January 2022 - Alsa-devel - mailman.alsa-project.org

slow snd_rawmidi_drain() for VirMidi devcies
by Stefan Sauer 07 Jan '22

07 Jan '22

hi, I've tried to link BitwigStudio to the webapp cables.gl over virmidi. Unfortunately Bitwig Studio only supports rawmidi. What I discovered is that there is a strange slowness when sending data to virmidi caused by snd_rawmidi_drain(). I've posted two tiny, self-contained c apps to: https://gist.github.com/ensonic/c7588b87fa6c1fa94a8f753b1e0aa394 See some examples below. 2 observations: * snd_rawmidi_type() is *not* reporting virmidi as VIRTUAL * snd_rawmidi_drain() takes about 60ms! on virtual vs. less that 0.1 ms on usb midi (I checked all my hw midi and the worst was avg=1ms on physical midi image unitor8) When comparing the implementations: https://github.com/alsa-project/alsa-lib/blob/master/src/rawmidi/rawmidi_vi… https://github.com/alsa-project/alsa-lib/blob/master/src/rawmidi/rawmidi_hw… I see that the hw one results in an IOCTL which I can see when striking the code and I wonder if this is the root cause? Why is rawmidi_virt.c not used for virmidi? >From poking at snd_rawmidi_open_conf() I have not yet figured where this is decided .... Stefan > amidi -l Dir Device Name IO hw:0,0,0 Scarlett 18i20 USB MIDI 1 IO hw:3,0,0 nanoKEY2 nanoKEY2 _ KEYBOARD IO hw:5,0,0 nanoKONTROL nanoKONTROL _ SLIDE IO hw:10,0 Virtual Raw MIDI (16 subdevices) IO hw:11,0 Virtual Raw MIDI (16 subdevices) # using direct i/o to virmidi - all good > ./rawmidi_oss /dev/midi11 0 Using device '/dev/midi11' without draining write took min= 0.0015 ms, avg= 0.0016 ms, max= 0.0110 ms > ./rawmidi_oss /dev/midi11 1 Using device '/dev/midi11' with draining write took min= 0.0015 ms, avg= 0.0017 ms, max= 0.0101 ms drain took min= 0.0001 ms, avg= 0.0001 ms, max= 0.0008 ms # using snd_rawmidi to virmidi - slow drain operations > ./rawmidi_alsa hw:11,0 0 Using device 'hw:11,0' without draining SND_RAWMIDI_TYPE_HW write took min= 0.0010 ms, avg= 0.0011 ms, max= 0.0056 ms > ./rawmidi_alsa hw:11,0 1 Using device 'hw:11,0' with draining SND_RAWMIDI_TYPE_HW write took min= 0.0016 ms, avg= 0.0040 ms, max= 0.0077 ms drain took min= 55.9951 ms, avg= 60.4330 ms, max= 64.0653 ms # using snd_rawmidi to usb hw - all good > ./rawmidi_alsa hw:3,0 0 Using device 'hw:3,0' without draining SND_RAWMIDI_TYPE_HW write took min= 0.0012 ms, avg= 0.0015 ms, max= 0.0121 ms > ./rawmidi_alsa hw:3,0 1 Using device 'hw:3,0' with draining SND_RAWMIDI_TYPE_HW write took min= 0.0024 ms, avg= 0.0032 ms, max= 0.0110 ms drain took min= 0.0293 ms, avg= 0.0636 ms, max= 0.2277 ms

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[PATCH 0/2] Add low power hibernation support to cs35l41
by Charles Keepax 07 Jan '22

