June 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fixing invalid fmt SND_SOC_DAIFMT_AC97 test
by Arnaud Mouiche 20 Sep '15

20 Sep '15

SND_SOC_DAIFMT_AC97 (6) has similar bits than SND_SOC_DAIFMT_DSP_A (4) and can't be tested with a simple AND operation Signed-off-by: Arnaud Mouiche <arnaud.mouiche(a)invoxia.com> --- sound/soc/fsl/fsl_ssi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6b0c8f7..c25a1e8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -249,7 +249,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids); static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private) { - return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97); + return (ssi_private->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) == + SND_SOC_DAIFMT_AC97; } static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) @@ -945,7 +946,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, CCSR_SSI_SCR_TCH_EN); } - if (fmt & SND_SOC_DAIFMT_AC97) + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97) fsl_ssi_setup_ac97(ssi_private); return 0; -- 1.9.1

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[alsa-devel] Intel SST on a Bay Trail tablet
by Antonio Ospite 18 Sep '15

18 Sep '15

Hi, I am trying to get the Intel SST driver working on a Teclast X98 Air 3G [1], it's a Bay Trail tablet. The tests below have been made with 4.0.0-rc1 and the firmware files from https://git.kernel.org/cgit/linux/kernel/git/firmware/linux-firmware.git/tr… The DSDT on this device is quite broken, and I am patching it to get stuff working. The audio device works just fine under Android, I don't know if that's because the Android kernel is ignoring the DSDT and is hard-coding some platform data. Anyhow, with the untouched DSDT[2] I was getting this message in dmesg[3]: [ 11.344095] platform byt-rt5640: Driver byt-rt5640 requests probe deferral [ 11.404831] resource sanity check: requesting [mem 0x91000000-0x911fffff], which spans more than ADMA0F28:00 [mem 0x91000000-0x91000fff] [ 11.404833] ------------[ cut here ]------------ [ 11.404849] WARNING: CPU: 0 PID: 213 at arch/x86/mm/ioremap.c:197 __ioremap_caller+0x22c/0x350() ... I guessed this was because the DSDT defined an ADMA0F28 device with a memory resource overlapping with the LPMA device. So I removed the overlapping resources (ADMA0F28 and SSPX0000) in the DSDT and then the driver attempted to boot the audio device but failed with this message in dmesg[4]: [ 11.602291] baytrail-pcm-audio baytrail-pcm-audio: ipc: error DSP boot timeout ... Then I noticed that the Interrupt resources of the LPMA device were out of order in the DSDT, so I put the Interrupt 0x18 as the first one[5] and now the driver can boot the device and upload the firmware to it[6]: [ 11.433460] baytrail-pcm-audio baytrail-pcm-audio: FW version: 04.05.13.a0 [ 11.433468] baytrail-pcm-audio baytrail-pcm-audio: Build type: a0 [ 11.433473] baytrail-pcm-audio baytrail-pcm-audio: Build date: Apr 2 2014 14:14:39 [ 11.448257] byt-rt5640 byt-rt5640: rt5640-aif1 <-> baytrail-pcm-audio mapping ok ... But still no sound. I can see the mixer controls in alsamixer and I tried to initialize them copying from the alsactl output from the Android kernel[7], but no, still no sound. I do not know if that matters, but the driver still emits this message: [ 11.398304] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [ 11.398314] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed -22 ... I noticed that on the Android kernel[8] the DMA channels for the ADMA0F28 device are actually configured, the intel_mid_dma driver sets up 4 channels for it along with 2 x 6 channels for the INTL9C60 device. On the mainline kernel only the INTL9C60 device gets configured, by the dw_dmac driver and with 2 x 8 channels. I do not know much about these details tho. Also, the codec driver on the Android kernel (byt_rt5642) seems to use one GPIO as an interrupt[9]: 390: 0 1 0 0 VLV-GPIO-gpio byt-codec-int Which is pin 4 on GPO2, but the codec driver in the mainline kernel does not seem to be using this; however AFAICS in sound/soc/codecs/rt5640.c a gpio pin in only used for suspend/resume, so it is not strictly needed if it's already in the correct state, right? Last question, what is the difference between having the device detected by sound/soc/intel/sst-acpi.c (like in my case) opposed to sound/soc/intel/sst/sst_acpi.c? I can see the 80860F28 id in both the files but they seem to load different firmwares. Any idea of what is missing? It's the DSDT which still needs adjusting, or are some changes needed (also) in the kernel driver? Thanks, Antonio [1] http://ao2.it/110 [2] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dsdt_sst_original.dsl [3] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_mainline_sst_ori… [4] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_mainline_sst_wit… [5] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dsdt_sst_without_ADMA0… [6] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_mainline_sst_wit… [7] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/android_asound.state [8] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_Android_Teclast_… [9] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/android_proc_interrupt… -- Antonio Ospite http://ao2.it A: Because it messes up the order in which people normally read text. See http://en.wikipedia.org/wiki/Posting_style Q: Why is top-posting such a bad thing?

