March 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: codecs: wsa881x: request gpio direction before setting
by Srinivas Kandagatla 16 Mar '20

16 Mar '20

Make sure that power down gpio direction is set to ouput before even setting it. Fixes: a0aab9e1404a ("ASoC: codecs: add wsa881x amplifier support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wsa881x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index 35b44b297f9e..1810e0775efe 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -1154,7 +1154,7 @@ static int wsa881x_probe(struct sdw_slave *pdev, wsa881x->sconfig.type = SDW_STREAM_PDM; pdev->prop.sink_ports = GENMASK(WSA881X_MAX_SWR_PORTS, 0); pdev->prop.sink_dpn_prop = wsa_sink_dpn_prop; - gpiod_set_value(wsa881x->sd_n, 1); + gpiod_direction_output(wsa881x->sd_n, 1); wsa881x->regmap = devm_regmap_init_sdw(pdev, &wsa881x_regmap_config); if (IS_ERR(wsa881x->regmap)) { -- 2.21.0

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[alsa-devel] [PATCH v3 0/9] x86: Easy way of detecting MS Surface 3
by Andy Shevchenko 16 Mar '20

16 Mar '20

While working on RTC regression, I noticed that we are using the same DMI check over and over in the drivers for MS Surface 3 platform. This series dedicated for making it easier in the same way how it's done for Apple machines. Changelog v3: - fixed typo in patch 5 (Jonathan) - returned back to if {} else {} condition in ASoC driver (Mark) - added Mark's Ack tag Changelog v2: - removed RTC patches for now (the fix will be independent to this series) - added couple more clean ups to arch/x86/kernel/quirks.c - redone DMI quirk to use driver_data instead of callback - simplified check in soc-acpi-intel-cht-match.c to be oneliner - added a new patch to cover rt5645 codec driver Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Andy Shevchenko (9): x86/platform: Rename x86/apple.h -> x86/machine.h x86/quirks: Add missed include to satisfy static checker x86/quirks: Introduce hpet_dev_print_force_hpet_address() helper x86/quirks: Join string literals back x86/quirks: Convert DMI matching to use a table x86/quirks: Add a DMI quirk for Microsoft Surface 3 platform/x86: surface3_wmi: Switch DMI table match to a test of variable ASoC: rt5645: Switch DMI table match to a test of variable ASoC: Intel: Switch DMI table match to a test of variable arch/x86/kernel/quirks.c | 91 +++++++++++++------ drivers/platform/x86/surface3-wmi.c | 16 +--- include/linux/platform_data/x86/apple.h | 14 +-- include/linux/platform_data/x86/machine.h | 20 ++++ sound/soc/codecs/rt5645.c | 14 ++- .../intel/common/soc-acpi-intel-cht-match.c | 28 +----- 6 files changed, 93 insertions(+), 90 deletions(-) create mode 100644 include/linux/platform_data/x86/machine.h -- 2.24.1

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[PATCH 0/3] ALSA USB MIDI: Predictable low latency for USB MIDI output
by Andreas Steinmetz 16 Mar '20

16 Mar '20

The following patch series fixes USB MIDI output starvation and allows to configure a predictable worst case output latency with a minimum of 15ms for 16x16 devices and 3.8ms for 4x4 devices (excluding USB transfer time and latencies of the interface device itself). The reasoning behind this patch series is first to fix the possible indefinite latency and then to allow to configure a predictable low latency to be able to use a cheap embedded Linux system (e.g. Raspberry Pi) in combination with a multiport USB MIDI interface as a drop in replacement for commercial hardware based MIDI routing and filtering products. This includes "real time" (live) usage. For this usage the guaranteed latency within the kernel must be as low as possible, depending on the requirements of the USB MIDI interface in use. Though the current state of the kernel code is acceptable in a fully controlled by a single user environment, it is not usable for uncontrolled standalone use as the driver behaviour then depends on unpredictable MIDI data streams and thus is unpredictable in behaviour itself. After applying this patch series the USB MIDI driver can be used as usual or it can be tuned to low latency when this is desired. >From a developer perspective the actual output rate of USB MIDI interfaces is typically slower than the expected 320us and even varies depending on how many ports are busy (e.g. a M-Audio Midisport 4x4 AE interface exhibits a varying output rate of 357us to 515us per byte depending on device load). Thus the driver needs a predictable low latency to allow userspace to use the rawmidi buffer fill state to estimate the expected output latency and schedule data accordingly. The following table shows relative processing time for varying driver configurations using the M-Audio Midisport 4x4 AE as the test device. For the parameters "outqueue" and "events" do see the descriptions of the following patches. | unpatched | outqueue=7 | outqueue=1 | outqueue=1 | | kernel | events=0 | events=0 | events=4 | --------+---------------+---------------+---------------+---------------+ 1 Port | 100% | 100% | 100% | 100.05% | --------+---------------+---------------+---------------+---------------+ 4 Ports | n/a | 100% | 99.97% | 100.36% | --------+---------------+---------------+---------------+---------------+ As can be seen from the above table the changes to output latency do affect throughput only marginally.

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[PATCH 0/2] ALSA: seq: Fix running status after receiving sysex
by Takashi Iwai 16 Mar '20

16 Mar '20

Hi, here are two small fixes for correcting the behavior of sequencer core wrt the event decoding after receiving a sysex message. Takashi === Takashi Iwai (2): ALSA: seq: virmidi: Fix running status after receiving sysex ALSA: seq: oss: Fix running status after receiving sysex sound/core/seq/oss/seq_oss_midi.c | 1 + sound/core/seq/seq_virmidi.c | 1 + 2 files changed, 2 insertions(+) -- 2.16.4

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BUG: ALSA snd-virmidi: snd-virmidi is not MIDI 1.0 specification compliant
by Andreas Steinmetz 16 Mar '20

16 Mar '20

snd-virmidi, as of kernel 5.4.23, does not adhere to the MIDI specification: >From MIDI 1.0 Detailed Specification, Document Version 4.2, Appendix A- 1 (RUNNING STATUS), Page A-2: "3. Buffer is cleared when a System Exclusive or Common status message is received." snd-virmidi does not clear the running status in case of a SysEx message. How to test: Load snd-virmidi module and get the card number of the module. In the following 5 is assumed as the card number and device 1 is used. You need two shells, in shell 1 do: aconnect 'Virtual Raw MIDI 5-1':0 'Virtual Raw MIDI 5-1':0 amidi -p hw:VirMIDI,1 -d Now, in shell 2 do: amidi -p hw:VirMIDI,1 -S '80 01 02 F0 01 F7 80 02 03' This sends a note off, followed by a SysEx, followed by another note off. The result in shell 1 is: 80 01 02 F0 01 F7 02 03 So snd-virmidi did not clear the running status. The correct result would be: 80 01 02 F0 01 F7 80 02 03

2 1

[PATCH] ASoC: Intel: boards: add stereo playback by woofer speaker
by mac.chiang＠intel.com 16 Mar '20

