March 2012 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH v3] sound/soc/lapis: add platform driver for ML7213
by Mark Brown 13 Jun '12

13 Jun '12

On Tue, Mar 06, 2012 at 02:48:05PM +0900, Tomoya MORINAGA wrote: > 2012年3月3日1:39 Mark Brown <broonie(a)opensource.wolfsonmicro.com>: > > I just merged Lars-Peter's dmaengine library code which has been on the > > list for a week or so - this should be updated to use that next time > > it's posted. That should save a lot of code from the driver and make > > sure it's following best practices for dmaengine use. > Sorry, we can't use this library. This sort of response really isn't on. Do you have some reason for saying this? Flat out refusing to do things with no reason is not useful. > >> +static struct ioh_i2s_data *i2s_data; > >> +static struct ioh_i2s_dma dmadata[MAX_I2S_CH]; > > Why are these needed, aren't they dynamically allocated by the driver? > You mean ASoC driver can't use global variable ? In general no Linux driver should be using a global variable except for things like this. > >> + case SNDRV_PCM_FORMAT_S32_LE: > >> + byte = 24; > >> + break; > > That looks wrong... are you sure you don't support S24_LE or something? > This is correct. > Because maximum transmit size of ML7213's I2S hw is 24bit. Note we often lay out 24 bit audio in 32 bit blocks. > >> +static struct platform_driver ioh_i2s_driver_plat = { > > > >> +static struct platform_driver ioh_dai_driver_plat = { > > > >> +static int ioh_i2s_pci_probe(struct pci_dev *pdev, > >> + const struct pci_device_id *id) > > Why are you creating these platform devices? I don't understand the > > function they serve. The code handling them looks to have quite a few > > problems but I'm not clear they should be there in the first place. > if these platform devices aren't used, device detection doesn't work correctly. > So, I added these. You've not actually mentioned the problem you were seeing... > >> + rv = request_irq(pdev->irq, ioh_i2s_irq, IRQF_SHARED, "ml7213_ioh", > >> + pdev); > >> + if (rv != 0) { > >> + printk(KERN_ERR "Failed to allocate irq\n"); > >> + goto out_irq; > >> + } > > Are you *sure* you're ready to handle interrupts at this point? > This is just registering interrupt handler. > As long as interrupt register is not enabled, the interrupt handler is > not called. > This is common request_irq description. > What's is your concern ? Apart from anything else you've got the interrupt requested as IRQF_SHARED so the interrupt could get called at any time. It's also not clear that you've got the hardware in a known good state.

3 28

[alsa-devel] [PATCH] alsa-lib/tlv: fix handling of raw value ranges
by Benoît Thébaudeau 02 Jun '12

