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

Functionality of pcm_notify in snd-aloop?
by Pavel Hofman 30 Apr '20

30 Apr '20

Hi, Please how is the module params pcm_notify supposed to be used, to do what the documentation says: Break capture when PCM format/rate/channels changes? Breaking capture side operation when the playback side changes the params is very useful, but I cannot find a way to use this param properly. When the capture side is open, the playback side cannot use a different parameter than the one currently used by the capture side (the configuration space is limited) - how can the param be changed then? Thanks a lot for help. Best regards, Pavel.

4 9

[PATCH] ALSA: pcm: oss: Avoid plugin buffer overflow
by Takashi Iwai 30 Apr '20

30 Apr '20

Each OSS PCM plugins allocate its internal buffer per pre-calculation of the max buffer size through the chain of plugins (calling src_frames and dst_frames callbacks). This works for most plugins, but the rate plugin might behave incorrectly. The calculation in the rate plugin involves with the fractional position, i.e. it may vary depending on the input position. Since the buffer size pre-calculation is always done with the offset zero, it may return a shorter size than it might be; this may result in the out-of-bound access as spotted by fuzzer. This patch addresses those possible buffer overflow accesses by simply setting the upper limit per the given buffer size for each plugin before src_frames() and after dst_frames() calls. Reported-by: syzbot+e1fe9f44fb8ecf4fb5dd(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000b25ea005a02bcf21@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_plugin.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 240e4702c098..c9401832967c 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -209,6 +209,8 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p if (stream == SNDRV_PCM_STREAM_PLAYBACK) { plugin = snd_pcm_plug_last(plug); while (plugin && drv_frames > 0) { + if (drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) drv_frames = plugin->src_frames(plugin, drv_frames); @@ -220,6 +222,8 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p plugin_next = plugin->next; if (plugin->dst_frames) drv_frames = plugin->dst_frames(plugin, drv_frames); + if (drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin = plugin_next; } } else @@ -248,11 +252,15 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc if (frames < 0) return frames; } + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_next; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_last(plug); while (plugin) { + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); -- 2.16.4

2 3

[RFC PATCH v2 0/2] Introduce multi PM domains helpers
by Daniel Baluta 21 Apr '20

21 Apr '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> i.MX8QXP/i.MX8QM has IPs that need multiple power domains to be up in order to work. In order to help drivers, we introduce multi PM domains helpers that are able to activate/deactivate multi PM domains. First patch introduces the helpers and second patch demonstrates how a driver can use them instead of hardcoding the PM domains handling. Changes since v1: (addressed Ranjani's comments) - enhanced description for dev_multi_pm_attach return value - renamed exit_unroll_pm label to exit_detach_pm Daniel Baluta (2): PM / domains: Introduce multi PM domains helpers ASoC: SOF: Use multi PM domains helpers drivers/base/power/common.c | 93 +++++++++++++++++++++++++++++++++++++ include/linux/pm_domain.h | 19 ++++++++ sound/soc/sof/imx/imx8.c | 60 ++++-------------------- 3 files changed, 121 insertions(+), 51 deletions(-) -- 2.17.1

4 6

[PATCH v3] ASoC: Intel: boards: split woofer and tweeter support
by mac.chiang＠intel.com 20 Apr '20

