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

[alsa-devel] [PATCH 0/2] Support non-ssi cpu-dai for sgtl5000
by Winter Wang 30 Aug '16

30 Aug '16

These patches support sgtl5000 to be attached to other non-ssi cpu-dais like SAI. Winter Wang (2): ASoC: imx-sgtl5000: Support non-ssi cpu-dai ARM: dts: modify sgtl5000 property to support non-ssi cpu-dai .../bindings/sound/imx-audio-sgtl5000.txt | 4 +-- arch/arm/boot/dts/imx25-pdk.dts | 2 +- arch/arm/boot/dts/imx51-babbage.dts | 2 +- arch/arm/boot/dts/imx53-m53evk.dts | 2 +- arch/arm/boot/dts/imx53-mba53.dts | 2 +- arch/arm/boot/dts/imx53-qsb-common.dtsi | 2 +- arch/arm/boot/dts/imx53-tx53.dtsi | 2 +- arch/arm/boot/dts/imx53-voipac-bsb.dts | 2 +- arch/arm/boot/dts/imx6dl-riotboard.dts | 2 +- arch/arm/boot/dts/imx6q-bx50v3.dtsi | 2 +- arch/arm/boot/dts/imx6q-gw5400-a.dts | 2 +- arch/arm/boot/dts/imx6q-h100.dts | 2 +- arch/arm/boot/dts/imx6q-icore-rqs.dts | 2 +- arch/arm/boot/dts/imx6q-tbs2910.dts | 2 +- arch/arm/boot/dts/imx6qdl-apalis.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-apf6dev.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-gw52xx.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-gw53xx.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-gw54xx.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-hummingboard.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-nit6xlite.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-nitrogen6_max.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-nitrogen6x.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-rex.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-sabrelite.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-tx6.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-wandboard.dtsi | 2 +- sound/soc/fsl/imx-sgtl5000.c | 41 ++++++++++++++-------- 28 files changed, 55 insertions(+), 42 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v6 0/2] amidi delay option
by Felipe F. Tonello 30 Aug '16

30 Aug '16

Small improvement to amidi, adding useful functionality. changes from v5: * wait until MIDI message is competely sent to sleep changes from v4: * simplified code changes from v3: * fixed a bug if memchr() returns NULL changes from v2: * only check of end of sysex (0xf7) * use memchr() instead of direct loop approach * fixed commit message changes from v1: * remove delay in between any MIDI message, only SysEx * added patch to use getopt_long() arguments Felipe F. Tonello (2): amidi: add delay option amidi: use GNU getopt_long() proper argument names amidi/amidi.1 | 14 ++++++++++ amidi/amidi.c | 82 ++++++++++++++++++++++++++++++++++++++++++++++------------- 2 files changed, 79 insertions(+), 17 deletions(-) -- 2.9.3

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Re: [alsa-devel] No sound for several minutes after reboot
by Takashi Iwai 30 Aug '16

30 Aug '16

On Sat, 27 Aug 2016 13:39:08 +0200, Markus Trippelsdorf wrote: > > On 2016.08.27 at 11:50 +0200, Markus Trippelsdorf wrote: > > Hi, > > > > since several kernel releases I get no sound for several minutes after > > reboot. After this delay it starts working normally. > > > > Is this some kind of speaker protection? Is there a parameter that I can > > set to get sound immediately? > > > > % aplay -l > > **** List of PLAYBACK Hardware Devices **** > > card 0: SB [HDA ATI SB], device 0: VT1708S Analog [VT1708S Analog] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > card 0: SB [HDA ATI SB], device 3: VT1708S Digital [VT1708S Digital] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > card 1: HDMI [HDA ATI HDMI], device 3: Generic Digital [Generic Digital] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > > > card 0, device 3 [VT1708S Digital] is the device that I use. It is a > > TOSLINK-S/PDIF link to my DAC. > > Here is the diff (bad (after reboot) vs. good (after a few minutes)): > > @@ -121,7 +121,7 @@ > Control: name="Surround Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 > - Amp-Out vals: [0x1c 0x1c] > + Amp-Out vals: [0x26 0x26] > Converter: stream=0, channel=0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > @@ -174,12 +174,12 @@ > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power states: D0 D1 D2 D3 > - Power: setting=D0, actual=D0 > + Power: setting=D3, actual=D3 > Connection: 1 > 0x1e So this looks like the only significant change. Others are about the volume level, and should lead to the complete mute. Could you try to adjust the power state manually while it happens? hda-verb /dev/snd/hwC0D0 0xXX SET_POWER 0x00 where 0xXX is filled with the node number of this widget. BTW, at the next time, take a diff with -p option. This will show the node ID in the diff. Takashi

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[alsa-devel] iMX6SX having SSI TXC from external and TXFS from internal, is it possible?
by Roberto Fichera 30 Aug '16