16 Mar '20

From: Mac Chiang <mac.chiang(a)intel.com> Separate woofer stereo speakers settings. Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/cml_rt1011_rt5682.c | 221 +++++++++++++++++++++++------ 1 file changed, 180 insertions(+), 41 deletions(-) diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index dd80d01..fc1bd94 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -30,6 +30,36 @@ #define CML_RT5682_CODEC_DAI "rt5682-aif1" #define NAME_SIZE 32 +#define SOF_RT1011_SPEAKER_WL BIT(0) +#define SOF_RT1011_SPEAKER_WR BIT(1) +#define SOF_RT1011_SPEAKER_TL BIT(2) +#define SOF_RT1011_SPEAKER_TR BIT(3) +#define SPK_CH 4 + +/* Default: Woofer+Tweeter speakers */ +static unsigned long sof_rt1011_quirk = SOF_RT1011_SPEAKER_WL | SOF_RT1011_SPEAKER_WR | + SOF_RT1011_SPEAKER_TL | SOF_RT1011_SPEAKER_TR; + +static int sof_rt1011_quirk_cb(const struct dmi_system_id *id) +{ + sof_rt1011_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id sof_rt1011_quirk_table[] = { + { + .callback = sof_rt1011_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + DMI_MATCH(DMI_PRODUCT_NAME, "Palkia"), + }, + .driver_data = (void *)(SOF_RT1011_SPEAKER_WL | + SOF_RT1011_SPEAKER_WR), + }, + { + } +}; + static struct snd_soc_jack hdmi_jack[3]; struct hdmi_pcm { @@ -48,15 +78,16 @@ struct card_private { static const struct snd_kcontrol_new cml_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("TL Ext Spk"), - SOC_DAPM_PIN_SWITCH("TR Ext Spk"), SOC_DAPM_PIN_SWITCH("WL Ext Spk"), SOC_DAPM_PIN_SWITCH("WR Ext Spk"), }; +static const struct snd_kcontrol_new cml_rt1011_tt_controls[] = { + SOC_DAPM_PIN_SWITCH("TL Ext Spk"), + SOC_DAPM_PIN_SWITCH("TR Ext Spk"), +}; + static const struct snd_soc_dapm_widget cml_rt1011_rt5682_widgets[] = { - SND_SOC_DAPM_SPK("TL Ext Spk", NULL), - SND_SOC_DAPM_SPK("TR Ext Spk", NULL), SND_SOC_DAPM_SPK("WL Ext Spk", NULL), SND_SOC_DAPM_SPK("WR Ext Spk", NULL), SND_SOC_DAPM_HP("Headphone Jack", NULL), @@ -64,6 +95,11 @@ static const struct snd_soc_dapm_widget cml_rt1011_rt5682_widgets[] = { SND_SOC_DAPM_MIC("SoC DMIC", NULL), }; +static const struct snd_soc_dapm_widget cml_rt1011_tt_widgets[] = { + SND_SOC_DAPM_SPK("TL Ext Spk", NULL), + SND_SOC_DAPM_SPK("TR Ext Spk", NULL), +}; + static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = { /*speaker*/ {"TL Ext Spk", NULL, "TL SPO"}, @@ -82,6 +118,12 @@ static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = { {"DMic", NULL, "SoC DMIC"}, }; +static const struct snd_soc_dapm_route cml_rt1011_tt_map[] = { + /*TL/TR speaker*/ + {"TL Ext Spk", NULL, "TL SPO" }, + {"TR Ext Spk", NULL, "TR SPO" }, +}; + static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) { struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); @@ -121,6 +163,35 @@ static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; }; +static int cml_rt1011_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret = 0; + struct snd_soc_card *card = rtd->card; + + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + + ret = snd_soc_add_card_controls(card, cml_rt1011_tt_controls, + ARRAY_SIZE(cml_rt1011_tt_controls)); + if (ret) + return ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, + cml_rt1011_tt_widgets, + ARRAY_SIZE(cml_rt1011_tt_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, cml_rt1011_tt_map, + ARRAY_SIZE(cml_rt1011_tt_map)); + + if (ret) + return ret; + } + + return ret; +} + static int cml_rt5682_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -192,31 +263,52 @@ static int cml_rt1011_hw_params(struct snd_pcm_substream *substream, * The feedback is captured for each codec individually. * Hence all 4 codecs use 1 Tx slot each for feedback. */ - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x4, 0x1, 4, 24); - if (ret < 0) - break; - } - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:02")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x1, 0x1, 4, 24); - if (ret < 0) - break; - } - /* TDM Rx slot 2 is used for Right Woofer & Tweeters pair */ - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x8, 0x2, 4, 24); - if (ret < 0) - break; - } - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:03")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x2, 0x2, 4, 24); - if (ret < 0) - break; + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x4, 0x1, 4, 24); + if (ret < 0) + break; + } + + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:02")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x1, 0x1, 4, 24); + if (ret < 0) + break; + } + + /* TDM Rx slot 2 is used for Right Woofer & Tweeters pair */ + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x8, 0x2, 4, 24); + if (ret < 0) + break; + } + + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:03")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x2, 0x2, 4, 24); + if (ret < 0) + break; + } + } else { + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x4, 0x1, 4, 24); + if (ret < 0) + break; + } + + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x8, 0x2, 4, 24); + if (ret < 0) + break; + } } + } if (ret < 0) dev_err(rtd->dev, @@ -302,10 +394,8 @@ SND_SOC_DAILINK_DEF(ssp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); SND_SOC_DAILINK_DEF(ssp1_codec, DAILINK_COMP_ARRAY( - /* WL */ COMP_CODEC("i2c-10EC1011:00", CML_RT1011_CODEC_DAI), - /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI), - /* TL */ COMP_CODEC("i2c-10EC1011:02", CML_RT1011_CODEC_DAI), - /* TR */ COMP_CODEC("i2c-10EC1011:03", CML_RT1011_CODEC_DAI))); + /* WL */ COMP_CODEC("i2c-10EC1011:00", CML_RT1011_CODEC_DAI), + /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI))); SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); @@ -399,6 +489,7 @@ static struct snd_soc_dai_link cml_rt1011_rt5682_dailink[] = { .dpcm_playback = 1, .dpcm_capture = 1, /* Capture stream provides Feedback */ .no_pcm = 1, + .init = cml_rt1011_spk_init, .ops = &cml_rt1011_ops, SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), }, @@ -413,14 +504,6 @@ static struct snd_soc_codec_conf rt1011_conf[] = { .dlc = COMP_CODEC_CONF("i2c-10EC1011:01"), .name_prefix = "WR", }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1011:02"), - .name_prefix = "TL", - }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1011:03"), - .name_prefix = "TR", - }, }; /* Cometlake audio machine driver for RT1011 and RT5682 */ @@ -442,10 +525,12 @@ static struct snd_soc_card snd_soc_card_cml = { static int snd_cml_rt1011_probe(struct platform_device *pdev) { + struct snd_soc_dai_link_component *rt1011_dais_components; + struct snd_soc_codec_conf *rt1011_dais_confs; struct card_private *ctx; struct snd_soc_acpi_mach *mach; const char *platform_name; - int ret; + int ret, i; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); if (!ctx) @@ -456,6 +541,59 @@ static int snd_cml_rt1011_probe(struct platform_device *pdev) snd_soc_card_cml.dev = &pdev->dev; platform_name = mach->mach_params.platform; + dmi_check_system(sof_rt1011_quirk_table); + + dev_info(&pdev->dev, "sof_rt1011_quirk = %lx\n", sof_rt1011_quirk); + + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + rt1011_dais_confs = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_codec_conf) * + SPK_CH, GFP_KERNEL); + + rt1011_dais_components = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_dai_link_component) * + SPK_CH, GFP_KERNEL); + + for (i = 0; i < SPK_CH; i++) { + rt1011_dais_confs[i].dlc.name = devm_kasprintf(&pdev->dev, + GFP_KERNEL, + "i2c-10EC1011:0%d", + i); + switch (i) { + case 0: + rt1011_dais_confs[i].name_prefix = "WL"; + break; + + case 1: + rt1011_dais_confs[i].name_prefix = "WR"; + break; + case 2: + rt1011_dais_confs[i].name_prefix = "TL"; + break; + case 3: + rt1011_dais_confs[i].name_prefix = "TR"; + break; + } + rt1011_dais_components[i].name = devm_kasprintf(&pdev->dev, + GFP_KERNEL, + "i2c-10EC1011:0%d", + i); + rt1011_dais_components[i].dai_name = CML_RT1011_CODEC_DAI; + } + + snd_soc_card_cml.codec_conf = rt1011_dais_confs; + snd_soc_card_cml.num_configs = SPK_CH; + + for (i = 0; i < ARRAY_SIZE(cml_rt1011_rt5682_dailink); i++) { + if (!strcmp(cml_rt1011_rt5682_dailink[i].codecs->dai_name, + CML_RT1011_CODEC_DAI)) { + cml_rt1011_rt5682_dailink[i].codecs = rt1011_dais_components; + cml_rt1011_rt5682_dailink[i].num_codecs = SPK_CH; + } + } + } + /* set platform name for each dailink */ ret = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cml, platform_name); @@ -483,5 +621,6 @@ MODULE_DESCRIPTION("Cometlake Audio Machine driver - RT1011 and RT5682 in I2S mo MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:cml_rt1011_rt5682"); -- 2.7.4