20 Apr '20

From: Mac Chiang <mac.chiang(a)intel.com> Support Woofer stereo speakers by default and optionally Tweeter stereo speakers with a DMI quirk Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- Changelog: v2: - considering woofer speakers settings as general. - separate out the tweeter speakers and consistency of woofer+tweeter when hardware-specific. - check NULL condition if devm_kzalloc are failed v3: - fixed multiple alignments - add invalid value return --- sound/soc/intel/boards/cml_rt1011_rt5682.c | 222 +++++++++++++++++++++++------ 1 file changed, 180 insertions(+), 42 deletions(-) diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index ed6c26a..2872345 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 speakers */ +static unsigned long sof_rt1011_quirk = SOF_RT1011_SPEAKER_WL | + SOF_RT1011_SPEAKER_WR; + +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, "Helios"), + }, + .driver_data = (void *)(SOF_RT1011_SPEAKER_WL | SOF_RT1011_SPEAKER_WR | + SOF_RT1011_SPEAKER_TL | SOF_RT1011_SPEAKER_TR), + }, + { + } +}; + 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), @@ -64,10 +95,13 @@ 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"}, - {"TR Ext Spk", NULL, "TR SPO"}, + /*WL/WR speaker*/ {"WL Ext Spk", NULL, "WL SPO"}, {"WR Ext Spk", NULL, "WR SPO"}, @@ -82,6 +116,12 @@ 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 +161,35 @@ 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) { @@ -191,30 +260,38 @@ 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; + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_WL | + SOF_RT1011_SPEAKER_WR)) { + 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; + } } - /* 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-10EC1011:02")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x1, 0x1, 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 (ret < 0) @@ -302,9 +379,7 @@ SND_SOC_DAILINK_DEF(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))); + /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI))); SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); @@ -398,6 +473,7 @@ 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), }, @@ -412,14 +488,6 @@ 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 */ @@ -441,10 +509,12 @@ 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_KERNEL); if (!ctx) @@ -455,6 +525,73 @@ 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); + + if (!rt1011_dais_confs) + return -ENOMEM; + + rt1011_dais_components = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_dai_link_component) * + SPK_CH, GFP_KERNEL); + + if (!rt1011_dais_components) + return -ENOMEM; + + for (i = 0; i < SPK_CH; i++) { + rt1011_dais_confs[i].dlc.name = devm_kasprintf(&pdev->dev, + GFP_KERNEL, + "i2c-10EC1011:0%d", + i); + + if (!rt1011_dais_confs[i].dlc.name) + return -ENOMEM; + + 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; + default: + return -EINVAL; + } + rt1011_dais_components[i].name = devm_kasprintf(&pdev->dev, + GFP_KERNEL, + "i2c-10EC1011:0%d", + i); + if (!rt1011_dais_components[i].name) + return -ENOMEM; + + 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); @@ -482,5 +619,6 @@ 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

2 1

[PATCH 0/2] ASoC: qdsp6: fix default FE dais and routings.
by Srinivas Kandagatla 20 Apr '20

20 Apr '20

QDSP6 Frontend dais can be configured to work in rx or tx or both rx/tx mode, however the default routing do not honour this DT configuration making sound card fail to probe. FE dais are also not fully honouring device tree configuration. Fix both of them. Originally issue was reported by Vinod Koul Srinivas Kandagatla (2): ASoC: qdsp6: q6asm-dai: only enable dais from device tree ASoC: qdsp6: q6routing: remove default routing sound/soc/qcom/qdsp6/q6asm-dai.c | 30 +++++++++++++++++++++++------- sound/soc/qcom/qdsp6/q6routing.c | 19 ------------------- 2 files changed, 23 insertions(+), 26 deletions(-) -- 2.21.0

4 11

[PATCH alsa-lib] conf: USB-Audio: Fix S/PDIF output of ASUS Xonar AE
by Ivan Mironov 17 Apr '20

17 Apr '20

ASUS Xonar AE is a PCI-Express card containing USB controller: USB controller [0c03]: ASMedia Technology Inc. ASM1042A USB 3.0 Host Controller [1b21:1142] and the actual USB sound card: ID 0b05:180f ASUSTek Computer, Inc. XONAR SOUND CARD As other Xonar USB sound cards, it uses second device for digital output. Signed-off-by: Ivan Mironov <mironov.ivan(a)gmail.com> diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 89811086db1d..3f4c421b7e2c 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -40,6 +40,7 @@ USB-Audio.pcm.iec958_device { "USB Sound Blaster HD" 1 "Xonar U7" 1 "ASUS XONAR U5" 1 + "XONAR SOUND CARD" 1 # The below don't have digital in/out, so prevent them from being opened. "Andrea PureAudio USB-SA Headset" 999 -- 2.25.1

2 1

[PATCH v2] slimbus: core: Set fwnode for a device when setting of_node
by Saravana Kannan 17 Apr '20

17 Apr '20

When setting the of_node for a newly created device, also set the fwnode. This allows fw_devlink feature to work for slimbus devices. Signed-off-by: Saravana Kannan <saravanak(a)google.com> --- drivers/slimbus/core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/core.c b/drivers/slimbus/core.c index 526e3215d8fe..44228a5b246d 100644 --- a/drivers/slimbus/core.c +++ b/drivers/slimbus/core.c @@ -163,8 +163,10 @@ static int slim_add_device(struct slim_controller *ctrl, INIT_LIST_HEAD(&sbdev->stream_list); spin_lock_init(&sbdev->stream_list_lock); - if (node) + if (node) { sbdev->dev.of_node = of_node_get(node); + sbdev->dev.fwnode = of_fwnode_handle(node); + } dev_set_name(&sbdev->dev, "%x:%x:%x:%x", sbdev->e_addr.manf_id, -- 2.25.1.696.g5e7596f4ac-goog

3 3

[PATCH v1] slimbus: core: Fix mismatch in of_node_get/put
by Saravana Kannan 17 Apr '20

