { > generic_hw_params(); > hw_dependant_hw_params(); > } > prepare(); > > > We would like to get feedback on these proposed solutions and get a > discussion going to see how best these requirements can be fitted in the > current Compressed ALSA framework (for mainlining). Can you explain your meta data? Be the change you want to see in the world:Gandhi

4 3

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Don't call kfree for memory allocated by devm_kzalloc
by Axel Lin 16 Mar '14

16 Mar '14

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/tlv320aic31xx.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index d3517a9..1953184 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1248,14 +1248,6 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return ret; } -static int aic31xx_i2c_remove(struct i2c_client *i2c) -{ - struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev); - - kfree(aic31xx); - return 0; -} - static const struct i2c_device_id aic31xx_i2c_id[] = { { "tlv320aic310x", AIC3100 }, { "tlv320aic311x", AIC3110 }, @@ -1274,7 +1266,6 @@ static struct i2c_driver aic31xx_i2c_driver = { .of_match_table = of_match_ptr(tlv320aic31xx_of_match), }, .probe = aic31xx_i2c_probe, - .remove = (aic31xx_i2c_remove), .id_table = aic31xx_i2c_id, }; -- 1.8.1.2

1 0

[alsa-devel] [RFC] [PATCH] firewire-lib: add support for AV/C deferred transaction
by Takashi Sakamoto 15 Mar '14

15 Mar '14

Clemens and Stefan, Can I request your comments about attached patch? This is for adding support of deferred transaction into snd-firewire-lib. I confirm some BeBoB devices use this type of transaction. If you have no negative comments, I'll add this patch for my series. For 'undefined interval' between INTERIM and final response, I want to use 'FCP_TIMEOUT_MS' (=125msec) again. In specifications which I can read, there is no limit for this interval. But I need to promise to finish this function for callers. According to AV/C general specification, targets can send INTERIM response one time. But this patch allows to accept several ITERIM responses for device quirks. I don't have devices which have such quirks, but there may be such devices and I want to make codes simpler. Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

2 2

[alsa-devel] [GIT PULL][PATCH 00/39 v2] Enhancement of support for Firewire devices
by Takashi Sakamoto 15 Mar '14

15 Mar '14

This pull-request is an update for my previous pull-request: [alsa-devel] [GIT PULL][PATCH 00/39] Enhancement of support for Firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073446.h… Updates since previous pull-request: firewire-lib: - use swap() instead of local macro - check NULL at memory allocation for sort table fireworks: - use bitmap.h for card slots - add 'firewire' subdirectory to card node in procfs - add sanity check of module parameter with clamp() bebob: - use bitmap.h for card slots - add 'firewire' subdirectory to card node in procfs - simplify codes for metering - use swap() instead of local macro - change order of last two commits ---------------------------------------------------------------- The following changes since commit 31278997add61ee13b709e274934c7b0085accce: ALSA: hda/realtek - Add headset quirk for Dell DT (2014-03-03 10:48:55 +0100) are available in the git repository at: https://github.com/takaswie/sound.git request for you to fetch changes up to 5b354e20ec3c64fa0a8fa7fbc7355201d881a8cc: bebob: Send a cue to load firmware for M-Audio Firewire series (2014-03-05 12:00:32 +0900) ---------------------------------------------------------------- Takashi Sakamoto (39): firewire-lib: Rename functions, structure, member for AMDTP firewire-lib: Add macros instead of fixed value for AMDTP firewire-lib: Add 'direction' member to 'amdtp_stream' structure firewire-lib: Split some codes into functions to reuse for both streams firewire-lib: Add support for AMDTP in-stream and PCM capture firewire-lib: Add support for MIDI capture/playback firewire-lib: Give syt value as parameter to handle_out_packet() firewire-lib: Add support for duplex streams synchronization in blocking mode firewire-lib: Add sort function for transmitted packet firewire-lib: Add transfer delay to synchronized duplex streams firewire-lib: Add support for channel mapping firewire-lib: Rename macros, variables and functions for CMP firewire-lib: Add 'direction' member to 'cmp_connection' structure firewire-lib: Add handling output connection by CMP firewire-lib: Add a new function to check others' connection firewire-lib: Add some AV/C general commands firewire-lib: Add quirks for Fireworks firewire-lib: Add a fallback at RCODE_CANCELLED fireworks: Add skelton for Fireworks based devices fireworks: Add transaction and some commands fireworks: Add connection and stream management fireworks: Add proc interface for debugging purpose fireworks: Add MIDI interface fireworks: Add PCM interface fireworks: Add hwdep interface fireworks: Add command/response functionality into hwdep interface bebob: Add skelton for BeBoB based devices bebob: Add commands and connections/streams management bebob: Add proc interface for debugging purpose bebob: Add MIDI interface bebob: Add PCM interface bebob: Add hwdep interface bebob: Prepare for device specific operations bebob: Add support for Terratec PHASE, EWS series and Aureon bebob: Add support for Yamaha GO series bebob: Add support for Focusrite Saffire/SaffirePro series bebob: Add support for M-Audio usual Firewire series bebob: Add support for M-Audio special Firewire series bebob: Send a cue to load firmware for M-Audio Firewire series include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 22 +- sound/firewire/Kconfig | 59 ++ sound/firewire/Makefile | 2 + sound/firewire/amdtp.c | 891 +++++++++++++++++------ sound/firewire/amdtp.h | 175 ++++- sound/firewire/bebob/Makefile | 4 + sound/firewire/bebob/bebob.c | 450 ++++++++++++ sound/firewire/bebob/bebob.h | 260 +++++++ sound/firewire/bebob/bebob_command.c | 354 +++++++++ sound/firewire/bebob/bebob_focusrite.c | 287 ++++++++ sound/firewire/bebob/bebob_hwdep.c | 197 +++++ sound/firewire/bebob/bebob_maudio.c | 828 +++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 170 +++++ sound/firewire/bebob/bebob_pcm.c | 451 ++++++++++++ sound/firewire/bebob/bebob_proc.c | 201 +++++ sound/firewire/bebob/bebob_stream.c | 869 ++++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 68 ++ sound/firewire/bebob/bebob_yamaha.c | 50 ++ sound/firewire/cmp.c | 229 ++++-- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 46 +- sound/firewire/fcp.c | 160 +++- sound/firewire/fcp.h | 21 + sound/firewire/fireworks/Makefile | 4 + sound/firewire/fireworks/fireworks.c | 326 +++++++++ sound/firewire/fireworks/fireworks.h | 223 ++++++ sound/firewire/fireworks/fireworks_command.c | 397 ++++++++++ sound/firewire/fireworks/fireworks_hwdep.c | 294 ++++++++ sound/firewire/fireworks/fireworks_midi.c | 179 +++++ sound/firewire/fireworks/fireworks_pcm.c | 477 ++++++++++++ sound/firewire/fireworks/fireworks_proc.c | 233 ++++++ sound/firewire/fireworks/fireworks_stream.c | 337 +++++++++ sound/firewire/fireworks/fireworks_transaction.c | 327 +++++++++ sound/firewire/lib.c | 6 +- sound/firewire/lib.h | 1 + sound/firewire/speakers.c | 93 +-- 37 files changed, 8307 insertions(+), 402 deletions(-) create mode 100644 sound/firewire/bebob/Makefile create mode 100644 sound/firewire/bebob/bebob.c create mode 100644 sound/firewire/bebob/bebob.h create mode 100644 sound/firewire/bebob/bebob_command.c create mode 100644 sound/firewire/bebob/bebob_focusrite.c create mode 100644 sound/firewire/bebob/bebob_hwdep.c create mode 100644 sound/firewire/bebob/bebob_maudio.c create mode 100644 sound/firewire/bebob/bebob_midi.c create mode 100644 sound/firewire/bebob/bebob_pcm.c create mode 100644 sound/firewire/bebob/bebob_proc.c create mode 100644 sound/firewire/bebob/bebob_stream.c create mode 100644 sound/firewire/bebob/bebob_terratec.c create mode 100644 sound/firewire/bebob/bebob_yamaha.c create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_hwdep.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c create mode 100644 sound/firewire/fireworks/fireworks_transaction.c

3 68

[alsa-devel] [PATCH] ASoC: max98090: Remove unused control_data field
by Lars-Peter Clausen 14 Mar '14

14 Mar '14

The driver assigns a value to the control_data field of the driver's state struct, but never reads it again. Which means it is unused and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/max98090.c | 1 - sound/soc/codecs/max98090.h | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 390fba1..1baa12c 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2343,7 +2343,6 @@ static int max98090_i2c_probe(struct i2c_client *i2c, max98090->devtype = id->driver_data; i2c_set_clientdata(i2c, max98090); - max98090->control_data = i2c; max98090->pdata = i2c->dev.platform_data; max98090->irq = i2c->irq; diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index 7e103f2..1a4e233 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1523,7 +1523,6 @@ struct max98090_priv { struct regmap *regmap; struct snd_soc_codec *codec; enum max98090_type devtype; - void *control_data; struct max98090_pdata *pdata; unsigned int sysclk; unsigned int bclk; -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Staticise non-exported symbols
by Mark Brown 14 Mar '14

14 Mar '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic31xx.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index dcdc5751048f..d3517a919776 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -129,7 +129,7 @@ static const struct regmap_range_cfg aic31xx_ranges[] = { }, }; -struct regmap_config aic31xx_i2c_regmap = { +static const struct regmap_config aic31xx_i2c_regmap = { .reg_bits = 8, .val_bits = 8, .writeable_reg = aic31xx_writeable, @@ -321,9 +321,9 @@ static const struct snd_kcontrol_new ldac_in_control = static const struct snd_kcontrol_new rdac_in_control = SOC_DAPM_ENUM("DAC Right Input", rdac_in_enum); -int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, - unsigned int mask, unsigned int wbits, int sleep, - int count) +static int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, + unsigned int mask, unsigned int wbits, int sleep, + int count) { unsigned int bits; int counter = count; @@ -1177,7 +1177,7 @@ static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) } #endif /* CONFIG_OF */ -void aic31xx_device_init(struct aic31xx_priv *aic31xx) +static void aic31xx_device_init(struct aic31xx_priv *aic31xx) { int ret, i; -- 1.9.0

3 3

[alsa-devel] [PATCH 00/18] edma/ASoC: dmaengine PCM for AM335x and AM447x
by Peter Ujfalusi 14 Mar '14

14 Mar '14

Hi, With this series AM335x and AM447x will use the dmaengine PCM for audio. The daVinci devices will keep using the davinci-pcm for now since I do not have means to test them but the code is written in a way that they can be switched to use the edma-pcm easily (DA850/OMAP-L138 has been tested and audio was working fine). To be able to use the dmaengine PCM I needed to make changes to the edma code in arch/arm/common/ and in drivers/dma/ : Adding support for DMA pause/resume Possibility to select non default event queue/TC for cyclic (audio) dma channels: all devices using the eDMA via dmaengine was assigned to the default EQ/TC (mmc, i2c, spi, etc, and audio). This is not optimal from system performance point of view since sharing the same EQ/TC can cause latency spikes for cyclic channels (long DMA transfers for MMC for example). While debugging the edma to get things sorted out I noticed that the debug was too verbose and the important information was hidden even when the we did not asked for verbose dmaengine debug. I have included some debug cleanups for the edma dmaengine driver also. On the ASoC side: the series has been tested on AM335x and AM447x. I have report that DA850/OMAP-L138 works with this series (which boots w/o DT). Regards, Peter --- Peter Ujfalusi (18): platform_data: edma: Be precise with the paRAM struct dma: edma: Add support for DMA_PAUSE/RESUME operation dma: edma: Set DMA_CYCLIC capability flag arm: common: edma: Select event queue 1 as default when booted with DT arm: common: edma: Save the number of event queues/TCs arm: common: edma: API to request non default queue for a channel DMA: edma: Use different eventq for cyclic channels dma: edma: Implement device_slave_caps callback dma: edma: Simplify direction configuration in edma_config_pset() dma: edma: Reduce debug print verbosity for non verbose debugging dma: edma: Prefix debug prints where the text were identical in prep callbacks dma: edma: Add channel number to debug prints dma: edma: Print the direction value as well when it is not supported ASoC: davinci: Add edma dmaengine platform driver ASoC: davinci-mcasp: Use dmaengine based platform driver for AM335x/447x ASoC: davinci-mcasp: Provide correct filter_data for dmaengine for non-DT boot ASoC: davinci-mcasp: Assign the dma_data earlier in dai_probe callback ASoC: davinci-mcasp: Place constraint on the period size based on FIFO config arch/arm/common/edma.c | 34 ++++++++++++- drivers/dma/edma.c | 92 ++++++++++++++++++++++++++-------- include/linux/platform_data/edma.h | 20 ++++---- sound/soc/davinci/Kconfig | 1 + sound/soc/davinci/Makefile | 2 +- sound/soc/davinci/davinci-mcasp.c | 100 ++++++++++++++++++++++++++++++------- sound/soc/davinci/edma-pcm.c | 77 ++++++++++++++++++++++++++++ sound/soc/davinci/edma-pcm.h | 26 ++++++++++ 8 files changed, 303 insertions(+), 49 deletions(-) create mode 100644 sound/soc/davinci/edma-pcm.c create mode 100644 sound/soc/davinci/edma-pcm.h -- 1.9.0

4 30

[alsa-devel] Low volume ad1983
by Raymond Yau 14 Mar '14

14 Mar '14

>>sam chan wrote: >> i'm having a problem with sound volume being too low. >> i **think** it's due to the levels of my channel map being >> too low..... >The channel map specifies which channel goes to which >physical output. >> my alsa-info... >> http://www.alsa-project.org/db/?f=3ba6278df3c6c4abe5196ad37f8015de7223ca8e >> Simple mixer control 'Master',0 >> Mono: Playback 37 [59%] [-39.00dB] [on] >That is pretty low. >> Simple mixer control 'Loopback Mixing',0 >> Item0: 'Enabled' >Why is this enabled? seem driver bug 1) Should the volume control and playback switch of line out be named as Front instead of PCM ? Node 0x05 [Pin Complex] wcaps 0x400185: Stereo Amp-Out Control: name="PCM Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="PCM Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Jack", index=0, device=0 Amp-Out caps: ofs=0x3d, nsteps=0x3f, stepsize=0x05, mute=1 Amp-Out vals: [0x23 0x23] Pincap 0x00000017: OUT Detect Trigger ImpSense Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 2) dB max of virtual master playback volume is different from the slave when slave dbmax of slaves > 0 control.1 { iface MIXER name 'PCM Playback Volume' value.0 61 value.1 61 comment { access 'read write' type INTEGER count 2 range '0 - 63' dbmin -9150 dbmax 300 dbvalue.0 0 dbvalue.1 0 } } control.3 { iface MIXER name 'Headphone Playback Volume' value.0 61 value.1 61 comment { access 'read write' type INTEGER count 2 range '0 - 63' dbmin -9150 dbmax 300 dbvalue.0 0 dbvalue.1 0 } } control.5 { iface MIXER name 'Speaker Playback Volume' value 61 comment { access 'read write' type INTEGER count 1 range '0 - 63' dbmin -9150 dbmax 300 dbvalue.0 0 } } control.17 { iface MIXER name 'Master Playback Volume' value 37 comment { access 'read write' type INTEGER count 1 range '0 - 63' dbmin -9450 dbmax 0 dbvalue.0 -3900 } }

3 3

[alsa-devel] please ignore my earlier message regarding R16 quirks
by James Miller 14 Mar '14

14 Mar '14

I'm sorry for my ignorance, but I realised that the device has been reporting itself as 1.1 due to the usb hub being 1.1. This is my first week of looking at usb, and its been a steep learning curve, I am finding my way around the code though, and hope to have something working at some point in the near future. I notice that some cards have boot up quirk functions and I am wondering if that is the problem as the R16 can be switched between its audio interface and a card reader. If I have any more questions, is this the correct list to ask them? Many thanks, James

1 0

[alsa-devel] [RFC] ASoC: core: Add support for DAI multicodec
by Benoit Cousson 13 Mar '14

13 Mar '14

From: Misael Lopez Cruz <misael.lopez(a)ti.com> DAI link assumes a one to one mapping between CPU DAI and CODEC. In some cases, the same CPU DAI can be connected to several codecs. This is the case for example, if you connect two mono codecs to the same I2S link in order to have a stereo card. The current ASoC implementation does not allow such setup. Add support for DAI links composed of a single CPU DAI and multiple CODECs. Sound cards have to pass the CODECs array in the corresponding DAI link through a new 'snd_soc_dai_link_codec' struct. Each CODEC in this array is described in the same manner single CODEC DAIs are (either DT/OF node or codec_name). CPU DAI in a multicodec DAI link can have more channels than what each CODEC has. The number of channels each CODEC is responsible for is machine specific, hence it's fixed up in machine drivers in a similar way to what DPCM does. Add few helpers to make the code more readable and fix a trailing space in the header as well. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> [bcousson(a)baylibre.com: Adapt the original series to 3.14] Signed-off-by: Benoit Cousson <bcousson(a)baylibre.com> --- Hi Mark and Liam, I'm not sure this is the only way to address that problem, but both Misa and myself were facing the same issue on different platforms, and we didn't find any straighforward way to do that. To avoid breaking all the exiting ASoC implementation, this approach is adding extra attributes for multiple codecs and does not enforce moving every single codec pointer to a multicodec array. The patch is based on 3.14-rc6. Comments are welcome. Thanks, Benoit --- include/sound/soc.h | 20 ++ sound/soc/soc-core.c | 511 ++++++++++++++++++++++++++++++++++----------------- sound/soc/soc-dapm.c | 39 ++-- sound/soc/soc-pcm.c | 412 ++++++++++++++++++++++++++++------------- 4 files changed, 677 insertions(+), 305 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a00147..c17351c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -818,6 +818,19 @@ struct snd_soc_platform_driver { int (*bespoke_trigger)(struct snd_pcm_substream *, int); }; +struct snd_soc_dai_link_codec { + const char *codec_name; + const struct device_node *codec_of_node; + const char *codec_dai_name; + + struct snd_soc_codec *codec; + struct snd_soc_dai *codec_dai; + + int (*hw_params_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); + struct snd_pcm_hw_params hw_params; +}; + struct snd_soc_platform { const char *name; int id; @@ -866,6 +879,10 @@ struct snd_soc_dai_link { const struct device_node *codec_of_node; /* You MUST specify the DAI name within the codec */ const char *codec_dai_name; + + struct snd_soc_dai_link_codec *codecs; + int num_codecs; + /* * You MAY specify the link's platform/PCM/DMA driver, either by * device name, or by DT/OF node, but not both. Some forms of link @@ -1055,6 +1072,9 @@ struct snd_soc_pcm_runtime { struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai; + struct snd_soc_dai_link_codec *codecs; + int num_codecs; + struct delayed_work delayed_work; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dpcm_root; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fe1df50..1a6c2d0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -557,8 +557,9 @@ static void codec2codec_close_delayed_work(struct work_struct *work) int snd_soc_suspend(struct device *dev) { struct snd_soc_card *card = dev_get_drvdata(dev); + struct snd_soc_dai_link_codec *codecs; struct snd_soc_codec *codec; - int i; + int i, j; /* If the initialization of this soc device failed, there is no codec * associated with it. Just bail out in this case. @@ -578,14 +579,17 @@ int snd_soc_suspend(struct device *dev) /* mute any active DACs */ for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *dai = card->rtd[i].codec_dai; - struct snd_soc_dai_driver *drv = dai->driver; + codecs = card->rtd[i].codecs; + for (j = 0; j < card->rtd[i].num_codecs; j++) { + struct snd_soc_dai *dai = codecs[j].codec_dai; + struct snd_soc_dai_driver *drv = dai->driver; - if (card->rtd[i].dai_link->ignore_suspend) - continue; + if (card->rtd[i].dai_link->ignore_suspend) + continue; - if (drv->ops->digital_mute && dai->playback_active) - drv->ops->digital_mute(dai, 1); + if (drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, 1); + } } /* suspend all pcms */ @@ -616,8 +620,12 @@ int snd_soc_suspend(struct device *dev) /* close any waiting streams and save state */ for (i = 0; i < card->num_rtd; i++) { + codecs = card->rtd[i].codecs; flush_delayed_work(&card->rtd[i].delayed_work); - card->rtd[i].codec->dapm.suspend_bias_level = card->rtd[i].codec->dapm.bias_level; + for (j = 0; j < card->rtd[i].num_codecs; j++) { + codecs[j].codec->dapm.suspend_bias_level = + codecs[j].codec->dapm.bias_level; + } } for (i = 0; i < card->num_rtd; i++) { @@ -700,8 +708,9 @@ static void soc_resume_deferred(struct work_struct *work) { struct snd_soc_card *card = container_of(work, struct snd_soc_card, deferred_resume_work); + struct snd_soc_dai_link_codec *codecs; struct snd_soc_codec *codec; - int i; + int i, j; /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, * so userspace apps are blocked from touching us @@ -762,14 +771,17 @@ static void soc_resume_deferred(struct work_struct *work) /* unmute any active DACs */ for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *dai = card->rtd[i].codec_dai; - struct snd_soc_dai_driver *drv = dai->driver; + codecs = card->rtd[i].codecs; + for (j = 0; j < card->rtd[i].num_codecs; j++) { + struct snd_soc_dai *dai = codecs[j].codec_dai; + struct snd_soc_dai_driver *drv = dai->driver; - if (card->rtd[i].dai_link->ignore_suspend) - continue; + if (card->rtd[i].dai_link->ignore_suspend) + continue; - if (drv->ops->digital_mute && dai->playback_active) - drv->ops->digital_mute(dai, 0); + if (drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, 0); + } } for (i = 0; i < card->num_rtd; i++) { @@ -851,14 +863,51 @@ EXPORT_SYMBOL_GPL(snd_soc_resume); static const struct snd_soc_dai_ops null_dai_ops = { }; +static struct snd_soc_codec *soc_find_codec( + const struct device_node *codec_of_node, + const char *codec_name) +{ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &codec_list, list) { + if (codec_of_node) { + if (codec->dev->of_node != codec_of_node) + continue; + } else { + if (strcmp(codec->name, codec_name)) + continue; + } + + return codec; + } + + return NULL; +} + +static struct snd_soc_dai *soc_find_codec_dai(struct snd_soc_codec *codec, + const char *codec_dai_name) +{ + struct snd_soc_dai *codec_dai; + + list_for_each_entry(codec_dai, &dai_list, list) { + if (codec->dev == codec_dai->dev && + !strcmp(codec_dai->name, codec_dai_name)) { + return codec_dai; + } + } + + return NULL; +} + static int soc_bind_dai_link(struct snd_soc_card *card, int num) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_codec *codec; + struct snd_soc_dai_link_codec *codecs = dai_link->codecs; struct snd_soc_platform *platform; - struct snd_soc_dai *codec_dai, *cpu_dai; + struct snd_soc_dai *cpu_dai; const char *platform_name; + int i; dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num); @@ -883,43 +932,34 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) return -EPROBE_DEFER; } - /* Find CODEC from registered CODECs */ - list_for_each_entry(codec, &codec_list, list) { - if (dai_link->codec_of_node) { - if (codec->dev->of_node != dai_link->codec_of_node) - continue; - } else { - if (strcmp(codec->name, dai_link->codec_name)) - continue; - } - - rtd->codec = codec; - - /* - * CODEC found, so find CODEC DAI from registered DAIs from - * this CODEC - */ - list_for_each_entry(codec_dai, &dai_list, list) { - if (codec->dev == codec_dai->dev && - !strcmp(codec_dai->name, - dai_link->codec_dai_name)) { + rtd->num_codecs = dai_link->num_codecs; + rtd->codecs = dai_link->codecs; - rtd->codec_dai = codec_dai; - } + /* Find CODEC from registered CODECs */ + for (i = 0; i < rtd->num_codecs; i++) { + codecs[i].codec = soc_find_codec(codecs[i].codec_of_node, + codecs[i].codec_name); + if (!codecs[i].codec) { + dev_err(card->dev, "ASoC: CODEC %s not registered\n", + codecs[i].codec_name); + return -EPROBE_DEFER; } + } - if (!rtd->codec_dai) { + for (i = 0; i < rtd->num_codecs; i++) { + codecs[i].codec_dai = soc_find_codec_dai( + codecs[i].codec, + codecs[i].codec_dai_name); + if (!codecs[i].codec_dai) { dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", - dai_link->codec_dai_name); + codecs[i].codec_dai_name); return -EPROBE_DEFER; } } - if (!rtd->codec) { - dev_err(card->dev, "ASoC: CODEC %s not registered\n", - dai_link->codec_name); - return -EPROBE_DEFER; - } + /* Single codec links expect codec and codec_dai in runtime data */ + rtd->codec = codecs[0].codec; + rtd->codec_dai = codecs[0].codec_dai; /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; @@ -994,8 +1034,9 @@ static void soc_remove_codec(struct snd_soc_codec *codec) static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; - int err; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int i, err; /* unregister the rtd device */ if (rtd->dev_registered) { @@ -1006,17 +1047,20 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) } /* remove the CODEC DAI */ - if (codec_dai && codec_dai->probed && - codec_dai->driver->remove_order == order) { - if (codec_dai->driver->remove) { - err = codec_dai->driver->remove(codec_dai); - if (err < 0) - dev_err(codec_dai->dev, - "ASoC: failed to remove %s: %d\n", - codec_dai->name, err); + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codecs[i].codec_dai; + if (codec_dai && codec_dai->probed && + codec_dai->driver->remove_order == order) { + if (codec_dai->driver->remove) { + err = codec_dai->driver->remove(codec_dai); + if (err < 0) + dev_err(codec_dai->dev, + "ASoC: failed to remove %s: %d\n", + codec_dai->name, err); + } + codec_dai->probed = 0; + list_del(&codec_dai->card_list); } - codec_dai->probed = 0; - list_del(&codec_dai->card_list); } /* remove the cpu_dai */ @@ -1044,9 +1088,9 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_codec *codec; + int i; /* remove the platform */ if (platform && platform->probed && @@ -1055,8 +1099,8 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, } /* remove the CODEC-side CODEC */ - if (codec_dai) { - codec = codec_dai->codec; + for (i = 0; i < rtd->num_codecs; i++) { + codec = rtd->codecs[i].codec; if (codec && codec->probed && codec->driver->remove_order == order) soc_remove_codec(codec); @@ -1239,45 +1283,106 @@ static void rtd_release(struct device *dev) kfree(dev); } +static int soc_aux_dev_init(struct snd_soc_card *card, int num) +{ + struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; + struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; + struct snd_soc_codec *codec; + const char *temp; + int ret; + + rtd->card = card; + + codec = soc_find_codec(NULL, aux_dev->codec_name); + if (!codec) + return -EPROBE_DEFER; + + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(codec->dapm.card); + + temp = codec->name_prefix; + codec->name_prefix = NULL; + + /* do machine specific initialization */ + if (aux_dev->init) { + ret = aux_dev->init(&codec->dapm); + if (ret < 0) + return ret; + } + + codec->name_prefix = temp; + + rtd->codec = codec; + + return 0; +} + +static int soc_dai_link_init(struct snd_soc_card *card, int num) +{ + struct snd_soc_dai_link *dai_link = &card->dai_link[num]; + struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; + const char **temp; + int i, ret; + + rtd->card = card; + + temp = devm_kzalloc(card->dev, rtd->num_codecs * sizeof(char *), + GFP_KERNEL); + if (!temp) + return -ENOMEM; + + for (i = 0; i < rtd->num_codecs; i++) { + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(codecs[i].codec->dapm.card); + + /* machine controls, routes and widgets are not prefixed */ + temp[i] = codecs[i].codec->name_prefix; + codecs[i].codec->name_prefix = NULL; + } + + /* do machine specific initialization */ + if (dai_link->init) { + ret = dai_link->init(rtd); + if (ret < 0) + return ret; + } + + for (i = 0; i < rtd->num_codecs; i++) + codecs[i].codec->name_prefix = temp[i]; + + devm_kfree(card->dev, temp); + + return 0; +} + static int soc_post_component_init(struct snd_soc_card *card, - struct snd_soc_codec *codec, int num, int dailess) { struct snd_soc_dai_link *dai_link = NULL; struct snd_soc_aux_dev *aux_dev = NULL; struct snd_soc_pcm_runtime *rtd; - const char *temp, *name; + const char *name; int ret = 0; if (!dailess) { dai_link = &card->dai_link[num]; rtd = &card->rtd[num]; name = dai_link->name; + ret = soc_dai_link_init(card, num); } else { aux_dev = &card->aux_dev[num]; rtd = &card->rtd_aux[num]; name = aux_dev->name; + ret = soc_aux_dev_init(card, num); } - rtd->card = card; - - /* machine controls, routes and widgets are not prefixed */ - temp = codec->name_prefix; - codec->name_prefix = NULL; - /* do machine specific initialization */ - if (!dailess && dai_link->init) - ret = dai_link->init(rtd); - else if (dailess && aux_dev->init) - ret = aux_dev->init(&codec->dapm); if (ret < 0) { dev_err(card->dev, "ASoC: failed to init %s: %d\n", name, ret); return ret; } - codec->name_prefix = temp; /* register the rtd device */ - rtd->codec = codec; - rtd->dev = kzalloc(sizeof(struct device), GFP_KERNEL); if (!rtd->dev) return -ENOMEM; @@ -1295,7 +1400,7 @@ static int soc_post_component_init(struct snd_soc_card *card, if (ret < 0) { /* calling put_device() here to free the rtd->dev */ put_device(rtd->dev); - dev_err(card->dev, + dev_err(rtd->dev, "ASoC: failed to register runtime device: %d\n", ret); return ret; } @@ -1304,13 +1409,13 @@ static int soc_post_component_init(struct snd_soc_card *card, /* add DAPM sysfs entries for this codec */ ret = snd_soc_dapm_sys_add(rtd->dev); if (ret < 0) - dev_err(codec->dev, + dev_err(rtd->dev, "ASoC: failed to add codec dapm sysfs entries: %d\n", ret); /* add codec sysfs entries */ ret = device_create_file(rtd->dev, &dev_attr_codec_reg); if (ret < 0) - dev_err(codec->dev, + dev_err(rtd->dev, "ASoC: failed to add codec sysfs files: %d\n", ret); #ifdef CONFIG_DEBUG_FS @@ -1332,9 +1437,9 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_platform *platform = rtd->platform; - int ret; + int i, ret; /* probe the CPU-side component, if it is a CODEC */ if (cpu_dai->codec && @@ -1345,12 +1450,14 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, return ret; } - /* probe the CODEC-side component */ - if (!codec_dai->codec->probed && - codec_dai->codec->driver->probe_order == order) { - ret = soc_probe_codec(card, codec_dai->codec); - if (ret < 0) - return ret; + /* probe the CODEC-side components */ + for (i = 0; i < rtd->num_codecs; i++) { + if (!codecs[i].codec->probed && + codecs[i].codec->driver->probe_order == order) { + ret = soc_probe_codec(card, codecs[i].codec); + if (ret < 0) + return ret; + } } /* probe the platform */ @@ -1364,24 +1471,61 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, return 0; } +static int soc_link_dai_widgets(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link, + struct snd_soc_dai *cpu_dai, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_dapm_widget *play_w, *capture_w; + int ret; + + /* link the DAI widgets */ + play_w = codec_dai->playback_widget; + capture_w = cpu_dai->capture_widget; + if (play_w && capture_w) { + ret = snd_soc_dapm_new_pcm(card, dai_link->params, + capture_w, play_w); + if (ret != 0) { + dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", + play_w->name, capture_w->name, ret); + return ret; + } + } + + play_w = cpu_dai->playback_widget; + capture_w = codec_dai->capture_widget; + if (play_w && capture_w) { + ret = snd_soc_dapm_new_pcm(card, dai_link->params, + capture_w, play_w); + if (ret != 0) { + dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", + play_w->name, capture_w->name, ret); + return ret; + } + } + + return 0; +} + + static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dapm_widget *play_w, *capture_w; - int ret; + int i, ret; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, num, order); /* config components */ cpu_dai->platform = platform; - codec_dai->card = card; cpu_dai->card = card; + for (i = 0; i < rtd->num_codecs; i++) + codecs[i].codec_dai->card = card; /* set default power off timeout */ rtd->pmdown_time = pmdown_time; @@ -1413,27 +1557,31 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } /* probe the CODEC DAI */ - if (!codec_dai->probed && codec_dai->driver->probe_order == order) { - if (codec_dai->driver->probe) { - ret = codec_dai->driver->probe(codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, - "ASoC: failed to probe CODEC DAI %s: %d\n", - codec_dai->name, ret); - return ret; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (!codec_dai->probed && + codec_dai->driver->probe_order == order) { + if (codec_dai->driver->probe) { + ret = codec_dai->driver->probe(codec_dai); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: failed to probe CODEC DAI %s: %d\n", + codec_dai->name, ret); + return ret; + } } - } - /* mark codec_dai as probed and add to card dai list */ - codec_dai->probed = 1; - list_add(&codec_dai->card_list, &card->dai_dev_list); + /* mark codec_dai as probed and add to card dai list */ + codec_dai->probed = 1; + list_add(&codec_dai->card_list, &card->dai_dev_list); + } } /* complete DAI probe during last probe */ if (order != SND_SOC_COMP_ORDER_LAST) return 0; - ret = soc_post_component_init(card, codec, num, 0); + ret = soc_post_component_init(card, num, 0); if (ret) return ret; @@ -1465,35 +1613,21 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) codec2codec_close_delayed_work); /* link the DAI widgets */ - play_w = codec_dai->playback_widget; - capture_w = cpu_dai->capture_widget; - if (play_w && capture_w) { - ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); - if (ret != 0) { - dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", - play_w->name, capture_w->name, ret); + for (i = 0; i < rtd->num_codecs; i++) { + ret = soc_link_dai_widgets(card, dai_link, + cpu_dai, codecs[i].codec_dai); + if (ret) return ret; - } - } - - play_w = cpu_dai->playback_widget; - capture_w = codec_dai->capture_widget; - if (play_w && capture_w) { - ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); - if (ret != 0) { - dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", - play_w->name, capture_w->name, ret); - return ret; - } } } } /* add platform data for AC97 devices */ - if (rtd->codec_dai->driver->ac97_control) - snd_ac97_dev_add_pdata(codec->ac97, rtd->cpu_dai->ac97_pdata); + for (i = 0; i < rtd->num_codecs; i++) { + if (codecs[i].codec_dai->driver->ac97_control) + snd_ac97_dev_add_pdata(codecs[i].codec->ac97, + rtd->cpu_dai->ac97_pdata); + } return 0; } @@ -1501,31 +1635,42 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) #ifdef CONFIG_SND_SOC_AC97_BUS static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) { - int ret; + int i, ret; + codec_dai; + codec; - /* Only instantiate AC97 if not already done by the adaptor - * for the generic AC97 subsystem. - */ - if (rtd->codec_dai->driver->ac97_control && !rtd->codec->ac97_registered) { - /* - * It is possible that the AC97 device is already registered to - * the device subsystem. This happens when the device is created - * via snd_ac97_mixer(). Currently only SoC codec that does so - * is the generic AC97 glue but others migh emerge. - * - * In those cases we don't try to register the device again. + for (i = 0; i < rtd->num_codecs; i++) { + codec = rtd->codec; + codec_dai = rtd->codec_dai; + + /* Only instantiate AC97 if not already done by the adaptor + * for the generic AC97 subsystem. */ - if (!rtd->codec->ac97_created) - return 0; + if (codec_dai->driver->ac97_control && + !codec->ac97_registered) { + /* + * It is possible that the AC97 device is already + * registered to the device subsystem. This happens when + * the device is created via snd_ac97_mixer(). Currently + * only SoC codec that does so is the generic AC97 glue + * but others migh emerge. + * + * In those cases we don't try to register the device + * again. + */ + if (!codec->ac97_created) + return 0; + + ret = soc_ac97_dev_register(codec); + if (ret < 0) { + dev_err(codec->dev, + "ASoC: AC97 device register failed: %d\n", + ret); + return ret; + } - ret = soc_ac97_dev_register(rtd->codec); - if (ret < 0) { - dev_err(rtd->codec->dev, - "ASoC: AC97 device register failed: %d\n", ret); - return ret; + codec->ac97_registered = 1; } - - rtd->codec->ac97_registered = 1; } return 0; } @@ -1582,7 +1727,7 @@ found: if (ret < 0) return ret; - ret = soc_post_component_init(card, codec, num, 1); + ret = soc_post_component_init(card, num, 1); out: return ret; @@ -3694,6 +3839,33 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); +static int snd_soc_init_multicodec(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) +{ + /* Legacy codec/codec_dai link is a single entry in multicodec */ + if (dai_link->codec_name || dai_link->codec_of_node || + dai_link->codec_dai_name) { + dai_link->num_codecs = 1; + + dai_link->codecs = devm_kzalloc(card->dev, + sizeof(struct snd_soc_dai_link_codec), + GFP_KERNEL); + if (!dai_link->codecs) + return -ENOMEM; + + dai_link->codecs[0].codec_name = dai_link->codec_name; + dai_link->codecs[0].codec_of_node = dai_link->codec_of_node; + dai_link->codecs[0].codec_dai_name = dai_link->codec_dai_name; + } + + if (!dai_link->codecs) { + dev_err(card->dev, "ASoC: DAI link has no CODECs\n"); + return -EINVAL; + } + + return 0; +} + /** * snd_soc_register_card - Register a card with the ASoC core * @@ -3702,7 +3874,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); */ int snd_soc_register_card(struct snd_soc_card *card) { - int i, ret; + int i, j, ret; if (!card->name || !card->dev) return -EINVAL; @@ -3710,22 +3882,29 @@ int snd_soc_register_card(struct snd_soc_card *card) for (i = 0; i < card->num_links; i++) { struct snd_soc_dai_link *link = &card->dai_link[i]; - /* - * Codec must be specified by 1 of name or OF node, - * not both or neither. - */ - if (!!link->codec_name == !!link->codec_of_node) { - dev_err(card->dev, - "ASoC: Neither/both codec name/of_node are set for %s\n", - link->name); - return -EINVAL; + ret = snd_soc_init_multicodec(card, link); + if (ret) { + dev_err(card->dev, "ASoC: failed to init multicodec\n"); + return ret; } - /* Codec DAI name must be specified */ - if (!link->codec_dai_name) { - dev_err(card->dev, - "ASoC: codec_dai_name not set for %s\n", - link->name); - return -EINVAL; + + for (j = 0; j < link->num_codecs; j++) { + /* + * Codec must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->codecs[j].codec_name == + !!link->codecs[j].codec_of_node) { + dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + /* Codec DAI name must be specified */ + if (!link->codecs[j].codec_dai_name) { + dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n", + link->name); + return -EINVAL; + } } /* diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b9dc6ac..f4b4e2a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2192,12 +2192,8 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, } EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power); -/* show dapm widget status in sys fs */ -static ssize_t dapm_widget_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t dapm_widget_show_codec(struct snd_soc_codec *codec, char *buf) { - struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); - struct snd_soc_codec *codec =rtd->codec; struct snd_soc_dapm_widget *w; int count = 0; char *state = "not set"; @@ -2250,6 +2246,22 @@ static ssize_t dapm_widget_show(struct device *dev, return count; } +/* show dapm widget status in sys fs */ +static ssize_t dapm_widget_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + struct snd_soc_codec *codec; + int i, count = 0; + + for (i = 0; i < rtd->num_codecs; i++) { + codec = rtd->codecs[i].codec; + count += dapm_widget_show_codec(codec, buf + count); + } + + return count; +} + static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); int snd_soc_dapm_sys_add(struct device *dev) @@ -3684,13 +3696,12 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) } } -static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, - int event) +static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *cpu_dai, + struct snd_soc_dai *codec_dai, + int stream, int event) { - struct snd_soc_dapm_widget *w_cpu, *w_codec; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { w_cpu = cpu_dai->playback_widget; @@ -3756,9 +3767,15 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int event) { struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int i; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - soc_dapm_stream_event(rtd, stream, event); + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codecs[i].codec_dai; + soc_dapm_stream_event(rtd, cpu_dai, codec_dai, stream, event); + } mutex_unlock(&card->dapm_mutex); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 47e1ce7..9d0bb9e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -7,7 +7,7 @@ * Copyright (C) 2010 Texas Instruments Inc. * * Authors: Liam Girdwood <lrg(a)ti.com> - * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -230,32 +230,67 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, } } -static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, - struct snd_soc_pcm_stream *codec_stream, - struct snd_soc_pcm_stream *cpu_stream) +static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_pcm_hardware *hw = &runtime->hw; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_driver *cpu_dai_drv = rtd->cpu_dai->driver; + struct snd_soc_dai_driver *codec_dai_drv; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; + struct snd_soc_pcm_stream *codec_stream; + struct snd_soc_pcm_stream *cpu_stream; + unsigned int chan_min = 0, chan_max = UINT_MAX; + unsigned int rate_min = 0, rate_max = UINT_MAX; + unsigned int rates = UINT_MAX; + unsigned int sig_bits; + u64 formats = ULLONG_MAX; + int i; - hw->channels_min = max(codec_stream->channels_min, - cpu_stream->channels_min); - hw->channels_max = min(codec_stream->channels_max, - cpu_stream->channels_max); - if (hw->formats) - hw->formats &= codec_stream->formats & cpu_stream->formats; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_stream = &cpu_dai_drv->playback; else - hw->formats = codec_stream->formats & cpu_stream->formats; - hw->rates = snd_pcm_rate_mask_intersect(codec_stream->rates, - cpu_stream->rates); + cpu_stream = &cpu_dai_drv->capture; - hw->rate_min = 0; - hw->rate_max = UINT_MAX; + /* first calculate min/max only for CODECs in the DAI link */ + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai_drv = codecs[i].codec_dai->driver; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &codec_dai_drv->playback; + else + codec_stream = &codec_dai_drv->capture; + chan_min = max(chan_min, codec_stream->channels_min); + chan_max = min(chan_max, codec_stream->channels_max); + rate_min = max(rate_min, codec_stream->rate_min); + rate_max = min_not_zero(rate_max, codec_stream->rate_max); + formats &= codec_stream->formats; + rates = snd_pcm_rate_mask_intersect(codec_stream->rates, rates); + } + + /* + * chan min/max cannot be enforced if there are multiple CODEC DAIs + * connected to a single CPU DAI, use CPU DAI's directly and let + * channel allocation be fixed up later + */ + if (rtd->num_codecs > 1) { + chan_min = cpu_stream->channels_min; + chan_max = cpu_stream->channels_max; + } + + hw->channels_min = max(chan_min, cpu_stream->channels_min); + hw->channels_max = min(chan_max, cpu_stream->channels_max); + if (hw->formats) + hw->formats &= formats & cpu_stream->formats; + else + hw->formats = formats & cpu_stream->formats; + hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_stream->rates); snd_pcm_limit_hw_rates(runtime); hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); - hw->rate_min = max(hw->rate_min, codec_stream->rate_min); + hw->rate_min = max(hw->rate_min, rate_min); hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max); - hw->rate_max = min_not_zero(hw->rate_max, codec_stream->rate_max); + hw->rate_max = min_not_zero(hw->rate_max, rate_max); } /* @@ -268,16 +303,18 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; - struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; - int ret = 0; + struct snd_soc_dai *codec_dai; + const char *codec_dai_name = "multicodec"; + int i, ret = 0; pinctrl_pm_select_default_state(cpu_dai->dev); - pinctrl_pm_select_default_state(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) + pinctrl_pm_select_default_state(codecs[i].codec_dai->dev); pm_runtime_get_sync(cpu_dai->dev); - pm_runtime_get_sync(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) + pm_runtime_get_sync(codecs[i].codec_dai->dev); pm_runtime_get_sync(platform->dev); mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -301,12 +338,17 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - if (codec_dai->driver->ops && codec_dai->driver->ops->startup) { - ret = codec_dai->driver->ops->startup(substream, codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, "ASoC: can't open codec" - " %s: %d\n", codec_dai->name, ret); - goto codec_dai_err; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops && codec_dai->driver->ops->startup) { + ret = codec_dai->driver->ops->startup(substream, + codec_dai); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: can't open codec %s: %d\n", + codec_dai->name, ret); + goto codec_dai_err; + } } } @@ -324,13 +366,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto dynamic; /* Check that the codec and cpu DAIs are compatible */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->playback, - &cpu_dai_drv->playback); - } else { - soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->capture, - &cpu_dai_drv->capture); - } + soc_pcm_init_runtime_hw(substream); + + if (rtd->num_codecs == 1) + codec_dai_name = rtd->codec_dai->name; if (soc_pcm_has_symmetry(substream)) runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; @@ -338,22 +377,22 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) ret = -EINVAL; if (!runtime->hw.rates) { printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n", - codec_dai->name, cpu_dai->name); + codec_dai_name, cpu_dai->name); goto config_err; } if (!runtime->hw.formats) { printk(KERN_ERR "ASoC: %s <-> %s No matching formats\n", - codec_dai->name, cpu_dai->name); + codec_dai_name, cpu_dai->name); goto config_err; } if (!runtime->hw.channels_min || !runtime->hw.channels_max || runtime->hw.channels_min > runtime->hw.channels_max) { printk(KERN_ERR "ASoC: %s <-> %s No matching channels\n", - codec_dai->name, cpu_dai->name); + codec_dai_name, cpu_dai->name); goto config_err; } - soc_pcm_apply_msb(substream, codec_dai); + soc_pcm_apply_msb(substream, codecs[0].codec_dai); soc_pcm_apply_msb(substream, cpu_dai); /* Symmetry only applies if we've already got an active stream. */ @@ -363,14 +402,17 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto config_err; } - if (codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream, codec_dai); - if (ret != 0) - goto config_err; + for (i = 0; i < rtd->num_codecs; i++) { + if (codecs[i].codec_dai->active) { + ret = soc_pcm_apply_symmetry(substream, + codecs[i].codec_dai); + if (ret != 0) + goto config_err; + } } pr_debug("ASoC: %s <-> %s info:\n", - codec_dai->name, cpu_dai->name); + codec_dai_name, cpu_dai->name); pr_debug("ASoC: rate mask 0x%x\n", runtime->hw.rates); pr_debug("ASoC: min ch %d max ch %d\n", runtime->hw.channels_min, runtime->hw.channels_max); @@ -380,14 +422,18 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) dynamic: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active++; - codec_dai->playback_active++; + for (i = 0; i < rtd->num_codecs; i++) + codecs[i].codec_dai->playback_active++; } else { cpu_dai->capture_active++; - codec_dai->capture_active++; + for (i = 0; i < rtd->num_codecs; i++) + codecs[i].codec_dai->capture_active++; } cpu_dai->active++; - codec_dai->active++; - rtd->codec->active++; + for (i = 0; i < rtd->num_codecs; i++) { + codecs[i].codec_dai->active++; + codecs[i].codec->active++; + } mutex_unlock(&rtd->pcm_mutex); return 0; @@ -396,10 +442,15 @@ config_err: rtd->dai_link->ops->shutdown(substream); machine_err: - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); + i = rtd->num_codecs; codec_dai_err: + while (--i >= 0) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + } + if (platform->driver->ops && platform->driver->ops->close) platform->driver->ops->close(substream); @@ -410,10 +461,13 @@ out: mutex_unlock(&rtd->pcm_mutex); pm_runtime_put(platform->dev); - pm_runtime_put(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) + pm_runtime_put(codecs[i].codec_dai->dev); pm_runtime_put(cpu_dai->dev); - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) { + if (!codecs[i].codec_dai->active) + pinctrl_pm_select_sleep_state(codecs[i].codec_dai->dev); + } if (!cpu_dai->active) pinctrl_pm_select_sleep_state(cpu_dai->dev); @@ -429,7 +483,7 @@ static void close_delayed_work(struct work_struct *work) { struct snd_soc_pcm_runtime *rtd = container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = rtd->codecs[0].codec_dai; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -457,36 +511,47 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dai *codec_dai; + int i; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active--; - codec_dai->playback_active--; + for (i = 0; i < rtd->num_codecs; i++) + codecs[i].codec_dai->playback_active--; } else { cpu_dai->capture_active--; - codec_dai->capture_active--; + for (i = 0; i < rtd->num_codecs; i++) + codecs[i].codec_dai->capture_active--; } cpu_dai->active--; - codec_dai->active--; - codec->active--; + for (i = 0; i < rtd->num_codecs; i++) { + codecs[i].codec_dai->active--; + codecs[i].codec->active--; + } /* clear the corresponding DAIs rate when inactive */ if (!cpu_dai->active) cpu_dai->rate = 0; - if (!codec_dai->active) - codec_dai->rate = 0; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (!codec_dai->active) + codec_dai->rate = 0; + } if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + } if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); @@ -496,8 +561,13 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (!rtd->pmdown_time || codec->ignore_pmdown_time || - rtd->dai_link->ignore_pmdown_time) { + int ignore = rtd->dai_link->ignore_pmdown_time || + !rtd->pmdown_time; + + for (i = 0; (i < rtd->num_codecs) && !ignore; i++) + ignore |= codecs[i].codec->ignore_pmdown_time; + + if (ignore) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, @@ -518,10 +588,13 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) mutex_unlock(&rtd->pcm_mutex); pm_runtime_put(platform->dev); - pm_runtime_put(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) + pm_runtime_put(codecs[i].codec_dai->dev); pm_runtime_put(cpu_dai->dev); - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) { + if (!codecs[i].codec_dai->active) + pinctrl_pm_select_sleep_state(codecs[i].codec_dai->dev); + } if (!cpu_dai->active) pinctrl_pm_select_sleep_state(cpu_dai->dev); @@ -537,9 +610,10 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; + struct snd_soc_dai *codec_dai; + int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -561,12 +635,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - if (codec_dai->driver->ops && codec_dai->driver->ops->prepare) { - ret = codec_dai->driver->ops->prepare(substream, codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, "ASoC: DAI prepare error: %d\n", - ret); - goto out; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops && codec_dai->driver->ops->prepare) { + ret = codec_dai->driver->ops->prepare(substream, + codec_dai); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: DAI prepare error: %d\n", ret); + goto out; + } } } @@ -589,7 +667,9 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(rtd, substream->stream, SND_SOC_DAPM_STREAM_START); - snd_soc_dai_digital_mute(codec_dai, 0, substream->stream); + for (i = 0; i < rtd->num_codecs; i++) + snd_soc_dai_digital_mute(codecs[i].codec_dai, 0, + substream->stream); out: mutex_unlock(&rtd->pcm_mutex); @@ -606,9 +686,10 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; + struct snd_soc_dai *codec_dai; + int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -625,13 +706,38 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (codec_dai->driver->ops && codec_dai->driver->ops->hw_params) { - ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, "ASoC: can't set %s hw params:" - " %d\n", codec_dai->name, ret); - goto codec_err; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_pcm_hw_params *codec_params; + + codec_dai = codecs[i].codec_dai; + codec_params = &codecs[i].hw_params; + + /* copy params for each codec */ + memcpy(codec_params, params, sizeof(struct snd_pcm_hw_params)); + + if (codecs[i].hw_params_fixup) { + ret = codecs[i].hw_params_fixup(rtd, codec_params); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: multicodec hw_params fixup failed %d\n", + ret); + goto codec_err; + } } + + if (codec_dai->driver->ops && + codec_dai->driver->ops->hw_params) { + ret = codec_dai->driver->ops->hw_params(substream, + codec_params, codec_dai); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: can't set %s hw params: %d\n", + codec_dai->name, ret); + goto codec_err; + } + } + + codec_dai->rate = params_rate(codec_params); } if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_params) { @@ -672,10 +778,16 @@ platform_err: cpu_dai->driver->ops->hw_free(substream, cpu_dai); interface_err: - if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); + i = rtd->num_codecs; codec_err: + while (--i >= 0) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + codec_dai->rate = 0; + } + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream); @@ -690,9 +802,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai; bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int i; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -703,16 +817,21 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) cpu_dai->sample_bits = 0; } - if (codec_dai->active == 1) { - codec_dai->rate = 0; - codec_dai->channels = 0; - codec_dai->sample_bits = 0; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->active == 1) + codec_dai->rate = 0; + codec_dai->channels = 0; + codec_dai->sample_bits = 0; } /* apply codec digital mute */ - if ((playback && codec_dai->playback_active == 1) || - (!playback && codec_dai->capture_active == 1)) - snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); + for (i = 0; i < rtd->num_codecs; i++) { + if ((playback && codecs[i].codec_dai->playback_active == 1) || + (!playback && codecs[i].codec_dai->capture_active == 1)) + snd_soc_dai_digital_mute(codecs[i].codec_dai, 1, + substream->stream); + } /* free any machine hw params */ if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) @@ -723,8 +842,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) platform->driver->ops->hw_free(substream); /* now free hw params for the DAIs */ - if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + } if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); @@ -737,14 +859,19 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (codec_dai->driver->ops && codec_dai->driver->ops->trigger) { - ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); - if (ret < 0) - return ret; + struct snd_soc_dai *codec_dai; + int i, ret; + + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops && codec_dai->driver->ops->trigger) { + ret = codec_dai->driver->ops->trigger(substream, + cmd, codec_dai); + if (ret < 0) + return ret; + } } if (platform->driver->ops && platform->driver->ops->trigger) { @@ -766,15 +893,20 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (codec_dai->driver->ops && - codec_dai->driver->ops->bespoke_trigger) { - ret = codec_dai->driver->ops->bespoke_trigger(substream, cmd, codec_dai); - if (ret < 0) - return ret; + struct snd_soc_dai *codec_dai; + int i, ret; + + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops && + codec_dai->driver->ops->bespoke_trigger) { + ret = codec_dai->driver->ops->bespoke_trigger(substream, + cmd, codec_dai); + if (ret < 0) + return ret; + } } if (platform->driver->bespoke_trigger) { @@ -799,11 +931,14 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai; struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t offset = 0; snd_pcm_sframes_t delay = 0; + snd_pcm_sframes_t codec_delay = 0; + int i; if (platform->driver->ops && platform->driver->ops->pointer) offset = platform->driver->ops->pointer(substream); @@ -811,11 +946,22 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops && cpu_dai->driver->ops->delay) delay += cpu_dai->driver->ops->delay(substream, cpu_dai); - if (codec_dai->driver->ops && codec_dai->driver->ops->delay) - delay += codec_dai->driver->ops->delay(substream, codec_dai); + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->ops && codec_dai->driver->ops->delay) + codec_delay = max(codec_delay, + codec_dai->driver->ops->delay(substream, + codec_dai)); + } + delay += codec_delay; + /* + * None of the existing platform drivers implement delay(), so + * for now the codec_dai of first multicodec entry is used + */ if (platform->driver->delay) - delay += platform->driver->delay(substream, codec_dai); + delay += platform->driver->delay(substream, + codecs[0].codec_dai); runtime->delay = delay; @@ -2129,22 +2275,29 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai_link_codec *codecs = rtd->codecs; + struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + int i; if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { playback = rtd->dai_link->dpcm_playback; capture = rtd->dai_link->dpcm_capture; } else { - if (codec_dai->driver->playback.channels_min && - cpu_dai->driver->playback.channels_min) - playback = 1; - if (codec_dai->driver->capture.channels_min && - cpu_dai->driver->capture.channels_min) - capture = 1; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = codecs[i].codec_dai; + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) + playback++; + if (codec_dai->driver->capture.channels_min && + cpu_dai->driver->capture.channels_min) + capture++; + } + capture = (rtd->num_codecs == capture); + playback = (rtd->num_codecs == playback); } if (rtd->dai_link->playback_only) { @@ -2170,7 +2323,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->dai_link->stream_name); else snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); + rtd->dai_link->stream_name, + (rtd->num_codecs > 1) ? + "multicodec" : rtd->codec_dai->name, num); ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback, capture, &pcm); @@ -2243,8 +2398,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) pcm->private_free = platform->driver->pcm_free; out: - dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", codec_dai->name, - cpu_dai->name); + dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", + (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name, + cpu_dai->name); return ret; } -- 1.8.3.2

