April 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: core: Fail codec probe if we fail to add controls
by Nariman Poushin 02 Apr '14

02 Apr '14

Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 35c0e71..7f9ae6c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1127,9 +1127,17 @@ static int soc_probe_codec(struct snd_soc_card *card, soc_init_codec_debugfs(codec); - if (driver->dapm_widgets) - snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, - driver->num_dapm_widgets); + if (driver->dapm_widgets) { + ret = snd_soc_dapm_new_controls(&codec->dapm, + driver->dapm_widgets, + driver->num_dapm_widgets); + + if (ret != 0) { + dev_err(codec->dev, + "Failed to create new controls %d\n", ret); + goto err_probe; + } + } /* Create DAPM widgets for each DAI stream */ list_for_each_entry(dai, &codec->component.dai_list, list) { -- 1.8.5.3

2 2

Re: [alsa-devel] Invalid PCM pointer with CMI8738 and pcm.c buffer underuns
by Clemens Ladisch 02 Apr '14

02 Apr '14

Andoru wrote: > The last time you replied you said that it's a hardware error. If it > is that way then how come the card otherwise works flawlessly in > Windows, and this doesn't happen on Debian when watching videos or > with certain programs? It's still a hardware error, but, apparently, it happens only when the Linux driver accesses it in a certain way. Which programs work? Which do not? Regards, Clemens

3 3

[alsa-devel] [PATCH bisect 0/2] ASoC: fsl_sai: Overwrite trigger()
by Nicolin Chen 02 Apr '14

02 Apr '14

This series of patches are bisected from the preivous version so as to apply the PATCH-1 onto asoc-v3.15-4. * The patches are generated by using '-U2' because the default '-U3' would conflict the baseline without fsl_sai_isr patches. Nicolin Chen (2): ASoC: fsl_sai: Fix buggy configurations in trigger() ASoC: fsl_sai: Separately enable interrupts for Tx and Rx streams sound/soc/fsl/fsl_sai.c | 43 +++++++++++++++++++++++-------------------- sound/soc/fsl/fsl_sai.h | 11 +++++++++++ 2 files changed, 34 insertions(+), 20 deletions(-) -- 1.8.4

2 9

[alsa-devel] pulseaudio and rcar-h2 sound
by Ben Dooks 01 Apr '14

01 Apr '14

I am currently investigating an issue with the rcar-h2 audio driver and a hang when running pulse-audio. From what I can see the issue seems to be an interaction with the driver and the regmap code. The current issue is that whatever is happening causes CPU0 to stall, thus nothing seems to be able to continue. The following errors are detected by the rcu code: INFO: rcu_sched self-detected stall on CPU 0: (1 GPs behind) idle=415/2/0 softirq=3076/3077 (t=10500 jiffies g=520 c=519 q=51) I have been through the regmap code and the locking looks ok. I also enabled as much of the locking checks as I could and it does not produce any useful output as to why it would be deadlocking. I can run the system fine if I replace the rsnd_read, rsnd_write and rsnd_bset code with versions which do not directly use regmap. Does anyone have any experience with this, or any further ideas on how to continue debugging? -- Ben Dooks http://www.codethink.co.uk/ Senior Engineer Codethink - Providing Genius

