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

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

22 Oct '19

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

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[alsa-devel] [PATCH 1/2] ASoC: adau1761: Add PGA Slew time control
by Alexandru Ardelean 01 Oct '19

01 Oct '19

From: Lars-Peter Clausen <lars(a)metafoo.de> The PGA Slew Time control allows to configure the rate with which the PGA gain control ramps up/down to the target setting. The PGA slew control is done via the ALC Control 0 register. There are 2 bits on that reg, that control PGA slew time, while the other bits control parts of the ALC (automatic level control) block. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- sound/soc/codecs/adau1761.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..428bf95ea648 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -29,6 +29,7 @@ #define ADAU1761_REC_MIXER_RIGHT1 0x400d #define ADAU1761_LEFT_DIFF_INPUT_VOL 0x400e #define ADAU1761_RIGHT_DIFF_INPUT_VOL 0x400f +#define ADAU1761_ALC_CTRL0 0x4011 #define ADAU1761_PLAY_LR_MIXER_LEFT 0x4020 #define ADAU1761_PLAY_MIXER_LEFT0 0x401c #define ADAU1761_PLAY_MIXER_LEFT1 0x401d @@ -72,6 +73,7 @@ static const struct reg_default adau1761_reg_defaults[] = { { ADAU1761_REC_MIXER_RIGHT0, 0x00 }, { ADAU1761_REC_MIXER_RIGHT1, 0x00 }, { ADAU1761_LEFT_DIFF_INPUT_VOL, 0x00 }, + { ADAU1761_ALC_CTRL0, 0x00 }, { ADAU1761_RIGHT_DIFF_INPUT_VOL, 0x00 }, { ADAU1761_PLAY_LR_MIXER_LEFT, 0x00 }, { ADAU1761_PLAY_MIXER_LEFT0, 0x00 }, @@ -148,6 +150,21 @@ static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_capture_bias_enum, ADAU17X1_REC_POWER_MGMT, 1, 0x3, adau1761_bias_select_text, adau1761_bias_select_values); +static const unsigned int adau1761_pga_slew_time_values[] = { + 3, 0, 1, 2, +}; + +static const char * const adau1761_pga_slew_time_text[] = { + "Off", + "24 ms", + "48 ms", + "96 ms", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_pga_slew_time_enum, + ADAU1761_ALC_CTRL0, 6, 0x3, adau1761_pga_slew_time_text, + adau1761_pga_slew_time_values); + static const struct snd_kcontrol_new adau1761_jack_detect_controls[] = { SOC_SINGLE("Speaker Auto-mute Switch", ADAU1761_DIGMIC_JACKDETECT, 4, 1, 0), @@ -162,6 +179,8 @@ static const struct snd_kcontrol_new adau1761_differential_mode_controls[] = { SOC_DOUBLE_R_TLV("PGA Boost Capture Volume", ADAU1761_REC_MIXER_LEFT1, ADAU1761_REC_MIXER_RIGHT1, 3, 2, 0, adau1761_pga_boost_tlv), + + SOC_ENUM("PGA Capture Slew Time", adau1761_pga_slew_time_enum), }; static const struct snd_kcontrol_new adau1761_single_mode_controls[] = { @@ -633,6 +652,7 @@ static bool adau1761_readable_register(struct device *dev, unsigned int reg) case ADAU1761_DEJITTER: case ADAU1761_CLK_ENABLE0: case ADAU1761_CLK_ENABLE1: + case ADAU1761_ALC_CTRL0: return true; default: break; -- 2.20.1

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[alsa-devel] [PATCH v5 0/5] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 18 Sep '19

18 Sep '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. Changes from v4 to v5: - synopsys/Kconfig: Remove the incorrect dependency change in v4. - rockchip/Kconfig: Add dependency of hdmi-codec when it is really need for jack support. Cheng-Yi Chiang (5): ASoC: hdmi-codec: Add an op to set callback function for plug event drm: bridge: dw-hdmi: Report connector status using callback drm: dw-hdmi-i2s: Use fixed id for codec device ASoC: rockchip_max98090: Add dai_link for HDMI ASoC: rockchip_max98090: Add HDMI jack support .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 13 +- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 41 ++++++- include/drm/bridge/dw_hdmi.h | 4 + include/sound/hdmi-codec.h | 17 +++ sound/soc/codecs/hdmi-codec.c | 46 +++++++ sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 3 +- sound/soc/rockchip/rockchip_max98090.c | 116 ++++++++++++++---- 8 files changed, 216 insertions(+), 27 deletions(-) -- 2.22.0.510.g264f2c817a-goog