2 5

[alsa-devel] [GIT PULL] ASoC updates for v3.15
by Mark Brown 13 Mar '14

13 Mar '14

The following changes since commit e00447fafbf7daf2cd49205b97e63d9734068a4f: ASoC: tlv320aic31xx: Add basic codec driver implementation (2014-03-13 12:09:51 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/tlv320aic31xx-fixes for you to fetch changes up to 9296f4da3bafa23d8b9abc5cd271a66ea8f90cd2: ASoC: tlv320aic31xx: Staticise non-exported symbols (2014-03-13 17:44:22 +0000) ---------------------------------------------------------------- ASoC: Fixes for tlv320aic31xx A few fixes for issues not yet identified by the zero day tester due to the last pull request being rushed in order to clean up the problem with the git mismerge. ---------------------------------------------------------------- Jyri Sarha (2): ASoC: tlv320aic31xx: Fix unused variable warning from aic31xx_clk_off ASoC: tlv320aic31xx: Remove snd_soc_codec_set_cache_io() call Mark Brown (1): ASoC: tlv320aic31xx: Staticise non-exported symbols sound/soc/codecs/tlv320aic31xx.c | 20 +++++--------------- 1 file changed, 5 insertions(+), 15 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Remove snd_soc_codec_set_cache_io() call
by Jyri Sarha 13 Mar '14

13 Mar '14

Remove snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP) call and codec->control_data = aic31xx->regmap assignment since that already done by core. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- This patch should fix the last minute breakkage of v3.15 ASoC updates build. sound/soc/codecs/tlv320aic31xx.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index c9f6989..625b184 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1051,18 +1051,9 @@ static int aic31xx_codec_probe(struct snd_soc_codec *codec) dev_dbg(aic31xx->dev, "## %s\n", __func__); aic31xx = snd_soc_codec_get_drvdata(codec); - codec->control_data = aic31xx->regmap; aic31xx->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - - if (ret != 0) { - dev_err(codec->dev, "snd_soc_codec_set_cache_io failed %d\n", - ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) { aic31xx->disable_nb[i].nb.notifier_call = aic31xx_regulator_event; -- 1.7.9.5

2 1

[alsa-devel] [PATCH v6] ASoC: tlv320aic31xx: Fix unused variable warning from aic31xx_clk_off
by Jyri Sarha 13 Mar '14

13 Mar '14

Fix "warning: unused variable 'aic31xx'" from function 'aic31xx_clk_off'. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index e626b56..c9f6989 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -945,7 +945,6 @@ static void aic31xx_clk_on(struct snd_soc_codec *codec) static void aic31xx_clk_off(struct snd_soc_codec *codec) { - struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); u8 mask = AIC31XX_PM_MASK; u8 off = 0; -- 1.7.9.5

2 1

[alsa-devel] [PATCH v6] ASoC: tlv320aic31xx: Add basic codec driver implementation
by Jyri Sarha 13 Mar '14

13 Mar '14

This commit adds a bare bones driver support for TLV320AIC31XX family audio codecs. The driver adds basic stereo playback trough headphone and speaker outputs and mono capture trough microphone inputs. The driver is currently missing support at least for mini DSP features and jack detection. I have tested the driver only on TLV320AIC3111, but based on the data sheets TLV320AIC3100, TLV320AIC3110, and TLV320AIC3120 should work Ok too. The base for the implementation was taken from: git@gitorious.org:ti-codecs/ti-codecs.git ajitk/topics/k3.10.1-aic31xx -branch at commit 77504eba0294764e9e63b4a0c696b44db187cd13. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Since v5 of this patch: - Fix "warning: unused variable 'aic31xx'" from function 'aic31xx_clk_off' .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 8 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic31xx.c | 1294 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ 6 files changed, 1627 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt new file mode 100644 index 0000000..74c66de --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt @@ -0,0 +1,61 @@ +Texas Instruments - tlv320aic31xx Codec module + +The tlv320aic31xx serial control bus communicates through I2C protocols + +Required properties: + +- compatible - "string" - One of: + "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp + "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp + "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP) + "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP) + "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP) + "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP) + +- reg - <int> - I2C slave address + + +Optional properties: + +- gpio-reset - gpio pin number used for codec reset +- ai31xx-micbias-vg - MicBias Voltage setting + 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V + 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V + 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD + If this node is not mentioned or if the value is unknown, then + micbias is set to 2.0V. +- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply, + DVDD-supply : power supplies for the device as covered in + Documentation/devicetree/bindings/regulator/regulator.txt + +CODEC output pins: + * HPL + * HPR + * SPL, devices with stereo speaker amp + * SPR, devices with stereo speaker amp + * SPK, devices with mono speaker amp + * MICBIAS + +CODEC input pins: + * MIC1LP + * MIC1RP + * MIC1LM + +The pins can be used in referring sound node's audio-routing property. + +Example: +#include <dt-bindings/sound/tlv320aic31xx-micbias.h> + +tlv320aic31xx: tlv320aic31xx@18 { + compatible = "ti,tlv320aic311x"; + reg = <0x18>; + + ai31xx-micbias-vg = <MICBIAS_OFF>; + + HPVDD-supply = <&regulator>; + SPRVDD-supply = <&regulator>; + SPLVDD-supply = <&regulator>; + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; +}; diff --git a/include/dt-bindings/sound/tlv320aic31xx-micbias.h b/include/dt-bindings/sound/tlv320aic31xx-micbias.h new file mode 100644 index 0000000..f5cb772 --- /dev/null +++ b/include/dt-bindings/sound/tlv320aic31xx-micbias.h @@ -0,0 +1,8 @@ +#ifndef __DT_TLV320AIC31XX_MICBIAS_H +#define __DT_TLV320AIC31XX_MICBIAS_H + +#define MICBIAS_2_0V 1 +#define MICBIAS_2_5V 2 +#define MICBIAS_AVDDV 3 + +#endif /* __DT_TLV320AIC31XX_MICBIAS_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e19b64f..af3c049 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -83,6 +83,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TAS5086 if I2C select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER + select SND_SOC_TLV320AIC31XX if I2C select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C @@ -428,6 +429,9 @@ config SND_SOC_TLV320AIC26 tristate depends on SPI +config SND_SOC_TLV320AIC31XX + tristate + config SND_SOC_TLV320AIC32X4 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 90ab24b..018e3f3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -75,6 +75,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o +snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o @@ -215,6 +216,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o +obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c new file mode 100644 index 0000000..bdc0d8b --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -0,0 +1,1294 @@ +/* + * ALSA SoC TLV320AIC31XX codec driver + * + * Copyright (C) 2014 Texas Instruments, Inc. + * + * Author: Jyri Sarha <jsarha(a)ti.com> + * + * Based on ground work by: Ajit Kulkarni <x0175765(a)ti.com> + * + * This package 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 PACKAGE IS PROVIDED AS IS AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + * + * The TLV320AIC31xx series of audio codec is a low-power, highly integrated + * high performance codec which provides a stereo DAC, a mono ADC, + * and mono/stereo Class-D speaker driver. + */ + +#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/gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <dt-bindings/sound/tlv320aic31xx-micbias.h> + +#include "tlv320aic31xx.h" + +static const struct reg_default aic31xx_reg_defaults[] = { + { AIC31XX_CLKMUX, 0x00 }, + { AIC31XX_PLLPR, 0x11 }, + { AIC31XX_PLLJ, 0x04 }, + { AIC31XX_PLLDMSB, 0x00 }, + { AIC31XX_PLLDLSB, 0x00 }, + { AIC31XX_NDAC, 0x01 }, + { AIC31XX_MDAC, 0x01 }, + { AIC31XX_DOSRMSB, 0x00 }, + { AIC31XX_DOSRLSB, 0x80 }, + { AIC31XX_NADC, 0x01 }, + { AIC31XX_MADC, 0x01 }, + { AIC31XX_AOSR, 0x80 }, + { AIC31XX_IFACE1, 0x00 }, + { AIC31XX_DATA_OFFSET, 0x00 }, + { AIC31XX_IFACE2, 0x00 }, + { AIC31XX_BCLKN, 0x01 }, + { AIC31XX_DACSETUP, 0x14 }, + { AIC31XX_DACMUTE, 0x0c }, + { AIC31XX_LDACVOL, 0x00 }, + { AIC31XX_RDACVOL, 0x00 }, + { AIC31XX_ADCSETUP, 0x00 }, + { AIC31XX_ADCFGA, 0x80 }, + { AIC31XX_ADCVOL, 0x00 }, + { AIC31XX_HPDRIVER, 0x04 }, + { AIC31XX_SPKAMP, 0x06 }, + { AIC31XX_DACMIXERROUTE, 0x00 }, + { AIC31XX_LANALOGHPL, 0x7f }, + { AIC31XX_RANALOGHPR, 0x7f }, + { AIC31XX_LANALOGSPL, 0x7f }, + { AIC31XX_RANALOGSPR, 0x7f }, + { AIC31XX_HPLGAIN, 0x02 }, + { AIC31XX_HPRGAIN, 0x02 }, + { AIC31XX_SPLGAIN, 0x00 }, + { AIC31XX_SPRGAIN, 0x00 }, + { AIC31XX_MICBIAS, 0x00 }, + { AIC31XX_MICPGA, 0x80 }, + { AIC31XX_MICPGAPI, 0x00 }, + { AIC31XX_MICPGAMI, 0x00 }, +}; + +static bool aic31xx_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC31XX_PAGECTL: /* regmap implementation requires this */ + case AIC31XX_RESET: /* always clears after write */ + case AIC31XX_OT_FLAG: + case AIC31XX_ADCFLAG: + case AIC31XX_DACFLAG1: + case AIC31XX_DACFLAG2: + case AIC31XX_OFFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG2: + case AIC31XX_INTRADCFLAG2: + return true; + } + return false; +} + +static bool aic31xx_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC31XX_OT_FLAG: + case AIC31XX_ADCFLAG: + case AIC31XX_DACFLAG1: + case AIC31XX_DACFLAG2: + case AIC31XX_OFFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG2: + case AIC31XX_INTRADCFLAG2: + return false; + } + return true; +} + +static const struct regmap_range_cfg aic31xx_ranges[] = { + { + .range_min = 0, + .range_max = 12 * 128, + .selector_reg = AIC31XX_PAGECTL, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +struct regmap_config aic31xx_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .writeable_reg = aic31xx_writeable, + .volatile_reg = aic31xx_volatile, + .reg_defaults = aic31xx_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(aic31xx_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .ranges = aic31xx_ranges, + .num_ranges = ARRAY_SIZE(aic31xx_ranges), + .max_register = 12 * 128, +}; + +#define AIC31XX_NUM_SUPPLIES 6 +static const char * const aic31xx_supply_names[AIC31XX_NUM_SUPPLIES] = { + "HPVDD", + "SPRVDD", + "SPLVDD", + "AVDD", + "IOVDD", + "DVDD", +}; + +struct aic31xx_disable_nb { + struct notifier_block nb; + struct aic31xx_priv *aic31xx; +}; + +struct aic31xx_priv { + struct snd_soc_codec *codec; + u8 i2c_regs_status; + struct device *dev; + struct regmap *regmap; + struct aic31xx_pdata pdata; + struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; + struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES]; + unsigned int sysclk; + int rate_div_line; +}; + +struct aic31xx_rate_divs { + u32 mclk; + u32 rate; + u8 p_val; + u8 pll_j; + u16 pll_d; + u16 dosr; + u8 ndac; + u8 mdac; + u8 aosr; + u8 nadc; + u8 madc; +}; + +/* ADC dividers can be disabled by cofiguring them to 0 */ +static const struct aic31xx_rate_divs aic31xx_divs[] = { + /* mclk rate pll: p j d dosr ndac mdac aors nadc madc */ + /* 8k rate */ + {12000000, 8000, 1, 8, 1920, 128, 48, 2, 128, 48, 2}, + {24000000, 8000, 2, 8, 1920, 128, 48, 2, 128, 48, 2}, + {25000000, 8000, 2, 7, 8643, 128, 48, 2, 128, 48, 2}, + /* 11.025k rate */ + {12000000, 11025, 1, 7, 5264, 128, 32, 2, 128, 32, 2}, + {24000000, 11025, 2, 7, 5264, 128, 32, 2, 128, 32, 2}, + {25000000, 11025, 2, 7, 2253, 128, 32, 2, 128, 32, 2}, + /* 16k rate */ + {12000000, 16000, 1, 8, 1920, 128, 24, 2, 128, 24, 2}, + {24000000, 16000, 2, 8, 1920, 128, 24, 2, 128, 24, 2}, + {25000000, 16000, 2, 7, 8643, 128, 24, 2, 128, 24, 2}, + /* 22.05k rate */ + {12000000, 22050, 1, 7, 5264, 128, 16, 2, 128, 16, 2}, + {24000000, 22050, 2, 7, 5264, 128, 16, 2, 128, 16, 2}, + {25000000, 22050, 2, 7, 2253, 128, 16, 2, 128, 16, 2}, + /* 32k rate */ + {12000000, 32000, 1, 8, 1920, 128, 12, 2, 128, 12, 2}, + {24000000, 32000, 2, 8, 1920, 128, 12, 2, 128, 12, 2}, + {25000000, 32000, 2, 7, 8643, 128, 12, 2, 128, 12, 2}, + /* 44.1k rate */ + {12000000, 44100, 1, 7, 5264, 128, 8, 2, 128, 8, 2}, + {24000000, 44100, 2, 7, 5264, 128, 8, 2, 128, 8, 2}, + {25000000, 44100, 2, 7, 2253, 128, 8, 2, 128, 8, 2}, + /* 48k rate */ + {12000000, 48000, 1, 8, 1920, 128, 8, 2, 128, 8, 2}, + {24000000, 48000, 2, 8, 1920, 128, 8, 2, 128, 8, 2}, + {25000000, 48000, 2, 7, 8643, 128, 8, 2, 128, 8, 2}, + /* 88.2k rate */ + {12000000, 88200, 1, 7, 5264, 64, 8, 2, 64, 8, 2}, + {24000000, 88200, 2, 7, 5264, 64, 8, 2, 64, 8, 2}, + {25000000, 88200, 2, 7, 2253, 64, 8, 2, 64, 8, 2}, + /* 96k rate */ + {12000000, 96000, 1, 8, 1920, 64, 8, 2, 64, 8, 2}, + {24000000, 96000, 2, 8, 1920, 64, 8, 2, 64, 8, 2}, + {25000000, 96000, 2, 7, 8643, 64, 8, 2, 64, 8, 2}, + /* 176.4k rate */ + {12000000, 176400, 1, 7, 5264, 32, 8, 2, 32, 8, 2}, + {24000000, 176400, 2, 7, 5264, 32, 8, 2, 32, 8, 2}, + {25000000, 176400, 2, 7, 2253, 32, 8, 2, 32, 8, 2}, + /* 192k rate */ + {12000000, 192000, 1, 8, 1920, 32, 8, 2, 32, 8, 2}, + {24000000, 192000, 2, 8, 1920, 32, 8, 2, 32, 8, 2}, + {25000000, 192000, 2, 7, 8643, 32, 8, 2, 32, 8, 2}, +}; + +static const char * const ldac_in_text[] = { + "Off", "Left Data", "Right Data", "Mono" +}; + +static const char * const rdac_in_text[] = { + "Off", "Right Data", "Left Data", "Mono" +}; + +static SOC_ENUM_SINGLE_DECL(ldac_in_enum, AIC31XX_DACSETUP, 4, ldac_in_text); + +static SOC_ENUM_SINGLE_DECL(rdac_in_enum, AIC31XX_DACSETUP, 2, rdac_in_text); + +static const char * const mic_select_text[] = { + "Off", "FFR 10 Ohm", "FFR 20 Ohm", "FFR 40 Ohm" +}; + +static const +SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, mic_select_text); + +static const +SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, mic_select_text); + +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_fgain_tlv, 0, 10, 0); +static const DECLARE_TLV_DB_SCALE(adc_cgain_tlv, -2000, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_pga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_SCALE(hp_drv_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(class_D_drv_tlv, 600, 600, 0); +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(sp_vol_tlv, -6350, 50, 0); + +/* + * controls to be exported to the user space + */ +static const struct snd_kcontrol_new aic31xx_snd_controls[] = { + SOC_DOUBLE_R_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL, + AIC31XX_RDACVOL, 0, -127, 48, 7, 0, dac_vol_tlv), + + SOC_SINGLE_TLV("ADC Fine Capture Volume", AIC31XX_ADCFGA, 4, 4, 1, + adc_fgain_tlv), + + SOC_SINGLE("ADC Capture Switch", AIC31XX_ADCFGA, 7, 1, 1), + SOC_DOUBLE_R_S_TLV("ADC Capture Volume", AIC31XX_ADCVOL, AIC31XX_ADCVOL, + 0, -24, 40, 6, 0, adc_cgain_tlv), + + SOC_SINGLE_TLV("Mic PGA Capture Volume", AIC31XX_MICPGA, 0, + 119, 0, mic_pga_tlv), + + SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 2, 1, 0), + SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv), + + SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL, + AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv), +}; + +static const struct snd_kcontrol_new aic311x_snd_controls[] = { + SOC_DOUBLE_R("Speaker Driver Playback Switch", AIC31XX_SPLGAIN, + AIC31XX_SPRGAIN, 2, 1, 0), + SOC_DOUBLE_R_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN, + AIC31XX_SPRGAIN, 3, 3, 0, class_D_drv_tlv), + + SOC_DOUBLE_R_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL, + AIC31XX_RANALOGSPR, 0, 0x7F, 1, sp_vol_tlv), +}; + +static const struct snd_kcontrol_new aic310x_snd_controls[] = { + SOC_SINGLE("Speaker Driver Playback Switch", AIC31XX_SPLGAIN, + 2, 1, 0), + SOC_SINGLE_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN, + 3, 3, 0, class_D_drv_tlv), + + SOC_SINGLE_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL, + 0, 0x7F, 1, sp_vol_tlv), +}; + +static const struct snd_kcontrol_new ldac_in_control = + SOC_DAPM_ENUM("DAC Left Input", ldac_in_enum); + +static const struct snd_kcontrol_new rdac_in_control = + SOC_DAPM_ENUM("DAC Right Input", rdac_in_enum); + +int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, + unsigned int mask, unsigned int wbits, int sleep, + int count) +{ + unsigned int bits; + int counter = count; + int ret = regmap_read(aic31xx->regmap, reg, &bits); + while ((bits & mask) != wbits && counter && !ret) { + usleep_range(sleep, sleep * 2); + ret = regmap_read(aic31xx->regmap, reg, &bits); + counter--; + } + if ((bits & mask) != wbits) { + dev_err(aic31xx->dev, + "%s: Failed! 0x%x was 0x%x expected 0x%x (%d, 0x%x, %d us)\n", + __func__, reg, bits, wbits, ret, mask, + (count - counter) * sleep); + ret = -1; + } + return ret; +} + +#define WIDGET_BIT(reg, shift) (((shift) << 8) | (reg)) + +static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(w->codec); + unsigned int reg = AIC31XX_DACFLAG1; + unsigned int mask; + + switch (WIDGET_BIT(w->reg, w->shift)) { + case WIDGET_BIT(AIC31XX_DACSETUP, 7): + mask = AIC31XX_LDACPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_DACSETUP, 6): + mask = AIC31XX_RDACPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_HPDRIVER, 7): + mask = AIC31XX_HPLDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_HPDRIVER, 6): + mask = AIC31XX_HPRDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_SPKAMP, 7): + mask = AIC31XX_SPLDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_SPKAMP, 6): + mask = AIC31XX_SPRDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_ADCSETUP, 7): + mask = AIC31XX_ADCPWRSTATUS_MASK; + reg = AIC31XX_ADCFLAG; + break; + default: + dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n", + w->name, __func__); + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + return aic31xx_wait_bits(aic31xx, reg, mask, mask, 5000, 100); + case SND_SOC_DAPM_POST_PMD: + return aic31xx_wait_bits(aic31xx, reg, mask, 0, 5000, 100); + default: + dev_dbg(w->codec->dev, + "Unhandled dapm widget event %d from %s\n", + event, w->name); + } + return 0; +} + +static const struct snd_kcontrol_new left_output_switches[] = { + SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0), + SOC_DAPM_SINGLE("From MIC1LP", AIC31XX_DACMIXERROUTE, 5, 1, 0), + SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 4, 1, 0), +}; + +static const struct snd_kcontrol_new right_output_switches[] = { + SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0), + SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 1, 1, 0), +}; + +static const struct snd_kcontrol_new p_term_mic1lp = + SOC_DAPM_ENUM("MIC1LP P-Terminal", mic1lp_p_enum); + +static const struct snd_kcontrol_new p_term_mic1rp = + SOC_DAPM_ENUM("MIC1RP P-Terminal", mic1rp_p_enum); + +static const struct snd_kcontrol_new p_term_mic1lm = + SOC_DAPM_ENUM("MIC1LM P-Terminal", mic1lm_p_enum); + +static const struct snd_kcontrol_new m_term_mic1lm = + SOC_DAPM_ENUM("MIC1LM M-Terminal", mic1lm_m_enum); + +static const struct snd_kcontrol_new aic31xx_dapm_hpl_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGHPL, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_hpr_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGHPR, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_spl_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGSPL, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_spr_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGSPR, 7, 1, 0); + +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* change mic bias voltage to user defined */ + snd_soc_update_bits(codec, AIC31XX_MICBIAS, + AIC31XX_MICBIAS_MASK, + aic31xx->pdata.micbias_vg << + AIC31XX_MICBIAS_SHIFT); + dev_dbg(codec->dev, "%s: turned on\n", __func__); + break; + case SND_SOC_DAPM_PRE_PMD: + /* turn mic bias off */ + snd_soc_update_bits(codec, AIC31XX_MICBIAS, + AIC31XX_MICBIAS_MASK, 0); + dev_dbg(codec->dev, "%s: turned off\n", __func__); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("DAC Left Input", + SND_SOC_NOPM, 0, 0, &ldac_in_control), + SND_SOC_DAPM_MUX("DAC Right Input", + SND_SOC_NOPM, 0, 0, &rdac_in_control), + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC Left", "Left Playback", + AIC31XX_DACSETUP, 7, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_DAC_E("DAC Right", "Right Playback", + AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0, + left_output_switches, + ARRAY_SIZE(left_output_switches)), + SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0, + right_output_switches, + ARRAY_SIZE(right_output_switches)), + + SND_SOC_DAPM_SWITCH("HP Left", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_hpl_switch), + SND_SOC_DAPM_SWITCH("HP Right", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_hpr_switch), + + /* Output drivers */ + SND_SOC_DAPM_OUT_DRV_E("HPL Driver", AIC31XX_HPDRIVER, 7, 0, + NULL, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_OUT_DRV_E("HPR Driver", AIC31XX_HPDRIVER, 6, 0, + NULL, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + + /* ADC */ + SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + + /* Input Selection to MIC_PGA */ + SND_SOC_DAPM_MUX("MIC1LP P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1lp), + SND_SOC_DAPM_MUX("MIC1RP P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1rp), + SND_SOC_DAPM_MUX("MIC1LM P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1lm), + + SND_SOC_DAPM_MUX("MIC1LM M-Terminal", SND_SOC_NOPM, 0, 0, + &m_term_mic1lm), + /* Enabling & Disabling MIC Gain Ctl */ + SND_SOC_DAPM_PGA("MIC_GAIN_CTL", AIC31XX_MICPGA, + 7, 1, NULL, 0), + + /* Mic Bias */ + SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC1LP"), + SND_SOC_DAPM_INPUT("MIC1RP"), + SND_SOC_DAPM_INPUT("MIC1LM"), +}; + +static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = { + /* AIC3111 and AIC3110 have stereo class-D amplifier */ + SND_SOC_DAPM_OUT_DRV_E("SPL ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUT_DRV_E("SPR ClassD", AIC31XX_SPKAMP, 6, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("Speaker Left", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spl_switch), + SND_SOC_DAPM_SWITCH("Speaker Right", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spr_switch), + SND_SOC_DAPM_OUTPUT("SPL"), + SND_SOC_DAPM_OUTPUT("SPR"), +}; + +/* AIC3100 and AIC3120 have only mono class-D amplifier */ +static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = { + SND_SOC_DAPM_OUT_DRV_E("SPK ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("Speaker", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spl_switch), + SND_SOC_DAPM_OUTPUT("SPK"), +}; + +static const struct snd_soc_dapm_route +aic31xx_audio_map[] = { + /* DAC Input Routing */ + {"DAC Left Input", "Left Data", "DAC IN"}, + {"DAC Left Input", "Right Data", "DAC IN"}, + {"DAC Left Input", "Mono", "DAC IN"}, + {"DAC Right Input", "Left Data", "DAC IN"}, + {"DAC Right Input", "Right Data", "DAC IN"}, + {"DAC Right Input", "Mono", "DAC IN"}, + {"DAC Left", NULL, "DAC Left Input"}, + {"DAC Right", NULL, "DAC Right Input"}, + + /* Mic input */ + {"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"}, + {"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"}, + {"MIC1LP P-Terminal", "FFR 40 Ohm", "MIC1LP"}, + {"MIC1RP P-Terminal", "FFR 10 Ohm", "MIC1RP"}, + {"MIC1RP P-Terminal", "FFR 20 Ohm", "MIC1RP"}, + {"MIC1RP P-Terminal", "FFR 40 Ohm", "MIC1RP"}, + {"MIC1LM P-Terminal", "FFR 10 Ohm", "MIC1LM"}, + {"MIC1LM P-Terminal", "FFR 20 Ohm", "MIC1LM"}, + {"MIC1LM P-Terminal", "FFR 40 Ohm", "MIC1LM"}, + + {"MIC1LM M-Terminal", "FFR 10 Ohm", "MIC1LM"}, + {"MIC1LM M-Terminal", "FFR 20 Ohm", "MIC1LM"}, + {"MIC1LM M-Terminal", "FFR 40 Ohm", "MIC1LM"}, + + {"MIC_GAIN_CTL", NULL, "MIC1LP P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1RP P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1LM P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1LM M-Terminal"}, + + {"ADC", NULL, "MIC_GAIN_CTL"}, + + /* Left Output */ + {"Output Left", "From Left DAC", "DAC Left"}, + {"Output Left", "From MIC1LP", "MIC1LP"}, + {"Output Left", "From MIC1RP", "MIC1RP"}, + + /* Right Output */ + {"Output Right", "From Right DAC", "DAC Right"}, + {"Output Right", "From MIC1RP", "MIC1RP"}, + + /* HPL path */ + {"HP Left", "Switch", "Output Left"}, + {"HPL Driver", NULL, "HP Left"}, + {"HPL", NULL, "HPL Driver"}, + + /* HPR path */ + {"HP Right", "Switch", "Output Right"}, + {"HPR Driver", NULL, "HP Right"}, + {"HPR", NULL, "HPR Driver"}, +}; + +static const struct snd_soc_dapm_route +aic311x_audio_map[] = { + /* SP L path */ + {"Speaker Left", "Switch", "Output Left"}, + {"SPL ClassD", NULL, "Speaker Left"}, + {"SPL", NULL, "SPL ClassD"}, + + /* SP R path */ + {"Speaker Right", "Switch", "Output Right"}, + {"SPR ClassD", NULL, "Speaker Right"}, + {"SPR", NULL, "SPR ClassD"}, +}; + +static const struct snd_soc_dapm_route +aic310x_audio_map[] = { + /* SP L path */ + {"Speaker", "Switch", "Output Left"}, + {"SPK ClassD", NULL, "Speaker"}, + {"SPK", NULL, "SPK ClassD"}, +}; + +static int aic31xx_add_controls(struct snd_soc_codec *codec) +{ + int ret = 0; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) + ret = snd_soc_add_codec_controls( + codec, aic311x_snd_controls, + ARRAY_SIZE(aic311x_snd_controls)); + else + ret = snd_soc_add_codec_controls( + codec, aic310x_snd_controls, + ARRAY_SIZE(aic310x_snd_controls)); + + return ret; +} + +static int aic31xx_add_widgets(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) { + ret = snd_soc_dapm_new_controls( + dapm, aic311x_dapm_widgets, + ARRAY_SIZE(aic311x_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic311x_audio_map, + ARRAY_SIZE(aic311x_audio_map)); + if (ret) + return ret; + } else { + ret = snd_soc_dapm_new_controls( + dapm, aic310x_dapm_widgets, + ARRAY_SIZE(aic310x_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic310x_audio_map, + ARRAY_SIZE(aic310x_audio_map)); + if (ret) + return ret; + } + + return 0; +} + +static int aic31xx_setup_pll(struct snd_soc_codec *codec, + struct snd_pcm_hw_params *params) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int bclk_n = 0; + int i; + + /* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */ + snd_soc_update_bits(codec, AIC31XX_CLKMUX, + AIC31XX_CODEC_CLKIN_MASK, AIC31XX_CODEC_CLKIN_PLL); + snd_soc_update_bits(codec, AIC31XX_IFACE2, + AIC31XX_BDIVCLK_MASK, AIC31XX_DAC2BCLK); + + for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) { + if (aic31xx_divs[i].rate == params_rate(params) && + aic31xx_divs[i].mclk == aic31xx->sysclk) + break; + } + + if (i == ARRAY_SIZE(aic31xx_divs)) { + dev_err(codec->dev, "%s: Sampling rate %u not supported\n", + __func__, params_rate(params)); + return -EINVAL; + } + + /* PLL configuration */ + snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK, + (aic31xx_divs[i].p_val << 4) | 0x01); + snd_soc_write(codec, AIC31XX_PLLJ, aic31xx_divs[i].pll_j); + + snd_soc_write(codec, AIC31XX_PLLDMSB, + aic31xx_divs[i].pll_d >> 8); + snd_soc_write(codec, AIC31XX_PLLDLSB, + aic31xx_divs[i].pll_d & 0xff); + + /* DAC dividers configuration */ + snd_soc_update_bits(codec, AIC31XX_NDAC, AIC31XX_PLL_MASK, + aic31xx_divs[i].ndac); + snd_soc_update_bits(codec, AIC31XX_MDAC, AIC31XX_PLL_MASK, + aic31xx_divs[i].mdac); + + snd_soc_write(codec, AIC31XX_DOSRMSB, aic31xx_divs[i].dosr >> 8); + snd_soc_write(codec, AIC31XX_DOSRLSB, aic31xx_divs[i].dosr & 0xff); + + /* ADC dividers configuration. Write reset value 1 if not used. */ + snd_soc_update_bits(codec, AIC31XX_NADC, AIC31XX_PLL_MASK, + aic31xx_divs[i].nadc ? aic31xx_divs[i].nadc : 1); + snd_soc_update_bits(codec, AIC31XX_MADC, AIC31XX_PLL_MASK, + aic31xx_divs[i].madc ? aic31xx_divs[i].madc : 1); + + snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr); + + /* Bit clock divider configuration. */ + bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) + / snd_soc_params_to_frame_size(params); + if (bclk_n == 0) { + dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n", + __func__); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_BCLKN, + AIC31XX_PLL_MASK, bclk_n); + + aic31xx->rate_div_line = i; + + dev_dbg(codec->dev, + "pll %d.%04d/%d dosr %d n %d m %d aosr %d n %d m %d bclk_n %d\n", + aic31xx_divs[i].pll_j, aic31xx_divs[i].pll_d, + aic31xx_divs[i].p_val, aic31xx_divs[i].dosr, + aic31xx_divs[i].ndac, aic31xx_divs[i].mdac, + aic31xx_divs[i].aosr, aic31xx_divs[i].nadc, + aic31xx_divs[i].madc, bclk_n); + + return 0; +} + +static int aic31xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *tmp) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + u8 data = 0; + + dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n", + __func__, params_format(params), params_width(params), + params_rate(params)); + + switch (params_width(params)) { + case 16: + break; + case 20: + data = (AIC31XX_WORD_LEN_20BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + case 24: + data = (AIC31XX_WORD_LEN_24BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + case 32: + data = (AIC31XX_WORD_LEN_32BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + default: + dev_err(codec->dev, "%s: Unsupported format %d\n", + __func__, params_format(params)); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_IFACE1, + AIC31XX_IFACE1_DATALEN_MASK, + data); + + return aic31xx_setup_pll(codec, params); +} + +static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + if (mute) { + snd_soc_update_bits(codec, AIC31XX_DACMUTE, + AIC31XX_DACMUTE_MASK, + AIC31XX_DACMUTE_MASK); + } else { + snd_soc_update_bits(codec, AIC31XX_DACMUTE, + AIC31XX_DACMUTE_MASK, 0x0); + } + + return 0; +} + +static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 iface_reg1 = 0; + u8 iface_reg3 = 0; + u8 dsp_a_val = 0; + + dev_dbg(codec->dev, "## %s: fmt = 0x%x\n", __func__, fmt); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface_reg1 |= AIC31XX_BCLK_MASTER | AIC31XX_WCLK_MASTER; + break; + default: + dev_alert(codec->dev, "Invalid DAI master/slave interface\n"); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_DSP_A: + dsp_a_val = 0x1; + case SND_SOC_DAIFMT_DSP_B: + /* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + iface_reg3 |= AIC31XX_BCLKINV_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + default: + return -EINVAL; + } + iface_reg1 |= (AIC31XX_DSP_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface_reg1 |= (AIC31XX_RIGHT_JUSTIFIED_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg1 |= (AIC31XX_LEFT_JUSTIFIED_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + default: + dev_err(codec->dev, "Invalid DAI interface format\n"); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_IFACE1, + AIC31XX_IFACE1_DATATYPE_MASK | + AIC31XX_IFACE1_MASTER_MASK, + iface_reg1); + snd_soc_update_bits(codec, AIC31XX_DATA_OFFSET, + AIC31XX_DATA_OFFSET_MASK, + dsp_a_val); + snd_soc_update_bits(codec, AIC31XX_IFACE2, + AIC31XX_BCLKINV_MASK, + iface_reg3); + + return 0; +} + +static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + + dev_dbg(codec->dev, "## %s: clk_id = %d, freq = %d, dir = %d\n", + __func__, clk_id, freq, dir); + + for (i = 0; aic31xx_divs[i].mclk != freq; i++) { + if (i == ARRAY_SIZE(aic31xx_divs)) { + dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n", + __func__, freq); + return -EINVAL; + } + } + + /* set clock on MCLK, BCLK, or GPIO1 as PLL input */ + snd_soc_update_bits(codec, AIC31XX_CLKMUX, AIC31XX_PLL_CLKIN_MASK, + clk_id << AIC31XX_PLL_CLKIN_SHIFT); + + aic31xx->sysclk = freq; + return 0; +} + +static int aic31xx_regulator_event(struct notifier_block *nb, + unsigned long event, void *data) +{ + struct aic31xx_disable_nb *disable_nb = + container_of(nb, struct aic31xx_disable_nb, nb); + struct aic31xx_priv *aic31xx = disable_nb->aic31xx; + + if (event & REGULATOR_EVENT_DISABLE) { + /* + * Put codec to reset and as at least one of the + * supplies was disabled. + */ + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) + gpio_set_value(aic31xx->pdata.gpio_reset, 0); + + regcache_mark_dirty(aic31xx->regmap); + dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__); + } + + return 0; +} + +static void aic31xx_clk_on(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + u8 mask = AIC31XX_PM_MASK; + u8 on = AIC31XX_PM_MASK; + + dev_dbg(codec->dev, "codec clock -> on (rate %d)\n", + aic31xx_divs[aic31xx->rate_div_line].rate); + snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, on); + mdelay(10); + snd_soc_update_bits(codec, AIC31XX_NDAC, mask, on); + snd_soc_update_bits(codec, AIC31XX_MDAC, mask, on); + if (aic31xx_divs[aic31xx->rate_div_line].nadc) + snd_soc_update_bits(codec, AIC31XX_NADC, mask, on); + if (aic31xx_divs[aic31xx->rate_div_line].madc) + snd_soc_update_bits(codec, AIC31XX_MADC, mask, on); + snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, on); +} + +static void aic31xx_clk_off(struct snd_soc_codec *codec) +{ + u8 mask = AIC31XX_PM_MASK; + u8 off = 0; + + dev_dbg(codec->dev, "codec clock -> off\n"); + snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, off); + snd_soc_update_bits(codec, AIC31XX_MADC, mask, off); + snd_soc_update_bits(codec, AIC31XX_NADC, mask, off); + snd_soc_update_bits(codec, AIC31XX_MDAC, mask, off); + snd_soc_update_bits(codec, AIC31XX_NDAC, mask, off); + snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, off); +} + +static int aic31xx_power_on(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + ret = regulator_bulk_enable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + if (ret) + return ret; + + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) { + gpio_set_value(aic31xx->pdata.gpio_reset, 1); + udelay(100); + } + regcache_cache_only(aic31xx->regmap, false); + ret = regcache_sync(aic31xx->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to restore cache: %d\n", ret); + regcache_cache_only(aic31xx->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + return ret; + } + return 0; +} + +static int aic31xx_power_off(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + regcache_cache_only(aic31xx->regmap, true); + ret = regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + + return ret; +} + +static int aic31xx_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__, + codec->dapm.bias_level, level); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + aic31xx_clk_on(codec); + break; + case SND_SOC_BIAS_STANDBY: + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + aic31xx_power_on(codec); + break; + case SND_SOC_BIAS_PREPARE: + aic31xx_clk_off(codec); + break; + default: + BUG(); + } + break; + case SND_SOC_BIAS_OFF: + aic31xx_power_off(codec); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int aic31xx_suspend(struct snd_soc_codec *codec) +{ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int aic31xx_resume(struct snd_soc_codec *codec) +{ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int aic31xx_codec_probe(struct snd_soc_codec *codec) +{ + int ret = 0; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + + dev_dbg(aic31xx->dev, "## %s\n", __func__); + + aic31xx = snd_soc_codec_get_drvdata(codec); + codec->control_data = aic31xx->regmap; + + aic31xx->codec = codec; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + + if (ret != 0) { + dev_err(codec->dev, "snd_soc_codec_set_cache_io failed %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) { + aic31xx->disable_nb[i].nb.notifier_call = + aic31xx_regulator_event; + aic31xx->disable_nb[i].aic31xx = aic31xx; + ret = regulator_register_notifier(aic31xx->supplies[i].consumer, + &aic31xx->disable_nb[i].nb); + if (ret) { + dev_err(codec->dev, + "Failed to request regulator notifier: %d\n", + ret); + return ret; + } + } + + regcache_cache_only(aic31xx->regmap, true); + regcache_mark_dirty(aic31xx->regmap); + + ret = aic31xx_add_controls(codec); + if (ret) + return ret; + + ret = aic31xx_add_widgets(codec); + + return ret; +} + +static int aic31xx_codec_remove(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + /* power down chip */ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF); + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) + regulator_unregister_notifier(aic31xx->supplies[i].consumer, + &aic31xx->disable_nb[i].nb); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_driver_aic31xx = { + .probe = aic31xx_codec_probe, + .remove = aic31xx_codec_remove, + .suspend = aic31xx_suspend, + .resume = aic31xx_resume, + .set_bias_level = aic31xx_set_bias_level, + .controls = aic31xx_snd_controls, + .num_controls = ARRAY_SIZE(aic31xx_snd_controls), + .dapm_widgets = aic31xx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic31xx_dapm_widgets), + .dapm_routes = aic31xx_audio_map, + .num_dapm_routes = ARRAY_SIZE(aic31xx_audio_map), +}; + +static struct snd_soc_dai_ops aic31xx_dai_ops = { + .hw_params = aic31xx_hw_params, + .set_sysclk = aic31xx_set_dai_sysclk, + .set_fmt = aic31xx_set_dai_fmt, + .digital_mute = aic31xx_dac_mute, +}; + +static struct snd_soc_dai_driver aic31xx_dai_driver[] = { + { + .name = "tlv320aic31xx-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC31XX_RATES, + .formats = AIC31XX_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AIC31XX_RATES, + .formats = AIC31XX_FORMATS, + }, + .ops = &aic31xx_dai_ops, + .symmetric_rates = 1, + } +}; + +#if defined(CONFIG_OF) +static const struct of_device_id tlv320aic31xx_of_match[] = { + { .compatible = "ti,tlv320aic310x" }, + { .compatible = "ti,tlv320aic311x" }, + { .compatible = "ti,tlv320aic3100" }, + { .compatible = "ti,tlv320aic3110" }, + { .compatible = "ti,tlv320aic3120" }, + { .compatible = "ti,tlv320aic3111" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tlv320aic31xx_of_match); + +static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) +{ + struct device_node *np = aic31xx->dev->of_node; + unsigned int value = MICBIAS_2_0V; + int ret; + + of_property_read_u32(np, "ai31xx-micbias-vg", &value); + switch (value) { + case MICBIAS_2_0V: + case MICBIAS_2_5V: + case MICBIAS_AVDDV: + aic31xx->pdata.micbias_vg = value; + break; + default: + dev_err(aic31xx->dev, + "Bad ai31xx-micbias-vg value %d DT\n", + value); + aic31xx->pdata.micbias_vg = MICBIAS_2_0V; + } + + ret = of_get_named_gpio(np, "gpio-reset", 0); + if (ret > 0) + aic31xx->pdata.gpio_reset = ret; +} +#else /* CONFIG_OF */ +static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) +{ +} +#endif /* CONFIG_OF */ + +void aic31xx_device_init(struct aic31xx_priv *aic31xx) +{ + int ret, i; + + dev_set_drvdata(aic31xx->dev, aic31xx); + + if (dev_get_platdata(aic31xx->dev)) + memcpy(&aic31xx->pdata, dev_get_platdata(aic31xx->dev), + sizeof(aic31xx->pdata)); + else if (aic31xx->dev->of_node) + aic31xx_pdata_from_of(aic31xx); + + if (aic31xx->pdata.gpio_reset) { + ret = devm_gpio_request_one(aic31xx->dev, + aic31xx->pdata.gpio_reset, + GPIOF_OUT_INIT_HIGH, + "aic31xx-reset-pin"); + if (ret < 0) { + dev_err(aic31xx->dev, "not able to acquire gpio\n"); + return; + } + } + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) + aic31xx->supplies[i].supply = aic31xx_supply_names[i]; + + ret = devm_regulator_bulk_get(aic31xx->dev, + ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + if (ret != 0) + dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret); + +} + +static int aic31xx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct aic31xx_priv *aic31xx; + int ret; + const struct regmap_config *regmap_config; + + dev_dbg(&i2c->dev, "## %s: %s codec_type = %d\n", __func__, + id->name, (int) id->driver_data); + + regmap_config = &aic31xx_i2c_regmap; + + aic31xx = devm_kzalloc(&i2c->dev, sizeof(*aic31xx), GFP_KERNEL); + if (aic31xx == NULL) + return -ENOMEM; + + aic31xx->regmap = devm_regmap_init_i2c(i2c, regmap_config); + + if (IS_ERR(aic31xx->regmap)) { + ret = PTR_ERR(aic31xx->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + aic31xx->dev = &i2c->dev; + + aic31xx->pdata.codec_type = id->driver_data; + + aic31xx_device_init(aic31xx); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, + aic31xx_dai_driver, + ARRAY_SIZE(aic31xx_dai_driver)); + + return ret; +} + +static int aic31xx_i2c_remove(struct i2c_client *i2c) +{ + struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev); + + kfree(aic31xx); + return 0; +} + +static const struct i2c_device_id aic31xx_i2c_id[] = { + { "tlv320aic310x", AIC3100 }, + { "tlv320aic311x", AIC3110 }, + { "tlv320aic3100", AIC3100 }, + { "tlv320aic3110", AIC3110 }, + { "tlv320aic3120", AIC3120 }, + { "tlv320aic3111", AIC3111 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); + +static struct i2c_driver aic31xx_i2c_driver = { + .driver = { + .name = "tlv320aic31xx-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tlv320aic31xx_of_match), + }, + .probe = aic31xx_i2c_probe, + .remove = (aic31xx_i2c_remove), + .id_table = aic31xx_i2c_id, +}; + +module_i2c_driver(aic31xx_i2c_driver); + +MODULE_DESCRIPTION("ASoC TLV320AIC3111 codec driver"); +MODULE_AUTHOR("Jyri Sarha"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h new file mode 100644 index 0000000..52ed57c --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -0,0 +1,258 @@ +/* + * ALSA SoC TLV320AIC31XX codec driver + * + * Copyright (C) 2013 Texas Instruments, Inc. + * + * This package 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 PACKAGE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + * + */ +#ifndef _TLV320AIC31XX_H +#define _TLV320AIC31XX_H + +#define AIC31XX_RATES SNDRV_PCM_RATE_8000_192000 + +#define AIC31XX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + + +#define AIC31XX_STEREO_CLASS_D_BIT 0x1 +#define AIC31XX_MINIDSP_BIT 0x2 + +enum aic31xx_type { + AIC3100 = 0, + AIC3110 = AIC31XX_STEREO_CLASS_D_BIT, + AIC3120 = AIC31XX_MINIDSP_BIT, + AIC3111 = (AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT), +}; + +struct aic31xx_pdata { + enum aic31xx_type codec_type; + unsigned int gpio_reset; + int micbias_vg; +}; + +/* Page Control Register */ +#define AIC31XX_PAGECTL 0x00 + +/* Page 0 Registers */ +/* Software reset register */ +#define AIC31XX_RESET 0x01 +/* OT FLAG register */ +#define AIC31XX_OT_FLAG 0x03 +/* Clock clock Gen muxing, Multiplexers*/ +#define AIC31XX_CLKMUX 0x04 +/* PLL P and R-VAL register */ +#define AIC31XX_PLLPR 0x05 +/* PLL J-VAL register */ +#define AIC31XX_PLLJ 0x06 +/* PLL D-VAL MSB register */ +#define AIC31XX_PLLDMSB 0x07 +/* PLL D-VAL LSB register */ +#define AIC31XX_PLLDLSB 0x08 +/* DAC NDAC_VAL register*/ +#define AIC31XX_NDAC 0x0B +/* DAC MDAC_VAL register */ +#define AIC31XX_MDAC 0x0C +/* DAC OSR setting register 1, MSB value */ +#define AIC31XX_DOSRMSB 0x0D +/* DAC OSR setting register 2, LSB value */ +#define AIC31XX_DOSRLSB 0x0E +#define AIC31XX_MINI_DSP_INPOL 0x10 +/* Clock setting register 8, PLL */ +#define AIC31XX_NADC 0x12 +/* Clock setting register 9, PLL */ +#define AIC31XX_MADC 0x13 +/* ADC Oversampling (AOSR) Register */ +#define AIC31XX_AOSR 0x14 +/* Clock setting register 9, Multiplexers */ +#define AIC31XX_CLKOUTMUX 0x19 +/* Clock setting register 10, CLOCKOUT M divider value */ +#define AIC31XX_CLKOUTMVAL 0x1A +/* Audio Interface Setting Register 1 */ +#define AIC31XX_IFACE1 0x1B +/* Audio Data Slot Offset Programming */ +#define AIC31XX_DATA_OFFSET 0x1C +/* Audio Interface Setting Register 2 */ +#define AIC31XX_IFACE2 0x1D +/* Clock setting register 11, BCLK N Divider */ +#define AIC31XX_BCLKN 0x1E +/* Audio Interface Setting Register 3, Secondary Audio Interface */ +#define AIC31XX_IFACESEC1 0x1F +/* Audio Interface Setting Register 4 */ +#define AIC31XX_IFACESEC2 0x20 +/* Audio Interface Setting Register 5 */ +#define AIC31XX_IFACESEC3 0x21 +/* I2C Bus Condition */ +#define AIC31XX_I2C 0x22 +/* ADC FLAG */ +#define AIC31XX_ADCFLAG 0x24 +/* DAC Flag Registers */ +#define AIC31XX_DACFLAG1 0x25 +#define AIC31XX_DACFLAG2 0x26 +/* Sticky Interrupt flag (overflow) */ +#define AIC31XX_OFFLAG 0x27 +/* Sticy DAC Interrupt flags */ +#define AIC31XX_INTRDACFLAG 0x2C +/* Sticy ADC Interrupt flags */ +#define AIC31XX_INTRADCFLAG 0x2D +/* DAC Interrupt flags 2 */ +#define AIC31XX_INTRDACFLAG2 0x2E +/* ADC Interrupt flags 2 */ +#define AIC31XX_INTRADCFLAG2 0x2F +/* INT1 interrupt control */ +#define AIC31XX_INT1CTRL 0x30 +/* INT2 interrupt control */ +#define AIC31XX_INT2CTRL 0x31 +/* GPIO1 control */ +#define AIC31XX_GPIO1 0x33 + +#define AIC31XX_DACPRB 0x3C +/* ADC Instruction Set Register */ +#define AIC31XX_ADCPRB 0x3D +/* DAC channel setup register */ +#define AIC31XX_DACSETUP 0x3F +/* DAC Mute and volume control register */ +#define AIC31XX_DACMUTE 0x40 +/* Left DAC channel digital volume control */ +#define AIC31XX_LDACVOL 0x41 +/* Right DAC channel digital volume control */ +#define AIC31XX_RDACVOL 0x42 +/* Headset detection */ +#define AIC31XX_HSDETECT 0x43 +/* ADC Digital Mic */ +#define AIC31XX_ADCSETUP 0x51 +/* ADC Digital Volume Control Fine Adjust */ +#define AIC31XX_ADCFGA 0x52 +/* ADC Digital Volume Control Coarse Adjust */ +#define AIC31XX_ADCVOL 0x53 + + +/* Page 1 Registers */ +/* Headphone drivers */ +#define AIC31XX_HPDRIVER 0x9F +/* Class-D Speakear Amplifier */ +#define AIC31XX_SPKAMP 0xA0 +/* HP Output Drivers POP Removal Settings */ +#define AIC31XX_HPPOP 0xA1 +/* Output Driver PGA Ramp-Down Period Control */ +#define AIC31XX_SPPGARAMP 0xA2 +/* DAC_L and DAC_R Output Mixer Routing */ +#define AIC31XX_DACMIXERROUTE 0xA3 +/* Left Analog Vol to HPL */ +#define AIC31XX_LANALOGHPL 0xA4 +/* Right Analog Vol to HPR */ +#define AIC31XX_RANALOGHPR 0xA5 +/* Left Analog Vol to SPL */ +#define AIC31XX_LANALOGSPL 0xA6 +/* Right Analog Vol to SPR */ +#define AIC31XX_RANALOGSPR 0xA7 +/* HPL Driver */ +#define AIC31XX_HPLGAIN 0xA8 +/* HPR Driver */ +#define AIC31XX_HPRGAIN 0xA9 +/* SPL Driver */ +#define AIC31XX_SPLGAIN 0xAA +/* SPR Driver */ +#define AIC31XX_SPRGAIN 0xAB +/* HP Driver Control */ +#define AIC31XX_HPCONTROL 0xAC +/* MIC Bias Control */ +#define AIC31XX_MICBIAS 0xAE +/* MIC PGA*/ +#define AIC31XX_MICPGA 0xAF +/* Delta-Sigma Mono ADC Channel Fine-Gain Input Selection for P-Terminal */ +#define AIC31XX_MICPGAPI 0xB0 +/* ADC Input Selection for M-Terminal */ +#define AIC31XX_MICPGAMI 0xB1 +/* Input CM Settings */ +#define AIC31XX_MICPGACM 0xB2 + +/* Bits, masks and shifts */ + +/* AIC31XX_CLKMUX */ +#define AIC31XX_PLL_CLKIN_MASK 0x0c +#define AIC31XX_PLL_CLKIN_SHIFT 2 +#define AIC31XX_PLL_CLKIN_MCLK 0 +#define AIC31XX_CODEC_CLKIN_MASK 0x03 +#define AIC31XX_CODEC_CLKIN_SHIFT 0 +#define AIC31XX_CODEC_CLKIN_PLL 3 +#define AIC31XX_CODEC_CLKIN_BCLK 1 + +/* AIC31XX_PLLPR, AIC31XX_NDAC, AIC31XX_MDAC, AIC31XX_NADC, AIC31XX_MADC, + AIC31XX_BCLKN */ +#define AIC31XX_PLL_MASK 0x7f +#define AIC31XX_PM_MASK 0x80 + +/* AIC31XX_IFACE1 */ +#define AIC31XX_WORD_LEN_16BITS 0x00 +#define AIC31XX_WORD_LEN_20BITS 0x01 +#define AIC31XX_WORD_LEN_24BITS 0x02 +#define AIC31XX_WORD_LEN_32BITS 0x03 +#define AIC31XX_IFACE1_DATALEN_MASK 0x30 +#define AIC31XX_IFACE1_DATALEN_SHIFT (4) +#define AIC31XX_IFACE1_DATATYPE_MASK 0xC0 +#define AIC31XX_IFACE1_DATATYPE_SHIFT (6) +#define AIC31XX_I2S_MODE 0x00 +#define AIC31XX_DSP_MODE 0x01 +#define AIC31XX_RIGHT_JUSTIFIED_MODE 0x02 +#define AIC31XX_LEFT_JUSTIFIED_MODE 0x03 +#define AIC31XX_IFACE1_MASTER_MASK 0x0C +#define AIC31XX_BCLK_MASTER 0x08 +#define AIC31XX_WCLK_MASTER 0x04 + +/* AIC31XX_DATA_OFFSET */ +#define AIC31XX_DATA_OFFSET_MASK 0xFF + +/* AIC31XX_IFACE2 */ +#define AIC31XX_BCLKINV_MASK 0x08 +#define AIC31XX_BDIVCLK_MASK 0x03 +#define AIC31XX_DAC2BCLK 0x00 +#define AIC31XX_DACMOD2BCLK 0x01 +#define AIC31XX_ADC2BCLK 0x02 +#define AIC31XX_ADCMOD2BCLK 0x03 + +/* AIC31XX_ADCFLAG */ +#define AIC31XX_ADCPWRSTATUS_MASK 0x40 + +/* AIC31XX_DACFLAG1 */ +#define AIC31XX_LDACPWRSTATUS_MASK 0x80 +#define AIC31XX_RDACPWRSTATUS_MASK 0x08 +#define AIC31XX_HPLDRVPWRSTATUS_MASK 0x20 +#define AIC31XX_HPRDRVPWRSTATUS_MASK 0x02 +#define AIC31XX_SPLDRVPWRSTATUS_MASK 0x10 +#define AIC31XX_SPRDRVPWRSTATUS_MASK 0x01 + +/* AIC31XX_INTRDACFLAG */ +#define AIC31XX_HPSCDETECT_MASK 0x80 +#define AIC31XX_BUTTONPRESS_MASK 0x20 +#define AIC31XX_HSPLUG_MASK 0x10 +#define AIC31XX_LDRCTHRES_MASK 0x08 +#define AIC31XX_RDRCTHRES_MASK 0x04 +#define AIC31XX_DACSINT_MASK 0x02 +#define AIC31XX_DACAINT_MASK 0x01 + +/* AIC31XX_INT1CTRL */ +#define AIC31XX_HSPLUGDET_MASK 0x80 +#define AIC31XX_BUTTONPRESSDET_MASK 0x40 +#define AIC31XX_DRCTHRES_MASK 0x20 +#define AIC31XX_AGCNOISE_MASK 0x10 +#define AIC31XX_OC_MASK 0x08 +#define AIC31XX_ENGINE_MASK 0x04 + +/* AIC31XX_DACSETUP */ +#define AIC31XX_SOFTSTEP_MASK 0x03 + +/* AIC31XX_DACMUTE */ +#define AIC31XX_DACMUTE_MASK 0x0C + +/* AIC31XX_MICBIAS */ +#define AIC31XX_MICBIAS_MASK 0x03 +#define AIC31XX_MICBIAS_SHIFT 0 + +#endif /* _TLV320AIC31XX_H */ -- 1.7.9.5

