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[alsa-devel] [PATCH] alsa-lib: pcm: segmentation fault snd_pcm_open
by Markus Seeber 08 Dec '17

08 Dec '17

Previously, calling snd_pcm_open with a null pointer argument for "name" caused a segmentation fault in snd_config_search_definition. The segmentation fault resulted from invoking undefined behavior by calling strchr with a null pointer argument. The issue can be reproduced by passing a null pointer for "name" to snd_pcm_open. To prevent this instance of strchr to be invoked with a null pointer argument, snd_config_search_definition now returns EINVAL if argument "name" is a null pointer. Signed-off-by: Markus Seeber <markus.seeber(a)spectralbird.de> --- src/conf.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/src/conf.c b/src/conf.c index e71bb972..e04b575b 100644 --- a/src/conf.c +++ b/src/conf.c @@ -5078,6 +5078,8 @@ int snd_config_search_definition(snd_config_t *config, { snd_config_t *conf; char *key; + if(name == NULL) + return -EINVAL; const char *args = strchr(name, ':'); int err; if (args) { -- 2.15.1

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[alsa-devel] [PATCH] ASoC: atmel-classd: select correct Kconfig symbol
by Arnd Bergmann 08 Dec '17

08 Dec '17

Commit 061981ff8cc8 ("ASoC: atmel: properly select dma driver state") changed the way that the dependencies are handled, but then the Class D amplifier support got merged, which used the old method. This seems to have triggered a very rare randconfig condition for me now, leading to a link error: sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': atmel_ssc_dai.c:(.text+0x79c): undefined reference to `atmel_pcm_dma_platform_register' atmel_ssc_dai.c:(.text+0x79c): relocation truncated to fit: R_AARCH64_CALL26 against undefined symbol `atmel_pcm_dma_platform_register' sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_put_audio': atmel_ssc_dai.c:(.text+0xf24): undefined reference to `atmel_pcm_dma_platform_unregister' atmel_ssc_dai.c:(.text+0xf24): relocation truncated to fit: R_AARCH64_CALL26 against undefined symbol `atmel_pcm_dma_platform_unregister' Changing it to select SND_ATMEL_SOC_SSC_DMA as intended rather than SND_ATMEL_SOC_DMA directly makes it work again. Fixes: e0a25b6d1862 ("ASoC: atmel-classd: add the Audio Class D Amplifier") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/atmel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 4a56f3dfba51..2d998e5c4fb5 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -64,7 +64,7 @@ config SND_AT91_SOC_SAM9X5_WM8731 config SND_ATMEL_SOC_CLASSD tristate "Atmel ASoC driver for boards using CLASSD" depends on ARCH_AT91 || COMPILE_TEST - select SND_ATMEL_SOC_DMA + select SND_ATMEL_SOC_SSC_DMA select REGMAP_MMIO help Say Y if you want to add support for Atmel ASoC driver for boards using -- 2.9.0

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Re: [alsa-devel] [Intel-gfx] [PATCH] drm/i915: Remove unused IRQ chip data of HDMI LPE audio
by Ville Syrjälä 08 Dec '17

08 Dec '17

On Fri, Dec 08, 2017 at 08:34:16AM +0000, Chen, Augustine wrote: > Dear Sirs, > Please kindly help reviewing this patch for the Bugzilla below. > https://bugs.freedesktop.org/show_bug.cgi?id=103731 Resend the patch using git send-email please. -- Ville Syrjälä Intel OTC

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[alsa-devel] [PATCH v2 0/2] fix race condition in rsnd_ssi_pointer_update
by jiada_wang＠mentor.com 08 Dec '17

08 Dec '17

From: Jiada Wang <jiada_wang(a)mentor.com> This patch set aims to fix the race condition in rsnd_ssi_pointer_update, between set of byte_pos and wrap it around when new buffer starts. Jiada Wang (2): ASoC: rsnd: ssi: fix race condition in rsnd_ssi_pointer_update ASoC: rsnd: ssi: remove unnesessary period_pos Changes for v2: * Updated commit message sound/soc/sh/rcar/ssi.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - Add vendor id for Cannonlake HDMI codec
by Guneshwor Singh 08 Dec '17

08 Dec '17

Cannonlake HDMI codec has the same nid as Geminilake. This adds the codec entry for it. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index c19c81d230bd..b4f1b6e88305 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -55,10 +55,11 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_kabylake(codec) ((codec)->core.vendor_id == 0x8086280b) #define is_geminilake(codec) (((codec)->core.vendor_id == 0x8086280d) || \ ((codec)->core.vendor_id == 0x80862800)) +#define is_cannonlake(codec) ((codec)->core.vendor_id == 0x8086280c) #define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec) \ || is_skylake(codec) || is_broxton(codec) \ - || is_kabylake(codec)) || is_geminilake(codec) - + || is_kabylake(codec)) || is_geminilake(codec) \ + || is_cannonlake(codec) #define is_valleyview(codec) ((codec)->core.vendor_id == 0x80862882) #define is_cherryview(codec) ((codec)->core.vendor_id == 0x80862883) #define is_valleyview_plus(codec) (is_valleyview(codec) || is_cherryview(codec)) @@ -3841,6 +3842,7 @@ HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x80862800, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), -- 2.15.1

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[alsa-devel] [PATCH v2] ASoC: Intel: Change kern log level to avoid unwanted messages
by Naveen Manohar 08 Dec '17

08 Dec '17

patch suppresses the warning message "control load not supported" as this is a debug information to help debug issues in topology. Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Acked-By: Vinod Koul <vinod.koul(a)intel.com> --- v1->v2: Added Intel sub-system sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index a072bcf..81923da 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2908,7 +2908,7 @@ static int skl_tplg_control_load(struct snd_soc_component *cmpnt, break; default: - dev_warn(bus->dev, "Control load not supported %d:%d:%d\n", + dev_dbg(bus->dev, "Control load not supported %d:%d:%d\n", hdr->ops.get, hdr->ops.put, hdr->ops.info); break; } -- 1.9.1

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[alsa-devel] [PATCH][next] ASoC: pcm186x: make pcm186x_dai_ops and pcm186x_adc_input_channel_sel_value static
by Colin King 07 Dec '17

07 Dec '17

From: Colin Ian King <colin.king(a)canonical.com> pcm186x_dai_ops and pcm186x_adc_input_channel_sel_value are local to the source and do not need to be in global scope, so make them static. Cleans up sparse warnings: symbol 'pcm186x_dai_ops' was not declared. Should it be static? symbol 'pcm186x_adc_input_channel_sel_value' was not declared. Should it be static? Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/pcm186x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index f7aa56e20169..cdb51427facc 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -59,7 +59,7 @@ static const struct snd_kcontrol_new pcm1865_snd_controls[] = { pcm186x_pga_tlv), }; -const unsigned int pcm186x_adc_input_channel_sel_value[] = { +static const unsigned int pcm186x_adc_input_channel_sel_value[] = { 0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x20, 0x30 @@ -475,7 +475,7 @@ static int pcm186x_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, return 0; } -const struct snd_soc_dai_ops pcm186x_dai_ops = { +static const struct snd_soc_dai_ops pcm186x_dai_ops = { .set_sysclk = pcm186x_set_dai_sysclk, .set_tdm_slot = pcm186x_set_tdm_slot, .set_fmt = pcm186x_set_fmt, -- 2.14.1

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[alsa-devel] [PATCH v3 00/10] Add Headphone Detection to TLV320AIC31xx Driver
by Andrew F. Davis 07 Dec '17

07 Dec '17

Hello all, This series has the end goal of adding headphone detection to the tlv320aic31xx driver. The first few patches are mostly cleanups. Then a couple bug fixes I noticed. Followed by adding interrupt handling and finally headphone detection. Still not sure if the IRQ patch is correct, I think we have no good way to share this IRQ line, so leaving it the way it is for now. Thanks, Andrew Changes from v2: - Rebased on -next - Added delta patch for GPIO header fix - Split platform data remove into fwnode and pdata removal - Rework reset patch to print error code - Use snd_soc_codec_set_jack to set jack Changes from v1: - Splitup the cleanup patch a bit more - Move the GPIO1 register fix patch before header cleanup so it can be taken back into stable - Added Acked-by - New patch dealing with regulator notifications - Various small touchups - Rebased on v4.15-rc1 Andrew F. Davis (10): ASoC: tlv320aic31xx: Fix GPIO header includes ASoC: tlv320aic31xx: Use fwnode APIs over raw OF calls ASoC: tlv320aic31xx: Remove platform data ASoC: tlv320aic31xx: Add MICBIAS off setting ASoC: tlv320aic31xx: Remove regulator notification handling ASoC: tlv320aic31xx: Reset registers during power up ASoC: tlv320aic31xx: Add short circuit detection support ASoC: tlv320aic31xx: Add overflow detection support ASoC: tlv320aic31xx: Add headphone/headset detection ASoC: tlv320aic31xx: Add button press detection .../devicetree/bindings/sound/tlv320aic31xx.txt | 1 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 1 + sound/soc/codecs/tlv320aic31xx.c | 263 +++++++++++++-------- sound/soc/codecs/tlv320aic31xx.h | 41 +++- 4 files changed, 196 insertions(+), 110 deletions(-) -- 2.15.0

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[alsa-devel] [PATCH v2 00/19] Add Headphone Detection to TLV320AIC31xx Driver
by Andrew F. Davis 07 Dec '17

07 Dec '17

Hello all, This series has the end goal of adding headphone detection to the tlv320aic31xx driver. The first few patches are mostly cleanups. Then a couple bug fixes I noticed. Followed by adding interrupt handling and finally headphone detection. This series (or at least patch #8) depend on this DT fix[0]. Thanks, Andrew Changes from v1: - Splitup the cleanup patch a bit more - Move the GPIO1 register fix patch before header cleanup so it can be taken back into stable - Added Acked-by - New patch dealing with regulator notifications - Various small touchups - Rebased on v4.15-rc1 + [0] [0] http://www.spinics.net/lists/kernel/msg2660968.html Andrew F. Davis (19): ASoC: tlv320aic31xx: File header and copyright cleanup ASoC: tlv320aic31xx: Change aic31xx_power_off return type to void ASoC: tlv320aic31xx: Move ACPI table next to OF table ASoC: tlv320aic31xx: General source formatting cleanup ASoC: tlv320aic31xx: Fix GPIO1 register definition ASoC: tlv320aic31xx: Reformat header file using GENMASK and BIT macros ASoC: tlv320aic31xx: Merge init function into probe ASoC: tlv320aic31xx: Switch GPIO handling to use gpiod_* API ASoC: tlv320aic31xx: Remove platform data ASoC: tlv320aic31xx: Add MICBIAS off setting ASoC: tlv320aic31xx: Check clock and divider before division ASoC: tlv320aic31xx: Add CODEC clock slave support ASoC: tlv320aic31xx: Fix inverted BCLK handling ASoC: tlv320aic31xx: Remove regulator notification handling ASoC: tlv320aic31xx: Reset registers during power up ASoC: tlv320aic31xx: Add short circuit detection support ASoC: tlv320aic31xx: Add overflow detection support ASoC: tlv320aic31xx: Add headphone/headset detection ASoC: tlv320aic31xx: Add button press detection .../devicetree/bindings/sound/tlv320aic31xx.txt | 1 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 1 + sound/soc/codecs/tlv320aic31xx.c | 422 ++++++++++-------- sound/soc/codecs/tlv320aic31xx.h | 496 ++++++++++----------- 4 files changed, 471 insertions(+), 449 deletions(-) rewrite sound/soc/codecs/tlv320aic31xx.h (83%) -- 2.15.0

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[alsa-devel] [RFC 00/10] Enable HDA Codec support on Intel Platforms (Series2)
by Rakesh Ughreja 07 Dec '17

07 Dec '17

Many Intel platforms (SKL,KBL) etc. in the market supports enahanced audio capabilities which also includes DSP processing. This patch carry forwads the works that is done in the previous series to enable HD Audio codecs on such platforms. This patch series adds ASoC HDA codec driver for Intel platforms. It is written by reusing the legacy HDA ALSA codec driver. Intention is to maximize the reuse and minimize the changes in the legacy HDA codec driver. I would like to receive feedback before proceeding further on this direction. TODO: - This series is tested on KBL based product (Dell XPS 13). - Basic playback is working with headset and speakers. - Capture operation is not tested. - More platforms and use cases coverage can be added once we have basic agreement in terms of the overall approach. FIXME: - KConfig changes does not look right, but I could not think of any proper way without making changes into legacy HDA codec driver. So need some help on this topic. Rakesh Ughreja (10): ASoC: Intel: Boards: Machine driver for Intel platforms ASoC: Intel: Skylake: Add entry in sst_acpi_mach for HDA codecs ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ALSA: hda - make some of the functions externally visible ASoC: hdac_hda: add ASoC based HDA codec driver ALSA: hda: add new API snd_hda_asoc_codec_new for ASoC codec drivers ASoC: hdac_hda: add DAI, widgets and related ops ASoC: hdac_hda: add runtime PM support ASoC: Intel: Boards: add support for HDA codecs include/sound/hdaudio_ext.h | 3 +- sound/hda/ext/hdac_ext_bus.c | 9 +- sound/pci/hda/Kconfig | 11 + sound/pci/hda/hda_codec.c | 109 +++++- sound/pci/hda/hda_codec.h | 32 +- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 633 +++++++++++++++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 16 + sound/soc/intel/boards/Kconfig | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_generic.c | 386 +++++++++++++++++++ sound/soc/intel/skylake/skl-pcm.c | 32 +- sound/soc/intel/skylake/skl.c | 79 +++- sound/soc/intel/skylake/skl.h | 10 +- 15 files changed, 1315 insertions(+), 26 deletions(-) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/intel/boards/skl_hda_generic.c -- 2.7.4

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[alsa-devel] [PATCH] alsa-utils: aplay: Adjust sample rate limits to support newer hardware
by Jussi Laako 07 Dec '17

07 Dec '17

There are number of devices that support up to 384 kHz sampling rate and some devices up to 768 kHz sampling rate. This patch increases sanity check limit to 768k in order to support testing of such hardware. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- aplay/aplay.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index dbae17c..6b740c2 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -635,10 +635,10 @@ int main(int argc, char *argv[]) error(_("invalid rate argument '%s'"), optarg); return 1; } - if (tmp < 300) + if (tmp < 1000) tmp *= 1000; rhwparams.rate = tmp; - if (tmp < 2000 || tmp > 192000) { + if (tmp < 2000 || tmp > 768000) { error(_("bad speed value %i"), tmp); return 1; } -- 2.7.4

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[alsa-devel] [PATCH] ALSA: usb-audio: Add native DSD support for Esoteric D-05X
by Jussi Laako 07 Dec '17

07 Dec '17

Adds VID:PID of Esoteric D-05X to the TEAC device id's. Renames the is_teac_50X_dac() function to is_teac_dsd_dac() to cover broader device family from the same corporation sharing the same USB audio implementation. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/quirks.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 77eecaa..a66ef57 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1166,10 +1166,11 @@ static bool is_marantz_denon_dac(unsigned int id) /* TEAC UD-501/UD-503/NT-503 USB DACs need a vendor cmd to switch * between PCM/DOP and native DSD mode */ -static bool is_teac_50X_dac(unsigned int id) +static bool is_teac_dsd_dac(unsigned int id) { switch (id) { case USB_ID(0x0644, 0x8043): /* TEAC UD-501/UD-503/NT-503 */ + case USB_ID(0x0644, 0x8044): /* Esoteric D-05X */ return true; } return false; @@ -1202,7 +1203,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, break; } mdelay(20); - } else if (is_teac_50X_dac(subs->stream->chip->usb_id)) { + } else if (is_teac_dsd_dac(subs->stream->chip->usb_id)) { /* Vendor mode switch cmd is required. */ switch (fmt->altsetting) { case 3: /* DSD mode (DSD_U32) requested */ @@ -1392,7 +1393,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, } /* TEAC devices with USB DAC functionality */ - if (is_teac_50X_dac(chip->usb_id)) { + if (is_teac_dsd_dac(chip->usb_id)) { if (fp->altsetting == 3) return SNDRV_PCM_FMTBIT_DSD_U32_BE; } -- 2.7.4

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[alsa-devel] [PATCH v2 0/6] ASoC: Intel: Skylake: Skylake driver updates
by Guneshwor Singh 07 Dec '17

07 Dec '17

This series includes fixes for HDMI widgets sysfs not updating, shim read optimization, fixes for dmic dev_type check and feature to configure DSP clock source. Guneshwor Singh (1): ASoC: Intel: Skylake: Do not check dev_type for dmic link type Pradeep Tewani (2): ASoC: Intel: Skylake: Parse vendor tokens to build A-State table ASoC: Intel: Skylake: Configure DSP clock source Puneeth Prabhu (2): ASoC: hdac_hdmi: Refresh sysfs during hdmi device probe ASoC: hdac_hdmi: Remove redundant assignments Subhransu S. Prusty (1): ASoC: Intel: Skylake: Remove second shim read in register_poll include/uapi/sound/snd_sst_tokens.h | 17 ++++++++++++- sound/soc/codecs/hdac_hdmi.c | 8 ++----- sound/soc/intel/common/sst-dsp.c | 4 +--- sound/soc/intel/skylake/skl-messages.c | 18 ++++++++++++++ sound/soc/intel/skylake/skl-nhlt.c | 15 ++++++++---- sound/soc/intel/skylake/skl-pcm.c | 6 +++++ sound/soc/intel/skylake/skl-sst-dsp.h | 3 +++ sound/soc/intel/skylake/skl-topology.c | 44 +++++++++++++++++++++++++++++++++- sound/soc/intel/skylake/skl.h | 17 +++++++++++++ 9 files changed, 116 insertions(+), 16 deletions(-) -- 2.15.1

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[alsa-devel] Multiple codecs on the same DAI
by Ricard Wanderlof 07 Dec '17

07 Dec '17

Hi, I'm studying a hardware implementation where an S/PDIF receiver is on the same I2S connection as codec. Essentially, the S/PDIF receiver echos the I2S capture data from the codec unless there is a digital input signal which it can lock on to. For playback, the I2S output from the CPU DAI goes directly to the codec. I'm a bit at a loss for how to model this in ALSA. I know it's possible to have multiple codecs on a card, but those cases seem to be either separate codecs for the left and right channels, or separate codecs for capture and playback. In my case the playback codec is always the same, but the capture codec can switch between the two avilable input methods (more or less automatically if the S/PDIF receiver is so configured). >From a user perspective, it's not important that ALSA knows where the data is actually coming from, so if I could just register the two "codecs" on the same dai_link at the same time it would be fine, but I'm just imagining that I'll run into problems when ALSA tries to set up streams with two different codecs being involved. (Of course, at some level, the user might be interested to know the signal routing, but that can be determined by reading appropriate status bits in the S/PDIF receiver for instance using amixer controls). Do you have any ideas about how to configure this in ALSA? /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH v4] ASoC: ak4613: call dummy write for PW_MGMT1/3 when Playback
by Kuninori Morimoto 07 Dec '17

07 Dec '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Power Down Release Command (PMVR, PMDAC, RSTN, PMDA1-PMDA6) which are located on PW_MGMT1 / PW_MGMT3 register must be write again after at least 5 LRCK cycle or later on each command. Otherwise, Playback volume will be 0dB. Basically, it should be 1. PowerDownRelease by Power Management1 <= call 1.x after 5LRCK 1.x Dummy write to Power Management1 2. PowerDownRelease by Power Management3 <= call 2.x after 5LRCK 2.x Dummy write to Power Management3 To avoid too many dummy write, this patch is merging these. 1. PowerDownRelease by Power Management1 2. PowerDownRelease by Power Management3 <= call after 5LRCK 2.x Dummy write to Power Management1/3 <= merge dummy write This patch adds dummy write when Playback Start timing. Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v3 -> v4 - use schedule_work() instead of schedule_delayed_work() sound/soc/codecs/ak4613.c | 60 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 60 insertions(+) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index ee9e822..8d2c90b 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -15,6 +15,7 @@ */ #include <linux/clk.h> +#include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -95,6 +96,9 @@ struct ak4613_priv { struct mutex lock; const struct ak4613_interface *iface; struct snd_pcm_hw_constraint_list constraint; + struct work_struct dummy_write_work; + struct snd_soc_component *component; + unsigned int rate; unsigned int sysclk; unsigned int fmt; @@ -392,6 +396,7 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } + priv->rate = rate; /* * FIXME @@ -467,11 +472,65 @@ static int ak4613_set_bias_level(struct snd_soc_component *component, return 0; } +static void ak4613_dummy_write(struct work_struct *work) +{ + struct ak4613_priv *priv = container_of(work, + struct ak4613_priv, + dummy_write_work); + struct snd_soc_component *component = priv->component; + unsigned int mgmt1; + unsigned int mgmt3; + + /* wait 5 LR clocks */ + udelay(5000000 / priv->rate); + + snd_soc_component_read(component, PW_MGMT1, &mgmt1); + snd_soc_component_read(component, PW_MGMT3, &mgmt3); + + snd_soc_component_write(component, PW_MGMT1, mgmt1); + snd_soc_component_write(component, PW_MGMT3, mgmt3); +} + +static int ak4613_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + /* + * FIXME + * + * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks + * from Power Down Release. Otherwise, Playback volume will be 0dB. + * To avoid complex multiple delay/dummy_write method from + * ak4613_set_bias_level() / SND_SOC_DAPM_DAC_E("DACx", ...), + * call it once here. + * + * But, unfortunately, we can't "write" here because here is atomic + * context (It uses I2C access for write). + * Thus, use delayed work with 0 delay to switch to normal context + * immediately, and wait 5 LR clocks and do dummy_write there. + */ + + if ((cmd != SNDRV_PCM_TRIGGER_START) && + (cmd != SNDRV_PCM_TRIGGER_RESUME)) + return 0; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + + priv->component = &codec->component; + schedule_work(&priv->dummy_write_work); + + return 0; +} + static const struct snd_soc_dai_ops ak4613_dai_ops = { .startup = ak4613_dai_startup, .shutdown = ak4613_dai_shutdown, .set_sysclk = ak4613_dai_set_sysclk, .set_fmt = ak4613_dai_set_fmt, + .trigger = ak4613_dai_trigger, .hw_params = ak4613_dai_hw_params, }; @@ -591,6 +650,7 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, priv->iface = NULL; priv->cnt = 0; priv->sysclk = 0; + INIT_WORK(&priv->dummy_write_work, ak4613_dummy_write); mutex_init(&priv->lock); -- 1.9.1

2 4

[alsa-devel] [PATCH V2] ARM: dts: bcm283x: Fix probing of bcm2835-i2s
by Stefan Wahren 06 Dec '17

06 Dec '17

Since 517e7a1537a ("ASoC: bcm2835: move to use the clock framework") the bcm2835-i2s requires a clock as DT property. Unfortunately the necessary DT change has never been applied. While we are at it also fix the first PCM register range to cover the PCM_GRAY register. This patch also fixes the dt-bindings accordlingly. Fixes: 517e7a1537a ("ASoC: bcm2835: move to use the clock framework") Signed-off-by: Stefan Wahren <stefan.wahren(a)i2se.com> Reviewed-by: Eric Anholt <eric(a)anholt.net> Tested-by: Matthias Reichl <hias(a)horus.com> --- Documentation/devicetree/bindings/dma/brcm,bcm2835-dma.txt | 4 ++-- Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt | 9 ++++----- arch/arm/boot/dts/bcm283x.dtsi | 4 ++-- 3 files changed, 8 insertions(+), 9 deletions(-) Changes in V2: * add Fixes, Reviewed-by and Tested-by tags diff --git a/Documentation/devicetree/bindings/dma/brcm,bcm2835-dma.txt b/Documentation/devicetree/bindings/dma/brcm,bcm2835-dma.txt index baf9b34..b6a8cc0 100644 --- a/Documentation/devicetree/bindings/dma/brcm,bcm2835-dma.txt +++ b/Documentation/devicetree/bindings/dma/brcm,bcm2835-dma.txt @@ -74,8 +74,8 @@ Example: bcm2835_i2s: i2s@7e203000 { compatible = "brcm,bcm2835-i2s"; - reg = < 0x7e203000 0x20>, - < 0x7e101098 0x02>; + reg = < 0x7e203000 0x24>; + clocks = <&clocks BCM2835_CLOCK_PCM>; dmas = <&dma 2>, <&dma 3>; diff --git a/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt b/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt index 65783de..7bb0362 100644 --- a/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt +++ b/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt @@ -2,9 +2,8 @@ Required properties: - compatible: "brcm,bcm2835-i2s" -- reg: A list of base address and size entries: - * The first entry should cover the PCM registers - * The second entry should cover the PCM clock registers +- reg: Should contain PCM registers location and length. +- clocks: the (PCM) clock to use - dmas: List of DMA controller phandle and DMA request line ordered pairs. - dma-names: Identifier string for each DMA request line in the dmas property. These strings correspond 1:1 with the ordered pairs in dmas. @@ -16,8 +15,8 @@ Example: bcm2835_i2s: i2s@7e203000 { compatible = "brcm,bcm2835-i2s"; - reg = <0x7e203000 0x20>, - <0x7e101098 0x02>; + reg = <0x7e203000 0x24>; + clocks = <&clocks BCM2835_CLOCK_PCM>; dmas = <&dma 2>, <&dma 3>; diff --git a/arch/arm/boot/dts/bcm283x.dtsi b/arch/arm/boot/dts/bcm283x.dtsi index dcde93c..f493761 100644 --- a/arch/arm/boot/dts/bcm283x.dtsi +++ b/arch/arm/boot/dts/bcm283x.dtsi @@ -396,8 +396,8 @@ i2s: i2s@7e203000 { compatible = "brcm,bcm2835-i2s"; - reg = <0x7e203000 0x20>, - <0x7e101098 0x02>; + reg = <0x7e203000 0x24>; + clocks = <&clocks BCM2835_CLOCK_PCM>; dmas = <&dma 2>, <&dma 3>; -- 2.7.4

2 1

[alsa-devel] [RFC PATCH 0/3] Add ASoC platforms directory
by Andrew F. Davis 06 Dec '17

06 Dec '17

Hello all, The first patch, I think, sums up well what I'm trying to do here with this series. The last two are a couple examples of what the moving would look like. The very end result would look something like: soc/ |_codecs/ |_codecs*.c |_platforms/ |_adi/ |_adi_i2s.c |_... |_amd/ |_.../ |_machines/ |_amd/ |_acp-rt5645.c |_... |_atmel/ |_.../ |_core_files*.c I would like to think the churn is minimal for this as git seems to handles renaming like this rather well, but I guess that is up to the maintainers. :) Thanks for your comments, Andrew Andrew F. Davis (3): ASoC: Add platforms directory ASoC: Platforms: Move Davinci platform drivers into platforms directory ASoC: Platforms: Move OMAP platform drivers into platforms directory MAINTAINERS | 1 + sound/soc/Kconfig | 3 ++ sound/soc/Makefile | 1 + sound/soc/davinci/Kconfig | 37 ----------------------- sound/soc/davinci/Makefile | 21 +++---------- sound/soc/omap/Kconfig | 29 ------------------ sound/soc/omap/Makefile | 13 -------- sound/soc/omap/am3517evm.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap-abe-twl6040.c | 4 +-- sound/soc/omap/omap-twl4030.c | 2 +- sound/soc/omap/omap3pandora.c | 2 +- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/rx51.c | 2 +- sound/soc/platforms/Kconfig | 10 ++++++ sound/soc/platforms/Makefile | 4 +++ sound/soc/platforms/davinci/Kconfig | 36 ++++++++++++++++++++++ sound/soc/{ => platforms}/davinci/Makefile | 5 --- sound/soc/{ => platforms}/davinci/davinci-i2s.c | 0 sound/soc/{ => platforms}/davinci/davinci-i2s.h | 0 sound/soc/{ => platforms}/davinci/davinci-mcasp.c | 0 sound/soc/{ => platforms}/davinci/davinci-mcasp.h | 0 sound/soc/{ => platforms}/davinci/davinci-vcif.c | 0 sound/soc/{ => platforms}/davinci/edma-pcm.c | 0 sound/soc/{ => platforms}/davinci/edma-pcm.h | 0 sound/soc/platforms/omap/Kconfig | 28 +++++++++++++++++ sound/soc/platforms/omap/Makefile | 13 ++++++++ sound/soc/{ => platforms}/omap/mcbsp.c | 0 sound/soc/{ => platforms}/omap/mcbsp.h | 0 sound/soc/{ => platforms}/omap/omap-dmic.c | 0 sound/soc/{ => platforms}/omap/omap-dmic.h | 0 sound/soc/{ => platforms}/omap/omap-hdmi-audio.c | 0 sound/soc/{ => platforms}/omap/omap-mcbsp.c | 0 sound/soc/{ => platforms}/omap/omap-mcbsp.h | 0 sound/soc/{ => platforms}/omap/omap-mcpdm.c | 0 sound/soc/{ => platforms}/omap/omap-mcpdm.h | 0 sound/soc/{ => platforms}/omap/omap-pcm.c | 0 38 files changed, 110 insertions(+), 109 deletions(-) rewrite sound/soc/davinci/Makefile (71%) create mode 100644 sound/soc/platforms/Kconfig create mode 100644 sound/soc/platforms/Makefile create mode 100644 sound/soc/platforms/davinci/Kconfig copy sound/soc/{ => platforms}/davinci/Makefile (76%) rename sound/soc/{ => platforms}/davinci/davinci-i2s.c (100%) rename sound/soc/{ => platforms}/davinci/davinci-i2s.h (100%) rename sound/soc/{ => platforms}/davinci/davinci-mcasp.c (100%) rename sound/soc/{ => platforms}/davinci/davinci-mcasp.h (100%) rename sound/soc/{ => platforms}/davinci/davinci-vcif.c (100%) rename sound/soc/{ => platforms}/davinci/edma-pcm.c (100%) rename sound/soc/{ => platforms}/davinci/edma-pcm.h (100%) create mode 100644 sound/soc/platforms/omap/Kconfig create mode 100644 sound/soc/platforms/omap/Makefile rename sound/soc/{ => platforms}/omap/mcbsp.c (100%) rename sound/soc/{ => platforms}/omap/mcbsp.h (100%) rename sound/soc/{ => platforms}/omap/omap-dmic.c (100%) rename sound/soc/{ => platforms}/omap/omap-dmic.h (100%) rename sound/soc/{ => platforms}/omap/omap-hdmi-audio.c (100%) rename sound/soc/{ => platforms}/omap/omap-mcbsp.c (100%) rename sound/soc/{ => platforms}/omap/omap-mcbsp.h (100%) rename sound/soc/{ => platforms}/omap/omap-mcpdm.c (100%) rename sound/soc/{ => platforms}/omap/omap-mcpdm.h (100%) rename sound/soc/{ => platforms}/omap/omap-pcm.c (100%) -- 2.15.0

2 11

[alsa-devel] [PATCH 00/11] ASoC: Intel: Skylake: Bug fixes for skylake driver
by Guneshwor Singh 06 Dec '17

06 Dec '17

This series include couple of bug fixes which include memory leak fixes, null pointer dereference fixes, fixes for errors generated by static checker, updown mixer module ipc format fix and use module uuid instead of module id which is required for module pins. Guneshwor Singh (2): ASoC: Intel: Skylake: Fix updown mixer module format ASoC: Intel: Skylake: Add channel map in updown mixer module IPC Pankaj Bharadiya (3): ASoC: Intel: Skylake: Fix potential NULL pointer dereference ASoC: Intel: Skylake: Fix uuid_module memory leak in failure case ASoC: Intel: Skylake: Check for NHLT ACPI header signature Sriram Periyasamy (2): ASoC: Intel: Skylake: Optimize UUID handling to fill pin info ASoC: Intel: Skylake: Add dynamic module id support Subhransu S. Prusty (4): ASoC: Intel: Skylake: Remove second shim read in register_poll ASoC: hdac_hdmi: Fix static checker warning for sprintf usage ASoC: hdac_hdmi: Fix possible NULL pointer dereference ASoC: hdac_hdmi: Fix possible memory leak on parse and map nid failure sound/soc/codecs/hdac_hdmi.c | 51 ++++++++++++++++++++++---- sound/soc/intel/common/sst-dsp.c | 4 +- sound/soc/intel/skylake/skl-messages.c | 9 +---- sound/soc/intel/skylake/skl-nhlt.c | 9 +++++ sound/soc/intel/skylake/skl-pcm.c | 3 +- sound/soc/intel/skylake/skl-sst-utils.c | 15 ++++++-- sound/soc/intel/skylake/skl-topology.c | 65 +++++++++++++++++++++------------ sound/soc/intel/skylake/skl-topology.h | 3 +- 8 files changed, 112 insertions(+), 47 deletions(-) -- 2.14.2

3 23

[alsa-devel] [PATCH 0/4] ASoC: Intel: Skylake: Skylake driver updates
by Guneshwor Singh 06 Dec '17

06 Dec '17

This series includes fixes for HDMI widgets sysfs not updating, shim read optimization and feature to configure DSP clock source. Pradeep Tewani (2): ASoC: Intel: Skylake: Parse vendor tokens to build A-State table ASoC: Intel: Skylake: Configure DSP clock source Puneeth Prabhu (1): ASoC: hdac_hdmi: Refresh sysfs during hdmi device probe Subhransu S. Prusty (1): ASoC: Intel: Skylake: Remove second shim read in register_poll include/uapi/sound/snd_sst_tokens.h | 17 ++++++++++++- sound/soc/codecs/hdac_hdmi.c | 1 + sound/soc/intel/common/sst-dsp.c | 4 +--- sound/soc/intel/skylake/skl-messages.c | 18 ++++++++++++++ sound/soc/intel/skylake/skl-pcm.c | 6 +++++ sound/soc/intel/skylake/skl-sst-dsp.h | 3 +++ sound/soc/intel/skylake/skl-topology.c | 44 +++++++++++++++++++++++++++++++++- sound/soc/intel/skylake/skl.h | 17 +++++++++++++ 8 files changed, 105 insertions(+), 5 deletions(-) -- 2.15.0

3 7

[alsa-devel] Applied "ASoC: hisilicon: remove duplicate includes" to the asoc tree
by Mark Brown 06 Dec '17

06 Dec '17

The patch ASoC: hisilicon: remove duplicate includes has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 769e40f0c17df53bac8999939993a280c1ea3dbd Mon Sep 17 00:00:00 2001 From: Pravin Shedge <pravin.shedge4linux(a)gmail.com> Date: Wed, 6 Dec 2017 23:09:00 +0530 Subject: [PATCH] ASoC: hisilicon: remove duplicate includes These duplicate includes have been found with scripts/checkincludes.pl but they have been removed manually to avoid removing false positives. Signed-off-by: Pravin Shedge <pravin.shedge4linux(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/hisilicon/hi6210-i2s.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 0c8f86d4020e..07a57209e055 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -36,7 +36,6 @@ #include <linux/of_irq.h> #include <linux/mfd/syscon.h> #include <linux/reset-controller.h> -#include <linux/clk.h> #include "hi6210-i2s.h" -- 2.15.0

1 0

[alsa-devel] [PATCH 30/45] sound: soc: hisilicon: remove duplicate includes
by Pravin Shedge 06 Dec '17

06 Dec '17

These duplicate includes have been found with scripts/checkincludes.pl but they have been removed manually to avoid removing false positives. Signed-off-by: Pravin Shedge <pravin.shedge4linux(a)gmail.com> --- sound/soc/hisilicon/hi6210-i2s.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 0c8f86d..07a5720 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -36,7 +36,6 @@ #include <linux/of_irq.h> #include <linux/mfd/syscon.h> #include <linux/reset-controller.h> -#include <linux/clk.h> #include "hi6210-i2s.h" -- 2.7.4

2 1

[alsa-devel] [PATCH v3 34/36] ALSA/dummy: Replace tasklet with softirq hrtimer
by Anna-Maria Gleixner 06 Dec '17

06 Dec '17

From: Thomas Gleixner <tglx(a)linutronix.de> The tasklet is used to defer the execution of snd_pcm_period_elapsed() to the softirq context. Using the HRTIMER_MODE_SOFT mode invokes the timer callback in softirq context as well which renders the tasklet useless. [o-takashi: avoid stall due to a call of hrtimer_cancel() on a callback of hrtimer] Signed-off-by: Thomas Gleixner <tglx(a)linutronix.de> Signed-off-by: Anna-Maria Gleixner <anna-maria(a)linutronix.de> Cc: alsa-devel(a)alsa-project.org Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Link: http://lkml.kernel.org/r/20170905161820.jtysvxtfleunbbmf@breakpoint.cc --- sound/drivers/dummy.c | 27 ++++++++++++--------------- 1 file changed, 12 insertions(+), 15 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 7b2b1f766b00..6ad2ff57833d 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -375,17 +375,9 @@ struct dummy_hrtimer_pcm { ktime_t period_time; atomic_t running; struct hrtimer timer; - struct tasklet_struct tasklet; struct snd_pcm_substream *substream; }; -static void dummy_hrtimer_pcm_elapsed(unsigned long priv) -{ - struct dummy_hrtimer_pcm *dpcm = (struct dummy_hrtimer_pcm *)priv; - if (atomic_read(&dpcm->running)) - snd_pcm_period_elapsed(dpcm->substream); -} - static enum hrtimer_restart dummy_hrtimer_callback(struct hrtimer *timer) { struct dummy_hrtimer_pcm *dpcm; @@ -393,7 +385,14 @@ static enum hrtimer_restart dummy_hrtimer_callback(struct hrtimer *timer) dpcm = container_of(timer, struct dummy_hrtimer_pcm, timer); if (!atomic_read(&dpcm->running)) return HRTIMER_NORESTART; - tasklet_schedule(&dpcm->tasklet); + /* + * In cases of XRUN and draining, this calls .trigger to stop PCM + * substream. + */ + snd_pcm_period_elapsed(dpcm->substream); + if (!atomic_read(&dpcm->running)) + return HRTIMER_NORESTART; + hrtimer_forward_now(timer, dpcm->period_time); return HRTIMER_RESTART; } @@ -403,7 +402,7 @@ static int dummy_hrtimer_start(struct snd_pcm_substream *substream) struct dummy_hrtimer_pcm *dpcm = substream->runtime->private_data; dpcm->base_time = hrtimer_cb_get_time(&dpcm->timer); - hrtimer_start(&dpcm->timer, dpcm->period_time, HRTIMER_MODE_REL); + hrtimer_start(&dpcm->timer, dpcm->period_time, HRTIMER_MODE_REL_SOFT); atomic_set(&dpcm->running, 1); return 0; } @@ -413,14 +412,14 @@ static int dummy_hrtimer_stop(struct snd_pcm_substream *substream) struct dummy_hrtimer_pcm *dpcm = substream->runtime->private_data; atomic_set(&dpcm->running, 0); - hrtimer_cancel(&dpcm->timer); + if (!hrtimer_callback_running(&dpcm->timer)) + hrtimer_cancel(&dpcm->timer); return 0; } static inline void dummy_hrtimer_sync(struct dummy_hrtimer_pcm *dpcm) { hrtimer_cancel(&dpcm->timer); - tasklet_kill(&dpcm->tasklet); } static snd_pcm_uframes_t @@ -465,12 +464,10 @@ static int dummy_hrtimer_create(struct snd_pcm_substream *substream) if (!dpcm) return -ENOMEM; substream->runtime->private_data = dpcm; - hrtimer_init(&dpcm->timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + hrtimer_init(&dpcm->timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL_SOFT); dpcm->timer.function = dummy_hrtimer_callback; dpcm->substream = substream; atomic_set(&dpcm->running, 0); - tasklet_init(&dpcm->tasklet, dummy_hrtimer_pcm_elapsed, - (unsigned long)dpcm); return 0; } -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: fsl_asrc: protect macro argument
by Stefan Agner 06 Dec '17

06 Dec '17

Protect macro argument with parentheses to avoid ambiguity. This fixes a warning seen with clang: warning: logical not is only applied to the left hand side of this comparison Signed-off-by: Stefan Agner <stefan(a)agner.ch> --- sound/soc/fsl/fsl_asrc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index 0f163abe4ba3..ec33dab4b909 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -57,7 +57,7 @@ #define REG_ASRDOC 0x74 #define REG_ASRDI(i) (REG_ASRDIA + (i << 3)) #define REG_ASRDO(i) (REG_ASRDOA + (i << 3)) -#define REG_ASRDx(x, i) (x == IN ? REG_ASRDI(i) : REG_ASRDO(i)) +#define REG_ASRDx(x, i) ((x) == IN ? REG_ASRDI(i) : REG_ASRDO(i)) #define REG_ASRIDRHA 0x80 #define REG_ASRIDRLA 0x84 -- 2.13.3

3 7

[alsa-devel] Applied "ASoC: rockchip: disable clock on error" to the asoc tree
by Mark Brown 06 Dec '17

06 Dec '17

The patch ASoC: rockchip: disable clock on error has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c7b92172a61b91936be985cb9bc499a4ebc6489b Mon Sep 17 00:00:00 2001 From: Stefan Potyra <Stefan.Potyra(a)elektrobit.com> Date: Wed, 6 Dec 2017 16:03:24 +0100 Subject: [PATCH] ASoC: rockchip: disable clock on error Disable the clocks in rk_spdif_probe when an error occurs after one of the clocks has been enabled previously. Found by Linux Driver Verification project (linuxtesting.org). Fixes: f874b80e1571 ASoC: rockchip: Add rockchip SPDIF transceiver driver Signed-off-by: Stefan Potyra <Stefan.Potyra(a)elektrobit.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/rockchip/rockchip_spdif.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index ee5055d47d13..a89fe9b6463b 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -322,26 +322,30 @@ static int rk_spdif_probe(struct platform_device *pdev) spdif->mclk = devm_clk_get(&pdev->dev, "mclk"); if (IS_ERR(spdif->mclk)) { dev_err(&pdev->dev, "Can't retrieve rk_spdif master clock\n"); - return PTR_ERR(spdif->mclk); + ret = PTR_ERR(spdif->mclk); + goto err_disable_hclk; } ret = clk_prepare_enable(spdif->mclk); if (ret) { dev_err(spdif->dev, "clock enable failed %d\n", ret); - return ret; + goto err_disable_clocks; } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(regs)) - return PTR_ERR(regs); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); + goto err_disable_clocks; + } spdif->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "hclk", regs, &rk_spdif_regmap_config); if (IS_ERR(spdif->regmap)) { dev_err(&pdev->dev, "Failed to initialise managed register map\n"); - return PTR_ERR(spdif->regmap); + ret = PTR_ERR(spdif->regmap); + goto err_disable_clocks; } spdif->playback_dma_data.addr = res->start + SPDIF_SMPDR; @@ -373,6 +377,10 @@ static int rk_spdif_probe(struct platform_device *pdev) err_pm_runtime: pm_runtime_disable(&pdev->dev); +err_disable_clocks: + clk_disable_unprepare(spdif->mclk); +err_disable_hclk: + clk_disable_unprepare(spdif->hclk); return ret; } -- 2.15.0

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[alsa-devel] [PATCH RESEND] ASoC: fsl_asrc: protect macro argument
by Stefan Agner 06 Dec '17

06 Dec '17

Protect macro argument with parentheses to avoid ambiguity. This fixes a warning seen with clang: warning: logical not is only applied to the left hand side of this comparison Signed-off-by: Stefan Agner <stefan(a)agner.ch> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_asrc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index 0f163abe4ba3..ec33dab4b909 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -57,7 +57,7 @@ #define REG_ASRDOC 0x74 #define REG_ASRDI(i) (REG_ASRDIA + (i << 3)) #define REG_ASRDO(i) (REG_ASRDOA + (i << 3)) -#define REG_ASRDx(x, i) (x == IN ? REG_ASRDI(i) : REG_ASRDO(i)) +#define REG_ASRDx(x, i) ((x) == IN ? REG_ASRDI(i) : REG_ASRDO(i)) #define REG_ASRIDRHA 0x80 #define REG_ASRIDRLA 0x84 -- 2.13.3

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[alsa-devel] [PATCH] SoC: rockchip: disable clock on error
by Stefan Potyra 06 Dec '17

06 Dec '17

Disable the clocks in rk_spdif_probe when an error occurs after one of the clocks has been enabled previously. Found by Linux Driver Verification project (linuxtesting.org). Fixes: f874b80e1571 ASoC: rockchip: Add rockchip SPDIF transceiver driver Signed-off-by: Stefan Potyra <Stefan.Potyra(a)elektrobit.com> --- sound/soc/rockchip/rockchip_spdif.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index ee5055d47d13..a89fe9b6463b 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -322,26 +322,30 @@ static int rk_spdif_probe(struct platform_device *pdev) spdif->mclk = devm_clk_get(&pdev->dev, "mclk"); if (IS_ERR(spdif->mclk)) { dev_err(&pdev->dev, "Can't retrieve rk_spdif master clock\n"); - return PTR_ERR(spdif->mclk); + ret = PTR_ERR(spdif->mclk); + goto err_disable_hclk; } ret = clk_prepare_enable(spdif->mclk); if (ret) { dev_err(spdif->dev, "clock enable failed %d\n", ret); - return ret; + goto err_disable_clocks; } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(regs)) - return PTR_ERR(regs); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); + goto err_disable_clocks; + } spdif->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "hclk", regs, &rk_spdif_regmap_config); if (IS_ERR(spdif->regmap)) { dev_err(&pdev->dev, "Failed to initialise managed register map\n"); - return PTR_ERR(spdif->regmap); + ret = PTR_ERR(spdif->regmap); + goto err_disable_clocks; } spdif->playback_dma_data.addr = res->start + SPDIF_SMPDR; @@ -373,6 +377,10 @@ static int rk_spdif_probe(struct platform_device *pdev) err_pm_runtime: pm_runtime_disable(&pdev->dev); +err_disable_clocks: + clk_disable_unprepare(spdif->mclk); +err_disable_hclk: + clk_disable_unprepare(spdif->hclk); return ret; } -- 2.15.1

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[alsa-devel] [PATCH v4 00/15] soundwire: Add a new SoundWire subsystem
by Vinod Koul 06 Dec '17

06 Dec '17

This patch series adds a new SoundWire subsystem which implements a new MIPI bus protocol 'SoundWire'. The SoundWire protocol is a robust, scalable, low complexity, low power, low latency, two-pin (clock and data) multi-drop bus that allows for the transfer of multiple audio streams and embedded control/commands. SoundWire provides synchronization capabilities and supports both PCM and PDM, multichannel data, isochronous and asynchronous modes. This series adds SoundWire Bus, IO transfers, DisCo (Discovery and Configuration) sysfs interface, regmap and Documentation summary This patch series is also available on git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git topic/patch_v4 v3: https://lkml.org/lkml/2017/11/30/160 v2: https://lkml.org/lkml/2017/11/10/216 v1: https://lkml.org/lkml/2017/10/18/1030 RFC: https://lkml.org/lkml/2016/10/21/395 Changes in v4: - Remove text licenses and add SPDX tags only with C99 style comments - make bus_type code as GPL v2.0 only Changes in v3: - Update the kernel-doc styles and fix included headers for files - handle dev_pm_domain_attach() for defered probe - remove OF placeholders - change regmap license to GPLv2 only Changes in v2: - move documentation into driver-api and do rst conversion - fix documentation comments - add SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) to all source files - rework the transfer logic and paging logic as commented on v1 - remove dummy sysfs fns - registration checks and fixes - remove slave check for regamp as that turned superfluous - remove depends SoundWire symbol - make modalias api arg const - use bitmap for tracking assigned - add counter for report present tracking todo: add the dt-bindings Sanyog Kale (4): Documentation: Add SoundWire summary soundwire: Add SoundWire MIPI defined registers soundwire: Add Slave status handling helpers soundwire: cdns: Add sdw_master_ops and IO transfer support Vinod Koul (11): soundwire: Add SoundWire bus type soundwire: Add Master registration soundwire: Add MIPI DisCo property helpers soundwire: Add IO transfer regmap: Add SoundWire bus support soundwire: Add slave status handling soundwire: Add sysfs for SoundWire DisCo properties soundwire: cdns: Add cadence library soundwire: intel: Add Intel Master driver soundwire: intel: Add Intel init module MAINTAINERS: Add SoundWire entry Documentation/driver-api/index.rst | 1 + Documentation/driver-api/soundwire/index.rst | 15 + Documentation/driver-api/soundwire/summary.rst | 205 ++++++ MAINTAINERS | 10 + drivers/Kconfig | 2 + drivers/Makefile | 1 + drivers/base/regmap/Kconfig | 4 + drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-sdw.c | 92 +++ drivers/soundwire/Kconfig | 37 + drivers/soundwire/Makefile | 18 + drivers/soundwire/bus.c | 958 +++++++++++++++++++++++++ drivers/soundwire/bus.h | 74 ++ drivers/soundwire/bus_type.c | 193 +++++ drivers/soundwire/cadence_master.c | 749 +++++++++++++++++++ drivers/soundwire/cadence_master.h | 48 ++ drivers/soundwire/intel.c | 347 +++++++++ drivers/soundwire/intel.h | 23 + drivers/soundwire/intel_init.c | 198 +++++ drivers/soundwire/mipi_disco.c | 374 ++++++++++ drivers/soundwire/slave.c | 115 +++ drivers/soundwire/sysfs.c | 343 +++++++++ include/linux/mod_devicetable.h | 6 + include/linux/regmap.h | 37 + include/linux/soundwire/sdw.h | 487 +++++++++++++ include/linux/soundwire/sdw_intel.h | 24 + include/linux/soundwire/sdw_registers.h | 194 +++++ include/linux/soundwire/sdw_type.h | 19 + scripts/mod/devicetable-offsets.c | 4 + scripts/mod/file2alias.c | 15 + 30 files changed, 4594 insertions(+) create mode 100644 Documentation/driver-api/soundwire/index.rst create mode 100644 Documentation/driver-api/soundwire/summary.rst create mode 100644 drivers/base/regmap/regmap-sdw.c create mode 100644 drivers/soundwire/Kconfig create mode 100644 drivers/soundwire/Makefile create mode 100644 drivers/soundwire/bus.c create mode 100644 drivers/soundwire/bus.h create mode 100644 drivers/soundwire/bus_type.c create mode 100644 drivers/soundwire/cadence_master.c create mode 100644 drivers/soundwire/cadence_master.h create mode 100644 drivers/soundwire/intel.c create mode 100644 drivers/soundwire/intel.h create mode 100644 drivers/soundwire/intel_init.c create mode 100644 drivers/soundwire/mipi_disco.c create mode 100644 drivers/soundwire/slave.c create mode 100644 drivers/soundwire/sysfs.c create mode 100644 include/linux/soundwire/sdw.h create mode 100644 include/linux/soundwire/sdw_intel.h create mode 100644 include/linux/soundwire/sdw_registers.h create mode 100644 include/linux/soundwire/sdw_type.h -- 2.7.4

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[alsa-devel] [PATCH v3 1/2] ASoC: codecs: Add PCM186x binding documentation
by Andrew F. Davis 06 Dec '17

06 Dec '17

Add the dt-binding documentation for the TI PCM186x 2ch and 4ch Audio ADCs With Universal Front End. Signed-off-by: Andrew F. Davis <afd(a)ti.com> Acked-by: Rob Herring <robh(a)kernel.org> --- Changes from v2: - None Changes from v1: - pcm186x@4a -> audio-codec@4a - Added Acked-by .../devicetree/bindings/sound/pcm186x.txt | 42 ++++++++++++++++++++++ 1 file changed, 42 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm186x.txt diff --git a/Documentation/devicetree/bindings/sound/pcm186x.txt b/Documentation/devicetree/bindings/sound/pcm186x.txt new file mode 100644 index 000000000000..1087f4855980 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm186x.txt @@ -0,0 +1,42 @@ +Texas Instruments PCM186x Universal Audio ADC + +These devices support both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "ti,pcm1862", + "ti,pcm1863", + "ti,pcm1864", + "ti,pcm1865" + + - reg : The I2C address of the device for I2C, the chip select + number for SPI. + + - avdd-supply: Analog core power supply (3.3v) + - dvdd-supply: Digital core power supply + - iovdd-supply: Digital IO power supply + See regulator/regulator.txt for more information + +CODEC input pins: + * VINL1 + * VINR1 + * VINL2 + * VINR2 + * VINL3 + * VINR3 + * VINL4 + * VINR4 + +The pins can be used in referring sound node's audio-routing property. + +Example: + + pcm186x: audio-codec@4a { + compatible = "ti,pcm1865"; + reg = <0x4a>; + + avdd-supply = <&reg_3v3_analog>; + dvdd-supply = <&reg_3v3>; + iovdd-supply = <&reg_1v8>; + }; -- 2.15.0

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[alsa-devel] [PATCH v2 1/2] ASoC: codecs: Add PCM186x binding documentation
by Andrew F. Davis 06 Dec '17

06 Dec '17

Add the dt-binding documentation for the TI PCM186x 2ch and 4ch Audio ADCs With Universal Front End. Signed-off-by: Andrew F. Davis <afd(a)ti.com> Acked-by: Rob Herring <robh(a)kernel.org> --- Changes from v1: - pcm186x@4a -> audio-codec@4a - Added Acked-by .../devicetree/bindings/sound/pcm186x.txt | 42 ++++++++++++++++++++++ 1 file changed, 42 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm186x.txt diff --git a/Documentation/devicetree/bindings/sound/pcm186x.txt b/Documentation/devicetree/bindings/sound/pcm186x.txt new file mode 100644 index 000000000000..1087f4855980 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm186x.txt @@ -0,0 +1,42 @@ +Texas Instruments PCM186x Universal Audio ADC + +These devices support both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "ti,pcm1862", + "ti,pcm1863", + "ti,pcm1864", + "ti,pcm1865" + + - reg : The I2C address of the device for I2C, the chip select + number for SPI. + + - avdd-supply: Analog core power supply (3.3v) + - dvdd-supply: Digital core power supply + - iovdd-supply: Digital IO power supply + See regulator/regulator.txt for more information + +CODEC input pins: + * VINL1 + * VINR1 + * VINL2 + * VINR2 + * VINL3 + * VINR3 + * VINL4 + * VINR4 + +The pins can be used in referring sound node's audio-routing property. + +Example: + + pcm186x: audio-codec@4a { + compatible = "ti,pcm1865"; + reg = <0x4a>; + + avdd-supply = <&reg_3v3_analog>; + dvdd-supply = <&reg_3v3>; + iovdd-supply = <&reg_1v8>; + }; -- 2.15.0

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Re: [alsa-devel] [PATCH 5/8] ASoC: uniphier: add support for UniPhier AIO driver
by Mark Brown 06 Dec '17

06 Dec '17

On Wed, Dec 06, 2017 at 03:03:18PM +0900, Katsuhiro Suzuki wrote: > > I'd expect this code to be structured more like a library - have a > > driver that handles the specific IPs then have it call into a shared > > block of code that does the generic bits. Though in this case the > > device specific bit looks like a couple of tiny data tables so I'm not > > sure it's worth making it conditional or separate at all. > Sorry... I agree your opinion, but I can't imagine the detail. > I think my driver has structure as follows (ex. startup): > DAI: uniphier_aio_startup()(a)aio-core.c > Lib: uniphier_aio_init()(a)aio-regctrl.c > SoC specific: uniphier_aio_ld11_spec(a)aio-ld11.c > Am I wrong? Would you mean split the functions in aio-regctl.[ch] to other > kernel module? I wonder if you could tell me the example from existing > drivers. I'll try to fix my driver like as it. One example is how all the drivers that use the generic dmaengine code instantiate their DMA drivers, or how all the drivers for CODECs that have both I2C and SPIi control interfaces instantiate - given that the device specific code here seems to be mostly data tables that's probably the closest thing. > > At least. I do think we need to get to the bottom of how flexible the > > hardware is first though. > Yes, indeed. This hardware is more flexible and complex, but now I (and our > company) don't use it. Of course, I don't want to hide some features of this > hardware from ALSA people. I should try to upstream all features in the future, > I think. My main concern here is to make sure that when you decide you need to use the more complex hardware that this can be done without too much pain to existing machines (and that they can benefit from as much of the enhanced functionality as is possible).

1 0

[alsa-devel] [PATCH v4 0/2] Add support for TI TAS6424 digital amplifier
by Andrew F. Davis 06 Dec '17

06 Dec '17

Hello all, This series adds support for the TAS6424 digital amplifier. A lot has changed from the last time this upstreaming was attempted, I've taken over this effort for Michael and I don't really have an exact change log but hopefully I've addressed all comments from v3. Thanks, Andrew Andreas Dannenberg (1): ASoC: Add support for TAS6424 digital amplifier Michael Stecklein (1): dt-bindings: sound: add bindings for TAS6424 .../devicetree/bindings/sound/ti,tas6424.txt | 20 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas6424.c | 707 +++++++++++++++++++++ sound/soc/codecs/tas6424.h | 144 +++++ 5 files changed, 881 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas6424.txt create mode 100644 sound/soc/codecs/tas6424.c create mode 100644 sound/soc/codecs/tas6424.h -- 2.15.0

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[alsa-devel] Question about soc-utils
by Kuninori Morimoto 06 Dec '17

06 Dec '17

Hi Mark, ALSA SoC ML In my understanding, "frame size" means "channel x physical width". linux/sound/core/pcm_native.c :: snd_pcm_hw_params() is doing so static int snd_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { ... => bits = snd_pcm_format_physical_width(runtime->format); runtime->sample_bits = bits; => bits *= runtime->channels; runtime->frame_bits = bits; ... } But, soc-utils::snd_soc_params_to_frame_size() is using snd_pcm_format_width() instead of snd_pcm_format_physical_width(), In "2ch x 24bit data with 32bit width" case, above "frame_bits" will be "64", but, below "frame size" will be "48", I think. Am I wrong ? int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params) { int sample_size; => sample_size = snd_pcm_format_width(params_format(params)); if (sample_size < 0) return sample_size; => return snd_soc_calc_frame_size(sample_size, params_channels(params), 1); } About "TDM slot", in my understanding it is TDM channel number. TDM6 = 6ch = 6 slot, TDM8 = 8ch = 8 slot. Am I wrong ?? I'm asking because soc-utils :: snd_soc_calc_bclk() / snd_soc_calc_frame_size() have both "channles" and "slot" parameter. What is this "tdm_slots" means ? int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots) int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH v4 0/7] ASoC: Intel: Skylake: Add a clk driver to enable ssp clks early
by Sriram Periyasamy 06 Dec '17

06 Dec '17

For certain platforms, clocks (mclk/sclk/fs) are required to be up before the stream start. Example: some codecs needs the mclk/sclk/fs to be enabled early for a successful clock synchronization. Some platforms require clock to be enabled at boot and be always ON. By sending set_dma_control IPC (with the i2s blobs queried from NHLT), these clocks can be enabled early after the firmware is downloaded. With this series, a virtual clock driver is created which provides interface to send the required IPCs from machine driver to enable the clocks. NHLT is parsed during probe and the clock information is populated. The pointer to blob is cached and sent along with the set_dma_control IPC structure during the clk prepare/unprepare callback. Clocks are created for a ssp if the nhlt table has endpoint configuration for that particular ssp. Skylake driver creates a platform driver with the clock information and register the clk ops callback. Kabylake machine driver uses the clock interface to enable the clocks early as it is required by the rt5663 driver for clock synchronization. v3 -> v4 - Add missing signed-offs v2 -> v3 - Moved the clk ops and IPCs from Skylake driver to clk driver and reordered commits accordingly - Add the support for extended I2S blob config which supports multiple mclk dividers in NHLT - Enable the clocks as well in DAPM PMU event instead of hw_params in machine drivers as confirmed by codec vendor - Do not register the clk if there is no valid clock source is avail in the I2S blob - Take care of error return in the clk driver - Address rest of the review comments and more optimization added - Fix the warning sound/soc/intel/skylake/skl.c:724:1-3: WARNING: PTR_ERR_OR_ZERO can be used reported by scripts/coccinelle/api/ptr_ret.cocci - Modified DSP replies as human readable to ease the debugging - Add firmware replies for MCLK/SCLK clocks if they are running already v1 -> v2 - Register parent clocks with skylake device. With the patch "clk: Add support for runtime PM" soon to be merged will help DSP to stay active on call to clock enable. Reference: (https://patchwork.kernel.org/patch/9911741/) - Fix the machine driver to enable clocks early for headphone playback path as well to fix a pop noise issue Harsha Priya (1): ASoC: Intel: kbl: Enable mclk and ssp sclk early Naveen M (1): ASoC: Intel: eve: Enable mclk and ssp sclk early Sriram Periyasamy (3): ASoC: Intel: Skylake: Parse nhlt and register clock device ASoC: Intel: Skylake: Add ssp clock driver ASoC: Intel: Skylake: Add extended I2S config blob support in Clock driver Subhransu S. Prusty (2): ASoC: Intel: Skylake: Make DSP replies more human readable ASoC: Intel: Skylake: Add FW reply for MCLK/SCLK IPC sound/soc/intel/Kconfig | 3 + sound/soc/intel/boards/Kconfig | 2 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 101 ++++- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 100 +++++ sound/soc/intel/skylake/Makefile | 5 + sound/soc/intel/skylake/skl-i2s.h | 95 +++++ sound/soc/intel/skylake/skl-messages.c | 1 + sound/soc/intel/skylake/skl-nhlt.c | 172 ++++++++ sound/soc/intel/skylake/skl-ssp-clk.c | 470 +++++++++++++++++++++ sound/soc/intel/skylake/skl-ssp-clk.h | 117 +++++ sound/soc/intel/skylake/skl-sst-ipc.c | 50 ++- sound/soc/intel/skylake/skl.c | 93 ++++ sound/soc/intel/skylake/skl.h | 10 + 13 files changed, 1207 insertions(+), 12 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-i2s.h create mode 100644 sound/soc/intel/skylake/skl-ssp-clk.c create mode 100644 sound/soc/intel/skylake/skl-ssp-clk.h -- 2.7.4

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[alsa-devel] [PATCH 0/7] ASoC: don't use rtd->codec - step [6/6]
by Kuninori Morimoto 06 Dec '17

06 Dec '17

Hi Mark These are step [6/6] patch-set for converting codec/platform to conponent. rtd/dai have ->codec now, but it will be removed soon. Thus we need to use ->component instead of it. Kuninori Morimoto (7): ASoC: don't use rtd->codec on soc_dev_attr_is_visible() ASoC: don't use rtd->codec on snd_soc_new_compress() ASoC: don't use rtd->codec on fsl-asoc-card ASoC: don't use rtd->codec on intel/skylake ASoC: don't use rtd->codec on Intel/haswell ASoC: don't use rtd->codec on qcom/apq8016_sbc ASoC: don't use rtd->codec on samsung/bells sound/soc/fsl/fsl-asoc-card.c | 4 ++-- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 4 ++-- sound/soc/qcom/apq8016_sbc.c | 8 ++++---- sound/soc/samsung/bells.c | 40 +++++++++++++++++++-------------------- sound/soc/soc-compress.c | 4 ++-- sound/soc/soc-core.c | 2 +- 7 files changed, 32 insertions(+), 32 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ALSA: pcm: prevent UAF in snd_pcm_info
by Nick Desaulniers 06 Dec '17

06 Dec '17

From: Robb Glasser <rglasser(a)google.com> When the device descriptor is closed, the `substream->runtime` pointer is freed. But another thread may be in the ioctl handler, case SNDRV_CTL_IOCTL_PCM_INFO. This case calls snd_pcm_info_user() which calls snd_pcm_info() which accesses the now freed `substream->runtime`. Signed-off-by: Robb Glasser <rglasser(a)google.com> Signed-off-by: Nick Desaulniers <ndesaulniers(a)google.com> --- sound/core/pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 9070f277f8db..09ee8c6b9f75 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -153,7 +153,9 @@ static int snd_pcm_control_ioctl(struct snd_card *card, err = -ENXIO; goto _error; } + mutex_lock(&pcm->open_mutex); err = snd_pcm_info_user(substream, info); + mutex_unlock(&pcm->open_mutex); _error: mutex_unlock(&register_mutex); return err; -- 2.15.0.531.g2ccb3012c9-goog

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[alsa-devel] [PATCH v8 00/13] Introduce framework for SLIMbus device driver
by srinivas.kandagatla＠linaro.org 06 Dec '17

06 Dec '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Thanks for everyone who reviewed v7 patchset, here is v8 with review comments addressed. SLIMbus (Serial Low Power Interchip Media Bus) is a specification developed by MIPI (Mobile Industry Processor Interface) alliance. SLIMbus is a 2-wire implementation, which is used to communicate with peripheral components like audio-codec. SLIMbus uses Time-Division-Multiplexing to accommodate multiple data channels, and control channel. Control channel has messages to do device-enumeration, messages to send/receive control-data to/from SLIMbus devices, messages for port/channel management, and messages to do bandwidth allocation. Framework is introduced to support multiple instances of the bus (1 controller per bus), and multiple slave devices per controller. SPI and I2C frameworks, and comments from last time when I submitted the patches were referred-to while working on this framework. These patchsets introduce device-management, OF helpers, and messaging APIs, controller driver for Qualcomm's SLIMbus controller, and clock-pause feature for entering/exiting low-power mode for SLIMbus. Framework patches to do channel, port and bandwidth management are work-in-progress and will be sent out once these initial patches are accepted. These patchsets were tested on Qualcomm Snapdragon processor board using the controller driver, and a test slave device. v7: https://lwn.net/Articles/739150/ Changes from v7 to v8: * Fixed few warnings from kerneldoc * Few cosmetic fixes suggested by Vinod * made SLIMbus string usage more consistent with spec * made probe mandatory for slim device * replaced explicit info and value element apis with more generic api * made driver_data aligned * cleaned up qcom controller driver as suggested by Charles and Jonathan Sagar Dharia (9): Documentation: Add SLIMbus summary dt-bindings: Add SLIMbus bindings slimbus: Add SLIMbus bus type slimbus: core: Add slim controllers support slimbus: Add messaging APIs to slimbus framework slimbus: Add support for 'clock-pause' feature dt-bindings: Add qcom slimbus controller bindings slimbus: qcom: Add Qualcomm Slimbus controller driver slimbus: qcom: Add runtime-pm support using clock-pause Srinivas Kandagatla (4): slimbus: core: add support to device tree helper regmap: add SLIMbus support slimbus: core: add common defines required for controllers MAINTAINERS: Add SLIMbus maintainer Documentation/devicetree/bindings/slimbus/bus.txt | 50 ++ .../devicetree/bindings/slimbus/slim-qcom-ctrl.txt | 39 ++ Documentation/driver-api/slimbus/index.rst | 15 + Documentation/driver-api/slimbus/summary.rst | 108 +++ MAINTAINERS | 8 + drivers/Kconfig | 2 + drivers/Makefile | 1 + drivers/base/regmap/Kconfig | 4 + drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-slimbus.c | 89 +++ drivers/slimbus/Kconfig | 22 + drivers/slimbus/Makefile | 9 + drivers/slimbus/core.c | 487 +++++++++++++ drivers/slimbus/messaging.c | 336 +++++++++ drivers/slimbus/qcom-ctrl.c | 756 +++++++++++++++++++++ drivers/slimbus/sched.c | 128 ++++ drivers/slimbus/slimbus.h | 268 ++++++++ include/linux/mod_devicetable.h | 13 + include/linux/regmap.h | 18 + include/linux/slimbus.h | 171 +++++ 20 files changed, 2525 insertions(+) create mode 100644 Documentation/devicetree/bindings/slimbus/bus.txt create mode 100644 Documentation/devicetree/bindings/slimbus/slim-qcom-ctrl.txt create mode 100644 Documentation/driver-api/slimbus/index.rst create mode 100644 Documentation/driver-api/slimbus/summary.rst create mode 100644 drivers/base/regmap/regmap-slimbus.c create mode 100644 drivers/slimbus/Kconfig create mode 100644 drivers/slimbus/Makefile create mode 100644 drivers/slimbus/core.c create mode 100644 drivers/slimbus/messaging.c create mode 100644 drivers/slimbus/qcom-ctrl.c create mode 100644 drivers/slimbus/sched.c create mode 100644 drivers/slimbus/slimbus.h create mode 100644 include/linux/slimbus.h -- 2.15.0

