August 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] sound: usb: get lock before calling usb_enable_autosuspend()
by Salah Triki 03 Aug '21

03 Aug '21

Based on the documentation of usb_enable_autosuspend(), the caller must hold udev's device lock. Signed-off-by: Salah Triki <salah.triki(a)gmail.com> --- sound/usb/card.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 2f6a62416c05..9290ba0acd5f 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -907,8 +907,13 @@ static void usb_audio_disconnect(struct usb_interface *intf) } } - if (chip->quirk_type & QUIRK_SETUP_DISABLE_AUTOSUSPEND) - usb_enable_autosuspend(interface_to_usbdev(intf)); + if (chip->quirk_type & QUIRK_SETUP_DISABLE_AUTOSUSPEND) { + struct usb_device *udev = interface_to_usbdev(intf); + + usb_lock_device(udev); + usb_enable_autosuspend(udev); + usb_unlock_device(udev); + } chip->num_interfaces--; if (chip->num_interfaces <= 0) { -- 2.25.1

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dmix optimization
by Giuliano Zannetti - ART S.p.A. 03 Aug '21

03 Aug '21

Hi, I'm trying to optimize the dmix because I'm working with a big number of channels (up to 16) and in this case the dmix has not a negligible impact on performance. I'm working with ALSA 1.1.9. I gave my first look to the generic_mix_areas_16_native function (https://github.com/alsa-project/alsa-lib/blob/v1.1.9/src/pcm/pcm_dmix_gener…). I would ask you if I can avoid to check, for each loop iteration, if the current dst sample is not 0. for (;;) { sample = *src; if (! *dst) { *sum = sample; *dst = *src; } else { sample += *sum; *sum = sample; if (sample > 0x7fff) sample = 0x7fff; else if (sample < -0x8000) sample = -0x8000; *dst = sample; } if (!--size) return; src = (signed short *) ((char *)src + src_step); dst = (signed short *) ((char *)dst + dst_step); sum = (signed int *) ((char *)sum + sum_step); } Could it be possible check for the first sample of the period only, as reported in the code below? My assumption is that if dst[0] is 0 also dst[1] ... dst[period-1] will be 0, and I don't need to check every time. This is already an optimization, but it could be also a starting point for other optimization based on my HW. But, first of all, I would ask to you if my assumption is right. if (! *dst) { for (;;) { sample = *src; *sum = sample; *dst = *src; if (!--size) return; src = (signed short *) ((char *)src + src_step); dst = (signed short *) ((char *)dst + dst_step); sum = (signed int *) ((char *)sum + sum_step); } } else { for (;;) { sample = *src; sample += *sum; *sum = sample; if (sample > 0x7fff) sample = 0x7fff; else if (sample < -0x8000) sample = -0x8000; *dst = sample; if (!--size) return; src = (signed short *) ((char *)src + src_step); dst = (signed short *) ((char *)dst + dst_step); sum = (signed int *) ((char *)sum + sum_step); } } Thank you! Best Regards, Giuliano

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[PATCH v3 0/4] ASoC: codecs: wcd934x: add Headset and button detection support
by Srinivas Kandagatla 03 Aug '21

03 Aug '21

This patchset adds support to MBHC(Multi Button Headset Control) block found in Qualcomm WCD codecs. MBHC support headset type detection, both Mechanical and electrical insert/removal detection along with 8 buttons detection, Over current interrupts on HPHL/R, Impedance Measurements on HPHL/R. Eventhough MBHC block supports things like OverCurrent detection, Currently its reported as a kernel debug message. Should this be reported as an uevent to userspace? like the way USB reports? Any suggestions? First patch adds a common mbhc driver and the second one wcd934x specific driver changes along with sdm845 soundcard related changes. Common wcd-mbhc-v2 driver should be reusable across multiple codecs like WCD9335, WCD934x, WCD937x and WCD938x. Most of the work is derived from downstream Qualcomm kernels. Credits to various Qualcomm authors from Patrick Lai's team who have contributed to this code. Changes since v2: - switched to EXPORT_SYMBOL_GPL from EXPORT_SYMBOL - converted one of the if else to switch case. Srinivas Kandagatla (4): ASoC: dt-bindings: wcd934x: add bindings for Headset Button detection ASoC: codecs: wcd: add multi button Headset detection support ASoC: codecs: wcd934x: add mbhc support ASoC: qcom: sdm845: add jack support for WCD934x .../bindings/sound/qcom,wcd934x.yaml | 30 + include/linux/mfd/wcd934x/registers.h | 57 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-mbhc-v2.c | 1475 +++++++++++++++++ sound/soc/codecs/wcd-mbhc-v2.h | 340 ++++ sound/soc/codecs/wcd934x.c | 884 +++++++++- sound/soc/qcom/sdm845.c | 8 + 8 files changed, 2785 insertions(+), 15 deletions(-) create mode 100644 sound/soc/codecs/wcd-mbhc-v2.c create mode 100644 sound/soc/codecs/wcd-mbhc-v2.h -- 2.21.0