2 1

[alsa-devel] ASoC updates for v3.15
by Mark Brown 13 Mar '14

13 Mar '14

The following changes since commit deeed33850c8a376addabbf971df433b2a1ba74c: Merge remote-tracking branches 'asoc/topic/wm8996', 'asoc/topic/wm9081' and 'asoc/topic/wm9705' into asoc-next (2014-03-12 23:05:31 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.15-2 for you to fetch changes up to 56fe48976db38ed3984e0e68b71e503e8179dd89: Merge remote-tracking branch 'asoc/topic/tlv320aic31xx' into asoc-next (2014-03-13 14:19:54 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.15 This is mostly a few additional fixes from Lars-Peter, a new driver and cleaning up a git failure with merging the Intel branch (combined with an xargs failure to pay attention to error codes). The history lists a bunch of additional commits for the branch but the content of those commits is actually present already but not recorded in history due to git failing. Unfortunately xargs is used in the merge script and it doesn't do a good job of noticing errors from the commands it invokes. ---------------------------------------------------------------- Dan Carpenter (4): ASoC: Intel: sst-firmware: missing curly braces (harmless) ASoC: intel: restore IRQs on error ASoC: intel: incorrect sizeof() ASoC: Baytrail: fix error handling in sst_byt_dsp_init() Jarkko Nikula (15): ASoC: Intel: Add common SST driver loader on ACPI systems ASoC: Intel: Move extended fw base and size fields in struct sst_pdata ASoC: Intel: sst-acpi: Request firmware before SST platform driver probing ASoC: Intel: sst-acpi: Add support for multiple machine drivers per platform ASoC: Intel: Add Baytrail SST ID and Baytrail specific register bits ASoC: Intel: Add Intel Baytrail SST DSP support ASoC: Intel: Add Intel Baytrail SST DSP IPC support ASoC: Intel: Add Intel Baytrail SST PCM platform driver ASoC: Intel: Add machine driver for Baytrail SST with RT5640 codec ASoC: Intel: Add Baytrail SST and byt-rt5640 machine driver probing ASoC: Intel: Add build support for Baytrail SST ASoC: Intel: sst-acpi: Fix Oops in case of missing firmware ASoC: Intel: byt-rt5640: Update internal mic and speaker kcontrol names ASoC: Intel: byt-rt5640: Use init time DAI format ASoC: Intel: Baytrail: Fix implicit declaration of function 'memcpy_fromio' Jyri Sarha (1): ASoC: tlv320aic31xx: Add basic codec driver implementation Lars-Peter Clausen (5): ASoC: ams-delta: Fix compile error ASoC: snappercl15: Convert to table based DAPM setup ASoC: pxa: magician: Convert to table based DAPM and control setup ASoC: pxa: tosa: Convert to table based DAPM and control setup ASoC: s6105-ipcam: Convert to table based DAPM setup Liam Girdwood (18): ASoC: Intel: Add a mfld prefix to Intel SST drivers. ASoC: Intel: Add Intel SST audio DSP low level shim driver. ASoC: Intel: Add Intel SST audio DSP Firmware loader. ASoC: Intel: Add trace support for generic SST IPC messages. ASoC: Intel: Add build support for Intel SST DSP core. ASoC: Intel: Add GFP_KERNEL flag to firmware DMA buffer. ASoC: Intel: Rename SST trace event header to be less generic. ASoC: Intel: Fix sparse warnings for firmware loader ASoC: Intel: Fix build for sst-dsp.c on PPC architecture ASoC: Intel: Add support for Haswell/Broadwell DSP ASoC: Intel: Add Haswell/Broadwell IPC ASoC: Intel: Add Haswell and Broadwell PCM platform driver ASoC: Intel: Add trace support for Haswell/Broadwell SST IPC messages. ASoC: Intel: Add build support for Haswell ADSP ASoC: Intel: Add Haswell Machine support ASoC: Intel: Check Haswell IPC process_reply/notification return value. ASoC: Intel: Use .dai_fmt for setting Haswell BE format. ASoC: Intel: Clean up indentation for Haswell machine driver/Kconfig Mark Brown (26): MAINTAINERS: Add entry for Cirrus CODEC drivers ASoC: io: Remove SPI support Merge remote-tracking branches 'asoc/topic/ak4671', 'asoc/topic/cs42l51' and 'asoc/topic/alc5623' into asoc-io ASoC: io: Remove SND_SOC_I2C ASoC: da732x: Replace hw_read usage with snd_soc_read() ASoC: io: Remove hw_read() operation Merge branches 'topic/ad193x', 'topic/tlv320aic23', 'topic/tlv320aic32x4', 'topic/wm8991', 'fix/si476x' and 'fix/88pm860' of git://git.kernel.org/.../broonie/sound into asoc-io Merge branch 'topic/dapm' of git://git.kernel.org/.../broonie/sound into asoc-pxa Merge tag 'asoc-v3.15' into asoc-intel Merge tag 'asoc-v3.14-rc6' into asoc-linus Merge remote-tracking branch 'asoc/fix/spear' into asoc-linus Merge tag 'asoc-v3.15' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/cirrus' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/io' into asoc-next Merge remote-tracking branch 'asoc/topic/jack' into asoc-next Merge remote-tracking branch 'asoc/topic/kirkwood' into asoc-next Merge remote-tracking branch 'asoc/topic/lm4857' into asoc-next Merge remote-tracking branch 'asoc/topic/lm49453' into asoc-next Merge remote-tracking branch 'asoc/topic/maintainers' into asoc-next Merge remote-tracking branch 'asoc/topic/max98088' into asoc-next Merge remote-tracking branch 'asoc/topic/max98095' into asoc-next Merge remote-tracking branch 'asoc/topic/pxa' into asoc-next Merge remote-tracking branch 'asoc/topic/s6000' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic31xx' into asoc-next Stephen Warren (1): ASoC: Intel: don't select RT5640 if !I2C Takashi Iwai (4): ASoC: lm4857: Use SOC_ENUM_SINGLE_EXT_DECL() ASoC: max98088: Use SOC_*_ENUM_SINGLE_DECL() ASoC: max98095: Use SOC_ENUM_SINGLE_DECL() ASoC: lm49453: Use SOC_ENUM_SINGLE_DECL() Thomas Petazzoni (2): ASoC: kirkwood: enable Kirkwood driver for mvebu platforms sound: ASoC: add ASoC board driver for Armada 370 DB Xiubo Li (6): ASoC: io: Clean up snd_soc_codec_set_cache_io() ASoC: core: Set the default I/O up try regmap. ASoC: sgtl5000: Simplify ASoC probe code ASoC: codec: Simplify ASoC probe code. ASoC: io: New signature for snd_soc_codec_set_cache_io() ASoC: core: Fix check before setting default I/O up try regmap xiangxiao (2): ASoC: add data field into snd_soc_jack_gpio ASoC: delay the initial jack detect by debounce_time .../bindings/sound/armada-370db-audio.txt | 24 + .../devicetree/bindings/sound/mvebu-audio.txt | 1 + .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + MAINTAINERS | 7 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 8 + include/sound/soc.h | 13 +- sound/soc/cirrus/snappercl15.c | 18 +- sound/soc/codecs/88pm860x-codec.c | 3 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad193x.c | 10 +- sound/soc/codecs/adau1373.c | 7 - sound/soc/codecs/adav80x.c | 7 - sound/soc/codecs/ak4535.c | 9 - sound/soc/codecs/ak4641.c | 8 - sound/soc/codecs/ak4642.c | 8 - sound/soc/codecs/ak4671.c | 12 +- sound/soc/codecs/alc5623.c | 7 - sound/soc/codecs/alc5632.c | 8 - sound/soc/codecs/cq93vc.c | 3 +- sound/soc/codecs/cs4270.c | 9 - sound/soc/codecs/cs42l51.c | 6 - sound/soc/codecs/cs42l52.c | 9 +- sound/soc/codecs/cs42l73.c | 11 +- sound/soc/codecs/da7210.c | 8 - sound/soc/codecs/da7213.c | 8 - sound/soc/codecs/da732x.c | 29 +- sound/soc/codecs/da9055.c | 8 - sound/soc/codecs/isabelle.c | 16 - sound/soc/codecs/lm4857.c | 3 +- sound/soc/codecs/lm49453.c | 31 +- sound/soc/codecs/max9768.c | 5 - sound/soc/codecs/max98088.c | 45 +- sound/soc/codecs/max98090.c | 8 - sound/soc/codecs/max98095.c | 56 +- sound/soc/codecs/max9850.c | 8 - sound/soc/codecs/mc13783.c | 4 +- sound/soc/codecs/ml26124.c | 10 - sound/soc/codecs/rt5631.c | 9 - sound/soc/codecs/rt5640.c | 8 - sound/soc/codecs/sgtl5000.c | 8 - sound/soc/codecs/si476x.c | 6 +- sound/soc/codecs/sn95031.c | 2 - sound/soc/codecs/ssm2518.c | 10 - sound/soc/codecs/ssm2602.c | 7 - sound/soc/codecs/sta32x.c | 14 - sound/soc/codecs/sta529.c | 10 - sound/soc/codecs/tlv320aic23.c | 8 - sound/soc/codecs/tlv320aic26.c | 2 - sound/soc/codecs/tlv320aic31xx.c | 1295 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ sound/soc/codecs/tlv320aic32x4.c | 2 - sound/soc/codecs/tlv320aic3x.c | 6 - sound/soc/codecs/tlv320dac33.c | 1 - sound/soc/codecs/wm2000.c | 2 - sound/soc/codecs/wm2200.c | 7 - sound/soc/codecs/wm5100.c | 7 - sound/soc/codecs/wm5102.c | 4 +- sound/soc/codecs/wm5110.c | 3 +- sound/soc/codecs/wm8350.c | 4 +- sound/soc/codecs/wm8400.c | 3 +- sound/soc/codecs/wm8510.c | 10 +- sound/soc/codecs/wm8523.c | 7 - sound/soc/codecs/wm8580.c | 6 - sound/soc/codecs/wm8711.c | 6 - sound/soc/codecs/wm8728.c | 11 +- sound/soc/codecs/wm8731.c | 7 - sound/soc/codecs/wm8737.c | 6 - sound/soc/codecs/wm8741.c | 6 - sound/soc/codecs/wm8750.c | 6 - sound/soc/codecs/wm8753.c | 7 - sound/soc/codecs/wm8770.c | 6 - sound/soc/codecs/wm8776.c | 6 - sound/soc/codecs/wm8804.c | 8 - sound/soc/codecs/wm8900.c | 8 +- sound/soc/codecs/wm8903.c | 10 +- sound/soc/codecs/wm8904.c | 9 - sound/soc/codecs/wm8940.c | 6 - sound/soc/codecs/wm8955.c | 8 - sound/soc/codecs/wm8960.c | 6 - sound/soc/codecs/wm8961.c | 7 - sound/soc/codecs/wm8962.c | 7 - sound/soc/codecs/wm8971.c | 6 - sound/soc/codecs/wm8974.c | 6 - sound/soc/codecs/wm8978.c | 8 +- sound/soc/codecs/wm8983.c | 6 - sound/soc/codecs/wm8985.c | 7 - sound/soc/codecs/wm8988.c | 8 - sound/soc/codecs/wm8990.c | 8 - sound/soc/codecs/wm8991.c | 8 - sound/soc/codecs/wm8993.c | 7 - sound/soc/codecs/wm8994.c | 3 +- sound/soc/codecs/wm8995.c | 7 - sound/soc/codecs/wm8996.c | 12 +- sound/soc/codecs/wm8997.c | 4 +- sound/soc/codecs/wm9081.c | 11 +- sound/soc/codecs/wm9090.c | 10 - sound/soc/intel/Kconfig | 4 +- sound/soc/kirkwood/Kconfig | 10 +- sound/soc/kirkwood/Makefile | 2 + sound/soc/kirkwood/armada-370-db.c | 120 ++ sound/soc/kirkwood/kirkwood-i2s.c | 1 + sound/soc/omap/ams-delta.c | 2 +- sound/soc/pxa/magician.c | 34 +- sound/soc/pxa/tosa.c | 35 +- sound/soc/s6000/s6105-ipcam.c | 28 +- sound/soc/soc-core.c | 14 +- sound/soc/soc-io.c | 64 +- sound/soc/soc-jack.c | 5 +- 109 files changed, 1958 insertions(+), 795 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/armada-370db-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h create mode 100644 sound/soc/kirkwood/armada-370-db.c

4 3

[alsa-devel] [PATCH v5] ASoC: tlv320aic31xx: Add basic codec driver implementation
by Jyri Sarha 13 Mar '14

13 Mar '14

This commit adds a bare bones driver support for TLV320AIC31XX family audio codecs. The driver adds basic stereo playback trough headphone and speaker outputs and mono capture trough microphone inputs. The driver is currently missing support at least for mini DSP features and jack detection. I have tested the driver only on TLV320AIC3111, but based on the data sheets TLV320AIC3100, TLV320AIC3110, and TLV320AIC3120 should work Ok too. The base for the implementation was taken from: git@gitorious.org:ti-codecs/ti-codecs.git ajitk/topics/k3.10.1-aic31xx -branch at commit 77504eba0294764e9e63b4a0c696b44db187cd13. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Since v4 version: - Remove MICBIAS_OFF DT parameter - Remove logging and add missing default: to aic31xx_dapm_power_event() - Take control, widget, and route adding errors into account - Don't try soft reset in power event handler - Remove route to MICBIAS from codec driver damp routing table and add it as output pin to DT doc - Update year in copyright message and change triple newlines to double - Remove internal DAC/ADC routes that were still haunting in the previous patch - Allow PLL reprogramming on the fly as it appears to work just fine BR, Jyri .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 8 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic31xx.c | 1295 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ 6 files changed, 1628 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt new file mode 100644 index 0000000..74c66de --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt @@ -0,0 +1,61 @@ +Texas Instruments - tlv320aic31xx Codec module + +The tlv320aic31xx serial control bus communicates through I2C protocols + +Required properties: + +- compatible - "string" - One of: + "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp + "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp + "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP) + "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP) + "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP) + "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP) + +- reg - <int> - I2C slave address + + +Optional properties: + +- gpio-reset - gpio pin number used for codec reset +- ai31xx-micbias-vg - MicBias Voltage setting + 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V + 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V + 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD + If this node is not mentioned or if the value is unknown, then + micbias is set to 2.0V. +- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply, + DVDD-supply : power supplies for the device as covered in + Documentation/devicetree/bindings/regulator/regulator.txt + +CODEC output pins: + * HPL + * HPR + * SPL, devices with stereo speaker amp + * SPR, devices with stereo speaker amp + * SPK, devices with mono speaker amp + * MICBIAS + +CODEC input pins: + * MIC1LP + * MIC1RP + * MIC1LM + +The pins can be used in referring sound node's audio-routing property. + +Example: +#include <dt-bindings/sound/tlv320aic31xx-micbias.h> + +tlv320aic31xx: tlv320aic31xx@18 { + compatible = "ti,tlv320aic311x"; + reg = <0x18>; + + ai31xx-micbias-vg = <MICBIAS_OFF>; + + HPVDD-supply = <&regulator>; + SPRVDD-supply = <&regulator>; + SPLVDD-supply = <&regulator>; + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; +}; diff --git a/include/dt-bindings/sound/tlv320aic31xx-micbias.h b/include/dt-bindings/sound/tlv320aic31xx-micbias.h new file mode 100644 index 0000000..f5cb772 --- /dev/null +++ b/include/dt-bindings/sound/tlv320aic31xx-micbias.h @@ -0,0 +1,8 @@ +#ifndef __DT_TLV320AIC31XX_MICBIAS_H +#define __DT_TLV320AIC31XX_MICBIAS_H + +#define MICBIAS_2_0V 1 +#define MICBIAS_2_5V 2 +#define MICBIAS_AVDDV 3 + +#endif /* __DT_TLV320AIC31XX_MICBIAS_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e19b64f..af3c049 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -83,6 +83,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TAS5086 if I2C select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER + select SND_SOC_TLV320AIC31XX if I2C select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C @@ -428,6 +429,9 @@ config SND_SOC_TLV320AIC26 tristate depends on SPI +config SND_SOC_TLV320AIC31XX + tristate + config SND_SOC_TLV320AIC32X4 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 90ab24b..018e3f3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -75,6 +75,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o +snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o @@ -215,6 +216,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o +obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c new file mode 100644 index 0000000..e60e37b --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -0,0 +1,1295 @@ +/* + * ALSA SoC TLV320AIC31XX codec driver + * + * Copyright (C) 2014 Texas Instruments, Inc. + * + * Author: Jyri Sarha <jsarha(a)ti.com> + * + * Based on ground work by: Ajit Kulkarni <x0175765(a)ti.com> + * + * This package 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 PACKAGE IS PROVIDED AS IS AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + * + * The TLV320AIC31xx series of audio codec is a low-power, highly integrated + * high performance codec which provides a stereo DAC, a mono ADC, + * and mono/stereo Class-D speaker driver. + */ + +#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/gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <dt-bindings/sound/tlv320aic31xx-micbias.h> + +#include "tlv320aic31xx.h" + +static const struct reg_default aic31xx_reg_defaults[] = { + { AIC31XX_CLKMUX, 0x00 }, + { AIC31XX_PLLPR, 0x11 }, + { AIC31XX_PLLJ, 0x04 }, + { AIC31XX_PLLDMSB, 0x00 }, + { AIC31XX_PLLDLSB, 0x00 }, + { AIC31XX_NDAC, 0x01 }, + { AIC31XX_MDAC, 0x01 }, + { AIC31XX_DOSRMSB, 0x00 }, + { AIC31XX_DOSRLSB, 0x80 }, + { AIC31XX_NADC, 0x01 }, + { AIC31XX_MADC, 0x01 }, + { AIC31XX_AOSR, 0x80 }, + { AIC31XX_IFACE1, 0x00 }, + { AIC31XX_DATA_OFFSET, 0x00 }, + { AIC31XX_IFACE2, 0x00 }, + { AIC31XX_BCLKN, 0x01 }, + { AIC31XX_DACSETUP, 0x14 }, + { AIC31XX_DACMUTE, 0x0c }, + { AIC31XX_LDACVOL, 0x00 }, + { AIC31XX_RDACVOL, 0x00 }, + { AIC31XX_ADCSETUP, 0x00 }, + { AIC31XX_ADCFGA, 0x80 }, + { AIC31XX_ADCVOL, 0x00 }, + { AIC31XX_HPDRIVER, 0x04 }, + { AIC31XX_SPKAMP, 0x06 }, + { AIC31XX_DACMIXERROUTE, 0x00 }, + { AIC31XX_LANALOGHPL, 0x7f }, + { AIC31XX_RANALOGHPR, 0x7f }, + { AIC31XX_LANALOGSPL, 0x7f }, + { AIC31XX_RANALOGSPR, 0x7f }, + { AIC31XX_HPLGAIN, 0x02 }, + { AIC31XX_HPRGAIN, 0x02 }, + { AIC31XX_SPLGAIN, 0x00 }, + { AIC31XX_SPRGAIN, 0x00 }, + { AIC31XX_MICBIAS, 0x00 }, + { AIC31XX_MICPGA, 0x80 }, + { AIC31XX_MICPGAPI, 0x00 }, + { AIC31XX_MICPGAMI, 0x00 }, +}; + +static bool aic31xx_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC31XX_PAGECTL: /* regmap implementation requires this */ + case AIC31XX_RESET: /* always clears after write */ + case AIC31XX_OT_FLAG: + case AIC31XX_ADCFLAG: + case AIC31XX_DACFLAG1: + case AIC31XX_DACFLAG2: + case AIC31XX_OFFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG2: + case AIC31XX_INTRADCFLAG2: + return true; + } + return false; +} + +static bool aic31xx_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC31XX_OT_FLAG: + case AIC31XX_ADCFLAG: + case AIC31XX_DACFLAG1: + case AIC31XX_DACFLAG2: + case AIC31XX_OFFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG2: + case AIC31XX_INTRADCFLAG2: + return false; + } + return true; +} + +static const struct regmap_range_cfg aic31xx_ranges[] = { + { + .range_min = 0, + .range_max = 12 * 128, + .selector_reg = AIC31XX_PAGECTL, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +struct regmap_config aic31xx_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .writeable_reg = aic31xx_writeable, + .volatile_reg = aic31xx_volatile, + .reg_defaults = aic31xx_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(aic31xx_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .ranges = aic31xx_ranges, + .num_ranges = ARRAY_SIZE(aic31xx_ranges), + .max_register = 12 * 128, +}; + +#define AIC31XX_NUM_SUPPLIES 6 +static const char * const aic31xx_supply_names[AIC31XX_NUM_SUPPLIES] = { + "HPVDD", + "SPRVDD", + "SPLVDD", + "AVDD", + "IOVDD", + "DVDD", +}; + +struct aic31xx_disable_nb { + struct notifier_block nb; + struct aic31xx_priv *aic31xx; +}; + +struct aic31xx_priv { + struct snd_soc_codec *codec; + u8 i2c_regs_status; + struct device *dev; + struct regmap *regmap; + struct aic31xx_pdata pdata; + struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; + struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES]; + unsigned int sysclk; + int rate_div_line; +}; + +struct aic31xx_rate_divs { + u32 mclk; + u32 rate; + u8 p_val; + u8 pll_j; + u16 pll_d; + u16 dosr; + u8 ndac; + u8 mdac; + u8 aosr; + u8 nadc; + u8 madc; +}; + +/* ADC dividers can be disabled by cofiguring them to 0 */ +static const struct aic31xx_rate_divs aic31xx_divs[] = { + /* mclk rate pll: p j d dosr ndac mdac aors nadc madc */ + /* 8k rate */ + {12000000, 8000, 1, 8, 1920, 128, 48, 2, 128, 48, 2}, + {24000000, 8000, 2, 8, 1920, 128, 48, 2, 128, 48, 2}, + {25000000, 8000, 2, 7, 8643, 128, 48, 2, 128, 48, 2}, + /* 11.025k rate */ + {12000000, 11025, 1, 7, 5264, 128, 32, 2, 128, 32, 2}, + {24000000, 11025, 2, 7, 5264, 128, 32, 2, 128, 32, 2}, + {25000000, 11025, 2, 7, 2253, 128, 32, 2, 128, 32, 2}, + /* 16k rate */ + {12000000, 16000, 1, 8, 1920, 128, 24, 2, 128, 24, 2}, + {24000000, 16000, 2, 8, 1920, 128, 24, 2, 128, 24, 2}, + {25000000, 16000, 2, 7, 8643, 128, 24, 2, 128, 24, 2}, + /* 22.05k rate */ + {12000000, 22050, 1, 7, 5264, 128, 16, 2, 128, 16, 2}, + {24000000, 22050, 2, 7, 5264, 128, 16, 2, 128, 16, 2}, + {25000000, 22050, 2, 7, 2253, 128, 16, 2, 128, 16, 2}, + /* 32k rate */ + {12000000, 32000, 1, 8, 1920, 128, 12, 2, 128, 12, 2}, + {24000000, 32000, 2, 8, 1920, 128, 12, 2, 128, 12, 2}, + {25000000, 32000, 2, 7, 8643, 128, 12, 2, 128, 12, 2}, + /* 44.1k rate */ + {12000000, 44100, 1, 7, 5264, 128, 8, 2, 128, 8, 2}, + {24000000, 44100, 2, 7, 5264, 128, 8, 2, 128, 8, 2}, + {25000000, 44100, 2, 7, 2253, 128, 8, 2, 128, 8, 2}, + /* 48k rate */ + {12000000, 48000, 1, 8, 1920, 128, 8, 2, 128, 8, 2}, + {24000000, 48000, 2, 8, 1920, 128, 8, 2, 128, 8, 2}, + {25000000, 48000, 2, 7, 8643, 128, 8, 2, 128, 8, 2}, + /* 88.2k rate */ + {12000000, 88200, 1, 7, 5264, 64, 8, 2, 64, 8, 2}, + {24000000, 88200, 2, 7, 5264, 64, 8, 2, 64, 8, 2}, + {25000000, 88200, 2, 7, 2253, 64, 8, 2, 64, 8, 2}, + /* 96k rate */ + {12000000, 96000, 1, 8, 1920, 64, 8, 2, 64, 8, 2}, + {24000000, 96000, 2, 8, 1920, 64, 8, 2, 64, 8, 2}, + {25000000, 96000, 2, 7, 8643, 64, 8, 2, 64, 8, 2}, + /* 176.4k rate */ + {12000000, 176400, 1, 7, 5264, 32, 8, 2, 32, 8, 2}, + {24000000, 176400, 2, 7, 5264, 32, 8, 2, 32, 8, 2}, + {25000000, 176400, 2, 7, 2253, 32, 8, 2, 32, 8, 2}, + /* 192k rate */ + {12000000, 192000, 1, 8, 1920, 32, 8, 2, 32, 8, 2}, + {24000000, 192000, 2, 8, 1920, 32, 8, 2, 32, 8, 2}, + {25000000, 192000, 2, 7, 8643, 32, 8, 2, 32, 8, 2}, +}; + +static const char * const ldac_in_text[] = { + "Off", "Left Data", "Right Data", "Mono" +}; + +static const char * const rdac_in_text[] = { + "Off", "Right Data", "Left Data", "Mono" +}; + +static SOC_ENUM_SINGLE_DECL(ldac_in_enum, AIC31XX_DACSETUP, 4, ldac_in_text); + +static SOC_ENUM_SINGLE_DECL(rdac_in_enum, AIC31XX_DACSETUP, 2, rdac_in_text); + +static const char * const mic_select_text[] = { + "Off", "FFR 10 Ohm", "FFR 20 Ohm", "FFR 40 Ohm" +}; + +static const +SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, mic_select_text); + +static const +SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, mic_select_text); + +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_fgain_tlv, 0, 10, 0); +static const DECLARE_TLV_DB_SCALE(adc_cgain_tlv, -2000, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_pga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_SCALE(hp_drv_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(class_D_drv_tlv, 600, 600, 0); +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(sp_vol_tlv, -6350, 50, 0); + +/* + * controls to be exported to the user space + */ +static const struct snd_kcontrol_new aic31xx_snd_controls[] = { + SOC_DOUBLE_R_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL, + AIC31XX_RDACVOL, 0, -127, 48, 7, 0, dac_vol_tlv), + + SOC_SINGLE_TLV("ADC Fine Capture Volume", AIC31XX_ADCFGA, 4, 4, 1, + adc_fgain_tlv), + + SOC_SINGLE("ADC Capture Switch", AIC31XX_ADCFGA, 7, 1, 1), + SOC_DOUBLE_R_S_TLV("ADC Capture Volume", AIC31XX_ADCVOL, AIC31XX_ADCVOL, + 0, -24, 40, 6, 0, adc_cgain_tlv), + + SOC_SINGLE_TLV("Mic PGA Capture Volume", AIC31XX_MICPGA, 0, + 119, 0, mic_pga_tlv), + + SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 2, 1, 0), + SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv), + + SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL, + AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv), +}; + +static const struct snd_kcontrol_new aic311x_snd_controls[] = { + SOC_DOUBLE_R("Speaker Driver Playback Switch", AIC31XX_SPLGAIN, + AIC31XX_SPRGAIN, 2, 1, 0), + SOC_DOUBLE_R_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN, + AIC31XX_SPRGAIN, 3, 3, 0, class_D_drv_tlv), + + SOC_DOUBLE_R_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL, + AIC31XX_RANALOGSPR, 0, 0x7F, 1, sp_vol_tlv), +}; + +static const struct snd_kcontrol_new aic310x_snd_controls[] = { + SOC_SINGLE("Speaker Driver Playback Switch", AIC31XX_SPLGAIN, + 2, 1, 0), + SOC_SINGLE_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN, + 3, 3, 0, class_D_drv_tlv), + + SOC_SINGLE_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL, + 0, 0x7F, 1, sp_vol_tlv), +}; + +static const struct snd_kcontrol_new ldac_in_control = + SOC_DAPM_ENUM("DAC Left Input", ldac_in_enum); + +static const struct snd_kcontrol_new rdac_in_control = + SOC_DAPM_ENUM("DAC Right Input", rdac_in_enum); + +int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, + unsigned int mask, unsigned int wbits, int sleep, + int count) +{ + unsigned int bits; + int counter = count; + int ret = regmap_read(aic31xx->regmap, reg, &bits); + while ((bits & mask) != wbits && counter && !ret) { + usleep_range(sleep, sleep * 2); + ret = regmap_read(aic31xx->regmap, reg, &bits); + counter--; + } + if ((bits & mask) != wbits) { + dev_err(aic31xx->dev, + "%s: Failed! 0x%x was 0x%x expected 0x%x (%d, 0x%x, %d us)\n", + __func__, reg, bits, wbits, ret, mask, + (count - counter) * sleep); + ret = -1; + } + return ret; +} + +#define WIDGET_BIT(reg, shift) (((shift) << 8) | (reg)) + +static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(w->codec); + unsigned int reg = AIC31XX_DACFLAG1; + unsigned int mask; + + switch (WIDGET_BIT(w->reg, w->shift)) { + case WIDGET_BIT(AIC31XX_DACSETUP, 7): + mask = AIC31XX_LDACPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_DACSETUP, 6): + mask = AIC31XX_RDACPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_HPDRIVER, 7): + mask = AIC31XX_HPLDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_HPDRIVER, 6): + mask = AIC31XX_HPRDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_SPKAMP, 7): + mask = AIC31XX_SPLDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_SPKAMP, 6): + mask = AIC31XX_SPRDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_ADCSETUP, 7): + mask = AIC31XX_ADCPWRSTATUS_MASK; + reg = AIC31XX_ADCFLAG; + break; + default: + dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n", + w->name, __func__); + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + return aic31xx_wait_bits(aic31xx, reg, mask, mask, 5000, 100); + case SND_SOC_DAPM_POST_PMD: + return aic31xx_wait_bits(aic31xx, reg, mask, 0, 5000, 100); + default: + dev_dbg(w->codec->dev, + "Unhandled dapm widget event %d from %s\n", + event, w->name); + } + return 0; +} + +static const struct snd_kcontrol_new left_output_switches[] = { + SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0), + SOC_DAPM_SINGLE("From MIC1LP", AIC31XX_DACMIXERROUTE, 5, 1, 0), + SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 4, 1, 0), +}; + +static const struct snd_kcontrol_new right_output_switches[] = { + SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0), + SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 1, 1, 0), +}; + +static const struct snd_kcontrol_new p_term_mic1lp = + SOC_DAPM_ENUM("MIC1LP P-Terminal", mic1lp_p_enum); + +static const struct snd_kcontrol_new p_term_mic1rp = + SOC_DAPM_ENUM("MIC1RP P-Terminal", mic1rp_p_enum); + +static const struct snd_kcontrol_new p_term_mic1lm = + SOC_DAPM_ENUM("MIC1LM P-Terminal", mic1lm_p_enum); + +static const struct snd_kcontrol_new m_term_mic1lm = + SOC_DAPM_ENUM("MIC1LM M-Terminal", mic1lm_m_enum); + +static const struct snd_kcontrol_new aic31xx_dapm_hpl_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGHPL, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_hpr_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGHPR, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_spl_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGSPL, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_spr_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGSPR, 7, 1, 0); + +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* change mic bias voltage to user defined */ + snd_soc_update_bits(codec, AIC31XX_MICBIAS, + AIC31XX_MICBIAS_MASK, + aic31xx->pdata.micbias_vg << + AIC31XX_MICBIAS_SHIFT); + dev_dbg(codec->dev, "%s: turned on\n", __func__); + break; + case SND_SOC_DAPM_PRE_PMD: + /* turn mic bias off */ + snd_soc_update_bits(codec, AIC31XX_MICBIAS, + AIC31XX_MICBIAS_MASK, 0); + dev_dbg(codec->dev, "%s: turned off\n", __func__); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("DAC Left Input", + SND_SOC_NOPM, 0, 0, &ldac_in_control), + SND_SOC_DAPM_MUX("DAC Right Input", + SND_SOC_NOPM, 0, 0, &rdac_in_control), + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC Left", "Left Playback", + AIC31XX_DACSETUP, 7, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_DAC_E("DAC Right", "Right Playback", + AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0, + left_output_switches, + ARRAY_SIZE(left_output_switches)), + SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0, + right_output_switches, + ARRAY_SIZE(right_output_switches)), + + SND_SOC_DAPM_SWITCH("HP Left", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_hpl_switch), + SND_SOC_DAPM_SWITCH("HP Right", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_hpr_switch), + + /* Output drivers */ + SND_SOC_DAPM_OUT_DRV_E("HPL Driver", AIC31XX_HPDRIVER, 7, 0, + NULL, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_OUT_DRV_E("HPR Driver", AIC31XX_HPDRIVER, 6, 0, + NULL, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + + /* ADC */ + SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + + /* Input Selection to MIC_PGA */ + SND_SOC_DAPM_MUX("MIC1LP P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1lp), + SND_SOC_DAPM_MUX("MIC1RP P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1rp), + SND_SOC_DAPM_MUX("MIC1LM P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1lm), + + SND_SOC_DAPM_MUX("MIC1LM M-Terminal", SND_SOC_NOPM, 0, 0, + &m_term_mic1lm), + /* Enabling & Disabling MIC Gain Ctl */ + SND_SOC_DAPM_PGA("MIC_GAIN_CTL", AIC31XX_MICPGA, + 7, 1, NULL, 0), + + /* Mic Bias */ + SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC1LP"), + SND_SOC_DAPM_INPUT("MIC1RP"), + SND_SOC_DAPM_INPUT("MIC1LM"), +}; + +static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = { + /* AIC3111 and AIC3110 have stereo class-D amplifier */ + SND_SOC_DAPM_OUT_DRV_E("SPL ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUT_DRV_E("SPR ClassD", AIC31XX_SPKAMP, 6, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("Speaker Left", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spl_switch), + SND_SOC_DAPM_SWITCH("Speaker Right", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spr_switch), + SND_SOC_DAPM_OUTPUT("SPL"), + SND_SOC_DAPM_OUTPUT("SPR"), +}; + +/* AIC3100 and AIC3120 have only mono class-D amplifier */ +static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = { + SND_SOC_DAPM_OUT_DRV_E("SPK ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("Speaker", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spl_switch), + SND_SOC_DAPM_OUTPUT("SPK"), +}; + +static const struct snd_soc_dapm_route +aic31xx_audio_map[] = { + /* DAC Input Routing */ + {"DAC Left Input", "Left Data", "DAC IN"}, + {"DAC Left Input", "Right Data", "DAC IN"}, + {"DAC Left Input", "Mono", "DAC IN"}, + {"DAC Right Input", "Left Data", "DAC IN"}, + {"DAC Right Input", "Right Data", "DAC IN"}, + {"DAC Right Input", "Mono", "DAC IN"}, + {"DAC Left", NULL, "DAC Left Input"}, + {"DAC Right", NULL, "DAC Right Input"}, + + /* Mic input */ + {"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"}, + {"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"}, + {"MIC1LP P-Terminal", "FFR 40 Ohm", "MIC1LP"}, + {"MIC1RP P-Terminal", "FFR 10 Ohm", "MIC1RP"}, + {"MIC1RP P-Terminal", "FFR 20 Ohm", "MIC1RP"}, + {"MIC1RP P-Terminal", "FFR 40 Ohm", "MIC1RP"}, + {"MIC1LM P-Terminal", "FFR 10 Ohm", "MIC1LM"}, + {"MIC1LM P-Terminal", "FFR 20 Ohm", "MIC1LM"}, + {"MIC1LM P-Terminal", "FFR 40 Ohm", "MIC1LM"}, + + {"MIC1LM M-Terminal", "FFR 10 Ohm", "MIC1LM"}, + {"MIC1LM M-Terminal", "FFR 20 Ohm", "MIC1LM"}, + {"MIC1LM M-Terminal", "FFR 40 Ohm", "MIC1LM"}, + + {"MIC_GAIN_CTL", NULL, "MIC1LP P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1RP P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1LM P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1LM M-Terminal"}, + + {"ADC", NULL, "MIC_GAIN_CTL"}, + + /* Left Output */ + {"Output Left", "From Left DAC", "DAC Left"}, + {"Output Left", "From MIC1LP", "MIC1LP"}, + {"Output Left", "From MIC1RP", "MIC1RP"}, + + /* Right Output */ + {"Output Right", "From Right DAC", "DAC Right"}, + {"Output Right", "From MIC1RP", "MIC1RP"}, + + /* HPL path */ + {"HP Left", "Switch", "Output Left"}, + {"HPL Driver", NULL, "HP Left"}, + {"HPL", NULL, "HPL Driver"}, + + /* HPR path */ + {"HP Right", "Switch", "Output Right"}, + {"HPR Driver", NULL, "HP Right"}, + {"HPR", NULL, "HPR Driver"}, +}; + +static const struct snd_soc_dapm_route +aic311x_audio_map[] = { + /* SP L path */ + {"Speaker Left", "Switch", "Output Left"}, + {"SPL ClassD", NULL, "Speaker Left"}, + {"SPL", NULL, "SPL ClassD"}, + + /* SP R path */ + {"Speaker Right", "Switch", "Output Right"}, + {"SPR ClassD", NULL, "Speaker Right"}, + {"SPR", NULL, "SPR ClassD"}, +}; + +static const struct snd_soc_dapm_route +aic310x_audio_map[] = { + /* SP L path */ + {"Speaker", "Switch", "Output Left"}, + {"SPK ClassD", NULL, "Speaker"}, + {"SPK", NULL, "SPK ClassD"}, +}; + +static int aic31xx_add_controls(struct snd_soc_codec *codec) +{ + int ret = 0; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) + ret = snd_soc_add_codec_controls( + codec, aic311x_snd_controls, + ARRAY_SIZE(aic311x_snd_controls)); + else + ret = snd_soc_add_codec_controls( + codec, aic310x_snd_controls, + ARRAY_SIZE(aic310x_snd_controls)); + + return ret; +} + +static int aic31xx_add_widgets(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) { + ret = snd_soc_dapm_new_controls( + dapm, aic311x_dapm_widgets, + ARRAY_SIZE(aic311x_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic311x_audio_map, + ARRAY_SIZE(aic311x_audio_map)); + if (ret) + return ret; + } else { + ret = snd_soc_dapm_new_controls( + dapm, aic310x_dapm_widgets, + ARRAY_SIZE(aic310x_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic310x_audio_map, + ARRAY_SIZE(aic310x_audio_map)); + if (ret) + return ret; + } + + return 0; +} + +static int aic31xx_setup_pll(struct snd_soc_codec *codec, + struct snd_pcm_hw_params *params) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int bclk_n = 0; + int i; + + /* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */ + snd_soc_update_bits(codec, AIC31XX_CLKMUX, + AIC31XX_CODEC_CLKIN_MASK, AIC31XX_CODEC_CLKIN_PLL); + snd_soc_update_bits(codec, AIC31XX_IFACE2, + AIC31XX_BDIVCLK_MASK, AIC31XX_DAC2BCLK); + + for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) { + if (aic31xx_divs[i].rate == params_rate(params) && + aic31xx_divs[i].mclk == aic31xx->sysclk) + break; + } + + if (i == ARRAY_SIZE(aic31xx_divs)) { + dev_err(codec->dev, "%s: Sampling rate %u not supported\n", + __func__, params_rate(params)); + return -EINVAL; + } + + /* PLL configuration */ + snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK, + (aic31xx_divs[i].p_val << 4) | 0x01); + snd_soc_write(codec, AIC31XX_PLLJ, aic31xx_divs[i].pll_j); + + snd_soc_write(codec, AIC31XX_PLLDMSB, + aic31xx_divs[i].pll_d >> 8); + snd_soc_write(codec, AIC31XX_PLLDLSB, + aic31xx_divs[i].pll_d & 0xff); + + /* DAC dividers configuration */ + snd_soc_update_bits(codec, AIC31XX_NDAC, AIC31XX_PLL_MASK, + aic31xx_divs[i].ndac); + snd_soc_update_bits(codec, AIC31XX_MDAC, AIC31XX_PLL_MASK, + aic31xx_divs[i].mdac); + + snd_soc_write(codec, AIC31XX_DOSRMSB, aic31xx_divs[i].dosr >> 8); + snd_soc_write(codec, AIC31XX_DOSRLSB, aic31xx_divs[i].dosr & 0xff); + + /* ADC dividers configuration. Write reset value 1 if not used. */ + snd_soc_update_bits(codec, AIC31XX_NADC, AIC31XX_PLL_MASK, + aic31xx_divs[i].nadc ? aic31xx_divs[i].nadc : 1); + snd_soc_update_bits(codec, AIC31XX_MADC, AIC31XX_PLL_MASK, + aic31xx_divs[i].madc ? aic31xx_divs[i].madc : 1); + + snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr); + + /* Bit clock divider configuration. */ + bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) + / snd_soc_params_to_frame_size(params); + if (bclk_n == 0) { + dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n", + __func__); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_BCLKN, + AIC31XX_PLL_MASK, bclk_n); + + aic31xx->rate_div_line = i; + + dev_dbg(codec->dev, + "pll %d.%04d/%d dosr %d n %d m %d aosr %d n %d m %d bclk_n %d\n", + aic31xx_divs[i].pll_j, aic31xx_divs[i].pll_d, + aic31xx_divs[i].p_val, aic31xx_divs[i].dosr, + aic31xx_divs[i].ndac, aic31xx_divs[i].mdac, + aic31xx_divs[i].aosr, aic31xx_divs[i].nadc, + aic31xx_divs[i].madc, bclk_n); + + return 0; +} + +static int aic31xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *tmp) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + u8 data = 0; + + dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n", + __func__, params_format(params), params_width(params), + params_rate(params)); + + switch (params_width(params)) { + case 16: + break; + case 20: + data = (AIC31XX_WORD_LEN_20BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + case 24: + data = (AIC31XX_WORD_LEN_24BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + case 32: + data = (AIC31XX_WORD_LEN_32BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + default: + dev_err(codec->dev, "%s: Unsupported format %d\n", + __func__, params_format(params)); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_IFACE1, + AIC31XX_IFACE1_DATALEN_MASK, + data); + + return aic31xx_setup_pll(codec, params); +} + +static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + if (mute) { + snd_soc_update_bits(codec, AIC31XX_DACMUTE, + AIC31XX_DACMUTE_MASK, + AIC31XX_DACMUTE_MASK); + } else { + snd_soc_update_bits(codec, AIC31XX_DACMUTE, + AIC31XX_DACMUTE_MASK, 0x0); + } + + return 0; +} + +static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 iface_reg1 = 0; + u8 iface_reg3 = 0; + u8 dsp_a_val = 0; + + dev_dbg(codec->dev, "## %s: fmt = 0x%x\n", __func__, fmt); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface_reg1 |= AIC31XX_BCLK_MASTER | AIC31XX_WCLK_MASTER; + break; + default: + dev_alert(codec->dev, "Invalid DAI master/slave interface\n"); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_DSP_A: + dsp_a_val = 0x1; + case SND_SOC_DAIFMT_DSP_B: + /* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + iface_reg3 |= AIC31XX_BCLKINV_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + default: + return -EINVAL; + } + iface_reg1 |= (AIC31XX_DSP_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface_reg1 |= (AIC31XX_RIGHT_JUSTIFIED_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg1 |= (AIC31XX_LEFT_JUSTIFIED_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + default: + dev_err(codec->dev, "Invalid DAI interface format\n"); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_IFACE1, + AIC31XX_IFACE1_DATATYPE_MASK | + AIC31XX_IFACE1_MASTER_MASK, + iface_reg1); + snd_soc_update_bits(codec, AIC31XX_DATA_OFFSET, + AIC31XX_DATA_OFFSET_MASK, + dsp_a_val); + snd_soc_update_bits(codec, AIC31XX_IFACE2, + AIC31XX_BCLKINV_MASK, + iface_reg3); + + return 0; +} + +static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + + dev_dbg(codec->dev, "## %s: clk_id = %d, freq = %d, dir = %d\n", + __func__, clk_id, freq, dir); + + for (i = 0; aic31xx_divs[i].mclk != freq; i++) { + if (i == ARRAY_SIZE(aic31xx_divs)) { + dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n", + __func__, freq); + return -EINVAL; + } + } + + /* set clock on MCLK, BCLK, or GPIO1 as PLL input */ + snd_soc_update_bits(codec, AIC31XX_CLKMUX, AIC31XX_PLL_CLKIN_MASK, + clk_id << AIC31XX_PLL_CLKIN_SHIFT); + + aic31xx->sysclk = freq; + return 0; +} + +static int aic31xx_regulator_event(struct notifier_block *nb, + unsigned long event, void *data) +{ + struct aic31xx_disable_nb *disable_nb = + container_of(nb, struct aic31xx_disable_nb, nb); + struct aic31xx_priv *aic31xx = disable_nb->aic31xx; + + if (event & REGULATOR_EVENT_DISABLE) { + /* + * Put codec to reset and as at least one of the + * supplies was disabled. + */ + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) + gpio_set_value(aic31xx->pdata.gpio_reset, 0); + + regcache_mark_dirty(aic31xx->regmap); + dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__); + } + + return 0; +} + +static void aic31xx_clk_on(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + u8 mask = AIC31XX_PM_MASK; + u8 on = AIC31XX_PM_MASK; + + dev_dbg(codec->dev, "codec clock -> on (rate %d)\n", + aic31xx_divs[aic31xx->rate_div_line].rate); + snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, on); + mdelay(10); + snd_soc_update_bits(codec, AIC31XX_NDAC, mask, on); + snd_soc_update_bits(codec, AIC31XX_MDAC, mask, on); + if (aic31xx_divs[aic31xx->rate_div_line].nadc) + snd_soc_update_bits(codec, AIC31XX_NADC, mask, on); + if (aic31xx_divs[aic31xx->rate_div_line].madc) + snd_soc_update_bits(codec, AIC31XX_MADC, mask, on); + snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, on); +} + +static void aic31xx_clk_off(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + u8 mask = AIC31XX_PM_MASK; + u8 off = 0; + + dev_dbg(codec->dev, "codec clock -> off\n"); + snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, off); + snd_soc_update_bits(codec, AIC31XX_MADC, mask, off); + snd_soc_update_bits(codec, AIC31XX_NADC, mask, off); + snd_soc_update_bits(codec, AIC31XX_MDAC, mask, off); + snd_soc_update_bits(codec, AIC31XX_NDAC, mask, off); + snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, off); +} + +static int aic31xx_power_on(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + ret = regulator_bulk_enable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + if (ret) + return ret; + + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) { + gpio_set_value(aic31xx->pdata.gpio_reset, 1); + udelay(100); + } + regcache_cache_only(aic31xx->regmap, false); + ret = regcache_sync(aic31xx->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to restore cache: %d\n", ret); + regcache_cache_only(aic31xx->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + return ret; + } + return 0; +} + +static int aic31xx_power_off(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + regcache_cache_only(aic31xx->regmap, true); + ret = regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + + return ret; +} + +static int aic31xx_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__, + codec->dapm.bias_level, level); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + aic31xx_clk_on(codec); + break; + case SND_SOC_BIAS_STANDBY: + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + aic31xx_power_on(codec); + break; + case SND_SOC_BIAS_PREPARE: + aic31xx_clk_off(codec); + break; + default: + BUG(); + } + break; + case SND_SOC_BIAS_OFF: + aic31xx_power_off(codec); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int aic31xx_suspend(struct snd_soc_codec *codec) +{ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int aic31xx_resume(struct snd_soc_codec *codec) +{ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int aic31xx_codec_probe(struct snd_soc_codec *codec) +{ + int ret = 0; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + + dev_dbg(aic31xx->dev, "## %s\n", __func__); + + aic31xx = snd_soc_codec_get_drvdata(codec); + codec->control_data = aic31xx->regmap; + + aic31xx->codec = codec; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + + if (ret != 0) { + dev_err(codec->dev, "snd_soc_codec_set_cache_io failed %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) { + aic31xx->disable_nb[i].nb.notifier_call = + aic31xx_regulator_event; + aic31xx->disable_nb[i].aic31xx = aic31xx; + ret = regulator_register_notifier(aic31xx->supplies[i].consumer, + &aic31xx->disable_nb[i].nb); + if (ret) { + dev_err(codec->dev, + "Failed to request regulator notifier: %d\n", + ret); + return ret; + } + } + + regcache_cache_only(aic31xx->regmap, true); + regcache_mark_dirty(aic31xx->regmap); + + ret = aic31xx_add_controls(codec); + if (ret) + return ret; + + ret = aic31xx_add_widgets(codec); + + return ret; +} + +static int aic31xx_codec_remove(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + /* power down chip */ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF); + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) + regulator_unregister_notifier(aic31xx->supplies[i].consumer, + &aic31xx->disable_nb[i].nb); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_driver_aic31xx = { + .probe = aic31xx_codec_probe, + .remove = aic31xx_codec_remove, + .suspend = aic31xx_suspend, + .resume = aic31xx_resume, + .set_bias_level = aic31xx_set_bias_level, + .controls = aic31xx_snd_controls, + .num_controls = ARRAY_SIZE(aic31xx_snd_controls), + .dapm_widgets = aic31xx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic31xx_dapm_widgets), + .dapm_routes = aic31xx_audio_map, + .num_dapm_routes = ARRAY_SIZE(aic31xx_audio_map), +}; + +static struct snd_soc_dai_ops aic31xx_dai_ops = { + .hw_params = aic31xx_hw_params, + .set_sysclk = aic31xx_set_dai_sysclk, + .set_fmt = aic31xx_set_dai_fmt, + .digital_mute = aic31xx_dac_mute, +}; + +static struct snd_soc_dai_driver aic31xx_dai_driver[] = { + { + .name = "tlv320aic31xx-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC31XX_RATES, + .formats = AIC31XX_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AIC31XX_RATES, + .formats = AIC31XX_FORMATS, + }, + .ops = &aic31xx_dai_ops, + .symmetric_rates = 1, + } +}; + +#if defined(CONFIG_OF) +static const struct of_device_id tlv320aic31xx_of_match[] = { + { .compatible = "ti,tlv320aic310x" }, + { .compatible = "ti,tlv320aic311x" }, + { .compatible = "ti,tlv320aic3100" }, + { .compatible = "ti,tlv320aic3110" }, + { .compatible = "ti,tlv320aic3120" }, + { .compatible = "ti,tlv320aic3111" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tlv320aic31xx_of_match); + +static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) +{ + struct device_node *np = aic31xx->dev->of_node; + unsigned int value = MICBIAS_2_0V; + int ret; + + of_property_read_u32(np, "ai31xx-micbias-vg", &value); + switch (value) { + case MICBIAS_2_0V: + case MICBIAS_2_5V: + case MICBIAS_AVDDV: + aic31xx->pdata.micbias_vg = value; + break; + default: + dev_err(aic31xx->dev, + "Bad ai31xx-micbias-vg value %d DT\n", + value); + aic31xx->pdata.micbias_vg = MICBIAS_2_0V; + } + + ret = of_get_named_gpio(np, "gpio-reset", 0); + if (ret > 0) + aic31xx->pdata.gpio_reset = ret; +} +#else /* CONFIG_OF */ +static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) +{ +} +#endif /* CONFIG_OF */ + +void aic31xx_device_init(struct aic31xx_priv *aic31xx) +{ + int ret, i; + + dev_set_drvdata(aic31xx->dev, aic31xx); + + if (dev_get_platdata(aic31xx->dev)) + memcpy(&aic31xx->pdata, dev_get_platdata(aic31xx->dev), + sizeof(aic31xx->pdata)); + else if (aic31xx->dev->of_node) + aic31xx_pdata_from_of(aic31xx); + + if (aic31xx->pdata.gpio_reset) { + ret = devm_gpio_request_one(aic31xx->dev, + aic31xx->pdata.gpio_reset, + GPIOF_OUT_INIT_HIGH, + "aic31xx-reset-pin"); + if (ret < 0) { + dev_err(aic31xx->dev, "not able to acquire gpio\n"); + return; + } + } + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) + aic31xx->supplies[i].supply = aic31xx_supply_names[i]; + + ret = devm_regulator_bulk_get(aic31xx->dev, + ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + if (ret != 0) + dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret); + +} + +static int aic31xx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct aic31xx_priv *aic31xx; + int ret; + const struct regmap_config *regmap_config; + + dev_dbg(&i2c->dev, "## %s: %s codec_type = %d\n", __func__, + id->name, (int) id->driver_data); + + regmap_config = &aic31xx_i2c_regmap; + + aic31xx = devm_kzalloc(&i2c->dev, sizeof(*aic31xx), GFP_KERNEL); + if (aic31xx == NULL) + return -ENOMEM; + + aic31xx->regmap = devm_regmap_init_i2c(i2c, regmap_config); + + if (IS_ERR(aic31xx->regmap)) { + ret = PTR_ERR(aic31xx->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + aic31xx->dev = &i2c->dev; + + aic31xx->pdata.codec_type = id->driver_data; + + aic31xx_device_init(aic31xx); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, + aic31xx_dai_driver, + ARRAY_SIZE(aic31xx_dai_driver)); + + return ret; +} + +static int aic31xx_i2c_remove(struct i2c_client *i2c) +{ + struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev); + + kfree(aic31xx); + return 0; +} + +static const struct i2c_device_id aic31xx_i2c_id[] = { + { "tlv320aic310x", AIC3100 }, + { "tlv320aic311x", AIC3110 }, + { "tlv320aic3100", AIC3100 }, + { "tlv320aic3110", AIC3110 }, + { "tlv320aic3120", AIC3120 }, + { "tlv320aic3111", AIC3111 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); + +static struct i2c_driver aic31xx_i2c_driver = { + .driver = { + .name = "tlv320aic31xx-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tlv320aic31xx_of_match), + }, + .probe = aic31xx_i2c_probe, + .remove = (aic31xx_i2c_remove), + .id_table = aic31xx_i2c_id, +}; + +module_i2c_driver(aic31xx_i2c_driver); + +MODULE_DESCRIPTION("ASoC TLV320AIC3111 codec driver"); +MODULE_AUTHOR("Jyri Sarha"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h new file mode 100644 index 0000000..52ed57c --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -0,0 +1,258 @@ +/* + * ALSA SoC TLV320AIC31XX codec driver + * + * Copyright (C) 2013 Texas Instruments, Inc. + * + * This package 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 PACKAGE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + * + */ +#ifndef _TLV320AIC31XX_H +#define _TLV320AIC31XX_H + +#define AIC31XX_RATES SNDRV_PCM_RATE_8000_192000 + +#define AIC31XX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + + +#define AIC31XX_STEREO_CLASS_D_BIT 0x1 +#define AIC31XX_MINIDSP_BIT 0x2 + +enum aic31xx_type { + AIC3100 = 0, + AIC3110 = AIC31XX_STEREO_CLASS_D_BIT, + AIC3120 = AIC31XX_MINIDSP_BIT, + AIC3111 = (AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT), +}; + +struct aic31xx_pdata { + enum aic31xx_type codec_type; + unsigned int gpio_reset; + int micbias_vg; +}; + +/* Page Control Register */ +#define AIC31XX_PAGECTL 0x00 + +/* Page 0 Registers */ +/* Software reset register */ +#define AIC31XX_RESET 0x01 +/* OT FLAG register */ +#define AIC31XX_OT_FLAG 0x03 +/* Clock clock Gen muxing, Multiplexers*/ +#define AIC31XX_CLKMUX 0x04 +/* PLL P and R-VAL register */ +#define AIC31XX_PLLPR 0x05 +/* PLL J-VAL register */ +#define AIC31XX_PLLJ 0x06 +/* PLL D-VAL MSB register */ +#define AIC31XX_PLLDMSB 0x07 +/* PLL D-VAL LSB register */ +#define AIC31XX_PLLDLSB 0x08 +/* DAC NDAC_VAL register*/ +#define AIC31XX_NDAC 0x0B +/* DAC MDAC_VAL register */ +#define AIC31XX_MDAC 0x0C +/* DAC OSR setting register 1, MSB value */ +#define AIC31XX_DOSRMSB 0x0D +/* DAC OSR setting register 2, LSB value */ +#define AIC31XX_DOSRLSB 0x0E +#define AIC31XX_MINI_DSP_INPOL 0x10 +/* Clock setting register 8, PLL */ +#define AIC31XX_NADC 0x12 +/* Clock setting register 9, PLL */ +#define AIC31XX_MADC 0x13 +/* ADC Oversampling (AOSR) Register */ +#define AIC31XX_AOSR 0x14 +/* Clock setting register 9, Multiplexers */ +#define AIC31XX_CLKOUTMUX 0x19 +/* Clock setting register 10, CLOCKOUT M divider value */ +#define AIC31XX_CLKOUTMVAL 0x1A +/* Audio Interface Setting Register 1 */ +#define AIC31XX_IFACE1 0x1B +/* Audio Data Slot Offset Programming */ +#define AIC31XX_DATA_OFFSET 0x1C +/* Audio Interface Setting Register 2 */ +#define AIC31XX_IFACE2 0x1D +/* Clock setting register 11, BCLK N Divider */ +#define AIC31XX_BCLKN 0x1E +/* Audio Interface Setting Register 3, Secondary Audio Interface */ +#define AIC31XX_IFACESEC1 0x1F +/* Audio Interface Setting Register 4 */ +#define AIC31XX_IFACESEC2 0x20 +/* Audio Interface Setting Register 5 */ +#define AIC31XX_IFACESEC3 0x21 +/* I2C Bus Condition */ +#define AIC31XX_I2C 0x22 +/* ADC FLAG */ +#define AIC31XX_ADCFLAG 0x24 +/* DAC Flag Registers */ +#define AIC31XX_DACFLAG1 0x25 +#define AIC31XX_DACFLAG2 0x26 +/* Sticky Interrupt flag (overflow) */ +#define AIC31XX_OFFLAG 0x27 +/* Sticy DAC Interrupt flags */ +#define AIC31XX_INTRDACFLAG 0x2C +/* Sticy ADC Interrupt flags */ +#define AIC31XX_INTRADCFLAG 0x2D +/* DAC Interrupt flags 2 */ +#define AIC31XX_INTRDACFLAG2 0x2E +/* ADC Interrupt flags 2 */ +#define AIC31XX_INTRADCFLAG2 0x2F +/* INT1 interrupt control */ +#define AIC31XX_INT1CTRL 0x30 +/* INT2 interrupt control */ +#define AIC31XX_INT2CTRL 0x31 +/* GPIO1 control */ +#define AIC31XX_GPIO1 0x33 + +#define AIC31XX_DACPRB 0x3C +/* ADC Instruction Set Register */ +#define AIC31XX_ADCPRB 0x3D +/* DAC channel setup register */ +#define AIC31XX_DACSETUP 0x3F +/* DAC Mute and volume control register */ +#define AIC31XX_DACMUTE 0x40 +/* Left DAC channel digital volume control */ +#define AIC31XX_LDACVOL 0x41 +/* Right DAC channel digital volume control */ +#define AIC31XX_RDACVOL 0x42 +/* Headset detection */ +#define AIC31XX_HSDETECT 0x43 +/* ADC Digital Mic */ +#define AIC31XX_ADCSETUP 0x51 +/* ADC Digital Volume Control Fine Adjust */ +#define AIC31XX_ADCFGA 0x52 +/* ADC Digital Volume Control Coarse Adjust */ +#define AIC31XX_ADCVOL 0x53 + + +/* Page 1 Registers */ +/* Headphone drivers */ +#define AIC31XX_HPDRIVER 0x9F +/* Class-D Speakear Amplifier */ +#define AIC31XX_SPKAMP 0xA0 +/* HP Output Drivers POP Removal Settings */ +#define AIC31XX_HPPOP 0xA1 +/* Output Driver PGA Ramp-Down Period Control */ +#define AIC31XX_SPPGARAMP 0xA2 +/* DAC_L and DAC_R Output Mixer Routing */ +#define AIC31XX_DACMIXERROUTE 0xA3 +/* Left Analog Vol to HPL */ +#define AIC31XX_LANALOGHPL 0xA4 +/* Right Analog Vol to HPR */ +#define AIC31XX_RANALOGHPR 0xA5 +/* Left Analog Vol to SPL */ +#define AIC31XX_LANALOGSPL 0xA6 +/* Right Analog Vol to SPR */ +#define AIC31XX_RANALOGSPR 0xA7 +/* HPL Driver */ +#define AIC31XX_HPLGAIN 0xA8 +/* HPR Driver */ +#define AIC31XX_HPRGAIN 0xA9 +/* SPL Driver */ +#define AIC31XX_SPLGAIN 0xAA +/* SPR Driver */ +#define AIC31XX_SPRGAIN 0xAB +/* HP Driver Control */ +#define AIC31XX_HPCONTROL 0xAC +/* MIC Bias Control */ +#define AIC31XX_MICBIAS 0xAE +/* MIC PGA*/ +#define AIC31XX_MICPGA 0xAF +/* Delta-Sigma Mono ADC Channel Fine-Gain Input Selection for P-Terminal */ +#define AIC31XX_MICPGAPI 0xB0 +/* ADC Input Selection for M-Terminal */ +#define AIC31XX_MICPGAMI 0xB1 +/* Input CM Settings */ +#define AIC31XX_MICPGACM 0xB2 + +/* Bits, masks and shifts */ + +/* AIC31XX_CLKMUX */ +#define AIC31XX_PLL_CLKIN_MASK 0x0c +#define AIC31XX_PLL_CLKIN_SHIFT 2 +#define AIC31XX_PLL_CLKIN_MCLK 0 +#define AIC31XX_CODEC_CLKIN_MASK 0x03 +#define AIC31XX_CODEC_CLKIN_SHIFT 0 +#define AIC31XX_CODEC_CLKIN_PLL 3 +#define AIC31XX_CODEC_CLKIN_BCLK 1 + +/* AIC31XX_PLLPR, AIC31XX_NDAC, AIC31XX_MDAC, AIC31XX_NADC, AIC31XX_MADC, + AIC31XX_BCLKN */ +#define AIC31XX_PLL_MASK 0x7f +#define AIC31XX_PM_MASK 0x80 + +/* AIC31XX_IFACE1 */ +#define AIC31XX_WORD_LEN_16BITS 0x00 +#define AIC31XX_WORD_LEN_20BITS 0x01 +#define AIC31XX_WORD_LEN_24BITS 0x02 +#define AIC31XX_WORD_LEN_32BITS 0x03 +#define AIC31XX_IFACE1_DATALEN_MASK 0x30 +#define AIC31XX_IFACE1_DATALEN_SHIFT (4) +#define AIC31XX_IFACE1_DATATYPE_MASK 0xC0 +#define AIC31XX_IFACE1_DATATYPE_SHIFT (6) +#define AIC31XX_I2S_MODE 0x00 +#define AIC31XX_DSP_MODE 0x01 +#define AIC31XX_RIGHT_JUSTIFIED_MODE 0x02 +#define AIC31XX_LEFT_JUSTIFIED_MODE 0x03 +#define AIC31XX_IFACE1_MASTER_MASK 0x0C +#define AIC31XX_BCLK_MASTER 0x08 +#define AIC31XX_WCLK_MASTER 0x04 + +/* AIC31XX_DATA_OFFSET */ +#define AIC31XX_DATA_OFFSET_MASK 0xFF + +/* AIC31XX_IFACE2 */ +#define AIC31XX_BCLKINV_MASK 0x08 +#define AIC31XX_BDIVCLK_MASK 0x03 +#define AIC31XX_DAC2BCLK 0x00 +#define AIC31XX_DACMOD2BCLK 0x01 +#define AIC31XX_ADC2BCLK 0x02 +#define AIC31XX_ADCMOD2BCLK 0x03 + +/* AIC31XX_ADCFLAG */ +#define AIC31XX_ADCPWRSTATUS_MASK 0x40 + +/* AIC31XX_DACFLAG1 */ +#define AIC31XX_LDACPWRSTATUS_MASK 0x80 +#define AIC31XX_RDACPWRSTATUS_MASK 0x08 +#define AIC31XX_HPLDRVPWRSTATUS_MASK 0x20 +#define AIC31XX_HPRDRVPWRSTATUS_MASK 0x02 +#define AIC31XX_SPLDRVPWRSTATUS_MASK 0x10 +#define AIC31XX_SPRDRVPWRSTATUS_MASK 0x01 + +/* AIC31XX_INTRDACFLAG */ +#define AIC31XX_HPSCDETECT_MASK 0x80 +#define AIC31XX_BUTTONPRESS_MASK 0x20 +#define AIC31XX_HSPLUG_MASK 0x10 +#define AIC31XX_LDRCTHRES_MASK 0x08 +#define AIC31XX_RDRCTHRES_MASK 0x04 +#define AIC31XX_DACSINT_MASK 0x02 +#define AIC31XX_DACAINT_MASK 0x01 + +/* AIC31XX_INT1CTRL */ +#define AIC31XX_HSPLUGDET_MASK 0x80 +#define AIC31XX_BUTTONPRESSDET_MASK 0x40 +#define AIC31XX_DRCTHRES_MASK 0x20 +#define AIC31XX_AGCNOISE_MASK 0x10 +#define AIC31XX_OC_MASK 0x08 +#define AIC31XX_ENGINE_MASK 0x04 + +/* AIC31XX_DACSETUP */ +#define AIC31XX_SOFTSTEP_MASK 0x03 + +/* AIC31XX_DACMUTE */ +#define AIC31XX_DACMUTE_MASK 0x0C + +/* AIC31XX_MICBIAS */ +#define AIC31XX_MICBIAS_MASK 0x03 +#define AIC31XX_MICBIAS_SHIFT 0 + +#endif /* _TLV320AIC31XX_H */ -- 1.7.9.5