6 20

[alsa-devel] [GIT PULL]: Cannonlake audio firware
by Vinod Koul 05 Dec '17

05 Dec '17

Hi, Please PULL to recieve the Cannonlake audio firmware. The following changes since commit 17e6288135d4500f9fe60224dce2b46d850c346b: brcm: update firmware for bcm4358 (2017-11-25 10:15:53 -0500) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git cnl for you to fetch changes up to 89c62115f14dd8f414dfd2702913a6f1fcac3287: linux-firmware: intel: Add Cannonlake audio firmware (2017-12-04 09:04:57 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: intel: Add Cannonlake audio firmware WHENCE | 5 +++++ intel/dsp_fw_cnl.bin | 1 + intel/dsp_fw_cnl_v1191.bin | Bin 0 -> 583852 bytes 3 files changed, 6 insertions(+) create mode 120000 intel/dsp_fw_cnl.bin create mode 100644 intel/dsp_fw_cnl_v1191.bin -- ~Vinod

2 1

[alsa-devel] [PATCH 1/3] conf/ucm: Rearrange Makefile.am
by Takashi Iwai 05 Dec '17

05 Dec '17

Just rearrange and split to multiple lines to make future patches cleaner. No actual code change. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- configure.ac | 16 ++++++++-------- src/conf/ucm/Makefile.am | 16 +++++++++++++++- 2 files changed, 23 insertions(+), 9 deletions(-) diff --git a/configure.ac b/configure.ac index 22d65bb84882..83724e07459c 100644 --- a/configure.ac +++ b/configure.ac @@ -691,19 +691,20 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/cards/Makefile \ src/conf/pcm/Makefile \ src/conf/ucm/Makefile \ + src/conf/ucm/broadwell-rt286/Makefile \ + src/conf/ucm/broxton-rt298/Makefile \ + src/conf/ucm/bytcr-rt5651/Makefile \ + src/conf/ucm/chtrt5645/Makefile \ src/conf/ucm/DAISY-I2S/Makefile \ + src/conf/ucm/DB410c/Makefile \ + src/conf/ucm/GoogleNyan/Makefile \ src/conf/ucm/PandaBoard/Makefile \ src/conf/ucm/PandaBoardES/Makefile \ - src/conf/ucm/SDP4430/Makefile \ - src/conf/ucm/tegraalc5632/Makefile \ src/conf/ucm/PAZ00/Makefile \ - src/conf/ucm/GoogleNyan/Makefile \ - src/conf/ucm/broadwell-rt286/Makefile \ + src/conf/ucm/SDP4430/Makefile \ src/conf/ucm/skylake-rt286/Makefile \ + src/conf/ucm/tegraalc5632/Makefile \ src/conf/ucm/VEYRON-I2S/Makefile \ - src/conf/ucm/chtrt5645/Makefile \ - src/conf/ucm/DB410c/Makefile \ - src/conf/ucm/bytcr-rt5651/Makefile \ src/conf/topology/Makefile \ src/conf/topology/broadwell/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ @@ -711,7 +712,6 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/topology/bxtrt298/Makefile \ alsalisp/Makefile aserver/Makefile \ test/Makefile test/lsb/Makefile \ - src/conf/ucm/broxton-rt298/Makefile \ utils/Makefile utils/alsa-lib.spec utils/alsa.pc) dnl Create asoundlib.h dynamically according to configure options diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 2bd4ebe761ae..060d025f1bb5 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1,15 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 skylake-rt286 VEYRON-I2S chtrt5645 DB410c broxton-rt298 bytcr-rt5651 +SUBDIRS=\ +broadwell-rt286 \ +broxton-rt298 \ +bytcr-rt5651 \ +chtrt5645 \ +DAISY-I2S \ +DB410c \ +GoogleNyan \ +PandaBoard \ +PandaBoardES \ +PAZ00 \ +SDP4430 \ +skylake-rt286 \ +tegraalc5632 \ +VEYRON-I2S -- 2.15.0

1 2

[alsa-devel] [alsa-lib][PATCH 3/4] pcm: linear, route: handle linear formats with 20-bit sample on 4 bytes
by Maciej S. Szmigiero 05 Dec '17

05 Dec '17

The previous patch has added 20-bit PCM formats SND_PCM_FORMAT_{S,U}20 to alsa-lib. We need to extend the linear format conversion code with handling of these sample formats so they can also be utilized by applications that only recognize the more typical ones like SND_PCM_FORMAT_S16. Since the conversion arrays are indexed by a format bit width divided by 8 the easiest way to handle these formats is to treat them like they were 40-bit wide (the next free integer multiple of 8 bits). This doesn't create a collision risk with a future format since there can't really be a 40-bit sample format that occupies 4 bytes. Make sure we use the getput conversion method for these formats since a direct conversion from / to them is not supported. While we are at it let's also add asserts on a format bit width value in snd_pcm_linear_get_index() and snd_pcm_linear_put_index() received from snd_pcm_format_width() so we won't try to access memory outside a conversion array if for some reason this function had returned a value that is not in the expected range. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- src/pcm/pcm_linear.c | 16 +++++++++++++--- src/pcm/pcm_route.c | 6 ++++-- src/pcm/plugin_ops.h | 50 ++++++++++++++++++++++++++++++++++++++++++++++---- 3 files changed, 63 insertions(+), 9 deletions(-) diff --git a/src/pcm/pcm_linear.c b/src/pcm/pcm_linear.c index 0d61e927323f..e4973a308004 100644 --- a/src/pcm/pcm_linear.c +++ b/src/pcm/pcm_linear.c @@ -90,6 +90,7 @@ int snd_pcm_linear_get_index(snd_pcm_format_t src_format, snd_pcm_format_t dst_f endian = 0; pwidth = snd_pcm_format_physical_width(src_format); width = snd_pcm_format_width(src_format); + assert(width >= 8 && width <= 32); if (pwidth == 24) { switch (width) { case 24: @@ -100,8 +101,11 @@ int snd_pcm_linear_get_index(snd_pcm_format_t src_format, snd_pcm_format_t dst_f default: width = 2; break; } - return width * 4 + endian * 2 + sign + 16; + return width * 4 + endian * 2 + sign + 20; } else { + if (width == 20) + width = 40; + width = width / 8 - 1; return width * 4 + endian * 2 + sign; } @@ -121,6 +125,7 @@ int snd_pcm_linear_put_index(snd_pcm_format_t src_format, snd_pcm_format_t dst_f endian = 0; pwidth = snd_pcm_format_physical_width(dst_format); width = snd_pcm_format_width(dst_format); + assert(width >= 8 && width <= 32); if (pwidth == 24) { switch (width) { case 24: @@ -131,8 +136,11 @@ int snd_pcm_linear_put_index(snd_pcm_format_t src_format, snd_pcm_format_t dst_f default: width = 2; break; } - return width * 4 + endian * 2 + sign + 16; + return width * 4 + endian * 2 + sign + 20; } else { + if (width == 20) + width = 40; + width = width / 8 - 1; return width * 4 + endian * 2 + sign; } @@ -303,7 +311,9 @@ static int snd_pcm_linear_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) if (err < 0) return err; linear->use_getput = (snd_pcm_format_physical_width(format) == 24 || - snd_pcm_format_physical_width(linear->sformat) == 24); + snd_pcm_format_physical_width(linear->sformat) == 24 || + snd_pcm_format_width(format) == 20 || + snd_pcm_format_width(linear->sformat) == 20); if (linear->use_getput) { if (pcm->stream == SND_PCM_STREAM_PLAYBACK) { linear->get_idx = snd_pcm_linear_get_index(format, SND_PCM_FORMAT_S32); diff --git a/src/pcm/pcm_route.c b/src/pcm/pcm_route.c index 1f485a8e79f9..bbcc6118b593 100644 --- a/src/pcm/pcm_route.c +++ b/src/pcm/pcm_route.c @@ -565,10 +565,12 @@ static int snd_pcm_route_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) } if (err < 0) return err; - /* 3 bytes formats? */ + /* 3 bytes or 20-bit formats? */ route->params.use_getput = (snd_pcm_format_physical_width(src_format) + 7) / 8 == 3 || - (snd_pcm_format_physical_width(dst_format) + 7) / 8 == 3; + (snd_pcm_format_physical_width(dst_format) + 7) / 8 == 3 || + snd_pcm_format_width(src_format) == 20 || + snd_pcm_format_width(dst_format) == 20; route->params.get_idx = snd_pcm_linear_get_index(src_format, SND_PCM_FORMAT_S32); route->params.put_idx = snd_pcm_linear_put_index(SND_PCM_FORMAT_S32, dst_format); route->params.conv_idx = snd_pcm_linear_convert_index(src_format, dst_format); diff --git a/src/pcm/plugin_ops.h b/src/pcm/plugin_ops.h index 5a6016adb13a..a784d9c2b320 100644 --- a/src/pcm/plugin_ops.h +++ b/src/pcm/plugin_ops.h @@ -21,6 +21,12 @@ #ifndef SX_INLINES #define SX_INLINES +static inline uint32_t sx20(uint32_t x) +{ + if(x&0x00080000) + return x|0xFFF00000; + return x&0x000FFFFF; +} static inline uint32_t sx24(uint32_t x) { if(x&0x00800000) @@ -341,7 +347,7 @@ conv_1234_123C: as_u32(dst) = as_u32c(src) ^ 0x80; goto CONV_END; #ifdef GET16_LABELS /* src_wid src_endswap sign_toggle */ -static void *const get16_labels[4 * 2 * 2 + 4 * 3] = { +static void *const get16_labels[5 * 2 * 2 + 4 * 3] = { &&get16_1_10, /* 8h -> 16h */ &&get16_1_90, /* 8h ^> 16h */ &&get16_1_10, /* 8s -> 16h */ @@ -350,6 +356,7 @@ static void *const get16_labels[4 * 2 * 2 + 4 * 3] = { &&get16_12_92, /* 16h ^> 16h */ &&get16_12_21, /* 16s -> 16h */ &&get16_12_A1, /* 16s ^> 16h */ + /* 4 byte formats */ &&get16_0123_12, /* 24h -> 16h */ &&get16_0123_92, /* 24h ^> 16h */ &&get16_1230_32, /* 24s -> 16h */ @@ -358,6 +365,10 @@ static void *const get16_labels[4 * 2 * 2 + 4 * 3] = { &&get16_1234_92, /* 32h ^> 16h */ &&get16_1234_43, /* 32s -> 16h */ &&get16_1234_C3, /* 32s ^> 16h */ + &&get16_0123_12_20, /* 20h -> 16h */ + &&get16_0123_92_20, /* 20h ^> 16h */ + &&get16_1230_32_20, /* 20s -> 16h */ + &&get16_1230_B2_20, /* 20s ^> 16h */ /* 3bytes format */ &&get16_123_12, /* 24h -> 16h */ &&get16_123_92, /* 24h ^> 16h */ @@ -390,6 +401,10 @@ get16_1234_12: sample = as_u32c(src) >> 16; goto GET16_END; get16_1234_92: sample = (as_u32c(src) >> 16) ^ 0x8000; goto GET16_END; get16_1234_43: sample = bswap_16(as_u32c(src)); goto GET16_END; get16_1234_C3: sample = bswap_16(as_u32c(src) ^ 0x80); goto GET16_END; +get16_0123_12_20: sample = as_u32c(src) >> 4; goto GET16_END; +get16_0123_92_20: sample = (as_u32c(src) >> 4) ^ 0x8000; goto GET16_END; +get16_1230_32_20: sample = bswap_32(as_u32c(src)) >> 4; goto GET16_END; +get16_1230_B2_20: sample = (bswap_32(as_u32c(src)) >> 4) ^ 0x8000; goto GET16_END; get16_123_12: sample = _get_triple(src) >> 8; goto GET16_END; get16_123_92: sample = (_get_triple(src) >> 8) ^ 0x8000; goto GET16_END; get16_123_32: sample = _get_triple_s(src) >> 8; goto GET16_END; @@ -407,7 +422,7 @@ get16_123_B2_18: sample = (_get_triple_s(src) >> 2) ^ 0x8000; goto GET16_END; #ifdef PUT16_LABELS /* dst_wid dst_endswap sign_toggle */ -static void *const put16_labels[4 * 2 * 2 + 4 * 3] = { +static void *const put16_labels[5 * 2 * 2 + 4 * 3] = { &&put16_12_1, /* 16h -> 8h */ &&put16_12_9, /* 16h ^> 8h */ &&put16_12_1, /* 16h -> 8s */ @@ -416,6 +431,7 @@ static void *const put16_labels[4 * 2 * 2 + 4 * 3] = { &&put16_12_92, /* 16h ^> 16h */ &&put16_12_21, /* 16h -> 16s */ &&put16_12_29, /* 16h ^> 16s */ + /* 4 byte formats */ &&put16_12_0120, /* 16h -> 24h */ &&put16_12_0920, /* 16h ^> 24h */ &&put16_12_0210, /* 16h -> 24s */ @@ -424,6 +440,10 @@ static void *const put16_labels[4 * 2 * 2 + 4 * 3] = { &&put16_12_9200, /* 16h ^> 32h */ &&put16_12_0021, /* 16h -> 32s */ &&put16_12_0029, /* 16h ^> 32s */ + &&put16_12_0120_20, /* 16h -> 20h */ + &&put16_12_0920_20, /* 16h ^> 20h */ + &&put16_12_0210_20, /* 16h -> 20s */ + &&put16_12_0290_20, /* 16h ^> 20s */ /* 3bytes format */ &&put16_12_120, /* 16h -> 24h */ &&put16_12_920, /* 16h ^> 24h */ @@ -456,6 +476,10 @@ put16_12_1200: as_u32(dst) = (uint32_t)sample << 16; goto PUT16_END; put16_12_9200: as_u32(dst) = (uint32_t)(sample ^ 0x8000) << 16; goto PUT16_END; put16_12_0021: as_u32(dst) = (uint32_t)bswap_16(sample); goto PUT16_END; put16_12_0029: as_u32(dst) = (uint32_t)bswap_16(sample) ^ 0x80; goto PUT16_END; +put16_12_0120_20: as_u32(dst) = sx20((uint32_t)sample << 4); goto PUT16_END; +put16_12_0920_20: as_u32(dst) = sx20((uint32_t)(sample ^ 0x8000) << 4); goto PUT16_END; +put16_12_0210_20: as_u32(dst) = bswap_32(sx20((uint32_t)sample << 4)); goto PUT16_END; +put16_12_0290_20: as_u32(dst) = bswap_32(sx20((uint32_t)(sample ^ 0x8000) << 4)); goto PUT16_END; put16_12_120: _put_triple(dst, (uint32_t)sample << 8); goto PUT16_END; put16_12_920: _put_triple(dst, (uint32_t)(sample ^ 0x8000) << 8); goto PUT16_END; put16_12_021: _put_triple_s(dst, (uint32_t)sample << 8); goto PUT16_END; @@ -478,7 +502,7 @@ put16_12_029_18: _put_triple_s(dst, (uint32_t)(sample ^ 0x8000) << 2); goto PUT1 #ifdef GET32_LABELS /* src_wid src_endswap sign_toggle */ -static void *const get32_labels[4 * 2 * 2 + 4 * 3] = { +static void *const get32_labels[5 * 2 * 2 + 4 * 3] = { &&get32_1_1000, /* 8h -> 32h */ &&get32_1_9000, /* 8h ^> 32h */ &&get32_1_1000, /* 8s -> 32h */ @@ -487,6 +511,7 @@ static void *const get32_labels[4 * 2 * 2 + 4 * 3] = { &&get32_12_9200, /* 16h ^> 32h */ &&get32_12_2100, /* 16s -> 32h */ &&get32_12_A100, /* 16s ^> 32h */ + /* 4 byte formats */ &&get32_0123_1230, /* 24h -> 32h */ &&get32_0123_9230, /* 24h ^> 32h */ &&get32_1230_3210, /* 24s -> 32h */ @@ -495,6 +520,10 @@ static void *const get32_labels[4 * 2 * 2 + 4 * 3] = { &&get32_1234_9234, /* 32h ^> 32h */ &&get32_1234_4321, /* 32s -> 32h */ &&get32_1234_C321, /* 32s ^> 32h */ + &&get32_0123_1230_20, /* 20h -> 32h */ + &&get32_0123_9230_20, /* 20h ^> 32h */ + &&get32_1230_3210_20, /* 20s -> 32h */ + &&get32_1230_B210_20, /* 20s ^> 32h */ /* 3bytes format */ &&get32_123_1230, /* 24h -> 32h */ &&get32_123_9230, /* 24h ^> 32h */ @@ -531,6 +560,10 @@ get32_1234_1234: sample = as_u32c(src); goto GET32_END; get32_1234_9234: sample = as_u32c(src) ^ 0x80000000; goto GET32_END; get32_1234_4321: sample = bswap_32(as_u32c(src)); goto GET32_END; get32_1234_C321: sample = bswap_32(as_u32c(src) ^ 0x80); goto GET32_END; +get32_0123_1230_20: sample = as_u32c(src) << 12; goto GET32_END; +get32_0123_9230_20: sample = (as_u32c(src) << 12) ^ 0x80000000; goto GET32_END; +get32_1230_3210_20: sample = bswap_32(as_u32c(src)) << 12; goto GET32_END; +get32_1230_B210_20: sample = (bswap_32(as_u32c(src)) << 12) ^ 0x80000000; goto GET32_END; get32_123_1230: sample = _get_triple(src) << 8; goto GET32_END; get32_123_9230: sample = (_get_triple(src) << 8) ^ 0x80000000; goto GET32_END; get32_123_3210: sample = _get_triple_s(src) << 8; goto GET32_END; @@ -553,7 +586,7 @@ __conv24_get: goto *put; #ifdef PUT32_LABELS /* dst_wid dst_endswap sign_toggle */ -static void *const put32_labels[4 * 2 * 2 + 4 * 3] = { +static void *const put32_labels[5 * 2 * 2 + 4 * 3] = { &&put32_1234_1, /* 32h -> 8h */ &&put32_1234_9, /* 32h ^> 8h */ &&put32_1234_1, /* 32h -> 8s */ @@ -562,6 +595,7 @@ static void *const put32_labels[4 * 2 * 2 + 4 * 3] = { &&put32_1234_92, /* 32h ^> 16h */ &&put32_1234_21, /* 32h -> 16s */ &&put32_1234_29, /* 32h ^> 16s */ + /* 4 byte formats */ &&put32_1234_0123, /* 32h -> 24h */ &&put32_1234_0923, /* 32h ^> 24h */ &&put32_1234_3210, /* 32h -> 24s */ @@ -570,6 +604,10 @@ static void *const put32_labels[4 * 2 * 2 + 4 * 3] = { &&put32_1234_9234, /* 32h ^> 32h */ &&put32_1234_4321, /* 32h -> 32s */ &&put32_1234_4329, /* 32h ^> 32s */ + &&put32_1234_0123_20, /* 32h -> 20h */ + &&put32_1234_0923_20, /* 32h ^> 20h */ + &&put32_1234_3210_20, /* 32h -> 20s */ + &&put32_1234_3290_20, /* 32h ^> 20s */ /* 3bytes format */ &&put32_1234_123, /* 32h -> 24h */ &&put32_1234_923, /* 32h ^> 24h */ @@ -607,6 +645,10 @@ put32_1234_1234: as_u32(dst) = sample; goto PUT32_END; put32_1234_9234: as_u32(dst) = sample ^ 0x80000000; goto PUT32_END; put32_1234_4321: as_u32(dst) = bswap_32(sample); goto PUT32_END; put32_1234_4329: as_u32(dst) = bswap_32(sample) ^ 0x80; goto PUT32_END; +put32_1234_0123_20: as_u32(dst) = sx20(sample >> 12); goto PUT32_END; +put32_1234_0923_20: as_u32(dst) = sx20((sample ^ 0x80000000) >> 12); goto PUT32_END; +put32_1234_3210_20: as_u32(dst) = bswap_32(sx20(sample >> 12)); goto PUT32_END; +put32_1234_3290_20: as_u32(dst) = bswap_32(sx20((sample ^ 0x80000000) >> 12)); goto PUT32_END; put32_1234_123: _put_triple(dst, sample >> 8); goto PUT32_END; put32_1234_923: _put_triple(dst, (sample ^ 0x80000000) >> 8); goto PUT32_END; put32_1234_321: _put_triple_s(dst, sample >> 8); goto PUT32_END;

3 6

[alsa-devel] [PATCH] [v2] ASoC: atmel-classd: improve Kconfig dependencies
by Arnd Bergmann 05 Dec '17

05 Dec '17

Commit 061981ff8cc8 ("ASoC: atmel: properly select dma driver state") changed the way that the dependencies are handled, but then the Class D amplifier support got merged, which used the old method. This seems to have triggered a very rare randconfig condition for me now, leading to a link error: sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': atmel_ssc_dai.c:(.text+0x79c): undefined reference to `atmel_pcm_dma_platform_register' atmel_ssc_dai.c:(.text+0x79c): relocation truncated to fit: R_AARCH64_CALL26 against undefined symbol `atmel_pcm_dma_platform_register' sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_put_audio': atmel_ssc_dai.c:(.text+0xf24): undefined reference to `atmel_pcm_dma_platform_unregister' atmel_ssc_dai.c:(.text+0xf24): relocation truncated to fit: R_AARCH64_CALL26 against undefined symbol `atmel_pcm_dma_platform_unregister' This changes the SND_ATMEL_SOC_DMA option so it gets built whenever one of the two users (SND_ATMEL_SOC_SSC_DMA and SND_ATMEL_SOC_CLASSD) is enabled, with the twist that we make it built-in when SND_ATMEL_SOC_SSC is built-in, even if one of the other two is a module. I think this captures best what the real dependency in the hardware is, and it simplifies the logic a bit. Fixes: e0a25b6d1862 ("ASoC: atmel-classd: add the Audio Class D Amplifier") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- v2: rewrite the change completely, after running into a new regression. I'm putting it out on the list so we can review the change together, but I'm still doing more testing at the same time. If I don't reply by Dec 6, nothing new has come up. --- sound/soc/atmel/Kconfig | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 4a56f3dfba51..af9a9096d50b 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -20,8 +20,8 @@ config SND_ATMEL_SOC_SSC_PDC config SND_ATMEL_SOC_DMA tristate select SND_SOC_GENERIC_DMAENGINE_PCM - default m if SND_ATMEL_SOC_SSC_DMA=m && SND_ATMEL_SOC_SSC=m - default y if SND_ATMEL_SOC_SSC_DMA=y || (SND_ATMEL_SOC_SSC_DMA=m && SND_ATMEL_SOC_SSC=y) + default y if SND_ATMEL_SOC_SSC=y && (SND_ATMEL_SOC_SSC_DMA=m || SND_ATMEL_SOC_CLASSD=m) + default SND_ATMEL_SOC_CLASSD || SND_ATMEL_SOC_SSC_DMA config SND_ATMEL_SOC_SSC_DMA tristate @@ -64,7 +64,6 @@ config SND_AT91_SOC_SAM9X5_WM8731 config SND_ATMEL_SOC_CLASSD tristate "Atmel ASoC driver for boards using CLASSD" depends on ARCH_AT91 || COMPILE_TEST - select SND_ATMEL_SOC_DMA select REGMAP_MMIO help Say Y if you want to add support for Atmel ASoC driver for boards using -- 2.9.0

2 1

Re: [alsa-devel] [PATCH 5/8] ASoC: uniphier: add support for UniPhier AIO driver
by Mark Brown 05 Dec '17

05 Dec '17

On Tue, Dec 05, 2017 at 01:48:39PM +0900, Katsuhiro Suzuki wrote: Please fix your mail client to word wrap within paragraphs at something substantially less than 80 columns. Doing this makes your messages much easier to read and reply to. > > Is there a mux in the SoC here? > Sorry for confusing, It's not mux. > uniphier_srcport_reset() resets HW SRC (sampling rate converter) block. > Audio data out ports of UniPhier audio system have HW SRC. Is the SRC just a single block sitting between the DMA and the external audio port or is there more going on? Some of the other code made me think the hardware was more flexible than this (all the writing to registers with names like RXSEL for example). > > > +#endif /* CONFIG_SND_SOC_UNIPHIER_LD11 */ > > Why is there an ifdef here? There's no other conditional code in here, > > it seems pointless. > This config is used to support or not LD11 SoC. > aio-ld11.c is not build and 'uniphier_aio_ldxx_spec' is undefined if this config is disabled. > > aio-ld11.c defines SoC dependent resources (port, HW ring buffer, DMA ch, etc.) > and fixed settings. > I know it's better to move such information into device-tree, but register areas of > UniPhier's audio system is very strange and interleaved. It's hard to split each nodes... I'd expect this code to be structured more like a library - have a driver that handles the specific IPs then have it call into a shared block of code that does the generic bits. Though in this case the device specific bit looks like a couple of tiny data tables so I'm not sure it's worth making it conditional or separate at all. > > This looks awfully like compressed audio support... should there be > > integration with the compressed audio API/ > Thanks, I'll try it. Is there Documentation in sound/designes/compress-offload.rst? > And best sample is... Intel's driver? Yes. > (Summary) > I think I should fix as follows: > - Split DMA, DAI patches from large one > - Validate parameters in hw_params > - Add description about HW SRC (or fix bad name 'srcport') > - Add comments about uniphier_aiodma_irq() > - Expose clocking and audio routing to userspace, or at the very > least machine driver configuration > - Support compress-audio API for S/PDIF > and post V2. At least. I do think we need to get to the bottom of how flexible the hardware is first though.

1 0

[alsa-devel] [alsa-lib][PATCH 2/4] pcm: add and describe SND_PCM_FORMAT_{S, U}20
by Maciej S. Szmigiero 05 Dec '17

05 Dec '17

This patch adds and describes in various functions that query format properties SND_PCM_FORMAT_{S,U}20 formats that were recently added to the kernel as SNDRV_PCM_FORMAT_{S,U}20. These formats are similar to existing 20-bit PCM formats SND_PCM_FORMAT_{S,U}20_3, however they occupy 4 bytes instead of 3. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- include/pcm.h | 20 ++++++++++++++++++-- src/pcm/pcm.c | 10 ++++++++++ src/pcm/pcm_local.h | 4 ++++ src/pcm/pcm_misc.c | 41 ++++++++++++++++++++++++++++++++++++++--- 4 files changed, 70 insertions(+), 5 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index e05777a7c221..5939c1dab37d 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -175,6 +175,14 @@ typedef enum _snd_pcm_format { SND_PCM_FORMAT_MPEG, /** GSM */ SND_PCM_FORMAT_GSM, + /** Signed 20bit Little Endian in 4bytes format, LSB justified */ + SND_PCM_FORMAT_S20_LE, + /** Signed 20bit Big Endian in 4bytes format, LSB justified */ + SND_PCM_FORMAT_S20_BE, + /** Unsigned 20bit Little Endian in 4bytes format, LSB justified */ + SND_PCM_FORMAT_U20_LE, + /** Unsigned 20bit Big Endian in 4bytes format, LSB justified */ + SND_PCM_FORMAT_U20_BE, /** Special */ SND_PCM_FORMAT_SPECIAL = 31, /** Signed 24bit Little Endian in 3bytes format */ @@ -239,7 +247,11 @@ typedef enum _snd_pcm_format { /** Float 64 bit CPU endian */ SND_PCM_FORMAT_FLOAT64 = SND_PCM_FORMAT_FLOAT64_LE, /** IEC-958 CPU Endian */ - SND_PCM_FORMAT_IEC958_SUBFRAME = SND_PCM_FORMAT_IEC958_SUBFRAME_LE + SND_PCM_FORMAT_IEC958_SUBFRAME = SND_PCM_FORMAT_IEC958_SUBFRAME_LE, + /** Signed 20bit in 4bytes format, LSB justified, CPU Endian */ + SND_PCM_FORMAT_S20 = SND_PCM_FORMAT_S20_LE, + /** Unsigned 20bit in 4bytes format, LSB justified, CPU Endian */ + SND_PCM_FORMAT_U20 = SND_PCM_FORMAT_U20_LE #elif __BYTE_ORDER == __BIG_ENDIAN /** Signed 16 bit CPU endian */ SND_PCM_FORMAT_S16 = SND_PCM_FORMAT_S16_BE, @@ -258,7 +270,11 @@ typedef enum _snd_pcm_format { /** Float 64 bit CPU endian */ SND_PCM_FORMAT_FLOAT64 = SND_PCM_FORMAT_FLOAT64_BE, /** IEC-958 CPU Endian */ - SND_PCM_FORMAT_IEC958_SUBFRAME = SND_PCM_FORMAT_IEC958_SUBFRAME_BE + SND_PCM_FORMAT_IEC958_SUBFRAME = SND_PCM_FORMAT_IEC958_SUBFRAME_BE, + /** Signed 20bit in 4bytes format, LSB justified, CPU Endian */ + SND_PCM_FORMAT_S20 = SND_PCM_FORMAT_S20_BE, + /** Unsigned 20bit in 4bytes format, LSB justified, CPU Endian */ + SND_PCM_FORMAT_U20 = SND_PCM_FORMAT_U20_BE #else #error "Unknown endian" #endif diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 1efb8b9d4e8c..e9ebf383c31b 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1780,6 +1780,10 @@ static const char *const snd_pcm_format_names[] = { FORMAT(IMA_ADPCM), FORMAT(MPEG), FORMAT(GSM), + FORMAT(S20_LE), + FORMAT(S20_BE), + FORMAT(U20_LE), + FORMAT(U20_BE), FORMAT(SPECIAL), FORMAT(S24_3LE), FORMAT(S24_3BE), @@ -1814,6 +1818,8 @@ static const char *const snd_pcm_format_aliases[SND_PCM_FORMAT_LAST+1] = { FORMAT(FLOAT), FORMAT(FLOAT64), FORMAT(IEC958_SUBFRAME), + FORMAT(S20), + FORMAT(U20), }; static const char *const snd_pcm_format_descriptions[] = { @@ -1842,6 +1848,10 @@ static const char *const snd_pcm_format_descriptions[] = { FORMATD(IMA_ADPCM, "Ima-ADPCM"), FORMATD(MPEG, "MPEG"), FORMATD(GSM, "GSM"), + FORMATD(S20_LE, "Signed 20 bit Little Endian in 4 bytes, LSB justified"), + FORMATD(S20_BE, "Signed 20 bit Big Endian in 4 bytes, LSB justified"), + FORMATD(U20_LE, "Unsigned 20 bit Little Endian in 4 bytes, LSB justified"), + FORMATD(U20_BE, "Unsigned 20 bit Big Endian in 4 bytes, LSB justified"), FORMATD(SPECIAL, "Special"), FORMATD(S24_3LE, "Signed 24 bit Little Endian in 3bytes"), FORMATD(S24_3BE, "Signed 24 bit Big Endian in 3bytes"), diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 776f8cd3e302..3d95e1749169 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -984,6 +984,10 @@ const snd_config_t *snd_pcm_rate_get_default_converter(snd_config_t *root); (1U << SND_PCM_FORMAT_S16_BE) | \ (1U << SND_PCM_FORMAT_U16_LE) | \ (1U << SND_PCM_FORMAT_U16_BE) | \ + (1U << SND_PCM_FORMAT_S20_LE) | \ + (1U << SND_PCM_FORMAT_S20_BE) | \ + (1U << SND_PCM_FORMAT_U20_LE) | \ + (1U << SND_PCM_FORMAT_U20_BE) | \ (1U << SND_PCM_FORMAT_S24_LE) | \ (1U << SND_PCM_FORMAT_S24_BE) | \ (1U << SND_PCM_FORMAT_U24_LE) | \ diff --git a/src/pcm/pcm_misc.c b/src/pcm/pcm_misc.c index 5420b1895713..7ee0dac771af 100644 --- a/src/pcm/pcm_misc.c +++ b/src/pcm/pcm_misc.c @@ -38,6 +38,8 @@ int snd_pcm_format_signed(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_S8: case SNDRV_PCM_FORMAT_S16_LE: case SNDRV_PCM_FORMAT_S16_BE: + case SNDRV_PCM_FORMAT_S20_LE: + case SNDRV_PCM_FORMAT_S20_BE: case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_S24_BE: case SNDRV_PCM_FORMAT_S32_LE: @@ -52,6 +54,8 @@ int snd_pcm_format_signed(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U8: case SNDRV_PCM_FORMAT_U16_LE: case SNDRV_PCM_FORMAT_U16_BE: + case SNDRV_PCM_FORMAT_U20_LE: + case SNDRV_PCM_FORMAT_U20_BE: case SNDRV_PCM_FORMAT_U24_LE: case SNDRV_PCM_FORMAT_U24_BE: case SNDRV_PCM_FORMAT_U32_LE: @@ -126,6 +130,8 @@ int snd_pcm_format_little_endian(snd_pcm_format_t format) switch (format) { case SNDRV_PCM_FORMAT_S16_LE: case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S20_LE: + case SNDRV_PCM_FORMAT_U20_LE: case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_U24_LE: case SNDRV_PCM_FORMAT_S32_LE: @@ -144,6 +150,8 @@ int snd_pcm_format_little_endian(snd_pcm_format_t format) return 1; case SNDRV_PCM_FORMAT_S16_BE: case SNDRV_PCM_FORMAT_U16_BE: + case SNDRV_PCM_FORMAT_S20_BE: + case SNDRV_PCM_FORMAT_U20_BE: case SNDRV_PCM_FORMAT_S24_BE: case SNDRV_PCM_FORMAT_U24_BE: case SNDRV_PCM_FORMAT_S32_BE: @@ -218,6 +226,10 @@ int snd_pcm_format_width(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U18_3LE: case SNDRV_PCM_FORMAT_U18_3BE: return 18; + case SNDRV_PCM_FORMAT_S20_LE: + case SNDRV_PCM_FORMAT_S20_BE: + case SNDRV_PCM_FORMAT_U20_LE: + case SNDRV_PCM_FORMAT_U20_BE: case SNDRV_PCM_FORMAT_S20_3LE: case SNDRV_PCM_FORMAT_S20_3BE: case SNDRV_PCM_FORMAT_U20_3LE: @@ -289,6 +301,10 @@ int snd_pcm_format_physical_width(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U24_3LE: case SNDRV_PCM_FORMAT_U24_3BE: return 24; + case SNDRV_PCM_FORMAT_S20_LE: + case SNDRV_PCM_FORMAT_S20_BE: + case SNDRV_PCM_FORMAT_U20_LE: + case SNDRV_PCM_FORMAT_U20_BE: case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_S24_BE: case SNDRV_PCM_FORMAT_U24_LE: @@ -350,6 +366,10 @@ ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples) case SNDRV_PCM_FORMAT_U24_3LE: case SNDRV_PCM_FORMAT_U24_3BE: return samples * 3; + case SNDRV_PCM_FORMAT_S20_LE: + case SNDRV_PCM_FORMAT_S20_BE: + case SNDRV_PCM_FORMAT_U20_LE: + case SNDRV_PCM_FORMAT_U20_BE: case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_S24_BE: case SNDRV_PCM_FORMAT_U24_LE: @@ -393,6 +413,8 @@ uint64_t snd_pcm_format_silence_64(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_S8: case SNDRV_PCM_FORMAT_S16_LE: case SNDRV_PCM_FORMAT_S16_BE: + case SNDRV_PCM_FORMAT_S20_LE: + case SNDRV_PCM_FORMAT_S20_BE: case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_S24_BE: case SNDRV_PCM_FORMAT_S32_LE: @@ -415,12 +437,16 @@ uint64_t snd_pcm_format_silence_64(snd_pcm_format_t format) #ifdef SNDRV_LITTLE_ENDIAN case SNDRV_PCM_FORMAT_U16_LE: return 0x8000800080008000ULL; + case SNDRV_PCM_FORMAT_U20_LE: + return 0x0008000000080000ULL; case SNDRV_PCM_FORMAT_U24_LE: return 0x0080000000800000ULL; case SNDRV_PCM_FORMAT_U32_LE: return 0x8000000080000000ULL; case SNDRV_PCM_FORMAT_U16_BE: return 0x0080008000800080ULL; + case SNDRV_PCM_FORMAT_U20_BE: + return 0x0000080000000800ULL; case SNDRV_PCM_FORMAT_U24_BE: return 0x0000800000008000ULL; case SNDRV_PCM_FORMAT_U32_BE: @@ -440,12 +466,16 @@ uint64_t snd_pcm_format_silence_64(snd_pcm_format_t format) #else case SNDRV_PCM_FORMAT_U16_LE: return 0x0080008000800080ULL; + case SNDRV_PCM_FORMAT_U20_LE: + return 0x0000080000000800ULL; case SNDRV_PCM_FORMAT_U24_LE: return 0x0000800000008000ULL; case SNDRV_PCM_FORMAT_U32_LE: return 0x0000008000000080ULL; case SNDRV_PCM_FORMAT_U16_BE: return 0x8000800080008000ULL; + case SNDRV_PCM_FORMAT_U20_BE: + return 0x0008000000080000ULL; case SNDRV_PCM_FORMAT_U24_BE: return 0x0080000000800000ULL; case SNDRV_PCM_FORMAT_U32_BE: @@ -653,11 +683,13 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int return 0; } -static const int linear_formats[4][2][2] = { +static const int linear_formats[5][2][2] = { { { SNDRV_PCM_FORMAT_S8, SNDRV_PCM_FORMAT_S8 }, { SNDRV_PCM_FORMAT_U8, SNDRV_PCM_FORMAT_U8 } }, { { SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_FORMAT_S16_BE }, { SNDRV_PCM_FORMAT_U16_LE, SNDRV_PCM_FORMAT_U16_BE } }, + { { SNDRV_PCM_FORMAT_S20_LE, SNDRV_PCM_FORMAT_S20_BE }, + { SNDRV_PCM_FORMAT_U20_LE, SNDRV_PCM_FORMAT_U20_BE } }, { { SNDRV_PCM_FORMAT_S24_LE, SNDRV_PCM_FORMAT_S24_BE }, { SNDRV_PCM_FORMAT_U24_LE, SNDRV_PCM_FORMAT_U24_BE } }, { { SNDRV_PCM_FORMAT_S32_LE, SNDRV_PCM_FORMAT_S32_BE }, @@ -706,12 +738,15 @@ snd_pcm_format_t snd_pcm_build_linear_format(int width, int pwidth, int unsignd, case 16: width = 1; break; - case 24: + case 20: width = 2; break; - case 32: + case 24: width = 3; break; + case 32: + width = 4; + break; default: return SND_PCM_FORMAT_UNKNOWN; }

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[alsa-devel] usx2y: "should not be here with counts=40" then hard lockup (linux 4.9.0)
by Jaime T 05 Dec '17

05 Dec '17

Hi all. I'm running a fresh and cleanly-installed debian stable (stretch) with my tascam us-122. After several hours of perfect sound, the kernel (4.9.0-2-686) emits: Dec 4 22:30:20 piano kernel: [14707.140766] should not be here with counts=40 Dec 4 22:30:24 piano jackd[793]: ALSA: poll time out, polled for 3999073 usecs Dec 4 22:30:24 piano jackd[793]: JackAudioDriver::ProcessAsync: read error, stopping... and my jack server stops working, followed soon after by a hard lockup. I've run alsa-info.sh and the results are available at http://www.alsa-project.org/db/?f=e2385306145a37956360def8fe06e8c0f951839d Should I log a bug at https://bugzilla.kernel.org/ ? Many thanks, Jaime

1 0

[alsa-devel] New codec support for ALC257
by Kailang 05 Dec '17

05 Dec '17

Hi Takashi, This patch was support new codec for ALC257. BR, Kailang

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[alsa-devel] [PATCH v6 00/13] Add STM32 DFSDM support
by Arnaud Pouliquen 04 Dec '17

04 Dec '17

Hello, Here is a new version to fix kbuild test warnings Main deltas V6 vs V5: - Fix warning reported by kbuild test in : include/linux/iio/consumer.h sound/soc/stm/stm32_adfsdm.c Main deltas V5 vs V4: - Integrate ASOC DAI as a subnode of the DFSDM. - Add in kernel consumer interface to allow to manipulate attribute. Context reminder: ----------------- DFSDM peripheral is a peripheral that allows to connect some sigma delta ADCs or PDM microphones via a SPI or Manchester bus. DFSDM integrates digital filters to offer up to 24 bits final resolution. In term of SW architecture. 2 use-cases have to be supported: 1) Sigma delta ADC conversion through IIO framework. Sigma delta ADC is handled by generic sigma delta modulator driver. DFSDM peripheral is binded to a SD modulator ADC using the IIO HW consumer interface. Please notice that IIO HW consumer interface has be proposed by Lars, but is part of this patchset with Lars's agreement. User interface is IIO one. Notice that this patch-set propose only a raw conversion, to simplify review. Buffer and trigger management will be added in next patch-sets. 2) PDM microphone record through ALSA framework. PDM microphone is handled by ASOC Generic DMIC codec driver. ADFSDM ASOC DAI driver is binded to IIO driver using the IIO consumer interface ADFSDM ASOC DAI driver is binded to a PDM microphone ASOC component using ASOC Of_graph. User interface is ALSA one. As IIO DMA management is not adapted to an audio realtime stream. A specific DMA management has been implemented in IIO driver for audio purposes. Regards, Arnaud Arnaud Pouliquen (12): docs: driver-api: add iio hw consumer section IIO: hw_consumer: add devm_iio_hw_consumer_alloc IIO: inkern: API for manipulating channel attributes IIO: Add DT bindings for sigma delta adc modulator IIO: ADC: add sigma delta modulator support IIO: add DT bindings for stm32 DFSDM filter IIO: ADC: add stm32 DFSDM core support IIO: ADC: add STM32 DFSDM sigma delta ADC support IIO: ADC: add stm32 DFSDM support for PDM microphone IIO: consumer: allow to set buffer sizes ASoC: add bindings for stm32 DFSDM filter ASoC: stm32: add DFSDM DAI support Lars-Peter Clausen (1): iio: Add hardware consumer buffer support .../ABI/testing/sysfs-bus-iio-dfsdm-adc-stm32 | 16 + .../bindings/iio/adc/sigma-delta-modulator.txt | 13 + .../bindings/iio/adc/st,stm32-dfsdm-adc.txt | 127 ++ .../devicetree/bindings/sound/st,stm32-adfsdm.txt | 62 + Documentation/driver-api/iio/hw-consumer.rst | 51 + Documentation/driver-api/iio/index.rst | 1 + drivers/iio/adc/Kconfig | 37 + drivers/iio/adc/Makefile | 3 + drivers/iio/adc/sd_adc_modulator.c | 81 ++ drivers/iio/adc/stm32-dfsdm-adc.c | 1232 ++++++++++++++++++++ drivers/iio/adc/stm32-dfsdm-core.c | 318 +++++ drivers/iio/adc/stm32-dfsdm.h | 319 +++++ drivers/iio/buffer/Kconfig | 10 + drivers/iio/buffer/Makefile | 1 + drivers/iio/buffer/industrialio-buffer-cb.c | 11 + drivers/iio/buffer/industrialio-hw-consumer.c | 248 ++++ drivers/iio/inkern.c | 18 +- include/linux/iio/adc/stm32-dfsdm-adc.h | 28 + include/linux/iio/consumer.h | 39 +- include/linux/iio/hw-consumer.h | 22 + sound/soc/stm/Kconfig | 11 + sound/soc/stm/Makefile | 3 + sound/soc/stm/stm32_adfsdm.c | 386 ++++++ 23 files changed, 3031 insertions(+), 6 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-iio-dfsdm-adc-stm32 create mode 100644 Documentation/devicetree/bindings/iio/adc/sigma-delta-modulator.txt create mode 100644 Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt create mode 100644 Documentation/driver-api/iio/hw-consumer.rst create mode 100644 drivers/iio/adc/sd_adc_modulator.c create mode 100644 drivers/iio/adc/stm32-dfsdm-adc.c create mode 100644 drivers/iio/adc/stm32-dfsdm-core.c create mode 100644 drivers/iio/adc/stm32-dfsdm.h create mode 100644 drivers/iio/buffer/industrialio-hw-consumer.c create mode 100644 include/linux/iio/adc/stm32-dfsdm-adc.h create mode 100644 include/linux/iio/hw-consumer.h create mode 100644 sound/soc/stm/stm32_adfsdm.c -- 2.7.4

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[alsa-devel] [PATCH] ASoC: amd: added error checks in dma driver
by Vijendar Mukunda 04 Dec '17

04 Dec '17

added additional error checks in acp dma driver v2: printed error codes for acp init & acp deinit failure cases. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 35 ++++++++++++++++++++++++++++++----- 1 file changed, 30 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index fb09578..29be517 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -848,6 +848,9 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + if (!rtd) + return -EINVAL; + buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd->acp_mmio, substream->stream); @@ -873,6 +876,8 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + if (!rtd) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { config_acp_dma_channel(rtd->acp_mmio, SYSRAM_TO_ACP_CH_NUM, PLAYBACK_START_DMA_DESCR_CH12, @@ -1082,7 +1087,11 @@ static int acp_audio_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, audio_drv_data); /* Initialize the ACP */ - acp_init(audio_drv_data->acp_mmio, audio_drv_data->asic_type); + status = acp_init(audio_drv_data->acp_mmio, audio_drv_data->asic_type); + if (status) { + dev_err(&pdev->dev, "ACP Init failed status:%d\n", status); + return status; + } status = snd_soc_register_platform(&pdev->dev, &acp_asoc_platform); if (status != 0) { @@ -1099,9 +1108,12 @@ static int acp_audio_probe(struct platform_device *pdev) static int acp_audio_remove(struct platform_device *pdev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(&pdev->dev); - acp_deinit(adata->acp_mmio); + status = acp_deinit(adata->acp_mmio); + if (status) + dev_err(&pdev->dev, "ACP Deinit failed status:%d\n", status); snd_soc_unregister_platform(&pdev->dev); pm_runtime_disable(&pdev->dev); @@ -1111,9 +1123,14 @@ static int acp_audio_remove(struct platform_device *pdev) static int acp_pcm_resume(struct device *dev) { u16 bank; + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_init(adata->acp_mmio, adata->asic_type); + status = acp_init(adata->acp_mmio, adata->asic_type); + if (status) { + dev_err(dev, "ACP Init failed status:%d\n", status); + return status; + } if (adata->play_stream && adata->play_stream->runtime) { /* For Stoney, Memory gating is disabled,i.e SRAM Banks @@ -1145,18 +1162,26 @@ static int acp_pcm_resume(struct device *dev) static int acp_pcm_runtime_suspend(struct device *dev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_deinit(adata->acp_mmio); + status = acp_deinit(adata->acp_mmio); + if (status) + dev_err(dev, "ACP Deinit failed status:%d\n", status); acp_reg_write(0, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } static int acp_pcm_runtime_resume(struct device *dev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_init(adata->acp_mmio, adata->asic_type); + status = acp_init(adata->acp_mmio, adata->asic_type); + if (status) { + dev_err(dev, "ACP Init failed status:%d\n", status); + return status; + } acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } -- 2.7.4

2 2

[alsa-devel] [PATCH 1/2] ASoC: Intel: Skylake: Make local functions static
by Guneshwor Singh 04 Dec '17

04 Dec '17

Some functions are local to the source and do not need to be in global scope, so make them static. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/soc/intel/skylake/skl-nhlt.c | 4 ++-- sound/soc/intel/skylake/skl.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 6734a439196b..e2121ddc8ef0 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -283,7 +283,7 @@ void skl_nhlt_remove_sysfs(struct skl *skl) * stores all possible rates supported in a rate table for the corresponding * sclk/sclkfs. */ -void skl_get_ssp_clks(struct skl *skl, struct skl_ssp_clk *ssp_clks, +static void skl_get_ssp_clks(struct skl *skl, struct skl_ssp_clk *ssp_clks, struct nhlt_fmt *fmt, u8 id) { struct skl_i2s_config_blob_legacy *i2s_config; @@ -374,7 +374,7 @@ void skl_get_ssp_clks(struct skl *skl, struct skl_ssp_clk *ssp_clks, } } -void skl_get_mclk(struct skl *skl, struct skl_ssp_clk *mclk, +static void skl_get_mclk(struct skl *skl, struct skl_ssp_clk *mclk, struct nhlt_fmt *fmt, u8 id) { struct skl_i2s_config_blob_legacy *i2s_config; diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 9e90a50dd9cd..90cc207869c7 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -568,7 +568,7 @@ struct skl_clk_parent_src *skl_get_parent_clk(u8 clk_id) return NULL; } -void init_skl_xtal_rate(int pci_id) +static void init_skl_xtal_rate(int pci_id) { switch (pci_id) { case 0x9d70: -- 2.15.1

3 4

[alsa-devel] [PATCH 9/9] ASoC: eukrea-tlv320: Remove include line of fsl_ssi.h
by Nicolin Chen 04 Dec '17

04 Dec '17

The machine driver links both imx-ssi (legacy non-DT driver) and fsl_ssi (up-to-date DT based driver). So It also includes both imx-ssi.h and fsl_ssi.h header files. This creates a limitation for two header files -- they can't define anything with identical names. Since the eukrea-tlv320 machine driver now does not really need anything being defined in the fsl_ssi header file, and it's also going to take some time to clean up two SSI drivers, this patch takes a quick way to remove the #include fsl_ssi.h line for now. It can be added back once the header files are done refactoring. The eukrea-tlv320 driver is still compiled successfully without any erorr using imx_v6_v6_defconfig, after removing it. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/eukrea-tlv320.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 84ef638..191426a 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -29,7 +29,6 @@ #include "../codecs/tlv320aic23.h" #include "imx-ssi.h" -#include "fsl_ssi.h" #include "imx-audmux.h" #define CODEC_CLOCK 12000000 -- 2.7.4

2 3

[alsa-devel] ASoC: nau8825: improve semaphore control
by John Hsu 04 Dec '17

04 Dec '17

After reviewing the crosstalk protection, there are two flaws at semaphore control. The first one is that the semaphore releases are not enough; and the other is that down_interruptible has an risk to make the ISR sleep. Therefore, the driver add more releases before the funcitons return. Take down_trylock to replace down_interruptible. The ISR can control the protection as well and never sleep by semaphore. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 38 +++++++++++++++++++++++--------------- 1 file changed, 23 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 7947b11..2aea642 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -245,13 +245,14 @@ static const unsigned short logtable[256] = { * tasks are allowed to acquire the semaphore, calling this function will * put the task to sleep. If the semaphore is not released within the * specified number of jiffies, this function returns. - * Acquires the semaphore without jiffies. If no more tasks are allowed - * to acquire the semaphore, calling this function will put the task to - * sleep until the semaphore is released. * If the semaphore is not released within the specified number of jiffies, - * this function returns -ETIME. - * If the sleep is interrupted by a signal, this function will return -EINTR. - * It returns 0 if the semaphore was acquired successfully. + * this function returns -ETIME. If the sleep is interrupted by a signal, + * this function will return -EINTR. It returns 0 if the semaphore was + * acquired successfully. + * + * Acquires the semaphore without jiffies. Try to acquire the semaphore + * atomically. Returns 0 if the semaphore has been acquired successfully + * or 1 if it it cannot be acquired. */ static int nau8825_sema_acquire(struct nau8825 *nau8825, long timeout) { @@ -262,8 +263,8 @@ static int nau8825_sema_acquire(struct nau8825 *nau8825, long timeout) if (ret < 0) dev_warn(nau8825->dev, "Acquire semaphore timeout\n"); } else { - ret = down_interruptible(&nau8825->xtalk_sem); - if (ret < 0) + ret = down_trylock(&nau8825->xtalk_sem); + if (ret) dev_warn(nau8825->dev, "Acquire semaphore fail\n"); } @@ -1258,8 +1259,10 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, regmap_read(nau8825->regmap, NAU8825_REG_DAC_CTRL1, &osr); osr &= NAU8825_DAC_OVERSAMPLE_MASK; if (nau8825_clock_check(nau8825, substream->stream, - params_rate(params), osr)) + params_rate(params), osr)) { + nau8825_sema_release(nau8825); return -EINVAL; + } regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_DAC_SRC_MASK, osr_dac_sel[osr].clk_src << NAU8825_CLK_DAC_SRC_SFT); @@ -1267,8 +1270,10 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, regmap_read(nau8825->regmap, NAU8825_REG_ADC_RATE, &osr); osr &= NAU8825_ADC_SYNC_DOWN_MASK; if (nau8825_clock_check(nau8825, substream->stream, - params_rate(params), osr)) + params_rate(params), osr)) { + nau8825_sema_release(nau8825); return -EINVAL; + } regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_ADC_SRC_MASK, osr_adc_sel[osr].clk_src << NAU8825_CLK_ADC_SRC_SFT); @@ -1285,8 +1290,10 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, bclk_div = 1; else if (bclk_fs <= 128) bclk_div = 0; - else + else { + nau8825_sema_release(nau8825); return -EINVAL; + } regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, NAU8825_I2S_LRC_DIV_MASK | NAU8825_I2S_BLK_DIV_MASK, ((bclk_div + 1) << NAU8825_I2S_LRC_DIV_SFT) | bclk_div); @@ -1306,6 +1313,7 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, val_len |= NAU8825_I2S_DL_32; break; default: + nau8825_sema_release(nau8825); return -EINVAL; } @@ -1324,8 +1332,6 @@ static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); unsigned int ctrl1_val = 0, ctrl2_val = 0; - nau8825_sema_acquire(nau8825, 3 * HZ); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: ctrl2_val |= NAU8825_I2S_MS_MASTER; @@ -1367,6 +1373,8 @@ static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return -EINVAL; } + nau8825_sema_acquire(nau8825, 3 * HZ); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK | NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK, @@ -1713,7 +1721,7 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) int ret; nau8825->xtalk_protect = true; ret = nau8825_sema_acquire(nau8825, 0); - if (ret < 0) + if (ret) nau8825->xtalk_protect = false; } /* Startup cross talk detection process */ @@ -2399,7 +2407,7 @@ static int __maybe_unused nau8825_resume(struct snd_soc_codec *codec) regcache_sync(nau8825->regmap); nau8825->xtalk_protect = true; ret = nau8825_sema_acquire(nau8825, 0); - if (ret < 0) + if (ret) nau8825->xtalk_protect = false; enable_irq(nau8825->irq); -- 2.6.4

2 1

[alsa-devel] ASoC: nau8825: improve crosstalk measurement protection
by John Hsu 04 Dec '17

04 Dec '17

The sequence of crosstalk measurement is not robust. The driver add more protections to make it stronger. The improvements as follows: (1)Give backup table the default value. The values are the same with the default of hardware registers. The impact will decline once restoring the backup table in the wrong way. (2)Add xtalk_baktab_initialized flag. The flag can keep the initiation status of backup table. It helps to backup and restore the backup-table correctly. (3)Add cancel parameter in the restore function. The volume ramping should do in crosstalk DONE state only. Otherwise, the delay action is not allowed because the restore may happen during the jack eject interruption. (4)Add xtalk_protect condition check before scheduling work in ISR. It makes sure the semaphore hold during the crosstalk measurement. The sequence is still under protection from playback interference. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 45 +++++++++++++++++++++++++++++++-------------- sound/soc/codecs/nau8825.h | 1 + 2 files changed, 32 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 603cd72..7947b11 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -194,10 +194,10 @@ static const struct reg_default nau8825_reg_defaults[] = { /* register backup table when cross talk detection */ static struct reg_default nau8825_xtalk_baktab[] = { - { NAU8825_REG_ADC_DGAIN_CTRL, 0 }, + { NAU8825_REG_ADC_DGAIN_CTRL, 0x00cf }, { NAU8825_REG_HSVOL_CTRL, 0 }, - { NAU8825_REG_DACL_CTRL, 0 }, - { NAU8825_REG_DACR_CTRL, 0 }, + { NAU8825_REG_DACL_CTRL, 0x00cf }, + { NAU8825_REG_DACR_CTRL, 0x02cf }, }; static const unsigned short logtable[256] = { @@ -454,22 +454,32 @@ static void nau8825_xtalk_backup(struct nau8825 *nau8825) { int i; + if (nau8825->xtalk_baktab_initialized) + return; + /* Backup some register values to backup table */ for (i = 0; i < ARRAY_SIZE(nau8825_xtalk_baktab); i++) regmap_read(nau8825->regmap, nau8825_xtalk_baktab[i].reg, &nau8825_xtalk_baktab[i].def); + + nau8825->xtalk_baktab_initialized = true; } -static void nau8825_xtalk_restore(struct nau8825 *nau8825) +static void nau8825_xtalk_restore(struct nau8825 *nau8825, bool cause_cancel) { int i, volume; + if (!nau8825->xtalk_baktab_initialized) + return; + /* Restore register values from backup table; When the driver restores - * the headphone volumem, it needs recover to original level gradually - * with 3dB per step for less pop noise. + * the headphone volume in XTALK_DONE state, it needs recover to + * original level gradually with 3dB per step for less pop noise. + * Otherwise, the restore should do ASAP. */ for (i = 0; i < ARRAY_SIZE(nau8825_xtalk_baktab); i++) { - if (nau8825_xtalk_baktab[i].reg == NAU8825_REG_HSVOL_CTRL) { + if (!cause_cancel && nau8825_xtalk_baktab[i].reg == + NAU8825_REG_HSVOL_CTRL) { /* Ramping up the volume change to reduce pop noise */ volume = nau8825_xtalk_baktab[i].def & NAU8825_HPR_VOL_MASK; @@ -479,6 +489,8 @@ static void nau8825_xtalk_restore(struct nau8825 *nau8825) regmap_write(nau8825->regmap, nau8825_xtalk_baktab[i].reg, nau8825_xtalk_baktab[i].def); } + + nau8825->xtalk_baktab_initialized = false; } static void nau8825_xtalk_prepare_dac(struct nau8825 *nau8825) @@ -644,7 +656,7 @@ static void nau8825_xtalk_clean_adc(struct nau8825 *nau8825) NAU8825_POWERUP_ADCL | NAU8825_ADC_VREFSEL_MASK, 0); } -static void nau8825_xtalk_clean(struct nau8825 *nau8825) +static void nau8825_xtalk_clean(struct nau8825 *nau8825, bool cause_cancel) { /* Enable internal VCO needed for interruptions */ nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); @@ -660,7 +672,7 @@ static void nau8825_xtalk_clean(struct nau8825 *nau8825) NAU8825_I2S_MS_MASK | NAU8825_I2S_LRC_DIV_MASK | NAU8825_I2S_BLK_DIV_MASK, NAU8825_I2S_MS_SLAVE); /* Restore value of specific register for cross talk */ - nau8825_xtalk_restore(nau8825); + nau8825_xtalk_restore(nau8825, cause_cancel); } static void nau8825_xtalk_imm_start(struct nau8825 *nau8825, int vol) @@ -779,7 +791,7 @@ static void nau8825_xtalk_measure(struct nau8825 *nau8825) dev_dbg(nau8825->dev, "cross talk sidetone: %x\n", sidetone); regmap_write(nau8825->regmap, NAU8825_REG_DAC_DGAIN_CTRL, (sidetone << 8) | sidetone); - nau8825_xtalk_clean(nau8825); + nau8825_xtalk_clean(nau8825, false); nau8825->xtalk_state = NAU8825_XTALK_DONE; break; default: @@ -822,7 +834,7 @@ static void nau8825_xtalk_cancel(struct nau8825 *nau8825) if (nau8825->xtalk_enable && nau8825->xtalk_state != NAU8825_XTALK_DONE) { cancel_work_sync(&nau8825->xtalk_work); - nau8825_xtalk_clean(nau8825); + nau8825_xtalk_clean(nau8825, true); } /* Reset parameters for cross talk suppression function */ nau8825_sema_reset(nau8825); @@ -1705,8 +1717,11 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) nau8825->xtalk_protect = false; } /* Startup cross talk detection process */ - nau8825->xtalk_state = NAU8825_XTALK_PREPARE; - schedule_work(&nau8825->xtalk_work); + if (nau8825->xtalk_protect) { + nau8825->xtalk_state = + NAU8825_XTALK_PREPARE; + schedule_work(&nau8825->xtalk_work); + } } else { /* The cross talk suppression shouldn't apply * in the headset with high impedance. Thus, @@ -1733,7 +1748,8 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) nau8825->xtalk_event_mask = event_mask; } } else if (active_irq & NAU8825_IMPEDANCE_MEAS_IRQ) { - if (nau8825->xtalk_enable) + /* crosstalk detection enable and process on going */ + if (nau8825->xtalk_enable && nau8825->xtalk_protect) schedule_work(&nau8825->xtalk_work); clear_irq = NAU8825_IMPEDANCE_MEAS_IRQ; } else if ((active_irq & NAU8825_JACK_INSERTION_IRQ_MASK) == @@ -2570,6 +2586,7 @@ static int nau8825_i2c_probe(struct i2c_client *i2c, */ nau8825->xtalk_state = NAU8825_XTALK_DONE; nau8825->xtalk_protect = false; + nau8825->xtalk_baktab_initialized = false; sema_init(&nau8825->xtalk_sem, 1); INIT_WORK(&nau8825->xtalk_work, nau8825_xtalk_work); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 199d6ea..f7e7321 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -477,6 +477,7 @@ struct nau8825 { bool xtalk_protect; int imp_rms[NAU8825_XTALK_IMM]; int xtalk_enable; + bool xtalk_baktab_initialized; /* True if initialized. */ }; int nau8825_enable_jack_detect(struct snd_soc_codec *codec, -- 2.6.4

2 1

[alsa-devel] [PATCH 0/4] ALSA SoC TWL4030: Adjustments for two function implementations
by SF Markus Elfring 04 Dec '17

04 Dec '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 23 Nov 2017 22:02:44 +0100 A few update suggestions were taken into account from static source code analysis. Markus Elfring (4): Delete an error message for a failed memory allocation in twl4030_get_pdata() Use common error handling code in twl4030_get_pdata() Improve a size determination in two functions Delete an unnecessary variable initialisation in twl4030_get_pdata() sound/soc/codecs/twl4030.c | 18 +++++++----------- 1 file changed, 7 insertions(+), 11 deletions(-) -- 2.15.0

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[alsa-devel] [PATCH 0/3] ALSA: usb-audio: fix OOB and return value check
by Jaejoong Kim 04 Dec '17

04 Dec '17

Hi, This patch series fixes the out-of-bound error caused by the return value of usb_string(). It was descovered by KASAN. The Patch 1 is the V2 about http://www.spinics.net/lists/alsa-devel/msg69487.html Chanes in V2: - put an explicit error bail out(by Takashi iwai) Patch1 was founded by connecting the following product. http://www.lg.com/uk/lg-friends/lg-AFD-1200 I found that it only check if the return value from usb_string() is always zero while modifying OOB KASAN message. So instead of making the modifications to OOB to V2, I sent a patch series. I am sorry to break the mail thread. Thanks jaejoong Jaejoong Kim (3): ALSA: usb-audio: Fix out-of-bound error ALSA: usb-audio: Fix return value check for usb_string() ALSA: usb-audio: Add check return value for usb_string() sound/usb/mixer.c | 41 ++++++++++++++++++++++++----------------- 1 file changed, 24 insertions(+), 17 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 0/2] Add devicetree for Axentia Nattis
by Peter Rosin 03 Dec '17

03 Dec '17

Hi! I'd like to add a devicetree for our Nattis to the kernel. The Nattis is a device for showing departures for public transportation (optionally including a text-to-speech module for the visually impaired). I'm a bit unsure if the tfa9879 sound codec binding is needed, but I suppose it belongs in trivial-devices? Cheers, Peter Peter Rosin (2): dt-bindings: trivial: add tfa9879 device ARM: dts: at91: add devicetree for the Axentia Nattis with Natte power Documentation/devicetree/bindings/arm/axentia.txt | 9 + .../devicetree/bindings/trivial-devices.txt | 1 + MAINTAINERS | 2 + arch/arm/boot/dts/Makefile | 1 + arch/arm/boot/dts/at91-natte.dtsi | 245 +++++++++++++++++++ arch/arm/boot/dts/at91-nattis-2-natte-2.dts | 258 +++++++++++++++++++++ 6 files changed, 516 insertions(+) create mode 100644 arch/arm/boot/dts/at91-natte.dtsi create mode 100644 arch/arm/boot/dts/at91-nattis-2-natte-2.dts -- 2.11.0

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[alsa-devel] [asoc:topic/tlv320aic31xx 10/13] sound/soc/codecs/tlv320aic31xx.c:1025:4: error: implicit declaration of function 'gpiod_set_value'
by kbuild test robot 02 Dec '17

02 Dec '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/tlv320aic31xx head: dcb407b257af06fa58b0544ec01ec9e0d3927e02 commit: b6b247cd5e37560e410c88b108e7408dafe60c15 [10/13] ASoC: tlv320aic31xx: Switch GPIO handling to use gpiod_* API config: x86_64-randconfig-r0-12021500 (attached as .config) compiler: gcc-6 (Debian 6.4.0-9) 6.4.0 20171026 reproduce: git checkout b6b247cd5e37560e410c88b108e7408dafe60c15 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound/soc/codecs/tlv320aic31xx.c: In function 'aic31xx_regulator_event': >> sound/soc/codecs/tlv320aic31xx.c:1025:4: error: implicit declaration of function 'gpiod_set_value' [-Werror=implicit-function-declaration] gpiod_set_value(aic31xx->gpio_reset, 1); ^~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic31xx.c: In function 'aic31xx_i2c_probe': >> sound/soc/codecs/tlv320aic31xx.c:1338:24: error: implicit declaration of function 'devm_gpiod_get_optional' [-Werror=implicit-function-declaration] aic31xx->gpio_reset = devm_gpiod_get_optional(aic31xx->dev, "reset", ^~~~~~~~~~~~~~~~~~~~~~~ >> sound/soc/codecs/tlv320aic31xx.c:1339:13: error: 'GPIOD_OUT_LOW' undeclared (first use in this function) GPIOD_OUT_LOW); ^~~~~~~~~~~~~ sound/soc/codecs/tlv320aic31xx.c:1339:13: note: each undeclared identifier is reported only once for each function it appears in cc1: some warnings being treated as errors vim +/gpiod_set_value +1025 sound/soc/codecs/tlv320aic31xx.c 1011 1012 static int aic31xx_regulator_event(struct notifier_block *nb, 1013 unsigned long event, void *data) 1014 { 1015 struct aic31xx_disable_nb *disable_nb = 1016 container_of(nb, struct aic31xx_disable_nb, nb); 1017 struct aic31xx_priv *aic31xx = disable_nb->aic31xx; 1018 1019 if (event & REGULATOR_EVENT_DISABLE) { 1020 /* 1021 * Put codec to reset and as at least one of the 1022 * supplies was disabled. 1023 */ 1024 if (aic31xx->gpio_reset) > 1025 gpiod_set_value(aic31xx->gpio_reset, 1); 1026 1027 regcache_mark_dirty(aic31xx->regmap); 1028 dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__); 1029 } 1030 1031 return 0; 1032 } 1033 --- 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 v2] ASoC: eukrea-tlv320: Remove include line of fsl_ssi.h
by Nicolin Chen 02 Dec '17