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[PATCH v2 0/6] soundwire/ASoC: abstract platform-dependent bases
by Bard Liao 02 Aug '21

02 Aug '21

shim base and alh base are platform-dependent. This series suggests to use variables for those bases. It allows us to use different bases for new platforms. v2: - Update the commit message of "soundwire: move intel sdw register definitions to sdw_intel.h" Bard Liao (6): soundwire: move intel sdw register definitions to sdw_intel.h ASoC: SOF: intel: add sdw_shim/alh_base to sof_intel_dsp_desc ASoC: SOF: intel: hda: remove HDA_DSP_REG_SNDW_WAKE_STS definition ASoC: SOF: intel: move sof_intel_dsp_desc() forward ASoC: SOF: intel: add snd_sof_dsp_check_sdw_irq ops soundwire: intel: introduce shim and alh base drivers/soundwire/intel.c | 74 ------------------------ drivers/soundwire/intel_init.c | 14 ++--- include/linux/soundwire/sdw_intel.h | 87 +++++++++++++++++++++++++++++ sound/soc/sof/intel/cnl.c | 6 ++ sound/soc/sof/intel/hda.c | 39 ++++++++----- sound/soc/sof/intel/hda.h | 8 ++- sound/soc/sof/intel/icl.c | 3 + sound/soc/sof/intel/shim.h | 3 + sound/soc/sof/intel/tgl.c | 12 ++++ 9 files changed, 149 insertions(+), 97 deletions(-) -- 2.17.1

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[PATCH] ASoC: tlv320aic32x4: make array clocks static, makes object smaller
by Colin King 02 Aug '21

02 Aug '21

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the array clocks on the stack but instead it static. Makes the object code smaller by 316 bytes. Before: text data bss dec hex filename 63668 28264 0 91932 1671c ./sound/soc/codecs/tlv320aic32x4.o After: text data bss dec hex filename 62616 29000 0 91616 165e0 ./sound/soc/codecs/tlv320aic32x4.o (gcc version 10.2.0) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/tlv320aic32x4.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 254a016cb1f3..d31ffb6d5b4b 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -714,7 +714,7 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, unsigned long adc_clock_rate, dac_clock_rate; int ret; - struct clk_bulk_data clocks[] = { + static struct clk_bulk_data clocks[] = { { .id = "pll" }, { .id = "nadc" }, { .id = "madc" }, @@ -878,7 +878,7 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component, { int ret; - struct clk_bulk_data clocks[] = { + static struct clk_bulk_data clocks[] = { { .id = "madc" }, { .id = "mdac" }, { .id = "bdiv" }, @@ -994,7 +994,7 @@ static int aic32x4_component_probe(struct snd_soc_component *component) u32 tmp_reg; int ret; - struct clk_bulk_data clocks[] = { + static struct clk_bulk_data clocks[] = { { .id = "codec_clkin" }, { .id = "pll" }, { .id = "bdiv" }, @@ -1144,7 +1144,7 @@ static int aic32x4_tas2505_component_probe(struct snd_soc_component *component) u32 tmp_reg; int ret; - struct clk_bulk_data clocks[] = { + static struct clk_bulk_data clocks[] = { { .id = "codec_clkin" }, { .id = "pll" }, { .id = "bdiv" }, -- 2.31.1

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[PATCH v2 0/6] ASoC: Intel: bytcr_rt5640: Fix HP ElitePad 1000 G2 audio routing
by Hans de Goede 02 Aug '21

02 Aug '21

Changes in v2: - Only set lineout_string if BYT_RT5640_LINEOUT is set, since BYT_RT5640_LINEOUT_AS_HP2 only works if the lineout is enabled in the first place Original cover-letter: The HP Elitepad 1000 G2 has 2 headset jacks: 1. on the dock which uses the output of the codecs built-in HP-amp + the standard IN2 input which is always used with the headset-jack. 2. on the tablet itself, this uses the line-out of the codec, combined with an external HP-amp + IN1 for the headset-mic. This series adds support for this, resolving: https://bugzilla.kernel.org/show_bug.cgi?id=213415 Note this series does not add jack-detect support. I plan to add that with a follow-up series when I can make some time to implement that. Regards, Hans Hans de Goede (6): ASoC: Intel: bytcr_rt5640: Move "Platform Clock" routes to the maps for the matching in-/output ASoC: Intel: bytcr_rt5640: Add line-out support ASoC: Intel: bytcr_rt5640: Add a byt_rt5640_get_codec_dai() helper ASoC: Intel: bytcr_rt5640: Add support for a second headphones output ASoC: Intel: bytcr_rt5640: Add support for a second headset mic input ASoC: Intel: bytcr_rt5640: Fix HP ElitePad 1000 G2 quirk sound/soc/intel/boards/bytcr_rt5640.c | 118 ++++++++++++++++++++++---- 1 file changed, 102 insertions(+), 16 deletions(-) -- 2.31.1

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Linux daemon +ALSA+pulseaudio problem
by Valery Savchuk 02 Aug '21

02 Aug '21

Hello! I am developing RadioAmateur software AlterFRN-client http://alterfrn.ucoz.ru/ which is using ALSA on Linux. The problem is: the AlterFRN-client daemon (background mode) can not initialize ALSA if pusleaudio is used/running also. In foreground mode all things are OK. AlsaInfo is attached also.

