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

[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] ALSA: au88x0: remove some dead code
by Jason Yan 17 Apr '20

17 Apr '20

Fix the following gcc warning: sound/pci/au88x0/au88x0_a3ddata.c:62:25: warning: ‘A3dHrirDImpulse’ defined but not used [-Wunused-const-variable=] static const a3d_Hrtf_t A3dHrirDImpulse = { ^~~~~~~~~~~~~~~ sound/pci/au88x0/au88x0_a3ddata.c:50:25: warning: ‘A3dHrirSatTest’ defined but not used [-Wunused-const-variable=] static const a3d_Hrtf_t A3dHrirSatTest = { ^~~~~~~~~~~~~~ sound/pci/au88x0/au88x0_a3ddata.c:33:25: warning: ‘A3dHrirOnes’ defined but not used [-Wunused-const-variable=] static const a3d_Hrtf_t A3dHrirOnes = { ^~~~~~~~~~~ Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/pci/au88x0/au88x0_a3d.c | 23 ------------------- sound/pci/au88x0/au88x0_a3ddata.c | 38 ------------------------------- 2 files changed, 61 deletions(-) diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index 2db183f8826a..4d3b79a34c69 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -512,31 +512,8 @@ static void a3dsrc_ProgramPipe(a3dsrc_t * a) /* Single spike leads to identity transfer function. */ a3dsrc_SetHrtfCurrent(a, A3dHrirImpulse, A3dHrirImpulse); a3dsrc_SetHrtfTarget(a, A3dHrirImpulse, A3dHrirImpulse); - - /* Test: Sounds saturated. */ - //a3dsrc_SetHrtfCurrent(a, A3dHrirSatTest, A3dHrirSatTest); - //a3dsrc_SetHrtfTarget(a, A3dHrirSatTest, A3dHrirSatTest); } -/* VDB = Vortex audio Dataflow Bus */ -#if 0 -static void a3dsrc_ClearVDBData(a3dsrc_t * a, unsigned long aa) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - - // ((aa >> 2) << 8) - (aa >> 2) - hwwrite(vortex->mmio, - a3d_addrS(a->slice, A3D_SLICE_VDBDest) + (a->source << 2), 0); - hwwrite(vortex->mmio, - a3d_addrS(a->slice, - A3D_SLICE_VDBDest + 4) + (a->source << 2), 0); - /* - hwwrite(vortex->mmio, 0x19c00 + (((aa>>2)*255*4)+aa)*8, 0); - hwwrite(vortex->mmio, 0x19c04 + (((aa>>2)*255*4)+aa)*8, 0); - */ -} -#endif - /* A3D HwSource stuff. */ static void vortex_A3dSourceHw_Initialize(vortex_t * v, int source, int slice) diff --git a/sound/pci/au88x0/au88x0_a3ddata.c b/sound/pci/au88x0/au88x0_a3ddata.c index 18623cb6bc52..cc41ea67a6d3 100644 --- a/sound/pci/au88x0/au88x0_a3ddata.c +++ b/sound/pci/au88x0/au88x0_a3ddata.c @@ -30,44 +30,6 @@ static const a3d_Hrtf_t A3dHrirImpulse = { 0, 0, 0 }; -static const a3d_Hrtf_t A3dHrirOnes = { - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, - 0x7fff, - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, - 0x7fff, - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, - 0x7fff, - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, - 0x7fff, - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff -}; - -static const a3d_Hrtf_t A3dHrirSatTest = { - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, - 0x7fff, - 0x8001, 0x8001, 0x8001, 0x8001, 0x8001, 0x8001, 0x8001, 0x8001, - 0x8001, - 0x8001, - 0x7fff, 0x0000, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0, 0, - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 -}; - -static const a3d_Hrtf_t A3dHrirDImpulse = { - 0, 0x7fff, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0, 0, - 0, 0, 0, 0, - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0, 0, - 0, 0, 0 -}; - static const a3d_ItdDline_t A3dItdDlineZeros = { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 -- 2.21.1

2 2

[PATCH] fix sw_params handling of timestamp types in direct plugins
by sylvain.bertrand＠gmail.com 17 Apr '20