07 Jan '22

This patch series adds support for the low power hibernation feature on cs35l41. This allows the DSP memory to be retained whilst the device enters a very low power state. These patches will cause some very minor conflicts with Lucas's currently outstanding work on the HDA version of cs35l41. Whilst things will still build (well now I fixed my silly mistake), this patch adds a test key function his code will now have to call. It looks like Lucas's patch might get a respin based on the comments from Andy, in which case I guess we can pull the small change into the next version of it. But either way these patches are not urgent so I am happy if they sit around until Lucas's stuff is merged too. Thanks, Charles Charles Keepax (2): ASoC: cs35l41: Update handling of test key registers ASoC: cs35l41: Add support for hibernate memory retention mode include/sound/cs35l41.h | 7 ++ sound/soc/codecs/cs35l41-i2c.c | 1 + sound/soc/codecs/cs35l41-lib.c | 73 +++++++------ sound/soc/codecs/cs35l41-spi.c | 1 + sound/soc/codecs/cs35l41.c | 233 +++++++++++++++++++++++++++++++++++++---- sound/soc/codecs/cs35l41.h | 4 + 6 files changed, 268 insertions(+), 51 deletions(-) -- 2.11.0

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[PATCH 0/5] ASoC: Xilinx fixes
by Robert Hancock 07 Jan '22

07 Jan '22

There are drivers in mainline for the Xilinx Audio Formatter and Xilinx I2S IP cores. However, because of a few issues, these were only really usable with Xilinx's xlnx_pl_snd_card top-level driver, which is not in mainline (and not suitable for mainline). The fixes in this patchset, for the simple-card layer as well as the Xilinx drivers, now allow these drivers to be properly used with simple-card without any out-of-tree support code. Robert Hancock (5): ASoC: xilinx: xlnx_formatter_pcm: Make buffer bytes multiple of period bytes ASoC: xilinx: xlnx_formatter_pcm: Handle sysclk setting ASoC: xilinx: xlnx_i2s.c: Handle sysclk setting ASoC: simple-card: fix probe failure on platform component ASoC: simple-card-utils: Set sysclk on all components sound/soc/generic/simple-card-utils.c | 11 +++ sound/soc/generic/simple-card.c | 52 +++++++------ sound/soc/xilinx/xlnx_formatter_pcm.c | 44 ++++++++++- sound/soc/xilinx/xlnx_i2s.c | 104 +++++++++++++++++--------- 4 files changed, 150 insertions(+), 61 deletions(-) -- 2.31.1

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[PATCH] ALSA: hda: Fix dependency on ASoC cs35l41 codec
by Takashi Iwai 07 Jan '22

07 Jan '22

The recently added support for CS35L41 codec unconditionally selects CONFIG_SND_SOC_CS35L41_LIB, but this can't work unless the top-level CONFIG_SND_SOC is enabled. This patch adds the proper dependency. Fixes: 7b2f3eb492da ("ALSA: hda: cs35l41: Add support for CS35L41 in HDA systems") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 84cefc006f29..68effb74866c 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -97,6 +97,7 @@ config SND_HDA_SCODEC_CS35L41 config SND_HDA_SCODEC_CS35L41_I2C tristate "Build CS35L41 HD-audio side codec support for I2C Bus" depends on ACPI + depends on SND_SOC select SND_HDA_GENERIC select SND_SOC_CS35L41_LIB select SND_HDA_SCODEC_CS35L41 @@ -110,6 +111,7 @@ comment "Set to Y if you want auto-loading the side codec driver" config SND_HDA_SCODEC_CS35L41_SPI tristate "Build CS35L41 HD-audio codec support for SPI Bus" depends on ACPI + depends on SND_SOC select SND_HDA_GENERIC select SND_SOC_CS35L41_LIB select SND_HDA_SCODEC_CS35L41 -- 2.31.1

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[PATCH v3] ALSA: intel_hdmi: Check for error num after setting mask
by Jiasheng Jiang 07 Jan '22