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[PATCH AUTOSEL 5.4 25/35] ASoC: stm32: sai: manage rebind issue
by Sasha Levin 16 Mar '20

16 Mar '20

From: Olivier Moysan <olivier.moysan(a)st.com> [ Upstream commit 0d6defc7e0e437a9fd53622f7fd85740f38d5693 ] The commit e894efef9ac7 ("ASoC: core: add support to card rebind") allows to rebind the sound card after a rebind of one of its component. With this commit, the sound card is actually rebound, but may be no more functional. The following problems have been seen with STM32 SAI driver. 1) DMA channel is not requested: With the sound card rebind the simplified call sequence is: stm32_sai_sub_probe snd_soc_register_component snd_soc_try_rebind_card snd_soc_instantiate_card devm_snd_dmaengine_pcm_register The problem occurs because the pcm must be registered, before snd_soc_instantiate_card() is called. Modify SAI driver, to change the call sequence as follows: stm32_sai_sub_probe devm_snd_dmaengine_pcm_register snd_soc_register_component snd_soc_try_rebind_card 2) DMA channel is not released: dma_release_channel() is not called when devm_dmaengine_pcm_release() is executed. This occurs because SND_DMAENGINE_PCM_DRV_NAME component, has already been released through devm_component_release(). devm_dmaengine_pcm_release() should be called before devm_component_release() to avoid this problem. Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component() explicitly from SAI driver, to have the right sequence. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> Message-Id: <20200304102406.8093-1-olivier.moysan(a)st.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 30bcd5d3a32a8..10eb4b8e8e7ee 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1543,20 +1543,20 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(&pdev->dev, &stm32_component, - &sai->cpu_dai_drv, 1); + ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); + if (ret) { + dev_err(&pdev->dev, "Could not register pcm dma\n"); + return ret; + } + + ret = snd_soc_register_component(&pdev->dev, &stm32_component, + &sai->cpu_dai_drv, 1); if (ret) return ret; if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) conf = &stm32_sai_pcm_config_spdif; - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, conf, 0); - if (ret) { - dev_err(&pdev->dev, "Could not register pcm dma\n"); - return ret; - } - return 0; } @@ -1565,6 +1565,8 @@ static int stm32_sai_sub_remove(struct platform_device *pdev) struct stm32_sai_sub_data *sai = dev_get_drvdata(&pdev->dev); clk_unprepare(sai->pdata->pclk); + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } -- 2.20.1

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[PATCH AUTOSEL 5.5 29/41] ASoC: stm32: sai: manage rebind issue
by Sasha Levin 16 Mar '20

16 Mar '20

From: Olivier Moysan <olivier.moysan(a)st.com> [ Upstream commit 0d6defc7e0e437a9fd53622f7fd85740f38d5693 ] The commit e894efef9ac7 ("ASoC: core: add support to card rebind") allows to rebind the sound card after a rebind of one of its component. With this commit, the sound card is actually rebound, but may be no more functional. The following problems have been seen with STM32 SAI driver. 1) DMA channel is not requested: With the sound card rebind the simplified call sequence is: stm32_sai_sub_probe snd_soc_register_component snd_soc_try_rebind_card snd_soc_instantiate_card devm_snd_dmaengine_pcm_register The problem occurs because the pcm must be registered, before snd_soc_instantiate_card() is called. Modify SAI driver, to change the call sequence as follows: stm32_sai_sub_probe devm_snd_dmaengine_pcm_register snd_soc_register_component snd_soc_try_rebind_card 2) DMA channel is not released: dma_release_channel() is not called when devm_dmaengine_pcm_release() is executed. This occurs because SND_DMAENGINE_PCM_DRV_NAME component, has already been released through devm_component_release(). devm_dmaengine_pcm_release() should be called before devm_component_release() to avoid this problem. Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component() explicitly from SAI driver, to have the right sequence. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> Message-Id: <20200304102406.8093-1-olivier.moysan(a)st.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 30bcd5d3a32a8..10eb4b8e8e7ee 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1543,20 +1543,20 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(&pdev->dev, &stm32_component, - &sai->cpu_dai_drv, 1); + ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); + if (ret) { + dev_err(&pdev->dev, "Could not register pcm dma\n"); + return ret; + } + + ret = snd_soc_register_component(&pdev->dev, &stm32_component, + &sai->cpu_dai_drv, 1); if (ret) return ret; if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) conf = &stm32_sai_pcm_config_spdif; - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, conf, 0); - if (ret) { - dev_err(&pdev->dev, "Could not register pcm dma\n"); - return ret; - } - return 0; } @@ -1565,6 +1565,8 @@ static int stm32_sai_sub_remove(struct platform_device *pdev) struct stm32_sai_sub_data *sai = dev_get_drvdata(&pdev->dev); clk_unprepare(sai->pdata->pclk); + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } -- 2.20.1

1 0

[PATCH AUTOSEL 5.5 09/41] ASoC: meson: g12a: add tohdmitx reset
by Sasha Levin 16 Mar '20

16 Mar '20

From: Jerome Brunet <jbrunet(a)baylibre.com> [ Upstream commit 22946f37557e27697aabc8e4f62642bfe4a17fd8 ] Reset the g12a hdmi codec glue on probe. This ensure a sane startup state. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Link: https://lore.kernel.org/r/20200221121146.1498427-1-jbrunet@baylibre.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/meson/g12a-tohdmitx.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c index 9cfbd343a00c8..8a0db28a6a406 100644 --- a/sound/soc/meson/g12a-tohdmitx.c +++ b/sound/soc/meson/g12a-tohdmitx.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <sound/pcm_params.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <sound/soc.h> #include <sound/soc-dai.h> @@ -378,6 +379,11 @@ static int g12a_tohdmitx_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; void __iomem *regs; struct regmap *map; + int ret; + + ret = device_reset(dev); + if (ret) + return ret; regs = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(regs)) -- 2.20.1

1 0

[PATCH v2 00/10] ASoC: Intel: machine driver updates for 5.7
by Pierre-Louis Bossart 15 Mar '20