17 Apr '20

Also, remove some unnecessary NULL checks. The functions in question already do NULL checks. Signed-off-by: Saravana Kannan <saravanak(a)google.com> --- drivers/slimbus/core.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/drivers/slimbus/core.c b/drivers/slimbus/core.c index 44228a5b246d..ae1e248a8fb8 100644 --- a/drivers/slimbus/core.c +++ b/drivers/slimbus/core.c @@ -162,11 +162,8 @@ static int slim_add_device(struct slim_controller *ctrl, sbdev->ctrl = ctrl; INIT_LIST_HEAD(&sbdev->stream_list); spin_lock_init(&sbdev->stream_list_lock); - - if (node) { - sbdev->dev.of_node = of_node_get(node); - sbdev->dev.fwnode = of_fwnode_handle(node); - } + sbdev->dev.of_node = of_node_get(node); + sbdev->dev.fwnode = of_fwnode_handle(node); dev_set_name(&sbdev->dev, "%x:%x:%x:%x", sbdev->e_addr.manf_id, @@ -285,6 +282,7 @@ EXPORT_SYMBOL_GPL(slim_register_controller); /* slim_remove_device: Remove the effect of slim_add_device() */ static void slim_remove_device(struct slim_device *sbdev) { + of_node_put(sbdev->dev.of_node); device_unregister(&sbdev->dev); } -- 2.25.1.696.g5e7596f4ac-goog

2 1

[PATCH] ASoC: qcom: lpass-cpu: support full duplex operation
by Srinivas Kandagatla 16 Apr '20

16 Apr '20

From: Takahide Higuchi <takahidehiguchi(a)gmail.com> This patch fixes a bug where playback on bidirectional I2S interface stops when we start recording on the same interface. We use regmap_update_bits instead of regmap_write so that we will not clear SPKEN and SPKMODE bits when we start/stop recording. Signed-off-by: Takahide Higuchi <takahidehiguchi(a)gmail.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-apq8016.c | 2 ++ sound/soc/qcom/lpass-cpu.c | 24 ++++++++++++++++++------ sound/soc/qcom/lpass-lpaif-reg.h | 2 +- 3 files changed, 21 insertions(+), 7 deletions(-) diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 6575da549237..85079c697faa 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -121,6 +121,8 @@ static struct snd_soc_dai_driver apq8016_lpass_cpu_dai_driver[] = { }, .probe = &asoc_qcom_lpass_cpu_dai_probe, .ops = &asoc_qcom_lpass_cpu_dai_ops, + .symmetric_samplebits = 1, + .symmetric_rates = 1, }, }; diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index dbce7e92baf3..dc8acb380b6f 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -72,6 +72,7 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, snd_pcm_format_t format = params_format(params); unsigned int channels = params_channels(params); unsigned int rate = params_rate(params); + unsigned int mask; unsigned int regval; int bitwidth, ret; @@ -81,6 +82,9 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, return bitwidth; } + mask = LPAIF_I2SCTL_LOOPBACK_MASK | + LPAIF_I2SCTL_WSSRC_MASK | + LPAIF_I2SCTL_BITWIDTH_MASK; regval = LPAIF_I2SCTL_LOOPBACK_DISABLE | LPAIF_I2SCTL_WSSRC_INTERNAL; @@ -100,6 +104,7 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mask |= LPAIF_I2SCTL_SPKMODE_MASK | LPAIF_I2SCTL_SPKMONO_MASK; switch (channels) { case 1: regval |= LPAIF_I2SCTL_SPKMODE_SD0; @@ -127,6 +132,7 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } } else { + mask |= LPAIF_I2SCTL_MICMODE_MASK | LPAIF_I2SCTL_MICMONO_MASK; switch (channels) { case 1: regval |= LPAIF_I2SCTL_MICMODE_SD0; @@ -155,9 +161,9 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, } } - ret = regmap_write(drvdata->lpaif_map, - LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), - regval); + ret = regmap_update_bits(drvdata->lpaif_map, + LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), + mask, regval); if (ret) { dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); return ret; @@ -178,11 +184,17 @@ static int lpass_cpu_daiops_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + unsigned int mask; int ret; - ret = regmap_write(drvdata->lpaif_map, - LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), - 0); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mask = LPAIF_I2SCTL_SPKMODE_MASK; + else + mask = LPAIF_I2SCTL_MICMODE_MASK; + + ret = regmap_update_bits(drvdata->lpaif_map, + LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), + mask, 0); if (ret) dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); diff --git a/sound/soc/qcom/lpass-lpaif-reg.h b/sound/soc/qcom/lpass-lpaif-reg.h index 3d74ae123e9d..7a2b9cf99976 100644 --- a/sound/soc/qcom/lpass-lpaif-reg.h +++ b/sound/soc/qcom/lpass-lpaif-reg.h @@ -56,7 +56,7 @@ #define LPAIF_I2SCTL_MICMODE_6CH (7 << LPAIF_I2SCTL_MICMODE_SHIFT) #define LPAIF_I2SCTL_MICMODE_8CH (8 << LPAIF_I2SCTL_MICMODE_SHIFT) -#define LPAIF_I2SCTL_MIMONO_MASK GENMASK(3, 3) +#define LPAIF_I2SCTL_MICMONO_MASK GENMASK(3, 3) #define LPAIF_I2SCTL_MICMONO_SHIFT 3 #define LPAIF_I2SCTL_MICMONO_STEREO (0 << LPAIF_I2SCTL_MICMONO_SHIFT) #define LPAIF_I2SCTL_MICMONO_MONO (1 << LPAIF_I2SCTL_MICMONO_SHIFT) -- 2.21.0

