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

[alsa-devel] After kernel upgrade to v ~~> 4.1.0, no audio output unless UNPLUG/RE-PLUGIN of audio jack on card; OK with earlier kernels
by robert.devanna＠nospammail.net 06 Jul '15

06 Jul '15

I run Opensuse 13.2/64. My audio system has been / is ALSA, not PulseAudio. I have installed rpm -qa | grep -i alsa alsa-utils-1.0.29-143.1.x86_64 alsa-plugins-jack-1.0.29-99.1.x86_64 alsa-firmware-1.0.29-33.1.noarch alsa-devel-1.0.29-218.1.x86_64 alsa-plugins-1.0.29-99.1.x86_64 alsa-1.0.29-218.1.x86_64 alsa-oss-1.0.28-51.1.x86_64 alsa-docs-1.0.29-218.1.noarch It's run fine for ages. In the past week -- after a bunch of zypper (distro packages) upgrades, and no I don't know which one specifically -- I now have an audio problem. When I boot my machine there's no audio output. Every alsa/sound diagnostic I ran said everything's OK. Took me awhile to figure out that simply UNPLUGGING the audio jack from the mobo socket, then replugging it in fixes the audio -- immediately. Chatting in #irc, suggestion was to check kernel version dependency. I'm currently on latest stable kernel 4.1.X. uname -rm 4.1.1-1.ga46abf6-desktop x86_64 I am absolutely sure that this was NOT a problem in the 4.0.X kernel branch. I am pretty-darn-sure that this was NOT a problem in earlier 4.1.X branch. Unfortunately, I only had access to 4.1.1-1.1.ga46abf6 4.1.0-3.g5ee367d and 3.16.7-21.1.x86_64 no longer any earlier 4.1.X or 4.0.X kernels. Testing each of those 4.1.1-1.1.ga46abf6 --> FAILs 4.1.0-3.g5ee367d --> FAILs 3.16.7-21.1.x86_64 --> WORKs so at least from 3.16.7 -> 4.1.0 there's a kernel dependency. Again, I'm pretty certain that this was NOT a problem with earlier 4.1.X and 4.0.X kernels; just can't prove it. As instructed in IRC, I DL'd and exec'd the 'alsa-info.sh' script. I ran the script in 2 cases: [1] boot to console, kernel 4.1.1-1.1.ga46abf6, no audio output [2] boot to console, kernel 4.1.1-1.1.ga46abf6, unplug & replug, OK audio output First, the DIFF between the two shows there IS a change on UNPLUG/REPLUG of the jack, DIFF: ------------------------------------------ --- alsa-info.txt.BEFORE 2015-07-05 12:32:44.973554471 -0700 +++ alsa-info.txt.AFTER 2015-07-05 12:34:01.935900433 -0700 @@ -3,7 +3,7 @@ !!ALSA Information Script v 0.4.64 !!################################ -!!Script ran on: Sun Jul 5 19:32:43 UTC 2015 +!!Script ran on: Sun Jul 5 19:34:00 UTC 2015 !!Linux Distribution @@ -332,7 +332,7 @@ Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 - Power: setting=D3, actual=D3 + Power: setting=D0, actual=D0 Connection: 1 0x16 Node 0x1d [Pin Complex] wcaps 0x40058d: Stereo Amp-Out @@ -602,7 +602,7 @@ crw-rw---- 1 root audio 116, 6 Jul 5 12:31 /dev/snd/hwC0D0 crw-rw---- 1 root audio 116, 10 Jul 5 12:31 /dev/snd/hwC2D0 crw-rw---- 1 root audio 116, 4 Jul 5 12:31 /dev/snd/pcmC0D0c -crw-rw---- 1 root audio 116, 3 Jul 5 12:32 /dev/snd/pcmC0D0p +crw-rw---- 1 root audio 116, 3 Jul 5 12:33 /dev/snd/pcmC0D0p crw-rw---- 1 root audio 116, 5 Jul 5 12:31 /dev/snd/pcmC0D3p crw-rw---- 1 root audio 116, 12 Jul 5 12:31 /dev/snd/pcmC1D0c crw-rw---- 1 root audio 116, 8 Jul 5 12:31 /dev/snd/pcmC2D3p ------------------------------------------ Since the output's too big for the list, apparently, I pastebin'd the full script output [1] -> http://pastebin.com/jUyTRrFG [2] -> http://pastebin.com/4V8YCguV - Rob