02 Jun '12

In case of a TLV dB range with all items having raw value ranges strictly within the main raw value range reported by the driver, snd_tlv_convert_from_dB() returned one of the main raw range boundaries, which was outside all dB range items. E.g.: static const unsigned int bd3753x_vol_fader_gain_att_tlv[] = { TLV_DB_RANGE_HEAD(2), 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), 48, 142, TLV_DB_SCALE_ITEM(-7900, 100, 0), }; SOC_SINGLE_TLV("Main Volume", BD3753X_VOL_GAIN, 0, 255, 1, bd3753x_vol_fader_gain_att_tlv), snd_tlv_convert_from_dB(tlv, 0, 255, 1500, &value, 1) returned 255 instead of the expected 142. or: static const unsigned int max9850_tlv[] = { TLV_DB_RANGE_HEAD(4), 0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0), 0x20, 0x33, TLV_DB_SCALE_ITEM(-4150, 200, 0), 0x34, 0x37, TLV_DB_SCALE_ITEM(-150, 100, 0), 0x38, 0x3f, TLV_DB_SCALE_ITEM(250, 50, 0), }; SOC_SINGLE_TLV("Headphone Volume", MAX9850_VOLUME, 0, 0x3f, 1, max9850_tlv), snd_tlv_convert_from_dB(tlv, 0, 0x3f, -7450, &value, 0) returned 0 instead of the expected 0x18. Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau(a)advansee.com> --- alsa-lib/src/control/tlv.c +++ alsa-lib/src/control/tlv.c @@ -291,41 +291,37 @@ int snd_tlv_convert_from_dB(unsigned int { switch (tlv[0]) { case SND_CTL_TLVT_DB_RANGE: { - long dbmin, dbmax, prev_rangemax; + long dbmin, dbmax, prev_submax; unsigned int pos, len; len = int_index(tlv[1]); - if (len > MAX_TLV_RANGE_SIZE) - return -EINVAL; - if (snd_tlv_get_dB_range(tlv, rangemin, rangemax, - &dbmin, &dbmax)) + if (len < 6 || len > MAX_TLV_RANGE_SIZE) return -EINVAL; - if (db_gain <= dbmin) { - *value = rangemin; - return 0; - } else if (db_gain >= dbmax) { - *value = rangemax; - return 0; - } pos = 2; - prev_rangemax = 0; - while (pos + 4 <= len) { - rangemin = (int)tlv[pos]; - rangemax = (int)tlv[pos + 1]; + prev_submax = 0; + do { + long submin, submax; + submin = (int)tlv[pos]; + submax = (int)tlv[pos + 1]; + if (rangemax < submax) + submax = rangemax; if (!snd_tlv_get_dB_range(tlv + pos + 2, - rangemin, rangemax, + submin, submax, &dbmin, &dbmax) && db_gain >= dbmin && db_gain <= dbmax) return snd_tlv_convert_from_dB(tlv + pos + 2, - rangemin, rangemax, + submin, submax, db_gain, value, xdir); else if (db_gain < dbmin) { - *value = xdir ? rangemin : prev_rangemax; + *value = xdir || pos == 2 ? submin : prev_submax; return 0; } - prev_rangemax = rangemax; + prev_submax = submax; + if (rangemax == submax) + break; pos += int_index(tlv[pos + 3]) + 4; - } - return -EINVAL; + } while (pos + 4 <= len); + *value = prev_submax; + return 0; } case SND_CTL_TLVT_DB_SCALE: { int min, step, max;

4 24

[alsa-devel] [PATCH] snd-usb-us122l: Fix missing NULL checks
by Karsten Wiese 27 Apr '12

27 Apr '12

Fix missing NULL checks in usb_stream_hwdep_poll() and usb_stream_hwdep_ioctl(). Wake up poll waiters before returning from usb_stream_hwdep_ioctl(). Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> Cc: stable(a)kernel.org --- sound/usb/usx2y/us122l.c | 41 ++++++++++++++++++++--------------------- 1 files changed, 20 insertions(+), 21 deletions(-) diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 91bb296..7abda8d 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -269,29 +269,26 @@ static unsigned int usb_stream_hwdep_poll(struct snd_hwdep *hw, struct file *file, poll_table *wait) { struct us122l *us122l = hw->private_data; - struct usb_stream *s = us122l->sk.s; unsigned *polled; unsigned int mask; poll_wait(file, &us122l->sk.sleep, wait); - switch (s->state) { - case usb_stream_ready: - if (us122l->first == file) - polled = &s->periods_polled; - else - polled = &us122l->second_periods_polled; - if (*polled != s->periods_done) { - *polled = s->periods_done; - mask = POLLIN | POLLOUT | POLLWRNORM; - break; + mask = POLLIN | POLLOUT | POLLWRNORM | POLLERR; + if (mutex_trylock(&us122l->mutex)) { + struct usb_stream *s = us122l->sk.s; + if (s && s->state == usb_stream_ready) { + if (us122l->first == file) + polled = &s->periods_polled; + else + polled = &us122l->second_periods_polled; + if (*polled != s->periods_done) { + *polled = s->periods_done; + mask = POLLIN | POLLOUT | POLLWRNORM; + } else + mask = 0; } - /* Fall through */ - mask = 0; - break; - default: - mask = POLLIN | POLLOUT | POLLWRNORM | POLLERR; - break; + mutex_unlock(&us122l->mutex); } return mask; } @@ -377,6 +374,7 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, { struct usb_stream_config *cfg; struct us122l *us122l = hw->private_data; + struct usb_stream *s; unsigned min_period_frames; int err = 0; bool high_speed; @@ -422,18 +420,18 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, snd_power_wait(hw->card, SNDRV_CTL_POWER_D0); mutex_lock(&us122l->mutex); + s = us122l->sk.s; if (!us122l->master) us122l->master = file; else if (us122l->master != file) { - if (memcmp(cfg, &us122l->sk.s->cfg, sizeof(*cfg))) { + if (!s || memcmp(cfg, &s->cfg, sizeof(*cfg))) { err = -EIO; goto unlock; } us122l->slave = file; } - if (!us122l->sk.s || - memcmp(cfg, &us122l->sk.s->cfg, sizeof(*cfg)) || - us122l->sk.s->state == usb_stream_xrun) { + if (!s || memcmp(cfg, &s->cfg, sizeof(*cfg)) || + s->state == usb_stream_xrun) { us122l_stop(us122l); if (!us122l_start(us122l, cfg->sample_rate, cfg->period_frames)) err = -EIO; @@ -444,6 +442,7 @@ unlock: mutex_unlock(&us122l->mutex); free: kfree(cfg); + wake_up_all(&us122l->sk.sleep); return err; } -- 1.7.2.3

