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

[PATCH 0/1] sound: rawmidi: Add framing mode
by David Henningsson 25 Mar '21

25 Mar '21

Hi, When writing an application that records midi (and e g saves it to disk), ultra-low latency isn't really needed, all we need to know is exactly when the midi message came in. The application can then wake up once a second or so, to write the incoming data, including an accurate timestamp for every event, to disk. As far as I can see, the rawmidi interface does not support such a feature at all. There is a snd_rawmidi_status syscall, but its timestamp field is not even filled by the kernel (!). But even if that was fixed, it would not fix the problem as there could be several midi events in the buffer with different timestamps. You could use the seq interface, it does support timestamps, but I can see at least two potential problems with this: First, the seq code runs in a work queue, not in the actual IRQ. This means that midi event is timestamped too late, especially so if the work is delayed for some reason. Second, seq hard-codes the timestamp type to monotonic - there is no monotonic_raw, so the timestamp would be affected by NTP changes. Also, the timespec uses 32-bit for sec and nsec, but I suspect this is less of a problem (unless people constantly record midi for sixty years or so...). So here's a patch that adds a new "framing" mode that stuffs all MIDI data into 16 byte frames, 8 bytes of timestamp, one byte length, up to seven bytes of data. I'll follow up with an alsa-lib patch if this gets merged. David Henningsson (1): sound: rawmidi: Add framing mode include/sound/rawmidi.h | 1 + include/uapi/sound/asound.h | 18 ++++++++++++++- sound/core/rawmidi.c | 45 ++++++++++++++++++++++++++++++++++++- 3 files changed, 62 insertions(+), 2 deletions(-) -- 2.25.1

