March 2016 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH 4/4] ALSA: dice: force to add two pcm devices for listed models
by Takashi Sakamoto 07 Mar '16

07 Mar '16

On Mar 7 2016 22:43, Stefan Richter wrote: > On Mar 07 Takashi Sakamoto wrote: >> Thanks for your report. Totally: >> * The patched snd-dice still works fine with Saffire Pro 24. >> * The pached snd-dice successfully manages several isochronous streams >> without code bugs (i.e. kernel NULL pointer dereference or >> continuous locking) with Saffire Pro 24/40. >> * The pached snd-dice still has mute issue with Saffire Pro 40. > > I concur. While I was only able to perform just a few tests yet, there > doesn't appear to be any regression, and this patchset adds support for > the previously inaccessible channels at the second transmitter and > receiver. > >> Anyway, the result cannot block applying this patchset for 4.6. I'll >> post it in this night for merging. > > Thanks. I posted. Thanks for your test ;) >> Well, about the mute issue, I have no idea to fix it at all. At least, >> for my test units, the patched snd-dice works fine without such issues. >> Therefore, it's nearly model-dependent issue and I have no way to do for >> it. FYI, I tested with: >> * OHCI 1394 host controllers >> * VIA VT6315 >> * TI XIO2213 >> * Dice II units >> * TC Electronic ImpactTwin >> * TC Electronic Konnekt24D >> * TC Electronic Konnekt8 >> * Dice Jr. units >> * TC Electronic DesktopKonnect6 >> * Dice Mini units >> * Focusrite Saffire Pro 26 > > FWIW, Saffire PRO 24 and 40 are based on Dice Junior according to FFADO's > device database. If desired I could open up my 24 and 40 to check. If you don't mind it, please. (Because once opening your unit, usually, you cannot get repair service from sellers.) And I wrote wrong for Dice Jr./Mini units. Correctly: * Dice Jr. units * TC Electronic DesktopKonnect6 * Focusrite Saffire Pro 26 * I have no units with Dice Mini (Oh...). Anyway, all of my units with Dice Jr. works fine. So we can judge that Pro 40 has its own quirk which we don't know. > Furthermore FYI, the FFADO userspace streaming driver has got the following > two known issues with the PRO 40: > http://subversion.ffado.org/ticket/326 > playback at 96 kHz is choppy (but fine at 44.1, 48, 88.2 kHz; > reproduced by multiple users, 100% reproducible by myself too) > http://subversion.ffado.org/ticket/372 > MIDI playback problem (perhaps dependent on firmware revision; > cannot reproduce this myself due to lack of MIDI equipment) Hm. With a lack of something like packet dump or detailed case-investigation, I can do nothing for the unit. I have no fashion to lost my time for meaningless talks, sorry. For our information, I put your result of hinawa-dice-common-cui from Saffire Pro 40. (meaningless lines are omitted.) $ ./hinawa-dice-common-cui 00130e040140dd28 current-status owner address: ffc1000100000000 global enabled: False clock status: locked: 1 nominal: 48000 clock source detection: locked: slip: measured rate: 48000 dice version: 1.0.12.0 latest notification: 0x00000010 clock source: Internal sampling rate: 48000 nickname: Pro40-00dd28 Transmitted streams: Stream 0: iso-channel: 4294967295 speed: 2 pcm: 10 midi: 1 iec60958: caps: 0 enable: 0 formation: 0: IP 1 1: IP 2 2: IP 3 3: IP 4 4: IP 5 5: IP 6 6: IP 7 7: IP 8 8: SPDIF L 9: SPDIF R Stream 1: iso-channel: 4294967295 speed: 2 pcm: 10 midi: 0 iec60958: caps: 0 enable: 0 formation: 0: ADAT 1 1: ADAT 2 2: ADAT 3 3: ADAT 4 4: ADAT 5 5: ADAT 6 6: ADAT 7 7: ADAT 8 8: Loop 1 9: Loop 2 Received streams: Stream 0: iso-channel: 4294967295 start: 0 pcm: 12 midi: 1 iec60958: caps: 0 enable: 0 formation: 0: Mon 1 1: Mon 2 2: Line 3 3: Line 4 4: Line 5 5: Line 6 6: Line 7 7: Line 8 8: Line 9 9: Line 10 10: SPDIF L 11: SPDIF R Stream 1: iso-channel: 4294967295 start: 0 pcm: 8 midi: 0 iec60958: caps: 0 enable: 0 formation: 0: ADAT 1 1: ADAT 2 2: ADAT 3 3: ADAT 4 4: ADAT 5 5: ADAT 6 6: ADAT 7 7: ADAT 8 Synchronization infomation: clock-source: Internal locked: 1 sampling-rate: 48000 ADAT: data-bits: 0 no-data: 1 $ ./hinawa-dice-common-cui 00130e040140dd28 sampling-rate set 96000 $ ./hinawa-dice-common-cui 00130e040140dd28 current-status owner address: ffc1000100000000 global enabled: False clock status: locked: 1 nominal: 96000 clock source detection: locked: slip: measured rate: 96000 dice version: 1.0.12.0 latest notification: 0x00000010 clock source: Internal sampling rate: 96000 nickname: Pro40-00dd28 Transmitted streams: Stream 0: iso-channel: 4294967295 speed: 2 pcm: 16 midi: 1 iec60958: caps: 0 enable: 0 formation: 0: IP 1 1: IP 2 2: IP 3 3: IP 4 4: IP 5 5: IP 6 6: IP 7 7: IP 8 8: SPDIF L 9: SPDIF R 10: ADAT 1 11: ADAT 2 12: ADAT 3 13: ADAT 4 14: Loop 1 15: Loop 2 Received streams: Stream 0: iso-channel: 4294967295 start: 0 pcm: 16 midi: 1 iec60958: caps: 0 enable: 0 formation: 0: Mon 1 1: Mon 2 2: Line 3 3: Line 4 4: Line 5 5: Line 6 6: Line 7 7: Line 8 8: Line 9 9: Line 10 10: SPDIF L 11: SPDIF R 12: ADAT 1 13: ADAT 2 14: ADAT 3 15: ADAT 4 Synchronization infomation: clock-source: Internal locked: 1 sampling-rate: 96000 ADAT: data-bits: 0 no-data: 1 Regards Takashi Sakamoto

