October 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: rsnd: dma: fix SSI9 4/5/6/7 busif dma address
by twischer＠de.adit-jv.com 22 Oct '19

22 Oct '19

From: Jiada Wang <jiada_wang(a)mentor.com> Currently each SSI unit 's busif dma address is calculated by following calculation formulation: 0xec540000 + 0x1000 * id + busif / 4 * 0xA000 + busif % 4 * 0x400 But according to user manual 41.1.4 Register Configuration ssi9 4/5/6/7 busif data register address (SSI9_4_BUSIF/SSI9_5_BUSIF/SSI9_6_BUSIF/SSI9_7_BUSIF) are out of this rule. This patch updates the calculation formulation to correct ssi9 4/5/6/7 busif data register address Fixes: commit 5e45a6fab3b9 ("ASoc: rsnd: dma: Calculate dma address with consider of BUSIF") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- sound/soc/sh/rcar/dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 0324a5c..28f65eb 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -508,10 +508,10 @@ static struct rsnd_mod_ops rsnd_dmapp_ops = { #define RDMA_SSI_I_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8) #define RDMA_SSI_O_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc) -#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400) - (0x4000 * ((i) / 9) * ((j) / 4))) #define RDMA_SSIU_O_N(addr, i, j) RDMA_SSIU_I_N(addr, i, j) -#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400) - (0x4000 * ((i) / 9) * ((j) / 4))) #define RDMA_SSIU_O_P(addr, i, j) RDMA_SSIU_I_P(addr, i, j) #define RDMA_SRC_I_N(addr, i) (addr ##_reg - 0x00500000 + (0x400 * i)) -- 2.7.4

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[alsa-devel] Recording on WM9712
by Matej Kupljen 22 Oct '19

22 Oct '19

Hi all, I have an iMX6q board with external AC97 WM9712 codec connected to SSI on iMX6. The card is detected correctly and it is using generic AC97 codec driver, not the specific one for WM9712. I was able to hard code the values into fsl-asoc-card.c and fsl_ssi.c files, but my problem still remains. I can playback WAV file normally, also speaker-test works as expected. The problem is when I try to record something from Line In. It does record the file, but when I playback it, it is "too slow". I tried different sample rates, but the problem is only worse (strange beeping in the recording). The command for recording is: # arecord -f dat -vv /tmp/record.wav (I tired also with -M option, but there is no difference). If I copy the recorded file to PC and use audacity to play it back using 1.5x speed, the recording seems O.K. Checking codec register values trough /proc/asound/ac97audio/codec97#0/ac97#0-0+regs it seems that the sample rates are set O.K. to 48kHz. Any idea what might be wrong? Is there anything else I can check? Thanks and BR, Matej

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[alsa-devel] [REGRESSION 5.3] WARNING: pulseaudio/1297 still has locks held!
by Russell King - ARM Linux admin 22 Oct '19

22 Oct '19

When upgrading from 5.2 to 5.3, I now encounter this error after boot: ==================================== WARNING: pulseaudio/1297 still has locks held! 5.3.0+ #1826 Not tainted ------------------------------------ 1 lock held by pulseaudio/1297: #0: ee815308 (&hcp->lock){....}, at: hdmi_codec_startup+0x20/0x130 stack backtrace: CPU: 0 PID: 1297 Comm: pulseaudio Not tainted 5.3.0+ #1826 Hardware name: Marvell Dove (Cubox) [<c0017b4c>] (unwind_backtrace) from [<c0014d10>] (show_stack+0x10/0x14) [<c0014d10>] (show_stack) from [<c00a2d18>] (futex_wait_queue_me+0x13c/0x19c) [<c00a2d18>] (futex_wait_queue_me) from [<c00a3630>] (futex_wait+0x184/0x24c) [<c00a3630>] (futex_wait) from [<c00a5e1c>] (do_futex+0x334/0x598) [<c00a5e1c>] (do_futex) from [<c00a62e8>] (sys_futex_time32+0x118/0x180) [<c00a62e8>] (sys_futex_time32) from [<c0009000>] (ret_fast_syscall+0x0/0x54) Exception stack(0xebd31fa8 to 0xebd31ff0) 1fa0: 00000000 ffffffff 000c8748 00000189 00000001 00000000 1fc0: 00000000 ffffffff 00000000 000000f0 00000000 00000000 00000000 00056200 1fe0: 000000f0 beac03a8 b6d6c835 b6d6f456 This seems to be caused by: eb1ecadb7f67 ("ASoC: hdmi-codec: re-introduce mutex locking") which holds a mutex and returns to userspace. Meanwhile, userspace issues a futex operation, which ends up in futex_wait_queue_me(). futex_wait_queue_me() can wait, and if the system will be hibernating, is a potential hibernation point. One of the requirements there is that no locks shall be held, and the freezer code verifies that by calling debug_check_no_locks_held(). With lockdep enabled, it reveals that hdmi-codec is indeed holding a lock. Reverting this commit solves the problem. -- RMK's Patch system: https://www.armlinux.org.uk/developer/patches/ FTTC broadband for 0.8mile line in suburbia: sync at 12.1Mbps down 622kbps up According to speedtest.net: 11.9Mbps down 500kbps up