07 Jan '22

To maintain the consistency of the code, it should be better to add the sanity check after calling dma_set_mask_and_coherent(), like tegra_pcm_dma_allocate() in `sound/soc/tegra/tegra_pcm.c`. Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- Changelog v2 -> v3 * Change 1. Remove the fixes tag and change the message. * Change 2. Correct the patch to fit the lastest code. --- sound/x86/intel_hdmi_audio.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 378826312abe..1c94eaff1931 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1750,7 +1750,9 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) card_ctx->irq = irq; /* only 32bit addressable */ - dma_set_mask_and_coherent(&pdev->dev, DMA_BIT_MASK(32)); + ret = dma_set_mask_and_coherent(&pdev->dev, DMA_BIT_MASK(32)); + if (ret) + return ret; init_channel_allocations(); -- 2.25.1

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Re: Re: [PATCH v2] ALSA: intel_hdmi: Check for error num after setting maski
by Jiasheng Jiang 07 Jan '22

07 Jan '22

On Thu, Jan 06, 2022 at 10:58:20PM +0800, Takashi Iwai wrote: > The build fails with the latest code, unfortunately. > This function simply returns an error, as it was changed to a devres > allocation some time ago. Thanks, I have already send a v3 fit for the latest code. But I still think the v2 is useful for the previous code. Actually, this one is suitable for the linux-stable-5.13.9. Therefore, I believe it might be better to apply the v2 to the previous version in order to maintain the consistency. Sincerely thanks, Jiang

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[PATCH] ASoC: wcd9335: Keep a RX port value for each SLIM RX mux
by Yassine Oudjana 06 Jan '22

06 Jan '22

Currently, rx_port_value is a single unsigned int that gets overwritten when slim_rx_mux_put() is called for any RX mux, then the same value is read when slim_rx_mux_get() is called for any of them. This results in slim_rx_mux_get() reporting the last value set by slim_rx_mux_put() regardless of which SLIM RX mux is in question. Turn rx_port_value into an array and store a separate value for each SLIM RX mux. Signed-off-by: Yassine Oudjana <y.oudjana(a)protonmail.com> --- sound/soc/codecs/wcd9335.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index bc5d68c53e5a..1e60db4056ad 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -341,7 +341,7 @@ struct wcd9335_codec { int reset_gpio; struct regulator_bulk_data supplies[WCD9335_MAX_SUPPLY]; - unsigned int rx_port_value; + unsigned int rx_port_value[WCD9335_RX_MAX]; unsigned int tx_port_value; int hph_l_gain; int hph_r_gain; @@ -1269,10 +1269,11 @@ static const struct snd_kcontrol_new sb_tx8_mux = static int slim_rx_mux_get(struct snd_kcontrol *kc, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kc); - struct wcd9335_codec *wcd = dev_get_drvdata(dapm->dev); + struct snd_soc_dapm_widget *w = snd_soc_dapm_kcontrol_widget(kc); + struct wcd9335_codec *wcd = dev_get_drvdata(w->dapm->dev); + u32 port_id = w->shift; - ucontrol->value.enumerated.item[0] = wcd->rx_port_value; + ucontrol->value.enumerated.item[0] = wcd->rx_port_value[port_id]; return 0; } @@ -1286,9 +1287,9 @@ static int slim_rx_mux_put(struct snd_kcontrol *kc, struct snd_soc_dapm_update *update = NULL; u32 port_id = w->shift; - wcd->rx_port_value = ucontrol->value.enumerated.item[0]; + wcd->rx_port_value[port_id] = ucontrol->value.enumerated.item[0]; - switch (wcd->rx_port_value) { + switch (wcd->rx_port_value[port_id]) { case 0: list_del_init(&wcd->rx_chs[port_id].list); break; @@ -1309,11 +1310,11 @@ static int slim_rx_mux_put(struct snd_kcontrol *kc, &wcd->dai[AIF4_PB].slim_ch_list); break; default: - dev_err(wcd->dev, "Unknown AIF %d\n", wcd->rx_port_value); + dev_err(wcd->dev, "Unknown AIF %d\n", wcd->rx_port_value[port_id]); goto err; } - snd_soc_dapm_mux_update_power(w->dapm, kc, wcd->rx_port_value, + snd_soc_dapm_mux_update_power(w->dapm, kc, wcd->rx_port_value[port_id], e, update); return 0; -- 2.34.1