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[alsa-devel] [PATCH] ASoC: cs35l32: avoid uninitialized variable access
by Arnd Bergmann 07 Mar '16

07 Mar '16

gcc warns about the possibilty of accessing a property read from devicetree in cs35l32_i2c_probe() when it has not been initialized because CONFIG_OF is disabled: sound/soc/codecs/cs35l32.c: In function 'cs35l32_i2c_probe': sound/soc/codecs/cs35l32.c:278:2: warning: 'val' may be used uninitialized in this function [-Wmaybe-uninitialized] The code is actually correct because it checks the dev->of_node variable first and we know this is NULL here when CONFIG_OF is disabled, but Russell King noticed that it's broken when we probe the device using DT, and the properties are absent. The code already has some checking for incorrect values, and I keep that checking unchanged here, but add an additional check for an error returned by the property accessor functions that now gets handled the same way as incorrect data in the properties. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/cs35l32.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 44c30fe3e315..c490dc74121b 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -274,21 +274,24 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, if (of_property_read_u32(np, "cirrus,sdout-share", &val) >= 0) pdata->sdout_share = val; - of_property_read_u32(np, "cirrus,boost-manager", &val); + if (of_property_read_u32(np, "cirrus,boost-manager", &val)) + val = -1u; + switch (val) { case CS35L32_BOOST_MGR_AUTO: case CS35L32_BOOST_MGR_AUTO_AUDIO: case CS35L32_BOOST_MGR_BYPASS: case CS35L32_BOOST_MGR_FIXED: - pdata->boost_mng = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,boost-manager DT value %d\n", val); pdata->boost_mng = CS35L32_BOOST_MGR_BYPASS; } - of_property_read_u32(np, "cirrus,sdout-datacfg", &val); + if (of_property_read_u32(np, "cirrus,sdout-datacfg", &val)) + val = -1u; switch (val) { case CS35L32_DATA_CFG_LR_VP: case CS35L32_DATA_CFG_LR_STAT: @@ -296,13 +299,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_DATA_CFG_LR_VPSTAT: pdata->sdout_datacfg = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,sdout-datacfg DT value %d\n", val); pdata->sdout_datacfg = CS35L32_DATA_CFG_LR; } - of_property_read_u32(np, "cirrus,battery-threshold", &val); + if (of_property_read_u32(np, "cirrus,battery-threshold", &val)) + val = -1u; switch (val) { case CS35L32_BATT_THRESH_3_1V: case CS35L32_BATT_THRESH_3_2V: @@ -310,13 +315,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_BATT_THRESH_3_4V: pdata->batt_thresh = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,battery-threshold DT value %d\n", val); pdata->batt_thresh = CS35L32_BATT_THRESH_3_3V; } - of_property_read_u32(np, "cirrus,battery-recovery", &val); + if (of_property_read_u32(np, "cirrus,battery-recovery", &val)) + val = -1u; switch (val) { case CS35L32_BATT_RECOV_3_1V: case CS35L32_BATT_RECOV_3_2V: -- 2.7.0