17 Apr '20

In pcms using direct plugins (dmix/dsnoop/dshare), the timestamp type could be different from the terminating hw plugin, then the kernel driver. Be sure such pcms have plugins using consistently the same timestamp type. signed-off-by: Sylvain Bertrand <sylvain.bertrand(a)legeek.net> --- src/conf/alsa.conf | 1 + src/conf/pcm/dmix.conf | 4 ++++ src/conf/pcm/dsnoop.conf | 4 ++++ src/pcm/pcm_direct.c | 36 +++++++++++++++++++++++++++++++++++- src/pcm/pcm_direct.h | 2 ++ src/pcm/pcm_dmix.c | 4 ++++ src/pcm/pcm_dshare.c | 4 ++++ src/pcm/pcm_dsnoop.c | 4 ++++ src/pcm/pcm_local.h | 2 ++ 9 files changed, 60 insertions(+), 1 deletion(-) diff --git a/src/conf/alsa.conf b/src/conf/alsa.conf index a091b810..0e01c887 100644 --- a/src/conf/alsa.conf +++ b/src/conf/alsa.conf @@ -69,6 +69,7 @@ defaults.pcm.minperiodtime 5000 # in us defaults.pcm.ipc_key 5678293 defaults.pcm.ipc_gid audio defaults.pcm.ipc_perm 0660 +defaults.pcm.tstamp_type "default" defaults.pcm.dmix.max_periods 0 defaults.pcm.dmix.channels 2 defaults.pcm.dmix.rate 48000 diff --git a/src/conf/pcm/dmix.conf b/src/conf/pcm/dmix.conf index 7fa5c8b2..50e573da 100644 --- a/src/conf/pcm/dmix.conf +++ b/src/conf/pcm/dmix.conf @@ -56,6 +56,10 @@ pcm.!dmix { @func refer name defaults.pcm.ipc_perm } + tstamp_type { + @func refer + name defaults.pcm.tstamp_type + } slave { pcm { type hw diff --git a/src/conf/pcm/dsnoop.conf b/src/conf/pcm/dsnoop.conf index abbd44f7..f4336e5f 100644 --- a/src/conf/pcm/dsnoop.conf +++ b/src/conf/pcm/dsnoop.conf @@ -49,6 +49,10 @@ pcm.!dsnoop { @func refer name defaults.pcm.ipc_perm } + tstamp_type { + @func refer + name defaults.pcm.tstamp_type + } slave { pcm { type hw diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 54d99005..aa60a477 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -991,8 +991,11 @@ int snd_pcm_direct_hw_free(snd_pcm_t *pcm ATTRIBUTE_UNUSED) return 0; } -int snd_pcm_direct_sw_params(snd_pcm_t *pcm ATTRIBUTE_UNUSED, snd_pcm_sw_params_t * params ATTRIBUTE_UNUSED) +int snd_pcm_direct_sw_params(snd_pcm_t *pcm, snd_pcm_sw_params_t *params) { + if (params->tstamp_type != pcm->tstamp_type) + return -EINVAL; + /* values are cached in the pcm structure */ return 0; } @@ -1318,6 +1321,15 @@ int snd_pcm_direct_initialize_slave(snd_pcm_direct_t *dmix, snd_pcm_t *spcm, str return ret; } + if (dmix->tstamp_type != -1) { + ret = snd_pcm_sw_params_set_tstamp_type(spcm, &sw_params, + dmix->tstamp_type); + if (ret < 0) { + SNDERR("unable to set tstamp type"); + return ret; + } + } + if (dmix->type != SND_PCM_TYPE_DMIX && dmix->type != SND_PCM_TYPE_DSHARE) goto __skip_silencing; @@ -1878,6 +1890,7 @@ int snd_pcm_direct_parse_open_conf(snd_config_t *root, snd_config_t *conf, rec->var_periodsize = 0; rec->direct_memory_access = 1; rec->hw_ptr_alignment = SND_PCM_HW_PTR_ALIGNMENT_AUTO; + rec->tstamp_type = -1; /* read defaults */ if (snd_config_search(root, "defaults.pcm.dmix_max_periods", &n) >= 0) { @@ -1941,6 +1954,27 @@ int snd_pcm_direct_parse_open_conf(snd_config_t *root, snd_config_t *conf, continue; } + if (strcmp(id, "tstamp_type") == 0) { + const char *str; + err = snd_config_get_string(n, &str); + if (err < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + if (strcmp(str, "default") == 0) + rec->tstamp_type = -1; + else if (strcmp(str, "gettimeofday") == 0) + rec->tstamp_type = SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY; + else if (strcmp(str, "monotonic") == 0) + rec->tstamp_type = SND_PCM_TSTAMP_TYPE_MONOTONIC; + else if (strcmp(str, "monotonic_raw") == 0) + rec->tstamp_type = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW; + else { + SNDERR("The field tstamp_type is invalid : %s", str); + return -EINVAL; + } + continue; + } if (strcmp(id, "ipc_gid") == 0) { char *group; char *endp; diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 221edbe1..8a236970 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -173,6 +173,7 @@ struct snd_pcm_direct { unsigned int recoveries; /* mirror of executed recoveries on slave */ int direct_memory_access; /* use arch-optimized buffer RW */ snd_pcm_direct_hw_ptr_alignment_t hw_ptr_alignment; + int tstamp_type; /* cached from conf, can be -1(default) on top of real types */ union { struct { int shmid_sum; /* IPC global sum ring buffer memory identification */ @@ -357,6 +358,7 @@ struct snd_pcm_direct_open_conf { int var_periodsize; int direct_memory_access; snd_pcm_direct_hw_ptr_alignment_t hw_ptr_alignment; + int tstamp_type; snd_config_t *slave; snd_config_t *bindings; }; diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index d533f40c..843fa316 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -1038,6 +1038,7 @@ int snd_pcm_dmix_open(snd_pcm_t **pcmp, const char *name, dmix->ipc_key = opts->ipc_key; dmix->ipc_perm = opts->ipc_perm; dmix->ipc_gid = opts->ipc_gid; + dmix->tstamp_type = opts->tstamp_type; dmix->semid = -1; dmix->shmid = -1; @@ -1237,6 +1238,9 @@ pcm.name { # roundup # rounddown # auto (default) + tstamp_type STR # timestamp type + # STR can be one of the below strings : + # default, gettimeofday, monotonic, monotonic_raw slave STR # or slave { # Slave definition diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 59448cfb..6a99452b 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -723,6 +723,7 @@ int snd_pcm_dshare_open(snd_pcm_t **pcmp, const char *name, dshare->ipc_key = opts->ipc_key; dshare->ipc_perm = opts->ipc_perm; dshare->ipc_gid = opts->ipc_gid; + dshare->tstamp_type = opts->tstamp_type; dshare->semid = -1; dshare->shmid = -1; @@ -929,6 +930,9 @@ pcm.name { # roundup # rounddown # auto (default) + tstamp_type STR # timestamp type + # STR can be one of the below strings : + # default, gettimeofday, monotonic, monotonic_raw slave STR # or slave { # Slave definition diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 24f472c7..c64df381 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -591,6 +591,7 @@ int snd_pcm_dsnoop_open(snd_pcm_t **pcmp, const char *name, dsnoop->ipc_key = opts->ipc_key; dsnoop->ipc_perm = opts->ipc_perm; dsnoop->ipc_gid = opts->ipc_gid; + dsnoop->tstamp_type = opts->tstamp_type; dsnoop->semid = -1; dsnoop->shmid = -1; @@ -780,6 +781,9 @@ pcm.name { # roundup # rounddown # auto (default) + tstamp_type STR # timestamp type + # STR can be one of the below strings : + # default, gettimeofday, monotonic, monotonic_raw slave STR # or slave { # Slave definition diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 05ed935f..89d4125b 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -928,6 +928,8 @@ int INTERNAL(snd_pcm_hw_params_set_buffer_size_last)(snd_pcm_t *pcm, snd_pcm_hw_ int snd_pcm_sw_params_set_tstamp_mode(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_tstamp_t val); int INTERNAL(snd_pcm_sw_params_get_tstamp_mode)(const snd_pcm_sw_params_t *params, snd_pcm_tstamp_t *val); +int snd_pcm_sw_params_set_tstamp_type(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_tstamp_type_t val); +int snd_pcm_sw_params_get_tstamp_type(const snd_pcm_sw_params_t *params, snd_pcm_tstamp_type_t *val); int snd_pcm_sw_params_set_avail_min(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val); int INTERNAL(snd_pcm_sw_params_get_avail_min)(const snd_pcm_sw_params_t *params, snd_pcm_uframes_t *val); int snd_pcm_sw_params_set_start_threshold(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val);