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[PATCH] ASoC: amd: acp: acp-mach: Change default RT1019 amp dev id
by Ajit Kumar Pandey 06 Jan '22

06 Jan '22

RT1019 components was initially registered with i2c1 and i2c2 but now changed to i2c0 and i2c1 in most of our AMD platforms. Change default rt1019 components to 10EC1019:00 and 10EC1019:01 which is aligned with most of AMD machines. Any exception to rt1019 device ids in near future board design can be handled using dmi based quirk for that machine. Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> --- sound/soc/amd/acp/acp-mach-common.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 03d8d1af14b3..c9caade5cb74 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -293,8 +293,8 @@ static const struct snd_soc_ops acp_card_rt5682s_ops = { /* Declare RT1019 codec components */ SND_SOC_DAILINK_DEF(rt1019, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC1019:01", "rt1019-aif"), - COMP_CODEC("i2c-10EC1019:02", "rt1019-aif"))); + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC1019:00", "rt1019-aif"), + COMP_CODEC("i2c-10EC1019:01", "rt1019-aif"))); static const struct snd_soc_dapm_route rt1019_map_lr[] = { { "Left Spk", NULL, "Left SPO" }, @@ -303,11 +303,11 @@ static const struct snd_soc_dapm_route rt1019_map_lr[] = { static struct snd_soc_codec_conf rt1019_conf[] = { { - .dlc = COMP_CODEC_CONF("i2c-10EC1019:01"), + .dlc = COMP_CODEC_CONF("i2c-10EC1019:00"), .name_prefix = "Left", }, { - .dlc = COMP_CODEC_CONF("i2c-10EC1019:02"), + .dlc = COMP_CODEC_CONF("i2c-10EC1019:01"), .name_prefix = "Right", }, }; -- 2.25.1

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[PATCH 0/3] ASoC: imx-card: several improvement and fixes
by Shengjiu Wang 06 Jan '22

06 Jan '22

Several improvement and fixes for AK codecs supported on i.MX platfroms Shengjiu Wang (3): ASoC: imx-card: Need special setting for ak4497 on i.MX8MQ ASoC: imx-card: Fix mclk calculation issue for akcodec ASoC: imx-card: improve the sound quality for low rate sound/soc/fsl/imx-card.c | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) -- 2.17.1

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[PATCH] ASoC: fsl_asrc: refine the check of available clock divider
by Shengjiu Wang 06 Jan '22