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[alsa-devel] [GIT PULL] ASoC fixes for v4.5
by Mark Brown 07 Mar '16

07 Mar '16

The following changes since commit fc77dbd34c5c99bce46d40a2491937c3bcbd10af: Linux 4.5-rc6 (2016-02-28 08:41:20 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.5-rc6 for you to fetch changes up to 02db17ff376893f65f45974b0897e70e6695034e: Merge remote-tracking branches 'asoc/fix/wm8985', 'asoc/fix/wm8994', 'asoc/fix/wm8996', 'asoc/fix/wm9081' and 'asoc/fix/wm9713' into asoc-linus (2016-03-05 21:27:13 +0900) ---------------------------------------------------------------- ASoC: Fixes for v4.5 This is far too big a set of fixes for this late in the release cycle but the overwhelming bulk is essentially the same simple fix from Takashi for a cut'n'pasted 64 bit cleanliness issue in the userspace interface where drivers were accessing things using the wrong element in a union which worked OK on 32 bit platforms as the correct element happened to be aligned the same way but with 64 bit platforms ABIs are different and the two members of the union are laid out in different places. They aren't all tagged to stable since some of these chips have vanishingly little chance of being used in 64 bit systems. The other changes are: - A fix for Qualcomm devices to work on big endian systems. The original change is actually correct but triggered a bug in regmap which is too invasive to fix for this cycle and can be worked around by just letting regmap pick the default. - A fix for the Samsung I2S driver locking which wasn't using IRQ safe spinlocks when it needed to. - A fix for the new Intel Sky Lake driver forgetting that C pointer arithmetic takes the type of the pointer into consideration. - A revert of a change to the FSL SSI driver that broke some systems. - A fix for the cleanup path of the wm9713 driver. - A fix for some incorrect register definitions in the ADAU17x1 driver that caused misclocking in some configurations. - A fix for the tracepoints for jack detection to avoid using an internal field of the core jack structure which is no longer present in all configurations. - A fix for another of the new Intel drivers which tried to write to a string literal. ---------------------------------------------------------------- Alan (1): ASoC: Intel: Skylake: fix pointer scaling Andreas Irestål (1): ASoC: adau17x1: Fix incorrect BCLK ratio definitions Arnd Bergmann (1): ASoC: trace: fix printing jack name Carlo Caione (1): ASoC: cht_bsw_rt5645: Fix writing to string literal Charles Keepax (1): ASoC: samsung: Use IRQ safe spin lock calls Fabio Estevam (1): ASoC: fsl_ssi: Go back to explicit register defaults Mark Brown (6): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branches 'asoc/fix/ab8500', 'asoc/fix/adau17x1', 'asoc/fix/adsp', 'asoc/fix/cs42l51', 'asoc/fix/da732x' and 'asoc/fix/fsl-ssi' into asoc-linus Merge remote-tracking branches 'asoc/fix/jack', 'asoc/fix/max98088', 'asoc/fix/max98095', 'asoc/fix/omap', 'asoc/fix/pxa' and 'asoc/fix/qcom-be' into asoc-linus Merge remote-tracking branches 'asoc/fix/samsung', 'asoc/fix/tlv320dac33', 