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[alsa-devel] [PATCH v2 1/2] ASoC: soc-compress: add a config item for soc-compress
by Jie Yang 15 Sep '15

15 Sep '15

We don't always need soc-compress in soc, here add a config item SND_SOC_COMPRESS, please add 'select SND_SOC_COMPRESS' to driver Kconfig when it is needed. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- include/sound/soc.h | 7 +++++++ sound/soc/Kconfig | 5 ++++- sound/soc/Makefile | 3 ++- sound/soc/intel/Kconfig | 1 + 4 files changed, 14 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 93df8bf..cc1cd4f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -440,7 +440,14 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, int snd_soc_platform_write(struct snd_soc_platform *platform, unsigned int reg, unsigned int val); int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); +#if IS_ENABLED(CONFIG_SND_SOC_COMPRESS) int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); +#else +static inline int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) +{ + return -EPERM; +} +#endif struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, const char *dai_link, int stream); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index e2828e1..a124759 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -9,7 +9,6 @@ menuconfig SND_SOC select SND_JACK if INPUT=y || INPUT=SND select REGMAP_I2C if I2C select REGMAP_SPI if SPI_MASTER - select SND_COMPRESS_OFFLOAD ---help--- If you want ASoC support, you should say Y here and also to the @@ -30,6 +29,10 @@ config SND_SOC_GENERIC_DMAENGINE_PCM bool select SND_DMAENGINE_PCM +config SND_SOC_COMPRESS + tristate + select SND_COMPRESS_OFFLOAD + # All the supported SoCs source "sound/soc/adi/Kconfig" source "sound/soc/atmel/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index a0e1ee6..184c1e6 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,6 +1,7 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o +snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o snd-soc-core-objs += soc-topology.o +snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 791953f..e559174 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -12,6 +12,7 @@ config SND_MFLD_MACHINE config SND_SST_MFLD_PLATFORM tristate + select SND_SOC_COMPRESS config SND_SST_IPC tristate -- 1.9.1

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[alsa-devel] [PATCH v3 0/9] asoc: Add audio for sti platforms
by Arnaud Pouliquen 09 Sep '15