15 Mar '20

Fix HDaudio/HDMI Kconfig dependencies, cleanup code, and add support for 2 new configs (DA7219+Maxim 98360A and RT5692+RT1015) Note that the last patch is the first case where we added the Reviewed-by tags automagically from the list of GitHub approvers. Feedback/comments welcome on the format, and thanks to Ranjani for the nice script! Changes since v1: Change base to broonie/for-5.7 Rebase and fix for_each_codec_dais() macro in Patch 9. Add new patch to remove GFP_ATOMIC Guennadi Liakhovetski (3): ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable initialisations ASoC: Intel: skylake: (cosmetic) remove redundant variable initialisations ASoC: Intel: (cosmetic) simplify structure member access Kai Vehmanen (4): ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms ASoC: SOF: Intel: hda: remove SND_SOC_SOF_HDA_COMMON_HDMI_CODEC Pierre-Louis Bossart (1): ASoC: Intel: don't use GFP_ATOMIC for machine driver contexts Yong Zhi (2): ASoC: Intel: sof_da7219_max98373: Add support for max98360a speaker amp ASoC: Intel: sof_rt5682: Add rt1015 speaker amp support sound/soc/codecs/hdac_hdmi.c | 6 +- sound/soc/intel/boards/Kconfig | 20 ++-- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 2 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 4 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- .../soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 78 +++++++++++-- sound/soc/intel/boards/sof_pcm512x.c | 38 ++++-- sound/soc/intel/boards/sof_rt5682.c | 110 +++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 34 +++++- sound/soc/intel/skylake/skl.c | 4 +- sound/soc/sof/intel/Kconfig | 11 -- sound/soc/sof/intel/hda.c | 3 +- 26 files changed, 269 insertions(+), 71 deletions(-) base-commit: 101247a3b86e1cc0e382b7e887a56176290fc957 -- 2.20.1

3 14

[PATCH Fix for Kingston HyperX Amp (0951:16d8) 0/2 (Try #2)]
by crwulff＠gmail.com 14 Mar '20

14 Mar '20

From: Chris Wulff <crwulff(a)gmail.com> This patch set is modified to use an inline function for the mixer control interface lookup. The attempt at preventing modifications to a PCM that has previously been registered has been changed to defer registration instead, and has been split into a separate patch. This approach now works with pulseaudio.

2 3

Re: Re: [alsa-devel] [PATCH 4/7] ALSA: intel8x0: constify snd_pcm_ops structures
by sandra luna 14 Mar '20

14 Mar '20

1 0

[PATCH 00/16] SoundWire: cadence: add clock stop and fix programming sequences
by Pierre-Louis Bossart 14 Mar '20

14 Mar '20

To make progress with SoundWire support, this patchset provides the missing support for clock stop modes, and revisits all Cadence Master register settings. The current code is for some reason not aligned with internal documentation and hardware recommended flows, specifically for multi-link operation. Pierre-Louis Bossart (12): soundwire: cadence: s/update_config/config_update soundwire: cadence: handle error cases with CONFIG_UPDATE soundwire: cadence: mask Slave interrupt before stopping clock soundwire: cadence: merge routines to clear/set bits soundwire: cadence: move clock/SSP related inits to dedicated function soundwire: cadence: make SSP interval programmable soundwire: cadence: reorder MCP_CONFIG settings soundwire: cadence: enable NORMAL operation in cdns_init() soundwire: cadence: remove PREQ_DELAY assignment soundwire: cadence: remove automatic command retries soundwire: cadence: commit changes in the exit_reset() sequence soundwire: cadence: multi-link support Rander Wang (4): soundwire: cadence: simplifiy cdns_init() soundwire: cadence: add interface to check clock status soundwire: cadence: add clock_stop/restart routines soundwire: cadence: fix a io timeout issue in S3 test drivers/soundwire/cadence_master.c | 297 ++++++++++++++++++++++++----- drivers/soundwire/cadence_master.h | 9 +- drivers/soundwire/intel.c | 2 +- 3 files changed, 261 insertions(+), 47 deletions(-) base-commit: 5de79ba865d7770c3bdde7c266ed425832764aac -- 2.20.1

2 26

[alsa-devel] [PATCH 0/6] ASoC: soc-acpi: add support for SoundWire-based machines
by Pierre-Louis Bossart 14 Mar '20

14 Mar '20

SoundWire support on Intel platforms relies on ACPI _ADR fields exposed in DSDT tables, with a 64-bit value defining which link a Slave device is connected to and its partId/manufacturerID/uniqueID. The existing definitions based on a globally-visible _HID used for I2C/I2S are no longer sufficient and need to be extended to detect during the probe steps which machine driver should be selected. This patchset adds the structure changes, and a set of static tables already used by Intel developers and customers. These tables are provided ahead of time, but they will only be used once all SoundWire-related parts are merged upstream and SoundWire is enabled in the builds. For reference, the complete set of 100+ patches required for SoundWire on Intel platforms is available here: https://github.com/thesofproject/linux/pull/1692 Bard Liao (1): ASoC: Intel: common: add match tables for ICL w/ SoundWire Pierre-Louis Bossart (4): ASoC: soc-acpi: add _ADR-based link descriptors ASoC: Intel: common: soc-acpi: declare new tables for SoundWire ASoC: Intel: common: add match tables for TGL w/ SoundWire ASoC: SOF: Intel: reference SoundWire machine lists Rander Wang (1): ASoC: Intel: common: add match tables for CNL/CFL/CML w/ SoundWire include/sound/soc-acpi-intel-match.h | 6 ++ include/sound/soc-acpi.h | 21 ++++ .../intel/common/soc-acpi-intel-cfl-match.c | 5 + .../intel/common/soc-acpi-intel-cml-match.c | 89 +++++++++++++++++ .../intel/common/soc-acpi-intel-cnl-match.c | 5 + .../intel/common/soc-acpi-intel-icl-match.c | 98 +++++++++++++++++++ .../intel/common/soc-acpi-intel-tgl-match.c | 51 +++++++++- sound/soc/sof/sof-pci-dev.c | 5 + 8 files changed, 278 insertions(+), 2 deletions(-) base-commit: b2e2a13796889e10ba0390ab338949ba63d89462 -- 2.20.1

4 20

[PATCH 3/3] ALSA USB MIDI: Make amount of MIDI events per output URB selectable by user
by Andreas Steinmetz 14 Mar '20

14 Mar '20

This patch introduces a new module parameter "events" which allows userspace to limit the amount of MIDI events per output URB. An "events" value of 0 which is the default means "use wMaxPacketSize". Out of range values for "events" are clamped to the nearest valid value. It is assumed that the previous patches are applied and "outqueue=1" is used. Looking then at a four port USB MIDI interface with a wMaxPacketSize of 64 and assuming an optimal output rate of 320us/byte the following is true: 16 MIDI events PER URB Up to 3 bytes per MIDI event (3 bytes are assumed below) A driver queue of 1 URB which is completely filled. Resulting latency is 16*3*1*320us=15.36ms which still is in the audible range (a latency of less than 5ms is required, think playing a keyboard to MIDI in and using MIDI out to control a piano module and using headphones, while using another port for sysex transfers). After applying this patch and selecting the valid minimum event value of 4 (there must be at least space for one event per port to assert proper output balancing) tests show that the throughput isn't really affected, the worst case latency however drops to 3.84ms which is acceptable. In case of the largest class compliant devices which do have 16 ports the worst case latency is then 15.36ms which is tolerable. If this is still to long it could be fixed by introducing additional complexity to the balancing of snd_usbmidi_standard_output() at some later stage. As the device default is used by default no implications are expected for existing users. By making the parameter writable via sysfs it is possible to do selective output URB event count management from userspace via a script that manages USB authorize and the module option. Signed-off-by: Andreas Steinmetz <ast(a)domdv.de> --- a/sound/usb/midi.c 2020-03-13 20:18:33.443265194 +0100 +++ b/sound/usb/midi.c 2020-03-13 20:42:49.445546326 +0100 @@ -79,9 +79,12 @@ MODULE_LICENSE("Dual BSD/GPL"); static int outqueue = OUTPUT_URBS; +static int events; module_param(outqueue, int, 0644); -MODULE_PARM_DESC(outqueue, "Outgoing queue size, 1-7 (default: 7)."); +MODULE_PARM_DESC(outqueue, "Outgoing MIDI queue size, 1-7 (default: 7)."); +module_param(events, int, 0644); +MODULE_PARM_DESC(events, "MIDI events per queue element, 0=device default (default: 0)."); struct usb_ms_header_descriptor { @@ -1426,6 +1429,17 @@ switch (umidi->usb_id) { default: ep->max_transfer = usb_maxpacket(umidi->dev, pipe, 1); + if (events) { + int ev = 0; + + for (i = 0; i < 0x10; ++i) + if (ep_info->out_cables & (1 << i)) + ev++; + if (events > ev) + ev = events; + if (ev < ep->max_transfer >> 2) + ep->max_transfer = ev << 2; + } break; /* * Various chips declare a packet size larger than 4 bytes, but