02 Dec '17

The machine driver links both imx-ssi (legacy non-DT driver) and fsl_ssi (up-to-date DT based driver). So It also includes both imx-ssi.h and fsl_ssi.h header files. This creates a limitation for two header files -- they can't define anything with identical names. Since the eukrea-tlv320 machine driver now does not really need anything being defined in the fsl_ssi header file, and it's also going to take some time to clean up two SSI drivers, this patch takes a quick way to remove the #include fsl_ssi.h line for now. It can be added back once the header files are done refactoring. The eukrea-tlv320 driver is still compiled successfully without any erorr using imx_v6_v7_defconfig, after removing it. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changelog v2: * Fixed a typo in the commit log at imx_v6_v7_defconfig sound/soc/fsl/eukrea-tlv320.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 84ef638..191426a 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -29,7 +29,6 @@ #include "../codecs/tlv320aic23.h" #include "imx-ssi.h" -#include "fsl_ssi.h" #include "imx-audmux.h" #define CODEC_CLOCK 12000000 -- 2.7.4

1 0

[alsa-devel] [RFC 0/6] Enable HDA Codec support on Intel Platforms (Series1)
by Rakesh Ughreja 01 Dec '17

01 Dec '17

Enhanced HD Audio capabilities introduced in HD Audio controllers are added in backward compatible way i.e. it does not change the behavior of the controller with respect to HDA Bus and HDA devices. Since there is no change in the hardware with respect to the HDA bus and HDA device it is more appropriate to represent the same in software as well. In order to align software representation with hardware it makes more sense to have common data structures across device, bus and driver for both enhanced HDA controllers and legacy HDA controllers. This patch series removes the hdac_ext_device structure, hdac_ext_bus and hdac_ext_driver data structures so that legacy and enhanced HDaudio capabilities can be handled in a backwards-compatible way without separate definitions. Once this clean-up is complete, additional patches will allow for HDaudio codec support in an ASoC framework without the need to develop new codec drivers, thereby enabling the use of the Intel DSP on more platforms (currently limited to hdmi). Rakesh Ughreja (6): ASoC: hdac_hdmi: introduce macro to access HDMI private data ASoC: hdac_hdmi: clean up hdac_ext_device variable names ASoC: hdac_hdmi: clean up hdac_device variable names ALSA: hdac: Replace struct hdac_ext_device with struct hdac_device ALSA: hdac: Remove usage of struct hdac_ext_bus and use hdac_bus instead ALSA: hdac: Remove usage of struct hdac_ext_driver, use hdac_driver instead include/sound/hdaudio.h | 21 ++ include/sound/hdaudio_ext.h | 119 ++------ sound/hda/ext/hdac_ext_bus.c | 76 +++-- sound/hda/ext/hdac_ext_controller.c | 55 ++-- sound/hda/ext/hdac_ext_stream.c | 104 +++---- sound/soc/codecs/hdac_hdmi.c | 519 ++++++++++++++++----------------- sound/soc/intel/skylake/skl-messages.c | 50 ++-- sound/soc/intel/skylake/skl-nhlt.c | 8 +- sound/soc/intel/skylake/skl-pcm.c | 112 ++++--- sound/soc/intel/skylake/skl-topology.c | 18 +- sound/soc/intel/skylake/skl-topology.h | 6 +- sound/soc/intel/skylake/skl.c | 176 +++++------ sound/soc/intel/skylake/skl.h | 7 +- 13 files changed, 571 insertions(+), 700 deletions(-) -- 2.7.4

3 13

[alsa-devel] [PATCH 1/1] timecounter: Make cyclecounter struct part of timecounter struct
by Sagar Arun Kamble 01 Dec '17

01 Dec '17

There is no real need for the users of timecounters to define cyclecounter and timecounter variables separately. Since timecounter will always be based on cyclecounter, have cyclecounter struct as member of timecounter struct. Suggested-by: Chris Wilson <chris(a)chris-wilson.co.uk> Signed-off-by: Sagar Arun Kamble <sagar.a.kamble(a)intel.com> Cc: Chris Wilson <chris(a)chris-wilson.co.uk> Cc: John Stultz <john.stultz(a)linaro.org> Cc: Thomas Gleixner <tglx(a)linutronix.de> Cc: Stephen Boyd <sboyd(a)codeaurora.org> Cc: linux-kernel(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: netdev(a)vger.kernel.org Cc: intel-wired-lan(a)lists.osuosl.org Cc: linux-rdma(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: kvmarm(a)lists.cs.columbia.edu --- arch/microblaze/kernel/timer.c | 20 ++++++------ drivers/clocksource/arm_arch_timer.c | 19 ++++++------ drivers/net/ethernet/amd/xgbe/xgbe-dev.c | 3 +- drivers/net/ethernet/amd/xgbe/xgbe-ptp.c | 9 +++--- drivers/net/ethernet/amd/xgbe/xgbe.h | 1 - drivers/net/ethernet/broadcom/bnx2x/bnx2x.h | 1 - drivers/net/ethernet/broadcom/bnx2x/bnx2x_main.c | 20 ++++++------ drivers/net/ethernet/freescale/fec.h | 1 - drivers/net/ethernet/freescale/fec_ptp.c | 30 +++++++++--------- drivers/net/ethernet/intel/e1000e/e1000.h | 1 - drivers/net/ethernet/intel/e1000e/netdev.c | 27 ++++++++-------- drivers/net/ethernet/intel/e1000e/ptp.c | 2 +- drivers/net/ethernet/intel/igb/igb.h | 1 - drivers/net/ethernet/intel/igb/igb_ptp.c | 25 ++++++++------- drivers/net/ethernet/intel/ixgbe/ixgbe.h | 1 - drivers/net/ethernet/intel/ixgbe/ixgbe_ptp.c | 17 +++++----- drivers/net/ethernet/mellanox/mlx4/en_clock.c | 28 ++++++++--------- drivers/net/ethernet/mellanox/mlx4/mlx4_en.h | 1 - .../net/ethernet/mellanox/mlx5/core/lib/clock.c | 34 ++++++++++---------- drivers/net/ethernet/qlogic/qede/qede_ptp.c | 20 ++++++------ drivers/net/ethernet/ti/cpts.c | 36 ++++++++++++---------- drivers/net/ethernet/ti/cpts.h | 1 - include/linux/mlx5/driver.h | 1 - include/linux/timecounter.h | 4 +-- include/sound/hdaudio.h | 1 - kernel/time/timecounter.c | 28 ++++++++--------- sound/hda/hdac_stream.c | 7 +++-- virt/kvm/arm/arch_timer.c | 6 ++-- 28 files changed, 163 insertions(+), 182 deletions(-) diff --git a/arch/microblaze/kernel/timer.c b/arch/microblaze/kernel/timer.c index 7de941c..b7f89e9 100644 --- a/arch/microblaze/kernel/timer.c +++ b/arch/microblaze/kernel/timer.c @@ -199,27 +199,25 @@ static u64 xilinx_read(struct clocksource *cs) return (u64)xilinx_clock_read(); } -static struct timecounter xilinx_tc = { - .cc = NULL, -}; - static u64 xilinx_cc_read(const struct cyclecounter *cc) { return xilinx_read(NULL); } -static struct cyclecounter xilinx_cc = { - .read = xilinx_cc_read, - .mask = CLOCKSOURCE_MASK(32), - .shift = 8, +static struct timecounter xilinx_tc = { + .cc.read = xilinx_cc_read, + .cc.mask = CLOCKSOURCE_MASK(32), + .cc.mult = 0, + .cc.shift = 8, }; static int __init init_xilinx_timecounter(void) { - xilinx_cc.mult = div_sc(timer_clock_freq, NSEC_PER_SEC, - xilinx_cc.shift); + struct cyclecounter *cc = &xilinx_tc.cc; + + cc->mult = div_sc(timer_clock_freq, NSEC_PER_SEC, cc->shift); - timecounter_init(&xilinx_tc, &xilinx_cc, sched_clock()); + timecounter_init(&xilinx_tc, sched_clock()); return 0; } diff --git a/drivers/clocksource/arm_arch_timer.c b/drivers/clocksource/arm_arch_timer.c index 57cb2f0..31543e5 100644 --- a/drivers/clocksource/arm_arch_timer.c +++ b/drivers/clocksource/arm_arch_timer.c @@ -179,11 +179,6 @@ static u64 arch_counter_read_cc(const struct cyclecounter *cc) .flags = CLOCK_SOURCE_IS_CONTINUOUS, }; -static struct cyclecounter cyclecounter __ro_after_init = { - .read = arch_counter_read_cc, - .mask = CLOCKSOURCE_MASK(56), -}; - struct ate_acpi_oem_info { char oem_id[ACPI_OEM_ID_SIZE + 1]; char oem_table_id[ACPI_OEM_TABLE_ID_SIZE + 1]; @@ -915,7 +910,10 @@ static u64 arch_counter_get_cntvct_mem(void) return ((u64) vct_hi << 32) | vct_lo; } -static struct arch_timer_kvm_info arch_timer_kvm_info; +static struct arch_timer_kvm_info arch_timer_kvm_info = { + .timecounter.cc.read = arch_counter_read_cc, + .timecounter.cc.mask = CLOCKSOURCE_MASK(56), +}; struct arch_timer_kvm_info *arch_timer_get_kvm_info(void) { @@ -925,6 +923,7 @@ struct arch_timer_kvm_info *arch_timer_get_kvm_info(void) static void __init arch_counter_register(unsigned type) { u64 start_count; + struct cyclecounter *cc = &arch_timer_kvm_info.timecounter.cc; /* Register the CP15 based counter if we have one */ if (type & ARCH_TIMER_TYPE_CP15) { @@ -943,10 +942,10 @@ static void __init arch_counter_register(unsigned type) clocksource_counter.flags |= CLOCK_SOURCE_SUSPEND_NONSTOP; start_count = arch_timer_read_counter(); clocksource_register_hz(&clocksource_counter, arch_timer_rate); - cyclecounter.mult = clocksource_counter.mult; - cyclecounter.shift = clocksource_counter.shift; - timecounter_init(&arch_timer_kvm_info.timecounter, - &cyclecounter, start_count); + + cc->mult = clocksource_counter.mult; + cc->shift = clocksource_counter.shift; + timecounter_init(&arch_timer_kvm_info.timecounter, start_count); /* 56 bits minimum, so we assume worst case rollover */ sched_clock_register(arch_timer_read_counter, 56, arch_timer_rate); diff --git a/drivers/net/ethernet/amd/xgbe/xgbe-dev.c b/drivers/net/ethernet/amd/xgbe/xgbe-dev.c index e107e18..5005c87 100644 --- a/drivers/net/ethernet/amd/xgbe/xgbe-dev.c +++ b/drivers/net/ethernet/amd/xgbe/xgbe-dev.c @@ -1622,8 +1622,7 @@ static int xgbe_config_tstamp(struct xgbe_prv_data *pdata, xgbe_set_tstamp_time(pdata, 0, 0); /* Initialize the timecounter */ - timecounter_init(&pdata->tstamp_tc, &pdata->tstamp_cc, - ktime_to_ns(ktime_get_real())); + timecounter_init(&pdata->tstamp_tc, ktime_to_ns(ktime_get_real())); return 0; } diff --git a/drivers/net/ethernet/amd/xgbe/xgbe-ptp.c b/drivers/net/ethernet/amd/xgbe/xgbe-ptp.c index d06d260..5ea4edf 100644 --- a/drivers/net/ethernet/amd/xgbe/xgbe-ptp.c +++ b/drivers/net/ethernet/amd/xgbe/xgbe-ptp.c @@ -126,7 +126,7 @@ static u64 xgbe_cc_read(const struct cyclecounter *cc) { struct xgbe_prv_data *pdata = container_of(cc, struct xgbe_prv_data, - tstamp_cc); + tstamp_tc.cc); u64 nsec; nsec = pdata->hw_if.get_tstamp_time(pdata); @@ -211,7 +211,7 @@ static int xgbe_settime(struct ptp_clock_info *info, spin_lock_irqsave(&pdata->tstamp_lock, flags); - timecounter_init(&pdata->tstamp_tc, &pdata->tstamp_cc, nsec); + timecounter_init(&pdata->tstamp_tc, nsec); spin_unlock_irqrestore(&pdata->tstamp_lock, flags); @@ -228,7 +228,7 @@ void xgbe_ptp_register(struct xgbe_prv_data *pdata) { struct ptp_clock_info *info = &pdata->ptp_clock_info; struct ptp_clock *clock; - struct cyclecounter *cc = &pdata->tstamp_cc; + struct cyclecounter *cc = &pdata->tstamp_tc.cc; u64 dividend; snprintf(info->name, sizeof(info->name), "%s", @@ -263,8 +263,7 @@ void xgbe_ptp_register(struct xgbe_prv_data *pdata) cc->mult = 1; cc->shift = 0; - timecounter_init(&pdata->tstamp_tc, &pdata->tstamp_cc, - ktime_to_ns(ktime_get_real())); + timecounter_init(&pdata->tstamp_tc, ktime_to_ns(ktime_get_real())); /* Disable all timestamping to start */ XGMAC_IOWRITE(pdata, MAC_TSCR, 0); diff --git a/drivers/net/ethernet/amd/xgbe/xgbe.h b/drivers/net/ethernet/amd/xgbe/xgbe.h index ad102c8..2445103 100644 --- a/drivers/net/ethernet/amd/xgbe/xgbe.h +++ b/drivers/net/ethernet/amd/xgbe/xgbe.h @@ -1168,7 +1168,6 @@ struct xgbe_prv_data { struct ptp_clock_info ptp_clock_info; struct ptp_clock *ptp_clock; struct hwtstamp_config tstamp_config; - struct cyclecounter tstamp_cc; struct timecounter tstamp_tc; unsigned int tstamp_addend; struct work_struct tx_tstamp_work; diff --git a/drivers/net/ethernet/broadcom/bnx2x/bnx2x.h b/drivers/net/ethernet/broadcom/bnx2x/bnx2x.h index 352beff..f164fe0 100644 --- a/drivers/net/ethernet/broadcom/bnx2x/bnx2x.h +++ b/drivers/net/ethernet/broadcom/bnx2x/bnx2x.h @@ -1827,7 +1827,6 @@ struct bnx2x { struct ptp_clock *ptp_clock; struct ptp_clock_info ptp_clock_info; struct work_struct ptp_task; - struct cyclecounter cyclecounter; struct timecounter timecounter; bool timecounter_init_done; struct sk_buff *ptp_tx_skb; diff --git a/drivers/net/ethernet/broadcom/bnx2x/bnx2x_main.c b/drivers/net/ethernet/broadcom/bnx2x/bnx2x_main.c index 91e2a75..83624ad 100644 --- a/drivers/net/ethernet/broadcom/bnx2x/bnx2x_main.c +++ b/drivers/net/ethernet/broadcom/bnx2x/bnx2x_main.c @@ -13850,7 +13850,7 @@ static int bnx2x_ptp_settime(struct ptp_clock_info *ptp, DP(BNX2X_MSG_PTP, "PTP settime called, ns = %llu\n", ns); /* Re-init the timecounter */ - timecounter_init(&bp->timecounter, &bp->cyclecounter, ns); + timecounter_init(&bp->timecounter, ns); return 0; } @@ -15254,7 +15254,7 @@ void bnx2x_set_rx_ts(struct bnx2x *bp, struct sk_buff *skb) /* Read the PHC */ static u64 bnx2x_cyclecounter_read(const struct cyclecounter *cc) { - struct bnx2x *bp = container_of(cc, struct bnx2x, cyclecounter); + struct bnx2x *bp = container_of(cc, struct bnx2x, timecounter.cc); int port = BP_PORT(bp); u32 wb_data[2]; u64 phc_cycles; @@ -15269,13 +15269,13 @@ static u64 bnx2x_cyclecounter_read(const struct cyclecounter *cc) return phc_cycles; } -static void bnx2x_init_cyclecounter(struct bnx2x *bp) +static void bnx2x_init_cyclecounter(struct cyclecounter *cc) { - memset(&bp->cyclecounter, 0, sizeof(bp->cyclecounter)); - bp->cyclecounter.read = bnx2x_cyclecounter_read; - bp->cyclecounter.mask = CYCLECOUNTER_MASK(64); - bp->cyclecounter.shift = 0; - bp->cyclecounter.mult = 1; + memset(cc, 0, sizeof(*cc)); + cc->read = bnx2x_cyclecounter_read; + cc->mask = CYCLECOUNTER_MASK(64); + cc->shift = 0; + cc->mult = 1; } static int bnx2x_send_reset_timesync_ramrod(struct bnx2x *bp) @@ -15511,8 +15511,8 @@ void bnx2x_init_ptp(struct bnx2x *bp) * unload / load (e.g. MTU change) while it is running. */ if (!bp->timecounter_init_done) { - bnx2x_init_cyclecounter(bp); - timecounter_init(&bp->timecounter, &bp->cyclecounter, + bnx2x_init_cyclecounter(&bp->timecounter.cc); + timecounter_init(&bp->timecounter, ktime_to_ns(ktime_get_real())); bp->timecounter_init_done = 1; } diff --git a/drivers/net/ethernet/freescale/fec.h b/drivers/net/ethernet/freescale/fec.h index 5385074..d54b501 100644 --- a/drivers/net/ethernet/freescale/fec.h +++ b/drivers/net/ethernet/freescale/fec.h @@ -549,7 +549,6 @@ struct fec_enet_private { struct ptp_clock_info ptp_caps; unsigned long last_overflow_check; spinlock_t tmreg_lock; - struct cyclecounter cc; struct timecounter tc; int rx_hwtstamp_filter; u32 base_incval; diff --git a/drivers/net/ethernet/freescale/fec_ptp.c b/drivers/net/ethernet/freescale/fec_ptp.c index f814397..b1261d1 100644 --- a/drivers/net/ethernet/freescale/fec_ptp.c +++ b/drivers/net/ethernet/freescale/fec_ptp.c @@ -186,13 +186,14 @@ static int fec_ptp_enable_pps(struct fec_enet_private *fep, uint enable) * ptp counter, which maybe cause 32-bit wrap. Since the * (NSEC_PER_SEC - (u32)ts.tv_nsec) is less than 2 second. * We can ensure the wrap will not cause issue. If the offset - * is bigger than fep->cc.mask would be a error. + * is bigger than fep->tc.cc.mask would be a error. */ - val &= fep->cc.mask; + val &= fep->tc.cc.mask; writel(val, fep->hwp + FEC_TCCR(fep->pps_channel)); /* Calculate the second the compare event timestamp */ - fep->next_counter = (val + fep->reload_period) & fep->cc.mask; + fep->next_counter = (val + fep->reload_period) & + fep->tc.cc.mask; /* * Enable compare event when overflow */ val = readl(fep->hwp + FEC_ATIME_CTRL); @@ -211,7 +212,8 @@ static int fec_ptp_enable_pps(struct fec_enet_private *fep, uint enable) * the third timestamp. Refer the TCCR register detail in the spec. */ writel(fep->next_counter, fep->hwp + FEC_TCCR(fep->pps_channel)); - fep->next_counter = (fep->next_counter + fep->reload_period) & fep->cc.mask; + fep->next_counter = (fep->next_counter + fep->reload_period) & + fep->tc.cc.mask; } else { writel(0, fep->hwp + FEC_TCSR(fep->pps_channel)); } @@ -233,7 +235,7 @@ static int fec_ptp_enable_pps(struct fec_enet_private *fep, uint enable) static u64 fec_ptp_read(const struct cyclecounter *cc) { struct fec_enet_private *fep = - container_of(cc, struct fec_enet_private, cc); + container_of(cc, struct fec_enet_private, tc.cc); const struct platform_device_id *id_entry = platform_get_device_id(fep->pdev); u32 tempval; @@ -276,14 +278,14 @@ void fec_ptp_start_cyclecounter(struct net_device *ndev) writel(FEC_T_CTRL_ENABLE | FEC_T_CTRL_PERIOD_RST, fep->hwp + FEC_ATIME_CTRL); - memset(&fep->cc, 0, sizeof(fep->cc)); - fep->cc.read = fec_ptp_read; - fep->cc.mask = CLOCKSOURCE_MASK(31); - fep->cc.shift = 31; - fep->cc.mult = FEC_CC_MULT; + memset(&fep->tc.cc, 0, sizeof(fep->tc.cc)); + fep->tc.cc.read = fec_ptp_read; + fep->tc.cc.mask = CLOCKSOURCE_MASK(31); + fep->tc.cc.shift = 31; + fep->tc.cc.mult = FEC_CC_MULT; /* reset the ns time counter */ - timecounter_init(&fep->tc, &fep->cc, ktime_to_ns(ktime_get_real())); + timecounter_init(&fep->tc, ktime_to_ns(ktime_get_real())); spin_unlock_irqrestore(&fep->tmreg_lock, flags); } @@ -434,11 +436,11 @@ static int fec_ptp_settime(struct ptp_clock_info *ptp, /* Get the timer value based on timestamp. * Update the counter with the masked value. */ - counter = ns & fep->cc.mask; + counter = ns & fep->tc.cc.mask; spin_lock_irqsave(&fep->tmreg_lock, flags); writel(counter, fep->hwp + FEC_ATIME); - timecounter_init(&fep->tc, &fep->cc, ns); + timecounter_init(&fep->tc, ns); spin_unlock_irqrestore(&fep->tmreg_lock, flags); mutex_unlock(&fep->ptp_clk_mutex); return 0; @@ -570,7 +572,7 @@ static irqreturn_t fec_pps_interrupt(int irq, void *dev_id) /* Update the counter; */ fep->next_counter = (fep->next_counter + fep->reload_period) & - fep->cc.mask; + fep->tc.cc.mask; event.type = PTP_CLOCK_PPS; ptp_clock_event(fep->ptp_clock, &event); diff --git a/drivers/net/ethernet/intel/e1000e/e1000.h b/drivers/net/ethernet/intel/e1000e/e1000.h index 2311b31..b59f82a 100644 --- a/drivers/net/ethernet/intel/e1000e/e1000.h +++ b/drivers/net/ethernet/intel/e1000e/e1000.h @@ -340,7 +340,6 @@ struct e1000_adapter { unsigned long tx_hwtstamp_start; struct work_struct tx_hwtstamp_work; spinlock_t systim_lock; /* protects SYSTIML/H regsters */ - struct cyclecounter cc; struct timecounter tc; struct ptp_clock *ptp_clock; struct ptp_clock_info ptp_clock_info; diff --git a/drivers/net/ethernet/intel/e1000e/netdev.c b/drivers/net/ethernet/intel/e1000e/netdev.c index 9f18d39..c9f7ba3 100644 --- a/drivers/net/ethernet/intel/e1000e/netdev.c +++ b/drivers/net/ethernet/intel/e1000e/netdev.c @@ -3536,7 +3536,7 @@ s32 e1000e_get_base_timinca(struct e1000_adapter *adapter, u32 *timinca) incperiod = INCPERIOD_96MHZ; incvalue = INCVALUE_96MHZ; shift = INCVALUE_SHIFT_96MHZ; - adapter->cc.shift = shift + INCPERIOD_SHIFT_96MHZ; + adapter->tc.cc.shift = shift + INCPERIOD_SHIFT_96MHZ; break; case e1000_pch_lpt: if (er32(TSYNCRXCTL) & E1000_TSYNCRXCTL_SYSCFI) { @@ -3544,13 +3544,13 @@ s32 e1000e_get_base_timinca(struct e1000_adapter *adapter, u32 *timinca) incperiod = INCPERIOD_96MHZ; incvalue = INCVALUE_96MHZ; shift = INCVALUE_SHIFT_96MHZ; - adapter->cc.shift = shift + INCPERIOD_SHIFT_96MHZ; + adapter->tc.cc.shift = shift + INCPERIOD_SHIFT_96MHZ; } else { /* Stable 25MHz frequency */ incperiod = INCPERIOD_25MHZ; incvalue = INCVALUE_25MHZ; shift = INCVALUE_SHIFT_25MHZ; - adapter->cc.shift = shift; + adapter->tc.cc.shift = shift; } break; case e1000_pch_spt: @@ -3559,7 +3559,7 @@ s32 e1000e_get_base_timinca(struct e1000_adapter *adapter, u32 *timinca) incperiod = INCPERIOD_24MHZ; incvalue = INCVALUE_24MHZ; shift = INCVALUE_SHIFT_24MHZ; - adapter->cc.shift = shift; + adapter->tc.cc.shift = shift; break; } return -EINVAL; @@ -3569,13 +3569,13 @@ s32 e1000e_get_base_timinca(struct e1000_adapter *adapter, u32 *timinca) incperiod = INCPERIOD_24MHZ; incvalue = INCVALUE_24MHZ; shift = INCVALUE_SHIFT_24MHZ; - adapter->cc.shift = shift; + adapter->tc.cc.shift = shift; } else { /* Stable 38400KHz frequency */ incperiod = INCPERIOD_38400KHZ; incvalue = INCVALUE_38400KHZ; shift = INCVALUE_SHIFT_38400KHZ; - adapter->cc.shift = shift; + adapter->tc.cc.shift = shift; } break; case e1000_82574: @@ -3584,7 +3584,7 @@ s32 e1000e_get_base_timinca(struct e1000_adapter *adapter, u32 *timinca) incperiod = INCPERIOD_25MHZ; incvalue = INCVALUE_25MHZ; shift = INCVALUE_SHIFT_25MHZ; - adapter->cc.shift = shift; + adapter->tc.cc.shift = shift; break; default: return -EINVAL; @@ -3955,8 +3955,7 @@ static void e1000e_systim_reset(struct e1000_adapter *adapter) /* reset the systim ns time counter */ spin_lock_irqsave(&adapter->systim_lock, flags); - timecounter_init(&adapter->tc, &adapter->cc, - ktime_to_ns(ktime_get_real())); + timecounter_init(&adapter->tc, ktime_to_ns(ktime_get_real())); spin_unlock_irqrestore(&adapter->systim_lock, flags); /* restore the previous hwtstamp configuration settings */ @@ -4389,7 +4388,7 @@ static u64 e1000e_sanitize_systim(struct e1000_hw *hw, u64 systim) static u64 e1000e_cyclecounter_read(const struct cyclecounter *cc) { struct e1000_adapter *adapter = container_of(cc, struct e1000_adapter, - cc); + tc.cc); struct e1000_hw *hw = &adapter->hw; u32 systimel, systimeh; u64 systim; @@ -4449,10 +4448,10 @@ static int e1000_sw_init(struct e1000_adapter *adapter) /* Setup hardware time stamping cyclecounter */ if (adapter->flags & FLAG_HAS_HW_TIMESTAMP) { - adapter->cc.read = e1000e_cyclecounter_read; - adapter->cc.mask = CYCLECOUNTER_MASK(64); - adapter->cc.mult = 1; - /* cc.shift set in e1000e_get_base_tininca() */ + adapter->tc.cc.read = e1000e_cyclecounter_read; + adapter->tc.cc.mask = CYCLECOUNTER_MASK(64); + adapter->tc.cc.mult = 1; + /* tc.cc.shift set in e1000e_get_base_tininca() */ spin_lock_init(&adapter->systim_lock); INIT_WORK(&adapter->tx_hwtstamp_work, e1000e_tx_hwtstamp_work); diff --git a/drivers/net/ethernet/intel/e1000e/ptp.c b/drivers/net/ethernet/intel/e1000e/ptp.c index b366885..03d5f2a 100644 --- a/drivers/net/ethernet/intel/e1000e/ptp.c +++ b/drivers/net/ethernet/intel/e1000e/ptp.c @@ -222,7 +222,7 @@ static int e1000e_phc_settime(struct ptp_clock_info *ptp, /* reset the timecounter */ spin_lock_irqsave(&adapter->systim_lock, flags); - timecounter_init(&adapter->tc, &adapter->cc, ns); + timecounter_init(&adapter->tc, ns); spin_unlock_irqrestore(&adapter->systim_lock, flags); return 0; diff --git a/drivers/net/ethernet/intel/igb/igb.h b/drivers/net/ethernet/intel/igb/igb.h index 9284569..4eac4f2 100644 --- a/drivers/net/ethernet/intel/igb/igb.h +++ b/drivers/net/ethernet/intel/igb/igb.h @@ -565,7 +565,6 @@ struct igb_adapter { unsigned long last_rx_timestamp; unsigned int ptp_flags; spinlock_t tmreg_lock; - struct cyclecounter cc; struct timecounter tc; u32 tx_hwtstamp_timeouts; u32 tx_hwtstamp_skipped; diff --git a/drivers/net/ethernet/intel/igb/igb_ptp.c b/drivers/net/ethernet/intel/igb/igb_ptp.c index 841c2a0..0745eff 100644 --- a/drivers/net/ethernet/intel/igb/igb_ptp.c +++ b/drivers/net/ethernet/intel/igb/igb_ptp.c @@ -79,7 +79,7 @@ /* SYSTIM read access for the 82576 */ static u64 igb_ptp_read_82576(const struct cyclecounter *cc) { - struct igb_adapter *igb = container_of(cc, struct igb_adapter, cc); + struct igb_adapter *igb = container_of(cc, struct igb_adapter, tc.cc); struct e1000_hw *hw = &igb->hw; u64 val; u32 lo, hi; @@ -96,7 +96,7 @@ static u64 igb_ptp_read_82576(const struct cyclecounter *cc) /* SYSTIM read access for the 82580 */ static u64 igb_ptp_read_82580(const struct cyclecounter *cc) { - struct igb_adapter *igb = container_of(cc, struct igb_adapter, cc); + struct igb_adapter *igb = container_of(cc, struct igb_adapter, tc.cc); struct e1000_hw *hw = &igb->hw; u32 lo, hi; u64 val; @@ -330,7 +330,7 @@ static int igb_ptp_settime_82576(struct ptp_clock_info *ptp, spin_lock_irqsave(&igb->tmreg_lock, flags); - timecounter_init(&igb->tc, &igb->cc, ns); + timecounter_init(&igb->tc, ns); spin_unlock_irqrestore(&igb->tmreg_lock, flags); @@ -1126,10 +1126,10 @@ void igb_ptp_init(struct igb_adapter *adapter) adapter->ptp_caps.gettime64 = igb_ptp_gettime_82576; adapter->ptp_caps.settime64 = igb_ptp_settime_82576; adapter->ptp_caps.enable = igb_ptp_feature_enable; - adapter->cc.read = igb_ptp_read_82576; - adapter->cc.mask = CYCLECOUNTER_MASK(64); - adapter->cc.mult = 1; - adapter->cc.shift = IGB_82576_TSYNC_SHIFT; + adapter->tc.cc.read = igb_ptp_read_82576; + adapter->tc.cc.mask = CYCLECOUNTER_MASK(64); + adapter->tc.cc.mult = 1; + adapter->tc.cc.shift = IGB_82576_TSYNC_SHIFT; adapter->ptp_flags |= IGB_PTP_OVERFLOW_CHECK; break; case e1000_82580: @@ -1145,10 +1145,10 @@ void igb_ptp_init(struct igb_adapter *adapter) adapter->ptp_caps.gettime64 = igb_ptp_gettime_82576; adapter->ptp_caps.settime64 = igb_ptp_settime_82576; adapter->ptp_caps.enable = igb_ptp_feature_enable; - adapter->cc.read = igb_ptp_read_82580; - adapter->cc.mask = CYCLECOUNTER_MASK(IGB_NBITS_82580); - adapter->cc.mult = 1; - adapter->cc.shift = 0; + adapter->tc.cc.read = igb_ptp_read_82580; + adapter->tc.cc.mask = CYCLECOUNTER_MASK(IGB_NBITS_82580); + adapter->tc.cc.mult = 1; + adapter->tc.cc.shift = 0; adapter->ptp_flags |= IGB_PTP_OVERFLOW_CHECK; break; case e1000_i210: @@ -1289,8 +1289,7 @@ void igb_ptp_reset(struct igb_adapter *adapter) igb_ptp_write_i210(adapter, &ts); } else { - timecounter_init(&adapter->tc, &adapter->cc, - ktime_to_ns(ktime_get_real())); + timecounter_init(&adapter->tc, ktime_to_ns(ktime_get_real())); } out: spin_unlock_irqrestore(&adapter->tmreg_lock, flags); diff --git a/drivers/net/ethernet/intel/ixgbe/ixgbe.h b/drivers/net/ethernet/intel/ixgbe/ixgbe.h index 468c355..5c391a0 100644 --- a/drivers/net/ethernet/intel/ixgbe/ixgbe.h +++ b/drivers/net/ethernet/intel/ixgbe/ixgbe.h @@ -738,7 +738,6 @@ struct ixgbe_adapter { unsigned long last_rx_ptp_check; unsigned long last_rx_timestamp; spinlock_t tmreg_lock; - struct cyclecounter hw_cc; struct timecounter hw_tc; u32 base_incval; u32 tx_hwtstamp_timeouts; diff --git a/drivers/net/ethernet/intel/ixgbe/ixgbe_ptp.c b/drivers/net/ethernet/intel/ixgbe/ixgbe_ptp.c index ae312c4..6e9f2c0 100644 --- a/drivers/net/ethernet/intel/ixgbe/ixgbe_ptp.c +++ b/drivers/net/ethernet/intel/ixgbe/ixgbe_ptp.c @@ -179,7 +179,7 @@ static void ixgbe_ptp_setup_sdp_x540(struct ixgbe_adapter *adapter) { struct ixgbe_hw *hw = &adapter->hw; - int shift = adapter->hw_cc.shift; + int shift = adapter->hw_tc.cc.shift; u32 esdp, tsauxc, clktiml, clktimh, trgttiml, trgttimh, rem; u64 ns = 0, clock_edge = 0; @@ -237,7 +237,7 @@ static void ixgbe_ptp_setup_sdp_x540(struct ixgbe_adapter *adapter) /** * ixgbe_ptp_read_X550 - read cycle counter value - * @hw_cc: cyclecounter structure + * @cc: cyclecounter structure * * This function reads SYSTIME registers. It is called by the cyclecounter * structure to convert from internal representation into nanoseconds. We need @@ -245,10 +245,10 @@ static void ixgbe_ptp_setup_sdp_x540(struct ixgbe_adapter *adapter) * result of SYSTIME is 32bits of "billions of cycles" and 32 bits of * "cycles", rather than seconds and nanoseconds. */ -static u64 ixgbe_ptp_read_X550(const struct cyclecounter *hw_cc) +static u64 ixgbe_ptp_read_X550(const struct cyclecounter *cc) { struct ixgbe_adapter *adapter = - container_of(hw_cc, struct ixgbe_adapter, hw_cc); + container_of(cc, struct ixgbe_adapter, hw_tc.cc); struct ixgbe_hw *hw = &adapter->hw; struct timespec64 ts; @@ -285,7 +285,7 @@ static u64 ixgbe_ptp_read_X550(const struct cyclecounter *hw_cc) static u64 ixgbe_ptp_read_82599(const struct cyclecounter *cc) { struct ixgbe_adapter *adapter = - container_of(cc, struct ixgbe_adapter, hw_cc); + container_of(cc, struct ixgbe_adapter, hw_tc.cc); struct ixgbe_hw *hw = &adapter->hw; u64 stamp = 0; @@ -508,7 +508,7 @@ static int ixgbe_ptp_settime(struct ptp_clock_info *ptp, /* reset the timecounter */ spin_lock_irqsave(&adapter->tmreg_lock, flags); - timecounter_init(&adapter->hw_tc, &adapter->hw_cc, ns); + timecounter_init(&adapter->hw_tc, ns); spin_unlock_irqrestore(&adapter->tmreg_lock, flags); if (adapter->ptp_setup_sdp) @@ -1164,7 +1164,7 @@ void ixgbe_ptp_start_cyclecounter(struct ixgbe_adapter *adapter) /* need lock to prevent incorrect read while modifying cyclecounter */ spin_lock_irqsave(&adapter->tmreg_lock, flags); - memcpy(&adapter->hw_cc, &cc, sizeof(adapter->hw_cc)); + memcpy(&adapter->hw_tc.cc, &cc, sizeof(adapter->hw_tc.cc)); spin_unlock_irqrestore(&adapter->tmreg_lock, flags); } @@ -1195,8 +1195,7 @@ void ixgbe_ptp_reset(struct ixgbe_adapter *adapter) ixgbe_ptp_start_cyclecounter(adapter); spin_lock_irqsave(&adapter->tmreg_lock, flags); - timecounter_init(&adapter->hw_tc, &adapter->hw_cc, - ktime_to_ns(ktime_get_real())); + timecounter_init(&adapter->hw_tc, ktime_to_ns(ktime_get_real())); spin_unlock_irqrestore(&adapter->tmreg_lock, flags); adapter->last_overflow_check = jiffies; diff --git a/drivers/net/ethernet/mellanox/mlx4/en_clock.c b/drivers/net/ethernet/mellanox/mlx4/en_clock.c index 0247885..35987b5 100644 --- a/drivers/net/ethernet/mellanox/mlx4/en_clock.c +++ b/drivers/net/ethernet/mellanox/mlx4/en_clock.c @@ -38,13 +38,13 @@ /* mlx4_en_read_clock - read raw cycle counter (to be used by time counter) */ -static u64 mlx4_en_read_clock(const struct cyclecounter *tc) +static u64 mlx4_en_read_clock(const struct cyclecounter *cc) { struct mlx4_en_dev *mdev = - container_of(tc, struct mlx4_en_dev, cycles); + container_of(cc, struct mlx4_en_dev, clock.cc); struct mlx4_dev *dev = mdev->dev; - return mlx4_read_clock(dev) & tc->mask; + return mlx4_read_clock(dev) & cc->mask; } u64 mlx4_en_get_cqe_ts(struct mlx4_cqe *cqe) @@ -138,7 +138,7 @@ static int mlx4_en_phc_adjfreq(struct ptp_clock_info *ptp, s32 delta) write_seqlock_irqsave(&mdev->clock_lock, flags); timecounter_read(&mdev->clock); - mdev->cycles.mult = neg_adj ? mult - diff : mult + diff; + mdev->clock.cc.mult = neg_adj ? mult - diff : mult + diff; write_sequnlock_irqrestore(&mdev->clock_lock, flags); return 0; @@ -207,7 +207,7 @@ static int mlx4_en_phc_settime(struct ptp_clock_info *ptp, /* reset the timecounter */ write_seqlock_irqsave(&mdev->clock_lock, flags); - timecounter_init(&mdev->clock, &mdev->cycles, ns); + timecounter_init(&mdev->clock, ns); write_sequnlock_irqrestore(&mdev->clock_lock, flags); return 0; @@ -274,17 +274,17 @@ void mlx4_en_init_timestamp(struct mlx4_en_dev *mdev) seqlock_init(&mdev->clock_lock); - memset(&mdev->cycles, 0, sizeof(mdev->cycles)); - mdev->cycles.read = mlx4_en_read_clock; - mdev->cycles.mask = CLOCKSOURCE_MASK(48); - mdev->cycles.shift = freq_to_shift(dev->caps.hca_core_clock); - mdev->cycles.mult = - clocksource_khz2mult(1000 * dev->caps.hca_core_clock, mdev->cycles.shift); - mdev->nominal_c_mult = mdev->cycles.mult; + memset(&mdev->clock.cc, 0, sizeof(mdev->clock.cc)); + mdev->clock.cc.read = mlx4_en_read_clock; + mdev->clock.cc.mask = CLOCKSOURCE_MASK(48); + mdev->clock.cc.shift = freq_to_shift(dev->caps.hca_core_clock); + mdev->clock.cc.mult = + clocksource_khz2mult(1000 * dev->caps.hca_core_clock, + mdev->clock.cc.shift); + mdev->nominal_c_mult = mdev->clock.cc.mult; write_seqlock_irqsave(&mdev->clock_lock, flags); - timecounter_init(&mdev->clock, &mdev->cycles, - ktime_to_ns(ktime_get_real())); + timecounter_init(&mdev->clock, ktime_to_ns(ktime_get_real())); write_sequnlock_irqrestore(&mdev->clock_lock, flags); /* Configure the PHC */ diff --git a/drivers/net/ethernet/mellanox/mlx4/mlx4_en.h b/drivers/net/ethernet/mellanox/mlx4/mlx4_en.h index 1856e27..e301dcf 100644 --- a/drivers/net/ethernet/mellanox/mlx4/mlx4_en.h +++ b/drivers/net/ethernet/mellanox/mlx4/mlx4_en.h @@ -422,7 +422,6 @@ struct mlx4_en_dev { spinlock_t uar_lock; u8 mac_removed[MLX4_MAX_PORTS + 1]; u32 nominal_c_mult; - struct cyclecounter cycles; seqlock_t clock_lock; struct timecounter clock; unsigned long last_overflow_check; diff --git a/drivers/net/ethernet/mellanox/mlx5/core/lib/clock.c b/drivers/net/ethernet/mellanox/mlx5/core/lib/clock.c index fa8aed6..8cb6838 100644 --- a/drivers/net/ethernet/mellanox/mlx5/core/lib/clock.c +++ b/drivers/net/ethernet/mellanox/mlx5/core/lib/clock.c @@ -64,7 +64,7 @@ enum { static u64 read_internal_timer(const struct cyclecounter *cc) { - struct mlx5_clock *clock = container_of(cc, struct mlx5_clock, cycles); + struct mlx5_clock *clock = container_of(cc, struct mlx5_clock, tc.cc); struct mlx5_core_dev *mdev = container_of(clock, struct mlx5_core_dev, clock); @@ -122,7 +122,7 @@ static int mlx5_ptp_settime(struct ptp_clock_info *ptp, unsigned long flags; write_lock_irqsave(&clock->lock, flags); - timecounter_init(&clock->tc, &clock->cycles, ns); + timecounter_init(&clock->tc, ns); write_unlock_irqrestore(&clock->lock, flags); return 0; @@ -177,8 +177,8 @@ static int mlx5_ptp_adjfreq(struct ptp_clock_info *ptp, s32 delta) write_lock_irqsave(&clock->lock, flags); timecounter_read(&clock->tc); - clock->cycles.mult = neg_adj ? clock->nominal_c_mult - diff : - clock->nominal_c_mult + diff; + clock->tc.cc.mult = neg_adj ? clock->nominal_c_mult - diff : + clock->nominal_c_mult + diff; write_unlock_irqrestore(&clock->lock, flags); return 0; @@ -281,8 +281,8 @@ static int mlx5_perout_configure(struct ptp_clock_info *ptp, write_lock_irqsave(&clock->lock, flags); nsec_now = timecounter_cyc2time(&clock->tc, cycles_now); nsec_delta = ns - nsec_now; - cycles_delta = div64_u64(nsec_delta << clock->cycles.shift, - clock->cycles.mult); + cycles_delta = div64_u64(nsec_delta << clock->tc.cc.shift, + clock->tc.cc.mult); write_unlock_irqrestore(&clock->lock, flags); time_stamp = cycles_now + cycles_delta; field_select = MLX5_MTPPS_FS_PIN_MODE | @@ -440,8 +440,8 @@ void mlx5_pps_event(struct mlx5_core_dev *mdev, write_lock_irqsave(&clock->lock, flags); nsec_now = timecounter_cyc2time(&clock->tc, cycles_now); nsec_delta = ns - nsec_now; - cycles_delta = div64_u64(nsec_delta << clock->cycles.shift, - clock->cycles.mult); + cycles_delta = div64_u64(nsec_delta << clock->tc.cc.shift, + clock->tc.cc.mult); clock->pps_info.start[pin] = cycles_now + cycles_delta; schedule_work(&clock->pps_info.out_work); write_unlock_irqrestore(&clock->lock, flags); @@ -454,6 +454,7 @@ void mlx5_pps_event(struct mlx5_core_dev *mdev, void mlx5_init_clock(struct mlx5_core_dev *mdev) { struct mlx5_clock *clock = &mdev->clock; + struct cyclecounter *cc = &clock->tc.cc; u64 ns; u64 frac = 0; u32 dev_freq; @@ -464,21 +465,18 @@ void mlx5_init_clock(struct mlx5_core_dev *mdev) return; } rwlock_init(&clock->lock); - clock->cycles.read = read_internal_timer; - clock->cycles.shift = MLX5_CYCLES_SHIFT; - clock->cycles.mult = clocksource_khz2mult(dev_freq, - clock->cycles.shift); - clock->nominal_c_mult = clock->cycles.mult; - clock->cycles.mask = CLOCKSOURCE_MASK(41); + cc->read = read_internal_timer; + cc->shift = MLX5_CYCLES_SHIFT; + cc->mult = clocksource_khz2mult(dev_freq, cc->shift); + clock->nominal_c_mult = cc->mult; + cc->mask = CLOCKSOURCE_MASK(41); - timecounter_init(&clock->tc, &clock->cycles, - ktime_to_ns(ktime_get_real())); + timecounter_init(&clock->tc, ktime_to_ns(ktime_get_real())); /* Calculate period in seconds to call the overflow watchdog - to make * sure counter is checked at least once every wrap around. */ - ns = cyclecounter_cyc2ns(&clock->cycles, clock->cycles.mask, - frac, &frac); + ns = cyclecounter_cyc2ns(cc, cc->mask, frac, &frac); do_div(ns, NSEC_PER_SEC / 2 / HZ); clock->overflow_period = ns; diff --git a/drivers/net/ethernet/qlogic/qede/qede_ptp.c b/drivers/net/ethernet/qlogic/qede/qede_ptp.c index 9b2280b..95bb8a8 100644 --- a/drivers/net/ethernet/qlogic/qede/qede_ptp.c +++ b/drivers/net/ethernet/qlogic/qede/qede_ptp.c @@ -34,7 +34,6 @@ struct qede_ptp { const struct qed_eth_ptp_ops *ops; struct ptp_clock_info clock_info; - struct cyclecounter cc; struct timecounter tc; struct ptp_clock *clock; struct work_struct work; @@ -132,7 +131,7 @@ static int qede_ptp_settime(struct ptp_clock_info *info, /* Re-init the timecounter */ spin_lock_bh(&ptp->lock); - timecounter_init(&ptp->tc, &ptp->cc, ns); + timecounter_init(&ptp->tc, ns); spin_unlock_bh(&ptp->lock); return 0; @@ -196,7 +195,7 @@ static u64 qede_ptp_read_cc(const struct cyclecounter *cc) u64 phc_cycles; int rc; - ptp = container_of(cc, struct qede_ptp, cc); + ptp = container_of(cc, struct qede_ptp, tc.cc); edev = ptp->edev; rc = ptp->ops->read_cc(edev->cdev, &phc_cycles); if (rc) @@ -428,14 +427,13 @@ static int qede_ptp_init(struct qede_dev *edev, bool init_tc) * unload / load (e.g. MTU change) while it is running. */ if (init_tc) { - memset(&ptp->cc, 0, sizeof(ptp->cc)); - ptp->cc.read = qede_ptp_read_cc; - ptp->cc.mask = CYCLECOUNTER_MASK(64); - ptp->cc.shift = 0; - ptp->cc.mult = 1; - - timecounter_init(&ptp->tc, &ptp->cc, - ktime_to_ns(ktime_get_real())); + memset(&ptp->tc.cc, 0, sizeof(ptp->tc.cc)); + ptp->tc.cc.read = qede_ptp_read_cc; + ptp->tc.cc.mask = CYCLECOUNTER_MASK(64); + ptp->tc.cc.shift = 0; + ptp->tc.cc.mult = 1; + + timecounter_init(&ptp->tc, ktime_to_ns(ktime_get_real())); } return rc; diff --git a/drivers/net/ethernet/ti/cpts.c b/drivers/net/ethernet/ti/cpts.c index e7b76f6..b8fe843 100644 --- a/drivers/net/ethernet/ti/cpts.c +++ b/drivers/net/ethernet/ti/cpts.c @@ -182,7 +182,7 @@ static u64 cpts_systim_read(const struct cyclecounter *cc) u64 val = 0; struct cpts_event *event; struct list_head *this, *next; - struct cpts *cpts = container_of(cc, struct cpts, cc); + struct cpts *cpts = container_of(cc, struct cpts, tc.cc); cpts_write32(cpts, TS_PUSH, ts_push); if (cpts_fifo_read(cpts, CPTS_EV_PUSH)) @@ -224,7 +224,7 @@ static int cpts_ptp_adjfreq(struct ptp_clock_info *ptp, s32 ppb) timecounter_read(&cpts->tc); - cpts->cc.mult = neg_adj ? mult - diff : mult + diff; + cpts->tc.cc.mult = neg_adj ? mult - diff : mult + diff; spin_unlock_irqrestore(&cpts->lock, flags); @@ -268,7 +268,7 @@ static int cpts_ptp_settime(struct ptp_clock_info *ptp, ns = timespec64_to_ns(ts); spin_lock_irqsave(&cpts->lock, flags); - timecounter_init(&cpts->tc, &cpts->cc, ns); + timecounter_init(&cpts->tc, ns); spin_unlock_irqrestore(&cpts->lock, flags); return 0; @@ -447,7 +447,7 @@ int cpts_register(struct cpts *cpts) cpts_write32(cpts, CPTS_EN, control); cpts_write32(cpts, TS_PEND_EN, int_enable); - timecounter_init(&cpts->tc, &cpts->cc, ktime_to_ns(ktime_get_real())); + timecounter_init(&cpts->tc, ktime_to_ns(ktime_get_real())); cpts->clock = ptp_clock_register(&cpts->info, cpts->dev); if (IS_ERR(cpts->clock)) { @@ -486,6 +486,7 @@ void cpts_unregister(struct cpts *cpts) static void cpts_calc_mult_shift(struct cpts *cpts) { + struct cyclecounter *cc = &cpts->tc.cc; u64 frac, maxsec, ns; u32 freq; @@ -494,7 +495,7 @@ static void cpts_calc_mult_shift(struct cpts *cpts) /* Calc the maximum number of seconds which we can run before * wrapping around. */ - maxsec = cpts->cc.mask; + maxsec = cc->mask; do_div(maxsec, freq); /* limit conversation rate to 10 sec as higher values will produce * too small mult factors and so reduce the conversion accuracy @@ -507,18 +508,19 @@ static void cpts_calc_mult_shift(struct cpts *cpts) dev_info(cpts->dev, "cpts: overflow check period %lu (jiffies)\n", cpts->ov_check_period); - if (cpts->cc.mult || cpts->cc.shift) + if (cc->mult || cc->shift) return; - clocks_calc_mult_shift(&cpts->cc.mult, &cpts->cc.shift, + clocks_calc_mult_shift(&cc->mult, &cc->shift, freq, NSEC_PER_SEC, maxsec); frac = 0; - ns = cyclecounter_cyc2ns(&cpts->cc, freq, cpts->cc.mask, &frac); + ns = cyclecounter_cyc2ns(cc, freq, cc->mask, &frac); dev_info(cpts->dev, "CPTS: ref_clk_freq:%u calc_mult:%u calc_shift:%u error:%lld nsec/sec\n", - freq, cpts->cc.mult, cpts->cc.shift, (ns - NSEC_PER_SEC)); + freq, cc->mult, cc->shift, + (ns - NSEC_PER_SEC)); } static int cpts_of_parse(struct cpts *cpts, struct device_node *node) @@ -527,13 +529,13 @@ static int cpts_of_parse(struct cpts *cpts, struct device_node *node) u32 prop; if (!of_property_read_u32(node, "cpts_clock_mult", &prop)) - cpts->cc.mult = prop; + cpts->tc.cc.mult = prop; if (!of_property_read_u32(node, "cpts_clock_shift", &prop)) - cpts->cc.shift = prop; + cpts->tc.cc.shift = prop; - if ((cpts->cc.mult && !cpts->cc.shift) || - (!cpts->cc.mult && cpts->cc.shift)) + if ((cpts->tc.cc.mult && !cpts->tc.cc.shift) || + (!cpts->tc.cc.mult && cpts->tc.cc.shift)) goto of_error; return 0; @@ -569,15 +571,15 @@ struct cpts *cpts_create(struct device *dev, void __iomem *regs, clk_prepare(cpts->refclk); - cpts->cc.read = cpts_systim_read; - cpts->cc.mask = CLOCKSOURCE_MASK(32); + cpts->tc.cc.read = cpts_systim_read; + cpts->tc.cc.mask = CLOCKSOURCE_MASK(32); cpts->info = cpts_info; cpts_calc_mult_shift(cpts); - /* save cc.mult original value as it can be modified + /* save tc.cc.mult original value as it can be modified * by cpts_ptp_adjfreq(). */ - cpts->cc_mult = cpts->cc.mult; + cpts->cc_mult = cpts->tc.cc.mult; return cpts; } diff --git a/drivers/net/ethernet/ti/cpts.h b/drivers/net/ethernet/ti/cpts.h index 73d73fa..a7174eb 100644 --- a/drivers/net/ethernet/ti/cpts.h +++ b/drivers/net/ethernet/ti/cpts.h @@ -117,7 +117,6 @@ struct cpts { struct ptp_clock *clock; spinlock_t lock; /* protects time registers */ u32 cc_mult; /* for the nominal frequency */ - struct cyclecounter cc; struct timecounter tc; int phc_index; struct clk *refclk; diff --git a/include/linux/mlx5/driver.h b/include/linux/mlx5/driver.h index a886b51..c81c615 100644 --- a/include/linux/mlx5/driver.h +++ b/include/linux/mlx5/driver.h @@ -780,7 +780,6 @@ struct mlx5_pps { struct mlx5_clock { rwlock_t lock; - struct cyclecounter cycles; struct timecounter tc; struct hwtstamp_config hwtstamp_config; u32 nominal_c_mult; diff --git a/include/linux/timecounter.h b/include/linux/timecounter.h index 2496ad4..6daca06 100644 --- a/include/linux/timecounter.h +++ b/include/linux/timecounter.h @@ -62,7 +62,7 @@ struct cyclecounter { * @frac: accumulated fractional nanoseconds */ struct timecounter { - const struct cyclecounter *cc; + struct cyclecounter cc; u64 cycle_last; u64 nsec; u64 mask; @@ -98,7 +98,6 @@ static inline void timecounter_adjtime(struct timecounter *tc, s64 delta) /** * timecounter_init - initialize a time counter * @tc: Pointer to time counter which is to be initialized/reset - * @cc: A cycle counter, ready to be used. * @start_tstamp: Arbitrary initial time stamp. * * After this call the current cycle register (roughly) corresponds to @@ -106,7 +105,6 @@ static inline void timecounter_adjtime(struct timecounter *tc, s64 delta) * the time stamp counter by the number of elapsed nanoseconds. */ extern void timecounter_init(struct timecounter *tc, - const struct cyclecounter *cc, u64 start_tstamp); /** diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 68169e3..3061f44 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -445,7 +445,6 @@ struct hdac_stream { unsigned long start_wallclk; /* start + minimum wallclk */ unsigned long period_wallclk; /* wallclk for period */ struct timecounter tc; - struct cyclecounter cc; int delay_negative_threshold; struct list_head list; diff --git a/kernel/time/timecounter.c b/kernel/time/timecounter.c index 8afd789..7919acb 100644 --- a/kernel/time/timecounter.c +++ b/kernel/time/timecounter.c @@ -18,11 +18,10 @@ #include <linux/export.h> #include <linux/timecounter.h> -void timecounter_init(struct timecounter *tc, - const struct cyclecounter *cc, - u64 start_tstamp) +void timecounter_init(struct timecounter *tc, u64 start_tstamp) { - tc->cc = cc; + struct cyclecounter *cc = &tc->cc; + tc->cycle_last = cc->read(cc); tc->nsec = start_tstamp; tc->mask = (1ULL << cc->shift) - 1; @@ -43,17 +42,18 @@ void timecounter_init(struct timecounter *tc, */ static u64 timecounter_read_delta(struct timecounter *tc) { + struct cyclecounter *cc = &tc->cc; u64 cycle_now, cycle_delta; u64 ns_offset; /* read cycle counter: */ - cycle_now = tc->cc->read(tc->cc); + cycle_now = cc->read(cc); /* calculate the delta since the last timecounter_read_delta(): */ - cycle_delta = (cycle_now - tc->cycle_last) & tc->cc->mask; + cycle_delta = (cycle_now - tc->cycle_last) & cc->mask; /* convert to nanoseconds: */ - ns_offset = cyclecounter_cyc2ns(tc->cc, cycle_delta, + ns_offset = cyclecounter_cyc2ns(cc, cycle_delta, tc->mask, &tc->frac); /* update time stamp of timecounter_read_delta() call: */ @@ -89,10 +89,10 @@ static u64 cc_cyc2ns_backwards(const struct cyclecounter *cc, return ns; } -u64 timecounter_cyc2time(struct timecounter *tc, - u64 cycle_tstamp) +u64 timecounter_cyc2time(struct timecounter *tc, u64 cycle_tstamp) { - u64 delta = (cycle_tstamp - tc->cycle_last) & tc->cc->mask; + struct cyclecounter *cc = &tc->cc; + u64 delta = (cycle_tstamp - tc->cycle_last) & cc->mask; u64 nsec = tc->nsec, frac = tc->frac; /* @@ -100,11 +100,11 @@ u64 timecounter_cyc2time(struct timecounter *tc, * than tc->cycle_last, detect when it is too far in the * future and treat it as old time stamp instead. */ - if (delta > tc->cc->mask / 2) { - delta = (tc->cycle_last - cycle_tstamp) & tc->cc->mask; - nsec -= cc_cyc2ns_backwards(tc->cc, delta, tc->mask, frac); + if (delta > cc->mask / 2) { + delta = (tc->cycle_last - cycle_tstamp) & cc->mask; + nsec -= cc_cyc2ns_backwards(cc, delta, tc->mask, frac); } else { - nsec += cyclecounter_cyc2ns(tc->cc, delta, tc->mask, &frac); + nsec += cyclecounter_cyc2ns(cc, delta, tc->mask, &frac); } return nsec; diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index e1472c7..9426c1a 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -467,7 +467,8 @@ int snd_hdac_stream_set_params(struct hdac_stream *azx_dev, static u64 azx_cc_read(const struct cyclecounter *cc) { - struct hdac_stream *azx_dev = container_of(cc, struct hdac_stream, cc); + struct hdac_stream *azx_dev = container_of(cc, struct hdac_stream, + tc.cc); return snd_hdac_chip_readl(azx_dev->bus, WALLCLK); } @@ -476,7 +477,7 @@ static void azx_timecounter_init(struct hdac_stream *azx_dev, bool force, u64 last) { struct timecounter *tc = &azx_dev->tc; - struct cyclecounter *cc = &azx_dev->cc; + struct cyclecounter *cc = &azx_dev->tc.cc; u64 nsec; cc->read = azx_cc_read; @@ -496,7 +497,7 @@ static void azx_timecounter_init(struct hdac_stream *azx_dev, cc->shift = 0; nsec = 0; /* audio time is elapsed time since trigger */ - timecounter_init(tc, cc, nsec); + timecounter_init(tc, nsec); if (force) { /* * force timecounter to use predefined value, diff --git a/virt/kvm/arm/arch_timer.c b/virt/kvm/arm/arch_timer.c index 4151250..f3505bd 100644 --- a/virt/kvm/arm/arch_timer.c +++ b/virt/kvm/arm/arch_timer.c @@ -53,7 +53,7 @@ static void kvm_timer_update_irq(struct kvm_vcpu *vcpu, bool new_level, u64 kvm_phys_timer_read(void) { - return timecounter->cc->read(timecounter->cc); + return timecounter->cc.read(&timecounter->cc); } static void soft_timer_start(struct hrtimer *hrt, u64 ns) @@ -138,7 +138,7 @@ static u64 kvm_timer_compute_delta(struct arch_timer_context *timer_ctx) if (now < cval) { u64 ns; - ns = cyclecounter_cyc2ns(timecounter->cc, + ns = cyclecounter_cyc2ns(&timecounter->cc, cval - now, timecounter->mask, &timecounter->frac); @@ -734,7 +734,7 @@ int kvm_timer_hyp_init(void) info = arch_timer_get_kvm_info(); timecounter = &info->timecounter; - if (!timecounter->cc) { + if (!timecounter->cc.mask) { kvm_err("kvm_arch_timer: uninitialized timecounter\n"); return -ENODEV; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: tfa9879: Export OF device ID as module alias
by Javier Martinez Canillas 01 Dec '17

01 Dec '17

The I2C core always reports a MODALIAS of the form i2c:<foo> even if the device was registered via OF, this means that exporting the OF device ID table device aliases in the module is not needed. But in order to change how the core reports modaliases to user-space, it's better to export it. Before this patch: $ modinfo sound/soc/codecs/snd-soc-tfa9879.ko | grep alias alias: i2c:tfa9879 After this patch: $ modinfo sound/soc/codecs/snd-soc-tfa9879.ko | grep alias alias: i2c:tfa9879 alias: of:N*T*Cnxp,tfa9879C* alias: of:N*T*Cnxp,tfa9879 Signed-off-by: Javier Martinez Canillas <javierm(a)redhat.com> --- sound/soc/codecs/tfa9879.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c index f8dd67ca0744..e7ca764b5729 100644 --- a/sound/soc/codecs/tfa9879.c +++ b/sound/soc/codecs/tfa9879.c @@ -316,6 +316,7 @@ static const struct of_device_id tfa9879_of_match[] = { { .compatible = "nxp,tfa9879", }, { } }; +MODULE_DEVICE_TABLE(of, tfa9879_of_match); static struct i2c_driver tfa9879_i2c_driver = { .driver = { -- 2.14.3

3 2

[alsa-devel] [PATCH 1/5 v5] ASoC: ep93xx-ac97: Fix platform_get_irq's error checking
by Arvind Yadav 01 Dec '17

01 Dec '17

The platform_get_irq() function returns negative if an error occurs. zero or positive number on success. platform_get_irq() error checking for zero is not correct. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- changes in v2 : irq was unsigned. so changed it to signed. changes in v3 : Add failure case '<= 0' instead of '< 0'. IRQ0 is not valid. changes in v4 : Return -ENODEV insted of irq. changes in v5 : Add separate error for irq == 0 and irq < 0. sound/soc/cirrus/ep93xx-ac97.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index bbf7a92..cd5a939 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -365,7 +365,7 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) { struct ep93xx_ac97_info *info; struct resource *res; - unsigned int irq; + int irq; int ret; info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); @@ -378,8 +378,8 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) return PTR_ERR(info->regs); irq = platform_get_irq(pdev, 0); - if (!irq) - return -ENODEV; + if (irq <= 0) + return irq < 0 ? irq : -ENODEV; ret = devm_request_irq(&pdev->dev, irq, ep93xx_ac97_interrupt, IRQF_TRIGGER_HIGH, pdev->name, info); -- 2.7.4

2 9

[alsa-devel] Applied "ASoC: tlv320aic31xx: Reformat header file using GENMASK and BIT macros" to the asoc tree
by Mark Brown 01 Dec '17