2 5

Re: [alsa-devel] [PATCH] ASoC: sgtl5000: Enable VAG when DAC/ADC up
by Mark Brown 25 Apr '12

25 Apr '12

On Thu, Mar 29, 2012 at 03:17:00PM +0800, Zeng Zhaoming wrote: > So the code should be: > + {"MIC_IN", NULL, "Mic Bias}, /* mic bias --> mic_in */ > {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ > -+ {"Mic Bias", NULL, "MIC_IN"}, /* mic_in --> mic bias */ > -+ {"Capture Mux", "MIC_IN", "Mic Bias"}, /* mic bias --> adc_mux */ > and platform code > + {"Mic Bias", NULL, "MIC"}, > - {"MIC_IN", NULL, "MIC"}, No, the bias should be a supply for the microphone - ensuring that this happens is the whole point of the patch I sent the other day.

3 5

Re: [alsa-devel] [PATCH 0/6] snd-usb endpoint rework, version 6
by Felix Homann 12 Apr '12

12 Apr '12

Happy New Year! That being said, I've tested a kernel without Daniel's patch set. It doesn't work either. The error messages are only slightly different: Jan 1 11:52:31 weasel mtp-probe: checking bus 1, device 4: "/sys/devices/pci0000:00/0000:00:1a.7/usb1/1-1" Jan 1 11:52:31 weasel mtp-probe: bus: 1, device: 4 was not an MTP device Jan 1 11:52:31 weasel kernel: [ 161.880773] usbcore: registered new interface driver snd-usb-audio Jan 1 11:52:34 weasel kernel: [ 164.284980] cannot submit datapipe for urb 0, error -28: not enough bandwidth So, It's not related to Daniel's rework. But nevertheless, something's broken...

3 9

[alsa-devel] paravirtualized alsa kernel driver for XEN
by Stefano Panella 11 Apr '12

11 Apr '12

Hi, I am Stefano Panella, I am new to the list and I would like to take the opportunity to ask some questions since I am trying to write a paravirtualized alsa driver for XEN. If all goes well I would also like to upstream it on linux. I have been reading the documentation on "Writing an ALSA Driver" and I am still not completely clear on the meaning of the "pointer" callback in the pcm operations. The description say: "This callback is called when the PCM middle layer inquires the current hardware position on the buffer." My question are: 1) In case of a playback stream, is the pointer referring to wich sample is currently playing on the DAC or to which is it the last frame read by the HW from the alsa memory buffer? 2) What does the pointer mean in case of a capture stream? Is it the position of the current frame on the ADC or is the latest frame written into the alsa buffer? 3) in case it is the frame on the DAC/ADC, what happens if the callback does not return the real DAC/ADC frame position but an approximate value, let say rounded to 64 frames only? 4) is there any test I could run to check I have implemented correctly the "pointer" callback? Or any application which would need very high "pointer" precision like frame precision? Thanks very much in advance, Stefano

3 9

[alsa-devel] [RFC] Fix internal mic for Lenovo Ideapad U300s
by David Henningsson 07 Apr '12