1 0

[PATCH 2/3] ALSA USB MIDI: Make amount of output URBs selectable by user
by Andreas Steinmetz 14 Mar '20

14 Mar '20

This patch introduces a new module parameter "outqueue" which allows userspace to set the amount of output URBs is a range from 1 to 7 with 7 being the default (illegal values are clamped to the next valid value). It is assumed that the previous patch has been applied. Looking then at a multiport USB MIDI interface with a wMaxPacketSize of 64 and assuming an optimal output rate of 320us/byte the following is true: 16 MIDI events PER URB Up to 3 bytes per MIDI event (3 bytes are assumed below) A driver queue of 7 URBs which are completely filled. Resulting latency is 16*3*7*320us=107.52ms So if all URBs are filled by e.g. a sysex transfer on one port, a realtime message delivered by userspace for another port will be delayed by 107.52ms which is far too long. After applying this patch and selecting a queue of only 1 URB tests show that the throughput isn't really affected, the latency however drops to 16*3*1*320us=15.36ms which is considerably better. As the default is not touched no implications are expected for existing users. By making the parameter writable via sysfs it is possible to do selective output URB queue size management from userspace via a script that manages USB authorize and the module option. Signed-off-by: Andreas Steinmetz <ast(a)domdv.de> --- a/sound/usb/midi.c 2020-03-13 19:19:28.614798593 +0100 +++ b/sound/usb/midi.c 2020-03-13 19:37:08.097411436 +0100 @@ -45,6 +45,7 @@ #include <linux/slab.h> #include <linux/timer.h> #include <linux/usb.h> +#include <linux/moduleparam.h> #include <linux/wait.h> #include <linux/usb/audio.h> #include <linux/module.h> @@ -77,6 +78,11 @@ MODULE_DESCRIPTION("USB Audio/MIDI helper module"); MODULE_LICENSE("Dual BSD/GPL"); +static int outqueue = OUTPUT_URBS; + +module_param(outqueue, int, 0644); +MODULE_PARM_DESC(outqueue, "Outgoing queue size, 1-7 (default: 7)."); + struct usb_ms_header_descriptor { __u8 bLength; @@ -141,6 +147,7 @@ } urbs[OUTPUT_URBS]; unsigned int active_urbs; unsigned int drain_urbs; + unsigned int out_urbs; int max_transfer; /* size of urb buffer */ struct tasklet_struct tasklet; unsigned int next_urb; @@ -335,7 +342,7 @@ break; ep->active_urbs |= 1 << urb_index; } - if (++urb_index >= OUTPUT_URBS) + if (++urb_index >= ep->out_urbs) urb_index = 0; if (urb_index == ep->next_urb) break; @@ -1364,7 +1371,7 @@ { unsigned int i; - for (i = 0; i < OUTPUT_URBS; ++i) + for (i = 0; i < ep->out_urbs; ++i) if (ep->urbs[i].urb) { free_urb_and_buffer(ep->umidi, ep->urbs[i].urb, ep->max_transfer); @@ -1397,7 +1404,14 @@ return -ENOMEM; ep->umidi = umidi; - for (i = 0; i < OUTPUT_URBS; ++i) { + if (outqueue < 1) + ep->out_urbs = 1; + else if (outqueue > OUTPUT_URBS) + ep->out_urbs = OUTPUT_URBS; + else + ep->out_urbs = outqueue; + + for (i = 0; i < ep->out_urbs; ++i) { ep->urbs[i].urb = usb_alloc_urb(0, GFP_KERNEL); if (!ep->urbs[i].urb) { err = -ENOMEM; @@ -1434,7 +1448,7 @@ ep->max_transfer = 9; break; } - for (i = 0; i < OUTPUT_URBS; ++i) { + for (i = 0; i < ep->out_urbs; ++i) { buffer = usb_alloc_coherent(umidi->dev, ep->max_transfer, GFP_KERNEL, &ep->urbs[i].urb->transfer_dma); @@ -1525,7 +1539,7 @@ if (ep->out) tasklet_kill(&ep->out->tasklet); if (ep->out) { - for (j = 0; j < OUTPUT_URBS; ++j) + for (j = 0; j < ep->out->out_urbs; ++j) usb_kill_urb(ep->out->urbs[j].urb); if (umidi->usb_protocol_ops->finish_out_endpoint) umidi->usb_protocol_ops->finish_out_endpoint(ep->out);

1 0

BUG: unable to handle kernel NULL pointer dereference in corrupted (5)
by syzbot 14 Mar '20

14 Mar '20

Hello, syzbot found the following crash on: HEAD commit: 30bb5572 Merge tag 'ktest-v5.6' of git://git.kernel.org/pu.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=15d300b1e00000 kernel config: https://syzkaller.appspot.com/x/.config?x=a5295e161cd85b82 dashboard link: https://syzkaller.appspot.com/bug?extid=8b0e78e390d1715b0f4e compiler: clang version 10.0.0 (https://github.com/llvm/llvm-project/ c2443155a0fb245c8f17f2c1c72b6ea391e86e81) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=11a57709e00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=13de9e91e00000 The bug was bisected to: commit 271213ef4d0d3a3b80d4cf95c5f2bebb5643e666 Author: Takashi Iwai <tiwai(a)suse.de> Date: Tue Dec 10 06:34:50 2019 +0000 ALSA: pcxhr: Support PCM sync_stop bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=1566b08de00000 final crash: https://syzkaller.appspot.com/x/report.txt?x=1766b08de00000 console output: https://syzkaller.appspot.com/x/log.txt?x=1366b08de00000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+8b0e78e390d1715b0f4e(a)syzkaller.appspotmail.com Fixes: 271213ef4d0d ("ALSA: pcxhr: Support PCM sync_stop") BUG: kernel NULL pointer dereference, address: 0000000000000086 #PF: supervisor instruction fetch in kernel mode #PF: error_code(0x0010) - not-present page PGD a27fa067 P4D a27fa067 PUD a2185067 PMD 0 Oops: 0010 [#1] PREEMPT SMP KASAN CPU: 1 PID: 9168 Comm: syz-executor418 Not tainted 5.6.0-rc5-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 RIP: 0010:0x86 Code: Bad RIP value. RSP: 0018:ffffc90002c9f998 EFLAGS: 00010086 RAX: ffffc90002c9f9c8 RBX: fffffe0000000000 RCX: ffff8880a31b6280 RDX: 0000000000000000 RSI: 0000000000000001 RDI: 0000000000000000 RBP: 0000000000000ec0 R08: ffffffff839888d3 R09: ffffffff811c7d9a R10: ffff8880a31b6280 R11: 0000000000000002 R12: dffffc0000000000 R13: fffffe0000000ec8 R14: ffffffff880016f0 R15: fffffe0000000ecb FS: 0000000000981880(0000) GS:ffff8880ae900000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 000000000000005c CR3: 00000000986d2000 CR4: 00000000001426e0 Call Trace: Modules linked in: CR2: 0000000000000086 ---[ end trace 7c78bc94cfc0a37d ]--- RIP: 0010:0x86 Code: Bad RIP value. RSP: 0018:ffffc90002c9f998 EFLAGS: 00010086 RAX: ffffc90002c9f9c8 RBX: fffffe0000000000 RCX: ffff8880a31b6280 RDX: 0000000000000000 RSI: 0000000000000001 RDI: 0000000000000000 RBP: 0000000000000ec0 R08: ffffffff839888d3 R09: ffffffff811c7d9a R10: ffff8880a31b6280 R11: 0000000000000002 R12: dffffc0000000000 R13: fffffe0000000ec8 R14: ffffffff880016f0 R15: fffffe0000000ecb FS: 0000000000981880(0000) GS:ffff8880ae900000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 000000000000005c CR3: 00000000986d2000 CR4: 00000000001426e0 --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. For information about bisection process see: https://goo.gl/tpsmEJ#bisection syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