'asoc/fix/wl1273', 'asoc/fix/wm8753', 'asoc/fix/wm8904', 'asoc/fix/wm8958' and 'asoc/fix/wm8983' into asoc-linus Merge remote-tracking branches 'asoc/fix/wm8985', 'asoc/fix/wm8994', 'asoc/fix/wm8996', 'asoc/fix/wm9081' and 'asoc/fix/wm9713' into asoc-linus Robert Jarzmik (1): ASoC: wm9713: fix regmap free path Stephen Boyd (1): ASoC: qcom: Don't specify LE device endianness Takashi Iwai (25): ASoC: dapm: Fix ctl value accesses in a wrong type ASoC: pxa: corgi: Fix enum ctl accesses in a wrong type ASoC: pxa: magician: Fix enum ctl accesses in a wrong type ASoC: pxa: poodle: Fix enum ctl accesses in a wrong type ASoC: pxa: spitz: Fix enum ctl accesses in a wrong type ASoC: pxa: tosa: Fix enum ctl accesses in a wrong type ASoC: omap: n810: Fix enum ctl accesses in a wrong type ASoC: omap: rx51: Fix enum ctl accesses in a wrong type ASoC: intel: mfld: Fix enum ctl accesses in a wrong type ASoC: cs42l51: Fix enum ctl accesses in a wrong type ASoC: da732x: Fix enum ctl accesses in a wrong type ASoC: ab8500: Fix enum ctl accesses in a wrong type ASoC: max98088: Fix enum ctl accesses in a wrong type ASoC: max98095: Fix enum ctl accesses in a wrong type ASoC: tlv320dac33: Fix enum ctl accesses in a wrong type ASoC: wl1273: Fix enum ctl accesses in a wrong type ASoC: wm8753: Fix enum ctl accesses in a wrong type ASoC: wm8904: Fix enum ctl accesses in a wrong type ASoC: wm8958: Fix enum ctl accesses in a wrong type ASoC: wm8983: Fix enum ctl accesses in a wrong type ASoC: wm8985: Fix enum ctl accesses in a wrong type ASoC: wm8994: Fix enum ctl accesses in a wrong type ASoC: wm8996: Fix enum ctl accesses in a wrong type ASoC: wm9081: Fix enum ctl accesses in a wrong type ASoC: wm_adsp: Fix enum ctl accesses in a wrong type include/trace/events/asoc.h | 8 +++---- sound/soc/codecs/ab8500-codec.c | 8 +++---- sound/soc/codecs/adau17x1.h | 6 ++--- sound/soc/codecs/cs42l51.c | 8 +++---- sound/soc/codecs/da732x.c | 8 +++---- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98095.c | 4 ++-- sound/soc/codecs/tlv320dac33.c | 9 ++++--- sound/soc/codecs/wl1273.c | 13 +++++----- sound/soc/codecs/wm8753.c | 6 ++--- sound/soc/codecs/wm8904.c | 4 ++-- sound/soc/codecs/wm8958-dsp2.c | 8 +++---- sound/soc/codecs/wm8983.c | 14 +++++------ sound/soc/codecs/wm8985.c | 14 +++++------ sound/soc/codecs/wm8994.c | 4 ++-- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 8 +++---- sound/soc/codecs/wm9713.c | 2 +- sound/soc/codecs/wm_adsp.c | 8 +++---- sound/soc/fsl/fsl_ssi.c | 42 ++++++++++++++++----------------- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 ++++- sound/soc/intel/boards/mfld_machine.c | 16 ++++++------- sound/soc/intel/skylake/skl-topology.c | 2 +- sound/soc/omap/n810.c | 18 +++++++------- sound/soc/omap/rx51.c | 18 +++++++------- sound/soc/pxa/corgi.c | 12 +++++----- sound/soc/pxa/magician.c | 6 ++--- sound/soc/pxa/poodle.c | 12 +++++----- sound/soc/pxa/spitz.c | 12 +++++----- sound/soc/pxa/tosa.c | 12 +++++----- sound/soc/qcom/lpass-cpu.c | 1 - sound/soc/samsung/i2s.c | 21 ++++++++++------- sound/soc/soc-dapm.c | 8 +++---- 33 files changed, 162 insertions(+), 160 deletions(-)