07 Apr '12

The internal mic input is phase inverted on one channel. To avoid people in userspace summing the channels together and get zero result, use a separate mixer control for the inverted channel. BugLink: https://bugs.launchpad.net/bugs/903853 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Inverted Internal Mic, take 2. I refined my idea a little. By calling the left channel "Internal Mic" and the right channel "Inverted Internal Mic", we're giving the user a choice whether to activate it or keeping it muted. For PulseAudio, the quick fix is just to make sure the "Inverted Internal Mic" control is muted. If we in the long run want to actually handle this and swap the phase around, we now also have a reasonable indicator if we should do this, by checking if the "Inverted Internal Mic" control exists. For the ALSA init DB (i e, should the "Inverted Internal Mic" be initially muted?) that is no big deal for me, since this will be overwritten by PulseAudio later anyway. I'm also open for better names than "Inverted Internal Mic" if you have any. As with the previous version, this is a draft version of the patch, and could be done slightly more elegant I guess. What do you say? sound/pci/hda/patch_conexant.c | 45 +++++++++++++++++++++++++++++---------- 1 files changed, 33 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e6eafb1..cd3859b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -142,6 +142,7 @@ struct conexant_spec { unsigned int asus:1; unsigned int pin_eapd_ctrls:1; unsigned int single_adc_amp:1; + unsigned int fixup_stereo_dmic:1; unsigned int adc_switching:1; @@ -4107,9 +4108,9 @@ static int cx_auto_init(struct hda_codec *codec) static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, const char *dir, int cidx, - hda_nid_t nid, int hda_dir, int amp_idx) + hda_nid_t nid, int hda_dir, int amp_idx, int chs) { - static char name[32]; + static char name[44]; static struct snd_kcontrol_new knew[] = { HDA_CODEC_VOLUME(name, 0, 0, 0), HDA_CODEC_MUTE(name, 0, 0, 0), @@ -4119,7 +4120,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; - knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx, + knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, chs, amp_idx, hda_dir); knew[i].subdevice = HDA_SUBDEV_AMP_FLAG; knew[i].index = cidx; @@ -4138,7 +4139,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, } #define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \ - cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0) + cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0, 3) #define cx_auto_add_pb_volume(codec, nid, str, idx) \ cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT) @@ -4216,14 +4217,24 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, int i; for (i = 0; i < spec->num_adc_nids; i++) { + int chs = 3; hda_nid_t adc_nid = spec->adc_nids[i]; int idx = get_input_connection(codec, adc_nid, nid); if (idx < 0) continue; if (spec->single_adc_amp) idx = 0; + + if (spec->fixup_stereo_dmic && (strcmp("Internal Mic", label) == 0)) { + /* Make two independent kcontrols */ + int err = cx_auto_add_volume_idx(codec, "Inverted Internal Mic", pfx, + cidx, adc_nid, HDA_INPUT, idx, 2); + if (err < 0) + return err; + chs = 1; + } return cx_auto_add_volume_idx(codec, label, pfx, - cidx, adc_nid, HDA_INPUT, idx); + cidx, adc_nid, HDA_INPUT, idx, chs); } return 0; } @@ -4405,22 +4416,30 @@ static void apply_pincfg(struct hda_codec *codec, const struct cxt_pincfg *cfg) } -static void apply_pin_fixup(struct hda_codec *codec, +enum { + CXT_PINCFG_LENOVO_X200, + CXT_FIXUP_STEREO_DMIC, +}; + +static void apply_fixup(struct hda_codec *codec, const struct snd_pci_quirk *quirk, const struct cxt_pincfg **table) { + struct conexant_spec *spec = codec->spec; + quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (quirk) { + if (quirk && table[quirk->value]) { snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n", quirk->name); apply_pincfg(codec, table[quirk->value]); } + if (quirk->value == CXT_FIXUP_STEREO_DMIC) { + snd_printdd(KERN_INFO "hda_codec: applying internal mic workaround for %s\n", + quirk->name); + spec->fixup_stereo_dmic = 1; + } } -enum { - CXT_PINCFG_LENOVO_X200, -}; - static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ { 0x17, 0x21a11000 }, /* dock-mic */ @@ -4431,10 +4450,12 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { static const struct cxt_pincfg *cxt_pincfg_tbl[] = { [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200, + [CXT_FIXUP_STEREO_DMIC] = NULL, }; static const struct snd_pci_quirk cxt_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), + SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), {} }; @@ -4477,7 +4498,7 @@ static int patch_conexant_auto(struct hda_codec *codec) break; } - apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); + apply_fixup(codec, cxt_fixups, cxt_pincfg_tbl); /* Show mute-led control only on HP laptops * This is a sort of white-list: on HP laptops, EAPD corresponds -- 1.7.9.1

2 3

[alsa-devel] [PATCH 1/2] ASoC: fsl: remove redundant Kconfig option SND_SOC_FSL_SSI
by Shawn Guo 04 Apr '12

