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How to set channel balance in amixer?
by GitHub issues - opened 04 Dec '20

04 Dec '20

alsa-project/alsa-utils issue #73 was opened from quantum-booty: When I change volumn using keyboard media keys, the left and right channel becomes imbalanced. ![image](https://user-images.githubusercontent.com/28958399/101188833-23befc00-364e-11eb-8816-efd94b9514d4.png) I currently have this mapping in i3, I would like to add command to balance the channels after I change the volumn. bindsym XF86AudioRaiseVolume exec --no-startup-id "amixer set Master '5%+,5%+'" Issue URL : https://github.com/alsa-project/alsa-utils/issues/73 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] [v2] ASoC: cros_ec_codec: fix uninitialized memory read
by Arnd Bergmann 04 Dec '20

04 Dec '20

From: Arnd Bergmann <arnd(a)arndb.de> gcc points out a memory area that is copied to a device but not initialized: sound/soc/codecs/cros_ec_codec.c: In function 'i2s_rx_event': arch/x86/include/asm/string_32.h:83:20: error: '*((void *)&p+4)' may be used uninitialized in this function [-Werror=maybe-uninitialized] 83 | *((int *)to + 1) = *((int *)from + 1); Change the length of the command to only pass down the part of the structure that has been initialized, as Tzung-Bi Shih explains that only that member is meant to be used. Cc: Tzung-Bi Shih <tzungbi(a)google.com> Fixes: 727f1c71c780 ("ASoC: cros_ec_codec: refactor I2S RX") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/cros_ec_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 58894bf47514..6ec673573c70 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -348,7 +348,7 @@ static int i2s_rx_event(struct snd_soc_dapm_widget *w, } return send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX, - (uint8_t *)&p, sizeof(p), NULL, 0); + &p.cmd, sizeof(p.cmd), NULL, 0); } static struct snd_soc_dapm_widget i2s_rx_dapm_widgets[] = { -- 2.27.0

2 2

Behringer WING usb audio - cyclic xruns dependent on periods/buffers
by Ben Bell 03 Dec '20

03 Dec '20

I've got a Behringer WING digital mixer which is equipped with a USB interface supporting 48 in, 48 out at 44.1 or 48kHz. It's plugged into a USB3 interface and for the most part it seems to work well as a class-compliant audio interface, but I'm struggling to eliminate xruns and it's starting to feel either driver or hardware quirk related issue. I've done all the usual things -- I'm running a PREEMPT_RT kernel, set up the realtime priorities of Jack and the relevant USB IRQ to no avail. I'm doing most of the debugging remotely over ssh with X shutdown and most other processes on the system stopped. There's nothing plugged into the computer beyond network, keyboard, mouse and the audio interface. I'm currently testing on kernel 5.10.0-rc5 with preempt-rt patches, but I've previously seen the same things on 5.4-rt (Debian backports) and 5.8 (self-compiled). The xruns are characterised by bursts of "retire_capture_urb" warnings in the kernel logs (and in particular the frame status for everything in those URBs is EXDEV, meaning I think that the frames haven't been consumed by the time it's retired). I've patched sound/driver/pcm.c to provide me with a bit more debugging information and it looks like usually it's almost always between 7-10 consecutive calls to retire_capture_urb that are affected. The bursts of retire_capture_urb warnings seem roughly cyclic, with a cycle time that is dependent on the combination of the frames/period setting and the number of periods per buffer, though it doesn't appear to be a strictly linear thing: 512/2 (~170s cycle); 256/2 (~94s); 128/3 (~65s); 64/13 (~270s). I can't immediately see any definitive smoking gun in the urb or interrupt counts, but there is some grouping in the timings between the xruns. Taking 64 frames, 5 periods, it's usually 50s, but sometimes 60s: 21:47:57 50s since xrun 4 21:48:47 50s since xrun 5 21:49:46 59s since xrun 6 21:50:46 60s since xrun 7 21:51:36 50s since xrun 8 21:52:36 60s since xrun 9 21:53:36 60s since xrun 10 21:54:37 61s since xrun 11 21:55:27 50s since xrun 12 21:56:27 60s since xrun 13 21:57:27 60s since xrun 14 21:58:17 50s since xrun 15 21:59:16 59s since xrun 16 22:00:06 50s since xrun 17 22:00:56 50s since xrun 18 ... If I count the URB numbers (sorry, I'm shaky on the terminology, but I'm counting each URB which is retired) I often (but not always) see the same number of URBs passing between xrun bursts. Currently at 64 frames/5 periods, it's often exactly 50042 or 49923 urbs between bursts. The delay between starting Jack and encountering the first burst is not predictable -- it's not just a whole cycle -- but stopping jack for e.g 30s then starting again delays the next burst by the same amount. So it does seem related to something in the audio streaming rather than anything else going on on the system interfering. It's as if something is slowly slipping out of sync between the Wing and the kernel until they need to resync, but I don't really know enough about USB to have any deeper insight. All the tests above happen to be at 44.1kHz with the Wing set to 48 in, 48 out, but changing to 48kHz or 2/2 IO doesn't seem to cure anything. One more data point which may not be relevant: if I switch off or unplug the Wing while Jack is running, the system gradually locks up over a period of a few seconds. I can get something out of the console briefly afterwards but within about ten seconds it's completely unresponsive. I imagine that pulling out a USB device from a realtime thread isn't a kind thing to do, but I don't recall this happening with other interfaces. Any help or insights into where I should be looking (I'm a newbie in kernel space) appreciated. If someone wants remote access to the box to investigate in realtime we could probably figure something out. bjb