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[alsa-devel] [PATCH] ASoC: stm32: sai: fix master clock management
by Olivier Moysan 16 Apr '20

16 Apr '20

When master clock is used, master clock rate is set exclusively. Parent clocks of master clock cannot be changed after a call to clk_set_rate_exclusive(). So the parent clock of SAI kernel clock must be set before. Ensure also that exclusive rate operations are balanced in STM32 SAI driver. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 64 +++++++++++++++++++++++++++++++------------ 1 file changed, 47 insertions(+), 17 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 3dd54bc54fa1..77aa38183955 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -70,6 +70,7 @@ #define SAI_IEC60958_STATUS_BYTES 24 #define SAI_MCLK_NAME_LEN 32 +#define SAI_RATE_11K 11025 /** * struct stm32_sai_sub_data - private data of SAI sub block (block A or B) @@ -311,6 +312,25 @@ static int stm32_sai_set_clk_div(struct stm32_sai_sub_data *sai, return ret; } +static int stm32_sai_set_parent_clock(struct stm32_sai_sub_data *sai, + unsigned int rate) +{ + struct platform_device *pdev = sai->pdev; + struct clk *parent_clk = sai->pdata->clk_x8k; + int ret; + + if (!(rate % SAI_RATE_11K)) + parent_clk = sai->pdata->clk_x11k; + + ret = clk_set_parent(sai->sai_ck, parent_clk); + if (ret) + dev_err(&pdev->dev, " Error %d setting sai_ck parent clock. %s", + ret, ret == -EBUSY ? + "Active stream rates conflict\n" : "\n"); + + return ret; +} + static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, unsigned long *prate) { @@ -492,25 +512,29 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai, struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); int ret; - if (dir == SND_SOC_CLOCK_OUT) { + if (dir == SND_SOC_CLOCK_OUT && sai->sai_mclk) { ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, (unsigned int)~SAI_XCR1_NODIV); if (ret < 0) return ret; - dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq); - sai->mclk_rate = freq; + /* If master clock is used, set parent clock now */ + ret = stm32_sai_set_parent_clock(sai, freq); + if (ret) + return ret; - if (sai->sai_mclk) { - ret = clk_set_rate_exclusive(sai->sai_mclk, - sai->mclk_rate); - if (ret) { - dev_err(cpu_dai->dev, - "Could not set mclk rate\n"); - return ret; - } + ret = clk_set_rate_exclusive(sai->sai_mclk, freq); + if (ret) { + dev_err(cpu_dai->dev, + ret == -EBUSY ? + "Active streams have incompatible rates" : + "Could not set mclk rate\n"); + return ret; } + + dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq); + sai->mclk_rate = freq; } return 0; @@ -917,11 +941,13 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, int div = 0, cr1 = 0; int sai_clk_rate, mclk_ratio, den; unsigned int rate = params_rate(params); + int ret; - if (!(rate % 11025)) - clk_set_parent(sai->sai_ck, sai->pdata->clk_x11k); - else - clk_set_parent(sai->sai_ck, sai->pdata->clk_x8k); + if (!sai->sai_mclk) { + ret = stm32_sai_set_parent_clock(sai, rate); + if (ret) + return ret; + } sai_clk_rate = clk_get_rate(sai->sai_ck); if (STM_SAI_IS_F4(sai->pdata)) { @@ -1080,9 +1106,13 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, SAI_XCR1_NODIV); - clk_disable_unprepare(sai->sai_ck); + /* Release mclk rate only if rate was actually set */ + if (sai->mclk_rate) { + clk_rate_exclusive_put(sai->sai_mclk); + sai->mclk_rate = 0; + } - clk_rate_exclusive_put(sai->sai_mclk); + clk_disable_unprepare(sai->sai_ck); spin_lock_irqsave(&sai->irq_lock, flags); sai->substream = NULL; -- 2.7.4

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