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[alsa-devel] [PATCH] ASoC: rsnd: fixup 6ch settings to 8ch
by Kuninori Morimoto 30 Aug '19

30 Aug '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd need to use 8ch clock settings for 6ch for TDM. Otherwise, it can't work correctly. This patch fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 18 +++++++++++++----- sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/sh/rcar/ssi.c | 2 ++ 3 files changed, 16 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4fe83e61..37cb615 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -300,6 +300,18 @@ int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, return chan; } +int rsnd_channel_normalization(int chan) +{ + if ((chan > 8) || (chan < 0)) + return 0; + + /* TDM Extend Mode needs 8ch */ + if (chan == 6) + chan = 8; + + return chan; +} + int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, struct snd_pcm_hw_params *params) { @@ -312,11 +324,7 @@ int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, if (rsnd_runtime_is_multi_ssi(io)) chan /= rsnd_rdai_ssi_lane_get(rdai); - /* TDM Extend Mode needs 8ch */ - if (chan == 6) - chan = 8; - - return chan; + return rsnd_channel_normalization(chan); } int rsnd_runtime_is_multi_ssi(struct rsnd_dai_stream *io) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 0e6ef4e1..7727add 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -446,6 +446,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, struct device_node *playback, struct device_node *capture); +int rsnd_channel_normalization(int chan); #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, diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index f5afab6..44bda21 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -303,6 +303,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_runtime_is_tdm_split(io)) chan = rsnd_io_converted_chan(io); + chan = rsnd_channel_normalization(chan); + main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx); if (!main_rate) { dev_err(dev, "unsupported clock rate\n"); -- 2.7.4

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[alsa-devel] [BUG] bdw-rt5650 DSP boot timeout
by Jon Flatley 28 Aug '19

28 Aug '19

I've been working on upstreaming the bdw-rt5650 machine driver for the Acer Chromebase 24 (buddy). There seems to be an issue when first setting the hardware controls that appears to be crashing the DSP: [ 51.424554] haswell-pcm-audio haswell-pcm-audio: FW loaded, mailbox readback FW info: type 01, - version: 00.00, build 77, source commit id: 876ac6906f31a43b6772b23c7c983ce9dcb18a19 ... [ 84.924666] haswell-pcm-audio haswell-pcm-audio: error: audio DSP boot timeout IPCD 0x0 IPCX 0x0 [ 85.260655] haswell-pcm-audio haswell-pcm-audio: ipc: --message timeout-- ipcx 0x83000000 isr 0x00000000 ipcd 0x00000000 imrx 0x7fff0000 [ 85.273609] haswell-pcm-audio haswell-pcm-audio: error: stream commit failed [ 85.279746] System PCM: error: failed to commit stream -110 [ 85.285388] haswell-pcm-audio haswell-pcm-audio: ASoC: haswell-pcm-audio hw params failed: -110 [ 85.293963] System PCM: ASoC: hw_params FE failed -110 This happens roughly 50% of the time when first setting hardware controls after a reboot. The other 50% of the time the DSP comes up just fine and audio works fine thereafter. Adding "#define DEBUG 1" to sound/soc/intel/haswell/sst-haswell-ipc.c makes the issue occur much less frequently in my testing. Seems like a subtle timing issue. There were timing issues encountered during the bringup of the 2015 chromebook pixel (samus) which uses the bdw-rt5677 machine driver. Those were slightly different, and manifested during repeated arecords. Both devices use the same revision of the sst2 firmware. Any ideas for how to debug this? Thanks, Jon

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[alsa-devel] [RFC PATCH 00/40] soundwire: updates for 5.4
by Pierre-Louis Bossart 23 Aug '19