09 Sep '15

version 3 : - patch dependency: ASoC: simple card: set cpu-dai sysclk with mclk-fs (Arnaud Pouliquen) - Remark not applied: Use Russell King's patch "sound/core: add IEC958 channel status helper". => This allows to create settings based on runtime, but not compatible with iec958 controls exposed to userland (settings overwritten by snd_pcm_create_iec958_consumer) - main corrections: - rework DAI CPU declaration to use generic dma - Supress abstraction layer for uniperiph player/reader - use dapm for DAC powerup/down version 2: corrections after first review - bindings: . suppress useless dts declaration - uniperif_player.c: . Suppress standby mode in uniplayer ( will be part of a dedicated patch) . Deliver IEC and clock adjustement controls in separate patch - uniperif_reader.c: . kept as separate file as behavior and feature not same than uniplayer . minor corrections - codec/sti-sas.c . implement DAPM for DAC . rework registers declaration to separate device and driver part version 1: - First patches submission Arnaud Pouliquen (9): ASoC: sti: add binding for ASoC driver ASoC: sti: Add uniperipheral header file ASoC: sti: Add CPU DAI driver for playback ASoC: sti: Add CPU DAI driver for capture ASoC: sti: Add uniperipheral dai driver ASoC: Add ability to build sti drivers ASoC: Codec: Add sti platform codec ASoC: sti: Add clock adjustement control ASoC: sti: Add IEC control .../devicetree/bindings/sound/st,sti-asoc-card.txt | 155 +++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sti-sas.c | 627 ++++++++++ sound/soc/sti/Kconfig | 11 + sound/soc/sti/Makefile | 4 + sound/soc/sti/sti_uniperif.c | 254 ++++ sound/soc/sti/uniperif.h | 1229 ++++++++++++++++++++ sound/soc/sti/uniperif_player.c | 1109 ++++++++++++++++++ sound/soc/sti/uniperif_reader.c | 364 ++++++ 12 files changed, 3762 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt create mode 100644 sound/soc/codecs/sti-sas.c create mode 100644 sound/soc/sti/Kconfig create mode 100644 sound/soc/sti/Makefile create mode 100644 sound/soc/sti/sti_uniperif.c create mode 100644 sound/soc/sti/uniperif.h create mode 100644 sound/soc/sti/uniperif_player.c create mode 100644 sound/soc/sti/uniperif_reader.c -- 1.9.1

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[alsa-devel] Fwd: [PATCH 1/1] SPI : spi-pxa2xx : fix spi init of WM510205 codec via ACPI (resend)
by Christian Hartmann 28 Aug '15

28 Aug '15

Hi Mika, Dan et al. @Dan : thank you, I will use get_maintainer.pl as of now. @Mika: you mean I should prepare a new DSDT table and in this case overwrite the SPI value there with the 0 and load it (the DSDT table ) into a custom kernel build ?? I will do this (cause all what I want is to enable the codec and to hear sound) currently I have in the extracted DSDT table from this device the SPI1 controller for the "AUDI" device as seen in this snippet : Scope (_SB) { Device (LPEA) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "80860F28" /* Intel SST Audio DSP */) // _HID: Hardware ID Name (_CID, "80860F28" /* Intel SST Audio DSP */) // _CID: Compatible ID Name (_DDN, "Intel(R) Low Power Audio Controller - 80860F28") // _DDN: DOS Device Name Name (_SUB, "17AA7004") // _SUB: Subsystem ID Name (_UID, One) // _UID: Unique ID Name (_DEP, Package (0x01) // _DEP: Dependencies { ^SPI1.AUDI }) Name (_PR0, Package (0x01) // _PR0: Power Resources for D0 { PLPE }) .... some lines later follows the AUDI device with Device (AUDI) { Name (_HID, "WM510205") // _HID: Hardware ID Name (_CID, "WM510205") // _CID: Compatible ID Name (_DDN, "Wolfson Microelectronics Audio WM5102") // _DDN: DOS Device Name Method (_CRS, 0, NotSerialized) // _CRS: Current Resource Settings { Name (SBUF, ResourceTemplate () { SpiSerialBus (0x0001, PolarityLow, FourWireMode, 0x08, ControllerInitiated, 0x007A1200, ClockPolarityLow, ClockPhaseFirst, "\\_SB.SPI1", 0x00, ResourceConsumer, , ) GpioInt (Edge, ActiveLow, ExclusiveAndWake, PullNone, 0x0000, "\\_SB.GPO2", 0x00, ResourceConsumer, , ) { // Pin list 0x0004 } GpioIo (Exclusive, PullDefault, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.I2C7.PMIC", 0x00, ResourceConsumer, , ) { // Pin list 0x0003 } GpioIo (Exclusive, PullDefault, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.GPO1", 0x00, ResourceConsumer, , ) { // Pin list 0x0017 } }) Return (SBUF) /* \_SB_.SPI1.AUDI._CRS.SBUF */ } .... I t makes sense to me, to change the current value(s) in the DSDT and I have read about it to use a custom DSDT table but I am not 100% sure yet (until I have tested it). If I understand you correct, than all I have to do is to change the 'SPI1' string in this dsdt to SPI0 like ^SPI1.AUDI to ^SPI0.AUDI and this block here SpiSerialBus (0x0001, PolarityLow, FourWireMode, 0x08, ControllerInitiated, 0x007A1200, ClockPolarityLow, ClockPhaseFirst, "\\_SB.SPI1", 0x00, ResourceConsumer, , becomes SpiSerialBus (0x0000, PolarityLow, FourWireMode, 0x08, ControllerInitiated, 0x007A1200, ClockPolarityLow, ClockPhaseFirst, "\\_SB.SPI0", 0x00, ResourceConsumer, , For this case, could you please have a look in the whole DSDT table that I extract from this device? If nothing else seems relevant and the current system dsdt has to be changed as suggested, I will try this out as soon as possible @Jarkko : as you suggested , I attach the current system DSDT as a binary. I have also disassembled, and if someone needs it, I can also paste it somewhere else as a reference After all I will assembly the new dsdt and include it in my custom kernel builld as documented, but will wait for some new infos etc. thank you for you answers @all cheers chris 2015-06-25 9:36 GMT+02:00 Mika Westerberg <mika.westerberg(a)linux.intel.com>: > On Thu, Jun 25, 2015 at 09:29:31AM +0200, Christian Hartmann wrote: >> Hello Maintainer(s)s of SPI, >> >> hope I got all of you maintainers now. This email was already sent out >> last week to the alsa-devel list and I resent it here to you. >> >> I have one patch that enables the spi initialization of the WM510205 audio codec >> Without the patch, this message indicates that the codec chip fails >> with the current chipselect number in spi-pxa2xx: >> >> >> [ 0.296256] pxa2xx-spi 80860F0E:00: cs1 >= max 1 >> [ 0.296270] spi_master spi32766: failed to add SPI device >> WM510205:00 from ACPI >> >> So raise num_chipselect by one to enable the codec. The WM510202 is on >> chipselect = 1. > > The hardware only has one chipselect. > > You can fix this by specifying the correct chipselect in the ACPI > SpiSerialBus connector: > > SpiSerialBus (0, PolarityLow, FourWireMode, 8, ControllerInitiated, .. > > First parameter is the chipselect.