2 4

[PATCH v3] sound: rawmidi: Add framing mode
by David Henningsson 25 Mar '21

25 Mar '21

This commit adds a new framing mode that frames all MIDI data into 16-byte frames with a timestamp. The main benefit is that we can get accurate timestamps even if userspace wakeup and processing is not immediate. Signed-off-by: David Henningsson <coding(a)diwic.se> --- Changes in v3: * Bumped protocol version * More checks for invalid combinations in set_params * Frame: Timestamp is 64 bit for seconds and 3 bytes for data * Sets clock and framing individually. (Clock_type is unsigned char instead of clockid_t to avoid padding issues.) * Supports both monotonic and monotonic_raw timestamps * Moved spin lock to after timestamp retreival. Both to decrease time spent in spin lock, and to get more accurate timestamp in case the spin lock is actually busy. include/sound/rawmidi.h | 2 ++ include/uapi/sound/asound.h | 21 +++++++++++-- sound/core/rawmidi.c | 63 +++++++++++++++++++++++++++++++++++-- 3 files changed, 82 insertions(+), 4 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 334842daa904..b0057a193c31 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -81,6 +81,8 @@ struct snd_rawmidi_substream { bool opened; /* open flag */ bool append; /* append flag (merge more streams) */ bool active_sensing; /* send active sensing when close */ + u8 framing; /* whether to frame input data */ + clockid_t clock_type; /* clock source to use for input framing */ int use_count; /* use counter (for output) */ size_t bytes; struct snd_rawmidi *rmidi; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 535a7229e1d9..a965a975aec3 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -710,7 +710,7 @@ enum { * Raw MIDI section - /dev/snd/midi?? */ -#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 1) +#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 2) enum { SNDRV_RAWMIDI_STREAM_OUTPUT = 0, @@ -736,12 +736,29 @@ struct snd_rawmidi_info { unsigned char reserved[64]; /* reserved for future use */ }; +enum { + SNDRV_RAWMIDI_FRAMING_NONE = 0, + SNDRV_RAWMIDI_FRAMING_TSTAMP, + SNDRV_RAWMIDI_FRAMING_LAST = SNDRV_RAWMIDI_FRAMING_TSTAMP, +}; + +#define SND_RAWMIDI_FRAMING_DATA_LENGTH 3 + +struct snd_rawmidi_framing_tstamp { + unsigned long long tv_sec; /* seconds */ + unsigned int tv_nsec; /* nanoseconds */ + unsigned char length; + unsigned char data[SND_RAWMIDI_FRAMING_DATA_LENGTH]; +}; + struct snd_rawmidi_params { int stream; size_t buffer_size; /* queue size in bytes */ size_t avail_min; /* minimum avail bytes for wakeup */ unsigned int no_active_sensing: 1; /* do not send active sensing byte in close() */ - unsigned char reserved[16]; /* reserved for future use */ + unsigned char framing; /* For input data only, frame incoming data */ + unsigned char clock_type; /* Type of clock to use for framing, same as clockid_t */ + unsigned char reserved[14]; /* reserved for future use */ }; #ifndef __KERNEL__ diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index aca00af93afe..269363286b3b 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -683,6 +683,8 @@ static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) return -EINVAL; + if (params->framing == SNDRV_RAWMIDI_FRAMING_TSTAMP && params->buffer_size & 0xf) + return -EINVAL; if (params->avail_min < 1 || params->avail_min > params->buffer_size) return -EINVAL; if (params->buffer_size != runtime->buffer_size) { @@ -720,7 +722,16 @@ EXPORT_SYMBOL(snd_rawmidi_output_params); int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params *params) { + if (params->framing) { + if (params->framing > SNDRV_RAWMIDI_FRAMING_LAST) + return -EINVAL; + /* framing requires a valid clock type */ + if (params->clock_type != CLOCK_MONOTONIC_RAW && params->clock_type != CLOCK_MONOTONIC) + return -EINVAL; + } snd_rawmidi_drain_input(substream); + substream->framing = params->framing; + substream->clock_type = params->clock_type; return resize_runtime_buffer(substream->runtime, params, true); } EXPORT_SYMBOL(snd_rawmidi_input_params); @@ -963,6 +974,45 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, return -ENOIOCTLCMD; } +static int receive_with_tstamp_framing(struct snd_rawmidi_substream *substream, + const unsigned char *buffer, int src_count, struct timespec64 *tstamp) +{ + struct snd_rawmidi_runtime *runtime = substream->runtime; + struct snd_rawmidi_framing_tstamp frame; + struct snd_rawmidi_framing_tstamp *dest_ptr; + + int dest_frames = 0; + int frame_size = sizeof(struct snd_rawmidi_framing_tstamp); + + frame.tv_sec = tstamp->tv_sec; + frame.tv_nsec = tstamp->tv_nsec; + + if (snd_BUG_ON(runtime->hw_ptr & 0xf || runtime->buffer_size & 0xf || frame_size != 0x10)) + return -EINVAL; + while (src_count > 0) { + if ((int)(runtime->buffer_size - runtime->avail) < frame_size) { + runtime->xruns += src_count; + return dest_frames * frame_size; + } + if (src_count >= SND_RAWMIDI_FRAMING_DATA_LENGTH) + frame.length = SND_RAWMIDI_FRAMING_DATA_LENGTH; + else { + frame.length = src_count; + memset(frame.data, 0, SND_RAWMIDI_FRAMING_DATA_LENGTH); + } + memcpy(frame.data, buffer, frame.length); + buffer += frame.length; + src_count -= frame.length; + dest_ptr = (struct snd_rawmidi_framing_tstamp *) (runtime->buffer + runtime->hw_ptr); + *dest_ptr = frame; + runtime->avail += frame_size; + runtime->hw_ptr += frame_size; + runtime->hw_ptr %= runtime->buffer_size; + dest_frames++; + } + return dest_frames * frame_size; +} + /** * snd_rawmidi_receive - receive the input data from the device * @substream: the rawmidi substream @@ -977,6 +1027,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count) { unsigned long flags; + struct timespec64 ts64; int result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -987,8 +1038,15 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, "snd_rawmidi_receive: input is not active!!!\n"); return -EINVAL; } - spin_lock_irqsave(&runtime->lock, flags); - if (count == 1) { /* special case, faster code */ + if (substream->framing == SNDRV_RAWMIDI_FRAMING_TSTAMP) { + if (substream->clock_type == CLOCK_MONOTONIC_RAW) + ktime_get_raw_ts64(&ts64); + else + ktime_get_ts64(&ts64); + spin_lock_irqsave(&runtime->lock, flags); + result = receive_with_tstamp_framing(substream, buffer, count, &ts64); + } else if (count == 1) { /* special case, faster code */ + spin_lock_irqsave(&runtime->lock, flags); substream->bytes++; if (runtime->avail < runtime->buffer_size) { runtime->buffer[runtime->hw_ptr++] = buffer[0]; @@ -999,6 +1057,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, runtime->xruns++; } } else { + spin_lock_irqsave(&runtime->lock, flags); substream->bytes += count; count1 = runtime->buffer_size - runtime->hw_ptr; if (count1 > count) -- 2.25.1