01 Dec '17

The patch ASoC: tlv320aic31xx: Reformat header file using GENMASK and BIT macros has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 12eb4d66ba2e14072b54f37f5a4a6f70457e228a Mon Sep 17 00:00:00 2001 From: "Andrew F. Davis" <afd(a)ti.com> Date: Wed, 29 Nov 2017 15:32:47 -0600 Subject: [PATCH] ASoC: tlv320aic31xx: Reformat header file using GENMASK and BIT macros We also move the comments describing the registers to after the register definition to remove non-uniform vertical white-space, this makes cross-referencing with the datasheet much easier. Signed-off-by: Andrew F. Davis <afd(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.h | 320 ++++++++++++++++----------------------- 1 file changed, 133 insertions(+), 187 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 6efea0485392..15ac7cba86fe 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -10,20 +10,21 @@ #define AIC31XX_RATES SNDRV_PCM_RATE_8000_192000 -#define AIC31XX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ - | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE \ - | SNDRV_PCM_FMTBIT_S32_LE) +#define AIC31XX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) - -#define AIC31XX_STEREO_CLASS_D_BIT 0x1 -#define AIC31XX_MINIDSP_BIT 0x2 -#define DAC31XX_BIT 0x4 +#define AIC31XX_STEREO_CLASS_D_BIT BIT(1) +#define AIC31XX_MINIDSP_BIT BIT(2) +#define DAC31XX_BIT BIT(3) enum aic31xx_type { AIC3100 = 0, AIC3110 = AIC31XX_STEREO_CLASS_D_BIT, AIC3120 = AIC31XX_MINIDSP_BIT, - AIC3111 = (AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT), + AIC3111 = AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT, DAC3100 = DAC31XX_BIT, DAC3101 = DAC31XX_BIT | AIC31XX_STEREO_CLASS_D_BIT, }; @@ -36,222 +37,167 @@ struct aic31xx_pdata { #define AIC31XX_REG(page, reg) ((page * 128) + reg) -/* Page Control Register */ -#define AIC31XX_PAGECTL AIC31XX_REG(0, 0) +#define AIC31XX_PAGECTL AIC31XX_REG(0, 0) /* Page Control Register */ /* Page 0 Registers */ -/* Software reset register */ -#define AIC31XX_RESET AIC31XX_REG(0, 1) -/* OT FLAG register */ -#define AIC31XX_OT_FLAG AIC31XX_REG(0, 3) -/* Clock clock Gen muxing, Multiplexers*/ -#define AIC31XX_CLKMUX AIC31XX_REG(0, 4) -/* PLL P and R-VAL register */ -#define AIC31XX_PLLPR AIC31XX_REG(0, 5) -/* PLL J-VAL register */ -#define AIC31XX_PLLJ AIC31XX_REG(0, 6) -/* PLL D-VAL MSB register */ -#define AIC31XX_PLLDMSB AIC31XX_REG(0, 7) -/* PLL D-VAL LSB register */ -#define AIC31XX_PLLDLSB AIC31XX_REG(0, 8) -/* DAC NDAC_VAL register*/ -#define AIC31XX_NDAC AIC31XX_REG(0, 11) -/* DAC MDAC_VAL register */ -#define AIC31XX_MDAC AIC31XX_REG(0, 12) -/* DAC OSR setting register 1, MSB value */ -#define AIC31XX_DOSRMSB AIC31XX_REG(0, 13) -/* DAC OSR setting register 2, LSB value */ -#define AIC31XX_DOSRLSB AIC31XX_REG(0, 14) +#define AIC31XX_RESET AIC31XX_REG(0, 1) /* Software reset register */ +#define AIC31XX_OT_FLAG AIC31XX_REG(0, 3) /* OT FLAG register */ +#define AIC31XX_CLKMUX AIC31XX_REG(0, 4) /* Clock clock Gen muxing, Multiplexers*/ +#define AIC31XX_PLLPR AIC31XX_REG(0, 5) /* PLL P and R-VAL register */ +#define AIC31XX_PLLJ AIC31XX_REG(0, 6) /* PLL J-VAL register */ +#define AIC31XX_PLLDMSB AIC31XX_REG(0, 7) /* PLL D-VAL MSB register */ +#define AIC31XX_PLLDLSB AIC31XX_REG(0, 8) /* PLL D-VAL LSB register */ +#define AIC31XX_NDAC AIC31XX_REG(0, 11) /* DAC NDAC_VAL register*/ +#define AIC31XX_MDAC AIC31XX_REG(0, 12) /* DAC MDAC_VAL register */ +#define AIC31XX_DOSRMSB AIC31XX_REG(0, 13) /* DAC OSR setting register 1, MSB value */ +#define AIC31XX_DOSRLSB AIC31XX_REG(0, 14) /* DAC OSR setting register 2, LSB value */ #define AIC31XX_MINI_DSP_INPOL AIC31XX_REG(0, 16) -/* Clock setting register 8, PLL */ -#define AIC31XX_NADC AIC31XX_REG(0, 18) -/* Clock setting register 9, PLL */ -#define AIC31XX_MADC AIC31XX_REG(0, 19) -/* ADC Oversampling (AOSR) Register */ -#define AIC31XX_AOSR AIC31XX_REG(0, 20) -/* Clock setting register 9, Multiplexers */ -#define AIC31XX_CLKOUTMUX AIC31XX_REG(0, 25) -/* Clock setting register 10, CLOCKOUT M divider value */ -#define AIC31XX_CLKOUTMVAL AIC31XX_REG(0, 26) -/* Audio Interface Setting Register 1 */ -#define AIC31XX_IFACE1 AIC31XX_REG(0, 27) -/* Audio Data Slot Offset Programming */ -#define AIC31XX_DATA_OFFSET AIC31XX_REG(0, 28) -/* Audio Interface Setting Register 2 */ -#define AIC31XX_IFACE2 AIC31XX_REG(0, 29) -/* Clock setting register 11, BCLK N Divider */ -#define AIC31XX_BCLKN AIC31XX_REG(0, 30) -/* Audio Interface Setting Register 3, Secondary Audio Interface */ -#define AIC31XX_IFACESEC1 AIC31XX_REG(0, 31) -/* Audio Interface Setting Register 4 */ -#define AIC31XX_IFACESEC2 AIC31XX_REG(0, 32) -/* Audio Interface Setting Register 5 */ -#define AIC31XX_IFACESEC3 AIC31XX_REG(0, 33) -/* I2C Bus Condition */ -#define AIC31XX_I2C AIC31XX_REG(0, 34) -/* ADC FLAG */ -#define AIC31XX_ADCFLAG AIC31XX_REG(0, 36) -/* DAC Flag Registers */ -#define AIC31XX_DACFLAG1 AIC31XX_REG(0, 37) +#define AIC31XX_NADC AIC31XX_REG(0, 18) /* Clock setting register 8, PLL */ +#define AIC31XX_MADC AIC31XX_REG(0, 19) /* Clock setting register 9, PLL */ +#define AIC31XX_AOSR AIC31XX_REG(0, 20) /* ADC Oversampling (AOSR) Register */ +#define AIC31XX_CLKOUTMUX AIC31XX_REG(0, 25) /* Clock setting register 9, Multiplexers */ +#define AIC31XX_CLKOUTMVAL AIC31XX_REG(0, 26) /* Clock setting register 10, CLOCKOUT M divider value */ +#define AIC31XX_IFACE1 AIC31XX_REG(0, 27) /* Audio Interface Setting Register 1 */ +#define AIC31XX_DATA_OFFSET AIC31XX_REG(0, 28) /* Audio Data Slot Offset Programming */ +#define AIC31XX_IFACE2 AIC31XX_REG(0, 29) /* Audio Interface Setting Register 2 */ +#define AIC31XX_BCLKN AIC31XX_REG(0, 30) /* Clock setting register 11, BCLK N Divider */ +#define AIC31XX_IFACESEC1 AIC31XX_REG(0, 31) /* Audio Interface Setting Register 3, Secondary Audio Interface */ +#define AIC31XX_IFACESEC2 AIC31XX_REG(0, 32) /* Audio Interface Setting Register 4 */ +#define AIC31XX_IFACESEC3 AIC31XX_REG(0, 33) /* Audio Interface Setting Register 5 */ +#define AIC31XX_I2C AIC31XX_REG(0, 34) /* I2C Bus Condition */ +#define AIC31XX_ADCFLAG AIC31XX_REG(0, 36) /* ADC FLAG */ +#define AIC31XX_DACFLAG1 AIC31XX_REG(0, 37) /* DAC Flag Registers */ #define AIC31XX_DACFLAG2 AIC31XX_REG(0, 38) -/* Sticky Interrupt flag (overflow) */ -#define AIC31XX_OFFLAG AIC31XX_REG(0, 39) -/* Sticy DAC Interrupt flags */ -#define AIC31XX_INTRDACFLAG AIC31XX_REG(0, 44) -/* Sticy ADC Interrupt flags */ -#define AIC31XX_INTRADCFLAG AIC31XX_REG(0, 45) -/* DAC Interrupt flags 2 */ -#define AIC31XX_INTRDACFLAG2 AIC31XX_REG(0, 46) -/* ADC Interrupt flags 2 */ -#define AIC31XX_INTRADCFLAG2 AIC31XX_REG(0, 47) -/* INT1 interrupt control */ -#define AIC31XX_INT1CTRL AIC31XX_REG(0, 48) -/* INT2 interrupt control */ -#define AIC31XX_INT2CTRL AIC31XX_REG(0, 49) -/* GPIO1 control */ -#define AIC31XX_GPIO1 AIC31XX_REG(0, 51) - +#define AIC31XX_OFFLAG AIC31XX_REG(0, 39) /* Sticky Interrupt flag (overflow) */ +#define AIC31XX_INTRDACFLAG AIC31XX_REG(0, 44) /* Sticy DAC Interrupt flags */ +#define AIC31XX_INTRADCFLAG AIC31XX_REG(0, 45) /* Sticy ADC Interrupt flags */ +#define AIC31XX_INTRDACFLAG2 AIC31XX_REG(0, 46) /* DAC Interrupt flags 2 */ +#define AIC31XX_INTRADCFLAG2 AIC31XX_REG(0, 47) /* ADC Interrupt flags 2 */ +#define AIC31XX_INT1CTRL AIC31XX_REG(0, 48) /* INT1 interrupt control */ +#define AIC31XX_INT2CTRL AIC31XX_REG(0, 49) /* INT2 interrupt control */ +#define AIC31XX_GPIO1 AIC31XX_REG(0, 51) /* GPIO1 control */ #define AIC31XX_DACPRB AIC31XX_REG(0, 60) -/* ADC Instruction Set Register */ -#define AIC31XX_ADCPRB AIC31XX_REG(0, 61) -/* DAC channel setup register */ -#define AIC31XX_DACSETUP AIC31XX_REG(0, 63) -/* DAC Mute and volume control register */ -#define AIC31XX_DACMUTE AIC31XX_REG(0, 64) -/* Left DAC channel digital volume control */ -#define AIC31XX_LDACVOL AIC31XX_REG(0, 65) -/* Right DAC channel digital volume control */ -#define AIC31XX_RDACVOL AIC31XX_REG(0, 66) -/* Headset detection */ -#define AIC31XX_HSDETECT AIC31XX_REG(0, 67) -/* ADC Digital Mic */ -#define AIC31XX_ADCSETUP AIC31XX_REG(0, 81) -/* ADC Digital Volume Control Fine Adjust */ -#define AIC31XX_ADCFGA AIC31XX_REG(0, 82) -/* ADC Digital Volume Control Coarse Adjust */ -#define AIC31XX_ADCVOL AIC31XX_REG(0, 83) - +#define AIC31XX_ADCPRB AIC31XX_REG(0, 61) /* ADC Instruction Set Register */ +#define AIC31XX_DACSETUP AIC31XX_REG(0, 63) /* DAC channel setup register */ +#define AIC31XX_DACMUTE AIC31XX_REG(0, 64) /* DAC Mute and volume control register */ +#define AIC31XX_LDACVOL AIC31XX_REG(0, 65) /* Left DAC channel digital volume control */ +#define AIC31XX_RDACVOL AIC31XX_REG(0, 66) /* Right DAC channel digital volume control */ +#define AIC31XX_HSDETECT AIC31XX_REG(0, 67) /* Headset detection */ +#define AIC31XX_ADCSETUP AIC31XX_REG(0, 81) /* ADC Digital Mic */ +#define AIC31XX_ADCFGA AIC31XX_REG(0, 82) /* ADC Digital Volume Control Fine Adjust */ +#define AIC31XX_ADCVOL AIC31XX_REG(0, 83) /* ADC Digital Volume Control Coarse Adjust */ /* Page 1 Registers */ -/* Headphone drivers */ -#define AIC31XX_HPDRIVER AIC31XX_REG(1, 31) -/* Class-D Speakear Amplifier */ -#define AIC31XX_SPKAMP AIC31XX_REG(1, 32) -/* HP Output Drivers POP Removal Settings */ -#define AIC31XX_HPPOP AIC31XX_REG(1, 33) -/* Output Driver PGA Ramp-Down Period Control */ -#define AIC31XX_SPPGARAMP AIC31XX_REG(1, 34) -/* DAC_L and DAC_R Output Mixer Routing */ -#define AIC31XX_DACMIXERROUTE AIC31XX_REG(1, 35) -/* Left Analog Vol to HPL */ -#define AIC31XX_LANALOGHPL AIC31XX_REG(1, 36) -/* Right Analog Vol to HPR */ -#define AIC31XX_RANALOGHPR AIC31XX_REG(1, 37) -/* Left Analog Vol to SPL */ -#define AIC31XX_LANALOGSPL AIC31XX_REG(1, 38) -/* Right Analog Vol to SPR */ -#define AIC31XX_RANALOGSPR AIC31XX_REG(1, 39) -/* HPL Driver */ -#define AIC31XX_HPLGAIN AIC31XX_REG(1, 40) -/* HPR Driver */ -#define AIC31XX_HPRGAIN AIC31XX_REG(1, 41) -/* SPL Driver */ -#define AIC31XX_SPLGAIN AIC31XX_REG(1, 42) -/* SPR Driver */ -#define AIC31XX_SPRGAIN AIC31XX_REG(1, 43) -/* HP Driver Control */ -#define AIC31XX_HPCONTROL AIC31XX_REG(1, 44) -/* MIC Bias Control */ -#define AIC31XX_MICBIAS AIC31XX_REG(1, 46) -/* MIC PGA*/ -#define AIC31XX_MICPGA AIC31XX_REG(1, 47) -/* Delta-Sigma Mono ADC Channel Fine-Gain Input Selection for P-Terminal */ -#define AIC31XX_MICPGAPI AIC31XX_REG(1, 48) -/* ADC Input Selection for M-Terminal */ -#define AIC31XX_MICPGAMI AIC31XX_REG(1, 49) -/* Input CM Settings */ -#define AIC31XX_MICPGACM AIC31XX_REG(1, 50) - -/* Bits, masks and shifts */ +#define AIC31XX_HPDRIVER AIC31XX_REG(1, 31) /* Headphone drivers */ +#define AIC31XX_SPKAMP AIC31XX_REG(1, 32) /* Class-D Speakear Amplifier */ +#define AIC31XX_HPPOP AIC31XX_REG(1, 33) /* HP Output Drivers POP Removal Settings */ +#define AIC31XX_SPPGARAMP AIC31XX_REG(1, 34) /* Output Driver PGA Ramp-Down Period Control */ +#define AIC31XX_DACMIXERROUTE AIC31XX_REG(1, 35) /* DAC_L and DAC_R Output Mixer Routing */ +#define AIC31XX_LANALOGHPL AIC31XX_REG(1, 36) /* Left Analog Vol to HPL */ +#define AIC31XX_RANALOGHPR AIC31XX_REG(1, 37) /* Right Analog Vol to HPR */ +#define AIC31XX_LANALOGSPL AIC31XX_REG(1, 38) /* Left Analog Vol to SPL */ +#define AIC31XX_RANALOGSPR AIC31XX_REG(1, 39) /* Right Analog Vol to SPR */ +#define AIC31XX_HPLGAIN AIC31XX_REG(1, 40) /* HPL Driver */ +#define AIC31XX_HPRGAIN AIC31XX_REG(1, 41) /* HPR Driver */ +#define AIC31XX_SPLGAIN AIC31XX_REG(1, 42) /* SPL Driver */ +#define AIC31XX_SPRGAIN AIC31XX_REG(1, 43) /* SPR Driver */ +#define AIC31XX_HPCONTROL AIC31XX_REG(1, 44) /* HP Driver Control */ +#define AIC31XX_MICBIAS AIC31XX_REG(1, 46) /* MIC Bias Control */ +#define AIC31XX_MICPGA AIC31XX_REG(1, 47) /* MIC PGA*/ +#define AIC31XX_MICPGAPI AIC31XX_REG(1, 48) /* Delta-Sigma Mono ADC Channel Fine-Gain Input Selection for P-Terminal */ +#define AIC31XX_MICPGAMI AIC31XX_REG(1, 49) /* ADC Input Selection for M-Terminal */ +#define AIC31XX_MICPGACM AIC31XX_REG(1, 50) /* Input CM Settings */ + +/* Bits, masks, and shifts */ /* AIC31XX_CLKMUX */ -#define AIC31XX_PLL_CLKIN_MASK 0x0c -#define AIC31XX_PLL_CLKIN_SHIFT 2 -#define AIC31XX_PLL_CLKIN_MCLK 0 -#define AIC31XX_CODEC_CLKIN_MASK 0x03 -#define AIC31XX_CODEC_CLKIN_SHIFT 0 -#define AIC31XX_CODEC_CLKIN_PLL 3 -#define AIC31XX_CODEC_CLKIN_BCLK 1 - -/* AIC31XX_PLLPR, AIC31XX_NDAC, AIC31XX_MDAC, AIC31XX_NADC, AIC31XX_MADC, - AIC31XX_BCLKN */ -#define AIC31XX_PLL_MASK 0x7f -#define AIC31XX_PM_MASK 0x80 +#define AIC31XX_PLL_CLKIN_MASK GENMASK(3, 2) +#define AIC31XX_PLL_CLKIN_SHIFT (2) +#define AIC31XX_PLL_CLKIN_MCLK 0x00 +#define AIC31XX_PLL_CLKIN_BCKL 0x01 +#define AIC31XX_PLL_CLKIN_GPIO1 0x02 +#define AIC31XX_PLL_CLKIN_DIN 0x03 +#define AIC31XX_CODEC_CLKIN_MASK GENMASK(1, 0) +#define AIC31XX_CODEC_CLKIN_SHIFT (0) +#define AIC31XX_CODEC_CLKIN_MCLK 0x00 +#define AIC31XX_CODEC_CLKIN_BCLK 0x01 +#define AIC31XX_CODEC_CLKIN_GPIO1 0x02 +#define AIC31XX_CODEC_CLKIN_PLL 0x03 + +/* AIC31XX_PLLPR */ +/* AIC31XX_NDAC */ +/* AIC31XX_MDAC */ +/* AIC31XX_NADC */ +/* AIC31XX_MADC */ +/* AIC31XX_BCLKN */ +#define AIC31XX_PLL_MASK GENMASK(6, 0) +#define AIC31XX_PM_MASK BIT(7) /* AIC31XX_IFACE1 */ -#define AIC31XX_WORD_LEN_16BITS 0x00 -#define AIC31XX_WORD_LEN_20BITS 0x01 -#define AIC31XX_WORD_LEN_24BITS 0x02 -#define AIC31XX_WORD_LEN_32BITS 0x03 -#define AIC31XX_IFACE1_DATALEN_MASK 0x30 -#define AIC31XX_IFACE1_DATALEN_SHIFT (4) -#define AIC31XX_IFACE1_DATATYPE_MASK 0xC0 +#define AIC31XX_IFACE1_DATATYPE_MASK GENMASK(7, 6) #define AIC31XX_IFACE1_DATATYPE_SHIFT (6) #define AIC31XX_I2S_MODE 0x00 #define AIC31XX_DSP_MODE 0x01 #define AIC31XX_RIGHT_JUSTIFIED_MODE 0x02 #define AIC31XX_LEFT_JUSTIFIED_MODE 0x03 -#define AIC31XX_IFACE1_MASTER_MASK 0x0C -#define AIC31XX_BCLK_MASTER 0x08 -#define AIC31XX_WCLK_MASTER 0x04 +#define AIC31XX_IFACE1_DATALEN_MASK GENMASK(5, 4) +#define AIC31XX_IFACE1_DATALEN_SHIFT (4) +#define AIC31XX_WORD_LEN_16BITS 0x00 +#define AIC31XX_WORD_LEN_20BITS 0x01 +#define AIC31XX_WORD_LEN_24BITS 0x02 +#define AIC31XX_WORD_LEN_32BITS 0x03 +#define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2) +#define AIC31XX_BCLK_MASTER BIT(2) +#define AIC31XX_WCLK_MASTER BIT(3) /* AIC31XX_DATA_OFFSET */ -#define AIC31XX_DATA_OFFSET_MASK 0xFF +#define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0) /* AIC31XX_IFACE2 */ -#define AIC31XX_BCLKINV_MASK 0x08 -#define AIC31XX_BDIVCLK_MASK 0x03 +#define AIC31XX_BCLKINV_MASK BIT(3) +#define AIC31XX_BDIVCLK_MASK GENMASK(1, 0) #define AIC31XX_DAC2BCLK 0x00 #define AIC31XX_DACMOD2BCLK 0x01 #define AIC31XX_ADC2BCLK 0x02 #define AIC31XX_ADCMOD2BCLK 0x03 /* AIC31XX_ADCFLAG */ -#define AIC31XX_ADCPWRSTATUS_MASK 0x40 +#define AIC31XX_ADCPWRSTATUS_MASK BIT(6) /* AIC31XX_DACFLAG1 */ -#define AIC31XX_LDACPWRSTATUS_MASK 0x80 -#define AIC31XX_RDACPWRSTATUS_MASK 0x08 -#define AIC31XX_HPLDRVPWRSTATUS_MASK 0x20 -#define AIC31XX_HPRDRVPWRSTATUS_MASK 0x02 -#define AIC31XX_SPLDRVPWRSTATUS_MASK 0x10 -#define AIC31XX_SPRDRVPWRSTATUS_MASK 0x01 +#define AIC31XX_LDACPWRSTATUS_MASK BIT(7) +#define AIC31XX_HPLDRVPWRSTATUS_MASK BIT(5) +#define AIC31XX_SPLDRVPWRSTATUS_MASK BIT(4) +#define AIC31XX_RDACPWRSTATUS_MASK BIT(3) +#define AIC31XX_HPRDRVPWRSTATUS_MASK BIT(1) +#define AIC31XX_SPRDRVPWRSTATUS_MASK BIT(0) /* AIC31XX_INTRDACFLAG */ -#define AIC31XX_HPSCDETECT_MASK 0x80 -#define AIC31XX_BUTTONPRESS_MASK 0x20 -#define AIC31XX_HSPLUG_MASK 0x10 -#define AIC31XX_LDRCTHRES_MASK 0x08 -#define AIC31XX_RDRCTHRES_MASK 0x04 -#define AIC31XX_DACSINT_MASK 0x02 -#define AIC31XX_DACAINT_MASK 0x01 +#define AIC31XX_HPLSCDETECT BIT(7) +#define AIC31XX_HPRSCDETECT BIT(6) +#define AIC31XX_BUTTONPRESS BIT(5) +#define AIC31XX_HSPLUG BIT(4) +#define AIC31XX_LDRCTHRES BIT(3) +#define AIC31XX_RDRCTHRES BIT(2) +#define AIC31XX_DACSINT BIT(1) +#define AIC31XX_DACAINT BIT(0) /* AIC31XX_INT1CTRL */ -#define AIC31XX_HSPLUGDET_MASK 0x80 -#define AIC31XX_BUTTONPRESSDET_MASK 0x40 -#define AIC31XX_DRCTHRES_MASK 0x20 -#define AIC31XX_AGCNOISE_MASK 0x10 -#define AIC31XX_OC_MASK 0x08 -#define AIC31XX_ENGINE_MASK 0x04 +#define AIC31XX_HSPLUGDET BIT(7) +#define AIC31XX_BUTTONPRESSDET BIT(6) +#define AIC31XX_DRCTHRES BIT(5) +#define AIC31XX_AGCNOISE BIT(4) +#define AIC31XX_SC BIT(3) +#define AIC31XX_ENGINE BIT(2) /* AIC31XX_DACSETUP */ -#define AIC31XX_SOFTSTEP_MASK 0x03 +#define AIC31XX_SOFTSTEP_MASK GENMASK(1, 0) /* AIC31XX_DACMUTE */ -#define AIC31XX_DACMUTE_MASK 0x0C +#define AIC31XX_DACMUTE_MASK GENMASK(3, 2) /* AIC31XX_MICBIAS */ -#define AIC31XX_MICBIAS_MASK 0x03 -#define AIC31XX_MICBIAS_SHIFT 0 +#define AIC31XX_MICBIAS_MASK GENMASK(1, 0) +#define AIC31XX_MICBIAS_SHIFT 0 #endif /* _TLV320AIC31XX_H */ -- 2.15.0

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[alsa-devel] [PATCH 00/17] Add Headphone Detection to TLV320AIC31xx Driver
by Andrew F. Davis 01 Dec '17

01 Dec '17

Hello all, This series has the end goal of adding headphone detection to the tlv320aic31xx driver. The first few patches are mostly cleanups. Then a couple bug fixes I noticed. Followed by adding interrupt handling and finally headphone detection. The last two, as their commit name recommends, should not be taken and are included in-case someone wants to evaluate headphone detection using the EVM for this device wired to a Beaglebone Black. This series (or at least patch #5) depend on this DT fix[0]. Thanks, Andrew [0]https://www.spinics.net/lists/kernel/msg2644651.html Andrew F. Davis (17): ASoC: tlv320aic31xx: General source formatting cleanup ASoC: tlv320aic31xx: Reformat header file using GENMASK and BIT macros ASoC: tlv320aic31xx: Fix GPIO1 register definition ASoC: tlv320aic31xx: Merge init function into probe ASoC: tlv320aic31xx: Switch GPIO handling to use gpiod_* API ASoC: tlv320aic31xx: Remove platform data ASoC: tlv320aic31xx: Add MICBIAS off setting ASoC: tlv320aic31xx: Check clock and divider before division ASoC: tlv320aic31xx: Add CODEC clock slave support ASoC: tlv320aic31xx: Fix inverted BCLK handling ASoC: tlv320aic31xx: Reset registers during probe ASoC: tlv320aic31xx: Add short circuit detection support ASoC: tlv320aic31xx: Add overflow detection support ASoC: tlv320aic31xx: Add headphone/headset detection ASoC: tlv320aic31xx: Add button press detection NOT FOR MERGING: Add TLV320DAC3101 to BBB for testing NOT FOR MERGING: Add demo jack detection policy for testing .../devicetree/bindings/sound/tlv320aic31xx.txt | 1 + arch/arm/boot/dts/am335x-boneblack.dts | 106 +++++ include/dt-bindings/sound/tlv320aic31xx-micbias.h | 1 + sound/soc/codecs/tlv320aic31xx.c | 400 +++++++++++----- sound/soc/codecs/tlv320aic31xx.h | 504 ++++++++++----------- 5 files changed, 623 insertions(+), 389 deletions(-) rewrite sound/soc/codecs/tlv320aic31xx.h (83%) -- 2.15.0

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[alsa-devel] Guarantee a 0-terminated string for all usages of strncpy()
by Wischer, Timo (ADITG/ESB) 01 Dec '17

01 Dec '17

Hi all, see attached a patch for ensuring a 0-terminated string when using strncpy(). I observed this issue in the ext control implementation. Best regards Timo Wischer Advanced Driver Information Technology GmbH Engineering Software Base (ADITG/ESB) Robert-Bosch-Str. 200 31139 Hildesheim Germany Tel. +49 5121 49 6938 Fax +49 5121 49 6999 twischer(a)de.adit-jv.com ADIT is a joint venture company of Robert Bosch GmbH/Robert Bosch Car Multimedia GmbH and DENSO Corporation Sitz: Hildesheim, Registergericht: Amtsgericht Hildesheim HRB 3438 Geschäftsführung: Wilhelm Grabow, Ken Yaguchi

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Re: [alsa-devel] [PATCH RESEND 1/2] ASoC: Intel: kbl: Remove option of choosing CH count based on pdata
by Mark Brown 01 Dec '17

01 Dec '17

On Thu, Nov 30, 2017 at 11:44:43AM +0530, Naveen M wrote: > Regarding the Query: > As Vinod mentioned, I'm working on Chrome OS user-space stack that uses > customized audio server "CRAS". > These patches are verified on a board that has only 2 dmics. > > Giving the control to CRAS, enables to select the desired dmic using > channel-map configuration in user-space. > Ex:Rear/Front mic selection for World & Front facing camera scenario People do run non-ChromeOS installs on their Chromebooks so we can't just assume that there's nobody out there with a regular distro on this hardware - a user like this reported the problems with the ABI for topology files breaking for example.

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[alsa-devel] [PATCH v3] ASoC: ak4613: call dummy write for PW_MGMT1/3 when Playback
by Kuninori Morimoto 01 Dec '17

01 Dec '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Power Down Release Command (PMVR, PMDAC, RSTN, PMDA1-PMDA6) which are located on PW_MGMT1 / PW_MGMT3 register must be write again after at least 5 LRCK cycle or later on each command. Otherwise, Playback volume will be 0dB. Basically, it should be 1. PowerDownRelease by Power Management1 <= call 1.x after 5LRCK 1.x Dummy write to Power Management1 2. PowerDownRelease by Power Management3 <= call 2.x after 5LRCK 2.x Dummy write to Power Management3 To avoid too many dummy write, this patch is merging these. 1. PowerDownRelease by Power Management1 2. PowerDownRelease by Power Management3 <= call after 5LRCK 2.x Dummy write to Power Management1/3 <= merge dummy write This patch adds dummy write when Playback Start timing. Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v2 -> v3 - use delayed_work with 0 delay Sakamoto-san This patch uses delayed_work with 0 delay, thus, call-back function will be called imminently. There was 0 jiffies delay between trigger <-> callback in my test. sound/soc/codecs/ak4613.c | 60 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 60 insertions(+) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index ee9e822..a7856d5 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -15,6 +15,7 @@ */ #include <linux/clk.h> +#include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -95,6 +96,9 @@ struct ak4613_priv { struct mutex lock; const struct ak4613_interface *iface; struct snd_pcm_hw_constraint_list constraint; + struct delayed_work dummy_write_work; + struct snd_soc_component *component; + unsigned int rate; unsigned int sysclk; unsigned int fmt; @@ -392,6 +396,7 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } + priv->rate = rate; /* * FIXME @@ -467,11 +472,65 @@ static int ak4613_set_bias_level(struct snd_soc_component *component, return 0; } +static void ak4613_dummy_write(struct work_struct *work) +{ + struct ak4613_priv *priv = container_of(work, + struct ak4613_priv, + dummy_write_work.work); + struct snd_soc_component *component = priv->component; + unsigned int mgmt1; + unsigned int mgmt3; + + /* wait 5 LR clocks */ + udelay(5000000 / priv->rate); + + snd_soc_component_read(component, PW_MGMT1, &mgmt1); + snd_soc_component_read(component, PW_MGMT3, &mgmt3); + + snd_soc_component_write(component, PW_MGMT1, mgmt1); + snd_soc_component_write(component, PW_MGMT3, mgmt3); +} + +static int ak4613_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + /* + * FIXME + * + * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks + * from Power Down Release. Otherwise, Playback volume will be 0dB. + * To avoid complex multiple delay/dummy_write method from + * ak4613_set_bias_level() / SND_SOC_DAPM_DAC_E("DACx", ...), + * call it once here. + * + * But, unfortunately, we can't "write" here because here is atomic + * context (It uses I2C access for write). + * Thus, use delayed work with 0 delay to switch to normal context + * immediately, and wait 5 LR clocks and do dummy_write there. + */ + + if ((cmd != SNDRV_PCM_TRIGGER_START) && + (cmd != SNDRV_PCM_TRIGGER_RESUME)) + return 0; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + + priv->component = &codec->component; + schedule_delayed_work(&priv->dummy_write_work, 0); + + return 0; +} + static const struct snd_soc_dai_ops ak4613_dai_ops = { .startup = ak4613_dai_startup, .shutdown = ak4613_dai_shutdown, .set_sysclk = ak4613_dai_set_sysclk, .set_fmt = ak4613_dai_set_fmt, + .trigger = ak4613_dai_trigger, .hw_params = ak4613_dai_hw_params, }; @@ -591,6 +650,7 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, priv->iface = NULL; priv->cnt = 0; priv->sysclk = 0; + INIT_DELAYED_WORK(&priv->dummy_write_work, ak4613_dummy_write); mutex_init(&priv->lock); -- 1.9.1

2 3

[alsa-devel] Doc.../sound/soc/platform.rst: snd_soc_platform_driver out of sync
by Andrey Utkin 01 Dec '17

01 Dec '17

Hi maintainers of Sound and/or Docs. Documentation/sound/soc/platform.rst quotes outdated definition of "struct snd_soc_platform_driver". I spotted this because it shows like there are suspend/resume function pointers while this is no longer true. I don't have a good idea how to update the doc, I am just sure that mere update of the struct definition is not the way to go. Hope this report helps. Thanks for you work!

1 0

[alsa-devel] [PATCH v4 2/2] ASoC: fsl_ssi: add 20-bit sample format for AC'97 and use it for capture
by Maciej S. Szmigiero 01 Dec '17

01 Dec '17

When testing AC'97 capture on UDOO board (currently the only user of fsl_ssi driver in the AC'97 mode) it become obvious that there is a massive distortion above certain, small input signal. This problem has been traced to silicon errata ERR003778: "In AC97, 16-bit mode, received data is shifted by 4-bit locations" that has "No fix scheduled". This errata suggests a workaround of doing a 4-bit shift back in SDMA script for this specific operation mode, however our SDMA scripts are shared between various SoC peripherals so we can't really modify them. There is a simple way to avoid this problem, however, that is to disallow recording in 16-bit mode and only support it in AC'97-native 20-bit mode. We have to use a 4-byte format for this since SSI FIFOs do not allow 3-byte accesses (and these aren't supported by imx-sdma driver anyway). With this change the capture distortion is gone. We can also add this format as an additional one supported for playback, using this opportunity to make sure that we use CPU-endian-native formats in AC'97 mode as we already do in I2S mode. There is no problem in using different bit widths in playback and capture in AC'97 mode so allow this, too. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: Adapt format name to changes in the first patch from this series. Changes from v2, v3: None. sound/soc/fsl/fsl_ssi.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 20ef09e1a395..c350117c8e31 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1278,14 +1278,15 @@ static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S20, }, .capture = { .stream_name = "AC97 Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + /* 16-bit capture is broken (errata ERR003778) */ + .formats = SNDRV_PCM_FMTBIT_S20, }, .ops = &fsl_ssi_dai_ops, }; @@ -1557,11 +1558,12 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - if (!fsl_ssi_is_ac97(ssi_private)) + if (!fsl_ssi_is_ac97(ssi_private)) { ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_samplebits = 1; + } ssi_private->cpu_dai_drv.symmetric_channels = 1; - ssi_private->cpu_dai_drv.symmetric_samplebits = 1; } /* Determine the FIFO depth. */

2 5

[alsa-devel] [PATCH v2] ASoC: ak4613: call dummy write for PW_MGMT1/3 when Playback
by Kuninori Morimoto 01 Dec '17

01 Dec '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Power Down Release Command (PMVR, PMDAC, RSTN, PMDA1-PMDA6) which are located on PW_MGMT1 / PW_MGMT3 register must be write again after at least 5 LRCK cycle or later on each command. Otherwise, Playback volume will be 0dB. Basically, it should be 1. PowerDownRelease by Power Management1 <= call 1.x after 5LRCK 1.x Dummy write to Power Management1 2. PowerDownRelease by Power Management3 <= call 2.x after 5LRCK 2.x Dummy write to Power Management3 To avoid too many dummy write, this patch is merging these. 1. PowerDownRelease by Power Management1 2. PowerDownRelease by Power Management3 <= call after 5LRCK 2.x Dummy write to Power Management1/3 <= merge dummy write This patch adds dummy write when Playback Start timing. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - add more explain on log area - call dummy write after 5LRCK sound/soc/codecs/ak4613.c | 59 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 59 insertions(+) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index ee9e822..937cff4 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -15,6 +15,7 @@ */ #include <linux/clk.h> +#include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -95,6 +96,9 @@ struct ak4613_priv { struct mutex lock; const struct ak4613_interface *iface; struct snd_pcm_hw_constraint_list constraint; + struct delayed_work dummy_write_work; + struct snd_soc_component *component; + unsigned int rate; unsigned int sysclk; unsigned int fmt; @@ -392,6 +396,7 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } + priv->rate = rate; /* * FIXME @@ -467,11 +472,64 @@ static int ak4613_set_bias_level(struct snd_soc_component *component, return 0; } +static void ak4613_dummy_write(struct work_struct *work) +{ + struct ak4613_priv *priv = container_of(work, + struct ak4613_priv, + dummy_write_work.work); + struct snd_soc_component *component = priv->component; + unsigned int mgmt1; + unsigned int mgmt3; + + snd_soc_component_read(component, PW_MGMT1, &mgmt1); + snd_soc_component_read(component, PW_MGMT3, &mgmt3); + + snd_soc_component_write(component, PW_MGMT1, mgmt1); + snd_soc_component_write(component, PW_MGMT3, mgmt3); +} + +static int ak4613_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned long delay; + + /* + * FIXME + * + * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks + * from Power Down Release. Otherwise, Playback volume will be 0dB. + * To avoid complex multiple delayed_work call from + * ak4613_set_bias_level() / SND_SOC_DAPM_DAC_E("DACx", ...), + * call it once here. + */ + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + break; + default: + return 0; + } + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + + delay = 5000000 / priv->rate; + priv->component = &codec->component; + schedule_delayed_work(&priv->dummy_write_work, + usecs_to_jiffies(delay)); + + return 0; +} + static const struct snd_soc_dai_ops ak4613_dai_ops = { .startup = ak4613_dai_startup, .shutdown = ak4613_dai_shutdown, .set_sysclk = ak4613_dai_set_sysclk, .set_fmt = ak4613_dai_set_fmt, + .trigger = ak4613_dai_trigger, .hw_params = ak4613_dai_hw_params, }; @@ -591,6 +649,7 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, priv->iface = NULL; priv->cnt = 0; priv->sysclk = 0; + INIT_DELAYED_WORK(&priv->dummy_write_work, ak4613_dummy_write); mutex_init(&priv->lock); -- 1.9.1

2 2

[alsa-devel] [PATCH v3 00/15] soundwire: Add a new SoundWire subsystem
by Vinod Koul 30 Nov '17

30 Nov '17

This patch series adds a new SoundWire subsystem which implements a new MIPI bus protocol 'SoundWire'. The SoundWire protocol is a robust, scalable, low complexity, low power, low latency, two-pin (clock and data) multi-drop bus that allows for the transfer of multiple audio streams and embedded control/commands. SoundWire provides synchronization capabilities and supports both PCM and PDM, multichannel data, isochronous and asynchronous modes. This series adds SoundWire Bus, IO transfers, DisCo (Discovery and Configuration) sysfs interface, regmap and Documentation summary This patch series is also available on git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git topic/patch_v3 v2: https://lkml.org/lkml/2017/11/10/216 v1: https://lkml.org/lkml/2017/10/18/1030 RFC: https://lkml.org/lkml/2016/10/21/395 Changes in v3: - Update the kernel-doc styles and fix included headers for files - handle dev_pm_domain_attach() for defered probe - remove OF placeholders - change regmap license to GPLv2 only Changes in v2: - move documentation into driver-api and do rst conversion - fix documentation comments - add SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) to all source files - rework the transfer logic and paging logic as commented on v1 - remove dummy sysfs fns - registration checks and fixes - remove slave check for regamp as that turned superfluous - remove depends SoundWire symbol - make modalias api arg const - use bitmap for tracking assigned - add counter for report present tracking todo: add the dt-bindings Sanyog Kale (4): Documentation: Add SoundWire summary soundwire: Add SoundWire MIPI defined registers soundwire: Add Slave status handling helpers soundwire: cdns: Add sdw_master_ops and IO transfer support Vinod Koul (11): soundwire: Add SoundWire bus type soundwire: Add Master registration soundwire: Add MIPI DisCo property helpers soundwire: Add IO transfer regmap: Add SoundWire bus support soundwire: Add slave status handling soundwire: Add sysfs for SoundWire DisCo properties soundwire: cdns: Add cadence library soundwire: intel: Add Intel Master driver soundwire: intel: Add Intel init module MAINTAINERS: Add SoundWire entry Documentation/driver-api/index.rst | 1 + Documentation/driver-api/soundwire/index.rst | 15 + Documentation/driver-api/soundwire/summary.rst | 205 +++++ MAINTAINERS | 10 + drivers/Kconfig | 2 + drivers/Makefile | 1 + drivers/base/regmap/Kconfig | 4 + drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-sdw.c | 104 +++ drivers/soundwire/Kconfig | 37 + drivers/soundwire/Makefile | 18 + drivers/soundwire/bus.c | 1006 ++++++++++++++++++++++++ drivers/soundwire/bus.h | 123 +++ drivers/soundwire/bus_type.c | 241 ++++++ drivers/soundwire/cadence_master.c | 797 +++++++++++++++++++ drivers/soundwire/cadence_master.h | 95 +++ drivers/soundwire/intel.c | 395 ++++++++++ drivers/soundwire/intel.h | 71 ++ drivers/soundwire/intel_init.c | 246 ++++++ drivers/soundwire/mipi_disco.c | 422 ++++++++++ drivers/soundwire/slave.c | 162 ++++ drivers/soundwire/sysfs.c | 390 +++++++++ include/linux/mod_devicetable.h | 6 + include/linux/regmap.h | 37 + include/linux/soundwire/sdw.h | 542 +++++++++++++ include/linux/soundwire/sdw_intel.h | 71 ++ include/linux/soundwire/sdw_registers.h | 242 ++++++ scripts/mod/devicetable-offsets.c | 4 + scripts/mod/file2alias.c | 15 + 29 files changed, 5263 insertions(+) create mode 100644 Documentation/driver-api/soundwire/index.rst create mode 100644 Documentation/driver-api/soundwire/summary.rst create mode 100644 drivers/base/regmap/regmap-sdw.c create mode 100644 drivers/soundwire/Kconfig create mode 100644 drivers/soundwire/Makefile create mode 100644 drivers/soundwire/bus.c create mode 100644 drivers/soundwire/bus.h create mode 100644 drivers/soundwire/bus_type.c create mode 100644 drivers/soundwire/cadence_master.c create mode 100644 drivers/soundwire/cadence_master.h create mode 100644 drivers/soundwire/intel.c create mode 100644 drivers/soundwire/intel.h create mode 100644 drivers/soundwire/intel_init.c create mode 100644 drivers/soundwire/mipi_disco.c create mode 100644 drivers/soundwire/slave.c create mode 100644 drivers/soundwire/sysfs.c create mode 100644 include/linux/soundwire/sdw.h create mode 100644 include/linux/soundwire/sdw_intel.h create mode 100644 include/linux/soundwire/sdw_registers.h -- 2.7.4

3 19

[alsa-devel] [PATCH v2 0/8] Remove use of "gpio-reset" from DT
by Andrew F. Davis 30 Nov '17

30 Nov '17

Hello all, I was working on fixing up the tlv320aic31xx driver when I noticed its gpio reset was not working, it was due to this driver looking for "gpio-reset" instead of the usual "reset-gpio". A quick check shows this mistake is rare and only copied by one other audio CODECs: $ git grep "reset-gpio" | wc -l 630 $ git grep "gpio-reset" | wc -l 6 Luckliy the two effected drivers only use this reset line when power has been cut to the device, so not only were these optional properties but they had no real functional effect anyway. Lets just fix this before is spreads to drivers were it matters. I've also added fixes tags to each patch so it can be individually back-ported to where this bug was introduced if one wanted to. I'm hoping this can sit on next for a while to get the most testing, just in case I'm wrong about this not breaking anything. Thanks, Andrew Changes from v1: - Move old property to "Deprecated properties" section in binding - Keep backwards compatibility for now in driver with warning - Re-order patches to keep series bisectable - Use "reset-gpios" over "reset-gpio" - Drop taken patch - Rebased on v4.15-rc1 Andrew F. Davis (8): ASoC: tlv320aic31xx: Fix typo in DT binding documentation ASoC: tlv320aic3x: Fix typo in DT binding documentation ASoC: tlv320aic31xx: Fix the reset GPIO OF name ASoC: tlv320aic3x: Fix the reset GPIO OF name ARM: dts: am335x-pepper: Fix the audio CODEC's reset pin ARM: dts: imx6: RDU2: Fix the audio CODEC's reset pin ARM: dts: imx: Fix the audio CODEC's reset pin ARM: dts: omap3-n900: Fix the audio CODEC's reset pin Documentation/devicetree/bindings/sound/tlv320aic31xx.txt | 9 ++++++++- Documentation/devicetree/bindings/sound/tlv320aic3x.txt | 10 +++++++++- arch/arm/boot/dts/am335x-pepper.dts | 2 +- arch/arm/boot/dts/imx6qdl-gw5903.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-zii-rdu2.dtsi | 4 ++-- arch/arm/boot/dts/omap3-n900.dts | 4 ++-- sound/soc/codecs/tlv320aic31xx.c | 11 +++++++++-- sound/soc/codecs/tlv320aic3x.c | 15 +++++++++++---- 8 files changed, 43 insertions(+), 14 deletions(-) -- 2.15.0

3 18

[alsa-devel] [PATCH] ALSA: usb-audio: Change the semantics of the enable option
by Tamaki Nishino 30 Nov '17

30 Nov '17

This patch changes the semantics of the enable option for snd-usb-audio in order to allow users to disable a device specified by either or both of the vendor id and the product id. Signed-off-by: Tamaki Nishino <otamachan(a)gmail.com> --- sound/usb/card.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 23d1d23..8018d56 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -585,15 +585,24 @@ static int usb_audio_probe(struct usb_interface *intf, * now look for an empty slot and create a new card instance */ for (i = 0; i < SNDRV_CARDS; i++) - if (enable[i] && ! usb_chip[i] && + if (!usb_chip[i] && (vid[i] == -1 || vid[i] == USB_ID_VENDOR(id)) && (pid[i] == -1 || pid[i] == USB_ID_PRODUCT(id))) { - err = snd_usb_audio_create(intf, dev, i, quirk, - id, &chip); - if (err < 0) + if (enable[i]) { + err = snd_usb_audio_create(intf, dev, i, quirk, + id, &chip); + if (err < 0) + goto __error; + chip->pm_intf = intf; + break; + } else if (vid[i] != -1 || pid[i] != -1) { + dev_info(&dev->dev, + "device (%04x:%04x) is disabled\n", + USB_ID_VENDOR(id), + USB_ID_PRODUCT(id)); + err = -ENOENT; goto __error; - chip->pm_intf = intf; - break; + } } if (!chip) { dev_err(&dev->dev, "no available usb audio device\n"); -- 2.7.4

2 1

[alsa-devel] [RFC PATCH v2 0/7] Fix Intel audio Kconfig issues
by Pierre-Louis Bossart 30 Nov '17

30 Nov '17

The first patch implements what Linus, Takashi and Mark requested: a top-level selector defaulting to 'y' to easily filter all other options and with no impact on code generation. There should be no functionality change and will avoid breaking audio for people using make oldnoconfig. The rest of the patch series does a more in-depth cleanup. It was tested on Baytrail/Cherrytrail/Skylake platforms with no regressions observed and no reports of any compilation issues with 0-day or randconfig. The 4th patch is really the most important one, there were nested configs which made no sense to me. I don't know the history which led to such complicated stuff but simpler is better. The last 3 patches are just clean-ups of the machine driver configs, for some reason there is no consistency in the settings so I tried to apply common sense and use the same rules. At Andy Shevchenko's suggestion, I also replaced the broken dependency on X86_INTEL_LPSS by MFD_INTEL_LPSS for Skylake+ machines, which looks ok but needs to be tested further, hence the RFCv2 status (this change is fine for Chrome, Ubuntu and other distros but better be cautious with a second set of tests with randconfig and on Skylake) Changes since initial RFC: Removed default n Added help text for HASWELL, BAYTRAIL (legacy) and SKYLAKE options Made top level machine driver selection dependent on INTEL_SST_TOPLEVEL. Added help text for PCI and HIFI2 platforms Replaced X86_INTEL_LPSS by MFD_INTEL_LPSS for Skylake+ devices Fixed a couple of indentation issues Pierre-Louis Bossart (7): ASoC: Intel: Fix Kconfig with top-level selector ASoC: Intel: Kconfig: Simplify-clarify ACPI/PCI dependencies ASoC: Intel: document what Kconfig options do ASoC: Intel: Fix nested/unnecessary Kconfig dependencies ASoC: Intel: boards: align Kconfig dependencies for Haswell/Broadwell ASoC: Intel: boards: align Kconfig configurations for HiFi2 ASoC: Intel: boards: align/fix SKL/BXT/KBL Kconfigs sound/soc/intel/Kconfig | 118 +++++++++++++++++++++++-------- sound/soc/intel/boards/Kconfig | 155 ++++++++++++++++++++++------------------- 2 files changed, 172 insertions(+), 101 deletions(-) -- 2.14.1

3 13

[alsa-devel] [PATCH] ASoC: codecs: msm8916-wcd: Fix supported formats
by srinivas.kandagatla＠linaro.org 30 Nov '17

30 Nov '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This codec is configurable for only 16 bit and 32 bit samples, so reflect this in the supported formats also remove 24bit sample from supported list. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- sound/soc/codecs/msm8916-wcd-digital.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 5f3c42c4f74a..066ea2f4ce7b 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -267,7 +267,7 @@ #define MSM8916_WCD_ANALOG_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000) #define MSM8916_WCD_ANALOG_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S32_LE) static int btn_mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_BTN_4; diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index a10a724eb448..13354d6304a8 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -194,7 +194,7 @@ SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_48000) #define MSM8916_WCD_DIGITAL_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S32_LE) struct msm8916_wcd_digital_priv { struct clk *ahbclk, *mclk; @@ -645,7 +645,7 @@ static int msm8916_wcd_digital_hw_params(struct snd_pcm_substream *substream, RX_I2S_CTL_RX_I2S_MODE_MASK, RX_I2S_CTL_RX_I2S_MODE_16); break; - case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: snd_soc_update_bits(dai->codec, LPASS_CDC_CLK_TX_I2S_CTL, TX_I2S_CTL_TX_I2S_MODE_MASK, TX_I2S_CTL_TX_I2S_MODE_32); -- 2.15.0

2 1

[alsa-devel] [PATCH] ASoC: nau8825: set clear_irq when imm IRQ happened
by John Hsu 30 Nov '17

30 Nov '17

Although the crosstalk is disabled, it is better to set clear_irq properly when the impedance measurement interrupt happens. It can avoid that the driver clears other IRQs by accident if the active_irq has another IRQ events. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index d3c1a02..603cd72 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1733,10 +1733,9 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) nau8825->xtalk_event_mask = event_mask; } } else if (active_irq & NAU8825_IMPEDANCE_MEAS_IRQ) { - if (nau8825->xtalk_enable) { + if (nau8825->xtalk_enable) schedule_work(&nau8825->xtalk_work); - clear_irq = NAU8825_IMPEDANCE_MEAS_IRQ; - } + clear_irq = NAU8825_IMPEDANCE_MEAS_IRQ; } else if ((active_irq & NAU8825_JACK_INSERTION_IRQ_MASK) == NAU8825_JACK_INSERTION_DETECTED) { /* One more step to check GPIO status directly. Thus, the -- 2.6.4

3 3

[alsa-devel] [PATCH v4 1/2] ALSA: pcm: add SNDRV_PCM_FORMAT_{S, U}20
by Maciej S. Szmigiero 30 Nov '17

30 Nov '17

This format is similar to existing SNDRV_PCM_FORMAT_{S,U}20_3 that keep 20-bit PCM samples in 3 bytes, however i.MX6 platform SSI FIFO does not allow 3-byte accesses (including DMA) so a 4-byte (more conventional) format is needed for it. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: Drop "_4" suffix from these formats since they aren't non-standard ones, use empty format slots starting from format number 25 for them, add information that they are LSB justified formats. Changes from v2: Adapt a comment in sound/core/pcm_misc.c so it still refers to the same sample formats as before. Changes from v3: Make the comment style in include/uapi/sound/asound.h match the style of comments at other sample formats. Corresponding alsa-lib changes will be posted as soon as this patch is merged on the kernel side, to keep alsa-lib and kernel synchronized. include/sound/pcm.h | 8 ++++++++ include/sound/soc-dai.h | 2 ++ include/uapi/sound/asound.h | 9 +++++++++ sound/core/pcm_misc.c | 19 ++++++++++++++++++- 4 files changed, 37 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 24febf9e177c..e054c583d3b3 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -169,6 +169,10 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_IMA_ADPCM _SNDRV_PCM_FMTBIT(IMA_ADPCM) #define SNDRV_PCM_FMTBIT_MPEG _SNDRV_PCM_FMTBIT(MPEG) #define SNDRV_PCM_FMTBIT_GSM _SNDRV_PCM_FMTBIT(GSM) +#define SNDRV_PCM_FMTBIT_S20_LE _SNDRV_PCM_FMTBIT(S20_LE) +#define SNDRV_PCM_FMTBIT_U20_LE _SNDRV_PCM_FMTBIT(U20_LE) +#define SNDRV_PCM_FMTBIT_S20_BE _SNDRV_PCM_FMTBIT(S20_BE) +#define SNDRV_PCM_FMTBIT_U20_BE _SNDRV_PCM_FMTBIT(U20_BE) #define SNDRV_PCM_FMTBIT_SPECIAL _SNDRV_PCM_FMTBIT(SPECIAL) #define SNDRV_PCM_FMTBIT_S24_3LE _SNDRV_PCM_FMTBIT(S24_3LE) #define SNDRV_PCM_FMTBIT_U24_3LE _SNDRV_PCM_FMTBIT(U24_3LE) @@ -202,6 +206,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_FLOAT SNDRV_PCM_FMTBIT_FLOAT_LE #define SNDRV_PCM_FMTBIT_FLOAT64 SNDRV_PCM_FMTBIT_FLOAT64_LE #define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE +#define SNDRV_PCM_FMTBIT_S20 SNDRV_PCM_FMTBIT_S20_LE +#define SNDRV_PCM_FMTBIT_U20 SNDRV_PCM_FMTBIT_U20_LE #endif #ifdef SNDRV_BIG_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_BE @@ -213,6 +219,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_FLOAT SNDRV_PCM_FMTBIT_FLOAT_BE #define SNDRV_PCM_FMTBIT_FLOAT64 SNDRV_PCM_FMTBIT_FLOAT64_BE #define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE +#define SNDRV_PCM_FMTBIT_S20 SNDRV_PCM_FMTBIT_S20_BE +#define SNDRV_PCM_FMTBIT_U20 SNDRV_PCM_FMTBIT_U20_BE #endif struct snd_pcm_file { diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 58acd00cae19..d970879944fc 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -102,6 +102,8 @@ struct snd_compr_stream; SNDRV_PCM_FMTBIT_S16_BE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S20_3BE |\ + SNDRV_PCM_FMTBIT_S20_LE |\ + SNDRV_PCM_FMTBIT_S20_BE |\ SNDRV_PCM_FMTBIT_S24_3LE |\ SNDRV_PCM_FMTBIT_S24_3BE |\ SNDRV_PCM_FMTBIT_S32_LE |\ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index c227ccba60ae..07d61583fd02 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -214,6 +214,11 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_IMA_ADPCM ((__force snd_pcm_format_t) 22) #define SNDRV_PCM_FORMAT_MPEG ((__force snd_pcm_format_t) 23) #define SNDRV_PCM_FORMAT_GSM ((__force snd_pcm_format_t) 24) +#define SNDRV_PCM_FORMAT_S20_LE ((__force snd_pcm_format_t) 25) /* in four bytes, LSB justified */ +#define SNDRV_PCM_FORMAT_S20_BE ((__force snd_pcm_format_t) 26) /* in four bytes, LSB justified */ +#define SNDRV_PCM_FORMAT_U20_LE ((__force snd_pcm_format_t) 27) /* in four bytes, LSB justified */ +#define SNDRV_PCM_FORMAT_U20_BE ((__force snd_pcm_format_t) 28) /* in four bytes, LSB justified */ +/* gap in the numbering for a future standard linear format */ #define SNDRV_PCM_FORMAT_SPECIAL ((__force snd_pcm_format_t) 31) #define SNDRV_PCM_FORMAT_S24_3LE ((__force snd_pcm_format_t) 32) /* in three bytes */ #define SNDRV_PCM_FORMAT_S24_3BE ((__force snd_pcm_format_t) 33) /* in three bytes */ @@ -248,6 +253,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_LE #define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_LE #define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE +#define SNDRV_PCM_FORMAT_S20 SNDRV_PCM_FORMAT_S20_LE +#define SNDRV_PCM_FORMAT_U20 SNDRV_PCM_FORMAT_U20_LE #endif #ifdef SNDRV_BIG_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_BE @@ -259,6 +266,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_BE #define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_BE #define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE +#define SNDRV_PCM_FORMAT_S20 SNDRV_PCM_FORMAT_S20_BE +#define SNDRV_PCM_FORMAT_U20 SNDRV_PCM_FORMAT_U20_BE #endif typedef int __bitwise snd_pcm_subformat_t; diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 9be81025372f..c4eb561d2008 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -163,13 +163,30 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { .width = 32, .phys = 32, .le = 0, .signd = 0, .silence = { 0x69, 0x69, 0x69, 0x69 }, }, - /* FIXME: the following three formats are not defined properly yet */ + /* FIXME: the following two formats are not defined properly yet */ [SNDRV_PCM_FORMAT_MPEG] = { .le = -1, .signd = -1, }, [SNDRV_PCM_FORMAT_GSM] = { .le = -1, .signd = -1, }, + [SNDRV_PCM_FORMAT_S20_LE] = { + .width = 20, .phys = 32, .le = 1, .signd = 1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_S20_BE] = { + .width = 20, .phys = 32, .le = 0, .signd = 1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_U20_LE] = { + .width = 20, .phys = 32, .le = 1, .signd = 0, + .silence = { 0x00, 0x00, 0x08, 0x00 }, + }, + [SNDRV_PCM_FORMAT_U20_BE] = { + .width = 20, .phys = 32, .le = 0, .signd = 0, + .silence = { 0x00, 0x08, 0x00, 0x00 }, + }, + /* FIXME: the following format is not defined properly yet */ [SNDRV_PCM_FORMAT_SPECIAL] = { .le = -1, .signd = -1, },

3 5

[alsa-devel] [PATCH] ALSA: usb-audio: Fix the enable parameter behavior
by Tamaki Nishino 30 Nov '17

30 Nov '17

The enable parameter doesn't seem to work as expected. Even when the enable parameter is set to false in /etc/modprobe.d/alsa-base.conf like "options snd-usb-audio index=-2 vid=0x1415 pid=0x1d27 enable=0", the specified device is not disabled. This patch fixes the enable parameter behavior to disable the device correctly. Signed-off-by: Tamaki Nishino <otamachan(a)gmail.com> --- sound/usb/card.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 23d1d23..8018d56 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -585,15 +585,24 @@ static int usb_audio_probe(struct usb_interface *intf, * now look for an empty slot and create a new card instance */ for (i = 0; i < SNDRV_CARDS; i++) - if (enable[i] && ! usb_chip[i] && + if (!usb_chip[i] && (vid[i] == -1 || vid[i] == USB_ID_VENDOR(id)) && (pid[i] == -1 || pid[i] == USB_ID_PRODUCT(id))) { - err = snd_usb_audio_create(intf, dev, i, quirk, - id, &chip); - if (err < 0) + if (enable[i]) { + err = snd_usb_audio_create(intf, dev, i, quirk, + id, &chip); + if (err < 0) + goto __error; + chip->pm_intf = intf; + break; + } else if (vid[i] != -1 || pid[i] != -1) { + dev_info(&dev->dev, + "device (%04x:%04x) is disabled\n", + USB_ID_VENDOR(id), + USB_ID_PRODUCT(id)); + err = -ENOENT; goto __error; - chip->pm_intf = intf; - break; + } } if (!chip) { dev_err(&dev->dev, "no available usb audio device\n"); -- 2.7.4

2 2

[alsa-devel] WARNING in initialize_timer
by syzbot 30 Nov '17

30 Nov '17

Hello, syzkaller hit the following crash on 43570f0383d6d5879ae585e6c3cf027ba321546f git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/master compiler: gcc (GCC) 7.1.1 20170620 .config is attached Raw console output is attached. Unfortunately, I don't have any reproducer for this bug yet. binder: 4519:4521 ioctl 40046205 2 returned -22 binder: 4519:4521 ioctl c0106403 20008ff0 returned -22 WARNING: CPU: 0 PID: 4528 at sound/core/seq/seq_timer.c:358 initialize_timer+0x22d/0x290 sound/core/seq/seq_timer.c:358 Kernel panic - not syncing: panic_on_warn set ... CPU: 0 PID: 4528 Comm: syz-executor7 Not tainted 4.15.0-rc1+ #199 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x194/0x257 lib/dump_stack.c:53 panic+0x1e4/0x41c kernel/panic.c:183 __warn+0x1dc/0x200 kernel/panic.c:547 report_bug+0x211/0x2d0 lib/bug.c:184 fixup_bug.part.11+0x37/0x80 arch/x86/kernel/traps.c:177 fixup_bug arch/x86/kernel/traps.c:246 [inline] do_error_trap+0x2d7/0x3e0 arch/x86/kernel/traps.c:295 do_invalid_op+0x1b/0x20 arch/x86/kernel/traps.c:314 invalid_op+0x18/0x20 arch/x86/entry/entry_64.S:930 RIP: 0010:initialize_timer+0x22d/0x290 sound/core/seq/seq_timer.c:358 RSP: 0018:ffff8801c7dbf880 EFLAGS: 00010012 RAX: 0000000000010000 RBX: ffff8801cdb75800 RCX: ffffffff840e956d RDX: 00000000000006b6 RSI: ffffc90001f35000 RDI: ffff8801cdb75850 RBP: ffff8801c7dbf8a8 R08: ffffffff86445678 R09: 1ffff10038fb7eb9 R10: ffff8801c97be2c0 R11: 0000000000000001 R12: 0000000000000000 R13: 0000000000000286 R14: ffff8801cdb75850 R15: ffff8801d92ecb40 seq_timer_start sound/core/seq/seq_timer.c:391 [inline] snd_seq_timer_start+0x147/0x2b0 sound/core/seq/seq_timer.c:405 snd_seq_queue_process_event sound/core/seq/seq_queue.c:698 [inline] snd_seq_control_queue+0x2b9/0x640 sound/core/seq/seq_queue.c:759 event_input_timer+0x25/0x30 sound/core/seq/seq_system.c:118 snd_seq_deliver_single_event.constprop.11+0x2fb/0x940 sound/core/seq/seq_clientmgr.c:621 snd_seq_deliver_event+0x176/0x840 sound/core/seq/seq_clientmgr.c:822 snd_seq_client_enqueue_event+0x2b6/0x420 sound/core/seq/seq_clientmgr.c:940 snd_seq_write+0x34d/0x720 sound/core/seq/seq_clientmgr.c:1069 __vfs_write+0xef/0x970 fs/read_write.c:480 vfs_write+0x18f/0x510 fs/read_write.c:544 SYSC_write fs/read_write.c:589 [inline] SyS_write+0xef/0x220 fs/read_write.c:581 entry_SYSCALL_64_fastpath+0x1f/0x96 RIP: 0033:0x4529d9 RSP: 002b:00007fb9be36ec58 EFLAGS: 00000212 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 0000000000758248 RCX: 00000000004529d9 RDX: 0000000000000030 RSI: 0000000020e6ffd0 RDI: 000000000000001c RBP: 00000000000005a2 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000212 R12: 00000000006f57d0 R13: 00000000ffffffff R14: 00007fb9be36f6d4 R15: 000000000000002a Dumping ftrace buffer: (ftrace buffer empty) Kernel Offset: disabled Rebooting in 86400 seconds.. --- This bug is generated by a dumb bot. It may contain errors. See https://goo.gl/tpsmEJ for details. Direct all questions to syzkaller(a)googlegroups.com. Please credit me with: Reported-by: syzbot <syzkaller(a)googlegroups.com> syzbot will keep track of this bug report. Once a fix for this bug is committed, please reply to this email with: #syz fix: exact-commit-title To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

2 1

Re: [alsa-devel] about the dual codecs on Lenovo p520
by Hui Wang 30 Nov '17

30 Nov '17

Hello Takashi, On the Lenovo P520, there are two analog codecs, the codec#0 is alc662, and lineout, linein and mic-in are on the rear panel and are connected to codec#0, the codec#1 is alc235, it connects an internal speaker, front-panel headset jack (headphone+mic). After booting up, only the lineout on codec#0 works, if i manually change value for some amixer controls (pulseaudio can't handle new controls like "Front-Panel"), the linein and mic-in on codec#0 also can work. But all ports on codec#1 can't work no matter how I change the value of the amixer controls. And you mentioned UCM profile, where can i find it and use it? the alsa-info.txt for this machine is at here： http://pastebin.ubuntu.com/26078174/ Thanks, Hui. On 2017年11月30日 15:45, Kailang wrote: > For the UCM frofile was provided by Takashi. > UCM profile was include in ALSA-LIB. > You could ask Takashi. > > >> -----Original Message----- >> From: Hui Wang [mailto:hui.wang@canonical.com] >> Sent: Thursday, November 30, 2017 3:30 PM >> To: Kailang >> Subject: about the dual codecs on Lenovo p520 >> >> Hi Kailang, >> >> >> We have a Lenovo P520, but the audio on this machine does not >> work, this machine has two codecs alc662 (codec#0) for >> rear-panel and alc235 >> (codec#1) for front-panel. Basically both the output and >> input can't work (only lineout on rear-panel works), it is >> because the pulseaudio can't set "Front-Panel ...". >> >> And I found you sent a patch to support dual codecs on P520, >> and you mentioned UCM in the commit header. Where can I find >> that UCM, or how can I make other ports ( like speaker on >> codec#1, headset port on codec#1, line-in and mic-in on codec#0) work? >> >> commit ca169cc2f9e1f8ed9c867b197a49d6dd05e5436d >> Author: Kailang Yang <kailang(a)realtek.com> >> Date: Tue Apr 25 16:17:40 2017 +0800 >> >> ALSA: hda/realtek - Add Dual Codecs support for Lenovo P520/420 >> >> Lenovo P520/420 build with two codecs. >> ALC233 for front panel. >> ALC662 for rear panel. >> This patch will rename capture name for slove conflicts. >> And create a card longname for UCM profile. >> >> Signed-off-by: Kailang Yang <kailang(a)realtek.com> >> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> >> >> >> Thanks, >> >> Hui. >> >> >> >> ------Please consider the environment before printing this e-mail. >>

3 2

Re: [alsa-devel] [PATCH RESEND 1/2] ASoC: Intel: kbl: Remove option of choosing CH count based on pdata
by Mark Brown 30 Nov '17

30 Nov '17

On Wed, Nov 15, 2017 at 03:37:42PM +0530, Naveen Manohar wrote: > Remove option of choosing channel count based on pdata for > kbl_rt5663_max98927 machine driver. > As User-space expects 4ch DMIC data and uses channel-map in ucm to > figure-out the desired channel to use, Hence removing driver change > to allow choosing channels. To repeat my previous question which I don't seem to see a response to: | Which userspace and are you sure there are no boards that only have two | of the mics?