04 Apr '12

While commit 606d620 (ASoC: imx: merge sound/soc/imx into sound/soc/fsl) adds SND_SOC_FSL_SSI outside "menuconfig SND_POWERPC_SOC" to make it visible for both PowerPC and ARM/IMX, it forgot removing the one inside "menuconfig SND_POWERPC_SOC". Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/fsl/Kconfig | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index aa14fc9..d23d612 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -16,10 +16,6 @@ if SND_POWERPC_SOC config SND_MPC52xx_DMA tristate -config SND_SOC_FSL_SSI - tristate - depends on FSL_SOC - config SND_SOC_POWERPC_DMA tristate depends on FSL_SOC -- 1.7.5.4

2 2

[alsa-devel] [PATCH] ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV
by Brian Austin 03 Apr '12

03 Apr '12

Some codecs namely Cirrus Logic Codecs have a way of wrapping the dB scale around 0dB without 0dB being in the middle. Rework of SOC_DOUBLE_R_SX_TLV to be more consistent with other asoc tlv macros. Add single register macro : SOC_SINGLE_SX_TLV. Use snd_soc_info_volsw for .info Use snd_soc_get_volsw_sx, snd_soc_put_volsw_sx for single and double. kcontrols for CS42L51 and CS42L73 are adjusted to these new TLV Macros. The max value is determined by: (number of steps) +1 for 0dB +max from codec datasheet. Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- include/sound/soc.h | 48 +++++++----- sound/soc/codecs/cs42l51.c | 6 +- sound/soc/codecs/cs42l73.c | 26 +++--- sound/soc/soc-core.c | 174 ++++++++++++++++++++----------------------- 4 files changed, 125 insertions(+), 129 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2ebf787..1e17e04 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -55,6 +55,18 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_SINGLE_SX_TLV(xname, xreg, xshift, xmin, xmax, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array),\ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw_sx,\ + .put = snd_soc_put_volsw_sx, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xreg, \ + .shift = xshift, .rshift = xshift, \ + .max = xmax, .min = xmin} } #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ @@ -85,6 +97,18 @@ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_DOUBLE_R_SX_TLV(xname, xreg, xrreg, xshift, xmin, xmax, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw_sx, \ + .put = snd_soc_put_volsw_sx, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xrreg, \ + .shift = xshift, .rshift = xshift, \ + .max = xmax, .min = xmin} } #define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ @@ -171,20 +195,6 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&xenum } -#define SOC_DOUBLE_R_SX_TLV(xname, xreg_left, xreg_right, xshift,\ - xmin, xmax, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_READWRITE, \ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_2r_sx, \ - .get = snd_soc_get_volsw_2r_sx, \ - .put = snd_soc_put_volsw_2r_sx, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg_left, \ - .rreg = xreg_right, .shift = xshift, \ - .min = xmin, .max = xmax} } - #define SND_SOC_BYTES(xname, xbase, xregs) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \ @@ -413,6 +423,10 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); #define snd_soc_get_volsw_2r snd_soc_get_volsw #define snd_soc_put_volsw_2r snd_soc_put_volsw +int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, @@ -421,12 +435,6 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_limit_volume(struct snd_soc_codec *codec, const char *name, int max); -int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); -int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); -int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index a8bf588..85d6069 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -141,15 +141,15 @@ static const struct soc_enum cs42l51_chan_mix = static const struct snd_kcontrol_new cs42l51_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, - 7, 0xffffff99, 0x18, adc_pcm_tlv), + 6, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("PCM Playback Switch", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL, - 8, 0xffffff19, 0x18, aout_tlv), + 0, 0x34, 0xE4, aout_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, - 7, 0xffffff99, 0x18, adc_pcm_tlv), + 6, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("ADC Mixer Switch", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 78979b3..d39b3e7 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -402,37 +402,37 @@ static const struct snd_kcontrol_new ear_amp_ctl = static const struct snd_kcontrol_new cs42l73_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", - CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, - 0xffffffC1, 0x0C, hpaloa_tlv), + CS42L73_HPAAVOL, CS42L73_HPBAVOL, 0, + 0x41, 0x4B, hpaloa_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, - CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv), SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, - CS42L73_MICBPREPGABVOL, 