2 2

[alsa-devel] zoom r16 quirks
by James Miller 13 Mar '14

13 Mar '14

Hi, I am trying to get the zoom R16 working with alsa for full duplex operation. I have been able to get the 8 channels of capture working after following jmancine's advice and using the following quirk on ifnum 4: { .ifnum = 4, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_LE, .channels = 8, .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 44100, .rate_max = 96000, .nr_rates = 4, .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 } } }, However, what I discovered was that the ifnum 4 in this case is related to the udev matching rules for the device number, so when I plugged in the zoom, it was being assigned to device 4, which was then matched to the ifnum 4. If I unplugged and replugged the device, it was then designated as device 5 in the udev tree, and wasn't working. I read in quirks-table.h the following: (To work with * hotplugging, bDeviceClass must be set to USB_CLASS_PER_INTERFACE.) so I changed the matching rule macro to : USB_DEVICE_INTERFACE_CLASS(0x1686, 0x00dd, USB_CLASS_AUDIO) to attempt to match the device during udev assignment (in case the device class wasn't returned from the device properly) and then set .binterfaceclass = USB_CLASS_PER_INTERFACE below that before the quirk data. I think that I am heading in the correct direction, although I was wondering if you could confirm the following: Am I correct in assuming that the .ifnum in the quirk is related to udev device matching rules? When I write zoom-quirks.h in the way that I am currently (see below), I get a incorrect maxpacket assignment for interface 3 (midi) and I traced this to: dev_warn(ddev, "config %d interface %d altsetting %d " "bulk endpoint 0x%X has invalid maxpacket %d\n", in config.c line 259. The above bit of code is selected in conditions where the device is usb high speed but I have kern.log reporting that the device is class 1.1. Its a 24 bit chip, and jmancine reports that the device is being incorrectly assigned a 32bit le format for playback when it should be 24bit le, so perhaps this is because of a protocol mismatch? When I have the device successfully recognised the next time, I will check in format.c that it is being assigned the correct bit format, but I am uncertain as to how to confirm what bprotocol to use. Any ideas? Here is the zoom-quirks.h that I am currently using. { /* ZOOM R16 in USB 2.0 mode */ USB_DEVICE_INTERFACE_CLASS(0x1686, 0x00dd, USB_CLASS_AUDIO), .bDeviceClass = USB_CLASS_PER_INTERFACE, .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 1, /*PLAYBACK*/ .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_LE, .channels = 2, .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .endpoint = 0x03, .ep_attr = (USB_ENDPOINT_XFER_ISOC && USB_ENDPOINT_SYNC_ADAPTIVE && SND_USB_ENDPOINT_TYPE_DATA), .rate_min = 44100, .rate_max = 96000, .nr_rates = 4, .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 } } }, { .ifnum = 2, /*CAPTURE*/ .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_LE, .channels = 8, .iface = 2, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .endpoint = 0x84, .ep_attr = (USB_ENDPOINT_XFER_ISOC && USB_ENDPOINT_SYNC_SYNC && USB_ENDPOINT_USAGE_DATA), .rate_min = 44100, .rate_max = 96000, .nr_rates = 4, .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 } } }, { .ifnum = 3, .type = QUIRK_MIDI_FIXED_ENDPOINT, }, { .ifnum = -1, }, } } }, Thanks for any help!! -- James Stewart Miller Bsc(hons) Psych.

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix pop noise on Dell Inspiron 7537
by Takashi Iwai 13 Mar '14

13 Mar '14

Dell Inspiron 7537 shows loud pop noises at booting or shutdown. They can be reduced by avoiding the shutup call that is used for this codec. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=71861 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b50114176736..c31254a20331 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3868,6 +3868,7 @@ enum { ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, + ALC283_FIXUP_DELL1_SHUTUP, ALC269_FIXUP_HEADSET_MODE, ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC269_FIXUP_ASUS_X101_FUNC, @@ -4087,6 +4088,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC283_FIXUP_DELL1_SHUTUP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_no_shutup, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + }, [ALC269_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode, @@ -4322,7 +4329,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x05f9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f9, "Dell Inspiron 7537", ALC283_FIXUP_DELL1_SHUTUP), SND_PCI_QUIRK(0x1028, 0x05fb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.9.0

1 0

[alsa-devel] [GIT PULL] ASoC updates for v3.15
by Mark Brown 13 Mar '14

13 Mar '14

The following changes since commit fa389e220254c69ffae0d403eac4146171062d08: Linux 3.14-rc6 (2014-03-09 19:41:57 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.15 for you to fetch changes up to deeed33850c8a376addabbf971df433b2a1ba74c: Merge remote-tracking branches 'asoc/topic/wm8996', 'asoc/topic/wm9081' and 'asoc/topic/wm9705' into asoc-next (2014-03-12 23:05:31 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.15 Quite a busy release for ASoC this time, more on janitorial work than exciting new features but welcome nontheless: - Lots of cleanups from Takashi for enumerations; the original API for these was error prone so he's refactored lots of code to use more modern APIs which avoid issues. - Elimination of the ASoC level wrappers for I2C and SPI moving us closer to converting to regmap completely and avoiding some randconfig hassle. - Provide both manually and transparently locked DAPM APIs rather than a mix of the two fixing some concurrency issues. - Start converting CODEC drivers to use separate bus interface drivers rather than having them all in one file helping avoid dependency issues. - DPCM support for Intel Haswell and Bay Trail platforms. - Lots of work on improvements for simple-card, DaVinci and the Renesas rcar drivers. - New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the CSR SiRF SoC. ---------------------------------------------------------------- Aaro Koskinen (1): ASoC: n810: fix init with DT boot Adam Thomson (1): ASoC: da9055: Add DT support for CODEC Alexander Shiyan (1): ASoC: cirrus: Remove excess dependencies on SND_SOC Bo Shen (2): ASoC: atmel_ssc_dai: make option to choose clock Binding: atmel-ssc: add option to choose clock Charles Keepax (33): ASoC: wm5102: Improve EQ coefficient controls ASoC: wm5110: Improve EQ coefficient controls ASoC: wm8997: Improve EQ coefficient controls ASoC: wm_adsp: Add debug print to note that the DSP has shutdown ASoC: wm5102: Correct typo in EQ coefficient sizes ASoC: wm5110: Correct type in EQ coefficient sizes ASoC: wm8997: Correct typo in EQ coefficient sizes ASoC: dapm: Add helpers to lock/unlock DAPM mutex ASoC: dapm: Add unlocked version of snd_soc_dapm_sync ASoC: adav80x: Update locking around use of DAPM pin API ASoC: wm5100: Update locking around use of DAPM pin API ASoC: wm8962: Update locking around use of DAPM pin API ASoC: wm8994: Update locking around use of DAPM pin API ASoC: wm8996: Update locking around use of DAPM pin API ASoC: mfld_machine: Update locking around use of DAPM pin API ASoC: ams-delta: Update locking around use of DAPM pin API ASoC: omap: n810: Update locking around use of DAPM pin API ASoC: omap: rx51: Update locking around use of DAPM pin API ASoC: pxa: corgi: Update locking around use of DAPM pin API ASoC: pxa: magician: Update locking around use of DAPM pin API ASoC: pxa: spitz: Update locking around use of DAPM pin API ASoC: pxa: tosa: Update locking around use of DAPM pin API ASoC: wm_adsp: Split firmware load into smaller chunks ASoC: wm_adsp: Correct type specifier in printf ASoC: arizona: An OUTDIV of 1 is not valid, avoid this ASoC: arizona: Add defines for FLL configuration constants ASoC: arizona: Move set of OUTDIV in to arizona_apply_fll ASoC: arizona: Move calculation of FLL configuration ASoC: arizona: Don't pass Fout into arizona_calc_fll ASoC: arizona: Calculate OUTDIV first ASoC: arizona: Calculate FLL gain last mfd: arizona: Add support for new fratio encoding ASoC: arizona: Support new fratio encoding on the wm5110 rev D Christian Engelmayer (3): ASoC: wm8991: Remove unused pointer in wm8991_probe() ASoC: wm8995: Remove unused pointer in hp_supply_event() ASoC: wm8993: Remove unused pointer in wm8993_remove() Daniel Mack (1): ASoC: cs4271: claim reset GPIO in bus probe function Denis Carikli (1): ASoC: eukrea-tlv320: Add DT support. Fabio Estevam (4): ASoC: wm8737: Use IS_ENABLED() macro ASoC: wm8983: Use IS_ENABLED() macro ASoC: fsl_esai: Check the return value from clk_prepare_enable() ASoC: fsl: imx-pcm-fiq: Remove unneeded 'out' label Jean-Francois Moine (8): ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code Jyri Sarha (3): ASoC: davinci-evm: Add named clock reference to DT bindings ASoC: davinci-mcasp: Set BCLK divider if McASP is BCLK master ASoC: tlv320aic32x4: Sort Makefile in alphabetic order Kuninori Morimoto (58): ASoC: rsnd: tidyup register naming of BUSIF_MODE ASoC: rsnd: cleanup debug information method ASoC: rsnd: remove meaningless rsnd_ssi_non() ASoC: rsnd: control SCU ops in probe timing ASoC: rsnd: add rsnd_scu_init(), and separate init/start ASoC: rsnd: remove meaningless function parameter ASoC: rsnd: merge rsnd_scu_start/stop() and rsnd_scu_transfer_start/stop() ASoC: rsnd: rsnd_dai_is_clk_master() can be shared ASoC: rsnd: remove pin sync option ASoC: rsnd: SSI_MODE0/1 settings goes to scu.c ASoC: rsnd: remove ssiu from ssi.c ASoC: rsnd: rename rsnd_scu_convert_rate_ctrl() ASoC: rsnd: explain SRC bypass mode settings in comment ASoC: rsnd: remove duplicate *priv from rsnd_dma ASoC: rsnd: non 0 is error on probe ASoC: rsnd: fixup Gen2 module naming ASoC: rsnd: don't use schedule_work() when rsnd_dma_start() ASoC: rsnd: SCU should be called before SSI ASoC: rsnd: clarify scu.c area ASoC: rsnd: Merge macros in scu.c ASoC: rsnd: rsnd_scu_hpbif_is_enable() become macro ASoC: rsnd: merge SRC clock timing/setting ASoC: rsnd: add rsnd_ssi_is_play() ASoC: rsnd: extracts Gen1/Gen2 common parts ASoC: rsnd: remove SSI dependent DMAEngine callback ASoC: rsnd: rsnd_ssi_probe() goes forwarder than rsnd_scu_probe() ASoC: rsnd: add Gen2 SRC and DMAEngine support ASoC: rsnd: use device dependency clock ASoC: rsnd: tidyup original for_each_rsnd_xxx macro ASoC: rsnd: set DIV_EN register on rsnd_adg_set_convert_clk_gen2() ASoC: rsnd: print error if there is SRC settings mismatch ASoC: simple-card: cpu_dai_name creates confusion when DT case ASoC: rsnd: move priv member settings to upper side ASoC: rsnd: move rsnd_mod_call() macro ASoC: rsnd: remove verbose function parameter ASoC: rsnd: remove verbose debug message from scu/ssi ASoC: rsnd: unify rdai naming ASoC: rsnd: tidyup RSND_SSI_xxx flags ASoC: rsnd: run rsnd_path_init() when probe() timing ASoC: rsnd: use mod array instead of list on rdai ASoC: rsnd: get ssi/scu from rsnd_dai_stream ASoC: rsnd: use devm_clk_get() instead of clk_get() ASoC: rsnd: use function pointer for each probe ASoC: rsnd: remove unused SSI_CONTROL ASoC: rsnd: modify rsnd_adg_ssi_ws_timing_gen2() parameter ASoC: rsnd: share reg_field and reduce memory ASoC: rsnd: add struct rsnd_dai_platform_info ASoC: rsnd: Get correct SCU ID ASoC: rsnd: add rsnd_scu_enable_ssi_irq() ASoC: rsnd: call rsnd_scu_ssi_mode_init() from SSI ASoC: rsnd: add probe/remove callback on rsnd_mod_ops ASoC: rsnd: use mod probe method on SCU ASoC: rsnd: use mod probe method on SSI ASoC: rsnd: nothing to do on rsnd_dai_remove() ASoC: rsnd: remove all rsnd_xxx_remove() ASoC: rsnd: rename scu to src ASoC: simple-card: tidyup cpu/codec dai_fmt settings for non-DT ASoC: simple-card: card name can be option Lars-Peter Clausen (58): ASoC: h1940_uda1380: Don't modify runtime->hw ASoC: rx1950_uda1380: Don't modify runtime->hw ASoC: cs4271: Remove outdated comment ASoC: cs42l73: Don't mix SNDRV_PCM_RATE_KNOT with specific rates ASoC: cs42l73: Constify rate constraints ASoC: ssm2602: Constify rate constraints ASoC: twl6040: Constify rate constraints ASoC: wm8741: Constify rate constraints ASoC: wm8988: Constify rate constraints ASoC: dapm: Consistently use unsigned int for register values ASoC: ad193x: Split SPI and I2C code into different modules ASoC: ssm2602: Split SPI and I2C code into different modules ASoC: adav80x: Use devm_kzalloc() ASoC: adav80x: Split SPI and I2C code into different modules ASoC: Add ADAU1977 CODEC driver ASoC: pxa: Remove superfluous locking ASoC: da732x: Use da732x->regmap instead of codec->control_data ASoC: wm8996: Replace codec->control_data with wm8996->regmap ASoC: wm8962: Replace codec->control_data with wm8962->regmap ASoC: Remove unused 'reg2' field from soc_enum struct ASoC: Consolidate enum and value enum controls ASoC: dapm: Consolidate MUXs and value MUXs ASoC: Add macros for defining virtual enums ASoC: adau1373: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: max98090: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: mc13783: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8994: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8995: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: dapm: Consolidate MUXs and virtual MUXs ASoC: dapm: Break dapm_set_path_status() appart ASoC: neo1973_wm8753: Post gta01 support removal cleanup ASoC: neo1973_wm8753: Convert to table based setup ASoC: pxa: Pass correct DAPM context to {corgi,poodle,spitz}_ext_control ASoC: pxa: e740_wm9705: Convert to table based DAPM setup ASoC: pxa: e750_wm9705: Convert to table based DAPM setup ASoC: pxa: e800_wm9712: Convert to table based DAPM setup ASoC: pxa: mioa701_wm9713: Convert to table based DAPM setup ASoC: pxa: zylonite: Convert to table based DAPM setup ASoC: wm1133-ev1: Convert to table based DAPM setup ASoC: wm8753: Remove superfluous 'codec->cache_sync = 1' ASoC: wm{5102, 5110, 8997}: Replace codec->control_data with arizona->regmap ASoC: ak4104: Remove superfluous codec->control_data initialization ASoC: Handle ignore_pmdown_time for CODEC to CODEC links ASoC: Add helper functions for PCM runtime 'active' management ASoC: Fix active count tracking for CODEC to CODEC links ASoC: Add helper function to check whether a CODEC is active ASoC: Add component pointer to the DAI struct ASoC: Move active count from CODEC to component ASoC: Move ignore_pmdown_time from CODEC to component ASoC: sam9g20_wm8731: Convert to table based DAPM setup ASoC: si476x: Fix IO setup ASoC: 88pm860: Fix IO setup ASoC: Consolidate snd_soc_register_dai() and snd_soc_register_dais() ASoC: Pass CODEC to snd_soc_register_dais() ASoC: Add a per component dai list ASoC: migor: Convert to table based DAPM setup ASoC: spear: spdif_out: Fix mute control ASoC: Fix use after free Mark Brown (60): Merge branch 'fix/davinci' of git://git.kernel.org/.../broonie/sound into HEAD Merge branch 'topic/sign' of git://git.kernel.org/.../broonie/sound into asoc-tlv320aic32x4 ASoC: wm8962: Hold a runtime PM reference while handling interrupts ASoC: wm8962: Check if we runtime resume the device when starting FLL ASoC: wm8962: Clean up error handling for failed FLL start ASoC: wm8962: Reinitialise the IRQ completion rather than just trying it ASoC: ak4554: Add to SND_SOC_ALL_CODECS ASoC: pcm1681: Convert to params_width() ASoC: pcm1792a: Convert to params_width() ASoC: codecs: Make OF supported CODECs visible in Kconfig ASoC: pcm512x: Add PCM512x driver ASoC: pcm512x: More constification ASoC: codecs: Put the CODEC drivers in a menu ASoC: wm8770: Depend on SPI only ASoC: pcm512x: Implement paging support ASoC: pcm512x: Implement analogue volume control Merge branch 'topic/of' of git://git.kernel.org/.../broonie/sound into asoc-simple ASoC: tobermory: Stop the FLL if we fail to switch SYSCLK to it ASoC: cs42l51: Don't log if we fail to allocate memory ASoC: cs42l51: Convert to direct regmap API usage ASoC: ak4671: Convert to table based control init ASoC: ak4671: Convert to direct regmap API usage ASoC: pcm512x: Add regmap select Merge remote-tracking branch 'asoc/topic/adav80x' into asoc-adau1977 ALSA: Export snd_pcm_constraint_mask64() Merge branch 'fix/dapm' of git://git.kernel.org/.../broonie/sound into asoc-dapm ASoC: alc5623: Convert to direct regmap API usage Merge branch 'topic/of' of git://git.kernel.org/.../broonie/sound into asoc-simple ASoC: da732x: Remove leftover cache size setting Merge branches 'topic/dapm', 'topic/sign', 'topic/adau1373', 'topic/max98090', 'topic/mc13783', 'fix/wm8994' and 'topic/wm8995' of git://git.kernel.org/.../broonie/sound into asoc-enum ASoC: fsl: Don't select FIQ from Eukrea Merge branch 'topic/pcm' of git://git.kernel.org/.../broonie/sound into asoc-enum Merge tag 'parse-val' of git://git.kernel.org/.../broonie/regmap into asoc-core Merge branch 'topic/tlv320aic23' of git://git.kernel.org/.../broonie/sound into asoc-samsung Merge branch 'topic/tlv320aic23' of git://git.kernel.org/.../broonie/sound into asoc-fsl ASoC: pcm512x: Split out bus drivers ASoC: pcm512x: Fix duplicate const warning Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Merge remote-tracking branches 'asoc/fix/88pm860', 'asoc/fix/omap' and 'asoc/fix/si476x' into asoc-linus Merge branch 'topic/pcm' of git://git.kernel.org/.../broonie/sound into asoc-core ASoC: dapm: Staticise dapm_mark_dirty() ASoC: dapm: Sprinkle lockdep asserts through the code ASoC: tlv320aic23: Remove spurious bits per word setting Merge tag 'asoc-v3.14-rc6' into asoc-linus Merge remote-tracking branch 'asoc/fix/spear' into asoc-linus Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/enum' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1681' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1792a' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-next Merge remote-tracking branches 'asoc/topic/88pm860x', 'asoc/topic/ad1836', 'asoc/topic/ad193x', 'asoc/topic/adau1977', 'asoc/topic/ak4104', 'asoc/topic/ak4641', 'asoc/topic/ak4671' and 'asoc/topic/alc5623' into asoc-next Merge remote-tracking branches 'asoc/topic/arizona', 'asoc/topic/atmel', 'asoc/topic/cirrus', 'asoc/topic/cs4271', 'asoc/topic/cs42l51', 'asoc/topic/cs42l52', 'asoc/topic/cs42l73' and 'asoc/topic/da7210' into asoc-next Merge remote-tracking branches 'asoc/topic/da7213', 'asoc/topic/da732x', 'asoc/topic/da9055', 'asoc/topic/davinci', 'asoc/topic/fsl', 'asoc/topic/fsl-esai', 'asoc/topic/fsl-sai' and 'asoc/topic/fsl-spdif' into asoc-next Merge remote-tracking branches 'asoc/topic/ml26124', 'asoc/topic/of', 'asoc/topic/omap', 'asoc/topic/pxa' and 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/samsung', 'asoc/topic/sgtl5000', 'asoc/topic/sh', 'asoc/topic/simple', 'asoc/topic/sirf', 'asoc/topic/sn95031', 'asoc/topic/ssm2602' and 'asoc/topic/stac9766' into asoc-next Merge remote-tracking branches 'asoc/topic/tlv320aic23', 'asoc/topic/tlv320aic26', 'asoc/topic/tlv320aic32x4', 'asoc/topic/tlv320aic3x', 'asoc/topic/tlv320dac33', 'asoc/topic/twl4030', 'asoc/topic/twl6040', 'asoc/topic/uda1380', 'asoc/topic/wl1273' and 'asoc/topic/wm2200' into asoc-next Merge remote-tracking branches 'asoc/topic/wm5100', 'asoc/topic/wm8523', 'asoc/topic/wm8731', 'asoc/topic/wm8737', 'asoc/topic/wm8741', 'asoc/topic/wm8753', 'asoc/topic/wm8903', 'asoc/topic/wm8904', 'asoc/topic/wm8940' and 'asoc/topic/wm8955' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8961', 'asoc/topic/wm8962', 'asoc/topic/wm8974', 'asoc/topic/wm8983', 'asoc/topic/wm8988', 'asoc/topic/wm8990', 'asoc/topic/wm8991', 'asoc/topic/wm8993' and 'asoc/topic/wm8994' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8996', 'asoc/topic/wm9081' and 'asoc/topic/wm9705' into asoc-next Markus Pargmann (9): ASoC: core: Add signed register volume control logic ASoC: core: Add SOC_DOUBLE_R_S_TLV ASoC: tlv320aic32x4: Use gpio_is_valid ASoC: tlv320aic32x4: Use signed int mixer controls ASoC: tlv320aic32x4: DT support ASoC: tlv320aic32x4: Support for master clock ASoC: tlv320aic32x4: Support for regulators ASoC: tlv320aic32x4: Rearrange clock tree shutdown ASoC: tlv320aic3x: Remove tlv320aic32x4 from compatibles of tlv320aic3x Max Filippov (3): ASoC: tlv320aic23: add support for SPI control mode ASoC: update Kconfig of AIC23 users to select I2C variant ASoC: tlv320aic23: add missing EXPORT_SYMBOLs Nenghua Cao (3): ASoC: core: fix coccinelle warnings regmap: add regmap_parse_val api ASoC: core: use regmap's parse_val to do endian translation Patrick Lai (1): ASoC: pcm: free path list before exiting from error conditions Paul Bolle (1): ASoC: smdk_wm8994: Fix typo "wm8894" Peter Ujfalusi (6): ASoC: davinci-mcasp: Return value handling cleanup for mcasp_common_hw_param() ASoC: davinci-mcasp: Move pm callbacks from platform device to soc_dai_driver ASoC: davinci-mcasp: Code cleanup in davinci_mcasp_hw_params() ASoC: davinci-mcasp: Rename platform data struct ASoC: davinci-mcasp: Remove long lines from suspend/resume callbacks ASoC: davinci-mcasp: Rename the pointer to davinci dma_params as dma_params Rongjun Ying (3): ASoC: sirf: Add SiRF internal audio codec driver ASoC: sirf: Add SiRF audio port driver is used by SiRF internal audio codec ASoC: sirf: Add SiRF audio card Sachin Kamat (6): ASoC: samsung: h1940_uda1380: Remove unused variable ASoC: samsung: h1940_uda1380: Fix build warning ASoC: samsung: rx1950_uda1380: Remove unused variable ASoC: cs42l51: Remove unused variable ASoC: samsung: Remove invalid file reference ASoC: s3c24xx: Remove invalid file reference Takashi Iwai (72): ASoC: da732x: Remove superfluous DA732X_SOC_ENUM_DOUBLE_R() ASoC: ssm2602: Omit superfluous elements in input select array ASoC: wm8990: Fix the wrong number of enum items ASoC: wm8991: Fix the wrong number of enum items ASoC: ad193x: Use SOC_ENUM_SINGLE_DECL() ASoC: Rename soc_enum.max field with items ASoC: twl4030: Clean up duplicated code ALSA: da732x: Remove superfluous DA732X_SOC_ENUM_DOUBLE_R() ASoC: Add const to SOC_ENUM_*_DECL() macros ASoC: adau1373: Remove superfluous const ASoC: lm49453: Remove superfluous const ASoC: mc13783: Remove superfluous const ASoC: rt5631: Remove superfluous const ASoC: rt5640: Remove superfluous const ASoC: ssm2518: Remove superfluous const ASoC: sta529: Remove superfluous const ASoC: wm8804: Remove superfluous const ASoC: wm8978: Remove superfluous const ASoC: wm8983: Remove superfluous const ASoC: wm8985: Remove superfluous const ASoC: wm8995: Remove superfluous const ASoC: arizona: Fix wrong number of items in enum ctls ASoC: adau1373: Use SOC_ENUM_SINGLE_DECL() ASoC: alc5623: Use SOC_ENUM_SINGLE_DECL() ASoC: cs42l52: Use SOC_ENUM_SINGLE_DECL() ASoC: da7210: Use SOC_ENUM_SINGLE_DECL() ASoC: da7213: Use SOC_ENUM_SINGLE_DECL() ASoC: da9055: Use SOC_ENUM_SINGLE_DECL() ASoC: max98090: Use SOC_ENUM_SINGLE_DECL() ASoC: mc13783: Use SOC_ENUM_SINGLE_DECL() ASoC: sgtl5000: Use SOC_ENUM_SINGLE_DECL() ASoC: sn95031: Use SOC_ENUM_SINGLE_DECL() ASoC: stac9766: Use SOC_ENUM_{SINGLE|DOUBLE}_DECL() ASoC: tlv320aic26: Use SOC_ENUM_SINGLE_DECL() ASoC: twl4030: Use SOC_ENUM_SINGLE_DECL() ASoC: twl6040: Use SOC_ENUM_SINGLE_DECL() ASoC: uda1380: Use SOC_ENUM_SINGLE_DECL() ASoC: wl1273: Use SOC_ENUM_SINGLE_EXT_DECL() ASoC: wm2200: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8523: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8731: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8737: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8903: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8904: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8940: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8961: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8962: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8974: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8993: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8995: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8996: Use SOC_ENUM_SINGLE_DECL() ASoC: wm_hubs: Use SOC_ENUM_SINGLE_DECL() ASoC: omap: Use SOC_ENUM_SINGLE_EXT_DECL() ASoC: 88pm860x: Use SOC_ENUM_SINGLE_DECL() ASoC: ak4641: Use SOC_ENUM_SINGLE_DECL() ASoC: ak4671: Use SOC_ENUM_SINGLE_DECL() ASoC: alc5632: Use SOC_ENUM_SINGLE_DECL() ASoC: wm2200: Use SOC_ENUM_SINGLE_*_DECL() ASoC: da732x: Use SOC_ENUM_SINGLE_DECL() ASoC: ml26124: Use SOC_ENUM_SINGLE_DECL() ASoC: tlv320aic23: Use SOC_ENUM_SINGLE_DECL() ASoC: wm5100: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8955: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8988: Use SOC_ENUM_SINGLE_DECL() ASoC: wm9081: Use SOC_ENUM_SINGLE_DECL() ASoC: wm9705: Use SOC_ENUM_SINGLE_DECL() ASoC: ad1836: Use SOC_ENUM_SINGLE_DECL() ASoC: cs42l73: Use SOC_ENUM_SINGLE_DECL() ASoC: tlv320dac33: Use SOC_ENUM_SINGLE_*_DECL() ASoC: wm5102: Use ARRAY_SIZE() for SOC_VALUE_ENUM_SINGLE() ASoC: wm8997: Use ARRAY_SIZE() for SOC_VALUE_ENUM_SINGLE() ASoC: cs4271: Fix build error without CONFIG_SPI_MASTER Thomas Petazzoni (1): ASoC: cs42l51: add Device Tree binding to cs42l51 Xiang Xiao (2): ASoC: dapm: Power off all widgets in the snd_soc_dapm_shutdown ASoC: dapm: Reorder the bias update sequence Xiubo Li (19): ASoC: core: set_tdm_slot() will return -ENOTSUPP if no operation provided ASoC: fsl: use snd_soc_dai_init_dma_data() ASoC: simple-card: fix __asoc_simple_card_dai_init ASoC: simple-card: simplify the daifmt code ASoC: simple-card: Add snd_card's name parsing from DT node support ASoC: fsl-sai: convert to use regmap API for Freeacale SAI ASoC: add snd_soc_of_parse_audio_simple_widgets for DT ASoC: binding: add widgets.txt ASoC: simple-card: add off-codec widgets supports. ASoC: simple-card: for new properties documenting and usage ASoC: fsl: make sure the regmap-mmio is actually enabled ASoC: fsl-spdif: big-endian support ASoC: fsl-esai: big-endian support ASoC: binding: add tdm-slot.txt ASoC: core: add TDM slot parsing from DT supports ASoC: simple-card: add slot information parsing supports ASoC: fsl-sai: fix Freescale SAI DAI format setting. ASoC: fsl-sai: Add SND_SOC_DAIFMT_DSP_A/B support. ASoC: cache: Do the codec->reg_cache zero pionter check .../devicetree/bindings/i2c/trivial-devices.txt | 1 + .../devicetree/bindings/misc/atmel-ssc.txt | 8 + Documentation/devicetree/bindings/sound/da9055.txt | 22 + .../bindings/sound/davinci-evm-audio.txt | 9 +- .../devicetree/bindings/sound/eukrea-tlv320.txt | 21 + .../devicetree/bindings/sound/fsl,esai.txt | 5 + .../devicetree/bindings/sound/fsl,spdif.txt | 5 + .../devicetree/bindings/sound/pcm512x.txt | 30 + .../devicetree/bindings/sound/simple-card.txt | 17 +- .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 + .../devicetree/bindings/sound/sirf-audio-port.txt | 20 + .../devicetree/bindings/sound/sirf-audio.txt | 41 + .../devicetree/bindings/sound/tdm-slot.txt | 20 + .../devicetree/bindings/sound/tlv320aic32x4.txt | 30 + .../devicetree/bindings/sound/tlv320aic3x.txt | 1 - .../devicetree/bindings/sound/widgets.txt | 20 + drivers/base/regmap/regmap.c | 12 + drivers/misc/atmel-ssc.c | 6 + include/linux/atmel-ssc.h | 1 + include/linux/mfd/arizona/registers.h | 6 +- include/linux/platform_data/adau1977.h | 45 + include/linux/platform_data/asoc-s3c.h | 3 +- include/linux/platform_data/asoc-s3c24xx_simtec.h | 3 +- include/linux/platform_data/davinci_asp.h | 4 +- include/linux/regmap.h | 9 + include/sound/rcar_snd.h | 36 +- include/sound/simple_card.h | 6 +- include/sound/soc-dai.h | 3 + include/sound/soc-dapm.h | 40 +- include/sound/soc.h | 126 +- include/trace/events/hswadsp.h | 384 +++++ include/trace/events/intel-sst.h | 148 ++ sound/core/pcm_lib.c | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/Kconfig | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 13 +- sound/soc/atmel/sam9g20_wm8731.c | 20 +- sound/soc/blackfin/Kconfig | 9 +- sound/soc/cirrus/Kconfig | 4 +- sound/soc/codecs/88pm860x-codec.c | 119 +- sound/soc/codecs/Kconfig | 183 +- sound/soc/codecs/Makefile | 33 +- sound/soc/codecs/ad1836.c | 4 +- sound/soc/codecs/ad193x-i2c.c | 54 + sound/soc/codecs/ad193x-spi.c | 48 + sound/soc/codecs/ad193x.c | 144 +- sound/soc/codecs/ad193x.h | 7 + sound/soc/codecs/adau1373.c | 32 +- sound/soc/codecs/adau1977-i2c.c | 59 + sound/soc/codecs/adau1977-spi.c | 76 + sound/soc/codecs/adau1977.c | 1018 +++++++++++ sound/soc/codecs/adau1977.h | 37 + sound/soc/codecs/adav801.c | 53 + sound/soc/codecs/adav803.c | 50 + sound/soc/codecs/adav80x.c | 152 +- sound/soc/codecs/adav80x.h | 7 + sound/soc/codecs/ak4104.c | 2 - sound/soc/codecs/ak4641.c | 16 +- sound/soc/codecs/ak4671.c | 240 ++- sound/soc/codecs/ak4671.h | 2 - sound/soc/codecs/alc5623.c | 117 +- sound/soc/codecs/alc5632.c | 40 +- sound/soc/codecs/arizona.c | 325 ++-- sound/soc/codecs/cs4271.c | 63 +- sound/soc/codecs/cs42l51.c | 86 +- sound/soc/codecs/cs42l52.c | 92 +- sound/soc/codecs/cs42l73.c | 55 +- sound/soc/codecs/da7210.c | 20 +- sound/soc/codecs/da7213.c | 151 +- sound/soc/codecs/da732x.c | 167 +- sound/soc/codecs/da732x.h | 3 - sound/soc/codecs/da9055.c | 92 +- sound/soc/codecs/lm49453.c | 16 +- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98090.c | 172 +- sound/soc/codecs/max98095.c | 4 +- sound/soc/codecs/mc13783.c | 20 +- sound/soc/codecs/ml26124.c | 12 +- sound/soc/codecs/pcm1681.c | 15 +- sound/soc/codecs/pcm1792a.c | 33 +- sound/soc/codecs/pcm512x-i2c.c | 71 + sound/soc/codecs/pcm512x-spi.c | 69 + sound/soc/codecs/pcm512x.c | 589 +++++++ sound/soc/codecs/pcm512x.h | 171 ++ sound/soc/codecs/rt5631.c | 75 +- sound/soc/codecs/rt5640.c | 75 +- sound/soc/codecs/sgtl5000.c | 10 +- sound/soc/codecs/si476x.c | 2 +- sound/soc/codecs/sirf-audio-codec.c | 533 ++++++ sound/soc/codecs/sirf-audio-codec.h | 75 + sound/soc/codecs/sn95031.c | 44 +- sound/soc/codecs/ssm2518.c | 14 +- sound/soc/codecs/ssm2602-i2c.c | 57 + sound/soc/codecs/ssm2602-spi.c | 41 + sound/soc/codecs/ssm2602.c | 173 +- sound/soc/codecs/ssm2602.h | 14 + sound/soc/codecs/sta529.c | 2 +- sound/soc/codecs/stac9766.c | 38 +- sound/soc/codecs/tlv320aic23-i2c.c | 59 + sound/soc/codecs/tlv320aic23-spi.c | 56 + sound/soc/codecs/tlv320aic23.c | 71 +- sound/soc/codecs/tlv320aic23.h | 6 + sound/soc/codecs/tlv320aic26.c | 5 +- sound/soc/codecs/tlv320aic32x4.c | 232 ++- sound/soc/codecs/tlv320dac33.c | 34 +- sound/soc/codecs/twl4030.c | 101 +- sound/soc/codecs/twl6040.c | 17 +- sound/soc/codecs/uda1380.c | 40 +- sound/soc/codecs/wl1273.c | 9 +- sound/soc/codecs/wm2200.c | 18 +- sound/soc/codecs/wm5100.c | 36 +- sound/soc/codecs/wm5102.c | 28 +- sound/soc/codecs/wm5110.c | 19 +- sound/soc/codecs/wm8523.c | 4 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8731.c | 4 +- sound/soc/codecs/wm8737.c | 50 +- sound/soc/codecs/wm8741.c | 34 +- sound/soc/codecs/wm8753.c | 5 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8903.c | 108 +- sound/soc/codecs/wm8904.c | 77 +- sound/soc/codecs/wm8940.c | 20 +- sound/soc/codecs/wm8955.c | 11 +- sound/soc/codecs/wm8958-dsp2.c | 8 +- sound/soc/codecs/wm8961.c | 16 +- sound/soc/codecs/wm8962.c | 80 +- sound/soc/codecs/wm8974.c | 4 +- sound/soc/codecs/wm8978.c | 30 +- sound/soc/codecs/wm8983.c | 45 +- sound/soc/codecs/wm8985.c | 39 +- sound/soc/codecs/wm8988.c | 62 +- sound/soc/codecs/wm8990.c | 41 +- sound/soc/codecs/wm8991.c | 44 +- sound/soc/codecs/wm8993.c | 66 +- sound/soc/codecs/wm8994.c | 52 +- sound/soc/codecs/wm8995.c | 43 +- sound/soc/codecs/wm8996.c | 75 +- sound/soc/codecs/wm8997.c | 25 +- sound/soc/codecs/wm9081.c | 23 +- sound/soc/codecs/wm9705.c | 12 +- sound/soc/codecs/wm_adsp.c | 50 +- sound/soc/codecs/wm_hubs.c | 16 +- sound/soc/davinci/davinci-evm.c | 58 +- sound/soc/davinci/davinci-mcasp.c | 196 ++- sound/soc/fsl/Kconfig | 12 +- sound/soc/fsl/eukrea-tlv320.c | 108 +- sound/soc/fsl/fsl_esai.c | 32 +- sound/soc/fsl/fsl_sai.c | 332 ++-- sound/soc/fsl/fsl_sai.h | 48 +- sound/soc/fsl/fsl_spdif.c | 9 +- sound/soc/fsl/imx-pcm-fiq.c | 7 +- sound/soc/fsl/wm1133-ev1.c | 11 +- sound/soc/generic/simple-card.c | 245 +-- sound/soc/intel/Kconfig | 42 +- sound/soc/intel/Makefile | 27 +- sound/soc/intel/byt-rt5640.c | 187 ++ sound/soc/intel/haswell.c | 235 +++ sound/soc/intel/mfld_machine.c | 65 +- sound/soc/intel/sst-acpi.c | 284 ++++ sound/soc/intel/sst-baytrail-dsp.c | 372 ++++ sound/soc/intel/sst-baytrail-ipc.c | 867 ++++++++++ sound/soc/intel/sst-baytrail-ipc.h | 69 + sound/soc/intel/sst-baytrail-pcm.c | 422 +++++ sound/soc/intel/sst-dsp-priv.h | 309 ++++ sound/soc/intel/sst-dsp.c | 385 +++++ sound/soc/intel/sst-dsp.h | 233 +++ sound/soc/intel/sst-firmware.c | 587 +++++++ sound/soc/intel/sst-haswell-dsp.c | 517 ++++++ sound/soc/intel/sst-haswell-ipc.c | 1785 ++++++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 488 ++++++ sound/soc/intel/sst-haswell-pcm.c | 872 ++++++++++ sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} | 8 +- .../intel/{sst_platform.c => sst-mfld-platform.c} | 8 +- .../intel/{sst_platform.h => sst-mfld-platform.h} | 4 +- sound/soc/omap/Kconfig | 4 +- sound/soc/omap/ams-delta.c | 55 +- sound/soc/omap/n810.c | 26 +- sound/soc/omap/rx51.c | 22 +- sound/soc/pxa/corgi.c | 49 +- sound/soc/pxa/e740_wm9705.c | 10 +- sound/soc/pxa/e750_wm9705.c | 10 +- sound/soc/pxa/e800_wm9712.c | 19 +- sound/soc/pxa/magician.c | 26 +- sound/soc/pxa/mioa701_wm9713.c | 19 +- sound/soc/pxa/poodle.c | 7 +- sound/soc/pxa/spitz.c | 58 +- sound/soc/pxa/tosa.c | 32 +- sound/soc/pxa/zylonite.c | 17 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/h1940_uda1380.c | 7 +- sound/soc/samsung/neo1973_wm8753.c | 168 +- sound/soc/samsung/rx1950_uda1380.c | 5 - sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/tobermory.c | 2 + sound/soc/sh/migor.c | 19 +- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/adg.c | 228 ++- sound/soc/sh/rcar/core.c | 306 ++-- sound/soc/sh/rcar/gen.c | 98 +- sound/soc/sh/rcar/rsnd.h | 195 ++- sound/soc/sh/rcar/scu.c | 384 ----- sound/soc/sh/rcar/src.c | 687 ++++++++ sound/soc/sh/rcar/ssi.c | 332 ++-- sound/soc/sirf/Kconfig | 14 + sound/soc/sirf/Makefile | 5 + sound/soc/sirf/sirf-audio-port.c | 194 +++ sound/soc/sirf/sirf-audio-port.h | 62 + sound/soc/sirf/sirf-audio.c | 156 ++ sound/soc/soc-cache.c | 13 +- sound/soc/soc-compress.c | 65 +- sound/soc/soc-core.c | 582 ++++--- sound/soc/soc-dapm.c | 465 ++--- sound/soc/soc-pcm.c | 112 +- sound/soc/spear/spdif_out.c | 10 +- sound/soc/tegra/Kconfig | 2 +- 217 files changed, 17528 insertions(+), 4570 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/da9055.txt create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm512x.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-port.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/tdm-slot.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt create mode 100644 Documentation/devicetree/bindings/sound/widgets.txt create mode 100644 include/linux/platform_data/adau1977.h create mode 100644 include/trace/events/hswadsp.h create mode 100644 include/trace/events/intel-sst.h create mode 100644 sound/soc/codecs/ad193x-i2c.c create mode 100644 sound/soc/codecs/ad193x-spi.c create mode 100644 sound/soc/codecs/adau1977-i2c.c create mode 100644 sound/soc/codecs/adau1977-spi.c create mode 100644 sound/soc/codecs/adau1977.c create mode 100644 sound/soc/codecs/adau1977.h create mode 100644 sound/soc/codecs/adav801.c create mode 100644 sound/soc/codecs/adav803.c create mode 100644 sound/soc/codecs/pcm512x-i2c.c create mode 100644 sound/soc/codecs/pcm512x-spi.c create mode 100644 sound/soc/codecs/pcm512x.c create mode 100644 sound/soc/codecs/pcm512x.h create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h create mode 100644 sound/soc/codecs/ssm2602-i2c.c create mode 100644 sound/soc/codecs/ssm2602-spi.c create mode 100644 sound/soc/codecs/tlv320aic23-i2c.c create mode 100644 sound/soc/codecs/tlv320aic23-spi.c create mode 100644 sound/soc/intel/byt-rt5640.c create mode 100644 sound/soc/intel/haswell.c create mode 100644 sound/soc/intel/sst-acpi.c create mode 100644 sound/soc/intel/sst-baytrail-dsp.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.h create mode 100644 sound/soc/intel/sst-baytrail-pcm.c create mode 100644 sound/soc/intel/sst-dsp-priv.h create mode 100644 sound/soc/intel/sst-dsp.c create mode 100644 sound/soc/intel/sst-dsp.h create mode 100644 sound/soc/intel/sst-firmware.c create mode 100644 sound/soc/intel/sst-haswell-dsp.c create mode 100644 sound/soc/intel/sst-haswell-ipc.c create mode 100644 sound/soc/intel/sst-haswell-ipc.h create mode 100644 sound/soc/intel/sst-haswell-pcm.c rename sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} (96%) rename sound/soc/intel/{sst_platform.c => sst-mfld-platform.c} (99%) rename sound/soc/intel/{sst_platform.h => sst-mfld-platform.h} (97%) delete mode 100644 sound/soc/sh/rcar/scu.c create mode 100644 sound/soc/sh/rcar/src.c create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio-port.c create mode 100644 sound/soc/sirf/sirf-audio-port.h create mode 100644 sound/soc/sirf/sirf-audio.c