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[alsa-devel] [PATCH v6 0/9] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 22 Oct '19

22 Oct '19

Hey, here's the 6th round for this series that adapts SOF to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is to unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. v6 changes: - Since v5, changes in patches 002, 003 and 005. - Fixed a bug with S3 suspend/resume during HDMI playback on Intel ICL platforms. As part of the bug fix, cleaned up the new display power control code added in v5. Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) Kai Vehmanen (9): ALSA: hda/hdmi - implement mst_no_extra_pcms flag ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: Intel: skl-hda-dsp-generic: fix include guard name ASoC: SOF: Intel: add support for snd-hda-codec-hdmi ASoC: Intel: bxt-da7219-max98357a: common hdmi codec support ASoC: Intel: glk_rt5682_max98357a: common hdmi codec support ASoC: intel: sof_rt5682: common hdmi codec support ASoC: Intel: bxt_rt298: common hdmi codec support include/sound/hda_codec.h | 1 + include/sound/soc-acpi.h | 2 + sound/pci/hda/patch_hdmi.c | 19 ++- sound/soc/codecs/hdac_hda.c | 114 ++++++++++++++++-- sound/soc/codecs/hdac_hda.h | 13 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 11 ++ sound/soc/intel/boards/bxt_rt298.c | 11 ++ sound/soc/intel/boards/glk_rt5682_max98357a.c | 11 ++ sound/soc/intel/boards/hda_dsp_common.h | 97 +++++++++++++++ sound/soc/intel/boards/skl_hda_dsp_common.c | 10 +- sound/soc/intel/boards/skl_hda_dsp_common.h | 27 ++++- sound/soc/intel/boards/skl_hda_dsp_generic.c | 1 + sound/soc/intel/boards/sof_rt5682.c | 11 ++ sound/soc/sof/intel/Kconfig | 10 ++ sound/soc/sof/intel/hda-codec.c | 22 +++- sound/soc/sof/intel/hda.c | 6 + sound/soc/sof/intel/hda.h | 6 +- 17 files changed, 345 insertions(+), 27 deletions(-) create mode 100644 sound/soc/intel/boards/hda_dsp_common.h -- 2.17.1

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[alsa-devel] sof-skl_hda: add device definition to speaker disable sequence
by GitHub pull_request - opened 22 Oct '19

22 Oct '19

alsa-project/alsa-ucm-conf pull request #3 was opened from juimonen: For example Pulseaudio is not able to run the disable sequence if the device is not defined. So add the definition. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/3 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/3.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[alsa-devel] [asoc:for-5.5 122/122] sound/soc/codecs/tas2562.c:195:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer
by kbuild test robot 22 Oct '19