1 0

Re: [alsa-devel] twl4030 latency update
by Peter Ujfalusi 01 Apr '14

01 Apr '14

Hi Leonardo, On 04/01/2014 12:41 PM, Leonardo Gabrielli wrote: > Dear Peter, > I actually managed to nearly halve the latency with McBSP2 and jackd with a > little trick: requesting 4 channels audio. Of course two will be zero, but the > FIFO will grow up more quick. Yes, this is expected. The FIFO is lost (or word) based. If you have mono audio, it is 1280 samples long, in stereo it is 640 samples with 4 channel it can hold up to 320 samples. > Outcome: > 4 channels, 44100, 64 input --> jack --> output takes 10.1ms latency, i.e.: > - 2 periods for input: 128 L/R frames > - 1280 FIFO words / 4 channels = 320 frames > Without the trick the latency was 17.4ms :) What you can also try is to reduce the max_rx_thres to let's say 10 samples (20 or 40 depending on the number of channels you are using). Keep the max_tx_thres as you have been using. > > CPU load as increased slightly in jackd, but of course any other jack client > will stay the same assumed that the 2 fake extra channels are unused. > > In this experiment sample size = 16-bit. If I try to default to 32bit nothing > changes (so I assume the FIFO words are 32-bit wide and can contain either > 16-bit zeropadded samples or 32, which I remember from the AM37xx tech guide > but I'm too lazy to check again ;) ) Yes it is like that. > Glitches may happen when some regular user tasks (scp, curl, ...) request some > cpu. I may try the new 3.14 kernel with SCHED_DEADLINE to see if jackd and > clients can really be not preempted by user tasks... > > Best > > Leonardo > > > > On 26/03/2014 13:51, Peter Ujfalusi wrote: >> On 03/26/2014 11:45 AM, Leonardo Gabrielli wrote: >>> On 26/03/2014 09:26, Peter Ujfalusi wrote: >>>> The McBSP2 FIFO will be always there. There's nothing can be done on that. >>>> The >>>> size on McBSP2 is 1280 words -> 640 stereo samples, ie ~29ms with 22050, >>>> 14.5ms with 44100. >>>> >>>> If you are staying in element mode this means that it is granted that the >>>> sample at the DMA pointer will out on the i2s line about the mentioned times. >>>> This is the delay caused by the FIFO itself. From where the rest is coming >>>> I'm >>>> not really sure. >>> BTW: I forgot to mention: the latency listed in my previous email is >>> input+output (i.e. I record pulses from the beagleboard input jack and the >>> delayed version to the beagleboard output jack). The twl4030 analog and >>> digital loopback features have been of course disabled, in order to get the >>> total latency due from A/D to D/A. >> This means that the McBSP latency in worst case is 1280 + selected rx >> threshold in words (so /2 in case of stereo.) If you lower the rx threshold >> you decrease the latency on the capture side. On the playback side there's >> nothing can be done. >> >>> So just to get confirm I understood the McBSP mechanism well: even though I >>> can transfer to/from DMA samples in bursts of <threshold> length, each sample >>> will always "travel along" the whole FIFO buffer length, (as if in a delay >>> line) and thus they will always have 640samples delay? >> On the playback side this is pretty much true. On capture side the threshold >> means that DMA will read from FIFO when threshold amount is available in it. >> >>> Would it be possible to workaround this, e.g. by putting 4-channel audio >>> frames instead of stereo frames in the FIFO (with 2 channels unused), in order >>> to fill up the FIFO more quickly and have less latency? Or is it pure craze? >> >From the FIFO McBSP takes data word by word. If you play stereo, you need to >> have stereo data in the FIFO. You can not skip two words with McBSP. >> >> The thing I tried for playback and did not worked AFAIR: >> In general the idea was to configure DMA to send threshold/channel to every >> request while configuring the McBSP threshold register to be 1280 - threshold. >> In case of threshold 80 (40 stereo samples) it would play out: >> transfer 40 samples to FIFO per DMA request >> assert the DMA request when we have space for 1260 (630 samples). The number >> is just a guess, keeping 10 samples in FIFO sounds safe enough >> This would keep the FIFO fill between 10 and 50 samples. >> But this does not work, I think McBSP is counting the received words also and >> deasserts the DMA request based on this count and not the FIFO level. >> >> Another thing which would be even more complicated is to play with the McBSP >> threshold runtime. With the same 40 sample: >> DMA is to transfer 40 samples per DMA requests. >> start >> 1. McBSP threshold to 80 >> 2. in dma interrupt callback McBSP threshold to 1260 >> 3. in McBSP warning interrupt (that we will be reaching the threshold soon) >> back to 80 >> 4. goto 2 >> >> If we could do the step between 3 and 4 within one sample time this might work >> but as soon as you are late the thing will fail. >> >> I know this is working in realtime systems like in DSPs and non linux >> systems... >> > -- Péter