3 2

[alsa-devel] [PATCH] ASoC: amd: added error checks in dma driver
by Vijendar Mukunda 30 Nov '17

30 Nov '17

added additional error checks in acp dma driver Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 39 ++++++++++++++++++++++++++++++++++----- 1 file changed, 34 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index fb09578..71ab5b5 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -848,6 +848,9 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + if (!rtd) + return -EINVAL; + buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd->acp_mmio, substream->stream); @@ -873,6 +876,8 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + if (!rtd) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { config_acp_dma_channel(rtd->acp_mmio, SYSRAM_TO_ACP_CH_NUM, PLAYBACK_START_DMA_DESCR_CH12, @@ -1082,7 +1087,11 @@ static int acp_audio_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, audio_drv_data); /* Initialize the ACP */ - acp_init(audio_drv_data->acp_mmio, audio_drv_data->asic_type); + status = acp_init(audio_drv_data->acp_mmio, audio_drv_data->asic_type); + if (status) { + dev_err(&pdev->dev, "ACP Init failed\n"); + return status; + } status = snd_soc_register_platform(&pdev->dev, &acp_asoc_platform); if (status != 0) { @@ -1099,9 +1108,14 @@ static int acp_audio_probe(struct platform_device *pdev) static int acp_audio_remove(struct platform_device *pdev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(&pdev->dev); - acp_deinit(adata->acp_mmio); + status = acp_deinit(adata->acp_mmio); + if (status) { + dev_err(&pdev->dev, "ACP Deinit failed\n"); + return status; + } snd_soc_unregister_platform(&pdev->dev); pm_runtime_disable(&pdev->dev); @@ -1111,9 +1125,14 @@ static int acp_audio_remove(struct platform_device *pdev) static int acp_pcm_resume(struct device *dev) { u16 bank; + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_init(adata->acp_mmio, adata->asic_type); + status = acp_init(adata->acp_mmio, adata->asic_type); + if (status) { + dev_err(dev, "ACP Init failed\n"); + return status; + } if (adata->play_stream && adata->play_stream->runtime) { /* For Stoney, Memory gating is disabled,i.e SRAM Banks @@ -1145,18 +1164,28 @@ static int acp_pcm_resume(struct device *dev) static int acp_pcm_runtime_suspend(struct device *dev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_deinit(adata->acp_mmio); + status = acp_deinit(adata->acp_mmio); + if (status) { + dev_err(dev, "ACP Deinit failed\n"); + return status; + } acp_reg_write(0, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } static int acp_pcm_runtime_resume(struct device *dev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_init(adata->acp_mmio, adata->asic_type); + status = acp_init(adata->acp_mmio, adata->asic_type); + if (status) { + dev_err(dev, "ACP Init failed\n"); + return status; + } acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } -- 2.7.4

3 2

[alsa-devel] alsa-plugins slibtool build failure
by Hunter S 30 Nov '17

30 Nov '17

Hi I tired to build alsa-plugins on Slackware64-current with slibtool instead of the GNU libtool and received this build failure. Full log - https://pastebin.com/ebjG60wk With help from #midipix @ freenode which maintains slibtool I found that adding '-pthread' to arcam-av/Makefile.am will allow building alsa-plugins with both libtool and slibtool. Here is a patch that does that, can it be applied to future versions of alsa-plugins? https://pastebin.com/w5T6FkZy Thanks,Hunter

1 0

[alsa-devel] [alsa-lib][PATCH] pcm: add SNDRV_PCM_FORMAT_{S,U}20
by Maciej S. Szmigiero 30 Nov '17

30 Nov '17

This format is similar to an existing 20-bit PCM format SNDRV_PCM_FORMAT_{S,U}20_3, however it occupies 4 bytes instead of 3. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- include/pcm.h | 20 ++++++++++++++++++-- include/sound/asound.h | 9 +++++++++ src/pcm/pcm.c | 10 ++++++++++ src/pcm/pcm_linear.c | 16 +++++++++++++--- src/pcm/pcm_local.h | 4 ++++ src/pcm/pcm_misc.c | 41 ++++++++++++++++++++++++++++++++++++++--- src/pcm/pcm_plug.c | 11 +++++++++++ src/pcm/pcm_route.c | 6 ++++-- src/pcm/plugin_ops.h | 50 ++++++++++++++++++++++++++++++++++++++++++++++---- 9 files changed, 153 insertions(+), 14 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index e05777a7c221..5939c1dab37d 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -175,6 +175,14 @@ typedef enum _snd_pcm_format { SND_PCM_FORMAT_MPEG, /** GSM */ SND_PCM_FORMAT_GSM, + /** Signed 20bit Little Endian in 4bytes format, LSB justified */ + SND_PCM_FORMAT_S20_LE, + /** Signed 20bit Big Endian in 4bytes format, LSB justified */ + SND_PCM_FORMAT_S20_BE, + /** Unsigned 20bit Little Endian in 4bytes format, LSB justified */ + SND_PCM_FORMAT_U20_LE, + /** Unsigned 20bit Big Endian in 4bytes format, LSB justified */ + SND_PCM_FORMAT_U20_BE, /** Special */ SND_PCM_FORMAT_SPECIAL = 31, /** Signed 24bit Little Endian in 3bytes format */ @@ -239,7 +247,11 @@ typedef enum _snd_pcm_format { /** Float 64 bit CPU endian */ SND_PCM_FORMAT_FLOAT64 = SND_PCM_FORMAT_FLOAT64_LE, /** IEC-958 CPU Endian */ - SND_PCM_FORMAT_IEC958_SUBFRAME = SND_PCM_FORMAT_IEC958_SUBFRAME_LE + SND_PCM_FORMAT_IEC958_SUBFRAME = SND_PCM_FORMAT_IEC958_SUBFRAME_LE, + /** Signed 20bit in 4bytes format, LSB justified, CPU Endian */ + SND_PCM_FORMAT_S20 = SND_PCM_FORMAT_S20_LE, + /** Unsigned 20bit in 4bytes format, LSB justified, CPU Endian */ + SND_PCM_FORMAT_U20 = SND_PCM_FORMAT_U20_LE #elif __BYTE_ORDER == __BIG_ENDIAN /** Signed 16 bit CPU endian */ SND_PCM_FORMAT_S16 = SND_PCM_FORMAT_S16_BE, @@ -258,7 +270,11 @@ typedef enum _snd_pcm_format { /** Float 64 bit CPU endian */ SND_PCM_FORMAT_FLOAT64 = SND_PCM_FORMAT_FLOAT64_BE, /** IEC-958 CPU Endian */ - SND_PCM_FORMAT_IEC958_SUBFRAME = SND_PCM_FORMAT_IEC958_SUBFRAME_BE + SND_PCM_FORMAT_IEC958_SUBFRAME = SND_PCM_FORMAT_IEC958_SUBFRAME_BE, + /** Signed 20bit in 4bytes format, LSB justified, CPU Endian */ + SND_PCM_FORMAT_S20 = SND_PCM_FORMAT_S20_BE, + /** Unsigned 20bit in 4bytes format, LSB justified, CPU Endian */ + SND_PCM_FORMAT_U20 = SND_PCM_FORMAT_U20_BE #else #error "Unknown endian" #endif diff --git a/include/sound/asound.h b/include/sound/asound.h index 5f659126f066..6041cab27a23 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -213,6 +213,11 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_IMA_ADPCM ((__force snd_pcm_format_t) 22) #define SNDRV_PCM_FORMAT_MPEG ((__force snd_pcm_format_t) 23) #define SNDRV_PCM_FORMAT_GSM ((__force snd_pcm_format_t) 24) +#define SNDRV_PCM_FORMAT_S20_LE ((__force snd_pcm_format_t) 25) /* in four bytes, LSB justified */ +#define SNDRV_PCM_FORMAT_S20_BE ((__force snd_pcm_format_t) 26) /* in four bytes, LSB justified */ +#define SNDRV_PCM_FORMAT_U20_LE ((__force snd_pcm_format_t) 27) /* in four bytes, LSB justified */ +#define SNDRV_PCM_FORMAT_U20_BE ((__force snd_pcm_format_t) 28) /* in four bytes, LSB justified */ +/* gap in the numbering for a future standard linear format */ #define SNDRV_PCM_FORMAT_SPECIAL ((__force snd_pcm_format_t) 31) #define SNDRV_PCM_FORMAT_S24_3LE ((__force snd_pcm_format_t) 32) /* in three bytes */ #define SNDRV_PCM_FORMAT_S24_3BE ((__force snd_pcm_format_t) 33) /* in three bytes */ @@ -247,6 +252,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_LE #define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_LE #define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE +#define SNDRV_PCM_FORMAT_S20 SNDRV_PCM_FORMAT_S20_LE +#define SNDRV_PCM_FORMAT_U20 SNDRV_PCM_FORMAT_U20_LE #endif #ifdef SNDRV_BIG_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_BE @@ -258,6 +265,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_BE #define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_BE #define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE +#define SNDRV_PCM_FORMAT_S20 SNDRV_PCM_FORMAT_S20_BE +#define SNDRV_PCM_FORMAT_U20 SNDRV_PCM_FORMAT_U20_BE #endif typedef int __bitwise snd_pcm_subformat_t; diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 1efb8b9d4e8c..e9ebf383c31b 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1780,6 +1780,10 @@ static const char *const snd_pcm_format_names[] = { FORMAT(IMA_ADPCM), FORMAT(MPEG), FORMAT(GSM), + FORMAT(S20_LE), + FORMAT(S20_BE), + FORMAT(U20_LE), + FORMAT(U20_BE), FORMAT(SPECIAL), FORMAT(S24_3LE), FORMAT(S24_3BE), @@ -1814,6 +1818,8 @@ static const char *const snd_pcm_format_aliases[SND_PCM_FORMAT_LAST+1] = { FORMAT(FLOAT), FORMAT(FLOAT64), FORMAT(IEC958_SUBFRAME), + FORMAT(S20), + FORMAT(U20), }; static const char *const snd_pcm_format_descriptions[] = { @@ -1842,6 +1848,10 @@ static const char *const snd_pcm_format_descriptions[] = { FORMATD(IMA_ADPCM, "Ima-ADPCM"), FORMATD(MPEG, "MPEG"), FORMATD(GSM, "GSM"), + FORMATD(S20_LE, "Signed 20 bit Little Endian in 4 bytes, LSB justified"), + FORMATD(S20_BE, "Signed 20 bit Big Endian in 4 bytes, LSB justified"), + FORMATD(U20_LE, "Unsigned 20 bit Little Endian in 4 bytes, LSB justified"), + FORMATD(U20_BE, "Unsigned 20 bit Big Endian in 4 bytes, LSB justified"), FORMATD(SPECIAL, "Special"), FORMATD(S24_3LE, "Signed 24 bit Little Endian in 3bytes"), FORMATD(S24_3BE, "Signed 24 bit Big Endian in 3bytes"), diff --git a/src/pcm/pcm_linear.c b/src/pcm/pcm_linear.c index 0d61e927323f..e4973a308004 100644 --- a/src/pcm/pcm_linear.c +++ b/src/pcm/pcm_linear.c @@ -90,6 +90,7 @@ int snd_pcm_linear_get_index(snd_pcm_format_t src_format, snd_pcm_format_t dst_f endian = 0; pwidth = snd_pcm_format_physical_width(src_format); width = snd_pcm_format_width(src_format); + assert(width >= 8 && width <= 32); if (pwidth == 24) { switch (width) { case 24: @@ -100,8 +101,11 @@ int snd_pcm_linear_get_index(snd_pcm_format_t src_format, snd_pcm_format_t dst_f default: width = 2; break; } - return width * 4 + endian * 2 + sign + 16; + return width * 4 + endian * 2 + sign + 20; } else { + if (width == 20) + width = 40; + width = width / 8 - 1; return width * 4 + endian * 2 + sign; } @@ -121,6 +125,7 @@ int snd_pcm_linear_put_index(snd_pcm_format_t src_format, snd_pcm_format_t dst_f endian = 0; pwidth = snd_pcm_format_physical_width(dst_format); width = snd_pcm_format_width(dst_format); + assert(width >= 8 && width <= 32); if (pwidth == 24) { switch (width) { case 24: @@ -131,8 +136,11 @@ int snd_pcm_linear_put_index(snd_pcm_format_t src_format, snd_pcm_format_t dst_f default: width = 2; break; } - return width * 4 + endian * 2 + sign + 16; + return width * 4 + endian * 2 + sign + 20; } else { + if (width == 20) + width = 40; + width = width / 8 - 1; return width * 4 + endian * 2 + sign; } @@ -303,7 +311,9 @@ static int snd_pcm_linear_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) if (err < 0) return err; linear->use_getput = (snd_pcm_format_physical_width(format) == 24 || - snd_pcm_format_physical_width(linear->sformat) == 24); + snd_pcm_format_physical_width(linear->sformat) == 24 || + snd_pcm_format_width(format) == 20 || + snd_pcm_format_width(linear->sformat) == 20); if (linear->use_getput) { if (pcm->stream == SND_PCM_STREAM_PLAYBACK) { linear->get_idx = snd_pcm_linear_get_index(format, SND_PCM_FORMAT_S32); diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 776f8cd3e302..3d95e1749169 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -984,6 +984,10 @@ const snd_config_t *snd_pcm_rate_get_default_converter(snd_config_t *root); (1U << SND_PCM_FORMAT_S16_BE) | \ (1U << SND_PCM_FORMAT_U16_LE) | \ (1U << SND_PCM_FORMAT_U16_BE) | \ + (1U << SND_PCM_FORMAT_S20_LE) | \ + (1U << SND_PCM_FORMAT_S20_BE) | \ + (1U << SND_PCM_FORMAT_U20_LE) | \ + (1U << SND_PCM_FORMAT_U20_BE) | \ (1U << SND_PCM_FORMAT_S24_LE) | \ (1U << SND_PCM_FORMAT_S24_BE) | \ (1U << SND_PCM_FORMAT_U24_LE) | \ diff --git a/src/pcm/pcm_misc.c b/src/pcm/pcm_misc.c index 5420b1895713..7ee0dac771af 100644 --- a/src/pcm/pcm_misc.c +++ b/src/pcm/pcm_misc.c @@ -38,6 +38,8 @@ int snd_pcm_format_signed(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_S8: case SNDRV_PCM_FORMAT_S16_LE: case SNDRV_PCM_FORMAT_S16_BE: + case SNDRV_PCM_FORMAT_S20_LE: + case SNDRV_PCM_FORMAT_S20_BE: case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_S24_BE: case SNDRV_PCM_FORMAT_S32_LE: @@ -52,6 +54,8 @@ int snd_pcm_format_signed(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U8: case SNDRV_PCM_FORMAT_U16_LE: case SNDRV_PCM_FORMAT_U16_BE: + case SNDRV_PCM_FORMAT_U20_LE: + case SNDRV_PCM_FORMAT_U20_BE: case SNDRV_PCM_FORMAT_U24_LE: case SNDRV_PCM_FORMAT_U24_BE: case SNDRV_PCM_FORMAT_U32_LE: @@ -126,6 +130,8 @@ int snd_pcm_format_little_endian(snd_pcm_format_t format) switch (format) { case SNDRV_PCM_FORMAT_S16_LE: case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S20_LE: + case SNDRV_PCM_FORMAT_U20_LE: case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_U24_LE: case SNDRV_PCM_FORMAT_S32_LE: @@ -144,6 +150,8 @@ int snd_pcm_format_little_endian(snd_pcm_format_t format) return 1; case SNDRV_PCM_FORMAT_S16_BE: case SNDRV_PCM_FORMAT_U16_BE: + case SNDRV_PCM_FORMAT_S20_BE: + case SNDRV_PCM_FORMAT_U20_BE: case SNDRV_PCM_FORMAT_S24_BE: case SNDRV_PCM_FORMAT_U24_BE: case SNDRV_PCM_FORMAT_S32_BE: @@ -218,6 +226,10 @@ int snd_pcm_format_width(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U18_3LE: case SNDRV_PCM_FORMAT_U18_3BE: return 18; + case SNDRV_PCM_FORMAT_S20_LE: + case SNDRV_PCM_FORMAT_S20_BE: + case SNDRV_PCM_FORMAT_U20_LE: + case SNDRV_PCM_FORMAT_U20_BE: case SNDRV_PCM_FORMAT_S20_3LE: case SNDRV_PCM_FORMAT_S20_3BE: case SNDRV_PCM_FORMAT_U20_3LE: @@ -289,6 +301,10 @@ int snd_pcm_format_physical_width(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U24_3LE: case SNDRV_PCM_FORMAT_U24_3BE: return 24; + case SNDRV_PCM_FORMAT_S20_LE: + case SNDRV_PCM_FORMAT_S20_BE: + case SNDRV_PCM_FORMAT_U20_LE: + case SNDRV_PCM_FORMAT_U20_BE: case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_S24_BE: case SNDRV_PCM_FORMAT_U24_LE: @@ -350,6 +366,10 @@ ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples) case SNDRV_PCM_FORMAT_U24_3LE: case SNDRV_PCM_FORMAT_U24_3BE: return samples * 3; + case SNDRV_PCM_FORMAT_S20_LE: + case SNDRV_PCM_FORMAT_S20_BE: + case SNDRV_PCM_FORMAT_U20_LE: + case SNDRV_PCM_FORMAT_U20_BE: case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_S24_BE: case SNDRV_PCM_FORMAT_U24_LE: @@ -393,6 +413,8 @@ uint64_t snd_pcm_format_silence_64(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_S8: case SNDRV_PCM_FORMAT_S16_LE: case SNDRV_PCM_FORMAT_S16_BE: + case SNDRV_PCM_FORMAT_S20_LE: + case SNDRV_PCM_FORMAT_S20_BE: case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_S24_BE: case SNDRV_PCM_FORMAT_S32_LE: @@ -415,12 +437,16 @@ uint64_t snd_pcm_format_silence_64(snd_pcm_format_t format) #ifdef SNDRV_LITTLE_ENDIAN case SNDRV_PCM_FORMAT_U16_LE: return 0x8000800080008000ULL; + case SNDRV_PCM_FORMAT_U20_LE: + return 0x0008000000080000ULL; case SNDRV_PCM_FORMAT_U24_LE: return 0x0080000000800000ULL; case SNDRV_PCM_FORMAT_U32_LE: return 0x8000000080000000ULL; case SNDRV_PCM_FORMAT_U16_BE: return 0x0080008000800080ULL; + case SNDRV_PCM_FORMAT_U20_BE: + return 0x0000080000000800ULL; case SNDRV_PCM_FORMAT_U24_BE: return 0x0000800000008000ULL; case SNDRV_PCM_FORMAT_U32_BE: @@ -440,12 +466,16 @@ uint64_t snd_pcm_format_silence_64(snd_pcm_format_t format) #else case SNDRV_PCM_FORMAT_U16_LE: return 0x0080008000800080ULL; + case SNDRV_PCM_FORMAT_U20_LE: + return 0x0000080000000800ULL; case SNDRV_PCM_FORMAT_U24_LE: return 0x0000800000008000ULL; case SNDRV_PCM_FORMAT_U32_LE: return 0x0000008000000080ULL; case SNDRV_PCM_FORMAT_U16_BE: return 0x8000800080008000ULL; + case SNDRV_PCM_FORMAT_U20_BE: + return 0x0008000000080000ULL; case SNDRV_PCM_FORMAT_U24_BE: return 0x0080000000800000ULL; case SNDRV_PCM_FORMAT_U32_BE: @@ -653,11 +683,13 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int return 0; } -static const int linear_formats[4][2][2] = { +static const int linear_formats[5][2][2] = { { { SNDRV_PCM_FORMAT_S8, SNDRV_PCM_FORMAT_S8 }, { SNDRV_PCM_FORMAT_U8, SNDRV_PCM_FORMAT_U8 } }, { { SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_FORMAT_S16_BE }, { SNDRV_PCM_FORMAT_U16_LE, SNDRV_PCM_FORMAT_U16_BE } }, + { { SNDRV_PCM_FORMAT_S20_LE, SNDRV_PCM_FORMAT_S20_BE }, + { SNDRV_PCM_FORMAT_U20_LE, SNDRV_PCM_FORMAT_U20_BE } }, { { SNDRV_PCM_FORMAT_S24_LE, SNDRV_PCM_FORMAT_S24_BE }, { SNDRV_PCM_FORMAT_U24_LE, SNDRV_PCM_FORMAT_U24_BE } }, { { SNDRV_PCM_FORMAT_S32_LE, SNDRV_PCM_FORMAT_S32_BE }, @@ -706,12 +738,15 @@ snd_pcm_format_t snd_pcm_build_linear_format(int width, int pwidth, int unsignd, case 16: width = 1; break; - case 24: + case 20: width = 2; break; - case 32: + case 24: width = 3; break; + case 32: + width = 4; + break; default: return SND_PCM_FORMAT_UNKNOWN; } diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index f1d7792db709..abf6f1ab9751 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -138,6 +138,17 @@ static const snd_pcm_format_t linear_preferred_formats[] = { SND_PCM_FORMAT_S24_LE, SND_PCM_FORMAT_U24_LE, #endif +#ifdef SND_LITTLE_ENDIAN + SND_PCM_FORMAT_S20_LE, + SND_PCM_FORMAT_U20_LE, + SND_PCM_FORMAT_S20_BE, + SND_PCM_FORMAT_U20_BE, +#else + SND_PCM_FORMAT_S20_BE, + SND_PCM_FORMAT_U20_BE, + SND_PCM_FORMAT_S20_LE, + SND_PCM_FORMAT_U20_LE, +#endif #ifdef SND_LITTLE_ENDIAN SND_PCM_FORMAT_S24_3LE, SND_PCM_FORMAT_U24_3LE, diff --git a/src/pcm/pcm_route.c b/src/pcm/pcm_route.c index 1f485a8e79f9..bbcc6118b593 100644 --- a/src/pcm/pcm_route.c +++ b/src/pcm/pcm_route.c @@ -565,10 +565,12 @@ static int snd_pcm_route_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) } if (err < 0) return err; - /* 3 bytes formats? */ + /* 3 bytes or 20-bit formats? */ route->params.use_getput = (snd_pcm_format_physical_width(src_format) + 7) / 8 == 3 || - (snd_pcm_format_physical_width(dst_format) + 7) / 8 == 3; + (snd_pcm_format_physical_width(dst_format) + 7) / 8 == 3 || + snd_pcm_format_width(src_format) == 20 || + snd_pcm_format_width(dst_format) == 20; route->params.get_idx = snd_pcm_linear_get_index(src_format, SND_PCM_FORMAT_S32); route->params.put_idx = snd_pcm_linear_put_index(SND_PCM_FORMAT_S32, dst_format); route->params.conv_idx = snd_pcm_linear_convert_index(src_format, dst_format); diff --git a/src/pcm/plugin_ops.h b/src/pcm/plugin_ops.h index 5a6016adb13a..a784d9c2b320 100644 --- a/src/pcm/plugin_ops.h +++ b/src/pcm/plugin_ops.h @@ -21,6 +21,12 @@ #ifndef SX_INLINES #define SX_INLINES +static inline uint32_t sx20(uint32_t x) +{ + if(x&0x00080000) + return x|0xFFF00000; + return x&0x000FFFFF; +} static inline uint32_t sx24(uint32_t x) { if(x&0x00800000) @@ -341,7 +347,7 @@ conv_1234_123C: as_u32(dst) = as_u32c(src) ^ 0x80; goto CONV_END; #ifdef GET16_LABELS /* src_wid src_endswap sign_toggle */ -static void *const get16_labels[4 * 2 * 2 + 4 * 3] = { +static void *const get16_labels[5 * 2 * 2 + 4 * 3] = { &&get16_1_10, /* 8h -> 16h */ &&get16_1_90, /* 8h ^> 16h */ &&get16_1_10, /* 8s -> 16h */ @@ -350,6 +356,7 @@ static void *const get16_labels[4 * 2 * 2 + 4 * 3] = { &&get16_12_92, /* 16h ^> 16h */ &&get16_12_21, /* 16s -> 16h */ &&get16_12_A1, /* 16s ^> 16h */ + /* 4 byte formats */ &&get16_0123_12, /* 24h -> 16h */ &&get16_0123_92, /* 24h ^> 16h */ &&get16_1230_32, /* 24s -> 16h */ @@ -358,6 +365,10 @@ static void *const get16_labels[4 * 2 * 2 + 4 * 3] = { &&get16_1234_92, /* 32h ^> 16h */ &&get16_1234_43, /* 32s -> 16h */ &&get16_1234_C3, /* 32s ^> 16h */ + &&get16_0123_12_20, /* 20h -> 16h */ + &&get16_0123_92_20, /* 20h ^> 16h */ + &&get16_1230_32_20, /* 20s -> 16h */ + &&get16_1230_B2_20, /* 20s ^> 16h */ /* 3bytes format */ &&get16_123_12, /* 24h -> 16h */ &&get16_123_92, /* 24h ^> 16h */ @@ -390,6 +401,10 @@ get16_1234_12: sample = as_u32c(src) >> 16; goto GET16_END; get16_1234_92: sample = (as_u32c(src) >> 16) ^ 0x8000; goto GET16_END; get16_1234_43: sample = bswap_16(as_u32c(src)); goto GET16_END; get16_1234_C3: sample = bswap_16(as_u32c(src) ^ 0x80); goto GET16_END; +get16_0123_12_20: sample = as_u32c(src) >> 4; goto GET16_END; +get16_0123_92_20: sample = (as_u32c(src) >> 4) ^ 0x8000; goto GET16_END; +get16_1230_32_20: sample = bswap_32(as_u32c(src)) >> 4; goto GET16_END; +get16_1230_B2_20: sample = (bswap_32(as_u32c(src)) >> 4) ^ 0x8000; goto GET16_END; get16_123_12: sample = _get_triple(src) >> 8; goto GET16_END; get16_123_92: sample = (_get_triple(src) >> 8) ^ 0x8000; goto GET16_END; get16_123_32: sample = _get_triple_s(src) >> 8; goto GET16_END; @@ -407,7 +422,7 @@ get16_123_B2_18: sample = (_get_triple_s(src) >> 2) ^ 0x8000; goto GET16_END; #ifdef PUT16_LABELS /* dst_wid dst_endswap sign_toggle */ -static void *const put16_labels[4 * 2 * 2 + 4 * 3] = { +static void *const put16_labels[5 * 2 * 2 + 4 * 3] = { &&put16_12_1, /* 16h -> 8h */ &&put16_12_9, /* 16h ^> 8h */ &&put16_12_1, /* 16h -> 8s */ @@ -416,6 +431,7 @@ static void *const put16_labels[4 * 2 * 2 + 4 * 3] = { &&put16_12_92, /* 16h ^> 16h */ &&put16_12_21, /* 16h -> 16s */ &&put16_12_29, /* 16h ^> 16s */ + /* 4 byte formats */ &&put16_12_0120, /* 16h -> 24h */ &&put16_12_0920, /* 16h ^> 24h */ &&put16_12_0210, /* 16h -> 24s */ @@ -424,6 +440,10 @@ static void *const put16_labels[4 * 2 * 2 + 4 * 3] = { &&put16_12_9200, /* 16h ^> 32h */ &&put16_12_0021, /* 16h -> 32s */ &&put16_12_0029, /* 16h ^> 32s */ + &&put16_12_0120_20, /* 16h -> 20h */ + &&put16_12_0920_20, /* 16h ^> 20h */ + &&put16_12_0210_20, /* 16h -> 20s */ + &&put16_12_0290_20, /* 16h ^> 20s */ /* 3bytes format */ &&put16_12_120, /* 16h -> 24h */ &&put16_12_920, /* 16h ^> 24h */ @@ -456,6 +476,10 @@ put16_12_1200: as_u32(dst) = (uint32_t)sample << 16; goto PUT16_END; put16_12_9200: as_u32(dst) = (uint32_t)(sample ^ 0x8000) << 16; goto PUT16_END; put16_12_0021: as_u32(dst) = (uint32_t)bswap_16(sample); goto PUT16_END; put16_12_0029: as_u32(dst) = (uint32_t)bswap_16(sample) ^ 0x80; goto PUT16_END; +put16_12_0120_20: as_u32(dst) = sx20((uint32_t)sample << 4); goto PUT16_END; +put16_12_0920_20: as_u32(dst) = sx20((uint32_t)(sample ^ 0x8000) << 4); goto PUT16_END; +put16_12_0210_20: as_u32(dst) = bswap_32(sx20((uint32_t)sample << 4)); goto PUT16_END; +put16_12_0290_20: as_u32(dst) = bswap_32(sx20((uint32_t)(sample ^ 0x8000) << 4)); goto PUT16_END; put16_12_120: _put_triple(dst, (uint32_t)sample << 8); goto PUT16_END; put16_12_920: _put_triple(dst, (uint32_t)(sample ^ 0x8000) << 8); goto PUT16_END; put16_12_021: _put_triple_s(dst, (uint32_t)sample << 8); goto PUT16_END; @@ -478,7 +502,7 @@ put16_12_029_18: _put_triple_s(dst, (uint32_t)(sample ^ 0x8000) << 2); goto PUT1 #ifdef GET32_LABELS /* src_wid src_endswap sign_toggle */ -static void *const get32_labels[4 * 2 * 2 + 4 * 3] = { +static void *const get32_labels[5 * 2 * 2 + 4 * 3] = { &&get32_1_1000, /* 8h -> 32h */ &&get32_1_9000, /* 8h ^> 32h */ &&get32_1_1000, /* 8s -> 32h */ @@ -487,6 +511,7 @@ static void *const get32_labels[4 * 2 * 2 + 4 * 3] = { &&get32_12_9200, /* 16h ^> 32h */ &&get32_12_2100, /* 16s -> 32h */ &&get32_12_A100, /* 16s ^> 32h */ + /* 4 byte formats */ &&get32_0123_1230, /* 24h -> 32h */ &&get32_0123_9230, /* 24h ^> 32h */ &&get32_1230_3210, /* 24s -> 32h */ @@ -495,6 +520,10 @@ static void *const get32_labels[4 * 2 * 2 + 4 * 3] = { &&get32_1234_9234, /* 32h ^> 32h */ &&get32_1234_4321, /* 32s -> 32h */ &&get32_1234_C321, /* 32s ^> 32h */ + &&get32_0123_1230_20, /* 20h -> 32h */ + &&get32_0123_9230_20, /* 20h ^> 32h */ + &&get32_1230_3210_20, /* 20s -> 32h */ + &&get32_1230_B210_20, /* 20s ^> 32h */ /* 3bytes format */ &&get32_123_1230, /* 24h -> 32h */ &&get32_123_9230, /* 24h ^> 32h */ @@ -531,6 +560,10 @@ get32_1234_1234: sample = as_u32c(src); goto GET32_END; get32_1234_9234: sample = as_u32c(src) ^ 0x80000000; goto GET32_END; get32_1234_4321: sample = bswap_32(as_u32c(src)); goto GET32_END; get32_1234_C321: sample = bswap_32(as_u32c(src) ^ 0x80); goto GET32_END; +get32_0123_1230_20: sample = as_u32c(src) << 12; goto GET32_END; +get32_0123_9230_20: sample = (as_u32c(src) << 12) ^ 0x80000000; goto GET32_END; +get32_1230_3210_20: sample = bswap_32(as_u32c(src)) << 12; goto GET32_END; +get32_1230_B210_20: sample = (bswap_32(as_u32c(src)) << 12) ^ 0x80000000; goto GET32_END; get32_123_1230: sample = _get_triple(src) << 8; goto GET32_END; get32_123_9230: sample = (_get_triple(src) << 8) ^ 0x80000000; goto GET32_END; get32_123_3210: sample = _get_triple_s(src) << 8; goto GET32_END; @@ -553,7 +586,7 @@ __conv24_get: goto *put; #ifdef PUT32_LABELS /* dst_wid dst_endswap sign_toggle */ -static void *const put32_labels[4 * 2 * 2 + 4 * 3] = { +static void *const put32_labels[5 * 2 * 2 + 4 * 3] = { &&put32_1234_1, /* 32h -> 8h */ &&put32_1234_9, /* 32h ^> 8h */ &&put32_1234_1, /* 32h -> 8s */ @@ -562,6 +595,7 @@ static void *const put32_labels[4 * 2 * 2 + 4 * 3] = { &&put32_1234_92, /* 32h ^> 16h */ &&put32_1234_21, /* 32h -> 16s */ &&put32_1234_29, /* 32h ^> 16s */ + /* 4 byte formats */ &&put32_1234_0123, /* 32h -> 24h */ &&put32_1234_0923, /* 32h ^> 24h */ &&put32_1234_3210, /* 32h -> 24s */ @@ -570,6 +604,10 @@ static void *const put32_labels[4 * 2 * 2 + 4 * 3] = { &&put32_1234_9234, /* 32h ^> 32h */ &&put32_1234_4321, /* 32h -> 32s */ &&put32_1234_4329, /* 32h ^> 32s */ + &&put32_1234_0123_20, /* 32h -> 20h */ + &&put32_1234_0923_20, /* 32h ^> 20h */ + &&put32_1234_3210_20, /* 32h -> 20s */ + &&put32_1234_3290_20, /* 32h ^> 20s */ /* 3bytes format */ &&put32_1234_123, /* 32h -> 24h */ &&put32_1234_923, /* 32h ^> 24h */ @@ -607,6 +645,10 @@ put32_1234_1234: as_u32(dst) = sample; goto PUT32_END; put32_1234_9234: as_u32(dst) = sample ^ 0x80000000; goto PUT32_END; put32_1234_4321: as_u32(dst) = bswap_32(sample); goto PUT32_END; put32_1234_4329: as_u32(dst) = bswap_32(sample) ^ 0x80; goto PUT32_END; +put32_1234_0123_20: as_u32(dst) = sx20(sample >> 12); goto PUT32_END; +put32_1234_0923_20: as_u32(dst) = sx20((sample ^ 0x80000000) >> 12); goto PUT32_END; +put32_1234_3210_20: as_u32(dst) = bswap_32(sx20(sample >> 12)); goto PUT32_END; +put32_1234_3290_20: as_u32(dst) = bswap_32(sx20((sample ^ 0x80000000) >> 12)); goto PUT32_END; put32_1234_123: _put_triple(dst, sample >> 8); goto PUT32_END; put32_1234_923: _put_triple(dst, (sample ^ 0x80000000) >> 8); goto PUT32_END; put32_1234_321: _put_triple_s(dst, sample >> 8); goto PUT32_END;

2 2

[alsa-devel] [alsa-lib][PATCH 4/4] pcm: plug: add SND_PCM_FORMAT_{S, U}20 to linear_preferred_formats
by Maciej S. Szmigiero 30 Nov '17

30 Nov '17

This commit adds the recently added formats SND_PCM_FORMAT_{S,U}20 to the linear_preferred_formats array in pcm_plug. Let's give them lower priority than more standard S24 formats but a higher priority than less typical 3-byte versions. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- src/pcm/pcm_plug.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index f1d7792db709..abf6f1ab9751 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -138,6 +138,17 @@ static const snd_pcm_format_t linear_preferred_formats[] = { SND_PCM_FORMAT_S24_LE, SND_PCM_FORMAT_U24_LE, #endif +#ifdef SND_LITTLE_ENDIAN + SND_PCM_FORMAT_S20_LE, + SND_PCM_FORMAT_U20_LE, + SND_PCM_FORMAT_S20_BE, + SND_PCM_FORMAT_U20_BE, +#else + SND_PCM_FORMAT_S20_BE, + SND_PCM_FORMAT_U20_BE, + SND_PCM_FORMAT_S20_LE, + SND_PCM_FORMAT_U20_LE, +#endif #ifdef SND_LITTLE_ENDIAN SND_PCM_FORMAT_S24_3LE, SND_PCM_FORMAT_U24_3LE,

1 0

[alsa-devel] [alsa-lib][PATCH 1/4] asound.h: add SNDRV_PCM_FORMAT_{S, U}20
by Maciej S. Szmigiero 30 Nov '17

30 Nov '17

This patch adds definitions of 20-bit, 4-byte PCM formats SNDRV_PCM_FORMAT_{S,U}20, that were recently added to the kernel, to alsa-lib's own copy of asound.h. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- include/sound/asound.h | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/include/sound/asound.h b/include/sound/asound.h index 5f659126f066..6041cab27a23 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -213,6 +213,11 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_IMA_ADPCM ((__force snd_pcm_format_t) 22) #define SNDRV_PCM_FORMAT_MPEG ((__force snd_pcm_format_t) 23) #define SNDRV_PCM_FORMAT_GSM ((__force snd_pcm_format_t) 24) +#define SNDRV_PCM_FORMAT_S20_LE ((__force snd_pcm_format_t) 25) /* in four bytes, LSB justified */ +#define SNDRV_PCM_FORMAT_S20_BE ((__force snd_pcm_format_t) 26) /* in four bytes, LSB justified */ +#define SNDRV_PCM_FORMAT_U20_LE ((__force snd_pcm_format_t) 27) /* in four bytes, LSB justified */ +#define SNDRV_PCM_FORMAT_U20_BE ((__force snd_pcm_format_t) 28) /* in four bytes, LSB justified */ +/* gap in the numbering for a future standard linear format */ #define SNDRV_PCM_FORMAT_SPECIAL ((__force snd_pcm_format_t) 31) #define SNDRV_PCM_FORMAT_S24_3LE ((__force snd_pcm_format_t) 32) /* in three bytes */ #define SNDRV_PCM_FORMAT_S24_3BE ((__force snd_pcm_format_t) 33) /* in three bytes */ @@ -247,6 +252,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_LE #define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_LE #define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE +#define SNDRV_PCM_FORMAT_S20 SNDRV_PCM_FORMAT_S20_LE +#define SNDRV_PCM_FORMAT_U20 SNDRV_PCM_FORMAT_U20_LE #endif #ifdef SNDRV_BIG_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_BE @@ -258,6 +265,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_BE #define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_BE #define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE +#define SNDRV_PCM_FORMAT_S20 SNDRV_PCM_FORMAT_S20_BE +#define SNDRV_PCM_FORMAT_U20 SNDRV_PCM_FORMAT_U20_BE #endif typedef int __bitwise snd_pcm_subformat_t;

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Fix the enable parameter behavior
by Tamaki Nishino 29 Nov '17

29 Nov '17

The enable parameter doesn't seem to work as expected. Even when the enable parameter is set to false in /etc/modprobe.d/alsa-base.conf like "options snd-usb-audio index=-2 vid=0x1415 pid=0x1d27 enable=0", the specified device is not disabled. This patch fixes the enable parameter behavior to disable the device correctly. Signed-off-by: Tamaki Nishino <otamachan(a)gmail.com> --- sound/usb/card.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 23d1d23..8018d56 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -585,15 +585,24 @@ static int usb_audio_probe(struct usb_interface *intf, * now look for an empty slot and create a new card instance */ for (i = 0; i < SNDRV_CARDS; i++) - if (enable[i] && ! usb_chip[i] && + if (!usb_chip[i] && (vid[i] == -1 || vid[i] == USB_ID_VENDOR(id)) && (pid[i] == -1 || pid[i] == USB_ID_PRODUCT(id))) { - err = snd_usb_audio_create(intf, dev, i, quirk, - id, &chip); - if (err < 0) + if (enable[i]) { + err = snd_usb_audio_create(intf, dev, i, quirk, + id, &chip); + if (err < 0) + goto __error; + chip->pm_intf = intf; + break; + } else if (vid[i] != -1 || pid[i] != -1) { + dev_info(&dev->dev, + "device (%04x:%04x) is disabled\n", + USB_ID_VENDOR(id), + USB_ID_PRODUCT(id)); + err = -ENOENT; goto __error; - chip->pm_intf = intf; - break; + } } if (!chip) { dev_err(&dev->dev, "no available usb audio device\n"); -- 2.7.4

1 0

[alsa-devel] [PATCH v2 2/2] ASoC: fsl_ssi: add 20-bit sample format for AC'97 and use it for capture
by Maciej S. Szmigiero 29 Nov '17

29 Nov '17

When testing AC'97 capture on UDOO board (currently the only user of fsl_ssi driver in the AC'97 mode) it become obvious that there is a massive distortion above certain, small input signal. This problem has been traced to silicon errata ERR003778: "In AC97, 16-bit mode, received data is shifted by 4-bit locations" that has "No fix scheduled". This errata suggests a workaround of doing a 4-bit shift back in SDMA script for this specific operation mode, however our SDMA scripts are shared between various SoC peripherals so we can't really modify them. There is a simple way to avoid this problem, however, that is to disallow recording in 16-bit mode and only support it in AC'97-native 20-bit mode. We have to use a 4-byte format for this since SSI FIFOs do not allow 3-byte accesses (and these aren't supported by imx-sdma driver anyway). With this change the capture distortion is gone. We can also add this format as an additional one supported for playback, using this opportunity to make sure that we use CPU-endian-native formats in AC'97 mode as we already do in I2S mode. There is no problem in using different bit widths in playback and capture in AC'97 mode so allow this, too. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: Adapt format name to changes in the first patch from this series. sound/soc/fsl/fsl_ssi.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 20ef09e1a395..c350117c8e31 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1278,14 +1278,15 @@ static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S20, }, .capture = { .stream_name = "AC97 Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + /* 16-bit capture is broken (errata ERR003778) */ + .formats = SNDRV_PCM_FMTBIT_S20, }, .ops = &fsl_ssi_dai_ops, }; @@ -1557,11 +1558,12 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - if (!fsl_ssi_is_ac97(ssi_private)) + if (!fsl_ssi_is_ac97(ssi_private)) { ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_samplebits = 1; + } ssi_private->cpu_dai_drv.symmetric_channels = 1; - ssi_private->cpu_dai_drv.symmetric_samplebits = 1; } /* Determine the FIFO depth. */

3 3

[alsa-devel] [PATCH v3 0/7] ASoC: Intel: Skylake: Add a clk driver to enable ssp clks early
by Sriram Periyasamy 29 Nov '17

29 Nov '17

For certain platforms, clocks (mclk/sclk/fs) are required to be up before the stream start. Example: some codecs needs the mclk/sclk/fs to be enabled early for a successful clock synchronization. Some platforms require clock to be enabled at boot and be always ON. By sending set_dma_control IPC (with the i2s blobs queried from NHLT), these clocks can be enabled early after the firmware is downloaded. With this series, a virtual clock driver is created which provides interface to send the required IPCs from machine driver to enable the clocks. NHLT is parsed during probe and the clock information is populated. The pointer to blob is cached and sent along with the set_dma_control IPC structure during the clk prepare/unprepare callback. Clocks are created for a ssp if the nhlt table has endpoint configuration for that particular ssp. Skylake driver creates a platform driver with the clock information and register the clk ops callback. Kabylake machine driver uses the clock interface to enable the clocks early as it is required by the rt5663 driver for clock synchronization. v2 -> v3 - Moved the clk ops and IPCs from Skylake driver to clk driver and reordered commits accordingly - Add the support for extended I2S blob config which supports multiple mclk dividers in NHLT - Enable the clocks as well in DAPM PMU event instead of hw_params in machine drivers as confirmed by codec vendor - Do not register the clk if there is no valid clock source is avail in the I2S blob - Take care of error return in the clk driver - Address rest of the review comments and more optimization added - Fix the warning sound/soc/intel/skylake/skl.c:724:1-3: WARNING: PTR_ERR_OR_ZERO can be used reported by scripts/coccinelle/api/ptr_ret.cocci - Modified DSP replies as human readable to ease the debugging - Add firmware replies for MCLK/SCLK clocks if they are running already v1 -> v2 - Register parent clocks with skylake device. With the patch "clk: Add support for runtime PM" soon to be merged will help DSP to stay active on call to clock enable. Reference: (https://patchwork.kernel.org/patch/9911741/) - Fix the machine driver to enable clocks early for headphone playback path as well to fix a pop noise issue Harsha Priya (1): ASoC: Intel: kbl: Enable mclk and ssp sclk early Naveen M (1): ASoC: Intel: eve: Enable mclk and ssp sclk early Sriram Periyasamy (3): ASoC: Intel: Skylake: Parse nhlt and register clock device ASoC: Intel: Skylake: Add ssp clock driver ASoC: Intel: Skylake: Add extended I2S config blob support in Clock driver Subhransu S. Prusty (2): ASoC: Intel: Skylake: Make DSP replies more human readable ASoC: Intel: Skylake: Add FW reply for MCLK/SCLK IPC sound/soc/intel/Kconfig | 3 + sound/soc/intel/boards/Kconfig | 2 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 101 ++++- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 100 +++++ sound/soc/intel/skylake/Makefile | 5 + sound/soc/intel/skylake/skl-i2s.h | 95 +++++ sound/soc/intel/skylake/skl-messages.c | 1 + sound/soc/intel/skylake/skl-nhlt.c | 172 ++++++++ sound/soc/intel/skylake/skl-ssp-clk.c | 470 +++++++++++++++++++++ sound/soc/intel/skylake/skl-ssp-clk.h | 117 +++++ sound/soc/intel/skylake/skl-sst-ipc.c | 50 ++- sound/soc/intel/skylake/skl.c | 93 ++++ sound/soc/intel/skylake/skl.h | 10 + 13 files changed, 1207 insertions(+), 12 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-i2s.h create mode 100644 sound/soc/intel/skylake/skl-ssp-clk.c create mode 100644 sound/soc/intel/skylake/skl-ssp-clk.h -- 2.7.4

2 8

[alsa-devel] [PATCH 0/2] Update codec driver for dual mics configuration
by Carlo Caione 29 Nov '17

29 Nov '17

From: Carlo Caione <carlo(a)endlessm.com> The RT5651 codec has 3 analog input ports. Some laptops (like the KIANO laptop) have two different analog mics on two ports, leaving the headset on the remaining one. We need to add a new quirk to support also this configuration. Carlo Caione (2): ASoC: intel: byt: Introduce new map for dual mics ASoC: intel: byt: Fix quirk for KIANO laptop sound/soc/intel/boards/bytcr_rt5651.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) -- 2.14.1

2 6

[alsa-devel] [PATCH] ALSA: usb-audio: Fix the enable parameter behavior
by Tamaki Nishino 29 Nov '17

29 Nov '17

The enable parameter doesn't seem to work as expected. Even when the enable parameter is set to false in /etc/modprobe.d/alsa-base.conf like "options snd-usb-audio index=-2 vid=0x1415 pid=0x1d27 enable=0", the specified device is not disabled. This patch fixes the enable parameter behavior to disable the device correctly. Signed-off-by: Tamaki Nishino <otamachan(a)gmail.com> --- sound/usb/card.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 23d1d23..8018d56 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -585,15 +585,24 @@ static int usb_audio_probe(struct usb_interface *intf, * now look for an empty slot and create a new card instance */ for (i = 0; i < SNDRV_CARDS; i++) - if (enable[i] && ! usb_chip[i] && + if (!usb_chip[i] && (vid[i] == -1 || vid[i] == USB_ID_VENDOR(id)) && (pid[i] == -1 || pid[i] == USB_ID_PRODUCT(id))) { - err = snd_usb_audio_create(intf, dev, i, quirk, - id, &chip); - if (err < 0) + if (enable[i]) { + err = snd_usb_audio_create(intf, dev, i, quirk, + id, &chip); + if (err < 0) + goto __error; + chip->pm_intf = intf; + break; + } else if (vid[i] != -1 || pid[i] != -1) { + dev_info(&dev->dev, + "device (%04x:%04x) is disabled\n", + USB_ID_VENDOR(id), + USB_ID_PRODUCT(id)); + err = -ENOENT; goto __error; - chip->pm_intf = intf; - break; + } } if (!chip) { dev_err(&dev->dev, "no available usb audio device\n"); -- 2.7.4

1 0

[alsa-devel] Applied "ASoC: soc-core: add missing EXPORT_SYMBOL_GPL() for snd_soc_disconnect_sync" to the asoc tree
by Mark Brown 29 Nov '17

29 Nov '17

The patch ASoC: soc-core: add missing EXPORT_SYMBOL_GPL() for snd_soc_disconnect_sync has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From df532185e8720baff120f55eb46058d270445d56 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 29 Nov 2017 02:38:27 +0000 Subject: [PATCH] ASoC: soc-core: add missing EXPORT_SYMBOL_GPL() for snd_soc_disconnect_sync Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9047046c749c..345baa4f10c0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1401,6 +1401,7 @@ void snd_soc_disconnect_sync(struct device *dev) snd_card_disconnect_sync(component->card->snd_card); } +EXPORT_SYMBOL_GPL(snd_soc_disconnect_sync); /** * snd_soc_add_dai_link - Add a DAI link dynamically -- 2.15.0

1 0

[alsa-devel] [PATCH] ASoC: rsnd: TDM 6ch needs 8ch clock for hw refine
by Kuninori Morimoto 29 Nov '17

29 Nov '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Renesas sound needs 8ch clock if TDM 6ch mode, and needs 2ch clock for 6ch or 8ch sound if Multi SSI mode. And these are related to before/after CTU (= Channel Transfer Unit). To calculate these we already has rsnd_runtime_channel_for_ssi() which returns runtime necessary channels. But, it based on runtime->channels which is not yet set when hw refine. We need to use hw_params instead of runtime->xxx when hw refine, and it is not needed after runtime was set. This patch adds new hw_params on rsnd_dai_stream, and it will be removed on rsnd_hw_params(). This is very temporary durty code, but it seems no choice at this point. Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 92 +++++++++++++++++++++++++++++++++++------------- sound/soc/sh/rcar/rsnd.h | 15 ++++++-- 2 files changed, 80 insertions(+), 27 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index a40f6a0..fa3d911 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -197,16 +197,27 @@ int rsnd_io_is_working(struct rsnd_dai_stream *io) return 0; } -int rsnd_runtime_channel_original(struct rsnd_dai_stream *io) +int rsnd_runtime_channel_original_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params) { struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - return runtime->channels; + /* + * params will be added when refine + * see + * __rsnd_soc_hw_rule_rate() + * __rsnd_soc_hw_rule_channels() + */ + if (params) + return params_channels(params); + else + return runtime->channels; } -int rsnd_runtime_channel_after_ctu(struct rsnd_dai_stream *io) +int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params) { - int chan = rsnd_runtime_channel_original(io); + int chan = rsnd_runtime_channel_original_with_params(io, params); struct rsnd_mod *ctu_mod = rsnd_io_to_mod_ctu(io); if (ctu_mod) { @@ -219,12 +230,13 @@ int rsnd_runtime_channel_after_ctu(struct rsnd_dai_stream *io) return chan; } -int rsnd_runtime_channel_for_ssi(struct rsnd_dai_stream *io) +int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params) { struct rsnd_dai *rdai = rsnd_io_to_rdai(io); int chan = rsnd_io_is_play(io) ? - rsnd_runtime_channel_after_ctu(io) : - rsnd_runtime_channel_original(io); + rsnd_runtime_channel_after_ctu_with_params(io, params) : + rsnd_runtime_channel_original_with_params(io, params); /* Use Multi SSI */ if (rsnd_runtime_is_ssi_multi(io)) @@ -616,8 +628,6 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: - rsnd_dai_stream_init(io, substream); - ret = rsnd_dai_call(init, io, priv); if (ret < 0) goto dai_trigger_end; @@ -639,7 +649,6 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, ret |= rsnd_dai_call(quit, io, priv); - rsnd_dai_stream_quit(io); break; default: ret = -EINVAL; @@ -784,8 +793,9 @@ static int rsnd_soc_hw_rule(struct rsnd_priv *priv, return snd_interval_refine(iv, &p); } -static int rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule, + int is_play) { struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); @@ -793,25 +803,37 @@ static int rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_soc_dai *dai = rule->private; struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture; /* * possible sampling rate limitation is same as * 2ch if it supports multi ssi + * and same as 8ch if TDM 6ch (see rsnd_ssi_config_init()) */ ic = *ic_; - if (1 < rsnd_rdai_ssi_lane_get(rdai)) { - ic.min = 2; - ic.max = 2; - } + ic.min = + ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params); return rsnd_soc_hw_rule(priv, rsnd_soc_hw_rate_list, ARRAY_SIZE(rsnd_soc_hw_rate_list), &ic, ir); } +static int rsnd_soc_hw_rule_rate_playback(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + return __rsnd_soc_hw_rule_rate(params, rule, 1); +} + +static int rsnd_soc_hw_rule_rate_capture(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + return __rsnd_soc_hw_rule_rate(params, rule, 0); +} -static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule, + int is_play) { struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); @@ -819,22 +841,34 @@ static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, struct snd_soc_dai *dai = rule->private; struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture; /* * possible sampling rate limitation is same as * 2ch if it supports multi ssi + * and same as 8ch if TDM 6ch (see rsnd_ssi_config_init()) */ ic = *ic_; - if (1 < rsnd_rdai_ssi_lane_get(rdai)) { - ic.min = 2; - ic.max = 2; - } + ic.min = + ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params); return rsnd_soc_hw_rule(priv, rsnd_soc_hw_channels_list, ARRAY_SIZE(rsnd_soc_hw_channels_list), ir, &ic); } +static int rsnd_soc_hw_rule_channels_playback(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + return __rsnd_soc_hw_rule_channels(params, rule, 1); +} + +static int rsnd_soc_hw_rule_channels_capture(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + return __rsnd_soc_hw_rule_channels(params, rule, 0); +} + static const struct snd_pcm_hardware rsnd_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | @@ -859,6 +893,8 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, int ret; int i; + rsnd_dai_stream_init(io, substream); + /* * Channel Limitation * It depends on Platform design @@ -886,11 +922,17 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, * It depends on Clock Master Mode */ if (rsnd_rdai_is_clk_master(rdai)) { + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - rsnd_soc_hw_rule_rate, dai, + is_play ? rsnd_soc_hw_rule_rate_playback : + rsnd_soc_hw_rule_rate_capture, + dai, SNDRV_PCM_HW_PARAM_CHANNELS, -1); snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - rsnd_soc_hw_rule_channels, dai, + is_play ? rsnd_soc_hw_rule_channels_playback : + rsnd_soc_hw_rule_channels_capture, + dai, SNDRV_PCM_HW_PARAM_RATE, -1); } @@ -915,6 +957,8 @@ static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream, * call rsnd_dai_call without spinlock */ rsnd_dai_call(nolock_stop, io, priv); + + rsnd_dai_stream_quit(io); } static const struct snd_soc_dai_ops rsnd_soc_dai_ops = { diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 57cd2bc..ad65235 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -399,9 +399,18 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, struct device_node *playback, struct device_node *capture); -int rsnd_runtime_channel_original(struct rsnd_dai_stream *io); -int rsnd_runtime_channel_after_ctu(struct rsnd_dai_stream *io); -int rsnd_runtime_channel_for_ssi(struct rsnd_dai_stream *io); +#define rsnd_runtime_channel_original(io) \ + rsnd_runtime_channel_original_with_params(io, NULL) +int rsnd_runtime_channel_original_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params); +#define rsnd_runtime_channel_after_ctu(io) \ + rsnd_runtime_channel_after_ctu_with_params(io, NULL) +int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params); +#define rsnd_runtime_channel_for_ssi(io) \ + rsnd_runtime_channel_for_ssi_with_params(io, NULL) +int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params); int rsnd_runtime_is_ssi_multi(struct rsnd_dai_stream *io); int rsnd_runtime_is_ssi_tdm(struct rsnd_dai_stream *io); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: dma.c: spin lock is no longer needed in IRQ handler
by Kuninori Morimoto 29 Nov '17

29 Nov '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> DMA handler had needed to calculate pointer before, but it doesn't need now. Thus, we can remove unnecessary spin lock from DMAC handler. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 18 ------------------ 1 file changed, 18 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 4d750bdf..41de234 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -71,25 +71,7 @@ struct rsnd_dma_ctrl { static void __rsnd_dmaen_complete(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - bool elapsed = false; - unsigned long flags; - - /* - * Renesas sound Gen1 needs 1 DMAC, - * Gen2 needs 2 DMAC. - * In Gen2 case, it are Audio-DMAC, and Audio-DMAC-peri-peri. - * But, Audio-DMAC-peri-peri doesn't have interrupt, - * and this driver is assuming that here. - */ - spin_lock_irqsave(&priv->lock, flags); - if (rsnd_io_is_working(io)) - elapsed = true; - - spin_unlock_irqrestore(&priv->lock, flags); - - if (elapsed) rsnd_dai_period_elapsed(io); } -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: nau8825: disable crosstalk by default
by John Hsu 29 Nov '17

29 Nov '17

The driver makes the crosstalk funciton disabled by default which can simplify the codec function. The platform may not need this funciton and reduce the potential risk. Therefore, We change the property "nuvoton,crosstalk-bypass" to "nuvoton,crosstalk-enable". The crosstalk measurement is enabled if the property is set. Otherwise, it is disabled. Besides, add more condition in the entry point of the crosstalk sequence to disable the function completely. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8825.txt | 4 ++-- sound/soc/codecs/nau8825.c | 23 ++++++++++++---------- sound/soc/codecs/nau8825.h | 2 +- 3 files changed, 16 insertions(+), 13 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt index 2f5e973..d16d968 100644 --- a/Documentation/devicetree/bindings/sound/nau8825.txt +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -69,7 +69,7 @@ Optional properties: - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - nuvoton,crosstalk-bypass: make crosstalk function bypass if set. + - nuvoton,crosstalk-enable: make crosstalk function enable if set. - clocks: list of phandle and clock specifier pairs according to common clock bindings for the clocks described in clock-names @@ -98,7 +98,7 @@ Example: nuvoton,short-key-debounce = <2>; nuvoton,jack-insert-debounce = <7>; nuvoton,jack-eject-debounce = <7>; - nuvoton,crosstalk-bypass; + nuvoton,crosstalk-enable; clock-names = "mclk"; clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>; diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 714ce17..d3c1a02 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -815,11 +815,12 @@ static void nau8825_xtalk_work(struct work_struct *work) static void nau8825_xtalk_cancel(struct nau8825 *nau8825) { - /* If the xtalk_protect is true, that means the process is still - * on going. The driver forces to cancel the cross talk task and + /* If the crosstalk is eanbled and the process is on going, + * the driver forces to cancel the crosstalk task and * restores the configuration to original status. */ - if (nau8825->xtalk_protect) { + if (nau8825->xtalk_enable && nau8825->xtalk_state != + NAU8825_XTALK_DONE) { cancel_work_sync(&nau8825->xtalk_work); nau8825_xtalk_clean(nau8825); } @@ -1686,7 +1687,7 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { if (nau8825_is_jack_inserted(regmap)) { event |= nau8825_jack_insert(nau8825); - if (!nau8825->xtalk_bypass && !nau8825->high_imped) { + if (nau8825->xtalk_enable && !nau8825->high_imped) { /* Apply the cross talk suppression in the * headset without high impedance. */ @@ -1732,8 +1733,10 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) nau8825->xtalk_event_mask = event_mask; } } else if (active_irq & NAU8825_IMPEDANCE_MEAS_IRQ) { - schedule_work(&nau8825->xtalk_work); - clear_irq = NAU8825_IMPEDANCE_MEAS_IRQ; + if (nau8825->xtalk_enable) { + schedule_work(&nau8825->xtalk_work); + clear_irq = NAU8825_IMPEDANCE_MEAS_IRQ; + } } else if ((active_irq & NAU8825_JACK_INSERTION_IRQ_MASK) == NAU8825_JACK_INSERTION_DETECTED) { /* One more step to check GPIO status directly. Thus, the @@ -2440,8 +2443,8 @@ static void nau8825_print_device_properties(struct nau8825 *nau8825) nau8825->jack_insert_debounce); dev_dbg(dev, "jack-eject-debounce: %d\n", nau8825->jack_eject_debounce); - dev_dbg(dev, "crosstalk-bypass: %d\n", - nau8825->xtalk_bypass); + dev_dbg(dev, "crosstalk-enable: %d\n", + nau8825->xtalk_enable); } static int nau8825_read_device_properties(struct device *dev, @@ -2506,8 +2509,8 @@ static int nau8825_read_device_properties(struct device *dev, &nau8825->jack_eject_debounce); if (ret) nau8825->jack_eject_debounce = 0; - nau8825->xtalk_bypass = device_property_read_bool(dev, - "nuvoton,crosstalk-bypass"); + nau8825->xtalk_enable = device_property_read_bool(dev, + "nuvoton,crosstalk-enable"); nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 8aee5c86..199d6ea 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -476,7 +476,7 @@ struct nau8825 { int xtalk_event_mask; bool xtalk_protect; int imp_rms[NAU8825_XTALK_IMM]; - int xtalk_bypass; + int xtalk_enable; }; int nau8825_enable_jack_detect(struct snd_soc_codec *codec, -- 2.6.4

3 7

[alsa-devel] [PATCH v2 2/2] ASoC: fsl_ssi: call _fsl_ssi_set_dai_fmt() just once in AC'97 mode
by Maciej S. Szmigiero 29 Nov '17

29 Nov '17

In AC'97 mode we configure and start SSI RX / TX on probe path via a call to _fsl_ssi_set_dai_fmt() function. We don't need to call this function again later and in fact don't want to do it since this function temporarily sets STCR, SRCR and SCR to some intermediate values. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: The SACCST setup code was split out into a separate commit. sound/soc/fsl/fsl_ssi.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 375aaaf6080d..70df6bf832df 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1116,6 +1116,9 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + if (fsl_ssi_is_ac97(ssi_private)) + return 0; + return _fsl_ssi_set_dai_fmt(cpu_dai->dev, ssi_private, fmt); }

3 2

[alsa-devel] [PATCH v2 1/2] ASoC: fsl_ssi: only enable proper channel slots in AC'97 mode
by Maciej S. Szmigiero 29 Nov '17

29 Nov '17

We need to make sure that only proper channel slots (in SACCST register) are enabled at playback start time since some AC'97 CODECs (like VT1613 on UDOO board) were observed requesting via SLOTREQ spurious ones just after an AC'97 link is started but before the CODEC is configured by its driver. When a bit for some channel slot is set in a SLOTREQ request then SSI sets the relevant bit in SACCST automatically, which then 'sticks' until it is manually unset. The SACCST register is not writable directly, we have to use SACCDIS and SACCEN registers to configure it instead (these aren't normal registers: writing a '1' bit at some position in SACCEN sets the relevant bit in SACCST; SACCDIS operates in a similar way but allows unsetting bits in SACCST). Theoretically, this should be necessary only for the very first playback but since some CODECs are so untrustworthy and extra channel slots enabled mean ruined playback let's play safe here and make sure that no extra slots are enabled in SACCST every time a playback is started. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: Split out this part from "fsl_ssi: call _fsl_ssi_set_dai_fmt() just once in AC'97 mode" commit, describe the problem and its solution better both in the commit message and in the code, move the SACCST setup code into a separate function and call it from TX config instead of doing it from trigger handler function. sound/soc/fsl/fsl_ssi.c | 52 +++++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 46 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 48bb850a34d9..375aaaf6080d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -574,8 +574,54 @@ static void fsl_ssi_rx_config(struct fsl_ssi_private *ssi_private, bool enable) fsl_ssi_config(ssi_private, enable, &ssi_private->rxtx_reg_val.rx); } +static void fsl_ssi_tx_ac97_saccst_setup(struct fsl_ssi_private *ssi_private) +{ + struct regmap *regs = ssi_private->regs; + + /* no SACC{ST,EN,DIS} regs on imx21-class SSI */ + if (!ssi_private->soc->imx21regs) { + /* + * Note that these below aren't just normal registers. + * They are a way to disable or enable bits in SACCST + * register: + * - writing a '1' bit at some position in SACCEN sets the + * relevant bit in SACCST, + * - writing a '1' bit at some position in SACCDIS unsets + * the relevant bit in SACCST register. + * + * The two writes below first disable all channels slots, + * then enable just slots 3 & 4 ("PCM Playback Left Channel" + * and "PCM Playback Right Channel"). + */ + regmap_write(regs, CCSR_SSI_SACCDIS, 0xff); + regmap_write(regs, CCSR_SSI_SACCEN, 0x300); + } +} + static void fsl_ssi_tx_config(struct fsl_ssi_private *ssi_private, bool enable) { + /* + * Why are we setting up SACCST everytime we are starting a + * playback? + * Some CODECs (like VT1613 CODEC on UDOO board) like to + * (sometimes) set extra bits in their SLOTREQ requests. + * When a bit is set in a SLOTREQ request then SSI sets the + * relevant bit in SACCST automatically (it is enough if a bit was + * set in a SLOTREQ just once, bits in SACCST are 'sticky'). + * If an extra slot gets enabled that's a disaster for playback + * because some of normal left or right channel samples are + * redirected instead to this extra slot. + * + * A workaround implemented in fsl-asoc-card of setting an + * appropriate CODEC register so that slots 3 & 4 (the normal + * stereo playback slots) are used for S/PDIF seems to mostly fix + * this issue on the UDOO board but since this CODEC is so + * untrustworthy let's play safe here and make sure that no extra + * slots are enabled every time a playback is started. + */ + if (enable && fsl_ssi_is_ac97(ssi_private)) + fsl_ssi_tx_ac97_saccst_setup(ssi_private); + fsl_ssi_config(ssi_private, enable, &ssi_private->rxtx_reg_val.tx); } @@ -630,12 +676,6 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) regmap_write(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV); - /* no SACC{ST,EN,DIS} regs on imx21-class SSI */ - if (!ssi_private->soc->imx21regs) { - regmap_write(regs, CCSR_SSI_SACCDIS, 0xff); - regmap_write(regs, CCSR_SSI_SACCEN, 0x300); - } - /* * Enable SSI, Transmit and Receive. AC97 has to communicate with the * codec before a stream is started.

3 2

[alsa-devel] [PATCH] ASoC: fsl_asrc: Fix typo in a field define
by Nicolin Chen 29 Nov '17

29 Nov '17

ASRFSTi_IAEi has an 11-bit offset as its _SHIFT macro defines. So this patch just fixes that. Reported-by: Laurent Charpentier <laurent.charpentier(a)nxp.com> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_asrc.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index 0f163ab..52c27a35 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -260,8 +260,8 @@ #define ASRFSTi_OUTPUT_FIFO_SHIFT 12 #define ASRFSTi_OUTPUT_FIFO_MASK (((1 << ASRFSTi_OUTPUT_FIFO_WIDTH) - 1) << ASRFSTi_OUTPUT_FIFO_SHIFT) #define ASRFSTi_IAEi_SHIFT 11 -#define ASRFSTi_IAEi_MASK (1 << ASRFSTi_OAFi_SHIFT) -#define ASRFSTi_IAEi (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_IAEi_MASK (1 << ASRFSTi_IAEi_SHIFT) +#define ASRFSTi_IAEi (1 << ASRFSTi_IAEi_SHIFT) #define ASRFSTi_INPUT_FIFO_WIDTH 7 #define ASRFSTi_INPUT_FIFO_SHIFT 0 #define ASRFSTi_INPUT_FIFO_MASK ((1 << ASRFSTi_INPUT_FIFO_WIDTH) - 1) -- 2.1.4

3 2

[alsa-devel] [PATCH] ASoC: soc-utils: add const to dummy_codec
by Kuninori Morimoto 29 Nov '17

29 Nov '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Let's align dummy_platform and dummy_codec Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index e30aacb..bcd3da2 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -288,7 +288,7 @@ static int dummy_dma_open(struct snd_pcm_substream *substream) .ops = &dummy_dma_ops, }; -static struct snd_soc_codec_driver dummy_codec; +static const struct snd_soc_codec_driver dummy_codec; #define STUB_RATES SNDRV_PCM_RATE_8000_192000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ -- 1.9.1

3 2

[alsa-devel] [PATCH 1/5 v4] ASoC: ep93xx-ac97: Fix platform_get_irq's error checking
by Arvind Yadav 29 Nov '17

29 Nov '17

The platform_get_irq() function returns negative if an error occurs. zero or positive number on success. platform_get_irq() error checking for zero is not correct. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- changes in v2 : irq was unsigned. so changed it to signed. changes in v3 : Add failure case '<= 0' instead of '< 0'. IRQ0 is not valid. changes in v4 : Return -ENODEV insted of irq. sound/soc/cirrus/ep93xx-ac97.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index bbf7a92..efeecee 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -365,7 +365,7 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) { struct ep93xx_ac97_info *info; struct resource *res; - unsigned int irq; + int irq; int ret; info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); @@ -378,7 +378,7 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) return PTR_ERR(info->regs); irq = platform_get_irq(pdev, 0); - if (!irq) + if (irq <= 0) return -ENODEV; ret = devm_request_irq(&pdev->dev, irq, ep93xx_ac97_interrupt, -- 2.7.4

4 7

[alsa-devel] [PATCH 0/2] ALSA: nm256: Fine-tuning for three function implementations
by SF Markus Elfring 29 Nov '17

29 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 16 Nov 2017 18:00:18 +0100 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Adjust five function calls together with a variable assignment Use common error handling code in snd_nm256_probe() sound/pci/nm256/nm256.c | 48 +++++++++++++++++++++++++++++------------------- 1 file changed, 29 insertions(+), 19 deletions(-) -- 2.15.0

3 43

[alsa-devel] [PATCH 00/24] ALSA: ctl: refactoring compat layer
by Takashi Sakamoto 29 Nov '17

29 Nov '17

Hi, ALSA control core supports system call compatibility layer because some structures in ALSA control interface includes members of 'long' and 'pointer' types and change own layout according to ABIs. In this point, it's enough for the layer to have handlers for structures on ABIs with 'ILP32' data model. A recent commit of 6236d8bb2afc ('ALSA: ctl: Fix ioctls for X32 ABI') clear that System V ABI for i386 architecture is unique than the other ABIs with 'ILP32' data model. On this ABI, machine type for storage class of 'long long' type has 4 bytes alignment. This is different from the other ABIs. In current implementation of this layer, the same structure is used for i386 ABI and the other ABI with 'ILP32' data model except for x32 ABI. Macro is used to switch between these. But in a view of data model, it's better to define structure for i386 ABI uniquely against the other ABIs including x32 for simplicity. Additionally, this layer includes some points to be improved: - cancel allocation in user space from kernel land - reduce kernel stack usage This patchset is my attempts to improve the layer. For this purpose, this patchset introduces a local framework to describe consistent method to convert and process data for differences of structure layout. Takashi Sakamoto (24): ALSA: ctl: introduce local framework to handle compat ioctl requests ALSA: ctl: add serializer/deserializer of 'elem_list' structure for 32bit ABI compatibility ALSA: ctl: add serializer/deserializer of 'elem_info' structure for 32bit ABI compatibility ALSA: ctl: add serializer/deserializer of 'elem_value' structure for modern 32 bit ABIs compatibility ALSA: ctl: add serializer/deserializer of 'elem_value' structure for i386 ABI compatibility ALSA: ctl: change prototype of local function for ELEM_LIST ioctl ALSA: ctl: add a helper function to allocate for ELEM_LIST request ALSA: ctl: cancel allocation in user space for ELEM_LIST request ALSA: ctl: unify calls of D0-wait function for ELEM_INFO request ALSA: ctl: unify calls of D0-wait function for ELEM_READ request ALSA: ctl: unify calls of D0-wait function for ELEM_WRITE request ALSA: ctl: allocation of elem_info data for ELEM_INFO request ALSA: ctl: change prototype of local function for ELEM_WRITE ioctl ALSA: ctl: change prototype of local function for ELEM_READ ioctl ALSA: ctl: allocation of elem_info data for ELEM_ADD/ELEM_REPLACE requests ALSA: ctl: add replace helper function to allocate own buffer ALSA: ctl: move removal code to replace helper function ALSA: ctl: replacement for compat ELEM_LIST operation for any 32 bit ABI ALSA: ctl: replacement for compat ELEM_INFO operation for any 32 bit ABI ALSA: ctl: replacement for compat ELEM_ADD operation for any 32 bit ABI ALSA: ctl: replacement for compat ELEM_REPLACE operation for any 32 bit ABI ALSA: ctl: replacement for compat ELEM_READ/ELEM_WRITE operation for i386 ABI ALSA: ctl: replacement for compat ELEM_READ/ELEM_WRITE operation for modern 32 bit ABI ALSA: ctl: code cleanup for compat handler sound/core/control.c | 242 ++++++++----- sound/core/control_compat.c | 837 +++++++++++++++++++++++++------------------- 2 files changed, 630 insertions(+), 449 deletions(-) -- 2.14.1

2 30

[alsa-devel] amixer sset volume: Left/Right race
by Eric Rannaud 29 Nov '17

29 Nov '17

In my ~/.xbindkeysrc I have "amixer set Master 1000+" XF86AudioRaiseVolume "amixer set Master 1000-" XF86AudioLowerVolume When I hold the "volume up" key on my keyboard, the left and right channel volumes quickly get out of sync. This is reproducible with this shell loop: for i in $(seq 20); do amixer sset Master 1000+ & done But not with this loop: for i in $(seq 20); do amixer sset Master 1000+; done It appears that xbindkeys invokes multiple amixer concurrently, like the first shell loop, and that amixer/ALSA is somehow setting the channel volumes one at a time, presumably in an unprotected read-modify-update critical section. I'm not familiar with how this works, and where the blame might lay. However, pamixer (https://github.com/cdemoulins/pamixer/blob/master/pulseaudio.cc) uses /usr/include/pulse/pulseaudio.h and pa_context_set_sink_volume_by_index() directly and does not have this problem. for i in $(seq 20); do pamixer --increase 1 & done Thanks. P.S. The "Bug Tracker" link on the left of http://alsa-project.org/main/index.php/Main_Page is broken.