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[PATCH] ASoC: rsnd: make some arrays static const, makes object smaller
by Colin King 02 Aug '21

02 Aug '21

From: Colin Ian King <colin.king(a)canonical.com> Don't populate arrays on the stack but instead them static const. Makes the object code smaller by 242 bytes. Before: text data bss dec hex filename 23827 8764 0 32591 7f4f ./sound/soc/sh/rcar/ssi.o After: text data bss dec hex filename 23361 8988 0 32349 7e5d ./sound/soc/sh/rcar/ssi.o gcc version 10.2.0) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sh/rcar/ssi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 27f34ca6059d..87e606f688d3 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -165,7 +165,7 @@ static void rsnd_ssi_status_check(struct rsnd_mod *mod, static u32 rsnd_ssi_multi_secondaries(struct rsnd_dai_stream *io) { - enum rsnd_mod_type types[] = { + static const enum rsnd_mod_type types[] = { RSND_MOD_SSIM1, RSND_MOD_SSIM2, RSND_MOD_SSIM3, @@ -228,7 +228,7 @@ unsigned int rsnd_ssi_clk_query(struct rsnd_dai *rdai, int param1, int param2, int *idx) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); - int ssi_clk_mul_table[] = { + static const int ssi_clk_mul_table[] = { 1, 2, 4, 8, 16, 6, 12, }; int j, ret; @@ -1079,7 +1079,7 @@ static void rsnd_ssi_connect(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { struct rsnd_dai *rdai = rsnd_io_to_rdai(io); - enum rsnd_mod_type types[] = { + static const enum rsnd_mod_type types[] = { RSND_MOD_SSI, RSND_MOD_SSIM1, RSND_MOD_SSIM2, -- 2.31.1

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[PATCH v2] ASoC: amd: Fix reference to PCM buffer address
by Takashi Iwai 02 Aug '21

02 Aug '21

PCM buffers might be allocated dynamically when the buffer preallocation failed or a larger buffer is requested, and it's not guaranteed that substream->dma_buffer points to the actually used buffer. The driver needs to refer to substream->runtime->dma_addr instead for the buffer address. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Fix build error of renoir driver sound/soc/amd/acp-pcm-dma.c | 2 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 143155a840ac..cc1ce6f22caa 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -969,7 +969,7 @@ static int acp_dma_hw_params(struct snd_soc_component *component, acp_set_sram_bank_state(rtd->acp_mmio, 0, true); /* Save for runtime private data */ - rtd->dma_addr = substream->dma_buffer.addr; + rtd->dma_addr = runtime->dma_addr; rtd->order = get_order(size); /* Fill the page table entries in ACP SRAM */ diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 8148b0d22e88..597d7c4b2a6b 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -286,7 +286,7 @@ static int acp3x_dma_hw_params(struct snd_soc_component *component, pr_err("pinfo failed\n"); } size = params_buffer_bytes(params); - rtd->dma_addr = substream->dma_buffer.addr; + rtd->dma_addr = substream->runtime->dma_addr; rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); config_acp3x_dma(rtd, substream->stream); return 0; diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index bd20622b0933..0391c28dd078 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -242,7 +242,7 @@ static int acp_pdm_dma_hw_params(struct snd_soc_component *component, return -EINVAL; size = params_buffer_bytes(params); period_bytes = params_period_bytes(params); - rtd->dma_addr = substream->dma_buffer.addr; + rtd->dma_addr = substream->runtime->dma_addr; rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); config_acp_dma(rtd, substream->stream); init_pdm_ring_buffer(MEM_WINDOW_START, size, period_bytes, -- 2.26.2

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[PATCH 0/5] ASoC: soc-topology: cleanup cppcheck warning
by Kuninori Morimoto 02 Aug '21

02 Aug '21

Hi Mark Now I'm posting audio-graph-card2 patch-set, but it seems it needs longer discussion. Thus I want to post more easy patch first, and reduce my local patches. These are cppcheck warning cleanup patches for soc-topology. Kuninori Morimoto (5): ASoC: soc-topology: cleanup cppcheck warning at soc_tplg_process_headers() ASoC: soc-topology: cleanup cppcheck warning at soc_tplg_dai_elems_load() ASoC: soc-topology: cleanup cppcheck warning at soc_tplg_dapm_widget_elems_load() ASoC: soc-topology: cleanup cppcheck warning at soc_tplg_kcontrol_elems_load() ASoC: soc-topology: cleanup cppcheck warning at snd_soc_find_dai_link() sound/soc/soc-topology.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) -- 2.25.1

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