30 Aug '16

Hi, I'm playing with a custom iMX6SX board having the SSI connected to a SLIC in TDM mode. SSI has TXC (PCLK) coming from an external oscillator running at 2.048MHz while TXFS (FSYNC) is set to be generated from internal or at least this is what I'm trying to do. The SSI documentation is not really clear in case, so wondering to know if it is possible to generate FSYNC with this particular configuration: SSI set in I2S normal mode, synchronous transfers, network mode, TXC taken from external oscillator. The relevant part reported in iMX6SXRM doc (62.8.4 SSI Clocking) is: "In master and synchronous mode, the unused port SRCK is used as network clock (oversampling clock) enabled by the SCR register bit 15, SYS_CLK_EN. This network clock is an oversampling clock of the frame sync clock (STFS). In this mode, the word length (WL), Prescaler Range (PSR), Prescaler Modulus (PM) and Frame rate (DC) selects the ratio of network clock to sampling clock STFS. In case of I2S mode, the network clock (oversampling clock) can be made available on this port if the SYS_CLK_EN bit is set." Accordingly with Figure 62-22 in the iMX6SX reference manual network clock on SRCK can be activated only when both TXDIR and SYS_CLK_EN are set. But if TXDIR is set I cannot get STCK from external oscillator because SSI is now generating it from internal. I've also changed the way how audmux is routing FYSNC, internal port is SSI1 and external is SSI6 int audmux_config(unsigned int master, unsigned int slave) { unsigned int ptcr, pdcr; ptcr = IMX_AUDMUX_V2_PTCR_SYN /* Synchronous mode */ | IMX_AUDMUX_V2_PTCR_TCSEL(slave); /* transmit clock select from external port */ pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(slave); audmux_v2_configure_port(master, ptcr, pdcr); /* configure internal port */ ptcr = IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSDIR /* Transmit Frame Sync from Internal */ | IMX_AUDMUX_V2_PTCR_TFSEL(master); /* from master port */ pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(master); audmux_v2_configure_port(slave, ptcr, pdcr); /* configure external port */ return 0; } and SSI1 is set as: scr = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_SYN | /* Synchronous mode */ CCSR_SSI_SCR_NET; /* Network mode to enable TDM */ stcr = CCSR_SSI_STCR_TXBIT0 | /* LSB Aligned */ CCSR_SSI_STCR_TFDIR | /* Transmit Frame Direction Internal - generated internally */ CCSR_SSI_STCR_TFEN0 | /* Enable TX FIFO0 */ CCSR_SSI_STCR_TSCKP; /* Transmit Clock Polarity - Data Clocked out on falling edge */ Doesn't anyone know if it's possible to have this particular TDM configuration? Thanks in advance, Roberto Fichera.

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[alsa-devel] [PATCH 0/4] ASoC: dapm: Various fixes for dapm kcontrols
by Chen-Yu Tsai 30 Aug '16

30 Aug '16

Hi everyone, This series is a bunch of fixes for some problems I encountered while experimenting with various dapm kcontrols for Allwinner A31 support in sun4i-codec. Patch 1 fixes a possible uninitialized variable usage in snd_soc_dapm_get_volsw(). This doesn't show up as a compiler warning but I'm fairly certain it might happen. Patch 2 fixes setting the value for an enum kcontrol's second channel. Patch 3 fixes kcontrol creation for output driver widgets. An output driver widget is the same as a PGA widget, just with a different priority. Patch 4 adds an error message if anyone attempts to add hardware backed kcontrols on PGA widgets. Originally kcontrols on PGAs weren't supported at all. Support for them on virtual widgets were added later, but a guard for real widgets was missing. Regards ChenYu Chen-Yu Tsai (4): ASoC: dapm: Fix possible uninitialized variable in snd_soc_dapm_get_volsw() ASoC: dapm: Fix value setting for _ENUM_DOUBLE MUX's second channel ASoC: dapm: Fix kcontrol creation for output driver widget ASoC: dapm: Give error when adding hardware backed controls for PGA widgets sound/soc/soc-dapm.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) -- 2.9.3

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[alsa-devel] [PATCH v2] ASoC: core: fix shift used for second item in snd_soc_get_enum_double
by Jaswinder Jassal 30 Aug '16

30 Aug '16

Incorrect shift value was being used to extract the second item. Signed-off-by: Jaswinder Jassal <jjassal(a)opensource.wolfsonmicro.com> --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index a513a34..9fc1a7b 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -77,7 +77,7 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, 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_r) & e->mask; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = item; } -- 2.1.4

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[alsa-devel] Range of pitchbend MIDI events
by Maurizio Berti 30 Aug '16