22 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: c173dba44c2d2ec2adaa4e607e9729b74236bd57 commit: c173dba44c2d2ec2adaa4e607e9729b74236bd57 [122/122] ASoC: tas2562: Introduce the TAS2562 amplifier reproduce: # apt-get install sparse # sparse version: v0.6.1-rc1-42-g38eda53-dirty git checkout c173dba44c2d2ec2adaa4e607e9729b74236bd57 make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/tas2562.c:195:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/soc/codecs/tas2562.c:202:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/soc/codecs/tas2562.c:209:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer >> sound/soc/codecs/tas2562.c:246:58: sparse: sparse: incorrect type in argument 2 (different base types) @@ expected int bitwidth @@ got restricted snd_pcm_formint bitwidth @@ >> sound/soc/codecs/tas2562.c:246:58: sparse: expected int bitwidth >> sound/soc/codecs/tas2562.c:246:58: sparse: got restricted snd_pcm_format_t sound/soc/codecs/tas2562.c:472:39: sparse: sparse: restricted snd_pcm_format_t degrades to integer vim +195 sound/soc/codecs/tas2562.c 189 190 static int tas2562_set_bitwidth(struct tas2562_data *tas2562, int bitwidth) 191 { 192 int ret; 193 194 switch (bitwidth) { > 195 case SNDRV_PCM_FORMAT_S16_LE: 196 snd_soc_component_update_bits(tas2562->component, 197 TAS2562_TDM_CFG2, 198 TAS2562_TDM_CFG2_RXWLEN_MASK, 199 TAS2562_TDM_CFG2_RXWLEN_16B); 200 tas2562->v_sense_slot = tas2562->i_sense_slot + 2; 201 break; 202 case SNDRV_PCM_FORMAT_S24_LE: 203 snd_soc_component_update_bits(tas2562->component, 204 TAS2562_TDM_CFG2, 205 TAS2562_TDM_CFG2_RXWLEN_MASK, 206 TAS2562_TDM_CFG2_RXWLEN_24B); 207 tas2562->v_sense_slot = tas2562->i_sense_slot + 4; 208 break; 209 case SNDRV_PCM_FORMAT_S32_LE: 210 snd_soc_component_update_bits(tas2562->component, 211 TAS2562_TDM_CFG2, 212 TAS2562_TDM_CFG2_RXWLEN_MASK, 213 TAS2562_TDM_CFG2_RXWLEN_32B); 214 tas2562->v_sense_slot = tas2562->i_sense_slot + 4; 215 break; 216 217 default: 218 dev_info(tas2562->dev, "Not supported params format\n"); 219 } 220 221 ret = snd_soc_component_update_bits(tas2562->component, 222 TAS2562_TDM_CFG5, 223 TAS2562_TDM_CFG5_VSNS_EN | TAS2562_TDM_CFG5_VSNS_SLOT_MASK, 224 TAS2562_TDM_CFG5_VSNS_EN | tas2562->v_sense_slot); 225 if (ret < 0) 226 return ret; 227 228 ret = snd_soc_component_update_bits(tas2562->component, 229 TAS2562_TDM_CFG6, 230 TAS2562_TDM_CFG6_ISNS_EN | TAS2562_TDM_CFG6_ISNS_SLOT_MASK, 231 TAS2562_TDM_CFG6_ISNS_EN | tas2562->i_sense_slot); 232 if (ret < 0) 233 return ret; 234 235 return 0; 236 } 237 238 static int tas2562_hw_params(struct snd_pcm_substream *substream, 239 struct snd_pcm_hw_params *params, 240 struct snd_soc_dai *dai) 241 { 242 struct snd_soc_component *component = dai->component; 243 struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); 244 int ret; 245 > 246 ret = tas2562_set_bitwidth(tas2562, params_format(params)); 247 if (ret) { 248 dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); 249 return ret; 250 } 251 252 ret = tas2562_set_samplerate(tas2562, params_rate(params)); 253 if (ret) 254 dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); 255 256 return ret; 257 } 258 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ASoC: compress: fix unsigned integer overflow check
by Srinivas Kandagatla 21 Oct '19

21 Oct '19

From: Xiaojun Sang <xsang(a)codeaurora.org> Parameter fragments and fragment_size are type of u32. U32_MAX is the correct check. Signed-off-by: Xiaojun Sang <xsang(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/core/compress_offload.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 41905afada63..f34ce564d92c 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -528,7 +528,7 @@ static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ if (params->buffer.fragment_size == 0 || - params->buffer.fragments > INT_MAX / params->buffer.fragment_size || + params->buffer.fragments > U32_MAX / params->buffer.fragment_size || params->buffer.fragments == 0) return -EINVAL; -- 2.21.0

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[alsa-devel] [PATCH 1/2] ASoC: msm8916-wcd-analog: Fix RX1 selection in RDAC2 MUX
by Stephan Gerhold 21 Oct '19