06 Jan '22

According to RM, the clock divider range is from 1 to 8, clock prescaling ratio may be any power of 2 from 1 to 128. So the supported divider is not all the value between 1 and 1024, just limited value in that range. Create table for the supported divder and add function to check the clock divider is available by comparing with the table. Fixes: d0250cf4f2ab ("ASoC: fsl_asrc: Add an option to select internal ratio mode") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc.c | 69 +++++++++++++++++++++++++++++++++------- 1 file changed, 58 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 24b41881a68f..d7d1536a4f37 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -19,6 +19,7 @@ #include "fsl_asrc.h" #define IDEAL_RATIO_DECIMAL_DEPTH 26 +#define DIVIDER_NUM 64 #define pair_err(fmt, ...) \ dev_err(&asrc->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) @@ -101,6 +102,55 @@ static unsigned char clk_map_imx8qxp[2][ASRC_CLK_MAP_LEN] = { }, }; +/* + * According to RM, the divider range is 1 ~ 8, + * prescaler is power of 2 from 1 ~ 128. + */ +static int asrc_clk_divider[DIVIDER_NUM] = { + 1, 2, 4, 8, 16, 32, 64, 128, /* divider = 1 */ + 2, 4, 8, 16, 32, 64, 128, 256, /* divider = 2 */ + 3, 6, 12, 24, 48, 96, 192, 384, /* divider = 3 */ + 4, 8, 16, 32, 64, 128, 256, 512, /* divider = 4 */ + 5, 10, 20, 40, 80, 160, 320, 640, /* divider = 5 */ + 6, 12, 24, 48, 96, 192, 384, 768, /* divider = 6 */ + 7, 14, 28, 56, 112, 224, 448, 896, /* divider = 7 */ + 8, 16, 32, 64, 128, 256, 512, 1024, /* divider = 8 */ +}; + +/* + * Check if the divider is available for internal ratio mode + */ +static bool fsl_asrc_divider_avail(int clk_rate, int rate, int *div) +{ + u32 rem, i; + u64 n; + + if (div) + *div = 0; + + if (clk_rate == 0 || rate == 0) + return false; + + n = clk_rate; + rem = do_div(n, rate); + + if (div) + *div = n; + + if (rem != 0) + return false; + + for (i = 0; i < DIVIDER_NUM; i++) { + if (n == asrc_clk_divider[i]) + break; + } + + if (i == DIVIDER_NUM) + return false; + + return true; +} + /** * fsl_asrc_sel_proc - Select the pre-processing and post-processing options * @inrate: input sample rate @@ -330,12 +380,12 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool use_ideal_rate) enum asrc_word_width input_word_width; enum asrc_word_width output_word_width; u32 inrate, outrate, indiv, outdiv; - u32 clk_index[2], div[2], rem[2]; + u32 clk_index[2], div[2]; u64 clk_rate; int in, out, channels; int pre_proc, post_proc; struct clk *clk; - bool ideal; + bool ideal, div_avail; if (!config) { pair_err("invalid pair config\n"); @@ -415,8 +465,7 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool use_ideal_rate) clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; clk_rate = clk_get_rate(clk); - rem[IN] = do_div(clk_rate, inrate); - div[IN] = (u32)clk_rate; + div_avail = fsl_asrc_divider_avail(clk_rate, inrate, &div[IN]); /* * The divider range is [1, 1024], defined by the hardware. For non- @@ -425,7 +474,7 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool use_ideal_rate) * only result in different converting speeds. So remainder does not * matter, as long as we keep the divider within its valid range. */ - if (div[IN] == 0 || (!ideal && (div[IN] > 1024 || rem[IN] != 0))) { + if (div[IN] == 0 || (!ideal && !div_avail)) { pair_err("failed to support input sample rate %dHz by asrck_%x\n", inrate, clk_index[ideal ? OUT : IN]); return -EINVAL; @@ -436,13 +485,12 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool use_ideal_rate) clk = asrc_priv->asrck_clk[clk_index[OUT]]; clk_rate = clk_get_rate(clk); if (ideal && use_ideal_rate) - rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); + div_avail = fsl_asrc_divider_avail(clk_rate, IDEAL_RATIO_RATE, &div[OUT]); else - rem[OUT] = do_div(clk_rate, outrate); - div[OUT] = clk_rate; + div_avail = fsl_asrc_divider_avail(clk_rate, outrate, &div[OUT]); /* Output divider has the same limitation as the input one */ - if (div[OUT] == 0 || (!ideal && (div[OUT] > 1024 || rem[OUT] != 0))) { + if (div[OUT] == 0 || (!ideal && !div_avail)) { pair_err("failed to support output sample rate %dHz by asrck_%x\n", outrate, clk_index[OUT]); return -EINVAL; @@ -621,8 +669,7 @@ static void fsl_asrc_select_clk(struct fsl_asrc_priv *asrc_priv, clk_index = asrc_priv->clk_map[j][i]; clk_rate = clk_get_rate(asrc_priv->asrck_clk[clk_index]); /* Only match a perfect clock source with no remainder */ - if (clk_rate != 0 && (clk_rate / rate[j]) <= 1024 && - (clk_rate % rate[j]) == 0) + if (fsl_asrc_divider_avail(clk_rate, rate[j], NULL)) break; } -- 2.17.1

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