1 0

Patch for some Roland devices' USB digital audio
by Lucas 25 Mar '21

25 Mar '21

A couple days ago, I found this excellent informational post, and that Mike Oliphant's patch is in mainline now: http://blog.nostatic.org/2020/01/getting-boss-gt-1-effects-processor-to.html Since I was fairly certain it applies to the Roland devices I own, I added my USB IDs. Wonderfully, now USB audio capture seems to be working perfectly from the three devices I can test: the INTEGRA-7, Boutique D-05, & VG-99! I don't know if the D-05 and VG-99 have digital audio playback/output, as I only tried it with the INTEGRA-7. It does work, but only when the INTEGRA-7's "Sampling Rate" is set to "44.1 kHz" and "Ext Part Source Select" is set to "USB AUDIO". I searched a bit for other Roland devices people have had USB digital audio issues with over the years, and took the liberty of adding them to the patch too. Likely there are still many more, especially in the Boutique line, but I couldn't find their USB IDs. I hope this can be added to the mainline kernel too, as this support should help many that have struggled. I, certainly, have been hoping somebody would figure this out for quite a while, and even bought a USB through/monitoring device to try to develop it myself (sadly, the software supporting that has its own bugs, so I probably should've just used wireshark). Suffice it to say, I'm quite happy to have this figured out, and to not have to delve into stuff I know nothing about! :) Thanks, Lucas

1 0

[PATCH v5 0/6] Add audio driver base on rpmsg on i.MX platform
by Shengjiu Wang 25 Mar '21

25 Mar '21

On Asymmetric multiprocessor, there is Cortex-A core and Cortex-M core, Linux is running on A core, RTOS is running on M core. The audio hardware device can be controlled by Cortex-M device, So audio playback/capture can be handled by M core. Rpmsg is the interface for sending and receiving msg to and from M core, that we can create a virtual sound on Cortex-A core side. A core will tell the Cortex-M core sound format/rate/channel, where is the data buffer, what is the period size, when to start, when to stop and when suspend or resume happen, each of this behavior there is defined rpmsg command. Especially we designed the low power audio case, that is to allocate a large buffer and fill the data, then Cortex-A core can go to sleep mode, Cortex-M core continue to play the sound, when the buffer is consumed, Cortex-M core will trigger the Cortex-A core to wakeup to fill data. changes in v5: - remove unneeded property in binding doc and driver - update binding doc according to Rob's comments. - Fix link issue reported by kernel test robot changes in v4: - remove the sound card node, merge the property to cpu dai node according to Rob's comments. - sound card device will be registered by cpu dai driver. - Fix do_div issue reported by kernel test robot changes in v3: - add local refcount for clk enablement in hw_params() - update the document according Rob's comments changes in v2: - update codes and comments according to Mark's comments Shengjiu Wang (6): ASoC: soc-component: Add snd_soc_pcm_component_ack ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg ASoC: dt-bindings: fsl_rpmsg: Add binding doc for rpmsg audio device ASoC: imx-audio-rpmsg: Add rpmsg_driver for audio channel ASoC: imx-pcm-rpmsg: Add platform driver for audio base on rpmsg ASoC: imx-rpmsg: Add machine driver for audio base on rpmsg .../devicetree/bindings/sound/fsl,rpmsg.yaml | 108 +++ include/sound/soc-component.h | 3 + sound/soc/fsl/Kconfig | 30 + sound/soc/fsl/Makefile | 6 + sound/soc/fsl/fsl_rpmsg.c | 279 ++++++ sound/soc/fsl/fsl_rpmsg.h | 35 + sound/soc/fsl/imx-audio-rpmsg.c | 140 +++ sound/soc/fsl/imx-pcm-rpmsg.c | 918 ++++++++++++++++++ sound/soc/fsl/imx-pcm-rpmsg.h | 512 ++++++++++ sound/soc/fsl/imx-rpmsg.c | 150 +++ sound/soc/soc-component.c | 14 + sound/soc/soc-pcm.c | 2 + 12 files changed, 2197 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml create mode 100644 sound/soc/fsl/fsl_rpmsg.c create mode 100644 sound/soc/fsl/fsl_rpmsg.h create mode 100644 sound/soc/fsl/imx-audio-rpmsg.c create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.c create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.h create mode 100644 sound/soc/fsl/imx-rpmsg.c -- 2.27.0