3 3

[alsa-devel] [PATCH] ASoC: Intel: don't select RT5640 if !I2C
by Stephen Warren 13 Mar '14

13 Mar '14

From: Stephen Warren <swarren(a)nvidia.com> The rt5640 driver won't compile without I2C enabled. Hence, the Intel Haswell and Baytrail+RT5640 ASoC drivers must also depend on I2C, since these select RT5640. This solves: sound/soc/codecs/rt5640.c:2220:1: warning: data definition has no type or storage class [enabled by default] sound/soc/codecs/rt5640.c:2220:1: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int] sound/soc/codecs/rt5640.c:2220:1: warning: parameter names (without types) in function declaration [enabled by default] sound/soc/codecs/rt5640.c:2210:26: warning: ‘rt5640_i2c_driver’ defined but not used [-Wunused-variable] Reported-by: Jim Davis <jim.epost(a)gmail.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/intel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 4577b69fcf2c..3c81b3891209 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -32,7 +32,7 @@ config SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help @@ -43,7 +43,7 @@ config SND_SOC_INTEL_HASWELL_MACH config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C select SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5640 help -- 1.8.1.5

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic23: Remove spurious bits per word setting
by Mark Brown 12 Mar '14

12 Mar '14

From: Mark Brown <broonie(a)linaro.org> regmap should handle any byte ordering issues required, it is looking for a byte stream from the bus, so don't set 16 bits per word. This is likely to have tested out OK due to use of an unmerged SPI controller driver. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic23-spi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic23-spi.c b/sound/soc/codecs/tlv320aic23-spi.c index 585aea436c6a..3b387e41d75d 100644 --- a/sound/soc/codecs/tlv320aic23-spi.c +++ b/sound/soc/codecs/tlv320aic23-spi.c @@ -25,7 +25,6 @@ static int aic23_spi_probe(struct spi_device *spi) dev_dbg(&spi->dev, "probing tlv320aic23 spi device\n"); - spi->bits_per_word = 16; spi->mode = SPI_MODE_0; ret = spi_setup(spi); if (ret < 0) -- 1.9.0

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Split firmware load into smaller chunks
by Charles Keepax 12 Mar '14

12 Mar '14

The firmware files can be quite large and allocating the whole firmware a single DMA safe buffer can be problematic if the system is under a high memory load. Ease the requirements slightly by writing the firmware out in page sized chunks. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 50 ++++++++++++++++++++++++++++--------------- 1 files changed, 32 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f9fd564..937af6f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -684,24 +684,38 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - buf = wm_adsp_buf_alloc(region->data, - le32_to_cpu(region->len), - &buf_list); - if (!buf) { - adsp_err(dsp, "Out of memory\n"); - ret = -ENOMEM; - goto out_fw; - } + size_t to_write = PAGE_SIZE; + size_t remain = le32_to_cpu(region->len); + const u8 *data = region->data; + + while (remain > 0) { + if (remain < PAGE_SIZE) + to_write = remain; + + buf = wm_adsp_buf_alloc(data, + to_write, + &buf_list); + if (!buf) { + adsp_err(dsp, "Out of memory\n"); + ret = -ENOMEM; + goto out_fw; + } - ret = regmap_raw_write_async(regmap, reg, buf->buf, - le32_to_cpu(region->len)); - if (ret != 0) { - adsp_err(dsp, - "%s.%d: Failed to write %d bytes at %d in %s: %d\n", - file, regions, - le32_to_cpu(region->len), offset, - region_name, ret); - goto out_fw; + ret = regmap_raw_write_async(regmap, reg, + buf->buf, + to_write); + if (ret != 0) { + adsp_err(dsp, + "%s.%d: Failed to write %d bytes at %d in %s: %d\n", + file, regions, + to_write, offset, + region_name, ret); + goto out_fw; + } + + data += to_write; + reg += to_write / 2; + remain -= to_write; } } -- 1.7.2.5

2 3

[alsa-devel] snd_pcm_drain causes repeated playback of last buffer contents
by Sanjeev Chugh 12 Mar '14

12 Mar '14

Hello, I'm not an alsa expert at all but trying to learn alsa. I'm experiencing an issue where on snd_pcm_drain() the slave PCM driven via plugin DSHARE is not filled with silence. Result is endless playback of buffer content until pcm is closed. I'm trying to analyze what could be the reason for this. In my understanding in Alsa, audio data is played as a unit of period. Once playing one period is finished, then next period will be played. If period has lesser frames than the period size, audio data previously played will be re-played until pcm is closed out. Could it be a reason for this continuous playback when snd_pcm_drain is called ? I'm not even sure if it is desired behavior that silence be padded after driver drains the pcm :( I guess it is too big as a bug to be ignored. Thanks Pthread

1 0

[alsa-devel] [PATCH] ASoC: Fix use after free
by Lars-Peter Clausen 12 Mar '14

12 Mar '14

Freeing the current list element while iterating over the list will cause a use after free since the iterator function will still use the current element to look up the next. Use list_for_each_safe() and remove the element from the list before freeing it to avoid this. Fixes: 1438c2f60b ("ASoC: Add a per component dai list") Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f67cef4..dac616d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3928,11 +3928,12 @@ static inline char *fmt_multiple_name(struct device *dev, */ static void snd_soc_unregister_dais(struct snd_soc_component *component) { - struct snd_soc_dai *dai; + struct snd_soc_dai *dai, *_dai; - list_for_each_entry(dai, &component->dai_list, list) { + list_for_each_entry_safe(dai, _dai, &component->dai_list, list) { dev_dbg(component->dev, "ASoC: Unregistered DAI '%s'\n", dai->name); + list_del(&dai->list); kfree(dai->name); kfree(dai); } -- 1.8.0

2 1

[alsa-devel] [PATCH v5-resend 0/7] ASoC: add CSR SiRFSoC sound drivers
by RongJun Ying 12 Mar '14

12 Mar '14

From: Rongjun Ying <rongjun.ying(a)csr.com> This patchset adds CSR SiRFSoC sound drivers including: 1. The SiRF internal audio codec driver 2. The SiRF audio port driver 3. The SiRF internal audio card driver 4. The SiRF I2S controller driver 5. The SiRF hdmi card driver 6. The SiRF USP controller(DSP mode) driver 7. The SiRF bt-sco card driver -v5: 1. Drop all inlines. 2. Reordering the Kconfig and Makefile 3. Remove the sirf_audio_codec_reg_bits struct, use the new controls instead it. 4. Add some SND_SOC_DAPM_OUT_DRV instead of HP and SPK enable driver 5. Add audio codec clock supply instead of adc event callback 6. Fixed playback and capture can't concurrent work bug. Rongjun Ying (7): ASoC: sirf: Add SiRF internal audio codec driver SoC: sirf: Add SiRF audio port driver is used by SiRF internal audio codec ASoC: sirf: Add SiRF audio card ASoC: sirf: Add SiRF I2S driver ASoC: sirf: Add hdmi card ASoC: sirf: Add usp driver which is used by dsp mode SoC: sirf: Add bt-sco card .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 + .../devicetree/bindings/sound/sirf-audio-port.txt | 20 + .../devicetree/bindings/sound/sirf-audio.txt | 41 ++ .../devicetree/bindings/sound/sirf-bt-sco.txt | 18 + .../devicetree/bindings/sound/sirf-hdmi.txt | 15 + .../devicetree/bindings/sound/sirf-i2s.txt | 27 + .../devicetree/bindings/sound/sirf-usp.txt | 27 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/sirf-audio-codec.c | 533 ++++++++++++++++++++ sound/soc/codecs/sirf-audio-codec.h | 75 +++ sound/soc/sirf/Kconfig | 34 ++ sound/soc/sirf/Makefile | 13 + sound/soc/sirf/sirf-audio-port.c | 194 +++++++ sound/soc/sirf/sirf-audio-port.h | 62 +++ sound/soc/sirf/sirf-audio.c | 156 ++++++ sound/soc/sirf/sirf-bt-sco.c | 99 ++++ sound/soc/sirf/sirf-hdmi.c | 98 ++++ sound/soc/sirf/sirf-i2s.c | 434 ++++++++++++++++ sound/soc/sirf/sirf-i2s.h | 88 ++++ sound/soc/sirf/sirf-usp.c | 448 ++++++++++++++++ sound/soc/sirf/sirf-usp.h | 282 +++++++++++ 24 files changed, 2689 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-port.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-bt-sco.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-hdmi.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio-port.c create mode 100644 sound/soc/sirf/sirf-audio-port.h create mode 100644 sound/soc/sirf/sirf-audio.c create mode 100644 sound/soc/sirf/sirf-bt-sco.c create mode 100644 sound/soc/sirf/sirf-hdmi.c create mode 100644 sound/soc/sirf/sirf-i2s.c create mode 100644 sound/soc/sirf/sirf-i2s.h create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h -- 1.7.5.4

2 19

[alsa-devel] HDA jack retask for Debian 7.4 NOT working
by Subodh Chiwate 12 Mar '14

12 Mar '14

Hi, I am trying to get surround 7.1 on debian 7.4 Wheezy. As guided by hda-jack-retask readme I built it from source on my system. But the changes are not reflecting and no errors by utility as well. I am getting Alsa 1.0.25 from the repository. Do I need to build the latest version from source to get hdajackretask as part of alsa-tools ? Any other suggestions to get the 7.1 working are welcome. Thanks in advance. -- Regards Subodh

2 2

[alsa-devel] [PATCH] ASoC: simple-card: overwrite DAIFMT_MASTER of cpu_dai->fmt
by Nicolin Chen 12 Mar '14

12 Mar '14

The current simple-card driver separates the daimft for cpu_dai and codec_dai. So we might get different values for them (0x4003 and 0x1003 for example): asoc-simple-card sound-cs42888.12: cpu : 2024000.esai / 4003 / 132000000 asoc-simple-card sound-cs42888.12: codec : cs42888 / 1003 / 24576000 asoc-simple-card sound-cs42888.12: cs42888 <-> 2024000.esai mapping ok It's pretty fair to do it for DAIFMT_INV since two dais might have differnt definitions in their drivers even though it'd be better to keep it same if we could. But for DAIFMT_MASTER bit, we should keep them identical. Otherwise, it'll make one of dai work in an incorrect mode. Thus this patch fixes it by overwriting the DAIFMT_MASTER bit of cpu_dai->fmt with the one of codec_dai->fmt since we defined DAIFMT_MASTER basing on CODEC at the first place. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/generic/simple-card.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 5dd4769..f9a3c8a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -157,6 +157,8 @@ static int asoc_simple_card_parse_of(struct device_node *node, struct device *dev) { struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; + struct asoc_simple_dai *codec_dai = &priv->codec_dai; + struct asoc_simple_dai *cpu_dai = &priv->cpu_dai; struct device_node *np; char *name; int ret; @@ -189,7 +191,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) ret = asoc_simple_card_sub_parse_of(np, priv->daifmt, - &priv->cpu_dai, + cpu_dai, &dai_link->cpu_of_node, &dai_link->cpu_dai_name); if (ret < 0) @@ -200,12 +202,16 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) ret = asoc_simple_card_sub_parse_of(np, priv->daifmt, - &priv->codec_dai, + codec_dai, &dai_link->codec_of_node, &dai_link->codec_dai_name); if (ret < 0) return ret; + /* overwrite DAIFMT_MASTER of cpu_dai since it's based on CODEC */ + cpu_dai->fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + cpu_dai->fmt |= codec_dai->fmt & SND_SOC_DAIFMT_MASTER_MASK; + if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name) return -EINVAL; @@ -227,12 +233,12 @@ static int asoc_simple_card_parse_of(struct device_node *node, dev_dbg(dev, "platform : %04x\n", priv->daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", dai_link->cpu_dai_name, - priv->cpu_dai.fmt, - priv->cpu_dai.sysclk); + cpu_dai->fmt, + cpu_dai->sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", dai_link->codec_dai_name, - priv->codec_dai.fmt, - priv->codec_dai.sysclk); + codec_dai->fmt, + codec_dai->sysclk); /* * soc_bind_dai_link() will check cpu name -- 1.8.4

3 6

[alsa-devel] [PATCHv3 resend 0/3] Simplify the CODEC ASoC probe code.
by Xiubo Li 12 Mar '14

12 Mar '14

3 5

[alsa-devel] Creating single-card driver for multi-codec (TLV320AIC34) board - is it possible at the machine level?
by Matt Kleffner 12 Mar '14

12 Mar '14

Hi ALSA developers, I'm developing an ALSA driver for a new board with two TLV320AIC34 chips, which to a driver is the equivalent of 4 TLV320AIC31/2/3 (TLV320AIC3X driver) codecs. The platform is OMAP4, with the OMAP as I2C master. TDM mode with one McBSP port is used, and it is desired that the four codecs appear as one 8-channel codec (clocks are shared and data buses are connected together). It is OK if there are 4 separate mixers, one for each codec. However, when starting/stopping audio all codecs should act together and audio data presented to the ALSA user interface just as with a single 8-channel sound card. My questions have two parts: one on binding together the codecs into a "single card", and another on alterations to support an external sample-clock generator. BONDING CODECS TOGETHER It is desired to write the driver at the "machine" level (ignore the external-clock modification for now), without modifying the ALSA SoC core or the TLV320AIC3X driver. Is this possible? I've searched the internet and looked through the source code, and I have the following gaps in understanding: The following TI link seems to suggest such TLV320AIC34 operation may already be possible with ALSA: http://e2e.ti.com/support/data_converters/audio_converters/w/design_notes/2… However, one of the threads pointed in that link: http://comments.gmane.org/gmane.linux.alsa.devel/84899 suggests such support is not in the ALSA SoC core or the TLV320AIC3X driver. While it is mentioned that if configured appropriately, "you'll end up with a single card", later it is mentioned that "you're probably trying to break new ground here. You're trying to bind all the CODECs together and treat them as one digital interface, currently the code does a series of point to point links multiplexed on a bus only." What does point-to-point link mean in this context? What does "you'll end up with a single card" mean if all 8 channels are not exposed in one card? If it is new ground, then what does the rx51 driver do (it is often referenced as a starting point for this type of binding effort), which interfaces to an AIC34 codec? The threads referenced are a few years old. Is it now possible to write such a driver at the machine level, or must I additionally either modify the core or write a new codec driver (as the original poster did)? ALTERATIONS TO SUPPORT AN EXTERNAL SAMPLE-CLOCK GENERATOR The sample clock is being generated by a single, I2C-configurable clock chip. So any sample-rate configuration must be sent to the clock chip. Can this be handled purely in the machine layer, or do I need to modify the AIC3X driver? I understand I need to use the regmap API to do such things, and ALSA uses this API, but I do not yet know much about it. Any pointers to documentation on doing this besides the source code are appreciated. A key newbie question on regmap (and Linux I2C) as needed by the driver the following: Does the regmap-i2c or i2c driver "own" the I2C bus so that the each device on the bus can be read from/written to via the regmap-i2c or i2c driver without the device's driver having to request-take-release the bus on each access? Thanks, - Matt

1 0

[alsa-devel] [PATCH v2] ASoC: cs42888: Add codec driver support
by Nicolin Chen 12 Mar '14