2 3

[bug report] ASoC: mchp-spdifrx: add driver for SPDIF RX
by Dan Carpenter 03 Dec '20

03 Dec '20

Hello Codrin Ciubotariu, The patch ef265c55c1ac: "ASoC: mchp-spdifrx: add driver for SPDIF RX" from Oct 2, 2020, leads to the following static checker warning: sound/soc/atmel/mchp-spdifrx.c:468 mchp_spdifrx_hw_params() warn: 'dev->gclk' not released on lines: 468. sound/soc/atmel/mchp-spdifrx.c 442 params_format(params)); 443 return -EINVAL; 444 } 445 446 if (dev->gclk_enabled) { 447 clk_disable_unprepare(dev->gclk); 448 dev->gclk_enabled = 0; 449 } 450 ret = clk_set_min_rate(dev->gclk, params_rate(params) * 451 SPDIFRX_GCLK_RATIO_MIN + 1); 452 if (ret) { 453 dev_err(dev->dev, 454 "unable to set gclk min rate: rate %u * ratio %u + 1\n", 455 params_rate(params), SPDIFRX_GCLK_RATIO_MIN); 456 return ret; 457 } 458 ret = clk_prepare_enable(dev->gclk); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 459 if (ret) { 460 dev_err(dev->dev, "unable to enable gclk: %d\n", ret); 461 return ret; 462 } 463 dev->gclk_enabled = 1; 464 465 dev_dbg(dev->dev, "GCLK range min set to %d\n", 466 params_rate(params) * SPDIFRX_GCLK_RATIO_MIN + 1); 467 468 return regmap_write(dev->regmap, SPDIFRX_MR, mr); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Smatch is complaining that if the regmap_write() fails then we should disable and unprepare the "dev->gclk". 469 } regards, dan carpenter

2 2

[PATCH] ASoC: intel: sof_rt5682: Add support for tgl_rt1011_rt5682
by Brent Lu 03 Dec '20