2 7

[PATCH] ASoC: rt1015: Add bclk detection and dc detection
by Jack Yu 25 Mar '21

25 Mar '21

Add bclk detection and dc detection before playback. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 18 +++++++++++++++++- sound/soc/codecs/rt1015.h | 10 ++++++++++ 2 files changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 10a7b7745f4b..9238f12999aa 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -669,8 +669,23 @@ static int rt1015_amp_drv_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); + unsigned int ret, ret2; switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = snd_soc_component_read(component, RT1015_CLK_DET); + ret2 = snd_soc_component_read(component, RT1015_SPK_DC_DETECT1); + if (!((ret >> 15) & 0x1)) { + snd_soc_component_update_bits(component, RT1015_CLK_DET, + RT1015_EN_BCLK_DET_MASK, RT1015_EN_BCLK_DET); + dev_dbg(component->dev, "BCLK Detection Enabled.\n"); + } + if (!((ret2 >> 12) & 0x1)) { + snd_soc_component_update_bits(component, RT1015_SPK_DC_DETECT1, + RT1015_EN_CLA_D_DC_DET_MASK, RT1015_EN_CLA_D_DC_DET); + dev_dbg(component->dev, "Class-D DC Detection Enabled.\n"); + } + break; case SND_SOC_DAPM_POST_PMU: if (rt1015->hw_config == RT1015_HW_28) schedule_delayed_work(&rt1015->flush_work, msecs_to_jiffies(10)); @@ -690,7 +705,8 @@ static const struct snd_soc_dapm_widget rt1015_dapm_widgets[] = { r1015_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_OUT_DRV_E("Amp Drv", SND_SOC_NOPM, 0, 0, NULL, 0, - rt1015_amp_drv_event, SND_SOC_DAPM_POST_PMU), + rt1015_amp_drv_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_OUTPUT("SPO"), }; diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index 2aeaf65ba793..14344532048e 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -209,6 +209,11 @@ #define RT1015_PLL_K_MASK (RT1015_PLL_K_MAX) #define RT1015_PLL_K_SFT 0 +/* 0x0020 */ +#define RT1015_EN_BCLK_DET_MASK (0x1 << 15) +#define RT1015_EN_BCLK_DET (0x1 << 15) +#define RT1015_DIS_BCLK_DET (0x0 << 15) + /* 0x007a */ #define RT1015_ID_MASK 0xff #define RT1015_ID_VERA 0x0 @@ -374,6 +379,11 @@ #define RT1015_PWR_SWR (0x1 << 12) #define RT1015_PWR_SWR_BIT 12 +/* 0x0519 */ +#define RT1015_EN_CLA_D_DC_DET_MASK (0x1 << 12) +#define RT1015_EN_CLA_D_DC_DET (0x1 << 12) +#define RT1015_DIS_CLA_D_DC_DET (0x0 << 12) + /* 0x1300 */ #define RT1015_PWR_CLSD (0x1 << 12) #define RT1015_PWR_CLSD_BIT 12 -- 2.29.0

2 1

[PATCH 1/2] ASoC: fsl-asoc-card: Add support for WM8958 codec
by Shengjiu Wang 25 Mar '21

