September 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC] ALSA: Reduce delay for a blocking pcm_drain.
by Dylan Reid 30 Sep '13

30 Sep '13

This patch addresses two issues related to calling snd_pcm_drain. If no_period_wakeup is set, then snd_pcm_drain would wait MAX_SCHEDULE_TIMEOUT jiffies, without a wakeup pending this will leave the calling task blocked indefinitely. Instead block for double the buffer size played back at a low sample rate (4k chosen arbitrarily). Also if the stream is running with period wakeups but with a long period, the delay could be seconds. If only a small part of the buffer is being used, this is unnecessary. Instead wait for the remaining samples to play out, plus one millisecond. This allows systems that fill the buffer only part way to still do a blocking drain before closing to avoid losing samples. Change-Id: I84ee1a52d9bdf80ea6065fb9533e565d3e6c8021 Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- This can be worked around in user space as well by using a nonblocking drain and sleeping based on how many samples are queued. Would that be the right thing to do or is this (or something similar) worth fixing here. To test this I opened a pcm and set no period wakeups, played a few samples and then called drain, which hung indefinitely. Using a USB headset with a max period size of 128k but only 15ms of the buffer filled, having the drain time shorter without losing samples makes a blocking call feasible. --- sound/core/pcm_native.c | 17 +++++++---------- 1 file changed, 7 insertions(+), 10 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index f4aaf5a..a5f0695 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1481,7 +1481,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, } for (;;) { - long tout; + long tout, tout_ms; struct snd_pcm_runtime *to_check; if (signal_pending(current)) { result = -ERESTARTSYS; @@ -1505,16 +1505,13 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, snd_pcm_stream_unlock_irq(substream); up_read(&snd_pcm_link_rwsem); snd_power_unlock(card); - if (runtime->no_period_wakeup) - tout = MAX_SCHEDULE_TIMEOUT; - else { - tout = 10; - if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; - tout = max(t, tout); - } - tout = msecs_to_jiffies(tout * 1000); + tout_ms = runtime->buffer_size * 2 * 1000 / 4000; + if (runtime->rate) { + snd_pcm_sframes_t hw_avail; + hw_avail = snd_pcm_playback_hw_avail(runtime); + tout_ms = 1 + hw_avail * 1000 / runtime->rate; } + tout = msecs_to_jiffies(tout_ms); tout = schedule_timeout_interruptible(tout); snd_power_lock(card); down_read(&snd_pcm_link_rwsem); -- 1.8.1.3.605.g02339dd

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[alsa-devel] [PATCH] Enable internal mic on a Thinkpad machine
by David Henningsson 30 Sep '13

30 Sep '13

From: Kailang <kailang(a)realtek.com> More thorough testing showed that these verbs were necessary to improve quality of the internal mic. Patch originally from Realtek. Cc: stable(a)kernel.org (3.11) BugLink: https://bugs.launchpad.net/bugs/1231931 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bc07d36..c4d54a9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3531,6 +3531,7 @@ enum { ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, ALC282_FIXUP_ASUS_TX300, + ALC283_FIXUP_INT_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -3790,6 +3791,16 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc282_fixup_asus_tx300, }, + [ALC283_FIXUP_INT_MIC] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x20, AC_VERB_SET_COEF_INDEX, 0x1a}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0011}, + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3874,7 +3885,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x17aa, 0x501a, "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), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), -- 1.7.9.5

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Re: [alsa-devel] pcm_rate API question
by Rob Sykes 30 Sep '13

30 Sep '13

> From: Rob Sykes <aquegg(a)yahoo.co.uk> >To: "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org> >Sent: Friday, 27 September 2013, 16:28 >Subject: Re: [alsa-devel] pcm_rate API question > >> From: Clemens Ladisch <clemens(a)ladisch.de> >> To: Rob Sykes <aquegg(a)yahoo.co.uk>; "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org> >> Cc: >> Sent: Friday, 27 September 2013, 13:38 >> Subject: Re: [alsa-devel] pcm_rate API question >> >> Rob Sykes wrote: >>> void (*convert_s16)(void *obj, int16_t *dst, unsigned int >> dst_frames, >>> const int16_t *src, unsigned int src_frames); >>> >>> is the plugin obliged always to consume src_frames and generate dst_frames >> samples? >> >> Yes. src_frames and dst_frames are the period sizes of the respective >> devices; their values will never change, and they define the actual >> conversion ratio. > >Thanks, Clemens. Typically there is some buffering/delay within a resampler -- how does the pcm_rate API allow for this? E.g. from the libsamplerate FAQ: > >Q6 : I'm use the SRC_SINC_* converters and up-sampling by a ratio of 2. I reset the converter and put in 1000 samples and I expect to get 2000 samples out, but I'm getting less than that. Why? >The short answer is that there is a transport delay inside the converter itself. Long answer follows. By way of example, the first time you call src_process() you might only get 1900 samples out. > However, after that first call all subsequent calls will probably get you about 2000 samples out for every 1000 samples you put in. Also, just noticed that when performing a rate conversion from 44100 -> 48000, the buffer sizes used by alsa are 5512 and 6000, however the actual ratio for the required conversion is 5512.5 to 6000. If, as you say above, the buffer sizes define the conversion ratio, then this seems to imply that alsa is unable to perform accurate rate conversion for one of the most common uses! — am I missing something here? A pointer to the API documentation, or perhaps some relevant threads, would be very useful. Cheers, Rob