1 0

[alsa-devel] [PATCH 0/2] ASoC: snd_soc_disconnect_sync()
by Kuninori Morimoto 28 Nov '17

28 Nov '17

Hi Mark, Takashi-san ALSA is now have snd_card_disconnect_sync() on Takashi-san's topic/card-disconnect branch. These are ASoC version of it, and Renesas sound driver uses it. Otherwise, kernel will be Oops if unbinded during playback. Kuninori Morimoto (2): ASoC: add snd_soc_disconnect_sync() ASoC: rsnd: call snd_soc_disconnect_sync() when remove include/sound/soc.h | 2 ++ sound/soc/sh/rcar/core.c | 2 ++ sound/soc/soc-core.c | 10 ++++++++++ 3 files changed, 14 insertions(+) -- 1.9.1

2 4

[alsa-devel] Applied "ASoC: rsnd: ssiu: clear SSI_MODE for non TDM Extended modes" to the asoc tree
by Mark Brown 28 Nov '17

28 Nov '17

The patch ASoC: rsnd: ssiu: clear SSI_MODE for non TDM Extended modes has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a91d7fb97092d6b840af5899ded3b389603fd7f1 Mon Sep 17 00:00:00 2001 From: Jiada Wang <jiada_wang(a)mentor.com> Date: Tue, 28 Nov 2017 16:05:13 +0900 Subject: [PATCH] ASoC: rsnd: ssiu: clear SSI_MODE for non TDM Extended modes register SSI_MODE is set when SSI works in TDM Extended, but it isn't reset when SSI starts to work in other modes, thus causes issues. This patch clearss SSI_MODE register when SSI works in modes other than TDM Extended. Fixes: 186fadc132f0 ("ASoC: rsnd: add TDM Extend Mode support") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/ssiu.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4d948757d300..6ff8a36c2c82 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -125,6 +125,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, { int hdmi = rsnd_ssi_hdmi_port(io); int ret; + u32 mode = 0; ret = rsnd_ssiu_init(mod, io, priv); if (ret < 0) @@ -136,9 +137,11 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, * see * rsnd_ssi_config_init() */ - rsnd_mod_write(mod, SSI_MODE, 0x1); + mode = 0x1; } + rsnd_mod_write(mod, SSI_MODE, mode); + if (rsnd_ssi_use_busif(io)) { rsnd_mod_write(mod, SSI_BUSIF_ADINR, rsnd_get_adinr_bit(mod, io) | -- 2.15.0

1 0

[alsa-devel] [PATCH] ALSA: drivers: make array 'names' const, reduces object code size
by Colin King 28 Nov '17

28 Nov '17

From: Colin Ian King <colin.king(a)canonical.com> Don't populate array 'names' on the stack but instead make them static. Makes the object code smaller by 50 bytes: Before: text data bss dec hex filename 21237 9192 1120 31549 7b3d linux/sound/drivers/dummy.o After: text data bss dec hex filename 21095 9280 1120 31495 7b07 linux/sound/drivers/dummy.o (gcc version 7.2.0 x86_64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/drivers/dummy.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 7b2b1f766b00..69db45bc0197 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -830,7 +830,7 @@ static int snd_dummy_capsrc_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el static int snd_dummy_iobox_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *info) { - const char *const names[] = { "None", "CD Player" }; + static const char *const names[] = { "None", "CD Player" }; return snd_ctl_enum_info(info, 1, 2, names); } -- 2.14.1

5 7

[alsa-devel] ["PATCH alsa-lib"] Change snd_dlopen() function to return the error string
by Jaroslav Kysela 28 Nov '17

28 Nov '17

The dlopen() function might fail also for another reason than a missing file, thus return the error string from dlerror(). Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- include/global.h | 2 +- src/Versions.in | 5 +++++ src/conf.c | 13 ++++++----- src/dlmisc.c | 60 +++++++++++++++++++++++++++++++++++++++---------- src/hwdep/hwdep.c | 6 ++--- src/mixer/simple_abst.c | 10 ++++----- src/pcm/pcm_hooks.c | 8 +++---- src/pcm/pcm_meter.c | 6 ++--- src/rawmidi/rawmidi.c | 6 ++--- src/seq/seq.c | 6 ++--- src/timer/timer.c | 6 ++--- src/timer/timer_query.c | 6 ++--- 12 files changed, 88 insertions(+), 46 deletions(-) diff --git a/include/global.h b/include/global.h index 41680829..d73d333a 100644 --- a/include/global.h +++ b/include/global.h @@ -97,7 +97,7 @@ extern struct snd_dlsym_link *snd_dlsym_start; /** \brief Returns the version of a dynamic symbol as a string. */ #define SND_DLSYM_VERSION(version) __STRING(version) -void *snd_dlopen(const char *file, int mode); +void *snd_dlopen(const char *file, int mode, char *errbuf, size_t errbuflen); void *snd_dlsym(void *handle, const char *name, const char *version); int snd_dlclose(void *handle); diff --git a/src/Versions.in b/src/Versions.in index 8d2dd116..2f9e1697 100644 --- a/src/Versions.in +++ b/src/Versions.in @@ -129,3 +129,8 @@ ALSA_0.9.7 { @SYMBOL_PREFIX@alsa_lisp_*; } ALSA_0.9.5; +ALSA_1.1.6 { + global: + + @SYMBOL_PREFIX@snd_dlopen; +} ALSA_1.1.6; diff --git a/src/conf.c b/src/conf.c index c410bfb8..e71bb972 100644 --- a/src/conf.c +++ b/src/conf.c @@ -982,6 +982,7 @@ static int get_freestring(char **string, int id, input_t *input) case '.': if (!id) break; + /* fall through */ case ' ': case '\f': case '\t': @@ -3477,7 +3478,7 @@ static int snd_config_hooks_call(snd_config_t *root, snd_config_t *config, snd_c { void *h = NULL; snd_config_t *c, *func_conf = NULL; - char *buf = NULL; + char *buf = NULL, errbuf[256]; const char *lib = NULL, *func_name = NULL; const char *str; int (*func)(snd_config_t *root, snd_config_t *config, snd_config_t **dst, snd_config_t *private_data) = NULL; @@ -3537,11 +3538,11 @@ static int snd_config_hooks_call(snd_config_t *root, snd_config_t *config, snd_c buf[len-1] = '\0'; func_name = buf; } - h = snd_dlopen(lib, RTLD_NOW); + h = snd_dlopen(lib, RTLD_NOW, errbuf, sizeof(errbuf)); func = h ? snd_dlsym(h, func_name, SND_DLSYM_VERSION(SND_CONFIG_DLSYM_VERSION_HOOK)) : NULL; err = 0; if (!h) { - SNDERR("Cannot open shared library %s", lib); + SNDERR("Cannot open shared library %s (%s)", lib, errbuf); err = -ENOENT; } else if (!func) { SNDERR("symbol %s is not defined inside %s", func_name, lib); @@ -4471,7 +4472,7 @@ static int _snd_config_evaluate(snd_config_t *src, { int err; if (pass == SND_CONFIG_WALK_PASS_PRE) { - char *buf = NULL; + char *buf = NULL, errbuf[256]; const char *lib = NULL, *func_name = NULL; const char *str; int (*func)(snd_config_t **dst, snd_config_t *root, @@ -4530,12 +4531,12 @@ static int _snd_config_evaluate(snd_config_t *src, buf[len-1] = '\0'; func_name = buf; } - h = snd_dlopen(lib, RTLD_NOW); + h = snd_dlopen(lib, RTLD_NOW, errbuf, sizeof(errbuf)); if (h) func = snd_dlsym(h, func_name, SND_DLSYM_VERSION(SND_CONFIG_DLSYM_VERSION_EVALUATE)); err = 0; if (!h) { - SNDERR("Cannot open shared library %s", lib); + SNDERR("Cannot open shared library %s (%s)", lib, errbuf); err = -ENOENT; goto _errbuf; } else if (!func) { diff --git a/src/dlmisc.c b/src/dlmisc.c index f154ebd0..fca843e1 100644 --- a/src/dlmisc.c +++ b/src/dlmisc.c @@ -43,12 +43,18 @@ struct snd_dlsym_link *snd_dlsym_start = NULL; * \brief Opens a dynamic library - ALSA wrapper for \c dlopen. * \param name name of the library, similar to \c dlopen. * \param mode mode flags, similar to \c dlopen. + * \param errbuf a string buffer for the error message \c dlerror. + * \param errbuflen a length of the string buffer for the error message. * \return Library handle if successful, otherwise \c NULL. * * This function can emulate dynamic linking for the static build of * the alsa-lib library. In that case, \p name is set to \c NULL. */ -void *snd_dlopen(const char *name, int mode) +#ifndef DOXYGEN +void *INTERNAL(snd_dlopen)(const char *name, int mode, char *errbuf, size_t errbuflen) +#else +void *snd_dlopen(const char *name, int mode, char *errbuf, size_t errbuflen) +#endif { #ifndef PIC if (name == NULL) @@ -73,24 +79,49 @@ void *snd_dlopen(const char *name, int mode) * via ld.so.conf. */ void *handle = NULL; - char *filename; + char *filename = NULL; if (name && name[0] != '/') { - filename = malloc(sizeof(ALSA_PLUGIN_DIR) + 1 + strlen(name) + 1); - strcpy(filename, ALSA_PLUGIN_DIR); - strcat(filename, "/"); - strcat(filename, name); - handle = dlopen(filename, mode); - free(filename); + filename = alloca(sizeof(ALSA_PLUGIN_DIR) + 1 + strlen(name) + 1); + if (filename) { + strcpy(filename, ALSA_PLUGIN_DIR); + strcat(filename, "/"); + strcat(filename, name); + handle = dlopen(filename, mode); + if (!handle) { + /* if the filename exists and cannot be opened */ + /* return immediately */ + if (access(filename, X_OK) == 0) + goto errpath; + } + } } - if (!handle) + if (!handle) { handle = dlopen(name, mode); + if (!handle) + goto errpath; + } return handle; +errpath: + if (errbuf) + snprintf(errbuf, errbuflen, "%s: %s", filename, dlerror()); + return NULL; + #else return NULL; #endif } +#ifndef DOXYGEN +void *INTERNAL(snd_dlopen_old)(const char *name, int mode) +{ + return INTERNAL(snd_dlopen)(name, mode, NULL, 0); +} +#endif + +use_symbol_version(__snd_dlopen_old, snd_dlopen, ALSA_0.9); +use_default_symbol_version(__snd_dlopen, snd_dlopen, ALSA_1.1.6); + /** * \brief Closes a dynamic library - ALSA wrapper for \c dlclose. * \param handle Library handle, similar to \c dlclose. @@ -229,6 +260,7 @@ void *snd_dlobj_cache_get(const char *lib, const char *name, struct list_head *p; struct dlobj_cache *c; void *func, *dlobj; + char errbuf[256]; snd_dlobj_lock(); list_for_each(p, &pcm_dlobj_list) { @@ -247,11 +279,15 @@ void *snd_dlobj_cache_get(const char *lib, const char *name, } } - dlobj = snd_dlopen(lib, RTLD_NOW); + errbuf[0] = '\0'; + dlobj = snd_dlopen(lib, RTLD_NOW, + verbose ? errbuf : 0, + verbose ? sizeof(errbuf) : 0); if (dlobj == NULL) { if (verbose) - SNDERR("Cannot open shared library %s", - lib ? lib : "[builtin]"); + SNDERR("Cannot open shared library %s (%s)", + lib ? lib : "[builtin]", + errbuf); snd_dlobj_unlock(); return NULL; } diff --git a/src/hwdep/hwdep.c b/src/hwdep/hwdep.c index 29350936..7aa475c0 100644 --- a/src/hwdep/hwdep.c +++ b/src/hwdep/hwdep.c @@ -41,7 +41,7 @@ static int snd_hwdep_open_conf(snd_hwdep_t **hwdep, snd_config_t *hwdep_conf, int mode) { const char *str; - char buf[256]; + char buf[256], errbuf[256]; int err; snd_config_t *conf, *type_conf = NULL; snd_config_iterator_t i, next; @@ -116,12 +116,12 @@ static int snd_hwdep_open_conf(snd_hwdep_t **hwdep, #ifndef PIC snd_hwdep_open_symbols(); #endif - h = snd_dlopen(lib, RTLD_NOW); + h = snd_dlopen(lib, RTLD_NOW, errbuf, sizeof(errbuf)); if (h) open_func = snd_dlsym(h, open_name, SND_DLSYM_VERSION(SND_HWDEP_DLSYM_VERSION)); err = 0; if (!h) { - SNDERR("Cannot open shared library %s", lib); + SNDERR("Cannot open shared library %s (%s)", lib, errbuf); err = -ENOENT; } else if (!open_func) { SNDERR("symbol %s is not defined inside %s", open_name, lib); diff --git a/src/mixer/simple_abst.c b/src/mixer/simple_abst.c index 73528393..32a3f621 100644 --- a/src/mixer/simple_abst.c +++ b/src/mixer/simple_abst.c @@ -66,7 +66,7 @@ static int try_open(snd_mixer_class_t *class, const char *lib) class_priv_t *priv = snd_mixer_class_get_private(class); snd_mixer_event_t event_func; snd_mixer_sbasic_init_t init_func = NULL; - char *xlib, *path; + char *xlib, *path, errbuf[256]; void *h; int err = 0; @@ -81,9 +81,9 @@ static int try_open(snd_mixer_class_t *class, const char *lib) strcpy(xlib, path); strcat(xlib, "/"); strcat(xlib, lib); - h = snd_dlopen(xlib, RTLD_NOW); + h = snd_dlopen(xlib, RTLD_NOW, errbuf, sizeof(errbuf)); if (h == NULL) { - SNDERR("Unable to open library '%s'", xlib); + SNDERR("Unable to open library '%s' (%s)", xlib, errbuf); free(xlib); return -ENXIO; } @@ -114,7 +114,7 @@ static int try_open_full(snd_mixer_class_t *class, snd_mixer_t *mixer, class_priv_t *priv = snd_mixer_class_get_private(class); snd_mixer_event_t event_func; snd_mixer_sfbasic_init_t init_func = NULL; - char *xlib, *path; + char *xlib, *path, errbuf[256]; void *h; int err = 0; @@ -128,7 +128,7 @@ static int try_open_full(snd_mixer_class_t *class, snd_mixer_t *mixer, strcat(xlib, "/"); strcat(xlib, lib); /* note python modules requires RTLD_GLOBAL */ - h = snd_dlopen(xlib, RTLD_NOW|RTLD_GLOBAL); + h = snd_dlopen(xlib, RTLD_NOW|RTLD_GLOBAL, errbuf, sizeof(errbuf)); if (h == NULL) { SNDERR("Unable to open library '%s'", xlib); free(xlib); diff --git a/src/pcm/pcm_hooks.c b/src/pcm/pcm_hooks.c index adb7230b..1f08bd78 100644 --- a/src/pcm/pcm_hooks.c +++ b/src/pcm/pcm_hooks.c @@ -341,7 +341,7 @@ even if the specified control doesn't exist. static int snd_pcm_hook_add_conf(snd_pcm_t *pcm, snd_config_t *root, snd_config_t *conf) { int err; - char buf[256]; + char buf[256], errbuf[256]; const char *str, *id; const char *lib = NULL, *install = NULL; snd_config_t *type = NULL, *args = NULL; @@ -424,12 +424,12 @@ static int snd_pcm_hook_add_conf(snd_pcm_t *pcm, snd_config_t *root, snd_config_ install = buf; snprintf(buf, sizeof(buf), "_snd_pcm_hook_%s_install", str); } - h = snd_dlopen(lib, RTLD_NOW); + h = snd_dlopen(lib, RTLD_NOW, errbuf, sizeof(errbuf)); install_func = h ? snd_dlsym(h, install, SND_DLSYM_VERSION(SND_PCM_DLSYM_VERSION)) : NULL; err = 0; if (!h) { - SNDERR("Cannot open shared library %s", - lib ? lib : "[builtin]"); + SNDERR("Cannot open shared library %s (%s)", + lib ? lib : "[builtin]", errbuf); err = -ENOENT; } else if (!install_func) { SNDERR("symbol %s is not defined inside %s", install, diff --git a/src/pcm/pcm_meter.c b/src/pcm/pcm_meter.c index 53685094..75df09ca 100644 --- a/src/pcm/pcm_meter.c +++ b/src/pcm/pcm_meter.c @@ -605,7 +605,7 @@ int snd_pcm_meter_open(snd_pcm_t **pcmp, const char *name, unsigned int frequenc static int snd_pcm_meter_add_scope_conf(snd_pcm_t *pcm, const char *name, snd_config_t *root, snd_config_t *conf) { - char buf[256]; + char buf[256], errbuf[256]; snd_config_iterator_t i, next; const char *id; const char *lib = NULL, *open_name = NULL, *str = NULL; @@ -670,11 +670,11 @@ static int snd_pcm_meter_add_scope_conf(snd_pcm_t *pcm, const char *name, open_name = buf; snprintf(buf, sizeof(buf), "_snd_pcm_scope_%s_open", str); } - h = snd_dlopen(lib, RTLD_NOW); + h = snd_dlopen(lib, RTLD_NOW, errbuf, sizeof(errbuf)); open_func = h ? dlsym(h, open_name) : NULL; err = 0; if (!h) { - SNDERR("Cannot open shared library %s", lib); + SNDERR("Cannot open shared library %s (%s)", lib, errbuf); err = -ENOENT; } else if (!open_func) { SNDERR("symbol %s is not defined inside %s", open_name, lib); diff --git a/src/rawmidi/rawmidi.c b/src/rawmidi/rawmidi.c index 7180becd..88d43017 100644 --- a/src/rawmidi/rawmidi.c +++ b/src/rawmidi/rawmidi.c @@ -162,7 +162,7 @@ static int snd_rawmidi_open_conf(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp snd_config_t *rawmidi_conf, int mode) { const char *str; - char buf[256]; + char buf[256], errbuf[256]; int err; snd_config_t *conf, *type_conf = NULL; snd_config_iterator_t i, next; @@ -239,12 +239,12 @@ static int snd_rawmidi_open_conf(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp #ifndef PIC snd_rawmidi_open_symbols(); #endif - h = snd_dlopen(lib, RTLD_NOW); + h = snd_dlopen(lib, RTLD_NOW, errbuf, sizeof(errbuf)); if (h) open_func = snd_dlsym(h, open_name, SND_DLSYM_VERSION(SND_RAWMIDI_DLSYM_VERSION)); err = 0; if (!h) { - SNDERR("Cannot open shared library %s", lib); + SNDERR("Cannot open shared library %s (%s)", lib, errbuf); err = -ENOENT; } else if (!open_func) { SNDERR("symbol %s is not defined inside %s", open_name, lib); diff --git a/src/seq/seq.c b/src/seq/seq.c index 630f52c3..983c4fab 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -823,7 +823,7 @@ static int snd_seq_open_conf(snd_seq_t **seqp, const char *name, int streams, int mode) { const char *str; - char buf[256]; + char buf[256], errbuf[256]; int err; snd_config_t *conf, *type_conf = NULL; snd_config_iterator_t i, next; @@ -899,12 +899,12 @@ static int snd_seq_open_conf(snd_seq_t **seqp, const char *name, #ifndef PIC snd_seq_open_symbols(); #endif - h = snd_dlopen(lib, RTLD_NOW); + h = snd_dlopen(lib, RTLD_NOW, errbuf, sizeof(errbuf)); if (h) open_func = snd_dlsym(h, open_name, SND_DLSYM_VERSION(SND_SEQ_DLSYM_VERSION)); err = 0; if (!h) { - SNDERR("Cannot open shared library %s", lib); + SNDERR("Cannot open shared library %s (%s)", lib, errbuf); err = -ENOENT; } else if (!open_func) { SNDERR("symbol %s is not defined inside %s", open_name, lib); diff --git a/src/timer/timer.c b/src/timer/timer.c index e027c903..8c49737e 100644 --- a/src/timer/timer.c +++ b/src/timer/timer.c @@ -76,7 +76,7 @@ static int snd_timer_open_conf(snd_timer_t **timer, snd_config_t *timer_conf, int mode) { const char *str; - char buf[256]; + char buf[256], errbuf[256]; int err; snd_config_t *conf, *type_conf = NULL; snd_config_iterator_t i, next; @@ -150,12 +150,12 @@ static int snd_timer_open_conf(snd_timer_t **timer, #ifndef PIC snd_timer_open_symbols(); #endif - h = snd_dlopen(lib, RTLD_NOW); + h = snd_dlopen(lib, RTLD_NOW, errbuf, sizeof(errbuf)); if (h) open_func = snd_dlsym(h, open_name, SND_DLSYM_VERSION(SND_TIMER_DLSYM_VERSION)); err = 0; if (!h) { - SNDERR("Cannot open shared library %s", lib); + SNDERR("Cannot open shared library %s (%s)", lib, errbuf); err = -ENOENT; } else if (!open_func) { SNDERR("symbol %s is not defined inside %s", open_name, lib); diff --git a/src/timer/timer_query.c b/src/timer/timer_query.c index d999d21b..6739f3c4 100644 --- a/src/timer/timer_query.c +++ b/src/timer/timer_query.c @@ -33,7 +33,7 @@ static int snd_timer_query_open_conf(snd_timer_query_t **timer, snd_config_t *timer_conf, int mode) { const char *str; - char buf[256]; + char buf[256], errbuf[256]; int err; snd_config_t *conf, *type_conf = NULL; snd_config_iterator_t i, next; @@ -108,12 +108,12 @@ static int snd_timer_query_open_conf(snd_timer_query_t **timer, #ifndef PIC snd_timer_query_open_symbols(); #endif - h = snd_dlopen(lib, RTLD_NOW); + h = snd_dlopen(lib, RTLD_NOW, errbuf, sizeof(errbuf)); if (h) open_func = snd_dlsym(h, open_name, SND_DLSYM_VERSION(SND_TIMER_QUERY_DLSYM_VERSION)); err = 0; if (!h) { - SNDERR("Cannot open shared library %s", lib); + SNDERR("Cannot open shared library %s (%s)", lib, errbuf); err = -ENOENT; } else if (!open_func) { SNDERR("symbol %s is not defined inside %s", open_name, lib); -- 2.13.6

3 6

[alsa-devel] [PATCH 0/3] ALSA: korg1212: Fine-tuning for three function implementations
by SF Markus Elfring 28 Nov '17

28 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 16 Nov 2017 12:42:42 +0100 Three update suggestions were taken into account from static source code analysis. Markus Elfring (3): Adjust eight function calls together with a variable assignment Delete a duplicate function call "release_firmware" in snd_korg1212_create() Use common error handling code in two functions sound/pci/korg1212/korg1212.c | 85 ++++++++++++++++++++++++------------------- 1 file changed, 47 insertions(+), 38 deletions(-) -- 2.15.0

2 4

[alsa-devel] [PATCH] ALSA: gus: Delete an error message for a failed memory allocation in snd_gf1_dma_transfer_block()
by SF Markus Elfring 28 Nov '17

28 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 22 Nov 2017 17:43:25 +0100 Omit an extra message for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/isa/gus/gus_dma.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index 36c27c832360..7f95f452f106 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -201,10 +201,9 @@ int snd_gf1_dma_transfer_block(struct snd_gus_card * gus, struct snd_gf1_dma_block *block; block = kmalloc(sizeof(*block), atomic ? GFP_ATOMIC : GFP_KERNEL); - if (block == NULL) { - snd_printk(KERN_ERR "gf1: DMA transfer failure; not enough memory\n"); + if (!block) return -ENOMEM; - } + *block = *__block; block->next = NULL; -- 2.15.0

2 1

[alsa-devel] [PATCH linux-next v2 1/1] ASoC: rsnd: ssiu: clear SSI_MODE for non TDM Extended modes
by Jiada Wang 28 Nov '17

28 Nov '17

register SSI_MODE is set when SSI works in TDM Extended, but it isn't reset when SSI starts to work in other modes, thus causes issues. This patch clearss SSI_MODE register when SSI works in modes other than TDM Extended. Fixes: 186fadc132f0 ("ASoC: rsnd: add TDM Extend Mode support") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/ssiu.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4d94875..6ff8a36 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -125,6 +125,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, { int hdmi = rsnd_ssi_hdmi_port(io); int ret; + u32 mode = 0; ret = rsnd_ssiu_init(mod, io, priv); if (ret < 0) @@ -136,9 +137,11 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, * see * rsnd_ssi_config_init() */ - rsnd_mod_write(mod, SSI_MODE, 0x1); + mode = 0x1; } + rsnd_mod_write(mod, SSI_MODE, mode); + if (rsnd_ssi_use_busif(io)) { rsnd_mod_write(mod, SSI_BUSIF_ADINR, rsnd_get_adinr_bit(mod, io) | -- 2.9.3

2 1

[alsa-devel] [PATCH v7 00/13] Introduce framework for SLIMbus device driver
by srinivas.kandagatla＠linaro.org 28 Nov '17

28 Nov '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Thanks for everyone who reviewed v6 patchset, here is v7 with review comments addressed. SLIMbus (Serial Low Power Interchip Media Bus) is a specification developed by MIPI (Mobile Industry Processor Interface) alliance. SLIMbus is a 2-wire implementation, which is used to communicate with peripheral components like audio-codec. SLIMbus uses Time-Division-Multiplexing to accommodate multiple data channels, and control channel. Control channel has messages to do device-enumeration, messages to send/receive control-data to/from slimbus devices, messages for port/channel management, and messages to do bandwidth allocation. Framework is introduced to support multiple instances of the bus (1 controller per bus), and multiple slave devices per controller. SPI and I2C frameworks, and comments from last time when I submitted the patches were referred-to while working on this framework. These patchsets introduce device-management, OF helpers, and messaging APIs, controller driver for Qualcomm's slimbus controller, and clock-pause feature for entering/exiting low-power mode for SLIMbus. Framework patches to do channel, port and bandwidth management are work-in-progress and will be sent out once these initial patches are accepted. These patchsets were tested on Qualcomm Snapdragon processor board using the controller driver, and a test slave device. v6: https://lwn.net/Articles/735693/ Changes from v6 to v7: * Various improvements in commenting style and kerneldoc suggested by various reviewers. * Removed usage of semaphores. * Split patches into more reviewable sizes. * Move buffer management into the controller driver. * use IDR for TID transactions. * use IDA for controller ids. * Remove lot of redundant code. * remove framer booted call for now, which can be added in future. * Added slim_readb/writeb and slim_read/write apis for simplified value element interface. Inspired by Soundwire patches. * Added slim device state so the the drivers which are probed after the device is announced, can get device state. * Lots of code cleanup. * move summary to .rst format. * move from callbacks to completions for async reads. * split slimbus.h into device and controller specific parts. Sagar Dharia (9): Documentation: Add SLIMbus summary dt-bindings: Add SLIMbus bindings slimbus: Add SLIMbus bus type slimbus: core: Add slim controllers support slimbus: Add messaging APIs to slimbus framework slimbus: Add support for 'clock-pause' feature dt-bindings: Add qcom slimbus controller bindings slimbus: qcom: Add Qualcomm Slimbus controller driver slimbus: qcom: Add runtime-pm support using clock-pause Srinivas Kandagatla (4): slimbus: core: add support to device tree helper regmap: add SLIMbus support slimbus: core: add common defines required for controllers MAINTAINERS: Add SLIMbus maintainer Documentation/devicetree/bindings/slimbus/bus.txt | 50 ++ .../devicetree/bindings/slimbus/slim-qcom-ctrl.txt | 43 ++ Documentation/driver-api/slimbus/index.rst | 15 + Documentation/driver-api/slimbus/summary.rst | 108 +++ MAINTAINERS | 8 + drivers/Kconfig | 2 + drivers/Makefile | 1 + drivers/base/regmap/Kconfig | 4 + drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-slimbus.c | 95 +++ drivers/slimbus/Kconfig | 22 + drivers/slimbus/Makefile | 9 + drivers/slimbus/core.c | 500 +++++++++++++ drivers/slimbus/messaging.c | 448 ++++++++++++ drivers/slimbus/qcom-ctrl.c | 791 +++++++++++++++++++++ drivers/slimbus/sched.c | 128 ++++ drivers/slimbus/slimbus.h | 276 +++++++ include/linux/mod_devicetable.h | 13 + include/linux/regmap.h | 18 + include/linux/slimbus.h | 167 +++++ 20 files changed, 2699 insertions(+) create mode 100644 Documentation/devicetree/bindings/slimbus/bus.txt create mode 100644 Documentation/devicetree/bindings/slimbus/slim-qcom-ctrl.txt create mode 100644 Documentation/driver-api/slimbus/index.rst create mode 100644 Documentation/driver-api/slimbus/summary.rst create mode 100644 drivers/base/regmap/regmap-slimbus.c create mode 100644 drivers/slimbus/Kconfig create mode 100644 drivers/slimbus/Makefile create mode 100644 drivers/slimbus/core.c create mode 100644 drivers/slimbus/messaging.c create mode 100644 drivers/slimbus/qcom-ctrl.c create mode 100644 drivers/slimbus/sched.c create mode 100644 drivers/slimbus/slimbus.h create mode 100644 include/linux/slimbus.h -- 2.15.0

8 46

[alsa-devel] [PATCH 0/6] ALSA: pcm: check parameter on snd_pcm_running()
by Kuninori Morimoto 28 Nov '17

28 Nov '17

Hi Takashi-san, Mark snd_pcm_running() is using "substream" and "substream->runtime" pointer, no check. These patches adds its check in function, and removes duplicate checks from each drivers. Not super important, but can be cleanup Kuninori Morimoto (6): ALSA: pcm: check parameter on snd_pcm_running() ALSA: pdaudiocf: remove unneeded check for snd_pcm_running() ASoC: dwc: remove unneeded check for snd_pcm_running() ASoC: omap-hdmi-audio: remove unneeded check for snd_pcm_running() ASoC: xtfpga-i2s: remove unneeded check for snd_pcm_running() ASoC: rsnd: remove unneeded check for snd_pcm_running() include/sound/pcm.h | 3 +++ sound/pcmcia/pdaudiocf/pdaudiocf_irq.c | 2 +- sound/soc/dwc/dwc-pcm.c | 2 +- sound/soc/omap/omap-hdmi-audio.c | 3 +-- sound/soc/sh/rcar/core.c | 5 +---- sound/soc/xtensa/xtfpga-i2s.c | 4 ++-- 6 files changed, 9 insertions(+), 10 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/3] ALSA: sgio2audio: Fine-tuning for two function implementations
by SF Markus Elfring 28 Nov '17

28 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 11 Nov 2017 20:16:32 +0100 Three update suggestions were taken into account from static source code analysis. Markus Elfring (3): Use common error handling code in two functions Improve a size determination in snd_sgio2audio_create() Adjust two null pointer checks in snd_sgio2audio_create() sound/mips/sgio2audio.c | 53 ++++++++++++++++++++++++------------------------- 1 file changed, 26 insertions(+), 27 deletions(-) -- 2.15.0

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[alsa-devel] [PATCH 0/3] ALSA: hal2: Fine-tuning for four function implementations
by SF Markus Elfring 28 Nov '17

28 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 11 Nov 2017 19:06:54 +0100 Three update suggestions were taken into account from static source code analysis. Markus Elfring (3): Use common error handling code in hal2_probe() Improve a size determination in hal2_create() Fix a typo in two comment lines sound/mips/hal2.c | 36 +++++++++++++++++------------------- 1 file changed, 17 insertions(+), 19 deletions(-) -- 2.15.0

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[alsa-devel] [PATCH 0/2][RESEND] ASoC: add/use snd_soc_disconnect_sync()
by Kuninori Morimoto 28 Nov '17

28 Nov '17

Hi Mark These are resend of add/use snd_soc_disconnect_sync() which is based on ALSA snd_card_disconnect_sync(). snd_card_disconnect_sync() was added Takashi-san's sound-4.15-rc1 branch. (or topic/card-disconnect branch) Without these patches, at least Renesas sound driver will hung up if user disconnect driver during playback. Kuninori Morimoto (2): ASoC: add snd_soc_disconnect_sync() ASoC: rsnd: call snd_soc_disconnect_sync() when remove include/sound/soc.h | 2 ++ sound/soc/sh/rcar/core.c | 2 ++ sound/soc/soc-core.c | 10 ++++++++++ 3 files changed, 14 insertions(+) -- 1.9.1

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[alsa-devel] [PATCH 00/14] ASoC: use snd_soc_component_init_regmap() - step [5/6]
by Kuninori Morimoto 28 Nov '17

28 Nov '17

Hi Mark These are step [5/6] for prepare of convert component of platform/codec. These uses snd_soc_component_init_regmap(), and remove .get_regmap(). These are duplicate feature. Kuninori Morimoto (14): ASoC: use snd_soc_component_init_regmap() on wm8998 ASoC: use snd_soc_component_init_regmap() on wm8997 ASoC: use snd_soc_component_init_regmap() on wm8994 ASoC: use snd_soc_component_init_regmap() on wm8400 ASoC: use snd_soc_component_init_regmap() on wm8350 ASoC: use snd_soc_component_init_regmap() on wm5110 ASoC: use snd_soc_component_init_regmap() on wm5102 ASoC: use snd_soc_component_init_regmap() on si476x ASoC: use snd_soc_component_init_regmap() on mc13783 ASoC: use snd_soc_component_init_regmap() on cq93vc ASoC: use snd_soc_component_init_regmap() on cs47l24 ASoC: use snd_soc_component_init_regmap() on 88pm860x ASoC: use snd_soc_component_init_regmap() on msm8916 ASoC: use snd_soc_component_init_regmap() on atmel-classd sound/soc/atmel/atmel-classd.c | 6 ------ sound/soc/codecs/88pm860x-codec.c | 9 +-------- sound/soc/codecs/cq93vc.c | 10 ++++++---- sound/soc/codecs/cs47l24.c | 12 +++--------- sound/soc/codecs/mc13783.c | 9 +++------ sound/soc/codecs/msm8916-wcd-analog.c | 8 ++------ sound/soc/codecs/si476x.c | 9 ++++++--- sound/soc/codecs/wm5102.c | 11 +++-------- sound/soc/codecs/wm5110.c | 12 +++--------- sound/soc/codecs/wm8350.c | 10 ++-------- sound/soc/codecs/wm8400.c | 9 +-------- sound/soc/codecs/wm8994.c | 10 ++-------- sound/soc/codecs/wm8997.c | 11 +++-------- sound/soc/codecs/wm8998.c | 12 +++--------- 14 files changed, 38 insertions(+), 100 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v3 1/2] ALSA: pcm: add SNDRV_PCM_FORMAT_{S, U}20
by Maciej S. Szmigiero 27 Nov '17

27 Nov '17

This format is similar to existing SNDRV_PCM_FORMAT_{S,U}20_3 that keep 20-bit PCM samples in 3 bytes, however i.MX6 platform SSI FIFO does not allow 3-byte accesses (including DMA) so a 4-byte (more conventional) format is needed for it. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: Drop "_4" suffix from these formats since they aren't non-standard ones, use empty format slots starting from format number 25 for them, add information that they are LSB justified formats. Changes from v2: Adapt a comment in sound/core/pcm_misc.c so it still refers to the same sample formats as before. Corresponding alsa-lib changes will be posted as soon as this patch is merged on the kernel side, to keep alsa-lib and kernel synchronized. include/sound/pcm.h | 8 ++++++++ include/sound/soc-dai.h | 2 ++ include/uapi/sound/asound.h | 9 +++++++++ sound/core/pcm_misc.c | 19 ++++++++++++++++++- 4 files changed, 37 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 24febf9e177c..e054c583d3b3 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -169,6 +169,10 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_IMA_ADPCM _SNDRV_PCM_FMTBIT(IMA_ADPCM) #define SNDRV_PCM_FMTBIT_MPEG _SNDRV_PCM_FMTBIT(MPEG) #define SNDRV_PCM_FMTBIT_GSM _SNDRV_PCM_FMTBIT(GSM) +#define SNDRV_PCM_FMTBIT_S20_LE _SNDRV_PCM_FMTBIT(S20_LE) +#define SNDRV_PCM_FMTBIT_U20_LE _SNDRV_PCM_FMTBIT(U20_LE) +#define SNDRV_PCM_FMTBIT_S20_BE _SNDRV_PCM_FMTBIT(S20_BE) +#define SNDRV_PCM_FMTBIT_U20_BE _SNDRV_PCM_FMTBIT(U20_BE) #define SNDRV_PCM_FMTBIT_SPECIAL _SNDRV_PCM_FMTBIT(SPECIAL) #define SNDRV_PCM_FMTBIT_S24_3LE _SNDRV_PCM_FMTBIT(S24_3LE) #define SNDRV_PCM_FMTBIT_U24_3LE _SNDRV_PCM_FMTBIT(U24_3LE) @@ -202,6 +206,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_FLOAT SNDRV_PCM_FMTBIT_FLOAT_LE #define SNDRV_PCM_FMTBIT_FLOAT64 SNDRV_PCM_FMTBIT_FLOAT64_LE #define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE +#define SNDRV_PCM_FMTBIT_S20 SNDRV_PCM_FMTBIT_S20_LE +#define SNDRV_PCM_FMTBIT_U20 SNDRV_PCM_FMTBIT_U20_LE #endif #ifdef SNDRV_BIG_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_BE @@ -213,6 +219,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_FLOAT SNDRV_PCM_FMTBIT_FLOAT_BE #define SNDRV_PCM_FMTBIT_FLOAT64 SNDRV_PCM_FMTBIT_FLOAT64_BE #define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE +#define SNDRV_PCM_FMTBIT_S20 SNDRV_PCM_FMTBIT_S20_BE +#define SNDRV_PCM_FMTBIT_U20 SNDRV_PCM_FMTBIT_U20_BE #endif struct snd_pcm_file { diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 58acd00cae19..d970879944fc 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -102,6 +102,8 @@ struct snd_compr_stream; SNDRV_PCM_FMTBIT_S16_BE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S20_3BE |\ + SNDRV_PCM_FMTBIT_S20_LE |\ + SNDRV_PCM_FMTBIT_S20_BE |\ SNDRV_PCM_FMTBIT_S24_3LE |\ SNDRV_PCM_FMTBIT_S24_3BE |\ SNDRV_PCM_FMTBIT_S32_LE |\ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index c227ccba60ae..7385024041d2 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -214,6 +214,11 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_IMA_ADPCM ((__force snd_pcm_format_t) 22) #define SNDRV_PCM_FORMAT_MPEG ((__force snd_pcm_format_t) 23) #define SNDRV_PCM_FORMAT_GSM ((__force snd_pcm_format_t) 24) +#define SNDRV_PCM_FORMAT_S20_LE ((__force snd_pcm_format_t) 25) /* \ */ +#define SNDRV_PCM_FORMAT_S20_BE ((__force snd_pcm_format_t) 26) /* | */ +#define SNDRV_PCM_FORMAT_U20_LE ((__force snd_pcm_format_t) 27) /* | in four bytes, */ +#define SNDRV_PCM_FORMAT_U20_BE ((__force snd_pcm_format_t) 28) /* / LSB justified */ +/* gap in the numbering for a future standard linear format */ #define SNDRV_PCM_FORMAT_SPECIAL ((__force snd_pcm_format_t) 31) #define SNDRV_PCM_FORMAT_S24_3LE ((__force snd_pcm_format_t) 32) /* in three bytes */ #define SNDRV_PCM_FORMAT_S24_3BE ((__force snd_pcm_format_t) 33) /* in three bytes */ @@ -248,6 +253,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_LE #define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_LE #define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE +#define SNDRV_PCM_FORMAT_S20 SNDRV_PCM_FORMAT_S20_LE +#define SNDRV_PCM_FORMAT_U20 SNDRV_PCM_FORMAT_U20_LE #endif #ifdef SNDRV_BIG_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_BE @@ -259,6 +266,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_BE #define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_BE #define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE +#define SNDRV_PCM_FORMAT_S20 SNDRV_PCM_FORMAT_S20_BE +#define SNDRV_PCM_FORMAT_U20 SNDRV_PCM_FORMAT_U20_BE #endif typedef int __bitwise snd_pcm_subformat_t; diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 9be81025372f..c4eb561d2008 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -163,13 +163,30 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { .width = 32, .phys = 32, .le = 0, .signd = 0, .silence = { 0x69, 0x69, 0x69, 0x69 }, }, - /* FIXME: the following three formats are not defined properly yet */ + /* FIXME: the following two formats are not defined properly yet */ [SNDRV_PCM_FORMAT_MPEG] = { .le = -1, .signd = -1, }, [SNDRV_PCM_FORMAT_GSM] = { .le = -1, .signd = -1, }, + [SNDRV_PCM_FORMAT_S20_LE] = { + .width = 20, .phys = 32, .le = 1, .signd = 1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_S20_BE] = { + .width = 20, .phys = 32, .le = 0, .signd = 1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_U20_LE] = { + .width = 20, .phys = 32, .le = 1, .signd = 0, + .silence = { 0x00, 0x00, 0x08, 0x00 }, + }, + [SNDRV_PCM_FORMAT_U20_BE] = { + .width = 20, .phys = 32, .le = 0, .signd = 0, + .silence = { 0x00, 0x08, 0x00, 0x00 }, + }, + /* FIXME: the following format is not defined properly yet */ [SNDRV_PCM_FORMAT_SPECIAL] = { .le = -1, .signd = -1, },

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[alsa-devel] [PATCH] ALSA: ice1724: Fix resume issues with Prodigy 7.1 HiFi
by Yussuf Khalil 27 Nov '17

27 Nov '17

There are two issues after resuming from suspend on the Audiotrak Prodigy 7.1 HiFi: - the output volume is set to 100% - microphone input isn't working anymore This patch fixes these issues by reinitializing both codecs of the device and restoring the previous volumes during resume. Signed-off-by: Yussuf Khalil <dev(a)pp3345.net> --- sound/pci/ice1712/prodigy_hifi.c | 131 ++++++++++++++++++++++++++++++--------- 1 file changed, 102 insertions(+), 29 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 2697402b5195..8dabd4d0211d 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -965,13 +965,32 @@ static int prodigy_hd2_add_controls(struct snd_ice1712 *ice) return 0; } +static void wm8766_init(struct snd_ice1712 *ice) +{ + static unsigned short wm8766_inits[] = { + WM8766_RESET, 0x0000, + WM8766_DAC_CTRL, 0x0120, + WM8766_INT_CTRL, 0x0022, /* I2S Normal Mode, 24 bit */ + WM8766_DAC_CTRL2, 0x0001, + WM8766_DAC_CTRL3, 0x0080, + WM8766_LDA1, 0x0100, + WM8766_LDA2, 0x0100, + WM8766_LDA3, 0x0100, + WM8766_RDA1, 0x0100, + WM8766_RDA2, 0x0100, + WM8766_RDA3, 0x0100, + WM8766_MUTE1, 0x0000, + WM8766_MUTE2, 0x0000, + }; + unsigned int i; -/* - * initialize the chip - */ -static int prodigy_hifi_init(struct snd_ice1712 *ice) + for (i = 0; i < ARRAY_SIZE(wm8766_inits); i += 2) + wm8766_spi_write(ice, wm8766_inits[i], wm8766_inits[i + 1]); +} + +static void wm8776_init(struct snd_ice1712 *ice) { - static unsigned short wm_inits[] = { + static unsigned short wm8776_inits[] = { /* These come first to reduce init pop noise */ WM_ADC_MUX, 0x0003, /* ADC mute */ /* 0x00c0 replaced by 0x0003 */ @@ -982,7 +1001,76 @@ static int prodigy_hifi_init(struct snd_ice1712 *ice) WM_POWERDOWN, 0x0008, /* All power-up except HP */ WM_RESET, 0x0000, /* reset */ }; - static unsigned short wm_inits2[] = { + unsigned int i; + + for (i = 0; i < ARRAY_SIZE(wm8776_inits); i += 2) + wm_put(ice, wm8776_inits[i], wm8776_inits[i + 1]); +} + +#ifdef CONFIG_PM_SLEEP +static int prodigy_hifi_resume(struct snd_ice1712 *ice) +{ + static unsigned short wm8776_reinit_registers[] = { + WM_MASTER_CTRL, + WM_DAC_INT, + WM_ADC_INT, + WM_OUT_MUX, + WM_HP_ATTEN_L, + WM_HP_ATTEN_R, + WM_PHASE_SWAP, + WM_DAC_CTRL2, + WM_ADC_ATTEN_L, + WM_ADC_ATTEN_R, + WM_ALC_CTRL1, + WM_ALC_CTRL2, + WM_ALC_CTRL3, + WM_NOISE_GATE, + WM_ADC_MUX, + /* no DAC attenuation here */ + }; + struct prodigy_hifi_spec *spec = ice->spec; + int i, ch; + + mutex_lock(&ice->gpio_mutex); + + /* reinitialize WM8776 and re-apply old register values */ + wm8776_init(ice); + schedule_timeout_uninterruptible(1); + for (i = 0; i < ARRAY_SIZE(wm8776_reinit_registers); i++) + wm_put(ice, wm8776_reinit_registers[i], + wm_get(ice, wm8776_reinit_registers[i])); + + /* reinitialize WM8766 and re-apply volumes for all DACs */ + wm8766_init(ice); + for (ch = 0; ch < 2; ch++) { + wm_set_vol(ice, WM_DAC_ATTEN_L + ch, + spec->vol[2 + ch], spec->master[ch]); + + wm8766_set_vol(ice, WM8766_LDA1 + ch, + spec->vol[0 + ch], spec->master[ch]); + + wm8766_set_vol(ice, WM8766_LDA2 + ch, + spec->vol[4 + ch], spec->master[ch]); + + wm8766_set_vol(ice, WM8766_LDA3 + ch, + spec->vol[6 + ch], spec->master[ch]); + } + + /* unmute WM8776 DAC */ + wm_put(ice, WM_DAC_MUTE, 0x00); + wm_put(ice, WM_DAC_CTRL1, 0x90); + + mutex_unlock(&ice->gpio_mutex); + return 0; +} +#endif + +/* + * initialize the chip + */ +static int prodigy_hifi_init(struct snd_ice1712 *ice) +{ + static unsigned short wm8776_defaults[] = { WM_MASTER_CTRL, 0x0022, /* 256fs, slave mode */ WM_DAC_INT, 0x0022, /* I2S, normal polarity, 24bit */ WM_ADC_INT, 0x0022, /* I2S, normal polarity, 24bit */ @@ -1010,22 +1098,6 @@ static int prodigy_hifi_init(struct snd_ice1712 *ice) WM_DAC_MUTE, 0x0000, /* DAC unmute */ WM_ADC_MUX, 0x0003, /* ADC unmute, both CD/Line On */ }; - static unsigned short wm8766_inits[] = { - WM8766_RESET, 0x0000, - WM8766_DAC_CTRL, 0x0120, - WM8766_INT_CTRL, 0x0022, /* I2S Normal Mode, 24 bit */ - WM8766_DAC_CTRL2, 0x0001, - WM8766_DAC_CTRL3, 0x0080, - WM8766_LDA1, 0x0100, - WM8766_LDA2, 0x0100, - WM8766_LDA3, 0x0100, - WM8766_RDA1, 0x0100, - WM8766_RDA2, 0x0100, - WM8766_RDA3, 0x0100, - WM8766_MUTE1, 0x0000, - WM8766_MUTE2, 0x0000, - }; - struct prodigy_hifi_spec *spec; unsigned int i; @@ -1052,16 +1124,17 @@ static int prodigy_hifi_init(struct snd_ice1712 *ice) ice->spec = spec; /* initialize WM8776 codec */ - for (i = 0; i < ARRAY_SIZE(wm_inits); i += 2) - wm_put(ice, wm_inits[i], wm_inits[i+1]); + wm8776_init(ice); schedule_timeout_uninterruptible(1); - for (i = 0; i < ARRAY_SIZE(wm_inits2); i += 2) - wm_put(ice, wm_inits2[i], wm_inits2[i+1]); + for (i = 0; i < ARRAY_SIZE(wm8776_defaults); i += 2) + wm_put(ice, wm8776_defaults[i], wm8776_defaults[i + 1]); - /* initialize WM8766 codec */ - for (i = 0; i < ARRAY_SIZE(wm8766_inits); i += 2) - wm8766_spi_write(ice, wm8766_inits[i], wm8766_inits[i+1]); + wm8766_init(ice); +#ifdef CONFIG_PM_SLEEP + ice->pm_resume = &prodigy_hifi_resume; + ice->pm_suspend_enabled = 1; +#endif return 0; } -- 2.14.3

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[alsa-devel] [PATCH 0/4] ASoC: wm2000: Adjustments for wm2000_i2c_probe()
by SF Markus Elfring 27 Nov '17

27 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 24 Nov 2017 08:26:56 +0100 A few update suggestions were taken into account from static source code analysis. Markus Elfring (4): Delete an error message for a failed memory allocation One function call less in wm2000_i2c_probe() after error detection Fix a typo in a comment line Improve a size determination sound/soc/codecs/wm2000.c | 28 ++++++++++++---------------- 1 file changed, 12 insertions(+), 16 deletions(-) -- 2.15.0

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[alsa-devel] [PATCH 0/2] ASoC: wm8903: Adjustments for wm8903_i2c_probe()
by SF Markus Elfring 27 Nov '17

27 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 24 Nov 2017 10:44:55 +0100 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Delete an error message for a failed memory allocation Improve two size determinations sound/soc/codecs/wm8903.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) -- 2.15.0

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[alsa-devel] [PATCH 0/3] ASoC: da7218: Adjustments for two function implementations
by SF Markus Elfring 27 Nov '17

27 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 23 Nov 2017 21:00:12 +0100 Three update suggestions were taken into account from static source code analysis. Markus Elfring (3): Delete two error messages for a failed memory allocation in da7218_of_to_pdata() Use common error handling code in da7218_of_to_pdata() Improve a size determination in da7218_i2c_probe() sound/soc/codecs/da7218.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) -- 2.15.0

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[alsa-devel] [PATCH 0/2] ASoC: da7213: Adjustments for two function implementations
by SF Markus Elfring 27 Nov '17

27 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 23 Nov 2017 18:28:38 +0100 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Delete an error message for a failed memory allocation in da7213_fw_to_pdata() Improve a size determination in da7213_i2c_probe() sound/soc/codecs/da7213.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) -- 2.15.0

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[alsa-devel] [PATCH 0/2] ASoC: cs42l56: Adjustments for cs42l56_i2c_probe()
by SF Markus Elfring 27 Nov '17

27 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 23 Nov 2017 09:38:24 +0100 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Delete an error message for a failed memory allocation Improve two size determinations sound/soc/codecs/cs42l56.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) -- 2.15.0

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[alsa-devel] [PATCH 0/2] ASoC: cs42l52: Adjustments for cs42l52_i2c_probe()
by SF Markus Elfring 27 Nov '17

27 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 22 Nov 2017 22:15:43 +0100 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Delete an error message for a failed memory allocation Improve two size determinations sound/soc/codecs/cs42l52.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) -- 2.15.0

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[alsa-devel] Applied "ASoC: ts3a227e: Map BTN_0 to KEY_PLAYPAUSE" to the asoc tree
by Mark Brown 27 Nov '17

27 Nov '17

The patch ASoC: ts3a227e: Map BTN_0 to KEY_PLAYPAUSE has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From af0f6c5820845b66c76296f281da5cf916d7e094 Mon Sep 17 00:00:00 2001 From: Benson Leung <bleung(a)chromium.org> Date: Wed, 22 Nov 2017 12:56:41 -0800 Subject: [PATCH] ASoC: ts3a227e: Map BTN_0 to KEY_PLAYPAUSE The Android 3.5mm Headset jack specification mentions that BTN_0 should be mapped to KEY_MEDIA, but this is less logical than KEY_PLAYPAUSE, which has much broader userspace support. For example, the Chrome OS userspace now supports KEY_PLAYPAUSE to toggle play/pause of videos and audio, but does not handle KEY_MEDIA. Furthermore, Android itself now supports KEY_PLAYPAUSE equivalently, as the new USB headset spec requires KEY_PLAYPAUSE for BTN_0. https://source.android.com/devices/accessories/headset/usb-headset-spec Signed-off-by: Benson Leung <bleung(a)chromium.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ts3a227e.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 738e04b09116..1271e7e1fc78 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -241,7 +241,7 @@ int ts3a227e_enable_jack_detect(struct snd_soc_component *component, { struct ts3a227e *ts3a227e = snd_soc_component_get_drvdata(component); - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); -- 2.15.0

1 0

[alsa-devel] Applied "ASoC: rk3399_gru_sound: Map BTN_0 to KEY_PLAYPAUSE" to the asoc tree
by Mark Brown 27 Nov '17

27 Nov '17

The patch ASoC: rk3399_gru_sound: Map BTN_0 to KEY_PLAYPAUSE has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From af1b1cefd735c919d3185ce06b549c2b121067ba Mon Sep 17 00:00:00 2001 From: Benson Leung <bleung(a)chromium.org> Date: Wed, 22 Nov 2017 12:56:42 -0800 Subject: [PATCH] ASoC: rk3399_gru_sound: Map BTN_0 to KEY_PLAYPAUSE The Android 3.5mm Headset jack specification mentions that BTN_0 should be mapped to KEY_MEDIA, but this is less logical than KEY_PLAYPAUSE, which has much broader userspace support. For example, the Chrome OS userspace now supports KEY_PLAYPAUSE to toggle play/pause of videos and audio, but does not handle KEY_MEDIA. Furthermore, Android itself now supports KEY_PLAYPAUSE equivalently, as the new USB headset spec requires KEY_PLAYPAUSE for BTN_0. https://source.android.com/devices/accessories/headset/usb-headset-spec Signed-off-by: Benson Leung <bleung(a)chromium.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/rockchip/rk3399_gru_sound.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index d64fbbd50544..fa6cd1de828b 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -206,7 +206,8 @@ static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } - snd_jack_set_key(rockchip_sound_jack.jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key( + rockchip_sound_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_jack_set_key( rockchip_sound_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); snd_jack_set_key( -- 2.15.0

1 0

[alsa-devel] Applied "ASoC: qcom: apq8016-sbc: Map BTN_0 to KEY_PLAYPAUSE" to the asoc tree
by Mark Brown 27 Nov '17

27 Nov '17

The patch ASoC: qcom: apq8016-sbc: Map BTN_0 to KEY_PLAYPAUSE has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 5f6d1df8ced6bb52453149f96d9b7005058641b6 Mon Sep 17 00:00:00 2001 From: Benson Leung <bleung(a)chromium.org> Date: Wed, 22 Nov 2017 12:56:43 -0800 Subject: [PATCH] ASoC: qcom: apq8016-sbc: Map BTN_0 to KEY_PLAYPAUSE The Android 3.5mm Headset jack specification mentions that BTN_0 should be mapped to KEY_MEDIA, but this is less logical than KEY_PLAYPAUSE, which has much broader userspace support. For example, the Chrome OS userspace now supports KEY_PLAYPAUSE to toggle play/pause of videos and audio, but does not handle KEY_MEDIA. Furthermore, Android itself now supports KEY_PLAYPAUSE equivalently, as the new USB headset spec requires KEY_PLAYPAUSE for BTN_0. https://source.android.com/devices/accessories/headset/usb-headset-spec Signed-off-by: Benson Leung <bleung(a)chromium.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/apq8016_sbc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index d49adc822a11..03851fedd1e2 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -92,7 +92,7 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) jack = pdata->jack.jack; - snd_jack_set_key(jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_jack_set_key(jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); -- 2.15.0

1 0

[alsa-devel] Applied "ASoC: Intel: kbl_rt5663_rt5514_max98927: Map BTN_0 to KEY_PLAYPAUSE" to the asoc tree
by Mark Brown 27 Nov '17

27 Nov '17

The patch ASoC: Intel: kbl_rt5663_rt5514_max98927: Map BTN_0 to KEY_PLAYPAUSE has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8ee649283b1e542aedba007f6c828d6767c48e0d Mon Sep 17 00:00:00 2001 From: Benson Leung <bleung(a)chromium.org> Date: Wed, 22 Nov 2017 12:56:39 -0800 Subject: [PATCH] ASoC: Intel: kbl_rt5663_rt5514_max98927: Map BTN_0 to KEY_PLAYPAUSE The Android 3.5mm Headset jack specification mentions that BTN_0 should be mapped to KEY_MEDIA, but this is less logical than KEY_PLAYPAUSE, which has much broader userspace support. For example, the Chrome OS userspace now supports KEY_PLAYPAUSE to toggle play/pause of videos and audio, but does not handle KEY_MEDIA. Furthermore, Android itself now supports KEY_PLAYPAUSE equivalently, as the new USB headset spec requires KEY_PLAYPAUSE for BTN_0. https://source.android.com/devices/accessories/headset/usb-headset-spec Signed-off-by: Benson Leung <bleung(a)chromium.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index e7672831bc49..38512f0d1a73 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -195,7 +195,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) } jack = &ctx->kabylake_headset; - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); -- 2.15.0

1 0

[alsa-devel] Applied "ASoC: Intel: kbl_rt5663_max98927: Map BTN_0 to KEY_PLAYPAUSE" to the asoc tree
by Mark Brown 27 Nov '17

27 Nov '17

The patch ASoC: Intel: kbl_rt5663_max98927: Map BTN_0 to KEY_PLAYPAUSE has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 38a5882e4292d135cebabad0b56c9420dfdd80a5 Mon Sep 17 00:00:00 2001 From: Benson Leung <bleung(a)chromium.org> Date: Wed, 22 Nov 2017 12:56:40 -0800 Subject: [PATCH] ASoC: Intel: kbl_rt5663_max98927: Map BTN_0 to KEY_PLAYPAUSE The Android 3.5mm Headset jack specification mentions that BTN_0 should be mapped to KEY_MEDIA, but this is less logical than KEY_PLAYPAUSE, which has much broader userspace support. For example, the Chrome OS userspace now supports KEY_PLAYPAUSE to toggle play/pause of videos and audio, but does not handle KEY_MEDIA. Furthermore, Android itself now supports KEY_PLAYPAUSE equivalently, as the new USB headset spec requires KEY_PLAYPAUSE for BTN_0. https://source.android.com/devices/accessories/headset/usb-headset-spec Signed-off-by: Benson Leung <bleung(a)chromium.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 6f9a8bcf20f3..94a34db4f8c0 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -225,7 +225,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) } jack = &ctx->kabylake_headset; - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); -- 2.15.0

1 0

[alsa-devel] [PATCH 0/3] ASoC: stm32: sai: fixes related to synchro feature
by Olivier Moysan 27 Nov '17

27 Nov '17

Fix sai binding and simplify code related to synchro feature. This change allows to remove stm32_sai_remove() function and use devm_of_platform_populate(). Olivier Moysan (3): ASoC: stm32: fix sync property description in SAI bindings ASoC: stm32: sai: simplify sync modes management ASoC: stm32: sai: use devm_of_platform_populate() .../devicetree/bindings/sound/st,stm32-sai.txt | 12 ++- sound/soc/stm/stm32_sai.c | 114 ++++----------------- 2 files changed, 29 insertions(+), 97 deletions(-) -- 1.9.1

3 7

[alsa-devel] [PATCH 0/3] ASoC-cs42l73: Adjustments for two function implementations
by SF Markus Elfring 27 Nov '17

27 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 22 Nov 2017 16:38:32 +0100 Three update suggestions were taken into account from static source code analysis. Markus Elfring (3): Delete an error message for a failed memory allocation in cs42l73_i2c_probe() Improve two size determinations in cs42l73_i2c_probe() Fix a typo in two comments sound/soc/codecs/cs42l73.c | 20 ++++++++------------ 1 file changed, 8 insertions(+), 12 deletions(-) -- 2.15.0

3 7

[alsa-devel] [PATCH] ASoC: wm0010: Delete an error message for a failed memory allocation in wm0010_boot()
by SF Markus Elfring 27 Nov '17

27 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 22 Nov 2017 17:17:48 +0100 Omit an extra message for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/codecs/wm0010.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 4f5f5710b569..0147d2fb7b0a 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -655,11 +655,8 @@ static int wm0010_boot(struct snd_soc_codec *codec) ret = -ENOMEM; len = pll_rec.length + 8; out = kzalloc(len, GFP_KERNEL | GFP_DMA); - if (!out) { - dev_err(codec->dev, - "Failed to allocate RX buffer\n"); + if (!out) goto abort; - } img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!img_swap) -- 2.15.0

3 2

[alsa-devel] [PATCH 0/2] ASoC: cs35l32: Adjustments for cs35l32_i2c_probe()
by SF Markus Elfring 27 Nov '17

27 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 22 Nov 2017 20:49:45 +0100 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Delete two error messages for a failed memory allocation Improve two size determinations sound/soc/codecs/cs35l32.c | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) -- 2.15.0

3 6

[alsa-devel] [PATCH 0/2] ASoC: cs35l34: Adjustments for cs35l34_i2c_probe()
by SF Markus Elfring 27 Nov '17

27 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 22 Nov 2017 21:23:45 +0100 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Delete two error messages for a failed memory allocation Improve two size determinations sound/soc/codecs/cs35l34.c | 19 ++++++------------- 1 file changed, 6 insertions(+), 13 deletions(-) -- 2.15.0

3 5

[alsa-devel] [PATCH] ASoC: amd: Add error checking to probe function
by Guenter Roeck 27 Nov '17

27 Nov '17

The acp_audio_dma does not perform sufficient error checking in its probe function. This can result in crashes if a critical error path is encountered. Fixes: 7c31335a03b6a ("ASoC: AMD: add AMD ASoC ACP 2.x DMA driver") Cc: Alex Deucher <alexander.deucher(a)amd.com> Cc: Dominik Behr <dbehr(a)chromium.org> Cc: Daniel Kurtz <djkurtz(a)chromium.org> Signed-off-by: Guenter Roeck <linux(a)roeck-us.net> --- I didn't add an error check to acp_init() since I was not sure if its return value is ignored on purpose. sound/soc/amd/acp-pcm-dma.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 9f521a55d610..b5e41df6bb3a 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1051,6 +1051,11 @@ static int acp_audio_probe(struct platform_device *pdev) struct resource *res; const u32 *pdata = pdev->dev.platform_data; + if (!pdata) { + dev_err(&pdev->dev, "Missing platform data\n"); + return -ENODEV; + } + audio_drv_data = devm_kzalloc(&pdev->dev, sizeof(struct audio_drv_data), GFP_KERNEL); if (audio_drv_data == NULL) @@ -1058,6 +1063,8 @@ static int acp_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); audio_drv_data->acp_mmio = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(audio_drv_data->acp_mmio)) + return PTR_ERR(audio_drv_data->acp_mmio); /* The following members gets populated in device 'open' * function. Till then interrupts are disabled in 'acp_init' -- 2.7.4

5 5

[alsa-devel] [PATCH 1/2] ASoC: fsl_ssi: AC'97 ops need regmap, clock and cleaning up on failure
by Maciej S. Szmigiero 27 Nov '17

27 Nov '17

AC'97 ops (register read / write) need SSI regmap and clock, so they have to be set after them. We also need to set these ops back to NULL if we fail the probe. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index dad80b4b0cfc..a71bb8391f61 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1460,12 +1460,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) sizeof(fsl_ssi_ac97_dai)); fsl_ac97_data = ssi_private; - - ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); - if (ret) { - dev_err(&pdev->dev, "could not set AC'97 ops\n"); - return ret; - } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1576,6 +1570,14 @@ static int fsl_ssi_probe(struct platform_device *pdev) return ret; } + if (fsl_ssi_is_ac97(ssi_private)) { + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + goto error_ac97_ops; + } + } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_ssi_component, &ssi_private->cpu_dai_drv, 1); if (ret) { @@ -1659,6 +1661,10 @@ static int fsl_ssi_probe(struct platform_device *pdev) fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); error_asoc_register: + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + +error_ac97_ops: if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private);

3 3

[alsa-devel] [PATCH 2/2] ASoC: fsl_ssi: serialize AC'97 register access operations
by Maciej S. Szmigiero 27 Nov '17

27 Nov '17

AC'97 register access operations (both read and write) on SSI use a one, shared set of SSI registers for AC'97 register address and data. This means that only one such access is possible at a time and so all these operations need to be serialized. Since an AC'97 register access operation in this driver takes 100us+ let's use a mutex for this. Use this opportunity to also change a default value returned from AC'97 register read function from -1 to 0, since that's what AC'97 specs require to be returned when unknown / undefined registers are read. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 26 ++++++++++++++++++++++---- 1 file changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index a71bb8391f61..9dea1b16de82 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -38,6 +38,7 @@ #include <linux/ctype.h> #include <linux/device.h> #include <linux/delay.h> +#include <linux/mutex.h> #include <linux/slab.h> #include <linux/spinlock.h> #include <linux/of.h> @@ -265,6 +266,8 @@ struct fsl_ssi_private { u32 fifo_watermark; u32 dma_maxburst; + + struct mutex ac97_reg_lock; }; /* @@ -1262,11 +1265,13 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, if (reg > 0x7f) return; + mutex_lock(&fsl_ac97_data->ac97_reg_lock); + ret = clk_prepare_enable(fsl_ac97_data->clk); if (ret) { pr_err("ac97 write clk_prepare_enable failed: %d\n", ret); - return; + goto ret_unlock; } lreg = reg << 12; @@ -1280,6 +1285,9 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, udelay(100); clk_disable_unprepare(fsl_ac97_data->clk); + +ret_unlock: + mutex_unlock(&fsl_ac97_data->ac97_reg_lock); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1287,16 +1295,18 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, { struct regmap *regs = fsl_ac97_data->regs; - unsigned short val = -1; + unsigned short val = 0; u32 reg_val; unsigned int lreg; int ret; + mutex_lock(&fsl_ac97_data->ac97_reg_lock); + ret = clk_prepare_enable(fsl_ac97_data->clk); if (ret) { pr_err("ac97 read clk_prepare_enable failed: %d\n", ret); - return -1; + goto ret_unlock; } lreg = (reg & 0x7f) << 12; @@ -1311,6 +1321,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, clk_disable_unprepare(fsl_ac97_data->clk); +ret_unlock: + mutex_unlock(&fsl_ac97_data->ac97_reg_lock); return val; } @@ -1571,6 +1583,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) } if (fsl_ssi_is_ac97(ssi_private)) { + mutex_init(&ssi_private->ac97_reg_lock); ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); if (ret) { dev_err(&pdev->dev, "could not set AC'97 ops\n"); @@ -1665,6 +1678,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) snd_soc_set_ac97_ops(NULL); error_ac97_ops: + if (fsl_ssi_is_ac97(ssi_private)) + mutex_destroy(&ssi_private->ac97_reg_lock); + if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); @@ -1683,8 +1699,10 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); - if (fsl_ssi_is_ac97(ssi_private)) + if (fsl_ssi_is_ac97(ssi_private)) { snd_soc_set_ac97_ops(NULL); + mutex_destroy(&ssi_private->ac97_reg_lock); + } return 0; }

3 3

[alsa-devel] [PATCH] ASoC: pcm512x-spi: add missing MODULE_DESCRIPTION/AUTHOR/LICENSE
by Jesse Chan 27 Nov '17

27 Nov '17

Signed-off-by: Jesse Chan <jc(a)linux.com> --- sound/soc/codecs/pcm512x-spi.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/pcm512x-spi.c b/sound/soc/codecs/pcm512x-spi.c index 25c63510ae15..7cdd2dc4fd79 100644 --- a/sound/soc/codecs/pcm512x-spi.c +++ b/sound/soc/codecs/pcm512x-spi.c @@ -70,3 +70,7 @@ static struct spi_driver pcm512x_spi_driver = { }; module_spi_driver(pcm512x_spi_driver); + +MODULE_DESCRIPTION("ASoC PCM512x codec driver - SPI"); +MODULE_AUTHOR("Mark Brown <broonie(a)kernel.org>"); +MODULE_LICENSE("GPL v2"); -- 2.14.1