1 0

[alsa-devel] [PATCH] ASoC: core: Fail probe if we fail to add dai widgets
by Nariman Poushin 01 Apr '14

01 Apr '14

Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b322cf2..5cf7107 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1132,8 +1132,15 @@ static int soc_probe_codec(struct snd_soc_card *card, driver->num_dapm_widgets); /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(dai, &codec->component.dai_list, list) - snd_soc_dapm_new_dai_widgets(&codec->dapm, dai); + list_for_each_entry(dai, &codec->component.dai_list, list) { + ret = snd_soc_dapm_new_dai_widgets(&codec->dapm, dai); + + if (ret != 0) { + dev_err(codec->dev, + "Failed to create DAI widgets %d\n", ret); + goto err_probe; + } + } codec->dapm.idle_bias_off = driver->idle_bias_off; -- 1.8.5.3

2 1

[alsa-devel] [PATCH] ASoC: alc5632: fix uninit var in alc5632_probe()
by Stephen Warren 01 Apr '14

01 Apr '14

From: Stephen Warren <swarren(a)nvidia.com> alc5632_probe() returns ret, yet it is not initialized or set anywhere. This ends up causing the function to appear to fail, and audio not to work on the Toshiba AC100, with my compiler at least. This function used to set ret in all cases, but recent cleanup removed that. Cc: Xiubo Li <Li.Xiubo(a)freescale.com> Fixes: 5d6be5aa6bec ("ASoC: codec: Simplify ASoC probe code.") Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/alc5632.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index ec071a6306ef..3ee8d4e41a99 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1061,7 +1061,6 @@ static int alc5632_resume(struct snd_soc_codec *codec) static int alc5632_probe(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); - int ret; /* power on device */ alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1075,7 +1074,7 @@ static int alc5632_probe(struct snd_soc_codec *codec) return -EINVAL; } - return ret; + return 0; } /* power down chip */ -- 1.8.1.5

3 2

[alsa-devel] [PATCH] ASoC: DAPM: Add support for multi register mux
by Arun Shamanna Lakshmi 01 Apr '14