21 Oct '19

According to the PM8916 Hardware Register Description, CDC_D_CDC_CONN_HPHR_DAC_CTL has only a single bit (RX_SEL) to switch between RX1 (0) and RX2 (1). It is not possible to disable it entirely to achieve the "ZERO" state. However, at the moment the "RDAC2 MUX" mixer defines three possible values ("ZERO", "RX2" and "RX1"). Setting the mixer to "ZERO" actually configures it to RX1. Setting the mixer to "RX1" has (seemingly) no effect. Remove "ZERO" and replace it with "RX1" to fix this. Fixes: 585e881e5b9e ("ASoC: codecs: Add msm8916-wcd analog codec") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/codecs/msm8916-wcd-analog.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 667e9f73aba3..e3d311fb510e 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -306,7 +306,7 @@ struct pm8916_wcd_analog_priv { }; static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" }; -static const char *const rdac2_mux_text[] = { "ZERO", "RX2", "RX1" }; +static const char *const rdac2_mux_text[] = { "RX1", "RX2" }; static const char *const hph_text[] = { "ZERO", "Switch", }; static const struct soc_enum hph_enum = SOC_ENUM_SINGLE_VIRT( @@ -321,7 +321,7 @@ static const struct soc_enum adc2_enum = SOC_ENUM_SINGLE_VIRT( /* RDAC2 MUX */ static const struct soc_enum rdac2_mux_enum = SOC_ENUM_SINGLE( - CDC_D_CDC_CONN_HPHR_DAC_CTL, 0, 3, rdac2_mux_text); + CDC_D_CDC_CONN_HPHR_DAC_CTL, 0, 2, rdac2_mux_text); static const struct snd_kcontrol_new spkr_switch[] = { SOC_DAPM_SINGLE("Switch", CDC_A_SPKR_DAC_CTL, 7, 1, 0) -- 2.23.0

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[alsa-devel] [PATCH v2] ASoC: samsung: i2s: Document clocks macros
by Marek Szyprowski 21 Oct '19

21 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Document clocks macros with their description from 'Documentation/devicetree/bindings/sound/samsung-i2s.txt' Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- v2: - Added proper commit message and description - Moved comments to the right side Best regards, Maciej Falkowski --- include/dt-bindings/sound/samsung-i2s.h | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/include/dt-bindings/sound/samsung-i2s.h b/include/dt-bindings/sound/samsung-i2s.h index 77545f14c379..250de0d6c734 100644 --- a/include/dt-bindings/sound/samsung-i2s.h +++ b/include/dt-bindings/sound/samsung-i2s.h @@ -2,8 +2,14 @@ #ifndef _DT_BINDINGS_SAMSUNG_I2S_H #define _DT_BINDINGS_SAMSUNG_I2S_H -#define CLK_I2S_CDCLK 0 -#define CLK_I2S_RCLK_SRC 1 -#define CLK_I2S_RCLK_PSR 2 +#define CLK_I2S_CDCLK 0 /* the CDCLK (CODECLKO) gate clock */ + +#define CLK_I2S_RCLK_SRC 1 /* the RCLKSRC mux clock (corresponding to + * RCLKSRC bit in IISMOD register) + */ + +#define CLK_I2S_RCLK_PSR 2 /* the RCLK prescaler divider clock + * (corresponding to the IISPSR register) + */ #endif /* _DT_BINDINGS_SAMSUNG_I2S_H */ -- 2.17.1

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[alsa-devel] [PATCH v5 0/4] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 21 Oct '19

21 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). ASoC: cros_ec_codec: support WoV - makes cros_ec_codec support WoV. ASoC: mediatek: mt6358: support WoV - sets necessary registers on mt6358 to support WoV. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV - make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. Changes from v4: - remove the first 6 commits which have merged - fix a compile error by adding Kconfig dependency https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156992.h… Changes from v3: - fix a compile error by adding ifdef https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156750.h… - fix a coccinelle warning https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156754.h… Changes from v2: - rebase upon to "don't use snd_pcm_ops" series https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156170.h… - fix sparse errors https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156328.h… - use "reg" for SHM binding https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156657.h… Changes from v1: - fix a compile error and make kbuild bot happy https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156315.h… Tzung-Bi Shih (4): ASoC: cros_ec_codec: support WoV ASoC: mediatek: mt6358: support WoV ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 1 + .../linux/platform_data/cros_ec_commands.h | 69 ++ sound/soc/codecs/Kconfig | 2 + sound/soc/codecs/cros_ec_codec.c | 700 +++++++++++++++++- sound/soc/codecs/mt6358.c | 105 +++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 946 insertions(+), 5 deletions(-) -- 2.23.0.866.gb869b98d4c-goog

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