03 Dec '20

This patch adds the driver data for two rt1011 speaker amplifiers on SSP1 and rt5682 on SSP0 for TGL platform. DAI format for rt1011 is leveraged from cml_rt1011_rt5682 which is 4-slot tdm with 100fs bclk. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/sof_realtek_common.c | 137 ++++++++++++++++++ sound/soc/intel/boards/sof_realtek_common.h | 24 +++ sound/soc/intel/boards/sof_rt5682.c | 26 +++- .../intel/common/soc-acpi-intel-tgl-match.c | 13 ++ 6 files changed, 198 insertions(+), 5 deletions(-) create mode 100644 sound/soc/intel/boards/sof_realtek_common.c create mode 100644 sound/soc/intel/boards/sof_realtek_common.h diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index dddb672a6d55..b58b9b60d37e 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -443,6 +443,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) select SND_SOC_MAX98373_I2C + select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT5682_I2C select SND_SOC_DMIC diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index a58e4d22e9c8..5f03e484b215 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -18,7 +18,7 @@ snd-soc-sst-byt-cht-cx2072x-objs := bytcht_cx2072x.o snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o -snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o sof_maxim_common.o +snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o sof_maxim_common.o sof_realtek_common.o snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o hda_dsp_common.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c new file mode 100644 index 000000000000..b3bc41385436 --- /dev/null +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -0,0 +1,137 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2020 Intel Corporation. All rights reserved. + +#include <linux/kernel.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <uapi/sound/asound.h> +#include "../../codecs/rt1011.h" +#include "sof_realtek_common.h" + +/* + * Current only 2-amp configuration is supported for rt1011 + */ +static const struct snd_soc_dapm_route rt1011_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left SPO" }, + { "Right Spk", NULL, "Right SPO" }, +}; + +/* + * Make sure device's Unique ID follows this configuration: + * + * Two speakers: + * 0: left, 1: right + * Four speakers: + * 0: Woofer left, 1: Woofer right + * 2: Tweeter left, 3: Tweeter right + */ +static struct snd_soc_codec_conf rt1011_codec_confs[] = { + { + .dlc = COMP_CODEC_CONF(RT1011_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(RT1011_DEV1_NAME), + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link_component rt1011_dai_link_components[] = { + { + .name = RT1011_DEV0_NAME, + .dai_name = RT1011_CODEC_DAI, + }, + { + .name = RT1011_DEV1_NAME, + .dai_name = RT1011_CODEC_DAI, + }, +}; + +struct { + unsigned int tx; + unsigned int rx; +} rt1011_tdm_mask[] = { + {.tx = 0x4, .rx = 0x1}, + {.tx = 0x8, .rx = 0x2}, +}; + +static int rt1011_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int srate, i, ret = 0; + + srate = params_rate(params); + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + /* 100 Fs to drive 24 bit data */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1011_PLL1_S_BCLK, + 100 * srate, 256 * srate); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set pll, ret %d\n", + ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT1011_FS_SYS_PRE_S_PLL1, + 256 * srate, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set sysclk, ret %d\n", + ret); + return ret; + } + + if (i >= ARRAY_SIZE(rt1011_tdm_mask)) { + dev_err(codec_dai->dev, "invalid codec index %d\n", + i); + return -ENODEV; + } + + ret = snd_soc_dai_set_tdm_slot(codec_dai, rt1011_tdm_mask[i].tx, + rt1011_tdm_mask[i].rx, 4, + params_width(params)); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set tdm slot, ret %d\n", + ret); + return ret; + } + } + + return 0; +} + +static const struct snd_soc_ops rt1011_ops = { + .hw_params = rt1011_hw_params, +}; + +static int rt1011_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, rt1011_dapm_routes, + ARRAY_SIZE(rt1011_dapm_routes)); + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +void sof_rt1011_dai_link(struct snd_soc_dai_link *link) +{ + link->codecs = rt1011_dai_link_components; + link->num_codecs = ARRAY_SIZE(rt1011_dai_link_components); + link->init = rt1011_init; + link->ops = &rt1011_ops; +} + +void sof_rt1011_codec_conf(struct snd_soc_card *card) +{ + card->codec_conf = rt1011_codec_confs; + card->num_configs = ARRAY_SIZE(rt1011_codec_confs); +} diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h new file mode 100644 index 000000000000..87cb3812b926 --- /dev/null +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Copyright(c) 2020 Intel Corporation. + */ + +/* + * This file defines data structures used in Machine Driver for Intel + * platforms with Realtek Codecs. + */ +#ifndef __SOF_REALTEK_COMMON_H +#define __SOF_REALTEK_COMMON_H + +#include <sound/soc.h> + +#define RT1011_CODEC_DAI "rt1011-aif" +#define RT1011_DEV0_NAME "i2c-10EC1011:00" +#define RT1011_DEV1_NAME "i2c-10EC1011:01" +#define RT1011_DEV2_NAME "i2c-10EC1011:02" +#define RT1011_DEV3_NAME "i2c-10EC1011:03" + +void sof_rt1011_dai_link(struct snd_soc_dai_link *link); +void sof_rt1011_codec_conf(struct snd_soc_card *card); + +#endif /* __SOF_REALTEK_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 891f908659f5..8b1ca2da9bb9 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -24,6 +24,7 @@ #include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" #include "sof_maxim_common.h" +#include "sof_realtek_common.h" #define NAME_SIZE 32 @@ -41,10 +42,11 @@ #define SOF_RT5682_NUM_HDMIDEV_MASK (GENMASK(12, 10)) #define SOF_RT5682_NUM_HDMIDEV(quirk) \ ((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK) -#define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(13) -#define SOF_RT1015_SPEAKER_AMP_100FS BIT(14) -#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(15) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(16) +#define SOF_RT1011_SPEAKER_AMP_PRESENT BIT(13) +#define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(14) +#define SOF_RT1015_SPEAKER_AMP_100FS BIT(15) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(16) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(17) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -741,6 +743,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].codecs = max98360a_component; links[id].num_codecs = ARRAY_SIZE(max98360a_component); links[id].init = speaker_codec_init; + } else if (sof_rt5682_quirk & + SOF_RT1011_SPEAKER_AMP_PRESENT) { + sof_rt1011_dai_link(&links[id]); } else { links[id].codecs = max98357a_component; links[id].num_codecs = ARRAY_SIZE(max98357a_component); @@ -851,6 +856,8 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) sof_max98373_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) + sof_rt1011_codec_conf(&sof_audio_card_rt5682); dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, dmic_be_num, hdmi_num); @@ -931,6 +938,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT1015_SPEAKER_AMP_100FS | SOF_RT5682_SSP_AMP(1)), }, + { + .name = "tgl_rt1011_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1011_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, { } }; @@ -948,6 +964,7 @@ module_platform_driver(sof_audio) MODULE_DESCRIPTION("SOF Audio Machine driver"); MODULE_AUTHOR("Bard Liao <bard.liao(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); +MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:sof_rt5682"); MODULE_ALIAS("platform:tgl_max98357a_rt5682"); @@ -955,3 +972,4 @@ MODULE_ALIAS("platform:jsl_rt5682_rt1015"); MODULE_ALIAS("platform:tgl_max98373_rt5682"); MODULE_ALIAS("platform:jsl_rt5682_max98360a"); MODULE_ALIAS("platform:cml_rt1015_rt5682"); +MODULE_ALIAS("platform:tgl_rt1011_rt5682"); diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 9f243e60b95c..84cc6326979a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -310,6 +310,11 @@ static struct snd_soc_acpi_codecs tgl_max98373_amp = { .codecs = {"MX98373"} }; +static struct snd_soc_acpi_codecs tgl_rt1011_amp = { + .num_codecs = 1, + .codecs = {"10EC1011"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { { .id = "10EC1308", @@ -335,6 +340,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { .sof_fw_filename = "sof-tgl.ri", .sof_tplg_filename = "sof-tgl-max98373-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "tgl_rt1011_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &tgl_rt1011_amp, + .sof_fw_filename = "sof-tgl.ri", + .sof_tplg_filename = "sof-tgl-rt1011-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_tgl_machines); -- 2.17.1