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[alsa-devel] [PATCH v5 0/6] Add dw_hdmi i2s audio support
by Yakir Yang 25 Aug '15

25 Aug '15

As Russell have send an series patches for dw_hdmi audio, and most important it works well on rockchip platform. Cause some patches have been achieve in Russell's series, so I abondon some patches in this verison. Here are the abondon patches that already be achieved: * drm: bridge/dw_hdmi: adjust n/cts setting order * drm: bridge/dw_hdmi: set ncts_atomic_write & cts_manual * drm: bridge/dw_hdmi: combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() * drm: bridge/dw_hdmi: add enable/disable to dw_hdmi_audio callbacks And I also abondon some interrupt patches that is not a common bug in dw_hdmi driver, but is a rockchip platform bug which HDMI registers would reset by CRU module after system suspend/resume (in this case HPD interrupt mask would reset, then driver can't respond hot plug event after system resume). Here are the abondon patches that used to "fix" dw_hdmi interrupt bug: * drm: bridge/dw_hdmi: add irq control to suspend/resume * drm: bridge/dw_hdmi: wrap irq control in fucntions * drm: rockchip/dw_hdmi_rockchip: add resume/suspend support Here are the leave list that I have make in my series: - add n/cts combinations for more display modes - enable audio support when sink device is HDMI monitor and has audio - rename dw_hdmi-ahb-audio.h to dw_hdmi-audio.h - add dw_hdmi i2s audio driver - add rockchip_hdmi_audio sound card driver - add dt-bindings for documentation for rockchip_hdmi_audio driver Changes in v5: - make more words in commit message. - Take Russell suggest that create "sink_has_audio" and "sink_is_hdmi" in struct hdmi, and keep vmode's mdvi changeless. - Config fc_invidconf to HDMI mode when sink device is HDMI monitor. - Take Mark Brown suggest that remove jack_status recorded, report jack status directly when hdmi plug happend, and remove devm_kfree and devm_free_irq. - Correct the MODULE_LICENSE to "GPL v2" - Move source from sound/soc/codecs to drivers/gpu/drm/bridge/ - Rename dai driver name to "dw-hdmi-i2s-hifi" - Take Mark Brown suggest that remove the no useful code in probe func, and remove the snd_soc_dai_set_fmt() snd_soc_unregister_card(), remove those noisy dev_info() - Rename codec_dai_name to "dw-hdmi-i2s-hifi" Changes in v4: - Take Dainel suggest that introduce drm_detect_monitor_audio() to judge whether source should config audio. - Replace delaywork with irq thread, and add suspend/resume interfaces, replace "dw-hdmi-audio" with consecutive strings. - Add ".pm = &snd_soc_pm_ops," Changes in v3: - Keep audio format config function in dw-hdmi-audio driver and remove audio_config & get_connect_status functions, move jack control to dw-hdmi-audio completely. - Delete the operation of jack in rockchip-hdmi-audio driver, get ready to switch to simple-audio-card driver. - modify cpu-of-node to i2s-controller Changes in v2: - Update dw_hdmi audio control interfaces, and adjust jack report process - give "codec-name" & "codec-dai-name" an const name - remove codec-name and codec-dai-name - rename rockchip,rockchip-hdmi-audio.txt to rockchip,rockchip-dw-hdmi-audio.txt Yakir Yang (6): drm: bridge/dw_hdmi: add audio support for more display resolutions drm: bridge/dw_hdmi: enable audio when sink device is HDMI and has audio drm: bridge/dw_hdmi: rename dw_hdmi-ahb-audio.h to dw_hdmi-audio.h drm: bridge/dw_hdmi-i2s-audio: add audio driver ASoC: rockchip/rockchip-hdmi-audio: add sound driver for hdmi audio dt-bindings: Add documentation for rockchip_hdmi_audio driver .../sound/rockchip,rockchip-dw-hdmi-audio.txt | 12 + drivers/gpu/drm/bridge/Kconfig | 9 + drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.c | 2 +- .../{dw_hdmi-ahb-audio.h => dw_hdmi-audio.h} | 0 drivers/gpu/drm/bridge/dw_hdmi-i2s-audio.c | 398 +++++++++++++++++++++ drivers/gpu/drm/bridge/dw_hdmi.c | 100 +++++- drivers/gpu/drm/bridge/dw_hdmi.h | 3 + sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_hdmi_audio.c | 138 +++++++ 11 files changed, 658 insertions(+), 16 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt rename drivers/gpu/drm/bridge/{dw_hdmi-ahb-audio.h => dw_hdmi-audio.h} (100%) create mode 100644 drivers/gpu/drm/bridge/dw_hdmi-i2s-audio.c create mode 100644 sound/soc/rockchip/rockchip_hdmi_audio.c -- 2.1.2

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[alsa-devel] [patch] ALSA: hdsp: silence a sprinft() overflow warning
by Dan Carpenter 22 Aug '15

22 Aug '15

card->shortname is a 32 char string so the sprintf() can theoretically overflow. snd_rawmidi_new() can accept strings up to 64 bytes long. I have made the temporay buf[] array 40 bytes long and changed the sprintf() to snprintf(). Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index c19e021..c1fe1d3 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -1526,7 +1526,7 @@ static struct snd_rawmidi_ops snd_hdsp_midi_input = static int snd_hdsp_create_midi (struct snd_card *card, struct hdsp *hdsp, int id) { - char buf[32]; + char buf[40]; hdsp->midi[id].id = id; hdsp->midi[id].rmidi = NULL; @@ -1537,7 +1537,7 @@ static int snd_hdsp_create_midi (struct snd_card *card, struct hdsp *hdsp, int i hdsp->midi[id].pending = 0; spin_lock_init (&hdsp->midi[id].lock); - sprintf (buf, "%s MIDI %d", card->shortname, id+1); + snprintf(buf, sizeof(buf), "%s MIDI %d", card->shortname, id + 1); if (snd_rawmidi_new (card, buf, id, 1, 1, &hdsp->midi[id].rmidi) < 0) return -1;