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[alsa-devel] [PATCH] arm64: dts: mt8173: Add afe device node
by Koro Chen 06 Jul '15

06 Jul '15

This adds afe (audio front end) device node to the MT8173 dtsi file. This patch is based on Matthias's tree: https://github.com/mbgg/linux-mediatek branch: v4.2-next/arm64 Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- arch/arm64/boot/dts/mediatek/mt8173.dtsi | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/arch/arm64/boot/dts/mediatek/mt8173.dtsi b/arch/arm64/boot/dts/mediatek/mt8173.dtsi index 0696f8f..f8afab4 100644 --- a/arch/arm64/boot/dts/mediatek/mt8173.dtsi +++ b/arch/arm64/boot/dts/mediatek/mt8173.dtsi @@ -14,6 +14,7 @@ #include <dt-bindings/clock/mt8173-clk.h> #include <dt-bindings/interrupt-controller/irq.h> #include <dt-bindings/interrupt-controller/arm-gic.h> +#include <dt-bindings/power/mt8173-power.h> #include <dt-bindings/reset-controller/mt8173-resets.h> #include "mt8173-pinfunc.h" @@ -393,6 +394,37 @@ #size-cells = <0>; status = "disabled"; }; + + afe: mt8173-afe-pcm@11220000 { + compatible = "mediatek,mt8173-afe-pcm"; + reg = <0 0x11220000 0 0x1000>; + interrupts = <GIC_SPI 134 IRQ_TYPE_EDGE_FALLING>; + power-domains = <&scpsys MT8173_POWER_DOMAIN_AUDIO>; + clocks = <&infracfg CLK_INFRA_AUDIO>, + <&topckgen CLK_TOP_AUDIO_SEL>, + <&topckgen CLK_TOP_AUD_INTBUS_SEL>, + <&topckgen CLK_TOP_APLL1_DIV0>, + <&topckgen CLK_TOP_APLL2_DIV0>, + <&topckgen CLK_TOP_I2S0_M_SEL>, + <&topckgen CLK_TOP_I2S1_M_SEL>, + <&topckgen CLK_TOP_I2S2_M_SEL>, + <&topckgen CLK_TOP_I2S3_M_SEL>, + <&topckgen CLK_TOP_I2S3_B_SEL>; + clock-names = "infra_sys_audio_clk", + "top_pdn_audio", + "top_pdn_aud_intbus", + "bck0", + "bck1", + "i2s0_m", + "i2s1_m", + "i2s2_m", + "i2s3_m", + "i2s3_b"; + assigned-clocks = <&topckgen CLK_TOP_AUD_1_SEL>, + <&topckgen CLK_TOP_AUD_2_SEL>; + assigned-clock-parents = <&topckgen CLK_TOP_APLL1>, + <&topckgen CLK_TOP_APLL2>; + }; }; }; -- 1.8.1.1.dirty

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[alsa-devel] [PATCH 1/3] ASoC: fsl_esai: Add driver suspend and resume to support MEGA Fast
by Zidan Wang 06 Jul '15