3 2

[PATCH] ASoC: amd: add missed checks for clk_get()
by Chuhong Yuan 03 Dec '20

03 Dec '20

cz_da7219_init() does not check the return value of clk_get(), while da7219_clk_enable() calls clk_set_rate() to dereference the pointers. Add checks to fix the problems. Fixes: bb24a31ed584 ("ASoC: AMD: Configure wclk and bclk of master codec") Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/amd/acp-da7219-max98357a.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index a7702e64ec51..d5185cd3106b 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -74,7 +74,12 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) } da7219_dai_wclk = clk_get(component->dev, "da7219-dai-wclk"); + if (IS_ERR(da7219_dai_wclk)) + return PTR_ERR(da7219_dai_wclk); + da7219_dai_bclk = clk_get(component->dev, "da7219-dai-bclk"); + if (IS_ERR(da7219_dai_bclk)) + return PTR_ERR(da7219_dai_bclk); ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_LINEOUT | -- 2.26.2

2 1

[PATCH 1/2] ASoC: fsl-asoc-card: Add support for si476x codec
by Shengjiu Wang 03 Dec '20

03 Dec '20

The si476x codec is used for FM radio function on i.MX6 auto board, it only supports recording function. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index a2dd3b6b7fec..f62f81ceab0d 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -131,6 +131,13 @@ static const struct snd_soc_dapm_route audio_map_tx[] = { {"CPU-Playback", NULL, "ASRC-Playback"}, }; +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* 1st half -- Normal DAPM routes */ + {"CPU-Capture", NULL, "Capture"}, + /* 2nd half -- ASRC DAPM routes */ + {"ASRC-Capture", NULL, "CPU-Capture"}, +}; + /* Add all possible widgets into here without being redundant */ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_LINE("Line Out Jack", NULL), @@ -653,6 +660,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.slot_width = 32; priv->card.dapm_routes = audio_map_tx; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); + } else if (of_device_is_compatible(np, "fsl,imx-audio-si476x")) { + codec_dai_name = "si476x-codec"; + 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 { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; @@ -869,6 +881,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-wm8960", }, { .compatible = "fsl,imx-audio-mqs", }, { .compatible = "fsl,imx-audio-wm8524", }, + { .compatible = "fsl,imx-audio-si476x", }, {} }; MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); -- 2.27.0