5, 0xffffff35, - 0x34, micpga_tlv), + CS42L73_MICBPREPGABVOL, 5, 0x34, + 0x24, micpga_tlv), SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, CS42L73_MICBPREPGABVOL, 6, 1, 1), SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, - CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv), + CS42L73_IPBDVOL, 0, 0xA0, 0x6C, ipd_tlv), SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", - CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5, - 0xE4, hl_tlv), + CS42L73_HLADVOL, CS42L73_HLBDVOL, + 0, 0x34, 0xE4, hl_tlv), SOC_SINGLE_TLV("ADC A Boost Volume", CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv), SOC_SINGLE_TLV("ADC B Boost Volume", - CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), + CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), - SOC_SINGLE_TLV("Speakerphone Digital Playback Volume", - CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv), + SOC_SINGLE_SX_TLV("Speakerphone Digital Volume", + CS42L73_SPKDVOL, 0, 0x34, 0xE4, hl_tlv), - SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume", - CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv), + SOC_SINGLE_SX_TLV("Ear Speaker Digital Volume", + CS42L73_ESLDVOL, 0, 0x34, 0xE4, hl_tlv), SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, 1, 1), diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a4deebc..548fb05 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2531,6 +2531,87 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_volsw); /** + * snd_soc_get_volsw_sx - single mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a single mixer control, or a double mixer + * control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mask = (1 << (fls(min + max) - 1)) - 1; + + ucontrol->value.integer.value[0] = + ((snd_soc_read(codec, reg) >> shift) - min) & mask; + + if (snd_soc_volsw_is_stereo(mc)) + ucontrol->value.integer.value[1] = + ((snd_soc_read(codec, reg2) >> rshift) - min) & mask; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); + +/** + * snd_soc_put_volsw_sx - double mixer set callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to set the value of a double mixer control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mask = (1 << (fls(min + max) - 1)) - 1; + int err; + unsigned short val, val_mask, val2 = 0; + + val_mask = mask << shift; + val = (ucontrol->value.integer.value[0] + min) & mask; + val = val << shift; + + if (snd_soc_update_bits_locked(codec, reg, val_mask, val)) + return err; + + if (snd_soc_volsw_is_stereo(mc)) { + val_mask = mask << rshift; + val2 = (ucontrol->value.integer.value[1] + min) & mask; + val2 = val2 << rshift; + + if (snd_soc_update_bits_locked(codec, reg2, val_mask, val2)) + return err; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx); + +/** * snd_soc_info_volsw_s8 - signed mixer info callback * @kcontrol: mixer control * @uinfo: control element information @@ -2651,99 +2732,6 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_limit_volume); -/** - * snd_soc_info_volsw_2r_sx - double with tlv and variable data size - * mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Returns 0 for success. - */ -int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int max = mc->max; - int min = mc->min; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = max-min; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r_sx); - -/** - * snd_soc_get_volsw_2r_sx - double with tlv and variable data size - * mixer get callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_2r_sx(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); - unsigned int mask = (1<<mc->shift)-1; - int min = mc->min; - int val = snd_soc_read(codec, mc->reg) & mask; - int valr = snd_soc_read(codec, mc->rreg) & mask; - - ucontrol->value.integer.value[0] = ((val & 0xff)-min) & mask; - ucontrol->value.integer.value[1] = ((valr & 0xff)-min) & mask; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r_sx); - -/** - * snd_soc_put_volsw_2r_sx - double with tlv and variable data size - * mixer put callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_2r_sx(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); - unsigned int mask = (1<<mc->shift)-1; - int min = mc->min; - int ret; - unsigned int val, valr, oval, ovalr; - - val = ((ucontrol->value.integer.value[0]+min) & 0xff); - val &= mask; - valr = ((ucontrol->value.integer.value[1]+min) & 0xff); - valr &= mask; - - oval = snd_soc_read(codec, mc->reg) & mask; - ovalr = snd_soc_read(codec, mc->rreg) & mask; - - ret = 0; - if (oval != val) { - ret = snd_soc_write(codec, mc->reg, val); - if (ret < 0) - return ret; - } - if (ovalr != valr) { - ret = snd_soc_write(codec, mc->rreg, valr); - if (ret < 0) - return ret; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r_sx); - int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { -- 1.7.9.1

2 1

[alsa-devel] [PATCH 00/17] ASoC: tegra: Add Tegra30 support
by Stephen Warren 03 Apr '12