2 1

[PATCH] ASoC: soc-pcm: dpcm: Only allow playback/capture if supported
by Stephan Gerhold 17 Apr '20

17 Apr '20

At the moment, PCM devices for DPCM are only created based on the dpcm_playback/capture parameters of the DAI link, without considering if the CPU/FE DAI is actually capable of playback/capture. Normally the dpcm_playback/capture parameter should match the capabilities of the CPU DAI. However, there is no way to set that parameter from the device tree (e.g. with simple-audio-card or qcom sound cards). dpcm_playback/capture are always both set to 1. This causes problems when the CPU DAI does only support playback or capture. Attemting to open that PCM device with an unsupported stream type then results in a null pointer dereference: Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128 Internal error: Oops: 96000044 [#1] PREEMPT SMP CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1 pc : invalidate_paths_ep+0x30/0xe0 lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 Call trace: invalidate_paths_ep+0x30/0xe0 snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 dpcm_path_get+0x38/0xd0 dpcm_fe_dai_open+0x70/0x920 snd_pcm_open_substream+0x564/0x840 snd_pcm_open+0xfc/0x228 snd_pcm_capture_open+0x4c/0x78 snd_open+0xac/0x1a8 ... ... because the DAI playback/capture_widget is not set in that case. We could add checks there to fix the problem (maybe we should anyway), but much easier is to not expose the device as playback/capture in the first place. Attemting to use that device would always fail later anyway. Add checks for snd_soc_dai_stream_valid() to the DPCM case to avoid exposing playback/capture if it is not supported. Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- This can be reproduced with q6asm-dai with a device tree configuration like: &q6asmdai { dai@0 { reg = <0>; direction = <Q6ASM_DAI_RX>; }; }; Then a simple "arecord output.wav" will result in the crash above, since it attempts to use that FE DAI for capture (even though it is limited to playback). --- sound/soc/soc-pcm.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 454735f8fa92..77a680da366f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2911,8 +2911,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) int i; if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { - playback = rtd->dai_link->dpcm_playback; - capture = rtd->dai_link->dpcm_capture; + cpu_dai = asoc_rtd_to_cpu(rtd, 0); + if (rtd->num_cpus > 1) { + dev_err(rtd->dev, + "DPCM doesn't support Multi CPU yet\n"); + return -EINVAL; + } + + playback = rtd->dai_link->dpcm_playback && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK); + capture = rtd->dai_link->dpcm_capture && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE); } else { /* Adapt stream for codec2codec links */ int cpu_capture = rtd->dai_link->params ? -- 2.26.1

3 4

sof-hda-dsp: Reverse order and priority of headset and headphone mic
by GitHub pull_request - opened 17 Apr '20

17 Apr '20

alsa-project/alsa-ucm-conf pull request #21 was opened from juimonen: Headset mic is the analog input source in 99% of the cases compared to headphone mic, so change it's priority higher. Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/21 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/21.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[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

3 3

[PATCH v1 1/2] dt-bindings: sound: add Microsemi ZL38060 binding
by Sven Van Asbroeck 16 Apr '20