23 Aug '19

The existing upstream code allows for SoundWire devices to be enumerated and managed by the bus, but streaming is not currently supported. Bard Liao, Rander Wang and I did quite a bit of integration/validation work to close this gap and we now have SoundWire streaming + basic power managemement on Intel CometLake and IceLake reference boards. These changes are still preliminary and should not be merged as is, but it's time to start reviews. While the number of patches is quite large, each of the changes is quite small. SOF driver changes will be submitted shortly as well but are still being validated. ClockStop modes and synchronized playback on multiple links are not supported for now and will likely be part of the next cycle (dependencies on codec drivers and multi-cpu DAI support). Acknowledgements: This work would not have been possible without the support of Slawomir Blauciak and Tomasz Lauda on the SOF side, currently being reviewed, see https://github.com/thesofproject/sof/pull/1638 Comments and feedback welcome! Bard liao (1): soundwire: include mod_devicetable.h to avoid compiling warnings Pierre-Louis Bossart (38): soundwire: add debugfs support soundwire: cadence_master: add debugfs register dump soundwire: cadence_master: align debugfs to 8 digits soundwire: intel: add debugfs register dump soundwire: intel: move interrupt enable after interrupt handler registration soundwire: intel: prevent possible dereference in hw_params soundwire: intel: fix channel number reported by hardware soundwire: intel: remove BIOS work-arounds soundwire: cadence_master: fix usage of CONFIG_UPDATE soundwire: cadence_master: remove useless wrapper soundwire: cadence_master: simplify bus clash interrupt clear soundwire: cadence_master: revisit interrupt settings soundwire: cadence_master: fix register definition for SLAVE_STATE soundwire: cadence_master: fix definitions for INTSTAT0/1 soundwire: cadence_master: handle multiple status reports per Slave soundwire: cadence_master: improve startup sequence with link hw_reset soundwire: bus: use runtime_pm_get_sync/pm when enabled soundwire: bus: split handling of Device0 events soundwire: bus: improve dynamic debug comments for enumeration soundwire: prototypes for suspend/resume soundwire: export helpers to find row and column values soundwire: stream: fix disable sequence soundwire: cadence_master: use BIOS defaults for frame shape soundwire: intel: use BIOS information to set clock dividers soundwire: Add Intel resource management algorithm soundwire: intel: handle disabled links soundwire: intel_init: add kernel module parameter to filter out links soundwire: cadence_master: add kernel parameter to override interrupt mask soundwire: intel: move shutdown() callback and don't export symbol soundwire: intel: add helper for initialization soundwire: intel: Add basic power management support soundwire: intel: ignore disabled links for suspend/resume soundwire: intel: export helper to exit reset soundwire: intel: disable interrupts on suspend soundwire: cadence_master: add hw_reset capability in debugfs soundwire: cadence_master: make clock stop exit configurable on init soundwire: intel: add pm_runtime support soundwire: intel: add delay on restart for enumeration Rander Wang (1): soundwire: cadence_master: fix divider setting in clock register drivers/soundwire/Makefile | 4 +- drivers/soundwire/algo_dynamic_allocation.c | 403 ++++++++++++++++++++ drivers/soundwire/bus.c | 44 ++- drivers/soundwire/bus.h | 77 +++- drivers/soundwire/bus_type.c | 3 + drivers/soundwire/cadence_master.c | 365 ++++++++++++++---- drivers/soundwire/cadence_master.h | 12 +- drivers/soundwire/debugfs.c | 156 ++++++++ drivers/soundwire/intel.c | 381 +++++++++++++++++- drivers/soundwire/intel_init.c | 14 + drivers/soundwire/slave.c | 1 + drivers/soundwire/stream.c | 53 ++- include/linux/soundwire/sdw.h | 15 + 13 files changed, 1414 insertions(+), 114 deletions(-) create mode 100644 drivers/soundwire/algo_dynamic_allocation.c create mode 100644 drivers/soundwire/debugfs.c -- 2.20.1

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[alsa-devel] [PATCH 0/5] ASoC: atmel: extend SSC support
by Michał Mirosław 23 Aug '19