03 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> This series adds basic support for 16-bit stereo on Tegra30. This will be enabled for the Cardhu board in a separate series to arch/arm/mach-tegra. I don't believe this depends on anything that won't be in 3.4. The related arch/arm/mach-tegra series will have some dependencies. There are a couple patches in here which also touch arch/arm/mach-tegra. I believe it should be fine to apply these to the ASoC tree without issue (I don't anticpate conflicts) but as you (Mark) mentioned, putting them into a separate branch just in case might be a good idea. Stephen Warren (17): ASoC: tegra: remove open-coded clk reference counting ASoC: tegra: ensure clocks are enabled when touching registers ASoC: tegra: remove unnecessary includes ASoC: tegra: add description to a couple Kconfig options ASoC: tegra: fix Kconfig SND_SOC_TEGRA_ALC5632 indentation ASoC: tegra: fix some checkpatch warnings ASoC: tegra: introduce separate Kconfig variable for DAS driver ASoC: tegra: make Tegra20 drivers depend on Tegra20 support ASoC: tegra: sort Makefile into common and per-SoC files ASoC: tegra: rename Tegra20-specific driver files ASoC: tegra: complete Tegra->Tegra20 renaming ASoC: tegra: utils: add support for Tegra30 devices ASoC: tegra: set a sensible initial clock rate ASoC: tegra: add tegra30-ahub driver ASoC: tegra: add tegra30-i2s driver ASoC: tegra: add Kconfig and Makefile support for Tegra30 ASoC: tegra+wm8903 machine: support Tegra30 .../bindings/sound/tegra-audio-wm8903.txt | 8 +- .../devicetree/bindings/sound/tegra30-ahub.txt | 18 + .../devicetree/bindings/sound/tegra30-i2s.txt | 12 + arch/arm/mach-tegra/board-dt-tegra20.c | 6 +- arch/arm/mach-tegra/devices.c | 6 +- arch/arm/mach-tegra/tegra2_clocks.c | 4 +- sound/soc/tegra/Kconfig | 61 ++- sound/soc/tegra/Makefile | 18 +- sound/soc/tegra/tegra20_das.c | 261 +++++++++ sound/soc/tegra/tegra20_das.h | 135 +++++ sound/soc/tegra/tegra20_i2s.c | 462 +++++++++++++++ sound/soc/tegra/tegra20_i2s.h | 165 ++++++ sound/soc/tegra/tegra20_spdif.c | 369 ++++++++++++ sound/soc/tegra/tegra20_spdif.h | 472 ++++++++++++++++ sound/soc/tegra/tegra30_ahub.c | 593 ++++++++++++++++++++ sound/soc/tegra/tegra30_ahub.h | 489 ++++++++++++++++ sound/soc/tegra/tegra30_i2s.c | 538 ++++++++++++++++++ sound/soc/tegra/tegra30_i2s.h | 242 ++++++++ sound/soc/tegra/tegra_alc5632.c | 6 +- sound/soc/tegra/tegra_asoc_utils.c | 35 +- sound/soc/tegra/tegra_asoc_utils.h | 10 +- sound/soc/tegra/tegra_das.c | 261 --------- sound/soc/tegra/tegra_das.h | 135 ----- sound/soc/tegra/tegra_i2s.c | 464 --------------- sound/soc/tegra/tegra_i2s.h | 166 ------ sound/soc/tegra/tegra_spdif.c | 369 ------------ sound/soc/tegra/tegra_spdif.h | 473 ---------------- sound/soc/tegra/tegra_wm8903.c | 39 +- sound/soc/tegra/trimslice.c | 10 +- 29 files changed, 3896 insertions(+), 1931 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tegra30-ahub.txt create mode 100644 Documentation/devicetree/bindings/sound/tegra30-i2s.txt create mode 100644 sound/soc/tegra/tegra20_das.c create mode 100644 sound/soc/tegra/tegra20_das.h create mode 100644 sound/soc/tegra/tegra20_i2s.c create mode 100644 sound/soc/tegra/tegra20_i2s.h create mode 100644 sound/soc/tegra/tegra20_spdif.c create mode 100644 sound/soc/tegra/tegra20_spdif.h create mode 100644 sound/soc/tegra/tegra30_ahub.c create mode 100644 sound/soc/tegra/tegra30_ahub.h create mode 100644 sound/soc/tegra/tegra30_i2s.c create mode 100644 sound/soc/tegra/tegra30_i2s.h delete mode 100644 sound/soc/tegra/tegra_das.c delete mode 100644 sound/soc/tegra/tegra_das.h delete mode 100644 sound/soc/tegra/tegra_i2s.c delete mode 100644 sound/soc/tegra/tegra_i2s.h delete mode 100644 sound/soc/tegra/tegra_spdif.c delete mode 100644 sound/soc/tegra/tegra_spdif.h

2 43