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Re: [alsa-devel] [RFC] ALSA: Reduce delay for a blocking pcm_drain.
by Raymond Yau 30 Sep '13

30 Sep '13

> > This can be worked around in user space as well by using a nonblocking drain and > sleeping based on how many samples are queued. Would that be the right thing to > do or is this (or something similar) worth fixing here. > > To test this I opened a pcm and set no period wakeups, played a few samples and > then called drain, which hung indefinitely. Using a USB headset with a > max period size of 128k but only 15ms of the buffer filled, having the > drain time shorter without losing samples makes a blocking call > feasible. > Do you mean that you tested on a patched version of USB audio driver since only three drivers support SNDRV_PCM_INFO_NO_PERIOD_WAKEUP http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound?qt=gr…

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Re: [alsa-devel] Bug: Conexant CX20588, Kernel 3.11, Win8 changes GPIO settings
by Takashi Iwai 30 Sep '13

30 Sep '13

At Fri, 27 Sep 2013 22:38:22 +0200, Christopher wrote: > > I attached the alsa-info. This issue seems to be specific to the driver > provided by Acer (with thx, equalizer etc.) > Thanks for your help. Thanks. Could you check whether the patch below works? Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix GPIO for Acer Aspire 3830TG Acer Aspire 3830TG seems requiring GPIO bit 0 as the primary mute control. When a machine is booted after Windows 8, the GPIO pin is turned off and it results in the silent output. This patch adds the manual fixup of GPIO bit 0 for this model. Reported-by: Christopher <DIDI2002(a)web.de> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4edd2d0..ec68eae 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3231,6 +3231,7 @@ enum { CXT_FIXUP_INC_MIC_BOOST, CXT_FIXUP_HEADPHONE_MIC_PIN, CXT_FIXUP_HEADPHONE_MIC, + CXT_FIXUP_GPIO1, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -3375,6 +3376,15 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_headphone_mic, }, + [CXT_FIXUP_GPIO1] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x01 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01 }, + { 0x01, AC_VERB_SET_GPIO_DATA, 0x01 }, + { } + }, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -3384,6 +3394,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), -- 1.8.4

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[alsa-devel] [PATCH 1/2] [RESEND] snd-ac97-codec: Add ID for TI TLV320AIC27 codec
by Ondrej Zary 30 Sep '13

30 Sep '13

Add 0x54584e03 ID for TI TLV320AIC27 AC'97 codec according to datasheet: http://www.ti.com/lit/ds/slas253a/slas253a.pdf The weird thing is that the chip is physically marked 320AD91. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/ac97/ac97_codec.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 445ca48..bf578ba2 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -175,6 +175,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54524106, 0xffffffff, "TR28026", NULL, NULL }, { 0x54524108, 0xffffffff, "TR28028", patch_tritech_tr28028, NULL }, // added by xin jin [07/09/99] { 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)] +{ 0x54584e03, 0xffffffff, "TLV320AIC27", NULL, NULL }, { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF -- Ondrej Zary

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[alsa-devel] [PATCH] caiaq: LED support for Maschine Controller
by Hannes Gräuler 30 Sep '13