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Re: [alsa-devel] ERROR: "isa_io_base" [sound/pci/vx222/snd-vx222.ko] undefined!
by Tomi Valkeinen 07 Mar '16

07 Mar '16

On 06/03/16 14:38, kbuild test robot wrote: > Hi Nicolai, > > FYI, the error/warning still remains. > > tree: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git master > head: 67944024c1cdd897e49a09b0d6af3ea38d1388ca > commit: 48fd8ecf29e3c1199432e173858f5ca4bc529738 framebuffer: disable vgacon on microblaze arch > date: 7 months ago > config: microblaze-allmodconfig (attached as .config) > reproduce: > wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross > chmod +x ~/bin/make.cross > git checkout 48fd8ecf29e3c1199432e173858f5ca4bc529738 > # save the attached .config to linux build tree > make.cross ARCH=microblaze > > All errors (new ones prefixed by >>): > >>> ERROR: "isa_io_base" [sound/pci/vx222/snd-vx222.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/trident/snd-trident.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-via82xx.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-via82xx-modem.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-sonicvibes.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-maestro3.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-fm801.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-es1968.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-es1938.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-ens1371.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-ens1370.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-cmipci.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-azt3328.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-als300.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/riptide/snd-riptide.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/pcxhr/snd-pcxhr.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/oxygen/snd-oxygen-lib.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/lx6464es/snd-lx6464es.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/ice1712/snd-ice1724.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/ice1712/snd-ice1712.ko] undefined! I didn't look at this closely, but sounds like disabling vgacon on microblaze made sound drivers fail because they or microblaze does not enable something in the kernel config. Tomi

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[alsa-devel] [PATCH] ASoC: rt5616: add missing mute control for HPVOL
by John Lin 07 Mar '16

07 Mar '16

Add missing kcontrol for HPVOL mute control. Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5616.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index 3704d07..f527b5b 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -309,6 +309,8 @@ static const struct snd_kcontrol_new rt5616_snd_controls[] = { /* Headphone Output Volume */ SOC_DOUBLE("HP Playback Switch", RT5616_HP_VOL, RT5616_L_MUTE_SFT, RT5616_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("HPVOL Playback Switch", RT5616_HP_VOL, + RT5616_VOL_L_SFT, RT5616_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5616_HP_VOL, RT5616_L_VOL_SFT, RT5616_R_VOL_SFT, 39, 1, out_vol_tlv), /* OUTPUT Control */ -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: rsrc-card: tidyup convert-channels explain
by Kuninori Morimoto 07 Mar '16

07 Mar '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt index 5abebf7..255ece3 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt @@ -30,7 +30,7 @@ Optional subnode properties: - frame-inversion : bool property. Add this if the dai-link uses frame clock inversion. - convert-rate : platform specified sampling rate convert -- convert-channels : platform specified channel size convert +- convert-channels : platform specified converted channel size (2 - 8 ch) - audio-prefix : see audio-routing - audio-routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the connection's sink, -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: rt5640: Set PLL src according to source
by Bard Liao 07 Mar '16

07 Mar '16

rt5640_set_dai_pll set pll source according to given source and dai id. However, the pll source should be set according to given source only. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6cd84fb..863c190 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1902,21 +1902,12 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_MCLK); break; case RT5640_PLL1_S_BCLK1: + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK1); + break; case RT5640_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 & RT5640_U_IF1) { - snd_soc_update_bits(codec, RT5640_GLB_CLK, - RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK1); - } - if (dai_sel & RT5640_U_IF2) { - snd_soc_update_bits(codec, RT5640_GLB_CLK, - RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK2); - } + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK2); break; default: dev_err(codec->dev, "Unknown PLL source %d\n", source); -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 0/7] ASoC: rsnd: update SRC activates conditions
by Kuninori Morimoto 07 Mar '16