30 Aug '16

Hello, I was playing with pyalsa and I realized an unexpected behavior with pitchbend events. The pitchbend range should be, by MIDI standard, between -8192 and +8191, or 0-16384. Being a 3 byte event, it should ignore any subsequent byte received. Using pyalsa I'm able to create events with values greater than 127 and they are translated using as many bytes as possible; in fact, if I create a note event with a note value of 8193, I receive a value 1 (using the first byte of a 2 byte value with 7bit format). That's fine, I suppose it's by architecture (I'm not a programmer, I just play around with python, I'm just assuming this). The problem comes with pitchbend values, since it looks like alsa accepts values greater than 8191 and does not ignore the exceeding bytes as it should; that is, I'm able to send 8192, and that's what vkeybd does. The transformation here is the issue, since receiving programs seem to translate only the latest two bytes, that's why vkeybd (and other software too) shouldn't have a 8192 value. That can be confusing, for example, using a2jmidi: the end of the pitchbend scale is suddenly a 0 value; I know that vkeybd is not the ultimate software in music production, but, still, it shouldn't be able to do so, I think :) I don't know if, maybe, there's some less known aspect regarding pitchbend, and, as I mentioned before, I don't know a thing about programming, nor how actually works ALSA under the hood. Anyway, I think there should be some kind of range control, at least with pyalsa, if not by alsa at all. What do you think? Regards, Maurizio -- È difficile avere una convinzione precisa quando si parla delle ragioni del cuore. - "Sostiene Pereira", Antonio Tabucchi http://www.jidesk.net

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[alsa-devel] [PATCH v2] ALSA: hda - Enable subwoofer on Dell Inspiron 7559
by Kai-Heng Feng 30 Aug '16

30 Aug '16

The subwoofer on Inspiron 7559 was disabled originally. Applying a pin fixup to node 0x1b can enable it and make it work. Old pin: 0x411111f0 New pin: 0x90170151 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7100f05..2a3dd18 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4855,6 +4855,7 @@ enum { ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, ALC298_FIXUP_SPK_VOLUME, + ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER, }; static const struct hda_fixup alc269_fixups[] = { @@ -5516,6 +5517,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, }, + [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x90170151 }, + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5560,6 +5570,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), -- 2.9.3

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[alsa-devel] [PATCH 00/12] ASoC: hdac: Add hdac generic driver
by Subhransu S. Prusty 30 Aug '16

30 Aug '16

HDA devices generically can be modelled with DAPM in ASoC. We would like to create a framework for the HDA devices in ASoC. This will follow HDA standard. This will be registered as generic "virtual" class device (Not implemented in this series). If there is no match found for vendor id and device id, then the class driver will be loaded. For vendor specific devices, vendor quirks will be loaded based on the match. Additional widgets, controls and route will be created in a vendor specific way through vendor ops. This series adds a framework in ASoC to model HDA codecs. HDA widgets are enumerated and one or multiple DAPM widget(s) are created. Connection list is queried for each widget to identify the connection between two endpoints and modelled using DAPM graph. Set of event handlers are defined for each widget type. Based on DAPM events required verbs are sent to program codec. Finally a function is exported to query for the device endpoint configuration to create machine controls. Features yet to be added: - Optional features of widgets will be added using additional dapm widgets or controls - Pin retasking will be done by adding all possible dapm route map - Class driver to load the generic driver if no match found - Jack detection Subhransu S. Prusty (12): ALSA: hdac: Add codec helper library ALSA: hda - Add macro to test pin widget's input capability ASoC: hdac: Add a generic hdac driver framework ASoC: hdac: Create DAPM model for HDA widgets ALSA: hdac: Move 'to_hda_ext_device' helper to core ASoC: hdac_hdmi: use get_edev helper from core ASoC: hdac: Build DAPM graph by querying through widget connection list ASoC: hdac: Register widget event handlers ALSA: hda - macro to get default config device of pin widgets ASoC: hdac: Export API to create machine controls ALSA: hdac: Implement a match function ASoC: hdac: Add coefficient init callback for RT286 include/sound/hdaudio.h | 1 + include/sound/hdaudio_ext.h | 1 + sound/hda/ext/Makefile | 3 +- sound/hda/ext/hdac_codec.c | 182 ++++ sound/hda/ext/hdac_codec.h | 60 ++ sound/hda/ext/hdac_ext_bus.c | 12 + sound/hda/local.h | 13 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_generic.c | 1754 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hdac_generic.h | 36 + sound/soc/codecs/hdac_hdmi.c | 8 +- 12 files changed, 2069 insertions(+), 8 deletions(-) create mode 100644 sound/hda/ext/hdac_codec.c create mode 100644 sound/hda/ext/hdac_codec.h create mode 100644 sound/soc/codecs/hdac_generic.c create mode 100644 sound/soc/codecs/hdac_generic.h -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - Enable subwoofer on Dell Inspiron 7559
by Kai-Heng Feng 30 Aug '16

30 Aug '16

The subwoofer on Inspiron 7559 does not work originally. Applying a pin fixup can make it work. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7100f05..2a3dd18 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4855,6 +4855,7 @@ enum { ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, ALC298_FIXUP_SPK_VOLUME, + ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER, }; static const struct hda_fixup alc269_fixups[] = { @@ -5516,6 +5517,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, }, + [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x90170151 }, + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5560,6 +5570,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), -- 2.9.3

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