06 Jul '15

For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of modules during system suspend and resume procedure. Thus, SAI needs to save all the values of registers before the system suspend and restore them after the system resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_esai.c | 69 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 69 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 5c75971..b749a29 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -684,6 +684,32 @@ static bool fsl_esai_readable_reg(struct device *dev, unsigned int reg) } } +static bool fsl_esai_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ESAI_ETDR: + case REG_ESAI_ERDR: + case REG_ESAI_ESR: + case REG_ESAI_TFSR: + case REG_ESAI_RFSR: + case REG_ESAI_TX0: + case REG_ESAI_TX1: + case REG_ESAI_TX2: + case REG_ESAI_TX3: + case REG_ESAI_TX4: + case REG_ESAI_TX5: + case REG_ESAI_RX0: + case REG_ESAI_RX1: + case REG_ESAI_RX2: + case REG_ESAI_RX3: + case REG_ESAI_SAISR: + return true; + default: + return false; + } + +} + static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -722,7 +748,9 @@ static const struct regmap_config fsl_esai_regmap_config = { .max_register = REG_ESAI_PCRC, .readable_reg = fsl_esai_readable_reg, + .volatile_reg = fsl_esai_volatile_reg, .writeable_reg = fsl_esai_writeable_reg, + .cache_type = REGCACHE_RBTREE, }; static int fsl_esai_probe(struct platform_device *pdev) @@ -853,10 +881,51 @@ static const struct of_device_id fsl_esai_dt_ids[] = { }; MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids); +#if CONFIG_PM_SLEEP +static int fsl_esai_suspend(struct device *dev) +{ + struct fsl_esai *esai = dev_get_drvdata(dev); + + regcache_cache_only(esai->regmap, true); + regcache_mark_dirty(esai->regmap); + + return 0; +} + +static int fsl_esai_resume(struct device *dev) +{ + struct fsl_esai *esai = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(esai->regmap, false); + + /* FIFO reset for safety */ + regmap_update_bits(esai->regmap, REG_ESAI_TFCR, + ESAI_xFCR_xFR, ESAI_xFCR_xFR); + regmap_update_bits(esai->regmap, REG_ESAI_RFCR, + ESAI_xFCR_xFR, ESAI_xFCR_xFR); + + ret = regcache_sync(esai->regmap); + if (ret) + return ret; + + /* FIFO reset done */ + regmap_update_bits(esai->regmap, REG_ESAI_TFCR, ESAI_xFCR_xFR, 0); + regmap_update_bits(esai->regmap, REG_ESAI_RFCR, ESAI_xFCR_xFR, 0); + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_esai_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(fsl_esai_suspend, fsl_esai_resume) +}; + static struct platform_driver fsl_esai_driver = { .probe = fsl_esai_probe, .driver = { .name = "fsl-esai-dai", + .pm = &fsl_esai_pm_ops, .of_match_table = fsl_esai_dt_ids, }, }; -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - add codec ID for Broxton display audio codec
by libin.yang＠intel.com 06 Jul '15

06 Jul '15

From: Libin Yang <libin.yang(a)intel.com> This patch adds codec ID (0x8086280a) and module alias for Broxton display codec. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2f24338..9515891 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3527,6 +3527,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x80862807, .name = "Haswell HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862808, .name = "Broadwell HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862809, .name = "Skylake HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x8086280a, .name = "Broxton HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862882, .name = "Valleyview2 HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862883, .name = "Braswell HDMI", .patch = patch_generic_hdmi }, @@ -3591,6 +3592,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862806"); MODULE_ALIAS("snd-hda-codec-id:80862807"); MODULE_ALIAS("snd-hda-codec-id:80862808"); MODULE_ALIAS("snd-hda-codec-id:80862809"); +MODULE_ALIAS("snd-hda-codec-id:8086280a"); MODULE_ALIAS("snd-hda-codec-id:80862880"); MODULE_ALIAS("snd-hda-codec-id:80862882"); MODULE_ALIAS("snd-hda-codec-id:80862883"); -- 1.9.1

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[alsa-devel] [PATCH] ALSA: oxygen: Fix S/PDIF muting
by Laurence Darby 05 Jul '15