12 Mar '14

This patch adds support for the Cirrus Logic CS42888 Audio CODEC that has four 24-bit A/D and eight 24-bit D/A converters. [ CS42888 supports both I2C and SPI control ports. As initial patch, this patch only adds the support for I2C. ] Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v2: * Separated driver into I2C and main part drivers. * Added error out if failed to get clock. * Enabled Auto-mute function as default. * Disabled runtime-pm in remove(). * Added return value checking for regcache_sync(). * Fixed some trival parts like 'const' as Lars mentioned. .../devicetree/bindings/sound/cs42888.txt | 27 ++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42888-i2c.c | 68 +++ sound/soc/codecs/cs42888.c | 536 +++++++++++++++++++++ sound/soc/codecs/cs42888.h | 213 ++++++++ 6 files changed, 858 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs42888.txt create mode 100644 sound/soc/codecs/cs42888-i2c.c create mode 100644 sound/soc/codecs/cs42888.c create mode 100644 sound/soc/codecs/cs42888.h diff --git a/Documentation/devicetree/bindings/sound/cs42888.txt b/Documentation/devicetree/bindings/sound/cs42888.txt new file mode 100644 index 0000000..7736527 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42888.txt @@ -0,0 +1,27 @@ +CS42888 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42888" + + - reg : the I2C address of the device for I2C + + - clocks : phandle to the clock source for MCLK + + - clock-names : must contain "mclk". + + - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +Example: + +codec: cs42888@48 { + compatible = "cirrus,cs42888"; + reg = <0x48>; + clocks = <&codec_mclk 0>; + clock-names = "mclk"; + VA-supply = <&reg_audio>; + VD-supply = <&reg_audio>; + VLS-supply = <&reg_audio>; + VLC-supply = <&reg_audio>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 32d7a6f..1642578 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,6 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI + select SND_SOC_CS42888_I2C if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if I2C select SND_SOC_DA7213 if I2C @@ -303,6 +304,15 @@ config SND_SOC_CS4271 tristate "Cirrus Logic CS4271 CODEC" depends on SND_SOC_I2C_AND_SPI +config SND_SOC_CS42888 + tristate + +config SND_SOC_CS42888_I2C + tristate "Cirrus Logic CS42888 CODEC (I2C)" + depends on I2C + select SND_SOC_CS42888 + select REGMAP_I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cb46c4c..f5612fb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -30,6 +30,8 @@ snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o +snd-soc-cs42888-objs := cs42888.o +snd-soc-cs42888-i2c-objs := cs42888-i2c.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -178,6 +180,8 @@ obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o +obj-$(CONFIG_SND_SOC_CS42888) += snd-soc-cs42888.o +obj-$(CONFIG_SND_SOC_CS42888_I2C) += snd-soc-cs42888-i2c.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/cs42888-i2c.c b/sound/soc/codecs/cs42888-i2c.c new file mode 100644 index 0000000..54813bc --- /dev/null +++ b/sound/soc/codecs/cs42888-i2c.c @@ -0,0 +1,68 @@ +/* + * Cirrus Logic CS42888 Audio CODEC Digital Audio Interface (DAI) I2C driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "cs42888.h" + +static int cs42888_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + u32 ret = cs42888_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &cs42888_regmap_config)); + if (ret) + return ret; + + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + + return 0; +} + +static int cs42888_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + pm_runtime_disable(&i2c->dev); + + return 0; +} + +static struct i2c_device_id cs42888_i2c_id[] = { + {"cs42888", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42888_i2c_id); + +static struct i2c_driver cs42888_i2c_driver = { + .driver = { + .name = "cs42888", + .owner = THIS_MODULE, + .pm = &cs42888_pm, + }, + .probe = cs42888_i2c_probe, + .remove = cs42888_i2c_remove, + .id_table = cs42888_i2c_id, +}; + +module_i2c_driver(cs42888_i2c_driver); + +MODULE_DESCRIPTION("Cirrus Logic CS42888 ALSA SoC Codec I2C Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42888.c b/sound/soc/codecs/cs42888.c new file mode 100644 index 0000000..8cd3d4a --- /dev/null +++ b/sound/soc/codecs/cs42888.c @@ -0,0 +1,536 @@ +/* + * Cirrus Logic CS42888 Audio CODEC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "cs42888.h" + +#define CS42888_NUM_SUPPLIES 4 +static const char *const cs42888_supply_names[CS42888_NUM_SUPPLIES] = { + "VA", + "VD", + "VLS", + "VLC", +}; + +#define CS42888_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct cs42888_priv { + struct regulator_bulk_data supplies[CS42888_NUM_SUPPLIES]; + struct regmap *regmap; + struct clk *clk; + + bool slave_mode; + unsigned long sysclk; +}; + +/* -127.5dB to 0dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +/* -64dB to 24dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(adc_tlv, -6400, 50, 0); + +static const char *const cs42888_adc_single[] = { "Differential", "Single-Ended" }; +static const char *const cs42888_szc[] = { "Immediate Change", "Zero Cross", + "Soft Ramp", "Soft Ramp on Zero Cross" }; + +static const struct soc_enum adc1_single_enum = + SOC_ENUM_SINGLE(CS42888_ADCCTL, 4, 2, cs42888_adc_single); +static const struct soc_enum adc2_single_enum = + SOC_ENUM_SINGLE(CS42888_ADCCTL, 3, 2, cs42888_adc_single); +static const struct soc_enum dac_szc_enum = + SOC_ENUM_SINGLE(CS42888_TXCTL, 5, 4, cs42888_szc); +static const struct soc_enum adc_szc_enum = + SOC_ENUM_SINGLE(CS42888_TXCTL, 0, 4, cs42888_szc); + +static const struct snd_kcontrol_new cs42888_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", CS42888_VOLAOUT1, + CS42888_VOLAOUT2, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", CS42888_VOLAOUT3, + CS42888_VOLAOUT4, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC3 Playback Volume", CS42888_VOLAOUT5, + CS42888_VOLAOUT6, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC4 Playback Volume", CS42888_VOLAOUT7, + CS42888_VOLAOUT8, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_S_TLV("ADC1 Capture Volume", CS42888_VOLAIN1, + CS42888_VOLAIN2, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE_R_S_TLV("ADC2 Capture Volume", CS42888_VOLAIN3, + CS42888_VOLAIN4, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("DAC1 Invert Switch", CS42888_DACINV, 0, 1, 1, 0), + SOC_DOUBLE("DAC2 Invert Switch", CS42888_DACINV, 2, 3, 1, 0), + SOC_DOUBLE("DAC3 Invert Switch", CS42888_DACINV, 4, 5, 1, 0), + SOC_DOUBLE("DAC4 Invert Switch", CS42888_DACINV, 6, 7, 1, 0), + SOC_DOUBLE("ADC1 Invert Switch", CS42888_ADCINV, 0, 1, 1, 0), + SOC_DOUBLE("ADC2 Invert Switch", CS42888_ADCINV, 2, 3, 1, 0), + SOC_SINGLE("ADC High-Pass Filter Switch", CS42888_ADCCTL, 7, 1, 1), + SOC_SINGLE("DAC De-emphasis Switch", CS42888_ADCCTL, 5, 1, 0), + SOC_ENUM("ADC1 Single Ended Mode Switch", adc1_single_enum), + SOC_ENUM("ADC2 Single Ended Mode Switch", adc2_single_enum), + SOC_SINGLE("DAC Single Volume Control Switch", CS42888_TXCTL, 7, 1, 0), + SOC_ENUM("DAC Soft Ramp & Zero Cross Control Switch", dac_szc_enum), + SOC_SINGLE("DAC Auto Mute Switch", CS42888_TXCTL, 4, 1, 0), + SOC_SINGLE("Mute ADC Serial Port Switch", CS42888_TXCTL, 3, 1, 0), + SOC_SINGLE("ADC Single Volume Control Switch", CS42888_TXCTL, 2, 1, 0), + SOC_ENUM("ADC Soft Ramp & Zero Cross Control Switch", adc_szc_enum), +}; + +static const struct snd_soc_dapm_widget cs42888_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC1", "Playback", CS42888_PWRCTL, 1, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", CS42888_PWRCTL, 2, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", CS42888_PWRCTL, 3, 1), + SND_SOC_DAPM_DAC("DAC4", "Playback", CS42888_PWRCTL, 4, 1), + + SND_SOC_DAPM_OUTPUT("AOUT1L"), + SND_SOC_DAPM_OUTPUT("AOUT1R"), + SND_SOC_DAPM_OUTPUT("AOUT2L"), + SND_SOC_DAPM_OUTPUT("AOUT2R"), + SND_SOC_DAPM_OUTPUT("AOUT3L"), + SND_SOC_DAPM_OUTPUT("AOUT3R"), + SND_SOC_DAPM_OUTPUT("AOUT4L"), + SND_SOC_DAPM_OUTPUT("AOUT4R"), + + SND_SOC_DAPM_ADC("ADC1", "Capture", CS42888_PWRCTL, 5, 1), + SND_SOC_DAPM_ADC("ADC2", "Capture", CS42888_PWRCTL, 6, 1), + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + + SND_SOC_DAPM_SUPPLY("PWR", CS42888_PWRCTL, 0, 1, NULL, 0), +}; + +static const struct snd_soc_dapm_route cs42888_dapm_routes[] = { + /* Playback */ + { "AOUT1L", NULL, "DAC1" }, + { "AOUT1R", NULL, "DAC1" }, + { "DAC1", NULL, "PWR" }, + + { "AOUT2L", NULL, "DAC2" }, + { "AOUT2R", NULL, "DAC2" }, + { "DAC2", NULL, "PWR" }, + + { "AOUT3L", NULL, "DAC3" }, + { "AOUT3R", NULL, "DAC3" }, + { "DAC3", NULL, "PWR" }, + + { "AOUT4L", NULL, "DAC4" }, + { "AOUT4R", NULL, "DAC4" }, + { "DAC4", NULL, "PWR" }, + + /* Capture */ + { "ADC1", NULL, "AIN1L" }, + { "ADC1", NULL, "AIN1R" }, + { "ADC1", NULL, "PWR" }, + + { "ADC2", NULL, "AIN2L" }, + { "ADC2", NULL, "AIN2R" }, + { "ADC2", NULL, "PWR" }, +}; + +struct cs42888_ratios { + unsigned int ratio; + unsigned char speed; + unsigned char mclk; +}; + +static const struct cs42888_ratios cs42888_ratios[] = { + { 64, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_256(4) }, + { 96, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_384(4) }, + { 128, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_512(4) }, + { 192, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_768(4) }, + { 256, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_256(1) }, + { 384, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_384(1) }, + { 512, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_512(1) }, + { 768, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_768(1) }, + { 1024, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_1024(1) } +}; + +static int cs42888_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + cs42888->sysclk = freq; + + return 0; +} + +static int cs42888_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + u32 val; + + /* Set DAI format */ + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + val = CS42888_INTF_DAC_DIF_LEFTJ | CS42888_INTF_ADC_DIF_LEFTJ; + break; + case SND_SOC_DAIFMT_I2S: + val = CS42888_INTF_DAC_DIF_I2S | CS42888_INTF_ADC_DIF_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = CS42888_INTF_DAC_DIF_RIGHTJ | CS42888_INTF_ADC_DIF_RIGHTJ; + break; + default: + dev_err(codec->dev, "unsupported dai format\n"); + return -EINVAL; + } + + regmap_update_bits(cs42888->regmap, CS42888_INTF, + CS42888_INTF_DAC_DIF_MASK | + CS42888_INTF_ADC_DIF_MASK, val); + + /* Set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs42888->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs42888->slave_mode = false; + break; + default: + dev_err(codec->dev, "unsupported master/slave mode\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42888_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 cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 ratio = cs42888->sysclk / params_rate(params); + u32 i, fm, val, mask; + + for (i = 0; i < ARRAY_SIZE(cs42888_ratios); i++) { + if (cs42888_ratios[i].ratio == ratio) + break; + } + + if (i == ARRAY_SIZE(cs42888_ratios)) { + dev_err(codec->dev, "unsupported sysclk ratio\n"); + return -EINVAL; + } + + mask = CS42888_FUNCMOD_MFREQ_MASK; + val = cs42888_ratios[i].mclk; + + fm = cs42888->slave_mode ? CS42888_FM_AUTO : cs42888_ratios[i].speed; + + regmap_update_bits(cs42888->regmap, CS42888_FUNCMOD, + CS42888_FUNCMOD_xC_FM_MASK(tx) | mask, + CS42888_FUNCMOD_xC_FM(tx, fm) | val); + + return 0; +} + +static int cs42888_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(cs42888->regmap, CS42888_DACMUTE, + CS42888_DACMUTE_ALL, mute ? CS42888_DACMUTE_ALL : 0); + + return 0; +} + +static const struct snd_soc_dai_ops cs42888_dai_ops = { + .set_fmt = cs42888_set_dai_fmt, + .set_sysclk = cs42888_set_dai_sysclk, + .hw_params = cs42888_hw_params, + .digital_mute = cs42888_digital_mute, +}; + +static struct snd_soc_dai_driver cs42888_dai = { + .name = "cs42888", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42888_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42888_FORMATS, + }, + .ops = &cs42888_dai_ops, +}; + +static const struct reg_default cs42888_reg[] = { + { 0x01, 0x01 }, /* Chip I.D. and Revision Register */ + { 0x02, 0x00 }, /* Power Control */ + { 0x03, 0xF0 }, /* Functional Mode */ + { 0x04, 0x46 }, /* Interface Formats */ + { 0x05, 0x00 }, /* ADC Control & DAC De-Emphasis */ + { 0x06, 0x10 }, /* Transition Control */ + { 0x07, 0x00 }, /* DAC Channel Mute */ + { 0x08, 0x00 }, /* Volume Control AOUT1 */ + { 0x09, 0x00 }, /* Volume Control AOUT2 */ + { 0x0a, 0x00 }, /* Volume Control AOUT3 */ + { 0x0b, 0x00 }, /* Volume Control AOUT4 */ + { 0x0c, 0x00 }, /* Volume Control AOUT5 */ + { 0x0d, 0x00 }, /* Volume Control AOUT6 */ + { 0x0e, 0x00 }, /* Volume Control AOUT7 */ + { 0x0f, 0x00 }, /* Volume Control AOUT8 */ + { 0x10, 0x00 }, /* DAC Channel Invert */ + { 0x11, 0x00 }, /* Volume Control AIN1 */ + { 0x12, 0x00 }, /* Volume Control AIN2 */ + { 0x13, 0x00 }, /* Volume Control AIN3 */ + { 0x14, 0x00 }, /* Volume Control AIN4 */ + { 0x17, 0x00 }, /* ADC Channel Invert */ + { 0x18, 0x00 }, /* Status Control */ + { 0x1a, 0x00 }, /* Status Mask */ + { 0x1b, 0x00 }, /* MUTEC Pin Control */ +}; + +static bool cs42888_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42888_STATUS: + return true; + default: + return false; + } +} + +static bool cs42888_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42888_CHIPID: + case CS42888_STATUS: + return false; + default: + return true; + } +} + +const struct regmap_config cs42888_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42888_LASTREG, + .reg_defaults = cs42888_reg, + .num_reg_defaults = ARRAY_SIZE(cs42888_reg), + .volatile_reg = cs42888_volatile_register, + .writeable_reg = cs42888_writeable_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(cs42888_regmap_config); + +static int cs42888_codec_probe(struct snd_soc_codec *codec) +{ + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + /* Mute all DAC channels */ + regmap_write(cs42888->regmap, CS42888_DACMUTE, CS42888_DACMUTE_ALL); + + return 0; +} + +static const struct snd_soc_codec_driver cs42888_driver = { + .probe = cs42888_codec_probe, + .idle_bias_off = true, + + .controls = cs42888_snd_controls, + .num_controls = ARRAY_SIZE(cs42888_snd_controls), + .dapm_widgets = cs42888_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42888_dapm_widgets), + .dapm_routes = cs42888_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs42888_dapm_routes), +}; + +int cs42888_probe(struct device *dev, struct regmap *regmap) +{ + struct cs42888_priv *cs42888; + int ret, val, i; + + cs42888 = devm_kzalloc(dev, sizeof(*cs42888), GFP_KERNEL); + if (cs42888 == NULL) + return -ENOMEM; + + dev_set_drvdata(dev, cs42888); + + cs42888->clk = devm_clk_get(dev, "mclk"); + if (IS_ERR(cs42888->clk)) { + dev_err(dev, "failed to get the clock: %ld\n", + PTR_ERR(cs42888->clk)); + return -EINVAL; + } + + cs42888->sysclk = clk_get_rate(cs42888->clk); + + for (i = 0; i < ARRAY_SIZE(cs42888->supplies); i++) + cs42888->supplies[i].supply = cs42888_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, + ARRAY_SIZE(cs42888->supplies), cs42888->supplies); + if (ret) { + dev_err(dev, "failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + /* Make sure hardware reset done */ + msleep(5); + + cs42888->regmap = regmap; + if (IS_ERR(cs42888->regmap)) { + ret = PTR_ERR(cs42888->regmap); + dev_err(dev, "failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + /* + * We haven't marked the chip revision as volatile due to + * sharing a register with the right input volume; explicitly + * bypass the cache to read it. + */ + regcache_cache_bypass(cs42888->regmap, true); + + /* Validate the chip ID */ + regmap_read(cs42888->regmap, CS42888_CHIPID, &val); + if (val < 0) { + dev_err(dev, "failed to get device ID: %x", val); + ret = -EINVAL; + goto err_enable; + } + + /* The top four bits of the chip ID should be 0000 */ + if ((val & CS42888_CHIPID_CHIP_ID_MASK) != 0x00) { + dev_err(dev, "unmatched chip ID: %d\n", + val & CS42888_CHIPID_CHIP_ID_MASK); + ret = -EINVAL; + goto err_enable; + } + + dev_info(dev, "found device, revision %X\n", + val & CS42888_CHIPID_REV_ID_MASK); + + regcache_cache_bypass(cs42888->regmap, false); + + ret = snd_soc_register_codec(dev, &cs42888_driver, &cs42888_dai, 1); + if (ret) { + dev_err(dev, "failed to register codec:%d\n", ret); + goto err_enable; + } + + regcache_cache_only(cs42888->regmap, true); + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + + return ret; +} +EXPORT_SYMBOL_GPL(cs42888_probe); + +#ifdef CONFIG_PM_RUNTIME +static int cs42888_runtime_resume(struct device *dev) +{ + struct cs42888_priv *cs42888 = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(cs42888->clk); + if (ret) { + dev_err(dev, "failed to enable mclk: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + goto err_clk; + } + + /* Make sure hardware reset done */ + msleep(5); + + regcache_cache_only(cs42888->regmap, false); + + ret = regcache_sync(cs42888->regmap); + if (ret) { + dev_err(dev, "failed to sync regmap: %d\n", ret); + goto err_bulk; + } + + return 0; + +err_bulk: + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); +err_clk: + clk_disable_unprepare(cs42888->clk); + + return ret; +} + +static int cs42888_runtime_suspend(struct device *dev) +{ + struct cs42888_priv *cs42888 = dev_get_drvdata(dev); + + regcache_cache_only(cs42888->regmap, true); + + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + + clk_disable_unprepare(cs42888->clk); + + return 0; +} +#endif + +const struct dev_pm_ops cs42888_pm = { + SET_RUNTIME_PM_OPS(cs42888_runtime_suspend, cs42888_runtime_resume, NULL) +}; +EXPORT_SYMBOL_GPL(cs42888_pm); + +const struct of_device_id cs42888_of_match[] = { + { .compatible = "cirrus,cs42888", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, cs42888_of_match); +EXPORT_SYMBOL_GPL(cs42888_of_match); + +MODULE_DESCRIPTION("Cirrus Logic CS42888 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42888.h b/sound/soc/codecs/cs42888.h new file mode 100644 index 0000000..eda55ac --- /dev/null +++ b/sound/soc/codecs/cs42888.h @@ -0,0 +1,213 @@ +/* + * cs42888.h - Cirrus Logic CS42888 Audio CODEC driver header file + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _CS42888_H +#define _CS42888_H + +extern const struct dev_pm_ops cs42888_pm; +extern const struct regmap_config cs42888_regmap_config; +int cs42888_probe(struct device *dev, struct regmap *regmap); + +/* CS42888 register map */ +#define CS42888_CHIPID 0x01 /* Chip ID */ +#define CS42888_PWRCTL 0x02 /* Power Control */ +#define CS42888_FUNCMOD 0x03 /* Functional Mode */ +#define CS42888_INTF 0x04 /* Interface Formats */ +#define CS42888_ADCCTL 0x05 /* ADC Control */ +#define CS42888_TXCTL 0x06 /* Transition Control */ +#define CS42888_DACMUTE 0x07 /* DAC Mute Control */ +#define CS42888_VOLAOUT1 0x08 /* Volume Control AOUT1 */ +#define CS42888_VOLAOUT2 0x09 /* Volume Control AOUT2 */ +#define CS42888_VOLAOUT3 0x0A /* Volume Control AOUT3 */ +#define CS42888_VOLAOUT4 0x0B /* Volume Control AOUT4 */ +#define CS42888_VOLAOUT5 0x0C /* Volume Control AOUT5 */ +#define CS42888_VOLAOUT6 0x0D /* Volume Control AOUT6 */ +#define CS42888_VOLAOUT7 0x0E /* Volume Control AOUT7 */ +#define CS42888_VOLAOUT8 0x0F /* Volume Control AOUT8 */ +#define CS42888_DACINV 0x10 /* DAC Channel Invert */ +#define CS42888_VOLAIN1 0x11 /* Volume Control AIN1 */ +#define CS42888_VOLAIN2 0x12 /* Volume Control AIN2 */ +#define CS42888_VOLAIN3 0x13 /* Volume Control AIN3 */ +#define CS42888_VOLAIN4 0x14 /* Volume Control AIN4 */ +#define CS42888_ADCINV 0x17 /* ADC Channel Invert */ +#define CS42888_STATUSCTL 0x18 /* Status Control */ +#define CS42888_STATUS 0x19 /* Status */ +#define CS42888_STATUSM 0x1A /* Status Mask */ +#define CS42888_MUTEC 0x1B /* MUTEC Pin Control */ + +#define CS42888_FIRSTREG CS42888_CHIPID +#define CS42888_LASTREG CS42888_MUTEC +#define CS42888_NUMREGS (CS42888_LASTREG - CS42888_FIRSTREG + 1) +#define CS42888_I2C_INCR 0x80 + +/* Chip I.D. and Revision Register (Address 01h) */ +#define CS42888_CHIPID_CHIP_ID_MASK 0xF0 +#define CS42888_CHIPID_REV_ID_MASK 0x0F + +/* Power Control (Address 02h) */ +#define CS42888_PWRCTL_PDN_ADC2_SHIFT 6 +#define CS42888_PWRCTL_PDN_ADC2_MASK (1 << CS42888_PWRCTL_PDN_ADC2_SHIFT) +#define CS42888_PWRCTL_PDN_ADC2 (1 << CS42888_PWRCTL_PDN_ADC2_SHIFT) +#define CS42888_PWRCTL_PDN_ADC1_SHIFT 5 +#define CS42888_PWRCTL_PDN_ADC1_MASK (1 << CS42888_PWRCTL_PDN_ADC1_SHIFT) +#define CS42888_PWRCTL_PDN_ADC1 (1 << CS42888_PWRCTL_PDN_ADC1_SHIFT) +#define CS42888_PWRCTL_PDN_DAC4_SHIFT 4 +#define CS42888_PWRCTL_PDN_DAC4_MASK (1 << CS42888_PWRCTL_PDN_DAC4_SHIFT) +#define CS42888_PWRCTL_PDN_DAC4 (1 << CS42888_PWRCTL_PDN_DAC4_SHIFT) +#define CS42888_PWRCTL_PDN_DAC3_SHIFT 3 +#define CS42888_PWRCTL_PDN_DAC3_MASK (1 << CS42888_PWRCTL_PDN_DAC3_SHIFT) +#define CS42888_PWRCTL_PDN_DAC3 (1 << CS42888_PWRCTL_PDN_DAC3_SHIFT) +#define CS42888_PWRCTL_PDN_DAC2_SHIFT 2 +#define CS42888_PWRCTL_PDN_DAC2_MASK (1 << CS42888_PWRCTL_PDN_DAC2_SHIFT) +#define CS42888_PWRCTL_PDN_DAC2 (1 << CS42888_PWRCTL_PDN_DAC2_SHIFT) +#define CS42888_PWRCTL_PDN_DAC1_SHIFT 1 +#define CS42888_PWRCTL_PDN_DAC1_MASK (1 << CS42888_PWRCTL_PDN_DAC1_SHIFT) +#define CS42888_PWRCTL_PDN_DAC1 (1 << CS42888_PWRCTL_PDN_DAC1_SHIFT) +#define CS42888_PWRCTL_PDN_SHIFT 0 +#define CS42888_PWRCTL_PDN_MASK (1 << CS42888_PWRCTL_PDN_SHIFT) +#define CS42888_PWRCTL_PDN (1 << CS42888_PWRCTL_PDN_SHIFT) + +/* Functional Mode (Address 03h) */ +#define CS42888_FUNCMOD_DAC_FM_SHIFT 6 +#define CS42888_FUNCMOD_DAC_FM_WIDTH 2 +#define CS42888_FUNCMOD_DAC_FM_MASK (((1 << CS42888_FUNCMOD_DAC_FM_WIDTH) - 1) << CS42888_FUNCMOD_DAC_FM_SHIFT) +#define CS42888_FUNCMOD_DAC_FM(v) ((v) << CS42888_FUNCMOD_DAC_FM_SHIFT) +#define CS42888_FUNCMOD_ADC_FM_SHIFT 4 +#define CS42888_FUNCMOD_ADC_FM_WIDTH 2 +#define CS42888_FUNCMOD_ADC_FM_MASK (((1 << CS42888_FUNCMOD_ADC_FM_WIDTH) - 1) << CS42888_FUNCMOD_ADC_FM_SHIFT) +#define CS42888_FUNCMOD_ADC_FM(v) ((v) << CS42888_FUNCMOD_ADC_FM_SHIFT) +#define CS42888_FUNCMOD_xC_FM_MASK(x) ((x) ? CS42888_FUNCMOD_DAC_FM_MASK : CS42888_FUNCMOD_ADC_FM_MASK) +#define CS42888_FUNCMOD_xC_FM(x, v) ((x) ? CS42888_FUNCMOD_DAC_FM(v) : CS42888_FUNCMOD_ADC_FM(v)) +#define CS42888_FUNCMOD_MFREQ_SHIFT 1 +#define CS42888_FUNCMOD_MFREQ_WIDTH 3 +#define CS42888_FUNCMOD_MFREQ_MASK (((1 << CS42888_FUNCMOD_MFREQ_WIDTH) - 1) << CS42888_FUNCMOD_MFREQ_SHIFT) +#define CS42888_FUNCMOD_MFREQ_256(s) ((0 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_384(s) ((1 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_512(s) ((2 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_768(s) ((3 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_1024(s) ((4 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) + +#define CS42888_FM_SINGLE 0 +#define CS42888_FM_DOUBLE 1 +#define CS42888_FM_QUAD 2 +#define CS42888_FM_AUTO 3 + +/* Interface Formats (Address 04h) */ +#define CS42888_INTF_FREEZE_SHIFT 7 +#define CS42888_INTF_FREEZE_MASK (1 << CS42888_INTF_FREEZE_SHIFT) +#define CS42888_INTF_FREEZE (1 << CS42888_INTF_FREEZE_SHIFT) +#define CS42888_INTF_AUX_DIF_SHIFT 6 +#define CS42888_INTF_AUX_DIF_MASK (1 << CS42888_INTF_AUX_DIF_SHIFT) +#define CS42888_INTF_AUX_DIF (1 << CS42888_INTF_AUX_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_SHIFT 3 +#define CS42888_INTF_DAC_DIF_WIDTH 3 +#define CS42888_INTF_DAC_DIF_MASK (((1 << CS42888_INTF_DAC_DIF_WIDTH) - 1) << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_LEFTJ (0 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_I2S (1 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_RIGHTJ (2 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_ONELINE_20 (4 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_ONELINE_24 (6 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_TDM (7 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_SHIFT 0 +#define CS42888_INTF_ADC_DIF_WIDTH 3 +#define CS42888_INTF_ADC_DIF_MASK (((1 << CS42888_INTF_ADC_DIF_WIDTH) - 1) << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_LEFTJ (0 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_I2S (1 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_RIGHTJ (2 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_ONELINE_20 (4 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_ONELINE_24 (6 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_TDM (7 << CS42888_INTF_ADC_DIF_SHIFT) + +/* ADC Control & DAC De-Emphasis (Address 05h) */ +#define CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 +#define CS42888_ADCCTL_ADC_HPF_FREEZE_MASK (1 << CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42888_ADCCTL_ADC_HPF_FREEZE (1 << CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42888_ADCCTL_DAC_DEM_SHIFT 5 +#define CS42888_ADCCTL_DAC_DEM_MASK (1 << CS42888_ADCCTL_DAC_DEM_SHIFT) +#define CS42888_ADCCTL_DAC_DEM (1 << CS42888_ADCCTL_DAC_DEM_SHIFT) +#define CS42888_ADCCTL_ADC1_SINGLE_SHIFT 4 +#define CS42888_ADCCTL_ADC1_SINGLE_MASK (1 << CS42888_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC1_SINGLE (1 << CS42888_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC2_SINGLE_SHIFT 3 +#define CS42888_ADCCTL_ADC2_SINGLE_MASK (1 << CS42888_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC2_SINGLE (1 << CS42888_ADCCTL_ADC2_SINGLE_SHIFT) + +/* Transition Control (Address 06h) */ +#define CS42888_TXCTL_DAC_SNGVOL_SHIFT 7 +#define CS42888_TXCTL_DAC_SNGVOL_MASK (1 << CS42888_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42888_TXCTL_DAC_SNGVOL (1 << CS42888_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SHIFT 5 +#define CS42888_TXCTL_DAC_SZC_WIDTH 2 +#define CS42888_TXCTL_DAC_SZC_MASK (((1 << CS42888_TXCTL_DAC_SZC_WIDTH) - 1) << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_IC (0 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_ZC (1 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SR (2 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SRZC (3 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_AMUTE_SHIFT 4 +#define CS42888_TXCTL_AMUTE_MASK (1 << CS42888_TXCTL_AMUTE_SHIFT) +#define CS42888_TXCTL_AMUTE (1 << CS42888_TXCTL_AMUTE_SHIFT) +#define CS42888_TXCTL_MUTE_ADC_SP_SHIFT 3 +#define CS42888_TXCTL_MUTE_ADC_SP_MASK (1 << CS42888_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42888_TXCTL_MUTE_ADC_SP (1 << CS42888_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42888_TXCTL_ADC_SNGVOL_SHIFT 2 +#define CS42888_TXCTL_ADC_SNGVOL_MASK (1 << CS42888_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42888_TXCTL_ADC_SNGVOL (1 << CS42888_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SHIFT 0 +#define CS42888_TXCTL_ADC_SZC_MASK (((1 << CS42888_TXCTL_ADC_SZC_WIDTH) - 1) << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_IC (0 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_ZC (1 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SR (2 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SRZC (3 << CS42888_TXCTL_ADC_SZC_SHIFT) + +/* DAC Channel Mute (Address 07h) */ +#define CS42888_DACMUTE_AOUT(n) (0x1 << n) +#define CS42888_DACMUTE_ALL 0xff + +/* Status Control (Address 18h)*/ +#define CS42888_STATUSCTL_INI_SHIFT 2 +#define CS42888_STATUSCTL_INI_WIDTH 2 +#define CS42888_STATUSCTL_INI_MASK (((1 << CS42888_STATUSCTL_INI_WIDTH) - 1) << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_ACTIVE_HIGH (0 << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_ACTIVE_LOW (1 << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_OPEN_DRAIN (2 << CS42888_STATUSCTL_INI_SHIFT) + +/* Status (Address 19h)*/ +#define CS42888_STATUS_DAC_CLK_ERR_SHIFT 4 +#define CS42888_STATUS_DAC_CLK_ERR_MASK (1 << CS42888_STATUS_DAC_CLK_ERR_SHIFT) +#define CS42888_STATUS_ADC_CLK_ERR_SHIFT 3 +#define CS42888_STATUS_ADC_CLK_ERR_MASK (1 << CS42888_STATUS_ADC_CLK_ERR_SHIFT) +#define CS42888_STATUS_ADC2_OVFL_SHIFT 1 +#define CS42888_STATUS_ADC2_OVFL_MASK (1 << CS42888_STATUS_ADC2_OVFL_SHIFT) +#define CS42888_STATUS_ADC1_OVFL_SHIFT 0 +#define CS42888_STATUS_ADC1_OVFL_MASK (1 << CS42888_STATUS_ADC1_OVFL_SHIFT) + +/* Status Mask (Address 1Ah) */ +#define CS42888_STATUS_DAC_CLK_ERR_M_SHIFT 4 +#define CS42888_STATUS_DAC_CLK_ERR_M_MASK (1 << CS42888_STATUS_DAC_CLK_ERR_M_SHIFT) +#define CS42888_STATUS_ADC_CLK_ERR_M_SHIFT 3 +#define CS42888_STATUS_ADC_CLK_ERR_M_MASK (1 << CS42888_STATUS_ADC_CLK_ERR_M_SHIFT) +#define CS42888_STATUS_ADC2_OVFL_M_SHIFT 1 +#define CS42888_STATUS_ADC2_OVFL_M_MASK (1 << CS42888_STATUS_ADC2_OVFL_M_SHIFT) +#define CS42888_STATUS_ADC1_OVFL_M_SHIFT 0 +#define CS42888_STATUS_ADC1_OVFL_M_MASK (1 << CS42888_STATUS_ADC1_OVFL_M_SHIFT) + +/* MUTEC Pin Control (Address 1Bh) */ +#define CS42888_MUTEC_MCPOLARITY_SHIFT 1 +#define CS42888_MUTEC_MCPOLARITY_MASK (1 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MCPOLARITY_ACTIVE_LOW (0 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MCPOLARITY_ACTIVE_HIGH (1 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MUTEC_ACTIVE_SHIFT 0 +#define CS42888_MUTEC_MUTEC_ACTIVE_MASK (1 << CS42888_MUTEC_MUTEC_ACTIVE_SHIFT) +#define CS42888_MUTEC_MUTEC_ACTIVE (1 << CS42888_MUTEC_MUTEC_ACTIVE_SHIFT) +#endif /* _CS42888_H */ -- 1.8.4

6 8

[alsa-devel] About snd_dmaengine_pcm_trigger()
by Kuninori Morimoto 12 Mar '14

12 Mar '14

Hi Mark, Lars and all I need your opinion Now, I'm working for ${LINUX}/sound/soc/sh/fsi and ${LINUX}/sound/soc/sh/rcar drirvers. These drivers are supporting DMAEngine transfer, but it is using original DMAEngine method, not using snd_dmaengine_pcm_trigger() or snd_dmaengine_pcm_register(). I was requested to use it from Mark. The reason why I couldn't use it was our DMAEngine didn't have "cyclic" tranfer support which is used on dmaengine_pcm_prepare_and_submit() But now, I created cyclic support on our DMA driver (on my local PC at this point) Then, I noticed our drivers still can't use snd_dmaengine_pcm_xxx() method. 1st, our device needs PIO tranfer support too. Current PIO/DMA transfer are sharing many methods/functions. Code will become difficult to read if it is separated forcibly. (to using snd_dmaengine_pcm_xxx()) 2nd, our device DMA ON/OFF timing has relation ship to other register settings. Unfortunately, our device is picky, it needs like this HW init setting -> DMA ON -> HW start setting It can be solved if snd_dmaengine_pcm_xxx() has callbacks, but many callbacks are needed... 3rd, our device needs special method on snd_pcm_ops int the future. From my point of view, our driver can replace to use "cyclic" DMA transfer instead of current original DMAEngine method. I guess, it is not difficult. But, using snd_dmaengine_pcm_xxx() is difficult. But, what do you think ?

3 10

[alsa-devel] [PATCH] ak4642: show error if register write fails
by Ben Dooks 11 Mar '14

11 Mar '14

The calls to snd_soc_update_bits() and snd_soc_write() return error codes if the calls fails. The ak4642 driver discards these errors which means that the user is not informed if there is an issue with the bus providing the register access. Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> --- sound/soc/codecs/ak4642.c | 62 ++++++++++++++++++++++++++++++++++------------- 1 file changed, 45 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 4717fa9..15795f6 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -225,6 +225,34 @@ static const struct reg_default ak4648_reg[] = { { 36, 0x00 }, { 37, 0x88 }, { 38, 0x88 }, { 39, 0x08 }, }; +/* wrapper functions to show any errors to updating register values */ + +static inline int ak4642_update_bits(struct snd_soc_codec *codec, + unsigned int reg, + unsigned int mask, unsigned int val) +{ + int ret = snd_soc_update_bits(codec, reg, mask, val); + + if (ret < 0) { + pr_info("%s: error %d writing %04x (%08x, mask %08x)\n", + codec->name, ret, reg, mask, val); + } + + return ret; +} + +static inline int ak4642_soc_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + int ret = snd_soc_write(codec, reg, value); + + if (ret < 0) + pr_info("%s: error %d writing %04x to %08x\n", + codec->name, ret, reg, value); + + return ret; +} + static int ak4642_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -242,8 +270,8 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p97. */ - snd_soc_write(codec, L_IVC, 0x91); /* volume */ - snd_soc_write(codec, R_IVC, 0x91); /* volume */ + ak4642_soc_write(codec, L_IVC, 0x91); /* volume */ + ak4642_soc_write(codec, R_IVC, 0x91); /* volume */ } else { /* * start stereo input @@ -258,11 +286,11 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p94. */ - snd_soc_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0); - snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); - snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); - snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL); - snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR); + ak4642_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0); + ak4642_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); + ak4642_soc_write(codec, ALC_CTL1, ALC | LMTH0); + ak4642_update_bits(codec, PW_MGMT1, PMADL, PMADL); + ak4642_update_bits(codec, PW_MGMT3, PMADR, PMADR); } return 0; @@ -277,9 +305,9 @@ static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, if (is_play) { } else { /* stop stereo input */ - snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0); - snd_soc_update_bits(codec, PW_MGMT3, PMADR, 0); - snd_soc_update_bits(codec, ALC_CTL1, ALC, 0); + ak4642_update_bits(codec, PW_MGMT1, PMADL, 0); + ak4642_update_bits(codec, PW_MGMT3, PMADR, 0); + ak4642_update_bits(codec, ALC_CTL1, ALC, 0); } } @@ -312,7 +340,7 @@ static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, default: return -EINVAL; } - snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll); + ak4642_update_bits(codec, MD_CTL1, PLL_MASK, pll); return 0; } @@ -339,8 +367,8 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) default: return -EINVAL; } - snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data); - snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); + ak4642_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data); + ak4642_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); /* format type */ data = 0; @@ -357,7 +385,7 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) default: return -EINVAL; } - snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data); + ak4642_update_bits(codec, MD_CTL1, DIF_MASK, data); return 0; } @@ -411,7 +439,7 @@ static int ak4642_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate); + ak4642_update_bits(codec, MD_CTL2, FS_MASK, rate); return 0; } @@ -421,10 +449,10 @@ static int ak4642_set_bias_level(struct snd_soc_codec *codec, { switch (level) { case SND_SOC_BIAS_OFF: - snd_soc_write(codec, PW_MGMT1, 0x00); + ak4642_soc_write(codec, PW_MGMT1, 0x00); break; default: - snd_soc_update_bits(codec, PW_MGMT1, PMVCM, PMVCM); + ak4642_update_bits(codec, PW_MGMT1, PMVCM, PMVCM); break; } codec->dapm.bias_level = level; -- 1.9.0

4 9

Re: [alsa-devel] Zoom R16
by Jason Mancine 11 Mar '14

11 Mar '14

Can't seem to compile 3.11 or 3.12 for the distro I am using (AVLInux)... tested it in Ubuntu Studio 13.10 and the quirk didn't seem to have any effect. The output of lsusb is the same, but the device doesn't show up as a sound card. On Sun, Nov 24, 2013 at 9:18 AM, Daniel Mack <daniel(a)zonque.org> wrote: > On Sun, Nov 24, 2013 at 09:03:47AM -0500, Jason Mancine wrote: > > Thanks, I will give it a try. By recent kernel do you mean latest > mainline > > or should my distro's 3.6.11 be new enough? > > Anything >= 3.11 should do. > > In further replies, please do not drop the alsa-devel mailing list. > > > Thanks, > Daniel >

3 15

[alsa-devel] [PATCH] ALSA: sis7019: Simplify dependencies
by Jean Delvare 11 Mar '14

11 Mar '14

Kconfig symbol X86_32 was introduced in October 2005, it's about time to use it :-) Signed-off-by: Jean Delvare <jdelvare(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/pci/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-3.14-rc6.orig/sound/pci/Kconfig 2014-03-11 11:29:10.346528331 +0100 +++ linux-3.14-rc6/sound/pci/Kconfig 2014-03-11 14:23:58.996511875 +0100 @@ -797,7 +797,7 @@ config SND_RME9652 config SND_SIS7019 tristate "SiS 7019 Audio Accelerator" - depends on X86 && !X86_64 + depends on X86_32 select SND_AC97_CODEC select ZONE_DMA help -- Jean Delvare SUSE L3 Support

2 1

[alsa-devel] [PATCH] ALSA: cs5535audio: Also needed on MIPS
by Jean Delvare 11 Mar '14

11 Mar '14

The CS5536 companion chipset is not only used on 32-bit x86 systems as I originally thought, it is also used on MIPS Loongson/Lemote 2 systems. So let the snd-cs5535audio driver be built on MIPS too. Signed-off-by: Jean Delvare <jdelvare(a)suse.de> Cc: Jaya Kumar <jayakumar.alsa(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Ralf Baechle <ralf(a)linux-mips.org> --- Takashi: the powerpc reference was irrelevant as the defconfig file I mentioned includes all possible sound drivers. Ralf: do you happen to know if any other CS55xx chip was used on MIPS systems? Or any non-x86 systems for that matter. sound/pci/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-3.14-rc6.orig/sound/pci/Kconfig 2014-03-10 21:09:12.142069639 +0100 +++ linux-3.14-rc6/sound/pci/Kconfig 2014-03-11 11:29:10.346528331 +0100 @@ -286,7 +286,7 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" - depends on X86_32 || COMPILE_TEST + depends on X86_32 || MIPS || COMPILE_TEST select SND_PCM select SND_AC97_CODEC help -- Jean Delvare SUSE L3 Support

2 1

[alsa-devel] [PATCH] ALSA: cs553*: Fix dependencies
by Jean Delvare 11 Mar '14

11 Mar '14

The CS5530, CS5535 and CS5536 chipsets are companions of the Geode series of processors, which are 32-bit x86 processors. So the snd-cs5530 and snd-cs5535audio drivers are only needed on this architecture, except for build testing purpose. Signed-off-by: Jean Delvare <jdelvare(a)suse.de> Cc: Jaya Kumar <jayakumar.alsa(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) --- linux-3.14-rc6.orig/sound/pci/Kconfig 2014-02-09 16:53:07.699734368 +0100 +++ linux-3.14-rc6/sound/pci/Kconfig 2014-03-10 20:52:21.908690341 +0100 @@ -276,7 +276,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" - depends on ISA_DMA_API + depends on ISA_DMA_API && (X86_32 || COMPILE_TEST) select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. @@ -286,6 +286,7 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" + depends on X86_32 || COMPILE_TEST select SND_PCM select SND_AC97_CODEC help -- Jean Delvare SUSE L3 Support

2 3

[alsa-devel] [PATCH] sound: sound_core: Reduce checkpatch noise
by Joe Perches 11 Mar '14

11 Mar '14

Mostly whitespace cleanups (spaces around operators, remove trailing spaces and unnecessary blank lines, etc) and o Reduce indent of switch/case labels o Use pr_<level> for printks, add pr_fmt o Move braces to appropriate locations o Wrap arguments to 80 columns o Remove unnecessary braces Mostly done with checkpatch --fix and some typing. Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/sound_core.c | 183 ++++++++++++++++++++++++----------------------------- 1 file changed, 82 insertions(+), 101 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 11e953a..952ec83 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -4,6 +4,8 @@ * is used OSS or emulation of it. */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + /* * First, the common part. */ @@ -68,7 +70,7 @@ module_exit(cleanup_soundcore); #ifdef CONFIG_SOUND_OSS_CORE /* * OSS sound core handling. Breaks out sound functions to submodules - * + * * Author: Alan Cox <alan(a)lxorguk.ukuu.org.uk> * * Fixes: @@ -80,9 +82,9 @@ module_exit(cleanup_soundcore); * 2 of the License, or (at your option) any later version. * * -------------------- - * + * * Top level handler for the sound subsystem. Various devices can - * plug into this. The fact they don't all go via OSS doesn't mean + * plug into this. The fact they don't all go via OSS doesn't mean * they don't have to implement the OSS API. There is a lot of logic * to keeping much of the OSS weight out of the code in a compatibility * module, but it's up to the driver to rember to load it... @@ -111,8 +113,7 @@ module_exit(cleanup_soundcore); #define SOUND_STEP 16 -struct sound_unit -{ +struct sound_unit { int unit_minor; const struct file_operations *unit_fops; struct sound_unit *next; @@ -151,15 +152,14 @@ extern int msnd_pinnacle_init(void); #ifdef CONFIG_SOUND_OSS_CORE_PRECLAIM static int preclaim_oss = 1; #else -static int preclaim_oss = 0; +static int preclaim_oss; #endif module_param(preclaim_oss, int, 0444); static int soundcore_open(struct inode *, struct file *); -static const struct file_operations soundcore_fops = -{ +static const struct file_operations soundcore_fops = { /* We must have an owner or the module locking fails */ .owner = THIS_MODULE, .open = soundcore_open, @@ -171,72 +171,72 @@ static const struct file_operations soundcore_fops = * join into it. Called with the lock asserted */ -static int __sound_insert_unit(struct sound_unit * s, struct sound_unit **list, const struct file_operations *fops, int index, int low, int top) +static int __sound_insert_unit(struct sound_unit *s, struct sound_unit **list, + const struct file_operations *fops, int index, + int low, int top) { - int n=low; + int n = low; if (index < 0) { /* first free */ - while (*list && (*list)->unit_minor<n) - list=&((*list)->next); + while (*list && (*list)->unit_minor < n) + list = &((*list)->next); - while(n<top) - { + while (n < top) { /* Found a hole ? */ - if(*list==NULL || (*list)->unit_minor>n) + if (*list == NULL || (*list)->unit_minor > n) break; - list=&((*list)->next); - n+=SOUND_STEP; + list = &((*list)->next); + n += SOUND_STEP; } - if(n>=top) + if (n >= top) return -ENOENT; } else { - n = low+(index*16); + n = low + (index * 16); while (*list) { - if ((*list)->unit_minor==n) + if ((*list)->unit_minor == n) return -EBUSY; - if ((*list)->unit_minor>n) + if ((*list)->unit_minor > n) break; - list=&((*list)->next); + list = &((*list)->next); } - } - + } + /* * Fill it in */ - - s->unit_minor=n; - s->unit_fops=fops; - + + s->unit_minor = n; + s->unit_fops = fops; + /* * Link it */ - - s->next=*list; - *list=s; - - + + s->next = *list; + *list = s; + + return n; } /* * Remove a node from the chain. Called with the lock asserted */ - -static struct sound_unit *__sound_remove_unit(struct sound_unit **list, int unit) + +static struct sound_unit *__sound_remove_unit(struct sound_unit **list, + int unit) { - while(*list) - { - struct sound_unit *p=*list; - if(p->unit_minor==unit) - { - *list=p->next; + while (*list) { + struct sound_unit *p = *list; + if (p->unit_minor == unit) { + *list = p->next; return p; } - list=&(p->next); + list = &(p->next); } - printk(KERN_ERR "Sound device %d went missing!\n", unit); + pr_err("Sound device %d went missing!\n", unit); return NULL; } @@ -251,7 +251,10 @@ static DEFINE_SPINLOCK(sound_loader_lock); * list. Acquires locks as needed */ -static int sound_insert_unit(struct sound_unit **list, const struct file_operations *fops, int index, int low, int top, const char *name, umode_t mode, struct device *dev) +static int sound_insert_unit(struct sound_unit **list, + const struct file_operations *fops, int index, + int low, int top, const char *name, umode_t mode, + struct device *dev) { struct sound_unit *s = kmalloc(sizeof(*s), GFP_KERNEL); int r; @@ -263,7 +266,7 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati retry: r = __sound_insert_unit(s, list, fops, index, low, top); spin_unlock(&sound_loader_lock); - + if (r < 0) goto fail; else if (r < SOUND_STEP) @@ -305,7 +308,7 @@ fail: * completed the removal before their file operations become * invalid. */ - + static void sound_remove_unit(struct sound_unit **list, int unit) { struct sound_unit *p; @@ -349,7 +352,7 @@ static struct sound_unit *chains[SOUND_STEP]; * register_sound_special_device - register a special sound node * @fops: File operations for the driver * @unit: Unit number to allocate - * @dev: device pointer + * @dev: device pointer * * Allocate a special sound device by minor number from the sound * subsystem. @@ -357,7 +360,6 @@ static struct sound_unit *chains[SOUND_STEP]; * Return: The allocated number is returned on success. On failure, * a negative error code is returned. */ - int register_sound_special_device(const struct file_operations *fops, int unit, struct device *dev) { @@ -367,69 +369,65 @@ int register_sound_special_device(const struct file_operations *fops, int unit, char _name[16]; switch (chain) { - case 0: + case 0: name = "mixer"; break; - case 1: + case 1: name = "sequencer"; if (unit >= SOUND_STEP) goto __unknown; max_unit = unit + 1; break; - case 2: + case 2: name = "midi"; break; - case 3: + case 3: name = "dsp"; break; - case 4: + case 4: name = "audio"; break; - case 5: + case 5: name = "dspW"; break; - case 8: + case 8: name = "sequencer2"; if (unit >= SOUND_STEP) goto __unknown; max_unit = unit + 1; break; - case 9: + case 9: name = "dmmidi"; break; - case 10: + case 10: name = "dmfm"; break; - case 12: + case 12: name = "adsp"; break; - case 13: + case 13: name = "amidi"; break; - case 14: + case 14: name = "admmidi"; break; - default: - { - __unknown: - sprintf(_name, "unknown%d", chain); - if (unit >= SOUND_STEP) - strcat(_name, "-"); - name = _name; - } +__unknown: + default: + sprintf(_name, "unknown%d", chain); + if (unit >= SOUND_STEP) + strcat(_name, "-"); + name = _name; break; } return sound_insert_unit(&chains[chain], fops, -1, unit, max_unit, name, S_IRUSR | S_IWUSR, dev); } - EXPORT_SYMBOL(register_sound_special_device); int register_sound_special(const struct file_operations *fops, int unit) { return register_sound_special_device(fops, unit, NULL); } - EXPORT_SYMBOL(register_sound_special); /** @@ -443,13 +441,11 @@ EXPORT_SYMBOL(register_sound_special); * Return: On success, the allocated number is returned. On failure, * a negative error code is returned. */ - int register_sound_mixer(const struct file_operations *fops, int dev) { return sound_insert_unit(&chains[0], fops, dev, 0, 128, "mixer", S_IRUSR | S_IWUSR, NULL); } - EXPORT_SYMBOL(register_sound_mixer); /** @@ -469,14 +465,13 @@ int register_sound_midi(const struct file_operations *fops, int dev) return sound_insert_unit(&chains[2], fops, dev, 2, 130, "midi", S_IRUSR | S_IWUSR, NULL); } - EXPORT_SYMBOL(register_sound_midi); /* * DSP's are registered as a triple. Register only one and cheat * in open - see below. */ - + /** * register_sound_dsp - register a DSP device * @fops: File operations for the driver @@ -491,13 +486,11 @@ EXPORT_SYMBOL(register_sound_midi); * Return: On success, the allocated number is returned. On failure, * a negative error code is returned. */ - int register_sound_dsp(const struct file_operations *fops, int dev) { return sound_insert_unit(&chains[3], fops, dev, 3, 131, "dsp", S_IWUSR | S_IRUSR, NULL); } - EXPORT_SYMBOL(register_sound_dsp); /** @@ -508,13 +501,10 @@ EXPORT_SYMBOL(register_sound_dsp); * register_sound_special(). The unit passed is the return value from * the register function. */ - - void unregister_sound_special(int unit) { sound_remove_unit(&chains[unit % SOUND_STEP], unit); } - EXPORT_SYMBOL(unregister_sound_special); /** @@ -524,12 +514,10 @@ EXPORT_SYMBOL(unregister_sound_special); * Release a sound device that was allocated with register_sound_mixer(). * The unit passed is the return value from the register function. */ - void unregister_sound_mixer(int unit) { sound_remove_unit(&chains[0], unit); } - EXPORT_SYMBOL(unregister_sound_mixer); /** @@ -539,12 +527,10 @@ EXPORT_SYMBOL(unregister_sound_mixer); * Release a sound device that was allocated with register_sound_midi(). * The unit passed is the return value from the register function. */ - void unregister_sound_midi(int unit) { sound_remove_unit(&chains[2], unit); } - EXPORT_SYMBOL(unregister_sound_midi); /** @@ -556,25 +542,21 @@ EXPORT_SYMBOL(unregister_sound_midi); * * Both of the allocated units are released together automatically. */ - void unregister_sound_dsp(int unit) { sound_remove_unit(&chains[3], unit); } - - EXPORT_SYMBOL(unregister_sound_dsp); static struct sound_unit *__look_for_unit(int chain, int unit) { struct sound_unit *s; - - s=chains[chain]; - while(s && s->unit_minor <= unit) - { - if(s->unit_minor==unit) + + s = chains[chain]; + while (s && s->unit_minor <= unit) { + if (s->unit_minor == unit) return s; - s=s->next; + s = s->next; } return NULL; } @@ -586,14 +568,13 @@ static int soundcore_open(struct inode *inode, struct file *file) struct sound_unit *s; const struct file_operations *new_fops = NULL; - chain=unit&0x0F; - if(chain==4 || chain==5) /* dsp/audio/dsp16 */ - { - unit&=0xF0; - unit|=3; - chain=3; + chain = unit & 0x0F; + if (chain == 4 || chain == 5) { /* dsp/audio/dsp16 */ + unit &= 0xF0; + unit |= 3; + chain = 3; } - + spin_lock(&sound_loader_lock); s = __look_for_unit(chain, unit); if (s) @@ -636,7 +617,7 @@ static int soundcore_open(struct inode *inode, struct file *file) replace_fops(file, new_fops); if (file->f_op->open) - err = file->f_op->open(inode,file); + err = file->f_op->open(inode, file); return err; } @@ -656,7 +637,7 @@ static int __init init_oss_soundcore(void) { if (preclaim_oss && register_chrdev(SOUND_MAJOR, "sound", &soundcore_fops) == -1) { - printk(KERN_ERR "soundcore: sound device already in use.\n"); + pr_err("sound device already in use\n"); return -EBUSY; }

2 1

[alsa-devel] [PATCH] Revert "ALSA: emu10k1: Fix warning: "CCR" redefined"
by Geert Uytterhoeven 11 Mar '14

11 Mar '14

This reverts commit 83fc3bc09518d42e8f5073e2a65884701dfadf19. sh-specific "CCR" and "CCR2" have been prefixed by "SH_" in commit a5f6ea29f9a918403629f8369ae55fac6b09cb53 ('sh: prefix sh-specific "CCR" and "CCR2" by "SH_"'). Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- include/sound/emu10k1.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index dfb42ca6d043..c46908c1bb3f 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -436,8 +436,6 @@ #define CCCA_CURRADDR_MASK 0x00ffffff /* Current address of the selected channel */ #define CCCA_CURRADDR 0x18000008 -/* undefine CCR to avoid conflict with the definition for SH */ -#undef CCR #define CCR 0x09 /* Cache control register */ #define CCR_CACHEINVALIDSIZE 0x07190009 #define CCR_CACHEINVALIDSIZE_MASK 0xfe000000 /* Number of invalid samples cache for this channel */ -- 1.7.9.5

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v3.14
by Mark Brown 11 Mar '14

11 Mar '14

The following changes since commit fa389e220254c69ffae0d403eac4146171062d08: Linux 3.14-rc6 (2014-03-09 19:41:57 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-rc6 for you to fetch changes up to 7f35afd44b14a39307757629ebc0a199aade52d9: Merge remote-tracking branches 'asoc/fix/88pm860', 'asoc/fix/omap' and 'asoc/fix/si476x' into asoc-linus (2014-03-10 11:30:56 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.14 A few things here: - Avoid memory leaks in error cases with DPCM, this code has never been that well tested in mainline due to the lack of mainline drivers but we now have one queued for the merge window! - Fix the N810 audio driver to load when booted with DT since the platform was converted to DT during the merge window. - Fixes for initialisation of some MFD drivers that are probably unused in mainline ---------------------------------------------------------------- Aaro Koskinen (1): ASoC: n810: fix init with DT boot Lars-Peter Clausen (2): ASoC: si476x: Fix IO setup ASoC: 88pm860: Fix IO setup Mark Brown (2): Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Merge remote-tracking branches 'asoc/fix/88pm860', 'asoc/fix/omap' and 'asoc/fix/si476x' into asoc-linus Patrick Lai (1): ASoC: pcm: free path list before exiting from error conditions sound/soc/codecs/88pm860x-codec.c | 3 +++ sound/soc/codecs/si476x.c | 2 +- sound/soc/omap/n810.c | 4 +++- sound/soc/soc-pcm.c | 3 +++ 4 files changed, 10 insertions(+), 2 deletions(-)

2 1

[alsa-devel] Playstation Eye won't accept rate greater than 16k, 48k device.
by Benjamin Gorman 10 Mar '14

10 Mar '14

Hello, I'm working with a playstation eye and I'm attempting to grab the raw audio in order to perform some processing using it's 4 channel microphone array. It's listed by the vendor as being a 48khz device. Using Audacity on OSX I can record 4 channels at 44,100hz. However, under linux (raspberry pi running debian) I can only record in 16khz using ALSA. arecord -vv /dev/null -r 16000 -f S16_LE -c 4 -D iec958:CARD=CameraB409241,DEV=0 /dev/null < /dev/null Any higher rate won't be accepted. I tried this using the ALSA C API and also under the PortAudio API. Both without luck. Any ideas on how to patch the driver in order to allow for 44100hz recording? The device is clearly capable of it. Regards, Benjamin The University of Dundee is a registered Scottish Charity, No: SC015096

1 0

[alsa-devel] [PATCH] ASOC: spear: spdif_out: Fix mute control
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

For controls registers with snd_soc_add_dai_controls snd_kcontrol_chip() returns a pointer to the DAI, not to the CODEC. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/spear/spdif_out.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index fe99f46..19cca04 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -213,10 +213,7 @@ static int spdif_digital_mute(struct snd_soc_dai *dai, int mute) static int spdif_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct snd_soc_card *card = codec->card; - struct snd_soc_pcm_runtime *rtd = card->rtd; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); ucontrol->value.integer.value[0] = host->saved_params.mute; @@ -226,10 +223,7 @@ static int spdif_mute_get(struct snd_kcontrol *kcontrol, static int spdif_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct snd_soc_card *card = codec->card; - struct snd_soc_pcm_runtime *rtd = card->rtd; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); if (host->saved_params.mute == ucontrol->value.integer.value[0]) -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: davinci-evm: Register machine level DAPM elements with the card
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

Machine level DAPM widgets and routes should be registered in the card's DAPM context, rather than in the CODEC's context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/davinci/davinci-evm.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 5e3bc3c..8d10244 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -95,23 +95,24 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Logic for a aic3x as connected on a davinci-evm */ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_card *card = rtd->card; struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; struct device_node *np = codec->card->dev->of_node; int ret; /* Add davinci-evm specific widgets */ - snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, + snd_soc_dapm_new_controls(&card->dapm, aic3x_dapm_widgets, ARRAY_SIZE(aic3x_dapm_widgets)); if (np) { - ret = snd_soc_of_parse_audio_routing(codec->card, - "ti,audio-routing"); + ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing"); if (ret) return ret; } else { /* Set up davinci-evm specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(&card->dapm, audio_map, + ARRAY_SIZE(audio_map)); } /* not connected */ @@ -120,10 +121,10 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "HPRCOM"); /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin(dapm, "Line Out"); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_enable_pin(dapm, "Line In"); + snd_soc_dapm_enable_pin(&card->dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(&card->dapm, "Line Out"); + snd_soc_dapm_enable_pin(&card->dapm, "Mic Jack"); + snd_soc_dapm_enable_pin(&card->dapm, "Line In"); return 0; } -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: migor: Convert to table based DAPM setup
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/sh/migor.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 5014a88..c58c252 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -136,19 +136,6 @@ static const struct snd_soc_dapm_route audio_map[] = { { "Mic Bias", NULL, "External Microphone" }, }; -static int migor_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, migor_dapm_widgets, - ARRAY_SIZE(migor_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - /* migor digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link migor_dai = { .name = "wm8978", @@ -158,7 +145,6 @@ static struct snd_soc_dai_link migor_dai = { .platform_name = "siu-pcm-audio", .codec_name = "wm8978.0-001a", .ops = &migor_dai_ops, - .init = migor_dai_init, }; /* migor audio machine driver */ @@ -167,6 +153,11 @@ static struct snd_soc_card snd_soc_migor = { .owner = THIS_MODULE, .dai_link = &migor_dai, .num_links = 1, + + .dapm_widgets = migor_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(migor_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *migor_snd_device; -- 1.8.0

2 1

[alsa-devel] integration into ASoC
by Maxime Ripard 10 Mar '14

10 Mar '14

Hi Liam, Mark, I have a sound IP that is part of an SoC that I'm willing to write a driver for. This IP is made of a few registers to control the sampling rate, if we're using mono/stereo, plus two fifos, one for playback, one for capture, that can be seed with data. The data are then taken, go through a DAC, and the outer interface of the IP are directly analog signals (so the DAC/ADC are directly in the SoC, and the only interface we have is plain registers). From what I understood from ASoC, you have mostly three components, the DAI, the codec and the platform that plumbers the two first together. Here, my understanding is that it's pretty much the whole three in a single IP. Should such a hardware block be handled into ASoC, and if yes, how? If not, which other framework should be used? Thanks! Maxime -- Maxime Ripard, Free Electrons Embedded Linux, Kernel and Android engineering http://free-electrons.com

3 7

[alsa-devel] [PATCH] add Steinberg UR44 USB Audio Interface support to quirks-table.
by Cyrus Harmon 10 Mar '14

10 Mar '14

* The patch is based on Clemens Ladisch's patch to support the UR22 (http://linuxaudio.org/mailarchive/lad/2013/2/23/197831), with the device ID modified for the UR44. The patch was tested with kernel versions 3.12.11 and 3.12.12 and a PREEMPT RT-patched 3.12.12-rt kernel. Basic recording and playback appear to work. * NB: On one of the systems that I used to test the patch, I was never able to achieve xrun-free JACK performance at -p256 -n3 as I would invariably see xruns roughly once every minute. But this is a new box (Intel haswell NUC) and it seems there are many possible sources of the unacceptable latency. On an older i3 box, the performance was much better. Signed-off-by: Cyrus Harmon <cyrus(a)bobobeach.com> --- sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f652b10..987ca01 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -436,6 +436,36 @@ YAMAHA_DEVICE(0x105d, NULL), } } }, +{ + USB_DEVICE(0x0499, 0x150e), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Yamaha", */ + /* .product_name = "Steinberg UR44", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_YAMAHA + }, + { + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), -- 1.8.3.2

1 0

[alsa-devel] [PATCHv3 0/3] Simplify the CODEC ASoC probe code.
by Xiubo Li 10 Mar '14

10 Mar '14

5 12

[alsa-devel] [PATCH 0/9] Update FLL calculations on Arizona devices
by Charles Keepax 10 Mar '14

10 Mar '14

Newer versions of the Arizona IP have several differences in the FLL configuration. The reference and synchroniser paths have slightly difference FRATIO encodings and the expanded encoding on the reference path should be used to avoid integer mode operation. This series of patches does some refactoring on the FLL configuration code, previously calculation of the FLL configuration was done before we knew whether this would be applied to reference or synchroniser path. These patches add some checks to make sure the requested clock can be accomodated and then does the calculations once we knew where they will be applied. Additionally, we add support for the encodings on the newer IP. Thanks, Charles Charles Keepax (9): ASoC: arizona: An OUTDIV of 1 is not valid, avoid this ASoC: arizona: Add defines for FLL configuration constants ASoC: arizona: Move set of OUTDIV in to arizona_apply_fll ASoC: arizona: Move calculation of FLL configuration ASoC: arizona: Don't pass Fout into arizona_calc_fll ASoC: arizona: Calculate OUTDIV first ASoC: arizona: Calculate FLL gain last mfd: arizona: Add support for new fratio encoding ASoC: arizona: Support new fratio encoding on the wm5110 rev D include/linux/mfd/arizona/registers.h | 6 +- sound/soc/codecs/arizona.c | 254 ++++++++++++++++++++++----------- 2 files changed, 171 insertions(+), 89 deletions(-) -- 1.7.2.5