16 Apr '20

Add devicetree binding for the Microsemi ZL38060 Connected Home Audio Processor. Signed-off-by: Sven Van Asbroeck <TheSven73(a)gmail.com> --- Tree: v5.7-rc1 .../devicetree/bindings/sound/zl38060.yaml | 90 +++++++++++++++++++ 1 file changed, 90 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/zl38060.yaml diff --git a/Documentation/devicetree/bindings/sound/zl38060.yaml b/Documentation/devicetree/bindings/sound/zl38060.yaml new file mode 100644 index 000000000000..8e1976bee76b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/zl38060.yaml @@ -0,0 +1,90 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/zl38060.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ZL38060 Connected Home Audio Processor from Microsemi. + +description: | + The ZL38060 is a "Connected Home Audio Processor" from Microsemi, + which consists of a Digital Signal Processor (DSP), several Digital + Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs. + +maintainers: + - Jaroslav Kysela <perex(a)perex.cz> + - Takashi Iwai <tiwai(a)suse.com> + +properties: + compatible: + const: mscc,zl38060 + + reg: + description: + SPI device address. + maxItems: 1 + + spi-max-frequency: + maximum: 24000000 + + reset-gpios: + description: + A GPIO line handling reset of the chip. As the line is active low, + it should be marked GPIO_ACTIVE_LOW (see ../gpio/gpio.txt) + maxItems: 1 + + mscc,load-firmware: + description: + If present, the driver will attempt to load firmware into the chip via + the SPI interface. + type: boolean + + '#gpio-cells': + const: 2 + + gpio-controller: true + + '#sound-dai-cells': + const: 0 + + amp-en-gpios: + description: + An optional GPIO line which is active when the codec is playing back + audio. Can be used to control an external amplifier. + maxItems: 1 + + mscc,amp-startup-delay-ms: + description: + An optional delay between amp-en-gpio becoming active, and the start + of audio playback. In milliseconds. + $ref: /schemas/types.yaml#/definitions/uint32 + +required: + - compatible + - reg + - '#gpio-cells' + - gpio-controller + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + spi0 { + #address-cells = <1>; + #size-cells = <0>; + + codec: zl38060@0 { + gpio-controller; + #gpio-cells = <2>; + #sound-dai-cells = <0>; + compatible = "mscc,zl38060"; + reg = <0>; + spi-max-frequency = <12000000>; + reset-gpios = <&gpio1 0 GPIO_ACTIVE_LOW>; + amp-en-gpios = <&codec 3 GPIO_ACTIVE_HIGH>; + mscc,amp-startup-delay-ms = <12>; + mscc,load-firmware; + }; + }; -- 2.17.1

2 4

[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

2 3

[PATCH 1/3] ASoC: codecs: hdac_hdmi: Fix incorrect use of list_for_each_entry
by Amadeusz Sławiński 16 Apr '20

16 Apr '20

If we don't find any pcm, pcm will point at address at an offset from the the list head and not a meaningful structure. Fix this by returning correct pcm if found and NULL if not. Found with coccinelle. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/codecs/hdac_hdmi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index fba9b749839d..f26b77faed59 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -142,14 +142,14 @@ static struct hdac_hdmi_pcm * hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, struct hdac_hdmi_cvt *cvt) { - struct hdac_hdmi_pcm *pcm = NULL; + struct hdac_hdmi_pcm *pcm; list_for_each_entry(pcm, &hdmi->pcm_list, head) { if (pcm->cvt == cvt) - break; + return pcm; } - return pcm; + return NULL; } static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, -- 2.17.1

2 4

Volume is not restored after boot
by GitHub issues - opened 16 Apr '20

16 Apr '20

alsa-project/alsa-utils issue #39 was opened from Oscar65: I have archlinux updated system last sunday. ``` $ uname -a Linux mypc 5.6.3-arch1-1 #1 SMP PREEMPT Wed, 08 Apr 2020 07:47:16 +0000 x86_64 GNU/Linux ``` I checked with alsa-restore.service and alsa-state.service. None of them work as expected. My sound card is: ``` # lshw -C sound *-multimedia description: Audio device product: 82801H (ICH8 Family) HD Audio Controller vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 02 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: driver=snd_hda_intel latency=0 resources: irq:29 memory:ffaf8000-ffafbfff ``` Every time that my system boots the volume levels are different than last time a powered off. Issue URL : https://github.com/alsa-project/alsa-utils/issues/39 Repository URL: https://github.com/alsa-project/alsa-utils

1 0