01 Apr '14

Modify soc_enum struct to handle pointers for reg and mask. Add dapm get and put APIs for multi register mux with one hot encoding. Signed-off-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> --- include/sound/soc-dapm.h | 10 ++++ include/sound/soc.h | 22 +++++-- sound/soc/soc-core.c | 12 ++-- sound/soc/soc-dapm.c | 143 +++++++++++++++++++++++++++++++++++++++++----- 4 files changed, 162 insertions(+), 25 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ef78f56..983b0ab 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -305,6 +305,12 @@ struct device; .get = snd_soc_dapm_get_enum_double, \ .put = snd_soc_dapm_put_enum_double, \ .private_value = (unsigned long)&xenum } +#define SOC_DAPM_ENUM_WIDE(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_wide, \ + .put = snd_soc_dapm_put_enum_wide, \ + .private_value = (unsigned long)&xenum } #define SOC_DAPM_ENUM_VIRT(xname, xenum) \ SOC_DAPM_ENUM(xname, xenum) #define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \ @@ -378,6 +384,10 @@ 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_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_put_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, diff --git a/include/sound/soc.h b/include/sound/soc.h index 5878410..5c274c4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -177,18 +177,23 @@ {.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 = &(int){(xreg)}, .shift_l = xshift_l, .shift_r = xshift_r, \ .items = xitems, .texts = xtexts, \ - .mask = xitems ? roundup_pow_of_two(xitems) - 1 : 0} + .mask = &(unsigned int){(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_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 = &(int){(xreg)}, .shift_l = xshift_l, .shift_r = xshift_r, \ + .mask = &(unsigned int){(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_VALUE_ENUM_WIDE(xregs, xmasks, xnum_regs, xitems, xtexts, xvalues) \ +{ .reg = xregs, .mask = xmasks, .num_regs = xnum_regs, \ + .items = xitems, .texts = xtexts, .values = xvalues, .reg_width = 32 } #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts) #define SOC_ENUM(xname, xenum) \ @@ -293,6 +298,9 @@ #define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \ const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \ ARRAY_SIZE(xtexts), xtexts, xvalues) +#define SOC_VALUE_ENUM_WIDE_DECL(name, xregs, xmasks, xnum_regs, xtexts, xvalues) \ + const struct soc_enum name = SOC_VALUE_ENUM_WIDE(xregs, xmasks, xnum_regs, \ + ARRAY_SIZE(xtexts), xtexts, xvalues) #define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \ @@ -1098,13 +1106,15 @@ struct soc_mreg_control { /* enumerated kcontrol */ struct soc_enum { - int reg; + int *reg; unsigned char shift_l; unsigned char shift_r; unsigned int items; - unsigned int mask; + unsigned int *mask; const char * const *texts; const unsigned int *values; + unsigned int reg_width; + unsigned int num_regs; }; /** diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cd52d52..aba0094 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; } @@ -2636,15 +2636,15 @@ 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); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c8a780d..4d2b35c 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 @@ -2903,15 +2903,15 @@ 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; } @@ -2945,25 +2945,25 @@ 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.reg = e->reg[0]; update.mask = mask; update.val = val; card->update = &update; @@ -2984,6 +2984,123 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); /** + * snd_soc_dapm_get_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 registers to set the enumerated items. The enumerated + * items are referred as values. + * Can be used for handling bit field coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_dapm_get_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, val, bit_pos = 0, reg_idx; + + 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->mask[reg_idx]; + if (val != 0) { + bit_pos = ffs(val) + (e->reg_width * reg_idx); + break; + } + } + + ucontrol->value.enumerated.item[0] = + snd_soc_enum_val_to_item(e, bit_pos); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_wide); + +/** + * snd_soc_dapm_put_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 registers to set the enumerated items. The enumerated + * items are referred as values. + * Can be used for handling bit field coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_dapm_put_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 *item = ucontrol->value.enumerated.item; + unsigned int change = 0, reg_idx = 0, value, bit_pos; + struct snd_soc_dapm_update update; + int ret = 0, reg_val = 0, i; + + if (item[0] >= e->items) + return -EINVAL; + + value = snd_soc_enum_item_to_val(e, item[0]); + + if (value) { + /* get the register index and value to set */ + reg_idx = (value - 1) / e->reg_width; + bit_pos = (value - 1) % e->reg_width; + reg_val = BIT(bit_pos); + } + + for (i = 0; i < e->num_regs; i++) { + if (i == reg_idx) { + change = snd_soc_test_bits(codec, e->reg[i], + e->mask[i], reg_val); + + } else { + /* accumulate the change to update the DAPM path + when none is selected */ + change += snd_soc_test_bits(codec, e->reg[i], + e->mask[i], 0); + + /* clear the register when not selected */ + snd_soc_write(codec, e->reg[i], 0); + } + } + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + if (change) { + update.kcontrol = kcontrol; + update.reg = e->reg[reg_idx]; + update.mask = e->mask[reg_idx]; + update.val = reg_val; + 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; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_wide); + +/** * snd_soc_dapm_info_pin_switch - Info for a pin switch * * @kcontrol: mixer control -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH] ASoC: Add support for multi register mux
by Songhee Baek 01 Apr '14