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[alsa-devel] BugReport: Dice works partially [Broken]
by Stanislav Sergiienko 16 Aug '15

16 Aug '15

Hi, I am a user of TC Electronic DesktopKonnekt6 and latest kernel update 3.19.2 add partial support for my device and thus broke my usual sound workflow. So i want to make good bug report to make device usable on latest kernels. With 3.19.2 kernel device is successfully recognized by system and even sometimes can play sound(!). But always, after some time sound is breaks (usually after 30-90 seconds) and to restore it i must restart pulseaudio(v6.0) and application. So device becomes unusable on latests kernels. Earlier i use my sound card through ffado + jackdbus + pulseaudio. Now it's impossible, so i have to downgrade kernel. Please request any additional info that you need to fix problem and i will post it here soon. Also i can test any "bleeding edge" changes thru dkms as described here: https://github.com/takaswie/snd-firewire-improve Related bug(not mine): https://bugs.archlinux.org/task/44258 Some useful info: cat /proc/asound/cards 0 [DesktopKonnekt6]: DICE - DesktopKonnekt6 TC Electronic DesktopKonnekt6 (serial 12941) at fw1.0, S400 cat /proc/asound/card0/dice sections: global: offset 10, size 90 tx: offset 100, size 142 rx: offset 242, size 282 ext_sync: offset 524, size 4 unused2: offset 0, size 0 global: owner: ffc1:000100000000 notification: 00000040 nick name: DesktopKonnekt6 clock select: internal 44100 enable: 1 status: locked 44100 ext status: 00000000 sample rate: 44100 version: 1.0.4.0 clock caps: 44100 48000 88200 96000 176400 192000 arx1 arx2 internal clock source names: Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\INTERNAL\\ tx 0: iso channel: 0 audio channels: 6 midi ports: 0 speed: S400 names: ch 1\ch 2\ - \ - \ - \ - \\ ac3 caps: 00000000 ac3 enable: 00000000 rx 0: iso channel: 1 sequence start: 0 audio channels: 6 midi ports: 0 names: main out 1\main out 2\phones out 1\phones out 2\ - \ - \\ ac3 caps: 00000000 ac3 enable: 00000000 ext status: clock source: internal locked: 1 rate: 44100 adat user data: 0

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[alsa-devel] [PATCH 1/6] ASoC: fsl_ssi: enable IPG clock during AC'97 reg access
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

IPG clock have to be enabled during AC'97 CODEC register access in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c7647e0..9c46c7d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1127,10 +1127,17 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, struct regmap *regs = fsl_ac97_data->regs; unsigned int lreg; unsigned int lval; + int ret; if (reg > 0x7f) return; + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 write clk_prepare_enable failed: %d\n", + ret); + return; + } lreg = reg << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1141,6 +1148,8 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, regmap_update_bits(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_RDWR_MASK, CCSR_SSI_SACNT_WR); udelay(100); + + clk_disable_unprepare(fsl_ac97_data->clk); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1151,6 +1160,14 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short val = -1; u32 reg_val; unsigned int lreg; + int ret; + + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 read clk_prepare_enable failed: %d\n", + ret); + return -1; + } lreg = (reg & 0x7f) << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1162,6 +1179,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, regmap_read(regs, CCSR_SSI_SACDAT, &reg_val); val = (reg_val >> 4) & 0xffff; + clk_disable_unprepare(fsl_ac97_data->clk); + return val; }

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[alsa-devel] [PATCH 2/6] ASoC: fsl_ssi: AC'97 DAI driver needs probe method too
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

AC'97 DAI driver struct need the same probe method as I2S one to setup DMA params in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9c46c7d..2ce9e1d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1101,6 +1101,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = { static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .bus_control = true, + .probe = fsl_ssi_dai_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 2,

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