3 2

[PATCH v2] ASoC: dt-bindings: google, cros-ec-codec: Fix dtc warnings in example
by Rob Herring 13 Mar '20

13 Mar '20

Extra dtc warnings (roughly what W=1 enables) are now enabled by default when building the binding examples. These were fixed treewide in 5.6-rc5, but the newly added google,cros-ec-codec schema adds some new warnings: Documentation/devicetree/bindings/sound/google,cros-ec-codec.example.dts:17.28-21.11: Warning (unit_address_vs_reg): /example-0/reserved_mem: node has a reg or ranges property, but no unit name Documentation/devicetree/bindings/sound/google,cros-ec-codec.example.dts:22.19-32.11: Warning (unit_address_vs_reg): /example-0/cros-ec@0: node has a unit name, but no reg property Documentation/devicetree/bindings/sound/google,cros-ec-codec.example.dts:26.37-31.15: Warning (unit_address_vs_reg): /example-0/cros-ec@0/ec-codec: node has a reg or ranges property, but no unit name Fixing the above, then results in: Documentation/devicetree/bindings/sound/google,cros-ec-codec.example.dts:26.13-23: Warning (reg_format): /example-0/cros-ec@0:reg: property has invalid length (4 bytes) (#address-cells == 1, #size-cells == 1) Documentation/devicetree/bindings/sound/google,cros-ec-codec.example.dts:27.37-32.15: Warning (unit_address_vs_reg): /example-0/cros-ec@0/ec-codec: node has a reg or ranges property, but no unit name Fix these warnings and adjust the node name to 'audio-codec' while we're touching the node name. Fixes: eadd54c75f1e ("dt-bindings: Convert the binding file google, cros-ec-codec.txt to yaml format.") Cc: Cheng-Yi Chiang <cychiang(a)chromium.org> Cc: Guenter Roeck <groeck(a)chromium.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Benson Leung <bleung(a)chromium.org> Cc: alsa-devel(a)alsa-project.org Reviewed-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> Signed-off-by: Rob Herring <robh(a)kernel.org> --- v2: - s/ec-codec/audio-codec/ --- .../bindings/sound/google,cros-ec-codec.yaml | 27 +++++++++++-------- 1 file changed, 16 insertions(+), 11 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml index 94a85d0cbf43..d62284288f23 100644 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -44,19 +44,24 @@ additionalProperties: false examples: - | - reserved_mem: reserved_mem { + reserved_mem: reserved-mem@52800000 { compatible = "shared-dma-pool"; - reg = <0 0x52800000 0 0x100000>; + reg = <0x52800000 0x100000>; no-map; }; - cros-ec@0 { - compatible = "google,cros-ec-spi"; - #address-cells = <2>; - #size-cells = <1>; - cros_ec_codec: ec-codec { - compatible = "google,cros-ec-codec"; - #sound-dai-cells = <1>; - reg = <0x0 0x10500000 0x80000>; - memory-region = <&reserved_mem>; + spi { + #address-cells = <1>; + #size-cells = <0>; + cros-ec@0 { + compatible = "google,cros-ec-spi"; + #address-cells = <2>; + #size-cells = <1>; + reg = <0>; + cros_ec_codec: audio-codec@10500000 { + compatible = "google,cros-ec-codec"; + #sound-dai-cells = <1>; + reg = <0x0 0x10500000 0x80000>; + memory-region = <&reserved_mem>; + }; }; }; -- 2.20.1

2 2

[PATCH] ASoC: max98357a: Add ACPI HID MAX98360A
by Yong Zhi 13 Mar '20

13 Mar '20

Maxim MAX98360A audio amplifier is functionally identical to MAX98357A, add ACPI ID "MAX98360A" for driver reuse. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/codecs/max98357a.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 74f20114297c..a8bd793a7867 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -133,6 +133,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id); #ifdef CONFIG_ACPI static const struct acpi_device_id max98357a_acpi_match[] = { { "MX98357A", 0 }, + { "MX98360A", 0 }, {}, }; MODULE_DEVICE_TABLE(acpi, max98357a_acpi_match); -- 2.7.4

2 1

alsatplg (libasound.a) segmentation fault using AFL
by GitHub issues - opened 13 Mar '20