25 Mar '21

WM8958 codec is used on some i.MX based platform. So add it support in this generic driver. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/Kconfig | 2 ++ sound/soc/fsl/fsl-asoc-card.c | 17 +++++++++++++++-- 2 files changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index c71c6024320b..0917d65d6921 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -310,6 +310,8 @@ config SND_SOC_FSL_ASOC_CARD select SND_SOC_FSL_ESAI select SND_SOC_FSL_SAI select SND_SOC_FSL_SSI + select SND_SOC_WM8994 + select MFD_WM8994 help ALSA SoC Audio support with ASRC feature for Freescale SoCs that have ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888, diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index f62f81ceab0d..c62bfd1c3ac7 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -25,6 +25,7 @@ #include "../codecs/sgtl5000.h" #include "../codecs/wm8962.h" #include "../codecs/wm8960.h" +#include "../codecs/wm8994.h" #define CS427x_SYSCLK_MCLK 0 @@ -37,12 +38,14 @@ /** * struct codec_priv - CODEC private data * @mclk_freq: Clock rate of MCLK + * @free_freq: Clock rate of MCLK for hw_free() * @mclk_id: MCLK (or main clock) id for set_sysclk() * @fll_id: FLL (or secordary clock) id for set_sysclk() * @pll_id: PLL id for set_pll() */ struct codec_priv { unsigned long mclk_freq; + unsigned long free_freq; u32 mclk_id; u32 fll_id; u32 pll_id; @@ -235,10 +238,10 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) priv->streams &= ~BIT(substream->stream); if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) { - /* Force freq to be 0 to avoid error message in codec */ + /* Force freq to be free_freq to avoid error message in codec */ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), codec_priv->mclk_id, - 0, + codec_priv->free_freq, SND_SOC_CLOCK_IN); if (ret) { dev_err(dev, "failed to switch away from FLL: %d\n", ret); @@ -665,6 +668,15 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; priv->card.dapm_routes = audio_map_rx; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx); + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8958")) { + codec_dai_name = "wm8994-aif1"; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + priv->codec_priv.mclk_id = WM8994_FLL_SRC_MCLK1; + priv->codec_priv.fll_id = WM8994_SYSCLK_FLL1; + priv->codec_priv.pll_id = WM8994_FLL1; + priv->codec_priv.free_freq = priv->codec_priv.mclk_freq; + priv->card.dapm_routes = NULL; + priv->card.num_dapm_routes = 0; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; @@ -882,6 +894,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-mqs", }, { .compatible = "fsl,imx-audio-wm8524", }, { .compatible = "fsl,imx-audio-si476x", }, + { .compatible = "fsl,imx-audio-wm8958", }, {} }; MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); -- 2.27.0

2 2

[PATCH 1/3] ASoC:codec:max98373: Changed amp shutdown register as volatile
by Ryan Lee 25 Mar '21

25 Mar '21

0x20FF(amp global enable) register was defined as non-volatile, but it is not. Overheating, overcurrent can cause amp shutdown in hardware. 'regmap_write' compare register readback value before writing to avoid same value writing. 'regmap_read' just read cache not actual hardware value for the non-volatile register. When amp is internally shutdown by some reason, next 'AMP ON' command can be ignored because regmap think amp is already ON. Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98373-i2c.c | 1 + sound/soc/codecs/max98373-sdw.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c index 85f6865019d4..ddb6436835d7 100644 --- a/sound/soc/codecs/max98373-i2c.c +++ b/sound/soc/codecs/max98373-i2c.c @@ -446,6 +446,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg) case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK: case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK: case MAX98373_R20B6_BDE_CUR_STATE_READBACK: + case MAX98373_R20FF_GLOBAL_SHDN: case MAX98373_R21FF_REV_ID: return true; default: diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index d8c47667a9ea..f3a12205cd48 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -220,6 +220,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg) case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK: case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK: case MAX98373_R20B6_BDE_CUR_STATE_READBACK: + case MAX98373_R20FF_GLOBAL_SHDN: case MAX98373_R21FF_REV_ID: /* SoundWire Control Port Registers */ case MAX98373_R0040_SCP_INIT_STAT_1 ... MAX98373_R0070_SCP_FRAME_CTLR: -- 2.17.1

2 4

[PATCH v2] ASoC: wm8960: Remove bitclk relax condition in wm8960_configure_sysclk
by Shengjiu Wang 25 Mar '21