2 13

[alsa-devel] [PATCH v2 1/3] ASoC: Consolidate snd_soc_register_dai() and snd_soc_register_dais()
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

snd_soc_register_dais() has basically the same code as snd_soc_register_dai(), but running in a loop. The only difference is that snd_soc_register_dai() calls fmt_single_name() to generate the DAIs name and snd_soc_register_dais() calls fmt_multiple_name(). This patch pushes the check in __snd_soc_register_component() which decides whether to call snd_soc_register_dai() or snd_soc_register_dais() to snd_soc_register_dais() to decide which naming scheme to use. This allows us to remove snd_soc_register_dai(). The patch also updates snd_soc_register_dais() to unregister every DAI it finds for the component rather than trying to unregister one DAI for each DAI that was registered. Both have the same result since there won't be more DAIs than what have been registered, but the former is easier to implement. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Fixed compile error that slipped into patch 1, but was fixed in patch 3 --- sound/soc/soc-core.c | 154 ++++++++++++++------------------------------------- 1 file changed, 42 insertions(+), 112 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8ddb15c..ae2b7cc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3897,100 +3897,47 @@ static inline char *fmt_multiple_name(struct device *dev, } /** - * snd_soc_register_dai - Register a DAI with the ASoC core + * snd_soc_unregister_dai - Unregister DAIs from the ASoC core * - * @component: The component the DAIs are registered for - * @dai_drv: DAI driver to use for the DAIs + * @component: The component for which the DAIs should be unregistered */ -static int snd_soc_register_dai(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv) +static void snd_soc_unregister_dais(struct snd_soc_component *component) { - struct device *dev = component->dev; - struct snd_soc_codec *codec; - struct snd_soc_dai *dai; - - dev_dbg(dev, "ASoC: dai register %s\n", dev_name(dev)); - - dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL); - if (dai == NULL) - return -ENOMEM; - - /* create DAI component name */ - dai->name = fmt_single_name(dev, &dai->id); - if (dai->name == NULL) { - kfree(dai); - return -ENOMEM; - } - - dai->component = component; - dai->dev = dev; - dai->driver = dai_drv; - dai->dapm.dev = dev; - if (!dai->driver->ops) - dai->driver->ops = &null_dai_ops; + struct snd_soc_dai *dai, *_dai; mutex_lock(&client_mutex); + list_for_each_entry_safe(dai, _dai, &dai_list, list) { + if (dai->dev != component->dev) + continue; - list_for_each_entry(codec, &codec_list, list) { - if (codec->dev == dev) { - dev_dbg(dev, "ASoC: Mapped DAI %s to CODEC %s\n", - dai->name, codec->name); - dai->codec = codec; - break; - } - } - - if (!dai->codec) - dai->dapm.idle_bias_off = 1; - - list_add(&dai->list, &dai_list); - - mutex_unlock(&client_mutex); - - dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name); - - return 0; -} - -/** - * snd_soc_unregister_dai - Unregister a DAI from the ASoC core - * - * @dai: DAI to unregister - */ -static void snd_soc_unregister_dai(struct device *dev) -{ - struct snd_soc_dai *dai; + list_del(&dai->list); - list_for_each_entry(dai, &dai_list, list) { - if (dev == dai->dev) - goto found; + dev_dbg(component->dev, "ASoC: Unregistered DAI '%s'\n", + dai->name); + kfree(dai->name); + kfree(dai); } - return; - -found: - mutex_lock(&client_mutex); - list_del(&dai->list); mutex_unlock(&client_mutex); - - dev_dbg(dev, "ASoC: Unregistered DAI '%s'\n", dai->name); - kfree(dai->name); - kfree(dai); } /** - * snd_soc_register_dais - Register multiple DAIs with the ASoC core + * snd_soc_register_dais - Register a DAI with the ASoC core * * @component: The component the DAIs are registered for * @dai_drv: DAI driver to use for the DAIs * @count: Number of DAIs + * @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the + * parent's name. */ static int snd_soc_register_dais(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv, size_t count) + struct snd_soc_dai_driver *dai_drv, size_t count, + bool legacy_dai_naming) { struct device *dev = component->dev; struct snd_soc_codec *codec; struct snd_soc_dai *dai; - int i, ret = 0; + unsigned int i; + int ret; dev_dbg(dev, "ASoC: dai register %s #%Zu\n", dev_name(dev), count); @@ -4002,21 +3949,32 @@ static int snd_soc_register_dais(struct snd_soc_component *component, goto err; } - /* create DAI component name */ - dai->name = fmt_multiple_name(dev, &dai_drv[i]); + /* + * Back in the old days when we still had component-less DAIs, + * instead of having a static name, component-less DAIs would + * inherit the name of the parent device so it is possible to + * register multiple instances of the DAI. We still need to keep + * the same naming style even though those DAIs are not + * component-less anymore. + */ + if (count == 1 && legacy_dai_naming) { + dai->name = fmt_single_name(dev, &dai->id); + } else { + dai->name = fmt_multiple_name(dev, &dai_drv[i]); + if (dai_drv[i].id) + dai->id = dai_drv[i].id; + else + dai->id = i; + } if (dai->name == NULL) { kfree(dai); - ret = -EINVAL; + ret = -ENOMEM; goto err; } dai->component = component; dai->dev = dev; dai->driver = &dai_drv[i]; - if (dai->driver->id) - dai->id = dai->driver->id; - else - dai->id = i; dai->dapm.dev = dev; if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; @@ -4025,8 +3983,7 @@ static int snd_soc_register_dais(struct snd_soc_component *component, list_for_each_entry(codec, &codec_list, list) { if (codec->dev == dev) { - dev_dbg(dev, - "ASoC: Mapped DAI %s to CODEC %s\n", + dev_dbg(dev, "ASoC: Mapped DAI %s to CODEC %s\n", dai->name, codec->name); dai->codec = codec; break; @@ -4040,33 +3997,18 @@ static int snd_soc_register_dais(struct snd_soc_component *component, mutex_unlock(&client_mutex); - dev_dbg(dai->dev, "ASoC: Registered DAI '%s'\n", dai->name); + dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name); } return 0; err: - for (i--; i >= 0; i--) - snd_soc_unregister_dai(dev); + snd_soc_unregister_dais(component); return ret; } /** - * snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core - * - * @dai: Array of DAIs to unregister - * @count: Number of DAIs - */ -static void snd_soc_unregister_dais(struct device *dev, size_t count) -{ - int i; - - for (i = 0; i < count; i++) - snd_soc_unregister_dai(dev); -} - -/** * snd_soc_register_component - Register a component with the ASoC core * */ @@ -4097,19 +4039,7 @@ __snd_soc_register_component(struct device *dev, cmpnt->dai_drv = dai_drv; cmpnt->num_dai = num_dai; - /* - * snd_soc_register_dai() uses fmt_single_name(), and - * snd_soc_register_dais() uses fmt_multiple_name() - * for dai->name which is used for name based matching - * - * this function is used from cpu/codec. - * allow_single_dai flag can ignore "codec" driver reworking - * since it had been used snd_soc_register_dais(), - */ - if ((1 == num_dai) && allow_single_dai) - ret = snd_soc_register_dai(cmpnt, dai_drv); - else - ret = snd_soc_register_dais(cmpnt, dai_drv, num_dai); + ret = snd_soc_register_dais(cmpnt, dai_drv, num_dai, allow_single_dai); if (ret < 0) { dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret); goto error_component_name; @@ -4164,7 +4094,7 @@ void snd_soc_unregister_component(struct device *dev) return; found: - snd_soc_unregister_dais(dev, cmpnt->num_dai); + snd_soc_unregister_dais(cmpnt); mutex_lock(&client_mutex); list_del(&cmpnt->list); -- 1.8.0

2 3

[alsa-devel] [PATCH 1/3] ASoC: omap-abe-twl6040: No need to register DMIC routes seperatly
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

When using table based DAPM setup there is no need to register DAPM elements for different sub-components separately. The widgets will be registered before the first sub-component is initialized, the routes are only added after the last sub-component has been initialized, meaning everything will be available when it is needed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/omap/omap-abe-twl6040.c | 19 ++++--------------- 1 file changed, 4 insertions(+), 15 deletions(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index ebb1390..5011bfa 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -163,6 +163,10 @@ static const struct snd_soc_dapm_route audio_map[] = { {"AFML", NULL, "Line In"}, {"AFMR", NULL, "Line In"}, + + /* DMIC routing */ + {"DMic", NULL, "Digital Mic"}, + {"Digital Mic", NULL, "Digital Mic1 Bias"}, }; static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) @@ -196,20 +200,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static const struct snd_soc_dapm_route dmic_audio_map[] = { - {"DMic", NULL, "Digital Mic"}, - {"Digital Mic", NULL, "Digital Mic1 Bias"}, -}; - -static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - return snd_soc_dapm_add_routes(dapm, dmic_audio_map, - ARRAY_SIZE(dmic_audio_map)); -} - /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link abe_twl6040_dai_links[] = { { @@ -229,7 +219,6 @@ static struct snd_soc_dai_link abe_twl6040_dai_links[] = { .codec_dai_name = "dmic-hifi", .platform_name = "omap-pcm-audio", .codec_name = "dmic-codec", - .init = omap_abe_dmic_init, .ops = &omap_abe_dmic_ops, }, }; -- 1.8.0

3 11

[alsa-devel] [PATCH v2 0/4] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver
by Jyri Sarha 10 Mar '14

10 Mar '14

Since the RFC version of the patches [1] I have addressed Mark Browns comments [2] as described in my mail here [3]. Besr regards, Jyri [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073289.h… [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073703.html http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073704.html http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073705.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073807.html Jyri Sarha (4): ASoC: tlv320aic31xx: Add basic codec driver implementation ASoC: davinci-evm: Add AM43xx-EPOS-EVM audio support ASoC: davinci: Add SND_AM43XX_SOC_EPOS_EVM build option ASoC: tlv320aic32x4: Sort Makefile in alphabetic order .../bindings/sound/davinci-evm-audio.txt | 9 +- .../devicetree/bindings/sound/tlv320aic31xx.txt | 58 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 9 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 6 +- sound/soc/codecs/tlv320aic31xx.c | 1345 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ sound/soc/davinci/Kconfig | 12 + sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 41 + 10 files changed, 1738 insertions(+), 5 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h -- 1.7.9.5

4 20

[alsa-devel] [PATCH] ASoC: 88pm860: Fix IO setup
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

The 88pm860 is a MFD device and the CODEC driver is using the regmap struct of the parent device, hence automatic IO setup will not work and we need to manually call snd_soc_codec_set_cache_io(). The issue was introduced in commit f9ded3b2e7 ("ASoC: 88pm860x: Use regmap for I/O"). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/88pm860x-codec.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 697d24b..8703244 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1328,6 +1328,9 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x->codec = codec; codec->control_data = pm860x->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (ret) + return ret; for (i = 0; i < 4; i++) { ret = request_threaded_irq(pm860x->irq[i], NULL, -- 1.8.0

2 1

[alsa-devel] [PATCH RFC 0/2] Fix simple-card *-master DT parameter handling
by Jyri Sarha 10 Mar '14

10 Mar '14

The schematics of bitclock-master and frame-master DT parameters should depend on whether they are found from a cpu-dai or codec sub-node. - bitclock-master in cpu-dai node means Codec-Bitclock-Slave - frame-master in cpu-dai node means Codec-Frame-Slave - bitclock-master in codec node means Codec-Bitclock-Master - frame-master in codec node means Codec-Frame-Master For example in a cpu-dai mode bitclock-master parameter should produce SND_SOC_DAIFMT_CBS_* daifmt flags and a codec node SND_SOC_DAIFMT_CBM_* flags. Best regards, Jyri Jyri Sarha (2): ASoC: core: Add is_cpu_dai_node-parameter to snd_soc_of_parse_daifmt() ASoC: simple-card: Take snd_soc_of_parse_daifmt() change in to account sound/soc/generic/simple-card.c | 13 ++++++++----- sound/soc/soc-core.c | 8 +++++++- 2 files changed, 15 insertions(+), 6 deletions(-) -- 1.7.9.5

1 2

[alsa-devel] [PATCH] ASoC: si476x: Fix IO setup
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

The si476x is a MFD device and the CODEC driver is using the regmap struct of the parent device, hence automatic IO setup will not work and we need to manually call snd_soc_codec_set_cache_io(). The issue was introduced commit d6173df35f ("ASoC: si476x: Remove custom register I/O implementation") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/si476x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 52e7cb0..fa2b8e0 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -210,7 +210,7 @@ out: static int si476x_codec_probe(struct snd_soc_codec *codec) { codec->control_data = dev_get_regmap(codec->dev->parent, NULL); - return 0; + return snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); } static struct snd_soc_dai_ops si476x_dai_ops = { -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: sam9g20_wm8731: Convert to table based DAPM setup
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. While we are at it also remove the snd_soc_dapm_enable_pin() in the init callback. Pins are enabled by default. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/atmel/sam9g20_wm8731.c | 20 +++++++------------- 1 file changed, 7 insertions(+), 13 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index f15bff1..174bd54 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -155,25 +155,14 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) return ret; } - /* Add specific widgets */ - snd_soc_dapm_new_controls(dapm, at91sam9g20ek_dapm_widgets, - ARRAY_SIZE(at91sam9g20ek_dapm_widgets)); - /* Set up specific audio path interconnects */ - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - /* not connected */ snd_soc_dapm_nc_pin(dapm, "RLINEIN"); snd_soc_dapm_nc_pin(dapm, "LLINEIN"); -#ifdef ENABLE_MIC_INPUT - snd_soc_dapm_enable_pin(dapm, "Int Mic"); -#else - snd_soc_dapm_nc_pin(dapm, "Int Mic"); +#ifndef ENABLE_MIC_INPUT + snd_soc_dapm_nc_pin(&rtd->card->dapm, "Int Mic"); #endif - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - return 0; } @@ -194,6 +183,11 @@ static struct snd_soc_card snd_soc_at91sam9g20ek = { .dai_link = &at91sam9g20ek_dai, .num_links = 1, .set_bias_level = at91sam9g20ek_set_bias_level, + + .dapm_widgets = at91sam9g20ek_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(at91sam9g20ek_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), }; static int at91sam9g20ek_audio_probe(struct platform_device *pdev) -- 1.8.0

3 2

[alsa-devel] [PATCH] ASoC: simple-card: Fix property name for simple-audio-card, format
by Peter Ujfalusi 10 Mar '14

10 Mar '14

The "format" part of the property name is missing from the code. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7cabcc5c8703..064b404ed252 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -165,7 +165,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, snd_soc_of_parse_card_name(&priv->snd_card, "simple-audio-card,name"); /* get CPU/CODEC common format via simple-audio-card,format */ - priv->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & + priv->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,format") & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); /* off-codec widgets */ -- 1.9.0

3 4

[alsa-devel] [PATCH v4] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver
by Jyri Sarha 10 Mar '14

10 Mar '14

Since the v3 version of the patches: - Dropped "ASoC: tlv320aic32x4: Sort Makefile in alphabetic order" out - Changed DAMP widget "Mic Bias" to "MICBIAS" Best regards, Jyri Jyri Sarha (1): ASoC: tlv320aic31xx: Add basic codec driver implementation .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 9 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic31xx.c | 1343 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ 6 files changed, 1677 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h -- 1.7.9.5

4 14

[alsa-devel] 5.1 sound system works as 2.0 on all kernels newer than 3.2.21
by giperon＠land.ru 10 Mar '14

10 Mar '14

Hi, my 5.1 sound system works as 2.0 on all kernels newer than 3.2.21. Control "Channel Mode" is missing in "Alsamixer". I use the option "3stack-6ch" (module snd-hda-intel, Realtetsk codec ALC882) for kernel 3.2.21. "Alsa-info" for kernels 3.2.21 and 3.13.6 attached.

2 1

[alsa-devel] About ALSA SoC cpu/codec asymmetry
by Kuninori Morimoto 10 Mar '14

10 Mar '14

Hi Mark, Stephen, Lars Now, ALSA SoC is using "component" style, and its image is like this. ALSA SoC - platform - component (as CPU) - codec - component It is still not symmetry, and confusable. I guess this is understandable. ALSA SoC - platform - cpu - component - codec - component If we add "cpu", it needs "cpu name", same as current "codec name" on soc_bind_dai_link(). I guess it can keep compatible if it allows "NULL" name on "cpu name" at this point. (but print warning ?) What do you think ? Best regards --- Kuninori Morimoto

2 5

[alsa-devel] [PATCH] ASoC: simple-card: card name can be option
by Kuninori Morimoto 10 Mar '14

10 Mar '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_card.name is now option on DT case. non-DT case can follow same style, and it is understandable from platform point of view. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7cabcc5..5dd4769 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -287,7 +287,6 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } if (!cinfo->name || - !cinfo->card || !cinfo->codec_dai.name || !cinfo->codec || !cinfo->platform || @@ -296,7 +295,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return -EINVAL; } - priv->snd_card.name = cinfo->card; + priv->snd_card.name = (cinfo->card) ? cinfo->card : cinfo->name; dai_link->name = cinfo->name; dai_link->stream_name = cinfo->name; dai_link->platform_name = cinfo->platform; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/3] ASoC: Consolidate snd_soc_register_dai() and snd_soc_register_dais()
by Lars-Peter Clausen 09 Mar '14

09 Mar '14

snd_soc_register_dais() has basically the same code as snd_soc_register_dai(), but running in a loop. The only difference is that snd_soc_register_dai() calls fmt_single_name() to generate the DAIs name and snd_soc_register_dais() calls fmt_multiple_name(). This patch pushes the check in __snd_soc_register_component() which decides whether to call snd_soc_register_dai() or snd_soc_register_dais() to snd_soc_register_dais() to decide which naming scheme to use. This allows us to remove snd_soc_register_dai(). The patch also updates snd_soc_register_dais() to unregister every DAI it finds for the component rather than trying to unregister one DAI for each DAI that was registered. Both have the same result since there won't be more DAIs than what have been registered, but the former is easier to implement. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Should go on top of topic/pcm --- sound/soc/soc-core.c | 153 ++++++++++++++------------------------------------- 1 file changed, 41 insertions(+), 112 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8ddb15c..2fe0338 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3897,100 +3897,46 @@ static inline char *fmt_multiple_name(struct device *dev, } /** - * snd_soc_register_dai - Register a DAI with the ASoC core + * snd_soc_unregister_dai - Unregister DAIs from the ASoC core * - * @component: The component the DAIs are registered for - * @dai_drv: DAI driver to use for the DAIs + * @component: The component for which the DAIs should be unregistered */ -static int snd_soc_register_dai(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv) +static void snd_soc_unregister_dais(struct snd_soc_component *component) { - struct device *dev = component->dev; - struct snd_soc_codec *codec; - struct snd_soc_dai *dai; - - dev_dbg(dev, "ASoC: dai register %s\n", dev_name(dev)); - - dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL); - if (dai == NULL) - return -ENOMEM; - - /* create DAI component name */ - dai->name = fmt_single_name(dev, &dai->id); - if (dai->name == NULL) { - kfree(dai); - return -ENOMEM; - } - - dai->component = component; - dai->dev = dev; - dai->driver = dai_drv; - dai->dapm.dev = dev; - if (!dai->driver->ops) - dai->driver->ops = &null_dai_ops; + struct snd_soc_dai *dai, *_dai; mutex_lock(&client_mutex); + list_for_each_entry_safe(dai, _dai, &dai_list, list) { + if (dai->dev != component->dev) + continue; - list_for_each_entry(codec, &codec_list, list) { - if (codec->dev == dev) { - dev_dbg(dev, "ASoC: Mapped DAI %s to CODEC %s\n", - dai->name, codec->name); - dai->codec = codec; - break; - } - } - - if (!dai->codec) - dai->dapm.idle_bias_off = 1; - - list_add(&dai->list, &dai_list); - - mutex_unlock(&client_mutex); - - dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name); - - return 0; -} - -/** - * snd_soc_unregister_dai - Unregister a DAI from the ASoC core - * - * @dai: DAI to unregister - */ -static void snd_soc_unregister_dai(struct device *dev) -{ - struct snd_soc_dai *dai; + list_del(&dai->list); - list_for_each_entry(dai, &dai_list, list) { - if (dev == dai->dev) - goto found; + dev_dbg(dev, "ASoC: Unregistered DAI '%s'\n", dai->name); + kfree(dai->name); + kfree(dai); } - return; - -found: - mutex_lock(&client_mutex); - list_del(&dai->list); mutex_unlock(&client_mutex); - - dev_dbg(dev, "ASoC: Unregistered DAI '%s'\n", dai->name); - kfree(dai->name); - kfree(dai); } /** - * snd_soc_register_dais - Register multiple DAIs with the ASoC core + * snd_soc_register_dais - Register a DAI with the ASoC core * * @component: The component the DAIs are registered for * @dai_drv: DAI driver to use for the DAIs * @count: Number of DAIs + * @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the + * parent's name. */ static int snd_soc_register_dais(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv, size_t count) + struct snd_soc_dai_driver *dai_drv, size_t count, + bool legacy_dai_naming) { struct device *dev = component->dev; struct snd_soc_codec *codec; struct snd_soc_dai *dai; - int i, ret = 0; + unsigned int i; + int ret; dev_dbg(dev, "ASoC: dai register %s #%Zu\n", dev_name(dev), count); @@ -4002,21 +3948,32 @@ static int snd_soc_register_dais(struct snd_soc_component *component, goto err; } - /* create DAI component name */ - dai->name = fmt_multiple_name(dev, &dai_drv[i]); + /* + * Back in the old days when we still had component-less DAIs, + * instead of having a static name, component-less DAIs would + * inherit the name of the parent device so it is possible to + * register multiple instances of the DAI. We still need to keep + * the same naming style even though those DAIs are not + * component-less anymore. + */ + if (count == 1 && legacy_dai_naming) { + dai->name = fmt_single_name(dev, &dai->id); + } else { + dai->name = fmt_multiple_name(dev, &dai_drv[i]); + if (dai_drv[i].id) + dai->id = dai_drv[i].id; + else + dai->id = i; + } if (dai->name == NULL) { kfree(dai); - ret = -EINVAL; + ret = -ENOMEM; goto err; } dai->component = component; dai->dev = dev; dai->driver = &dai_drv[i]; - if (dai->driver->id) - dai->id = dai->driver->id; - else - dai->id = i; dai->dapm.dev = dev; if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; @@ -4025,8 +3982,7 @@ static int snd_soc_register_dais(struct snd_soc_component *component, list_for_each_entry(codec, &codec_list, list) { if (codec->dev == dev) { - dev_dbg(dev, - "ASoC: Mapped DAI %s to CODEC %s\n", + dev_dbg(dev, "ASoC: Mapped DAI %s to CODEC %s\n", dai->name, codec->name); dai->codec = codec; break; @@ -4040,33 +3996,18 @@ static int snd_soc_register_dais(struct snd_soc_component *component, mutex_unlock(&client_mutex); - dev_dbg(dai->dev, "ASoC: Registered DAI '%s'\n", dai->name); + dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name); } return 0; err: - for (i--; i >= 0; i--) - snd_soc_unregister_dai(dev); + snd_soc_unregister_dais(component); return ret; } /** - * snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core - * - * @dai: Array of DAIs to unregister - * @count: Number of DAIs - */ -static void snd_soc_unregister_dais(struct device *dev, size_t count) -{ - int i; - - for (i = 0; i < count; i++) - snd_soc_unregister_dai(dev); -} - -/** * snd_soc_register_component - Register a component with the ASoC core * */ @@ -4097,19 +4038,7 @@ __snd_soc_register_component(struct device *dev, cmpnt->dai_drv = dai_drv; cmpnt->num_dai = num_dai; - /* - * snd_soc_register_dai() uses fmt_single_name(), and - * snd_soc_register_dais() uses fmt_multiple_name() - * for dai->name which is used for name based matching - * - * this function is used from cpu/codec. - * allow_single_dai flag can ignore "codec" driver reworking - * since it had been used snd_soc_register_dais(), - */ - if ((1 == num_dai) && allow_single_dai) - ret = snd_soc_register_dai(cmpnt, dai_drv); - else - ret = snd_soc_register_dais(cmpnt, dai_drv, num_dai); + ret = snd_soc_register_dais(cmpnt, dai_drv, num_dai, allow_single_dai); if (ret < 0) { dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret); goto error_component_name; @@ -4164,7 +4093,7 @@ void snd_soc_unregister_component(struct device *dev) return; found: - snd_soc_unregister_dais(dev, cmpnt->num_dai); + snd_soc_unregister_dais(cmpnt); mutex_lock(&client_mutex); list_del(&cmpnt->list); -- 1.8.0

1 2

[alsa-devel] [PATCH v3 0/2] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver
by Jyri Sarha 09 Mar '14

09 Mar '14

Since the v2 version of the patches [1] I have dropped davinci-evm related patches since I plan to start using simple-card for machine driver. I have also addressed Mark Brown's comments [2] as I described in my mail here [3]. Best regards, Jyri [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073809.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073814.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073992.html Jyri Sarha (2): ASoC: tlv320aic32x4: Sort Makefile in alphabetic order ASoC: tlv320aic31xx: Add basic codec driver implementation .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 9 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 6 +- sound/soc/codecs/tlv320aic31xx.c | 1343 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ 6 files changed, 1679 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h -- 1.7.9.5

2 4

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Rename the pointer to davinci dma_params as dma_params
by Peter Ujfalusi 09 Mar '14

09 Mar '14

It is quite confusing to name the pointer to davinci_pcm_dma_params as dma_data. Rename the pointer to avoid confusion. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 44 +++++++++++++++++++-------------------- 1 file changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 35b9b5505f07..b0ae0677f023 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1025,7 +1025,7 @@ nodata: static int davinci_mcasp_probe(struct platform_device *pdev) { - struct davinci_pcm_dma_params *dma_data; + struct davinci_pcm_dma_params *dma_params; struct resource *mem, *ioarea, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; @@ -1094,41 +1094,41 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (dat) mcasp->dat_port = true; - dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; - dma_data->asp_chan_q = pdata->asp_chan_q; - dma_data->ram_chan_q = pdata->ram_chan_q; - dma_data->sram_pool = pdata->sram_pool; - dma_data->sram_size = pdata->sram_size_playback; + dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + dma_params->asp_chan_q = pdata->asp_chan_q; + dma_params->ram_chan_q = pdata->ram_chan_q; + dma_params->sram_pool = pdata->sram_pool; + dma_params->sram_size = pdata->sram_size_playback; if (dat) - dma_data->dma_addr = dat->start; + dma_params->dma_addr = dat->start; else - dma_data->dma_addr = mem->start + pdata->tx_dma_offset; + dma_params->dma_addr = mem->start + pdata->tx_dma_offset; /* Unconditional dmaengine stuff */ - mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = dma_data->dma_addr; + mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = dma_params->dma_addr; res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (res) - dma_data->channel = res->start; + dma_params->channel = res->start; else - dma_data->channel = pdata->tx_dma_channel; + dma_params->channel = pdata->tx_dma_channel; - dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; - dma_data->asp_chan_q = pdata->asp_chan_q; - dma_data->ram_chan_q = pdata->ram_chan_q; - dma_data->sram_pool = pdata->sram_pool; - dma_data->sram_size = pdata->sram_size_capture; + dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; + dma_params->asp_chan_q = pdata->asp_chan_q; + dma_params->ram_chan_q = pdata->ram_chan_q; + dma_params->sram_pool = pdata->sram_pool; + dma_params->sram_size = pdata->sram_size_capture; if (dat) - dma_data->dma_addr = dat->start; + dma_params->dma_addr = dat->start; else - dma_data->dma_addr = mem->start + pdata->rx_dma_offset; + dma_params->dma_addr = mem->start + pdata->rx_dma_offset; /* Unconditional dmaengine stuff */ - mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = dma_data->dma_addr; + mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = dma_params->dma_addr; if (mcasp->version < MCASP_VERSION_3) { mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE; - /* dma_data->dma_addr is pointing to the data port address */ + /* dma_params->dma_addr is pointing to the data port address */ mcasp->dat_port = true; } else { mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE; @@ -1136,9 +1136,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (res) - dma_data->channel = res->start; + dma_params->channel = res->start; else - dma_data->channel = pdata->rx_dma_channel; + dma_params->channel = pdata->rx_dma_channel; /* Unconditional dmaengine stuff */ mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = "tx"; -- 1.9.0

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic23: add missing EXPORT_SYMBOLs
by Max Filippov 09 Mar '14

09 Mar '14

This fixes the following build errors when aic23 is configured as module: >> ERROR: "tlv320aic23_probe" >> [sound/soc/codecs/snd-soc-tlv320aic23-i2c.ko] undefined! >> ERROR: "tlv320aic23_regmap" >> [sound/soc/codecs/snd-soc-tlv320aic23-i2c.ko] undefined! Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- sound/soc/codecs/tlv320aic23.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 455250f..a934add 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -59,6 +59,7 @@ const struct regmap_config tlv320aic23_regmap = { .num_reg_defaults = ARRAY_SIZE(tlv320aic23_reg), .cache_type = REGCACHE_RBTREE, }; +EXPORT_SYMBOL(tlv320aic23_regmap); static const char *rec_src_text[] = { "Line", "Mic" }; static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; @@ -634,6 +635,7 @@ int tlv320aic23_probe(struct device *dev, struct regmap *regmap) return snd_soc_register_codec(dev, &soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1); } +EXPORT_SYMBOL(tlv320aic23_probe); MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); -- 1.8.1.4

2 1

[alsa-devel] [PATCH] ASoC: s6105-ipcam: Convert to table based DAPM setup
by Lars-Peter Clausen 08 Mar '14

08 Mar '14

Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. While we are at it also remove the snd_soc_dapm_enable_pin() in the init callback, since pins are enabled by default. Also drop the snd_soc_dapm_sync() calls, since they are ignored by the core anyway until the card has been fully instantiated. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/s6000/s6105-ipcam.c | 28 ++++++++++------------------ 1 file changed, 10 insertions(+), 18 deletions(-) diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 945e8ab..b2fb37c 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -104,8 +104,8 @@ static int output_type_get(struct snd_kcontrol *kcontrol, static int output_type_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = kcontrol->private_data; - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_card *card = kcontrol->private_data; + struct snd_soc_dapm_context *dapm = &card->dapm; unsigned int val = (ucontrol->value.enumerated.item[0] != 0); char *differential = "Audio Out Differential"; char *stereo = "Audio Out Stereo"; @@ -137,13 +137,7 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* Add s6105 specific widgets */ - snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, - ARRAY_SIZE(aic3x_dapm_widgets)); - - /* Set up s6105 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + struct snd_soc_card *card = rtd->card; /* not present */ snd_soc_dapm_nc_pin(dapm, "MONO_LOUT"); @@ -157,17 +151,10 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "RLOUT"); snd_soc_dapm_nc_pin(dapm, "HPRCOM"); - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Audio In"); - /* must correspond to audio_out_mux.private_value initializer */ - snd_soc_dapm_disable_pin(dapm, "Audio Out Differential"); - snd_soc_dapm_sync(dapm); - snd_soc_dapm_enable_pin(dapm, "Audio Out Stereo"); - - snd_soc_dapm_sync(dapm); + snd_soc_dapm_disable_pin(&card->dapm, "Audio Out Differential"); - snd_ctl_add(codec->card->snd_card, snd_ctl_new1(&audio_out_mux, codec)); + snd_ctl_add(card->snd_card, snd_ctl_new1(&audio_out_mux, card)); return 0; } @@ -190,6 +177,11 @@ static struct snd_soc_card snd_soc_card_s6105 = { .owner = THIS_MODULE, .dai_link = &s6105_dai, .num_links = 1, + + .dapm_widgets = aic3x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic3x_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct s6000_snd_platform_data s6105_snd_data __initdata = { -- 1.8.0

1 0

[alsa-devel] UCM: How does its api relate to ALSA lib api?
by Kamil Wekalski 07 Mar '14

07 Mar '14

Where will be then best to ask these questions?

1 0

[alsa-devel] compress: dpcm_path_get called in wrong place
by noman pouigt 07 Mar '14

07 Mar '14

Shouldn't dpcm_path_get be called earlier ? I think it should be called earlier as happens in soc-pcm file. static int soc_compr_open_fe(struct snd_compr_stream *cstream) { if (dpcm_path_get(fe, stream, &list) <= 0) { dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); In current code we open platform driver and then machine startup function gets called. After this we check for valid path which i think should happen earlier i.e. in the beginning of soc_compr_open_fe function. If you agree, then i will send a patch.

2 1

Re: [alsa-devel] [GIT PULL v2] Enhancement of support for Firewire devices
by Takashi Iwai 07 Mar '14

07 Mar '14

At Wed, 05 Mar 2014 12:13:03 +0900, Takashi Sakamoto wrote: > > This pull-request is an update for my previous pull-request: > [alsa-devel] [GIT PULL][PATCH 00/39] Enhancement of support for Firewire > devices > http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073446.h… > > Updates since previous pull-request: > firewire-lib: > - use swap() instead of local macro > - check NULL at memory allocation for sort table > fireworks: > - use bitmap.h for card slots > - add 'firewire' subdirectory for card node in procfs > - add sanity check of module parameter with clamp() > bebob: > - use bitmap.h for card slots > - add 'firewire' subdirectory for card node in procfs > - simplify codes for metering > - use swap() instead of local macro > - change order of last two commits Clemens, could you review this new series? thanks, Takashi

2 1

[alsa-devel] [TINYCOMPRESS][PATCH 1/1] compress: no need to set metadata before calling next_track
by Richard Fitzgerald 07 Mar '14

07 Mar '14

The metadata is mainly for MP3 gapless playback, since the MP3 audio stream does not contain enough information to enable gapless. Other audio formats do not necessarily require any additional metadata so we should allow next_track to be called without any metadata. Signed-off-by: Zhao Weijia <weijia.zhao(a)capelabs.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- compress.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/compress.c b/compress.c index 15dfdb7..0c9ecd2 100644 --- a/compress.c +++ b/compress.c @@ -534,8 +534,6 @@ int compress_next_track(struct compress *compress) if (!is_compress_running(compress)) return oops(compress, ENODEV, "device not ready"); - if (!compress->gapless_metadata) - return oops(compress, EPERM, "metadata not set"); if (ioctl(compress->fd, SNDRV_COMPRESS_NEXT_TRACK)) return oops(compress, errno, "cannot set next track\n"); compress->next_track = 1; -- 1.7.2.5

2 7

[alsa-devel] [PATCH] ALSA: hda - Fix loud click noise with IdeaPad 410Y
by Takashi Iwai 07 Mar '14

07 Mar '14

Lenovo IdeaPad 410Y with ALC282 codec makes loud click noises at boot and shutdown. Also, it wrongly misdetects the acpi_thinkpad hook. This patch adds a device-specific fixup for disabling the shutup callback that is the cause of the click noise and also avoiding the thinpad_helper calls. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=71511 Reported-and-tested-by: Guilherme Amadio <guilherme.amadio(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 850296a1e0ff..8d0a84436674 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3616,6 +3616,19 @@ static void alc_fixup_auto_mute_via_amp(struct hda_codec *codec, } } +static void alc_no_shutup(struct hda_codec *codec) +{ +} + +static void alc_fixup_no_shutup(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct alc_spec *spec = codec->spec; + spec->shutup = alc_no_shutup; + } +} + static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3844,6 +3857,7 @@ enum { ALC269_FIXUP_HP_GPIO_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, + ALC269_FIXUP_NO_SHUTUP, ALC286_FIXUP_SONY_MIC_NO_PRESENCE, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, @@ -4020,6 +4034,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC269_FIXUP_NO_SHUTUP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_no_shutup, + }, [ALC269_FIXUP_LENOVO_DOCK] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -4405,6 +4423,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 1.9.0

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Re: [alsa-devel] Call for testing: ALSA driver for Fireworks/BeBoB based devices
by Brand, Mark (Mr) (Summerstrand Campus South) 07 Mar '14

07 Mar '14

Hi Takashi, Thank you once again! Your comments have been very helpful. I now understand how to run ffado-dbus-server/ffado-mixer alongside your ALSA driver. I have pulled your revised Ozonic mixer from ffado-mixer, and when I now invoke ffado-mixer, it properly detects the Ozonic. I can also verify that the mixer control of my audio pass-through (which is plugged into inputs 3 & 4) works as expected. However, I have not yet been able to get audio playback (using "aplay -D plughw:2,0 Desktop/test.wav"). The .wav file is a short, 44.1/16 mono file saved from Audacity, if that's relevant. The terminal output of aplay seems to reflect successful playback, and I hear audible clicks as the file starts and ends its playback. The pass-through is also momentarily muted as aplay starts and stops. I have experimented with every conceivable combination of ffado-mixer settings, ultimately returning to the default settings that it was installed with. When aplay executes, the mixer app shows check marks next to both "Outgoing" and "Incoming". I have been testing the latest git version of the driver (updated this morning), and installing/uninstalling by DKMS. Please, is there anything you can suggest that I should try next? Is there anything obvious that I appear to have overlooked? I will try with a wider range of test materials tomorrow, as well as trying other ALSA apps, but I wanted to get this posted in case that's not the issue. I also recently saw some comments suggesting that pulseaudio be killed. Could that be the issue? I will investigate this tomorrow too. Thank you sincerely, Mark Brand. ________________________________________ From: Takashi Sakamoto [o-takashi(a)sakamocchi.jp] Sent: Sunday, February 23, 2014 9:55 AM To: Brand, Mark (Mr) (Summerstrand Campus South) Cc: alsa-devel(a)alsa-project.org; ffado-devel(a)lists.sf.net Subject: Re: Call for testing: ALSA driver for Fireworks/BeBoB based devices Hi Mark, Thanks for your testing. Ozonic is one of devices for which I test. > My testing is probably not particularly rigorous, and I would > sincerely appreciate your advice on what steps to follow > in order to verify your driver's functionality. Try playback/capture PCM samples/MIDI messages with ALSA applications. > In particular, you mention "(to control mixer channels please use > FFADO upstream)". My drivers don't give ways to control device's internal mixer. So you need the other stuff for this purpose. Please refer to this page. You can see some screenshots of ffado-mixer for 'M-Audio Ozonic'. I already commited these improvement to current FFADO trunk except for FW1814/ProjectMiX: MaudioBebob http://subversion.ffado.org/wiki/MaudioBebob > I have not yet set up FFADO on this machine, but I also see under > "Note": "- Don't use simultaneously 'ALSA PCM/MIDI playback/capture' > and 'jackd with Firewire (FFADO) backend'. Both of them try > connecting to the device when another is running." Would you please > explain? How should one rely on the "upstream" mixer functionality > while keeping jackd out of ALSA's way? Devices which my drivers support handle a pair of streams over IEEE1394 bus. PCM samples/MIDI messages are multiplexed into these streams. And snd-fireworks/snd-bebob uses 'duplex streams synchronization'. As a result: - ALSA PCM/MIDI functionality and jackd with firewire backend conflict for its streaming functionality. But: - ALSA PCM/MIDI fuctionality and ffado-dbus-server/ffado-mixer do not interrupt each other. > I open Audacity, which now lists "Ozonic: Ozonic PCM (hw:2,0)" as > ALSA input and output options. However, no audio input is shown when > I try to monitor the audio input I confirm that Audacity has such bugs to detect ALSA PCM character devices. In detail, its PortAudio v19 codes includes bugs. I have no will to investigate this bug because it's not within my work for these drivers. > $aplay -D hw:2,0 test.wav > (where test.wav has been saved from Audacity) > ... and this responds with: > aplay: main:722: audio open error: Input/output error According to aplay codes, snd_pcm_open() generate this error. I guess that another application still uses PCM character device for Ozonic. > $ aplay -D hw:2,0 Desktop/test.wav > Playing WAVE 'Desktop/test.wav' : Signed 16 bit Little Endian, Rate > 44100 Hz, Mono aplay: set_params:1239: Channels count not available You MUST add 'c' option to indicate the number of channels with 'hw' PCM device. For convinience, use 'plughw' PCM device instead, like: $ aplay -D plughw:2,0 Desktop/test.wav I note an item: - A few days ago, I fixed bug to fail to start stream for Ozonic. See: bebob: fix mappling for MIDI conformant data in stream channels https://github.com/takaswie/snd-firewire-improve/commit/545ab7068f1d53b2b17… Thanks Takashi Sakamoto o-takashi(a)sakamocchi.jp ________________________________ NOTICE: Please note that this eMail, and the contents thereof, is subject to the standard NMMU eMail disclaimer which may be found at: http://www.nmmu.ac.za/disclaimer/email.htm

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[alsa-devel] Controlling the cards order
by Olivier Langlois 07 Mar '14

07 Mar '14

Hi, I have added a second sound device in my system and it unexpectedly took the first card slot. I have found that it is easy to control the order as long as the devices use different drivers by modifying /etc/modprobe.d/alsa.conf ( http://alsa.opensrc.org/MultipleCards ) but in my situation I have 2 different devices using the same driver. I think that there is hope that I find an answer by digging the more general topic of ordering PCI devices drivers loading but if someone has few pointers to share, any help is welcome! Here are my 2 devices that I would like to change the order: lano1106@whippet2 /proc/asound $ cat cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xfbf20000 irq 84 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfbe60000 irq 88 thank you for your time, Olivier

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[alsa-devel] [PATCH] ASoC: pcm512x: Split out bus drivers
by Mark Brown 07 Mar '14

07 Mar '14

From: Mark Brown <broonie(a)linaro.org> Move to the new style of defining the bus interfaces in separate modules in order to simplify dependencies. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/Kconfig | 18 +++++- sound/soc/codecs/Makefile | 4 ++ sound/soc/codecs/pcm512x-i2c.c | 71 ++++++++++++++++++++++ sound/soc/codecs/pcm512x-spi.c | 69 +++++++++++++++++++++ sound/soc/codecs/pcm512x.c | 133 +++-------------------------------------- sound/soc/codecs/pcm512x.h | 9 +++ 6 files changed, 175 insertions(+), 129 deletions(-) create mode 100644 sound/soc/codecs/pcm512x-i2c.c create mode 100644 sound/soc/codecs/pcm512x-spi.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f2383eb..3a6046f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -66,7 +66,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 - select SND_SOC_PCM512x if SND_SOC_I2C_AND_SPI + select SND_SOC_PCM512x_I2C if I2C + select SND_SOC_PCM512x_SPI if SPI_MASTER select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C @@ -362,8 +363,19 @@ config SND_SOC_PCM3008 tristate config SND_SOC_PCM512x - tristate "Texas Instruments PCM512x CODECs" - select REGMAP + tristate + +config SND_SOC_PCM512x_I2C + tristate "Texas Instruments PCM512x CODECs - I2C" + depends on I2C + select SND_SOC_PCM512x + select REGMAP_I2C + +config SND_SOC_PCM512x_SPI + tristate "Texas Instruments PCM512x CODECs - SPI" + depends on SPI_MASTER + select SND_SOC_PCM512x + select REGMAP_SPI config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6af7a55..ac7025c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -54,6 +54,8 @@ snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-pcm512x-objs := pcm512x.o +snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o +snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -197,6 +199,8 @@ obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o +obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o +obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c new file mode 100644 index 0000000..4d62230 --- /dev/null +++ b/sound/soc/codecs/pcm512x-i2c.c @@ -0,0 +1,71 @@ +/* + * Driver for the PCM512x CODECs + * + * Author: Mark Brown <broonie(a)linaro.org> + * Copyright 2014 Linaro Ltd + * + * 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/init.h> +#include <linux/module.h> +#include <linux/i2c.h> + +#include "pcm512x.h" + +static int pcm512x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return pcm512x_probe(&i2c->dev, regmap); +} + +static int pcm512x_i2c_remove(struct i2c_client *i2c) +{ + pcm512x_remove(&i2c->dev); + return 0; +} + +static const struct i2c_device_id pcm512x_i2c_id[] = { + { "pcm5121", }, + { "pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); + +static const struct of_device_id pcm512x_of_match[] = { + { .compatible = "ti,pcm5121", }, + { .compatible = "ti,pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm512x_of_match); + +static struct i2c_driver pcm512x_i2c_driver = { + .probe = pcm512x_i2c_probe, + .remove = pcm512x_i2c_remove, + .id_table = pcm512x_i2c_id, + .driver = { + .name = "pcm512x", + .owner = THIS_MODULE, + .of_match_table = pcm512x_of_match, + .pm = &pcm512x_pm_ops, + }, +}; + +module_i2c_driver(pcm512x_i2c_driver); + +MODULE_DESCRIPTION("ASoC PCM512x codec driver - I2C"); +MODULE_AUTHOR("Mark Brown <broonie(a)linaro.org>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm512x-spi.c b/sound/soc/codecs/pcm512x-spi.c new file mode 100644 index 0000000..f297058 --- /dev/null +++ b/sound/soc/codecs/pcm512x-spi.c @@ -0,0 +1,69 @@ +/* + * Driver for the PCM512x CODECs + * + * Author: Mark Brown <broonie(a)linaro.org> + * Copyright 2014 Linaro Ltd + * + * 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/init.h> +#include <linux/module.h> +#include <linux/spi/spi.h> + +#include "pcm512x.h" + +static int pcm512x_spi_probe(struct spi_device *spi) +{ + struct regmap *regmap; + int ret; + + regmap = devm_regmap_init_spi(spi, &pcm512x_regmap); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + return ret; + } + + return pcm512x_probe(&spi->dev, regmap); +} + +static int pcm512x_spi_remove(struct spi_device *spi) +{ + pcm512x_remove(&spi->dev); + return 0; +} + +static const struct spi_device_id pcm512x_spi_id[] = { + { "pcm5121", }, + { "pcm5122", }, + { }, +}; +MODULE_DEVICE_TABLE(spi, pcm512x_spi_id); + +static const struct of_device_id pcm512x_of_match[] = { + { .compatible = "ti,pcm5121", }, + { .compatible = "ti,pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm512x_of_match); + +static struct spi_driver pcm512x_spi_driver = { + .probe = pcm512x_spi_probe, + .remove = pcm512x_spi_remove, + .id_table = pcm512x_spi_id, + .driver = { + .name = "pcm512x", + .owner = THIS_MODULE, + .of_match_table = pcm512x_of_match, + .pm = &pcm512x_pm_ops, + }, +}; + +module_spi_driver(pcm512x_spi_driver); diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index a035ba8..0014d14 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -18,11 +18,9 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/clk.h> -#include <linux/i2c.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> -#include <linux/spi/spi.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/tlv.h> @@ -375,7 +373,7 @@ static const struct regmap_range_cfg pcm512x_range = { .window_start = 0, .window_len = 0x100, }; -static const struct regmap_config pcm512x_regmap = { +const struct regmap_config pcm512x_regmap = { .reg_bits = 8, .val_bits = 8, @@ -391,14 +389,7 @@ static const struct regmap_config pcm512x_regmap = { .cache_type = REGCACHE_RBTREE, }; -static const struct of_device_id pcm512x_of_match[] = { - { .compatible = "ti,pcm5121", }, - { .compatible = "ti,pcm5122", }, - { } -}; -MODULE_DEVICE_TABLE(of, pcm512x_of_match); - -static int pcm512x_probe(struct device *dev, struct regmap *regmap) +int pcm512x_probe(struct device *dev, struct regmap *regmap) { struct pcm512x_priv *pcm512x; int i, ret; @@ -510,8 +501,9 @@ err: pcm512x->supplies); return ret; } +EXPORT_SYMBOL_GPL(pcm512x_probe); -static void pcm512x_remove(struct device *dev) +void pcm512x_remove(struct device *dev) { struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); @@ -522,6 +514,7 @@ static void pcm512x_remove(struct device *dev) regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies), pcm512x->supplies); } +EXPORT_SYMBOL_GPL(pcm512x_remove); static int pcm512x_suspend(struct device *dev) { @@ -585,122 +578,10 @@ static int pcm512x_resume(struct device *dev) return 0; } -static const struct dev_pm_ops pcm512x_pm_ops = { +const struct dev_pm_ops pcm512x_pm_ops = { SET_RUNTIME_PM_OPS(pcm512x_suspend, pcm512x_resume, NULL) }; - -#if IS_ENABLED(CONFIG_I2C) -static int pcm512x_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct regmap *regmap; - - regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap); - if (IS_ERR(regmap)) - return PTR_ERR(regmap); - - return pcm512x_probe(&i2c->dev, regmap); -} - -static int pcm512x_i2c_remove(struct i2c_client *i2c) -{ - pcm512x_remove(&i2c->dev); - return 0; -} - -static const struct i2c_device_id pcm512x_i2c_id[] = { - { "pcm5121", }, - { "pcm5122", }, - { } -}; -MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); - -static struct i2c_driver pcm512x_i2c_driver = { - .probe = pcm512x_i2c_probe, - .remove = pcm512x_i2c_remove, - .id_table = pcm512x_i2c_id, - .driver = { - .name = "pcm512x", - .owner = THIS_MODULE, - .of_match_table = pcm512x_of_match, - .pm = &pcm512x_pm_ops, - }, -}; -#endif - -#if defined(CONFIG_SPI_MASTER) -static int pcm512x_spi_probe(struct spi_device *spi) -{ - struct regmap *regmap; - int ret; - - regmap = devm_regmap_init_spi(spi, &pcm512x_regmap); - if (IS_ERR(regmap)) { - ret = PTR_ERR(regmap); - return ret; - } - - return pcm512x_probe(&spi->dev, regmap); -} - -static int pcm512x_spi_remove(struct spi_device *spi) -{ - pcm512x_remove(&spi->dev); - return 0; -} - -static const struct spi_device_id pcm512x_spi_id[] = { - { "pcm5121", }, - { "pcm5122", }, - { }, -}; -MODULE_DEVICE_TABLE(spi, pcm512x_spi_id); - -static struct spi_driver pcm512x_spi_driver = { - .probe = pcm512x_spi_probe, - .remove = pcm512x_spi_remove, - .id_table = pcm512x_spi_id, - .driver = { - .name = "pcm512x", - .owner = THIS_MODULE, - .of_match_table = pcm512x_of_match, - .pm = &pcm512x_pm_ops, - }, -}; -#endif - -static int __init pcm512x_modinit(void) -{ - int ret = 0; - -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&pcm512x_i2c_driver); - if (ret) { - printk(KERN_ERR "Failed to register pcm512x I2C driver: %d\n", - ret); - } -#endif -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&pcm512x_spi_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register pcm512x SPI driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(pcm512x_modinit); - -static void __exit pcm512x_exit(void) -{ -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&pcm512x_i2c_driver); -#endif -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&pcm512x_spi_driver); -#endif -} -module_exit(pcm512x_exit); +EXPORT_SYMBOL_GPL(pcm512x_pm_ops); MODULE_DESCRIPTION("ASoC PCM512x codec driver"); MODULE_AUTHOR("Mark Brown <broonie(a)linaro.org>"); diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h index ac4a52c..6ee76aa 100644 --- a/sound/soc/codecs/pcm512x.h +++ b/sound/soc/codecs/pcm512x.h @@ -17,6 +17,9 @@ #ifndef _SND_SOC_PCM512X #define _SND_SOC_PCM512X +#include <linux/pm.h> +#include <linux/regmap.h> + #define PCM512x_VIRT_BASE 0x100 #define PCM512x_PAGE_LEN 0x100 #define PCM512x_PAGE_BASE(n) (PCM512x_VIRT_BASE + (PCM512x_PAGE_LEN * n)) @@ -159,4 +162,10 @@ #define PCM512x_AGBR_SHIFT 0 #define PCM512x_AGBL_SHIFT 4 +extern const struct dev_pm_ops pcm512x_pm_ops; +extern const struct regmap_config pcm512x_regmap; + +int pcm512x_probe(struct device *dev, struct regmap *regmap); +void pcm512x_remove(struct device *dev); + #endif -- 1.9.0

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[alsa-devel] [PATCH] ASoC: pcm512x: Fix duplicate const warning
by Mark Brown 07 Mar '14

07 Mar '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm512x.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 3a0bbb6..a035ba8 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -203,10 +203,10 @@ static const unsigned int pcm512x_dsp_program_values[] = { 7, }; -static const SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program, - PCM512x_DSP_PROGRAM, 0, 0x1f, - pcm512x_dsp_program_texts, - pcm512x_dsp_program_values); +static SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program, + PCM512x_DSP_PROGRAM, 0, 0x1f, + pcm512x_dsp_program_texts, + pcm512x_dsp_program_values); static const char * const pcm512x_clk_missing_text[] = { "1s", "2s", "3s", "4s", "5s", "6s", "7s", "8s" -- 1.9.0

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[alsa-devel] [PATCH v2 0/2] add support for SPI control mode to tlv320aic23
by Max Filippov 07 Mar '14

07 Mar '14

Hi, this series splits I2C-specific part of tlv320aic23 codec driver into a separate file and adds SPI control mode support. It also updates Kconfigs of existing tlv320aic23 users so that I2C variant is selected. Changes v1->v2: - split i2c interface into separate file; - implement spi interface in a separate file; - update Kconfigs of existing users. Max Filippov (2): ASoC: tlv320aic23: add support for SPI control mode ASoC: update Kconfig of AIC23 users to select I2C variant sound/soc/atmel/Kconfig | 2 +- sound/soc/cirrus/Kconfig | 2 +- sound/soc/codecs/Kconfig | 11 ++++++- sound/soc/codecs/Makefile | 4 +++ sound/soc/codecs/tlv320aic23-i2c.c | 59 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23-spi.c | 57 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23.c | 55 +++++++---------------------------- sound/soc/codecs/tlv320aic23.h | 6 ++++ sound/soc/fsl/Kconfig | 2 +- sound/soc/omap/Kconfig | 4 +-- sound/soc/samsung/Kconfig | 2 +- sound/soc/tegra/Kconfig | 2 +- 12 files changed, 154 insertions(+), 52 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic23-i2c.c create mode 100644 sound/soc/codecs/tlv320aic23-spi.c -- 1.8.1.4

2 3

[alsa-devel] [PATCH] ASoC: Intel: Check Haswell IPC process_reply/notification return value.
by Liam Girdwood 07 Mar '14

07 Mar '14

Check the return value for error when processing replies and notifications. The patch 22981243589c: "ASoC: Intel: Add Haswell/Broadwell IPC" from > Feb 20, 2014, leads to the following imaginary static checker warning: > > sound/soc/intel/sst-haswell-ipc.c:898 hsw_irq_thread() > warn: this is always true. > > sound/soc/intel/sst-haswell-ipc.c > 895 /* Handle Immediate reply from DSP Core */ > 896 handled = hsw_process_reply(hsw, ipcx); > ^^^^^^^^^^^^^^^^^ > Returns 1 on success/error and -EIO on error. > > 897 > 898 if (handled) { > 899 /* clear DONE bit - tell DSP we have completed */ > 900 sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX, > 901 SST_IPCX_DONE, 0); > 902 > 903 /* unmask Done interrupt */ > 904 sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, > 905 SST_IMRX_DONE, 0); > 906 } > Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 1f1576a..f46bb4d 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -895,7 +895,7 @@ static irqreturn_t hsw_irq_thread(int irq, void *context) /* Handle Immediate reply from DSP Core */ handled = hsw_process_reply(hsw, ipcx); - if (handled) { + if (handled > 0) { /* clear DONE bit - tell DSP we have completed */ sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX, SST_IPCX_DONE, 0); @@ -913,7 +913,7 @@ static irqreturn_t hsw_irq_thread(int irq, void *context) handled = hsw_process_notification(hsw); /* clear BUSY bit and set DONE bit - accept new messages */ - if (handled) { + if (handled > 0) { sst_dsp_shim_update_bits_unlocked(sst, SST_IPCD, SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); -- 1.8.3.2

2 5

[alsa-devel] [PATCH] ASoC: wm_adsp: Correct type specifier in printf
by Charles Keepax 07 Mar '14

07 Mar '14

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 937af6f..bb5f7b4 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -706,7 +706,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) to_write); if (ret != 0) { adsp_err(dsp, - "%s.%d: Failed to write %d bytes at %d in %s: %d\n", + "%s.%d: Failed to write %zd bytes at %d in %s: %d\n", file, regions, to_write, offset, region_name, ret); -- 1.7.2.5

2 1

Re: [alsa-devel] ASoC: wm5110: Extend SYSCLK patch file for rev D
by Ben Hutchings 06 Mar '14

06 Mar '14

I'm looking at commit 34354792432b6e0a3b156819a1a19716c50d3473 and include/linux/mfd/arizona/registers.h, and wondering what exactly this the register patch is changing. It looks like there are some registers not named elsewhere (addresses 0x0049, 0x004a, 0x0057, 0x0058). What about the higher addresses? Are these more registers, or is this patching firmware in device memory? Ben. -- Ben Hutchings The world is coming to an end. Please log off.