23 Aug '19

This series improves support for various configurations using SSC module as implemented in Atmel SAMA5Dx SoCs. Patches are: 1. enable SSC in Kconfig for audio-graph-card support 2-3. implement left-justified data mode 4-5. fix shared FSYNC source for slave mode Patches against tiwai/sound/for-next tree. You can also pull from https://rere.qmqm.pl/git/linux branch: atmel-ssc Michał Mirosław (5): ASoC: atmel: enable SSC_PCM_DMA in Kconfig ASoC: atmel_ssc_dai: rework DAI format configuration ASoC: atmel_ssc_dai: implement left-justified data mode ASoC: atmel_ssc_dai: split TX/RX FS constants ASoC: atmel_ssc_dai: Enable shared FSYNC source in frame-slave mode .../devicetree/bindings/misc/atmel-ssc.txt | 4 + drivers/misc/atmel-ssc.c | 2 + include/linux/atmel-ssc.h | 1 + sound/soc/atmel/Kconfig | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 306 ++++++------------ sound/soc/atmel/atmel_ssc_dai.h | 9 +- 6 files changed, 120 insertions(+), 204 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH 2/3] input: Add SW_UNSUPPORT_INSERT define
by bgoswami＠codeaurora.org 22 Aug '19

22 Aug '19

From: Banajit Goswami <bgoswami(a)codeaurora.org> Some devices may not support specific type of input devices. For example, when a headset or extension cable with GND/MIC swap is plugged into a headset jack that does not support the headset/cable, it needs to be reported with a corresponding input event. Also, increase the max values for INPUT_DEVICE_ID_SW_MAX and SW_MAX, to accommodate future extension of the number of event. Signed-off-by: Gopikrishnaiah Anandan <agopik(a)codeaurora.org> Signed-off-by: Banajit Goswami <bgoswami(a)codeaurora.org> --- include/linux/mod_devicetable.h | 2 +- include/uapi/linux/input-event-codes.h | 3 ++- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/linux/mod_devicetable.h b/include/linux/mod_devicetable.h index 448621c..7586099 100644 --- a/include/linux/mod_devicetable.h +++ b/include/linux/mod_devicetable.h @@ -299,7 +299,7 @@ struct pcmcia_device_id { #define INPUT_DEVICE_ID_LED_MAX 0x0f #define INPUT_DEVICE_ID_SND_MAX 0x07 #define INPUT_DEVICE_ID_FF_MAX 0x7f -#define INPUT_DEVICE_ID_SW_MAX 0x0f +#define INPUT_DEVICE_ID_SW_MAX 0x1f #define INPUT_DEVICE_ID_PROP_MAX 0x1f #define INPUT_DEVICE_ID_MATCH_BUS 1 diff --git a/include/uapi/linux/input-event-codes.h b/include/uapi/linux/input-event-codes.h index 85387c7..960fa86 100644 --- a/include/uapi/linux/input-event-codes.h +++ b/include/uapi/linux/input-event-codes.h @@ -808,7 +808,8 @@ #define SW_LINEIN_INSERT 0x0d /* set = inserted */ #define SW_MUTE_DEVICE 0x0e /* set = device disabled */ #define SW_PEN_INSERTED 0x0f /* set = pen inserted */ -#define SW_MAX 0x0f +#define SW_UNSUPPORT_INSERT 0x10 /* set = unsupported device inserted */ +#define SW_MAX 0x1f #define SW_CNT (SW_MAX+1) /* -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] What to do when a device clears the buffer in the start trigger?
by Paul Pawlowski 22 Aug '19

22 Aug '19

Hello, I have a device which clears the DMA buffer when I command it to start the playback. How can this be workarounded? I found the SNDRV_PCM_INFO_DOUBLE flag, but it doesn't seem that it changes alsa/alsa-lib behaviour. I wasn't able to find any drivers which shared this quirk. I'd like to avoid double buffering if possible, as once the stream is actually started it's not required. Thank you, Paul Pawlowski

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[alsa-devel] [PATCH v1] ASoC: Intel: Skylake: Print constant literals from format specifier
by Andy Shevchenko 15 Aug '19

15 Aug '19

Instead of using two additional "%s" specifiers, put the constant string literals directly to the format specifier. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/skylake/skl-sst.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 13c636dece56..f3e1399450db 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -421,8 +421,7 @@ static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid) int ret = 0; char mod_name[64]; /* guid str = 32 chars + 4 hyphens */ - snprintf(mod_name, sizeof(mod_name), "%s%pUL%s", - "intel/dsp_fw_", guid, ".bin"); + snprintf(mod_name, sizeof(mod_name), "intel/dsp_fw_%pUL.bin", guid); module_entry = skl_module_get_from_id(ctx, mod_id); if (module_entry == NULL) { -- 2.20.1

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