05 Jul '15

The S/PDIF output was muted whenever audio wasn't playing which resulted in a clicking noise from the DAC when resuming (similar to the issue in 3bef1c377d1b (ALSA: hda - stop setup_dig_out_stream() causing clicks)). Also, the mixer control wasn't actually controlling the S/PDIF output. - Fix the initial state of the mixer to match the initial muted state. - Mute the output at shutdown. - Delete the code that modifies OXYGEN_SPDIF_OUT_ENABLE, and modify it only in oxygen_update_spdif_source(), so only the mixer control controls it. Signed-off-by: Laurence Darby <ldarby(a)tuffmail.com> --- Patch is against master, tested on 4.0. sound/pci/oxygen/oxygen_lib.c | 5 ++++- sound/pci/oxygen/oxygen_mixer.c | 9 +++++---- sound/pci/oxygen/oxygen_pcm.c | 15 +-------------- 3 files changed, 10 insertions(+), 19 deletions(-) diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index b4ef580..2cca908 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -373,7 +373,7 @@ static void oxygen_init(struct oxygen *chip) for (i = 0; i < 8; ++i) chip->dac_volume[i] = chip->model.dac_volume_min; chip->dac_mute = 1; - chip->spdif_playback_enable = 1; + chip->spdif_playback_enable = 0; chip->spdif_bits = OXYGEN_SPDIF_C | OXYGEN_SPDIF_ORIGINAL | (IEC958_AES1_CON_PCM_CODER << OXYGEN_SPDIF_CATEGORY_SHIFT); chip->spdif_pcm_bits = chip->spdif_bits; @@ -575,6 +575,9 @@ static void oxygen_shutdown(struct oxygen *chip) spin_lock_irq(&chip->reg_lock); chip->interrupt_mask = 0; chip->pcm_running = 0; + chip->spdif_playback_enable = 0; + oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL, + OXYGEN_SPDIF_OUT_ENABLE); oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); spin_unlock_irq(&chip->reg_lock); diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 6492bca..20a2781 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -252,7 +252,6 @@ void oxygen_update_spdif_source(struct oxygen *chip) old_control = oxygen_read32(chip, OXYGEN_SPDIF_CONTROL); old_routing = oxygen_read16(chip, OXYGEN_PLAY_ROUTING); if (chip->pcm_active & (1 << PCM_SPDIF)) { - new_control = old_control | OXYGEN_SPDIF_OUT_ENABLE; new_routing = (old_routing & ~OXYGEN_PLAY_SPDIF_MASK) | OXYGEN_PLAY_SPDIF_SPDIF; oxygen_rate = (old_control >> OXYGEN_SPDIF_OUT_RATE_SHIFT) @@ -265,13 +264,15 @@ void oxygen_update_spdif_source(struct oxygen *chip) oxygen_rate = oxygen_read16(chip, OXYGEN_I2S_MULTICH_FORMAT) & OXYGEN_I2S_RATE_MASK; new_control = (old_control & ~OXYGEN_SPDIF_OUT_RATE_MASK) | - (oxygen_rate << OXYGEN_SPDIF_OUT_RATE_SHIFT) | - OXYGEN_SPDIF_OUT_ENABLE; + (oxygen_rate << OXYGEN_SPDIF_OUT_RATE_SHIFT); } else { - new_control = old_control & ~OXYGEN_SPDIF_OUT_ENABLE; new_routing = old_routing; oxygen_rate = OXYGEN_RATE_44100; } + if (chip->spdif_playback_enable) + new_control = old_control | OXYGEN_SPDIF_OUT_ENABLE; + else + new_control = old_control & ~OXYGEN_SPDIF_OUT_ENABLE; if (old_routing != new_routing) { oxygen_write32(chip, OXYGEN_SPDIF_CONTROL, new_control & ~OXYGEN_SPDIF_OUT_ENABLE); diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index aa2ebd1..f4a4a48 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -477,8 +477,6 @@ static int oxygen_spdif_hw_params(struct snd_pcm_substream *substream, mutex_lock(&chip->mutex); spin_lock_irq(&chip->reg_lock); - oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL, - OXYGEN_SPDIF_OUT_ENABLE); oxygen_write8_masked(chip, OXYGEN_PLAY_FORMAT, oxygen_format(hw_params) << OXYGEN_SPDIF_FORMAT_SHIFT, OXYGEN_SPDIF_FORMAT_MASK); @@ -544,17 +542,6 @@ static int oxygen_hw_free(struct snd_pcm_substream *substream) return snd_pcm_lib_free_pages(substream); } -static int oxygen_spdif_hw_free(struct snd_pcm_substream *substream) -{ - struct oxygen *chip = snd_pcm_substream_chip(substream); - - spin_lock_irq(&chip->reg_lock); - oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL, - OXYGEN_SPDIF_OUT_ENABLE); - spin_unlock_irq(&chip->reg_lock); - return oxygen_hw_free(substream); -} - static int oxygen_prepare(struct snd_pcm_substream *substream) { struct oxygen *chip = snd_pcm_substream_chip(substream); @@ -669,7 +656,7 @@ static struct snd_pcm_ops oxygen_spdif_ops = { .close = oxygen_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = oxygen_spdif_hw_params, - .hw_free = oxygen_spdif_hw_free, + .hw_free = oxygen_hw_free, .prepare = oxygen_prepare, .trigger = oxygen_trigger, .pointer = oxygen_pointer, -- 2.3.5