2 2

[alsa-devel] [PATCH] ASoC: tlv320aic23: add support for SPI control mode
by Max Filippov 06 Mar '14

06 Mar '14

tlv320aic23 chip control interface may work in either I2C or SPI mode depending on the MODE pin state. Functionality and register layout are independent of the control mode. Provide i2c and spi driver entry points when CONFIG_I2C and CONFIG_SPI_MASTER are enabled respectively. Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/tlv320aic23.c | 58 +++++++++++++++++++++++++++++++++++++++--- 2 files changed, 56 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..6d82de5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -71,7 +71,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TAS5086 if I2C - select SND_SOC_TLV320AIC23 if I2C + select SND_SOC_TLV320AIC23 if SND_SOC_I2C_AND_SPI select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 5d430cc..2418260 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -26,6 +26,7 @@ #include <linux/i2c.h> #include <linux/regmap.h> #include <linux/slab.h> +#include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -617,12 +618,13 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon), }; +#ifdef CONFIG_I2C /* * If the i2c layer weren't so broken, we could pass this kind of data * around */ -static int tlv320aic23_codec_probe(struct i2c_client *i2c, - const struct i2c_device_id *i2c_id) +static int tlv320aic23_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) { struct aic23 *aic23; int ret; @@ -661,12 +663,62 @@ static struct i2c_driver tlv320aic23_i2c_driver = { .driver = { .name = "tlv320aic23-codec", }, - .probe = tlv320aic23_codec_probe, + .probe = tlv320aic23_i2c_probe, .remove = __exit_p(tlv320aic23_i2c_remove), .id_table = tlv320aic23_id, }; module_i2c_driver(tlv320aic23_i2c_driver); +#endif /* CONFIG_I2C */ + +#ifdef CONFIG_SPI_MASTER +static int aic23_spi_probe(struct spi_device *spi) +{ + struct aic23 *aic23; + int ret; + + dev_dbg(&spi->dev, "probing tlv320aic23 spi device\n"); + + spi->bits_per_word = 16; + spi->mode = SPI_MODE_0; + ret = spi_setup(spi); + if (ret < 0) + return ret; + + /* Allocate driver data */ + aic23 = devm_kzalloc(&spi->dev, sizeof(struct aic23), GFP_KERNEL); + if (!aic23) + return -ENOMEM; + + aic23->regmap = devm_regmap_init_spi(spi, &tlv320aic23_regmap); + if (IS_ERR(aic23->regmap)) + return PTR_ERR(aic23->regmap); + + /* Initialize the driver data */ + dev_set_drvdata(&spi->dev, aic23); + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1); + return ret; +} + +static int aic23_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver aic23_spi = { + .driver = { + .name = "tlv320aic23", + .owner = THIS_MODULE, + }, + .probe = aic23_spi_probe, + .remove = aic23_spi_remove, +}; + +module_spi_driver(aic23_spi); +#endif /* CONFIG_SPI_MASTER */ MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); -- 1.8.1.4

3 5

[alsa-devel] [PATCH 1/7] ASoC: Handle ignore_pmdown_time for CODEC to CODEC links
by Lars-Peter Clausen 06 Mar '14

06 Mar '14

For CODEC to CODEC links we should only immediately power down if both CODECs are configured to ignore the power down delay. Factor the logic for this into a helper function that can be used for both compressed and normal PCMs. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 2 ++ sound/soc/soc-compress.c | 3 +-- sound/soc/soc-pcm.c | 27 +++++++++++++++++++++++++-- 3 files changed, 28 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a00147..93c31c7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -413,6 +413,8 @@ struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, const char *dai_link); +bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd); + /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params); diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 5e9690c..ef585af 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -235,8 +235,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) cpu_dai->runtime = NULL; if (cstream->direction == SND_COMPRESS_PLAYBACK) { - if (!rtd->pmdown_time || codec->ignore_pmdown_time || - rtd->dai_link->ignore_pmdown_time) { + if (snd_soc_runtime_ignore_pmdown_time(rtd)) { snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 47e1ce7..f098c80 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -35,6 +35,30 @@ #define DPCM_MAX_BE_USERS 8 /** + * snd_soc_runtime_ignore_pmdown_time() - Check whether to ignore the power down delay + * @rtd: The ASoC PCM runtime that should be checked. + * + * This function checks whether the power down delay should be ignored for a + * specific PCM runtime. Returns true if the delay is 0, if it the DAI link has + * been configured to ignore the delay, or if none of the components benefits + * from having the delay. + */ +bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd) +{ + bool ignore = true; + + if (!rtd->pmdown_time || rtd->dai_link->ignore_pmdown_time) + return true; + + if (rtd->cpu_dai->codec) + ignore &= rtd->cpu_dai->codec->ignore_pmdown_time; + + ignore &= rtd->codec_dai->codec->ignore_pmdown_time; + + return ignore; +} + +/** * snd_soc_set_runtime_hwparams - set the runtime hardware parameters * @substream: the pcm substream * @hw: the hardware parameters @@ -496,8 +520,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (!rtd->pmdown_time || codec->ignore_pmdown_time || - rtd->dai_link->ignore_pmdown_time) { + if (snd_soc_runtime_ignore_pmdown_time(rtd)) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, -- 1.8.0

2 7

[alsa-devel] [PATCH] ASoC: dapm: Sprinkle lockdep asserts through the code
by Mark Brown 06 Mar '14

06 Mar '14

From: Mark Brown <broonie(a)linaro.org> Try to spot locking issues by asserting that the DAPM mutex is held when it should be. There's a bit of fun due to us not requiring the lock to be held prior to the card being instantiated which mean we need to wrap the assert in some paths and this isn't methodical by any stretch of the imagination. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- I've only compile tested this so it's entirely possible there's some oversights here - at this point I'm mostly just throwing it out there for people to play with as a work in progress. sound/soc/soc-dapm.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3728e21..c8a780d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -111,6 +111,12 @@ static int dapm_down_seq[] = { [snd_soc_dapm_post] = 14, }; +static void dapm_assert_locked(struct snd_soc_dapm_context *dapm) +{ + if (dapm->card && dapm->card->instantiated) + lockdep_assert_held(&dapm->card->dapm_mutex); +} + static void pop_wait(u32 pop_time) { if (pop_time) @@ -144,6 +150,8 @@ static bool dapm_dirty_widget(struct snd_soc_dapm_widget *w) static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) { + dapm_assert_locked(w->dapm); + if (!dapm_dirty_widget(w)) { dev_vdbg(w->dapm->dev, "Marking %s dirty due to %s\n", w->name, reason); @@ -356,6 +364,8 @@ static void dapm_reset(struct snd_soc_card *card) { struct snd_soc_dapm_widget *w; + lockdep_assert_held(&card->dapm_mutex); + memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); list_for_each_entry(w, &card->widgets, list) { @@ -1750,6 +1760,8 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) ASYNC_DOMAIN_EXCLUSIVE(async_domain); enum snd_soc_bias_level bias; + lockdep_assert_held(&card->dapm_mutex); + trace_snd_soc_dapm_start(card); list_for_each_entry(d, &card->dapm_list, list) { @@ -2045,6 +2057,8 @@ static int soc_dapm_mux_update_power(struct snd_soc_card *card, struct snd_soc_dapm_path *path; int found = 0; + lockdep_assert_held(&card->dapm_mutex); + /* find dapm widget path assoc with kcontrol */ dapm_kcontrol_for_each_path(path, kcontrol) { if (!path->name || !e->texts[mux]) @@ -2095,6 +2109,8 @@ static int soc_dapm_mixer_update_power(struct snd_soc_card *card, struct snd_soc_dapm_path *path; int found = 0; + lockdep_assert_held(&card->dapm_mutex); + /* find dapm widget path assoc with kcontrol */ dapm_kcontrol_for_each_path(path, kcontrol) { found = 1; @@ -2260,6 +2276,8 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); + dapm_assert_locked(dapm); + if (!w) { dev_err(dapm->dev, "ASoC: DAPM unknown pin %s\n", pin); return -EINVAL; -- 1.9.0

1 0

[alsa-devel] [PATCH] ASoC: dapm: Staticise dapm_mark_dirty()
by Mark Brown 06 Mar '14

06 Mar '14

From: Mark Brown <broonie(a)linaro.org> The function is not called outside soc-dapm.c so there is no need for it to be exported. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc-dapm.h | 1 - sound/soc/soc-dapm.c | 3 +-- 2 files changed, 1 insertion(+), 3 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 04d32d8..ef78f56 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -449,7 +449,6 @@ int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); /* Mostly internal - should not normally be used */ -void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason); void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm); /* dapm path query */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0d8ba30..3728e21 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -142,7 +142,7 @@ static bool dapm_dirty_widget(struct snd_soc_dapm_widget *w) return !list_empty(&w->dirty); } -void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) +static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) { if (!dapm_dirty_widget(w)) { dev_vdbg(w->dapm->dev, "Marking %s dirty due to %s\n", @@ -150,7 +150,6 @@ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) list_add_tail(&w->dirty, &w->dapm->card->dapm_dirty); } } -EXPORT_SYMBOL_GPL(dapm_mark_dirty); void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm) { -- 1.9.0

1 0

[alsa-devel] [PATCH] ASoC: cs4271: Fix build error without CONFIG_SPI_MASTER
by Takashi Iwai 06 Mar '14

06 Mar '14

cs4271_common_probe() is called from cs4271_i2c_probe() but defined in CONFIG_SPI_MASTER block, thus it results in a build error when CONFIG_SPI_MASTER=n: sound/soc/codecs/cs4271.c:721:2: error: implicit declaration of function ‘cs4271_common_probe’ [-Werror=implicit-function-declaration] Move the function out of #if block. Fixes: d6cf89ee07cb ('ASoC: cs4271: claim reset GPIO in bus probe function') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/cs4271.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 96c309777208..aef4965750c7 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -612,22 +612,6 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .num_dapm_routes = ARRAY_SIZE(cs4271_dapm_routes), }; -#if defined(CONFIG_SPI_MASTER) - -static const struct regmap_config cs4271_spi_regmap = { - .reg_bits = 16, - .val_bits = 8, - .max_register = CS4271_LASTREG, - .read_flag_mask = 0x21, - .write_flag_mask = 0x20, - - .reg_defaults = cs4271_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), - .cache_type = REGCACHE_RBTREE, - - .volatile_reg = cs4271_volatile_reg, -}; - static int cs4271_common_probe(struct device *dev, struct cs4271_private **c) { @@ -658,6 +642,22 @@ static int cs4271_common_probe(struct device *dev, return 0; } +#if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config cs4271_spi_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = CS4271_LASTREG, + .read_flag_mask = 0x21, + .write_flag_mask = 0x20, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_spi_probe(struct spi_device *spi) { struct cs4271_private *cs4271; -- 1.9.0

4 3

[alsa-devel] [patch] ASoC: Baytrail: fix error handling in sst_byt_dsp_init()
by Dan Carpenter 06 Mar '14

06 Mar '14

Calling "kfree(byt)" is a double free because that was allocated with devm_kzalloc(). There were a couple places which leak "byt->msg". That memory is allocated in msg_empty_list_init(). Fixes: f7d01fd6754c ('ASoC: Intel: Add Intel Baytrail SST DSP IPC support') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index c12e194bbc6b..d0eaeee21be4 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -796,7 +796,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) err = msg_empty_list_init(byt); if (err < 0) - goto list_err; + return -ENOMEM; /* start the IPC message thread */ init_kthread_worker(&byt->kworker); @@ -806,7 +806,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) if (IS_ERR(byt->tx_thread)) { err = PTR_ERR(byt->tx_thread); dev_err(byt->dev, "error failed to create message TX task\n"); - goto list_err; + goto err_free_msg; } init_kthread_work(&byt->kwork, sst_byt_ipc_tx_msgs); @@ -816,7 +816,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) byt->dsp = sst_dsp_new(dev, &byt_dev, pdata); if (byt->dsp == NULL) { err = -ENODEV; - goto list_err; + goto err_free_msg; } /* keep the DSP in reset state for base FW loading */ @@ -848,9 +848,9 @@ boot_err: sst_fw_free(byt_sst_fw); fw_err: sst_dsp_free(byt->dsp); +err_free_msg: kfree(byt->msg); -list_err: - kfree(byt); + return err; } EXPORT_SYMBOL_GPL(sst_byt_dsp_init);

3 2

[alsa-devel] UCM: How does its api relate to ALSA lib api?
by kaweka 05 Mar '14

05 Mar '14

Hi, I am a newcomer if it goes to following topics: ALSA, audio in Linux, audio in embedded Linux. However several years of experience in audio development on other platforms. As next a development of audio applications in embedded Linux is the task. However have few gaps in understanding of the audio system architecture. It would be helpful to have answers to following questions. For any hints lot of thanks in advance! * There is well-known handshaking application-ALSA lib based on amonst others snd_pcm_open snd_pcm_hw_params_alloca snd_pcm_hw_params_any snd_pcm_hw_params_set_... snd_pcm_hw_params snd_pcm_hw_params_get_... snd_pcm_hw_params_is_... snd_pcm_hw_params_can_... snd_pcm_close * Does the UCM and its api complement the api and handshaking as above. Or is it rather a question of replacement? * If neither nor, how to describe best the relation between UCM api and ALSA lib api? * How does the entire handshaking application-audio system look like if to use ALSA and UCM api? * An use case seems to be just a container for verb, device, modifier. Is an use case introducing any functionality to be found neither in verb, nor in device, nor in modifier? * For what reason does the term / definition of use case exists at all. Will verb, device, modifier not be enough? * What is the difference between switching to other use case and modifying a use case? Why not to switch to other use case in situations where modifier is used? * QoS api seems to introduce some redundancy. If voice, or music, or other - this information can be gathered from use case. A use case does already read voice, music, other! Why is qos api necessary? Best Regards kaweka

1 0

[alsa-devel] Add "cval->res = 384" quirk for Logitech Webcam C500
by Simon 05 Mar '14

05 Mar '14

Hello, Since some time I am successfully using this workaround http://mailman.alsa-project.org/pipermail/alsa-devel/2011-March/037657.html to solve the problem for the Logitech C500 Webcam (046d:0807). Would it be possible to add this device to the list included in the mixer.c file? Thanks

2 1

[alsa-devel] [patch] ALSA: echoaudio: use after free on error
by Dan Carpenter 05 Mar '14

05 Mar '14

There are some places where we dereference "chip" in the error message but we've already freed it. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 166ec0c120d4..9f10c9e0df5e 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1990,8 +1990,8 @@ static int snd_echo_create(struct snd_card *card, if ((chip->iores = request_mem_region(chip->dsp_registers_phys, sz, ECHOCARD_NAME)) == NULL) { - snd_echo_free(chip); dev_err(chip->card->dev, "cannot get memory region\n"); + snd_echo_free(chip); return -EBUSY; } chip->dsp_registers = (volatile u32 __iomem *) @@ -1999,8 +1999,8 @@ static int snd_echo_create(struct snd_card *card, if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { - snd_echo_free(chip); dev_err(chip->card->dev, "cannot grab irq\n"); + snd_echo_free(chip); return -EBUSY; } chip->irq = pci->irq; @@ -2012,8 +2012,8 @@ static int snd_echo_create(struct snd_card *card, if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), sizeof(struct comm_page), &chip->commpage_dma_buf) < 0) { - snd_echo_free(chip); dev_err(chip->card->dev, "cannot allocate the comm page\n"); + snd_echo_free(chip); return -ENOMEM; } chip->comm_page_phys = chip->commpage_dma_buf.addr; @@ -2291,8 +2291,8 @@ static int snd_echo_resume(struct device *dev) if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { - snd_echo_free(chip); dev_err(chip->card->dev, "cannot grab irq\n"); + snd_echo_free(chip); return -EBUSY; } chip->irq = pci->irq;

3 3

Re: [alsa-devel] [GIT PULL v2] Enhancement of support for Firewire devices
by Takashi Iwai 05 Mar '14

05 Mar '14

At Wed, 05 Mar 2014 12:13:03 +0900, Takashi Sakamoto wrote: > > This pull-request is an update for my previous pull-request: > [alsa-devel] [GIT PULL][PATCH 00/39] Enhancement of support for Firewire > devices > http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073446.h… > > Updates since previous pull-request: > firewire-lib: > - use swap() instead of local macro > - check NULL at memory allocation for sort table > fireworks: > - use bitmap.h for card slots > - add 'firewire' subdirectory for card node in procfs > - add sanity check of module parameter with clamp() > bebob: > - use bitmap.h for card slots > - add 'firewire' subdirectory for card node in procfs > - simplify codes for metering > - use swap() instead of local macro > - change order of last two commits Could you submit v2 patches to ML? The patches need review, and I want get them acked by Clemens before applying. thanks, Takashi > > ---------------------------------------------------------------- > The following changes since commit 31278997add61ee13b709e274934c7b0085accce: > > ALSA: hda/realtek - Add headset quirk for Dell DT (2014-03-03 10:48:55 > +0100) > > are available in the git repository at: > > https://github.com/takaswie/sound.git request > > for you to fetch changes up to 5b354e20ec3c64fa0a8fa7fbc7355201d881a8cc: > > bebob: Send a cue to load firmware for M-Audio Firewire series > (2014-03-05 12:00:32 +0900) > > ---------------------------------------------------------------- > Takashi Sakamoto (39): > firewire-lib: Rename functions, structure, member for AMDTP > firewire-lib: Add macros instead of fixed value for AMDTP > firewire-lib: Add 'direction' member to 'amdtp_stream' structure > firewire-lib: Split some codes into functions to reuse for both > streams > firewire-lib: Add support for AMDTP in-stream and PCM capture > firewire-lib: Add support for MIDI capture/playback > firewire-lib: Give syt value as parameter to handle_out_packet() > firewire-lib: Add support for duplex streams synchronization in > blocking mode > firewire-lib: Add sort function for transmitted packet > firewire-lib: Add transfer delay to synchronized duplex streams > firewire-lib: Add support for channel mapping > firewire-lib: Rename macros, variables and functions for CMP > firewire-lib: Add 'direction' member to 'cmp_connection' structure > firewire-lib: Add handling output connection by CMP > firewire-lib: Add a new function to check others' connection > firewire-lib: Add some AV/C general commands > firewire-lib: Add quirks for Fireworks > firewire-lib: Add a fallback at RCODE_CANCELLED > fireworks: Add skelton for Fireworks based devices > fireworks: Add transaction and some commands > fireworks: Add connection and stream management > fireworks: Add proc interface for debugging purpose > fireworks: Add MIDI interface > fireworks: Add PCM interface > fireworks: Add hwdep interface > fireworks: Add command/response functionality into hwdep interface > bebob: Add skelton for BeBoB based devices > bebob: Add commands and connections/streams management > bebob: Add proc interface for debugging purpose > bebob: Add MIDI interface > bebob: Add PCM interface > bebob: Add hwdep interface > bebob: Prepare for device specific operations > bebob: Add support for Terratec PHASE, EWS series and Aureon > bebob: Add support for Yamaha GO series > bebob: Add support for Focusrite Saffire/SaffirePro series > bebob: Add support for M-Audio usual Firewire series > bebob: Add support for M-Audio special Firewire series > bebob: Send a cue to load firmware for M-Audio Firewire series > > include/uapi/sound/asound.h | 4 +- > include/uapi/sound/firewire.h | 22 +- > sound/firewire/Kconfig | 59 ++ > sound/firewire/Makefile | 2 + > sound/firewire/amdtp.c | 891 > +++++++++++++++++------ > sound/firewire/amdtp.h | 175 ++++- > sound/firewire/bebob/Makefile | 4 + > sound/firewire/bebob/bebob.c | 450 ++++++++++++ > sound/firewire/bebob/bebob.h | 260 +++++++ > sound/firewire/bebob/bebob_command.c | 354 +++++++++ > sound/firewire/bebob/bebob_focusrite.c | 287 ++++++++ > sound/firewire/bebob/bebob_hwdep.c | 197 +++++ > sound/firewire/bebob/bebob_maudio.c | 828 > +++++++++++++++++++++ > sound/firewire/bebob/bebob_midi.c | 170 +++++ > sound/firewire/bebob/bebob_pcm.c | 451 ++++++++++++ > sound/firewire/bebob/bebob_proc.c | 201 +++++ > sound/firewire/bebob/bebob_stream.c | 869 > ++++++++++++++++++++++ > sound/firewire/bebob/bebob_terratec.c | 68 ++ > sound/firewire/bebob/bebob_yamaha.c | 50 ++ > sound/firewire/cmp.c | 229 ++++-- > sound/firewire/cmp.h | 14 +- > sound/firewire/dice.c | 46 +- > sound/firewire/fcp.c | 160 +++- > sound/firewire/fcp.h | 21 + > sound/firewire/fireworks/Makefile | 4 + > sound/firewire/fireworks/fireworks.c | 326 +++++++++ > sound/firewire/fireworks/fireworks.h | 223 ++++++ > sound/firewire/fireworks/fireworks_command.c | 397 ++++++++++ > sound/firewire/fireworks/fireworks_hwdep.c | 294 ++++++++ > sound/firewire/fireworks/fireworks_midi.c | 179 +++++ > sound/firewire/fireworks/fireworks_pcm.c | 477 ++++++++++++ > sound/firewire/fireworks/fireworks_proc.c | 233 ++++++ > sound/firewire/fireworks/fireworks_stream.c | 337 +++++++++ > sound/firewire/fireworks/fireworks_transaction.c | 327 +++++++++ > sound/firewire/lib.c | 6 +- > sound/firewire/lib.h | 1 + > sound/firewire/speakers.c | 93 +-- > 37 files changed, 8307 insertions(+), 402 deletions(-) > create mode 100644 sound/firewire/bebob/Makefile > create mode 100644 sound/firewire/bebob/bebob.c > create mode 100644 sound/firewire/bebob/bebob.h > create mode 100644 sound/firewire/bebob/bebob_command.c > create mode 100644 sound/firewire/bebob/bebob_focusrite.c > create mode 100644 sound/firewire/bebob/bebob_hwdep.c > create mode 100644 sound/firewire/bebob/bebob_maudio.c > create mode 100644 sound/firewire/bebob/bebob_midi.c > create mode 100644 sound/firewire/bebob/bebob_pcm.c > create mode 100644 sound/firewire/bebob/bebob_proc.c > create mode 100644 sound/firewire/bebob/bebob_stream.c > create mode 100644 sound/firewire/bebob/bebob_terratec.c > create mode 100644 sound/firewire/bebob/bebob_yamaha.c > create mode 100644 sound/firewire/fireworks/Makefile > create mode 100644 sound/firewire/fireworks/fireworks.c > create mode 100644 sound/firewire/fireworks/fireworks.h > create mode 100644 sound/firewire/fireworks/fireworks_command.c > create mode 100644 sound/firewire/fireworks/fireworks_hwdep.c > create mode 100644 sound/firewire/fireworks/fireworks_midi.c > create mode 100644 sound/firewire/fireworks/fireworks_pcm.c > create mode 100644 sound/firewire/fireworks/fireworks_proc.c > create mode 100644 sound/firewire/fireworks/fireworks_stream.c > create mode 100644 sound/firewire/fireworks/fireworks_transaction.c >

2 3

[alsa-devel] [PATCH] ALSA: emu10k1: Fix possible NULL dereference
by Takashi Iwai 05 Mar '14

05 Mar '14

The previous dev_err() conversion resulted in a code that may give NULL dereference in snd_emu10k1_ptr_write(). Since it's a sanity check, better to be replaced with a debug macro like other places in this driver. Fixes: 6f002b02166c ('ALSA: emu10k1: Use standard printk helpers') Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/emu10k1/io.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index 81c75b4528d9..706b4f0c6806 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -71,11 +71,8 @@ void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned i unsigned long flags; unsigned int mask; - if (!emu) { - dev_err(emu->card->dev, "ptr_write: emu is null!\n"); - dump_stack(); + if (snd_BUG_ON(!emu)) return; - } mask = emu->audigy ? A_PTR_ADDRESS_MASK : PTR_ADDRESS_MASK; regptr = ((reg << 16) & mask) | (chn & PTR_CHANNELNUM_MASK); -- 1.9.0

1 0

[alsa-devel] [patch] ALSA: lola: NULL deref on allocation error
by Dan Carpenter 05 Mar '14

05 Mar '14

"chip" is NULL here. We don't need a printk here because kmalloc() has it built in. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index d63434d8c3fb..68824cdd137d 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -586,7 +586,6 @@ static int lola_create(struct snd_card *card, struct pci_dev *pci, chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (!chip) { - dev_err(chip->card->dev, "cannot allocate chip\n"); pci_disable_device(pci); return -ENOMEM; }

2 1

Re: [alsa-devel] ALSA: emu10k1: Use standard printk helpers
by Dan Carpenter 05 Mar '14

05 Mar '14

Hello Takashi Iwai, The patch 6f002b02166c: "ALSA: emu10k1: Use standard printk helpers" from Feb 25, 2014, leads to the following static checker warning: sound/pci/emu10k1/io.c:75 snd_emu10k1_ptr_write() error: we previously assumed 'emu' could be null (see line 74) sound/pci/emu10k1/io.c 68 void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data) 69 { 70 unsigned int regptr; 71 unsigned long flags; 72 unsigned int mask; 73 74 if (!emu) { ^^^^ 75 dev_err(emu->card->dev, "ptr_write: emu is null!\n"); ^^^^^^^^^^^^^^ Dereference. 76 dump_stack(); 77 return; 78 } regards, dan carpenter

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Add missing loopback merge path for AD1884/1984 codecs
by Takashi Iwai 05 Mar '14

05 Mar '14

The mixer widget (NID 0x20) of AD1884 and AD1984 codecs isn't connected directly to the actual I/O paths but only via another mixer widget (NID 0x21). We need a similar fix as we did for AD1882. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index df3652ad15ef..498971095869 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1092,6 +1092,7 @@ static int patch_ad1884(struct hda_codec *codec) spec = codec->spec; spec->gen.mixer_nid = 0x20; + spec->gen.mixer_merge_nid = 0x21; spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); -- 1.9.0

1 1

[alsa-devel] Xonar DG backports
by Clemens Ladisch 05 Mar '14

05 Mar '14

Hi Roman, I backporting the bugfixes in your patches to the stable kernels. However, I'm unable to see what was changed to enable the headphone outputs. What particular thing did the old driver wrong? Regards, Clemens

2 1

[alsa-devel] Audio-Bug
by Silver Death 05 Mar '14

05 Mar '14

Hi there, many people have this bug: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1200925 I would fix it, but I have no knowledge in C and about the kernel internal. I also didn't found any documentation which I could use to translate the datasheet into the source file. So I gentle ask you, if you could as a maintainer fix this bug or at least give me some information/instruction/documentation. Thanks in advance. With best regards, Simon

1 0

Re: [alsa-devel] Question on compress offload framework memory corruption
by Banajit Goswami 05 Mar '14

05 Mar '14

Hi Liam, > On Thu, Dec 19, 2013 at 10:55:08AM +0000, Mark Brown wrote: > > On Wed, Dec 18, 2013 at 01:17:51PM -0000, gsantosh(a)codeaurora.org wrote: > > > Hi, > > > > > > I have following questions in the compressed offload framework API. > > > > > > static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, > > > struct snd_compr_params *params) { ... > > > hw_params = kzalloc(sizeof(*hw_params), GFP_KERNEL); > > > if (hw_params == NULL) > > > return -ENOMEM; > > > /*1st question is, what is the use of above allocated memory I do > > > not see this being used in this function*/ > > > > The code you're quoting there isn't in mainline. I've also added > > Vinod to the CCs, you probably want to include him in any discussion > > of the compressed API. > > That's right, this is from DPCM compressed updated done by Liam, so he is > the best person to comment on this bit, and somehow his address in CC has > gone bad, i have added him in to list > But yes looking at the source, i agree that two are interlinked. > I guess we need to poupulate the hw_params and then use that to copy, that > would be the right thing to do here and yes this is not correct and not > sure how this is working... Continuing the discussion on the "memcpy" function in Santosh's 2nd question- > > > memcpy(&fe->dpcm[fe_substream->stream].hw_params, params, > > > sizeof(struct snd_pcm_hw_params)); > > > > > > /* 2nd question is > > > in above memcpy there is parameter mismatch > > > &fe->dpcm[fe_substream->stream].hw_params is of structure type struct > > > snd_pcm_hw_params > > > > > > params argument is of the structure type struct snd_compr_params > > > the definition of the two structures are different, how this is > > > working? > > > this will over right the destination with junk value which will return > > > in error when this memory is accessed, after memcpy this cpu_dai > > > hw_params > > > returing with EINVAL error, please explain why this is done this way. > > > */ I can see that a later version of the patch adds the function soc_compr_set_params_fe() at- http://permalink.gmane.org/gmane.linux.alsa.devel/117716 with the below code replaces the "memcpy" function with a "memset"- + /* + * Create an empty hw_params for the BE as the machine driver must + * fix this up to match DSP decoder and ASRC configuration. + * I.e. machine driver fixup for compressed BE is mandatory. + */ + memset(&fe->dpcm[fe_substream->stream].hw_params, 0, + sizeof(struct snd_pcm_hw_params)); It's evident that the code relies on machine driver fixup function to populate the parameters of hw_params before propagating those to back-end. I think rather than that, like Vinod also mentioned, the hw_params should have been populated with individual parameters which are relevant. Let me know your comment on this. Thanks, Banajit The Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation

2 1

[alsa-devel] Update patch for codecs alias name for Dell
by Kailang 05 Mar '14

05 Mar '14

Hi Takashi, Dell assigned more codecs for alias name. I update in attach patch. Thanks. Kailang

3 7

[alsa-devel] [PATCH v5 0/7] ASoC: add CSR SiRFSoC sound drivers
by RongJun Ying 05 Mar '14

05 Mar '14

From: Rongjun Ying <rongjun.ying(a)csr.com> This patchset adds CSR SiRFSoC sound drivers including: 1. The SiRF internal audio codec driver 2. The SiRF audio port driver 3. The SiRF internal audio card driver 4. The SiRF I2S controller driver 5. The SiRF hdmi card driver 6. The SiRF USP controller(DSP mode) driver 7. The SiRF bt-sco card driver -v5: 1. Drop all inlines. 2. Reordering the Kconfig and Makefile 3. Remove the sirf_audio_codec_reg_bits struct, use the new controls instead it. 4. Add some SND_SOC_DAPM_OUT_DRV instead of HP and SPK enable driver 5. Add audio codec clock supply instead of adc event callback 6. Fixed playback and capture can't concurrent work bug. Rongjun Ying (7): ASoC: sirf: Add SiRF internal audio codec driver ASoC: sirf: Add SiRF audio port driver is used by SiRF internal audio codec ASoC: sirf: Add SiRF audio card ASoC: sirf: Add SiRF I2S driver ASoC: sirf: Add hdmi card ASoC: sirf: Add usp driver which is used by dsp mode ASoC: sirf: Add bt-sco card .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 + .../devicetree/bindings/sound/sirf-audio-port.txt | 20 + .../devicetree/bindings/sound/sirf-audio.txt | 41 ++ .../devicetree/bindings/sound/sirf-bt-sco.txt | 18 + .../devicetree/bindings/sound/sirf-hdmi.txt | 15 + .../devicetree/bindings/sound/sirf-i2s.txt | 27 + .../devicetree/bindings/sound/sirf-usp.txt | 27 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sirf-audio-codec.c | 534 ++++++++++++++++++++ sound/soc/codecs/sirf-audio-codec.h | 75 +++ sound/soc/sirf/Kconfig | 34 ++ sound/soc/sirf/Makefile | 13 + sound/soc/sirf/sirf-audio-port.c | 194 +++++++ sound/soc/sirf/sirf-audio-port.h | 62 +++ sound/soc/sirf/sirf-audio.c | 156 ++++++ sound/soc/sirf/sirf-bt-sco.c | 99 ++++ sound/soc/sirf/sirf-hdmi.c | 98 ++++ sound/soc/sirf/sirf-i2s.c | 434 ++++++++++++++++ sound/soc/sirf/sirf-i2s.h | 88 ++++ sound/soc/sirf/sirf-usp.c | 448 ++++++++++++++++ sound/soc/sirf/sirf-usp.h | 282 +++++++++++ 24 files changed, 2691 insertions(+), 1 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-port.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-bt-sco.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-hdmi.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio-port.c create mode 100644 sound/soc/sirf/sirf-audio-port.h create mode 100644 sound/soc/sirf/sirf-audio.c create mode 100644 sound/soc/sirf/sirf-bt-sco.c create mode 100644 sound/soc/sirf/sirf-hdmi.c create mode 100644 sound/soc/sirf/sirf-i2s.c create mode 100644 sound/soc/sirf/sirf-i2s.h create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h -- 1.7.5.4

2 8

[alsa-devel] [PATCH 00/15] ASoC: fsl-ssi: Driver cleanup
by Markus Pargmann 05 Mar '14

05 Mar '14

Hi, Here is a series of cleanups for the fsl-ssi driver. It would be great if someone with PPC hardware and this fsl-ssi unit could please test this series. I am not sure if the regmap patch works as expected on PPC as well. I tested this series on imx53 quick starter board and a custom board. Both with SSI as I2S slave. Regards, Markus Markus Pargmann (15): ASoC: fsl-ssi: Use regmap ASoC: fsl-ssi: Remove fsl_ssi_setup ASoC: fsl-ssi: Move debugging to seperate file ASoC: fsl-ssi: Use dev_name for DAI driver struct ASoC: fsl-ssi: Move imx-specific probe to seperate function ASoC: fsl-ssi: Remove useless DMA code ASoC: fsl-ssi: baud clock error handling ASoC: fsl-ssi: Cleanup probe function ASoC: fsl-ssi: Remove unnecessary variables from ssi_private ASoC: fsl-ssi: reorder and document fsl_ssi_private ASoC: fsl-ssi: Fix register values when disabling ASoC: fsl-ssi: Only enable baudclk when used ASoC: fsl-ssi: Set default dai-fmts ASoC: fsl-ssi: Transmit enable synchronization ASoC: fsl-ssi: Update binding documentation .../devicetree/bindings/sound/fsl,ssi.txt | 21 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_ssi.c | 1167 ++++++++------------ sound/soc/fsl/fsl_ssi.h | 112 +- sound/soc/fsl/fsl_ssi_dbg.c | 163 +++ 6 files changed, 750 insertions(+), 716 deletions(-) create mode 100644 sound/soc/fsl/fsl_ssi_dbg.c -- 1.8.5.3

4 21

[alsa-devel] [PATCH 00/10] Simplify the code of TDM slot setting
by Xiubo Li 05 Mar '14

05 Mar '14

Xiubo Li (10): ASoC: core: Add snd_soc_dai_set_tdm_slot_xlate(). ASoC: fsl-utils: Add fsl_asoc_of_xlate_tdm_slot_mask() support. ASoC: fsl-esai: Add .of_xlate_tdm_slot_mask() support. ASoC: fsl-ssi: Add .of_xlate_tdm_slot_mask() support. ASoC: imx-ssi: Add .of_xlate_tdm_slot_mask() support. ASoC: simple-card: Use snd_soc_dai_set_tdm_slot_xlate() ASoC: blackfin: bf5xx-ad1836: Use snd_soc_dai_set_tdm_slot_xlate() ASoC: blackfin: bf5xx-ad193x: Use snd_soc_dai_set_tdm_slot_xlate() ASoC: fsl: eukrea-tlv320: Use snd_soc_dai_set_tdm_slot_xlate() ASoC: fsl: wm1133-ev1: Use snd_soc_dai_set_tdm_slot_xlate() include/sound/soc-dai.h | 3 +++ sound/soc/blackfin/bf5xx-ad1836.c | 2 +- sound/soc/blackfin/bf5xx-ad193x.c | 4 ++-- sound/soc/fsl/Kconfig | 3 +++ sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 ++ sound/soc/fsl/fsl_ssi.c | 2 ++ sound/soc/fsl/fsl_utils.c | 27 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_utils.h | 4 +++- sound/soc/fsl/imx-ssi.c | 2 ++ sound/soc/fsl/wm1133-ev1.c | 11 +---------- sound/soc/generic/simple-card.c | 7 +++---- sound/soc/soc-core.c | 33 ++++++++++++++++++++++++++++----- 13 files changed, 78 insertions(+), 24 deletions(-) -- 1.8.4

4 14

[alsa-devel] [PATCH] ASoC: simple-card: tidyup cpu/codec dai_fmt settings for non-DT
by Kuninori Morimoto 05 Mar '14

05 Mar '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> 30d0341e7da0c012f64fb290dd1153360fb49a8d (ASoC: simple-card: simplify the daifmt code) simplify cpu/codec dai_fmt which consists from dai specific format + common format. But, it didn't care about non-DT case. This patch fixes it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 034a2b7..7cabcc5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -307,6 +307,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) sizeof(priv->cpu_dai)); memcpy(&priv->codec_dai, &cinfo->codec_dai, sizeof(priv->codec_dai)); + + priv->cpu_dai.fmt |= cinfo->daifmt; + priv->codec_dai.fmt |= cinfo->daifmt; } /* -- 1.7.9.5

2 1

[alsa-devel] [PATCH v4-resend 0/7] ASoC: add CSR SiRFSoC sound drivers
by RongJun Ying 05 Mar '14

05 Mar '14

From: Rongjun Ying <rongjun.ying(a)csr.com> This patchset adds CSR SiRFSoC sound drivers including: 1. The SiRF internal audio codec driver 2. The SiRF audio port driver 3. The SiRF internal audio card driver 4. The SiRF I2S controller driver 5. The SiRF hdmi card driver 6. The SiRF USP controller(DSP mode) driver 7. The SiRF bt-sco card driver -v4: 1. Use the generic dmaengine pcm codes instead of sirf pcm driver. 2. Add TLV information into internal audio codec driver. 3. Use the MMIO regmap instead of custom read and write callbacks. 4. Remove spin_lock. 5. Change some variable names. 6. The SiRF internal audio driver split into codec, port. 7. Add binding documents. Rongjun Ying (7): ASoC: sirf: Add SiRF internal audio codec driver ASoC: sirf: Add SiRF audio port driver is used by SiRF internal audio codec ASoC: sirf: Add SiRF audio card ASoC: sirf: Add SiRF I2S driver ASoC: sirf: Add hdmi card ASoC: sirf: Add usp driver which is used by dsp mode ASoC: sirf: Add bt-sco card .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 + .../devicetree/bindings/sound/sirf-audio-port.txt | 20 + .../devicetree/bindings/sound/sirf-audio.txt | 41 ++ .../devicetree/bindings/sound/sirf-bt-sco.txt | 18 + .../devicetree/bindings/sound/sirf-hdmi.txt | 15 + .../devicetree/bindings/sound/sirf-i2s.txt | 27 + .../devicetree/bindings/sound/sirf-usp.txt | 27 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sirf-audio-codec.c | 585 ++++++++++++++++++++ sound/soc/codecs/sirf-audio-codec.h | 75 +++ sound/soc/sirf/Kconfig | 34 ++ sound/soc/sirf/Makefile | 13 + sound/soc/sirf/sirf-audio-port.c | 194 +++++++ sound/soc/sirf/sirf-audio-port.h | 62 ++ sound/soc/sirf/sirf-audio.c | 156 ++++++ sound/soc/sirf/sirf-bt-sco.c | 99 ++++ sound/soc/sirf/sirf-hdmi.c | 98 ++++ sound/soc/sirf/sirf-i2s.c | 438 +++++++++++++++ sound/soc/sirf/sirf-i2s.h | 88 +++ sound/soc/sirf/sirf-usp.c | 448 +++++++++++++++ sound/soc/sirf/sirf-usp.h | 282 ++++++++++ 24 files changed, 2745 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-port.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-bt-sco.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-hdmi.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio-port.c create mode 100644 sound/soc/sirf/sirf-audio-port.h create mode 100644 sound/soc/sirf/sirf-audio.c create mode 100644 sound/soc/sirf/sirf-bt-sco.c create mode 100644 sound/soc/sirf/sirf-hdmi.c create mode 100644 sound/soc/sirf/sirf-i2s.c create mode 100644 sound/soc/sirf/sirf-i2s.h create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h -- 1.7.5.4

4 18

[alsa-devel] [PATCH] ASoC: pcm: free path list before exiting from error conditions
by Patrick Lai 05 Mar '14

05 Mar '14

dpcm_path_get() allocates dynamic memory to hold path list. Corresponding dpcm_path_put() must be called to free the memory. dpcm_path_put() is not called under several error conditions. This leads to memory leak. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 47e1ce7..28522bd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1989,6 +1989,7 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); if (paths < 0) { + dpcm_path_put(&list); dev_warn(fe->dev, "ASoC: %s no valid %s path\n", fe->dai_link->name, "playback"); mutex_unlock(&card->mutex); @@ -2018,6 +2019,7 @@ capture: paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); if (paths < 0) { + dpcm_path_put(&list); dev_warn(fe->dev, "ASoC: %s no valid %s path\n", fe->dai_link->name, "capture"); mutex_unlock(&card->mutex); @@ -2082,6 +2084,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) fe->dpcm[stream].runtime = fe_substream->runtime; if (dpcm_path_get(fe, stream, &list) <= 0) { + dpcm_path_put(&list); dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); } -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, hosted by The Linux Foundation

2 1

[alsa-devel] [RFC][PATCH 0/3] Remove set_cache_io entirely from ASoC probe
by Xiubo Li 05 Mar '14

05 Mar '14

Xiubo Li (3): ASoC: core: Move the default regmap I/O setting to snd_soc_register_codec() ASoc: codec: remove the set_cache_io() entirely from CODEC ASoC probe. ASoC: core: remove the default regmap I/O setting up from soc_probe_codec() include/sound/soc.h | 3 +++ sound/soc/codecs/cq93vc.c | 6 ++++-- sound/soc/codecs/mc13783.c | 10 ++-------- sound/soc/codecs/wm5102.c | 6 ++---- sound/soc/codecs/wm5110.c | 6 ++---- sound/soc/codecs/wm8350.c | 6 ++++-- sound/soc/codecs/wm8400.c | 6 ++++-- sound/soc/codecs/wm8994.c | 4 ++-- sound/soc/codecs/wm8997.c | 7 ++----- sound/soc/soc-core.c | 21 +++++++++++---------- 10 files changed, 36 insertions(+), 39 deletions(-) -- 1.8.4

3 5

[alsa-devel] [PATCH] ASoC: ak4104: Remove superfluous codec->control_data initialization
by Lars-Peter Clausen 05 Mar '14

05 Mar '14

The driver uses automatic IO setup, which will also initialize the control_data field of the CODEC, no need to do it manually. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ak4104.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index b4819dc..10adf25 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -174,8 +174,6 @@ static int ak4104_probe(struct snd_soc_codec *codec) struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = ak4104->regmap; - /* set power-up and non-reset bits */ ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, -- 1.8.0

3 2

[alsa-devel] [PATCH] ASoC: io: Remove hw_read() operation
by Mark Brown 05 Mar '14

05 Mar '14

From: Mark Brown <broonie(a)linaro.org> We now no longer have any users of hw_read() in the kernel so remove the code in order to prevent any new users being added. Users should be using regmap. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc.h | 1 - sound/soc/soc-io.c | 13 ------------- 2 files changed, 14 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 56c4c71..dc36331 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -706,7 +706,6 @@ struct snd_soc_codec { /* codec IO */ void *control_data; /* codec control (i2c/3wire) data */ hw_write_t hw_write; - unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); void *reg_cache; diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index c3b6a0a..3c1de9c 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -26,18 +26,6 @@ static int hw_write(struct snd_soc_codec *codec, unsigned int reg, return regmap_write(codec->control_data, reg, value); } -static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) -{ - int ret; - unsigned int val; - - ret = regmap_read(codec->control_data, reg, &val); - if (ret == 0) - return val; - else - return -1; -} - /** * snd_soc_codec_set_cache_io: Set up standard I/O functions. * @@ -64,7 +52,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, int ret; codec->write = hw_write; - codec->read = hw_read; switch (control) { case SND_SOC_REGMAP: -- 1.9.0

2 6

[alsa-devel] [PATCH] ASoC: fsl: Don't select FIQ from Eukrea
by Mark Brown 05 Mar '14

05 Mar '14

From: Mark Brown <broonie(a)linaro.org> There is no point in using FIQ if DMA is available (it is selected) and selecting FIQ currently breaks the build on non-i.MX platforms. If FIQ is actually required the build will need to be restricted or have a select for the relevant FIQ code adding. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index a7d6ab6..b244830 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -175,7 +175,6 @@ config SND_SOC_EUKREA_TLV320 || (OF && ARM) depends on I2C select SND_SOC_TLV320AIC23 - select SND_SOC_IMX_PCM_FIQ select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI select SND_SOC_FSL_SSI -- 1.9.0

1 0

[alsa-devel] [PATCH] ASoC: rsnd: Fix warnings due to improper printk formats
by Laurent Pinchart 04 Mar '14

04 Mar '14

Use the %pap printk specifier to print resource_size_t variables. This fixes warnings on platforms where resource_size_t has a different size than int. Signed-off-by: Laurent Pinchart <laurent.pinchart+renesas(a)ideasonboard.com> --- sound/soc/sh/rcar/gen.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index add088b..f505b1f 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -283,13 +283,13 @@ static int rsnd_gen2_probe(struct platform_device *pdev, return ret; dev_dbg(dev, "Gen2 device probed\n"); - dev_dbg(dev, "SRU : %08x => %p\n", scu_res->start, + dev_dbg(dev, "SRU : %pap => %p\n", &scu_res->start, gen->base[RSND_GEN2_SCU]); - dev_dbg(dev, "ADG : %08x => %p\n", adg_res->start, + dev_dbg(dev, "ADG : %pap => %p\n", &adg_res->start, gen->base[RSND_GEN2_ADG]); - dev_dbg(dev, "SSIU : %08x => %p\n", ssiu_res->start, + dev_dbg(dev, "SSIU : %pap => %p\n", &ssiu_res->start, gen->base[RSND_GEN2_SSIU]); - dev_dbg(dev, "SSI : %08x => %p\n", ssi_res->start, + dev_dbg(dev, "SSI : %pap => %p\n", &ssi_res->start, gen->base[RSND_GEN2_SSI]); return 0; -- Regards, Laurent Pinchart

1 0