2 2

[alsa-devel] [PATCH] ASoC: rt5514: Add the sanity check for the driver_data in the resume function
by Oder Chiou 27 Nov '17

27 Nov '17

If the rt5514 spi driver is loaded, but the snd_soc_platform_driver is not loaded by the correct DAI settings, the NULL pointer will be gotten by snd_soc_platform_get_drvdata in the resume function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index a7604b9..4339f23 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -483,10 +483,13 @@ static int __maybe_unused rt5514_resume(struct device *dev) if (device_may_wakeup(dev)) disable_irq_wake(irq); - if (rt5514_dsp->substream) { - rt5514_spi_burst_read(RT5514_IRQ_CTRL, (u8 *)&buf, sizeof(buf)); - if (buf[0] & RT5514_IRQ_STATUS_BIT) - rt5514_schedule_copy(rt5514_dsp); + if (rt5514_dsp) { + if (rt5514_dsp->substream) { + rt5514_spi_burst_read(RT5514_IRQ_CTRL, (u8 *)&buf, + sizeof(buf)); + if (buf[0] & RT5514_IRQ_STATUS_BIT) + rt5514_schedule_copy(rt5514_dsp); + } } return 0; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: remove duplicated flag setting in fsl_ssi_setup_reg_vals()
by Maciej S. Szmigiero 27 Nov '17

27 Nov '17

We don't need to set CCSR_SSI_SIER_RFF0_EN / CCSR_SSI_SIER_TFE0_EN bits in reg->rx.sier / reg->tx.sier variables in a non-AC'97 mode considering we had just initialized these variables to these very values unconditionally a few lines earlier. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f2f51e06e22c..48bb850a34d9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -597,9 +597,7 @@ static void fsl_ssi_setup_reg_vals(struct fsl_ssi_private *ssi_private) if (!fsl_ssi_is_ac97(ssi_private)) { reg->rx.scr = CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE; - reg->rx.sier |= CCSR_SSI_SIER_RFF0_EN; reg->tx.scr = CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE; - reg->tx.sier |= CCSR_SSI_SIER_TFE0_EN; } if (ssi_private->use_dma) {

3 2

[alsa-devel] [PATCH 0/2] ASoC: Correct DT binding examples IRQ usage for da7218/9
by Adam Thomson 27 Nov '17

27 Nov '17

Currently in the DT bindings documents for DA7218 and DA7219, the examples for bindings usage show ACTIVE_HIGH IRQ usage. For both devices this is incorrect as the devices themselves only suport ACTIVE_LOW. This patch set rectifies the inconsistency. Adam Thomson (2): ASoC: da7219: Correct IRQ level in DT binding example ASoC: da7218: Correct IRQ level in DT binding example Documentation/devicetree/bindings/sound/da7218.txt | 2 +- Documentation/devicetree/bindings/sound/da7219.txt | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) -- 1.9.1

3 6

[alsa-devel] [PATCH v2 00/14] soundwire: Add a new SoundWire subsystem
by Vinod Koul 27 Nov '17

27 Nov '17

This patch series adds a new SoundWire subsystem which implements a new MIPI bus protocol 'SoundWire'. The SoundWire protocol is a robust, scalable, low complexity, low power, low latency, two-pin (clock and data) multi-drop bus that allows for the transfer of multiple audio streams and embedded control/commands. SoundWire provides synchronization capabilities and supports both PCM and PDM, multichannel data, isochronous and asynchronous modes. This series adds SoundWire Bus, IO transfers, DisCo (Discovery and Configuration) sysfs interface, regmap and Documentation summary This patch series is also available on git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git topic/patch_v2 v1: https://lkml.org/lkml/2017/10/18/1030 RFC: https://lkml.org/lkml/2016/10/21/395 Changes in v2: - move documentation into driver-api and do rst conversion - fix documentation comments - add SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) to all source files - rework the transfer logic and paging logic as commented on v1 - remove dummy sysfs fns - registration checks and fixes - remove slave check for regamp as that turned superfluous - remove depends SoundWire symbol - make modalias api arg const - use bitmap for tracking assigned - add counter for report present tracking todo: add the dt-bindings Sanyog Kale (4): Documentation: Add SoundWire summary soundwire: Add SoundWire MIPI defined registers soundwire: Add Slave status handling helpers soundwire: cdns: Add sdw_master_ops and IO transfer support Vinod Koul (10): soundwire: Add SoundWire bus type soundwire: Add Master registration soundwire: Add MIPI DisCo property helpers soundwire: Add IO transfer regmap: Add SoundWire bus support soundwire: Add slave status handling soundwire: Add sysfs for SoundWire DisCo properties soundwire: cdns: Add cadence library soundwire: intel: Add Intel Master driver soundwire: intel: Add Intel init module Documentation/driver-api/index.rst | 1 + Documentation/driver-api/soundwire/index.rst | 15 + Documentation/driver-api/soundwire/summary.rst | 205 +++++ drivers/Kconfig | 2 + drivers/Makefile | 1 + drivers/base/regmap/Kconfig | 4 + drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-sdw.c | 140 ++++ drivers/soundwire/Kconfig | 36 + drivers/soundwire/Makefile | 18 + drivers/soundwire/bus.c | 1013 ++++++++++++++++++++++++ drivers/soundwire/bus.h | 126 +++ drivers/soundwire/bus_type.c | 245 ++++++ drivers/soundwire/cadence_master.c | 791 ++++++++++++++++++ drivers/soundwire/cadence_master.h | 96 +++ drivers/soundwire/intel.c | 391 +++++++++ drivers/soundwire/intel.h | 72 ++ drivers/soundwire/intel_init.c | 232 ++++++ drivers/soundwire/mipi_disco.c | 422 ++++++++++ drivers/soundwire/slave.c | 173 ++++ drivers/soundwire/sysfs.c | 388 +++++++++ include/linux/mod_devicetable.h | 6 + include/linux/regmap.h | 37 + include/linux/soundwire/sdw.h | 563 +++++++++++++ include/linux/soundwire/sdw_intel.h | 74 ++ include/linux/soundwire/sdw_registers.h | 242 ++++++ scripts/mod/devicetable-offsets.c | 4 + scripts/mod/file2alias.c | 15 + 28 files changed, 5313 insertions(+) create mode 100644 Documentation/driver-api/soundwire/index.rst create mode 100644 Documentation/driver-api/soundwire/summary.rst create mode 100644 drivers/base/regmap/regmap-sdw.c create mode 100644 drivers/soundwire/Kconfig create mode 100644 drivers/soundwire/Makefile create mode 100644 drivers/soundwire/bus.c create mode 100644 drivers/soundwire/bus.h create mode 100644 drivers/soundwire/bus_type.c create mode 100644 drivers/soundwire/cadence_master.c create mode 100644 drivers/soundwire/cadence_master.h create mode 100644 drivers/soundwire/intel.c create mode 100644 drivers/soundwire/intel.h create mode 100644 drivers/soundwire/intel_init.c create mode 100644 drivers/soundwire/mipi_disco.c create mode 100644 drivers/soundwire/slave.c create mode 100644 drivers/soundwire/sysfs.c create mode 100644 include/linux/soundwire/sdw.h create mode 100644 include/linux/soundwire/sdw_intel.h create mode 100644 include/linux/soundwire/sdw_registers.h -- 2.7.4

5 35

[alsa-devel] [PATCH v2 1/2] ALSA: pcm: add SNDRV_PCM_FORMAT_{S, U}20
by Maciej S. Szmigiero 27 Nov '17

27 Nov '17

This format is similar to existing SNDRV_PCM_FORMAT_{S,U}20_3 that keep 20-bit PCM samples in 3 bytes, however i.MX6 platform SSI FIFO does not allow 3-byte accesses (including DMA) so a 4-byte (more conventional) format is needed for it. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: Drop "_4" suffix from these formats since they aren't non-standard ones, use empty format slots starting from format number 25 for them, add information that they are LSB justified formats. Corresponding alsa-lib changes will be posted as soon as this patch is merged on the kernel side, to keep alsa-lib and kernel synchronized. include/sound/pcm.h | 8 ++++++++ include/sound/soc-dai.h | 2 ++ include/uapi/sound/asound.h | 9 +++++++++ sound/core/pcm_misc.c | 16 ++++++++++++++++ 4 files changed, 35 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 24febf9e177c..e054c583d3b3 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -169,6 +169,10 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_IMA_ADPCM _SNDRV_PCM_FMTBIT(IMA_ADPCM) #define SNDRV_PCM_FMTBIT_MPEG _SNDRV_PCM_FMTBIT(MPEG) #define SNDRV_PCM_FMTBIT_GSM _SNDRV_PCM_FMTBIT(GSM) +#define SNDRV_PCM_FMTBIT_S20_LE _SNDRV_PCM_FMTBIT(S20_LE) +#define SNDRV_PCM_FMTBIT_U20_LE _SNDRV_PCM_FMTBIT(U20_LE) +#define SNDRV_PCM_FMTBIT_S20_BE _SNDRV_PCM_FMTBIT(S20_BE) +#define SNDRV_PCM_FMTBIT_U20_BE _SNDRV_PCM_FMTBIT(U20_BE) #define SNDRV_PCM_FMTBIT_SPECIAL _SNDRV_PCM_FMTBIT(SPECIAL) #define SNDRV_PCM_FMTBIT_S24_3LE _SNDRV_PCM_FMTBIT(S24_3LE) #define SNDRV_PCM_FMTBIT_U24_3LE _SNDRV_PCM_FMTBIT(U24_3LE) @@ -202,6 +206,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_FLOAT SNDRV_PCM_FMTBIT_FLOAT_LE #define SNDRV_PCM_FMTBIT_FLOAT64 SNDRV_PCM_FMTBIT_FLOAT64_LE #define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE +#define SNDRV_PCM_FMTBIT_S20 SNDRV_PCM_FMTBIT_S20_LE +#define SNDRV_PCM_FMTBIT_U20 SNDRV_PCM_FMTBIT_U20_LE #endif #ifdef SNDRV_BIG_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_BE @@ -213,6 +219,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_FLOAT SNDRV_PCM_FMTBIT_FLOAT_BE #define SNDRV_PCM_FMTBIT_FLOAT64 SNDRV_PCM_FMTBIT_FLOAT64_BE #define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE +#define SNDRV_PCM_FMTBIT_S20 SNDRV_PCM_FMTBIT_S20_BE +#define SNDRV_PCM_FMTBIT_U20 SNDRV_PCM_FMTBIT_U20_BE #endif struct snd_pcm_file { diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 58acd00cae19..d970879944fc 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -102,6 +102,8 @@ struct snd_compr_stream; SNDRV_PCM_FMTBIT_S16_BE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S20_3BE |\ + SNDRV_PCM_FMTBIT_S20_LE |\ + SNDRV_PCM_FMTBIT_S20_BE |\ SNDRV_PCM_FMTBIT_S24_3LE |\ SNDRV_PCM_FMTBIT_S24_3BE |\ SNDRV_PCM_FMTBIT_S32_LE |\ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index c227ccba60ae..7385024041d2 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -214,6 +214,11 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_IMA_ADPCM ((__force snd_pcm_format_t) 22) #define SNDRV_PCM_FORMAT_MPEG ((__force snd_pcm_format_t) 23) #define SNDRV_PCM_FORMAT_GSM ((__force snd_pcm_format_t) 24) +#define SNDRV_PCM_FORMAT_S20_LE ((__force snd_pcm_format_t) 25) /* \ */ +#define SNDRV_PCM_FORMAT_S20_BE ((__force snd_pcm_format_t) 26) /* | */ +#define SNDRV_PCM_FORMAT_U20_LE ((__force snd_pcm_format_t) 27) /* | in four bytes, */ +#define SNDRV_PCM_FORMAT_U20_BE ((__force snd_pcm_format_t) 28) /* / LSB justified */ +/* gap in the numbering for a future standard linear format */ #define SNDRV_PCM_FORMAT_SPECIAL ((__force snd_pcm_format_t) 31) #define SNDRV_PCM_FORMAT_S24_3LE ((__force snd_pcm_format_t) 32) /* in three bytes */ #define SNDRV_PCM_FORMAT_S24_3BE ((__force snd_pcm_format_t) 33) /* in three bytes */ @@ -248,6 +253,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_LE #define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_LE #define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE +#define SNDRV_PCM_FORMAT_S20 SNDRV_PCM_FORMAT_S20_LE +#define SNDRV_PCM_FORMAT_U20 SNDRV_PCM_FORMAT_U20_LE #endif #ifdef SNDRV_BIG_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_BE @@ -259,6 +266,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_BE #define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_BE #define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE +#define SNDRV_PCM_FORMAT_S20 SNDRV_PCM_FORMAT_S20_BE +#define SNDRV_PCM_FORMAT_U20 SNDRV_PCM_FORMAT_U20_BE #endif typedef int __bitwise snd_pcm_subformat_t; diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 9be81025372f..c62bfe27106f 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -170,6 +170,22 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { [SNDRV_PCM_FORMAT_GSM] = { .le = -1, .signd = -1, }, + [SNDRV_PCM_FORMAT_S20_LE] = { + .width = 20, .phys = 32, .le = 1, .signd = 1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_S20_BE] = { + .width = 20, .phys = 32, .le = 0, .signd = 1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_U20_LE] = { + .width = 20, .phys = 32, .le = 1, .signd = 0, + .silence = { 0x00, 0x00, 0x08, 0x00 }, + }, + [SNDRV_PCM_FORMAT_U20_BE] = { + .width = 20, .phys = 32, .le = 0, .signd = 0, + .silence = { 0x00, 0x08, 0x00, 0x00 }, + }, [SNDRV_PCM_FORMAT_SPECIAL] = { .le = -1, .signd = -1, },

2 2

[alsa-devel] [PATCH v3 2/2] ASoC: fsl_ssi: add 20-bit sample format for AC'97 and use it for capture
by Maciej S. Szmigiero 27 Nov '17

27 Nov '17

When testing AC'97 capture on UDOO board (currently the only user of fsl_ssi driver in the AC'97 mode) it become obvious that there is a massive distortion above certain, small input signal. This problem has been traced to silicon errata ERR003778: "In AC97, 16-bit mode, received data is shifted by 4-bit locations" that has "No fix scheduled". This errata suggests a workaround of doing a 4-bit shift back in SDMA script for this specific operation mode, however our SDMA scripts are shared between various SoC peripherals so we can't really modify them. There is a simple way to avoid this problem, however, that is to disallow recording in 16-bit mode and only support it in AC'97-native 20-bit mode. We have to use a 4-byte format for this since SSI FIFOs do not allow 3-byte accesses (and these aren't supported by imx-sdma driver anyway). With this change the capture distortion is gone. We can also add this format as an additional one supported for playback, using this opportunity to make sure that we use CPU-endian-native formats in AC'97 mode as we already do in I2S mode. There is no problem in using different bit widths in playback and capture in AC'97 mode so allow this, too. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: Adapt format name to changes in the first patch from this series. Changes from v2: None. sound/soc/fsl/fsl_ssi.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 20ef09e1a395..c350117c8e31 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1278,14 +1278,15 @@ static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S20, }, .capture = { .stream_name = "AC97 Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + /* 16-bit capture is broken (errata ERR003778) */ + .formats = SNDRV_PCM_FMTBIT_S20, }, .ops = &fsl_ssi_dai_ops, }; @@ -1557,11 +1558,12 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - if (!fsl_ssi_is_ac97(ssi_private)) + if (!fsl_ssi_is_ac97(ssi_private)) { ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_samplebits = 1; + } ssi_private->cpu_dai_drv.symmetric_channels = 1; - ssi_private->cpu_dai_drv.symmetric_samplebits = 1; } /* Determine the FIFO depth. */

1 0

[alsa-devel] [PATCH] ASoC: fsl_ssi: Override bit clock rate based on slot number
by Nicolin Chen 26 Nov '17

26 Nov '17

The set_sysclk() now is used to override the output bit clock rate. But this is not a common way to implement a set_dai_sysclk(). And this creates a problem when a general machine driver (simple-card for example) tries to do set_dai_sysclk() by passing an input clock rate for the baud clock instead of setting the bit clock rate as fsl_ssi driver expected. So this patch solves this problem by firstly removing set_sysclk() since the hw_params() can calculate the bit clock rate. Secondly, in order not to break those TDM use cases which previously might have been using set_sysclk() to override the bit clock rate, this patch changes the driver to override it based on the slot number. The patch also removes an obsolete comment of the dir parameter. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_ssi.c | 26 ++++++++------------------ 1 file changed, 8 insertions(+), 18 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 64598d1..3657c88 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -197,12 +197,12 @@ struct fsl_ssi_soc_data { * @use_dma: DMA is used or FIQ with stream filter * @use_dual_fifo: DMA with support for both FIFOs used * @fifo_deph: Depth of the SSI FIFOs + * @slots: number of slots * @rxtx_reg_val: Specific register settings for receive/transmit configuration * * @clk: SSI clock * @baudclk: SSI baud clock for master mode * @baudclk_streams: Active streams that are using baudclk - * @bitclk_freq: bitclock frequency set by .set_dai_sysclk * * @dma_params_tx: DMA transmit parameters * @dma_params_rx: DMA receive parameters @@ -233,12 +233,12 @@ struct fsl_ssi_private { bool use_dual_fifo; bool has_ipg_clk_name; unsigned int fifo_depth; + unsigned int slots; struct fsl_ssi_rxtx_reg_val rxtx_reg_val; struct clk *clk; struct clk *baudclk; unsigned int baudclk_streams; - unsigned int bitclk_freq; /* regcache for volatile regs */ u32 regcache_sfcsr; @@ -700,8 +700,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, * Note: This function can be only called when using SSI as DAI master * * Quick instruction for parameters: - * freq: Output BCLK frequency = samplerate * 32 (fixed) * channels - * dir: SND_SOC_CLOCK_OUT -> TxBCLK, SND_SOC_CLOCK_IN -> RxBCLK. + * freq: Output BCLK frequency = samplerate * 32 (fixed) * slots (or channels) */ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai, @@ -716,9 +715,9 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, unsigned int freq; bool baudclk_is_used; - /* Prefer the explicitly set bitclock frequency */ - if (ssi_private->bitclk_freq) - freq = ssi_private->bitclk_freq; + /* Generate bit clock based on the slot or channel number */ + if (ssi_private->slots) + freq = ssi_private->slots * 32 * params_rate(hw_params); else freq = params_channels(hw_params) * 32 * params_rate(hw_params); @@ -805,16 +804,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, return 0; } -static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); - - ssi_private->bitclk_freq = freq; - - return 0; -} - /** * fsl_ssi_hw_params - program the sample size * @@ -1121,6 +1110,8 @@ static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, regmap_update_bits(regs, CCSR_SSI_SCR, CCSR_SSI_SCR_SSIEN, val); + ssi_private->slots = slots; + return 0; } @@ -1191,7 +1182,6 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .hw_params = fsl_ssi_hw_params, .hw_free = fsl_ssi_hw_free, .set_fmt = fsl_ssi_set_dai_fmt, - .set_sysclk = fsl_ssi_set_dai_sysclk, .set_tdm_slot = fsl_ssi_set_dai_tdm_slot, .trigger = fsl_ssi_trigger, }; -- 2.7.4

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[alsa-devel] Issue with the Realtek ALC 898 codec on the Clevo P751DM2-G and the ESS Sabre Audio Jack for the headphone output.
by Dennis Mungai 26 Nov '17

26 Nov '17

Hello there, I have Ubuntu 16.04LTS installed on the Clevo P751DM2-G, also marketed as the Schenker XMG U507, the System76's Serval WS (2017 model), the Eurocom Sky X4E2, among others. The sound card is the Realtek ALC898 with the following output configuration: 1. One SPDIF/headphone combo jack (to the right) 2. One Line Out. 3. One Line-In. 4. One Mic-In. Background information: On Windows, one can use the Realtek HD Audio control panel pop-up menu on jack detection to set the output type when an audio device is plugged in to an appropriate jack, combined with Sound Blaster X-Fi MB5 software to enable post-processing effects such as Bass, noise cancellation, etc. What I have tried so far: 1. Setting the model definition at /etc/modprobe.d/sound.conf with the following entries: options snd-hda-intel model=no-primary-hp enable=1 index=0 Using such a line worked before on an MSI GS43VR 6RE Phantom Pro that also has the same (or a similar) ESS Sabre Audio DAC component. Expected behavior: When plugged into the microphone jack, the ESS Sabre DAC should be activated and sound should be routed to the headset. Observed anomaly: Sound comes out through the speakers instead. However, the line-out output works. In that case, when plugged into the line out, Audio works as expected. I was able to report this as a bug to Launchpad, with the information below: https://bugs.launchpad.net/bugs/1731940 ProblemType: Bug DistroRelease: Ubuntu 16.04 Package: alsa-base 1.0.25+dfsg-0ubuntu5 Uname: Linux 4.14.0 x86_64 NonfreeKernelModules: nvidia_uvm nvidia_drm nvidia_modeset nvidia ApportVersion: 2.20.1-0ubuntu2.10 Architecture: amd64 AudioDevicesInUse: USER PID ACCESS COMMAND /dev/snd/controlC1: lin 2199 F.... pulseaudio /dev/snd/controlC0: lin 2199 F.... pulseaudio CurrentDesktop: GNOME Date: Mon Nov 13 17:34:53 2017 InstallationDate: Installed on 2017-07-25 (111 days ago) InstallationMedia: Ubuntu 16.04.2 LTS "Xenial Xerus" - Release amd64 (20170215.2) PackageArchitecture: all SourcePackage: alsa-driver Symptom: audio Symptom_AlsaPlaybackTest: ALSA playback test through plughw:PCH failed Symptom_Card: Built-in Audio - HDA Intel PCH Symptom_DevicesInUse: USER PID ACCESS COMMAND /dev/snd/controlC1: gdm 1812 F.... pulseaudio lin 2199 F.... pulseaudio /dev/snd/controlC0: gdm 1812 F.... pulseaudio lin 2199 F.... pulseaudio Symptom_Jack: Speaker, Internal Symptom_Type: No sound at all Title: [P7xxDM2(-G), Realtek ALC898, Speaker, Internal] No sound at all UpgradeStatus: No upgrade log present (probably fresh install) dmi.bios.date: 12/21/2016 dmi.bios.vendor: American Megatrends Inc. dmi.bios.version: 1.06.04 dmi.board.asset.tag: Tag 12345 dmi.board.name: P7xxDM2(-G) dmi.board.vendor: Notebook dmi.board.version: Not Applicable dmi.chassis.asset.tag: No Asset Tag dmi.chassis.type: 10 dmi.chassis.vendor: No Enclosure dmi.chassis.version: N/A dmi.modalias: dmi:bvnAmericanMegatrendsInc.:bvr1.06.04:bd12/21/2016:svnNotebook:pnP7xxDM2(-G):pvrNotApplicable:rvnNotebook:rnP7xxDM2(-G):rvrNotApplicable:cvnNoEnclosure:ct10:cvrN/A: dmi.product.family: Not Applicable dmi.product.name: P7xxDM2(-G) dmi.product.version: Not Applicable dmi.sys.vendor: Notebook ** Affects: alsa-driver (Ubuntu) Importance: Undecided Status: New Regards, Dennis.

1 0

[alsa-devel] LADSPA ALSA plugin
by Robert Bielik 26 Nov '17

26 Nov '17

Hi all, Started experimenting with using LADSPA plugins with the ALSA LADSPA PCM plugin, and it works nicely. However, it seems only the PCM is interface is exposed from the plugin. Question is if there is a way to expose the LADSPA plugin parameters directly to alsamixer ? Regards /Rob

1 1

[alsa-devel] [PATCH] ASoC: amd: added error checks in dma driver
by Vijendar Mukunda 24 Nov '17

24 Nov '17

added error checks in acp dma driver Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Akshu Agrawal <Akshu.Agrawal(a)amd.com> Signed-off-by: Guenter Roeck <groeck(a)chromium.org> --- sound/soc/amd/acp-pcm-dma.c | 45 ++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 40 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 17d76fa..804e659 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -848,6 +848,9 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + if (!rtd) + return -EINVAL; + buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd->acp_mmio, substream->stream); @@ -873,6 +876,8 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + if (!rtd) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { config_acp_dma_channel(rtd->acp_mmio, SYSRAM_TO_ACP_CH_NUM, PLAYBACK_START_DMA_DESCR_CH12, @@ -1066,6 +1071,10 @@ static int acp_audio_probe(struct platform_device *pdev) struct resource *res; const u32 *pdata = pdev->dev.platform_data; + if (!pdata) { + dev_err(&pdev->dev, "Missing platform data\n"); + return -ENODEV; + } audio_drv_data = devm_kzalloc(&pdev->dev, sizeof(struct audio_drv_data), GFP_KERNEL); if (audio_drv_data == NULL) @@ -1074,6 +1083,8 @@ static int acp_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); audio_drv_data->acp_mmio = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(audio_drv_data->acp_mmio)) + return PTR_ERR(audio_drv_data->acp_mmio); /* The following members gets populated in device 'open' * function. Till then interrupts are disabled in 'acp_init' * and device doesn't generate any interrupts. @@ -1099,7 +1110,11 @@ static int acp_audio_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, audio_drv_data); /* Initialize the ACP */ - acp_init(audio_drv_data->acp_mmio, audio_drv_data->asic_type); + status = acp_init(audio_drv_data->acp_mmio, audio_drv_data->asic_type); + if (status) { + dev_err(&pdev->dev, "ACP Init failed\n"); + return status; + } status = snd_soc_register_platform(&pdev->dev, &acp_asoc_platform); if (status != 0) { @@ -1116,9 +1131,14 @@ static int acp_audio_probe(struct platform_device *pdev) static int acp_audio_remove(struct platform_device *pdev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(&pdev->dev); - acp_deinit(adata->acp_mmio); + status = acp_deinit(adata->acp_mmio); + if (status) { + dev_err(&pdev->dev, "ACP Deinit failed\n"); + return status; + } snd_soc_unregister_platform(&pdev->dev); pm_runtime_disable(&pdev->dev); @@ -1128,9 +1148,14 @@ static int acp_audio_remove(struct platform_device *pdev) static int acp_pcm_resume(struct device *dev) { u16 bank; + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_init(adata->acp_mmio, adata->asic_type); + status = acp_init(adata->acp_mmio, adata->asic_type); + if (status) { + dev_err(dev, "ACP Init failed\n"); + return status; + } if (adata->play_stream && adata->play_stream->runtime) { /* For Stoney, Memory gating is disabled,i.e SRAM Banks @@ -1162,18 +1187,28 @@ static int acp_pcm_resume(struct device *dev) static int acp_pcm_runtime_suspend(struct device *dev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_deinit(adata->acp_mmio); + status = acp_deinit(adata->acp_mmio); + if (status) { + dev_err(dev, "ACP Deinit failed\n"); + return status; + } acp_reg_write(0, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } static int acp_pcm_runtime_resume(struct device *dev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_init(adata->acp_mmio, adata->asic_type); + status = acp_init(adata->acp_mmio, adata->asic_type); + if (status) { + dev_err(dev, "ACP Init failed\n"); + return status; + } acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } -- 2.7.4

4 5

[alsa-devel] [PATCH] SoC: rt5514: Make sure the DMIC delay will be happened after normal SUPPLY widgets power on
by Oder Chiou 24 Nov '17

24 Nov '17

The patch makes sure the DMIC delay will be happened after normal SUPPLY widgets power on. If there are some platforms that provide the MCLK using the SUPPLY widget, it will make sure the delay time is helpful. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 61ccbc6..198df01 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -567,7 +567,7 @@ static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SUPPLY_S("DMIC CLK", 1, SND_SOC_NOPM, 0, 0, rt5514_set_dmic_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("ADC CLK", RT5514_CLK_CTRL1, -- 2.7.4

2 1

[alsa-devel] [PATCH] test: correct emulation for channel-map TLV
by Takashi Sakamoto 24 Nov '17

24 Nov '17

Current implementation of channel-map TLV on test program is not valid. Furthermore, it brings buffer-over-run due to byte counting. This commit fixes it. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- test/user-ctl-element-set.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) diff --git a/test/user-ctl-element-set.c b/test/user-ctl-element-set.c index 668831a8..e94152b9 100644 --- a/test/user-ctl-element-set.c +++ b/test/user-ctl-element-set.c @@ -126,33 +126,39 @@ static void change_int_elem_members(struct elem_set_trial *trial, static int allocate_int_elem_set_tlv(struct elem_set_trial *trial, unsigned int **tlv) { - unsigned int len, pos; + unsigned int count, pos; unsigned int i, j; struct chmap_entry *entry; /* Calculate size of TLV packet for channel-mapping information. */ - len = 0; + count = 0; for (i = 1; i <= 25; ++i) { - len += sizeof(struct chmap_entry); - len += i * sizeof(unsigned int); + count += 2; /* sizeof(struct chmap_entry). */ + count += i; /* struct chmap_entry.maps. */ } - *tlv = malloc(len); - if (*tlv == NULL) + *tlv = malloc((2 + count) * sizeof(unsigned int)); + if (!*tlv) return -ENOMEM; /* * Emulate channel-mapping information in in-kernel implementation. * Here, 25 entries are for each different channel. */ - pos = 0; - for (i = 1; i <= 25 && pos < len; ++i) { + (*tlv)[0] = SNDRV_CTL_TLVT_CONTAINER; + (*tlv)[1] = count * sizeof(unsigned int); + pos = 2; + + for (i = 1; i <= 25 && pos < count; ++i) { entry = (struct chmap_entry *)&(*tlv)[pos]; + entry->type = SNDRV_CTL_TLVT_CHMAP_FIXED; entry->length = i * sizeof(unsigned int); + pos += 2; + for (j = 0; j < i; ++j) entry->maps[j] = SND_CHMAP_MONO + j; - pos += sizeof(struct chmap_entry) + i * sizeof(unsigned int); + pos += i; } return 0; -- 2.14.1

2 1

[alsa-devel] [PATCH v2] ASoC: rt5514: Make sure the DMIC delay will be happened after normal SUPPLY widgets power on
by Oder Chiou 24 Nov '17

24 Nov '17

The patch makes sure the DMIC delay will be happened after normal SUPPLY widgets power on. If there are some platforms that provide the MCLK using the SUPPLY widget, it will make sure the delay time is helpful. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 61ccbc6..198df01 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -567,7 +567,7 @@ static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SUPPLY_S("DMIC CLK", 1, SND_SOC_NOPM, 0, 0, rt5514_set_dmic_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("ADC CLK", RT5514_CLK_CTRL1, -- 2.7.4

1 0

Re: [alsa-devel] Focusrite Scarlett 6i6 gen1 - input handling fix
by Jens Verwiebe 23 Nov '17

23 Nov '17

Dammit, sorry, i better add the patch as textfile (?) Formatting looks good here, but ml shows spaces where i expect tabs. New try: The Scarlett 6i6 has no padding on rear inputs 3/4 but a gainstage (Low/Hi). Adding this functionality Signed-off-by: Jens Verwiebe <info at jensverwiebe.de> 1 file changed, 21 insertions(+), 2 deletions(-) -- Jens Verwiebe Allerskehre 44 - 22309 Hamburg Tel.: +49 40 68 78 50 mobile: +49 172 400 49 07 mailto: info(a)jensverwiebe.de web: http://www.jensverwiebe.de

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[alsa-devel] [PATCH 1/2] ALSA: pcm: add SNDRV_PCM_FORMAT_{S, U}20_4
by Maciej S. Szmigiero 23 Nov '17

23 Nov '17

This format is similar to existing SNDRV_PCM_FORMAT_{S,U}20_3 that keep 20-bit PCM samples in 3 bytes, however i.MX6 platform SSI FIFO does not allow 3-byte accesses (including DMA) so a 4-byte format is needed for it. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- include/sound/pcm.h | 8 ++++++++ include/sound/soc-dai.h | 2 ++ include/uapi/sound/asound.h | 10 +++++++++- sound/core/pcm_misc.c | 16 ++++++++++++++++ 4 files changed, 35 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 24febf9e177c..7ad2d3f0934f 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -191,6 +191,10 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_DSD_U32_LE _SNDRV_PCM_FMTBIT(DSD_U32_LE) #define SNDRV_PCM_FMTBIT_DSD_U16_BE _SNDRV_PCM_FMTBIT(DSD_U16_BE) #define SNDRV_PCM_FMTBIT_DSD_U32_BE _SNDRV_PCM_FMTBIT(DSD_U32_BE) +#define SNDRV_PCM_FMTBIT_S20_4LE _SNDRV_PCM_FMTBIT(S20_4LE) +#define SNDRV_PCM_FMTBIT_U20_4LE _SNDRV_PCM_FMTBIT(U20_4LE) +#define SNDRV_PCM_FMTBIT_S20_4BE _SNDRV_PCM_FMTBIT(S20_4BE) +#define SNDRV_PCM_FMTBIT_U20_4BE _SNDRV_PCM_FMTBIT(U20_4BE) #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE @@ -202,6 +206,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_FLOAT SNDRV_PCM_FMTBIT_FLOAT_LE #define SNDRV_PCM_FMTBIT_FLOAT64 SNDRV_PCM_FMTBIT_FLOAT64_LE #define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE +#define SNDRV_PCM_FMTBIT_S20_4 SNDRV_PCM_FMTBIT_S20_4LE +#define SNDRV_PCM_FMTBIT_U20_4 SNDRV_PCM_FMTBIT_U20_4LE #endif #ifdef SNDRV_BIG_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_BE @@ -213,6 +219,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_FLOAT SNDRV_PCM_FMTBIT_FLOAT_BE #define SNDRV_PCM_FMTBIT_FLOAT64 SNDRV_PCM_FMTBIT_FLOAT64_BE #define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE +#define SNDRV_PCM_FMTBIT_S20_4 SNDRV_PCM_FMTBIT_S20_4BE +#define SNDRV_PCM_FMTBIT_U20_4 SNDRV_PCM_FMTBIT_U20_4BE #endif struct snd_pcm_file { diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 58acd00cae19..16aec0cc96f5 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -102,6 +102,8 @@ struct snd_compr_stream; SNDRV_PCM_FMTBIT_S16_BE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S20_3BE |\ + SNDRV_PCM_FMTBIT_S20_4LE |\ + SNDRV_PCM_FMTBIT_S20_4BE |\ SNDRV_PCM_FMTBIT_S24_3LE |\ SNDRV_PCM_FMTBIT_S24_3BE |\ SNDRV_PCM_FMTBIT_S32_LE |\ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index c227ccba60ae..69b661816491 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -236,7 +236,11 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_DSD_U32_LE ((__force snd_pcm_format_t) 50) /* DSD, 4-byte samples DSD (x32), little endian */ #define SNDRV_PCM_FORMAT_DSD_U16_BE ((__force snd_pcm_format_t) 51) /* DSD, 2-byte samples DSD (x16), big endian */ #define SNDRV_PCM_FORMAT_DSD_U32_BE ((__force snd_pcm_format_t) 52) /* DSD, 4-byte samples DSD (x32), big endian */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U32_BE +#define SNDRV_PCM_FORMAT_S20_4LE ((__force snd_pcm_format_t) 53) /* in four bytes */ +#define SNDRV_PCM_FORMAT_S20_4BE ((__force snd_pcm_format_t) 54) /* in four bytes */ +#define SNDRV_PCM_FORMAT_U20_4LE ((__force snd_pcm_format_t) 55) /* in four bytes */ +#define SNDRV_PCM_FORMAT_U20_4BE ((__force snd_pcm_format_t) 56) /* in four bytes */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_U20_4BE #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE @@ -248,6 +252,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_LE #define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_LE #define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE +#define SNDRV_PCM_FORMAT_S20_4 SNDRV_PCM_FORMAT_S20_4LE +#define SNDRV_PCM_FORMAT_U20_4 SNDRV_PCM_FORMAT_U20_4LE #endif #ifdef SNDRV_BIG_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_BE @@ -259,6 +265,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_BE #define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_BE #define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE +#define SNDRV_PCM_FORMAT_S20_4 SNDRV_PCM_FORMAT_S20_4BE +#define SNDRV_PCM_FORMAT_U20_4 SNDRV_PCM_FORMAT_U20_4BE #endif typedef int __bitwise snd_pcm_subformat_t; diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 9be81025372f..132ee41edae1 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -229,6 +229,22 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { .width = 5, .phys = 8, .le = -1, .signd = -1, .silence = {}, }, + [SNDRV_PCM_FORMAT_S20_4LE] = { + .width = 20, .phys = 32, .le = 1, .signd = 1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_S20_4BE] = { + .width = 20, .phys = 32, .le = 0, .signd = 1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_U20_4LE] = { + .width = 20, .phys = 32, .le = 1, .signd = 0, + .silence = { 0x00, 0x00, 0x08, 0x00 }, + }, + [SNDRV_PCM_FORMAT_U20_4BE] = { + .width = 20, .phys = 32, .le = 0, .signd = 0, + .silence = { 0x00, 0x08, 0x00, 0x00 }, + }, };

3 6

[alsa-devel] [PATCH] ASoC: nau8825: disable crosstalk by default
by John Hsu 23 Nov '17

23 Nov '17

The driver makes the crosstalk funciton disabled by default which can simplify the codec function. The platform may not need this funciton and reduce the potential risk. Therefore, We change the property "nuvoton,crosstalk-bypass" to "nuvoton,crosstalk-enable". The crosstalk measurement is enabled if the property is set. Otherwise, it is disabled. Besides, add more condition in the entry point of the crosstalk sequence to disable the function completely. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- Documentation/devicetree/bindings/sound/nau8825.txt | 4 ++-- sound/soc/codecs/nau8825.c | 21 +++++++++++++-------- sound/soc/codecs/nau8825.h | 2 +- 3 files changed, 16 insertions(+), 11 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt index 2f5e973..d16d968 100644 --- a/Documentation/devicetree/bindings/sound/nau8825.txt +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -69,7 +69,7 @@ Optional properties: - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - nuvoton,crosstalk-bypass: make crosstalk function bypass if set. + - nuvoton,crosstalk-enable: make crosstalk function enable if set. - clocks: list of phandle and clock specifier pairs according to common clock bindings for the clocks described in clock-names @@ -98,7 +98,7 @@ Example: nuvoton,short-key-debounce = <2>; nuvoton,jack-insert-debounce = <7>; nuvoton,jack-eject-debounce = <7>; - nuvoton,crosstalk-bypass; + nuvoton,crosstalk-enable; clock-names = "mclk"; clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>; diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 714ce17..2d6c536 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -792,6 +792,10 @@ static void nau8825_xtalk_work(struct work_struct *work) struct nau8825 *nau8825 = container_of( work, struct nau8825, xtalk_work); + /* block when crosstalk disabled */ + if (!nau8825->xtalk_enable) + return; + nau8825_xtalk_measure(nau8825); /* To determine the cross talk side tone gain when reach * the impedance measure state. @@ -815,11 +819,12 @@ static void nau8825_xtalk_work(struct work_struct *work) static void nau8825_xtalk_cancel(struct nau8825 *nau8825) { - /* If the xtalk_protect is true, that means the process is still - * on going. The driver forces to cancel the cross talk task and + /* If the crosstalk is eanbled and the process is on going, + * the driver forces to cancel the cross talk task and * restores the configuration to original status. */ - if (nau8825->xtalk_protect) { + if (nau8825->xtalk_enable && nau8825->xtalk_protect && + nau8825->xtalk_state != NAU8825_XTALK_DONE) { cancel_work_sync(&nau8825->xtalk_work); nau8825_xtalk_clean(nau8825); } @@ -1686,7 +1691,7 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { if (nau8825_is_jack_inserted(regmap)) { event |= nau8825_jack_insert(nau8825); - if (!nau8825->xtalk_bypass && !nau8825->high_imped) { + if (nau8825->xtalk_enable && !nau8825->high_imped) { /* Apply the cross talk suppression in the * headset without high impedance. */ @@ -2440,8 +2445,8 @@ static void nau8825_print_device_properties(struct nau8825 *nau8825) nau8825->jack_insert_debounce); dev_dbg(dev, "jack-eject-debounce: %d\n", nau8825->jack_eject_debounce); - dev_dbg(dev, "crosstalk-bypass: %d\n", - nau8825->xtalk_bypass); + dev_dbg(dev, "crosstalk-enable: %d\n", + nau8825->xtalk_enable); } static int nau8825_read_device_properties(struct device *dev, @@ -2506,8 +2511,8 @@ static int nau8825_read_device_properties(struct device *dev, &nau8825->jack_eject_debounce); if (ret) nau8825->jack_eject_debounce = 0; - nau8825->xtalk_bypass = device_property_read_bool(dev, - "nuvoton,crosstalk-bypass"); + nau8825->xtalk_enable = device_property_read_bool(dev, + "nuvoton,crosstalk-enable"); nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 8aee5c86..199d6ea 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -476,7 +476,7 @@ struct nau8825 { int xtalk_event_mask; bool xtalk_protect; int imp_rms[NAU8825_XTALK_IMM]; - int xtalk_bypass; + int xtalk_enable; }; int nau8825_enable_jack_detect(struct snd_soc_codec *codec, -- 2.6.4

2 4

[alsa-devel] [PATCH] ALSA: hda: Add Raven PCI ID
by Vijendar Mukunda 23 Nov '17

23 Nov '17

This commit adds PCI ID for Raven platform v2: Comment added for Raven Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 01eb1dc..44babe5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2454,6 +2454,9 @@ static const struct pci_device_id azx_ids[] = { /* AMD Hudson */ { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + /* AMD Raven */ + { PCI_DEVICE(0x1022, 0x15e3), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x0002), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, -- 2.7.4

2 1

Re: [alsa-devel] AMD Raven Platform
by Kailang 23 Nov '17

23 Nov '17

Hi Takashi, Attach patch for fix AMD no sound issue. It also could solve Intel ALC700 no sound issue. Thanks. BR, Kailang > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Tuesday, November 21, 2017 2:40 PM > To: Kailang > Subject: Re: AMD Raven Platform > > On Tue, 21 Nov 2017 04:13:16 +0100, > Kailang wrote: > > > > Hi Takashi, > > > > AMD guys add below patch. To support their new HDA controller. > > But it still no sound. > > This platform codec is ALC701. > > But ALC701 was worked with Intel platform on Intel site. > > My test platform was ICH7. ALC701 and ICH7 was also worked > on my site. > > > > I check alsa-info dump as attach file. It's very well. > > > > The patch that AMD guys assigned ATI_SB flag. > > I think no sound issue maybe the HDA controller setting relation. > > > > Could you can find any difference? > > First off, if the main target is the onboard analog I/O, I'd > pass index=1,0 option to snd-hda-intel module to swap the > HDMI/DP and the analog I/O. This will allow testing with the > native ALSA API more easily. > > Did you pass need some COEF setups as we applied for Intel > reference board? There is even another patch that isn't > still included to upstream (the difference is COEF 0x4a with > bit 0x2000). > > In the controller side, one thing that comes to my mind about > AMD is the PCI snooping. Try AZX_DCAPS_PRESET_ATI_HDMI_NS > instead as a test. > The essential difference with this preset is > AZX_DCAPS_SNOOP_OFF bits. Other than that, the recent Intel > chip requires the different slot assignment, and > AZX_DCAPS_SEPARATE_STREAM_TAG is used. But this is set > automatically also when the number of streams is greater than > 15, too. (You'll see a warning.) > > Feel free to put me to Cc in the communication with AMD > people directly. > > > thanks, > > Takashi > > > > > Many Thanks. > > Kailang > > --- a/sound/pci/hda/hda_intel.c > > +++ b/sound/pci/hda/hda_intel.c > > @@ -2454,6 +2454,8 @@ static const struct pci_device_id > azx_ids[] = { > > /* AMD Hudson */ > > { PCI_DEVICE(0x1022, 0x780d), > > .driver_data = AZX_DRIVER_GENERIC | > AZX_DCAPS_PRESET_ATI_SB > > }, > > + { PCI_DEVICE(0x1022, 0x15e3), > > + .driver_data = AZX_DRIVER_GENERIC | > AZX_DCAPS_PRESET_ATI_SB > > + }, > > /* ATI HDMI */ > > { PCI_DEVICE(0x1002, 0x0002), > > .driver_data = AZX_DRIVER_ATIHDMI_NS | > > AZX_DCAPS_PRESET_ATI_HDMI_NS }, > > ------Please consider the environment before printing this e-mail. >

2 1

[alsa-devel] [PATCH] ALSA: hda: Add Raven PCI ID
by Vijendar Mukunda 23 Nov '17

23 Nov '17

This commit adds PCI ID for Raven platform Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 01eb1dc..9c7d479 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2454,6 +2454,8 @@ static const struct pci_device_id azx_ids[] = { /* AMD Hudson */ { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + { PCI_DEVICE(0x1022, 0x15e3), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x0002), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, -- 2.7.4

2 1

[alsa-devel] [PATCH 0/7] mark expected switch fall-throughs
by Gustavo A. R. Silva 23 Nov '17

23 Nov '17

In preparation to enabling -Wimplicit-fallthrough, this patchset aims to mark switch cases where we are expecting to fall through. Thanks! Gustavo A. R. Silva (7): ASoC: msm8916-wcd-analog: mark expected switch fall-through ASoC: tlv320aic23: mark expected switch fall-through ASoC: tlv320dac31xx: mark expected switch fall-through ASoC: tpa6130a2: mark expected switch fall-through ASoC: wm8753: mark expected switch fall-throughs ASoC: wm8993: mark expected switch fall-throughs ASoC: wm8994: mark expected switch fall-throughs sound/soc/codecs/msm8916-wcd-analog.c | 1 + sound/soc/codecs/tlv320aic23.c | 1 + sound/soc/codecs/tlv320aic31xx.c | 2 +- sound/soc/codecs/tpa6130a2.c | 1 + sound/soc/codecs/wm8753.c | 4 ++-- sound/soc/codecs/wm8993.c | 2 ++ sound/soc/codecs/wm8994.c | 2 ++ 7 files changed, 10 insertions(+), 3 deletions(-) -- 2.7.4

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[alsa-devel] [GIT PULL]: Geminilake audio firmware update
by Vinod Koul 23 Nov '17

23 Nov '17

Hi, Please PULL to receive update for GLK audio firmware. The following changes since commit bf04291309d3169c0ad3b8db52564235bbd08e30: WHENCE: Add new qed firmware (2017-10-09 18:03:26 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git glk for you to fetch changes up to b964279f2d9955a6ce00d3a672b273ae4d7a362f: linux-firmware: intel: Update Geminilake audio firmware (2017-11-07 10:25:50 +0530) ---------------------------------------------------------------- Pradeep Tewani (1): linux-firmware: intel: Update Geminilake audio firmware WHENCE | 4 +++- intel/dsp_fw_glk.bin | 2 +- intel/dsp_fw_glk_v2880.bin | Bin 0 -> 505608 bytes 3 files changed, 4 insertions(+), 2 deletions(-) create mode 100644 intel/dsp_fw_glk_v2880.bin Thanks -- ~Vinod

2 1

[alsa-devel] [v5 RESEND 00/17] Add support for Cirrus Logic CS47L35/L85/L90/L91 codecs
by Richard Fitzgerald 23 Nov '17

23 Nov '17

Patches still waiting for review/ack: #8 regulator #9 irqchip #14-17 ASoC Summary of changes since V4: - Fixed review comments on MFD code - Fixed dependency error in core codec driver - minor bugfix to core codec driver Richard Fitzgerald (17): mfd: madera: Add register definitions for Cirrus Logic Madera codecs mfd: madera: Add DT bindings for Cirrus Logic Madera codecs mfd: madera: Add common support for Cirrus Logic Madera codecs mfd: madera: Register map tables for Cirrus Logic CS47L35 mfd: madera: Register map tables for Cirrus Logic CS47L85 mfd: madera: Register map tables for Cirrus Logic CS47L90/91 regulator: arizona-micsupp: Add support for Cirrus Logic Madera codecs regulator: arizona-ldo1: Add support for Cirrus Logic Madera codecs irqchip: Add driver for Cirrus Logic Madera codecs pinctrl: madera: Add DT bindings for Cirrus Logic Madera codecs pinctrl: madera: Add driver for Cirrus Logic Madera codecs gpio: madera: Support Cirrus Logic Madera class codecs ASoC: madera: Add DT bindings for Cirrus Logic Madera codecs ASoC: madera: Add common support for Cirrus Logic Madera codecs ASoC: cs47l35: Add codec driver for Cirrus Logic CS47L35 ASoC: cs47l85: Add codec driver for Cirrus Logic CS47L85 ASoC: cs47l90: Add codec driver for Cirrus Logic CS47L90 Documentation/devicetree/bindings/mfd/madera.txt | 102 + .../bindings/pinctrl/cirrus,madera-pinctrl.txt | 99 + .../bindings/regulator/arizona-regulator.txt | 3 +- Documentation/devicetree/bindings/sound/madera.txt | 67 + MAINTAINERS | 23 + drivers/gpio/Kconfig | 6 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-madera.c | 196 + drivers/irqchip/Kconfig | 3 + drivers/irqchip/Makefile | 1 + drivers/irqchip/irq-madera.c | 301 ++ drivers/mfd/Kconfig | 48 + drivers/mfd/Makefile | 13 + drivers/mfd/cs47l35-tables.c | 1610 +++++++ drivers/mfd/cs47l85-tables.c | 3010 +++++++++++++ drivers/mfd/cs47l90-tables.c | 2675 ++++++++++++ drivers/mfd/madera-core.c | 599 +++ drivers/mfd/madera-i2c.c | 139 + drivers/mfd/madera-spi.c | 138 + drivers/mfd/madera.h | 44 + drivers/pinctrl/Kconfig | 1 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/cirrus/Kconfig | 15 + drivers/pinctrl/cirrus/Makefile | 11 + drivers/pinctrl/cirrus/pinctrl-cs47l35.c | 46 + drivers/pinctrl/cirrus/pinctrl-cs47l85.c | 60 + drivers/pinctrl/cirrus/pinctrl-cs47l90.c | 58 + drivers/pinctrl/cirrus/pinctrl-madera.c | 1074 +++++ drivers/pinctrl/cirrus/pinctrl-madera.h | 40 + drivers/regulator/Kconfig | 15 +- drivers/regulator/arizona-ldo1.c | 82 +- drivers/regulator/arizona-micsupp.c | 71 +- include/dt-bindings/sound/madera.h | 27 + include/linux/irqchip/irq-madera-pdata.h | 23 + include/linux/irqchip/irq-madera.h | 98 + include/linux/mfd/madera/core.h | 196 + include/linux/mfd/madera/pdata.h | 61 + include/linux/mfd/madera/registers.h | 3987 ++++++++++++++++++ include/sound/madera-pdata.h | 61 + sound/soc/codecs/Kconfig | 23 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/cs47l35.c | 1748 ++++++++ sound/soc/codecs/cs47l85.c | 2722 ++++++++++++ sound/soc/codecs/cs47l90.c | 2661 ++++++++++++ sound/soc/codecs/madera.c | 4439 ++++++++++++++++++++ sound/soc/codecs/madera.h | 470 +++ 46 files changed, 27066 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/madera.txt create mode 100644 Documentation/devicetree/bindings/pinctrl/cirrus,madera-pinctrl.txt create mode 100644 Documentation/devicetree/bindings/sound/madera.txt create mode 100644 drivers/gpio/gpio-madera.c create mode 100644 drivers/irqchip/irq-madera.c create mode 100644 drivers/mfd/cs47l35-tables.c create mode 100644 drivers/mfd/cs47l85-tables.c create mode 100644 drivers/mfd/cs47l90-tables.c create mode 100644 drivers/mfd/madera-core.c create mode 100644 drivers/mfd/madera-i2c.c create mode 100644 drivers/mfd/madera-spi.c create mode 100644 drivers/mfd/madera.h create mode 100644 drivers/pinctrl/cirrus/Kconfig create mode 100644 drivers/pinctrl/cirrus/Makefile create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l35.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l85.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l90.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-madera.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-madera.h create mode 100644 include/dt-bindings/sound/madera.h create mode 100644 include/linux/irqchip/irq-madera-pdata.h create mode 100644 include/linux/irqchip/irq-madera.h create mode 100644 include/linux/mfd/madera/core.h create mode 100644 include/linux/mfd/madera/pdata.h create mode 100644 include/linux/mfd/madera/registers.h create mode 100644 include/sound/madera-pdata.h create mode 100644 sound/soc/codecs/cs47l35.c create mode 100644 sound/soc/codecs/cs47l85.c create mode 100644 sound/soc/codecs/cs47l90.c create mode 100644 sound/soc/codecs/madera.c create mode 100644 sound/soc/codecs/madera.h -- 2.11.0

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[alsa-devel] [PATCH 2/2] ASoC: fsl_ssi: add 20-bit sample format for AC'97 and use it for capture
by Maciej S. Szmigiero 22 Nov '17

22 Nov '17

When testing AC'97 capture on UDOO board (currently the only user of fsl_ssi driver in the AC'97 mode) it become obvious that there is a massive distortion above certain, small input signal. This problem has been traced to silicon errata ERR003778: "In AC97, 16-bit mode, received data is shifted by 4-bit locations" that has "No fix scheduled". This errata suggests a workaround of doing a 4-bit shift back in SDMA script for this specific operation mode, however our SDMA scripts are shared between various SoC peripherals so we can't really modify them. There is a simple way to avoid this problem, however, that is to disallow recording in 16-bit mode and only support it in AC'97-native 20-bit mode. We have to use a 4-byte format for this since SSI FIFOs do not allow 3-byte accesses (and these aren't supported by imx-sdma driver anyway). With this change the capture distortion is gone. We can also add this format as an additional one supported for playback, using this opportunity to make sure that we use CPU-endian-native formats in AC'97 mode as we already do in I2S mode. There is no problem in using different bit widths in playback and capture in AC'97 mode so allow this, too. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 20ef09e1a395..2aed089fdd76 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1278,14 +1278,15 @@ static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S20_4, }, .capture = { .stream_name = "AC97 Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + /* 16-bit capture is broken (errata ERR003778) */ + .formats = SNDRV_PCM_FMTBIT_S20_4, }, .ops = &fsl_ssi_dai_ops, }; @@ -1557,11 +1558,12 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - if (!fsl_ssi_is_ac97(ssi_private)) + if (!fsl_ssi_is_ac97(ssi_private)) { ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_samplebits = 1; + } ssi_private->cpu_dai_drv.symmetric_channels = 1; - ssi_private->cpu_dai_drv.symmetric_samplebits = 1; } /* Determine the FIFO depth. */

1 0

[alsa-devel] Regression with a91d66129fb9 ("ALSA: hda - Fix incorrect TLV callback check introduced during set_fs() removal")
by Laura Abbott 22 Nov '17

22 Nov '17

Hi, Fedora got a bug report (https://bugzilla.redhat.com/show_bug.cgi?id=1512853) that Line Out stopped working between 4.13.9 and 4.13.10. Reverting 82d745a55779 ("ALSA: hda - Fix incorrect TLV callback check introduced during set_fs() removal") fixed the problem. I didn't ask the reporter to test on 4.14 since I didn't see anything explicitly tagged as fixing the issue. Any ideas? Thanks, Laura

2 3

[alsa-devel] [PATCH] Emit an error message when dlopen fails.
by Cedric Roux 22 Nov '17

22 Nov '17

Signed-off-by: Cedric Roux <sed(a)free.fr> --- src/dlmisc.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git src/dlmisc.c src/dlmisc.c index f154ebd0..9e8e401e 100644 --- src/dlmisc.c +++ src/dlmisc.c @@ -81,10 +81,14 @@ void *snd_dlopen(const char *name, int mode) strcat(filename, "/"); strcat(filename, name); handle = dlopen(filename, mode); + if (!handle) + SNDERR("snd_dlopen: %s: %s", filename, dlerror()); free(filename); } if (!handle) handle = dlopen(name, mode); + if (!handle) + SNDERR("snd_dlopen: %s: %s", name, dlerror()); return handle; #else return NULL; -- 2.15.0

4 6

[alsa-devel] [PATCH] ASoC: rt5645: reset RT5645_AD_DA_MIXER at probe
by Bard Liao 22 Nov '17

22 Nov '17

RT5645_AD_DA_MIXER (0x29) register will not be reset to default after SW reset. So we have to write it to its default value in i2c_probe. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index b7eef6a..27a3d3d 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3815,6 +3815,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, regmap_read(regmap, RT5645_VENDOR_ID, &val); rt5645->v_id = val & 0xff; + regmap_write(rt5645->regmap, RT5645_AD_DA_MIXER, 0x8080); + ret = regmap_register_patch(rt5645->regmap, init_list, ARRAY_SIZE(init_list)); if (ret != 0) -- 2.7.4

1 0

[alsa-devel] [PATCH RFC] ARM: dts: bcm283x: Fix probing of bcm2835-i2s
by Stefan Wahren 21 Nov '17

21 Nov '17

Since 517e7a1537a ("ASoC: bcm2835: move to use the clock framework") the bcm2835-i2s requires a clock as DT property. Unfortunately the necessary DT change has never been applied. While we are at it also fix the first PCM register range to cover the PCM_GRAY register. This patch also fixes the dt-bindings accordlingly. Signed-off-by: Stefan Wahren <stefan.wahren(a)i2se.com> --- Documentation/devicetree/bindings/dma/brcm,bcm2835-dma.txt | 4 ++-- Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt | 9 ++++----- arch/arm/boot/dts/bcm283x.dtsi | 4 ++-- 3 files changed, 8 insertions(+), 9 deletions(-) Hi, this is only probe tested with a RPi 1B. I don't have the sound hardware to test the drivers function. diff --git a/Documentation/devicetree/bindings/dma/brcm,bcm2835-dma.txt b/Documentation/devicetree/bindings/dma/brcm,bcm2835-dma.txt index baf9b34..b6a8cc0 100644 --- a/Documentation/devicetree/bindings/dma/brcm,bcm2835-dma.txt +++ b/Documentation/devicetree/bindings/dma/brcm,bcm2835-dma.txt @@ -74,8 +74,8 @@ Example: bcm2835_i2s: i2s@7e203000 { compatible = "brcm,bcm2835-i2s"; - reg = < 0x7e203000 0x20>, - < 0x7e101098 0x02>; + reg = < 0x7e203000 0x24>; + clocks = <&clocks BCM2835_CLOCK_PCM>; dmas = <&dma 2>, <&dma 3>; diff --git a/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt b/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt index 65783de..7bb0362 100644 --- a/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt +++ b/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt @@ -2,9 +2,8 @@ Required properties: - compatible: "brcm,bcm2835-i2s" -- reg: A list of base address and size entries: - * The first entry should cover the PCM registers - * The second entry should cover the PCM clock registers +- reg: Should contain PCM registers location and length. +- clocks: the (PCM) clock to use - dmas: List of DMA controller phandle and DMA request line ordered pairs. - dma-names: Identifier string for each DMA request line in the dmas property. These strings correspond 1:1 with the ordered pairs in dmas. @@ -16,8 +15,8 @@ Example: bcm2835_i2s: i2s@7e203000 { compatible = "brcm,bcm2835-i2s"; - reg = <0x7e203000 0x20>, - <0x7e101098 0x02>; + reg = <0x7e203000 0x24>; + clocks = <&clocks BCM2835_CLOCK_PCM>; dmas = <&dma 2>, <&dma 3>; diff --git a/arch/arm/boot/dts/bcm283x.dtsi b/arch/arm/boot/dts/bcm283x.dtsi index 431dcfc..01733a9 100644 --- a/arch/arm/boot/dts/bcm283x.dtsi +++ b/arch/arm/boot/dts/bcm283x.dtsi @@ -391,8 +391,8 @@ i2s: i2s@7e203000 { compatible = "brcm,bcm2835-i2s"; - reg = <0x7e203000 0x20>, - <0x7e101098 0x02>; + reg = <0x7e203000 0x24>; + clocks = <&clocks BCM2835_CLOCK_PCM>; dmas = <&dma 2>, <&dma 3>; -- 2.7.4

3 6

[alsa-devel] [PATCH 0/4] Harden USB-audio parsers
by Takashi Iwai 21 Nov '17

21 Nov '17

Hi, as recently spotted by syzkaller, the parser codes in USB audio driver don't check the length of the descriptor unit before actually accessing the extra field, which may lead to out-of-bound access error. This patchset tries to address these by adding proper sanity checks. Takashi === Takashi Iwai (4): ALSA: usb-audio: Add sanity checks to FE parser ALSA: usb-audio: Fix potential out-of-bound access at parsing SU ALSA: usb-audio: Fix potential zero-division at parsing FU ALSA: usb-audio: Add sanity checks in v2 clock parsers sound/usb/clock.c | 9 ++++++--- sound/usb/mixer.c | 19 ++++++++++++++++--- 2 files changed, 22 insertions(+), 6 deletions(-) -- 2.15.0

1 4

[alsa-devel] usb/sound: use-after-free in __uac_clock_find_source
by Andrey Konovalov 21 Nov '17

21 Nov '17

Hi! I've got the following report while fuzzing the kernel with syzkaller. On commit e1d1ea549b57790a3d8cf6300e6ef86118d692a3 (4.15-rc1). This actually looks more like an out-of-bounds with large offset than a use-after-free due to unrelated alloc and free stack traces. ================================================================== BUG: KASAN: use-after-free in __uac_clock_find_source+0xddd/0xe40 Read of size 1 at addr ffff8800699ca00f by task kworker/1:0/18 CPU: 1 PID: 18 Comm: kworker/1:0 Not tainted 4.14.0-57501-g9284d204d604 #119 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:17 dump_stack+0xe1/0x157 lib/dump_stack.c:53 print_address_description+0x71/0x234 mm/kasan/report.c:252 kasan_report_error mm/kasan/report.c:351 kasan_report+0x173/0x270 mm/kasan/report.c:409 __asan_report_load1_noabort+0x19/0x20 mm/kasan/report.c:427 __uac_clock_find_source+0xddd/0xe40 sound/usb/clock.c:221 snd_usb_clock_find_source+0x41/0x50 sound/usb/clock.c:276 set_sample_rate_v2 sound/usb/clock.c:367 snd_usb_init_sample_rate+0x3b6/0xa40 sound/usb/clock.c:430 create_fixed_stream_quirk+0x510/0x8f0 sound/usb/quirks.c:192 snd_usb_create_quirk+0xa6/0x120 sound/usb/quirks.c:560 create_composite_quirk+0x1e0/0x420 sound/usb/quirks.c:59 snd_usb_create_quirk+0xa6/0x120 sound/usb/quirks.c:560 usb_audio_probe+0x1220/0x1f70 sound/usb/card.c:618 usb_probe_interface+0x324/0x940 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:424 driver_probe_device+0x564/0x820 drivers/base/dd.c:566 __device_attach_driver+0x25d/0x2d0 drivers/base/dd.c:662 bus_for_each_drv+0xff/0x160 drivers/base/bus.c:463 __device_attach+0x1ab/0x2a0 drivers/base/dd.c:719 device_initial_probe+0x1f/0x30 drivers/base/dd.c:766 bus_probe_device+0x1fc/0x2a0 drivers/base/bus.c:523 device_add+0xc27/0x15a0 drivers/base/core.c:1835 usb_set_configuration+0xd55/0x17a0 drivers/usb/core/message.c:1967 generic_probe+0xbb/0x120 drivers/usb/core/generic.c:174 usb_probe_device+0xab/0x100 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:424 driver_probe_device+0x564/0x820 drivers/base/dd.c:566 __device_attach_driver+0x25d/0x2d0 drivers/base/dd.c:662 bus_for_each_drv+0xff/0x160 drivers/base/bus.c:463 __device_attach+0x1ab/0x2a0 drivers/base/dd.c:719 device_initial_probe+0x1f/0x30 drivers/base/dd.c:766 bus_probe_device+0x1fc/0x2a0 drivers/base/bus.c:523 device_add+0xc27/0x15a0 drivers/base/core.c:1835 usb_new_device+0x7fa/0x1090 drivers/usb/core/hub.c:2538 hub_port_connect drivers/usb/core/hub.c:5000 hub_port_connect_change drivers/usb/core/hub.c:5106 port_event drivers/usb/core/hub.c:5212 hub_event_impl+0x17bc/0x3440 drivers/usb/core/hub.c:5324 hub_event+0x38/0x50 drivers/usb/core/hub.c:5222 process_one_work+0x944/0x15f0 kernel/workqueue.c:2112 worker_thread+0xef/0x10d0 kernel/workqueue.c:2246 kthread+0x367/0x420 kernel/kthread.c:238 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:437 Allocated by task 5253: save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 kasan_kmalloc+0xc4/0xe0 mm/kasan/kasan.c:551 kasan_slab_alloc+0x12/0x20 mm/kasan/kasan.c:489 slab_post_alloc_hook mm/slab.h:442 slab_alloc_node mm/slub.c:2725 slab_alloc mm/slub.c:2733 kmem_cache_alloc+0xd3/0x270 mm/slub.c:2738 kmem_cache_zalloc ./include/linux/slab.h:678 get_empty_filp+0xac/0x350 fs/file_table.c:123 path_openat+0x36/0x2860 fs/namei.c:3505 do_filp_open+0x13f/0x1d0 fs/namei.c:3563 do_sys_open+0x362/0x4c0 fs/open.c:1059 SYSC_open fs/open.c:1077 SyS_open+0x32/0x40 fs/open.c:1072 entry_SYSCALL_64_fastpath+0x23/0x9a arch/x86/entry/entry_64.S:203 Freed by task 17: save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 kasan_slab_free+0x71/0xc0 mm/kasan/kasan.c:524 slab_free_hook mm/slub.c:1391 slab_free_freelist_hook mm/slub.c:1412 slab_free mm/slub.c:2968 kmem_cache_free+0xb7/0x2f0 mm/slub.c:2990 file_free_rcu+0x67/0x90 fs/file_table.c:50 __rcu_reclaim kernel/rcu/rcu.h:195 rcu_do_batch kernel/rcu/tree.c:2758 invoke_rcu_callbacks kernel/rcu/tree.c:3012 __rcu_process_callbacks kernel/rcu/tree.c:2979 rcu_process_callbacks+0x4df/0xcf0 kernel/rcu/tree.c:2996 __do_softirq+0x2e0/0x88e kernel/softirq.c:285 The buggy address belongs to the object at ffff8800699ca000 which belongs to the cache filp of size 488 The buggy address is located 15 bytes inside of 488-byte region [ffff8800699ca000, ffff8800699ca1e8) The buggy address belongs to the page: page:ffffea0001a67280 count:1 mapcount:0 mapping: (null) index:0x0 compound_mapcount: 0 flags: 0x100000000008100(slab|head) raw: 0100000000008100 0000000000000000 0000000000000000 00000001000c000c raw: dead000000000100 dead000000000200 ffff88006ca97980 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8800699c9f00: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ffff8800699c9f80: 00 00 05 fc fc fc fc fc fc fc fc fc fc fc fc fc >ffff8800699ca000: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff8800699ca080: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8800699ca100: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ==================================================================

2 1

[alsa-devel] usb/sound: slab-out-of-bounds in parse_audio_unit
by Andrey Konovalov 21 Nov '17