30 Sep '13

This patch adds LED support for the Native Instruments Maschine Controller. It adds ALSA controls for dimming the LEDs of all buttons and the backlight of the two displays. Signed-off-by: Hannes Gräuler <hgraeule(a)uos.de> --- sound/usb/caiaq/control.c | 92 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/caiaq/device.c | 25 +++++++++++++ sound/usb/caiaq/device.h | 5 +++ 3 files changed, 122 insertions(+) diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index ae6b50f..f65fc09 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -28,6 +28,7 @@ #include "control.h" #define CNT_INTVAL 0x10000 +#define MASCHINE_BANK_SIZE 32 static int control_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -105,6 +106,10 @@ static int control_put(struct snd_kcontrol *kcontrol, USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1)) cmd = EP1_CMD_DIMM_LEDS; + if (cdev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER)) + cmd = EP1_CMD_DIMM_LEDS; + if (pos & CNT_INTVAL) { int i = pos & ~CNT_INTVAL; @@ -121,6 +126,20 @@ static int control_put(struct snd_kcontrol *kcontrol, usb_sndbulkpipe(cdev->chip.dev, 8), cdev->ep8_out_buf, sizeof(cdev->ep8_out_buf), &actual_len, 200); + } else if (cdev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER)) { + + int bank = 0; + int offset = 0; + + if (i >= MASCHINE_BANK_SIZE) { + bank = 0x1e; + offset = MASCHINE_BANK_SIZE; + } + + snd_usb_caiaq_send_command_bank(cdev, cmd, bank, + cdev->control_state + offset, + MASCHINE_BANK_SIZE); } else { snd_usb_caiaq_send_command(cdev, cmd, cdev->control_state, sizeof(cdev->control_state)); @@ -490,6 +509,74 @@ static struct caiaq_controller kontrols4_controller[] = { { "LED: FX2: Mode", 133 | CNT_INTVAL }, }; +static struct caiaq_controller maschine_controller[] = { + { "LED: Pad 1", 3 | CNT_INTVAL }, + { "LED: Pad 2", 2 | CNT_INTVAL }, + { "LED: Pad 3", 1 | CNT_INTVAL }, + { "LED: Pad 4", 0 | CNT_INTVAL }, + { "LED: Pad 5", 7 | CNT_INTVAL }, + { "LED: Pad 6", 6 | CNT_INTVAL }, + { "LED: Pad 7", 5 | CNT_INTVAL }, + { "LED: Pad 8", 4 | CNT_INTVAL }, + { "LED: Pad 9", 11 | CNT_INTVAL }, + { "LED: Pad 10", 10 | CNT_INTVAL }, + { "LED: Pad 11", 9 | CNT_INTVAL }, + { "LED: Pad 12", 8 | CNT_INTVAL }, + { "LED: Pad 13", 15 | CNT_INTVAL }, + { "LED: Pad 14", 14 | CNT_INTVAL }, + { "LED: Pad 15", 13 | CNT_INTVAL }, + { "LED: Pad 16", 12 | CNT_INTVAL }, + + { "LED: Mute", 16 | CNT_INTVAL }, + { "LED: Solo", 17 | CNT_INTVAL }, + { "LED: Select", 18 | CNT_INTVAL }, + { "LED: Duplicate", 19 | CNT_INTVAL }, + { "LED: Navigate", 20 | CNT_INTVAL }, + { "LED: Pad Mode", 21 | CNT_INTVAL }, + { "LED: Pattern", 22 | CNT_INTVAL }, + { "LED: Scene", 23 | CNT_INTVAL }, + + { "LED: Shift", 24 | CNT_INTVAL }, + { "LED: Erase", 25 | CNT_INTVAL }, + { "LED: Grid", 26 | CNT_INTVAL }, + { "LED: Right Bottom", 27 | CNT_INTVAL }, + { "LED: Rec", 28 | CNT_INTVAL }, + { "LED: Play", 29 | CNT_INTVAL }, + { "LED: Left Bottom", 32 | CNT_INTVAL }, + { "LED: Restart", 33 | CNT_INTVAL }, + + { "LED: Group A", 41 | CNT_INTVAL }, + { "LED: Group B", 40 | CNT_INTVAL }, + { "LED: Group C", 37 | CNT_INTVAL }, + { "LED: Group D", 36 | CNT_INTVAL }, + { "LED: Group E", 39 | CNT_INTVAL }, + { "LED: Group F", 38 | CNT_INTVAL }, + { "LED: Group G", 35 | CNT_INTVAL }, + { "LED: Group H", 34 | CNT_INTVAL }, + + { "LED: Auto Write", 42 | CNT_INTVAL }, + { "LED: Snap", 43 | CNT_INTVAL }, + { "LED: Right Top", 44 | CNT_INTVAL }, + { "LED: Left Top", 45 | CNT_INTVAL }, + { "LED: Sampling", 46 | CNT_INTVAL }, + { "LED: Browse", 47 | CNT_INTVAL }, + { "LED: Step", 48 | CNT_INTVAL }, + { "LED: Control", 49 | CNT_INTVAL }, + + { "LED: Top Button 1", 57 | CNT_INTVAL }, + { "LED: Top Button 2", 56 | CNT_INTVAL }, + { "LED: Top Button 3", 55 | CNT_INTVAL }, + { "LED: Top Button 4", 54 | CNT_INTVAL }, + { "LED: Top Button 5", 53 | CNT_INTVAL }, + { "LED: Top Button 6", 52 | CNT_INTVAL }, + { "LED: Top Button 7", 51 | CNT_INTVAL }, + { "LED: Top Button 8", 50 | CNT_INTVAL }, + + { "LED: Note Repeat", 58 | CNT_INTVAL }, + + { "Backlight Display", 59 | CNT_INTVAL } +}; + static int add_controls(struct caiaq_controller *c, int num, struct snd_usb_caiaqdev *cdev) { @@ -553,6 +640,11 @@ int snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *cdev) ret = add_controls(kontrols4_controller, ARRAY_SIZE(kontrols4_controller), cdev); break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): + ret = add_controls(maschine_controller, + ARRAY_SIZE(maschine_controller), cdev); + break; } return ret; diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 1a61dd1..bc55f70 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -235,6 +235,31 @@ int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *cdev, cdev->ep1_out_buf, len+1, &actual_len, 200); } +int snd_usb_caiaq_send_command_bank(struct snd_usb_caiaqdev *cdev, + unsigned char command, + unsigned char bank, + const unsigned char *buffer, + int len) +{ + int actual_len; + struct usb_device *usb_dev = cdev->chip.dev; + + if (!usb_dev) + return -EIO; + + if (len > EP1_BUFSIZE - 2) + len = EP1_BUFSIZE - 2; + + if (buffer && len > 0) + memcpy(cdev->ep1_out_buf+2, buffer, len); + + cdev->ep1_out_buf[0] = command; + cdev->ep1_out_buf[1] = bank; + + return usb_bulk_msg(usb_dev, usb_sndbulkpipe(usb_dev, 1), + cdev->ep1_out_buf, len+2, &actual_len, 200); +} + int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, int rate, int depth, int bpp) { diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index ad102fa..ab0f752 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -128,5 +128,10 @@ int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *cdev, unsigned char command, const unsigned char *buffer, int len); +int snd_usb_caiaq_send_command_bank(struct snd_usb_caiaqdev *cdev, + unsigned char command, + unsigned char bank, + const unsigned char *buffer, + int len); #endif /* CAIAQ_DEVICE_H */ -- 1.8.4