01 Apr '14

> -----Original Message----- > From: Songhee Baek > Sent: Friday, March 28, 2014 11:10 AM > To: 'Lars-Peter Clausen' > Cc: Arun Shamanna Lakshmi; 'lgirdwood(a)gmail.com'; 'broonie(a)kernel.org'; > 'swarren(a)wwwdotorg.org'; 'alsa-devel(a)alsa-project.org'; 'tiwai(a)suse.de'; > 'linux-kernel(a)vger.kernel.org' > Subject: RE: [alsa-devel] [PATCH] ASoC: Add support for multi register mux > > > > > 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. These get/put functions are updated from previous mail, now it works for multi register mux, please review these function whether I can add in current put/get function. #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); break; } } } 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, val, update_idx = 0; unsigned int mask; struct snd_soc_dapm_update update; int ret = 0, reg_val; if (item[0] >= e->items) return -EINVAL; val = 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; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l; mask |= e->mask[0] << e->shift_r; } if (e->num_regs < 2) { value = val; goto update_reg; } for (i = 0; i < e->num_regs; i++) { reg_val = val - 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.

4 6

Re: [alsa-devel] [PATCH bisect 2/2] ASoC: fsl_sai: Separately enable interrupts for Tx and Rx streams
by Nicolin Chen 01 Apr '14

01 Apr '14

On Tue, Apr 01, 2014 at 11:48:16AM +0800, Wang Dongsheng-B40534 wrote: > > > > -----Original Message----- > > From: Nicolin Chen [mailto:Guangyu.Chen@freescale.com] > > Sent: Tuesday, April 01, 2014 11:14 AM > > To: Wang Dongsheng-B40534 > > Cc: broonie(a)kernel.org; alsa-devel(a)alsa-project.org; Xiubo Li-B47053; linuxppc- > > dev(a)lists.ozlabs.org; linux-kernel(a)vger.kernel.org; timur(a)tabi.org > > Subject: Re: [PATCH bisect 2/2] ASoC: fsl_sai: Separately enable interrupts for > > Tx and Rx streams > > > > On Tue, Apr 01, 2014 at 11:25:02AM +0800, Wang Dongsheng-B40534 wrote: > > > > Subject: [PATCH bisect 2/2] ASoC: fsl_sai: Separately enable interrupts for > > Tx > > > > and Rx streams > > > > > > > > We only enable one side interrupt for each stream since over/underrun > > > > on the opposite stream would be resulted from what we previously did, > > > > enabling TERE but remaining FRDE disabled, even though the xrun on the > > > > opposite direction will not break the current stream. > > > > > > > > Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> > > > > Acked-by: Xiubo Li <Li.Xiubo(a)freescale.com> > > > > --- > > > > sound/soc/fsl/fsl_sai.c | 8 ++++++-- > > > > sound/soc/fsl/fsl_sai.h | 1 + > > > > 2 files changed, 7 insertions(+), 2 deletions(-) > > > > > > > > diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c > > > > index bdfd497..d64c33f 100644 > > > > --- a/sound/soc/fsl/fsl_sai.c > > > > +++ b/sound/soc/fsl/fsl_sai.c > > > > @@ -397,4 +397,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream > > > > *substream, int cmd, > > > > > > > > regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > > > + FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS); > > > > + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > > > FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); > > > > break; > > > > @@ -404,4 +406,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream > > > > *substream, int cmd, > > > > regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > > > FSL_SAI_CSR_FRDE, 0); > > > > + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > > > + FSL_SAI_CSR_xIE_MASK, 0); > > > > > > > > if (!(tcsr & FSL_SAI_CSR_FRDE || rcsr & FSL_SAI_CSR_FRDE)) { > > > > @@ -464,6 +468,6 @@ 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, FSL_SAI_FLAGS); > > > > - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, FSL_SAI_FLAGS); > > > > + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); > > > > + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); > > > > > > Why are you remove this macro? Don't use magic number. > > > > It's pretty clear that the so-called magic number is to clear the settings > > in the registers for driver init as what this driver did at the first place > > -- no offense but I don't think you would ask this if you check the git-log > > of the driver. > > > ~FSL_SAI_MASK is better than 0x0. And you also replace 0xffffffff. I would later send a patch to reset SAI for a true init instead of these lines but not within this patch as it's focusing on the interrupts enabling. So please don't grasp the mask here. Just let me continue. Thank you, Nicolin Chen

1 0