13 Mar '20

alsa-project/alsa-lib issue #37 was opened from tysonite: I was playing around with [AFL](https://fuzzing-project.org/tutorial3.html) tonight on one of my pet projects. And after it found few crashes, I've decided to fuzz one of open-source projects. The `alsatplg` tool just looked simple enough to exercise it with fuzzing tool. I made a simple Dockerfile that runs AFL on `alsatplg`: ``` FROM ubuntu:18.04 ENV LANG C.UTF-8 RUN apt-get update && \ apt-get install -y apt-utils && \ apt-get install -y afl git build-essential m4 autoconf automake libtool RUN cd / RUN git clone https://github.com/alsa-project/alsa-lib.git RUN cd alsa-lib && \ libtoolize --force --copy --automake && \ aclocal && \ autoheader && \ automake --foreign --copy --add-missing && \ autoconf && \ export CFLAGS="-O2 -Wall -W -Wunused-const-variable=0 -pipe -g" && \ export CC=afl-gcc && \ ./configure --disable-aload && \ make && \ make install \ && cd / RUN apt-get install -y gettext ncurses-base libncurses5 libncurses5-dev pkg-config RUN git clone https://github.com/alsa-project/alsa-utils.git RUN cd alsa-utils && \ export CC=afl-gcc && \ ./gitcompile && \ make install && \ cd / RUN mkdir in #RUN cp alsa-utils/speaker-test/samples/Noise.wav in RUN echo "Hello" > in/input.txt CMD ["afl-fuzz", "-i", "in", "-o", "out", "alsatplg", "-c", "@@", "-o", "/output"] ``` After around 10-15 minutes running on my core i7 laptop, it generated a sequence of bytes that leads to crash. If you want to try it by yourself just run `docker build -t alsa/dev .` followed by `docker run alsa/dev`, and wait a bit. When crash happened, the input data can be copied from the container by running `docker cp <container_id>:/out .`. An example of input data that lead to SIGSEGV: [id:000000,sig:11,src:000325,op:arith8,pos:48,val:-26.txt](https://github.co… And stack trace based on it: ``` Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `alsatplg -c out/crashes/id:000000,sig:11,src:000325,op:arith8,pos:48,val:-26 -o'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007fcb65e05ca8 in snd_config_delete () from /usr/lib/x86_64-linux-gnu/libasound.so.2 (gdb) bt #0 0x00007fcb65e05ca8 in snd_config_delete () from /usr/lib/x86_64-linux-gnu/libasound.so.2 #1 0x00007fcb65e06479 in ?? () from /usr/lib/x86_64-linux-gnu/libasound.so.2 #2 0x00007fcb65e064ba in ?? () from /usr/lib/x86_64-linux-gnu/libasound.so.2 #3 0x00007fcb65e0661c in ?? () from /usr/lib/x86_64-linux-gnu/libasound.so.2 #4 0x00007fcb65e818c4 in snd_tplg_build_file () from /usr/lib/x86_64-linux-gnu/libasound.so.2 #5 0x00005587bce0ab6a in ?? () #6 0x00007fcb65a07b97 in __libc_start_main (main=0x5587bce0aa10, argc=5, argv=0x7ffcfa707628, init=<optimized out>, fini=<optimized out>, rtld_fini=<optimized out>, stack_end=0x7ffcfa707618) at ../csu/libc-start.c:310 #7 0x00005587bce0ac4a in ?? () (gdb) bt full #0 0x00007fcb65e05ca8 in snd_config_delete () from /usr/lib/x86_64-linux-gnu/libasound.so.2 No symbol table info available. #1 0x00007fcb65e06479 in ?? () from /usr/lib/x86_64-linux-gnu/libasound.so.2 No symbol table info available. #2 0x00007fcb65e064ba in ?? () from /usr/lib/x86_64-linux-gnu/libasound.so.2 No symbol table info available. #3 0x00007fcb65e0661c in ?? () from /usr/lib/x86_64-linux-gnu/libasound.so.2 No symbol table info available. #4 0x00007fcb65e818c4 in snd_tplg_build_file () from /usr/lib/x86_64-linux-gnu/libasound.so.2 No symbol table info available. #5 0x00005587bce0ab6a in ?? () No symbol table info available. #6 0x00007fcb65a07b97 in __libc_start_main (main=0x5587bce0aa10, argc=5, argv=0x7ffcfa707628, init=<optimized out>, fini=<optimized out>, rtld_fini=<optimized out>, stack_end=0x7ffcfa707618) at ../csu/libc-start.c:310 self = <optimized out> __self = <optimized out> result = <optimized out> unwind_buf = {cancel_jmp_buf = {{jmp_buf = {0, -5452963434713232627, 94041477786656, 140724510160416, 0, 0, -2259219850243519731, -2248813385476519155}, mask_was_saved = 0}}, priv = {pad = {0x0, 0x0, 0x7fcb660ee733 <_dl_init+259>, 0x7fcb660d6370}, data = {prev = 0x0, cleanup = 0x0, canceltype = 1712252723}}} not_first_call = <optimized out> #7 0x00005587bce0ac4a in ?? () No symbol table info available. ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/37 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/6] ASoC: SOF: multi-cpu dais, IPC and Intel improvements for 5.7
by Pierre-Louis Bossart 13 Mar '20

13 Mar '20

Corrections for compatibility with the multi-cpu dai check for min-channels, IPC simplifications, and removal of unnecessary boot-related code for Intel platforms. Amery Song (2): ASoC: SOF: Intel: hda: remove unnecessary ROM IPC filter function ASoC: SOF: Intel: remove unnecessary waitq before loading firmware Bard Liao (2): ASoC: SOF: Intel: hda-dai: add stream capability ASoC: SOF: topology: connect dai widget to all cpu-dais Karol Trzcinski (2): ASoC: SOF: Make sof_ipc_ext_data enum more rigid ASoC: SOF: Remove SOF_IPC_EXT_DMA_BUFFER include/sound/sof/info.h | 22 +------- sound/soc/sof/intel/cnl.c | 5 -- sound/soc/sof/intel/hda-dai.c | 96 ++++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda-ipc.c | 20 +------ sound/soc/sof/intel/hda-loader.c | 3 - sound/soc/sof/intel/hda.c | 3 - sound/soc/sof/intel/hda.h | 1 - sound/soc/sof/loader.c | 6 -- sound/soc/sof/sof-priv.h | 5 -- sound/soc/sof/topology.c | 8 ++- 10 files changed, 108 insertions(+), 61 deletions(-) base-commit: 101247a3b86e1cc0e382b7e887a56176290fc957 -- 2.20.1

2 13

[PATCH 00/10] ASoC: SOF: Intel: add SoundWire support
by Pierre-Louis Bossart 13 Mar '20

13 Mar '20

This patchset provides the support for SoundWire support on Intel CometLake, IcelLake and TigerLake RVP platforms and form-factor devices to be released 'soon'. The bulk of the code is about detecting a valid SoundWire configuration from ACPI, and implementing the interfaces suggested in '[PATCH 0/8] soundwire: remove platform devices, add SOF interfaces' for interrupts, PCI wakes and clock-stop configurations. Since that SoundWire series is stuck with no resolution, the build support for SOF w/ SoundWire is not provided for now, and fall-back functions will be used. This code is tested on a daily basis in the SOF tree and is not expected to change. If audio maintainers will only accept functional code, which isn't unreasonable, I would kindly ask that they reach out to Vinod Koul. Bard Liao (1): ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers Pierre-Louis Bossart (7): ASoC: soc-acpi: expand description of _ADR-based devices ASoC: SOF: Intel: add SoundWire configuration interface ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks Rander Wang (2): ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread include/sound/soc-acpi.h | 39 +- include/sound/sof/dai-intel.h | 18 +- .../intel/common/soc-acpi-intel-cml-match.c | 87 +++- .../intel/common/soc-acpi-intel-icl-match.c | 97 ++++- .../intel/common/soc-acpi-intel-tgl-match.c | 49 ++- sound/soc/sof/intel/hda-dsp.c | 2 + sound/soc/sof/intel/hda-loader.c | 31 ++ sound/soc/sof/intel/hda.c | 399 ++++++++++++++++++ sound/soc/sof/intel/hda.h | 66 +++ 9 files changed, 728 insertions(+), 60 deletions(-) base-commit: 101247a3b86e1cc0e382b7e887a56176290fc957 -- 2.20.1

2 12

Applied "ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI" to the asoc tree
by Mark Brown 13 Mar '20