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[alsa-devel] ASoC updates for v3.12
by Mark Brown 30 Sep '13

30 Sep '13

The following changes since commit 4a10c2ac2f368583138b774ca41fac4207911983: Linux 3.12-rc2 (2013-09-23 15:41:09 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.12-rc2 for you to fetch changes up to 267666ea3bd6e804c2c8bbafbe628026be963d06: Merge remote-tracking branch 'asoc/fix/max98095' into asoc-linus (2013-09-27 14:56:24 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.12 Nothing too exciting here, all driver specific except for the fix from Liam for DPCM systems which have both front and back end DAIs which is not yet used by anything in mainline. ---------------------------------------------------------------- Dan Carpenter (3): ASoC: max98095: a couple array underflows ASoC: ab8500-codec: info leak in anc_status_control_put() ASoC: 88pm860x: array overflow in snd_soc_put_volsw_2r_st() Liam Girdwood (1): ASoC: core: Only add platform DAI widgets once. Mark Brown (6): Merge remote-tracking branch 'asoc/fix/88pm860x' into asoc-linus Merge remote-tracking branch 'asoc/fix/ab8500' into asoc-linus Merge remote-tracking branch 'asoc/fix/bfin' into asoc-linus Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus Merge remote-tracking branch 'asoc/fix/max98095' into asoc-linus Philipp Zabel (2): ASoC: imx-sgtl5000: do not use devres on a foreign device ASoC: imx-sgtl5000: Fix uninitialized pointer use in error path Valentin Ilie (1): ASoC: blackfin: Add missing break statement to bf6xx sound/soc/blackfin/bf6xx-i2s.c | 1 + sound/soc/codecs/88pm860x-codec.c | 3 +++ sound/soc/codecs/ab8500-codec.c | 7 ++++++- sound/soc/codecs/max98095.c | 4 ++-- sound/soc/fsl/imx-sgtl5000.c | 7 +++++-- sound/soc/soc-core.c | 1 - 6 files changed, 17 insertions(+), 6 deletions(-)