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[alsa-devel] How to Configure Appropriate runtime->hw in ALSA Driver
by Azizul Hakim 05 Jul '15

05 Jul '15

What is the best way to set the *runtime->hw* parameter in ALSA driver. >From sample ALSA driver tutorial I understand we have to declare a *structure snd_pcm_hardware* where we define all sound parameters like *STEREO/MONO, RATES* etc. Now at run time we can play sound files of different parameters. How ALSA behaves if the audio file parameters doesn't match with the *structure_snd_pcm_hardware* parameters? So if we set MONO in our structure, but play a audio file with STEREO data, does ALSA automatically configure the data according to its structure or we're going to get some kind of disruption? If we want our driver to play any sound file, what should be the appropriate *structure snd_pcm_hardware *parameters? Thanks -Azizul Hakim

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[alsa-devel] [PATCH 1/6] ASoC: rt286: Prefix hexadecimal ID register value with 0x in error print
by Jarkko Nikula 03 Jul '15

03 Jul '15

Make it obvious that unexpected value read from ID register is printed in hexadecimal. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 5c43e263b2c1..2f39843ebd14 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1157,7 +1157,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, } if (val != RT286_VENDOR_ID && val != RT288_VENDOR_ID) { dev_err(&i2c->dev, - "Device with ID register %x is not rt286\n", val); + "Device with ID register %#x is not rt286\n", val); return -ENODEV; } -- 2.1.4

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[alsa-devel] [PATCH 0/6] Remove redundant spi driver bus initialization
by Antonio Borneo 03 Jul '15

03 Jul '15

This cleanup was already completed between end 2011 and early 2012 with a patch series from Lars-Peter Clausen: https://lkml.org/lkml/2011/11/24/190 Later on new redundant initialization re-appeared here and there. Time to cleanup again. And, yes, I'm lazy! I copy-paste the exact same commit message from Lars-Peter; only minor reformat to stay in 75 char/line and fix the name of spi_register_driver(). Regards, Antonio Antonio Borneo (6): ASoC: wm0010: Remove redundant spi driver bus initialization iio: ssp_sensors: Remove redundant spi driver bus initialization staging: mt29f_spinand: Remove redundant spi driver bus initialization net: ieee802154: Remove redundant spi driver bus initialization wireless: cw1200: Remove redundant spi driver bus initialization [media] s5c73m3: Remove redundant spi driver bus initialization drivers/iio/common/ssp_sensors/ssp_dev.c | 1 - drivers/media/i2c/s5c73m3/s5c73m3-spi.c | 1 - drivers/net/ieee802154/cc2520.c | 1 - drivers/net/ieee802154/mrf24j40.c | 1 - drivers/net/wireless/cw1200/cw1200_spi.c | 1 - drivers/staging/mt29f_spinand/mt29f_spinand.c | 1 - sound/soc/codecs/wm0010.c | 1 - 7 files changed, 7 deletions(-) -- 2.4.4

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Re: [alsa-devel] Intel Baytrail Linux Audio
by Koul, Vinod 03 Jul '15