07 Mar '16

Hi Mark Current Renesas R-Car sound SRC had cared only Playback, because we had heard that it can't use under Capture. But it was telephone-game. accurately, SRC has Sync/Async mode, and it can't use Sync mode when Capture. And, SRC activates conditions is related to CMD, not DVC. These patches update these. Kuninori Morimoto (7): ASoC: rsnd: don't enable SRC Sync mode when Capture with CMD ASoC: rsnd: SRC sync mode is not related to clock master mode ASoC: rsnd: SRC function parameter uses "mod" instead of "src" ASoC: rsnd: rename rsnd_enable_sync_convert() to rsnd_src_sync_is_enabled() ASoC: rsnd: add rsnd_src_get_in/out_rate() ASoC: rsnd: SRC TIMSEL support for Capture ASoC: rsnd: CMD TIMSEL support for Capture sound/soc/sh/rcar/adg.c | 209 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++----------------------------------------------------------- sound/soc/sh/rcar/rsnd.h | 18 ++++++------ sound/soc/sh/rcar/src.c | 95 +++++++++++++++++++++++++++++++------------------------------ sound/soc/sh/rcar/ssi.c | 5 ++-- 4 files changed, 177 insertions(+), 150 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [RFT][PATCH 0/4] ALSA: dice: enabled to handle several streams
by Takashi Sakamoto 07 Mar '16

07 Mar '16

Hi, This patchset adds support for whole available isochronous streams to ALSA dice driver. As a result, for some models, additional PCM substreams are available via subdevice of ALSA PCM character device. In configuration space of alsa-lib, it's represented like 'hw:0,0,1'(card/device/subdevice). It's soon in the last week of this developing cycle, while I hope to finish my work for ALSA dice driver by this patchset. Unfortunately, this patchset adds huge changes and includes possibilities of regression. I'm happy to receive your reports to test with any of your Dice units. For testers, I push whole patches to 'dice-several-streams' branch in my private repository. The changed driver is available from this branch. I recommend to use DKMS to maintain the driver, instead of installing by your own. https://github.com/takaswie/snd-firewire-improve/tree/dice-several-streams To check whether your units support several isochronous streams, please read output from /proc/asound/cardX/dice. When seeing 'tx1' (input) or 'rx1' (output), the additional subdevices are available. For example, $ arecord -l ... card 1: Pro26000f5a [Pro26-000f5a], device 0: DICE [Pro26-000f5a] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 In above case, you can use two subdevices for PCM capture, thus: * arecord -D plughw:1,0,0 file1.wav * arecord -D plughw:1,0,1 file2.wav In the middle of next week, I plan to post this patchset again for merging. So it's nice to test till then. Regards Takashi Sakamoto (4): ALSA: dice: have two sets of isochronous resources/streams ALSA: dice: handle whole available isochronous streams ALSA: dice: handle several PCM substreams when any isochronous streams are available ALSA: dice: force to add two pcm devices for listed models sound/firewire/dice/dice-midi.c | 8 +- sound/firewire/dice/dice-pcm.c | 110 +++++++---- sound/firewire/dice/dice-stream.c | 400 ++++++++++++++++++++++++-------------- sound/firewire/dice/dice.c | 41 ++++ sound/firewire/dice/dice.h | 33 +++- 5 files changed, 401 insertions(+), 191 deletions(-) -- 2.7.0

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[alsa-devel] [PATCH] ASoC: rt5616: add kcontrol for headphone channel switch
by John Lin 07 Mar '16

07 Mar '16

Add kcontrol for headphone channel switch. Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5616.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index 3704d07..b370993 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -309,6 +309,8 @@ static const struct snd_kcontrol_new rt5616_snd_controls[] = { /* Headphone Output Volume */ SOC_DOUBLE("HP Playback Switch", RT5616_HP_VOL, RT5616_L_MUTE_SFT, RT5616_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("HP Channel Switch", RT5616_HP_VOL, + RT5616_VOL_L_SFT, RT5616_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5616_HP_VOL, RT5616_L_VOL_SFT, RT5616_R_VOL_SFT, 39, 1, out_vol_tlv), /* OUTPUT Control */ -- 1.8.1.1.439.g50a6b54

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