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Re: [alsa-devel] Realtek ALC286 SONY Vaio Pro 13 Ubuntu 13.10
by David Henningsson 30 Sep '13

30 Sep '13

(Adding alsa-devel to CC) On 09/28/2013 05:53 AM, Yuval Levy wrote: > Sirs: > > I have been pointed to your patch to implement headset jack > functionality for some Dell hw [1] after reporting and investigating > soud support issues [2] with a SONY Vaio Pro 13 (Realtek ALC286). This different headset mode support is likely unrelated to your problem. And they don't cover your codec (ALC286), so you would need another set of verbs too, things that only Realtek knows. > Output to internal loudspeakers and HDMI works as expected, but neither > internal microphone nor the headset work (and they do work in Windows, > so it is not a hardware issue). An important distinction is here: to they work in Windows without specific Realtek drivers installed, or do you require Realtek's Windows drivers for this support? Second, to do some actual triaging. I see you have been running hda-jack-sense-test. Could you run it all three scenarios: 1) nothing connected 2) headphone (without mic) connected 3) headset (heaphone + mic combo) connected ...and submit the result? In ideal circumstances, 0x21 should report "Yes" in scenario 2 and 3, and 0x18 should only report yes in scenario 3. If this is not the result you get, try using hda-jack-retask to turn off the jack detection: in Ubuntu 13.10 just install the alsa-tools-gui package (no need to use my ppa). Click "advanced override", find pins 0x18 or 0x21, change "jack detection" from "yes" to "no" and click apply. Now you should be able to manually choose between internal and external microphone, as well as between headphones as speakers. See if this helps. > As you can see in the bug thread, I have been helped in the diagnosis by > a very friendly expert, but it seems that we have reached the limit and > without support from Realtek or Sony the issue will go unresolved. I'm afraid Sony is not a company that actively supports Linux for the desktop machines, sorry. Canonical, which seems to have the most contracts for preloading Linux these days, has a list of certified hardware here: http://www.ubuntu.com/certification/ > Would you kindly help me find out what need to be done for the Realtek > ALC286 in the SONY Vaio Pro 13 to be supported in Ubuntu Linux? > > Many thanks, > Yuv > > [1] > <https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p…> > [2] <https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1228689> > -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] [PATCH] ASoC: si476x: Remove custom register I/O implementation
by Mark Brown 28 Sep '13

28 Sep '13

From: Mark Brown <broonie(a)linaro.org> The current si476x I/O implementation wraps the regmap for the core with functions that make the register map cache only when the device is powered down. This implementation appears to be incomplete since there is no code to synchronise the cache so writes done while the core is powered down will be ignored, the device will only be configured if it is powered. A better and more idiomatic approach would be to have the MFD manage the cache, making the device cache only when it powers things down. This also allows ASoC to use the standard regmap helpers for the device which helps remove the ASoC custom ones so do convert to do that. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/si476x.c | 46 +--------------------------------------------- 1 file changed, 1 insertion(+), 45 deletions(-) diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 38f3b10..ec29d26 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -60,48 +60,6 @@ enum si476x_pcm_format { SI476X_PCM_FORMAT_S24_LE = 6, }; -static unsigned int si476x_codec_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - int err; - unsigned int val; - struct si476x_core *core = codec->control_data; - - si476x_core_lock(core); - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, true); - - err = regmap_read(core->regmap, reg, &val); - - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, false); - si476x_core_unlock(core); - - if (err < 0) - return err; - - return val; -} - -static int si476x_codec_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int val) -{ - int err; - struct si476x_core *core = codec->control_data; - - si476x_core_lock(core); - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, true); - - err = regmap_write(core->regmap, reg, val); - - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, false); - si476x_core_unlock(core); - - return err; -} - static const struct snd_soc_dapm_widget si476x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LOUT"), SND_SOC_DAPM_OUTPUT("ROUT"), @@ -239,7 +197,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, static int si476x_codec_probe(struct snd_soc_codec *codec) { - codec->control_data = i2c_mfd_cell_to_core(codec->dev); + codec->control_data = dev_get_regmap(codec->dev.parent); return 0; } @@ -268,8 +226,6 @@ static struct snd_soc_dai_driver si476x_dai = { static struct snd_soc_codec_driver soc_codec_dev_si476x = { .probe = si476x_codec_probe, - .read = si476x_codec_read, - .write = si476x_codec_write, .dapm_widgets = si476x_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(si476x_dapm_widgets), .dapm_routes = si476x_dapm_routes, -- 1.8.4.rc3

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