13 Mar '20

The patch ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 5c82813ce43e3a2017c43b32a57dc7a8802d9ad4 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Date: Thu, 12 Mar 2020 14:48:50 -0500 Subject: [PATCH] ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI Having a reverse dependency to a config that has its own additional dependencies, is generally not recommended. And this applies to select statements for SND_HDA_CODEC_HDMI, e.g. the case where SND_HDA and SND_SOC_SOF_HDA are built as modules, but the machine driver is built-in, leading to compile errors (reported as i386-randconfig-e003-20200206). Give up on trying to define different dependencies based on SOF/SST selection, and simply add a "depends on" for SND_HDA_CODEC_HDMI. This fixes the issue with randconfigs. Only downside is that SND_HDA_CODEC_HDMI may be built unnecessarily in some cases, but this seems like the lesser evil. Fixes: aa2b4a5 ('ASoC: Intel: boards: fix incorrect HDMI Kconfig dependency') Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200312194859.4051-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/Kconfig | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 755e1de19df9..67d85a7be559 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -289,7 +289,6 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_DMIC - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_HDAC_HDMI config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON @@ -302,6 +301,7 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH tristate "Broxton with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON help This adds support for ASoC machine driver for Broxton-P platforms @@ -402,6 +402,7 @@ config SND_SOC_INTEL_GLK_DA7219_MAX98357A_MACH tristate "GLK with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON help This adds support for ASoC machine driver for Geminilake platforms @@ -413,10 +414,10 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH tristate "GLK with RT5682 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_RT5682 select SND_SOC_MAX98357A select SND_SOC_DMIC - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_HDAC_HDMI help This adds support for ASoC machine driver for Geminilake platforms @@ -430,7 +431,7 @@ if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH tristate "SKL/KBL/BXT/APL with HDA Codecs" - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC + depends on SND_HDA_CODEC_HDMI select SND_SOC_HDAC_HDMI select SND_SOC_DMIC # SND_SOC_HDAC_HDA is already selected @@ -448,9 +449,9 @@ config SND_SOC_INTEL_SOF_RT5682_MACH depends on I2C && ACPI depends on (SND_SOC_SOF_HDA_LINK && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) + depends on SND_HDA_CODEC_HDMI select SND_SOC_RT5682 select SND_SOC_DMIC - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_HDAC_HDMI help This adds support for ASoC machine driver for SOF platforms @@ -490,11 +491,11 @@ config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH tristate "CML with RT1011 and RT5682 in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_RT1011 select SND_SOC_RT5682 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC help This adds support for ASoC machine driver for SOF platform with RT1011 + RT5682 I2S codec. @@ -509,10 +510,10 @@ config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH tristate "SOF with DA7219 and MAX98373 in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_DA7219 select SND_SOC_MAX98373 select SND_SOC_DMIC - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC help This adds support for ASoC machine driver for SOF platforms with DA7219 + MAX98373 I2S audio codec. -- 2.20.1

1 0

Applied "ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI" to the asoc tree
by Mark Brown 13 Mar '20

13 Mar '20

The patch ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4399afd21a0169a125a16c2dbba68f7d9a2cffdb Mon Sep 17 00:00:00 2001 From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Date: Thu, 12 Mar 2020 14:48:51 -0500 Subject: [PATCH] ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI Having a reverse dependency to a config that has its own additional dependencies, is generally not recommended. And this applies to select statements for SND_HDA_CODEC_HDMI, e.g. the case where SND_HDA and SND_SOC_SOF_HDA are built as modules, but the machine driver is built-in, leading to compile errors (reported as i386-randconfig-e003-20200206). Give up on trying to define different dependencies based on SOF/SST selection, and simply add a "depends on" for SND_HDA_CODEC_HDMI. This fixes the issue with randconfigs. Only downside is that SND_HDA_CODEC_HDMI may be built unnecessarily in some cases, but this seems like the lesser evil. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200312194859.4051-3-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 67d85a7be559..6833ef548710 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -464,8 +464,8 @@ config SND_SOC_INTEL_SOF_PCM512x_MACH depends on I2C && ACPI depends on (SND_SOC_SOF_HDA_AUDIO_CODEC && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) + depends on SND_HDA_CODEC_HDMI select SND_SOC_PCM512x_I2C - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC help This adds support for ASoC machine driver for SOF platforms with TI PCM512x I2S audio codec. -- 2.20.1

1 0

Applied "ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms" to the asoc tree
by Mark Brown 13 Mar '20

13 Mar '20

The patch ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 15a5a89597e57e67d4dde1d57fa85105c5930bb3 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Date: Thu, 12 Mar 2020 14:48:52 -0500 Subject: [PATCH] ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms Make HDMI optional for APL and later platforms. If no HDMI codec is found on the HDA bus, the graphics side driver is missing or correct codec driver is not part of kernel build, codec_mask reflects this and HDMI is disabled. The DSP topology will still have the links for HDMI, so connect these to dummy codec to avoid failures in topology loading. This change also fixes a kernel oops that was triggered if sof_pcm512x was used with SOF configured to use hdac-hdmi (can be done via "use_common_hdmi=0" or by selecting CONFIG_SND_SOC_SOF_HDA_COMMON_HDMI_CODEC=n). This is not a supported configuration. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200312194859.4051-4-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/sof_pcm512x.c | 38 +++++++++++++++++++++------- 1 file changed, 29 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c index 626153bd71e7..4ce707b6eb79 100644 --- a/sound/soc/intel/boards/sof_pcm512x.c +++ b/sound/soc/intel/boards/sof_pcm512x.c @@ -27,6 +27,8 @@ #define SOF_PCM512X_SSP_CODEC(quirk) ((quirk) & GENMASK(3, 0)) #define SOF_PCM512X_SSP_CODEC_MASK (GENMASK(3, 0)) +#define IDISP_CODEC_MASK 0x4 + /* Default: SSP5 */ static unsigned long sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(5); @@ -40,6 +42,7 @@ struct sof_hdmi_pcm { struct sof_card_private { struct list_head hdmi_pcm_list; + bool idisp_codec; }; static int sof_pcm512x_quirk_cb(const struct dmi_system_id *id) @@ -136,6 +139,9 @@ static int sof_card_late_probe(struct snd_soc_card *card) if (list_empty(&ctx->hdmi_pcm_list)) return -EINVAL; + if (!ctx->idisp_codec) + return 0; + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, head); return hda_dsp_hdmi_build_controls(card, pcm->codec_dai->component); @@ -214,7 +220,8 @@ SND_SOC_DAILINK_DEF(dmic_component, static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int dmic_be_num, - int hdmi_num) + int hdmi_num, + bool idisp_codec) { struct snd_soc_dai_link_component *idisp_components; struct snd_soc_dai_link_component *cpus; @@ -316,11 +323,19 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, if (!links[id].cpus->dai_name) goto devm_err; - idisp_components[i - 1].name = "ehdaudio0D2"; - idisp_components[i - 1].dai_name = devm_kasprintf(dev, - GFP_KERNEL, - "intel-hdmi-hifi%d", - i); + /* + * topology cannot be loaded if codec is missing, so + * use the dummy codec if needed + */ + if (idisp_codec) { + idisp_components[i - 1].name = "ehdaudio0D2"; + idisp_components[i - 1].dai_name = + devm_kasprintf(dev, GFP_KERNEL, + "intel-hdmi-hifi%d", i); + } else { + idisp_components[i - 1].name = "snd-soc-dummy"; + idisp_components[i - 1].dai_name = "snd-soc-dummy-dai"; + } if (!idisp_components[i - 1].dai_name) goto devm_err; @@ -341,8 +356,8 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, static int sof_audio_probe(struct platform_device *pdev) { + struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct snd_soc_dai_link *dai_links; - struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; int ret, ssp_codec; @@ -360,6 +375,11 @@ static int sof_audio_probe(struct platform_device *pdev) } else { dmic_be_num = 2; #if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) + if (mach->mach_params.common_hdmi_codec_drv && + (mach->mach_params.codec_mask & IDISP_CODEC_MASK)) + ctx->idisp_codec = true; + + /* links are always present in topology */ hdmi_num = 3; #endif } @@ -374,7 +394,8 @@ static int sof_audio_probe(struct platform_device *pdev) sof_audio_card_pcm512x.num_links = 1 + dmic_be_num + hdmi_num; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, - dmic_be_num, hdmi_num); + dmic_be_num, hdmi_num, + ctx->idisp_codec); if (!dai_links) return -ENOMEM; @@ -383,7 +404,6 @@ static int sof_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); sof_audio_card_pcm512x.dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; /* set platform name for each dailink */ ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_pcm512x, -- 2.20.1

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