03 Jul '15

1 0

Re: [alsa-devel] Intel Baytrail Linux Audio
by Koul, Vinod 03 Jul '15

03 Jul '15

On 3 Jul 2015 4:43 pm, Nicolas George <george(a)nsup.org> wrote: > > Le quintidi 15 messidor, an CCXXIII, Antonio Ospite a écrit : > > snd-soc-sst-bytcr-rt5640 will be loaded automatically if you use > > snd-intel-sst-acpi instead of snd-soc-sst-acpi, you can blacklist > > the latter using a file in /etc/modprobe.d (look up the details), or > > just remove snd-soc-sst-acpi.ko from the installation dir as a dirty > > hack. > > Indeed. I always loaded snd-soc-sst-acpi before snd-intel-sst-acpi, so I > could not observe it gives a different result. > > > In my tests I used the code from mainline linux. > > Ok. That is what I am doing too. > > > Note that different drivers expect different firmwares, you cannot mix > > them: > > > > snd-soc-sst-acpi -> fw_sst_0f28.bin-48kHz_i2s_master > > snd-intel-sst-acpi -> fw_sst_0f28.bin > > snd-intel-sst-acpi -> fw_sst_0f28_ssp0.bin (renamed to fw_sst_0f28.bin) > > Ok. I re-ran the tests like that. > > After loading snd-intel-sst-acpi, the kernel says: > > [ 35.160330] intel_sst_acpi 80860F28:00: LPE base: 0x90a00000 size:0x200000 > [ 35.160339] intel_sst_acpi 80860F28:00: IRAM base: 0x90ac0000 > [ 35.160372] intel_sst_acpi 80860F28:00: DRAM base: 0x90b00000 > [ 35.160384] intel_sst_acpi 80860F28:00: SHIM base: 0x90b40000 > [ 35.160395] intel_sst_acpi 80860F28:00: Mailbox base: 0x90b44000 > [ 35.160405] intel_sst_acpi 80860F28:00: DDR base: 0x20000000 > [ 35.161373] intel_sst_acpi 80860F28:00: Got drv data max stream 25 > > After loading snd-soc-sst-bytcr-rt5640 it says: > > [ 69.130424] bytt100_rt5640 bytt100_rt5640: snd-soc-dummy-dai <-> media-cpu-dai mapping ok > [ 69.130482] compress asoc: snd-soc-dummy-dai <-> compress-cpu-dai mapping ok > [ 69.130536] bytt100_rt5640 bytt100_rt5640: rt5640-aif1 <-> ssp2-port mapping ok > [ 69.130650] bytt100_rt5640 bytt100_rt5640: Connecting non-supply widget to supply widget is not supported (Int Mic -> LDO2) > [ 69.130657] bytt100_rt5640 bytt100_rt5640: ASoC: no dapm match for Int Mic --> (null) --> LDO2 > [ 69.130663] bytt100_rt5640 bytt100_rt5640: ASoC: Failed to add route Int Mic -> direct -> LDO2 > [ 69.165392] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 0 doesn't have DAPM widget!!! > [ 69.165427] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 1 doesn't have DAPM widget!!! > [ 69.165460] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 2 doesn't have DAPM widget!!! > [ 69.165494] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 3 doesn't have DAPM widget!!! > [ 69.165521] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_0 doesn't have DAPM widget!!! > [ 69.165551] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_1 doesn't have DAPM widget!!! > [ 69.165583] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_0 doesn't have DAPM widget!!! > [ 69.165618] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_1 doesn't have DAPM widget!!! > > The sound card appears, but then if I try to play something: > > aplay -D hw:0 -c 2 -r 48000 -f S16_LE /dev/urandom > aplay: set_params:1297: Unable to install hw params: > ACCESS: RW_INTERLEAVED > FORMAT: S16_LE > SUBFORMAT: STD > SAMPLE_BITS: 16 > FRAME_BITS: 32 > CHANNELS: 2 > RATE: 48000 > PERIOD_TIME: 125000 > PERIOD_SIZE: 6000 > PERIOD_BYTES: 24000 > PERIODS: 4 > BUFFER_TIME: 500000 > BUFFER_SIZE: 24000 > BUFFER_BYTES: 96000 > TICK_TIME: 0 > > [ 95.164749] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port This is dpcm enabled driver So please route the FE to BE and it should work... Vinod > > The messages are exactly the same with both firmwares (fw_sst_0f28.bin and > fw_sst_0f28_ssp0.bin). > > Maybe I forgot to load another module? > > Side note: snd-intel-sst-acpi and snd-soc-rt5640 both have usage count 1 > without dependent modules, that requires a reboot between each test. > > Thanks for your help, it helped me make progress even if it still is not > success. > > Regards, > > -- > Nicolas George

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