21 Nov '17

Hi! I've got the following report while fuzzing the kernel with syzkaller. On commit e1d1ea549b57790a3d8cf6300e6ef86118d692a3 (4.15-rc1). ================================================================== BUG: KASAN: slab-out-of-bounds in parse_audio_unit+0x2e08/0x31d0 Read of size 1 at addr ffff88006998ff20 by task kworker/1:1/1495 CPU: 1 PID: 1495 Comm: kworker/1:1 Not tainted 4.14.0-57501-g9284d204d604 #119 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:17 dump_stack+0xe1/0x157 lib/dump_stack.c:53 print_address_description+0x71/0x234 mm/kasan/report.c:252 kasan_report_error mm/kasan/report.c:351 kasan_report+0x173/0x270 mm/kasan/report.c:409 __asan_report_load1_noabort+0x19/0x20 mm/kasan/report.c:427 parse_audio_feature_unit sound/usb/mixer.c:1472 parse_audio_unit+0x2e08/0x31d0 sound/usb/mixer.c:2213 snd_usb_mixer_controls+0x2ec/0x750 sound/usb/mixer.c:2320 snd_usb_create_mixer+0x209/0xbe0 sound/usb/mixer.c:2566 create_standard_mixer_quirk+0x61/0x80 sound/usb/quirks.c:516 snd_usb_create_quirk+0xa6/0x120 sound/usb/quirks.c:560 create_composite_quirk+0x1e0/0x420 sound/usb/quirks.c:59 snd_usb_create_quirk+0xa6/0x120 sound/usb/quirks.c:560 usb_audio_probe+0x1220/0x1f70 sound/usb/card.c:618 usb_probe_interface+0x324/0x940 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:424 driver_probe_device+0x564/0x820 drivers/base/dd.c:566 __device_attach_driver+0x25d/0x2d0 drivers/base/dd.c:662 bus_for_each_drv+0xff/0x160 drivers/base/bus.c:463 __device_attach+0x1ab/0x2a0 drivers/base/dd.c:719 device_initial_probe+0x1f/0x30 drivers/base/dd.c:766 bus_probe_device+0x1fc/0x2a0 drivers/base/bus.c:523 device_add+0xc27/0x15a0 drivers/base/core.c:1835 usb_set_configuration+0xd55/0x17a0 drivers/usb/core/message.c:1967 generic_probe+0xbb/0x120 drivers/usb/core/generic.c:174 usb_probe_device+0xab/0x100 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:424 driver_probe_device+0x564/0x820 drivers/base/dd.c:566 __device_attach_driver+0x25d/0x2d0 drivers/base/dd.c:662 bus_for_each_drv+0xff/0x160 drivers/base/bus.c:463 __device_attach+0x1ab/0x2a0 drivers/base/dd.c:719 device_initial_probe+0x1f/0x30 drivers/base/dd.c:766 bus_probe_device+0x1fc/0x2a0 drivers/base/bus.c:523 device_add+0xc27/0x15a0 drivers/base/core.c:1835 usb_new_device+0x7fa/0x1090 drivers/usb/core/hub.c:2538 hub_port_connect drivers/usb/core/hub.c:5000 hub_port_connect_change drivers/usb/core/hub.c:5106 port_event drivers/usb/core/hub.c:5212 hub_event_impl+0x17bc/0x3440 drivers/usb/core/hub.c:5324 hub_event+0x38/0x50 drivers/usb/core/hub.c:5222 process_one_work+0x944/0x15f0 kernel/workqueue.c:2112 worker_thread+0xef/0x10d0 kernel/workqueue.c:2246 kthread+0x367/0x420 kernel/kthread.c:238 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:437 Allocated by task 1495: save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 kasan_kmalloc+0xc4/0xe0 mm/kasan/kasan.c:551 __kmalloc+0x1bc/0x300 mm/slub.c:3762 kmalloc ./include/linux/slab.h:504 usb_get_configuration+0x299/0x4e60 drivers/usb/core/config.c:856 usb_enumerate_device drivers/usb/core/hub.c:2371 usb_new_device+0xab1/0x1090 drivers/usb/core/hub.c:2507 hub_port_connect drivers/usb/core/hub.c:5000 hub_port_connect_change drivers/usb/core/hub.c:5106 port_event drivers/usb/core/hub.c:5212 hub_event_impl+0x17bc/0x3440 drivers/usb/core/hub.c:5324 hub_event+0x38/0x50 drivers/usb/core/hub.c:5222 process_one_work+0x944/0x15f0 kernel/workqueue.c:2112 worker_thread+0xef/0x10d0 kernel/workqueue.c:2246 kthread+0x367/0x420 kernel/kthread.c:238 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:437 Freed by task 5251: save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 kasan_slab_free+0x71/0xc0 mm/kasan/kasan.c:524 slab_free_hook mm/slub.c:1391 slab_free_freelist_hook mm/slub.c:1412 slab_free mm/slub.c:2968 kfree+0xf2/0x2e0 mm/slub.c:3899 kzfree+0x2d/0x40 mm/slab_common.c:1482 aa_free_task_context+0x13a/0x170 security/apparmor/context.c:54 apparmor_cred_free+0x38/0x70 security/apparmor/lsm.c:58 security_cred_free+0x4d/0x90 security/security.c:995 put_cred_rcu+0x72/0x2a0 kernel/cred.c:117 __rcu_reclaim kernel/rcu/rcu.h:195 rcu_do_batch kernel/rcu/tree.c:2758 invoke_rcu_callbacks kernel/rcu/tree.c:3012 __rcu_process_callbacks kernel/rcu/tree.c:2979 rcu_process_callbacks+0x4df/0xcf0 kernel/rcu/tree.c:2996 __do_softirq+0x2e0/0x88e kernel/softirq.c:285 The buggy address belongs to the object at ffff88006998ff00 which belongs to the cache kmalloc-32 of size 32 The buggy address is located 0 bytes to the right of 32-byte region [ffff88006998ff00, ffff88006998ff20) The buggy address belongs to the page: page:ffffea0001a663c0 count:1 mapcount:0 mapping: (null) index:0x0 flags: 0x100000000000100(slab) raw: 0100000000000100 0000000000000000 0000000000000000 0000000180550055 raw: ffffea0001aea900 0000000200000002 ffff88006c803980 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff88006998fe00: fc fc fb fb fb fb fc fc 00 00 00 fc fc fc fb fb ffff88006998fe80: fb fb fc fc fb fb fb fb fc fc fb fb fb fb fc fc >ffff88006998ff00: 00 00 00 00 fc fc fb fb fb fb fc fc fb fb fb fb ^ ffff88006998ff80: fc fc 00 00 00 00 fc fc fb fb fb fb fc fc fc fc ffff880069990000: 00 00 00 00 00 00 00 00 00 00 00 00 fc fc fc fc ==================================================================

2 1

Re: [alsa-devel] WARNING in snd_timer_user_info_compat
by Takashi Iwai 21 Nov '17

21 Nov '17

On Tue, 21 Nov 2017 16:27:58 +0100, syzbot wrote: > > Hello, > > syzkaller hit the following crash on > c8a0739b185d11d6e2ca7ad9f5835841d1cfc765 > git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/master > compiler: gcc (GCC) 7.1.1 20170620 > .config is attached > Raw console output is attached. > C reproducer is attached > syzkaller reproducer is attached. See https://goo.gl/kgGztJ > for information about syzkaller reproducers > > > Kernel panic - not syncing: panic_on_warn set ... > > CPU: 1 PID: 3039 Comm: syzkaller881964 Not tainted 4.14.0+ #98 > Hardware name: Google Google Compute Engine/Google Compute Engine, > BIOS Google 01/01/2011 > Call Trace: > __dump_stack lib/dump_stack.c:17 [inline] > dump_stack+0x194/0x257 lib/dump_stack.c:53 > panic+0x1e4/0x41c kernel/panic.c:183 > __warn+0x1dc/0x200 kernel/panic.c:547 > report_bug+0x211/0x2d0 lib/bug.c:184 > fixup_bug.part.11+0x37/0x80 arch/x86/kernel/traps.c:177 > fixup_bug arch/x86/kernel/traps.c:246 [inline] > do_error_trap+0x2d7/0x3e0 arch/x86/kernel/traps.c:295 > do_invalid_op+0x1b/0x20 arch/x86/kernel/traps.c:314 > invalid_op+0x18/0x20 arch/x86/entry/entry_64.S:926 > RIP: 0010:snd_timer_user_info_compat.isra.6+0x3be/0x490 > sound/core/timer_compat.c:69 Aha, it's a superfluous WARN_ON() usage from snd_BUG_ON(). Now syzkaller turned on CONFIG_SND_DEBUG=y, so this surfaced. The fix patch is below. I'm going to queue this. #syz fix: ALSA: timer: Remove kernel warning at compat ioctl error paths thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: timer: Remove kernel warning at compat ioctl error paths Some timer compat ioctls have NULL checks of timer instance with snd_BUG_ON() that bring up WARN_ON() when the debug option is set. Actually the condition can be met in the normal situation and it's confusing and bad to spew kernel warnings with stack trace there. Let's remove snd_BUG_ON() invocation and replace with the simple checks. Also, correct the error code to EBADFD to follow the native ioctl error handling. Reported-by: syzbot <syzkaller(a)googlegroups.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer_compat.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c index 59127b6ef39e..e00f7e399e46 100644 --- a/sound/core/timer_compat.c +++ b/sound/core/timer_compat.c @@ -66,11 +66,11 @@ static int snd_timer_user_info_compat(struct file *file, struct snd_timer *t; tu = file->private_data; - if (snd_BUG_ON(!tu->timeri)) - return -ENXIO; + if (!tu->timeri) + return -EBADFD; t = tu->timeri->timer; - if (snd_BUG_ON(!t)) - return -ENXIO; + if (!t) + return -EBADFD; memset(&info, 0, sizeof(info)); info.card = t->card ? t->card->number : -1; if (t->hw.flags & SNDRV_TIMER_HW_SLAVE) @@ -99,8 +99,8 @@ static int snd_timer_user_status_compat(struct file *file, struct snd_timer_status32 status; tu = file->private_data; - if (snd_BUG_ON(!tu->timeri)) - return -ENXIO; + if (!tu->timeri) + return -EBADFD; memset(&status, 0, sizeof(status)); status.tstamp.tv_sec = tu->tstamp.tv_sec; status.tstamp.tv_nsec = tu->tstamp.tv_nsec; -- 2.15.0

1 0

Re: [alsa-devel] KASAN: use-after-free Read in snd_timer_user_info_compat
by Takashi Iwai 21 Nov '17

21 Nov '17

On Fri, 27 Oct 2017 10:34:01 +0200, syzbot wrote: > > Hello, > > syzkaller hit the following crash on > 33d930e59a98fa10a0db9f56c7fa2f21a4aef9b9 > git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/master > compiler: gcc (GCC) 7.1.1 20170620 > .config is attached > Raw console output is attached. > > syzkaller reproducer is attached. See https://goo.gl/kgGztJ > for information about syzkaller reproducers > > > BUG: KASAN: use-after-free in > snd_timer_user_info_compat.isra.6+0x46c/0x480 > sound/core/timer_compat.c:71 > Read of size 8 at addr ffff8801d0dd9d00 by task syz-executor0/3342 OK, this is a real bug, and obviously a forgotten code path from the previous fixes. The patch below should address it, but I checked only compile-tested, so far. Let me know if it really works. thanks, Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: timer: Add missing mutex lock for compat ioctls The races among ioctl and other operations were protected by the commit af368027a49a ("ALSA: timer: Fix race among timer ioctls") and later fixes, but one code path was forgotten in the scenario: the 32bit compat ioctl. As syzkaller recently spotted, a very similar use-after-free may happen with the combination of compat ioctls. The fix is simply to apply the same ioctl_lock to the compat_ioctl callback, too. Fixes: af368027a49a ("ALSA: timer: Fix race among timer ioctls") Reference: http://lkml.kernel.org/r/089e082686ac9b482e055c832617@google.com Reported-by: syzbot <bot+e5f3c9783e7048a74233054febbe9f1bdf54b6da(a)syzkaller.appspotmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer_compat.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c index 6a437eb66115..59127b6ef39e 100644 --- a/sound/core/timer_compat.c +++ b/sound/core/timer_compat.c @@ -133,7 +133,8 @@ enum { #endif /* CONFIG_X86_X32 */ }; -static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg) +static long __snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, + unsigned long arg) { void __user *argp = compat_ptr(arg); @@ -153,7 +154,7 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns case SNDRV_TIMER_IOCTL_PAUSE: case SNDRV_TIMER_IOCTL_PAUSE_OLD: case SNDRV_TIMER_IOCTL_NEXT_DEVICE: - return snd_timer_user_ioctl(file, cmd, (unsigned long)argp); + return __snd_timer_user_ioctl(file, cmd, (unsigned long)argp); case SNDRV_TIMER_IOCTL_GPARAMS32: return snd_timer_user_gparams_compat(file, argp); case SNDRV_TIMER_IOCTL_INFO32: @@ -167,3 +168,15 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns } return -ENOIOCTLCMD; } + +static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, + unsigned long arg) +{ + struct snd_timer_user *tu = file->private_data; + long ret; + + mutex_lock(&tu->ioctl_lock); + ret = __snd_timer_user_ioctl_compat(file, cmd, arg); + mutex_unlock(&tu->ioctl_lock); + return ret; +} -- 2.14.2

2 1

[alsa-devel] [PATCH] SoC: intel: byt: Introduce new custom IN2 map
by Carlo Caione 21 Nov '17

21 Nov '17

From: Carlo Caione <carlo(a)endlessm.com> Introduce a new custom dapm routes map to quirk platforms with the internal mic connected to IN2P. Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- sound/soc/intel/boards/bytcr_rt5651.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 3076bfc0db5e..1dad5c98c9ef 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -37,6 +37,7 @@ enum { BYT_RT5651_DMIC_MAP, BYT_RT5651_IN1_MAP, + BYT_RT5651_IN2_MAP, }; #define BYT_RT5651_MAP(quirk) ((quirk) & GENMASK(7, 0)) @@ -58,6 +59,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk DMIC_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_MAP) dev_info(dev, "quirk IN1_MAP enabled"); + if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_MAP) + dev_info(dev, "quirk IN2_MAP enabled"); if (byt_rt5651_quirk & BYT_RT5651_DMIC_EN) dev_info(dev, "quirk DMIC enabled"); if (byt_rt5651_quirk & BYT_RT5651_MCLK_EN) @@ -143,7 +146,6 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { {"ssp2 Rx", NULL, "AIF1 Capture"}, {"Headset Mic", NULL, "micbias1"}, /* lowercase for rt5651 */ - {"IN2P", NULL, "Headset Mic"}, {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, {"Speaker", NULL, "LOUTL"}, @@ -151,15 +153,23 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { }; static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic_map[] = { + {"IN2P", NULL, "Headset Mic"}, {"DMIC L1", NULL, "Internal Mic"}, {"DMIC R1", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = { {"Internal Mic", NULL, "micbias1"}, + {"IN2P", NULL, "Headset Mic"}, {"IN1P", NULL, "Internal Mic"}, }; +static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_map[] = { + {"Internal Mic", NULL, "micbias1"}, + {"IN1P", NULL, "Headset Mic"}, + {"IN2P", NULL, "Internal Mic"}, +}; + static const struct snd_kcontrol_new byt_rt5651_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -246,6 +256,10 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5651_intmic_in1_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_map); break; + case BYT_RT5651_IN2_MAP: + custom_map = byt_rt5651_intmic_in2_map; + num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_map); + break; default: custom_map = byt_rt5651_intmic_dmic_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic_map); -- 2.14.2

3 5

[alsa-devel] [PATCH v2] ALSA: core: update tstamp only if audio_tstamp changed
by Henrik Eriksson 21 Nov '17

21 Nov '17

commit 3179f6200188 ("ALSA: core: add .get_time_info") had a side effect of changing the behaviour of the PCM runtime tstamp. Prior to this change tstamp was not updated by snd_pcm_update_hw_ptr0() unless the hw_ptr had moved, after this change tstamp was always updated. For an application using alsa-lib, doing snd_pcm_readi() followed by snd_pcm_status() to estimate the age of the read samples by subtracting status->avail * [sample rate] from status->tstamp this change degraded the accuracy of the estimate on devices where the pcm hw does not provide a granular hw_ptr, e.g., devices using soc-generic-dmaengine-pcm.c and a dma-engine with residue_granularity DMA_RESIDUE_GRANULARITY_DESCRIPTOR. The accuracy of the estimate depended on the latency between the PCM hw completing a period and the driver called snd_pcm_period_elapsed() to notify ALSA core, typically determined by interrupt handling latency. After the change the accuracy of the estimate depended on the latency between the PCM hw completing a period and the application calling snd_pcm_status(), determined by the scheduling of the application process. The maximum error of the estimate is one period length in both cases, but the error average and variance is smaller when it depends on interrupt latency. Instead of always updating tstamp, update it only if audio_tstamp changed. Fixes: 3179f6200188 ("ALSA: core: add .get_time_info") Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Henrik Eriksson <henrik.eriksson(a)axis.com> --- Changes since v1: - Use Pierre-Louis Bossart's suggested fix - Clean up commit log sound/core/pcm_lib.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a93a4235a332..10e7ef7a8804 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -248,8 +248,10 @@ static void update_audio_tstamp(struct snd_pcm_substream *substream, runtime->rate); *audio_tstamp = ns_to_timespec(audio_nsecs); } - runtime->status->audio_tstamp = *audio_tstamp; - runtime->status->tstamp = *curr_tstamp; + if (!timespec_equal(&runtime->status->audio_tstamp, audio_tstamp)) { + runtime->status->audio_tstamp = *audio_tstamp; + runtime->status->tstamp = *curr_tstamp; + } /* * re-take a driver timestamp to let apps detect if the reference tstamp -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: call _fsl_ssi_set_dai_fmt() just once in AC'97 mode
by Maciej S. Szmigiero 21 Nov '17

21 Nov '17

In AC'97 mode we configure and start SSI RX / TX on probe path via a call to _fsl_ssi_set_dai_fmt() function. We don't need to call this function again later and in fact don't want to do it since this function temporarily sets STCR, SRCR and SCR to some intermediate values. We need to make sure, however, that only proper channel slots are enabled at playback start time since some AC'97 CODECs (like VT1613) were observed requesting via SLOTREQ (and so enabling at SSI) spurious ones just after an AC'97 link is started but before the CODEC is configured by its driver. Theoretically, this should be necessary only for the very first playback but let's play safe here and make sure that no extra slots are enabled every time a playback is started. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 48bb850a34d9..dad80b4b0cfc 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -630,12 +630,6 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) regmap_write(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV); - /* no SACC{ST,EN,DIS} regs on imx21-class SSI */ - if (!ssi_private->soc->imx21regs) { - regmap_write(regs, CCSR_SSI_SACCDIS, 0xff); - regmap_write(regs, CCSR_SSI_SACCEN, 0x300); - } - /* * Enable SSI, Transmit and Receive. AC97 has to communicate with the * codec before a stream is started. @@ -1076,6 +1070,9 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + if (fsl_ssi_is_ac97(ssi_private)) + return 0; + return _fsl_ssi_set_dai_fmt(cpu_dai->dev, ssi_private, fmt); } @@ -1149,9 +1146,16 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* no SACC{ST,EN,DIS} regs on imx21-class SSI */ + if (fsl_ssi_is_ac97(ssi_private) && + !ssi_private->soc->imx21regs) { + regmap_write(regs, CCSR_SSI_SACCDIS, 0xff); + regmap_write(regs, CCSR_SSI_SACCEN, 0x300); + } + fsl_ssi_tx_config(ssi_private, true); - else + } else fsl_ssi_rx_config(ssi_private, true); break;

2 4

[alsa-devel] [PATCH alsa-tools] hdajacksensetest: add gitcompile to EXTRA_DIST
by Tanu Kaskinen 20 Nov '17

20 Nov '17

The gitcompile script is used from the top-level Makefile, so it needs to be included in the tarball. Signed-off-by: Tanu Kaskinen <tanuk(a)iki.fi> --- hdajacksensetest/Makefile.am | 1 + 1 file changed, 1 insertion(+) diff --git a/hdajacksensetest/Makefile.am b/hdajacksensetest/Makefile.am index 795373c..893b4e1 100644 --- a/hdajacksensetest/Makefile.am +++ b/hdajacksensetest/Makefile.am @@ -1,3 +1,4 @@ +EXTRA_DIST = gitcompile MYNAME = hdajacksensetest AUTOMAKE_OPTIONS = foreign bin_PROGRAMS = hdajacksensetest -- 2.14.2

2 1

[alsa-devel] [PATCH] ALSA: line6: add support for POD HD DESKTOP
by Hans P. Möller Ebner 20 Nov '17

20 Nov '17

Add support for the Line6 POD HD500X multi effect processor for playback and capture (in/out audio) through USB. Signed-off-by: Hans P. Moller <hmoller(a)uc.cl> --- sound/usb/line6/podhd.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index 956f847..ba4f71f 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -39,7 +39,8 @@ enum { LINE6_PODHD500_1, LINE6_PODX3, LINE6_PODX3LIVE, - LINE6_PODHD500X + LINE6_PODHD500X, + LINE6_PODHDDESKTOP }; struct usb_line6_podhd { @@ -377,6 +378,7 @@ static const struct usb_device_id podhd_id_table[] = { { LINE6_IF_NUM(0x414A, 0), .driver_info = LINE6_PODX3 }, { LINE6_IF_NUM(0x414B, 0), .driver_info = LINE6_PODX3LIVE }, { LINE6_IF_NUM(0x4159, 0), .driver_info = LINE6_PODHD500X }, + { LINE6_IF_NUM(0x4156, 0), .driver_info = LINE6_PODHDDESKTOP }, {} }; @@ -463,6 +465,18 @@ static const struct line6_properties podhd_properties_table[] = { .ep_audio_r = 0x86, .ep_audio_w = 0x02, }, + [LINE6_PODHDDESKTOP] = { + .id = "PODHDDESKTOP", + .name = "POD HDDESKTOP", + .capabilities = LINE6_CAP_CONTROL + | LINE6_CAP_PCM | LINE6_CAP_HWMON, + .altsetting = 1, + .ep_ctrl_r = 0x81, + .ep_ctrl_w = 0x01, + .ctrl_if = 1, + .ep_audio_r = 0x86, + .ep_audio_w = 0x02, + }, }; /* -- 2.7.4

3 3

[alsa-devel] [PATCH] ASoC: cht_bsw_rt5645: Use common error handling code in cht_codec_fixup()
by SF Markus Elfring 20 Nov '17

20 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 19 Nov 2017 09:53:56 +0100 * Add a jump target so that a specific error message is stored only once at the end of this function implementation. * Replace two calls of the function "dev_err" by goto statements. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 18d129caa974..bf3ffc2b0d8f 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -357,20 +357,16 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS ); - if (ret < 0) { - dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); - return ret; - } + if (ret < 0) + goto report_failure; ret = snd_soc_dai_set_fmt(rtd->codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS ); - if (ret < 0) { - dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); - return ret; - } + if (ret < 0) + goto report_failure; ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 16); if (ret < 0) { @@ -403,6 +399,10 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, } } return 0; + +report_failure: + dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); + return ret; } static int cht_aif1_startup(struct snd_pcm_substream *substream) -- 2.15.0

3 2

Re: [alsa-devel] BUG: unable to handle kernel paging request in snd_seq_oss_readq_puts
by Dmitry Vyukov 20 Nov '17

20 Nov '17

On Wed, Nov 1, 2017 at 9:38 PM, syzbot <bot+31681772ec7a18dde8d3f8caf475f361a89b9514(a)syzkaller.appspotmail.com> wrote: > Hello, > > syzkaller hit the following crash on > fc2e8b1a47c14b22c33eb087fca0db58e1f4ed0e > git://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git/master > compiler: gcc (GCC) 7.1.1 20170620 > .config is attached > Raw console output is attached. > C reproducer is attached > syzkaller reproducer is attached. See https://goo.gl/kgGztJ > for information about syzkaller reproducers This also happened on more recent commits, including upstream 9c323bff13f92832e03657cabdd70d731408d621 (Oct 20): device gre0 entered promiscuous mode BUG: unable to handle kernel paging request at ffffc90001688000 IP: snd_seq_oss_readq_puts+0x146/0x210 sound/core/seq/oss/seq_oss_readq.c:112 PGD 1dac75067 P4D 1dac75067 PUD 1dac76067 PMD 1c1dd3067 PTE 0 Oops: 0000 [#1] SMP KASAN Dumping ftrace buffer: (ftrace buffer empty) Modules linked in: CPU: 1 PID: 18947 Comm: syz-executor2 Not tainted 4.14.0-rc5+ #51 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 task: ffff8801d92c8140 task.stack: ffff8801c2ef0000 RIP: 0010:snd_seq_oss_readq_puts+0x146/0x210 sound/core/seq/oss/seq_oss_readq.c:112 RSP: 0018:ffff8801c2ef6db0 EFLAGS: 00010246 RAX: ffffed00385dedbf RBX: ffffc90001688000 RCX: ffff8801c2ef6df9 RDX: 000000000000554e RSI: ffffffff83d4c556 RDI: 0000000000000282 RBP: ffff8801c2ef6e60 R08: 0000000000000001 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 00000000400000a0 R13: ffff8801c2ef6e38 R14: ffffc90001688001 R15: dffffc0000000000 FS: 0000000000000000(0000) GS:ffff8801db300000(0063) knlGS:00000000f76ffb40 CS: 0010 DS: 002b ES: 002b CR0: 0000000080050033 CR2: ffffc90001688000 CR3: 00000001d0650000 CR4: 00000000001406e0 DR0: 0000000020000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000600 Call Trace: send_midi_event sound/core/seq/oss/seq_oss_midi.c:616 [inline] snd_seq_oss_midi_input+0xd39/0x1040 sound/core/seq/oss/seq_oss_midi.c:535 snd_seq_oss_event_input+0x15d/0x220 sound/core/seq/oss/seq_oss_event.c:439 snd_seq_deliver_single_event.constprop.11+0x2fb/0x940 sound/core/seq/seq_clientmgr.c:621 deliver_to_subscribers sound/core/seq/seq_clientmgr.c:676 [inline] snd_seq_deliver_event+0x318/0x790 sound/core/seq/seq_clientmgr.c:807 snd_seq_kernel_client_dispatch+0x11e/0x150 sound/core/seq/seq_clientmgr.c:2313 dummy_input+0x2c4/0x400 sound/core/seq/seq_dummy.c:104 snd_seq_deliver_single_event.constprop.11+0x2fb/0x940 sound/core/seq/seq_clientmgr.c:621 snd_seq_deliver_event+0x12c/0x790 sound/core/seq/seq_clientmgr.c:818 snd_seq_dispatch_event+0x105/0x5b0 sound/core/seq/seq_clientmgr.c:892 snd_seq_check_queue.part.3+0x38e/0x510 sound/core/seq/seq_queue.c:285 snd_seq_check_queue sound/core/seq/seq_queue.c:357 [inline] snd_seq_enqueue_event+0x32d/0x3d0 sound/core/seq/seq_queue.c:363 snd_seq_client_enqueue_event+0x21b/0x420 sound/core/seq/seq_clientmgr.c:951 kernel_client_enqueue.part.10+0xb5/0xd0 sound/core/seq/seq_clientmgr.c:2252 kernel_client_enqueue sound/core/seq/seq_clientmgr.c:2242 [inline] snd_seq_kernel_client_enqueue_blocking+0xcf/0x110 sound/core/seq/seq_clientmgr.c:2279 insert_queue sound/core/seq/oss/seq_oss_rw.c:189 [inline] snd_seq_oss_write+0x5fe/0xa80 sound/core/seq/oss/seq_oss_rw.c:148 odev_write+0x64/0x90 sound/core/seq/oss/seq_oss.c:177 __vfs_write+0xef/0x970 fs/read_write.c:479 __kernel_write+0xf8/0x350 fs/read_write.c:500 write_pipe_buf+0x175/0x220 fs/splice.c:797 splice_from_pipe_feed fs/splice.c:502 [inline] __splice_from_pipe+0x328/0x730 fs/splice.c:626 splice_from_pipe+0x1e9/0x330 fs/splice.c:661 default_file_splice_write+0x40/0x90 fs/splice.c:809 do_splice_from fs/splice.c:851 [inline] do_splice fs/splice.c:1147 [inline] SYSC_splice fs/splice.c:1402 [inline] SyS_splice+0x7b7/0x1610 fs/splice.c:1382 do_syscall_32_irqs_on arch/x86/entry/common.c:329 [inline] do_fast_syscall_32+0x3f2/0xf05 arch/x86/entry/common.c:391 entry_SYSENTER_compat+0x51/0x60 arch/x86/entry/entry_64_compat.S:124 RIP: 0023:0xf7f03c79 RSP: 002b:00000000f76ff01c EFLAGS: 00000296 ORIG_RAX: 0000000000000139 RAX: ffffffffffffffda RBX: 0000000000000014 RCX: 0000000000000000 RDX: 0000000000000013 RSI: 0000000000000000 RDI: 000000005813dd57 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Code: d9 4c 8d 73 01 48 c1 e8 03 83 e1 07 42 0f b6 04 38 38 c8 7f 08 84 c0 0f 85 bf 00 00 00 48 8b 85 58 ff ff ff 48 8b 8d 68 ff ff ff <41> 0f b6 5e ff 0f b6 00 83 e1 07 38 c8 7f 08 84 c0 0f 85 89 00 RIP: snd_seq_oss_readq_puts+0x146/0x210 sound/core/seq/oss/seq_oss_readq.c:112 RSP: ffff8801c2ef6db0 CR2: ffffc90001688000 ---[ end trace c7c98a599517e76d ]--- > IP: snd_seq_oss_readq_puts+0x146/0x210 > sound/core/seq/oss/seq_oss_readq.c:112 > PGD 1dac75067 P4D 1dac75067 PUD 1dac76067 PMD 1c1cd3067 PTE 0 > Oops: 0000 [#1] SMP KASAN > Dumping ftrace buffer: > (ftrace buffer empty) > Modules linked in: > CPU: 0 PID: 3012 Comm: syzkaller659314 Not tainted 4.14.0-rc1-next-20170918+ > #24 > Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS > Google 01/01/2011 > task: ffff8801ce930000 task.stack: ffff8801ce650000 > RIP: 0010:snd_seq_oss_readq_puts+0x146/0x210 > sound/core/seq/oss/seq_oss_readq.c:112 > RSP: 0018:ffff8801ce657208 EFLAGS: 00010246 > RAX: ffffed0039ccae4a RBX: ffffc900016f4000 RCX: ffff8801ce657251 > RDX: 0000000000000000 RSI: 1ffff10039d26117 RDI: 0000000000000282 > RBP: ffff8801ce6572b8 R08: ffff8801d807ab50 R09: 0000000000000000 > R10: 0000000000000000 R11: 0000000000000000 R12: 000000003fffffa4 > R13: ffff8801ce657290 R14: ffffc900016f4001 R15: dffffc0000000000 > FS: 00007f9079b19700(0000) GS:ffff8801db200000(0000) knlGS:0000000000000000 > CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 > CR2: ffffc900016f4000 CR3: 00000001ce47d000 CR4: 00000000001406f0 > DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 > DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 > Call Trace: > send_midi_event sound/core/seq/oss/seq_oss_midi.c:616 [inline] > snd_seq_oss_midi_input+0xd39/0x1040 sound/core/seq/oss/seq_oss_midi.c:535 > snd_seq_oss_event_input+0x15d/0x220 sound/core/seq/oss/seq_oss_event.c:439 > snd_seq_deliver_single_event.constprop.11+0x2fb/0x940 > sound/core/seq/seq_clientmgr.c:621 > deliver_to_subscribers sound/core/seq/seq_clientmgr.c:676 [inline] > snd_seq_deliver_event+0x318/0x790 sound/core/seq/seq_clientmgr.c:807 > snd_seq_kernel_client_dispatch+0x11e/0x150 > sound/core/seq/seq_clientmgr.c:2309 > dummy_input+0x2c4/0x400 sound/core/seq/seq_dummy.c:104 > snd_seq_deliver_single_event.constprop.11+0x2fb/0x940 > sound/core/seq/seq_clientmgr.c:621 > snd_seq_deliver_event+0x12c/0x790 sound/core/seq/seq_clientmgr.c:818 > snd_seq_dispatch_event+0x105/0x5b0 sound/core/seq/seq_clientmgr.c:892 > snd_seq_check_queue.part.3+0x38e/0x510 sound/core/seq/seq_queue.c:285 > snd_seq_check_queue sound/core/seq/seq_queue.c:357 [inline] > snd_seq_enqueue_event+0x32d/0x3d0 sound/core/seq/seq_queue.c:363 > snd_seq_client_enqueue_event+0x21b/0x420 sound/core/seq/seq_clientmgr.c:951 > kernel_client_enqueue.part.10+0xb5/0xd0 sound/core/seq/seq_clientmgr.c:2248 > kernel_client_enqueue sound/core/seq/seq_clientmgr.c:2238 [inline] > snd_seq_kernel_client_enqueue_blocking+0xcf/0x110 > sound/core/seq/seq_clientmgr.c:2275 > insert_queue sound/core/seq/oss/seq_oss_rw.c:189 [inline] > snd_seq_oss_write+0x5fe/0xa80 sound/core/seq/oss/seq_oss_rw.c:148 > odev_write+0x64/0x90 sound/core/seq/oss/seq_oss.c:177 > __vfs_write+0xef/0x970 fs/read_write.c:479 > vfs_write+0x18f/0x510 fs/read_write.c:543 > SYSC_write fs/read_write.c:588 [inline] > SyS_write+0xef/0x220 fs/read_write.c:580 > entry_SYSCALL_64_fastpath+0x1f/0xbe > RIP: 0033:0x446469 > RSP: 002b:00007f9079b18dc8 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 > RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 0000000000446469 > RDX: 0000000000000150 RSI: 0000000020e7f000 RDI: 0000000000000003 > RBP: 0000000000000082 R08: 00007f9079b19700 R09: 00007f9079b19700 > R10: 00007f9079b19700 R11: 0000000000000246 R12: 0000000000000000 > R13: 00007ffe543fcc1f R14: 00007f9079b199c0 R15: 0000000000000000 > Code: d9 4c 8d 73 01 48 c1 e8 03 83 e1 07 42 0f b6 04 38 38 c8 7f 08 84 c0 > 0f 85 bf 00 00 00 48 8b 85 58 ff ff ff 48 8b 8d 68 ff ff ff <41> 0f b6 5e ff > 0f b6 00 83 e1 07 38 c8 7f 08 84 c0 0f 85 89 00 > RIP: snd_seq_oss_readq_puts+0x146/0x210 > sound/core/seq/oss/seq_oss_readq.c:112 RSP: ffff8801ce657208 > CR2: ffffc900016f4000 > ---[ end trace cc8c763aca15dec8 ]--- > > > --- > This bug is generated by a dumb bot. It may contain errors. > See https://goo.gl/tpsmEJ for details. > Direct all questions to syzkaller(a)googlegroups.com. > Please credit me with: Reported-by: syzbot <syzkaller(a)googlegroups.com> > > syzbot will keep track of this bug report. > Once a fix for this bug is committed, please reply to this email with: > #syz fix: exact-commit-title > To mark this as a duplicate of another syzbot report, please reply with: > #syz dup: exact-subject-of-another-report > If it's a one-off invalid bug report, please reply with: > #syz invalid > Note: if the crash happens again, it will cause creation of a new bug > report. > Note: all commands must start from beginning of the line. > > -- > You received this message because you are subscribed to the Google Groups > "syzkaller-bugs" group. > To unsubscribe from this group and stop receiving emails from it, send an > email to syzkaller-bugs+unsubscribe(a)googlegroups.com. > To view this discussion on the web visit > https://groups.google.com/d/msgid/syzkaller-bugs/001a1140b6b825822a055cf02f…. > For more options, visit https://groups.google.com/d/optout.

2 7

[alsa-devel] [alsa-lib][PATCH] pcm: route: Fix use_getput flag computation for 3 byte formats
by Maciej S. Szmigiero 20 Nov '17

20 Nov '17

Commit de63b942acf520 ("pcm: route: Use get/put labels for all 3 byte formats") wanted to make the route plugin use get / put labels not only for 24-bit physical width formats but also for 18 and 20-bit ones. There was, however, a typo in that commit so a check for these widths didn't really work. Let's fix it now. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- src/pcm/pcm_route.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_route.c b/src/pcm/pcm_route.c index 999a2eed3a25..1f485a8e79f9 100644 --- a/src/pcm/pcm_route.c +++ b/src/pcm/pcm_route.c @@ -567,8 +567,8 @@ static int snd_pcm_route_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) return err; /* 3 bytes formats? */ route->params.use_getput = - (snd_pcm_format_physical_width(src_format) + 7) / 3 == 3 || - (snd_pcm_format_physical_width(dst_format) + 7) / 3 == 3; + (snd_pcm_format_physical_width(src_format) + 7) / 8 == 3 || + (snd_pcm_format_physical_width(dst_format) + 7) / 8 == 3; route->params.get_idx = snd_pcm_linear_get_index(src_format, SND_PCM_FORMAT_S32); route->params.put_idx = snd_pcm_linear_put_index(SND_PCM_FORMAT_S32, dst_format); route->params.conv_idx = snd_pcm_linear_convert_index(src_format, dst_format); -- 2.14.1

2 1

[alsa-devel] [RFC PATCH] ALSA: core: do not update tstamp when hw ptr has not moved
by Henrik Eriksson 20 Nov '17

20 Nov '17

Hello This reverts a change that was part of a larger commit. That change broke timestamps in some application code for us as outlined in the log, and reverting it obviously fixes this. Since the original change did not explain why the behaviour was modified I can't speak for the effects on that use case. Regards, /henrik >8------------------------------------------------------8< Commit 3179f62001880e588e229db3006a59ad87b7792a ("ALSA: core: add .get_time_info") made an undocumented change to the behaviour of the PCM runtime tstamp. Prior to this change the tstamp was not updated by snd_pcm_update_hw_ptr0() unless the hw ptr had moved, after this change the tstamp is always updated. For an application, using alsa-lib, doing snd_pcm_readi() followed by snd_pcm_status() to estimate the age of the read samples by subtracting status->avail * [sample rate] from status->tstamp this change degraded the accuracy of that estimate on devices where the pcm hw does not provide a granular hw pointer. On a device using soc-generic-dmaengine-pcm.c and a dma-engine with residue_granularity DMA_RESIDUE_GRANULARITY_DESCRIPTOR the accuracy of the estimate depended on the latency between the PCM hw completing a period and when the driver called snd_pcm_period_elapsed() to notify ALSA core, typically determined by interrupt handling latency. After this change the accuracy of the estimate is determined by the latency between the PCM hw completing a period and the application calling snd_pcm_status(), determined by the scheduling of the application process. Revert the part of commit 3179f62001880e588e229db3006a59ad87b7792a ("ALSA: core: add .get_time_info") that changed be behaviour of snd_pcm_update_hw_ptr0() when the PCM hw pointer has not moved. Fixes: 3179f6200188 ("ALSA: core: add .get_time_info") Signed-off-by: Henrik Eriksson <henrik.eriksson(a)axis.com> --- sound/core/pcm_lib.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a93a4235a332..2fbaf9a974cc 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -437,10 +437,8 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, } no_delta_check: - if (runtime->status->hw_ptr == new_hw_ptr) { - update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp); + if (runtime->status->hw_ptr == new_hw_ptr) return 0; - } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) -- 2.11.0 -- henrik eriksson fixed box cameras firmware, base video products, axis communications ab

3 6

[alsa-devel] strange code in pcm_route.c for 3 byte formats
by Maciej S. Szmigiero 20 Nov '17

20 Nov '17

Hi all, While reading alsa-lib sources I've spotted some strange code in src/pcm/pcm_route.c: /* 3 bytes formats? */ route->params.use_getput = (snd_pcm_format_physical_width(src_format) + 7) / 3 == 3 || (snd_pcm_format_physical_width(dst_format) + 7) / 3 == 3; Since snd_pcm_format_physical_width() returns a width in bits it seems to me that this would mean formats that are between 2 and 4 bits (inclusive) wide. Before commit de63b942acf520 ("pcm: route: Use get/put labels for all 3 byte formats") this read as: route->params.use_getput = snd_pcm_format_physical_width(src_format) == 24 || snd_pcm_format_physical_width(dst_format) == 24; The commit message of that commit says that this change was supposed to set this flag also for 18 and 20-bit physical width formats. Is there a mistake in this code or am I wrong somewhere in my thinking? Best regards, Maciej

2 2

[alsa-devel] Again about regression introduced by commit 70d435ba1cd6 ("ASoC: imx-pcm-dma: simplify pcm_config")
by Kalle Kankare 20 Nov '17

20 Nov '17

Hi, The regression was previously discussed on this mailing list last month (2017-10-04). As in the previous discussion, this commit restricts the supported sound formats to S24_LE. This problem is experienced in our iMX6q based hardware with SGTL5000 sound chip. I.e.: # arecord -l | grep 'card 0' -A2 card 0: imx6qba16sgtl50 [imx6q-ba16-sgtl5000], device 0: HiFi sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 In kernel 4.11.0: # arecord --dump-hw-params -D hw:0,0 2>&1 | grep '^FORMAT' FORMAT: S16_LE And in now in kernel 4.13.3: # arecord --dump-hw-params -D hw:0,0 2>&1 | grep '^FORMAT' FORMAT: S24_LE By reverting the mentioned commit 70d435ba1cd6, the format returned to S16_LE. However, I'm unsure if that is the proper way to solve this issue. I.e. does reverting the mentioned commit cause some other unwanted effect ? What would be the best solution that would make the S16_LE format supported again for SGTL5000 ?

2 2

[alsa-devel] [PATCH] ASoC: bells: Use common error handling code in bells_set_bias_level_post()
by SF Markus Elfring 20 Nov '17

20 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 19 Nov 2017 15:00:22 +0100 * Add a jump target so that a specific error message is stored only once at the end of this function implementation. * Replace two calls of the macro "pr_err" by goto statements. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/samsung/bells.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 34deba461ae1..fc9a71523b3e 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -120,18 +120,14 @@ static int bells_set_bias_level_post(struct snd_soc_card *card, switch (level) { case SND_SOC_BIAS_STANDBY: ret = snd_soc_codec_set_pll(codec, WM5102_FLL1, 0, 0, 0); - if (ret < 0) { - pr_err("Failed to stop FLL: %d\n", ret); - return ret; - } + if (ret < 0) + goto report_failure; if (bells->asyncclk_rate) { ret = snd_soc_codec_set_pll(codec, WM5102_FLL2, 0, 0, 0); - if (ret < 0) { - pr_err("Failed to stop FLL: %d\n", ret); - return ret; - } + if (ret < 0) + goto report_failure; } break; @@ -142,6 +138,10 @@ static int bells_set_bias_level_post(struct snd_soc_card *card, dapm->bias_level = level; return 0; + +report_failure: + pr_err("Failed to stop FLL: %d\n", ret); + return ret; } static int bells_late_probe(struct snd_soc_card *card) -- 2.15.0

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek: Add headset mic support for Intel NUC Skull Canyon
by Kai-Heng Feng 20 Nov '17

20 Nov '17

Users have been using knob "model=dell-headset-multi" on Intel Skull Canyon for a while. Add the equivalent quirk, ALC269_FIXUP_DELL1_MIC_NO_PRESENCE for Skull Canyon. BugLink: https://bugs.launchpad.net/bugs/1732034 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1797110cb096..5bb5623339dc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6455,6 +6455,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { ALC225_STANDARD_PINS, {0x12, 0xb7a60130}, {0x1b, 0x90170110}), + SND_HDA_PIN_QUIRK(0x10ec0233, 0x8086, "Intel NUC Skull Canyon", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x1b, 0x01111010}, + {0x1e, 0x01451130}, + {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60140}, {0x14, 0x90170110}, -- 2.14.1

2 1

[alsa-devel] [PATCH] ASoC: wm8804: Use common error handling code in wm8804_probe()
by SF Markus Elfring 20 Nov '17

20 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 19 Nov 2017 07:55:49 +0100 * Add a jump target so that a specific error message is stored only once at the end of this function implementation. * Replace two calls of the function "dev_err" by goto statements. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/codecs/wm8804.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index fc69b87443d8..a45812417909 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -626,16 +626,12 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) gpiod_set_value_cansleep(wm8804->reset, 1); ret = regmap_read(regmap, WM8804_RST_DEVID1, &id1); - if (ret < 0) { - dev_err(dev, "Failed to read device ID: %d\n", ret); - goto err_reg_enable; - } + if (ret < 0) + goto report_read_failure; ret = regmap_read(regmap, WM8804_DEVID2, &id2); - if (ret < 0) { - dev_err(dev, "Failed to read device ID: %d\n", ret); - goto err_reg_enable; - } + if (ret < 0) + goto report_read_failure; id2 = (id2 << 8) | id1; @@ -674,6 +670,8 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) return 0; +report_read_failure: + dev_err(dev, "Failed to read device ID: %d\n", ret); err_reg_enable: regulator_bulk_disable(ARRAY_SIZE(wm8804->supplies), wm8804->supplies); return ret; -- 2.15.0

2 1

[alsa-devel] [PATCH 0/4] ALSA SoC PCM: Fine-tuning for some function implementations
by SF Markus Elfring 19 Nov '17

19 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 19 Nov 2017 21:12:34 +0100 A few update suggestions were taken into account from static source code analysis. Markus Elfring (4): Improve unlocking of a mutex in soc_dpcm_runtime_update() Add some blank lines for better code readability Add some spaces for better code readability Improve a size determination in dpcm_be_connect() sound/soc/soc-pcm.c | 60 ++++++++++++++++++++++++++++++++--------------------- 1 file changed, 36 insertions(+), 24 deletions(-) -- 2.15.0

1 4

[alsa-devel] [PATCH 0/2] ASoC-Rockchip: Fine-tuning for rk_aif1_hw_params()
by SF Markus Elfring 19 Nov '17

19 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 19 Nov 2017 13:34:56 +0100 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Use common error handling code Delete two unnecessary variable initialisations sound/soc/rockchip/rockchip_max98090.c | 13 +++++-------- sound/soc/rockchip/rockchip_rt5645.c | 13 +++++-------- 2 files changed, 10 insertions(+), 16 deletions(-) -- 2.15.0

1 2

[alsa-devel] [PATCH] ASoC: sst-firmware: Improve unlocking of a mutex in sst_block_alloc_scratch()
by SF Markus Elfring 19 Nov '17

19 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 19 Nov 2017 10:45:36 +0100 * Add a jump target so that a call of the function "mutex_unlock" is stored only once in this function implementation. * Replace three calls by goto statements. * Increase the lock scope. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/intel/common/sst-firmware.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 79a9fdf94d38..665e66670512 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -1097,8 +1097,8 @@ int sst_block_alloc_scratch(struct sst_dsp *dsp) if (dsp->scratch_size == 0) { dev_info(dsp->dev, "no modules need scratch buffer\n"); - mutex_unlock(&dsp->mutex); - return 0; + ret = 0; + goto unlock; } /* allocate blocks for module scratch buffers */ @@ -1127,21 +1127,21 @@ int sst_block_alloc_scratch(struct sst_dsp *dsp) } if (ret < 0) { dev_err(dsp->dev, "error: can't alloc scratch blocks\n"); - mutex_unlock(&dsp->mutex); - return ret; + goto unlock; } ret = block_list_prepare(dsp, &dsp->scratch_block_list); if (ret < 0) { dev_err(dsp->dev, "error: scratch block prepare failed\n"); - mutex_unlock(&dsp->mutex); - return ret; + goto unlock; } /* assign the same offset of scratch to each module */ dsp->scratch_offset = ba.offset; + ret = dsp->scratch_size; +unlock: mutex_unlock(&dsp->mutex); - return dsp->scratch_size; + return ret; } EXPORT_SYMBOL_GPL(sst_block_alloc_scratch); -- 2.15.0

1 0

[alsa-devel] [PATCH] ASoC: audio-graph-scu-card: Use common error handling code in asoc_graph_card_parse_of()
by SF Markus Elfring 19 Nov '17

19 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 19 Nov 2017 08:51:41 +0100 Add a jump target so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/generic/audio-graph-scu-card.c | 22 +++++++++------------- 1 file changed, 9 insertions(+), 13 deletions(-) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index a967aa143d51..c4b999fba7e6 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -231,10 +231,8 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, NULL, &daifmt); - if (ret < 0) { - of_node_put(cpu_port); - goto parse_of_err; - } + if (ret < 0) + goto put_node; } dai_idx = 0; @@ -266,17 +264,13 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) /* Back-End (= Codec) */ ret = asoc_graph_card_dai_link_of(codec_ep, priv, daifmt, dai_idx++, 0); - if (ret < 0) { - of_node_put(cpu_port); - goto parse_of_err; - } + if (ret < 0) + goto put_node; } else { /* Front-End (= CPU) */ ret = asoc_graph_card_dai_link_of(cpu_ep, priv, daifmt, dai_idx++, 1); - if (ret < 0) { - of_node_put(cpu_port); - goto parse_of_err; - } + if (ret < 0) + goto put_node; } } } @@ -285,8 +279,10 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) if (ret) goto parse_of_err; - ret = 0; + return 0; +put_node: + of_node_put(cpu_port); parse_of_err: return ret; } -- 2.15.0

1 0

[alsa-devel] [PATCH] ALSA: via82xx: Use common error handling code in snd_via82xx_create()
by SF Markus Elfring 19 Nov '17

19 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 22 Aug 2017 19:37:26 +0200 Add jump targets so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/via82xx_modem.c | 33 +++++++++++++++++++-------------- 1 file changed, 19 insertions(+), 14 deletions(-) diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 55f79b2599e7..8659776887b2 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -1110,8 +1110,8 @@ static int snd_via82xx_create(struct snd_card *card, return err; if ((chip = kzalloc(sizeof(*chip), GFP_KERNEL)) == NULL) { - pci_disable_device(pci); - return -ENOMEM; + err = -ENOMEM; + goto disable_device; } spin_lock_init(&chip->reg_lock); @@ -1121,30 +1121,27 @@ static int snd_via82xx_create(struct snd_card *card, if ((err = pci_request_regions(pci, card->driver)) < 0) { kfree(chip); - pci_disable_device(pci); - return err; + goto disable_device; } chip->port = pci_resource_start(pci, 0); if (request_irq(pci->irq, snd_via82xx_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); - snd_via82xx_free(chip); - return -EBUSY; + err = -EBUSY; + goto free_chip; } chip->irq = pci->irq; if (ac97_clock >= 8000 && ac97_clock <= 48000) chip->ac97_clock = ac97_clock; synchronize_irq(chip->irq); - if ((err = snd_via82xx_chip_init(chip)) < 0) { - snd_via82xx_free(chip); - return err; - } + err = snd_via82xx_chip_init(chip); + if (err < 0) + goto free_chip; - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_via82xx_free(chip); - return err; - } + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) + goto free_chip; /* The 8233 ac97 controller does not implement the master bit * in the pci command register. IMHO this is a violation of the PCI spec. @@ -1153,6 +1150,14 @@ static int snd_via82xx_create(struct snd_card *card, *r_via = chip; return 0; + +disable_device: + pci_disable_device(pci); + goto exit; +free_chip: + snd_via82xx_free(chip); +exit: + return err; } -- 2.14.0

3 6

[alsa-devel] [PATCH] ALSA: cs5530: Use common error handling code in snd_cs5530_probe()
by SF Markus Elfring 18 Nov '17

18 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 18 Nov 2017 22:10:19 +0100 Add a jump target so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/cs5530.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index 0a8cf94c4858..674572edf2d2 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -269,23 +269,24 @@ static int snd_cs5530_probe(struct pci_dev *pci, return err; err = snd_cs5530_create(card, pci, &chip); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto free_card; strcpy(card->driver, "CS5530"); strcpy(card->shortname, "CS5530 Audio"); sprintf(card->longname, "%s at 0x%lx", card->shortname, chip->pci_base); err = snd_card_register(card); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto free_card; + pci_set_drvdata(pci, card); dev++; return 0; + +free_card: + snd_card_free(card); + return err; } static struct pci_driver cs5530_driver = { -- 2.15.0

1 0

[alsa-devel] [PATCH 0/2] ALSA-Digigram VX222: Fine-tuning for four function implementations
by SF Markus Elfring 18 Nov '17

18 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 18 Nov 2017 21:27:50 +0100 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Adjust ten function calls together with a variable assignment Use common error handling code in two functions sound/pci/vx222/vx222.c | 53 +++++++++++++++++++++++++-------------------- sound/pci/vx222/vx222_ops.c | 19 +++++++++------- 2 files changed, 41 insertions(+), 31 deletions(-) -- 2.15.0

1 2

[alsa-devel] [PATCH 0/3] ALSA-Trident 4DWave: Fine-tuning for nine function implementations
by SF Markus Elfring 18 Nov '17

18 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 18 Nov 2017 19:50:12 +0100 A few update suggestions were taken into account from static source code analysis. Markus Elfring (3): Adjust 34 function calls together with a variable assignment Use common error handling code in two functions Use common error handling code in snd_trident_mixer() sound/pci/trident/trident.c | 68 ++++++------ sound/pci/trident/trident_main.c | 221 ++++++++++++++++++++++----------------- 2 files changed, 163 insertions(+), 126 deletions(-) -- 2.15.0

1 3

[alsa-devel] [PATCH 0/3] ALSA-S3 SonicVibes: Fine-tuning for five function implementations
by SF Markus Elfring 18 Nov '17

18 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 18 Nov 2017 16:25:43 +0100 A few update suggestions were taken into account from static source code analysis. Markus Elfring (3): Adjust 15 function calls together with a variable assignment Use common error handling code in snd_sonic_probe() Use common error handling code in snd_sonicvibes_create() sound/pci/sonicvibes.c | 144 +++++++++++++++++++++++++++---------------------- 1 file changed, 79 insertions(+), 65 deletions(-) -- 2.15.0

1 3

[alsa-devel] [PATCH 0/2] ALSA-RME Digi96: Fine-tuning for four function implementations
by SF Markus Elfring 18 Nov '17

18 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 18 Nov 2017 13:45:43 +0100 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Adjust 11 function calls together with a variable assignment Improve unlocking of an IRQ in snd_rme96_capture_hw_params() sound/pci/rme96.c | 91 +++++++++++++++++++++++++++++++------------------------ 1 file changed, 51 insertions(+), 40 deletions(-) -- 2.15.0

1 2

[alsa-devel] [PATCH 1/3 v3] ASoC: ep93xx-ac97: Fix platform_get_irq's error checking
by Arvind Yadav 18 Nov '17

18 Nov '17

The platform_get_irq() function returns negative if an error occurs. zero or positive number on success. platform_get_irq() error checking for zero is not correct. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- changes in v2 : irq was unsigned. so changed it to signed. changes in v3 : Add failure case '<= 0' instead of '< 0'. IRQ0 is not valid. sound/soc/cirrus/ep93xx-ac97.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index bbf7a92..b86b83b 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -365,7 +365,7 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) { struct ep93xx_ac97_info *info; struct resource *res; - unsigned int irq; + int irq; int ret; info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); @@ -378,8 +378,8 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) return PTR_ERR(info->regs); irq = platform_get_irq(pdev, 0); - if (!irq) - return -ENODEV; + if (irq <= 0) + return irq; ret = devm_request_irq(&pdev->dev, irq, ep93xx_ac97_interrupt, IRQF_TRIGGER_HIGH, pdev->name, info); -- 2.7.4

1 4

[alsa-devel] [PATCH 1/5 v2] ASoC: ep93xx-ac97: Fix platform_get_irq's error checking
by Arvind Yadav 18 Nov '17

18 Nov '17

The platform_get_irq() function returns negative if an error occurs. zero or positive number on success. platform_get_irq() error checking for zero is not correct. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- changes in v2 : irq was unsigned. so changed it to signed. sound/soc/cirrus/ep93xx-ac97.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index bbf7a92..b86b83b 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -365,7 +365,7 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) { struct ep93xx_ac97_info *info; struct resource *res; - unsigned int irq; + int irq; int ret; info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); @@ -378,8 +378,8 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) return PTR_ERR(info->regs); irq = platform_get_irq(pdev, 0); - if (!irq) - return -ENODEV; + if (irq < 0) + return irq; ret = devm_request_irq(&pdev->dev, irq, ep93xx_ac97_interrupt, IRQF_TRIGGER_HIGH, pdev->name, info); -- 2.7.4

2 5

[alsa-devel] [PATCH 0/3] ALSA-RME Digi32: Fine-tuning for seven function implementations
by SF Markus Elfring 17 Nov '17

17 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 17 Nov 2017 22:08:44 +0100 A few update suggestions were taken into account from static source code analysis. Markus Elfring (3): Adjust 15 function calls together with a variable assignment Use common error handling code in snd_rme32_probe() Improve unlocking of an IRQ in two functions sound/pci/rme32.c | 127 +++++++++++++++++++++++++++++++----------------------- 1 file changed, 74 insertions(+), 53 deletions(-) -- 2.15.0

1 3

Re: [alsa-devel] BUG: unable to handle kernel paging request in snd_seq_oss_readq_puts
by Dmitry Vyukov 17 Nov '17

17 Nov '17

On Fri, Nov 17, 2017 at 9:10 PM, syzbot <bot+31681772ec7a18dde8d3f8caf475f361a89b9514(a)syzkaller.appspotmail.com> wrote: > Hello, > > syzbot has tested the proposed patch but the reproducer still triggered > crash: > no output from test machine please ignore this, this testing of patch testing

1 0

[alsa-devel] Missing surround channels with Nvidia HDMI & snd-hda-intel driver
by VDR User 17 Nov '17

17 Nov '17

Hi. I just noticed that I'm missing some surround channels and I think it may be a bug in the alsa drivers because my current setup used to work just fine. I'm running Debian Testing, stable kernel 4.13.12, alsa-utils 1.1.3-1. My ~/.asoundrc is: ~$ cat .asoundrc pcm.!default "hdmi:NVidia,1" ~$ I am not setting anything on my kernel load line: echo 'Loading Linux 4.13.12-amd ...' linux /vmlinuz-4.13.12-amd root=UUID=6df5733d-b089-450c-9ec6-661dd013c7c0 ro echo 'Loading initial ramdisk ...' initrd /initrd.img-4.13.12-amd The driver being used is snd-hda-intel with an Nvidia GT520 video card. And lastly I do not have Pulse or a desktop environment. This is a minimal HTPC setup that runs only an xserver to provide video output, with audio/video being sent to my surround receiver via HDMI. The first thing I noticed was mpv-player was downmixing 5.1 to 3.1 and giving the following log message: [ao/alsa] trying to set ALSA channel map: FL FR FC LFE RL RR [ao/alsa] channel map reported by ALSA: FL FR FC LFE UNKNOWN UNKNOWN [ao/alsa] which we understand as: fl-fr-fc-lfe-na-na [ao/alsa] using the ALSA channel map. [ao/alsa] hw pausing supported: yes [ao/alsa] buffersize: 2720 samples [ao/alsa] period size: 160 samples [ao/alsa] device buffer: 2720 samples. [ao/alsa] using soft-buffer of 8820 samples. [cplayer] AO: [alsa] 44100Hz fl-fr-fc-lfe-na-na (3.1) 6ch s32 [cplayer] AO: Description: ALSA audio output [af] Adding filter lavrresample [lavrresample] Remix: 5.1 -> 3.1 [af] Audio filter chain: [af] [in] 44100Hz 5.1 6ch floatp [af] [lavrresample] 44100Hz fl-fr-fc-lfe-na-na (3.1) 6ch s32 [a] [af] [out] 44100Hz fl-fr-fc-lfe-na-na (3.1) 6ch s32 [af] [ao] 44100Hz fl-fr-fc-lfe-na-na (3.1) 6ch s32 That makes it look like Rear Left/Right channels are absent. I think ran speaker-test. 4 and 5 show as "Unknown" but correctly audibly announced as "front center" and "subwoofer" respectively: ~$ speaker-test -c 6 -t wav speaker-test 1.1.3 Playback device is default Stream parameters are 48000Hz, S16_LE, 6 channels WAV file(s) Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 5440 Period size range from 32 to 2720 Using max buffer size 5440 Periods = 4 was set period_size = 1088 was set buffer_size = 5440 0 - Front Left 1 - Front Right 3 - Rear Right 2 - Rear Left 4 - Unknown 5 - Unknown Additional information that may be of use: ~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 0/1 Subdevice #0: subdevice #0 ~$ cat /proc/asound/card1/eld#0.1 monitor_present 1 eld_valid 1 monitor_name SONY AVAMP connection_type HDMI eld_version [0x2] CEA-861D or below edid_version [0x3] CEA-861-B, C or D manufacture_id 0xd94d product_id 0xf02 port_id 0x200 support_hdcp 0 support_ai 0 audio_sync_delay 0 speakers [0x5f] FL/FR LFE FC RL/RR RC RLC/RRC sad_count 12 sad0_coding_type [0xc] MLP (Dolby TrueHD) sad0_channels 2 sad0_rates [0x1ec0] 44100 48000 88200 96000 176400 192000 sad1_coding_type [0xc] MLP (Dolby TrueHD) sad1_channels 6 sad1_rates [0x1ec0] 44100 48000 88200 96000 176400 192000 sad2_coding_type [0xc] MLP (Dolby TrueHD) sad2_channels 8 sad2_rates [0x6c0] 44100 48000 88200 96000 sad3_coding_type [0xb] DTS-HD sad3_channels 2 sad3_rates [0x1ec0] 44100 48000 88200 96000 176400 192000 sad4_coding_type [0xb] DTS-HD sad4_channels 6 sad4_rates [0x1ec0] 44100 48000 88200 96000 176400 192000 sad5_coding_type [0xb] DTS-HD sad5_channels 8 sad5_rates [0x6c0] 44100 48000 88200 96000 sad6_coding_type [0x1] LPCM sad6_channels 2 sad6_rates [0x1ee0] 32000 44100 48000 88200 96000 176400 192000 sad6_bits [0xe0000] 16 20 24 sad7_coding_type [0x1] LPCM sad7_channels 6 sad7_rates [0x1ee0] 32000 44100 48000 88200 96000 176400 192000 sad7_bits [0xe0000] 16 20 24 sad8_coding_type [0x1] LPCM sad8_channels 8 sad8_rates [0x1ee0] 32000 44100 48000 88200 96000 176400 192000 sad8_bits [0xe0000] 16 20 24 sad9_coding_type [0x2] AC-3 sad9_channels 6 sad9_rates [0xe0] 32000 44100 48000 sad9_max_bitrate 680000 sad10_coding_type [0x7] DTS sad10_channels 6 sad10_rates [0x6e0] 32000 44100 48000 88200 96000 sad10_max_bitrate 1536000 sad11_coding_type [0xa] E-AC-3/DD+ (Dolby Digital Plus) sad11_channels 8 sad11_rates [0xc0] 44100 48000 02:00.0 VGA compatible controller: NVIDIA Corporation GF119 [GeForce GT 520] (rev a1) (prog-if 00 [VGA controller]) Subsystem: eVga.com. Corp. GF119 [GeForce GT 520] Flags: bus master, fast devsel, latency 0, IRQ 25 Memory at fd000000 (32-bit, non-prefetchable) [size=16M] Memory at f0000000 (64-bit, prefetchable) [size=128M] Memory at fa000000 (64-bit, prefetchable) [size=32M] I/O ports at ec00 [size=128] [virtual] Expansion ROM at 000c0000 [disabled] [size=128K] Capabilities: <access denied> Kernel driver in use: nvidia Kernel modules: nvidia 02:00.1 Audio device: NVIDIA Corporation GF119 HDMI Audio Controller (rev a1) Subsystem: eVga.com. Corp. GF119 HDMI Audio Controller Flags: bus master, fast devsel, latency 0, IRQ 17 Memory at fea7c000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel

2 14

Re: [alsa-devel] [GIT PULL] sound updates for 4.15-rc1
by Takashi Iwai 17 Nov '17

17 Nov '17

[Adding more people and alsa-devel to Cc] On Wed, 15 Nov 2017 03:40:09 +0100, Linus Torvalds wrote: > > On Tue, Nov 14, 2017 at 6:51 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > please pull sound updates for v4.15-rc1 from: > > Hmm. Making "oldconfig" on my laptop with this, my > SND_SOC_INTEL_SKYLAKE went away. > > And the reason seems to be that new SND_SOC_INTEL_SST_TOPLEVEL config option. > > Which has no help associated with it. > > This is not a friendly thing to do to people. It basically breaks > existing setups for no documented reason, and with no explanation. > > Please fix the config situation. At the very least, add documentation. Sorry about that. I saw Vinod already submitted a patch to add the help text to CONFIG_SND_SOC_INTEL_SST_TOPLEVEL, so the least fix should go in soon. But now looking at these changes, I noticed a few things, too: - With the introduction of SND_SOC_INTEL_SST_TOPLEVEL, keeping SND_SOC_INTEL_COMMON and SND_SOC_INTEL_MACH individually doesn't make much sense. They can be dropped and replaced with SND_SOC_INTEL_SST_TOPLEVEL as a further cleanup. - ... or, make SND_SOC_INTEL_SST_TOPLEVEL=y as default, if this is considered to be a top-level filter config (like the network vendor kconfig items). In that case, the reverse-selection of SND_SOC_INTEL_COMMON and SND_SOC_INTEL_MACH should be avoided, but they should be selected from the actual drivers instead. And I believe there are a few more possible cleanups / fixes in the messy Intel ASoC Kconfigs. For example, SND_SOC_INTEL_SST is almost always set. The only exception is via SND_SST_ATOM_HIFI2_PLATFORM. But all machine drivers using Atom Hifi2 do set SND_SST_IPC_ACPI, which also requires SND_SOC_INTEL_SST. Further looking at this, we see that the only entry that does *not* require SND_SOC_INTEL_SST is the case with SND_MFLD_MACHINE in sound/soc/intel/boards. And now more interesting part -- there is no corresponding entry in Makefile. That is, this kconfig is effectively dead! The source code mfld_machine.c exists, but it's just a place holder now. The code was supposed to be integrated into atom directory by the commit b97169da0699, but it seems forgotten to be updated. Hmm... Takashi

4 13

Re: [alsa-devel] [PATCH] ASoC: Intel: mfld: allow mfld machine to be build
by Vinod Koul 17 Nov '17

17 Nov '17

On Wed, Nov 15, 2017 at 08:41:18AM -0600, Pierre-Louis Bossart wrote: > On 11/15/17 4:52 AM, Vinod Koul wrote: > >Commit b97169da0699: ("ASoC: Intel: create atom folder and move atom > >platform files in") missed adding Kconfig and Makefile support for > >MFLD machine driver, so add it while alos fixing the relative patch to > >codec header > > Does anyone test this code? I don't think anyone on the Intel side has a > working platform any longer... That is good question, we had boards for these but not sure if this is needed. This was not compiling and noone noticed, so I dont mind removing these bits. Any objections.. -- ~Vinod

3 2

[alsa-devel] [PATCH] ALSA: synth: emux: remove redundant test for r <= 13
by Colin King 17 Nov '17

17 Nov '17

From: Colin Ian King <colin.king(a)canonical.com> The calculation r = (3 - ((rate >> 6) & 3)) * 3 results in r being 0, 3, 6 or 9 and so the check (13 > r) is always true and hence we can remove the redundant check and the else path. Detected by CoverityScan, CID#744415 ("Logically dead code") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/synth/emux/soundfont.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 31a4ea94830e..0beb5a737c8b 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -858,10 +858,7 @@ calc_gus_envelope_time(int rate, int start, int end) p = rate & 0x3f; t = end - start; if (t < 0) t = -t; - if (13 > r) - t = t << (13 - r); - else - t = t >> (r - 13); + t = t << (13 - r); return (t * 10) / (p * 441); } -- 2.14.1

4 3

[alsa-devel] [PATCH] ALSA: usb-audio: uac1: Invalidate ctl on interrupt
by Julian Scheel 17 Nov '17

17 Nov '17

When an interrupt occurs, the value of at least one of the belonging controls should have changed. To make sure they get re-read from device on the next read, invalidate the cache. This was correctly implemented for uac2 already, but missing for uac1. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- sound/usb/mixer.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 91bc8f18791e..1beb5b4d027e 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2330,9 +2330,14 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid) { struct usb_mixer_elem_list *list; - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) + for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) { + struct usb_mixer_elem_info *info = + (struct usb_mixer_elem_info *)list; + /* invalidate cache, so the value is read from the device */ + info->cached = 0; snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &list->kctl->id); + } } static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer, -- 2.15.0

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