25 Mar '21

The call sequence in wm8960_configure_clocking is ret = wm8960_configure_sysclk(); if (ret >= 0) goto configure_clock; .... ret = wm8960_configure_pll(); configure_clock: ... wm8960_configure_sysclk is called before wm8960_configure_pll, as there is bitclk relax on both functions, so wm8960_configure_sysclk always return success, then wm8960_configure_pll() never be called. With this case: aplay -Dhw:0,0 -d 5 -r 48000 -f S24_LE -c 2 audio48k24b2c.wav the required bitclk is 48000 * 24 * 2 = 2304000, bitclk got from wm8960_configure_sysclk is 3072000, but if go to wm8960_configure_pll. it can get correct bitclk 2304000. So bitclk relax condition should be removed in wm8960_configure_sysclk, then wm8960_configure_pll can be called, and there is also bitclk relax function in wm8960_configure_pll. Fixes: 3c01b9ee2ab9 ("ASoC: codec: wm8960: Relax bit clock computation") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- changes in v2: - Find the real reason for the case, just patially remove bitclk relax sound/soc/codecs/wm8960.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index df351519a3a6..847ca16b9841 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -608,10 +608,6 @@ static const int bclk_divs[] = { * - lrclk = sysclk / dac_divs * - 10 * bclk = sysclk / bclk_divs * - * If we cannot find an exact match for (sysclk, lrclk, bclk) - * triplet, we relax the bclk such that bclk is chosen as the - * closest available frequency greater than expected bclk. - * * @wm8960: codec private data * @mclk: MCLK used to derive sysclk * @sysclk_idx: sysclk_divs index for found sysclk @@ -629,7 +625,7 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, { int sysclk, bclk, lrclk; int i, j, k; - int diff, closest = mclk; + int diff; /* marker for no match */ *bclk_idx = -1; @@ -653,12 +649,6 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, *bclk_idx = k; break; } - if (diff > 0 && closest > diff) { - *sysclk_idx = i; - *dac_idx = j; - *bclk_idx = k; - closest = diff; - } } if (k != ARRAY_SIZE(bclk_divs)) break; -- 2.27.0

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[PATCH 0/3] ASoC: soc.h: small cleanups
by Kuninori Morimoto 25 Mar '21

25 Mar '21

Hi Mark These are small cleanups for soc.h Kuninori Morimoto (3): ASoC: soc.h: add asoc_link_to_cpu/codec/platform() macro ASoC: soc.h: fixup return timing for snd_soc_fixup_dai_links_platform_name() ASoC: soc.h: return error if multi platform at snd_soc_fixup_dai_links_platform_name() include/sound/soc.h | 34 ++++++++++++++++++++++++++++------ 1 file changed, 28 insertions(+), 6 deletions(-) -- 2.25.1 Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH v2 1/2] ALSA: usb-audio: Carve out connector value checking into a helper
by Kai-Heng Feng 25 Mar '21

25 Mar '21

This is preparation for next patch, no functional change intended. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v2: - Only return early when ret < 0. sound/usb/mixer.c | 34 +++++++++++++++++++++++++--------- 1 file changed, 25 insertions(+), 9 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index b004b2e63a5d..5a2d9a768f70 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1446,13 +1446,11 @@ static int mixer_ctl_master_bool_get(struct snd_kcontrol *kcontrol, return 0; } -/* get the connectors status and report it as boolean type */ -static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int get_connector_value(struct usb_mixer_elem_info *cval, + char *name, int *val) { - struct usb_mixer_elem_info *cval = kcontrol->private_data; struct snd_usb_audio *chip = cval->head.mixer->chip; - int idx = 0, validx, ret, val; + int idx = 0, validx, ret; validx = cval->control << 8 | 0; @@ -1467,21 +1465,24 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC2_CS_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, &uac2_conn, sizeof(uac2_conn)); - val = !!uac2_conn.bNrChannels; + if (val) + *val = !!uac2_conn.bNrChannels; } else { /* UAC_VERSION_3 */ struct uac3_insertion_ctl_blk uac3_conn; ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC2_CS_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, &uac3_conn, sizeof(uac3_conn)); - val = !!uac3_conn.bmConInserted; + if (val) + *val = !!uac3_conn.bmConInserted; } snd_usb_unlock_shutdown(chip); if (ret < 0) { - if (strstr(kcontrol->id.name, "Speaker")) { - ucontrol->value.integer.value[0] = 1; + if (name && strstr(name, "Speaker")) { + if (val) + *val = 1; return 0; } error: @@ -1491,6 +1492,21 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, return filter_error(cval, ret); } + return ret; +} + +/* get the connectors status and report it as boolean type */ +static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *cval = kcontrol->private_data; + int ret, val; + + ret = get_connector_value(cval, kcontrol->id.name, &val); + + if (ret < 0) + return ret; + ucontrol->value.integer.value[0] = val; return 0; } -- 2.30.2

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