3 3

[PATCH] ASoC: fsl_audmix: Remove unneeded data field
by Fabio Estevam 03 Dec '20

03 Dec '20

The .data field is only used to pass the string name to platform_device_register_data(). Pass the string name directly to make the code simpler. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/fsl/fsl_audmix.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index 7ad5925772e8..8dc44dec7956 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -455,7 +455,6 @@ static const struct regmap_config fsl_audmix_regmap_config = { static const struct of_device_id fsl_audmix_ids[] = { { .compatible = "fsl,imx8qm-audmix", - .data = "imx-audmix", }, { /* sentinel */ } }; @@ -465,17 +464,9 @@ static int fsl_audmix_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct fsl_audmix *priv; - const char *mdrv; - const struct of_device_id *of_id; void __iomem *regs; int ret; - of_id = of_match_device(fsl_audmix_ids, dev); - if (!of_id || !of_id->data) - return -EINVAL; - - mdrv = of_id->data; - priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; @@ -510,10 +501,10 @@ static int fsl_audmix_probe(struct platform_device *pdev) goto err_disable_pm; } - priv->pdev = platform_device_register_data(dev, mdrv, 0, NULL, 0); + priv->pdev = platform_device_register_data(dev, "imx-audmix", 0, NULL, 0); if (IS_ERR(priv->pdev)) { ret = PTR_ERR(priv->pdev); - dev_err(dev, "failed to register platform %s: %d\n", mdrv, ret); + dev_err(dev, "failed to register platform: %d\n", ret); goto err_disable_pm; } -- 2.17.1

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[PATCH 0/4] ASoC/SoundWire: add Realtek SoundWire SDCA codec
by Bard Liao 03 Dec '20

03 Dec '20

Add Realtek rt1316 and rt711-sdca vender-specific drivers plus some fixup for rt715-sdca driver. Jack Yu (2): ASoC/SoundWire: rt715-sdca: use rt715_sdca prefix for symbols ASoC/SoundWire: rt715-sdca: merge mute/volume operations Shuming Fan (2): ASoC/SoundWire: rt1316: Add RT1316 SDCA vendor-specific driver ASoC/SoundWire: rt711-sdca: Add RT711 SDCA vendor-specific driver sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/rt1316-sdw.c | 756 +++++++++++++++ sound/soc/codecs/rt1316-sdw.h | 115 +++ sound/soc/codecs/rt711-sdca-sdw.c | 424 +++++++++ sound/soc/codecs/rt711-sdca-sdw.h | 101 ++ sound/soc/codecs/rt711-sdca.c | 1482 +++++++++++++++++++++++++++++ sound/soc/codecs/rt711-sdca.h | 246 +++++ sound/soc/codecs/rt715-sdca-sdw.c | 12 +- sound/soc/codecs/rt715-sdca.c | 323 ++++--- sound/soc/codecs/rt715-sdca.h | 4 +- 11 files changed, 3316 insertions(+), 164 deletions(-) create mode 100644 sound/soc/codecs/rt1316-sdw.c create mode 100644 sound/soc/codecs/rt1316-sdw.h create mode 100644 sound/soc/codecs/rt711-sdca-sdw.c create mode 100644 sound/soc/codecs/rt711-sdca-sdw.h create mode 100644 sound/soc/codecs/rt711-sdca.c create mode 100644 sound/soc/codecs/rt711-sdca.h -- 2.17.1

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[PATCH v3 1/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Dec '20

03 Dec '20

If stop by underrun and DPCM BE disconnection is run simultaneously, data abort can be occurred by the sequence below. CPU0 CPU1 dpcm_be_dai_trigger(): dpcm_be_disconnect(): for_each_dpcm_be(fe, stream, dpcm) { spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); struct snd_soc_pcm_runtime *be = dpcm->be; <-- Accessing freed memory To prevent this situation, dpcm_lock should be acquired during iteration of dpcm list in dpcm_be_dai_trigger(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcab9527ba3d..7c5d950a8628 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2073,6 +2073,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, struct snd_pcm_substream *substream, int cmd) { @@ -2092,8 +2095,10 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; + unsigned long flags; int ret = 0; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -2113,7 +2118,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2123,7 +2128,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2133,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2142,12 +2147,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2155,12 +2160,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2168,17 +2173,20 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } + if (ret) + break; } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); return ret; } @@ -2916,10 +2924,9 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; int ret = 1; - unsigned long flags; int i; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); + lockdep_assert_held(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -2933,17 +2940,12 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, } } } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to do this BE DAI */ return ret; } -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { const enum snd_soc_dpcm_state state[] = { @@ -2954,6 +2956,23 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + unsigned long flags; + int ret; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = dpcm_can_be_free_stop(fe, be, stream); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* @@ -2963,6 +2982,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { + unsigned long flags; + int ret; + const enum snd_soc_dpcm_state state[] = { SND_SOC_DPCM_STATE_START, SND_SOC_DPCM_STATE_PAUSED, @@ -2970,6 +2992,10 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, SND_SOC_DPCM_STATE_PREPARE, }; - return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); -- 2.21.0

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